| 00:00.24 | vlt | JT: Got it. But what does the NTBA do there to stabilze the connection? |
| 00:00.39 | J4k3 | http://www.csgnetwork.com/phone1freqtable.html |
| 00:00.40 | J4k3 | what I need. |
| 00:00.59 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 00:01.05 | JT | vlt: err the ntba/nt1 is a modem sort of device |
| 00:01.18 | JT | converts 2 wire U interface to telco to 4 wire local S/T bus |
| 00:02.53 | vlt | JT: In my case (like shown in the diagram) it connects two 4-wire cables on the same bus ... strange world ... |
| 00:03.05 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:04.14 | JT | vlt: what diagram? |
| 00:04.27 | JT | i'm not sure how well asterisk works with sharing the S-bus |
| 00:06.27 | obnauticus | is anyone here using fwdOUT, because i got a question...is it working? |
| 00:06.28 | obnauticus | lol |
| 00:08.32 | rtcg | By what term is the following process is known? An inbound call rings an extension and upon no answer gets put on hold while the cell phone number associated with said extension is dialed. The inbound callerid info is read to the owner of the cell phone with a request to accept the call otherwise the call is send to voicemail? |
| 00:08.45 | vlt | JT: The diagram on http://www.pro-linux.de/work/asterisk/asterisk-1.html I pasted above ... |
| 00:09.36 | JT | vlt: went you exceed about 10 billion lines, the IRCD cuts off your text... |
| 00:10.18 | rtcg | Automated supervised transfer is the best term I can come up with...only I'm not getting too many google hits with that term. |
| 00:10.31 | JT | s/went/when/ |
| 00:11.03 | rob0 | would be handy if the * could send the cell user a txt msg |
| 00:11.37 | JT | i'm sure you could make it send smses using an external application upon certain events |
| 00:12.50 | vlt | rob0: You can use asterisk's built in CURL library to send an sms over an http(s) gateway (like www.fullsms.de in Germany) ... |
| 00:15.18 | *** join/#asterisk GMitre (n=qzblo@200-234-185-201.usr.microlinknet.com.br) |
| 00:16.10 | rtcg | The process as I envision it would be to park an incoming call, place a separate outbound call where festival could playback the CALLERID(name) and number and present an opportunity to pick up the parked call. If after X amount of time the call is not picked up, then the parked call is routed to voicemail. |
| 00:17.09 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:18.09 | GMitre | I made a callcenter with asterisk and when u call an backgruound intro play´s saying to you type a number to entry on a sector but i need to listen all the message before i type the number, if i type the number before the final message nothing heapens |
| 00:23.09 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 00:27.50 | *** join/#asterisk dlynes_laptop (n=dlynes@s209-121-50-177.bc.hsia.telus.net) |
| 00:27.52 | *** join/#asterisk dlynes_ (n=dlynes@s209-121-50-177.bc.hsia.telus.net) |
| 00:34.18 | sweeper | kombi: dunno |
| 00:36.29 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:37.16 | *** join/#asterisk dlynes__ (n=dlynes@s209-121-50-177.bc.hsia.telus.net) |
| 00:40.42 | *** join/#asterisk saftsack (n=saftsack@pD9E070D0.dip.t-dialin.net) |
| 00:42.07 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:42.52 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 00:50.15 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:51.39 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
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| 01:01.31 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:07.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:09.02 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 01:15.10 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 01:16.07 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:16.51 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 01:17.24 | nDuff | Anyone know where one can get a list of local long distance prefixes (or genuinely local prefixes... one way or the other)? |
| 01:18.52 | rtcg | what's a "local long distance" prefix?? |
| 01:19.00 | rtcg | is it local or is it long distance?? |
| 01:21.06 | Err | you can almost certainly find the LATA rules for your area via google (your state's public utility commission likely stores this information) |
| 01:22.17 | *** join/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 01:22.18 | rtcg | http://puck.nether.net/npa-nxx/ might be of SOME use... |
| 01:22.30 | n3glv | any #freepbx ops here? |
| 01:22.34 | n3glv | groogs, ? |
| 01:22.52 | [TK]D-Fender | n3glv, LOL |
| 01:23.37 | n3glv | got some jack-off who has auto-rejoin on, but can't stay on for more than 1min |
| 01:23.37 | nDuff | rtcg: "local long distance" == long distance numbers with the same area code (which need a 1+<area code> to be dialed first). |
| 01:23.55 | rtcg | OH! is THAT what those are called!!! |
| 01:23.57 | n3glv | so, been filling all our logs with ON/OfF messages |
| 01:23.59 | [TK]D-Fender | n3glv, Trolls in #freepbx? For real!? |
| 01:24.45 | [TK]D-Fender | n3glv, From what I hear there is next to no trffic in #freepbx anyways... you'd thingk you WANTED a little logging action.... |
| 01:24.54 | n3glv | he has done this under one or two nicks for about 12 hrs |
| 01:25.30 | rtcg | nDuff: http://www.nanpa.com/area_codes/index.html has downloadable links of area code databases..but I don't know how to find out what is in your LATA other than to ask your service provider. |
| 01:25.32 | Err | heh, I suspect that the concept of 'local long distance' varies based on your LATA - there are two classes of long distance in my area, both within the same area code, along with local (free) calling as well |
| 01:25.53 | [TK]D-Fender | n3glv, go complain to a FreeNode Net-op then |
| 01:27.26 | JT | n3glv: the irc channel sounds as well maintained and supported as freepbx itself |
| 01:27.40 | Qwell | JT: that was below the belt |
| 01:27.43 | Qwell | I like it :p |
| 01:27.51 | JT | ;) |
| 01:28.14 | [TK]D-Fender | JT : Indeed... I only showed false concern & disbelief that anything could be amiss there! |
| 01:28.28 | n3glv | well, we _have_ drawn some ppl over to #elastix and #voipcoop |
| 01:28.37 | [TK]D-Fender | JT : Who told you you could just say the TRUTH like that! |
| 01:28.50 | JT | n3glv: what on earth are those channels? |
| 01:30.42 | *** part/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net) |
| 01:32.32 | n3glv | JT_, elastix is * 1.4.4 and centos 5 |
| 01:32.42 | JT | never heard of it |
| 01:33.07 | n3glv | voipcoop is our co-op channel |
| 01:33.16 | n3glv | www.voipcoop.org |
| 01:37.37 | *** join/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net) |
| 01:43.33 | *** join/#asterisk andrewc (n=andrewc@216.160.70.198) |
| 01:45.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:49.11 | rtcg | well...gonna change nicks to stop the join flood in #freepbx...since I'm not actively chatting... have a day! |
| 01:49.33 | JT | nice nick |
| 01:51.09 | asterisknerds2 | surely you jest! |
| 01:51.22 | JT | hehe |
| 01:52.36 | asterisknerds2 | what does "as sip provider here.." mean when registering? (ref: ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here.) |
| 01:52.38 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 01:53.58 | asterisknerds2 | shouldn't register => user:password@host be enough?? |
| 01:55.17 | rtcg | ah...someone else has taken on the mantle of nick squatter. |
| 02:10.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:12.50 | flenders | rtcg: you do the '/exten' if you want to have something different than the 's' extension handling your incoming calls on that context |
| 02:13.04 | rtcg | got it! which I don't... so that's good. |
| 02:17.30 | rtcg | can a sip trunk provider pass along any OTHER type of information that I could use to further route the call to a different trunk? ...so that calls originally destined for [one context] that get 'hunt group transferred' to a sip trunk handling [another context] could be identified and put back into the correct [one context]? |
| 02:22.07 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 02:22.20 | KDan | Hello again |
| 02:23.08 | KDan | the agi record_file function... it says that the return value is: |
| 02:23.09 | KDan | return: see evaluate for return information. ['result'] is -1 on error, 0 on hangup, otherwise a decimal value of the DTMF tone. Use chr() to convert to ASCII. |
| 02:23.44 | KDan | that would imply, to me, that if the user hangs up, the function returns 0 and the agi process continues on its way... but instead it seems to be just dying immediately? |
| 02:24.03 | KDan | I guess asterisk kills it... is there a way to stop asterisk from being quite so murderous in these circumstances? |
| 02:24.38 | *** join/#asterisk wundaboy (n=pat@c-24-21-71-88.hsd1.mn.comcast.net) |
| 02:24.55 | wundaboy | why am I getting this: WARNING[27481]: chan_iax2.c:7118 socket_read: Call rejected by 66.227.100.30: No such context/extension |
| 02:25.27 | KDan | wundaboy: there's no context defined for the IAX url you gave to your DID |
| 02:25.39 | KDan | e.g. if you gave it guest@yourserver.tld |
| 02:25.46 | KDan | you need a [guest] entry in your iax.conf |
| 02:25.56 | wundaboy | i need that in order to dial a landline? |
| 02:26.05 | KDan | oh - this is dialing out? |
| 02:26.09 | wundaboy | yeah |
| 02:26.11 | KDan | hmm |
| 02:26.16 | KDan | don't know then.. |
| 02:26.18 | KDan | sorry |
| 02:26.23 | wundaboy | anyone else? |
| 02:26.43 | watchy | anyone here got the newest polycom firmware layin around |
| 02:27.41 | watchy | i'll trade hugs for it? |
| 02:27.52 | wundaboy | dosent freedomphones have it? |
| 02:28.07 | watchy | it doesnt have the newest for some reason |
| 02:29.13 | watchy | trade netsex for it? |
| 02:31.02 | wundaboy | i dont have the binary |
| 02:31.14 | wundaboy | why am i getting this: WARNING[27481]: chan_iax2.c:5074 authenticate_verify: requested inkey 'jnctn' for RSA authentication does not exist |
| 02:31.38 | KDan | re: my issue, i have tried calling the php script through DeadAGI instead of AGI, but even so it still kills the script |
| 02:31.43 | wundaboy | figured it out |
| 02:35.26 | KDan | woo |
| 02:35.27 | KDan | i found it |
| 02:35.41 | KDan | had to use DeadAGI *and* follow the instructions here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI |
| 02:35.50 | KDan | ie disable SIGHUP signals in the php script |
| 02:36.13 | `Sean | Jun 26 22:37:23 WARNING[5124]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm |
| 02:36.13 | `Sean | Jun 26 22:37:23 WARNING[5124]: file.c:824 ast_streamfile: Unable to open vm-rec-name (format g729): No such file or directory |
| 02:36.13 | `Sean | <PROTECTED> |
| 02:36.26 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:38.37 | `Sean | can somoene please help me here is the error http://pastebin.ca/591269 |
| 02:40.19 | [TK]D-Fender | `Sean, You have no G729 codec or available transcoders. |
| 02:40.44 | `Sean | But Conference isn't set to use G729 tho |
| 02:40.52 | `Sean | it was working well just 5 minutes ago |
| 02:41.55 | [TK]D-Fender | `Sean, It isn't liying about the codecs involved. |
| 02:44.45 | *** join/#asterisk anthm (n=anthm@dhcp64-134-34-214.bwic.chi.wayport.net) |
| 02:44.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:45.52 | [hC] | This is interesting. a sip trunk provider has indicated that they use one ip for SIP messaging and another for RTP. do i need to configure asterisk to handle this somehow? |
| 02:46.19 | `Sean | arfh i found out the problem |
| 02:46.22 | `Sean | stupid digium |
| 02:46.22 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-226.bwic.chi.wayport.net) |
| 02:46.25 | `Sean | wont let g729 encoders work |
| 02:46.28 | `Sean | on a different NIC |
| 02:46.35 | `Sean | so i had to use same old one |
| 02:46.35 | `Sean | :( |
| 02:46.43 | fbffff | hey SwK |
| 02:46.46 | `Sean | i dont see why they dont just use some kinda login method |
| 02:46.46 | [hC] | the license is based upon the sum of all the NIC's in the system |
| 02:46.47 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:46.58 | `Sean | hC this nic is NWQ |
| 02:47.00 | `Sean | *NEW |
| 02:47.06 | [hC] | so? |
| 02:47.22 | [hC] | if you change any NICs in or out it screws with the license |
| 02:47.31 | [hC] | as i said, the license is based on the SUM of ALL the nic's in the box |
| 02:47.41 | `Sean | lol |
| 02:48.02 | [hC] | i didnt say it was a good idea, im just explaining it to you |
| 02:48.46 | J4k3 | `Sean: I believe you're legal to use a 3rd party g729 codec as long as you're within the limits of the original purchase |
| 02:48.49 | J4k3 | but |
| 02:48.51 | J4k3 | I'm no lawyer |
| 02:48.58 | J4k3 | and your country/town/luck may vary |
| 02:49.25 | `Sean | lol |
| 02:49.36 | `Sean | argh! |
| 02:49.41 | [hC] | you can use the intel g729 codec which is just a reference implementation, but it is not legal in the US to use it for commercial use |
| 02:49.51 | `Sean | midas well use the cracked version of g729 that i got |
| 02:49.55 | `Sean | next time i install asterisk :p |
| 02:51.25 | `Sean | thats gay my DID's dont even use g729 |
| 02:51.28 | `Sean | im using them to conf |
| 02:51.41 | `Sean | yet just cause g729 wont work asterisk wont let me have incoming calls |
| 02:53.46 | J4k3 | your incoming provider not allow anything except g729? |
| 02:55.47 | `Sean | dude didww |
| 02:55.51 | `Sean | doesnt even offer g729 |
| 02:56.01 | `Sean | they send calls via ulaw |
| 02:56.10 | `Sean | so i dont see why conference would have had a heart attack |
| 02:56.11 | `Sean | :| |
| 02:56.27 | [hC] | you're calling into a DID and straight into a meetme? |
| 02:56.37 | [hC] | via ulaw |
| 02:56.47 | `Sean | Yes |
| 02:56.52 | JT | `Sean: it'd be more useful to get some sip debug output too |
| 02:57.06 | [hC] | then it should work |
| 02:57.14 | [hC] | if what you're saying is true, that has no relevance to g729 |
| 02:57.20 | CrashSys | G729 is 8kbit right? |
| 02:57.25 | `Sean | Yers |
| 02:58.38 | J4k3 | are there any better-than-telco-quality free codecs? |
| 02:58.42 | CrashSys | Anyone know of any good link-testing tools for VoIP? Something I can set-up at the endpoints and have them stream back and forth for stats? |
| 02:58.51 | JT | J4k3: g.722 |
| 02:59.19 | [TK]D-Fender | J4k3, What would be the point? |
| 03:00.01 | J4k3 | [TK]D-Fender: both ends being voip endpoints |
| 03:00.18 | J4k3 | I personally find skype to sound wonderful between a couple decent systems with good headsets |
| 03:00.20 | [TK]D-Fender | `Sean, show a pastebin with real detail of the gcodecs, because it didn't mention G.729 for nothing. |
| 03:00.26 | J4k3 | and afaik, its a much higher bitrate |
| 03:00.36 | JT | J4k3: g.722 |
| 03:00.39 | CrashSys | Asterisk is only 8k sampling tho isn't it? |
| 03:00.43 | JT | correct |
| 03:00.46 | [TK]D-Fender | IINM G.722 isn't exactly free... |
| 03:01.33 | CrashSys | So telco quality is all your going to get with asterisk... regardless of codec... |
| 03:01.40 | CrashSys | unless better-than-telco is lower bitrate... |
| 03:01.43 | JT | CrashSys: not if endpoints reinvite |
| 03:01.53 | JT | or don't use asterisk |
| 03:01.55 | CrashSys | Hmm... p2p maybe... |
| 03:02.21 | CrashSys | Shoutcast = Better-then-telco quality |
| 03:02.29 | JT | also not telephony :) |
| 03:02.33 | CrashSys | And it's got a built-in one-way conferencing engine! |
| 03:02.38 | rtcg | `Sean: I had to 'disallow=all' and 'allow=ulaw' to get my * server to quit referencing g729 |
| 03:02.40 | JT | icecast kthx |
| 03:02.41 | JT | ;) |
| 03:02.43 | CrashSys | With multiple codecs! |
| 03:02.48 | [hC] | any of you guys here implement screen pop solutions w/ asterisk? |
| 03:02.48 | [TK]D-Fender | CrashSys, everyone is entitled to my opinion! |
| 03:03.03 | CrashSys | D-Fender: yes master, i'm sorry... |
| 03:03.06 | `Sean | ahh thanks rtcg |
| 03:03.37 | rtcg | `Sean: I had to do it for each of my SIP trunks. |
| 03:03.38 | J4k3 | well, theres skype, it sounds awesome.. no idea what speed its actually running it |
| 03:03.41 | J4k3 | er at |
| 03:03.46 | J4k3 | it seems to use more bandwidth than g711 |
| 03:05.04 | CrashSys | I dont see much value in audio resolution... as long as it's clear... |
| 03:05.28 | CrashSys | Unless I wanted to rock-out to some awesome elevator music... |
| 03:05.42 | JT | J4k3: are you going to actually acknowledge we've said g.722 about 3 times now |
| 03:05.46 | JT | enough about skype :P |
| 03:10.35 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:11.27 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-218-123.dsl.irvnca.pacbell.net) |
| 03:17.57 | *** join/#asterisk ZX81 (n=matt@161.29.135.2) |
| 03:18.01 | ZX81 | hi all |
| 03:18.19 | JT | hello :) |
| 03:18.43 | ZX81 | anyone able to help with a red alarm on zaptel car? |
| 03:18.43 | rob0 | Hey, it's a Matt. If he doesn't work at Digium, he will soon. |
| 03:18.49 | ZX81 | :) |
| 03:19.09 | JT | how's nz? |
| 03:19.13 | ZX81 | I've got a quad span card - set up as E1 - crossover cable between ports 3 and 4 |
| 03:19.17 | ZX81 | heh cold |
| 03:19.19 | ZX81 | but good |
| 03:19.22 | ZX81 | a bit boring |
| 03:19.23 | ZX81 | :) |
| 03:19.41 | ZX81 | I can't get the card out of red alarm |
| 03:19.57 | JT | it's freezing here too |
| 03:20.02 | ZX81 | you'd think a crossover between ports 3 and 4 would bring both up as green no? |
| 03:20.03 | JT | today is very cold |
| 03:20.03 | ZX81 | really? |
| 03:20.09 | ZX81 | bet its colder here |
| 03:20.10 | ZX81 | :) |
| 03:20.12 | rob0 | Red alarm, I had that once, I think the card died. |
| 03:20.15 | JT | yes, sydney isn't much warmer |
| 03:20.28 | ZX81 | rob0 its a new card |
| 03:20.32 | JT | ZX81: what's zaptel.conf and zapata.conf like? |
| 03:20.39 | ZX81 | meh pretty clean |
| 03:20.45 | ZX81 | bchans,dchans in zaptel |
| 03:20.51 | ZX81 | pri_cpe in zapata |
| 03:20.54 | [TK]D-Fender | bad question, WORSE answer.... |
| 03:21.01 | [TK]D-Fender | *shudder* |
| 03:21.02 | JT | well can you pastebin them full? |
| 03:21.02 | ZX81 | :) |
| 03:21.06 | [TK]D-Fender | BETTER |
| 03:21.09 | ZX81 | heh from console |
| 03:21.13 | ZX81 | am in server farm |
| 03:21.15 | ZX81 | :) |
| 03:21.29 | JT | transcribe? ;) |
| 03:21.34 | [TK]D-Fender | s/can you/get off your ass and/ |
| 03:21.47 | [TK]D-Fender | ZX81, yes... definately out to pasture... |
| 03:21.50 | [TK]D-Fender | :) |
| 03:21.56 | ZX81 | reads like span=1,0,0,ccs,hdb3,crc4 |
| 03:22.05 | ZX81 | same for 2-3 |
| 03:22.12 | JT | well |
| 03:22.15 | ZX81 | and then 4 is ,1 for timing |
| 03:22.24 | *** join/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 03:22.28 | JT | which one is acting as network? |
| 03:22.30 | ZX81 | then bchan=1-15, dchan=16,bchan=17-31 |
| 03:22.35 | ZX81 | 4 as network |
| 03:22.40 | ZX81 | 3 as cpe |
| 03:22.53 | ZX81 | but I'd expect it to at least go over to yellow alarm no? |
| 03:22.54 | JT | you have timing wrong then |
| 03:22.55 | [TK]D-Fender | ZX81, your timing is backwards. |
| 03:23.05 | JT | network GIVES clock |
| 03:23.10 | ZX81 | real? yeah |
| 03:23.13 | JT | cpe receives |
| 03:23.20 | JT | 0 = give clock |
| 03:23.25 | ZX81 | ok |
| 03:23.26 | JT | 1 = priority 1 receive |
| 03:23.27 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-226.bwic.chi.wayport.net) |
| 03:24.16 | JT | ZX81: and zapata.conf? |
| 03:24.25 | ZX81 | changing now |
| 03:24.27 | ZX81 | sec |
| 03:26.04 | ZX81 | does zapata make a diff with a red alarm? |
| 03:26.19 | ZX81 | I thought red alarm means cable down etc |
| 03:26.39 | JT | usually that's what it means |
| 03:26.47 | JT | what cable pinout did you use? |
| 03:26.51 | JT | what is the card |
| 03:26.55 | ZX81 | 1,2 -> 4,5 |
| 03:26.58 | ZX81 | 4,5 -> 1,2 |
| 03:27.08 | ZX81 | quad span PRI with echo can |
| 03:27.16 | ZX81 | can't keep up with their version numbers |
| 03:27.23 | JT | digium? |
| 03:27.23 | ZX81 | is like 407 or something |
| 03:27.27 | ZX81 | yeah |
| 03:27.29 | ZX81 | of course |
| 03:27.31 | ZX81 | :) |
| 03:27.37 | JT | that's nowhere near a given here |
| 03:27.38 | rtcg | Can callerid name and number be set on the same line? set(CALLERID(name)="Some Name" CALLERID(number)=<2125551212>) |
| 03:27.55 | ZX81 | heh yeah |
| 03:28.04 | ZX81 | rtcg yes but network may not recognise it |
| 03:28.08 | ZX81 | internally you can |
| 03:28.11 | rob0 | set(CALLERID(all)=...) ? |
| 03:28.12 | ZX81 | oh |
| 03:28.17 | ZX81 | in same line of asterisk |
| 03:28.21 | Corydon76-home | rtcg: Set(CALLERID(all)=name <number>) |
| 03:28.27 | ZX81 | yep |
| 03:28.41 | rtcg | ZX81: who would the network not recognize it? |
| 03:28.55 | JT | ZX81: well the cable sounds ok |
| 03:29.02 | rob0 | Some providers might not pass the name. |
| 03:29.07 | ZX81 | :) if you send callerid name out to the telephone company they won't transmit it |
| 03:29.09 | ZX81 | yah |
| 03:29.27 | ZX81 | JT yeah - they had a crossover rj45 end here |
| 03:29.31 | rob0 | Telco wom't pass ANY callerID. |
| 03:29.33 | ZX81 | with 1,2 going to 4 and 5 |
| 03:29.41 | rtcg | ah...which is what I'm seeing... |
| 03:29.55 | ZX81 | rob0 yeah we can set cid number here |
| 03:29.58 | rtcg | well at LEAST I can pass the calleridnum...thx. |
| 03:30.13 | ZX81 | but only to a DDI we own |
| 03:30.15 | JT | rob0: err callerid number can be passed if done right |
| 03:30.21 | ZX81 | hey also |
| 03:30.31 | JT | in the us, most telcos let you set the callerid to anything |
| 03:30.32 | rtcg | I need to take a break though. Calling wrong numbers this late at night is..... very not good. |
| 03:30.52 | ZX81 | when I modprobed in zaptel I got an error about userspace firmware or whatever being unavailable for vpm450 or some such |
| 03:30.56 | ZX81 | .128 |
| 03:31.03 | ZX81 | but it got installed by the make install |
| 03:31.22 | rtcg | at least the callerid was messed up when I mis-dialed. :) |
| 03:31.25 | ZX81 | I'm assuming thats something to do with octasic chip for echo can |
| 03:31.32 | JT | yes |
| 03:32.02 | ZX81 | did you know that zttool still says linux support services :) |
| 03:32.11 | ZX81 | (c) 2002 |
| 03:32.12 | ZX81 | :) |
| 03:32.30 | JT | heaps of the source does |
| 03:33.01 | ZX81 | yeah but not normally gui stuff |
| 03:33.17 | JT | ZX81: how's the embedded box going? |
| 03:33.39 | ZX81 | yeah good - nice for a market that doesn't know asterisk |
| 03:33.51 | ZX81 | asterisk people can just build their own |
| 03:33.52 | ZX81 | :) |
| 03:34.33 | JT | can i ask what sort of board it has? |
| 03:34.34 | ZX81 | trying to compile in support for this PIKA board that does FXO with echo can in DSP (no CPU left over) |
| 03:34.35 | ZX81 | :) |
| 03:34.39 | ZX81 | soekris |
| 03:34.44 | ZX81 | 4801 |
| 03:34.45 | ZX81 | :) |
| 03:34.50 | JT | ah they seem popular for that stuff |
| 03:34.59 | ZX81 | yeah really solid |
| 03:35.04 | JT | and pci slot |
| 03:35.07 | ZX81 | yep |
| 03:35.15 | ZX81 | also mini pci which could be used for wifi |
| 03:35.20 | ZX81 | but we're not using that yet |
| 03:35.25 | JT | they need a mini-pci g.729/etc transcoder board that doesn't cost the earth |
| 03:35.32 | ZX81 | yeah agree |
| 03:35.56 | ZX81 | or just ship the audio off to a server for processing - oh no wait :) |
| 03:36.06 | JT | all the embedded stuff is still too pissweak to transcode |
| 03:36.07 | ZX81 | hehe |
| 03:36.12 | ZX81 | yep |
| 03:36.13 | ZX81 | for sure |
| 03:36.16 | JT | even to gsm |
| 03:36.20 | ZX81 | a few channels of ulaw to gsm |
| 03:36.22 | ZX81 | but not many |
| 03:36.33 | ZX81 | kick ass if its the same all the way through |
| 03:36.33 | JT | ulaw, i thought you were in nz |
| 03:36.37 | ZX81 | heh |
| 03:36.39 | ZX81 | yeah |
| 03:36.42 | ZX81 | we transcode |
| 03:36.50 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:36.59 | JT | but nz is alaw |
| 03:37.07 | ZX81 | on e1 yeah |
| 03:37.23 | JT | yes |
| 03:37.25 | ZX81 | but my only e1's are in a cisco box grrr |
| 03:37.35 | ZX81 | cos I can't get the red alarm to go away |
| 03:37.36 | ZX81 | :) |
| 03:37.44 | JT | the whole telco network would use alaw |
| 03:37.47 | ZX81 | man its cold in the server room! |
| 03:37.51 | JT | heh |
| 03:37.58 | JT | i think it's colder outside |
| 03:38.02 | ZX81 | well yeah, but hardly any encoding on analogue :) |
| 03:38.04 | ZX81 | yeah |
| 03:38.06 | ZX81 | probably |
| 03:38.28 | ZX81 | its actually 17 at the top of the rack in front of me, but the fans are pointed right at me |
| 03:38.31 | ZX81 | :) |
| 03:38.38 | JT | analogue gets converted to alaw when it hits the exchange heh |
| 03:38.44 | ZX81 | :) |
| 03:39.01 | JT | it's 13degC in sydney city right now |
| 03:39.09 | ZX81 | *** You have new email |
| 03:39.11 | ZX81 | 13!!!!!!!!!! |
| 03:39.13 | ZX81 | summer! |
| 03:39.14 | ZX81 | :) |
| 03:39.19 | JT | lies |
| 03:39.23 | ZX81 | yesterday we had a high of 5 |
| 03:39.24 | rob0 | Not a good time to go down under. |
| 03:39.24 | ZX81 | :) |
| 03:39.49 | ZX81 | when I played at winter solstice party last weekend it was -5 :) |
| 03:39.58 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:40.03 | JT | nice |
| 03:40.05 | ZX81 | anywayz better make a move - giving up on pri for now |
| 03:40.10 | JT | it was 0 at about 6am |
| 03:40.13 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:40.16 | rtcg | you mean summer solstice ..right??? |
| 03:40.18 | rtcg | :) |
| 03:40.20 | ZX81 | need some sleep before the americas cup tonight |
| 03:40.22 | JT | ZX81: have any pri debug output? |
| 03:40.23 | ZX81 | winter |
| 03:40.27 | ZX81 | nah |
| 03:40.29 | ZX81 | nothing really |
| 03:40.48 | ZX81 | unnumbered frame |
| 03:40.54 | ZX81 | set aßs |
| 03:40.56 | ZX81 | hmm |
| 03:41.02 | JT | going only one way? |
| 03:41.08 | ZX81 | set asyncronous balanced mode extended |
| 03:41.44 | ZX81 | hmmm sec |
| 03:42.47 | JT | ZX81: this card is brand new, is the jumper set correctly? |
| 03:43.02 | nephfl | i have a problem, i had 4 analog trunks temporarily i have replaced it with a voip provider, and forwaded the main number to a did number, problem is that it was also a hunt group/rollover on the analog trunk, anyone know how i can get more than one line to connect from this number? |
| 03:43.42 | ZX81 | JT: yeah set it first |
| 03:43.45 | ZX81 | zap is down at the mo |
| 03:43.52 | ZX81 | will sort it and back in a sec |
| 03:44.03 | Nuitari | nephfl: you'd have to change the whole group to the proper numbers for the voip |
| 03:44.05 | JT | hmm |
| 03:44.55 | nephfl | my telco says they cant do that (short of transferring the numbers), and since this is temporary, i dont know what else i can do |
| 03:45.06 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:45.23 | nephfl | so, there is no way to cascade rollover numbers through the provider? |
| 03:46.01 | ZX81 | zap back up |
| 03:46.11 | Nuitari | forward all of them to voip ? |
| 03:46.36 | *** part/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net) |
| 03:46.52 | ZX81 | pri intense debug span 3 and 4 gives same messages |
| 03:46.57 | ZX81 | and still red alarm |
| 03:48.09 | JT | ZX81: umm, what is in zapata.conf? |
| 03:48.20 | ZX81 | spans 1,3,4 - provisioned, in alarm, down,activer |
| 03:48.22 | ZX81 | er |
| 03:48.28 | nephfl | the forward breaks the huntgroup with my telco, so forwards from each of the analog lines wont help |
| 03:48.32 | ZX81 | signalling=pri_net for 4 |
| 03:48.39 | ZX81 | pri_cpe for 3 |
| 03:48.56 | ZX81 | grrr@selinux spewing all over my screen |
| 03:48.57 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:48.58 | ZX81 | :) |
| 03:49.05 | JT | disable selinux :) |
| 03:49.09 | ZX81 | heh yeah |
| 03:49.13 | ZX81 | its on warn |
| 03:49.17 | ZX81 | so kinda disabled |
| 03:49.18 | ZX81 | :) |
| 03:49.32 | ZX81 | I'm going to start up again at home |
| 03:49.36 | ZX81 | see you soon |
| 03:49.42 | JT | ok |
| 03:49.48 | ZX81 | bye |
| 03:52.03 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 04:07.17 | *** join/#asterisk gthing (n=gthing32@72.8.88.137) |
| 04:07.38 | gthing | Hello, does anyone know where I can find a hardware guide for asterisk? |
| 04:08.06 | gthing | I searched around quite a bit |
| 04:08.19 | gthing | but couldn't find a generic page that basically just said "Buy this and everything will work" |
| 04:08.39 | gthing | I am planning on using asterisknow |
| 04:09.49 | [TK]D-Fender | gthing, I wouldn't if I were you. Its not ready to do all the work for you and you'll be finding yourself without much support. |
| 04:09.57 | Nuitari | gthing: you need a computer, usually some network connection or some fxo/fxs cards. You can have wired or voip phones. |
| 04:10.33 | JT | s/wired/analogue/ |
| 04:10.36 | gthing | So am I going to be better off going with the regular asterisk project for now? or asterisk for windows? |
| 04:10.47 | JT | asterisk for windows really doesn't exist |
| 04:10.49 | [TK]D-Fender | gthing, you're going to have to learn a fair bit about * to get things up and running, or hire a consultant. |
| 04:11.19 | Nuitari | the first thing to learn would be to ask precise questions... |
| 04:11.24 | [TK]D-Fender | gthing, there is no "asterisk for windows", in essence, and yes, PLAIN Asterisk is the way to go. |
| 04:11.34 | remmo | vanilla |
| 04:11.49 | [TK]D-Fender | gthing, With that in mind I'm willing to spare you a few odd minutes to help with your hardware questions. |
| 04:11.56 | gthing | Okay, cool - I was looking at AsteriskWin32 - I guess that's a different project |
| 04:12.20 | [TK]D-Fender | gthing, That is jsut running under CYGWIN. AKA bullshit. |
| 04:12.28 | gthing | Gotcha ;) |
| 04:12.47 | [TK]D-Fender | gthing, So lets move on. |
| 04:12.52 | [TK]D-Fender | gthing, Where are you located? |
| 04:12.55 | gthing | Utah |
| 04:13.43 | [TK]D-Fender | gthing, Ok, what kind of lines are you looking to use with *? A) Normal analog phone lines. B) Digital link to telco (T1/PRI) C) ITSP (VoIP Provider like Vonage, etc) |
| 04:13.56 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 04:14.00 | gthing | Vonage |
| 04:14.23 | [TK]D-Fender | gthing, I would specific advise against THEM personally, but you are considering running on pure VoIP? |
| 04:14.49 | gthing | [TK]D-Fender, That's right |
| 04:15.09 | [TK]D-Fender | gthing, This is not advised for business, that IS what you are planning for correct? |
| 04:15.18 | gthing | [TK]D-Fender, That's correct |
| 04:15.19 | [TK]D-Fender | gthing, Ok, moving on... |
| 04:15.37 | [TK]D-Fender | gthing, How big a pipe do you have to the internet (up/down)? |
| 04:16.30 | gthing | [TK]D-Fender, We will be on last mile fiber (hasn't been hooked up yet) 15mbit up/down (although real world speed seem to hover around half that) |
| 04:16.39 | [TK]D-Fender | gthing, I take that as "formidable". |
| 04:16.41 | [TK]D-Fender | next... |
| 04:17.09 | gthing | [TK]D-Fender, Yea, it's nice - and in Provo, Utah of all places :) |
| 04:17.16 | [TK]D-Fender | gthing, How many phones do you need? Do you have an extra RJ45 jack at each station? |
| 04:17.38 | gthing | [TK]D-Fender, Right now it will be 2 phones, within the next few months 4 phones |
| 04:18.02 | [TK]D-Fender | gthing, Ok starting very small and growing to still tiny :) |
| 04:18.13 | gthing | [TK]D-Fender, that's right :) |
| 04:18.21 | [TK]D-Fender | gthing, Any issues using a power brick for your phone? |
| 04:18.38 | gthing | [TK]D-Fender, What do you mean? |
| 04:19.28 | gthing | [TK]D-Fender, If you mean has it caused disruption or problems with the service, I don't know yet |
| 04:19.47 | gthing | [TK]D-Fender, I just signed up for Vonage tonight and don't have my internet connection set up yet |
| 04:20.15 | JT | vonage is crap |
| 04:20.33 | [TK]D-Fender | I mean to power the phone, do you mind a brick to plug into the wall verses being hoowed into a powered switch? |
| 04:20.40 | gthing | JT, Is there a better service in the price range? |
| 04:20.42 | [TK]D-Fender | powered* |
| 04:20.46 | JT | dozens |
| 04:20.51 | gthing | [TK]D-Fender, I do not mind a brick |
| 04:20.52 | JT | vonage is amongst the worst |
| 04:20.53 | [TK]D-Fender | JT : Let this sit for a bit, ok? |
| 04:20.59 | JT | [TK]D-Fender: ? |
| 04:22.21 | [TK]D-Fender | gthing, Are these 2 starter phone for more "exec" types? |
| 04:22.39 | gthing | [TK]D-Fender, No, just your basic cordless phones |
| 04:22.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:22.51 | gthing | [TK]D-Fender, I sense the list of my mistakes already growing |
| 04:23.16 | [TK]D-Fender | gthing, Ok, so you'd like to use just 2 boring cordless phones as extensions to your system? |
| 04:23.30 | gthing | [TK]D-Fender, That's right |
| 04:24.13 | [TK]D-Fender | gthing, perfectly fine, hrees the ONLY thing you needs aside from a WIMPY PC : http://www.telephonydepot.com/product_p/105-054-212.htm |
| 04:26.10 | gthing | [TK]D-Fender, Okay - so this allows me to connect the phones to the network, ya? |
| 04:26.22 | [TK]D-Fender | gthing, Correct |
| 04:26.34 | gthing | [TK]D-Fender, And is there some kind of switch I need? |
| 04:26.37 | _VoiceMeUp_COM | lol |
| 04:26.51 | _VoiceMeUp_COM | my cuba term is chaper via tdm pri's then all providers i have for intl |
| 04:26.52 | gthing | [TK]D-Fender, Each computer is the switch |
| 04:27.13 | [TK]D-Fender | gthing, They will be able to talk SIP to * and * will talk to your provider |
| 04:27.28 | [TK]D-Fender | gthing, Nothing special at all. |
| 04:27.31 | JT | _VoiceMeUp_COM: i could hardly understand that |
| 04:27.41 | [TK]D-Fender | gthing, Just plug that box on the same LAN as your server and you're in business |
| 04:28.16 | [TK]D-Fender | gthing, Full PBX for 75$ + cheap PC (I could buy one for 100$) and you already have the phones I gather |
| 04:28.54 | gthing | [TK]D-Fender, Okay so it goes phone line in ---> SPA2102 ---> asterisk servers ---> (via IP) to stations |
| 04:29.07 | gthing | [TK]D-Fender, right? |
| 04:29.53 | [TK]D-Fender | gthing, No it goes : (voip provider) ----> (Asterisk Server) ------ (Linksys SPA-2102) ----> (your analog cordless phones). |
| 04:30.38 | gthing | [TK]D-Fender, Okay |
| 04:31.18 | gthing | [TK]D-Fender, And the 2102 and * server can differentiate between the phones connected to it? |
| 04:32.28 | [TK]D-Fender | gthing, Yes, it has 2 ports each of which can operate completely independant of the other |
| 04:32.50 | gthing | [TK]D-Fender, Okay, so when I expand to 4 phones, I will need an additional or different box - gotcha |
| 04:33.39 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:33.48 | [TK]D-Fender | gthing, Depending on what kind of phones and style, etc you want. |
| 04:34.15 | [TK]D-Fender | gthing, You said you were expecting to use your 2 cordless, so I showed you a VERY inexpensive option to let you do just that |
| 04:34.24 | gthing | [TK]D-Fender, That's perfect |
| 04:34.28 | [TK]D-Fender | gthing, I suspect you deisre that kind of mobility for a few sets. |
| 04:34.54 | sweeper | ok, I need a decent distro that's not a pita to install. I'm too tired for gentoo >.< |
| 04:35.56 | snuff-work | centos is easy install for server |
| 04:36.10 | [TK]D-Fender | sweeper, I concure. |
| 04:36.14 | sweeper | yea, centos is what I'm uninstalling |
| 04:36.14 | [TK]D-Fender | sweeper, I concur* |
| 04:36.25 | [TK]D-Fender | UNINSTALLING? |
| 04:36.29 | JT | sweeper: debian |
| 04:36.34 | sweeper | overwriting more like |
| 04:36.40 | Nuitari | sweeper: gentoo |
| 04:36.41 | gthing | [TK]D-Fender, So that's the hardware - now what about vonage? |
| 04:36.50 | sweeper | actually, since I don't need zaptel, I think I'm gonna go with freebsd~ |
| 04:36.57 | gthing | [TK]D-Fender, It sounds like anyone around here would be willing to give me a rant :) |
| 04:37.17 | [TK]D-Fender | gthing, They are a flaming pile of shit, do not want anything to do with ahrdware they don't provide, and are threating to be wpied off the face of the planet. |
| 04:37.25 | [TK]D-Fender | gthing, Pick someone else :) |
| 04:37.37 | sweeper | yea, vonage is usually a ripoff |
| 04:37.43 | [TK]D-Fender | gthing, ACN is solid. telix is respected a fair bit as well. |
| 04:37.46 | gthing | [TK]D-Fender, Can you make some recommendations - I'm trying to stay in the price range |
| 04:37.51 | sweeper | I mean, you can get 1.5 cents a minute from lots of terminators |
| 04:38.12 | sweeper | who the hell talks for 1000 minutes a month? :P |
| 04:38.45 | gthing | sweeper, we will be spending a lot of time on the phone |
| 04:38.48 | sweeper | I probably don't even TALK talk that much, let alone on the phone :D |
| 04:39.01 | [TK]D-Fender | sweeper, VoicePulse Connect goes <.01$ for most US 48 |
| 04:39.23 | BSD_Tech | man this day is a pisser |
| 04:39.33 | sweeper | [TK]D-Fender: ooooo, that's nice |
| 04:39.37 | BSD_Tech | I cant find the box with all my dev cd's |
| 04:40.23 | [TK]D-Fender | sweeper, 1 hr/day = 1800 / month :) How about ANY BUSINESS? |
| 04:40.41 | gthing | So if I were to take a survey with people in this chatroom, who would end up on top as the best provider in my price range? ACN or Telix or someone else? |
| 04:40.42 | sweeper | so, not me >.> |
| 04:40.50 | [TK]D-Fender | 1800 * .02 (MAX!!!) = 36$ <- So who gives a shit? |
| 04:41.29 | [TK]D-Fender | gthing, Teliax has the most positive reviews from people I respect here. |
| 04:42.09 | BSD_Tech | what the uptime with vp like these days is it reliable |
| 04:42.27 | sweeper | oh man I love those US rates |
| 04:42.28 | [TK]D-Fender | gthing, I have clients using ACN, and others with VoicePulse, both happy. YMMV <- |
| 04:42.36 | BSD_Tech | ?me thinks teliax is a pain in the rump |
| 04:42.52 | [TK]D-Fender | BSD_Tech, what aspect specifcally? |
| 04:43.04 | BSD_Tech | but then again I use to work there and got screwed over |
| 04:43.20 | _VoiceMeUp_COM | waht your price range ? |
| 04:43.20 | [TK]D-Fender | Qwell : Fear not, you definitely count amongst them. |
| 04:43.37 | sweeper | hopefully I'll be cool enough to buy minutes from level3 someday :D |
| 04:43.45 | [TK]D-Fender | BSD_Tech, OH YEAH, my keyboard! |
| 04:43.53 | [TK]D-Fender | BSD_Tech, got a few? ;) |
| 04:44.01 | Qwell | sweeper: They won't even talk to you unless you meet some minimum usage criteria |
| 04:44.07 | BSD_Tech | about to go to bed I waited for you |
| 04:44.14 | BSD_Tech | we can try this fast |
| 04:44.22 | BSD_Tech | I am wiped |
| 04:44.26 | [TK]D-Fender | BSD_Tech, I pretty much forgot and you were so quiet! |
| 04:44.26 | sweeper | Qwell: I know, that's what I was referring to :P |
| 04:44.32 | [TK]D-Fender | BSD_Tech, another day then? |
| 04:44.35 | BSD_Tech | ok |
| 04:44.41 | BSD_Tech | I will not go far |
| 04:44.42 | sweeper | "yea, I'll be wanting 100k minutes this month. and 300 DIDs |
| 04:44.45 | BSD_Tech | I promiss |
| 04:44.45 | sweeper | ASAP plz |
| 04:44.56 | [TK]D-Fender | BSD_Tech, I should be home at a sane hour tomorrow. |
| 04:44.57 | Qwell | sweeper: yeah, 100k is nothing |
| 04:45.02 | _VoiceMeUp_COM | 500 minutes etc 9.99 |
| 04:45.06 | sweeper | I typed that and then I thought about it |
| 04:45.07 | BSD_Tech | tk how is vp these days is it reliable |
| 04:45.08 | gthing | For the size of my business - it would be best to go with an unlimited residential plan, ya? That's what vonage set me up on (seeing as how there is almost no difference between there residential and business plans except price) |
| 04:45.11 | _VoiceMeUp_COM | think we even offer 1 did |
| 04:45.17 | sweeper | ok |
| 04:45.17 | _VoiceMeUp_COM | dorr trying to get underline off |
| 04:45.19 | Qwell | a full T1 in a month can do about a million I believe it was |
| 04:45.20 | sweeper | 10M! |
| 04:45.26 | [TK]D-Fender | BSD_Tech, I don't have any more modern reports on them now. |
| 04:45.33 | *** part/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 04:45.34 | BSD_Tech | ok |
| 04:45.38 | _VoiceMeUp_COM | ggggggg |
| 04:45.40 | _VoiceMeUp_COM | ok sorry |
| 04:45.43 | sweeper | whatever |
| 04:45.45 | BSD_Tech | night kids |
| 04:45.53 | _VoiceMeUp_COM | nicgh tech |
| 04:46.06 | sweeper | if this restful voip thing takes off, maybe I'll hire mark spencer to write me a good pbx or something... ;) |
| 04:46.10 | Qwell | 993600 minutes per month on a PRI |
| 04:46.14 | Qwell | assuming 24/7/30 |
| 04:46.26 | _VoiceMeUp_COM | yeah more like 10 hours |
| 04:46.37 | _VoiceMeUp_COM | 8 to 13pm 14pm to 6pm 7pm to 9pm |
| 04:46.44 | Qwell | sure |
| 04:46.53 | _VoiceMeUp_COM | thats 12 lol |
| 04:47.04 | gthing | [TK]D-Fender, well I'll be honest ACN scares me because it says something about MLMs - which I steer clear of or die trying |
| 04:47.05 | _VoiceMeUp_COM | but you where talking on ful usage.. |
| 04:47.09 | _VoiceMeUp_COM | so i guesyour right |
| 04:47.28 | [TK]D-Fender | gthing, Oh no... not the MLM BS side.. I know big-leagers who jsut use their service direct. |
| 04:47.30 | _VoiceMeUp_COM | wow i was wodnering who you talking about |
| 04:47.32 | _VoiceMeUp_COM | acn lol |
| 04:47.38 | sweeper | see, I have a dream |
| 04:47.50 | _VoiceMeUp_COM | yeah everyone i know went to meeteings paid 500$ and know trying to get it back scamming theyr family |
| 04:48.13 | [TK]D-Fender | gthing, I was approached by one of their ilk once..... don't worry, they're well on their way to becomeing AMWAY MILLIONAIRES thanks to me! ;) |
| 04:48.15 | sweeper | a dream where I'm buying cisco pri->sip gateways by the dozen, just to get good rates on DIDs and local termination |
| 04:48.23 | _VoiceMeUp_COM | also said they where going to do internet access broadband and voip for 2 years now |
| 04:48.33 | JT | dreaming of buying cisco? that's pretty sad |
| 04:48.53 | _VoiceMeUp_COM | u can get a cheak 38XX for like 1k i think |
| 04:48.57 | _VoiceMeUp_COM | cheap |
| 04:48.58 | sweeper | JT: find me a better sip gateway for the price, and I'll dream of something else |
| 04:49.12 | sweeper | you can stick 8 pri cards in those things |
| 04:49.24 | JT | in how many RU? |
| 04:49.27 | sweeper | 1 |
| 04:49.38 | sweeper | you could do the same thing with a sangoma |
| 04:49.48 | sweeper | but then you're running on pc hardware |
| 04:49.49 | JT | you could in theory make your own box in 1RU that does 8, yes :) |
| 04:49.53 | _VoiceMeUp_COM | sangoma 108d is how much appro X? |
| 04:49.54 | JT | there are other sip gateways |
| 04:50.01 | sweeper | yes, but better? |
| 04:50.03 | JT | i'm get to be convinced cisco is the best |
| 04:50.11 | JT | _VoiceMeUp_COM: USD$4500 iirc |
| 04:50.13 | [TK]D-Fender | JT : You mean like the AudioCodes Mediant 2000.... ALREADY DOES? |
| 04:50.13 | sweeper | from what I've seen, those things are pretty rock solid |
| 04:50.16 | _VoiceMeUp_COM | cheap |
| 04:50.24 | _VoiceMeUp_COM | let me open price list ill confirm |
| 04:50.25 | sweeper | arg audiocodes |
| 04:50.29 | sweeper | fuck them man |
| 04:50.36 | [TK]D-Fender | sweeper, they work. |
| 04:50.40 | sweeper | well |
| 04:50.43 | _VoiceMeUp_COM | stay away from mediatrix too |
| 04:50.48 | [TK]D-Fender | _VoiceMeUp_COM, A108d = $$$ |
| 04:50.48 | sweeper | their 8-port fxos sure as hell don't |
| 04:50.53 | sweeper | I had 4 of them |
| 04:51.06 | sweeper | and was on the phone with audiocods AND the vendor for 6+ hours |
| 04:51.12 | [TK]D-Fender | _VoiceMeUp_COM, Mediatrix makes GREAT analog gateways, but their PRI stuff is BULLSHIT. |
| 04:51.17 | sweeper | vnc, ssh into asterisk box, the whole nine yards |
| 04:51.23 | sweeper | NOTHING |
| 04:51.34 | JT | [TK]D-Fender: so who makes good pri gateways? |
| 04:51.36 | sweeper | finally went with that crazy linksys thing that kinda runs asterisk |
| 04:51.39 | [TK]D-Fender | sweeper, I've set up a Mediant 1000. YUO = suck :) |
| 04:51.47 | _VoiceMeUp_COM | 108d 4.5k |
| 04:51.50 | _VoiceMeUp_COM | approx |
| 04:51.51 | sweeper | <PROTECTED> |
| 04:52.02 | _VoiceMeUp_COM | cisco 53xx 25k |
| 04:52.02 | *** join/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net) |
| 04:52.03 | JT | _VoiceMeUp_COM: that's what i said isn't it ;) |
| 04:52.06 | _VoiceMeUp_COM | loaded |
| 04:52.14 | _VoiceMeUp_COM | jt yeas you are right |
| 04:52.15 | sweeper | _VoiceMeUp_COM: you'd want echo cancellation in hardware, with 8 ports mang |
| 04:52.25 | [TK]D-Fender | _VoiceMeUp_COM, Well if you jsut want G711, the Sangoma is fine, but NO BOX you you put it in would survive transcoding it all :) |
| 04:52.26 | _VoiceMeUp_COM | 108D;) |
| 04:52.31 | JT | _VoiceMeUp_COM: add $2-3k for a really good server chassis |
| 04:52.36 | _VoiceMeUp_COM | bladecenter |
| 04:52.38 | _VoiceMeUp_COM | as we use |
| 04:52.40 | _VoiceMeUp_COM | 4gig nics |
| 04:52.41 | JT | for a fair comparison |
| 04:52.43 | [TK]D-Fender | _VoiceMeUp_COM, Which explains the cost on the appliance ones :) |
| 04:52.46 | _VoiceMeUp_COM | 4 power supplys |
| 04:52.51 | _VoiceMeUp_COM | redundancy top notch |
| 04:52.54 | _VoiceMeUp_COM | 14 in 7 |
| 04:52.57 | _VoiceMeUp_COM | 14 in 7u |
| 04:53.06 | sweeper | or buy 2 ciscos for half the price :o |
| 04:53.18 | JT | 14 what? |
| 04:53.23 | sweeper | blades |
| 04:53.24 | JT | whose bladecentre? |
| 04:53.27 | _VoiceMeUp_COM | 14 bblaes in 7u |
| 04:53.27 | J4k3 | sweeper: maybe he's using P2-400 blades? |
| 04:53.30 | J4k3 | ebay surplus 4 life |
| 04:53.31 | J4k3 | !! |
| 04:53.36 | sweeper | w3rd |
| 04:53.38 | _VoiceMeUp_COM | ibm |
| 04:53.49 | sweeper | dude when I get a basement |
| 04:53.50 | _VoiceMeUp_COM | i use 2 for pci expensions that hold 2 card each |
| 04:53.56 | sweeper | I'm gonna rape ebay for that kind of stuff |
| 04:53.56 | _VoiceMeUp_COM | to make 4 pri's with failover |
| 04:53.59 | _VoiceMeUp_COM | so i use 12 blades |
| 04:54.13 | _VoiceMeUp_COM | dual xeon 2.8 |
| 04:54.25 | sweeper | also |
| 04:54.29 | JT | i'm not convinced about blades |
| 04:54.30 | sweeper | the ciscos scale |
| 04:54.35 | JT | if the whole things die |
| 04:54.37 | JT | you'll be very sad |
| 04:54.41 | sweeper | with 4 psus? |
| 04:54.43 | JT | s/things/thing/ |
| 04:54.49 | JT | sweeper: there's more to a server than psus |
| 04:54.55 | _VoiceMeUp_COM | trying to read dmesg |
| 04:55.02 | _VoiceMeUp_COM | <PROTECTED> |
| 04:55.08 | JT | also datacentres hate blades |
| 04:55.17 | sweeper | because they use less space :P |
| 04:55.19 | _VoiceMeUp_COM | is ee 4 |
| 04:55.24 | sweeper | they probably keel the AC tho |
| 04:55.24 | _VoiceMeUp_COM | so dual xoen is HT i guess |
| 04:55.37 | _VoiceMeUp_COM | yeah and you need a 240 DROP instead of 120v |
| 04:55.43 | _VoiceMeUp_COM | so they charge you extra |
| 04:55.45 | sweeper | really? wtf |
| 04:55.49 | _VoiceMeUp_COM | but 14 u in 7 is nice |
| 04:55.54 | _VoiceMeUp_COM | stack 2 blade per rack |
| 04:55.56 | J4k3 | 48v 4 life. |
| 04:56.02 | sweeper | haha 48v |
| 04:56.11 | sweeper | J4k3 lives on a boat or an oil rig? :P |
| 04:56.18 | JT | sweeper: umm, they absolutely kill most datacentres' power density setups |
| 04:56.23 | J4k3 | sweeper: optimally I'd live in a decom'd CO. |
| 04:56.23 | _VoiceMeUp_COM | plus every thing dual and raided.. or quadruple like powersupplies |
| 04:56.33 | J4k3 | that'd be the bomb dizzle |
| 04:56.35 | sweeper | JT: so they use less space :D |
| 04:56.35 | _VoiceMeUp_COM | dont think that thing could die if i hamemred it down |
| 04:56.37 | JT | sweeper: most datacentres don't have enough power to fill the whol place with racks full of blades |
| 04:56.50 | J4k3 | JT: or hvac |
| 04:56.56 | JT | right |
| 04:56.57 | sweeper | pfft |
| 04:57.01 | _VoiceMeUp_COM | 1800w fans |
| 04:57.02 | _VoiceMeUp_COM | lol |
| 04:57.10 | _VoiceMeUp_COM | cools off 2 rows behind it |
| 04:57.13 | sweeper | so I'm supposed to be sorry because their infrastructure isn't up to supporting me? |
| 04:57.21 | J4k3 | outside air venting ftw |
| 04:57.30 | _VoiceMeUp_COM | arctic |
| 04:57.36 | J4k3 | who cares about the AC going out when its NEGATIVE EIGHTY outside. |
| 04:57.37 | _VoiceMeUp_COM | wonder who gets assigned to go fix things there |
| 04:57.38 | JT | sweeper: no, you're supposed to be sorry for buying the dream without seeing if it's the reality ;) |
| 04:57.44 | J4k3 | _VoiceMeUp_COM: people live there |
| 04:57.55 | _VoiceMeUp_COM | yeah.. still ;) |
| 04:58.00 | sweeper | JT: when I can afford a blade, I'll be able to afford the colo for it :P |
| 04:58.02 | _VoiceMeUp_COM | save on cooling that for sure |
| 04:58.08 | _VoiceMeUp_COM | then you need lots of fiber |
| 04:58.09 | J4k3 | http://www.paulwberg.com/tuktoyaktuk |
| 04:58.12 | sweeper | hurricane electric is my top choice atm~ |
| 04:58.18 | J4k3 | the fiber might already be there |
| 04:58.21 | J4k3 | in places |
| 04:58.22 | sweeper | but it's mostly because they have a cool name XD |
| 04:58.24 | _VoiceMeUp_COM | hear bad things |
| 04:58.31 | JT | anyway, if the blade chassis dies, you're screwed, long story short |
| 04:58.32 | _VoiceMeUp_COM | get serverbeach since htey got peer1 acquired |
| 04:58.37 | _VoiceMeUp_COM | they push alot on peer1 |
| 04:58.42 | [TK]D-Fender | _VoiceMeUp_COM, What else do your Rice Crispies say to you?! ;) |
| 04:58.59 | J4k3 | I didn't do any voip with it, though |
| 04:59.15 | J4k3 | JT: buy two, they're small |
| 04:59.16 | `Sean | Hey [TK]D-Fender do you have a soloution for this |
| 04:59.16 | `Sean | <PROTECTED> |
| 04:59.20 | J4k3 | and colo them across the world from each other |
| 04:59.20 | JT | he is only about 7 hops from australia, it's a good location |
| 04:59.29 | J4k3 | and seamlessly fail over when the poo hits the fan |
| 04:59.42 | `Sean | im basicly in a conference local one wich is setup on my server i wanna call someone else basicly 3 way and its avoiding is there a wya to stop it from doing that? |
| 05:00.06 | JT | J4k3: i'd rather get a bunch of seperate boxes not sharing so much infrastructure, and in a few datacentres :) |
| 05:00.07 | _VoiceMeUp_COM | jt your right |
| 05:00.12 | _VoiceMeUp_COM | if chassis dies |
| 05:00.14 | [TK]D-Fender | `Sean, It clearly doesn't want to let you so STOP. |
| 05:00.17 | _VoiceMeUp_COM | ;) |
| 05:00.46 | `Sean | [TK]D-Fender there must be a way tho |
| 05:00.51 | J4k3 | JT: same here. |
| 05:00.53 | [TK]D-Fender | `Sean, What phone are you on that causes that? |
| 05:00.53 | JT | i think enterprise servers/mainframes are better options than blades at the high end of town |
| 05:00.59 | [TK]D-Fender | `Sean, Thats what DENIAL is for... |
| 05:01.02 | sweeper | J4k3: yea man, run the linuxha heartbeat over serial over ip, and the giggle with glee when someone drops a packet and your backup tries to take over services from the primary XD |
| 05:01.03 | JT | an IBM Z series would go down nicely |
| 05:01.03 | `Sean | [TK]D-Fender Zap1 |
| 05:01.10 | `Sean | Analogue phone |
| 05:01.15 | sweeper | JT: nah man, it's all about the GoogleWAy |
| 05:01.16 | _VoiceMeUp_COM | anyone tried the mac servers ? |
| 05:01.19 | _VoiceMeUp_COM | they nice but... |
| 05:01.21 | sweeper | lots and lots of whiteboxes XD |
| 05:01.23 | _VoiceMeUp_COM | anycomments ? |
| 05:01.26 | _VoiceMeUp_COM | apart the price |
| 05:01.29 | JT | sweeper: that's definitely one way |
| 05:01.43 | J4k3 | voip is too nit-picky for junk pcs |
| 05:01.46 | JT | as far as x86 goes, i've been most impressed with the NEC stuff |
| 05:01.53 | sweeper | J4k3: nonono |
| 05:01.53 | JT | running dual mobos/etc in lockstep |
| 05:01.57 | J4k3 | web serving/databases are pretty lax when it comes to timing |
| 05:02.03 | JT | high availability servers |
| 05:02.03 | _VoiceMeUp_COM | wow |
| 05:02.07 | sweeper | DIGIUM HARDWARE is too picky |
| 05:02.15 | _VoiceMeUp_COM | tell mw where |
| 05:02.16 | [TK]D-Fender | `Sean, Clearly Zaptel doesn't like working that hard to synch so many confreences. *TFB* get something else. |
| 05:02.19 | sweeper | please don't lump all of voip together :P |
| 05:02.21 | JT | if cpu or motherboard fails, the hardware backplane disables it and only uses the other motherboard |
| 05:02.27 | [TK]D-Fender | Zaptel FXS = ASS. |
| 05:02.34 | sweeper | s/fxs// |
| 05:02.35 | J4k3 | sweeper: I have a pure ip setup on a shitty system that loves to hiccup, the only thing I can get it to link to is timing slip |
| 05:02.35 | _VoiceMeUp_COM | so anyone tries Apple Xserve ? |
| 05:02.38 | Supaplex | kiss my FXS! |
| 05:02.49 | J4k3 | I've since switched to a nice stable intel system that doesn't misbehave. |
| 05:02.51 | `Sean | [TK]D-Fender im trying to call via SIP the only the phone is via ZAP all incoing outgoing calls are done via SIP termination |
| 05:02.53 | JT | they look like a waste of money |
| 05:03.02 | JT | especially since xserve is now intel, BORING |
| 05:03.10 | sweeper | J4k3: honestly, it's gotta be a really, really shitty system if it did that :v |
| 05:03.11 | _VoiceMeUp_COM | ahah |
| 05:03.12 | JT | the advantage they used to have is now gone |
| 05:03.12 | _VoiceMeUp_COM | true |
| 05:03.23 | sweeper | I mean, if your USB stuff is crapping out? :V |
| 05:03.24 | JT | low power usage, decent risc performance |
| 05:03.27 | _VoiceMeUp_COM | waht that nec model ? |
| 05:03.28 | J4k3 | sweeper: very shitty. |
| 05:03.28 | JT | now they're the same shit |
| 05:03.28 | _VoiceMeUp_COM | jt |
| 05:03.33 | JT | 5800 iirc |
| 05:03.33 | [TK]D-Fender | `Sean, You phone is Zaptel, sos the timer for your Meetme. They can't cooperate. JUST F'N DEAL WITH IT. |
| 05:03.45 | J4k3 | via km400 ddr chipset athlon xp 2000+ |
| 05:03.54 | J4k3 | with good cl2.5 PC2700 in it. |
| 05:03.54 | sweeper | bahaha |
| 05:03.59 | JT | _VoiceMeUp_COM: yep, 5800 series NEC |
| 05:04.14 | _VoiceMeUp_COM | http://crs-usa.com/nec-5000.asp |
| 05:04.15 | _VoiceMeUp_COM | ? |
| 05:04.16 | _VoiceMeUp_COM | kik\ |
| 05:04.17 | J4k3 | it'd randomly take 100 ms-ish (guessing) naps and whatnot |
| 05:04.18 | _VoiceMeUp_COM | j/k |
| 05:04.25 | sweeper | awsome |
| 05:04.50 | JT | http://www.nec.co.jp/express/products/f_tolerant/ |
| 05:05.26 | JT | i saw them at cebit here, they were awesome |
| 05:05.34 | JT | watecooled cpus, hotswap mobos |
| 05:05.36 | _VoiceMeUp_COM | hmm i had necct.co.jp for soemreason |
| 05:05.38 | _VoiceMeUp_COM | google mix |
| 05:05.47 | _VoiceMeUp_COM | LOL |
| 05:05.50 | _VoiceMeUp_COM | look at the tile |
| 05:05.51 | JT | you can split the 2 mobos just like splitting a raid 1 array |
| 05:05.56 | _VoiceMeUp_COM | FAULT tolerant server |
| 05:06.02 | _VoiceMeUp_COM | Running on windows server 2003 rc2 |
| 05:06.05 | JT | so you can take one offline to upgrade software |
| 05:06.07 | _VoiceMeUp_COM | AKA until you put windows |
| 05:06.09 | JT | heh |
| 05:06.10 | _VoiceMeUp_COM | its failt tolerant |
| 05:06.32 | JT | i thought they were blades at first |
| 05:06.41 | _VoiceMeUp_COM | JT you made my day |
| 05:06.48 | _VoiceMeUp_COM | pushing this to partner |
| 05:07.03 | _VoiceMeUp_COM | they come with a high price i assume |
| 05:07.16 | JT | yeah, around $20k AUD |
| 05:07.21 | _VoiceMeUp_COM | aud=usd |
| 05:07.31 | JT | probably be less in the us |
| 05:07.32 | J4k3 | usd = worthless |
| 05:07.33 | sweeper | what the fuck? watercooling in a production system? |
| 05:07.33 | _VoiceMeUp_COM | lol nevermind goign to xe.com |
| 05:07.36 | sweeper | shoot me now |
| 05:07.39 | sweeper | like, really |
| 05:07.47 | sweeper | what datacenter would have you? |
| 05:07.49 | J4k3 | water cooling in racks is pretty tight |
| 05:07.54 | JT | sweeper: they need it due to the space density requirements |
| 05:07.55 | J4k3 | sweeper: a smart one |
| 05:07.56 | J4k3 | its cheaper. |
| 05:08.01 | JT | sweeper: the water stays internal |
| 05:08.05 | JT | it doesn't go anyway |
| 05:08.07 | JT | anywhere |
| 05:08.10 | J4k3 | ahh, bummer. |
| 05:08.12 | JT | it's just a heat pipe system |
| 05:08.14 | _VoiceMeUp_COM | 1 wilshire |
| 05:08.15 | J4k3 | the proper way to do this is pump it all out. |
| 05:08.21 | sweeper | it's cheaper until it sprays all over the rack :v |
| 05:08.21 | _VoiceMeUp_COM | maybe |
| 05:08.33 | _VoiceMeUp_COM | has open sahr eon peer1 plus lots of asia singapore etc routes |
| 05:08.34 | J4k3 | sweeper: pull a vacuum on the system. |
| 05:08.38 | _VoiceMeUp_COM | university link there |
| 05:08.40 | sweeper | bah |
| 05:08.45 | J4k3 | and circulate |
| 05:08.47 | sweeper | just do what the VSAT amps do |
| 05:08.48 | J4k3 | wouldn't require much |
| 05:08.52 | J4k3 | put an alarm on it when it leaks |
| 05:08.58 | sweeper | ac unit IN THE BOX :D |
| 05:09.04 | J4k3 | haha |
| 05:10.20 | sweeper | and then there's the ones that take a direct feed from the HVAC |
| 05:10.45 | _VoiceMeUp_COM | i want 7 of 9 |
| 05:11.01 | sweeper | I did when I was 13 |
| 05:11.06 | _VoiceMeUp_COM | lol |
| 05:11.12 | _VoiceMeUp_COM | shes has mainfram access |
| 05:11.17 | sweeper | then I realized that metal is not conducive to good sex |
| 05:11.29 | JT | lies |
| 05:11.31 | sweeper | especially not when it assimilates you for not using a condom |
| 05:11.33 | _VoiceMeUp_COM | tdepends on voltage and where its paplied |
| 05:11.33 | _VoiceMeUp_COM | ;) |
| 05:11.55 | JT | sweeper: i've seen the stuff direct connected to water chiller pipes |
| 05:12.04 | JT | big render farm in a datacentre here |
| 05:12.07 | *** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-89.myvzw.com) |
| 05:12.12 | JT | do stuff for major animated movies |
| 05:12.13 | _VoiceMeUp_COM | no way |
| 05:12.18 | _VoiceMeUp_COM | awesome |
| 05:12.29 | sweeper | I dunno |
| 05:12.29 | JT | like happy feet, etc |
| 05:12.33 | sweeper | that takes balls |
| 05:12.39 | JT | or proper planning |
| 05:12.42 | JT | your choice :) |
| 05:12.48 | sweeper | nah, balls |
| 05:13.11 | sweeper | one person drops a wrench, bam, 5-10 servers gone |
| 05:13.57 | JT | i think they have hundreds of nodes, but were having serious problems in their old cage that was just process a/c cooled |
| 05:14.07 | JT | overheating problems |
| 05:14.24 | sweeper | well, they still have to pump out the same amount of heat |
| 05:14.30 | sweeper | the compressor and stuff has to do the same amount of work |
| 05:14.42 | JT | water has a higher heat density |
| 05:14.46 | sweeper | yea |
| 05:14.49 | JT | much more efficient |
| 05:15.00 | JT | and it gets rid of the big fan blower units |
| 05:15.14 | sweeper | you still need big fan blower units, dude |
| 05:15.25 | JT | not so much |
| 05:15.50 | [TK]D-Fender | ok, checkout time... later all |
| 05:16.23 | _VoiceMeUp_COM | right now blade1 poushing only 2100 wats |
| 05:16.39 | JT | the whole bladecentre? |
| 05:16.44 | _VoiceMeUp_COM | yeah |
| 05:16.48 | _VoiceMeUp_COM | checking |
| 05:16.58 | _VoiceMeUp_COM | got 4 times 1875 W |
| 05:17.08 | _VoiceMeUp_COM | Allocated Power (Max) 1275W 820W |
| 05:17.59 | _VoiceMeUp_COM | 1 72% |
| 05:18.03 | _VoiceMeUp_COM | 2 73% |
| 05:18.08 | _VoiceMeUp_COM | runnign 73% speed |
| 05:18.08 | _VoiceMeUp_COM | nice |
| 05:18.09 | _VoiceMeUp_COM | lol |
| 05:18.13 | _VoiceMeUp_COM | temp 27 c |
| 05:18.17 | _VoiceMeUp_COM | cool |
| 05:18.20 | _VoiceMeUp_COM | ok im out night |
| 05:20.24 | *** join/#asterisk robby___ (n=robby@203.63.126.9) |
| 05:21.14 | robby___ | Hi all, getting a weird crash with asterisk 1.2.7 |
| 05:21.25 | robby___ | very intermittant, doesnt appear to be anything out of the ordinary on the console when it happens |
| 05:21.31 | _VoiceMeUp_COM | maybe the sip thing ? |
| 05:21.32 | robby___ | app_queue.so will just go away |
| 05:21.34 | _VoiceMeUp_COM | with fake contact |
| 05:21.38 | _VoiceMeUp_COM | ah |
| 05:21.49 | _VoiceMeUp_COM | bad wav ? |
| 05:22.03 | robby___ | doing show queues on the console just brings up a blank new line, reload app_queue.so doesnt bring it back |
| 05:22.09 | robby___ | i actually have to kill and restart asterisk |
| 05:22.10 | _VoiceMeUp_COM | think one bug with a wav was messing up something before in like 2 months ago |
| 05:22.35 | robby___ | _VoiceMeUp_COM: that in regard to me? or someone else |
| 05:22.35 | _VoiceMeUp_COM | id start by reducing number of agents quees to 1 and see if happens |
| 05:22.42 | _VoiceMeUp_COM | noyou i guess |
| 05:22.50 | _VoiceMeUp_COM | not even sure it applie |
| 05:22.52 | _VoiceMeUp_COM | s |
| 05:30.45 | gthing | [TK]D-Fender, thanks for all your help - you've given me everything I needed to know (for now :D ) |
| 05:32.44 | flenders | -!- [TK]D-Fender [n=Joe@64.235.216.2] has quit |
| 05:34.25 | gthing | oh |
| 05:34.33 | gthing | :0 |
| 05:35.47 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 05:43.48 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-226.bwic.chi.wayport.net) |
| 05:48.13 | *** join/#asterisk gardo (n=gardo@121.97.209.126) |
| 05:49.36 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@83.223.131.104) |
| 05:49.44 | aiksa[LV] | morning! |
| 05:50.26 | aiksa[LV] | Could anyone guide me if i should report this as a bug to the tracker or this is known issue... |
| 05:50.55 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 05:51.00 | aiksa[LV] | I have a setup where agents logs in to asterisk through AMI |
| 05:51.17 | zeeesh | hi all |
| 05:51.23 | aiksa[LV] | the devices they use to acept incomming calls - BT 102 |
| 05:51.27 | aiksa[LV] | zeeesh, hi |
| 05:51.39 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 05:51.59 | *** join/#asterisk vn (n=nostalge@bas5-quebec14-1128557048.dsl.bell.ca) |
| 05:52.45 | aiksa[LV] | ]now - if due to a network problems asterisk cant connect one of of those devices (no route to host) sometimes these channels leave hanging till the box is restarted |
| 05:52.54 | aiksa[LV] | is this known issue? |
| 05:58.54 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:02.24 | rue_mohr | Asterisk SIP Setup A basic HOW-To on configuring SIP, Extensions, and Voicemail .conf files |
| 06:02.34 | rue_mohr | can anyone tell me where I can find that howto? |
| 06:02.43 | rue_mohr | http://www.voip-info.org/wiki-Asterisk |
| 06:02.46 | rue_mohr | link on there is dead |
| 06:03.49 | *** part/#asterisk dhakatel (n=root@58.65.224.5) |
| 06:06.18 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:12.56 | *** join/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 06:16.27 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 06:21.54 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:22.54 | swagger | rue_mohr: check out trixbox |
| 06:24.28 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:34.19 | *** join/#asterisk oej (n=olle@174.82-134-80.bkkb.no) |
| 06:34.38 | |R | anyone tried the Twin phone with an asterisk setup? |
| 06:35.39 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 06:38.15 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-165-37.columbus.res.rr.com) |
| 06:40.30 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:54.40 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-226.bwic.chi.wayport.net) |
| 07:00.45 | *** join/#asterisk jmcadam (i=jmcadam@60-240-83-50-nsw-pppoe.tpgi.com.au) |
| 07:01.30 | jmcadam | hey is anyone who knows anything about bug #8507 around? |
| 07:02.31 | snuff-work | M8507 |
| 07:08.00 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 07:12.21 | mvanbaak | snuff-work: that only works in #asterisk-dev |
| 07:18.32 | snuff-work | i realised after i did it ;) |
| 07:18.36 | snuff-work | no muffinman |
| 07:24.41 | vlt | Hello. How can I set a queue member to a lower priority (so that it is only called when the others are busy/congested? |
| 07:26.41 | vlt | Ok, found it: Just add a priority value: "member => Zap/g1/10,1" for example. |
| 07:41.16 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 07:44.08 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
| 07:44.28 | *** join/#asterisk kova (n=kova@tech.quentris.com) |
| 07:44.47 | kova | hello |
| 07:46.18 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
| 07:48.13 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 07:49.46 | mightnare | just a simple question guys... if i do Set(variablename=value), will the value persist even after the channels dies? |
| 07:53.59 | Corydon76-home | No |
| 07:54.35 | Corydon76-home | If you want a global variable, though, you can use SetGlobalVar (deprecated in 1.4) or Set(GLOBAL(var)=value) |
| 07:55.26 | Corydon76-home | Global variables persist until restart |
| 07:55.50 | Corydon76-home | If you want variables that last longer, use a database (see func_odbc.conf in 1.4) |
| 07:56.30 | awk | anyone know if there is going to bea new asterisk book released soon |
| 07:56.39 | awk | something with details about 1.4, etc |
| 07:56.48 | Corydon76-home | awk: yes, by the end of the summer |
| 07:57.11 | Corydon76-home | O'Reilly, TFOT, 2nd edition |
| 07:57.16 | awk | ok, so by the end of our winter :) |
| 07:57.22 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 07:57.29 | Corydon76-home | within the next 2 months |
| 07:57.55 | awk | nice |
| 07:58.04 | Corydon76-home | At last check, the final proof was in the hands of the author team |
| 07:58.30 | awk | so will you guys make the ebook available to the public or will that be charged |
| 07:58.45 | Corydon76-home | Not at first, but eventually, yes |
| 07:59.24 | awk | great, I dislike reading books on the screen anyway :D I should take a pic of my library 1 day :) |
| 07:59.28 | awk | anyway & |
| 08:01.00 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:15.17 | *** join/#asterisk gardo (n=gardo@121.97.209.126) |
| 08:17.06 | *** join/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 08:17.46 | Phuntom | hi ya! need help with A. :-) |
| 08:19.52 | Phuntom | any alive? |
| 08:21.17 | HarryR | Possibly... |
| 08:21.36 | Phuntom | :-) |
| 08:22.47 | Phuntom | how can i get monitored ststus of the clients? |
| 08:23.02 | HarryR | see who's online and their status? |
| 08:23.12 | Phuntom | yeah |
| 08:23.47 | Phuntom | i type sip show users and get in status column - unmonitored |
| 08:24.03 | Phuntom | how can i switch this function? |
| 08:24.06 | HarryR | ah, dont think I can help with that |
| 08:25.16 | Phuntom | i see :-( |
| 08:31.06 | YonahW-Work | Phuntom: I think qualify=yes in sip.conf will monitor the status |
| 08:32.25 | *** join/#asterisk dec_ (n=tom@unaffiliated/dec) |
| 08:33.51 | Phuntom | in which section? |
| 08:35.49 | YonahW-Work | each client |
| 08:37.09 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-226.bwic.chi.wayport.net) |
| 08:38.19 | Phuntom | no such section, or you mean i have to write this in each section? |
| 08:39.14 | *** join/#asterisk saftsack (n=saftsack@pD9E07409.dip.t-dialin.net) |
| 08:39.37 | YonahW-Work | yes |
| 08:39.53 | JT | couldn't you just set it in general? |
| 08:40.49 | Phuntom | no |
| 08:41.10 | JT | why not? |
| 08:41.15 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:43.05 | Phuntom | btw anyone know is there in asterisknow normal text editor excl. vi? |
| 08:44.09 | Strom_M | excl.? |
| 08:44.19 | Phuntom | excluding |
| 08:44.27 | berktr | hello |
| 08:44.33 | Strom_M | welcome to IRC, where you don't have to abbreviate things |
| 08:44.54 | berktr | when i place a call using asterisk, i see -- Called PSTN/902323768056 on my asterisk CLI |
| 08:45.04 | berktr | my phone starts ringing |
| 08:45.10 | berktr | but i don't see it ringing on cli |
| 08:45.14 | berktr | and when i answer the phone |
| 08:45.22 | berktr | it still doesn't show it answered |
| 08:45.44 | berktr | i hear the ringing tone on my sip phone but the other phone is off hook |
| 08:45.49 | berktr | why is this? |
| 08:46.46 | Strom_M | berktr: how are you connecting to the PSTN, and what's the Dial() line in extensions.conf? |
| 08:48.04 | berktr | PSTN is the alias for my provider's sip peer account |
| 08:48.20 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 08:49.02 | Strom_M | berktr: and what's the Dial() line in extensions.conf? |
| 08:49.26 | berktr | SIP/PSTN/905333585867|60|RrL(3600000:61000:30000)) |
| 08:49.31 | berktr | for that number |
| 08:49.53 | berktr | this is normally working |
| 08:49.59 | berktr | but sometimes this problem occurs |
| 08:50.20 | berktr | it was working for 2 weeks |
| 08:50.31 | berktr | however today, 15 mins ago the problem started |
| 08:50.34 | Strom_M | why are you using the r flag? |
| 08:51.38 | berktr | because sometimes PSTN is not working and my system switches to other provicer |
| 08:51.40 | berktr | provider |
| 08:52.01 | berktr | and during this time, the ringing tone helps me to fool the caller |
| 08:52.30 | Strom_M | yeah, that's dumb |
| 08:52.31 | Strom_M | don't do that |
| 08:52.48 | berktr | you don't know how stupid the people that i work here are |
| 08:53.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:53.47 | berktr | anyways |
| 08:53.50 | berktr | any ideas? |
| 08:53.51 | Strom_M | well how about turning it off temporarily for testing purposes? |
| 08:53.59 | berktr | i've done that |
| 08:54.01 | berktr | no change |
| 08:54.06 | Strom_M | well then bitch at your provider |
| 08:54.28 | berktr | my primary,secondary and third provider have the same problem |
| 08:54.36 | berktr | so i don't think it's a provider related problem |
| 08:54.40 | Strom_M | did you do *anything* to the system? |
| 08:55.42 | berktr | jeez |
| 08:55.46 | berktr | it is working back again... |
| 08:55.47 | berktr | grrr |
| 08:56.05 | Strom_M | what version of asterisk are you running? |
| 08:56.08 | berktr | 1.4 |
| 08:56.20 | berktr | 1.4.3 |
| 08:56.29 | Strom_M | 1.4.5 is out now |
| 08:56.52 | berktr | sure but i use freebsd and have to wait for ports collection to be updated |
| 08:56.59 | Strom_M | boners |
| 08:57.20 | berktr | and in case of a system problem |
| 08:57.24 | berktr | i will be kicked out |
| 08:57.27 | berktr | for sure |
| 08:57.41 | Strom_M | i'd blame the provider |
| 08:57.49 | berktr | they fuckin don't care |
| 08:57.50 | Strom_M | they may all share the same upstream provider |
| 08:58.00 | berktr | maybe yeah |
| 08:58.24 | *** join/#asterisk zver (n=zveruga@ido-rtr2.distance.ru) |
| 08:58.39 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:59.49 | zver | hello. I get many errors in CLI: chan_zap.c:4888 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 28, when i call over E1. Why ? |
| 09:00.36 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 09:08.44 | cheshair | hi guys! |
| 09:16.28 | *** join/#asterisk DJ_Stefan0 (n=sluxor@203-59-81-230.dyn.iinet.net.au) |
| 09:17.27 | berktr | how can i set asterisk to send 40bytes payload for h323? |
| 09:18.19 | DJ_Stefan0 | my soft phones are registering with my PBX , my hardware phone isnt.. any ideas? Do I need to specify something for hardware phones in the configuration? |
| 09:19.47 | DJ_Stefan0 | i am configuring all to run using SIP.. so I wrote a simple sip.conf and extensions.conf |
| 09:21.13 | creativx | turn on sip debugging DJ_Stefan0 |
| 09:21.17 | creativx | and see if the hardphone even reaches aterisk |
| 09:21.36 | DJ_Stefan0 | its reaching, i get a tone |
| 09:21.39 | DJ_Stefan0 | it just doesnt register |
| 09:21.54 | Phuntom | how to reset asterisk? |
| 09:22.22 | Phuntom | im debugging but it send options packet all the time to switched off host |
| 09:22.46 | creativx | the hardphone will give you a tone on its own |
| 09:23.02 | creativx | it can be disconnected and still give you a tone.. atleast mine does |
| 09:23.21 | DJ_Stefan0 | ah i see |
| 09:25.09 | creativx | go to asterisk console and try dial <ext> where <ext> is the hardphone |
| 09:25.18 | creativx | no wait |
| 09:25.22 | creativx | nevermind, thats useless |
| 09:25.33 | creativx | i would turn on sip debugging, pull the power cord on the hardphone and look |
| 09:27.28 | berktr | SIP vs H323 |
| 09:27.31 | berktr | what do you say? |
| 09:32.22 | JT | no-one uses H.323 in asterisk |
| 09:32.27 | JT | by comparison |
| 09:33.28 | *** join/#asterisk Mike_TK (n=Mike@nat0.yes.ko.if.ua) |
| 09:37.41 | berktr | i need to use h323 in asterisk |
| 09:37.43 | berktr | what should i do |
| 09:40.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:41.14 | Mike_TK | berktr Download asterisk-addons for your * Version and follow instructions inside |
| 09:43.56 | *** join/#asterisk booray (n=ray@64.70.85.100) |
| 09:47.46 | booray | I'm running svn-trunk-r54552 (about two weeks before 1.4.1) right now as it worked fabulously when I set it up.. are there any advantages to moving to 1.4.5 now? or, is there a summary of major changes in point releases, or should I read through the entire 300 page changelog? thanks for any info |
| 09:50.18 | booray | I suppose everyone's asleep |
| 09:50.26 | booray | it's 2:50am anyway |
| 09:50.30 | booray | wtf am I doing awake? |
| 09:51.17 | *** join/#asterisk Bananaskin (n=Banana@81-86-102-88.dsl.pipex.com) |
| 09:51.30 | Mike_TK | Yea... zzzzzzzzzzzz |
| 09:52.53 | booray | ha |
| 09:53.06 | booray | well I suppose it's only about half of the changelog |
| 09:54.00 | Mike_TK | You know... it's better don't touch if it works OK. |
| 09:56.54 | booray | True, however there are features that we want to add which may benefit from updating. |
| 09:57.03 | booray | I'll have to re-examine the list and compare it to the changelog |
| 09:57.04 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-165-37.columbus.res.rr.com) |
| 09:57.29 | booray | but I do agree with you Mike |
| 09:57.52 | Mike_TK | booray I don't think 1.4.1 - 1.4.5 changelog is so big... and you are gona to sleep, so you can read it to sleep better |
| 09:57.53 | *** part/#asterisk Bananaskin (n=Banana@81-86-102-88.dsl.pipex.com) |
| 09:58.07 | *** join/#asterisk mkl1525 (n=qwertz@pD9533214.dip0.t-ipconnect.de) |
| 09:59.44 | mkl1525 | Hi, trying to configure QoS for my ip phones. had a look with tcpdump and there were port combinations 51980/19198 and 63174/18162 but on my firewall the ports are 5004,5060,8000-8019,10000. so can anyone give me a hint which ports are used for sip/voip traffic? |
| 09:59.50 | booray | my scrollbar is about 45% the way down of my web browser window with plenty of room to scroll up. |
| 10:00.38 | booray | mkl1525: my understanding was that sip=5060, but it's hard to argue with tcpdump |
| 10:00.58 | Mike_TK | mkl1525: on asterisk what's port to use defined in file: rtp.conf. Each phone model is have own settings |
| 10:01.32 | pj_ | and some phones have a "use a random port" setting too |
| 10:02.07 | Mike_TK | mkl1525: so take a look at your phone config. You need to look for something like RTP Port range settings. |
| 10:02.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:02.44 | mkl1525 | Mike_TK, thanks will have a look |
| 10:05.06 | negativeduck | mkl, if you are doing QOS (not security) look in the source tree and read ip-tos.txt, a good way to setup qos is based on the TOS bit set in the IP packets for more refinded you can do limit to traffic destined to/from your asterisk box. |
| 10:06.38 | Mike_TK | in modern it's called DSCP |
| 10:07.28 | Mike_TK | ToS field is a part of DSCP value |
| 10:08.02 | Mike_TK | Here is how it can be done on Cisco switched / routers http://www.cisco.com/warp/public/105/dscpvalues.html |
| 10:08.34 | Mike_TK | but if you are talking about firewall than probably it's related to security |
| 10:10.00 | *** join/#asterisk mihinomenest (i=MMIV@cerebus.clandestineresearch.com) |
| 10:11.39 | negativeduck | eh, old habbits die hard and all that jazz. |
| 10:18.04 | e-ddie | i'm not into jazz, so that's not a problem |
| 10:24.28 | creativx | tos = pos |
| 10:28.37 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
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| 10:43.27 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 10:55.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 10:55.35 | ZaVoid | sup all |
| 10:56.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:59.59 | ZaVoid | anyone good with extension? |
| 11:08.40 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net) |
| 11:09.06 | Dovid | when using the H option in the dial command when I hit * where does the call go to ? |
| 11:10.10 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 11:21.01 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:21.37 | pigpen | Dovid, "Allow the caller to hang up by dialing *" |
| 11:21.43 | pigpen | http://www.voip-info.org/wiki-Asterisk+cmd+dial |
| 11:25.15 | *** join/#asterisk yassaccan (n=yassacca@admin224.hgo.se) |
| 11:25.36 | *** join/#asterisk jacq (n=jal@61.12.17.165) |
| 11:28.24 | pigpen | I need an opinion. |
| 11:28.51 | pigpen | My Father in Law may be taking a Job in India, and he wants to do some sort of voip. |
| 11:29.22 | pigpen | I have an asterisk setup and I am very familiar with the setup. But part of me wants to dump him over to skype. :) |
| 11:30.14 | awk | well that is your choice.. |
| 11:30.16 | pigpen | But, I will still have to support him. So. Since I have never used skype, nor have I used voip in India (connecting to a US server), I am needing some advice. |
| 11:30.38 | awk | how hard can it be, let him register a phone to your asterisk box |
| 11:30.42 | awk | and vwala ree calls |
| 11:30.47 | awk | s/ree/free |
| 11:30.57 | vn | ehm...wouldn't there me some kind of lag? |
| 11:31.02 | vn | india to us.. |
| 11:31.16 | vn | s/me/be |
| 11:31.21 | pigpen | That is was I was wondering...certianly skype isn't using some magical codec. |
| 11:31.40 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-136-74.fastres.net) |
| 11:31.44 | pigpen | But yeah..idefisk and a bluetooth headset and it is done. |
| 11:31.44 | awk | its all to do with compression |
| 11:31.55 | awk | and that isn;t even the problem its to do with the speed of your connection |
| 11:32.09 | pigpen | ie: in the US? |
| 11:32.15 | vn | and quality and number of hops |
| 11:32.30 | awk | hops still isn't such an issue, jitter is |
| 11:32.39 | awk | but I doubt any of this is a problem... |
| 11:32.44 | pigpen | Well, the US side (in San Antonio) is no issue. 400MB dual homed. |
| 11:32.57 | pigpen | but yeah...hops/latency. |
| 11:33.02 | awk | well i doubt either side will have a problem |
| 11:33.23 | pigpen | I have read a few postings about india, and all have been pretty good. |
| 11:33.37 | vn | depends, where is he going in india? |
| 11:34.01 | awk | still on dialup you could have a clear call |
| 11:34.13 | awk | 1 call registering to your server |
| 11:34.19 | pigpen | Well, for sure I don't know. But it must be pretty major, as he will be flying big jets. |
| 11:34.29 | vn | must be Bangalore |
| 11:34.32 | *** part/#asterisk h4mm3r` (n=h4mm3r@85-18-136-74.fastres.net) |
| 11:34.37 | pigpen | yeah...I think so. |
| 11:34.58 | pigpen | Shit, my evdo card works better than my business class cable modem sometimes. |
| 11:36.03 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 11:36.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:37.13 | Dovid | I think india has some tough laws against VOIP |
| 11:37.38 | Dovid | pigpen: when I call some one (I am the caller) and I press * the call does not hang up |
| 11:39.30 | pigpen | h allows the callee to hang up |
| 11:39.38 | pigpen | H allows the caller to hang up |
| 11:39.45 | pigpen | maybe you have it backwards. |
| 11:39.56 | pigpen | but for that matter, I have never found a use for it. |
| 11:40.10 | pigpen | In regards to India, yeah, I was just reading an article about it. |
| 11:40.33 | pigpen | I think it is mainly endpoint services they are concerned about. |
| 11:40.56 | Dovid | yup. i am a tad tired |
| 11:43.38 | lilalinux | Is fax possible with hfc cards and mISDN? Is everything integrated in the kernel, or do I need to compile custom modules (like zaphfc for bristuff)? |
| 11:48.18 | pigpen | I have found that faxing is very reliable using iaxmodem with hylafax. |
| 11:48.46 | DrAk0 | pigpen, asterisk 1.4 ? |
| 11:49.16 | pigpen | Actually, I haven't upgraded mine to 1.4 yet...but I need to get a few customers up to it. |
| 11:49.20 | pigpen | shouldn't be any different. |
| 11:49.37 | DrAk0 | pigpen, but i tried hylyfax with iaxmodem |
| 11:49.51 | DrAk0 | pigpen, it was very unrelieable i was using t.30 tho.. |
| 11:49.59 | pigpen | Asterisk just dumps the call off to an iax extension, which is iaxmodem, which is answered by hylafax. |
| 11:50.12 | e-ddie | start using mails instead |
| 11:50.40 | e-ddie | that fax crap is stoneage technology |
| 11:50.49 | pigpen | Well, I have two deployments. One supporting faxing for a faculty of 100 people. The other is for about 75. |
| 11:51.03 | pigpen | I haven't touched either one for about 9 months. |
| 11:51.19 | e-ddie | i wouldnt touch the fax one anytime |
| 11:51.30 | creativx | mail is crap too |
| 11:51.33 | pigpen | e-ddie, yeah..I hate faxing. |
| 11:51.34 | creativx | with all the spam these days |
| 11:51.38 | creativx | its almost worse than faxing |
| 11:52.19 | creativx | so fun when corporate mail filterers decide to block all .pdf attachments due to an exploit... ofcourse in the most silent of all manners, no explanations |
| 11:52.22 | e-ddie | creativx: true... |
| 11:53.04 | pigpen | kick the corporate IT people, that is stupid. |
| 11:53.47 | creativx | well |
| 11:55.10 | pigpen | I know that some have reported troubles with iaxmodem, mainly in setting it up. |
| 11:55.18 | pigpen | but mine has been working great. |
| 11:55.35 | pigpen | I will note: I kill iaxmodem and hylafax every AM. |
| 11:56.16 | pigpen | The first few weeks I would fine something wasn't quite right after about 5 days. Cycling it resolved the issue. |
| 11:56.25 | pigpen | Fluke? who knows. but it works. |
| 11:57.58 | creativx | humm |
| 11:58.12 | creativx | when adding a tos param to an iax peer, should a reload do the trick or do i need a stop now |
| 11:58.46 | pigpen | reload of iax should be fine, then reconnect. |
| 12:00.28 | creativx | reconnect ? reload does that automatically doesnt it |
| 12:00.49 | pigpen | that part was kinda of a "why not" |
| 12:00.51 | pigpen | try it. |
| 12:00.58 | pigpen | try it....without that is. |
| 12:01.07 | pigpen | if not, the drop the conn, then reconnect. |
| 12:01.21 | pigpen | I can't remember if it does it live. |
| 12:01.26 | creativx | ive never dropped an iax2 connection |
| 12:01.29 | creativx | so how would i do that |
| 12:01.29 | creativx | =) |
| 12:02.00 | pigpen | shit. good question. |
| 12:02.14 | pigpen | :) |
| 12:02.27 | creativx | yeah exactly |
| 12:02.35 | pigpen | Well dont' do it. |
| 12:02.39 | pigpen | haha |
| 12:02.39 | creativx | doesnt look like it.. tcpdumps shows tos 0x0 for all udp packets to the iax2 peer |
| 12:03.00 | pigpen | well, you may want to drop the connection to allow it to reconnect. |
| 12:03.01 | pigpen | :) |
| 12:03.24 | creativx | restarting when convenient.. |
| 12:03.24 | creativx | :) |
| 12:03.50 | pigpen | yeah...lets face it, if you are screwing with trunks, after hours is kind of an assumed. |
| 12:03.56 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:04.08 | creativx | i can torture our users during business hours |
| 12:04.08 | creativx | :) |
| 12:04.25 | Err | it should be noted that much/most of the internet ignores ToS bits |
| 12:04.42 | creativx | yea |
| 12:04.44 | Err | ...if your direct ISP cares, or you're using them internally, that's fine - but don't expect anyone upstream to give a whit :-P |
| 12:05.01 | creativx | i have 10mbit fiber to my isp and if i remember correctly they care about my tos bits |
| 12:05.04 | creativx | all the way to the nix |
| 12:05.14 | creativx | and there the gblx/alter net takes over |
| 12:05.15 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:05.57 | creativx | hmm restart didnt help either |
| 12:09.16 | *** join/#asterisk oej (n=olle@ti112220a340-2859.bb.online.no) |
| 12:11.23 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 12:16.44 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:19.31 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 12:19.39 | Corydon76-home | creativx: are you running Asterisk as root or as another user? |
| 12:27.27 | creativx | as root |
| 12:27.34 | creativx | all my root is belong to.. me |
| 12:28.07 | Corydon76-home | Okay, just checking one possible source of why setting tos might not work |
| 12:28.26 | Corydon76-home | Non-root users cannot currently set tos |
| 12:28.53 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 12:29.03 | creativx | yeah i see |
| 12:29.53 | awk | Corydon76-home that isn't entirely true.. |
| 12:30.36 | Corydon76-home | awk: for all of the bits you're likely to set for voip, it is |
| 12:30.50 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 12:31.50 | Corydon76-home | Nobody is likely to set mincost on their voip packets |
| 12:32.28 | Corydon76-home | That's just saying "if it's possible for the connection to screw up, make sure it screws up" |
| 12:32.29 | *** join/#asterisk hi365_m (i=HydraIRC@212.199.22.207.forward.012.net.il) |
| 12:32.51 | Corydon76-home | hmmm, actually, that doesn't sound half bad for testing purposes |
| 12:32.54 | creativx | i tried 0x18 |
| 12:33.11 | hi365_m | is there a celing number of static extensions that a queue will be able to function with? |
| 12:33.38 | awk | Corydon76-home heh, true.. |
| 12:33.54 | hi365_m | like after x amount of extensions it will start acting flaky? |
| 12:34.20 | Corydon76-home | Uh, how are queues and extensions even related? |
| 12:35.20 | Corydon76-home | I don't see any possible relationship between the number of extensions and the behavior of queues |
| 12:37.04 | Corydon76-home | hi365_m: what behavior are you seeing that is causing you to grasp at straws? |
| 12:38.06 | hi365_m | mmm, calls that come in from a q dissconect after a certain amount of time, also freezeing the telephone/softclient in the prosses |
| 12:40.46 | Corydon76-home | That sounds more like a queue timeout |
| 12:41.14 | hi365_m | that would limit the amount of time that a caller can talk to an agent? |
| 12:42.01 | creativx | no |
| 12:42.04 | creativx | only unanswered calls |
| 12:42.15 | creativx | if a call has been answered it can go on forever |
| 12:43.17 | mosty | is there a way to get asterisk to restart the B channels on a particular span? |
| 12:45.21 | *** join/#asterisk CVirus (n=GoD@213.212.224.7) |
| 12:49.31 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 12:55.17 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 12:55.56 | YonahW-Work | hey after every call through a pri with debug on i get a list of all the channels having successfully been restarted. Is this normal? |
| 12:56.33 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 12:57.47 | ZaVoid | dunno |
| 12:57.54 | ZaVoid | maybe :) |
| 12:59.46 | ZaVoid | anyone decent with extensions? |
| 13:00.11 | ZaVoid | exten => _0XX.,1,Dial(SIP/p3ptest/${EXTEN}&SIP/SIPp2ptest/${EXTEN}) |
| 13:00.12 | *** join/#asterisk saftsack (n=saftsack@pD9E07409.dip.t-dialin.net) |
| 13:00.14 | ZaVoid | this isn't working right |
| 13:01.10 | [TK]D-Fender | YonahW-Work: "resetinterval=never" |
| 13:01.20 | YonahW-Work | thanks |
| 13:01.46 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 13:01.48 | zdrulio | hellp |
| 13:01.58 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 13:02.00 | [TK]D-Fender | ZaVoid: the FORMAT is legal, your CONFIGURATION can be a completely dirrernt thing. |
| 13:02.00 | zdrulio | * hello ! |
| 13:02.08 | ZaVoid | hey fender |
| 13:02.11 | ZaVoid | just figured it out acutally |
| 13:02.17 | ZaVoid | i had SIPp2ptest |
| 13:02.21 | kombi_ | why might it be that ${ANSWEREDTIME} contains no value? |
| 13:02.30 | ZaVoid | needed to remove the SIP |
| 13:02.33 | ZaVoid | fat fingered |
| 13:02.41 | [TK]D-Fender | zdrulio: Usually the same thing.... |
| 13:03.07 | zdrulio | [TK]D-Fender: |
| 13:03.08 | zdrulio | ;:) |
| 13:03.58 | kombi_ | is ${ANSWEREDTIME} not active under any kind of circumstances? |
| 13:04.53 | kombi_ | like in IVR? |
| 13:05.50 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 13:07.15 | cheshair | hi guys! problems with my extensions.conf: s, i, t, ... predefined extensions don't work, any hint? |
| 13:07.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.38 | ZaVoid | heres a stupid question... |
| 13:07.47 | cheshair | i'm just told: "404 not found" |
| 13:07.54 | ZaVoid | in sip.conf.. can i use any characters in [somthing123] to name peers? |
| 13:08.12 | ZaVoid | like . and - |
| 13:08.20 | ZaVoid | or is ther ea list of legal characters somewhere? |
| 13:08.41 | [TK]D-Fender | cheshair: make sure your device is even pointed at the right CONTEXT |
| 13:09.00 | [TK]D-Fender | ZaVoid: DON'T get creative... |
| 13:09.05 | ZaVoid | lol |
| 13:09.35 | [TK]D-Fender | cheshair: And then when you're done banging your head against the wall, pastebin your SIP.CONF and EXTENSIONS.CONF |
| 13:09.38 | kombi_ | maybe I got this wrong, to get the total time a caller spend in i.e. an IVR, do you use ${ANSWEREDTIME} at all? |
| 13:10.18 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 13:10.19 | cheshair | [TK]D-Fender: i guess you remeber me 'cause of your ulcera... :-) |
| 13:11.06 | [TK]D-Fender | kombi_: NO |
| 13:11.16 | russellb | [TK]D-Fender: do you live here? |
| 13:11.33 | kombi_ | Fender: ok.. but what instead? |
| 13:11.35 | [TK]D-Fender | russellb: No, but I DIE a little each day.... |
| 13:11.46 | russellb | ha. |
| 13:12.13 | [TK]D-Fender | russellb: I may even find myself reaching ManxPower's level of BOFH-ness (he'd be so pround) |
| 13:12.22 | [TK]D-Fender | proud* |
| 13:13.00 | zdrulio | i`m looking for call manager. something like fonality but open source and free. any ideas ? |
| 13:13.15 | [TK]D-Fender | zdrulio: please define "call manager" |
| 13:13.20 | *** join/#asterisk geoaxis (n=geoaxis@unaffiliated/geoaxis) |
| 13:13.24 | geoaxis | hello people |
| 13:13.41 | kombi_ | Fender? |
| 13:14.20 | *** join/#asterisk oej (n=olle@ti112220a340-2859.bb.online.no) |
| 13:15.11 | cheshair | the point is: if i write "exten => 1,1,Playback(digits/1)" everything works: i digit 1 on my softphone and i get the right answer. |
| 13:15.18 | [TK]D-Fender | kombi_: ANSWEREDTIME = Total time - RINGING TIME. |
| 13:15.49 | cheshair | if i write "exten => s,1,Playback(digits/1)" i get "404" error no matter the number i dial |
| 13:17.54 | kombi_ | Fender: that's what I figured, just way does it not contain a value |
| 13:17.57 | [TK]D-Fender | cheshair: I have no proof you have that in the right ****CONTEXT**** at all, that your configure your PHONE right, or ANYTHING. |
| 13:18.01 | NovceGuru | Could somebody hook me up with a DID for testing purposes? |
| 13:18.04 | kombi_ | sorry way->why |
| 13:18.18 | [TK]D-Fender | cheshair: Your point is LOST. PASTEBIN all of those configs I mentioned. |
| 13:18.20 | [TK]D-Fender | ~pb |
| 13:18.21 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:18.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:18.35 | cheshair | [TK]D-Fender: right |
| 13:18.50 | [TK]D-Fender | kombi_: What tells you that this is an actual variable you can access? |
| 13:19.09 | [TK]D-Fender | NovceGuru: What's to test? You sue don't need a DID for anything... |
| 13:19.12 | [TK]D-Fender | sure* |
| 13:19.13 | *** join/#asterisk mazpe (n=email@66.184.205.67) |
| 13:19.34 | kombi_ | Fender: isn't that what variables are for..,) |
| 13:19.54 | [TK]D-Fender | kombi_: who says THAT is a VALID one MANAGED by *? |
| 13:19.59 | mocker | [TK]D-Fender: Are you ever not helping on #asterisk?? :) |
| 13:20.06 | NovceGuru | [TK]D-Fender, I dont know :P just wanted to see if I could get inbound call routing and stuff setup before I buy a provider (who is a good provider of a basic DID btw) |
| 13:20.17 | NovceGuru | yeah I always seem him here mocker :) |
| 13:20.25 | NovceGuru | GET A JOB! =P |
| 13:20.28 | [TK]D-Fender | mocker: Any time I'm at home or work I'm on. Sometimes idle a bit. |
| 13:20.33 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:20.37 | [TK]D-Fender | I *AM* at work :) |
| 13:20.42 | [TK]D-Fender | I multi-task :) |
| 13:20.51 | NovceGuru | Nice |
| 13:21.03 | kombi_ | Fender: hmm, the docs? |
| 13:21.04 | mocker | [TK]D-Fender: You have a blog or anything where you talk about asterisk stuff? |
| 13:21.05 | [TK]D-Fender | NovceGuru: You can. Get to it. |
| 13:21.18 | [TK]D-Fender | mocker: I have it on my to-do list. |
| 13:21.23 | mocker | [TK]D-Fender: hah. |
| 13:21.29 | mocker | I have those lists too. |
| 13:21.33 | NovceGuru | [TK]D-Fender, get a job? *confused* |
| 13:22.08 | [TK]D-Fender | NovceGuru>[TK]D-Fender, I dont know :P just wanted to see if I could get inbound call routing and stuff setup before I buy a provider (who is a good provider of a basic DID btw) <------[TK]D-Fender>NovceGuru: You can. Get to it. |
| 13:22.39 | [TK]D-Fender | NovceGuru: No need for a DID to test incoming call processing. to * ANY call is just a call, it depends on context & exten. |
| 13:22.53 | [TK]D-Fender | NovceGuru: You can set up your IVR and jsut dial it from your phone |
| 13:23.03 | [TK]D-Fender | NovceGuru: a call is a call is a call. |
| 13:23.33 | [TK]D-Fender | mocker: I'm planning on a stupidly basic WordPress install which shouldn't take that long but I've lacked motivation. |
| 13:23.49 | [TK]D-Fender | kombi_: Link me.... |
| 13:24.08 | *** join/#asterisk version5 (i=version5@nat/ibm/x-0a5e0cdf790b41b5) |
| 13:24.10 | mocker | [TK]D-Fender: I've forced myself to. |
| 13:24.12 | NovceGuru | I see (maybe) so if I have "local" phones calling each other it's about the same as setting up with a VOIP provider channel (sorry, i'm not up with the terms yet) |
| 13:24.16 | cheshair | should you happen to have a few minutes: http://pastebin.ca/591862 |
| 13:24.24 | mocker | [TK]D-Fender: Now I just collect links to other people's blogs w/ things that I'm interested in. |
| 13:24.49 | kombi_ | Fender: http://www.voip-info.org/wiki-Asterisk+variables |
| 13:24.54 | [TK]D-Fender | NovceGuru: I'm saying you can just call direct into your IVR. its an exten like any other. |
| 13:24.55 | cheshair | the problem is i can't make predefined ext work |
| 13:25.53 | NovceGuru | [TK]D-Fender, you mean calling 500 to test? |
| 13:26.38 | mocker | NovceGuru: 600,1,Playback(tt-monkeys) |
| 13:26.40 | mocker | Or something like that.. |
| 13:27.10 | mocker | Or setup two SIP softphones and have them call each other. |
| 13:27.16 | [TK]D-Fender | kombi_: That does not exist in 1.4 |
| 13:27.20 | NovceGuru | right, I have done that, just wanted to move onto the next step :D |
| 13:27.38 | mocker | NovceGuru: Umm. |
| 13:27.43 | kombi_ | Fender: 1.2 here |
| 13:27.57 | mocker | Try e164.org and get an enum number routed? |
| 13:27.57 | kombi_ | after painfully downgrading.. |
| 13:28.01 | NovceGuru | mocker, not that I have mastered that step, by any means |
| 13:28.07 | mocker | I think it's e164.org that gives those.. |
| 13:28.13 | [TK]D-Fender | cheshair: And how are you dialing "s" and why would you WANT TO? I think you are unaware of its PURPOSE.... |
| 13:29.07 | Dovid | TK: any way to have VAD enabled in asterisk ? |
| 13:29.29 | cheshair | [TK]D-Fender: i suppose "s" stands for "any number you dial"... am i wrong? |
| 13:29.48 | [TK]D-Fender | kombi_: I don't see it in 1.2 either... |
| 13:29.54 | [TK]D-Fender | cheshair: YES |
| 13:30.02 | [TK]D-Fender | cheshair: Back to thr drawing board for yOU! |
| 13:30.17 | cheshair | [TK]D-Fender: i see |
| 13:30.34 | [TK]D-Fender | cheshair: Go re-read everything on PATTERNS, and STANDARD EXTENSIONS. |
| 13:30.41 | version5 | hey guys, i want to use a phone (probably sip or some other voip protocol) to control a menu driven system on my pc. i.e when a 1 is pressed the server will recognise that and call some program on the system. |
| 13:30.58 | kombi_ | Fender: hmmm, what do you use then? |
| 13:30.59 | [TK]D-Fender | Dovid: You have GCC... get to work! |
| 13:31.02 | version5 | i assume i could script this in asterisk? |
| 13:31.25 | *** join/#asterisk flashnet (i=flashnet@gateway/tor/x-e780f4616bde13a7) |
| 13:31.27 | [TK]D-Fender | kombi_: this is Asterisk 101 stuff.... and you should not be using an "any number" match if at all possible. |
| 13:31.27 | cheshair | [TK]D-Fender: ok, see you veeery soon, thank you! |
| 13:31.51 | [TK]D-Fender | s/kombi_/cheshair / |
| 13:31.59 | purserj | hmm anyone heard of elastix? |
| 13:32.15 | Dovid | what does gcc have to do with it ? |
| 13:32.17 | NovceGuru | version5, tail -f the logs for <client> dialing a string? :\ thats my lame first though |
| 13:32.27 | [TK]D-Fender | version5: sure |
| 13:32.53 | NovceGuru | thought* |
| 13:33.18 | mocker | Woo, iaxmodem is going somewhat better than rxfax |
| 13:33.38 | [TK]D-Fender | mocker: I've got to get around to that too.. much higher on my list. |
| 13:33.55 | mocker | [TK]D-Fender: It doesn't seem to bad. |
| 13:34.19 | mocker | Took about half a day w/ pretty much zero hylafax experience. |
| 13:34.19 | [TK]D-Fender | mocker: RxFax has done nothing but crash on my since 1.2.7.1 |
| 13:34.27 | mocker | rxfax is pretty much crap. |
| 13:34.27 | cheshair | [TK]D-Fender: i don't understand what you mean when you talk about * 101 and "any number", however i'm sure i'll see when i'll learn more. just a last question: am i wrong too if i suppose that "i" extension will match any call to not existent numbers? |
| 13:34.28 | mocker | :( |
| 13:34.35 | [TK]D-Fender | mocker: I might use a couple of pointers later if you're around |
| 13:34.42 | mocker | sure thing. |
| 13:34.47 | [TK]D-Fender | cheshair: No. |
| 13:35.03 | [TK]D-Fender | cheshair: "i" is only used by IVR's. |
| 13:35.13 | Dovid | TK: I am lost what does gcc have to do with VAD ? Do i need to build a special package ? |
| 13:35.42 | [TK]D-Fender | cheshair: "s" is used by Macro's & IVR's typically, and as the incoming exten for analog channels or others that don't target a NUMBERED exten. |
| 13:35.59 | [TK]D-Fender | Dovid: translation, go code it YOURSELF ;) |
| 13:36.08 | Dovid | ah ok |
| 13:36.14 | *** join/#asterisk CVirus (n=GoD@213.212.224.7) |
| 13:36.25 | Dovid | so asterisk dosent have it :) |
| 13:37.18 | kombi_ | here's an easy one: how do I write a punched in key value into a variable? |
| 13:37.50 | mocker | Anyone know if the dCAP cert is up to 1.4 yet? |
| 13:38.04 | [TK]D-Fender | Dovid: DUH <- |
| 13:38.06 | cheshair | [TK]D-Fender: hm that doesn't sound too much easy to me... anyway i'll have more tries and read the manual, tahnk you! see you soon |
| 13:38.28 | [TK]D-Fender | kombi_: What do you think is already HOLDING this value? :) |
| 13:38.51 | [TK]D-Fender | cheshair: its all in the... BOOK |
| 13:38.53 | [TK]D-Fender | ~book |
| 13:38.54 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:39.18 | [TK]D-Fender | cheshair: if you can't figure out these few tiny things, Asterisk is the LEAST of your worries. |
| 13:39.55 | kombi_ | Fender: well, variables I suppose, but what are they? |
| 13:40.35 | [TK]D-Fender | kombi_: You need to think yhe entire process through as to what you want to do. Your questions are coming out as tiny broken fragment with no coherence. |
| 13:41.43 | *** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) |
| 13:41.48 | kombi_ | Fender: IVR that records key strokes |
| 13:41.53 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:41.53 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 13:41.58 | tamp4x | pbx_find_extension: Maximum PBX stack exceeded .... anyone know how to get around this error? |
| 13:42.18 | tamp4x | it seems contexts and what not are not beign added to the dial plan |
| 13:42.19 | [TK]D-Fender | kombi_: "show application read" <----------- |
| 13:42.30 | cheshair | [TK]D-Fender: i ran into such problem while reading ",ch05.20886 Page 85 Wednesday, August 31, 2005 4:56 PM" of that book. i guess it's MY fault, that's sure... however i just hoped some experienced user could hint me the right way... it's ok |
| 13:43.20 | kombi_ | thanks! |
| 13:44.19 | DarylVOIP | Anyone know how I can run a PHP agi script in the dialplan and continue on with the dialplan without waiting for output form the script (fork it)? |
| 13:44.41 | DarylVOIP | I only see one sketch example on voip-info about perl, and I don't really understand the fundamentals behind it. |
| 13:44.49 | [TK]D-Fender | cheshair: "exten => _X.,1," will capture any number dialed that is at least 2 digits or longer. |
| 13:44.51 | DarylVOIP | (otherwise I'd be able to figure it out in PHP) |
| 13:45.14 | [TK]D-Fender | DarylVOIP: try asing in #per l. This is #asterisk |
| 13:45.34 | DarylVOIP | I know it's #asterisk. |
| 13:45.41 | DarylVOIP | I'm asking how to fork the diaplan. |
| 13:45.44 | DarylVOIP | with a PHP agi. |
| 13:46.02 | DarylVOIP | The ASTERISK dialplan. |
| 13:46.19 | [TK]D-Fender | DarylVOIP: Oh NOW you tell us that it has something to do with *! |
| 13:46.20 | cheshair | [TK]D-Fender: that's good! thank you! anyway i think the point is i have to understand what "s" and "i" really mean and when i am suppose to use them. back to the manual |
| 13:46.37 | [TK]D-Fender | DarylVOIP: Please clarify "fork the dialplan" |
| 13:46.53 | DarylVOIP | DarylVOIP |
| 13:46.53 | DarylVOIP | : |
| 13:46.53 | DarylVOIP | Anyone know how I can run a PHP agi script in the dialplan and continue on with the dialplan without waiting for output form the script (fork it)? |
| 13:46.57 | [TK]D-Fender | cheshair: I won't keep you from it. |
| 13:47.08 | DarylVOIP | I'm not sure how that isn't clear. Let me know what else you need me to carify. |
| 13:47.09 | mocker | DarylVOIP: You can just call AGI scripts w/ the agi() application. |
| 13:47.12 | [TK]D-Fender | DarylVOIP: NO <- |
| 13:47.37 | [TK]D-Fender | DarylVOIP: You cannot. Calls are not THREADED |
| 13:47.39 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 13:47.47 | [TK]D-Fender | DarylVOIP: Painfully LINEAR |
| 13:48.06 | mocker | Hmm. |
| 13:48.16 | DarylVOIP | I'm pastebinning my dialplan....that will make this easier to explain. |
| 13:48.17 | mocker | What if the script exited, but forked off another process? |
| 13:48.35 | DarylVOIP | Ugh..pastebin is slow today. |
| 13:48.38 | [TK]D-Fender | DarylVOIP: I KNOW what you want and I told your stright up that you cannot. |
| 13:49.03 | DarylVOIP | So the voip-info wiki is wrong? |
| 13:49.05 | DarylVOIP | "If you don't want Asterisk to wait until the script finishes you can fork the script off to return to dialplan excution: Here's how to accomplish this in PERL: |
| 13:49.06 | DarylVOIP | <PROTECTED> |
| 13:49.06 | DarylVOIP | <PROTECTED> |
| 13:49.17 | DarylVOIP | I'm just trying to figure out how it work for PHP AGIs. |
| 13:49.41 | [TK]D-Fender | DarylVOIP: What are you expecting your AGI to do afterwards? |
| 13:49.48 | *** join/#asterisk bintut (n=bintut@cm63.gamma177.maxonline.com.sg) |
| 13:49.50 | Err | well of course you could fork the script from within itself, as long as you return immediately in the other process - that would leave a process spun off, with no communication with asterisk, though |
| 13:50.09 | [TK]D-Fender | err : yes, that is what I'm implying. |
| 13:50.21 | DarylVOIP | It's a call-back. Basically the AGI is making a callback based on the ANI of the incoming call. The dialplan continues to play a progress tone and then dump the initial call. |
| 13:50.39 | DarylVOIP | I want the AGI to fire before the call is dumped because it has to do DB looups for authentication. |
| 13:50.47 | [TK]D-Fender | DarylVOIP: Not really the way to do that in the first place... |
| 13:50.47 | Err | yes, I see exactly what you're saying - there's nothing (asterisk-wise) that you can do *usefully* with such a setup - but if all you want to do is spawn a process when you hit a certain place in the dialplan, you can |
| 13:50.58 | DarylVOIP | I'm timing it so that it won't actually place the call sooner than x seconds before the intial call is received. |
| 13:51.01 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:51.04 | Err | if you felt the need to build some giant IPC interaction gob of crap external to asterisk, you could |
| 13:51.13 | shido6 | I do that in the diaplan and a call file. then when I call in I use disa for auth but im sure u can use a db for auth |
| 13:51.18 | DarylVOIP | OK....so what's the right way? Maybe I'm looking at this from the wrong angle. |
| 13:51.23 | [TK]D-Fender | DarylVOIP: Use "h" to catch the disconnect, and have that fire off a .call" file. |
| 13:51.24 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 13:51.40 | [TK]D-Fender | DarylVOIP: No need for AGI even if you think about it right. |
| 13:51.42 | DarylVOIP | I suppose I could move it there. |
| 13:52.00 | DarylVOIP | There is a need for an AGI for other reasons. |
| 13:52.16 | [TK]D-Fender | DarylVOIP: like? |
| 13:52.26 | HarryR | [TK]D-Fender: how about because the asterisk dialplan is a nightmare :) |
| 13:52.34 | DarylVOIP | (I'm looking up the ANI, verifying if the account associated with it is enabled for callback, verifying the credit in the account, etc.......non * specific stuff) |
| 13:52.44 | DarylVOIP | Back-end billing stuff basically. |
| 13:52.44 | [TK]D-Fender | HarryR: On your fruitless explanations count as such ;) |
| 13:52.51 | *** join/#asterisk rogerz (n=highvolt@rrcs-24-213-232-12.nys.biz.rr.com) |
| 13:53.01 | [TK]D-Fender | DarylVOIP: Ok, yeah, fine THAT then :) |
| 13:53.02 | *** join/#asterisk wglenncamp (n=wglennca@cblmdm72-240-183-54.buckeyecom.net) |
| 13:53.04 | DarylVOIP | lol |
| 13:53.18 | [TK]D-Fender | DarylVOIP: Do it in a dead-agi on "h" making sure THEY hang up. |
| 13:53.56 | DarylVOIP | THat makes sense....and much easier than what I was getting myself into. Thanks.....I'll give that a shot now. |
| 13:54.23 | DarylVOIP | Oh....but the only problem with that is that I'm never answering the call. Hopefullt it will still work. |
| 13:54.39 | DarylVOIP | I send busy as soon as I get the ANI. |
| 13:54.56 | *** part/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 13:55.02 | DarylVOIP | (so the caller doesn't get billed, which would severely reduce the point of a callback) |
| 13:56.33 | *** join/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 13:58.11 | cheshair | guys, does anybody in here know what's wrong with: http://pastebin.ca/591905 i'm expecting to be told "one" TWO times. on the contrary the first line (the one with the "s") seems not to work at all |
| 13:58.26 | [TK]D-Fender | DarylVOIP: you need for them to hang up to tinitiate, so just look a Wait |
| 13:58.57 | [TK]D-Fender | cheshair: You. Do. NOT. Dial. "s"! |
| 13:59.32 | DarylVOIP | [TK]D-Fender: Looks fine with a DeadAGI on h - seems to work. Thanks! Now I have to get all of my DB lookups done. |
| 14:00.08 | cheshair | [TK]D-Fender: i see, but i read similar exmples on the guide, and those should work |
| 14:00.22 | kombi_ | where do you set the path for prompts? |
| 14:01.06 | cheshair | [TK]D-Fender: if you want i can paste the code i read on the guide |
| 14:01.29 | [TK]D-Fender | cheshair: No. |
| 14:01.41 | mocker | woo. |
| 14:02.17 | [TK]D-Fender | cheshair: FORGET about "s". You dhave failed to understand it. it is NOT a catch-all, and you will NEVER execut BOTH in a context. only ONE thing can be executed on a MATCH. |
| 14:02.25 | *** join/#asterisk mazpe (n=email@68.152.128.30) |
| 14:02.32 | [TK]D-Fender | mocker: Stacking up IAXModems? |
| 14:02.50 | mocker | Yeah. |
| 14:03.31 | cheshair | [TK]D-Fender: are you able to help me to understand what the guide says at page 85, without getting angry? |
| 14:03.33 | mocker | Don't want to get busy signals. |
| 14:03.39 | [TK]D-Fender | cheshair: You can't be "A" and "B" at the same time, so "1" and "s" will never be called simultaneously. |
| 14:04.04 | cheshair | [TK]D-Fender: i see |
| 14:04.05 | [TK]D-Fender | mocker: One client of mine is running 3. I'm not sure what the limit is on these. |
| 14:04.10 | ManxPower | Extension "s" is matched when the call comes into Asterisk with no destination extensions. Examples of this is if a call comes in on an FXO port, or from a VoIP carrier that does not send the destination number. IAXTel is like this. "s" does not stand for "start". "s" stands for "stupid". |
| 14:04.18 | mocker | [TK]D-Fender: heh, I just setup six.. |
| 14:04.22 | mocker | So.. we'll see. |
| 14:04.26 | cheshair | [TK]D-Fender: still i can't see why the guide does st similar |
| 14:04.40 | mocker | I think I'm going to use the commandline 'sendfax' app to do some brute force load testing. |
| 14:04.42 | cheshair | [TK]D-Fender: (similar to my eyes) |
| 14:04.50 | mocker | send a fax every minute for an hour or something. |
| 14:05.15 | [TK]D-Fender | cheshair: they are not. they may DO the same thing, but both will not trigger at the same time. |
| 14:05.19 | CVirus | The SPA400 features the ability to connect up to four (4) standard analog telephones lines to a Linksys Voice System (LVS) VoIP network |
| 14:05.20 | coppice | why not send 50 faxes at once? |
| 14:05.29 | CVirus | what's a LVS ? |
| 14:05.30 | CVirus | http://www.telephonyware.com/telephonyware/tw00358.html |
| 14:05.33 | mocker | coppice: I only have 6 modems.. |
| 14:05.49 | coppice | well that's not much of a test :-) |
| 14:05.51 | mocker | And I want to make sure they hang up and clear the line. |
| 14:05.57 | [TK]D-Fender | CVirus: "inksys Voice System" <- do you really like answering your own questions even as you ASK them?! |
| 14:06.14 | CVirus | hehe |
| 14:06.31 | CVirus | [TK]D-Fender: I read the rest of the description and i found the feature i want .. thanks anyways |
| 14:06.34 | [TK]D-Fender | CVirus: They make their own PBX, and thats what it is designed for. |
| 14:06.46 | rue_mohr | heh, kphone is registered wth asterisk, and I dont know how to make a call with it.... |
| 14:06.57 | [TK]D-Fender | CVirus: It can also be used with *, but you can't pick a SPECIFIC channel to dial out on. |
| 14:07.03 | cheshair | [TK]D-Fender: so what's the point of using all those "s"s, "i"s and "t"s? (as i see in the guide itself) |
| 14:07.09 | [TK]D-Fender | CVirus: It treats them all as a pool only. |
| 14:07.24 | [TK]D-Fender | cheshair: Stop now, keep reading, and learn how to make an actual IVR. |
| 14:07.54 | CVirus | [TK]D-Fender: please clarify that last point |
| 14:08.21 | [TK]D-Fender | cheshair: these are not things you DIAL. Once you dial with your phone you may end up on "s" to start your IVR, and these other extens come into play. they are NOT for when you FIRST dial in, they are for AFTER * answers the call to present you a menu |
| 14:08.46 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:08.51 | [TK]D-Fender | CVirus: With a zaptel card you can tell it to use Port 3 for a call. with the SPA400 you can only tell it to DIAL, not to use a specific line. |
| 14:09.01 | [TK]D-Fender | CVirus: It can ONLY treat them as being all the same. |
| 14:09.37 | cheshair | [TK]D-Fender: that's coming clearer... i'll think about it, thanks |
| 14:10.32 | cheshair | [TK] |
| 14:10.32 | cheshair | ooops |
| 14:11.31 | *** join/#asterisk anthm (n=anthm@dhcp64-134-34-214.bwic.chi.wayport.net) |
| 14:11.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.06 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 14:14.15 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 14:14.19 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 14:14.46 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 14:15.55 | [TK]D-Fender | Mercestes: it was. |
| 14:16.16 | [TK]D-Fender | Mercestes: I was still holding on when I reg'd here years ago and never dropped it. |
| 14:16.27 | lirakis | When I have a context that is an IVR or some thing.. i get wierd destinations in my CDR's .. like s, and 1, 2,3 .. whatever digit they press. What do i set to make the cdr's show what I want in dst? I have tried set(EXTEN=whatiwant) and set(CDR(dst)=whatiwant) but niether seem to work |
| 14:16.38 | Mercestes | oh, can I join your clan? |
| 14:16.46 | Err | wow, I can't imagine why linksys would build such a limited device, since individually-addressable ports would've been no more trouble |
| 14:17.20 | mazpe | zaptel drivers is what i need to install my Sangoma A101D T1 PRI card? |
| 14:18.14 | [TK]D-Fender | Mercestes: l0lxors n00b! |
| 14:18.23 | mocker | mazpe: Both zaptel and sangoma drivers probably. |
| 14:18.26 | [TK]D-Fender | mazpe: libpri & zaptel |
| 14:18.34 | [TK]D-Fender | mazpe: and of course... Wanpipe |
| 14:18.58 | Mercestes | speaking of n00b. anyone know how to set a static env variable for root? |
| 14:19.12 | coppice | shouldn't they change it to Wantube, to be more up to date? |
| 14:19.27 | mocker | heh |
| 14:19.45 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:20.21 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:20.27 | mazpe | [TK]D-Fender: thanks |
| 14:21.39 | CVirus | [TK]D-Fender: I'm an asterisk n00b ... anyways .. suppose I got 4 phone lines connected to the SPA400 and it is connected to a network containing an asterisk server and 4 ATA's connected to 4 analog phones .... Now ... someone calls one of the lines connected to the SPA .. can I configure asterisk to connect the call to ATA no. 1 while if a phone comes to line no. 2, asterisk connects it to ATA no. 2 ... is this possible ? |
| 14:21.49 | *** join/#asterisk Zaggynl^Laptop (i=az@145.89.181.85) |
| 14:22.06 | CVirus | while if a phone call comes to |
| 14:22.28 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-245.bwic.chi.wayport.net) |
| 14:22.30 | [TK]D-Fender | CVirus: it sends calls to *, and calls going OUT will just pick the first free line. You can't tell it to use a SPECIFIC one. |
| 14:22.35 | [TK]D-Fender | CVirus: That is ALL. |
| 14:23.09 | CVirus | [TK]D-Fender: there are no calls going out .. thanks for helping :-) |
| 14:24.08 | *** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 14:26.30 | *** join/#asterisk saftsack (n=saftsack@pD9E07409.dip.t-dialin.net) |
| 14:27.31 | *** join/#asterisk sosoriosv (n=salvador@200.31.160.4) |
| 14:27.49 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 14:29.00 | cheshair | hi guys, im having a bloody fight against "s", "t" and other predefined extensions in my extensions.conf. i'm having some attempts while calling myself. maybe "s" and friends do not work 'cause i should make calls from another channel? |
| 14:29.48 | mosty | cheshair, try describing the problem |
| 14:29.53 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 14:30.08 | cheshair | hi mosty, here comes my problem |
| 14:30.10 | littleball | hi does cisco as5300 SIP supporting good or not? |
| 14:30.23 | littleball | can i use SIP to connect asterisk to as5300? |
| 14:30.23 | sosoriosv | hello, i need a good manual to begin in asterisk |
| 14:30.28 | littleball | within the same subnetwork |
| 14:30.52 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-94a8099b3799015c) |
| 14:30.56 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:31.32 | mocker | Ok, cron job in place. |
| 14:31.35 | mosty | ~book |
| 14:31.36 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:31.39 | mocker | Sending 2 faxes every 3 minutes. |
| 14:32.36 | sosoriosv | thank jbot i am checking |
| 14:33.15 | mosty | sosoriosv, it's a bit old but that's about all i know of that is a reasonable intro |
| 14:33.16 | [TK]D-Fender | mocker: I'd suggest getting an actual fax board for outgoing calls off Hylafax. I do trust you're using it to its full potential... |
| 14:34.00 | sosoriosv | thank mosty |
| 14:34.05 | mocker | [TK]D-Fender: Actually, I'm not planning on doing many outgoing.. |
| 14:34.14 | mocker | Just using sendfax for load testing. |
| 14:34.16 | Mike_TK | Hi. Can anyone say me why asterisk uses so much file descriptors? It's growing exponentially: 10 SIP 2 SIP calls = ~2000FDs, 20 = 5000, 30 = 10k, 50 =~ 25k, 100=~90k, 300 =~ 750k, 500 =~ 3300k |
| 14:34.18 | *** part/#asterisk version5 (i=version5@nat/ibm/x-0a5e0cdf790b41b5) |
| 14:34.34 | mocker | i.e. I want to make sure this is actually stable. :) |
| 14:34.46 | Mercestes | hylafax or asterisk? |
| 14:34.50 | Mercestes | or sendfax? |
| 14:36.08 | coppice | or a place where horses live? |
| 14:36.41 | Mercestes | lmao |
| 14:36.50 | Mercestes | That must be it. |
| 14:37.51 | cheshair | mosty, the problem is i can't understand what "s", "t" and friends work. i had some attempts with http://pastebin.ca/591957, which i took from the *'s guide. |
| 14:38.52 | mocker | Mercestes: yes, all of the above. |
| 14:38.56 | mosty | cheshair, asterisk puts calls in the s extension of a particular context when it doesn't know what number was dialed, eg a call from an analogue phone line. calls go the the t extension if there is a timeout |
| 14:39.12 | cheshair | mosty, i expected to receive "enter-ext-of-person" greeting as soon as i answer the phone (to myself) |
| 14:39.20 | cheshair | mosty, i see |
| 14:39.34 | mosty | cheshair, where is the call coming from? |
| 14:39.50 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 14:40.00 | Mercestes | mocker, I would say that from experience that hylafax is very stable. |
| 14:40.06 | cheshair | mosty, so i was right when i said "s, t and friends deal with analogue phone calls" |
| 14:40.35 | cheshair | mosty, the point is i'm having some attempts just calling myself just on the same computer |
| 14:40.43 | Uatec | hey, where can i get alternative voicepacks for the voicemail ? |
| 14:40.52 | Uatec | i found some alternative voicepacks, but they didn't do the voicemail |
| 14:40.56 | cheshair | mosty, i use a two lines softphone and i call myself |
| 14:41.09 | cheshair | mosty, i guess that's my problem! |
| 14:41.10 | mosty | cheshair, t doesn't have to have anything to do with analogue lines, neither does s. |
| 14:41.11 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:41.16 | mazpe | anyone knows of a good tutorial that talks about Zaptel/PRI installation and usage? |
| 14:41.21 | mosty | cheshair, what number are you dialing to begin the call? |
| 14:41.25 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 14:41.40 | Mercestes | mazpe: http://www.google.com |
| 14:41.50 | Mercestes | mazpe: Try entering in keywords like asterisk zaptel and asterisk pri |
| 14:41.57 | mazpe | ;) |
| 14:42.13 | cheshair | mosty, 101 |
| 14:42.20 | mocker | Mercestes: Yeah, I'm actually more concerned if iaxmodem is stable. |
| 14:42.27 | mosty | cheshair, then use 101 instead of s in your dialplan |
| 14:42.28 | mocker | I know that hylafax has been around forever. |
| 14:42.45 | Mike_TK | mocker: I have an installation that sends 96 concurent outbound faxes, that delivers ~ 100 000 pages per day |
| 14:42.58 | mosty | cheshair, since asterisk knows you dialed 101, it puts the call in the 101 extension (not the s extension) |
| 14:43.00 | CVirus | A network consisting of asterisk server + 4 ATA's connected to 4 analog phones ... Can I configure asterisk to forward an incoming call to the first free analog phone ? |
| 14:43.14 | Mike_TK | mocker: It working stable enough |
| 14:43.15 | cheshair | mosty, i see, anyway when am i supposed to follow the guide and really use the s? |
| 14:43.44 | mocker | Mike_TK: Are you a member of TK's clan? |
| 14:43.51 | mocker | :) |
| 14:43.53 | Mike_TK | mocker: no |
| 14:44.06 | Mercestes | mocker: I didn't have many problems with iaxmodem but I don't think it's beign developed right now. |
| 14:44.06 | mocker | Mike_TK: You're using iaxmodem? |
| 14:44.12 | berktr | how can i kill a sip channel? |
| 14:44.22 | Nugget | berktr: shutdown -h now :) |
| 14:44.26 | berktr | let's say i want to kill 688e3c-c0a8 channel |
| 14:44.28 | berktr | :D |
| 14:44.44 | Mike_TK | mocker: no - t38modem connected to cisco 5400 over SIP |
| 14:44.47 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:44.49 | mocker | Mike_TK: Ahh. |
| 14:44.56 | Mike_TK | sry |
| 14:45.00 | Mike_TK | mocker: over h323 |
| 14:45.08 | mosty | cheshair, use the s extension for calls coming from an analogue phone line (or any source that doesn't tell asterisk what number was dialed), or but doing a Goto that jumps to extension s somewhere |
| 14:45.09 | mocker | Mike_TK: Yeah, that sounds like a decent fax solution. |
| 14:45.17 | Uatec | no body have any ideas? |
| 14:45.40 | cheshair | mosty, ok, now i see |
| 14:45.55 | cheshair | mosty, maaany thanks! |
| 14:46.03 | berktr | any answers for me :( ? |
| 14:46.11 | CVirus | any answers for me too ? |
| 14:46.38 | Mike_TK | berktr: soft hangup, but it takes an asterisk call ID, not SIP call ID |
| 14:46.50 | berktr | how can i learn a call id? |
| 14:47.28 | berktr | soft hangup 688e3c-c0a8 => 688e3c-c0a8 is not a known channel |
| 14:47.45 | berktr | however when i type sip show channels |
| 14:47.50 | berktr | i see that channel there, with ACK status |
| 14:48.00 | cheshair | see u later guys and maaany thanks to you all!! |
| 14:48.03 | *** part/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 14:48.47 | Mike_TK | soft hangup and press TAB after it. it will show you a list of channels |
| 14:49.17 | Mike_TK | berktr: or use 'show channels' |
| 14:49.33 | berktr | asterisk*CLI> soft hangup SIP/3003-0877e000 |
| 14:49.34 | berktr | Requested Hangup on channel 'SIP/3003-0877e000' |
| 14:49.42 | berktr | well, how can i force it to hangup |
| 14:49.45 | berktr | still active |
| 14:49.54 | CVirus | A network consisting of asterisk server + 4 ATA's connected to 4 analog phones ... Can I configure asterisk to forward an incoming call to the first free analog phone ? |
| 14:50.54 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:52.10 | mosty | yes |
| 14:52.20 | mosty | setup a call queue |
| 14:53.00 | *** part/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 14:53.13 | mosty | or limit the number of simultaneous calls to each sip account to 1, then just dial them in order and it will fall through to the first one that isn't busy |
| 14:56.07 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:57.01 | *** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 14:57.27 | littleball | hello , from the asterisk log file, i found "No D-channels available! Using Primary channel 78 as D-channel anyway!". what does it mean? |
| 14:57.41 | [TK]D-Fender | mocker: Don't need a lot.or ChanIsAvail them sequentially and dial. |
| 14:57.42 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 14:57.59 | mocker | [TK]D-Fender: Does not compute. |
| 14:58.36 | mocker | [TK]D-Fender: I have a macro that sequentially dials the modems. |
| 14:58.41 | mocker | If that's what you're asking. |
| 14:58.51 | mosty | littleball, does "pri show span <number>" say that the span is up? |
| 14:59.28 | [TK]D-Fender | mocker: You always think I'm talking exclusinvely with YOU... thats the problem ;) |
| 14:59.33 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 14:59.57 | mocker | Ahh, suprious tab completion. |
| 15:00.00 | [TK]D-Fender | mocker: Actually... that WAS your nick |
| 15:00.04 | [TK]D-Fender | Stupid auto-complete! |
| 15:00.29 | littleball | mosty, Status: Provisioned, Down, Active |
| 15:00.40 | littleball | what does this mean? |
| 15:00.44 | Uatec | hey, where can i get alternative voicepacks for the voicemail ? |
| 15:00.45 | Uatec | i found some alternative voicepacks, but they didn't do the voicemail |
| 15:01.33 | mosty | littleball, status down probably means that asterisk isn't seeing data from the PRI line. are you using a sangoma card or digium? |
| 15:02.05 | littleball | digium |
| 15:02.19 | littleball | mosty, but it works. i can make calls |
| 15:02.40 | mosty | if you can make and receive calls over that line then ignore it |
| 15:03.00 | littleball | mosty, but the problem is that some times, e1 hang and need to reset |
| 15:03.11 | [TK]D-Fender | littleball: Means you've got synch, but no d-chan active |
| 15:03.13 | *** join/#asterisk rantsh (n=rsmith@201.210.16.238) |
| 15:03.22 | rantsh | hello people |
| 15:03.43 | [TK]D-Fender | littleball: pastebin your configs, and tells us your card model. |
| 15:04.12 | *** join/#asterisk alteregoz (n=evang@mail2.johnstoncom.com) |
| 15:04.15 | littleball | [TK]D-Fender, paste to where? |
| 15:04.17 | littleball | ok |
| 15:04.23 | rantsh | anyone knows where I can get a tutorial to apply an asterisk patch |
| 15:04.40 | mocker | littleball: Is it for sure a PRI? |
| 15:04.57 | littleball | where to paste? pastebin.org? |
| 15:05.06 | littleball | wrong site, i think |
| 15:05.21 | *** join/#asterisk Scrumps (n=scrumpy@smurfnet.xs4all.nl) |
| 15:05.22 | javar | pastebin.ca |
| 15:05.22 | [TK]D-Fender | ~pb |
| 15:05.22 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:05.29 | sergee | seems like my GXV3000 died.. :( are there any way to restore it? boot from usb flash? |
| 15:05.35 | rantsh | I'm using patch -p1 asterisk-1.4.5-patch but it remains idle and does nothing |
| 15:05.38 | [TK]D-Fender | littleball: You've been here long enough, you should know better |
| 15:06.02 | [TK]D-Fender | sergee: Go ask Lazerus :) |
| 15:06.09 | [TK]D-Fender | ~gs |
| 15:06.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:06.13 | littleball | :-) i know. pastebin.com. wait just slow |
| 15:06.23 | [TK]D-Fender | littleball: .ca <- |
| 15:06.36 | [TK]D-Fender | littleball: Where do you see anything refer to the .com one?! |
| 15:06.45 | sergee | [TK]D-Fender: i didn't see any other SIP videophones, which works... |
| 15:07.00 | [TK]D-Fender | sergee: Welcome to the world of "tunnel vision" |
| 15:07.29 | [TK]D-Fender | sergee: ACN sells one, Tornado's M20, plenty of others. You clearly aren't trying very hard. |
| 15:07.43 | [TK]D-Fender | NEXT!@!!@!@!@ (c) BKW |
| 15:09.12 | *** part/#asterisk alteregoz (n=evang@mail2.johnstoncom.com) |
| 15:09.19 | lirakis | arg.. i cant get a new DID to route properly |
| 15:09.26 | littleball | ttp://paste.lisp.org/display/43537 |
| 15:09.34 | littleball | [TK]D-Fender, ttp://paste.lisp.org/display/43537. |
| 15:10.12 | littleball | mocker, yes, pri |
| 15:10.17 | [TK]D-Fender | littleball: and in repeating STILL lacked 1 char... |
| 15:10.24 | [TK]D-Fender | littleball: Zaptel please... |
| 15:10.51 | littleball | STILL? |
| 15:11.32 | penguinFunk | making your links unclickable! |
| 15:11.33 | littleball | http://paste.lisp.org/display/43537#1 |
| 15:11.45 | *** part/#asterisk sosoriosv (n=salvador@200.31.160.4) |
| 15:11.55 | rantsh | any help with patches? |
| 15:12.10 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 15:12.18 | littleball | [TK]D-Fender, i have 4 E1 |
| 15:12.25 | [TK]D-Fender | littleball: ....... |
| 15:12.32 | file | rantsh: patch -p1 < asterisk-1.4.5.patch |
| 15:12.34 | file | notice the < |
| 15:13.02 | rantsh | I get an error there |
| 15:13.26 | *** join/#asterisk ccesario (n=ccesario@ns1.unialco.com.br) |
| 15:13.28 | rantsh | "can't find file to patch at input line 5" |
| 15:13.41 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:14.06 | littleball | http://paste.lisp.org/display/43537#3 |
| 15:15.04 | mosty | on a PRI channel, does the telco end just start sending audio as soon as the call is accepted but before it is answered? ie is there any signal sent by the telco when the remote end answers? |
| 15:15.18 | littleball | [TK]D-Fender, any idea? |
| 15:15.30 | mosty | rantsh, make sure you're in the correct directory when running that command |
| 15:15.37 | [TK]D-Fender | littleball: ZAPTEL.CONF please..... |
| 15:16.04 | mosty | littleball, which distribution are you using? |
| 15:16.08 | brettnem | !seen mitcheloc |
| 15:16.15 | brettnem | ~seen mitcheloc |
| 15:16.47 | jbot | mitcheloc <n=mitchelo@titaniumsoft.net> was last seen on IRC in channel #asterisk, 19d 10h 50m 51s ago, saying: 'jcmoore: thanks, testing now'. |
| 15:16.47 | littleball | http://paste.lisp.org/display/43537#5 |
| 15:16.47 | rantsh | mosty: ok, I'm standing in the source code directory |
| 15:16.47 | littleball | [TK]D-Fender, any idea? http://paste.lisp.org/display/43537#5 |
| 15:16.53 | rantsh | mosty: and this is where I saved the patch, was I supposed to save it somewhere else? |
| 15:17.26 | mosty | rantsh, depends how the patch was created. look for docs where the patch came from |
| 15:17.37 | berktr | how to reduce the echo on the conversation |
| 15:17.45 | berktr | i mean how can i get rid of the echo |
| 15:17.46 | mosty | rantsh, also try without -p1 |
| 15:18.24 | penguinFunk | berktr: what hardware? digital line or analogue? |
| 15:18.28 | [TK]D-Fender | littleball: What do you have on each port? |
| 15:18.49 | lirakis | is there any way to see what DID is being sent to my pbx? I am having problems. Basically if i dial a DID from within my network it routes fine to a conference room i created... but if i dial it outside.. it goes to my default context.. so .. i feel like it is not being sent .. the way i am anticipating it |
| 15:18.54 | littleball | on port 1 and 2, it is switch |
| 15:19.02 | rantsh | mosty: ok, I'll try that |
| 15:19.02 | littleball | port 3 and 4 connect to cisco as5300 |
| 15:19.21 | rantsh | cuz I got the file from http://ftp1.digium.com/pub/asterisk/ and there's no docs there |
| 15:19.48 | CVirus | A network consisting of asterisk server + 4 ATA's connected to 4 analog phones ... Can I configure asterisk to forward an incoming call to the first free analog phone ? |
| 15:20.33 | [TK]D-Fender | littleball: So you provide timing to the Cisco? basically putting * betweent he telco & Cisco? |
| 15:21.03 | [TK]D-Fender | lirakis: Go configure the user for the call to come in on. |
| 15:21.04 | littleball | no. |
| 15:21.32 | littleball | i set to 0 from zaptel.conf file |
| 15:21.36 | littleball | on port 3 and 4 |
| 15:21.43 | penguinFunk | berktr: if your using an FXO card with analogue lines try fxotune |
| 15:24.08 | [TK]D-Fender | littleball: is span 1 the only one with a problem? |
| 15:24.13 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 15:24.14 | littleball | [TK]D-Fender, any idea? the strange thing is that although line 1 and 2 is Down, but no problem. line 3 and 4 is up but sometimes get hang |
| 15:24.30 | littleball | span 3 and 4 has problem. span 1 and 2 no problem |
| 15:24.52 | [TK]D-Fender | Not sure, you might want to try upgrading your * and zaptel, and check with the telco to see what they have to say |
| 15:25.14 | littleball | 1 and 2 connect to telco |
| 15:25.23 | littleball | althoug it is down, but no problem |
| 15:25.31 | [TK]D-Fender | littleball: if you're CPE to all of your equipment, then you shouldn't have "0" for the timing on those last 2 ports. |
| 15:25.38 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 15:25.56 | [TK]D-Fender | littleball: "0" mean * is PROVIDING timing, which isn't a great idea. |
| 15:27.23 | sergee | [TK]D-Fender: ACN seems to be a provider, do you know a model of video phone they sell? |
| 15:27.26 | littleball | i know, originally, it is not "0". because 3 and 4 connect to voip provider (not telco) and they told me it is due to timing issue, then decide to chagne to 0 |
| 15:27.44 | [TK]D-Fender | sergee: Go to their site. Go look, go ASK. |
| 15:28.00 | [TK]D-Fender | littleball: Ok, not sure what to do now then. |
| 15:28.12 | sergee | [TK]D-Fender: thanks |
| 15:28.51 | *** join/#asterisk Dovid (n=Dovid@79.178.24.155) |
| 15:29.46 | rantsh | mosty: thanks man, using p0 solved it |
| 15:34.43 | NovceGuru | http://connect.voicepulse.com/ charges 11/month/did, I can't tack on additional numbers and keep the same amount of channels, anybody suggest someone I can? |
| 15:34.57 | NovceGuru | s/I/that |
| 15:35.08 | Dovid | TK: How hard do you think it will be to add VAD to asterisk ? |
| 15:35.24 | Qwell[] | Dovid: if it were easy, it would already be done |
| 15:35.33 | [TK]D-Fender | Dovid: I don't code much, let alone for *. |
| 15:35.36 | galeras | Sirs: in a pure Lan environment, can cheap switches affect the quality of voice? |
| 15:35.41 | Qwell[] | galeras: yes |
| 15:36.28 | *** join/#asterisk xezz (n=asdasd@85.75.173.3) |
| 15:36.54 | [TK]D-Fender | galeras: In any sane scenario it shouldn't matter |
| 15:36.54 | *** join/#asterisk SwK_ (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 15:38.29 | xezz | hello, is there a sip phone like linksys spa 941 in example that supports a vpn client on its menu ? |
| 15:39.14 | Uatec | hey, i've got a new voice pack for asterisk, but all the files are *.g723, not *.gsm, if i swap them over, will asterisk still find the right files? |
| 15:39.25 | galeras | my customer has a lot of broascast because he has a lot of workgroups :(. can better switches (like 3com) help to solve this? |
| 15:39.32 | [TK]D-Fender | Uatec: * doesn't really support G.723 |
| 15:39.42 | mosty | xezz, voip over vpn probably introduces more latency than you want |
| 15:39.44 | festr__ | hello, anyone has expirience with sangoma 102D or 104D with hw echo? i'm trying to google but nothing usefull. so i'm asking here: does sangoma hw EC turn off when detecting fax? |
| 15:39.45 | [TK]D-Fender | galeras: No. |
| 15:39.53 | NovceGuru | Uatec, they should offer .gsm, if you payed for it |
| 15:39.58 | Uatec | hah, we didn't |
| 15:40.02 | Dovid | Qwell: How much time do you think it would take to write it and can I contract out Digium to do it ? |
| 15:40.02 | Uatec | doesn't really? or doesnt? |
| 15:40.05 | [TK]D-Fender | festr__: Yes |
| 15:40.09 | Uatec | how about wavs? |
| 15:40.10 | NovceGuru | you get them from voicevector? |
| 15:40.14 | festr__ | [TK]D-Fender: but it does not work for me :( |
| 15:40.15 | Uatec | i do have the plain old wavs |
| 15:40.23 | Uatec | nah |
| 15:40.30 | festr__ | [TK]D-Fender: my faxes are bad quality and echo does not turn off. any trick how to do this? |
| 15:40.32 | NovceGuru | you can probably convert those to sln easily |
| 15:40.38 | Uatec | the voicevector english english voice example was horribly american |
| 15:41.17 | NovceGuru | Uatec, I bet you could throw all the files in foobar2000 and convert them quickly |
| 15:41.18 | [TK]D-Fender | festr__: pastebin your wanpipe config |
| 15:41.20 | [TK]D-Fender | ~pb |
| 15:41.21 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:41.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 15:41.25 | festr__ | [TK]D-Fender: ok |
| 15:41.42 | Uatec | NovceGuru, bearing in mind i'm al azy bastard |
| 15:42.03 | NovceGuru | Uatec, make a simple for each script and use sox |
| 15:42.17 | NovceGuru | or a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm" resample –ql |
| 15:42.25 | festr__ | [TK]D-Fender: http://www.pastebin.ca/592079 |
| 15:42.26 | NovceGuru | for |
| 15:42.35 | *** join/#asterisk bbryant (i=brett@nat/digium/x-ee2698b086a44c21) |
| 15:43.11 | NovceGuru | or `sln |
| 15:44.03 | Err | heh, if you're going to bother using a search-and-replace with sed, why not just let it add the extension you want? :-) |
| 15:45.02 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:46.32 | festr__ | [TK]D-Fender: any idea? |
| 15:46.48 | rantsh | if I have an asterisk 1.4.5 and I want to go back to 1.4.4 it should be enogh if I recompile and reinstall the older one for it to overwrite the newer version |
| 15:46.50 | rantsh | right? |
| 15:47.38 | [TK]D-Fender | festr__: your LBO looks high. Pastebin your zaptel & zapata |
| 15:47.56 | [TK]D-Fender | rantsh: Yes |
| 15:48.55 | rantsh | [tk]D-fender: thanks |
| 15:49.06 | festr__ | [TK]D-Fender: http://www.pastebin.ca/592089 |
| 15:50.47 | festr__ | [TK]D-Fender: why LBO should be problem when voice is OK and PRI is digital without errors or clicks? the problem is, that HW echocan is not turning off like software echocan |
| 15:51.35 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:52.30 | [TK]D-Fender | festr__: the EC can hide a lot of problems... you may want to try setting to 0 if you are really close to your demarc |
| 15:52.48 | festr__ | [TK]D-Fender: yes i'm very close |
| 15:52.52 | [TK]D-Fender | jitterbuffers=4 <- ummm ICK? |
| 15:52.56 | festr__ | [TK]D-Fender: where to set it to 0? |
| 15:53.04 | festr__ | [TK]D-Fender: what is ICK? |
| 15:53.10 | [TK]D-Fender | echocancelwhenbridged = yes <- I wouldn't bother |
| 15:53.34 | [TK]D-Fender | festr__: And you should have "faxdetect=both" |
| 15:53.37 | festr__ | [TK]D-Fender: you recommend echocancelwhenbridged=no? |
| 15:53.46 | [TK]D-Fender | festr__: Maybe comment it out. |
| 15:53.50 | festr__ | [TK]D-Fender: but it will jump to fax extension? |
| 15:54.03 | festr__ | [TK]D-Fender: when i do faxdetect=both? |
| 15:54.06 | [TK]D-Fender | festr__: if available, but it will at least aid in detecting. |
| 15:54.26 | [TK]D-Fender | festr__: the exten is not needed. |
| 15:54.27 | Dovid | i have a stupid question. what is phone.conf used for ? |
| 15:54.30 | festr__ | [TK]D-Fender: good point i'll try put faxdetect=both first |
| 15:55.09 | festr__ | [TK]D-Fender: but, are you 100% sure that sangoma hw EC can turn off when detects fax? |
| 15:55.13 | Dovid | i am sorry. i meant to ask what is "Linux Telephony devices" |
| 15:55.22 | zdrulio | can i record a msgs ? |
| 15:55.24 | CVirus | A network consisting of asterisk server + 4 ATA's connected to 4 analog phones ... Can I configure asterisk to forward an incoming call to the first free analog phone ? |
| 15:55.46 | [TK]D-Fender | festr__: Yes, I run an A104d myself |
| 15:55.58 | [TK]D-Fender | CVirus: For the 15th time YES |
| 15:55.59 | mosty | CVirus, i answered that already, scroll up |
| 15:56.11 | [TK]D-Fender | CVirus: Get off your ass, install * and get to work! |
| 15:56.14 | festr__ | [TK]D-Fender: any debug messages or way to debug if it turns off? |
| 15:56.40 | [TK]D-Fender | festr__: I think core debug will show it, or jsut regular. |
| 15:57.02 | [TK]D-Fender | festr__: I believe there is a wanrouter tool to see the status of a channel as well |
| 15:58.17 | festr__ | [TK]D-Fender: faxdetect: Echo Cancellation: 128 taps, currently ON |
| 15:58.25 | festr__ | [TK]D-Fender: so it didnt turn off |
| 15:59.08 | festr__ | [TK]D-Fender: i've compared core debug on digium PRI and there is message it is turning off echocancel |
| 15:59.29 | festr__ | [TK]D-Fender: i've almost the same configuration and sangoma hw echocan stays on when faxing :( |
| 15:59.51 | CVirus | [TK]D-Fender: you never addressed my nickname explicitly ... sorry though |
| 16:01.07 | [TK]D-Fender | festr__: Call them up, their suport is great |
| 16:01.43 | festr__ | [TK]D-Fender: i'll rather write them (not native language :) but thank you for help |
| 16:02.09 | [TK]D-Fender | festr__: What languages do you speak? |
| 16:02.16 | festr__ | [TK]D-Fender: czech |
| 16:02.29 | [TK]D-Fender | festr__: Ok, that MIGHT be more troublesome :/ |
| 16:02.37 | [TK]D-Fender | festr__: But do ask them |
| 16:02.49 | *** join/#asterisk yxa (n=lonari@bb116-14-8-181.singnet.com.sg) |
| 16:02.55 | festr__ | [TK]D-Fender: i think yes :) i'm writing mail. i'm curious that noone have this issue when googling |
| 16:03.24 | [TK]D-Fender | festr__: Guess it's just YOU :| |
| 16:03.29 | yxa | any one has experience with the 1 port TE110P or TE120P? without echo canc, do they sound terrible? |
| 16:03.33 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:04.00 | festr__ | yxa: i'm recomend you sangoma |
| 16:04.17 | festr__ | yxa: i've irq issues with pci te110p |
| 16:04.21 | [TK]D-Fender | yxa: I would nver buy a card without HWEC. Feel free if you enjoy recreational Russian Roulette |
| 16:04.44 | festr__ | if you use MARK2 and aggressive echocan it sounds allright |
| 16:05.00 | festr__ | and it catches all echo issues in our environments |
| 16:05.05 | [TK]D-Fender | festr__: Still hit-or-miss. Not a game I enjoy |
| 16:05.06 | festr__ | but, hw echo is hw echo |
| 16:05.26 | festr__ | this is why we have invested to sangoma hw echo :) |
| 16:06.51 | yxa | dang, thanks guys. i almost bought it |
| 16:07.10 | yxa | sangoma cards are not that cheaper than digium |
| 16:07.20 | dansmith | so are sangoma analog cards (TDM400 clones) worth it or is better to stick with digium? |
| 16:07.30 | mosty | dansmith, they are better |
| 16:07.39 | dansmith | mosty: which are? |
| 16:07.43 | mosty | sangoma |
| 16:07.56 | dansmith | better, yet cheaper than digium? that's kinda scary :) |
| 16:08.04 | mosty | better echocan, better drivers, better hardware |
| 16:08.12 | yxa | sangoma cards are NOT cheaper. at least where i'm at |
| 16:08.14 | mosty | better diagnostic utils |
| 16:08.17 | coppice | just a pain to install |
| 16:08.35 | dansmith | I thought the analog cards were clones of the TDM400.. they use different drivers? |
| 16:08.39 | mosty | coppice, i wrote a short shell script to do it, it's not so bad |
| 16:08.42 | festr__ | better irq handling |
| 16:08.57 | coppice | the sangoma cards are not remotely like the digium ones |
| 16:09.01 | yxa | mosty why not post the install script somewhere |
| 16:09.19 | dansmith | oh, I think I'm thinking of openvox |
| 16:09.41 | dansmith | whoops :) |
| 16:09.44 | coppice | mosty: OK, then. just what ansers to their dumb questions will get you set up for R2? |
| 16:09.46 | mosty | yxa: all the info is on the sangoma wiki, most of the script is local specifics, ie which dir to compile in, where zaptel is etc |
| 16:10.03 | mosty | what's R2? |
| 16:10.18 | Uatec | NovceGuru, i've got foobar2000 installed, what format can i convert these files to that asterisk will accept without complaining? |
| 16:11.19 | s0ck | anyone used the asterisk manager interface? |
| 16:11.28 | mosty | s0ck, plenty of people have |
| 16:11.54 | s0ck | using an 'Action: Originate' to pass a call from a win32 app. it dials the end point, when they pick up, it rings the local extension |
| 16:11.57 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 16:12.02 | s0ck | i'd rather it dial the local extension first |
| 16:12.15 | s0ck | hoping someone can point me in the right direction |
| 16:13.16 | mosty | can you just swap the order of the src and dest? |
| 16:13.22 | s0ck | didn't appear to work |
| 16:13.23 | *** join/#asterisk oej (n=olle@193.214.121.128) |
| 16:13.25 | s0ck | i thought it would |
| 16:13.48 | Uatec | s0ck, i'm going to write something like you are writing in a couple of weeks. but no yet :\ |
| 16:14.26 | s0ck | i place exten above the channel but it still dialled the channel first |
| 16:14.34 | s0ck | wondering if originate is the wrong action to be using |
| 16:15.01 | mosty | s0ck, i guess the problem is it might not make sense to enter an extension if there's no channel connected already |
| 16:15.20 | s0ck | hmm ;/ |
| 16:17.05 | HarryR | Has anybody managed to integrate a gsm femtocell with asterisk? |
| 16:17.31 | JT | mosty: in answer to your earlier question about audio and PRIs |
| 16:17.44 | JT | mosty: i think what you're after is early media |
| 16:19.02 | mosty | JT: since asterisk gives a billsec value in CDR entries for PRI calls, i guess the answer must be yes |
| 16:19.16 | zdrulio | how can i record a calls ? |
| 16:19.25 | lirakis | zdrulio: monitor() |
| 16:19.35 | JT | mosty: billing time only starts when a call is deemed answered |
| 16:20.15 | mosty | jt: so asterisk guesses when that is? or is it known precisely? |
| 16:20.38 | JT | mosty: on analogue, that's as soon as the call is made, on pri, it is known |
| 16:20.49 | JT | via Q.931 messages |
| 16:20.57 | lirakis | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor |
| 16:21.26 | JT | however you must realise a far end pbx may answer the call and then give you another ringing indication of its own, from which point it'd be considered answered by the telco |
| 16:21.28 | JT | and ast |
| 16:22.26 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 16:22.28 | JT | even if no human or record message has played |
| 16:22.34 | JT | recorded |
| 16:23.44 | festr__ | [TK]D-Fender: so, it does not detect fax and does not turn off echocan in SW mode ... |
| 16:23.57 | festr__ | [TK]D-Fender: i've somewhere bug :) |
| 16:26.13 | [TK]D-Fender | s0ck: its not the ORDER. You need to change your CHANNEL. |
| 16:26.45 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:27.09 | s0ck | [TK]D-Fender: can you expand on that m8? |
| 16:27.10 | FuriousGeorge | hey all |
| 16:27.21 | s0ck | originally it was channel > exten |
| 16:27.34 | [TK]D-Fender | s0ck: your CHANNEL is what you dial. CHANGE IT. Use your imagination, you should already know what to do. |
| 16:27.36 | s0ck | i've also tried channel > channel (where both are sip) |
| 16:28.21 | s0ck | you are obviously more familiar with the manager interface than me :P |
| 16:28.30 | [TK]D-Fender | s0ck: Originate (just like a .call file) *Dial's* and Channel, and upon answer bridges to a specific point in your dialplan. |
| 16:28.51 | [TK]D-Fender | s0ck: this has NOTHING to do with AMI so much as realizing WHAT you are dialing. |
| 16:28.53 | s0ck | we are talking ami syntax now, not dialplan, yeh? |
| 16:29.28 | [TK]D-Fender | s0ck: I am 100% aware of what you want to do. Your poblem is that the type of channel you are dialing is SIP <- |
| 16:29.48 | s0ck | right... |
| 16:29.51 | [TK]D-Fender | s0ck: There's a HINT for you. Go think of what other types of CHANNELS you could put there. |
| 16:30.19 | [TK]D-Fender | s0ck: Naturally the right answer is probably the LAST thing to come to you... |
| 16:30.24 | s0ck | hehe |
| 16:30.34 | s0ck | i have no zap/iax trunks on this box |
| 16:30.44 | s0ck | unless im thinking in totally the wrong direction? |
| 16:31.03 | [TK]D-Fender | s0ck: 2 down! Right direction.. those ARE channel types. Keep going through the list! |
| 16:31.16 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 16:32.47 | s0ck | im confused ;/ everything is sip here |
| 16:33.15 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 16:33.28 | [TK]D-Fender | s0ck: You effectively want to use dialplan on BOTH sides of this call. What channel type supports that? |
| 16:33.35 | *** part/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 16:33.56 | [TK]D-Fender | s0ck: SIP is a means to an end, not the end itself :) |
| 16:34.14 | s0ck | pardon my ignorance, i'm fairly new to this |
| 16:34.20 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 16:34.52 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 16:35.04 | [TK]D-Fender | s0ck: just keep churning throught he list of channel types, I've even enlarged the hint there.... |
| 16:35.16 | s0ck | thanks :) |
| 16:35.18 | [TK]D-Fender | s0ck: the answer isn't FAR.... |
| 16:36.29 | vader-- | wonder what it would cost to get the voicemail message changed from Comedian Mail to a custom message |
| 16:36.44 | vader-- | with the same women's voice |
| 16:37.02 | Qwell[] | vader--: about $10 |
| 16:37.18 | Qwell[] | if it's a short phrase |
| 16:37.25 | HarryR | Qwell[]: she charges more than that doesn't she? |
| 16:37.57 | Qwell[] | $12 for 1-15 words |
| 16:38.18 | HarryR | Are numerics one word? |
| 16:38.19 | Qwell[] | http://www.digium.com/en/products/voice/ |
| 16:38.30 | Qwell[] | Allison is the one that does the original English voicemail prompts. |
| 16:38.45 | HarryR | ah I see |
| 16:41.28 | lirakis | i use http://www.research.att.com/~ttsweb/tts/demo.php for my voice prompts |
| 16:41.29 | lirakis | :p |
| 16:41.38 | HarryR | cheap git :) |
| 16:43.17 | vader-- | we just some plantronics cs70ns |
| 16:43.21 | vader-- | wireless headsets |
| 16:43.26 | vader-- | with the lifter kit |
| 16:43.28 | vader-- | they are pretty nice |
| 16:43.39 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:45.49 | [TK]D-Fender | Qwell[]: How is Corporate America supposed to take * seriously when we have "Comedian Mail"? :) |
| 16:49.46 | lirakis | i am having an issue with routing a DID. Please see http://pastebin.ca/592190 |
| 16:50.58 | lirakis | I have a DID from my provider (telasip) and I have a SIP DID routed to me from RNK. When I dial either DID from a phone on my network, they route fine. When I dial the RNK DID from a phone not on my network.. it hits my pbx then goes to my [default] context |
| 16:51.22 | lirakis | .. why is it going to [default] when dialed from an external phone??? any help is greatly appreciated |
| 16:51.42 | [TK]D-Fender | lirakis: Well.... without seeing your sip.conf or having any idea what your networking scenario is like, what kind of help do you expect? |
| 16:52.02 | [TK]D-Fender | lirakis: [inbound] is NOT some magical name with reserved properties |
| 16:52.29 | lirakis | [TK]D-Fender: so.. I should route my calls in [default] ? |
| 16:52.37 | lirakis | duh ... i didnt even try that |
| 16:52.46 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 16:52.55 | [TK]D-Fender | lirakis: ... you haven't shown us your SIP config. Where they heck do you think you TELL it whgere to go?! |
| 16:53.07 | [TK]D-Fender | *sigh* |
| 16:53.09 | *** join/#asterisk O_Zone (n=pbx@host91-245.pool8252.interbusiness.it) |
| 16:53.12 | O_Zone | hi all |
| 16:54.28 | lirakis | [TK]D-Fender: [general] |
| 16:54.28 | lirakis | conext=default |
| 16:54.36 | lirakis | i suspect .. its going to default |
| 16:55.05 | [TK]D-Fender | lirakis: ITSP'sshould have their own user to receive calls under defining auth params and context. |
| 16:55.28 | lirakis | ITSP ?? |
| 16:55.31 | Qwell[] | ~itsp |
| 16:55.32 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 16:55.33 | lirakis | oh |
| 16:55.35 | lirakis | i got it |
| 16:55.38 | *** join/#asterisk _0penser_ (n=Administ@202.4.107.19) |
| 16:55.49 | [TK]D-Fender | lirakis: And I still don't know if your networking scenario is workable since you are spending all this time giving a really POOR description of your config instead of just providing it. |
| 16:56.18 | lirakis | .. well its got a lot of clients and auth in it.. it would take a some time to clean it for public consumtion |
| 16:56.28 | [TK]D-Fender | Qwell[]: So... "Comedian Mail".... who's laughing now? The joke is long dead. Pass it on... |
| 16:56.28 | lirakis | .. let me try this.. if it doesnt work.. i will provide my conf |
| 16:56.42 | Qwell[] | [TK]D-Fender: most people don't even know what the joke *IS* |
| 16:56.45 | [TK]D-Fender | lirakis: I would gess about 30 seconds. |
| 16:56.52 | [TK]D-Fender | Qwell[]: EXACTLY. |
| 16:57.42 | irule | what ports may I forward from my adsl modem to my * server so that I can connect to it from another location over the internet? |
| 16:57.59 | vn | uhm..some of my IP phones are having high pitch background noises sometimes, and I'm not changing anything on my systems...what could be the cause? t-storms? |
| 16:58.29 | rob0 | irule, that would depend entirely upon what protocols you'll be using. |
| 16:58.54 | lirakis | [TK]D-Fender: yeah i got it figured out |
| 16:59.22 | mosty | irule, what kind of phones? |
| 16:59.50 | irule | xten |
| 17:00.25 | mosty | what kind of headset? |
| 17:00.37 | *** part/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 17:01.08 | [TK]D-Fender | irule: 5060,10000-20000 |
| 17:01.21 | rob0 | udp |
| 17:01.25 | [TK]D-Fender | yup |
| 17:04.36 | lirakis | [TK]D-Fender: .. sorry its been a while since I set up any routes on my * box. This RNK DID is not associated with any kind of providor... I have a block that have been pointed to a SBC.. and i just provisioned this DID on the SBC to go to my * box for testing.. so it goes to the context set in sip.conf.. which .. is [default] ... my telasip trunk has a context of its own that has routing set up.. |
| 17:05.22 | [TK]D-Fender | lirakis: You set up your accounts.... you should have a user for it to land on to control the context and auth. |
| 17:05.51 | skymeyer | evening, any chance on letting asterisk use SIP over TCP to connect to Exchange Unified Message server ? |
| 17:06.15 | lirakis | [TK]D-Fender: .. so i assume i could create an "account" just using the host variable.. and point it to a context.. so any DID coming from my SBC goes to that context? |
| 17:06.35 | [TK]D-Fender | lirakis: More or less, yeah |
| 17:06.39 | lirakis | okay |
| 17:06.40 | lirakis | cool |
| 17:06.48 | [TK]D-Fender | skymeyer: No. * does not support SIP over TCP. |
| 17:06.49 | rob0 | I think someone's working on SIP-over-TCP, but last I heard, still alpha stage. |
| 17:07.15 | lirakis | [TK]D-Fender: thanks.. thats useful information... i just dont do this stuff day in/out enough to come up with it immediately.. it takes some tinkering for me to get it |
| 17:07.17 | lirakis | :p |
| 17:07.17 | skymeyer | [TK]D-Fender: ok thanks, saves me a lot of research because it wasnt clear on the net |
| 17:07.21 | Err | ugh, fixed-rate streams over TCP - what a terrible idea |
| 17:07.27 | rob0 | indeed |
| 17:07.38 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 17:07.47 | skymeyer | Err: SIP = signalisation and not the voice transport ;) |
| 17:07.49 | rob0 | Sounds like something Microsoft would implement :) |
| 17:07.54 | [TK]D-Fender | err : There is no streaming in SIP. |
| 17:08.08 | Err | ah, you mean just for the SIP part, and not the streams it negotiates |
| 17:08.14 | [TK]D-Fender | err : and it CAN be TCP just fine. |
| 17:08.24 | Err | sure, I agree |
| 17:08.29 | [TK]D-Fender | err : RTP is ANOTHER matter |
| 17:08.35 | skymeyer | Err: mostly used to protect with SSL/TLS |
| 17:08.47 | Err | right right - I wasn't thinking - too many acronyms :-) |
| 17:08.47 | skymeyer | whatever, thanks anyway [TK]D-Fender for the info ;) |
| 17:08.58 | [TK]D-Fender | skymeyer: np. |
| 17:09.13 | [TK]D-Fender | skymeyer: M$ = dumb, but I'm sure I didn't have to say it.. |
| 17:09.36 | skymeyer | [TK]D-Fender: thats for sure ;) customers asked this to test it out though |
| 17:09.51 | [TK]D-Fender | To test what, * or M$? :) |
| 17:10.08 | skymeyer | connecting asterisk to unified message server from M$ |
| 17:10.22 | skymeyer | but UM seems to only support SIP over TCP :s |
| 17:10.24 | [TK]D-Fender | skymeyer: Oh... its too late for them then.... |
| 17:10.29 | skymeyer | :) |
| 17:10.30 | skymeyer | hehe |
| 17:11.27 | *** join/#asterisk osiris250 (i=q42e3ajb@bsd02.evansengineering.net) |
| 17:12.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:13.28 | *** join/#asterisk oej (n=olle@193.214.121.128) |
| 17:14.07 | vn | can * run in a vm? |
| 17:14.31 | Corydon76-work | It can, yes |
| 17:14.37 | rob0 | why not? |
| 17:14.42 | vn | any problems? |
| 17:14.50 | Corydon76-work | I wouldn't run it in production, but yes, you can. |
| 17:14.58 | rob0 | If it's set up wrong, lots of problems. :) |
| 17:15.01 | vn | dunno, was thinking there could be hardware support problems |
| 17:15.12 | rob0 | what hardware? |
| 17:15.23 | Corydon76-work | Not everybody runs it with telephony hardware |
| 17:15.26 | vn | well my only good machine's on windows so... |
| 17:15.26 | Err | heh, there very well may be issues if you're trying to use internal interface cards - but that wasn't your question :-) |
| 17:15.40 | vn | didn't choose the hardware yet |
| 17:15.53 | rob0 | ~ata |
| 17:15.53 | jbot | i guess ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 17:16.17 | Corydon76-work | I wouldn't run a production server on Windows, either |
| 17:16.38 | vn | yea but I've been told that IAX ATAs are crap and SIP has problems with NATing |
| 17:16.43 | vn | well it's for personal use |
| 17:17.13 | tzafrir | vn, that's mainly because there are very few of them |
| 17:17.21 | tzafrir | (IAX ATAs) |
| 17:17.22 | rob0 | If your SIP ATA is on the same subnet, why NAT it? |
| 17:17.28 | Corydon76-work | Personal use, one FXO channel? Rescue a machine from the dumpster |
| 17:17.46 | tzafrir | FXO channel or FXS channel? |
| 17:18.14 | vn | PO...wha's the term already heh |
| 17:18.16 | Corydon76-work | presumably FXO if he's connecting to a phone line |
| 17:18.19 | vn | analog phone |
| 17:18.27 | vn | no, I won't have a phone line |
| 17:18.32 | vn | only internet |
| 17:18.44 | Corydon76-work | Then FXS or SIP phone |
| 17:19.23 | vn | rob0: ain't I forced to NAT it if I NAT my computers, use a router+ a firewall? |
| 17:19.26 | Corydon76-work | I'd only use an ATA if you absolutely needed to use an old analog phone that wasn't next to the server |
| 17:19.53 | vn | can a 486 be a * derver? |
| 17:19.57 | vn | server* |
| 17:20.03 | Corydon76-work | vn: you can use SIP on the LAN and IAX2 to your call provider |
| 17:20.21 | vn | and no NAT problem? |
| 17:20.25 | Corydon76-work | I wouldn't provision anything less than a P133 |
| 17:20.42 | rob0 | vn, you have your * at, say, 192.168.5.5, and your ATA at 192.168.5.6 on the same Ethernet segment. Why NAT that? |
| 17:20.45 | tzafrir | vn, hmm, well, basically, sortof, yes |
| 17:20.46 | Corydon76-work | with 64MB RAM |
| 17:20.55 | Corydon76-work | That rules out a 486 |
| 17:21.17 | tzafrir | vn, if you're really easy on it (e.g: no transcoding of compressed codecs, not too many channels and such) |
| 17:21.43 | vn | I'll have 10MBps line so no problem..hehe |
| 17:22.03 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 17:22.17 | vn | so should I simply use an ATA or not? |
| 17:22.34 | Corydon76-work | Get a SIP phone instead |
| 17:22.44 | vn | I'm reticent to power another computer in my room simply for this |
| 17:22.49 | irule | anybody have some sort of script that will report what language files are missing that do exist in another lang? |
| 17:23.14 | Corydon76-work | vn: so you like jittery audio? |
| 17:23.15 | *** join/#asterisk oej (n=olle@193.214.121.128) |
| 17:23.32 | vn | if it was that easy...I'm hard of hearing and need specific phone equipment on wich I need strong volume and the possibility to adjust the tone |
| 17:23.38 | vn | ATA = jittery? |
| 17:23.56 | Corydon76-work | No, running on Windows under a VM = possible source of jitter |
| 17:24.01 | rob0 | VM running under windows ... yes |
| 17:24.09 | rob0 | or on a 486 |
| 17:24.12 | vn | lol ok yeah that was just an idea like that |
| 17:24.30 | Corydon76-work | especially if the windows machine is doing more than just sitting idle |
| 17:25.14 | vn | well I never blew my memory limit and never used the cpu at 100%...it's a Kentsfield QX6700 |
| 17:25.17 | *** join/#asterisk Tili (n=tili@203.170.74.167) |
| 17:25.46 | `Sean | Hey Corydon76-home i was wondering if you knew anything about how to get callerID information even if the caller is calling via *67 |
| 17:26.06 | vn | wait I got a p2 2600 128 MB RAM |
| 17:26.11 | vn | that could do fine I guess |
| 17:26.17 | vn | 266* |
| 17:26.17 | irule | an idle windows is never just idling |
| 17:26.32 | rob0 | People who want to accomplish real things in Unix generally devote real hardware to the matter; no getting around that. Of course simple things like routing can be done on dinosaurs. |
| 17:26.39 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 17:26.53 | rob0 | Yeah, a p266 with ATAs should be okay. |
| 17:27.04 | rob0 | for small-scale use :) |
| 17:27.11 | rob0 | ata/SIP phone |
| 17:27.16 | vn | yeah I do know that...but why complicating the task when it can be easy |
| 17:27.19 | irule | just a couple phones |
| 17:27.24 | vn | just one |
| 17:27.38 | *** join/#asterisk adeeln (i=adeeln@c-67-161-185-121.hsd1.ca.comcast.net) |
| 17:28.38 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 17:28.55 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 17:29.43 | adeeln | while compiling asterisk 1.2.17, i seem to be getting a bunch of undefined symbols in just about all the apps/modules...how can i resolve them? |
| 17:29.53 | vn | so I'd need a fxo IAX2 card + ATA, right? |
| 17:30.08 | vn | fxo or fxs...I'm always confused |
| 17:31.46 | rob0 | vn: Corydon76-work recommended a SIP phone. But if ATA, you want FXS (station), not FXO (office). |
| 17:31.54 | anonymouz666 | what means wildcard ? |
| 17:32.01 | anonymouz666 | why all digium card are wildcards? |
| 17:32.45 | mike_jh | Cause they're wild |
| 17:32.49 | mike_jh | Untamed |
| 17:32.53 | anonymouz666 | born to be wild |
| 17:32.58 | Corydon76-work | That's like asking why all Boeing jet models start with 7 |
| 17:33.10 | anonymouz666 | there is nothing related to modules? |
| 17:34.19 | mike_jh | Corydon76-work: What about Dreamliner? |
| 17:34.36 | Strom_M | 787 |
| 17:34.49 | Corydon76-work | A.D. |
| 17:34.56 | Strom_M | or was that the 777 |
| 17:34.57 | vn | rob0: well I have no idea if I'd have way more control on the sound with the SIP phone than my actual phone |
| 17:35.04 | Eradan | Anyone have much negative / positive experience with the Linksys 942 / 962's ? ... compariing them to a 7960 / 7970 ? |
| 17:35.18 | mike_jh | What about the F22? |
| 17:35.41 | Corydon76-work | mike_jh: it takes passengers? |
| 17:35.47 | vn | I have a 70dB amplification actually, which I want to keep, with the tone setting |
| 17:37.24 | Strom_M | Eradan: I have a 942 and a 7960...the cisco phone is much higher build quality |
| 17:37.47 | mike_jh | You said planes, you made no mention of passengers |
| 17:38.00 | mike_jh | But since you mention it, the DC10? |
| 17:38.05 | Corydon76-work | Tangent |
| 17:38.25 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 17:39.05 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:39.27 | Eradan | Strom_M: functionally ... do you notice much difference ? ... i have a 70 and love it .. but i'm looking at mass rollout and i like the form factor of the 942 and the fact that it has a 1.5m jack, but was really looking at functions / user experience. SO thank you much. |
| 17:39.35 | *** join/#asterisk havoc (n=havoc@saturn.chaillet.net) |
| 17:39.36 | [TK]D-Fender | Eradan: I wouldn't suggest EITHER |
| 17:39.44 | Eradan | [TK]D-Fender: what would you suggest ? |
| 17:39.52 | Corydon76-work | Polycom |
| 17:39.57 | [TK]D-Fender | Eradan: Cisco over-priced, Linksys = underfunctioned. |
| 17:40.04 | [TK]D-Fender | Polcyom > All. |
| 17:40.06 | havoc | damn, big channel |
| 17:40.37 | Corydon76-work | It's reportedly the largest channel on Freenode |
| 17:40.39 | Eradan | [TK]D-Fender i don't care about price. I care about functionality. What polycom is comparable to a 60 or 942 ? |
| 17:40.39 | Strom_M | yeah, i'd say polycom |
| 17:40.54 | [TK]D-Fender | Eradan: THIS is what you want then : http://www.telephonydepot.com/product_p/105-058-320.htm |
| 17:41.19 | Eradan | [TK]D-Fender: damn those phones are ugly. |
| 17:41.32 | vn | is there somewhere I can do say...try a sip phone for 30 days and then send it back if I don,t like it, pay it if I like it? |
| 17:41.37 | Corydon76-work | If you don't care about price, then get a Polycom 650 |
| 17:41.51 | vn | Corydon76-home: there's more ppl on ##linux |
| 17:41.53 | [TK]D-Fender | Eradan: Not really, seen far worse. Linksys lacks on functions & quality... |
| 17:42.10 | Eradan | [TK]D-Fender: how are the polycom for configuration ? |
| 17:42.18 | Corydon76-work | vn: might have been "most active", I dunno |
| 17:42.32 | [TK]D-Fender | Eradan: A bit complex at the start, but extremely flexible |
| 17:42.55 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:43.08 | Eradan | Corydon76-work: the 650 color ? |
| 17:43.14 | vn | can you mix the sound with the configuration of a SIP phone? |
| 17:43.14 | Corydon76-work | black |
| 17:43.36 | Eradan | Corydon76-work Does polycom offer something that competes with the 70 ? |
| 17:43.56 | [TK]D-Fender | Eradan: Competes HOW? |
| 17:43.59 | Corydon76-work | Dunno, what is it that you like about the 70? |
| 17:44.23 | Eradan | [TK]D-Fender color touch screen w/ the high quality voice. |
| 17:44.41 | [TK]D-Fender | Eradan: Polycom supports sane paging, no-charge firmware, presence support, dual-hedset connections, *JOIN/SPLIT* |
| 17:44.42 | Corydon76-work | No, there's no touch screen AFAIK |
| 17:44.57 | [TK]D-Fender | Eradan: As for quality Polycom is an easy tie with Cisco if not one-upping. |
| 17:44.58 | Eradan | Good sound quality, touch screen, programable directories. |
| 17:45.09 | Eradan | [TK]D-Fender Very good to know. |
| 17:45.20 | Eradan | Although it sucks ... cause now i gotta order some polycom to test :( |
| 17:45.27 | Eradan | that'll push my roll out back a week :( |
| 17:45.31 | Corydon76-work | Realize that Polycom was Cisco's original vendor for rebranded phones... |
| 17:45.35 | [TK]D-Fender | Eradan: It IS a phone you know.... if you want to pay a fortune for something your uses pick up to talk to someone, thats your perogative. |
| 17:45.48 | Eradan | [TK]D-Fender :P ... point well taken. |
| 17:46.38 | *** join/#asterisk friedrich| (n=friedric@e177253231.adsl.alicedsl.de) |
| 17:46.50 | Eradan | Holy schkit .. the sidecar is SWEET. |
| 17:47.30 | Corydon76-work | You can add 3 sidecars, if that floats your boat |
| 17:47.37 | Eradan | yah ... saw that ... very cool. |
| 17:47.43 | Eradan | i was gonna do a softphone for the receptionist. |
| 17:48.00 | Corydon76-work | Then again, the SNOM 320 with a sidecar is less expensive |
| 17:48.04 | Eradan | But i can't get one that works worth a crap. |
| 17:48.52 | [TK]D-Fender | Corydon76-work: On the topic of sidecars, NOBODY touches Aastras LCD one... |
| 17:49.11 | [TK]D-Fender | Corydon76-Esp as its paged, and better presence support. |
| 17:49.26 | Corydon76-work | Given that SNOM is as enthusiastic about the community as Digium, I fully expect SNOM to integrate better with Asterisk in the coming years |
| 17:49.30 | Eradan | [TK]D-Fender When you say 'presence' support ... what's that referring to ? |
| 17:49.44 | [TK]D-Fender | Eradan: Lighted indicators so you can see phones in use, etc |
| 17:49.51 | Eradan | [TK]D-Fender Gotcha ... |
| 17:49.56 | Eradan | That's what i assumed. |
| 17:50.03 | Eradan | I realy like the web panel. |
| 17:50.15 | Eradan | although flash has a little to be desired. |
| 17:50.25 | Corydon76-work | Uh, web panel? |
| 17:50.29 | Eradan | much more informative than the sidecar and much cheaper :P |
| 17:50.30 | [TK]D-Fender | http://www.telephonydepot.com/product_p/105-057-560m.htm <- 20 at a time, 3 pages, backlight, coloured indicators, and CHEAP. |
| 17:50.40 | Corydon76-work | You mean the configuration screen or the minibrowser? |
| 17:50.51 | [TK]D-Fender | Corydon76-work: No, he means FOP |
| 17:50.58 | Eradan | the flash pannel |
| 17:51.08 | Corydon76-work | oh, fop |
| 17:51.17 | *** join/#asterisk jm|home (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 17:51.24 | Corydon76-work | I thought he was still talking about the Polycom |
| 17:51.46 | Eradan | sorry :( |
| 17:51.52 | brettnem | [TK]D-Fender: but what about BLA for polycom :) |
| 17:52.03 | [TK]D-Fender | Corydon76-work: But just look at Aastra's.... now if only their handset wasn't a weightless POS, not have those rubbery-ass buttons, and Polycom's call handling! |
| 17:52.20 | [TK]D-Fender | brettnem: Referring to BLF/Presence? |
| 17:52.30 | *** join/#asterisk Dantix (n=Dante@200.68.70.100) |
| 17:52.30 | *** join/#asterisk mazpe (n=email@68.152.128.30) |
| 17:52.34 | brettnem | [TK]D-Fender: No, shared lines |
| 17:52.38 | [TK]D-Fender | brettnem: Its decent, but pricey for attendant modules. |
| 17:52.46 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:52.52 | brettnem | like presence with ringing |
| 17:53.15 | [TK]D-Fender | brettnem: SLA = 2-bit hack that really doesn't cut it and it next to worthless for anyone NOT using a sidecar and having masochistic tendencies ;) |
| 17:53.33 | brettnem | yes, I know. I've gone down that road :) |
| 17:53.41 | brettnem | I need real BLA |
| 17:53.50 | [TK]D-Fender | brettnem: as far as ASTERISK's is concerned :) Now when someone gets around to doing TRUE SIP-B well thats ANOTHER matter :) |
| 17:54.09 | brettnem | And I'm real disappointed in the polycom LCS stuff over some of the other standards |
| 17:54.35 | [TK]D-Fender | brettnem: I Q&A'd that understanding direct from russellb while explaining where the people who guided him towards that "solution" went horribly wrong. |
| 17:54.38 | brettnem | [TK]D-Fender: Have you made real BLA work with the pcoms outside of asterisk? |
| 17:54.55 | [TK]D-Fender | brettnem: I don't have a platform that supports it. |
| 17:55.13 | [TK]D-Fender | brettnem: Either way, there are still true issues with it. |
| 17:55.28 | brettnem | I tried with OpenSER.. It seems to work off the BroadCom BLA stuff more than the sipping specs |
| 17:55.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 17:55.58 | brettnem | ie: subscription with line-sieze event |
| 17:56.15 | [TK]D-Fender | brettnem: With Polycom you're still limited to 12 regs on a 601. At best you could push 6 linekeys for the primary reg, and push the rest to an attendant module. |
| 17:56.16 | Dantix | hi, I have an openvox a1200p board with 1 fxo and 2 fxs modules, just working... my problem is the zap channel (FXO) is countinuously taking itself off hook. At full log I see each time the channel go off hook: Jun 27 14:55:22 NOTICE[4363] chan_zap.c: Got event 18 (Ring Begin)... seems the channels is "viewing" a ring tone does not exist.. where can I start to trouble this issue? |
| 17:56.18 | brettnem | I was going to try to hack it up with OpenSER |
| 17:56.26 | festr__ | [TK]D-Fender: so, i've disabled hw echocan and i'm trying to send fax over TDM and it is bad too.. i've tryid the same fax and ATA on another server with digium and it works... any ideas? voice is good without any hearable issues |
| 17:56.28 | [TK]D-Fender | brettnem: That'd leave you 11 to watch |
| 17:56.46 | [TK]D-Fender | festr__: What versions of wanpipe, asterisk, and zaptel? |
| 17:56.47 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 17:57.07 | brettnem | yeah, I'm just interested in doing 2 or 3 properly for starters.. I have to setup a presence button and a second sip registration for each line to watch now.. it's very confusing for the customer |
| 17:57.08 | [TK]D-Fender | brettnem: that might do but I'm not qualified |
| 17:57.20 | festr__ | [TK]D-Fender: asterisk 1.4.5, zaptel latest svn and wanpipe wanpipe-3.1.2.p5 becuase of kernel 2.6.21.5c2 |
| 17:57.24 | [TK]D-Fender | brettnem: CrapTASTIC! |
| 17:57.38 | brettnem | yeah, no kiddin |
| 17:57.43 | [TK]D-Fender | festr__: You're WAY too bleeding-edge... ease off |
| 17:57.58 | festr__ | [TK]D-Fender: sorry dont understand |
| 17:58.30 | brettnem | I'm half temped to suggest an ATA, with hardwired bridged lines and good ol fashioned business telephones |
| 17:58.34 | festr__ | [TK]D-Fender: i think that i should downgrade to 2.6.18 and wanpipe wanpipe-2.3.4-10 |
| 17:59.01 | brettnem | I guess not many people want bridged lines |
| 17:59.12 | *** join/#asterisk gardo (n=gardo@121.97.209.126) |
| 18:00.58 | *** part/#asterisk _0penser_ (n=Administ@202.4.107.19) |
| 18:04.03 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 18:05.41 | Eradan | [TK]D-Fender Which polycom phone did you say had the split non-rj9 headset ports ? |
| 18:06.00 | *** join/#asterisk eatmypiano (n=eatmypia@host86-132-181-229.range86-132.btcentralplus.com) |
| 18:07.55 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 18:10.23 | [TK]D-Fender | Eradan: 320/330 have 2.5 & RJ9 |
| 18:11.48 | Eradan | [TK]D-Fender any experience with a non-amplified headset on those models ? |
| 18:12.03 | [TK]D-Fender | Eradan: Not personally, I know 1 guy though. I could ask |
| 18:12.17 | Eradan | [TK]D-Fender if possible that would be awesome |
| 18:12.32 | [TK]D-Fender | Eradan: Not online now... I look around later |
| 18:13.08 | Eradan | [TK]D-Fender Thanks. |
| 18:13.15 | Eradan | [TK]D-Fender Gonna order one today for testing. |
| 18:13.48 | mazpe | so after i install libpri, zaptel drivers, wanpipe, wanpipe-utils and ran setup-sangoma.. my T1 PRI card shold be setup. |
| 18:13.56 | mazpe | so i have to create trunks for the channels? |
| 18:14.43 | [TK]D-Fender | mazpe: Have you ever installed and configured Asterisk before? |
| 18:15.00 | mazpe | first time with voicepluse. |
| 18:15.02 | [TK]D-Fender | Eradan: Keep in mind its PoE only, but you can order a brick for it. |
| 18:15.12 | mazpe | first time with a T1 PRI |
| 18:15.19 | [TK]D-Fender | mazpe: then you've done this all BACKWARDS. |
| 18:15.29 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:15.31 | mazpe | i'm realizing that. |
| 18:15.39 | [TK]D-Fender | mazpe: Time to actually LEARN how to use *. Go read THE BOOK for a while |
| 18:15.41 | [TK]D-Fender | ~book |
| 18:15.42 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:16.00 | mazpe | reading is good =) |
| 18:16.13 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-17-38.bflony.east.verizon.net) |
| 18:16.22 | [TK]D-Fender | Eradan: If you are considering the brick, I'd say "don't" and jsut get a single PoE injector. Its recyclable |
| 18:16.34 | Eradan | [TK]D-Fender nah ... POE only. |
| 18:16.46 | Eradan | the 320 fits the bill perfectly ... it's just ugly. |
| 18:16.47 | [TK]D-Fender | Eradan: didn't know if you were equiped |
| 18:17.00 | [TK]D-Fender | Eradan: Get a BT 101 then ;) |
| 18:17.10 | Eradan | [TK]D-Fender ? 101 ? |
| 18:17.29 | *** join/#asterisk joe_bag_odonuts (n=erne@dsl027-164-165.atl1.dsl.speakeasy.net) |
| 18:17.32 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-30988135680.htm |
| 18:17.33 | Corydon76-work | Grandstream Budgetone |
| 18:17.49 | rantsh | is anyone here using sox to record calls on asterisk? |
| 18:18.02 | SuPrSluG | man asterisk has been a real peach here lately. now I'm getting voicemails cut off/short for no apparent reason. |
| 18:18.03 | [TK]D-Fender | rantsh: No. |
| 18:18.04 | rantsh | actually asterisk 1.4.x? |
| 18:18.09 | Nugget | rantsh: not since digium added MixMonitor() |
| 18:18.10 | Eradan | [TK]D-Fender Even uglier :P |
| 18:18.16 | Strom_M | the Grandstream "what, did you buy this out of the clearance bin at wal-mart?" Budgetone |
| 18:18.23 | Eradan | [TK]D-Fender I think the 320 is perfect, ordering one now. |
| 18:18.27 | [TK]D-Fender | Eradan: Oh I'n sorry... I forgot the closing tag! |
| 18:18.31 | [TK]D-Fender | </sarcasm> |
| 18:18.32 | rantsh | is MixMonitor really better? |
| 18:18.42 | Nugget | it doesn't require a working sox install, so yes, I think it is. |
| 18:18.44 | SuPrSluG | I seem to be in asterisk bizarro world |
| 18:18.46 | irule | exten => s,n,GotoIf($["${DEFTIMEOUT}" = "5"]?h:n) does not work, exten => s,n,GotoIf($["${DEFTIMEOUT}" = "5"]?h) does work, is there a different thing for next or should I just leave it like the second? the CLI gives me a warning to checn the documentation :s |
| 18:19.49 | [TK]D-Fender | irule: h & n are not valid choices. |
| 18:20.13 | Eradan | [TK]D-Fender What's interesting ... i've budgeted for ~ 250 / phone ... i could squeeze the 550's in : ) ... |
| 18:20.33 | [TK]D-Fender | Eradan: 550 = total waste. I'd sooner hear you go for th 650 :) |
| 18:20.49 | Eradan | [TK]D-Fender oh ? |
| 18:20.57 | [TK]D-Fender | Eradan: Polycom has created a categor killer with the IP 320/330 |
| 18:21.26 | [TK]D-Fender | Eradan: 650 can have sidecar expansion, USB, 2 more line-keys, etc, for about $40 more. 550 = WASTE |
| 18:21.32 | Eradan | yah .. you just can't beat that pricepoint for a POE phone. |
| 18:21.45 | [TK]D-Fender | Eradan: Don't skimp out to buy a dead end at that margin. |
| 18:21.55 | Eradan | [TK]D-Fender yah ... /agree |
| 18:22.45 | AndrewGearhart | [TK]D-Fender: I have to sheepishly repeat a question I've asked you before. |
| 18:23.02 | AndrewGearhart | [TK]D-Fender: best softphone out there? |
| 18:23.40 | Eradan | AndrewGearhart: find one ... and let me know ... as i havn't found anything decent!. |
| 18:24.17 | AndrewGearhart | Eradan: heh. what have you tried? |
| 18:24.32 | Eradan | AndrewGearhart: xlite and sjphone. |
| 18:24.47 | AndrewGearhart | what were your complaints with them? |
| 18:24.58 | Eradan | AndrewGearhart: voice quality. |
| 18:25.09 | Eradan | AndrewGearhart both sounded like poop. |
| 18:25.17 | AndrewGearhart | doh |
| 18:25.26 | rantsh | going back to the recording issue for a second here, I succesfully record mine with monitor and mixmonitor |
| 18:26.00 | rantsh | but for some reason where calls are being recorded and both UA are using the same codec there's no audio on one of the ways |
| 18:26.12 | *** join/#asterisk javar (n=javar@69.79.134.24) |
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| 18:27.57 | *** part/#asterisk Nuitari (n=Nuitari@melchior.nuitari.net) |
| 18:30.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:31.03 | DEac- | i heared about an echo-server. how can i realize an echo-server in my-asterisk? |
| 18:31.07 | rantsh | no one knows what could be going wrong? |
| 18:31.46 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:32.05 | rantsh | I know [tk]d-fender tried helping me yesterday (I really apreciate it) but I can't seem to see what the problem could be |
| 18:34.30 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:34.53 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:35.16 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:35.53 | SuPrSluG | ok i'm looking at a voicemail message .txt file duration=46. so, why would playback be a problem? |
| 18:37.57 | awk | sorry, what way can i do conversions to mp3 for voicemail |
| 18:38.20 | awk | or would i have to script it to pass to a conversion process before mailing the voicemail to me? |
| 18:38.30 | Corydon76-work | DEac-: show application echo |
| 18:38.31 | *** join/#asterisk AntiLoop (n=bostjan@84-255-253-30.static.dsl.t-2.net) |
| 18:38.35 | AntiLoop | hi all |
| 18:38.50 | DEac- | Corydon76-work: thanks |
| 18:39.32 | AntiLoop | i'd like to set sipusers => pgsql,asterisk,sipusers extconfig.conf, but i don't know where to set pgsql parameters like hostname,... ? |
| 18:39.40 | AntiLoop | or do i need odbc for that ? |
| 18:41.35 | [TK]D-Fender | AndrewGearhart: eyeBeam |
| 18:42.14 | [TK]D-Fender | rantsh: You did NOT prove to me what codecs were in play now did I see FULL CLI output. |
| 18:42.28 | AndrewGearhart | [TK]D-Fender: thanks! |
| 18:42.47 | [TK]D-Fender | s/now/nor/ |
| 18:43.36 | rantsh | [TK]D-Fender: I know I didn't send you enough data |
| 18:44.02 | *** part/#asterisk havoc (n=havoc@saturn.chaillet.net) |
| 18:44.09 | *** join/#asterisk bmg505 (n=leon@196.209.179.191) |
| 18:44.12 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:44.23 | [TK]D-Fender | rantsh: So stop thinking you've done everything to find out why. You were getting G.729 issues and it is not lying that they do NOT match |
| 18:45.05 | rantsh | [TK]D-Fender: but it's a weird case, no matter what codec I use as long as I'm recording one of the parties can't listen... comment the Monitor or MixMonitor line and everything is peachy |
| 18:45.48 | rantsh | [TK]D-Fender: g729? I've never come here asking about g729 [yet] |
| 18:45.53 | *** join/#asterisk racker2 (n=matias@mail.rack2.com.ar) |
| 18:46.19 | [TK]D-Fender | rantsh: pretty sure I saw that in there, but anyways, enough TALK about this. Pastebin some real backup if you want help. |
| 18:47.00 | irule | how can I limit the recording time time for a voicemail |
| 18:47.01 | awk | ok, somebody please answer my question,. saving voicemail in mp3 format? what is the process or where can I find docummentation |
| 18:47.07 | rantsh | [TK]D-Fender: ok ok no need to get angry man :) |
| 18:47.16 | [TK]D-Fender | irule: its in the sample file... go read it again |
| 18:47.42 | irule | ok :s |
| 18:47.54 | DEac- | awk: use a better format is a tipp of me. mp3 isn't for speakings |
| 18:48.05 | [TK]D-Fender | awk: * can't save in it natively. You can call an external process to do the conversion if you want. Go read the sample to see where ythe hooks are. |
| 18:48.33 | [TK]D-Fender | awk: given how small the wav's are, there's no point to mp3 anyways. |
| 18:48.35 | awk | [TK]D-Fender I allready said I can pass it for conversion, that isn't what i want to do.. |
| 18:48.46 | Mercestes | yay, thank you russelb! :) |
| 18:48.47 | awk | DEac- I beg to differ |
| 18:48.47 | [TK]D-Fender | awk: TFB, there is no native way. |
| 18:49.03 | awk | [TK]D-Fender so i'm stuck with wav or gsm? |
| 18:49.24 | [TK]D-Fender | awk: You're struck with any format * can ENCODE in. |
| 18:49.40 | awk | its not small enough for dial up users |
| 18:50.08 | awk | never mind, i'll script it to do the conversion for me.. thanks for the reply |
| 18:50.16 | [TK]D-Fender | awk: You think MP3 is smaller? What is smaller than GSM? |
| 18:50.28 | [TK]D-Fender | awk: thats insane |
| 18:50.47 | awk | gsm is small yes |
| 18:51.27 | awk | then it requires further coding to my cl;ients application. its fine.., i'll work it out 1 way or another |
| 18:51.40 | DEac- | speex can be the smallest |
| 18:51.41 | rantsh | [tk]D-Fender: you wanted the output of core set debug 10 or so right? |
| 18:52.28 | [TK]D-Fender | 226K = 140sec VM. @ 3.3K/s (c/mon here...) thats 68 sec to DL. 2/1 ratio! |
| 18:52.43 | [TK]D-Fender | rantsh: and "sip show channels", etc. |
| 18:52.57 | awk | [TK]D-Fender that is small |
| 18:52.59 | [TK]D-Fender | awk: "further coding"? |
| 18:53.06 | rantsh | [tk]D-Fender: well that one I did too... |
| 18:53.09 | [TK]D-Fender | awk: thats in WAV perfectly playable. |
| 18:53.13 | *** join/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 18:53.13 | awk | guess i never did my research properly |
| 18:53.22 | *** join/#asterisk racker2 (n=matias@mail.rack2.com.ar) |
| 18:53.23 | [TK]D-Fender | awk: I wasn't even talking GSM |
| 18:53.24 | cheshair | hi *boys!! :-) |
| 18:53.28 | [TK]D-Fender | awk: Evidently |
| 18:53.30 | rantsh | [TK]D-Fender: http://pastebin.ca/592361 |
| 18:53.41 | [TK]D-Fender | awk: *I* just wasted my time finding this out NOW. |
| 18:53.53 | awk | :) |
| 18:54.10 | racker2 | i have a problem to compile openh323 :( |
| 18:54.26 | awk | ignorance is bliss *shrugs* :) |
| 18:54.47 | Mercestes | racker2, oh, I know what your problem is...your trying to compile openh323. rm all instances of h323 from your system and you should be fine. |
| 18:55.39 | [TK]D-Fender | rantsh: Now host 123456.wav for us to hear. |
| 18:56.31 | [TK]D-Fender | awk: at 1.6K/s they could realistically STREAM it on dialup. |
| 18:56.57 | racker2 | no, i rimember yet when i run ./configure i see this erros checking for C++ compiler default output... configure: error: C++ compiler cannot create executables |
| 18:57.16 | rantsh | [tk]d-fender: ok I'll make a comprensible recording, I'll do some more pastebin-ing and host the audio... give me a sec please |
| 18:58.01 | awk | [TK]D-Fender hehe, true.. |
| 18:59.17 | J4k3 | oh my fucking god |
| 18:59.25 | J4k3 | grandstream sucks with no fucking limit to the vacuum |
| 18:59.34 | J4k3 | THEY NEED TO DIE. |
| 19:00.07 | rantsh | [tk]d-fender: http://pastebin.ca/592376 |
| 19:00.15 | J4k3 | are there any sip phones that will actually REPORT hold status. |
| 19:00.19 | rantsh | [tk]d-fender: now I'll try and host the recording |
| 19:00.23 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:00.46 | rantsh | if I figure out how :s |
| 19:00.50 | J4k3 | this piece of shit... I can't tell when a call is on hold or not |
| 19:00.52 | [TK]D-Fender | J4k3: as in? |
| 19:01.11 | [TK]D-Fender | J4k3: Ah... |
| 19:01.13 | [TK]D-Fender | ~gs |
| 19:01.14 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:01.15 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 19:01.30 | J4k3 | the phones work ok |
| 19:01.33 | J4k3 | but the designers and programmers |
| 19:01.37 | J4k3 | need to be taken out back and shot |
| 19:01.37 | [TK]D-Fender | J4k3: Polycom > All. You should have haerd that a few dozen times already |
| 19:01.54 | J4k3 | [TK]D-Fender: do I have to jump through hoops to configure the phones? |
| 19:02.01 | [TK]D-Fender | J4k3: GS = Garbage |
| 19:02.10 | J4k3 | (polycom) |
| 19:02.18 | J4k3 | I'm replacing these phones... I always planned to |
| 19:02.24 | watchy | J4k3: get the inital setup for polycoms aint so easy but once thats does every new phone is almost plug and play |
| 19:02.26 | J4k3 | but I never realized how bad they SUCK |
| 19:02.57 | kiscokid | I just got a Grandstream 2000xp and I have problems with it trying to enter voicemail passwords and going through the v-mail tree |
| 19:03.01 | J4k3 | considering I need a whole grand total of 2 'good' phones (I'll put a GS in my bedroom or whatever) |
| 19:03.08 | J4k3 | thats tough |
| 19:03.20 | J4k3 | I was looking at cisco |
| 19:03.28 | watchy | i don't like cisco personally |
| 19:03.40 | kiscokid | SPA/Linksys? |
| 19:03.42 | lirakis | i dislike cisco as well |
| 19:03.43 | watchy | i got 2 7940s and they just dont impres me |
| 19:03.45 | J4k3 | me either... they're the 'easy answer' in most cases |
| 19:03.58 | lirakis | people always give me crap for it.. but i really like the Grandstream GXP-2000's |
| 19:04.11 | lirakis | Im really looking foward to picking up my GXP-2020 |
| 19:04.23 | watchy | i love my polycom 501s and 601s |
| 19:04.30 | kiscokid | Lirakis: do you have any problems with it? |
| 19:04.39 | [TK]D-Fender | J4k3: Seriously Polcyom devalidates jsut about everything out there. |
| 19:04.48 | lirakis | no I have no issues with any graand stream products |
| 19:04.58 | J4k3 | lirakis: GS "works" but GS doesn't work well. |
| 19:04.58 | lirakis | .. if you are looking for enterprise.. go ahead get a polycom |
| 19:05.13 | kiscokid | lirakis: are you using it with Asterisk? |
| 19:05.15 | J4k3 | if "better than grandstream" is enterprise, I need the presidential package :) |
| 19:05.28 | lirakis | J4k3: i disagree. The only thing I dont like about GS is thier speakerphones sucked until the BT200 and GXP2020 |
| 19:05.35 | J4k3 | lirakis: the bigger problem is the crappyness of the device. |
| 19:05.43 | lirakis | kiscokid: i do use it with asterisk |
| 19:05.47 | [TK]D-Fender | J4k3: IP 320 is good enough for almost everyone, and at $95 we're not talking a lot of money |
| 19:05.48 | rantsh | [tk]d-fender: I think my irc client (Xchat on linux) doesn't support this... either that or I'm more stupid than I thought |
| 19:05.56 | lirakis | J4k3: people say that.. i dont know where they are coming from |
| 19:05.58 | J4k3 | [TK]D-Fender: yeah... $95 isn't bad |
| 19:06.00 | [TK]D-Fender | rantsh: IMPOSSIBLE |
| 19:06.01 | lirakis | J4k3: ive never had an issue |
| 19:06.06 | J4k3 | I'm saving $300/mo easily by switching to voip |
| 19:06.17 | J4k3 | lirakis: how about an "on hold" light. |
| 19:06.18 | [TK]D-Fender | J4k3: OUCH, how the hell? |
| 19:06.23 | kiscokid | lirakis: can you send me the config you use? |
| 19:06.28 | lirakis | J4k3: and as far as simplicity.. and ease of use.. they are infinitely better than cisco/linksys/polycom |
| 19:06.33 | [TK]D-Fender | J4k3: on All Polycom's, covered |
| 19:06.44 | J4k3 | [TK]D-Fender: $1.50/mo DIDs to replace a bunch of $30/mo/per-call (2 at least per jump) remote call forwarding. |
| 19:06.57 | lirakis | J4k3: i think people got a bad taste from GS "back in the day" .. but they have improoved significantly in the past 2 years |
| 19:07.01 | [TK]D-Fender | J4k3: Ah... yeah, funky needs jack it up |
| 19:07.10 | J4k3 | lirakis: I wouldn't call this bt101 'easy to use' |
| 19:07.15 | J4k3 | easy to configure, yes |
| 19:07.25 | watchy | i had a bt101 |
| 19:07.26 | J4k3 | easy to use.. it makes me want to throw it across the room |
| 19:07.26 | [TK]D-Fender | J4k3: IP 320 is right up your ally and you get a top notch phone. |
| 19:07.35 | watchy | its in my trash can in the kitchen currently |
| 19:07.36 | lirakis | J4k3: thats a $40 phone.. i mean.. you want super cheap.. you get super cheap. I have 2 bt-101's |
| 19:07.36 | J4k3 | [TK]D-Fender: sweet... I'll get a couple and give them a go |
| 19:07.46 | lirakis | J4k3: they work fine.. they configure quickly.. |
| 19:07.47 | J4k3 | watchy: take that shit out before it stinks! |
| 19:07.53 | watchy | haha |
| 19:08.01 | [TK]D-Fender | J4k3: Yes there can be a speed bump in the initial config, but I wouldn't by a Lada just because I'd have to drive down 2 more streets for a Mercedes. |
| 19:08.04 | lirakis | J4k3: .. and its just a phone.. you dont get a bt-101 for anything except to pick up and make a phone call |
| 19:08.08 | watchy | my house keeper threw it away thinking it was an old ass phone |
| 19:08.27 | J4k3 | lirakis: yeah... thats what we bought them for... just a 'proof of concept' that ended up working so well we went live with it before I got some real phones. |
| 19:08.32 | [TK]D-Fender | watchy: BarbieTone strikes again! |
| 19:08.36 | J4k3 | normally I don't answer the phone... I hate telephones |
| 19:08.43 | watchy | then it got hamburger helper dumped on it. so i decided it wasnt worth fishing out of the trash |
| 19:08.45 | J4k3 | I did this afternoon and OMG that phone pissed me off. |
| 19:08.49 | lirakis | J4k3: i would not recomend bt-101's for anything but simple home use |
| 19:09.10 | lirakis | J4k3: gxp-2000 .. absolutely a high quality extremely high value phone |
| 19:09.13 | J4k3 | lirakis: yeah. they'll work great in bedrooms. |
| 19:09.24 | J4k3 | if someone calls me in my bedroom, they're most likely to get hung up on anyways |
| 19:09.27 | [TK]D-Fender | lirakis: I would SERIOUSLY perfec even an ATA over GS. |
| 19:09.29 | J4k3 | :D |
| 19:09.52 | [TK]D-Fender | ATA's are entirely viable. |
| 19:09.54 | watchy | i just got a email " Earline Greenberg ?Premature Ejaculation – Learn How To Cure It Now " anyone want it forwarded to them? |
| 19:09.58 | lirakis | J4k3: exactly.. i gave one to my GF when she was in college.. she spilled stuff on it.. dropped it on the ground lots of times.. and it still works fine. |
| 19:10.04 | watchy | i'll hook you up |
| 19:10.19 | lirakis | [TK]D-Fender: .. i generally dislike ATA's |
| 19:10.21 | J4k3 | lirakis: the BT101 in the other room had an entire cup of coffee poured through it |
| 19:10.28 | J4k3 | didn't do the speakerphone any more favors :) |
| 19:10.28 | watchy | J4k3: haha |
| 19:10.40 | J4k3 | opened it up |
| 19:10.41 | watchy | you seem to abuse your phones |
| 19:10.45 | J4k3 | realzied that the board was like.... 2"x2" |
| 19:10.50 | J4k3 | and the phone is one big empty box. |
| 19:11.07 | adeeln | i'm having trouble compiling app_rxfax & app_txfax...any pointers or guides someone can point me to? i think the one i'm using is outdated |
| 19:11.08 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 19:11.20 | [TK]D-Fender | lirakis: At least I can plug a decent phone onto it and even go cordless. |
| 19:11.26 | J4k3 | watchy: haha... nah. I don't abuse stuff. other people here do :) |
| 19:11.33 | lirakis | [TK]D-Fender: absolutely .. for cordless |
| 19:11.43 | lirakis | [TK]D-Fender: there are no quality cordless voip phones that i know of |
| 19:12.00 | watchy | lir: i agree. I have a linksys its a piece of shit |
| 19:12.07 | *** join/#asterisk bmg505 (n=leon@196.209.179.191) |
| 19:12.17 | J4k3 | there are |
| 19:12.22 | J4k3 | but it requires buying a cellphone with wifi |
| 19:12.29 | J4k3 | so you can get a REAL cpu that can handle the job |
| 19:12.41 | J4k3 | all the "sip wifi phones" out there have *shit* processors that can't keep up. |
| 19:12.45 | watchy | can a cingular 8125 do sip wifi? |
| 19:12.57 | J4k3 | watchy: does it have wifi and run a semi-modern OS? |
| 19:13.06 | J4k3 | watchy: if so, its about a 95% chance of "yes" |
| 19:13.08 | watchy | it runs winderz and it has wifi |
| 19:13.12 | J4k3 | yep |
| 19:13.19 | J4k3 | load a wm5 voip client |
| 19:13.23 | [TK]D-Fender | lirakis: Yup |
| 19:13.26 | [TK]D-Fender | ~wifisip |
| 19:13.27 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 19:13.30 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:13.39 | rantsh | [tk]d-fender: Just checked, either way it's recording until I pick up the phone |
| 19:13.42 | watchy | i'll test it tonight j4ke |
| 19:13.44 | lirakis | .. in general I think that people discount the quality of the gxp-2000 series.... i mean i have a cisco 7940 here at work.. it has a bunch of lines.. but you can only register to 1 server.. wtf is that about??? GXP-2000 is just a great valued phone. And i suspect the gxp2020 will have fixed the speaker phone issues... which .. would be the ONLY flaw against them in my book |
| 19:13.48 | J4k3 | I had a friend bring over a nokia e60, worked great |
| 19:14.03 | J4k3 | couldn't get a fuckin GSM signal here, but the wifi worked great ;) |
| 19:14.05 | lirakis | yeah .. wifi phones are novelty items |
| 19:14.07 | watchy | im heading to work. i just took a shower and i'm about 6 hours late |
| 19:14.12 | J4k3 | (the e60 doesn't have 850 GSM) |
| 19:14.16 | rantsh | [tk]d-fender: so it's just a couple or rings "doooooooo ... doooooooo" and then it cuts |
| 19:14.19 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:14.20 | lirakis | watchy: hope you still have a job |
| 19:14.33 | watchy | lirakis: yea these bitches cant fire me |
| 19:14.37 | rantsh | [tk]d-fender: (as you can notice I'm trying to figure out dcc commands still) |
| 19:14.39 | J4k3 | lirakis: only because the manufacturers are stupid... Senao should be spitting out sexy IP PBX phones like mad. |
| 19:14.39 | watchy | they dunno what linux even is |
| 19:14.42 | J4k3 | but they're not. |
| 19:14.43 | [TK]D-Fender | lirakis: 7940 has a crappy SIP implementation, no presence and costs way too much. Polycom offers a lot more features at a much lower cost with GOOD quality manufacturing, and call handling that is unmatched. |
| 19:14.44 | lirakis | watchy: lol |
| 19:14.52 | DEac- | ah, i've a problem with a cisco 7970. it should know sip, but it can't i read, that i need to update the firmware, but i haven't a callmanager |
| 19:14.54 | vlt | Hello. I have this extension: "555,1,Dial(Zap/g1/555,10,tT) | 555,2,Dial(Zap/g2/555,10,tT)". When it rings g1 I get "Zap/1-1 is ringing" on *CLI and the caller hears a ringing tone. After the timeout it calls g2, I get "Zap/4-1 is ringing" but don't hear the ringing tone anymore. Any idea why? |
| 19:14.59 | ez` | there is a way to assign unused softkey of my polycom ip 300 to something else , example : tranfert key ... ??? |
| 19:15.04 | watchy | lir: i run a wireless isp and i'm the only guy who has access to any of the systems or any clue |
| 19:15.09 | [TK]D-Fender | rantsh: I did NOT say DCC it to me. HOST IT |
| 19:15.16 | J4k3 | watchy: you sound as bad as me! I don't wander into work til noon |
| 19:15.21 | J4k3 | not like I'm useful for anything anyways |
| 19:15.24 | J4k3 | I'm paid to IRC! |
| 19:15.28 | watchy | same here |
| 19:15.48 | lirakis | [TK]D-Fender: .. yeah .. i agree .. polycom is better than cisco/linksys/sipura.. and .. i think they are "nicer" phones that grandstreams... but configureing them sucks balls... and for the price/feature difference.. unless you are going enterprise.. i pick grandstream |
| 19:15.50 | watchy | before i leave someone be kind and gimme the newest polycom sip fw. so i dont gotta call voip supply |
| 19:16.06 | rantsh | I know but this is not mIRC, I have an option to set up a download directory but not an upload one |
| 19:16.16 | rantsh | I'm looking for another irc client now |
| 19:16.17 | DEac- | what's with snom? |
| 19:16.17 | [TK]D-Fender | ez`: You HAVE a transfer soft-key while in a call... |
| 19:16.35 | [TK]D-Fender | lirakis: Personally it take me about 1 minute to configure a Polycom.... |
| 19:16.45 | lirakis | rantsh: try pidgin... the new gaim .. it has irc, aim, google chat, yahoo, and msn .. all in one client |
| 19:16.57 | ez` | [TK]D-Fender, i know ut not visible; people asking to see this key always ... |
| 19:17.02 | [TK]D-Fender | lirakis: Unbox, assemble, 1 minute to configure, WALK AWAY. DONE |
| 19:17.05 | ez` | ut= but |
| 19:17.10 | rantsh | lirakis: thanks I'll do that |
| 19:17.20 | lirakis | [TK]D-Fender: they arent as bad as the linksys BS.. where you have to tftp an xml file etc. |
| 19:17.21 | [TK]D-Fender | ez`: there is no point isn seeing it when you can't USE it. |
| 19:17.27 | *** part/#asterisk adeeln (i=adeeln@c-67-161-185-121.hsd1.ca.comcast.net) |
| 19:17.45 | [TK]D-Fender | lirakis: You can provision Linksys via HTTP as well... |
| 19:17.58 | [TK]D-Fender | lirakis: They aren't bad to configure, they are just 2nd rate. |
| 19:18.05 | lirakis | [TK]D-Fender: .. i assume youve seen the web interface... ?? its terrible |
| 19:18.16 | [TK]D-Fender | lirakis: I suggest Linksys in countries where Polycom's import pricing is nasty |
| 19:18.18 | ez` | [TK]D-Fender, while talking we dont see the key; we have to push [more] then [trasnfert] ... thats why i am asking it |
| 19:18.26 | [TK]D-Fender | lirakis: Yeah... only MORONS use it ;) |
| 19:18.35 | rantsh | I'll leave now so I can go to pidgin brb [tk]d-fender |
| 19:18.55 | [TK]D-Fender | ez`: Well.. you always see it, its just not on the FIRST page. And NO, you can't rearrange them. |
| 19:19.28 | [TK]D-Fender | lirakis: People who try the shortcut to everything and use Polycom's Web interface SHOULD be dragged out and shot. |
| 19:19.36 | ez` | [TK]D-Fender, i agree they are so lazy , but there is really no way to assign it ????? |
| 19:19.38 | lirakis | [TK]D-Fender: .. yeah as i have said.. i am willing to concede polycom makes a better product.. but .. as i have said.. unless you are going into a business environment.. i think the GXP-2000 and now.. GXP-2020 have a much better feature/value ratio.. thats all |
| 19:20.14 | lirakis | [TK]D-Fender: right and if you are provisioning large sets of clients.. it will be via tftp anyway |
| 19:21.37 | [TK]D-Fender | lirakis: Well lets compare the GXP-2000 & IP 320 for typical use : Polycom has better call handling, better audio quality, better physical feel, more stable firmware, PoE, Speakerphone, *2* kinds of headset jacks, for starters. |
| 19:21.42 | De_Mon | I can't figure out why this queue doesn't time out when specified |
| 19:21.43 | De_Mon | http://pastebin.ca/592424 |
| 19:21.54 | [TK]D-Fender | lirakis: And thats comparing about $75 for a GXP, and $95 for an IP 320. |
| 19:22.45 | mocker | Woo. |
| 19:23.00 | mocker | Jungle Disk is awesome. |
| 19:23.03 | lirakis | [TK]D-Fender: the 320 is a two line phone.. single server capable .. etc. |
| 19:23.07 | mocker | (not asterisk related at all) |
| 19:23.16 | lirakis | [TK]D-Fender: you are paying more for less |
| 19:23.26 | [TK]D-Fender | lirakis: No, 2-line, 4 calls max, and supports MULTIPLE servers. |
| 19:24.06 | [TK]D-Fender | lirakis: See big thing is that Polycom supports multiple calls PER LIN-KEY. |
| 19:24.25 | [TK]D-Fender | lirakis: My IP 301 supports a minimum of 10 calls :) |
| 19:24.40 | lirakis | [TK]D-Fender: so does grandstream... you can have up to 11 calls on a gxp-2000 |
| 19:24.48 | [TK]D-Fender | lirakis: When I reg'd my 501 to multiple clients I had 5 calls each on 3 distinct reg's |
| 19:25.20 | [TK]D-Fender | lirakis: Not compare EITHER of these phones # of calls supported, and BOTH cover business needs. |
| 19:25.22 | [TK]D-Fender | now* |
| 19:25.43 | [TK]D-Fender | lirakis: So if the GXP wins on sheer volume alone, all those other features still stack against it. |
| 19:25.47 | lirakis | [TK]D-Fender: .. plus the grandstream has all the programable buttons.. supports asterisk blf.. .etc. |
| 19:25.49 | watchy | i wouldnt implement anything but a polycom personally |
| 19:25.58 | [TK]D-Fender | lirakis: and that is niether a small list nor a petty one. |
| 19:26.28 | watchy | does the grandstream support precense? |
| 19:26.34 | [TK]D-Fender | lirakis: And Polycom Supports BLF as well. (not as many IMMEDIATE lit indicators for sure. But then we're talking receptionist type stuff anyways. |
| 19:26.40 | [TK]D-Fender | watchy: It does. |
| 19:26.54 | watchy | wtf is BLF? |
| 19:26.58 | [TK]D-Fender | lirakis: But Polycom's BLF has been solid for years |
| 19:27.07 | [TK]D-Fender | watchy: BLF = Presence |
| 19:27.10 | [TK]D-Fender | ~blf |
| 19:27.11 | jbot | somebody said blf was Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 19:27.14 | watchy | oh |
| 19:27.34 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:27.56 | watchy | yea i got 5 601s at a business with 2 sidecars each. BLF is very useful for them |
| 19:28.05 | cheshair | hi guys, could you please tell me what's the best way to make the very first steps with asterisk? more precisely: i'm reading the first one hundred pages of *toft.pdf and having some tests. the point is i have no appropriate hardware and so i'm having my tests using a sip channel from a softphone on a localhost * installation. does it make sense? |
| 19:28.25 | [TK]D-Fender | De_Mon: agent dial timeout + Queue Timeout = time your call lasted |
| 19:29.09 | [TK]D-Fender | cheshair: Set up a minimal dialplan and your soft phone. Do Echo tests, voicemail, etc./ Install another soft phone so you can at least call FROM one to the other. |
| 19:29.25 | lirakis | [TK]D-Fender: .. i wasnt really talking about meeting "business needs" .. i was talking about the quantity and variety of features the GXP-2000 supports... Again.. i think that if you took a polycom with the "same" on paper feature set.. it would cost a lot more than a gxp2000. Sure there is other stuff to consider.. but for my (and i believe most ) purposes outside of enterprise deployment... it is a great phone and a great value |
| 19:30.16 | cheshair | [TK]D-Fender: perfect, that's what i'll do. thanks!! |
| 19:30.17 | [TK]D-Fender | lirakis: Actually... its pretty mucht he same feature-set. Just more total calls on the GXP, but then again you don't have the JOIN/SPLIt that polycom does for handling.. |
| 19:30.49 | J4k3 | there are voip 'end users' that aren't just freakish vonage customers?! :D |
| 19:30.51 | [TK]D-Fender | lirakis: List the actual FEATURES the GXP has over Polycom and we'll see. |
| 19:31.07 | [TK]D-Fender | J4k3: Tons |
| 19:31.08 | *** join/#asterisk ibob63 (n=james@dsl-217-155-69-86.zen.co.uk) |
| 19:31.16 | [TK]D-Fender | J4k3: .... of BOTH ;) |
| 19:31.19 | De_Mon | [TK]D-Fender I have a queue that works correctly with the same timeout value (dialplan timeout is 5seconds) |
| 19:31.47 | [TK]D-Fender | De_Mon: I would set your agent time < queue time... |
| 19:31.52 | De_Mon | [TK]D-Fender the only difference I noticed is that the working queue does not have retry=40. commented out that line and it works as expected now |
| 19:32.00 | [TK]D-Fender | J4k3: Why not? |
| 19:32.05 | J4k3 | Child A: "I want a phone in my room" <begin construction project> |
| 19:32.14 | J4k3 | [TK]D-Fender: can't even get a real cordless phone yet? :) |
| 19:32.16 | [TK]D-Fender | De_Mon: Ah |
| 19:32.22 | [TK]D-Fender | J4k3: huh?! |
| 19:32.33 | J4k3 | most people aren't willing to do the plug-this-box-into-that-box-then-plug-into-another-box-and-hope-it-all-works |
| 19:32.40 | De_Mon | so it's waiting the retry length before hanging up (realizing it was supposed to hang up after 20 seconds? |
| 19:33.04 | [TK]D-Fender | J4k3: Ask yourself how clueless you'd have to be to implement this. Homes don't have PBX's! |
| 19:33.07 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 19:33.17 | rob0 | A VoIP home is indeed a highly geekful project, not suited for the masses unless they go with Vonage. |
| 19:33.23 | ibob63 | hi guys. I am just about to attempt to install an asterisk system using their B410P quad bri isdn card. Has anyone had any experience with this? |
| 19:33.26 | [TK]D-Fender | J4k3: As for "VoIP", well yeah, thats what vonage is for! Idiots who only need a preconfigured ATA |
| 19:33.29 | J4k3 | [TK]D-Fender: correct... but one can buy a few thousand minutes for the price of one residential pots line. |
| 19:33.45 | lirakis | sorry .. phone call |
| 19:33.54 | J4k3 | (residential lines are up to about $20/mo after taxes here) |
| 19:33.55 | lirakis | .. some times i actually have to use the phone lol |
| 19:34.03 | De_Mon | J4k3 and when your power goes out you gotta use the cell phone to report it ;P |
| 19:34.25 | J4k3 | De_Mon: I know enough people with that problem on POTS |
| 19:34.34 | [TK]D-Fender | J4k3: Cheaper than here actually. but yeah VoIP termination is cheaper on LD per volume often. Depends WHERE, and how long |
| 19:34.35 | De_Mon | J4k3 eh? |
| 19:34.39 | J4k3 | due to not owning a real line-powered phone... cordless this, answering machine that. |
| 19:34.59 | [TK]D-Fender | De_Mon: I run SIP only at home.... I'm fine with Cell as last resort... |
| 19:35.00 | rob0 | Oh yeah, POTS is having trouble competing. I had fun telling SBC / AT&T when I got new service in May. :) |
| 19:35.03 | De_Mon | well thats just being stupid, nothing in the world can fix that |
| 19:35.05 | J4k3 | [TK]D-Fender: "universally cheaper than pstn" :) |
| 19:35.33 | *** join/#asterisk techie (n=gus@antibala.net) |
| 19:35.38 | [TK]D-Fender | J4k3: Not if all calls are your daughter talking for HOURS with her best friend and noone else needing the phone ;) |
| 19:35.41 | J4k3 | our "small business/home" voip falls over to cellular in events that suck |
| 19:35.43 | J4k3 | of course |
| 19:35.52 | J4k3 | usually, if we're down the cellular is too due to it being fiber related. |
| 19:36.35 | lirakis | [TK]D-Fender: .. ditto .. i have sip only.. and a blackberry that is backup |
| 19:36.37 | J4k3 | [TK]D-Fender: thats why kids all have cingular now, it seems. |
| 19:36.43 | J4k3 | unlimited m2m ;0 |
| 19:36.56 | [TK]D-Fender | J4k3: m2m? |
| 19:37.00 | J4k3 | mobile to mobile |
| 19:37.03 | [TK]D-Fender | J4k3: ah |
| 19:37.10 | [TK]D-Fender | J4k3: ABUSABLE ;) |
| 19:37.14 | lirakis | i need to setup e911 though.. :\ .. i havent done that yet.. |
| 19:37.35 | J4k3 | [TK]D-Fender: yeah... if the bluetooth hookup stuff wasn't such a hack it'd be VERY abusable. |
| 19:37.36 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:37.47 | rob0 | exten => 911,1,Playback(tt-monkeysintro) |
| 19:37.57 | lirakis | lol |
| 19:37.59 | J4k3 | "screw buying extensions, I'll use the cellular network!" |
| 19:38.03 | lirakis | yeah.. not so good |
| 19:38.27 | Ryushin | I remember that there was a new asteriskgui by digium a few month back. I cannnot find it now. Any idea on where I can try this out? |
| 19:38.55 | cheshair | [TK]D-Fender: which are the very first pieces of hardware i'll need to have a more serious * box? maybe i'll need some sort of zaptel cards, won't i? |
| 19:39.00 | rob0 | Ryushin: www.asterisk.org has a link! |
| 19:39.27 | cheshair | [TK]D-Fender: just to have an idea about what i shall have to buy |
| 19:39.41 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:40.27 | Ryushin | rob0: The only link I've been seeing is for asterisknow which does include the gui, but I don't wish to use a dedicated asterisk distro. |
| 19:40.42 | rob0 | oh, doesn't it include the gui? |
| 19:40.48 | rob0 | I didn't know |
| 19:41.01 | rob0 | DOES ... sorry, misread |
| 19:41.11 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 19:41.46 | Ryushin | I remember reading that the gui was for 1.4, and now that I'm on 1.4, I wanted to try it out. |
| 19:42.15 | *** join/#asterisk Dantix (n=Dante@200.68.70.100) |
| 19:42.38 | rob0 | SwK, Chicago? You're not still in HSV? |
| 19:43.26 | J4k3 | ooey gui |
| 19:44.24 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 19:44.27 | Dantix | hi, have an openvox a1200 card. It's detecting ringing tone on its fxo port but the line idle (nobody is calling it), how can I fix it? |
| 19:44.48 | Qwell[] | ~ygwypf |
| 19:44.49 | jbot | i guess ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 19:44.56 | Qwell[] | Dantix: ^^^ |
| 19:45.21 | Dantix | Qwell[]: what do you mean? |
| 19:45.25 | *** join/#asterisk sashion (n=sdgsdg@dsl-242-91-154.telkomadsl.co.za) |
| 19:45.42 | *** join/#asterisk troy- (n=troy@DCC.SEND.startkeylogger.000.telephreak.org) |
| 19:45.58 | troy- | i am getting the warning message: [Jun 27 15:15:59] WARNING[11232]: chan_iax2.c:3792 iax2_send: No private structure for packet? |
| 19:46.05 | troy- | any idea what it means or how to fix it? |
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| 19:48.49 | [TK]D-Fender | cheshair: Depends what you want to do. |
| 19:50.37 | *** join/#asterisk Zion800 (i=Glitter1@guest-wireless-207-151-252-033.usc.edu) |
| 19:51.39 | Ryushin | I guess if you want to use the gui, you have to use asterisknow. So I guess I won't be trying it out. |
| 19:51.42 | sashion | Is there anyway to reject a call from PRI even when the extension exists ? |
| 19:52.17 | cheshair | [TK]D-Fender: i see... well at first i'll have a lot of tests with softphones and localhost * installation, then i think i'll buy some pieces of hw to set up some more realistic scenarios... e.g. something which will allow me to use * at home, on my analog line... |
| 19:53.19 | [TK]D-Fender | cheshair: several options. Learn the basics for free and we can talk hardware after |
| 19:53.44 | [TK]D-Fender | sashion: Just "congestion" it |
| 19:54.01 | cheshair | [TK]D-Fender: that's a good roadmap, ok! :-) |
| 19:54.30 | troy- | does nvfaxdetect work well? |
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| 19:55.49 | *** join/#asterisk javb (n=javb@190.80.236.79) |
| 19:56.04 | javb | made some changes to zapata.conf, how to i apply them? |
| 19:56.30 | [TK]D-Fender | javb: "reload chan_zap.so |
| 19:56.44 | javb | from asterisk console? |
| 19:57.08 | javb | ok. |
| 19:57.37 | Zion800 | I'm having a problem with ChanIsAvail in Asterisk 1.4.4. For some reason, no matter what, the AvailStatus is always 0 (Unknown). In Asterisk 1.2.18 it worked perfectly. Is this a bug? |
| 19:57.51 | [TK]D-Fender | Zion800: Pastebin in-call proof |
| 19:58.00 | Zion800 | ok |
| 19:58.03 | sashion | [TK]D-Fender: hmmm and if the Avaya system doesn't understand that in the vector programming? |
| 19:58.03 | javb | I have this problem.. i can pick up call comming from another exten, but CANT coming from a zap channel, mean with pick up (USING COMMAND PICKUP) |
| 19:58.13 | *** part/#asterisk Dantix (n=Dante@200.68.70.100) |
| 19:58.13 | javb | to pick up an exten ringing on another exten |
| 19:58.34 | javb | have options correctly set in zapata, but still cant. |
| 19:58.36 | [TK]D-Fender | sashion: Dunno.... |
| 19:58.45 | javb | my grandstream says "603" error. |
| 19:58.51 | javb | NEED HELP for real. |
| 19:58.57 | javb | My card is TDM400P |
| 19:58.58 | [TK]D-Fender | sashion: * was not the smartest at LOOKING at calls without giving SOME sort of "trying" result back. |
| 19:59.35 | Zion800 | [TK]D-Fender: http://pastebin.ca/592492 |
| 20:00.12 | Zion800 | [TK]D-Fender: You can see I NoOp the Avail Status |
| 20:00.29 | sashion | [TK]D-Fender: hmmm now we don't want to admit the fact that asterisk isn't smart. |
| 20:00.34 | sashion | that would spoil the plan |
| 20:01.11 | [TK]D-Fender | Zion800: Ah yes, THAT... well for one thing you aren't testing is the phone is on ANY calls (missing a parm for that), you also need to set "call-limit=[somehealthy big number here]" for your SIP devices. |
| 20:01.18 | [TK]D-Fender | Zion800: Apply it and you should be happy |
| 20:01.20 | lirakis | okay guys... im done. ttyl... thanks for earlier [TK]D-Fender |
| 20:01.34 | [TK]D-Fender | lirakis: I'll wait for your feedback after :) |
| 20:01.41 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 20:01.42 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:01.44 | javb | :s |
| 20:01.46 | [TK]D-Fender | lirakis: For that list of what the GS has that Polycom DOESN'T |
| 20:02.08 | Zion800 | [TK]D-Fender: Thanks! I'll try setting the call limit |
| 20:03.29 | Zion800 | [TK]D-Fender: I had actually already set a call-limit=100 for each device, and even in the general section of sip.conf... |
| 20:03.40 | sashion | [TK]D-Fender: Should one answer a channel before sending a Busy or Congestion? Or should one rather not answer the channel |
| 20:04.16 | javb | any idea guys? |
| 20:04.21 | *** join/#asterisk friedrich| (n=friedric@e177253231.adsl.alicedsl.de) |
| 20:05.03 | [TK]D-Fender | sashion: No answer first |
| 20:05.23 | Zion800 | [TK]D-Fender: How do I set a parm for checking if a phone is on any calls? (i dont even know what a PARM is..) |
| 20:05.28 | vlt | Hello. Any idea why I don't hear the ringing tone anymore when priority changes from "exten =>555,1,Dial(Zap/g1/555,10,tT)" to "exten => 555,2,Dial(Zap/g2/555,10,tT)"? I get "Zap/... is ringing" on *CLI everytime. |
| 20:07.07 | *** join/#asterisk anthony] (n=anthony@212.201.189.72.cfl.res.rr.com) |
| 20:07.10 | [TK]D-Fender | Zion800: "show application zhanisavail" |
| 20:07.14 | [TK]D-Fender | Zion800: "show application chanisavail" |
| 20:08.51 | Zion800 | [TK]D-Fender: Ah...so I need to set the 's' option? |
| 20:10.59 | [TK]D-Fender | Zion800: Yo begin to learn... |
| 20:14.56 | Zion800 | [TK]D-Fender: haha...i looked at it 100 times...but didnt realize the little tidbit of information :-) Just out of curiosity...did I set up the call-limit corectly? Can it be put in the general section of sip.conf? or does it have to go under each extension? |
| 20:15.12 | [TK]D-Fender | Zion800: under EACH, not up top |
| 20:15.17 | Zion800 | ok |
| 20:17.58 | sashion | Oh joys |
| 20:18.06 | sashion | q.931 for complete idiots |
| 20:18.09 | sashion | I lost my book |
| 20:24.53 | AndrewGearhart | Eradan: did you see [TK]D-Fender recommended eyebeam? |
| 20:24.56 | AndrewGearhart | [TK]D-Fender: thanks |
| 20:25.06 | AndrewGearhart | I'm outta here for the day... wife sprained her ankle... ugh. |
| 20:25.48 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 20:25.52 | sci_05 | afternoon all |
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| 20:39.22 | Mercestes | ~itsp |
| 20:39.22 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 20:41.30 | *** join/#asterisk timofonic (n=evil@unaffiliated/timofonic) |
| 20:41.34 | timofonic | Hello |
| 20:41.43 | Mercestes | hello... |
| 20:42.22 | timofonic | I want to know about the X100P thingie and winmodems for using them as FXO cards |
| 20:42.56 | timofonic | I want to connect my phone line to a PC and using it with Asterisk. Any list of compatible cards? |
| 20:44.45 | Zion800 | www.digium.com |
| 20:45.40 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 20:46.16 | timofonic | Zion800: Too expensive, even if it's the official sponsor of the Asterisk project |
| 20:46.36 | timofonic | I mean cheap winmodems, like the x100 "clones" using the Intel Ambient chips |
| 20:49.26 | SuPrSluG | timofonic:it's a crapshoot w/ those. some work ok and some are nothing but headaches. |
| 20:50.19 | Corydon76-work | Also realize how ironic it is to say that Digium gear is "too expensive" |
| 20:50.45 | Corydon76-work | Before Digium, the equivalent cards in the market cost at least 5 times as much |
| 20:50.51 | Nugget | heh |
| 20:51.08 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 20:51.21 | Corydon76-work | Some of them still do |
| 20:51.25 | Ryushin | So is the common way for rolling out night mode on polycom phones to have the main phone that rings be transferred to an extension that is set up for after hours speech, etc. |
| 20:51.51 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 20:51.53 | vlt | The problem that I don't hear the ringing tone anymore when priority changes from "exten =>555,1,Dial(Zap/g1/555,10,tT)" to "exten => 555,2,Dial(Zap/g2/555,10,tT)" only appears when calling in via SIP. IAX2 and ZAP work fine. Hmmmmm ... |
| 20:52.13 | sashion | vlt: add r after you tT |
| 20:52.34 | vlt | sashion: I'll try ... |
| 20:54.19 | vlt | sashion: Yes, that seems to work. What does it do? |
| 20:55.30 | sashion | vlt: plays rining :P |
| 20:55.41 | sashion | s/rining/ringing\g |
| 20:56.08 | Mercestes | Ryushin, I would say the common way has something to do with gotoiftime |
| 20:56.57 | Ryushin | Yea, I set that up first. But the company wants to have it happen in a manual mode. |
| 20:57.13 | Mercestes | define manual mode |
| 20:57.23 | Mercestes | they wanna push buttons? |
| 20:57.33 | vlt | sashion: Does it ANSWER the call or is it a normal "ringing state"? |
| 20:57.47 | sashion | vlt: just plays ringing to the callee |
| 20:57.51 | Ryushin | They have one phone, they put it in night mode on the phone, and while it's in this mode, asterisk will behave appropriately. |
| 20:57.55 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 20:57.55 | sashion | vlt: Doesn't answer the call |
| 20:58.22 | Ryushin | I mean, they want to set of one of their phones to be the master for a night mode method. |
| 20:58.26 | Mercestes | Ryushin, I use some System() commands calling scripts that change my dialplan and do an asterisk -rx 'extensions reload" for that. |
| 20:58.47 | Mercestes | I just modify an #include => file for that. |
| 20:59.36 | Ryushin | Are the calling scripts somehow querying a phone? Does the polycom even have a night mode configuration? |
| 20:59.47 | Mercestes | they press some buttons on the phone. |
| 21:00.08 | Mercestes | You could also code a softkey or a hardkey to call that extension to do the same thing. |
| 21:00.12 | Mercestes | and label it "night mode." |
| 21:00.30 | Mercestes | and you can even display "night mode" on the LCD if you wish. |
| 21:00.51 | Ryushin | Yea, I saw that done by Unity. I had just never seen it before. |
| 21:01.03 | Mercestes | but it would still just call an extension in asterisk which ran some shell scripts which modified an #include file to modify your dialing plan accordingly |
| 21:01.52 | stoffell_w | Mercestes, hm, do you have any idea where to find more info on coding a softkey with a label? (example, night mode ) |
| 21:02.00 | Ryushin | So I essentially have to maintain two extensions files. One for day, and one for night. |
| 21:02.34 | Ryushin | With the only difference being one include statement. |
| 21:02.35 | stoffell_w | Ryushin, or use variables.. and use the asterisk db to store values.. (like: nightmode = 0 or 1) |
| 21:02.43 | Mercestes | stoffell_w, The admin guide, honestly. It reads like the Rosetta Stone but it's in there. You can also use the existing sip.cfg as an example. |
| 21:03.05 | Mercestes | Ryushin: that's what I do. Astdb is probably a much cleaner solution but, I managed with shell scripts. |
| 21:03.12 | stoffell_w | Mercestes, okay, will have another (;-)) look into the beast ;) tnx, at least you confirm it's possible, so thanks! |
| 21:03.21 | Mercestes | np. :) |
| 21:03.34 | Ryushin | Yea, time to go digging and have more fun. :) |
| 21:03.52 | Mercestes | just about every button on the polycom can be configed to do just about anything you want (or be disabled entirely) it's mainly a matter of how to do it. |
| 21:03.55 | SuPrSluG | what causes outbound calling delay. when a call is placed the called party doesn't hear the person calling for the 1st few seconds. sometimes they hangup. happened on both zap and sip channels |
| 21:03.56 | stoffell_w | without digging it wouldn't be that much fun, now would it? ;) |
| 21:03.59 | Mercestes | that sip.cfg pretty much *is* th eprogramming for that phone. |
| 21:04.04 | Ryushin | Am I antiquated that I use text files for everything and don't use a database. :) |
| 21:04.25 | timofonic | SuPrSluG: I prefer headaches but cheap hardware, I have no bucks |
| 21:04.33 | stoffell_w | Mercestes, true. and the new way of provisioning as explaned in the latest guide is a pretty great way to do it cleanly and easily |
| 21:04.56 | SuPrSluG | timofonic:pick one up one ebay then |
| 21:05.47 | SuPrSluG | timofonic: or scavenge from a used hardware place. |
| 21:09.19 | *** part/#asterisk cheshair (n=fabio@host-78-12-100-200.cust-adsl.tiscali.it) |
| 21:13.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 21:22.23 | *** part/#asterisk clive- (n=pirch@dsl-242-179-161.telkomadsl.co.za) |
| 21:25.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:25.40 | vlt | "Accepting overlap voice call from '555' to '888' on channel 0/1, span 1", "Starting simple switch on 'Zap/1-1'". What does overlap mean here and what is simple switch? Before I changed my setup yesterday it was just "Accepting voice call from ..." |
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| 21:36.45 | shido6 | wow |
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| 21:51.04 | *** join/#asterisk rg2112 (n=rob@64.163.131.18) |
| 21:51.54 | rg2112 | hi all. anyone have experience with using digium cards to connect to Adtran 850? |
| 21:52.07 | *** join/#asterisk ruied_ (n=ruied@bl7-211-29.dsl.telepac.pt) |
| 21:52.09 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 21:52.33 | irule | * is sending email to hotmail accounts with /usr/sbin/sendmail -t, that is neat but, email server is not even installed, is it all lost, or are those email messages saved somewhere? |
| 21:53.04 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 21:53.14 | ZX81 | irule: is sendmail installed? |
| 21:53.34 | irule | nop |
| 21:53.46 | irule | should I? |
| 21:53.51 | ZX81 | yep |
| 21:53.56 | rg2112 | no mail server at all? qmail? postfix? |
| 21:54.32 | irule | none, debian installs exim as default but at installation I tell it to configure localhost with no internet options |
| 21:56.12 | russellb | you don't need a mail server on the box for this to work |
| 21:56.16 | russellb | all sendmail does is ..... send mail |
| 21:56.27 | russellb | you don't need a mail server installed on your laptop to send email from your mail client |
| 21:58.11 | holiday_42 | irule:not familiar with exim, but there should be record in logs as to the fate of the emails |
| 21:58.39 | holiday_42 | irule: such as /var/log/maillog |
| 22:00.04 | irule | Ill investigate, thanks |
| 22:00.26 | irule | I dont know exim though, so I may just replace it with sendmail ;) |
| 22:00.47 | holiday_42 | i converted from sendmail to postfix myself |
| 22:02.04 | *** join/#asterisk hi365_m (i=HydraIRC@212.199.22.88.forward.012.net.il) |
| 22:02.13 | Strom_M | i converted from sendmail to OpenFreeSendMail 2.6.15.62.7.2 because sendmail wasn't quite technically FOSSy enough for me (it violated the GPL if used in conjunction with a wet razorblade retrieved from a bathtub in Palm Springs) |
| 22:02.30 | hi365_m | does the zttest apply to other cards too (sangoma) or is it digium specific? |
| 22:04.48 | Strom_M | hi365_m: well here's a simple test |
| 22:04.55 | Strom_M | does it work with the sangoma card? |
| 22:05.07 | rob0 | Exim, like all major MTAs, includes a sendmail CLI binary for sending mail. |
| 22:05.56 | *** join/#asterisk oej (n=olle@86.85-200-224.bkkb.no) |
| 22:06.25 | hi365_m | Strom_M: its giving results, but nothing that ide like to see, so im wondering if i could just igonor the results or not |
| 22:06.56 | Strom_M | well, how far off from reality are the results, and what problem are you attempting to diagnose? |
| 22:06.59 | ruied_ | I'm trying to compile mISDN with 'make' but it reports: "make: *** /lib/modules/2.6.18-4-686/build: File or directory not found" what could be the problem? is it missing something? |
| 22:07.17 | ruied_ | I'm using debian etch |
| 22:07.22 | Strom_M | do you have the kernel headers installed, ruied_ ? |
| 22:07.46 | ruied_ | Strom_M, no, just the kernel source... |
| 22:09.16 | Strom_M | well theres your problem :) |
| 22:10.22 | ruied_ | Strom_M, thanks... going to install it... |
| 22:10.28 | ZX81 | OT: anyone know a good text editor for windows with SSH support (i.e. upload/download files) |
| 22:11.00 | holiday_42 | secureCRT, but it's not free |
| 22:11.09 | shido6 | BBedit |
| 22:11.16 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 22:11.27 | ruied_ | Strom_M, when do I need kernel headers and when do I need the kernel source? I'm a litle lost about this... can you point me some good reading stuff? |
| 22:11.44 | ZX81 | got securecrt would prefer to actually edit files here then upload - had Zeus programmers editor but licence expired |
| 22:11.49 | Strom_M | ... |
| 22:11.50 | Strom_M | why not just use putty and then use vim on whatever machine you're ssh'd into? |
| 22:12.00 | ZX81 | auto indent etc |
| 22:12.01 | ZX81 | :) |
| 22:12.08 | ZX81 | folding sections |
| 22:12.14 | galeras | is possible in zapata.conf to redefine groups for same zap channels? |
| 22:12.22 | ZX81 | anyway its OT - I'll ask google :) |
| 22:12.31 | shido6 | using group, galeras ? |
| 22:12.37 | galeras | yes |
| 22:13.05 | galeras | i mean g0->zap/1-15 g2->zap/1-15 zap/17-31 |
| 22:13.26 | shido6 | why would you want to do that? yuo can. |
| 22:13.31 | Strom_M | ruied_: you need kernel headers if you're compiling modules against the kernel. you need kernel source if you're compiling the kernel. |
| 22:13.31 | shido6 | you CAN do that |
| 22:14.16 | ZX81 | I've still got friggin red alarms on pri spans connected together with crossover cable, anyone care to read a pastebin? |
| 22:14.32 | Strom_M | ZX81: t1 crossover cable? |
| 22:15.40 | *** join/#asterisk tomlobato (n=tomlobat@201-27-209-159.dsl.telesp.net.br) |
| 22:15.43 | hi365_m | Strom_M: im having an assorment of issues - cals coming from a queue sometime "freeze" and then dissconect (the phone requires a reboot), pops and clics on ivr's, and ocasional pop and clicks during conversasions |
| 22:15.51 | ZX81 | Strom_M: E1 |
| 22:15.54 | ZX81 | made it up myself |
| 22:15.57 | ZX81 | 1,2 -> 4,5 |
| 22:16.15 | ruied_ | Strom_M, but to have the mISDN module, don't I need some kind of kernel patch so I can use the mISDN as module recognized by the kernel? this is what is confusing me.... |
| 22:16.28 | Corydon76-work | 1 to 4 and 2 to 5, right? |
| 22:16.35 | *** join/#asterisk oej (n=olle@86.85-200-224.bkkb.no) |
| 22:16.39 | ZX81 | yeo |
| 22:16.43 | galeras | shido6: i need 60 lines for predictive dialer and 30 lines for other use, both using same dial pattern |
| 22:16.43 | ZX81 | brb |
| 22:16.44 | ZX81 | digium support answered |
| 22:16.46 | ZX81 | :)( |
| 22:17.03 | ibob63 | Does anyone have experience with the quad isdn card b410p? |
| 22:17.34 | k31th | ibob63: not yet, maybe soon tho |
| 22:17.56 | k31th | Can i get commercial support from digium ? |
| 22:18.12 | Strom_M | ZX81: did you test the cable? |
| 22:18.16 | shido6 | sure you can. |
| 22:18.20 | k31th | Nice |
| 22:18.53 | Strom_M | ruied_: no, you compile it against the headers and then you load it |
| 22:18.54 | Strom_M | simple |
| 22:19.32 | ibob63 | does anyone know where I can download libtermcap ? Does anyone make .deb for this? |
| 22:19.40 | ruied_ | Strom_M, hmm, ok! going to compile it! thanks :) |
| 22:19.55 | shido6 | for your source question, checkout http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation |
| 22:19.58 | shido6 | for starters |
| 22:20.29 | Strom_M | welcome to #asterisk, where it's apparently "Linux 101" day |
| 22:21.14 | holiday_42 | :) |
| 22:21.22 | hi365_m | Strom_M: im having an assorment of issues - cals coming from a queue sometime "freeze" and then dissconect (the phone requires a reboot), pops and clics on ivr's, and ocasional pop and clicks during conversasions |
| 22:21.37 | holiday_42 | seems like everyday is "linux 101" day |
| 22:22.34 | hi365_m | Strom_M: so i decited to run zttest, thinking perhaps its a timing issue, but i really didnt like what i saw |
| 22:22.50 | Strom_M | hi365_m: well, beats me. i have no experience with the sangoma cards except remembering that the one I did set up for a client was a colossal pain in the ass to work with |
| 22:23.08 | hi365_m | gotchya. thanks anyway |
| 22:23.49 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-107-173-57.ny325.east.verizon.net) |
| 22:27.49 | *** join/#asterisk davidcsi (n=davidcsi@82.158.35.53.dyn.user.ono.com) |
| 22:28.06 | *** part/#asterisk hi365_m (i=HydraIRC@212.199.22.88.forward.012.net.il) |
| 22:28.30 | davidcsi | anyone knows how to know if call sent via PERL AGI has connected? event-like?? |
| 22:28.34 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 22:30.16 | irule | <PROTECTED> |
| 22:30.56 | Strom_M | /var/log/asterisk/cdr-csv/ |
| 22:31.19 | Strom_M | locate cdr | grep asterisk |
| 22:31.37 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:32.00 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 22:32.31 | aptura | Is it possible to play a voice mail so both parties can listen to it? |
| 22:32.46 | Strom_M | what do you mean "both parties" |
| 22:32.49 | shido6 | "both parties" ? |
| 22:33.00 | Strom_M | in a call to voicemail, there's you and there's the voicemail server |
| 22:33.06 | aptura | sip session between two people. |
| 22:33.12 | Mercestes | You could conference call into the voicemail system. |
| 22:33.22 | Mercestes | or forward the voicemail |
| 22:33.23 | shido6 | or copy the mesage itself |
| 22:33.28 | shido6 | to somewhere else |
| 22:33.29 | aptura | call brb |
| 22:33.35 | Mercestes | or email it to multiple users |
| 22:33.45 | shido6 | or play it on a website |
| 22:33.50 | shido6 | or stream it somewhere |
| 22:33.58 | shido6 | pick 12 |
| 22:34.00 | shido6 | :) |
| 22:34.19 | snuff-work | does the g729 card have a timer.. ztdummy isn't cutting it |
| 22:34.30 | shido6 | g729 card is for 1.4 |
| 22:34.34 | Strom_M | aptura: yeah, it's called THREE-WAY CALLING |
| 22:34.34 | Strom_M | it's only been around since the 1960s |
| 22:34.43 | ZX81 | patrick (DigiumSupport) is helping me fix my red alarms :) yay! |
| 22:34.53 | Strom_M | snuff-work: yes, you get timing from the TC400B |
| 22:35.08 | Qwell[] | Strom_M: You sure about that? |
| 22:35.18 | Strom_M | well...uh....I would assume so...... |
| 22:35.22 | Qwell[] | Without knowing, I would say no |
| 22:35.26 | Strom_M | i'm wrong, aren't I |
| 22:35.31 | Qwell[] | I know that the TDM400p with no modules does not |
| 22:36.16 | davidcsi | TDM400 DOES have a timing |
| 22:36.22 | Qwell[] | Yes, with modules it does. |
| 22:36.26 | Qwell[] | Without modules it does not. |
| 22:36.43 | snuff-work | mmm.. might have to put it into this machine to find out.. |
| 22:36.54 | davidcsi | there is absolutely no way of doing TDM without a timing reference |
| 22:37.05 | Qwell[] | snuff-work: If you want a board just for timing, you could get a 1 port TDM400p.. |
| 22:37.21 | davidcsi | you can get timing with usb |
| 22:37.30 | aptura | I just thought it would be a interesting feature to just get into vm and play it for the other caller I called. |
| 22:37.34 | snuff-work | but i only have a 1 ru server with 1 pci-x slot |
| 22:37.43 | Qwell[] | snuff-work: that could prove difficult then... |
| 22:37.53 | aptura | btw |
| 22:38.05 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 22:38.48 | aptura | Now this is a little odd but my ip500 is not rining when a call comes in. It will attempt to ring but with a duration of .5ms and then the call goes to vm. I have made no changes to create this behavior. |
| 22:39.28 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-150-67.dsl.ipltin.ameritech.net) |
| 22:39.45 | aptura | Guess no one has seen that issue yet. |
| 22:40.24 | shido6 | the cli should read something like, cannot create channel of type sip |
| 22:40.28 | shido6 | then move to the next priority |
| 22:40.50 | aptura | let me check |
| 22:40.55 | perf3kt | SIP/2.0 404 Not Found |
| 22:41.14 | Strom_M | SIP/2.0 808 Hawaii |
| 22:42.04 | perf3kt | that is the error that I recieve when I attempt to call into my * box |
| 22:42.26 | JT | cool |
| 22:42.48 | snuff-work | mm.. ok if i have to go usb timing to get 1/2 decent clock will be a shame :( |
| 22:43.16 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:43.54 | *** join/#asterisk seele_ (n=seele@webserver.datawareltda.com) |
| 22:46.30 | JT | snuff-work: usb timing won't be that decent either |
| 22:47.00 | JT | why is it a shame? |
| 22:47.51 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
| 22:48.51 | davidcsi | snuff-work, why do you need a timer for? |
| 22:49.41 | snuff-work | so meetme work smoother |
| 22:49.45 | snuff-work | *works |
| 22:50.27 | davidcsi | oh ok... but for that my guess is usb is just fine... unless we're talking heavy traffic here... |
| 22:50.35 | davidcsi | do you have a TDM card? |
| 22:50.36 | *** part/#asterisk ibob63 (n=james@dsl-217-155-69-86.zen.co.uk) |
| 22:52.27 | snuff-work | in one box i have a TE110P 1xE1 card.. but the other is just going to have a g729 card and the servers only support 1 pci-x because its 1ru |
| 22:53.59 | javar | someone can help with sangoma card? |
| 22:54.32 | JT | davidcsi: weird guess |
| 22:54.46 | JT | davidcsi: meetme likes zap timing best, from a real card |
| 22:55.47 | aptura | getting a no route to destination. http://www.pastebin.ca/592808 but the second error is rather interesting. |
| 22:55.50 | davidcsi | usb is a real timing device, only not as accurate as tdm, but for small applications its just fine, i'm doing 45k min/day with iax trunking using usb timing |
| 22:57.01 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 22:57.08 | Aces1Up | any one here run a calling card business? |
| 22:57.14 | Aces1Up | I would like some advice. |
| 22:57.24 | davidcsi | aces1up: ask |
| 22:57.35 | aptura | Everyone has at one time or another |
| 22:57.36 | JT | javar: i'm sure some people can. |
| 22:57.40 | JT | ~question |
| 22:57.41 | jbot | somebody said question was If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 22:58.02 | Aces1Up | just curious how many users-per-channel a calling card company has typically supported. |
| 22:58.15 | JT | err, users per channel, what? |
| 22:58.18 | JT | 1? |
| 22:58.24 | davidcsi | good "what" |
| 22:58.28 | davidcsi | what do you mean?= |
| 22:58.43 | JT | if they use pri for outgoing, 1 user takes 2 channels |
| 22:58.53 | javar | hi JT, I have a A200. I'm trying to install the drivers but i got this error: ./Setup: 973: Syntax error: Bad substitution |
| 22:59.10 | davidcsi | oh, that... it really depends on what traffic you think you'll have... |
| 22:59.15 | JT | javar: is this wanpipe? |
| 22:59.23 | javar | JT, yes |
| 22:59.24 | davidcsi | its a matter of dimensioning really. |
| 22:59.31 | JT | javar: latest version/ |
| 22:59.31 | Aces1Up | david say i have 1 line incoming, how many users can typically use that 1 line in a 24-hour period without getting excessive busy signals. |
| 22:59.47 | javar | JT, wanpipe-3.1.0 |
| 22:59.48 | Aces1Up | how much can i load it. |
| 23:00.15 | JT | javar: yeah i have no idea if that's the latest or not, anyway, sounds like a bug, try another versiob |
| 23:00.41 | Aces1Up | i know its a measurement that has to be made after traffic patterns are seen, but i would think there is a general measurement or rule of thumb for that sort of thing. |
| 23:00.46 | J4k3 | hmm interesting... idefisk's in-call dtmf won't work with my cellphone's voicemail, but xlite's will |
| 23:00.58 | davidcsi | again, depends on your traffic, but you gotta overdimensing the access, otherwise you'll have people calling that can't get through... the standard is like 250k/month per E1 (30channels) do your math... |
| 23:01.01 | J4k3 | I guess idefisk is trying to do inband and the xlite is doing oob |
| 23:01.01 | Aces1Up | i was thinking 5 users would be too many. but not sure. |
| 23:01.39 | JT | J4k3: different protocols too |
| 23:01.50 | JT | i thought iax was out of band only |
| 23:01.51 | davidcsi | but again, tha ALSO depends on your asr, etc... i usually do 60%inbound 40%outbound |
| 23:02.10 | J4k3 | JT: I'm using idefisk in sip mode. |
| 23:02.13 | javar | JT, i tried with wanpipe-2.3.4-10, but same issue |
| 23:02.18 | J4k3 | so... that might be the problem too, I doubt its optimized for sip |
| 23:02.35 | JT | J4k3: didn't know it could do sip |
| 23:03.01 | J4k3 | JT: yep.. it'll do both (pretty slick really) |
| 23:03.16 | JT | J4k3: all soft phones are awful :) |
| 23:03.17 | Aces1Up | david average-...-ratio ? ASR? |
| 23:03.30 | JT | Aces1Up: google |
| 23:03.43 | JT | javar: umm what system is this on? |
| 23:03.43 | davidcsi | aces: yeah |
| 23:04.50 | davidcsi | fact is you can have, say 1 T1 inbound and 1 outboud, all channels of inband could be in use while only 20 of the outbound are in use... depends on ACD/ASR and so on. |
| 23:05.28 | Aces1Up | yeh i looked up on google my bad, i'm lazy. |
| 23:05.40 | javar | JT, ubuntu server |
| 23:05.42 | davidcsi | if you think you'll have some traffic, get 4TDM, on these things you can never overdo it... |
| 23:05.55 | *** join/#asterisk MACscr (n=MACscr@adsl-75-17-76-236.dsl.peoril.sbcglobal.net) |
| 23:06.17 | Aces1Up | david well, i was just going to run it all voip. |
| 23:06.21 | davidcsi | OR, get the access provider to deliver the access via VoIP |
| 23:06.22 | MACscr | Can anyone recommend a data center in the US to get a server to through asterisk on? Im going ot be connecting it to a server in Germany |
| 23:06.35 | davidcsi | Aces: that's the best solution... |
| 23:06.36 | Hogie | softlayer |
| 23:06.47 | JT | javar: what commands did you issue to install the sangoma? |
| 23:06.59 | MACscr | My client already has one server at LayeredTech, so i was thinking about them |
| 23:07.05 | javar | JT, /usr/src/wanpipe-3.1.0$ sudo ./Setup install |
| 23:07.13 | javar | JT, ./Setup: 973: Syntax error: Bad substitution |
| 23:07.31 | JT | javar: what is on line 973? |
| 23:07.54 | Hogie | MACscr: Softlayer has Xeon 3220 (quadcore Core 2 Duo based) + 2gb ram for $179 with 2TB transfer atm |
| 23:08.04 | JT | do the main us-eu links go out via the us west or east coast? |
| 23:08.23 | javar | JT, name=${name// /} |
| 23:08.27 | Aces1Up | david hrmm that figure 250k/e1 a month, is there a number of users for that |
| 23:08.28 | Aces1Up | ? |
| 23:08.58 | MACscr | Hogie, thats nice, but a smaller server for around $100 would work just fine with the amount of traffic we will be doing |
| 23:09.00 | davidcsi | aces: can't say for sure... concurrent users you mean? |
| 23:09.09 | Aces1Up | yes |
| 23:09.22 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 23:09.45 | Aces1Up | well acually just users period, how many users have access to create the 250k/e1/mo |
| 23:10.02 | Hogie | MAC: ah, okay. They dont carry anything slower than that really |
| 23:10.03 | javar | JT, line 972: elif [ $BASH_SUPPORT -gt 1 ]; then |
| 23:10.37 | JT | javar: err |
| 23:10.54 | JT | javar: that or the line with name=? what's what? |
| 23:11.16 | javar | JT, Line 973: name=${name// /} |
| 23:11.20 | davidcsi | 28-30 at peak time, of course... there's really no way to calculate that... it depends on the traffic profile... i.e. if most of the traffic you do is for instance Colombia, then few users, as they talk a lot and use up the card very fast... same goes for turists... |
| 23:11.21 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.125) |
| 23:11.53 | jaxxan | hey guys, is there a way to send a fax using asterisk |
| 23:12.13 | jaxxan | is it possible to hook up a fax machine to it to send a fax is what i mean |
| 23:12.20 | Aces1Up | david i see thanks. |
| 23:13.00 | davidcsi | aces: but you're already doing it via voip, right? divide whatever monthly traffic you got by 250k, then multiply by 2 to know how many T1/E1 you actually need. |
| 23:13.07 | davidcsi | its the best way to go. |
| 23:17.50 | Aces1Up | david thank you very much these figures will help me greatly, is there a site with these type of measurements for different calling areas? |
| 23:18.02 | Aces1Up | or what would i google for that sort of info? |
| 23:18.39 | JT | telecommunications traffic engineering |
| 23:18.47 | davidcsi | no that i know of... i would google something like "[country] average call duration" |
| 23:20.41 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 23:20.48 | thevoke | :) |
| 23:22.18 | javar | JT, any idea? |
| 23:26.51 | mvanbaak | niterz all |
| 23:28.11 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 23:30.57 | J4k3 | asterisk eats another telemarketer! |
| 23:31.51 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 23:32.31 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 23:32.33 | canberk | helloooo |
| 23:33.21 | canberk | i set asterisk to use g729:20 codec in h323.conf however asterisk is always trying to have the conversation with 230ms payload |
| 23:33.26 | canberk | why do you think is this |
| 23:33.57 | davidcsi | don't know... but i wouldn't use h323 on asterisk... :S |
| 23:34.25 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 23:34.29 | canberk | well, why davidcsi |
| 23:35.02 | davidcsi | it can't hold much traffic, it eats up the ram, etc, etc... |
| 23:35.03 | JT | J4k3: do you use Zapateller too? |
| 23:35.18 | canberk | but in case you must use, there is no way out :( |
| 23:35.41 | davidcsi | i do protocol converion with yate on the same machine. |
| 23:35.46 | JT | canberk: because no-one uses H.323 on asterisk and it's poorly developed? |
| 23:36.55 | canberk | possibly yeah, but i guess there must be a solution because it is impossible to have a h323 conversation with 230bytes payload |
| 23:37.16 | JT | canberk: umm davidcsi just mentioned a solution |
| 23:37.29 | JT | also, never assume a solution, or at least a free one exists |
| 23:39.43 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:40.38 | kiscokid | In the dialplan how do you refer to the extension of the caller. I want to use this as a parameter for VoiceMailMain |
| 23:40.41 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 23:42.40 | J4k3 | JT: nah... I just let them get hung up on |
| 23:43.26 | J4k3 | s 23.137 ms |
| 23:44.05 | davidcsi | anyone knows how the callback in perl agi works?? can't get it to work! |
| 23:44.05 | kiscokid | ? |
| 23:47.05 | davidcsi | kiscokid, 1.2 and before: ${CALLERID(num)} |
| 23:48.28 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 23:49.48 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:51.05 | JT | J4k3: cool |
| 23:51.18 | JT | J4k3: so routerboards.... which ones have you used? |
| 23:51.44 | kiscokid | davidcsi: I am on 1.4 |
| 23:51.45 | J4k3 | JT: 112, 133, 133c, 532+502 |
| 23:51.56 | JT | not 150 then |
| 23:52.00 | JT | considering the 150 |
| 23:52.01 | J4k3 | not yet |
| 23:52.07 | davidcsi | kisco: http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 23:52.08 | JT | i don't need wireless/mini pci |
| 23:52.09 | J4k3 | I have a use for them |
| 23:52.21 | kiscokid | ok, thanks |
| 23:52.24 | J4k3 | just haven't purchased 'em yet |
| 23:52.29 | JT | J4k3: do you only run routeros? |
| 23:52.56 | J4k3 | JT: on the routerboards yes... |
| 23:53.13 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 23:53.25 | JT | J4k3: i am interested in running linux |
| 23:53.37 | J4k3 | yeah... I can't imagine it being a problem |
| 23:53.44 | J4k3 | supported mips chip |
| 23:53.55 | J4k3 | well, the 150 might be different... its a different chip I think |
| 23:53.56 | J4k3 | hrm |
| 23:54.20 | MACscr | Would a 2.4ghz celeron with 512mb of ram be fine for asterisk? With a light call parttern? |
| 23:54.47 | J4k3 | wow, one of these gs bt101's makes a horrible static sound on occasion after its been active for a few minutes |
| 23:54.47 | davidcsi | mac, very light, no transcoding... sure |
| 23:55.23 | J4k3 | (I called milliwatt and let it run for a few minutes) |
| 23:55.23 | MACscr | Hmm, i guess i would need transcoding if i used a high compression codec, right? |
| 23:56.01 | davidcsi | no, if you change from on codec (receive) to another(send) |
| 23:56.11 | kiscokid | davidcsi: that worked |
| 23:56.31 | JT | davidcsi: you could do some transcoding on a 2.4GHz for sure |
| 23:57.00 | MACscr | Im wanting to do a setup between an asterisk in germany and one in the US |
| 23:57.13 | MACscr | Shopping for a DC in the US right now that uses Global Crossing |
| 23:57.27 | MACscr | And where i can get a decent price on a basic server |
| 23:57.29 | davidcsi | JT, sure you can, but very light is what i mean. |
| 23:57.56 | JT | MACscr: co-lo is usually cheaper than dedicated, but obviously may be more inconvenient |
| 23:58.25 | J4k3 | hmm... this static noise I just got, again... almost sounds like a telco timing problem |
| 23:58.33 | J4k3 | and I've only ever noticed it on calls from this area... |
| 23:59.42 | fetcher | anyone have SIP accounts with i2telecom.com (aka Voicestick)? Wondering if they're down all over, or just for me |