| 00:00.18 | TrentCreek | one like this one: 710 Western Union Telegraph Company - Southern USA |
| 00:00.32 | TrentCreek | Does that mean it is only used by WU? |
| 00:00.38 | TrentCreek | or by their customers? |
| 00:01.30 | TrentCreek | well they explain it below....TTY |
| 00:04.13 | eppigy | holla |
| 00:04.20 | TrentCreek | crayola |
| 00:09.51 | *** join/#asterisk asteriskwow (n=elastixr@196.211.34.2) |
| 00:10.00 | asteriskwow | hi there |
| 00:10.09 | asteriskwow | need some help please anyone active? |
| 00:10.17 | tiberius_ | anyone have recommendations for SIP trunking providers? |
| 00:11.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:11.49 | TrentCreek | Corydon76-dig: yes, the list is a bit outdatd |
| 00:12.27 | TrentCreek | at least it is a start |
| 00:14.15 | asteriskwow | if i dial a external number that is busy it tells me that " Got SIP response 486 "Busy here" back from 196.3.177.233" but i dont get ingage tone on my sip phone, how can i change my dialplan to use this info to play ingage tone to me? |
| 00:14.35 | *** join/#asterisk variable_office (n=variable@fs0.iswan.net) |
| 00:14.52 | variable_office | is there a way to read the 'mailbox' value for a user in sip.conf from the extensions? |
| 00:16.01 | asteriskwow | i am dialing over a sip trunks to my home house number, i just get a dead silence than my phone disconnects, would be nice to hear ingage tone if number dialed is busy |
| 00:16.19 | [TK]D-Fender | variable_office: "core show function SIPPEER" |
| 00:16.41 | [TK]D-Fender | asteriskwow: "core show application busy" |
| 00:16.47 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-217-50.lns10.mel6.internode.on.net) |
| 00:16.55 | [TK]D-Fender | asteriskwow: ANSWER first |
| 00:17.02 | variable_office | thanks! |
| 00:19.21 | TrentCreek | Corydon76-dig: a litte research shows that list predates 1991 :-D |
| 00:20.50 | *** join/#asterisk mattzerah (n=matt@ozvoip.dsl.onthenet.net) |
| 00:24.14 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 00:24.44 | Corydon76-dig | TrentCreek: it cannot |
| 00:24.48 | asteriskwow | [TK]D-Fender: do you mean i have to but a answer first in my dialplan,,? sorry if it sounds dumb still new to asterisk |
| 00:25.03 | Corydon76-dig | That list is on the web and 1991 predates the web |
| 00:25.26 | [TK]D-Fender | asteriskwow: Yes, have * answer the call first, then call busy. You might need to Playback a least a little recording (silence/1 should do) |
| 00:25.26 | TrentCreek | Corydon76-dig: why not? it lists area code 905 as mexico city...they stopped using it in 1991 |
| 00:25.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:26.04 | asteriskwow | [TK]D-Fender-how do answe the call first if it is an outgoing call? |
| 00:26.09 | TrentCreek | Corydon76-dig: Much data that was available via archie and gopher was later in in web format |
| 00:26.32 | [TK]D-Fender | asteriskwow: * answers the call from your PHONES before calling OUT <- |
| 00:26.44 | [TK]D-Fender | asteriskwow: There are *2* calls involved |
| 00:27.54 | *** join/#asterisk mrsci (n=mrsci@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 00:29.01 | asteriskwow | meaning when i dial out it picks up(answers) a sip channel on my side, then connects to another sip channel at the Sip trunk provider side |
| 00:29.45 | asteriskwow | [TK]D-Fender so the call on my side must be live(answered) before i will get ingage tone? |
| 00:29.59 | [TK]D-Fender | asteriskwow: When you call from your phone, * accepts the call from the phone. that is 1 end. then it hits dialplan. your PHONE has not been "answered". In your dialplan it dials out. When the other side says "busy", * just passes that along and your phone sasy "click* |
| 00:30.41 | [TK]D-Fender | asteriskwow: If you have * ANSWER your phone then when the other side says "sory, busy", in your dialplan you can pass back an AUDIO indication of "busy" |
| 00:30.58 | asteriskwow | [TK]D-Fender thx i learned something new ,, ill give it a go |
| 00:41.17 | asteriskwow | [TK]D-Fender do i still have to use the application busy? |
| 00:41.48 | [TK]D-Fender | asteriskwow: Yes. |
| 00:42.37 | asteriskwow | thx |
| 00:44.41 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 00:48.15 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5f728c8c5fd06ee6) |
| 00:49.12 | *** join/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 00:50.53 | root52 | So question. I am interested in some asterisk training. Is it true that the only training offered is what is listed on the digium web site? I have done some googleing and could not find any collages or tech schools that offer training closer to home. |
| 00:51.07 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 00:55.20 | [TK]D-Fender | root52: many places offering the boot camp & dCAP cert training. If they aren't local, then you'll have to travel. |
| 00:55.37 | [TK]D-Fender | root52: If you're looking for a physical class |
| 00:57.09 | root52 | [TK]D-Fender: Thanks. I was just unsure if the only "official" training was that listed on the web site I will look for other classes. Thanks!! |
| 01:01.53 | keebler | When setting up a shared network between Asterisk and local LAN, how much Bandwidth would be appropriate to allot to Asterisk for 6 simultaneous internal calls? Asterisk will have priority otherwise, but would it be beneficial or practical to setup a Quota for it? Say, 2MBps of LAN, and the rest to data? |
| 01:02.46 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:02.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:03.56 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:04.01 | *** join/#asterisk neurosys (n=vinix@c-71-196-8-216.hsd1.fl.comcast.net) |
| 01:04.03 | keebler | Or should i just set the Asterisk Server's MAC address as the priority for QoS? |
| 01:04.12 | [TK]D-Fender | keebler: 1 call in SIP+ULAW = 85kpbs. You're on a LAN. It doesn't matter :P |
| 01:04.26 | keebler | Wireless lan. |
| 01:04.27 | keebler | Sorry |
| 01:05.21 | [TK]D-Fender | keebler: 510 kbps for 6 calls. thats 0.510 megabits. How fast is your WLAN? |
| 01:05.31 | [TK]D-Fender | keebler: Seriously..... |
| 01:05.32 | keebler | And Just testing my lan right now, I was getting 11KBs per call. |
| 01:06.00 | [TK]D-Fender | keebler: You cal sucks. |
| 01:06.03 | [TK]D-Fender | calc* |
| 01:06.15 | keebler | I'm using the Router's Bandwidth monitor. |
| 01:06.38 | keebler | Want me to take a Screenshot? |
| 01:06.40 | [TK]D-Fender | keebler: ULAW = 64kpbs, UDP overhead = 20kbps |
| 01:06.58 | [TK]D-Fender | keebler: I don't doubt that it SAYS that. Whatever... its just WRONG |
| 01:07.04 | keebler | Heh |
| 01:07.22 | keebler | So g.711u as the codec doesn't mean jack? |
| 01:07.37 | [TK]D-Fender | keebler: G.711u = ULAW |
| 01:07.54 | keebler | Okay, then how do I go about testing this crap properly? |
| 01:08.14 | [TK]D-Fender | [20:04]<[TK]D-Fender>keebler: 1 call in SIP+ULAW = 85kpbs. You're on a LAN. It doesn't matter :P |
| 01:08.19 | [TK]D-Fender | [20:06]<[TK]D-Fender>keebler: ULAW = 64kpbs, UDP overhead = 20kbps |
| 01:08.29 | [TK]D-Fender | [20:05]<[TK]D-Fender>keebler: 510 kbps for 6 calls. thats 0.510 megabits. How fast is your WLAN? |
| 01:08.32 | keebler | Okay, thats what its supposed to do.... |
| 01:08.38 | keebler | I want to physically chart it. |
| 01:08.43 | [TK]D-Fender | keebler: Can it get much clearer? :) |
| 01:08.58 | keebler | I need justification... |
| 01:08.58 | [TK]D-Fender | keebler: Bits are bits dammit! |
| 01:09.04 | keebler | Right.... |
| 01:09.22 | [TK]D-Fender | keebler: If you want validation go to #psychotherapy |
| 01:09.36 | keebler | But, why the hell is it saying I'm using 11KBps ? Its pretty damn accurate when monitoring this ISO I'm downloading. |
| 01:10.00 | [TK]D-Fender | keebler: Do we support your router? I doubt it highly! |
| 01:10.30 | [TK]D-Fender | keebler: Maybe because its better at calculating bitter numbers |
| 01:10.35 | [TK]D-Fender | bigger* |
| 01:10.49 | keebler | Then you contradict your previous statement. |
| 01:10.53 | keebler | If "bits are Bits" |
| 01:11.01 | keebler | It shouldn't matter the size. |
| 01:11.24 | keebler | Its DDWRT, everyone uses it in here. |
| 01:12.54 | keebler | And since I'm only using Asterisk during the call. Its relative to this channel... since My monitor says its 11KB for just a voice call, And I'm more inclined to believe it works properly, then I need to find out what in either Asterisk or the ATA's is forcing such large numbers. |
| 01:22.59 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 01:29.08 | *** join/#asterisk RB2 (n=RB2@pool-71-172-128-195.nwrknj.east.verizon.net) |
| 01:29.08 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 01:37.40 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 01:39.09 | keebler | So realistically 128Kbps is used for every 2 channels? |
| 01:40.36 | keebler | erm 168? |
| 01:40.52 | keebler | Nvm |
| 01:40.59 | rob0 | Everyone here uses dd-wrt? |
| 01:40.59 | keebler | Gah. I'm going to bed. |
| 01:41.00 | [TK]D-Fender | keebler: math <- |
| 01:41.07 | [TK]D-Fender | rob0: Statistically... no |
| 01:41.12 | keebler | rob0: Yes. |
| 01:41.14 | rob0 | whew |
| 01:41.16 | keebler | Haha |
| 01:41.18 | keebler | :P |
| 01:41.20 | [TK]D-Fender | I don't |
| 01:41.24 | [TK]D-Fender | keebler: Liar ;p |
| 01:41.26 | rob0 | me neither |
| 01:41.51 | [TK]D-Fender | rob0: Any other questions you already have the answer to you feel like asking anyway? :) |
| 01:42.32 | rob0 | Why is the sky blue? |
| 01:42.34 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:42.50 | [TK]D-Fender | rob0: Because if it were green we wouldn't know when to stop mowing :p |
| 01:43.03 | [TK]D-Fender | NEXT!@!@@!@ (c) BKW |
| 01:49.59 | *** join/#asterisk RAiDENZ (n=raiden@wnpgmb014rw-ad01-9-223.dynamic.mts.net) |
| 01:50.52 | RAiDENZ | Is there a way to tell in asterisk which end of the call hung up? |
| 01:53.18 | [TK]D-Fender | in your dialplan you can use "g" in dial, or the "h" exten |
| 01:59.50 | RAiDENZ | I have an H extension an tried a dumpchan but doesn't give any information to see which end hung up |
| 02:02.37 | [TK]D-Fender | RAiDENZ: If you use "g" in your dial You'll know they hung up. |
| 02:02.48 | [TK]D-Fender | RAiDENZ: And can flag the CDR accordinging. |
| 02:02.53 | [TK]D-Fender | RAiDENZ: And can flag the CDR accordingly |
| 02:03.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 02:04.23 | *** join/#asterisk criten (n=criten@216.75.233.220.exetel.com.au) |
| 02:04.37 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-250-127.lns10.mel6.internode.on.net) |
| 02:13.58 | keebler | How many calls/extensions can the Asterisk enabled WRT54G support? |
| 02:21.56 | BadHAL | :o |
| 02:22.00 | BadHAL | probably not a whole lot |
| 02:24.35 | frogonwheels | keebler: at a guess 2 active calls. |
| 02:25.09 | keebler | Yeah. Just curious. I wouldn't think about deploying it in a corporate environment. I've just got an extra WRT54 lying here and was considering tinkering with it. |
| 02:25.38 | frogonwheels | keebler: My asus wl500gp has handled a meetme with 2 external, 1 internal line + one other call concurrently. |
| 02:25.58 | frogonwheels | keebler: it possibly could handle more. |
| 02:26.04 | keebler | That the one with 32MB ram? |
| 02:26.18 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 02:26.39 | frogonwheels | keebler: erm.. yep. |
| 02:27.03 | keebler | Heh. *laughs at the measly 8MB on this WRT) |
| 02:27.15 | harry_v | looks like callerID 911 fraud had made its way to msnbc |
| 02:27.26 | frogonwheels | keebler: depends what else you got running:) |
| 02:27.50 | harry_v | how many simultanious calls can you make on those units? |
| 02:28.02 | keebler | Thats what I just asked. :) |
| 02:28.12 | keebler | Our guess is 2. |
| 02:28.29 | frogonwheels | keebler: obviously no transcoding. |
| 02:28.35 | keebler | yea |
| 02:29.20 | harry_v | Figured that was the case. |
| 02:29.37 | frogonwheels | keebler: You using openwrt? |
| 02:29.45 | [TK]D-Fender | Could probably support more calls outside that meetme |
| 02:30.11 | frogonwheels | [TK]D-Fender: possibly not the WRT54G though. |
| 02:30.32 | [TK]D-Fender | frogonwheels: More on any platform :) |
| 02:30.34 | frogonwheels | [TK]D-Fender: I'm guessing a few calls concurrently. but never tried. |
| 02:31.05 | [TK]D-Fender | I have seen the number "4" thrown around regarding WRT's |
| 02:32.15 | frogonwheels | keebler: I've got Asterisk package changes for OpenWRT to break it up into smaller bits so you can fit it in an image. |
| 02:32.28 | frogonwheels | keebler: I have my * installed on an USB drive under OpenWRT. |
| 02:32.40 | frogonwheels | keebler: oBviously, not an option for the WRT. |
| 02:33.32 | keebler | Yeah. I'll just stick with this MSIWind desktop as the PBX |
| 02:33.56 | keebler | Its small enough.. |
| 02:34.06 | keebler | I will eventually go iTX. |
| 02:34.32 | ricko73 | if you don't need a pci device, the ALIX boards are nice |
| 02:36.09 | [TK]D-Fender | If you do, Soekris is a good option, and supports better storage |
| 02:36.59 | [TK]D-Fender | Soekris is kinda pricy, but the combination of functionality is really nice for the size & power reeq's |
| 02:37.25 | RAiDENZ | [TK]D-Fender is there a way in the h extension to see who hung up instead of using the "g" option in dial? |
| 02:37.40 | [TK]D-Fender | I haven't really seen anything interesting or new in 2008 for the embedded space like these... |
| 02:38.00 | keebler | Yeah, this MSIWind doesn't have any PCI slots. But its bigger than need be because it has a 5.25" slot for an internal DVDRM. |
| 02:38.15 | keebler | The board itself is tiny. |
| 02:38.39 | [TK]D-Fender | RAiDENZ: If you use the "g" exten you can set the CDR userfield or something to imply the callee hung up. if this value is NOT et you can then infer that the CALLER hung up. |
| 02:39.00 | [TK]D-Fender | keebler: link? |
| 02:39.23 | keebler | http://www.newegg.com/Product/Product.aspx?Item=N82E16856167032 |
| 02:39.28 | keebler | Not bad for $140 |
| 02:39.47 | keebler | Useless as HTPC though. |
| 02:40.27 | [TK]D-Fender | keebler: Depends on your needs I guess... |
| 02:40.34 | mrsci | keeble, I have looked at that system how well does it run? |
| 02:40.43 | mrsci | How many calls can it handle? |
| 02:41.25 | keebler | mrsci: I have FreeBSD installed on a 4GB CF Card, running Asterisk 1.4.21 and atm 8 extensions. Only tested 2 concurrent calls, but its not bad. |
| 02:41.26 | [TK]D-Fender | keebler: Would work for my HTPC needs |
| 02:41.28 | keebler | SILENT. |
| 02:41.36 | [TK]D-Fender | keebler: But thats what My * does already :) |
| 02:42.36 | keebler | [TK]D-Fender: The Intel ATOM can't reliably play 720p, let alone 1080p. Not to mention it only has VGA and its an IntelGMA945. Other than that, just basic movie watching on a VGA enabled LCD, (My 40" Sony), it did pretty good. |
| 02:43.06 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 02:43.11 | [TK]D-Fender | keebler: I use VGA on my setup as well.... better card though... |
| 02:43.17 | [TK]D-Fender | actually... scratch that :) |
| 02:43.30 | [TK]D-Fender | keebler: VIA KT400 onboard :) |
| 02:43.31 | keebler | I'm going to get one of these to play around with... http://www.mini-box.com/Mini-Box-M200-LCD |
| 02:43.49 | keebler | Maybe get it with the Jetway board and minipci slot. |
| 02:44.04 | keebler | And throw an 802.11a/b/g card in. |
| 02:44.32 | keebler | Although, I don't like the idea of running my Router and any sort of server on the same box. |
| 02:49.36 | keebler | OOH. |
| 02:49.43 | keebler | Dual Core Intel Atom |
| 02:49.43 | keebler | http://www.mini-box.com/Intel-D945GCLF2-Mini-ITX-Motherboard |
| 02:57.32 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 02:57.39 | ZX81 | is digium website down? |
| 02:57.44 | ZX81 | http://www.digium.com/ |
| 02:57.51 | ZX81 | or just for me? |
| 02:57.58 | rob0 | down |
| 02:58.25 | ZX81 | heh |
| 02:58.25 | ZX81 | k |
| 03:00.18 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 03:18.01 | mrsci | i can't get to the digium site either |
| 03:18.41 | Qwell | maintenance, I believe |
| 03:19.08 | Qwell | yep |
| 03:21.40 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:22.36 | *** join/#asterisk sohum (n=sohum@114.72.191.67) |
| 03:23.10 | sohum | ok. I don't quite understand what Asterisk does, but is it possible to use it to plug a phone into a modem and use that as a mic/speakers? |
| 03:23.45 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:24.29 | trnzmeta | so you're going to turn asterisk to an overglorified recorder? |
| 03:24.57 | sohum | if there's an easier way to do it, I'm be willing to listen. this is just the first thing that came to mind |
| 03:25.27 | trnzmeta | what do you need to do? |
| 03:25.48 | sohum | be able to use a physical POTS phone as a speaker/mic pair |
| 03:27.04 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:27.37 | trnzmeta | so a phone call comes in, and using your phone, you can go hands free? |
| 03:28.15 | *** part/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 03:29.04 | sohum | no - I just want to be able to use the speaker and mic on the phone as inputs and outputs on my machine, theoretically using the modem to connect the phone and the machine |
| 03:32.24 | [TK]D-Fender | sohum: Modem = dead end. Useless. Speaker & mic on a soundcard would be using a soft-phone |
| 03:32.26 | [TK]D-Fender | ~softphone |
| 03:32.27 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 03:32.49 | sohum | ah |
| 03:32.56 | sohum | fair enough, thank you |
| 03:33.02 | *** part/#asterisk sohum (n=sohum@114.72.191.67) |
| 03:33.06 | [TK]D-Fender | sohum: Ah, if you want to use a phone a speaker & mic I have actually seen a device for this... |
| 03:34.10 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:36.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:37.05 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:37.41 | iaxy | TK.... |
| 03:38.26 | iaxy | I turned qulify back on and I am getting sip OPTIONS messages every 1 minute about. |
| 03:38.31 | iaxy | Normal? |
| 03:38.38 | [TK]D-Fender | iaxy: Yup, thats what it does |
| 03:38.41 | jql | that's what it does |
| 03:39.09 | [TK]D-Fender | slaps jql |
| 03:39.14 | [TK]D-Fender | jql: Cheap knock-off! |
| 03:39.22 | iaxy | haha |
| 03:39.51 | iaxy | Why does it say from unknown. kinda silly that it doesn't know itself. |
| 03:39.53 | jql | I snooze, I lose |
| 03:40.00 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 03:40.03 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:40.06 | jql | I even loose |
| 03:42.25 | harry_v | seems my ip500 is having fits again trying to locate the server. tested the server with same username/password for that account and can login. |
| 03:43.02 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:45.36 | harry_v | did get this error in the polycom log 0202031732|app1 |4|00|Loaded application sip.ld successfully, errors 0x4000. |
| 03:46.26 | [TK]D-Fender | harry_v: Could be an incompatible config issue. |
| 03:46.35 | [TK]D-Fender | harry_v: (or corrupted) |
| 03:48.16 | harry_v | possibly. I am installing tftp just to see if its not some odd ftp issue. |
| 03:48.20 | harry_v | compare the two |
| 03:48.59 | harry_v | does tftp use the same home/user directory as ftp? |
| 03:50.07 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 03:51.09 | [TK]D-Fender | harry_v: TFTP doesn't have "users" |
| 03:51.14 | [TK]D-Fender | harry_v: or passwords |
| 03:51.24 | [TK]D-Fender | harry_v: thats why its "trivial" |
| 03:51.24 | harry_v | im sure. |
| 03:51.29 | harry_v | :) |
| 03:51.39 | [TK]D-Fender | tftp = suck |
| 03:51.49 | harry_v | I used it 10 years ago on cisco but never on linux. |
| 03:52.02 | harry_v | and why is that? |
| 03:52.48 | [TK]D-Fender | harry_v: all the above... lack of timestamps for updates, not having multiple accounts so you can split firmwares. Or having to worry about possibly conflicting files. |
| 03:53.07 | harry_v | I see |
| 03:53.12 | [TK]D-Fender | harry_v: Total waste |
| 03:53.19 | harry_v | probebly best to stay away then. |
| 03:53.27 | [TK]D-Fender | harry_v: FTP is dead-easy to set up on any platform. |
| 03:53.38 | harry_v | ohh sure. it was working before. |
| 03:53.41 | harry_v | just not this time. |
| 03:54.05 | harry_v | Changed the CFG files to match the mac info of the phone. |
| 03:54.07 | harry_v | brb |
| 03:55.04 | frogonwheels | harry_v: not that tftp doesn't have some uses.. however they _are_ pretty limited. |
| 03:57.14 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:01.15 | *** join/#asterisk Gopaul (n=chatzill@59.97.120.98) |
| 04:02.49 | harry_v | now if the files were misconfigured would it simulate a failed to ftp server message? |
| 04:04.24 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:10.27 | keebler | Anyone here try installing Asterisk on a Gumstix? I'm reading about it now, but looking for first hand experience. |
| 04:10.42 | harry_v | could not contact boot server. user/pass of phone same as ftp server. Man I dont know. Might have to work on this tomarro its getting late. |
| 04:11.20 | harry_v | keebler, I went to a sit in session with the developer of gumstix and he said somone in the community did so that. |
| 04:11.27 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:11.56 | keebler | harry_v: Yeah. I know. I'm reading about it now. Just hoping more than one person has tried it. |
| 04:12.13 | keebler | A 400mhz Xscale processor sounds promising. |
| 04:12.26 | harry_v | yea no kidding. spy pbx |
| 04:13.56 | keebler | in deed |
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| 04:17.56 | LemensTS | I want to connect SIP/305 with test-asm.agi, i got this: $asm->Originate(SIP/305, NULL, NULL, NULL, DeadAGI, test-asm.agi); |
| 04:18.12 | LemensTS | it gives me: Response = Error; Message = Originate failed |
| 04:18.27 | LemensTS | This is running a php file from bash... |
| 04:18.35 | LemensTS | php-asmanager |
| 04:19.53 | LemensTS | I cant find anyone originating a call into an agi app |
| 04:19.57 | LemensTS | on google |
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| 04:32.18 | [TK]D-Fender | LemensTS: You seem to have forgotting the concept of typed variables & parameters.... |
| 04:32.45 | [TK]D-Fender | LemensTS: Something in there should be a STRING |
| 04:34.02 | trnzmeta | guys: any one had the experience where they make a phone call, everything goes well |
| 04:34.33 | trnzmeta | but when you hang up, the phone rings and connects the two parties again |
| 04:34.45 | trnzmeta | however the call is dead |
| 04:35.40 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:35.44 | [TK]D-Fender | trnzmeta: Some real details & debug would help |
| 04:36.33 | trnzmeta | going through the logs now |
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| 04:50.09 | IPconfig | hey guys , quick question i have tried installing asteriskNow ver 1.0.2 (32bit) but it bogs out with a bug error while its installing , anyone had that issue with the current asteriskNow from the website ? |
| 04:50.46 | keebler | Here you go. Asterisk on the Gumstix... supports 40 concurrent calls. apparently http://the-edge.blogspot.com/2005/10/worldss-smallest-ip-pbx-at-astricon.html |
| 04:51.29 | keebler | That was dated 2005 though |
| 04:51.45 | keebler | I think all development of Astlinux for the Gumstix has stopped. |
| 04:51.51 | keebler | Shame, very powerful system |
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| 04:52.44 | IPconfig | anyone had that issue at all .. |
| 04:52.51 | IPconfig | this being the version offered right of the website |
| 04:53.08 | [TK]D-Fender | IPconfig: upgrade regardless... 1.0.2 was rPath PITA... |
| 04:53.22 | [TK]D-Fender | IPconfig: New one is considerably better |
| 04:53.43 | [TK]D-Fender | IPconfig: Or do what the rest of us will tell you anyway and roll your own on a distro of yourchoice. |
| 04:54.54 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:55.08 | keebler | Agreed |
| 04:56.05 | keebler | AsteriskNow 1.5 worked fine except a small issue with ethernet driver support. |
| 04:56.19 | [TK]D-Fender | keebler: for which? |
| 04:56.36 | keebler | MSIWind. Realtek chip |
| 04:57.17 | keebler | Both 1.5 and 1.0.2 |
| 04:57.38 | [TK]D-Fender | keebler: Odd... |
| 04:57.49 | keebler | I got annoyed with all of the distro's though. |
| 04:57.51 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 04:58.02 | keebler | Not a fan of GUI anyways. |
| 04:58.31 | keebler | Trixbox and PBXinAFLash didn't support my NIC |
| 04:58.33 | keebler | as well |
| 04:59.14 | [TK]D-Fender | keebler: What does? |
| 04:59.25 | [TK]D-Fender | keebler: these are all CentOS based |
| 04:59.58 | keebler | FreeBSD is the only thing that worked right out of the box... well, Ubuntu would have worked too. |
| 05:00.06 | keebler | But I prefer FBSD. |
| 05:00.36 | keebler | Only "problem" is. Asterisk PORT is only version 1.4.21 |
| 05:00.42 | keebler | But meh. |
| 05:00.48 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 05:01.19 | keebler | My application is so basic and simple, I'm sure I wouldn't notice any benefit from .23 |
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| 05:57.44 | Gopaul_ | how to get the Q931 packets from asterisk log |
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| 06:28.44 | ruben23 | hi...i installed asterisk 1.4 and registered my sip account on it...problem is i cant get a call outgoing...my asterisk CLI display sip registered..when i call no activity...then call failed..iahve my extensions.conf setup also.. |
| 06:33.38 | keebler | Not an Asterisk question exactly... but does anyone know what is faster? a "High Speed" 4GB CF Flash card, or a Standard 4GB SD Flash Card? |
| 06:34.03 | keebler | Currently using a 120x CF Flash card with FBSD/Asterisk. |
| 06:34.30 | keebler | But looking at going to a more embedded route and would like to use an SD Card, but I'm afraid it would be too slow for constant Asterisk use. |
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| 07:31.05 | zeeesh | tring to use simple calling card AGI from here "www.dynx.net/ASTERISK/AGI/ccard/agi-ccard.agi " what shud be the extensions call ... like can i "exten => _X.,1,DeadAGI(pina.pl|${CDR(accountcode)}|${EXTEN})"?? |
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| 08:02.54 | ruben23 | hi |
| 08:03.07 | ruben23 | how to see call logs on asterisk.. |
| 08:04.39 | kaldemar | by default you'll find them in /var/log/asterisk/cdr-csv/Master.csv |
| 08:12.50 | kaldemar | ruben23: but you won't find any help there to your problem. in cli, give command "set verbose 10" before making a call to see what happens when you dial. |
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| 10:01.13 | Chris-NB | hi |
| 10:01.21 | Chris-NB | anyone using snom with vlans? |
| 10:02.28 | Chris-NB | I've two vlans, one for voice, one for data. the phones should get the ip via dhcp. put they dont ... |
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| 10:02.45 | Chris-NB | do I have to enter the ID of the vlan into the phone to get an ip address via dhcp? |
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| 10:38.26 | fiddur | Hmm, asterisk svn branch 1.6.0 doesn't close sound files after Background... with a lot of menues, that leads to unability to open new sockets due to "Too many open files" after a while. |
| 10:38.28 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 10:47.01 | dan__t | That sucks. |
| 10:49.00 | fiddur | Yes. Really does. I'm trying to figure out where it should be closed... but perhaps I should just post a bug-report... |
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| 11:19.52 | *** join/#asterisk ultrav1olet (n=telnet@rtk-gw.prm.ru) |
| 11:21.21 | ultrav1olet | When I call someone else (Me -> Asterisk ServerA -> Asterisk ServerB -> SIP Provider -> Someone else) there's no sound and I get some error messages (Channel 'IAX2/intaster-13' unable to transfer; Channel 'IAX2/intaster-13' unable to transfer) - can anyone help? |
| 11:23.04 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
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| 11:23.59 | ultrav1olet | Me = Softphone (Zoiper) |
| 11:24.18 | ultrav1olet | between server A and server B there's a IAX2 channel |
| 11:25.59 | Milad | I want to pass a agent number to AGI which run after agent in Queue pickup phone, I check I did not find any normal way, I want to rewrite module app_queue but I don't know how pass for example member->membername to pbx_exec |
| 11:26.09 | Milad | anyone has a clue ? |
| 11:30.09 | kaldemar | ultrav1olet: notransfer=yes into iax.conf's |
| 11:31.13 | ultrav1olet | kaldemar: the problem is worse - I cannot even call from Server B directly |
| 11:32.06 | ultrav1olet | I mean I can call but there's absolutely no sound in any direction |
| 11:34.54 | kaldemar | is A<->B working? |
| 11:42.24 | kaldemar | is zoiper<->B working? |
| 11:55.03 | ultrav1olet | Now I have connected to ServerB directly and then I try to call a person behind ServerA - while he can hear me perfectly, I don't hear him at all |
| 11:55.54 | ultrav1olet | In a console I see this message (repeated twice): Operating with different codecs 4[0x4 (ulaw)] 2[0x2 (gsm)] , can't native bridge... |
| 12:09.56 | *** join/#asterisk BugKhaM (n=BugKhaM@125.25.86.40.adsl.dynamic.totbb.net) |
| 12:10.51 | BugKhaM | I have a user trying to register with wrong sip password. Is it possible to see that password from asterisk console? |
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| 12:22.16 | *** join/#asterisk kalib (n=kalib@200.253.26.159) |
| 12:22.47 | kalib | Hi people.. I'm having troubles.. I'm not receiving calls from the external world.. How can I check what's goin on? |
| 12:23.32 | kalib | when I do try to call to my number.. It looks like busy.. but it's not busy.. |
| 12:23.42 | ultrav1olet | kalib: asterisk -rvvvvv and see what's going on |
| 12:23.50 | nachox | guys, is it possible to use ZRTP+SRTP in asterisk? |
| 12:23.59 | kalib | ultrav1olet, I'm on it.. but I got no error message.. that's the problem |
| 12:24.06 | nachox | or some form of end to end encryption for the connection |
| 12:27.11 | kalib | ultrav1olet, I can call the external world... But I can't receive.. |
| 12:28.49 | jpmcallister | ultrav1olet: like kaldemar suggested, try notransfer=yes into iax.conf |
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| 12:37.01 | ultrav1olet | jpmcallister: in [general] or for interserver communications only? On both servers? |
| 12:37.13 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 12:37.14 | casix | hello |
| 12:38.22 | casix | hello, I wants that who recieve a call can transfer it. I have set the Tt options but if I call to a mobil number (using a sip trunk) asterisk don't transfer the call but it recieves the dtmfs from the mobil phone (I see its on the log file) what i'm doing wrong? |
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| 12:42.56 | ruben23 | hi |
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| 13:11.08 | ultrav1olet | all my problems are due to our ducking ISP :( |
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| 13:46.03 | boch | hello, anyone knows if phpagi-2.14 works fine with ast 1.6 ? |
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| 13:55.03 | guax | boch, yes, works fine for me |
| 13:56.39 | boch | do you know what permission needs the user to make calls using Originate command? cause the phpagi api is returning "permission denied" |
| 13:56.57 | boch | already give "call" permission but nothing |
| 13:57.52 | guax | humm, you are using ami not agi interface |
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| 13:59.52 | boch | guax, im using the phpagi API to connect to the asterisk manager interface, login is ok but cant perform Originate action |
| 14:00.16 | boch | manager debug is very poor |
| 14:02.25 | guax | Originate Call (privilege: call,all) |
| 14:02.32 | guax | boch, sure you gave the right permissions? |
| 14:03.07 | boch | yes, tried with call |
| 14:03.20 | boch | seems originate is the right privilege.. |
| 14:06.41 | guax | if you give read and write permission to call, i dont see why it returns permission denied, perhaps a wrong login, not sure |
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| 14:17.29 | Dovid | hi. i am using stream_file in an agi and the value never comes back to what I pressed in. |
| 14:17.52 | Dovid | I tried using get_data but that did pick up in my # entries |
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| 15:15.07 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 15:15.16 | Katty | ello |
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| 15:19.58 | casix | hello, I wants that who recieve a call can transfer it. I have set the Tt options but if I call to a mobil number (using a sip trunk) asterisk don't transfer the call but it recieves the dtmfs from the mobil phone (I see its on the log file) what i'm doing wrong? |
| 15:20.41 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:20.56 | *** join/#asterisk ang-st (n=ang-st@rsbac/developer/ang-st) |
| 15:21.05 | ang-st | bonjour |
| 15:21.12 | ang-st | sorry hello :) |
| 15:22.19 | ang-st | did someone already use asterisk thru openvpn ? |
| 15:22.32 | stintel | yes |
| 15:22.45 | *** join/#asterisk mrsci (n=mrsci@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
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| 15:23.34 | ang-st | i have strange problem: i can register, init call, even answer but i have no sound when i dial another client |
| 15:24.15 | ang-st | but when i call my test_call with a playback wav ... i can hear it |
| 15:25.48 | ang-st | and it only happend thru the vpn (it work like a charm in local) |
| 15:29.07 | *** part/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 15:31.06 | zeeesh | compiling Asterisk 1.4.19 .. after installation .. unable to c any modules save in /usr/lib/asterisk/modules direcotry .. could not found any error while compling .. can anybody guide why its happening? |
| 15:32.08 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 15:33.23 | icebrew54 | ang-st: what vpn type? |
| 15:33.47 | icebrew54 | ang-st: openvpn "scratches" up our asterisk connection |
| 15:36.26 | *** join/#asterisk freckle (n=chatzill@195.74.96.118) |
| 15:36.50 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:37.24 | freckle | I am trying to find away of marking a call in progress (possibly with a channel variable) so I can flag the call in post hangup processing, anyone have any ideas on how to do this? |
| 15:37.49 | Katty | ello |
| 15:38.06 | ang-st | icebrew54: openvpn ... |
| 15:38.33 | icebrew54 | ang-st: for our point to point, we switched to ipsec and had good luck |
| 15:38.54 | icebrew54 | I use iax2 overseas...180 ms ping....and it sounds like I'm calling next door |
| 15:39.06 | Kobaz | icebrew54: are you using tcp or udp |
| 15:39.21 | icebrew54 | icebrew54: I believe we were using udp |
| 15:39.24 | icebrew54 | err Kobaz :P |
| 15:39.33 | Kobaz | believe? |
| 15:39.43 | Kobaz | well if your using tcp, packet loss is going to destroy your voip |
| 15:40.08 | ang-st | icebrew54: i would prefer use openvpn but in case i can't manage to make it work maybe i'll switch to ipsec |
| 15:40.17 | Kobaz | make sure #1) openvpn is using udp #2) you've diable compression |
| 15:40.18 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-08ebf7e36a817977) |
| 15:40.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:40.31 | icebrew54 | yeah I'd check with Kobaz, he sounds like he knows a little more |
| 15:40.42 | icebrew54 | I know we had complications with it, so we switched to ipsec for performance benefits |
| 15:40.48 | Kobaz | ipsex |
| 15:40.56 | icebrew54 | decreased our ping by 20-30ms |
| 15:40.58 | Kobaz | ipsec is a royal pain in the ass, in general |
| 15:41.02 | icebrew54 | yeah it is... |
| 15:41.23 | ang-st | Kobaz: it can be compression so |
| 15:41.42 | Kobaz | compression will kill voip as well |
| 15:42.07 | ang-st | ok i'll disable it and try again |
| 15:42.09 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 15:42.11 | Kobaz | if you have any recurring packet loss, you will lose more voip packets than if you weren't compressing |
| 15:42.11 | ang-st | thanks for the point |
| 15:42.23 | Kobaz | compression will put 10 packets into one |
| 15:42.34 | Kobaz | so if you drop that packet, now 10 voip packets are gone |
| 15:42.38 | Kobaz | rather than just one |
| 15:42.43 | ang-st | at now i have no packet loss |
| 15:42.51 | Kobaz | so you will get long dropouts instead of a tiny little blip |
| 15:43.19 | Kobaz | yeah but you'll always have packet loss, given time approaching infinity |
| 15:43.36 | ang-st | lol sure |
| 15:43.47 | ang-st | yep but why it work with a playback() |
| 15:44.18 | ang-st | and not with a client ? |
| 15:44.20 | Kobaz | it sounds fine with playback, but not extension to extension? |
| 15:44.25 | ang-st | yep |
| 15:44.31 | Kobaz | sounds like codecs or buffering |
| 15:44.38 | *** join/#asterisk path_ (n=path@190.21.120.197) |
| 15:45.00 | ang-st | in debug rtp it seems no packet tx after the client answer |
| 15:45.24 | Kobaz | no rtp at all? |
| 15:45.38 | Kobaz | is this the same vpn conversation? |
| 15:45.40 | ang-st | 5/6 packet until hang up |
| 15:45.40 | Kobaz | or something new? |
| 15:45.40 | Kobaz | heh |
| 15:45.53 | Kobaz | mmm |
| 15:46.15 | Kobaz | with ipsec or openvpn? |
| 15:46.22 | ang-st | openvpn |
| 15:47.05 | Kobaz | have you ran some generic tools like mtr |
| 15:47.08 | *** join/#asterisk adr|an (n=xpl@unaffiliated/adrianxxx) |
| 15:47.21 | ang-st | not yet |
| 15:48.16 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 15:48.19 | ang-st | but as until answer every thing goes fine i thought it was rtp related |
| 15:48.28 | ang-st | (i'm quite new to asterisk= |
| 15:48.29 | ang-st | ) |
| 15:51.24 | ang-st | i'll put a wireshark to see if something strange occur |
| 15:51.32 | ang-st | (wrong route or so) |
| 15:52.04 | ang-st | anyway will try try also without compression |
| 15:52.12 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 15:52.14 | Kobaz | yeah |
| 15:52.43 | ang-st | thanks for all that points :) |
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| 15:52.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:52.53 | Kobaz | np |
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| 15:57.52 | *** join/#asterisk bcochofel (n=bcochofe@adonis.iportalmais.pt) |
| 15:58.45 | bcochofel | Hi, I'm trying to integrate asterisk with openser and I'm trying to put this under an oracle DB. Can anyone help me with the config for openserdbctl (v1.3) for unixodbc? |
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| 16:04.34 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 16:06.57 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 16:07.12 | *** join/#asterisk skirmisha (n=viki@90.154.203.57) |
| 16:08.12 | skirmisha | hello guys |
| 16:08.21 | skirmisha | i have strange problem with asterisk |
| 16:08.31 | skirmisha | it is doing a lot of re-transmisions |
| 16:08.38 | skirmisha | what could cause this problem? |
| 16:09.01 | manxpower | skirmisha: network problems cause that |
| 16:10.11 | skirmisha | could it be from STP ? |
| 16:10.35 | skirmisha | or could it be from nat config in asterisk? |
| 16:12.09 | manxpower | not sure what you mean by "STP", but yes, if it does not work at all then the retransmittions could be caused by NAT issues. |
| 16:15.02 | skirmisha | STP spanning tree protocol in switch |
| 16:15.54 | skirmisha | strange thing is that i can register, but options packages that are send by * to test if peer is alive are retransmited |
| 16:15.59 | skirmisha | and this is very strange |
| 16:16.00 | Kobaz | i always thought stp was stone temple pilots |
| 16:16.12 | skirmisha | sip_poke_noanswer is coming up |
| 16:16.16 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 16:19.52 | manxpower | registration = phone -> Asterisk. OPTIONS = Asterisk -> phone |
| 16:20.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:21.18 | skirmisha | yes correct |
| 16:21.28 | skirmisha | so when i set qualify to yes |
| 16:21.35 | skirmisha | i always see unreachable |
| 16:21.37 | manxpower | so you have two different directions, one works, one does not. |
| 16:21.42 | skirmisha | and sip debug shows retransmision |
| 16:21.46 | manxpower | that should tell you something right there. |
| 16:21.53 | manxpower | pastebin the output |
| 16:22.00 | *** join/#asterisk norwolf (n=aj@kontoret.n4f.no) |
| 16:23.12 | norwolf | Hi, I'm having a lot of problems with a newly purchased Wildcard TE122P in use in Norway (with euroisdn).. I've tried loads of different configs, but the status led blinks red no matter what config used.. any experiences with that? |
| 16:23.31 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:27.45 | manxpower | norwolf: pastebin the output of ztcfg -vvv |
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| 16:27.59 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:28.34 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:28.49 | norwolf | manxpower: http://pastebin.com/m232fb544 |
| 16:29.23 | norwolf | when running zttool, it reports Alarms: RED |
| 16:29.47 | boch | can i set an specified name server to asterisk ? |
| 16:30.01 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:30.48 | [TK]D-Fender | boch: ...HUH? |
| 16:30.54 | manxpower | <PROTECTED> |
| 16:31.01 | manxpower | boch: no. |
| 16:31.12 | boch | right |
| 16:32.03 | rwaite | hey all |
| 16:32.10 | norwolf | manxpower: I haven't got the command zap (this is a default installation of trixbox with the exception of some editor packages) |
| 16:32.49 | rob0 | ~trixbox |
| 16:32.50 | jbot | well, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:33.16 | norwolf | jbot: ah, ok :) I'll reinstall with a clean centos tomorrow then :) |
| 16:33.56 | norwolf | ah |
| 16:34.01 | norwolf | bot.. damn me :) |
| 16:35.02 | norwolf | something tells me installing asterisk after 30hours of coding isn't that smart.. better start fresh tomorrow. sorry for wasting your time until now :-) |
| 16:36.06 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 16:36.16 | rwaite | centos noooo, debian. |
| 16:36.24 | rwaite | centos leaves a dirty taste in my mouth |
| 16:37.03 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 16:37.03 | mchou | I am a bit mystified. I get no ringback after dialing to one extension (* says the extension is ringing) and the person on the other end has also verified the phone rings |
| 16:37.34 | manxpower | mchou: make sure you have a valid /etc/asterisk/indications.conf |
| 16:37.45 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:37.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:38.02 | mchou | manxpower: huh?? It works for all other extensions I've tried |
| 16:38.15 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 16:38.31 | mchou | only ONE extension doesnt seem to work right.... |
| 16:39.18 | manxpower | mchou: if the line gets answered before the Dial then Asterisk must use inband indications -- configured by indications.conf |
| 16:39.59 | manxpower | if not then you get messages on the console"Dont know inidcation 14" or something similar and the caller won't hear ringback tones. |
| 16:40.23 | rwaite | manxpower: could that also cause the problem i'm running into where it seems like the other person picks up, but i still hear ringing for a few seconds (while they hear silence) |
| 16:40.45 | manxpower | rwaite: no |
| 16:41.00 | manxpower | Maybe you need canreinvite=no |
| 16:41.09 | mchou | manxpower: no, I dont think the line gets answered. * says "Asterisk is making progress passing channel blah blah to channel foo bar...." |
| 16:41.15 | rwaite | my theory was it had something to do with how asterisk was 'emulating' the ring |
| 16:41.54 | manxpower | mchou: That is my suggestion. take it or leave it. |
| 16:41.58 | mchou | lol |
| 16:42.09 | mchou | take it or leave it? |
| 16:42.30 | manxpower | <-- does not work for Digium |
| 16:42.38 | mchou | manxpower: who do you think you are? St. Peter? |
| 16:42.41 | rwaite | what does 'progressinband' handle? |
| 16:42.53 | manxpower | mchou: No. I'm someone that volunteers his time to help people here. |
| 16:43.09 | manxpower | If you don't want to take my advice, that is fine. |
| 16:43.24 | mog | hugs manxpower |
| 16:43.28 | mchou | manxpower: right. Saying "take it or leave it" is superfluous |
| 16:43.33 | rwaite | manxpower has helped me a lot. |
| 16:43.48 | rwaite | so has tk |
| 16:44.08 | manxpower | rwaite: On SIP and PRI it tries (and many times fails) to provide inband indications |
| 16:44.16 | drmessano | mchou: Nonsense, that's hyperbole |
| 16:44.33 | rwaite | indications meaning ringing, busy tone, etc? |
| 16:44.42 | manxpower | correct |
| 16:44.45 | mchou | drmessano: well, whatever it is, manxpower is full of himself |
| 16:45.03 | rwaite | i saw a random mailing list post saying to use progressinband=no for a polycom phone, so i'm trying that |
| 16:45.33 | rwaite | well i'd hope he was full of himself. if he were full of someone else, that would be gross. mixing fluids and all that |
| 16:45.35 | mchou | drmessano: considering he didnt even bother to let me explain the "problem" beyond two sentences |
| 16:45.37 | wonderworld | damn, in 10 years the pstn will be based on random mailing list suggestions ;) |
| 16:45.38 | [TK]D-Fender | rwaite: Don't forget that your call has *2* legs |
| 16:46.06 | [TK]D-Fender | rwaite: While your outbound call my have OOB progress, your call from the phone to * may be ANSWERED and INBAND |
| 16:46.23 | [TK]D-Fender | rwaite: So * will still pass it back as indications.conf audio |
| 16:46.53 | [TK]D-Fender | wonderworld: Its already a century-old horse build by committee |
| 16:46.54 | rwaite | hmm. how would i force * to do it OOB? or is there no way |
| 16:47.37 | *** join/#asterisk ingenius (n=alektro@200.73.174.225) |
| 16:47.46 | [TK]D-Fender | rwaite: If you've answered the its game over. |
| 16:48.08 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 16:48.22 | rwaite | so * has a bias toward inband? |
| 16:48.25 | ingenius | Hi |
| 16:49.04 | manxpower | rwaite: no. |
| 16:49.32 | rwaite | in this doc, i see a lot of stuff like 'send 180 ringing' is this in the sip spec that i havent read yet? |
| 16:49.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:49.39 | rwaite | i need to start that. |
| 16:50.38 | manxpower | send 180 ringing is OOB |
| 16:50.51 | zeeesh | while compling getting this error: "/usr/bin/ld: cannot find -lcap collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1"? |
| 16:52.21 | manxpower | zeeesh: you must be doing something odd. asterisk does not use libcap |
| 16:52.58 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:55.50 | *** join/#asterisk n3hxs (n=HAMming@71.39.159.200) |
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| 16:59.00 | jaytee | file are you around? |
| 16:59.05 | Qwell | file: hide! |
| 16:59.09 | file | >_> |
| 16:59.10 | file | <_< |
| 16:59.29 | file | jaytee: yessssssssss? |
| 16:59.43 | jaytee | can you take a look at a grammar file and point me to the error of my ways? |
| 16:59.54 | file | I have not looked at grammars in months |
| 17:00.00 | file | and do not remember 'em :) |
| 17:00.16 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 17:00.19 | jaytee | oh, ok |
| 17:02.01 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:03.16 | ingenius | Hi guys ... i'm looking for an agi example do the following, call some number and play an audio file |
| 17:06.28 | wonderworld | ingenius: maybe you want to use call-files. generate a call file to call the number and send the call into an extension that simply does a Playback() and hangs up. |
| 17:07.44 | ingenius | wonderworld: sounds good .. let me check |
| 17:08.09 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 17:09.08 | *** join/#asterisk elred (i=sauron@fucksheep.org) |
| 17:09.12 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:10.27 | elred | Hello. Is it possible to force in my extensions.conf to return some SIP code. Like you do with Busy() function, but what I want is send back 480 (Temporarily not available) and 603 (Declined) ? Thanks |
| 17:11.09 | *** join/#asterisk Khratos (n=khratos@190.166.103.146) |
| 17:11.32 | Khratos | Good afternoon |
| 17:11.45 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:12.44 | ingenius | wonderworld: it's simple is like a call back service... why i don't thing this before.... my dead brain .. |
| 17:19.58 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 17:22.22 | rwaite | ingenius: agi is very simple, assuming you're familiar with the programming language you create the agi script with. i suggest you look at the agi chapters in the asterisk: the future of telephony book |
| 17:22.34 | rwaite | they'll get you up and running fairly quickly |
| 17:22.59 | ingenius | rwaite: thanks i will |
| 17:23.22 | rwaite | (it's freely available, too) |
| 17:24.04 | rwaite | http://cachefly.oreilly.com/books/9780596510480.pdf |
| 17:25.44 | ingenius | rwaite: Cool! :) |
| 17:26.01 | elred | damn I paid 30$ for it |
| 17:26.08 | elred | (for the online ebook copy) |
| 17:26.31 | Qwell | elred: what? |
| 17:26.56 | elred | Qwell : for the "Asterisk, the futur of telephony" book |
| 17:26.58 | Qwell | elred: Please explain |
| 17:27.07 | Qwell | From who? What were the conditions? |
| 17:27.13 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 17:27.35 | elred | I have buy it from the oreilly.com online webpage, I ignored it was freely available |
| 17:27.39 | Qwell | oh |
| 17:27.47 | elred | anyway it's a very nice book to begin with, not a waste of money |
| 17:28.14 | Qwell | elred: I just wanted to make sure the money didn't go somewhere besides O'Reilly/the authors |
| 17:28.18 | Qwell | that would have been bad |
| 17:29.03 | Juggie | how would you dial distincitively on an outbound call? |
| 17:30.41 | rwaite | with a nice suite, and some tea? |
| 17:30.46 | rwaite | raised pinky, or course |
| 17:30.50 | Juggie | hah :) |
| 17:30.57 | rwaite | wow look at my typing today. |
| 17:31.00 | elred | well, nobody know how can I force in my extensions.conf to return a choosed SIP code ? (I need to send a 603 -declined-) |
| 17:31.29 | manxpower | elred: You can't. |
| 17:31.43 | manxpower | Asterisk is not designed to do protocol specific things. |
| 17:31.49 | Juggie | any ideas, eg, if a certain number called you, you could ring the phone a certain way. |
| 17:32.14 | manxpower | You may be able to figure out what AST_HANGUPCAUSE maps to 603 declined, but that's about it. Those sorts of things are the job of a SIP Proxy, not a B2BUA (like Asterisk) |
| 17:32.15 | rwaite | methinks that would make it too easy to break the protocol rolling your own stuff |
| 17:32.47 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:33.40 | elred | manxpower : ok, I understand. Thanks a lot |
| 17:33.59 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:34.12 | rwaite | is OpenSIPS meant to replace asterisk? the site says it includes 'application-level functionalies' |
| 17:34.53 | manxpower | rwaite: no. |
| 17:35.04 | manxpower | if it's not SIP then openSIPs won't handle it. |
| 17:35.33 | elred | I have no idee how to write a dialplan using OpenSips |
| 17:36.16 | elred | OpenSIPs is good as a SIP proxy, I think it better stay it. |
| 17:38.28 | manxpower | you don't really write "dialplans" in OpenSIPs, you just route calls and audio based on rules and SIP headers. |
| 17:39.05 | rwaite | hmm. |
| 17:40.07 | manxpower | http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/ |
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| 17:48.38 | rob0 | tzafrir_laptop: update re: dahdi-linux in 2.6.{27.7,28.2}: I built my kernel with CONFIG_HIGH_RES_TIMERS and CONFIG_HZ=1000, but I still didn't get the needed symbols ('rtc_((un)?register|control)'), although they're in the rtc.c code. |
| 17:48.43 | *** join/#asterisk sack (n=sack@158.Red-79-148-189.dynamicIP.rima-tde.net) |
| 17:49.17 | tzafrir_laptop | rob0, it shouldn't be looking for RTC in the first place |
| 17:49.23 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 17:49.28 | rob0 | I'll have to figure out why those symbols are not being exported. |
| 17:49.40 | tzafrir_laptop | Are you sure it is built vs. the right kernel source tree? |
| 17:51.27 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:53.04 | rob0 | I didn't even try dahdi-linux on this yet, since I don't have the symbols it said it needs. |
| 17:58.39 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 17:59.34 | kannan | hello, on faxing eith asterisk on t.38, is there any reliable (as in telco grade) solution at all, or is it better to go for PSTN faxing only with hylafax. We need to large volume faxing so its important for me to decide correctly. With our current voip provider we are failing many faxes |
| 18:00.19 | rwaite | Hmm. What would the best way to have a line (the receptionist) send calls to a queue but only if the receptionist has done something to indicate that she will not be in? |
| 18:00.49 | manxpower | rwaite: first define what she does to inidcate that |
| 18:00.55 | rwaite | My first thought was to have her set her phone to dnd, but would i be able to differentiate between that and her just being on a call? |
| 18:01.28 | manxpower | rwaite: set up extension to let her call to set an astdb flag, then check that flag before routing the call to her |
| 18:01.37 | rwaite | well i was thinking of rolling my own thing, so when she dials *657567567 it will create a touch file in /tmp, but that seems fragile |
| 18:01.48 | rwaite | oic |
| 18:01.59 | rwaite | good thinking |
| 18:02.04 | *** join/#asterisk Greek-Boy (n=greek@41.222.92.254) |
| 18:02.28 | manxpower | *2929 (*AWAY) |
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| 18:04.20 | rwaite | my only concern with that is then she'll need to do it again to un-away herself. but with no visual indication on the phone that she's away, i'm afraid she'll forget. |
| 18:04.29 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:05.07 | rwaite | maybe if i test for the same astdb key and give her an error on any outbound calls |
| 18:05.17 | rwaite | or automatically mark her back if she makes any calls. |
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| 18:08.02 | rwaite | hmm. or maybe just mark her back on the following day. |
| 18:08.08 | rwaite | so many possibilities. |
| 18:09.50 | manxpower | You can if(away = yes) { ring phone for 1 second then send to queue } |
| 18:10.14 | [TK]D-Fender | manxpower: I often do *86 fo *VM for outside hidden VM access via primary IVR's |
| 18:10.28 | manxpower | that way the phone gives one ring only at the operator phone. Think of it as a "reminder ring" like you get from the telco when you get a call on a line that has callforward enabled. |
| 18:10.51 | rwaite | ok, so always ring the phone. if she's there, she should obviously answer |
| 18:10.55 | *** part/#asterisk kalib (n=kalib@200.253.26.159) |
| 18:11.07 | manxpower | rwaite: nope, make it so short she can't answer |
| 18:11.18 | rwaite | i see |
| 18:11.34 | rwaite | to force her to unaway herself |
| 18:11.38 | manxpower | they annoyance should remind her to set her status as available |
| 18:12.16 | manxpower | also you can have the EXACT same dialplan stuff for avail .vs. away, just change the timeout |
| 18:12.37 | rwaite | so if i set a 1s timeout, it should do the quick ring? |
| 18:12.47 | manxpower | rwaite: experiment with it |
| 18:12.52 | manxpower | it all depends on the phone. |
| 18:12.54 | rwaite | i will. cool. |
| 18:14.58 | *** join/#asterisk mrsci (n=mrsci@166.128.97.85) |
| 18:16.57 | rwaite | man qos is overrated |
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| 18:22.22 | workdraft | tried this tutorial. http://www.asteriskguru.com/tutorials/asterisk_gui.html but still doesnt work or accessible via -p 8088 |
| 18:23.13 | manxpower | maybe you can ask on the asterisk gui channel |
| 18:30.16 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 18:30.36 | rwaite | i am very bad at bandwidth. i always think ulaw uses this ungodly amount of b/w, but really it uses like 6Kb, which is nothing really |
| 18:31.02 | Qwell | rwaite: ~64kbit/s plus overhead |
| 18:31.13 | rwaite | or wait, i think i even got that wrong |
| 18:31.56 | rwaite | so 64kbit would be about 8kilobytes? |
| 18:32.02 | manxpower | 64Kbps is 1/24th of a T-1 |
| 18:32.10 | manxpower | well kbps at l;east |
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| 18:32.34 | rwaite | so on a cable connection, i should be able to handle tons of ulaw connections |
| 18:32.39 | manxpower | correct |
| 18:32.40 | rwaite | theoretically |
| 18:33.03 | harry_v | if your isp does not limit your rtp |
| 18:33.14 | rwaite | rtp? |
| 18:33.35 | Gido-E | rwaite on the internet your ulaw is only an rtp tunnel |
| 18:33.45 | Qwell | Gido-E: ...what? |
| 18:33.53 | CoffeeIV | I have an asterisk office phone system setup, and recently people have moved from using SIP phones on their desks to having their internal extension forwarded to their cell phone. The problem is they want features like transfering calls, and I would think you could do that with a hook flash, but I don't even know how to send a hook flash from a cellphone. How do other people handle this ? |
| 18:34.16 | rwaite | well no, because this is iax. and if my isp limited anything of the sort, i'd be getting a new isp |
| 18:34.16 | manxpower | rwaite: RTP == audio protocol for most VoIP protocols |
| 18:34.41 | rwaite | that's what i thought |
| 18:34.52 | Qwell | what exactly is an rtp tunnel? |
| 18:35.22 | Gido-E | Qwell for the tcp/ip model it is only the applicatioln layer |
| 18:35.41 | rwaite | actually i think tunnel has a very specific and well-defined meaning |
| 18:36.00 | jaytee | "it's a series of tubes!" |
| 18:36.09 | rwaite | it's not a big truck |
| 18:36.21 | rwaite | you can't just dump stuff in it |
| 18:36.41 | jaytee | moles are in my backyard :-( |
| 18:38.52 | bmoraca | bunch of savages on the internet...arg |
| 18:39.24 | bmoraca | 3 of my servers have been the target of an SSH botnet trying to brute-force my password...woo... |
| 18:40.54 | drmessano | Erm |
| 18:40.56 | drmessano | sorry about that |
| 18:41.00 | drmessano | I just turned one off |
| 18:41.36 | jaytee | "Hakka Palle!!!!" |
| 18:42.46 | *** join/#asterisk endemic (n=endemic@orion.onvox.net) [NETSPLIT VICTIM] |
| 18:43.20 | *** join/#asterisk dsp2877 (n=ds@118.100.13.60) |
| 18:43.27 | dsp2877 | hi all |
| 18:43.29 | drmessano | Sorry, been a bad day |
| 18:43.36 | jaytee | same here |
| 18:43.49 | drmessano | I may be getting my gall bladder sucked out |
| 18:44.05 | dsp2877 | am trying to compile zaptel 1.4.21 with a 2.6.26.8 kernel but have some syntax errors, is it not compatible with newer kernels is it |
| 18:44.26 | jaytee | they suck them out now? i thought they cut them out? |
| 18:44.50 | drmessano | 2 holes now I think.. little cutting, little SLURP and its gone |
| 18:44.53 | errr | bmoraca, you should setup that pam module that works with ssh and iptables that if you get the passwd wrong x number of trys it will block incoming connections from the ip for x ammount of time |
| 18:45.05 | bmoraca | i am using PAM |
| 18:45.12 | rwaite | dsp2877: err, no i actually think it should be fine |
| 18:45.32 | rwaite | speaking of kernels. i need to compile 2.6.27 sometime soon now |
| 18:45.41 | bmoraca | i just set hosts.allow to deny everyone...i don't need to use SSH to access the servers anyway |
| 18:46.04 | jaytee | drmessano, sorry to hear about that. hope everything works out ok |
| 18:46.11 | errr | running ssh on a non standard port helps that too |
| 18:46.40 | drmessano | jaytee: Should be fine. Just pissed me off.. I had made a commitment to my Gall Bladder. Sad that we wont be making it to the finish line together :( |
| 18:46.43 | dsp2877 | rwaite : am actually trying to just compile ztdummy |
| 18:47.01 | drmessano | jaytee: I feel like I am losing a part of me. Wait, I am |
| 18:47.14 | dsp2877 | not sure why its giving some error at In function a??init_modulea??: |
| 18:47.50 | tzafrir_laptop | dsp2877, what errors? |
| 18:48.05 | jaytee | drmessano, you should have listened to your mother when she told you not to swallow your gum. |
| 18:48.05 | tzafrir_laptop | Do you mean zaptel 1.4.12 ? 1.4.12.1 ? |
| 18:48.39 | dsp2877 | sorry tzafrir , your right its 1.4.12 |
| 18:48.45 | dsp2877 | 1.4.12.1 |
| 18:48.58 | dsp2877 | i have a 1.4.21 asterisk installation just getting confused tehre |
| 18:49.14 | drmessano | jaytee: I got my lessons from Mom mixed up.. apparently had that one and the birds and the bees confused. That poor girl I had my first time with. I bet she still tells people about it. |
| 18:49.57 | dsp2877 | tzafrir : i started with configure, all was fine there. then i did make menuselect to select only ztdummy, then make , |
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| 18:50.26 | *** part/#asterisk pyite[A] (n=pyite@c-69-181-46-99.hsd1.ca.comcast.net) |
| 18:50.26 | dsp2877 | during the make i get some errors |
| 18:50.28 | dsp2877 | make[3]: *** [/usr/src/zaptel-1.4.12.1/kernel/ztdummy.o] Error 1 |
| 18:50.28 | dsp2877 | make[2]: *** [_module_/usr/src/zaptel-1.4.12.1/kernel] Error 2 |
| 18:50.28 | dsp2877 | make[2]: Leaving directory `/usr/src/kernels/linux-2.6.26.8-rt15' |
| 18:50.40 | dsp2877 | sorry for the pasting |
| 18:50.44 | *** join/#asterisk kc2tnk (n=fskrotzk@host198.textwise.com) |
| 18:50.56 | jaytee | drmessano, with those lewd and lascivious thoughts it's a sure thing you're going to Hell. Rest easy though! I'll save you a good seat :-) |
| 18:51.03 | rwaite | those lines tell us nothing, only that it failed while compiling the ztdummy.o object |
| 18:51.09 | dsp2877 | ok.. |
| 18:51.10 | rwaite | paste the full output to a pastebin |
| 18:51.18 | dsp2877 | ok one sec |
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| 18:52.21 | dsp2877 | http://pastebin.ca/1325563 |
| 18:52.27 | drmessano | jaytee: If theres anyone I want to rot in hell with, dude, I am so there with you.. Minus the rainbow laced rollerskates. |
| 18:52.32 | dsp2877 | just the start from the point of configure till the make |
| 18:52.59 | jaytee | drmessano, lol |
| 18:53.54 | drmessano | jaytee: Nevermind, i'm in for the rollerskates too.. and 24/7 Skating Rink songs from the 80s |
| 18:54.21 | rwaite | well. first i would look at ztdummy.c, line 320 |
| 18:54.43 | rwaite | maybe there is a bug in that file? |
| 18:54.48 | tzafrir_laptop | dsp2877, and what were the errors right above it? |
| 18:56.03 | dsp2877 | tzafrir : thats the only errors there it nothing else above it just the compile information |
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| 18:57.13 | dsp2877 | rwaite : line 320 of ztdummy.c just says : ztd->counter = 0; |
| 18:57.25 | dsp2877 | not sure what that means |
| 18:57.26 | rwaite | hold up |
| 18:57.26 | RypPn | dsp2877 for me zaptel-1.4.12 wont build on a kernel newer than 2.6.25 |
| 18:57.49 | dsp2877 | ryppn: okay i think its the same here as well |
| 18:59.00 | RypPn | saying that I'm using dahdi now and I'm still on 2.6.25 cos it barfs on newer versions, but thats more to do with wanpipe sucking |
| 18:59.02 | rwaite | that sucks |
| 19:00.01 | dsp2877 | hmm |
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| 19:10.29 | kfife | Can someone give me a pointer: I'm trying to update my hardphone's display using sip update. Can someone point me to the asterisk syntax for such a thing? I'm stumped. |
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| 19:16.07 | rwaite | kfife: no clue, i spent a few afternoons trying to get that on my polycom phones and i could not get it to work. |
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| 19:19.41 | kfife | rwaite: thanks. The idea is to use sip update to populate the CNAM on outdial. I know it can be done, because ZipDX does it, |
| 19:21.01 | kfife | rwaite: ... it's a great idea for a number of reasons, including confirming that you have not misdialed, and it makes it easier to manage parties in an ad-hoc conference using a sip hard phone |
| 19:22.02 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 19:23.28 | kfife | Anybody know which version of asterisk will drop its dependency on span dsp for app_fax? |
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| 19:34.49 | kfife | The channel is awfully quiet today! |
| 19:35.04 | mog | dances |
| 19:35.11 | rwaite | every day i get less sleep, and i sit here at work and everything just blurs |
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| 20:06.46 | Katty | )= |
| 20:06.48 | Katty | i'm bummed. |
| 20:06.58 | Katty | 2.6 Million lay offs in 2008 |
| 20:07.07 | Katty | rioting in France and UK over economy hardships. |
| 20:07.30 | Katty | North Korea has just abandoned a few peace treaties with South Korea |
| 20:07.43 | Katty | Russia threatening to leave the UN over propetry disputes of the north pole |
| 20:08.27 | Katty | not to mention the clashing of civil and religious bliefs that are running rampant all over the place |
| 20:08.44 | jasonwoot | Relax.... Obama is our white knight... he's going to fix everything |
| 20:09.04 | jasonwoot | I could be wrong, but I'm pretty sure he is both the way, and the light |
| 20:09.28 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 20:09.30 | Katty | I'm not sure Obama's going to fix anything. |
| 20:09.34 | Katty | I imagine he really wants to. |
| 20:09.52 | Katty | but what can obama really do to straighten out North Korea and South Korea? |
| 20:09.56 | Katty | if they go to war, china will get involved |
| 20:09.58 | Katty | and that will just be Bad |
| 20:10.52 | russellb | pretty sure there might be a #politics, or #depressing-discussion that would welcome this topic |
| 20:10.53 | russellb | :-p |
| 20:10.59 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 20:11.26 | Katty | :< |
| 20:11.31 | jasonwoot | I kid Katty, I kid.... the shit sandwich is still flying through the air, and hasn't reached apogee yet |
| 20:11.32 | rob0 | NK & SK both really want peace with one another. And big bux from abroad. |
| 20:11.37 | kaldemar | would videoconferencing be as depressing a subject? |
| 20:11.48 | Katty | i think it'd be worse, kaldemar |
| 20:11.59 | russellb | it might be depressing, but at least it is about asterisk :) |
| 20:12.04 | Katty | i honestly dunno how we're gonna fix this mess. |
| 20:12.17 | Katty | the unemployment rate hasn't been this low since WWII |
| 20:12.30 | kfife | Can someone give me a pointer: I'm trying to update my hardphone's display using sip update. Can someone point me to the asterisk syntax for such a thing? I'm stumped. |
| 20:12.43 | bmoraca | obama is only going to make things worse. entitlement programs, big government, and government spending is what got us IN to this mess. having more of it is not going to bring us out. |
| 20:12.49 | Katty | damn you reaganomics!!! |
| 20:12.54 | russellb | stop the politics discussion now please |
| 20:12.58 | kfife | The idea is to use sip update to populate the CNAM on outdial. |
| 20:13.09 | bmoraca | sorry :) |
| 20:13.17 | Katty | russellb: why? don't you want to be mopey with me? |
| 20:13.35 | cvnet | can u have a sip users be identified by its IP and not user/pass ?> |
| 20:13.35 | russellb | kfife: you may want to check out issue 8824 on bugs.digium.com .... there is a branch that supports connected party info updating |
| 20:13.43 | bmoraca | political discussion on the internet is never a good idea |
| 20:13.47 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:13.49 | jasonwoot | I can be mopey AND talk about asterisk... to me, they are closely related |
| 20:13.58 | kfife | russellb: thanks! |
| 20:14.25 | bmoraca | cvnet: yes...but why? |
| 20:14.44 | cvnet | I need to set it up, how do I do that? |
| 20:14.54 | bmoraca | use the "host" setting |
| 20:15.15 | cvnet | hum ok, let me do some reading on host, thanks |
| 20:15.32 | bmoraca | cvnet: it's in sip.conf...check the wiki |
| 20:15.42 | beek | kfife: In one of the after-conference VUC sessions you had mentioned a click-to-dial app for Windows that you liked. What was that? |
| 20:15.45 | cvnet | pl thamx |
| 20:16.01 | *** join/#asterisk maqr (n=maqr@httpcraft/hax) |
| 20:16.08 | cvnet | ok thanks |
| 20:16.29 | bmoraca | i love it when people put their finger in the wrong place when they type...it's so much fun |
| 20:17.02 | *** join/#asterisk john_fbac (n=john_fba@75.sub-70-220-144.myvzw.com) |
| 20:17.37 | *** join/#asterisk workdraft (n=iyanski@203.215.94.239) |
| 20:18.24 | john_fbac | I'm trying to put a plan together for an upgrade from asterisk 1.2.24 to 1.4 or higher. what is this transition like; what things do I need to prepare for? |
| 20:18.54 | [TK]D-Fender | john_fbac: Everything breaking |
| 20:19.33 | john_fbac | :) that's what I figured. is it best to just start over and build from scratch, rather than trying to upgrade? |
| 20:19.34 | [TK]D-Fender | john_fbac: You know that asking what you just did is like asking us to read the UPGRADE.TXT file to you line by line over IRC... right? |
| 20:19.36 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 20:19.42 | [TK]D-Fender | john_fbac: Read the damn docs :p |
| 20:20.09 | john_fbac | I deserved that, thx. |
| 20:20.09 | [TK]D-Fender | john_fbac: Which means go download the version you're looking at, make sure you aren't doing things the 1.0 way anywhere or you're relly in trouble, etc |
| 20:20.56 | [TK]D-Fender | john_fbac: Plenty of info right in the tarball... |
| 20:20.56 | workdraft | any recommended ip phones for less than a hundred units? |
| 20:20.56 | john_fbac | okay. thank you! |
| 20:20.56 | [TK]D-Fender | workdraft: Same as those for more than 100 units :) |
| 20:21.14 | workdraft | any recommendations? or any favorites? |
| 20:21.28 | maqr | thought not strictly asterisk-related, maybe someone here can point me in the right direction... i have a polycom phone which until today worked fine, but now can't acquire a dhcp address. has anyone seen this kind of behavior before? |
| 20:21.36 | [TK]D-Fender | workdraft: Given your area Linksys is probably the most cost effective choice. |
| 20:21.55 | [TK]D-Fender | workdraft: Polycom is a better product, but IIRC the margin isn't work it there. |
| 20:22.09 | kfife | beek: That must have been ADA |
| 20:22.10 | [TK]D-Fender | maqr: VLAN issue perhaps? |
| 20:22.17 | workdraft | ok. thnx for the inputs |
| 20:22.24 | beek | kfife: That's the one that Digium now owns? |
| 20:22.38 | maqr | [TK]D-Fender: i don't see how, everything is identical to the way it was before, it's just hooked up to a little linksys home router |
| 20:23.06 | [TK]D-Fender | maqr: So the phone says it failed to pull an IP? |
| 20:23.08 | maqr | [TK]D-Fender: the LED says it's plugged in, and unfortunately i don't have a good way of getting between the linksys and the phone to sniff out the dhcp traffic |
| 20:23.09 | kfife | beek: Correct. I was originally looking for a windows shell extension type implementation, but with ADA's TAPI support it may be just as effective. |
| 20:23.16 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 20:23.39 | maqr | [TK]D-Fender: yeah, i picked it up to make a call and got a fast busy when i tried to dial anything, so i pulled the AC to reboot it, plugged it back in, and now it won't get an IP |
| 20:23.51 | beek | kfife: I am using noojee's firefox plugin, but I'm interested in something integrated in 'doze. Thanks for the info! |
| 20:24.02 | kfife | beek: NP! |
| 20:24.03 | maqr | [TK]D-Fender: it's not in the linksys 'dhcp table' on the router even |
| 20:24.17 | [TK]D-Fender | maqr: that isn't what I'd call "real" |
| 20:24.26 | [TK]D-Fender | maqr: Look on the PHONE ITSELF |
| 20:24.32 | kfife | Anyone know which version of Asterisk will drop its dependence on SpanDSP for app_fax? |
| 20:24.44 | [TK]D-Fender | kfife: Odss are... NONE |
| 20:24.47 | [TK]D-Fender | Odds* |
| 20:24.57 | *** join/#asterisk mnicholson_ (n=matthew@adsl-163-40-99.hsv.bellsouth.net) |
| 20:24.57 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 20:24.58 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 20:24.58 | *** join/#asterisk HeMan (n=jimmy@193.12.106.19) |
| 20:24.58 | *** join/#asterisk vader-- (n=me@c-71-225-195-86.hsd1.nj.comcast.net) |
| 20:24.58 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 20:25.07 | maqr | [TK]D-Fender: it says "Server Address Resolving..." , then if i click "next", it says "IP Address Resolving...", and the red light blinks a lot |
| 20:25.14 | [TK]D-Fender | kfife: Who do you think at Digium is expert enough at DSP's that will dedicate the time to reinvent this wheel? |
| 20:25.24 | [TK]D-Fender | maqr: "click"? Pardon? |
| 20:25.29 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 20:25.52 | maqr | [TK]D-Fender: well, there's 3 buttons, two of them have words above them on the display... one is 'prev' and the other is 'next' |
| 20:26.20 | maqr | [TK]D-Fender: maybe 'push' is the right word to use, but i'm activating the buttons for 'next' and 'prev' to see the status of the phone, and it's definitely inidicating it isn't getting an IP |
| 20:26.23 | [TK]D-Fender | maqr: That sounds more like a PROVISIONING issue... not a DHCP issue |
| 20:27.11 | kfife | [TK]D-Fender: It surprised me when I heard it, but I believe it was Steve Sokal in a visit with the VOIP Users Conference. |
| 20:27.21 | maqr | [TK]D-Fender: when i first turned the phone on, months ago, it got an IP automatically and i set it up via the web interface |
| 20:27.31 | kfife | IIRC |
| 20:27.53 | [TK]D-Fender | maqr: If you see the web interface what does that tell you about LOOKING at it again? Where did it get that IP from if you've succeeded in going back in? |
| 20:28.02 | [TK]D-Fender | maqr: I said ON THE PHONE. |
| 20:28.13 | [TK]D-Fender | maqr: As in the LCD menu, not some web interface. |
| 20:29.47 | maqr | [TK]D-Fender: i can't get to the web interface now... and actually, i think i found the issue, i plugged in another dhcp device and that won't get an address either... must be something messed up with the linksys |
| 20:29.55 | kfife | [TK]D-Fender: Perhaps it would be forked and included with the distribution? Not sure. Like I said, I was surprised. |
| 20:30.06 | [TK]D-Fender | maqr: Good... no fix your router :) |
| 20:30.08 | [TK]D-Fender | now* |
| 20:30.16 | rue_work | :) |
| 20:30.25 | [TK]D-Fender | rue_work: .... |
| 20:30.29 | [TK]D-Fender | TRABAJO |
| 20:30.38 | [TK]D-Fender | channel eppigy |
| 20:30.41 | [TK]D-Fender | (s) |
| 20:30.55 | rue_work | what does TRABAJO stand for? |
| 20:31.02 | maqr | [TK]D-Fender: thanks, sometimes i have to talk it out :p |
| 20:31.09 | rue_work | no M so its not manual related |
| 20:31.25 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 20:31.30 | [TK]D-Fender | rue_work: Go translate... we all had to to make sense of his ramblings :0 |
| 20:32.08 | rue_work | Thats Rediculous And Bad As Jerry's Output? |
| 20:32.52 | bmoraca | so, rue_work, how's your HWEC working out? |
| 20:33.01 | rue_work | its not |
| 20:33.12 | rue_work | would prolly help if I had it plugged in |
| 20:33.24 | rue_work | but I cant right now |
| 20:33.29 | bmoraca | oh? |
| 20:33.37 | rue_work | I havn't recieved it yet |
| 20:33.47 | bmoraca | wow, long shipping delay |
| 20:34.04 | rue_work | I'm in a town in west canada, it takes a LONG time for things to get here |
| 20:34.09 | rue_work | I expect about 2 weeks |
| 20:34.26 | bmoraca | ahhh |
| 20:35.05 | rue_work | I wish oslec were more usable |
| 20:35.51 | [TK]D-Fender | rue_work: CRAZY. Purolator can't possibly take more than 5 days TOPS. |
| 20:36.36 | rue_work | this aint the big city |
| 20:36.39 | rue_work | "" |
| 20:36.49 | [TK]D-Fender | rue_work: thats why I sad 5 and not 2 <- |
| 20:36.51 | [TK]D-Fender | said* |
| 20:37.33 | rue_work | the only comany that can get things here in less than a week seeme to be digikey, somehow they get stuff here in 2 days |
| 20:37.52 | rue_work | nobody knows how they do it |
| 20:38.05 | [TK]D-Fender | rue_work: What city exactly? |
| 20:38.21 | rue_work | well, you will be able to find sechelt on a map |
| 20:38.27 | rue_work | coast of bc canada |
| 20:39.21 | [TK]D-Fender | rue_work: I see it... |
| 20:39.33 | [TK]D-Fender | rue_work: Yup... its a truck-stop. |
| 20:39.53 | rue_work | not really, ferry on either end |
| 20:39.54 | [TK]D-Fender | rue_work: Still.. Puro can't take more than 5 days.. that'd be crazy |
| 20:40.10 | [TK]D-Fender | rue_work: Ferry, or that long northern detour |
| 20:40.35 | *** join/#asterisk aatmaa_ (i=aatma@118.103.233.15) |
| 20:40.42 | [TK]D-Fender | rue_work: Which... yeah.. that'd suck |
| 20:40.51 | rue_work | the equip is comming out of east canada, "Williams" is the ONLY canadian place I was able to find that could provide digium equip there is nothing on the west coast |
| 20:41.34 | [TK]D-Fender | rue_work: Yeah they are the Canadian distributor... see if you went with Sangoma, they're right out of Markham and you could have gone direct :) |
| 20:41.45 | rue_work | ok, are my configs ready for the hwec? |
| 20:42.05 | rue_work | markham? |
| 20:42.31 | [TK]D-Fender | ON |
| 20:42.39 | rue_work | ah |
| 20:42.59 | [TK]D-Fender | rue_work: Yes, the card only requires "echocancel=yes" the driver does the rest. |
| 20:42.59 | rue_work | whats sangoma? |
| 20:43.12 | [TK]D-Fender | rue_work: .... |
| 20:43.14 | rue_work | hmmm |
| 20:43.53 | rue_work | is that a knockoff or an alternative |
| 20:44.00 | bmoraca | i like sangoma cards, but it seems like wanrouter is a weird driver model |
| 20:44.07 | bmoraca | rue_work: alternative |
| 20:44.32 | aatmaa_ | have anyone used vicidialer. Does it costs money ? |
| 20:44.51 | kannan | aatmaa_, i am using |
| 20:44.55 | kannan | its free |
| 20:45.09 | kannan | see #vicidial |
| 20:45.21 | rue_work | looks like sangoma is twice the price |
| 20:46.05 | rue_work | I dont like the way they do card ganging, I'm trying to keep the pc's small |
| 20:46.29 | rue_work | (this one is an EVO) |
| 20:46.38 | [TK]D-Fender | rue_work: Same price pretty much |
| 20:47.06 | bmoraca | they have an 8 port PRI card |
| 20:47.15 | [TK]D-Fender | yup |
| 20:47.22 | rue_work | dont want pri, not available here |
| 20:47.29 | bmoraca | i know, i was just saying |
| 20:47.40 | [TK]D-Fender | rue_work: FFS its a miracle you HAVE POTS at all and aren't on sat-phones :p |
| 20:47.41 | rue_work | :) |
| 20:47.51 | bmoraca | lol |
| 20:48.05 | rue_work | too many people still using rotary |
| 20:48.21 | [TK]D-Fender | bmoraca: I'd say that rue_work is "just left of nowhere".... but he's got OCEAN on the left :p |
| 20:48.31 | rue_work | haha |
| 20:49.04 | [TK]D-Fender | rue_work: You've gained a little pity from me at least... |
| 20:49.23 | rue_work | ok, I need to get one phone to register again (I offended it) and need to tune dialplans |
| 20:49.41 | seanmh | I have a 7960 where if I have the CFwdALL set to my cell phone and I don't pickup my cellphone Asterisk voicemail picksup as opposed to my cell phone voicemail. Any idea how to change this? |
| 20:49.50 | seanmh | 7960 with a SIP load, btw |
| 20:50.25 | [TK]D-Fender | seanmh: If * VM picks up its because * gave up ringing before your Cell carrier did. |
| 20:50.34 | [TK]D-Fender | seanmh: Change your dial timeout |
| 20:53.15 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 20:55.47 | seanmh | [TK]D-Fender, I changed it to 120 seconds and it still picks up voicemail a lot sooner |
| 20:56.03 | *** join/#asterisk Jeff_Phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 20:56.10 | Jeff_Phillips | Hello |
| 20:57.21 | rue_work | ok aastra question, there are 3 lines in the config for line 1 'phone number' 'caller id' and 'auth name' it SEEMS if I change ANY of them, the phone is unable to log into asterisk, I need/want calls from the phone to come up with a name, not 14 (the account name) |
| 20:59.37 | rue_work | I have found that the aastra is REALLY bad at naming paramiters what their really for. |
| 21:00.16 | rue_work | it almost seems that they just grab them at random, but within limits |
| 21:00.16 | Jeff_Phillips | I have an issue where if I try to dial *2 or ## to transfer a call, the DTMF tone "hangs" such that the caller thinks I'm pressing and holding the button indefinately until they hang up, and the system fails to recognize the key sequence as being a valid feature code so I am unable to transfer the call |
| 21:00.47 | rue_work | Jeff_Phillips, does the tone come from the phone or asterisk |
| 21:00.52 | rue_work | analog phone? |
| 21:01.13 | Jeff_Phillips | They are analog phones connected through an Audiocodes MP-124 24-channel FXS gateway |
| 21:01.31 | rue_work | I wasn't able to get me analog phones to trigger transfers on anything but a hook flash |
| 21:01.31 | Jeff_Phillips | I've also seen it on one of the two extensions I have wired up using an SPA-2000 |
| 21:01.43 | rue_work | ok, if the tone sticks its the phone, has to be. |
| 21:01.56 | Jeff_Phillips | no it only sticks on one end of the call |
| 21:02.00 | rue_work | sounds like my setup at home, I'm using a mainstreet channelbank |
| 21:02.30 | rue_work | I had to use hookflash for transfers |
| 21:02.34 | Jeff_Phillips | If I place an outbound call I have no problem navigating touch tone menus of various businesses |
| 21:02.39 | rue_work | I suggest you use hookflash |
| 21:02.41 | Jeff_Phillips | how do you use the hookflash for transfers? |
| 21:02.46 | Jeff_Phillips | I thought I had to dial the feature codes |
| 21:02.57 | Jeff_Phillips | i tried using a hookflash and lost the call but I figured I did it wrong |
| 21:03.05 | [TK]D-Fender | seanmh: Some more details would help and I trust nothing without a clear pastbin... |
| 21:03.10 | rue_work | you flahs the hook and it takes you to you pickup context, you dial your extension and hang up |
| 21:03.27 | Jeff_Phillips | Can you do an attended transfer somehow this way? |
| 21:03.41 | Jeff_Phillips | or will the hookswitch only perform a blind transfer? |
| 21:03.52 | rue_work | you can to attended |
| 21:03.59 | rue_work | it is by default |
| 21:04.08 | Jeff_Phillips | okay let me try... |
| 21:04.10 | Jeff_Phillips | hang on |
| 21:04.11 | [TK]D-Fender | Jeff_Phillips: For you go read your admin guide... the gateway itself provider transfer capabilities. |
| 21:04.34 | [TK]D-Fender | providers* |
| 21:04.44 | *** join/#asterisk Khratos (n=khratos@190.166.103.180) |
| 21:04.56 | rue_work | [TK]D-Fender, you know anything on the aastra issue? |
| 21:05.12 | rue_work | to my understanding the callerid data comes from the phone |
| 21:05.26 | rue_work | I suppose I can have the dialplan overwrite it |
| 21:05.55 | Jeff_Phillips | hmm well the hookflash transfer seems to work |
| 21:06.04 | rue_work | :) |
| 21:06.10 | Jeff_Phillips | thanks |
| 21:06.18 | Jeff_Phillips | that will at least get my boss off my back for a while |
| 21:06.27 | rue_work | why did you use analog phones? |
| 21:06.46 | Jeff_Phillips | because we already had tons of them wired up all through the plant |
| 21:07.13 | rue_work | so voip phones cost too much then? |
| 21:07.20 | Jeff_Phillips | and it's a really dusty shop enviroment. Any fancy electronics we put in will die within a month |
| 21:07.27 | rue_work | ok |
| 21:07.45 | Jeff_Phillips | where as this way I can put cheap $5 phones in, and just keep replacing them, while keeping the device they are wired to locked away in a nice clean closet somewhere |
| 21:07.46 | rue_work | do you have a receptionist using a single line analog phone? |
| 21:07.59 | rue_work | no i can understand the env thing |
| 21:08.08 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 21:08.10 | Jeff_Phillips | for the moment, i want to get her a fancy IP phone for her desk |
| 21:08.29 | rue_work | so she has call notify? |
| 21:08.38 | Jeff_Phillips | they moved the office into a different building and asked me to wire up the phones. That was my excuse to get us to switch over to an IP phone system |
| 21:08.41 | rue_work | single line only ? |
| 21:08.49 | rue_work | :) |
| 21:09.26 | Jeff_Phillips | we used to have two POTS lines wired everywhere and 2-line phones everywhere. I switched out one of the POTS lines a long time ago with a VoIP service to cut long distance, so everyone presses line 2 to make a long distance call. |
| 21:09.39 | Jeff_Phillips | Then I switched everything over so each phone has its own extension and you dont' use the "line 2" button anymore |
| 21:09.47 | rue_work | ah so you can use the 2line set and give her both then |
| 21:10.04 | Jeff_Phillips | well I got all kinds of complaints from people saying they couldn't make any outgoing calls. I've explained it 100 times that you dont' have to push line 2 anymore......... |
| 21:10.08 | rue_work | call in progress and call incomming |
| 21:10.16 | rue_work | :) |
| 21:10.17 | Jeff_Phillips | but they can't comprehend it for some reason so I had to rip out all the 2 line phones and put in just 1 line phones |
| 21:10.24 | rue_work | wire them both togethor |
| 21:10.41 | Jeff_Phillips | yeah I can do that in the office |
| 21:10.58 | rue_work | say you fond a fix online, and they have to pay the consultant who devised it $400 |
| 21:11.04 | Jeff_Phillips | actually it is already wired that way but it confused the daylights out of them when I had line-1 be ext 110 and line-2 be ext 111 |
| 21:11.17 | rue_work | nono, same pair |
| 21:11.29 | Jeff_Phillips | wait, what?? |
| 21:11.53 | rue_work | put pair 2 in parallel with pair 1 for the nitwitts, and make it a seperat line for reception |
| 21:12.20 | Jeff_Phillips | oh that's a good idea to use up these 2 line phones until they all die off |
| 21:12.27 | rue_work | ;) |
| 21:13.09 | rue_work | what continent you in? |
| 21:13.18 | Jeff_Phillips | I'm in the US |
| 21:13.42 | rue_work | hmm any of the 2line sets in at all descent shape? |
| 21:13.51 | rue_work | looking for some |
| 21:14.00 | Jeff_Phillips | they're kinda rugged |
| 21:14.31 | Jeff_Phillips | and i have a boss who won't part with anything even if it's stuff we don't have a use for |
| 21:14.34 | rue_work | thinks |
| 21:14.37 | Jeff_Phillips | we have to wait until he's gone to use the dumpster |
| 21:14.46 | Jeff_Phillips | lol |
| 21:14.56 | rue_work | I'z gonna suggest buying them from ya |
| 21:15.18 | Jeff_Phillips | well if they were mine I'd go for it. Being his I'd have to sneak them out of here |
| 21:15.38 | rue_work | ok, did someone say that the reason echo isn't a problem with a channelbank is cause its nativly bridged? |
| 21:15.38 | Jeff_Phillips | anyway there's one other issue I can't figure out |
| 21:15.53 | rue_work | whats that |
| 21:16.13 | Jeff_Phillips | the extensions that get used the most often tend to do this thing at random times where they just cease having a dial-tone |
| 21:16.24 | Jeff_Phillips | you can still make and receive calls. You just don't hear a dial tone when you pick it up |
| 21:16.39 | Jeff_Phillips | and, if it is an extension that is an agent of a queue, it triggers the queue to fail |
| 21:16.58 | Jeff_Phillips | what's really goofy is that if I set the fail option on the queue to direct the call to that same extension, it rings |
| 21:17.02 | rue_work | is immediate yes or no? |
| 21:17.09 | Jeff_Phillips | immediate |
| 21:17.13 | Jeff_Phillips | For example... |
| 21:17.22 | rue_work | might be picking the click up as a pulse dial |
| 21:17.23 | Jeff_Phillips | I have a queue that rings 110, 112, and 113 simultaneously |
| 21:17.38 | Jeff_Phillips | If you're on the phone (on any one of these) the other two ring |
| 21:17.46 | Jeff_Phillips | 110 spontaneously ceased having a dial tone |
| 21:17.50 | Jeff_Phillips | but if you call it directly it rings |
| 21:17.56 | Jeff_Phillips | if you place an outbound call, it works |
| 21:18.03 | Jeff_Phillips | But if you call the queue, then it was going to voice mail |
| 21:18.03 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 21:18.11 | Jeff_Phillips | I changed the fail option from voice mail to 110 |
| 21:18.33 | Jeff_Phillips | So now if all three have a dial tone then all three will ring when a call hits the queue |
| 21:18.41 | rue_work | er... |
| 21:18.56 | Jeff_Phillips | but if 110 quits having a dial tone, then only 110 rings and 112 and 113 do not |
| 21:19.04 | rue_work | not sure there, wanna pastebin your extensions.conf? |
| 21:19.17 | Jeff_Phillips | okay. |
| 21:20.18 | Jeff_Phillips | is it okay to just paste it here? (worried about flooding) |
| 21:20.32 | rue_work | no |
| 21:20.33 | bmoraca | ~pb |
| 21:20.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 21:20.35 | rue_work | use pastebin |
| 21:21.14 | Jeff_Phillips | OOHhh I thought that was a typo |
| 21:21.17 | Jeff_Phillips | sorry |
| 21:22.07 | rue_work | if all internal extensions are 2 digit, shall the dialplan be XX ? |
| 21:22.21 | rue_work | ont eh phone |
| 21:22.37 | rue_work | so it automatically does the call instead of pressing send? |
| 21:23.52 | Jeff_Phillips | http://pastebin.com/d3c8c2d08 |
| 21:23.56 | Jeff_Phillips | that's kinda cool |
| 21:24.53 | Jeff_Phillips | I would think so... My phones are analgo so no "send" key. But, mine it just works by timing. If you dial a short number and pause then it will connect to an extension, if you dial a long number and pause it will try to make an outgoing call |
| 21:25.03 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-482168f84765e85e) |
| 21:25.12 | rue_work | no this is for my problem, my phones are voip |
| 21:25.44 | rue_work | yikes, cant help much with the dialplan [TK]D-Fender ? |
| 21:25.51 | Jeff_Phillips | yeah i get you, sorry |
| 21:26.40 | rue_work | wholy gazoo I'm glad I write my own dialplans by hand |
| 21:26.51 | rue_work | er extensions.conf |
| 21:27.09 | Jeff_Phillips | lol well i thought the trixbox thing would make it easier |
| 21:27.11 | Jeff_Phillips | now i'm not so sure |
| 21:27.32 | Jeff_Phillips | i'm a newbie at this phone stuff |
| 21:27.48 | rue_work | sounds like your doing ok so far |
| 21:28.28 | Jeff_Phillips | lol, thanks |
| 21:29.00 | rue_work | well you managed to get asterisk, a T1 card, and a channelbank all talking and thats non-trivial |
| 21:29.23 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 21:29.24 | kaldemar | as is getting someone here to debug trixbox |
| 21:30.00 | rue_work | From: "Val" <sip:11@192.168.1.10>;tag=39D34B9C-20472983 <-- where does that quoted name come from on an aastra? |
| 21:31.11 | hardwire | any fft masters ? |
| 21:31.27 | rue_work | Im not a master, but I might be as close as you get |
| 21:32.00 | hardwire | woo |
| 21:32.03 | Jeff_Phillips | Well I don't have a T1 -- it would cost us a fortune. |
| 21:32.10 | hardwire | well have a good day then ;P |
| 21:32.13 | rue_work | how did you hook up the channelbank? |
| 21:32.25 | Jeff_Phillips | The MP-124 is just for internal extensions -- 24 FXS ports |
| 21:32.39 | rue_work | how do you hook it to asterisk? |
| 21:32.44 | Jeff_Phillips | Ethernet |
| 21:33.01 | Jeff_Phillips | I have an Openvox PCI card with for analog FXO ports I use for the POTS line |
| 21:33.18 | Jeff_Phillips | Plus some sip providers for outbound calling |
| 21:34.00 | Jeff_Phillips | i have two POTS lines and two VOIP lines that I can't setup in asterisk because they aren't SIP, they are SPA so I have to use the box the provider sent me which gives me analog RJ-11 jacks, and connect those to the FXO ports |
| 21:34.13 | *** join/#asterisk telecos (n=sergio@160.167.219.87.dynamic.jazztel.es) |
| 21:34.15 | rue_work | oh |
| 21:34.37 | x86 | Jeff_Phillips: SPA is just a model of Linksys SIP ATA's... |
| 21:35.04 | Jeff_Phillips | yeah it's a linksys but it has some goofy GSM protocal |
| 21:35.04 | Jeff_Phillips | T-mobile |
| 21:35.15 | x86 | Asterisk supports the GSM codec |
| 21:35.29 | Jeff_Phillips | Oh really? Well where do I put the SIM cards? |
| 21:35.39 | Jeff_Phillips | because if I take them out of the little linksys box it won't work anymore ;-P |
| 21:35.42 | x86 | codec != radio |
| 21:35.56 | Pan3D | SPA started with Sipura |
| 21:35.59 | x86 | you can buy GSM radios for Asterisk too |
| 21:36.08 | x86 | Pan3D: until Linksys bought them ;) |
| 21:36.11 | Jeff_Phillips | no it doesn't actually have a radio |
| 21:36.13 | Pan3D | yeah |
| 21:36.21 | Jeff_Phillips | What they do is backhaul GSM protocal over IP |
| 21:36.22 | Pan3D | I've got an original Sipura one |
| 21:36.26 | rue_work | the default asterisk hold music is funkey! |
| 21:36.31 | Jeff_Phillips | so t-mobile's network sees it the same way it sees a call that comes from a cell phone tower |
| 21:36.36 | Jeff_Phillips | but it is actually coming to them via IP |
| 21:36.41 | Jeff_Phillips | and I get an RJ-11 analog jack out of it |
| 21:36.45 | x86 | Jeff_Phillips: that's retarded |
| 21:36.50 | Jeff_Phillips | That's what I thought |
| 21:37.06 | Jeff_Phillips | but the service is only $10/month per line unlimited to the whole US, and I got two free airline tickets out of the deal |
| 21:37.42 | Jeff_Phillips | it's cheap and it works like a POTS line |
| 21:38.07 | Jeff_Phillips | and they are able to do number portablity here and get local numbers in my rate center |
| 21:38.16 | Jeff_Phillips | which most providers can't do |
| 21:38.21 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 21:38.23 | harry_v | Jeff, what line? |
| 21:38.38 | Jeff_Phillips | T-mobile's "@Home" service |
| 21:38.42 | harry_v | ohhh |
| 21:39.24 | Jeff_Phillips | I had to buy a prepaid cell phone at the dollar store for $15 and activate it to get a local number, then port it to this thing |
| 21:39.33 | Jeff_Phillips | they couldn't give me a local number directly unless I already had one |
| 21:39.41 | Jeff_Phillips | so I threw the cell phone away two days later |
| 21:40.15 | Jeff_Phillips | We live in a town with a lame local phone company that is stubborn to let any other providers do anything here |
| 21:40.36 | Jeff_Phillips | So I could either pay $50 / month per line plus 5 cents a minute |
| 21:40.47 | Jeff_Phillips | Or I can pay $10 / line plus zero cents a minute by doing it this way |
| 21:40.52 | rue_work | SIP/2.0 403 Authentication user name does not match account name hmmm |
| 21:41.04 | Jeff_Phillips | but I have to put up with the fact that they give me this VoIP device that converts it to analog, only for me to plug it into an FXO card and convert it back to IP |
| 21:41.30 | Jeff_Phillips | rue -- ?huh |
| 21:42.59 | rue_work | check_auth: username mismatch, have <14>, digest has <14c> <-- what is this digest thing! |
| 21:43.34 | rue_work | SIP/2.0 403 Authentication user name does not match account name <-- please tell me what two strings dont match!!! |
| 21:43.42 | rue_work | or is that whats above? |
| 21:43.47 | rue_work | whats digest!? |
| 21:44.03 | rue_work | Qwell, bmoraca ? |
| 21:45.28 | Jeff_Phillips | oh shoot i didn't realize the time |
| 21:45.40 | Jeff_Phillips | gotta go before they get on me for unauthorized overtime |
| 21:46.10 | kaldemar | rue_work: look at the SIP debug and you'll figure it out. |
| 21:46.53 | Katty | does the name John Wicks ring a bell with anyone? |
| 21:46.58 | rue_work | here is what I see... 'phone number' is the account (>>14<<@192.168...") 'caller id' is the alias (>>"14b"<< <sip:14@...) and 'auth name' is the "digest username" |
| 21:47.02 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 21:47.40 | rue_work | so, if I want a phone to come up with someone name instead of their account name |
| 21:48.28 | rue_work | kaldemar, I dont understand what digest *IS* though |
| 21:48.29 | n3hxs | Katty, Garage band Christmas? |
| 21:48.56 | Katty | some how i think that's someone else. |
| 21:49.16 | Katty | some person named John Wicks working for Inter-tel wanted to follow me on twitter |
| 21:49.21 | Katty | thought maybe it was one of you people |
| 21:49.32 | n3hxs | I don't twitter. |
| 21:49.38 | carrar | Whats your twitter page? |
| 21:49.41 | errr | Katty, he is a sales man for intertel I bet |
| 21:49.47 | Katty | errr: probably. |
| 21:49.52 | Pan3D | yeah, that smacks of sales |
| 21:49.57 | Katty | carrar: you don't need to know. |
| 21:50.08 | carrar | not that interesting? :) |
| 21:50.12 | Katty | no. |
| 21:50.14 | carrar | heh |
| 21:50.14 | Katty | i'm horribly boring. |
| 21:50.18 | carrar | awesome |
| 21:50.52 | kaldemar | rue_work: take a look at rfc 3261 for example |
| 21:51.16 | rue_work | oooh, must I know every neuance of how it works to use it?? |
| 21:51.21 | Pan3D | rue_work: digest is authentication type |
| 21:51.34 | *** part/#asterisk Jeff_Phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 21:51.44 | rue_work | hmm, that requires having 3 usernames that match? |
| 21:51.47 | errr | Katty, do a google for that guys name + intertel he is a spammer spamming: http://forums.socalphonepros.com/ |
| 21:52.24 | Katty | fun. |
| 21:53.35 | kaldemar | rue_work: no, digest only requires the digest username, but asterisk has other requirements too to authenticate a caller. |
| 21:53.41 | rue_work | its interesting that when the polycom phone logs in its "myname" <sip: myaccount@myip> and with asastra its just <sip: myaccount@myip> |
| 21:53.54 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 21:54.08 | rue_work | and I cant find aparamiter on the aastra to submitt that |
| 21:54.15 | _charly_ | hi :) |
| 21:57.06 | rue_work | hi |
| 21:57.54 | rue_work | anyone used to configuring aastra sets? |
| 21:58.42 | rue_work | so frustrating when the paramiters aren't what they say they are |
| 22:00.45 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 22:01.31 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 22:01.40 | manxpower | I'm looking for some suggestions, using manager, to keep track of the state (in use, avail, etc) of a device using the manager interface. |
| 22:05.55 | manxpower | Whoo! Whoo! I made everyone stop talking. |
| 22:06.21 | rue_work | sorry, I'm on company time making phone system |
| 22:06.33 | rue_work | where did our support people go?? |
| 22:06.33 | beek | kfife: are you still around? |
| 22:07.41 | kfife | beek: indeed |
| 22:08.31 | beek | kfife: Do you use users.conf to configure ADA or do you configure the files manually? |
| 22:08.56 | *** part/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 22:09.17 | kfife | Haven't actually gone life with it. Have you seed the configuration docs? They're pretty detialed. |
| 22:09.50 | beek | kfife: They are if you're using users.conf, which I don't. |
| 22:10.03 | beek | The problem I'm having is that I don't grok where the password goes in the ADA client on the PC. |
| 22:10.18 | beek | When I use noojee, the configuration screen asks for a password so getting this to work is trivial. |
| 22:10.28 | beek | ADA doesn't appear on the client to have anywhere to throw a password. |
| 22:10.30 | kfife | beek: I see. I don't use users.conf either. |
| 22:10.59 | kfife | My ADA client propmts me with a username & Pass |
| 22:11.01 | beek | I can use 134@myasteriskbox for the server name to get the correct extension. |
| 22:11.10 | beek | Really? Mine hasn't done that. |
| 22:11.51 | beek | kfife: Does it do it every time you start your 'doze box? |
| 22:12.06 | kfife | What happens when you double click on the system tray icon? |
| 22:12.18 | kfife | I'm running 1.0. You? |
| 22:12.18 | beek | kfife: nevermind, I'm a dumbshit. |
| 22:12.25 | kfife | no worries. |
| 22:12.36 | beek | kfife: Mine must have it cached somewhere. If I tell it to log off then it seems to have this right. |
| 22:13.10 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 22:13.10 | beek | kfife: sorry to have bothered you.l |
| 22:13.10 | kfife | No worries. I may have some TAPI questions. Are you going to be using TAPI to drive ADA at all? |
| 22:13.10 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-175-28.telkomadsl.co.za) |
| 22:14.16 | beek | kfife: I don't think so. We don't use outlook here, and I can't imagine that I have any other programs that would use it. |
| 22:14.35 | kfife | NP. Thanks! |
| 22:14.47 | beek | thank you |
| 22:19.06 | *** join/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 22:19.47 | *** part/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 22:21.58 | *** join/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 22:22.27 | xacatecas | on the Dial() application - is there a description anywhere of the individual values for the DIALSTATUS variable? |
| 22:23.09 | xacatecas | and also, the traditional behaviour of Dial() was to terminate the dialplan if a call got answered, this seems to have changed, am I correct? |
| 22:27.25 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 22:38.34 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 22:40.54 | dominic1 | hi, just one problem I get this string from the database Dial(SIP/chw\,12\,t) |
| 22:41.07 | dominic1 | before the upgrade to 1.6 the result was Dial(SIP/chw,12,t) |
| 22:41.31 | dominic1 | can I change this? The system automatically escapes this stuff |
| 22:41.36 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 22:42.02 | dominic1 | this isn't working to: Dial(SIP/chw|12|t) |
| 22:43.19 | jplank | tell me if this make sense to anyone, polycom 550, first call comes in everything works, second call comes in and while the call is ringing, the person can't here the first call anymore, if you put the first call on hold, and then go to the second call, the first call hear hold music, but you can't hear the second caller. if you then put the second caller on hold, and go back to the first, everything starts working |
| 22:47.34 | rue_work | jplank, what kind of firewall is the data going though |
| 22:48.33 | rue_work | jplank, you have to answer or we cant help |
| 22:48.46 | dominic1 | escapecommas=no |
| 22:48.50 | dominic1 | was my solution |
| 22:48.52 | dominic1 | wonderful |
| 22:50.17 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 22:50.50 | *** join/#asterisk ibercom (i=d9d85170@gateway/web/ajax/mibbit.com/x-92bd17dc947cf9ba) |
| 22:53.01 | manxpower | jplank: sounds like you are sending the call to the same line appearance |
| 22:56.52 | ibercom | Anybody know if vmail.cgi support voicemail with odbc ? |
| 22:59.41 | dominic1 | hi, is the handling of macros different in asterisk 1.6? |
| 23:00.02 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 23:00.40 | edibrac | has anyone had problems with a sangoma card...then switched to digium and all was good? |
| 23:00.43 | bmoraca | i wonder if asterisk would work through a hardware loadbalancer |
| 23:00.47 | edibrac | for T1 PRI connections. |
| 23:01.06 | edibrac | because, from what I hear/read, it's usually the other case.. .going from Digium to Sangoma? |
| 23:02.26 | manxpower | edibrac: The current generations of both the Digium and the Sangoma cards are good. |
| 23:02.30 | edibrac | and was the case for me, where I was getting HDLC errors and did everything I could find in the mailing list to troubleshoot -- but with the PCI sangoma A101 in (instead of PCI-E Digium TE121), there are no more HDLC errors. This is after testing in 3 difference supermicro servers |
| 23:02.42 | manxpower | Older Digium cards did have significant issues in a few system |
| 23:03.01 | dominic1 | sind I upgraded to asterisk 1.6 I get this error when executing a macro: possible infinite loop detected. Returning early. |
| 23:03.05 | edibrac | the TE121 is fairly recent though -- have you heard about that model specifically? |
| 23:03.19 | jplank | sorry I went afk |
| 23:03.27 | jplank | rue_work: no nat involved |
| 23:03.30 | manxpower | edibrac: "older card" = 0 in the middle digit |
| 23:03.54 | jplank | manxpower: The phone has all 4 line appearances programmed and 1 registration |
| 23:04.03 | edibrac | our current theory is that somehow Sangoma A101 is more lenient or resilient against low-level problems |
| 23:04.07 | rue_work | jplank, its a rtp problem for sure |
| 23:04.21 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 23:04.40 | jplank | the csutomer said the call automatically goes to hold |
| 23:06.15 | jplank | im kind of interested in what manxpower said "sounds like you are sending the call to the same line appearance" but I wouldn't know how to do that on purpose, let alone on accident, and I have a bunch of other systems out with the same config, and they dont have the problem, so I'm stomped |
| 23:06.46 | dominic1 | ? |
| 23:07.40 | manxpower | jplank: calls.per.line.appearance=1 or something like the in the polycom configs |
| 23:07.49 | jplank | let me take a look |
| 23:07.52 | *** part/#asterisk mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 23:07.55 | jplank | should be 1 right? |
| 23:08.05 | manxpower | yup |
| 23:08.14 | manxpower | try it and see if it help |
| 23:08.38 | jplank | call.callsPerLineKey is blank |
| 23:08.45 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 23:11.12 | edibrac | hmm.. now that i've dealt with my HDLC problems...I'm getting crackling sounds |
| 23:11.18 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:12.12 | rob0 | mmmm, rice krispies! |
| 23:12.42 | *** part/#asterisk ibercom (i=d9d85170@gateway/web/ajax/mibbit.com/x-92bd17dc947cf9ba) |
| 23:13.38 | jplank | manxpower: I don't know if thats it because a) the second line does show up on the second line appearance, b) I used the same sip.cfg from a working system and I just tested it on the other system and it doesn't happen c) seems to only happen on the ip550s not the 320s. But I'll try anyway |
| 23:13.51 | jplank | a) the second CALL* |
| 23:15.51 | ruben23 | what is the firewall setup of asterisk behind a gateway firewall...using g729 codec.... |
| 23:16.03 | jplank | wouldn't I want to keep that setting as default so the customer can conference in multiple people and only use one line key? |
| 23:16.28 | dominic1 | ARNING[26030]: app_macro.c:201 _macro_exec: No such context 'macro-quick-conference-start' for macro 'quick-conference-start' |
| 23:16.41 | dominic1 | I don't understand why this worked in 1.4 |
| 23:16.44 | dominic1 | and not in 1.6 |
| 23:17.35 | edibrac | hmm could this in /etc/zaptel.conf, cause crackling sounds if the timing is incorrect: span=1,0,0,esf,b8zs |
| 23:18.03 | edibrac | should be span=1,1,0,esf,b8zs for a regular US PRI where timing source is the telco ...AFAIK |
| 23:23.51 | manxpower | dominic1: Do you HAVE a [macro-quick-conference-start] line in extensions.conf |
| 23:24.39 | dominic1 | Hurray, I found the problem, there isn't any macro quick-conference-start anymore. I changed it to context macro-quick-conference-start |
| 23:24.44 | dominic1 | then it worked |
| 23:24.47 | manxpower | edibrac: The typical symptom of wrong sync source is audio blips, usually causing faxes to fail |
| 23:24.50 | dominic1 | wonderful |
| 23:24.55 | dominic1 | thank you manx |
| 23:27.04 | *** part/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 23:30.37 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:33.44 | macli | hi I set TRUNKMSD=0 in extensions.conf, stop and start asterisk, dialplan show globals shows TRUNKMSD=1, I am running asterisk-1.6.1-rc1 |
| 23:34.50 | jplank | grrr I really think there is something wrong with this asterisk |
| 23:35.15 | jplank | after a reboot, asterisk didn't start up right away, and theres no color in the cli |
| 23:36.32 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:39.55 | *** join/#asterisk bgmarete (n=bgmarete@196.201.208.129) |
| 23:41.23 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 23:43.31 | dominic1 | what does that mean: Rejected connect attempt from 172.17.1.11, requested/capability 0x8/0x6008 incompatible with our capability 0xe703 |
| 23:43.52 | manxpower | usually that means "no codecs can be agreed on" |
| 23:44.31 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:44.47 | dominic1 | but what is the codec 0xe703? |
| 23:45.03 | dominic1 | both server are configured with disallow=all; allow=alaw |
| 23:45.04 | rue_work | is it a bitmask for a few of the,? |
| 23:45.06 | Qwell | dominic1: It's several combined |
| 23:45.41 | iaxy | hi guys!! |
| 23:46.23 | iaxy | how do you set the rtp for a IAX? gots to set it to 20ms and what is it set default as? |
| 23:46.28 | rue_work | Qwell, you a polycom or aastra guru? |
| 23:47.05 | rue_work | (guru&aatra) | (guru&polycom) |
| 23:47.39 | carrar | Nice PIPE |
| 23:47.49 | dominic1 | seems my asterisk always wants to use slin16 as codec |
| 23:47.50 | rue_work | its not a pipe! its a or |
| 23:47.51 | dominic1 | why? |
| 23:47.58 | carrar | heh |
| 23:48.11 | rue_work | dominic1, did you compile asterisk? |
| 23:48.27 | dominic1 | yes |
| 23:48.41 | rue_work | were you missing a library it needed for the other codecs? |
| 23:48.50 | rue_work | you would see it in the configure |
| 23:48.55 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:49.08 | dominic1 | sip is using my setting alaw |
| 23:49.27 | dominic1 | only the interconnect of my two asteriskboxes always wants slin |
| 23:49.27 | rue_work | it would look like : do I have this? NO do I have that? NO do I have other? NO |
| 23:49.43 | rue_work | yea, can your asterisk handle anything else |
| 23:50.11 | dominic1 | yes I can there are a lot of entries in core show translations |
| 23:50.30 | rue_work | ok, good, your beyond my help |
| 23:50.54 | dominic1 | and I wanted to use alaw on the iax trunk, but only slin seems to work with iax |
| 23:50.59 | dominic1 | mh... |
| 23:51.34 | rue_work | anyone an aastra or polycom guru? atleast say no... |
| 23:51.51 | iaxy | who knows how to set the rtp packet size? |
| 23:53.14 | carrar | for what phone? |
| 23:53.20 | rue_work | ip601 |
| 23:53.36 | rue_work | I cant get it to auto match and dial |
| 23:54.00 | rue_work | aka the dialplan is set to XX and when I hit '12' it just sits there looking stupid |
| 23:54.09 | carrar | voice.audioProfile.G711Mu.payloadSize="20" |
| 23:54.37 | rue_work | on the polycom I set <dialplan dialplan.applyToUserDial="1" dialplan.digitmap="xx" /> |