| 00:07.22 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 00:08.42 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 00:12.48 | *** join/#asterisk digitalirony (n=digitali@shellium/member/digitalirony) |
| 00:28.39 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 00:29.35 | p3nguin_ | ecrane: What are you trying to configure, specifically? |
| 00:29.53 | p3nguin_ | ecrane: Lots of the settings can be done from the base (not the handset) in the menu. |
| 00:32.02 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 00:33.32 | kyleh | Hey, anyone on have any type of experience with festival + asterisk? |
| 00:34.04 | TSM | i got it to work, sounds horible though |
| 00:34.26 | TSM | is there any better free ones |
| 00:34.38 | kyleh | like festival? |
| 00:35.08 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 00:35.08 | kyleh | im having trouble with mine. when i call festival in the extensions.conf |
| 00:35.39 | kyleh | it will connect to the festival server then disconnect right away. leaving asterisk just hanging there |
| 00:35.53 | TSM | is the festival server running? |
| 00:35.57 | kyleh | ya |
| 00:36.15 | kyleh | have it running the in background and it says client connected |
| 00:36.29 | kyleh | then it disconnects right after it says that |
| 00:36.35 | TSM | i had that issue |
| 00:36.58 | kyleh | ya, any tips on getting past it? |
| 00:37.46 | TSM | did you modify the /usr/share/festival/festival.scm as indicated on the voip-info page/ |
| 00:38.42 | kyleh | http://www.voip-info.org/wiki/view/Asterisk+festival+installation |
| 00:38.48 | kyleh | from that page? |
| 00:40.15 | TSM | yup i did method 1 |
| 00:40.50 | kyleh | ya i pasted that stuff at the end of festival.scm |
| 00:40.57 | leifmadsen | cepstral is a million times better than festival, however not free (but very reasonably priced) |
| 00:42.01 | kyleh | it wouldnt have anything to do with the set italian voice part would it? |
| 00:42.21 | kyleh | i left that chunk out |
| 00:42.23 | TSM | i have english voice |
| 00:42.49 | TSM | i left it commented out |
| 00:43.19 | Katty | http://www.youtube.com/watch?v=qtrvraTNOfA :> |
| 00:46.09 | kyleh | dang wonder what the problem could be then] |
| 00:46.38 | kyleh | i tried useing the perl script method mentioned in that write-up |
| 00:46.44 | kyleh | no luck with that either |
| 00:48.33 | *** join/#asterisk gilevy (n=gil@24.10.28.163) |
| 00:48.57 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 00:49.27 | gilevy | i am able to forward a call for about 15 seconds and then i get this error :chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission |
| 00:49.48 | *** join/#asterisk Gokee2 (n=gokee2@24-113-159-168.wavecable.com) |
| 00:49.52 | gilevy | i am able to talk on the phone fine from both ends |
| 00:50.07 | gilevy | but it still produces that error and hangs up |
| 00:50.42 | gilevy | i am able to forward a call for about 15 seconds and then i get this error :chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmissionanybody have an idea of what to do/try? |
| 00:53.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 01:06.48 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 01:11.19 | Katty | yawns |
| 01:11.53 | ChannelZ | stretches |
| 01:12.36 | p3nguin_ | throws things |
| 01:14.51 | Katty | dances with p3nguin_ |
| 01:14.59 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:16.20 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 01:17.10 | jblack | scratches his butt and looks around the room, bleary eyed |
| 01:17.15 | jblack | Is it spring yet? |
| 01:21.36 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 01:22.17 | doolittlework | hi there can one use the ChanIsAvail for groups like exten = |
| 01:22.35 | doolittlework | chanisavail(zap/g1)? |
| 01:24.23 | gilevy | where can i find doc/sip-retransmit.txt? |
| 01:24.45 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 01:26.13 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:27.38 | p3nguin_ | jblack: You can go back to sleep, it's still October. |
| 01:30.26 | *** join/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) |
| 01:37.03 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
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| 01:49.27 | *** join/#asterisk kuku1 (n=ingo@67.175.3.155) |
| 01:51.44 | kuku1 | I have 2 NICs. On one, I have the local network, on the other, I have the connection XO's SIp proxy. Now if I di dial(sip/whatever@proxyip) it shows the call originating from the original ip, I need the call to show coming from the ip of the second nic. Is there a way to force asterisk to use an interface to orginate a call ? |
| 01:55.07 | voipmonk | you can use ser |
| 01:56.39 | *** join/#asterisk Jumpie (i=n3rdz@c-76-100-241-4.hsd1.md.comcast.net) |
| 02:07.14 | *** part/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 02:08.55 | *** join/#asterisk theangryamoeba (n=agn0g3ni@c-98-212-197-126.hsd1.il.comcast.net) |
| 02:15.22 | kuku1 | can I install on the same serwer ? |
| 02:21.08 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 02:21.08 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:25.21 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 02:25.22 | *** mode/#asterisk [+o file] by ChanServ |
| 02:29.30 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 02:30.15 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:31.56 | *** join/#asterisk korcan (n=korcan@adsl-99-33-107-250.dsl.sfldmi.sbcglobal.net) |
| 02:40.56 | drmessano | KARMIC IS ALMOST HERE! |
| 02:43.31 | ChannelZ | jizzes in his pants |
| 02:45.39 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 02:45.56 | *** join/#asterisk OrNix (n=ornix@91.151.249.47) |
| 02:48.11 | carrar | Y*A*W*N |
| 02:59.13 | *** part/#asterisk theangryamoeba (n=agn0g3ni@c-98-212-197-126.hsd1.il.comcast.net) |
| 03:03.44 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 03:07.55 | [TK]D-Fender | pukes in his mouth |
| 03:09.13 | loather-work | that's kind of disgusting. |
| 03:10.10 | [TK]D-Fender | loather-work: Loaded joke if you're familiar with SNL |
| 03:10.34 | loather-work | i haven't watched SNL for probably 15 years |
| 03:14.47 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:20.04 | *** join/#asterisk jayrod422 (n=jarrod@pool-96-235-30-58.pitbpa.fios.verizon.net) |
| 03:20.28 | [TK]D-Fender | loather-work: The original : http://www.youtube.com/watch?v=4pXfHLUlZf4 |
| 03:20.39 | jayrod422 | does anyone know how to solve issues where asterisk wont recoginize a peer behind a nat by ip or user/pass - ex. http://pastebin.com/m34dc42f0 |
| 03:20.52 | *** join/#asterisk dmz (n=dmz@h96-61-156-43.mtjltn.dsl.dynamic.tds.net) |
| 03:21.33 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-215-181.dsl.lsan03.sbcglobal.net) |
| 03:23.14 | *** join/#asterisk raden (n=chatzill@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 03:23.45 | raden | Katty: OLA |
| 03:25.10 | [TK]D-Fender | jayrod422: You don't have a peer with the name, and your default doesn't assume they are NAT'd which is a bad move. |
| 03:25.12 | [TK]D-Fender | ~sipnat |
| 03:25.13 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:25.15 | [TK]D-Fender | ^^^^ |
| 03:25.21 | raden | OMFG lag |
| 03:25.21 | [TK]D-Fender | jayrod422: Go follow the guide |
| 03:25.29 | carrar | then jizz in your pants |
| 03:25.45 | jayrod422 | lol |
| 03:27.36 | jayrod422 | fender - what would the peer name be from what you have seen in the SIP packets (the ip?) |
| 03:27.39 | [TK]D-Fender | carrar / loather-work : and the response... http://www.youtube.com/watch?v=DJsQcnB6GC0 |
| 03:27.54 | [TK]D-Fender | jayrod422: From: |
| 03:27.55 | jayrod422 | or username |
| 03:28.18 | [TK]D-Fender | jayrod422: <sip:FEDCOM@209.195.155.137> |
| 03:33.42 | Maliuta | afternoon *ers |
| 03:34.20 | Maliuta | I just read a post to a sysadmin list I am on concerned about "vishing" attacks on * servers |
| 03:34.36 | Maliuta | seemed a bit preposterous to me |
| 03:34.40 | loather-work | [TK]D-Fender: hilarious. |
| 03:34.57 | carrar | I'm vishing on your server right now |
| 03:35.10 | raden | anyone tell me if $95 a good deal for Cisco 2610XM Router with 128MB DRAM / 48MB Flash |
| 03:35.25 | carrar | It's ok |
| 03:35.54 | carrar | depends what you need it for I guess |
| 03:36.58 | carrar | perfect for a T1/Ethernet router |
| 03:37.31 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 03:44.48 | raden | carrar: CISCO lab |
| 03:45.03 | raden | otherwise 2950's with wic t1 for $50 each i dont know mem specs though |
| 03:46.08 | carrar | 2950 switch? |
| 03:46.23 | raden | Router |
| 03:46.37 | raden | I got 2x 2950 switches for $140 i overpaid but best i could find |
| 03:46.37 | carrar | 2950 is a switch |
| 03:46.50 | raden | 2650 sorry |
| 03:46.53 | raden | ROUTER 2650 |
| 03:46.55 | trogs | raden: you should look at dynamips. software emulation of cisco routers. |
| 03:47.12 | trogs | if you wanna do lab work etc. |
| 03:47.40 | trogs | 2600 router pretty old these days so you should be able to get em pretty cheap on ebay etc |
| 03:48.08 | raden | they any decent though I can get 3 with T1 cards for $150 shipped |
| 03:49.24 | loather-work | the T1 boards alone are worth that |
| 03:50.01 | raden | for real ? |
| 03:50.09 | carrar | Just use your production network as your lab :) |
| 03:52.00 | raden | carrar: all HP :( |
| 03:58.21 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 04:07.11 | *** join/#asterisk dkirker (n=dkirker@129.65.204.143) |
| 04:14.16 | p3nguin_ | raden: Where can I get one with a some ethernet ports? |
| 04:17.19 | raden | p3nguin_: ??? |
| 04:19.24 | carrar | ethernet is over rated |
| 04:19.24 | p3nguin_ | what? |
| 04:20.35 | raden | what you wrote me did not make sense |
| 04:21.16 | p3nguin_ | Why not? I thought it made perfect sense. |
| 04:21.29 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 04:22.02 | p3nguin_ | You're talking about getting a router with a WIC for $50... I want one with either two or three ethernet ports. For cheap, of course. |
| 04:24.12 | raden | good luck |
| 04:24.23 | raden | trying to find cheap routers for ccna lab |
| 04:24.32 | raden | dudes in cisco tell me 2650 not good enough |
| 04:24.51 | p3nguin_ | But you said you can get them for $50, so I'm asking you where I can get the router for the price. |
| 04:25.35 | raden | http://cgi.ebay.com/Cisco-2650-T1-Router-2620-upgrade-w-WIC-1DSU-T1_W0QQitemZ170394024625QQcmdZViewItemQQptZCOMP_EN_Routers?hash=item27ac467ab1 |
| 04:26.54 | p3nguin_ | The card alone is worth $139.00. |
| 04:28.03 | p3nguin_ | Is there a 10/100 WIC available, though? That router only has one Ethernet port. |
| 04:31.24 | raden | p3nguin_: I been looking I saw some earlier |
| 04:31.35 | raden | p3nguin_: you think that router enough for a CCNA starter lab ? |
| 04:31.52 | p3nguin_ | Probably. |
| 04:32.51 | p3nguin_ | We don't use T1 cards, though. We connect the routers together for labs via serial crossover cables. |
| 04:33.33 | p3nguin_ | WIC-2T cards |
| 04:34.44 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 04:34.45 | p3nguin_ | The academy courses have you hooking three routers together for the labs, so serial makes sense. |
| 04:35.33 | loather-work | i hate those things |
| 04:35.41 | loather-work | with the HDB-60 connnectors |
| 04:35.59 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 04:37.36 | p3nguin_ | I don't think that's the right connector. |
| 04:37.52 | loather-work | oh yeah. the 2t has those stupid blue ones. |
| 04:38.33 | p3nguin_ | The small ones |
| 04:38.49 | p3nguin_ | I think they are called "smart" |
| 04:42.32 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:49.41 | jayrod422 | in asterisk 1.6 how do you turn sip debbuging on |
| 04:50.44 | raden | sip set debug on |
| 04:51.02 | jayrod422 | ah |
| 04:51.03 | jayrod422 | set |
| 04:54.00 | raden | p3nguin_: Cisco 2610 w/ NM-4B-U <<<< |
| 04:55.01 | p3nguin_ | 4-port ISDN BRI network module with integrated Network Termination 1 (NT1)-U interface |
| 04:55.36 | raden | Cisco WIC-4ESW 4-Port 10/100 w/ 1-YEAR WARRANTY! << $44 |
| 04:56.25 | jayrod422 | can anyone look at http://pastebin.com/m4e55656b i am having a hell of time trying to get a asterisk box behind a nat to work with another that is not, ive tried ip auth, register, and host=ip and for some reason asterisk cannot find these peers in sip.conf |
| 04:56.45 | dlynes | Does anyone have any Prince Edward Island DIDs? |
| 04:57.59 | dlynes | jayrod422: one thing...register line must be in the general section |
| 04:58.25 | dlynes | jayrod422: you've got it in a peer context |
| 04:59.08 | jayrod422 | it is in general |
| 04:59.18 | dlynes | jayrod422: not according to your pastebin |
| 04:59.33 | p3nguin_ | I'm thinking I need the HWIC-2FE, but I'm not certain. |
| 04:59.34 | dlynes | jayrod422: according to your pastebin, it's in fedcomoffice |
| 05:00.19 | jayrod422 | thats on the main server |
| 05:00.42 | dlynes | jayrod422: in that case, then...you don't have a general context |
| 05:00.47 | jayrod422 | the register command is on the remote box behind the nat |
| 05:00.52 | dlynes | jayrod422: either that, or you haven't pasted the entire files |
| 05:00.53 | jayrod422 | and i do have it in general |
| 05:01.03 | jayrod422 | yeah |
| 05:01.17 | jayrod422 | i left out all the commented out stiff on the remote box from the default asterisk sip.conf |
| 05:01.43 | dlynes | jayrod422: can you do it as two separate pastebins, and just paste the two files, verbatim, with the passwords (and hostnames, if wanted) scrubbed? |
| 05:01.52 | jayrod422 | k |
| 05:01.53 | jayrod422 | 1 sec |
| 05:01.56 | dlynes | jayrod422: and scrub the usernames, too |
| 05:03.37 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 05:04.37 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 05:05.13 | p3nguin_ | raden: The HWIC-2FE is the card I would want, but it doesn't work on the 2600 (does on 2800, though). |
| 05:05.38 | raden | 2x 2950-24's 3x 2650's w/ T1 WICS TOTAL SPENT $260 |
| 05:05.56 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:06.01 | jayrod422 | ok the remote is http://pastebin.com/m24f912bd and the main is http://pastebin.com/m5e2dbfca |
| 05:06.09 | *** join/#asterisk thazza (n=thazza@ntechm.lnk.telstra.net) |
| 05:06.35 | PMantis | Quick question: If I download the latest 1.6.1 tarball vs checking out asterisk from the 1.6.1 branch, is there any difference at all? (besides the .svn dirs) |
| 05:07.25 | dlynes | PMantis: yes...depends on whether you check it out based on the branch tag, or not |
| 05:07.44 | dlynes | PMantis: if you check out based on the branch tag for 1.6.1.8, it'll be the same as the 1.6.1.8 release |
| 05:08.01 | dlynes | PMantis: otherwise, it'll be what's going to be 1.6.1.9 |
| 05:08.42 | dlynes | jayrod422: remote = box that registers to the 'main', right? |
| 05:08.48 | jayrod422 | yes |
| 05:08.54 | dlynes | jayrod422: remote is also the one behind a router, right? |
| 05:09.01 | jayrod422 | yepper |
| 05:09.16 | PMantis | dlynes: OK, Understood. I'm a new SVN user, can you show the syntax for that? I have to get a handle on the branching features for my own project, too. |
| 05:09.19 | dlynes | jayrod422: and when you say it's not working....how is it not working? |
| 05:09.33 | jayrod422 | it cannot find the peer in sip.conf |
| 05:10.00 | jayrod422 | look back in my 1st pastebin post |
| 05:10.07 | jayrod422 | i put the sip debug in there |
| 05:10.19 | jayrod422 | i also tried this just by ip and no dice |
| 05:10.45 | dlynes | PMantis: this doesn't cover everything, but it should get you started with the basics...I haven't used it since it was using cvs, so I'm kinda out of date with it, but here you go: http://www.asterisk.org/developer/resources/svn |
| 05:11.01 | jayrod422 | http://pastebin.com/m4e55656b line 86 |
| 05:13.00 | dlynes | jayrod422: fwiw, canreinvite=no is set that way, because asterisk cannot reinvite across a nat |
| 05:13.09 | dlynes | jayrod422: nothign to do with cisco |
| 05:13.21 | PMantis | dlynes: I still have some gray around this. That's a standard svn checkout, I was expecting to see a -r1.6.1.8 or something like that. :) |
| 05:14.16 | dlynes | jayrod422: can I also see your remote's dialplan? |
| 05:14.23 | jayrod422 | some of it |
| 05:14.25 | jayrod422 | sure |
| 05:14.28 | jayrod422 | 1 sec |
| 05:14.31 | dlynes | jayrod422: and if you can, highlight the section of the dialplan where you're dialing |
| 05:15.08 | PMantis | dlynes: According to the viewer (http://svnview.digium.com/svn/asterisk/branches/1.6.1/) I don't see where it mentions the 4th level of the version number, only to the 3rd (1.6.1). I must be missing something simple. |
| 05:15.54 | jayrod422 | its always going to the default of main-incoming |
| 05:16.01 | jayrod422 | and not from-prepay customer |
| 05:16.55 | dlynes | PMantis: it's not going to be something as simple as -r1.6.1.8 |
| 05:17.09 | dlynes | PMantis: you're going to need to know the revision number to do it that way, not the tag |
| 05:17.21 | PMantis | reads further into svn-book.pdf |
| 05:19.10 | dlynes | PMantis: for instance, it looks like it might be -r226384 for v1.6.1.8 |
| 05:19.57 | dlynes | jayrod422: sorry...you've lost me |
| 05:20.23 | dlynes | jayrod422: I'm looking for the dialplan from the client, and have the section where you call 'main' highlighted |
| 05:20.29 | PMantis | dlynes: I have a project that includes *, and that project is under SVN. If someone updates chan-sip.c for a security risk, it seems silly to download the entire tarball to fix the one file. Just trying to simplify management. Perhaps the svn:externals property is what I want... |
| 05:20.34 | dlynes | jayrod422: i'm not interested in the dialplan on main at this point |
| 05:20.55 | jayrod422 | here it is |
| 05:21.03 | jayrod422 | er shit |
| 05:21.05 | jayrod422 | you want remote |
| 05:21.15 | PMantis | dlynes: May I ask how you found that rev number for 1.6.1.8? |
| 05:21.58 | dlynes | PMantis: erm...actually -r226386 |
| 05:22.24 | dlynes | PMantis: I went to the link you posted, and found the latest revision number under the 'Rev.' column |
| 05:22.43 | PMantis | looks |
| 05:22.45 | jayrod422 | heres teh remote http://pastebin.com/m38709937 |
| 05:23.14 | jayrod422 | and here is the main on for what its worth http://pastebin.com/m65384974 |
| 05:23.46 | DND | guys, how can i patch indications.comf |
| 05:23.54 | dlynes | jayrod422: you mean these two lines:? |
| 05:24.00 | [TK]D-Fender | DND: You don't patch a config file.... |
| 05:24.01 | dlynes | [from-prepay-customer] |
| 05:24.03 | dlynes | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.fed-com.com) |
| 05:24.08 | jayrod422 | thats all remote needs |
| 05:24.10 | jayrod422 | yep |
| 05:24.14 | dlynes | ok |
| 05:24.44 | jayrod422 | the main issue is the main box doesnt match these peers and since it does not it fails the call |
| 05:25.12 | jayrod422 | does it on 1.4 latest and 1.6 latest |
| 05:25.27 | dlynes | jayrod422: Now, why wouldn't you do exten => _1NXXNXXXXXX,1,Dial(SIP/sip.fed-com.com/${EXTEN})? |
| 05:25.59 | PMantis | dlynes: Hmm, the revision you mention is at the top of the page by "Directory revision:". Something tells me this is not coincidence. |
| 05:25.59 | jayrod422 | old habits hard to break |
| 05:26.33 | dlynes | PMantis: probably not coincidence, no :) |
| 05:26.53 | dlynes | jayrod422: change it to that format, to make things a little more predictable |
| 05:27.07 | [TK]D-Fender | jayrod422: From: "FEDCOM101" <sip:FEDCOM101@192.168.1.3>;tag=as7940b945 <- since when is this supposed to match a peer? |
| 05:27.08 | dlynes | jayrod422: it forces your username and password that way |
| 05:27.13 | [TK]D-Fender | jayWhich one exactly? |
| 05:27.31 | jayrod422 | its not |
| 05:27.32 | dlynes | [TK]D-Fender: it doesn't...he's got his usernames mismatches on both sides |
| 05:27.42 | [TK]D-Fender | jayrod422: And stop using FQDN looking sip.conf entriy names |
| 05:28.05 | [TK]D-Fender | dylI know.. and I pointed this out TWO HOURS AGO |
| 05:28.21 | dlynes | [TK]D-Fender: which is why i'm getting him to use the dial format I suggested, so that it makes it easier for him to find his problems on his own |
| 05:29.17 | PMantis | dlynes: Thanks for your time - much appreciated! |
| 05:29.27 | DND | [TK]D-Fender sorry but i need to add a signalling for one country |
| 05:30.03 | dlynes | PMantis: no problem.... [TK]D-Fender might be able to help you out to let you know how to check out 1.6.1.8 without having to know the revision number...he's more seasoned with asterisk than I am |
| 05:30.14 | jayrod422 | ok so i make that change to the dial command but still same thing |
| 05:30.19 | jayrod422 | on the remote |
| 05:30.29 | dlynes | jayrod422: yes..it'll be the same problem, or quite similar |
| 05:30.38 | dlynes | jayrod422: I'm just getting you headed in the right direction |
| 05:30.52 | PMantis | dlynes: Yeah, but it's too easy to get on his bad side... ;) |
| 05:31.23 | dlynes | PMantis: nah...he's a pussycat, as long as he doesn't have to keep telling you the same thing over and over again |
| 05:31.41 | PMantis | lol |
| 05:31.52 | [TK]D-Fender | Checkout time.... consider it a gentle mercy ;) |
| 05:31.58 | jayrod422 | now why the heck am i seeing this on the main box [Oct 29 01:24:53] NOTICE[11564]: chan_sip.c:18454 handle_request_invite: Failed to authenticate device "FEDCOM101" <sip:FEDCOM101@192.168.1.3>;tag=as07863ba4 |
| 05:31.58 | jayrod422 | <PROTECTED> |
| 05:32.16 | dlynes | jayrod422: Now, do you see a sip.fed-com.com username on the server side? |
| 05:32.52 | jayrod422 | sip show peers |
| 05:32.52 | jayrod422 | fedcomoffice/FEDCOMO (Unspecified) D N 5060 UNKNOWN |
| 05:33.18 | dlynes | jayrod422: I didn't ask for sip show peers...I merely asked if you saw that username on the server side |
| 05:33.27 | jayrod422 | oh |
| 05:33.35 | dlynes | jayrod422: so, you don't see it on the server side |
| 05:33.58 | dlynes | jayrod422: so let's rename it to something non-fqdn, and make it something meaningful |
| 05:34.06 | PMantis | bangs his head on the table for jayrod422 |
| 05:34.15 | jayrod422 | No matching peer for 'FEDCOM101' from '96.235.30.58:1024' |
| 05:34.15 | jayrod422 | [Oct 29 01:28:55] NOTICE[11564]: chan_sip.c:18454 handle_request_invite: Failed to authenticate device "FEDCOM101" <sip:FEDCOM101@192.168.1.3>;tag=as56ca854e |
| 05:34.27 | jayrod422 | lol |
| 05:34.34 | jayrod422 | ok |
| 05:35.04 | dlynes | jayrod422: for sake of simplicity, considering your hostname is 'laptop', let's call the peer 'laptop', so change every occurence of 'sip.fed-com.com' on the remote side to 'laptop' |
| 05:35.13 | jayrod422 | k |
| 05:35.37 | dlynes | jayrod422: also change the value of username in the new laptop peer to 'laptop' as well |
| 05:35.44 | dlynes | jayrod422: and change the type to 'friend' |
| 05:36.03 | dlynes | jayrod422: then comment out the 'FEDCOM101' friend section |
| 05:36.17 | dlynes | jayrod422: so that way we only have one possibility on the remote end |
| 05:36.24 | dlynes | jayrod422: less confusing that way |
| 05:36.38 | dlynes | jayrod422: lemme know when you're done |
| 05:36.46 | jayrod422 | ok |
| 05:36.55 | jayrod422 | so everything on the remote change to no fqdn |
| 05:37.18 | jayrod422 | what about the host= field |
| 05:37.23 | jayrod422 | for the outbound route |
| 05:37.57 | dlynes | jayrod422: now, change your register line to 'register => laptop:a@sip.fed-com.com' |
| 05:38.15 | jayrod422 | k |
| 05:38.23 | dlynes | jayrod422: host field should remain 'host=sip.fed-com.com' |
| 05:39.12 | dlynes | jayrod422: now, at your command prompt on the remote end (linux command prompt), type 'dig sip.fed-com.com' to make sure your box can resolve that hostname |
| 05:39.39 | dlynes | jayrod422: is it ok with that hostname? |
| 05:39.53 | jayrod422 | yes |
| 05:40.11 | dlynes | jayrod422: ok...save your sip.conf file and your dialplan |
| 05:40.25 | dlynes | jayrod422: then give me a pastebin of the new versions of both of them on your remote |
| 05:40.44 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 05:40.46 | dlynes | jayrod422: i want to make sure all of that's good before we even touch the main machine's file |
| 05:42.05 | jayrod422 | ok everything on the laptop (remote) machine is cool |
| 05:42.21 | dlynes | jayrod422: i don't trust you |
| 05:42.25 | jayrod422 | lol |
| 05:42.31 | jayrod422 | rofl |
| 05:42.33 | dlynes | gimme the pastebins |
| 05:42.35 | jayrod422 | ok ok |
| 05:44.51 | jayrod422 | http://pastebin.com/m539f26bb |
| 05:52.35 | jayrod422 | ok so what did changing from fqdn on the remote do? |
| 05:55.13 | dlynes | jayrod422: now, for some reason the section in [....] is 'sip-out', not 'laptop' like I asked for |
| 05:55.59 | dlynes | Also, for some reason you have your sip.conf file duplicated |
| 05:56.01 | jayrod422 | you want to context also calledlapptop |
| 05:56.19 | dlynes | So you have two general contexts, two authentication contexts, and two sip-out contexts |
| 05:56.24 | jayrod422 | wtf did i just write |
| 05:56.32 | jayrod422 | lol |
| 05:56.34 | dlynes | you tell me |
| 05:56.35 | jayrod422 | no |
| 05:56.39 | jayrod422 | just 1 |
| 05:56.42 | jayrod422 | its 2 am here |
| 05:57.00 | dlynes | Did you hit Ctrl-V twice, or something? |
| 05:57.08 | jayrod422 | probably |
| 05:57.19 | jayrod422 | ok you want to know whats crazy |
| 05:57.23 | dlynes | Also, to uncomplicate things |
| 05:57.28 | jayrod422 | its f*cking working |
| 05:57.34 | jayrod422 | once we changed the names |
| 05:57.40 | dlynes | Perform these two regexes on your script: |
| 05:57.50 | jayrod422 | and i also took the liberty of changing the context and username on the server |
| 05:58.05 | dlynes | :%s/^\s*;.*$//g |
| 05:58.15 | dlynes | and :g/^\s*$/d |
| 05:58.35 | dlynes | that'll get rid of lines that are nothing but comments |
| 05:58.41 | dlynes | So that your files are easier to read |
| 05:58.55 | dlynes | it also gets rid of blank lines |
| 05:58.56 | jayrod422 | how do you do that in 1 command perl -? |
| 05:59.14 | dlynes | jayrod422: no idea...I just use vi or vim |
| 05:59.33 | dlynes | those are both ex commands |
| 05:59.55 | jayrod422 | anyway whenever i renamed the user stuff and the context shits working |
| 06:00.07 | dlynes | First one blanks out lines with comments...the second removes all blank lines |
| 06:00.30 | jayrod422 | but im still not understanding what was wrong with what i originally had other than the @fqdn in the dial string |
| 06:00.39 | dlynes | jayrod422: yes...amazing stuff happens when both sides actually match |
| 06:00.51 | jayrod422 | what didnt match before though |
| 06:00.54 | dlynes | jayrod422: it was obvious that there was no attention to detail when you typed in the two sides |
| 06:01.06 | dlynes | jayrod422: you didn't even have your typos consistent |
| 06:01.10 | jayrod422 | lol |
| 06:01.33 | jayrod422 | i mixed two different drugs earlier today to get where i am right now |
| 06:01.41 | dlynes | jayrod422: and you had a friend and a peer created on the laptop side, and you should have only needed a friend |
| 06:01.53 | jayrod422 | i been fucking with cisco all day and moved to this in the middle of the night |
| 06:02.10 | dlynes | jayrod422: sleep does amazing things |
| 06:02.20 | jayrod422 | lol |
| 06:02.25 | *** join/#asterisk Tim_Toady (n=moi@adsl54-170.kln.forthnet.gr) |
| 06:02.32 | dlynes | jayrod422: not to mention a drug-free mind |
| 06:02.55 | dlynes | btw |
| 06:02.55 | jayrod422 | i only mixed some anti flu drug with moosehead |
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| 06:03.33 | dlynes | the fqdn in the dial string and in your sip context were not correct, because it makes it confusing for you, and makes it confusing for anyone trying to help you |
| 06:03.42 | dlynes | asterisk doesn't care...it'll still work |
| 06:03.49 | dlynes | it's just not the correct way to do it |
| 06:04.30 | dlynes | and on that note...you said moosehead? |
| 06:04.39 | dlynes | You're from New Brunswick or Nova Scotia? |
| 06:04.51 | dlynes | jayrod422: ? |
| 06:05.15 | jayrod422 | neither |
| 06:05.17 | jayrod422 | pittsburgh |
| 06:05.21 | dlynes | ah |
| 06:05.32 | dlynes | and you drink moosehead? |
| 06:05.33 | jayrod422 | we have all the northern beers here |
| 06:05.35 | dlynes | very strange |
| 06:05.38 | jayrod422 | yea it was on sale |
| 06:05.56 | jayrod422 | a few bars here have it on tap also |
| 06:06.17 | dlynes | I guess it's more popular than labatt's blue or molson canadian? |
| 06:06.27 | jayrod422 | no its not |
| 06:06.34 | dlynes | btw |
| 06:06.35 | jayrod422 | labatts is the most popular around here |
| 06:06.43 | jayrod422 | then molson |
| 06:06.44 | dlynes | Don't get Alexander Keith's |
| 06:06.48 | dlynes | It's quite disgusting |
| 06:06.52 | jayrod422 | then some cheap shit i cant remember the name |
| 06:07.02 | jayrod422 | then maybe moosehead |
| 06:07.22 | dlynes | I think Nova Scotia just ships that crap outside the province...don't think anyone in Nova Scotia actually drinks Alexander Keith's |
| 06:07.33 | dlynes | Same thing with Schooner beer |
| 06:07.36 | jayrod422 | ive never heard of it |
| 06:07.40 | jayrod422 | or schooner |
| 06:07.51 | jayrod422 | our local shit is iron city |
| 06:08.07 | dlynes | Sounds like Steeler beer (Hamilton) |
| 06:08.07 | jayrod422 | ive never seen it outside this area |
| 06:08.10 | jayrod422 | lol |
| 06:08.15 | jayrod422 | its is the steelers office beer |
| 06:08.18 | dlynes | Hamilton's a major steel city, too |
| 06:08.21 | jayrod422 | official |
| 06:08.38 | dlynes | Iron City is the Steeler's official beer? |
| 06:08.41 | dlynes | Or Steeler beer is? |
| 06:08.44 | jayrod422 | btw what didnt match in my original pastebin |
| 06:08.49 | jayrod422 | steelers official beer |
| 06:08.50 | jayrod422 | http://pastebin.com/m4e55656b |
| 06:09.15 | dlynes | FEDCOM0 vs FEDCOM101 |
| 06:09.24 | jayrod422 | FEDCOM101 is a cisco phone |
| 06:09.51 | dlynes | jayrod422: then I guess you just happened to get lucky that everything's working |
| 06:09.54 | jayrod422 | FEDCOMO was what i was using for the two boxes to talk |
| 06:10.07 | dlynes | because you don't have a 'laptop' peer on the server side |
| 06:10.10 | dlynes | erm rather user |
| 06:10.18 | dlynes | or friend for that matter |
| 06:10.59 | dlynes | jayrod422: ah...so fedcom101 is a cisco phone connected to your laptop then? |
| 06:11.02 | jayrod422 | do you know if the context names have to be the same on the both side |
| 06:11.04 | jayrod422 | yes |
| 06:11.07 | dlynes | ah |
| 06:11.18 | dlynes | jayrod422: that's been my experience, yes |
| 06:11.26 | jayrod422 | that may have been what it was |
| 06:11.29 | dlynes | jayrod422: although, I'm far from being an expert on sip |
| 06:11.40 | dlynes | jayrod422: so i'm not quite sure which is which in the sip dialog |
| 06:11.56 | jayrod422 | while i know with other systems (not opensource) you just need a user/pass and no context is used |
| 06:12.18 | dlynes | jayrod422: well, asterisk seems to put that sip context into the context of the sip dialog, too |
| 06:12.46 | dlynes | jayrod422: I think one is the username and the other is the 'user' |
| 06:12.52 | dlynes | or something like that |
| 06:13.03 | jayrod422 | that seems to be the case |
| 06:13.22 | jayrod422 | oh well its working for now |
| 06:13.28 | dlynes | it was quite confusing the first time I tried to get asterisk to talk to an aastra phone |
| 06:13.41 | dlynes | because aastra's got three different things it uses |
| 06:13.43 | dlynes | btw |
| 06:13.51 | jayrod422 | my problem has just been other asterisk boxes behind nats |
| 06:13.56 | dlynes | fwiw, there's another value you can use for some stubborn devices, too |
| 06:14.00 | dlynes | fromuser=.... |
| 06:14.21 | jayrod422 | ill have to try that |
| 06:14.29 | dlynes | in case you're having issues with the remote end using your caller id as part of the user authentication |
| 06:14.32 | jayrod422 | ive been trying to get toshibas to talk through nat for awhile |
| 06:14.35 | jayrod422 | and that might work |
| 06:14.51 | dlynes | toshiba makes a voip phone? |
| 06:14.56 | jayrod422 | pbx |
| 06:15.04 | jayrod422 | and voip phones |
| 06:15.34 | jayrod422 | i have customers running them and cant get sip trunk up on them with asterisk on my side |
| 06:15.36 | dlynes | it's like a Panasonic TDA-300? |
| 06:15.41 | jayrod422 | i have to put them on our nextone |
| 06:15.54 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 06:16.22 | dlynes | or like a BCM? |
| 06:16.48 | jayrod422 | ive never seen teh panasonic one |
| 06:17.08 | jayrod422 | like a bcm |
| 06:17.10 | dlynes | If so, you might find that it's got its own proprietary methods for getting across a firewall that won't be compatible with anything other than itself |
| 06:17.41 | dlynes | Which is what Nortel does with its BCM systems and its SIP phones |
| 06:17.56 | dlynes | erm SIP-'compatible' phones |
| 06:18.07 | jayrod422 | have you got nortel pbx to talk to asterisk and a sip trunking provider? |
| 06:18.08 | dlynes | that being said |
| 06:18.17 | jayrod422 | as a |
| 06:18.19 | jayrod422 | shit wtf |
| 06:18.34 | dlynes | Aastra SIP phones support the Nortel version of firewall navigation |
| 06:19.18 | dlynes | nope...I've done very little with bcm |
| 06:19.50 | dlynes | Every time I've touched it, it's either old Meridian digital handsets, or someone else was running the IP end of things |
| 06:20.09 | dlynes | And they were only using it to hook up branch offices that were also using Nortel BCMs |
| 06:20.25 | jayrod422 | we have had some customer who couldnt figure out how to connect it with our systems and just gave it on voip providers in general |
| 06:20.26 | dlynes | so sip-intercommunication was a non-issue |
| 06:20.50 | jayrod422 | oh |
| 06:20.51 | jayrod422 | shit |
| 06:20.54 | dlynes | s/gave it on/gave up on/? |
| 06:20.58 | jayrod422 | gave up |
| 06:21.09 | dlynes | anwyays |
| 06:21.11 | dlynes | need sleep |
| 06:21.14 | dlynes | ttfn |
| 06:21.17 | jayrod422 | well i gotta sleep if im gonna be in by 9 tommorrow |
| 06:21.23 | jayrod422 | so thanks for your help |
| 06:21.29 | jayrod422 | ttyl |
| 06:21.30 | dlynes | i just need to walk from bed to the desk |
| 06:21.34 | jayrod422 | lol |
| 06:21.37 | dlynes | I've got a pretty long commute |
| 06:21.45 | jayrod422 | i have 2 flights of stairs |
| 06:21.50 | jayrod422 | so |
| 06:21.53 | jayrod422 | lata |
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| 06:40.18 | ChannelZ | hmm I think this is the first time I've seen amazon mp3 tracks at $1.29 instead of $.99 |
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| 06:46.46 | drmessano | YAY KARMIC IS HE..... oh, not just yet |
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| 07:24.47 | ChannelZ | They still have 2 days! |
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| 07:46.53 | cerwik | hi there |
| 07:47.03 | cerwik | can DAHDI compiles under 2.6.31? |
| 07:53.43 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:56.24 | cerwik | I am getting error about : error: ?struct net_device? has no member named ?do_ioctl? |
| 07:56.31 | cerwik | when compiling wanpipe |
| 07:56.35 | cerwik | any suggestion? |
| 08:02.27 | DigitalFlux | Guys |
| 08:02.36 | DigitalFlux | if i get the following line on the CLI |
| 08:02.49 | DigitalFlux | where i am trying to evaluate a GoToIf statement |
| 08:02.56 | DigitalFlux | GotoIf("SIP/500-08a76f60", "0?AgentLoggedIn:ContinueAddingAgent") |
| 08:03.03 | DigitalFlux | the "0" here .. |
| 08:03.16 | DigitalFlux | means the statement got 0 as a result |
| 08:03.25 | DigitalFlux | or that means false ? |
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| 08:09.09 | cerwik | nobody has wanpipe on 2.6.3x? |
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| 08:16.42 | tzafrir | dahdi trunk compiles under 2.6.31 |
| 08:17.01 | tzafrir | cerwik, dahdi trunk compiles under 2.6.31 |
| 08:17.17 | kaldemar | DigitalFlux: it's false |
| 08:18.44 | cerwik | i am using the dahdi from ubnut that seems fine but wanpipe is not compiling, so you mean wanpipe trunk? |
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| 09:02.17 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 09:02.19 | thomas | hiho |
| 09:02.33 | thomas | i try to dial a number and have a error like: [Oct 29 10:02:05] WARNING[10639]: chan_sip.c:2994 create_addr: No such host: 41079070 |
| 09:02.39 | thomas | my dialplan: http://paste.keks.be/4473/txt |
| 09:02.45 | *** join/#asterisk Tim_Toady (n=moi@adsl264-78.kln.forthnet.gr) |
| 09:02.56 | thomas | why "No such host: 41079070" ? |
| 09:03.05 | thomas | this number is in context intern-tm-js-fk { .. ? |
| 09:03.51 | thomas | any ideas? |
| 09:13.13 | thomas | huhu? |
| 09:13.53 | mort_gib | thomas: check that from and to are in the same context |
| 09:14.41 | thomas | mort_gib: ok |
| 09:14.50 | thomas | how i can say with dial command |
| 09:14.55 | thomas | other extention? |
| 09:15.07 | thomas | example: Dial(SIP/41074849/intern-tm-js-fk); ? |
| 09:15.13 | thomas | but, my question is: |
| 09:15.20 | thomas | i have include: intern-tm-js-fk; |
| 09:15.25 | thomas | and: 41074849 => { |
| 09:15.26 | thomas | <PROTECTED> |
| 09:15.26 | thomas | <PROTECTED> |
| 09:15.38 | thomas | how doeslnt work? mort_gib ? |
| 09:16.05 | thomas | ah |
| 09:16.15 | thomas | prio: the same context and THEN the include-context |
| 09:16.37 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:16.51 | thomas | mort_gib: how i can say: Dial(SIP/41074849) in context X ? |
| 09:17.46 | mort_gib | A sip user in context x can't dial a sip device in context y unless you include |
| 09:17.59 | thomas | mort_gib: what is the best solution? |
| 09:18.01 | thomas | jump? |
| 09:18.30 | kaldemar | thomas: Dial(Local/41074849@context) <-- is that what you mean? |
| 09:18.40 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 09:19.08 | kaldemar | thomas: are you trying to call an extension in the dialplan or dial somewhere with SIP directly? |
| 09:19.30 | thomas | kaldemar: [Oct 29 10:19:21] WARNING[26196]: chan_sip.c:3005 create_addr: No such host: intern-tm-js-fk |
| 09:19.40 | thomas | when i try: Dial(SIP/41074849@intern-tm-js-fk); |
| 09:19.43 | thomas | ah Local? |
| 09:21.20 | thomas | perfect :-) |
| 09:27.29 | thomas | kaldemar: emm, can you help me with a other question? :-) |
| 09:29.23 | kaldemar | thomas: just ask the question, i'll help if i can. |
| 09:29.28 | thomas | :-) |
| 09:29.29 | thomas | ok |
| 09:29.35 | *** join/#asterisk angryuser (n=angryuse@193.251.186.99) |
| 09:30.01 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 09:30.40 | thomas | kaldemar: http://paste.keks.be/4474/txt |
| 09:30.57 | thomas | kaldemar: when i dial out i have I'm on tzhe context: 08941079070 |
| 09:31.16 | thomas | but i would like when i dial 41074849 example then not via Dial(SIP/${EXTEN}@11111111); |
| 09:31.28 | thomas | i like over context intern-tm-js-fk { |
| 09:31.37 | thomas | kaldemar: is it posible or i need other rules? |
| 09:33.39 | kaldemar | i have trouble understanding what you mean. |
| 09:33.43 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 09:34.02 | thomas | kaldemar: when i dial: 0891234567 then over _X. => { |
| 09:34.04 | thomas | <PROTECTED> |
| 09:34.21 | thomas | but when i dial 41074849 then over 41074849 => { |
| 09:34.26 | thomas | context: intern-tm-js-fk |
| 09:34.32 | thomas | is it posible? |
| 09:35.12 | kaldemar | replace Dial(SIP/${EXTEN}@11111111); with Dial(Local/${EXTEN}@intern-tm-js-fk); |
| 09:36.11 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 09:37.57 | thomas | kaldemar: ok.. |
| 09:38.06 | thomas | kaldemar: _XX. == is minimum 3 numbers? or 2 ? |
| 09:38.17 | kaldemar | 3 |
| 09:38.29 | thomas | k |
| 09:39.35 | kaldemar | when a call lands in a context, it's extensions are matched first. includes are checked after extensions. so now your include is useless since you match everything with _X. |
| 09:40.55 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:42.35 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 09:44.03 | *** join/#asterisk TimToady_ (n=moi@adsl355-103.kln.forthnet.gr) |
| 09:47.09 | *** join/#asterisk thazza (n=thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 09:57.42 | thomas | kaldemar: ok. thank you |
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| 10:06.57 | *** part/#asterisk ZX81 (n=Matt_Rid@121.74.235.218) |
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| 10:09.39 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 10:11.27 | niekie | Hmm.. |
| 10:13.48 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
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| 10:22.10 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-235-218.telstraclear.net) |
| 10:22.43 | DND | guys i need some help configuring te121 card. i think its more complicated than setting up my aex card |
| 10:26.00 | *** join/#asterisk albertoandrade (n=albertoa@189.4.48.141) |
| 10:30.07 | *** join/#asterisk coppice (n=chatzill@host86-139-183-205.range86-139.btcentralplus.com) |
| 10:31.35 | Fister | okay, i feel like a fool. i got sip calls with slin16 now. |
| 10:31.42 | Fister | and audio works both ways. |
| 10:34.53 | *** join/#asterisk pukkita (n=pukkita@80.59.10.137) |
| 10:35.00 | pukkita | hiya all |
| 10:35.31 | pukkita | looking for Siempens Optipoint 410 SIP firmware, does anyone have a copy by chance? |
| 10:36.00 | ppc | not I |
| 10:36.05 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 10:39.42 | pukkita | thanks anyway :) |
| 10:45.44 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
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| 10:52.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:04.45 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 11:05.32 | TSM2 | which file does asterisk take its default local from? |
| 11:09.41 | *** join/#asterisk Da-Geek (n=Da-Geek@nat/redhat/x-hyemmxjedfrjsjpi) |
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| 11:12.07 | Gido-E | DND did you fix it aleardy? |
| 11:12.36 | drcarumas | hi guys, i'm trying to change asterisk default sound as global rule. I've tried to put language=pt_BR in [general] at sip.conf , and in peer, and got nothing. For now i'm setting it with "set(CHANNEL(language)" but i dont want that. Thanks for your help. |
| 11:13.16 | Gido-E | drcarumas i would try, the name of the dir in the sound package |
| 11:13.40 | drcarumas | the name of the dir? |
| 11:13.43 | drcarumas | the path? |
| 11:13.56 | drcarumas | like language=/xxxx/xxxx/xxx ? |
| 11:14.28 | Gido-E | it creates a dir (that is only for that language), in the language files. That dir name |
| 11:15.50 | drcarumas | i have tthe sounds working on the correct dir , i guess, cause wen I use Set(CHANNEL(language)=pt_BR) in dialplan the audios are playing in the right language. But still i'm having problems with that. For istance, when i use saydigits it only says alf digits in PT, then it goes to EN. lol . Thanks |
| 11:16.49 | TSM2 | which file does asterisk take its default language from? |
| 11:17.31 | Chainsaw | drcarumas: Is it falling back to english because your portuguese sound file collection is incomplete? |
| 11:17.37 | drcarumas | TSM2, hmm good question. how can i see that? |
| 11:18.50 | drcarumas | Chainsaw, no because if i change the number order it will play ok the first number. For instance, 1234 : it will say in PT one and other in english. If I change the order like 4321 it will say in PT 4 and other numbers in english. |
| 11:19.04 | drcarumas | so i assume that the audios are all ok. |
| 11:19.19 | Chainsaw | drcarumas: Interesting, So it's reverting to a default. |
| 11:19.40 | drcarumas | that's why i wanted to force the default language do PT so i dont have this kind of issue, and try to understand what's going on. |
| 11:19.48 | Chainsaw | drcarumas: Do you need one of these: /etc/asterisk/sip.conf:language=en_uk; Default language setting for all users/peers |
| 11:20.38 | TSM2 | if it does not find the audio file in the sub language folder, will it default to the standard US prompts? |
| 11:20.48 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 11:20.50 | drcarumas | Chainsaw, i've tried that. Under [general] . But i don't have nothing set there now. I don't know where i assumes the default language=en. |
| 11:20.52 | TSM2 | also whats the best format to have the audio files in, there are sever |
| 11:20.55 | TSM2 | several |
| 11:21.15 | Chainsaw | Mine are in GSM. |
| 11:21.41 | drcarumas | Hmmm ok i think i know what could be the problem |
| 11:22.08 | drcarumas | i have my peer allow=gsm only and I have audios in wav |
| 11:22.41 | drcarumas | That could be a problem eheheh! :| |
| 11:23.01 | TSM2 | i keep reading that its best to go to SLN format |
| 11:23.08 | drcarumas | however it's weird that it's playing the first audio right and others not. |
| 11:23.30 | drcarumas | TSM2, sorry for my n00biness , what's SLN format? |
| 11:23.50 | TSM2 | aparently its asterisk native format |
| 11:25.53 | drcarumas | In conclusion to my inital question, if I go to sip.conf and under [general] put there language=pt_BR it will change de default language to that? I don't need to force this config in other places right? |
| 11:25.53 | DND | Gido-E, sorry i was out supporting |
| 11:26.06 | DND | no i havent make it work till now |
| 11:26.33 | DND | still 31 channels to configure |
| 11:26.48 | drcarumas | I''ll remove all my Set(CHANNEL(language) from my dialplan and try with that config in sip.conf. And see how it goes. |
| 11:26.50 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 11:27.32 | Fister | ok, i'm ready to post my patch on the bugtracker. anyone have any advice so that i'm not the new guy who annoys people? |
| 11:29.19 | DND | Gido-E, our telco provide dsome info on span parameters |
| 11:29.51 | TSM2 | drcaruman: seems so, ive just tried it, the language you put there has to be the same as the subfolder name in /var/lib/asterisk/sounds/ |
| 11:30.28 | TSM2 | put all the files in that subfolder, if asterisk cant find what it wants from there it will revert back to the default files in /var/lib/asterisk/sounds/ |
| 11:30.49 | garymc | Yo, now im using ISDN30 in our office whats gonna be the best way to send faxes? |
| 11:31.05 | Chainsaw | garymc: Sending faxes over a PRI? |
| 11:31.06 | garymc | cos apparently i cant link one to the isdn 30 |
| 11:31.27 | Chainsaw | garymc: Not directly, no. |
| 11:31.28 | drcarumas | TSM2, thanks for your help. :) |
| 11:32.12 | garymc | so whats my best option. I thought faxing was old hat, and the last 3 companys ive spoke to have asked me to send them a fax!!!! |
| 11:32.30 | kaldemar | garymc: get an analog line for that and plug the fax machine directly into it. |
| 11:32.31 | DND | so our fax machines cannot directly send faxes? |
| 11:32.51 | DND | unless we buy the linksys device?> |
| 11:32.57 | garymc | not through PRI, i made this mistake this month |
| 11:33.18 | garymc | Kaldemar Ive got an anolouge line, ill have to use that then |
| 11:33.35 | Chainsaw | garymc: Fax has a legal status. |
| 11:33.35 | *** join/#asterisk lemmy (n=markus@eclipse/developer/Technology/lemmy) |
| 11:34.17 | Chainsaw | garymc: It is as good as sending a registered letter as far as the court is concerned. If you have a fax TX report (which they can have authenticated by the fax vendor), they received your note. </story> |
| 11:34.21 | *** join/#asterisk tareKhoury (n=tarekhou@tony11-128-131.inter.net.il) |
| 11:34.26 | Chainsaw | garymc: Much better then the legal can of worm an e-mail is. |
| 11:34.31 | DND | ChanServ, so in order to sendfax using a fax machine, we need to buy a device that can talk to asterisk first? |
| 11:34.59 | Chainsaw | I'm not convinced ChanServ is going to be a lot of help here, DND. |
| 11:35.06 | coppice | the snag is a fax tx report is totally bogus :-) |
| 11:35.36 | Chainsaw | coppice: Actually, it's not. It contains at least 30% of the original document, the machine serial number and has several distinct properties that a vendor can check. |
| 11:36.10 | Chainsaw | coppice: (Usually a line or dot pattern that looks benign but contains an encoded verification string) |
| 11:36.18 | coppice | it only contains verifiable information about the sending machine. it tells you nothing about where the fax went |
| 11:36.41 | Chainsaw | coppice: It tells you the phone number it went out to, and whether that machine confirmed that it received the document. |
| 11:37.03 | Chainsaw | coppice: If it did, that's the end of your responsibility. The other side was notified of what you had to say. This holds up in court. Strongly I might add. |
| 11:37.08 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-71-162-62-253.altnpa.east.verizon.net) |
| 11:37.09 | lemmy | Hi, I have a problem with Asterisk (1.4.21) and a pirelli dp-l10. Call signaling works fine, but when an incoming call is accepted at the Pirelli no speech comes through. NAT can't be an issue since Asterisk and the phone are connected over a VPN. Any hints where I can't start debugging this? |
| 11:37.24 | coppice | which actually tells you nothing. telex used to be accepted by courts before fax, and the reports from those were totally fakable |
| 11:38.01 | Chainsaw | coppice: If given the choice, I'd be more inclined to accept a fax TX report then an e-mail server log. |
| 11:39.09 | coppice | they are both about as useless as each other for verifying anything. courts just happen to accept the fax report as if its meaningful |
| 11:41.29 | Chainsaw | coppice: And it is, you can get an outside expert to verify it. The guy from Xerox/Brother/Canon has no bias one way or the other. Just a "yes, one of ours" or "no, that's fake because of X". |
| 11:41.34 | drcarumas | Guys is this suposed to give me empty result? exten => 12,1,NoOp(${LANGUAGE}) |
| 11:42.03 | coppice | Chainsaw: but it doesn't tell you if the FAX actually reached the other end. |
| 11:42.36 | kaldemar | drcarumas: if using 1.6, yes. |
| 11:42.51 | drcarumas | kaldemar, 1.4 here. thanks |
| 11:42.55 | Chainsaw | coppice: It does. Other end says "Okay, I have received your X pages which is all you said you have. Thank you, come again" |
| 11:43.28 | *** join/#asterisk kazaa_lite (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 11:43.30 | DND | sorry chainsaw, that was for ou |
| 11:43.32 | drcarumas | kaldemar, it seems that it's not assuming my language="anylanguage" , under general in sip.conf. |
| 11:44.11 | coppice | Chainsaw: not really. lets say the fax machine was out of paper. typically a modern machine will accept a few FAXes and hold them in RAM. whether they ever reach a piece of paper and get read is very uncertain |
| 11:44.29 | Chainsaw | DND: In my case, the fax machine is attached to a Patton Smartnode 4118 (8x FXO) device. It is then sent out on a British Telecom BRI by a Patton Smartnode 4634. |
| 11:45.06 | Chainsaw | coppice: The fax standard clearly states that a fax that is out of paper and does not have non-volatile memory available send a reject code. It does not accept the transmission. |
| 11:46.03 | coppice | Chainsaw: the FAX spec says absolutely nothing about buffering. in real machines its usually a RAM buffer. turn off the machine and its lost |
| 11:46.21 | tareKhoury | hello mates, who should i talk 2 to get help about asterisk SIP errors ? |
| 11:46.22 | kaldemar | drcarumas: ${SIPPEER(language)} will show you the language setting for a sip peer. LANGUAGE was deprecated to 1.4 anyway, so it's better not to use it. |
| 11:46.37 | Chainsaw | coppice: It does actually. Non-volatile storage is allowed, volatile storage is only allowed if there is paper available. |
| 11:47.03 | kaldemar | drcarumas: sorry, ${SIPPEER(<peername>,language)} |
| 11:47.04 | coppice | Chainsaw: can you point to the relevant section in the specs? |
| 11:47.19 | Chainsaw | coppice: Can you link me to the specification document that you are using? |
| 11:47.34 | drcarumas | kaldemar, should i put that with NoOp and get the current channel language? |
| 11:47.36 | TSM2 | garymc: you will need a ATA that supprts t28 |
| 11:47.45 | Chainsaw | TSM2: 38 :) |
| 11:47.46 | coppice | Chainsaw: The ITU specs, of course |
| 11:47.52 | kaldemar | drcarumas: yes |
| 11:48.28 | Chainsaw | coppice: That's not a link. Please try again. |
| 11:48.31 | drcarumas | ok, now i have language=fr , under [general] in sip.conf. If all goes well i should get language=fr with NoOp? am i correct? |
| 11:48.32 | garymc | TSM2 : an ATA? |
| 11:49.01 | Chainsaw | garymc: A little box that converts between SIP and an analog phone line. |
| 11:49.15 | Chainsaw | garymc: Or multiple phone lines, if it is a particularly fancy one. |
| 11:49.18 | tareKhoury | any ideas what causes this error : Unable to create/find SIP channel for this INVITE ????????????????? |
| 11:49.37 | garymc | where can i find one of those, how much are they, and how easy to install? |
| 11:49.47 | coppice | Chainsaw: http://www.itu.int/rec/T-REC-T.30/en of course. There is no other FAX spec for the top level of FAXing |
| 11:50.20 | kaldemar | tareKhoury: show a CLI output of the call with sip debug enabled. use a pastebin: |
| 11:50.24 | kaldemar | ~pb |
| 11:50.25 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 11:50.46 | coppice | Chainsaw: and few countries have any approval process for most of the protocols in a FAX machine, so machines can't be expected to follow T.30 to the letter, anyway. |
| 11:50.50 | tareKhoury | ohh |
| 11:50.56 | tareKhoury | can`t i send you a log gile? |
| 11:51.08 | tareKhoury | ok i`ll try to use this pastebin |
| 11:51.10 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:51.27 | Chainsaw | coppice: "Where the called destination is an automatic facsimile terminal which is not ready or not able to operate, the call should not be answered automatically." |
| 11:51.47 | coppice | Chainsaw: there isn't even any common practice about handling in a fax to e-mail system |
| 11:51.58 | drcarumas | kaldemar, ok that worked, thanks for the help. |
| 11:52.16 | coppice | Chainsaw: if a machine has RAM space it will answer and accept the call. nothing is said about storage being non-volatile |
| 11:52.20 | Chainsaw | coppice: That isn't a standardised process, correct. |
| 11:52.30 | Chainsaw | coppice: Are you aware of the concept of case law? |
| 11:52.53 | coppice | nothing is standardised at all, really. T.30 is a recommendation, not a standard |
| 11:53.03 | Chainsaw | coppice: It is a recommendation in force. |
| 11:53.25 | Chainsaw | coppice: Which like specific RFC documents from the IETF, is a standard. |
| 11:53.30 | coppice | I am aware of case law, and I am aware of the long and bogus history of how that worked out with telex. FAX is even worse |
| 11:53.59 | Chainsaw | coppice: I hope for your sake that you never appear before a judge with this argument. |
| 11:54.42 | coppice | friends have had to appear in front of judges to say yah or nah on telex log reports, and they found the whole thing a farce |
| 11:54.53 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 11:56.06 | pukkita | does anyone have SIP firmware for siempens optipoint? |
| 11:56.07 | Chainsaw | coppice: Standard forensics still apply. Make the alleged terminal produce another failure report and compare ink & paper. |
| 11:56.50 | Chainsaw | pukkita: Most Siemens handsets I've seen will automatically update their firmware from the Siemens website. Unless it's not based on "chagall" firmware? |
| 11:57.38 | coppice | Chainsaw: with telex that was really trivial. the only reason friends were able to shoot down telex logs was because the fraudster was sloppy. it takes more knowledge to fake faxes, but few machines offer even the slightest security |
| 11:59.08 | Chainsaw | coppice: The court doesn't work on the basis that a verification system must be infallible. It works on the basis that defrauding the verification system is a difficult and involved process. |
| 12:00.11 | tareKhoury | any ideas what causes this error : Unable to create/find SIP channel for this INVITE ??? SIP DEBUG at http://pastebin.com/d3f43b2f8 |
| 12:00.32 | coppice | that's the whole point. it isn't hard at all. you can easily construct a FAX and send it to most FAX machines, especially the thermal paper roll type, so it comes out precisely like a report |
| 12:00.56 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:00.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:03.29 | *** join/#asterisk scalex000 (n=chatzill@200.88.91.76) |
| 12:04.05 | tareKhoury | i need help solving this sip error : WARNING[15567]: chan_sip.c:3940 retrans_pkt: Maximum retries exceeded on transmission ?? |
| 12:04.45 | tareKhoury | sip debug can be found at http://pastebin.com/d3f43b2f8 |
| 12:05.57 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:07.00 | angryuser | tareKhoury, so there is no nat between you and the client ? |
| 12:07.28 | tareKhoury | the client is a dedicated server hosting ( CentOS 5.3 ) |
| 12:07.36 | tareKhoury | i tried using nat=yes , nat = no |
| 12:07.39 | tareKhoury | nat = route |
| 12:07.49 | tareKhoury | the problem still exists |
| 12:07.55 | scalex000 | good morning I need to know how to unregister all SIP |
| 12:08.32 | angryuser | tareKhoury, the question is, its there nat between you and cient ? |
| 12:08.34 | tareKhoury | could it be a connection lag? since i`m from israel and the server is in the UK |
| 12:08.39 | Fister | just stop registering, they will time out |
| 12:08.52 | tareKhoury | yes |
| 12:08.54 | tareKhoury | there is |
| 12:08.57 | tareKhoury | a nat |
| 12:10.30 | angryuser | tareKhoury, please set nat=yes and do a sip trace, also please tell me what is your local net adress (192.168xxxx else?) |
| 12:10.55 | tareKhoury | my local is 10.0.0.3 |
| 12:11.11 | tareKhoury | i will do a nat=yes and debug again |
| 12:11.24 | angryuser | tareKhoury, have you set localnet=xxx in you sip.conf ? |
| 12:11.32 | tareKhoury | no i did not |
| 12:11.37 | angryuser | your* |
| 12:11.51 | angryuser | tareKhoury, do it before doing a sip trace |
| 12:12.02 | tareKhoury | okay 1 min |
| 12:12.53 | angryuser | scalex000, remove register lines and do reload |
| 12:13.08 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 12:13.12 | scalex000 | ok |
| 12:14.44 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 12:14.46 | tareKhoury | angryuser : should it be localnet=10.0.0.3/255.0.0.0 ?? |
| 12:15.19 | angryuser | tareKhoury, localnet=10.0.0.0/255.0.0.0 (sure of mask ?) |
| 12:15.32 | tareKhoury | yes sure |
| 12:15.48 | angryuser | tareKhoury, then this is it |
| 12:16.25 | *** join/#asterisk robl^laptop (n=robl@m4c5336d0.tmodns.net) |
| 12:17.37 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-235-218.telstraclear.net) |
| 12:17.45 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-235-218.telstraclear.net) |
| 12:18.18 | TSM2 | im trying to temporarly disable my voicemail, but even though ive deactivated OnBusy/Noanswer etc the system still goes to VM, any ideas/ |
| 12:18.27 | tareKhoury | angryuser: the problem still exists .. should i debug? |
| 12:18.42 | angryuser | tareKhoury, yes pastebin |
| 12:18.47 | tareKhoury | ok 1 min |
| 12:20.06 | beek | TSM2: Yes, open up the CLI, ensure that verbose is at least 3, then watch what happens to see how it's getting to the voicemail app. |
| 12:21.52 | [TK]D-Fender | TSM2: Wrong channel <- |
| 12:23.03 | tareKhoury | angryuser : http://pastebin.com/m2b8191b7 |
| 12:23.07 | tareKhoury | here is the debug |
| 12:23.07 | beek | [TK]D-Fender: What's he got for a system? |
| 12:23.29 | [TK]D-Fender | beek: Usual prime offender |
| 12:23.32 | TSM2 | hees complaining as usual because someone has come here with a freepbx install |
| 12:24.05 | [TK]D-Fender | TSM2: Yess, because that is dialplan flow and FreePBX owns your sorry ass and isn't supported here |
| 12:24.12 | [TK]D-Fender | TSM2: This is not an "Asterisk problem" |
| 12:24.16 | beek | And yet I wonder why I can't buy parts for my Honda at the local Chevy dealership. |
| 12:24.58 | [TK]D-Fender | TSM2: And its not that your running FreePBX, its that your problem is with t, and not * itself |
| 12:25.00 | TSM2 | never said it was, but dialplans are asterisk and considering that this is a asterisk board, even debuggins comes under asterisk |
| 12:25.20 | TSM2 | these are custom dialplans but dialplans none the less |
| 12:25.32 | TSM2 | go tell anyone with a custom dialplan to go to their own board |
| 12:25.34 | [TK]D-Fender | TSM2: No, your dialplan is created by FreePBX, not by you. Yuo don't like how it builds them, take it up with the people who maintain it. |
| 12:25.42 | angryuser | tareKhoury, so the 213.8.128.131 is you uk provider ? |
| 12:25.52 | [TK]D-Fender | TSM2: If this is custom, feel free to pastebin something to show |
| 12:25.57 | tareKhoury | the UK server yes |
| 12:28.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:28.21 | angryuser | tareKhoury, are you are sure that he is using 4558 for sip ? |
| 12:29.17 | tareKhoury | angryuser: i own the server in the UK with asterisk on it |
| 12:29.33 | tareKhoury | and im trying to reach it through a softphone |
| 12:29.35 | tareKhoury | from here |
| 12:29.47 | tareKhoury | to test my application before routing voip numbers to it |
| 12:30.44 | garymc | Hi, my 0800 number went live today. It is assigned to a number in my DID range. Can i make certain extensions show as a 0800 number when that extension makes a call out? |
| 12:31.05 | [TK]D-Fender | garymc: That too is a #freepbx question .... |
| 12:31.09 | angryuser | tareKhoury, looks like you are sending ok for invite but you do not get any reply, do you have ports 10 000 - 20 000 5060 udp opened ? |
| 12:31.10 | [TK]D-Fender | garymc: Move along :) |
| 12:31.11 | garymc | ive tried setting it in DID on that extension but it shows up as the number in my DID range |
| 12:31.14 | garymc | whoops |
| 12:31.17 | garymc | wrong room |
| 12:31.21 | TSM2 | garymc: only if your provider allows you to specify that as a CID |
| 12:31.24 | Chainsaw | garymc: It depends on how BT have set up your PRI. In most cases they will filter out any numbers you present that aren't in the DID range. |
| 12:31.40 | *** join/#asterisk gsiener (n=gsiener@d-63-245-118-26.batelnet.bs) |
| 12:31.51 | [TK]D-Fender | garymc: and the answer is "yes" |
| 12:31.52 | tareKhoury | angryuser: the UK hosting told me that there is no closed ports, ALL OPEN, how can u really test if they are |
| 12:31.57 | [TK]D-Fender | garymc: "how" is their concern |
| 12:32.12 | tareKhoury | angryuser: i disabled the iptables firewall also |
| 12:32.17 | *** join/#asterisk tumisho (n=tumisho@196-209-87-115-rrba-esr-3.dynamic.isadsl.co.za) |
| 12:32.17 | garymc | ok thankks |
| 12:32.21 | angryuser | tareKhoury, yea, but for example centos has firewall enable by default, check it |
| 12:33.57 | tareKhoury | i disabled the iptables, i`ll check the SElinux |
| 12:34.35 | angryuser | tareKhoury, and explain in details you sip config, it is like sip softphone > nat > public server > provider , or something else ? |
| 12:35.09 | angryuser | or softphone > server > nat > provider ? |
| 12:35.55 | tareKhoury | okay i`ll try to check if okay now, then i will explain everything |
| 12:36.11 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 12:36.25 | angryuser | nah its is second variant |
| 12:38.40 | tareKhoury | angryuser: it`s like that softphone > nat > UK SERVER with asterisk on it |
| 12:39.11 | angryuser | ok |
| 12:45.48 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 12:46.42 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:48.20 | radovan | is there a way to create an active-active asterisk cluster with one iax trunk from telco? |
| 12:49.08 | radovan | something like iax trunk will be active only on one node, and if that node fails, second will start to register itself for incomming calls |
| 12:49.40 | angryuser | radovan, yes drbd + hearbeat |
| 12:49.51 | angryuser | and mysql replication if needed |
| 12:50.09 | angryuser | *heartbeat |
| 12:50.22 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 12:52.58 | [TK]D-Fender | angryuser: How will that mitigate selective * registration? |
| 12:53.14 | radovan | angryuser: that's active-passive |
| 12:53.42 | radovan | i want both nodes active, that's isn't a problem, since I have mysql cluster and dundi |
| 12:54.06 | radovan | but i can't figure out a reliable way to failover iax trunk from telco |
| 12:54.30 | ManxPower-work | You would have to make the failover server use the same MAC/IP. |
| 12:55.24 | *** join/#asterisk ska (n=ska@cpe-70-124-88-58.austin.res.rr.com) |
| 12:55.57 | radovan | ManxPower-work: that's not the way... |
| 12:56.02 | angryuser | [TK]D-Fender, it is possible to trigger reloads and many other stuff from heartbeat, moreover "service" could be iax trunks, it is not 100% active-active but why would he need that ? in my exp swich could take 1-2 sec |
| 12:56.02 | *** join/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 12:56.09 | radovan | just need to migrate iax registration |
| 12:56.20 | moos3 | has anyone installed asterfax with asterisk 1.6.0.5? |
| 12:57.28 | moos3 | I'm in need of a fax to email for asterisk |
| 12:57.39 | moos3 | can anyone suggest something? |
| 12:58.08 | angryuser | radovan, why cant you have a small out of service for 1-2 sec ? big system ? |
| 12:58.41 | *** part/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 12:58.43 | [TK]D-Fender | moos3: that version is not quite old and many serious exploits are in the wild. I highly recommend upgrading |
| 12:58.59 | radovan | angryuser: because of hotline. and it's quite a challenge :) |
| 12:59.00 | ManxPower-work | you're still going to lose all calls when the failover happens |
| 12:59.16 | radovan | i'm aware of that |
| 12:59.19 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 12:59.26 | angryuser | radovan, so you hotline works for 24/7 ? |
| 12:59.30 | angryuser | your* |
| 12:59.55 | radovan | yeah, 24x7 |
| 13:00.11 | moos3 | [TK]D-Fender: what version do you recommand? |
| 13:00.38 | [TK]D-Fender | moos3: 1.6.0.15. as per the topic |
| 13:00.44 | moos3 | k |
| 13:01.11 | moos3 | [TK]D-Fender: can I do the update in a Rolling fasion? |
| 13:01.20 | *** join/#asterisk wam_ (i=wam@unaffiliated/wam) |
| 13:01.22 | moos3 | I'm new to administration of asterisk |
| 13:01.23 | *** part/#asterisk lemmy (n=markus@eclipse/developer/Technology/lemmy) |
| 13:01.31 | [TK]D-Fender | moos3: No idea what you mean by that exactly... |
| 13:01.40 | [TK]D-Fender | moos3: DL, extract, compile, install |
| 13:01.56 | moos3 | keep up the server up while doing the compile and install |
| 13:02.11 | [TK]D-Fender | moos3: You can ruight up until it comes to to restart it to put it in effect |
| 13:02.18 | [TK]D-Fender | time* |
| 13:03.20 | moos3 | [TK]D-Fender: ok cool, We did a install from Rpm on our new server, What do you recommend for the fax-to-email part of it |
| 13:03.41 | [TK]D-Fender | moos3: What are you using now? |
| 13:04.14 | angryuser | radovan, well i know some systems capable of adding/removing nodes dynamicly but it is difficult to achieve this with asterisk, at least you need to think about this functionnality when building the system at the beginning, the easyest way is to ask your telco try to join 2 ip adress when first fails |
| 13:04.19 | moos3 | [TK]D-Fender: nothing, at the moment, but we are suggested asterfax |
| 13:04.45 | [TK]D-Fender | moos3: What are you using for PSTN connectivity? |
| 13:05.00 | moos3 | [TK]D-Fender: T1 with 23 lines |
| 13:05.34 | moos3 | [TK]D-Fender: we have a digium te220 |
| 13:05.37 | [TK]D-Fender | moos3: IAXModem + Hylafax |
| 13:06.15 | moos3 | [TK]D-Fender: K, is there anything special I need to compile for the IAXModem? |
| 13:06.41 | radovan | angryuser: that's the reason why I'm asking |
| 13:07.09 | [TK]D-Fender | moos3: get Googleing. |
| 13:07.15 | ManxPower-work | Now if you were using SIP and no NAT (i.e. reinvites) you might be able to keep calls up when the server fails over. |
| 13:07.29 | moos3 | [TK]D-Fender: K thanks for the help |
| 13:07.46 | radovan | angryuser: I can't get two trunks from telco, I asked for it and they are unable to provide it. |
| 13:09.42 | angryuser | radovan, you can do it with sip proxy, but with iax it is dead |
| 13:10.48 | angryuser | radovan, and i am curious how many calls do you get a 4h of the morning if you can not afford 2 sec downtime |
| 13:12.29 | ManxPower-work | Maybe Asterisk is not the correct solution for radovan? |
| 13:13.13 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 13:13.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.55 | *** join/#asterisk Skeeter- (i=Skeeter-@c216.218.2-65.clta.globetrotter.net) |
| 13:15.01 | Skeeter- | good morning everyone |
| 13:15.04 | angryuser | The cheapest out-of-the-box softswich with active-active i saw was around 80k thats too much for 2 seconds for me |
| 13:16.09 | Skeeter- | i purchased some g729 codec, problem is that is i get more calls then registered codec in between my servers, calls cant be made, if i had allow;g729,gsm will g729 be used in priority |
| 13:17.01 | ManxPower-work | Skeeter-: no, Buy more licenses or create an additional connection between your two servers and handle "license failover" in your dialplan. |
| 13:17.16 | Katty | :> |
| 13:18.46 | TSM2 | Skeeter-: as manxpower said, make two trunks between the servers, first trunk for your g729 stuff and put a max calls limit on it equal to ammount of licences you have, then have another trunk for gsm g711 etc... then create a plan that uses them in that order |
| 13:19.46 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:19.47 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:19.49 | ManxPower-work | Licenses are so cheap you might as well just buy extras |
| 13:20.03 | radovan | Skeeter-: write it in two lines |
| 13:20.06 | radovan | allow=g729 |
| 13:20.13 | radovan | allow=gsm |
| 13:20.17 | TSM2 | what is the licencing rules on that, which countries does g729 apply in? |
| 13:20.20 | ManxPower-work | radovan: that won't work |
| 13:20.32 | radovan | hmm, since which version? |
| 13:20.36 | ManxPower-work | TSM2: according to the ITU, all countries |
| 13:20.51 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-mpcxaphwjtkvjgma) |
| 13:20.52 | radovan | working for me on 1.4.x |
| 13:21.17 | Skeeter- | radovan: tahnks for the tip iw ill give it a try |
| 13:21.19 | TSM2 | http://www.howlertech.com/products/howlets/pricing/ |
| 13:21.22 | ManxPower-work | radovan: Asterisk will allow the call to be processed (and fail) if you run out of licenses |
| 13:21.35 | Skeeter- | TSM2: whtnks to you too, this fives me 2 solutions |
| 13:23.24 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:23.24 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:23.56 | radovan | ManxPower-work: that's quite a bug, if it cannot check if there is free lincence to make a call and fallback to another codec |
| 13:24.18 | ManxPower-work | radovan: not at all. Asterisk will happily to g729 PASSTHRU if you have no licenses. |
| 13:24.22 | radovan | angryuser: i can afford downtime for 1-2 minute, i'm just currious |
| 13:24.30 | ManxPower-work | Passthru is pretty useless to most people, however |
| 13:27.38 | [TK]D-Fender | [09:19]<radovan>allow=g729 |
| 13:27.40 | [TK]D-Fender | [09:20]<radovan>allow=gsm |
| 13:27.41 | [TK]D-Fender | This is pointless |
| 13:27.43 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 13:28.05 | [TK]D-Fender | If G.729 is the preferred choice, it will be the ONLY choice. |
| 13:28.23 | [TK]D-Fender | There is no such thing as "failover to other codec when licenses run out" |
| 13:29.02 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 13:29.25 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
| 13:32.57 | Chainsaw | TSM2: "32-bit linux binary". What is this, 2003? |
| 13:34.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:34.47 | *** join/#asterisk af_ (n=getsmart@88-149-240-182.dynamic.ngi.it) |
| 13:37.03 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 13:39.30 | *** join/#asterisk knctrnl (n=aembrey@76.164.169.130) |
| 13:41.49 | Skeeter- | There is no such thing as "failover to other codec when licenses run out" |
| 13:42.08 | Skeeter- | [TK]D-Fender: you suggest another trunk with another codec?? |
| 13:42.16 | *** join/#asterisk Elwell (n=aelwell@freenode/staff/elwell) |
| 13:44.41 | Elwell | Hi folks, I have a [fedora] section in sip.conf with my fedoraproject details, what magic do I need so that my ekiga softphone ([andrew] in sip.conf) can connect to sip:conferences@fedoraproject.org (at the moment it only works wit sip:extension@myasteriskserver.adddress |
| 13:45.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:45.49 | *** join/#asterisk Grof (n=grof@89.201.165.226) |
| 13:47.20 | moos3 | [TK]D-Fender: ok I have 1.6.0.15 up and running, I have soundsp installed, but I'm not sure how to set up the iaxmodem |
| 13:48.34 | ManxPower-work | Elwell: if you connect to "sip:conferences@fedoraproject.org" then you are bypassing Asterisk. |
| 13:48.49 | Elwell | aaah OK |
| 13:50.08 | Elwell | so presumably I'd need to make sure I have local copy of the authentication. dammit. was hoping it'd all happen in *. /me RTFMs |
| 13:50.43 | ManxPower-work | Elwell: doing it the way you do now should work |
| 13:51.21 | *** join/#asterisk jtrimmer (n=jtrimmer@75.151.66.133) |
| 13:51.54 | Elwell | ManxPower-work: but my local ekiga doesn't have my fedora account in, just the authentcation to get to my * server. if I bypass that then it needs to authenticate me to fedora |
| 13:52.03 | *** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 13:52.15 | ManxPower-work | Elwell: on Asterisk exten =>666,1,Dial(SIP/fedora) |
| 13:52.20 | ManxPower-work | dial 666 in your ekiga phone. |
| 13:52.21 | ManxPower-work | done! |
| 13:53.00 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 13:54.51 | _trine | can someone explain why when say I dial a 6 figure phone number like 123456 it connects with the other phone but there is no ringing tone in my phone ear piece but when I dial say 01652 245678 there is a ringing tone before the phone connects as you would normally expect |
| 13:55.33 | _trine | whats should I be looking at to correct this behavior |
| 13:56.15 | _trine | quit |
| 13:56.19 | _trine | opps |
| 13:56.28 | jtrimmer | I have an asterisk server behind a sonicwall tz190 and a remote extension behind a linksys wrt310n. I can make calls from the extension to any phone on the same network as asterisk and out through the pots. audio works fine both directions. If I try to call the remote extension though it tells me it is unavailable. Also in freepbx panel that extension is half greyed out. I think it might be the linksys blocking traffic but |
| 13:56.28 | jtrimmer | though I might ask if anyone else might of seen this before. |
| 13:56.54 | Skeeter- | ManxPower-work: could you help me with my CID issue |
| 13:57.28 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:58.14 | knctrnl | I am bringing callers in and asking for and recording their employee ID into a varible as well as data from a few other questions I am asking them. I am echoing this data to a file. I am then asking them to record more information which I record to an mp3. Whats the easiest way to pick these files up and send as an email after the call is finished? |
| 14:02.10 | ManxPower-work | ~freepbx |
| 14:02.11 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:04.01 | _trine | ManxPower-work: why should I get a ringing tone on some calls and not others depending on the length of the number,, any ideas for me |
| 14:05.14 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 14:05.15 | *** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM] |
| 14:05.15 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 14:05.15 | *** join/#asterisk Nivex (i=ac1f0304@user-0c2h3cu.cable.mindspring.com) [NETSPLIT VICTIM] |
| 14:07.07 | *** part/#asterisk jtrimmer (n=jtrimmer@75.151.66.133) |
| 14:07.41 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:08.47 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 14:09.09 | [TK]D-Fender | knctrnl: E-Mailing is not an * task. Call an external script in the "h" exten on termination of the call |
| 14:10.38 | knctrnl | Also does anyone have any suggestions of a first language to learn if I wanted to do AGI? There are so many options and I dont know how to do any programing. Is one easier to learn than the other? |
| 14:10.46 | *** part/#asterisk thazza (n=thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 14:10.50 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:11.46 | [TK]D-Fender | knctrnl: AGI doesn't care about the language. Pick one you feel will be most useful for you for other things |
| 14:12.37 | [TK]D-Fender | knctrnl: PHP is common for web stuff, Perl is nore OS-centric and gives you better dev control, etc... its a personal thing |
| 14:13.12 | [TK]D-Fender | knctrnl: I know some PHP, so that's what I would use, but there are load issues at a certain point |
| 14:14.05 | Katty | hands out bowls of oatmeal |
| 14:14.30 | beek | Katty: I hope that they're original oats (no processing) and not "quick" oats. |
| 14:15.11 | Katty | no, they're old fashioned oats |
| 14:15.42 | *** join/#asterisk fainsys (n=fainsys@32.134.103.103) |
| 14:16.36 | Katty | put peanut butter and banana in mine :> |
| 14:18.25 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 14:19.54 | beek | I just use bananas, raisins and cinammon |
| 14:20.11 | beek | accepts bowl from Katty |
| 14:20.25 | beek | Mmmmm... thanks! |
| 14:20.51 | Katty | :> |
| 14:21.18 | beek | [TK]D-Fender: God uses Perl |
| 14:21.35 | Katty | something tells me DNA wasn't written in Perl |
| 14:21.38 | Carlos_PHX | looks at hemp, veggie, and peanut butter smoothie on his desk... |
| 14:21.49 | ManxPower-work | MMmmmm...hemp. |
| 14:21.54 | [TK]D-Fender | beek: http://xkcd.com/378/ |
| 14:21.55 | beek | Katty: no, but the sequence for DNA was processed using Perl. |
| 14:22.00 | Katty | hehehe |
| 14:22.49 | beek | :D |
| 14:24.05 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:24.38 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 14:24.43 | Katty | can't finish oatmeal :/ |
| 14:26.30 | tzafrir | beek meant http://xkcd.com/224/ |
| 14:27.06 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:28.18 | *** join/#asterisk fainsys (n=fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 14:28.40 | *** join/#asterisk wam_ (i=wam@unaffiliated/wam) |
| 14:31.47 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:32.19 | eppigy | TRABAJO |
| 14:32.42 | Katty | HELLO THAR DAVE |
| 14:32.52 | eppigy | hiya :> |
| 14:33.11 | Katty | hugs eppigy |
| 14:33.19 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 14:33.33 | jaytee | hugs Katty |
| 14:33.57 | jaytee | Buenos Dias, Senor Dave! |
| 14:34.01 | beek | tzafrir: EXCELLENT! :D |
| 14:34.20 | beek | LISP: Lots of Irritating, Spurious Parenthesis |
| 14:34.25 | beek | Morning jaytee |
| 14:34.49 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 14:35.26 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-71-162-62-253.altnpa.east.verizon.net) |
| 14:35.37 | torrancew | hey all, how bulky is the config for setting up a numeric prompt, and using that to direct unanswered calls to voicemail? |
| 14:35.41 | eppigy | holds Katty close |
| 14:35.47 | torrancew | or more specifically, what does that entail? |
| 14:35.57 | eppigy | HOLA SENOR JAYTEE |
| 14:36.43 | [TK]D-Fender | torrancew: basic IVR. couple of lines of dialplan. |
| 14:37.04 | moos3 | how does one get app_txfax to build |
| 14:37.12 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 14:37.26 | torrancew | could you give me the syntax for just 1 number? perhaps, have 1 dial SIP/john? |
| 14:38.01 | beek | ~book |
| 14:38.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:38.36 | torrancew | cool, didn't find it in there at a glance. i'll look closer |
| 14:38.55 | torrancew | btw, thanks. you guys make one of the most helpful channels i've found on here |
| 14:47.57 | TheDavidFactor | http://xkcd.com/293/ |
| 14:49.22 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 14:50.19 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
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| 14:50.19 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:50.30 | neurosys | Got my Polycom SoundPoint 550 today. Not to happy with the speakphone tho. Everyone complains i sound real far :( |
| 14:50.49 | *** join/#asterisk chendy (n=chatzill@121.15.28.224) |
| 14:51.58 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:52.06 | [TK]D-Fender | neurosys: Depends what you, your room,, your gain settings, and their environment look like as well as codecs, etc |
| 14:53.03 | *** join/#asterisk fainsys_ (n=fainsys@32.134.103.103) |
| 14:53.13 | neurosys | [TK]D-Fender: Gonna play with it. I rule out enviroment because the Linksys SPA-942 had no trouble. I'll keep playin tho :) |
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| 14:54.52 | Katty | http://sciencewins.files.wordpress.com/2009/06/droids1.jpg |
| 14:55.18 | neurosys | Lol! |
| 14:58.38 | *** join/#asterisk Gnutoo (n=gnutoo@host133-133-dynamic.54-79-r.retail.telecomitalia.it) |
| 14:58.57 | Gnutoo | hi, I bet there is no software for making a gtalk server? |
| 14:59.10 | TheDavidFactor | that's funny! |
| 14:59.25 | Katty | well i'm sure there is. |
| 14:59.32 | Gnutoo | ah ok |
| 14:59.33 | Katty | and i'm sure google has copyrights all over it. |
| 15:00.04 | Gnutoo | lol ....I wasn't very clear...can I make my own gtalk server in some way? |
| 15:00.22 | Katty | why don't you write google and ask. |
| 15:01.02 | Gnutoo | mmm |
| 15:02.53 | *** join/#asterisk ircast (n=elmarp@p5492D1DF.dip.t-dialin.net) |
| 15:04.20 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:04.58 | radovan | how can I debug problems with voice quality? |
| 15:05.15 | radovan | i have "rc_avpair_new: unknown attribute 1490026597" in daemon.log a scrambled voice on all calls |
| 15:07.12 | *** join/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 15:13.07 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
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| 15:23.33 | ircast | Hi, I'm new to asterisk and I'm looking for a docu of the asterisk database. Any hint or link? |
| 15:23.36 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 15:23.53 | ManxPower-work | ~doc |
| 15:23.54 | infobot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 15:24.15 | *** join/#asterisk robl^laptop (n=robl@208.54.83.71) |
| 15:24.27 | ircast | THX |
| 15:25.05 | torrancew | in extensions.conf, can i assign and use an "ALLINTERNAL" variable that is equal to ${VAR1}&${VAR2}, etc, or would that barf? (VAR1 = SIP/line1, etc) |
| 15:26.02 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp176-22.adsl.forthnet.gr) |
| 15:26.08 | ManxPower-work | torrancew: yes, you can do |
| 15:26.24 | torrancew | ManxPower-work: thanks, wanted to make sure my dialplan wouldn't barf before i implemented it |
| 15:26.51 | ManxPower-work | torrancew: have you read the Asterisk book? |
| 15:27.07 | ManxPower-work | torrancew: depends on where in the dialplan y |
| 15:27.13 | torrancew | i've used it as a refernce |
| 15:27.24 | torrancew | i haven't had time to read it cover to cover, but i'm working on it |
| 15:27.45 | ManxPower-work | torrancew: did you read the docs in the asterisk source? |
| 15:27.48 | torrancew | they decided to tell me they wanted asterisk pbx, and have it implemented, within the course of about 1 week |
| 15:27.59 | ManxPower-work | torrancew: you will fail |
| 15:28.00 | torrancew | same situation there |
| 15:28.11 | torrancew | well, we're riding ok so far |
| 15:28.26 | torrancew | i'm just implementing their desired features in phases |
| 15:28.33 | torrancew | and trying to watch my back along the way |
| 15:29.14 | p3nguin_ | gnutoo: It's google talk just an xmpp server? You can certainly make your own xmpp setup -- install jabberd. |
| 15:29.20 | torrancew | ManxPower-work: i appreciate the advice though, and if i get the other projects i'm on out of the way, i'll read the docs this weekend |
| 15:29.24 | ManxPower-work | People should spend at least a month learning Asterisk, then another 2 - 3 months in testing before deploying in production |
| 15:29.40 | torrancew | ManxPower-work: i don't disagree |
| 15:30.53 | *** join/#asterisk twanny796 (n=chatzill@c141-183.i01-4.onvol.net) |
| 15:31.12 | twanny796 | anylink to a good book on Euro ISDN? |
| 15:38.33 | Carlos_PHX | Ah, another day, another stupid reason for RTP to be broken. Fun fun fun. |
| 15:38.50 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
| 15:38.51 | Carlos_PHX | Customer's firewall thought our traffic was a port scan and auto-blacklisted our Asterisk server. |
| 15:39.06 | torrancew | if i don't create a busy or unavailable message for a VM account, how can i expect a call to VoiceMail(EXTEN) to act? |
| 15:40.24 | Carlos_PHX | Do you mean you don't create either, or create only one? |
| 15:41.25 | torrancew | Carlos_PHX: don't create either |
| 15:41.34 | raden_work | anyone interested in a HP procurve 1800-24G ? I have 2 for sale |
| 15:42.15 | moos3 | how can i see what t1 channels are in use? |
| 15:44.08 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
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| 15:44.45 | ryduh | gooooood morning! |
| 15:45.02 | outtolunc | morn'n |
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| 15:47.04 | [TK]D-Fender | torrancew: regardless about what you create, you aren't specifying which one to play |
| 15:47.25 | torrancew | [TK]D-Fender: right |
| 15:48.29 | torrancew | what i'm trying to figure out is, a) can i have someone leave a VM without a prompt being played by the VM application, and b) if not, how can i specify an existing audio file for a VM account's greeting |
| 15:49.28 | [TK]D-Fender | torrancew: A) yes |
| 15:49.46 | [TK]D-Fender | torrancew: To choose which one is played, you CHOOSE it when you call Voicemail() |
| 15:49.57 | [TK]D-Fender | torrancew: "core show application voicemail" |
| 15:50.13 | torrancew | [TK]D-Fender: right, VoiceMail(exten,[u|b]) |
| 15:50.30 | [TK]D-Fender | torrancew: Good... progress... |
| 15:51.15 | torrancew | [TK]D-Fender: that's what i needed. thanks once more |
| 15:51.40 | torrancew | [TK]D-Fender: how might i specify an existing file for a greeting in voicemail.conf? |
| 15:52.20 | torrancew | (i feel that VoiceMail(exten,s) will be fine, but the boss may soon decide "wait, in addition to the ivr, i want _this_ played too" |
| 15:52.37 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 15:56.47 | torrancew | and digging through the sample voicemail.conf, i don't see any greeting parameter |
| 15:56.51 | torrancew | also couldn't find one in the book |
| 15:58.21 | Gnutoo | ok thanks but gtalk also do voice |
| 16:02.41 | torrancew | adoes anyone know the parameter to set an account's greeting file (if possible) in voicemail.conf? |
| 16:03.07 | torrancew | *face-palm. never mind, i can do a playback, and a voicemail(exten,s) |
| 16:03.18 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
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| 16:07.40 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:08.14 | ryduh | torrancew: :) The words face-palm make me crack up a little inside every time I read them |
| 16:10.17 | moos3 | whats the best fax to email for * |
| 16:10.34 | *** part/#asterisk ircast (n=elmarp@p5492D1DF.dip.t-dialin.net) |
| 16:13.28 | jaytee | Katty, you around? |
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| 16:16.57 | angryuser | moos3, depends, what do you use for faxing ? |
| 16:18.30 | moos3 | well its all incoming faxes |
| 16:18.46 | moos3 | I cant seem to get txfax to build with * 1.6.0.15 |
| 16:19.12 | moos3 | am I doing something wrong? I have got the dsp built and installed |
| 16:19.24 | angryuser | moos3, so you fax dont work and you want fax to email ? |
| 16:19.34 | angryuser | your* |
| 16:19.56 | moos3 | well I'm trying to get it build first but wanted some information |
| 16:20.41 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 16:20.47 | angryuser | moos3, do you want to fax over ip or over bri T1/E1 ? |
| 16:21.00 | moos3 | when i try to build asterisk i get this http://pastie.org/pastes/674999 |
| 16:21.09 | moos3 | over my T1 |
| 16:21.55 | angryuser | moos3, drop it, install hylafax with iaxmodem and a very nice web interface called avantfax |
| 16:22.03 | torrancew | what's a good audio conversion program to convert into .gsm format? |
| 16:22.21 | jaytee | torrancew, sox |
| 16:22.31 | jaytee | or you can do file convert from the CLI |
| 16:22.38 | angryuser | moos3, my users are happy with it |
| 16:22.49 | torrancew | jaytee: sox streams or does true conversion? |
| 16:22.55 | moos3 | angryuser: my concern is we are talking alike 10K+ of incoming faxes |
| 16:23.02 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:23.05 | angryuser | moos3, and ? |
| 16:23.09 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:34b2:7abd:b2c9:8871) |
| 16:23.09 | jaytee | torrancew, true conversion |
| 16:23.14 | cusco | help!! |
| 16:23.15 | moos3 | it should handle that? |
| 16:23.16 | cusco | http://paste.debian.net/50277/ |
| 16:23.21 | cusco | see those errors? |
| 16:23.27 | cusco | Oct 29 16:18:59] ERROR[25864] chan_dahdi.c: !! Got reject for frame 5, but we have nothing -- resetting |
| 16:23.40 | cusco | all of our pstn calls are constantly going down |
| 16:23.41 | angryuser | moos3, 10k per month ? |
| 16:23.46 | moos3 | no a day |
| 16:24.09 | moos3 | we are a document collection point various clients |
| 16:24.24 | drcarumas | torrancew,example : "sox xxxxxx.wav -r 8000 -a xxxxx.gsm" |
| 16:24.31 | torrancew | drcarumas: thanks alog |
| 16:24.33 | torrancew | alot* |
| 16:24.36 | angryuser | moos3, hylafax will handle that not sure that avantfax is suitable for the mass |
| 16:25.18 | angryuser | moos3, replace avantfax with native hylafax clients |
| 16:25.23 | *** join/#asterisk moy (n=moy@74.12.131.148) |
| 16:25.47 | ryduh | angryuser: how many faxes do you send/receive a day with 1 hylafax server? |
| 16:25.47 | moos3 | we are going to email them to a database processing script |
| 16:26.28 | angryuser | ryduh, the maximum i had is around 1 k the server was cold |
| 16:27.01 | moos3 | nice |
| 16:27.32 | torrancew | hmm |
| 16:27.37 | torrancew | sox doesn't like my input formats |
| 16:27.45 | ryduh | angryuser: how often do you have problems? how often do you have to spend time to maintain it? we currently use an email to outgoing fax system myfax.com and we'd love to host it ourselves |
| 16:27.50 | torrancew | looks like the guy sent me MS ADPCM |
| 16:28.11 | drcarumas | convert it to wav or mp3 |
| 16:28.39 | angryuser | moos3, with 10 k you need really think of the scructure of your setup how to build HA and save faxes |
| 16:28.52 | drcarumas | torrancew, check "man sox" if he can handle that format |
| 16:29.15 | angryuser | ryduh, the faxes are sent by mail and 1 time a week there is a script to launch |
| 16:29.34 | torrancew | drcarumas: don't have to - he's the one that complained about it ;-) |
| 16:29.37 | moos3 | angryuser: I have a high proformance san cluster, and a quadcore box with 4 gigs of ram, and single t1 |
| 16:29.49 | moos3 | what should I put this setting to Maximum number of concurrent jobs to a destination [1]? |
| 16:29.55 | angryuser | ryduh, ...with crontab, actually i have not looked for a year xD |
| 16:30.12 | jaytee | wow, it's not everyday you read about someone using a ferret as a weapon. http://www.msnbc.msn.com/id/33531981/ns/us_news-weird_news/ |
| 16:30.15 | drcarumas | torrancew, oh ok. EHeh! well go for it :) |
| 16:30.25 | angryuser | ryduh, i dont work for the guys i setup it anymore, but they would call me if something gone wrong |
| 16:30.34 | torrancew | drcarumas: also, it was in wav format, just wrong encoding i suppose, so i'll ship it out to flac, if sox has a problem with that, i have a problem with sox |
| 16:30.50 | drcarumas | torrancew, i use audacity for some audio editing try that, if you dont have any other audio editing tool. |
| 16:30.59 | torrancew | drcarumas: i've got a few, i'll get it liced |
| 16:31.01 | torrancew | licked* |
| 16:31.10 | torrancew | i hate macbook keyboards |
| 16:32.13 | angryuser | moos3, i cant tell you, you need to test i coz i had never 10k per day, what i can say hylafax was here for a years, and it was used in a large setups, google a bit |
| 16:32.22 | *** join/#asterisk wam_ (i=wam@unaffiliated/wam) |
| 16:32.23 | ryduh | angryuser: that's comforting. do you only run one server? |
| 16:33.41 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 16:33.49 | angryuser | ryduh, yes one server, the faxes actually stored for a week locally(raid1) , then send to a remote storage (just a mount) with redundant hdds |
| 16:33.51 | moos3 | angryuser: cool thx, with is the best way to install iaxmodem is to build from source on centos? |
| 16:33.53 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 16:34.12 | angryuser | moos3, yes from sources its is simple |
| 16:35.15 | ryduh | lol how did Hylafax skip version 5 altogether. from version 4.4.4 to 6.0.0 |
| 16:35.19 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:35.29 | angryuser | moos3, dont drop avantfax ask a avantfax dev if they have same large setups with their web interface, it is really handy and usefull |
| 16:35.47 | angryuser | ryduh, there is hylafax and hylafax+ (fork) |
| 16:35.56 | angryuser | ryduh, not the same thing |
| 16:37.33 | ryduh | angryuser: so v5.* is hylafax+? Do you know the main difference between the two? |
| 16:37.34 | moos3 | ok cool |
| 16:41.36 | angryuser | ryduh, hylafax+ pretend to be more advanced, but both are pretty stable |
| 16:41.50 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:43.01 | ryduh | moos3: you accept over 10k faxes a day on a T1 line and one server? |
| 16:43.17 | moos3 | thats the plan |
| 16:43.43 | ryduh | moos3: no, how much do you currently handle? |
| 16:43.53 | moos3 | it might not be a 10K at first, we are setting up a trail run which should be about 2500 a day |
| 16:43.58 | angryuser | moos3, i know a pro who is doing fax setups for providers, buy you have to pay $$ |
| 16:44.13 | moos3 | aww |
| 16:44.17 | ryduh | angryuser: are we talking $$ or $$$$$$? |
| 16:44.43 | angryuser | ryduh, ~$$$$ |
| 16:45.37 | Wollie | Does anyone knows how to disable Packet2Packet bridging? When I make a call sometimes only 1 side hears audio. Everytime that happens this is in the log: |
| 16:45.40 | Wollie | Hi James, |
| 16:45.43 | Wollie | I've been following your blog throughout the year and although I never left any comment before (I'm just not such a big commenter) I really love it. I'm living in Costa Rica (haven't you noticed me in your statistics?) and Formula1 coverage here is nihil. Live broadcasting is horrible with commentators from the studio that manage to talk through all the radio broadcasts. I'm so thankfull for internet and your blog so I can keep following Formula1 like how I |
| 16:45.47 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 16:45.50 | Wollie | I will definately order your book! Just to be sure it arrives I'll have it delivered to friends in The Netherlands. The mailservice here is not so reliable :/. |
| 16:45.54 | Wollie | sorry, not that :) |
| 16:45.57 | Wollie | this: |
| 16:45.59 | Wollie | Packet2Packet bridging SIP/201-00802b40 and SIP/301-007f9b40 |
| 16:46.07 | Wollie | stupid Windows that doesn't copy when you select text ;) |
| 16:46.23 | moos3 | lol thanks for the help to get started :) |
| 16:46.52 | ryduh | angryuser: I wish we could justify spending that but we can't right now. Guess it will be a trial and error kind of development |
| 16:46.54 | file | Packet2Packet bridging is an optimized way of exchanging the RTP inside of Asterisk, it literally reads in the RTP, and sends it directly out |
| 16:47.08 | file | it shouldn't be possible for that to cause one way audio |
| 16:47.12 | hesco | I am working to add a new DID to my existing * server. iax2 show registry | peers, plus the registration statements are posted at: http://pastebin.com/d10278ae6. My existing DID is routing incoming calls appropriately, but the new DID is failing to make its presence known in my logs, much less by ringing my desk. Can anyone advise me what I may be missing, please? |
| 16:48.00 | angryuser | ryduh, i am not selling anything so i wish you good luck! |
| 16:48.00 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 16:48.12 | Wollie | file, the weird thing is that when I call 201 FROM 301 it works good, without Packet2Packet. |
| 16:48.23 | Wollie | this one works: |
| 16:48.24 | Wollie | -- Called 201 |
| 16:48.24 | Wollie | <PROTECTED> |
| 16:48.24 | Wollie | <PROTECTED> |
| 16:48.31 | Wollie | this one doesn't: |
| 16:48.31 | Wollie | <PROTECTED> |
| 16:48.31 | Wollie | <PROTECTED> |
| 16:48.32 | Wollie | <PROTECTED> |
| 16:48.32 | Wollie | <PROTECTED> |
| 16:48.41 | ryduh | wollie: use pastebin.com |
| 16:49.06 | file | in the first case Asterisk would be constructing new RTP packets, in the second one Asterisk would be passing it through |
| 16:49.15 | file | it is possible that one endpoint doesn't like the RTP packets the other side is creating |
| 16:49.38 | ryduh | angryuser: We've got a T1 at the office with 4 lines. Would you have any idea of what kind of load we could handle using 2 lines? |
| 16:50.29 | angryuser | ryduh, two T1 2x24 ? |
| 16:50.43 | Wollie | can it be because the 2 phones are different? |
| 16:50.50 | angryuser | ryduh, or E1 2x30 ? |
| 16:51.06 | Wollie | one is a Linksys and the other one a 'Callmyway Lan Phone 88' |
| 16:51.10 | ryduh | angryuser: I'm sorry I don't know much about T1. We have a T1 line with 4 DIDs |
| 16:51.52 | superbeef | ryduh: US or UK? |
| 16:52.26 | ryduh | superbeef: US |
| 16:52.50 | ryduh | We're also using the T1 for internet |
| 16:52.54 | angryuser | ryduh, dids is one thing, but how many lines ? one T1(usa) is 24 max europe is E1 its 30, also of course you have have one T1 but less lines (look at you telco contract) |
| 16:53.03 | superbeef | ryduh: oh.. it's a fractional |
| 16:53.20 | torrancew | do VM extensions need to be numeric? |
| 16:53.35 | moos3 | angryuser: the peername and secert do I have to set that in sip? the instructions are a little on the bad side |
| 16:53.59 | superbeef | ryduh: you need to ask your provider how it divides it to determine the amount of voice channels.. some providers will add and remove channels for data as voice calls increase |
| 16:54.25 | angryuser | moos3, you have config for iaxmodem /etc/iaxmodem make them mach to iax.conf |
| 16:55.16 | moos3 | ok cool thanks |
| 16:55.17 | torrancew | or can they be alphabetic strings? |
| 16:55.38 | angryuser | moos3, here is the tuto for trixbox but you can pull what you need: http://www.trixbox.org/forums/trixbox-forums/share-your-trixbox-success-stories/trixbox-2-3-0-3-postfix-iaxmodem-hylafax-an |
| 17:00.23 | moos3 | angryuser: cool thx, I have the iaxmodem working I believe when I run the iaxmodem command no errors |
| 17:00.51 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
| 17:01.06 | *** join/#asterisk garymc (i=garymc@host86-164-37-163.range86-164.btcentralplus.com) |
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| 17:02.04 | KavanS | can anyone tell me how long a "ring" takes? |
| 17:02.20 | KavanS | so I can time my rings/dial time correctly? |
| 17:02.24 | KavanS | googles |
| 17:02.33 | superbeef | 6 seconds i think... 3 on 3 off |
| 17:02.43 | superbeef | what's google say |
| 17:02.54 | angryuser | KavanS, depends on country, look at indications.conf |
| 17:02.59 | KavanS | google says little...redefining my search terms |
| 17:09.12 | *** join/#asterisk _cgc (n=_cgc@tequila.lemon-computing.com) |
| 17:09.20 | _cgc | hi everyone |
| 17:09.23 | moos3 | I can't figure out why my registration is failing |
| 17:09.39 | moos3 | I have added them to iax.conf |
| 17:09.46 | moos3 | and reloaded asterisk |
| 17:09.50 | _cgc | does anyone have any experience with isdn2e lines? |
| 17:13.52 | Chainsaw | _cgc: You mean european ISDN BRI lines? |
| 17:14.08 | _cgc | yes, is it possible to set the outbound callerid based on call groups on a isdn2e line using asterisk 1.4 |
| 17:14.24 | Chainsaw | _cgc: It is dependent on your provider letting you. |
| 17:14.28 | _cgc | and using zaptel |
| 17:14.59 | Chainsaw | _cgc: (Most will filter and drop anything that isn't in your assigned number range) |
| 17:15.44 | _cgc | i can do it with the isdn30 lines by setting the callerid in extensions.conf but i am using dahdi for these lines, im trying to set the callerid to numbers that have already been assigned to me |
| 17:16.00 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 17:16.23 | _cgc | for some reason with my isdn2e lines, the callerid always goes to the main number |
| 17:16.27 | Chainsaw | _cgc: DAHDI & Zaptel are fairly similar still, when it comes to ISDN configuration. |
| 17:16.41 | Chainsaw | _cgc: Have you checked with your telco? They may just be filtering it. |
| 17:17.27 | _cgc | so i should be able to set the callerid the same way as on a isdn30 line, if it doesn't work then its probably due to the telco? |
| 17:18.02 | angryuser | _cgc, are you sure that you are setting it ? try to set ${Callerid(num)} manually to see |
| 17:18.18 | tzafrir | _cgc, you can set both the group and the caller ID in the dialplan |
| 17:18.48 | _cgc | ok, ill have a closer look at my extension.conf, thanks every1 :) |
| 17:18.59 | _trine | could someone shed some light on why I have an outgoing ring tone on some telephone numbers and not others? |
| 17:19.38 | angryuser | _trine, using sip ? |
| 17:19.50 | _trine | yes |
| 17:20.01 | _trine | it always the same numbers |
| 17:20.07 | _trine | my local ones |
| 17:20.13 | angryuser | _trine, look a sip trace, do you have 180 Ringing send to you every time ? |
| 17:20.18 | angryuser | sent* |
| 17:20.27 | *** join/#asterisk SLCarrijo (n=scarrijo@interuna.novamerica.com.br) |
| 17:20.33 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 17:20.54 | _trine | angryuser: I don't have an 'r' in there no |
| 17:21.36 | _trine | is that what you meant |
| 17:21.57 | acxty | Hi guys, Does someone know a callcenter software that works with asterisk and linux based |
| 17:22.15 | _trine | if i dial 01253 234567 I get a ring tone as the phone is trying to connect |
| 17:22.19 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 17:22.26 | angryuser | acxty, depends on need, what features ? |
| 17:22.31 | _trine | but if i dial a local number like 234567 I don't |
| 17:22.45 | _trine | I'm a new user |
| 17:23.18 | angryuser | _trine, yes we know, try to do a sip debug on the call to see a difference |
| 17:23.33 | acxty | angryuser, What I want to do is to monitor the calls the agents make, how much time, to what number they call, |
| 17:23.43 | moos3 | [2009-10-29 13:22:56] ERROR[28181]: chan_iax2.c:4474 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user fax requirecalltoken=no |
| 17:23.44 | _trine | angryuser: could you give me a clue on the command for that |
| 17:23.44 | acxty | angryuser, If for a sale department |
| 17:23.51 | moos3 | angryuser: ideas on that? |
| 17:24.31 | angryuser | acxty, so it is for outbound ? do you need predictive ? |
| 17:24.39 | acxty | angryuser, Also to hear the conversations of the agents without them to notice |
| 17:24.48 | acxty | angryuser, also for inbound |
| 17:25.26 | ManxPower-work | moos3: did you try adding requiretoken=no to the iax.conf [fax] section? |
| 17:25.58 | angryuser | acxty, do you nead a call generator from leads ? |
| 17:26.13 | acxty | yes |
| 17:26.41 | moos3 | yes I just did and everything seems cool now expect a error about not being dynamic |
| 17:26.50 | angryuser | _trine what is you * version ? |
| 17:26.54 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 17:27.35 | angryuser | moos3, set host=127.0.0.1 |
| 17:28.02 | acxty | angryuser, what software so you recommend? |
| 17:28.51 | angryuser | acxty, there is only one free called vicidal and all others out-of-the-box payed, or you can hire a pro to do a custom install |
| 17:29.14 | angryuser | acxty, how many agents ? |
| 17:29.30 | _trine | angryuser: Asterisk 1.4.25.1 |
| 17:29.57 | _trine | it's running inside my router for personal use only |
| 17:30.19 | angryuser | _trine, then sip debug peer Peername where peername is you provider peer name |
| 17:30.44 | _trine | tnks I'll give that a try |
| 17:30.47 | angryuser | _trine, and use "help" it is explained there |
| 17:30.50 | acxty | angryuser, a small callcenter 5 the most, which one is the free |
| 17:31.33 | ariel_ | <PROTECTED> |
| 17:31.45 | _trine | angryuser: I did that and guess what |
| 17:32.02 | _trine | it then gave the ring tone :S |
| 17:32.44 | angryuser | ariel_, ah yes! they got a call center addon, but i ahve never used predictive, have any good exp with it ? |
| 17:33.38 | _trine | angryuser: thank you |
| 17:33.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:34.59 | moos3 | angryuser: even with setting the host [2009-10-29 13:33:37] NOTICE[28175]: chan_iax2.c:7363 register_verify: Peer 'fax' is not dynamic (from 127.0.0.1) |
| 17:35.06 | angryuser | _trine, welcome, but we did nothing, it eventually it will come back |
| 17:35.17 | angryuser | moos3, hm let me pull out my iax config |
| 17:35.33 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 17:35.44 | ManxPower-work | moos3: if your device registers then host=dynamic. If the host does not register then host=ip.of.de.vice |
| 17:35.45 | _trine | angryuser: would there be a reason for it to return? |
| 17:36.43 | moos3 | here is what mine looks like http://pastie.org/675322 |
| 17:37.22 | ariel_ | angryuser: and acxty, it works, but all around most call centers need to go with some type of custom settings. As I have never seen anything fit everyone |
| 17:38.09 | acxty | ariel_, which one you recommend, doesn't if it is free or not |
| 17:38.22 | ryduh | I'm looking to replace our current Key System with VoIP phones. What phones would you recommend for a small office (10 employees)? |
| 17:38.31 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:38.50 | ariel_ | acxty: I would start with Elastix see that works for you, but all around there is no real good prediticted dialer out there that is free. |
| 17:38.58 | ryduh | Without being too horribly expensive |
| 17:39.18 | ariel_ | ryduh: polycom |
| 17:39.32 | ryduh | Or, which phones should I stay away from? Grandstream? |
| 17:40.20 | ManxPower-work | ~phones |
| 17:40.21 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 17:40.29 | ariel_ | ryduh: in my view if you want it to work then you stay with Polycom |
| 17:40.49 | angryuser | moos3, here it is http://pastie.org/675334 |
| 17:40.49 | ryduh | thanks :) |
| 17:41.12 | bmoraca | ryduh: Polycom 330s aren't really all that expensive anyway...and the 430 is kind of nice, too (but more expensive) |
| 17:41.18 | Anth8708 | Any polycom guys here with Enhanced BLF experience? |
| 17:41.18 | ryduh | And is the best place to buy polycoms at voipsupply? |
| 17:41.23 | moos3 | ManxPower-work: even with dynamic no go |
| 17:41.28 | bmoraca | or you can go way back in the day and just get some dirt cheap 501s |
| 17:42.00 | ManxPower-work | moos3: I don't believe that. The message is clear. |
| 17:42.12 | scalex000 | hi friend, I setup 2 asterisk using IAX with encryption,(1 asterisk behind nat, another public address), in the asterisk that is behind nat I get a notice chan_iax2.c:8382 socket_process: Packet Decrypt Failed! |
| 17:42.13 | angryuser | ariel_, well you can gem most of you need, but if you want more pull out Cdr records and analyse them with third party |
| 17:42.19 | angryuser | get* |
| 17:42.47 | ariel_ | angryuser: there are over 1000 plus ways to do thing with asterisk. |
| 17:42.48 | *** join/#asterisk twanny796 (n=user@85.232.220.146) |
| 17:43.38 | moos3 | there is finally worked |
| 17:43.42 | moos3 | workign |
| 17:43.54 | [TK]D-Fender | bmoraca: IP 450 largely devalidates the 430, and in many cases the 330 as well |
| 17:43.56 | ryduh | Why in the world do the polycom phones not come with power adapters |
| 17:44.20 | [TK]D-Fender | ryduh: Because common corporate deployments run PoE <- |
| 17:44.41 | Chainsaw | Indeed, it's all PoE in our offices. WLAN APs & phones alike. |
| 17:44.43 | [TK]D-Fender | ryduh: Why increase the cost of the majority market unnecessarily? |
| 17:44.49 | bmoraca | ryduh: depends where you buy them...where i buy them, they all come with AC adapters |
| 17:45.12 | [TK]D-Fender | ryduh: And indeed there are SKU's with the PS included |
| 17:46.23 | angryuser | new aastra phone is coming out soon, color touch display and usb host, they dont say for what, maybe camera |
| 17:46.34 | ryduh | Good to know. Is it feasible to run PoE in a small office? Would I just need to get a PoE switch to run it? |
| 17:46.38 | angryuser | i hope that it is camera |
| 17:46.49 | Chainsaw | ryduh: Yes, generally just replacing your switch will suffice. |
| 17:46.49 | scalex000 | hello who know how to register iax again |
| 17:46.51 | ryduh | bmoraca: where do you buy them? |
| 17:48.23 | bmoraca | the majority of my deployments are not PoE...they tend to be 4-5 phones and use a hosted PBX, so they don't end up wanting to spend money on PoE switches, etc. Although, with polycoms and more recent Cisco phones, you can get away with using much less expensive PoE switches |
| 17:48.35 | *** join/#asterisk VooDooNOFX (n=Joe@rrcs-24-43-123-93.west.biz.rr.com) |
| 17:48.35 | moos3 | angryuser: faxsetup is telling me Sorry, the device is currently in use by another program. |
| 17:48.37 | bmoraca | ryduh: i buy them from one of my regular distributors, techdata |
| 17:48.43 | [TK]D-Fender | ryduh: Same place as everything else |
| 17:48.49 | moos3 | when I tell it the iaxmodem ttyIAX0 |
| 17:49.06 | bmoraca | ryduh: they tend to be 15-20% less expensive than the other internet-based wholesalers |
| 17:49.06 | moos3 | is there something I'm doing wrong? |
| 17:49.16 | angryuser | moos3, dont use faxsetup you got a config in iaxmode source fo hylafax |
| 17:49.30 | moos3 | ok cool thx |
| 17:49.35 | torrancew | has anyone configured a Linksys PAP2 Voip adapter with asterisk? |
| 17:49.50 | bmoraca | i'm sure almost everyone has |
| 17:50.22 | torrancew | bmoraca: well put |
| 17:50.35 | torrancew | bmoraca: the config for it is a bit confusing for me |
| 17:50.45 | bmoraca | do you have a specific question or are you just taking a survey? |
| 17:51.17 | torrancew | i do, but it'll have to wait |
| 17:51.21 | torrancew | boss is calling me out for a bit |
| 17:53.27 | *** join/#asterisk Skeeter- (i=Skeeter-@c216.218.2-65.clta.globetrotter.net) |
| 17:53.57 | Skeeter- | i can intercept(pickup) any calls, except those made from the IAX trunk |
| 17:54.04 | *** join/#asterisk cherva (n=cherva@78.128.16.162) |
| 17:57.08 | moos3 | angryuser: the config.ttyIAX file right? |
| 17:57.57 | *** join/#asterisk |Cybex| (n=John@212.178.82.20) |
| 17:58.44 | angryuser | moos3, yes |
| 17:59.35 | *** join/#asterisk jayrod422 (n=chatzill@pool-96-235-30-58.pitbpa.fios.verizon.net) |
| 18:02.04 | scalex000 | I need help |
| 18:03.20 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 18:03.25 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:07.21 | angryuser | scalex000, try to do iax2 reload |
| 18:07.55 | scalex000 | angryuser, not work, not register again |
| 18:08.14 | scalex000 | angryuser, its strange I can call from one side |
| 18:09.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:12.34 | *** join/#asterisk gabec (n=gabec@cerberus.franklinamerican.com) |
| 18:14.33 | gabec | Anyone here tried interop: sipxecs with asterisk as media gateway ? |
| 18:14.58 | *** part/#asterisk jayrod422 (n=chatzill@pool-96-235-30-58.pitbpa.fios.verizon.net) |
| 18:16.22 | [8none1] | names |
| 18:17.30 | ryduh | bmoraca: do you have a personal techdata account or do you use your businesses techdata account? I don't currently do much service that would require a distributor but now that I'm getting into * it does sound fun to install local * systems for a fee. |
| 18:21.56 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 18:23.16 | angryuser | scalex000, can you try to debug your iax communication to see why it is not registering ? |
| 18:24.43 | scalex000 | angryuser, yes, but I do not understand I will try to do it, thansk |
| 18:24.54 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-oraobdzoxdbkkekw) |
| 18:31.37 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:32.25 | *** join/#asterisk moy (n=moy@74.12.131.148) |
| 18:32.50 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 18:37.46 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 18:40.27 | moos3 | angryuser: how can I test hylafax |
| 18:40.46 | angryuser | moos3, download a hylafax native client |
| 18:41.16 | moos3 | ok cool |
| 18:41.26 | moos3 | I want to test it coming into itself |
| 18:41.28 | angryuser | moos3, add a user and send |
| 18:41.40 | moos3 | ok |
| 18:46.51 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:48.20 | angryuser | moos3, you dont need a client for incoming |
| 18:48.40 | angryuser | moos3, and dont forget to rung a getty on you modem |
| 18:50.55 | moos3 | angryuser: ok |
| 18:51.17 | angryuser | moos3, run faxstat to be sure of the state |
| 18:52.17 | moos3 | angryuser: ok its running |
| 18:53.01 | moos3 | it just rings and rings |
| 18:53.28 | moos3 | exten => 7032,1,Dial(IAX2/fax) |
| 18:53.31 | moos3 | is in my extensions |
| 18:55.37 | angryuser | faxstat says what ? |
| 18:56.54 | angryuser | moos3, have you pointed it to a good tty under /dev ? |
| 18:57.01 | moos3 | Modem ttyIAX0 (+1.703.373.7032): Waiting for modem to come ready |
| 18:57.03 | moos3 | yeah |
| 18:57.25 | *** join/#asterisk coppice (n=chatzill@host86-139-183-205.range86-139.btcentralplus.com) |
| 18:57.33 | angryuser | moos3, so it is not Ready ? |
| 18:58.24 | moos3 | iaxmodem is running |
| 18:58.58 | angryuser | so getty pointing to what ? |
| 18:59.11 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 18:59.38 | moos3 | angryuser: in the config? |
| 19:00.06 | moos3 | GettyArgs: "-h %l dx_%s" |
| 19:00.10 | moos3 | you mean that? |
| 19:00.27 | angryuser | moos3, no, look at my getty in inittab : 7:2345:respawn:/usr/sbin/faxgetty ttyIAX |
| 19:00.52 | moos3 | ttyIAX0 |
| 19:01.04 | moos3 | Modem ttyIAX0 (+1.703.373.7032): Initializing server |
| 19:01.08 | angryuser | moos3, so it is running ? |
| 19:01.13 | moos3 | yeah |
| 19:01.22 | moos3 | and now back to waiting |
| 19:01.29 | angryuser | ok nice |
| 19:01.36 | angryuser | moos3, this is what we want |
| 19:01.36 | moos3 | Modem ttyIAX0 (+1.703.373.7032): Waiting for modem to come ready |
| 19:01.43 | angryuser | crap |
| 19:01.53 | angryuser | no this xD |
| 19:02.07 | moos3 | ok I'm getting fax tones |
| 19:02.13 | moos3 | when i call it |
| 19:02.18 | angryuser | check if you /dev/ttyIAX0 exist |
| 19:02.57 | moos3 | [root@pbx1:/var/spool/hylafax/etc] stat /dev/ttyIAX0 |
| 19:02.57 | moos3 | <PROTECTED> |
| 19:02.57 | moos3 | <PROTECTED> |
| 19:02.57 | moos3 | Device: 11h/17dInode: 539088 Links: 1 |
| 19:02.57 | moos3 | Access: (0777/lrwxrwxrwx) Uid: ( 0/ root) Gid: ( 0/ root) |
| 19:02.57 | moos3 | Access: 2009-10-29 15:02:46.855096334 -0400 |
| 19:02.59 | moos3 | Modify: 2009-10-29 14:58:03.635020334 -0400 |
| 19:03.01 | moos3 | Change: 2009-10-29 14:58:03.635020334 -0400 |
| 19:03.03 | moos3 | yup it exists |
| 19:04.01 | moos3 | faxstat is saying Modem ttyIAX0 (+1.703.373.7032): Running and idle |
| 19:04.06 | moos3 | now |
| 19:04.07 | angryuser | hm it should not be Waiting for modem to come ready, but just ready |
| 19:04.17 | angryuser | ok nice this is it |
| 19:04.21 | angryuser | send |
| 19:04.38 | Katty | oh man, i am so stuffed. |
| 19:04.40 | angryuser | go to /var/spool/hylafax/log |
| 19:04.48 | angryuser | tail -f "latest log" |
| 19:05.26 | moos3 | <PROTECTED> |
| 19:05.47 | angryuser | moos3, look at hylafax logs |
| 19:06.07 | moos3 | I only have c0000001 c0000002 seqf in there |
| 19:06.21 | angryuser | c0000002 |
| 19:06.32 | moos3 | thats where I pulled that from |
| 19:06.41 | angryuser | moos3, pastebin all |
| 19:06.45 | moos3 | ok |
| 19:07.31 | moos3 | http://pastie.org/675555 |
| 19:07.47 | Skeeter- | does the hint feature can be used over a IAX trunk |
| 19:09.55 | hesco | I'm working to configure a new DID, its now being appropriately routed to my telephony server, but now its being answered by the wrong context. Relevant excerpts from my configuration are posted at: http://pastebin.com/d6bad3d9f Can anyone here please suggest how I might untangle this issue, please? |
| 19:13.49 | moos3 | angry what I'm I doing wrong |
| 19:13.54 | moos3 | you want my configs? |
| 19:16.48 | angryuser | moos3, got some job wait a sec |
| 19:16.55 | moos3 | np |
| 19:17.00 | moos3 | thanks for all th help |
| 19:18.47 | angryuser | moos3, you are limited codec to ulaw or alaw ? |
| 19:19.19 | angryuser | moos3, not sure which one is used in usa, its ulaw maybe |
| 19:19.44 | angryuser | moos3, check iaxmode config for codec |
| 19:20.34 | moos3 | k |
| 19:20.41 | moos3 | I can use either one |
| 19:22.14 | Kobaz | hmm |
| 19:22.23 | Kobaz | how do i get the wanpipe version that's currently loaded |
| 19:22.33 | Kobaz | modinfo on the module just reports the kernel version |
| 19:22.41 | Nugget | puts that in his wanpipe and smokes it |
| 19:22.54 | moos3 | where is that config in asterisk's iax.conf? |
| 19:24.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:24.34 | Kobaz | <PROTECTED> |
| 19:24.35 | Kobaz | sexy |
| 19:25.10 | Chainsaw | Wait for 4200, 42, 42 :) |
| 19:25.11 | angryuser | mog, in iaxmodem conf |
| 19:25.22 | angryuser | moos3, where you set pass |
| 19:25.24 | mog | tab failure |
| 19:25.35 | angryuser | mog, !Fail xD |
| 19:26.12 | Kobaz | Chainsaw: heh |
| 19:27.15 | *** join/#asterisk mascool (n=mascool@75-145-232-137-Michigan.hfc.comcastbusiness.net) |
| 19:27.21 | angryuser | moos3, i got to go soon so speed up xD |
| 19:27.37 | mascool | does anyone know how to force a firmware upgrade on a aastra 480i ? |
| 19:27.59 | mascool | or even a downgrade ? |
| 19:29.57 | Kobaz | ah |
| 19:30.09 | Kobaz | i ran strings on wanpipe.ko, and it has the version in there |
| 19:30.24 | angryuser | moos3, i am really hungry so i got to go |
| 19:30.28 | moos3 | lol ok |
| 19:30.31 | moos3 | thanks for the help |
| 19:30.38 | *** join/#asterisk Micc (n=Micc@c-71-231-123-28.hsd1.wa.comcast.net) |
| 19:30.41 | moos3 | i have that set to ulaw |
| 19:30.48 | angryuser | moos3, test then |
| 19:30.50 | moos3 | but I can change it to anything you want |
| 19:31.05 | angryuser | moos3, hm normally it should work with alaw |
| 19:31.10 | moos3 | request id is 5 (group id 5) for host localhost (1 file) |
| 19:31.15 | twanny796 | .. |
| 19:31.23 | moos3 | when i use sendfax to test |
| 19:31.42 | angryuser | moos3, try a normal fax not sendfax |
| 19:31.42 | twanny796 | how do I get help on functions in the console? |
| 19:31.53 | angryuser | twanny796, "Help" |
| 19:32.15 | twanny796 | angryuser: say on echo() |
| 19:33.05 | angryuser | twanny796, its not a joke! tupe help and look at core show funtion(s) part |
| 19:33.10 | angryuser | type* |
| 19:33.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:34.24 | twanny796 | how do I get help on a function? |
| 19:34.28 | angryuser | twanny796, you mean linux console ? |
| 19:34.41 | moos3 | k |
| 19:34.43 | twanny796 | in asterisk CLI |
| 19:35.02 | angryuser | by all see you later |
| 19:35.09 | moos3 | later |
| 19:38.42 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:38.59 | Micc | How do I fix a polycom to accept numbers that start with 00? It seems the default dial plan for the polycom doesn't allow anything more than 00 to be pressed. |
| 19:41.38 | Kobaz | Micc: edit sip.cfg, change the digitmap |
| 19:42.20 | Micc | Kobaz, right, but whats a good generic digitmap? Or how would I allow everything? |
| 19:42.36 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 19:43.24 | TSM | Micc: setup somthing like .T i think |
| 19:44.28 | TSM | this is a dial plan i have for UK stuff, <digitmap dialplan.digitmap="112|999|1234|7x|[2-6]xxT|0[0-9].T|020[7-9]xxxxxxx|07xxxxxxxxx|*[1-9]x|**2xxx|*0x.T" dialplan.digitmap.timeOut="1|1|1|1|3|3|1|1|1|1|1"/> |
| 19:44.29 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 19:49.11 | Kobaz | Micc x.T |
| 19:49.57 | torrancew | ok, can anyone help me figure out the configuration for a Linksys PAP2 with asterisk? |
| 19:50.43 | TSM | torrancew: xml config? or web config? |
| 19:56.27 | [TK]D-Fender | Micc: "x.T|*x.T|#x.T" impossiblematchhandling="2" |
| 19:57.50 | TSM | what is the impossiblematchhandling? when x.T would match anything anyway |
| 19:58.18 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:59.50 | *** join/#asterisk kannan (i=kannan@121.245.36.54) |
| 20:02.09 | kannan | hello all. I want to connect 2 E-1 PRI cards (each card is a single span digium) and connect 1 to PSTN and the other with a E-1 cable to an existing PBX. so one will be PRI CPE and the other PRI NET, i presume. whether it is possible to configure in dahdi and zapata.conf lie this in the same server , or does it require 2 separate asterisk boxes? |
| 20:02.47 | *** join/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 20:03.13 | torrancew | sorry, TSM web config |
| 20:03.52 | moos3 | anyone using hylafax on centos 5 |
| 20:04.08 | moos3 | angryuser: you back? |
| 20:05.36 | [TK]D-Fender | kannan: No issue |
| 20:06.31 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 20:07.03 | kannan | [TK}D-Fender , thanks. |
| 20:07.28 | kannan | [TK]D-Fender , i mean, ty |
| 20:08.52 | kannan | does the E1 cable for the Net side card have to be purchased from Digium ( or can we use a self crimped cat5 )? |
| 20:09.42 | Kobaz | i don't know about e1, but t1 is just straight through cat5 and rj45 |
| 20:09.54 | moos3 | as long as its a striaght though cable you should be fine |
| 20:10.03 | [TK]D-Fender | kannan: You should have a crossover cable (you can crimp yourself) for * -> PBX, and a straight Cat5 to the SmartJack |
| 20:10.39 | kannan | oh ok, thanks again all |
| 20:12.26 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 20:17.45 | *** join/#asterisk Druken (n=jdumais@70.54.242.169) |
| 20:18.33 | Druken | anyone around that feels like explaining periodic anouncements to me? |
| 20:19.08 | Druken | it looks like you should be able to have more than one and use them as advertisements for your company, but i can't figure it out totally, seems to always only play the one file |
| 20:19.59 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:20.26 | Chainsaw | Druken: Some advice from someone who's had to endure being on hold a lot... |
| 20:20.51 | Chainsaw | Druken: When the music stops, it should be because you have an agent available to connect my call to. Music stopping for an advertisement is *annoying* |
| 20:20.58 | Druken | hehe my customers are never on hold for more than about 30 seconds.. hehe i just want to play |
| 20:22.01 | Druken | chances are they will never actually hear them... but doesn't mean i don't want to try and make it work.. why? because i can :) what better reason... |
| 20:22.02 | p3nguin_ | druken: queues.conf, queue-thankyou = or periodic-announce = |
| 20:22.52 | Druken | p3nguin_: yes, but how do i configure more than one periodic-announce ? it has an option to play them randomly... |
| 20:23.26 | hesco | Thought I'd try this again if I may . . . I'm setting up a new DID, which is being appropriately routed by my provider to my telephony server, but is being answered by the wrong context. Relevant excerpts from my configuration are posted at: http://pastebin.com/d6bad3d9f Can anyone here please suggest how I might untangle this issue, please? |
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| 20:27.49 | Druken | hesco: try talking to your provider, see if they are specifing the default context, or an even simplier way, just put in a goto exten as the did in default to go where you want it |
| 20:31.53 | kyleh | anyone using festival with asterisk? my system just hangs once it hits the festival command in extensions.conf |
| 20:32.04 | kyleh | cant figure out how to get past this |
| 20:32.15 | kyleh | just wondering if anyone has ran into same problem |
| 20:33.41 | Micc | Is there any way to tell asterisk to match exact extensions before dynamic extensions? Like exten 300 vs _NX. ? |
| 20:34.57 | voipmonk | sure there is, you just have to get creative with your dialplan logic :) |
| 20:35.43 | *** part/#asterisk SirFoxey (n=SirFoxey@unaffiliated/sirfoxey) |
| 20:35.48 | Druken | hmm, i thought asterisk did that by default... |
| 20:35.49 | hesco | Druken: I would have thought that this line from iax2.conf would do that for me. 'context=from-diamondcard-16783211145' |
| 20:35.52 | ryduh | kyleh: pastebin your extensions.conf |
| 20:36.19 | Druken | hesco: yes well, logic and theory are sometimes diffrent |
| 20:37.00 | ryduh | Micc: I'm not sure about the matching order inside a context but maybe you could try separating them into multiple contexts and then include the less specific context |
| 20:38.08 | Micc | ryduh, well I want my include parkinglot to match before the other extensions. |
| 20:38.35 | Micc | ryduh, maybe by putting the _NX. in another context and including it after the parking lot it might work. |
| 20:38.37 | hesco | So would I use something like a GoToIf(${some_channel_variable},<my_DID>,my_DIDs_context:priority) ? |
| 20:38.38 | p3nguin_ | ~patternmatching |
| 20:38.59 | hesco | and if so, what would that channel variable be? |
| 20:39.01 | ryduh | Micc: That's what I was trying to explain |
| 20:39.21 | *** part/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 20:39.40 | p3nguin_ | micc: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 20:39.58 | p3nguin_ | ~dialplanpatters |
| 20:40.01 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:40.03 | p3nguin_ | ~dialplanpatterns |
| 20:40.09 | p3nguin_ | shrug |
| 20:40.23 | ryduh | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting |
| 20:40.39 | ryduh | Micc: ^ |
| 20:40.41 | p3nguin_ | Yeah! That's what I wanted. |
| 20:40.58 | p3nguin_ | ~sorting |
| 20:41.06 | p3nguin_ | ~sortorder |
| 20:41.14 | ryduh | ~google ;) |
| 20:41.22 | kyleh | ryduh: http://pastebin.com/d522bc1dc |
| 20:41.31 | hesco | perhaps: ${DNID} ??? |
| 20:42.49 | kyleh | line 523 is where it calls festival |
| 20:42.51 | ryduh | kyleh: pastebin some * debug output when it's hanging |
| 20:43.18 | ryduh | kyleh: are you positive festival is setup correctly? |
| 20:43.44 | p3nguin_ | micc: 300 should match before _NX. matches, anyway, as long as you aren't doing an include for another context. |
| 20:44.00 | kyleh | followed instructions here http://www.voip-info.org/wiki/view/Asterisk+Festival+installation |
| 20:44.05 | kyleh | could have something wrong |
| 20:44.10 | kyleh | but festival is up and running |
| 20:44.33 | kyleh | when asterisk calls festival..i get a (client connected) |
| 20:44.40 | kyleh | on festival server |
| 20:44.56 | kyleh | it connects and then right away disconnects |
| 20:44.57 | ryduh | kyleh: no errors, it just hangs? |
| 20:45.07 | kyleh | no errors in asterisk |
| 20:45.14 | kyleh | just hangs |
| 20:45.23 | kyleh | but i get output from festival server |
| 20:45.28 | kyleh | saying client connected, then disconnected |
| 20:45.35 | p3nguin_ | kyleh: I pulled this off my production box: http://pastebin.ca/1648356 |
| 20:45.42 | kyleh | thanks |
| 20:45.59 | kaldemar | hesco: make a peer that actually matches the incoming call |
| 20:46.17 | p3nguin_ | kyleh: I have a feeling your problem isn't with * but is with festival. |
| 20:46.27 | kyleh | i feel it is too |
| 20:46.36 | kyleh | its running on ubuntu |
| 20:46.45 | kyleh | so i just apt-get installed festival that way |
| 20:46.56 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 20:47.00 | kyleh | festival: Festival Speech Synthesis System: 1.96:beta July 2004 |
| 20:47.04 | kyleh | thats my version |
| 20:47.09 | p3nguin_ | Did you start the server? |
| 20:47.13 | kyleh | ya |
| 20:47.15 | p3nguin_ | Did you configure it before you started it? |
| 20:47.31 | kyleh | the .scm file? |
| 20:47.44 | p3nguin_ | You'll probably want to make some changes before starting it. |
| 20:47.50 | Peaceful | Are you supposed to explicitly return an exit code from a macro? I keep getting this error at the end of my "dialout" macro after the call hangs up (normally): |
| 20:47.51 | Peaceful | <PROTECTED> |
| 20:48.09 | kyleh | changes to the festival.scm file you mean? |
| 20:48.16 | kyleh | that is the only thing i have touched on the festival side |
| 20:48.24 | kaldemar | Peaceful: that's not an error. it's just verbosity. |
| 20:48.29 | p3nguin_ | THe only thing I set in /etc/festival.scm is the voice_default. |
| 20:48.52 | Peaceful | kaldemar: Oh, good point. |
| 20:48.54 | kyleh | ic |
| 20:49.09 | Peaceful | kaldemar: Yep, setting verbosity to 0 stops it. |
| 20:49.12 | Peaceful | slaps forehead |
| 20:49.15 | p3nguin_ | kyleh: But in the user file of the user who runs festival server, I have some other things configured. |
| 20:49.34 | ecrane | Anyone know if IAXModem can be used to make modem/dialup calls? |
| 20:50.02 | Peaceful | ecrane: iaxmodem just emulates a hardware modem |
| 20:50.07 | p3nguin_ | kyleh: http://pastebin.ca/1648361 |
| 20:50.16 | kyleh | thanks p3nguin_ |
| 20:50.17 | Peaceful | ecrane: so you can use any software that will work with a software modem |
| 20:50.31 | Peaceful | *hardware modem |
| 20:50.34 | p3nguin_ | kyleh: Without telling festival to use alsa, mine doesn't seem to work. |
| 20:51.00 | p3nguin_ | kyleh: You might be able to stick all that into the main scm file and it'll be fine. |
| 20:51.14 | p3nguin_ | kyleh: I think I'll try that right now. |
| 20:52.43 | kyleh | ill try it too. i dont have a signal on my phone right now tho so i cant test mine right now |
| 20:52.55 | ecrane | Peaceful: Thanks, but do you happen to know if it can be used to connect to another modem (E.g. dialup to my local ISP)? Or does it just support the fax protocols.... |
| 20:53.19 | p3nguin_ | kyleh: If I run it as root, it will not work. If I run /usr/bin/festival --server as a regular user, it works fine. |
| 20:53.28 | kyleh | oh |
| 20:53.33 | Peaceful | ecrane: Don't know. |
| 20:53.44 | kyleh | that could be problem there |
| 20:53.46 | p3nguin_ | kyleh: I'm going to see if I can figure out why that happens. |
| 20:54.03 | ecrane | Peaceful: ok, thanks. I guess I'll have to read the documentation ^^. |
| 20:54.20 | p3nguin_ | kyleh: As root, it says: {FND} Feature Token_Method not defined |
| 20:56.46 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:59.49 | Druken | [TK]D-Fender: can you specify multiple periodic-announce's? |
| 21:00.30 | [TK]D-Fender | Druken: not in terms of sequence as in this @ 30, that @ 120 |
| 21:00.55 | p3nguin_ | kyleh: Even using sudo (as root) to run it as the other user, it still doesn't work. I have to run it as the other user or it fails. |
| 21:01.08 | kyleh | k |
| 21:01.12 | kyleh | i was running as sudo |
| 21:01.23 | p3nguin_ | sudo -u root? |
| 21:01.38 | kyleh | no i just spawned a sudo shell for my user name |
| 21:01.39 | Katty | so. i have a serious problem i need help with. |
| 21:01.41 | kyleh | sudo -s |
| 21:01.49 | Katty | i have pasta and chicken, and...a very serious lack of ideas for dinner. |
| 21:01.53 | p3nguin_ | I was root and running it as sudo -u otheruser festival. |
| 21:02.11 | kyleh | and it did or didnt work that way? |
| 21:02.23 | p3nguin_ | kyleh: It did not work that way, even though I think it should have. |
| 21:02.30 | kyleh | k |
| 21:02.42 | p3nguin_ | kyleh: Just run it as your regular user and see what happens. |
| 21:02.48 | kyleh | ya im trying |
| 21:03.07 | kyleh | someone messed with the extensions file and its got other problems now. ugh |
| 21:03.14 | p3nguin_ | Also, make sure that your phone peer has a context that includes recordings. |
| 21:03.30 | kyleh | thanks for the help. hope i can get it figured out now |
| 21:11.10 | kyleh | p3nguin_: when you put that stuff you have in your .festivalrc file into the festival.scm file did it work? |
| 21:11.23 | kyleh | or does it only work as a file in your home dir? |
| 21:11.54 | p3nguin_ | kyleh: It only works as the regular user, for some reason. I didn't have time to try to resolve it. Just run it as the regular user. |
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| 21:12.04 | kyleh | k |
| 21:15.54 | hesco | kaldemar: I have again reviewed the peer intended to catch these incoming calls and route them to the appropriate context. It is labeled as [<new_acct_number>] in my representation of it in the iax.conf excerpt available at: http://pastebin.com/d6bad3d9f I just used cut-and-paste from my provider's website to make sure I had the account number and secret correct. if those are correct, can you say anything about how that fails to match the |
| 21:15.54 | hesco | incoming number? |
| 21:16.52 | kaldemar | no, unless you show a call |
| 21:18.25 | TSM | does anyone use the LDAP directory in the polycom phones? |
| 21:19.21 | Druken | [TK]D-Fender: nope, not looking in sequence, just random files, like a company advertisement |
| 21:19.39 | [TK]D-Fender | Druken: Nope, not happening |
| 21:20.01 | hesco | I could show a call, but it only shows the default context picking up the inbound call which ought to be routed elsewhere., pastebin of that log coming. |
| 21:20.24 | Druken | [TK]D-Fender: technically or morally? hehe |
| 21:20.53 | hesco | http://pastebin.com/d386b2481 |
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| 21:21.48 | [TK]D-Fender | Druken: Technically |
| 21:22.03 | Druken | [TK]D-Fender: so what does random-periodic-announce=yes |
| 21:22.03 | Druken | do ? |
| 21:22.14 | kaldemar | hesco: with iax debug enabled... |
| 21:22.15 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 21:23.03 | bmoraca | wow...someone just actually tried to make an unauthorized call off of my sip server...first time :P |
| 21:23.40 | [TK]D-Fender | Druken: non-fixed interval perhaps... I'm not 100% sure, so go ahead and read the sample configs a few dozen times to see if they added something I missed |
| 21:24.11 | Druken | been there, done that.. hehe i'm not gettin it :( |
| 21:24.19 | Druken | i'll let you know if i do eventually find something |
| 21:25.18 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-235-218.telstraclear.net) |
| 21:27.26 | Druken | <PROTECTED> |
| 21:34.09 | Druken | ahh, sweet! |
| 21:34.53 | Druken | [TK]D-Fender: when specifing your periodic-announce, seperate filename with | and use the random and it randomly picks one to play :) |
| 21:37.04 | ecrane | p3nguin_: I don't know enough about the software, but it could be an environment variable that is set in the user's home folder/shell. I need to re-read about sudo; if my suspicion is correct, then running it FROM root as the user would not be enough, you'd have to do something like sudo -i <username> <command>. |
| 21:40.08 | ecrane | wait that is giving me problems too.. dang... |
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| 21:45.37 | p3nguin_ | ecrane: Good call. sudo -u myuser -i /usr/bin/festival --server seems to work. |
| 21:45.48 | kyleh | p3nguin_: i think i found the problem |
| 21:46.00 | kyleh | its with the sound card |
| 21:46.10 | kyleh | or atleast configureing festival to use it |
| 21:46.26 | kyleh | festival> (SayText "hi") |
| 21:46.26 | kyleh | Wave save: can't open output file "/dev/audio" |
| 21:46.35 | p3nguin_ | Oh. |
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| 21:46.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:46.49 | kyleh | i tried what you used |
| 21:47.01 | kyleh | your audio_command and stuff |
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| 21:50.29 | kyleh | still doesnt work. get a bunch of ALSA errors |
| 21:50.52 | kyleh | atleast i found the problem i think |
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| 21:53.20 | bmoraca | has anyone heard of an ISP screwing with RTP packets such that fax tone detection would fail? |
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| 21:54.31 | jblack | bmoraca: I suppose anything that introduced high enough latency could make fax undoable. |
| 21:55.08 | bmoraca | it's the only other possibility i can think of... |
| 21:55.24 | bmoraca | i take the same system and move it to a different ISP and everything works |
| 21:55.55 | bmoraca | the ISP that doesn't work is two bonded T1s...the ISP that does is AT&T Uverse... |
| 21:56.05 | bmoraca | i'd have expected the opposite if anything |
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| 22:04.38 | Get_The_Fish | hello all |
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| 22:14.27 | Get_The_Fish | I have been reading around, and I am unsure of a compelling reason to use a macro in a dialplan. I understand what they do and how to use them, I just dont see why you would use one over a context, etc... can someone point me in the right direction? |
| 22:14.29 | ryduh | did anyone else see that mass exodus? |
| 22:14.29 | Get_The_Fish | yeah |
| 22:14.29 | Get_The_Fish | some funkyness goin on... saw a mass exodus and re-join |
| 22:14.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 22:14.30 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:14.30 | Get_The_Fish | I thought it was just my interwebz |
| 22:14.30 | russellb | it's referred to as a netsplit |
| 22:14.30 | russellb | the joys of IRC. |
| 22:14.30 | thehar | russellb: ! |
| 22:14.30 | Get_The_Fish | ah |
| 22:14.31 | thehar | hellos! |
| 22:14.31 | russellb | zomg thehar !!!!! |
| 22:14.31 | thehar | russellb: !!!!! |
| 22:14.31 | russellb | <3 |
| 22:14.31 | thehar | !!!11oneone! |
| 22:14.31 | thehar | <3 |
| 22:14.31 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 22:14.31 | thehar | goes back to documentation hell |
| 22:14.32 | thehar | just sayin hi! |
| 22:14.33 | lesouvage | just a quesstion: is it possible to have meetme() not running well just because the hardware is to old (lets say a 6 years old full blown server). btw also just saying hi! ;-) |
| 22:15.04 | Get_The_Fish | anyone? Bueller? |
| 22:15.26 | Druken | go fish... |
| 22:15.35 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:15.35 | ryduh | Get_The_Fish: Are you a programmer? |
| 22:15.42 | Get_The_Fish | negative |
| 22:15.49 | ryduh | go fish and get drunk |
| 22:15.55 | lesouvage | The problem is that the sound seems to run 4 times slower then it should while running the newest version of Asterisk 1.4 and the fitting dahdi version. |
| 22:16.15 | lesouvage | dahdi_dummy |
| 22:16.36 | Get_The_Fish | damn man, just a question... and I have a working dialplan, I was just askin |
| 22:16.41 | ryduh | Get_The_Fish: it would be clear if you were a programmer. They are similar to functions. They allow you to abstract something you do many times into one line. Cleans up your dialplan |
| 22:16.52 | ryduh | functions or subroutines* |
| 22:17.30 | Druken | i had my dialplan dynamic, but i never used macros... |
| 22:17.40 | lesouvage | Get_The_Fish: I just missed your question |
| 22:17.41 | Get_The_Fish | I understand that. But whats the difference between doing that and creating a context and routing into and out of that context when you need that function is what I am asking |
| 22:17.41 | ryduh | Get_The_Fish: have you looked at the macro examples? |
| 22:17.50 | Druken | i'm with fishy there, i don't see anything special about a macro that i can't do with a normal dialplan |
| 22:18.55 | ryduh | Get_The_Fish: the fact that you don't need to route out if you use gosub |
| 22:18.56 | Get_The_Fish | yes |
| 22:18.57 | Get_The_Fish | I have a couple of them working in fact. I understand what they do, I was just wondering what the real advantage of a macro is versus just using a context |
| 22:19.02 | lesouvage | Get_The_Fish: And I'm in the answering mood |
| 22:19.02 | ryduh | Get_The_Fish: you don't have to keep track of which priority you need to route back to which is very nice |
| 22:19.35 | Get_The_Fish | ah, ok, so thats an advantage there... that's the kind of thing that I was looking for, thanks ryduh |
| 22:20.45 | Get_The_Fish | I need to look some more, because I dont really use them, but dont they do some funky things to CDRs? |
| 22:20.57 | lesouvage | Get_The_Fish: you can write generetic code that can be started from all different places in the dial plan. Little snaps of code that just do a specific thing. |
| 22:21.15 | ryduh | Get_The_Fish: look at Gosub instead of Macros. Macros are deprecated in the latest version of * |
| 22:21.41 | lesouvage | ryduh: you can't be serious about that. |
| 22:21.45 | Get_The_Fish | yes, they are |
| 22:21.51 | russellb | it's not going away |
| 22:21.54 | russellb | GoSub is just more efficient |
| 22:21.59 | russellb | implementation wise |
| 22:22.22 | Get_The_Fish | good to know. |
| 22:22.28 | ryduh | Get_The_Fish: Here's an example of how I use GoSub: http://pastebin.com/d2c7c3c07 |
| 22:22.39 | lesouvage | russellb: so now there is a gosub and a return that do the tric? |
| 22:22.50 | russellb | nods |
| 22:24.41 | Get_The_Fish | ok, interesting... I think that I got somewhat confused with the s extension and what it does to CDR's. CDR's are like one of the most important things to us, so I go way out of my way to keep them clean. |
| 22:25.04 | Get_The_Fish | I was thinking that a macro would do the same thing to a CDR that the "s" extension would do, is this not correct? |
| 22:25.35 | ryduh | Get_The_Fish: I'm not sure I don't mess with our CDRs much |
| 22:25.45 | Get_The_Fish | ah ok |
| 22:25.55 | lesouvage | russellb: what do you mean by "nods" |
| 22:26.03 | russellb | "yes" |
| 22:26.06 | Get_The_Fish | unfortunately I have to... makes life a little interesting |
| 22:26.17 | ryduh | Get_The_Fish: what do you do? |
| 22:26.24 | Qwell | russellb: "air quotes" |
| 22:26.25 | Get_The_Fish | we are a call center |
| 22:26.44 | Get_The_Fish | so we mine the CDR's for patterns, productivity stats, etc etc |
| 22:27.19 | ryduh | Get_The_Fish: are you involved with the mining algorithms? |
| 22:27.35 | Get_The_Fish | to a point, yes |
| 22:27.55 | ryduh | Get_The_Fish: What do you use to find patterns? |
| 22:28.05 | ryduh | Get_The_Fish: I just took a data mining class last year |
| 22:28.16 | Get_The_Fish | yeah it's not that complicated really |
| 22:28.27 | Get_The_Fish | or sophisticated, I should say :) |
| 22:28.41 | *** join/#asterisk jasonpr (n=jasonpr@wsip-98-188-201-109.om.om.cox.net) |
| 22:29.14 | Get_The_Fish | generally it's something like the biz guys saying "lets look at the relationship between these 4 metrics", some SQL magic happens and reports are spit out. |
| 22:29.23 | Get_The_Fish | then they go to lunch. |
| 22:29.33 | Get_The_Fish | next week, same thing :) |
| 22:29.44 | ryduh | Get_The_Fish: ah lol |
| 22:29.52 | jasonpr | any know why asterisk dies when I get over 80 sip-sip calls? (1.6.1.6 core 2 duo 8gb ram) |
| 22:30.10 | jasonpr | i'm using SIPP to send the calls |
| 22:30.21 | Get_The_Fish | what do the asterisk logs say? |
| 22:30.31 | jasonpr | console? |
| 22:31.01 | Get_The_Fish | try the debug log(s) first |
| 22:31.03 | Katty | http://42ndrecipestreet.blogspot.com/2009/10/clean-out-fridge-pasta.html <- dinner. |
| 22:31.10 | jasonpr | /var/log/asterisk/messages and the console doesn't say anything that stands out |
| 22:31.23 | jasonpr | debug logs? |
| 22:32.15 | lesouvage | jasonpr: /var/log/asterisk/messages |
| 22:32.26 | lesouvage | sorry |
| 22:32.32 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
| 22:33.00 | jasonpr | yeah there's nothing in the messages file. Just some stuff about odbc which I'm bypassing for the calls |
| 22:35.25 | lesouvage | Katty: that doesn't look to bad. It depends of your place on the planet if it is time to eat it. |
| 22:36.04 | ecrane | Get_The_Fish: I'm familiar with that biz guys thing where they look at relationships between metrics. They are using the principle that 'correlation is causation', right? |
| 22:36.31 | ecrane | oh dang; he left ;< |
| 22:37.17 | ryduh | i think ecrane got sad |
| 22:38.25 | lesouvage | Katty: this was my live music of this evening http://www.ustream.tv/recorded/2448695 |
| 22:44.18 | lesouvage | If you want the image go to http://www.ustream.tv/lesouvage and click the recording. |
| 22:50.04 | bmoraca | rock and roll band, everybody's waving... |
| 22:50.07 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:52.12 | lesouvage | bmoraca: just a little bar with 20 guests and a small band playing like there life is at stake. We should sip enable them so they can invite the whole world ack ack ack |
| 23:01.42 | ryduh | I'll paste this again: bmoraca: do you have a personal techdata account or do you use your businesses techdata account? I don't currently do much service that would require a distributor but now that I'm getting into * it does sound fun to install local * systems for a fee. |
| 23:04.10 | *** join/#asterisk Get_The_Fish (n=IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 23:04.30 | Get_The_Fish | sorry, my interwebz decided to take a long lunch |
| 23:05.19 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 23:07.06 | Get_The_Fish | so, does anyone know anything about the alternateext field in users.conf? It doesnt appear to be documented anywhere that I can find. |
| 23:09.11 | bmoraca | ryduh: no, the company i work for uses techdata for probably 80% of all our supplies...cisco, HP server/workstations, polycom phones, etc |
| 23:09.58 | ryduh | bmoraca: I checked out there customer form. There's a $100 deposit that's refunded if you do over $1500 in the first 120 days. How often do you buy? |
| 23:10.10 | bmoraca | ryduh: just about every day |
| 23:11.30 | ryduh | can't believes he used there instead of their |
| 23:11.51 | bmoraca | you're just full of typos, aren't you? |
| 23:12.05 | ryduh | oh god |
| 23:12.13 | ryduh | I think it's chocolate time |
| 23:12.46 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 23:13.19 | bmoraca | ryduh: if you're doing small-time installs and such, you're probably fine with something like voiplink.com or voip-supply.com or the like...especially if you're having the customer buy the stuff themselves. if you're into the resale end of things, you'll want better margins and a place like techdata will give it to you |
| 23:13.20 | ryduh | bmoraca: Could you check the price for a IP 450 ? |
| 23:14.34 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 23:14.45 | raden_work | where does asterisk keep voicemail files ? |
| 23:14.50 | ryduh | bmoraca: I could see myself getting into VoIP installations |
| 23:15.51 | bmoraca | raden_work: whatever you set as your astspooldir in asterisk.conf |
| 23:16.03 | raden_work | bmoraca, thanks |
| 23:16.03 | Get_The_Fish | raden_work: typical is /var/spool/asterisk/voicemail/<vm context name>/<mailbox #> |
| 23:16.18 | Get_The_Fish | that is pretty much the default |
| 23:16.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 23:16.54 | bmoraca | ryduh: i can get IP 450s for $175 w/o power supply or $190 with |
| 23:17.12 | ryduh | bmoraca: thanks |
| 23:17.16 | [TK]D-Fender | \o/ |
| 23:17.21 | [TK]D-Fender | upgrade sucess |
| 23:19.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:19.24 | ryduh | [TK]D-Fender: What did you upgrade? |
| 23:19.49 | [TK]D-Fender | ryduh: Ubunto 8.10 -> 9.10 |
| 23:21.25 | ryduh | [TK]D-Fender: How do you like it so far? |
| 23:22.43 | [TK]D-Fender | ryduh: boots a bit faster, screen res is better, switching betwwen apps isn't so jittery. Going to desktop doesn't bug the way it used to, the icons loko a lot better, fonts look a little better (I think...) I gt new FF, OOo, etc.... |
| 23:23.16 | [TK]D-Fender | ryduh: minor PITA not having > SVGA post-install w/o DL-ing the proprietary driver... |
| 23:23.24 | [TK]D-Fender | ryduh: Once in, all is gold |
| 23:23.55 | *** join/#asterisk blackgecko (n=blackgec@189.135.203.40) |
| 23:24.05 | ryduh | [TK]D-Fender: Sounds like a success minus the SVGA setback |
| 23:25.07 | blackgecko | anyone here has deployed an asterisk solutions for up to 400 simultaneous calls ? can you share your experience |
| 23:25.11 | [TK]D-Fender | ryduh: Honestly I figured I'd get hit with that... thing is it fely like guessing to get the right driver a bit. |
| 23:25.23 | bmoraca | just don't ever, ever try to use ubuntu (or any Linux you intend to run xwindows on) with a Radeon HD 2400/2600 video card |
| 23:25.38 | ryduh | [TK]D-Fender: Don't you just love that? |
| 23:25.55 | ryduh | I've switched to Mac and absolutely love it. |
| 23:25.59 | [TK]D-Fender | ryduh: Seriously if that's my main gripe, it still destroys Windows :) |
| 23:26.18 | ryduh | I still run Ubuntu on a server at home and Windows on a laptop though |
| 23:27.04 | ryduh | [TK]D-Fender: very true |
| 23:28.40 | bmoraca | does anyone here have a recommendation for a good, relatively cheap multi-T1 (or T3) media gateway appliance |
| 23:29.25 | Get_The_Fish | so, does anyone know anything about the alternateext field in users.conf? It doesnt appear to be documented anywhere that I can find. |
| 23:29.35 | bmoraca | ~users.conf |
| 23:29.36 | infobot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 23:29.45 | Get_The_Fish | lol |
| 23:29.49 | ryduh | Get_The_Fish: you could look in the asterisk source code |
| 23:29.51 | ryduh | lol |
| 23:29.52 | russellb | infobot: forget users.conf |
| 23:29.52 | infobot | i forgot users.conf, russellb |
| 23:29.54 | Get_The_Fish | ugh |
| 23:30.00 | bmoraca | lol |
| 23:30.27 | Get_The_Fish | I was hoping that alternateext could be used in sip.conf, actually. It would be nice to have that functionality |
| 23:30.45 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 23:31.06 | bmoraca | $17k for an 8 T1 AudioCodes gateway is way steep...i can get an AS5400 for that... |
| 23:31.43 | russellb | infobot: users.conf is an Asterisk configuration file that was primarily created for the AsteriskGUI project. It is intended as a simple configuration interface for users with basic PBX functionality, not as a replacement for other configuration methods. |
| 23:31.44 | infobot | russellb: okay |
| 23:32.06 | bmoraca | toat's way too nice |
| 23:32.27 | bmoraca | not sure where that o came from... |
| 23:32.57 | Get_The_Fish | I hear that... it's the little "alternateext" at the bottom that peaked my interest. I dont use it, but I saw that the other day, and the gears began turning. |
| 23:33.05 | russellb | sure |
| 23:33.14 | russellb | I was just replacing the troll's versions of the description :-) |
| 23:33.25 | Get_The_Fish | Arent most of the configuration options in there also in the applicable configuration file? |
| 23:33.32 | *** join/#asterisk QaDeS_ (n=mklaus@p4FC7298A.dip0.t-ipconnect.de) |
| 23:33.50 | Get_The_Fish | I am going to have to dig through source code arent I. Dammit. |
| 23:34.39 | russellb | it's really the other way around |
| 23:34.41 | [TK]D-Fender | back later... |
| 23:34.43 | russellb | anything in sip.conf is valid in users.conf |
| 23:34.55 | russellb | but there may be some things specific to users.conf not supported elsewhere |
| 23:35.04 | Get_The_Fish | ah. Damn |
| 23:35.05 | Get_The_Fish | ok |
| 23:35.20 | Get_The_Fish | I am guessing alternateext is probably one of them. :) |
| 23:36.02 | russellb | goes to look at what that is |
| 23:36.19 | russellb | Yes, that is specific to users.conf |
| 23:36.36 | blackgecko | whats the biggest asterisk implementation you have done ? |
| 23:36.45 | russellb | The equivalent is just to add extensions in your dialplan, though |
| 23:36.53 | russellb | I'm not sure how that option makes sense for sip.conf |
| 23:37.04 | russellb | blackgecko: about 1 million lines of code? |
| 23:37.06 | russellb | :-p |
| 23:37.18 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 23:37.57 | Get_The_Fish | I was eyeballing it for a hotdesking-esq type feature here |
| 23:38.06 | *** join/#asterisk lewellyn (n=lewellyn@greenviolet/lewellyn) |
| 23:38.40 | raden_work | Why when i play a wav file in the voicemail directory is it blank ? |
| 23:39.24 | russellb | raden_work: what version of asterisk |
| 23:39.36 | raden_work | 1.6 |
| 23:39.41 | russellb | 1.6.what |
| 23:39.46 | manxpower | raden_work: how big is the file? |
| 23:39.57 | russellb | 1.6.what.what |
| 23:40.00 | *** part/#asterisk dD0T (n=dD0T@unaffiliated/dd0t) |
| 23:40.00 | russellb | if it's not recent, update |
| 23:40.17 | russellb | there have been some improvements to help prevent blank voicemails in the last year at some point i think |
| 23:41.53 | blackgecko | @russelb: sorry i dont get it, i meant of terminals or simultaneous calls |
| 23:42.29 | russellb | I know, I was kidding. |
| 23:42.36 | raden_work | mangala, like 100k |
| 23:43.05 | russellb | stories of 10s of thousands of endpoints, and millions of minutes a month are commonplace these days, though |
| 23:43.16 | raden_work | i moved from bosses voice mail to my voice mail directory and i can hear it just cant play it as a wav file |
| 23:43.25 | cusco | hi |
| 23:43.27 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:43.39 | hardwire | raden_work: permz |
| 23:43.40 | russellb | raden_work: then whatever player you're trying to use doesn't support the specific encoding used in the wav ... |
| 23:43.41 | ryduh | raden_work: You're stealing voicemails?! Shame on you |
| 23:44.07 | Get_The_Fish | no, thats just funny |
| 23:44.18 | cusco | I set up asterisk on a test machine, and installed asterisk-gui |
| 23:44.23 | raden_work | ryduh, yeah cause some fuck head customer said I said something and he heard it in a voicemail and yada yada now i Know the truth |
| 23:44.45 | cusco | now when I compilled it I enabled jabber and gtalk, it shows up on asterisk gui and set up an account |
| 23:44.46 | raden_work | And now I know it aint true so he can stop his lil torturess games |
| 23:44.53 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:45.00 | cusco | but my extension does not ring when I get acall on gtalk |
| 23:45.08 | cusco | asterisk cli shows the incomming jabbercall |
| 23:45.48 | blackgecko | @russelb: is it possible to get to 400 simultaneos calls with just one single server ? i was thinking more of a ser + asterisk solution but dont know |
| 23:48.18 | *** join/#asterisk DavidR2008 (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 23:50.27 | ryduh | blackgecko: how much are you willing to pay for that 'one single server' ? |
| 23:51.04 | TSM | is there not the law of deminishing returns with asterisk |
| 23:51.20 | russellb | it is very possible, yes. |
| 23:51.25 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 23:51.49 | TSM | also there are far more vairables, what codecs, SIP incomming or T1 incomming etc... |
| 23:53.34 | *** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net) |
| 23:53.45 | Micc | All of my customers on qwest dsl today are having problems. |
| 23:54.04 | Micc | and it looks like one of qwest's main routers is having some issues with dropped packets. |
| 23:54.15 | Micc | Is it normal though for a major backbone router to have packet loss? |
| 23:54.25 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:54.49 | manxpower | Micc: no. Keep in mind ICMP (ping) is almost always very low priority in the router. |
| 23:54.54 | TSM | not realy, mabey they have a routing problem |
| 23:54.58 | Micc | Here is one of their local routers that is loosing packets. 71.217.184.246 |
| 23:55.42 | Micc | manxpower, thats what I was thinking too, but then it would also point to being busy if its dropping some packets. |
| 23:55.45 | TSM | yet, no lost packets tested from the UK |
| 23:56.01 | manxpower | Micc: *nod* |
| 23:58.42 | Katty | hi. |
| 23:59.28 | Chainsaw | Micc: It's not always the router itself that's at fault. |
| 23:59.36 | Chainsaw | Micc: It might well be attached to a congested link. |
