| 00:00.19 | Snoogan | so do i need to reboot? |
| 00:00.21 | Snoogan | or reload asterisk? |
| 00:00.57 | [TK]D-Fender | Snoogan: You need to completely RECOMPILE * |
| 00:01.11 | ManxPower | Snoogan: it looks like there were no errors on the dahdi install |
| 00:03.57 | Snoogan | [TK]D-Fender: what do you mean completely recompile? Can you tell i am just learning linux :) i have instructions i'm following to install the dahdi andthen dahdi tools, libpri, asterisk, addons then freepbx |
| 00:04.33 | Snoogan | do i need to go back to the beginning? |
| 00:04.55 | ManxPower | Snoogan: you really need to know linux before you can expect to be able to do much with Asterisk. |
| 00:05.29 | Snoogan | well i'm trying |
| 00:06.24 | Snoogan | so far i can make and receive calls :) [TK]D-Fender helped me with a problem registering a trunk. Seems it was my modem giving grief. I've got this far... i'm determined to nut this out and get it right. |
| 00:10.39 | Mango_ | Snoogan: I'm in the same boat. It's fun isn't it? :) |
| 00:10.56 | Mango_ | Just picked me up an Asus WL-520GU the other day. I'm impressed with how well it runs Asterisk. |
| 00:12.01 | ScarEye_ | Mango_: How many calls can the WL-520GU handle? |
| 00:12.16 | [TK]D-Fender | Snoogan: When * is compiled it checks for DAHDI. If it isn't there MeetMe, etc won't be compiled in, and it won't matter if you simply compile & install DAHDI after |
| 00:12.27 | [TK]D-Fender | Snoogan: So you have to recompile & isntall * again |
| 00:13.19 | Mango_ | ScarEye_: I'm not sure yet but the load is 0.01 with one call in progress. I don't do transcoding though. |
| 00:13.39 | ScarEye_ | Mango_: Just using SIP |
| 00:14.16 | Mango_ | ScarEye_: Yes, that's right. Interestingly, if I do other stuff on the router (like SMB) the audio is choppy, even though there's free memory and the CPU load is only at 50%. I haven't figured that one out yet. |
| 00:14.49 | ScarEye_ | Mango_: What are you running on that router? OpenWRT? |
| 00:14.56 | Mango_ | Tomato |
| 00:15.00 | ScarEye_ | ok |
| 00:15.29 | Mango_ | But, if I do browsing and downloading with my computer, through the router, it's fine. |
| 00:15.49 | ScarEye_ | Mango_: Who are you using to terminate your calls? |
| 00:15.50 | Mango_ | That reminds me of another question. Is it possible to reinvite the audio if the phone is behind NAT, if you do port forwarding? |
| 00:16.21 | *** join/#asterisk Iamnach0 (i=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net) |
| 00:16.26 | Mango_ | VoIP.ms at the moment but now that I have Asterisk I'll probably be looking at other providers just because I can :) |
| 00:17.54 | Mango_ | Yourself? |
| 00:19.11 | ScarEye_ | Mango_: I am searching around. I am looking for my business. |
| 00:19.11 | Snoogan | Mango_: yes, i having heaps of fun learning this stuff. Maybe oneday i'll know it well enough to help like the good folk in this channel |
| 00:19.23 | Snoogan | i'm off to recompile and see how it goes. |
| 00:19.34 | ScarEye_ | Mango_: I got like 130 stores that I want to setup astrisk with. |
| 00:19.42 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:19.45 | ScarEye_ | I am researhin right now |
| 00:20.40 | [TK]D-Fender | Mango_: the phone would have to be pretty smart too. |
| 00:21.04 | Mango_ | ScarEye_: What part of the world are you in? |
| 00:21.04 | [TK]D-Fender | Mango_: So don't bet on it |
| 00:21.04 | Mango_ | [TK]D-Fender: Oh? |
| 00:21.04 | ScarEye_ | Mango_ NY |
| 00:21.06 | ScarEye_ | UA |
| 00:21.08 | ScarEye_ | USA |
| 00:21.44 | Mango_ | ScarEye_: You may also want to check out CallCentric. They're a little bit on the expensive side but they are reliable and their support is great. And their servers are in New York. |
| 00:22.12 | Mango_ | VoIP.ms is more flexible though. There are lots of basic PBX functions that you can use without even having Asterisk. |
| 00:23.29 | ScarEye_ | Mango_: I am thinking about broadvoice for unlimited and have a second company as failover |
| 00:25.02 | Mango_ | That's cool. Where are their SIP servers, if you know? |
| 00:26.41 | Mango_ | east coast, it looks like |
| 00:27.06 | ScarEye_ | I think they might be all over |
| 00:27.10 | Mango_ | hmm |
| 00:27.11 | ScarEye_ | but east coast is best |
| 00:27.16 | Mango_ | for you :P |
| 00:27.16 | ScarEye_ | for me at least |
| 00:27.18 | ScarEye_ | yea |
| 00:28.21 | Mango_ | http://www.dslreports.com/gbu if you're interested. |
| 00:35.30 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 00:35.51 | *** join/#asterisk PhunTelTek (n=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 00:40.06 | Snoogan | if i recompile asterisk, do i need to recompile freepbx afterwards? |
| 00:42.36 | [TK]D-Fender | Snoogan: No |
| 00:44.00 | *** part/#asterisk korihor (n=korihor@190.205.243.246) |
| 00:45.13 | Snoogan | thanks [TK]D-Fender :) |
| 00:47.09 | Snoogan | dahdi, dahdi_tools, libpri, asterisk + asterisk_addons have all been "make clean" and recompiled |
| 00:47.37 | Snoogan | when i type dahdi_cfg i still get the error line 0: Unable to open master device '/dev/dahdi/ctl' |
| 00:47.53 | Snoogan | dahdi_monitor |
| 00:47.56 | PhunTelTek | FYI. Broadvoice blames my registration problems on Asterisk. They blacklisted my IP from too many registrations. |
| 00:47.58 | Snoogan | oops wrong window |
| 00:52.30 | [TK]D-Fender | Snoogan: What card(s) do you have? |
| 00:52.58 | [TK]D-Fender | PhunTelTek: \o/ |
| 00:53.19 | [TK]D-Fender | PhunTelTek: When I last saw it it was still bad |
| 00:56.00 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:58.28 | Snoogan | TDM400 with 1 FXO and 1 FXS module |
| 00:58.43 | [TK]D-Fender | Snoogan: modprobe wctdm |
| 00:58.47 | [TK]D-Fender | Snoogan: modprobe zaptel |
| 00:58.54 | [TK]D-Fender | Snoogan: dahdi_cfg -vvvv |
| 00:58.56 | PhunTelTek | saw what? |
| 00:59.08 | [TK]D-Fender | PhunTelTek: Your configs |
| 00:59.15 | ScarEye_ | anyone here know a good how to for setting up asterisk on a server (CentOS) ? |
| 00:59.18 | [TK]D-Fender | PhunTelTek: the register line formatting was all wrong |
| 00:59.29 | [TK]D-Fender | ScarEye_: just enter "centos" on the WIKI |
| 00:59.31 | [TK]D-Fender | ~wikis |
| 00:59.32 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 01:00.56 | *** join/#asterisk LeddyHM (n=NONE@you.cant.hack.thisbox.org) |
| 01:01.01 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:01.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:02.25 | *** join/#asterisk DrZeus (n=bobzilla@201.224.147.76) |
| 01:02.29 | DrZeus | hi all |
| 01:02.41 | ScarEye_ | [TK]D-Fender: What's the wiki url? |
| 01:02.56 | PhunTelTek | 440319xxxx:yyyyyyyyyy@sip.broadvoice.com/440319xxxx This is right isn't it? |
| 01:04.39 | ScarEye_ | sometimes google sucks |
| 01:05.26 | [TK]D-Fender | PhunTelTek: No. |
| 01:05.35 | ScarEye_ | Seriously, what's the difference between 1.4 and 1.6 ? |
| 01:05.40 | [TK]D-Fender | ScarEye_: LOOK UP |
| 01:05.44 | [TK]D-Fender | ScarEye_: 0.2 |
| 01:05.58 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 01:06.38 | Alfio | ScarEye_ 1.6 its more newer than 1.4 |
| 01:06.59 | PhunTelTek | how about this one? |
| 01:07.01 | PhunTelTek | 440319xxxx@sip.broadvoice.com:yyyyyyyyyy:440319xxxx@sip.broadvoice.com |
| 01:07.37 | [TK]D-Fender | ScarEye_: Go read the docs in the 1.6 tarball |
| 01:07.41 | [TK]D-Fender | PhunTelTek: much better |
| 01:08.00 | [TK]D-Fender | PhunTelTek: + /yourdidhere at the end |
| 01:08.18 | ScarEye_ | Alfio: But for production use should I just stick with 1.4? |
| 01:08.33 | [TK]D-Fender | ScarEye_: 1.6.0 is fairly stable |
| 01:08.34 | ScarEye_ | Fender: K I will look up the tarball |
| 01:08.49 | ScarEye_ | Fender: What are you running? |
| 01:08.49 | Alfio | ScarEye_ <[TK]D-Fender> ScarEye_: LOOK UP |
| 01:09.01 | PhunTelTek | they both work. |
| 01:09.04 | ScarEye_ | ya |
| 01:09.04 | Alfio | he is running trixbox |
| 01:09.08 | ScarEye_ | pj |
| 01:09.10 | ScarEye_ | oh |
| 01:09.14 | [TK]D-Fender | Alfio: And? |
| 01:09.32 | Alfio | ScarEye_ he dosent like to talk about but he is running trixbox |
| 01:09.43 | Alfio | :) |
| 01:10.09 | ScarEye_ | hehe |
| 01:12.04 | *** join/#asterisk spiffcow (n=spiffcow@c-71-59-217-253.hsd1.or.comcast.net) |
| 01:15.29 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 01:16.52 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 01:18.38 | *** join/#asterisk Kumbang (n=timeline@rusnas.paume.itb.ac.id) |
| 01:18.40 | PhunTelTek | i tried defaultexpirey maxexpirey. nothing changed the refresh time. |
| 01:22.37 | Snoogan | [TK]D-Fender: FATAL: Module wctdm not found. |
| 01:23.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:25.29 | [TK]D-Fender | Snoogan: reboot |
| 01:25.43 | [TK]D-Fender | PhunTelTek: Is it registering? |
| 01:27.31 | PhunTelTek | it is today. it came back for no apparent reason. |
| 01:32.37 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:33.20 | BenCecka | I'm in the market for SIP trunking service. anybody here have rave reviews or buyer bewares? |
| 01:33.23 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-66942cf15e2319e4) |
| 01:34.24 | Alfio | ~itsplist-us |
| 01:34.25 | infobot | hmm... itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 01:37.34 | Snoogan | [TK]D-Fender: http://pastebin.com/mba34a55 |
| 01:39.19 | [TK]D-Fender | Snoogan: modprobe dahdi |
| 01:39.38 | [TK]D-Fender | Snoogan: Snoogan barrign that, insmod wctdm |
| 01:39.44 | BenCecka | Alfio: thank you. I've tried 2 of these on the list but will check out the others as well. |
| 01:40.05 | Alfio | BenCecka no prob |
| 01:40.21 | Alfio | thx to the infobot |
| 01:41.57 | Snoogan | insmod: can't read 'wctdm': No such file or directory |
| 01:42.18 | Snoogan | modprob dahdi return: FATAL: Module dahdi not found. |
| 01:45.36 | [TK]D-Fender | Snoogan: this is post reboot? |
| 01:47.52 | Snoogan | yes |
| 01:50.04 | Snoogan | not sure if this helps http://pastebin.com/m4a8a30d8 its the contents of the modules file in /etc/dahdi |
| 01:50.26 | Snoogan | i would expect it should have wctdm instead of wctdm24xxp, since its a tdm400p card |
| 01:54.14 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 01:54.20 | [TK]D-Fender | Snoogan: TDM400 or TDM410? |
| 01:54.41 | [TK]D-Fender | Snoogan: the TDM410 (new gen) uses the TDM2400 driver, not wctdm |
| 01:55.15 | [TK]D-Fender | Snoogan: Does yours have the lower port for the EC module? |
| 01:55.25 | [TK]D-Fender | Snoogan: if so its a TDM410, |
| 01:56.08 | Snoogan | just checked. TDM410P |
| 01:56.10 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 01:56.56 | [TK]D-Fender | Snoogan: then you should modprobe wctdm2400 |
| 01:57.57 | Snoogan | FATAL: Module wctdm2400 not found. |
| 01:58.01 | [TK]D-Fender | Snoogan: then you should modprobe wctdm24xxp |
| 01:58.04 | [TK]D-Fender | my bad |
| 01:58.07 | [TK]D-Fender | see above |
| 01:58.19 | Snoogan | FATAL: Module wctdm24xxp not found. |
| 01:58.22 | Snoogan | :( |
| 02:00.09 | [TK]D-Fender | Snoogan: I'd call up Digium support if I were you |
| 02:01.04 | Snoogan | righteo |
| 02:06.37 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:10.03 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
| 02:25.58 | Snoogan | [TK]D-Fender: could this be a problem? |
| 02:25.58 | Snoogan | Under your /lib/modules dir, you may end up with an extra directory put there by Asterisk. |
| 02:25.59 | Snoogan | Create a link from the Asterisk modules dir into the original modules directory (asterisk wont |
| 02:25.59 | Snoogan | find the new modules otherwise). |
| 02:26.10 | Snoogan | this is in the intructions i was following |
| 02:28.20 | Qwell | Snoogan: what? |
| 02:28.33 | Qwell | <PROTECTED> |
| 02:28.42 | [TK]D-Fender | Snoogan: What OS? |
| 02:29.06 | Qwell | Asterisk doesn't care about kernel modules. Whoever wrote that is wrong. |
| 02:29.36 | [TK]D-Fender | Qwell: DAHDI issues getting a TDM410P runnign |
| 02:29.47 | Qwell | install DAHDI |
| 02:31.03 | [TK]D-Fender | Qwell: thats his problem... can't seem to get the kernel modules loading |
| 02:31.18 | Qwell | ls -l /lib/modules/`uname -r`/dahdi/ |
| 02:31.32 | [TK]D-Fender | ^^ far more Linux-savvy than I |
| 02:37.42 | *** join/#asterisk webman (n=adamg@124.246.8.196.static.nexnet.net.au) |
| 02:45.47 | carrar | Qwell is letting out the super secret linux commands!! |
| 02:45.50 | carrar | heh |
| 02:46.21 | carrar | ls owns you!! |
| 02:50.12 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:52.15 | Snoogan | ls -l /lib/modules/`uname -r`/dahdi/ didn't work |
| 02:52.30 | Snoogan | under lib/modules there are two directories |
| 02:53.01 | Snoogan | 2.6.26 and 2.6.26-2-686 |
| 02:53.22 | Snoogan | under 2.6.26 there is dadhi and misc |
| 02:53.40 | carrar | heh |
| 02:53.44 | Snoogan | in the other, there is more stuff |
| 02:54.00 | Snoogan | i'm thinking dahdi is in the wrong place |
| 02:54.30 | webman | does anyone here use asterisk 1.4.x with more than 100 concurrent calls ? I'm getting very high CPU load (>40) |
| 02:55.27 | [TK]D-Fender | webman: And what are those calls doing? |
| 02:55.54 | [TK]D-Fender | webman: Because Digium was making 4-port PRI cards handling 120 channels for almost the past decaude. |
| 02:56.04 | [TK]D-Fender | webman: so 100 "calls" doesn't say much |
| 02:56.21 | webman | ooops, sorry, Zap <-> sip or IAX2 calls with codec conversion to g729 for all voip channels |
| 02:57.44 | webman | yes, this machine has been handling these calls for a long time, this problem only started happening when we upgraded from asterisk 1.2 CVS (very old version) to the current 1.4.26 |
| 02:58.56 | webman | upgraded asterisk, asterisk-addons, codec_g729 and zaptel to the current versions |
| 03:03.47 | [TK]D-Fender | webman: Well the codec conversion certainly is the heaviest part I can see. What about recording? MeetMe? |
| 03:03.58 | [TK]D-Fender | webman: What are the system specs? |
| 03:04.13 | webman | at around 80 active calls, the cpu is around 1.4, but at 90 active calls, the cpu spikes to 25 or more... at over 100 calls, the cpu goes even higher... when the cpu spikes we get dropped audio... |
| 03:04.31 | webman | there is no meetme at all |
| 03:05.20 | webman | dual cpu AMD Opteron(tm) Processor 244 with 2G RAM (no swap used and 1.7G cached) |
| 03:05.56 | webman | also, no monitoring/recording of calls at all |
| 03:06.02 | [TK]D-Fender | webman: I might try varios G729 .so to see if there is an optimization failure in one of them causing a spike |
| 03:06.11 | webman | in fact most modules are "noload" |
| 03:09.30 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 03:10.08 | webman | well, using the codec_g729a-1.4_3.1.3-opteron_sse3_64 I get a illegal instruction on asterisk startup (and crash) so I am back on codec_g729a-1.4_3.1.3-opteron_64 version... I don't see any other version which is likely to perform better? Is there one you suggest I try? |
| 03:11.58 | [TK]D-Fender | webman: I'd start with the basics like (sorry to say) 386, and work your way up |
| 03:13.10 | webman | I think most of them would be in-compatible with my CPU... I could try the pentium m, the 3m and the 4m I suppose those should work... |
| 03:14.20 | [TK]D-Fender | webman: Beyond that I would consider calling up Digium support. Just make sure to have catalog'd the outcome of these |
| 03:15.44 | webman | btw do you recall there used to be a command like show translation table recalc <seconds> which would show how long each translation took for a sample of seconds long ? |
| 03:17.57 | [TK]D-Fender | webman: I don't have any real debuggin skills in this side of things beyond what I've suggested already... |
| 03:19.39 | carrar | webman, something like "core show translation recalc 10" |
| 03:19.53 | webman | well, that didn't go so well... all the pentium codec g729's didn't work at all... "no translation from zap to g729"... |
| 03:20.45 | *** join/#asterisk xa0z (n=Interex@24-171-72-49.dhcp.mtvr.il.charter.com) |
| 03:21.20 | xa0z | Anyone in here use, or know if I can use a voice modem for ptsn communication rather than phone cards? |
| 03:21.20 | webman | yep, for that I get 2ms to translate 1sec of data from anything to g729 |
| 03:22.35 | webman | which I think means I should be able to deal with 500 channels assuming there was no CPU needed for asterisk/overheads... |
| 03:23.24 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:24.04 | Corydon76-dig | xa0z: you certainly can. Which side do you want to talk and which side do you want to listen? Because most voice modems are half-duplex ONLY. |
| 03:24.37 | [TK]D-Fender | xa0z: In case you missed that its pretty much a "NO" |
| 03:25.01 | carrar | All you need is a soldering iron and some sound cards!! |
| 03:25.07 | [TK]D-Fender | ~savemoney |
| 03:25.08 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 03:25.11 | [TK]D-Fender | z0mg!!! |
| 03:25.20 | carrar | heh |
| 03:25.24 | *** part/#asterisk xa0z (n=Interex@24-171-72-49.dhcp.mtvr.il.charter.com) |
| 03:25.29 | carrar | bwhahah |
| 03:25.40 | [TK]D-Fender | RUN FORREST RUN!!! |
| 03:26.25 | Snoogan | i think i have my dahdi issues resolved |
| 03:26.31 | Corydon76-dig | Apparently he didn't like that answer too much |
| 03:26.32 | Snoogan | can anyone suggest a way to check? |
| 03:26.39 | carrar | ahah yeah |
| 03:26.50 | Corydon76-dig | Snoogan: make a call? |
| 03:27.15 | Corydon76-dig | Snoogan: what was it doing when it didn't work? |
| 03:27.41 | Snoogan | couldn't load dahdi modules |
| 03:27.56 | Corydon76-dig | Snoogan: Right, so load them and make a call |
| 03:28.26 | Snoogan | i'm using freepbx, so am i right in thinking that i should make a new zaptel extension? |
| 03:28.42 | Corydon76-dig | ~freepbx |
| 03:28.43 | 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 |
| 03:29.15 | Snoogan | ok |
| 03:29.22 | Snoogan | how do i check my dahdi channels? |
| 03:29.35 | [TK]D-Fender | Snoogan: USE THEM |
| 03:30.01 | webman | Snnogan: try "dahdi show channels" |
| 03:30.54 | Snoogan | ok, thank you. I have lights at the back of the TDM410P card :) will go experiment and see how i go. |
| 03:31.02 | Snoogan | thank you everyone for your help so far |
| 03:38.10 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.35.4) |
| 03:43.15 | *** join/#asterisk spiffcow (n=spiffcow@c-71-59-217-253.hsd1.or.comcast.net) |
| 03:45.06 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 03:48.03 | *** join/#asterisk wwalker (n=wwalker@72.249.1.66) |
| 03:49.40 | wwalker | is there a way for me to make callA (originate or call file), run thru and IVR, then, separately, make callB (originate) do some IVR, then connect the two channels? If so, what command does the bridging (has to be 1.4, not 1.6) |
| 03:56.37 | [TK]D-Fender | wwalker: 2 separate orighinates + MeetMe |
| 03:59.02 | PhunTelTek | Scheduling destruction of SIP dialog '2586ef347942e21c18d6152a2664944c@127.0.0.1' in 32000 ms (Method: REGISTER) <------I can't find how to change the 32000 ms reregister. |
| 03:59.50 | PhunTelTek | broadvoice says to change it, but they don't say how. I might just change THEM. |
| 04:01.04 | wwalker | [TK]D-Fender: I was afraid I'd end up there. not a bad place, just was hoping for a simpler solution. meetme or app_conference better? |
| 04:01.13 | wwalker | I've only used meetme |
| 04:01.39 | [TK]D-Fender | wwalker: Whichever |
| 04:03.21 | [TK]D-Fender | PhunTelTek: defaultexpirey / maxexpirey |
| 04:04.30 | PhunTelTek | I put those in sip_custom.conf it looks like it is ignoring them. |
| 04:04.57 | [TK]D-Fender | PhunTelTek: go make sure they appear under [general] |
| 04:04.57 | PhunTelTek | defaultexpirey=600 |
| 04:04.57 | PhunTelTek | maxexpirey=3600 |
| 04:06.00 | [TK]D-Fender | PhunTelTek: and that isn't 32000ms to re-register,..... |
| 04:06.20 | *** join/#asterisk xa0z (n=Interex@24-171-72-49.dhcp.mtvr.il.charter.com) |
| 04:06.35 | xa0z | does anyone here have the link to the story where the guy called Cisco about the license agreement? |
| 04:06.56 | [TK]D-Fender | ~ciscolicense |
| 04:06.57 | infobot | rumour has it, ciscolicense is unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html |
| 04:07.26 | xa0z | Thank you. |
| 04:08.13 | *** part/#asterisk xa0z (n=Interex@24-171-72-49.dhcp.mtvr.il.charter.com) |
| 04:08.46 | PhunTelTek | what does that 32000 ms represent? |
| 04:09.41 | [TK]D-Fender | PhunTelTek: how long * will sit around waiting for an answer to the register request |
| 04:09.58 | [TK]D-Fender | PhunTelTek: Go look at the actual register attempts |
| 04:10.20 | PhunTelTek | sip.broadvoice.com:5060 440319xxxx 23 Registered Mon, 27 Jul 2009 00:08:51 The refresh shows 23 seconds. |
| 04:11.28 | [TK]D-Fender | PhunTelTek: Go look at the actual register attempts <------------- |
| 04:14.09 | PhunTelTek | I'll shut off the extensions so i don't have to wade through all that other dialog. |
| 04:17.29 | PhunTelTek | it still tries to talk to the extensions after they've been disconnected. |
| 04:18.47 | [TK]D-Fender | PhunTelTek: "it"? |
| 04:20.23 | PhunTelTek | asterisk/trixbox whichever is in control of the show. |
| 04:21.03 | [TK]D-Fender | PhunTelTek: Let me know when you've got something to actually show. |
| 04:21.21 | PhunTelTek | will do, thanx |
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| 04:27.52 | tzafrir_laptop | Something for Alison: http://xkcd.com/615/ |
| 04:30.19 | [TK]D-Fender | LOL |
| 04:30.24 | [TK]D-Fender | Comedy Gold |
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| 04:49.13 | PhunTelTek | http://pastebin.com/db15b2fe this shows the registrations. |
| 04:51.17 | webman | is there some way to find out which part of asterisk is consuming CPU ? I am considering installing the hardware g729 (TC400B) but if it isn't the codec consuming CPU then this won't fix it either ... |
| 04:52.15 | Qwell | webman: re your question earlier.. get the benchg729 util... |
| 04:52.41 | webman | qwell: did that, it told me to use "opteron" version which is the one I am already using ... |
| 04:53.14 | Qwell | [TK]D-Fender: I'm amused somebody actually asked for that link |
| 04:53.38 | Qwell | wonder how many hits it's gotten |
| 04:55.33 | webman | hmmm, the only other major change I suppose is the echo canceller, 1.2 used a different EC compared to 1.4 I think .... for now I reduced the EC from 128 to 64, but it channel usage hasn't gone over 40 for a while.... |
| 04:58.22 | webman | or looking at the product specs, perhaps an upgrade from the TE410p to a TE405p would give "Because the TE405P improves I/O speed by up to 10 times, the result is reduced CPU usage and increased card density per server. " |
| 04:59.56 | webman | nevermind, I think the 405 and 410 are the same, the 405 is 3.3V and the 410 is 5V |
| 05:04.33 | [TK]D-Fender | Correct |
| 05:05.04 | [TK]D-Fender | webman: I would not think this is an EC issue |
| 05:05.56 | PhunTelTek | http://pastebin.com/db15b2fe <-----broadvoice registrations |
| 05:06.30 | [TK]D-Fender | PhunTelTek: Line 14 says * asks for 600, line 30 says BV wants 30. Thats THEIR decision |
| 05:07.08 | PhunTelTek | Then their tech support is full of it. |
| 05:07.53 | [TK]D-Fender | PhunTelTek: Send them the debug. |
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| 05:09.05 | PhunTelTek | so moving defaultexpirey=600 to sip_general_custom worked, but they are ignoring it. So how can they say i register too often? |
| 05:09.38 | *** join/#asterisk tpeers (n=peers@static-97-27.blueline.mg) |
| 05:09.45 | [TK]D-Fender | PhunTelTek: Look in CLI to see how often you do. |
| 05:10.38 | PhunTelTek | i'm watchin gthe debug. it really is every 30 sec |
| 05:12.50 | [TK]D-Fender | PhunTelTek: So pass it on to them. You offer 600, THEY decide 30.. Tell 'em to F-Off |
| 05:14.12 | PhunTelTek | thanx, now i have evidence. |
| 05:23.21 | [TK]D-Fender | Checkout time, later all |
| 05:24.26 | PhunTelTek | g-nite |
| 05:24.26 | PhunTelTek | thanx for your patience |
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| 05:53.33 | darkmadda | quick question i have an outbound route, but i want to play a sound to the line dialing out (not to the outgoing route). How to? |
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| 06:35.50 | webman | if I set call-limit=22 why does "sip show inuse" show the Limit as 99 ? any ideas? |
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| 06:38.29 | darkmadda | in freepbx how do i put a call on hold from an analog line? |
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| 06:43.48 | webman | darkmadda: hookflash should do it I think... |
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| 06:44.11 | webman | darkmadda: or do you mean how to transfer them to park.... |
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| 06:49.21 | webman | which is the preferred/stable version of asterisk currently? 1.6.0.10 or 1.4.26 ? |
| 06:50.53 | kaldemar | both of them. |
| 06:51.20 | kaldemar | call-limit is deprecated anyway, so i wouldn't put too much effort in that. use group functions to limit calls. |
| 06:52.01 | webman | I thought call-limit was still valid in 1.6 but had been fixed/improved ? |
| 06:52.42 | webman | I'm trying to minimise my extensions.conf complexity to try and reduce CPU utilisation, not that it has made a big difference.... |
| 06:53.17 | kaldemar | it works in 1.6 but will be removed in the next version as stated in the sample configuration file and UPGRADE-1.6.txt |
| 06:53.33 | mvanbaak | webman: you wont gain a lot there |
| 06:54.56 | webman | mvanbaak: nope, I didn't :( mainly I moved from using the MYSQL command to using realtime to lookup the values... shortened my extensions a reasonable amount, and reduced the number of connections from ast -> mysql ... other than that no diff |
| 06:56.31 | WindowsUser | anyone know where i can get a toll free DID with cheap incoming from canada? I dont need to accept calls from the usa |
| 06:56.32 | mvanbaak | asterisk dialplan processing is not that hard on the CPU |
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| 07:06.47 | BigM | morning all |
| 07:07.25 | BigM | When somebody calls to a script and the script will make an outbound what happend if one of the two hangup? Will the script terminate? |
| 07:08.48 | kaldemar | what kind of a script? do you mean a dialplan extension? |
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| 07:23.16 | BigM | or php |
| 07:27.50 | kaldemar | depends on what you use and how you originate the outbound call. |
| 07:28.02 | kaldemar | it doesn't have to terminate. |
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| 07:40.58 | WindowsUser | BigM: if the callee hangs up you may have to have your script exit |
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| 07:51.33 | BigM | okay :) |
| 07:52.54 | BigM | another question |
| 07:53.56 | BigM | with dial, do you put somebody DIRECT through or is there another command which will call someone and that you had to connect by hand (like connect(channelNumber) or something like that) |
| 07:54.08 | BigM | just a clean callout with no other actions |
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| 07:57.30 | kaldemar | BigM: app Dial takes a channel as a parameter, among others. |
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| 08:00.58 | BigM | but you had to enter the phoneNumber of the caller or am I wrong? |
| 08:01.24 | BigM | correction.. |
| 08:01.42 | BigM | You had to enter another phoneNumber which yuo wanna call... |
| 08:01.44 | kaldemar | you don't have to enter anything regarding the caller. |
| 08:01.56 | BigM | so |
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| 08:02.31 | BigM | The caller will be on channel 8, I will make another call on 9 (to an employee) let the employee enter a code and if true, change channel? |
| 08:03.50 | eans | How to start the OP Panel? |
| 08:04.14 | BigM | eans, is that a question to me? |
| 08:04.48 | eans | to anyone who may help me :D |
| 08:04.59 | BigM | o// I thought that it was a reaction on my question xd |
| 08:05.07 | eans | lol |
| 08:05.15 | kaldemar | BigM: the caller on channel 8 lands to the context assigned to channel 8 and there is a Dial command that dials channel 9. |
| 08:06.24 | BigM | so youŕe using 2 lines instead of 1 ? |
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| 08:21.54 | BigM | maybe someone can help me out now... |
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| 08:22.41 | BigM | A user calls and will be putted through to an employee. When the employee answered the phone he/she must enter a PIN. can someone help me out with that? |
| 08:27.30 | kaldemar | core show application Dial in asterisk CLI will give you all sorts of options for the Dial command, including U(), which lets you execute an extension for the called channel before connecting the call. that extension could handle the pin with Authenticate (core show application Authenticate). |
| 08:28.49 | BigM | okay ty :) |
| 08:29.28 | BigM | kaldemar, can you give an example? |
| 08:31.35 | kaldemar | read up on dialplans, learn what is a context, an extension and how they work, and you'll get it. |
| 08:31.39 | kaldemar | ~thebook |
| 08:31.39 | infobot | methinks thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
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| 08:32.54 | BigM | mornin TommyBotten |
| 08:33.26 | TommyBotten | Hi BigM |
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| 09:28.07 | superc | did anyone successfully run asterisk with app_fax or hylafax/opal/t38modem to send a fax via a t.38 enabled sip carrier? |
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| 09:36.35 | mvanbaak | fax is obsolete. use email |
| 09:36.48 | superc | thanks mvanbaak |
| 09:38.13 | mvanbaak | :) |
| 09:38.28 | VooDooNOFX | tell that to the banking industry :D |
| 09:38.42 | kaldemar | banking industry is obsolete |
| 09:38.49 | mvanbaak | whehehe |
| 09:38.57 | kaldemar | use squirrel skins |
| 09:39.12 | mvanbaak | we switched from money to cocaine |
| 09:39.40 | kaldemar | a whole new perspective to using money |
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| 09:40.14 | superc | if I read the wikis I get the impression that asterisk with either hylafax or the builtin spandsp/fax must work quite well... so probably its a configuration issue.. but whats wrong then? |
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| 09:49.04 | mvanbaak | superc: without more detail we cannot even guess |
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| 09:50.16 | superc | I reportet it as bug: https://issues.asterisk.org/view.php?id=15578 |
| 09:53.07 | mvanbaak | maybe the problem is QSC ? |
| 09:54.01 | superc | I don't think because its working with atas... qsc is not only a small hut |
| 09:54.34 | DigitalDaz21 | Hi all, I am looking for help using an outbound proxy, can anyone help? |
| 09:56.29 | DigitalDaz21 | I''l post it anyway so anyone joining can look it over... |
| 09:57.02 | DigitalDaz21 | I am trying to get BT Business Broadband Voice working with asterisk and freepbx and have come accross the following problem. |
| 09:57.02 | DigitalDaz21 | The service uses an outbound proxy. In my case the registrar bmnha-01.bt.com is not resoveable by DNS. |
| 09:57.02 | DigitalDaz21 | The outbound proxy is www.bbvservice-560129.bt.com:5060 ... |
| 09:57.26 | DigitalDaz21 | If I set the trunk the way the way I believe it should be configured, it does not work. I immediately get an all circuits are busy and there appears to be no sip activity. |
| 09:57.26 | DigitalDaz21 | If I then replace bmnha-01.bt.com with sipgate.co.uk and reload, I correctly get sip traffic to the proxy but of course it fails. |
| 09:57.26 | DigitalDaz21 | If I now put the original bmnha-01.bt.com back in as the host and reload, everything works perfectly as it should. |
| 09:57.26 | DigitalDaz21 | I should add that I have also tried a random string as the hostname eg hfjgktu.com and again there is no sip traffic, its almost as if I need to reload after entering a publicly resoveable DNS name to "kickstart" it first. None of this survives a restart. |
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| 10:26.13 | *** mode/#asterisk [+o Qwell] by irc.freenode.net |
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| 11:18.23 | BigM | hi there |
| 11:19.02 | BigM | When I use param U in the dial command where do I had to add the function which should executed? |
| 11:22.30 | BigM | TommyBotten, do you have an idea? |
| 11:22.32 | Alfio | BigM what the "U" parameter its suposed to do? |
| 11:22.51 | BigM | <kaldemar> core show application Dial in asterisk CLI will give you all sorts of options for the Dial command, including U(), which lets you execute an extension for the called channel before connecting the call. that extension could handle the pin with Authenticate (core show application Authenticate). |
| 11:23.18 | BigM | Let me explain the situation |
| 11:23.28 | BigM | A customer make a call |
| 11:23.37 | BigM | The system checks if there is a free employee |
| 11:24.17 | BigM | if there is some found, make a call to that employee but before the scripts makes the brigde to eachother the employee must enter a key |
| 11:24.27 | BigM | 4 digits in this situation |
| 11:26.35 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 11:27.08 | Alfio | BigM did you see that option int core show application Dial? |
| 11:27.18 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 11:28.24 | BigM | int core? |
| 11:30.30 | Alfio | BigM did you make core show application Dial? |
| 11:32.11 | yangp | I am wondering has anyone been selected for the use of ENUM numbers ? |
| 11:33.15 | *** join/#asterisk E-bola (n=bola@smtp.techbiz.dk) |
| 11:33.46 | BigM | is that a function? |
| 11:36.37 | E-bola | If i have a need to show my users 2 informations when their phone rings, do anybody know how i can do that? They need to see both the caller ID and the line the call came in from (Every user has 2 direct telephone numbers corresponding to 2 different companies, so they need to answer the phones differently depending on which company the caller called) |
| 11:36.45 | E-bola | They need to use snom 320 or 360 models |
| 11:38.06 | E-bola | By default the snom phones shows it with BLF light's, but the users need to see it in the display |
| 11:40.10 | Alfio | BigM did i wrote core show applications or core show funtions? |
| 11:40.54 | BigM | What I ment, it is an function what you need to execute to see what you need :) |
| 11:41.02 | BigM | is it* |
| 11:41.50 | BigM | but what I see, when you use the dial command you can enter flags... |
| 11:44.13 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 11:45.30 | Dovid | anyone here from the UK ? |
| 11:49.54 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 11:56.54 | TommyBotten | BigM: Sorry... kinda busy today. Could you try to describe what you are trying to achieve? |
| 11:57.09 | BigM | okay no problem :) |
| 11:57.12 | BigM | What I have |
| 11:57.13 | *** join/#asterisk gornati (n=cams@201-87-101-62.static-corp.ajato.com.br) |
| 11:57.36 | BigM | A caller calls to the application and will be put through to an employee |
| 11:58.40 | BigM | this mornin (a few hours ago) I asked how I could fix that the employee had to enter a PIN before the bridge was completed. Kaldemar told me something about U(). But Now Iḿ looking in the dialcommand but I cannot find u() |
| 11:59.18 | BigM | or I find u as Unavailable |
| 12:00.23 | TommyBotten | I've never heard about the U command/function |
| 12:00.51 | TommyBotten | Never the less.. the PIN code.. is that bound to the incoming caller or the destination callee? |
| 12:01.34 | BigM | the employee which should called had to enter a pin before he/she could talk to the customer which calls the number |
| 12:01.51 | TommyBotten | Ah... Now I see |
| 12:02.02 | TommyBotten | hmm... in that case, I guess a local channel is the answer |
| 12:02.09 | TommyBotten | or one answer at least |
| 12:02.17 | *** join/#asterisk webman (n=adamg@200.179.233.220.static.exetel.com.au) |
| 12:02.18 | BigM | how you mean local channel? |
| 12:02.47 | TommyBotten | What you do is that when you have a incoming call, you spawn a local channel that calls up the employee and asks for the PIN code |
| 12:02.56 | webman | with asterisk 1.6.0 and dahdi do you need libpri ? It seems latest version is 1.4 only ?? |
| 12:03.17 | TommyBotten | when the PIN is correct, the local channel is connected to the callers (SIP/whatever) channel. |
| 12:03.29 | TommyBotten | webman: Do you need LibPRI? :p |
| 12:03.49 | TommyBotten | webman: As in.. what features do you need from it? |
| 12:04.13 | BigM | the version is: Asterisk PBX 1.4.18 |
| 12:04.27 | BigM | and we've got some php 5 cli |
| 12:04.49 | TommyBotten | BigM: That should not matter. |
| 12:05.00 | BigM | ok |
| 12:05.26 | TommyBotten | But did you understand my suggestion to you? |
| 12:05.35 | tzafrir_laptop | webman, you need libpri >= 1.4.4 for asterisk 1.6.0 (though just get the latest) |
| 12:06.23 | webman | TommyBotten: AFAIK, you need libpri to use the digium PRI cards with asterisk ... ?? |
| 12:06.36 | webman | tzafrir_laptop: thanks |
| 12:06.39 | TommyBotten | webman: True |
| 12:06.43 | BigM | TommyBotten, lik: dial(ZAP/g1/w,phonenumber,10) and replace g1 by a free channel? |
| 12:07.05 | TommyBotten | exactly |
| 12:07.06 | BigM | I know what you mean but How I can execute it in my script is the second question :) |
| 12:07.09 | BigM | ^^ |
| 12:07.11 | TommyBotten | hehe |
| 12:07.15 | TommyBotten | execute it in your script? |
| 12:07.23 | BigM | is the php guy ) |
| 12:07.33 | TommyBotten | ehhh |
| 12:07.49 | TommyBotten | I have no idea. I use the dialplan for what its meant for |
| 12:07.51 | tzafrir_laptop | BigM, hmm... g1 should give you "a free channel" in group1 |
| 12:08.58 | BigM | ok now iḿ confused.. |
| 12:09.28 | BigM | what if g1 gives a free channel, than I spawn the callee already in a free channel but how can I do the trick than? |
| 12:12.45 | TommyBotten | tzafrir_laptop: What is the easiest way to bridge channels in 1.4.x? |
| 12:14.21 | tzafrir_laptop | BigM, Zap/g1 means "the first channel that belongs to group 1 that is available" |
| 12:15.00 | tzafrir_laptop | BigM, so I don't understand your question |
| 12:15.02 | BigM | yeah I know, but I need some time between the pick up and the bridge |
| 12:15.34 | BigM | when you as employee will answer the the in comming call you must enter your personal PIN |
| 12:16.24 | E-bola | Anybody here uses snom 360's? |
| 12:16.31 | BigM | now if I pick up the phone as employee I don't need to enter a pin, but thatś not what I want |
| 12:16.41 | BigM | I want that the employee had to enter a pin |
| 12:17.01 | BigM | if hey/she fails for 3 timesthan somecode. Bur first the pin :) |
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| 12:20.19 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:30.24 | BigM | but if the caller is in a channel en the employee in another channel How can I fix that the employee had to enter the PIN. My working dial will directly bridge the employee |
| 12:31.13 | BigM | so what is the way to make some time between the pickup and the bridge |
| 12:36.38 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:38.23 | BigM | can I do something with M(x [^arg]) |
| 12:38.44 | BigM | executes the macro |
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| 12:38.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:39.32 | TommyBotten | BigM: Sure, but if the user enters the wrong pin code, the connection will be broken |
| 12:39.37 | TommyBotten | hence, the need for a local channel |
| 12:40.15 | [TK]D-Fender | Unless you ask a few times... |
| 12:41.31 | [TK]D-Fender | And the wording doesn't make much sense. What does "caller in a channel" have to do with "Dial"? what are these calls (are there really 2 already in progress?) really doing? |
| 12:43.11 | BigM | if there is a free employee the system must call him/her |
| 12:43.44 | TommyBotten | So you need a queue? |
| 12:44.15 | TommyBotten | First... let me ask why. Why does an employee need to press the PIN code? |
| 12:44.24 | BigM | no, if there is no free employee there will be played a message like: At this moment there is no free employee |
| 12:44.58 | [TK]D-Fender | BigM: "core show application chanisavail" <- |
| 12:45.21 | [TK]D-Fender | BigM: Again, why does your employee have to enter a pin? |
| 12:45.26 | BigM | That was the first question, but the employee is a listen consultant. That mean, if you have problems or you're suicidal you can call with them and the be sure that they are who they are they all need to enter a pin |
| 12:45.33 | BigM | I was typing xd |
| 12:45.55 | TommyBotten | Ah |
| 12:46.02 | TommyBotten | In that case, I think I have a better solution |
| 12:46.26 | [TK]D-Fender | BigM: M() <- |
| 12:46.27 | TommyBotten | You use the queue features as is ... and the agent must sign in - using the pin code |
| 12:46.34 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 12:46.46 | TommyBotten | Then he/she can answer calls. And afterwards he/she signs out |
| 12:46.48 | BigM | [TK]D-Fender, were can I find that "core show application chanisavail" what the hack is that "core show application"? |
| 12:46.52 | TommyBotten | and will no longer be called |
| 12:47.03 | [TK]D-Fender | TommyBotten: BigM Go type into * CLI and read the INSTRUCTIONS <- |
| 12:47.29 | TommyBotten | [TK]D-Fender: ? |
| 12:47.46 | [TK]D-Fender | TommyBotten: Yes? |
| 12:47.56 | [TK]D-Fender | BigM: Go type into * CLI and read the INSTRUCTIONS <- |
| 12:48.01 | [TK]D-Fender | TommyBotten: Sorry, bad aim. |
| 12:48.18 | TommyBotten | [TK]D-Fender: No worries ;) |
| 12:48.26 | [TK]D-Fender | TommyBotten: Queu is a nifty idea but harder to control the forced exit if they're all on the phone. |
| 12:49.08 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 12:49.11 | [TK]D-Fender | TommyBotten: And while they might have to enter a pin to "log in" I'm sure he wants to auth on every call in case the guy slips away for a cigarette break |
| 12:49.27 | TommyBotten | Hmm.. Sounds fair |
| 12:49.46 | [TK]D-Fender | TommyBotten: Only because the situation sounds that serious. |
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| 12:51.45 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:30a7:9641:2ac0:762d) |
| 12:51.47 | cusco_ | hi |
| 12:52.03 | BigM | TommyBotten, you new solution is not workin' because you must be sure that the callee did answer the phone. What if there is a voicemail or he/she has multiply chats on one phone |
| 12:52.32 | TommyBotten | Multiple calls is easy to detect |
| 12:53.02 | BigM | and when the employee has a voicemail enabled on the mobile phone and starts after 3 rings... |
| 12:53.08 | [TK]D-Fender | BigM: "core show application chanisavail" <- |
| 12:53.23 | BigM | in the terminal? |
| 12:53.35 | [TK]D-Fender | BigM: And are you now expanding this by telling us that the people you are calling are REMOTE on the PSTN? |
| 12:53.38 | [TK]D-Fender | BigM: YES |
| 12:53.53 | *** join/#asterisk jmacz (n=mcorb@201.244.174.55) |
| 12:54.23 | BigM | they are remote. So they can sitting @ home or in the forrest or maybe at the top of a mountain :P |
| 12:54.25 | *** join/#asterisk cusco__ (n=tralala@2001:0:53aa:64c:3427:8713:2ac0:762d) |
| 12:54.28 | cusco__ | hi |
| 12:54.38 | cusco__ | I dunno if you read wha I said, probably not |
| 12:54.54 | cusco__ | boss is running an asterisk server here, he is not present... we were expereincing some noise in the calls... i see loads of warnings in CLI such as: |
| 12:55.00 | cusco__ | i think this is my colleague that is trying to make up a FAX system... |
| 12:55.07 | cusco__ | the file is there, I do not understand the warning |
| 12:55.11 | [TK]D-Fender | BigM: And you call them via the PSTN? |
| 12:55.13 | cusco__ | [Jul 27 13:47:50] WARNING[31253]: pbx_spool.c:384 scan_service: Unable to open /var/spool/asterisk/outgoing/210358656.call: No such file or directory, deleting |
| 12:55.47 | cusco__ | the file seems allright, why do I get that warning? |
| 12:56.03 | BigM | <PROTECTED> |
| 12:56.08 | [TK]D-Fender | cusco_: how si the file getting there? |
| 12:56.21 | cusco__ | being generated by callmenow.php |
| 12:56.58 | [TK]D-Fender | BigM: then you NEED M() and "chanisavail" is of no use to you. use "core show functions like GROUP" o see what you can do to limit calls sent to them |
| 12:57.14 | [TK]D-Fender | BigM: And if they are busy and have VM, you're screwed |
| 12:57.26 | [TK]D-Fender | BigM: Unless you wait for the M() to fail and timeout |
| 12:57.54 | [TK]D-Fender | cusco_ : WOW.... like we know what that script is doing.... GO LOOK |
| 12:59.16 | *** join/#asterisk QaDeS (n=mklaus@p4FC72E72.dip0.t-ipconnect.de) |
| 12:59.26 | TommyBotten | cusco__: In that case it's more of a php/script error, and not really Asterisk |
| 12:59.56 | TommyBotten | [TK]D-Fender: Voicemail detection? ;) |
| 13:00.41 | [TK]D-Fender | TommyBotten: Since he has to sit in M()... not much point... 30s timeout on answer tops is about the best he could expect. And could count no PIN as a "log out" if needed |
| 13:00.53 | TommyBotten | True.. true |
| 13:01.05 | TommyBotten | I'm thinking technology. not solution :D |
| 13:01.13 | BigM | core show functions like GROUP << I did that but what do you wanna know? |
| 13:01.31 | cusco__ | TommyBotten: [TK]D-Fender but the file is there... whi is it erroring it? |
| 13:01.35 | cusco__ | why |
| 13:01.43 | cusco__ | why can't asterisk open it |
| 13:02.07 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:02.32 | *** join/#asterisk pwden (n=domin8@pool-72-80-206-49.nycmny.east.verizon.net) |
| 13:02.35 | cusco__ | http://pastebin.com/mf697028 |
| 13:03.28 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:03.34 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:03.40 | TommyBotten | cusco__: Permissions perhaps? |
| 13:03.48 | [TK]D-Fender | TommyBotten: My next guess |
| 13:04.06 | pwden | anyone have a PBX with IVR or just something I can use to look professional for a businsess that I get 0 calls for currently? |
| 13:04.07 | TommyBotten | Beat you too it ;) |
| 13:05.27 | [TK]D-Fender | TommyBotten: Yeah, I'm uber-multi-tasking right now... |
| 13:05.28 | TommyBotten | pwden: Could you rephrase. I'm not sure I understood you correctly |
| 13:05.43 | TommyBotten | [TK]D-Fender: I can't multi-task to save my life. |
| 13:05.52 | [TK]D-Fender | TommyBotten: He wants someone to host an IVR for him probably pretty much for free |
| 13:06.01 | pwden | I have a web design businsess and I would like to setup a professional Corporate PBX-like system. |
| 13:06.31 | *** join/#asterisk eliyahud (n=eliyahu@77.126.64.188) |
| 13:06.36 | [TK]D-Fender | pwnHave you considered installing and configuring ASTERISK? I hear its AWESOME |
| 13:07.00 | TommyBotten | pwden: I see. This channel is for the asterisk technology, and not for service providers in general. |
| 13:07.17 | pwden | ya just don't have a server or anything 24/7 |
| 13:07.33 | TommyBotten | pwden: If you are looking to do it yourself from scracth, then sure Asterisk is great... but it might be a bit much |
| 13:07.52 | [TK]D-Fender | pwden: Go set one up |
| 13:07.55 | pwden | its a bit too much, I hacked MagicJack to an IVR system but its not reliable |
| 13:08.28 | pwden | I need something small scale and I need it by today / tommorrow so not feasible right now with no server and experience. |
| 13:08.47 | leifmadsen | I suggest finding an ITSP that does hosted PBX's then |
| 13:08.48 | [TK]D-Fender | pwden: My rates are very accessible ;) |
| 13:09.01 | BigM | [TK]D-Fender, I can use m() with function ? |
| 13:09.12 | [TK]D-Fender | BigM: Huh? |
| 13:09.18 | leifmadsen | huh's as well |
| 13:09.24 | BigM | lol |
| 13:09.48 | BigM | when I use m() in the dial command |
| 13:09.55 | pwden | I'm looking to score $3k+ contracts and not spend much time on the call because I pay for toll free time. I am willing to give 5% on every finished contract. |
| 13:09.56 | BigM | what can I enter between () ? |
| 13:10.10 | leifmadsen | BigM: anything |
| 13:10.12 | *** join/#asterisk bsteele (n=bbsteele@lb1.ihets.org) |
| 13:10.13 | [TK]D-Fender | BigM: PARAMEterS |
| 13:10.17 | leifmadsen | yes, you can put a function there |
| 13:10.28 | leifmadsen | you can put functions anywhere variables can go |
| 13:10.46 | BigM | and when the return of that function is true, the bridge will be made or else error? end of call? |
| 13:10.55 | BigM | *shocked* |
| 13:11.01 | [TK]D-Fender | BigM: "core show application dial" <- |
| 13:11.16 | [TK]D-Fender | BigM: You choose how you exit that macro and how you want to handle things. |
| 13:13.22 | BigM | that core show application dial I read before :) |
| 13:13.53 | [TK]D-Fender | BigM: read it again. And again.... |
| 13:13.56 | [TK]D-Fender | ~osmosis |
| 13:13.56 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 13:14.28 | BigM | pwnd :) And itś true, when youŕe reading it again you will know it beter after a few times |
| 13:15.10 | TommyBotten | :D |
| 13:15.12 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:18.07 | coppice | read it often enough, and you'll be able to recite it by heart. that won't necessarily mean any of it sunk in, though :-\ |
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| 13:18.54 | [TK]D-Fender | coppice: Yes, but the inevitable aneurysm STILL beings silence ;) |
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| 13:40.57 | Dovid | anyone from the UK here ? |
| 13:41.31 | coppice | being from the UK is preferable to being in the UK |
| 13:41.36 | pwden | technically I am if you trace my bloodline so how can I help you |
| 13:41.52 | pwden | m8 |
| 13:42.40 | [TK]D-Fender | pwden: You have a blood trail leading back to the UK? |
| 13:42.49 | [TK]D-Fender | pwden: unload res_exsanguination.so |
| 13:42.53 | Dovid | lol. |
| 13:43.02 | Dovid | i need an 09 number in the UK to test something ehre |
| 13:43.08 | pwden | :o |
| 13:43.35 | coppice | since the UK is a island a trail of blood from it sounds a little hard to achieve |
| 13:43.37 | pwden | I just got a spam mail from a company that does that |
| 13:43.55 | pwden | coppice: good point |
| 13:44.26 | pwden | 's family is very thick-blooded |
| 13:44.59 | coppice | warfarin is good for that |
| 13:45.07 | [TK]D-Fender | coppice: Chunnel <- |
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| 13:48.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:50.32 | yangp | I am wondering why do I get declined call by calling a SIP address 82@My-Asterisk-IP -> [Jul 27 15:49:29] NOTICE[17009]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '82' rejected because extension not found. |
| 13:50.47 | eric_hill | [TK]D-Fender: Hopefully this will demonstrate my SIP RPID issue: http://pastebin.com/m45e0adbd |
| 13:51.00 | yangp | I made exten => 82,n,Dial(SIP/82,130,rtk) |
| 13:51.15 | [TK]D-Fender | yangp: And where is your 1 priority? |
| 13:51.27 | yangp | exten => 82,1,Answer() |
| 13:51.28 | [TK]D-Fender | yangp: You can't jsut start with "n" |
| 13:51.39 | [TK]D-Fender | yangp: Then its not in the context the call is landing in. |
| 13:51.52 | yangp | but maybe its in a wrong context...which context is default for SIP to SIP calls ? |
| 13:51.59 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
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| 13:53.02 | [TK]D-Fender | yangp: Whatever you set it to be under [general] |
| 13:54.19 | ariel_ | morning everyone |
| 13:57.36 | TommyBotten | Good morning.. or afternoon |
| 13:59.06 | ruyo | Or night even. :o |
| 14:00.26 | yangp | [TK]D-Fender: under [general] settings in extensions.conf |
| 14:00.45 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:00.47 | yangp | I added this exten lines underneath its still failing |
| 14:01.14 | yangp | calling over PSTN will ring 82 in another context |
| 14:01.25 | [TK]D-Fender | yangp: ... [general] is SIP.CONF <--- the CONTEXT you specify.. |
| 14:01.32 | yangp | agh |
| 14:01.43 | yangp | sorry |
| 14:05.03 | *** join/#asterisk awk_r (n=awk@nat/digium/x-4541873b2870d7bf) |
| 14:06.12 | webman | could someone running asterisk 1.6.0 paste the output of core show function CALLERPRES please.... it doesn't seem to be documented on the voip-info wiki as yet and I want to update my extensions.conf before upgrading |
| 14:06.22 | yangp | I still don't quite understand the ENUM procedure. When adding a PSTN number on to e164.org then I can make all sort of aliases inside their DNS, but in which form should I then redistribute the info ? Like ENUM:+<country><area><number> ? |
| 14:07.12 | eric_hill | webman: http://pastebin.com/m6d6bce96 |
| 14:07.35 | webman | eric_hill: thanks |
| 14:09.01 | yangp | or asking in another way - What ENUM info should I place into my signature ? |
| 14:10.24 | webman | So, does this line look right? exten => 7565,n,${CALLERPRES()=prohib} |
| 14:11.37 | kaldemar | webman: no |
| 14:11.56 | kaldemar | exten => 7565,n,Set(CALLERPRES()=prohib) |
| 14:11.57 | *** join/#asterisk ingenius (n=alektro@host191.190-224-106.telecom.net.ar) |
| 14:12.03 | eric_hill | webman: exten => 7565,n,Set(CALLERPRES=prohib) |
| 14:12.43 | webman | eric_hill: doesn't that set the channel variable?? That doesn't use the function?? or does it ? |
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| 14:13.41 | eric_hill | webman: Test it out. That should work... |
| 14:14.23 | [TK]D-Fender | eric_hill: Looks like the RPID comes in fine at the start, and the Cisco tries to override it after |
| 14:15.02 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 14:15.12 | eric_hill | TK: I assume you mean the "pending" then "Hill Eric" later? Strangely, I don't see "pending" on the phone display either. |
| 14:15.22 | webman | eric/kaldemar ok, will do... thanks.... |
| 14:30.59 | ruyo | I was wondering if anyone could help me with the following problem. |
| 14:31.20 | ruyo | I have a asterisk box connected to a ISDN line and a ISDN pbx. |
| 14:31.28 | leifmadsen | eric_hill: think you need to use the () to tell the parser it is a function |
| 14:31.53 | leifmadsen | although that is probably something to test out and write down |
| 14:32.22 | ruyo | If I make a call from a phone connected to the PBX through Asterisk bridging it to the ISDN line, the callee can't ear me, but I can ear him. |
| 14:32.51 | ruyo | If someone calls me, again, isdn line -> asterisk -> pbx -> phone, all works great. |
| 14:33.10 | ruyo | Can this be some kind of buggy mISDN bridging? |
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| 14:37.15 | t_corr | In GotoIfTime, would 0:00 represent 12:00am? |
| 14:37.27 | Alfio | yes |
| 14:37.56 | Alfio | t_corr yes |
| 14:37.56 | t_corr | Thanks |
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| 14:39.18 | ramindia | [intra]lanman: Hi |
| 14:39.20 | jaytee | sounds like an earing problem |
| 14:39.55 | [intra]lanman | hi ramindia |
| 14:40.06 | *** part/#asterisk xjester (n=ghostshe@c-98-196-97-191.hsd1.tx.comcast.net) |
| 14:40.33 | ramindia | how can user directly get in to his voicemail box, without saying com.....mailbox..... then iam getting you have X number of messages....i try to google but not able to get the right answer |
| 14:41.15 | Alfio | ramindia with s |
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| 14:41.49 | ramindia | Alfio: iam using like this "exten => 999,1,VoicemailMain(${CALLERIDNUM})" where is the S Goes |
| 14:42.08 | Alfio | at last |
| 14:42.11 | Alfio | ,s |
| 14:42.19 | ramindia | (${CALLERIDNUM},s) |
| 14:42.20 | Alfio | after ${} |
| 14:42.25 | ramindia | is this correct |
| 14:42.26 | Alfio | yes |
| 14:42.35 | ramindia | give me second let me try |
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| 14:43.49 | ramindia | Alfio: iam getting same message " Executing [999@default:1] VoiceMailMain("SIP/6930614565-09a55118", "|s") in new stack" |
| 14:43.56 | Alfio | exten => 1234,2,VoiceMail(777@mios,sl) |
| 14:44.01 | Alfio | not in voicemail main |
| 14:44.07 | Alfio | its for the extensions |
| 14:44.24 | Alfio | you want to skip vm-intro |
| 14:44.25 | Alfio | ? |
| 14:44.26 | ramindia | No iam directly sending to voicemailmain |
| 14:45.20 | jaytee | hmmm, looks like Google Voice and Gizmo5 are back with SIP again: http://nerdvittles.com/?p=630 |
| 14:45.28 | ramindia | Alfio: its playing this "<SIP/6930614565-09a55118> Playing 'vm-login' (language 'en')" like com....mailbox.............later it says you have X mesages, i dont want to hear that first comm........mailbox |
| 14:48.00 | ramindia | [intra]lanman: any advice |
| 14:49.11 | [TK]D-Fender | :p |
| 14:49.11 | [TK]D-Fender | TIMING |
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| 14:49.40 | ramindia | [intra]lanman: Timing ? |
| 14:49.45 | [intra]lanman | ramindia: yeah, i have some... but it wouldn't be welcome in this channel ;-) |
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| 14:51.18 | ramindia | [intra]lanman: then which channel ? |
| 14:51.31 | [TK]D-Fender | ramindia: Yeah, if you don't want it to ask for the mailbox you have to SPECIFY the mailbox |
| 14:52.33 | ramindia | [TK]D-Fender: picking up mailbox is not a problem, but before it says to the user you have X mesage, i want to disable commedian mailbox mesage |
| 14:52.54 | [TK]D-Fender | ramindia: What should it do instead? |
| 14:53.09 | ramindia | just play you have X messages |
| 14:53.19 | [TK]D-Fender | ramindia: You sign in to check your VM's... shouldn't it tell you how many new ones you have waiting? |
| 14:53.30 | [TK]D-Fender | ramindia: Then go replace the recording <- |
| 14:53.54 | ramindia | i dont want to play this message "<SIP/6930614565-09a55118> Playing 'vm-login' (language 'en')" |
| 14:55.36 | [TK]D-Fender | ramindia: then REPLACE the recording. |
| 14:55.46 | [TK]D-Fender | ramindia: there is no configuration option to disable it. |
| 14:56.50 | ramindia | in 1.2.x i use "exten => 999,1,VoicemailMain(${CALLERIDNUM})" i dont hear that commedian mailbiox |
| 14:57.00 | ramindia | now iam testing with 1.4.X |
| 14:57.27 | [TK]D-Fender | ramindia: because that variable NO LONGEr EXISTS |
| 14:57.38 | [TK]D-Fender | ramindia: It was deprecated in 1.2 and removed in 1.4 |
| 14:57.46 | [TK]D-Fender | ramindia: ${CALLERID(num)} |
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| 14:59.05 | ramindia | [TK]D-Fender: let me try |
| 15:00.03 | ramindia | [TK]D-Fender: thats did the Tricks..................... thanks... |
| 15:00.32 | ramindia | :-[ |
| 15:01.36 | ramindia | [TK]D-Fender: is there any place i can find depricated commands from 1.2.X to 1.4.X |
| 15:01.55 | [TK]D-Fender | ramindia: In the wonderful docs included in yuor tarball |
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| 15:02.38 | ramindia | [TK]D-Fender: :P |
| 15:03.09 | *** join/#asterisk fernandojdk (n=fernando@189.88.64.188) |
| 15:03.38 | fernandojdk | hi all |
| 15:05.06 | fernandojdk | i have one question: In my asterisk, i have configure Realtime from users and peers, but, when i try to register one user, the asterisk return a error "Not found". Do asterisk suport to a register trought realtime config? |
| 15:05.10 | thehar | Anyone recall when MinWait was removed from QueueAdd for the ami action? |
| 15:09.14 | thehar | ah it was never available.. appears it was custom in our old pbx |
| 15:10.20 | *** join/#asterisk sevard (i=sev@216.164.6.24) |
| 15:10.26 | sevard | Does anyone here use google voice? |
| 15:11.27 | sevard | I was wondering if anyone has figured out a way to call your GV # with SIP. Is there a npxnxxxxxx@googlevoice.com uri sip gateway? |
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| 15:11.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:12.03 | kaldemar | fernandojdk: yes it does. have you configured extconfig.conf? |
| 15:12.56 | fernandojdk | yes |
| 15:13.00 | kaldemar | how? |
| 15:14.05 | fernandojdk | kaldema: sippers => mysql,general,sippeers |
| 15:14.13 | fernandojdk | kaldema: sipusers => mysql,general,sipusers |
| 15:14.45 | kaldemar | and you have installed res_config_mysql from addons? |
| 15:14.50 | fernandojdk | yes |
| 15:14.55 | fernandojdk | all works fine |
| 15:15.14 | fernandojdk | but, i can't registrate any sipuser |
| 15:15.19 | kaldemar | the table exists? the peer exists in the table? does asterisk make a query? |
| 15:15.51 | fernandojdk | i'm not see if asterisk make a query |
| 15:16.02 | fernandojdk | however, sippeers works fine in realtime scene |
| 15:16.17 | ramindia | fernandojdk: check at * cli realtime mysql status |
| 15:16.56 | fernandojdk | yes, this is conected |
| 15:17.10 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:17.26 | fernandojdk | when i make a call, the call go fine, and the peer is match in the database |
| 15:18.09 | fernandojdk | but, when i try to register with X-Lite i have the message: ...... No matching peer found |
| 15:19.18 | kaldemar | the call that goes through is from asterisk to the peer? |
| 15:19.20 | ramindia | fernandojdk: check the passwords and enable debug and see what happends |
| 15:19.37 | *** join/#asterisk ArchGT (n=ArchGT@190.149.103.56) |
| 15:19.41 | fernandojdk | ok |
| 15:19.59 | kaldemar | sounds like an x-lite configuration issue. |
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| 15:22.10 | pif | anyone using chan_capi with 1.6 ? |
| 15:23.12 | fernandojdk | tha asterisk make a query, however, the query search in the peers database, and not in sipusers database |
| 15:23.16 | fernandojdk | any idea? |
| 15:25.13 | leifmadsen | you register to a sippeer -- not sipuser |
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| 15:26.04 | fernandojdk | right, how i register as sipuser? |
| 15:27.11 | leifmadsen | you don't... |
| 15:27.29 | leifmadsen | sipuser is used for placing calls, the sippeer structure is where the 'host' definition is applied |
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| 15:27.44 | leifmadsen | host=dynamic |
| 15:28.02 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:28.41 | fernandojdk | ok, thanks, i'm try another solution |
| 15:28.43 | fernandojdk | thanks all |
| 15:33.20 | tzafrir_laptop | pif, I see it just got into Debian Unstable. That's about as much as I know |
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| 15:49.59 | DRoBeR | Good evening. |
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| 15:52.04 | *** join/#asterisk ddickenson (n=android@64-194-103-171.wco.clearwire-dns.net) |
| 15:55.40 | ddickenson | remote call pickup. I have uncommented the line from features.conf and added "pickupgroup=#" to the users I wanted in sip.conf. what am I missing? |
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| 15:57.36 | eric_hill | ~pb |
| 15:57.37 | 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://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:57.54 | sevard | Can anyone do a call test to a SIP number for me? |
| 15:58.29 | kaldemar | ddickenson: callgroup definitions |
| 15:58.46 | eric_hill | sevard: What's the number? |
| 15:59.24 | ddickenson | kaldemar: as in callgroup=# also in sip.conf? if so I added that too and still nothing |
| 16:00.55 | kaldemar | ddickenson: yes, as in that. pastebin sip.conf and show a failed pickup. |
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| 16:04.41 | ddickenson | kaldemar: pastebin.com/m2b0c8446 |
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| 16:08.53 | ddickenson | kaldemar: failed pickup doesn't give any output to the cli... |
| 16:10.06 | kaldemar | what peer is the callee and what tries to pick up? |
| 16:10.43 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 16:10.59 | ddickenson | actually it is a ring group that I'm trying to pickup. |
| 16:11.13 | *** join/#asterisk many (n=many@dslb-094-217-199-161.pools.arcor-ip.net) |
| 16:11.26 | ddickenson | the group has the first 4 peers in it though |
| 16:11.59 | ddickenson | and the calling party is coming from outside on the pstn |
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| 16:12.56 | kaldemar | which peer tries to pick up the call? |
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| 16:14.25 | ddickenson | I've tried from "phc-01-0046-0609" and "Drew_Cell" |
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| 16:15.02 | kaldemar | and sip show peer <peer> for those show the groups right? |
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| 16:17.01 | ddickenson | well actually the group is just defined in extensions . conf as exten => 200,1,Dial() and it dials som variables that are tied to those first 4 users. it just dials them all at once for incoming calls |
| 16:17.53 | kaldemar | that doesn't have anything to do with callgroups and pickupgroups |
| 16:18.39 | kaldemar | sip show peer shows Callgroup and Pickupgroup values for a peer. do those exist? |
| 16:18.40 | ddickenson | oh, wait I see what you're saying. yeah it shows callgroup: 1I> and pickupgroup: 1I> |
| 16:20.02 | *** join/#asterisk geek_cl (n=geek@pc-81-222-86-200.cm.vtr.net) |
| 16:21.12 | ddickenson | and some just show pickupgroup: 1... why the extra characters? |
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| 16:25.57 | eric_hill | After digging through the source a bit, get_rpid_num is only called to extract the Remote-Party-ID from a SIP message for the initial invite. Subsequent packets are ignored. |
| 16:28.26 | kaldemar | ddickenson: the extra characters look abnormal |
| 16:29.11 | ddickenson | yeah, I logged in with that user (softphone) and those characters went away |
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| 16:32.37 | ddickenson | do I need any kind of "pickup" extension in my dialplan? I figured the features.conf like that adds the *8 for pickup |
| 16:32.55 | ddickenson | would take care of that |
| 16:33.08 | kaldemar | ddickenson: no, features.conf takes care of that. |
| 16:34.50 | ddickenson | is there some know problem with cisco phones in this? |
| 16:36.21 | kaldemar | look at sip debug to know if something even happens when you try it. |
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| 16:39.17 | *** join/#asterisk ddickenson_ (n=ddickens@166.191.16.149) |
| 16:39.43 | ddickenson_ | Got disconnected... |
| 16:40.48 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 16:41.17 | ramindia | [TK]D-Fender: i have another quick question in 1.2.X iam using "exten => _X.,1,Voicemail(${EXTEN})" so when the user not available it say person extension not available leave mesge....but in 1.4.x it directly leave message " how can i config to say person not available leave message" |
| 16:41.46 | [TK]D-Fender | ramindia: "core show application voicemail" |
| 16:44.05 | ramindia | is this this correct ............exten => _X.,1,Voicemail(${EXTEN},u) |
| 16:44.13 | kmem | i cant seem to change my musiconhold from the default tunes :( |
| 16:44.52 | [TK]D-Fender | ramindia: Sure looks better |
| 16:46.46 | ramindia | [TK]D-Fender: thanks again it worked, before i use to use U before syntax |
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| 16:58.25 | garymc | Hi |
| 16:58.38 | garymc | Was wondering if anyone could aswer a couple of questions for me |
| 16:58.44 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 16:58.59 | garymc | I have a basic client server setup on a Fedora 10 machine |
| 16:59.14 | garymc | Can I put asterisk on it with minimal hardware |
| 16:59.24 | garymc | Like one IE card |
| 16:59.43 | garymc | and plug a usb headset or phone into my client comps and get a phone line to them |
| 16:59.51 | garymc | Looking to do this as cheap as possible |
| 17:00.05 | [TK]D-Fender | garymc: IE card? |
| 17:00.10 | garymc | sorry |
| 17:00.17 | garymc | the card that connects to my phone line |
| 17:00.35 | [TK]D-Fender | garymc: Sure. |
| 17:00.44 | garymc | so its possible |
| 17:00.51 | garymc | that card costs about £210 |
| 17:01.00 | garymc | is that all id need |
| 17:01.01 | [TK]D-Fender | garymc: Just get a compatible FXO interface and you're good to go |
| 17:01.10 | [TK]D-Fender | garymc: Linksys SPA-3102 <- |
| 17:01.10 | garymc | how do i know whats compatible |
| 17:02.15 | garymc | right |
| 17:02.24 | garymc | so I want to record all incoming calls |
| 17:02.36 | garymc | should I build a seperate server and put Astlinux on it? |
| 17:02.47 | garymc | then plug that server into my ethernet switch |
| 17:02.48 | garymc | ? |
| 17:02.59 | garymc | so my other server can see it and all the clients |
| 17:03.16 | garymc | see it starts to get a little complex for my brain, but i need to know how it works |
| 17:03.33 | garymc | also im not sure how my isdn line will connect in? |
| 17:03.43 | garymc | i got 2 isdn line with BT |
| 17:04.26 | webman | does anyone know if there is a config option to stop a polycom from stripping the # from the end of a dial string ? |
| 17:04.34 | [TK]D-Fender | garymc: You don't need 2 servers for this |
| 17:04.51 | [TK]D-Fender | webman: "removeendofdial" |
| 17:04.52 | garymc | even if im saving calls and using other apps on the server |
| 17:05.00 | garymc | by 10 employees at once? |
| 17:05.27 | [TK]D-Fender | garymc: My home server is my router, web/file/ftp server, runs *, is my HTPC and makes me COFFEE |
| 17:05.46 | garymc | i wish i knew what all that meant |
| 17:06.24 | garymc | i just want as many phone lines as possible as cheaply as possible |
| 17:06.25 | webman | garymc: most decent hardware will do whatever you want on 10 concurrent channels... |
| 17:06.44 | garymc | and you say the ;linksys SPA-3102 will do the job? |
| 17:06.50 | garymc | no ech stuff |
| 17:06.53 | garymc | *echo |
| 17:07.02 | *** join/#asterisk galeras (n=galeras@186.80.186.161) |
| 17:07.03 | webman | garymc: more than 20 or 30 channels and you might start to have issues unless you get good hardware |
| 17:07.40 | garymc | will it handle normal calls like anolouge and digital? |
| 17:08.01 | garymc | and fire them over to my client computer phones plugged into the client comps via usb? |
| 17:08.11 | BlargMaN00 | What is the best way to change the MOH in a conference room?? |
| 17:08.16 | [TK]D-Fender | garymc: You should really start to be VERY clear about what kind of lines, and how many you intend to use |
| 17:09.08 | garymc | hmmm |
| 17:09.10 | garymc | ok |
| 17:09.18 | garymc | Ive got 2 ISDN lines coming into my office |
| 17:09.28 | garymc | using BT versatility |
| 17:09.36 | webman | garymc: more than 4 lines, and I would strongly suggest to use digital lines... preferably ISDN |
| 17:09.38 | garymc | "dont know if youve heard of that"? |
| 17:09.43 | [TK]D-Fender | garymc: then you need a completely different interface |
| 17:09.50 | garymc | ok |
| 17:09.54 | garymc | what would i need? |
| 17:10.07 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:10.08 | garymc | I appreciate the advice by the way |
| 17:10.19 | [TK]D-Fender | garymc: I don't know BRI interfaces too well, but I know on the economy side AVM fritz is used by many |
| 17:10.31 | webman | garymc: if you intend on expanding to more than 4 channels (2 lines) then you should get a ISDN PRI (10 channel) now, and save buying double hardware later... |
| 17:10.38 | [TK]D-Fender | garymc: Digium & sangoma also make interfaces for this |
| 17:11.01 | garymc | yes |
| 17:11.13 | webman | TKD: avm fritz gave a lot of problems when trying to use more than one in a single machine.... we ended up replacing it for the digium B410p |
| 17:11.21 | garymc | See i dont know what I need for this, if i explain my setup maybe you could advise what I need |
| 17:11.40 | webman | mind you that also gave some problems..... but digium were good to support it/help get it installed/working |
| 17:11.56 | garymc | Im starting with a App Server to launch 10 Client computer off |
| 17:12.20 | garymc | I want to connect those client comps 10 of them with a phone line via usb |
| 17:12.37 | garymc | also I need to record all incoming and outgoing calls |
| 17:12.43 | garymc | and find them easily |
| 17:12.53 | garymc | what would I need to do this? |
| 17:13.25 | webman | garymc: I would go for the digium b410p because digium will help you with install/configure ... |
| 17:13.48 | garymc | is that expensive? |
| 17:14.04 | webman | also, a semi decent (ie, any current model) server, with reasonable speed HDD, and a reliable power supply |
| 17:14.32 | webman | garymc: dunno, it does up to 4 x ISDN BRI, you should look for prices locally... |
| 17:14.40 | garymc | so should I have a seperate server to do this from my app one? |
| 17:14.40 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:14.48 | garymc | I found a price |
| 17:14.51 | webman | what does the "app" server do ? |
| 17:14.53 | garymc | about £500 |
| 17:15.05 | webman | yep, sounds about right... |
| 17:15.18 | garymc | It has programs on it and client comps connect off it via the network card in them |
| 17:15.29 | webman | so it does file sharing ? |
| 17:15.29 | garymc | so the clients use the networks resources and programs |
| 17:15.39 | garymc | yes |
| 17:16.12 | garymc | The take details off customers and input them via a webpage on an intranet and all gets save don a mysql database |
| 17:16.18 | webman | you could share this same computer, but I would suggest you record your phone calls to a different drive, and it should be SATA/SCSI/SAS not an old IDE disk |
| 17:16.42 | garymc | so I could use the one server for all this stuff |
| 17:16.49 | [TK]D-Fender | garymc: And more |
| 17:16.56 | garymc | just need a seperate scsi drive for call recording |
| 17:17.13 | garymc | and the asterix software will do all this |
| 17:17.30 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:17.39 | webman | garymc: yes.... and no... :) You should be able to, but everything is always a "it depends" ... on your current cpu, ram, HDD, how demanding your php/mysql/etc apps are, etc etc.... |
| 17:18.16 | garymc | well Im gonna get a quad core chip with about 8 to 16 gig ram |
| 17:18.28 | garymc | and lots of HDD space |
| 17:18.30 | webman | garymc: basically, if you want to start cheap, then use the same PC, then later, you might either upgrade the PC, or else get a new one... but at least you will know the concept works in real life |
| 17:18.50 | garymc | yeah I need to test all this before I purchase the server |
| 17:18.54 | webman | the main prob with recording is the disk bandwidth/etc... |
| 17:19.04 | garymc | disk bandwidth? |
| 17:19.08 | *** join/#asterisk bmg505 (n=leon@41-195-68-93.access.uunet.co.za) |
| 17:19.11 | webman | but usually that is only a problem when talking about 100 channels or more :) |
| 17:19.24 | garymc | bandwidth of my phone connection/internet |
| 17:19.33 | garymc | or the disk drive itself |
| 17:19.44 | webman | garymc: no, bandwidth between the HDD and the motherboard |
| 17:19.51 | garymc | hmmm |
| 17:19.58 | garymc | never heard of that |
| 17:20.06 | garymc | iam a bit of a newbie though |
| 17:20.15 | webman | garymc: that is why I said to use a different disk.... saves head seeking/etc... |
| 17:20.22 | garymc | yes |
| 17:20.51 | *** join/#asterisk MACscr (n=Mark@98.214.100.212) |
| 17:20.57 | webman | garymc: I'm just giving you the worst case scenario... some big call centers get multiple machines, use ram drives, and all sorts of tricks to get call recording to work |
| 17:21.16 | webman | garymc: but you aren't in that category with only 10 channels or less :) |
| 17:21.22 | MACscr | what is the recommended software these days for windows? Im not happy with pangolin |
| 17:21.34 | MACscr | whoops, meant to say softphone |
| 17:21.52 | webman | MACscr: I use diax... but nobody else seems to like it :) |
| 17:22.20 | garymc | webman: cool |
| 17:22.36 | garymc | So there are phones that plug into a client comps usb drive? |
| 17:23.10 | grandpapadot | We record thousands of calls an hour to our SAN in real-time and it works great. |
| 17:23.16 | [TK]D-Fender | garymc: No. |
| 17:23.21 | grandpapadot | Asterisk 1.4, dual opterons. |
| 17:23.30 | MACscr | well, one thing that I really hate about pangolin is that you have to have the phone number perfect, it doesn't parse out hyphens, periods, or convert letters to numbers. Also, when adjusting its microphone settings, it doesn't do it just for itself, it changes the main windows mic settings. So I constantly have to adjust it |
| 17:23.31 | grandpapadot | From just under 40 servers. |
| 17:23.37 | [TK]D-Fender | garymc: Ethernet (SIP) yes, USB = dumb audio device for use with a soft-phone |
| 17:23.41 | garymc | so how would I connect all my phones for my client comps up? |
| 17:24.07 | garymc | Ok id need an Ethernet switch big enough |
| 17:24.11 | garymc | ? |
| 17:24.14 | [TK]D-Fender | garymc: Yup |
| 17:24.17 | garymc | hmmm |
| 17:24.19 | garymc | ok |
| 17:24.32 | [TK]D-Fender | garymc: Or get phones with a passthrough switched port |
| 17:24.38 | LeddyHM | wow, we must have the same schedule tk |
| 17:24.48 | LeddyHM | both idle till ~the same time |
| 17:24.57 | webman | garymc: you either use a proper voip phone (I love the polycom ones, the extra money is worth it IMHO) or else you can use a mic/headset with a software phone on the PC |
| 17:25.33 | grandpapadot | We've found that CounterPath's Bria + a really good USB headset work best with asterisk and it supports g729. |
| 17:25.42 | grandpapadot | Polycom is my pick as well. |
| 17:25.58 | webman | garymc: you will get much better audio quality and reliability by using a polycom phone... I can't comment on any other brand except grandstream, and they were crap (again, in my opinion)... |
| 17:26.26 | grandpapadot | Hey also the Polycom 330 has a mini-dic headphone/mic plug and that phone works great with mobile phone headsets. |
| 17:26.36 | grandpapadot | .. and are about $110 |
| 17:26.59 | webman | grandpapadot: what version of ast do you use? and how are your calls delivered to your system? (dahdi/voip/other)? |
| 17:27.25 | grandpapadot | 1.4.26, SIP |
| 17:27.34 | grandpapadot | and we're transcoding a lot |
| 17:27.57 | generalhan | ariel_: you around? |
| 17:28.33 | ariel_ | generalhan: yes |
| 17:28.33 | webman | grandpapadot: hmmm, I just upgraded to that version last week, and have had so many load issues that tonight I am upgrading to 1.6.0 to see if it is any better |
| 17:29.15 | webman | load issues means more than 30 to 40 concurrent calls bridged from zap(pri) to mix of SIP and IAX2 |
| 17:29.32 | generalhan | ariel_: i was talking to you last week about the HP DL160 you said you use as an asterisk server ... i was wondering how you got power back to the PCI cage on that server. everyone i talk to at HP says its not possible (of course) so i was wondering if you remembered where you pulled that power from |
| 17:30.03 | webman | I think the main issue is that a lot of the channels are very short lived.... ie, dial, busy, dial, busy, dial, etc... |
| 17:30.42 | webman | but of course, it all worked perfectly under an old 1.2 version (except for the crash once every few days recently)... |
| 17:30.43 | grandpapadot | webman: I've been half paying attention, what kind of load issues? We have our systems limited to around 100 channels and they average 80-100 all day long with full recording and probably more than half the calls transcoded g729->ulaw. |
| 17:31.26 | webman | grandpapadot: well, 100% are transcoded from alaw -> g729, there is no apps/monitoring/recording/etc ... |
| 17:31.49 | grandpapadot | What distro? |
| 17:31.50 | garymc | webman so these phone will plug into the same switch as the client comps? |
| 17:32.01 | webman | the problem is when we get over 40 calls, (sometimes less) we get load average shooting up to 40 or more |
| 17:32.03 | garymc | So basically each employee bay will take up 2 ethernet ports |
| 17:32.29 | grandpapadot | garymc: Most decent IP phones have a built-in two port ethernet bridge. |
| 17:32.31 | webman | garymc: yes, they can do, depending on the model you can plug the computer into the second ethernet on the phone |
| 17:32.44 | webman | grandpapadot: debian stable (lenny) |
| 17:32.45 | garymc | ahhhh ok |
| 17:33.01 | webman | grandpapadot: dual CPU opteron with 2G RAM |
| 17:33.10 | *** join/#asterisk darkmadda (n=dylan@c-76-27-95-83.hsd1.ut.comcast.net) |
| 17:33.25 | grandpapadot | webman: We're running debian (etch) latest, dual opterons with k7 kernel, no special asterisk 1.4 compile-time settings, but we do have a lot of custom modules such as our load balancing stuff. |
| 17:33.48 | grandpapadot | webman: And the CPU stats stay around 25-30% all day long. |
| 17:34.02 | webman | grandpapadot: I have no special asterisk settings, except to noload almost all modules :) |
| 17:34.06 | grandpapadot | webman: They frankly perform extremely well (looking for wood to knock on). |
| 17:34.12 | darkmadda | how to i put someone on hold when i'm using an analog for (connected to a fxs) |
| 17:34.33 | webman | I *know* it should all work really well, I just don't know how to find out what is chewing the CPU ....... |
| 17:34.38 | grandpapadot | webman: Got anything funky running on debian? We don't have any lenny systems deployed ... |
| 17:34.45 | webman | darkmadda: flash |
| 17:34.54 | MACscr | if I forward my DID at my provider to my cellphone, think I could get a SMS message to that number, but at my cell? |
| 17:35.15 | grandpapadot | webman: etch is just so damn stable we haven't even considered upgrading. |
| 17:35.48 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 17:35.53 | grandpapadot | MACscr: You would have to have an SMS gateway plus connectivity to a provider going to forward SMS, they don't go over the phone network. |
| 17:36.10 | webman | grandpapadot: just mysql, ntpd, stund, lighttpd, exim4 (dumb, no traffic), sshd, syslog, thats it (and asterisk) |
| 17:36.27 | grandpapadot | webman: hrm... what hardware specs? |
| 17:36.31 | darkmadda | webman: flash? |
| 17:37.00 | MACscr | grandpapadot: the reason why I ask, is that I would like to use skype every so often for phone calls (using the actual skype app), but I want to set the caller id to my actual pbx system. But in order to do that, skype sends a text message to that number to verify it |
| 17:37.07 | webman | dual opteron 244 with 2GB RAM, SAS drives 72G * 2 in RAID1 (hardware raid) |
| 17:37.37 | grandpapadot | webman: Ok, that's pretty close to ours. You using the k7 kernel? |
| 17:37.48 | webman | darkmadd: hook flash (basically hangup for .5 seconds |
| 17:37.52 | grandpapadot | webman: 2.6.18-6-k7 |
| 17:38.16 | webman | grandpapadot: on lenny it is 2.6.26-2-amd64 |
| 17:38.28 | grandpapadot | So you're running asterisk in a 64-bit environment? |
| 17:38.42 | webman | grandpapadot: yes, 64 bit |
| 17:39.06 | grandpapadot | webman: I would focus my research on how well asterisk 1.4 runs in 64-bit, that's out of my experience range. |
| 17:39.23 | grandpapadot | webman: And also look into the g729 module and 64-bit. |
| 17:39.32 | webman | I really don't see why it would run worse than 32bit ?? |
| 17:39.48 | grandpapadot | webman: Like I said, we're running 32-bit on etch and we're getting like twice your capacity with no issues at all. |
| 17:39.53 | *** join/#asterisk propellerhead (n=yogurt2u@155-38-17-190.fibertel.com.ar) |
| 17:39.54 | webman | I tried all the g729 modules, and the only one that worked was the opteron one |
| 17:40.06 | grandpapadot | webman: consider running 32-bit ... |
| 17:40.17 | grandpapadot | webman: and frankly etch which you're at it ... |
| 17:40.40 | grandpapadot | webman: You're not going to see a human-noticeable performance difference running 64-bit kernels right now. |
| 17:40.40 | webman | grandpapadot: yes, I realise your system is similar in spec, and you get *much* better performance... under etch with 1.2 asterisk the load never went above 1.0... ever |
| 17:41.13 | grandpapadot | Anyone care to chime in on the state of asterisk on 64-bit systems? Any other experiences in channel? |
| 17:41.26 | Qwell | grandpapadot: works just fine.. |
| 17:41.28 | Chainsaw | grandpapadot: I run 1.2.32 on AMD64. No problems to report. |
| 17:41.30 | webman | the main reason to upgrade to lenny was for the newer kernel... we were using a custom 2.6.11 because the original etch one wouldn't work on the hardware |
| 17:41.31 | MACscr | grandpapadot: yeah, works fine |
| 17:41.51 | MACscr | webman: I prefer to use enterprise os's for phone systems, such as centos |
| 17:42.12 | grandpapadot | @Qwell: Which g729 transcoder module are you using? |
| 17:42.35 | Qwell | grandpapadot: none, BUT, if you want to figure out which one to use... grab the benchg729 util from downloads.digium.com |
| 17:42.51 | grandpapadot | Qwell: tnx |
| 17:43.14 | grandpapadot | webman: Have you tried that benchg729 util? It basically profiles your system and tells you which g729 module to use, just a thought. |
| 17:44.05 | Qwell | grandpapadot: he did |
| 17:44.11 | webman | I used the benchg729 and it told me to use the opteron one, I also tried a few other versions, but none of the others worked at all (just crashed asterisk on startup) |
| 17:44.23 | generalhan | ariel_: any recollection on how you accomplished that task ? |
| 17:47.29 | webman | like I said before, the only other thing I can think of is that it just doesn't handle lots of very short calls as well as 1.2.x did .... |
| 17:47.36 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:47.44 | grandpapadot | webman: Well I'm out of ideas, lol. If it were me, and I didn't want to switch to 32-bit or go back to etch, i would start looking at how asterisk/zaptel/libpri were compiled, maybe there are some optimizations for your hardware in there somewhere. |
| 17:48.42 | webman | or it sometimes gets 'stuck' and gets a big backlog, and then catches up again a few mins later ... |
| 17:48.48 | darkmadda | is there a command to turn on/off a mwi? |
| 17:48.55 | webman | grandpapadot: well, I'm going to try asterisk 1.6.0.11 + dahdi/etc... if this doesn't work out, then I'll probably go back to the original 1.2 version and see what happens.... |
| 17:50.49 | grandpapadot | webman: Good luck! |
| 17:51.37 | webman | grandpapadot: thanks... I'll find out tomorrow when the calls start :) tonight I'm just testing with one channel to make sure I didn't mess up the config :) |
| 17:52.10 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 17:59.00 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 18:01.31 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 18:02.07 | webman | hmmm, grandpapadot: can you compare this please http://www.pastebin.ca/1509126 |
| 18:02.12 | grey-monkey | Hey, what do you guys think about recompiling the kernel to optimize asterisk? |
| 18:02.37 | webman | or anyone else, just "core show translation recalc 20" output, specifically the g729 values |
| 18:02.40 | grey-monkey | This guy says that asterisk postgres db schema is buggy in 1.6 (http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian). Is that true? |
| 18:02.52 | grey-monkey | Has anyone seen this article? |
| 18:03.33 | webman | grey-monkey: I haven't seen it.... is he referring to 1.6.0 or 1.6.1 or 1.6.2 ? |
| 18:03.59 | grey-monkey | He just says 1.6+ |
| 18:04.31 | grey-monkey | He's downloading the latest from the svn trunk in the tutorial |
| 18:04.56 | webman | grey-monkey: well, if anyone finds a bug in any open source project (like asterisk) the least they could do is submit a bug report .... if they haven't, you should remind them to do so |
| 18:07.19 | webman | hmmm, is there 1000 microseconds in one millisecond ? |
| 18:08.07 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 18:10.08 | jdnWEST | Anyone have PRI's or data service from TWTC? |
| 18:10.13 | *** part/#asterisk darkmadda (n=dylan@c-76-27-95-83.hsd1.ut.comcast.net) |
| 18:10.26 | grey-monkey | Isn't there a way to reload asterisk config without taking down the server? |
| 18:10.38 | eric_hill | grey-monkey: What config? |
| 18:10.48 | eric_hill | grey-monkey: dialplan reload, sip reload, etc. |
| 18:10.56 | grey-monkey | yes. All of them. |
| 18:11.09 | eric_hill | grey-monkey: and reload won't work? |
| 18:11.22 | grey-monkey | That's my question |
| 18:11.39 | eric_hill | grey-monkey: reload doesn't "take down the server". |
| 18:11.40 | webman | grey-monkey: yes, just use "reload" |
| 18:11.57 | grey-monkey | webman: Great. Thanks! |
| 18:12.08 | eric_hill | grey-monkey: And reloading the *whole* configuration is quite a bit overkill for just changing the dialplan... |
| 18:12.09 | webman | eric_hill: well, it'll drop your sip realtime cached entries ... :) |
| 18:13.17 | webman | eric_hill: yes, usually you only need to reload one file at a time.... worst case you can also use "restart now" to take kill asterisk and restart immediately... (a few seconds downtime) |
| 18:20.00 | *** join/#asterisk intel352 (n=intel352@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:22.19 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 18:22.29 | levity | hello folks, what would be the meaning "SIP/2.0 489 Bad event" would somebody have a look at my pastebin? -> http://pastebin.com/d62f970c0 it doesn't seem to be causing any technical difficulty, just looks like an error message and would like to fix it if possible |
| 18:22.59 | webman | thanks for the help guys... I gotta grab some sleep so that in 4/5 hours when people call me to say how well it is working I'll be awake enough to smile :) |
| 18:23.16 | [TK]D-Fender | levity: * doesn't respond to OPTIONS packets |
| 18:23.39 | [TK]D-Fender | levity: (positively). It is enough to work as a keep-alive however |
| 18:24.45 | levity | [TK]D-Fender: what can I do to fix it? |
| 18:25.00 | [TK]D-Fender | levity: there is nothing to fix. this is not "broken" |
| 18:25.37 | levity | ok thanks [TK]D-Fender, just was being paranoid ;) |
| 18:25.55 | [TK]D-Fender | levity: Oh, we're still out to get you ;) |
| 18:26.58 | levity | appreciate the help |
| 18:27.32 | kaldemar | that looks like an answer to a NOTIFY, with an Event header that * doesn't understand. |
| 18:28.45 | levity | kaldemar: so its my ITSPs fault? |
| 18:29.12 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 18:31.28 | kaldemar | or your fault if you don't support what they send (and if you ask them, for sure). depends on the point of view. |
| 18:32.44 | levity | kaldemar & [TK]D-Fender thanks again, must be going |
| 18:33.16 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 18:34.46 | *** join/#asterisk sjobeck (n=Adium@32.159.11.40) |
| 18:35.10 | *** part/#asterisk sjobeck (n=Adium@32.159.11.40) |
| 18:37.03 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 18:37.50 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:38.51 | *** join/#asterisk SkramX2 (i=mark@phalse.2600.COM) |
| 18:39.10 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 18:42.55 | *** join/#asterisk vicscandl (n=vicscand@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 18:54.23 | *** join/#asterisk niros (i=c075ec1d@gateway/web/freenode/x-b92d742b40512dce) |
| 18:54.33 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:58.02 | Psychobilly | anyone has any experience with openvox cards for asterisk? Are they any good? |
| 18:58.35 | *** part/#asterisk SkramX2 (i=mark@phalse.2600.COM) |
| 19:01.53 | MACscr | [TK]D-Fender: Does my DID/sip provider need to have the ability to do SMS or does just my pbx? |
| 19:04.00 | [TK]D-Fender | Psychobilly: YMMV, but don't expect much by way of sympaties if yourun into problems and start hitting walls trying to get tech support |
| 19:04.46 | Psychobilly | YMMV? :) |
| 19:04.56 | [TK]D-Fender | ~ymmv |
| 19:04.57 | infobot | rumour has it, ymmv is Your Mileage May Vary |
| 19:05.06 | Psychobilly | thx |
| 19:05.51 | Psychobilly | im looking for some mini-pci solution for embedded boxes, do u have any suggestions? |
| 19:06.40 | [TK]D-Fender | Psychobilly: Nope. |
| 19:06.48 | *** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 19:06.51 | [TK]D-Fender | Psychobilly: They may be one of the few who do this |
| 19:07.32 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 19:08.52 | Psychobilly | i see |
| 19:11.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:11.44 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 19:12.50 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 19:25.24 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 19:25.44 | timeshell | Greetings |
| 19:28.18 | grandpapadot | Greetings, mighty baud warrior! |
| 19:31.00 | *** join/#asterisk sjobeck (n=Adium@68.178.19.156) |
| 19:32.30 | timeshell | Having trouble getting a dahdi channel to show a custom caller ID |
| 19:32.57 | timeshell | In dahdi-channels.conf I have set the callerid=Back Door |
| 19:33.08 | timeshell | But it comes up on the phones as Unknown |
| 19:34.24 | [TK]D-Fender | timeshell : first, use quotes, second pastebin the call attempt (CLI w/ noop), and configs |
| 19:34.46 | timeshell | Yah, just trying quotes now |
| 19:34.49 | timeshell | just as ec |
| 19:34.53 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:39.27 | tzafrir_laptop | timeshell, maybe 'callerid' must have a number part? |
| 19:39.39 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:39.41 | timeshell | Tried that. |
| 19:39.43 | tzafrir_laptop | callerid = Back Door <> |
| 19:39.50 | timeshell | I tried callerid="Back Door" <0000000000> |
| 19:43.02 | Qwell | No quotes ;p |
| 19:44.43 | timeshell | Bah |
| 19:46.02 | timeshell | Qwell Nope |
| 19:46.04 | timeshell | That didn't work |
| 19:46.24 | timeshell | Does callerid apply to both fxs_ls and fxs_ks signalling? |
| 19:47.39 | *** join/#asterisk iflux (i=iconicfl@www.kevinlynn.com) |
| 19:48.13 | mmattice | chan_dahdi.c:10728 dahdi_pri_error: 1 !! Got reject for frame 0, retransmitting frame 0 now, updating n_r! |
| 19:48.19 | mmattice | timing issue? |
| 19:49.24 | timeshell | [TK]D-Fender |
| 19:49.26 | timeshell | [TK]D-Fender http://www.pastebin.ca/1509241 |
| 19:50.18 | [TK]D-Fender | timeshell : signalling=fxs_ks <- PARDON? |
| 19:50.31 | [TK]D-Fender | timeshell : You can't set CID on FXO port |
| 19:50.52 | timeshell | 1. That's why I asked. 2. Why not? |
| 19:51.04 | timeshell | The incoming is from a phone intercom |
| 19:51.18 | timeshell | I want it to be set to something specific. |
| 19:51.49 | [TK]D-Fender | timeshell : Ok, INBOUND. I also don'[t see quotes, and I wouldn't trust that you restarted * for this to take effect. |
| 19:52.14 | timeshell | Well 1. I ALWAYS shut down and restart Asterisk for changes. |
| 19:52.22 | timeshell | 2. I tried with both quotes AND without. |
| 19:53.49 | [TK]D-Fender | timeshellwhat do you see on it currently? |
| 19:54.53 | timeshell | Just Unknown. |
| 19:54.55 | timeshell | However |
| 19:55.31 | timeshell | I just noticed that the gui put callerid in the dahdi-channels.conf twice, with the second one being blank. |
| 19:55.33 | mmattice | nobody's clueful on the pri's today? |
| 19:55.41 | timeshell | It might be overriding it. Gonna delete it and try again |
| 19:56.23 | jaytee | gui? when did Asterisk get a gui? |
| 19:57.15 | timeshell | Nope. That didn't work |
| 19:57.18 | kaldemar | that's freepbx |
| 19:57.51 | [TK]D-Fender | timeshell : I also se you changing the callerpres. |
| 19:58.01 | [TK]D-Fender | timeshell : trust[-1] |
| 19:58.08 | [TK]D-Fender | timeshell : remove that and test |
| 19:58.26 | timeshell | That should be an ExecIf where the callerid =""\ |
| 19:58.54 | [TK]D-Fender | timeshell : -- Executing [s@DID_trunk_3:2] ExecIf("DAHDI/5-1", "1,Set,CALLERID(all)=unknown <0000000>") in new stack < - 1 |
| 19:59.13 | [TK]D-Fender | timeshell : Code I didn't get to see.... |
| 19:59.15 | timeshell | Yah I saw that |
| 20:01.06 | timeshell | exten = s,1,ExecIf($[ "${CALLERID(num)}"="" ],SetCallerPres,unavailable) |
| 20:01.08 | timeshell | exten = s,2,ExecIf($[ "${CALLERID(num)}"="" ],Set,CALLERID(all)=unknown <0000000>) |
| 20:01.36 | timeshell | Is <0000000000> == "" ?? |
| 20:04.34 | timeshell | Yah, it apparently is that callerpres thing. |
| 20:04.38 | timeshell | bah |
| 20:09.21 | [TK]D-Fender | timeshell : whitespace = BAD, and always NoOp Your callerid rather than guess. And it won't be ""... you set it in chan_dahdi to have zeroes <- |
| 20:09.44 | timeshell | Well that's what I thought too. |
| 20:09.56 | timeshell | BUT, it's definitely what's overriding my settings. |
| 20:10.06 | timeshell | When I commented the execif's out, it came up on the phones correctly. |
| 20:10.16 | [TK]D-Fender | timeshell : Yup. You eally ought to stop doing this to yourself... |
| 20:10.21 | [TK]D-Fender | really* |
| 20:10.29 | timeshell | Doing what? |
| 20:10.35 | timeshell | Using the GUI? |
| 20:10.46 | timeshell | I'm using the GUI because it's my JOB to./ |
| 20:11.15 | [TK]D-Fender | timeshell : So that code is 100% generated? |
| 20:11.22 | timeshell | Mostly. |
| 20:11.24 | [TK]D-Fender | timeshellGo bitch to awk_r :p |
| 20:11.36 | timeshell | I would if he was here. |
| 20:11.39 | [TK]D-Fender | timeshel: lMOSTLY huh... what part is "mostly"? |
| 20:11.51 | timeshell | Some of it is still hacked to make it work correctly. |
| 20:11.57 | timeshell | Here's another hack for me. |
| 20:12.06 | [TK]D-Fender | timeshell: Um... it doesn't work ;) |
| 20:12.31 | timeshell | Microsoft doesn't work either. People are happy when it LOOKs like it works. |
| 20:13.26 | timeshell | Anyway, Just gonna comment those lines for now. |
| 20:15.20 | *** join/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 20:16.11 | cryptanthus | Is there someone who could help me with installation of Wanpipe drivers for a Sangoma A200 card? |
| 20:16.37 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 20:17.17 | cryptanthus | /var/log/messages shows that Error: TDM Voice prot not compiled during installation process. |
| 20:19.49 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:21.35 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 20:22.39 | *** join/#asterisk K3rN3L (n=dam@infapen.com) |
| 20:22.45 | K3rN3L | hi everybody |
| 20:22.47 | K3rN3L | i have a question |
| 20:22.58 | K3rN3L | how i can make a auto dial out with a Zap channel? |
| 20:23.27 | K3rN3L | i can auto dial out to the sip extension and when this is answer i play a sound, but with a Zap channel i cant do this |
| 20:23.32 | grandpapadot | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 20:23.40 | *** join/#asterisk mphill (n=mphill@174.37.19.92) |
| 20:24.16 | K3rN3L | yep grandpapadot i read this but if i use a zap channel the asterisk play the sound before the call its answer |
| 20:24.31 | mphill | anyone used the snom m3 and aastra 420 dect phones and have an opinion? |
| 20:25.25 | [TK]D-Fender | K3rN3L: You can if your channel supoprts CALL PROgRESS. |
| 20:25.29 | ACK-NAK | No PERSONAL experience with either but word on the street is that you want a siemens gigaset |
| 20:25.51 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:25.58 | [TK]D-Fender | K3rN3L: However "callprogress=yes" is also synonymous with "disconnectmycallsatrandom=yes" |
| 20:26.01 | K3rN3L | [TK]D-Fender: and sorry but how i can know if my channel support call progress? |
| 20:26.07 | ACK-NAK | mphill: No PERSONAL experience with either but word on the street is that you want a siemens gigaset |
| 20:26.17 | mphill | interesting |
| 20:26.47 | [TK]D-Fender | K3rN3L: this is an option on analog and is unreliable. Things will eventually go wrong, but your situation might be survivable |
| 20:26.56 | ACK-NAK | mphill: Fabulous sound, FXO failover, WIDEBAND support, 6 sip registrations. CHEAP. |
| 20:26.58 | K3rN3L | [TK]D-Fender: trying.. |
| 20:27.03 | cryptanthus | Does anyone have experience with Sangoma A200 Cards using Fedora? |
| 20:27.21 | timeshell | [TK]D-Fender Ok. Here's the thing. This code should be correct. CALLERID(num) is valid and shouldn't be equal to "" since it was set to <0000000000> like we said. HOWEVER, notice that CLI said it resolved to true (or 1) here : -- Executing [s@DID_trunk_3:1] ExecIf("DAHDI/5-1", "1,SetCallerPres,unavailable") in new stack |
| 20:27.23 | ACK-NAK | mphill: I own the gigaset after saying no to the Aastra and Snom & polycom offeerings |
| 20:27.35 | timeshell | Why would that be? |
| 20:27.59 | mphill | ACK-NAK: thanks, wideband is a nice feature to have for sure. how is the range? |
| 20:28.31 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 20:29.02 | [TK]D-Fender | timeshell : I don't see your updated code, its execution, or the dump showing the contents of CID prior to manipulation & testing |
| 20:29.22 | ACK-NAK | Great. About the same as any DECT device. |
| 20:29.32 | timeshell | nm |
| 20:29.35 | ACK-NAK | Can someone clarify a simple concept in chan_dahdi.conf? I don't see the [contexts] that I see in the other conf files, that groups settings together. How is this controlled in chan_dahdi? Is this done by the "group=..." parameter itself? For example channels 1-23 are CPE signalling and 25-47 are NET signalling. Each group should start in a different context. Where does one grouping end, and the other one start? |
| 20:30.15 | [TK]D-Fender | timeshell : C'mon, you're asking for a judgement and not showing the crucial bits. What are you expecting here? |
| 20:30.51 | timeshell | A little logic. You saw the original code and its result. It should be correct from what I can see. |
| 20:31.16 | [TK]D-Fender | ACK-NAK: group=1 channel =>1-23 group=2 channel =>25-47 |
| 20:31.30 | [TK]D-Fender | timeshell: No way am I ever going to trust this blind. |
| 20:31.32 | kaldemar | ACK-NAK: chan_dahdi.conf doesn't have contexts like other files. channel parameters are configured above channel lines and apply until otherwised configured. |
| 20:31.41 | [TK]D-Fender | timeshell: Too much BS always happens int he background. |
| 20:32.09 | jaytee | channel => should be the last line for each group and each group should start with group="somenumbergoeshere" |
| 20:32.11 | [TK]D-Fender | timeshell: Because things always go as planned and changes are always done perfectly. |
| 20:32.21 | [TK]D-Fender | </not> |
| 20:32.23 | kaldemar | ACK-NAK: and group parameters only has effect when you dial a group of channels, it is not something to group channels with configuration wise. |
| 20:32.43 | ACK-NAK | [TK]D-Fender: Thanks. so in other words, any parameters I specify accrue to the previous group=... until I specify a different group? I think I've got it. Thanks! |
| 20:32.52 | K3rN3L | [TK]D-Fender: |
| 20:32.58 | K3rN3L | i put this in my zapata.conf |
| 20:33.17 | K3rN3L | channel => 1-8 |
| 20:33.18 | K3rN3L | callprogress = yes |
| 20:33.46 | K3rN3L | i need reboot asterisk? |
| 20:33.55 | K3rN3L | or only reload in asterisk -rvvvv |
| 20:34.02 | kaldemar | K3rN3L: restart |
| 20:34.16 | [TK]D-Fender | K3rN3L: parameters have to appear BEFORE the channel declaraion. |
| 20:34.28 | [TK]D-Fender | K3rN3L: Fix your configs, then restart * <- |
| 20:34.56 | K3rN3L | Ok [TK]D-Fender |
| 20:35.01 | [TK]D-Fender | Well, times up. I'm out for a while |
| 20:35.01 | K3rN3L | thanks trying again |
| 20:35.05 | K3rN3L | thks kaldemar |
| 20:39.25 | cryptanthus | Has anyone used any Sangoma telephony cards on Linux? |
| 20:39.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:40.13 | *** join/#asterisk galeras (n=galeras@186.80.186.161) |
| 20:41.35 | raden_work | version 1.6 insecure=vary doesnt work ? |
| 20:41.39 | raden_work | very |
| 20:41.51 | raden_work | [Jul 27 15:39:10] WARNING[21424]: chan_sip.c:20193 set_insecure_flags: Unknown insecure mode 'very' on line 15 |
| 20:41.54 | Qwell | raden_work: nope. there were deprecation notices in 1.4... |
| 20:42.00 | raden_work | omfg |
| 20:42.24 | Qwell | and it told you it was removed in UPGRADE.txt |
| 20:42.29 | Qwell | which you read...right? |
| 20:43.10 | Qwell | okay, I lied. it isn't there... |
| 20:43.58 | eric_hill | Anyone going to be at asterisk advanced training Aug 24-28 in LV? |
| 20:44.48 | raden_work | Qwell, i got it figured out changed to invite instead |
| 20:44.58 | Qwell | port,invite |
| 20:45.23 | file | actually very is equivalent to invite,port |
| 20:45.35 | file | Qwell: ohsnapz |
| 20:45.41 | Qwell | SLOOOOOOW |
| 20:46.18 | raden_work | ? |
| 20:46.53 | *** join/#asterisk sjobeck (n=Adium@68.178.19.156) |
| 20:48.07 | *** part/#asterisk sjobeck (n=Adium@68.178.19.156) |
| 20:51.05 | ACK-NAK | Sometimes, when I tell asterisk to reload the dial plan it hangs for ten to fifteen seconds. It seems to happen at random times. During that 'hang' time it doesn't always route calls properly. I've seen this across many versions of 1.4 and 1.6 for a few years. Can anyone shed light on what's happeing? |
| 20:52.19 | raden_work | I have a dialtone asterisk says SIP is registered but dial out nothing happens cant dial in |
| 20:52.22 | galeras | I will apreciate any suggestion to generate a cdr record for an auto-dial out call, my callfile and extensions.conf are at http://pastebin.ca/1509295. Thanks |
| 20:53.50 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 20:54.31 | *** join/#asterisk ingenius (n=alektro@host191.190-224-106.telecom.net.ar) |
| 20:55.34 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:56.22 | DigitalDaz21 | Hi, can anyone tell me how asterisk determines a sip trunk is unavailable without first trying it? |
| 21:00.29 | mmattice | how can one get asterisk to log 'pri debug span' output? |
| 21:00.34 | grandpapadot | DigitalDaz21: There really is no concept of SIP trunks, only SIP channels. Can you ellaborate on your problem? The more information you give, the better help you'll receive from the experts in this channel. |
| 21:00.58 | ACK-NAK | Has anybody seen my keys? They were right here on the table. |
| 21:01.16 | ACK-NAK | I thougt I'd ask. You guys seem to know everything else! |
| 21:01.53 | DigitalDaz21 | yep, sure, I posted it in detail earlier but got no reply so I'm trying to work it out myself but here it is... |
| 21:02.30 | DigitalDaz21 | I am trying to get BT Business Broadband Voice working with asterisk and freepbx and have come accross the following problem. |
| 21:02.30 | DigitalDaz21 | The service uses an outbound proxy. In my case the registrar bmnha-01.bt.com is not resoveable by DNS. |
| 21:02.30 | DigitalDaz21 | The outbound proxy is www.bbvservice-560129.bt.com:5060 |
| 21:02.30 | DigitalDaz21 | If I set the trunk the way the way I believe it should be configured, it does not work. I immediately get an all circuits are busy and there appears to be no sip activity. |
| 21:02.30 | DigitalDaz21 | If I then replace bmnha-01.bt.com with sipgate.co.uk and reload, I correctly get sip traffic to the proxy but of course it fails. |
| 21:02.46 | kaldemar | mmattice: a command called script is useful for that |
| 21:02.56 | DigitalDaz21 | If I now put the original bmnha-01.bt.com back in as the host and reload, everything works perfectly as it should. |
| 21:02.56 | DigitalDaz21 | I should add that I have also tried a random string as the hostname eg hfjgktu.com and again there is no sip traffic, its almost as if I need to reload after entering a publicly resoveable DNS name to "kickstart" it first. None of this survives a restart. |
| 21:03.19 | mmattice | kaldemar: screen can log too. rather hackish |
| 21:04.04 | raden_work | does my asterisk server have to have a static IP ? |
| 21:04.39 | DigitalDaz21 | I'm using freepbx so I'm trying to initially work out why the invites are not sent, whether it is an asterisk or freepbx problem |
| 21:05.47 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:07.43 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 21:07.45 | galeras | I will apreciate any suggestion to generate a cdr record for an auto-dial out call, my callfile and extensions.conf are at http://pastebin.ca/1509295. Thanks |
| 21:08.01 | BlargMaN00 | is there a variable that tells you what current priority you are on?? |
| 21:09.05 | ManxPower | BlargMaN00: should be in channelvariables.txt in the Asterisk source. However, now that we have priority labels, the idea of priorities is largely obsolete. |
| 21:09.30 | BlargMaN00 | nevermind... i found it... |
| 21:10.12 | BlargMaN00 | ManxPower: not exactly... I have a scripting need for it... will save me several lines of dialplan code... |
| 21:10.12 | ManxPower | These days if you care about the priority number, chances are you are doing something wrong 8-| |
| 21:11.12 | *** join/#asterisk d4rkstar (n=bruno@ip-233-233.sn2.eutelia.it) |
| 21:11.31 | ACK-NAK | "dahdi show channels" doesn't show the right contexts as I would expect. 2-23 are 'default' A quick pointer would be very much appreciated. http://pastebin.com/m69f52c9d |
| 21:12.52 | ACK-NAK | Strangely 1 is correct, and 25-47 are also as I would expect. |
| 21:12.53 | kaldemar | ACK-NAK: configuration parameters must be above channel lines |
| 21:13.01 | ManxPower | ACK-NAK: you have things backwards. Set the options THEN specify the channels |
| 21:13.36 | ManxPower | a option is set until you override it. They are not reset by a group= or channels= line. |
| 21:13.37 | ACK-NAK | so in other words, param, param, param, group, channel(s)? |
| 21:13.45 | ManxPower | group is a "param" |
| 21:13.49 | timeshell | [TK]D-Fender : Evidently callerid=Name <number> doesn't set the number. I have put 4 noops in my code as indicated here: http://www.pastebin.ca/1509328 |
| 21:14.35 | timeshell | However, the callerid is defined as: http://www.pastebin.ca/1509330 |
| 21:15.39 | ManxPower | all group= does is specify what channels are used when you specify g1 or g2, etc as the channel. |
| 21:15.51 | ManxPower | it does not "group" anything in the config file. |
| 21:16.20 | ACK-NAK | In other words, like this? http://pastebin.com/m7a0799bc |
| 21:16.36 | ACK-NAK | In other words, like this? http://pastebin.com/m7a0799bc |
| 21:17.43 | ManxPower | yes, other than the fact the 2nd echocancel=yes is duplicated (it is inherited by the first instance of that option). Leave it if you want to readabity |
| 21:18.13 | ManxPower | remember once set and option applies to all channels that follow the option until you reset it. |
| 21:19.18 | *** join/#asterisk raden (n=tanning@69-179-99-17.stat.centurytel.net) |
| 21:19.20 | ACK-NAK | I see. I forgot that one. I put 'switchtype=national' for the reason you mention. |
| 21:19.24 | raden | Name/username Host Dyn Nat ACL Port Status |
| 21:19.24 | raden | 101/101 (Unspecified) D 5060 Unmonitored |
| 21:19.59 | ManxPower | raden: device [101] is not registered. If it was registered there would be an IP instead of (Unspecified) |
| 21:20.35 | raden | first time setting this up i have a aastra 9133i what would i be missing in settings ? |
| 21:20.51 | ManxPower | raden: I've never used an aastra |
| 21:29.49 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 21:36.05 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:36.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:38.07 | *** join/#asterisk sjobeck (n=Adium@32.159.144.33) |
| 21:38.22 | ScarEye | Hey guys, I am in the misdt of putting something togehter for my company for VoIP and I am thinking *asterisk*, we have like 100 stores and 1 main corporate office. I was just wondering should I throw up a asterisk box in each location or just have it in one or two locations and have the SIP phones point to the asterisk boxes? |
| 21:39.19 | raden | matters how many phones per store |
| 21:39.22 | raden | bandwith etc.... |
| 21:40.02 | ScarEye | 2 phones per store, but like 4 devices that transmit data to other terminals |
| 21:40.36 | ScarEye | like debit cards machines |
| 21:40.49 | ScarEye | only 1 call's will be taking place at a time |
| 21:40.53 | ScarEye | voice calls |
| 21:41.08 | ScarEye | I have 512KBps upload |
| 21:41.13 | ManxPower | don't expect credit card machines, fax, or other data to work with VoIP |
| 21:41.35 | scoof | work *over* VoIP |
| 21:41.54 | ScarEye | Dman I thought that problem would have been fixed now. |
| 21:41.55 | ManxPower | scoof: over, under or sideways |
| 21:42.01 | mwalling | ScarEye: what problem? |
| 21:42.02 | ScarEye | yea we get the picture |
| 21:42.05 | scoof | ManxPower: modern CC-terminals can speak IP themselves today |
| 21:42.15 | ManxPower | ScarEye: VoIP is optimized for voice, not for modem sounds. |
| 21:42.15 | scoof | at least here in Denmark |
| 21:42.20 | sjobeck | hey, hi, all, hope youre healthy, wealthy, wise. can i ask a question about customizing signalling at the zaptel/dahdi level? i need to send & receive "winks" to a channelbank over E&M. |
| 21:42.35 | ScarEye | ManxPower: your right, but I thought we would have able to do so. |
| 21:42.38 | ScarEye | by now |
| 21:42.43 | mwalling | ScarEye: you realize you're encoding digial signals as an analog signal, then compressing that, then decompressing that and turning it back into digital? |
| 21:42.48 | ScarEye | cause I tried it like 4-5 years ago and it didn't work |
| 21:43.02 | ManxPower | heck, MUSIC doesn't work well over compressed codecs |
| 21:43.03 | ScarEye | mwalling: point |
| 21:43.27 | mwalling | why not just send it digital all the way? |
| 21:43.32 | mwalling | or at least as far as you can |
| 21:43.44 | ScarEye | mwalling: Don't have that option atiquated equipment |
| 21:43.45 | ManxPower | mwalling: because IP CC machines are expensive |
| 21:43.52 | ScarEye | Yea, my spelling sucks |
| 21:44.02 | ScarEye | and that too. |
| 21:44.08 | mwalling | ManxPower: Wendys has POTS CC swipers |
| 21:44.15 | mwalling | er, POS |
| 21:44.21 | mwalling | wrong PO.* acronym |
| 21:44.34 | mwalling | wow, nevermind, i am failing too many ways |
| 21:44.55 | ScarEye | so bascially all these CC type terminals we will need POTS |
| 21:45.09 | ManxPower | ScarEye: they are all basically modems. |
| 21:45.22 | ScarEye | yea that I understand |
| 21:46.55 | ScarEye | okay, so, if I have 2 phones per location (and I have enough b/w) should I just setup one asterisk server? and have all these SIP phones point to this one astersk server? |
| 21:47.02 | ScarEye | or should I throw one up in each location |
| 21:47.30 | ScarEye | and have like to VoIP carriers just in case one VoIP company goes belly up |
| 21:47.31 | mmattice | ScarEye: your customers would love you if you could transition to CC auth over IP |
| 21:47.32 | *** join/#asterisk s14ck (n=s14ck@190-76-100-84.dyn.movilnet.com.ve) |
| 21:47.42 | ScarEye | mmattice: My company would love me |
| 21:47.48 | ScarEye | shit I might get a raise |
| 21:47.49 | ScarEye | lol |
| 21:47.54 | ScarEye | sorry for my languauge |
| 21:48.47 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:48.47 | mmattice | it _is_ doable if your clearinghouse is capable. |
| 21:48.57 | *** part/#asterisk sjobeck (n=Adium@32.159.144.33) |
| 21:49.40 | vicscandl | how "bleeding edge" is 1.6 vs 1.4? |
| 21:50.10 | raden | asterisk server ip would be my sip server that i would program my ip phone to right ? |
| 21:50.38 | eric_hill | vicscandl: I had some asterisk crashdumps with 1.6.0.6, but 1.6.0.10 has (for the month or two) been 100% stable. |
| 21:50.55 | vicscandl | eric_hill: thanks |
| 21:51.28 | mwalling | at my old resturuant, our processor gave us a discount on the fee if we switched to ip, and it almost paid for the internet (we added stuff like online ordering too) |
| 21:52.33 | raden | Name/username Host Dyn Nat ACL Port Status |
| 21:52.33 | raden | 101/101 (Unspecified) D 5060 Unmonitored |
| 21:52.39 | ACK-NAK | ManxPower: Thanks for the help earlier. Any ideas as to why the channel 1 context would be different than 2-23? Channel 1 is the same as the context set for 25-47. http://pastebin.com/m1b58be5b |
| 21:52.45 | raden | why would the host be unspecified |
| 21:53.04 | ACK-NAK | ManxPower: dahdi show channels: http://pastebin.com/m2b074341 |
| 21:53.56 | ACK-NAK | ManxPower: I tried swapping the order of group and channel, and it didn't change anything. |
| 21:56.00 | ManxPower | ACK-NAK: unload chan_dahdi.so and then load chan_dahdi.so A few of the option won't update on just a plain reload. |
| 21:56.48 | ManxPower | ACK-NAK: What did I tell you about options. If you set an option AFTER the channel= line then it won't apply to the channel line |
| 21:57.02 | ManxPower | group is an option just like any other. |
| 21:58.14 | ManxPower | "group = 2" does nothing at all since it is below any channel= lines. |
| 21:59.01 | ManxPower | bbiab |
| 21:59.57 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:00.05 | explody | anyone have a recommendation on rhino vs. openvox? |
| 22:00.29 | ACK-NAK | ManxPower: I see. I changed it only after it didn't 'take' clutching at straws :-) |
| 22:00.52 | *** join/#asterisk yziquel (i=55da623a@gateway/web/freenode/x-550027a92384eab8) |
| 22:01.44 | ACK-NAK | ManxPower: Am I correct in assuming the file is parsed from the bottom up? |
| 22:07.46 | mmattice | I guess I shouldn't be too concerned about this: "asterisk[4356]: rc_avpair_new: unknown attribute 1490026597" |
| 22:07.49 | explody | how about hardware echo cancellation vs. none or software... if this would be running all of a company's voice traffic is it at all worthwhile to try without hardware or is it really just necessary? |
| 22:13.19 | raden | Name/username Host Dyn Nat ACL Port Status |
| 22:13.19 | raden | 101/101 (Unspecified) D 5060 Unmonitored |
| 22:13.19 | raden | callcentric/17772445766 204.11.192.36 5080 Unmonitored |
| 22:13.32 | raden | what exactly does host unspecified mean ????? how can i fix it ? |
| 22:13.41 | vicscandl | Qwell: good location for asterisk RPM's? |
| 22:14.14 | vicscandl | ~rpm |
| 22:14.15 | infobot | Red Hat's package management system. URL: http://www.rpm.org/ |
| 22:14.20 | vicscandl | <= idiot |
| 22:16.56 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:17.59 | Qwell | vicscandl: packages.asterisk.org, for AsteriskNOW RPMs |
| 22:18.23 | Qwell | http://packages.asterisk.org/centos/5/current/x86_64/RPMS/asterisknow-version-1.5.0-1_centos5.noarch.rpm |
| 22:18.23 | vicscandl | Qwell: thanks, as always |
| 22:18.29 | Qwell | install that and just yum install what you need |
| 22:18.46 | vicscandl | Qwell: thanks, wanted a good source for my RPM's.. :) |
| 22:19.37 | raden | :( i have a dial tone cant dial out cant dial in anyone ???? |
| 22:26.16 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 22:26.37 | grandpapadot | raden: What does asterisk say? |
| 22:26.45 | raden | as far as ? |
| 22:26.56 | raden | I have a dial tone on my phone i can dial but nothing happens |
| 22:27.11 | raden | my provider in the log is not showing anything being attempted to dial |
| 22:27.44 | ManxPower | raden: ip phones provide their own dialtone, not the provider |
| 22:28.01 | ManxPower | they also collect the digits then send them all at once to the provider |
| 22:28.11 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 22:28.29 | raden | so how do i know the digits are getting from my phone to asterisk ? |
| 22:28.44 | ManxPower | raden: you don't unless you look at Asterisk's sip debug. |
| 22:28.55 | ManxPower | or unless you see something on the Asterisk console |
| 22:29.11 | mmattice | anybody know tips or tricks to getting dtmf to be reliable over a PRI? |
| 22:29.18 | ManxPower | You can see how important it is for you to know how to configure your phone. |
| 22:29.49 | ManxPower | mmattice: receiving DTMF from calls into the PRI or sending DTMF out on calls going out the PRI? |
| 22:29.59 | mmattice | ManxPower: receiving |
| 22:30.24 | ManxPower | mmattice: the number one thing that impacts that is rxgain in the zaptel/dahdi config |
| 22:30.24 | *** join/#asterisk pazof (i=paul@reverse-81.fdn.fr) |
| 22:30.34 | mmattice | half the time it's not picking up digits in our IVR |
| 22:31.12 | ManxPower | if rxgain is too high, you can get distorted DTMF received. Too low and asterisk may detect a single digit as 2 digits. Also turn off relaxdtmf if it is set. It usually causes more issues than it fixes. |
| 22:32.31 | ManxPower | I've been using Asterisk since late 2001, using PRIs with Asterisk since early 2003. |
| 22:34.05 | mmattice | ah, so I probably need to go and tune rx and tx gain? |
| 22:34.36 | drmessano | thinks of asterisk as a big PRI <> GoogleTalk adapter |
| 22:34.44 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:35.07 | ManxPower | mmattice: Yes. Getting the gains right can be hard to do. If they are out of whack, you get echo, dtmf issues, and fax problems. |
| 22:35.38 | raden | how can i tell if asterisk is getting what im dialing on my phone ? |
| 22:36.04 | ManxPower | raden: "sip debug on" |
| 22:36.45 | raden | where do i put that ? |
| 22:36.55 | ManxPower | mmattice: start out with rxgain=0 and txgain=0, then work from there. (I'd start by doing rxgain=2) |
| 22:37.01 | ManxPower | raden: in the Asterisk console. |
| 22:37.10 | raden | command not found |
| 22:37.17 | ManxPower | then you are not in the Asterisk console. |
| 22:37.35 | ManxPower | "asterisk -rvvv" gets you connected to the existing Asterisk process |
| 22:38.00 | ManxPower | You really need to read the Asterisk book. |
| 22:38.09 | ManxPower | ~book |
| 22:38.10 | 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 |
| 22:38.59 | raden | sip set debug on |
| 22:39.33 | ManxPower | the format of the command is different between asterisk versions |
| 22:39.47 | raden | i have the first version and the second one isnt much help either |
| 22:39.59 | raden | technology bla |
| 22:40.00 | ManxPower | you have 0.65? |
| 22:40.10 | raden | no the book |
| 22:40.16 | raden | running 1.6.0.10 |
| 22:40.28 | ManxPower | Ah. the book should at least help you understand how to get into the console. |
| 22:40.35 | raden | im in console |
| 22:40.41 | raden | i can use console |
| 22:40.56 | raden | can i send output to a file ? |
| 22:41.49 | ACK-NAK | ManxPower: Thanks for your suggestions. In order to have channel 1 appear in the correct context (the same context as 2-23 instead of same as 25-47), I had to reverse the order of channels/groups. from this: http://pastebin.com/m4ca48539 to this: http://pastebin.com/m7d72dd83. Why would that be the case? |
| 22:41.54 | ManxPower | <PROTECTED> |
| 22:42.21 | ACK-NAK | ManxPower - I gave you those pastebins backwards. |
| 22:42.48 | ACK-NAK | I should have said from this (wrong) http://pastebin.com/m7d72dd83, to this (right) http://pastebin.com/m4ca48539 |
| 22:43.12 | ManxPower | ACK-NAK: I suspect you can't change the context without restarting Asterisk (or at least chan_dahdi). |
| 22:43.48 | ACK-NAK | I actually stopped asterisk and dahdi, and restarted both. |
| 22:43.57 | raden | how can i set debug just for 192.168.1.101 ? |
| 22:44.44 | ManxPower | raden: you can't unless the device is registered. |
| 22:44.47 | ACK-NAK | ManxPower: so it appeared that I was able to change the context just as you suggested by unloading/reloading, but the first channel was always in the wrong context until I reversed the appearance in chan_dahdi.conf. and I'm puzzled by that. |
| 22:45.00 | ManxPower | in that case its something like sip set debug peer blah.bla.bla.bla |
| 22:45.39 | ManxPower | ACK-NAK: what channels do you want in which contexts? |
| 22:46.12 | raden | ManxPower, registered howso ? |
| 22:46.12 | ACK-NAK | ManxPower: 1-23 in inbound-pri, 25-47 outbound-pri. |
| 22:47.57 | ManxPower | ACK-NAK: you are missing the [trunkgroups] section at the top of the file. I bet that is really confuzzling the config parser |
| 22:48.41 | ManxPower | raden: If an IP phone is on a dynamic IP address the phone must tell Asterisk what IP it is on. This is done using a process called "registration". I'm sorry, I cannot help you further. |
| 22:49.11 | raden | Name/username Host Dyn Nat ACL Port Status |
| 22:49.11 | raden | 101/101 192.168.1.101 5060 Unmonitored |
| 22:49.11 | raden | callcentric/17772445766 204.11.192.37 5080 Unmonitored |
| 22:49.19 | ManxPower | there, it is registered. |
| 22:49.19 | raden | seems like no one can ever help :( |
| 22:49.44 | ManxPower | raden: you are asking such basic questions nobody really wants to spend the time tutoring you. |
| 22:49.56 | ACK-NAK | ManxPower: We're not doing NFAS, do I still need it? |
| 22:50.03 | ManxPower | ACK-NAK: yes, leave it empty |
| 22:51.06 | ManxPower | unload and load chan_dahdi.so after you fix the config. I've heard leaving out a required config section causes cancer and can tear a hole in space/time. |
| 22:51.27 | ACK-NAK | ManxPower: I'll try putting back [trunkgroups] empty & re-reverse the order of the channel parameters, to see if it makes the order irrelavent. |
| 22:52.00 | ManxPower | ACK-NAK: What you are seeing does not make any sense at all, so I suspect the lack of [trunkgroups] is the problem. |
| 22:52.46 | *** part/#asterisk eric256 (n=Administ@c-67-165-208-191.hsd1.co.comcast.net) |
| 22:52.51 | ManxPower | raden: With Asterisk you really need to know Networking, Telecom, SIP, Asterisk, Linux. It's not an easy thing to learn, so don't be surprised if you have a steep learning curve. |
| 22:54.01 | raden | im just trying to get 1 phone working with asterisk lol networking and linux easy asterisk install was easy the configuration seems straight forward but i cant tell if its the phone or asterisk where my config is wrong |
| 22:54.17 | raden | well im going to order the 2nd edition of that book |
| 22:54.25 | ACK-NAK | ManxPower: I agree. It makes no sense, but after a quick test, the order appears to matter with or without [trunkgroups]. Why is there a channel called pseudo? |
| 22:54.31 | ManxPower | raden: getting one phone working is almost as hard as getting 200 phones working. |
| 22:54.38 | ManxPower | ACK-NAK: pseudo is used for conferencing |
| 22:54.47 | raden | you for real ? |
| 22:55.04 | ManxPower | raden: yup. Once you learn to set up one phone, you know how to setup the other 199 phones. |
| 22:55.33 | ManxPower | The DIALPLAN (extensions.conf) is more complex with 200 phones, but the basic sip.conf setup is very similar between phones. |
| 22:56.15 | ACK-NAK | ManxPower. I see. is this file parsed from the top-down or from the bottom-up? |
| 22:56.54 | ManxPower | ACK-NAK: think of it as bottom-up. That's not technically true, but thinking of it that way helps people understand it. |
| 22:57.36 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:59.08 | ManxPower | It might look like a windows .ini file, but it's not parsed that way |
| 23:01.25 | ManxPower | raden: just download the 2nd edition |
| 23:01.44 | ManxPower | if it helps, buy the actual book (so all the cool people get a few cents) |
| 23:10.39 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 23:19.29 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 23:33.21 | ACK-NAK | ManxPower: Thanks for your help! |
| 23:42.55 | raden | what does status unomonitored mean ? |
| 23:44.21 | raden | anyone ever configured aastra phones for asterisk ? |
| 23:44.41 | *** join/#asterisk Alfio (n=Amunoz@190.94.58.24) |
| 23:47.45 | ManxPower | unmonitored means you do not have qualify=yes. |
| 23:53.21 | *** join/#asterisk iewebguy (n=mark@65.19.81.253) |
| 23:57.01 | *** part/#asterisk vicscandl (n=vicscand@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |