| 00:00.05 | [TK]D-Fender | asdx, On both ends? |
| 00:00.10 | asdx | [TK]D-Fender: yeh |
| 00:00.20 | jero | How can I tell asterisk to notify presence-enabled phones when one of them places an outgoing call ? |
| 00:00.24 | [TK]D-Fender | asdx, then fix your headsets & soudcard gains. |
| 00:00.33 | [TK]D-Fender | asdx, because thats accoustic feedback. |
| 00:00.46 | asdx | [TK]D-Fender: ok, thanks |
| 00:00.49 | katsuodo | [TK]D-Fender almost forgot will use tdm805 card for office with (2) company |
| 00:01.05 | [TK]D-Fender | jero, make sure your phones are "type=peer", "call-limit=99" |
| 00:01.27 | jero | [TK]D-Fender: they are peers and call-limit=4 |
| 00:02.03 | jero | [TK]D-Fender: and limitonpeers=yes |
| 00:02.07 | [TK]D-Fender | jero, pastebin "show channels concise" and "show hints" |
| 00:02.11 | jero | k |
| 00:02.14 | [TK]D-Fender | jero, remove that last one |
| 00:03.33 | jero | oh |
| 00:03.55 | jero | let me place an outgoing call |
| 00:04.03 | [TK]D-Fender | jero, and up to 99 the other one |
| 00:04.14 | jero | okay |
| 00:09.21 | jero | [TK]D-Fender: http://pastebin.ca/778868 |
| 00:12.11 | katsuodo | [TK]D-Fender simultaneous call = 20 |
| 00:14.00 | weazahl | katsuodo: can so few stations make so many calls. you use 5 analogue lines. so how many simultanious calls do you think you make now? |
| 00:14.25 | [TK]D-Fender | jero, pastebin your dialplan.... |
| 00:14.30 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 00:14.50 | [TK]D-Fender | katsuodo, that measn you expect everyone to be on the phone 100% of the time... |
| 00:14.52 | weazahl | unless you are doing predictive dialing. you will probably never get close to thay |
| 00:15.36 | weazahl | i have 32 stations and never say more than a load of 8 |
| 00:15.45 | [TK]D-Fender | jero, and your sip.conf entry |
| 00:23.20 | *** join/#asterisk Giofe (n=chatzill@cliente37.amx.com.pe) |
| 00:23.41 | Giofe | :) |
| 00:25.31 | jero | [TK]D-Fender: here they are http://pastebin.ca/778876 |
| 00:25.50 | *** join/#asterisk kotyagin (n=knkbox@ppp85-140-239-38.pppoe.mtu-net.ru) |
| 00:26.26 | asdx | i'm using gsm now, can i increase the bitrate? |
| 00:26.37 | [TK]D-Fender | asdx, nope |
| 00:26.45 | [TK]D-Fender | asdx, pick another codec |
| 00:27.21 | *** join/#asterisk implicit (n=implicit@207.181.11.96) |
| 00:27.54 | weazahl | i switched ulaw to gsm without telling anyone. then asked if anyone noticed any changes in voice quality a few days later. nope. |
| 00:28.43 | weazahl | most people are DEAF, point and case, stock car stereo systems. people turn them up to like 80%THD and think that is cool. |
| 00:29.52 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 00:35.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:38.29 | kotyagin | you damn right !!! |
| 00:39.25 | weazahl | i know. that is why i have 1500W RMS in my car, it does go loud. but at reasonable levels, it sounds unbelievable |
| 00:39.35 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 00:41.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:42.04 | [TK]D-Fender | BBIAB |
| 00:42.13 | *** join/#asterisk exothermic (n=miles@68-189-133-163.dhcp.wlwl.wa.charter.com) |
| 00:42.59 | kotyagin | most companies in Russia use g.729 in their VoIP networks... And they happy with quality... |
| 00:43.31 | exothermic | We are having trouble with fonality boxes connecting to our asterisk server. The DTMF tones are not getting to fonality. All of our other peers work fine though. |
| 00:43.59 | exothermic | In turn the other peers on the fonality side work as well. |
| 00:44.08 | exothermic | Everything is set to use rfc2833 |
| 00:44.18 | exothermic | Anyone have any idea what is going on? |
| 00:44.33 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 00:45.54 | kotyagin | Did you try to use SIP INFO, to transmit DTMF ? |
| 00:47.07 | exothermic | No I did not. |
| 00:47.08 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:49.16 | exothermic | how would you relay that with asterisk? |
| 00:49.28 | kotyagin | imo, this is the best way to send dtmf... try to use dtmf=info |
| 00:49.41 | kotyagin | sorry |
| 00:49.43 | exothermic | ahh ok let me try that. |
| 00:49.47 | kotyagin | dtmfmode=info |
| 00:49.52 | kotyagin | on both sides |
| 00:50.08 | exothermic | most devices support this? |
| 00:50.19 | exothermic | and carriers? |
| 00:51.42 | kotyagin | not all devices support this feature, but i think you should try |
| 00:52.16 | *** part/#asterisk cyberpass2 (n=mataz@ppp-64-219-79-16.dsl.hstntx.swbell.net) |
| 00:52.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:53.06 | exothermic | looks like it works |
| 00:53.13 | exothermic | Wow that is the oddest thing |
| 00:53.36 | exothermic | we have one version of asterisk that can send dtmf via 2833 to that fonality box and one that can't |
| 00:53.56 | exothermic | 1.4.11 fails. |
| 00:53.57 | Qwell | exothermic: try calling Fonality. We can't know what they did to butcher the configs |
| 00:54.28 | Qwell | if they even support it anymore |
| 00:54.46 | Qwell | they change direction every year |
| 00:54.48 | exothermic | Ya think they will support me since it isn't even my box ;)? |
| 00:55.02 | Qwell | who knows |
| 00:55.07 | exothermic | 1.4.2 work with the dtmf. |
| 00:55.38 | Qwell | they use something stupid like 1.0.9 on their boxes |
| 00:55.44 | Qwell | so, it doesn't surprise me that dtmf doesn't work |
| 00:56.12 | kotyagin | burn your fonality ^) |
| 00:56.17 | exothermic | Well it works on what seems to be everything except my 1.4.11 setup. |
| 00:56.29 | Qwell | try 1.4.13 - 1.4.11 may have had a bug |
| 00:56.34 | Qwell | erm, 1.4.14 |
| 00:56.34 | exothermic | ya it is just a clients setup. |
| 00:56.59 | exothermic | Well I have no other peers that have dtmf issues with this 1.4.11 setup. |
| 00:57.07 | exothermic | which is why it is odd. |
| 00:57.22 | exothermic | Looking back through change logs to see if there is anything done to dtmf |
| 00:57.30 | exothermic | Just is odd all around. |
| 00:57.30 | Qwell | there has been, iirc |
| 00:57.34 | asdx | G711 (alaw and ulaw) |
| 00:57.37 | asdx | are those codecs free |
| 00:57.39 | asdx | ? |
| 00:57.42 | exothermic | asdx: yes |
| 00:57.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:58.00 | Qwell | asdx: everything supports at least one of those |
| 00:58.23 | exothermic | Qwell: That statement might be a little broad. |
| 00:58.34 | kotyagin | asdx: most often both... |
| 00:58.36 | Qwell | nah, it's pretty true |
| 00:58.37 | asdx | zoiper says UNKNOWN when i try to use that |
| 00:59.45 | asdx | but they say G.711 is supported |
| 01:00.07 | exothermic | u or a |
| 01:00.28 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 01:00.28 | asdx | u/a |
| 01:00.42 | kotyagin | asdx: try to dump sip trace\ |
| 01:01.05 | kotyagin | asdx: especially SDP offer from invite from zoiper |
| 01:01.22 | asdx | kotyagin: i'm using iax2 |
| 01:02.15 | *** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 01:02.31 | mackes | hey |
| 01:02.35 | kotyagin | asdx: mmm, i think codec list will we same in both SIP and IAX, but SIP trace much more easer to read |
| 01:02.49 | asdx | kotyagin: ok, thanks :) |
| 01:03.03 | mackes | how do you perform a SIP trace? sniff the wire? |
| 01:03.41 | b_d | :) |
| 01:03.50 | kotyagin | mackes: something like tcpdump or wireshark at PBX side, or at switch with port mirroring |
| 01:04.10 | mackes | OK, thats what I thought |
| 01:04.28 | mackes | I wasnt sure if Asterisk had a tool that showed debug |
| 01:04.39 | *** join/#asterisk shido6 (i=shido6@74-130-126-198.dhcp.insightbb.com) |
| 01:04.44 | b_d | you will still have to have a protocol analyzer attached to the spanned port on the switch |
| 01:04.53 | kotyagin | mackes: sip set debug |
| 01:05.56 | mackes | So, Wireshark installed on your Asterisk server |
| 01:06.03 | mackes | sip debug? |
| 01:06.14 | kotyagin | b_d: wireshark works fine |
| 01:06.46 | mackes | Ahh, thats the stuff |
| 01:06.55 | mackes | I forgot about that command |
| 01:07.33 | mackes | I know that Snom will show the same debug on the other side of the conversation, Will Polycom or Aastra? |
| 01:08.11 | kotyagin | mackes: dunno :( |
| 01:09.32 | kotyagin | mackes: defenetely the most powerful thing is wireshark at asterisk, then ssh -X and we don't need any port mirroring... |
| 01:11.24 | b_d | wireshark IS a protocol analyzer |
| 01:11.52 | mackes | ssh -X, how does that help? |
| 01:14.21 | kotyagin | mackes: with ssh -X host you can run wireshark(and other X applications) from remote so it helps a lot :) |
| 01:15.22 | *** part/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca) |
| 01:16.45 | mackes | ssh -X will allow you to excute an appication on a remote machine that is installed on your local machine-- right? |
| 01:16.49 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:17.22 | b_d | kotyagin, mackes: that is assuming you have X11 running on your pbx, in the absence of X11 you may just have to use something like tshark supplied with the proper arguments to filter out unwanted chatter |
| 01:19.04 | mackes | Ahh... yep. I use poundkey.. no X |
| 01:20.30 | *** join/#asterisk bintut (n=chatzill@cm246.gamma178.maxonline.com.sg) |
| 01:20.36 | mackes | I find an old laptop with a NIC, and a hub, and ethereal placed infront of a machine does the trick |
| 01:21.30 | b_d | mackes: crude, but effective |
| 01:22.36 | mackes | Or a mirrored port on a good switch does nicely as well |
| 01:22.36 | bintut | i am trying to connect 2 asterisk server using iax2 but i can't make them connect. do i need to create ssl certificate for iax2 first in order for them to register each other? |
| 01:22.44 | mackes | no |
| 01:22.56 | mackes | Just an account on each for them to connect with |
| 01:23.29 | mackes | I have trouble getting my caller idea to carry correctly between systems with iax |
| 01:25.39 | *** part/#asterisk b_d (n=brian@209.240.42.151) |
| 01:35.57 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 01:43.16 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 01:43.50 | mackes | quiet all of the sudden |
| 01:44.05 | mackes | I think grandstream rockS! |
| 01:44.23 | mackes | (Chum in the water) |
| 01:44.51 | mackes | #help |
| 01:45.50 | bintut | anyone here can point me to the right url where i can read the documentation about the iax.conf and if they are similar with the sip.conf configuration? |
| 01:48.38 | BBHoss | bintut: i think there are some examples in the book |
| 01:50.18 | bintut | BBHoss: actually, i have a copy of the Asterisk: The Future of Telephony 2/e already.. i followed the book on how to connect 2 asterisk servers but with no luck |
| 01:50.34 | BBHoss | with iax2? |
| 01:50.58 | bintut | BBHoss: my current connection with the remote asterisk box is within an openvpn tunnel |
| 01:51.03 | bintut | BBHoss: yes, using iax2 |
| 01:52.01 | bintut | BBHoss: do you think that the iax.conf and the sip.conf have a similar directives? |
| 01:52.14 | BBHoss | the way i do it is setup a peer and a user on each box, then connect them together |
| 01:52.36 | BBHoss | bintut: similar, but not really more than that |
| 01:54.07 | bintut | BBHoss: yeah, that's the same thing that was said in the book.. but, it doesn't say the externip directive |
| 01:54.25 | BBHoss | bintut: so you've followed the guide starting on page 111 to a T |
| 01:54.41 | bintut | BBHoss: yes |
| 01:55.11 | BBHoss | i don't believe externIP is required |
| 01:55.13 | bintut | BBHoss: although, i found that there's a typo error.. |
| 01:55.20 | BBHoss | you just have to make sure that 4569 is forwarded to the * box on both ends |
| 01:55.46 | bintut | BBHoss: remember that i am connecting to the remote asterisk box through openvpn |
| 01:56.06 | BBHoss | is there a reason its going over VPN? |
| 01:56.06 | bintut | BBHoss: and with that, my ip address as well as on the other side is different already |
| 01:56.27 | bintut | BBHoss: just to tunnel it |
| 01:56.41 | BBHoss | why not run it over the net? security? |
| 01:56.55 | bintut | BBHoss: i'll try adding an externip directive on my iax.conf first |
| 01:57.05 | BBHoss | ok w/e |
| 01:57.35 | BBHoss | even though its sip onlky |
| 01:57.53 | BBHoss | brb |
| 02:00.41 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 02:01.39 | BBHoss | bintut: look here also: http://astrecipes.net/index.php?n=204 |
| 02:03.16 | *** join/#asterisk __freedom__lover (n=eduardo@201-26-101-47.dsl.telesp.net.br) |
| 02:03.30 | bintut | BBHoss: thanks.. but that config is almost the same with mine |
| 02:05.35 | *** join/#asterisk DarkDlx (n=darkdll@21.pool85-53-207.dynamic.orange.es) |
| 02:06.25 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 02:06.28 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:06.29 | *** join/#asterisk asdx (n=diego@adsl-149-212.click.com.py) |
| 02:13.49 | DarkRift | I've seen once a websiet that had the asterisk soruce code online with the explanation of all it's functions and in what file they are, anyone know what I'm talking about ? I can't seem to find it anymore |
| 02:15.15 | Qwell | DarkRift: asking the same question in multiple places is a bit rude |
| 02:15.50 | Qwell | DarkRift: make progdoc, from the asterisk source dir |
| 02:15.53 | Qwell | progdocs |
| 02:15.57 | DarkRift | I agree, but not everyone is on the same chan |
| 02:16.08 | Qwell | that's exactly why it's rude |
| 02:16.48 | DarkRift | Just trying to get information from most people I can |
| 02:17.00 | DarkRift | Hummm it was a website more than the progodocs, but let me check that out |
| 02:18.10 | DarkRift | It had all the function on the left side, and on the right side the commands, their parameters and a explanation of what they do |
| 02:18.20 | DarkRift | generating the progdocs tho atm |
| 02:25.55 | DarkRift | There it is, doxygen trunk |
| 02:30.48 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 02:30.58 | UserReg_CL | HI!!! Good day!! |
| 02:34.02 | UserReg_CL | (Hola, a todos) |
| 02:34.18 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:34.40 | asdx | UserReg_CL: hola |
| 02:35.48 | UserReg_CL | asdx: holas... mejor no hablamos en español pues algunos comienzan a "gruñir" jjajaj "lol" |
| 02:37.35 | *** join/#asterisk ManxPower (n=manxpowe@36.sub-70-197-245.myvzw.com) |
| 02:46.19 | *** join/#asterisk angom_h (n=Angel@201.170.35.218) |
| 02:46.26 | UserReg_CL | Helpme !!! |
| 02:46.35 | UserReg_CL | "need asign x minutes a one user sip" ¿how? |
| 02:47.06 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:49.29 | ManxPower | UserReg_CL: there is no easy way |
| 02:51.08 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 02:52.18 | UserReg_CL | ManxPower: |
| 02:56.47 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 02:58.35 | *** join/#asterisk [zender] (n=ender@72.52.23.4) |
| 03:00.31 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-179.dynamic.qsc.de) |
| 03:01.04 | *** join/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 03:01.17 | dongs | what do i add in sip.conf to allow receive of calls to anything@mysipproxyip? |
| 03:01.28 | dongs | like incoming context or something. |
| 03:02.01 | phix | hey |
| 03:03.06 | *** join/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:05.15 | katsuodo | [TK]D-Fender and weazhl it is multinational fashion company and no I do not expect them all to be on phone all the time. I asked about call volume and they can give no answer they do not know |
| 03:06.24 | coppice | if I aska fashion company about call volume I'd expect them to say "really really loud" |
| 03:08.20 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:08.38 | *** join/#asterisk Raky-2 (n=rakaveli@d220-237-207-199.dsl.nsw.optusnet.com.au) |
| 03:08.58 | Raky-2 | hi guys, i was wondering if it's possible to bind an m3u stream, like icecast to an extension? |
| 03:09.09 | Raky-2 | so let's say, someone dials 100, they will hear the web stream. |
| 03:09.13 | Raky-2 | is that possible? |
| 03:10.28 | [N00B] | i am in need of some help resolving an issue. I have been trying for nearly 6 weeks to resolve it. I could really use some help: |
| 03:10.28 | [N00B] | here is the error: |
| 03:10.28 | [N00B] | WARNING[11598] res_monitor.c: Execute of ( nice -n 19 soxmix "//dev/shm/1194888906.22392-in.wav" "//dev/shm/11948 |
| 03:10.35 | [N00B] | I have verified that sox is installed |
| 03:10.53 | [N00B] | i ran: |
| 03:10.55 | [N00B] | which sox |
| 03:10.55 | [N00B] | /usr/bin/sox |
| 03:11.05 | [N00B] | so sox is there |
| 03:11.12 | UserReg_CL | helpme: need asign x minutes a one user sip ¿? |
| 03:12.39 | [N00B] | here is the file details of where i am trying to send the file to: |
| 03:12.40 | [N00B] | <PROTECTED> |
| 03:12.40 | [N00B] | lrwxrwxrwx 1 root root 9 Oct 30 15:36 monitor -> /dev/shm/ |
| 03:13.17 | [N00B] | <PROTECTED> |
| 03:13.18 | [N00B] | drwxrwxrwt 3 root root 80 Nov 17 20:10 shm |
| 03:13.37 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 03:14.14 | [N00B] | could it possibly have to do with the "//dev" in the error message? |
| 03:14.16 | katsuodo | coppice I know understand |
| 03:14.35 | [N00B] | i dont know how that would have happened. my symlink is to /dev/shm |
| 03:15.23 | [N00B] | Raky-2: it is possible. There is a link somewhere on the wiki regarding how to do that. |
| 03:16.18 | [N00B] | Raky-2: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES |
| 03:16.19 | Raky-2 | i've only seem to have found links that say how to broadcast from an asterisk extension |
| 03:16.40 | Raky-2 | i just want to be able to feed it an m3u, and have it play back to the user. |
| 03:16.54 | [N00B] | user= phone? |
| 03:17.13 | Raky-2 | well, something like this |
| 03:17.30 | Raky-2 | exten => 655,3,SetMusicOnHold(stream) |
| 03:17.48 | Raky-2 | where stream in musiconhold.conf = stream => quietmp3:/var/lib/asterisk/moh/stream,http://partydome.us:8000/liljon.ogg.m3u |
| 03:17.54 | [N00B] | then look at the link i added. the instructions even include an example on how to do what you are looking for |
| 03:18.40 | dongs | lol, ogg. |
| 03:18.41 | *** part/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 03:18.42 | dongs | what the fuck. |
| 03:18.55 | dongs | i guess the only possible excuse is that since its 8khz mono it doesnt matter. |
| 03:19.05 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 03:19.26 | *** join/#asterisk Miamiman302 (n=pirch@c-76-109-187-45.hsd1.fl.comcast.net) |
| 03:19.37 | *** part/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 03:20.08 | Raky-2 | thanks i'll give it a shot |
| 03:33.39 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 03:37.08 | UserReg_CL | mmm |
| 03:37.09 | asdx | if i want to make a call to a PSTN line i just do: extern => _test,1,Dial(SIP/NUMBER) right? |
| 03:37.52 | phix | hey, AMR NB is am audio codec right? and does asterisk support it? |
| 03:38.49 | phix | My issue is I only have ulaw or alaw available on my SIP ip phone (Nokia E65). It support AMR NB too but I have nfi what that codec is. |
| 03:39.38 | phix | Believe it or not but the Nokia E65 doesn't use GSM for some reason, even though it is my preferred codec in asterisk |
| 03:39.44 | coppice | AMR NB is the main codec used for 3G phones, and is growing in use for GSM |
| 03:39.51 | UserReg_CL | asdx: nop extern => _test,1,Dial(canal/${EXTEN}) |
| 03:39.52 | phix | ok thank you |
| 03:40.02 | phix | coppice: :) I knew it had something to do with 3G |
| 03:40.24 | phix | coppice: any advice? or havn't you played around with mobile SIP phones/ |
| 03:40.45 | coppice | the codec things like asterisk loosely call GSM is rarely used by GSM networks these days |
| 03:40.57 | phix | I mean ulaw works great in my house, I just figure if I used a compressed codec I could get a bit more range out of the phone |
| 03:41.34 | phix | currently I get about 20 Mtrs range |
| 03:41.51 | coppice | most GSM phones that do VoIP are set up for UMA environments. AMR is the codec of choice there |
| 03:41.52 | phix | (not direct line of sight, through a brick wall :)) |
| 03:54.29 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 03:56.56 | mackes | na,,, GSM wont help with range. |
| 03:57.08 | mackes | Your AP will not drop below 1Mb |
| 03:57.31 | mackes | So, ULaw will be just fine |
| 04:01.10 | *** part/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp) |
| 04:04.18 | coppice | a quick google says the E65 supports iLBC and G.729 |
| 04:04.49 | *** part/#asterisk stubert (n=stu@techtools.actusa.net) |
| 04:06.03 | asdx | i'm trying to dial a pstn number and i get: == Everyone is busy/congested at this time (1:0/0/1) |
| 04:13.06 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 04:13.07 | weazahl | did anyone know that external hard disk are great for raising bread |
| 04:15.29 | wothinn | Neat idea. I prefer to do a slow rise, though. :-) |
| 04:22.16 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 04:23.09 | [N00B] | asdx: did you have a question to go with that? |
| 04:23.24 | asdx | [N00B]: yeh it was a question |
| 04:23.26 | UserReg_CL | HI: what distro linux install for asterisk recommend ? |
| 04:23.38 | [N00B] | that was a statement |
| 04:23.43 | [N00B] | what is your question |
| 04:23.52 | asdx | [N00B]: yeah |
| 04:24.00 | asdx | [N00B]: well, i'm trying to dial a PSTN phone number... |
| 04:24.06 | asdx | [N00B]: and i get that, why? |
| 04:24.20 | [N00B] | you need to provide more information. |
| 04:24.28 | asdx | right, let me paste my configs |
| 04:24.47 | [N00B] | also describe your setup and how the call is being made |
| 04:25.02 | [N00B] | you using T1? PRI? POTS? |
| 04:25.08 | [N00B] | gotta give us the info |
| 04:25.10 | *** join/#asterisk ming_zym (n=ming_zym@124.14.235.101) |
| 04:25.17 | asdx | [N00B]: i don't have special hardware, just a voip provider (Teliax) |
| 04:25.33 | [N00B] | voip=IAX2? SIP? what? |
| 04:26.54 | asdx | [N00B]: i have my linux desktop with a softphone (zoiper), my asterisk box is in another computer, and i'm trying to route my calls to the teliax server and then go out to PSTN, i'm using IAX2 |
| 04:26.55 | [N00B] | asdx: asterisk version? |
| 04:27.12 | UserReg_CL | exten => s,1,... "s"? is for call in ?? |
| 04:27.17 | *** join/#asterisk _theHub (n=_theHub@ool-43577a99.dyn.optonline.net) |
| 04:27.33 | asdx | [N00B]: latest |
| 04:27.44 | [N00B] | configs? |
| 04:27.48 | asdx | 1min |
| 04:29.00 | UserReg_CL | please.. need one example for using GOTO command Dial |
| 04:29.31 | [N00B] | exten => blah,1,Goto(context|extension|priority) |
| 04:29.52 | UserReg_CL | when use Goto command ? |
| 04:30.02 | [N00B] | UserReg_CL: please refer to google or the asterisk wiki |
| 04:30.27 | [N00B] | you use it when you want your dialplan to "goto" something else |
| 04:31.54 | UserReg_CL | N0000: and assign one total time for one user SIP for example: for user SIP/1001 assign 100 minutes for one month |
| 04:32.43 | [N00B] | there is no one here by that username |
| 04:32.48 | [N00B] | what are you trying to ask? |
| 04:33.25 | UserReg_CL | need assign for one user SIP one total time for call to pstn net |
| 04:33.34 | UserReg_CL | (sorry by bad english, I am talk spanish) |
| 04:34.14 | [N00B] | sounds like you are trying to do billing... if so please, once again, refer to google or the asterisk wiki on some sort of a howto. There are programs to do that for you and examples of dialplan options. |
| 04:34.43 | [N00B] | asdx: you still alive? |
| 04:35.28 | [N00B] | UserReg_CL: I would refer you also to the asterisk cdr and cdr_mysql |
| 04:36.37 | UserReg_CL | traslated.... |
| 04:38.24 | [N00B] | ....? |
| 04:38.24 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 04:38.29 | UserReg_CL | mmm not.. in cdr look total time for user... need assign one time (x minutes) for all call to pstn for month |
| 04:39.18 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 04:39.40 | [N00B] | you need to know your total minutes to the pstn? |
| 04:39.46 | UserReg_CL | not |
| 04:40.16 | [N00B] | then i havent got a clue what you are asking |
| 04:40.44 | UserReg_CL | need what one user, example john, have x minutes for month for all call to pstn (example: john 100 minutes for month for call to pstn) |
| 04:40.49 | asdx | [N00B]: yeah http://pastebin.ca/779078 |
| 04:41.36 | [N00B] | UserReg_CL: anyone here speak spanish? otherwise, sorry, I cant help you. it still sounds to me like you need cdr data |
| 04:41.38 | asdx | [N00B]: i'm trying to dial 59521201964... |
| 04:42.08 | [N00B] | asdx: first of all, do an IAX2 show registry at the cli> |
| 04:42.17 | asdx | k |
| 04:42.42 | [N00B] | output? |
| 04:43.57 | jameswf-home | jbot: newbe |
| 04:44.03 | jameswf-home | jbot: newb |
| 04:44.04 | jbot | Don't bother telling us you're a "newb" or a "n00b". We can tell. |
| 04:44.52 | asdx | [N00B]: http://pastebin.com/m2679d6ab |
| 04:45.03 | [N00B] | jameswf-home: sorry, have i done something to offend you? that seems kinda out of the blue |
| 04:45.50 | jameswf-home | lol I am only offended by orange skittles |
| 04:45.58 | coppice | what do you expect when you walk around with a "KICK ME" sign on your back :-) |
| 04:46.43 | [N00B] | asdx: can you give me the cli output from start to finish when you place that call? |
| 04:46.53 | asdx | [N00B]: ok |
| 04:47.01 | [N00B] | need to see more than just the error... need to see what is happening before hand. |
| 04:47.09 | UserReg_CL | mmm |
| 04:47.13 | jameswf-home | jbot: dropdatabase; |
| 04:47.13 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 04:47.23 | jameswf-home | I love that |
| 04:48.20 | coppice | I love that cartoon where the kid that has a weird name containing punctuation marks that wrecks the databases they enter it into. |
| 04:50.02 | jameswf-home | my wife wouldnt let me name any of our kids that |
| 04:50.14 | asdx | [N00B]: http://pastebin.com/m5863f4ba |
| 04:50.26 | jameswf-home | should have married a geek lol |
| 04:50.31 | asdx | [N00B]: interesting, when i dial "test" i get a incoming call |
| 04:51.01 | jameswf-home | I probably couldbnt share the bandwith though |
| 04:51.03 | asdx | [N00B]: i guess is because i'm doing Dial(IAX2/user) |
| 04:51.26 | asdx | [N00B]: but i want to do a incoming call trough that pstn phone number |
| 04:51.28 | tzafrir_home | http://xkcd.com/327/ |
| 04:51.48 | coppice | yeah, that's the one |
| 04:51.48 | [N00B] | that is your inbound number? or you are trying to dial it outbound from asterisk? |
| 04:52.07 | asdx | [N00B]: i'm trying to do a outbound call |
| 04:52.25 | [N00B] | ok... your cli looks like it worked. i did not see your error |
| 04:52.40 | jameswf-home | I like http://xkcd.com/330/ |
| 04:53.09 | jameswf-home | its been a running joke @ work |
| 04:53.45 | asdx | [N00B]: but in my softphone i see "Incoming call, waiting for answer" |
| 04:53.52 | asdx | [N00B]: shouldnt be the other way around? |
| 04:54.00 | [N00B] | is that your inbound number also? |
| 04:54.05 | jameswf-home | we also have a 3 month running gaim of where's waldo, a 6'' waldo gets hidden in strange places |
| 04:54.39 | asdx | [N00B]: no |
| 04:54.50 | [N00B] | send me your whole extensions.conf |
| 04:55.19 | coppice | I think this one http://xkcd.com/334/ really gets to the heart of his humour |
| 04:55.44 | asdx | [N00B]: that's my whole extensions.conf, it starts in line 14 http://pastebin.ca/779078 |
| 04:55.52 | UserReg_CL | good night all |
| 04:56.01 | jameswf-home | asdx: my wife didnt like that one lol |
| 04:56.21 | jameswf-home | she liked the cuddle bed with the robots |
| 04:56.22 | asdx | jameswf-home: ? |
| 04:56.22 | asdx | lol |
| 04:57.43 | jameswf-home | mythbusters did bus behind jet ... neat |
| 04:59.17 | coppice | http://xkcd.com/317/ |
| 05:00.58 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 05:04.07 | alephcom_ | this is annoying, my agi script is totally messed up. :-( |
| 05:04.24 | alephcom_ | I guess I should either keep others hands out of it or else comment more. |
| 05:04.52 | ectospasm | documentation is only bemoaned when it's too late (-; |
| 05:14.33 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 05:17.46 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 05:33.09 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 05:34.25 | jameswf-home | its an mmrprpg you r tard |
| 05:34.36 | jameswf-home | jbot: rtard |
| 05:35.43 | jameswf-home | jbot: wow |
| 05:35.44 | jbot | I have no life | Lets go raid! |
| 05:35.49 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 05:36.30 | *** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 05:38.10 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 05:38.50 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 05:43.39 | *** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 05:45.42 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 05:47.16 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 05:47.32 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 05:47.40 | Zuchmir | can i use a dialogic card with asterisk |
| 05:48.44 | Zuchmir | is there any editing software for GSM? |
| 05:49.44 | alephcom_ | Zechmir: I usually use wavepad for gsm files |
| 05:50.39 | *** join/#asterisk echosyp (n=fu69@adsl-70-255-87-110.dsl.lbcktx.sbcglobal.net) |
| 05:52.22 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:52.44 | Zuchmir | alephcom_: thanks |
| 05:52.46 | tzafrir_home | Zuchmir, what sort of editing? |
| 05:53.12 | Zuchmir | sound editor |
| 05:53.23 | tzafrir_home | Zuchmir, sox has quite a few capabilities for automated editing |
| 05:57.22 | mackes | Ok, Everyones thoughts SER vs OpenSER |
| 05:58.22 | tzafrir_home | OpenSER has extra four letters |
| 05:59.00 | echosyp | i need help with a project of mine |
| 05:59.08 | echosyp | i want to offer free phone service to my tenants |
| 05:59.23 | echosyp | 14 units per building |
| 05:59.33 | echosyp | 16* |
| 06:00.22 | echosyp | i was thinking of using the existing phone lines |
| 06:01.21 | echosyp | i don't know where to start, there are alot of options/services to check out |
| 06:04.00 | echosyp | someone help me out here |
| 06:04.14 | fakhir | ~thebook |
| 06:04.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 06:05.46 | *** join/#asterisk [zender] (n=ender@72.52.23.3) |
| 06:07.36 | echosyp | notex |
| 06:07.39 | echosyp | noted* |
| 06:10.27 | Zuchmir | tzafrir_home: thanks, i know about sox, i was looking for manual editing |
| 06:12.57 | Zuchmir | looks like wavepad does what i need |
| 06:19.50 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 06:27.17 | echosyp | any other reference suggestions? |
| 06:39.22 | *** join/#asterisk kmhunt (i=Ment0r@12-214-162-192.client.mchsi.com) |
| 06:43.04 | [TK]D-Fender | echosyp, that should do it |
| 06:43.34 | echosyp | k |
| 06:44.31 | echosyp | will an 8mb isp pipe support 48 phones |
| 06:44.38 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:46.13 | fakhir | echosyp, depends on how many concurrent calls, codecs, ... |
| 06:49.09 | echosyp | gotcha |
| 06:49.20 | echosyp | doubt there will be htat many concurrent calls |
| 06:49.21 | [TK]D-Fender | echosyp, easily |
| 06:53.21 | BBHoss | no |
| 06:55.30 | *** join/#asterisk bmg505 (n=leon@196.209.177.134) |
| 06:56.57 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 06:57.50 | BBHoss | ok, back now |
| 06:59.21 | echosyp | i will be talking to you guys more soon |
| 06:59.27 | echosyp | im gonna read up and go to sleep |
| 07:02.00 | BBHoss | does asterisk still support aes128 encryption on iax2? |
| 07:05.00 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 07:05.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 07:15.54 | *** part/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 07:25.25 | obnauticus | Where's the default AGI directory again? |
| 07:25.33 | obnauticus | and BBHoss I got it all 100% working :) |
| 07:26.19 | obnauticus | Nevermind I found itL /var/lib/asterisk/agi-bin/ |
| 07:26.58 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 07:37.01 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 07:38.28 | *** join/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 07:45.45 | *** join/#asterisk timothywcrane_ (n=hanif@ip72-207-2-65.sd.sd.cox.net) |
| 07:45.58 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 07:47.04 | timothywcrane_ | I want to use asterisk to set up a telephone order taking system. How does it recognize touchtone signals for recording them in the case of cc #s? |
| 07:47.13 | timothywcrane_ | so I can write them to file |
| 07:47.28 | timothywcrane_ | or am I way off base? |
| 07:47.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:47.38 | luke-jr | yeah, you should use HTTPS for that |
| 07:47.48 | timothywcrane_ | thank you |
| 07:47.51 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 07:48.40 | timothywcrane_ | not a secure server for inline, but for people to call in to a IVR system and enter cc # to order products |
| 07:48.48 | timothywcrane_ | onl;ine |
| 07:49.03 | timothywcrane_ | ^$#^ must be tired |
| 07:49.31 | tzafrir | timothywcrane_, that's really the basic IVR Asterisk can do |
| 07:49.40 | tzafrir | do you have an Asterisk system running? |
| 07:49.59 | luke-jr | timothywcrane_: phones are insecure |
| 07:50.08 | timothywcrane_ | no not yet, I was looking at it and Yate. I am on my desktop doing research for my server projects |
| 07:50.43 | phix | so does asterisk support the AMR NB codec? or is it a commerical codec? |
| 07:50.59 | timothywcrane_ | I know this , but for some reason, some of my customers would rather cost me lots of money than enter it into a secure site. |
| 07:51.02 | phix | prioritery even |
| 07:51.05 | tzafrir | If you look at the "demo" sample in the sample configuration that comes with asterisk, |
| 07:51.38 | tzafrir | then just add an option with '#' instead of '1' , '2', or '3' as there are lready there |
| 07:52.01 | tzafrir | phix, AMR has patent issues |
| 07:52.46 | tzafrir | timothywcrane_, order by phone? or order through the web? |
| 07:53.40 | timothywcrane_ | order by phone, trying to integrate with skype and crm |
| 07:53.41 | phix | tzafrir: aawww :( |
| 07:53.53 | phix | I guess I will use ulaw on my mobile phone then |
| 07:54.09 | phix | hmmm it supports iblc |
| 07:54.19 | phix | isn't that a terriable codec? |
| 07:54.32 | tzafrir | phix, what about g726, gsm, ilbc, speex? none of them supported? |
| 07:55.17 | phix | suprisenly gsm isn't, I thought it would support it as it is a mobile phone |
| 07:55.54 | phix | ummm g729 I think it does as well, but I dont really want to buy a licence, although they are only $15AU |
| 07:56.19 | tzafrir | (per channel) |
| 07:56.38 | phix | yeah |
| 07:56.44 | tzafrir | What I really don't like about such licenses is the licensing overhead: |
| 07:56.50 | phix | I only need one channel to my phone |
| 07:57.10 | phix | ditto, and the fact yuo need to pay $15 AU again if you change your hardware too often |
| 07:57.55 | tzafrir | or call Digium (not sure of the exact procedure), but still a headache |
| 07:58.42 | *** join/#asterisk izaak (n=izaak@bas1-montreal02-1096726242.dsl.bell.ca) |
| 07:58.52 | phix | Capabilities: us - 0x1e (gsm|ulaw|alaw|g726), peer - audio=0x50c (ulaw|alaw|g729|ilbc) |
| 07:58.57 | phix | tzafrir: agreed |
| 07:59.40 | tzafrir | ilbc is nice |
| 07:59.54 | coppice | ilbc is brain dead |
| 08:00.02 | phix | doesn't it sound like your speaking to a robot? |
| 08:00.02 | izaak | anyone wanna shout their recommended OS for asterisk? i'm leaning towards debian, but i'm worried about running testing/unstable |
| 08:00.14 | phix | izaak: I use Debian Etch |
| 08:00.21 | BBHoss | izaak: im on etch right now |
| 08:00.21 | phix | izaak: run stable |
| 08:00.29 | BBHoss | no problems here |
| 08:00.31 | tzafrir | ah, I knew there was a reason you missed ilbc |
| 08:00.35 | phix | izaak: Asterisk 1.4 is on Etch |
| 08:00.47 | izaak | phix: hm? really? |
| 08:00.51 | phix | izaak: yes |
| 08:00.58 | tzafrir | ilbc's license is a bit to shoddy for Debian |
| 08:00.58 | coppice | ilbc sounds comparable to G.729, but it takes twive the bit rate to achieve that |
| 08:01.10 | izaak | phix, what method did you use? |
| 08:01.14 | phix | izaak: oops, on it isn't |
| 08:01.16 | phix | <PROTECTED> |
| 08:01.16 | phix | ii asterisk 1.2.13~dfsg-2etch1 Open Source Private Branch Exchange (PBX) |
| 08:01.21 | izaak | yeah. |
| 08:01.28 | phix | oh well :) what is wrong with 1.2? |
| 08:01.33 | izaak | no imap voicemail :P |
| 08:01.43 | phix | imap voicemail? |
| 08:01.57 | tzafrir | izaak, you actually use imap voicemail? |
| 08:02.04 | phix | izaak: well use pinning then and set it up to only use testing for asterisk |
| 08:02.17 | tzafrir | This is something that takes quite a bit of voodoo to set up, IIRC |
| 08:02.21 | phix | izaak: that way most of your system is running stable |
| 08:02.40 | izaak | no, but i like the idea. a major annoyance of mine is that people love e-mail notification so much they often forget to clear out their messages. then some people prefer checking their messages with their phones. imap voicemail unites the two. |
| 08:02.46 | phix | izaak: you know about pinning right? /etc/apt/preferences ? |
| 08:02.48 | tzafrir | phix, Asterisk 1.4 is still not in testing . Some people are working hard on that: |
| 08:03.05 | phix | tzafrir: :O |
| 08:03.08 | izaak | phix: i've used it once before in ubuntu. |
| 08:03.10 | tzafrir | http://packages.qa.debian.org/a/asterisk.html |
| 08:03.12 | izaak | it should be there soon |
| 08:03.15 | phix | izaak: in that case use unstable for asterisk only ;) |
| 08:03.26 | tzafrir | http://bjorn.haxx.se/debian/testing.pl?package=asterisk |
| 08:03.47 | tzafrir | There are some backports at http://buildserver.net |
| 08:03.55 | phix | izaak: or just be like me and be comtempt with 1.2 |
| 08:04.07 | tzafrir | (including imap support) |
| 08:04.21 | izaak | so, by pinning unstable asterisk, i'll end up with a handful of unstable dependencies right? (no problem, just confirming...) |
| 08:05.16 | tzafrir | izaak, one problem is the newer libc-client |
| 08:05.47 | izaak | mm yeah, i think i'll just stick with etch and wait :) |
| 08:06.02 | izaak | thanks for bouncing the idea |
| 08:06.33 | tzafrir | anyway, right now Etch does get security updates. Be sure to update your package |
| 08:12.43 | izaak | i'm excited about this new system i have in mind - an Intel D201GLY2 w/ TDM400 connecting a couple telephone lines (IAX) to a WRAP 2C3, the main office PBX |
| 08:12.47 | izaak | debian on 1GB CF on both |
| 08:13.20 | coppice | is the Intel D201GLY2 the ITX board that takes too much power? |
| 08:13.23 | izaak | er, not a WRAP but the newer ALIX... |
| 08:14.14 | izaak | it's uATX, but yeah it's not quite as low as the VIA boards. but it is fan-less. i will still put it in a nice case with fans, though. |
| 08:14.57 | coppice | intel has what looks like a nice ITX board at a low price. then you find it takes 30W |
| 08:16.50 | izaak | yeah but it performs way better than the via chipsets with a bigger cache and better FPU performance |
| 08:17.32 | izaak | i figure it will be nicer for driving a TDM400p, and i'll put the really important PBX functions on the ultra-low ALIX |
| 08:17.50 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 08:18.59 | coppice | but it won't go in any ITX box I ever saw |
| 08:19.41 | coppice | give VIA several times the power budget and I expect they'll go faster too :-) |
| 08:20.42 | izaak | but there are a good selection of micro atx cases, either mini tower or thin desktop. for example antec boxes have better ventilation than most itx cases i've seen. |
| 08:21.20 | coppice | few things run at a sane temperature without at least a little fanning |
| 08:21.47 | izaak | yeah. |
| 08:21.51 | coppice | the secret is to find things with big slow fans |
| 08:22.21 | izaak | i'm using this one - antec 2480 - two 120cm |
| 08:22.51 | coppice | 120cm definitely classifies as big :-) |
| 08:36.05 | *** part/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 08:37.56 | *** join/#asterisk marl (n=marl@89.241.242.164) |
| 08:38.11 | marl | good morning folks :) |
| 08:42.55 | marl | can anyone tell me if there is a way to set a varable within a context in extensions.conf that can be set once, and will hold for any extensions dial within that context? i have a setup like this : http://www.pastebin.ca/779207 |
| 08:43.49 | marl | im just not sure how to do the SetVar line, so it sets the var but doesnt affect any of the extensions that follow |
| 08:46.43 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
| 08:51.23 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 08:54.06 | timothywcrane_ | anyone familiar with FreePBX? |
| 08:55.06 | marl | wats the prob, not used it much, but will try :) |
| 09:03.44 | timothywcrane_ | not a prob, just looking for a thumbs up or down . |
| 09:03.55 | marl | go on |
| 09:04.04 | marl | u thinking of installing it? |
| 09:04.23 | timothywcrane_ | loking to integrate pbx order by phone function and cms for e commerce |
| 09:04.46 | timothywcrane_ | yeah but I want to get all the tools together that I want before I do an install |
| 09:04.50 | BBHoss | freepbx is harder to troubleshoot usually than asterisk |
| 09:05.04 | timothywcrane_ | goods advice, |
| 09:05.11 | marl | freepbx is good in some ways, as it has a good frontend, but if u want to add any code to it, its a BIG PAIN! |
| 09:05.21 | BBHoss | ~trixbox |
| 09:05.21 | jbot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support, and thus you will find little help here for it. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 09:05.43 | timothywcrane_ | is astericks really hard to set up for someone familiar with Linux |
| 09:05.44 | marl | lol |
| 09:05.53 | marl | nope, easy |
| 09:06.01 | BBHoss | its all in what you want to do with it |
| 09:06.05 | marl | so long as u rmemeber to build the sample files as well :) |
| 09:06.14 | timothywcrane_ | good off to find the perfect cms for my projec then |
| 09:06.29 | timothywcrane_ | project |
| 09:06.42 | timothywcrane_ | thanks marl, jbot |
| 09:06.54 | marl | is it cms or crm |
| 09:06.56 | marl | ? |
| 09:07.01 | BBHoss | crm probably |
| 09:07.08 | marl | thought so |
| 09:07.16 | BBHoss | sugarcrm is commonly included, you could try that |
| 09:07.19 | marl | sugar_crm could be a good starting poiint |
| 09:07.22 | marl | lol |
| 09:07.25 | timothywcrane_ | the crm is prob going to be Sugar, seems to integrate well, the CMS is for the website |
| 09:07.40 | BBHoss | i like drupal |
| 09:08.20 | timothywcrane_ | I'm used to Joomla, but I know Drupal is more extensive in developement and has less holes |
| 09:08.42 | marl | im sure i read an article on a basic http * front end available from * svn, but i cant find it now, anyone know were it is, or was i just dreaming? |
| 09:08.45 | coppice | I like Gruyere for its holes |
| 09:09.05 | timothywcrane_ | bye guys |
| 09:26.46 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:36.51 | *** join/#asterisk GuyOCanada (i=GuyOCana@75.155.220.205) |
| 09:38.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:39.17 | tzafrir | marl, start with the svnbook |
| 09:39.49 | GuyOCanada | I am building a dialplan and need some help with a scenario like this (the user calls in dials the vip support extension. is being asked for a numeric user id, after the userid is being asked for a numeric password and if the info is correct he will be redirected to another context. (I am clueless on how to do it) can anyone point me to any page that has a tutorial for stuff like that |
| 09:39.56 | tzafrir | http://svnbook.red-bean.com/ |
| 09:40.50 | tzafrir | GuyOCanada, there are a bunch of handy dialplan applications: |
| 09:40.58 | tzafrir | Read, Authenticate , and such |
| 09:43.08 | marl | tzafrir, its not how to get the stuff from svn, i can do that bit, its that ive lost the link to the page about the application :( |
| 09:45.07 | tzafrir | marl, it's a module for apache |
| 09:46.11 | tzafrir | On Debian, it would be something of the sort of: apt-get install libapache2-svn |
| 09:47.54 | marl | its the asterisk http application that i cant find! i read an article about a new http frontend for * i can remember that it was available via svn, but i cant remember the name of the * applicaion :( |
| 09:49.19 | tzafrir | marl, asterisk has a built-in httpd as of 1.4 |
| 09:49.56 | tzafrir | The asterisk-gui is technically mostly just a bunch of javascript pages served from that httpd |
| 09:50.28 | tzafrir | asterisk comes with e.g. a sample astmon.js to be used from that built-in httpd |
| 09:50.38 | GuyOCanada | Read:Read a variable in the form for DTMF tones as pressed by the caller |
| 09:50.39 | tzafrir | but you may be looking for the asterisk-gui: |
| 09:50.50 | GuyOCanada | can anyone explain that? |
| 09:51.10 | tzafrir | svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui |
| 09:51.48 | tzafrir | GuyOCanada, are you familiar with shell scripts? |
| 09:51.56 | GuyOCanada | yes |
| 09:52.12 | tzafrir | well, this is esencially like the shell's read |
| 09:52.29 | tzafrir | waits for the user to "type" in a value |
| 09:52.46 | *** join/#asterisk salzh (n=salzh@124.77.5.180) |
| 09:52.51 | tzafrir | That value then goes to a variable |
| 09:54.08 | GuyOCanada | Well i dont get how it would help me authenticating a user |
| 09:55.14 | tzafrir | It would help you get a value if you have other means of authentication that are better than the ones provided by Authenticate |
| 09:55.19 | tzafrir | or VMAuthenticate |
| 09:56.06 | marl | ok, im going to repost this question, just incase anyone has woken up who may be able to help me :) |
| 09:56.08 | marl | can anyone tell me if there is a way to set a varable within a context in extensions.conf that can be set once, and will hold for any extensions dial within that context? i have a setup like this : http://www.pastebin.ca/779207 |
| 09:56.16 | marl | <PROTECTED> |
| 09:57.08 | marl | i originally posted it at 0830 my time, hopefully there may be some early risers, up now, hangovers permitting :) |
| 09:57.34 | GuyOCanada | well That means for every user i have to set a diff. variable :) that would not be good |
| 09:59.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:03.38 | GuyOCanada | whats the difference between a hangup and a softhangup |
| 10:04.00 | tzafrir | GuyOCanada, have you really looked at Authenticate? |
| 10:06.29 | GuyOCanada | i was reading http://www.voip-info.org/wiki-Asterisk+cmd+authenticate |
| 10:07.20 | tzafrir | GuyOCanada, [core] show application Authenticate |
| 10:07.43 | tzafrir | And use tab completion |
| 10:07.49 | tzafrir | (from the asterisk CLI) |
| 10:08.51 | GuyOCanada | tzafrir: i read that its the same on the page |
| 10:09.40 | tzafrir | marl, variables are local to the channel (except globals) |
| 10:09.41 | GuyOCanada | Maybe I can use authenticate twice? first for the username and then for the password? |
| 10:10.41 | tzafrir | GuyOCanada,consider Read twice and doing your own custom authentication, then |
| 10:11.14 | GuyOCanada | Well I wish there was an easy way to read from my mysql table :) |
| 10:14.02 | GuyOCanada | A question about DISA |
| 10:14.40 | marl | tzafrir, can a var not be local to a context ? |
| 10:14.50 | tzafrir | marl, no |
| 10:14.53 | GuyOCanada | I have mapped extension 9 to authenticate (authenticate has a 6 digit password) and authenticate is connected to DISA without a pasword |
| 10:15.05 | GuyOCanada | how would i select which context it goes to? |
| 10:16.09 | marl | thanks :( will now go and write a couple of macros to do the job :( |
| 10:24.37 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
| 10:34.40 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:35.46 | marl | im trying to find a good howto on creating your own functions within *, i need to be able to return a string from the function, any pointers? cant find anything about creating your own functions :( |
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| 11:03.24 | *** join/#asterisk bintut (n=chatzill@cm246.gamma178.maxonline.com.sg) |
| 11:03.47 | bintut | hello all.. |
| 11:04.03 | *** join/#asterisk NirS (n=chatzill@87.68.157.68) |
| 11:04.48 | NirS | good morning all |
| 11:04.50 | NirS | anybody home ? |
| 11:05.13 | bintut | i am trying to connect 2 asterisk servers using iax2 over an openvpn tunnel but i can't make them work. my asterisk is in version 1.4.13 and the remote asterisk box is in version 1.2.4(?) |
| 11:05.24 | bintut | NirS: i am at home.. |
| 11:05.52 | NirS | hey bintut |
| 11:06.03 | bintut | hello NirS.. :) |
| 11:06.48 | NirS | can you paste your iax2.conf contexts to pastebin.com - I'll have a look at it |
| 11:07.30 | bintut | NirS: ok.. for a while.. |
| 11:11.09 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 11:26.51 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 11:27.55 | tzafrir | Hi NirS |
| 11:28.16 | tzafrir | Morning?? |
| 11:28.33 | bintut | NirS: it's here => http://www.privatepaste.com/141646WsIT |
| 11:28.45 | bintut | hello tzafrir.. :) |
| 11:40.14 | bintut | NirS? |
| 11:41.11 | NirS | sorry |
| 11:41.25 | NirS | lets take a lok |
| 11:42.32 | NirS | bintut, when you perform an Asterisk-2-Asterisk connection, you're not really required to perform a registration from one to the other |
| 11:42.34 | NirS | it is enough to define the peers with the proper IP addresses, and that's usually it |
| 11:43.00 | NirS | if you want to use passwords, simply embed that information into the dial string on each side, or dial via the peer context, that would make life much simpler |
| 11:44.55 | bintut | NirS: ok. actually, i just got that config from the asterisk: the future of telephone 2nd edition book |
| 11:45.02 | *** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu) |
| 11:45.23 | NirS | I see |
| 11:47.29 | bintut | NirS: but, do you think there's something wrong with my config that both of them cannot register to each other? |
| 11:47.46 | NirS | hold on |
| 11:48.00 | *** join/#asterisk implicit (n=implicit@207.181.11.96) |
| 11:48.48 | Mavvie | exten => _8226231[0123456789],Macro(call-int-deskfax,612${EXTEN}) |
| 11:49.05 | Mavvie | throw an error damned, extensions loader, when you see one! |
| 11:49.10 | NirS | bintut, what are the exact error you are receiving ? |
| 11:50.26 | NirS | btw, anyone ever encountered an issue with PHPAGI where it passes back variable to the script wrong ? |
| 11:50.30 | bintut | i don't get any error message.. based on my "iax2 show registry" on my asterisk box, it says that the state is still on Request Sent |
| 11:51.02 | NirS | hmmm... in that case, you have a networking issue, not an Asterisk issue |
| 11:51.10 | NirS | sounds like your VPN isn't completely up or something |
| 11:52.12 | bintut | actually, i'm connected to the remote server through vpn |
| 11:52.42 | NirS | well, are you sure that you're running a full VPN, and not a per port tunnel ? |
| 11:53.07 | bintut | NirS: yes |
| 11:53.46 | NirS | can you issue 'iax2 debug on' to your 1.2.X server, and tell me what it says |
| 11:55.06 | bintut | its very noisy |
| 11:55.19 | NirS | hmmmm.... |
| 11:55.24 | NirS | define noisy |
| 12:00.17 | bintut | NirS: http://www.privatepaste.com/e21d3sNVeL |
| 12:00.36 | bintut | NirS: well, so many output.. |
| 12:01.33 | NirS | Well, I think that at this point I would need some access to your boxes to look into |
| 12:03.13 | bintut | i don't own the remote box. it's my friend's box. |
| 12:03.29 | bintut | NirS: what do you want me to do instead? |
| 12:04.11 | NirS | well, unless you run a tcpdump and a full debug of your asterisk, and paste it somewhere, I can't really help you |
| 12:04.12 | NirS | I need a better view of what's going on |
| 12:06.45 | bintut | ok |
| 12:06.54 | bintut | i'll get a tcpdump capture |
| 12:16.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 12:40.34 | Greek-Boy | what is the pattern for the range 300 to 399? |
| 12:40.51 | Greek-Boy | _3YY ? |
| 12:41.00 | NirS | no |
| 12:41.13 | NirS | _3XX |
| 12:41.32 | Greek-Boy | sorry, i'm getting rusty with my dialplan rules |
| 12:41.39 | Greek-Boy | dont practice them often enough... |
| 12:41.53 | Greek-Boy | and what does ._3XX mean again>? |
| 12:43.49 | marl | can anyone tell me how to use an external program to return a value in the following : exten => _0[1-9].,1,MixMonitor(external-script(${context})-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN}.wav) basicly i want the external-script to accept ${context} and return a path that will be used with the rst of the mix monitor command |
| 12:46.35 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 12:47.08 | marl | geek-boy, isnt it that the number has to start with a 3 and be 3 digits long? |
| 12:47.51 | marl | sorry greek-boy that was ment for you :) |
| 12:50.17 | Greek-Boy | yes, start with 3 and be 3 digits long |
| 12:50.41 | Greek-Boy | thats _3XX, right? |
| 12:51.33 | marl | yup sorry _3XX is 3 digit starting with 3 |
| 12:51.55 | Greek-Boy | and .3 is anything starting with 3 |
| 12:54.12 | Greek-Boy | what happens if I add _3XX and add 307 as well to the dialplan? |
| 12:55.38 | marl | think it depends on which comes first |
| 12:56.03 | marl | if 307 comes first, then it will take priority, cant be certain thow sorry |
| 12:56.05 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128685020.dsl.bell.ca) |
| 12:58.19 | Greek-Boy | thanks |
| 12:59.17 | marl | isnt the .3 any number ending in 3 btw? |
| 13:03.53 | tzafrir | _3. |
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| 13:12.21 | *** join/#asterisk Qb3rt (n=eric@modemcable156.182-80-70.mc.videotron.ca) |
| 13:12.55 | Mavvie | > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) |
| 13:13.00 | Mavvie | my telco doesn't like that one... |
| 13:16.03 | *** join/#asterisk _theHub (n=_theHub@ool-43577a99.dyn.optonline.net) |
| 13:18.40 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 13:18.58 | skirmisha | can someone tell me what is func_core.so module doing |
| 13:19.04 | tzafrir | Mavvie, what version of Asterisk is it? |
| 13:19.07 | skirmisha | and do i need it loaded everytime |
| 13:19.27 | Mavvie | tzafrir: I'm further now, it's coming from the hylafax server who wants it. |
| 13:19.46 | Mavvie | tzafrir: but it's an 1.2-r34160M version from a year ago. |
| 13:21.10 | skirmisha | ??? |
| 13:21.28 | tzafrir | skirmisha, all sorts of functions |
| 13:21.34 | Mavvie | that 3.1KHz is part of the audio-fax service. somehow that is. |
| 13:22.00 | tzafrir | strings /usr/lib/asterisk/modules/func_core.so |less |
| 13:22.06 | Mavvie | tzafrir: since the hylafax is talking to the E1 card as a normal modem, I guess that it's the patton card... |
| 13:22.22 | skirmisha | tzafrir is it so important to load it? can i miss it? |
| 13:22.35 | tzafrir | skirmisha, you would normally want it |
| 13:22.54 | skirmisha | does it open any tcp or udp port? |
| 13:22.59 | tzafrir | Unless you're very short of memory |
| 13:23.00 | tzafrir | no |
| 13:23.46 | skirmisha | it is strange because it gets in conflict when u try to run 2 diff daemons on same box |
| 13:24.07 | tzafrir | right, it seems to be gone in 1.4, right? |
| 13:26.24 | skirmisha | yes |
| 13:26.57 | skirmisha | what about if u have 1.2 and 1.4 |
| 13:27.07 | skirmisha | is it func_core substracted in 1.4 |
| 13:27.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:27.56 | skirmisha | i mean replaced |
| 13:28.17 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 13:30.42 | bintut | gtg now.. |
| 13:31.24 | skirmisha | tzafrir does it require some other module to be loaded as well |
| 13:31.40 | skirmisha | because when i try to load func_core.so it crash |
| 13:31.55 | tzafrir | skirmisha, what version of Asterisk do you use? recently upgraded from 1.2 to 1.4? |
| 13:32.18 | skirmisha | 1.2 |
| 13:32.29 | skirmisha | but i am loading only modules i need |
| 13:32.41 | skirmisha | so probably i miss some module |
| 13:32.50 | tzafrir | what error does it give you when it crashes? |
| 13:33.15 | skirmisha | dies with error code 1 |
| 13:34.07 | tzafrir | any more specific error message? in the logs? |
| 13:34.16 | skirmisha | let me see |
| 13:37.49 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:39.32 | skirmisha | nope |
| 13:40.24 | tzafrir | ls -lt /usr/lib/asterisk/modules |
| 13:40.43 | tzafrir | was func_core.so built at a distinctively different time? |
| 13:41.38 | skirmisha | i think i found it |
| 13:45.32 | skirmisha | func_core hasn't been build at all |
| 13:58.44 | tzafrir | there's an explicit load=> for it> |
| 13:58.46 | tzafrir | ? |
| 13:59.00 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
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| 14:23.29 | puzzled | hi |
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| 14:36.43 | Qb3rt | i am working on a project for a bank and i want to know... Is it possible to sniff the network traffic while there is a phone conversation using asterisk and listen to the conversation?? |
| 14:37.55 | tzafrir | Qb3rt, sure. experiment with wireshark. |
| 14:38.14 | Qb3rt | is there a manner to encrypt the conversavion?? |
| 14:38.48 | tzafrir | Asterisk supports IAX encryption. This currently only seems to work between Asterisk servers |
| 14:39.11 | Qb3rt | ok :/ thanks for your help |
| 14:39.13 | rob0 | or, encrypt the IP layer in any of several ways (I like openvpn). |
| 14:39.37 | Qb3rt | yeah.... good idea :) |
| 14:40.08 | rob0 | Note that Ethernet sniffing requires access to the media, so it might not be a real-world concern. |
| 14:40.15 | tzafrir | There's SIP over TLS. But this won't encrypt the audio itself |
| 14:40.52 | tzafrir | and also usually won't encrypt dialed digits |
| 14:43.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:52.31 | Greek-Boy | or make it a standard to use a linux firewall at each branch with openvpn support |
| 15:02.38 | rob0 | Linux is a good choice, but OpenBSD might look more pleasing to the technically ignorant. (Not a slam against OBSD, just that their secure-by-default policies are easier to understand, and appeal to those who wouldn't know that a Linux can be made similarly secure.) |
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| 15:57.58 | jtexter3 | I have a site running 1.4.13. The past 2 days, conferences have started to fail with "Unable to open pseudo device". A simple restart of Asterisk clears up the error |
| 15:58.36 | jtexter3 | The error logged is no such device or address. This is a system with 2 TE412P's loaded, and appropriate modules |
| 15:59.02 | jtexter3 | Anyone come across something like this before? |
| 15:59.47 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 16:00.36 | JayTee52 | Buenos Dias, Asteristas! Viva la revolucion! :-) |
| 16:01.23 | Greek-Boy | anyone know a good asterisk test server available on the internet? google doesn't return anything an digium's demo server misery.digium.com only plays IVR's. |
| 16:01.53 | [TK]D-Fender | Greek-Boy, what is a "test server"? |
| 16:02.30 | JayTee52 | a test server? I'm running Asterisk on a 733mhz clone with an Intel mobo. Just use whatever you got that's half way fast. |
| 16:02.49 | Greek-Boy | a demo or test server that one can dial into and mess around with the applications available |
| 16:02.58 | JayTee52 | [TK]D-Fender, morning dude! |
| 16:03.02 | [TK]D-Fender | Greek-Boy, lol |
| 16:03.18 | Greek-Boy | TK dont u think it would be useful for users out there? |
| 16:03.24 | [TK]D-Fender | Greek-Boy, Sure, so everyone can run System(rm -rf /) |
| 16:03.32 | [TK]D-Fender | Greek-Boy, Install it yourself you lazy ass! :p |
| 16:03.41 | Greek-Boy | i'm thinking of finding these and creating a database |
| 16:03.42 | Greek-Boy | lol |
| 16:03.58 | Greek-Boy | I just wanna make easier accessiblity to asterisk demoing and testing |
| 16:04.01 | JayTee52 | or use AsteriskNOW if you're looking for something "no brainer, no frills" |
| 16:04.15 | [TK]D-Fender | Greek-Boy, What.... and you think they're going to PAY to provide you the bandwidth for this? Or PSTN conenctivity? Oh, and what happens when everyone wants to "play" at once" |
| 16:04.20 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 16:04.27 | Greek-Boy | TK they can't do rm -rf /, its a test just for voip. only IAX and SIP |
| 16:04.27 | JayTee52 | lol |
| 16:04.30 | [TK]D-Fender | Greek-Boy, This isn't some collaboration suite you know... |
| 16:04.45 | Greek-Boy | yeah I suppose you have a point |
| 16:04.46 | [TK]D-Fender | Greek-Boy, "show application system" <---- |
| 16:05.21 | Greek-Boy | digium uses gsm codec on their demo server. |
| 16:05.34 | [TK]D-Fender | Greek-Boy, it is ludicrous to consider the tought of a test server. Next thing you know people will use it as a DDOS point generating calls and pissing people off. |
| 16:05.35 | Greek-Boy | and ofcourse IAX2 protocol |
| 16:06.07 | rob0 | Hmmm. I did System(rm -rf /), and now I can't login. |
| 16:06.08 | Greek-Boy | [TK]D-Fender: As always, you have knocked some sense into me. |
| 16:06.16 | JayTee52 | rob0, hahaha |
| 16:06.19 | Greek-Boy | lol rob0 |
| 16:06.24 | JayTee52 | hope ya got a backup! |
| 16:06.27 | Greek-Boy | rob0: I know u kidding |
| 16:06.43 | Greek-Boy | lol! @ ClueBat |
| 16:06.57 | Greek-Boy | u sure its a registered trademark? |
| 16:07.16 | rob0 | Yes, thank you, I'll be performing here all week. |
| 16:07.24 | Qwell | ~cluebat [TK]D-Fender |
| 16:07.24 | jbot | ACTION pulls out a ClueBat (tm) and thwaps [TK]D-Fender. |
| 16:07.29 | Greek-Boy | I finally got my hands on a WIP330 today and tested it. It sucks! |
| 16:07.29 | [TK]D-Fender | :O |
| 16:07.34 | JayTee52 | rob0, hey! try this one out in a terminal. :(){ :|:& };: |
| 16:07.41 | Qwell | 330 runs Windows, doesn't it? |
| 16:08.04 | Qwell | 300 is the good one, I think |
| 16:08.04 | [TK]D-Fender | Qwell, scary isnt it ;) |
| 16:08.11 | [TK]D-Fender | Qwell, 300 = ick |
| 16:08.15 | Qwell | that's so unlike linksys |
| 16:08.15 | Greek-Boy | yeah it runs Windows CE. but it lacks features |
| 16:08.22 | [TK]D-Fender | Qwell, no transfer, conference, etc, and slow processor |
| 16:08.23 | Qwell | Greek-Boy: redundant |
| 16:08.44 | Qwell | rob0: :() { :|: }:; |
| 16:08.53 | Qwell | rob0: run that in a shell |
| 16:09.05 | JayTee52 | Qwell, day late and dollar short |
| 16:09.15 | Qwell | oh, d'oh |
| 16:09.24 | Qwell | and I missed the backgrounding |
| 16:09.27 | Qwell | I fail |
| 16:09.34 | JayTee52 | s'ok |
| 16:09.44 | Greek-Boy | i can't wait to see a Wimax mobile phone. And I'm not talking about the Samsung Wibro. I actually want to see a phone based on IEEE802.16e |
| 16:10.50 | Qwell | 802.16? |
| 16:10.54 | Qwell | silly people |
| 16:11.24 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:12.19 | JayTee52 | aren't they using that alot in the UK? |
| 16:28.19 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 16:28.55 | *** join/#asterisk snazm (n=snazm@78.147.13.67) |
| 16:35.53 | Greek-Boy | i'm trying to help some idiot with qmailrocks installation |
| 16:36.03 | Greek-Boy | the whole point of QMR is to be a no-frills install |
| 16:36.04 | Greek-Boy | lol |
| 16:37.46 | *** join/#asterisk asdx (n=diego@adsl-152-126.click.com.py) |
| 16:38.07 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 16:43.52 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 16:44.35 | [TK]D-Fender | "IDC now releases 'Idiots for Dummies'" |
| 16:46.47 | tzafrir | Greek-Boy, but why not just use postfix? |
| 16:47.13 | JayTee52 | [TK]D-Fender, lol |
| 16:48.32 | Greek-Boy | lol |
| 16:48.46 | unixdog | lol |
| 16:49.14 | unixdog | Postfix is nice |
| 16:49.31 | unixdog | and sendmail is stilll apain and a half at times |
| 16:49.40 | Greek-Boy | yes |
| 16:49.48 | Greek-Boy | but qmail is more secure and faster |
| 16:50.09 | unixdog | qmail is a big pain in the asss |
| 16:50.25 | Corydon76-dig | qmail... my favorite cause of spam backscatter |
| 16:50.58 | Corydon76-dig | because it accepts all messages, then generates a false bounce |
| 16:51.31 | Corydon76-dig | instead of just rejecting the email in the first place |
| 16:52.58 | Greek-Boy | if its setup with Clam AV and spamassassin its not bad |
| 16:54.04 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 16:54.29 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-70-196-184.myvzw.com) |
| 16:54.45 | Corydon76-dig | Yes, but unless you set that up yourself, you're violating the qmail distribution license |
| 16:55.54 | unixdog | spam its whats good with eggs and toast |
| 16:56.57 | Corydon76-dig | More than 99% of the mail that hits my server is rejected at SMTP time |
| 16:57.27 | Corydon76-dig | meaning it never goes beyond the spam zombie |
| 17:00.00 | rob0 | qmail more secure ... haha. |
| 17:01.27 | rob0 | DJB is such a hypocrite. He says qmail's #2 on the 'net, which is probably wrong in any event. And every single one of those qmails is patched up, so none qualify for his bogus "guarantee". |
| 17:01.56 | rob0 | BTW he has put it in public domain as of a couple weeks ago. |
| 17:03.15 | rob0 | qmail is crap, though. It has a long way to go to catch up with modern MTAs. Perhaps now someone will try to make that happen, but still, way behind. |
| 17:03.55 | mosty | the more i use postfix, the more i like it |
| 17:04.19 | JayTee52 | the more I eat bacon, the more I like it |
| 17:04.20 | cpm | the more i abuse rob0, the more i like it |
| 17:04.27 | cpm | mmmm, bacon! |
| 17:04.34 | rob0 | Mmmm, abuse!! |
| 17:05.15 | JayTee52 | bacon, it's the magic fairy dust of food groups. sprinkle it on a baked potato or a salad and a side dish becomes and entree |
| 17:05.37 | tzafrir | But then again quite a few people claim that you can't really put stuff in the public domain |
| 17:05.47 | tzafrir | So cpm: don't |
| 17:05.57 | rob0 | Whew, that's good, I didn't want to be PD. |
| 17:06.03 | JayTee52 | otherwise most people would put their mother-in-law in the public domain. |
| 17:06.30 | rob0 | My m-i-l is in the ground, so a bit late for that. |
| 17:06.41 | cpm | dang. Sorry |
| 17:06.45 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 17:06.53 | rob0 | So's my ex-m-i-l, and she was definitely no loss. |
| 17:07.10 | Greek-Boy | whats the best MTA then? |
| 17:07.17 | cpm | postfix |
| 17:07.18 | tzafrir | is .mil named after m-i-l ? |
| 17:07.21 | cpm | or netcat |
| 17:07.23 | cpm | :) |
| 17:07.28 | rob0 | netcat++ |
| 17:07.37 | cpm | that's two votes for nc |
| 17:07.55 | cpm | wait, deja-vu all over again! |
| 17:08.18 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 17:08.24 | bobkare | postfix is nice. exim isn't bad either |
| 17:09.24 | rob0 | Postfix is less likely to be hit by major security flaws, but yes, Exim has good points. |
| 17:09.30 | Greek-Boy | what spam utility do u prefer to use with postfix? |
| 17:09.55 | cpm | I like my spam fried, with scrambled eggs, , , and bacon on the side |
| 17:10.09 | Greek-Boy | lol |
| 17:10.12 | tzafrir | Greek-Boy, mailman? |
| 17:10.39 | rob0 | reject_non_fqdn_helo_hostname, reject_invalid_helo_hostname, reject_rbl_client zen.spamhaus.org |
| 17:10.40 | Greek-Boy | i'm looking for something that will generate a spam digest and e-mail a list of spam to a user and let him release false positive if he/she finds any |
| 17:10.58 | rob0 | Ah, I just want to reject the spam. |
| 17:11.50 | rob0 | check_cpm_access static:REJECT |
| 17:13.01 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.251) |
| 17:14.33 | CBU[^_^]M`` | how can i transfer calls if im using an analog phone on the FXS port? |
| 17:15.53 | tzafrir | CBU[^_^]M``, using flash |
| 17:16.13 | tzafrir | (a short disconnect, or the "flash" button on most phones) |
| 17:16.28 | mosty | CBU[^_^]M``, features.conf |
| 17:16.48 | tzafrir | that is also an option |
| 17:17.41 | CBU[^_^]M`` | thanks |
| 17:20.01 | *** join/#asterisk js_ (i=js@194.17.31.204) |
| 17:20.47 | js_ | right now i have a modem that answer calls to a certain number, and a script that plays a wave file and then makes an insert in a mysql database.. is it possible to replace this with sip and asterisk? |
| 17:21.00 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:21.28 | mosty | yes |
| 17:22.03 | reber | hi all. Has anyone used asterisk_gui of openwrt here ? |
| 17:26.41 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:27.07 | Greek-Boy | whose using a good billing system for asterisk? |
| 17:27.28 | rob0 | I do. I just send all the bills to cpm. ;) |
| 17:28.14 | rob0 | cpm: your payment is late. :( |
| 17:32.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:34.13 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 17:34.23 | *** join/#asterisk Arkonek (n=arkon@aadk54.neoplus.adsl.tpnet.pl) |
| 17:34.26 | mosty | wanpipemon segfaults on my amd64 box, is there anything special i need to do when building wanpipe on amd64? |
| 17:35.40 | unixdog | yes |
| 17:35.53 | unixdog | send sangoma a bug report email |
| 17:36.04 | unixdog | and alex will look ingot it on monday |
| 17:36.26 | unixdog | and in the bug report put the flavor of linux and the machine info |
| 17:36.38 | mosty | techdesk@sangoma.com? |
| 17:36.41 | Qwell | or stop buying hardware that requires silly binary modules :D |
| 17:36.53 | unixdog | support@sangoma.com |
| 17:37.01 | unixdog | is a good start |
| 17:38.23 | mosty | Qwell, wanpipe is binary only? why does it take so long to compile then? |
| 17:39.26 | Greek-Boy | so with regard to dns I guess you guys also prefer bind compared to tinydns? |
| 17:39.27 | unixdog | qwell is a digium whore.. there for its it's no digium its wrong |
| 17:40.10 | mosty | i tried digium pri cards, but they were impossible to debug. sangoma cards have been a lot easier until now |
| 17:40.13 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:40.42 | Arkonek | hi, i have a problem with mfcr2 on my box, can someone help with it? chan_unicall is loading, there is no problem. Bits are being set to idle (as it should be) but i recive only zeros form my teleco and they have DISA alarm :( |
| 17:41.31 | unixdog | well even asterisk still has a few issue on amd64 |
| 17:41.46 | unixdog | the world is dragging its feet changing over |
| 17:42.23 | file | unixdog: issues like what? |
| 17:42.31 | unixdog | I use sangoma because they officialy support bsd |
| 17:42.59 | unixdog | well on bsd . codec issues |
| 17:43.09 | unixdog | and having to do a lot of patching |
| 17:43.46 | _ys | unixdog are You see bsd sangoma architecture? |
| 17:43.55 | unixdog | ilbc and speex and gsm on amd64 have sounded like siince 1.4.10 on amd64 |
| 17:44.24 | unixdog | I have seen the brand new driver gettting ready to go into the ports tree |
| 17:44.30 | unixdog | I helped work on it |
| 17:44.33 | unixdog | and test it |
| 17:45.01 | unixdog | and I have a a200 and a 101a |
| 17:45.19 | unixdog | and they work better on bsd thne they do on linux I think |
| 17:46.12 | unixdog | but I have heard of issues on amd64 and ia64 with sangomas older drivers |
| 17:47.02 | unixdog | but not sure if they still exist in the latest driver.. I will have to crack open a amd4 with bsd64 and test |
| 17:47.36 | Arkonek | so? somone can help with this mfc/r2? :( |
| 17:47.40 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 17:48.40 | unixdog | I think chan_unicall is a third party addon |
| 17:48.53 | _ys | unixdog may be, http://lists.digium.com/mailman/listinfo/asterisk-bsd is for You? |
| 17:49.05 | unixdog | I am on the list |
| 17:49.17 | Arkonek | yes it is, but it |
| 17:49.18 | unixdog | I also have a project going called DaemonSwitch |
| 17:49.37 | unixdog | and it is about to have its first major iso |
| 17:49.39 | Arkonek | is chan or zaptelm problem if on the line they getting DISA? |
| 17:50.13 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 17:50.33 | ManxPower | Arkonek: we don't know. DISA is a MFC specific thing. |
| 17:50.45 | ManxPower | I would suggest searching the mailing lists. |
| 17:50.48 | ManxPower | ~mailinglist |
| 17:50.48 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:51.02 | unixdog | DISA in asterisk means direct inword system access |
| 17:51.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:51.12 | unixdog | its a function in the dial plan |
| 17:51.59 | Arkonek | :) i was searching and found no sollution |
| 17:52.08 | Arkonek | here it is distance service alarm |
| 17:52.33 | Arkonek | thanks for help i will try to ask somone from mfcr2 lib |
| 17:53.26 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 17:53.30 | unixdog | asterisk has come along way. and the fact it really almost works out of the box on bsd has made a big diff. |
| 17:53.45 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 17:53.55 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 17:53.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:54.04 | file | unixdog: does it not build out of the box on bsd? |
| 17:55.35 | unixdog | it does but with some small issues latly |
| 17:55.44 | unixdog | so we have been back to patching |
| 17:55.47 | file | like? |
| 17:56.01 | unixdog | looking for the current list of issues |
| 17:58.07 | unixdog | we have now 15 patch files in /usr/ports/net/asterisk |
| 17:58.24 | unixdog | that you can look to see what we patch to many to list |
| 18:00.15 | unixdog | we have now 15 patch files in /usr/ports/net/asterisk/files |
| 18:01.13 | unixdog | I am currently working to test the current patches today |
| 18:03.11 | unixdog | but bsd is also in a ports freeze right now so there may be new patches I have yet to see |
| 18:08.07 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:10.38 | *** join/#asterisk Greek-Boy (n=email@41.221.58.2) |
| 18:13.20 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 18:14.53 | Greek-Boy | aghhhh! these damn linksys spa942's. I tried everything I found on the wiki for remote provisioning but nothing works |
| 18:16.03 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:17.41 | *** join/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 18:18.42 | salviadud | you guys watching football? |
| 18:26.32 | *** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro) |
| 18:26.41 | alejandro | Good afternoon. |
| 18:27.07 | alejandro | I try to convert a wav file to gsm, but when asterisk plays the gsm file its sounds lower, any clue ? |
| 18:27.25 | puzzled | Greek-Boy, I have that working. lemme find the files I use |
| 18:28.52 | Greek-Boy | puzzled: i will be more than greatful :) |
| 18:29.18 | [TK]D-Fender | alejandro, convert how? |
| 18:29.43 | _ys | sox? |
| 18:29.46 | Greek-Boy | alejandro, check your gain settings on whatever converter u using? what u using btw |
| 18:29.48 | alejandro | yes, using sox |
| 18:30.08 | Greek-Boy | check your settings and paramaters |
| 18:30.18 | alejandro | wav sounds good but it's using 44k |
| 18:31.04 | Greek-Boy | remember, gsm is highly compressed |
| 18:31.48 | *** join/#asterisk Giofe (n=Giovanni@cliente37.amx.com.pe) |
| 18:32.01 | [TK]D-Fender | alejandro, you shouldn't be using 44khz with * |
| 18:32.35 | _ys | wny need sound files in compressed format? You have superfluous CPU time for transcoding? |
| 18:33.13 | puzzled | Greek-Boy, http://pastebin.ca/780908 Also check that the files are readable (chmod 644 <files>) and pass a few "-v" to tftpserver so you can see in /var/log/messages what is going on. And power the phone down, wait 30 secs and up again. I've seen a reboot not picking up any changes |
| 18:33.48 | alejandro | [TK]D-Fender: yes, i know, that it's why i'm converting, but sounds lower and i can't hear it well |
| 18:34.05 | [TK]D-Fender | alejandro, Why are you converting at all? |
| 18:35.28 | Greek-Boy | puzzled: thanks a lot |
| 18:35.33 | Greek-Boy | puzzled: which firmware u using? |
| 18:35.34 | alejandro | because I want to use with playback/background |
| 18:35.45 | puzzled | Greek-Boy, 5.1.8 |
| 18:35.53 | [TK]D-Fender | alejandro, just use the GSM, why bother converting? |
| 18:35.58 | puzzled | Greek-Boy, with a 941 by the way |
| 18:36.17 | alejandro | [TK]D-Fender: because I have a wav in 44khz, and * just plays 8khz |
| 18:36.40 | [TK]D-Fender | alejandro, Sorry, I got the order backwards... |
| 18:36.46 | Greek-Boy | puzzled: did u generate those templates from http://phone_IP/admin/spacfg.xml? |
| 18:36.48 | [TK]D-Fender | alejandro, nvm |
| 18:37.09 | puzzled | Greek-Boy, nope, copy paste from various places on the Net |
| 18:37.38 | Greek-Boy | hmmmm |
| 18:37.39 | Greek-Boy | i see |
| 18:37.53 | Greek-Boy | puzzled: have you tried using the SPC utility? |
| 18:38.17 | puzzled | nope, once it worked I thought it was better to no touch it anymore :) |
| 18:43.18 | Greek-Boy | puzzled: must the mac addres be in lower case? |
| 18:43.29 | puzzled | Greek-Boy, it is here and it works |
| 18:45.06 | Greek-Boy | k |
| 18:56.40 | *** join/#asterisk noway909 (n=vvv@scandic887.host.songnetworks.se) |
| 18:56.48 | noway909 | hi |
| 18:57.16 | noway909 | on cli,i do , sip show registry and see a sip phone to be registered |
| 18:57.28 | noway909 | but when i do sip show peers, the phone is not there |
| 18:58.00 | asdx | ~book |
| 18:58.00 | jbot | 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 |
| 18:58.23 | noway909 | the phone is pingable and "very correctly" configured with asterisk as a friend and currently being tested with a new wifi access point. the phone is a wifi phone. |
| 18:58.58 | mamep | hello, can someone help me with ooh323 channel |
| 19:00.30 | noway909 | <PROTECTED> |
| 19:00.30 | noway909 | <noway909> i want to know what is the criteria behind on adding these phones to the list. e.g. i have a wifi phone that is listed in registry but sometimes it appears in the sip show peers and then i can call it but other times it is not listed under sip show peers and i cannot call it but yet i can ping it at that point in time. |
| 19:01.19 | noway909 | sip show peers, i dont the see the ip of that sip phone in the list, but yet i can ping it, but cannot call it over sip. |
| 19:02.08 | noway909 | whats going on, it works fine with other wired sip phones. the wifi phone is stable as well. |
| 19:02.55 | *** join/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 19:03.41 | unixdog | ok 1.4.14 now compiling |
| 19:03.44 | unixdog | from ports |
| 19:03.47 | unixdog | we will see |
| 19:04.46 | mamep | is it possible to have user / pass authentication with ooh323? |
| 19:05.40 | [TK]D-Fender | noway909, "sip show registry" shows where ASTERISK has registered to. This is NOT a list of devices registered TO asterisk. |
| 19:05.57 | Greek-Boy | puzzled: good news man. I got it working with the SPC utility |
| 19:06.08 | asdx | [TK]D-Fender: what "registry" means? |
| 19:06.16 | asdx | in * |
| 19:07.45 | asdx | register* |
| 19:08.08 | asdx | Registration entails sending a REGISTER request to a special type of UAS known as a registrar. A registrar acts as the front end to the location service for a domain, reading and writing mappings based on the contents of REGISTER requests. This location service is then typically consulted by a proxy server that is responsible for routing requests for that domain. |
| 19:09.38 | salviadud | check this out. I have asterisk box a, and asterisk box b, box a can handle 50 channels, and when channel 51 comes up, it should go to box b |
| 19:09.58 | salviadud | should i setup an iax trunk to do that? |
| 19:11.38 | salviadud | you know, just a dial out to box b, with a context that dials what the channel wanted, but on the ther box, so i can, share the load? |
| 19:11.55 | [TK]D-Fender | "sip show registery" has NOTHING to do with SIP phones you want to have conenct to *. |
| 19:12.15 | [TK]D-Fender | it is for * registering to an **ITSP** |
| 19:13.16 | Strom_M | *************************************SEE?******************************************** |
| 19:13.37 | salviadud | Strom_M what do you mean by that? |
| 19:14.03 | Strom_M | it's just a question of time |
| 19:14.09 | Strom_M | it's running out for you |
| 19:15.41 | mamep | is it possible to have user / pass authentication with ooh323? |
| 19:18.43 | [TK]D-Fender | mamep, "unload chan_brokenrecord.so" |
| 19:20.06 | mamep | and? |
| 19:24.30 | mamep | [TK]D-Fender : i'm not sure if ooh323 supports user/pass authentication |
| 19:41.14 | puzzled | Greek-Boy, congrats! |
| 19:46.51 | *** join/#asterisk MoutaPT (n=mmouta@a213-22-40-62.cpe.netcabo.pt) |
| 19:48.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:56.55 | *** join/#asterisk timothywcrane_ (n=hanif@ip72-207-2-65.sd.sd.cox.net) |
| 19:57.58 | timothywcrane_ | I have to ask, I have a cable modem that I think is SIP comp. will I still need a PCI card to get POTS comp. with asterisk? |
| 19:59.22 | Greek-Boy | thanks puzzled |
| 20:01.10 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:01.38 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:01.41 | puzzled | Greek-Boy, y're welcome |
| 20:02.20 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 20:05.57 | MoutaPT | hi guys, anyone here experienced with heavy load using asterisk and redfone? |
| 20:07.05 | *** join/#asterisk PepOSX (n=pepOSX@190.72.146.88) |
| 20:09.35 | MoutaPT | <PROTECTED> |
| 20:09.58 | timothywcrane_ | let me google brb |
| 20:10.16 | MoutaPT | if it does, you just need to register your packetcable sip cablemodem as a sip endpoint on asterisk |
| 20:10.26 | MoutaPT | u don't need nothing else |
| 20:10.38 | MoutaPT | check the bootfile you r providing for ur cablemodem |
| 20:10.51 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 20:27.38 | Greek-Boy | puzzled: u back? |
| 20:28.49 | *** join/#asterisk hijacked (i=G3Ey@66.255.220.17) |
| 20:30.34 | tzafrir_home | mamep, I could not find such a a config option in that channel |
| 20:31.21 | mamep | anyway to use h323 channel with user / pass based authentication? |
| 20:32.37 | timothywcrane_ | how I can Icheck the bootfile, the web interface shows nothing interesting |
| 20:34.48 | timothywcrane_ | let me call Cox, though the last time I heard, they'll probably wnat to know what CHANNEL I have my computer TURNED TO, and do I want extra Musak with that lol |
| 20:36.55 | MoutaPT | timothywcrane_ SNMP walk |
| 20:37.01 | MoutaPT | the cable modem |
| 20:37.23 | timothywcrane_ | Surfboard SB5101 |
| 20:38.31 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 20:41.45 | timothywcrane_ | I'm slow, what is _SNMP walk |
| 20:43.23 | timothywcrane_ | well I called COX, the first lady hung up on me as soon as I asked my question lol, the second time, the rep told me he thought I would have to upgrade to a 5120 for SIP. I heard the 5100 works though, does this make sense. First time dealing with researching a PBX over cable system |
| 20:43.39 | timothywcrane_ | you guys have a paste bin/ lol |
| 20:43.42 | fujin_ | snmpwalk will walk snmp |
| 20:43.43 | fujin_ | ~pb |
| 20:43.44 | jbot | A Pastebin is a place to paste your stuff 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 |
| 20:44.14 | timothywcrane_ | well what do you know, I just got the paste bot message |
| 20:44.21 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:44.21 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:45.07 | blitzrage | giggity |
| 20:48.45 | timothywcrane_ | darn only rpm and exe. Advise: should I alien the rpm? |
| 20:48.53 | timothywcrane_ | for .deb |
| 20:49.27 | *** part/#asterisk MoutaPT (n=mmouta@a213-22-40-62.cpe.netcabo.pt) |
| 20:52.09 | timothywcrane_ | SVN pulls it all, everything they got, and when I browse I don't know what to pull, think I'll try to Alien bianary |
| 20:57.59 | *** join/#asterisk peanut- (n=tokarev@50ae.net) |
| 20:59.43 | *** join/#asterisk shido6 (i=shido6@74-130-126-198.dhcp.insightbb.com) |
| 21:00.32 | timothywcrane_ | crud alien refuses to convert the scropts |
| 21:02.37 | timothywcrane_ | now I have to search out each file to delete or make dir wont work with --scripts variable |
| 21:09.37 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 21:11.52 | timothywcrane_ | well got it to deb, but will not install, have to compile source. |
| 21:15.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:15.46 | *** join/#asterisk jtexter3 (n=jamest@nat.bloommg.com) |
| 21:16.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:17.55 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 21:19.50 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 21:23.01 | fujin_ | ugh |
| 21:23.07 | fujin_ | I generally don't deb asterisk. |
| 21:23.11 | fujin_ | or, binary it at all tbh |
| 21:23.14 | De_Mon | oh, it looks like he wasnt trying to do asterisk but something else? |
| 21:23.25 | fujin_ | it's one of those things I always have and will build from source |
| 21:23.26 | *** join/#asterisk `Sean (i=Sean@CPE002211569301-CM0011e6be76d9.cpe.net.cable.rogers.com) |
| 21:23.47 | De_Mon | you've had problems with packaged builds? |
| 21:24.00 | fujin_ | no, I just prefer to build |
| 21:24.06 | fujin_ | so that I can build asterisk-addons, aswell |
| 21:24.16 | fujin_ | and add stuff like russellb's func_devstate 1.4 backport |
| 21:24.28 | fujin_ | although, I do make use of the init script from Ubuntu's .deb :P |
| 21:24.45 | De_Mon | oh, haven't looked at asterisk-addons lately |
| 21:24.56 | fujin_ | app_addon_sql_mysql is quite handy |
| 21:25.04 | fujin_ | I prefer it to the odbc stuff. |
| 21:25.09 | De_Mon | func_odbc not good enough? |
| 21:25.32 | fujin_ | I dunno, I've never been able to get it working properly, or I'm just not persistent enough |
| 21:25.40 | fujin_ | app_addon_sql_mysql is easy |
| 21:25.52 | fujin_ | and a little more raw (which I prefer) than odbc |
| 21:26.10 | De_Mon | heh, odbc wasnt any trouble for me, guess I'm just good like that |
| 21:26.44 | fujin_ | I dunno, even with sane configuration, it wasn't happy |
| 21:26.48 | fujin_ | and I've no need for an abstraction layer |
| 21:34.38 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:35.32 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 21:37.54 | *** join/#asterisk ta^3 (n=tacvbo@201.128.232.238) |
| 21:40.27 | Greek-Boy | puzzled: Everything good with my SPA942's except I cant upload a ringtone to them. Any ideas? |
| 21:40.44 | fujin_ | Greek-Boy: there's a tool for doing it |
| 21:40.51 | fujin_ | clicky button magic |
| 21:40.57 | fujin_ | I don't recall where I found it, though. |
| 21:40.58 | Greek-Boy | but its for the old phone |
| 21:41.03 | Greek-Boy | i've got it |
| 21:41.03 | fujin_ | yes, it works anyway. |
| 21:41.14 | Greek-Boy | it generates the ringtone but doesn't upload it |
| 21:41.15 | Greek-Boy | i've tried it |
| 21:41.16 | fujin_ | THat's how I upload ringtones to all of mine |
| 21:41.39 | fujin_ | Greek-Boy: it starts up a tftp server on the local machine as a server, and the phone clients to it |
| 21:41.42 | fujin_ | are you behind a firewall at all? |
| 21:42.06 | *** join/#asterisk orsonork (n=orsonork@190.128.168.24) |
| 21:44.47 | asdx | orsonork: hi |
| 21:44.59 | orsonork | asdx: hello |
| 21:45.03 | orsonork | asterisk rules |
| 21:45.18 | asdx | orsonork: yes, it does ;) |
| 21:45.50 | *** join/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 21:46.02 | Greek-Boy | nope |
| 21:46.40 | salviadud | I got this huuuge question |
| 21:46.45 | Greek-Boy | fujin: is it called ringtone.exe? |
| 21:46.46 | salviadud | when I do sip show channels |
| 21:46.57 | salviadud | it tells me how many active sip channels i got right |
| 21:47.02 | fujin_ | Greek-Boy: I don't recall, it's for the SPA841 though |
| 21:47.14 | salviadud | so, if i want to run a simple bash script to evaluate that number... |
| 21:47.27 | salviadud | how do I write it? |
| 21:47.34 | fujin_ | google:bash scripting |
| 21:47.37 | fujin_ | && learn |
| 21:47.38 | fujin_ | && write |
| 21:47.42 | salviadud | i know bash |
| 21:47.50 | fujin_ | oh, you do? |
| 21:47.51 | salviadud | i don't know if bash can talk to the asterisk cli |
| 21:48.00 | fujin_ | It doesn't need to talk to the asterisk cli. |
| 21:48.11 | fujin_ | asterisk -rx "sip show channels"|tail -1 |
| 21:48.12 | salviadud | really? |
| 21:48.22 | orsonork | yes |
| 21:48.25 | salviadud | thanx fujin |
| 21:48.57 | fujin_ | asterisk -rx "sip show channels"|tail -1|awk '{print $1;}' |
| 21:49.22 | fujin_ | will parse that number |
| 21:49.25 | fujin_ | no bash involved |
| 21:49.26 | fujin_ | :P |
| 21:49.29 | salviadud | great! |
| 21:49.42 | salviadud | i can set that number to a variable |
| 21:49.51 | salviadud | then I can use the GotoIf app |
| 21:50.00 | fujin_ | indeed |
| 21:50.03 | salviadud | and say, if I got like 50 sip channels running |
| 21:50.18 | salviadud | number 51 would go through an IAX trunk |
| 21:50.27 | salviadud | and to another asterisk pbx |
| 21:50.44 | salviadud | I need to do a load balancing thingy, so this might work, thanx a lot fujin_ |
| 21:51.02 | fujin_ | There are probably *other* better ways to do that |
| 21:51.07 | fujin_ | with $GROUP, $GROUP_COUNT |
| 21:51.08 | fujin_ | iirc. |
| 21:52.23 | fujin_ | ${GROUP_COUNT(${MACRO_EXTEN}@agents)}=0 |
| 21:52.35 | Greek-Boy | fujin: so the PC u run it from needs to have a tftp server running? |
| 21:52.45 | salviadud | so, i would need to play with agents.conf for $group_count ? |
| 21:53.07 | fujin_ | salviadud: no, it's fine |
| 21:53.50 | fujin_ | that's just an example |
| 21:53.55 | fujin_ | Greek-Boy: no, the software spawns one |
| 21:55.00 | Greek-Boy | firewall is the problem |
| 21:55.08 | Greek-Boy | what port does TFTP use? 69 TCP and UDP? |
| 21:55.52 | JT | just a tip: hardly anything uses both TCP AND UDP on the same port |
| 21:56.15 | Greek-Boy | JT: sorry i meant TCP or UDP? |
| 21:56.26 | JT | ah |
| 21:58.09 | Greek-Boy | i think its TCP |
| 21:58.51 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:00.52 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:02.25 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:03.23 | De_Mon | what the heck are groups |
| 22:04.05 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:05.50 | lesouvage | I'm looking for the in asterisk 1.4 deprecated variable ${DNID} What is the replacement for this variable in 1.4? |
| 22:06.52 | remmo | heyya |
| 22:06.52 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:08.47 | *** join/#asterisk Tebi (n=tero@gw.aller.fi) |
| 22:09.22 | Greek-Boy | fujin |
| 22:09.29 | fujin_ | Greek-Boy: tftp is UDP (69) |
| 22:09.30 | Greek-Boy | i just learned a better way without using the tool |
| 22:09.33 | Greek-Boy | http://phone-ip/ringtone1?tftp://tftpserver-ip/ringtone1.dat |
| 22:09.34 | fujin_ | oh? do tell |
| 22:09.41 | fujin_ | That's handy. |
| 22:09.45 | Greek-Boy | works wonders |
| 22:09.47 | Greek-Boy | yeah |
| 22:09.50 | Greek-Boy | but |
| 22:10.01 | De_Mon | lesouvage ${CALLERID(dnid)} possibly |
| 22:10.04 | Greek-Boy | u still need the ringtone.exe utility to convert the tone to sipura format |
| 22:10.32 | Greek-Boy | The phone also gets the tone name from a header in the .dat file |
| 22:10.45 | Greek-Boy | the utility has some sort of signature that it also puts in the .dat |
| 22:12.43 | *** join/#asterisk LakeSolon (n=blake@12-202-201-70.client.mchsi.com) |
| 22:14.41 | lesouvage | De_Mon: thanks it's working. I was afraid that 330 phones would not be reachable tomorow |
| 22:15.53 | De_Mon | lesouvage good to hear |
| 22:15.59 | Greek-Boy | what was the problem lesouvage? |
| 22:17.06 | lesouvage | I have migrated to 1.4 this weekend, did a lot of testing but overlooked a scrip with de ${DNID} variable that is deprecated in 1.4. The script wasn't working anymore but is up and running again <;-) |
| 22:18.03 | Greek-Boy | oh |
| 22:18.06 | Greek-Boy | yeah |
| 22:18.23 | Greek-Boy | i made the same mistake upgrading to 1.4 from 1.2 without reading UPGRADE.txt |
| 22:18.24 | Greek-Boy | lol |
| 22:18.26 | Greek-Boy | big mistake |
| 22:18.33 | Greek-Boy | especially in a production environment |
| 22:20.48 | unixdog | ok whats going on major packet loss today |
| 22:20.58 | unixdog | 4 diff backbones |
| 22:23.12 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 22:23.20 | JayTee52 | man, I love Asterisk!!! |
| 22:24.03 | JayTee52 | the more I mess with the dialplan logic the more I think there's next to nothing that can't be done with this software. |
| 22:26.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:26.35 | *** join/#asterisk _theHub (n=_theHub@ool-43577a99.dyn.optonline.net) |
| 22:28.18 | *** join/#asterisk grimsy (n=chatzill@60-241-200-26.static.tpgi.com.au) |
| 22:31.13 | *** part/#asterisk unixdog (n=unixdog@adsl-69-234-184-228.dsl.irvnca.pacbell.net) |
| 22:34.55 | JT | JayTee52: you'll find it... eventually ;) |
| 22:35.45 | JayTee52 | well, yeah. I guess I shouldn't expect that it'll whip me up some scrambled eggs and bacon or wash my car for me :-) |
| 22:36.16 | JayTee52 | but coming from the Nortel Meridian telecom world, this is soooo much more flexible. |
| 22:38.42 | [TK]D-Fender | Scrambled eggs....very doable |
| 22:40.45 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 22:44.48 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:45.11 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 22:50.29 | JayTee52 | I installed a new TDM400P card to replace a bunch of old crap X100 cards and the call quality is much better without having to do any tweaking of echo cancellation. |
| 22:52.55 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:55.24 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 23:00.48 | De_Mon | hang on, I feel a surpprise face comming on |
| 23:02.10 | fujin_ | :O |
| 23:07.57 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 23:10.44 | JayTee52 | this sinus cold has me wanting to drill holes in my skull |
| 23:14.35 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 23:15.35 | fujin_ | Do it, put it on youtube. |
| 23:15.47 | JayTee52 | lol |
| 23:16.26 | JayTee52 | I still haven't seen the two girls, one cup video that's been all over the net yet. From what I hear I'm lucky. |
| 23:17.09 | fujin_ | Yes, I wouldn't suggest searching it out. Although, I can point you in the right direction |
| 23:17.12 | fujin_ | I think it's 2girls1cup.com? |
| 23:17.17 | fujin_ | Used to be on cupchicks.com aswell |
| 23:17.49 | hmmhesays | disgusting |
| 23:18.14 | hmmhesays | I was scarred for life after my girlfriend decked me for asking her to do that |
| 23:18.21 | fujin_ | ROFL |
| 23:18.29 | fujin_ | Why would you ask that? |
| 23:18.37 | fujin_ | "haha, this will be a funny joke" |
| 23:18.42 | fujin_ | O_o |
| 23:19.00 | hmmhesays | well obviously cause I had just wrapped the couch in plastic |
| 23:19.09 | fujin_ | dude |
| 23:19.29 | `Sean | JayTee52 try callweaver :) |
| 23:19.54 | JayTee52 | callweaver? I'll have to google it |
| 23:20.13 | hmmhesays | callweaver.org isn't it? |
| 23:20.19 | fujin_ | gah |
| 23:20.22 | fujin_ | derived from asterisk |
| 23:20.23 | fujin_ | = die in a fire |
| 23:20.26 | fujin_ | learn2asterisk |
| 23:20.30 | `Sean | ~callweaver |
| 23:20.30 | jbot | i guess callweaver is something that started off as a fork of Asterisk (b the name of openpbx), but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in Asterisk. see http://callweaver.org/ for more info, or join #callweaver |
| 23:20.45 | `Sean | fujin_ callweaver is better then you think. |
| 23:20.47 | `Sean | it was a fork |
| 23:20.49 | hmmhesays | i've used it in a few apps |
| 23:20.58 | hmmhesays | especially where faxing is involved |
| 23:21.08 | `Sean | its pretty good actualy |
| 23:21.14 | `Sean | less problems in crasterisk |
| 23:21.14 | `Sean | :) |
| 23:21.29 | fujin_ | allWeaver has emerged as the undisputed leader in T38 support. |
| 23:21.30 | fujin_ | lol |
| 23:21.32 | fujin_ | wonder who wrote that |
| 23:21.37 | `Sean | steve |
| 23:21.54 | hmmhesays | well that fact that he had a heavy hand in developing it I would say yes |
| 23:21.57 | JayTee52 | I'm probably going to stick with Asterisk now that my Microsoft worshipping boss has actually decided he likes it and will buy support from Digium and send me to a class or two. |
| 23:22.01 | `Sean | and fujin_ its true learn to see facts rather the open your mouth at things you dont know about :) |
| 23:22.13 | fujin_ | No thanks |
| 23:22.16 | fujin_ | #callweaver |
| 23:22.17 | fujin_ | eof |
| 23:22.34 | JayTee52 | I can't wait for 1.6 because it's supposed to have SIP/TCP support so I can dump sipX as a gateway to Exchange UM |
| 23:22.42 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 23:22.47 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 23:22.58 | hmmhesays | you're using sipx huh? there are a lot of tutorials using SER |
| 23:23.49 | JayTee52 | the tutorial I found used sipX and it worked. In the time frame I was dealing with (faster! faster! what's taking you so long?) it was the best option. |
| 23:24.34 | hmmhesays | that is obviously the only way to work isn't it? |
| 23:25.24 | JayTee52 | if you work for someone like my boss it is. He's always right even when he's dead wrong. He wouldn't know a kernel module from a kernel of corn stuck in his colon. |
| 23:25.56 | hmmhesays | I'm not sure if I want to drop 50 bucks on super mario galaxy |
| 23:26.03 | obnauticus | Anyoen else here having problems getting their MD5 string from VoipJet? |
| 23:27.20 | JayTee52 | I've gotta go to Walgreen's and find some better sinus meds. Crap I've been using isn't working. |
| 23:27.30 | JayTee52 | be back later. |
| 23:36.25 | salviadud | mario galaxy |
| 23:36.32 | salviadud | i think that game's gonna kick ass |
| 23:36.44 | fujin_ | street fighter world |
| 23:36.46 | fujin_ | looks awesome |
| 23:36.51 | fujin_ | www.streetfighterworld.com |
| 23:37.00 | fujin_ | and all the old streetfighters being remade in hidef 1080p for ps3 |
| 23:37.01 | fujin_ | =want |
| 23:39.30 | macTijn | lol |
| 23:41.30 | macTijn | mmm |
| 23:41.33 | macTijn | trailer looks nice |
| 23:43.09 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 23:43.48 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 23:47.46 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:49.10 | mosty | how do i tell asterisk 1.4 to pick a sip caller's preferred codec and not it's own, for an incoming call? |
| 23:52.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 23:52.47 | *** join/#asterisk icewaterman (n=immagine@i538742B6.versanet.de) |
| 23:54.58 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 23:55.19 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 23:58.25 | fujin_ | mosty: disallow=alll, allow=preferred_codec |
| 23:58.28 | fujin_ | under that callers definition |
| 23:58.41 | mosty | thanks |