| 00:00.02 | khlaed | p3nguin: no! say 30000-40000 |
| 00:00.08 | khlaed | is that possible? |
| 00:00.21 | p3nguin | heffer: Yes, and lvlolvlo told you which file to make the changes in. |
| 00:07.42 | p3nguin | wait... heffer, disregard. |
| 00:07.55 | p3nguin | khlaed: Yes, and lvlolvlo told you which file to make the changes in. |
| 00:12.52 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:23.56 | killown | brb |
| 00:25.19 | *** join/#asterisk sourcode (~code@ppp-58-8-229-221.revip2.asianet.co.th) |
| 00:28.41 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 00:35.12 | *** join/#asterisk wonderworld (~ww@port-92-201-211-226.dynamic.qsc.de) |
| 00:53.50 | *** join/#asterisk daxt (~daxt@112.135.77.194) |
| 00:54.04 | daxt | hi guys who is using mydivert here ? |
| 01:05.57 | *** join/#asterisk bjhaid (~abejide@41.220.69.19) |
| 01:06.40 | khlaed | where can i get pbxware support? |
| 01:06.50 | khlaed | #pbxware = empty |
| 01:06.51 | nightwalk | I've tried grepping the output of 'asterisk -d -vvvvvvv' for ERROR, and I've disabled all of the extraneous modules causing those lines. Something is still making this asterisk server exit 0, though. Any ideas on how to pinpoint what's wrong? It's 1.8.3, but the configs were ported over from 1.6.x, btw |
| 01:08.44 | nestAr | another question this SPA525G2, every so often, it unregisters, reloads the config and reregisters.. |
| 01:08.50 | nestAr | would that be related to |
| 01:09.14 | nestAr | <Resync_Periodic> |
| 01:09.16 | nestAr | ? |
| 01:10.33 | p3nguin | Note: there is no Asterisk 1.8.3 yet. |
| 01:11.08 | nestAr | that's got to be it.. |
| 01:13.05 | nightwalk | p3nguin: that's what rawhide calls it, anyway. It's probably a git checkout or something |
| 01:13.50 | nightwalk | In any case, the behavior is the same going all the way back to 1.8.0 |
| 01:14.53 | *** join/#asterisk daxt (~daxt@112.135.89.76) |
| 01:15.07 | p3nguin | nightwalk: If you launch asterisk with "asterisk -f -G asterisk -U asterisk", how long does it take before it dies and exits? |
| 01:17.58 | p3nguin | or if you wanted lots of verbosity, asterisk -vvvv -G asterisk -U asterisk |
| 01:18.11 | nightwalk | p3nguin: less than two seconds. It prints a bunch of diagnostic messages, but the only thing that looks like it might be cause for asterisk to exit is 'jack server is not running or cannot be started' |
| 01:22.17 | p3nguin | nightwalk: Do you have any jack modules? Are you trying to use anything related to jack? |
| 01:24.24 | nightwalk | Nope, don't need chan_oss, jack, etc. Not for this install, anyway |
| 01:25.37 | nightwalk | Disabling it doesn't help though, if that's what you're thinking. Just did it. |
| 01:26.44 | nightwalk | The weird thing is, it keeps trying to connect to an X server during startup. What the heck does it need X for? |
| 01:27.58 | carrar | back door |
| 01:28.06 | *** join/#asterisk daxt (~daxt@112.135.92.55) |
| 01:28.12 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:29.16 | riddlebox | when I try to make a call with gvoice with my gizmo number or my IPKALL number it fails, but it works from the call button on the google voice website? |
| 01:31.12 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:33.48 | p3nguin | riddlebox: Are you asking if that's what happens, or are you telling us? |
| 01:34.18 | riddlebox | p3nguin: sorry, that is what happens, has anyone else had this problem? |
| 01:34.42 | p3nguin | riddlebox: You're calling to your gv number from the PSTN, right? |
| 01:35.30 | nightwalk | Wow, kind of silly. I just found the culprit: "WARNING[16261]: pbx.c:5480 ast_register_application2: Already have an application 'Directory'" |
| 01:36.18 | nightwalk | I added 'noload's for two out of the three app_directory_*.so's, and it works now. Seems like that WARNING should actually be an ERROR if it's a fatal condition :/ |
| 01:37.15 | riddlebox | p3nguin: I have no problems calling my GV number from the PSTN, it rings to my gizmo account which is registered to my asterisk server, the issue I have is using the gvoice command where I use "gvoice -b -e name@gmail.com -p password call numbertobecalled ringbacknumber(my gizmo #) |
| 01:38.08 | p3nguin | riddlebox: And where do you run this gvoice -b ... command? |
| 01:38.27 | riddlebox | p3nguin: from my cli on my asterisk server |
| 01:39.13 | riddlebox | it is part of the pygooglevoice-0.5 package |
| 01:39.34 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:42.28 | p3nguin | riddlebox: I think you would be much better off upgrading your asterisk to the current version in the 1.8 branch and using chan_gtalk to call out directly rather than tinkering around with very old scripts that aren't needed anymore. |
| 01:43.20 | *** join/#asterisk daxt (~daxt@112.135.92.55) |
| 01:45.07 | riddlebox | p3nguin: I guess so |
| 01:46.29 | *** join/#asterisk Woinch (bd4ad49d@gateway/web/freenode/ip.189.74.212.157) |
| 01:46.42 | Woinch | boa noite, alguem aqui fala portugues? |
| 01:49.03 | p3nguin | woinch: Consider this: The channel's name in in English. The channel's topic is in English. |
| 01:49.24 | p3nguin | I can't understand why you would join us and then start speaking in a language which isn't English. |
| 01:49.30 | nestAr | lol |
| 01:50.35 | Woinch | i just only checking if there are some one who speaks my language! :) |
| 01:54.29 | *** join/#asterisk daxt (~daxt@112.135.66.240) |
| 01:58.40 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 01:58.59 | p3nguin | There's another one that needs JOINS/QUITS put on ignore. Ugh! |
| 02:03.32 | SaiSoma|AtHome | hey guys, anyone done any interfacing * with handheld radios? not HAM radio stuff, but standard handhelds, like consumer level motorola's? |
| 02:35.19 | drdru | ok created new sip.conf and extensions.conf - now I can make calls between two users, but I hear no audio |
| 02:35.36 | p3nguin | drdru: That's almost always a NAT problem. |
| 02:37.01 | drdru | I have nat=yes on both entries |
| 02:37.02 | drdru | http://pastebin.com/74C3Q8Er |
| 02:37.29 | p3nguin | But you don't have canreinvite=no on them. |
| 02:38.03 | drdru | what does that do |
| 02:38.19 | JerJer | does not attempt to optimize the audio path |
| 02:38.20 | p3nguin | And you pasted your sip.conf under where you were trying to show the extensions.conf. |
| 02:38.37 | p3nguin | Is asterisk directly connected, or also behind a NAT? |
| 02:39.43 | drdru | asterisk is on an Internet accessible host |
| 02:39.44 | drdru | so no nat |
| 02:39.48 | p3nguin | okay |
| 02:39.48 | drdru | but I am on a nat |
| 02:40.04 | drdru | and the two devices I'm calling between are on the same network |
| 02:41.00 | drdru | added canreinvite=no but still no audio |
| 02:41.18 | p3nguin | Did you run "sip reload" after you saved the changes in sip.conf? |
| 02:41.46 | drdru | I did service asterisk restart |
| 02:41.48 | JerJer | smells like a firewall or ALG issue |
| 02:45.18 | drdru | what ports need to be opened on the server? just 5060/udp ? |
| 02:45.30 | JerJer | not just no |
| 02:45.39 | JerJer | you first should have no firewall |
| 02:45.53 | p3nguin | You also have to have rtp ports open. Usually it's 10000-20000, but confirm it in rtp.conf. |
| 02:46.08 | JerJer | then once you know it works, start locking things down (as ISPs can block shit too) |
| 02:46.10 | p3nguin | No firewall? What kind of masochist are you? |
| 02:46.15 | drdru | udp? |
| 02:46.18 | p3nguin | yes |
| 02:46.29 | drdru | yeah, I'll probably keep the firewall enabled :-P |
| 02:46.37 | drdru | just want to know what ports I have to open |
| 02:46.37 | JerJer | good luck |
| 02:46.47 | drdru | k |
| 02:48.03 | drdru | sick |
| 02:48.10 | drdru | flushed the firewall and now I got audio |
| 02:48.11 | drdru | :) |
| 02:48.12 | drdru | thanks |
| 02:49.11 | JerJer | i blame firewall first |
| 02:49.28 | p3nguin | I usually blame the user. |
| 02:49.33 | JerJer | saves lots of time to know it works without one, then to spend all day finding out your ISP is blocking it |
| 02:50.00 | JerJer | i've worked with some pretty shady ISPs |
| 02:50.31 | drdru | so now I just need to figure out which ports to open |
| 02:50.37 | drdru | looks like rtp.conf says 10-20k |
| 02:50.55 | drdru | I forget my iptables syntax :( |
| 02:52.11 | p3nguin | probably something like iptables -I INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT |
| 02:53.46 | p3nguin | I usually will include the input interface, as well. -i eth0 for example |
| 02:54.48 | drdru | figured it out |
| 02:54.50 | drdru | I think you're right |
| 02:55.08 | p3nguin | Yeah, I wasn't guessing. |
| 02:55.13 | drdru | -A INPUT -m state --state NEW -m udp -p udp --dport 10000:20000 -j ACCEPT |
| 02:55.15 | drdru | hehe |
| 02:55.25 | drdru | I didn't see your message until I had already made the change |
| 02:55.30 | p3nguin | Exclude that state stuff. |
| 02:56.29 | drdru | why |
| 02:56.36 | p3nguin | For one, it's UDP. |
| 02:56.39 | p3nguin | That should be enough. |
| 02:56.41 | drdru | ok |
| 02:56.42 | drdru | true |
| 02:58.11 | drdru | I start a video call from one device, but I don't see video on the other device |
| 02:58.32 | drdru | I added videosupport=yes |
| 02:58.32 | drdru | to general |
| 03:09.14 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 03:12.51 | *** join/#asterisk dtoronto (~dtoronto@173-14-233-70-Utah.hfc.comcastbusiness.net) |
| 03:14.20 | dtoronto | Hi, setting up a distributed call center and want to use Tigase, do the advanced options work with console install and if so which options are required to run PubSub? |
| 03:16.30 | Sertys | huh |
| 03:16.39 | Sertys | that's supposed to be an asterisk chan, not xmpp |
| 03:16.49 | Sertys | or whatever else is tigase |
| 03:17.02 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup) |
| 03:17.23 | dzup2 | hello, is there a SIP phone that works of terminal ? |
| 03:17.38 | dtoronto | sorry, trying to set up a distributed asterisk install using an xmpp server for device state |
| 03:18.00 | p3nguin | dzup2: What does "works of terminal" mean? |
| 03:18.24 | p3nguin | dzup2: Are you possibly asking about a softphone that runs without a GUI? |
| 03:18.40 | dzup2 | like i like that phone to be use by a intercom so i can send input from a phone to dsp device |
| 03:18.50 | dzup2 | p3nguin: exactly |
| 03:19.06 | dzup2 | p3nguin: a sip without graphics |
| 03:19.20 | dzup2 | p3nguin: a sip phone without graphics |
| 03:20.17 | dzup2 | pbx1teq ~]# ekiga |
| 03:20.17 | dzup2 | (ekiga:4489): Gtk-WARNING **: cannot open display: <--because ekiga runs with Gtk, so i can not use that on my centos linux |
| 03:20.46 | dtoronto | also I have a centralized voicemail server but can't figure out how to get MWI working with the extensions, can anyone point me in the right direction to get that working? |
| 03:21.07 | p3nguin | You _could_ use ekiga on CentOS, but it does require graphics. |
| 03:21.11 | dzup2 | ...or if there is a way to send input audio from soundcard to the speakers connected to the soundcard, i dont know if possible that |
| 03:21.41 | Sertys | dzup2: just run a framebuffer |
| 03:21.44 | Sertys | on a Xvnc |
| 03:21.55 | Sertys | s/on/or/ |
| 03:22.13 | Sertys | infobot: yup |
| 03:22.13 | infobot | well, I feel sorry for you |
| 03:22.22 | p3nguin | lol |
| 03:22.24 | dzup2 | that way i can jackin my boombox to the JACK then output that as a moh, is that possible or is there any tutorial i can read? |
| 03:23.49 | Sertys | infobot: i appreciate some cares about me, even if it expresses in regret |
| 03:23.49 | infobot | You appreciate some cares about me, even if it expresses in regret? |
| 03:24.03 | Sertys | lol |
| 03:24.19 | dzup2 | Sertys: Xvnc but am not interest it in workingh my gui, i need to output the soundcard from my mic tyo spéaker ...that way if i install a softphone with auto-answer i can use my soundcard as a intercom, understand? |
| 03:25.40 | Sertys | dzup2: then use vnc to stream the linein and mpg123 with a custom MoH |
| 03:25.59 | dzup2 | hmm i follow up some tutorials ...i can get the audio alright thru out a .wav file, but i need a way to pip dsp thru speakears ... |
| 03:26.21 | dzup2 | pipe* |
| 03:26.26 | Sertys | u geeks are always overcomplicating |
| 03:26.41 | dzup2 | Sertys: srry, how is done? |
| 03:27.50 | dzup2 | mic -> soundcard -> ---> soundcard ->speakers ...in realtime |
| 03:28.02 | khlaed | where is rtp.conf located? |
| 03:28.30 | p3nguin | khlaed: Normally in /etc/asterisk/ along with ALL THE OTHER ASTERISK FILES. |
| 03:28.39 | dzup2 | ...that way i figure if i have a SIP phone with autoasnwer on ...i can answer the phone ...then my PAGE will sound out my CPU speakers ... |
| 03:29.55 | khlaed | i see. it's woomera.conf |
| 03:30.10 | p3nguin | What's a woomera? |
| 03:30.38 | khlaed | it's the replacement of rtp.conf for pbxware |
| 03:31.17 | p3nguin | Okay, but what is a woomera? |
| 03:31.46 | khlaed | maybe this will help - http://wiki.sangoma.com/WoomeraFreepbx |
| 03:32.03 | khlaed | or this Woomera may refer to: Woomera (spear-thrower), an Indigenous Australian spear-throwing tool. Woomera, South Australia |
| 03:37.40 | daxt | anyone who is using mydivert ? |
| 03:46.12 | khlaed | daxt: not me |
| 03:48.54 | *** join/#asterisk daxt (~daxt@112.135.87.115) |
| 03:52.47 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 03:58.40 | khlaed | is there a command to monitor calls from terminal? |
| 03:58.48 | p3nguin | Define "monitor" |
| 03:59.03 | khlaed | monitoring live calls |
| 03:59.06 | p3nguin | Define "monitor" |
| 03:59.23 | khlaed | see text output |
| 03:59.28 | p3nguin | Yes, there is. |
| 03:59.31 | khlaed | of logs |
| 03:59.33 | khlaed | live |
| 03:59.42 | p3nguin | sip show channels, sip show channel <channel> |
| 04:00.06 | p3nguin | core show channels [concise|verbose] , core show channel <channel> |
| 04:00.53 | p3nguin | Also, you can use sip debug to see the sip data during a call. |
| 04:06.21 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 04:06.40 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 04:07.30 | drdru | is it possible to capture audio from live call? |
| 04:07.37 | p3nguin | yes |
| 04:07.46 | drdru | "Our calls may be recorded for quality purposes"... |
| 04:07.55 | drdru | what about injecting audio? |
| 04:08.02 | p3nguin | yes |
| 04:08.07 | drdru | :) |
| 04:08.09 | drdru | how? |
| 04:08.31 | p3nguin | Do you actually want to record an active call, or just spy on it with your own ear(s)? |
| 04:08.58 | drdru | record and stream it somewhere else |
| 04:09.10 | drdru | (that stream won't be sip though) |
| 04:09.11 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 04:09.21 | p3nguin | If you want to set up the recording when the call it being started, use MixMonitor() in dialplan. |
| 04:09.49 | drdru | what will that do? |
| 04:09.59 | p3nguin | record the call |
| 04:10.37 | drdru | to a file? |
| 04:10.42 | p3nguin | yes |
| 04:10.42 | drdru | I want to have the live stream |
| 04:11.11 | p3nguin | Then you won't be recording it, you'll be spying on it. Use ChanSpy() or ExtenSpy() for that. |
| 04:11.25 | drdru | and⦠I want to capture each channel independently |
| 04:11.29 | p3nguin | However, you can record why spying. |
| 04:11.33 | drdru | cool |
| 04:11.37 | p3nguin | s/why/while/ |
| 04:12.01 | drdru | hah |
| 04:12.10 | drdru | s/hah/heh |
| 04:12.13 | drdru | oops |
| 04:12.16 | p3nguin | sed: -e expression #1, Unterminated `s' command |
| 04:12.18 | drdru | s/oops/oooops/ |
| 04:12.30 | drdru | funny bot |
| 04:12.48 | drdru | ok I still have 3 other problems I need to resolve |
| 04:12.57 | drdru | voicemail, external dialing, and video chat |
| 04:13.13 | drdru | I tried to setup voicemail.conf based on an example I found |
| 04:13.24 | p3nguin | ChanSpy() also has a whisper option to allow inserting audio into the channel you're listening to. |
| 04:13.29 | drdru | but it doesn't do anything when I try to call that user if that user is not logged in |
| 04:13.39 | drdru | p3nguin: sweet |
| 04:14.30 | p3nguin | Voicemail is going to be accessed through the dialplan. voicemail.conf doesn't affect if the user is present on the system or not. |
| 04:16.19 | drdru | http://pastebin.com/cUdpScQp |
| 04:18.49 | p3nguin | I can see why outbound calling doesn't work... |
| 04:19.00 | p3nguin | You don't have a SIP peer by the name of VoIPProvider to send calls to. |
| 04:19.29 | drdru | even calls to other sip providers? |
| 04:19.42 | drdru | I thought I only needed a voip provider to terminate to pstn |
| 04:19.46 | p3nguin | You don't have any listed. |
| 04:20.08 | drdru | so I need to have each sip provider listed? |
| 04:20.22 | *** join/#asterisk Smirker (~x@14-202-69-225.static.tpgi.com.au) |
| 04:20.39 | p3nguin | Your calling rules (known as extensions) show that you are sending calls through a peer by the name of VoIPProvider as configured in sip.conf. |
| 04:20.39 | Smirker | i've got a Voice Enabled Port service that is SIP based. the engineers there tell me i can register directly to the proxy without any user id/password and all my indials will go through that one registration |
| 04:20.45 | Smirker | any idea how to do that? |
| 04:21.07 | Smirker | (where my static IP address is my authentication) |
| 04:21.51 | drdru | p3nguin: ok, I just have that in because it was part of the example |
| 04:21.59 | p3nguin | Also, your voicemail mailbox names are okay, but it will be almost impossible for the people to enter alpha password when trying to check his or her voicemail. |
| 04:22.02 | p3nguin | oh |
| 04:22.56 | drdru | p3nguin: ok, I'll change the passwords - but the voicemail never kicks in - it just says the call is declined if I don't answer |
| 04:23.33 | p3nguin | drdru: On your Asterisk CLI, run "core set verbose 4" and then make a call to one of the phones. |
| 04:24.38 | drdru | ok |
| 04:24.40 | p3nguin | I'm also curious how you expect people to call other people with an obvious lack of numeric extensions. Most phones cannot dial names. |
| 04:24.58 | drdru | it's not a traditional phone service that I'm setting up |
| 04:25.02 | drdru | it's mobile |
| 04:25.05 | drdru | iphone based |
| 04:25.08 | p3nguin | I can't pick up my handset and dial "john". |
| 04:25.33 | drdru | yeah |
| 04:25.43 | drdru | is it possible to have alias extensions? |
| 04:25.47 | drdru | like 111 = john |
| 04:25.48 | drdru | ? |
| 04:25.59 | p3nguin | If you and all your users can, then I can work with that you have. |
| 04:26.17 | p3nguin | Yes, it's very simple: exten => 111,1,Goto(john,1) |
| 04:26.53 | p3nguin | Now extension 111 is John's extension. |
| 04:27.03 | carrar | dial "john" == dial 5646 |
| 04:27.18 | p3nguin | Then you'll need to create extension 5646. |
| 04:27.32 | p3nguin | Either way, it can work, but you need to make sure your users can dial the extensions you have. |
| 04:27.49 | drdru | sure |
| 04:27.52 | drdru | makes sense |
| 04:28.01 | carrar | drdru, you could create a prefix like dial 8 for dial by last name |
| 04:28.12 | p3nguin | That's how I decide on extensions... my name is Rob, so my extension is 762. |
| 04:28.13 | carrar | or just use the directory service |
| 04:28.57 | drdru | http://pastebin.com/mdFGvjfD |
| 04:29.18 | carrar | ~book |
| 04:29.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 04:29.31 | p3nguin | Yeah, I guess in the absence of numeric extens, the directory would still be capable of calling an alpha exten. |
| 04:30.00 | p3nguin | No application 'Voicemail' |
| 04:30.27 | p3nguin | Make sure you load app_voicemail.so and then try it again. |
| 04:30.44 | drdru | how? |
| 04:30.59 | p3nguin | module load app_voicemail.so |
| 04:31.35 | p3nguin | Are you using autoload for the modules? |
| 04:31.48 | drdru | I have no idea |
| 04:31.53 | drdru | this is a vanilla install |
| 04:31.58 | drdru | so is it on by default? |
| 04:32.06 | p3nguin | Well, there is no default. |
| 04:32.10 | drdru | autoload=yes |
| 04:32.13 | p3nguin | You have to actually create confs. |
| 04:32.17 | carrar | ~book |
| 04:32.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 04:32.23 | p3nguin | ~newbook |
| 04:32.23 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 04:32.29 | carrar | drdru, please go read |
| 04:33.03 | p3nguin | With autoload enabled, I would expect that your voicemail modules will load on each startup, as long as you have the conf for them. |
| 04:33.52 | drdru | I had no idea how powerful asterisk was |
| 04:34.15 | drdru | can't say I've had a real use for it until now |
| 04:34.16 | p3nguin | For the most parts, its limitations are your limitations. |
| 04:34.16 | carrar | SUPAH POWER |
| 04:34.21 | drdru | but this thing is pretty amazing |
| 04:34.35 | carrar | It's BEYOND amazing |
| 04:34.35 | drdru | the tiny amount I've learned so far |
| 04:34.37 | drdru | is pretty cool |
| 04:34.46 | p3nguin | How about that book? What do you think of it? |
| 04:35.05 | drdru | wow |
| 04:35.33 | drdru | is it possible to set the language of your sip phone, so the server knows what language to talk to it? |
| 04:35.41 | carrar | too bad you can't search the new book |
| 04:35.43 | drdru | (for multilingual systems) |
| 04:35.55 | carrar | yeah for pdf's |
| 04:35.58 | carrar | yay |
| 04:36.33 | p3nguin | I haven't configured multilingual Asterisk, so I'm not sure. I know you can set the language globally, though. |
| 04:36.37 | drdru | seems like a web-based configuration app would be a real nice thing to have for asterisk |
| 04:36.43 | drdru | there has to be one |
| 04:36.52 | carrar | no web thanks |
| 04:36.55 | drdru | but it would probably require patching? |
| 04:36.59 | p3nguin | There are some, but they ruin Asterisk. |
| 04:37.05 | carrar | go create it |
| 04:37.19 | p3nguin | the Asterisk GUI and FreePBX are the two most common web GUIs, I guess. |
| 04:37.46 | p3nguin | FreePBX totally ruins the power of Asterisk. I can't say about Asterisk GUI. |
| 04:42.39 | drdru | damn |
| 04:43.21 | drdru | I uncommented load => app_voicemail_plain.so |
| 04:43.26 | drdru | restarted asterisk |
| 04:43.51 | drdru | still getting No application 'Voicemail' |
| 04:44.00 | p3nguin | If you are using autoload, you really don't need to be manually loading modules until you know what you're doing. |
| 04:44.53 | p3nguin | Make sure there is no noload for app_voicemail. There isn't really any reason it would be in there, but since you obviously never read the book, there's no telling what is actually in the file. |
| 04:46.27 | drdru | k |
| 04:46.41 | ChannelZ | what is app_voicemail_plain ? |
| 04:46.47 | p3nguin | How about you do what I said to load the module and see how that works out? |
| 04:47.03 | p3nguin | (2230.50) <p3nguin> module load app_voicemail.so |
| 04:47.31 | p3nguin | If that does work, then maybe we can figure out why it isn't autoloading. |
| 04:48.46 | drdru | No such file or directory |
| 04:49.03 | drdru | [Feb 18 04:48:35] WARNING[3969]: loader.c:387 load_dynamic_module: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: cannot open shared object file: No such file or directory |
| 04:49.17 | p3nguin | find /usr/lib/asterisk -name \*voicemail\* |
| 04:49.37 | drdru | no folder /usr/lib/asterisk |
| 04:49.47 | p3nguin | maybe directory? |
| 04:50.07 | Smirker | howdy |
| 04:50.18 | Smirker | is it possible to register to a SIP proxy without a user id/any kind of credentials? |
| 04:50.33 | drdru | sorry, directory is there, but there is no app_voicemail.so inside the modules directory |
| 04:50.34 | ChannelZ | what would the "register" do then? |
| 04:50.35 | Smirker | apparently i have a service with multiple inbound/outbound numbers that authenticates me based on IP address |
| 04:50.47 | p3nguin | /etc/asterisk/asterisk.conf configures were asterisk looks for files. If you aren't using the normal paths, you'd better reconfigure asterisk to use the paths where stuff really is. |
| 04:50.51 | Smirker | but i can't register to it as it gives me authentication errors |
| 04:51.03 | Smirker | however the voice "engineers" at the company tell me i can do it without a user id |
| 04:51.06 | Smirker | which confuses me. |
| 04:51.15 | carrar | based on IP |
| 04:51.24 | ChannelZ | static IP |
| 04:51.30 | Smirker | yeah. |
| 04:51.44 | ChannelZ | that's on their end |
| 04:51.45 | carrar | excellent for hacking free service |
| 04:51.55 | carrar | unless the SIP is over a VPN |
| 04:51.59 | p3nguin | drdru: You're going to have to get the voicemail stuff, either by getting a package or go compile the voicemail module from source. How did you install asterisk? |
| 04:52.01 | Smirker | eventually it will be onnet |
| 04:52.06 | ChannelZ | perhaps they are matching you by your peer name (From) |
| 04:52.11 | drdru | yum install asterisk |
| 04:52.12 | drdru | I thikn |
| 04:52.25 | Smirker | potentially |
| 04:52.33 | Smirker | what would my register statement look like without a user id, though? |
| 04:52.44 | p3nguin | drdru: Did you at least use the official asterisk packages from whatever repo they exist in? |
| 04:53.03 | p3nguin | smirker: If you are using IP auth, there's no reason to register. |
| 04:53.05 | carrar | I swear there was a book out there |
| 04:53.11 | carrar | about asterisk |
| 04:53.17 | Smirker | p3nguin: how would i receive phone calls without registering? |
| 04:53.25 | carrar | PDF no less |
| 04:53.27 | p3nguin | smirker: by IP, obviously |
| 04:53.29 | drdru | p3nguin: I can't find the entry in my bash history to see how I installed it |
| 04:53.30 | drdru | crap |
| 04:53.30 | carrar | and HTML |
| 04:53.47 | Smirker | p3nguin: i never thought of that :) |
| 04:53.50 | ChannelZ | books are for squares |
| 04:53.59 | drdru | p3nguin: I'm pretty sure I did yum install asterisk |
| 04:54.02 | p3nguin | smirker: SIP registration is for telling the other side how to contact your system. If the other side already knows how to reach you by static IP address, you have no reason to register. |
| 04:54.10 | drdru | it defaults to the fedora core repo |
| 04:54.58 | Smirker | p3nguin: thanks, that clears up some questions. i'll try that out then. |
| 04:55.03 | Smirker | i may, however, be back. |
| 04:57.09 | drdru | asterisk-1.6.2.16.1-1.fc14.i686 |
| 04:57.12 | p3nguin | drdru: I'm finding it hard to believe that the voicemail module would be excluded in the asterisk package. If that really is the case, check for another package to provide voicemail. |
| 04:57.59 | p3nguin | "yum search asterisk" should show you all asterisk stuff. Look for voicemail. |
| 04:58.27 | drdru | asterisk-voicemail-plain.i686 : Store voicemail on the local filesystem |
| 04:58.35 | drdru | it also has -imap and -odbc |
| 04:58.43 | drdru | probably start with plain |
| 04:59.01 | p3nguin | Install that sucker. Also, take a look at the package contents. |
| 04:59.25 | drdru | it installed app_voicemail_plain.so |
| 04:59.33 | p3nguin | After you see that it has a voicemail module and it is in /usr/lib/asterisk/modules/ restart asterisk. |
| 04:59.59 | p3nguin | Then you can use "module show like voicemail" to see what is loaded. |
| 05:00.44 | drdru | says it's loaded |
| 05:00.45 | p3nguin | Make sure you undo anything you did in modules.conf. |
| 05:01.23 | drdru | I did |
| 05:01.25 | drdru | [Feb 18 05:00:51] WARNING[4070]: file.c:650 ast_openstream_full: File vm-theperson does not exist in any format |
| 05:01.31 | drdru | [Feb 18 05:00:51] WARNING[4070]: file.c:956 ast_streamfile: Unable to open vm-theperson (format 0x4 (ulaw)): No such file or directory |
| 05:01.35 | p3nguin | Okay, so now you're missing sounds. |
| 05:01.39 | p3nguin | Install sounds packages. |
| 05:02.55 | *** join/#asterisk tessier (~treed@kernel-panic/copilotco) |
| 05:03.46 | tessier | Hello all. My box gets hammered by people trying to brute force SIP accounts all the time. I just firewall them off. But today someone has been hammering away on a valid extension. Normally it is random guessing at common extensions and passwords. How could they know that particular extension (397) exists? |
| 05:04.13 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 05:04.43 | p3nguin | tessier: Let's start by using asterisk terminology. Do you mean they are trying to bruteforce a sip device by the name of 397, or do you actually mean they are calling extension 397? |
| 05:05.46 | tessier | p3nguin: Not sure. Which would this be? chan_sip.c: Registration from '"397" <sip:397@216.105.40.123>' |
| 05:05.49 | tessier | failed for '129.206.120.28' - Wrong password |
| 05:05.58 | tessier | Device, I think. |
| 05:06.04 | p3nguin | tessier: That looks like a device to me. |
| 05:06.19 | p3nguin | tessier: Do you have alwaysauthreject set to yes in your sip.conf? If not, they could use that to figure out what does and doesn't exist. |
| 05:07.21 | p3nguin | tessier: One very common mechanism to use to block out these repeated attempts is fail2ban. You configure it to add blocking rules to iptables after several bad auths. |
| 05:07.29 | tessier | p3nguin: I do not have that option specified at all. What is the default? It looks like it defaults to yes in the config. |
| 05:07.48 | tessier | p3nguin: Right. I have been trying to set up fail2ban for a couple of days. It isn't banning for some reason. |
| 05:08.10 | *** join/#asterisk baldrailers (~Adium@27.110.205.119) |
| 05:08.27 | p3nguin | tessier: By default, it is most likely set to no. That allows existing and non-existing devices to return different answers on failure. If you set that to yes, all failures return the same thing, so there is no way to know what does and does not really exist. |
| 05:08.59 | p3nguin | tessier: Is it doing anything at all? Can you pastebin iptables -L for me to look at? |
| 05:09.30 | tessier | Yeah, looks like it defaults to no from some googling. The example in the config has it set to yes but commented out which usually indicates a default. I guess not in this case. |
| 05:09.33 | tessier | enables it |
| 05:12.22 | p3nguin | I don't see any reason it shouldn't be enabled by default, but it's not. |
| 05:12.40 | tessier | As for fail2ban...Here's my iptables -L: http://copilotco.com/paste/104/ |
| 05:12.47 | tessier | Would love any help you can provide on debugging it. |
| 05:14.10 | p3nguin | Alright, it's clear that you didn't configure fail2ban for asterisk. |
| 05:14.19 | tessier | I am trying for just ssh first. |
| 05:14.23 | tessier | I haven't even attempted asterisk. |
| 05:14.37 | p3nguin | http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 05:14.44 | tessier | Yes, I've already read it. Thanks. :) |
| 05:14.59 | p3nguin | You'll basically just copy and paste from this page into your system. |
| 05:15.19 | p3nguin | Unless someone changed it recently, it's a good page. |
| 05:21.57 | tessier | Actually...maybe fail2ban is working...I thought that since I am not seeing any entries in my iptables -L that it wasn't banning anything. But I see that it is unbanning after just 10 minutes... |
| 05:22.50 | p3nguin | You can adjust the time as needed. |
| 05:24.17 | tessier | I just did. |
| 05:24.22 | p3nguin | If you see the fail2ban-ASTERISK chain and a jump to it from INPUT chain, there's a chance it's going to be working correctly. You need to ensure that the log format is fixed like it needs to be before it will read anything, though. |
| 05:28.09 | *** join/#asterisk daxt (~daxt@112.135.91.81) |
| 05:32.11 | *** join/#asterisk tyman (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 05:32.20 | *** part/#asterisk tyman (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 05:32.26 | tessier | Yeay, fail2ban seems to be working. I have the chains anyway. I just increased the ban time so I'll see some iptables entries. |
| 05:34.10 | fauxalliance | tessier, PortSentry is equally awesome |
| 05:34.48 | *** join/#asterisk tyman_ (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 05:36.53 | *** join/#asterisk tyman (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
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| 05:40.54 | p3nguin | fauxalliance: Does it also write rules in iptables? |
| 05:44.09 | fauxalliance | p3nguin, socket binding? Packet filter? KILL_ROUTE! |
| 05:44.57 | fauxalliance | p3nguin, it 'compliments' fail2ban |
| 05:45.15 | p3nguin | It says nice things to fail2ban? |
| 05:45.24 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 05:45.48 | fauxalliance | p3nguin, i dunno... ask logsentry |
| 05:45.56 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 05:46.08 | p3nguin | Perhaps it complements fail2ban? |
| 05:46.35 | fauxalliance | tight...tighter...tightest |
| 05:47.16 | fauxalliance | p3nguin, can fail2ban detect and block stealth scans? |
| 05:47.39 | p3nguin | I can't say for sure. |
| 05:47.49 | fauxalliance | p3nguin, the answer is no... it cannot |
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| 06:22.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 06:37.36 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 07:13.40 | drdru | damn, still having trouble with voicemail - it can't find the sound files |
| 07:13.42 | drdru | Feb 18 07:12:49] WARNING[4390]: file.c:650 ast_openstream_full: File vm-theperson does not exist in any format |
| 07:13.43 | drdru | [Feb 18 07:12:49] WARNING[4390]: file.c:956 ast_streamfile: Unable to open vm-theperson (format 0x4 (ulaw)): No such file or directory |
| 07:14.29 | drdru | but I have g711a, g711u, g723, g729, sln, wav for en_GB vm-theperson |
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| 07:27.06 | eMBee | good afternoon |
| 07:27.51 | eMBee | has a problem with a calling queue that occasionally does not hang up a line so that the line appears busy continuously |
| 07:29.14 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:29.22 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:30.37 | p3nguin | embee: What do you mean that the queue does not hang up a line? |
| 07:32.34 | eMBee | it means that despite the call being terminated asterisk keeps the status of the channels involved as Up |
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| 07:39.08 | dzup2 | ok guy now i have sound of my mic jack to speaker on the same soundcard device, now how do i send the sip/ext input to that little program? |
| 07:39.15 | dzup2 | ok guys now i have sound of my mic jack to speaker on the same soundcard device, now how do i send the sip/ext input to that little program? |
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| 07:46.56 | jkroon | hi guys, is it possible to publish srv records for iax similar to that of sip? eg, for sip I can publish _sip._udp.${domain} to inform it where to register to, assign priorities and others, is something similar possible for IAX/2 |
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| 07:56.47 | drmessano | http://www.voip-info.org/wiki/view/DNS+SRV |
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| 07:58.10 | jkroon | drmessano, thanks. a simple yes would have also been sufficient and now i feel like an id10t error anyway. |
| 07:59.03 | titter | jkroon: I have successfully had IAX setup this way, but if you are using it for failover purposes, it doesn't work as expected. At least for me it would never jump to the next record if the first became unreachable |
| 07:59.28 | kaldemar | dzup2: is it a must that the announcement comes out of the intercom when spoken? |
| 08:01.01 | drmessano | Um ok |
| 08:01.27 | *** join/#asterisk weinerk (~irc@bzq-79-183-206-242.red.bezeqint.net) |
| 08:04.05 | weinerk | hi |
| 08:04.20 | weinerk | Hi, can someone help re: asterisk AGI male/female in another language? |
| 08:04.43 | weinerk | Asterisk ver 1.4.33, |
| 08:04.43 | weinerk | If I try this: [SAY NUMBER 123 ""] - it counts in female gender. |
| 08:04.43 | weinerk | If I try this: [SAY NUMBER 123 "" "m"] - I get an error: |
| 08:04.43 | weinerk | 520-Invalid command syntax. Proper usage follows: |
| 08:06.46 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:06.48 | schmidts | good morning |
| 08:10.48 | jkroon | drmessano, interesting question that has cropped up now, let's say I've got two servers at two different priorities and a qualify=100, now lets say the latency on the preferred priority goes to 150ms, will asterisk then automatically jump to the alternate server? |
| 08:11.53 | jkroon | titter, I take it you're referring to the scenario I just pointed out directly above. I can't imagine it won't fail over if the upstream stops responding entirely. |
| 08:13.25 | titter | jkroon: I briefly looked at the scenario, but I couldn't get it to work the way I wanted ... sip works fine, iax just didn't play as nicely |
| 08:16.20 | jkroon | interesting. i happen to have it such that I can test at the moment, my secondary seems to be behind a bit of insane latency (albeit with low jitter) |
| 08:16.33 | jkroon | so I can probably see if it does or doesn't. will take some effort though. |
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| 08:25.28 | ChannelZ | weinerk: I don't even think gender is implemented |
| 08:29.25 | justdave | is there a way to trigger an event to happen in the background via dialplan? Like if a certain number gets dialed, I want it to dial a different number and play a recording to whoever answers, but still place the original call without waiting for the second dialout to complete |
| 08:30.10 | drmessano | Originate should do that |
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| 08:53.50 | titter | jkroon: I was stopping the Asterisk service on my primary pbx before placing into production to cause the trunk to fail. The far ends would never attempt to try the next record, rather they would sit as unreachable. I believe if I reloaded iax it would then go to my secondary. I tried messing with some of the expiry settings, but didn't get it to work. The redundancy for iax wasn't needed for production, so I haven't messed with it si |
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| 09:08.11 | WIMPy | Can anyone tell me how to pass a parameter containing commas to Macro()? |
| 09:10.58 | mzb | ',,,,,,,,' ? |
| 09:11.40 | WIMPy | No, and \, doesn't work either. |
| 09:11.54 | WIMPy | Which seems to work in applications |
| 09:12.25 | WIMPy | Which seems to work in other applications. |
| 09:12.48 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:22.15 | kaldemar | WIMPy: use Gosub and "" around the argument. |
| 09:22.29 | kaldemar | app Macro doesn't seem to care about such things but Gosub does. |
| 09:23.09 | WIMPy | Interesting to see that things get parsed differently. |
| 09:23.23 | WIMPy | Will try Gosub. |
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| 09:27.18 | WIMPy | Yes that works. But has very interesting side effects. |
| 09:27.19 | kaldemar | it picks the "" (or \ if \, is used) into the argument value, but they can be easily removed with func FILTER. |
| 09:27.58 | kaldemar | like ${FILTER(a-zA-Z \,,${ARG1})} or something similar. |
| 09:28.35 | WIMPy | Dial doesn't seem to get upset about the quotes. |
| 09:28.52 | WIMPy | But I get s as caller ID now, which is pretty interesting. |
| 09:35.21 | *** join/#asterisk sacoetzee (~s_a_coetz@dsl-241-228-69.telkomadsl.co.za) |
| 09:35.41 | sacoetzee | hey everybody |
| 09:37.36 | sacoetzee | strange thing happening with a couple of SNOMs at a client, this place has about 20-30 extensions, but 5 or 6 of them are forever coming up as forbidden with attempting to transfer. Ast version is 1.6.2.16.1. Anyone ever seen this before? |
| 09:39.50 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:41.35 | WIMPy | This caller ID thing is freaking me out sinve weeks. The interesting thing is that core show channels verbose also shows the dialed extension as caller ID on the (Outgoing Line), wich I think it |
| 09:41.40 | WIMPy | didn't do before. |
| 09:41.57 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 09:43.26 | *** part/#asterisk sacoetzee (~s_a_coetz@dsl-241-228-69.telkomadsl.co.za) |
| 09:46.33 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:55.26 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:55.32 | eMBee | good afternoon |
| 09:57.30 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 10:01.23 | *** join/#asterisk shamelessn00b (~chatzilla@203.175.64.42) |
| 10:01.39 | shamelessn00b | Hi, can anyone help me out regarding a nat related issue? |
| 10:01.51 | shamelessn00b | I'm unable to register a sip client on my asterisk server hosted on ec2 |
| 10:02.54 | shamelessn00b | output of sip show settings |
| 10:02.55 | shamelessn00b | http://pastebin.com/MbYL190N |
| 10:04.17 | shamelessn00b | I'm using asterisk 1.8.2.3 |
| 10:07.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d96a:2770:ee73:cd39) |
| 10:11.14 | kaldemar | shamelessn00b: are you getting some error? |
| 10:12.49 | plundra | Anyone familiar with SPA9xx-devices? I've configured two accounts and while testing on my (942), I can dial both extensions/accounts as expected. No matter if one of them is in use or whatever. But on ANOTHER 942, with the same firmware and no significaant difference in configuration. I get BUSY directly when trying to dial the secondary extension on the phone. |
| 10:13.29 | plundra | I've done a diff on the spacfg.xml files on both phones, and apart from different network-settings and (of course) accounts, it's the same. |
| 10:14.20 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 10:15.33 | joobie | shamelessn00b, you have a asterisk server in the ec2? i thought ec2 was just storagE?? |
| 10:16.29 | shamelessn00b | ebs is the storage |
| 10:16.43 | joobie | can you have ISDN termination |
| 10:16.44 | shamelessn00b | kaldemar: no, i'm not even getting anything on sip debug |
| 10:16.45 | joobie | in wc3 |
| 10:16.51 | shamelessn00b | on my asterisk box |
| 10:16.51 | joobie | like do they let you put an ISDN card on ur box |
| 10:17.43 | shamelessn00b | http://pastebin.com/9s0qq6Cv |
| 10:18.00 | shamelessn00b | output of netstat -nap |grep asterisk |
| 10:18.19 | joobie | sip show settings |
| 10:18.28 | shamelessn00b | http://pastebin.com/MbYL190N |
| 10:18.32 | shamelessn00b | already pasted that |
| 10:18.53 | joobie | sip set debyg on |
| 10:18.59 | joobie | then try connect |
| 10:19.04 | shamelessn00b | no output |
| 10:19.12 | shamelessn00b | on console |
| 10:19.13 | joobie | tcpdump |
| 10:19.17 | joobie | the int |
| 10:19.21 | joobie | do u see ur ip hit the box |
| 10:19.23 | kaldemar | shamelessn00b: then your problem is not related to asterisk. either you have misconfigured the client or you have a network issue. |
| 10:19.25 | joobie | ? |
| 10:20.01 | shamelessn00b | would it be a possibility that my ISP is filtering udp traffic on port 5060 |
| 10:20.15 | joobie | it could |
| 10:20.16 | joobie | be |
| 10:20.19 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 10:20.23 | joobie | it could also be that a rat ate your cable |
| 10:20.28 | shamelessn00b | lol |
| 10:20.28 | joobie | BUT.. try tcpdump |
| 10:20.35 | shamelessn00b | then I won't be talking to you |
| 10:20.39 | joobie | perhaps |
| 10:20.49 | joobie | tcp has error control |
| 10:20.51 | joobie | udp doesnt |
| 10:20.51 | joobie | :P |
| 10:20.56 | shamelessn00b | ok, lemme run tcpdump |
| 10:21.01 | joobie | youuu couuulld bee laggeddddd |
| 10:21.05 | joobie | ;P |
| 10:21.08 | joobie | im just fuken aroudn bro |
| 10:21.11 | joobie | had a scotch |
| 10:21.11 | joobie | or two |
| 10:21.14 | joobie | so bitt waked. |
| 10:21.16 | shamelessn00b | figured :p |
| 10:21.17 | joobie | bout to tgame tho |
| 10:21.21 | joobie | try tcpdump |
| 10:21.23 | shamelessn00b | DotA? |
| 10:21.25 | joobie | that will tell u if ur isp is filtering |
| 10:21.29 | joobie | or if ur phone is misconfigured |
| 10:21.30 | joobie | wtf |
| 10:21.32 | joobie | how did u kno? |
| 10:21.33 | joobie | wc3 |
| 10:21.34 | joobie | dota |
| 10:21.52 | shamelessn00b | you mentioned wc3, I've been DotAing since 6.35 |
| 10:21.54 | shamelessn00b | lol |
| 10:21.55 | joobie | hahah |
| 10:21.58 | joobie | wana game ?:P |
| 10:22.03 | joobie | u from AU? |
| 10:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:32.01 | kaii | when asterisk is faced with a "SIP OPTIONS" message from a peer that is directed to its ip-address (e.g. "To: <sip:10.0.0.65>") asterisk responds with "404 Not Found". |
| 10:32.22 | kaii | i would like asterisk to respond OPTIONS with 200 OK .. but how? |
| 10:34.48 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
| 10:34.50 | Seb^ | Hi |
| 10:35.01 | Seb^ | I restarted my asterisk box and now my pri card has vanished |
| 10:35.04 | Seb^ | (not physically) |
| 10:35.28 | Seb^ | if I try to do pri show spans it errors "No such command 'pri show spans' (type 'help pri show' for other possible commands)" |
| 10:35.35 | Seb^ | dahdi services are running |
| 10:35.39 | Seb^ | can anyone help? |
| 10:35.56 | Seb^ | I've not made any config changes |
| 10:36.01 | Seb^ | and ive restarted it before just fine. |
| 10:36.18 | Seb^ | the card has a green light on the back |
| 10:36.22 | Seb^ | as if the span is up |
| 10:36.28 | Seb^ | but asterisk doesnt see it. |
| 10:41.16 | Seb^ | the system can see the card |
| 10:41.16 | Seb^ | 0d:03.0 Communication controller: Digium, Inc. Wildcard TE210P/TE212P dual-span T1/E1/J1 card 3.3V (rev 02) |
| 10:41.30 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:43.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:44.05 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 10:45.33 | benngard | any that can give me a hand with AddQueueMember? |
| 10:45.48 | benngard | i have : [agents] |
| 10:45.48 | benngard | exten => 647,hint,SIP/0424449647&SIP/0106024647 |
| 10:45.58 | benngard | core show hints, works |
| 10:46.21 | benngard | but, AddQueueMember(Helsinborg,Local/647@agents,1,,Lisbeth Mingert Nilsson,SIP/0424449647&SIP/0106024647) dont do what i excpect |
| 10:49.51 | kaii | Seb^: seems like asterisk was unable to load chan_dahdi .. enable verbosity and try "module reload chan_dahdi.so" |
| 10:50.04 | kaii | it should give you an error |
| 10:50.45 | kaii | benngard: well.. what do you expect it to do? |
| 10:51.11 | benngard | call both sip phones when i dial "helsingborg" |
| 10:51.57 | benngard | core show hint 647 |
| 10:51.57 | benngard | <PROTECTED> |
| 10:52.03 | benngard | thats fine |
| 10:52.06 | kaii | benngard: do you also have an "exten => 647,1,SIP/0424449647&SIP/0106024647"? or only "exten => 647,hint,..." in "[agents]" |
| 10:52.27 | benngard | hmm, w8 |
| 10:52.43 | kaii | because param2 of addqueuemember() is the interface to dial .. and when you dont have that exten you specified in the param, it will most likely never be able to ring |
| 10:55.41 | benngard | thx, stupid me, did forgot that :( |
| 10:55.49 | kaii | also, you dont need the last paramter .. have you read http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember |
| 10:56.08 | benngard | did add exten => 647,1,Dial(SIP/0424449647&SIP/0106024647) to agents |
| 10:58.11 | benngard | [agents] |
| 10:58.11 | benngard | exten => 647,hint,SIP/0424449647&SIP/0106024647 |
| 10:58.11 | benngard | exten => 647,1,Dial(SIP/0424449647&SIP/0106024647) |
| 10:58.37 | benngard | exten => 1/0424449647,n,AddQueueMember(Helsinborg,Local/647@agents,1,,Lisbeth Mingert Nilsson) |
| 10:58.46 | benngard | now it works as i want, THX! |
| 11:07.57 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
| 11:13.25 | saxa | hi, i´m using * 1.8.2.3 with SIP and DAHDI, when I try to call from the SIP phones I do get a looooong delay before I can hear the line signal, also * it call the called number but then everything is muted, on the other side the party answers and doesnt hear anything. Any ideas where should I look for a sollution ? |
| 11:15.08 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 11:15.51 | saxa | btw, i´m using grandstream GXP285 and GXP2120 |
| 11:16.01 | saxa | sip phones |
| 11:17.08 | no1peanut | I am trying to set up an ivr where caller should enter a 4 digit code ending it with pound key - I know I can do this with read - but caller should also be able to press the asterisk key to go back in menu - and this should be without ending with the pound key - Anyone know how to do this ? |
| 11:19.26 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 11:20.52 | BlackBishop | exited non-zero on .... |
| 11:20.54 | BlackBishop | is that a bad thing ? |
| 11:21.18 | BlackBishop | I mean, everything works just as expected but .. that message looks like it shouldn't exit with non-zero |
| 11:22.16 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:25.52 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 11:29.58 | *** join/#asterisk xLP (~xLP@mail-out.lpcorp.fr) |
| 11:29.58 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:31.25 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 11:33.33 | *** join/#asterisk dank14 (~dank14@78.32.115.1) |
| 11:37.28 | *** join/#asterisk mzahariev (adminimini@piem-i-povrashtame-v.unixsol.org) |
| 11:39.51 | *** join/#asterisk Olivier_54 (~Olivier_5@maxicours.pck.nerim.net) |
| 11:39.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:40.44 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 11:42.47 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 11:42.54 | *** join/#asterisk Akiraa (~Akira@82.76.238.207) |
| 11:43.24 | saxa | another question is, how can I activate the verbose messages in the CLI in 1.8.x ? |
| 11:44.06 | saxa | i just get a prompt with 1.8 but with 1.4.x or 1.6.x i was getting everytime i got an event a description of it |
| 11:44.19 | saxa | which is no longer happening with 1.8 |
| 11:46.16 | saxa | btw, tunning asterisk -rvvvvvv |
| 11:46.31 | saxa | just changes the verbosity level, but nothing happens |
| 11:46.55 | saxa | i tried also to use core set verbose 100 |
| 11:46.57 | Jasnejac | saxa: what do you have set for console in logger.conf? |
| 11:47.19 | saxa | Jasnejac: hi, i never touched that file, let me check |
| 11:47.59 | saxa | Jasnejac: i do not have that file at all in my /etc/asterisk |
| 11:48.11 | saxa | do i need it ? |
| 11:48.33 | Jasnejac | it tells both the comsole and logging files what to report/store |
| 11:49.50 | saxa | yup I see |
| 11:49.52 | Jasnejac | could you give me the output of "logger show channels" from the CLI please? |
| 11:50.05 | saxa | let me try to restart * with it in /etc/asterisk |
| 11:50.29 | saxa | braserv*CLI> logger show channels |
| 11:50.29 | saxa | Channel Type Status Configuration |
| 11:50.29 | saxa | ------- ---- ------ ------------- |
| 11:50.29 | saxa | braserv*CLI> |
| 11:50.29 | saxa | braserv*CLI> logger show channels |
| 11:50.31 | saxa | Channel Type Status Configuration |
| 11:50.33 | saxa | ------- ---- ------ ------------- |
| 11:50.36 | saxa | braserv*CLI> |
| 11:50.38 | saxa | braserv*CLI> logger show channels |
| 11:50.41 | saxa | Channel Type Status Configuration |
| 11:50.43 | saxa | ------- ---- ------ ------------- |
| 11:50.46 | saxa | braserv*CLI> |
| 11:50.48 | saxa | braserv*CLI> logger show channels |
| 11:50.51 | saxa | Channel Type Status Configuration |
| 11:50.53 | saxa | ------- ---- ------ ------------- |
| 11:50.56 | saxa | braserv*CLI> |
| 11:50.59 | saxa | sorry |
| 11:51.01 | saxa | it pasted to fast |
| 11:51.07 | saxa | Jasnejac: as you can see nothing for now |
| 11:51.23 | saxa | i will restart * with this file in place |
| 11:51.28 | Jasnejac | good plan |
| 11:53.22 | saxa | http://pastebin.com/SQJ4bvvs |
| 11:53.31 | saxa | thats what I get now |
| 11:53.51 | saxa | but it still doesnt show any actividy |
| 11:53.56 | saxa | activity |
| 11:54.08 | Jasnejac | hmm. at least it looks right noe |
| 11:54.11 | Jasnejac | now* |
| 11:54.14 | saxa | ok |
| 11:54.55 | saxa | ok, now seems it works |
| 11:55.12 | saxa | its showing when i try to call |
| 11:55.18 | saxa | good, many thanks Jasnejac |
| 11:55.26 | *** join/#asterisk jcongdon (~jcongdon@rrcs-71-43-205-195.se.biz.rr.com) |
| 11:55.29 | Jasnejac | you can add debug, dtmf and fax reporting as well |
| 11:55.41 | Jasnejac | just modify the .conf file and do "logger reload" at the CLI |
| 11:56.03 | saxa | ok |
| 11:59.19 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 12:00.15 | saxa | ok, now the problem of everything being muted still persists. Need to find out why. |
| 12:00.18 | jcongdon | Hello everyone... looking for someone that can point me towards an ADSI guide of somesort. Google has been no help this time :( |
| 12:07.09 | xLP | is it normal that a channel variable disappears after a goto? |
| 12:08.10 | wdoekes2 | after a Local dial, yes, after a goto, no |
| 12:08.12 | BlackBishop | depends on the Goto ?! |
| 12:08.20 | BlackBishop | what wdoekes2 says ! |
| 12:08.37 | BlackBishop | learned about inheritance yesterday |
| 12:08.42 | xLP | -- Executing [a24@from-pstn:1] Set("SIP/bluesip-a24-out-00000025", "__FROM_DID=adria24") in new stack |
| 12:08.48 | xLP | => as you can see, it's inherited :) |
| 12:09.00 | wdoekes2 | that should survive a local dial too, indeed |
| 12:09.06 | xLP | and later |
| 12:09.08 | xLP | -- Executing [s@lp-do-not-answer-nor-ring:1] NoOp("SIP/bluesip-a24-out-00000025", "We got a call from "+493907716157" <+493907716157> to -. We will simply ignore it.") in new stack |
| 12:09.14 | BlackBishop | yup .. ${FROM_DID} should survive |
| 12:09.14 | xLP | same channel... |
| 12:09.35 | xLP | and here's the corresponding dialplan line |
| 12:09.36 | xLP | exten => s,1,NoOp(We got a call from ${CALLERID(all)} to ${__FROM_DID} -${DID}. We will simply ignore it.) |
| 12:09.42 | wdoekes2 | remove the __ |
| 12:09.44 | BlackBishop | yup |
| 12:09.49 | wdoekes2 | that was easy ;) |
| 12:09.55 | BlackBishop | __ is not a part of the variable |
| 12:10.03 | BlackBishop | or at least .. it shouldn't be ! |
| 12:10.07 | xLP | oh, in the NoOp ? |
| 12:10.19 | BlackBishop | yup |
| 12:10.25 | wdoekes2 | indeed.. you don't use it when dereferencing it, only when setting *to* it |
| 12:12.16 | xLP | simple, indeed :D |
| 12:12.18 | xLP | many thanks! |
| 12:12.23 | BlackBishop | you welcome |
| 12:12.26 | BlackBishop | I guess |
| 12:12.27 | BlackBishop | :D |
| 12:12.31 | BlackBishop | helped ! yeey |
| 12:12.34 | xLP | hihi |
| 12:12.51 | xLP | I feel ashamed now that I didn't think about trying that... |
| 12:12.56 | xLP | this is a 2 weeks old issue |
| 12:13.08 | xLP | (not really an issue, the value is only used in a NoOp) |
| 12:13.20 | xLP | (but I don't like not understanding something) |
| 12:26.10 | kaii | when asterisk is faced with a "SIP OPTIONS" message from a peer that is directed to its ip-address (e.g. "To: <sip:10.0.0.65>") asterisk responds with "404 Not Found". |
| 12:26.19 | kaii | i would like asterisk to respond OPTIONS with 200 OK .. but how? |
| 12:31.29 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 12:32.07 | shamelessn00b | I'm able to connect to my asterisk box that's behind NAT if I use a SIP Phone that uses TCP for SIP instead of UDP |
| 12:32.49 | fauxalliance | reads that again... slowly |
| 12:34.05 | shamelessn00b | Should I explain? |
| 12:34.38 | kaii | shamelessn00b: check your firewall then ? |
| 12:35.03 | shamelessn00b | traffic is going out from my box, where I have the soft phone |
| 12:35.31 | shamelessn00b | its not reaching my asterisk box though, I have port 5060 opened on both sides, both for TCP and UDP |
| 12:39.30 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:44.07 | kaldemar | shamelessn00b: did you dump the interface for incoming traffic with tcpdump already? |
| 12:45.55 | saxa | == Using SIP RTP CoS mark 5 <--- what exactly that means ? |
| 12:46.21 | saxa | mark 5 means the level ? |
| 12:47.08 | *** join/#asterisk shamelessn00b (~chatzilla@203.175.64.42) |
| 12:47.23 | *** join/#asterisk Carko23 (BloW@84-255-241-35.static.t-2.net) |
| 12:48.38 | jcongdon | I have an interesting caller id issue. I have no issues receiving internally on SIP or voicemail. However, our DAHDI extensions only receive the caller id information about 50% of the time. Any thoughts where to look. I think I have tried tweaking everything in the asterisk configs. |
| 12:48.50 | kaldemar | saxa: it's the 802.1p CoS value for RTP traffic. configured with cos_audio in sip.conf. |
| 12:48.55 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 12:49.44 | saxa | kaldemar: thx |
| 12:55.17 | Carko23 | can anyone please give me some advice on how to set up the SIP trunk for incoming calls if all I got from the provider was their hostname and phone numbers and they said no registration is needed? |
| 13:00.15 | *** join/#asterisk wonderworld (~ww@port-92-201-59-129.dynamic.qsc.de) |
| 13:01.24 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-69.telkomadsl.co.za) |
| 13:01.46 | jkroon | if there is anybody from digium present - the SSL cert on login.digium.com has expired. |
| 13:10.42 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 13:14.38 | *** join/#asterisk filo1234 (~filo@unaffiliated/filo1234) |
| 13:14.48 | filo1234 | hi to all |
| 13:18.00 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 13:18.23 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
| 13:18.31 | filo1234 | i trying to use qutecom like a client VOIP, when i configure i get a connection error, i think that probabily i need to activate something on astersik but i don't know what is for...i tracked my network for to follow packets and i found an error about X-Wengo-Ping request http://imagebin.org/138582 how can i solve it? |
| 13:18.45 | filo1234 | using ekig it run at all |
| 13:18.48 | filo1234 | ekiga* |
| 13:20.39 | jcongdon | filo1234: what do you mean like a client VOIP? Are you trying to use Qutecom internally with an Asterisk box? |
| 13:20.59 | jcongdon | We use it here without issue, as long as sip is configured in asterisk. |
| 13:21.13 | jcongdon | The one thing I have found is not to set caller id within the softphone. |
| 13:21.26 | filo1234 | jcongdon: yes like pc client on LAN |
| 13:21.49 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:23.09 | filo1234 | jcongdon: http://pastebin.com/qvJkx7CP this a log from wireshark when i try to connect to asterisk server |
| 13:23.39 | jcongdon | you guys are way smarter than me with wireshark and all. I have never used that |
| 13:24.30 | jcongdon | Did you set caller id in Qutecom? |
| 13:24.41 | jcongdon | if so, try removing it and restarting |
| 13:26.44 | filo1234 | jcongdon:i have just reset confguration several times and reconfigured it... |
| 13:26.56 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:27.26 | filo1234 | my id and password are internal telephone number like id 130 pass 130 and server asterisk is a local ip address |
| 13:27.38 | filo1234 | and itre run good with ekiga |
| 13:27.44 | filo1234 | it* |
| 13:28.09 | jcongdon | filo1234: just to make sure we are clear, I am not talking about asterisk now. In Qutecom, make sure you do not set caller id or caller name |
| 13:28.25 | filo1234 | i think that qutecom ask for a dummy registration request but asterisk cannot resolve it |
| 13:28.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:31.20 | filo1234 | jcongdon: and what i put in config?? |
| 13:31.57 | jcongdon | you said sip was already working with ekiga, correct? |
| 13:32.05 | filo1234 | yes |
| 13:32.12 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:32.20 | jcongdon | try using the same login information your use in that for testing |
| 13:32.42 | filo1234 | jcongdon: i making it all morning :D |
| 13:32.43 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 13:32.49 | jcongdon | I don't know why the callerid and caller name caused problems. That may not even be it, as that was a problem I had years ago. |
| 13:32.52 | filo1234 | jcongdon: i'm using same config |
| 13:33.32 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
| 13:33.47 | philfine | Is it possible to connect a cellphone to asterisk to make and receive calls ? |
| 13:35.14 | philfine | I have 2 Siemens S45 which I beleive are able to perform modem requests through cable connection |
| 13:36.19 | WIMPy | Not that way. Either chan_mobile, chan_datacard or via some external gateway. |
| 13:37.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:37.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:37.54 | kaldemar | philfine: chan_mobile allows that via bluetooth. |
| 13:38.08 | kaldemar | oops. i'm an echo. |
| 13:38.10 | philfine | Ok, I am checking that now :D |
| 13:38.27 | philfine | No, WIMPy didn't specify that :) |
| 13:38.45 | anonymouz666 | kaldemar: chan_mobile works? :P |
| 13:41.03 | ssureshot | in the chan_dahdi.conf is a # also a comment? |
| 13:41.09 | Carko23 | Can anyone see from this log why would I get the "The number is not in service" for an inbound SIP call when Allow anonymous calls is disabled and "any DID/CID" rule is created: http://www.shrani.si/f/2A/K5/vDUaA4h/tmpae39.jpg ? |
| 13:42.06 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:42.59 | philfine | ssureshot: What comment ? |
| 13:43.46 | mzb | Carko23: nothing like a nice blurry and (overly) short debug *image* to finish off the day ;) |
| 13:43.56 | Carko23 | :PPP |
| 13:44.03 | mzb | does absolute timeout of 15 secs make much sense? |
| 13:44.16 | kaldemar | anonymouz666: last time i tried it, it did. |
| 13:44.31 | kaldemar | ssureshot: no, only ;. |
| 13:44.48 | ssureshot | I'm trying to include dahdi-channels.conf |
| 13:45.05 | kaldemar | ssureshot: #include dahdi-channels.conf <--- |
| 13:45.10 | ssureshot | so is #include /etc/asterisk/dahdi |
| 13:45.15 | ssureshot | ah ok |
| 13:46.46 | *** join/#asterisk sourcode (~code@ppp-58-8-229-221.revip2.asianet.co.th) |
| 13:47.45 | ssureshot | I'm getting chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. .. any ideas on where to look with this? |
| 13:48.00 | ssureshot | dahdi configuration issue? |
| 13:48.05 | ssureshot | asterisk config issue? |
| 13:48.09 | WIMPy | Configute the correct signalling. |
| 13:49.03 | ssureshot | the signaling is exactly as it was using zapata from the old box though.. shouldn't it stay the same? |
| 13:49.38 | WIMPy | Yes, but it's obviousely wrong. Or you left a loopback plug in that port. |
| 13:49.41 | manji | ssureshot: ssureshot are you sure the other end is NT ? |
| 13:50.14 | manji | it's currently TE - TE I reckon |
| 13:50.28 | manji | unless I misunderstood |
| 13:50.36 | ssureshot | well I havn't verified it with the provider I'm gong from the config of the crashed box |
| 13:50.38 | WIMPy | That's what it says. |
| 13:51.28 | WIMPy | If you're connected to a telco line, make sure noone put a loopback in when you were down. |
| 13:51.56 | ssureshot | ah let me go check the dmark |
| 13:54.55 | ssureshot | no errors so let me give them a call thanks for the direction |
| 14:00.25 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 14:04.32 | *** join/#asterisk philfine_ (~philfine@56.4.166.178.rev.vodafone.pt) |
| 14:05.15 | filo1234 | jcongdon: solved... |
| 14:05.34 | jcongdon | filo1234: what was it? |
| 14:13.19 | ssureshot | WIMPy: is that normal for the to automate a loopback when my end goes down? |
| 14:13.24 | filo1234 | jcongdon: for some reason in qutecom, cannot authenticate at that server ip ( that is own server gateway ) but it want dns server |
| 14:13.28 | filo1234 | :| |
| 14:14.35 | WIMPy | ssureshot: I don't know your telco. But I've heard some doing such things. At least if the situation exists for londer or at the wrong time. |
| 14:15.02 | ssureshot | it's been down 2 weeks and it's att |
| 14:15.02 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:15.14 | ssureshot | backup box obiously |
| 14:15.28 | WIMPy | Ask them. |
| 14:15.57 | ssureshot | yeah I got a call in to them now.. |
| 14:30.39 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 14:35.03 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:38.56 | jcongdon | I am using a Zhone Z-plex channel bank. I know the signalling is fxsloopstart. I should be using fxo_ks in my chan_dahdi.conf, right? Or should I use fxo_ls? |
| 14:46.45 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:46.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:47.43 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 14:51.53 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 14:53.07 | *** join/#asterisk last1 (~dood@modemcable238.94-200-24.mc.videotron.ca) |
| 14:53.47 | last1 | what is the default MySQL table in which Asterisk holds its data when doing Set(DB(..)) for example ? |
| 14:54.09 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-awkptsqoigbfgvzy) |
| 14:54.35 | WIMPy | None. It's AstDB. |
| 14:55.15 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 14:55.56 | *** join/#asterisk tash (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 14:56.06 | jcongdon | finally solved half of my CallerID problems.... Dialing direct DID to desk phone, caller id works. |
| 14:56.24 | jcongdon | Dialing through the company main number and entering an extension, caller id doesn't work. |
| 14:56.30 | jcongdon | any idea? |
| 14:57.29 | tash | if this line appears in a .call file: StartRetry: 12916 1 (1298041000) ... and I touch the file to give it a timestamp of, say, right now .... the call doesn't go out. What happens, I assume, is that Asterisk reads the file and if it sees StartRetry in the future, it ignores the call. Is that right? |
| 14:57.46 | tash | oh, the StartRetry is 30 minutes in the future, for example. |
| 14:58.16 | WIMPy | Why do you change a file that was written back by Asterisk? |
| 14:58.36 | kaldemar | jcongdon: how do you enter the extension? what goes on in the dialplan? |
| 14:58.38 | tash | just testing |
| 14:59.41 | tash | I'm testing WaitExten. I'm not entering any digits on purpose, then the next rule in my dialplan is to move the .call to a directory and timestamp it for 30 minutes in the future, to retry the call. I opened the .call after all that was done and saw the StartRetry, which looking at it now doesn't seem to be 30 minutes in the future, I think it was 5 |
| 15:00.17 | tash | my goal is: if a user doesn't answer their phone ... or even if they do and don't press 1 to confirm they heard the message, I want to call them back in, say, 30 minutes or so |
| 15:01.32 | jcongdon | kaldemar: http://www.bowlingball.com/dialplan.php |
| 15:01.59 | jcongdon | kaldemar: that's a miniature version |
| 15:02.09 | jcongdon | I think it has the necessary information |
| 15:02.12 | last1 | is it possible with AstDB to update an entire family of keys ? |
| 15:02.20 | last1 | ie, not just a specific family/key |
| 15:02.30 | last1 | I want all keys within family to have value 1 for example |
| 15:07.03 | tash | can someone give me some advice ^^ |
| 15:09.09 | philfine | Is chan_mobile suppose to come with asterisk 1.8 source ? |
| 15:09.24 | wdoekes2 | tash: why don't you create a new callfile if the he-has-confirmed-the-message conditions aren't met? |
| 15:09.25 | philfine | Cannot find it in menuselect |
| 15:09.27 | jcongdon | tash: why not create a separate script that will generate the .call file at the appropriate time |
| 15:10.30 | tash | wdoekes2 and jcongdon: I didn't think of that. Why can't I just touch the file for 30 minutes in the future? |
| 15:10.54 | tash | If I understand why that doesn't work, then maybe I will do what you say....if that is a better solution (you guys know more than I do I'm sure) |
| 15:11.02 | jcongdon | tash: that may work, but I don't think asterisk cares about the timestamp |
| 15:11.04 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 15:11.12 | tash | doesn't it though? |
| 15:11.19 | jcongdon | I really don't know for sure |
| 15:11.35 | kaldemar | jcongdon: try removing the answer |
| 15:11.36 | tash | I mean, if I touch -t a .call file for 30 minutes in the future when I create the fall and drop it in the outgoing dir, it doesn't trigger the call until that time |
| 15:11.47 | jcongdon | I have always done outgoing calls on demand, meaning put the file in the directory as soon as I am ready for it to fire |
| 15:11.58 | jcongdon | I see |
| 15:12.04 | WIMPy | It usually does care about the timestamp. But the timestamp inside the file is more important as it seems. |
| 15:12.20 | tash | but, in my dial plan I'm doing touch -d "30 minutes" <filename> |
| 15:12.38 | tash | WIMPy: yeah, that's what I'm coming to conclude ... I need to open the file and look at that |
| 15:13.32 | tash | WIMPy: the original .call doesn't have a timestamp in the file. It has Channel, Maxretries, Retrytime, Context, Callerid, Extension, SetVar, and Waittime |
| 15:14.26 | philfine | Anyone ? WIMPy ? |
| 15:14.38 | wdoekes2 | tash: doesn't * simply not look at the original file after moving it? you seem to say that creating a new file would work.. |
| 15:14.49 | *** join/#asterisk erwinator (~papogi@124.105.166.166) |
| 15:15.29 | *** part/#asterisk last1 (~dood@modemcable238.94-200-24.mc.videotron.ca) |
| 15:15.30 | jcongdon | philfine: it's in my /etc/asterisk directory already |
| 15:15.40 | WIMPy | philfine: No, they are both not included. |
| 15:15.43 | jcongdon | I assume it came with 1.8 addons |
| 15:15.56 | philfine | The config file yes, but not the module |
| 15:16.03 | jcongdon | gotcha |
| 15:16.16 | philfine | WIMPy: Do I need both or just one ? |
| 15:16.32 | tash | wdoekes2: when the original call is placed, and I answer the phone, and don't press a digit to confirm I heard the message ... then the dial plan executes System to copy the file and touch it, and move it back to *'s outgoing dir ... then at that time, it seems * doesn't ever read it, b/c I don't see anything in the asterisk console when that time comes |
| 15:16.32 | philfine | Just one |
| 15:16.36 | WIMPy | Depends on your hardware. |
| 15:16.39 | philfine | Other is for cards |
| 15:16.41 | philfine | Thanks |
| 15:19.20 | jkroon | interesting. |
| 15:19.41 | jkroon | asterisk definitely doesn't handle srv records well when it comes to IAX/2 support. |
| 15:20.26 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 15:21.34 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 15:22.32 | philfine | WIMPy: where do I get chan_mobile ? |
| 15:23.02 | WIMPy | google? |
| 15:23.32 | philfine | Thats something I already done. Curiously chan_movile is in the source |
| 15:24.37 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:24.42 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:25.50 | WIMPy | Oh, never noticed that. |
| 15:27.42 | wdoekes2 | I don't know tash: something must be different from a "real new" file versus your copy, the sane answer being the content |
| 15:30.05 | WIMPy | was under the impression that it was a little more experimental. |
| 15:30.43 | tash | wdoekes2: The only thing different is a line like: StartRetry: 23730 1 (1298042982) |
| 15:31.08 | tash | If I knew more about that line, maybe I could figure it out. I know the number in parentheses is epoch time, not sure what 23730 1 is |
| 15:31.17 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 15:31.23 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:33.23 | sbrath | Anyone ever seen this senario: call 1 comes in Digium pots 1/1,and is answered. While that call is active a second call comes in, and is routed to another phone. When the dahdi-1/2 channel hangs-up. 4 seconds later the dahdi-1/1 channel hangs up... This is very odd.. |
| 15:35.27 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 15:43.09 | *** join/#asterisk odin917 (~Gavin_Sil@64.61.105.226) |
| 15:43.13 | odin917 | hey all |
| 15:43.28 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 15:44.11 | wdoekes2 | tash: why don't you just remove the line then? sed -e '/^StartRetry:/d' |
| 15:44.11 | *** join/#asterisk matach (~matach@248-7.79-83.cust.bluewin.ch) |
| 15:44.11 | odin917 | does nayone know how I can set my outbound callerid as one specific number (not a number i have my trunk registered as" but still identify myself as the trunk number to my voip provider? |
| 15:45.56 | odin917 | i was just on the phone with my voip provider and he said the invite came accross as my set callerid = 2125551234 and contact = 2125551234, but my trunk is registered as 2125552222 |
| 15:46.14 | odin917 | he said he is throwing out the call because that "contact" field needs to reference my trunk # |
| 15:48.50 | tash | wdoekes2: I can definitely try that. Can you explain the sed example? I've not much experience with sed |
| 15:51.14 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 15:51.40 | *** join/#asterisk matach (~matach@83.79.7.248) |
| 15:54.30 | *** join/#asterisk matach_ (~matach@248-7.79-83.cust.bluewin.ch) |
| 15:58.41 | *** join/#asterisk polycomSlave (~chatzilla@8.224.32.34) |
| 16:07.16 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:16.15 | ChannelZ | odin917: Set(CALLERID(num)=2125552222) |
| 16:18.38 | tash | interesting, when I answer a call that Asterisk places to my cell, and do an ls -la on /var/spool/asterisk/outgoing, the timestamp on the file automatically changes to 10 minutes in the future. Why is that? |
| 16:18.41 | jcongdon | in chan_dahdi.conf should I change to dahdichan= instead of cahnnel => is there a difference? |
| 16:21.56 | ChannelZ | only if you are defining the channels in a different context thing than [channels] |
| 16:22.46 | ChannelZ | a "section" I guess, not a context.. whatever they call them |
| 16:23.33 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 16:25.43 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 16:27.28 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 16:28.04 | odin917 | ChannelZ: I have set the callerID in the extension |
| 16:28.37 | odin917 | my problem is that when the call connects it shows the invite to my provider as that number instead of the trunk registration number so the provider throws out the call |
| 16:29.08 | odin917 | he says he sees callerid and "contact" seperate, but they are both my outbound CID |
| 16:36.46 | *** join/#asterisk Leddy (leddy@aqua.surgehost.net) |
| 16:39.25 | tash | wdoekes2: using sed on that file would probably work, except the modified time on the file changes as soon as you change the file. So, if you don't want the call to go out unitl 20 minutes from now or so, and you sed the file, it will go out as soon as you save the file |
| 16:45.25 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 16:49.38 | ssureshot | Ive got a yellow alarm then a second later alarms are cleared.. it keeps fluctuating like this .. |
| 16:50.05 | ssureshot | just looking for a baseline to start with.. I'm new to this config and am being thrown under the buss |
| 16:50.34 | *** join/#asterisk areay (~boss@78-86-178-210.zone2.bethere.co.uk) |
| 16:53.04 | *** join/#asterisk appleboy (~AppleBoy@about/cooking/nakedchef/apple/tarts) |
| 16:53.21 | appleboy | hi file, you there? |
| 16:53.55 | file | I exist in a form that is always present in one form or another. |
| 16:54.04 | jaytee | :-) |
| 16:54.31 | file | Some restrictions apply on the form. Please see site for details. |
| 16:54.41 | jaytee | he is the Kwisatz Haderach, the one who can be many places at once |
| 16:55.33 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:58.08 | n3hxs | has a wood file, a criminal file and a bastard file. ;) |
| 16:59.47 | fauxalliance | files himself under bastard |
| 17:01.35 | n3hxs | wood file along with the rest of the bastard criminals, but escapes. |
| 17:06.27 | WIMPy | ssureshot: So how did you get there? |
| 17:06.36 | ssureshot | if I use the dahdi-tool and choose loop what does that do |
| 17:06.42 | areay | hi all.. i'm configuring asterisk as a sip client (peer) behind NAT. i have forwarded the SIP and RTP ports on my router, and specified the externip, localnet, and nat options in sip.conf. yet my registration is still expiring. here's my sip debug: http://pastebin.com/TEx0FhNP ...and my sip.conf: http://pastebin.com/w21iecFV |
| 17:07.06 | ssureshot | WIMPy, actaully it's a new device zorcom usb.. bank my boss bought a few months back |
| 17:07.51 | WIMPy | Did you get rid of a loop? |
| 17:08.13 | ssureshot | WIMPy: it was giving yellow then clearing but once I ran the dahdi-tool and chose loop it's now giving me the same loop error so I believe it's in the dahdi config |
| 17:08.35 | ssureshot | Wimpy: att tested and all looks good with the line |
| 17:10.11 | WIMPy | What are you doing with dahdi-tool? |
| 17:10.43 | ssureshot | I thought I was testing.. I open up dahdi-tool choose the card and loop |
| 17:11.13 | WIMPy | Why do you do that? |
| 17:11.16 | ssureshot | so I beleive I put in in a loop state |
| 17:11.32 | WIMPy | So would I. |
| 17:11.46 | ssureshot | so the question is how to I get it out of that state |
| 17:11.51 | WIMPy | But it sounds unlikely you want it to. |
| 17:12.00 | *** join/#asterisk [Outcast]_ (~anonymous@64.202.62.5) |
| 17:12.27 | WIMPy | Hmm. I never tried a software loop. |
| 17:12.40 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:12.49 | appleboy | file: the ssl cert on neutrino expired I think |
| 17:12.57 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:13.05 | file | it didn't. |
| 17:13.09 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 17:13.15 | file | it's a different server. |
| 17:13.22 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 17:13.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:13.31 | file | cert is good until 2012-02-11 |
| 17:13.39 | appleboy | ah |
| 17:13.57 | ssureshot | so I guess no idea on how to remove that loop then aye |
| 17:14.07 | ssureshot | bummer |
| 17:15.34 | appleboy | err, mail.joshua-colp.com's cert |
| 17:16.31 | file | Valid from 2/10/2011 to 2/10/2012 |
| 17:17.10 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 17:18.30 | areay | nobody knows? maybe i should just assign the asterisk box an external ip.. was kinda hoping to accomplish this behind NAT |
| 17:18.38 | appleboy | did anything else change? pop on 995 over ssl gives connection refused, the kerio gives untrusted cert |
| 17:18.40 | areay | i'm sure i've had it working this way before... |
| 17:19.08 | WIMPy | ssureshot: If in doubt, just restart it. |
| 17:20.52 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 17:21.50 | ssureshot | going to grab a quick lunch,, fyi.. I purged dahdi reinstalled ane rebooted and now I'm back to the alarms so thats good... Ill keep you posted |
| 17:22.30 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 17:24.02 | areay | no worries, i'll have to figure it out myself... thanks to all the helpful people who responded :-P. |
| 17:30.18 | *** join/#asterisk tash (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 17:30.22 | tash | Looking for some advice: I am debating whether to have Asterisk attempt to detect voicemail/answering machine, or if I don't receive DTMF during a phone call from a user (such as pressing 1 to confirm they heard the call), place the call again at a later time. The latter could be costly. So, does answering machine/voicemail detection work well? I've never used it. |
| 17:32.15 | *** join/#asterisk matach (~matach@248-7.79-83.cust.bluewin.ch) |
| 17:32.36 | *** join/#asterisk matach (~matach@248-7.79-83.cust.bluewin.ch) |
| 17:35.22 | appleboy | file: anything else besides the cert change? |
| 17:35.31 | file | it's a completely new server. |
| 17:35.59 | appleboy | oh |
| 17:36.10 | appleboy | did the ports change though? |
| 17:36.17 | file | no |
| 17:39.42 | Kobaz | hahahaha |
| 17:39.44 | Kobaz | wtf |
| 17:40.02 | Kobaz | you know the polycom kirk wireless phones |
| 17:40.09 | Kobaz | well, i just got this email |
| 17:40.15 | Kobaz | We mistakenly stuck the sticky hanger to the forklift instead of the back of the phone. How can I get another one? |
| 17:44.23 | appleboy | I figured it out :) file, you're the best! |
| 17:44.30 | appleboy | lol Kobaz |
| 17:44.31 | file | bananas. |
| 17:54.28 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 17:58.57 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 18:03.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 18:07.01 | *** join/#asterisk shamelessn00b (~chatzilla@175.110.242.37) |
| 18:07.38 | shamelessn00b | Hi, using asterisk 1.8.2.3, finally managed to register a SIP client on my asterisk box on EC2, however when I make a call I get this: Insufficient information for SDP (m= not found) |
| 18:08.02 | Qwell | sounds like you've got a terrible SIP client |
| 18:08.40 | shamelessn00b | xlite |
| 18:09.08 | shamelessn00b | been using it for ages |
| 18:10.22 | shamelessn00b | using it on TCP SIP, UDP SIP not working for me |
| 18:14.32 | shamelessn00b | any clue |
| 18:14.40 | shamelessn00b | you try registering?? |
| 18:15.25 | Tozz_ | xlite works fine on asterisk |
| 18:15.32 | Tozz_ | something on your setup is wrong |
| 18:16.12 | shamelessn00b | what could that be? |
| 18:17.26 | tash | what causes StartRetry to be appended to a .call file? |
| 18:19.56 | Tozz_ | shamelessn00b: some wicked SIP proxy or something, I dont know ;) |
| 18:20.32 | shamelessn00b | it sure is wicked |
| 18:20.39 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 18:20.52 | shamelessn00b | I have the sip account registered on my asterisk box, over TCP though |
| 18:21.00 | Tozz_ | maybe Amazon is fucking something up |
| 18:22.09 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:22.46 | shamelessn00b | I doubt it, I'm running a tcpdump on that machine, and I can see the packets coming on that machine |
| 18:32.23 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 18:32.41 | shamelessn00b | I did an originate from asterisk cli to my SIP client |
| 18:32.51 | tash | grr, I really don't understand this... |
| 18:33.10 | shamelessn00b | originate SIP/3001 application playback demo-congrats |
| 18:33.25 | shamelessn00b | after I answer the call, I just hear 2 beeps |
| 18:35.01 | tash | placing outbound calls, if the user doesn't press 1 to confirm they heard the message, I do the following: http://pastie.org/1579708 ... then I have a cron that moves .call files in that directory to /var/spool/asterisk/outgoing <---- does anyone see a problem with this? |
| 18:35.46 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 18:36.00 | tash | what seems to happen, is that as soon as the .call file(s) are moved from that tmp dir to /var/spool/asterisk/outgoing is that a line: StartRetry: 11353 1 (1298052420), for example, gets appended to the .call file and a call isn't placed. |
| 18:36.12 | Tozz_ | shamelessn00b: but the packet might be corrupt |
| 18:36.13 | tash | then, every 5 minutes, a DelayedRetry line is added to that .call file |
| 18:36.39 | tash | can someone please help me with this? |
| 18:36.49 | shamelessn00b | Tozz_: inbound to the client is working, sort of |
| 18:36.54 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 18:40.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:51.47 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 19:01.46 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:03.09 | *** join/#asterisk Arsenick- (~y@fedora/Arsenick) |
| 19:06.03 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:06.35 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:06.51 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:13.32 | *** join/#asterisk denon (~denon@unaffiliated/denon) |
| 19:15.01 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:16.18 | shamelessn00b | Ok call is progressing now, but I don't hear any audio |
| 19:18.43 | Tozz_ | NAT ? |
| 19:19.55 | *** part/#asterisk denon (~denon@unaffiliated/denon) |
| 19:20.58 | shamelessn00b | yes, but I already configured sip.conf |
| 19:21.03 | shamelessn00b | modified* |
| 19:24.46 | tessier | fauxalliance: Wow, blast from the past. I used to run portsentry many years ago. |
| 19:25.06 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:25.27 | *** join/#asterisk mserve (~mserve@asams.mserve.com) |
| 19:27.40 | mserve | How might I go about getting the context of an extension from inside the dialplan? I have the caller enter their extension and then I would like to check their context before going further on in the dialplan. |
| 19:27.59 | Tozz_ | (i think) sippeer |
| 19:28.17 | Tozz_ | jep |
| 19:28.22 | Tozz_ | http://www.voip-info.org/wiki/view/Asterisk+func+sippeer |
| 19:28.42 | mserve | Thanks! |
| 19:28.53 | Tozz_ | yw |
| 19:29.31 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 19:34.46 | shamelessn00b | Tozz_: what to do |
| 19:35.07 | shamelessn00b | my SIP clients have private IPs as well |
| 19:39.09 | mserve | When a caller dials a number and then is sent in to a macro, how can I store the dialed number in a variable? |
| 19:39.26 | mserve | for use in the macro |
| 19:42.35 | mserve | Nevermind...got it |
| 19:46.16 | *** join/#asterisk e_dot_zilla99 (~e_dot_zil@8.224.32.34) |
| 19:46.19 | e_dot_zilla99 | hi |
| 19:48.08 | *** join/#asterisk andyoutside (cec00d82@gateway/web/freenode/ip.206.192.13.130) |
| 19:49.37 | andyoutside | ok why when using queues and having it show hold time does it some times changes the caller id to the telephone number and other times does not? I prefer it not change it. |
| 20:01.56 | e_dot_zilla99 | ~ |
| 20:08.48 | *** mode/#asterisk [+q e_dot_zilla99!*@*] by Qwell |
| 20:08.55 | Qwell | Ban evasion is not smart. |
| 20:09.54 | thehar | heh |
| 20:10.01 | thehar | Qwell 1 Others 0 |
| 20:10.35 | *** mode/#asterisk [+b #asterisk!*@*] by Qwell |
| 20:10.44 | *** mode/#asterisk [-bb #asterisk!*@* *!*@i-evolve.net] by Qwell |
| 20:10.54 | *** mode/#asterisk [-b *!*@i-evolve.net] by Qwell |
| 20:11.00 | Qwell | stupid xchat |
| 20:11.11 | *** mode/#asterisk [+b *!*@8.224.32.0/24] by Qwell |
| 20:11.17 | *** mode/#asterisk [+b *!*@*i-evolve.net] by Qwell |
| 20:11.32 | *** mode/#asterisk [-b t_dot_zilla!*@*] by Qwell |
| 20:12.00 | *** mode/#asterisk [-b *!*@8.224.32.0/24] by Qwell |
| 20:12.03 | *** mode/#asterisk [+b *!*@8.224.32.0/23] by Qwell |
| 20:12.14 | Qwell | there |
| 20:16.20 | ssureshot | what tools do I have available with zaptel to find out if it sees the channels ? |
| 20:16.42 | jaytee | zap show channels? |
| 20:17.08 | Qwell | <e_dot_zilla99> power tripping nerd |
| 20:17.09 | Qwell | >.> |
| 20:17.17 | leifmadsen | Qwell: you have no mercy :) |
| 20:17.24 | leifmadsen | wonders how long till he gets a msg |
| 20:17.27 | Qwell | he's been banned like 4 different times |
| 20:17.55 | jaytee | did anyone file a bug report for the bug that's up e_dot_zilla99's rear? |
| 20:18.36 | leifmadsen | it isn't a bug :) |
| 20:19.26 | jaytee | I wish xchat had a button you could click on after selecting a user in the chat list that would stream a wav file to that person's system that plays R. Lee Ermey from Full Metal Jacket screaming "What is your major malfunction, numbnuts?" |
| 20:19.58 | Qwell | aol did, heh |
| 20:20.04 | Qwell | wasn't a button though |
| 20:20.21 | Qwell | you could play sounds to a room, and it was annoying as crap. |
| 20:20.48 | Qwell | people constantly used it to play a:\blah, so the drive would grind with no disk |
| 20:20.54 | jaytee | I vaguely remember that despite having trying to banish any memories of using AoHell and Prodigy back in the early days |
| 20:28.18 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:34.56 | *** join/#asterisk cnu (cnu@2001:470:1f0b:ea::10) |
| 20:35.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:43.35 | andyoutside | ok why when using queues and having it show hold time does it some times changes the caller id to the telephone number and other times does not? I prefer it not change it. |
| 20:46.55 | andyoutside | any ideas? |
| 20:48.44 | *** kick/#asterisk [e_dot_zilla99!~north@pdpc/sponsor/digium/Qwell] by Qwell (now it is permanent.) |
| 20:57.24 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 21:03.34 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 21:04.06 | *** join/#asterisk carrar (~tim@2604:5000:10::4) |
| 21:07.51 | *** join/#asterisk Plazma (~Plazma@freenode/staff/plazma) |
| 21:08.25 | odin917 | how do i set the "contact" field as soemthing different from my outbound callerid when my SIP invite goes through to my provider? |
| 21:09.47 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 21:10.55 | odin917 | a tech at my sip trunk provider tells me that their system is not accepting my outbound call attempt when I forge my outbound caller ID because womsewhere in the invite i also have to identify myself as the number my trunk is registered at. |
| 21:11.17 | odin917 | s/womswhere/somewhere |
| 21:12.24 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 21:14.35 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:17.07 | killown | is there a way to turn of SIP METHOD OPTIONS in asterisk? |
| 21:17.10 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 21:19.01 | _Corey_ | odin917: Who's your provider? |
| 21:21.05 | killown | _Corey_, my provider does not support method option |
| 21:23.28 | Qwell | Your provider doesn't support much of anything. You really need to just switch. Seriously. Stop wasting peoples time. |
| 21:23.48 | dzup2 | hello, i manage to use my intercom work thruout my PBX speakers using a text based SIP phone and i find out i have to use that program with screen in order to keep it running, so i made a /etc/rc.d/intercomd script ( http://pastebin.com/hR4jLBtd ) , but am not sure, starts up before asterisk does, is there a way i can modify runlevels to start my script after asterisk starts and to use screen /etc/rc.d/intercomd start ... ? in which order i have to ru |
| 21:24.50 | killown | Qwell, please, do me a favor /ignore me |
| 21:25.58 | _Corey_ | killown: I really don't know what you're referring to, I was asking odin917. (I've dealt with is problem with a few carriers.) |
| 21:26.05 | *** join/#asterisk ZX81_ (~ZX81_@121-74-236-177.telstraclear.net) |
| 21:26.52 | killown | _Corey_ ok man i will give up of asterisk, there is no good brazilian provider, you win |
| 21:27.35 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:27.38 | Tozz_ | perhaps you can try freepbx! |
| 21:27.39 | ZX81_ | hey all, having a strange problem whereby Asterisk is cancelling an outbound call after a 302 redirect - i.e. Linksys SPA is on call forward, tells Asterisk to redirect to an external number, makes the call, cancels after 10 seconds. Dial has timeout of 60 seconds - Asterisk 1.6.2 and Asterisk 1.8 - latest SVN |
| 21:27.54 | _Corey_ | Um, ok. My mother always told me I was a winner. I guess that clinches it. :) |
| 21:28.23 | odin917 | _Corey_: broadsoft |
| 21:28.28 | ZX81_ | ran with full debug and nothing obvious is happening |
| 21:28.44 | ZX81_ | I mean it uses the local channel and I'd prefer to be able to add a /n to it but can't |
| 21:28.45 | _Corey_ | odin917: ahh... well you might be in luck then |
| 21:28.50 | ZX81_ | other than that no idea |
| 21:29.48 | _Corey_ | odin917: One carrier running Broadsoft requires "P-Charge Identity" be added to the SIP INVITE with the BTN |
| 21:30.13 | _Corey_ | They've said: "When P-Charge Identity is not present, we take the FROM field in the SIP Header and use it as the billing telephone number." |
| 21:30.27 | odin917 | awesome |
| 21:30.34 | _Corey_ | From != Billable Number? Refuse call. |
| 21:30.49 | odin917 | yea because its one trunk, they dont know who to bill |
| 21:30.55 | _Corey_ | May not be the same on all flavors of Broadsoft, so your mileage may vary. |
| 21:31.04 | odin917 | how do i set p-charge indentity? |
| 21:31.42 | _Corey_ | SIPAddHeader |
| 21:31.43 | odin917 | please excuse my ignorance of the verbiage, this is my first asterisk setup |
| 21:32.37 | _Corey_ | ouch... probably: SIPAddHeader(P-Charge Identity: BTN_HERE) |
| 21:32.53 | _Corey_ | You may need to contact their support if it doesn't work. |
| 21:32.56 | *** part/#asterisk killown (~killown@unaffiliated/killown) |
| 21:33.14 | _Corey_ | I think that's all we ended up doing |
| 21:34.26 | odin917 | can i jam that in the context of the trunk somewhow? |
| 21:34.50 | _Corey_ | Stuff it in before you Dial() |
| 21:35.09 | odin917 | *duck* im only using the web interface so far |
| 21:35.28 | _Corey_ | oh, um, FreePBX? |
| 21:35.32 | odin917 | yea |
| 21:35.34 | _Corey_ | ouch |
| 21:36.35 | _Corey_ | well, I don't believe it's possible within the GUI. You're going to have to add a route to [from-internal-custom] or something. You may want to try the #FreePBX room for specific guidance there |
| 21:37.58 | odin917 | thanks corey |
| 21:39.17 | _Corey_ | no problem, good luck with it |
| 21:43.41 | odin917 | _Corey_: files found in /etc/asterisk/ with the _custom in then will not be touched by updates to freepbx, where should i add this sipaddheader line to? which file? |
| 21:52.33 | odin917 | exten custom got it |
| 21:52.34 | odin917 | thanks |
| 22:33.26 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 22:36.02 | ChannelZ | Hmm. Do I need to do anything special in my DAHDI config for it to transmit Caller ID to an FXS port? |
| 22:40.27 | *** join/#asterisk neurosys (~neurosys@69.198.143.74) |
| 22:40.37 | JerJer | let the channel ring twice, then answer |
| 22:40.43 | neurosys | How can I call all extens except the one that initiated the call? |
| 22:41.04 | *** join/#asterisk navaismo (~navaismo@evdomip-203-96-202.iusacell.net) |
| 22:41.09 | paulc | neurosys: with some clever AGI scripting? |
| 22:41.19 | ChannelZ | yeah, there's no built-in way to do that |
| 22:41.39 | neurosys | ugh ok ok |
| 22:41.44 | Katty | hi ChannelZ |
| 22:41.58 | ChannelZ | hey Katty |
| 22:42.01 | ChannelZ | word up |
| 22:42.37 | Katty | hugs ChannelZ |
| 22:42.44 | Katty | just waitin for 5 |
| 22:42.48 | Katty | got a busy night |
| 22:43.08 | ChannelZ | Oh? |
| 22:43.12 | ChannelZ | anything fun? |
| 22:44.13 | thehar | hugs Katty |
| 22:44.26 | Katty | hugs thehar |
| 22:44.28 | thehar | yay |
| 22:44.35 | Katty | ChannelZ: rollerderby practice, then goin out downtown |
| 22:44.52 | ChannelZ | Ah cool |
| 22:45.02 | ChannelZ | roller derby is awesome |
| 22:45.13 | Katty | i'm not a skater |
| 22:45.18 | Katty | jsut an official |
| 22:45.27 | Qwell | she flags people she doesn't like |
| 22:45.34 | ChannelZ | hehe |
| 22:45.35 | Qwell | that's totally what I would do |
| 22:45.40 | Katty | hehe no i'm not a ref |
| 22:45.46 | ChannelZ | Seen Whip It? |
| 22:45.48 | Qwell | how do you officiate? |
| 22:45.49 | Katty | of course |
| 22:45.54 | Katty | Qwell: i keep track of penalties |
| 22:45.55 | ChannelZ | I liked that movie |
| 22:46.03 | Qwell | I see |
| 22:46.03 | Katty | and hand out bandaids |
| 22:46.07 | Katty | hello kitty bandaids |
| 22:46.13 | ChannelZ | sweet |
| 22:49.48 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 22:49.52 | carrar | Are they PINK? |
| 22:52.31 | ChannelZ | uhh, HELLO KITTY |
| 22:55.02 | *** join/#asterisk neurosys (~neurosys@69.198.143.74) |
| 22:55.51 | Qwell | carrar: fuchsia |
| 22:56.14 | carrar | I don't think I am comfortable with fuchsia |
| 22:56.14 | Qwell | Katty: I have a better suggestion for this weekend! |
| 22:57.08 | navaismo | ¬¬ |
| 23:01.36 | Katty | Qwell: i have to go. i'll be on aim |
| 23:01.42 | Katty | see ya guys next week |
| 23:02.30 | ChannelZ | happy friday! |
| 23:06.27 | carrar | WHere on AIM |
