| 00:00.27 | navaismo | or use fail2ban or similar tool |
| 00:00.31 | navaismo | datarecal, ^^ |
| 00:02.32 | jpsharp | apurvtwr: Are you trying to use gtalk with the configs I gave you? |
| 00:03.05 | knarfly | Okay what was the trick to get Music on Hold working. I have dahdi setup and it worked before, but tonight I can see that the music starts but I can't hear anything. |
| 00:04.03 | apurvtwr | i tried that .. but it wasn't working. so started looking up the web for the res_jabber and chan_gtalk that you talked about |
| 00:04.28 | *** join/#asterisk qbx (~qbx@unaffiliated/qubix) |
| 00:05.07 | apurvtwr | now .. with navaismo input i finally just now managed to call gtalk to phone. |
| 00:06.16 | apurvtwr | that is.. if I originate call from gtalk then it is received on my SIP phone. but I also want to call from my phone and receive that on SIP |
| 00:07.03 | apurvtwr | i have exten => s,n,Dial(SIP/1000,20,D(:1)) in my google-inbound context |
| 00:08.05 | apurvtwr | the guy giving in the video said that google needs a DTMF input '1' to start the call.. so this strange looking command really just waits for 20 secs for my SIP to pick up |
| 00:08.20 | apurvtwr | and if it does it sends 1 itself |
| 00:08.36 | apurvtwr | i am calling it strange because I just installed asterisk 3 days back |
| 00:08.45 | *** part/#asterisk qbx (~qbx@unaffiliated/qubix) |
| 00:08.47 | apurvtwr | might be very common command ;-) |
| 00:09.23 | navaismo | apurvtwr, http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 00:09.48 | navaismo | seriously IM getting crazy trying to compile the dahdi module in the raspi, so I have downloaded the rpi-patches sources, then edited some files in the kernel directory 3.5.2-3.fc17.armv5tel |
| 00:10.08 | navaismo | <PROTECTED> |
| 00:10.10 | navaismo | but the raspi insist in use the 3.1.9 kernel on reboot |
| 00:10.10 | navaismo | so the dahdi module still unavailable |
| 00:10.25 | navaismo | any basic compilations hints? |
| 00:11.19 | apurvtwr | navaismo: I do have a SIP registered as 1000.. are you hinting that there might be a problem with SIP/1000? |
| 00:11.35 | apurvtwr | or just giving me a pointer to Dial command? |
| 00:12.08 | navaismo | just giving a pointer to the dial command usage and options |
| 00:12.24 | apurvtwr | hm. i see. thanks :-) |
| 00:13.14 | apurvtwr | anyway. I will give this inbound thing one more shot. navaismo thanks a ton. I had been struggling for hours. |
| 00:14.52 | WIMPy | Does SIP allow for a 100 Trying after a 183 Session Progress? |
| 00:16.00 | navaismo | <PROTECTED> |
| 00:17.22 | carrar | NO |
| 00:17.40 | carrar | WIMPy, only if you are sending a NEW unrelated call :) |
| 00:17.45 | WIMPy | Thought so. |
| 00:18.23 | carrar | the onyl response will be a ACK after a 200 OK |
| 00:19.27 | carrar | also |
| 00:19.37 | carrar | could send a RE-INVITE |
| 00:19.47 | WIMPy | It ended with a busy after an inband busy. |
| 00:19.50 | carrar | which would make a new 100 trying in return |
| 00:25.30 | WIMPy | Also it seems trange to send an 100 Trying before a 407 Porxy Authentication Required. |
| 00:26.51 | carrar | yeah that would be strange |
| 00:27.11 | carrar | How can you send a call without first being authenticated (if that is required to do so to send a call) |
| 00:27.48 | carrar | though technically, registering is really for people to find you |
| 00:27.57 | carrar | not so much for you to send a call |
| 00:28.02 | carrar | however that is used that way also |
| 00:28.04 | WIMPy | NFI. Sofia-sip/1.12.11 seems to think it's a good ide. |
| 00:28.23 | WIMPy | That was on INVITE, not on register. |
| 00:28.49 | carrar | 407 is in response to a REGISTER or INVITE? |
| 00:29.35 | WIMPy | The full call was >INVITE <100 Trying <407 AuthReq >ACK >INVITE <100 Trying <183 Progress <100 Trying (pause with inband busy) <486 Busy |
| 00:30.22 | carrar | nice |
| 00:36.58 | *** join/#asterisk Solarbaby (~Solarbaby@adsl-108-83-110-107.dsl.lsan03.sbcglobal.net) |
| 00:43.34 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 00:44.52 | *** join/#asterisk naikrovek (~Mordecai@unaffiliated/naikrovek) |
| 00:45.59 | carrar | Wimpy, it is ok to send a 100 trying |
| 00:46.13 | *** part/#asterisk Solarbaby (~Solarbaby@adsl-108-83-110-107.dsl.lsan03.sbcglobal.net) |
| 00:46.18 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 00:46.38 | carrar | that just stops the additional INVITES |
| 00:46.40 | WIMPy | The first or the third? |
| 00:46.54 | carrar | then the UAS can send a 407 |
| 00:47.03 | carrar | but if can also just send a 407 also |
| 00:47.19 | carrar | typically more so then often you just send a 407 |
| 00:47.26 | WIMPy | Ok, so that for authentication schemes that might take some time? |
| 00:47.54 | carrar | if you have to go look it up |
| 00:47.57 | carrar | and that takes time |
| 00:48.01 | carrar | jsut fire back a 100 |
| 00:48.09 | WIMPy | Yes, makes sense. |
| 00:48.09 | carrar | to stop the INVITES from continually coming in |
| 00:48.28 | WIMPy | But that's certainly not going to happen on a soho pbx. |
| 00:48.30 | carrar | also lets them know you "hear them" :) |
| 00:49.00 | carrar | heh |
| 00:49.44 | carrar | I think the rule of thumb is 200ms |
| 00:49.55 | knarfly | what does facility not subscribed error message mean? |
| 00:50.01 | carrar | if you're going to take more then 200ms to figure it out |
| 00:50.04 | carrar | send a 100 |
| 00:50.56 | *** join/#asterisk CrazyTux[m] (~Brandon@107-199-131-100.lightspeed.irvnca.sbcglobal.net) |
| 00:51.05 | carrar | knarfly, ISDN error? |
| 00:51.40 | WIMPy | IAX register |
| 00:51.59 | WIMPy | And that doesn;t mean anything to me. |
| 00:52.02 | carrar | no idea about IAX |
| 00:52.12 | knarfly | carrar: but I don't have ISDN...my * server is on my LAN |
| 00:52.21 | carrar | but for ISDN it means the circuit probably isn't completely setup |
| 00:52.22 | carrar | possibly |
| 00:52.49 | knarfly | X-Lite works fine but Zoiper is having troubles |
| 00:52.58 | carrar | simple |
| 00:53.05 | carrar | don't use zoiper ;) |
| 00:53.12 | carrar | heh |
| 00:53.13 | WIMPy | No. |
| 00:53.19 | knarfly | why didn't I think of that 8-) |
| 00:53.24 | carrar | hehe |
| 00:53.38 | WIMPy | It could be a supplementary service only available as a paid option. |
| 00:53.40 | carrar | Whats the SIP error with Zoiper? |
| 00:53.46 | carrar | not registering? |
| 00:53.49 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 00:53.54 | knarfly | I like Zoiper....X-Lite now comes with builtin commercials |
| 00:54.06 | carrar | coutner path is so easy to use |
| 00:54.09 | carrar | counter path |
| 00:54.11 | WIMPy | likes zoiper as well. |
| 00:54.46 | carrar | try blink |
| 00:54.58 | knarfly | it registers but then when I try to call it says that host failed to authenticate |
| 00:55.23 | carrar | so client isn't setup correctly perhaps |
| 00:55.28 | WIMPy | Cann you call the other way? |
| 00:55.32 | carrar | needs to do proxy auth |
| 00:55.40 | WIMPy | I thought it happened when registering. |
| 00:55.56 | knarfly | I've double checked the account settings...paswd is ok, username is ok |
| 00:56.06 | WIMPy | But if it registers and only calls fail it might just be a bad message for a codec issue. |
| 00:56.11 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 00:56.15 | knarfly | I can't call it from the console either...let me try again |
| 00:56.43 | carrar | Where did you put the SIP TRace? |
| 00:56.48 | carrar | I must have missed it |
| 00:57.33 | knarfly | console dial 201 |
| 00:57.34 | knarfly | No such extension '201' in context 'default' |
| 00:57.55 | knarfly | my default context is empty as I was warned not to setup anything in default |
| 00:58.23 | WIMPy | Good idea. |
| 00:58.30 | WIMPy | So specify the correct context when calling. |
| 00:58.56 | carrar | using free zoiper or zoiper biz? |
| 00:58.57 | knarfly | how do I do that from the console...what is the syntax? |
| 00:59.06 | knarfly | carrar: FREE! |
| 00:59.10 | carrar | heh |
| 00:59.11 | carrar | OF COURSE! |
| 00:59.23 | WIMPy | should be @context |
| 00:59.26 | knarfly | I'm a cheap MF |
| 01:00.04 | knarfly | how do I dial the correct context from the console? |
| 01:00.16 | carrar | help |
| 01:00.45 | WIMPy | ext@context |
| 01:00.51 | knarfly | but I learned my lesson about buying Grandstream phones too! Don't do it unless you want bricks laying around |
| 01:01.09 | carrar | everyone has to have one laying arounf |
| 01:01.10 | carrar | d |
| 01:01.22 | carrar | I got one in the basement with a sidecar someplace |
| 01:01.35 | WIMPy | I have an Allne 7960 instead. |
| 01:01.40 | WIMPy | Allnet |
| 01:01.54 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 01:02.09 | WIMPy | The 7950 was usable, the 7960 is not. |
| 01:02.20 | carrar | I have never seen a 7950 |
| 01:02.21 | knarfly | thanks...console dial 201@internal got a ring in my headphones and I answered the call...of course the console can't talk back to myself...I will attempt to transfer it now |
| 01:02.43 | knarfly | Still need to get the MOH working like it was before...I likes that feature of Asterisk |
| 01:03.02 | WIMPy | channel originate 201@internal application sayunixtime |
| 01:10.44 | knarfly | WIMPy: that doesn't work...I get a long useage text |
| 01:10.49 | knarfly | brb |
| 01:11.21 | WIMPy | Err, right. |
| 01:11.35 | WIMPy | That should be iax2/peername instead of the extension. |
| 01:11.54 | WIMPy | or loopback/... |
| 01:12.43 | *** join/#asterisk iamcorbin (~corbin@c-98-209-175-2.hsd1.mi.comcast.net) |
| 01:12.55 | *** join/#asterisk mjordan (~mjordan@adsl-98-81-110-80.hsv.bellsouth.net) |
| 01:12.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:21.03 | iamcorbin | I'm brand new to asterisk and trying to get a simple google voice account setup, is this the only reference for doing so: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ? |
| 01:24.27 | *** join/#asterisk ZX81 (~ZX81@97.104.212.126) |
| 01:24.41 | *** join/#asterisk drcode (~user1@213.57.57.2) |
| 01:24.46 | drcode | hi all |
| 01:24.48 | drcode | whats up |
| 01:25.11 | drcode | I am trying to use linphone h264 with asterisk , is it possible? |
| 01:25.57 | WIMPy | Should |
| 01:26.45 | drcode | it seems that no video |
| 01:26.59 | WIMPy | Did you allow video? |
| 01:27.10 | drcode | I am using elastix |
| 01:27.12 | drcode | yes |
| 01:27.14 | drcode | sip.conf |
| 01:27.52 | drcode | I need to speak or somthing? |
| 01:27.57 | knarfly | WIMPy: channel originate iax2/201 application sayunixtime just times out after 30 seconds...no reply from the extension, no ring |
| 01:28.08 | drcode | mybe it dosnt start the video , I am using video conf |
| 01:28.57 | WIMPy | knarfly: But console dial works? |
| 01:30.37 | knarfly | WIMPy: that's affirmative |
| 01:31.15 | knarfly | but there is no sound card in this server so I get loads of "Unable to open DSP device /dev/dsp:" messages on the CLI |
| 01:31.28 | WIMPy | Interesting. What do you do at extension 201 other than Dial(iax2/201)? |
| 01:31.57 | knarfly | I'd like to have 201 answer the call and then console transfer it to extension 600??? |
| 01:32.41 | knarfly | WIMPy: I'm trying to setup a method for my client in Toronto to call me when we need to discuss biz using my Asterisk server. |
| 01:33.28 | WIMPy | originate can be used with an extension as well. |
| 01:33.40 | WIMPy | But maybe we should look back at the original issue. |
| 01:34.02 | WIMPy | So registering works and receiving calls also works, but no outgoing calls? |
| 01:35.11 | knarfly | that's about it...I don't understand why it will register but then the system says the host failed to authenticate...can't call voicemail extension or do an echo test...etc.! |
| 01:35.22 | drcode | dose asterisk video conf work good? |
| 01:35.29 | drcode | with h264? |
| 01:35.40 | WIMPy | Did you check codec configurations? |
| 01:35.57 | WIMPy | drcode: There's no real video conference. |
| 01:36.16 | drcode | I see |
| 01:36.35 | drcode | if I have ext mcu |
| 01:36.53 | drcode | can I forward registered users to ext mcu? |
| 01:36.55 | knarfly | I'm all green on that...I know that I had this working about a year ago. When I lost my job i let it go to retrain for other tasks. I'm dusting it all back off as I now have a use for it |
| 01:37.30 | WIMPy | Something must have changed. |
| 01:38.38 | knarfly | FreeBSD-8.2-STABLE to FreeBSD-8.3-STABLE...but that really shouldn't have upset the cart so much. If X-Lite works right out of the box I can't understand why Zoiper is not |
| 01:39.18 | WIMPy | You're using two different protocolls. |
| 01:39.31 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 01:40.07 | knarfly | Yes, but even with SIP on Zoiper is still has the same issue...btw this is a much newer version of Zoiper |
| 01:41.05 | knarfly | I was able to do an echo test using SIP with Zoiper using callwithus.com...but I couldn't hear any of the other calls to my client, nor could he hear me. |
| 01:41.21 | WIMPy | Err, you get the same message on both iax and sip? |
| 01:43.26 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 01:44.15 | knarfly | SIP on Zoiper with different account: Received SIP subscribe for peer without mailbox: 101 |
| 01:44.36 | knarfly | 101 has a mailbox...I used to use it all the time |
| 01:44.47 | WIMPy | That's notmal if you don;t have a mailbox defined. |
| 01:45.10 | WIMPy | That peer obviousely hasn't. |
| 01:47.58 | knarfly | OK the echo test went perfect although this microphone on this headset has got to go. I will try this again tomorrow when I dig out the one and only Grandstream phone which their f&^%$##@ firmware upgrade didn't brick. |
| 01:48.09 | knarfly | that was using SIP on Zoiper |
| 01:48.36 | *** join/#asterisk CrazyTux[m] (~Brandon@107-199-131-100.lightspeed.irvnca.sbcglobal.net) |
| 01:48.46 | knarfly | the iax account is still failing to authenticate |
| 01:49.28 | knarfly | but the cli says Registered IAX2 '201' (AUTHENTICATED) at 10.0.0.109:4569 |
| 01:50.37 | drcode | thnx WIMPy |
| 01:51.51 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.27.213) |
| 02:04.19 | knarfly | <PROTECTED> |
| 02:04.19 | knarfly | [Sep 11 22:03:39] NOTICE[1278]: chan_iax2.c:10894 socket_process: Host 10.0.0.109 failed to authenticate as 201 |
| 02:04.36 | knarfly | hows that for a contradiction of terms |
| 02:05.58 | knarfly | goes off to have sex with Heidi Klum 8-) |
| 02:09.04 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
| 02:22.19 | carrar | moo |
| 02:26.39 | jpsharp | meow |
| 02:40.37 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 02:45.14 | ChannelZ | clucks |
| 02:50.07 | iamcorbin | I'm new to asterisk and using this setup ( https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ) to try and get google voice working but I am not receiving any incoming calls and outgoing calls just give an extension error. Anyone have any advice on how to proceed or other resources to look at? |
| 02:57.48 | [TK]D-Fender | You should probably SHOW us this complete call with the error so we have something to at least guess with... |
| 02:57.49 | [TK]D-Fender | ~pb |
| 02:57.50 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:57.52 | [TK]D-Fender | ^ |
| 02:59.45 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 03:03.16 | iamcorbin | I just changed the setting in gmail to "Only allow people that I've explicitly approved to chat with me and see when I'm online." and now I'm getting a bunch of XMPP received from 'google' message in the asterisk CLI |
| 03:03.22 | iamcorbin | but still not receiving calls |
| 03:08.44 | jpsharp | does "jabber show connections" show you properly registered? |
| 03:09.00 | iamcorbin | here are the messages I'm getting : http://pastebin.com/xAkiyJQh |
| 03:09.01 | jpsharp | and does "gtalk show sheddings" show the right data? |
| 03:09.07 | jpsharp | er, "gtalk show settings" |
| 03:09.32 | iamcorbin | is there a way to stop all those XMPP messages |
| 03:09.35 | iamcorbin | I'm getting flooded |
| 03:09.43 | ChannelZ | yeah turn off jabber debug |
| 03:09.50 | ChannelZ | jabber set debug off |
| 03:10.27 | iamcorbin | jabber doesn't work, but xmpp set debug off worked |
| 03:10.35 | iamcorbin | and xmpp show connections shows that I am connected |
| 03:10.41 | ChannelZ | oh are you using v11 beta? |
| 03:10.44 | jpsharp | Oh, you're using 11 |
| 03:10.45 | iamcorbin | yes |
| 03:10.49 | ChannelZ | ok |
| 03:10.54 | iamcorbin | I compiled it earlier |
| 03:11.01 | jpsharp | hasn't been that masochistic yet. |
| 03:11.33 | iamcorbin | lol, I read I needed it for google voice |
| 03:11.49 | ChannelZ | GV works in other versions |
| 03:11.54 | jpsharp | I use 10.3.1 |
| 03:12.05 | ChannelZ | I ran it under 1.8 |
| 03:12.16 | iamcorbin | oh |
| 03:12.26 | ChannelZ | They've just redeveloped it a bit in 11 |
| 03:12.34 | iamcorbin | gotcha |
| 03:12.45 | ChannelZ | There was a confusing period of Jabber and Jingle and GTalk |
| 03:12.59 | iamcorbin | I'm new to all of this |
| 03:13.20 | ChannelZ | so anyway it seems to be talking to your account |
| 03:13.30 | iamcorbin | what are all those debug messages? |
| 03:13.41 | ChannelZ | mostly presence messages |
| 03:14.04 | iamcorbin | so how could I further debug why it isn't getting the incoming calls? |
| 03:14.39 | ChannelZ | well first do you have verbose turned on a little? And then what happens, any errors or you get no traffic at all? |
| 03:15.05 | iamcorbin | every once in a while I get: res_xmpp.c:3087 xmpp_pak_presence: Received presence information about '0ql9x34lkclwi0d0nw5yfmn7yc@public.talk.google.com' despite not having them in roster on client 'google' |
| 03:15.30 | iamcorbin | different x@public.talk.google.com though |
| 03:15.51 | iamcorbin | all WARNING[31596] though |
| 03:16.33 | iamcorbin | I'm running CLI with asterisk -vvvr |
| 03:16.34 | ChannelZ | Yeah that's probably just buddies. Like I assume this is an existing Google account you are using |
| 03:16.40 | iamcorbin | yes |
| 03:17.02 | ChannelZ | I believe it's basically saying "I just got status from a person I don't know about" |
| 03:17.15 | iamcorbin | ok, it didn't seem like that was the problem |
| 03:17.24 | ChannelZ | Is your Google Voice account setup to receive calls, or is it forwarding them to some other number? |
| 03:17.36 | iamcorbin | it receives calls |
| 03:17.41 | iamcorbin | it is the only number I have |
| 03:17.54 | iamcorbin | I cancelled my sprint account a couple months ago and ported my number to GV |
| 03:18.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:18.31 | iamcorbin | I've been using a couple different android apps to make and receive calls (but usually can't answer them, so that's why I'm setting asterisk up) |
| 03:18.38 | ChannelZ | ok so what happens when someone calls that number? Just nothing? |
| 03:18.54 | iamcorbin | it rings if I have one of the third party apps open on my android |
| 03:19.03 | iamcorbin | or if I'm on gmail on my laptop |
| 03:19.07 | ChannelZ | I mea on the Asterisk console, you see absolutely nothing? |
| 03:19.20 | iamcorbin | with xmpp debug off or on? |
| 03:19.50 | ChannelZ | well either really |
| 03:19.57 | iamcorbin | nothing with it off, trying again with it on |
| 03:20.39 | iamcorbin | I can't tell if any of the messages in debug are any different |
| 03:20.48 | iamcorbin | no different errors though |
| 03:21.20 | iamcorbin | or WARNINGS rather |
| 03:21.50 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 03:23.58 | ChannelZ | well you should get some sort of <iq> message with your number listed in there @voice.google.com as an initiator, etc |
| 03:25.34 | iamcorbin | just copied all the debug stuff...no "<iq>" message |
| 03:25.51 | iamcorbin | and no "voice.google.com" |
| 03:25.53 | ChannelZ | I'm not sure what to suggest besides 'check your config', having not run v11 and being that it is beta I'm not sure if it's known working/not working |
| 03:26.21 | ChannelZ | I assume you just followed the configs from wiki.asterisk.org |
| 03:26.31 | iamcorbin | the config I'm using is : https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 03:31.18 | ChannelZ | if you're not getting any significant XMPP messages on a call, either it's not registering to Google correctly or Google is not sending the messages to you for reasons unknown |
| 03:33.40 | iamcorbin | alright, thanks for the advice |
| 03:34.21 | ChannelZ | That you're getting presence info at least says you're talking to them |
| 03:34.54 | ChannelZ | I'd just double check your Google Voice settings that it is indeed set to forward calls to your Google Chat at that gmail address |
| 03:35.16 | iamcorbin | I just saw a <iq> line |
| 03:35.24 | iamcorbin | but I wasn't getting a phone call as far as I know |
| 03:35.52 | ChannelZ | well there are many iq messages |
| 03:36.16 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 03:37.12 | ChannelZ | an incoming call should be something like <iq from="+15552221111@voice.google.com" .... > and then there's a <ses:session type="initiate"> and a few others |
| 03:38.27 | ChannelZ | I should build 11 at home for the hell of it |
| 03:38.59 | iamcorbin | oh wait, I am getting an iq error: http://pastebin.com/YJD6vh75 |
| 03:40.51 | iamcorbin | feature-not-implemented? |
| 03:42.11 | ChannelZ | hmm |
| 03:42.43 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 03:43.07 | iamcorbin | it doesn't seem to be giving that every time I call though |
| 03:43.38 | hrolf | Hi # asterisk, I'm trying to connect two Asterisk boxes through SIP trunking and it works one way, i.e. from serverA to serverB but when I call from serverB to serverA then it gives error "chan_sip.c:18458 handle_response_invite: Failed to authenticate on INVITE to ..." |
| 03:45.11 | ChannelZ | both sides need a configured peer |
| 03:45.54 | carrar | and perhaps some BEER |
| 03:46.22 | iamcorbin | beer with a peer |
| 03:46.55 | hrolf | http://pastie.org/4705548 Config and SIP trace of one end |
| 03:47.06 | hrolf | http://pastie.org/4705560 another machine |
| 03:47.20 | hrolf | I have configured peer but works one way and not the other way |
| 03:48.15 | iamcorbin | looks like those <iq> errors weren't coming from my calls they are poping up occationally without me calling myself from another phone |
| 03:49.50 | ChannelZ | yeah because whatever server peer 1101 is is rejecting it (192.168.3.114) |
| 03:50.53 | ChannelZ | It looks like it's asking you to authenticate but you have no secret or anything configured for that peer so it just fails |
| 03:51.15 | ChannelZ | The other server should be complaining in a different (and probably more informative) way |
| 03:52.15 | iamcorbin | ChannelZ: was that message for me? |
| 03:52.21 | ChannelZ | no sorry |
| 03:52.49 | iamcorbin | ok |
| 03:53.07 | ChannelZ | But to you, I'm compiling v11 though and will see if I can switch over without many config changes, see if it works for me.. |
| 03:53.18 | iamcorbin | alright, awesome! |
| 03:53.39 | iamcorbin | which version are you using right now? |
| 03:54.23 | ChannelZ | 10.7.x |
| 03:57.16 | hrolf | ChannelZ: You mean I should add username= and secret= ? |
| 03:58.15 | ChannelZ | Well presumably yes but the point being that it's not authenticating on the other side for whatever peer it's matching over there |
| 03:58.36 | ChannelZ | we're only seeing half of the picture so I can't say for sure what the other side is doing |
| 03:58.53 | hrolf | ChannelZ: The username and secret should be same for both peers, right? |
| 03:59.26 | hrolf | ChannelZ: I have pasted the SIP trace and configuration of the other side too. |
| 03:59.35 | hrolf | http://pastie.org/4705560 first |
| 03:59.42 | hrolf | http://pastie.org/4705548 second |
| 04:00.06 | ChannelZ | oh I missed it sorry |
| 04:02.31 | ChannelZ | So it's presumably supposed to match the "avaya" peer based on the IP but it's not because it's coming from "101" |
| 04:03.35 | ChannelZ | I think the way you have it now if you changed the [avaya] to type=friend it would match by IP and work |
| 04:03.49 | ChannelZ | errr sorry type=user |
| 04:03.57 | hrolf | Let me try |
| 04:05.18 | hrolf | ChannelZ: Nope, same error. |
| 04:05.34 | ChannelZ | I'm also not sure what your registersip=yes is, that's not an option I've ever heard of |
| 04:05.53 | ChannelZ | hmm isn't 192.168.3.5 the machine you're trying to call from? |
| 04:06.13 | hrolf | Yes, .5 is the machine I'm calling from and .114 is the target. |
| 04:06.14 | ChannelZ | oh... actually you do have a peer called 101 too it's probably matching that one |
| 04:06.35 | ChannelZ | you have some cross-config happening here |
| 04:07.16 | hrolf | Yes, .114 have three SIP extensions, 1) avaya type=peer, 2) 100 and 3) 101 both type=friend. |
| 04:08.10 | hrolf | ChannelZ: |
| 04:08.20 | hrolf | ChannelZ: I removed the two peers and it worked. |
| 04:08.24 | hrolf | ChannelZ: Strange? |
| 04:08.27 | ChannelZ | yeah you have two matching peers |
| 04:08.34 | ChannelZ | it was actually matching the [101] one |
| 04:08.40 | ChannelZ | I missed the line in the paste |
| 04:08.41 | ChannelZ | Found peer '101' for '101' from 192.168.3.5:5060 |
| 04:08.56 | ChannelZ | However that peer has a secret, but your calling machine did not (thus the auth fail) |
| 04:09.09 | hrolf | ChannelZ: How is it doing the matching? |
| 04:09.24 | ChannelZ | well that's what the peer type= does |
| 04:09.57 | ChannelZ | either by IP, by the From: or by a bit of both. The docs are terrible |
| 04:10.28 | hrolf | ChannelZ: It isn't documented? |
| 04:10.34 | ChannelZ | It is, poorly |
| 04:11.02 | ChannelZ | I can never remember which is which and the sample sip.conf is pretty much wrong |
| 04:11.41 | ChannelZ | I believe type=user matches by the From: header, type=peer matches by IP, and type=friend tries to match a From: first and then an IP failing that. |
| 04:12.26 | ChannelZ | But you had two peers that could possibly match, [101] by From: and [avaya] by the static IP you had in there |
| 04:12.57 | ChannelZ | and it was matching that one first |
| 04:13.23 | ChannelZ | ("that one" being 101) |
| 04:13.56 | hrolf | ChannelZ: I see. |
| 04:14.42 | hrolf | ChannelZ: Anyways, thanks. Was stuck on it the whole day (yesterday.) |
| 04:16.32 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 04:18.40 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 04:19.34 | ChannelZ | sure |
| 04:20.35 | ChannelZ | iamcorbin: well my dialplan crashed and burned but I at least got a slew of messages when I tried to call |
| 04:21.13 | iamcorbin | is yours quite different from : https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ? |
| 04:22.03 | ChannelZ | not much. Perhaps you should pastebin your xmpp.conf and motif.con (blank your password) |
| 04:23.31 | ChannelZ | interesting.. now it's NOT working. |
| 04:24.34 | iamcorbin | http://pastebin.com/KpqBQs1M |
| 04:24.42 | iamcorbin | hmm |
| 04:25.29 | ChannelZ | I seem to be receiving an identical call setup message but Asterisk is sitting doing nothing. |
| 04:26.38 | ChannelZ | reloaded.. call 1 worked.. |
| 04:26.52 | ChannelZ | call 2 worked |
| 04:26.53 | ChannelZ | hmm |
| 04:26.57 | ChannelZ | now it seems fine |
| 04:28.29 | iamcorbin | and my xmpp.conf and motif.conf? |
| 04:28.53 | ChannelZ | they look ok, but let me try one thing |
| 04:30.37 | ChannelZ | in your xmpp.conf, change [default] to [general] - dunno if that's busting the parser or what |
| 04:31.24 | ChannelZ | I think it is |
| 04:31.35 | iamcorbin | oops, that is already [general] |
| 04:31.42 | iamcorbin | I typed that line in |
| 04:31.49 | iamcorbin | since there are lots of comments between them |
| 04:31.50 | ChannelZ | in the patebin you mean? |
| 04:31.58 | iamcorbin | yeah, that was a typo in pastebin |
| 04:32.54 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-58-179.bois.qwest.net) |
| 04:32.58 | iamcorbin | and all the things under [default] in motif.conf were already there, I only added lines [google] on |
| 04:33.14 | iamcorbin | do I need to change any of those defaults? |
| 04:34.10 | ChannelZ | not necessarily no. I have none at the moment |
| 04:34.12 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-bckijnifwimtlbmd) |
| 04:35.12 | iamcorbin | I'll try commenting them |
| 04:35.18 | iamcorbin | how do you reload xmpp.conf? |
| 04:35.38 | ChannelZ | module reload res_xmpp |
| 04:36.39 | ChannelZ | remove priority=1 |
| 04:37.05 | ChannelZ | well no that's not it. This thing is just generally misbehaving. |
| 04:37.13 | ChannelZ | Don't bother reloading, stop asterisk completely. |
| 04:37.19 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 04:37.21 | ChannelZ | The module seems to sort of die or something. |
| 04:38.58 | iamcorbin | so it's not working for you either? |
| 04:40.24 | ChannelZ | no, it is now |
| 04:41.05 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 04:41.20 | ChannelZ | In playing around trying to break it, the module seemed to just konk out. When I'd do a core stop, it'd quit but then I'd see an asterisk task running still |
| 04:41.52 | ChannelZ | but it's all working now |
| 04:41.57 | iamcorbin | core stop doesn't work for me |
| 04:42.00 | iamcorbin | how do I shut it down? |
| 04:42.05 | ChannelZ | core stop gracefully |
| 04:42.19 | ChannelZ | then ps ax |grep asterisk and make sure it's not still hung |
| 04:42.50 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 04:42.55 | iamcorbin | weird |
| 04:42.59 | iamcorbin | it is still running |
| 04:43.07 | iamcorbin | but I go to kill {process number} and it changes numbers |
| 04:43.16 | ChannelZ | killall asterisk |
| 04:43.20 | iamcorbin | nevermind |
| 04:43.22 | ChannelZ | are you running safe_asterisk |
| 04:43.31 | iamcorbin | that was just the grep command |
| 04:43.32 | iamcorbin | doh |
| 04:43.35 | iamcorbin | it is stopped |
| 04:43.36 | ChannelZ | hahah |
| 04:44.11 | iamcorbin | and then I'm starting it running just "asterisk" and then "asterisk -vvvr" is that proper? |
| 04:44.59 | ChannelZ | mostly |
| 04:45.19 | iamcorbin | mostly? |
| 04:45.34 | ChannelZ | it's fine |
| 04:45.45 | iamcorbin | alright |
| 04:45.47 | ChannelZ | I forgot it starts detached by default |
| 04:48.33 | iamcorbin | anything else different between our confs? |
| 04:49.35 | ChannelZ | Not really. Mine are clean though, not with the sample junk. And I don't have the template in xmpp.conf |
| 04:49.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:50.42 | ChannelZ | http://pastebin.com/zpkHemF8 |
| 04:52.15 | iamcorbin | and those are your entire files? |
| 04:53.35 | ChannelZ | yup |
| 04:53.49 | iamcorbin | I'll try getting rid of all the extra junk in mine |
| 04:54.03 | ChannelZ | Do you have any other devices logged in to that account at the same time? IMs, apps on your phone... |
| 04:54.22 | *** join/#asterisk kannan (~chatzilla@123.238.235.147) |
| 04:54.39 | iamcorbin | google talk has been logged in |
| 04:54.57 | iamcorbin | that's just txt though |
| 04:55.05 | iamcorbin | would that matter? |
| 04:55.35 | ChannelZ | dunno, maybe. I'm just not sure why it wouldn't be sending you an xmpp on a call |
| 04:56.08 | iamcorbin | what does your xmpp look like when it receives? |
| 04:56.19 | ChannelZ | I know it's a stupid question but that is really the gmail account associated with your Google voice number,right? |
| 04:57.05 | iamcorbin | yep, only have one gmail account |
| 04:58.15 | *** join/#asterisk kannan (~chatzilla@123.238.235.147) |
| 04:58.21 | ChannelZ | http://pastebin.com/gB2HDd6P |
| 05:04.18 | iamcorbin | ok, so I'm looking for an initiate or a telephone-event? |
| 05:06.06 | ChannelZ | well yeah it's the first giant <iq> message you should get |
| 05:08.12 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 05:11.31 | iamcorbin | ::smacks head:: |
| 05:11.35 | iamcorbin | sorry to waste your time |
| 05:12.06 | iamcorbin | I just went into my list of running apps on my nexus 7 and killed all google things |
| 05:12.11 | iamcorbin | something was intercepting it |
| 05:12.19 | iamcorbin | it comes in and connects and is working now! |
| 05:12.35 | iamcorbin | I also trimmed down the configs |
| 05:13.15 | iamcorbin | now I just need to figure out how to get outgoing working |
| 05:13.30 | iamcorbin | it gives me an extension error, but I'm assuming I'm missing something in my config |
| 05:14.28 | iamcorbin | can I see your outgoing GV extension? |
| 05:15.51 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:16.36 | ChannelZ | sorry doing other things |
| 05:16.49 | iamcorbin | no prob |
| 05:16.59 | ChannelZ | I don't have one, but you should basically be able to Dial(motif/google/12345) |
| 05:17.22 | iamcorbin | then how would I set that up to be able to dial any number from my sip phone? |
| 05:17.35 | iamcorbin | I don't know enough about dialplans yet |
| 05:17.36 | ChannelZ | motif being the channel tech, google is the account (which is [google] in your configs) and 12345 is the number to dial, which probably has to be formatted as 1xxxXXXxxxx or something |
| 05:18.06 | iamcorbin | I get that I can call Dial in extension.conf but how do I send that from my SIP phone? |
| 05:18.13 | ChannelZ | well something like exten => _NXXNXXXXXX,1,Dial(Motif/google/1${EXTEN}) |
| 05:18.23 | iamcorbin | using the Xs? |
| 05:18.34 | ChannelZ | yeah, N and X are patterns |
| 05:19.10 | iamcorbin | oh ok |
| 05:19.45 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 05:20.12 | iamcorbin | the "Calling using Google" doc gives: exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 05:20.18 | iamcorbin | that's what I'm trying to use |
| 05:20.41 | iamcorbin | I made an [outgoing-motif] section under [incoming-motif] in extensions.conf |
| 05:21.13 | iamcorbin | so I need to link that to a SIP phone somehow? |
| 05:22.05 | ChannelZ | well that extension has to exist in whatever context the device you're using is in |
| 05:22.48 | iamcorbin | alright, I'm doing that completely wrong then |
| 05:23.44 | iamcorbin | does that '_' mean I have to dial a '_' |
| 05:23.49 | ChannelZ | hmm this beta is flakey |
| 05:24.01 | ChannelZ | no _ means "this extension is a pattern" |
| 05:24.20 | *** join/#asterisk CrazyTux[m] (~Brandon@107-199-131-100.lightspeed.irvnca.sbcglobal.net) |
| 05:24.23 | ChannelZ | which makes X mean "any number 0-9" |
| 05:24.35 | iamcorbin | and N? |
| 05:24.52 | ChannelZ | 2-9 |
| 05:25.20 | iamcorbin | there we go |
| 05:25.23 | iamcorbin | just needed to dial 1 |
| 05:26.40 | iamcorbin | sweet, all good now. Thanks for the help |
| 05:27.41 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 05:28.42 | ChannelZ | sure |
| 05:30.29 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 05:30.33 | fling | wow! |
| 05:30.39 | fling | it just started to work |
| 05:30.43 | fling | 15 minutes ago |
| 05:30.52 | fling | peer is reachable and registered |
| 05:30.59 | ChannelZ | whowhaa? |
| 05:31.05 | fling | I did not touched it 2 days |
| 05:31.24 | fling | [Sep 12 12:15:03] NOTICE[4233]: chan_sip.c:13139 sip_reg_timeout: -- Registration for 'xxx' timed out, trying again (Attempt #8624) |
| 05:31.36 | fling | [Sep 12 12:15:16] NOTICE[4233]: chan_sip.c:20798 handle_response_peerpoke: Peer 'xxx' is now Reachable. (118ms / 2000ms) |
| 05:31.43 | fling | it is a miracle! :p |
| 05:31.45 | ChannelZ | You must have Official Soviet Russia Internet |
| 05:31.53 | fling | ChannelZ: yes |
| 05:32.14 | ChannelZ | (are you really in RU?) |
| 05:32.17 | fling | yes |
| 05:32.21 | ChannelZ | oh ahahahah |
| 05:32.23 | fling | привет, я из России |
| 05:32.25 | ChannelZ | I was just being mean |
| 05:32.29 | ChannelZ | who knew! |
| 05:32.29 | fling | ChannelZ: woot? |
| 05:32.32 | fling | umm |
| 05:32.45 | ChannelZ | sorry no offense |
| 05:32.49 | fling | hehe :p |
| 05:33.10 | ChannelZ | I guess I shouldn't knock russian internet, it seems to work OK based on the amount of spam I get from *.ru hosts... :| |
| 05:33.17 | *** join/#asterisk kannan (~chatzilla@122.174.38.132) |
| 05:33.37 | fling | ChannelZ: I have a lots of spam too, still learning my spamassassin from a honeypot |
| 05:34.28 | fling | ChannelZ: the only change I made two days ago is iptables rule on all of my routers > -A FORWARD -i metro -o some_tunnel -m state --state RELATED,ESTABLISHED -j ACCEPT |
| 05:35.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:35.11 | fling | ChannelZ: may it be related to the miracle? :p |
| 05:37.37 | ChannelZ | hard to say |
| 05:38.11 | ChannelZ | if a bad entry got stuck in the NAT table or something |
| 05:41.05 | fling | hmm hmm |
| 05:41.35 | fling | ChannelZ: I have 100Mb port at home so russian internets are fine |
| 05:41.48 | fling | ChannelZ: but I have insanely big pings to asia |
| 05:42.01 | ChannelZ | oh is that where that phone is? |
| 05:42.06 | ChannelZ | (or whatever it is) |
| 05:42.12 | fling | ChannelZ: and with my current ISP I have 50ms to europe (Moscow) |
| 05:42.24 | fling | ChannelZ: yes |
| 05:42.44 | fling | s/50ms/130ms/ |
| 05:42.50 | fling | is sad |
| 05:43.10 | fling | ChannelZ: so should I keep this iptables rule? |
| 05:43.44 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 05:44.29 | fling | ChannelZ: may you please ping smtown.com? I have 400ms |
| 05:45.07 | CrazyTux[m] | fling: 305 |
| 05:45.17 | fling | CrazyTux[m]: where are you from? |
| 05:45.23 | CrazyTux[m] | fling: West Coast USA. |
| 05:45.45 | fling | smtown.com is near my location but my ping is longer :\ |
| 05:45.49 | CrazyTux[m] | aka Irvine, CA. |
| 05:48.14 | ChannelZ | ~200 avg for me in CO USA |
| 05:51.04 | hrolf | Where can I configure the recording path in Asterisk? (I use MixMonitor.) |
| 05:52.26 | ChannelZ | just put the path in your MixMonitor() |
| 05:53.06 | ChannelZ | (absolute paths are allowed) |
| 05:53.32 | fling | ChannelZ: do I need to use -m state --state RELATED,ESTABLISHED -j ACCEPT on my nat router? |
| 05:53.36 | hrolf | ChannelZ: There is no configuration for the default location? |
| 05:53.48 | ChannelZ | Yes, it's in asteris.conf I think |
| 05:54.24 | ChannelZ | usually /var/spool/asterisk and then it goes in the 'monitor' directory there |
| 05:55.19 | ChannelZ | fling: I'll go with "yes"? |
| 05:55.38 | fling | ChannelZ: ok :p thanks |
| 05:56.02 | fling | thank you guys for your help! now I can call |
| 05:56.42 | ChannelZ | rolled back to 10.7 |
| 05:57.15 | hrolf | ChannelZ: You involved in asterisk dev too? |
| 05:57.15 | fling | ChannelZ: you can have 100Mb port at home for about $30/month |
| 05:57.20 | ChannelZ | dunno what was causing 11 to behave so badly but I don't have time to ffffuuu with it |
| 05:57.27 | fling | ChannelZ: if you live in Russia :p |
| 05:57.32 | ChannelZ | No, just a user |
| 05:58.12 | ChannelZ | And I'm in the US. I pay ~$80 for 12mbit. 1st world country and everything. |
| 05:58.30 | ChannelZ | and that's not even symmetric |
| 06:00.15 | fling | ChannelZ: http://lurkmore.so/images/d/dd/Live_in_russia.jpg |
| 06:00.29 | ChannelZ | heheeh |
| 06:03.01 | fling | ChannelZ: Meanwhile in Russia > http://maps.yandex.ru/-/CVeinWoF |
| 06:04.45 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
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| 06:25.11 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:25.13 | schmidts | good morning |
| 06:30.44 | ChannelZ | whoo! |
| 06:38.26 | *** join/#asterisk Corydon76-home (six@c-69-137-80-31.hsd1.tn.comcast.net) |
| 06:38.27 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 06:54.08 | mealstrom | does anyone use jitsi sip phone? |
| 06:58.31 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 07:13.27 | fling | I have some old phones |
| 07:13.43 | fling | Do I need to buy a sip pata converter/router or sip phones? |
| 07:14.06 | fling | (people do not want to use softphones) |
| 07:20.52 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 07:22.22 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 07:28.32 | fling | is D-link DVG-5402SP good? |
| 07:31.29 | schmidts | fling when you say old phones you mean pots phones? i dont know the d-link well but i mostly use cisco pap2t ATAs and they work very well |
| 07:31.48 | fling | schmidts: give me a link please |
| 07:38.38 | *** join/#asterisk alsuren (~dlaban@80.169.133.251) |
| 07:39.24 | *** join/#asterisk bobby_ct (hidden-use@mail.imagemakers.co.za) |
| 07:40.02 | bobby_ct | Morning all anyone awake |
| 07:41.53 | schmidts | fling www.cisco.com ;) |
| 07:41.54 | fling | how to change first number from 8 to 7 in dial? exten => _8.,1,Dial(SIP/${EXTEN}@pbx,40) |
| 07:41.59 | fling | schmidts: lol :P |
| 07:42.31 | fling | how to sed in ${EXTEN} ? |
| 07:42.50 | schmidts | fling http://www.cisco.com/en/US/products/ps11977/index.html |
| 07:43.22 | schmidts | fling dont do a dial, just use a goto like exten => _8.,1,Goto(7${EXTEN:1},1) |
| 07:43.57 | fling | schmidts: will it work instead of dial? :x |
| 07:44.22 | schmidts | even better cause asterisk will not start a whole new call for it, just move this call to the new extension |
| 07:45.59 | fling | schmidts: http://dpaste.com/799796/ ? |
| 07:46.02 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 07:46.07 | ChannelZ | How is the Goto a better solution? |
| 07:46.31 | ChannelZ | I mean I guess if you have a ton of dialplan afterwords to recycle.. |
| 07:46.42 | fling | so is this ^ fine if I want to be able to call with 7 and with 8? |
| 07:47.10 | ChannelZ | ok then nm |
| 07:47.21 | fling | ChannelZ: umm? :p |
| 07:47.29 | fling | am I doing it right? |
| 07:47.31 | ChannelZ | I answered my own question after your paste |
| 07:47.42 | fling | great! |
| 07:47.42 | ChannelZ | yeah |
| 07:47.55 | fling | ChannelZ: schmidts: thanks |
| 07:48.31 | ChannelZ | assuming you want it to be that if you dial a number starting with a 7 it gets changed to an 8 even in the actual dial? |
| 07:48.51 | ChannelZ | or the other way around I mean |
| 07:49.02 | ChannelZ | (8 to 7) |
| 07:50.51 | fling | 8 should be changed to 7 |
| 07:51.19 | fling | am I using right priority numbers? http://dpaste.com/799797/ |
| 07:51.46 | ChannelZ | Just curious, is that a steering digit, or what is the significance of 8 or 7? (country thing?) |
| 07:52.13 | fling | 7 is a country digit, sip provider wants so use 7 |
| 07:52.15 | ChannelZ | looks like it |
| 07:52.37 | fling | 8 is a common pstn thing to call out from a town to another town |
| 07:53.11 | ChannelZ | ah. |
| 07:53.45 | fling | so if you use pstn you dial 8 then town code… |
| 07:54.03 | fling | and if you use sip you dial country code then town code |
| 07:54.41 | fling | 383 is a town code I want to add to a number if it is _XXXXXXX |
| 07:55.42 | ChannelZ | make sense |
| 07:55.46 | ChannelZ | s/make/makes/ |
| 07:56.07 | fling | sedbot! |
| 07:57.08 | schmidts | sedbot FTW! ;) |
| 07:57.10 | fling | right, it works, I've simplified my dialplan a lot. I'm removing lines |
| 07:57.51 | schmidts | fling thats also why you use goto, its just easier, but there is a much bigger background about it which goes deeper into asterisk call handling itself |
| 07:57.59 | mealstrom | fling: -- _8. -- thats bad practice. number 800 goes there |
| 07:58.17 | fling | right, removing :p |
| 07:58.29 | *** part/#asterisk bobby_ct (hidden-use@mail.imagemakers.co.za) |
| 07:59.32 | schmidts | fling if you want to make sure use something like this exten => _8[1-9].,1, ... |
| 07:59.43 | mealstrom | fling: -> im going from short to long numbers . like XXXXX -> +380citycodeXXXXX. |
| 08:00.03 | fling | schmidts: umm? |
| 08:00.26 | mealstrom | fling: from dpate.com look like you are from ukraine ? |
| 08:00.29 | schmidts | with this extension it only matches to numbers which dont start with 80xxx like 812 |
| 08:00.42 | fling | schmidts: got it, sorry |
| 08:00.55 | fling | mealstrom: from Russian Federation |
| 08:00.59 | ChannelZ | _8Z. |
| 08:01.01 | schmidts | and about priority numbers you can also use same => |
| 08:01.09 | mealstrom | fling: i was close :D |
| 08:01.21 | fling | ChannelZ: Z is 'not zero'? |
| 08:01.26 | ChannelZ | no, 'same' is for extension, priority is 'n' |
| 08:01.33 | ChannelZ | Z is 1-9 shorthand |
| 08:01.41 | ChannelZ | X is 0-9, Z is 1-9, N is 2-9 |
| 08:02.06 | schmidts | ChannelZ yes didnt say it clear enough ;) |
| 08:02.12 | schmidts | fling what i meant looks something like this http://dpaste.com/799802/ |
| 08:02.44 | fling | wait I need to call my ISP and ask how should I dial if I call to 800 |
| 08:02.51 | fling | usualy we dial 8800 here |
| 08:05.18 | fling | whoops isp girl said I'm not allowed to call 800 numbers |
| 08:05.44 | *** join/#asterisk racho (~racho@46.40.123.204) |
| 08:06.02 | ChannelZ | sinner! |
| 08:06.12 | fling | noooo!!! :p |
| 08:11.27 | fling | Which standard sound file do I need if I want digital girl to say something like 'You are not allowed to call this number'? |
| 08:17.19 | ChannelZ | pbx-invalid maybe, or invalid |
| 08:18.16 | ChannelZ | heh those are both the same on mine. Hrm. |
| 08:18.27 | *** join/#asterisk mutex7c (~mutex7c@80.150.220.106) |
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| 08:21.48 | fling | ChannelZ: thanks |
| 08:22.32 | ChannelZ | you probably want to record your own that's more relevant else someone just keeps trying over and over |
| 08:22.59 | fling | or I may use tt-weasels |
| 08:23.07 | ChannelZ | access-denied might be a good one followed by tt-monkeys |
| 08:23.15 | fling | right :p |
| 08:24.51 | ChannelZ | sorry-cant-let-you-do-that |
| 08:26.29 | ChannelZ | that-is-not-rec-phn-num |
| 08:26.40 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:27.36 | ChannelZ | eww bleh I had a ton of gsm duplicates in my sounds. |
| 08:27.37 | fling | no russian language for this sound :p |
| 08:27.57 | ChannelZ | Oh. Well it might not even be a standard one. I'm using a Pat Fleet sound set. |
| 08:27.58 | fling | why are you using gsm? do I need gsm? |
| 08:28.19 | ChannelZ | I wasn't, I just wiped them out 5 seconds ago. Probably got installed by accident at some point |
| 08:28.38 | fling | I have net-misc/asterisk-core-sounds net-misc/asterisk-extra-sounds net-misc/asterisk-moh-opsound installed |
| 08:28.57 | fling | I use only alaw, it works fine |
| 08:29.00 | ChannelZ | If you have lots of devices using gsm it's beneficial to have them so Asterisk doesn't have to transcode, but otherwise there's no reason |
| 08:29.13 | fling | right… |
| 08:29.18 | ChannelZ | it sounds like ass anyway |
| 08:32.49 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:34.56 | fling | how to automagically wait few seconds and dial 0 dtmf tone when I call to some number? |
| 08:35.21 | *** join/#asterisk rolandow (~roland@92.68.81.83) |
| 08:36.21 | ChannelZ | look at the D option of Dial |
| 08:36.22 | rolandow | is anybody familiar with monitoring software like solarwinds? |
| 08:36.47 | ChannelZ | not sure if that will work for your purposes |
| 08:39.17 | ChannelZ | hmm www.solarwinds.com is hosed |
| 08:40.01 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 08:40.40 | rolandow | what do you mean with hosed? |
| 08:41.04 | ChannelZ | not working. It's back now. |
| 08:41.04 | rolandow | we received spam from solarwinds about a voip quality monitor .. something like that.. |
| 08:41.23 | rolandow | i hate spam :) |
| 08:41.30 | ChannelZ | This is the first time I've heard of them |
| 08:41.37 | rolandow | but on the other hand i was wondering if people are using tools to monitor the quality |
| 08:41.37 | ChannelZ | Don't encourage them then by giving them money |
| 08:41.45 | rolandow | i don't |
| 08:41.56 | rolandow | i just want to know if there are similar tools to do proper monitoring |
| 08:42.19 | rolandow | a lot of people here are complaining about dropped calls or bad quality .. i think they just need to get used to some changes :) |
| 08:42.35 | rolandow | but it would be nice to be able to constantly monitor this |
| 08:43.25 | rolandow | maybe there are tools who can detect that the rtp stream has hickup's .. |
| 08:43.36 | rolandow | or temporary increase of jitter |
| 08:43.48 | rolandow | or whatever variables there are to make it a good phonecall |
| 08:45.00 | ChannelZ | guess it depends on which way the hiccups are. You can only control jitter on incoming audio |
| 08:45.34 | rolandow | exactly .. so i need to know if there actually ARE hickups .. or are people just complaining for nothing? |
| 08:45.40 | rolandow | and if there are: where do they come from? |
| 08:46.07 | rolandow | so the whole idea of monitoring this, that at least i liked about the solarwinds advertisement :) |
| 08:47.07 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:fdf3:f3ae:a3fd:bd5f) |
| 08:49.08 | ChannelZ | understandable |
| 08:51.06 | *** part/#asterisk mealstrom (~mealstrom@46.63.71.254) |
| 08:52.25 | rolandow | hm... there are several opmanagers, but they need cisco ip sla |
| 08:56.35 | rolandow | maybe something like voipmonitor.org ? |
| 09:08.00 | fling | ChannelZ: exten => _00000,1,Dial(SIP/00000@zadarma,40,D(0)) |
| 09:08.10 | rolandow | hm.. is that true? it would have the last value of the array then?? |
| 09:08.15 | rolandow | damn |
| 09:08.16 | fling | ChannelZ: ^ it is not dialing additional 0 |
| 09:08.17 | rolandow | wrong window again |
| 09:11.35 | fling | schmidts: how can I simplify this? > http://dpaste.com/799834/ |
| 09:14.35 | *** join/#asterisk kannan (~chatzilla@123.238.235.147) |
| 09:15.40 | kannan | hello in sip setting, when an attempted register is made, what is the setting to be made so that the sip user cannot know whether the username at least is valid ? is it alwaysauthreject=yes ? |
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| 09:31.36 | wdoekes | kannan: correct. and that setting is on by default |
| 09:31.58 | kannan | wdoekes thanks, |
| 09:36.20 | RiceCracker | in asterisk 1.8 is there a bug with Anonymous@anonymous.invalid showing everytime there is an Anonymous call? |
| 09:36.24 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 09:37.05 | RiceCracker | or is there somewhere where I can change this incoming caller ID to just show "Unavailable" or "Blocked" |
| 09:37.07 | RiceCracker | ? |
| 09:37.17 | kaldemar | RiceCracker: are you asking if it is a bug that you see Anonymous@anonymous.invalid? |
| 09:37.25 | fling | kaldemar: hello |
| 09:38.23 | kaldemar | fling: howdy. |
| 09:38.34 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 09:39.08 | kaldemar | RiceCracker: to change that you'd have to modify the source code. |
| 09:41.06 | wdoekes | or modify the dialplan so setcallerpres=allowed and set the callerid(num)=Blocked |
| 09:42.42 | fling | kaldemar: I want to send 0 dtmf tone but it is not sending > exten => _00000,1,Dial(SIP/00000@zadarma,40,D(0)) |
| 09:42.50 | fling | kaldemar: do I need to remove timeout? |
| 09:43.10 | kaldemar | removing timeout won't help. |
| 09:43.33 | fling | hmm hmm |
| 09:43.44 | RiceCracker | I wanted to change blocked CID's to display as Blocked instead of coming in as "From Anonymous Anonymous@anonymous.invalid" |
| 09:43.53 | fling | kaldemar: what am I doing wrong? :p |
| 09:44.23 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 09:44.32 | gavimobile | I was so sure that I was already in the channel |
| 09:44.40 | kaldemar | fling: what do you want to do exactly? |
| 09:44.49 | RiceCracker | when i google search how to change that i seem to keep coming across forms saying there is a bug etc... |
| 09:44.59 | gavimobile | if a call doesn't go through all the time, who would be to blame? the itsp? |
| 09:45.09 | fling | kaldemar: I call support and digital girl says press '0' to connect the operator |
| 09:45.27 | fling | kaldemar: I want asterisk to dial that '0' dtmf automagically when I call |
| 09:45.29 | gavimobile | the call was charged for by the itsp though, just didn't go through on my end |
| 09:45.43 | kaldemar | fling: try adding some w's before the "0". a w means 500 msec pause. |
| 09:45.50 | fling | oh! great |
| 09:46.24 | RiceCracker | Fling: "digital girl" name is Allison :P |
| 09:46.31 | fling | RiceCracker: why? |
| 09:46.54 | RiceCracker | because that is her real name |
| 09:47.05 | RiceCracker | you would have to ask her parents why she was named that :P |
| 09:47.06 | fling | kaldemar: ,D(wwww0) < this works, I do not hear Allison anymore |
| 09:47.35 | fling | RiceCracker: she is not in the channel, how can I ask her parents :p |
| 09:49.26 | RiceCracker | Kaldemar: so pretty much nothing i can do about chaning the display of blocked CID from Anonymous to Blocked or Unknown? |
| 09:49.58 | fling | I have few similar parts it my extensions like Dial,Playback,Hangup. I want to simplify it, may I somehow join it together? > http://dpaste.com/799844/ |
| 09:50.15 | fling | kaldemar: thanks :) |
| 09:51.12 | kaldemar | fling: np |
| 09:51.42 | kaldemar | RiceCracker: wdoekes gave you an approach |
| 09:52.36 | RiceCracker | oh i see, let me try |
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| 10:48.05 | racho | how can i get a sip trunk's name in an agi script? |
| 10:57.41 | kaldemar | racho: when? |
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| 11:16.58 | knarfly | 8-) |
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| 11:21.26 | knarfly | I keep getting facility not subscribed error with Zoiper using IAX...anyone else have this issue |
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| 11:40.25 | knarfly | <PROTECTED> |
| 11:40.25 | knarfly | So if the account is registered and Authenticated, why do I keep seeing Host 10.0.0.109 failed to authenticate as 201? |
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| 11:52.36 | gavimobile | how do I make a dial plan rule for several numbers, for example exten = 106,... && exten = 107,... && exten = 108,... |
| 11:55.33 | gavimobile | nevermind, I got a better idea |
| 11:58.10 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-58-179.bois.qwest.net) |
| 12:01.45 | kaldemar | gavimobile: use patterns. exten => _10[678],1,... |
| 12:06.18 | blitzrage | gavimobile: check the Dialplan Basics chapter at asteriskdocs.org for information on dialplan 101 |
| 12:10.43 | knarfly | needs help with iax |
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| 12:33.22 | gavimobile | thanks folks |
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| 12:47.50 | gavimobile | kaldemar: so if I want to create a dialplan for 100-108, I can do this? exten => _1[108]X,1, to Matches a single character from the range of digits specified. In this case, the pattern matches a single 1, as well as any number in the range 0, 6, 8. |
| 12:48.05 | gavimobile | ? |
| 12:48.55 | MLNoah | gavi: what you specified would match 100-109, 110-119, and 180-189 |
| 12:49.06 | MLNoah | if you want to match 100-108, you want _10[0-8] |
| 12:49.17 | MLNoah | or _10X and have a 109 extension to overload so that 109 doesn't match _10X |
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| 12:49.42 | gavimobile | I have now _10X but I don't want them to call 109 |
| 12:49.52 | gavimobile | I see |
| 12:49.56 | blitzrage | then create a 109,1,NoOp() exten |
| 12:50.00 | gavimobile | I like _10[0-8] better |
| 12:50.02 | blitzrage | most specific will match first |
| 12:50.06 | gavimobile | what do you say? |
| 12:50.59 | MLNoah | personally i find _10X and having a 109,1,NoOp() or 109,1,Goto(i,1) to be easier for me to come back and read later |
| 12:51.05 | MLNoah | but either way is functional. |
| 12:54.29 | gavimobile | MLNoah: is that method depreciated or something? |
| 12:54.39 | gavimobile | I make a lot of notes... |
| 12:54.56 | gavimobile | and I like putting use to all different types of situations so I can apply it to other places where I need to later on |
| 12:55.29 | MLNoah | to my knowledge, either way is supported. people will put forth using Verbose() instead of NoOp() for throw-away lines, but all of what I listed is supported. |
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| 12:59.19 | gavimobile | MLNoah: thanks again |
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| 14:21.14 | gavimobile | im having trouble with the voicemail options |
| 14:21.54 | gavimobile | when I call a certain extention the voicemail plays my custom greeting than after it plays vm-intro |
| 14:22.25 | gavimobile | I changed the options to us and su which solves the problem if I call locally to that extentions voice mailbox |
| 14:22.39 | gavimobile | but if I call from my cellphone it doesn't solve the problem |
| 14:23.45 | [TK]D-Fender | * knows nothing about "cellphone" |
| 14:23.57 | [TK]D-Fender | the channel calling in has nothing to do with this |
| 14:24.41 | gavimobile | [TK]D-Fender: my incoming did forwards to my pbx, my pbx forwards to a time condition I created. if out of time condition it sends the caller to a voicemailbox |
| 14:25.02 | gavimobile | if it is in the time condition, the call is sent to a queue. |
| 14:25.10 | [TK]D-Fender | gavimobile, None of that means anything. VoiceMail does what it is told. It is not dependent on some property of the calling channel. |
| 14:26.20 | gavimobile | [TK]D-Fender: well its good to know that its not dependent on the channel, but im still clueless to know why I don't hear vm-intro when I don't call from internal |
| 14:26.36 | [TK]D-Fender | gavimobile, Where is the debug? |
| 14:26.50 | [TK]D-Fender | gavimobile, Where are the folder dumps? |
| 14:28.41 | gavimobile | [TK]D-Fender: here is an internal call which works fine |
| 14:28.43 | gavimobile | http://pastebin.com/qVzQbJNA |
| 14:28.49 | gavimobile | now ill get the debug from an external call |
| 14:33.06 | gavimobile | [TK]D-Fender: I found the problem! |
| 14:33.08 | gavimobile | thanks anyways |
| 14:33.27 | [TK]D-Fender | Helps when you actually look.... |
| 14:34.50 | gavimobile | of course I looked [TK]D-Fender.. |
| 14:35.00 | gavimobile | there was a goto which the timecondition was forwarding the user |
| 14:35.10 | gavimobile | so Voicemail application was in my dialplan twice |
| 14:35.17 | gavimobile | with that specific extention |
| 14:35.36 | [TK]D-Fender | and all of that is just useless "story" that you tell us while coming in here telling us VoiceMail is broken. |
| 14:35.51 | gavimobile | [TK]D-Fender: good point |
| 14:36.13 | gavimobile | but at least you helped me gain the confidence to solve one more problem |
| 14:36.15 | gavimobile | :-p |
| 14:36.19 | gavimobile | thanks again! |
| 14:36.53 | [TK]D-Fender | I think you mean "kicked into looking at what I'm actually doing"... |
| 14:37.08 | [TK]D-Fender | But sure, I'll accept some credit for that at least... |
| 14:37.33 | *** join/#asterisk MariusKarthaus (~quassel@541F8611.cm-5-8c.dynamic.ziggo.nl) |
| 14:40.51 | seather | is there any easy-ish way to log call quality for calls made over SIP? |
| 14:41.15 | *** join/#asterisk consolas (~consolas@a81-84-246-186.static.cpe.netcabo.pt) |
| 14:41.17 | consolas | hey guys! |
| 14:41.19 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:41.43 | consolas | How can I debug the url that is calling my dialplan? I mean, I want to get exactly the url that is getting sent to the asterisk. |
| 14:44.20 | [TK]D-Fender | consolas, "sip set debug on". |
| 14:44.40 | consolas | Im in the shell now running the asterisk -rvvvvv |
| 14:44.47 | consolas | Oh, just enable debug and that is it? |
| 14:44.47 | consolas | nice |
| 14:46.29 | consolas | wow, too much information now |
| 14:46.30 | consolas | ahah |
| 14:48.20 | MariusKarthaus | Hi, we have an asterisk server with a few grandstreams connected to it. A calling customer (on a dial in number via SIP trunk) is put in a queue. The phones subscribe to the queue using AddQueueMember() in the extensions.conf. Sometimes when a person is picked up on phoneX and then transferred to phoneY, the caller gets stuck in the queue and we are unable to get them back. Any ideas what could be going wrong? |
| 14:48.35 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:51.02 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:53.50 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 14:56.19 | *** join/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 15:01.23 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 15:03.45 | mcargile | I am using Asterisk 1.4.39.2 (waiting on a good window to update) and am seeing this warning constantly in my logs "RTCP Read too short". I have never seen this before and am curious what it means. We have setup two cross connects recently with other offices and I am wondering if something is wrong. |
| 15:07.08 | *** join/#asterisk navaismo (~navaismo@189.144.249.74) |
| 15:17.46 | consolas | Well still strugling here, guys after configuring a service with asterisk that accepts a curl request, how can I, in the asterisk machine, get the complete url that its receiving? |
| 15:18.02 | consolas | I can enable sip debug but it has everything except the url |
| 15:19.53 | kaldemar | consolas: you'll see it on the INVITE line in sip debug. |
| 15:20.07 | consolas | INVITE line? Ok, let me try that then |
| 15:20.47 | [TK]D-Fender | consolas, Asterisk doesn't receive "curl" requests. Asterisk is not a web server. |
| 15:21.05 | consolas | Yes that much I understand but the cURL is executed |
| 15:21.18 | consolas | and asterisk is receiving the variables, I can see the number to call variable for isntance |
| 15:21.22 | kaldemar | i thought is was a typo. |
| 15:21.25 | [TK]D-Fender | consolas, Is it? We have no idea what you're doing. You aren't showing us |
| 15:21.26 | [TK]D-Fender | ~pb |
| 15:21.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:21.27 | [TK]D-Fender | ^ |
| 15:22.06 | consolas | Fender: http://i.chzbgr.com/completestore/2011/11/10/55228ff4-7e60-499d-95d5-df5fb433b963.jpg |
| 15:22.09 | consolas | indeed! ahah |
| 15:22.57 | consolas | I just inherit this and I'm making it work, I wish I had more time, that is why I came here just to check out this |
| 15:22.59 | consolas | thanks guys |
| 15:28.10 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 15:39.35 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 15:42.21 | *** join/#asterisk _omer (~omer@58-27-165-190.wateen.net) |
| 15:42.24 | _omer | hello |
| 15:43.16 | apurvtwr | hi jpsharp you there? |
| 15:43.47 | *** join/#asterisk alsuren (~dlaban@80.169.133.251) |
| 15:44.05 | _omer | when I type "asterisk" at shell prompt, it runs asterisk ... I want to see what commands it runs in background .... or can I add more commands before running asterisk...... I think it's linux question. |
| 15:45.11 | [TK]D-Fender | you normally run * as a daemon, not live like that |
| 15:45.25 | [TK]D-Fender | safe_asterisk & |
| 15:45.33 | [TK]D-Fender | and then conenct to it once it's running |
| 15:45.57 | _omer | ok ... |
| 15:46.00 | [TK]D-Fender | you question about "commands it runs in the background" makes no sense. please rephrase |
| 15:47.54 | _omer | actually ... I have installed asterisk in a folder like "/abc/xyz/" .... so I run asterisk like "/abc/xyz/sbin/asterisk" then "/abc/xyz/sbin/asterisk -vvvr" |
| 15:48.02 | _omer | it is working fine. |
| 15:48.20 | _omer | but when I also type "asterisk" ... it also run old installation I think. |
| 15:48.33 | _omer | so I want to run /abc/xyz/sbin/asterisk by typing only "asterisk" |
| 15:48.37 | _omer | you know what I mean ? |
| 15:48.43 | Qwell | so update your PATH |
| 15:48.57 | _omer | how and where I can find that path ? |
| 15:49.10 | Qwell | This is a very basic shell thing. |
| 15:49.42 | _omer | could not google about it because I dont know what to find :) |
| 15:50.29 | Qwell | http://linux.about.com/od/linux101/l/blnewbie3_1_4.htm |
| 15:50.40 | _omer | thanks... let me check. |
| 15:51.29 | [TK]D-Fender | You shouldn't have multiple *'s on your system.... |
| 15:53.03 | _omer | I think I did not uninstall old installation properly. that is why "asterisk" commands runs asterisk differently |
| 15:53.48 | _omer | by the way, I have a question, If I have installed multiple asterisk in multiple folders, so Can I run different instances from their specific folders? |
| 15:53.51 | _omer | in the same machine? |
| 15:54.43 | Qwell | man asterisk |
| 15:54.51 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:54.53 | Qwell | see -C |
| 15:55.29 | _omer | aww... looks good |
| 15:55.56 | *** part/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 16:07.29 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 16:19.59 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:24.44 | consolas | hey guys: exten => s,n,Dial(SIP/${destino}@${prov},40) |
| 16:24.56 | consolas | I can get the destino variable and the prov variable I know for a fact that is getting sent. |
| 16:25.09 | consolas | but asterisk is not getting it |
| 16:27.51 | [TK]D-Fender | consolas, those are channel variables. If they are filled in with anything it's because YOU filled them in earlier in your dialplan. |
| 16:28.04 | *** join/#asterisk CrazyTux[m] (~Brandon@107-199-131-100.lightspeed.irvnca.sbcglobal.net) |
| 16:28.08 | [TK]D-Fender | consolas, And youa re not showing use the call. |
| 16:28.13 | consolas | yeah, the destino is coming from the curl |
| 16:28.18 | consolas | so is the prob |
| 16:28.19 | consolas | prov |
| 16:28.49 | [TK]D-Fender | consolas, PASTEBIN <=- Show us. |
| 16:28.51 | [TK]D-Fender | ~pb |
| 16:28.52 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:28.55 | consolas | roger that |
| 16:34.08 | consolas | http://pastebin.com/8tijSYju |
| 16:34.29 | consolas | the destiny gets the value correctly, prov doesn't |
| 16:34.59 | consolas | this is supposed to call a number and when I pick up, it calls the other number. IT works if I force a provider |
| 16:36.01 | Qwell | You didn't show where you're setting those... |
| 16:36.43 | Katty | hi Qwell |
| 16:36.50 | [TK]D-Fender | consolas, And I said show us the CALL. Not just your dialplan code. I want to see it EXECUTE. |
| 16:37.00 | Qwell | Katty: ohai |
| 16:37.05 | Katty | how's the ladyfriend. |
| 16:37.17 | Qwell | she be well |
| 16:37.19 | Katty | :> |
| 16:37.29 | Katty | and you? doing well too? |
| 16:37.34 | consolas | Well I inherit this, I though the curl to the asterisk would set this variables, this is pretty much what I've learned so far |
| 16:37.34 | Qwell | yar |
| 16:37.38 | Katty | horay! |
| 16:37.41 | consolas | oh the call! |
| 16:37.44 | consolas | that I can do |
| 16:37.55 | kaldemar | consolas: what do you mean by curl? |
| 16:37.57 | consolas | Ill pastebin both: the not working, and the working |
| 16:38.09 | consolas | Im sending the numbers through a php script with curl |
| 16:41.32 | [TK]D-Fender | consolas, We don't see this "curl" we don't see yuor call. Yuo are talking about things and not showing us. |
| 16:41.51 | [TK]D-Fender | consolas, Nothing in that dialplan you pasted tells us anything about PHP, curl, etc. |
| 16:41.57 | consolas | found it |
| 16:41.57 | consolas | https://www.asterisk.org/astdocs/node211.html |
| 16:42.02 | consolas | this is what I use |
| 16:42.12 | [TK]D-Fender | consolas, Don't show us a story. Show us YOUR call. |
| 16:42.27 | consolas | yes, pastebin now the "working call" |
| 16:44.57 | consolas | http://pastebin.com/JcSqawf1 this is the call that works, as soon as I change in the dialplan the provider siptel to the variable sent by the url, it stops working (and the url is sending the variable, that is for sure) |
| 16:46.24 | *** join/#asterisk joel (~chatzilla@unaffiliated/joel) |
| 16:46.27 | *** join/#asterisk qbx (0c19f064@gateway/web/freenode/ip.12.25.240.100) |
| 16:46.29 | n3hxs | Katty |
| 16:46.38 | n3hxs | morning.... er Afternoon. |
| 16:47.23 | [TK]D-Fender | consolas, Why are we not seeing this "URL" you are calling? Why are we not seeing that failure that is supposedly generates? |
| 16:47.25 | Katty | n3hxs: howdy (= |
| 16:47.50 | n3hxs | Did you behave while I was on vacation in the Caribbean? |
| 16:49.11 | consolas | http://pastebin.com/n9VLAcNc |
| 16:49.13 | consolas | this is the url |
| 16:49.29 | consolas | and the dialplan is what I showed you before |
| 16:49.50 | consolas | and now its working, if I change the @siptel on the dial plan to @${MY_VAR}, it doesn't work anymore |
| 16:49.51 | kaldemar | prov=siptel means nothing. |
| 16:49.54 | Qwell | You're missing something there... |
| 16:50.02 | consolas | oh my bad |
| 16:50.03 | kaldemar | Variable=destiny=921231212&prov=siptel <-- see the difference? |
| 16:50.08 | consolas | I did't put the VARIABLE |
| 16:50.08 | consolas | yes |
| 16:50.10 | consolas | absolutely |
| 16:50.37 | kaldemar | FYI, that's called originating a call via AMI. |
| 16:51.40 | [TK]D-Fender | consolas, http://IP:8088/asterisk/mxml?action=originate&channel=Local/55911231212@dial-out&context=macro-answered&exten=s&priority=1&Variable=destiny=921231212&prov=siptel <--- this doesn't SAY MY_VAR. So of course if you change the name in the dialplan IT WON'T MATCH |
| 16:52.04 | consolas | fender: the my var was just to explain this better |
| 16:52.08 | consolas | I know the var name must match |
| 16:52.14 | consolas | but yes, the problem was msising the VARIABLE= |
| 16:52.19 | consolas | thanks a lot guys! |
| 16:52.21 | [TK]D-Fender | show us actual failing code & URL. |
| 16:52.31 | consolas | for you this is a no brainer but like I said, I'm fairly new to this |
| 16:52.36 | consolas | Fender: it works, thanks mate |
| 16:52.41 | consolas | Appreciate the help guys, really |
| 16:52.42 | consolas | do |
| 16:52.53 | consolas | Gonna do some more tests, again, thank you very much for your concern and attention |
| 16:52.57 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:53.32 | [TK]D-Fender | consolas, Don't waste people's time showing fake things, and different samples that do work. Show us the ones that don't. |
| 16:54.02 | consolas | hey mate, I wans't showing fake things, easy there |
| 16:54.08 | consolas | I was showing what I had |
| 16:54.11 | *** join/#asterisk xJustTheTip (~xJustTheT@50-192-155-18-static.hfc.comcastbusiness.net) |
| 16:54.25 | consolas | I just didnt paste everything I have here or else you guys had to go through a lot of things that didnt matter |
| 16:54.26 | [TK]D-Fender | You said X doesn't work. You weren't showing us X |
| 16:54.39 | consolas | I showed dialplan, url and the call itself |
| 16:54.46 | consolas | right? |
| 16:54.56 | [TK]D-Fender | and now its working, if I change the @siptel on the dial plan to @${MY_VAR}, it doesn't work anymore <--- I did not see full dialplan with this |
| 16:55.00 | xJustTheTip | hi all |
| 16:55.04 | [TK]D-Fender | or the call to it |
| 16:55.21 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:55.47 | consolas | the my_var I had before with the prov, and I showed the url with prov. But I thought saing MY_VAR would be easier to udnerstand, my bad then |
| 16:56.17 | [TK]D-Fender | FAKE var names where we can't tell what you're talking about. yeah... no reason we should be confused there.... |
| 16:57.00 | xJustTheTip | after installing AsteriskNOW-2.0.2-x86_64-DVD on ESXi, the default admin/admin isn't working... googled all over and tried several other default un/pw's no luck... any help would be super. |
| 16:57.06 | [TK]D-Fender | moves on more productive matters. |
| 16:57.19 | Qwell | xJustTheTip: isn't working, where? |
| 16:57.43 | xJustTheTip | freepbx web login |
| 16:58.05 | Qwell | You would definitely be the first to suggest that such a problem might exist. |
| 16:58.18 | Qwell | That's something I would have heard about. |
| 16:59.13 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:59.14 | consolas | <kaldemar> FYI, that's called originating a call via AMI. |
| 16:59.18 | consolas | Thanks, noted!:) |
| 17:07.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:11.23 | consolas | do you guys have like small IVR or custom made pbx at home, using asterisk? |
| 17:11.29 | consolas | this seems very powerful |
| 17:11.36 | Qwell | any and all of the above. |
| 17:13.36 | consolas | haha |
| 17:13.55 | consolas | what do you have at home with this? like a voip phone in every room for isntance? |
| 17:14.12 | Qwell | sure |
| 17:14.47 | consolas | nice! |
| 17:16.28 | jpsharp | I have an IP phone on my desk, voip software on my iphones, and an ATA with a cordless phone on it. |
| 17:17.44 | *** join/#asterisk Vince-0 (~Vince-0@41-135-7-154.dsl.mweb.co.za) |
| 17:19.24 | consolas | Well I have a voip app as well, mobilevoip |
| 17:19.24 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 17:19.28 | consolas | but I had nothing to do with it |
| 17:19.36 | consolas | still it would be fun to have a custom ivr at home |
| 17:19.45 | consolas | "to reach room, press 1" to reach kitchen, press 2" |
| 17:19.46 | consolas | ahah |
| 17:20.21 | jpsharp | I used the heck out of it to deflect bill collectors. And then I used the recording ability to hold those same bill collectors to the fire. |
| 17:21.13 | consolas | ahaha awesome |
| 17:21.33 | consolas | I will get an old computer and start doing some stuff with this |
| 17:21.45 | consolas | Voipbuster is the most common +provider, I assume |
| 17:24.32 | consolas | and what do you think about the google voip number? I think it was a service where you would add all your phone numbers and just give to people the "google" number |
| 17:24.45 | consolas | seemed a very easy to implement custom voip solution |
| 17:24.48 | consolas | but pretty awesome |
| 17:28.22 | [TK]D-Fender | Google voice likes to break semi-regularly and we get far more questions than answers here. |
| 17:28.33 | [TK]D-Fender | Standard ITSPs are much easier to support |
| 17:34.30 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 17:38.52 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 17:42.45 | consolas | cool! |
| 17:42.53 | consolas | I think Im going to try this |
| 17:44.16 | jpsharp | I use google voice for voice and Gafachi for fax support. |
| 17:45.23 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-202.nwrknj.fios.verizon.net) |
| 17:45.29 | Dovid | Asterisk-SCF project canned? |
| 17:45.59 | navaismo | uh? |
| 17:46.09 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 17:46.11 | eXcAliBuR | hey |
| 17:46.25 | eXcAliBuR | does anyone here know what i'm doing? because I don't >:( |
| 17:47.38 | qbx | Well that's when people know best. |
| 17:48.34 | navaismo | depends on what are you doing or trying to do |
| 17:53.16 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-ryykyicvraduvyog) |
| 17:54.08 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 17:54.40 | Dovid | i was just told that asterisk-scf is now dead |
| 17:54.58 | *** join/#asterisk hiddenman (~akornilov@altlinux/developer/hiddenman) |
| 17:57.01 | hiddenman | hi all. guys, we've just got a stange problem with our Asterisk. it has been broken somehow. changing password doesn't help. we have dumped all the traffic and signalization. as far as i can see, that bad guy sends: REGISTER, gets 401 Unauth, then sends REGISTER again, get 403 Bad auth. after that he sends INVITE without any DIGEST and so on. and asterisk DOES call |
| 17:57.04 | hiddenman | how can this be? |
| 17:57.32 | hiddenman | we have insecure=invite,port for this account. could it be the problem? not sure |
| 17:59.00 | Greenlight | Howdy folks. Got a problem that I was hoping some of you would be able to advise on. We've a hosted dedicated server running asterisk doing circa 100k calls a day. For the last day or so users have been getting audio quality issues where by the remote party (connected via SIP wholesale provider) is getting intermittant drops in audio. We've isolated the issue to somehwere between our server |
| 17:59.00 | Greenlight | and our wholesale provider, but can't pin it down anymore. Our dedicated provider is acknowldging congestion on london routes where they peer with linx but pings all seem okay (10ms - 100ms). In an effort to sort things I've provisioned a server in Amsterdam and want to route our traffic vian this box. I was planning on using OpenVPN but this looks like it might be tricky as Asterisk will |
| 17:59.00 | Greenlight | have two external ips. the other option is to install Asterisk on the box in Amsterdam and use it that way - which method do you think would suit best? (Sorry for wall of text!) |
| 17:59.19 | hiddenman | he does send digest in the second REGISTER. but he gets 403 Bad auth, how can he send INVITE?! |
| 18:01.40 | mjordan | hiddenman: from sip.conf - ;insecure=invite ; Do not require authentication of incoming INVITEs |
| 18:02.17 | hiddenman | mjordan: so he can call without any problems?! that's our mistake? |
| 18:04.44 | Greenlight | Could Asterisk be configured to work with a public ip one it's NIC and another public IP via a VPN tunnel (NATed)? |
| 18:04.51 | mjordan | hiddenman: no idea. But you asked how Asterisk would allow an INVITE from a peer without requiring authentication. And if you tell Asterisk to allow INVITE requests from that peer without authenticating - which is what "insecure=invite" is going to do (hence "insecure") - then yes, they can send INVITE requests and Asterisk won't authenticate them. |
| 18:05.22 | hiddenman | mjordan: thank you! it seems you are right :( |
| 18:09.33 | hiddenman | mjordan: i'm not so familiar with SIP protocol. so as far as i understand, user doesn't need any registrations in case he wants to call? just send a correct INVITE and that's all, right? |
| 18:10.36 | Qwell | hiddenman: correct, registration is for telling the server where you can be called. |
| 18:10.56 | hiddenman | i mean he just have to know the correct username, that can call, sends an invite from this username@myasterisk with particular number and that's all. am i right? |
| 18:11.09 | Qwell | with insecure=invite, yes, that is correct. |
| 18:11.22 | hiddenman | that's a pity. our bad. thank you |
| 18:14.02 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 18:25.27 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 18:28.14 | eXcAliBuR | can i have line 1 on my phone connect to analog line 1? |
| 18:31.23 | eXcAliBuR | like a direct channel, so what I dial is being sent live on analog 1, and not waiting for the dial plan to re-route it |
| 18:31.24 | eXcAliBuR | ? |
| 18:32.30 | [TK]D-Fender | eXcAliBuR, Your phone would have to dial something to get * to reach |
| 18:33.08 | [TK]D-Fender | eXcAliBuR, So no.. it wouldn't be a "line" key. Yuo could make a SPEED DIAL dial something that your dialplan takes and just dials an empty number with and you'll do the rest yourself for. |
| 18:33.40 | eXcAliBuR | i just want to be able to see when one of my 2 lines are in use on the phone |
| 18:33.52 | eXcAliBuR | so if someone trys to dial they will know before dialing |
| 18:34.02 | [TK]D-Fender | Seeing if a line is in use has nothing to do with getting "striaght tone without dialing". |
| 18:34.31 | navaismo | ~blf |
| 18:34.32 | infobot | well, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 18:35.08 | [TK]D-Fender | 19:10.28eXcAliBuRI have a digium d50 phone, and 2 analog lines, I'd like to know if it's capable to showing when a line is in use? |
| 18:35.08 | [TK]D-Fender | 19:12.04[TK]D-FendereXcAliBuR, Set up a hint in your dialplan. Watch it. |
| 18:35.16 | [TK]D-Fender | eXcAliBuR, I told you all of this YESTERDAY |
| 18:35.25 | eXcAliBuR | yes but thats not the same question |
| 18:35.27 | [TK]D-Fender | And with any luck probably not the only time. |
| 18:35.43 | [TK]D-Fender | <[TK]D-Fender> 19:10.28 eXcAliBuR I have a digium d50 phone, and 2 analog lines, I'd like to know if it's capable to showing when a line is in use? <- Yes, the SAME THING |
| 18:35.50 | [TK]D-Fender | <eXcAliBuR> i just want to be able to see when one of my 2 lines are in use on the phone <- SAME |
| 18:36.04 | [TK]D-Fender | Show. When. Line. In. Use. |
| 18:36.19 | eXcAliBuR | but one is analog and one is sip |
| 18:36.20 | eXcAliBuR | :/ |
| 18:36.25 | [TK]D-Fender | same thing. |
| 18:36.38 | [TK]D-Fender | dialplan HINT to show pesence state for the device to watch |
| 18:36.44 | eXcAliBuR | oh |
| 18:37.08 | [TK]D-Fender | DAHDI channel, SIP Peer. same thing |
| 18:38.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:40.33 | eXcAliBuR | EEERRR i'm trying to think and nothing is happening |
| 18:40.34 | eXcAliBuR | >:( |
| 18:41.12 | Qwell | ~book |
| 18:41.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:41.16 | Qwell | Read that. |
| 18:42.47 | joel | eXcAliBuR: you're looking for key system functionality with a pbx. |
| 18:43.11 | eXcAliBuR | I have the book, i'm about to beat myself over the head with it |
| 18:43.20 | joel | eXcAliBuR: your key term is "shared line appearance" |
| 18:43.21 | eXcAliBuR | looks big enough to do some damage |
| 18:44.12 | eXcAliBuR | page 318 |
| 18:44.16 | eXcAliBuR | :/ |
| 18:44.25 | eXcAliBuR | shared line appearrance |
| 18:44.29 | eXcAliBuR | who knew |
| 18:44.33 | joel | me. |
| 18:46.26 | *** join/#asterisk hiddenman (~akornilov@altlinux/developer/hiddenman) |
| 18:47.57 | eXcAliBuR | oh i could kiss you |
| 18:48.04 | eXcAliBuR | it's telling me exactly what i want to hear |
| 18:48.44 | eXcAliBuR | i can't wait to get home, i need pills |
| 18:49.52 | jpsharp | Do you have enough to share? |
| 18:50.19 | eXcAliBuR | NO! |
| 18:50.23 | eXcAliBuR | they are all mine |
| 18:50.24 | eXcAliBuR | ! |
| 18:54.59 | eXcAliBuR | thank you |
| 18:55.09 | *** part/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 18:56.12 | *** part/#asterisk hiddenman (~akornilov@altlinux/developer/hiddenman) |
| 18:57.04 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:59.24 | [TK]D-Fender | And no, "Shared Line Appearance" is NOT it. You have just caused a new level of nightmare up his return |
| 19:00.07 | [TK]D-Fender | Because that relies on his having a speed dial w/ presences set up... and 1 tons of OTHER shit that really isn't required or manageable. |
| 19:00.24 | navaismo | O_o |
| 19:00.28 | navaismo | bad mood |
| 19:00.32 | [TK]D-Fender | He can't wrap his head around a 1-step thing.. this requires that + 15 more |
| 19:01.13 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:01.15 | [sr] | hwllou |
| 19:01.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:10.42 | *** join/#asterisk [netman] (~netman@67.red-83-36-39.dynamicip.rima-tde.net) |
| 19:11.09 | xJustTheTip | I reported a problem logging into web interface of freepbx after clean install of AsteriskNOW-2.0.2-x86_64-DVD on ESXi; Seems like a chrome issue. Firefox works fine. |
| 19:13.40 | carrar | that sucks |
| 19:18.41 | ChannelZ | tell FreePBX |
| 19:19.45 | *** part/#asterisk xJustTheTip (~xJustTheT@50-192-155-18-static.hfc.comcastbusiness.net) |
| 19:22.03 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 19:23.44 | joel | he parted before I could demand my kiss. |
| 19:24.20 | joel | [TK]D-Fender: you ever thought about taking a break from #asterisk? It obviously causes you great amounts of pain. |
| 19:25.54 | [TK]D-Fender | Life causes pain. This is just an on-set location :) |
| 19:26.32 | joel | That's a shame, I'm sorry to hear it. My life actually brings me joy. |
| 19:29.48 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 19:30.45 | ChkDigit | A joyful life sure can kill a thread. Perhaps everyone went outside to play? |
| 19:34.11 | jpsharp | I was outside weedeating, does that count? |
| 19:45.46 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 19:46.08 | drfreeze | Anyone deploy a new Polycom 650 phone? |
| 19:46.43 | drfreeze | I have a new set of phones and they don't appear to be checking the provisioning server |
| 19:47.24 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 19:48.47 | drfreeze | Saw one online statement that said they they require option 66 on DHCP |
| 19:49.08 | drfreeze | right now, I'm trying to set the via the html gui |
| 19:50.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:50.49 | _Corey_ | drfreeze: What does your scope option look like? |
| 19:53.51 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:56.14 | drfreeze | _Corey_: what are you referring to with 'scope' option? |
| 19:56.25 | _Corey_ | DHCP scope option 66 |
| 19:56.36 | drfreeze | Right now, I don't have control of DHCP. It is in hands of windows admin |
| 19:56.42 | _Corey_ | ah |
| 19:56.45 | drfreeze | so, it is not being set |
| 19:56.56 | drfreeze | I am setting that option on the phone |
| 19:57.34 | jpsharp | Ive seen my polycom ignore what the GUI says because it wasn't getting that option from the DHCP server |
| 19:57.43 | _Corey_ | I'm not paying full attention here, it sounded like you were trying to figure out why they weren't provisioning |
| 19:57.54 | drfreeze | jpsharp: new phone? Like within the last 4 months? |
| 19:58.10 | jpsharp | No, my IP-300 with new firmware. |
| 19:58.38 | drfreeze | _Corey_: I have a 550 that provisioned fine by just setting the Outbound Proxy Address and the Server 1 address from the web gui |
| 19:59.03 | drfreeze | but this one doesn't appear to be contacting the server with that same info |
| 19:59.07 | _Corey_ | drfreeze: Well, it sounds like it *authenticated* fine... provisioning involves it downloading a config file |
| 20:00.13 | drfreeze | right |
| 20:00.37 | _Corey_ | so, as far as manually configuring them... just sounds like you need to troubleshoot |
| 20:04.48 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 20:06.34 | drfreeze | yeah. doing remote troubleshooting |
| 20:09.43 | [TK]D-Fender | No, they don't require Op 66. |
| 20:12.55 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 20:16.39 | drfreeze | Must be a bug or a change to the web gui |
| 20:16.59 | drfreeze | phone works fine when configured via the menu |
| 20:25.24 | *** join/#asterisk eureka^ (~F@173-15-226-185-static.hfc.comcastbusiness.net) |
| 20:26.15 | eureka^ | hi. i'm having a problem with my phones not ringing when a second call is coming in, if the first call is on hold |
| 20:26.34 | eureka^ | asterisk 10.7.0, phone is a grandstream gxp2100 |
| 20:30.01 | WIMPy | What does the manual say? |
| 20:30.44 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 20:30.45 | eureka^ | phone or asterisk? |
| 20:30.56 | WIMPy | phone |
| 20:31.10 | eureka^ | i don't see anything relevant in the phone manual, but it's pretty bare :/ |
| 20:34.49 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 20:36.04 | nickgaw | Hi, With asterisk distributions like pbx in a flash on the raspberrypi can you use wireless sip phones so that more then one person at the same house that has an extention can receive and make a call? |
| 20:37.17 | WIMPy | If you have something that does wifi (wich you probably don't really want), yes. |
| 20:37.20 | *** join/#asterisk davemha (~IceChat77@66.135.230.101) |
| 20:37.34 | WIMPy | I didn;t understand the rest of the question. |
| 20:38.58 | nickgaw | what I mean is lets say you have three people in a house and each person has a sip extention and you have an IVR that when the caller calls they press the number of the person they wish to talk to then can all three people be talking to a different person at the same time or make an out going call at the same time? |
| 20:40.16 | navaismo | 3 simultaneous calls without transcoding on the Pi yes, |
| 20:40.22 | WIMPy | Yes, you can make a menu to select a phone to call. |
| 20:40.34 | navaismo | and without recording |
| 20:41.16 | nickgaw | for a small home do you think the raspberrypi is an ok system to use for asterisk? |
| 20:41.32 | navaismo | Yesterday I was testng it with SIPp and with more than 7 calls the audio get choppy |
| 20:41.37 | WIMPy | The usual plasticrouters don;t have more beef, and seem to easily manage transcoding. |
| 20:41.50 | navaismo | cpu goes from 0 to 7 |
| 20:42.06 | nickgaw | can you overclock the device? |
| 20:42.11 | pabelanger | why even bother with the raspberrypi |
| 20:42.14 | navaismo | a little yes |
| 20:42.29 | nickgaw | what does it require? |
| 20:42.35 | pabelanger | just setup a normal PC and install asterisk |
| 20:42.37 | navaismo | consume very tinny power, silence |
| 20:42.38 | pabelanger | problem solved |
| 20:43.09 | navaismo | and for 3 extensions is a very good choice instead a 300W old pc running all the day |
| 20:43.13 | pabelanger | Right, so from what I gather, you can do 3 calls |
| 20:43.18 | pabelanger | if you need more, get better hardware |
| 20:43.31 | pabelanger | also |
| 20:43.32 | navaismo | exactly |
| 20:43.42 | nickgaw | I like the size of the raspberrypi I was just wondering what would be required to overclock the device should you wish to do so? |
| 20:43.48 | pabelanger | do you really need to install PBXiaf on it? |
| 20:43.59 | pabelanger | again, you are using resources that you cannot spare |
| 20:44.27 | navaismo | Yes you can run more call if you avoid the MYSQL & PHP process more free memory |
| 20:44.48 | navaismo | but so far the 3 elements run nicely in the Pi |
| 20:44.55 | pabelanger | install emdebian, setup asterisk and away you go |
| 20:45.02 | nickgaw | I thought pbx in a flash requires mysql to work? |
| 20:45.09 | navaismo | also the monast (fop) run very nice |
| 20:46.00 | pabelanger | nickgaw: that's what I am saying, do you really want to be running a database server on such a small device? |
| 20:46.31 | nickgaw | probably not. is there a better option on the raspberrypi? |
| 20:46.37 | navaismo | and here is how you overclock the device http://fusionstrike.com/2012/overclock-raspberry-pi-cpu-850mhz-ram-500mhz |
| 20:47.02 | navaismo | For that price i dont think so, and i highlight THAT PRICE |
| 20:47.25 | navaismo | you can buy a cubieboard and alix board and setup as well |
| 20:47.37 | WIMPy | Some plastic router? They can be even cheaper. |
| 20:47.54 | navaismo | cheaper than 35USD? |
| 20:48.08 | WIMPy | And they might even have telephony hinterfaces included. |
| 20:48.16 | navaismo | well with taxes I buy it from $50USD |
| 20:48.53 | navaismo | but personally I have worked with the AlixD2D board and preffer the Pi |
| 20:49.19 | navaismo | You got an extra GPIO to play with it |
| 20:55.24 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 20:57.11 | Greenlight | Is "Locally Bridging" the same as packet2packet bridging? |
| 21:01.42 | nickgaw | Anyone know what boot loader the rasberrypi uses? |
| 21:02.26 | navaismo | boot loader? |
| 21:02.42 | nickgaw | well it has to boot up somehow? |
| 21:02.51 | navaismo | you mean grub |
| 21:02.59 | navaismo | or lilo" |
| 21:03.02 | nickgaw | yes or syslinux or lilo? |
| 21:03.20 | WIMPy | The one you install? |
| 21:03.23 | navaismo | weel the popular images use grub |
| 21:03.47 | nickgaw | could I always create my own image if the system supports arm7? |
| 21:04.09 | navaismo | yep you could create your custom image for arm |
| 21:04.14 | navaismo | a painful path |
| 21:04.24 | navaismo | google is your friend for that |
| 21:04.36 | nickgaw | have people done it before? |
| 21:04.38 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a0792.pool.mediaWays.net) |
| 21:05.53 | *** join/#asterisk qbx (0c19f064@gateway/web/freenode/ip.12.25.240.100) |
| 21:06.32 | navaismo | yes, there are falvors to install in your Pi |
| 21:07.03 | nickgaw | what are those? |
| 21:07.12 | navaismo | fedora-arm, raspbian, openelec, raspbmc etc etc, |
| 21:07.20 | navaismo | dependes on your needs |
| 21:07.39 | fenrus | <3 raspbian |
| 21:07.47 | nickgaw | could I always create something custom that is not already provided? |
| 21:08.37 | WIMPy | You can just go and program it yourself. Why do you need an OS? |
| 21:08.42 | navaismo | yes you can, you need to document yourself and create your image |
| 21:09.19 | nickgaw | do the people at pbx in a flash have an apt repository for normal x86 debian systems?\ |
| 21:10.08 | navaismo | dont know ,ask them |
| 21:10.27 | navaismo | but there a re an Image for use with asterisk+freepbx 2.10 |
| 21:10.29 | nickgaw | have tried can't get in contact with them |
| 21:11.59 | nickgaw | debian has asterisk but not freepbx in it's repositories any major reason for this? |
| 21:12.25 | navaismo | dont know, ask debian |
| 21:12.33 | navaismo | check this http://www.cl.cam.ac.uk/freshers/raspberrypi/tutorials/os/ |
| 21:12.54 | navaismo | and this http://www.raspberry-asterisk.org/ |
| 21:13.19 | navaismo | ready to dd that image |
| 21:13.28 | navaismo | if you dont want to make yourself |
| 21:14.59 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 21:15.31 | navaismo | sucks http://www.geek.com/articles/chips/supercomputer-built-with-raspberry-pi-and-lego-20120912/ |
| 21:23.29 | nickgaw | is the raspberrypi really the size of a credit card or are they just saying that? |
| 21:24.40 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:25.56 | beardy | Look at the dimensions and compare with a credit card. |
| 21:25.58 | beardy | ... |
| 21:26.24 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 21:27.37 | navaismo | is this channel of raspi? No, --> #raspberrypi |
| 21:28.34 | nickgaw | reading on the asterisk web sites it looks like they also sell fully built asterisk systems do you think those are worth the money? |
| 21:29.42 | jpsharp | THey're worth it if you need the support. |
| 21:30.02 | nickgaw | is that all you are really buying is the system and their support? |
| 21:30.45 | navaismo | you buy a stable hardware and software appliance and the excellent support |
| 21:31.52 | nickgaw | when there is so much free support out there for asterisk? that is probably meant for large companies correct? |
| 21:32.12 | jpsharp | There's free support, but it isn't guaranteed. |
| 21:32.44 | nickgaw | true but for a home user paying for support is it worth it can you just buy a support plan? |
| 21:32.57 | *** join/#asterisk MiserySoft (4f822780@gateway/web/freenode/ip.79.130.39.128) |
| 21:34.25 | jpsharp | You can pay a consultant or the like. |
| 21:34.45 | navaismo | feels like a job interview |
| 21:34.46 | nickgaw | what is the cheapest you have heard of? |
| 21:35.04 | jpsharp | $30/hour on up to $250/hour. |
| 21:35.30 | navaismo | Is the cheaper slut, damm it! |
| 21:35.39 | nickgaw | they are probably remote or would they need to come out? |
| 21:35.48 | jpsharp | Most definitely remote. |
| 21:35.55 | jpsharp | unless you want to pay the $250. |
| 21:36.22 | nickgaw | do I just search on google for them? |
| 21:36.49 | jpsharp | Google or ask here. |
| 21:37.08 | nickgaw | what web site does the $30 per hour? |
| 21:37.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:37.18 | jpsharp | I do consulting as I believe navismo does as well. |
| 21:37.58 | jpsharp | or navaismo, rather. |
| 21:38.45 | nickgaw | jpsharp do you have a web site or how would I get in touch with you if I needed this service? |
| 21:39.26 | nickgaw | naviasmo are you still here and do you do asterisk work? |
| 21:40.15 | navaismo | yes Im here, And i do asterisk work too. I guess everybody in the channel does |
| 21:40.55 | nickgaw | do you have a web site or how could I get in touch with you if I needed services done? |
| 21:40.56 | *** join/#asterisk aossama (~ao@41.233.166.236) |
| 21:42.50 | navaismo | Yes I will PM you |
| 21:42.59 | nickgaw | ok |
| 21:43.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:44.09 | jpsharp | I suppose I should put together a basic website for it. |
| 21:48.01 | carrar | higher a consultant to put together a website for you |
| 21:48.11 | carrar | hire even |
| 21:48.13 | carrar | heh |
| 21:49.24 | jpsharp | That's what I have a wife fore. |
| 21:49.26 | jpsharp | fore |
| 21:49.29 | jpsharp | for, dammit. |
| 21:52.48 | [TK]D-Fender | "fore". What you're supposed to yell to warn bystanders of an incoming ball |
| 21:53.13 | [TK]D-Fender | Instead I jsut kept muttering to myself "Man there aint no way that's gonna hit 'em" |
| 21:53.14 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 21:54.05 | nickgaw | for some reason my irc client quit on me |
| 22:07.45 | Greenlight | Do the batch settings in cdr.conf also apply to cdr_adaptive_odbc? |
| 22:24.17 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-wzcuslvikjyanvnl) |
| 22:27.35 | *** join/#asterisk zam (~gerardo.z@unaffiliated/danyel) |
| 22:30.41 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 22:35.37 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 22:39.10 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 22:59.34 | *** join/#asterisk YoMomma_ (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 23:10.35 | paulc | If a server has multiple CPUs in it, does Asterisk make use of more than one? |
| 23:11.23 | WIMPy | Yes. |
| 23:11.33 | WIMPy | But I don;t know how efficient it is. |
| 23:12.14 | paulc | doffs hat to WIMPy |
| 23:12.20 | WIMPy | OTOH hardware drivers do not or even run better on only one CPU. :-( |
| 23:13.07 | paulc | Experimenting with something at work, gonna get a colo'd server to throw some traffic at, and trying to figure out what kind of spec we need.. *puts finger to the wind* |
| 23:13.44 | WIMPy | I'ts not the traffic. |
| 23:13.51 | paulc | all SIP, no hardware.. so no harm in dual processor if the price is decent.. versus a beefy single processor perhaps.. I dunnooooo |
| 23:14.42 | WIMPy | Where do you get a single CPU system? Oter than virtual. |
| 23:15.56 | paulc | Single chip, multi-core? I was looking at 100tb.com |
| 23:16.05 | *** join/#asterisk cyborg-one (~cyborg-on@130-0-34-78.broadband.tenet.odessa.ua) |
| 23:27.12 | raden | how do I drop a phone number to voice mail every time |
| 23:28.01 | WIMPy | You mean always? |
| 23:29.09 | raden | yup |
| 23:29.12 | raden | caller id routing |
| 23:29.56 | WIMPy | You can use exten => exten/CallerID,prio,app |
| 23:30.11 | raden | ? |
| 23:30.18 | WIMPy | Or eny version of [exec|goto]if. |
| 23:30.51 | WIMPy | Extensions can not only match the called party number but optionally the calling party number as well. |
| 23:36.40 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
