| 00:01.46 | Captain_Proton | ok let me take another look |
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| 00:28.22 | mick_laptop | anyone know if you can repurpose old vonage phones as general purpose sip devices? |
| 00:35.32 | p3nguin | Only if they are or can be unlocked. |
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| 01:19.52 | mick_laptop | meh doesn't look like it |
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| 01:43.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:46.07 | Captain_Proton | p3nguin, thanks for your help I figure it out. wow that kicked my ass |
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| 02:53.12 | raden | how can i debug a call |
| 02:54.42 | p3nguin | Which channel technology? |
| 02:56.00 | raden | asterisk SIP |
| 02:56.04 | raden | i got it |
| 02:56.06 | *** join/#asterisk DACRepair (PJirc@d27-96-73-6.nap.wideopenwest.com) |
| 02:56.42 | DACRepair | Hey so Im having some issues getting outbound calls working with an sip trunk |
| 02:57.01 | DACRepair | asterisk is kinda new to me so i have no clue where to begin |
| 02:59.17 | raden | is there a reason asterisk needs to register with vitelity like every 5 seconds ? |
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| 03:41.23 | p3nguin | dacrepair: SIP doesn't trunk. |
| 03:43.00 | p3nguin | raden: I think every five seconds would be a bit too often, but every 120 might be reasonable. |
| 03:43.46 | raden | p3nguin, where do you set the inverval ? |
| 03:44.02 | raden | has about had it with phones today :( |
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| 03:45.39 | p3nguin | I'd guess the registertimeout setting in sip.conf could have an effect on it. |
| 03:45.53 | p3nguin | Wait, maybe it doesn't. |
| 03:46.11 | p3nguin | That value probably only makes it try when it hasn't registered successfully. |
| 03:46.44 | DACRepair | im using freepbx to configure asterisk atm |
| 03:46.49 | p3nguin | ~freepbx |
| 03:46.49 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:47.00 | DACRepair | ah |
| 03:47.02 | DACRepair | lol |
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| 03:48.26 | kaldemar | raden: either you have set it in the register statement or the expiry time comes from vitality. |
| 03:50.07 | kaldemar | raden: are they really REGISTER messages you see and are the registrations successful? |
| 03:50.27 | raden | the reguistration is successful |
| 03:50.35 | raden | but keeps destroying and rerregistering |
| 03:50.36 | raden | annoying |
| 03:50.48 | [TK]D-Fender | DACRepair: I've already answereed you in #freepbx |
| 03:52.30 | p3nguin | Turn off sip debug if you don't want the noise. |
| 03:53.58 | DACRepair | oh i didnt even notice lol |
| 04:06.33 | raden | i have my ports forwarded on my router to my phone but asterisk is selecting a different address ? |
| 04:07.18 | p3nguin | You don't forward ports to phones. |
| 04:08.32 | raden | always have in the past and never had a issue |
| 04:08.54 | p3nguin | Then you've been doing it wrong in the past. |
| 04:08.59 | raden | lol |
| 04:09.08 | raden | just leave the phones as nat and leave it at that ? |
| 04:09.31 | p3nguin | If they are behind NAT, then yes, they need to be set with nat enabled. |
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| 04:26.53 | raden | p3nguin, my asterisk box is behind a 1 to 1 NAT external IP gets forwarded directly to internal should nat in general be yes or no ? |
| 04:27.36 | p3nguin | If it is using a private address when the outside of the nat has a public address, I would set nat to yes. |
| 04:27.51 | raden | ok |
| 04:28.18 | raden | stupid phone will register to my one asterisk server at one office but not the other :( |
| 04:30.25 | raden | WTF |
| 04:30.52 | raden | i accidentially typed in the wrong IP address in my phone .75 instead of .78 and it showed up on my screen |
| 04:31.11 | raden | that registration failed :( |
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| 05:38.45 | raden | anyone have asterisk behind a static one to one nat ? |
| 05:39.19 | kaldemar | ~sipnat |
| 05:39.19 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
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| 05:54.26 | *** join/#asterisk ayrus (~ayrus@unaffiliated/ayrus) |
| 05:59.33 | ayrus | Hi, |
| 06:00.13 | *** join/#asterisk TheMan (~garry@wsip-24-234-154-105.lv.lv.cox.net) |
| 06:00.55 | ayrus | Hi. I have installed asterisk on my server. Now what i need to call to other countries. I mean should i now purchase DID or SIP minuites. |
| 06:08.42 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:23.48 | ChannelZ | Ok. You should do that. |
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| 06:33.03 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 06:33.40 | chris-NB | hi |
| 06:34.30 | chris-NB | I've a question concerning Call Completion Supplementary Services (ccss) |
| 06:34.56 | chris-NB | When A calls B, B is not available and A places a CallBack |
| 06:36.02 | chris-NB | later, B sees, that A called him and give him a ring, after they talked, A gets a new call, the callback for B |
| 06:37.05 | chris-NB | but, the just talked to each other, so the callback is unnecessary |
| 06:37.19 | chris-NB | anyone knows how to get round this? |
| 06:39.51 | ayrus | ChannelZ, I want to make an outbound call then what should i buy? a sip or did? |
| 06:40.45 | kaldemar | ayrus: those are not mutually exclusive |
| 06:41.42 | ayrus | kaldemar, Ok. so when i buy sip trunk then i will also get DID? |
| 06:42.22 | kaldemar | if that's what you buy. |
| 06:43.53 | kaldemar | buying a mere SIP connection somewhere without any further connectivity (=PSTN) would not make much sense. |
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| 07:55.42 | jww_ | hello. |
| 07:56.42 | jww_ | does somebody use a2billing ? |
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| 08:03.35 | tuxx- | hi guys, is it possible when monitoring a queue call, to get the filename to include the sip account that the queue has dialed? |
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| 08:18.31 | din3sh | hi all |
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| 09:28.53 | schmidts | hi folks |
| 09:32.13 | jww_ | hi. |
| 09:37.02 | tuxx- | <PROTECTED> |
| 09:37.03 | tuxx- | <PROTECTED> |
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| 09:40.45 | Valken | Hi Everybody. |
| 09:41.41 | Valken | As for an IVR, I need the user to enter a 4 number code. How can I tell asterisk that once the user pressed the fourth number, it has to go further? |
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| 09:58.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:02.42 | Sophira | If you're using the Authenticate() application, then there's an argument available to do just that. For example, 'exten => s,1,Authenticate(1428,,4)' will tell Asterisk that the user needs to enter 1428 to continue, and to continue after the fourth number is pressed. |
| 10:04.14 | Sophira | Otherwise, how are you currently doing it? |
| 10:04.54 | Valken | Sophira: thank for the information. However these number must simply be stored in a variable that will later be sent in an agi script. |
| 10:05.13 | Valken | which will connect to a web appliance to perform the check |
| 10:05.30 | Valken | I don't know if 1428 is ok or not. I just know I need 4 numbers. |
| 10:05.55 | Valken | Which might later be 3 or 5 depending on our infrastructure. |
| 10:06.53 | Sophira | Okay. |
| 10:07.09 | *** join/#asterisk polysics (~polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 10:07.54 | Sophira | It looks like Read() also has a maxdigits argument. https://wiki.asterisk.org/wiki/display/AST/Application_Read |
| 10:08.13 | Valken | Thank for the search. |
| 10:08.53 | Sophira | No probs! Let me know if you have any issues with that, or need an example. |
| 10:11.41 | Valken | Perfect. Working like a charm |
| 10:12.14 | Valken | depending where the number read will be use for ivr switching, depending where sent to the agi script for application purpose. |
| 10:13.00 | Valken | How can I use the pound key. I want the user to enter a number and confirm he's finished by pressing the pound key. |
| 10:13.03 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-253-067.sc.res.rr.com) |
| 10:15.09 | schmidts | Valken the '#' is the pound key, you should have it on your phone |
| 10:15.21 | Valken | yes, I know. :D |
| 10:15.30 | Chainsaw | Valken: GET_EXTEN & friends default to obeying # as "end of input", unless you specially configure them. |
| 10:15.42 | Valken | ah ok. Thank. |
| 10:15.57 | schmidts | valken sorry i only read your last sentence and didnt get what you want ;) |
| 10:15.58 | Chainsaw | Valken: The main thing is to ask for more input. So if you expect 4, ask for say... 6. That way it will wait for the # |
| 10:16.04 | Valken | yes indeed. ;) |
| 10:16.20 | schmidts | Chainsaw isnt it Read_Exten not Get_Exten? |
| 10:16.23 | Chainsaw | Valken: (Otherwise, if you ask for 4, and they give you 4, the command will return without waiting) |
| 10:16.30 | Chainsaw | schmidts: Possibly. |
| 10:16.38 | Chainsaw | schmidts: I don't have an Asterisk terminal open at this time. |
| 10:16.55 | schmidts | Chainsaw its ReadExten to be precise ;) |
| 10:16.57 | Valken | ReadExten in fact |
| 10:16.58 | Valken | ;) |
| 10:17.12 | schmidts | and it should need the # at the end: [Description] |
| 10:17.12 | schmidts | Reads a '#' terminated string of digits from the user into the given variable. |
| 10:17.12 | schmidts | Will set READEXTENSTATUS on exit with one of the following statuses: |
| 10:17.15 | Chainsaw | schmidts: That's great, thank you for your input. |
| 10:18.10 | Sophira | ReadExten isn't useful here though. Valken wants it in a variable. |
| 10:18.26 | Chainsaw | transfers the call to Sophira |
| 10:18.26 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f838.pool.mediaWays.net) |
| 10:19.16 | Sophira | Hee. |
| 10:19.45 | *** join/#asterisk freeman_u (~freeman@195.177.208.1) |
| 10:20.17 | Sophira | Oh, I suppose ReadExten does put it into a variable, but only if it's a valid extension. |
| 10:21.00 | schmidts | Sophira and if you use a context with an extension like _X! then the extension will allways be valid ;) |
| 10:21.52 | schmidts | but the normal Read Application with maxdigits set to 0 should also work, cause you have to enter the # key to finish the entered value |
| 10:23.26 | Sophira | Okay. I only started using and learning Asterisk a couple of days ago, and I haven't learned about extension pattern matching yet. |
| 10:23.54 | chris-NB | Hi |
| 10:24.04 | chris-NB | anyone using Call Completion Supplementary Services (ccss) |
| 10:24.24 | schmidts | Sophira i use asterisk since 7 years and still learn something new every day :D |
| 10:24.28 | Valken | Pfff, grrrr. I get carzy with some aspect of conf files. ;) |
| 10:24.35 | chris-NB | anyone tried this? what is implemented in Asterisk 10.3 |
| 10:24.50 | Valken | for instance goto can send me to another context but GotoIf doesn't seems to be able to do so. |
| 10:25.28 | chris-NB | Valken, GotoIf can do that as well |
| 10:25.39 | Valken | any clue? |
| 10:27.14 | Chainsaw | chris-NB: The newest feature I've implemented is connected line updates. |
| 10:27.47 | chris-NB | Valken: GotoIf($["${GOTO}" == "1"]?context,exten,1) |
| 10:28.06 | chris-NB | Chainsaw, what is that? how do you do that? |
| 10:28.22 | Chainsaw | chris-NB: It's caller ID "the other way". |
| 10:28.44 | chris-NB | Chainsaw, can't follow you :/ |
| 10:28.46 | Chainsaw | chris-NB: If I place an outbound call, and the number is in the CRM, it will put the appropriate contact & company name on it. |
| 10:28.47 | Valken | chris-NB: I tough we had to use goto too. :D |
| 10:28.49 | Valken | Thank |
| 10:29.09 | Chainsaw | chris-NB: On an inbound call, the system will send the name of the contact, and their status. (i.e. what office they're in, and whether you're getting a live person or a voicemail box) |
| 10:29.43 | chris-NB | Chainsaw, okay, nice feature. but doesn't have anything to do with ccss, does it? |
| 10:29.57 | chris-NB | Valken, you'r welcome |
| 10:30.03 | Chainsaw | chris-NB: No, but I dislike this deafening silence after someone asks a question. Just making conversation. |
| 10:30.07 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 10:30.21 | chris-NB | Chainsaw, *hehe, thanks :) |
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| 10:31.12 | chris-NB | Chainsaw, I got ccss working, but I've one situation where I miss something to automatic cancel a callback |
| 10:31.27 | chris-NB | When A calls B, B is not available and A places a CallBack |
| 10:31.47 | chris-NB | later, B sees, that A called him and give him a ring, after they talked, A gets a new call, the callback for B |
| 10:31.55 | chris-NB | but, the just talked to each other, so the callback is unnecessary |
| 10:32.06 | chris-NB | and I wan't to cancel it automatic |
| 10:32.40 | chris-NB | would be easy, if there is a app/function where I can check the callback/cc states |
| 10:32.48 | chris-NB | but I haven't found one :/ |
| 10:35.31 | *** join/#asterisk attila_lendvai (~attila_le@unaffiliated/attila-lendvai/x-3126965) |
| 10:36.04 | attila_lendvai | hi! I can't seem to find any information on what's the situation with DUNDi on OpenWRT. it used to be available in whiterussian, but I can't find it in backfire and I failed googling around... |
| 10:37.02 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 10:38.55 | chris-NB | Chainsaw, connected line is an interesting feature. thanks for pointing me there :) |
| 10:44.11 | Valken | Any recommendation for a load balancing system based on asterisk? |
| 10:44.30 | Valken | I heard that 500 simultaneous calls is quite the maximum asterisk can support. |
| 10:44.56 | *** join/#asterisk paGos (~omer@212.156.220.171) |
| 10:46.33 | Sophira | chris-NB: I didn't realise Asterisk could natively do callbacks. How are you doing it? |
| 10:47.49 | chris-NB | Sophira, As of 1.8 there is this functionality: https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29 |
| 10:48.15 | chris-NB | Sophira, works pretty good. But as I wrote, there is a drawback. |
| 10:48.30 | chris-NB | Sophira, as far as I have seen it |
| 10:49.27 | WIMPy | I think it mus already be 5 years since I last saw that working. |
| 10:49.39 | WIMPy | Nostalgia |
| 10:58.23 | *** join/#asterisk cyford (allen@c-24-99-33-93.hsd1.ga.comcast.net) |
| 10:59.59 | Valken | Have to go for lunch. See you later. |
| 11:05.03 | attila_lendvai | could someone please give me a piece of info I can follow up on about how to set up dundi in the latest openwrt? or am I looking for the wrong thing, dundi has been obsoleted by something else? |
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| 11:40.15 | *** join/#asterisk kjs (kjs@fedora/kjs) |
| 11:41.23 | kjs | hmmm, my asterisk ring tone sounds like a US ring but I am in the UK, is there a way to change this? |
| 11:42.06 | *** join/#asterisk biberao (~marco@unaffiliated/biberao) |
| 11:42.10 | biberao | hi |
| 11:42.36 | biberao | is there a way to make asterisk to call a number i want and call me and make us talk to each other? |
| 11:43.01 | leifmadsen | biberao: yes, it's done via Origination() |
| 11:43.11 | leifmadsen | or a callfile |
| 11:43.21 | leifmadsen | kjs: what technology? |
| 11:43.55 | Chainsaw | kjs: In indications.conf, you need to specify country=uk in [general] |
| 11:44.04 | biberao | leifmadsen: will it ask for any code a protection? |
| 11:44.07 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 11:44.14 | leifmadsen | biberao: you programm that in |
| 11:44.18 | leifmadsen | it's just dialplan |
| 11:44.27 | biberao | ok cool i gotta try that |
| 11:44.29 | leifmadsen | use Read() or something to check for a pin |
| 11:44.34 | leifmadsen | goes to breakfast now |
| 11:44.48 | biberao | i wonder it will work with my voip router which my isp provides |
| 11:44.51 | Chainsaw | kjs: http://pastebin.com/6QjtrBNg |
| 11:45.00 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:45.16 | leifmadsen | biberao: is it running Asterisk... ? |
| 11:45.29 | biberao | my isp's? |
| 11:45.32 | biberao | i dont know |
| 11:45.35 | biberao | the router isnt |
| 11:46.19 | leifmadsen | then why does the router matter? |
| 11:46.33 | biberao | leifmadsen: not the router itself |
| 11:46.37 | biberao | my isp configuration |
| 11:46.58 | biberao | they dont know i want to run that |
| 11:47.34 | biberao | leifmadsen: ill try it :) |
| 11:47.49 | biberao | i need to buy some connecters to split the phones |
| 11:48.36 | kjs | Chainsaw: thank you very much :) |
| 11:49.31 | kjs | Chainsaw: that info goes into the extension.conf - correct ? |
| 11:49.33 | Chainsaw | kjs: Any time. Those should sound right. leifmadsen has a point, in that you should doublecheck your country setting for any analog/ISDN boards in the server. |
| 11:49.36 | Chainsaw | kjs: No, indications.conf |
| 11:49.55 | biberao | leifmadsen: what i mean is that it already comes from voip system to another |
| 11:50.02 | kjs | ah ok, let me try |
| 11:50.26 | leifmadsen | biberao: if you can place a call already, then it won't be any different whatsoever |
| 11:51.05 | biberao | i havent configured yet |
| 11:51.26 | biberao | will try |
| 11:52.10 | biberao | leifmadsen: i have still try fax but either my printer fax doesnt work with it or my isp blocks |
| 11:54.55 | kjs | Chainsaw: that seems to have sorted it out, sounds much better. Probably didn't help the fact the location was set to the states ;) |
| 11:55.09 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:55.10 | Chainsaw | kjs: Indeed. If it sounds like in the movies, you're definitely on US. |
| 11:55.50 | kjs | clients were asking if we had gone on holiday ;) |
| 11:56.10 | Chainsaw | kjs: I put my Polycoms on Australia once by accident. That sounds even more wrong. |
| 11:57.08 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:58.07 | Chainsaw | kjs: But basically, you need to configure this anywhere where you generate or listen to call progress tones. So phone handsets, Asterisk itself (indications.conf, which you've now done) and DAHDI drivers for any analog/ISDN cards. |
| 11:59.07 | biberao | is it possible to test fax without connecting it to the phone port in the router? If its all voip already thru the router? (sorry if stupid question) |
| 12:00.30 | kjs | yeah, thanks - I have another problem, we have a bunch of lines here and i changed the inbound to display a name before ow it only shows the inbound trunk name on the voicmail emails. Is there a way to make it display the caller ID as well? |
| 12:01.55 | Chainsaw | kjs: How are you getting these calls in? ISDN BRI with BT? |
| 12:02.44 | kjs | Chainsaw: over SIP provider (gradwell). |
| 12:02.51 | *** join/#asterisk Arroyo1010 (~sdghhf@93-87-163-160.dynamic.isp.telekom.rs) |
| 12:02.57 | din3sh | Chainsaw: should loadzone= in zaptel.conf correspond to country= in indications.conf? |
| 12:03.16 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:03.27 | Chainsaw | din3sh: Mine does, yes. |
| 12:03.30 | Chainsaw | din3sh: uk on both. |
| 12:04.08 | Chainsaw | kjs: You probably need trustrpid=yes on the relevant SIP peer. |
| 12:04.16 | Chainsaw | kjs: Only on the Gradwell one, mind you. Not globally. |
| 12:04.19 | Arroyo1010 | hi guys. I'm using Asterisk 1.6 with asterisk-gui, and I'm very happy with it!. I have a question: Can anyone recommend a SIP trunking provider for these specs: 1 number/3trunks, about 7000 outbound minutes, 7000 inbound. USA and CAN only |
| 12:04.34 | Arroyo1010 | 7k/7k / monthly |
| 12:04.48 | Chainsaw | waits for Fender to use the bot ITSP listings |
| 12:05.24 | Arroyo1010 | We are curently using Skpye for Business/ Skype Connect, and we are not happy because they have some issues that they can't resolve |
| 12:05.38 | Arroyo1010 | Chainsaw: oh. nice :) |
| 12:05.43 | kjs | Chainsaw: it's setup like this currently: exten => 01225111111,3,Set(CALLERID(name)=MAIN NUMBER) |
| 12:05.45 | Arroyo1010 | wait, too xD |
| 12:07.15 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-10-14.bois.qwest.net) |
| 12:10.53 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 12:22.08 | Arroyo1010 | :'( |
| 12:22.27 | *** part/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 12:24.26 | [TK]D-Fender | Arroyo1010, ... |
| 12:24.30 | [TK]D-Fender | ~itsplist-us |
| 12:24.30 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 12:24.59 | [TK]D-Fender | Arroyo1010, les.net allows up to 5 calls at a time for their metered accounts. |
| 12:27.31 | [TK]D-Fender | Arroyo1010, But voip.ms = cheaper /min. Perhaps the most cost-effective option will have you splitting up inbound VS outbound. This has a benefit of not risking both if one goes down. |
| 12:27.39 | [TK]D-Fender | Arroyo1010, Shop around a bit |
| 12:29.03 | bulkorok | hi... the last 2 days asterisk 1.8.11.0 (compiled on Debian) segfaulted in libc-2.11.3.so twice a day. how can I check what was going on there? |
| 12:32.43 | leifmadsen | bulkorok: https://wiki.asterisk.org/wiki/display/AST/Debugging |
| 12:32.43 | Arroyo1010 | D-Boy: thank you so much. I have benn calling various voip/sip providers, and they are usually quite expencive. I will try the solutions listed on infobot |
| 12:37.01 | D-Boy | Arroyo1010 : you're welcome ! (you can thank who you want really to thank too :p) |
| 12:37.01 | *** join/#asterisk CyfordTechnologi (allen@c-24-99-33-93.hsd1.ga.comcast.net) |
| 12:37.01 | Arroyo1010 | [TK]D-Fender: i wanted to thank you. D-Boy thanks for correcting me ;) |
| 12:37.11 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 12:39.19 | [TK]D-Fender | Arroyo1010, There may some "unlimited" per-channel options that could pan out cheaper than meters as 14,000 @ $.01/min = $140 |
| 12:41.28 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-10-14.bois.qwest.net) |
| 12:42.09 | Arroyo1010 | [TK]D-Fender: so far, the cheapes option i've found for my needs is skype that we use now. 0.008/min, free inbound... but inbound does not work properly at random times during the day. it's on their end, they confirmed. they can't resolve the issue for a month now... |
| 12:42.29 | Arroyo1010 | the sound is excellent and oubound is flawless, though |
| 12:42.30 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:43.04 | Arroyo1010 | it comes to about 70$/month. The second cheapest solution so far is nextiva (i think) |
| 12:43.11 | [TK]D-Fender | Well a single account with les.net for inbound should do it then as they'll give you 5 channels for around $10/month |
| 12:43.14 | *** join/#asterisk cbdev (~cbdev@2a01:4f8:121:4083:1333:3333:3333:3337) |
| 12:43.30 | [TK]D-Fender | And you can keep skype for outbound, and maybe use les.net as a failover. |
| 12:43.40 | Arroyo1010 | [TK]D-Fender: thanks fo the tip, lemme check it out |
| 12:43.49 | [TK]D-Fender | As I said, hybrid solution might be the most cost-effective |
| 12:43.57 | [TK]D-Fender | And offer some backup |
| 12:44.04 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:44.08 | Arroyo1010 | awesome advice |
| 12:45.34 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:46.02 | *** join/#asterisk playmobil (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 12:50.49 | playmobil | Hi, I'm experimenting a fixed 12 seconds delay between Dial and first ring from ITSP. What can I suggest to them for eliminate this delay ? |
| 12:51.21 | Arroyo1010 | Ok so, If I use les.net for inbound (transfer my DID to les), and use Skype for outbound... can I force asterisk to send my callerID as a specific number? |
| 12:51.27 | *** join/#asterisk nosaj (~jbarinas@200.25.224.130) |
| 12:51.44 | Arroyo1010 | when i calll someone, they would get my les.net number |
| 12:51.48 | Arroyo1010 | is that possible? |
| 12:52.32 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:56.10 | p3nguin | It sends whatever you configure it to send. |
| 12:59.15 | [TK]D-Fender | Arroyo1010, Depends on what Skype permits. I don't know them.... |
| 12:59.34 | Arroyo1010 | gotcha |
| 12:59.47 | [TK]D-Fender | playmobil, Show us the call and tell us precisely what you're using for each leg of the call. |
| 13:03.07 | [TK]D-Fender | ~pb |
| 13:03.08 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:03.10 | [TK]D-Fender | ^^^ |
| 13:03.49 | Arroyo1010 | [TK]D-Fender: yeah, skype won't allow custom cid :) |
| 13:07.31 | din3sh | [TK]D-Fender: hi |
| 13:07.40 | *** join/#asterisk RogerH (545c6276@gateway/web/freenode/ip.84.92.98.118) |
| 13:07.55 | din3sh | [TK]D-Fender: have updated the BRI gateway firmware, no luck |
| 13:07.56 | din3sh | :S |
| 13:08.09 | [TK]D-Fender | replace it. |
| 13:08.18 | RogerH | Any plans to use a new Centos kernel in Asterisk Now (2.0 beta) ISO? The ISO I tried does not recognise my SATA CD/DVD drive and mainboard (Via Nano VE-900 ITX) has no IDE ports, only SATA |
| 13:08.28 | RogerH | Or is there a workaround |
| 13:08.41 | din3sh | debug 1.4 (working config) = http://paste2.org/p/1983124 |
| 13:09.06 | din3sh | debug 1.8 (not working) =http://paste2.org/p/1983125 |
| 13:09.07 | leifmadsen | RogerH: it's just CentOS... pretty sure you can just login and upgrade it like any other CentOS box... |
| 13:09.25 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yyfkxeypiaulvmap) |
| 13:09.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:09.40 | WIMPy | leifmadsen: Upgrade the CD? |
| 13:09.51 | din3sh | RogerH: yum -y install kernel-devel |
| 13:10.01 | leifmadsen | WIMPy: that's not what I said |
| 13:10.09 | leifmadsen | although I didn't read "ISO" before |
| 13:10.20 | leifmadsen | I should really stop trying to respond to stuff before I finish my first coffee |
| 13:10.38 | leifmadsen | points RogerH at Qwell |
| 13:10.47 | RogerH | My problem is first install on a new machine. Gets stuck at Unable to download kickstart file and cdrom:opt1-ks.cfg. Elastix does same thing |
| 13:10.59 | din3sh | [TK]D-Fender: can you try to have a look at the debugs and see if differences |
| 13:11.27 | din3sh | RogerH: use an extenal dvd drive |
| 13:11.33 | [TK]D-Fender | din3sh, Sorry, I've spent about as much time on that device as I care to. This has eaten up several weeks now. |
| 13:11.42 | din3sh | :/ |
| 13:11.54 | din3sh | ok nvm |
| 13:12.10 | RogerH | din3sh: Thanks. Don't have an external drive. Will have buy one. |
| 13:14.26 | RogerH | Thanks :-) |
| 13:14.46 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 13:15.29 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 13:17.12 | playmobil | [TK]D-Fender, about delay --> http://pastebin.com/gB1W0erX |
| 13:18.00 | [TK]D-Fender | playmobil, and I asked what was on each end.... |
| 13:19.12 | playmobil | [TK]D-Fender, I've not access to itsp system. |
| 13:23.04 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
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| 13:50.49 | acidfoo | is there any known bug with attended transfer ? (features.conf, option atxfer, and Dial() with option T) |
| 13:51.49 | acidfoo | because if the 'target of transfer' hangup, the other 2 (the transferer and the transferee) hangup |
| 13:53.46 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.94.105) |
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| 13:58.20 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 13:58.26 | ollii | hey |
| 13:58.49 | ollii | could someone recommend some sip to fxs ata like grandstream gxw40xx ? |
| 13:59.28 | ollii | I've tried several of them...linksys pap2t, grandstream ht502/gxw4004/8/24 ... none of them are really that good |
| 14:00.14 | acidfoo | define "good" |
| 14:00.23 | acidfoo | what are your expectation |
| 14:00.32 | [TK]D-Fender | Or at least what wasn't "good" about those others |
| 14:01.55 | Arroyo1010 | Can I register to more then one sip provider? |
| 14:03.05 | Arroyo1010 | ollii: I don't know where you live, but we recently purchased various models of Yealink phones and we are very happy. They are chinese, but the quality is really great. If you have them in your country, at least try and see some in action. |
| 14:03.34 | Arroyo1010 | quality: plastic, behaviour, software, menus, web administration - everything is just... right |
| 14:03.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:03.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.07 | [TK]D-Fender | Arroyo1010, Yes |
| 14:04.23 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:05.18 | Arroyo1010 | [TK]D-Fender: ty |
| 14:06.28 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 14:06.47 | ollii | ... wasn't clear..sorry |
| 14:06.57 | ollii | germany, we have tons of faxes |
| 14:07.11 | anonymouz666 | ollii: pap2t is indeed a good FXS ata. |
| 14:07.13 | ollii | T.38 and stable ... thats want i need |
| 14:07.23 | ollii | -want +what |
| 14:07.25 | Arroyo1010 | ollii: i'm an idiot lol. you were clear :) |
| 14:07.44 | anonymouz666 | pap2t does not support T.38 :/ |
| 14:08.00 | [TK]D-Fender | ollii, None of those devices suppotr T.38 AFAIK so you should expect failure, especially if you are using an ITSP that isn't your actual ISP as well and you don't have the best possible conditions. |
| 14:08.56 | [TK]D-Fender | ollii, How many do you need? All at one site? |
| 14:09.47 | ollii | gxw4004/8/24 support T.38 more or less |
| 14:10.12 | ollii | setup: BRI/PRI <-> Media-Gateway <-> Asterisk <-> SIP ATA FAX |
| 14:10.33 | ollii | there are fxs modules for our media gateway as well...but these are too expensive |
| 14:11.05 | ollii | some customers event want to start faxing via ata over wlan :S |
| 14:11.59 | [TK]D-Fender | ollii, your gatway will have to support T.38 as well, and if they are being cheap/stupid about it... then nothing you can do. |
| 14:12.14 | [TK]D-Fender | ollii, Technology < stupidity |
| 14:15.02 | ollii | Media-Gateway's t.38 is fine ... in combination with grandstream gxw4004 its working ... but the gxw often dies and needs a reboot...no reply to sip invites and so on |
| 14:15.08 | ollii | www.beronet.com |
| 14:16.54 | p3nguin | anonymouz666: The PAP2T is supposed to support T.38 ... so why would it not? |
| 14:18.27 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 14:18.30 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 14:18.35 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 14:18.45 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 14:18.59 | p3nguin | SPA2102 and SPA3102 surely do. |
| 14:25.03 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 14:27.52 | p3nguin | Wait... Are we talking about pass-through or direct T.38 support? |
| 14:29.41 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 14:31.06 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 14:32.10 | ollii | both... * 1.4 and * 1.8 |
| 14:32.42 | p3nguin | I'm also finding information that the PAP2 v2 does in fact support direct T.38. |
| 14:33.01 | ollii | the old one did not support t.38 |
| 14:34.58 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:36.24 | *** join/#asterisk Foxi352 (~Foxi352@v-172-4.access.restena.lu) |
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| 14:38.59 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 14:39.33 | p3nguin | Even thevoipconnection thinks the PAP2T is T.38 compatible, but I don't know if they are claiming it only works with pass-through or what. They don't go into specifics. |
| 14:41.07 | [TK]D-Fender | Coppice has stated that they are not T.38 for some time time... |
| 14:44.25 | playmobil | [TK]D-Fender, about the delay: if I define two extensions (for ex. 12 and 122); is there a timeout option for tell to asterisk "wait X seconds before matching an extension" ? |
| 14:45.27 | p3nguin | My provider does not support T.38, so I can't even see what happens if I try to accept a T.38 fax via PAP2T. |
| 14:47.45 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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| 14:53.12 | *** part/#asterisk Arroyo1010 (~sdghhf@93-87-163-160.dynamic.isp.telekom.rs) |
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| 14:53.51 | *** join/#asterisk RogerH (545c6276@gateway/web/freenode/ip.84.92.98.118) |
| 14:54.54 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 14:54.58 | treborsux | <treborsux> Why would phone not have any show that I have voicemail |
| 14:54.59 | treborsux | [10:54] <treborsux> the light does not come on nor does envolope show up |
| 14:54.59 | treborsux | [10:54] <treborsux> polycom phones |
| 14:54.59 | treborsux | [10:54] <treborsux> what do I have to change to make either the light or envolope show up |
| 14:55.12 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yyfkxeypiaulvmap) |
| 14:55.29 | Naikrovek | you did somethign to make it not show up, i'm betting |
| 14:57.32 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 14:58.02 | Sophira | Does Asterisk 1.8 support ICE? |
| 14:58.25 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:58.29 | Sophira | (I don't believe it's being used if so, and I don't know how to turn it on - does anyone else?) |
| 14:59.10 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gziwmyatzhjcwqkc) |
| 14:59.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:00.59 | FinboySlick | Hello people. As per my earlier question, I added SIP/account/number members to have a queue dial external numbers. I've tested the SIP/account/number bit with a Dial() statement to make sure it works but doing queue show myqueue lists this member as invalid. Any pointers as to how I might diagnose this? |
| 15:02.50 | FinboySlick | Essentially, I just did: queue add member SIP/Metaswitch/1235550169 to web |
| 15:03.04 | [TK]D-Fender | treborsux, Immediate guess : didn't define the mailbox for the peer |
| 15:03.41 | FinboySlick | [TK]D-Fender: thanks again for giving me that hint, btw. |
| 15:04.18 | [TK]D-Fender | FinboySlick, Show us |
| 15:04.23 | [TK]D-Fender | (for your more recent Q |
| 15:04.53 | FinboySlick | [TK]D-Fender: you mean the output of 'queue show web' ? |
| 15:04.53 | anonymouz666 | Sophira: it does not |
| 15:05.30 | [TK]D-Fender | FinboySlick, yes, before/aftter,, dump your peer, etc |
| 15:06.06 | Sophira | anonymouz666: Thanks. |
| 15:06.15 | FinboySlick | [TK]D-Fender: Ok, lemme put together a little pastebin. |
| 15:06.19 | anonymouz666 | Sophira: mediaproxy does support |
| 15:06.36 | anonymouz666 | but you have to use it with a SIP PROXY |
| 15:11.25 | rrittgarn | is there an easy way to play DTMF Tones into a call via the dialplan? |
| 15:11.42 | [TK]D-Fender | rrittgarn, "core show application senddtmf" |
| 15:11.51 | p3nguin | SendDTMF() |
| 15:12.04 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 15:12.09 | rrittgarn | tyty |
| 15:12.17 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 15:12.35 | Sophira | anonymouz666: Okay, thanks. I'm surprised Asterisk doesn't do it natively, but fair enough. |
| 15:12.54 | FinboySlick | [TK]D-Fender: How does one dump the peer? |
| 15:13.44 | [TK]D-Fender | FinboySlick, "sip show peer X" |
| 15:14.12 | FinboySlick | Ah, heh... I thought you meant drop him somehow. |
| 15:17.16 | *** join/#asterisk jrondeau (~jrondeau@24.214.205.162) |
| 15:17.31 | jrondeau | hello channel |
| 15:17.54 | FinboySlick | [TK]D-Fender: Hmmm... Looks like showing the peer doesn't work. I'll have a closer look at that bit. |
| 15:18.16 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 15:18.22 | [TK]D-Fender | FinboySlick, Yes, making sure your selection alone is sane should have been step #1 :) |
| 15:18.30 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 15:19.15 | *** join/#asterisk pithen (~ke-esc@155.229.209.170) |
| 15:19.20 | FinboySlick | [TK]D-Fender: Heh, I need to figure out what I messed up, it actually worked last week. Dial(SIP/Peer/number) still works though. |
| 15:19.56 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 15:20.05 | FinboySlick | [TK]D-Fender: Anyway, thanks for the brain jolt. |
| 15:21.03 | jrondeau | i thought it was Dial(Sip/Number@Peer) |
| 15:21.25 | FinboySlick | jrondeau: I think both are valid. |
| 15:21.26 | [TK]D-Fender | Tech/Peer-channel-group/number |
| 15:21.39 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 15:21.39 | [TK]D-Fender | proper consistent way |
| 15:21.56 | FinboySlick | Gaaah... I had done 'sip show peer SIP/mypeer' |
| 15:22.16 | FinboySlick | I'm beyond rusty. |
| 15:24.50 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 15:25.15 | CyBeRxIxO | is a Digium g729 paid licence compatible on an elastix box? |
| 15:25.24 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:26.50 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-073.mobile.uci.edu) |
| 15:27.38 | playmobil | there are two extensions: 12 and 122; is there a timeout to permit Asterisk wait X seconds for permit user digits 122 (without match 12 at second digit) ? |
| 15:28.51 | Qwell | playmobil: Yes, there is a timeout already |
| 15:29.18 | playmobil | @Qwell, where is defined ? which name ? |
| 15:29.22 | Qwell | I used to know what the option name was.. people used to ask all the time |
| 15:29.36 | p3nguin | TIMEOUT(digit) ? |
| 15:29.42 | Qwell | that's the one |
| 15:30.30 | p3nguin | If you have both extension 12 and extension 122 in the context where you are waiting for an extension, there will already be a fairly long timeout by default. |
| 15:30.37 | Qwell | it's like 500ms |
| 15:30.44 | Qwell | or, was. maybe 1000 now |
| 15:30.48 | playmobil | p3nguin, exists an option for configure it globally ? (asterisk.conf, features.conf, other) ? |
| 15:30.54 | p3nguin | dial plan only |
| 15:31.39 | Qwell | playmobil: really though, if you're running into issues like that, it's usually because your dialplan isn't designed properly |
| 15:31.47 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 15:32.09 | p3nguin | When you're waiting for that shorter extension to "go" while it is waiting for more digits, it seems like it's more like 4000 ms. |
| 15:32.22 | Qwell | p3nguin: that high? |
| 15:32.36 | p3nguin | Feels like it. I'd have to measure it to say what it really is. |
| 15:32.37 | playmobil | thanks to all. |
| 15:32.38 | Qwell | sure you don't have a higher pattern that would be hitting absolute? |
| 15:32.57 | CyBeRxIxO | p3nguin i bought g729 licence, could you help me on the installation? |
| 15:33.13 | Qwell | CyBeRxIxO: There are very detailed instructions in the README |
| 15:33.22 | Qwell | http://downloads.digium.com/pub/telephony/codec_g729/README |
| 15:33.36 | CyBeRxIxO | im on that, but where do i have to make that steps |
| 15:33.42 | CyBeRxIxO | on my laptop or on asterisk |
| 15:34.13 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 15:34.27 | p3nguin | Let's see... are you installing a codec in Asterisk or in your laptop? |
| 15:35.23 | CyBeRxIxO | on asterisknow |
| 15:35.39 | Qwell | yum install asterisk-codec_g729a |
| 15:35.41 | Qwell | done and done |
| 15:36.15 | p3nguin | Yum can install the key as well? |
| 15:36.24 | Qwell | actually |
| 15:36.51 | Qwell | CyBeRxIxO: find the "Digium Addons" page in FreePBX |
| 15:37.01 | Qwell | You can do everything you need. |
| 15:37.27 | CyBeRxIxO | no README way if i go trough digium addons? |
| 15:44.09 | *** join/#asterisk bchia (~Adium@nat/digium/x-hejnszdkxyjslfaa) |
| 15:44.46 | *** join/#asterisk iulhk (~iulhk@119.152.73.46) |
| 15:44.51 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 15:46.43 | FinboySlick | [TK]D-Fender: http://pastebin.com/W1qjZpL3 For some odd reason, it randomly started working. I removed the peer, re-added it, now it's invalid again. |
| 15:47.51 | CyBeRxIxO | i activated key trought Digium Add-on on asterisknow, how to verify? |
| 15:48.02 | CyBeRxIxO | is it istalled now? |
| 15:49.26 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:49.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 15:50.01 | FinboySlick | Maybe it's related to some sort of sip register/reregister timing? |
| 15:50.35 | p3nguin | cyberxixo: See if you can run "g729 show license" on the asterisk CLI. |
| 15:55.22 | *** join/#asterisk capt-rogers (~IceChat77@mail1.decisioningsolutions.com) |
| 15:55.43 | capt-rogers | how do i check trunk status ? |
| 15:57.15 | p3nguin | Define "trunk." |
| 15:57.30 | treborsux | http://pastebin.com/rAaXD1d7 |
| 15:57.34 | p3nguin | If you simply mean a PEER, then use "sip show peer XXX" where XXX is the peer name. |
| 15:58.17 | treborsux | http://pastebin.com/rAaXD1d7 |
| 16:05.10 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:06.07 | CyBeRxIxO | No such command 'g729 show license' |
| 16:08.00 | p3nguin | Do you have g729 show ... anything? |
| 16:08.23 | p3nguin | Do you have g729 ... anything? |
| 16:08.28 | p3nguin | Maybe you have to manually load the g729 module. |
| 16:08.47 | p3nguin | I do not use and cannot support FreePBX, so I have no clue what it does for you. |
| 16:12.38 | CyBeRxIxO | p3nguin are you linux expert? |
| 16:13.10 | p3nguin | Perhaps. Why do you ask? |
| 16:13.14 | Qwell | CyBeRxIxO: module load codec_g729a.so |
| 16:15.11 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:15.19 | CyBeRxIxO | is that a command i have to put on CLI? "module load codec_g729.so"? |
| 16:16.07 | Qwell | mjordan: your warning should have had exclamation points or something. maybe some <blink/> tags |
| 16:16.13 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 16:17.09 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:19.12 | *** join/#asterisk watchy (~yo@mail.spectra-tech.net) |
| 16:19.38 | watchy | anyone ever play with app_rtsp? |
| 16:21.54 | *** join/#asterisk vfabi (~fabi@178.76.123.249) |
| 16:26.37 | watchy | is ast realtime worth using nowadays? |
| 16:27.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:27.35 | Qwell | no |
| 16:27.49 | Qwell | there are much better ways to do the same sort of thing |
| 16:28.35 | *** join/#asterisk TheMan (~garry@66.237.29.132.ptr.us.xo.net) |
| 16:28.36 | watchy | can you give me one really quick? |
| 16:29.06 | Qwell | #exec is your friend |
| 16:29.14 | watchy | i am currently about to redo our asterisk based system at work. we have 3 servers some still running 1.2 |
| 16:29.21 | Qwell | shudders |
| 16:29.23 | sp00kz | ouch |
| 16:29.36 | watchy | but we are about to link our locations by fiber and i'm going to build 1 new box to replace 3 |
| 16:29.54 | Qwell | if it's only 1 box, why not just use flat configs? |
| 16:29.56 | sp00kz | anyone played with the digium phones? |
| 16:30.10 | Qwell | sp00kz: Yes! And they're awesome. And I'm completely biased. |
| 16:30.14 | watchy | qwell: lots of phones / devices / custom paging system |
| 16:30.16 | sp00kz | haha <3 |
| 16:30.19 | sp00kz | I have one on my desk, actually |
| 16:30.31 | Qwell | sp00kz: _Corey_ has been a pretty big advocate |
| 16:30.33 | sp00kz | curious if it's possible to get auto-complete working |
| 16:30.35 | watchy | we use alot of cyberdata paging equipment |
| 16:30.38 | Qwell | auto-complete? |
| 16:30.42 | treborsux | <PROTECTED> |
| 16:30.47 | sp00kz | recently typed numbers, yeah |
| 16:31.05 | sp00kz | my old ciscos/polycoms both did it, was very handy :P |
| 16:31.12 | watchy | digium phones better then polys? |
| 16:31.13 | Qwell | sp00kz: I don't recall seeing anything like that, however, that would be a fantastic feature request. |
| 16:31.21 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 16:31.37 | [TK]D-Fender | watchy, Not yet fom what I've heard |
| 16:31.42 | [TK]D-Fender | from* |
| 16:31.49 | sp00kz | certinly comparable watchy, not quite as ironed out |
| 16:31.51 | watchy | i didnt even know they made phones |
| 16:31.55 | sp00kz | it's very new |
| 16:32.09 | watchy | i use all polys here for phones, cyberdata equipment for paging |
| 16:32.33 | watchy | the digiums do look good though |
| 16:32.51 | rrittgarn | I have 6 of the digium phones here. They are pretty nifty |
| 16:33.15 | rrittgarn | I have them integrated with a switchvox smb setup and it was super easy to get everything working. They also have a lot of nifty little features working |
| 16:33.47 | sp00kz | indeed. built in voicemail/call parking management |
| 16:33.50 | sp00kz | very fancy |
| 16:34.11 | Qwell | on switchvox you also get queue magic and a few other things not yet implemented in DPMA |
| 16:37.08 | sp00kz | http://i.imgur.com/xvmlv.jpg |
| 16:37.13 | sp00kz | :P |
| 16:37.25 | Qwell | You should probably check who called you. |
| 16:37.29 | Qwell | I bet it's important. |
| 16:37.29 | sp00kz | naw |
| 16:37.47 | sp00kz | Doubtful |
| 16:38.16 | Qwell | also what is that monstrosity in the background? |
| 16:39.00 | sp00kz | a microtik board with openwrt, mounted to a antenna box housing |
| 16:39.06 | Qwell | ahh |
| 16:39.16 | Faustov | sp00kz: is there anything similar but DECT? |
| 16:39.29 | treborsux | You may have to add @context to the mailbox entry. This seems to fix things for many users. Note that this context is the context specified in voicemail.conf for the extension, not the context specified in sip.conf |
| 16:39.50 | raden | anyone have asterisk behind a cisco router with one to one nat ? |
| 16:39.58 | sp00kz | i know nothing about DECT |
| 16:39.58 | sp00kz | sorry |
| 16:40.01 | raden | Naikrovek, yoooo bro |
| 16:40.12 | treborsux | where is voicemail.cfg??????? |
| 16:41.47 | Naikrovek | wassup raden |
| 16:41.56 | raden | having asterisk / cisco issues |
| 16:42.01 | raden | Naikrovek, what u up to |
| 16:42.02 | Naikrovek | that sucks |
| 16:42.15 | Naikrovek | actually right now i'm reading up on microsoft lync |
| 16:42.16 | Qwell | Ciscos causing NAT problems? Now I've heard everything! |
| 16:42.22 | Qwell | wait, nm, heard that thousands of times before. |
| 16:42.25 | Naikrovek | and how we'll use it for IM & softphone and all that |
| 16:42.26 | raden | lync ? |
| 16:42.26 | sp00kz | :> |
| 16:42.42 | Naikrovek | yah |
| 16:42.48 | Naikrovek | specing out a phone system |
| 16:42.52 | treborsux | Does anyone know how to get the light to work on a polycom phone? |
| 16:42.57 | raden | Qwell, i have a 7206 doing a static one to one and lets just say asterisk can register with vitelity but a phone on outside cant seem to register in |
| 16:43.24 | raden | treborsux, the light for what ? |
| 16:43.41 | Faustov | Qwell: any idea if there are plans for digium phones using DECT? couldn't find anything on the website but the wired ones seem quite impressive |
| 16:43.54 | treborsux | the voicemail light |
| 16:43.56 | Naikrovek | he wants a light for voicemail notification, I think. |
| 16:44.10 | Naikrovek | you've turned it off; by default it's there, I'm sure. |
| 16:44.22 | treborsux | it is not |
| 16:44.29 | treborsux | i havent changed anything |
| 16:44.36 | treborsux | does not work on any phone |
| 16:44.41 | treborsux | polycom phones |
| 16:45.03 | Naikrovek | you have a config problem |
| 16:45.19 | *** join/#asterisk Beltechs (~beltechs@pool-72-87-189-166.lsanca.btas.verizon.net) |
| 16:45.22 | treborsux | i looked at this http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk but i dont know where it wants me to add @context |
| 16:45.25 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 16:45.37 | raden | treborsux, look at notifications |
| 16:45.55 | treborsux | context specified in voicemail.conf for the extension wher ethe heck is that? |
| 16:46.47 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 16:46.49 | treborsux | notifications? |
| 16:46.57 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 16:50.21 | biberao | what do you advise reading for an asterisk first timer? |
| 16:50.28 | Qwell | ~book |
| 16:50.28 | 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 |
| 16:50.31 | Qwell | that |
| 16:50.51 | treborsux | anyone on the making the envolope or the led light fo a voicemail??? |
| 16:51.14 | treborsux | you would trhink this would be on by default |
| 16:51.23 | Qwell | treborsux: Have you looked at the admin guide for the config you need there? Do you have a mailbox set in sip.conf in Asterisk? |
| 16:52.15 | adeel|work | is it possible to set multiple externip's in sip.conf? |
| 16:52.21 | treborsux | sip.conf in the tftp??? |
| 16:52.21 | biberao | thanks Qwell |
| 16:52.31 | CyBeRxIxO | http://downloads.digium.com/pub/register im on that |
| 16:52.34 | treborsux | all the boxes work |
| 16:52.42 | treborsux | that is all fione |
| 16:52.44 | CyBeRxIxO | how to make sure im on correct version if x86 or 64 |
| 16:53.04 | treborsux | just does not light light nor does in make the envolope |
| 16:53.10 | p3nguin | cyberxixo: core show version |
| 16:53.23 | FLeiXiuS | Is it possible to run channel funcitons in the background? Like RECORD? |
| 16:53.31 | CyBeRxIxO | ty a lot |
| 16:54.23 | *** join/#asterisk MarKsaitis (~MarKsaiti@027d5646.bb.sky.com) |
| 16:54.31 | CyBeRxIxO | n a x86_64 running Linux means any? |
| 16:54.47 | p3nguin | It meanst what it says. |
| 16:54.57 | p3nguin | x86_64 <----- |
| 16:55.17 | treborsux | what has to be done for the light for voicemail notification to weork? |
| 16:55.19 | CyBeRxIxO | oh ok i get it |
| 16:55.57 | Qwell | treborsux: Re-read what I said. I was very explicit. |
| 16:56.12 | watchy | is 10.x stable? |
| 16:56.22 | watchy | or should i go with 1.8 for a new box |
| 16:56.31 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:56.36 | Qwell | watchy: We don't label releases "stable". 10 is the current release branch. |
| 16:56.39 | Qwell | ~asteriskversions |
| 16:56.43 | p3nguin | watchy: I guess that depends on your purposes and desires. |
| 16:56.43 | Qwell | ~asterisk versions |
| 16:56.52 | Qwell | infobot: I don't like you anymore. |
| 16:56.52 | infobot | You don't like you anymore.? |
| 16:56.53 | p3nguin | watchy: I only use LTS for production boxes. |
| 16:57.03 | watchy | which is currently 1.8 right? |
| 16:57.05 | Qwell | watchy: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 16:57.06 | biberao | Qwell: is it possible to have asterisk get its info from rj45 and output to rj11? |
| 16:57.14 | Qwell | biberao: get what info? O.o |
| 16:57.23 | watchy | i wish app_transcode worked in 1.8 |
| 16:57.33 | Qwell | watchy: what is that supposed to do? |
| 16:57.41 | watchy | transcode video to another format |
| 16:57.49 | biberao | Qwell: to get the phone connection |
| 16:57.52 | biberao | its already voip |
| 16:58.02 | Qwell | ~ata |
| 16:58.02 | infobot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 16:58.12 | Qwell | biberao: Interfaces are explained well in the book |
| 16:58.13 | watchy | im trying to build a way to call a sip device and use voice to it, but show the video of an ip camera |
| 16:58.27 | watchy | like a video intercom without the video phone |
| 16:58.30 | biberao | oki |
| 16:58.51 | watchy | app_rtsp doesn't seem to like my camera output for some unknown reason |
| 16:58.52 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 16:59.46 | raden | Qwell, is there a cicso howto with asterisk cause i have stuff configured how id think it should be configuerd and no go ... works on 3 off brand routers i have though of course |
| 16:59.47 | Beltechs | http://www.lifesize.com/ |
| 17:00.39 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:01.32 | watchy | i was trying to do this video intercom thing mainly for my house so i can have it dial my cell and I can see whos at the door |
| 17:02.21 | p3nguin | raden: I had a Cisco SOHO router that I could not get to work well with RTP. I sold it. |
| 17:02.31 | raden | fml :( |
| 17:03.18 | raden | p3nguin, running a 7206 |
| 17:03.42 | raden | and on a traceroute it gets to the router main IP and bounces for 3 hops in the 7206 before getting to the next ip |
| 17:03.47 | p3nguin | I had problems with an 800 series. |
| 17:04.28 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 17:04.40 | raden | ill have to play with it today some more that is all i can do i gues |
| 17:04.52 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 17:04.53 | sp00kz | my only suggestion for you raden is to check (turn off?) the protocol fixups that cisco adds. e.g. http://www.cisco.com/en/US/docs/ios/12_2t/12_2t8/feature/guide/ftnatsip.html |
| 17:04.58 | brian98 | Evening all |
| 17:06.12 | p3nguin | It was only RTP that was an issue. No matter what I did, RTP packets always had the remote phones' private IP addresses rather than the public addresses. |
| 17:06.46 | raden | p3nguin, shit thats what i was seeing lasyt night |
| 17:06.55 | raden | be back in 2 hours gotta meet gf for lunch fml |
| 17:07.46 | p3nguin | sold the Cisco + installed a Linux-based solution = problem solved |
| 17:07.46 | *** join/#asterisk wonderworld (~ww@dsdf-4db5da91.pool.mediaWays.net) |
| 17:08.49 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-073.mobile.uci.edu) |
| 17:09.35 | treborsux | what file is mwi/ tag in??? |
| 17:09.53 | treborsux | http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk something is missing here |
| 17:10.00 | treborsux | i have done all it says |
| 17:10.19 | [TK]D-Fender | treborsux, You don't need to subscribe at all for MWI |
| 17:10.32 | treborsux | do i literally put @context in the voicemail.conf?? |
| 17:10.35 | [TK]D-Fender | treborsux, You aren't looking at your own peer configs. And take this back into #freepbx |
| 17:10.41 | treborsux | Fender please tell me what i need to do |
| 17:11.04 | p3nguin | (1151.23) <@Qwell> treborsux: Have you looked at the admin guide for the config you need there? Do you have a mailbox set in sip.conf in Asterisk? |
| 17:11.12 | p3nguin | clue ^ |
| 17:13.07 | brian98 | I'm sending calls to Asterisk from opensips and want to check in Asterisk if privacy flag is set in remote Party ID header and if it is then make callerid(num)='' - is this possible? Using asterisk 1.8.9.3 |
| 17:14.04 | brian98 | I can't see any documentation on using values from remote party id or p asserted identity and using them in extensions. |
| 17:14.17 | [TK]D-Fender | brian98, "core show function SIP_HEADER" |
| 17:14.22 | brian98 | Ty |
| 17:15.14 | Qwell | There are actually values in ${CALLERID()} for that stuff |
| 17:15.21 | Qwell | see core show function callerid |
| 17:15.39 | Qwell | I think. |
| 17:16.22 | *** join/#asterisk Spriggan (~ircap@excsupercol.supercable.net.co) |
| 17:16.24 | brian98 | I'm bridging between SIP and ISDN |
| 17:16.49 | brian98 | If it's SIP <> SIP and privacy=full is in the SIP header from the calling peer it goes to the called peer |
| 17:17.22 | brian98 | If I go to ISDN asterisk doesn't automatically look at the pai or rpid to check if privacy flag is set |
| 17:18.32 | Spriggan | Hi everyone, i need to get the Extension info via api. I need to know if the call is outgoing or incoming and the duration, anyone??? please ?? |
| 17:19.26 | p3nguin | What kind of API? |
| 17:19.36 | brian98 | why should it I guess.. |
| 17:24.56 | Spriggan | Asterisk manager API |
| 17:25.14 | FLeiXiuS | Is there any way to background functions in a dial plan? |
| 17:25.50 | leifmadsen | no |
| 17:26.00 | leifmadsen | Dialplan proceeds serially |
| 17:26.11 | FLeiXiuS | I want to record a channel, then enter a conference room. |
| 17:26.12 | leifmadsen | perhaps you can explain what you're trying to do |
| 17:26.17 | FLeiXiuS | Where only that channel is being recorded... |
| 17:26.42 | FLeiXiuS | I just want to record a 'special' user. |
| 17:26.43 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 17:26.55 | leifmadsen | FLeiXiuS: just use MixMonitor() |
| 17:27.44 | Spriggan | p3nguin: Asterisk manager API |
| 17:28.01 | leifmadsen | FLeiXiuS: actually I think I meant Monitor() as MixMonitor() mixes both streams |
| 17:28.04 | Qwell | Spriggan: You don't need AMI for that. All of that stuff is stored in CDR |
| 17:28.05 | FLeiXiuS | This will allow the dial plan to proceed |
| 17:28.08 | FLeiXiuS | leifmadsen, ^ ahh |
| 17:28.12 | leifmadsen | that's how Monitor and MixMonitor work yes |
| 17:28.26 | leifmadsen | if it didn't proceed with those applications then you'd never be able to record anything |
| 17:28.28 | FLeiXiuS | Excellent - i was using record and MeetMe's (r) option. |
| 17:28.36 | FLeiXiuS | Thats what I was wondering. |
| 17:28.39 | leifmadsen | no, you need to start recording prior to the meetme |
| 17:28.47 | FLeiXiuS | RIght, I get it. |
| 17:28.48 | FLeiXiuS | Let me try that |
| 17:29.46 | *** join/#asterisk din3sh (din3sh@41.136.241.235) |
| 17:30.05 | *** join/#asterisk chasing`Sol (~cS@197.134.239.198) |
| 17:30.29 | Spriggan | @Qwell, i need to get this info in real time. that's why im using astersik manager API |
| 17:31.15 | Qwell | Spriggan: well, there are plenty of events that would contain that info. You just need to figure out which ones, and get the details from them. |
| 17:31.37 | CyBeRxIxO | whats my operating system if i have asterisk now |
| 17:31.49 | Qwell | Linux. |
| 17:31.50 | CyBeRxIxO | "asterisknow" |
| 17:31.51 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 17:32.12 | CyBeRxIxO | GNU/Linux x86? |
| 17:32.41 | Qwell | Didn't you already figure out the architecture earlier? |
| 17:32.51 | Spriggan | Â Â Â @Qwell can you tell me what event can i get this info from? |
| 17:33.00 | Qwell | Spriggan: no, there are several |
| 17:33.58 | Spriggan | can u tell me please , where i can find documantation about the events in particular? |
| 17:34.27 | *** join/#asterisk TheMan (~garry@66.237.29.132.ptr.us.xo.net) |
| 17:36.10 | Spriggan | Â Â Â @Qwell: do you know if there is an astersik manager api channel? |
| 17:37.00 | *** join/#asterisk jeffik (~chatzilla@76-10-173-164.dsl.teksavvy.com) |
| 17:37.09 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:38.24 | ruben23 | hi guys i have set an audio for MOH on my asterisk but i get this error: ----> [Apr 18 01:35:46] WARNING[16129]: file.c:664 ast_openstream_full: File julieIB does not exist in any format<---------------already tried wav, gsm,mp3 stillt eh same any ideas..? |
| 17:38.44 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:39.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:39.41 | watchy | tk: what ftpd do you use to provision phones? |
| 17:39.50 | p3nguin | Linux is a kernel, not an operating system. |
| 17:40.21 | din3sh | hi p3nguin |
| 17:40.29 | p3nguin | yup |
| 17:40.35 | *** join/#asterisk albertoandrade (~albertoan@187.59.22.190) |
| 17:40.39 | din3sh | i tried the polycom as you suggested |
| 17:40.58 | watchy | anyone tried streaming iheartradio for moh? |
| 17:41.22 | Qwell | watchy: Doing so would violate all sorts of copyright laws. |
| 17:41.24 | din3sh | sorry to be boring you with my ongoing xfer issue |
| 17:41.24 | din3sh | :/ |
| 17:41.36 | watchy | qwell: thats depressing |
| 17:41.48 | watchy | what if i paid for siriusxm? |
| 17:41.53 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 17:42.07 | Qwell | you don't have a license to rebroadcast, nor do you have a license to play the music that they play. |
| 17:42.18 | watchy | what do most people use for moh? |
| 17:42.27 | coppice | you cannot be sirius |
| 17:42.37 | Qwell | watchy: stuff they can play legally |
| 17:42.47 | watchy | so crappy stuff |
| 17:42.55 | coppice | watchy: there are CDs made specifically as muzac |
| 17:42.56 | Qwell | or they pay tons of money to people like BMI |
| 17:43.33 | watchy | alot of the guys around here who sell phone systesm just pipe in the local fm easy listening station through phone in |
| 17:43.40 | ruben23 | idea guys |
| 17:43.49 | _Corey_ | bah, it's not that expensive... freeplaymusic for example is maybe $150 a year |
| 17:43.52 | Qwell | watchy: Feel free to do so, just know that you can and will be sued. |
| 17:44.22 | _Corey_ | There's a lot of creative commons stuff that's not awful though |
| 17:44.29 | p3nguin | can be, but probably won't ever be. |
| 17:44.36 | Qwell | p3nguin: You'd be surprised.. |
| 17:44.52 | watchy | i accept the "can" but we aren't that large of a corp i doubt anyone would care |
| 17:45.15 | watchy | im in camden arkansas and we make explosives |
| 17:45.21 | watchy | i doubt the riaa is gonna call me |
| 17:45.22 | Qwell | watchy: Let me put it this way. It would cost them *very little* to draft a C&D, and squeeze you for $50k. |
| 17:45.42 | malcolmd | opsound.org, creative commons licensed music that can be used for MoH |
| 17:45.55 | coppice | watchy: they sue little old ladies. why wouldn't they sue you? |
| 17:46.11 | watchy | they sue little old ladies whos grandsons torrented |
| 17:46.18 | watchy | they had a way to easily catch the old lady |
| 17:46.57 | watchy | im not saying im going to do it. its probably more trouble then its worth than to just get free songs at opsound.org |
| 17:47.11 | p3nguin | Calling you and listening to your moh seems like a pretty easy way to gather evidence. |
| 17:47.12 | watchy | but if the riaa is cold calling companies to listen to moh |
| 17:47.15 | watchy | they got some isues |
| 17:47.25 | Qwell | watchy: It would be very profitable to do so. |
| 17:47.30 | _Corey_ | watchy: Don't be surprised if they do it |
| 17:47.36 | watchy | maybe ill start a company doing that |
| 17:47.49 | Qwell | Pay somebody $40k/year to call random companies and listen to MoH for a minute. |
| 17:48.00 | Qwell | How many companies do you think that person could call? how much do you think their return would be? |
| 17:48.02 | watchy | "oh thats our song, lets sue" |
| 17:48.04 | Qwell | 100x? |
| 17:48.09 | watchy | probably |
| 17:48.13 | _Corey_ | If Shazam can idenfity any song in a few seconds, how hard do you think it would really be? |
| 17:48.13 | din3sh | p3nguin: can you take a look at this and tell me what's wrong in it: http://paste2.org/p/1983125 |
| 17:48.20 | din3sh | plz |
| 17:48.43 | watchy | qwell: what would you recommend for provision polys? ftp/ftps or http/https? |
| 17:48.49 | FLeiXiuS | leifmadsen, I cant seem to find a way to change the path the recording is stored in. |
| 17:48.54 | jrondeau | does anyone know is it possible to see the actual cause code, be it SIP or Q931 result from a failed call attempt issued via the Originate command using the Management Interface? |
| 17:49.15 | leifmadsen | watchy: either of the 's' methods is fine |
| 17:49.18 | p3nguin | din3sh: Does it work when you change to the Polycom phone? |
| 17:49.36 | din3sh | p3nguin: doesnt work |
| 17:49.41 | watchy | how do you recieve logs with https? |
| 17:49.41 | leifmadsen | FLeiXiuS: you'll have to check the documentation as I have to write a document now and can't look it up for you |
| 17:49.53 | din3sh | i hooked the same gateway back to 1.4, works fine |
| 17:50.02 | Qwell | FLeiXiuS: How are you recording? |
| 17:50.25 | WIMPy | jrondeau: You get the hangupcause. But for SIP there is no way to get the original result AFAIK. |
| 17:50.47 | jrondeau | the hang up cause is alway 16 normally clearing |
| 17:50.57 | watchy | anyone know if the sergio guy who coded app_rtsp hangs in here? |
| 17:50.58 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 17:50.59 | jrondeau | regardless of what actually happened |
| 17:51.18 | WIMPy | jrondeau: Using SIP? |
| 17:51.31 | jrondeau | yes mostly, but in some cases PSTN |
| 17:51.43 | *** join/#asterisk dijib (~dijib@bas10-kitchener06-1279682024.dsl.bell.ca) |
| 17:52.03 | WIMPy | jrondeau: If you call out ISDN it should be correct. |
| 17:52.29 | din3sh | watchy: /var/log/httpd check access logs |
| 17:52.32 | jrondeau | what about sip? is there no way to send the result code back though the AMI? |
| 17:52.56 | WIMPy | jrondeau: Not that I know. |
| 17:53.20 | watchy | din3sh: thanks |
| 17:53.53 | jrondeau | i am new to the source code, but how hard do you think it would be to send in the action id from the AMI to the sip channel structure and then just fire a seperate event at the end of the call containing the result code and the action id |
| 17:55.07 | din3sh | watchy: by default polycoms i think requests file on ftp |
| 17:55.13 | WIMPy | jrondeau: I'm not that in to the internals of chan_sip. I think it only translates the SIP results to causes. |
| 17:55.36 | watchy | din: yea they do i'm looking at doing a new system here at work trying to go with the best. currently i use ftp |
| 17:55.42 | watchy | im sure i should move to ftps |
| 17:56.43 | din3sh | you have to allow dhcp option 160 |
| 17:56.53 | watchy | anyone prov polys on 1 vlan and make them jump to another for service? |
| 17:57.00 | din3sh | for the phones to ask config files via ftp |
| 17:57.09 | watchy | ftp or ftps? |
| 17:57.10 | FLeiXiuS | Qwell, Monitor. http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
| 17:57.24 | watchy | im thinking ftps |
| 17:57.33 | din3sh | both would use same method i'd guess |
| 17:57.38 | watchy | yea i think so |
| 17:57.47 | watchy | im about to setup a lab at work to play with it |
| 17:57.54 | watchy | so i don't have to touch my production stuff |
| 17:59.04 | din3sh | of course |
| 17:59.15 | *** join/#asterisk Foxi352 (~Foxi352@v-172-4.access.restena.lu) |
| 17:59.22 | watchy | i got about 200 phones combined at 3 locations |
| 17:59.30 | watchy | about 50 of those are ATAs |
| 17:59.46 | watchy | and on top of that i have about 50 Cyberdata paging things |
| 18:00.56 | watchy | i wish i could get app_rtsp working with my cameras i have. i could do neat things |
| 18:01.51 | brian98 | [TK]D-Fender: the SIP_HEADER function is perfect. I'm using CUT \;,4 and that gets me to privacy=yes/no |
| 18:02.11 | brian98 | I'll do some more work on it , thanks for pointing me in right direction. |
| 18:05.22 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 18:05.44 | CyBeRxIxO | im stuck on 4.3 digium codec install |
| 18:05.52 | CyBeRxIxO | cant load or reload codec g729.so |
| 18:05.58 | CyBeRxIxO | any idea? |
| 18:12.17 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:13.46 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:16.40 | *** join/#asterisk rgsteele (~rgsteele@pool-108-36-107-86.phlapa.fios.verizon.net) |
| 18:20.38 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 18:23.51 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:30.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:30.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:42.37 | brian98 | If I want to check if a variable=yes and set callerID='' how can I do that? I can't quite figure the if statement in asterisk dialplan. |
| 18:43.13 | din3sh | any documentation how to read/interprete PRI debug logs? |
| 18:43.43 | brian98 | IF($CLI='YES'?[CallerID(num)=''][:]) ? |
| 18:43.44 | WIMPy | din3sh: Q.931 |
| 18:43.54 | WIMPy | Or just read the text :-) |
| 18:44.02 | din3sh | :D |
| 18:44.02 | [TK]D-Fender | $["${A}"="abc" || ${BA}"="123"] |
| 18:44.17 | [TK]D-Fender | brian98, go read channelvariables.text in your tarball |
| 18:44.22 | brian98 | ok |
| 18:44.24 | brian98 | sorry. |
| 18:44.30 | [TK]D-Fender | (sample above) |
| 18:44.34 | brian98 | thanks |
| 18:44.48 | din3sh | thnks WIMPy |
| 18:45.34 | p3nguin | brian98: You could use Set() and IF() or you could use ExecIf() and Set(). |
| 18:45.38 | *** join/#asterisk samandiriel (~samandiri@96.53.116.78) |
| 18:45.59 | brian98 | Thank you. |
| 18:46.28 | brian98 | I'll get there. New to asterisk dial plans.. |
| 18:46.33 | brian98 | thanks again |
| 18:46.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:47.49 | p3nguin | example: Set(CALLERID(num)=${IF($[${myVar}]?123:321)}) |
| 18:48.42 | p3nguin | example: ExecIf($[${myVar}]?Set(CALLERID(num)=123)) |
| 18:49.43 | p3nguin | Both methods check that myVar exists. |
| 18:49.54 | bchia | brian98 note that IF in Asterisk Dialplan is a "Dialplan function" and behaves differently than other languages |
| 18:49.55 | p3nguin | You could also compare it to the literal yes. |
| 18:50.06 | bchia | run "core show function IF" on the asterisk CLI for more info |
| 18:50.10 | brian98 | ok |
| 18:50.40 | brian98 | thanks guys. I'll hammer on! :) |
| 18:50.59 | WIMPy | It's like in the shell where the task is split between if an test, here it's IF or GotoIf and $[]. |
| 18:51.22 | p3nguin | s/GotoIf/ExecIf/ |
| 18:51.37 | WIMPy | Or that. |
| 18:51.40 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:52.27 | samandiriel | Need a little help getting a client to connect to a new Asterisk install (my second install, so not a complete noob). Keep getting "No matching peer found" message in CLI when trying to connect softphone... can't figure out why :( Troubleshooting suggestions? |
| 18:53.04 | talntid | paste your sip.conf |
| 18:53.07 | talntid | in a pastebin |
| 18:53.20 | samandiriel | talntid: will do, one mo |
| 18:53.21 | talntid | and paste the actual output of the CLI |
| 18:54.54 | samandiriel | talntid: http://pastebin.com/sHRUFBwF |
| 18:55.55 | samandiriel | [Apr 17 12:52:09] NOTICE[1567]: chan_sip.c:24929 handle_request_register: Registration from '"SPN Test" <sip:1000@freepbx.steelplus.com>' failed for '96.53.116.78:5060' - No matching peer found |
| 18:56.00 | talntid | ahh, we don't really support freePBX here.. |
| 18:56.04 | talntid | there is a #freepbx channel |
| 18:56.12 | samandiriel | Yah, I know-thought I'd try it out and see what it was like |
| 18:56.25 | watchy | its newb friendly |
| 18:56.26 | talntid | if you edit the file manually, it'll rewrite it on reload |
| 18:56.57 | samandiriel | thinking it's more trouble than it's worth, but I am not going to be maintaining it so needs to be pretty pretty |
| 18:57.18 | samandiriel | yes, but if I can figure out what's wrong with the basic config, I should be able to correct it via the GUI |
| 18:57.51 | samandiriel | there are also include files that one can edit manually that don't get overwritten |
| 18:58.04 | samandiriel | shit, I just posted my secret |
| 18:58.05 | talntid | yeah, i'm quite familiar |
| 18:58.07 | samandiriel | why I am so stupid? |
| 18:58.11 | talntid | I'm checking it out |
| 18:58.30 | talntid | one thing, not sure if it has anything to do with it.. that is interesting to me is the.. deny=0.0.0.0/0.0.0.0 |
| 18:59.15 | brian98 | p3nguin: is this wrong? exten=> _[0-9].,n,ExecIf($[${CLI}='no']?Set(CALLERID(num)='123')) |
| 18:59.33 | samandiriel | talntid: I checked that out - apparently it checks in order, so the permit at the end basically cancels it out |
| 18:59.40 | talntid | ah, yeah |
| 19:00.14 | samandiriel | driving me mad that I can't figure out why the client can't connect... should be easy-peasy |
| 19:00.48 | talntid | in the phone change "SPN Test" to 1000 |
| 19:00.50 | talntid | as a test. |
| 19:01.05 | samandiriel | sip show users also doesn't show any rows, which is odd to me |
| 19:02.05 | samandiriel | [Apr 17 13:01:55] NOTICE[1567]: chan_sip.c:24929 handle_request_register: Registration from '"TEST" <sip:1000@freepbx.steelplus.com>' failed for '96.53.116.78:5060' - No matching peer found |
| 19:04.28 | watchy | did you save your config |
| 19:04.56 | watchy | he said change spn test to 1000 |
| 19:04.58 | watchy | not test |
| 19:05.31 | samandiriel | ah, whups. thanks watchy |
| 19:06.36 | samandiriel | [Apr 17 13:06:30] NOTICE[1567]: chan_sip.c:24929 handle_request_register: Registration from '"1000" <sip:1000@freepbx.steelplus.com>' failed for '96.53.116.78:5060' - No matching peer found |
| 19:06.45 | samandiriel | not too surprising, I didn't think that would be it |
| 19:06.50 | watchy | you sure your saving your config in freepbx |
| 19:06.57 | watchy | just cause you add as peer doesnt save it |
| 19:07.03 | brian98 | exten=> _[0-9].,n,ExecIf($[${CLI} = no]?Set(CALLERID(num)='123')) Worked - THANKS guys! |
| 19:07.09 | watchy | you gotta goto the top right i think and click apply settings or something |
| 19:07.43 | *** join/#asterisk classix (~classix@silenceisdefeat.com) |
| 19:07.56 | samandiriel | watchy: there's nothing to save yet |
| 19:08.04 | talntid | yeah, nothing to save |
| 19:08.20 | [TK]D-Fender | brian98, Do not put quotes around 123 |
| 19:08.30 | brian98 | ok |
| 19:08.44 | brian98 | I am actually making it blank so don't do '' ? |
| 19:08.47 | [TK]D-Fender | brian98, and do put them around the 2 parts of your expression |
| 19:08.57 | talntid | i'm not sure, samandiriel. I'm all about plain ole *.. I know freepbx is at the core, but there's a lot of "noise" in that config.. the allows, denies, etc.. |
| 19:09.12 | talntid | I'd rip those out, sip reload, and see what happens |
| 19:09.28 | samandiriel | drat. now I have to run - thanks for looking at it a bit talntid, I'll try just building it from scratch later then and seeing if it will work that way |
| 19:09.48 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 19:09.59 | watchy | tk: have you played with ipv6 in *? |
| 19:10.18 | talntid | good luck, samandiriel |
| 19:10.19 | talntid | :) |
| 19:10.47 | [TK]D-Fender | watchy, Nope |
| 19:10.47 | watchy | i wonder if going ipv6 with a install would work without issues |
| 19:14.23 | *** join/#asterisk classix (~classix@silenceisdefeat.com) |
| 19:16.30 | watchy | i'm trying to decide if its easier to just edit ast config files manually or go with realtime. whats your opinion tk? |
| 19:17.34 | [TK]D-Fender | manual file = instant access |
| 19:18.08 | [TK]D-Fender | I never touch DB's without a solid reason to |
| 19:18.21 | watchy | yea but harder to write a interface for it right? |
| 19:18.52 | watchy | for adding devices etc |
| 19:20.11 | *** join/#asterisk irule (~irule@189.161.190.154) |
| 19:21.11 | irule | hi, how may I get started with asterisk? I am done setting up an spa3102, trunk and line1 register, but I geta busy signal and cant figure out how to make calls |
| 19:25.31 | *** join/#asterisk jsjc (~Adium@103.Red-2-136-100.dynamicIP.rima-tde.net) |
| 19:28.10 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 19:29.18 | *** join/#asterisk TheMan (~garry@66.237.29.132.ptr.us.xo.net) |
| 19:29.39 | woleium | lo peeps :-). I'm using Asterisk 1.8.7.1 on Centos (piaf). I've had complaints from users about unequal volume in conference calls, specifically that external callers coming in over POTS find it hard to hear other POTS callers, but can hear local SIP users fine. local SIP users can hear everyone fine. |
| 19:29.50 | woleium | Is there some kind of auto gain control I can enable in conferences? |
| 19:30.23 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:31.04 | woleium | Reading back I suspect it's probably a DAHDI win issue.. I remember reading about how to set that somewhere.. |
| 19:31.07 | *** join/#asterisk como|work (~como@66.186.188.78) |
| 19:31.25 | woleium | s/win/gain/ (autocorrect sux) |
| 19:31.45 | como|work | are there any big setbacks when trying to run an asterisk box within a virtual environment, like xenserver? |
| 19:31.46 | talntid | but VIM doesn't |
| 19:31.57 | talntid | como|work, some people will say yes... |
| 19:32.02 | talntid | but I have been doing it for 5 years, on Xen |
| 19:32.05 | talntid | and I love it. |
| 19:32.08 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 19:32.11 | talntid | I also have one on VMWare, and no issues there either |
| 19:32.16 | como|work | Mmm, good to hear. |
| 19:32.18 | woleium | como|work: there have traditionally been issues of jitter and lag, but it's supposed to be better now |
| 19:32.31 | jrondeau | i have several on vmware |
| 19:32.38 | como|work | I'm told there is a virtual/software "timing card" that needs to be configured that eats up a few more cpu cycles |
| 19:32.50 | talntid | IF you want to run meetme |
| 19:33.48 | [TK]D-Fender | <PROTECTED> |
| 19:33.49 | CyBeRxIxO | any asterisknow support? |
| 19:34.00 | [TK]D-Fender | And timing has nothing to do with gains, nor does VM |
| 19:34.02 | brian98 | Are there other functions in asterisk for cutting up variables? If I use cut I need to be sure that it's always the same amount of fields. How would I search for a particular word in a variable. For example how would I look for the yes of screen=yes in remote party id header |
| 19:34.03 | como|work | My biggest worry is disk i/o. I'm not really too in tune with asterisk since I've not maintained a box for it and more than 5 users in a few years, so I dont know what kind of i/o subsystem to be looking into |
| 19:34.12 | woleium | Obviously you have to be more careful about network utilisation - if another server spikes all your IO you may loose voice packets |
| 19:34.17 | brian98 | I am currently using cut 4 times on , then cut on=1 to get the yes or no. |
| 19:34.31 | brian98 | but that assumes that all ua's are sending every field in rpid... |
| 19:35.02 | [TK]D-Fender | brian98, Complicate looping, etc required... or do it in some external language via AGI, etc |
| 19:35.24 | woleium | brian98: grep -e ftw! |
| 19:35.28 | brian98 | I was thinking to loop until I find the privacy word then look for the next |
| 19:35.38 | brian98 | grep- e FTW indeed |
| 19:35.46 | *** join/#asterisk classix (~classix@silenceisdefeat.com) |
| 19:37.05 | woleium | grep -eo to be specific :-) |
| 19:37.20 | brian98 | or maybe I should tell anyone using RPID to feck off it was never an RFC and we will not honor it. |
| 19:37.40 | woleium | lol |
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| 20:09.10 | CyBeRxIxO | im stuck on 4.3 digium g729 codec installation |
| 20:09.27 | CyBeRxIxO | i can't load or reload g729 codec |
| 20:11.12 | leifmadsen | CyBeRxIxO: that is a commercial product of Digium so you should contact them for support |
| 20:11.37 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 20:19.41 | CyBeRxIxO | how do i fix the perl issue of "dahdi" |
| 20:19.49 | dwayne | leifmadsen, have any good linux softphone recommendations? |
| 20:20.00 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 20:20.08 | [TK]D-Fender | checkout time, BBIAB |
| 20:20.15 | leifmadsen | dwayne: yes |
| 20:20.30 | dwayne | what do you like ? :-) |
| 20:20.37 | leifmadsen | dwayne: I like Zoiper |
| 20:21.05 | dwayne | leifmadsen, thanks buddy |
| 20:21.17 | CyBeRxIxO | "Use of ininitialized value in concatenation(.)... and Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3360 |
| 20:21.30 | leifmadsen | that sounds like you're using a GUI |
| 20:21.59 | CyBeRxIxO | im using asterisknow |
| 20:22.06 | leifmadsen | see #asterisknow |
| 20:22.07 | raden | anyone know how to get IOS to work with asterisk 1 to 1 static nat ? |
| 20:22.10 | CyBeRxIxO | istalled twice and upgraded, same error |
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| 20:45.05 | irule | using asteirisk/spa3102, do I need to setup the outbound proxy? |
| 20:48.18 | irule | show sip peers sanys status UNREACHABLE, what does that mean? |
| 20:48.53 | *** join/#asterisk itstriz (~triz@216.55.8.194) |
| 20:50.35 | p3nguin | It means the device is unregistered, possibly because of a network or other unrelated problem. |
| 20:50.54 | p3nguin | It also means that you have qualify enabled for that peer. |
| 20:58.10 | *** join/#asterisk krotos (~d3v1l@host158-27-dynamic.8-87-r.retail.telecomitalia.it) |
| 20:58.23 | krotos | hi all guy |
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| 21:15.24 | irule | hi :s |
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| 21:15.54 | bluregard | good afternoon everyone |
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| 21:19.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 21:20.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 21:39.21 | FLeiXiuS | If I Set(AGC=(rx)16000) then use a GOTO afterwards, does that propagate down |
| 21:39.32 | Qwell | FLeiXiuS: yes |
| 21:39.37 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:39.41 | Qwell | Goto doesn't create a new channel. |
| 21:40.11 | FLeiXiuS | Excellent. |
| 21:40.21 | [TK]D-Fender | And * variable have no sense of scope |
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| 21:43.35 | p3nguin | fleixius: But your variable it will not necessarily be inherited by new channels that are spawned from the channel where you are setting the value. |
| 21:44.18 | FLeiXiuS | p3nguin, So a MeetMe conference would be instantiating a new channel |
| 21:44.29 | p3nguin | no |
| 21:45.04 | FLeiXiuS | ok so AGC should follow through a goto and in my meetme conf room. |
| 21:45.51 | [TK]D-Fender | FLeiXiuS: first your Set is wrong. Second... that app is news to me .... but if it is what it looks like it also only acts on that calling channel. |
| 21:46.48 | FLeiXiuS | [TK]D-Fender, My Set is wrong!? Bah its not complaining... |
| 21:47.15 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 21:47.34 | [TK]D-Fender | because you're setting a non-relevant channel variable, and not setting a function |
| 21:47.38 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:48.34 | p3nguin | You said Set(AGC=(rx)16000) but would be Set(AGC(rx)=16000) |
| 21:48.47 | FLeiXiuS | OH woops - yeah that was my bad. |
| 21:49.10 | FLeiXiuS | Set(AGC(rx)=${value}) is what I have. Should work. |
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| 21:50.32 | p3nguin | What's your purpose for using AGC()? |
| 21:51.05 | FLeiXiuS | p3nguin, Need to increase overall gain on the a SIP channel (speex codec) |
| 21:51.17 | p3nguin | Maybe you want to use VOLUME(), then. |
| 21:52.18 | p3nguin | It means amplitude, not volumetric capacity. |
| 21:55.00 | FLeiXiuS | p3nguin, Interesting - all of my search led me to AGC. Do you know the range of values volume can accept? |
| 21:55.38 | p3nguin | It's going to be able to increase or decrease by magnitudes, considering the value it takes is in dB. |
| 21:55.53 | p3nguin | So 3 is going to be double the amplitude. |
| 21:56.03 | p3nguin | 6 would be four times as loud. |
| 21:56.34 | p3nguin | Give it a 24 and you might pop a speaker. |
| 21:57.14 | p3nguin | And, as I have recently learned, you can apply the value to the existing channel and make a change in real time by using AMI. |
| 21:57.31 | p3nguin | Great for testing a change on-the-fly. |
| 21:58.09 | [TK]D-Fender | 6db = double.... |
| 21:58.21 | FLeiXiuS | Yeah I have an entire AMI interface. I'll definitely integrate into it. |
| 21:58.32 | [TK]D-Fender | 3db = significant perceived increase |
| 21:58.32 | p3nguin | If 0 is 100%, isn't 3 dB going to be 200%? |
| 21:59.20 | p3nguin | Perhaps I am trying to apply decibels improperly. |
| 21:59.27 | p3nguin | I don't think so, but it's possible. |
| 22:03.28 | FLeiXiuS | I assume I can do float values as well? 1.5, 2.0 etc |
| 22:03.59 | p3nguin | The help doesn't indicate one way or the other, but I'd imagine it would work. |
| 22:04.26 | p3nguin | I'm not sure why you'd need to, since half of a decibel isn't much change. |
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| 22:04.56 | FLeiXiuS | p3nguin, I'm offering a 'slider' for a range value. |
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| 22:05.18 | p3nguin | I guess at a higher value, .5 would provide considerable change. |
| 22:05.28 | p3nguin | 100.0 vs 100.5 |
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| 22:12.38 | raden | <PROTECTED> |
| 22:12.50 | raden | i have this coming up on console literally every 5 seconds |
| 22:12.55 | raden | how do i make it not do that ? |
| 22:13.15 | p3nguin | Turn down the verbose level below 4. |
| 22:16.02 | raden | why does it have to keep looking it up |
| 22:16.08 | raden | kinda ridiculous |
| 22:16.16 | raden | can i slow it down |
| 22:17.16 | p3nguin | Change the value in dnsmgr.conf. |
| 22:17.28 | raden | k... |
| 22:20.05 | raden | p3nguin, ok we will try that |
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| 22:25.00 | raden | ok that helps |
| 22:27.20 | p3nguin | Were you still seeing the lookups at verbose 3 and lower? |
| 22:27.28 | raden | no i wasnt |
| 22:27.50 | raden | just found it ridiculous that it was looking up soo often |
| 22:27.56 | raden | set dns manager on and set to 1200 |
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| 22:48.15 | p3nguin | If you wouldn't have been running above 3, you would have never known about it. |
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| 23:49.12 | raden | [Apr 17 18:47:37] WARNING[7120]: chan_sip.c:3714 __sip_xmit: sip_xmit of |
| 23:49.13 | raden | 0x184cff0 (len 1148) to 64.2.*.*:0 returned |
| 23:49.13 | raden | <PROTECTED> |
| 23:49.16 | raden | anyone ? |
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| 23:57.31 | raden | is there a way to define the port of a clients phone ? |
| 23:58.24 | p3nguin | The client port is configured in the phone and in the peer definition. |
| 23:58.50 | p3nguin | port=5061; for example |
