| 00:00.28 | alami | i'm confused, should i install asterisknow or elastix or just Asterisk on debian |
| 00:12.49 | alami | can some one help me to get a decision |
| 00:14.16 | Penguin | Do you prefer Debian or CentOS? |
| 00:14.29 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 00:17.15 | alami | Penguin: i have never use CentOS, but i know debian good |
| 00:17.37 | Penguin | Then you should probably install your favorite Debian and install asterisk on that. |
| 00:17.55 | Penguin | AsteriskNOW is CentOS. |
| 00:19.19 | alami | Penguin: yes that's the best idea i thing |
| 00:19.42 | alami | i thing i can install freepbx or elastix as GUI if i need |
| 00:20.02 | Penguin | That would be the worst thing you could do. |
| 00:20.34 | alami | Penguin: rearly? |
| 00:20.37 | alami | why? |
| 00:21.11 | alami | Penguin: you don't like GUI's? |
| 00:21.38 | Penguin | They obscure the actual system and waste my time. |
| 00:21.43 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:21.50 | tm1000 | alami: and he's a hater |
| 00:21.56 | tm1000 | and always says that |
| 00:22.10 | Penguin | Find one time I have ever said that. |
| 00:22.19 | alami | lol :) |
| 00:22.30 | tm1000 | Penguin: you always say the same thing, that you hate freepbx because "They obscure the actual system and waste my time." |
| 00:22.32 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 00:22.47 | tm1000 | but only about freepbx |
| 00:22.49 | tm1000 | anyways |
| 00:22.52 | WIMPy | There might be a reason. |
| 00:22.55 | Penguin | And I'm saying find ONE TIME that I ever said it before today. |
| 00:23.06 | navaismo | vomit and eat it again |
| 00:23.09 | tm1000 | Penguin: that would be a waste of my time, but yes you have said it |
| 00:23.09 | WIMPy | It's the same for any GUI. |
| 00:23.19 | tm1000 | alami: here's the REAL answer |
| 00:23.22 | navaismo | please stop using e***x word |
| 00:23.26 | Penguin | I'm only asking for one time. |
| 00:23.30 | tm1000 | elastix uses freepbx at this moment |
| 00:23.40 | tm1000 | asterisknow (the gui part) also uses freepbx |
| 00:23.43 | navaismo | *old* |
| 00:23.49 | tm1000 | asterisk by itself has no gui |
| 00:24.04 | tm1000 | so if you want to learn about asterisk then install asterisk |
| 00:24.10 | tm1000 | if you dont want to learn asterisk |
| 00:24.14 | tm1000 | then install a gui |
| 00:25.02 | alami | so i thing no need of gui if i can do lern and do everything without the gui |
| 00:25.08 | tm1000 | exactly |
| 00:26.30 | alami | but also the idea of install a gui after asterisk is not a good ideaß? |
| 00:27.14 | WIMPy | The point in time doesn't matter. |
| 00:27.29 | file | nothing is perfect for everyone. |
| 00:27.31 | tm1000 | alami: its up to you and what you want to accomplish |
| 00:27.46 | alami | because it obscure the actual system |
| 00:28.07 | alami | i want to make call conference with confme() |
| 00:28.27 | tm1000 | confme? |
| 00:28.32 | tm1000 | meetme? confbridge? |
| 00:28.35 | Penguin | meatbridge() |
| 00:28.45 | WIMPy | Is that something that existes somewhere or somethin you want to invent? |
| 00:28.47 | alami | but i can a little bit asterisk and i want to know some idea from profissional people |
| 00:29.03 | alami | yes tm1000: soory i mean meetme |
| 00:29.33 | alami | WIMPy: that exist already |
| 00:29.53 | tm1000 | alami: so you dont want to do any dialplan? |
| 00:30.07 | tm1000 | I realize there is a language barrier here but we are trying to figure out what you want to accomplish |
| 00:31.01 | alami | tm1000: i can write more |
| 00:31.03 | alami | :) |
| 00:31.20 | alami | but i don't know a lot about asterisk |
| 00:31.45 | alami | i have just install it more time, and make comunication between two hard or softphone with sip as protocol |
| 00:32.23 | alami | and now i want to make asterisk for conference between analog phone, or also voipphone |
| 00:32.30 | alami | tm1000:if it possible |
| 00:32.41 | alami | do you understand me now better? |
| 00:34.37 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 00:35.29 | tm1000 | alami: It's up to you. If you want to do it by hand then you are in the right channel. If you want something graphical then you are not in the right channel |
| 00:35.47 | tm1000 | that is what Penguin and WIMPy imply usually |
| 00:36.21 | tm1000 | guis are guis, the take everything and generalize it. if you want something specifically done that works only for you, then a gui usually wont cut it |
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| 00:36.35 | tm1000 | if you can work inside of cookie cutter standards then a gui will possibly work |
| 00:37.02 | file | mmm... cookies |
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| 03:56.54 | linocisco | hi all |
| 03:57.18 | linocisco | is there any asterisk based video intercom door lock system? |
| 03:58.15 | [TK]D-Fender | Pretty much 100% sure that's a no |
| 03:58.40 | [TK]D-Fender | Why would an intercom door lock .... be running Asterisk? |
| 04:01.58 | ChannelZ | so we can try to call Palestine through it |
| 04:02.10 | linocisco | [TK]D-Fender, the idea is elevator or guest user will type in keypad at elevator door, that will act as one extension to the desired apartment and through webcam, host can see who the vistor is and can unlock elevator door. |
| 04:03.15 | [TK]D-Fender | linocisco: the phone is not "asterisk based" You had a large misuse of words there. |
| 04:03.37 | [TK]D-Fender | linocisco: Anyway, there are SIP videophones out there all over the place |
| 04:05.07 | linocisco | [TK]D-Fender, door unlock will be used by using different unlock switch. asterisk is just for video intercom , between elevator door and room apartment |
| 04:05.21 | [TK]D-Fender | Clearly |
| 04:05.36 | [TK]D-Fender | Which is a completely separate solution for you to pick |
| 04:07.41 | linocisco | [TK]D-Fender, security lock/unlock will be different system , apartment users can have control over it. but video phone with web cam will be installed at elevator door to make phone call to room number. where there is video phone to see who is calling before answering or unlocking the elevator from inside |
| 04:08.41 | linocisco | [TK]D-Fender, the thing is i dont want to use complete proprietary home appliance system. I am thinking from opensource side if there is such app or setup |
| 04:09.11 | [TK]D-Fender | Youa re talking "open source" ... before finding the HARDWARE |
| 04:09.21 | [TK]D-Fender | Do not assume the software side first |
| 04:11.09 | linocisco | [TK]D-Fender, yes. if I buy proprietary system, there are some I have n't check. They would probably sell the whole package where we can't use asterisk |
| 04:11.46 | [TK]D-Fender | And how hard have you looked? |
| 04:12.23 | linocisco | [TK]D-Fender, not yet |
| 04:13.02 | [TK]D-Fender | linocisco: Then don't say "probably". Zero market research = zero right to assumptions.... |
| 04:15.14 | linocisco | [TK]D-Fender, i found some china and thai products. that are complete solutions. we can't buy separately |
| 04:16.06 | [TK]D-Fender | Keep shopping |
| 04:17.16 | linocisco | [TK]D-Fender, the reason i am asking here is there might be someone who had experienced this kind of setup. and want to know how. If noone knows, thanks anyway |
| 04:17.18 | [TK]D-Fender | I just found a ton |
| 04:17.41 | linocisco | [TK]D-Fender, tons of what, can we integrate with asterisk? |
| 04:18.00 | [TK]D-Fender | How do you "unlock" the elevator? |
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| 06:22.31 | jameswf | Locking an elevator seems like something that would piss off the fire Marshall |
| 06:30.20 | *** join/#asterisk jzu_ (jussi@alppi.unikko.org) |
| 06:30.25 | jzu_ | Yo |
| 06:30.44 | jzu_ | Any of you running RasPBX? |
| 06:30.55 | jzu_ | I'm thinking of building in-house phone trunk using it |
| 06:30.57 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 06:31.17 | jzu_ | with few Cisco IP phones on SIP firmware |
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| 07:33.08 | linocisco | jameswf, I dont lock or unlock using asterisk. it would be done by other devices. My idea of asterisk deployment is just video calling between elevator door and the room |
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| 08:30.19 | alberte | Hi All - I need some assistance setting the codec preference |
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| 09:12.12 | jaflong | Hi, Any ideas why the a=ice-ufrag, a=ice-pwd in the sdp does not get sent back to the client |
| 09:13.04 | kemmler | wat |
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| 09:54.16 | alberte | Jaflong -> what are the ice-ufrag attributes used for? |
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| 10:03.39 | jaflong | <alberte> icesupport for webrtc |
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| 10:24.58 | Oggu | What kind of hardware would be needed for asterisk to handle 100 calls (200 persons) at the same time? |
| 10:25.05 | Oggu | With recording |
| 10:25.38 | kemmler | crappy hardware |
| 10:25.42 | morsing | Pentium 1? |
| 10:26.09 | kemmler | a raspberry pi |
| 10:26.29 | morsing | Raspberry Pi isn't crappy hardware :-P |
| 10:27.04 | kemmler | lol, i actually like it quite a bit |
| 10:30.23 | jzu_ | yeah I'm about to buy Rasp for running private trunk within my fathers house.. |
| 10:30.28 | jzu_ | he needs 'in house' phone system, Rasp is enuff' |
| 10:30.37 | jzu_ | Rasp + couple Cisco IP phones |
| 10:31.07 | kemmler | did you get the camera module on yours morsing |
| 10:31.50 | *** join/#asterisk zapphir (~zapphir@c-2ec3253f-74736162.cust.telenor.se) |
| 10:32.07 | morsing | kemmler: I don't have one :-( Don't have the time to play around like I used to |
| 10:33.48 | zapphir | Hi, I can't get device state to work for my sip software phone. I add the x-lite user to a queue, an incoming call gets placed into the queue and connected to the agent, but device state is still "Not in use". I have tried to set callcounter to yes in sip.conf and call-limit to 1 for the agent, but with no result. Does anyone have any tips? |
| 10:40.02 | Oggu | I read somewhere about performance problems with many calls in asterisk? (and deadlocks) Is that something of past versions? |
| 10:40.45 | kemmler | something will all versions if you shit calls at like my bowel movements after indian food |
| 10:43.13 | Oggu | Meaning? ;) |
| 10:44.44 | kemmler | asterisk has trouble with bringing up loads of channels at once, maintaining them once they're up is easier |
| 11:01.02 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 11:10.26 | Oggu | And channel equals connection to one persons "phone"? |
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| 11:19.38 | kemmler | connection between asterisk and something |
| 11:23.15 | eirirs | when are PoE powered version of Raspberry Pi coming... |
| 11:27.47 | jzu_ | is it easy to setup in-house phone system with Asterisk? :-) |
| 11:28.03 | jzu_ | ie. I would be only dialing to extension numbers, phones located within the same building and connected to the same Asterisk server |
| 11:28.50 | kemmler | si, grab some ethernet phones and off you go |
| 11:29.04 | jzu_ | well, I'll be having Cisco's for free so... |
| 11:29.14 | kemmler | should be a piece of cake |
| 11:29.18 | jzu_ | yep |
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| 11:44.03 | jzu_ | does RasPBX run on 256MB Raspberry Pi? |
| 11:44.13 | jzu_ | or would it require the 512MB model for TWO phones? :-D |
| 11:44.40 | kemmler | asterisk is on the rpi's debian source if you just want to use that |
| 11:44.47 | kemmler | it'll suppôrt two phones without a problem |
| 11:48.26 | kemmler | uses about 60Mb of memory to run |
| 11:49.29 | kemmler | don't know what kind of milage you'll get out of a model A, but i do know the model A doesn't have ethernet |
| 11:49.56 | jzu_ | it's model B with Ethernet |
| 11:50.00 | jzu_ | 256MB Model B |
| 11:50.09 | kemmler | 512Mb model B |
| 11:50.19 | jzu_ | well, model A then but WITH ethernet :P |
| 11:50.21 | jzu_ | the first release |
| 11:50.24 | jzu_ | nowdays model B is 512MB |
| 11:50.27 | kemmler | actually I think both are 512 now |
| 11:50.28 | jzu_ | it used to be 256MB |
| 11:51.07 | kemmler | model B is worth the extra $10 imo |
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| 11:53.38 | *** join/#asterisk alami (~alami@unaffiliated/alami) |
| 11:54.25 | alami | hello, i have just install asterisk 12.1.1 and i when i run rasterisk i don't have any sip command, what i'm dowing wrong here |
| 11:56.09 | *** join/#asterisk Pbxman (501a96d0@gateway/web/freenode/ip.80.26.150.208) |
| 11:56.11 | Pbxman | hello |
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| 12:00.43 | kemmler | hello |
| 12:02.55 | jzu_ | alami: quite damn hard to tell :O |
| 12:03.11 | jzu_ | alami: even though I haven't ever used Asterisk yet your description is not helpful at all |
| 12:03.19 | jzu_ | alami: we would need pastebin of logs etc etc |
| 12:06.15 | alami | jzu_: i have just want to try asterisk 12.1.1 and i have install it from source on debian VM |
| 12:07.07 | alami | after instllation i make some extentions on sip.conf and when i try to run show sip peers or sip reload at rasterisk |
| 12:07.51 | alami | No such command 'sip reload' (type 'core show help sip reload' for other possible commands) |
| 12:12.04 | snmp | module show like sip |
| 12:12.07 | [TK]D-Fender | alami: "ls -la /etc/asterisk", "cat /etc/asterisk/modules.conf", "cat /etc/asterisk/asterisk.conf" |
| 12:12.09 | [TK]D-Fender | ~pb |
| 12:12.09 | 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. |
| 12:12.12 | [TK]D-Fender | ^^^ |
| 12:16.58 | alami | http://pastebin.com/YmjmZjLg |
| 12:17.35 | alami | http://pastebin.com/xmUTfMNp |
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| 12:19.05 | alami | http://pastebin.com/3Gxz801N |
| 12:19.57 | snmp | *CLI> module show like sip |
| 12:20.10 | snmp | alami: |
| 12:21.06 | alami | Module Description Use Count Status |
| 12:21.09 | alami | app_adsiprog.so Asterisk ADSI Programming Application 0 Running |
| 12:21.12 | alami | 1 modules loaded |
| 12:21.48 | snmp | module load chan_sip.so |
| 12:24.32 | snmp | alami: |
| 12:24.39 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:29.40 | snmp | loooks like the problem solved, nevermind pastebins) |
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| 12:56.42 | gartral | oook.. I'm in a bind here.. there's a slight problem in my networking that I think boils down to a race condition in asterisk.. namely, I have a dual stack network.. and that's causing a lot of issues.. how do I force asterisk to use one or the other stack? (preferably IPv4) |
| 13:00.03 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 13:02.57 | Oggu | How would I go about in asterisk to listen to incoming calls on a SIP number and give them to one of the connected agents who are free? |
| 13:06.59 | zapphir | Hi, I can't get device state to work for my sip software phone. I add the x-lite user to a queue, an incoming call gets placed into the queue and connected to the agent, but device state is still "Not in use". I have tried to set callcounter to yes (and counteronpeer=yes) in sip.conf and call-limit to 1 for the agent, but with no result. Does anyone have any tips about this (I am using Asterisk |
| 13:06.59 | zapphir | 11.5.1)? |
| 13:07.29 | [TK]D-Fender | Oggu: "core show application queue" <- |
| 13:09.04 | [TK]D-Fender | zapphir: Show us your queue, and the status dump prior to a call going to the queue and hitting that device |
| 13:09.28 | [TK]D-Fender | Oggu: and read the sample queues.conf config |
| 13:10.27 | alami | snmp:sorry i wasn't here |
| 13:10.29 | Oggu | Ok. And how do I do the whole "listen to incoming calls on a sip account" thing (from our SIP provider) |
| 13:10.30 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 13:11.25 | alami | snmp: WARNING[3989]: loader.c:523 load_dynamic_module: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No such file or directory |
| 13:12.26 | [TK]D-Fender | Oggu: Set up a peer to match them, set up an exten to match what they'll dial towards you. |
| 13:12.29 | [TK]D-Fender | ~book |
| 13:12.30 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:12.31 | [TK]D-Fender | ^^^ |
| 13:13.27 | zapphir | D-Fender: Not sure if this is what you meant, but this is the status of the queue before and during an incoming call: http://pastebin.com/UkcqP5YE |
| 13:17.20 | alami | [TK]D-Fender: can you help please? |
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| 13:46.39 | alami | snmp? |
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| 13:50.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:50.49 | *** join/#asterisk kchehab (~kchehab@77.42.241.68) |
| 13:50.56 | kchehab | hi ppl |
| 13:51.33 | newtonr | howdy |
| 13:52.08 | kchehab | User A is calling user B through asterisk ,i there a way in asterisk to send a BYE to User B and let user A hear a music on hold after 60 seconds for example |
| 13:55.27 | newtonr | kchehab, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AMI+Actions Sure, use AMI |
| 13:56.14 | newtonr | kchehab, look at Park,Hangup,Redirect and others |
| 13:57.02 | newtonr | kchehab, there are sections in the book http://shop.oreilly.com/product/0636920025894.do on AMI as well |
| 13:58.11 | kchehab | newtonr thanks but can you help me more by showing me the code or the function that can send a bye for leg B and let Lag A up ,since i didnt match any thing related ,i |
| 13:58.30 | kchehab | it seems like meet me ,mute function ? |
| 14:00.31 | newtonr | kchehab, I already mentioned a few to you. Hangup specifically will result in a BYE being send to an established SIP channel |
| 14:02.31 | newtonr | kchehab, parking A or putting A into a MeetMe conference or a ConfBridge conference could result in Music on Hold depending on how you configure things. |
| 14:02.51 | kchehab | newtonr yes i got it thanks |
| 14:03.35 | newtonr | I don't build things with AMI much, so that is all I got |
| 14:03.41 | newtonr | np |
| 14:04.06 | kchehab | newtonr i was waundering if it can be bulit by code not my manager file |
| 14:05.08 | newtonr | kchehab, both really require code. I supposed you mean with dialplan (extensions.conf) ? |
| 14:06.24 | newtonr | kchehab, Maybe, what event do you want to trigger B getting hung up and A getting parked/held ? |
| 14:06.28 | kchehab | newtonr yes extensions.conf |
| 14:07.28 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-qoyxucmzhyzncmwx) |
| 14:08.12 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:12.27 | alami | newtonr: can you help please? No such command 'show sip peers' (type 'core show help show sip' for other possible commands) |
| 14:12.56 | alami | i have just i new installation of asterisk LTS 11.8.1 on a debian wheezy |
| 14:13.55 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:14.02 | newtonr | alami, it is "sip show peers" , not "show sip peers" |
| 14:14.30 | newtonr | alami, tab complete works on the Asterisk CLI, start with the first word of a command and hit tab a few times if you are unsure. |
| 14:15.01 | newtonr | newtonr, you can also do "core show help" to see a list of commands available |
| 14:15.10 | gartral | hey all, after upgrading to asterisk 12-r410949 my gvoice connection no longer works and my cli is flooded with http://pastebin.com/L4MYV1mk |
| 14:15.46 | newtonr | newtonr, you can search through those from the linux CLI by doing something like "asterisk -rx "core show help" | grep -i sip" |
| 14:15.48 | alami | okay thanks a lot, now because befor i had 12.1 vbersion and i have a lot of problem with it, so i switch to 11.8.1 and now everything work |
| 14:15.51 | alami | thanks |
| 14:16.09 | newtonr | lol referenced myself in both my last messages, i need tea |
| 14:16.28 | alami | lol yes man |
| 14:16.40 | Chainsaw | newtonr: We shan't interrupt you whilst you are talking to yourself. That would be rude. |
| 14:16.49 | newtonr | alami, 12 has a lot of development going on, but if you find bugs in it, make sure to file them on the tracker |
| 14:17.17 | newtonr | Chainsaw, thank you kindly |
| 14:18.19 | alami | @newtonr:ok sure |
| 14:18.51 | newtonr | and now i go off to triage stuff on the tracker |
| 14:19.03 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 14:21.28 | *** join/#asterisk Fwny (~potato@192-0-250-120.cpe.teksavvy.com) |
| 14:23.39 | gartral | seriously guys.. what's going on with my set up here >.< |
| 14:24.09 | *** join/#asterisk serafie (~erin@nat/digium/x-tchizuejhbujxknx) |
| 14:25.27 | gartral | i also keep getting this error occasionally.. res_xmpp.c:2674 xmpp_client_requested_tls: TLS connection for client 'google' cannot be established. OpenSSL initialization f |
| 14:25.34 | gartral | ailed. |
| 14:36.36 | Penguin | |
| 14:37.23 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
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| 14:39.01 | gartral | Penguin: ctrl-L? |
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| 14:44.10 | *** join/#asterisk serafie (~erin@nat/digium/x-hsurnswnrwsjldqw) |
| 14:44.55 | gartral | hey all, after upgrading to asterisk 12-r410949 my gvoice connection no longer works and my cli is flooded with http://pastebin.com/L4MYV1mk i've also downgraded to 11.8.1 and i'm seeing the exact same issue |
| 14:46.46 | Penguin | gartral: I don't really know what it is, but my computer sometimes prints that when this window was in focus and I open a new window or change to another desktop. |
| 14:47.14 | Penguin | I figured it was a quirk of the desktop environment. |
| 14:53.43 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
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| 15:12.24 | Katty | hands out cupcakes. |
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| 15:43.21 | Chainsaw | Those are the best kind of cakes :) |
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| 17:02.40 | saint_ | hi all - is there a reason why a phone setup with DPMA does not ask for the voicemail password, when I press the MSGS button ? |
| 17:04.29 | [TK]D-Fender | go look at your call. |
| 17:08.53 | saint_ | i think it is before the call |
| 17:09.07 | saint_ | right when I push the MSGS button, the list of voicemail is displayed on the D70 screen. |
| 17:09.14 | saint_ | way before any sip is involved. |
| 17:11.51 | saint_ | anyone using DPMA around here ? |
| 17:12.19 | [TK]D-Fender | I'm not seeing an option for PW vs none |
| 17:13.12 | saint_ | me neither. if I call the voicemail number (Voicemail(xxx@default)) from this user, it asks for the PWD |
| 17:13.45 | saint_ | if I call from outside, and get to VoiceMailMain(xxx@default), it asks for the PWD |
| 17:14.04 | saint_ | but if I push MSGS , no password .. is this considered as a bug or feature request ? |
| 17:16.51 | saint_ | i found this: http://forums.digium.com/viewtopic.php?p=178259 .. but I *want* to have visual voicemail. Just with a password, that is it. |
| 17:23.09 | *** join/#asterisk Vendigroth (18d59c71@gateway/web/freenode/ip.24.213.156.113) |
| 17:27.09 | *** join/#asterisk marceloamorim (bd5ac048@gateway/web/freenode/ip.189.90.192.72) |
| 17:27.42 | marceloamorim | hi guys, I`m having a problem with /var/spool/asterisk/outgoing |
| 17:28.51 | Vendigroth | Is there a way to make asterisk not play the ring tone when using Dial() during transfer? sip-pstn we use sends back 180 Ringing, and i don't want it to ring. |
| 17:29.27 | marceloamorim | I need to do a callback program and I`m doing great, but I need to call with application Dial and data, but just work if I had Channel: SIP/name I need replace this channel for a Queue |
| 17:29.36 | [TK]D-Fender | Vendigroth: Have your dial play MoH instead with an empty class |
| 17:29.52 | [TK]D-Fender | Vendigroth: And the only way to detect is if it is a blind transfer |
| 17:30.02 | *** join/#asterisk makubi (~makubi@xdsl-87-78-48-99.netcologne.de) |
| 17:31.08 | [TK]D-Fender | marceloamorim: Channel: is what gets called. |
| 17:32.59 | marceloamorim | [TK]D-Fender: but I can call back for the X number with dial application and call for a queue at the same time like Channel: SIP/name? |
| 17:33.37 | Vendigroth | [TK]D-Fender: thanks, might have to MoH, i just thought it was a bit hack-ish. and what do you mean to detect? detect what? |
| 17:33.50 | [TK]D-Fender | marceloamorim: HUH?> |
| 17:34.12 | [TK]D-Fender | Vendigroth: that e call is the result of a transfer |
| 17:34.13 | marceloamorim | because if I put channel calling for X PSTN-number and the application: Dial I call for the queue, the asterisk`ll call for PSTN-number first and then call for the queue |
| 17:34.14 | [TK]D-Fender | the* |
| 17:34.23 | marceloamorim | sorry about my english, isn`t my main language |
| 17:35.04 | [TK]D-Fender | marceloamorim: I don't see where this queue is coming in. You are talking about too many different things at once, and out of order |
| 17:35.55 | marceloamorim | How can I explain to you, give me a sec |
| 17:37.30 | marceloamorim | my outgoing file is X-pstn-number.call and he has Channel: SIP/name, MaxRetries: 1, RetryTime: 60, WaitTime: 60, Applicatoin:Dial, Data: SIP/X-pstn-number.call@ipgateway,60,Ttr |
| 17:38.36 | marceloamorim | it works pretty well, but I need to replace my Channel: SIP/name to Queue: queuenumber, because this way I can put all my callcenter to wait the call |
| 17:42.56 | *** join/#asterisk ghost75 (~quassel@dslb-092-075-060-075.pools.arcor-ip.net) |
| 17:44.55 | navaismo | use local channel |
| 17:45.09 | navaismo | pointing to the queue dialplan |
| 17:45.24 | navaismo | or use application Queue |
| 17:46.21 | marceloamorim | I`m already using application Dial, can I use two parameters to Application? |
| 17:46.46 | marceloamorim | do you know how can I poiting to the queue dialplan navaismo ? |
| 17:47.05 | marceloamorim | Any suggestion is helpfull |
| 17:47.38 | navaismo | local/8000@mycontext assuming you dial 8000 to get into the queue |
| 17:48.26 | marceloamorim | I`ll try, thx guys |
| 17:52.16 | *** join/#asterisk jibone (~jibone@hlr-223-144.tm.net.my) |
| 18:00.41 | marceloamorim | it works navaismo |
| 18:00.45 | marceloamorim | thx a lot guys |
| 18:00.52 | marceloamorim | save my day |
| 18:03.21 | *** part/#asterisk jibone (~jibone@hlr-223-144.tm.net.my) |
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| 18:16.01 | valeech | Can anyone point me in the right direction to get two Video SIP phones to negotiate their resolution higher than QCIF (176x144)? |
| 18:17.04 | valeech | I have max bitrate set to about 10Mbps and still no love. |
| 18:17.19 | rrittgarn | anyone ever have issues using the Voicemail app with multiple mailboxes? I have it working on one server but not on another. They are both asterisk 11.8.0 Specific syntax is Voicemail(101@Context&102@Context) - Yields the personal greeting for the first mailbox and thats as far as it goes, doesn't deliver mail to the subsequent VMs |
| 18:18.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:18.55 | navaismo | valeech with h263p i guess it should accept more resolution |
| 18:19.46 | valeech | navaismo: it should, I should get at least VGA (640x480), right? |
| 18:20.00 | navaismo | yes |
| 18:21.24 | file | valeech, what version of Asterisk? |
| 18:22.12 | *** join/#asterisk serafie (~erin@nat/digium/x-olzxrkdvncuxdcfw) |
| 18:22.40 | valeech | If use a jitsi client to jitsi client I send/recv 630x480. If I jitsi to a desk phone, I recv 640x480 but send 176x144 on the desk phone |
| 18:22.49 | valeech | file: asterisk 11.8.1 |
| 18:23.36 | file | hrm, provided res_format_attr_h263 is loaded the SDP attributes should get passed through |
| 18:24.48 | valeech | file: how can I check that that is loaded? |
| 18:25.18 | file | module show like h263 |
| 18:26.45 | valeech | ok, it says two modules are loaded: |
| 18:26.46 | valeech | localhost*CLI> module show like h263 |
| 18:26.47 | valeech | Module Description Use Count |
| 18:26.48 | valeech | format_h263.so Raw H.263 data 0 |
| 18:26.49 | valeech | res_format_attr_h263.so H.263 Format Attribute Module 0 |
| 18:26.49 | valeech | 2 modules loaded |
| 18:27.16 | valeech | same output for h264 modules |
| 18:27.31 | Katty | glares at asterisk-stat, apache, and php |
| 18:27.32 | file | what is the SIP traffic? |
| 18:27.55 | valeech | H264, |
| 18:27.58 | Katty | file: i hope it's beer going to my tummy |
| 18:29.41 | Katty | beats asterisk-stat repeatedly with something...heavy. |
| 18:30.17 | file | Katty, try a... cookie! |
| 18:30.21 | Katty | YES |
| 18:30.22 | Katty | a COOKIE |
| 18:30.33 | Katty | oh. that means i have to make cookies first :< |
| 18:32.50 | Katty | sadly, i don't comprehend why it's not working |
| 18:33.08 | Katty | i know that apache is displaying php correctly |
| 18:35.18 | Katty | cdr_pgsql.c:201 pgsql_log: Reason: fe_sendauth: no password supplied <- hmm. file. yes i need more cookies. |
| 18:38.21 | valeech | would setting directrtpsetup=yes in sip.conf help? |
| 18:39.21 | file | no |
| 18:40.04 | *** join/#asterisk serafie (~erin@nat/digium/x-fumeievppmbbxwib) |
| 18:40.17 | Katty | waves to serafie |
| 18:41.52 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 18:48.53 | serafie | waves at Katty |
| 18:49.06 | Katty | how're you dear? |
| 18:51.17 | jwr__ | anyone have recommendations for a softphone that runs on linux for testing my asterisk setup? |
| 18:52.18 | [TK]D-Fender | jitsi |
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| 19:09.00 | navaismo | linphone & zoiper |
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| 19:20.19 | jwr__ | [TK]D-Fender: jitsi works perfectly. thanks. |
| 19:20.29 | ghost75 | when a change is done in source, will this be detected if i run only make or do i need to clear anything before? |
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| 20:12.40 | EdwinGrubbs | ~itsplist-us |
| 20:12.40 | 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 |
| 20:25.34 | serafie | Katty: sorry, was in a meeting! |
| 20:25.59 | serafie | Doing very well. Working on a new project that has less to do with Asterisk, so haven't been active in a long time. |
| 20:26.37 | file | serafie was banished from the engineering floor even! |
| 20:26.39 | file | BANISHED! |
| 20:27.03 | serafie | It's true. It's all men up there again. |
| 20:27.04 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 20:28.22 | serafie | still coding thing. |
| 20:28.27 | serafie | *though |
| 20:28.33 | serafie | still coding a thing though. |
| 20:40.37 | Katty | can i bribe anyone to paste the contents of their pg_hba.conf? |
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| 21:13.05 | Nugget | what are you trying to sort out? |
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| 22:27.10 | Ice_Strike | WIMPy You around? |
| 22:32.46 | WIMPy | yes |
| 22:33.05 | WIMPy | What do you want me to break? |
| 22:35.33 | Ice_Strike | WIMPy Other day you told me to touch the computer or something to see if the noise stop. Remember? |
| 22:36.32 | WIMPy | Oh, yes. |
| 22:37.18 | Ice_Strike | What i found out that when I touch a computer (full hand) the buzzing/humming noises is gone.. |
| 22:37.35 | Ice_Strike | Other option - put a handset cable on a computer - buzzing/humming noises is gone.. |
| 22:38.04 | Ice_Strike | What does that mean? |
| 22:38.10 | WIMPy | Sounds like you are causing the problem. |
| 22:38.18 | WIMPy | You need grounded :-) |
| 22:38.26 | Ice_Strike | grounded on where? |
| 22:38.37 | WIMPy | Ground |
| 22:38.49 | Penguin | If he puts his hand on the thing and the noise stops, it sounds more like his body is fixing it. |
| 22:39.09 | Ice_Strike | lol |
| 22:39.10 | WIMPy | No, on another thing. |
| 22:39.27 | Ice_Strike | Seriously, it happen to many agents. |
| 22:39.29 | WIMPy | So if he touches the conputer he gets himself a ground connection. |
| 22:39.52 | Penguin | Tie your computer case to earth ground. |
| 22:40.12 | WIMPy | is pretty sure that's where it is. |
| 22:40.29 | Ice_Strike | What is strange that it was buzzing for like 15 minutes.. then it stop itself. |
| 22:40.40 | Ice_Strike | Sometime it happen to other desk.. and back for forth. |
| 22:40.51 | WIMPy | Any heavy machinery nearby? |
| 22:41.02 | Penguin | From what is the noise emitted? |
| 22:41.35 | Ice_Strike | No heavy machinery nearby but there is a tram on the road. |
| 22:42.03 | WIMPy | Electric? |
| 22:42.08 | Ice_Strike | Yep |
| 22:42.18 | Penguin | From what is the noise emitted? |
| 22:42.20 | WIMPy | Trolley system? |
| 22:43.02 | Ice_Strike | Like that http://i3.manchestereveningnews.co.uk/incoming/article6393776.ece/ALTERNATES/s615/oldham-test-trams-6393776.jpg |
| 22:43.55 | Ice_Strike | WIMPy No issue on the first floor as far I am aware, only problem on the second floor. |
| 22:44.10 | WIMPy | Trolley systems can radiate severely. |
| 22:44.14 | Penguin | From what is the noise emitted? |
| 22:45.08 | Ice_Strike | Penguin I don't know. |
| 22:45.25 | Penguin | The noise is floating in the air? |
| 22:45.32 | WIMPy | Is it normal mains hum or somethin else? |
| 22:48.40 | Ice_Strike | Penguin I really don't know. I don't know where its interferring from. |
| 23:07.55 | Ice_Strike | WIMPy normal mains hum |
| 23:08.33 | Penguin | I'm not asking you to identify the source of the interference. I'm asking what produces the audible noise that you hear. |
| 23:09.36 | Penguin | Possible answers might include: a speaker, the case is resonating, or the sound is so loud that you cannot determine what outputs the sound. |
| 23:18.17 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 23:20.48 | Ice_Strike | It sounded like that http://www.youtube.com/watch?v=IIju2GlDjVE as you can hear |
| 23:20.59 | Ice_Strike | There is no speakers on the desk. |
| 23:25.30 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-bxphcfoyizzwjiyp) |
| 23:25.57 | Ice_Strike | WIMPy Are you suggesting that grounding (earthing) on the floor might fix it? |
| 23:27.20 | WIMPy | Or on the headset cables. |
| 23:27.47 | Ice_Strike | over 80 headset :O |
| 23:28.14 | Ice_Strike | How to ground on a headset cables? |
| 23:29.01 | WIMPy | I don't know your cables. |
| 23:37.31 | Ice_Strike | I get a electrian on Monday to look into it and see if they can ground the floor. |
| 23:37.52 | *** join/#asterisk drbrown (~chatzilla@remote.l3tech.com) |
| 23:38.47 | Ice_Strike | what tool will electrician need to check for ground issue? |
| 23:40.00 | drbrown | I am attempting to compile asterisk certified 11.6-cert2 on a brand new haswell i3 and am getting the following error: |
| 23:40.09 | drbrown | {standard input}: Assembler messages: |
| 23:40.11 | drbrown | {standard input}:596: Error: no such instruction: `vfnmadd312sd .LC5(%rip),%xmm2,%xmm0' |
| 23:40.23 | drbrown | Any idea what compiler flag I need to use to get around this issue? |
| 23:47.13 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
