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| 00:53.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 03:34.58 | mattwj2002 | hi asterisk :) |
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| 03:35.54 | mattwj2002 | I have most recently rediscovered the beauty of sip |
| 03:36.32 | mattwj2002 | how beautify it is :D |
| 03:36.39 | mattwj2002 | *beautiful |
| 03:38.26 | mattwj2002 | I have an unlocked cell phone with t-mobile's prepaid $30 service. |
| 03:38.41 | mattwj2002 | 100 minutes of talk 5 GB of data per month |
| 03:39.20 | mattwj2002 | I can talk/use service from my data to make sip calls much cheaper |
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| 04:00.53 | ChannelZ | it's beautiful when it works. |
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| 04:30.26 | snadge | i want to vent my frustrations about polycom devices |
| 04:30.39 | bsdice | well what mattwj2002 said won't work e.g. in Germany with t-mobile or congstar, because there the big magenta firewall throttles your UDP after 5 sec to 2 packets/sec |
| 04:30.54 | snadge | i cant figure them out.. is there like a polycom devices in a nutshell website |
| 04:31.09 | snadge | you go to their website.. you cant find what the current firmware version is, or where to download them from |
| 04:31.36 | snadge | the darned things, try to download their configuration file off a server.. ok fine.. lets just factory reset the damned thing.. except theres 3 different reset options |
| 04:45.10 | ChannelZ | http://support.polycom.com/PolycomService/support/us/support/voice/index.html click your phone and I see lots of firmware |
| 04:49.46 | snadge | ok so there are firmware updates |
| 04:52.17 | snadge | hes running 3.1.3 on his ip 301, and 4.1.4 on his ip 501, even though the support matrix says.. this isnt supported on this model |
| 05:02.16 | snadge | how do i stop it from trying to download its configuration file? |
| 05:02.45 | snadge | all i want to do .. is manually configure the username/password and ip that it registers to.. and not have it try to download any configuration |
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| 05:04.45 | ChannelZ | sorry I don't use polycoms but my impression is that's the way it always works. |
| 05:05.10 | ChannelZ | But I could be wrong. |
| 05:16.24 | snadge | that's pretty frustrating |
| 05:17.13 | snadge | i dont know the first thing about polycoms either.. i have just been tasked with moving some people off a pbx, and onto a new one.. i've created the new pbx, now its down to reconfiguring their phones to point to the new one |
| 05:17.49 | snadge | plus we dont have any polycoms in the office that I can play with |
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| 16:48.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 16:55.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 17:16.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 18:05.08 | boratynskikamil | Extension definition: |
| 18:05.16 | boratynskikamil | http://wklej.org/id/1282172/ |
| 18:05.34 | boratynskikamil | Error message: |
| 18:05.34 | boratynskikamil | http://wklej.org/id/1282176/ |
| 18:06.33 | boratynskikamil | Suggestion? |
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| 18:20.31 | dan_j | Here's an interesting one. Asterisk 11.5.1 in a data centre receiving calls from a sip provider. 1 x Gigaset N300 base unit which has 3 SIP registrations to the Asterisk server (and 3 Gigaset handsets connected to the base). Whenever one of the handsets (lets say Handset1) tries to do a SIP based blind transfer to any other handset (lets say Handset2), the receiving handset (Handset 2) only hears asterisk's on-hold music. |
| 18:21.19 | dan_j | But when I set up freepbx (also asterisk 11.5.1) on a local pc, it works fine. |
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| 18:25.03 | dan_j | When it happens, Asterisk CLI says Host 'x.x.x.x:5060' does not implement 'UPDATE' |
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| 18:31.59 | Neozonz|Disc | [TK]D-Fender, ok i figured it all out... except... if i connect to ldap what's my asterisk username? |
| 18:34.47 | Neozonz|Disc | my config : http://prntscr.com/2vhhku |
| 18:36.40 | [TK]D-Fender | Neozonz|Disc: I don't do LDAP |
| 18:36.59 | Neozonz|Disc | ah ok |
| 18:42.29 | navaismo | dan_j, have your tried adding the disalloed_methods=UPDATE in your peer config |
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| 18:45.45 | dan_j | navaismo: not yet. I didnt know that existed. However i've checked the conf files on the Freepbx installation, and there's nothing special about them. |
| 18:52.52 | dan_j | navaismo: it doesnt seem to have an effect. |
| 18:54.34 | navaismo | i made a typo not sure if you corrected is disallowed_methods=UPDATE, also you need to share the cli output with the sip debug via pastebin |
| 18:55.07 | dan_j | I've tried looking at the sip debug, but its hard because all the phones are based behind the same IP address. |
| 18:55.40 | dan_j | I'll pb it in a mo. |
| 19:04.20 | dan_j | Here's the pastebin of the failing transfer. http://pastebin.com/wjesCgsC |
| 19:05.18 | dan_j | The initial called peer is kesher_203 which answers the call. |
| 19:05.24 | dan_j | Then tries to transfer it to kesher_204 |
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| 19:15.05 | navaismo | not shure if opverlooked but i only saw a bunch of BUSY HERE BACK FROM |
| 19:16.32 | dan_j | I saw that. Not sure why its saying. It seems to be saying it in response to OPTIONS |
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| 19:18.18 | dan_j | For internal calls, I've got an Answer() in the dialplan so that I can play a different ringtone if the dialled extension is in use. |
| 19:18.33 | dan_j | Im just wondering if that can cause problems when trying to blind transfer a call. |
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| 19:22.59 | dan_j | Ah. ok. I've removed the Answer() and now blind transfer works perfectly. |
| 19:23.49 | dan_j | However, that leaves me with the question, is it possible to play a different ringing sound to a SIP device, without sending Answer() first? |
| 19:24.45 | dan_j | Ok. Tested and it seems to work. |
| 19:24.53 | dan_j | without Answer() |
| 19:25.29 | dan_j | I wonder why this issue only showed itself with the Gigaset N300? I've got loads of other makes and models that havent had any issue with blind transfers. |
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| 19:30.31 | pabelanger | anybody playing with UniMRCP and asterisk? |
| 19:33.13 | navaismo | not me |
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| 19:45.58 | dan_j | Is there a list of sip headers used for auto answer by the different manufacturers? I'm trying to get a gigaset handset to autoanswer. |
| 19:46.58 | navaismo | check the manual of the vendor |
| 19:47.35 | [TK]D-Fender | dan_j: voip-info WIKI might have a bunch. |
| 19:47.58 | [TK]D-Fender | When in doubt (and there never should be any) ... read the manual |
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| 19:49.12 | dan_j | The manual is written for users it seems. Not very technical. |
| 19:49.39 | [TK]D-Fender | good odds they have separate admin guide.... like evry other manufacturer |
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| 20:28.29 | Katty | LOGITECH HEADPHONES. |
| 20:32.37 | [TK]D-Fender | RANDOM NOUN. |
| 20:33.59 | navaismo | Katty, how is going that cdr-stats? |
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| 20:46.39 | iulhk | trying to make some call center setup, 2 agents logged in , when call comes in system, call automatically connected with agent, i want at the start both agent soft should be ring, whoever will press answer button call should be connected, how to possible? |
| 20:47.25 | uosiu | Hi all |
| 20:47.54 | uosiu | Is there built-in in asterisk to hang up incoming calls with "busy here" when destination queue is too long? |
| 20:48.21 | Penguin | It's your dial plan, so sure there is. |
| 20:53.37 | uosiu | Hmm, I'm a bit newbie in asterisk (I usually work a LAMP sysadmin). We have asterisk routing calls via SIP (upstream telephony provider is also on SIP) and I've found some routing rules in extensions.conf |
| 20:54.54 | [TK]D-Fender | uosiu: And all of that.. is up to you to determine how to process your calls |
| 20:57.00 | navaismo | iulhk, set your ring strategy to ringall |
| 20:58.18 | iulhk | <navaismo>: in queues.conf its already , strategy = ringall |
| 20:59.11 | uosiu | Inside there's a menu (to call sales- press 1, to call boss - press 2 etc) with content like this: http://wklej.to/Bn5Ge. I'm looking for something like "If QueueLength('rejestracja') >= 20; then ?rozlacz" |
| 20:59.48 | navaismo | uosiu, set the Queue timeout |
| 21:00.32 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 21:00.50 | Penguin | You can set a timeout value in the Queue() application or in the queue configuration file. |
| 21:01.32 | uosiu | navaismo: I'm targeting in one of two possible ways: #1) End a call with "Busy here" #2) Play wav file with "There's no way to answer your call. Try another day" |
| 21:02.29 | Penguin | If your timeout time expires, dialplan should continue. Run another application that does #1 or #2. |
| 21:02.33 | navaismo | as Penguin said you can set the timeout in the queue in the next priority you can set that audio, or use a digit to jump out of the queue and go to another context |
| 21:03.14 | uosiu | hmmm, OK |
| 21:03.18 | uosiu | I'll try that :) |
| 21:03.27 | iulhk | <navaismo>: agent logged in, listening music on hold, call comes in server and at agent end it doesn't ring, automatically establish channel, i want it should ring first, if agent accept then channel should be established, hope you understand my question? |
| 21:04.23 | navaismo | maybe your sip client has the autoanswer enabled |
| 21:04.42 | uosiu | Penguin: navaismo: How about max len in queues.conf? |
| 21:14.16 | navaismo | iulhk, aah you are the mode where agents loggin directly and wait for the call connected all the time, you need to switch to static agent or something in order to leave the agent's phone hangged and when call come in the phone ring and the agent will answer that |
| 21:16.21 | gusto | hi Penguin |
| 21:16.34 | uosiu | navaismo: I've set queue timeout, but I still get periodic announcements :/ |
| 21:16.42 | gusto | hi WIMPy |
| 21:16.52 | WIMPy | hi gusto |
| 21:17.02 | gusto | how are you? |
| 21:17.12 | *** join/#asterisk hrnt (hrnt@2001:4b98:dc0:41:216:3eff:fe3f:b792) |
| 21:17.37 | WIMPy | Failing to make a B410P do something usefull. |
| 21:18.02 | iulhk | <navaismo>: so in this case in queues.conf i have to use "member => SIP/agent1" instead of "member => Agent/@1" is that ? |
| 21:18.52 | WIMPy | The Linux driver throws lots of errors and refuses to load. DAHDI does load but fails to do anything usefull so far. |
| 21:22.00 | gusto | ah, you did buy some adapter |
| 21:22.30 | gusto | is it ISDN? |
| 21:22.44 | WIMPy | No, it was a donation. |
| 21:22.49 | gusto | to you? |
| 21:22.51 | gusto | who donated? |
| 21:22.52 | WIMPy | Yes. quad BRI. |
| 21:23.09 | WIMPy | To our project. |
| 21:23.21 | gusto | to asterisk? |
| 21:23.44 | WIMPy | luftschlossfabrik.de |
| 21:23.51 | gusto | i was quite surprised that ISDN does almost every g7xx codec |
| 21:23.59 | gusto | even g722 according to some books |
| 21:24.02 | gusto | is it true? |
| 21:24.08 | WIMPy | It doesn't. |
| 21:24.18 | WIMPy | Only G.711 and G.722. |
| 21:25.14 | gusto | well, that is enough |
| 21:25.22 | gusto | but i did read that also 739 |
| 21:26.02 | Katty | my brain has stopped functioning. |
| 21:26.17 | WIMPy | In a data call you can transmit whatever you want, but you will always have 64kbit/s. |
| 21:26.29 | gusto | aha |
| 21:27.52 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 21:35.38 | *** join/#asterisk xaristax (~xaristax@148.240.65.35) |
| 21:36.09 | xaristax | Ey can someone give me a guide im kinda lost everythings works gret but whet the phone transfer the call there is no sound on one side |
| 21:36.16 | xaristax | or is like cut |
| 21:37.58 | eirirs | <PROTECTED> |
| 21:42.57 | xaristax | is there a problem with call forward and g729 |
| 21:57.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:03.21 | Katty | i don't suppose any of you use labtech to monitor your asterisk boxes |
| 22:04.23 | Katty | listens to crickets |
| 22:06.30 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 22:06.52 | Katty | well if anyone is interested in documentation on that topic, let me know. i'll be working on it sometime this week. i think. |
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| 22:23.15 | mjordan | Katty: I'm not familiar with it, but I'd be interested in anything you wrote up |
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| 23:00.02 | carrar | looks at Katty |
| 23:00.18 | carrar | people pay for monitoring software? :) |
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| 23:49.01 | *** join/#asterisk Skinns (~Skinns@66.55.134.219) |
| 23:50.47 | Skinns | I was wondering if someone could answer some questions. I want to know if its possible to build an Asterisk server to act as a voice chat for a website. Using this client http://sipml5.org/index.html?svn=179 |
| 23:51.27 | Skinns | Was hopeing that when someone created a room it would create a conference and then allow people who have access to join it. |
| 23:51.36 | Skinns | obviously an interface would have to be developed |
| 23:55.14 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 23:55.33 | mattwj2002 | hi all |
| 23:55.34 | Skinns | Anyways if any experts could answer me that would be great. I may even have a job for someone to set this up, if interested you can email me at bradzazulak@gmail.com |
| 23:55.35 | mattwj2002 | :) |
| 23:56.12 | navaismo | Skinns, yes that is possible |
| 23:56.23 | Skinns | :) |
| 23:56.30 | navaismo | you need a version of asterisk supporting sip messages |
| 23:56.52 | navaismo | usualyy the sip chat is peer to peer |
| 23:56.53 | mattwj2002 | is there an astriskpi or anything similar? |
| 23:57.02 | Skinns | oh ok |
| 23:57.09 | navaismo | but i guess you can create something to send the sip message to all participants |
| 23:57.21 | Skinns | I am very new to this and have just been playing with it on a vps I setup |
| 23:57.30 | navaismo | mattwj2002, raspbx and uelastix |
| 23:57.39 | Skinns | I have a developer but he doesnt know much about sip and asterisk |
| 23:57.53 | mattwj2002 | thanks navaismo |
| 23:58.10 | navaismo | mattwj2002, you also can get the packes of asterisk from the OS repo or compile from sources |
| 23:58.22 | mattwj2002 | awesome |
| 23:58.22 | navaismo | Skinns, for sip messages is not so hard |
| 23:58.52 | mattwj2002 | obviously I am not going to run a company off of a rpi but I thought it might be fun for a project |
| 23:58.53 | mattwj2002 | :) |
| 23:59.01 | Skinns | Navaismo, do you think you would be interested in setting something like this up? And maybe just explaining to me and my developer how it works and what you have done? Obviously I am willing to pay for this service. |