| 00:09.14 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 00:09.14 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 00:32.59 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-55-81.om.om.cox.net) |
| 00:34.35 | *** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.68) |
| 00:40.29 | batphone | apparenly zaptel only works on the 2.6 kernel. is this correct? |
| 01:06.28 | *** join/#asterisk awkwardpenguin (~awkwardpe@172-222-167-081.dhcp.chtrptr.net) |
| 01:18.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:18.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:42.22 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:f56b:78f0:2d2c:3122) |
| 01:44.21 | *** join/#asterisk DaRock (~Thunderbi@mail.unitedinsong.com.au) |
| 02:59.15 | *** join/#asterisk corretico (~corretico@200.91.143.34) |
| 03:02.14 | *** join/#asterisk boris_t (~boris_t@109.248.217.2) |
| 03:16.49 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 04:10.48 | *** join/#asterisk PrezPusyGrab (~gbully@71-223-44-106.phnx.qwest.net) |
| 04:37.23 | *** join/#asterisk robmal (r@Noe-2-pt.tunnel.tserv28.waw1.ipv6.he.net) |
| 05:22.54 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 05:32.10 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 05:47.38 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 05:50.49 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:36.02 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 07:16.54 | *** join/#asterisk miralin (~Thunderbi@91.237.94.2) |
| 07:18.23 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 07:32.32 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:39.41 | *** join/#asterisk u0m3_ (~u0m3@86.127.131.167) |
| 07:42.13 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 07:50.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:03.52 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:13.43 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:20.33 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 08:37.35 | *** join/#asterisk areski (~areski@154.red-83-47-211.dynamicip.rima-tde.net) |
| 09:28.53 | *** join/#asterisk war9407 (war@pool-70-106-197-167.clppva.fios.verizon.net) |
| 09:57.14 | *** join/#asterisk Bhakimi (~textual@cpe-104-34-204-106.socal.res.rr.com) |
| 09:57.41 | Bhakimi | hi people, anyone has experience sending variables to the asterisk AMI via http? |
| 09:57.47 | Bhakimi | called AJAM i belive |
| 09:58.03 | Bhakimi | im trying to pass variables but im not sure how thats exacly possible |
| 09:59.01 | Bhakimi | http://xxx.xxx.xxx.xxx:8088/rawmanAsync=yes&action=Originate&Exten=8364&Context=test&Channel=Local/XXXXXXX@test&Priority=1&Variable=callerid=1111111|called_count=19999999 |
| 09:59.28 | Bhakimi | my concern is how im passing the different variables sicne i use a = again and that may cause a conflict in the url |
| 10:00.25 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 10:01.18 | *** join/#asterisk sotoz (~sotoz@095-097-255-066.static.chello.nl) |
| 10:08.03 | Bhakimi | http://xxx.xxx.xxx.xxx:8088/rawmanAsync=yes&action=Originate&Exten=8364&Context=test&Channel=Local/XXXXXXX@test&Priority=1&Variable=callerid=1111111|called_count=19999999 |
| 10:08.10 | Bhakimi | is this how i pass variables via ajam ? |
| 10:10.48 | sotoz | I'm getting this wierd situation where Asterisk (v14.0.2) is sending a re-INVITEs without the connection-address in the connection data (c= field) in the SDP. The c= field looks like 'c=IN IP4' although it should look like 'c=IN IP4 10.10.10.1' for example. |
| 10:10.50 | sotoz | Have you guys seen that before? |
| 10:16.36 | drmessano | Could be a bug, that 14 release is ancient |
| 10:17.05 | drmessano | 14.0.2 was released the week 14.0.0 came out |
| 10:17.31 | Bhakimi | drmessano do you know how to pass variables via asterisk AJAM ? |
| 10:17.48 | drmessano | Bhakimi: Nope |
| 10:17.58 | Bhakimi | where can i find documentation on it |
| 10:18.04 | Bhakimi | seems like there are none anywhere |
| 10:18.08 | drmessano | No idea |
| 10:19.45 | drmessano | Bhakimi: Why not use ARI? |
| 10:20.06 | Bhakimi | because its a JS app |
| 10:20.16 | Bhakimi | and we've depended on ajam for a while |
| 10:21.04 | drmessano | Eh ok |
| 10:23.23 | drmessano | Bhakimi: I think your syntax is incorrect |
| 10:23.51 | drmessano | http://xxx.xxx.xxx.xxx:8088/rawman?Async=yes&action=Originate&Exten=8364&Context=test&Channel=Local/XXXXXXX@test&Priority=1&Variable=callerid=1111111%7Ccalled_count=19999999 |
| 10:24.03 | drmessano | Missing the ? |
| 10:29.01 | Bhakimi | sorry i aded that also |
| 10:29.05 | Bhakimi | but it still didnt work |
| 10:29.12 | Bhakimi | the request goes through but the variables dont pass |
| 10:29.38 | Bhakimi | i assume because the key name is Variable=key=var|key2=var |
| 10:31.02 | *** join/#asterisk |REDfox| (5389f81d@gateway/web/freenode/ip.83.137.248.29) |
| 10:32.07 | |REDfox| | Hi all, looking for some help with a new FreePBX+Asterisk setup, relating to "critical packet" call drops |
| 11:14.25 | *** join/#asterisk juvenal_ (juvenal@gateway/shell/elitebnc/x-mndkqamvzmypyiov) |
| 11:14.27 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 11:14.33 | *** join/#asterisk tafa2 (~tafa2@t.ldn.dsrtnet.com) |
| 11:47.30 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:02.52 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 12:22.54 | sibiria | i have a very strange problem with asterisk (13.18.2) and EAGI |
| 12:23.12 | sibiria | we employ an EAGI-based solution for audio analysis which gets the audio stream fed to it as per EAGI |
| 12:23.36 | sibiria | but asterisk refuse to send audio to the application for one specific trunk |
| 12:23.54 | sibiria | asterisk itself picks up and records the audio of the call just fine, and the other end is just fine, too |
| 12:24.14 | sibiria | but not a single byte is coming across the file descriptor to the EAGI application with this specific trunk, and i cannot figure out why |
| 12:24.53 | sibiria | is there something in the SDP or so that can result in this? |
| 12:28.25 | sibiria | the trunk in question is configured identically to the others (which work), barring of course the different host |
| 12:39.18 | sibiria | this happens on both incoming and outgoing calls with this trunk |
| 12:44.45 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 12:55.16 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 12:55.16 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:04.41 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 13:11.21 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 13:36.20 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 13:48.49 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:10.12 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 14:13.28 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 14:38.02 | *** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
| 14:47.48 | *** join/#asterisk anonymouz666 (aafe5142@gateway/web/freenode/ip.170.254.81.66) |
| 14:57.27 | *** join/#asterisk DrunkPyromaniac (~Greg@cassie.qwop.net) |
| 14:59.06 | DrunkPyromaniac | I have a user that is either at his desk or on the go with a SIP softphone. What's the best way for him to change where his extension rings? |
| 15:03.04 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 15:03.15 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 15:17.42 | *** join/#asterisk igcewieling (~ewieling@speedy-02.nyigc.net) |
| 15:18.08 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 15:22.30 | [TK]D-Fender | DrunkPyromaniac, it rings wherever you Dial() and can be reached |
| 15:22.58 | Samot | DrunkPyromaniac: You can't. |
| 15:23.02 | Samot | Not easily. |
| 15:23.32 | Samot | He's either got 2 Chan_SIP extensions or a single PJSIP extension that has 2 contacts. |
| 15:23.48 | [TK]D-Fender | If you want multiple simultaneous devices to ring either use PJSIP ro set them up separately and dial them together. |
| 15:24.01 | Samot | You'd have to control how/which contact was dialed. |
| 15:24.40 | [TK]D-Fender | yup |
| 15:24.45 | Samot | "Contact/Device Priority" or whatever sales term they use isn't a single setting. |
| 15:25.00 | [TK]D-Fender | if he solidly wants to flip it and not just ring both. |
| 15:25.06 | Samot | Asterisk is also not the best option to do that. |
| 15:26.34 | DrunkPyromaniac | OK, thanks. |
| 15:31.09 | *** join/#asterisk vt (~vt@osa.vbc.su) |
| 15:31.50 | [TK]D-Fender | 2 devices and a little dialplan to tell it to flip to the other wouldn't bee too hard |
| 15:32.25 | DrunkPyromaniac | Yeah, I was thinking set a variable by dialing like ** or something then do an IF() in the dailplan. |
| 15:33.24 | [TK]D-Fender | yes |
| 15:33.58 | DrunkPyromaniac | I'll play around with it. |
| 15:34.17 | DrunkPyromaniac | I was just hoping there was a better way. |
| 15:34.43 | vt | hi, just a quick insight for a behaviour that I don't understand. Could you confirm me that type=auth \n auth_type=userpass \n password=ABC \n username=DEF in pjsip.conf is roughly equivalent to [DEF] \n host=dynamic \n secret=ABC in sip.conf ? |
| 15:36.29 | [TK]D-Fender | Sounds right |
| 15:42.41 | vt | I've got a working (apart for unrelated details) configuration using pjsip.conf, but with sip.conf I have "Wrong Password" with my local SIP phones. They are registering with MD5 Digest. Any quick clues ? |
| 15:50.22 | vt | ok, it seems related to the fact that I use a different username in the SIP From: and the Digest "username". I'll research by myself, thanks |
| 15:58.49 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hkjvvgjtakdgazzt) |
| 15:58.49 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:12.21 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:f56b:78f0:2d2c:3122) |
| 16:14.46 | *** join/#asterisk JohnWigley (~JohnWigle@johnwigley.plus.com) |
| 16:15.44 | JohnWigley | Hi, what's the correct config to get Asterisk to accept calls from a TLS registration to another box WITHOUT trying to match on IP address? Thanks |
| 16:15.59 | JohnWigley | I tried host = dynamic but that didn't work either. |
| 16:16.25 | JohnWigley | It seems as though that only works for UAs registering TO Asterisk, not when Asterisk registers to another server. |
| 16:20.28 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 16:38.28 | *** join/#asterisk beatdown (beatdown@violator.strangled.net) |
| 16:41.20 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 16:47.02 | igcewieling | JohnWigley: have you tried removing the host= line? |
| 16:47.43 | JohnWigley | I haven't, and can't easily since it's generated automatically by a PBX GUI (Wazo ex. Xivo) |
| 16:48.08 | JohnWigley | I shall try and work out a way to test it, thanks. |
| 17:02.37 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:02.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:06.39 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:37.38 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 17:37.38 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 17:37.55 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 17:37.55 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 17:38.43 | *** join/#asterisk thiagoc_ (~thiagoc@unaffiliated/thiagoc) |
| 17:39.41 | *** join/#asterisk corretico (~corretico@200.91.143.34) |
| 17:50.19 | *** part/#asterisk DrunkPyromaniac (~Greg@cassie.qwop.net) |
| 17:55.43 | *** join/#asterisk moose55 (d8bb0020@gateway/web/freenode/ip.216.187.0.32) |
| 17:57.49 | moose55 | question with digium pri card. I have a TE420 on my spare box and I want to test to make sure the card is still good. In the pdf from digium, they state to run ./patlooptest /dev/dahdi/1 60 and if any errors come back, its a bad card. now I receive the following: /dev/dahdi/1: Device or resource busy. Is that an error or an output. can you test a card with no PRI cables plugged into it? |
| 17:58.43 | [TK]D-Fender | Use a loopback cable or dongle |
| 17:59.40 | moose55 | thought so but instructions didnt say to put one in....thanks...I will be back |
| 18:00.48 | moose55 | regular T1 loopback jack work? |
| 18:01.11 | moose55 | 1-4; 2-5 |
| 18:03.12 | [TK]D-Fender | should |
| 18:13.02 | moose55 | pretty sure my card is defective |
| 18:16.38 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 18:18.38 | *** join/#asterisk cmendes0101 (~cmendes01@189.177.242.197) |
| 18:21.54 | *** join/#asterisk cmendes0101 (~cmendes01@189.177.242.197) |
| 19:08.23 | *** join/#asterisk jkroon (~jkroon@165.16.204.163) |
| 19:18.44 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 19:25.17 | *** join/#asterisk rathjeja (32cc550e@gateway/web/freenode/ip.50.204.85.14) |
| 19:37.01 | rathjeja | Y |
| 19:42.53 | *** join/#asterisk rathjeja_ (32cc550e@gateway/web/freenode/ip.50.204.85.14) |
| 19:43.26 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 19:54.50 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 19:58.06 | *** join/#asterisk miralin (~Thunderbi@91.237.94.2) |
| 20:05.54 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 20:08.50 | *** join/#asterisk rathjeja (~rathjeja@50-204-85-14-static.hfc.comcastbusiness.net) |
| 20:14.19 | *** part/#asterisk rathjeja (~rathjeja@50-204-85-14-static.hfc.comcastbusiness.net) |
| 20:23.06 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 20:50.19 | moose55 | anyone want to help me figure out my raid/harddrive/how asterisk is set up on centos... basically I have two drives in the system. Not sure about RAID either hardware or software...my drives are in a volgroup LVM2 on centos. |
| 20:50.30 | moose55 | I dont see the other drive and I dont think I have any raid |
| 20:50.55 | moose55 | more of a server assistance vs asterisk |
| 21:00.46 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 21:03.15 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:c5a3:df70:63f1:bef1) |
| 21:05.18 | vt | in the Monitor() documentation, I read « The channel's input and output voice packets are logged to files » and below, I see « file_format - optional, if not set, defaults to wav » |
| 21:05.51 | vt | does that mean that, when using Monitor(), my voice packets (in the RTP stream) will no longer be G.711 or G.729 or whatever, but WAV ? |
| 21:06.03 | vt | is WAV is a container and not a file format ? |
| 21:06.30 | vt | s/file/sound/ |
| 21:07.13 | vt | I would like to record what pass through the RTP stream, whatever the codec in use |
| 21:17.36 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:c5a3:df70:63f1:bef1) |
| 21:31.43 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 21:32.57 | *** part/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 21:53.56 | *** join/#asterisk jermudgeon-work (Elite17303@gateway/shell/elitebnc/x-ekixvsdasoqwnyqe) |
| 22:01.46 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 22:03.45 | jermudgeon-work | Wideband from streaming media â MP3Player works as expected when called from a dialplan, and resamples to 16khz beautifully. When dumped into a ConfBridge using 'channel originate' or a .call file, it resamples to 8khz. Is there a way to dump active channel variables (not using dialplan DumpChan) in order to compare, and perhaps pass parameters to the channel originate to override the 8k default? |
| 22:03.54 | jermudgeon-work | The conference itself is not restricted to 8k. |
| 22:18.40 | *** join/#asterisk areski (~areski@37.223.2.207) |
| 22:35.34 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 22:40.32 | *** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.68) |
| 23:06.26 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:14.16 | *** join/#asterisk ketas (~ketas@73-212-191-90.dyn.estpak.ee) |
| 23:28.30 | *** join/#asterisk hfb (~hfb@47.139.21.31) |
| 23:52.31 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:6553:d638:46d0:68b) |
| 23:57.59 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 23:58.00 | xochilpili | hi all |
| 23:58.13 | xochilpili | it's being a while... |