| 00:01.34 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
| 00:11.43 | sawgood | Hi: I have a need to transfer qty = 4 channels of G.729 from one box to another box (I have the) .lic file ... (any tips on what to do next) |
| 00:12.20 | sawgood | meaning: Do I have to unistall first on what is to be the 'old' box? |
| 00:17.27 | sawgood | got it1 |
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| 01:07.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 01:41.51 | Penguin | Can two ATAs talk to each other without some type of a call control system, such as asterisk, in between? |
| 01:45.46 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:50.53 | Penguin | My guess is that they can. Since SIP devices are both clients and servers, and if the ATAs can do IP dialing, I think one should be able to call the other directly. Am I thinking correctly about this? |
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| 03:34.38 | Vutral | Penguin, you mean direct ip calling? |
| 03:34.44 | Vutral | some sip phones have that |
| 03:34.56 | Penguin | ht704, for example. |
| 03:35.08 | Vutral | yes |
| 03:35.21 | Vutral | well id prefer a phone which supports zrtp today |
| 03:35.24 | Vutral | ^^ |
| 03:35.29 | Vutral | directly encrypted connection |
| 03:35.32 | Vutral | without wiretappers :D |
| 03:35.45 | Penguin | The question was: Can one HT704 ATA call another HT704 directly without asterisk or other call control? |
| 03:35.46 | Vutral | i dont trust govnerment it can protect industrial secrets |
| 03:35.47 | Vutral | ^^ |
| 03:35.52 | Penguin | I was pretty sure it would work. |
| 03:35.53 | *** join/#asterisk mintos (~mvaliyav@1.186.101.142) |
| 03:35.57 | Vutral | yes it can |
| 03:36.03 | Vutral | if you dont have to traverse nat and so |
| 03:36.19 | Vutral | if you have asterisk and reinviting is allowed |
| 03:36.31 | Vutral | they even make up a direct connection when under call control |
| 03:36.36 | Vutral | instead of sending evrything over the server |
| 03:37.11 | Penguin | There wouldn't be any "server." |
| 03:37.30 | Vutral | yeah |
| 03:37.31 | Penguin | It would just be one ATA on one end an another ATA on the other end. They would be configured on the same LAN. |
| 03:37.34 | Vutral | so no broker for nat traversal |
| 03:37.36 | Vutral | ^^ |
| 03:37.42 | Vutral | sure on the same lan its zero problem |
| 03:37.45 | Penguin | I figured if IP dialing was enabled, one could call the other. |
| 03:37.47 | Vutral | just enter the ip of the other phone |
| 03:37.49 | Vutral | and you can call it |
| 03:37.53 | Penguin | That's what I thought. |
| 03:45.43 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
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| 04:21.53 | redotis | Why is it that when you have one extension with two dial statements, back to back, that asterisk doesn't simply go to the next one after you hang up? |
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| 05:01.05 | kaldemar | redotis: you didn't tell it to do so with an option for app Dial. |
| 05:01.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:01.43 | kaldemar | redotis: also, who hangs up? caller or callee? |
| 05:06.07 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 05:07.38 | mae_tae | and request name is invlid |
| 05:08.33 | mae_tae | how do i resolve "method is unknown" error? |
| 05:09.47 | kaldemar | asterisk has no such error. what do you mean? |
| 05:12.38 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 05:15.00 | redotis | Either kaldemar |
| 05:15.18 | redotis | The first dial exits the extension |
| 05:15.21 | redotis | no matter what |
| 05:15.28 | redotis | well |
| 05:15.47 | redotis | if it successfully creates the channel |
| 05:16.03 | redotis | I would think it would just dial the next priority |
| 05:20.32 | kaldemar | if the caller hangs up, the dialplan execution continues in the hangup extension. if the callee hangs up, dialplan execution will continue but only if you use the g option in Dial. |
| 05:21.31 | kaldemar | is the destination answering the call? |
| 05:25.34 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:26.03 | kaldemar | actually it doesn't matter if the destination answers or not. g will make it continue to next priority anyway. |
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| 06:02.07 | redotis | hangup extension? |
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| 06:53.25 | *** join/#asterisk hurdman (~ygcheny@c3p0.r0b0t.fr) |
| 06:56.49 | hurdman | hello ! i have got an asterisk between two different network , one over ethernet and one over wifi. somethig like that Sip-A <== eth0 ==> Asterisk <== wlan0 ==> Sip-B . What should i set to call with video from Sip-A to Sip-B. Without any change into the sip.conf ( only peers ), i can talk but no video. i have try some change with directmedia and directrtpsetup, but i'm not sure of what i'm doing. any idea ? |
| 06:56.56 | *** join/#asterisk raub (~raub@ip70-171-22-39.ga.at.cox.net) |
| 07:00.42 | kaldemar | hurdman: videosupport=yes in sip.conf |
| 07:04.03 | hurdman | it's turn on :/ |
| 07:04.54 | hurdman | i have seen that i have to choose between no / update /yes for directmedia, but i 'm not sure to understand, when Asterisk IS the bridge, should i choose "no" ? |
| 07:05.12 | hurdman | to ever stay between the to peers ? |
| 07:06.00 | TazzNZ | hurdman: does "normal" calls work ? |
| 07:06.32 | hurdman | from Sip-A to Asterisk localhost for example ? |
| 07:06.39 | hurdman | yes |
| 07:06.40 | TazzNZ | and to sip-b |
| 07:07.19 | hurdman | it seems with default option, i have the sound only in one direction and no video |
| 07:07.26 | hurdman | but i can call |
| 07:07.34 | TazzNZ | any nat'ing inbetween ? |
| 07:07.55 | hurdman | nat'ing ? |
| 07:07.58 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 07:07.59 | TazzNZ | NAT |
| 07:08.04 | TazzNZ | firewalls |
| 07:08.13 | hurdman | no, there's no |
| 07:08.34 | TazzNZ | and ip forwarding enabled on the asterisk box ? |
| 07:08.43 | hurdman | yes |
| 07:08.53 | hurdman | echo 1 > /proc/sys/ipv4*** |
| 07:09.00 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 07:11.43 | hurdman | i use a version of linphone with h264 for video |
| 07:11.54 | hurdman | (that works with a direct call ) |
| 07:11.57 | hurdman | into linux |
| 07:12.35 | hurdman | should i had some route or some nat TazzNZ ? |
| 07:18.29 | *** join/#asterisk moy_ (~textual@63.133.202.2) |
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| 07:28.57 | hurdman | arf, my /proc/sys/net/ipv4/ip_forward is back to 0, TazzNZ , i should pass it to 1 ? |
| 07:30.23 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
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| 07:38.30 | TazzNZ | hurdman: if you doing directmedia then yes |
| 07:39.16 | TazzNZ | what is canreinvite set to ? |
| 07:39.19 | TazzNZ | in sip.conf |
| 07:39.24 | mae_tae | people, whats the cause of method is unknown? how do i resolve this? |
| 07:39.49 | TazzNZ | mae_tae: more info please ? |
| 07:41.01 | mae_tae | the problem is i cannot make external call when snort is activated, it blocks the softphone call |
| 07:41.22 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 07:41.28 | mae_tae | the reason why it is block its because of method is unknown & request name is invalid |
| 07:41.37 | kaldemar | mae_tae: by looking at asterisk and the call. sip set debug on. |
| 07:41.45 | mae_tae | but if snort is deactivated everything works fine |
| 07:41.46 | PLMg | hello, anyone know how to input amportal restart into crontab? (pathing) |
| 07:42.15 | kaldemar | PLMg: are you asking how to know the path to the command? |
| 07:42.52 | mae_tae | ok ill try |
| 07:42.53 | TazzNZ | mae_tae: is snort up-to-date ? |
| 07:43.11 | PLMg | uhm, yes and no. I tried with /amportal restart in crontab -e but I think I need to do something like /etc/init.d/asterisk but for amportal |
| 07:43.56 | mae_tae | TazzNZ: yes up to date, i wonder what rules will i disable to allow the softphone call |
| 07:44.06 | PLMg | short version, I want amportal restart to run at certain times |
| 07:44.22 | kaldemar | PLMg: you hadrly have the amportal in your root directory. "which amportal" will tell you where it is. |
| 07:44.52 | PLMg | ty |
| 07:44.57 | hurdman | TazzNZ: you |
| 07:45.02 | hurdman | TazzNZ: thx you* |
| 07:45.17 | TazzNZ | hurdman: no problem |
| 07:45.52 | hurdman | TazzNZ: you mean directmedia for careinvite ? :) i have set it now to "no" |
| 07:46.02 | hurdman | i have try update too |
| 07:46.31 | TazzNZ | if canreinvite=no then the call is forced via the Asterisk box |
| 07:46.46 | TazzNZ | you should then be able to disable ip forwarding |
| 07:48.05 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 07:48.51 | *** part/#asterisk PLMg (PLMg@78.96.151.225) |
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| 07:50.32 | hurdman | TazzNZ: ok, so in my case, with two different network and asterisk in the middle i should set it to no ? directrtpsetup too ? |
| 07:51.37 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.180.118) |
| 07:54.07 | TazzNZ | hurdman: imho, yes |
| 07:54.19 | TazzNZ | unless you have a very weak Asterisk box |
| 07:54.35 | TazzNZ | or you don't want the media to go via the Asterisk box |
| 07:54.49 | TazzNZ | which, if I understand you correctly, is impossible in your setup |
| 07:55.06 | TazzNZ | the difference is that Asterisk will "know" about the media |
| 07:55.15 | TazzNZ | where as the ip forward route, it is the kernel |
| 07:56.11 | hurdman | ok |
| 07:56.40 | hurdman | TazzNZ: my asterisk is "weak" but sufficiant ( a tiny arm ) |
| 08:02.04 | TazzNZ | well - at least you know the limits and if you start seeing high CPU usage, you know what to change |
| 08:03.07 | mae_tae | TazzNZ: would whatever block by snort would it also appear in asterisk console? |
| 08:06.39 | *** join/#asterisk kresp0 (~kresp0@136.Red-95-120-50.dynamicIP.rima-tde.net) |
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| 08:16.21 | mae_tae | any comment |
| 08:17.09 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:20.51 | *** join/#asterisk s14ck (uid6427@gateway/web/irccloud.com/x-bstrnqkyxucdekhq) |
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| 08:23.42 | mae_tae | asterisk is very difficult, huhu.. |
| 08:26.16 | mae_tae | (spp_sip) Request name is invalid - 10/09/13-16:14:18 <--- this is a sample error |
| 08:38.08 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 08:39.17 | mae_tae | TazzNZ: here's the captured data from the console http://pastebin.com/dhLCc9qX |
| 08:39.56 | mae_tae | kaldemar: captured result http://pastebin.com/dhLCc9qX |
| 08:40.07 | mae_tae | the one that is being block is 1152 |
| 08:41.38 | mae_tae | help me pleaseeee ;-) |
| 08:43.04 | kaldemar | mae_tae: your debug only has some qualify messages. it is useless. |
| 08:43.46 | mae_tae | i see |
| 08:50.16 | mae_tae | can i search from that sip set debug on the specific userID? |
| 08:51.49 | kaldemar | mae_tae: hopefully. |
| 08:53.37 | mae_tae | kaldemar: i mean what command will i type that i can only list specific id no? |
| 08:53.48 | mae_tae | im really new to asterisk |
| 08:53.52 | stanlly | hi, I'm having issues with Digium Wildcard TE110P. I've run dahdi_maint and patlooptest and see 2038 errors. So there is no doubt that the card is not ok, right? or could it be something else? |
| 08:54.13 | kaldemar | mae_tae: there is no debug filtering in asterisk if that's what you mean. save it all to a file and do what you wish afterwards. |
| 08:55.59 | mae_tae | kaldemar: yeah you're right but i cannot capture all screen displays because it soo fast & too long... |
| 09:00.21 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.180.118) |
| 09:00.37 | kaldemar | mae_tae: there are tools to help you with that. asterisk -vvvr | tee /tmp/ast_output.log |
| 09:01.26 | kaldemar | mae_tae: tee is a program that duplicates stdout to a file. |
| 09:01.42 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:01.50 | mae_tae | ok thanks kaldemar |
| 09:03.35 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
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| 09:26.44 | aberrios | are any of the presentations from AstriCon being livecast/streamed online? |
| 09:28.05 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.180.118) |
| 09:32.54 | *** join/#asterisk bombev (~User@73.109.Globcom.Net) |
| 09:38.47 | bombev | what is the best asterisk version 1.8 , 10 , 11 ? |
| 09:38.48 | bombev | :) |
| 09:47.25 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 09:51.32 | WIMPy | 11 |
| 09:58.50 | mirela666 | 1.8 :) |
| 10:01.37 | bombev | :) |
| 10:04.23 | bombev | WIMPy why do you prefer 11 and mirela666 why 1.8 ;) |
| 10:09.19 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:10.38 | davlefou | bombev, 1.8 seems an good choice, it packaged under ubuntu. |
| 10:12.03 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 10:12.39 | mirela666 | bombev: 1.8 has more Open Source spirit than 11 :D |
| 10:13.57 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 10:13.57 | bombev | :)) |
| 10:14.07 | bombev | mirela666 really :) |
| 10:16.10 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
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| 11:25.58 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 11:25.58 | *** mode/#asterisk [+o file] by ChanServ |
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| 11:47.54 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 11:49.30 | *** join/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 11:49.35 | DelphiWorld | hi people |
| 11:49.39 | DelphiWorld | anyone have sipvissious? |
| 11:49.59 | kesselklopfer79 | is there a trick to get features working on channel that where bridged with the manager action Bridge ? |
| 11:50.53 | *** part/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 11:52.56 | kesselklopfer79 | for example at the end of atxfer we brige the channel to get rid of Local/....;1 Local/....;2 but the next atxfer will not work |
| 12:00.49 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 12:02.38 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 12:06.06 | eirirs | do asterisk tag rtp packets with DSCP tags? |
| 12:08.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:13.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 12:39.22 | pigeonflight | I've setup a polycom soundpoint 321 it receives calls fine but cannot make calls. What might cause such a problem? |
| 12:41.04 | [TK]D-Fender | bad codec order, bad dialplan, bad auth, sun-spots, shift in the earth's magnetic axis, etc |
| 12:41.24 | [TK]D-Fender | You should look at the SIP debug and see what * sees |
| 12:41.30 | [TK]D-Fender | "sip set debug on" <- |
| 12:41.41 | tm1000 | bombev: 11 |
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| 12:44.38 | bombev | tm1000 why 11 |
| 12:44.39 | bombev | :) |
| 12:47.42 | tm1000 | confbridge improvements, chan motif, webrtc, |
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| 12:52.35 | [TK]D-Fender | Hangup handlers, T.38 termination, |
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| 12:57.42 | phix | firewall |
| 12:58.45 | Weezey | Have a problem where users' SIP adapter doesn't hear ringing when they dial out. |
| 12:59.25 | Weezey | SIP adapter has public IP, and inbound/outbound calling is working. |
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| 13:00.49 | [TK]D-Fender | if it is only ringing, it isn't a networking issue, its an indications issue |
| 13:00.58 | [TK]D-Fender | Weezey: pastebin the call with SIP debug |
| 13:01.00 | [TK]D-Fender | ~pb |
| 13:01.01 | 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. |
| 13:01.02 | [TK]D-Fender | ^^^ |
| 13:01.09 | phix | ---^ |
| 13:01.42 | phix | <PROTECTED> |
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| 13:05.20 | Weezey | [TK]D-Fender: I don't think they're in for the day, anything you can think of to look for? The device is provisioned the same way as all the others I have out there, and I thought it was natting but, it's their router and webif responds on their IP. |
| 13:05.47 | [TK]D-Fender | Weezey: You're asking for a diagnosis without an examination. |
| 13:05.58 | [TK]D-Fender | Weezey: You don't miraculously loos ONLY the ringing. |
| 13:06.19 | Weezey | They didn't lose it, they never had it. |
| 13:06.44 | [TK]D-Fender | Weezey: Wither it is trying to be sent in-band and * can't transcode it, or the channel that is in process of being bridged with it isn't passing any, etc |
| 13:07.01 | [TK]D-Fender | Or the phone's zone isn't set and donesn't know what to do.. |
| 13:07.14 | [TK]D-Fender | But you can't debug a probolem you can't see |
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| 13:11.51 | Weezey | [TK]D-Fender: gotcha. Any way you can think of to simulate an outbound call without a physical human there? |
| 13:12.18 | Weezey | as opposed to a mental human. |
| 13:12.41 | [TK]D-Fender | Weezey: I certainly know of no magic to make the phone call OUT.... and then have noone there to prove if there isa ringing sound <- |
| 13:13.12 | [TK]D-Fender | Weezey: Wait for actual people. |
| 13:13.25 | Weezey | k |
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| 13:43.20 | jansiva | we have an issue where 8901 conference umber |
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| 13:45.10 | jansiva | when people are in conference we are getting error |
| 13:45.22 | jansiva | some one ringing |
| 13:45.32 | jansiva | what could be the issue |
| 13:46.04 | Katty | lack of caffeinated beverage |
| 13:52.06 | [TK]D-Fender | jansiva: you need to give us a proper description and show the actual debug where you get the failure |
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| 14:03.23 | eirirs | lol |
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| 14:17.13 | *** join/#asterisk kchehab (~kchehab@77.42.241.66) |
| 14:18.00 | kchehab | good evening i can find in my asterisk full log config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 14:18.27 | kchehab | it was working fine before what can cause this problem |
| 14:19.20 | kchehab | just system has an un proper shutdown |
| 14:19.29 | kchehab | bu i can connect to mysql normally |
| 14:22.56 | davlefouAMD | kchehab, did you have load or preload it? |
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| 14:24.24 | kchehab | davlefouAMD yes i did ,now i am thinking to recompile it |
| 14:24.44 | kchehab | davlefouAMD 'MODULE LOAD RES_CONFIG_MYSQL' failed. |
| 14:26.48 | davlefouAMD | you should use mysql in you asterisk, it seems me more simple. |
| 14:27.24 | davlefouAMD | kchehab, why you don't use package? Under what os are you working? |
| 14:28.00 | kchehab | davlefouAMD i was using it and it was working |
| 14:28.29 | kchehab | davlefouAMD i know that its more than simple but my problem is why did it dissapear or cant be loaded any more |
| 14:29.11 | kchehab | davlefouAMD as i inform you it was working for 2 months but due to unproper shutdown this error appears |
| 14:29.45 | davlefouAMD | did you look you log? |
| 14:29.59 | davlefouAMD | what os are you using? |
| 14:30.28 | Katty | WINDERS. |
| 14:31.55 | kchehab | davlefouAMD i have a simple question ,i have a freepbx and i want to install a new version of asterisk ,what should be the make command --prefix =/usr ?? or |
| 14:32.12 | kchehab | to load in same directory |
| 14:33.29 | davlefouAMD | i don't you freepbx it seems me you must well separate twice version, better, erase old one |
| 14:34.14 | kchehab | davlefouAMD its for my friend and ask me to install it remotely ,some i was trying to compile asterisk in the same machine |
| 14:34.17 | kchehab | could this work |
| 14:34.52 | davlefouAMD | it seems me an very bad idea |
| 14:34.54 | kchehab | c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not availab |
| 14:35.05 | kchehab | what do this message extatly means ? |
| 14:35.36 | kchehab | could be a problem in dsn |
| 14:36.21 | [TK]D-Fender | Go prove the DSN is functional with the ame login |
| 14:37.24 | kchehab | module load res_config_mysql.so |
| 14:37.25 | kchehab | Loaded res_config_mysql.so |
| 14:39.18 | davlefouAMD | In realtime, how i translate that : register => fromuser@fromdomain:secret@host ? |
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| 14:50.22 | Penguin | ~freepbx |
| 14:50.22 | 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 |
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| 15:08.22 | user258467 | Hi, I would like to know in which files asterisk phones are configured becaus eI don't see it? |
| 15:08.49 | Penguin | What kind of "asterisk phones" are you talking about? |
| 15:09.11 | user258467 | Penguin, I mean asterisk users |
| 15:09.27 | Penguin | What is an asterisk user? |
| 15:10.03 | kchehab | [TK]D-Fender general connected to asterisk@127.0.0.1, port 3306 with username asteriskuser for 43 seconds |
| 15:10.05 | user258467 | I am in a small business there is approximatively 10 phones, I would like to know where are the phones settings |
| 15:10.10 | Penguin | Generally speaking, people do not use asterisk; people use phones. |
| 15:10.25 | Penguin | I'll ask you again: what kind of phones are you talking about? |
| 15:10.40 | kchehab | [TK]D-Fender sip show peers Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 15:10.52 | user258467 | Penguin, it is isco phones and Gigaset |
| 15:10.58 | [TK]D-Fender | kchehab: Show proper, and complete backup .... |
| 15:11.09 | Penguin | Cisco? |
| 15:11.14 | user258467 | Penguin, yes |
| 15:11.18 | Penguin | Using SIP or SCCP? |
| 15:11.28 | kchehab | [TK]D-Fender sorry i didnt get you ,what show proper ? |
| 15:11.35 | user258467 | Penguin, SIP i think |
| 15:11.38 | Penguin | Your config file in asterisk is sip.conf |
| 15:11.45 | [TK]D-Fender | kchehab: Show the COMPLETE commands you are using to validate your setup. |
| 15:11.50 | kchehab | ok |
| 15:11.55 | [TK]D-Fender | kchehab: You just pasted ONE line and I don't know what you did to get it |
| 15:12.18 | kchehab | [TK]D-Fender i have a problem while loading asterisk users real time for asterisk 11.5.1 |
| 15:12.45 | kchehab | printing this command in asterisk cli realtime mysql status |
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| 15:13.06 | kchehab | it shows that asterisk is already connected to mysql |
| 15:13.15 | kchehab | general connected to asterisk@127.0.0.1, port 3306 with username asteriskuser for 3 minutes |
| 15:13.55 | kchehab | while printing localhost*CLI> sip show peers i find this error config.c: Realtime mapping for 'iaxpeers' found to engine 'odbc', but the engine is not available |
| 15:14.16 | [TK]D-Fender | [11:13]kchehabit shows that asterisk is already connected to mysql <- so? |
| 15:14.36 | [TK]D-Fender | MySQL direct connection is NOT ***ODBC*** |
| 15:14.59 | kchehab | [TK]D-Fender ok how to start troubleshooting odbc please |
| 15:15.14 | [TK]D-Fender | kchehab: and I'm not seeing complete out from your debugging attmps. Where do I see you connecting via ODBC from CLI? |
| 15:15.43 | kchehab | how can i test odbc from cli ? |
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| 15:16.22 | [TK]D-Fender | kchehab: man isql <- |
| 15:16.45 | [TK]D-Fender | kchehab: Go read how to actually use ODBC. You should be better informed before starting to configure * to use it |
| 15:17.08 | kchehab | [TK]D-Fender thanks i will go for reading |
| 15:18.03 | kchehab | [TK]D-Fender isql DSN [UID [PWD]] [options] that what should i test :) |
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| 15:20.00 | rrittgarn | Morning Gentlemen. Wondering if you guys have any recommendation on hardware (preferably an appliance) that will have POTS connectivity, and connect back to a central asterisk server. The offices are in a location where good Internet isn't always available so the recommended appliance(es) need to support a bit of local routing. Thinking of pairing a Sangoma Vega 50 and Sangoma Vega 5000 together to try and figure thi |
| 15:22.34 | rrittgarn | tldr version: Anybody worked with the Sangoma Vega series hardware, and have any input on the capabilities of the 5000 unit in regards to a branch office scenario |
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| 15:38.04 | davlefouAMD | hi, in realtime with sip data base, i need to register some trunk to receive call, i don't undestand how i have to do? What have to put in data base? |
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| 16:39.51 | TazzNZ | davlefouAMD: can you get it working in sip.conf ? |
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| 16:45.22 | davlefouAMD | if i put the register ligne in sip.conf, but i would like to have it in sql data base. |
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| 16:46.37 | davlefouAMD | TazzNZ, how i do it on sql data base? |
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| 17:19.46 | beta2k | Hello all |
| 17:20.24 | beta2k | Can the console channel be used for paging and music on hold when not paging? |
| 17:21.56 | [TK]D-Fender | keep this in #freepbx |
| 17:22.27 | Weezey | Anyone have XMPP PubSub set up? |
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| 17:31.48 | anonymouz666 | Weezey: res_corosync, sorry |
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| 17:48.34 | Weezey | anonymouz666: that looks much simpler than res_xmpp was to set up. |
| 17:48.44 | Weezey | anonymouz666: where were you 3 months ago? |
| 17:49.51 | anonymouz666 | honolulu |
| 17:50.35 | Weezey | nice |
| 17:50.46 | Weezey | can you query the corosync data somehow? |
| 17:51.00 | Weezey | to make it accessible outside of asterisk? |
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| 17:53.13 | anonymouz666 | I am not aware of such kind of access using res_corosync, for me, it's just a way to transport device state/mwi for one box to another. |
| 17:53.20 | anonymouz666 | *from |
| 17:54.51 | anonymouz666 | or nodes |
| 17:54.59 | anonymouz666 | in my case I have 4 nodes using res_corosync |
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| 17:57.58 | Micc | what is the best windows app for testing RTP? I need to troubleshoot an audio problem for a customer, and my other tools are not seeing anything. I'd like to put something in place to make test calls and tell me the packet loss or jitter. |
| 17:58.12 | Micc | I'm willing to pay for a good program. |
| 17:59.07 | carrar | http://www.voipmonitor.org works well |
| 18:00.57 | davlefou | TazzNZ, hi, can you said me more about register with data base realtime? |
| 18:01.04 | Micc | carrar, Is this just a sniffer or can it actually place a test call? |
| 18:01.16 | carrar | just a sniffer |
| 18:02.45 | carrar | SIPP isn't windows but it can generate calls |
| 18:02.55 | carrar | do stress testing |
| 18:03.31 | Micc | carrar, oh voipmonitor is something I put on my asterisk server? |
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| 18:04.40 | carrar | Micc: you can put a collector on your asterisk box |
| 18:04.56 | carrar | could install voipmonitor on the server too if youw anted |
| 18:05.38 | carrar | http://www.voipmonitor.org/doc/Architecture |
| 18:06.19 | carrar | it could be all 1 single box |
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| 18:07.43 | Micc | ok, I see. I'll have to read up on that later. |
| 18:08.01 | Micc | I need something I can run on a laptop at the customers location. |
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| 18:09.06 | carrar | could deploy a remote collector |
| 18:09.11 | carrar | mirror their port |
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| 18:10.03 | Weezey | anonymouz666: cool, looks much more robust than xmpp. I have that working with a bunch of offices for a centralized queue system. but looking to extend voice mail and phone state efficiently. |
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| 18:45.00 | Penguin | Bizarre. Even though I have noload for res_smdi.so, it still tries to load up when starting asterisk. |
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| 21:13.55 | Shaan | hey guys i needed some help, i'm looking for a way to send faxes via VOIP, something similar to myfax.com, but for personal use can anyone suggest any solutions? |
| 21:15.44 | Shaan | how reliable is sendfax on asterisk? |
| 21:16.09 | Shaan | or is there another *nix based solution to sending faxes. one that preferably allows you to set your own Fax Headers. |
| 21:21.44 | navaismo | Fax for Asterisk, but fax on voip channels really sucks... hmm fax really sucks. Provider must support fax protocol like T38. You can try also with Hylafax+Avantfax+Iaxmodem without SIP lines |
| 21:22.16 | Shaan | but i used to use a regular fax machine with vonage works fine dont they use voip too? |
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| 21:42.33 | navaismo | idk |
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| 22:15.58 | raden | is there a way to have 2 extensions ring but one be delayed by 10 seconds without doing ring one ext for 10 seconds then ring both ? customer doesnt like the 2 caller id enteries |
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| 22:27.03 | WIMPy | Use a local channel and a wait. |
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| 22:53.41 | *** join/#asterisk iceyp (~icepick@watcher.vibecommunications.co.nz) |
| 22:54.04 | iceyp | hey guys, Im using asterisk to send fax's however when asterisk dial's the B party number it's not requesting T.38 |
| 22:54.13 | iceyp | how do I force asterisk to request T.38 on the call? |
| 22:54.39 | iceyp | I've tried Set(FAXOPT(gateway)=yes) and setting sip.conf:faxdetect = t38 |
| 23:13.28 | ChannelZ-Wk | Is it setup? Does your ITSP even support it? |
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| 23:18.29 | iceyp | yes they do |
| 23:18.38 | iceyp | the problem is when i dial the remote number my provider doesnt request T.38 |
| 23:18.43 | iceyp | We have to send the re-invite |
| 23:18.47 | iceyp | but asterisk isnt doing it |
| 23:19.02 | iceyp | with our other provider, if we try send a fax, the remote end requests the T.38 |
| 23:19.17 | iceyp | so in this case, my carrier doesnt do the re-invite they are expecting us to do the re-invite with T.38 |
| 23:21.38 | iceyp | so I guess I'm trying to figure out how to get Asterisk to establish T38 with a re-nvite |
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