| 00:07.33 | *** join/#asterisk spditner (~simon@192.171.42.137) |
| 00:15.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 01:05.02 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 01:12.23 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 01:37.29 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 01:40.01 | phix | G"day |
| 01:40.30 | phix | I am having issues with dialling a number from an analogue phone connected to a tdm400p |
| 01:41.18 | phix | There is a dial tone (except on one of the ports, even though there is a FXS module in it), asterisk console logs the phone when it goes off the hook but doesn't register when I dial a number |
| 01:41.48 | phix | after a few seconds I get an engaged signal of the analogue phone |
| 01:46.27 | *** join/#asterisk lanning_ (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 01:46.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:47.56 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 01:56.32 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 02:10.59 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 02:10.59 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:30.06 | dysinger | been hunting around for a while. I have asterisk 12 installed. Iâm looking for pointers on how to get vp8 video passthrough worknig. Any links or tips? |
| 02:31.25 | dysinger | All I can find is âPassthrough support for the video codec VP8 (and Opus for audio) was added in Asterisk 12.â |
| 02:31.48 | dysinger | Trying to get Linphone to work on a couple phones for video is not easy for a noob to asterisk |
| 02:31.52 | [TK]D-Fender | QAllow the codec. use it |
| 02:32.08 | dysinger | allow=vp8 ? |
| 02:32.31 | dysinger | searches the google for QAllow |
| 02:32.41 | [TK]D-Fender | Q was a typo... |
| 02:32.54 | dysinger | ok |
| 02:34.54 | dysinger | code says allow=vp8 should work |
| 02:34.58 | dysinger | crosses fingers |
| 02:38.13 | phix | ok I upgraded to asterisk 11 and that solved my issues with dahdi |
| 02:38.30 | phix | I now have a new issues, I can't dial my SIP provider now :/ |
| 02:38.34 | phix | app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 02:40.31 | [TK]D-Fender | Would be nice if * had somewhere to contact... |
| 02:41.47 | phix | [TK]D-Fender: so how do I debug this? |
| 02:41.54 | [TK]D-Fender | look at your peer. |
| 02:41.58 | phix | sip shoiw registry tells me it is registered |
| 02:42.12 | phix | thati s for incoming I guess |
| 02:42.13 | [TK]D-Fender | that has nothing to do with the ability to place a call |
| 02:45.35 | phix | ok, so how can I debug this? |
| 02:46.38 | [TK]D-Fender | look at your peer |
| 02:49.51 | phix | Retransmitting #4 (NAT) |
| 02:49.53 | phix | hmmm |
| 02:53.06 | phix | peer unreachable ":\ |
| 03:17.46 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:37.37 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 03:45.11 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 03:46.30 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 03:46.41 | *** join/#asterisk SGjunior (~sgjunior@out-pq-166.wireless.telus.com) |
| 03:55.58 | *** join/#asterisk Phunter (~Pghpunkid@ip68-3-220-113.ph.ph.cox.net) |
| 03:56.20 | Phunter | Queues arent doing any Music.. any ideas? |
| 03:57.01 | [TK]D-Fender | get some? |
| 03:57.08 | Phunter | there is audio files in /var/lib/asterisk/moh, have musiconhold.conf setup with "mode=files" and "directory=moh" and its not working. |
| 03:57.14 | Phunter | using right contexts.. |
| 03:57.40 | [TK]D-Fender | Show us |
| 03:57.44 | [TK]D-Fender | !pb |
| 03:57.45 | Phunter | And if The 'customer' stays in the queue too long, an agent rings the other person. |
| 03:57.49 | [TK]D-Fender | ~pb |
| 03:57.50 | 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. |
| 03:57.53 | [TK]D-Fender | ^^^ |
| 03:57.56 | Phunter | Hang on Ill Pastebin it |
| 04:00.40 | Phunter | http://pastebin.com/MFZUpV3J |
| 04:01.45 | Phunter | It provides this |
| 04:01.45 | Phunter | http://pastebin.com/vwXDKPiV |
| 04:01.58 | Phunter | the |
| 04:02.45 | Phunter | the 'customer' never hears the audio files, and the agent eventually appears to call the 'customer' |
| 04:03.03 | [TK]D-Fender | Show us the files too... |
| 04:03.16 | Phunter | <PROTECTED> |
| 04:03.18 | Phunter | ^^ |
| 04:03.37 | Phunter | I have this issue on another server, so I setup a test box at home to figure out why. |
| 04:03.40 | Phunter | Its slim. |
| 04:03.55 | Phunter | Oh wait, the audio listing? |
| 04:04.21 | [TK]D-Fender | show us the files in the folder |
| 04:05.03 | Phunter | http://pastebin.com/r7auUu1e |
| 04:05.13 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 04:05.52 | [TK]D-Fender | specify a full path for the directory |
| 04:05.58 | [TK]D-Fender | in your conf |
| 04:07.21 | Phunter | Still get nothing. |
| 04:07.27 | [TK]D-Fender | also "moh show files" , "moh show classes" |
| 04:08.34 | Phunter | http://pastebin.com/7cRQ2YJP |
| 04:09.26 | Phunter | I get audio when the 2 endpoints call each other, but not in queues. |
| 04:10.07 | [TK]D-Fender | what about a raw playback of them? |
| 04:11.04 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:11.10 | [TK]D-Fender | So far it's sounding like "no audio directly with my server", more than "MoH problem" |
| 04:11.37 | Phunter | like i said, the devices have comedia set, and it works that way.. |
| 04:11.44 | Phunter | OH |
| 04:11.47 | Phunter | hang on |
| 04:12.23 | Phunter | Okay nevermind. |
| 04:12.27 | Phunter | Was thinking RTP. |
| 04:12.48 | snadge | is anyone able to comment on how qualify time relates to ping? |
| 04:13.17 | snadge | ie.. is it normal to have a ping of 30ms.. but the devices at that site are showing qualify times of around 110-120ms |
| 04:13.17 | Phunter | Raw playback in extensions works fine when you specify the specific file. |
| 04:13.56 | [TK]D-Fender | [00:12]snadgeis anyone able to comment on how qualify time relates to ping? <- they don't. At all. |
| 04:13.56 | Phunter | <PROTECTED> |
| 04:14.07 | [TK]D-Fender | snadge: totally different protocol and OSI layer |
| 04:14.20 | snadge | yeah this is the first time i've looked making a comparison between the two |
| 04:14.45 | snadge | long story short.. theres a ddos on an upstream provider.. but traces and pings to customer equipment look good.. however they are complaining of call quality issues |
| 04:15.00 | snadge | im not saying thats unusual or wrong.. that obviously makes sense |
| 04:15.25 | [TK]D-Fender | Phunter: Use the full path in your "directory=" line |
| 04:15.30 | snadge | if i try to make my ping packets larger.. they dont make it through at all |
| 04:15.42 | Phunter | How do I do multiple files? |
| 04:15.55 | [TK]D-Fender | Phunter: Where? |
| 04:16.52 | Phunter | musiconhold. |
| 04:17.00 | [TK]D-Fender | you lost me... |
| 04:17.02 | Phunter | You said use the full path. |
| 04:17.04 | Phunter | I already am. |
| 04:17.10 | [TK]D-Fender | just put the full directory path there |
| 04:17.21 | Phunter | I did. It does not work. |
| 04:17.49 | [TK]D-Fender | test with "MusicOnHold in the dialplan |
| 04:19.27 | Phunter | ^ Does not work. |
| 04:21.43 | [TK]D-Fender | rhm |
| 04:21.47 | [TK]D-Fender | hrm even.... |
| 04:21.52 | Phunter | Wait. |
| 04:21.55 | Phunter | I think I figured it out. |
| 04:21.59 | Phunter | Have to Answer() first. |
| 04:22.01 | Phunter | WTF. |
| 04:22.02 | [TK]D-Fender | .... |
| 04:22.04 | [TK]D-Fender | yes |
| 04:22.08 | Phunter | .. |
| 04:22.13 | Phunter | facepalms. |
| 04:22.24 | [TK]D-Fender | So you can have a queue without explicitly answering the line first |
| 04:23.28 | Phunter | wow. |
| 04:23.33 | Phunter | This is amazing. |
| 04:24.04 | Phunter | Thank you for helping me think through this. |
| 04:25.14 | [TK]D-Fender | You're welcome. |
| 04:47.24 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 05:03.27 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:06.26 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 05:11.33 | *** join/#asterisk SirLagz (~SirLagz@ppp121-45-253-228.lns20.per2.internode.on.net) |
| 05:12.00 | SirLagz | Why would my asterisk@home server suddenly not work until I reload hte configuration ? It's virtualised if that would affect anything |
| 05:16.32 | [TK]D-Fender | NAT mappings closing due to lack of upkeep. Reload reopening the hole briefly only to seal up again. |
| 05:16.38 | [TK]D-Fender | Basically misconfiguration of a system |
| 05:16.43 | SirLagz | hmm |
| 05:16.50 | SirLagz | NAT mappings on the router ? |
| 05:17.05 | SirLagz | or on the server ? |
| 05:19.17 | [TK]D-Fender | improper server setings, lack or appropriate forwardings, etc |
| 05:19.29 | SirLagz | I see |
| 05:20.20 | SirLagz | Would the virtualisation have anything to do with that ? I don't recall having the same issue when the server wasn't virtualised |
| 05:20.56 | SirLagz | however I did reinstall from scratch when virtualising it...guess I'll need to have a dig around |
| 05:21.05 | [TK]D-Fender | this is all networking.... |
| 05:23.34 | SirLagz | what is all networking ? the issue that I'm having ? |
| 05:25.19 | [TK]D-Fender | "virtual" is not some magic sauce. It works on a reload because it's opening a hole that is getting sealed up again. |
| 05:25.44 | SirLagz | [TK]D-Fender: yes I'm aware of that. I'm wondering why I was not having the same issue when the server wasn't virtual though |
| 05:26.03 | [TK]D-Fender | things aren't pointing where they did, how they did. |
| 05:26.21 | SirLagz | True. might need to double check my router settings then, in case I forgot something |
| 05:26.29 | [TK]D-Fender | So go prove that you have satisfied *'s requirements for how it is is networked. |
| 05:28.29 | *** join/#asterisk roentgen (~none@openvpn/community/support/roentgen) |
| 05:31.56 | [TK]D-Fender | bedtime.... back tomorrow... |
| 05:32.01 | SirLagz | night |
| 06:08.46 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:12.35 | *** join/#asterisk XATRIX (~xatrix@77.88.209.171) |
| 06:23.53 | SirLagz | Why do some things say I don't need to forward ports while other things say I do =/ |
| 06:28.47 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:49.12 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 07:06.29 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:20.32 | XATRIX | Hi, i'm using external commercial service to provide me PSTN->SIP lines. So, i did a trunk to connect to, and it actually registered successfully |
| 07:20.59 | XATRIX | sip.zadarma.com:5060 N 44898 105 Registered Fri, 18 Jul 2014 10:19:21 |
| 07:21.20 | XATRIX | But, i can't do any calls in/out. |
| 07:22.04 | XATRIX | I suppose it because my peer detects me with incorrect IP addr |
| 07:22.39 | XATRIX | When i go to zadarma.com and see the SIP number statistics. I can see a session from my PBX but : |
| 07:23.07 | XATRIX | IP: 127.0.0.1, port: 5080, useragent: Elastix 3.0 |
| 07:23.26 | XATRIX | I suppose it won't send calls to my PBX because of incorrect IP |
| 07:24.08 | XATRIX | How can i correct it ? I have a KVM virt. container with bridged interface. I have a real-IP , not NAT |
| 07:32.38 | XATRIX | Can someone help me ? |
| 07:32.55 | SirLagz | XATRIX: someone will, just need to be patient. Someone will come on and know what the issue is |
| 07:33.47 | XATRIX | Ok, i'll be waiting here. I simply stuck with :( |
| 07:34.02 | SirLagz | XATRIX: do you have "externhost" set in /etc/asterisk/sip_general_additional.conf ? |
| 07:34.30 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 07:34.59 | XATRIX | I have only /etc/asterisk/sip_general_custom.conf , no /etc/asterisk/sip_general_additional.conf |
| 07:35.02 | XATRIX | I'm using Elastix |
| 07:35.36 | SirLagz | XATRIX: ah, I'm not familiar with Elastix, but there's probably an option somewhere to set the external host to your external IP address somewhere |
| 07:35.54 | XATRIX | yeap. i'm currently searching for |
| 07:40.41 | XATRIX | Yeap, i've added externhost=176.111.63.189 - but i still see the same 127.0.0.1 on zadarma.com account |
| 07:41.07 | SirLagz | XATRIX: did you reload asterisk ? |
| 07:41.52 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:510a:d7fd:d4c9:2a8b) |
| 07:41.55 | XATRIX | yes, i've restarted the service |
| 07:42.56 | SirLagz | XATRIX: hang on, are you behind a NAT or does this box have a direct connection out to the internet ? |
| 07:43.43 | XATRIX | It has direct. Actually it's a proxmox hypervisor + KVM vitual container, and i have a bridged interface |
| 07:44.09 | SirLagz | XATRIX: Direct out to the internet ? So if I poked that IP address on port 5060 your asterisk box should respond ? |
| 07:44.34 | XATRIX | I suppose it should |
| 07:44.50 | SirLagz | ok, just checking I hadn't gotten confused then lol |
| 07:44.58 | SirLagz | In that case, I'm not too sure, sorry |
| 07:48.48 | XATRIX | Hm...yea, the same for me |
| 07:48.56 | XATRIX | I can't understand what's happening with it |
| 07:51.06 | *** join/#asterisk CeBe (~CeBe@port-92-206-29-51.dynamic.qsc.de) |
| 08:00.26 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 08:06.47 | *** join/#asterisk wonderworld (~ww@191-231.197-178.cust.bluewin.ch) |
| 08:08.19 | XATRIX | Ok, i'll try another problem. I've made a dialplan for number 1111 as it mentioned in this article |
| 08:08.47 | XATRIX | http://nettips.ru/article/sip_elastix_zadarma.html |
| 08:09.57 | XATRIX | But when i do a call to 1111 - simply nothing happens. No sound completely. Phoner shows me B3 connected, counts some seconds , in: ----, out: A-Law |
| 08:14.16 | *** join/#asterisk nix8n82 (~AndChat27@67-130-74-235.dia.static.qwest.net) |
| 08:15.55 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 08:16.12 | XATRIX | That's a log of my call from Phoner : http://ur1.ca/hrus9 |
| 08:16.14 | ChannelZ | If your asterisk is behind a firewall, you need to set externaddr and localnet accordingly under [general] in sip.conf at minimum (or externip if you're running an old version) |
| 08:16.17 | XATRIX | Maybe someone can help me ? |
| 08:16.34 | XATRIX | ChannelZ: No! It's not. I has a direct access |
| 08:16.42 | stevePearPear | hi, I used MixMonitor with the b option âOnly save audio to the file while the channel is bridged.â However if the call wasnât bridge, Asterisk still save a voice recording of 44 byte. is there anyway I could prevent Asterisk to not save? |
| 08:16.52 | ChannelZ | Then is the device you're testing with behind NAT? |
| 08:17.32 | XATRIX | Yes, my PC with Phoner is behind NAT. But i can do some calls on internal extensions |
| 08:17.40 | XATRIX | Without any problems |
| 08:18.16 | ChannelZ | Is Asterisk actually remote to you, or on your LAN? |
| 08:19.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:20.45 | XATRIX | ChannelZ: http://imgur.com/qTXSmoT |
| 08:21.04 | XATRIX | That's what i have. Take a look please on a last line on the left square |
| 08:22.04 | XATRIX | ChannelZ: it's remote server |
| 08:22.17 | ChannelZ | I have no idea what that even means. It's impossible for a remote host to see your IP as 127.0.0.1 so I don't know where it's getting that from. |
| 08:22.39 | XATRIX | 192.168.1.5[my pc] ---- 77.88.209.171[gw] ---- 176.111.63.189[asterisk] |
| 08:23.03 | XATRIX | yes, that's what i'm talking about |
| 08:23.19 | XATRIX | free-189*CLI> sip show registry |
| 08:23.19 | XATRIX | Host dnsmgr Username Refresh State Reg.Time |
| 08:23.19 | XATRIX | sip.zadarma.com:5060 N 44898 105 Registered Fri, 18 Jul 2014 11:21:36 |
| 08:25.03 | ChannelZ | I don't know why it seems to think your port is 5080 either |
| 08:25.16 | ChannelZ | It'd be more instructive to see a sip debug of your asterisk doing the register. |
| 08:26.27 | XATRIX | How can i do it ? |
| 08:27.21 | XATRIX | I mean, how can i retry register ? |
| 08:28.16 | ChannelZ | sip unregister |
| 08:29.10 | ChannelZ | then you might have to sip reload to get it to try again. I can't really recall, I don't register out to anyone anymore.. |
| 08:29.11 | XATRIX | It asks me a <peer> , which one should i set ? |
| 08:29.51 | ChannelZ | oh.. that's for incomings. Hmm. |
| 08:30.49 | ChannelZ | maybe you can just sip reload and it'll re-register the outgoing one. |
| 08:30.55 | XATRIX | free-189*CLI> sip unregister sip.zadarma.com |
| 08:30.55 | XATRIX | Peer unknown: 'sip.zadarma.com'. Not unregistered. |
| 08:31.11 | XATRIX | But the trunk is still active |
| 08:31.28 | ChannelZ | What do you mean.. I thought you said it wasn't working. |
| 08:35.33 | XATRIX | No... Look, I have trunk which is reported by Asterisk like Registered. Zadarma SIP provider detects me as 127.0.0.1 . And no calls can be done even on a test numbers to 4444 or 1111 |
| 08:37.04 | ChannelZ | which means nothing to me (4444 and 1111). I don't know if your PC softphone even works between you and Asterisk. Make a test extension that does Answer(), Playback(hello-world), and then Echo() and call it from your PC. Do you get bi-directional audio (you hear 'hello world' and then can yell into the phone and hear yourself echo'd back)? |
| 08:37.29 | *** join/#asterisk af_ (~af@93-43-45-195.ip90.fastwebnet.it) |
| 08:38.31 | XATRIX | Ok, ill try |
| 08:38.41 | XATRIX | Gimme some time to |
| 08:42.22 | XATRIX | ChannelZ: also, can you take a look, seems like i caught REGISTER messages from asterisk to trunk ISP |
| 08:42.24 | XATRIX | http://fpaste.org/118945/40567288/ |
| 08:43.28 | ChannelZ | "Reliably Transmitting (NAT) to (null):" I've never seen that before.. null.. uhhhh |
| 08:44.00 | XATRIX | me neither |
| 08:44.24 | ChannelZ | Can you pb your sip.conf only removing your passwords? |
| 08:44.54 | XATRIX | Ok |
| 08:45.04 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 08:45.23 | ChannelZ | (you can delete secret= lines, and replace the PW with XXXXX in your register => lines so we can see the exact syntax used) |
| 08:46.26 | XATRIX | http://fpaste.org/118946/40567316/ |
| 08:46.44 | XATRIX | Seems like bindaddr ? |
| 08:48.25 | ChannelZ | yes |
| 08:49.16 | ChannelZ | they seem to have it setup for local access only which explains a lot. |
| 08:49.18 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 08:49.52 | XATRIX | Maybe i should set 0.0.0.0 ? |
| 08:49.59 | ChannelZ | Just comment that and outboundproxy out |
| 08:50.21 | XATRIX | I'm not sure about proxy because of |
| 08:50.27 | ChannelZ | You could also comment out the two port lines but that's sort of less important |
| 08:50.37 | XATRIX | [root@free-189 asterisk]# netstat -tpln | grep 5060 |
| 08:50.37 | XATRIX | tcp 0 0 176.111.63.189:5060 0.0.0.0:* LISTEN 23922/kamailio |
| 08:50.37 | XATRIX | tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 23922/kamailio |
| 08:51.01 | XATRIX | So, i guess outboundpoxy should be set |
| 08:51.25 | ChannelZ | wait.. all this is running through kamailo?? |
| 08:51.37 | XATRIX | tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 22956/asterisk |
| 08:51.49 | XATRIX | Sounds if it is |
| 08:52.30 | ChannelZ | shit I don't know if I can help you in this context. Elastix has set this up in some specific fashion, you'd really need to get support from them as to what's going on. |
| 08:53.26 | XATRIX | damn.. |
| 08:53.30 | ChannelZ | this is not a vanilla Asterisk setup |
| 08:54.40 | XATRIX | Damn, now zadarma detects me as 0.0.0.0 port 5080... |
| 08:54.41 | XATRIX | O_O |
| 08:56.15 | XATRIX | Ok, i'll go get latest FreePBX |
| 08:56.23 | ChannelZ | well they are apparently using kamailo as a proxy, so the localhost bindport on behalf of Asterisk is probably right. But what kamailo is doing to the traffic (which is maybe nothing) I have no idea. I've never used it. |
| 08:56.35 | ChannelZ | That's almost worse. |
| 08:56.41 | ChannelZ | Do you really need some crazy GUI? |
| 08:57.18 | XATRIX | I have 0.5% experience in VoIP |
| 08:57.25 | XATRIX | So... Yes |
| 08:58.50 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 08:59.00 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 09:00.12 | ChannelZ | Well do what you like but we don't really support FreePBX here directly, for similar reasons, it covers up asterisk in specific ways and is not condusive external manipulation/direct configuration |
| 09:00.39 | XATRIX | Damn :( |
| 09:00.50 | ChannelZ | FWIW I wrote http://burner.com/asterisk-primer as a means to get stock Asterisk up and running. It's not as complicated as it looks. |
| 09:00.50 | XATRIX | Anyway thanks a lot for help! |
| 09:01.21 | XATRIX | I'll take a look at the link you gave me |
| 09:05.05 | ChannelZ | Good luck in your endevours. Sorry I can't help more but I've never touched Elastix or kamailo |
| 09:06.20 | XATRIX | No problem. I'll try to test FreePBX. But thanks for the help! You did much for me! |
| 09:06.20 | ChannelZ | there is an #elastix for what it's worth, whether you'll find anyone alive I dunno. |
| 09:06.39 | XATRIX | They have a few users, which didn't answered me :) |
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| 11:35.40 | XATRIX | Mostly my trunk returns me : SIP/2.0 407 Proxy Authentication Required |
| 11:35.57 | XATRIX | It means it can't auth on my side or i can't auth there ? |
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| 11:59.14 | SirLagz | XATRIX: where are you getting that message from ? |
| 12:00.45 | XATRIX | SirLagz: from sip set debug on ? |
| 12:01.10 | XATRIX | I don't know, it's always busy or call is dropped :( |
| 12:01.37 | XATRIX | I think somehow my trunk works, but calls can't reach the external VoIP ISP |
| 12:01.46 | XATRIX | I have echo test number 4444 |
| 12:01.52 | XATRIX | I can't call it via trunk |
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| 12:03.54 | SirLagz | XATRIX: can you call another extension on that asterisk box ? |
| 12:04.25 | XATRIX | 120->110 ? |
| 12:06.18 | SirLagz | XATRIX: are tjpse 2 exts on your asterisk box ? |
| 12:06.38 | XATRIX | sip show users: |
| 12:06.40 | XATRIX | directorIM_pbx.parkovka.u pbx.parkovka.ua Yes Yes |
| 12:06.41 | XATRIX | admin1_pbx.parkovka.ua pbx.parkovka.ua Yes Yes |
| 12:06.47 | XATRIX | Is it ok to call between ? |
| 12:06.57 | SirLagz | yeah those 2. can you call between those 2 ? |
| 12:07.00 | XATRIX | Yeap |
| 12:11.00 | SirLagz | XATRIX: is this asterisk box a vps or something ? |
| 12:11.35 | XATRIX | http://fpaste.org/118985/14056854/ |
| 12:11.54 | XATRIX | SirLagz: It's on a proxmox hypervisor, KVM virtual container |
| 12:12.10 | XATRIX | Network connection is bridged. It's not NATed or routed |
| 12:12.34 | XATRIX | brctl - used for bridging, and it has real-IP |
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| 12:14.00 | XATRIX | And this is happening while i try to call 4444 (VoIP ISP test number) |
| 12:14.14 | XATRIX | It should provide me an Echo() |
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| 12:18.26 | XATRIX | SirLagz: any ideas ? |
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| 12:21.56 | SirLagz | no sorry =/ |
| 12:23.07 | XATRIX | :( |
| 12:25.44 | XATRIX | SirLagz: please look at the 361 line http://fpaste.org/118986/05685566/ |
| 12:26.11 | XATRIX | Which one auth does it require ? |
| 12:27.05 | SirLagz | sorry don't have a browser at the moment, I'll take a look at it a bit later |
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| 13:01.38 | SirLagz | XATRIX: I don't understand why your asterisk server is trying to connect to 127.0.0.1... |
| 13:02.26 | [TK]D-Fender | Local script like FOP, iSymphony, etc |
| 13:03.06 | SirLagz | Ah. maybe that's why he's getting a proxy authentication thingo. |
| 13:03.30 | XATRIX | Yeap, there's a proxy |
| 13:03.56 | XATRIX | http://fpaste.org/119002/68862914/ |
| 13:04.02 | XATRIX | Thta's the ports for |
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| 13:04.27 | [TK]D-Fender | Ah, should have just asked for the earlier debug... |
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| 13:19.25 | Katty | Qwell: are you back yet?! |
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| 15:36.18 | gavimobile | lately I've been having intermittant problems when initiating calls. im not sure if its me or my tisp. i just logged a sip call which took 1 minute and thirty seconds for the phone to ring from the moment I initiated the call from my sip client. here is a link with sip set debug on before I made the call.. any help or leads on this would be great http://pastebin.com/YFhJ6LA8 |
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| 16:18.54 | lbazan | morning |
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| 16:20.02 | lbazan | in FOP panel I dont have activity in DADHI trunks, not blinking, nothing... :( someone who can help me! :-) |
| 16:27.09 | jameswf | Why would res_calender get 301, when curl gets 200. http://pastebin.com/VCwz7m4d |
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| 16:56.07 | cusco | since asterisk 11 I've been having a few: bad magic number for object 0x7f17ec601228. Object is likely destroyed. |
| 16:56.10 | cusco | and then asterisk dies |
| 16:56.12 | cusco | :( |
| 16:57.45 | newtonr | if you can reproduce the situation it happens it, you might consider reading through this article https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging and gathering the needed info to file a bug report. |
| 16:57.51 | newtonr | *it happens in |
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| 17:26.14 | bibz | hi there |
| 17:26.46 | bibz | I'm still getting these errors |
| 17:26.47 | bibz | [2014-07-18 19:20:27] WARNING[5050][C-000000a6]: res_odbc.c:608 ast_odbc_direct_execute: SQL Execute error! Verifying connection to mw1 [mw1]... |
| 17:26.47 | bibz | [2014-07-18 19:20:27] ERROR[5050][C-000000a6]: cdr_odbc.c:162 odbc_log: CDR direct execute failed |
| 17:27.06 | bibz | res_odbc is configured properly |
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| 17:37.23 | rrittgarn | bibz: i know i had issues when I was missing columns / had them in the wrong oder. I use adaptive_cdr with realtime though, so not sure if thats different than your implementation |
| 17:38.07 | bibz | thanks for the advice. columns should be right.. |
| 17:38.20 | bibz | i have migrated a server and since the migration it doesnt work anymore |
| 17:38.35 | bibz | everything works but cdr-related stuff |
| 17:40.06 | bibz | ]The INSERT statement conflicted with the FOREIGN KEY constraint "FK_Peer_Status_OnlineDatum". The conflict occurred in database "mw1", table "dbo.OnlineDatum", column 'Datum'. (196) |
| 17:40.41 | bibz | what does this error tell me? I mean I'm quite good at english, but I just don't understand it |
| 17:40.51 | bibz | the database structure hasn't changed.. |
| 17:45.45 | bibz | ok, fixed it. |
| 17:46.33 | bibz | was missing an entry in the "linked" table. |
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| 18:38.22 | onixx | good friday everyone. I have been searching google for an answer for my mwi delay and to no success so far. I have multiple SIP phones and on most, the MWI light lights up within seconds of leaving a message, however, I have 2 phones that take quite some time to show MWI. |
| 18:39.05 | onixx | I turned on SIP debug on one of those phone and noticed that no SIP message is transmitted to the effect of new messages once I hang up after leaving the message |
| 18:39.33 | onixx | It takes some time to happen.... but eventually does and MWI light up ! |
| 18:49.39 | Chainsaw | <irker291> gentoo-x86: chainsaw net-misc/dahdi: The pciradio.c file still used interruptible_sleep_on_timeout, which is no longer available. Upstream fixed it in May but the commit is not in any release yet. This will now cope with 3.15 kernels. |
| 18:49.52 | Chainsaw | Perhaps it's time folks? 2.9.1 or something? |
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| 19:49.48 | Spartak | Hi. Need urgent help on restoring from backup, asterisk freepbx |
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| 20:11.04 | bibz | I'm getting "busy" on all calls.. what could it be?.. |
| 20:19.53 | [TK]D-Fender | Anything |
| 20:20.06 | [TK]D-Fender | If you'd like us to narrow it down, show us the call |
| 20:21.58 | bibz | ok, hold on. |
| 20:22.14 | bibz | its pretty weird.. I didn't change anything in the config.. calls worked for 2-3 hours |
| 20:22.19 | bibz | and now NO call gets through anymore |
| 20:22.25 | bibz | I'll PB the call, give me a sec |
| 20:25.40 | rjblack | Question from a newbie, Iâm going through the Digium Switchvox training and it mentioned this resource board. Are there any caveats that anyone wants to let me know about. (I am an expert in Avaya IP Office, NEAX 2000, and the HiPath 3000) |
| 20:27.18 | [TK]D-Fender | rjWe don't support SwitchVox here. |
| 20:27.28 | [TK]D-Fender | It's a closed GUI |
| 20:28.45 | bibz | ok, I rebooted now.. |
| 20:28.57 | bibz | can't record the call |
| 20:30.42 | [TK]D-Fender | checkout time, BBIAB |
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| 20:34.09 | bibz | http://pastebin.com/qqPqTGKK |
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| 22:09.45 | tom______ | have a starnge question, I have an asterisk server(currant) that send a call to another server. When the orig hits # I need it to do a goto. I have tried adding it to feature.conf but no luck. Can someone point me in the right direction |
| 22:14.13 | [TK]D-Fender | Hit WHEN is important... |
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