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| 08:05.54 | Zogot | Im trying to use the default features.conf [featuremap] but after I input my first DTMF input after being asked to input an extension to transfer to |
| 08:06.08 | Zogot | it immediately says its not a valid extension, it doesn't let me put in more than 1 dtmf |
| 08:06.17 | Zogot | tried in both 13.3 and 13.8 |
| 08:13.51 | Zogot | question, what context will the extensions attempted to be matched in, when using a DTMF transfer? |
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| 08:28.47 | Zogot | ok, figured out its the context the phone is in |
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| 08:46.10 | joj[] | Is it still possible to use users.conf with res_pjsip? |
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| 08:47.31 | joj[] | Nevermind, there is no reference in 13.9.1 source, so I guess the answer is no. |
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| 09:53.43 | [[thufir]] | in provisioning a Cisco IP phone, it comes with an entry for the dialplan: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) why is there a dialplan in a phone, and what does this dialplan do? |
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| 10:56.24 | MajesticFudgie | Any common reasons why the SPA3000 wouldn't be forwarding caller ID? |
| 10:56.50 | MajesticFudgie | It does show the caller id on the admin page for the webgui |
| 10:56.58 | MajesticFudgie | But asterisk simply doesn't recieve it |
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| 11:07.07 | mohan | Hi I need to write the recorded filename into the asterisk db can you please suggest a solution |
| 11:07.51 | mohan | I use goautodial with asterisk |
| 11:08.16 | mohan | in the asterisk db there is vicidial_list which it updates call status |
| 11:08.34 | mohan | along with call status I would also want the recorded file |
| 11:08.45 | mohan | please help |
| 11:15.39 | Zogot | is there no way to define the context to attempt the dtmf transfer in? |
| 11:15.39 | mohan | this post answered my question http://stackoverflow.com/questions/28942659/asterisk-call-recordings-to-mysql-database |
| 11:30.52 | mohan | and there is lead_id in vicidial_list which I want to access from dialplan |
| 11:31.05 | mohan | is there a way? |
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| 11:37.02 | mohan | ok i think i can update that with phone no which exists in the context |
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| 11:45.30 | mohan | before that step i want first call records in month wise folder how can i achieve that |
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| 13:06.31 | mohan | does freepbx writes call recording file location to databse |
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| 13:10.11 | Suisso | Hello, what is the best tool (.agi) "Text-to-Speech (TTS)" for Asterisk ? I tried to use the Google TTS (googletts.agi) but does not work. Can anyone help me please? Thank you. |
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| 13:23.40 | [TK]D-Fender | Google TTS works |
| 13:23.54 | [TK]D-Fender | "does not work" = did it wrong |
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| 13:24.47 | [TK]D-Fender | Other options are festival, cepstral, etc |
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| 13:48.01 | Zogot | [TK]D-Fender, Suisso: there is microsoft tts, mstts made by the same person who made googletts |
| 13:48.06 | Zogot | its what we use now |
| 13:48.16 | Zogot | official api also, better than the little hack that was googletts |
| 13:48.43 | markallen | does ast_frame::data::ptr always contain raw audio data when AST_FRAME_VOICE == ast::frame::frametype? |
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| 15:01.05 | jrun | pbx_function.so is missing under /usr/lib/asterisk/modules so loader.c see complains. installations is based off 13.9.1. any idea? |
| 15:03.20 | jrun | autoload is set to no. modules.conf comes from 'make smaple'. |
| 15:08.34 | mjordan | jrun: What's the specific error that you're getting? pbx_functions is part of the core, and isn't a loadable module. |
| 15:10.24 | jrun | https://gist.github.com/27ffa3a7d1529ba489a0914f41813410 |
| 15:10.29 | jrun | mjordan: ^ |
| 15:11.25 | mjordan | are you telling it to load pbx_functions? |
| 15:11.48 | mjordan | same with res_pjsip_log_forwarder.so - that module was removed awhile back |
| 15:12.37 | jrun | mjordan: modules.conf was written by 'make sample' |
| 15:13.02 | jrun | an yes it's explicitly asking for those to be loaded. i guess i can remove them from modules.conf, correct? |
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| 15:19.54 | mjordan | yes |
| 15:20.23 | mjordan | pbx_functions isn't a loadable module; the other one was removed. |
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| 15:20.55 | mjordan | modules.conf.sample - at least in the version I'm looking at (13.10.x), doesn't explicitly load any modules. So if those directives are in there, someone added them. |
| 15:22.10 | jrun | 13.9.1 is marked LTS (topic) so i based all work off of that. |
| 15:22.34 | jrun | i could move to 13.10 series if it helps. |
| 15:22.42 | mjordan | No, the file is the same between the versions. |
| 15:22.43 | jrun | not sure about the dev cycle here. |
| 15:22.48 | jrun | oh |
| 15:23.01 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 15:23.36 | mjordan | 13.10 is just the next bug fix release in the same major version. |
| 15:24.12 | mjordan | ah ha |
| 15:24.25 | mjordan | modules.conf in basic-pbx *does* explicitly load those two modules. |
| 15:24.29 | mjordan | That's a mistake in that config file. |
| 15:24.29 | jrun | so for production i guess we should stay on 13.x series since that's where the bug fixes go to |
| 15:24.53 | mjordan | jrun: 13 is an LTS version that is still under bug fix support. 11 is as well, but not for much longer. |
| 15:25.28 | jrun | hence our move to 13 actually... we're running away from 11 and chan_sip |
| 15:28.02 | mjordan | cool |
| 15:32.40 | jrun | btw, i'm not getting the old school gray/blue menuconf; it's the new nmenconf which is a bit hard to read that i get. any idea how to switch back? |
| 15:33.50 | jrun | i was hoping for something cool like nconfig (kernel's). |
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| 16:35.17 | MajesticFudgie | Any common reasons why the SPA3000 wouldn't be forwarding caller ID? My provider does send it, but for some reason the SPA3000 doesn't relay it to Asterisk, it shows on the SPA web gui as the last PSTN Caller in the format ", 0XXXXXXXXXX" |
| 16:36.24 | [TK]D-Fender | Show us the call. |
| 16:36.51 | MajesticFudgie | From asterisk? |
| 16:37.03 | [TK]D-Fender | Obvious first step |
| 16:40.39 | MajesticFudgie | Is there a way I can filter out the crap |
| 16:41.01 | MajesticFudgie | Finding it hard to get hold of the call logs as its in with all the other ongoing stuff |
| 16:41.05 | MajesticFudgie | reload |
| 16:41.08 | MajesticFudgie | w/c/ |
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| 16:42.23 | MajesticFudgie | http://paste.ubuntu.com/18035817/ is capture from Sngrep |
| 16:42.27 | mohan | hi good morning, good afternoon, good evening. I have 1 small question please clarify |
| 16:43.36 | [TK]D-Fender | Go look at your SPA settings |
| 16:43.53 | MajesticFudgie | What do I need to look for in them |
| 16:44.38 | [TK]D-Fender | Anything blatently labelled callerid |
| 16:45.02 | mohan | in asterisk there is a function MixMonitor in which there is also an option to specify command (.sh) want to understand from the process flow point that if the command fails would it interrupt asterisk dialing system |
| 16:45.25 | MajesticFudgie | In PSTN Line settings theres, VoIP Caller ID Pattern and PSTN Caller ID Pattern |
| 16:45.52 | [TK]D-Fender | picpaste.com < ALL the screens..... |
| 16:46.06 | [TK]D-Fender | Actually.. your PEER first |
| 16:46.07 | [TK]D-Fender | ^ |
| 16:46.21 | MajesticFudgie | Peer? |
| 16:46.22 | [TK]D-Fender | lets see what you made for this |
| 16:46.54 | mohan | this is important because i dont any system failure due to this extra feature i we want to add |
| 16:49.48 | MajesticFudgie | https://drive.google.com/open?id=0B3OjtkyPZhgsbUl3STQwNUZaRGM |
| 16:50.02 | MajesticFudgie | Screenshot of every page of my SPA Gui |
| 16:51.02 | MajesticFudgie | My providers BT UK btw |
| 16:51.19 | MajesticFudgie | Some of the settings are set to work with them |
| 16:54.46 | [TK]D-Fender | <[TK]D-Fender> Go look at your SPA settings |
| 16:54.51 | [TK]D-Fender | <[TK]D-Fender> Anything blatently labelled callerid |
| 16:55.00 | [TK]D-Fender | You've failed this.... |
| 16:55.10 | [TK]D-Fender | read your PSTN screen again |
| 16:57.03 | MajesticFudgie | I only see the ones I said about earlier |
| 16:57.09 | MajesticFudgie | What am I failing to notice? |
| 16:57.56 | [TK]D-Fender | <MajesticFudgie> In PSTN Line settings theres, VoIP Caller ID Pattern and PSTN Caller ID Pattern <- is see several others |
| 16:57.56 | MajesticFudgie | At least CTRL+F for Caller ID only shows those two |
| 16:58.32 | [TK]D-Fender | READ THE DAMN SCREEN |
| 16:58.37 | MajesticFudgie | I AM |
| 16:58.41 | MajesticFudgie | What are you referring to |
| 16:58.50 | [TK]D-Fender | ------------------------> CID <----------------------- |
| 16:59.00 | [TK]D-Fender | CALLER ................. ID\ |
| 16:59.11 | MajesticFudgie | Why didn't you just say look for CID |
| 16:59.19 | MajesticFudgie | How am I supposed to know CID stands for Caller ID? |
| 16:59.41 | [TK]D-Fender | This is the standard acronym |
| 16:59.47 | MajesticFudgie | Well I didn't know it |
| 16:59.52 | MajesticFudgie | Now I do |
| 16:59.55 | MajesticFudgie | I've enabled it and it works |
| 17:01.38 | jrun | what's the max limit for qualify_frequency? |
| 17:01.48 | jrun | in fact which header defines pjsip.conf options? |
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| 17:41.21 | mjordan | jrun: it's a bit more complex than that... |
| 17:41.37 | mjordan | jrun: but if you want to see where all that gets set up, *most* of it is in pjsip_configuration.c |
| 17:42.49 | mjordan | although, since that's an AoR setting, it's actually in res/res_pjsip/location.c |
| 17:43.00 | mjordan | https://github.com/asterisk/asterisk/blob/master/res/res_pjsip/location.c#L1148 |
| 17:44.28 | mjordan | max value is 86400. It'd be nice if 'config show help res_pjsip aor qualify_frequency' displayed that. Hrm. |
| 17:45.48 | jrun | (qualify_frequency == qualify) ? |
| 17:49.31 | mjordan | it is essentially the replacement for that option, yes |
| 17:49.38 | mjordan | 0 is 'just don't bother' qualifying |
| 17:49.46 | mjordan | any other value sets up the frequency at which you qualify the endpoint |
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| 18:08.48 | jrun | mjordan: thanks |
| 18:09.13 | jrun | sip_to_pjsip.py marks it as un-mapped though |
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| 18:13.18 | MapleDev | Hi |
| 18:18.16 | MapleDev | I was following this wiki page: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject specifically the Bundled Version part, but when I run ./configure --with-pjproject-bundled I receive this error: www.hastebin.com/ebayozegeq.vbs |
| 18:18.49 | MapleDev | A couple of guys had a similar issue in comments section, but I already have bzip2 installed |
| 18:19.10 | pa | frikking GV alive again |
| 18:19.16 | pa | wtf is going on @GOogle... |
| 18:20.11 | pa | is there anything equivalent to #ifdef for extensions.conf |
| 18:20.12 | pa | ? |
| 18:22.23 | [TK]D-Fender | What's your actual goal? |
| 18:25.46 | MapleDev | Mine is to install Asterisk from source |
| 18:26.55 | [TK]D-Fender | pa <- |
| 18:28.02 | pa | to use one or another line(instruction, or how are they called in extension.conf) depending on a variable |
| 18:28.21 | pa | s/use/enable |
| 18:29.01 | [TK]D-Fender | "core show application execif", "core show application gotoif" |
| 18:37.36 | Mimiko | Hello. I'm novice to asterisk. Recently I've to manage a client's asterisk and there where running 4 instances of asterisk. My question to comunity - why running on same phisical server 4 instances insteed of only one? |
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| 18:39.36 | shido6 | I can only speculate without further info Mimiko -- one would be that the decision was made out of lack of knowledge of how to build a proper multi tenant solution |
| 18:40.09 | *** join/#asterisk ipengineer (~zconkle@71.252.134.63) |
| 18:40.40 | Mimiko | shido6, maybe, as the manager said that previous admin maid this because one instance had some limitations. As all instances are configured the same except the port |
| 18:40.58 | Mimiko | the manager does not know why is that situation |
| 18:41.17 | shido6 | have you inherited this system, Mimiko ? |
| 18:41.34 | Mimiko | shido6, yes |
| 18:42.07 | shido6 | well, congratulations! its a clusterphuck! :) |
| 18:43.07 | shido6 | build a new system on another box and migrate as needed or put all new customers on the new box. Spend some time reading through the documentation and examples, ask questions here after you've read the documentation 7 times. |
| 18:43.23 | shido6 | enjoy! |
| 18:43.38 | Mimiko | shido6, yes that is a solution, but I had to convince manager to do this. its a production |
| 18:43.48 | shido6 | dont mess with prod |
| 18:43.57 | shido6 | play on your dev box |
| 18:43.58 | Mimiko | Did realy ver. 1.4 or 1.6 had some limitations per instance? |
| 18:44.15 | shido6 | the limitation is the developer |
| 18:44.26 | Mimiko | ah, ok then |
| 18:44.28 | shido6 | or the person who built that system |
| 18:44.58 | Mimiko | As I sussposed, all 4 instances will bit for resources on the server - no better performance |
| 18:51.31 | *** join/#asterisk areski (~areski@80.174.128.68.dyn.user.ono.com) |
| 18:58.47 | *** join/#asterisk sarge444 (~sarget@50-201-222-202-static.hfc.comcastbusiness.net) |
| 18:58.59 | sarge444 | <sarget> trying to mount sda2 on an elastix instalation drive on an unbuntu machine so i can delete some monitors. it has taken up the whole drive |
| 18:58.59 | sarge444 | <sarget> sudo mount /dev/sda2 /mnt it says cant do it |
| 18:58.59 | sarge444 | <sarget> but i can mount the boot partition sda1 |
| 18:58.59 | sarge444 | <sarget> why?? |
| 18:58.59 | sarge444 | <sarget> what am I doing wrong? |
| 19:00.34 | sarge444 | says lvm2_member cant mount |
| 19:00.42 | sarge444 | anyony here? |
| 19:01.59 | *** join/#asterisk ipengineer (~zconkle@71.252.134.63) |
| 19:10.25 | ||cw | sarge444: if it's lvm you need to import the vg. that's more linux/lvm specific, your distro support channels can help wit that |
| 19:10.52 | ||cw | but I'd start with "man vgimport" |
| 19:11.09 | shido6 | Mimiko - the instances are a non standard way to "partition" |
| 19:11.44 | Mimiko | shido6, to partition what? |
| 19:11.48 | shido6 | a service for multiple clients - its not horrible - |
| 19:12.15 | Mimiko | one instance cant do this? |
| 19:12.57 | shido6 | when instance 1 takes a break u have 3 other instances running so how many customers have you saved? - but if the box takes a break, well you're planning a slightly different day |
| 19:13.27 | shido6 | I would rethink that bcp plan |
| 19:13.50 | Mimiko | isnt better to have different phisical boxes for this? |
| 19:14.13 | shido6 | I dont know whats happening there so I cant answer that without "evidence" of whats running - what service(s) are being offered? |
| 19:15.51 | shido6 | are there any comments in the dialplan or notes or runbook that came with your inheritence? |
| 19:16.06 | Mimiko | clients have ids, each client has an id only in one instance, clients can call and receive calls |
| 19:16.21 | shido6 | is there a web interface? |
| 19:16.22 | Mimiko | shido6, nope, just the box and password |
| 19:16.31 | Mimiko | shido6, no interface, only cli |
| 19:17.18 | shido6 | do you want to consolidate it into a single box ? ( again don't "tamper" with prod ) |
| 19:29.05 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 19:29.09 | xochilpili | hi all |
| 19:30.21 | xochilpili | i have some strange behavior, i have created a new account (i had only one), and both accounts are registered as same ip in asterisk; then i create another extension, and register to the same asterisk, and again same ipaddress, which is not my public ipaddress, how is this possible? |
| 19:32.26 | shido6 | are these sip accounts on the asterisk system you are referring to? |
| 19:32.44 | shido6 | what has the same ip address? |
| 19:33.45 | shido6 | you can have several phones behind a router registering to a single asterisk server's ip address - |
| 19:33.57 | shido6 | can you be more specific xochilpili ? |
| 19:34.07 | shido6 | would you please be more specific, xochilpili ? |
| 19:34.19 | xochilpili | shido6, please, wait |
| 19:34.22 | xochilpili | let me paste |
| 19:34.24 | shido6 | whats the problem? |
| 19:34.25 | shido6 | NO |
| 19:34.28 | shido6 | pastebin.com |
| 19:34.30 | shido6 | please |
| 19:34.33 | xochilpili | yes, pastebin |
| 19:35.00 | xochilpili | shido6, what should i paste? |
| 19:38.30 | xochilpili | shido6, ? |
| 19:39.13 | shido6 | evidence of the problem |
| 19:39.20 | shido6 | sip debug |
| 19:39.38 | shido6 | are you having registration issues? |
| 19:40.27 | xochilpili | no, i can register, but in the sip show peers, 3 extensions has the same ipaddress which is not my public ipaddres |
| 19:41.11 | xochilpili | i give to my friend an extension, he is in other state, so, in the sip show peers, he has the same ipaddress |
| 19:41.43 | xochilpili | i dont know if im being clear, im not an english native |
| 19:42.32 | shido6 | show that in the pastebin |
| 19:42.36 | shido6 | sip show peers |
| 19:42.42 | xochilpili | oks |
| 19:42.47 | shido6 | and add a question in there with some arrows :) |
| 19:43.19 | shido6 | in case I get hit by a bus someone here will be able to help you as long as you paste that pastebin url here |
| 19:45.06 | xochilpili | shido6, yes, thanks, i deleted everything before in order to check iptables, fail2ban, asterisk conf, etc... |
| 19:45.21 | xochilpili | im creating again the extensions, in order to prove the issue and paste |
| 19:47.43 | xochilpili | shido6, http://pastebin.com/7kRbZhVf |
| 20:05.26 | shido6 | back |
| 20:06.24 | shido6 | thats a good damn question. |
| 20:06.43 | shido6 | xochilpili is the grandstream on the same subnet? |
| 20:06.55 | shido6 | ifconfig -a |
| 20:07.03 | shido6 | err hold on - |
| 20:07.59 | shido6 | curl ifconfig.co |
| 20:08.08 | shido6 | and add that to the pastebin |
| 20:08.27 | shido6 | do you have multiple wan interfaces on the edge? |
| 20:10.05 | shido6 | is the cell on the same subnet ( via wifi? ) |
| 20:13.15 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:13.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 21:01.48 | xochilpili | shido6, i solve it |
| 21:02.40 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
| 21:02.47 | xochilpili | the issue was that grandstream phone, i dont know why was registering that ip, so i search into my history router log, and i realize that that ip was my last one assigned from my ISP |
| 21:03.20 | xochilpili | i just reboot grandstream, remove the account of all lines, and restart asterisk directly from services, and now it's working ok |
| 21:03.31 | xochilpili | shido6, thanks for your time |
| 21:16.18 | *** part/#asterisk markallen (~mhorila@u217-44.talkpoint.com) |
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| 22:16.24 | lvlinux | n |
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| 23:22.22 | *** part/#asterisk kharwell (kharwell@nat/digium/x-asltzorjqsbrosae) |
| 23:31.47 | *** join/#asterisk freepbx_q (~freepbx_q@205.168.217.70) |
| 23:32.05 | *** join/#asterisk shido6 (~textual@CPE00fc8d94d3f3-CM00fc8d94d3f0.cpe.net.cable.rogers.com) |
| 23:32.22 | freepbx_q | Greetings, I have a question about Digium D40âs, the Presence state, and âUnavailable" |
| 23:32.39 | freepbx_q | I have actually turned off the presence module, just FYI |
| 23:33.03 | freepbx_q | But I have one extension that shows unavailable on the Digium phone, but it receives calls |
| 23:33.10 | freepbx_q | Also able to make calls. |
| 23:34.17 | freepbx_q | Using FreePBX 12.0.76.3 |
| 23:34.26 | freepbx_q | With current modules as of today. |
| 23:34.49 | freepbx_q | Does anyone have any idea where to look? Iâve tried suggestions from here: |
| 23:34.50 | freepbx_q | FreePBX 12.0.76.3 |
| 23:34.55 | freepbx_q | Oops |
| 23:34.59 | freepbx_q | http://community.freepbx.org/t/fix-for-stuck-status-unavailbable-offline-with-digium-phones/24920 |
| 23:35.12 | freepbx_q | and Here: http://community.freepbx.org/t/digium-dpma-pressence-messed-up-since-upgrade/24867 |
| 23:38.00 | *** join/#asterisk Emily_ (0c11aa15@wikipedia/RandomStringOfCharacters) |
| 23:38.11 | shido6 | turn the module back on - wait a few minutes and see if it changes - then disable it again - if it were me , I would block all other phones in iptables then perform the action - then watch the phone toggle - and disable it again and remove the block |
| 23:38.20 | shido6 | but thats me, they say I'm "cray-cray" |
| 23:38.56 | freepbx_q | Interesting idea. Thanks, Iâll try it. |
| 23:39.04 | Emily_ | hey, I've got an asterisk instance set up, and I can route calls between peers, and sip trunks, etc... but if I attempt to Playback(hello-world) nothing happens -- any ideas? |
| 23:39.27 | shido6 | yeah - show us the call. Use pasebin.com |
| 23:39.34 | shido6 | sip debug , please :) |
| 23:40.26 | shido6 | does hello-world exist as a sound file? - whats asterisk tell you when debug is set to 5 and verbose is set to 5 ? |
| 23:43.37 | Emily_ | yes hello-world.gsm exists, and asterisk seems to detect that it is gsm at the very least, since if the file is not there / or in a different format it complains |
| 23:45.51 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:47.22 | shido6 | thanks for the play by play - but... I'm afraid I need to see the evidence. |
| 23:48.04 | shido6 | show that part of the dial plan - and the debug |
| 23:48.39 | shido6 | please & thank you |
| 23:53.16 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 23:54.39 | Emily_ | http://pastebin.com/Jnm1pyVw |
| 23:55.37 | Emily_ | shido6: and the dialplan section: http://pastebin.com/GVDqgHYg |
| 23:56.05 | Emily_ | it executes the waits successfully -- I'm sort of assuming it's some codec issue, but the logs are not particularly friendly to someone new to the software :/ |
| 23:56.36 | shido6 | ok - so now show the sip debug |
| 23:57.10 | Emily_ | did that not contain the sip debug? |
| 23:57.17 | shido6 | no |
| 23:57.33 | shido6 | 5 lines od dialplan |
| 23:57.34 | shido6 | oh |
| 23:57.38 | shido6 | sorry |
| 23:58.39 | shido6 | holy codecs batman |
| 23:58.39 | Emily_ | that said, I think I may have copied the wrong section of logs... either that or pastebin screws up browser find |