| 00:06.30 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 00:14.35 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 00:14.48 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 00:16.19 | WiretapSeven | Hey guys, when someone calls one of my queues, every time the queue hits a busy agent, it tries to go to voicemail, then a DTMF tone is heard and this appears in the logs |
| 00:16.25 | WiretapSeven | <PROTECTED> |
| 00:16.26 | WiretapSeven | <PROTECTED> |
| 00:17.16 | WiretapSeven | when I used to use trixbox there was no problem but using asterisk/freepbx there is? |
| 00:47.36 | atan | Is there a way to not repeat myself on every line? I have exten => _1NXXNXXXXXX, dozens of times. Can I replace that with something that says "use what you see above" ? |
| 00:50.51 | ariel_ | don't understand your question, you make the dial plan it's up to you |
| 01:01.34 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
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| 01:04.37 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 01:08.20 | seraphie | atan: You could do something like exten => _1NXXNXXXXXX,1,GoTo(context,ld,1) Then execution would continue at an extension named "ld". However, yes, you do have to type the extension name every time. |
| 01:09.29 | atan | seraphie, thanks |
| 01:09.43 | atan | ariel_, it's just to avoid typing much like 'n' is |
| 01:09.59 | Andrew__M | atan: You can also call a Macro from a single line. But if you give us more details, we can understand your problem better. |
| 01:11.49 | Andrew__M | n is more to avoid renumbering if you insert or delete a line. |
| 01:12.35 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 01:13.18 | atan | Andrew__M, so I have like 10 numbers inside my inbound context |
| 01:13.32 | atan | When I add a new number it involves typing another 10 lines to setup the things I want |
| 01:13.41 | atan | I'd prefer not to type the same number over and over again |
| 01:14.03 | Andrew__M | Create a Macro, and call it each time. |
| 01:14.24 | atan | I wonder if it would be a totally awful idea to query a database on every incoming call to see how to handle it as opposed to the static file |
| 01:14.58 | atan | Then I could hide my functions directly within the mysql database and change it without a reload each time. |
| 01:15.05 | Andrew__M | You can use AstDB. It's built-in and relatively easy to use. |
| 01:15.34 | atan | Can I store Dial("SIP/1234") inside sql and have Asterisk run it once it have it stored in a ${var}? |
| 01:15.56 | Andrew__M | You can use mysql as well, but for 10 lines probably the Macro is easiest. |
| 01:16.43 | atan | Well it's 10 lines now. I want to build on so users can click into a web interface to update things like call forwarding and such. I just query for an array of actions and run them one-for-one that would rock my boat. |
| 01:16.53 | Andrew__M | Sure. But this is a perfect candidate for a Macro. |
| 01:17.06 | atan | Andrew__M, how does a user remotely update a macro? |
| 01:17.32 | atan | The page they change account details on is just an apache box that connects up to mysql with php. |
| 01:17.53 | ariel_ | atan: same way they do your 10 lines now |
| 01:17.54 | Andrew__M | I don't know. But they can update the AstDB, even from a voice menu. |
| 01:23.33 | atan | Now is there a way I can execute something store in a ${var}? Say ${var} = Dial('SIP/123', 30); |
| 01:24.09 | ariel_ | yes |
| 01:24.25 | ariel_ | dial(${var}) |
| 01:25.27 | atan | Dial is inside the ${var} |
| 01:25.49 | atan | The ${var} is actually "Dial(...)" or some other function |
| 01:26.21 | ariel_ | dial is a command |
| 01:27.37 | *** join/#asterisk nix8n82 (~nate@24.143.27.157) |
| 01:30.03 | atan | Yes exactly |
| 01:30.24 | atan | I want to store the commands/functions whatnot inside a ${var} to execute |
| 01:30.34 | atan | Not just the var it needs, the entire function |
| 01:31.11 | atan | I suppose I could just have a ton of vars and a ton of "if" functions |
| 01:31.35 | atan | If ${dial} exists Dial(${dial}, 30) |
| 01:31.53 | atan | If ${voicemail} exists Voicemail($...) |
| 01:32.22 | atan | But I'd prefer to just take my list of functions and execute them foreach in the order I have them in my database. |
| 01:33.08 | atan | If you know php this might make more sense I suppose: <?php $var = 'shutdown -h now'; exec($var); ?> |
| 01:34.28 | ariel_ | atan: yes but you still need some type of command |
| 01:34.34 | ariel_ | to do what you need |
| 01:35.37 | atan | So there's no way inside the dialplan to execute functions on the fly? |
| 01:36.17 | ariel_ | yes but you need commands |
| 01:36.57 | ariel_ | I use mysql in my dial plans all the time, it checks for valide pins, correct channels, I route them depending on what it needs and many different things |
| 01:37.17 | ariel_ | I also use it for my least cost routing and IVR's |
| 01:37.27 | atan | Set(action="Dial("SIP/123", 30)") then can I just NNN,n,${action} ? |
| 01:37.30 | ariel_ | but commands are just that some of them your going to have to use |
| 01:39.13 | atan | Hmm, nope that didn't fly. pbx_extension_helper: No application '${action}' for extension |
| 01:40.10 | ariel_ | lol, what application = dial are you going to use? |
| 01:44.37 | atan | Right now dial, but it could be swapped for voicemail or playback |
| 01:45.02 | atan | Exec() looks interesting. Is Exec(Dial()) how one might use it? |
| 01:45.23 | atan | "Allows an arbitrary application to be invoked even when not hardcoded into the dialplan." |
| 01:46.42 | atan | Ahh! Golden. Exec is just what I need. |
| 02:23.46 | *** join/#asterisk radic (~radic@dslb-178-002-211-177.pools.arcor-ip.net) |
| 02:52.45 | atan | http://www.voip-info.org/wiki/view/Asterisk+fax under "Can Asterisk act as a fax/voice switch? " can I use that with SIP somehow? |
| 02:53.00 | atan | I added answer() and a fax extension but it didn't pick it up like one might have expected. |
| 02:54.01 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 02:54.37 | WiretapSeven | atan, you're avoiding freepbx for a reason I assume, it sounds like it'd make your particular case much easier |
| 03:16.39 | atan | WiretapSeven, it's all gui-ish so I just stayed away. |
| 03:17.07 | WiretapSeven | atan, an aversion to GUI is no healthier than an aversion to CLI, each has their place |
| 03:17.14 | atan | When you grab a new version of asterisk and make menuselect does it already know what you had selected previously or do you need to go in and do it all up again? |
| 03:17.36 | WiretapSeven | atan, it will not automatically know |
| 03:18.55 | atan | Is there a way to save your old settings and have it pre-selected? |
| 03:21.00 | atan | Oh heck there isn't that many I suppose I can just go into each list. |
| 03:25.34 | *** part/#asterisk atan (~atan@unaffiliated/atan) |
| 03:38.25 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:39.01 | atan | When I take a call on say 555 and then send them to another extension is there a way to pass along some vars? |
| 03:39.42 | atan | In my case it's a fax. Comes in, gets detected and sent to fax,n,... but it doesn't have any of the details I need which were in the previous extension. |
| 03:40.48 | atan | Hmm "Prepending a single _ character to a variables name in SetVar will cause that variable to be inherited by channels created by the main channel." might have found my answer |
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| 07:55.48 | _omer | hello, I need to know how to run multiple asterisk instances in same computer ... |
| 07:57.05 | _omer | I have just read that I can do it with -c and changing directories from /etc/asterisk/asterick.conf but how to connect to Asterisk CLIs differently ? |
| 07:59.06 | gavimobile | _omer: wow, that sounds wild |
| 07:59.56 | gavimobile | misubah |
| 08:01.49 | _omer | misubah? |
| 08:03.25 | gavimobile | means complicated |
| 08:03.41 | gavimobile | for me at least |
| 08:04.39 | _omer | for me either :) |
| 08:04.56 | gavimobile | is omer really your name |
| 08:05.01 | _omer | yes |
| 08:05.05 | gavimobile | isreali? |
| 08:05.14 | _omer | Pakistani |
| 08:05.24 | gavimobile | I was sure it was middleeast |
| 08:05.33 | gavimobile | :-p |
| 08:05.34 | _omer | :D |
| 08:05.50 | _omer | is gavimobile your real name ? ;-) |
| 08:05.57 | gavimobile | I don't even know how to run apache in cgi mode |
| 08:06.11 | gavimobile | gavi is |
| 08:06.19 | _omer | is everybody sleeping here? |
| 08:06.29 | gavimobile | in the usa prob |
| 08:06.45 | gavimobile | try asking in freepbx or another asterisk related room |
| 08:06.49 | gavimobile | maybe they can help |
| 08:06.51 | _omer | apache thing sounds complicated to me |
| 08:07.09 | _omer | how about voice recognition ? |
| 08:07.11 | *** join/#asterisk teathsch (~chatzilla@108-73-146-32.lightspeed.irvnca.sbcglobal.net) |
| 08:07.18 | gavimobile | your trying to have asterisk several times on one machine |
| 08:07.25 | _omer | yes |
| 08:07.41 | gavimobile | I don't know how to help with that, but I don't understand why you would want to do that |
| 08:08.25 | _omer | I want to run asterisk-realtime with 10 different databases ... |
| 08:08.41 | _omer | I think I can use different directories per instance |
| 08:08.57 | _omer | using different sip , rtp and iax ports |
| 08:09.03 | gavimobile | sounds like your doing some hard core stuff there man |
| 08:09.18 | gavimobile | hurting my brain just reading |
| 08:09.29 | _omer | :D |
| 08:10.05 | _omer | I couldn't find any help ... neither on voice recognition......I wish if I had to work on extensions simply :-/ |
| 08:10.22 | gavimobile | there are many asterisk channels |
| 08:10.24 | teathsch | what's the preferred pastebin here? |
| 08:10.31 | gavimobile | maybe even packistani |
| 08:10.36 | gavimobile | #asterisk-pk or something |
| 08:11.06 | _omer | teatchsch: you can use any paste bin |
| 08:11.31 | teathsch | _omer.. cool.. i've been flamed before lolz |
| 08:12.03 | _omer | :D |
| 08:12.49 | _omer | due to paste bin thingy? |
| 08:13.55 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 08:14.07 | teathsch | due to using the not preferred one for the channel.. i was probably asking for it somehow though |
| 08:14.16 | teathsch | http://pastebin.com/TeJTyBtL |
| 08:14.41 | teathsch | it's a script to do Onion routing via iax2.. just finished it.. maybe some people would be interested |
| 08:15.06 | Dovid | how do I find out if asterisk RT supports outboundproxy= ? |
| 08:16.14 | _omer | teatchsch: why dont you put it on wiki ? |
| 08:18.54 | teathsch | never seen the wiki.asterisk before now.. always voip-info.org.. might do that |
| 08:36.03 | WiretapMac | teathsch: who cares which pastebin you use |
| 08:38.32 | teathsch | WiretapMac: #c++ |
| 08:38.39 | WiretapMac | >_> |
| 08:38.41 | WiretapMac | *Sigh* |
| 08:51.23 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 08:51.23 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 09:01.06 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:04.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:08.42 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:fdf7:e78c:db98:6a7c) |
| 09:14.16 | mzb | teathsch, fwiw there are better pastebins ;) |
| 09:14.29 | mzb | ~google openpaste |
| 09:28.09 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:30.14 | tzafrir | teathsch, data on top of voip? |
| 09:30.23 | tzafrir | or do I misunderstand you? |
| 09:31.02 | teathsch | no... think Tor |
| 09:31.42 | teathsch | anonymous calls |
| 09:32.30 | tzafrir | ah, the other way around. Sounds interesting |
| 09:34.11 | tzafrir | teathsch, I would imagine voip is sensetive to the delays (latency, but more importantly jitter) |
| 09:34.19 | tzafrir | Have you tried this in practice? |
| 09:34.30 | teathsch | it doesn't go over tor.. but it works like tor |
| 09:34.48 | tzafrir | same issue |
| 09:34.54 | teathsch | i've tested looping back to 127.0.0.1 lol |
| 09:35.06 | teathsch | yes, but it doesn't have the same delays due to tcp |
| 09:35.19 | tzafrir | the more you bounce the call, the more jitter you'll get |
| 09:37.23 | teathsch | yeah.. and depending on 3+ nodes being configured correctly |
| 09:37.51 | tzafrir | oh, and if you want to post code, don't use a pastebin |
| 09:38.03 | tzafrir | it's impossible to follow-up on fixes |
| 09:38.17 | tzafrir | just put up a small git repo somewhere |
| 09:38.42 | tzafrir | And hand over the direct link to the HEAD of the master branch |
| 09:42.48 | teathsch | i'm probably gonna do something like that |
| 09:44.15 | teathsch | but yeah, the jitter/latency issue is one of the biggest forseeable issues, but i think that if users are willing to tolerate an audio delay for anonymous calls, then it should work |
| 09:44.39 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 09:50.52 | gavimobile | hi tzafrir |
| 09:51.10 | tzafrir | Hi |
| 10:08.27 | gavimobile | folks, I did an update to my asterisk and im getting these errors, anyone have a clue? centos told me I need to ask here http://pastebin.com/F5FRSNHV |
| 10:09.18 | WiretapSeven | gavimobile, is your kernel up to date |
| 10:10.50 | gavimobile | WiretapSeven: prob not, but I have no expirence with kernels |
| 10:11.13 | WiretapSeven | gavimobile, your error is saying that your kernel is outdated, and 2.6.18 is very out of date |
| 10:11.29 | WiretapSeven | upgrade your kernel first |
| 10:12.15 | gavimobile | WiretapSeven: is it something complicated to explain here? |
| 10:12.21 | WiretapSeven | no |
| 10:12.28 | gavimobile | or its more than one line of code |
| 10:12.28 | WiretapSeven | you run the same command you did for asterisk |
| 10:12.38 | gavimobile | this is asterisknow |
| 10:12.44 | WiretapSeven | specifying 'kernel-i686' instead |
| 10:12.46 | gavimobile | freepbx is managing everything for me |
| 10:12.59 | gavimobile | WiretapSeven: where can I specify? /etc/init.d? |
| 10:13.04 | WiretapSeven | gavimobile, run 'uname -a' |
| 10:13.16 | gavimobile | Linux localhost.localdomain 2.6.18-238.5.1.el5 #1 SMP Fri Apr 1 18:42:32 EDT 2011 i686 athlon i386 GNU/Linux |
| 10:13.17 | WiretapSeven | paste the output here (its one line) |
| 10:13.24 | WiretapSeven | okay, right |
| 10:13.25 | gavimobile | too slow :-p |
| 10:13.28 | WiretapSeven | your kernel is _too_new_ |
| 10:13.31 | WiretapSeven | >_< |
| 10:13.33 | WiretapSeven | gotta love centos |
| 10:13.35 | gavimobile | to new? |
| 10:13.37 | gavimobile | lol |
| 10:13.54 | gavimobile | so I need to rollback on my old kernel |
| 10:13.58 | gavimobile | ? |
| 10:14.01 | WiretapSeven | yours is 2.6.18-238.5.1.el5, it expects 2.6.18-194.31.1.el5 |
| 10:14.31 | gavimobile | wow. like the bank.. only a few digits make a huge difference |
| 10:14.34 | WiretapSeven | tbqh, I find this way of dealing with kernels quite silly, the one time I prefer compiling your own |
| 10:14.43 | teathsch | just force it lol |
| 10:14.51 | WiretapSeven | btw, current 'stable' kernel as for kernel.org is 2.6.37 :P |
| 10:15.02 | WiretapSeven | 2.6.18 is at least a year or two old |
| 10:33.51 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 10:35.10 | gavimobile | WiretapSeven: how can I fix the problem man? |
| 10:35.18 | drmessano | Reboot the box |
| 10:35.31 | WiretapSeven | gavimobile, I'm a debian user, not a centos user |
| 10:35.41 | WiretapSeven | my asterisk pbx is debian based |
| 10:36.07 | drmessano | gavimobile, you have a pending kernel update.. reboot the box, then run yum again |
| 10:37.39 | gavimobile | drmessano: I've tried that |
| 10:37.46 | gavimobile | no luck :-( |
| 10:43.26 | *** join/#asterisk _omer (~omer@182.178.180.222) |
| 10:43.42 | _omer | AJAM ??? |
| 10:44.28 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 10:44.30 | [sr] | hi people |
| 10:44.45 | _omer | any one who can answer my stupid questions about AJAM ... :D |
| 10:50.24 | _omer | is AJAM asterisk example polling data after a certain time period or wht? |
| 11:07.21 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 11:07.23 | Aut0Exec | hi guys |
| 11:11.28 | Aut0Exec | how do I become l33t in asterisk? |
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| 12:11.55 | *** join/#asterisk ripon (~ripon@78-86-161-207.zone2.bethere.co.uk) |
| 12:14.04 | ripon | cant usethe , amportal restart or asterisk restart from the asterisk command line- so couldnt restart asterisk- so just switched the machine off and on |
| 12:14.50 | ripon | now two of my sipgate acccounts dont show as being registered - have i ballsed my machine up ? |
| 12:15.55 | ripon | since i couldnt use those commands ( commands not found ) |
| 12:16.39 | ripon | restarting ubuntu hasnt helped- any advice please///// anyone ? |
| 12:19.29 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
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| 13:01.10 | benngard | hmm, CALLERID(name)="Martela (fax)" shouldnt exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) give me just "Martela" it gives me "Martela " |
| 13:02.22 | S[r]us | hi. I'm experimentig with detecting distinctive ring cadences (asterisk 1.6.2.5; dahdi channel; usedistinctiveringdetection=yes). But, somehow, three of these cadences produce a very similar signature, making them impossible to distinguish in Asterisk. However, to the human ear they are very clearly distinguishable. |
| 13:02.56 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:10.39 | S[r]us | The three rings are basically (1) 1s on;4s off;repeat (2) .5s on;.5s off;.5s on;3.5s off;repeat (3) .5s on;.5s off;.5s on;.5s off;.5s on;2.5s off;repeat |
| 13:16.29 | S[r]us | all these produce a signature of 207,146,0 (or values that lie within a range of 2; not always identical) |
| 13:17.02 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 13:17.06 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 13:37.33 | *** join/#asterisk jakent (~john@ip72-205-7-182.dc.dc.cox.net) |
| 13:39.55 | *** join/#asterisk bintut (~bintut@cm193.kappa9.maxonline.com.sg) |
| 13:40.04 | bintut | waves |
| 13:41.04 | bintut | i'm trying to add the apt repository on my debian squeeze i386 following the https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 13:41.25 | bintut | but i don't have the command "add-apt-repository" |
| 13:46.23 | benngard | bintut: aptitude search repository <- not sure it is in there but i would guess so |
| 13:47.54 | benngard | wrong of me :( |
| 13:48.21 | bintut | benngard: it's not there.. anyway, i'm trying to add the repository manually to my /etc/apt/sources.list but i'm confused the in-between backticks |
| 13:48.50 | bintut | this => `lsb_release -cs`-proposed |
| 13:49.35 | bintut | i mean, this => `lsb_release -cs` |
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| 13:51.20 | bintut | nevermind.. |
| 13:51.33 | bintut | i already added the repo.. |
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| 15:09.53 | Aut0Exec | hi guys... besides going to those asterisk training courses... whats the best way to get really l33t in this stuff? |
| 15:14.02 | nix8n82 | Read, Develop theory, Test, Repeat. |
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| 15:47.17 | Andrew__M | The o'reilly book is really good, and the new edition will be out soon. It is available on the oreilly site for reading. |
| 15:47.52 | Andrew__M | But the Digium training courses are excellent as well. |
| 15:59.52 | Aut0Exec | awesome... |
| 16:00.00 | Aut0Exec | is the new oreilly book free for download too? |
| 16:00.04 | Aut0Exec | i have version 1.4 |
| 16:00.14 | Aut0Exec | oh not out yet |
| 16:00.16 | Aut0Exec | missed that |
| 16:03.38 | Aut0Exec | any of you guys make a career out of asterisk? |
| 16:03.48 | Aut0Exec | going around and setting up pbx servers for people? |
| 16:03.51 | Aut0Exec | companies etc etc |
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| 16:18.03 | Andrew__M | <Aut0Exec> The book was available on http://ofps.oreilly.com/ but I just checked and no longer there. |
| 16:18.43 | Aut0Exec | oh ok |
| 16:19.04 | Aut0Exec | Andrew__M: you work on pbx's as a career? |
| 16:20.39 | Andrew__M | It should be out in May both hard and e-formats. I don't know if it will be made available free. |
| 16:20.39 | Andrew__M | There are people and companies (other than Digium) that make a living of Asterisk. |
| 16:20.53 | Aut0Exec | ok |
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| 19:33.42 | tehrabbitt | hey, not sure if anyone in here is familiar with asterisk-gui but I was trying to use it and I have a few questions, mostly where it actually is modifying the .conf files and why specific dialplans don't work |
| 19:34.01 | tehrabbitt | like toll-free numbers work, but regular 7/10 digit numbers go to "exten not found" |
| 19:34.14 | tehrabbitt | I used the same syntax |
| 19:34.18 | tehrabbitt | well similar |
| 19:35.09 | tzafrir | tehrabbitt, it uses the manager interface's UpdateConfig action |
| 19:35.51 | tzafrir | It's somewhere in the Javascript code |
| 19:36.13 | tzafrir | pbx2.js, mostly |
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| 19:39.33 | Aut0Exec | hi guys... anyone here have cisco spa3102? I'm trying to pass thru caller id but for the caller id name.. the only name that passes thru is the name of my pstn sip setup..(pstn) |
| 19:39.52 | Aut0Exec | how do I pass thru the real caller id? |
| 19:39.55 | Aut0Exec | name at least |
| 19:40.00 | Aut0Exec | the number shows up fine |
| 19:40.30 | tehrabbitt | tzafrir, Ah, so basically I have a SCCP / Skinny phone, which I have operational (manually set up the .conf files since you can't set it up in the GUI). However I can't seem to figure out how to assign the phones to a context / dialplan |
| 19:41.00 | tzafrir | tehrabbitt, the GUI mostly works by editing users.conf |
| 19:42.27 | tehrabbitt | ah |
| 19:42.38 | tehrabbitt | tzafrir, so most of my actual config will be in users.conf? |
| 19:43.16 | tzafrir | yup |
| 19:43.54 | tzafrir | tehrabbitt, there's also #asterisk-gui . I have no idea if it is actually active |
| 19:51.14 | tehrabbitt | tzafrir, ah :-\ i'll take a look |
| 19:51.32 | tehrabbitt | tzafrir, my question though is really how can I make the GUI work alongside my custom configs in a way 0_o |
| 19:51.33 | tehrabbitt | lol |
| 19:52.02 | tehrabbitt | because I like the fact I can use the GUI to log in / view the status of things / quickly set up a line, but I also want to be able to have the customization that the regular .conf offers |
| 19:52.04 | tzafrir | tehrabbitt, what do you want to use the GUI for? |
| 19:52.24 | tzafrir | you can still use it to view the status of some things |
| 19:52.33 | tzafrir | for instance - calls |
| 19:53.20 | tehrabbitt | exactly |
| 19:54.01 | tehrabbitt | basically just want to use the GUI to be able to view the status of calls, registration, etc... without needing access to ssh (aka if i'm on the go or need to check the status of the server away from home) |
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| 20:04.11 | gruvfunk | Newb Question: learning AGI+PHP to build IVR (PBX fires off calls to potential providers, collects their DTMF and presents options back to the IVR caller -- can I accomplish this all via 1 .php script, theoretically? |
| 20:04.44 | gruvfunk | Or, if my inbound and outbound are different contexts in the dialplan, would I need to split this out into at least 2 .php scripts? |
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| 20:56.46 | rethus | anybody know, how i could easy debug php-agi scripts? |
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| 21:16.10 | teathsch | rethus: agi set debug on on the cli.. also use the VERBOSE command.. also when the script terminates right away, just run it on the cli to see what the syntax error is |
| 21:17.16 | teathsch | last part assumes php or other scripted language that behaves that way |
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| 21:39.21 | gruvfunk | How can I record user's voice into a file via PHP AGI? I'm currently testing with the following: |
| 21:40.23 | gruvfunk | fwrite(STDOUT, "ANSWER\n EXEC Playback greeting1\n SAY DIGITS $agi_callerid ''\n EXEC Playback greeting2\n RECORD FILE $agi_callerid 'gsm' '#' '12000' 'S=5'\n EXEC Playback greeting3\n"); |
| 21:40.54 | gruvfunk | but it doesn't stop to record, moves onto the last Exec Playback greeting3 |
| 21:43.58 | teathsch | try RECORD FILE \"${agi_callerid} gsm # 12000 S=5\" |
| 21:46.31 | gruvfunk | teathsch: same result |
| 21:46.56 | teathsch | agi set debug on |
| 21:48.05 | gruvfunk | getting a bunch of these: utils.c:1131 ast_carefulwrite: write() returned error: Broken pipe |
| 21:48.31 | gruvfunk | and on the Record specifically: |
| 21:48.48 | gruvfunk | >AGI Rx << RECORD FILE "1008 gsm # 12000 S=5" |
| 21:48.48 | gruvfunk | <PROTECTED> |
| 21:57.12 | teathsch | try with a lowercase s or try using EXEC Record instead |
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| 21:58.33 | gruvfunk | tried lower case s= |
| 21:58.41 | gruvfunk | same result, will try EXEC Record |
| 22:00.26 | Dsolution | can anyone help me get my firewall working for sip to asterisk? its a nokia 50i vpn gateway. ive confgured xtm firewalls before but this is a little bit more non friendly. im trying to get ports 80, 22, 21, sip and rtp. i just need one protocal working with some help. i can duplicate the rest. i guess im not understanding the syntax? |
| 22:02.56 | gruvfunk | teathsch: Exec Record -> works! |
| 22:03.09 | gruvfunk | I'll take it |
| 22:03.37 | gruvfunk | but if it's truly a syntax error, then it should be remediable, no? |
| 22:04.44 | teathsch | i dunno.. all i know is that EXEC usually makes my scripts more likely to behave the way i want |
| 22:05.23 | teathsch | i don't know how well supported those old commands are.. i think maybe they just leave them in for for backward compatibility |
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| 22:15.49 | WiretapSeven | Dsolution, does nokia have an unofficial support chan on freenode? probably get more help there if it does |
| 22:17.13 | Dsolution | they seem to only support phones |
| 22:17.15 | Dsolution | lol |
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| 23:11.23 | imcdona | Everything I've read suggests and Polycom phones are capable of doing remote call pickup on a BLF. See http://www.excaliburtech.net/archives/147 Has anyone got it to work? |
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