| 00:03.53 | rrittgarn | queue add member Local/8003913000@OutboundContext to MyQueue (from the CLI) accomplishes what you're looking for, I don't know anything about queue metrics though. |
| 00:04.16 | rrittgarn | your OutboundContext has to have an extension that matches the number you're dialing otherwise asterisk will call it an invalid extension |
| 00:04.24 | rrittgarn | (rightfully so) |
| 00:04.52 | rrittgarn | needs to learn how to read timestamps... 20 minutes late to the party apparently |
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| 01:24.19 | ipengineer | in Asterisk 12 with pjsip if I want to subscribe to MWI on voicemailbox 999 would I just set 999 in mailbox under ps_endpoints? The voicemail is realtime in the database |
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| 08:56.04 | boratynskikamil | Question. Do you have any proper right way to keep files clear? |
| 08:59.00 | wonderworld | morning |
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| 09:43.30 | fbnts | Hi, I am having a small problem when using MONITOR_EXEC in extensions.conf. My bash script it is calling is: http://pastebin.com/htBcbpQR |
| 09:44.02 | fbnts | all works fine until the last line which moves the new audio file into a sub-directory - its just not moving the file |
| 09:48.06 | boratynskikamil | Should I consider remove warnings like: [Feb 15 10:44:15] WARNING[2448]: chan_dahdi.c:17998 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. |
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| 09:53.15 | WIMPy | remove? |
| 09:53.51 | WIMPy | Do you have calls working or are you still searching for them? |
| 10:03.54 | boratynskikamil | WIMPy: Call are working... |
| 10:04.14 | boratynskikamil | WIMPy: The problem was chan_extra does not compile asterisk with PRI. |
| 10:04.17 | boratynskikamil | Ok, but, go on. |
| 10:04.24 | boratynskikamil | I load modules by myself. |
| 10:05.08 | boratynskikamil | I would like to make an order here and I load them. Where I do not I get such as warnings and some with xmldoc. |
| 10:09.13 | WIMPy | Oops. You got ignored. |
| 10:09.38 | boratynskikamil | WIMPy: By mistake? ;-P |
| 10:09.56 | *** join/#asterisk Szab100 (~Szab100@catv3EC94287.pool.t-online.hu) |
| 10:10.46 | WIMPy | Might have been a network issue. |
| 10:12.39 | boratynskikamil | 11:03 < boratynskikamil> WIMPy: Call are working... |
| 10:12.40 | boratynskikamil | 11:04 < boratynskikamil> WIMPy: The problem was chan_extra does not compile asterisk with PRI. |
| 10:12.42 | boratynskikamil | 11:04 < boratynskikamil> Ok, but, go on. |
| 10:12.45 | boratynskikamil | 11:04 < boratynskikamil> I load modules by myself. |
| 10:12.47 | boratynskikamil | 11:04 < boratynskikamil> I would like to make an order here and I load them. Where I do not I get such as warnings and some with xmldoc. |
| 10:12.50 | boratynskikamil | 11:09 < WIMPy> Oops. You got ignored. |
| 10:13.46 | WIMPy | What warnings? |
| 10:15.52 | Szab100 | Hi.. Is there anyone who is using asterisk-java? I would like to create a pbx system which process incoming calls & can originate outgoing ones based on user's input on the incoming channel. It's important to ring on the incoming leg while ringing on outbound leg, do something (play rec, get user input) on the outbound leg once it's answered then bridge the two calls.. After the call finishes |
| 10:15.53 | Szab100 | (one of the parties hung up), I would like to get proper CDR processing.. |
| 10:17.29 | Szab100 | Actually my question is NOT only for asterisk-java users.. So this is a general topic I think.. I am able to use AGI to handle incoming leg (and possibly the outgoing after answered via Local chans) + AMI to originate / AMI Events.. |
| 10:18.37 | WIMPy | I'm not exactely sure what the question is, but you can have two seperate calls and bridge them later on. |
| 10:18.39 | Szab100 | I would just need the best practice for this kind of use.. And the simplest way because I would like to handle it with the least threads as possible.. |
| 10:19.30 | Szab100 | Well, I tried to use AMI/Originate |
| 10:20.14 | Szab100 | but it's very complicated to follow the Events generated by this originate |
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| 10:20.30 | Szab100 | because there isn't any fixed IDs I can use |
| 10:21.07 | WIMPy | The channel name. |
| 10:21.38 | Szab100 | yes but that's changing.. |
| 10:21.48 | Szab100 | so I was using async originate |
| 10:22.03 | WIMPy | The channel name never changes. |
| 10:22.53 | WIMPy | But I have to admit that I'm not sure how obvious the link from your originate action to the resulting channel name is. Haven't originated via AMI for ages. |
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| 10:24.37 | boratynskikamil | WIMPy: http://wklej.org/id/1273313/ - like here. |
| 10:26.20 | WIMPy | Yes, that's the kind of warning that would only be needed once. |
| 10:26.33 | boratynskikamil | needed? |
| 10:26.41 | boratynskikamil | I would like to solve it. :-) |
| 10:26.56 | WIMPy | To tell you that the reload you just did might not have the desired effect. |
| 10:27.45 | boratynskikamil | WIMPy: The case is I didn't reload. |
| 10:27.48 | boratynskikamil | I loaded it. |
| 10:28.45 | WIMPy | SMaybe it can't tell the difference between initial load and reload? |
| 10:29.03 | boratynskikamil | So is it good I got these warning? |
| 10:29.06 | boratynskikamil | warnings* |
| 10:30.41 | WIMPy | Well, they tell you that those values cannot be changed on a reload. |
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| 11:16.05 | boratynskikamil | WIMPy: These modules are not reloaded. They are started. So? |
| 11:16.37 | WIMPy | >>Maybe it can't tell the difference between initial load and reload? |
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| 11:40.26 | boratynskikamil | WIMPy: So? Just leave it? |
| 11:42.56 | WIMPy | Safe to ignore (except for the message about reloads, off course). |
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| 14:22.37 | ruben23 | hi guys any way i cant set my asterisk 1.8 to have color on its logs same as the origanl asterisk CLI- which have white, pink, yellow and bluegreen text on its logs--->coz now my asterisk logs on CLI- are all white only...displaying |
| 14:26.12 | ruben23 | any idea guys...? what could be the issue |
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| 14:34.00 | HumpyDumpy | xcuse me, is it hard to set asterisk up for phone conferencing where people can enter there conference number to join the call? |
| 14:39.27 | WIMPy | The conferencing part is super easy. |
| 14:39.45 | HumpyDumpy | i see conf bridge makes it happen |
| 14:39.47 | WIMPy | Getting things up and running is debatable. |
| 14:39.48 | HumpyDumpy | i'm googling |
| 14:39.58 | WIMPy | yes |
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| 14:40.35 | HumpyDumpy | my connection speed is 14MB/s |
| 14:40.40 | HumpyDumpy | is that good enough for a few callers? |
| 14:40.43 | HumpyDumpy | like 5 |
| 14:40.45 | HumpyDumpy | or so |
| 14:41.03 | WIMPy | In both directions? |
| 14:41.13 | HumpyDumpy | ummm |
| 14:41.16 | HumpyDumpy | thats my down |
| 14:41.19 | HumpyDumpy | I don't know what my up is |
| 14:41.21 | HumpyDumpy | ;{ |
| 14:41.29 | WIMPy | Without additional complression you can plan about 100kbit/s per call and direction. |
| 14:42.10 | fauxalliance | HumpyDumpy: speedtest.net and bandcalc.com |
| 14:42.10 | WIMPy | s/complression/reduction/ |
| 14:42.15 | WIMPy | double wrong |
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| 15:03.20 | HumpyDumpy | so it's good both ways |
| 15:03.26 | HumpyDumpy | i just checked my upload |
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| 15:28.37 | HumpyDumpy | i'm trying to get asteriskgui working, but it's saying page not found when I goto my url:8088 |
| 15:28.47 | HumpyDumpy | i followed the guide |
| 15:28.50 | HumpyDumpy | and made all the changes |
| 15:29.00 | HumpyDumpy | but i don't think my web server knows what to do |
| 15:30.17 | HumpyDumpy | i don't see port 8088 open |
| 15:30.18 | HumpyDumpy | :( |
| 15:30.23 | HumpyDumpy | when i do a netstat -a |
| 15:30.54 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 15:33.24 | WIMPy | Try to ask in #asterisk-gui |
| 15:33.35 | HumpyDumpy | >:( |
| 15:33.40 | HumpyDumpy | that went well |
| 15:33.56 | HumpyDumpy | bites WIMPy really really hard |
| 15:34.43 | WIMPy | We don't support GIUs in here. |
| 15:34.53 | HumpyDumpy | maybe i should have installed from source instead of apt-get |
| 15:36.01 | HumpyDumpy | looks at WIMPy with sad eyes.... I'm sorry I bit you :{ |
| 15:37.54 | HumpyDumpy | time to reset my vps |
| 15:39.45 | HumpyDumpy | is it better 32 or 64 bit |
| 15:39.47 | HumpyDumpy | for asterisk? |
| 15:44.03 | HumpyDumpy | I think i broke DigitalOcean |
| 15:44.06 | HumpyDumpy | :{ |
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| 16:11.43 | bananapie | Hello, I want to upgrade my asterisk. How can I copy the config file from the old source directory to the new source directory so I don't have to redo menuconfig all over again? |
| 16:18.08 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
| 16:19.37 | BeachBall | I want my asterisk to ring 2 phones at the same time I added this - same => n,Dial(SIP/10000000000@flowroute,SIP/300) |
| 16:19.48 | BeachBall | it dials my cell number but not the sip/300 |
| 16:20.19 | bananapie | right |
| 16:20.26 | bananapie | it's & not , for multiple extensions |
| 16:20.44 | bananapie | , is for seperating arguments. All the extensions are part of the same extension |
| 16:20.47 | bananapie | so (SIP/10000000000@flowroute&SIP/30 |
| 16:22.30 | BeachBall | i c |
| 16:22.52 | bananapie | sorry, I mistyped that |
| 16:22.59 | bananapie | => n,Dial(SIP/10000000000@flowroute&SIP/300) |
| 16:25.07 | bananapie | does that help? |
| 16:26.42 | BeachBall | it's working now |
| 16:26.51 | BeachBall | it rings my office and cell |
| 16:26.52 | BeachBall | ;D |
| 16:29.48 | bananapie | :D |
| 16:29.56 | bananapie | Need anything else ? |
| 16:30.18 | BeachBall | not at the moment, i'm good thank you |
| 16:30.23 | BeachBall | tinkering with conf bridge |
| 16:32.08 | BeachBall | gently bounces |
| 16:33.10 | BeachBall | does asterisk have a decent IVR? |
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| 16:33.19 | BeachBall | like if I say stuff it will understand me? |
| 16:33.54 | bananapie | Speech recognition ? |
| 16:33.59 | BeachBall | yuh |
| 16:33.59 | BeachBall | ;D |
| 16:35.07 | bananapie | I can't help you with tat |
| 16:35.09 | bananapie | that* |
| 16:35.10 | bananapie | sorry |
| 16:47.02 | WIMPy | bananapie: menuselect.makeopts |
| 16:49.15 | bananapie | thanks :D |
| 16:49.50 | bananapie | is there an asterisk equivalent to 'make oldconfig' |
| 16:50.20 | WIMPy | doesn't know any |
| 16:50.44 | bananapie | ok, I'll just copy menuselect.makeopts and then run make menuconfig |
| 16:50.54 | bananapie | thanks for your help :D |
| 16:51.29 | WIMPy | BeachBall: Asterisk doesn't do speech recognition by itself. |
| 16:59.26 | BeachBall | where does asterisk hide it's sound files? |
| 17:00.09 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 17:00.30 | bananapie | /usr/share/asterisk/sounds or /var/lib/asterisk/sounds/ |
| 17:00.33 | bananapie | I can never remember |
| 17:01.02 | BeachBall | it's the first one |
| 17:01.02 | BeachBall | ;D |
| 17:01.03 | WIMPy | I/ve never had anything Asterisk under */share/*. |
| 17:01.04 | BeachBall | thanks |
| 17:01.37 | WIMPy | /usr/lib/asterisk or /var/lib/asterisk. |
| 17:02.45 | WIMPy | Broken bananapipe? *eg* |
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| 17:35.55 | BeachBall | how do i secure my asterisk so people can't access my dialplan? |
| 17:36.13 | BeachBall | [Feb 15 12:31:29] NOTICE[985]: chan_sip.c:22622 handle_request_invite: Call from '' (198.50.145.104:5093) to extension '100' rejected because extension not found in context 'default'. |
| 17:36.16 | BeachBall | i see things like that |
| 17:36.19 | BeachBall | and it's not me |
| 17:36.22 | BeachBall | it's a new install |
| 17:36.30 | [TK]D-Fender | don't allow anonymous calls. |
| 17:36.40 | [TK]D-Fender | use strong passwords for your peers |
| 17:36.51 | WIMPy | read the README-SERIOUSELY.bestpractices.txt |
| 17:37.07 | [TK]D-Fender | restrict peers to fixed hosts/networds where possible |
| 17:37.27 | [TK]D-Fender | use a log-scraper like fail2ban to check for brute force attempts and firewall them. |
| 17:37.46 | [TK]D-Fender | Firewall out everything except what you know should be good |
| 17:38.46 | WIMPy | Or better still don't have a connection to the internet. |
| 17:39.54 | [TK]D-Fender | Better yet, there's this cord that supplies power to your server. Better pull on that hard.... |
| 17:40.21 | *** part/#asterisk LiuYan (~LiuYan@222.125.134.157) |
| 17:40.38 | WIMPy | That's more to avoid failures than remote access. |
| 17:43.13 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 17:43.53 | BeachBall | going back to the don't allow anonymous calls bit |
| 17:43.55 | BeachBall | where might one put that |
| 17:44.16 | WIMPy | look for allowguests |
| 17:44.28 | BeachBall | i have that set to no |
| 17:44.30 | BeachBall | ;D |
| 17:45.31 | [TK]D-Fender | show us |
| 17:45.48 | WIMPy | Maybe you set a peer that doesn't need to authenticate? |
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| 18:23.02 | *** join/#asterisk ruben23 (~owner@122.55.73.118) |
| 18:23.43 | ruben23 | hi guys any help regarding my ubuntu PC where i install my headset pretty bad echo audio |
| 18:26.29 | pabelanger | Does your headset have echo cancelling? |
| 18:27.37 | ruben23 | <PROTECTED> |
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| 18:34.16 | ruben23 | pabelanger: still there..? |
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| 18:38.26 | pabelanger | ruben23, sounds like something in ubuntu is interfering with it. See what #ubuntu says |
| 18:40.25 | ruben23 | yea i tried plantronics usb headset, it still the same..so weird. |
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| 18:57.27 | [TK]D-Fender | That EC is software based. You are comparing totally different code |
| 18:57.35 | [TK]D-Fender | And they are not likely to be the same |
| 18:57.40 | [TK]D-Fender | Try another softphone |
| 18:57.41 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 19:11.57 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 19:48.38 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:52.07 | ipengineer | Hello all... I am trying to setup a sip trunk with IP based authentication with PJSIP and when I look at module show it is saying "res_pjsip_endpoint_identifier_ip.so" is Not Running. Asterisk is sending back a 401 to the invite. |
| 20:06.09 | ChannelZ | hmm |
| 20:06.57 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:07.13 | ChannelZ | I guess let's see your pjsip.conf |
| 20:12.44 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 20:44.52 | *** join/#asterisk wastonNope (18fb2a7b@gateway/web/freenode/ip.24.251.42.123) |
| 20:49.38 | wastonNope | übernoob here - just got AsteriskNOW box running w/ 4 port analog digium card. I have a consumer cordless vtech base plugged into an analog fsx port. After every call on the fsx port and after every config update through FreePBX, the analog phone receives a silent call that displays the asterisk configured extension name and number. |
| 20:50.09 | wastonNope | I would appreciate any pointers to what this signalling is called and how I might stop it from happening |
| 20:55.54 | [TK]D-Fender | An FXS port on what? |
| 20:55.57 | wastonNope | it seems as if asterisk is sending the name and extension number out the analog port via some type of caller-id protocol?! |
| 20:56.12 | wastonNope | fxs port on a 4 port analog digium pci-e card |
| 20:56.12 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 20:57.15 | [TK]D-Fender | pastebin all of your DAHDI configs |
| 20:57.17 | [TK]D-Fender | ~pb |
| 20:57.17 | 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. |
| 20:57.19 | [TK]D-Fender | ^^^ |
| 20:58.19 | wastonNope | will do - it is for the most part a stock asterisknow w/ freepbx install - where do I find these? /etc/dahdi? thanks for the help |
| 20:58.37 | wastonNope | I'm an asterisk noob, but I've been around the block |
| 21:00.27 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:01.28 | wastonNope | actually, the only file in /etc/dahdi that isn't virgin is system.conf that reads |
| 21:01.35 | wastonNope | fxsks=1-3fxoks=4 loadzone=us |
| 21:01.39 | [TK]D-Fender | PASTEBIN <- |
| 21:01.48 | wastonNope | alright |
| 21:01.56 | [TK]D-Fender | <PROTECTED> |
| 21:02.06 | wastonNope | it's 3 lines |
| 21:02.17 | [TK]D-Fender | <PROTECTED> |
| 21:02.27 | wastonNope | k |
| 21:03.02 | [TK]D-Fender | Also, did you plug the molex power connector to the card? |
| 21:03.25 | wastonNope | http://pastebin.com/6Sf9m5m6 |
| 21:03.46 | wastonNope | yes I did - everything works fine - can dial out, receive calls, etc |
| 21:04.08 | wastonNope | I just get a what looks like to be a call on the extension after every call or config update |
| 21:04.37 | [TK]D-Fender | What colour modelues and how many each on this card? |
| 21:04.57 | wastonNope | there are 4, 3 FXO 1 FXS |
| 21:05.14 | wastonNope | color, green and red, can't remember what color maps to what type |
| 21:05.49 | [TK]D-Fender | change your port 4 to fxols |
| 21:05.55 | wastonNope | I have analog POTS lines connected to the FXO's - I can dial out and receive calls on these and route to my analog extension on the FXS port |
| 21:06.21 | [TK]D-Fender | and provide the Asterisk dahdi configs |
| 21:06.36 | wastonNope | k |
| 21:06.37 | [TK]D-Fender | make sure you change both sets for what I have just sdvised |
| 21:08.04 | wastonNope | I'm I looking for /etc/asterisk/chan_dahdi* ? |
| 21:09.34 | [TK]D-Fender | .conf, and everything that it #include's |
| 21:10.05 | wastonNope | just to clarify, in system.conf I am changing last line to fxols=4 ? |
| 21:14.33 | wastonNope | [TK]D-Fender: http://pastebin.com/4f8PuNW7 |
| 21:15.37 | [TK]D-Fender | [16:09]wastonNopejust to clarify, in system.conf I am changing last line to fxols=4 ? <- yes |
| 21:15.45 | wastonNope | ok |
| 21:16.52 | [TK]D-Fender | your chan_dahdi setup looks pretty messy.... |
| 21:17.23 | [TK]D-Fender | from-analog is not a normal contest for FXO port (lines). They should point to from-trunk |
| 21:17.48 | [TK]D-Fender | unless you are using they "ZAP DID option for splitting routing |
| 21:17.59 | [TK]D-Fender | For which I don't recall that being the place to send them |
| 21:18.11 | [TK]D-Fender | As for your FXS port (phone), that should point to from-internal |
| 21:18.35 | [TK]D-Fender | Also ensure that you change"signalling=fxo_ks" to "fso_ls" to match what you've done in system.conf |
| 21:18.53 | [TK]D-Fender | You'll need to start * and DAHDI and restart both services to take efffect |
| 21:19.05 | wastonNope | ok - I'm assuming that is Loop Start as opposed to Kewl Start |
| 21:19.30 | wastonNope | I'll give it whirl - that's for your time and help |
| 21:30.11 | wastonNope | [TK]D-Fender: made the changes, restarted everything, everything works as before - still receiving some type of signalling to the analog phone after calls and on asterisk restarts. My analog phone now shows 9 missed calls. |
| 21:31.01 | wastonNope | if I pick up the receiver, listen to dialtone, then hang up, after a few seconds my analog phone's caller=id displays my asterisk extension name and number |
| 21:31.09 | wastonNope | and registers a missed call |
| 21:31.31 | [TK]D-Fender | You have to completely reset * AND DAHDI |
| 21:31.39 | wastonNope | I will reboot |
| 21:31.51 | [TK]D-Fender | a little overkill, but that should do it... |
| 21:33.31 | wastonNope | while rebooting, I kept on eye on the analog phone plugged into the fxs port - when asterisk started up, the analog phone registered a missed call from extension 30 |
| 21:33.42 | wastonNope | I have 10 missed calls now |
| 21:34.11 | wastonNope | I can not hear it ring, though |
| 21:35.59 | wastonNope | will start checking out what kind of logging facilities I have access to |
| 21:37.03 | wastonNope | the missed calls aren't showing up, by default, in /var/log/asterisk/* |
| 21:38.25 | [TK]D-Fender | verify that your changes haven't gotten overridden |
| 21:38.51 | wastonNope | k |
| 21:39.22 | wastonNope | system.conf was re-ordered, but fxols=4 persisted |
| 21:40.23 | wastonNope | singalling=fxo_ls and context=from-internal persisted as well |
| 21:40.45 | wastonNope | signalling |
| 21:52.08 | WIMPy | goes to start the same experiment |
| 21:53.24 | wastonNope | debug:5 shows nothing at the time of the mysterious caller-id missed call event after the hangup |
| 21:53.26 | wastonNope | http://pastebin.com/GaFB0VC4 |
| 21:53.45 | wastonNope | maybe someone else can see something that I'm missing |
| 22:01.52 | wastonNope | I will dump the audio of the fxs port - I should be able to hear this caller-id mubo jumbo bullsh*t? |
| 22:05.03 | wastonNope | yes - I can hear a caller-id blast after I hang up the phone |
| 22:05.20 | wastonNope | where should I paste this .wav? |
| 22:10.36 | WIMPy | I guess the interesting question is what kind of "feature" you're seeing there. |
| 22:12.41 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 22:12.58 | wastonNope | https://soundcloud.com/user378628101/asterisk-caller-id-blast-after |
| 22:13.12 | wastonNope | WIMPy: check it out and let me know what you think it is |
| 22:13.21 | wastonNope | [TK]D-Fender: ping |
| 22:13.41 | [TK]D-Fender | kill "callwaitingcallerid |
| 22:13.48 | WIMPy | Hering it won't tell me what might be configured. |
| 22:14.04 | [TK]D-Fender | Where are you located BTW? |
| 22:14.16 | wastonNope | United States |
| 22:14.31 | wastonNope | will kill cwcid |
| 22:14.44 | [TK]D-Fender | ok, you get the with calls from your analog phone to internal freepbx features as well? |
| 22:15.14 | wastonNope | I don't fully understand the question |
| 22:17.01 | wastonNope | [TK]D-Fender: I don't have "callwaiting" (Case insensitive) appearing anywhere in /etc |
| 22:21.35 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:22.14 | wastonNope | I'm assuming this is some type of feature to update the extension name and number for the display on certain types of phones |
| 22:22.36 | wastonNope | however, it is tripping up my phone |
| 22:22.38 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:23.25 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@sm3e210fc5.cust.navigue.com) |
| 22:23.33 | eXcAliBuR | is it ok if i just sit here, i feel smarter when i'm here :} |
| 22:23.51 | wastonNope | eXcAliBuR: put your smarts to use and help me out |
| 22:24.09 | [TK]D-Fender | No, he only FEELS smarter :) |
| 22:24.21 | wastonNope | just trying to help make it a reality |
| 22:25.12 | wastonNope | maybe I just need a SMARTER phone |
| 22:27.00 | WIMPy | WTF? Why does Asterisk complain about missing pjproject because I enabled chan_dahdi??? |
| 22:27.22 | wastonNope | permissions |
| 22:27.24 | wastonNope | it is always permissions |
| 22:28.00 | WIMPy | I don't see why I want or need pjproject. |
| 22:28.44 | wastonNope | should I open a bug, or post to the forum about this late caller-id issue? |
| 22:28.47 | eXcAliBuR | i need someone to think for me |
| 22:29.27 | [TK]D-Fender | WIMPy: https://fbcdn-sphotos-c-a.akamaihd.net/hphotos-ak-prn2/971561_515187928549534_2091874263_n.jpg |
| 22:29.45 | eXcAliBuR | asterisk answers the call - says please enter pin number (it's a waitexten) if it times out or is invalid i want it to playback that sound... and then loop |
| 22:29.55 | [TK]D-Fender | eXcAliBuR: that's what becoming an employer means.... |
| 22:30.03 | eXcAliBuR | i don't know how to make the loop happen |
| 22:30.07 | WIMPy | Gnnnnnaah |
| 22:30.10 | [TK]D-Fender | eXcAliBuR: Goto <- |
| 22:31.43 | eXcAliBuR | ok, now i've got a infi loop happening |
| 22:31.57 | eXcAliBuR | how do i set the number of times it can loop? |
| 22:31.58 | [TK]D-Fender | You should maybe look at where you are, and where you're going... |
| 22:32.37 | [TK]D-Fender | count each time it does. before Goto-ing back see if you've reached your limit |
| 22:32.46 | [TK]D-Fender | "core show application GotoIf" <- |
| 22:41.20 | *** join/#asterisk ruben23 (~owner@122.55.73.118) |
| 22:43.56 | eXcAliBuR | as soon as i dial a ext that is not in the dialplan it's goes beep boop and kills the call |
| 22:44.11 | eXcAliBuR | how can i have it so it will let me finish dialing what number i want before doing that? |
| 22:44.39 | WIMPy | What sense does it make to wait when there's no chance of success? |
| 22:44.59 | eXcAliBuR | so people can't guess the number based on when it fails |
| 22:45.30 | WIMPy | The number? |
| 22:46.00 | eXcAliBuR | there has to be a way to do it |
| 22:46.01 | WIMPy | Either don't use WaitExten or create a catchall extension. |
| 22:46.23 | eXcAliBuR | define how using waitexten is breaking it? |
| 22:46.38 | eXcAliBuR | or what could i use in place of waitexten |
| 22:46.49 | WIMPy | Because that's what WaitExten is there for. |
| 22:47.11 | WIMPy | It waits for the user to dial an extension. |
| 22:47.28 | WIMPy | Read would just wait for whatever you want. |
| 22:49.25 | WIMPy | Oh, that was evil. Asterisk uses absolute paths in the source. :-( |
| 22:50.38 | eXcAliBuR | ok so if i had exten 4 and 40 waitexten would dial 4 and not let ppl dial 40 |
| 22:50.39 | eXcAliBuR | right? |
| 22:50.47 | WIMPy | no |
| 22:51.01 | WIMPy | It would wait for another digit or a timeout. |
| 22:51.06 | eXcAliBuR | oh |
| 22:51.18 | WIMPy | Which is why overlapping extensions are bad. |
| 23:13.30 | bsdice | In Asterisk 11 dialplan if you Dial() and call completes normally (callee picks up, both then hang up), will Dial() return to dialplan and execute next command (like a Hangup(16)) or will only the h extension have any meaning at that point? |
| 23:14.20 | WIMPy | Unless you give the right option, Dial will not continue in the dialplan once someone answered. |
| 23:16.29 | bsdice | ahh g Option |
| 23:17.09 | WIMPy | There are several. |
| 23:17.53 | *** join/#asterisk lwizardl (~lwizardl@c-67-177-138-192.hsd1.mi.comcast.net) |
| 23:17.56 | lwizardl | hello |
| 23:18.41 | lwizardl | I just bought a used spa1001. I would like to use this for on asterisk. Is there a guide for unlocking and using it? google isn;t helping |
| 23:19.03 | bsdice | unlocking would be offtopic |
| 23:19.10 | bsdice | and careful here, trips mods up :) |
| 23:19.25 | WIMPy | If google doesn't know, I guess you have your answer. |
| 23:19.27 | lwizardl | k |
| 23:19.31 | bsdice | other than that for config see http://spakonfig.de/ |
| 23:20.12 | bsdice | should work if you have sip.conf in working order to let the spa login into asterisk |
| 23:21.30 | eXcAliBuR | thank you ppl |
| 23:21.33 | eXcAliBuR | nap time now |
| 23:22.11 | lwizardl | okay is that website like a default setup page for these ? kinda like the routerlogin.net ? |
| 23:22.29 | bsdice | its a website that shows settings for VOIP providers |
| 23:22.34 | lwizardl | k |
| 23:22.44 | bsdice | gives you a hint what you have to touch in order to get your spa working with your asterisk |
| 23:23.15 | lwizardl | ah ok |
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