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| 01:01.16 | LemensTS | can I use a digium echo canceller module on an openvox a400p? |
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| 01:05.43 | [TK]D-Fender | LemensTS: as in? |
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| 02:01.11 | hariom | I am trying to setup a small hello world application. When I call from my mobile to my asterisk box, I see that call is landing on the asterisk but it simply falls. This is what I get: http://pastebin.com/YtdihNGL |
| 02:02.44 | [TK]D-Fender | [Nov 6 07:18:54] -- Auto fallthrough, channel 'SIP/testsip-00000000' status is 'UNKNOWN' |
| 02:02.55 | [TK]D-Fender | You ran out of dialplan to execute for the call so it simply ended |
| 02:03.08 | [TK]D-Fender | "Nothing more? Ok, bye!" |
| 02:06.03 | hariom | [TK]D-Fender: After Answer() I am playing hello-world and then demo-thanks |
| 02:06.17 | [TK]D-Fender | Apparently you're not |
| 02:06.29 | hariom | What could be the issue? |
| 02:06.41 | [TK]D-Fender | Perhaps you did it wrong |
| 02:09.12 | hariom | [TK]D-Fender: http://pastebin.com/xWADk3b7 |
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| 02:12.59 | [TK]D-Fender | hates his system's random crashes.... |
| 02:17.10 | [TK]D-Fender | So, have you found the error yet? |
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| 02:31.38 | hariom | [TK]D-Fender: No. That still exists: http://pastebin.com/eAjgtW1X . Seeking help in resolving it |
| 02:32.54 | [TK]D-Fender | har"dialplan show" |
| 02:32.59 | [TK]D-Fender | hariom: "dialplan show" |
| 02:39.14 | hariom | [TK]D-Fender: http://pastebin.com/PmNrcHNf |
| 02:39.24 | [TK]D-Fender | harFirst I'm betting your attempt to minimize the modules you are loading is cutting off critical ones. |
| 02:40.52 | hariom | [TK]D-Fender: which ones? |
| 02:40.58 | [TK]D-Fender | Not sure offhand |
| 02:41.12 | [TK]D-Fender | Prove it first by just auto-loading all of them |
| 02:42.25 | hariom | [TK]D-Fender: If I comment all "load" and just use auto-load = yes then also it loads just 29 modules only |
| 02:42.43 | [TK]D-Fender | go check out which ones are even in your folder. |
| 02:45.44 | hariom | [TK]D-Fender: ok, I have loaded 186 modules (all that came with default installation) but still the same issue |
| 02:46.52 | hariom | [TK]D-Fender: Just to inform, it was working yesterday but today after restarting the process, it doesn't seem to work |
| 02:48.13 | hariom | I am running it as "asterisk" user which in not my default login uesr into the server. "asterisk" user has home directory but it is not in admin group (I can't sudo). |
| 02:50.11 | hariom | [TK]D-Fender:^ |
| 02:51.39 | hariom | [TK]D-Fender: I only have "asterisk" process in "ps aux" . Should there be "safe_asterisk" ? |
| 03:00.26 | hariom | any body? |
| 03:02.09 | [TK]D-Fender | How did you start *? |
| 03:02.12 | [TK]D-Fender | "core show settings". |
| 03:02.22 | [TK]D-Fender | then pastebin another call with core debug jacked up |
| 03:02.28 | [TK]D-Fender | "core set debug 10" |
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| 03:16.33 | hariom | [TK]D-Fender: http://pastebin.com/TSnKhJgX |
| 03:17.49 | [TK]D-Fender | You should have a huge log for the new call with core debug |
| 03:18.33 | ChannelZ | I think you probably need some of the bridge_* modules... |
| 03:22.39 | [TK]D-Fender | heads out for a while... |
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| 03:23.58 | hariom | [TK]D-Fender, ChannelZ: Do you see anything unusual. I first rebooted my system. After loggin in, I found "asterisk" process running but when I tried restarting this process go this msg: http://pastebin.com/r7S8CTJ4 |
| 03:25.16 | ChannelZ | that could just be the init script screwed up |
| 03:32.19 | ChannelZ | wait a minute something looks wrong with your extensions |
| 03:32.48 | ChannelZ | <PROTECTED> |
| 03:32.48 | ChannelZ | <PROTECTED> |
| 03:33.41 | ChannelZ | Those are two different extensions. Do you have some invisible/special character typed in the first one, or vice-versa? |
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| 03:41.01 | hariom | ChannelZ, [TK]D-Fender: Any other possibility you see? I have loaded all default modules (186 modules). In fact there is only one line in that section: autoload=yes. Rest of the things are commented. |
| 03:41.17 | ChannelZ | Read above |
| 03:41.23 | ChannelZ | your dialplan is messed up |
| 03:41.33 | hariom | I see |
| 03:44.48 | hariom | ChannelZ, [TK]D-Fender: That was the case. It works now |
| 03:44.54 | hariom | ChannelZ: big thanks |
| 03:50.16 | ChannelZ | sure |
| 03:56.47 | hariom | ChannelZ: Another strange issue. I have created one more extension and playing a beep sound when call lands on that extension. But when I call to that extension 2 number, call still goes to extension 1 and plays hello-world |
| 04:03.33 | hariom | ChannelZ, [TK]D-Fender: http://pastebin.com/2gqspMbM |
| 04:03.52 | hariom | When call to 3890126, it plays "hello-world" |
| 04:04.03 | hariom | It should actually play "beap" sound |
| 04:05.43 | hariom | ChannelZ, [TK]D-Fender: http://pastebin.com/Tf3hie2U . Pls ignore previous paste |
| 04:07.36 | hariom | Pls note that "beap" is a custom file and not beep |
| 04:08.06 | hariom | Any idea why it is not picking up correct extension? |
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| 05:02.08 | ChannelZ | show us verbose on the call. |
| 05:08.11 | ChannelZ | oh.. he pinged. |
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| 06:09.01 | Milos | pinggggggggg |
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| 07:05.16 | Penguin | I pinged, too! |
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| 07:33.34 | Milos | What is "connected line has changed. Saving it until answer for" |
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| 08:05.00 | ChannelZ | I think it means time and space bent. |
| 08:10.14 | Milos | lol |
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| 10:43.40 | paul_grozav | is there any device that I can use like this: myPhoneProvider <---> phoneLine <---> SOME_DEVICE <--->ethernetCable <---> asteriskServer ? I'd like to connect the phone line to that device and then connect the device to the server by a LAN cable. The idea is that I'd like to have something like an asterisk PCI card, that is outside of the computer/server |
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| 10:55.29 | jww | paul_grozav: I may be wrong but that looks like a modem, some are called 'ethernet modems' like the one given by ISP |
| 11:06.05 | workingcats | paulc, well what do you want to do on the phone line, calls? |
| 11:06.13 | workingcats | then you're looking for an asterisk server... |
| 11:06.21 | workingcats | tabfail |
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| 11:06.34 | workingcats | ah he's gone |
| 11:09.00 | WIMPy | You need a gateway. Some ATAs do have both an FXS and an FXO port. They would do. |
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| 14:02.55 | file | falls over into existence |
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| 14:40.09 | abourget | folks, I get a straight up crash when, using Asterisk 13, I put a channel in a Bridge .. |
| 14:40.27 | abourget | I'm using the ARI, using a brand new Golang API I'm writing, here: https://github.com/abourget/ari |
| 14:40.56 | abourget | when I run this line: https://github.com/abourget/ari/blob/master/birthday/incoming.go#L47 |
| 14:41.26 | abourget | a few seconds later, Asterisk just shuts down, but it ends with "Asterisk cleanly ending (0).".. which surprises me even more |
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| 14:42.19 | abourget | oh, note also that * is running from this container: https://github.com/abourget/asterisk-docker |
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| 14:53.37 | Navion | Anyone ever have trouble resetting the admin password on a Polycom phone? I did the 4 finger initialize with the MAC address as the password but it comes up saying "SORRY PLEASE TRY AGAIN" |
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| 14:53.43 | gesnaud | Hello there |
| 14:54.01 | gesnaud | I'm looking for an info |
| 14:54.09 | gesnaud | about generating calls in a queue |
| 14:54.20 | gesnaud | I want to make a callback system |
| 14:54.32 | gesnaud | I want to creat a callback queue |
| 14:54.53 | gesnaud | with many call waiting for an agent to be available |
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| 14:55.39 | gesnaud | Is that possible *without* making external number ringing? |
| 14:55.53 | gesnaud | The calls are just in the queue |
| 14:56.00 | gesnaud | no phones are ringing |
| 14:56.08 | gesnaud | an agent is available |
| 14:56.23 | gesnaud | then both agent's phone and external phone ringing |
| 14:56.39 | gesnaud | (or agent first then external phone) |
| 15:00.00 | [TK]D-Fender | [09:55]gesnaudIs that possible *without* making external number ringing? |
| 15:00.10 | [TK]D-Fender | [09:56]gesnaudthen both agent's phone and external phone ringing |
| 15:00.20 | [TK]D-Fender | First you say "no ringing", then you say "ringing" |
| 15:00.31 | [TK]D-Fender | Please rephrase your question |
| 15:02.51 | gesnaud | @[TK]D-Fender: Yes, sure |
| 15:03.00 | gesnaud | All agents are busy |
| 15:03.31 | gesnaud | an external number come in the queue |
| 15:04.15 | gesnaud | with the appropriate script, this external number ask for a callback, as the EWT is too long for him |
| 15:04.18 | gesnaud | okay? |
| 15:04.41 | [TK]D-Fender | Not very clear |
| 15:04.47 | [TK]D-Fender | Numbers don't ask for a callback |
| 15:05.00 | [TK]D-Fender | and "numbers" don't tend to enter a queue |
| 15:05.02 | [TK]D-Fender | a CHANNEL does |
| 15:05.35 | gesnaud | you are right |
| 15:05.39 | gesnaud | my bad |
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| 15:06.15 | gesnaud | a cutomer calls a number that is binded to the queue |
| 15:07.00 | [TK]D-Fender | There is also no "binding". If your dialplan sends them to Queue() then OK. |
| 15:07.20 | [TK]D-Fender | So you have a caller reach your system, and enters a Queue(). What then? |
| 15:07.46 | [TK]D-Fender | Normally Queue will use its "strategy" to find an agent to take their call. |
| 15:07.50 | [TK]D-Fender | What about this process? |
| 15:08.06 | gesnaud | all agent are IN USE state |
| 15:08.38 | gesnaud | so the system ask the caller for a callback (specific dialplan here) |
| 15:08.49 | [TK]D-Fender | Which means your caller will sit around till one is available, or they reach a timeout |
| 15:09.08 | gesnaud | No |
| 15:09.20 | [TK]D-Fender | Or they choose to exit the queue |
| 15:09.31 | gesnaud | The caller is invited to enter a number where to be called back |
| 15:09.56 | [TK]D-Fender | They can pick an option to exit, or wait until a set time limit. Take your pick. |
| 15:10.00 | gesnaud | this number is recorded by the system (a postgresql table actually) |
| 15:10.14 | gesnaud | yes that's it |
| 15:11.31 | gesnaud | My problem is to generate in a clean way this callback |
| 15:11.39 | gesnaud | I have my table |
| 15:12.24 | gesnaud | In fact, i'm using a crontab to detect when an agent is available in the queue |
| 15:13.14 | gesnaud | the when one is NOT IN USE, i take a record in my callback table, write a callme, and place it in /var/spool/asterisk |
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| 15:13.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:17.17 | [TK]D-Fender | Originate a Local channel that loops around asking something liek "Press 1 to call back this client', and dump that channel into the queue for the agents to respond to. |
| 15:17.34 | [TK]D-Fender | And have it dial them when your agent presses 1 after answering. |
| 15:21.27 | gesnaud | Yes :) |
| 15:21.44 | gesnaud | "Originate a Local channel that loops around asking something liek "Press 1 to call back this client', and dump that channel into the queue for the agents to respond to." >> how could you do that? |
| 15:22.48 | gesnaud | a local channel could "live" in a queue (without making ringing nobody), until an agent become available? |
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| 15:25.39 | [TK]D-Fender | Local channel is just dialplan. In this case, dialplan that keepsw asking the user to press 1 to accept. when they do you continue on to actually Dial() the callback number |
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| 15:57.20 | gesnaud | @[TK]D-Fender: that sounds nice, thx for your support! |
| 16:00.12 | [TK]D-Fender | You're welcome. |
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| 16:22.58 | Navion | Anyone ever have trouble resetting the admin password on a Polycom phone? I did the 4 finger initialize with the MAC address as the password but it comes up saying "SORRY PLEASE TRY AGAIN" |
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| 16:24.02 | Navion | I unplugged it and started it up with the 1-3-5-7 keys pressed. It asked for password. I put in the MAC address as the password. It said "FACTORY RESET" SORRY PLEASE TRY AGAIN". |
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| 16:30.09 | Navion | The silence is deafening. :) |
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| 16:58.02 | aphedox | does anyone know any high quality UK trunk providers that offer toll-free DIDs? |
| 16:58.10 | aphedox | currently using 10k minutes/mo |
| 16:59.03 | aphedox | preferrably with some sort of functional web management so I don't have to call them to change anything |
| 17:03.29 | gesnaud | @[TK]D-Fender: all my fellows are greatful to you!! |
| 17:03.46 | gesnaud | @[TK]D-Fender: that's working fine, no needs of crontab |
| 17:04.27 | [TK]D-Fender | gesnaud: You're welcome. |
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| 19:04.27 | net | is there any function where i can convert a hexa value to decimal value using dialplan? |
| 19:06.36 | [TK]D-Fender | not in a single command entirely within the dialplan. |
| 19:06.47 | [TK]D-Fender | I'd basically shell it out. |
| 19:08.15 | net | [TK]D-Fender: ok |
| 19:08.26 | Qwell | mmm, func_sprintf might be able to |
| 19:09.54 | Qwell | NoOp(${SPRINTF(%d,0x${MYHEXVAR})}) |
| 19:09.58 | WIMPy | I can see it working in the opposite direction. |
| 19:09.59 | Qwell | or something, possibly |
| 19:10.24 | WIMPy | I don't think Asterisk understands 0x. |
| 19:14.26 | Qwell | hrm, yeah, for the "numeric" (diouxXc) conversions, int is required |
| 19:14.48 | Qwell | funny enough though, oct would probably work |
| 19:15.36 | net | nope its not working.. actually my hex will be like this - 303233343633373732 |
| 19:15.56 | net | its giving a wrong value like this "-1823327708" |
| 19:15.59 | net | in negative |
| 19:16.03 | net | i think i have to use shell |
| 19:16.14 | Qwell | Which is why I said to add the 0x. Regardless though, it won't work with SPRINTF. |
| 19:18.17 | net | @Qwell: adding 0x results the value to 0 |
| 19:18.58 | [TK]D-Fender | Shell is probably the easiest way. |
| 19:19.14 | [TK]D-Fender | Or you could set up the logic in a stupid amount of dialplan ;) |
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| 20:24.44 | n3tctrl | having trouble setting up a sip to fxo gateway. when i dial a number the gateway light lights up but all i get is dial tone on the ip phone. any suggestions? |
| 20:25.46 | n3tctrl | nvr mnd. it started working. |
| 20:25.47 | n3tctrl | thanks! |
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| 21:41.21 | n3tctrl | can someone assist with asterisk to an overhead paging system? i have to configure a trunk port. |
| 21:42.11 | WIMPy | How do those two things fit together. |
| 21:42.53 | n3tctrl | overhead paging system uses a trunk port off the existing nortel. i need to replace it with an fxo gateway to duplicate, but i can't get my outbound route correct to connect the right channel on my fxo gateway. |
| 21:43.04 | n3tctrl | gives me the your call cannot be completed as dialed message. |
| 21:46.23 | n3tctrl | basically, i just need a trunk access code type of setup using 77 to pass the audio from the phone to the paging amplifier |
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| 21:58.55 | drmessano | n3tctrl, have the FXO gateway register as an extension, set the gateway to auto answer |
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| 22:09.38 | Micc | The buttons on the right side of a digium D70 are supposed to have lights too, right? When I use the smart blf file and change it from main to side it doesn't show a light. |
| 22:11.23 | WIMPy | What light? |
| 22:12.20 | WIMPy | You used the line keys as destination keys, or what it the starting point? |
| 22:12.27 | WIMPy | is |
| 22:12.52 | n3tctrl | drmessano, can i do this with an 8 port fxo? i am using a grandstream gxw4108, so maybe i just need a single port fxo? |
| 22:13.15 | drmessano | n3tctrl, just use 1 port |
| 22:13.46 | drmessano | Doesnt it have 8 "lines" to configure in the UI? |
| 22:13.51 | drmessano | Each one is a SIP client |
| 22:14.01 | drmessano | So register it to the box as an extension |
| 22:14.10 | n3tctrl | drmessano, i think i see where you are going with that! thank you! |
| 22:15.46 | n3tctrl | i always knew that voip was more capable than standard pbx's, but this project is really showing me just how incredibly flexible a voip system can be! |
| 22:24.41 | n3tctrl | so, how does one do a pickup group? |
| 22:27.57 | hardwire | carefully |
| 22:28.34 | Qwell | hardwire: Are you on the marijuanas? |
| 22:28.39 | n3tctrl | yeah so i see. i have an overhead ringer in a noisey manufacturing environment that is tied to an extension, and i need a way of picking up calls to that extension from every phone. |
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| 22:31.14 | n3tctrl | maybe setting up a ring group and send the ivr option to that ring group instead. then all the phones ring, and whoever picks up gets the call, right? |
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| 22:52.16 | hardwire | Qwell: no. Not for another 30 days. |
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| 22:52.40 | hardwire | Actually most of the people I know that are on the marijuanas up here already are pretty useless at work anyways... so I tend to avoid it. |
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