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| 03:39.24 | linocisco | is anybody using asterisk on QNAP? |
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| 04:27.55 | Visage | Hey guys, I've migrated from chan_sip to res_pjsip in Asterisk 13, and I'm wondering what the pjsip equivalent of the ${SIPDOMAIN} variable is, since it appears to be empty. I'm guessing it only existed for chan_sip. |
| 04:48.25 | wormEZ | is there a #beginers-asterisk ? |
| 04:55.01 | linocisco | wormEZ, Nope. |
| 04:55.33 | linocisco | wormEZ, https://www.youtube.com/watch?v=PfSL-kekuDE ( from 2013) |
| 04:56.07 | linocisco | wormEZ, https://www.youtube.com/watch?v=7UEBjdVZLw8&list=PLighc-2vlRgTgTGtRvSNA3Q_NqP_qTNF8&index=2 (2014) |
| 04:56.29 | linocisco | wormEZ, that answers most common...... |
| 04:56.49 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:59.15 | wormEZ | thanks |
| 05:01.46 | wormEZ | i am going to have to look busy somehow while i am at work trying to skive off and look at something more interesting :-) |
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| 05:04.58 | *** join/#asterisk cstk421 (~cstk421@99-20-229-203.lightspeed.brhmmi.sbcglobal.net) |
| 05:05.09 | cstk421 | has anyone installed/used homer sip capture ? |
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| 05:05.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 05:59.50 | wdoekes | ~polls |
| 05:59.50 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 06:00.00 | wdoekes | ~ask |
| 06:00.01 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:00.05 | wdoekes | cstk421: ^^ |
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| 06:01.14 | cstk421 | copy that apologies |
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| 08:49.46 | wasanzy | hello |
| 08:50.12 | wasanzy | pls is http://www.voip-info.org/wiki/ down? am not able to reach it |
| 08:54.44 | pa | works here |
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| 09:01.07 | wasanzy | ok |
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| 09:01.19 | wasanzy | maybe I will try different net |
| 09:01.47 | linocisco | wasanzy, down here |
| 09:02.52 | wasanzy | oh ok |
| 09:03.15 | wasanzy | what could be the problem? seems pa: is accessing a cached version |
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| 09:21.29 | wasanzy | is back now |
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| 09:57.27 | linocisco | for indoor wireless LAN with Asterisk traffic, what kind of APs and models are recommended and have no delay issues? we will probably use grandstream WiFi DECT phones and android/IPhone clients |
| 10:23.07 | *** part/#asterisk c0rnoTa (~c0rnoTa@89.17.43.185) |
| 10:28.38 | wasanzy | I want to let users record their voice using Record cmd. But I want them to press 1 to save the recording or 2 to cancel. what can I do? |
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| 10:35.33 | Gugge | linocisco: wifi dect phone? |
| 10:35.44 | linocisco | Gugge, yes. |
| 10:36.02 | Gugge | a phone using two different wireless transports? |
| 10:36.39 | Gugge | do you want to be able to move around between AP's without loosing packets? |
| 10:46.31 | Marquel | linocisco: afaik grandstream makes dect-phones, no wifi-support in them. |
| 10:47.12 | linocisco | Gugee, yes no interruption should be required |
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| 11:00.31 | Gugge | linocisco: then you need something controller based |
| 11:00.47 | wasanzy | I want to let users record their voice using Record cmd. But I want them to press 1 to save the recording or 2 to cancel. what can I do? |
| 11:01.22 | Gugge | wasanzy: record to a tmp file, then make dialplan logic that asks the user, and do something if they press 1 and something else if they press 2 |
| 11:01.46 | Gugge | linocisco: but you dont need wifi for dect phones |
| 11:02.03 | linocisco | so... |
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| 11:02.14 | wasanzy | Gugge: right |
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| 11:23.23 | wasanzy | also, I want to check the input from the user where they press 1 or *, something like Goif( $[ ${ActionItem} == '*' ] ? gotomain:playchannel) what is the fault variable, the inputs from user is stored? |
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| 11:24.57 | linocisco | Gugge, what do I do? |
| 11:24.58 | wasanzy | ${EXTEN}? |
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| 11:26.08 | alexises | Hi |
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| 11:27.51 | alexises | I have some issue with my BRI card |
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| 11:29.47 | alexises | I have 4 physical port connected to channel 1-3,2-6,7-9,10-12 |
| 11:30.07 | alexises | channel 2-6 and 7-9 work perfectly, not 1-3 and 10-12 |
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| 11:30.49 | alexises | configuration is the same between the two 4 interfes |
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| 11:36.36 | wasanzy | <PROTECTED> |
| 11:36.43 | wasanzy | ${EXTEN}? |
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| 11:45.18 | CrummyGummy | Hi, I have a NAT question. I've got a network containing soft phones and an asterisk server. The network is 10.1.0.0/16. The ast server is at 10.1.0.11. That server has an external nic that has 192.168.243.250/24 IP and connects to 192.168.243.10 at our sip provider. |
| 11:45.57 | CrummyGummy | When I make calls from my soft phones I can hear incoming talking but the destination phones can't here me. |
| 11:47.51 | CrummyGummy | Can anyone offer advice as to the nat setup here? I have simple iptables nat setup atm, I've tried a bunch of asterisk nat options. It's just really strange that the incoming voice is being transmitted but the outgoing packets aren't getting out. I'd assume the opposite would be true with nat issues. |
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| 12:46.46 | anonymouz666 | I am in |
| 12:46.58 | anonymouz666 | good morning. |
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| 13:27.12 | Milenco | hey, is there a way to exclude $-signs when using the SHELL() func? |
| 13:28.00 | Milenco | I want to run something through awk but Asterisk give me an error in extension logic error because it stops reading after the $-sign (in print $1;) |
| 13:28.24 | Milenco | doing it in a seperate script would probably work, but im just wondering if i can escape it |
| 13:29.41 | [TK]D-Fender | Tried the usual \ ? |
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| 14:37.43 | Micc | after upgrading from 1.8.9.2 to 1.8.28-cert3, asterisk seems to have reset its way of knowing if people have new or old voicemail messages. Everyone's voicemails are now new. Is there any way to fix this? |
| 14:38.47 | Milenco | yeah [TK]D-Fender, \ doesn't work. I've did it in an external script which works fine :) |
| 14:39.02 | Visage | With PJSIP and dialing a SIP URI, what variable do I use for the "domain.com" part of the URI? ${SIPDOMAIN} was a chan_sip thing, so I can't use that anymore. |
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| 14:46.30 | CrummyGummy | What versions of Asterisk support prac? |
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| 15:15.04 | CrummyGummy | PRACK rather |
| 15:16.03 | CrummyGummy | I see some indication that it should be in 1.8 but I'm running 11.7 and prac=yes doesn't seem to enable it. |
| 15:16.45 | anonymouz666 | I don't remember if there's PRACK support chan_sip.c |
| 15:16.56 | anonymouz666 | but I am sure there is a branch out there that implements it |
| 15:17.06 | anonymouz666 | look for OEJ's branch about prack support. |
| 15:17.19 | anonymouz666 | there's one for 1.8 and maybe another for 11 version. |
| 15:17.24 | [TK]D-Fender | I sure don't see that setting even in 13.... |
| 15:17.47 | [TK]D-Fender | So I'm not sure why you'd assume it's there... |
| 15:18.32 | CrummyGummy | There is a random email headed 'Testers needed for prac in 1.8' |
| 15:18.39 | CrummyGummy | I can't find it atm |
| 15:18.46 | CrummyGummy | maybe it is for a branch |
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| 15:19.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:23.04 | CrummyGummy | thanks anonymouz666 I'll dig around for that tomorrow. |
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| 15:38.45 | Visage | Well, I got around my issue by using PJSIP_HEADER(read,To) and stripping off the < and > to get the dialed URI. Probably not the best way to go about it, but it'll work for now. |
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| 15:41.24 | pouledodue | is there a channel dedicated to voip or similar on freenode |
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| 15:45.42 | [TK]D-Fender | pouledodue: No. VoIP is a generic concept for which there are many different protocols & implementations. |
| 15:48.54 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hrefdxqtlexoyxam) |
| 15:50.24 | Visage | Well, someone could start a ##voip topical channel, but it'd probably be very chaotic due to the above reason. :) |
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| 15:51.31 | anonymouz666 | Visage: you wanna talk about something generic or just a specific thing... let's say, sip. |
| 15:51.41 | anonymouz666 | there are a lot of places you can speak about it |
| 15:52.35 | Visage | Nah, I'm good. I'll stick to asterisk. :) |
| 15:56.07 | wasanzy | how can I execute two Asterisk Application on one line? like same => n,Playback(${GETHEARD_UPLOADSUCCESSFULL}) && Goto(getheard,100,1) |
| 15:56.27 | [TK]D-Fender | wasanzy: You can't |
| 15:56.32 | skrusty | you should just use background |
| 15:56.39 | [TK]D-Fender | wasanzy: 1 application per line./ Period |
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| 15:58.07 | wasanzy | ok |
| 15:59.05 | wasanzy | Record() can't accept 1, as input and end the recording? I have this Record(${GETHEARD_RECORDTEMP_PATH}temprecord-${RNDNUMBERS}:alaw,,180 y) |
| 15:59.13 | *** join/#asterisk michael-i (~michael-i@128.Red-88-19-15.staticIP.rima-tde.net) |
| 15:59.42 | wasanzy | and I wanted to press 1 for the recording to end and saved, 2 to end and remove the recording |
| 15:59.48 | michael-i | Is xmldoc now required in v13? I tried the usual configure trick of --disable-xmldoc but it still complains about missing that lib. |
| 16:00.44 | skrusty | wasanzy: by default you can only use # or * iirc |
| 16:00.49 | skrusty | to terminate the recording |
| 16:01.02 | wasanzy | nothing else? |
| 16:01.14 | skrusty | use option 't' to make the termiantor key '*' |
| 16:01.31 | skrusty | nope, not from what i can see in the docs |
| 16:01.31 | wasanzy | I want 1 not * |
| 16:01.44 | skrusty | as my mum always said, i want doesn't get :) |
| 16:02.31 | skrusty | you could terminate on ANY dtmf |
| 16:02.39 | skrusty | i.e. any number or #* |
| 16:02.47 | skrusty | using option 'y' |
| 16:03.01 | skrusty | which would simualte the termination by pressing 1 |
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| 16:45.34 | WIMPy | Oh. An exploded Asterisk at 1.5G. |
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| 17:11.13 | anonymouz666 | WIMPy: try to see the threads being used before killing it... backtrace process will show it. It was useful for me in the past. I discovered the most used threads and found the module responsible for the crash |
| 17:11.53 | WIMPy | It's an endless loop with Read() trying to play a non-existant file. |
| 17:13.13 | WIMPy | has just deleted 11G of log. |
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| 17:21.56 | michael_work | hey |
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| 17:23.03 | WIMPy | In the time, I've been at it, the log has grogessed by about half a minute. |
| 17:23.29 | michael_work | as i see g729 registration requires internet, am i right? what would be my procedure if i don't have any out trafic on some machines? |
| 17:23.48 | Qwell | michael_work: contact support |
| 17:24.07 | michael_work | Qwell, ok, thanks |
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| 17:26.41 | Johnny5 | michael_work: Manual license activation, contact Digium support |
| 17:26.52 | Johnny5 | late to the party |
| 17:30.25 | michael_work | thnaks Johnny5 |
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| 17:56.42 | michael-i | Is xmldoc required in 13 now? I tried disabling via the normal configure flag and it still complains about it missing. |
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| 18:02.50 | chillerz | Hi all, having some trouble with dodgy audio for a few seconds after picking up a transferred call. Using asterisk 13 (no nat) with snom 760 (behind nat) I have sip.confirm set as nat=force_rport,comedia |
| 18:03.04 | chillerz | Any ideas? |
| 18:03.16 | chillerz | Sip.conf* |
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| 18:30.57 | wasanzy | am trying to convert wav to alaw for use with asterisk sox airtel_ng/You_will_be_charge_n50.wav r 8000 -c 1 -e a-law you_will_be_charged.wav |
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| 18:31.20 | wasanzy | but am getting this error: sox FAIL formats: can't open input file `8000': No such file or directory |
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| 18:31.57 | Qwell | wasanzy: -r, not just r |
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| 18:34.37 | wasanzy | ok thank you |
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| 22:31.18 | lorenzo | hello, is there a way to execute something after one of these extensions is dialed? |
| 22:31.29 | lorenzo | http://pastebin.com/0icAA55U |
| 22:31.38 | WIMPy | That's the point of extensions. |
| 22:31.42 | lorenzo | like, if the call to 101 fails (busy, unavail, etc) play that audio |
| 22:33.07 | WIMPy | You probably want to use a Gosub or something to reuse for all extensions. |
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| 22:41.36 | lorenzo | ah okay |
| 22:41.42 | lorenzo | do I need Hangup() after Busy() too? |
| 22:41.55 | WIMPy | You want to. |
| 22:42.14 | WIMPy | Otherwise you risk falling through to another extension. |
| 22:42.24 | lorenzo | okay :) thanks |
| 22:42.53 | WIMPy | Which is something else you could take advantage of, instead of using a Gosub or the like. |
| 22:43.09 | WIMPy | But most people find that rather confusing. |
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| 23:35.13 | bfoote | hello! I'm on the hunt for an ata capable of ringing an old Western Electric telephone that's "unlocked" and would be able to talk to asterisk |
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| 23:36.35 | bfoote | I've been looking at the Grandstream ht702, the InnoMedia MTA 6328-4, the PAP2-NA and few others |
| 23:37.27 | bfoote | is there one (either those or others) which is en vogue and reasonably priced? |
| 23:38.03 | bfoote | I'm looking to connect to asterisk just for 'local use' for a small art project |
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| 23:41.04 | lorenzo | bfoote: cisco |
| 23:41.35 | bfoote | thanks lorenzo, which one do you like? |
| 23:42.35 | lorenzo | I've been using the spa122 just fine |
| 23:43.16 | lorenzo | you can get the spa112 which is simpler |
| 23:43.24 | lorenzo | and it's ata only, the 122 does some routing too |
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| 23:45.31 | bfoote | does the spa122 offer enough line power to trigger the ringer on an old bell rotary phone? |
| 23:50.08 | lorenzo | bfoote: sorry, no idea :/ |
| 23:51.19 | [TK]D-Fender | bfoote, SPA122 = 5 REN. SHould handle it fine |
| 23:51.37 | bfoote | fantastic |
| 23:51.39 | bfoote | thanks much |
| 23:52.28 | lorenzo | same for 112 |
| 23:53.24 | bfoote | are there features of a 122 which I should consider or for a simple setup can I "get away" with the 112? |
| 23:53.47 | lorenzo | if it's just an ata the 112 will do |
| 23:54.00 | lorenzo | the 122 does some routing too |
| 23:55.48 | lorenzo | it basically adds a wan port so you can connect e.g. your adsl modem and use the whole thing as a gateway |