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00:16.07 | iceyp | found it had to add z to the sendfax parameter |
00:16.13 | iceyp | :) cya |
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06:08.28 | volga629 | corosync for asterisk is only for 2 boxes located on same lan right ? |
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08:44.00 | linocisco | hello |
08:44.30 | linocisco | i have asterisk server at work. i want to save CDR on internet |
08:44.32 | linocisco | how to? |
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08:49.03 | PLMg | hello, did I enter the crontab job corectly? Amportal restart every day at 5am. 0 5 * * * /usr/local/sbin/amportal restart |
08:49.29 | PLMg | cause it didn't restart :( |
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09:08.55 | enzo | hello |
09:10.27 | enzo | Is it possible to define a directory when calling an agi ? like exten => _X.,2,AGI(some/path/to/script.agi) It will be relative to /usr/share/asterisk/agi-bin, so it will look for /usr/share/asterisk/agi-bin/some/path/to/script.agi, right ? |
09:17.38 | user258467 | Can someone help me to understand this asterisk extension confhttp://pastie.org/8391808 |
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09:18.53 | Ice_Strike | Hey |
09:26.00 | linocisco | user258467, what is not clear? |
09:28.45 | user258467 | Linkforsoad, are Answer(), Queue(), Hangup() built in functions? |
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09:30.59 | linocisco | user258467, Answer, and Hangup are builtin function. |
09:32.05 | user258467 | linocisco, and what about Queue()? |
09:32.09 | linocisco | user258467, Queue() is also DTMF based function, u can put parameter inside |
09:33.10 | linocisco | user258467, everyone is busy in Astricon 2013. I am the one who could not afford to attend such heaven event. so u can ask me and i can answer as far as I know. I m not so geek |
09:33.14 | user258467 | I don't know where sav3 is defined in the sample |
09:34.37 | user258467 | linocisco, why aren't you there :)? |
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09:35.03 | linocisco | user258467, honestly , no money |
09:35.26 | linocisco | user258467, but pity they dont have good internet at hotel to demonstrate. |
09:45.08 | user258467 | linocisco, I have a phone and I don't know its phone number what could I do to find it ? |
09:45.37 | user258467 | it is a gigaset a58h |
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09:54.10 | linocisco | user258467, it is ip phone. right? i dont know which voip service u scribed |
09:54.17 | linocisco | user258467, it is ip phone. right? i dont know which voip service u subscribed |
09:55.15 | Ice_Strike | Any clue what does this mean "${DB(record/${MACRO_EXTEN})}" |
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10:40.05 | linocisco | hi all |
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13:47.25 | volga629 | Hello Everyone is corosync for asterisk is only for 2 boxes located on same lan right |
13:47.28 | volga629 | ? |
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13:48.01 | anonymouz666 | 2, 3, 4 etc. |
13:48.34 | anonymouz666 | you gotta use multicast |
13:50.03 | volga629 | I wonder about distribution over the sites |
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13:50.22 | anonymouz666 | you mean WAN networks? |
13:50.39 | anonymouz666 | XMPP with PubSub |
13:51.20 | anonymouz666 | Astridevcon 2013 - "Kill the app_queue" |
13:51.22 | anonymouz666 | hehe |
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13:52.02 | volga629 | yes, I tried prosody and asterisk give constant errors, only option left tigase, but need rebuild rpm from centos is old |
13:52.43 | volga629 | in my case though vpn tunnels |
13:55.40 | volga629 | but prosody is better in my case, I don't know match about java in tigase |
13:56.45 | volga629 | and I can't find what asterisk exactly looking in PUB/SUB that I can check prosody support it |
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14:09.21 | Blashyrkh | is this the right place to ask about asterisk now? |
14:12.31 | Penguin | Maybe, maybe not. It depends on your question. |
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14:20.11 | ipalmer | hi all, I'm using asterisk 11.5 and using realtime. I've created an app that inserts endpoints into the sipfriends table and need to forcibly get them to register once I place them in the table without the endpoint rebooting. Is there a command to do this? |
14:21.28 | [TK]D-Fender | ipalmer: No, there is no such thing |
14:21.41 | [TK]D-Fender | ipalmer: Asterisk can't just make a phone register. |
14:22.01 | [TK]D-Fender | ipalmer: If the phone has some magic way of triggering this that's up to the phone. |
14:22.33 | [TK]D-Fender | ipalmer: for instance, Polycom's have a "remote reboot" command that could be targeted to it and pick up configs with updated info |
14:23.54 | ipalmer | D-Fender: so it's a case of configuring whatever I want to register to register itself then |
14:25.01 | [TK]D-Fender | ipalmer: depending on what you have. |
14:25.20 | [TK]D-Fender | ipalmer: But there is no generic :make him register to me!" concept |
14:25.49 | ipalmer | D-Fender: OK thanks |
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15:36.21 | Katty | if i put new audio files in the moh folder, do i need to reload anything? |
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15:41.21 | redotis | updated AsteriskNOW and it killed all my sound files. I'm starting to think the only way to manage an Asterisk box if by compiling it yourself. |
15:41.45 | Faustov | been there done that |
15:41.51 | redotis | I wanted to avoid that and just run straight Asterisk with AsteriskNOW and let them manage the updates but if they're going to kill my sound files. |
15:42.27 | Faustov | by introducing another layer between yourself and asterisk you are a) increasing the complexity b) losing flexibility |
15:42.30 | redotis | WTF is that all about? Does Digium push shit out like that so you move to their paid product? |
15:42.45 | redotis | Well it's not really that |
15:42.55 | redotis | I mean they could fuck up the compilation source too. |
15:43.04 | redotis | It's like they choose to do shit like that. |
15:46.01 | navaismo | wring updating asterisk cant kill sounds files |
15:46.13 | navaismo | you muts look what are you doing |
15:47.02 | navaismo | and no digium dont create SHIT opensource just to jump to the paid version, mostly the shitty admins screw the stuff |
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15:50.50 | [TK]D-Fender | The troll is strong with this one... |
15:50.58 | [TK]D-Fender | feels a disturbance in the farce... |
15:54.58 | Faustov | lol |
15:55.07 | Faustov | and at first I thgouth it was genuine |
15:59.09 | [TK]D-Fender | Oh, I'm sure it is... that's the worst part :) |
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16:03.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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16:11.55 | ipalmer | I'm trying to get an asterisk server to send calls one way to another asterisk server but am getting a failed to autheticate on the originating server. I'm sure I'm doing something stupid but can't see what, my config is here http://pastebin.com/aMYR0fw1 Any help would be appreciated Thanks |
16:12.40 | ChannelZ | do you actually have peers configured called "monkey" |
16:12.47 | ChannelZ | s/peers/a peer/ |
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16:13.04 | ChannelZ | oh RT. |
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16:13.26 | ChannelZ | but that's on the .2 box? You have nothing on .1? |
16:13.55 | ipalmer | no nothing on 1, I just want the calls to go from a to b and not from b to a |
16:14.11 | ChannelZ | but you're calling SIP/monkey and it has no idea what monkey is |
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16:14.41 | ipalmer | I thought that the monkey was the username being passed across |
16:15.18 | ipalmer | I thought the format was technology/username:password@server |
16:15.42 | ipalmer | sorry about that was supposed to be : |
16:18.02 | ChannelZ | sorry I'm delirious |
16:18.13 | redotis | navaismo...AsteriskNow update deleted ALL my sound files |
16:18.21 | redotis | yum update |
16:18.26 | redotis | it's their damn repository |
16:18.32 | redotis | so stfu |
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16:19.03 | ipalmer | ChannelZ: Huh? |
16:19.07 | redotis | It's happened to a lot of others. Google it beforeyou act like an ass. |
16:19.10 | ChannelZ | show us the failed auth |
16:19.46 | ipalmer | <PROTECTED> |
16:20.17 | navaismo | bows and roll eyes |
16:20.36 | ipalmer | and the dial that caused that [54321@internal:1] Dial("SIP/5000-00000408", "SIP/monkey:password@192.168.1.2,,T") |
16:20.46 | ChannelZ | .100? |
16:20.58 | redotis | Laughs at an ass that thinks he knows what he's talking about. |
16:21.09 | ipalmer | yeah i left the 00 off in the pastebin |
16:21.54 | ChannelZ | pastebin a SIP debug of the call |
16:23.20 | ipalmer | The pastebin is here http://pastebin.com/mxHL2PhA The usernames, passwords and ip addresses are different than the original pastebin |
16:24.11 | ChannelZ | yeah why |
16:25.14 | ipalmer | no real valid reason I suppose, just didn't want to put our companies internal stuff on a public channel |
16:25.34 | ipalmer | monkey = applianx, password = call |
16:26.25 | ChannelZ | 192.168 is unroutable... but anyway |
16:28.26 | ChannelZ | I never use the username/pw-in-a-Dial() syntax, not sure if it's not trying to auth by design or what... unless you cut off the paste because all I see is the original INVITE attempt and not the follow-up there should usually be after the 401. |
16:30.33 | ipalmer | maybe i did here's everything between dial and hangup http://pastebin.com/U6Mg5qQu |
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16:42.06 | ChannelZ | sorry I'm trying to do 4 things at once I probably shouldn't be trying to help you in this state :) |
16:42.28 | ipalmer | lol any help is appreciated |
16:42.41 | ChannelZ | Either that dial syntax is broken or something else is awry, it's trying to auth as your phone (5000) |
16:43.17 | ipalmer | what do you think is the best way forward? |
16:47.08 | ChannelZ | make a proper peer on the 'calling from' box |
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16:47.58 | ChannelZ | sorry errant return. Make a proper peer on the 'calling from' box for the 'calling to' box and Dial that |
16:49.15 | ipalmer | ok I'll do that. thanks for all your time |
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16:52.30 | runfromnowhere | So I'm hoping this isn't an insanely common question but I'm looking to get asterisk to record calls in two separate channels as opposed to just mixing them together. Trouble is I seem to be stuck working with v1.8 |
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16:53.50 | navaismo | are you using MONITOR cmd? |
16:54.04 | runfromnowhere | Yes, which I believe feeds into MixMonitor - let me confirm |
16:56.08 | Greenlight | Both MixMonitor AND Monitor can do this |
16:56.19 | runfromnowhere | The audio hook is MixMonitor and I'm setting the MONITOR_FILENAME but I'm not setting MONITOR_EXEC or anything |
16:56.49 | navaismo | use MOnitor instead mixmonitor |
16:56.54 | runfromnowhere | I've kind of inherited this system. We're also running FreePBX atop that so it looks like that's trying to manage all the config files |
16:57.23 | Greenlight | Hmm if you're using FreePBX you're likely best asking in #freepbx |
16:57.29 | Greenlight | ~freepbx |
16:57.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
16:57.32 | navaismo | right, ^ |
16:58.02 | runfromnowhere | Right, I was hoping to be able to configure something only on the Asterisk side |
16:58.12 | runfromnowhere | But if FreePBX being involved means they're the people to ask, I can head over there :) |
16:58.32 | Greenlight | Problem is that freepbx will try and overwrite changes you make |
16:58.59 | Greenlight | We did "hooK" into the recording stuff on freepbx systems, and there's a mechnism for doing overrides to their configs |
16:59.37 | navaismo | maybe in advanced settings menu, the setting "Use MixMonitor for Recordings" |
16:59.59 | Greenlight | And, I think Monitor is the way to go, as looking at the documentation, it was only from Asterisk 10 that MixMonitor began to support separate files (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MixMonitor) |
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17:00.54 | runfromnowhere | Ah great - thanks a ton |
17:01.05 | runfromnowhere | That's a good chunk of info for me to get started with :) |
17:02.22 | Greenlight | extensions_override_freepbx.conf I think is where we did stuff with recordings to change the default behaviour |
17:03.07 | Greenlight | If memory serves, there's a macro named something like "macro-record-check" which you can override, and then alter the recording method used |
17:03.29 | Greenlight | BUt, as I say, you're a lot safer asking in #freepbx |
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17:16.05 | monsterco | hi everyone - I am trying to get an Aastra phone to give me trouble shooting logs. I am using Kiwi Syslog and it's UDP port is set to 514 - I have set that in Aastra troubleshoot page with IP and debug levels all to 1 but I am not receiving anything. What could I be doing wrong? |
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17:24.22 | monsterco | anyone experienced with aastra syslog here? |
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17:27.30 | paulc | monsterco: Can you identify whether it's the Aastra not sending or your server not receiving? (how about running Wireshark to sniff the network?) |
17:28.47 | volga629 | Is this meesage possible network problem chan_sip.c:3905 __sip_xmit: sip_xmit of 0x7fdbdc01d3c0 (len 602) to ? |
17:29.22 | volga629 | client with TLS loosing registration each 2 min |
17:31.35 | monsterco | paulc - I am new to Syslog so let me get the fact straight first. I have kiwi syslog and I see port 516 UDP set to it. so, on the phone I should setup 516 UDP and then my windows pc IP and that's it? |
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17:32.02 | monsterco | UDP 514* |
17:32.57 | Chainsaw | Syslog. |
17:34.25 | paulc | monsterco: sounds reasonable. I've done syslog stuff with Sipura boxes but not the Aastra phones. I was trying to figure out which end to point the finger at.. if you can see the traffic in Wireshark it would suggest the PC/Kiwi isn't picking it up.. could be a windows firewall issue maybe |
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17:35.07 | monsterco | I just turned off firewall - so checking that again and I will reset the phone |
17:35.42 | monsterco | aastra GUI is relly bad sometimes - unresponsive etc |
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17:49.47 | paulc | monsterco: Yeah, I've had a few oddities with it in the past too, but the phones are pretty decent.. solid and stable.. that said, I haven't done any big roll outs with them. But they're decent as phones go |
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18:02.03 | redotis | Anyone know how to get a Snom to AutoAnswer for paging? |
18:02.13 | redotis | Snom 720 |
18:05.08 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:449b:f1a4:60f2:9790) |
18:07.16 | WIMPy | Their wiki. |
18:07.35 | WIMPy | Use standard headers or the phones config. Free choice. |
18:14.45 | redotis | I found it WIMPY. Thanks man |
18:14.49 | redotis | phones config huh? |
18:14.59 | redotis | Here's what worked for me. |
18:15.16 | WIMPy | There are several ways to do it. |
18:15.19 | redotis | exten => 2400,1,Set(VXML_URL=intercom=true) |
18:15.20 | redotis | exten => 2400,2,SIPAddHeader(Alert-Info: <http://www.notused.com>\;info=alert-autoanswer\;delay=0) |
18:15.20 | redotis | exten => 2400,3,Page(SIP/snom1&SIP/snom2) |
18:15.41 | redotis | and setting intercom to enabled on the phone and the intercome policy to always |
18:16.16 | redotis | I wonder what you have to do if you have multiple types of phones though. |
18:16.21 | redotis | with that sipheader stuff |
18:16.30 | WIMPy | If you want to page multiplephones, take a liik at multicastRTP. |
18:16.44 | redotis | what if they are all on the Internet though |
18:16.50 | redotis | not in your local network |
18:17.06 | redotis | anyhow...thanks a lot man |
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18:17.08 | redotis | :) |
18:17.11 | WIMPy | Well, then don't do multicast. |
18:17.15 | redotis | go to their wiki...derp |
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18:56.33 | bpietro | problem with twinkle softphone: deregister -> de-registration succeeded: 200 OK, register -> registration succeeded (expires = 3600 seconds), call (to test dialplan, MusicOnHold only) -> call failed 403 Forbidden? Why? Can pastebin all conf files and sip debug report |
18:56.56 | [TK]D-Fender | that'd be advisable |
18:57.12 | bpietro | ok, wait a moment |
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19:01.52 | bpietro | http://pastebin.com/q0acaJHX |
19:03.19 | [TK]D-Fender | bpietro: "sip show peers" |
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19:05.23 | bpietro | test-twinkle/test-twinkle 192.168.0.199 D a 5060 Unmonitored / Unmonitore 1 online |
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19:06.28 | [TK]D-Fender | bpietro: all of them... |
19:06.33 | [TK]D-Fender | PB it |
19:06.51 | zippo | how can I get the name of an outgoing channel name, as soon as it's created? |
19:07.01 | [TK]D-Fender | zippo: AMI |
19:07.26 | zippo | which command? |
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19:07.52 | bpietro | another two: acer-test/acer-test (Unspecified), gigaset-test (Unspecified) / Unmonitored 2 offline |
19:07.57 | [TK]D-Fender | zippo: It isn't a command, it is a NEWCHANNEL event |
19:08.16 | [TK]D-Fender | bpietro: proper clean pastebin please |
19:08.22 | bpietro | ok |
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19:10.23 | bpietro | http://pastebin.com/aFcPScRU but no more info the I wrote |
19:12.16 | [TK]D-Fender | bpietro: double check your account settings in twinkle |
19:14.26 | zippo | [TK]D-Fender: any idea how should I catch it and mount it as global variable? |
19:14.45 | bpietro | ok, tnx |
19:15.19 | [TK]D-Fender | zippo: You should probably give us a better description of the goal you are actually looking to accomplish in case this is not the best approach for doing so |
19:15.30 | WIMPy | zippo: maybe you should tell us what you really need instead of asking for some random detail. |
19:15.58 | zippo | ok |
19:16.53 | zippo | an outgoing call is made using Dial() application |
19:17.48 | zippo | need to catch that outgoing channel name immidiately, and use it on a completely different channel |
19:19.21 | WIMPy | wonders if that question sounds familiar. |
19:19.57 | WIMPy | How do you find that other channel or how does that other channel know about the existance of that call? |
19:20.01 | zippo | until now I used a script which check for it every second, but I think that's not the best approach |
19:20.59 | zippo | the outgoing channel rings some fixed SIP address |
19:21.48 | zippo | but asterisk appends changing data to the channel name, which I need to find |
19:22.07 | WIMPy | That doesn't answer any of my two questions. |
19:23.11 | bpietro | http://imagebin.org/273312 and http://imagebin.org/273313, I think it's OK but maybe I'm wrong |
19:23.17 | zippo | the other channel know about the call since it intentionally coded to look for it |
19:23.59 | WIMPy | So it can only be one call at a time? |
19:24.09 | zippo | yes |
19:25.21 | WIMPy | I'd wait for the newexten containing the dial and then setvar the info on the waiting channel and also redirect it somewhere so it doesn't have to loop. |
19:25.38 | zippo | so, is there a nice way to immidiately catch outgoing channel name used by Dial applcation? |
19:26.55 | WIMPy | Or even easier wait for the varset dialedpeernumber. |
19:27.19 | WIMPy | Just one event. |
19:29.31 | zippo | thanks. I'll google for dialedpeernumber |
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19:49.54 | monsterco | can anyone tell me what this line means: To: <sip:227%23@65.30.20.4:5060>;tag=as5f1bdca6 |
19:50.04 | monsterco | I have never seen a % - is that BLF or something? |
19:50.36 | jmetro | i would guess a typo |
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19:53.04 | WIMPy | It's an escape (URL encoding) |
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19:56.34 | zippo | WIMPy: that can't work since the outgoing channel is created when the Dial app starts running - and at that point it's impossible to execute any priority (like set var) |
19:57.09 | WIMPy | The dialplan application os calles Set(). SetVar is a manager command. |
19:57.36 | WIMPy | (and redirect) |
19:58.18 | zippo | so did you mean I shouod yous Setvar as AMI command? |
19:58.31 | WIMPy | yes |
20:00.12 | zippo | so need to catch the event first, then setvar |
20:01.10 | zippo | please tell me the command used to catch an event and I'll google for it |
20:01.49 | monsterco | WIMPy - why am I seeing that? the extension should be just 227 or 222... |
20:06.23 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
20:07.19 | pzn | after a Dial() command that failed (due to busy, unavailable, etc), how can I get the "error code/error messages"? which should I put in Noop() command to log this? |
20:10.31 | ChannelZ-Wk | ${DIALSTATUS} |
20:10.48 | zippo | pzn: NoOp( Dial Status: ${DIALSTATUS}) |
20:10.59 | ChannelZ-Wk | see 'core show application dial' |
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20:12.36 | zippo | does anyone know how to catch an event from AMI? |
20:16.26 | WIMPy | zippo: There's no command. It's an event. You just sit there and wait for it to happen. |
20:16.48 | WIMPy | monsterco: Looks like you 227# was called, nut just 227. |
20:17.15 | monsterco | WIMPy - you are right - it was forwarded - thanks |
20:17.24 | monsterco | now I have a different issue - it's one way audio |
20:17.29 | monsterco | I have sip debug posted here: http://paste.debian.net/55446/ |
20:18.07 | WIMPy | pzn: Additional information in HANGUPCAUSE and possibly in hash SIP_CAUSE. |
20:18.10 | monsterco | seems like other side is not receiving my NATed requests |
20:18.21 | pzn | zippo, WIMPy: thanks! |
20:19.14 | zippo | WIMPy: the problem I need to catch that programatticly... can't sit and wait |
20:19.44 | WIMPy | You start tour program befor it happens. |
20:20.02 | WIMPy | Usually you just let it run as a daemon. |
20:20.03 | zippo | zippo: I wish there was a way to automate this using 'expect' Linux program |
20:20.53 | WIMPy | Mightbe possible. |
20:20.56 | zippo | anyway, I see there is actually a 'waitevent' AMI command. |
20:21.00 | zippo | reading.. |
20:21.48 | zippo | http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-44.html |
20:21.55 | WIMPy | That's only for http. |
20:23.54 | zippo | I'm going to try it now with telnet |
20:23.55 | Nugget | telnet is eeeeeeevil! |
20:25.14 | jmetro | tenlet |
20:30.50 | monsterco | Usually is it good to keep NAT ALG set to ON on router or no? I am having one-way audio issues |
20:31.39 | WIMPy | That's usally causing more issues than it potentially solves. |
20:38.02 | monsterco | WIMPy - thanks. I will turn it off. CAn you tell me what is causing my one-way audio from this sip debug? - http://paste.debian.net/55446/ |
20:39.30 | WIMPy | I don't care about SIP issues any more. |
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20:44.28 | FreakWent | <PROTECTED> |
20:44.53 | FreakWent | The console just says the names of the audio files played, and shows no other relevant errors or info. |
20:45.37 | FreakWent | The text file named exists and looks correct to my untrained eye, it has info in there. Has anyone advice? |
20:49.16 | FreakWent | See, I reckon it's this: |
20:49.48 | FreakWent | "Set channel SIP/10-0000007a to write format ulaw" and that my sip.conf -- as suggested by the handset manufacturer -- has a number of codecs configured, but not uLaw. |
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20:55.56 | jmetro | so turn ulaw on |
20:57.28 | FreakWent | ok done |
20:57.29 | FreakWent | file.c: File digits/mon-5 does not exist in any format file.c: Unable to open digits/mon-5 (format 0x4 (ulaw)): No such file or directory say.c: Unable to play message digits/mon-5 |
20:57.33 | FreakWent | So there ya go. |
20:58.01 | FreakWent | So is it wasier to track down the right files, or can I tell Asterisk to not use ulaw and stick with alaw, which I'd prefer? |
20:59.23 | FreakWent | ... for context, confirming that all my digits/ and so probably all the other files are all suffixed with .gsm |
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21:05.30 | FreakWent | Well shutting ulaw out of sip_addit.... conf makes the same error but complainign about alaw |
21:05.48 | FreakWent | And predictably, shutting down alaw to gsm only makes the system unable to operate properlay at all |
21:05.49 | jmetro | you can convert them to alaw |
21:06.26 | FreakWent | I have gsm only. What I don't understand is why the system is happy to play all the menu prompts in gsm, the message in gsm, but refuses to use gsm to play the envelope. |
21:07.20 | FreakWent | It's annoying. However, I will use this opportunity to replace the sound files with an Australian English set instead of the US-accented defautls I think. |
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21:07.25 | FreakWent | Thank you very much for your time! |
21:11.51 | jmetro | asterisk also supports file conversion |
21:12.05 | jmetro | using "file convert /path/file.codec /path/file.newcodec" |
21:14.07 | FreakWent | Ha! Thankyou everyone for allowing me to go through this exercise so publicly. Attempting to use sox I discover "permission denied". chmod u+x on each subdirectory has rectified the problem. |
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21:14.16 | FreakWent | blushes and cringes away.... |
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23:49.00 | phix | ey, any one provisioned a Yealink before? |
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