| 00:00.28 | pdtpatrick_ | @p3nguin -- like i said im here for help and im not claiming to know more than i know. I do understand what extensions are and so far i've been able to navigate around. Im very new to the system so I don't understand what is your obsession with me knowing or not knowing. I might now understand fully what extensions are in terms of how asterisk use it but i understand the basics how it operates on the network. |
| 00:02.28 | p3nguin | Extensions aren't network mechanisms or devices. Extensions form the dial plan that does useful things when you dial numbers from your devices. |
| 00:03.07 | p3nguin | One of those useful things is to run the VoiceMail() application after having Dial()ed a device. |
| 00:03.43 | pdtpatrick_ | thanks for the information |
| 00:03.55 | pdtpatrick_ | but doesn't then extensions participate in the network? |
| 00:04.19 | p3nguin | Not really, no. |
| 00:04.57 | p3nguin | They don't participate on the network any more than your calculator does when you run calc.exe. |
| 00:05.19 | p3nguin | Extensions are rules that tell Asterisk how to process a call when you dial numbers from a phone. |
| 00:05.26 | pdtpatrick_ | ...currently the guy prior installed FreePBX and they are setting all the extensions on there and on the main box.. I have to setup voicemail.conf |
| 00:05.42 | pdtpatrick_ | all of which are done but when i dial that extension -- says call cannot be completed as dialed |
| 00:05.49 | p3nguin | FreePBX is outside the scope of this channel... |
| 00:06.04 | p3nguin | But what FreePBX controls is within the scope. |
| 00:06.40 | p3nguin | However, you aren't going to want to manually change Asterisk settings if you intend to continue using FreePBX. |
| 00:07.22 | pdtpatrick_ | FreePBX won't pick it up im guessing? or one overwrites the other? |
| 00:07.41 | p3nguin | If you no longer wish to use FreePBX, you can change voicemail.conf and extensions.conf and possibly have long-lasting success. |
| 00:08.34 | p3nguin | FreePBX will either ignore or overwrite any manual changes you make, unless you make them in accordance with FreePBX's methods (which is again outside the scope of this channel). |
| 00:09.10 | pdtpatrick_ | I might start working on that then. So in terms of getting this extensions to work (staying within this channel's scope) .. the two files I need to be concerned with are extensions.conf and voicemail.conf |
| 00:09.19 | pdtpatrick_ | do i need to reload anything from the cli once im done ? |
| 00:09.32 | pdtpatrick_ | usually when i change music on hold i have to do moh reload on the cli |
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| 00:09.59 | MikeH_ | is there a way to list identifiers for all channels? |
| 00:10.18 | p3nguin | If you change extensions.conf, you'll want to run "dialplan reload" and if you change voicemail.conf, you'll want to "module reload app_voicemail" |
| 00:10.47 | p3nguin | unless they finally added a voicemail reload command, which I don't see. |
| 00:11.18 | p3nguin | Alternatively, restart asterisk entirely to effect all changes. |
| 00:11.20 | cusco | MikeH_: what do you call identifiers? |
| 00:11.29 | cusco | MikeH_: core show channels concise ? |
| 00:11.51 | MikeH_ | cusco, I'm trying to get my head around FreePBX to be specific, adding trunks. |
| 00:12.10 | p3nguin | points to the FreePBX channel yet again. |
| 00:12.10 | cusco | I really dunno how freepbx works nor what you're trying to acomplish |
| 00:12.19 | cusco | but it seems a xyproblem |
| 00:12.28 | p3nguin | Second door on your left, sir. |
| 00:14.20 | pdtpatrick_ | p3nguin -- much appreciated |
| 00:15.49 | Preytell | can anyone explain, beyond the obvious what this error means: devicestate.c: No provider found, checking channel drivers for SIP |
| 00:16.29 | Preytell | I have a problem with asterisk 1.8.3 and polycom UC endpoints. They register after system boot, but then go away after a little over 1 hour. |
| 00:17.18 | Preytell | that error than causes this: Changing state for SIP/2134 - state 1 (Not in use), which is when the phone goes awol. |
| 00:19.37 | Preytell | funny thing is this all happens on the SIP/2.0 200 OK message back from the server, in which it states: PeerStatus: Registered. |
| 00:20.41 | Preytell | it happens to ALL endpoints at exactly the same time, a little over 1 hour after initial registration. |
| 00:21.20 | p3nguin | Have you tried any other user agent? |
| 00:21.20 | Preytell | I don't want to call bug, or deadlock if it's my own fault. |
| 00:21.38 | Preytell | yes, x-lite. |
| 00:21.48 | p3nguin | Does it do the same thing? |
| 00:21.58 | Preytell | yes. |
| 00:22.21 | p3nguin | Are these phones on the same LAN as Asterisk? |
| 00:22.51 | Preytell | yes and no, Asterisk is eth1 vlans to three subnets. |
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| 00:23.44 | Preytell | phones continue to work , i.e. making calls, just drop registration. |
| 00:24.13 | p3nguin | Right -- registration has nothing to do with making calls. |
| 00:24.19 | Preytell | correct. |
| 00:24.24 | Preytell | I mean nod. |
| 00:24.50 | p3nguin | What type of networking devices are you using to create the vlans? |
| 00:25.14 | Preytell | (don't laugh, it's not my choice) Dell Switches. |
| 00:25.26 | Preytell | PowerConnect 2824 |
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| 00:27.44 | Preytell | I HAVE thought of that, as I have another office on identical hardware, also running 1.8.3.3 but on HP 2848's, and they are not having this issue, but they also do not run the UC firmware from polycom yet. |
| 00:28.08 | p3nguin | X-lite isn't using the UC Polycom firmware either. |
| 00:28.20 | Preytell | sorry, didn't think about that. |
| 00:29.21 | p3nguin | I'd bet you it's a networking problem rather than a UA problem. |
| 00:30.01 | p3nguin | If possible, take your Asterisk computer, a couple Polycoms, and a PC with X-lite and put them all on a single dumb switch. |
| 00:30.06 | Preytell | not ruling that out, but it seems odd that rebooting the server fixes it, and the timing is near exact, withing a few minutes give or take. |
| 00:31.17 | Preytell | I'll set that up. See what happens. |
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| 00:59.58 | Preytell | Something I just noticed in the log: chan_sip.c: Reliably Transmitting (NAT) : Everywhere my conf NAT is disabled. And when things are working you see: chan_sip.c: Reliably Transmitting (no NAT), but when the phones drop it switches to NAT. Any ideas what would cause that? |
| 01:00.35 | p3nguin | I'll need to see your sip.conf. |
| 01:00.50 | Preytell | one moment. |
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| 01:04.44 | Preytell | http://pastebin.com/LgxS7AKi |
| 01:05.19 | p3nguin | More FreePBX? *sigh* |
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| 01:06.35 | Preytell | hehehe |
| 01:06.46 | p3nguin | I don't find it amusing at all. |
| 01:06.56 | p3nguin | Here I was trying to help, and you're wasting my time. |
| 01:06.59 | Preytell | I would blame them, but this is an asterisk issues, not freepbx. |
| 01:07.09 | p3nguin | No, it's a FreePBX issue. |
| 01:07.12 | Preytell | how so. |
| 01:07.18 | Preytell | they don't control registration. |
| 01:07.56 | p3nguin | FreePBX controls everything, and I should have been able to see sip.conf and find the NAT problem, but because it's FreePBX you'd have to show me half a dozen more confs, which I don't care to see. |
| 01:08.16 | p3nguin | THAT is how it's a FreePBX problem. |
| 01:08.56 | Preytell | none matter, except: http://pastebin.com/NmEKcRVV, and it contains nat=never. the only nat=yes in my entire config is on my trunk to flowroute. |
| 01:09.27 | WiretapWork_ | Preytell, ensure that externip is set, and localnet |
| 01:09.46 | Preytell | they are. and are correct. |
| 01:09.51 | WiretapWork_ | p3nguin, you really do hate freepbx don't you |
| 01:10.22 | p3nguin | No. I only hate when people try to fool me bringing their FreePBX shit into this Asterisk channel. |
| 01:10.56 | p3nguin | There's a reason we don't support FreePBX here. |
| 01:10.58 | Preytell | wow, ok. I have to WORK with asterisk, and it has to be SUPPORTED by staff that are not unix guru's. I am a RHCE, and Certified Network Engineer. They are not. |
| 01:11.35 | WiretapWork_ | Preytell, don't rage, there's no point |
| 01:12.27 | Preytell | Wow, make me wonder why I just spent $20,000 on digium boards over sangoma. I will take this up with Digium, I was just looking for a little help. |
| 01:12.28 | *** part/#asterisk Preytell (~Preytell@65.114.21.3) |
| 01:13.28 | p3nguin | I can imagine the laughing at Digium when they too find out he's using FreePBX. |
| 01:13.48 | WiretapWork_ | p3nguin, I don't think the hatorade was neccessary |
| 01:16.07 | p3nguin | Because I don't want to support FreePBX in a channel which specifically does not support FreePBX, that makes me a prick. Sure, that makes perfect sense to me. |
| 01:17.26 | Freeaqingme | why do people even choose to use freepbx if apparently there's no support for it? |
| 01:17.45 | p3nguin | There's some support for it... in the FreePBX channel. |
| 01:17.50 | *** join/#asterisk dr00d (~rtp.aster@b2A50.static.pacific.net.au) |
| 01:17.52 | Freeaqingme | heh |
| 01:18.01 | WiretapWork_ | p3nguin, your attitude makes you a prick, not your decision not to support it, that I can understand |
| 01:18.02 | Freeaqingme | crazy world |
| 01:18.03 | WiretapWork_ | but getting all well.... bitchy about it doesn't help anyone |
| 01:19.05 | mzb | I agree that more information is required. Some might think that p3nguin has been a bit rude with his response, but the bottom line is he can pick and choose who/how he helps ... that's his choice. |
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| 01:19.28 | WiretapWork_ | mzb, nobody was disputing that though |
| 01:19.35 | Freeaqingme | Wiretap, on the other hand, if you've been in this channel long enough (or a similar channel) you'll get tired of this kind of stuff and automatically turn a bit cynical |
| 01:19.49 | mzb | reads more scrollback |
| 01:19.54 | WiretapWork_ | Freeaqingme, I actually work with asterisk both with an without freepbx |
| 01:19.59 | dr00d | hi - i was wondering if anyone can tell me what is the best way to supply message waiting indication from asterisk to a legacy pbx ? i need to call a sip extension and then send some commands to turn mwi on and off for the legacy handsets ... |
| 01:20.39 | WiretapWork_ | Freeaqingme, and while I agree that it can/does make the config files a fucking nightmare to work with, in some situations, i.e. when it has to be supported by non-technical staff, there is some merit to it. |
| 01:20.43 | Freeaqingme | can you use sip notify through the pbx dr00d ? |
| 01:21.18 | Freeaqingme | WiretapWork_, there's merit to using it I suppose, but there's not so much merit if you then turn to an asterisk channel for support |
| 01:21.24 | Freeaqingme | (imho, of course) |
| 01:22.04 | WiretapWork_ | Freeaqingme, well, its not so illogical when you think about what freepbx is driven by, asterisk... most people would draw that link and think of freepbx as the web interface and asterisk as the driving software |
| 01:22.24 | WiretapWork_ | the distinction is not that clear in reality, but to newbies, it sure is gonna seem that way |
| 01:22.39 | Freeaqingme | WiretapWork_, yeah, but it's like joining #linux because someone has a problem with unity |
| 01:22.48 | WiretapWork_ | the one that amuses me has to be trixbox though :P |
| 01:23.06 | p3nguin | People come here asking for support for that, too. |
| 01:23.09 | dr00d | im not sure - the problem is that i need to call a voicemail extension on the legacy pbc and send *68504 for example to turn mwi for extension 504 on and #68504 to turn it off - i can do this by calling via an FXO gateway - but i cant figure out how to do this automatically when a voicemail message appears in an asterisk vm box |
| 01:23.40 | dr00d | we want to use asterisk voicemail for the legacy pbc btw |
| 01:23.45 | dr00d | pbx |
| 01:23.59 | WiretapWork_ | dr00d, use a macro to pull in the voicemail box instead of just using VoiceMail()? |
| 01:24.17 | dr00d | how do i do that |
| 01:24.37 | WiretapWork_ | have it check the message count before and after the voicemail call, if its greater, send an MWI |
| 01:24.43 | WiretapWork_ | no idea sorry :P I've not actually ever written a macro |
| 01:24.52 | dr00d | yes i know about that |
| 01:25.20 | p3nguin | How does the legacy PBX handle MWI? Does it handle it at all? |
| 01:25.50 | WiretapWork_ | p3nguin, after your line about trixbox support |
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| 01:26.14 | Thedr | Morning all |
| 01:26.33 | dr00d | yes - i have to go off hook on one of the legacy extensions configured for vm, then send for eg *68504 to turn on mwi for extension 504 |
| 01:26.44 | p3nguin | Oh, you have to send special DTMF. |
| 01:26.47 | dr00d | yes |
| 01:27.10 | WiretapWork_ | that is satisfyingly hacky :P |
| 01:27.10 | Thedr | Does anyone know if its possible to register a Cisco 7940 to 2 different asterisk servers? |
| 01:27.20 | WiretapWork_ | Thedr, should be able to do one per line |
| 01:27.21 | dr00d | so all i need to do is - in an inbound route - when a vm call comes in , to use the dial command to connect and send these dtmfs |
| 01:27.28 | WiretapWork_ | and IIRC theyre two-line |
| 01:27.36 | WiretapWork_ | just configure the second line to go to the other server |
| 01:27.54 | p3nguin | thedr: Are you using SCCP, MGCP, or SIP on the 7940? |
| 01:27.57 | Thedr | do you know the xml command to have a second proxy server? |
| 01:28.00 | Thedr | sip |
| 01:28.30 | dr00d | cant i somehow poll for new vm messages and then run a seperate process to call and send the dtmfs ? not as part of a dial plan ... |
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| 01:29.07 | WiretapWork_ | Thedr, are you using USECALLMANAGER at the moment for the proxy for the line buttons? |
| 01:29.38 | WiretapWork_ | dr00d, that sounds even hackier :P |
| 01:29.52 | dr00d | lol well do you have any suggestions ? |
| 01:30.13 | WiretapWork_ | dr00d, yes, I do, you saw it above |
| 01:30.22 | Thedr | at the moment I have proxy1_address: "192.x.x.x" |
| 01:30.29 | WiretapWork_ | that's not XML |
| 01:30.35 | Thedr | oh |
| 01:30.37 | WiretapWork_ | that's the old fashioned flatfile |
| 01:30.41 | Thedr | ah |
| 01:30.51 | Thedr | oops |
| 01:30.56 | WiretapWork_ | you've got proxy1_* and line1_* |
| 01:31.03 | WiretapWork_ | configure proxy2_* and line2_* |
| 01:31.12 | dr00d | do u mean - > dr00d, use a macro to pull in the voicemail box instead of just using VoiceMail()? ?? |
| 01:31.25 | Thedr | tried that |
| 01:31.55 | WiretapWork_ | dr00d, build a macro that does a check on the number of messages in the box, sends the call to voicemail, then when the caller rings off, checks the number again, compares it with the original, and sets MWI if its higher |
| 01:32.43 | dr00d | ok - i understand that - i just dont know how to build the macros |
| 01:32.45 | p3nguin | I have proxy1_address - proxy6_address in SIPDefault.cnf |
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| 01:33.05 | WiretapWork_ | dr00d, time to learn |
| 01:33.11 | dr00d | you are correct |
| 01:33.41 | Thedr | I'd rather keep the config in my seperate MAC files as there are only 2 phones that need seperate servers |
| 01:33.47 | dr00d | and then it needs to dial an extensionand send dtmfs after that |
| 01:33.51 | p3nguin | Everything line_* related is in SIP<MAC>.cnf |
| 01:34.24 | WiretapWork_ | dr00d, yep |
| 01:34.30 | dr00d | ok thansk |
| 01:34.32 | WiretapWork_ | dr00d, you'd do that as part of the macro |
| 01:34.39 | WiretapWork_ | or as part of a submacro |
| 01:34.45 | dr00d | ok ill go google macros |
| 01:35.01 | dr00d | thanks |
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| 03:31.24 | Besticles | I've browse everything I could on the net, I can't seem to find a solution to my problem. On certain PRI's I cannot get my outbound caller id set. I do have 2 diff ptsn. I have turned on pri intense debug, but I am not getting a complaint from what I can tell from the ptsn. I would really appreciate it if someone can look at the pri intense and see if they see something I dont. |
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| 03:31.34 | Besticles | My ptsn allows me to set my caller id number. |
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| 04:16.00 | dr00d | i have another question - when using the dial command, is it possible to hang up the call the dial plan rather than waiting for the other end to hang up, and so running the rest of the dial plan code ? |
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| 04:19.54 | p3nguin | dr00d: Take a look at Dial()'s option g. It could be useful to you. |
| 04:20.10 | dr00d | ok thanks |
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| 04:22.02 | dr00d | When the called party hangs up, continue to execute commands in the current context at the next priority... |
| 04:22.26 | dr00d | the problem is that i need to hang up the call from the dialplan not wait for the other end to hang up ... |
| 04:23.12 | dr00d | or ext the dial commmand some other way ... |
| 04:24.31 | p3nguin | In normal conditions, when a party hangs up, that's the end of the call. At that time, the h extension is executed. |
| 04:25.06 | dr00d | ok - all i am trying to do is call an extension, send some dtmf tones then hang up |
| 04:25.16 | dr00d | but from the dial plan tho |
| 04:25.18 | dr00d | and |
| 04:25.20 | dr00d | exten => 333,n,Dial(SIP/201,5,D(*68504)) |
| 04:25.34 | dr00d | this wont send the dtmf tones as per option d :( |
| 04:26.00 | dr00d | cries |
| 04:27.56 | p3nguin | If you call extension 333, the peer by the name of 201 as defined in sip.conf will be called and *68504 will be sent before the call between your phone and 201 gets bridged. |
| 04:28.19 | dr00d | oh |
| 04:28.36 | dr00d | so how can i send the dtmf after the call is connected then |
| 04:29.18 | p3nguin | The example we just discussed sends the tones after the call is connected, but before being bridged. |
| 04:29.48 | dr00d | ok i think i need to send it after the call is brided |
| 04:29.50 | p3nguin | Is SIP/201 the peer for your legacy PBX? |
| 04:29.53 | dr00d | bridged |
| 04:29.57 | dr00d | yes correct |
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| 04:30.27 | p3nguin | If you Dial(SIP/201), can you manually press the keys and get the result you're after? |
| 04:30.35 | dr00d | yes i can |
| 04:31.15 | p3nguin | What are you doing on the priorities before the Dial() that you showed me above? |
| 04:32.43 | dr00d | if you mean the dialplan code before the Dial(), i dont need to do anything, just simply dial, send dtmf, hang up |
| 04:33.02 | p3nguin | You can't start with priority n. |
| 04:33.10 | p3nguin | You have to have something before that line. |
| 04:33.32 | dr00d | exten => 333,1,Swift(Hi, this is extension 333 |
| 04:33.57 | dr00d | just that so i know im running the code i think i am |
| 04:34.03 | p3nguin | So extension 333 contains only those two lines? Swift() and Dial() |
| 04:34.05 | dr00d | with a ) at the end |
| 04:34.15 | dr00d | yes and hangup at the end so 3 lines |
| 04:34.22 | p3nguin | Okay, perfect. |
| 04:34.34 | dr00d | i sense that here is hope ... |
| 04:34.59 | p3nguin | Now I just want to know why the call needs to be bridged before the PBX accepts the DTMF. |
| 04:35.38 | p3nguin | Can you replace the SIP/201 with a peer that is another phone? That way when you call 333 you can answer the other phone and hear what your other PBX would be hearing. |
| 04:36.25 | p3nguin | Maybe you need to add some pause before sending the tones. Dial(SIP/201,5,D(wwww*68504)) |
| 04:36.30 | dr00d | this is for message waiting on the legacy pbx handsets - i need to send the dtmf tones to tell the pbx to switch on the message waiting |
| 04:36.44 | p3nguin | I remember from earlier. |
| 04:36.49 | dr00d | ok i will try that now |
| 04:37.04 | p3nguin | Try adding wwww before the * first. That'll be the easiest. |
| 04:37.07 | dr00d | ok |
| 04:37.33 | p3nguin | If still not working, change SIP/201 to another phone and answer the phone as if you are the PBX. |
| 04:37.41 | p3nguin | audible troubleshooting |
| 04:40.44 | *** join/#asterisk luisfelice (~luisfelic@190.39.218.64) |
| 04:43.16 | luisfelice | Hi, is it possible the set the host parameter in iax.conf with the fqdn of a phone? I would like that asterisk resolve the ip address of the host to see if it match but not using the IP address, instead I would like to use the fqdn, is it possible? |
| 04:43.27 | dr00d | ok i tried that - i can hear the dtmf tones when i pick up another phone - but it sounds like there are t lots of tones tho |
| 04:43.49 | p3nguin | luisfelice: You can use a host name instead of an IP address in the host parameter. |
| 04:43.50 | dr00d | it isnt doing what i want - if i use the dial pad on a phone and type the tones it works ok |
| 04:44.14 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
| 04:44.25 | p3nguin | dr00d: *68504 is six tones. You hear more than six? |
| 04:45.12 | p3nguin | luisfelice: host=phone1.domain.local |
| 04:46.09 | p3nguin | luisfelice: However... most phones will probably want to use registration. If your phone is registering, you'd probably want host=dynamic instead. |
| 04:46.35 | luisfelice | p3nguin: yes, I tried that but does not work, it doen't allow the phone to register |
| 04:46.40 | dr00d | i think i know why it isnt working - and no i hear **6688550044 |
| 04:46.54 | dr00d | i have to wait longer - ill just try that |
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| 04:48.40 | p3nguin | You can add more w to wait more. Each w is .5 second. |
| 04:50.35 | dr00d | ok im trying that now i think it will work |
| 04:51.14 | dr00d | the thing is - the more w s i put the longer it takes for the call to be answered ... |
| 04:51.25 | p3nguin | hmm |
| 04:51.49 | dr00d | cant i send the tones after the call is connected AND bridged ? |
| 04:51.50 | p3nguin | I don't think that's how it works. |
| 04:51.59 | dr00d | hmm |
| 04:52.57 | dr00d | it seems you have to wait about 5 seconds before you send the tones otherwise the pbx doesnt accept tones and do the mwi thing |
| 04:53.45 | dr00d | there is an answer i can tell |
| 04:55.04 | p3nguin | I suppose you could try running a macro which runs SendDTMF(). |
| 04:55.20 | p3nguin | 5 seconds = wwwwwwwwww |
| 04:57.15 | dr00d | ok - but how do i run this macro while the call is connected tho - ? because the dial command doesnt exit until the call is hung up ... |
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| 04:58.02 | dr00d | and i just found the S option which hangs up the call after so many seconds so that answers the other problem |
| 04:58.13 | p3nguin | I'm not sure if it would have a different result using the macro in the Dial command. |
| 04:59.24 | dr00d | oh - is this the answer ? - > Executes, via gosub, routine x on the called channel. This is similar to M above, but a gosub rather than a macro. |
| 04:59.35 | dr00d | option u(x) |
| 05:00.01 | p3nguin | My version does not have u, so I'm not familiar with it. |
| 05:00.05 | dr00d | ok |
| 05:00.11 | dr00d | i might try that |
| 05:00.21 | p3nguin | It sounds like it would do exactly what you want to do. |
| 05:00.35 | dr00d | im getting all excited now coz i think it is gonna work :) |
| 05:00.45 | p3nguin | I'd use SendDTMF() to send my tones. |
| 05:00.53 | dr00d | ok ill try that now |
| 05:01.07 | p3nguin | I just don't know the usage of that u option. |
| 05:02.59 | dr00d | is a subroutine a line of code in the current context such as exten => 333,n(send),SendDTMF(xxx) ? or a seperate macro |
| 05:04.19 | dr00d | its ok i got it figured out |
| 05:04.24 | dr00d | ill try it now |
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| 05:13.35 | dr00d | hey thanks heaps p3nguin that works - using the m option in the dial() command to run senddmtf with a few waits before hand - and i need to use the S option to hang up the call after a few seconds |
| 05:14.24 | p3nguin | Despite how hackish this method is, I'm glad it finally did what you wanted it to do. |
| 05:14.52 | dr00d | no probs - i can always refine it later |
| 05:15.33 | dr00d | gives p3nguin a big HUG |
| 05:16.02 | p3nguin | Watch it, I just got this shirt! |
| 05:18.43 | dr00d | lol |
| 05:18.45 | dr00d | sorry |
| 05:19.04 | dr00d | i think im ready to refine it now |
| 05:20.52 | dr00d | how can i run this macro when a new message arrive in the voicemail system ? at the momnent ive put this in an inbound route which is run whenever someone leaves a voicemail message - in this case they hear all the dtmf tones being sent and they have to wait while it all runs ... |
| 05:21.40 | dr00d | very hackish ... |
| 05:23.10 | p3nguin | Take a look at externnotify in voicemail.conf. |
| 05:23.16 | dr00d | ok |
| 05:23.22 | dr00d | oo i think ive seen that |
| 05:23.26 | dr00d | ill go loook |
| 05:24.04 | p3nguin | This is another option that I don't use ever, but it could potentially be useful in your situation. |
| 05:24.39 | dr00d | my brain has stretch marks from being excited, dissappointed, excited, more excited ... |
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| 05:38.32 | l1nuxman | where does asterisk specify the 'To:','From:',and 'CC:' fields? |
| 05:39.01 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 05:39.03 | kaushal | Hi |
| 05:39.25 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
| 05:39.52 | kaushal | Can someone please point me to the forum link wherein the Sangoma card doesnot work with the latest version of Ubuntu Kernel ? |
| 05:43.22 | dr00d | hey p3nguin - the next question is - can i run asterisk dial plan commands such as dial and senddtmf from a shell script ? |
| 05:43.41 | kaushal | kaldemar: you around ? |
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| 05:52.32 | schmidts | good morning |
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| 05:53.27 | jizzzum6 | where is asterisk 1.8.4? |
| 05:53.40 | jizzzum6 | http://www.asterisk.org/downloads is still showing 1.8.3.3 |
| 05:54.01 | jizzzum6 | anyone know? |
| 05:54.35 | schmidts | jizzzum6 try the link with "older asterisk versions" maybe you can find it in there |
| 05:55.24 | jizzzum6 | ah there it is... |
| 05:55.27 | jizzzum6 | thanks! |
| 05:55.45 | kaushal | hi schmidts |
| 05:56.23 | schmidts | hi kaushal |
| 05:56.33 | kaushal | schmidts: is there a known issue with Sangoma Card and the Ubuntu Kernel ? |
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| 06:06.25 | Wiretap | damnit, he quit |
| 06:06.30 | Wiretap | I had news for kaushal |
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| 07:27.45 | zkn | Hello |
| 07:28.51 | tuxx- | morning |
| 07:29.21 | Thedr | afternoon |
| 07:29.27 | zkn | I think I had a similar question before and at that time I was proved wrong by liefmadsen, hovever now I have now come across the same issue with channel variables getting lost when moving from one context to another... |
| 07:29.33 | zkn | only this time i'm using a macro |
| 07:30.32 | _schmidts | zkn a channel var wouldnt be lost if u just use a goto, only if you use a dial(LOCAL/ then it will be lost cause its a new "call" |
| 07:31.50 | zkn | I'm setting values to variables to Dial, and when Dial is executed with M option, then in the macro context these variables are lost, is that how it is supposed to be? |
| 07:31.56 | _schmidts | zkn if you want to make sure to have a channel var you should use something like this SET(__test=1234) instead of just SET(test=1234) with the "__" in front it will be a persistant var for this channel and also all bridged channels |
| 07:32.35 | _schmidts | zkn a macro is a little bit different, you can call the macro with arguments, maybe this will be the better way |
| 07:32.45 | zkn | _schmidts. mmm... okay. gotta try the persistent option |
| 07:33.26 | schmidts | you should try it with only one _ and also two __ cause there is a difference in how asterisk handle these vars then |
| 07:33.38 | zkn | okay |
| 07:36.17 | nunne | a sip phone sends a 302 temp. moved to it's context like Local/33333@users. but variables set before the 302 is recieved are gone.. why? |
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| 07:36.51 | nunne | i have a var set to redir=y, so it knows it's a redirection.. and doesnt try to call using the regular trunk |
| 07:37.06 | nunne | is there anyway to set the redirect-context? |
| 07:38.03 | kaldemar | a single _ means that the channel variable is inherited only once in created channels, two _'s means that they are inherited indefinitely. |
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| 07:39.27 | nunne | kaldemar, thanks! |
| 07:40.05 | kaldemar | nunne: actually, that was not an aswer for you, but it might help you too. :) |
| 07:46.43 | schmidts | has anyone of you tried 1.8 function CONNECTEDLINE? |
| 07:47.15 | schmidts | i want to update the information after an attendant transfer but i am not sure how to do this best ;) |
| 07:48.34 | zkn | mm..nope, preceeding variable name with _ or __ does not have any difference, when I do Dial(IAX2/"ph0n3numb3r",,M(sms)) then macro-sms will still not use the values set to variables in the context where Dial app resides.. in macro-sms context the same channel variables have no values anymore |
| 07:49.24 | schmidts | zkn then you have to use macro arguments |
| 07:49.50 | schmidts | i think this looks like this M(sms^ARG1^ARG2) |
| 07:50.20 | kaldemar | schmidts: iirc, the values should be updated upon a transfer automatically. |
| 07:50.38 | zkn | okay, i must admit i haven't yet figured out arguments... how do I know what argument will correspond to what value or how do I determine these? |
| 07:50.55 | schmidts | kaldemar thats what i thought too but it doesnt work :( |
| 07:53.22 | zkn | ok, reading the book : |
| 07:53.24 | zkn | :) |
| 07:53.41 | schmidts | zkn take a look at the extensions.conf.sampe file there you can find how to act with macro arguments and btw with core show application DIAL the part of the M you will see that you dont have "pbx features" in this macro and channel vars are part of these pbx features |
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| 08:02.11 | schmidts | kaldemar i didnt see any connectedline updates even in sip history :( |
| 08:02.35 | schmidts | are there other options i have to activate then sendrpid and trustrpid? |
| 08:05.03 | kaldemar | rpid_update=yes might be worth a try. |
| 08:05.37 | schmidts | will do so |
| 08:06.30 | schmidts | not really but maybe i am just doing it wrong |
| 08:14.17 | zkn | hmm.. i still don't see where macro arguments get their values from.. |
| 08:17.59 | kaldemar | zkn: from the line that exectes the macro. |
| 08:18.26 | kaldemar | zkn: how are you executing it? with the Macro app or as a Dial app option? |
| 08:18.43 | zkn | I do Dial(IAX2/"ph0n3numb3r",,M(sms)) |
| 08:19.45 | schmidts | zkn as i said above you have to put the arguments into the M statement like this |
| 08:20.02 | schmidts | M(sms^${EXTEN}^${smstext}) |
| 08:20.17 | schmidts | then you can access the arguments with ${ARG1} and ${ARG2} and so on |
| 08:20.23 | zkn | aah okay |
| 08:20.29 | zkn | making more sense now |
| 08:20.35 | schmidts | the ^ is the delimiter for these vars ;) |
| 08:20.46 | zkn | so i need to pass the variables to my macro separately |
| 08:21.01 | schmidts | yes |
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| 08:40.29 | schmidts | kaldemar when i take a look at handle_request_refer i didnt see anything related to connectedline maybe something like this isnt implemented? |
| 08:40.50 | atan | I'm getting ' Failed to parse contact info' when my phone tries to register |
| 08:45.58 | atan | http://www.spinics.net/lists/asterisk/msg09422.html |
| 08:46.12 | atan | Can I somehow override this in sip.conf? |
| 08:47.10 | schmidts | atan which asterisk version do you use? 1.0 or 1.2 i dont think this is still a problem |
| 08:47.29 | atan | The latest 1.8 |
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| 08:47.49 | schmidts | then a bug from year 2005 should be a prbolem ;) |
| 08:48.02 | schmidts | maybe you should do a sip debug to see what contact header you phone sends |
| 08:48.51 | atan | Oh no doubt the phone is sending the wrong headers :-( |
| 08:49.26 | schmidts | ;) |
| 08:50.01 | kaldemar | maybe pedantic=no would help if the phone sends crappy headers. |
| 08:52.14 | atan | register_verify: Failed to parse contact info |
| 08:52.19 | atan | Even with that =\ |
| 08:52.39 | schmidts | atan show us the ouput of sip debug then we can maybe see what cause this problem |
| 08:53.24 | atan | You got it. One momento! |
| 08:54.50 | atan | http://pastebin.com/b4b51EZ8 |
| 08:55.47 | atan | If it's of any help the device connecting is one of those Android phones. In the latest Google firmware they have "SIP support" =\ |
| 08:59.29 | schmidts | atan i meant the sip debug not the asterisk debug output ;) sip set debug peer 1234 |
| 09:01.38 | atan | You got it :-) |
| 09:05.03 | atan | Hmm. All it shows is chan_sip.c:13843 register_verify: Failed to parse contact info when it tries to register. |
| 09:05.20 | atan | I wonder if my nat thing has anything to do with it |
| 09:05.42 | schmidts | atan instead of 1234 you have to use your peer name ;) but maybe you will see something with sip set debug on |
| 09:06.21 | atan | Peer is 1133, I changed it to that when I ran it |
| 09:06.29 | atan | SIP Debugging Enabled for IP: 216... |
| 09:06.42 | atan | Okay now it appears registered. Hmm. Sec. |
| 09:07.01 | atan | And shows as registered. Strange. Let me see if it will take a call. |
| 09:07.39 | schmidts | :D |
| 09:08.12 | atan | <PROTECTED> |
| 09:08.19 | atan | <PROTECTED> |
| 09:10.17 | kaldemar | that is not likely to cause a no audio issue. |
| 09:13.36 | atan | Okay so it tries to register once, fails with can't parse contact then tries again in a few seconds and works |
| 09:47.53 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:52.19 | angryuser | Good day, i need a softphone which auto pick ups incoming calls, do you know one ? Thank you. |
| 09:53.00 | schmidts | angryuser i am not sure but i think ninja could do this |
| 09:53.29 | kaldemar | pickups or answers? |
| 09:53.41 | angryuser | Erf, answers |
| 09:53.58 | angryuser | kaldemar, ^ |
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| 09:59.03 | k3asd` | hi |
| 10:00.20 | angryuser | schmidts, ninja lite can not do this |
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| 10:24.00 | Jasnejac | angryuser: zoiper communicator supposedly supports server-ide auto answer |
| 10:24.18 | angryuser | Jasnejac, only in pro version |
| 10:24.47 | Jasnejac | angryuser: that's what I have. not seen it anywhere else |
| 10:25.16 | angryuser | Jasnejac, i've seend it on sjphone, kapanga phone or x-lite pro |
| 10:26.26 | angryuser | too old, or not free |
| 10:27.22 | Jasnejac | angryuser: how about setting the answerphone option in blink to zero seconds? |
| 10:27.49 | angryuser | Jasnejac, hm Blink, right forgot about this one |
| 10:28.22 | Jasnejac | blink is good. I like blink |
| 10:28.58 | angryuser | Jasnejac, yes, done by ag projects |
| 10:31.32 | angryuser | Jasnejac, i sont see the options you mentionned |
| 10:32.03 | angryuser | Jasnejac, there is ony answerring machine in there |
| 10:32.22 | Jasnejac | angryuser: under preferences->audio. may not work of course |
| 10:32.33 | angryuser | Jasnejac, it is written: |
| 10:32.42 | angryuser | Answer delay for the answerring machine |
| 10:33.09 | angryuser | Jasnejac, so it is useless |
| 10:34.10 | Jasnejac | not tried it but I was thinking maybe set to zero seconds, record a blank (very short message) and see what happens. if its no good I have no other suggestions |
| 10:43.30 | zkn | Hey, I've got one issue I'd like to discuss with you guys... maybe someone knows a trick or a way around this... so, thus far I have tried both with options M and U to run a subroutine with Dial app when the callee has answered the call.. what my subroutine does is that it will execute a shell script in the system that will send sms to the callee about the call, but the problem here is that the script takes time to execute and finish in the system and while it |
| 10:43.30 | zkn | <PROTECTED> |
| 10:45.32 | kaldemar | zkn: & |
| 10:47.41 | kaldemar | zkn: System(/path/to/script &) <-- makes it run in background and the system app immediately returns. |
| 10:47.54 | zkn | oh, i see... |
| 10:47.58 | zkn | gotta try that |
| 10:54.24 | zkn | kaldemar, yep, works now exactly how it should ! :) thanks a bunch... also thanks to schmidts, i've learned some very nice tricks today! |
| 11:03.40 | schmidts | your welcome ;) |
| 11:10.05 | zkn | any ideas what to check in the system when Asterisk has become unresponsive..e.g. not possible to register in to the server nor make calls when you have been registered previously nor does it restart with CLI command restart now |
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| 11:12.19 | zkn | also it is not accepting incoming calls... so i have to restart from init script, but before I do that I'd like to check what you would check in similar cases from opsys side what might be going on |
| 11:12.20 | cusco | cli |
| 11:12.27 | cusco | check the cli for warnings7errors |
| 11:12.35 | zkn | nope nothing |
| 11:12.45 | cusco | core set verbose 15 |
| 11:12.54 | cusco | dial to it from outside or something |
| 11:12.57 | cusco | and look at the error |
| 11:13.03 | zkn | it was 10 |
| 11:13.42 | cusco | and no error when you dial to it? |
| 11:13.58 | zkn | ok, now the server became responsive again.... so i'm thinking that some resource issue occurred |
| 11:14.07 | cusco | o.O |
| 11:14.21 | cusco | perhaps its not asterisk it self |
| 11:14.25 | cusco | but the OS? |
| 11:14.30 | cusco | or that system command you tun... |
| 11:14.32 | cusco | run |
| 11:14.43 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:14.48 | cusco | perhaps you should monitor the os |
| 11:15.46 | kaldemar | zkn: see what happened the last in logs to give you some kind of information on how to reproduce the issue. |
| 11:15.57 | zkn | yea... i guess something in the system, yep... although in htop everything looked OK - mem and cpu usage. for example.. |
| 11:18.41 | zkn | well, occasionally i do get some weird messaes |
| 11:18.46 | zkn | messages* |
| 11:22.26 | *** join/#asterisk Infin1ty|work (~erez@pdpc/supporter/active/infin1ty) |
| 11:22.29 | zkn | such as this: chan_iax2.c: Max retries exceeded to host XXX.XXX.XXX.XXX on IAX2/iaxtrunk-5922 (type = 6, subclass = 11, ts=899719, seqno=14) |
| 11:23.58 | zkn | but these i have seen occur when then inbound caller has ended the call but Asterisk did not hangup on the channel for callee |
| 11:24.13 | zkn | s/then/the |
| 11:25.43 | zkn | haven't found yet the setting that adjust this detection sensitivity |
| 11:26.16 | zkn | because when this particular error happens then that channel is locked |
| 11:26.29 | zkn | SIP trunk works, but IAX2 doesnt |
| 11:26.55 | zkn | until I manually hangup the request in CLI |
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| 11:40.32 | Diffen | Hello, Is it possible to implement a linear queue strategy in Asterisk 1.4? |
| 11:44.09 | kaii | what is a linear queue strategy ? |
| 11:44.28 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 11:45.32 | Diffen | first 3 signals on agent 1 then the queue call is moved to agent 2 for 3 signals and so on |
| 11:47.03 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 11:47.51 | QuantumSchema | Hey all! I've hit a pretty big wall after upgrading to 1.8.3.3 and I could use some help.. |
| 11:48.24 | QuantumSchema | Here's the pastebin for the dialplan... http://pastebin.com/CqUUDuZW |
| 11:49.46 | QuantumSchema | With the dial plan listed in the PB, I was able to hit 1 in the queue and it would send me to voicemail (7226). Now with 1.8.3.3, it either doesn't do anything or it changes the queue to a ringall.... |
| 11:49.48 | kaii | Diffen: you mean roundrobin ? |
| 11:50.00 | QuantumSchema | Does anything look wrong? |
| 11:51.15 | QuantumSchema | Sorry... missed a part... here's the updated PB... http://pastebin.com/uBsnUA6C |
| 11:51.30 | Diffen | kaii: well if round robin is that it always rings on agent 1 for 3 signals and then to agent 2 for 3 signals and so on, yes that´s what i mean. i thought round robin was that first agent 1 takes the first call, the second call will be directed to agent 2 and not agent 1. |
| 11:52.15 | kaii | QuantumSchema: which asterisk version did u use previously |
| 11:52.23 | sezuan | Is there an ISN test number? Both numbers on the freenum website, 1234*256 and 613*262, seem to be down. |
| 11:52.42 | leifmadsen | sezuan: I can set you one up real quick, hold please |
| 11:53.17 | QuantumSchema | 1.8.3.2 |
| 11:53.17 | kaii | Diffen: in future versions it will be like that, yes. roundrobin has been superseded by rrmemory (roundrobin with memory) which remembers the last position in the round robin and continues there in the next call. |
| 11:53.37 | kaii | QuantumSchema: sounds like a bug then somehow |
| 11:53.46 | kaii | QuantumSchema: you have verbose logs to look at? |
| 11:54.01 | kaii | Diffen: *future: >1.4 |
| 11:54.35 | Diffen | kaii ok so round robin in 1.4 is without the memory so it always starts on agent 1 (if that agent is avaiable) |
| 11:54.51 | kaii | if i remember correctly, yes. |
| 11:55.06 | Diffen | COOL |
| 11:55.46 | kaii | Diffen: oh noes, i just realised that it is only in 1.2.. all versions >1.2 do memory |
| 11:56.02 | QuantumSchema | Trying to pull it down now... 55MB |
| 11:56.05 | Diffen | kaii: crap! is it possible to remove the memory? |
| 11:56.12 | leifmadsen | sezuan: 1234*460 <-- try that |
| 11:56.12 | sezuan | leifmadsen: thanks |
| 11:56.47 | *** join/#asterisk jacc0 (~jacc0@94.157.231.110) |
| 11:56.51 | jacc0 | hi all :) |
| 11:57.31 | jacc0 | I am recieving an invite that shows rtp en srtp is supported |
| 11:57.53 | jacc0 | and asterisk cli says : Can't provide secure audio requested in SDP offer |
| 11:58.12 | jacc0 | but the invite also supports rtp |
| 11:58.19 | sezuan | leifmadsen: works. |
| 11:58.25 | jacc0 | the connection fails |
| 11:58.37 | leifmadsen | sezuan: enjoy :) |
| 11:58.42 | jacc0 | what can I do about it? |
| 11:58.49 | jacc0 | hi leifmadsen |
| 11:58.53 | kaii | Diffen: uhm try it out .. i think "deprecated" means its still in this version, but is going to be removed. in 1.6 there is strategy "linear". gjust give "roundrobin" a try |
| 11:59.44 | *** join/#asterisk derbasti (~Miranda@dslb-178-008-109-208.pools.arcor-ip.net) |
| 11:59.47 | leifmadsen | kaii: deprecated typically means there is a better way to do something, but the feature is not necessarily removed -- it's just not supported in favour of the newer, better method |
| 12:00.43 | kaii | leifmadsen: thanks for the clarification. but that implies "could be removed" |
| 12:01.56 | leifmadsen | kaii: "could", but not necessarily. The rule is typically to leave things in unless they get terribly broken or something. If they continue to work for someone, then they get left in for backwards compatibility. |
| 12:02.13 | leifmadsen | Personally, I'm not sure why 'roundrobin' is deprecated because rrmemory does something different |
| 12:02.27 | kaii | agreed |
| 12:02.41 | leifmadsen | I think that may have been a mistake, and I'm not sure it's truly "deprecated" anymore |
| 12:06.13 | creativx | queues.. always an issue |
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| 12:07.48 | QuantumSchema | Kaii: Here's the PB to some verbose logging.... http://pastebin.com/GaczH4TG |
| 12:07.58 | QuantumSchema | Please let me know if you need any more turned on... |
| 12:09.55 | basti1101 | hello, is it possible to use asterisk with tls for sip and checking the client certificate? i only see ways to verify the server certificate and the client is accepted by authname/password. but the client certificate would be nice too. is there a solution? |
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| 12:21.27 | kaii | QuantumSchema: woah that is very noisy. TLDR |
| 12:21.38 | QuantumSchema | TLDR? |
| 12:22.00 | kaii | QuantumSchema: reduce your test to 1234,Queue(mytestqueue) and the members of that queue to a single static member without local channels and macros and stuff |
| 12:22.26 | kaii | QuantumSchema: Too Long Didnt Read: 260KB of log for a single call test case? wtf. |
| 12:23.52 | *** join/#asterisk glam (~glam@113.87.99.192) |
| 12:27.14 | Diffen | kaii: ive tried the roundrobin and its not working good on my 1.4... have to figure something else out to get the queue calls linear on the agents. |
| 12:29.04 | *** join/#asterisk billmania (~bill@38.98.130.98) |
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| 12:33.34 | kaii | Diffen: what do you mean "not working good" |
| 12:34.32 | Diffen | kaii: well it acting as round robin :) random what agent who will get the call. Are there not anyway to decied that agent 1 always gets the first call in the queue in 1.4? |
| 12:36.04 | *** join/#asterisk jacc0 (~jacc0@94.157.231.110) |
| 12:37.50 | QuantumSchema | Looking through the logs it seems that it's never entering the queue6226-voicemail context.... |
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| 12:49.55 | billmania | Morning all. |
| 12:51.01 | billmania | What does asterisk call the feature whereby the user of SIP deskset A can retrieve a call which has been placed on HOLD on SIP deskset B and converse with that caller? |
| 12:51.10 | billmania | Is that "parking" or is that a "pick"? |
| 12:51.43 | kaii | QuantumSchema: yes thats what a quick search gave me too |
| 12:53.21 | kaldemar | billmania: parking |
| 12:53.32 | billmania | kaldemar: Thanks. |
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| 13:10.41 | basti1101 | is there a client certificate validation for sip over tls for asterisk? i can only find server certificate validation |
| 13:10.53 | russellb | no |
| 13:11.50 | Diffen | Is it possible to add a code that starts a meetme confernece? for example if I have 10 people that have called a meetme conference room are set on hold and when i want to start the conference I enter a code that starts the conference and all participants are sent to my conference room. |
| 13:12.22 | basti1101 | but, why do they generate client certifiactes in the asterisk-wiki? (https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial) |
| 13:12.25 | russellb | sort of ... there is an option to wait for a marked user to join before it starts |
| 13:13.12 | Diffen | russellb: ok so if there are 10 uses that are waiting they cant talk to each other until the marked user enter the room?` |
| 13:14.09 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 13:14.19 | leifmadsen | Diffen: exactly |
| 13:14.31 | leifmadsen | Diffen: just look at the options to MeetMe and look for the marked user stuff |
| 13:15.02 | blackdoor67 | any one know how to spoof call using astrisk |
| 13:15.19 | Diffen | leifmadsen: ok nice. there are not possible to change the marked user for a code? :) |
| 13:15.23 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 13:15.46 | kaldemar | blackdoor67: please elaborate |
| 13:17.01 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:18.29 | leifmadsen | Diffen: I don't understand your question |
| 13:18.45 | leifmadsen | oh, I know what you mean |
| 13:18.52 | Diffen | :) |
| 13:19.01 | leifmadsen | just have someone enter a code before joining the conference room or something |
| 13:19.38 | leifmadsen | I could probably think of something like an application map that would transfer the person to another part of the dialplan to enter a code, then enter them back in to the conference as the marked user |
| 13:19.44 | leifmadsen | you just have to be creative |
| 13:20.07 | Diffen | leifmadsen: unfortunately im not that good at Asterisk... yet I hope :) |
| 13:20.40 | leifmadsen | just keep playing and thinking outside the box :) |
| 13:20.53 | Diffen | leifmadsen: its not any hurry its just nice to not be needed to start a conference call that you need to use just one single phone. |
| 13:21.15 | Diffen | phone or extension. it would be nice if you could start the conference call from a cellphone |
| 13:21.21 | leifmadsen | damn you, now you got my interested in doing this for a cookbook recipe |
| 13:21.22 | Diffen | leifmadsen: will try to do that :D |
| 13:21.25 | leifmadsen | russellb: ^^^ |
| 13:21.39 | Diffen | leifmadsen: :D |
| 13:21.49 | russellb | leifmadsen: to the google doc? |
| 13:22.12 | Diffen | haha sounds like "to the bat mobile!" |
| 13:22.29 | leifmadsen | it's basically the same thing |
| 13:22.34 | leifmadsen | russellb: to the google doc!! |
| 13:22.52 | Diffen | :D |
| 13:22.52 | leifmadsen | russellb: I bet a bunch of recipes for ConfBridge() and MeetMe() could be gleamed... |
| 13:23.07 | russellb | leifmadsen: sounds like a fun chapter |
| 13:23.28 | russellb | a little early for ConfBridge() |
| 13:23.38 | leifmadsen | agreed |
| 13:23.42 | leifmadsen | could do a bunch of MeetMe() things though |
| 13:23.48 | russellb | yup |
| 13:23.53 | leifmadsen | and take some ideas for ConfBridge() for 1.10 even if we don't write them yet |
| 13:24.06 | leifmadsen | we're almost going to have to start marking things for specific branches |
| 13:24.15 | leifmadsen | "Works as of: 1.X" |
| 13:24.22 | russellb | yes |
| 13:24.24 | *** join/#asterisk moy (~moy@CPE003048b11058-CM00222d6b4d65.cpe.net.cable.rogers.com) |
| 13:24.29 | leifmadsen | moy: o/ |
| 13:24.51 | Diffen | while im at it, are the acd agents removed in 1.6 or have I got that totally wrong? |
| 13:25.44 | leifmadsen | AgentCallbackLogin() has been removed for a while |
| 13:25.51 | leifmadsen | you should build it using dialplan really |
| 13:26.02 | leifmadsen | there are docs even :) |
| 13:27.02 | Diffen | leifmadsen: hmm ok we used AgentCallbackLogin for users that wanted to use their cellphone in queues. Worked perfectly, do you have any nice suggestion on how to get something like that in 1.6? |
| 13:27.37 | leifmadsen | I presume you mean 1.6.2. |
| 13:27.51 | Diffen | yes |
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| 13:28.17 | leifmadsen | yes, use some dialplan to AddQueueMember() to the Queue() -- it can either be a SIP channel, or a Local channel, etc. |
| 13:29.25 | leifmadsen | russellb: idea added |
| 13:30.01 | russellb | yay |
| 13:31.21 | Diffen | leifmadsen: ok so i can use something like this: AddQueueMember(techsupport|$CALLERID(num)) |
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| 13:31.53 | leifmadsen | not quite, but close |
| 13:32.09 | leifmadsen | AddQueueMember(techsupport,SIP/my_itsp/${CALLERID(num)}) |
| 13:32.25 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
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| 13:36.08 | Diffen | leifmadsen: sweet, hmm and to see what members that are logged in i do a queue show queue-name? |
| 13:36.11 | Diffen | in the CLI |
| 13:37.20 | leifmadsen | queue show |
| 13:37.27 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 13:37.30 | Freeaqingme_ | ~thebook |
| 13:37.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
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| 13:38.40 | Diffen | leifmadsen: ok good. is it 2nd or 3rd edition of your book i should buy for 1.6? what do you recommend? |
| 13:38.50 | Diffen | both i guess but.. :D |
| 13:39.09 | Freeaqingme_ | I just regged http://bit.ly/TheAsteriskBook |
| 13:39.25 | Dovid | <PROTECTED> |
| 13:39.34 | Dovid | or does it mean i am listening on this port ? |
| 13:39.38 | *** join/#asterisk Eriatolc (~chatzilla@theleme.adsl.unimedia.fr) |
| 13:39.59 | Eriatolc | hi everyone |
| 13:40.25 | Eriatolc | I've a pb with Asterisk-gui |
| 13:40.36 | Eriatolc | can someone help ? |
| 13:41.29 | Freeaqingme_ | just askyour qestion(s) ;) |
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| 13:43.15 | leifmadsen | Diffen: we didn't write a book for the 1.6.x series specifically |
| 13:43.22 | leifmadsen | A:TDG is based on Asterisk 1.8 |
| 13:43.35 | leifmadsen | Eriatolc: there is #asterisk-gui as well |
| 13:44.13 | Eriatolc | leifmadsen: thx, i'm going to this chan :) |
| 13:45.11 | Eriatolc | i've "just" a 404 error when i'm trying to load the config/index.html page |
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| 13:49.20 | *** join/#asterisk Rewt` (rewt@192.94.73.16) |
| 13:49.25 | Lantizia | Hey what else can cause SIP 488 (not acceptable here) other than codec incompatibility (checked that) |
| 13:49.36 | Lantizia | or should I paste some logs? |
| 13:49.43 | Rewt` | is there a good reporting tool for Asterisk? |
| 13:49.44 | Lantizia | i.e. on a nopaste site :P |
| 13:50.03 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 13:50.17 | Freeaqingme_ | Rewt`, to report what? |
| 13:51.04 | Rewt` | usage, missed calls, etc |
| 13:53.15 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:53.36 | aberrios | Rewt`: What kind of environment are you wanting this for? Private use? Office Use? Queues? |
| 13:53.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:53.47 | Rewt` | Office use |
| 13:54.43 | aberrios | As long as its not getting pounded with tons of calls then FreePBX's interface for the CDR is nifty |
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| 13:57.50 | Rewt` | ok |
| 13:58.16 | Rewt` | depends on your definition of pounded |
| 13:58.20 | Rewt` | 200 calls a day |
| 13:59.09 | Freeaqingme_ | Rewt`, I'd suggest to just it out and see if it fits your needs ;) |
| 13:59.34 | aberrios | i'd call that not much |
| 13:59.58 | Dovid | hi all. I am having an issue where a device is sending rtp from a port other than where it is sending RTP from which seems to be confusing Asterisk |
| 14:00.26 | aberrios | I'd say we're a small call center and we hand 10,000 inbound a day, and that's not a huge amount, but i went with OrderlyStats since the management like all the nifty stats. |
| 14:00.46 | Dovid | when I send to the port that is sending RTP from and NOT the port that it says (example 2) then it works. if I send to what it says in the ACK and then to the source port of its packets then it does not work. here is a âgraphâ: http://pastebin.com/TxWALEgY |
| 14:01.03 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:04.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:04.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:06.14 | Freeaqingme_ | From ~thebook: "We recommend that new installations use cdr_adaptive_odbc instead [of cdr_mysql]", why is that? |
| 14:07.47 | leifmadsen | Freeaqingme: because it's better |
| 14:07.49 | leifmadsen | and more flexible |
| 14:08.03 | leifmadsen | and odbc is better supported than the native modules |
| 14:08.03 | Freeaqingme_ | kk |
| 14:08.16 | Dovid | leifmadsen: See my question ? |
| 14:08.25 | leifmadsen | Dovid: see when I connected? |
| 14:09.03 | Dovid | now i looked ;) |
| 14:09.05 | Dovid | when I send to the port that is sending RTP from and NOT the port that it says (example 2) then it works. if I send to what it says in the ACK and then to the source port of its packets then it does not work. here is a âgraphâ: http://pastebin.com/TxWALEgY |
| 14:09.33 | Dovid | seems that if the ack comes b4 i send RTP then asteisk sends to port of ack and then switches to port of remotes source. |
| 14:09.59 | Dovid | if i send rtp first and then i get ack from remote side with other port asterisk ignores it and it works. trying to see who is at fault here |
| 14:18.24 | Freeaqingme_ | leifmadsen, if I were to use cel_odbc, should I use the sql that's in cel/cel_mysql.c? |
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| 14:38.29 | pabelanger | ~asteriskreleases |
| 14:38.34 | pabelanger | ~asteriskrelease |
| 14:42.00 | leifmadsen | ~asteriskversions |
| 14:42.00 | infobot | [~asteriskversions] Always check the channel topic for updates, otherwise for the latest bundled listing of Asterisk & supporting packages by major release type ~asterisk1.2 / ~asterisk1.4 / ~asterisk1.6 |
| 14:42.09 | leifmadsen | ~asterisk1.6 |
| 14:42.09 | infobot | [~asterisk1.6] Asterisk 1.6.0-beta9 (2008/05/14), Addons 1.6.0-beta4 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.5 (2008/07/11) |
| 14:42.14 | leifmadsen | wtf.... |
| 14:42.21 | leifmadsen | wonder who added that |
| 14:42.29 | leifmadsen | ~asteriskversioning |
| 14:42.29 | infobot | asteriskversioning is, like, http://www.asterisk.org/asterisk-versions |
| 14:42.37 | leifmadsen | pabelanger: what were you looking for? |
| 14:43.19 | leifmadsen | infobot: no, asteriskversions is reply [~asteriskversions] Always check the channel topic for updates, otherwise for the latest bundled listing of Asterisk & supporting packages by major release type ~asterisk1.2 / ~asterisk1.4 / ~asterisk1.6 / ~asterisk1.8 |
| 14:43.19 | infobot | leifmadsen: okay |
| 14:43.25 | leifmadsen | ~asteriskversions |
| 14:43.25 | infobot | methinks asteriskversions is reply [~asteriskversions] Always check the channel topic for updates, otherwise for the latest bundled listing of Asterisk & supporting packages by major release type ~asterisk1.2 / ~asterisk1.4 / ~asterisk1.6 / ~asterisk1.8 |
| 14:43.30 | leifmadsen | ugh |
| 14:43.42 | leifmadsen | I always get that formatting screwed up |
| 14:45.25 | leifmadsen | infobot: forget asterisk1.2 |
| 14:45.25 | infobot | leifmadsen: i forgot asterisk1.2 |
| 14:45.26 | Jasnejac | has anyone tried using the sip session variables on 1.8.3.3? I had a problem with a Global Crossing trunk withg this yesterday - calls were cutting out after 15 mins. SIP trace showed asterisk was acting as UAC when initial negotiation told it it was UAS |
| 14:45.30 | leifmadsen | infobot: forget asterisk1.4 |
| 14:45.30 | infobot | i forgot asterisk1.4, leifmadsen |
| 14:45.32 | leifmadsen | infobot: forget asterisk1.6 |
| 14:45.32 | infobot | leifmadsen: i forgot asterisk1.6 |
| 14:45.35 | leifmadsen | infobot: forget asterisk1.8 |
| 14:45.35 | infobot | leifmadsen: i didn't have anything called 'asterisk1.8' to forget |
| 14:45.39 | leifmadsen | ~asteriskversions |
| 14:45.42 | leifmadsen | good |
| 14:56.11 | Lantizia | Hey can anyone see why I'm getting error 488 in this log? I've gone over it a dozen times... http://nopaste.me/raw/9792304984dcaa3659258f.txt |
| 14:56.39 | Lantizia | it's not codecs (first thing I've checked)... both phone and system are behind nat but the system does have nat=yes and the externip in sip_nat.conf |
| 14:57.22 | Lantizia | this is when my phone (1998) is attempting to dial out to 07654321000.... i just get "Not acceptable here" on the phone screen and in the logs as error 488 |
| 14:57.46 | Lantizia | that log has full rtp and sip debug on |
| 14:59.10 | Dovid | anyone know what the setting âconstantssrcâ is for in sip.conf ? |
| 15:05.26 | angryuser | Lantizia, pastebin detail of peer 1998 |
| 15:06.44 | Lantizia | angryuser, will do one sec (sip show peer?) |
| 15:06.59 | angryuser | Lantizia, sip.conf part |
| 15:07.11 | Lantizia | ok |
| 15:09.00 | Lantizia | angryuser, http://nopaste.me/raw/4822551714dcaa680a19aa.txt |
| 15:09.58 | angryuser | Lantizia, add allow=all, reload & call what happens ? |
| 15:10.55 | Lantizia | angryuser, same |
| 15:11.10 | angryuser | Lantizia, reloaded for sure ? |
| 15:11.20 | Lantizia | angryuser, positive |
| 15:11.46 | angryuser | Lantizia, remove lines deny=0.0.0.0/0.0.0.0 and permit |
| 15:12.05 | angryuser | Lantizia, reload & call |
| 15:12.53 | Lantizia | ok one sec |
| 15:13.24 | Lantizia | same again |
| 15:15.22 | angryuser | Lantizia, interesting |
| 15:15.33 | Lantizia | phones are fine when on site - but not remote |
| 15:16.57 | angryuser | Lantizia, strip the peer setup to minimum, secret, context username, try to call |
| 15:17.13 | Lantizia | ok one sec |
| 15:17.20 | angryuser | Lantizia, reload & re-register with the phone |
| 15:18.44 | Lantizia | angryuser, yup already on it - but i get the same |
| 15:18.49 | Lantizia | want new logs? |
| 15:19.01 | angryuser | Lantizia, if it the same no need |
| 15:20.28 | angryuser | Lantizia, http://lists.digium.com/pipermail/asterisk-bugs/2011-February/096654.html |
| 15:20.34 | angryuser | Looks familiar ? |
| 15:21.17 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:21.28 | angryuser | Lantizia, maybe something crypto related |
| 15:21.38 | angryuser | Lantizia, can you remove it for testing ? |
| 15:21.39 | *** join/#asterisk mersault (~mersault@acihip.tor4.dsl4u.ca) |
| 15:21.45 | Lantizia | will take a look - inbound calls don't work now |
| 15:21.52 | Lantizia | before they did on this extension |
| 15:22.11 | *** join/#asterisk cnu (cnu@2001:470:28:1fe::10) |
| 15:22.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:22.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:23.39 | mersault | quick question about macro and dialplan behaviour: if I have a GotoIf(condition?1000) at the end of a macro, and the context that uses the macro has an exten => h,1000,whatever entry, will a call that meets the condition in the GotoIf go from the macro to the desired priority in the calling context? |
| 15:23.47 | Lantizia | angryuser, what am i removing? |
| 15:26.29 | Lantizia | angryuser, ah I see encryption=no - will try it |
| 15:26.36 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:28.54 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:29.47 | Lantizia | angryuser, hmm even if i set encryption=no i still get crypto mentioned in the logs |
| 15:30.00 | Lantizia | i had no encryption=yes in my config though |
| 15:32.04 | Lantizia | angryuser, if I turn RTP encryption off on the snom it rings out! :P getting warm i think |
| 15:32.24 | *** join/#asterisk dprophit (dprophit@229.201.205.68.cfl.res.rr.com) |
| 15:33.00 | dprophit | I did an RPM install of 1.8 How can I get the sample conf files to create after RPM install? |
| 15:33.07 | l1nuxman | can someone help I can't get the 'dial' command to work on asterisk cmd line |
| 15:33.24 | leifmadsen | l1nuxman: what is the problem? |
| 15:33.37 | l1nuxman | leifmadsen, no such command found |
| 15:33.44 | leifmadsen | l1nuxman: what version of asterisk? |
| 15:33.48 | l1nuxman | 1.8 |
| 15:34.49 | leifmadsen | menuselect > Resource Files > res_clioriginate I think |
| 15:35.01 | l1nuxman | huh? |
| 15:35.13 | leifmadsen | oh wait, 'dial' is dependent upon a sound card, so you need chan_oss or chan_also |
| 15:35.24 | l1nuxman | I have chan_alsa loaded |
| 15:35.27 | leifmadsen | l1nuxman: you have to enable the appropriate module in menuselect |
| 15:35.35 | l1nuxman | wheres menuselect |
| 15:35.37 | l1nuxman | ? |
| 15:35.48 | leifmadsen | 'make menuselect' in the asterisk source |
| 15:36.10 | l1nuxman | I didn't compile from source |
| 15:36.15 | l1nuxman | I ran a package |
| 15:36.38 | leifmadsen | chan_alsa should do it -- make sure it's actually loaded and configured |
| 15:36.43 | leifmadsen | beyond that I'm not sure, I don't use that command |
| 15:37.00 | l1nuxman | leifmadsen, I did module show and it had chan_alsa there |
| 15:37.29 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:42.23 | dprophit | I've been getting help in #asterisk-gui no luck for resolution. http://pastebin.com/gsjwz33w is the manager.conf and GUI is stuck on "Checking write permissions for gui folder" |
| 15:43.24 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:49.54 | sled-dog | what are the top-tier voip providers? teliax and ??? |
| 15:53.33 | sled-dog | dprophit: (re)install asterisk18-configs (after you back up anything you want to keep in /etc/asterisk and any dahdi stuff, etc.) |
| 15:53.46 | sled-dog | <dprophit> I did an RPM install of 1.8 How can I get the sample conf files to create after RPM install? |
| 15:53.50 | sled-dog | for that |
| 16:01.30 | l1nuxman | no matter which module I load, chan_alsa.so or chan_oss.so I get No such command 'dial' (type 'core show help dial' for other possible commands) |
| 16:03.07 | dprophit | sled-dog you mean source compile asterisk? |
| 16:03.33 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 16:04.38 | malcolmd | 'core show application dial' will give you the help for the dial application (app_dial.so) if that returns nothing or an error, you don't have app_dial.so loaded, so do 'module load app_dial.so' |
| 16:07.08 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:16.49 | *** join/#asterisk pheex (~pheex@c-7c38e655.04-21-76737411.cust.bredbandsbolaget.se) |
| 16:20.15 | sled-dog | dprophit: I think you were asking about rpm, not source |
| 16:20.16 | leifmadsen | malcolmd: he wants the CLI command 'dial' |
| 16:20.36 | malcolmd | m'bad |
| 16:20.43 | leifmadsen | easy to confuse |
| 16:21.45 | russellb | or console dial |
| 16:24.09 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 16:24.58 | sled-dog | lke the time that i remoted in to my box at home and did a console dial... which never hung up, causing congestion to blare through my speakers at my wife. All. Day. :-) |
| 16:25.35 | russellb | nice. |
| 16:25.36 | Qwell | sled-dog: I did that with my alarm clock one day. |
| 16:26.10 | Qwell | Left for work early, forgot to stop it before I left. It went off at the normal time. My disabled mother-in-law was downstairs listening to it all day. |
| 16:26.21 | sled-dog | ow |
| 16:27.37 | *** join/#asterisk unixSnob (~unixSnob@212.117.169.230) |
| 16:28.18 | unixSnob | i'm having a very difficult time finding a SIP provider |
| 16:28.28 | unixSnob | i need one that will let me configure the outbound callerid text |
| 16:28.28 | KavanS | unixSnob, what country are you in? |
| 16:28.54 | unixSnob | KavanS: i'm in europe, but I make calls worldwide.. I don't care where the provider is |
| 16:29.52 | unixSnob | i thought a channel with so many asterisk users would have to know of a good SIP provider |
| 16:29.53 | KavanS | plenty of options |
| 16:30.19 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 16:30.24 | unixSnob | i'm not running asterisk myself.. so I suppose that narrows my options |
| 16:31.04 | russellb | ~itsplist-us |
| 16:31.05 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:31.34 | russellb | i don't know if infobot has a list for europe ... |
| 16:31.36 | russellb | ~itsplist |
| 16:31.46 | russellb | ~itsplist-eu |
| 16:31.51 | russellb | oh well |
| 16:31.58 | unixSnob | US should be fine |
| 16:32.05 | *** join/#asterisk moy (~moy@CPE002719f00364-CM00222d6b4d65.cpe.net.cable.rogers.com) |
| 16:32.11 | paulc | I've used telappliant in the UK and they're pretty good.. as well as some of the US providers on the list above ^^ |
| 16:32.28 | paulc | telappliant == voiptalk.org I think |
| 16:32.29 | unixSnob | every provider I encounter does not allow callerid configuring |
| 16:32.57 | paulc | unixSnob: Tends to be more restrictive in europe, whereas in the USA it's more like "anything goes" (despite a law to the contrary) |
| 16:33.20 | paulc | when I worked in the UK, using PRIs, you could only send caller ID that you owned (like a number from your DDI range) |
| 16:33.22 | *** join/#asterisk blackdoor67 (~admin@123.237.54.187) |
| 16:33.34 | unixSnob | paulc: well the new law in the US only prohibits fraudulent use of callerid manipulation |
| 16:34.11 | unixSnob | paulc: but from what I've seen, it looks like the law has scared some providers from offering callerid editing |
| 16:35.15 | paulc | unixSnob: True. Because there are many legitimate causes for changing caller ID. But I doubt many providers want to have that conversation with you. *sighs* ah if only common sense could prevail |
| 16:37.28 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 16:37.36 | unixSnob | russellb: that's just a list of what's popular.. but do you know if any of them specifically offer callerid editing? |
| 16:38.16 | l1nuxman | voicemail email you can specify your email@domain,secondemail@domain where the second is your pager right? But what if you want that to be a regular email address. Because it sends different data |
| 16:38.29 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 16:39.05 | gray_ | hi, does anyone know how to do a loop using the pri card. I heard of dahdi_loop but am not sure if that is what I need. |
| 16:39.47 | paulc | gray_ What kind of loop? Like dialing out on port A ends up with the arriving on port B? |
| 16:41.41 | leifmadsen | I think you need to physically run a cable between the ports |
| 16:42.08 | sled-dog | ser, openser, opensips, sipproxy, kamiwhatever.... which to choose? |
| 16:42.13 | *** join/#asterisk hfb (~hfb@pool-96-247-114-211.lsanca.dsl-w.verizon.net) |
| 16:42.16 | fish-bulb | gray_: if you mean a loopback for testing, you can do a fed different kinds of loopback in software using dahdi_maint |
| 16:42.25 | l1nuxman | does my question make sense? |
| 16:42.28 | leifmadsen | I know in the training classes jsmith had a neat little jumper for doing that so it came back in on the same port. Was just an RJ45 with a couple of pins wired on the same clip |
| 16:42.32 | fish-bulb | /s/fed/few |
| 16:42.33 | unixSnob | looks like flowroute.com allows callerid editing.. they say 'We transmit Caller-ID based on the presence of one of the following header fields in order of preference: "P-Asserted-Identity", "Remote-Party-ID" or "From:".' |
| 16:43.05 | unixSnob | how can a PAP2 user send a header? |
| 16:43.17 | fish-bulb | leifmadsen: yeah, superloopers I think is what they are called |
| 16:43.20 | leifmadsen | l1nuxman: yes it sends different data -- you'd have to create a distribution group on your server that emails multiple people to the single email address |
| 16:43.35 | leifmadsen | fish-bulb: that could be true :) |
| 16:43.38 | l1nuxman | hmmm |
| 16:43.55 | leifmadsen | l1nuxman: that is outside the realm of asterisk, so you have to deal with it on the service that is providing your email |
| 16:44.08 | gray_ | thanks guys, I will try dahdi_maint |
| 16:44.20 | l1nuxman | leifmadsen, you talking about sendmail now or my ISP? |
| 16:44.40 | gray_ | When trying to activate the spans I get red alarm and the provider says that my side is open and he cannot see anything. |
| 16:45.04 | fish-bulb | gray_: np. You won't need a physical connection for the localhost loopback, but the other two you will |
| 16:45.07 | gray_ | Provider suggested I run a loop on those ports so that he can see his own signals |
| 16:45.50 | gray_ | fish-bulb, this isn't an interface loopback, this is for PRI. I can do ifconfig eth0:1 <address> etc... but that is not what I am looking for |
| 16:45.51 | fish-bulb | gray_: ah, gotcha. I would either enable localhost loopback and run a patlooptest, or use a physical loopback plug and run patlooptest |
| 16:45.59 | gray_ | you used to be able to do this just using zttool |
| 16:46.13 | gray_ | ok |
| 16:46.17 | sezuan | unixSnob: to set the from header its usually called display name, user name, from user, etc.. |
| 16:46.20 | fish-bulb | gray_: I understand, that is what dahdi_maint allows you do. It is called localhost loopback |
| 16:46.28 | gray_ | I see |
| 16:46.30 | leifmadsen | l1nuxman: yes |
| 16:46.34 | gray_ | :) my misunderstanding |
| 16:46.35 | paulc | l1nuxman: your email setup. You can set up a local account on the box that then forwards to user1@domain.com and user2@somewhere-else.com |
| 16:46.35 | sezuan | something different than authentication user. |
| 16:46.57 | paulc | l1nuxman: then just set my-local-account@my-asterisk-box as the email destination in voicemail.conf |
| 16:47.01 | leifmadsen | l1nuxman: ^^^ what paulc said |
| 16:47.13 | fish-bulb | gray_: the zttool loopback was broken for as long as I can remember, so you should get better results with dahdi_maint =) . You can also do a network facing loop |
| 16:47.24 | fish-bulb | which is what the zttool loop was intended for, afaik |
| 16:50.00 | gray_ | fish-bulb, thank you for your help, that should do the trick. |
| 16:50.06 | fish-bulb | np |
| 16:56.06 | unixSnob | thanks sezuan. I just discovered that late versions of PAP2 firmware have a configurable "caller id header" field. |
| 16:56.47 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-63.vianet.ca) |
| 16:58.07 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 17:02.55 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 17:04.19 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59) |
| 17:04.35 | *** join/#asterisk serafie (~erin@nat/digium/x-vrxlwcioldmkkrus) |
| 17:05.27 | serafie | Is there any configuration in Asterisk that would affect the behavior or permissions of System() beyond what the operating system controls via permissions, et. al.? |
| 17:05.36 | leifmadsen | nope |
| 17:05.39 | leifmadsen | not that I'm aware of |
| 17:05.52 | leifmadsen | it's pretty much a direct extension of the shell I'm pretty sure |
| 17:06.16 | leifmadsen | can you execute the same thing from the CLI with ! as the leading char? |
| 17:06.21 | leifmadsen | !ls /tmp |
| 17:06.23 | leifmadsen | for example |
| 17:07.14 | serafie | OK. I'm seeing an issue where it appears that the user "asterisk" can create files in a certain directory on commandline but not through the System() dialplan application. |
| 17:07.32 | leifmadsen | I'd try SHELL() and see if you get anything idfferent |
| 17:07.46 | leifmadsen | also I think System() sets a channel variable that gives you some sort of result... |
| 17:07.51 | leifmadsen | might be useful? |
| 17:07.59 | serafie | only SUCCESS and FAILURE. |
| 17:08.09 | leifmadsen | ic |
| 17:08.13 | leifmadsen | and you're getting SUCCESS? |
| 17:08.31 | serafie | Ha. |
| 17:10.15 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 17:13.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:14.06 | *** join/#asterisk cerberus_za (~coert@196-210-218-151.dynamic.isadsl.co.za) |
| 17:18.03 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 17:19.57 | stope | I need to fire off an event after the channel hangs up and I've tried with the 'h' extension for when the call completes and even with the g switch in the Dial command but no dice |
| 17:20.13 | stope | It won't execute the next line in my dial plan |
| 17:20.50 | stope | I am using meetme and do have a dahdi channel running, is there another way to execute a command? I need to clear an entry in the asterisk database |
| 17:22.28 | citywok | stope: what happens if you NoOp(testing) in the h extension of the context, do you see that happen in the console? |
| 17:23.08 | stope | Yes, I'll see it but it fires too early. I'm using SLA to emulate the legacy key pbx |
| 17:23.38 | citywok | what do you mean it executes too early? it can't execute until the channel has been hung up |
| 17:23.42 | stope | when calling out, I set a flag for the line selected and ensure that if a call comes in, skip trying to dial the line in use |
| 17:24.00 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 17:25.32 | stope | I select a line, DISA provides the dial tone, number dialed, phone rings, call is anwered, h is executed, call is bridged with meet me |
| 17:26.00 | stope | I guess the 'h' gets executed as meetme takes over? |
| 17:26.11 | citywok | it's not supposed too. what version of *? |
| 17:26.29 | citywok | can you pastebin your context/dialplan? |
| 17:26.36 | stope | Asterisk 1.8.3.3 built by root |
| 17:26.43 | stope | ya, prepping it now..... |
| 17:27.59 | citywok | i just created a basic context with s,1,MeetMe() and h,1,NoOp(testing) and it didnt execute the NoOp until i hungup the call. |
| 17:30.59 | stope | here's the output: http://pastebin.ca/2056923 getting the dialplan now.... |
| 17:32.20 | Katty | boingboing |
| 17:32.22 | leifmadsen | citywok: yep that's what I'd expect |
| 17:32.40 | citywok | yea, exactly. that's not whath appens for him apparently. |
| 17:32.52 | Katty | hello my asterisk does not work at all |
| 17:32.57 | Katty | how to fix plz??? |
| 17:33.05 | citywok | can i haz astrixburger? |
| 17:33.08 | leifmadsen | Katty: klll yourself? |
| 17:33.24 | Katty | :< |
| 17:33.41 | *** join/#asterisk sereal (~jjrh@75.98.19.251) |
| 17:33.46 | leifmadsen | Katty: don't do that |
| 17:33.49 | Katty | :> |
| 17:34.08 | stope | http://pastebin.ca/2056926 |
| 17:34.14 | stope | there's my actual dial plan |
| 17:34.17 | sereal | is it possible to set a variable for a extension? |
| 17:34.56 | leifmadsen | yes |
| 17:35.33 | sereal | whats the syntax for that? |
| 17:35.43 | leifmadsen | probably exactly what you'd expect |
| 17:35.47 | citywok | sereal: depends how yoiu are doing it |
| 17:35.58 | Katty | are you doin it right?! |
| 17:36.17 | carrar | Doing it and doing it and doing it yeah! |
| 17:36.22 | Katty | :> |
| 17:36.24 | Katty | hugs carrar |
| 17:36.28 | sereal | I don't want global variables. I just want to set in the context like joe = 1234 and then just type joe,1,dial(sam/20) |
| 17:36.32 | leifmadsen | does it all night long |
| 17:36.32 | carrar | hugs kattroooo |
| 17:36.37 | carrar | err kattyrooo |
| 17:36.40 | Katty | so. i must share |
| 17:36.45 | stope | citywork: like I said, the 'h' fires soon as the call gets picked up and I'm not explicitly calling MeetMe |
| 17:36.48 | Katty | with you and leif |
| 17:37.00 | Katty | do either of you listen to 3 days grace? |
| 17:37.05 | leifmadsen | sereal: so set the channel variable, then use it |
| 17:37.09 | leifmadsen | Katty: I have yes |
| 17:37.28 | leifmadsen | exten => 1234,1,Set(joe=1234) |
| 17:37.37 | Katty | leifmadsen: you will appreciate it then ^_^ |
| 17:37.41 | leifmadsen | same => n,Goto(${joe},1) |
| 17:37.44 | carrar | let me see if i have that in my collection |
| 17:37.48 | leifmadsen | exten => ${joe},1,NoOp() |
| 17:38.16 | carrar | hrmm I don't have any of their tunes |
| 17:38.18 | serafie | leifmadsen: would that count as two priority 1 extensions for 1234? |
| 17:38.35 | carrar | they sound like metallica |
| 17:38.38 | sereal | okay... so I need to have a dummy extension to setup all my variables. |
| 17:38.40 | leifmadsen | serafie: ya that wasn't the greatest example :) |
| 17:38.56 | sereal | I thought there would be a way to just type variablename = value then use that from then on |
| 17:39.08 | leifmadsen | sereal: yes or do it via a global variable |
| 17:39.14 | leifmadsen | no that's not how the dialplan works |
| 17:39.22 | leifmadsen | you need to set variables via the Set() application |
| 17:39.57 | sereal | well I was trying to avoid having the exten = part since it seems silly if i'm setting variables because it has nothing to do with extensions. |
| 17:40.03 | citywok | Katty: was the 3 days grace question directed my way? if so, yes, but not in a long time. |
| 17:40.15 | leifmadsen | sereal: you can't avoid that part |
| 17:40.32 | Katty | citywok: i was actually talking to carrar, but that's ok |
| 17:40.34 | sereal | I understand, just seems silly to me. |
| 17:40.38 | Katty | citywok: i will share my excitement with you too! |
| 17:40.41 | citywok | stope: i just tested similar to what you are doing using ,,G and i don't get the hangup right away like you are |
| 17:40.46 | leifmadsen | sereal: I don't understand what is silly about it |
| 17:41.06 | leifmadsen | lines in the dialplan need to start with exten =>, that's all there is to it |
| 17:41.15 | leifmadsen | you can use same => as well though |
| 17:41.31 | leifmadsen | of course the first line needs to be defined though with exten => |
| 17:41.35 | stope | citywork: did you use DISA? |
| 17:42.10 | sereal | oh you can do exten = 1234, 1,bla() then exten = same, 2, bla() |
| 17:42.14 | citywok | stope: no |
| 17:42.21 | stope | let me restart * and see, maybe simple dialplan reload's aren't doing it |
| 17:42.25 | leifmadsen | sereal: yes |
| 17:42.29 | leifmadsen | no no |
| 17:42.35 | leifmadsen | exten => 1234,1,NoOp() |
| 17:42.37 | leifmadsen | same => n,NoOp() |
| 17:42.44 | sereal | ah okay. |
| 17:42.49 | sereal | thats even better |
| 17:42.50 | leifmadsen | same => n(hello),NoOp() |
| 17:42.54 | leifmadsen | same => n,Goto(hello) |
| 17:43.34 | stope | same thing after a restart :( |
| 17:43.49 | citywok | stope: i just tested it using Disa and it works properly |
| 17:43.58 | leifmadsen | sounds like a dialplan typo |
| 17:44.36 | leifmadsen | runs off to do some bug triage |
| 17:44.49 | sereal | My point about variables is that you should be able to have a context, set a bunch of variables with a straight = sign, and then be able to use them threw out that context. Set global variables the same way, and use them threw all contexts. The dialplan parser is only looking for exten and [ so this shouldn't be a problem. |
| 17:45.24 | leifmadsen | I don't see how that gives you anything different |
| 17:45.41 | leifmadsen | anyways, that's not how it works, so I'm moving on |
| 17:45.46 | citywok | leifmadsen: it lets you be lazy and less specific with what you are trying to do, making it harder to read the dialplan later on... sounds like a win-win tom e! |
| 17:45.46 | leifmadsen | let me know if you create a patch |
| 17:45.55 | leifmadsen | citywok: :0 |
| 17:46.58 | sereal | not being specific is the point. you can create a dialplan, and only change one line if you switch DIDS |
| 17:47.15 | leifmadsen | you can still do that |
| 17:47.26 | leifmadsen | you only have to specify the extension ones |
| 17:47.27 | leifmadsen | once* |
| 17:47.50 | sereal | yes, with 'same' it is a bit better. |
| 17:47.50 | citywok | Set(var=123), NoOp(${var}) |
| 17:48.20 | sereal | citywok, do you not need exten = before that though? |
| 17:48.35 | citywok | oooh, i see what you are trying to do, nevermind, ignore my comment |
| 17:48.43 | leifmadsen | I don't see the diff though |
| 17:48.45 | citywok | you just dont want to have to do exten=>13,1 exten => 13,2 |
| 17:48.49 | leifmadsen | you are using a line to specify the variable anyways |
| 17:48.59 | Katty | not that i've been paying attention |
| 17:49.00 | leifmadsen | citywok: you don't have to do that anyways |
| 17:49.13 | citywok | yea, you can use same |
| 17:49.14 | leifmadsen | exten => 1234,1,Set(variable=${EXTEN}) |
| 17:49.18 | Katty | but: exten => 13,n |
| 17:49.22 | leifmadsen | same => n,Goto(${variable},hello) |
| 17:49.28 | leifmadsen | you don't even need that! |
| 17:49.34 | leifmadsen | you only need exten => once |
| 17:49.50 | citywok | oh, wait you can use a variable for the exten => ${var} ? |
| 17:49.51 | sereal | the point though is that the exten = 1234,1 part is pointless and confusing if you are just using the 1234 extension to set a bunch of variables. |
| 17:49.52 | leifmadsen | only the first priority needs the extension defined |
| 17:50.16 | Katty | i'll define your extension in a minute. |
| 17:50.25 | leifmadsen | #exec /path/to/script-that-generates-what-I-want-on-a-single-line.sh |
| 17:51.05 | leifmadsen | sereal: you're going to have to give me a use case because I don't follow. If you're setting channel variables, you need to be executing dialplan SOMEWHERE |
| 17:53.01 | _Corey_ | sereal: I think it's confusing if you attempt to consider "exten => xxx" variable assignment |
| 17:53.45 | citywok | _Corey_: me too. I don't worry about simplifying that part of the dialplan b/c it's all autogenerated so i don't have to type it 20 times, and when something changes the generator re-creates it all from scratch. |
| 17:53.53 | sereal | okay, I do set(mike=1), set(joe=2) set(pete=3) |
| 17:53.54 | sereal | then I type |
| 17:53.54 | sereal | exten = pete,1,application() |
| 17:53.54 | sereal | which is equilivant to exten = 3,1,application except I can change pete=4 if his extension changes or for what ever reason with only editing one line |
| 17:54.19 | _Corey_ | no |
| 17:54.20 | leifmadsen | pete is not a variable |
| 17:54.28 | leifmadsen | only ${pete} is |
| 17:54.44 | leifmadsen | and what you're doing is setting global variables |
| 17:54.50 | leifmadsen | [globals] |
| 17:54.53 | leifmadsen | mike=1 |
| 17:54.56 | leifmadsen | <PROTECTED> |
| 17:55.01 | leifmadsen | exten => ${mike},1,NoOp() |
| 17:55.48 | sereal | yes - except it should also be context sensitive so you can use the variable again (such as if you have two companies on one pbx and there is two petes) |
| 17:56.06 | _Corey_ | sereal: this is why people don't use variables that way |
| 17:56.21 | leifmadsen | then use a GoSub() first |
| 17:56.42 | leifmadsen | anyways, what you're asking isn't possible |
| 17:57.13 | leifmadsen | you're asking for global variables at a context level |
| 17:57.23 | sereal | I'm asking for variable scope |
| 17:57.27 | citywok | sereal: companyone_mike |
| 17:57.31 | citywok | companytwo_mike |
| 17:57.37 | leifmadsen | yep |
| 17:57.38 | _Corey_ | it's best to try to understand best practices for dial plan design than to try to adapt it to some other kind of thing |
| 17:57.50 | citywok | if you run two companies in a pbx you should prepend something to all context, variable, peer, EVERYTHING |
| 17:58.07 | citywok | it'll be much cleaner that way, if that's what you are trying to do (i've never done it) |
| 17:58.26 | sereal | _Corey_ best answer I have heard. |
| 17:58.26 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.70) |
| 17:58.41 | leifmadsen | just use an AGI() for all your dialplan then |
| 17:59.06 | leifmadsen | then you get whatever tools your script can use |
| 18:00.28 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:01.49 | pdtpatrick | Question for you smart folks - Where do i need to look and see how many rings before going to voicemail. Is there a log I can watch as well ? |
| 18:02.00 | leifmadsen | pdtpatrick: look at the dialplan |
| 18:02.09 | leifmadsen | pdtpatrick: rings are defined based on time to the Dial() application |
| 18:02.28 | leifmadsen | call flow continues to Voicemail() when you tell it to |
| 18:03.45 | _Corey_ | pdtpatrick: ring =~ 4s |
| 18:03.46 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 18:03.59 | leifmadsen | uhhh... 6 seconds in north america |
| 18:04.01 | leifmadsen | 4 on 2 off |
| 18:04.02 | citywok | yea 6 |
| 18:04.14 | _Corey_ | d'oh |
| 18:06.31 | paulc | or 2 on and 4 off, even? ;) |
| 18:06.47 | leifmadsen | depends how you setup indications.conf :D |
| 18:06.55 | paulc | oh touche sir! ;) |
| 18:07.15 | paulc | 4 on 2 off if you want attention grabbing rings :) |
| 18:07.38 | _Corey_ | lol, i realize now that i count faster in my head than the call timer |
| 18:07.39 | paulc | (which is apparently why the UK system is Ring..Ring..Pauuuuuse... the double ring was more attention grabbing |
| 18:07.44 | _Corey_ | you guys are right... ;) |
| 18:08.10 | leifmadsen | paulc: http://ofps.oreilly.com/titles/9780596517342/asterisk-Initial.html#InitialConfig_id291715 <-- hacking indications.conf for fun and profit! |
| 18:08.50 | *** join/#asterisk MrNemus (~MrNemus@firewall.drgutah.com) |
| 18:09.18 | paulc | LOL That's awesome. And I'm totally going to try it right now - waiting for data to copy between drives makes it a slow morning so far... |
| 18:10.04 | leifmadsen | f:) |
| 18:12.20 | MrNemus | so I have a question about iax2 show netstats http://pastebin.com/cSwbgrbt |
| 18:12.31 | MrNemus | what could becausing the loss in packets |
| 18:16.10 | citywok | crappy network conditions... |
| 18:16.41 | leifmadsen | +1 |
| 18:16.47 | MrNemus | lol, its on a mpls .. |
| 18:16.51 | citywok | the same things that cause every other kind of packetloss |
| 18:17.00 | citywok | yes... what's your point? |
| 18:17.02 | MrNemus | well we only see packet loss with iax2 |
| 18:17.11 | leifmadsen | do you hear packetloss? |
| 18:17.16 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 18:17.17 | MrNemus | the sound quality is bad |
| 18:17.20 | citywok | mpls doesn't mean immune to bad network quality |
| 18:17.23 | leifmadsen | is the jitterbuffer on? |
| 18:17.26 | Eitan | what would u guys say is the absolute BEST 64bit OS to run for asterisk |
| 18:17.28 | MrNemus | yes I believe so |
| 18:17.28 | *** join/#asterisk |Physis| (~|Physis|@186.213.3.195) |
| 18:17.32 | leifmadsen | is it static or dynamic? |
| 18:17.40 | leifmadsen | Eitan: yes |
| 18:17.45 | citywok | Eitan: i'd say debian b/c i like debian :P |
| 18:17.53 | leifmadsen | Eitan: he said OS, not distribution |
| 18:17.54 | cusco | debian |
| 18:17.56 | citywok | i imagine most answers you get will be along the lines of: the one i use |
| 18:17.57 | leifmadsen | Eitan: so the answer is Linux. |
| 18:17.59 | cusco | oh |
| 18:18.04 | Eitan | lol |
| 18:18.05 | cusco | true |
| 18:18.22 | citywok | leifmadsen: hah, i just assumed he meant flavor :p |
| 18:18.25 | leifmadsen | I would not suggest Win7 64-bit |
| 18:18.30 | leifmadsen | citywok: :) |
| 18:18.38 | citywok | Win7 64bit + VMWare Player ftw |
| 18:18.51 | citywok | it works pretty well on my laptop :P |
| 18:18.52 | leifmadsen | well he obviously meant OS as in BSD vs Linux, because the actual distribution doesn't matter |
| 18:19.17 | leifmadsen | citywok: I have a computer dedicated to Win7 64-bit and I control it via Synergy from my linux laptop |
| 18:19.35 | MrNemus | so does "iax2 show netstat" show a real time report of packet loss? |
| 18:19.55 | leifmadsen | MrNemus: assuming it is accurate, I suppose so |
| 18:20.01 | citywok | leifmadsen: i can't use linux as a workstation b/c most of what i admin is windows servers |
| 18:20.14 | leifmadsen | citywok: ah I'm the opposite, so I can't use Windows as a workstation :) |
| 18:20.14 | citywok | well, i could, but my life is much easier when i simply use windows |
| 18:20.18 | leifmadsen | same |
| 18:20.29 | citywok | although i do have putty pinned to my task bar |
| 18:20.32 | leifmadsen | we're in the same ocean on different boats |
| 18:20.45 | citywok | and expandrive makes life easier for dealing with files |
| 18:21.03 | leifmadsen | I can't admin all my asterisk servers that easily from windows because putty annoys me when I need multiple tabs and such :) |
| 18:21.16 | leifmadsen | expandrive is new... |
| 18:21.18 | leifmadsen | googles |
| 18:21.23 | citywok | it used to be named sftpdrive |
| 18:21.30 | Eitan | putty is my favorite :) |
| 18:21.34 | citywok | i have 3 monitors, so i prefer putty to not be tabbed |
| 18:21.38 | file | leifmadsen, putty connection manager |
| 18:21.42 | leifmadsen | file: nice |
| 18:21.50 | citywok | default putty window size i can get 12 putty windows up with no overlap |
| 18:21.55 | _Corey_ | I'm a big fan of GridMove |
| 18:22.02 | _Corey_ | I use that to arrange my putty windows |
| 18:22.04 | Eitan | citywok : yup |
| 18:22.07 | MrNemus | how often are the averages taken in iax2 netstat? why is the kpkts zero across the board ? and some are over 100% doesn't make sense. |
| 18:22.10 | Eitan | multiple monitors works great |
| 18:22.35 | *** join/#asterisk Besticles (~larry@209-58-227-178.static-ip.telepacific.net) |
| 18:22.38 | leifmadsen | I have 3 monitors now too, the main one in the middle runs Linux and tabs in xterm ftw |
| 18:22.43 | citywok | _Corey_: have you used windows 7's snap to feature? |
| 18:22.51 | leifmadsen | MOAR ASTERISK |
| 18:22.54 | _Corey_ | Still on XP |
| 18:22.57 | Eitan | i run 3 as well |
| 18:23.06 | Eitan | ptty on the exteriors and windows in the center |
| 18:23.07 | citywok | ah, windows 7 does a pretty good job of snapping windows. Windows key -> arrow |
| 18:23.08 | Eitan | makes me happy |
| 18:23.31 | citywok | unfortunately i have 3 24's, i would prefer 3 22's now that they come 1920x1080 |
| 18:23.35 | Besticles | Is there a way to detect a beep after AMD() has finished? |
| 18:23.46 | citywok | too much turning my head gets annoying |
| 18:23.58 | _Corey_ | citywok: I like this one because I have an odd arrangement with 4 monitors... a central 22" ws and 2 19"ws rotated 90 degrees on either side |
| 18:24.01 | carrar | People still use MSFT? |
| 18:24.07 | leifmadsen | citywok: ya I have 17's flanking a 26 |
| 18:24.25 | _Corey_ | I have a hotkey to reposition/size the window to defined zones |
| 18:24.29 | citywok | doesn't the different resolutions drive you guys crazy? lol |
| 18:24.39 | citywok | s/doesn't/don't/ |
| 18:24.46 | Eitan | at one point i had a 24 up top center. and 3 19s down below |
| 18:24.49 | Eitan | it was pretty rad |
| 18:25.00 | _Corey_ | If you buy from the same manufacturer with the native resolution it usually looks normal |
| 18:25.11 | citywok | 1920x1080 (or 1200 is even better) or bust |
| 18:25.18 | citywok | even my 15" laptop is 1920x1200 |
| 18:25.28 | leifmadsen | I would love a panel of 4 monitors without borders |
| 18:25.32 | leifmadsen | 1 2 |
| 18:25.35 | leifmadsen | 3 4 |
| 18:25.37 | citywok | yea that would be baller |
| 18:25.41 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 18:25.43 | leifmadsen | but lined up correct..... :) |
| 18:25.57 | _Corey_ | I bought one of these: http://www.cotytech.com/content-product_info/product_id-2026/triple_monitor_desk_mount_spring_arm_quick_release_mount.html |
| 18:25.58 | citywok | on my desk at home i got a dual monitor mount and a pair of new 23" LED's, sooo nice |
| 18:26.00 | leifmadsen | kpfleming has a pair of 26's I believe beside each other in portrait mode |
| 18:26.09 | *** join/#asterisk jkroon (~jkroon@dsl-241-248-179.telkomadsl.co.za) |
| 18:26.16 | leifmadsen | citywok: you should just setup your monitors in portrait mode to save on the head turning |
| 18:26.19 | citywok | this guy: http://www.monoprice.com/products/product.asp?c_id=109&cp_id=10828&cs_id=1082808&p_id=5560&seq=1&format=2 |
| 18:26.22 | leifmadsen | citywok: I think that's your answer :) |
| 18:26.41 | citywok | 1080x1920 would be annoying, i like the width for coding |
| 18:26.51 | _Corey_ | That would have been cheaper, but this thing is slick |
| 18:26.54 | leifmadsen | portrait would be ideal for writing |
| 18:26.58 | citywok | eclipse :hearts: landscape |
| 18:26.59 | carrar | 2560x1600 x 2 |
| 18:27.00 | _Corey_ | I can swing a monitor around to show people stuff |
| 18:27.10 | leifmadsen | _Corey_: I don't have people in my office :) |
| 18:27.17 | sereal | that's pretty bad ass _Corey_ |
| 18:27.27 | citywok | _Corey_: yea i want one of those, but my mount was only $40 haha. |
| 18:27.41 | sereal | Little expensive though. |
| 18:27.51 | _Corey_ | I have the whole thing running via my laptop, so picture 3 monitors above the laptop |
| 18:28.02 | citywok | _Corey_: what laptop can drive 3 monitors? |
| 18:28.11 | sereal | one with dvi, vga and hdmi |
| 18:28.21 | _Corey_ | nope |
| 18:28.30 | sereal | like a thinkpad and docking bay |
| 18:28.31 | carrar | without smoking? |
| 18:28.31 | citywok | sereal: it likely wouldn't have a dac for each one |
| 18:28.41 | sereal | citywok, they usually do |
| 18:28.55 | _Corey_ | I use these: http://plugable.com/products/UGA-2K-A/ |
| 18:29.05 | _Corey_ | you can have up to 6 via usb 2.0 |
| 18:29.09 | sereal | ewww |
| 18:29.17 | citywok | ah okay, that's kind of what i figured. how's the performance? does it lag on youtube video's? lol |
| 18:29.33 | _Corey_ | not on youtube but i wouldn't do gaming with them |
| 18:29.45 | carrar | what about youporn? |
| 18:29.49 | citywok | lol |
| 18:29.49 | _Corey_ | lol |
| 18:29.56 | sereal | not that you need to game on 3 displays |
| 18:30.10 | _Corey_ | matrox markets something similar that's all vga |
| 18:30.22 | _Corey_ | designed for flight sim and other stuff |
| 18:30.31 | sereal | personally 3 wide screens seems like a bit too much, 1 wide screen and two 4:3 is probably better |
| 18:30.40 | leifmadsen | I'd like 3 monitors for Forza 2 :) |
| 18:30.44 | citywok | 3 wide screens takes up my entire desk, but i love it |
| 18:30.55 | citywok | that's why i wish they were 22's |
| 18:31.18 | leifmadsen | you should have 4 like this: | = | |
| 18:31.26 | _Corey_ | rotating the two side monitors keeps mine to a reasonable width |
| 18:31.38 | sereal | yeah I would probably do that Corey |
| 18:31.49 | citywok | _Corey_: does it hurt your neck looking up/down a lot? |
| 18:32.12 | citywok | i discovered that after sitting on an exercise ball for a while i didn't slouch much anymore, so i had to look down at my monitors all the time which started to hurt my neck quite a bit. |
| 18:32.13 | sereal | thing is I position my self very directly infront of my screen so the screens are eye level and I don't need to turn my head |
| 18:32.15 | leifmadsen | I look down too much -- I wish my monitors sat about 8 inches off the desk |
| 18:32.24 | sereal | I think I would rather have two and be able to rotate them from wide to portrate |
| 18:32.30 | citywok | leifmadsen: i grabbed 3 reams of paper... one for each monitor |
| 18:32.35 | _Corey_ | leifmadsen: that's the beauty of the bracket |
| 18:32.48 | sereal | leifmadsen: phone books brah |
| 18:32.56 | leifmadsen | so ugly ;) |
| 18:32.59 | citywok | heh, that would work too if i had any. |
| 18:33.05 | citywok | but my reams of paper aren't a whole lot prettier |
| 18:33.18 | leifmadsen | I just need to build a custom desk with a ledge |
| 18:33.21 | sereal | you can drape a sheet around it so it just looks like a colorful stand |
| 18:33.32 | carrar | martha style |
| 18:33.33 | citywok | leifmadsen: just get mounts :P |
| 18:33.33 | leifmadsen | sereal: now that's not a terrible idea.... |
| 18:33.39 | leifmadsen | citywok: no space :) |
| 18:33.40 | leifmadsen | haha |
| 18:33.47 | leifmadsen | BACK TO WORK SLACKERS! |
| 18:33.48 | citywok | how can you not have space for a mount? |
| 18:33.55 | sereal | but leifmadsen didn't you say no one comes in your office? |
| 18:33.59 | citywok | it made my desk at home so much bigger |
| 18:34.05 | carrar | haha |
| 18:34.06 | sereal | work? i'm on a train |
| 18:34.42 | paulc | IRC on a train - that's cool :-) |
| 18:34.51 | sereal | train has wifi |
| 18:35.11 | MrNemus | so no ideas on how iax2 show netstats works? |
| 18:35.17 | citywok | the new verizon 4G mifi puck is crazy nice |
| 18:35.26 | citywok | mine does 14/3 @ 75ms |
| 18:35.35 | sereal | nice. |
| 18:35.44 | _Corey_ | http://twitpic.com/4wcchm |
| 18:35.45 | sereal | get unlimited data? |
| 18:35.48 | citywok | my boss got 16/5 on his, but he was in the middle of nowhere |
| 18:36.04 | citywok | _Corey_: nice |
| 18:36.24 | citywok | is that eclipse in the bottom left, and the FoP on the top left? |
| 18:36.32 | _Corey_ | no, it's this chatroom |
| 18:36.33 | citywok | and skype bottom right? outlook top right? |
| 18:36.44 | _Corey_ | google cal on the top right |
| 18:36.57 | carrar | You have sunshine? |
| 18:36.59 | carrar | wtf |
| 18:37.06 | _Corey_ | lol |
| 18:37.07 | citywok | yea it's raining here... |
| 18:37.11 | citywok | f****** seattle |
| 18:37.25 | carrar | also in Bellevue |
| 18:37.49 | citywok | ah, yea i'm actually in redmond next to proclub |
| 18:38.12 | citywok | although my office doesn't have a window so i never have to see the rain :P |
| 18:38.13 | paulc | raining in Vancouver too.. www.katkam.ca - shite weather more often than not, remind me why I'm here again |
| 18:38.14 | _Corey_ | i'm visiting seattle in two weeks |
| 18:38.26 | paulc | _Corey_ Your screen setup is awesome |
| 18:38.33 | sereal | it's nice and sunny in ottawa |
| 18:38.35 | citywok | yea, i'm jealous of his setup :P |
| 18:39.10 | _Corey_ | i'm happy to have an appreciative audience... usually i'm forcing people to look at that picture on my iphone trying to explain how cool it is |
| 18:39.45 | citywok | haha, yea. how much does each monitor weigh? |
| 18:40.02 | citywok | my old dell 24 weight like 10 lbs, my new LED's weigh more like 4 or 5, super light. |
| 18:40.15 | citywok | s/weight/weighed/ |
| 18:40.26 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 18:40.27 | _Corey_ | hmm, dunno... they're just dell lcd's |
| 18:40.45 | |Physis| | I'm having trouble using the voicemail recording the data in the database postgresql using odbc, selecting MENUSELECT_OPTS_app_voicemail = ODBC_STORAGE. Displays the following error when I leave a message on voicemail: app_voicemail.c: 3661 insert_data_cb: Direct SQL Execute failed! ??? |
| 18:41.24 | sereal | well i'm here |
| 18:41.25 | sereal | peace |
| 18:41.55 | leifmadsen | |Physis|: did you create all the appropriate things to save the blob to the pgsql per the documentation? |
| 18:41.58 | leifmadsen | with pgsql it's not trivial |
| 18:42.06 | leifmadsen | because there is no "blob" element |
| 18:42.06 | carrar | mmm psql |
| 18:42.46 | leifmadsen | https://wiki.asterisk.org/wiki/display/AST/Storing+Voicemail+in+PostgreSQL+via+ODBC |
| 18:43.41 | |Physis| | leifmadsen, I put the column recording as BYTEA |
| 18:43.53 | leifmadsen | I can't help you further |
| 18:43.59 | |Physis| | leifmadsen, in this case would be the mistake? |
| 19:12.33 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:14.00 | |Physis| | [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)] |
| 19:15.18 | |Physis| | not enter data in voicemessages and writes the id of the recording \ lo_list |
| 19:15.28 | leifmadsen | did you follow the instructions per the wiki? |
| 19:15.48 | leifmadsen | I am not familiar with BYTEA -- can it actually store and retrieve the information correctly? |
| 19:16.30 | |Physis| | I did all the steps indicated by jared smith in wiki.asterisk.org |
| 19:17.05 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 19:17.07 | |Physis| | \lo_export 16599 /tmp/odcb-17652.gsm and play /tmp/odcb-17652.gsm |
| 19:17.39 | |Physis| | does not insert any data in the table voicemessages! |
| 19:17.42 | |Physis| | :( |
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| 19:30.33 | *** join/#asterisk shier_h2 (47f65014@gateway/web/freenode/ip.71.246.80.20) |
| 19:30.49 | shier_h2 | hi, is anyone having trouble with google voice outbound calling right now? |
| 19:31.53 | kuj | Google working fine here, just now |
| 19:32.05 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 19:33.00 | shier_h2 | right now it's giving me busy signals or telling me "your call can't be completed at this time" or just dropping the call.... ever experienced something like that? |
| 19:33.20 | eXcAliBuR | I want my asterisk box to call a bunch of people with a pre-recorded message... what should I be searching for? |
| 19:33.54 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 19:34.52 | eXcAliBuR | i tried searching with terms like phone chain, call list |
| 19:34.56 | eXcAliBuR | but not getting anywhere |
| 19:34.56 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 19:35.45 | shier_h2 | perhaps something like this eXcAliBuR http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message |
| 19:36.16 | eXcAliBuR | auto dial :} |
| 19:36.23 | eXcAliBuR | thank you |
| 19:36.27 | shier_h2 | no problem |
| 19:36.45 | *** join/#asterisk m_tadeu (~quassel@a89-152-50-173.cpe.netcabo.pt) |
| 19:36.55 | pdtpatrick | Question -- when i run "voicemail show users" there's an extension missing .. however that number exists in voicemail.conf and is setup properly |
| 19:37.16 | pdtpatrick | how can i tell asterisk read voicemail.conf and add this VM so it shows when i do voicemail show users ? |
| 19:37.28 | kuj | shier_h2: they haven't been the most stable for me either, but usually come back to life within an hour or so. sorry, don't have more info. But if it used to work and you didn't change your config, I would not start tinkering with it. |
| 19:38.44 | shier_h2 | kuj: thanks.... it's too bad it isn't more reliable... I guess you get what you pay for :) |
| 19:39.05 | kuj | yep. I didn;t pay much either :) |
| 19:39.38 | shier_h2 | any thoughts on how to have asterisk detect when google voice is down and have it switch over to another SIP trunking provider? |
| 19:39.41 | Wiretap | pdtpatrick, voicemail reload |
| 19:40.04 | leifmadsen | shier_h2: switch what over? |
| 19:40.24 | leifmadsen | shier_h2: all you can control is your outbound calls, and you can detect that with ${DIALSTATUS} to try dialing out another trunk |
| 19:41.15 | kuj | I'm checking dial status, and if it looks like a provider issue, I'll then fall through to dial the next cheapest for the dialed destination |
| 19:41.28 | shier_h2 | oh cool.... do you ahve an example of how that might look in the dial plan? |
| 19:42.21 | leifmadsen | Just Dial() |
| 19:42.32 | leifmadsen | check ${DIALSTATUS} value, then Dial() again |
| 19:43.45 | leifmadsen | just like you would for Voicemail() |
| 19:43.46 | leifmadsen | http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Deeper.html |
| 19:43.51 | leifmadsen | search for DIALSTATUS |
| 19:44.52 | kuj | leifmadsen: thanks for the great book! |
| 19:44.53 | shier_h2 | awesome, thanks |
| 19:45.53 | shier_h2 | is it possible to spoof outbound caller ID? Say I have multiple outbound trunks with different providers... can I set what the caller ID should look like? |
| 19:46.18 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 19:46.23 | leifmadsen | shier_h2: please don't call it spoofing, but yes, you can set the caller ID to anything you want -- it's up to your provider to allow you to do it though |
| 19:47.04 | shier_h2 | oh I see... yea spoofing is the wrong term... do you know if google voice will let you change it to whatever your main office number is... or do they lock it down |
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| 20:03.21 | pdtpatrick | Question - where in asterisk can you find the number of times the number will ring before going to VM ? |
| 20:06.11 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-153.mobile.uci.edu) |
| 20:08.03 | shier_h2 | I think it measures it in seconds... Dial(<device to dial>,<secs to ring>) |
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| 20:09.49 | pdtpatrick | Thanks. is there a config that would contain such information? manager.conf ? |
| 20:12.47 | leifmadsen | pdtpatrick: as alluded to earlier, just extensions.conf |
| 20:12.50 | leifmadsen | you control that, not asterisk |
| 20:12.58 | leifmadsen | however you program it, asterisk will do it |
| 20:14.14 | cusco | probably peer has interface SIP/X instead of Local/X |
| 20:15.33 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 20:15.34 | leifmadsen | o.O |
| 20:15.37 | leifmadsen | kfife: ohai |
| 20:15.45 | kfife | hey there! |
| 20:16.14 | kfife | Quick question: Is there a way to make automon give some sort of confirmation tone that it's recording? |
| 20:16.47 | kfife | either ongoing, or a one-time to let the party know that their DTMF's have registered? |
| 20:17.11 | leifmadsen | kfife: yes and no :) |
| 20:17.16 | leifmadsen | yes... but not trivially |
| 20:17.26 | kfife | What's it look like? |
| 20:17.38 | kfife | Do I have to define an application? |
| 20:17.49 | kfife | ...in features.conf? |
| 20:18.26 | leifmadsen | kfife: you basically have to trigger something to know that the channel is being recorded, monitor for that, and inject audio into the existing channel... |
| 20:18.46 | leifmadsen | kfife: you could start here: http://ofps.oreilly.com/titles/9781449303822/c03-AudioManipulation_id302347.html#TriggeringAudioViaDTMF |
| 20:18.49 | kuj | putty |
| 20:18.53 | leifmadsen | I had to do something like this for a client |
| 20:18.55 | kfife | Thanks. Lemmie take a look |
| 20:19.01 | kuj | oops, sorry. |
| 20:19.15 | kfife | Seems like it would be a nice addition to automon. |
| 20:20.00 | leifmadsen | agreed |
| 20:20.02 | kfife | Cool. Now I can make my own soundboard. |
| 20:20.04 | leifmadsen | I would welcome the patch :) |
| 20:20.32 | kfife | *8 = "Don't have a cow man!" |
| 20:20.39 | kfife | *9 = "Doh!" |
| 20:20.43 | leifmadsen | indeed :D |
| 20:20.52 | leifmadsen | ohai VUC :) |
| 20:21.13 | kfife | *7 = "Hey Karl! Put your clothes back on. Let's not have another incident" |
| 20:22.16 | leifmadsen | *7 is fun |
| 20:22.29 | _Corey_ | kfife: Be careful... I had a 'tt-monkeys' that I had forgotten about until someone discovered it during a conf. call :) |
| 20:22.45 | kfife | _Corey_ whoops! |
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| 20:26.08 | *** join/#asterisk dinesh___ (~dinesh@46-126-192-144.dynamic.hispeed.ch) |
| 20:27.13 | dinesh___ | Good evening everyone - I would like to install Asterisk on a Windows Server 2008 R2 (I had one previously running on Linux Gentoo) |
| 20:27.42 | dinesh___ | Does it run as a Windows Service ? I have found a project "asteriskwin32" |
| 20:28.22 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:28.32 | dinesh___ | but doesn't asterisk itself directly support windows ? |
| 20:30.48 | dinesh___ | "AsteriskWin32 0.66b build from Asterisk 1.2.26.2" |
| 20:30.53 | dinesh___ | sounds to me like a dead project |
| 20:32.54 | dinesh___ | uf it looks like i am going to have to migrate my sip.conf and extension.conf to Microsoft's Unified Communications :( |
| 20:33.41 | dinesh___ | the french Wikipedia article about Asterisk claims that it runs on Windows, but the english one mentions only AsteriskWin32 for Windows |
| 20:33.44 | WiretapWork | dinesh___, because something is preventing you from running a linux box? |
| 20:34.17 | paulc | dinesh__ Just run Asterisk on a Linux box. It works, it's less stress, it will be supported by your peers here, and everyone will be happier. |
| 20:34.59 | Nugget | I hate Linux just as much as the next guy, and even I run asterisk on Linux. |
| 20:35.12 | Nugget | anything else is just a bucket of pain and regret |
| 20:35.13 | WiretapWork | I can't see any real reason to run asterisk on windows when you can run it on something as simple as a wireless router or an old thin client (like I do) |
| 20:35.37 | dinesh___ | well my gentoo was totally broken, i didn't update it for years (since 2006). it became impossible to fix. windows on the other hand as windows update |
| 20:35.49 | dinesh___ | but yep everything was working perfectly fine for 4-5 years |
| 20:35.54 | Nugget | Yes, but Windows DOESN'T have Asterisk. |
| 20:36.01 | Nugget | which is more important to you? |
| 20:36.14 | dinesh___ | i'll have to think more then, i could create another linux VM perhaps |
| 20:36.18 | seraphie | Lots of Linux distros have automatic updates. |
| 20:36.23 | WiretapWork | dinesh___, apt-get dselect-upgrade |
| 20:36.23 | dinesh___ | perhaps something easier like ubuntu |
| 20:36.26 | WiretapWork | debian is for you :P |
| 20:36.38 | WiretapWork | ubuntu is an african word meaning "cant configure debian" |
| 20:36.47 | WiretapWork | you're a gentoo user, so that is in itself unlikely ;P |
| 20:37.00 | WiretapWork | I migrated from Gentoo to Debian a few years ago, never looked back |
| 20:37.12 | Nugget | well, to be fair, he's a *failed* gentoo user. :) |
| 20:37.40 | WiretapWork | Nugget, gentoo breaks if you do too many updates, thats what I didn't like about it |
| 20:38.06 | _Corey_ | just download AsteriskNow and pop in the CD |
| 20:38.32 | WiretapWork | that also works |
| 20:38.43 | _Corey_ | you need not be a Linux guy to get it going |
| 20:40.07 | dinesh___ | yep that is also a good option |
| 20:40.32 | dinesh___ | after all i would just have to maintain my sip.conf and extension.conf files, which shouldn't be hard |
| 20:41.04 | dinesh___ | i'll give that a try, thanks |
| 20:42.02 | paulc | May the force be with you :) |
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| 21:46.57 | WiretapWork | ~thebook |
| 21:46.57 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
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| 23:30.19 | weinerk | ? |
| 23:30.23 | carrar | ?? |
| 23:30.34 | kfife | Question: In features.conf, Dynamic_features, how does one pass comma separated values to an application? Can't use pipes. |
| 23:30.35 | WIMPy | ¿ |
| 23:31.09 | weinerk | Hi. Please help. |
| 23:31.09 | weinerk | I need to implement a mechanism of call confirmation from php AGI - |
| 23:31.10 | weinerk | incoming call on music hold - meanwhile try a sequence of a few destinations - |
| 23:31.10 | weinerk | if someone answers - announce to him the call and wait for "1" for acceptance - |
| 23:31.10 | weinerk | otherwise try next on list. |
| 23:31.10 | weinerk | If no answer - return to incoming call - call back later. |
| 23:31.10 | weinerk | I am not allowed to use freepbx ringgroups and the like. |
| 23:31.11 | weinerk | I tried this - problematic: |
| 23:31.11 | weinerk | agi->Dial("Local/14083334444@from-internal|20|M(confirmcustom^custom/Prompt^^)m(default)t"); problem is that incoming call ANSWERED as soon as dest. picks up the phone - |
| 23:31.12 | weinerk | so even if call not accepted - after timeout/hangup - |
| 23:31.12 | weinerk | it hangs up the incoming call also. |
| 23:31.16 | *** join/#asterisk nny (~Scott_2@174.107.201.103) |
| 23:31.23 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:32.08 | nny | hi quick polycomm question. Noticed that some phones on this network I am helping with won't reregister after a network drop. Now considering the network shouldn't drop, what's the 3.3.1 config section name called that deals with telling the phone to register after x timeout? |
| 23:32.19 | nny | i see: voIpProt.server.1.retryTimeOut="10" |
| 23:32.28 | nny | and reg.1.server.1.expires="30" in examples |
| 23:32.35 | nny | but wondering if it changed in 3.3.1 |
| 23:35.18 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 23:35.46 | Freeaqingme | nny, why not just contact polycom? |
| 23:36.46 | weinerk | Could someone point me to where I could get expert advice for pay? Thanks! |
| 23:38.53 | Freeaqingme | weinerk, uhm, I suppose 50 % of the users in here provide commercial services, otherwise you could also contact digium |
| 23:39.30 | Freeaqingme | weinerk, re your dial command, care elaborating where you found that? I kinda miss good docs on the dial app |
| 23:40.22 | Freeaqingme | weinerk, and to answer your question, I think you should be able to setup a separate call, and once the call is accepted you could bind the channels together through ami (I think) |
| 23:43.23 | nny | Freeaqingme: i can, i know this channel has a lot of polycomm users in it, seemed like an appropriate question |
| 23:44.34 | WiretapWork | I haven't been able to google up an answer to this yet, but with many commercial PBX systems, the receptionist can see on their attendant console that a line is ringing, and hit the button to answer the call on behalf of that person. is there a way to do this in asterisk? |
| 23:44.50 | Freeaqingme | nny, yeah, it definitely is appropriate (I think). But it's a commercial product with commercial support, so (given that at this time of day it usually is relatively quiet here), I'd just enjoy the support they provide if I were you ;) < nny |
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| 23:45.28 | WIMPy | WiretapWork: subscriptions and hints |
| 23:45.30 | nny | weinerk: look at Bridge |
| 23:45.59 | WiretapWork | WIMPy, I have that & BLF working, but does that actually allow you to answer-on-behalf by hitting the key as it flashes? |
| 23:46.07 | nny | http://www.voip-info.org/wiki/view/Asterisk+cmd+Bridge |
| 23:46.25 | WIMPy | WiretapWork: That depends on your phone. |
| 23:47.01 | WiretapWork | WIMPy, but if the phone is subscribed to the line in *, there's no special config I have to do for * to make it work, so long as the phone can do it it jus will? |
| 23:47.30 | WIMPy | It has to be in the phone, yes. |
| 23:47.57 | WiretapWork | well the phones are Cisco 79xx |
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| 23:56.34 | weinerk | Freeaqingme: |
| 23:56.34 | weinerk | 1) thanks for feedback |
| 23:56.34 | weinerk | 2) re: dial options - I found here and also by debuggin logs |
| 23:56.34 | weinerk | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 23:56.34 | weinerk | 3) re: separate call - how can I do that? |
| 23:56.35 | weinerk | 4) re: ami - I found this: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 23:56.35 | weinerk | but can you point me more directly where to look? |
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| 23:57.51 | weinerk | I kinda have a half baked solution that I think works ... |
| 23:58.19 | weinerk | actually something similar to what you are saying... |
| 23:58.41 | weinerk | but it is rather ugly and in any event I would like to run it by someone who knows what he is doing :-) unlike me |
| 23:58.49 | Freeaqingme | weinerk, your point #4 was an addition to #3. You should be able to use the ami originate command to initiate a new channel I think |
| 23:59.14 | weinerk | I will read up in a sec. |