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| 00:09.44 | mutilator | any (free) sip phone for android recommended? |
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| 00:25.42 | Penguin | mutilator: I like csipsimple. |
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| 01:27.28 | oli-werk | hey there, anyone around to give me a hand with some agent wrapuptime status? |
| 01:28.17 | [TK]D-Fender | ~ask |
| 01:28.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:28.29 | oli-werk | okay |
| 01:29.30 | oli-werk | how can we make wrapuptime set an agent status to paused, and unpause on completion? |
| 01:30.02 | oli-werk | what happens now is that 'agent show' and 'queue show' don't reflect that an agent is on wrapup |
| 01:30.03 | [TK]D-Fender | You don't |
| 01:30.46 | [TK]D-Fender | This would take a feature request for it to be added |
| 01:30.54 | oli-werk | the autodailer we have doesn't respect that an agent is in wrapup time |
| 01:30.58 | [TK]D-Fender | This is not something that exists as an option now. |
| 01:31.20 | oli-werk | do you have any suggestions for what we could do? |
| 01:31.27 | [TK]D-Fender | Perhaps you should deal with it in your dialer in the meantime. |
| 01:31.54 | oli-werk | thanks for your help |
| 01:32.13 | [TK]D-Fender | Or you coul use local channels for all of your member dialouts and include pause/unpause logic on your own. |
| 01:32.36 | [TK]D-Fender | #patienceforplanbhuhIguessnot |
| 01:36.47 | *** join/#asterisk infobot_ (ibot@rikers.org) |
| 01:36.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 02:01.40 | oli-werk | is there way to query asterisk for an agent status that will show if they are in wrapup? |
| 02:11.00 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 02:14.29 | Katty | beep boop. |
| 02:14.31 | [TK]D-Fender | <oli-werk> do you have any suggestions for what we could do? |
| 02:14.31 | [TK]D-Fender | <[TK]D-Fender> Perhaps you should deal with it in your dialer in the meantime. |
| 02:14.31 | Katty | whrrrrr. |
| 02:14.45 | [TK]D-Fender | For how to find out, perhaps this will give a clue : https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Agents |
| 02:14.55 | oli-werk | the dialer is written in php and spanish, so not really an option at the moment |
| 02:15.11 | Katty | hello there, fenderbender! |
| 02:15.38 | Katty | that was some fine looking sushi you posted awhile back. |
| 02:15.46 | [TK]D-Fender | What kind of "dialer" is this? |
| 02:17.19 | oli-werk | elastix 2.5 dialerd |
| 02:17.53 | Katty | eyes file |
| 02:17.57 | Katty | file: i have scissors. and a buzzor. |
| 02:18.01 | Katty | buzzer. |
| 02:18.01 | oli-werk | the problem we have is that it isn't aware of agents in wrapuptime, so it will dial external numbers |
| 02:18.08 | oli-werk | and because no agent is available due to wrapuptime |
| 02:18.15 | oli-werk | the callee is put onto hold music |
| 02:19.58 | oli-werk | yet, if an agent status is "paused" it won't dial |
| 02:20.11 | oli-werk | so without being able to modify the dailer, my options are: |
| 02:20.33 | oli-werk | 1) a script to check when an agent ends a call, manually pause them and unpause after a sleep period |
| 02:20.47 | oli-werk | 2) implement this in a dialplan somehow |
| 02:21.00 | oli-werk | or... some other solution i don't know of yet |
| 02:24.01 | [TK]D-Fender | Is that actually even using *'s Queue system? |
| 02:24.53 | oli-werk | *'s? |
| 02:24.56 | oli-werk | oh |
| 02:24.56 | oli-werk | yes |
| 02:24.56 | [TK]D-Fender | Asterisk |
| 02:25.08 | [TK]D-Fender | How locked down is the dialplan around this? |
| 02:25.29 | oli-werk | i don't know to be honest |
| 02:25.37 | oli-werk | willing to try anything |
| 02:26.15 | [TK]D-Fender | Go look at how that's made. |
| 02:26.43 | [TK]D-Fender | Maybe there is something in there you can modify to kill time manually instead of wrap-up |
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| 03:21.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 03:21.58 | ruben23 | hi guys any help i have tried calling an inbound number - but it just ring and wont ring my extension at all - which is on ring group and ring all policy.. |
| 03:22.01 | ruben23 | any idea guys..? |
| 03:23.57 | [TK]D-Fender | Asterisk has no such thing as a "policy" |
| 03:24.10 | [TK]D-Fender | or "ring-all" as a thing at all. |
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| 03:40.49 | ruben23 | <PROTECTED> |
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| 03:46.12 | [TK]D-Fender | If you say so... |
| 03:50.02 | *** join/#asterisk infobot_ (ibot@rikers.org) |
| 03:50.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 04:14.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:10.14 | alexw | Could anyone help with SIP originate |
| 06:10.31 | AnonGirl | . |
| 06:10.34 | alexw | . |
| 06:10.46 | alexw | ?paste |
| 06:10.49 | alexw | !paste |
| 06:11.24 | alexw | http://pastebin.com/xxmvhgju |
| 06:11.30 | alexw | I'm trying to call Ext 10 - and when Ext 10 picks up |
| 06:11.35 | alexw | It will dial 61422333444 |
| 06:12.14 | alexw | But what is happening is when I pick up it calls Ext 10 :S |
| 06:14.59 | [TK]D-Fender | it will not dial 61422333444 |
| 06:15.10 | [TK]D-Fender | Because you did not tell it to anywhere in there |
| 06:15.29 | alexw | Ooooh |
| 06:15.30 | alexw | I get it now |
| 06:15.39 | alexw | Okay it's working now :) |
| 06:16.03 | alexw | But it only calls once the outside number picks up |
| 06:16.07 | alexw | How do I get it to call through to the ext first - and then call the outside number? |
| 06:16.13 | alexw | I have EarlyMedia set to tru |
| 06:16.16 | alexw | true( |
| 06:16.23 | alexw | But this only calls through once it beings to ring |
| 06:17.00 | [TK]D-Fender | When the Channel: gets answered it gets dumped into the dilaplan where you tell it to go. |
| 06:17.04 | [TK]D-Fender | That is what this does. |
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| 06:19.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 06:20.09 | Penguin | Reverse your parameters. |
| 06:20.27 | alexw | iSymfony somehow does this |
| 06:20.32 | alexw | When I double click to call |
| 06:20.41 | alexw | Oooh |
| 06:20.42 | alexw | get it now |
| 06:20.44 | alexw | :) |
| 06:20.57 | alexw | so Exten should be the outgoing |
| 06:21.13 | Penguin | If you want that to be the behavior. |
| 06:21.25 | alexw | But now I get Extension does not exist. |
| 06:21.42 | alexw | 'Exten'=>'SIP/b612xxxxxxxx/61421xxxxxx', |
| 06:21.59 | Penguin | That isn't an extension. |
| 06:22.02 | Penguin | That's a channel. |
| 06:22.40 | Penguin | So you can use application Dial rather than Exten, or you can create dial plan that dials that channel. |
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| 06:26.02 | alexw | Still getting extension not found |
| 06:26.02 | alexw | http://pastebin.com/EQL7427Z |
| 06:26.37 | alexw | maybe because I'm using Context/Priority |
| 06:26.51 | alexw | Got it |
| 06:26.58 | alexw | Context/Priority was causing he issue |
| 06:27.08 | Penguin | Yeah, there's no context or priority. |
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| 06:28.18 | snadge | i cant find a straight answer about SIP and tcp.. theres a few vague blogs that are quite high up in the google rankings, which is a bit disturbing ;) |
| 06:28.53 | snadge | experience tells me that its just not worth it.. but perhaps in some limited situations it is? |
| 06:29.45 | cyford | yes depends on ya setup |
| 06:30.24 | snadge | it seems very obvious as to why UDP is better for the actual media stream.. jitter etc |
| 06:30.47 | snadge | i didnt realise that you could use tcp for just the signalling.. and udp for the media stream |
| 06:31.17 | snadge | so thats what leads me to wonder about the pros/cons for using tcp over udp for the signalling |
| 06:31.39 | cyford | if ur using alot of microsoft or binding with others that use tcp it makes things easier, also some mobile carriers block or re structure sip udp for there own needs, changing to tcp helps there too |
| 06:32.31 | snadge | also i've heard it is easier on mobile devices batteries |
| 06:32.45 | [TK]D-Fender | <snadge> i didnt realise that you could use tcp for just the signalling.. and udp for the media stream <- RTP is ALWAYS UDP |
| 06:33.26 | snadge | right.. so the provider i work for, simply doesnt accept tcp at all.. except on one experimental / testing server thats not part of the core network |
| 06:33.36 | snadge | im not sure what the logic is behind that |
| 06:33.43 | [TK]D-Fender | <snadge> so thats what leads me to wonder about the pros/cons for using tcp over udp for the signalling <- Because TCP is stateful and you don't need keep-alives for NAT mapping, etc which has nasty effects on battery life on cellphones for instance. |
| 06:34.02 | snadge | right |
| 06:42.52 | [TK]D-Fender | heads to bed |
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| 09:16.59 | ChannelZ | So here's a question I dunno if anyone has encountered. I have a friend who wants to switch to VoIP at his house but use his analog phones. He has a patch panel of sorts where his landline comes in and then splits to all his jacks throughout the house. Do these little ATAs like the SPA112 (or the old PAP2, 3102, etc) even have enough power to ring multiple phones if we patched it in in place of the land line, or will they explode? |
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| 09:22.46 | phix | ChannelZ: I use tdm400s, and they work fine for multiple phones |
| 09:23.39 | ChannelZ | Yeah but he's not going to run a server in his house and spend several hundred bucks on a TDM |
| 09:24.08 | phix | TDM is about $250 - $300 |
| 09:24.35 | phix | or you can get SPA ATA I guess, although I have only tested them with one phone |
| 09:24.55 | phix | I could plug it up to my home phones if you like and let you know if it explodes :) |
| 09:27.58 | ChannelZ | heh - I need to ask him tomorrow how many he has actually |
| 09:28.58 | phix | I have 3 phones in my house, between 5 - 20metres away |
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| 09:31.50 | ChannelZ | I guess the SPA112 has 2 ports so I know I could get at least 2 of them to work anyway |
| 09:33.43 | ChannelZ | Hmm.. or the Grandstream Handytone 704 has 4. Though I don't know anything about them, if they are horrible or not. |
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| 09:49.26 | MaliutaLap | ChannelZ: or a TDM can have X ports and all of them will work :P |
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| 11:02.29 | alex999 | Hello everyone... I have a problem with MWI... can someone give me an help? |
| 11:02.48 | WIMPy | No |
| 11:02.52 | WIMPy | ~ask |
| 11:02.53 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:02.56 | alex999 | thanks |
| 11:04.49 | alex999 | I have recently installed Asterisk 13.2.... everything seems to work fine except MWI... My phones don't blink and i do not see any sip notify message for MWI |
| 11:05.24 | WIMPy | Did you configure the mailbox for the sip account? |
| 11:05.55 | WIMPy | Is the phone subscribing to MWI? |
| 11:06.48 | alex999 | Obvioulsy I am using PJSIP with asterisk 13 and I have set MWI Subscription Type to unsolicited |
| 11:07.45 | alex999 | they are 4 snom 760, but the same is for bria 4 and other phones |
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| 11:12.10 | alex999 | any idea |
| 11:12.31 | WIMPy | has no idea about pjsip, sorry. |
| 11:13.05 | alex999 | thanks wimpy |
| 11:15.31 | WIMPy | If noone answers, try to ask again in a few hours when others are awake. |
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| 12:12.36 | alex999 | Hello Everyone... I have recently installed Asterisk 13.2.... everything seems to work fine except MWI... My phones don't blink and i do not see any sip notify message for MWI |
| 12:13.11 | alex999 | Looking though out asterisk verbose .... tere is no trace of any message.summary |
| 12:14.04 | alex999 | note asterisk is at 13.2 using pjsip |
| 12:15.36 | alex999 | subscription type is unsolicited |
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| 13:02.56 | alex9999 | Anybody.... |
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| 13:30.48 | volga629 | Hello Everyone is version 11 have support supportpath ? |
| 13:32.00 | WIMPy | what? |
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| 13:49.16 | [sID] | How learned from the Header from just "anonymous"? |
| 13:49.16 | [sID] | I mean because I have flexible dial plan to check whether the connection that comes is from anonymous |
| 13:49.24 | wdoekes | whether asterisk 11 has SIP Path support, I guess |
| 13:53.32 | [TK]D-Fender | [sID]: You'll know the connection is from "anonymous" when it comes into the context you specified under [general] instead of one's specified in your peers |
| 13:57.59 | volga629 | support path need it to route traffic to edge proxy and add received= line |
| 13:58.07 | volga629 | ok I tested it worked |
| 13:59.00 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 13:59.15 | volga629 | Path: <sip:outbound@proxy_ip;lr;received=sip:clientPub_ip:5063%3Btransport%3Dtls> |
| 13:59.40 | [sID] | [TK]D-Fender: I thought that something like this Set(FROM=${CUT(SIP_HEADER(From),@,1)}) |
| 13:59.40 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-jegngkfxbfiuxqtv) |
| 14:00.15 | [TK]D-Fender | [sID]: you need to be specific in what you're talking about. |
| 14:00.30 | [TK]D-Fender | [sID]: You said anonymous CONNECTION. Not CALLERID |
| 14:00.40 | [TK]D-Fender | [sID]: Very different things. |
| 14:01.20 | [TK]D-Fender | [sID]: For CallerID you can check the function or the privacy flags on the channel. |
| 14:01.56 | [sID] | ehhh |
| 14:02.15 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 14:02.19 | [sID] | I want to download from the Header information or the UDF is the word "Anonymous" and that's all. |
| 14:02.43 | [sID] | UDF=FROM |
| 14:05.51 | [TK]D-Fender | [sID]: CALLERID should hold that |
| 14:06.22 | [TK]D-Fender | [sID]: otherwise you could do it the hard way like you showed an attempt at |
| 14:07.36 | [sID] | [TK]D-Fender: |
| 14:07.36 | [sID] | FreePBX sends me to another asterisk (billing) in the form of hidden number FROM "anonymous" <sip: anonymous @> |
| 14:07.39 | [sID] | And now I need to know that if it sends is set in the dialplan this and that |
| 14:09.43 | [TK]D-Fender | [sID][TK]D-Fender: I thought that something like this Set(FROM=${CUT(SIP_HEADER(From),@,1)}) <- I DID just say you could do it this way... but you sholdn't HAVE to. |
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| 14:12.45 | [sID] | [TK]D-Fender: Why ? |
| 14:12.58 | [TK]D-Fender | [09:01][TK]D-Fender[sID]: For CallerID you can check the function or the privacy flags on the channel. |
| 14:13.00 | [TK]D-Fender | ^^^^^^^^^ |
| 14:14.39 | [sID] | [TK]D-Fender: |
| 14:14.40 | [sID] | Give an example |
| 14:16.38 | [TK]D-Fender | [sID]: "core show function CALLERID" , "core show function CHANNEL" |
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| 14:44.31 | alex999 | Hello Everyone... I have recently installed Asterisk 13.2.... everything seems to work fine except MWI... My phones don't blink and i do not see any sip notify message for MWI |
| 14:44.40 | alex999 | Looking though out asterisk verbose .... tere is no trace of any message.summary |
| 14:44.50 | alex999 | note asterisk is at 13.2 using pjsip |
| 14:44.59 | alex999 | subscription type is unsolicited |
| 14:45.38 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 14:45.48 | alex999 | do you know if currently there are some issues with MWI and Asterisk 13.2 |
| 14:49.08 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 14:49.11 | darkdrgn2k | hi all, is it at all possible to have a server side jitter buffer enabled for only a handfull of extensions? |
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| 15:15.40 | alex999 | Hello can someone help me with MWI |
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| 15:27.01 | *** join/#asterisk alex9999 (~alex999@host172-102-static.15-188-b.business.telecomitalia.it) |
| 15:39.27 | alex9999 | sorry ....can you please confirm you are reading me |
| 15:42.03 | file | yes. |
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| 15:56.27 | panamark | Hello am facing some problems with *69 feature code. How can i prepend 9 before the number so i can dial out? |
| 15:58.52 | [TK]D-Fender | * has no such code |
| 15:59.11 | [TK]D-Fender | As for dialing out.. you can pass whatever you want to peers you use to dial out. |
| 15:59.27 | [TK]D-Fender | Match your pattern, put the 9 in front of the number you pass |
| 16:01.03 | panamark | to be honest its a freepbx distro, maybe i should ask at #freepbx. Thought it was asterisk specific |
| 16:01.09 | panamark | Thanks in advance though |
| 16:01.11 | panamark | :) |
| 16:02.11 | [TK]D-Fender | Everything you dial there follow its rules and is something you need to keep in their channel |
| 16:02.42 | alex9999 | D-fender I am facing problems with asterisk 13.2 and MWI.... I do not receive notification ... do you know if with this version are there some issues with MWI |
| 16:02.55 | [TK]D-Fender | alexShow the actual debug for your call. |
| 16:03.02 | [TK]D-Fender | alex9999: Show the actual debug for your call. |
| 16:03.17 | alex9999 | side phone or side asterisk |
| 16:03.29 | [TK]D-Fender | alex9999: And lorsungcu_ asked you a question in #freepbx about this |
| 16:03.46 | [TK]D-Fender | alex9999: Asterisk always. And keep this in #freepbx |
| 16:04.31 | alex9999 | yes I did what he suggested me to do ... change the MWI subscription type in unsolicited... but nothing happened |
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| 17:12.00 | *** join/#asterisk tonsofpcs (~mythbuntu@rivendell/member/tonsofpcs) |
| 17:12.35 | tonsofpcs | hi, I want to test a call-center-esque idea and I was wondering if there was a good software sip client that I could run with a bunch of 'xml buttons' to simulate how a real button set would work |
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| 17:14.46 | [TK]D-Fender | what is a " real button set"? |
| 17:15.35 | tonsofpcs | a phone that has a bunch of buttons like a DSS console |
| 17:17.45 | [TK]D-Fender | Ok, but "XML buttons" is not a concept for * really. |
| 17:18.27 | tonsofpcs | basically I want to make a desk set run controls against * to handle putting calls on hold and transferring them to clients and pulling them to hold from clients, etc. |
| 17:19.15 | [TK]D-Fender | If you want to make a soft-phone there are plenty of free ones out there to learn from |
| 17:19.29 | tonsofpcs | I don't want to make a soft phone... |
| 17:19.31 | [TK]D-Fender | The actual act of transferring, hold, etc are documented based on standard protocols |
| 17:19.41 | [TK]D-Fender | What re you trying to make exactly? |
| 17:19.50 | tonsofpcs | I want to test functionality without actually buying a hard phone. |
| 17:19.58 | tonsofpcs | (firm phone?) |
| 17:20.17 | [TK]D-Fender | then just download a soft phone |
| 17:20.19 | [TK]D-Fender | plenty out there |
| 17:20.24 | [TK]D-Fender | ~sofphone |
| 17:20.27 | [TK]D-Fender | ~softphone |
| 17:20.28 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 17:21.55 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 17:25.44 | *** join/#asterisk Elio19 (~elio19@unaffiliated/elio19) |
| 17:29.22 | Elio19 | ~primer |
| 17:29.22 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 17:30.20 | tonsofpcs | basically I want one desk set that accepts a dozen or so calls and can pick them then park them, then another desk set that has 'pick-up' and park/hangup buttons but when it 'picks up', the call actually gets handed off to a different SIP client |
| 17:32.12 | [TK]D-Fender | tonsofpcs: that'd just be a softphone that can handle X amount of calls |
| 17:32.27 | [TK]D-Fender | Each one has their own limits as to hpow many they can do |
| 17:33.32 | [TK]D-Fender | Even just 2-3 is enough to prove what the flow looks like |
| 17:34.03 | [TK]D-Fender | As you said this was just to test functionality. |
| 17:34.25 | [TK]D-Fender | It'd work the same for 2 as for 200. |
| 17:34.34 | tonsofpcs | [TK]D-Fender: how do you make it so that 'picking' a call on one desk set actually makes it transfer to a different one? |
| 17:34.50 | tonsofpcs | and how do you control putting that other set back on-hook? |
| 17:34.59 | tonsofpcs | (it can go off-hook automatically) |
| 17:35.19 | [TK]D-Fender | SIP 302 redirect can throw a call to another location. |
| 17:35.43 | [TK]D-Fender | Also, the concept of "on-hook" does not exist in SIP and other similar protocols |
| 17:35.59 | tonsofpcs | hang up, disconnect, ... |
| 17:36.06 | [TK]D-Fender | There are many conepts you'll have to toss your understanding of right out the window for. |
| 17:36.14 | [TK]D-Fender | concepts* |
| 17:37.35 | tonsofpcs | so I need to find a desk set with fully programmable buttons in order to do redirects and drops? |
| 17:39.51 | tonsofpcs | and I presume I can use a live input for MOH? |
| 17:40.16 | *** join/#asterisk CeBe (~CeBe@port-92-200-110-115.dynamic.qsc.de) |
| 17:40.18 | [TK]D-Fender | Line-in for MoH can be used for Asterisk |
| 17:40.36 | [TK]D-Fender | This "fully programmable" thing you're mentioning isn't a "thing" |
| 17:40.49 | tonsofpcs | ok, how do I make buttons do this? |
| 17:41.16 | [TK]D-Fender | Any given phone device "soft, hard or otherwise" will have its "buttons" for ending a given call, transferring, etc |
| 17:41.24 | [TK]D-Fender | You don't make buttons do anything. |
| 17:41.28 | [TK]D-Fender | they HAVE a function already |
| 17:41.37 | [TK]D-Fender | You don't define a transfer button. They are CODED |
| 17:41.37 | tonsofpcs | right, I want specific functions on my buttons. |
| 17:41.50 | [TK]D-Fender | Then you're talking about making your won soft phone. |
| 17:41.55 | [TK]D-Fender | own* |
| 17:42.18 | [TK]D-Fender | There is none out there that I've ever seen that let you invent your own layout. |
| 17:43.36 | [TK]D-Fender | And there are certain call handling features that are not commonly implemented in ways that might be possible. |
| 17:45.32 | [TK]D-Fender | For instance a phone will use a redirect for a fixed "forward", but not offer an on-demand means of tossing calls arriving at your phone without answering first. |
| 17:46.00 | *** join/#asterisk DynamicFail (~DynamicFa@147.177.70.238) |
| 17:46.15 | DynamicFail | What is it called in asterisk where you can dynamically share numbers etc so you don't have to update tons of trunks each time you something gets added a branch |
| 17:46.47 | [TK]D-Fender | DynamicFail: 2 concepts over this : DUNDI, and e.168 |
| 17:47.40 | DynamicFail | [TK]D-Fender, perfect... do you know what cisco's similar system is called |
| 17:47.50 | [TK]D-Fender | DynamicFail: Nope. |
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| 18:11.08 | linocisco | which billing software is perfect for asterisk (free and commercial) |
| 18:11.09 | linocisco | ? |
| 18:11.22 | [TK]D-Fender | LOL |
| 18:11.37 | [TK]D-Fender | "Perfect" = automatic fail |
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| 18:21.50 | linocisco | which free softphones for asterisk has "call forward" and "transfer" feature available for free? |
| 18:22.33 | chokesmaster | Hi, I'm trying to configure tls on my Incredible pbx setup and I am not able to configure a Granstream GXP-2120 with SRTP but i get 488 Not Acceptable on the phone. |
| 18:26.32 | chokesmaster | Is anyone online right now? |
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| 18:30.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:33.36 | [TK]D-Fender | chokesmaster: Your extension and core settings are not matching what the other side is requesting |
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| 18:35.43 | chokesmaster | I found some documents that say i neet to add ignorecryptolifetime=yes for grandsstreams phones |
| 18:35.56 | chokesmaster | but it still doesnt work |
| 18:37.07 | chokesmaster | does the srtp ports neet to be udp as well? |
| 18:37.40 | malcolmd | an aside, the crypto lifetime bit should now work in trunk, thanks oej and mjordan |
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| 18:40.48 | [TK]D-Fender | chokesmaster: RTP is always UDP |
| 18:41.07 | Katty | slides file some monopoly money. |
| 18:41.15 | file | hi |
| 18:41.20 | Katty | hi. |
| 18:43.06 | chokesmaster | [TK]D-Fender: Ok |
| 18:50.06 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 18:54.58 | chokesmaster | As soon as I force encryption (SRTP) I get the 488 Not acceptable error but i can't see any settings in the grandstream phone... |
| 18:55.10 | mjordan | malcolmd: latest tip of the 11/13 branches too! |
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| 19:10.37 | malcolmd | ah, m'bad, didn't notice that it was pushed there :D |
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| 19:13.04 | file | pushes malcolmd to the 1.8 branch |
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| 19:15.59 | malcolmd | it's pretty dark down here |
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| 21:45.45 | gryffus | hello, i'm searching for a solution to VoIP/SIP for a small company. The focus is on security and future interconnecting with company's XMPP services. I have heard of Kamailio, OpenSIPS and FreePBX. My apology for asking in Asterisk channel, but i couldn't find any generoc VOIP channel on freenod. Can you guys recommend me one of the server software mentioned and tell me why? Thanks for any clarification and advice. |
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| 21:47.34 | WIMPy | They all have their purpose. What's your's exactely? |
| 21:50.16 | gryffus | Calling from mobile phones and computers with Z/SRTP and forwarding messages to and from XMPP. |
| 21:50.50 | gryffus | Nie web administration would be nice also |
| 21:51.27 | gryffus | Please specify if i get your question wrong |
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| 21:55.03 | gryffus | I won't need to call from VoIP to regular telephone network in the near future, but AFAIK all solutions can be connected to Asterisk in the future to do this job...? |
| 21:56.17 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:57.06 | jeev | any recommendations to plug asterisk into salesforce? they want their calls logged.. |
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| 23:35.38 | oli-werk | hey guys/girl, i can't find any information on how wrapuptime actually works |
| 23:35.46 | oli-werk | does anyone have a link to some documentation on it? |
| 23:36.52 | oli-werk | or perhaps could explain: 1) what agent (or extension?) attributes are modified when wrap up time is active |
| 23:37.38 | oli-werk | 2) is there a way to view whether an agent/extension is currently on wrap up time? agent show / queue show don't seem to change when it is active |
| 23:38.09 | oli-werk | nor does the asterisk full(.log) show any indication |
| 23:46.09 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:59.06 | newtonr | oli-werk, I can't answer your question off-hand, but any documentation for wrapuptime will be in the sample config file, on the wiki, in the Asterisk definitive guide, or in Asterisk command line help text for related applications, functions, etc. |
| 23:59.37 | newtonr | beyond that, you'll probably need someone to look at the source code |