| 00:00.02 | [TK]D-Fender | Could be |
| 00:00.12 | [TK]D-Fender | the phone answered... and it says "no can do" |
| 00:00.16 | [TK]D-Fender | So look at the phone. |
| 00:00.27 | [TK]D-Fender | Either it can't handle more.. or is being more actively rejected |
| 00:00.54 | mushroomed | [TK]D-Fender: I have an option named "Disable DND Button" and it is set to "No" |
| 00:01.00 | mushroomed | Should I set it to "YES" ? |
| 00:01.00 | ChannelZ-Wk | The Retransmissions are a bit of a bother as well |
| 00:01.06 | leifmadsen | sounds like the DND is not disabled then :) |
| 00:01.19 | [TK]D-Fender | mushroomed: that disbles the button.... that doesn't say if you're USING it or not. |
| 00:01.46 | mushroomed | [TK]D-Fender: When I pick up the handset I have no sound |
| 00:01.46 | [TK]D-Fender | So DND has not been "prevented"... and it is refusing the call... so it COULD be set.... |
| 00:02.00 | [TK]D-Fender | mushroomed: that's possibly another matter entirely |
| 00:02.19 | mushroomed | [TK]D-Fender: I'm going crazy |
| 00:02.26 | [TK]D-Fender | swiss__: Looks like DTMF is arriving late.... |
| 00:02.43 | [TK]D-Fender | swiss__: because we see it run out of time.. and then seem to get them. |
| 00:02.48 | swiss__ | im not worried about the softphone results, thats more just a sanity check for me than an actual use case |
| 00:03.04 | swiss__ | this is going to only call cell phones |
| 00:06.29 | mushroomed | Damn~ |
| 00:07.23 | [TK]D-Fender | mushroomed: restart the phone and lets look at the registration process and test immediately after |
| 00:07.25 | ChannelZ-Wk | it just sounds like it's not entirely really configured correctly |
| 00:12.32 | swiss__ | welp thanks anyway, ill be back tomorrow with the same questions. |
| 00:14.33 | [TK]D-Fender | lol |
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| 01:17.00 | puzzled | hi |
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| 02:22.39 | ChannelZ | ahoy |
| 02:23.51 | [TK]D-Fender | chips |
| 02:24.17 | ChannelZ | mmmm |
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| 09:11.25 | cusco | hi |
| 09:12.41 | cusco | so.. using realtime queue_members configuration, member has 'interface' and 'state_interface' |
| 09:13.59 | cusco | setting state_interface to hint:ext@context should be ok, right? |
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| 09:32.50 | bhavikpatel6842 | Hi all |
| 09:33.35 | bhavikpatel6842 | I want to integrate Webrtc and for that I am using asterisk 11.9.3 version. |
| 09:33.41 | bhavikpatel6842 | but getting issue. |
| 09:34.15 | bhavikpatel6842 | like not able to call using two parties. |
| 09:36.48 | *** join/#asterisk aurs (~aurs@228.84-48-61.nextgentel.com) |
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| 09:39.12 | bhavikpatel6842 | SORRY my asterisk version is Asterisk 11.9.0 |
| 09:40.43 | bhavikpatel6842 | http://pastebin.com/G5FVKcpp |
| 09:41.37 | bhavikpatel6842 | you can check my configuration. |
| 09:46.22 | aurs | bhavikpatel6842, you have "deny=0.0.0.0/0.0.0.0" but no permit? will that work? |
| 09:47.29 | bhavikpatel6842 | okey I remove that line |
| 09:47.33 | bhavikpatel6842 | let me test |
| 09:48.02 | aurs | I didn't catch what the problem was, but that didn't look right :) |
| 09:48.21 | bhavikpatel6842 | Yes I remove so now able to register. |
| 09:48.35 | bhavikpatel6842 | I am using SIPm5 demo for testing webrtc calls |
| 09:50.05 | bhavikpatel6842 | call is pass but when I accept call that time showing like call rejected. |
| 09:51.12 | bhavikpatel6842 | Got SIP response 603 "Failed to get local SDP" back from ip:18934 |
| 09:51.23 | bhavikpatel6842 | this error showing in asterisk |
| 09:53.17 | cusco | core set verbose 15 |
| 09:55.40 | *** join/#asterisk ghoti (~paul@scratch.it.ca) |
| 09:58.05 | bhavikpatel6842 | showing same not extra line of error |
| 09:58.48 | aurs | perhaps you can find something here https://issues.asterisk.org/jira/browse/ASTERISK-23191 |
| 09:58.58 | aurs | didn't read it for you though ;) |
| 10:00.31 | aurs | anyone else here using Aastra 675xi phones? I get "No service" on them on the first register, but it's ok on the re-register. (asterisk 1.8 and aastra fw 3.3) |
| 10:05.37 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 10:11.55 | bhavikpatel6842 | I checked this link but still have issue and getting same issue. |
| 10:12.09 | bhavikpatel6842 | is this issue by asterisk side or Browser side ? |
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| 10:17.51 | bhavikpatel6842 | Can any one have idea about how to solve Got SIP response 603 "Failed to get local SDP" back ? |
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| 11:48.54 | peetaur | How can I tell which file and line number causes this error? |
| 11:48.56 | peetaur | [2014-05-09 13:45:56] WARNING[16983][C-00000041]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>', expecting $end; Input: |
| 11:52.01 | Chainsaw | peetaur: I'm going to bet it's the dialplan in /etc/asterisk/extensions.conf |
| 11:52.54 | peetaur | ok |
| 11:53.00 | peetaur | in that file, is this good syntax? exten => 5001,1,Set(__RINGTIMER=${IF($[${DB(AMPUSER/5001/ringtimer)} > 0]?${DB(AMPUSER/5001/ringtimer)}:${RINGTIMER_DEFAULT})}) |
| 11:53.20 | peetaur | there is another line with 5701 instead of 5001 which looks the same other than the ext number, and doesn't cause that error ... so I don't have a clue |
| 11:54.39 | Chainsaw | peetaur: Comment out the line you're not sure about and reload. |
| 11:54.57 | Chainsaw | peetaur: That is how you find faults. You make a single change each time and that makes it worse or it makes it better. |
| 11:55.09 | peetaur | and here are the only 2 other lines with "> 0" on them exten => s,n,GotoIf($[${LEN(${INPUT})} > 0]?${INPUT},1:t,1) exten => s,n,GotoIf($[ ${TTL} > 0 ]?continue) |
| 11:55.55 | peetaur | the problem came from using FreePBX of one version, doing an upgrade, then a backup, then restoring to another server... the old extension I have is broken (anyone calling gets UNAVAIL error), but new extensions work fine |
| 11:56.15 | peetaur | so I think if I simply remove that line, it might just never reach the other line that has the problem |
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| 11:57.47 | jwwwww | Hello. |
| 11:59.13 | Chainsaw | peetaur: If you're using FreePBX to create a dial plan, you need to talk to them in #freepbx |
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| 12:31.34 | peetaur | FYI my issue was fixed by going in the FreePBX gui for the bad extension, then hitting save without any changes. |
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| 12:36.34 | Chainsaw | peetaur: Exactly, so Asterisk was not at fault. |
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| 12:38.49 | peetaur | I didn't think it was, just wanted to understand the syntax error. |
| 12:40.39 | peetaur | but might I suggest an improvement of printing the config file name and line number for errors? |
| 12:49.51 | marceloamorim | guys, I wish to know, and I´ll be honest, I never do any search about this, but anyone knows if exist any addon or functionality that makes me call through the usb connection |
| 12:51.52 | Chainsaw | marceloamorim: USB telephony adapters do exist. It would help to know what kind of line you wish to connect to. |
| 12:51.53 | WIMPy | That question is extremely vague, but so far the answer is yes. |
| 12:54.01 | marceloamorim | I wondering if I can substitute those gsm gateway when I have only one chip |
| 12:54.25 | WIMPy | Google for chan_dongle. |
| 12:59.32 | marceloamorim | nice guys, ty |
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| 13:22.50 | woopstar | Hi. I'm trying to enable encryption on a sipfriend, and then use webrtc2sip to do a call from a browser. I'm connecting fine, but when doing a call, the asterisk is saying: "Matched device setup to use SRTP, but request was not!". If i disable encryption for the friend, everything works. |
| 13:23.07 | woopstar | Any clues? |
| 13:25.55 | woopstar | It comes from a problem, where we get this error: SetRemoteDescription called with a session description without crypto enabled when trying to transfer a call |
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| 13:34.33 | jwwwww | I'm trying to setup asterisk so a user call in, enter a phone number that is called for him.I think I succed in reading the phone number, but the call doesn't ring the phone, and I get a sip error 'nobody picked up' . can somebody help please ? here are my sip.conf and part of extensions.conf and sip debug : http://dpaste.com/3MYY5K9/ |
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| 14:06.46 | [TK]D-Fender | jwwwww: Retransmitting #5 (NAT) to 194.51.85.76:5060: <- they're not answering |
| 14:10.40 | jwwwww | [TK]D-Fender: I asked their support earlier, they told me my account got 1 channel in and 1 channel out. |
| 14:12.10 | jwwwww | [TK]D-Fender: do you by any chance see something problematic in my setup ? I'm not sure at all it's ok. |
| 14:17.15 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 14:17.22 | [TK]D-Fender | Retransmitting #5 (NAT) to 194.51.85.76:5060: <- is that the right IP you resolved? |
| 14:17.36 | [TK]D-Fender | INVITE sip:ipp.fr SIP/2.0 <- you also aren't even dialing a NUMBER |
| 14:17.49 | [TK]D-Fender | To: <sip:ipp.fr> |
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| 14:21.09 | jwwwww | [TK]D-Fender: no this ip doesn't seems to be the right ip. ippi.fr doesn't resove to this. |
| 14:21.38 | [TK]D-Fender | taht doesn't say ippi.fr at all. |
| 14:22.00 | [TK]D-Fender | exten => s,n,Dial(SIP/${NUMBER}@ipp.fr,30) <- looks like you didn't pay attention to the URI you're dialing |
| 14:22.01 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:22.12 | [TK]D-Fender | ipp != ippi |
| 14:22.54 | jwwwww | I correct this, and make a new try. |
| 14:26.10 | jwwwww | [TK]D-Fender: off course this help a lot. I cannot call yet, but it's going further. thanks ! |
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| 15:11.43 | Mango45 | Is there a way to make a CLI shortcut for a series of commands so I don't have to type them out each time? |
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| 15:19.13 | newtonr | Mango45, I don't think so. You can edit cli_aliases.conf to configure an alias mapped to a single command |
| 15:20.03 | newtonr | Mango45, I'm sure there are probably terminal software that would let you make macros for text-based commands |
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| 15:22.54 | Mango45 | Thank you; I will check out cli_aliases.conf. :) |
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| 15:38.37 | marceloamorim | guys, simple question, is it possible to do more than once call for iax trunk? simultaneously |
| 15:38.59 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:40.16 | pabelanger | yes |
| 15:40.21 | pabelanger | enable trunking=yes |
| 15:40.27 | pabelanger | or trunk=yes |
| 15:40.30 | pabelanger | something like that |
| 15:40.58 | marceloamorim | nice, ty pabelanger |
| 15:41.10 | cusco | hey guys .. so I'm trying to set up hints |
| 15:41.18 | cusco | using real time queue members ... |
| 15:41.26 | cusco | state_interface is set to hint:ext@cotnext |
| 15:41.28 | cusco | rigth? |
| 15:41.44 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 15:41.55 | cusco | I have done so, and in queue show .. member is Invalid |
| 15:42.04 | Tuju | is there a plugin that could be used to ban ip after unsuccessful attempts? |
| 15:42.30 | cusco | there is fail2ban and you can find on the interwebs several hints for configuring its regural expressions |
| 15:42.42 | cusco | also there is a security log in logger.conf |
| 15:42.51 | Tuju | ack |
| 15:42.56 | Tuju | i look into those |
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| 15:47.30 | pabelanger | cusco, we've had good success using custom device state with local channels for queue members |
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| 15:48.40 | cusco | pabelanger: we haven't set any custom states yet ... |
| 15:48.59 | cusco | we use ale, and set up a list of interfaces with hint in the subscribed context |
| 15:49.04 | pabelanger | cusco, right, it just gives more flexability when the phones are not attached directly to asterisk |
| 15:49.13 | cusco | right... |
| 15:49.25 | cusco | but now in the realtime queue member, I have |
| 15:49.33 | cusco | interface=Local/150@agents |
| 15:49.54 | cusco | and on the state_interface where I had 'SIP/150' now have: 'hint:150@agents' |
| 15:50.12 | pabelanger | Ya, not sure about realtime, we don't use it. We wrong something ourselves using AMI commands for queueadd / queueremove |
| 15:50.23 | pabelanger | cusco, that should be right |
| 15:50.36 | cusco | err I looked at core show hints |
| 15:50.40 | pabelanger | we use Custom:exten@something |
| 15:50.41 | cusco | <PROTECTED> |
| 15:50.44 | cusco | that _ |
| 15:50.48 | pabelanger | ya |
| 15:50.51 | cusco | ok ok |
| 15:50.59 | pabelanger | so, maybe DB is adding it? |
| 15:51.07 | cusco | no I set it that way |
| 15:51.33 | pabelanger | so do hint:_150@gents in state_interface |
| 15:51.56 | cusco | thanks for your input pabelanger |
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| 16:06.26 | cusco | pabelanger: I changed the hint back to 150 |
| 16:06.40 | cusco | 150@agents : SIP/150 State:Idle Watchers 0 |
| 16:06.46 | *** join/#asterisk jonno11 (~jonno11@amigopod.rave.ac.uk) |
| 16:06.51 | cusco | but still shows as invalid |
| 16:07.00 | cusco | 150 (Local/150@agents) with penalty 10 (realtime) (Invalid) has taken no calls yet |
| 16:07.05 | *** join/#asterisk sorressean (~tyler@69.164.206.65) |
| 16:07.20 | sorressean | Does anyone know of anything that offers good minute rates for outbound calls to vietnam? |
| 16:09.10 | cusco | tried google voice? :p |
| 16:09.25 | cusco | pabelanger: how could I debug this situation? |
| 16:10.08 | sorressean | I need it through asterisk. :| |
| 16:10.29 | cusco | asterisk can connect to googlevoice |
| 16:11.02 | *** join/#asterisk jonno11 (~jonno11@amigopod.rave.ac.uk) |
| 16:11.15 | peetaur | sorressean: what is the best you found? |
| 16:12.53 | sorressean | peetaur: nothing so far. |
| 16:14.34 | peetaur | k... well the one I use has it for 10 cents http://www.vitelity.com/international_rates/ |
| 16:14.36 | scouture | cusco: that 'queue show' line doesn't say you're using that 150@agents hint. are you sure you've specified 'hint:150@agents' as the state_interface? |
| 16:15.31 | sorressean | scratch that. turns out the best is 47 cents. |
| 16:16.22 | scouture | cusco: you'd see something like.. "150 (Local/150@agents from hint:150@agents) with penalty 10 (realtime) ..." |
| 16:18.12 | scouture | cusco: add it manually from the cli as such: queue add member Local/150@agents/n to your-queue penalty 10 as 150 state_interface hint:150@agents |
| 16:24.52 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
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| 16:50.14 | woopstar | Hi guys. How do you get the channels for an extension from asterisk using agi/cli something... |
| 16:51.01 | WIMPy | Maybe you should clarify what you want to do before we try to answer. |
| 16:54.42 | woopstar | Sure.. Im trying to do an API that's able to park a call for an extension simply. Currently im using the "core show channels concise" and then extracting the extension from the information. But concise is deprecated and I dont think it's a good solution if you have thousands of channels. |
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| 17:00.35 | WIMPy | The best way is usually to use AMI. |
| 17:01.01 | WIMPy | But that extension thing is still rather unclear. |
| 17:06.27 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 17:07.14 | woopstar | well.. im trying to code an api, where you just use the extension number and then it parks the active call |
| 17:07.24 | woopstar | so i need to grab the active channel for the extension |
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| 17:14.39 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 17:14.46 | BKhan | Hi |
| 17:16.21 | BKhan | I am using asterisk 11.9 but hangup function is not working |
| 17:18.16 | *** join/#asterisk SuperNull (~YoMommaEa@24-148-101-238.ip.mhcable.com) |
| 17:18.38 | [TK]D-Fender | hangup isn't a function. |
| 17:19.21 | BKhan | http://pastebin.freeswitch.org/22518 |
| 17:19.29 | SuperNull | is the manager interface capable of providing a statistic for active call channels ? |
| 17:19.47 | SuperNull | trying to find a unified way to pull max concurrent call stats for our systems |
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| 17:24.30 | BKhan | D-Fender> did u check pastebin |
| 17:25.08 | WIMPy | BKhan: That URL requests a username and password. |
| 17:25.18 | WIMPy | SuperNull: You have to count yourself. |
| 17:26.39 | BKhan | http://pastebin.com/qQvSR9xa |
| 17:26.57 | BKhan | WIMPy: please check now |
| 17:27.49 | WIMPy | BKhan: The log is not for the same extension as the dialplan snippet above. |
| 17:28.03 | WIMPy | And I don;t see anything not working, either. |
| 17:29.18 | BKhan | oh sorry but using same code for *1 and *3 |
| 17:29.42 | BKhan | thats also amazing for me that apprently no error found |
| 17:29.45 | Qwell | Those are not the same at all. |
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| 17:41.12 | [TK]D-Fender | Your *3 extensions all start with "n". We have no proof where thos priorities fall at all |
| 17:41.27 | [TK]D-Fender | And no proof of what *1 looks like short of what we see executing. |
| 17:41.57 | [TK]D-Fender | You also say "hangup doesn't work", yet we don't see a single line after the hangup is called in your debug to prove that it didn't |
| 17:47.09 | *** join/#asterisk Geriatrix (~kvirc@mail.vancouveronthenet.com) |
| 17:47.36 | Geriatrix | hey guys - i need to set up bluetooh on my box to get asterisk working with it - can anyone give me a hand |
| 17:47.44 | Geriatrix | i amhaving issues with pairing password |
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| 19:03.23 | ipengineer | Does anyone know if ringinuse=no works with queues in Asterisk 12? The device is showing up as inuse in queue show but the call is still being sent to that device |
| 19:03.37 | [TK]D-Fender | Show us |
| 19:08.32 | ipengineer | [TK]D-Fender: I got it.. Dynamic agents apparently need to be logged out/logged back in if that option changes.. Whats weird is the device that had (ringinuse disabled) had been logged in the longest. |
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| 19:30.34 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.82.191) |
| 19:32.47 | vlad_starkov | Questions: 1) Is it possible to check whether current channel is being recorded? 2) If I call Monitor() multiple times for current channel how it will influence on current channel call recording? |
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| 19:37.22 | Mango45 | I would like to know this too. |
| 19:58.26 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 19:59.32 | kuku | I'm running 1.4 and have polycom phones. For some reason certain order polcyom phones when they try to login to get their voicemail, asterisk can't read their password - as if it wasn't entered |
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| 20:03.32 | Qwell | ~upgrade asterisk |
| 20:03.32 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:03.43 | kuku | Qwell: 1.8 ok ? |
| 20:07.29 | [TK]D-Fender | kuku: Almost cerrtainly wrong DTMF mode set |
| 20:07.37 | [TK]D-Fender | kuku: Just fix that first to be sure |
| 20:07.57 | *** join/#asterisk lecoq (uid32203@gateway/web/irccloud.com/x-aotuzheonehwesci) |
| 20:08.05 | [TK]D-Fender | kuku: And then still plan to migrate to something that isn't a huge security risk & filled with bugs that will never be fixed |
| 20:12.01 | Geriatrix | ok - i got bluetooth to "work" but my blackberry notsamsung cannot wonnect |
| 20:12.08 | Geriatrix | only the old nokia for which i dont' have a sim card |
| 20:12.33 | SuperNull | does 'sip show channels' show both calling party and called party audio channels or does it put both as a single channel ? |
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| 20:14.30 | kuku | [TK]D-Fender: I plan to - but I can't upgrade them mid day - would like to solve hte dmtl issue somehow. |
| 20:14.54 | kuku | dtmfmode = rfc2833 is set in sip.conf |
| 20:21.42 | *** join/#asterisk swiss__ (~swiss__@70.184.40.66) |
| 20:21.52 | swiss__ | hello again all |
| 20:22.27 | swiss__ | but especially [TK]D-Fender |
| 20:22.48 | [TK]D-Fender | kuku: Where? It's a big file. |
| 20:22.57 | [TK]D-Fender | kuku: Show us the peers & the call with SIP debug enabled |
| 20:33.24 | kuku | [TK]D-Fender: thats in the [general] of sip.conf |
| 20:33.47 | kuku | Tried inband as well. Also tried setting dtmfmode on the extension itself, auto as well. |
| 20:33.56 | kuku | ok - ill try to get a log |
| 20:34.45 | paulc | Joseph: Short of a court order we have no means of getting that detail. I have checked even with my supervisor just to confirm. And its the same result there is nothing we can do here to get those numbers. Even if I was to do a ticket to our admins they would just come back with the same result as I am currently providing. I will credit it for you then for this one time adjustment due to your tenure with us. |
| 20:34.52 | paulc | oops.. that'd be the wrong window then.. |
| 20:37.44 | malcolmd | heh |
| 20:40.09 | [TK]D-Fender | checkout time, BBL |
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| 22:16.14 | mushroomed | Hi, is there any way to record every call without pressing my automixmon feature key(s)? |
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| 22:47.25 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-169.ph.ph.cox.net) |
| 22:49.55 | RaMcHiP | Hello everyone! I have gotten my trunk setup now with all my extensions and all the trimmings. I am trying to integrate VTiger into the PBX so that I can use click to dial. In vtiger I have went under tools -> PBX Manager and added in my asterisk information. I created a user with an exension associated to it and when I click it number it says pick up the extensions receiver to dial but |
| 22:49.56 | RaMcHiP | when I pick up that extensions receiver, I do not get anything |
| 22:50.08 | RaMcHiP | Anyone able to assist me with this functionality? |
| 22:50.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:50.43 | RaMcHiP | PS I am using cisco 7940's |
| 22:51.47 | [TK]D-Fender | RaMcHiP: "I created a user with an exension associated to it" <- where, how? |
| 22:52.08 | [TK]D-Fender | And we know nothing about vtiger or whatit asks you to enter |
| 22:52.13 | RaMcHiP | In Vtiger under settings when logged into admin |
| 22:52.31 | RaMcHiP | I was just hoping someone had integrated it :) |
| 22:52.36 | [TK]D-Fender | You should refer to their docs as to what those are |
| 22:52.38 | RaMcHiP | I got asterisk working fantastically |
| 22:52.46 | KNERD | not since they make it closed source |
| 22:52.51 | KNERD | time to move onm |
| 22:52.57 | RaMcHiP | Ohhh? |
| 22:53.12 | RaMcHiP | What do you recommend KNERD |
| 22:53.12 | [TK]D-Fender | Last I heard vTiger was OSS |
| 22:53.49 | KNERD | well...they no longer will release any more open soure. Just like SugarCRM has done |
| 22:54.16 | RaMcHiP | ouch :) |
| 22:54.20 | RaMcHiP | Well then... |
| 22:54.41 | RaMcHiP | Any advice on a replacement open source CRM? |
| 22:54.42 | KNERD | i dont know what else is out there |
| 22:54.54 | KNERD | they were the 2 major ones for linux |
| 22:54.57 | RaMcHiP | It seemed like just sugar or Vtiger :) |
| 22:56.11 | KNERD | maybe http://www.opencrx.org/ ? |
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| 23:01.42 | RaMcHiP | Thanks KNERD! |
| 23:04.20 | KNERD | RaMcHiP: they are trying to say SugarCRm Community Edition will be continued, but not at the moment...they wont give a release time for version 7.x CE. |
| 23:04.22 | KNERD | http://forums.sugarcrm.com/f22/whats-going-community-edition-88583/ |
| 23:04.31 | KNERD | but nobody seems to believe them |
| 23:05.08 | KNERD | only a paid version of 7 is available now |
| 23:05.14 | KNERD | you may want to lok at that |
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| 23:16.21 | RaMcHiP | I really want to stay away from licensing.. Especially per seat :) |
| 23:16.39 | RaMcHiP | I am thinking that we can still use VTiger and just modify as we see fit |
| 23:16.48 | RaMcHiP | and fix any document security issues as they come along |
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| 23:23.31 | vlad_starkov | Questions: 1) Is it possible to check whether current channel is being recorded? 2) If I call Monitor() multiple times for current channel how it will influence on current channel call recording? |
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