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| 00:17.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.1 (2015/01/28), 11.15.1 (2015/01/28), 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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| 06:26.04 | ChannelZ | So THAT's what "serious hardware for serious voice systems" look like - http://red-fone.net/ |
| 06:26.29 | AnonGirl | wat |
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| 06:28.04 | [TK]D-Fender | <PROTECTED> |
| 06:28.14 | [TK]D-Fender | I remember when they had products... |
| 06:28.31 | [TK]D-Fender | Products I wouldn't touch with a 10ft pole. |
| 06:28.46 | [TK]D-Fender | glares at TDMoE |
| 06:29.19 | ChannelZ | is cleaning out his bookmarks |
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| 09:16.15 | pawiecki | hello! |
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| 09:40.38 | phix | hi gang |
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| 10:11.48 | [sID] | Hello |
| 10:15.14 | [sID] | I have a problem with the transfer CallerID respectively for the connection. |
| 10:15.19 | [sID] | It looks as follows |
| 10:15.21 | [sID] | Public phone number is provided with a wildcard asterisk calls I answered and there is bland tranafer |
| 10:15.26 | [sID] | and now I'm in the header |
| 10:15.37 | [sID] | FROM "947211111" <sip: 12345@192.168.1.1> |
| 10:15.52 | [sID] | and needs |
| 10:15.52 | [sID] | FROM "947211111" <sip: 947211111@192.168.1.1> |
| 10:15.52 | [sID] | How can I get one? |
| 10:15.57 | [sID] | I tried CallerID (num), (name), (all), (number) and nothing :( |
| 10:24.19 | Gugge | if you det tje callerid number, it should be the part after sip: and before @ |
| 10:24.25 | Gugge | s/tje/the/ |
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| 10:29.21 | bibz2 | hey there. |
| 10:29.22 | [sID] | I do not understand? |
| 10:29.38 | bibz2 | Is it possible to write the CDR in realtime onto 2 databases? |
| 10:29.59 | [sID] | no setting to what I wrote above is not doing me such entry |
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| 12:28.44 | ipalmer | Hi all, can anyone tell me if AMI has a default buffer size? |
| 12:29.50 | ipalmer | Hi all, can anyone tell me if ami has a default buffer size? |
| 12:30.00 | ipalmer | oops sorry, didn;t mean to post twice |
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| 14:01.59 | ipalmer | Can anyone tell me if AMI has a default buffer size when sending out events? |
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| 14:58.28 | loko- | Can anyone recommend a good *free* toll-free outbound calling service - needs to work with both Asterisk and also directly connected SIP phones (Cisco 7960). I currently use Flowroute and it works fine with some 800 numbers, but other numbers (Web ex main line) it fails on. |
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| 15:31.54 | dan_j | Hi. In the dialplan, which variable or function contains the name of the SIP Peer? |
| 15:32.11 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 15:32.29 | skrusty | dan_j: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 15:32.43 | WIMPy | function CHANNEL |
| 15:33.07 | przerull | hi everyone. quick question so __DYNAMIC_FEATURES only appears to work for me when I put it in the globals part of the dialplan. any ideas why that might be? |
| 15:33.24 | [TK]D-Fender | przerull: It doesn't only work then |
| 15:33.53 | [TK]D-Fender | przerull: It only works when the variable is set for the calling channel. Doesn't matter if you do it in the run-time processing, or as a global. |
| 15:34.00 | dan_j | WIMPy: Thanks. I thought that was going to be the case. I've been using ${CALLERID(num)} but I've discovered that that's incorrect. |
| 15:34.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:34.11 | [TK]D-Fender | przerull: But Asterisk does not assume you want to give EVERY feature you may define to EVERY call. |
| 15:34.20 | [TK]D-Fender | przerull: So you clearly have to specify which |
| 15:34.58 | [TK]D-Fender | dan_j: It COULD be correct.. if you happen to configure all the bits that way..... |
| 15:35.34 | dan_j | I know. But if a user incorrectly changes their config, then its a problem. |
| 15:35.47 | dan_j | Do I have to extract the peer name from CHANNEL(name)? |
| 15:36.02 | dan_j | Oh wait. DIdnt see peername |
| 15:36.39 | przerull | yeah well I set it in my dialplan and do a chandump to verify that it's set but when I enter the dtmf for the feature it doesn't run the application but passes the dtmf through |
| 15:36.42 | przerull | to the other channel |
| 15:38.23 | ipalmer | Can anyone tell me if AMI has a default buffer size when sending out events? |
| 15:40.42 | [TK]D-Fender | przerull: Go ahead and post us up something to look at... |
| 15:40.53 | przerull | thanks will do |
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| 15:53.40 | przerull | @D-Fender here's the link to the gist |
| 15:53.49 | przerull | https://gist.github.com/anonymous/f9d7ce76cf1aa55bec7c |
| 15:56.27 | [TK]D-Fender | Show the actual call |
| 15:56.30 | *** join/#asterisk rex___ (6ad03fd7@gateway/web/freenode/ip.106.208.63.215) |
| 15:57.42 | przerull | the full verbose output? |
| 15:57.46 | [TK]D-Fender | yes |
| 16:00.59 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 16:02.14 | rex___ | I am new to pbx admin. I have 4 store chain each having a PSTN number. when a customer calls the first 3 stores it gets forwarded to my pbx server where it is forwarded to appropriate inbound routes. but when the same customer calls the fourth store it doesnot reach my pbx but goes directly to pstn number. I presume telcos forward the dialed numbers to pbx. what am I doing wrong with the fourth setup, all being similiar except for |
| 16:02.22 | przerull | here's that verbose output |
| 16:02.26 | przerull | https://gist.github.com/anonymous/8b2b0743c814b08c9bfb |
| 16:02.38 | *** join/#asterisk _omer (~omer@184.175.79.212) |
| 16:03.00 | _omer | is there any replacement of "AgentCallBackLogin()" ? |
| 16:04.07 | przerull | the way the two calls are bridged is we do an originate in the ami script. when the far leg answers then it issues the bridge application pulling in the channel upon which we set the DYNAMIC_FEATURES |
| 16:07.20 | przerull | @rex your comment got cut off after the "all being similar except for" |
| 16:08.30 | [TK]D-Fender | _omer: "AddQueueMember" |
| 16:11.33 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 16:12.11 | _omer | [TK]D-Fender: I want to use Hot-Desking ... If agent having extension 100 wants to login from SIP Peer 200 then it means actual 100 peer is offline so other agents cannot transfer calls to him.....may be I am not delivering my confusion properly. I can try to explain it again.... |
| 16:12.34 | _omer | this is the issue if I use AddQueueMember |
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| 16:28.50 | przerull | @D-Fender, do you think the call to bridge that is calling my DYNAMIC_FEATURES |
| 16:30.01 | [TK]D-Fender | Was loking for something to parse out the ansi colour codign crap in there first... |
| 16:31.04 | przerull | oh yeah. I wish asterisk didn't do that |
| 16:31.17 | przerull | I just piped asterisk -r into tee |
| 16:31.30 | [TK]D-Fender | gah |
| 16:31.36 | [TK]D-Fender | that's why I just copy/paste from putty |
| 16:32.26 | WIMPy | had thought of actually converting the coloured output to html. |
| 16:32.58 | przerull | http://www.commandlinefu.com/commands/view/3584/remove-color-codes-special-characters-with-sed |
| 16:33.07 | przerull | that's a handy one for the bash profile |
| 16:33.07 | *** join/#asterisk cw1972 (~cw1972@host81-136-221-45.in-addr.btopenworld.com) |
| 16:33.16 | przerull | I'm going to post up another one with the codes removed |
| 16:34.28 | przerull | https://gist.github.com/anonymous/76dbebfa0b2abdf59318 |
| 16:36.55 | [TK]D-Fender | przerull: -- Executing [201135@inbound:14] MSet("SIP/carrier1_pipe-00000004", "__DYNAMIC_FEATURES=stop_recording#kill_recording") in new stack |
| 16:37.03 | [TK]D-Fender | przerull: Not sure about this use of "mset" |
| 16:37.30 | [TK]D-Fender | przerull: -- Executing [201135@inbound:15] MSet("SIP/carrier1_pipe-00000004", "features=""") in new stack <- also the double quotes = bleh |
| 16:37.44 | przerull | yeah that's ael's doing |
| 16:38.07 | [TK]D-Fender | przerull: Executing [201135@spark_outbound:42] Dial("Local/201135@spark_outbound-00000002;2", "SIP/carrier1_pipe/+19895720005,180,gb(set_hangup_handler^s^1(201135,outbound))") in new stack |
| 16:38.13 | [TK]D-Fender | przerull: THIS is the culprit |
| 16:38.29 | [TK]D-Fender | the actual dial is occurring in a nested local channel |
| 16:39.01 | [TK]D-Fender | SIP/carrier1_pipe-00000004 != Local/201135@spark_outbound-00000002;2 |
| 16:39.15 | przerull | hmmm but how does setting the hangup handler cause that? |
| 16:39.16 | *** join/#asterisk _omer (~omer@184.175.79.212) |
| 16:40.06 | przerull | lol I suppose I should provide my full extensions.ael |
| 16:40.21 | przerull | (i didn't earlier because It's a bit complicated and I thought it might be confusing) |
| 16:41.27 | przerull | https://gist.github.com/anonymous/41246e58f242f9434081 |
| 16:41.59 | [TK]D-Fender | Your dial occurs before that anyway.... |
| 16:42.07 | [TK]D-Fender | and that's already within a 2nd channel |
| 16:47.31 | przerull | so SIP/carrier1_pipe-00000004 is the channel name of the call where I dialed into my pbx. the ael call to agi://127.0.0.1:7771/set_features_finished?callid=201135&channel_id=90 originates |
| 16:47.41 | przerull | the following channels |
| 16:47.56 | przerull | Local/201135@spark_outbound-00000002;2 |
| 16:48.05 | przerull | and Local/201135@spark_outbound-00000002;1 |
| 16:48.14 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 16:48.22 | przerull | the spark_outbound channel actually does the dial |
| 16:48.50 | przerull | when the peer at the end of the dial answers the spark_inbound channel starts executing dialplan code |
| 16:49.04 | przerull | and executes the bridge |
| 16:49.06 | przerull | [201135@spark_inbound:206] Bridge("Local/201135@spark_outbound-00000002;1", "SIP/carrier1_pipe-00000004,F(inbound,201135,delay_finished)") |
| 16:49.18 | przerull | all of these channels have that hangup handler set |
| 16:49.52 | przerull | now what is interesting is that the dynamic features work for me when I make the DYNAMIC_FEATURES in the globals so all channels get it |
| 16:49.54 | _omer | [TK]D-Fender: Is this possible if a person can login to the queue from any table in a call center. AddQueueMember works but what happens to the internal calls? Actually person has not taken his original extension to that new table. AddQueueMember adds SIP Address (SIP/XXX) to the queue but basically agent's internal extension is offline.....you understand what I mean or Let me know and I can try to explain the confusion once again. |
| 16:50.17 | przerull | are you suggesting that I should be setting the dynamic feature on the spark_inbound channel instead? |
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| 16:56.07 | [TK]D-Fender | _omer: You're adding the wrong kind of CHANNEL. |
| 16:56.20 | [TK]D-Fender | _omer: You've forgotten what chan_agent is doing technically... |
| 16:57.18 | [TK]D-Fender | przerull: "originate" is a completely new thread with no inheritance. |
| 16:57.25 | xnaron | I have a tdm401p with a fxo and fxs. On my at&t analog phone on the fxs I notice that the line blinks and the caller ID for the extension and name for the extension appear on the phone a second or so after hanging up. What causes this? btw using elastix. |
| 16:57.37 | [TK]D-Fender | przerull: Looks at that dialplan you are executing for it./ |
| 16:58.40 | xnaron | (tdm410p) |
| 16:59.06 | przerull | lol that dialplan is a bit of a doozie isn't it |
| 17:01.37 | przerull | it appears as though there is some relationship between the dial and the dynamic features. I thought that it would be the inbound channel that needs to set the dynamic feature. |
| 17:02.00 | przerull | I've always had some difficulty in sharing information across channels in asterisk and tried getting around it with a bunch of agi and ami |
| 17:02.31 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 17:02.44 | przerull | I store all my state in the agi server and pass "commands" back to asteirsk by setting channel variables over agi |
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| 17:04.54 | [TK]D-Fender | You're acutally doing this in a NEW channel you originate. Nothing from the channel that starts this matters |
| 17:05.34 | [TK]D-Fender | You should be setting it IN the dialplan you are having that thing execute |
| 17:05.43 | [TK]D-Fender | You're doing it in the wrong place right now. |
| 17:07.16 | przerull | yeah I see now. I changed up my dialplan so I set the dynamic features inside that action=dial bit and it works now. I learned a lot about dynamic features today. Thanks a million D-Fender |
| 17:08.19 | przerull | I think I've seen you and Wimpy on here before. You guys are both super knowledgeable on asterisk. Are you guys Digium employees? |
| 17:08.24 | [TK]D-Fender | You're welcome. |
| 17:08.36 | [TK]D-Fender | Neither of us are. |
| 17:08.44 | [TK]D-Fender | Just long-time users |
| 17:09.50 | przerull | well thanks again. you've been a huge help. |
| 17:10.27 | _omer | [TK]D-Fender: I am really confused. I thought that if an agent can move to a new location then he can take his complete extension to the new seat, which means he is logged into the queue and he can also receive internal calls coming on his extension. |
| 17:11.06 | [TK]D-Fender | _omer: AgentCallBackLogin uses the DIALPLAN to find it's way to actually dialing a device |
| 17:11.23 | [TK]D-Fender | _omer: You should be using a LOCAL CHANNEL as your member to do the same with AQM |
| 17:13.04 | _omer | AQM? |
| 17:13.09 | [TK]D-Fender | AddQueueMember |
| 17:13.45 | _omer | so it means, Dialplan for internal calls need to be updated? |
| 17:14.00 | _omer | if I need a complete hot-desking feature |
| 17:15.03 | _omer | what about in case of "freepbx"? should I ask this question in freepbx room ? |
| 17:16.59 | [TK]D-Fender | is that what you're using? |
| 17:18.42 | _omer | in simple asterisk installation, Hot-Desking is not an issue. I can use AddQueueMember and use Local Channel and can also update Dialplan for internal calls so that an agent can move to any table in call center. He will be able to receive internal calls there and also log into the queue .... |
| 17:18.56 | _omer | but in case of FreePBX, I am confused about internal calls. |
| 17:19.59 | xnaron | Even if I go off hook and hang up...about 4 to 6 seconds later the phone flashes the line 2 light and the extension name and number appear on the display. If I eavesdrop on the line I hear the cid tones being sent from the fxs. I've also tried asterisk -vvvvvvvvvR to monitor and don't see anything after -- Hungup 'DAHDI/2-1' but about 4 to 6 seconds after it happens. I've googled the crap |
| 17:19.59 | xnaron | out of this trying to figure out why this happens... perhaps it is normal? |
| 17:20.56 | _omer | [TK]D-Fender : may be I've not delivered by confusion properly. |
| 17:21.00 | [TK]D-Fender | _omer: For freePBX I don't recall what kind of "agent" concept they implement. You'd have to ask in there... |
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| 17:23.57 | _omer | [TK]D-Fender : ok. thanks ... let me try :) |
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| 20:10.15 | rex_ | do we have a Windows GUI App that converts .wav, (eg, recorded with Microsoft Recorder) file to PCM Encoded file? |
| 20:14.35 | DivideBy0 | rex_: you have a lot of audacity to ask that in here :) http://audacity.sourceforge.net/ |
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| 20:18.05 | newtonr | DivideBy0, :| |
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| 20:23.06 | DivideBy0 | that's not funny? |
| 20:23.16 | DivideBy0 | sorry. it's been a long week |
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| 20:26.23 | newtonr | DivideBy0, okay I actually laughed at it. I do enjoy corny jokes. |
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| 20:27.44 | rex_ | well said |
| 20:27.46 | DivideBy0 | lemme tell you, it was hilarous when I typed it |
| 20:28.31 | DivideBy0 | rex_: I'm 99% sure you can actually do the PCM encoded directly from microsoft sound recorder. (except the windows 8 version, I couldn't find it in there) |
| 20:29.08 | rex_ | I know but I want a playback app for pcm |
| 20:29.24 | rex_ | i dont know whether it is possible |
| 20:30.43 | rex_ | because if I PCM Encoded, 16 Bits, at 8000Hz and reconvert back to wav file, the resultant file runs like speed train |
| 20:31.01 | rex_ | fast speech. is it normal? |
| 20:31.12 | DivideBy0 | audacity is File -> Import -> Raw Data, then you can listen/edit/slow down |
| 20:31.24 | DivideBy0 | but something else probably went wrong with the conversion or recording |
| 20:32.32 | rex_ | pbx says to encode at 8000Hz, when the .raw file is uploaded will the pbx play at the same rate of orig file? |
| 20:33.30 | DivideBy0 | I personally just save as a .wav, with pcm encoding inside it, not a ".raw" |
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| 21:02.36 | g-maurizi | I really need help. First time putting together a production asterisk/freepbx environment -- Grandstream GXP2160 phones (using Public Mode) which presents a username/password prompt on the screen -- phone user has to enter in extension # as username sip secret as password -- at the login screen the phone is NOT sip registered until the user logs in... I've got everything set up correctly in |
| 21:02.36 | g-maurizi | freepbx for Presence/BLF to work, and it does, but the BLF/MPK LED's on the phone stay green when the extension is not SIP registered. I've been racking my brain with this for 3 days now.DND turns the BLF red, they blink red when ringing, solid red when in call, green when available -- they just stay green when the extension is unregistered/powered off. |
| 21:03.55 | [TK]D-Fender | Then that is a phone issue |
| 21:04.26 | [TK]D-Fender | They should properly unsubscribe and not let an old status linger but they don't |
| 21:04.40 | [TK]D-Fender | You'll have to check Grandstream's resources for this |
| 21:04.48 | g-maurizi | How can I check the SIP register status of the extension to the moment? |
| 21:05.05 | g-maurizi | I see core show hints, but is there a more straight forward way ? |
| 21:05.20 | robmal | sip show subscriptions |
| 21:05.25 | g-maurizi | ty |
| 21:05.29 | [TK]D-Fender | Perhaps using iSymphony/FOP |
| 21:06.16 | [TK]D-Fender | robmal: That doesn't show who's online... that shows who is being WATCHED |
| 21:07.20 | robmal | Meh. "Try: 'sip show peers' and then 'sip show subscriptions'" - better? |
| 21:08.08 | g-maurizi | one recently logged out extension still shows idle in sip show subscriptions, but none of the other extensions are in that list, and they are solid green on the BLF's |
| 21:08.55 | robmal | So you suck at BLFing ;-) |
| 21:09.07 | g-maurizi | an interesting footnote, in my experimenting, throwing freepbx into user & device mode, and then logging off with the feature code, made the light shut off like I want it too, but only when in device & user mode, and using the log off feature code. |
| 21:09.57 | g-maurizi | I'm still learning. :) |
| 21:10.46 | robmal | Try to configure one phone via its web interface to subscribe to another ext and find what options are you missing |
| 21:13.01 | g-maurizi | robmal -- I'm not sure I understand. I have the phones in "public mode" so you actually subscribe to the extension using a username/password prompt, and logout when done, it's like a ghetto hotdesk mode on the grandstreams. |
| 21:13.43 | g-maurizi | The webUI shows "unregistered" and as far as I can see in asterisk/freepbx the extensions are not registered. |
| 21:14.01 | robmal | core set verbose 10 and try to figure out why ;-) |
| 21:14.24 | g-maurizi | lets see what that does. :) |
| 21:16.38 | g-maurizi | I go to log out from public mode on the grandstream, and see "sip 101 unregistered", log in again and see "unregistered sip 101" , i log back in at the public mode UI on the grandstream and see "registered sip 101" |
| 21:16.54 | g-maurizi | sorry, log in shows *registered sip 101 |
| 21:17.50 | g-maurizi | it looks legit. So the phones registering correctly when logging in/out. asterisk/freepbx is either reporting unregistered extensions as idle for some reason, or the BLF keys on this thing are just screwy. |
| 21:19.49 | g-maurizi | I tried changing the BLF to monitor a non existent extension and it shuts off the lamp. So some how the BLF key on this thing is detecting unregistered extensions as valid. |
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| 21:22.17 | g-maurizi | Is there a way that the BLF on this thing can tell that an extension is valid, if it isn't registered? this just strikes me as odd. Maybe more of a grandstream problem and less of a me problem. |
| 21:22.36 | [TK]D-Fender | I'm betting it's more like "not bothering to update them at all" |
| 21:23.10 | g-maurizi | On my end? I've upgraded firmware, and downgraded again just to verify the upgrade didn't bust the bLF lamps |
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| 21:25.21 | [TK]D-Fender | Not saying it's your fault.... |
| 21:25.24 | [TK]D-Fender | ~gs |
| 21:25.24 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:25.30 | [TK]D-Fender | ~grandstream |
| 21:25.30 | infobot | well, grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 21:26.44 | g-maurizi | These have been pretty nice/flexible so far.. to be honest, though complicated and not really approachable for a first timer. This BLF thing is the first thing that's got me beat. |
| 21:27.21 | g-maurizi | That and the instant message/xmpp is damn near impossible |
| 21:27.27 | [TK]D-Fender | Big catch is that you are using a hop on/off feature on the phone to act like hot-desking |
| 21:27.43 | [TK]D-Fender | Which is somehting virtually no-one does from a phone itself on that level |
| 21:28.02 | [TK]D-Fender | Those who do this on a "logical" leve tend not to get screwed so much. |
| 21:28.10 | g-maurizi | I may just go back to device & user mode |
| 21:28.11 | [TK]D-Fender | You just ran into some funny consequences |
| 21:28.14 | g-maurizi | yeah |
| 21:29.20 | robmal | Try this: https://www.grandstream.com/products/gxp_series/gxp2000/documents/gxp2000_interop_asterisk_blf.pdf |
| 21:29.42 | g-maurizi | ty. going to rtfm now. :) |
| 21:29.43 | [TK]D-Fender | That's not the issue |
| 21:29.50 | [TK]D-Fender | They LINGER after the user logs off... |
| 21:30.10 | [TK]D-Fender | And on that note... heading home... |
| 21:30.13 | [TK]D-Fender | BBIAB |
| 21:30.30 | g-maurizi | They don't linger... asterisk core sees the sip unregister event the second log-off is pressed. :) the phones just stupid |
| 21:34.28 | g-maurizi | robmal. ty. So I've limited the issue down I guess. It's for sure the GXP2160 is showing solid green if the extension is 'valid' regardless of its sip register state.. I just don't know how to find out why and fix it. ;oX |
| 21:35.30 | g-maurizi | Sigh, and I was really hoping it was some kind of polling interval or something. |
| 21:35.48 | robmal | Call the extension you're monitoring from another ext, the led will blink. |
| 21:36.05 | g-maurizi | They do. |
| 21:36.46 | g-maurizi | All other BLF functionality works as desired, ringing blinks red, on call solid red, available solid green, DND solid red. -- unregistered is just solid green. |
| 21:37.33 | robmal | So? |
| 21:37.59 | robmal | Print a sticker saying 'Green meens good' and carry on ;-) |
| 21:38.32 | g-maurizi | but green doesn't mean good if green means = offline and available at the same time! |
| 21:39.24 | g-maurizi | I need that light to stfu when sip unregister happens :oP |
| 21:39.25 | robmal | But BLF means BusyLineFunction. It's not busy when offline or available ;-) |
| 21:40.01 | g-maurizi | that's technically true. |
| 21:40.35 | robmal | So: 'Green means good' |
| 21:41.16 | g-maurizi | so the BLF shouldn't turn off, when the extension is offline? I'm honestly asking. |
| 21:41.37 | g-maurizi | Is this normal behaviour on alot of hardphones? |
| 21:43.01 | robmal | http://www.voip-info.org/wiki/view/Asterisk+presence says 'Extension and device state combined, no user state!' |
| 21:43.21 | g-maurizi | the BLF on these guys has alot of different settings, BLF, presence watcher, transfer, call park, record, intercom, dial DTMF.. ironically when I set it to "presence watcher" it stays solid green no matter what -- ringing, DND, doesn't trigger it. I have to set it to BLF or 'call park' to make extension presense seem to work. |
| 21:43.41 | WIMPy | g-maurizi: No. Usually it's off when available and on in any other situation. But every phone is different there. |
| 21:43.59 | g-maurizi | robmal - I understand this, but the device and extension being offline and unreachable is an extension AND device state. lol. |
| 21:44.04 | robmal | Yes. That's what separates good phones from the cheap ones ;-) |
| 21:44.10 | g-maurizi | damn |
| 21:44.13 | g-maurizi | that's the rub |
| 21:44.46 | WIMPy | On some phones you can even configure how to light the buttons in what situation. |
| 21:44.57 | g-maurizi | that would be nice. |
| 21:45.24 | WIMPy | But you still might not have all states available you're interested in. |
| 21:46.09 | g-maurizi | Is there some trick to figure out what device hints the different BLF modes on this particular phone listen to, for what lamp state? |
| 21:46.31 | WIMPy | Try it out. |
| 21:47.43 | g-maurizi | I have, I meant, would there be an official list some place, or could I maybe find that buried in the xml config? Alot of the "Multi Purpose Key" modes do nothing / do not react to SIP register/dnd/presence. |
| 21:48.22 | robmal | You should call Grandstream support and complain ;-) |
| 21:48.53 | WIMPy | See the manual. If tey want to tell you that where to expect it. |
| 21:48.58 | g-maurizi | I'm not usually an ask for help kinda guy, I'm generally a rtfm kinda guy. |
| 21:49.40 | robmal | I hope you never run into Unify Openstage series ;-) |
| 21:50.26 | WIMPy | No manual? |
| 21:50.36 | g-maurizi | Can I ask, what is different between *11/*12 user log on / log off feature codes when freepbx is in device & user mode, versus sip register/unregister when in extension(normal) mode? |
| 21:50.57 | g-maurizi | would the hints differ? |
| 21:51.05 | WIMPy | Yes. In #freepbx. |
| 21:51.28 | g-maurizi | They are lazy |
| 21:51.29 | robmal | It's a Siemens spinoff for UC. The docs for provisioning guide are 300+ pages, but there is (almost) no way to connect this junk to asterisk. |
| 21:51.42 | g-maurizi | ah |
| 21:52.19 | WIMPy | Siemens doesn't want to be mentioned with anythgn telephony sinve last year. |
| 21:52.55 | g-maurizi | Guess I could clone this container switch back to device & user mode and see whats different about user log in/off versus sip register. |
| 21:53.12 | WIMPy | I've only come across the TDM version. Looks nice/important/expensive. Sells great for that reason, but I didn't really like the UI. |
| 21:53.46 | g-maurizi | I'm not usually a UI person either and im really regretting that I went with a Web UI for my first VoIP deployment |
| 21:54.11 | robmal | How many extensions? |
| 21:54.31 | g-maurizi | the amount of time I've put into learning the quirks of freepbx would have been much better spent understanding asterisk |
| 21:54.34 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 21:54.36 | WIMPy | It's inconsistant. Sometimes you use the cursorwheel as cursor keys, sometime as a dial. |
| 21:55.00 | WIMPy | g-maurizi: That's what you would have got told in here. |
| 21:55.05 | g-maurizi | just 10 until we add the satellite offices |
| 21:56.12 | g-maurizi | I have a feeling grandstream made some funky tweaks to some asterisk core stuff in regards to some of the BLF functionality, they released their own $300 UCM device using asterisk/custom version of elastix... |
| 21:56.18 | robmal | Then the fun will start ;-) So, how many in total? |
| 21:56.55 | g-maurizi | something like 60 across 5 sites |
| 21:57.28 | robmal | Each site has its own dhcp server? |
| 21:57.40 | g-maurizi | dhcp forwarders, l2tp/ipsec |
| 21:57.52 | robmal | Have fun with that :-) |
| 21:58.00 | g-maurizi | I sure won't. |
| 21:58.08 | g-maurizi | sadly I get paid for this |
| 21:58.14 | g-maurizi | I don't do it for fun |
| 21:58.18 | robmal | By the hour, i hope? ;-) |
| 21:58.40 | g-maurizi | Me too |
| 21:59.18 | robmal | Well, it's going to be fun if you don't change it. |
| 21:59.59 | g-maurizi | I'm largely considering forsaking my roots and just buying their crappy little UCM just to make this BLF light stfu |
| 22:00.32 | robmal | The force weak in you is. |
| 22:00.43 | g-maurizi | sleep litle in me is |
| 22:01.09 | g-maurizi | im underpaid and overworked |
| 22:01.41 | WIMPy | Then tell them they get what they pay for :-) |
| 22:01.46 | robmal | A word of wisdom: if you didn't pick the phones - document every problem you've encountered with them and send it to the person responsible with proper CC. |
| 22:02.31 | g-maurizi | robmal: the phones picked us... they wanted alot of BLF keys and lines, and they didn't want to spend over $80 per unit |
| 22:03.09 | robmal | So now they have to pay for your extra hours. That's life ;-) |
| 22:03.13 | g-maurizi | yep |
| 22:07.00 | g-maurizi | alright, im going to go google helplessly for a while. thanks for the help! |
| 22:07.17 | robmal | Good luck. |
| 22:07.23 | g-maurizi | Ty. :) |
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| 22:33.30 | g-maurizi | So guess what. I found the solution finally: I have compared the 1.4 code to the old 1.2 and in line 7157 of chan_sip.c (case AST_EXTENSION_UNAVAILABLE) i have changed from: statestring "terminated" to statestring "confirmed" ... it seems that the gxp2160 wants statestring 'confirmed' to make the lamp stfu when devices fail to notify or sip unregister... |
| 22:34.24 | g-maurizi | looks like I'm going to have to patch chan_sip.c |
| 22:34.47 | robmal | Congrats ;-) |
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