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| 01:21.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.3 (2017/12/01), Standard: 15.1.3 (2017/12/01); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 10:25.28 | bittis | random question, how do i get the call id variable (C00000000) ? |
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| 11:57.58 | bllackjack | Can anyone help me out, I need to change the path where the recording (.wav) file is saved. |
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| 12:58.48 | mahafyi | bllackjack , you can change it in the dialplan entry, by default it uses the entry in asterisk.conf |
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| 13:28.52 | dnit | Hi when I load musiconhold module I am unable to whisper to agents who are idle ( using Chanpspy). |
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| 14:14.04 | bittis | hey guys, if anyone is in, trying to call a script with mixmonitor, and where command goes i want to have a bash script, setting the path there along with ^filename doesn't work |
| 14:14.07 | bittis | any ideas? |
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| 14:47.07 | sibiria | bittis: is the script's permissions set to executable, and does it have a proper shebang? |
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| 14:56.36 | cervajs2 | i cant find alembic change for pjsip webrtc=yes ? this is not supported in pjsip realtime? tnx |
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| 15:10.39 | seanbright | cervajs2: it's there: https://github.com/asterisk/asterisk/blob/15/contrib/ast-db-manage/config/versions/44ccced114ce_add_webrtc.py |
| 15:10.47 | seanbright | it's only available in asterisk 15+ |
| 15:10.50 | cervajs2 | is it for ast13? |
| 15:11.01 | seanbright | no, that option isn't available in asterisk 13 |
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| 16:21.43 | avb | hey guys, anyone had this situation when an endpoint is sending 0.0.0.0 to hold itself? |
| 16:21.54 | avb | having hard time asterisk to do that |
| 16:23.18 | avb | reinvite is on https://pastebin.com/a2bqeRdc |
| 16:23.37 | avb | what a mess |
| 16:23.44 | avb | c=IN IP4 0.0.0.0 |
| 16:23.49 | avb | a=sendrecv |
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| 16:42.06 | Samot | That's a hold |
| 16:42.10 | Samot | Or a return from hold. |
| 16:42.26 | Samot | That is not uncommon for holds. |
| 16:42.42 | Samot | The a=sendrecv is for putting a call on hold. |
| 16:43.03 | Samot | That's usually a=sendonly or something close to that |
| 16:49.39 | avb | yeh, their system is fcked up |
| 16:50.43 | Samot | Wait. |
| 16:50.48 | Samot | That's not what I said at all. |
| 16:51.03 | Samot | 11:23:45 AM A<avb> c=IN IP4 0.0.0.0 <-- This is common |
| 16:51.30 | Samot | 11:23:50 AM A<avb> a=sendrecv <-- This is generally a=sendonly from the device when putting a call on hold. |
| 16:51.54 | Samot | Now that's not to say what is there is wrong. |
| 16:52.22 | Samot | That's telling the other side to basically send the media from the remote endpoint to a "blackhole" |
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| 16:52.40 | Samot | So if they talk over the hold music, it never makes it back to the system. |
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| 16:55.41 | avb | Samot: and what whey are expecting with them is to mute a leg who is emmiting this message |
| 16:55.51 | avb | Samot: and what whey are expecting with that is to mute a leg who is emmiting this message |
| 16:56.19 | Samot | I put you on hold... |
| 16:56.19 | avb | it should be a=inactive for that i belive |
| 16:56.22 | Samot | No. |
| 16:56.24 | Samot | Not at all |
| 16:56.28 | avb | hu |
| 16:56.32 | Samot | That's not a valid option. |
| 16:56.34 | avb | you are calling me |
| 16:56.42 | avb | you are sending me this message |
| 16:56.53 | avb | in result, I am on hold |
| 16:57.05 | avb | while its expected to put you on hold |
| 16:57.24 | Samot | Let's try this again. |
| 16:57.34 | Samot | I answered your question about the SDP body |
| 16:57.36 | Samot | Now. |
| 16:57.46 | Samot | Are you having issues with putting/taking a call on hold? |
| 16:58.03 | avb | wrong leg is becoming on hold :) |
| 16:58.13 | Samot | What do you mean the "wrong leg"? |
| 16:58.14 | avb | thats what is the problem |
| 16:58.24 | avb | avb> you are calling me |
| 16:58.24 | avb | avb> you are sending me this message |
| 16:58.24 | avb | avb> in result, I am on hold |
| 16:58.24 | avb | avb> while its expected to put you on hold |
| 16:58.33 | Samot | No. |
| 16:58.43 | Samot | If I'm putting you on hold, that's what I would send to you |
| 16:58.49 | avb | right |
| 16:58.53 | Samot | To tell you that you can't send media back to me. |
| 16:58.59 | avb | right |
| 16:59.01 | Samot | But I'm sending media to you aka the hold music |
| 16:59.06 | Samot | So you should be on hold, not me. |
| 16:59.11 | avb | but what they want to achive with this message :) |
| 16:59.23 | Samot | The caller wants to hold themselves? |
| 16:59.27 | avb | right |
| 16:59.31 | Samot | Uhm. |
| 16:59.52 | Samot | Why? |
| 17:00.04 | avb | they have a GUI to control the call flow |
| 17:00.13 | avb | thats what hold button generates |
| 17:00.16 | Samot | The person who puts the call on hold wants to hear the hold music? |
| 17:00.55 | avb | in reality other leg is outdialed out number |
| 17:01.02 | avb | autodialed* |
| 17:01.04 | Samot | OK. |
| 17:01.12 | avb | so they want to put this leg on hold |
| 17:01.27 | Samot | So you are Originating a call from Asterisk.. |
| 17:01.34 | avb | its complicated |
| 17:01.36 | avb | not from asterisk |
| 17:01.44 | Samot | Well you need to explain. |
| 17:01.51 | avb | for asterisk it looks like a regular inbound call |
| 17:01.54 | Samot | Or you can't really get help. |
| 17:02.00 | Samot | Stop. |
| 17:02.08 | Samot | Where are you Originating this call FROM? |
| 17:02.13 | avb | on another platofmr |
| 17:02.16 | avb | on another platoform |
| 17:02.17 | Samot | How does this call GET to Asterisk? |
| 17:02.22 | avb | via sip trunk |
| 17:02.37 | Samot | OK. |
| 17:02.48 | Samot | So it's making a normal SIP call to Asterisk. |
| 17:03.09 | avb | right |
| 17:03.09 | Samot | And Asterisk has a peer for this platform. |
| 17:03.14 | avb | right |
| 17:03.45 | Samot | So Asterisk answers the call or does it just Dial() straight to a device? |
| 17:03.50 | avb | right |
| 17:03.53 | avb | answers |
| 17:04.02 | avb | i have tried staright dial too |
| 17:04.19 | Samot | When do you want this call "held"? |
| 17:05.04 | avb | when im receiving thi SDP, an expected behavoir is yes to hold this incoming leg |
| 17:05.12 | avb | when im receiving this SDP, an expected behavoir is yes to hold this incoming leg |
| 17:05.16 | Samot | That's not the answer. |
| 17:05.23 | Samot | When this call hits Asterisk.. |
| 17:05.33 | Samot | And enters the dialplan, when you do you want hold the call? |
| 17:05.36 | Samot | Immediately? |
| 17:05.39 | avb | no |
| 17:05.42 | Samot | After a recording is played back? |
| 17:05.46 | avb | no |
| 17:06.04 | avb | call gets answered by a an agent |
| 17:06.20 | avb | they are talking and then agent needs to put caller on hold |
| 17:06.29 | avb | he press a button in the gui |
| 17:06.46 | avb | im getting this SDP coming from trunk |
| 17:07.00 | avb | agents becomes muted |
| 17:07.11 | Samot | OK. |
| 17:07.18 | Samot | Asterisk doesn't "hold" calls. |
| 17:07.19 | avb | instead of leg from which message came from |
| 17:07.32 | Samot | Hold is method of SDP/media flow. |
| 17:07.37 | Samot | You need to PARK the call. |
| 17:07.47 | avb | i cant |
| 17:07.51 | Samot | OK |
| 17:07.59 | Samot | HOLD is a method of SDP/media flow. |
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| 17:08.04 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 17:08.05 | Samot | The channels are still bridged. |
| 17:08.06 | avb | im RECEIVING HOLD SDP |
| 17:08.09 | Samot | They are still connected. |
| 17:08.11 | Samot | STOP |
| 17:08.18 | Samot | Asterisk CANNOT HOLD CALLS |
| 17:08.33 | Samot | HOLD is not a status |
| 17:08.36 | Samot | of a channel |
| 17:08.41 | Samot | The channel is still bridged. |
| 17:08.45 | avb | and what is res_musiconhold |
| 17:08.48 | Samot | It changes the MEDIA FLOW |
| 17:09.01 | Samot | And triggers the media to hold the call |
| 17:09.08 | Samot | Asterisk doesn't trigger that |
| 17:09.14 | Samot | The endpoint does |
| 17:09.33 | Samot | If Asterisk answers an inbound call... and you want to, via AMI, "hold" that call... |
| 17:09.37 | Samot | you have to PARK it. |
| 17:09.40 | avb | so the endpoint is downloading mp3 from my asterisk |
| 17:09.47 | avb | and plays it to a caller |
| 17:09.51 | Samot | No. |
| 17:10.00 | Samot | With AMI you have to PARK the call |
| 17:10.03 | Samot | Period. |
| 17:10.06 | avb | Samot |
| 17:10.20 | avb | you help is very long and as always useless |
| 17:10.21 | avb | thank you |
| 17:10.28 | avb | you are a life saver |
| 17:10.36 | avb | :) |
| 17:10.41 | avb | i love talking to you |
| 17:10.52 | avb | https://raw.githubusercontent.com/asterisk/asterisk/01a8d9844b50e98af9974a2367f7c6d6fa69774f/channels/chan_sip.c |
| 17:11.36 | Samot | jcolpAsterisk DeveloperFeb 26 |
| 17:11.37 | Samot | You generally can't force a SIP phone to do something, unless it provides its own interface. You can only hope to replicate similar (may not be exact) behavior in Asterisk. |
| 17:11.50 | avb | https://pastebin.com/fQxHT1Y7 |
| 17:11.51 | Samot | You can only hope to replicate similar (may not be exact) behavior in Asterisk. <-- Note that |
| 17:12.03 | avb | get out of the bushes |
| 17:12.09 | avb | asterisk can do moh for years |
| 17:12.14 | Samot | ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) { |
| 17:12.16 | avb | without parking the call |
| 17:12.16 | Samot | ^^^ |
| 17:12.17 | Samot | SEE THAT |
| 17:12.21 | avb | yes :) |
| 17:12.25 | Samot | That's looking for SDP DETAILS |
| 17:12.30 | Samot | That the DEVICE SENDS |
| 17:12.52 | avb | i know that :) |
| 17:12.58 | avb | in my case its a trunk |
| 17:13.05 | Samot | So how do you, from the GUI, make the device send that? |
| 17:13.08 | Samot | To Asterisk? |
| 17:13.19 | Samot | From this platform GUI? |
| 17:13.24 | Samot | That the Agent is using |
| 17:13.35 | Samot | To somehow interject on his device? |
| 17:13.36 | avb | from this other platofrm which i have a trunk to |
| 17:13.48 | avb | and yes, on that side they generate it via some kind of an AMI |
| 17:13.59 | Samot | Which goes to Asterisk |
| 17:14.05 | Samot | And has to know which channel to hit |
| 17:14.06 | avb | and this event comes from the trunk to asterisk |
| 17:14.13 | Samot | And it sounds like they are hitting the wrong channel |
| 17:14.23 | avb | correct |
| 17:14.26 | Samot | Or sending the information in a way that makes Asterisk use the wrong channel |
| 17:14.33 | Samot | Because you are trying to replicate something from the DEVICE |
| 17:14.36 | Samot | That is not the DEVICE |
| 17:14.49 | Samot | 12:11:52 PM S<Samot> You can only hope to replicate similar (may not be exact) behavior in Asterisk. <-- Note that |
| 17:15.03 | avb | you got it |
| 17:15.04 | Samot | ^^ STraight from Joshua, an Asterisk Developers |
| 17:15.10 | Samot | So to REPLICATE this |
| 17:15.17 | Samot | You use AMI to PARK the call. |
| 17:15.50 | avb | damn, you are genious |
| 17:15.55 | Samot | And then AMI to UNPARK the call via the GUI when the Agent wants to take them off the call |
| 17:15.56 | avb | i can catch moh even |
| 17:16.13 | Samot | 12:10:21 PM A<avb> you help is very long and as always useless <-- Huh |
| 17:16.19 | avb | and then start MusicOnHold/Park bridged channel |
| 17:16.31 | avb | and on Unhold, do the same in reverse |
| 17:16.38 | avb | ugly, but will work :) |
| 17:16.45 | Samot | 12:15:51 PM A<avb> damn, you are genious <-- Wow |
| 17:16.47 | avb | thanks for a hint |
| 17:16.54 | avb | :)) |
| 17:17.00 | Samot | So I went from being a long winded useless guy to a genius. |
| 17:17.10 | Samot | In five minutes. |
| 17:17.11 | avb | Samot: you know i still love you :) |
| 17:17.18 | avb | from love to hate its only one step |
| 17:17.19 | avb | lol |
| 17:17.50 | avb | thank mate |
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| 19:00.46 | jjrh | Do folks here have any recommendations for software to do SIPREC? My googleing lead me to https://community.asterisk.org/t/asterisk-siprec/69032/4 which seems surprising to me |
| 19:02.14 | file | no change, noone has looked at the RFC, seen what needs to be done in Asterisk if anything |
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| 19:47.22 | jjrh | Thanks file |
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| 20:55.36 | dnit | Hi I am not able to whisper / barge on another agent when musiconhold is loaded. |
| 20:56.05 | Samot | Do you not get anything? |
| 20:56.09 | Samot | Or do you get MoH? |
| 20:57.42 | dnit | Not working logs : https://pastebin.com/sfdNtGBk , working logs : https://pastebin.com/SGh5FEZQ |
| 20:58.00 | dnit | Currently my moh is pointing to null. Will test with valid moh |
| 20:58.25 | Samot | So who are trying trying to spy on? |
| 20:59.00 | dnit | One agent spying on another. |
| 20:59.53 | Samot | OK |
| 21:00.07 | Samot | So if Agent A is on a call and puts it on hold... |
| 21:00.22 | Samot | Then Agent B wants to spy on Agent A... |
| 21:00.35 | Samot | The channel they are spying on is playing MoH |
| 21:00.57 | Samot | And has it's media set to sendonly or has the media IP set to 0.0.0.0 |
| 21:01.03 | Samot | So it doesn't receive media |
| 21:01.14 | Samot | So what are they spying on? |
| 21:01.21 | Samot | Who are they whispering to? |
| 21:01.55 | Samot | Because on the phone Agent A isn't on that line anymore. |
| 21:02.02 | dnit | If Agent A is waiting. and Agent B spies on agent A. B is able to hear A while A is not able to hear B ( while musiconhold module is loaded) |
| 21:02.09 | Samot | Right |
| 21:02.17 | Samot | Because when the phone sends a hold |
| 21:02.24 | Samot | It either sets the media to sendonly |
| 21:02.37 | Samot | Or sets the SDP media IP to 0.0.0.0 |
| 21:02.41 | dnit | But when I dont load musiconhold A is able to hear B. |
| 21:02.44 | Samot | So it doesn't receive media. |
| 21:02.49 | Samot | OK. |
| 21:05.32 | Samot | Oh. |
| 21:05.33 | Samot | HAH |
| 21:05.34 | Samot | Duh |
| 21:05.43 | Samot | Spy uses a conference bridge. |
| 21:05.47 | Samot | So yeah.. |
| 21:05.57 | Samot | With MoH is considers Agent A "talking" |
| 21:06.25 | Samot | So could be how it's handling that. |
| 21:06.50 | Samot | And the fact that recordings have no silence or pauses in them |
| 21:08.55 | dnit | When I have mode=quitemp3 then Agent A is not able to hear Agent B. |
| 21:09.17 | dnit | While when I have mode=file and valid directory Agent A is able to hear Agent B along with moh |
| 21:23.40 | dnit | Samot: Any specific reason behind this behaviour ? |
| 21:32.57 | *** join/#asterisk gtrmtx (~gtrmtx@74.197.222.10) |
| 21:33.46 | gtrmtx | so i know this is the room for asterisk, and believe me i would be using asterisk in this situation im having if i could, but by any chance does anyone in here have experience with Shoretel systems? |
| 21:39.23 | [TK]D-Fender | Always just start with your actual specific question. |
| 21:39.37 | gtrmtx | <PROTECTED> |
| 21:39.43 | gtrmtx | im monitoring the trunks and when they try to do it its not even hitting the trunks |
| 21:39.52 | gtrmtx | with** a caveat |
| 21:39.57 | gtrmtx | if they 'consult' the external user, aka put call on hold, call external number, and once it connects then transfer |
| 21:40.03 | gtrmtx | <PROTECTED> |
| 21:40.11 | gtrmtx | but blind transfers do not work at al |
| 21:40.15 | gtrmtx | all* |
| 21:41.57 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 21:42.29 | qakhan | is there any free Speech to talk engine for * |
| 21:45.45 | [TK]D-Fender | qakhan, talk IS speech |
| 21:48.05 | [TK]D-Fender | qakhan, And you're still (badly) aiming at questions that are easily and instantly answered by a Google search. |
| 21:55.44 | sibiria | qakhan: sphinx |
| 22:00.20 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:01.07 | Samot | gtrmtx: That's a Shoretel question. Blind transfers are something they need to handle. |
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| 22:22.17 | rrittgarn | anybody here work with the XML apps on Aastra/Mitel phones before? |
| 22:22.26 | rrittgarn | looking to see if its possible to control the MWI with an XML app |
| 22:23.22 | [TK]D-Fender | http://edocs.mitel.com/UG/EN/6800i%20Series/4.3%20SP1/58015044_XML%20API%20for%20Mitel%20SIP%20Phones%20Firmware%204.3.0%20SP1.pdf |
| 22:23.54 | [TK]D-Fender | 1st result from 5-word Google search |
| 22:25.33 | rrittgarn | apparently i failed at googling |
| 22:25.38 | rrittgarn | i appreciate the assistance |
| 22:28.06 | [TK]D-Fender | "mitel SIP XML programming guide" <- if that wasn't your search then you didn't really try aat all |
| 22:29.06 | rrittgarn | it wasn't it was aastra xml mwi |
| 22:29.19 | rrittgarn | and aastra xml administrator guide |
| 22:29.22 | rrittgarn | but again, thanks |
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| 22:44.09 | Samot | http://www.mitel.com/sites/default/files/41-001561-01_REV03_6800_Series_4.1.0_SP3_Admin_Guide.pdf |
| 22:44.29 | Samot | ^^ That was found with aastra xml administrator guide |
| 22:44.43 | Samot | A User can configure the MWI using the Mitel Web UI only. An Administrator can configure the |
| 22:44.43 | Samot | MWI on single or all lines using the configuration files or the Mitel Web UI |
| 22:44.51 | Samot | ^^ Which states that |
| 22:50.46 | rrittgarn | ok that makes sense |
| 22:50.58 | rrittgarn | and i see the stuff about explicit MWI URI, i'm just curious the syntax i guess |
| 22:52.06 | Samot | OK |
| 22:52.21 | Samot | SIP devices will use the host/OB proxy for things. |
| 22:52.47 | Samot | If you don't specify certain things they default to the SIP host. |
| 22:53.04 | Samot | Don't set a BLF/Attendant URI, uses the host |
| 22:53.21 | Samot | Don't set a Voicemail/MWI server, uses the host |
| 22:54.04 | rrittgarn | ok, so it's doing this in actual sip signalling and If i wanted to make MWI external of * i'd have to set up something like OpenSIPs to handle that? |
| 22:54.10 | Samot | Nothing ever states that the voicemail server lives on SIP server. |
| 22:54.17 | Samot | No. |
| 22:54.23 | Samot | It's having an external voicemail server. |
| 22:54.33 | Samot | SIP calls go to SIP server |
| 22:54.40 | Samot | Voicemail gets sent to voicemail server. |
| 22:54.47 | Samot | That's totally possible. |
| 22:54.57 | Samot | It is totally possible to have SIP services on one server |
| 22:55.02 | Samot | BLF services on another |
| 22:55.06 | Samot | and VM services on another. |
| 22:55.17 | Samot | Providing you with "one Voice service" |
| 22:55.39 | rrittgarn | yeah, which makes sense for a nice distributed environment. I'm still hung up on the mechanics of *how* the phone is doing it so i can figure out how to interface that into an additional system |
| 22:55.51 | Samot | The phone will have settings. |
| 22:56.04 | Samot | If the SIP proxy is 192.168.1.10 |
| 22:56.10 | Samot | And none of the other settings are set.. |
| 22:56.14 | rrittgarn | perfect world I was hoping to use the sip-notify portion to pull an XML to set the MWI on or off... but that's proving more difficult than I had hoped |
| 22:56.16 | Samot | They all default to 192.168.1.10 |
| 22:56.23 | Samot | No. |
| 22:56.29 | Samot | That's now how that works. |
| 22:56.39 | Samot | MWI is a SUBSCRIBE |
| 22:56.59 | Samot | You are subscribing to the MWI to get a NOTIFY about changes to it |
| 22:57.05 | rrittgarn | ok so its sip signalling (UDP 5060 traffic that adheres to RFC 3261) |
| 22:57.16 | Samot | They all use 5060 |
| 22:57.20 | Samot | Or the port that is needed |
| 22:57.45 | Samot | MWI is just another form of BLF |
| 22:58.04 | rrittgarn | except on Mitels i can control BLFs with XML messages |
| 22:58.08 | rrittgarn | as well as subscribes |
| 22:58.16 | Samot | I'm giving a comparison |
| 22:58.27 | Samot | When the voicemail box receives a NEW message.. |
| 22:58.31 | Samot | It sends a NOTIFY |
| 22:58.40 | rrittgarn | but MWI is 'special' in this case - at least I haven't found a nice call in the XML for it |
| 22:58.42 | Samot | You have to be subscribed to receive it |
| 22:58.50 | Samot | There isn't. |
| 22:58.51 | rrittgarn | yeah, i'm ok with going to a polled interval for this particular project |
| 22:59.01 | rrittgarn | i figured as much after reading through things |
| 22:59.37 | Samot | You want to turn MWI on and off.. |
| 22:59.38 | rrittgarn | Trying to offload MWI to something outside the asterisk box that the phones are registered to because they aren't keeping up and apparently that stupid blinking light is important to this customer |
| 22:59.45 | Samot | That has to be done with the config file. |
| 22:59.51 | Samot | And the phone needs to pull the new config |
| 23:00.01 | Samot | Well that's not Asterisk. |
| 23:00.15 | Samot | I just dealt with this at an office. |
| 23:00.28 | Samot | Put in new voice services for them.. |
| 23:00.44 | Samot | They wanted 25 BLF's per phone. |
| 23:00.51 | Samot | They only had 7 before. |
| 23:01.00 | Samot | Once we added all those BLFs.... |
| 23:01.10 | Samot | Their NAT went nuts. |
| 23:01.20 | Samot | Because the PBX is in the cloud. |
| 23:01.30 | Samot | And their router wasn't sufficent. |
| 23:01.31 | rrittgarn | interesting, because i would expect those messages to just traverse the same PAT @ the firewall |
| 23:01.37 | Samot | And their router wasn't sufficient. |
| 23:01.39 | Samot | Right |
| 23:01.48 | Samot | But it is creating MORE messages. |
| 23:01.51 | Samot | That have to NAT |
| 23:02.00 | Samot | If the router handles NAT crappy |
| 23:02.15 | Samot | I replaced their router with a real one. |
| 23:02.24 | Samot | An actual business/enterprise level router |
| 23:02.25 | rrittgarn | yeah, they are running a Checkpoint 5100 - enterprise grade |
| 23:02.27 | Samot | Problems went away |
| 23:02.37 | Samot | Well then, it could be a NAT issue. |
| 23:02.47 | Samot | Moving it to another server doesn't mean the NAT issues will go away |
| 23:02.54 | Samot | Doesn't mean that the MWI doesn't make it to them. |
| 23:02.54 | rrittgarn | what I'm seeing is * isn't actually updating the message counts correctly |
| 23:03.02 | Samot | How so? |
| 23:03.20 | Samot | Empty mailbox with 0 messages.. |
| 23:03.23 | Samot | Leave new message |
| 23:03.31 | Samot | What does the NOTIFY show? |
| 23:03.39 | rrittgarn | it increments properly on the new messages |
| 23:03.40 | Samot | Does it show 1/0? |
| 23:03.43 | Samot | OK. |
| 23:03.46 | rrittgarn | after deleting they dont go away |
| 23:03.51 | rrittgarn | LastMsgsSent : 1/0 |
| 23:03.53 | rrittgarn | when there are no messages |
| 23:03.59 | Samot | How are they deleted? |
| 23:04.08 | rrittgarn | you're about to throw your hands up |
| 23:04.12 | rrittgarn | It's real time |
| 23:04.20 | rrittgarn | DB entry is deleted |
| 23:04.22 | Samot | How do they delete the voicemail. |
| 23:04.24 | Samot | How do they delete the voicemail? |
| 23:04.35 | rrittgarn | DIal into VoicemailMain and hit 7 |
| 23:04.47 | Samot | And what is your pooling set to? |
| 23:04.49 | rrittgarn | OR they delete the row in the DB |
| 23:05.23 | Samot | What are the pool settings? |
| 23:05.31 | Samot | You need to have this set to yes |
| 23:05.39 | Samot | You are modifying things outside of app_voicemail |
| 23:05.58 | rrittgarn | in res_odbc? or in voicemail.conf ? |
| 23:06.26 | rrittgarn | i'm not familiar with that setting, and honestly am super appreciative that you didn't just throw your hands up like most others at this point... like seriously thank you for taking the time |
| 23:07.07 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/voicemail.conf.sample |
| 23:07.43 | Samot | Look at the poolmailboxes setting and the poolfeq setting |
| 23:07.54 | Samot | If that is set to no |
| 23:08.02 | rrittgarn | no, i had that one already |
| 23:08.07 | Samot | Then only interactions through app_voicemail will work. |
| 23:08.07 | rrittgarn | its set to yes and the default of 30 seconds |
| 23:08.26 | Samot | This is why I never store this crap in a DB |
| 23:08.40 | rrittgarn | yeah I'm working on getting it out of there, but it's what i've got for this project |
| 23:09.09 | rrittgarn | when you said pooling, i thought you were in reference to odbc pooling, which i have max_connections => 20 |
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