| 00:01.22 | BurmaSauce | Hi, any ideas why some audio files might sound broken? They're all OK on the command line, but for example if I dial *65 I get "your extension is one zero <robot voice>two</robot voice>" |
| 00:06.57 | cusco | is your system under stress? |
| 00:07.10 | cusco | are you running asterisk in the same machine as dialing? |
| 00:08.21 | WIMPy | was just about to ask if you wanted to know if the systyem on the phone is under stress. |
| 00:09.42 | BurmaSauce | It's a fairly low spec machine - actually it's a raspberry pi... but the system is otherwise unloaded |
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| 02:22.45 | cyphase | hey, i'm interested in setting up a pretty simple (i assume) asterisk installation for one phone line, then being able to integrate it with a custom web app. it's for a business, and the idea is to integrate call/voicemail data/recordings into a customer management interface. anyone have pointers as to how to talk to asterisk and get such data via api, or have asterisk push the data to the web app? |
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| 02:25.07 | [TK]D-Fender | cyphase: It's a question on when you want things to happen |
| 02:25.17 | [TK]D-Fender | cyphase: You can't talk about "integration" so loosely. |
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| 02:25.54 | [TK]D-Fender | cyphase: Try to give a more precise sample of what you wan tot do. |
| 02:26.19 | cyphase | [TK]D-Fender, i'm being intentionally loose as i don't know what the best way might be. that's why i gave some details on my use case |
| 02:26.38 | cyphase | sure |
| 02:26.48 | [TK]D-Fender | cyphase: You need to clarify the timing & flow |
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| 02:31.25 | cyphase | i want this business/customer management interface to be able to show information about calls and voicemails to and from customers. i would be storing all the information within the web app's database; i just need to know how to get the data out of asterisk. the data being, all calls (ingoing, outgoing, missed, answered), metadata such as time/length/etc, recordings of calls, etc |
| 02:31.28 | cyphase | sorry, brb |
| 02:32.23 | cyphase | (or not) back |
| 02:34.06 | [TK]D-Fender | That is CDR, and Asterisk can store it in a DB, or a usual flat text file |
| 02:34.10 | [TK]D-Fender | I'tts in the book |
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| 02:34.54 | [TK]D-Fender | As for voicemail.. those are typically just files as well. Now is what you are looking to do actually USING voicemail or were you trying to use that term while looking to implement just "some" way of recording a file? |
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| 02:41.28 | crumb | anyone with a callcentric account? |
| 02:41.29 | cyphase | [TK]D-Fender, i want to use voicemail, and have those files accessible in the web interface (specifically, accessible for me to add to the web interface). i also want all calls to be recorded as well |
| 02:43.55 | [TK]D-Fender | All doable |
| 02:44.02 | [TK]D-Fender | And other GUI's alrady offer this |
| 02:44.11 | crumb | i'm trying to figure out how to make did forward to multiple extensions using callcentric's webui |
| 02:44.11 | cyphase | well, yes |
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| 02:46.41 | cyphase | [TK]D-Fender, sure, i'm not just trying to re-implement a voicemail web interface, i want to integrate it with a custom web app, e.g. for showing past interactions with a customer, integrating the call data with other customer data, etc |
| 02:47.07 | [TK]D-Fender | Go for it. It's all jsut a bunch of dumb files. |
| 02:47.26 | [TK]D-Fender | There is no "API" for this |
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| 07:19.07 | hos7ein | hi |
| 07:21.20 | ChannelZ | ahoy |
| 07:22.21 | hos7ein | for create IAX Trunk, I First create IAX user Extension? |
| 07:27.26 | gerritfromsa | hos7ein, if you want it to work both way with 1 account you'll make it a friend account (type=friend) |
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| 07:51.57 | magespawn | good day all |
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| 07:54.58 | gerritfromsa | is away: Hmmmm ... |
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| 08:37.27 | _omer | Hello, I need expert suggestion: I have 2 channels/Calls in ConfBridge ... I want to hangup both when anyone of them hangsup first. Should I use SoftHangup Command or any other suggestion? |
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| 11:58.43 | danfromuk | Hi. I'm just trying to make my dialplans a bit more robust. Does CONGESTION on an outbound sip trunk call always mean that there is a fault with the sip trunk/provider? |
| 11:58.51 | danfromuk | Or can you get congestion for another reason? |
| 11:59.13 | danfromuk | I'm trying to work out when to try a backup sip trunk, and when the call is successful. |
| 11:59.58 | WIMPy | CONGESTION = "general call failure" |
| 12:00.13 | WIMPy | It can mean round about anything. |
| 12:02.04 | danfromuk | Is it fair to say that if I get congestion, i should try a different route and see if that also gets congestion? |
| 12:02.20 | Greenlight | You should probably look at the SIP error code |
| 12:02.36 | danfromuk | Is that available in the dialplan? |
| 12:02.37 | Greenlight | CONGESTION is a general error and can mean a numbe rof things |
| 12:02.39 | Greenlight | Yes. |
| 12:02.42 | WIMPy | That MIGHT help. |
| 12:03.01 | danfromuk | Currently, the error code is 500 "Service Unavailable" as displayed in the CLI |
| 12:03.10 | WIMPy | Unfortunatly the SIP responses are often less helpfull. |
| 12:03.27 | Greenlight | 500 is usually terminal |
| 12:03.41 | WIMPy | The trouble is, as with Asterisk HANGUPCAUSE, that you don;t get any information about where it happened. |
| 12:03.52 | Greenlight | That's true |
| 12:03.58 | WIMPy | That often makes the cause rather useless. |
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| 12:08.14 | danfromuk | It seems complex to retrieve the SIP response code. For now, I'll stick with retrying a different trunk when CONGESTION is returned. |
| 12:08.39 | gerritfromsa | danfromuk, are you using the ${DIALSTATUS} to get the CONGESTION ? |
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| 12:32.27 | danfromuk | gerritfromsa: yes |
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| 13:01.41 | gerritfromsa | dan's gone |
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| 13:33.12 | str8uplinux | is there a way to set the caller id to the name of the persion you're dialing instead of just the extension for internal calls? |
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| 13:38.54 | Greenlight | You can set the callerid to what you like, https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID |
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| 13:42.37 | str8uplinux | Greenlight: I guess i shouldn't have said callerid, i just want MY phone (dialing party) to display the name of the person I am calling (dialed party). So if i dial extension 200 I want to tie a name to it so on the DIALING phone it displays something like "John Smith - 200" |
| 13:45.00 | Greenlight | str8uplinux: That would depend on your phone I'd imagine, and it would need to do some sort of lookup, rahter than anything on the Asterisk side. You *may* be able to use CONNECTEDLINE function I suppose: https://wiki.asterisk.org/wiki/display/AST/Function_CONNECTEDLINE |
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| 14:05.59 | str8uplinux | Greenlight: thanks, i will give that a look! |
| 14:06.41 | str8uplinux | I have a sql table that has all of our extensions mapped to a name, so i was thinking i will have to write a agi script to do all of the "magic" |
| 14:07.15 | WIMPy | What chenneltype are you using? |
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| 14:07.29 | WIMPy | SIP will do it automagically. |
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| 14:21.57 | Rico | hi there |
| 14:22.13 | Rico | I'm having the same problem than this one : https://issues.asterisk.org/jira/browse/ASTERISK-19181 |
| 14:22.14 | LieutPants | [ASTERISK-19181] [Status: Open] SIP-Provider without "SIP/2.0 180 Ringing" no Audio - https://issues.asterisk.org/jira/browse/ASTERISK-19181 |
| 14:22.44 | Rico | but I don't use DAHDI and my asterisk version is 1.8.20.1 |
| 14:22.53 | Rico | does anybody know something about that ? |
| 14:23.10 | WIMPy | What you just posted. |
| 14:23.49 | WIMPy | But it might be a good idea to |
| 14:23.54 | WIMPy | ~upgrade asterisk |
| 14:23.54 | 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. |
| 14:24.44 | WIMPy | Also it might help if you tell what you use and not just what you don't use. |
| 14:25.13 | Rico | I have a SIP provider and a SIP phone behind nat on the other side |
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| 14:28.33 | str8uplinux | WIMPy: SIP should show the dialed parties' name on the phone? - I am using cisco 7961's |
| 14:29.13 | [TK]D-Fender | Rico: Your version is old and the key factor for that tracker post is different than your scenario |
| 14:29.25 | WIMPy | str8uplinux: If a call is answered by a SIP device, the caller will get the caller ID of the answering device, yes. |
| 14:29.27 | [TK]D-Fender | Rico: This does not help in pinpointing your problem. Show us your problem |
| 14:29.56 | [TK]D-Fender | str8uplinux: it will when you use the CONNECTEDLINE app |
| 14:30.19 | [TK]D-Fender | function* |
| 14:30.22 | WIMPy | [TK]D-Fender: No need. It happens automatically. |
| 14:30.35 | str8uplinux | WIMPy: it doesn't happen automatically for me. |
| 14:31.15 | WIMPy | You need to enable sendrpid. Probably to "pai". |
| 14:31.16 | str8uplinux | WIMPy: do you have to have something specific set on each extension in sip.conf? |
| 14:31.37 | WIMPy | If that doesn't show anything, you need to upgrade Asterisk or get better phones. |
| 14:31.50 | str8uplinux | better phones than cisco 7961's? |
| 14:31.51 | WIMPy | I don't remember since when it works. |
| 14:32.12 | Qwell | str8uplinux: anything is better than 79xx. Even Grandstream. |
| 14:32.19 | [TK]D-Fender | Cisco never cared much about SIP. Polycom > All |
| 14:32.29 | WIMPy | I think most SIP phones can't display that information. |
| 14:32.50 | str8uplinux | WIMPy: probably to "pia" ? |
| 14:32.56 | str8uplinux | pai* |
| 14:33.15 | WIMPy | sendrpid=yes or sendrpid=pai |
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| 14:46.46 | Rico | [TK]D-Fender: i can't reproduce the problem on-demand |
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| 14:47.19 | Rico | I've just seen that if I have no 180 ringing / 183 session progress, I have no audio |
| 14:50.06 | Greenlight | You should get a 183 if there's early media... |
| 14:50.21 | Rico | I don't |
| 14:50.24 | Greenlight | From your ITSP ? |
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| 14:50.46 | Rico | captures I've done is on client side |
| 14:50.50 | Rico | no on provider side |
| 14:50.56 | Greenlight | SO, your ITSP |
| 14:51.00 | Rico | s/no/not/ |
| 14:51.11 | Greenlight | *from* asterisk ? |
| 14:51.38 | [TK]D-Fender | Rico: * will only pass on progress if it gets progress. |
| 14:51.53 | [TK]D-Fender | Rico: Otherwise you're going to have to answer the call and force it inband with "r" |
| 14:51.54 | Rico | Greenlight: capture have been done between asterisk and the user's phone |
| 14:52.07 | Greenlight | Rico: SO, is this *from* asterisk or *from* ITSP ? |
| 14:52.24 | Greenlight | If it's from asterisk, then add either Ringing() or Progress() |
| 14:52.59 | Rico | after the Dial() ? |
| 14:53.05 | Rico | in any cases ? |
| 14:53.53 | Greenlight | Show us the call. |
| 14:54.37 | Rico | Greenlight: tcpdump capture ? asterisk logs ? |
| 14:55.07 | Greenlight | Just from the CLI initially |
| 14:55.25 | Rico | Id on't have it, I only can provide logs or tcpdump capture |
| 14:55.49 | Greenlight | Logs is fine then |
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| 14:55.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.21 | [TK]D-Fender | Rico: BEFORE the dial clearly... |
| 14:57.53 | Rico | mmh, I can pastebin it, but there's nothing interesting in it, just a : AGI Script Executing Application: (Dial) Options: (SIP/trunk/06....8954) |
| 14:58.36 | Rico | and the SIP/trunk-00283273 answered SIP/101-00283272 |
| 14:58.53 | Greenlight | Right, so you call FROM SIP/101 > Asterisk > SIP/trunk ? |
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| 14:59.15 | Greenlight | Okay, what's that AGI script doing the Dial for then ? |
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| 15:01.16 | Rico | Greenlight: a lot of stuff inthe AGI |
| 15:01.23 | Rico | but nothing can interfer with that |
| 15:02.09 | [TK]D-Fender | just look at the actual dial |
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| 15:03.12 | Rico | Yes, I'll try tu add ',,r' at the end |
| 15:03.18 | Rico | s/tu/to/ |
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| 15:06.26 | Rico | and a "Progress()" before the dial |
| 15:07.15 | Greenlight | No need for both. |
| 15:07.39 | Greenlight | I'm not sure if Asterisk will pass the audio, prior to a 180 or 183 though |
| 15:07.53 | Greenlight | Does seem very odd if your ITSP is doing that |
| 15:08.56 | Greenlight | It's kinda what 183 is *or* |
| 15:08.59 | Greenlight | *for* |
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| 15:52.30 | mirela666 | Hello, has anyone encoontered fax ATA problems with max datagram on Asterisk 1.6.2 |
| 15:53.00 | mirela666 | http://pastebin.com/QNh8qzyY |
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| 15:53.53 | Penguin | unsupported |
| 16:01.10 | mirela666 | I have forces maxdatagram in sip.conf : t38pt_udptl=yes,fec,maxdatagram=400 |
| 16:04.35 | Greenlight | ~upgrade asterisk |
| 16:04.35 | 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. |
| 16:04.52 | mirela666 | oki thx :) |
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| 16:17.36 | mirela666 | yep it was the older version... thanks :) |
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| 17:10.55 | anonymouz666 | sruffell: another case that a framer 2.1 is recognized with dahdi 2.3.0.1 and 2.6.1 does not found any frame. |
| 17:10.58 | anonymouz666 | framer |
| 17:11.18 | anonymouz666 | wct4xxp 0000:01:00.0: FALC Framer Version: Unknown (VSTR = 0x99) |
| 17:11.21 | dorphalsig | Hi. I'm running two * servers. One is connected to my SIP provider and the other handles all internal comms. However when I try calling through my SIP provider I get this error: |
| 17:11.52 | anonymouz666 | card 0: FALC framer is v2.1 or earlier. (DAHDI 2.3.0.1) |
| 17:13.05 | dorphalsig | Error No route from SIp/trlmextrunk to IAX2/main |
| 17:13.13 | dorphalsig | sorry |
| 17:13.24 | dorphalsig | no path to translate from SIP/telmextrunk to IAX2/main |
| 17:13.43 | Greenlight | The codces don't match, and you can't transcode |
| 17:14.04 | Greenlight | What codecs are you trying to use ? |
| 17:14.36 | dorphalsig | GSM |
| 17:14.37 | sruffell | anonymouz666: another case? What happened with the last case? |
| 17:14.50 | dorphalsig | and incoming I think is ulaw |
| 17:15.03 | Greenlight | dorphalsig: Don't think. Know. Go check. |
| 17:15.15 | anonymouz666 | sruffell: revert back to old dahdi |
| 17:16.07 | sruffell | If my memory serves, the last time was something with dahdi/system.conf or something like that? |
| 17:16.08 | sruffell | (but that wouldn't explain VSTR=0x99) |
| 17:16.46 | anonymouz666 | before that case I reported a case exactly as the same condition I am seeing now |
| 17:17.32 | dorphalsig | I havent specified any codec preference for my IP trunk |
| 17:17.58 | sruffell | would you be willing to bisect the version that is causing problems on your system? I have 2.1 framers in our test set here without issue so it will be hard for me to look into. (or set up a maintenance window where I can log into your system) |
| 17:18.26 | Greenlight | At a guess it's using something like g729, which by default Asterisk will not trasncode from/to. |
| 17:18.47 | Greenlight | Be explicit about which codec you want to use, and ensure they match, or you can transcode. |
| 17:19.12 | anonymouz666 | sruffell: are you available to take a look now? |
| 17:19.19 | dorphalsig | ok. I had not allowed gsm |
| 17:19.23 | dorphalsig | thanks for the helo |
| 17:19.24 | sruffell | yes (but not for too long) |
| 17:19.25 | dorphalsig | help |
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| 17:20.39 | anonymouz666 | sruffell: I can save your time, if there's instructions that I can follow, maybe that can speed up things.. |
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| 17:24.03 | sruffell | actually, if you could tell me if happens on 2.5.1 and on 2.6.2 that would be helpful. |
| 17:25.11 | anonymouz666 | 2.6.2 could be replaced by 2.7.x.x ? |
| 17:26.43 | sruffell | yeah…it's the 2.5.1 that I'm more interested in anyway. |
| 17:27.13 | sruffell | do you know what version of cards these are? |
| 17:27.54 | anonymouz666 | wct4xxp 0000:01:00.0: Found a Wildcard: Wildcard TE410P (2nd Gen) |
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| 17:35.06 | sruffell | I'm just hoping that the framer accesses do not need to be slowed down even more like when http://svnview.digium.com/svn/dahdi?view=revision&revision=10559 was committed. |
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| 17:45.37 | Katty | hello thar |
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| 17:58.35 | monsterco | I am hving a problem with meetme channel not working with psudo DAHDI - I have a sangoma timer device. Where should i look for configuration? |
| 17:58.36 | *** join/#asterisk vlad_sta_ (~vlad_star@91.233.188.226) |
| 17:58.50 | monsterco | Error: WARNING[973]: app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo device |
| 17:59.06 | monsterco | to give a fuller picture here are all the details paste: http://paste.debian.net/64146/ |
| 17:59.48 | Penguin | Do you have chan_dahdi loaded? |
| 18:00.32 | monsterco | Penguin - my understanding is that yes but I am not sure. How can I check? |
| 18:00.49 | monsterco | dahdi show channels: pseudo default default In Service |
| 18:01.23 | Penguin | That seems to indicate it is loaded. module show like dahdi to be sure. |
| 18:02.15 | monsterco | chan_dahdi.so DAHDI Telephony Driver w/PRI & SS7 & MFC 0 |
| 18:02.22 | monsterco | codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 |
| 18:02.30 | monsterco | res_timing_dahdi.so DAHDI Timing Interface 0 |
| 18:02.34 | monsterco | app_dahdiras.so DAHDI ISDN Remote Access Server 0 |
| 18:02.43 | monsterco | app_dahdibarge.so Barge in on DAHDI channel application 0 |
| 18:02.48 | monsterco | that's it ^^^^ |
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| 18:03.37 | monsterco | I may have to configure the psudo dahdi channels? think I saw an error like that somewhere |
| 18:05.31 | monsterco | Penguin - Dahdi show status: WANVTIME/1 (source: wanpipe_voicetime) UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 18:07.44 | Penguin | I don't remember having to configure anything for it to work. I remember installing dahdi and meetme worked. |
| 18:09.47 | monsterco | Penguin - this is on OpenVZ so HN is giving access to Dahdi. I am reading on this thread that my OpenVZ must be a 64bit just mother node or else it will fail - any experience with that? |
| 18:11.17 | Penguin | No. I have never tried it on a virtual machine. |
| 18:14.52 | monsterco | thanks - so there is nothing else that I should put in meetme.conf for example or anywhere else? |
| 18:16.39 | Penguin | I don't have any settings configured in meetme.conf. |
| 18:18.23 | Qwell | Why are you still using meetme? |
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| 18:27.57 | monsterco | @Qwell - what else can I use? |
| 18:28.04 | monsterco | I am on Asterisk 1.8.3 |
| 18:28.20 | Penguin | There's ConfBridge, but it isn't very robust in 1.8. |
| 18:28.37 | monsterco | is there binary for that? |
| 18:29.13 | Penguin | app_confbridge.so |
| 18:29.21 | Penguin | It's part of asterisk. |
| 18:32.40 | monsterco | Penguin - how can I check to see if it's installed? |
| 18:32.52 | Penguin | module show like confbridge |
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| 18:33.58 | monsterco | Module Description Use Count |
| 18:33.58 | monsterco | app_confbridge.so Conference Bridge Application 0 |
| 18:34.11 | Penguin | There you go. |
| 18:37.23 | monsterco | so, how can I replace meetme to that? |
| 18:41.16 | Penguin | core show application ConfBridge |
| 18:41.37 | Penguin | Edit your dial plan to use the ConfBridge app instead of the MeetMe app. |
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| 19:02.18 | monsterco | Penguin - thanks all working fine for now. Let's test and see... |
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| 21:04.55 | Katty | jigs through channel |
| 21:04.58 | Katty | jazzhands |
| 21:05.02 | Katty | jigs back out |
| 21:08.29 | LeLutin | WIMPy: I got reports from users since I disabled jitterbuffer on iax/sip that now sound doesn't just cut, it's repeating some seconds for some time then might be coming back and goes back to repeating again |
| 21:08.59 | LeLutin | ^ or if anyone else can help.. it's just that WIMPy was the one who helped me last time |
| 21:09.15 | LeLutin | I suspect removing the jb exposed the bigger underlying problem. |
| 21:15.55 | ipengineer | Does anyone know if hints work the same way with PJSIP as SIP module? exten => 101,hint,PJSIP/sip101_abcdefg |
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| 21:18.00 | mjordan | ipengineer: that should be the correct nomenclature. |
| 21:20.37 | ipengineer | mjordan: Ok for some reason that isn't working for me.. Anything else that needs to be done without setting that in the dial plan? I know in older version of asterisk we would have to do some trickery in SIP.conf but I don't see that mattering any more since we aren't using that module. |
| 21:23.17 | Katty | allah, peanut butter sammiches! |
| 21:23.23 | Katty | does allah like pb sammiches? |
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| 22:32.13 | ipengineer | gtjoseph: Are you working on the pjsip show endpoints output format? |
| 22:32.43 | gtjoseph | ipengineer: I am, yes |
| 22:34.07 | ipengineer | gtjoseph: Ok great.. yea I thought that looked like you ;) I guess the current expected behavior is for just the name to be returned when running that command? |
| 22:36.07 | gtjoseph | ipengineer: I'm trying to get as much of the volatile info in the list as I can but I'll take any feedback I can get as to what should be there. |
| 22:39.57 | ipengineer | gtjoseph: Ok sounds great.. I was looking through the CLI-specs on the wiki and thought they looked really good. I will dig through your paste |
| 22:40.19 | mjordan | ipengineer: Hm... res_pjsip_exten_state needs to be loaded obviously, and things would need to subscribe for the extension state still. Other than that, I wouldn't expect there to be anything else. putnopvut file or kharwell may be able to help you a bit more however |
| 22:40.34 | mjordan | ipengineer: you may need to go poke them in #asterisk-dev however |
| 22:41.14 | file | they do work the same way |
| 22:41.27 | gtjoseph | ipengineer: Yep, I've been in touch with kharwell and I'm useing much of the stuff he has ready for ami. |
| 22:42.29 | ipengineer | gtjoseph: Ok great. I am running on beta 1 so prob behind on some of the newer dev builds |
| 22:43.09 | putnopvut | ipengineer: a couple of things. 1) I've noticed sometimes that on start-up, device state of PJSIP devices may be "invalid" until an endpoint places a call. After that, all is well. 2) We only have support for the "presence" SIP event-package. If you are using a device that subscribes to the "dialog" event-package, we don't yet support that. |
| 22:43.18 | gtjoseph | ipengineer: kharwell's stuff is still in review so it's not even committed yet. he's hoping for this week then I can follow on with my stuff. |
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| 22:43.54 | ipengineer | file: So if res_pjsip_exten_state.so is running, which it is and I have "exten => 100,hint,PJSIP/sip100_abcdefg" set in the dial plan Im assuming that is all I would need... |
| 22:44.42 | file | even without res_pjsip_exten_state loaded it'll still keep track of device state, you just wouldn't be able to subscribe to it from SIP land ^_^ |
| 22:45.20 | ipengineer | putnopvut: Ok issue 2 may be my problem I will have to go look into that.. |
| 22:46.00 | ipengineer | file: Makes sense.. I am thinking the dialog event subscription may be the problem. |
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| 22:52.31 | cptreptile | Dear all, i am having top weird issue with hearing playbacks on ast 11.5, it was working perfectly...i am running it on VPS, i had an earlier staged duplicate VPS that has same issue, tested also with firewall off |
| 22:52.46 | ipengineer | putnopvut: So digging this is what I see in the console "No registered publish handler for event presence" |
| 22:53.07 | file | that's fine, we don't have anything written to accept PUBLISH events |
| 22:53.34 | ipengineer | file: and also "No registered subscribe handler for event dialog" |
| 22:53.49 | ipengineer | which is I think the same as what we were discussing earlier.. |
| 22:53.53 | file | yup |
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| 23:07.50 | ipengineer | file: putnopvut Thanks for the help.. Calling it a night here. Will keep an eye out for dialog event support |
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