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| 00:25.35 | edong23 | has anyone in here sucessfully set up a remote queue member as a cell phone? |
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| 00:28.26 | [TK]D-Fender | plenty |
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| 00:29.46 | edong23 | [TK]D-Fender: do you have... any... examples? or material i can red? |
| 00:29.54 | edong23 | i dont mind the research.. but it is limited online |
| 00:30.03 | [TK]D-Fender | There is nothing to read. |
| 00:30.04 | edong23 | and most of what i find talks about deprecated ways to do it |
| 00:30.17 | [TK]D-Fender | A device is a device is a device |
| 00:30.35 | edong23 | oh, true |
| 00:30.39 | edong23 | yeah, i can make it ring.. |
| 00:30.45 | [TK]D-Fender | The concept of "remote" doesn't actually say anythiing in and of itself. |
| 00:30.51 | edong23 | but i was thinking of... a way to make them able to log it in or out |
| 00:31.09 | [TK]D-Fender | logging in/out is no different |
| 00:31.27 | [TK]D-Fender | * has the same dialplan apps, the same AMI function calls, the same CLI commands... |
| 00:32.02 | edong23 | [TK]D-Fender: im not following... |
| 00:32.06 | edong23 | let me tell you my setup |
| 00:32.12 | edong23 | maybe youll see my confusion |
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| 00:32.58 | edong23 | i have a few queues set up. they are static. just a file with a list of devices that ring at a certain time. |
| 00:32.59 | [TK]D-Fender | You are certainly overthinging this |
| 00:33.10 | edong23 | maybe... |
| 00:33.11 | [TK]D-Fender | overthinking* |
| 00:33.33 | edong23 | i think... maybe i am a little |
| 00:33.45 | edong23 | but i need... different phones to be able to just "log in" to the queue... |
| 00:33.51 | edong23 | not always the same cellphone |
| 00:34.14 | edong23 | if im over thinking it... then maybe i just need sleep |
| 00:35.02 | [TK]D-Fender | What are the tools * gives you to "log in" to a queue? |
| 00:35.33 | edong23 | ive never done queues with login |
| 00:36.31 | edong23 | ah, agents.conf |
| 00:36.33 | edong23 | ? |
| 00:36.38 | [TK]D-Fender | that's the problem then. It has nothing to do with "remote" or "cell" at this point |
| 00:36.38 | edong23 | let me look into that |
| 00:36.45 | [TK]D-Fender | no, agents.conf is DEAD |
| 00:37.04 | [TK]D-Fender | "core show application AddQueueMember" |
| 00:37.11 | [TK]D-Fender | "core show application RemoveQueueMember" |
| 00:37.16 | [TK]D-Fender | "core show application PauseQueueMember" |
| 00:37.19 | [TK]D-Fender | "core show application UnPauseQueueMember" |
| 00:37.48 | edong23 | [TK]D-Fender: thanks. let me look into those |
| 00:37.52 | [TK]D-Fender | Dialplain apps you SHOULD know. Read the book on thiw section and give a thorough reading for all of *'s dialplan apps |
| 00:40.38 | edong23 | yes, [TK]D-Fender ill read this |
| 00:40.44 | edong23 | i see it though |
| 00:40.48 | edong23 | but im going to give it a good read |
| 00:40.49 | edong23 | thanks |
| 00:41.32 | [TK]D-Fender | be sure to go over all of *'s channel-types while you're at it.... |
| 00:44.16 | edong23 | yeah, ill dig through that pretty good too |
| 00:44.18 | edong23 | after my workout |
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| 01:08.49 | pabelanger | Is there any way to disable a SIP NOTIFY in asterisk? |
| 01:10.43 | pabelanger | I should be more specific, disable a SIP NOTIFY for Messages-Waiting: no |
| 01:13.31 | pabelanger | okay, so this: Notification only works for registered peers with mailbox= definitions in sip.conf |
| 01:16.54 | pabelanger | and fixed. |
| 01:18.19 | edong23 | lol |
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| 01:39.15 | pabelanger | anybody using res_corosync or XMPP for distributed device state? Any one work better? |
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| 02:36.07 | linagee | is there an easy way to wait for audio when asterisk is automatically originating calls? (I have a callback service, but don't want to start playing any sounds until I know someone has said "Hello?") |
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| 02:37.31 | linagee | WaitForSilence? |
| 02:37.48 | linagee | hrm. I almost want the opposite of that though... |
| 02:38.53 | linagee | ah. BackGroundDetect. now I just have to figure out how to use it. :) |
| 02:39.34 | pabelanger | AMD |
| 02:40.04 | linagee | nice. :) |
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| 03:46.56 | Kalidarn | okay that's weird, i have a crappy openvox card, it seems to work. my official digium card though is all crackley |
| 03:47.16 | Kalidarn | i did do an fxotune on it ie fxotune -i 4 |
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| 04:48.38 | Kalidarn | okay well i put the old openvox card in my new server and it also crackles. |
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| 05:07.16 | edong23 | Kalidarn: same server causes the crackle? |
| 05:07.44 | Kalidarn | 8yeah. |
| 05:07.54 | Kalidarn | and it only has one pci slot so i can't check it in another one |
| 05:08.07 | Kalidarn | so it's obviosuly not the card that is the issue |
| 05:13.41 | edong23 | Kalidarn: its possible the pci voltages are different |
| 05:13.48 | edong23 | from the old one, that i |
| 05:14.06 | Kalidarn | hmm |
| 05:14.21 | Kalidarn | probably the other board was an old one, this one is a haswell one |
| 05:14.40 | Kalidarn | haswell did all sorts of kinky things with power saving |
| 05:15.07 | edong23 | i thought haswell was doing away with legacy pci? |
| 05:15.19 | Kalidarn | well my board has one PCI slot on it |
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| 05:16.25 | eirirs_ | finally |
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| 05:18.31 | edong23 | Kalidarn: ah, its not haswell, it is the q85 chipset that does away with pci |
| 05:18.48 | edong23 | Kalidarn: does your "old" card have 2 slits? |
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| 05:29.09 | Kalidarn | edong23: it's a DZ87KLT-75K |
| 05:29.21 | Kalidarn | http://www.intel.com/content/www/us/en/motherboards/desktop-motherboards/desktop-board-dz87klt-75k.html |
| 05:29.41 | Kalidarn | Three PCIe x1slots |
| 05:29.43 | Kalidarn | One PCI slot |
| 05:31.41 | Kalidarn | and i was using a http://ark.intel.com/products/75044/Intel-Core-i5-4570S-Processor-6M-Cache-up-to-3_60-GHz on it |
| 05:35.13 | Kalidarn | the old board was an G31M-ES2L edong23 |
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| 05:54.18 | edong23 | your crd |
| 05:54.20 | edong23 | the card |
| 05:54.22 | edong23 | not hte board |
| 05:55.36 | Kalidarn | edong23: i have two |
| 05:55.42 | Kalidarn | one is a Wildcard TDM400P REV E/F and it's an openvox clone |
| 05:55.48 | Kalidarn | the other one is a legit Wildcard TDM400P REV I |
| 05:55.51 | Kalidarn | from Digium |
| 05:55.56 | Kalidarn | both have the same issue |
| 05:56.15 | edong23 | dude |
| 05:56.16 | edong23 | work with me |
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| 05:56.25 | edong23 | i was asking if they are 3.3v cards, or 5v cards |
| 05:56.29 | Kalidarn | oh :P |
| 05:56.31 | edong23 | do they ahve 2 slits in them? |
| 05:56.32 | edong23 | or 1? |
| 05:56.33 | Kalidarn | i'd have to check |
| 05:56.37 | Kalidarn | well they both fit in the slot |
| 05:56.58 | edong23 | im not even sure it matters |
| 05:57.49 | Kalidarn | edong23: ill look in the technical manual of the motherboard to see what voltage the slot is |
| 05:57.58 | Kalidarn | i think it will adjust depending on what the card wants would it not |
| 05:58.12 | edong23 | modern boards are likely 5 volts |
| 05:58.14 | edong23 | but again, i was asking about your old card |
| 05:58.18 | edong23 | it might be 3.3 |
| 05:58.24 | Kalidarn | oh. they would be the same i think |
| 05:58.26 | Kalidarn | im not sure |
| 05:58.28 | edong23 | no idea |
| 05:58.45 | Kalidarn | in any case it doesn't matter both the digium one and the openvox one do exactly the same thing in the haswell board |
| 05:59.10 | Kalidarn | where as the openvox one worked fine in the previous conroe board |
| 05:59.22 | edong23 | i woul be pretty curious about irqs though |
| 05:59.24 | Kalidarn | i haven't tried the digium one in there, but i expect it works too |
| 05:59.36 | Kalidarn | yes i looked at that it didn't seem to be sharing any |
| 05:59.37 | edong23 | did you check /proc/interrupts |
| 06:00.17 | Kalidarn | it's freebsd so i used vmstat -i http://pastebin.com/SWUWBNDJ |
| 06:00.59 | edong23 | dont know |
| 06:01.01 | edong23 | im heading to bed |
| 06:01.07 | Kalidarn | nighty night |
| 06:01.12 | Kalidarn | but yeah it has it's own IRQ |
| 06:01.20 | Kalidarn | so i would say it isn't sharing any interrupts? |
| 06:01.28 | Kalidarn | i don't really know too much about that |
| 06:01.40 | Kalidarn | nothing else seems to be on IRQ 18 |
| 06:02.54 | Kalidarn | and it's obviously not a problem with my dahdi settings, because it worked in the old motherboard the G31M-ES2L |
| 06:06.31 | bsdice | <PROTECTED> |
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| 07:17.43 | Kalidarn | i have an easy solution |
| 07:17.47 | Kalidarn | throw the PSTN card in the bin |
| 07:17.52 | Kalidarn | and go naked dsl |
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| 08:19.11 | linjan | hello! is there any way to accumulate CDR records, when ODBC database is offline? |
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| 08:23.19 | mirela666 | hm, can someone help me with pattarns? |
| 08:23.33 | mirela666 | I want to inclde in one line emergency numbers |
| 08:23.42 | mirela666 | (100,101,102,112) |
| 08:23.55 | mirela666 | but exten => _1[01][0-2],1,Macro(emergency-numbers) |
| 08:23.56 | mirela666 | <PROTECTED> |
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| 08:24.18 | mirela666 | matches 110 and 111 which I don't want t match |
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| 08:24.55 | mirela666 | can't thee be like : _10[0-2]|112 for example? |
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| 08:28.28 | linjan | mirela666: i would use two strings: _10[0-2],1,Macro... and next 112,1,Macro... |
| 08:28.56 | linjan | or maybe your situation requires only one? |
| 08:29.23 | wdoekes | Macro? Macro? |
| 08:29.46 | wdoekes | there is this thing called Gosub |
| 08:29.56 | linjan | wdoekes: GoSub is better, yeah |
| 08:31.24 | wdoekes | mirela666: if you want both on a single line, you can, if you use extra Ifs.. it depends on your dialplan if that looks better or not |
| 08:31.34 | wdoekes | usually it won't |
| 08:32.57 | mirela666 | oki thx :) |
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| 09:33.17 | frooker | hi everybody =) |
| 09:33.21 | frooker | i have a problem |
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| 09:33.26 | frooker | can anyone help me ? =) |
| 09:33.41 | WIMPy | No. |
| 09:33.46 | WIMPy | Impossible. |
| 09:34.16 | frooker | :( |
| 09:34.37 | frooker | i have a problem on agent login |
| 09:35.00 | frooker | i am using *111 [agent_extension] for login |
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| 09:35.19 | frooker | can i do it in originate aciton ? |
| 09:37.04 | WIMPy | You need to ask complete questions. |
| 09:38.03 | frooker | i want to originate a call and answer it in background can anyone help me ? |
| 09:39.20 | WIMPy | What does "answer in background" mean? |
| 09:39.40 | WIMPy | And is that stil the same question or are they different questions? |
| 09:39.54 | frooker | same question |
| 09:39.58 | frooker | i need that |
| 09:40.17 | WIMPy | What? |
| 09:40.49 | frooker | i want to make a call silently in background |
| 09:40.58 | frooker | and this is an agent login dial |
| 09:41.06 | frooker | i am using *111 to login agents |
| 09:41.48 | WIMPy | Stop mentioning random facts and tell us what you really want. The whole story in comprehensible terms. |
| 09:42.23 | frooker | when i originate *111435 my agent session starting but agent must answer this call i want to call a number and answer it in background |
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| 10:00.06 | skrusty | morning |
| 10:00.21 | WIMPy | Good morning |
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| 10:31.16 | wizbit | i am launching asterisk as a asterisk user now: |
| 10:31.17 | wizbit | asterisk 20250 4.5 0.6 2442592 25880 ? Ssl 10:30 0:00 /usr/sbin/asterisk -U asterisk |
| 10:31.38 | wizbit | however, when i start, i still get this message: Privilege escalation protection disabled! |
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| 10:59.28 | WIMPy | That has nothing to do with system permissions. Didn't you get the URL with the explanation? |
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| 14:14.18 | AsteriskUser | Hi All, I'm having a strange issue with early audio not passing back to phone from the provider. This issue only occurs when asterisk has abount 70 calls. I have a large hash_user, hash_peer and hash_dialogue along with large ulimits for files and processes. Audo works fine once the call is answeres and is not on every call. Can you offer any adivice on settings that may be causing this? |
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| 16:05.32 | Ice_Strike | Is there a way to find out What user has connected to a sip phone and what IP it came from? |
| 16:05.36 | Ice_Strike | Via log? |
| 16:07.16 | pabelanger | *CLI> sip show peer foo |
| 16:07.19 | pabelanger | will have it |
| 16:07.25 | pabelanger | or turn up verbose / debug logs |
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| 16:45.41 | AsteriskUser | Hi All, has anyone come across asterisk not passing one way audio for some calls? I can see audio passing from the provider but asterisk does not pass this on. Could this be a timing issue? |
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| 16:50.47 | wdoekes | AsteriskUser: do you have timing modules loaded at all (res_timing_*?)? and has the destination device sent any RTP yet? is the destination device behind nat? |
| 16:50.52 | wdoekes | rtp set debug on |
| 16:51.25 | salz212 | hi all, I have installed Asterisk12 and wanted to test opus and VP8 codecs.. is there sny patch available for Asterisk 12 for both codecs? |
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| 16:53.53 | AsteriskUser | yes I have res_timing_dahdi wich has a use count of 312 |
| 16:55.26 | AsteriskUser | looking at the version it is 2.6.1. Does anyone know if this has an issue with high usage? or is incompatible with 1.8.24? |
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| 17:27.42 | salz212 | hi all, I have installed Asterisk12 and wanted to test opus and VP8 codecs.. is there sny patch available for Asterisk 12 for both codecs? |
| 17:29.46 | mjordan | salz212: Asterisk 12 has pass through support only. If you want the actual codec, Google is your best friend. A quick google of "asterisk codec_opus" has lots of useful hits that should lead you to something. Asterisk does not support video codecs, so a codec_vp8 does not exist. |
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| 17:36.38 | salz212 | Thanks Matt, I was checking lminiero patch but its for 11.x .. do I need to modify it or just go with codec_opus.c? |
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| 17:49.16 | justdave | when I'm making a dundi lookup zone, and I have one server that has an exact number and another server that has a catchall for a range that includes it, is there a way to make the catchall get returned with a lower weight than it would if it were an exact match? |
| 17:50.26 | wizbit | what module is required so asterisk can play sounds like 'queue-youarenext.gsm' ? |
| 17:50.39 | wizbit | my setup works apart from that |
| 17:50.46 | wizbit | the log says its playing them but it doesnt |
| 17:50.52 | [TK]D-Fender | format_gsm.so |
| 17:50.59 | wizbit | ace :-) |
| 17:51.09 | justdave | only thing I can think of would be putting the catchall dialplan into a different context and having a separate dialplan entry for that context in the dundi config, but that section of the dialplan is generated and I don't really have control over what context it's in |
| 17:51.34 | wizbit | load => format_gsm.so |
| 17:53.39 | wizbit | hmm still not sound when the log shows this: |
| 17:53.39 | wizbit | Playing 'queue-youarenext.gsm' (language 'en') |
| 17:53.47 | mjordan | salz212: since pass through support was added, if you have a .so that implements a codec, you should just need to load it. |
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| 18:00.34 | [TK]D-Fender | wizbit: Is the file even there? |
| 18:00.50 | [TK]D-Fender | also, it should never say ".gsm' |
| 18:01.00 | [TK]D-Fender | You should never ever specify the EXTENSION of the file ot be played |
| 18:01.02 | wizbit | [TK]D-Fender: i got other sounds to play |
| 18:01.10 | [TK]D-Fender | It is selected automatically based on cheapest transcode |
| 18:01.15 | wizbit | ok |
| 18:01.39 | wizbit | is there a way to do live text to speech for phone messages? |
| 18:01.50 | wizbit | not live, just text to speech |
| 18:03.32 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 18:04.11 | [TK]D-Fender | wizbit: There are several engines you can add to * |
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| 18:05.01 | justdave | hmm, I can use a diaplan function to set the weight... is there a way to tell in a dialplan function whether the dialplan that matches a number uses a wildcard or not? |
| 18:05.24 | hfp | Hi, is it possible to mute these messages 'Received SIP subscribe for peer without mailbox:' ? |
| 18:05.36 | hfp | I don't want to remove the rest of the messages but this one is flooding the console |
| 18:06.36 | justdave | looks like dialplan_exists() only tells me if it exists or not, not how it matched it |
| 18:08.25 | wizbit | [TK]D-Fender: http://asterisk-espeak.sourceforge.net/ |
| 18:08.43 | wizbit | i wonder if that has a sexy female voice |
| 18:08.49 | justdave | valid_exten() is the same thing as dialplan_exists() ? |
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| 18:15.08 | [TK]D-Fender | pretty much |
| 18:15.48 | [TK]D-Fender | hfp: Not selectively. Try to synchronize your blinks so you don't feel so bad... |
| 18:16.43 | newtonr | hfp, you should configure the SIP devices to not attempt subscription if possible. That is, or create mailboxes for them, then you won't see that message. |
| 18:21.31 | hfp | [TK]D-Fender, newtonr: Alright, thanks |
| 18:22.47 | salz212 | Matt, I do need to test (just test for now) the opus transcoding as well, so that is why I wanted to find a patch for 12 if available I have found related patches on internet but they are for 11.v. By the way .so file is hard to find :S |
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| 18:28.59 | newtonr | salz212, you have to use the persons nick for most clients to notify the person. |
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| 18:43.22 | mjordan | salz212: the codec_opus module is not officially part of the Asterisk distribution. If you want to test it, you should contact the author of the module. If they aren't lurking in this channel, that's probably your best bet. |
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| 18:49.53 | FinboySlick | Anyone here familiar with encrypted configuration/provisioning on Polycom SoundPoint phones? This isn't strictly asterisk related but I seem to remember polycom phones being very popular around here. |
| 18:50.48 | FinboySlick | The phones seem to reject my encrypted config and there's likely something very obvious that I'm missing. |
| 18:56.12 | funky1 | is going nuts |
| 18:56.37 | funky1 | getting 481 error on sip trunk registration :/ |
| 19:04.21 | salz212 | mjordan: thanks. |
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| 20:10.19 | roramirez | hello |
| 20:11.29 | roramirez | how i can get dev state from manager? i am using asterisk 1.8.25 |
| 20:13.04 | *** join/#asterisk Tom_123 (89eddf0e@gateway/web/freenode/ip.137.237.223.14) |
| 20:13.25 | pabelanger | Newstate event |
| 20:13.29 | pabelanger | IIRC |
| 20:13.30 | Tom_123 | Anyone know how I can send DTMF signals on a trunk with * 1.8 |
| 20:14.00 | [TK]D-Fender | Tom_123: Same was as on any version |
| 20:14.13 | pabelanger | Tom_123, SendDTMF |
| 20:14.19 | Tom_123 | which is? I thought it would do it by default |
| 20:14.19 | [TK]D-Fender | Tom_123: The key question is WHEN is this decision to be made, and at what point of the processing. |
| 20:14.43 | [TK]D-Fender | Tom_123: You need to be a lot clearer as well... |
| 20:14.58 | [TK]D-Fender | Tom_123: Asterisk PASSES DTMF from end to end automatically. |
| 20:15.16 | [TK]D-Fender | Tom_123: is you want to INTRODUCE DTMF based on some other process then that is another matter. |
| 20:16.07 | Tom_123 | I just need to pass DTMF tones, but it doesn't seem to be working |
| 20:17.01 | Tom_123 | I have an exten => _790X,1,Dial(SIP/mytrunk/${EXTEN}, 30) |
| 20:17.02 | roramirez | ¿ is possible ? |
| 20:17.30 | [TK]D-Fender | Tom_123: what "tones"? |
| 20:17.38 | Tom_123 | * and # presses |
| 20:17.40 | [TK]D-Fender | Tom_123: that is not passing "tones... you are dialing a NUMBER to a SIP provider... |
| 20:17.42 | Katty | don't you get that tone with me |
| 20:17.58 | [TK]D-Fender | sweeps the mids! |
| 20:18.02 | [TK]D-Fender | hyaaaaaaa!!!!!!!!!!! |
| 20:18.19 | Tom_123 | ? |
| 20:18.24 | Katty | pats tk |
| 20:18.39 | Katty | tab complete fails on circ |
| 20:19.01 | [TK]D-Fender | Tom_123: first go prove that you are getting them properly by the channel that is placing that call |
| 20:19.18 | Tom_123 | what debug would show me the presses? |
| 20:19.27 | pabelanger | Tom_123, if you are trying to interact with something you need to use SendDTMF, after the channels is picked up. Or use playback to play the sound files of dtmf |
| 20:19.47 | [TK]D-Fender | Tom_123: Go make an IVR or something that expects to process them. |
| 20:20.01 | [TK]D-Fender | Tom_123: then if that goes through.l. it's wherever you are call through/to that is at fault |
| 20:21.30 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:22.02 | Katty | i feel like i should have a joke about danjenkins |
| 20:22.27 | [TK]D-Fender | But that would be redundant! |
| 20:22.30 | [TK]D-Fender | *zing*! |
| 20:22.32 | danjenkins | huh? |
| 20:22.33 | [TK]D-Fender | There. |
| 20:22.41 | Katty | but he was the one with the lifetime achievement award, not me |
| 20:22.59 | [TK]D-Fender | danjenkins: Random afternoon humour. Feel free to disregard. |
| 20:23.10 | Katty | alternatively, join in |
| 20:23.39 | danjenkins | i only just signed back in and just so happened to be mentioned... i can't see what was said ;) |
| 20:23.45 | Katty | shares a cookie with [TK]D-Fender |
| 20:24.01 | [TK]D-Fender | No-one should be there to receive a life-time achievement award.... they're still alive. Can't sum up their life if they're not dead yet. |
| 20:24.06 | Katty | danjenkins: we never say anything important anyway |
| 20:24.17 | Katty | [TK]D-Fender: no i mean espn lifetime |
| 20:24.25 | Katty | it's right on his wikipedia page |
| 20:24.44 | [TK]D-Fender | danjenkins: Nothing. that was all spawned randomly as you joined. Spontaneous target for random joke |
| 20:25.02 | danjenkins | oh |
| 20:25.04 | [TK]D-Fender | Katty: Check out he bro Leroy ;) |
| 20:25.07 | Katty | like fender's face. |
| 20:25.07 | [TK]D-Fender | his* |
| 20:25.36 | Katty | [TK]D-Fender: goober. |
| 20:33.21 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 20:41.21 | AsteriskUser | Hi, is there an issue with asterisk 1.8.23 up with not always processing early audio. 90% of calls progress however under heavy load no early audio is passed but when the call answers audio progresses... |
| 20:42.09 | Katty | AsteriskUser: hi |
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| 21:16.35 | FinboySlick | Anyone here who's had success getting Polycom encrypted config file provisioning working, please let me know. I'm not entirely sure on the proper way to make the phone aware of the encryption key. Does it have to be one-time provisioned through tftp? |
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| 21:30.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:36.24 | boom^time | Hey guys, I'm using AMI to originate calls to a local exten which then dials out. However I'm not seeing CDR's being created for the dialed out call or the extension/priority. Only the one calling the local channel and using the dial application. Am I forgetting something in the dialplan? |
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| 21:51.02 | snadge | you must have forgotten to charge the flux capacitor |
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| 21:54.09 | boom^time | snadge, I had a feeling it was something simple. |
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| 22:18.00 | snadge | i dont know enough about asterisk.. im almost as curious as you are about that answer |
| 22:19.01 | snadge | if i had to take a wild stab in the dark.. maybe the local extension is dialling out through a different server |
| 22:19.23 | snadge | so with that in mind, perhaps it might be something to do with the dial plan |
| 22:20.09 | snadge | i cant think of why a call could be initiated and a cdr not generated for it |
| 22:23.53 | [TK]D-Fender | Because unanswered calls by default do not leave CDR |
| 22:23.58 | [TK]D-Fender | there is a blatant setting for that |
| 22:25.53 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 22:27.00 | snadge | i just assumed the call was being answered ;) |
| 22:28.03 | snadge | i think i found another bug with freepbx 12 and iax as well.. the problem where the USER/PEER settings box wasn't appearing is fixed |
| 22:28.14 | snadge | but now if you go back into the trunk to edit it.. those boxes don't appear (i think) |
| 22:28.20 | snadge | i'll have to re-test that when I get into work |
| 22:29.12 | snadge | also.. is there a way to modify the freepbx configs directly, without using the gui.. ie.. you have console access to the box only |
| 22:29.28 | [TK]D-Fender | MySQL <- |
| 22:29.41 | snadge | i ended up using ssh to forward a local port through to port 80 |
| 22:29.53 | snadge | ok yeah, that makes sense.. duh ;) |
| 22:30.32 | snadge | i totally fixed this guys asterisk@home setup.. by getting him to type in an ssh command, with a remote port forward |
| 22:31.00 | snadge | even though he was an accountant with no linux experience whatsoever.. he told me he had a root shell, and his tech is awol |
| 22:32.29 | snadge | its not every day you have to talk a non techie through setting up a remote port forward over the phone.. but hey, it worked.. just in case anyone needs to do that ;) |
| 22:39.04 | Penguin | Write up the instructions and publish to your web site, which is indexed by googlebot. |
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