| 00:00.43 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 00:03.19 | pppingme | cusco seems like it wouldn't be that hard to combine into a new 3rd table, someone is probably doing it out there.. Is sql on a 3rd box, or one of the two asterisk boxes? |
| 00:08.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:08.31 | *** join/#asterisk mathi (Matthew@ip-62-235-253-219.dsl.scarlet.be) |
| 00:08.32 | mathi | hi |
| 00:09.05 | mathi | where can I find documentation (functions, applications, applications arguments, etc.) about Asterisk 1.4 ? |
| 00:19.12 | sruffell | http://doxygen.asterisk.org/asterisk1.4/index.html but why Asterisk 1.4? |
| 00:19.33 | sruffell | (the above link was found by following links from http://www.asterisk.org/community/documentation ) |
| 00:19.33 | navaismo | core show application/funtion <application/funtion> |
| 00:24.24 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 00:28.49 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 00:36.12 | dijib | [Nov 29 19:36:36] -- <SIP/23.30.88.89-00000042> Playing 'silence/1.ulaw' (language 'en') |
| 00:36.15 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:36.19 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:36.23 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:36.27 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:36.31 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:36.36 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:36.39 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 00:36.40 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:36.41 | kikohnl | ~pb |
| 00:36.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:36.44 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:36.48 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:36.52 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:36.56 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format ulaw since our native format has changed to 0x1000 (g722) |
| 00:37.00 | dijib | [Nov 29 19:36:36] NOTICE[869]: channel.c:4220 __ast_read: Dropping incompatible voice frame on SIP/23.30.88.89-00000042 of format g722 since our native format has changed to 0x4 (ulaw) |
| 00:37.57 | cusco | pppingme: sql is in a separate box |
| 00:38.16 | cusco | pppingme: I'mintending to use queue_log too |
| 00:38.30 | cusco | I'm hoping I don't get doubled uniqueids |
| 00:44.39 | SeRi | dijib you are a fucking retard! |
| 00:47.34 | kikohnl | SeRi, did you ever get your port 5060 issue with comcast fixed? |
| 00:48.13 | SeRi | kikohnl: Yeap. They put the modem on bridge mode. my upload is still screwed. |
| 00:49.25 | kikohnl | I wish I could get better upload than 1MB, when I have 21MB down :-( |
| 00:50.39 | qakhan | [TK]D-Fender u there |
| 00:51.27 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
| 00:51.30 | qakhan | i want to setup conf room, but i want to setup my own IVR on conf room and it authenticate pin# |
| 00:51.52 | p3nguin | What's with the spewing of crap? |
| 00:52.00 | dijib | #asterisk i a truely sorry for the great error in jedgjement i have mad in flooding your respectful channel. please forgive my asshat ways! |
| 00:52.10 | dijib | judgeent |
| 00:52.39 | p3nguin | judgment |
| 00:52.49 | p3nguin | Close, though. |
| 00:53.00 | SeRi | p3nguin: in your experience when an issue gets escalated to "NOC" for the eng. How long does it take for them to respond? |
| 00:53.08 | dijib | m k i and sometimes u dont like to work for me |
| 00:53.44 | SeRi | dijib you are a fucking retard! |
| 00:53.58 | p3nguin | NOC and engineering don't really go together in my opinion. |
| 00:54.12 | dijib | i think i woul dhave to somewhat agree |
| 00:54.26 | p3nguin | If a problem is escalated to an engineer, I'd expect to hear from him the following day. |
| 00:54.29 | SeRi | well I dont know comcast infrastructure I am just going by what they say |
| 00:54.41 | p3nguin | Even if he isn't able to come out, I'd still expect a call. |
| 00:54.48 | SeRi | ok |
| 00:55.19 | SeRi | My issue is waiting at the "NOC" for an eng. |
| 00:55.26 | SeRi | Thats the satus they claim |
| 00:55.35 | p3nguin | Engineers generally aren't as dumb and unprofessional as the tier 1 support people. |
| 00:55.57 | dijib | <---me |
| 00:56.07 | SeRi | ^^ yes you ^^ |
| 00:56.11 | dijib | im wearing my asshat well i think |
| 00:56.29 | SeRi | p3nguin: jump in for a sec to the conf |
| 00:56.34 | SeRi | I have a few questions |
| 00:56.35 | p3nguin | Maybe later. |
| 00:56.50 | SeRi | damn. |
| 00:56.58 | SeRi | by the way did you track the pkg? |
| 00:57.49 | p3nguin | I've got like a dozen packages to track, but for some reason I don't remember seeing that one on my list. |
| 00:57.52 | p3nguin | tracks it now. |
| 00:58.42 | p3nguin | It hasn't even been accepted into the system yet. |
| 00:58.51 | SeRi | ok |
| 00:58.53 | p3nguin | As in, not even scanned in at all. |
| 00:59.39 | SeRi | give it some time. It will update |
| 00:59.52 | p3nguin | I'm not worried. |
| 01:00.31 | SeRi | I am trully depress. |
| 01:00.48 | SeRi | This is an issue and I need a resolution |
| 01:00.54 | p3nguin | It should fade over time. |
| 01:01.40 | SeRi | trust me I am not going to get iver this |
| 01:02.03 | SeRi | s/iver/used to/ |
| 01:03.16 | SeRi | I all ready left a vmail to the sales rep. I want to cancelled |
| 01:03.24 | SeRi | it has nbot been 24hrs yet |
| 01:03.39 | SeRi | *cancell* |
| 01:06.03 | SeRi | I am out. going to go and crawl in to a corner and cry my self to sleep. |
| 01:06.26 | p3nguin | Are you really going to cancel, or was that just out of frustration? |
| 01:06.51 | dijib | you guys need 3rd party with network maintainence done by comcrap |
| 01:06.56 | dijib | sry sri. |
| 01:07.12 | SeRi | Frustration but nontheless something is going to happen. ether fixing or a reall cancellation |
| 01:07.23 | p3nguin | I would never let any person from ANY cable company touch my networks. |
| 01:08.00 | SeRi | p3nguin: Thats a good moto to live by. That fucker destroy my network |
| 01:08.04 | p3nguin | I don't even like that they touch the feedline bringing the service into the buildings. |
| 01:08.06 | SeRi | I wish you seen the mess eh made |
| 01:08.11 | p3nguin | What did he do? |
| 01:08.18 | SeRi | and my shit is wired perfectly |
| 01:08.39 | SeRi | he went and pull cables out the patch pannel and move shit around disconnected my servers |
| 01:08.46 | SeRi | so he can install THE FUCKING MODEM' |
| 01:08.52 | SeRi | omfg |
| 01:08.57 | p3nguin | What?! Why would someone do that?! |
| 01:08.58 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:09.00 | SeRi | I dont want to live any more |
| 01:09.05 | p3nguin | Did you tell him he could do that?` |
| 01:09.22 | SeRi | No i did not |
| 01:09.29 | p3nguin | Why would he do that? |
| 01:09.35 | p3nguin | That doesn't even make sense to do that. |
| 01:09.39 | SeRi | we got in to it and just turn in to shit quick |
| 01:09.49 | p3nguin | Did you yell at him? |
| 01:09.52 | SeRi | but me out of politness and been a bigger person let him finish |
| 01:10.13 | p3nguin | I would have never let that go. |
| 01:10.55 | SeRi | is still a mess in there. I fix what I could so far. |
| 01:11.19 | p3nguin | To install a modem or even a premise router, the patch panel should not be involved. You obviously had a modem before that, so all he needed to do was plug in the modem between the RG6 and the Cat 5e. |
| 01:11.21 | SeRi | I will rewire it when all my anger passes away. |
| 01:11.57 | SeRi | his argument was he could not get a good angle so he need to move cables |
| 01:12.56 | SeRi | I am done for the night |
| 01:13.02 | SeRi | I am going to cry on my sleep |
| 01:13.04 | SeRi | g/n all |
| 01:13.08 | Venaran | cya |
| 01:22.19 | *** join/#asterisk gusto (~gusto@ppp-93-104-68-243.dynamic.mnet-online.de) |
| 01:25.56 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 01:35.27 | *** join/#asterisk echo777 (~echo@71-82-226-158.dhcp.stpt.wi.charter.com) |
| 01:35.39 | echo777 | my CLI keeps freaking out and saying |
| 01:35.44 | echo777 | NOTICE[5707]: chan_sip.c:27641 handle_request_register: Registration from '<sip:1000@71.82.226.158>' failed for '188.161.19.68:56355' - No matching peer found |
| 01:37.07 | navaismo | seems like a sip attack |
| 01:37.27 | echo777 | how can that occur? |
| 01:37.56 | WIMPy | People want to make calls an let you pay for them. |
| 01:37.59 | navaismo | you have open to the world the sip port |
| 01:38.05 | WIMPy | That's perfectely normal. |
| 01:38.08 | echo777 | how do i resolve that |
| 01:38.21 | navaismo | dont open your the sip port if you dont need it |
| 01:38.23 | echo777 | and they are trying port 56414 and that not the port im using ha |
| 01:38.39 | WIMPy | generel internet secutity. |
| 01:38.48 | navaismo | set the allowguest=no if you dont need it |
| 01:39.05 | echo777 | í also have a jabber socket read error and im not using jabber how do i get rid of it? will allowguest=no hurt my config? |
| 01:39.07 | WIMPy | Pick anything from cut the internet connection to just ignore it. |
| 01:39.08 | navaismo | set strong password for your peers, dont use peers as same extensions |
| 01:40.32 | echo777 | ah they keep trying different ports, may i have to do with that fact that im using it with google voice? |
| 01:43.05 | WIMPy | It's the remote port and that has no meaning. |
| 01:43.47 | echo777 | i just dont see how it could be happening ha but it doesnt matter, theres nothing they can get into, i only have one extension and the password is strong |
| 01:47.55 | qakhan | i want to setup conf room, but i want to setup my own IVR on conf room and it authenticate pin# |
| 01:48.03 | navaismo | echo777, You underestimate the power of the attackers |
| 01:48.23 | echo777 | your not going to brute force something like lasfd3kljs |
| 01:49.13 | echo777 | its topped now anyway i believe |
| 01:49.32 | navaismo | qakhan, you can use playback for the greeting, then configure pins on meetme.conf |
| 01:49.32 | navaismo | im not an attacker |
| 01:55.28 | p3nguin | It's not the extension that they are trying to get into. |
| 01:55.33 | p3nguin | They are trying to register as your peer. |
| 02:07.45 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
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| 02:23.47 | dijib | who who here goes to rizon? |
| 02:24.03 | dijib | and again my appologies for being a tard and a hald |
| 02:24.05 | dijib | f |
| 02:44.52 | *** join/#asterisk pcAngel (~yoink@S0106602ad07d2b78.vc.shawcable.net) |
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| 02:56.49 | pcAngel | I've been having an issue getting ringinuse=no to work, but I think I just found the problem - is ringinuse=no unable to be set in [general] ? ie if only set there, will it cause other queues to default to ringinuse=no, or will it not affect the queues? |
| 03:01.58 | *** join/#asterisk elico (~Thunderbi@109.64.187.2) |
| 03:02.27 | pcAngel | Ahh that turned out to be accurate |
| 03:02.39 | pcAngel | damn that's been boggling me for years |
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| 03:09.03 | blizzow | do I want to run amportal start as root or asterisk? |
| 03:12.45 | qakhan | who provide cheap DIDs? |
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| 03:18.39 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 03:24.01 | asr33 | Hello folks, I've just upgraded my Asterisk to asterisk-1.8.14.1, I haven't changed any of my configurations from previous version, if I try dialing numbers internal/external I get this error message "WARNING[4839]: chan_sip.c:9208 process_sdp: We are requesting SRTP for audio, but they responded without it!" |
| 03:24.33 | pabelanger | warning != error |
| 03:25.06 | asr33 | is it possible to disable SRTP easily? |
| 03:25.37 | asr33 | yes warning ;-) |
| 03:27.48 | asr33 | I've tried to put "encryption=no" in [general] section of sip.conf without success? |
| 03:29.05 | asr33 | should I recompile with a special switch? |
| 03:30.00 | p3nguin | pcangel: Anything set in general should be the default value for all queues unless you override the value in the queue with another value. |
| 03:31.44 | p3nguin | I would think encryption=no would fix it. Make sure you run "sip reload" after you save that change. |
| 03:32.06 | asr33 | tries that |
| 03:32.44 | p3nguin | Also make sure you did not set it to yes in any peer. |
| 03:33.14 | asr33 | got ya, i'll have a look... |
| 03:34.35 | asr33 | still got the warning message :-( |
| 03:34.52 | p3nguin | That was my only guess. I don't use SRTP. |
| 03:35.15 | asr33 | wishes I didn't also ;-) |
| 03:44.35 | *** join/#asterisk Russ__ (~russ@ma15636d0.tmodns.net) |
| 03:45.59 | asr33 | I'm able to receive incoming calls but simply can't place outgoing calls |
| 03:53.01 | asr33 | p3nguin: I disabled SRTP on my smom 300 and now it works, please forgive me for disturbing you |
| 03:53.11 | *** join/#asterisk linuxnero (~linuxnero@119.92.147.65) |
| 03:56.10 | asr33 | happy, happy, happiness |
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| 04:31.43 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 04:46.46 | qakhan | can someone tell me good DID provider |
| 04:56.28 | Russ | teliax |
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| 07:54.17 | bombev | good morning to all |
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| 08:00.56 | ChannelZ | or in my case, goodnight |
| 08:09.24 | echo777 | hows it going guys? |
| 08:09.41 | echo777 | still up? |
| 08:10.42 | ChannelZ | up to no good |
| 08:10.44 | bombev | yes |
| 08:10.45 | bombev | :) |
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| 08:14.22 | bulkorok | up again... this morning :-/ |
| 08:14.42 | echo777 | where you from |
| 08:14.59 | bulkorok | DE |
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| 08:59.12 | *** join/#asterisk chris- (~chris@p5DD196C4.dip0.t-ipconnect.de) |
| 08:59.41 | chris- | hi |
| 09:00.51 | chris- | I wanted to implement in my AMI interface that it displays if I am connected or not. Sure I could make it with a variable, if I connect set it on 1... but if I restart the server the interface should see that it is not connected |
| 09:01.22 | chris- | is there anything I could use for it, I didn't find something yet |
| 09:07.05 | kaldemar | the manager interface will send you a shutdown event and your tcp connection will drop when you stop asterisk. |
| 09:07.39 | WIMPy | Good morning. |
| 09:08.35 | WIMPy | That question was rather unspecific. What connection? |
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| 09:19.12 | chris- | I mean, if I manually disconnect or connect to the server I can set a variable for it, but if I use core restart now for example the gui needs to know that it got disconnected |
| 09:19.59 | chris- | but I think kaldemar answered my question |
| 09:20.20 | chris- | thanks :) |
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| 12:18.33 | alsuren | when I call Dial(SIP/foo&SIP/bar,10), and SIP/foo explicitly rejects the call, SIP/bar currently keeps ringing. Is there any way to get a callback when SIP/foo hangs up so that I can drop the call? |
| 12:20.28 | WIMPy | callback? |
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| 12:21.19 | WIMPy | And otherwise I'd also like to see a BOB feature for Dial(). But you can do it in the dialplan if you need to. |
| 12:21.48 | WIMPy | Well, you can do it if you're only using SIP at least. |
| 12:23.26 | alsuren | BOB? |
| 12:24.43 | alsuren | WIMPy: I'm using phpagi, so being notified that one of the legs has been rejected would be good enough |
| 12:29.20 | alsuren | let's make the assumption that I'm only using sip for now: How would you do it? |
| 12:30.19 | WIMPy | BOB = Busy On Busy |
| 12:30.40 | WIMPy | And the dialplan solution is to check DEVICE_STATE before dialing. |
| 12:31.05 | WIMPy | AMI would also tell you about each single channel. |
| 12:31.28 | WIMPy | AGI won;t do. |
| 12:34.57 | alsuren | DEVICE_STATE doesn't sound like what I'm looking for. The use case is that each user has multiple devices that ring, and they all stop ringing and send the call to answerphone immediately if he rejects the call |
| 12:35.39 | WIMPy | Nope. That wouldn;t work on a manual reject. |
| 12:36.29 | WIMPy | The easiest solution is probably to give the user a key to transfer the call to VM instead of rejecting it. |
| 12:37.51 | alsuren | but how would that be implemented on the SIP side? |
| 12:38.38 | WIMPy | Depends on the phone in use. |
| 12:38.54 | alsuren | let's say I control my client |
| 12:39.26 | alsuren | so I can implement it however I want. |
| 12:41.25 | alsuren | if I wanted to add an option like AST_DIAL_OPTION_BUSY_ON_BUSY, I should follow https://wiki.asterisk.org/wiki/display/AST/New+Feature+Process, I guess? |
| 12:41.43 | alsuren | -> lunch |
| 12:43.01 | WIMPy | It might be pretty simple to add. |
| 12:44.33 | alsuren | I might pick your brains on the "send to voicemail button" idea after lunch. |
| 12:45.10 | WIMPy | is currently having lunch :-) |
| 12:50.02 | *** join/#asterisk spooky7 (~spooky@ppp-94-67-105-3.home.otenet.gr) |
| 12:52.06 | spooky7 | Good morning . Need some help . I have centos 6.3 iaxmodem , hylafax. and asterisk. I have put in /etc/inittab the next entry m0:2345:respawn:/usr/sbin/faxgetty ttyIAX0 but faxgetty doesn't tun . What am I doing wrong ? |
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| 13:00.28 | bulkorok | hey... I have a segfault with receivefax... somebody can help to inspect this?! |
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| 13:06.35 | ectospasm | bulkorok: is it Digium's FAX, or FFA? |
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| 13:12.57 | bulkorok | it's res_fax I suppose... currently we have spandsp |
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| 13:24.41 | p3nguin | Digium's fax IS ffa. |
| 13:27.42 | SeRi | p3nguin: whats going on |
| 13:28.20 | p3nguin | Just starting up. |
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| 13:29.43 | metiu | hi, I was asking some time ago about a problem with skewed packets with a confbridge with only 2 participants, A* 11.0.1. I was told to check which timing module I was using. I had timerfd, dahdi, pthread timing modules all enabled. I tried to leave just one of them enabled in turn, but the problem is still there: the audio is stuttering badly, due to skew |
| 13:29.58 | metiu | anything else I could check? |
| 13:33.42 | SeRi | man I had a bad dream that I moved to comcast business and everything broke... |
| 13:35.36 | SeRi | o wait that was not a dream. |
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| 13:45.24 | metiu | funny: timing test fails (1 tick in 2600ms) for some time, then after some tests it goes well (1004ms for 50 ticks) (pthread) |
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| 13:46.15 | kaldemar | metiu: i've mostly had bad results with pthread. are dahdi and timerfd also unusable for you? |
| 13:46.33 | metiu | trying right now |
| 13:47.42 | metiu | kaldemar: timerfd test is great, I'm trying a call now |
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| 13:54.14 | metiu | No luck: timing test passes with 1000ms for 50 ticks, but sound stutters badly and skew grows ... more than 6 seconds in 116 RTP packets |
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| 13:55.16 | qakhan | i setup conf room in asterisk and i also setup my own IVR |
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| 13:55.47 | qakhan | but when mine IVR ran after that asterisk ivr run to enter conf pin # |
| 13:55.53 | qakhan | what is the problem |
| 13:56.18 | alsuren | WIMPy: my workmate reminded me that the interactions between call forking and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics (Forcing bridged channels to be secure) is likely to be horrible. |
| 13:57.07 | qakhan | http://pastebin.com/T8HH88p5 |
| 13:58.50 | kaldemar | qakhan: what do you expect to happen? |
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| 14:05.44 | metiu | dahdi does the same |
| 14:05.47 | slav3_kitten | qakhan, why are you not using confbridge |
| 14:05.56 | metiu | should I file a bug report, or ask on the mailing list? |
| 14:06.26 | slav3_kitten | i'd ask an confirm it's a bug |
| 14:06.30 | slav3_kitten | not a conf issue |
| 14:06.31 | jmetro | slav3_kitten : notice he is using a | instead of commas. old version of asterisk? |
| 14:06.43 | slav3_kitten | good point |
| 14:08.27 | metiu | FWIW, the RTP packet analysis by wireshark is here: http://pastebin.com/06RWnpbh |
| 14:09.16 | qakhan | kaldemar i want to play my own ivr on conf room to enter conf pin # rather then play asterisk default IVR to enter conf pin # |
| 14:09.58 | jmetro | If you just want your own sound file, you could try replacing that part of it. |
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| 14:10.30 | [TK]D-Fender | qakhan, Just asking for a single field of input is a job for READ(), not an "ivr". |
| 14:10.52 | kaldemar | qakhan: then you must configure asterisk to read user input and not ask PIN for MeetMe. |
| 14:11.01 | [TK]D-Fender | qakhan, And what you put there is NOT a proper "ivr" by any definition there. It does not wait for input and then do anything with it. You have no EXTEN's in there to match anything or do anything with it |
| 14:11.44 | [TK]D-Fender | qakhan, and | <-- is not a proper delimiter for dialplan parameters since 1.4 |
| 14:14.08 | qakhan | kaldemar i didnt get you |
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| 14:17.52 | p3nguin | One sound prompt doesn't constitute an IVR? |
| 14:18.01 | qakhan | [TK]D-Fender where i put read(), |
| 14:18.18 | qakhan | after background()? |
| 14:18.56 | kaldemar | qakhan: do you know what the Background application does? |
| 14:19.00 | jmetro | p3nguin: i think he means because he is just asking for a conference pin, not displaying a list of options. basically authenticating not routing. |
| 14:19.49 | p3nguin | qakhan: Read() can play the same sounds that BackGround() can play. And it will wait for input. The input will be used to set a variable by Read(). |
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| 14:20.33 | qakhan | jmetro you are right, that is what i want to do |
| 14:21.37 | kaldemar | and for that purpose there is also app Authenticate. |
| 14:21.51 | p3nguin | Seems like a better app to be using for just a PIN. |
| 14:23.23 | [TK]D-Fender | qakhan, You don't use background. At all |
| 14:23.30 | [TK]D-Fender | qakhan, background is for IVR's |
| 14:23.34 | p3nguin | Actually, since Authenticate() can also play any sound files you want to play, it really comes down to how many different PINs you need to manage. |
| 14:23.52 | p3nguin | Usually Read() is for IVRs just as much. |
| 14:24.48 | [TK]D-Fender | p3nguin, Not quite "usual", but depending on what you have to match and how you deal with band input, sure... |
| 14:25.29 | p3nguin | I would even go as far as saying Read() is used MORE for IVRs than BackGround() is. |
| 14:25.47 | p3nguin | BackGround() is more for an auto-attendant. |
| 14:27.06 | [TK]D-Fender | p3nguin, Yes, I suppose I should should start differentiating IVR vs AA more. I still use IVR where AA is a more appropriate term |
| 14:27.18 | [TK]D-Fender | (terminology-wise) |
| 14:27.23 | qakhan | [TK]D-Fender how read() will authenticate conf pin # with meetme.conf |
| 14:27.42 | [TK]D-Fender | qakhan, There is nothing to validate against meetme.conf |
| 14:27.50 | p3nguin | If you are using MeetMe, it does it all for you. |
| 14:28.00 | p3nguin | Look at the syntax for the app. |
| 14:28.04 | [TK]D-Fender | qakhan, If you want to look that up then you wil lhave to do real scripting ourtside of * to parse that file. |
| 14:28.16 | p3nguin | MeetMe() does it. No scripting needed. |
| 14:28.30 | [TK]D-Fender | p3nguin, He doesn't want to use the PIN lookup in the app itself. His square wheel will be ROUNDER.... or so he believes. |
| 14:29.08 | p3nguin | Then he can't be helped. |
| 14:29.47 | [TK]D-Fender | p3nguin, He's been in here for a year & a quarter and can't get past 5 lines of dialplan. I concur. |
| 14:30.16 | p3nguin | Is it comedy or sadness? |
| 14:30.24 | paryl | speaking of MeetMe... I'm still having the same problem I asked about yesterday. No matter what combination of options I use, all regular users enter into the conference muted. Only admins can hear/be heard. |
| 14:30.59 | p3nguin | Don't use option 'm' in the dialplan and don't set any options for the conf room in meetme.conf. |
| 14:31.39 | paryl | p3nguin: i'm using realtime, if that matters, but the row in the database just has 'MeetMe' for the app... no options |
| 14:31.57 | p3nguin | Nothing in meetme.conf? |
| 14:32.00 | paryl | but that's the strange thing... none of the options i set (if i set any) seem to make any difference |
| 14:32.26 | p3nguin | There's no option that you can add to make it not muted. Not muted is the lack of option m. |
| 14:32.42 | paryl | the only things in meetme.conf are fuzzystart=300 and earlyalert=600 |
| 14:33.07 | p3nguin | What do you see being executed on the CLI when the meetme conf starts? |
| 14:34.05 | alsuren | Further to my earlier question, do the new https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers get called for each leg of a forked call (using Dial(SIP/foo&SIP/bar,10)) if they get rejected? |
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| 14:36.16 | [TK]D-Fender | alsuren, That is not "forked" |
| 14:37.28 | paryl | p3nguin: i don't see any meetme-specific options in the log. it shows that it's updating the member count for the channel, so it knows 2 people are joined... i don't see anything specifically saying channels were muted though |
| 14:38.12 | salz212 | Guys in Asterisk 11.0.1 is there any way to log AGI Verbose other than cli command. Previous I was using AGIVERBOSE = 9 in extensions.conf but it is not working now in 11.0.1 |
| 14:38.25 | salz212 | agi set verbose is temporary and its clumsy . |
| 14:38.26 | [TK]D-Fender | alsuren, Actaully.. reading that link... this is a new feature whose consequence I think I should read up on first... |
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| 14:43.05 | kaldemar | salz212: do you mean agi debug? |
| 14:44.26 | salz212 | yep, 'agi set debug on' server the same purpose. But I am looking for AGIVERBOSE=some_value in extensions.conf to work. Point is I am unable to log AGI logs in messages. |
| 14:45.18 | salz212 | The only way working is agi set debug on but it is also appending extra agi resource info which is clumsy and also not indented to be read. AGIVERBOSE=9 works in Asterisk 1.6.x |
| 14:45.36 | salz212 | Not sure if it works in 11 or not that is what I am here to confirm, |
| 14:45.40 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:45.52 | kaldemar | there is no such thing in vanilla asterisk. |
| 14:46.37 | kaldemar | salz212: what does you AGIVERBOSE do exactly? |
| 14:46.55 | kaldemar | salz212: is it a global variable or something under general? |
| 14:47.21 | salz212 | yes globals |
| 14:48.07 | kaldemar | salz212: what does you dialplan do with it? |
| 14:49.42 | p3nguin | Setting some arbitrary variable doesn't enable asterisk's magic mode, so the dial plan must be using it. |
| 14:50.10 | paryl | the meetme docs talk about needing the zaptel driver... i have all of the wct and dahdi modules for my card, and i was under the impression that they took the place of the zaptel drivers, so i don't have any zaptel modules loaded. could that be an issue for meetme? |
| 14:50.30 | p3nguin | What version of asterisk are you using? |
| 14:51.06 | paryl | p3nguin: 1.6.2.9-2+squeeze4 |
| 14:51.15 | kaldemar | paryl: zaptel was renamed to DAHDI years ago. DAHDI will do just fine. |
| 14:51.28 | paryl | kaldemar: ok, thanks for the confirmation :) |
| 14:51.33 | p3nguin | I don't think that version is old enough to be using zaptel. |
| 14:51.42 | jmetro | "asterisks magic mode" |
| 14:51.42 | metiu | kaldemar: do you think I can make any other test to timings, or should I file a bug report? |
| 14:52.07 | paryl | like i said though... meetme works fine as long as you use the admin passcode. my users are just getting sick of giving that code out |
| 14:52.33 | p3nguin | Let's look at what commands run in "admin mode" and "user mode." |
| 14:52.43 | p3nguin | Feel free to pastebin the verbose CLI output. |
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| 14:55.40 | paryl | i'm not sure if i'm missing something or not... i have debug and verbose both set to 3, and all i see is the main command from the dialplan, which is - Executing MeetMe("SIP/<number>-000008cc", "") |
| 14:56.04 | p3nguin | That looks like no options are being used. |
| 14:56.32 | paryl | right, which means it should all 'just work', right? |
| 14:56.40 | paryl | that's why i'm so confused by this issue |
| 14:57.18 | p3nguin | Unless there is some option set elsewhere, I would expect callers to be non-muted when they join. |
| 14:57.30 | paryl | the odd thing is, even if i set options, they don't make sense. I set M, which is supposed to be music on hold, but instead it takes the user to a 'record your name' prompt |
| 14:57.56 | p3nguin | Have you considered using a recent version? Maybe you have bugs. |
| 14:58.44 | paryl | i was wondering about that. i've just been using the one that is in the squeeze repo up till now, that's why it's kind of old |
| 14:59.14 | p3nguin | See if you can get a newer version out of one of the repos. |
| 14:59.42 | paryl | alright, i'll give that a try. thanks for your help! |
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| 15:05.13 | alsuren | [TK]D-Fender: yeah, I know: SIP call forking is more low-level than that. What term should I use instead of forking? |
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| 15:11.55 | alsuren | WIMPy: re "send to voicemail button": are you referring to the 't' option to Dial? "Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf" |
| 15:12.13 | alsuren | or something else? |
| 15:12.24 | WIMPy | alsuren: No to the phones transfer option. |
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| 15:12.45 | WIMPy | Features only work for active calls. |
| 15:12.50 | [TK]D-Fender | alsuren, They are really just spawned channels from the dial, but I'm not sure what you're actually trying to accomplish... when you linked that page it made me wonder what kind of complex thing you'd be trying to do given it's nature |
| 15:13.06 | p3nguin | its |
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| 15:13.55 | [TK]D-Fender | p3nguin, yeah yeah yeah.... |
| 15:14.24 | alsuren | WIMPy: so the client instantly picks up the call, and then transfers it to voicemail rather than rejecting it? |
| 15:15.02 | WIMPy | No it doesn't pick up. It deflects the call. |
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| 15:16.12 | alsuren | WIMPy: what SIP feature should I be googling for? |
| 15:16.32 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 15:16.45 | WIMPy | I think that's a REFER. |
| 15:18.03 | alsuren | looks into it |
| 15:20.31 | alsuren | [TK]D-Fender: I'm trying to accomplish parallel ringing of both SRTP-supporting and non-SRTP supporting clients |
| 15:21.31 | WIMPy | What do you (think you) have to care about for that? |
| 15:21.38 | [TK]D-Fender | alsuren, Ok, where does the use of hangup handlers come in? |
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| 15:22.40 | WIMPy | Hmm. Since the notify on subscription thing was fixed, I get a nice light show on the phone when restarting Asterisk. |
| 15:24.11 | alsuren | see: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics (Set(CHANNEL(secure_bridge_signaling)=1) then wait for the client to tell you to fuck off and redial with Set(CHANNEL(secure_bridge_signaling)=0)) |
| 15:25.15 | alsuren | [TK]D-Fender: I'm basically brainstorming here. It might not even work. If you know of a better way to do it, give me a shout. |
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| 15:33.22 | a1fa | morning |
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| 15:46.37 | ideaman55 | Can anyone tell me what the default MTU is for IAX2 in 1.8? |
| 15:47.23 | WIMPy | ideaman55: No, but I can tell you that the question doesn't make sense. |
| 15:47.34 | ideaman55 | How so? |
| 15:48.02 | WIMPy | IAX2 is not a transport for IP. |
| 15:48.04 | p3nguin | Wrong layer for such a setting. |
| 15:48.25 | ideaman55 | Then I guess I misunderstood what IAX2 set mtu meant |
| 15:48.30 | ideaman55 | What is that for? |
| 15:48.49 | [TK]D-Fender | ideaman55, On a scale of 1 to 10, what is your favourite color of the alphabet? |
| 15:50.12 | ideaman55 | Understood it's the wrong layer, so does setting not change anything, then? |
| 15:50.21 | WIMPy | Never noticed that setting. But it tells you that it will make Asterisk handlefragmentation so the IP layer doesn't have to. |
| 15:50.57 | WIMPy | So unless you know the transports used all along the way, it's probably best left off. |
| 15:50.59 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 15:51.09 | wltjr_ | anyone have experience with cisco spa112 and credit card processing? having to make several attempts at times to process cards |
| 15:51.21 | ideaman55 | Wimpy: thanks |
| 15:52.00 | WIMPy | If you have a big trunk it might save some load however. |
| 15:52.07 | [TK]D-Fender | wltjr_, trying to pump data over a VoIP device like that is asking for trouble. |
| 15:52.26 | WIMPy | wltjr_: That's not supposed to work at all. |
| 15:52.30 | [TK]D-Fender | wltjr_, There is a question of the code & networking condition jsut SPA>*, then there is the OTHER leg of the call. |
| 15:52.56 | wltjr_ | I used to do cc processing without problem using Cisco 827-4v |
| 15:53.21 | WIMPy | Pure luck. ger an IP terminal. |
| 15:53.21 | p3nguin | Even where we remove all other POTS phones and switch to VoIP only, we still leave one line for the credit card terminal. |
| 15:53.30 | wltjr_ | not sure if its network traffic, or the spa not being dialed in, thinking of switching the spa to tcp vs udp |
| 15:53.46 | [TK]D-Fender | wltjr_, And you haven't filled in ANY details about either leg of the call as provided now. |
| 15:54.24 | wltjr_ | [TK]D-Fender: sry what details would you like, there are lots |
| 15:55.15 | [TK]D-Fender | wltjr_, Codecs, networking, description of the other leg. |
| 15:56.19 | wltjr_ | [TK]D-Fender: lines come in via sip trunk, pretty sure just ulaw and t38, all sip, trunk and phones/ata |
| 15:57.15 | [TK]D-Fender | wltjr_, T.38 is for faxing, not for CC last I checked... |
| 15:57.21 | wltjr_ | [TK]D-Fender: internet is comcast cable 100Mbs/10mbps, network is highly used, but no where near saturation, line is shared by 3 networks, voip, office lan, and public wirelesss |
| 15:57.24 | [TK]D-Fender | wltjr_, and "pretty sure" doesn't cut it... |
| 15:57.47 | p3nguin | Comcast, you say? |
| 15:57.52 | p3nguin | There's yer problem. |
| 15:57.57 | wltjr_ | [TK]D-Fender: I am not changing any codecs, nor have any set, default from sip provided and within * |
| 15:58.01 | [TK]D-Fender | wltjr_, And without QoS to your provider provided end-to-end, expect failures |
| 15:58.32 | wltjr_ | [TK]D-Fender: I thought QoS might be an issue, presently not doing any, and considering a stand alone line for VOIP |
| 15:58.52 | p3nguin | You really need to set your codecs, otherwise a codec will be chosen all willy-nilly. |
| 15:59.18 | wltjr_ | [TK]D-Fender: aren't fax transmissions and credit cards similar? thought bought would need t38 support |
| 15:59.38 | p3nguin | If your ISP doesn't provide any type of QoS, you can still implement traffic shaping on your outbound traffic. |
| 16:00.09 | [TK]D-Fender | wltjr_, Similar != same |
| 16:00.27 | wltjr_ | in sip.conf I am only allowing ulaw, pretty sure t38 doesn't go in the same codecs area |
| 16:01.12 | coppice | fax transmissions are nothing like credit card ones, although some credit card transactions use a modified V.29 modem, which is a FAX modem |
| 16:02.08 | [TK]D-Fender | Which is ... hardware that can also do something that isn't the same. |
| 16:02.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:02.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:06.09 | wltjr_ | think I might just look to replace analog cc terminals with ones that work via the internet, that will correct most of the problem, not seeing as many if any problems with fax |
| 16:08.33 | *** join/#asterisk tuxfarva (hidden-use@2620:3f:8000::51) |
| 16:21.10 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:24.02 | *** join/#asterisk lukejt (~luke@149.241.199.189) |
| 16:27.43 | lukejt | Am I missing something obvious? Using AGI and channel variables such as DIALSTATUS and ANSWEREDTIME are only available when using the DIAL command? |
| 16:27.45 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 16:28.38 | Qwell | DIALSTATUS is the STATUS of the DIAL, so...there's that. |
| 16:29.02 | lukejt | If I answer the call, stream a file and hangup, they are missing, could anyone please explain why that would be the case? |
| 16:29.30 | WIMPy | Qwell just did. |
| 16:30.27 | lukejt | Qwell: Thanks, makes sense. |
| 16:30.30 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 16:31.25 | lukejt | WIMPy: Didn't realise the question was answered :) |
| 16:34.04 | pabelanger | Anybody know a good Polycom vendor around Kansas? Need to order / dropship a few phones |
| 16:34.49 | rrittgarn | e4voip is in MI and dropships |
| 16:34.59 | rrittgarn | (not close to kansas persay but its who we use) |
| 16:35.06 | rrittgarn | I can get you a contact if you'd like |
| 16:35.11 | rrittgarn | he was actually @ Astricon |
| 16:40.34 | SeRi | did somebody sa comcast? |
| 16:40.42 | SeRi | yeap thats a problem. |
| 16:43.55 | p3nguin | rrittgarn: per se |
| 16:44.23 | p3nguin | seri: Yes. |
| 16:46.30 | SeRi | lol |
| 16:46.38 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 16:46.43 | SeRi | my issue has caused a service outage for my area. |
| 16:46.46 | SeRi | how nice! |
| 16:47.00 | p3nguin | Good. At least that should make you feel better. |
| 16:47.17 | SeRi | lol well they are "working" on it. |
| 16:47.29 | p3nguin | They're always working on it. |
| 16:47.33 | SeRi | hahahaha |
| 16:47.49 | p3nguin | That's part of the reason it is always broken. |
| 16:49.02 | SeRi | so true |
| 16:50.52 | p3nguin | sccp set hold 161 on |
| 16:51.01 | p3nguin | There's another thing I like about my Cisco phones. |
| 16:51.09 | p3nguin | I can put a call on hold from the CLI. |
| 16:51.16 | SeRi | nice |
| 16:51.47 | lukejt | p3nguin: can you give an example of how that would be useful? I can't picture it |
| 16:52.10 | lukejt | just out of curiosity :) |
| 16:52.16 | p3nguin | I only have one use for it... |
| 16:52.54 | p3nguin | Each Friday, the VUC meets. I have a cron job that automatically dials in and my phone autoanswers if I am not on it... |
| 16:53.05 | p3nguin | I am not at my desk, but I just saw the call start to my phone. |
| 16:53.19 | p3nguin | Since I am not at my desk, I put it on hold until I go there. |
| 16:54.31 | p3nguin | I could have just as easily destroyed the call completely, and called back when I was ready to join the conference, but I prefer to just put it on hold. |
| 16:54.55 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 16:54.55 | WIMPy | And play MOH to the conference? |
| 16:55.06 | p3nguin | Of course not. That would be rude and silly. |
| 16:55.11 | lukejt | MOH disabled? |
| 16:55.13 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:4:0:414:11) |
| 16:55.29 | lukejt | nvm |
| 16:55.31 | WIMPy | But the usual behaviour when using SIP with Asterisk. |
| 16:55.47 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 16:57.03 | p3nguin | Even if moh would play, all calls are initially muted. |
| 16:57.55 | WIMPy | I guess they had Asterisk users before. |
| 16:58.42 | p3nguin | smirks |
| 16:59.59 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:00.11 | p3nguin | I would venture to say that a good majority uses asterisk. |
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| 17:10.18 | ghost75 | do i need to set callerid manually with set before i do a call with originate? |
| 17:10.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:10.42 | Qwell | Need? No. |
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| 17:10.58 | ghost75 | callerid doesnt appear in cdr |
| 17:11.00 | p3nguin | That depends on what you want to happen and on your dial plan. |
| 17:11.24 | ghost75 | without dialplan |
| 17:11.42 | p3nguin | How will you set the caller id without dial plan? |
| 17:12.00 | ghost75 | AMI |
| 17:12.34 | ghost75 | the originate gets started over webbrowser |
| 17:12.50 | p3nguin | Unless you have already originated the call, there is no channel on which to set a caller id. |
| 17:13.17 | ghost75 | the originate cmd also has no option to set callerid? |
| 17:13.34 | Qwell | ghost75: What does the documentation say? |
| 17:15.48 | ghost75 | ok i found |
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| 17:17.59 | p3nguin | If you want to set caller id, use one or two lines of dial plan and then origiante against a local channel using that new bit of dial plan. |
| 17:18.51 | p3nguin | I guess you'll need at least two: one for the Set() and one for a Dial(). |
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| 17:20.22 | ghost75 | callerid can be specified directly in the originate manager cmd |
| 17:20.40 | p3nguin | I was not aware of that. That would be very useful. |
| 17:20.48 | ghost75 | indeed |
| 17:20.54 | p3nguin | I rarely use the AMI. |
| 17:21.05 | p3nguin | I do most things via CLI or shell script. |
| 17:21.11 | p3nguin | (which uses the cli) |
| 17:21.23 | ghost75 | what was again cli command to display cmd help? |
| 17:21.28 | Qwell | help |
| 17:22.55 | jmetro | with what, qwell? =P |
| 17:23.02 | p3nguin | [CallerID:] <value> |
| 17:23.05 | p3nguin | Nice. |
| 17:23.09 | p3nguin | That could be very useful for me. |
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| 17:25.18 | ghost75 | wasnt this in help core? |
| 17:27.21 | p3nguin | I saw it in manager show command Originate. |
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| 17:27.47 | ghost75 | yes but i look the documentation of commands in dialplan or cli |
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| 17:49.01 | mathi | hi |
| 17:49.20 | mathi | anyone has experience with the Asterisk MyPBX embedded system ? |
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| 17:58.59 | psilikon | I'm using Asterisk 1.4 and it's 'Monitor' application to record meetme conferences "Monitor("Local/58600054@default-61f4,1", "wav|recording_name")". Sometimes the callee volume is extremely low and I can't figure out how to correct this. Can anyone shed any light on this? |
| 18:00.52 | pcAngel | When that's happened to me I've generally found out that the agent's microphone is not close enough to the mic on their headset |
| 18:00.59 | pcAngel | and in that case, that's how the customer actually hears them |
| 18:01.12 | pcAngel | oh sorry, callee |
| 18:01.24 | pcAngel | does the agent have to crank up their headset to hear them? |
| 18:02.26 | psilikon | There is a Sangoma a104d in that server which is only used at moment for timing since all trunks are voip. I am wondering if any changes I have made the 'chan_dahdi.conf' would affect volume levels since I thing dahdi is involved with the meetme rooms |
| 18:02.52 | psilikon | pcAngel: The live audio is fine. Only after a recording is made do I notice a problem. |
| 18:02.59 | pcAngel | dahdi is only used for timing when VOIP is involved |
| 18:03.09 | pcAngel | Btw I'm assuming you know how to switch to MixMonitor & are using Monitor for a specific reason-- is that accurate? |
| 18:03.17 | psilikon | Yes |
| 18:03.18 | pcAngel | MixMonitor has an app flag to increase the volume on one track |
| 18:03.57 | psilikon | pcAngel: I believe MixMonitor cannot be used on conferences, correct? |
| 18:06.03 | p3nguin | If you're using MeetMe, it has its own record option. |
| 18:07.44 | Qwell | ~upgrade asterisk |
| 18:07.44 | 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. |
| 18:07.46 | Qwell | psilikon: ^ |
| 18:07.59 | psilikon | It's a Vicidial system in which an agent has a web interface that can actually initiate a recording via button click and a ajax call. |
| 18:08.51 | psilikon | Unfortunately I can't upgrade the Asterisk version since 1.4 is the only supported version with vicidial. |
| 18:09.26 | pcAngel | psilikon: MixMonitor should work on the channel, I use that all the time, although my CRM platform uses Asterisk 10 |
| 18:09.39 | psilikon | I was kinda poking around and hoping for some insight to see if the Sangoma card was 'in the loop' when it comes to the Monitor application. |
| 18:10.21 | Qwell | psilikon: have them help you then |
| 18:10.52 | Qwell | 1.4 is dead. Nobody cares anymore. It's not worth trying to debug stuff like that, since it will never be fixed. |
| 18:11.07 | psilikon | Qwell: I hear ya. |
| 18:11.41 | *** join/#asterisk ThinkGNU- (~ThinkGNU@216.67.183.210) |
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| 18:12.13 | wltjr_ | I am having problems detecting when a line on a phone is in use so I can ring others that are not, I am doing it this way atm http://pastebin.com/54YEuJc3, I had to set busylevel and call-limit to 1 for the lines to show as busy, but it breaks call forwarding |
| 18:12.14 | ThinkGNU- | anyone know of a good VoIP gateway that has multiple ports and can have those ports configure as FXS or FXO? |
| 18:12.36 | Qwell | wltjr_: Set call-limit higher. |
| 18:12.50 | wltjr_ | Qwell: when I do, the line never shows as busy, and it rings the line in use |
| 18:13.13 | p3nguin | Turn off call waiting. |
| 18:13.17 | wltjr_ | Qwell: I can try, because I have to increase that for call transfering to work |
| 18:13.34 | wltjr_ | p3nguin: ah maybe thats it, but doesn't act like call waiting, just rings the same line/extension |
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| 18:14.18 | p3nguin | If call waiting is disabled on the phone, it cannot accept a second call. The DIALSTATUS will be BUSY if asterisk tries to Dial() it. |
| 18:15.19 | wltjr_ | p3nguin: ty good to know, now to find that setting and disable it |
| 18:15.19 | p3nguin | If the phone has call waiting enabled, the second call will behave exactly as the first in the event of a NOANSWER. |
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| 18:16.32 | wltjr_ | p3nguin: then I can lose the settings in sip.conf entirely I believe, since I did that to force the busy status |
| 18:17.08 | p3nguin | You can also use ChanIsAvail() to check if a channel is in use before attempting a Dial(). |
| 18:20.19 | *** join/#asterisk kontinuity (~kontinuit@122.167.218.155) |
| 18:20.31 | p3nguin | That's what I use to determine if FollowMe() should be executed or not. I got tired of my cell phone ringing while I was on my desk phone but didn't answer the call waiting. |
| 18:20.44 | wltjr_ | p3nguin: I was looking into that, but never got the right status, likely cause of call waiting |
| 18:21.04 | p3nguin | I use ChanIsAvail with option s. |
| 18:21.18 | p3nguin | Then I test for AVAILORIGCHAN. |
| 18:21.32 | p3nguin | If it is null, the phone is in use. |
| 18:21.40 | p3nguin | If there is a value, the phone is free. |
| 18:22.04 | wltjr_ | p3nguin: good to know, I might have to go that route, not sure yet |
| 18:23.55 | p3nguin | wltjr_: http://pastebin.com/8dzGfuXJ |
| 18:25.47 | p3nguin | It'll work. |
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| 18:29.13 | p3nguin | I'll even give you a hand with it if it doesn't work for you. |
| 18:33.59 | SeRi | so they claim is now fixed |
| 18:34.05 | p3nguin | Is it? |
| 18:34.15 | SeRi | I am at work. cant test right now |
| 18:34.30 | SeRi | I did called earlier and sip was still bad |
| 18:34.52 | SeRi | my wife soud like a robot |
| 18:35.00 | SeRi | so ill check later. |
| 18:35.26 | SeRi | pathload v2 |
| 18:35.32 | SeRi | seems to show good results |
| 18:35.46 | SeRi | 32MBps download |
| 18:35.53 | SeRi | 8MBps upload |
| 18:36.29 | p3nguin | Can't test it? lrn2ssh! |
| 18:36.44 | SeRi | I am in over ssh right |
| 18:36.46 | SeRi | now |
| 18:36.52 | SeRi | I just posted a test |
| 18:36.56 | p3nguin | Test it, then! |
| 18:37.07 | wltjr_ | p3nguin: thats interesting, I need to look at it a bit so I can understand it, not sure if I need to modify it for my needs |
| 18:37.17 | p3nguin | I guess I got confused by the statement, "I am at work. cant test right now." |
| 18:37.22 | SeRi | cant right now. In a conf at work. |
| 18:39.10 | p3nguin | wltjr_: It's the concept that you will use. |
| 18:47.09 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 18:48.03 | JasonL | [TK]D-Fender: Do you remember helping me the other day with PRI passthough and echo? We had turned off echo cancellation and restarted asterisk... we're still getting echo on all calls... any other ideas? |
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| 18:49.25 | rrittgarn | anybody work with Cisco SPA series much? When a caller gets a busy signal (using congestion(5)) the phone displays Service Unavailable, which is making the users think there is something wrong with the system, when in fact its just a busy number. |
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| 18:50.47 | p3nguin | wltjr_: Reload http://pastebin.com/8dzGfuXJ |
| 18:51.06 | SeRi | p3nguin: Yes. All fixed. well I have to say that shit cc business did do the trick. |
| 18:51.16 | SeRi | 3yrs of issues with home class |
| 18:51.22 | SeRi | 2 days on comcast business |
| 18:52.37 | [TK]D-Fender | JasonL, swap card slots, re-enable EC. Test card independant of bridging. |
| 18:52.49 | [TK]D-Fender | JasonL, Confirm Echo as during call VS recording. |
| 18:53.25 | [TK]D-Fender | rrittgarn, Congestion != number busy |
| 18:53.49 | [TK]D-Fender | Congestion can be a blockage in the path, not just the final number reporting busy |
| 18:54.52 | rrittgarn | ah so i assume using the Busy() application would probably solve that a bit better |
| 18:55.29 | p3nguin | It's probably a bad idea to use a Congestion when you mean Busy. |
| 18:55.35 | kaldemar | rrittgarn: or use Hangup with cause code 17. |
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| 18:56.43 | JasonL | [TK]D-Fender: yes, during call, and not the recording |
| 18:57.00 | JasonL | [TK]D-Fender: only one PCI-e slot in the chassis |
| 18:57.16 | *** part/#asterisk leedm777 (~leedm777@nat/digium/x-gkleefoordgywuqr) |
| 19:00.45 | [TK]D-Fender | JasonL, Oh well... what card? |
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| 19:01.56 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 19:02.17 | JasonL | TE420 |
| 19:02.31 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 19:02.33 | JasonL | with no echo cancellation |
| 19:03.06 | *** join/#asterisk tuxfarva (hidden-use@chtest46.gta.com) |
| 19:05.22 | Snivets | so, I'm trying to enable saycid (in the voicemail conf) for only a single client. If my root /asterisk/voicemail.conf has a bunch of #includes with specific files, and each of those files defines the client's extensions in a context specific to them, can I just put "saycid = yes" at the top of that context? |
| 19:05.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:05.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:06.23 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:07.19 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:09.02 | *** join/#asterisk ross` (~ross@spam.im) |
| 19:09.12 | ross` | does this channel like/approve of debian asterisk |
| 19:09.24 | ross` | or does debian asterisk make ppl angry here |
| 19:09.58 | p3nguin | Is debian asterisk different from the asterisk everyone else uses? |
| 19:11.19 | ross` | that is what i'm asking |
| 19:11.32 | ross` | i'm on a debian server wondering if i should build it from source or if debian's will work |
| 19:11.59 | p3nguin | The main thing is what version you will get/use if you choose one from a repo. |
| 19:12.19 | ross` | are there good reasons to get an older version |
| 19:12.22 | p3nguin | You need to get the most recent version in a branch to get support (if you need support). |
| 19:12.46 | p3nguin | Old versions have bugs. That is the reason to make newer versions. |
| 19:13.58 | ross` | Asterisk 1.6.2.9-2+squeeze8, Copyright (C) 1999 - 2010 Digium, Inc. and others. |
| 19:14.09 | ross` | i take it that version isn't supported |
| 19:14.55 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 19:17.53 | [TK]D-Fender | ross`, 1.6.x is no longer supported, and that is ancient even within its branch |
| 19:18.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:18.31 | ross` | does #asterisk support editing configuration files or using the asterisk cli or using asterisk --flags to do server operations |
| 19:20.24 | [TK]D-Fender | ross`, "server operations"? |
| 19:20.28 | [TK]D-Fender | ross`, huh? |
| 19:22.28 | ross` | for making changes |
| 19:22.33 | ross` | like user changes etc |
| 19:22.34 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 19:23.54 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:23.58 | [TK]D-Fender | ross`, user changes = config files. How do you think you'd do this without editing the configs? |
| 19:24.25 | [TK]D-Fender | ross`, You don't configure SIP peers via Asterisk CLI..... |
| 19:25.40 | *** join/#asterisk mathi (Matthew@ip-62-235-253-219.dsl.scarlet.be) |
| 19:25.59 | ross` | [TK]D-Fender: okay cool |
| 19:26.04 | ross` | i'm new to asterisk just thought i'd ask |
| 19:26.23 | ross` | irssi an irc client for instance their channel will not provide support for people editing their files manually |
| 19:26.38 | Qwell | ~book |
| 19:26.38 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:26.40 | ross` | so since this project is new to me i thought i'd ask before i step on any toes with future questions |
| 19:26.43 | mathi | hi |
| 19:26.54 | ross` | mathi: hello |
| 19:27.44 | *** join/#asterisk Sicelo900 (~user@unaffiliated/sicelo) |
| 19:27.49 | ross` | Qwell: i see thank you |
| 19:27.59 | ross` | Qwell: do you also know of any user tutorials that you could suggest? |
| 19:28.43 | jmetro | the book has a pretty good step-by-step for configuring some endpoints and making them work. |
| 19:29.14 | jmetro | like, from installing the server -> installing asterisk -> configure files -> configure endpoints |
| 19:29.15 | ross` | this will be a single server internal pbx system for about 10 users |
| 19:29.15 | mathi | I cannot find a lot of information on the internet about servers running asterisk, probably I am not sure what I am looking for... I developped an IVR that I want to install to some clients, now I need to buy a small server on which I can connect PSTN incoming lines. I have tested on a MyPBX but it comes with asterisk 1.4 only, a custom interface, and a custom Linux distribution |
| 19:29.44 | mathi | the ideal would be to find such a small machine but where I can install my own asterisk version, my own GUI, a normal Linux distribution, ... |
| 19:29.45 | Qwell | mathi: It doesn't matter. Figure out what you need, and buy something. |
| 19:29.50 | mathi | does that exist and what should I look for ? |
| 19:30.01 | ross` | mathi: are you a pro with asterisk? |
| 19:30.40 | ross` | mathi: i have server space if you can also teach me |
| 19:30.41 | mathi | no, I developped a quite spohisticated IVR, but new in telephony, especially all the hardware stuff... what should I look for ? |
| 19:31.03 | ross` | mathi: asterisk doesn't need that great hardware as far as i know |
| 19:31.07 | Qwell | mathi: it doesn't really matter |
| 19:31.10 | ross` | unless you have hundreds of people |
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| 19:31.43 | ross` | Qwell: what is the suggested hardware for 100 people on lossless call quality |
| 19:31.56 | mathi | well I don't want to put a classic desktop computer at the client's places ... look how nice this is for examle : http://www.yeastar.com/images/MyPBX_SOHO.png |
| 19:32.12 | mathi | Qwell, so what do you suggest me to do ? |
| 19:32.23 | Qwell | Buy something that fits your needs. |
| 19:32.33 | ross` | mathi: can't you fash the myPBX |
| 19:32.41 | mathi | Qwell, that's why I post a question ehre, I have no idea what would fit my needs |
| 19:32.45 | mathi | ross`, no idea ... |
| 19:32.50 | ross` | flash* |
| 19:32.53 | Qwell | mathi: How could we possibly know? |
| 19:33.08 | ross` | mathi: how many users will be on your system |
| 19:33.09 | ross` | ? |
| 19:34.00 | mathi | Qwell, because I described all what I want, there are servers like the MyPBX SOHO which are very cheap and compact, unfortunately it has Asterisk 1.4, a custom Linux, a custom GUI, ... are there any alternatives |
| 19:34.05 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 19:34.08 | mathi | ross`, from 1 to 10 maximum |
| 19:34.12 | Qwell | You haven't described a thing. |
| 19:34.22 | Qwell | 10 max? Go buy a Pentium 2. |
| 19:34.30 | Qwell | It doesn't matter. |
| 19:35.06 | ross` | make sure it has enough memory and bandwidth for the amount of users you plan on servicing and don't worry about the cpu is what Qwell is trying to say |
| 19:35.09 | ross` | :P |
| 19:35.11 | mathi | Qwell, for example I had no idea what a Pentium 2 was until you mentionned |
| 19:35.23 | Sicelo900 | huh? |
| 19:35.24 | ross` | mathi: google vps providor |
| 19:35.27 | Qwell | ~hafc |
| 19:35.27 | infobot | hmm... hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 19:35.46 | ross` | Qwell: ha, i like that one |
| 19:36.54 | mathi | pfffff |
| 19:37.08 | jmetro | build a pc in a custom box. stick whatever you want on it. |
| 19:37.24 | jmetro | custom everything means everything you want. |
| 19:37.56 | ross` | jmetro: it could be everything you don't want that no one else has because it's unstandardized crap :p |
| 19:38.03 | ross` | custom != what you want |
| 19:38.12 | ross` | hopefully it is tho :D |
| 19:38.12 | *** join/#asterisk areski (~areski@80.174.255.87.dyn.user.ono.com) |
| 19:38.37 | ross` | that's like saying my water bottle is custom |
| 19:38.37 | jmetro | i build custom all the time and its exactly what I want. learn your hardware and what your hardware can do on your own and you will never be questioning what is going on. |
| 19:38.42 | ross` | it has seven holes in it |
| 19:39.26 | jmetro | end users would put 7 holes in a water bottle, people who build custom water bottles know how many ounces it holds and how much water it outputs per sip |
| 19:40.35 | ross` | jmetro: an ounce is a denomination of weight |
| 19:40.44 | ross` | jmetro: fluid ounce is a denomination of volume |
| 19:40.48 | ross` | i think you want the latter |
| 19:40.59 | Qwell | ross`: force, not weight |
| 19:41.09 | jmetro | sorry my analogy didnt include technical specifications on oz vs fl-oz |
| 19:41.13 | Qwell | (okay, fine, it's both) |
| 19:41.32 | ross` | Qwell: ounce force is different |
| 19:42.14 | ross` | Qwell: the technicality you were looking was probably mass |
| 19:42.15 | ross` | :p |
| 19:42.40 | mathi | jmetro, so what do you have as hardware |
| 19:43.32 | ross` | mathi: google vps provider and get the cheapest one with atleast 50gb monthly bandwith and 1gb ram |
| 19:43.43 | mathi | ok i'm checking |
| 19:43.49 | ross` | this is not the place to ask unrelated pbx hardware questions |
| 19:44.05 | ross` | unless you are looking for a specific piece of hardware |
| 19:44.10 | ross` | then try to rephrase your question |
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| 19:47.00 | mathi | ross`, when I google vps provider I get as first results companies that host servers, I didn't mentionned I need a hosting? |
| 19:47.26 | ross` | mathi: then what do yo uneed |
| 19:47.43 | ross` | you need to rent or purchase a server to run asterisk on right |
| 19:48.58 | mathi | ross`, as you said, maybe this isn't the right channel, sorry for that... I just need a computer with cards to run my IVR (at the client's place, not hosted somewhere). I like the idea of having an embedded system like the MyPBX machines, as it is compact, and comes with cards. |
| 19:49.09 | Qwell | what cards? |
| 19:49.17 | mathi | unfortunately you are stuck with their software |
| 19:49.34 | mathi | cards for incoming PSTN |
| 19:49.39 | Qwell | mathi: http://store.digium.com/products.php?category_id=20 is your friend. |
| 19:49.47 | mathi | thank you I am checking |
| 19:50.03 | jmetro | oh yeah, dont forget digium products |
| 19:50.14 | mathi | that are the cards, and where are the small computers ? |
| 19:50.17 | mathi | :P |
| 19:50.18 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 19:50.28 | Qwell | mathi: http://google.com/ |
| 19:50.38 | *** join/#asterisk lroe (~roe@unaffiliated/roe) |
| 19:50.43 | lroe | Does linksys still make SIP phones? |
| 19:51.14 | jmetro | Qwell: wouldnt switchvox work? |
| 19:52.11 | Qwell | jmetro: He wants to run his own distro. He's wrong, since he doesn't know what he's doing. Yes, switchvox is ultimately what he wants though. |
| 19:52.42 | *** join/#asterisk Sicelo900 (~user@unaffiliated/sicelo) |
| 19:54.20 | mathi | I know what I am doing, I have put now a big computer at my client's place and my IVR is working perfectly fine, I want to replace with something that looks more professional |
| 19:54.42 | Qwell | looks more professional? really? |
| 19:55.02 | Robotman321 | its a phone system.. does it work? good, there you go.. amirit? |
| 19:59.00 | jmetro | mathi - take a look at http://www.starface.de/en/ |
| 19:59.45 | p3nguin | I think these embedded systems I use look pretty good. They'd look better if I painted them black and put the orange Asterisk logo on them, but I was advised to not do that. |
| 19:59.52 | *** join/#asterisk infinity_ (~brendon@web2.artsopolis.com) |
| 20:00.06 | infinity_ | anyone know how to tell dahdi to access the channels in reverse order? |
| 20:00.11 | infinity_ | instead of 1-23, do it 23-1 ? |
| 20:00.21 | p3nguin | Use a group. Use G instead of g. |
| 20:01.01 | p3nguin | I mean in the Dial(), of course. |
| 20:01.36 | infinity_ | ahh |
| 20:01.55 | infinity_ | i'm using freepbx so let me see. that helps a lot. at least i have a hint :) |
| 20:02.12 | p3nguin | I can't help you do it in FreePBX. |
| 20:02.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:02.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:02.50 | ghost75 | can spot any error: http://pastebin.com/bZxfDaV6 |
| 20:02.55 | infinity_ | <PROTECTED> |
| 20:03.02 | infinity_ | is that correct ? |
| 20:03.19 | infinity_ | instead of g0 ? |
| 20:03.35 | infinity_ | okay. looking good. |
| 20:10.06 | p3nguin | That was exactly what I meant, yes. |
| 20:13.02 | ghost75 | hmm why i get permission denied when i try to originate over ami |
| 20:13.24 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 20:13.52 | kaldemar | ghost75: your manager user does not have originate privileges. |
| 20:15.16 | ghost75 | aha :) |
| 20:17.51 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:18.39 | ghost75 | great, now it just tells me "failed" |
| 20:23.20 | *** join/#asterisk vimreaper (~vimreaper@rrcs-70-62-43-252.central.biz.rr.com) |
| 20:38.31 | ghost75 | wtf, when i originate a new call it first rings on the called number |
| 20:41.05 | p3nguin | "Channel" is called first, then application or extension will be run. |
| 20:42.15 | p3nguin | If that isn't what you want, you'll have to use a local channel to call the extension first, and then application Dial to call the channel second. |
| 20:42.58 | vimreaper | hey guys, if someone with gdb knowledge could look at this coredump for me id sure appreciate it http://pastie.org/private/ickgzuz6mvyfjbs8peirq Asterisk is crashing on us 3 times a day :/ |
| 20:43.41 | vimreaper | to give a little background, theres 300 phones connected to it all cisco phones. We have Chan-SCCP-b installed too |
| 20:46.37 | ghost75 | on cli it was working like this (ext was first ringing): originate SIP/$Fromextension extension $PhoneNumber@$Context |
| 20:48.49 | p3nguin | This is how it works: channel originate Tech/data extension exten@context |
| 20:48.57 | p3nguin | Tech/data is first called. |
| 20:49.08 | p3nguin | If it answers, extension exten@context will then be executed. |
| 20:49.18 | ghost75 | my above example is working but only in cli |
| 20:49.22 | p3nguin | If Tech/data does not answer, extension does not execute. |
| 20:50.05 | p3nguin | In your example, SIP/$Fromextension will first be called. If it answers, then extension $PhoneNumber@$Context will be executed. |
| 20:50.21 | p3nguin | SIP/$Fromextension being the "channel" |
| 20:50.32 | p3nguin | "Channel" is called FIRST. |
| 20:50.42 | ghost75 | so i am it goes different |
| 20:51.03 | ghost75 | in ami |
| 20:52.20 | ghost75 | this is then Dial/$EXTEN to make local channel? |
| 20:52.42 | ghost75 | crap |
| 20:52.55 | ghost75 | Local/$EXTEN |
| 20:53.42 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 20:54.34 | p3nguin | Local/exten@context |
| 20:55.17 | p3nguin | That turns a location in your dial plan into a channel that can be used in the same manner as SIP/whatever. |
| 20:55.40 | ghost75 | and after that the dial cmd to the number |
| 20:56.00 | ghost75 | wondering if this will work over ami |
| 20:56.13 | p3nguin | Replace "extension" with "application" and use Dial SIP/whatever |
| 20:56.39 | p3nguin | channel originate Local/1234@phones application Dial SIP/itsp/13145551212 |
| 20:57.07 | p3nguin | 1234@phones will first be called. If it answers, then 13145551212 will be called through a SIP peer named itsp. |
| 20:57.38 | p3nguin | It appears to work the same way in AMI. The command has the exact same parameters. |
| 20:59.15 | ghost75 | but i think ami will not wait |
| 20:59.28 | ghost75 | it will do dial immediately after originate |
| 20:59.47 | ghost75 | because there is no dialplan |
| 20:59.49 | p3nguin | I'm going to have to test it to ensure I know what I am talking about. |
| 21:00.15 | p3nguin | For what you said, there has to be dial plan. The local channel requires a bit of dial plan. |
| 21:00.42 | p3nguin | If you are using extension whatever@something, that requires dial plan. |
| 21:01.22 | ghost75 | in ami there is no @ i think |
| 21:01.35 | p3nguin | No, there is extension, context, and priority. |
| 21:01.42 | ghost75 | yes |
| 21:01.51 | p3nguin | If you use extension rather than application, you are executing dial plan. |
| 21:02.09 | p3nguin | If you use extension rather than application, you are required to supply extension, context, and priority. |
| 21:02.31 | ghost75 | never used "application" |
| 21:04.39 | p3nguin | You'll either use Exten, Context, and Priority WHICH UTILIZES DIAL PLAN, or you will use Application, the app name, and data to pass to the app. |
| 21:06.14 | p3nguin | So your saying that there is no dial plan precludes you from using anything BUT application. |
| 21:06.18 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 21:10.40 | ghost75 | then i try with dialplan |
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| 21:13.12 | ghost75 | gnarrr i cant because i dont have number in dialplan |
| 21:13.39 | ghost75 | seems to be difficult to do this webbased |
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| 21:33.44 | jacekowski | anyone on BT UK ISDN2E lines? |
| 21:35.12 | mathi | Qwell, why did you say earlier that I was wrong wanting my own linux distribution on the system? |
| 21:35.25 | Qwell | Because you don't know how. |
| 21:36.20 | mathi | heh? |
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| 21:48.33 | wltjr_ | p3nguin: can get a few minutes of your time in pm to go over that snipped you provided, looks like I might have to implement something like that to get the desired behaviour, having a hard time understanding the vars in your snippet |
| 21:49.13 | wltjr_ | p3nguin: this snippet -> http://pastebin.com/8dzGfuXJ |
| 21:51.47 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 21:52.12 | ThinkGNU- | does anyone know of any devices that can be used to bridge an analog connection over a network link? |
| 21:52.33 | ThinkGNU- | I know MultiVoIP makes them but I can't find any others(or can't verify that others do) |
| 21:54.24 | Qwell | ThinkGNU-: Is that somehow different from having ATAs at both ends? |
| 21:54.36 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 21:54.51 | ThinkGNU- | Can I have an ATA send calls to another ATA without having Asterisk running somewhere on that network? |
| 21:55.08 | ThinkGNU- | This would be a standalone setup |
| 21:55.11 | nny | working on hashing out a n HA setup using 2 servers. Any gotchas I should be aware of? |
| 21:59.13 | *** part/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
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| 22:04.42 | ThinkGNU- | Qwell, can a set of ATAs work independently like that? |
| 22:08.21 | tonikasch | bye |
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| 22:22.46 | *** part/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 22:27.45 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 22:28.08 | SuperNull | will a general reload on 1.4.x cause cdr_mysql to take new settings ? (userfield=1 from =0) |
| 22:36.31 | jpsharp | It should. |
| 22:38.07 | *** join/#asterisk Dibbler (~Dibbler@host86-150-172-31.range86-150.btcentralplus.com) |
| 22:38.24 | SuperNull | hurm. i think i recall of some other requirement to get userfield working in mysql .. then again im comparing 1.6 to 1.4 so .. hm. |
| 22:38.25 | SeRi | jpsharp: comcast business fix in 2 days what has been broken for 3yrs in residential. |
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| 22:41.24 | Tagor | When I convert my audio file with 'sox org.wav -c 1 -r 8000 output.gsm' I get a lot of noise. Is there a better way to convert it? |
| 22:41.27 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 22:42.13 | SuperNull | oh nice, looks like it is working how ever it doesnt do unique id.. no biggie, more concerned with proper account #s ;) |
| 22:48.31 | Tagor | Never mind, converted to sln and it sounds better :) |
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| 23:02.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:15.03 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 23:16.21 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:18.02 | p3nguin | wltjr_: I'm back if you're still around. |
| 23:19.36 | wltjr_ | p3nguin: yes still here, though dealing with qos issues as in needing it badly :) |
| 23:24.29 | p3nguin | The paste expired, so I'm not sure what you wanted me to see. |
| 23:25.41 | *** part/#asterisk ThinkGNU- (~ThinkGNU@216.67.183.210) |
| 23:31.03 | SuperNull | uhg. i hate asking stuff like this but its hard to find on that google machine.. when did 'regserver' become a standard sql field for the user table? |
| 23:50.59 | SuperNull | soooo where might one find legacy sources of ast? the site doesnt point out anything to old. |
| 23:52.58 | p3nguin | What will you use the old versions for? |
| 23:56.08 | SuperNull | need to see if an ancient version has regserver field support on ast realtime. |
| 23:56.23 | SuperNull | yes i know.. upgrade.. blah blah.. :) |
| 23:57.52 | p3nguin | http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 23:58.43 | SuperNull | thats where im looking yet.. i was fairly sure it was an asterisk addon yet i dont see anything relating to it in the addon tgz. |
