IRC log for #asterisk on 20120122

00:11.18*** join/#asterisk infobot (~infobot@rikers.org)
00:11.18*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.1 (2012/01/19), 1.8.8.2 (2012/01/19), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
00:12.38*** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
00:19.43*** join/#asterisk keepar (ayane@quality.webhosting.at.sunsaturn.com)
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01:01.09*** mode/#asterisk [+o leifmadsen] by ChanServ
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01:09.55*** join/#asterisk Ice_Strike (~Ice_Black@94-192-112-241.zone6.bethere.co.uk)
01:11.12Ice_StrikeHello
01:12.20Ice_StrikeI have managed to get asterisk working and it seem doing fine so far.
01:12.43Ice_StrikeNow I want to develop a call center website to allow agents to use the website
01:12.51Ice_StrikeWhich PHP libabry do you recommend?
01:36.36[TK]D-Fender~toywy
01:36.37infobotsomebody said toywy was The one you write yourself.
01:56.34carrarI have that one
01:56.43carrarIt's crap
01:57.10carrarAnd the author won't fix it
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02:50.15ChannelZhehe
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03:47.04WIMPyHa, I'm no longer alone. Someone googled for "dahdi pri G722".
03:51.14[TK]D-Fendermisery loves company ;)
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04:02.07WIMPyLook at the number of users in here.
04:05.38[TK]D-Fenderrawr
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04:26.00Dovidis a 180 with SDP "correct SIP"?
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04:38.21bbourdageCan you change the keys for delete message, and other items in voice mail, or are they hard coded ?
04:38.59Dovidbbourdage: Can you explain what you mean by "keys"
04:39.03[TK]D-Fenderbbourdage: vi app_voicemail.c
04:39.25[TK]D-FenderHard-coded.  Fortunately you have the code
04:39.35bbourdage7 for delete, would like to mimic our current system which is 3
04:39.43bbourdageThanks Fender, that helps a ton
04:39.51Dovidwow i am tired. didnt see the entire msg
05:43.18*** join/#asterisk Linux4Eric (~chatzilla@cpe-71-72-172-65.woh.res.rr.com)
05:46.38Linux4Ericis dahdi still required now that the linux kernel timer can be used?
05:46.55Linux4Ericon 1.8
06:01.24ChannelZyou still need it for audio mixing if you use MeetMe
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06:47.27Hadi_Hello.. anyone have experience with Asterisk and Polycom "Watch Buddy"?
06:48.14Hadi_We have a bunch of phones behind NAT (the asterisk machine is on a public IP outside) and having problems getting Watch Buddy to work on the Polycom IP 650 expansion module.
06:50.52Linux4EricHadi_, has this ever worked yet for you?
06:51.49Hadi_yes it works fine
06:52.01Linux4EricWhat I mean is, does the IP 650 make and receive calls?
06:52.03Hadi_but when there is lots of call volume on the ext being monitored
06:52.19*** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj)
06:52.31Hadi_the lights stop working... they either stay lit.. (as if the users on the phone) or they do not indicate that the ext is on the phone
06:52.48Hadi_when we reset the phone.. (reception phone) everything works... then after a while it stops again
06:52.56Linux4EricI am not familiar with watch buddy but I assume it is some sort of Busy Lamp that can show status of extensions?
06:53.08Hadi_yes exactly
06:54.53Linux4EricI have had much difficulty with NAT and polycom phones, but it sounds like NAT is working properly since your phones all work correctly
06:55.31Linux4Ericso watch buddy uses asterisk BLF "Hints", is that correct?
06:55.58Hadi_yes
06:56.59Linux4Ericyou may be running out of bandwidth with too many calls and that packet that changes the BLF on the phone might get dropped
06:57.23Hadi_there is a dedicated DSL line for 10 ext's
06:57.29Hadi_and we are using g729
06:58.26Hadi_It seems that when its not workingÂ… the phone send a REGISTER request but does not send the SUBSCRIBE as it should be.  The INVITE is coming to the phone before it manage to send the SUBSCRIBE.. In this case the PUBLISH and the NOTFY are not sent.
06:58.55Hadi_when we restart the phone... it starts to work again for a little while
06:59.22Linux4Erichmm. Once in a great while my BLF flip out for about 10 seconds but always return to normal.
07:00.09Linux4EricBy the way, I do the BLF on cisco/linksys phones and I have to change a setting that states its an Asterisk  server.  Anything like that on the IP650?
07:00.31Hadi_I don't think so...
07:00.42Hadi_The issue only with Polycom we have no issue with other phones
07:02.57Linux4Erichttp://www.asteriskguru.com/archives/image-vp258046.html
07:03.13Linux4EricThis states a firmware upgrade solved someone's problem
07:03.22Linux4Erica few posts down
07:04.45Hadi_let me check
07:04.56Hadi_weare running latest firmware
07:05.19*** part/#asterisk bbourdage (~bbourdage@174.33.93.154)
07:08.41Linux4EricI don't know then.  Definitely seems like it's the phone since it tries to send subscribe before registration
07:10.29Linux4Ericor doesn't send subscribe, have you confirmed that asterisk is sending device state change information about the particular extensions from the console?  WHat version of Asterisk are you running out of curiosity
07:11.50Hadi_1.6.1.20
07:18.41Linux4EricI am out of suggestions, seems like you've covered a lot of bases in your troubleshooting.  It's probably not an Asterisk issue, but either network (nat or otherwise) or Phone firmware bug
07:19.48Hadi_yes... it is not asterisk
07:20.01Hadi_was wondering if someone had same issue... perhaps some changes to the phone config will resolve it
07:21.46Linux4Ericyes i hear ya, but the fact that it works fine until lots of traffic makes me think that the settings are right.  Do you use QoS or any bandwith shaping on your DSL router?
07:22.39Hadi_no since the DSL is dedicated to the phones
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07:23.06Hadi_I'm wondering if setting the reception phone under DMZ settings of the router will resolve the issue
07:25.22Linux4Ericit would remove nat and do more like port forwarding so try it and see if different result.
07:26.07Linux4EricI gotta run, good luck
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11:23.05webczathello
11:23.13webczatHow to get documentation on asterisk10?
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11:42.42webczatshould I read sample configs?
11:57.44wdoekes2webczat: read the CHANGES and UPGRADE files, they'll tell you what's different from 1.8
11:58.04wdoekes2~book
11:58.05infobotAsterisk: 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
11:58.11webczatare config examples and command/app references enough to understand things?
11:58.27wdoekes210 isn't a completely different version
11:58.30webczator wiki for 1.8, maybe?
11:58.37wdoekes2it's just 1.8 with certain enhancements
11:59.59webczatokay. so wiki for 1.8? or maybe can I just read docs in sample configs and base on it?
12:10.19webczathmm what is call parking?
12:15.15*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
12:15.59jzawwebczat: ive migrated more or less the same confs from 1.2 - 1.4 - 1.6 - 10
12:16.09jzawediting new / changed stuff as ive been going on
12:16.35jzawthere have been changes but they always come up in the console / logs as errors so you can edit them
12:18.35webczatI am actually new to asterisk in general
12:19.43webczatand trying to configure it using, but without, sample configs. like handwriting configs, but reading samples for it
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12:30.41jzawnods
12:31.15jzawwebczat: thats how i started
12:31.17jzawactully
12:31.21jzawactually
12:31.32jzawwhat was useful was to take some asterisk distro
12:31.35jzawwith webgui
12:31.44jzawand add / remove  / edit extensions
12:31.51jzawand see how the actual textual confs changed
12:31.59jzawyou get a feel for the syntax that way
12:32.11jzawthen i went for my own roll out
12:32.41jzawalso extensions.ael is much much easier to read and especially easier to write
12:32.58webczatmhm
12:33.06jzawimho of course
12:33.08webczatnow, I don't actually understand what are parking lots/etc
12:33.29webczatand transfers. I mean, can't I do a transfer using my sip client?
12:50.06*** join/#asterisk RyuunoAelia (~any@2a02:120b:c3f6:a641:219:66ff:fedd:acbc)
12:50.18RyuunoAeliahello
12:50.41RyuunoAeliaI have a somewhat long question :
12:51.02RyuunoAeliaI have this fiber router http://www.swisscom.ch/res/internet/dsl/router/index.htm?languageId=en (which is in fact a rebrand of http://www.adbglobal.com/broadband/triple-play/prg-f4202n.html ).
12:51.24RyuunoAeliaThis router is imposed on me, and I am quite dubious about security. It has an Asterisk running as uid 0 facing internet (because the routeur has 2 analog rj-11 connectors for the phone). And when I connect via ssh (OpenSSH-4.2_p1 ...) it runs linux kernel version 2.6.16... and asterisk -V gives me "Asterisk" no version number...
12:51.29RyuunoAeliaIs there a way to know which version/security issues this Asterisk has ?
12:52.07RyuunoAelia(I am more interested in the security issues as this asterisk is internet-facing and the only way they give me to use my phone number...)
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13:28.56webczatwhat's the bridge module and what's the dial module?
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13:31.25AshuttoHello
13:32.00RyuunoAeliawebczat: it gets complicated because it uses some kernel module for an ADC-DAC from broadcom (the whole router is a broadcom SoC)
13:35.15RyuunoAeliawebczat: loaded modules are : res_adsi res_monitor res_features pbx_config chan_jdsp chan_sip app_dial app_openrg_cmd app_transfer app_voicemail
13:35.47RyuunoAeliachan_jdsp is the abstraction for the broadcom ADC-DAC
13:37.41RyuunoAeliaI asked my ISP to get the source code (as it is GPL), but they take their time... delaying with "we have to ask the higher-ups how to do it" and so on
13:39.48fileRyuunoAelia, a quick Google leads here - http://a1-forum.at/kombipaket-aonspeed-privatkunden/neue-prgav4202n-version-von-4-8-3-dwvv-tau-5-1-6-t970.html
13:40.04filelots of good info, maybe a link to source
13:41.40RyuunoAeliahard for me to tell I don't read german :D
13:42.24webczathemm a question: extension contexts are groups for extensions, does it mean I as an user have the assigned extension context or what?
13:46.00RyuunoAeliafile: is there no way to know except for me to find or get the sources ?
13:46.57fileif the core show version won't tell you then the binary doesn't know
13:47.08filewell, if you don't have a 'core show version' you can narrow it down
13:47.23filebecause the command changed from show version to core show version after... 1.4? I think? it was long ago
13:47.46RyuunoAeliaaoutch...
13:48.02RyuunoAelia"No such command 'core' (type 'help' for help)"
13:52.37RyuunoAeliaso I guess the kernel is not the only thing "really old" on my router...
13:53.30RyuunoAeliawell there is openssh too ... version 4.2_p1 is quite old too ...
13:54.47RyuunoAeliathat's something for a router that began *selling* 4 months ago ...
13:56.14filegenerally the software provided for socs contains really old software
13:56.21filethat is 'confirmed' to work happily
13:56.34fileand they don't want to go to the trouble of upgrading because of the cost in time and money
13:59.51RyuunoAeliayes... but I don't have any choice I am forced to use this thing... I can't event buy a fibre PCI-e card and plug it... plus the thing is a "firewall" (nice security)... and well the software provider is based in Israel ... the kind of things that makes me trust my hardware ^^
14:03.09RyuunoAeliaplus I had to buy this router... and they won't let upgrade it myself because everything is closed... you apparently you need to sign the firmware to get it into the SoC (I didn't even try because I don't even have the sources)
14:04.37RyuunoAeliawell complaining here won't change anything ^^
14:06.49RyuunoAeliaand from a security point of view considering that it's an Asterisk 1.4 or something are there remotely exploitable security issues ?
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14:18.35Neptuhej if i do a sequection in  a plsql scrip the whole script is a transaction??
14:19.10Neptuhow can i commit on some points??
14:19.19Neptuor rollback
14:34.11Neptuhow can i do several commits over a loop in a pl/sql??
14:42.19RyuunoAeliaNeptu: maye asking a pl/sql channel will get you an answer ^^
14:42.58Neptuthere is a channel only for postgres pl/sql??
14:46.22RyuunoAeliaNeptu: well there probably a postgres channel ^^
14:49.33webczatwhat happens if you set parkexcl_exclusive to yes?
15:09.17*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
15:16.46webczatI can't actually test everything yet and not everything is documented
15:18.19[TK]D-Fenderwebczat: Yes, devices are pointed to a context.  That says what that device can dial
15:19.16[TK]D-Fenderwebczatwhat's the bridge module and what's the dial module? <-  Bridge is a dialplan application that lets you hook into another call in progress.  Like a way of forcing a 3-way call, etc.
15:19.31[TK]D-Fenderwebczat: Many other ways to use that tool for other similar means.
15:19.56[TK]D-Fender[07:33]webczatnow, I don't actually understand what are parking lots/etc
15:19.58[TK]D-Fender[07:33]webczatand transfers. I mean, can't I do a transfer using my sip client?
15:20.30[TK]D-FenderParking lots are so you can trasfer a call you're on to a "holding area" so that you can pik it up froma completely different device by calling up the lot #
15:20.48[TK]D-FenderAnd not sure what you meant more precisely about your transfer question.
15:26.15webczat[TK]D-Fender: I understand parking lots. my question was about why there are multiple of them? and what's parkext_exclusive then? an intelligent parking or what?
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15:27.52[TK]D-FenderNot sure on the finer points, but picture using your system to host multiple companies.  You don't want company A picking up company B's parked calls.
15:27.58[TK]D-FenderAs for that parm... lets take a looks..
15:29.28[TK]D-FenderHrm, wording is a little vague...
15:30.01webczatI understand it like I can have two parking lots, one set on 701 to 799, one from 800 to 899, and both using number 1 for call parking
15:30.51[TK]D-Fenderdoesn't sound right...
15:31.34[TK]D-Fender"* Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access."
15:31.43[TK]D-Fenderthis is a little off as well..
15:32.20webczatmaybe I'm right in my understanding.
15:32.31webczathmm, the callee parks, not the caller, right?
15:33.02[TK]D-FenderThe caller parks the other call
15:33.12webczatahh
15:33.25[TK]D-FenderWe are talking.  I transfer you to the parking ext.  it reads the lot #.  I hang up and then you're handed off
15:33.53webczatand I can unpark then
15:34.02webczatwhat is the pickup dtmf?
15:34.19[TK]D-FenderYou dial an exten that calls ParkedCall()
15:34.43[TK]D-Fenderone is generated in the parking context based on that other setting to specify a pickup exten
15:34.50[TK]D-Fenderbut you can also make your own.
15:35.05webczatI meant the other parameter set to *8 by default
15:35.40[TK]D-FenderI'm not sure the point of features.conf for that method..
15:35.46[TK]D-FenderFeels kinda pointless
15:35.59[TK]D-Fenderwait.. that is CALL pickup, not parking
15:36.13webczatmhm
15:36.14[TK]D-Fendercall pickup is to hijack a call that is actually ringing somewhere else
15:36.25[TK]D-FenderI see your phone ringing and want to answer from mine
15:36.26webczatlike erm what?
15:36.43webczatah so then what do I do?
15:37.37[TK]D-FenderFor what?
15:37.59webczatHow do I pick the call up?
15:38.28[TK]D-Fender*8 is the default.  This also depends on your setting pickupgroup for your devices
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16:05.26Ashuttohow can i have a call stack trace for my dialplan in asterisk console?
16:08.35[TK]D-FenderAshutto: Can you clarify that request...
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16:20.12webczathmm
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16:24.36webczatthe subscribecontext parameter means that I can subscribe to receive notifications only for those extensions that are in the context?
16:25.13Ashutto[TK]D-Fender, in my console, if i call my asterisk, i'd like to see the dialplan execution plan (like "Dial(xxxx)", "Hangup()") in the calling order
16:32.31[TK]D-FenderAshutto: Then go connect to the * CLI and "core set verbose 10"
16:41.06Ashuttothanks ^_^
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16:49.49picard276hey guys im having an issue with a2billing
16:49.58picard276im not sure how to get it to pass to a2billing?
16:50.10picard276i have my incoming DID and i have that set to a custon Extension: a2billing,s,1
16:50.21picard276but it looks like this http://pastie.org/3231635
16:50.24picard276when a call comes in?
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16:53.03Ashutto[TK]D-Fender, it only says "Using SIP RTP CoS mark 5"
16:55.22picard276what?
16:57.34[TK]D-FenderAshutto: If you set the verbose level and you don't see anything then that is because it isn't even executing dialplan because a call (if any) is refused
16:57.49[TK]D-FenderAshutto: At which point you should enable SIP DEBUG and look at the actual attempt
16:57.55[TK]D-FenderAshutto: "sip set debug on"
17:00.08picard276anyone have any ideas
17:04.14[TK]D-FenderI'd start by proving the target exists
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17:28.03*** join/#asterisk phunteltek (~androirc@cpe-184-59-139-30.neo.res.rr.com)
17:29.08phunteltekHey guys it's good to be here
17:31.50phunteltekCan anyone recommend an app for android?
17:33.47phunteltekCan you would recommend a voice over ip application for android o s tablet
17:35.49phunteltek<PROTECTED>
17:36.09ikevinsipdroid
17:37.06phunteltek<PROTECTED>
17:37.37phunteltekAre there any issues with firewalls on the android tablet?
17:37.50phunteltek<PROTECTED>
17:38.02ikevinusing 3g or wifi?
17:38.09phunteltek<PROTECTED>
17:38.27ikevinvia wifi no firewall/nat problem on android
17:38.49ikevin(else if you have a firewall on your wifi equipment)
17:39.34ikevinby default android accept all incomings/outgoings connexion
17:39.47phunteltek<PROTECTED>
17:40.08phunteltek<PROTECTED>
17:40.25phunteltek<PROTECTED>
17:40.26ikevini you use public wifi you maybe need a thing like vpn to receive calls
17:40.34phunteltek<PROTECTED>
17:40.56ikevinor a stun server
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17:41.33phunteltekDo you typically need to root your android device?
17:41.43ikevinno
17:42.20ikevinstun server can be setup on sipdroid, and vpn can be setup via android
17:42.39ikevin(see settings > network > vpns)
17:42.51phunteltek<PROTECTED>
17:43.01ikevin=
17:43.16ikevinstun server can be more easy to find and configure
17:43.58phunteltekI would like to use vpn to get through my work network. They have an open wifi network there but I don't want them watching the traffic
17:44.27ikevinok, maybe openvpn can be a good & simple solution
17:44.37ikevinso, vpn take a lot of battery
17:45.00[TK]D-FenderSo far GV & Skype on your phone have nothing to do with your SIP agent on your phone vs your server.
17:45.01phunteltek<PROTECTED>
17:45.39[TK]D-Fenderphunteltek: Right now the first thing is : Why should we trust your server environment is even sane, let alone all proper?
17:45.43p3nguinUse an encryption method.
17:46.15[TK]D-Fenderphunteltek: Start further back and reall start looking at what is coming in and what you setup to allow it in.
17:46.27p3nguinIf VoIP over WiFi is bad, encrypted VoIP over WiFi must be horrid.
17:46.59[TK]D-Fenderp3nguin: Encryped should not be any worse.  Encapsulated perhaps....
17:47.02phunteltekI haven't had any problems with voice over wifi
17:49.04phunteltek<PROTECTED>
17:49.29phunteltek<PROTECTED>
17:49.55phunteltek<PROTECTED>
17:52.15phunteltekI have an android tablet with no phone or 3g
17:53.11phunteltek<PROTECTED>
17:54.37phunteltekMy asterisk box is behind a firewall though
17:55.02p3nguinAs it should be.
17:56.35phunteltekI use to get phone calls from a fake extension from ip in bulgaria
17:56.47phunteltek<PROTECTED>
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18:47.34grkbloodmy inbound calls are pretty choppy, the caller can hear me fine but im having trouble hearing them clearly, heres an example: http://files.joeshowradio.com/samples/inbound_sample.mp3
18:47.52grkblooddoes anyone know what might be causing this?
19:03.03p3nguinpoor downstream bandwidth, excessive downloads from PCs, no QoS on the network
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19:25.42grkbloodp3nguin, my speedtest results are 21.53Mpbs down/4.14 Mbps up/49 ms and im not downloading anything
19:31.17ketaspacket loss?
19:34.33grkbloodnope, it looks pretty solid
19:35.15ketaswho's provider?
19:35.57p3nguinWho is a provider?
19:36.45p3nguinOh, maybe you are asking who is the provider?
19:36.57ketaswtf?
19:37.20grkbloodvitelity
19:37.36grkbloodits prolly them, i need an east coast provider
19:37.53Vickstersanyone here has some experience with DAHDI?
19:37.54ketasis this some kind of asymmetric connection?
19:38.38grkbloodcallcentric is the only reputable provider i know of on the east coast, anyone in the southeast worth checking out?
19:38.46p3nguin"my speedtest results are 21.53Mpbs down/4.14 Mbps up"
19:39.02p3nguinSeems pretty asymmetric to me.
19:39.19p3nguinCall Centric isn't really reputable.
19:40.05grkbloodwhos worth checking out on the east coast then?
19:40.05p3nguinVitelity should be fine on the east as long as you use a PoP in the east instead of one on Denver.
19:40.24grkbloodp3nguin, thats there only PoP right now
19:40.30grkbloodive asked them about that
19:40.58grkbloodthey plan on adding more, but currently there only PoP is out of denver
19:41.03grkbloodtheir*
19:41.48p3nguinCheck out VoIP.ms, then.  They have one in Atlanta and one in Tampa.
19:43.03p3nguinI'm using them nearly exclusively.  I rarely have any issues with the service.
19:44.04grkbloodill check them out, hopefully they port toll free numbers
19:45.41p3nguinToll-Free number $25 Per number 2 to 4 Weeks * 800/855/866/877/888
19:45.45p3nguinYep, they do.
19:49.04grkbloodwhere do you see that they port numbers? im not seeign anythign about that in there faq
19:50.27p3nguinI'm logged in.  I clicked on DID Portability.
19:50.42p3nguinLook in their wiki.
19:51.41p3nguinhttp://wiki.voip.ms/article/Porting_a_Number#Numbers_portable
19:52.03p3nguinThey show a screenshot of what I just copied/pasted.
20:35.47ChannelZVicksters: For analog, yes
20:37.19VickstersI can't get it working. Everything looks to be fine and detected except I have no dial tone when I look at dahdi_monitor
20:38.34VickstersI've put the relevant configs on pastebin if you'd care to take a peek I could PM the links to you?
20:44.56ChannelZyeah
20:47.50ChannelZSo when you Dial() you're trying to monitor with dahdi_monitor and get nothing?
20:48.02VickstersYes.
20:48.08VickstersNothing at all.
20:48.56ChannelZWhat country?
20:49.00Vicksterscanada
20:49.51ChannelZhmm.. I assume they use kewlstart but I have no idea
20:50.10Vickstersits the same as the US
20:50.43ChannelZWhat happens if you call in rather than out?  Does it detect that OK?
20:51.39Vickstersnah it doesnt either
20:52.18Vickstersits like the card is totally dead but ive tried two of them and its the same result
20:52.22ChannelZhmm
20:53.48ChannelZand "dahdi show status" in the console shows the channel?
20:53.55Vickstersyeah.
20:54.04Vicksterstheres 0 and 1
20:54.31Vicksterser. thats show channels. status just shows that channel.
20:55.57ChannelZHmm.  Not sure what to tell you.  Perhaps the card is not right in the head... you're sure the line it's hooked up to/the cable works? :)
20:56.42Vickstersyeah ive tried the line with an actual phone and it works as advertised. i came here as a last resort. :/
21:04.34ChannelZSorry not sure what else to suggest. I don't have any experience with the x100p but it seems there are a number of clones, some probably more problematic than others
21:06.03Vickstersyeah thats for sure. this thing is a piece of crap. :)
21:06.30Vickstersthanks for your help anyway. im just puzzled as to why it doesnt work. this card used to be fine in another box with basically the same setup.
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21:11.52*** join/#asterisk dandate2 (~dan@222.127.52.58)
21:12.11DovidHi all
21:12.20dandate2i set my pap2-t voipgateway + router to bridge mode and now I cannot connect to its GUI, is this normal?
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22:38.54TehRabbitthey, my router died on me, and I had to replace it, ever since, now my asterisk setup refuses to work X_X any ideas?
22:39.02TehRabbittI can dial out, but there's no audio, so i'm thinking NAT issue
22:39.12TehRabbittbut I have the ports fowared properly so i'm not sure what to check next
22:41.10TehRabbittI have ports 10000-20000 fowarded as well as 5060-5061
22:43.31TehRabbittany ideas?
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22:45.26[TK]D-FenderIdea : show us
22:45.35[TK]D-FenderYour forwarding, the failed calls with SIP debug.  Everything
22:46.51TehRabbittok one sec
22:48.09ChannelZUgh.  I can still smell the stink of the Comcast guy's BO who was in my office yesterday.
22:48.22TehRabbittChannelZ: tht sucks X_X
22:49.25TehRabbitthm this is really weird, now it is randomly working.... i'm convinced NAT is voodoo shit
22:49.26TehRabbittlol
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23:25.09*** join/#asterisk dandate2 (~dan@222.127.170.179)
23:25.40dandate2when I call out from my pap2 there is about  an 8 second delay to connect the call, however when i call out from a softphone on the same network its instant?
23:26.49Vickstersdandate2: whats your dialplan like in the pap2?
23:27.48[TK]D-FenderDelay-ful
23:28.15dandate2(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
23:28.25Vickstersand what number are you trying to call?
23:28.44dandate2either an internal extention or through an outbound route is same delay
23:29.36[TK]D-FenderAnd I doubt either of those parrterns match what you've just shown us
23:30.08dandate2the dial plan?
23:32.41Vickstersyou need to modify your dialplan to match the numbers you try to call. here's a tutorial from a quick google search: http://www.netphonedirectory.com/pap2_dialplan.htm
23:33.59dandate2hmm the factory default dial plan doesnt cut it eh
23:34.04Vickstersnope!
23:34.19Vickstersi mean you need to customize it to your specific situation
23:34.51Vickstersotherwise it just cannot know what your environment is like..
23:40.32eZzhm, is there a way to record a full call flow (including ringing tones) using MixMonitor ?
23:41.01eZzsounds like MixMonitor is starting only when a channel is being bridged
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23:43.35[TK]D-FendereZz: answer the call first
23:43.53eZzI don't need to answer
23:45.12eZzI need a schema like this: 1) MixMonitor(/some/file), 2) Dial(Tech/Num ...), h,1,StopMixMonitor()
23:46.17[TK]D-FenderIf you want the ringing you will have to ANSWER first
23:46.36[TK]D-FenderOtherwise it could still be held OOB and you'll get nothing
23:47.16[TK]D-FenderAnd there is no need to stop MixMonitor in "h".  It should die automatically
23:47.37dandate2what is the dial plan that softphones use?
23:47.41p3nguinnone
23:47.43eZzIf I will answer - then how can I will have a fact than the user was answered ?
23:47.59eZzthan=that
23:50.28[TK]D-FenderThy wording that again...
23:50.31[TK]D-Fendertry*
23:50.48[TK]D-FenderAnd be clean about which call you are referring to.
23:50.48eZzbtw I just tried answer 1-st, then dial
23:50.53[TK]D-Fenderclear*
23:50.54[TK]D-Fendergah
23:50.59eZzthe result wan't changed
23:51.07[TK]D-FenderDial with "r"
23:51.28[TK]D-Fenderthat should gurantee it
23:52.21eZzI don't need a fake ringing
23:54.04eZzdang, even with 'r' the same result
23:56.23p3nguinYou would execute MixMonitor, then Answer, then Dial.  MixMonitor doesn't record when the line is not up.
23:56.49eZzexten => _X.,n,Answer()
23:56.49eZzexten => _X.,n,MixMonitor(/tmp/call-${STRFTIME(${EPOCH},,%Y-%m-%d)}-${ACTIVENUM}-${RAND(100,999)}.wav)
23:56.52eZzexten => _X.,n,Dial(SIP/9529,180)
23:56.55eZzwhat's wrong?
23:57.19eZzhm, I see
23:58.23p3nguinOf course if you Answer, then execute MixMonitor, it would begin recording right away because the line is up.
23:58.47eZzin this case I have channel up and extension (for user's call answer) is launching right after dial

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