| 00:05.16 | igcewieling | goes to the ATM Machine and enters his PIN number. |
| 00:25.04 | vlad_starkov | Question: Why can my Asterisk not properly set callerid, if I call Set(CALLERID(num)=1234567) before Dial()? |
| 00:25.11 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 00:25.46 | WIMPy | Too little information. |
| 00:25.56 | WIMPy | Guess: Your provider doesn't let you. |
| 00:26.12 | vlad_starkov | I have 2 Asterisk boxes installed and connected to each other |
| 00:26.23 | vlad_starkov | Box A and box B |
| 00:26.34 | vlad_starkov | My SIP client connected to box B |
| 00:26.34 | WIMPy | The it's yourself. |
| 00:27.10 | WIMPy | Do you have trustrpid? |
| 00:27.12 | vlad_starkov | I set caller id for my SIP client in dialplan on box B and I want to see that callerid that I've set on box A |
| 00:27.22 | vlad_starkov | WIMPy: probably not |
| 00:28.03 | vlad_starkov | WIMPy: Probably I have to add "sendrpid=yes" in my SIP client peer config on box B? |
| 00:29.13 | j4jackj | Oo er... finally set up a Cisco Catalyst 2950. |
| 00:29.15 | WIMPy | I prefer to use PAI. |
| 00:29.57 | WIMPy | But RPID will do the same. |
| 00:34.17 | vlad_starkov | WIMPy: Yay! |
| 00:34.47 | vlad_starkov | WIMPy: trustrpid=yes at box A AND sendrpid=yes at box B |
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| 04:39.15 | ChannelZ | it's oh so quiet |
| 04:52.08 | j4jackj | Hallo |
| 04:52.21 | j4jackj | Just another random Linux and Haiku user here. |
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| 05:49.00 | phix | j4jackj: Haiku? |
| 05:49.27 | phix | as in the japanese poem? |
| 05:55.20 | ChannelZ | as in the BeOS clone |
| 05:56.57 | j4jackj | ChannelZ: thank you |
| 05:57.04 | j4jackj | phix: see ChannelZ |
| 06:03.39 | phix | Why clone something that is already shit? now you have two pieces of shit |
| 06:03.46 | phix | unless it is a fork |
| 06:03.56 | phix | with the aim to make it not so shit |
| 06:08.19 | ChannelZ | BeOS was not shit |
| 06:13.01 | j4jackj | The aim in Haiku is to make a good base better. |
| 06:13.02 | j4jackj | You can't polish a polished thing, though. |
| 06:13.02 | j4jackj | Well you can, it just makes it worse. |
| 06:14.08 | j4jackj | It causes things to go downhill. |
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| 07:33.02 | TobSnyder | how is it possible to passthrough custom SIP header in invite ? |
| 07:36.59 | kaldemar | TobSnyder: func SIP_HEADER and app SIPAddHeader |
| 07:38.04 | TobSnyder | kaldemar: having two clients, A and B, both registered at asterisk - Client A adds a custom SIP Header when calling Client B (using Asterisk) - but it seems Asterisk is sending a completely new invite without that custom header to client B ? |
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| 07:44.59 | TobSnyder | found a solution in http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/264997 |
| 07:45.41 | TobSnyder | it seems asterisk can not forward all unknown (custom) headers itself, you have to tell asterisk which headers to look for and create them manually ba calling SipAddHeader before calling Dial() in dialplan?!? |
| 07:46.11 | kaldemar | TobSnyder: yes, asterisk is not a proxy. you need to read the header from A and add it before dialing B. |
| 07:55.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:55.37 | TobSnyder | ok thanks |
| 08:08.37 | kaldemar | and asterisk will not forward a single header anywhere, unless you tell it to in dialplan. that's because asterisk is a B2BUA and not a proxy. the second leg of the call is a whole new sip dialog and separate from the first incoming one. |
| 08:13.49 | *** join/#asterisk R1ck (~rick@vandenhof.eu) |
| 08:14.35 | R1ck | hi. I'm using the Queue application and change members with queurules.conf (setting penalty) - is there a way to execute another piece of dialplan (mainly: JabberSend) when penalties are changed? |
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| 08:19.29 | sadtimes | yo |
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| 08:20.09 | defswork | yo ? |
| 08:20.50 | sadtimes | well hi |
| 08:20.52 | sadtimes | or hello |
| 08:21.32 | sadtimes | wanted to ask if anyone has a problem with reloading config / scaling issues |
| 08:21.58 | sadtimes | I'm having an issue at the moment; I reload; and then all extensions start going lagged/unreachable |
| 08:22.00 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 08:22.28 | sadtimes | and I'm thinking that it's because it's trying to get through all that config; while it's still receiving requests |
| 08:22.50 | sadtimes | so once the service becomes available again the amount of traffic it receives causes issues |
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| 08:22.54 | sadtimes | restart the service and it's fine |
| 08:30.56 | magespawn | good morning |
| 08:31.59 | magespawn | ChannelZ: thanks for the link |
| 08:45.52 | sadtimes | I mean does everyone just do seperate reloads generally |
| 08:46.08 | sadtimes | sip reload / dialplan reload |
| 08:46.09 | sadtimes | and so on |
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| 08:59.19 | kaldemar | sadtimes: people usually reload what they need to reaload. if you change dialplan, don't issue a reload on all modules. |
| 09:05.13 | sadtimes | yeah fair enough |
| 09:06.02 | sadtimes | that's about what I thought but just wanted to confirm |
| 09:06.06 | sadtimes | thanks kaldemar :) |
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| 09:57.11 | SeanN | hi guys |
| 09:57.25 | SeanN | I'm hoping to get some help with CDR logging on Asterisk which isn't working |
| 09:57.46 | SeanN | it's enabled, was working for a while but has stopped |
| 09:58.03 | SeanN | in the CLI is seems to actually be disabled |
| 09:58.04 | SeanN | LON-MG-PBX03*CLI> cdr show status |
| 09:58.04 | SeanN | Call Detail Record (CDR) settings |
| 09:58.04 | SeanN | ---------------------------------- |
| 09:58.04 | SeanN | <PROTECTED> |
| 09:58.05 | SeanN | <PROTECTED> |
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| 09:58.18 | SeanN | can anyone offer any advice on this please? |
| 09:58.30 | SeanN | <PROTECTED> |
| 10:05.03 | SeanN | anyone? |
| 10:05.33 | R1ck | so.. enable it? |
| 10:07.24 | SeanN | it seems it is enabled, cdr.conf: enable = yes |
| 10:08.09 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:08.10 | SeanN | and this module is loaded: cdr_mysql.so MySQL CDR Backend 0 |
| 10:08.13 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:08.27 | SeanN | unless it needs to be enabled somewhere else too? |
| 10:10.15 | R1ck | if it was enabled, it should say Logging: enabled |
| 10:10.34 | R1ck | have you reloaded the config? |
| 10:11.41 | SeanN | yea, I've restarted the entire server too for good measure |
| 10:12.10 | SeanN | is there somewhere you need to tell Asterisk to observe cdr.conf? |
| 10:12.49 | R1ck | don't think so |
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| 10:31.20 | SeanN | R1ck, when Asterisk starts, it sees the cdr.conf file, then immediately says CDR logging is disabled: |
| 10:31.20 | SeanN | [Sep 12 11:30:18] VERBOSE[605] config.c: == Parsing '/etc/asterisk/cdr.conf': Found |
| 10:31.20 | SeanN | [Sep 12 11:30:18] NOTICE[605] cdr.c: CDR logging disabled, data will be lost. |
| 10:31.35 | SeanN | I do note there is no include statement for cdr_mysql.conf |
| 10:31.39 | SeanN | do you know if that is reuired? |
| 10:31.44 | SeanN | required* |
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| 10:43.23 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 10:47.34 | vbrinza | hi all. asterisk behind nat. 2 twinkle behind an other nat, both in same network. calling each other. 5060 udp port on asterisk for sip and 10000-10020 for rtp. when we call 4 udp ports from rtp range are initiated however if i use tcpdump to analyze the traffic, i don't see any package coming on them. what could be the issue? |
| 10:47.44 | vbrinza | ps. of course we don't hear each other .. |
| 10:53.47 | eZz | vbrinza: take a look to rtp.conf |
| 10:54.04 | eZz | rtpend=20000 |
| 10:54.07 | eZz | not 10020 |
| 10:54.49 | vbrinza | doesn't matter |
| 10:54.55 | vbrinza | i can define any port there |
| 10:55.02 | SeanN | ok, so I solved my issue, there was a csv section in cdr.conf, commented that out, all working now, seems if you specify this in cdr.conf, only the first section is observed, if you want multiple cdr destinations, they need to be sent in their respective files |
| 10:55.10 | vbrinza | important is not to interfere them with already used ones |
| 10:55.12 | eZz | true, but for debug I suggest to open it ツ |
| 10:55.40 | vbrinza | [general] |
| 10:55.40 | vbrinza | rtpstart=10000 |
| 10:55.40 | vbrinza | rtpend=10020 |
| 10:55.41 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:55.41 | vbrinza | ok |
| 10:55.47 | vbrinza | this its its content right now |
| 10:56.34 | eZz | have you debugged rtp inside asteris? |
| 10:57.39 | *** join/#asterisk sm1ly (~sm1ly@79.137.204.207) |
| 10:57.52 | eZz | vbrinza: I have exactly the same schema as you have, and this works to me great |
| 10:57.54 | vbrinza | i see nothing related to rtp |
| 10:58.03 | vbrinza | if i do rtp set debug on |
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| 11:13.41 | *** join/#asterisk orn (~orn@2a01:8280:101:0:1db5:1bca:57cc:8242) |
| 11:13.53 | orn | Is there any way to dial from the CLI after console dial was removed? |
| 11:15.32 | Greenlight | originate |
| 11:16.43 | orn | ahhh |
| 11:16.44 | orn | thank you |
| 11:16.47 | orn | :D |
| 11:17.04 | orn | i tried googling whether it had been renamed but i guess i didn't use the right keywords |
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| 11:18.46 | orn | and then just hangup on the request? |
| 11:18.53 | kaldemar | orn: "console dial" has not been removed. |
| 11:19.19 | orn | Maybe the module isn't loaded? Do you remember what it's called? |
| 11:19.58 | kaldemar | and the full command for cli originate is "channel originate". |
| 11:20.51 | kaldemar | orn: chan_oss.so, chan_alsa.so or chan_console.so provide the console dial function. |
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| 11:22.41 | orn | ah, it would seem that ubuntu has removed those channel modules from the default install o_O |
| 11:22.47 | orn | my fault for using pre-packaged installs |
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| 12:12.51 | bulkorok | hi |
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| 12:34.14 | *** join/#asterisk [SySteM] (~antoine@ip-202.net-89-2-251.rev.numericable.fr) |
| 12:34.49 | [SySteM] | Hello, anyone know a sip provider where i can change my callerid number ? |
| 12:37.29 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
| 12:38.08 | Greenlight | Which country are you in? Do you want to change to *any* number, or a specific number ? |
| 12:38.32 | [SySteM] | any |
| 12:38.38 | [SySteM] | in france |
| 12:38.47 | [SySteM] | but no problem to take international provider |
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| 12:39.39 | Greenlight | Presenting any number is going to be trickier, at least from my experience with providers here in UK. We didn't get that ability till we had a significant call volume and our carrier "trusted" us.. |
| 12:40.20 | [SySteM] | :( |
| 12:40.40 | Greenlight | It's open to abuse too much |
| 12:41.02 | Greenlight | What's the reason you need to be able to present *any* number? |
| 12:41.39 | [SySteM] | got 50 different number |
| 12:41.58 | [SySteM] | fixe and mobil |
| 12:42.07 | [SySteM] | in different operator |
| 12:42.34 | [SySteM] | and each operator want just 1 callerid |
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| 12:43.33 | Greenlight | I'm not sure I fully understand, sorry |
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| 12:58.57 | Katty | ARE YOU OPEN ON THURSDAYS |
| 13:11.38 | [TK]D-Fender | BANANAS |
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| 13:22.51 | phix | [TK]D-Fender: Peenut jelly and a baseball bat? |
| 13:23.19 | Katty | wonders what the baseball bat is for |
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| 13:44.10 | SuperNull | anyone see realtime have invalid passwords.. then you do a 'sip prune realtime all' and it works yet no database changes happened |
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| 13:44.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:50.49 | igcewieling | SuperNull: make sure to set a port in the DB for any peers which don't register |
| 13:51.14 | SuperNull | they all register. |
| 13:51.34 | SuperNull | i just did a mysqldump on my user table to confirm changes are not happening |
| 13:53.07 | igcewieling | SuperNull: sounds like you need to do some mysql debugging. |
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| 13:54.16 | SuperNull | fairly sure its not mysql ;) but. only have this issue on one of the 7 servers running against it.. this box is 1.8.21.0 so. |
| 13:54.23 | SuperNull | just to be certain im checking. |
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| 14:16.08 | trdillon1 | i have choppy music on hold. Is there a quick way to fix it? |
| 14:17.02 | [TK]D-Fender | trdillon1: Not without know what the actual situation is |
| 14:17.20 | trdillon1 | what do you need to know |
| 14:17.23 | [TK]D-Fender | trdillon1: We have no idea what you're using for MoH, or what you're listening to it on, and over |
| 14:17.35 | [TK]D-Fender | trdillon1: Because it could absolutely anything right now... |
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| 14:18.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:18.19 | igcewieling | [TK]D-Fender: you need to learn how to ignore dumb questions. |
| 14:18.46 | trdillon1 | We have a .wav file set up for music on hold. When calling over a cell phone the music is broken up |
| 14:18.54 | trdillon1 | asterisk 11.4 |
| 14:19.03 | trdillon1 | using sip |
| 14:19.05 | [TK]D-Fender | igcewieling: It's not a "dumb" question, just one asked having given no useful information upon which to provide a educated answer. |
| 14:19.27 | igcewieling | I disagree. Questions with no useful information are dumb questions. |
| 14:19.35 | [TK]D-Fender | trdillon1: Cellphone is a device... you need to be very specific about how that phone is actually talking to *. |
| 14:19.51 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:20.13 | [TK]D-Fender | trdillon1: and you should set up an alternate playback for that file and confirm its specs, etc |
| 14:20.50 | igcewieling | trdillon1: cell phones do not transport music very well, it will be doubly bad if you are using something like GSM or G729 to talk to your carrier. |
| 14:21.15 | [TK]D-Fender | igcewieling: We have no idea how it's actually being used yet... just a little more rope, k? |
| 14:21.34 | trdillon1 | we checked the file and it plays ok in a music player |
| 14:21.47 | igcewieling | [TK]D-Fender: won't do any good. |
| 14:21.52 | trdillon1 | it seems to happen over landlines as well but less often |
| 14:22.02 | [TK]D-Fender | trdillon1: doesn't mean much to *. Forget yyour player. * doesn't downmix poorly formatted files well |
| 14:22.20 | [TK]D-Fender | trYou are not giving useful descriptionas on what you are listening to this over. |
| 14:22.35 | [TK]D-Fender | trdillon1: You are not giving useful descriptions on what you are listening to this over. |
| 14:25.11 | trdillon1 | [TK]D-Fender, Asterisk talks to a VOIP router using GSM. The router is connected to a PRI and sends the traffic out using H323. |
| 14:26.26 | [TK]D-Fender | trdillon1: This is not yet clear. |
| 14:26.49 | [TK]D-Fender | trdillon1: Clarify how * talks to this "router. |
| 14:26.56 | *** join/#asterisk Welington (~Welington@mvx-200-196-57-166.mundivox.com) |
| 14:27.19 | [TK]D-Fender | trdillon1: Because I see 3 totally different techs involved. PRI does NOT talk "H.323" |
| 14:27.27 | [TK]D-Fender | trdillon1: Draw me a better chain |
| 14:27.36 | *** join/#asterisk Welington (~Welington@mvx-200-196-57-166.mundivox.com) |
| 14:27.45 | [TK]D-Fender | trdillon1: And be specific about what is within a local LAN segment, and what is OUTSIDE |
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| 15:42.19 | *** join/#asterisk paule32 (~paule32@dslb-188-106-229-158.pools.arcor-ip.net) |
| 15:42.26 | paule32 | hello |
| 15:42.37 | *** join/#asterisk NOT_guru (~chatzilla@24-241-103-142.static.stls.mo.charter.com) |
| 15:42.48 | paule32 | i running asterisk by scratch |
| 15:43.05 | paule32 | make a default context - hello world example |
| 15:43.17 | paule32 | start asterisk -r |
| 15:43.18 | Katty | hi paule32 |
| 15:43.41 | paule32 | and type in "console dial 1001" |
| 15:43.48 | paule32 | hello katty |
| 15:44.09 | paule32 | bit i get error |
| 15:44.13 | paule32 | but |
| 15:44.17 | Katty | and what's the error? |
| 15:44.42 | igcewieling | put the cli output if the failed call on a pastebin |
| 15:44.43 | igcewieling | ~pb |
| 15:44.43 | 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. |
| 15:44.45 | paule32 | no such extension 1001 in context default |
| 15:44.55 | Katty | well then you need to add one |
| 15:44.58 | igcewieling | pastebin the output of "dialplan show" as well. |
| 15:45.14 | Katty | that's like telling igcewieling to make a merange. |
| 15:45.16 | Greenlight | did you reload after editing it ? |
| 15:45.19 | Katty | without giving him the recipe for merange |
| 15:45.19 | igcewieling | and hope someone has time to help. |
| 15:45.35 | Katty | hugs igcewieling |
| 15:45.58 | NOT_guru | hello, quick question, I have a asterisk system at 2 locations, location A has extensions in the 1XXX range Location 2 has extensions in the 2000 range, when you dial into location A and get into their IVR you can dial an extension of 1XXX but since that locat PBX is not aware of the 2XXX range you can not direct dial a 2XXX extension. is DUNDi the way around this? or is there another way... |
| 15:46.00 | NOT_guru | ...around this I am not thinking of? |
| 15:46.27 | igcewieling | NOT_guru: tell the pbx about the other range |
| 15:46.29 | Katty | you can register them as sip devices on the local server |
| 15:46.39 | NOT_guru | NOTE: I have only setup asterisk in a single location setting in the past, so this is the first time I have bumbed into this |
| 15:46.47 | Katty | the remote device then checks in and goes HAI I"M OVER HERE KTHX |
| 15:47.15 | NOT_guru | well if I do that, my call routing will be broken as my outbound route states that 2XXX goes out trunk B |
| 15:47.30 | Katty | luckily you can change your routing |
| 15:47.32 | paule32 | hui lots of output from dialplan show |
| 15:47.40 | igcewieling | apparently it doesn't if the ivr doesn't know about the 2xxx range |
| 15:47.58 | Katty | that's the beauty of routing. |
| 15:48.04 | Katty | it's like a make your own story |
| 15:48.06 | paule32 | what is the interface port by configuring per web browser? |
| 15:48.29 | Katty | paule32: one does not simply configure asterisk per web browser |
| 15:48.33 | igcewieling | paule32: Asterisk does not have a GUI and cannot be configured by a web browser |
| 15:48.47 | NOT_guru | but then all the extensions at location B are constantly checking in to location A's pbx |
| 15:48.59 | igcewieling | NOT_guru: incorrect. |
| 15:49.31 | igcewieling | NOT_guru: pastebin the cli output if the failed IVR call and a successful normal call |
| 15:49.44 | NOT_guru | wait sorry igcwieling, I was refering to Katty's solution |
| 15:49.58 | paule32 | Katty: i have move the sample files in other directory, create an extension.conf |
| 15:50.01 | NOT_guru | uhm OK |
| 15:50.19 | NOT_guru | let me get those together |
| 15:50.23 | igcewieling | paule32: until you create those pastebins requested I cannot help further. |
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| 15:50.57 | igcewieling | NOT_guru: if you can call the 2xxx range from the 1xxx server, but the ivr doesn't see the 2xxx extensions then you have a simple configuration error |
| 15:51.03 | NOT_guru | and igcwieling when you say " tell the pbx about the other range" what do you mean by this? |
| 15:51.10 | [TK]D-Fender | [11:45]NOT_guruhello, quick question, I have a asterisk system at 2 locations, location A has extensions in the 1XXX range Location 2 has extensions in the 2000 range, when you dial into location A and get into their IVR you can dial an extension of 1XXX but since that locat PBX is not aware of the 2XXX range you can not direct dial a 2XXX extension. is DUNDi the way around this? or is... |
| 15:51.12 | [TK]D-Fender | ...there another way... <- ad exten(s) to dial over to the other server |
| 15:51.25 | igcewieling | exten => _2XXX,1,Dial(SIP/theotherserver/${EXTEN}) |
| 15:52.31 | NOT_guru | yes I can call extensions from location a to B its just the IVR that doesn't accept locations B's extensions from A and vice versa |
| 15:52.47 | igcewieling | NOT_guru: then TELL YOUR IVR about the extensions |
| 15:53.07 | [TK]D-Fender | NOT_guru: Add extens to match and dial the other server then |
| 15:53.10 | NOT_guru | I am reviewing now |
| 15:53.10 | igcewieling | usually that can be done with an include => but until you provide the requested pastebins I cannot help you further. |
| 15:53.42 | igcewieling | paule32: the included .conf.sample files are not designed to be a working system. |
| 15:53.53 | NOT_guru | I understand igcewieling, I was just clariufying while I gather info |
| 15:54.17 | [TK]D-Fender | NOT_guru: Shouldn't even need a pastebin. Where did you put them so that your IVR can see them? |
| 15:54.30 | NOT_guru | and D-Fender, sorry, I |
| 15:54.31 | NOT_guru | yes |
| 15:54.45 | [TK]D-Fender | NOT_guru: "where" <- |
| 15:54.56 | NOT_guru | I have a route between the locations, its the IVR that can't see them |
| 15:55.11 | [TK]D-Fender | NOT_guru: "route" is not a meaningful term here |
| 15:55.11 | igcewieling | NOT_guru: incorrect, your ivr is not seeing the "route" |
| 15:55.16 | [TK]D-Fender | NOT_guru: What you dial.. is DIALPLAN |
| 15:55.21 | NOT_guru | sorry |
| 15:55.50 | NOT_guru | I understand, I am using web interface terms from freepbx, and I meant outbound route |
| 15:56.01 | igcewieling | Maybe NOT_guru is one of those FreePBX users who wander into this channel by mistake? |
| 15:56.01 | NOT_guru | I am going to gather more info |
| 15:56.02 | [TK]D-Fender | NOT_guru: this is NOT a place for dialplan processing support for FreePBX |
| 15:56.03 | igcewieling | there we go |
| 15:56.08 | [TK]D-Fender | NOT_guru: And none of that is of use here |
| 15:56.11 | igcewieling | NOT_guru: sorry, I can't help you. |
| 15:56.12 | NOT_guru | was not a mistake |
| 15:56.13 | igcewieling | ~freepbx |
| 15:56.13 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 15:56.23 | NOT_guru | I was jst seeing if DUNDi would be a resolution |
| 15:56.27 | NOT_guru | and was asking that |
| 15:56.28 | [TK]D-Fender | NOT_guru: That GUI separate context and the rules for your calls, not YOU |
| 15:56.37 | NOT_guru | but you have given me more info than I expected sorry |
| 15:56.45 | [TK]D-Fender | NOT_guru: Don't ask those questions in here, that's what #freepbx is for |
| 15:56.50 | igcewieling | NOT_guru: dundi is a solution. an overly complicated and wrong solution, but a solution none the less. |
| 15:56.58 | [TK]D-Fender | NOT_guru: Our solutions are NOT FreePBX solutions |
| 15:57.01 | NOT_guru | ok |
| 15:57.07 | tm1000 | also of note. the two people are you are talking to also respond in #freepbx |
| 15:57.09 | NOT_guru | so there are better ways thats all I was looking for |
| 15:57.14 | NOT_guru | sorry I did not mean to offend |
| 15:57.38 | NOT_guru | I appreciate the time |
| 15:57.40 | [TK]D-Fender | NOT_guru: Not a question of "better", it's a question of the rules of how FreePBX works that you have to work in conjunction with |
| 15:57.48 | *** join/#asterisk navaismo (~navaismo@189.241.77.253) |
| 15:57.55 | [TK]D-Fender | NOT_guru: When you configure your own system you can do whatever you want immdiately |
| 15:58.33 | NOT_guru | I understand that I have alot more control in a pure asterisk enviroment, I was just wrapping my head around a problem I had never bumped into before |
| 15:59.10 | igcewieling | NOT_guru: there is nothing terrible about FreePBX. The terrible thing is asking for FreePBX help here. |
| 15:59.12 | NOT_guru | now I know I just need to lookup how to make the IVR aware of the other locations, and not worry as much about dundi |
| 15:59.29 | NOT_guru | again I was not trying to ask a freepbx specific question here |
| 15:59.45 | NOT_guru | I was just asking a general ( or what I thought was a general ) question |
| 15:59.50 | igcewieling | NOT_guru: If you are using FreePBX then all questions are FreePBx questions. |
| 15:59.50 | NOT_guru | sorry again |
| 16:00.25 | NOT_guru | I understand |
| 16:01.02 | NOT_guru | thank you again for the responces |
| 16:01.09 | NOT_guru | I will take this to the other channel |
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| 16:22.18 | davlefouAMD | hi, in my asterisk, i have an context for receive call with serval accounte code but when a revece an call, it all the time with name off the first account and with is account code |
| 16:22.47 | davlefouAMD | It seems ignore over line et ovh accounte code. |
| 16:26.08 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:38.21 | *** join/#asterisk giucio (~giucio@85.118.242.195) |
| 16:40.08 | Qwell | wat |
| 16:41.33 | giucio | Hi, I'm originating a call through AMI, doing a "Action: Originate, Channel: SIP/giucio", plus other parameters. Now, I'd like to use the GetVar() command through AMI. It needs a channel name. The thing is Asterisk doesn't use SIP/giucio as channel name, it builds up something like "SIP/giucio-0000000X" where X is a number. I have no way to know exactly what the channel name is, since it adds these numbers. |
| 16:42.10 | giucio | Is there a way to use GetVar knowing only what I know (which is: SIP/giucio) ? |
| 16:42.36 | WIMPy | Listen to the events you receive on AMI. |
| 16:42.46 | Greenlight | Yea, you can easily identify the actual channel that's been created. |
| 16:42.52 | Greenlight | And then use GetVar on that. |
| 16:43.14 | giucio | Greenlight: how do I can identify it? |
| 16:43.18 | Greenlight | Although you'll see VarSet events anyway, so there's little need to use GetVar explicitly |
| 16:43.40 | WIMPy | Indeed |
| 16:44.22 | Greenlight | There's a few ways to identify calls you've originated. What I tend to do is specify a guid as a variable in the originate command |
| 16:44.41 | Greenlight | Then listen for the VarSet with that guid in it, and use that to "link" the channel that's been created. |
| 16:44.50 | Greenlight | Although there's certainly other ways to do it |
| 16:45.45 | giucio | Greenlight: I see, it is kind of low-level, but I can certainly do it |
| 16:46.12 | Greenlight | It depends what you're looking to do with your application |
| 16:46.54 | Greenlight | I mean, as a quick and easy hack, you can list listen to the VarSet events, and filter based on channel names containg "SIP/giucio" |
| 16:47.17 | giucio | What I'm ultimately trying to do, is to set a channel variable through AMI to the CDR(duration) value |
| 16:47.17 | Greenlight | Assuming you have those events enabled in manager.conf |
| 16:47.44 | giucio | in order to to this, what I'm doing is: Originating the call, wait for the operator event which leads to the setting of this variable |
| 16:47.45 | WIMPy | Most CDR values are read-only. |
| 16:47.54 | giucio | get the CDR variable through AMI |
| 16:47.59 | giucio | and set the channel variable to it |
| 16:48.17 | Greenlight | That one gets set in the hangup |
| 16:48.21 | Greenlight | umm |
| 16:48.27 | Greenlight | Perhaps a UserEvent would be your best bet |
| 16:48.29 | [TK]D-Fender | giucio: tying to make a "time marker" for some point during a call? |
| 16:48.35 | Greenlight | Rather than playing around with varibles |
| 16:48.37 | giucio | [TK]D-Fender: exactly |
| 16:48.42 | giucio | I'm trying to put a time marker |
| 16:48.45 | Greenlight | Oh |
| 16:48.52 | [TK]D-Fender | giucio: What is your trigger for the marker? |
| 16:48.53 | Greenlight | I dont think duration gets updated realtime like that |
| 16:49.37 | giucio | [TK]D-Fender: Operators interact with a web app connected through AMI |
| 16:49.46 | giucio | it is a web application triggering the marker |
| 16:49.48 | WIMPy | I'm pretty sure it's not possible. YOu need to write that to another field. And yes, a userevent might be a good idea. |
| 16:50.03 | giucio | WIMPy: I have created a custom CDR field for that |
| 16:50.14 | WIMPy | Can't you add that to that web app? |
| 16:50.15 | giucio | What I'm trying to do is to put the CDR(duration) value into that field |
| 16:50.17 | Greenlight | You can get access to the duration, but I don't think via the CRD field |
| 16:50.30 | Greenlight | There's a variable that'll give you it though |
| 16:50.31 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:50.45 | Greenlight | I would suggest reading it directly using GetVar |
| 16:50.46 | giucio | Greenlight: if I look at the channel using the "core show channel <x>", the CDR(duration) gets updated in real time |
| 16:51.20 | Greenlight | giucio: Ahh, maybe you can then... for some reason I thought it only got updated during hangup |
| 16:51.42 | giucio | the thing is, it gets hard to know the channel name |
| 16:51.48 | Greenlight | Well |
| 16:51.55 | Greenlight | If you listen for events, no so. |
| 16:52.06 | Greenlight | BUt if your using a web app, and not staying connected... i can see the issue |
| 16:52.17 | Greenlight | eg, you page is connecting in, originating, and disconnecting ? |
| 16:52.19 | giucio | Greenlight: that's the thing, it's stateless |
| 16:52.22 | Greenlight | Yea |
| 16:52.27 | WIMPy | But doesn;t that web app need the channel name? |
| 16:52.53 | giucio | WIMPy: the web app would know the "SIP/giucio" |
| 16:52.57 | Greenlight | giucio: YOur best best is to wait for the channel to be created, *before* disconnecting |
| 16:53.07 | Greenlight | It'll only take a fraction of a second, so it'll be okay |
| 16:53.11 | giucio | what it doesn't know is the part that asterisk adds after it |
| 16:53.17 | giucio | Greenlight: I see |
| 16:53.33 | giucio | Greenlight: I could also receive events from other concurrently created calls though |
| 16:53.39 | Greenlight | And ? |
| 16:53.48 | [TK]D-Fender | [12:49]giucio[TK]D-Fender: Operators interact with a web app connected through AMI <- so the web operator is there to "tag" the call that THEY are on? |
| 16:53.57 | WIMPy | So what happens at that point you want to find? I'm pretty sure you would find an AMI event directly telling you. |
| 16:54.16 | giucio | [TK]D-Fender: yes, they need to tag a particular moment in their ongoing call |
| 16:54.40 | giucio | Greenlight: it gets a bit more laborious... I can do it, I was just hoping to find a faster way |
| 16:54.43 | Greenlight | You would originate the call, with a unique identifier (a guid), cahce that somewher |
| 16:55.37 | giucio | WIMPy: the point operators want to mark is when a sale process starts, so it's entirely human-driven |
| 16:56.16 | Greenlight | You're going to need to know the channel name for that. And you can't tell that at the point of originating the call unless you listen for those events. |
| 16:57.24 | giucio | Greenlight: maybe I can store the channel name in the dialplan using a global variable? |
| 16:57.39 | giucio | (guys, I started using asterisk a week ago, I might be saying nonsense here...) |
| 16:58.00 | Greenlight | giucio: Yes, you could. How would that help though ? |
| 16:58.11 | Greenlight | And what if there are multiple calls ? |
| 16:58.34 | [TK]D-Fender | giucio: tso depends if they could be involved with multiple channels at a time... as to how that we app would know which was "current" |
| 16:58.39 | igcewieling | giucio: why not use a cookie? |
| 16:58.49 | Greenlight | As I see the problem, you need to tie your web user, to the call they've originated. |
| 16:58.53 | giucio | Greenlight: I might set a "X_SIP_giucio" variable to "SIP/giucio-00000x |
| 16:58.55 | [TK]D-Fender | giucio: There are call-state flags, but that fails on local 3-way calling, etc |
| 16:59.10 | Greenlight | giucio: You're only having ONE call active at a time ? |
| 16:59.19 | giucio | Greenlight: one per operator |
| 16:59.25 | igcewieling | you could also write a daemon to connect to AMI and keep track of stuff, then your web script can talk to your daemon. that is what I do for a couple of things |
| 16:59.26 | [TK]D-Fender | giucio: otherwise you could just rely on that with the "base" name and set an agent # variable. Get a channel list, and scan EACH of them for the match to ID them. |
| 16:59.46 | giucio | which means, SIP/giucio, SIP/giucio1, SIP/giucio2 will have each a different channel |
| 16:59.56 | Greenlight | Yea, if you've got one call per peer, you can at least rely on the start of the channel name matchning the peer name |
| 17:00.13 | giucio | [TK]D-Fender: That's another idea, retrieving the channel list through ami |
| 17:00.28 | giucio | Ok I hope there's a commadn for that |
| 17:00.43 | giucio | yup |
| 17:00.43 | Greenlight | Surely that's more complicated than just listening for the channel name in the first place ? |
| 17:00.44 | giucio | nice |
| 17:01.23 | giucio | Greenlight: I'd need to store the channel association somewhere, in order to be available to the afterwards |
| 17:01.30 | giucio | s/the// |
| 17:01.34 | igcewieling | Greenlight: we are talking about trying to do stateful stuff with a web app, it is going to be horribly complicated and ugly no matter what you do |
| 17:01.42 | Greenlight | igcewieling: +1 |
| 17:01.59 | giucio | I think the channel list scanning approach might be simpler |
| 17:02.08 | Greenlight | giucio: As igcewieling says, you could use a cookie or even a session variable. Or a db. |
| 17:02.10 | giucio | in this specific situation |
| 17:02.19 | ChannelZ-Wk | mmmm cookie |
| 17:02.32 | igcewieling | I have a daemon which watches AMI events and updates an internal list of peers with their registration state, my web script queries that on a socket |
| 17:02.47 | Greenlight | Yup that's how i do it as well. |
| 17:02.59 | Greenlight | I'd never had the webserver connecting to AMI directly |
| 17:04.00 | Greenlight | giucio: The only problem I see is if you ever get "stale" channels sitting around, or if an agent has a call on hold etc. |
| 17:04.20 | Greenlight | You're relying on there only ever being ONE channel starting with "SIP/giucio" |
| 17:04.29 | igcewieling | daemons are great to use with web apps. I have one which handles "route locking" so if someone is editing a route, others cannot, stuff like that. |
| 17:05.03 | Greenlight | igcewieling: Why not use a database for that ? |
| 17:05.40 | giucio | Greenlight: uhm I see what you mean |
| 17:05.48 | igcewieling | Greenlight: because then the client would have to constantly query the database to check to see if a user has a route open. |
| 17:06.00 | Greenlight | igcewieling: Ahh good point |
| 17:06.16 | Greenlight | igcewieling: Ever played with SingalR? |
| 17:06.21 | igcewieling | Greenlight: I use long running "comet-like" model for these sorts of things |
| 17:06.24 | Greenlight | *signalR |
| 17:06.33 | igcewieling | never heard of it. |
| 17:07.19 | Greenlight | Microsoft just gobbled it up, but don't let that put you off. It's very cool javascript library for realtime two way communication on webpages, without polling |
| 17:07.46 | Greenlight | I can call a method server side, and *instantly* that fires a javacscript client side function |
| 17:08.27 | igcewieling | Greenlight: ah. my solution is simple in design, small in code size, and already written |
| 17:08.59 | Greenlight | Of course. Just made me thing of it. It's some really nice tech |
| 17:09.08 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 17:09.10 | igcewieling | besides I'd rather eat glass than voluntarily use MS products. Before you /CTCP info me, no, I do not have a choice in running MS windows for work stuff. |
| 17:09.34 | Greenlight | I hear ya |
| 17:09.35 | igcewieling | (for some work stuff that is) |
| 17:10.12 | igcewieling | I get sub-second latency on my existing stuff which is good enough. |
| 17:10.30 | Greenlight | You're web app polls the deamon ? |
| 17:10.36 | Greenlight | *your |
| 17:10.54 | igcewieling | Greenlight: no, they have a long running ajax connection to the daemon on the server |
| 17:11.21 | Greenlight | Ahh nice, same idea as what SignalR has abstracted then really |
| 17:11.48 | Greenlight | Right ... time to leave the office. Laters! |
| 17:11.57 | igcewieling | *nod* since my stuff doesn't have to be cross-browser compatible the code can be a lot more simple too. |
| 17:18.08 | *** join/#asterisk joesmoe (~joesmoe@204.45.29.84) |
| 17:28.58 | paule32 | i have found this site: |
| 17:29.00 | paule32 | http://www.howtoforge.com/asterisk_pbx_linux_p2 |
| 17:29.33 | paule32 | then i copied, the svn files /config to my http server dir |
| 17:29.53 | paule32 | make changes in manager/http.conf |
| 17:30.34 | giucio | Greenlight: I went for the solution you proposed (I believe): I'm originating the call and I'm listening for NewChannel events |
| 17:30.38 | paule32 | the site will occor in webbrowser, but i can't login |
| 17:32.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.73) |
| 17:36.34 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 17:37.41 | [TK]D-Fender | paule32: that GUI has been dead for YEARS. |
| 17:38.40 | [TK]D-Fender | paulc: And that guide is referencing Asterisk 1.4 which was released in *** 2006 *** |
| 17:39.10 | paulc | wakes up? say what? huh? |
| 17:39.18 | paulc | oh.. not me |
| 17:39.23 | paulc | breathes a sigh of relief |
| 17:39.28 | [TK]D-Fender | paule32: And that guide is referencing Asterisk 1.4 which was released in *** 2006 *** |
| 17:39.32 | [TK]D-Fender | yeah, bad aim |
| 17:39.37 | paulc | s'all good :) |
| 17:39.52 | paulc | I got excited for a minute there (doesn't take much eh?) :-) |
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| 17:51.51 | navaismo | if i have accept_outofcall_messages=yes and using DPMA how can I edit the context to allow other clients to use the MESSAGE context? |
| 17:53.36 | *** join/#asterisk SGjunior (~sgjunior@out-pq-134.wireless.telus.com) |
| 17:55.35 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
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| 18:26.49 | *** join/#asterisk paule32 (~paul@dslb-188-106-229-158.pools.arcor-ip.net) |
| 18:26.55 | paule32 | so now on linux |
| 18:27.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.110) |
| 18:27.25 | paule32 | http://pastebin.com/5qWq88Vy |
| 18:27.44 | paule32 | this is the output of dialplan show |
| 18:33.52 | paule32 | how can i make a call |
| 18:33.56 | paule32 | on console |
| 18:33.59 | paule32 | ? |
| 18:36.07 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.110) |
| 18:45.39 | j4jackj | vlad_sta_: your ghost has died |
| 18:48.01 | [TK]D-Fender | paulc: You have nothing you CAN call. There is no dialplan you can execute. |
| 18:48.27 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 18:48.43 | paulc | wakes up again |
| 18:48.47 | [TK]D-Fender | paulc: the dialplan (extensions.conf) configures how * will process calls you send it... you have NO rules of any kind there. * has no instructions on how to handle anything. |
| 18:48.53 | [TK]D-Fender | paule32: ^ |
| 18:48.56 | [TK]D-Fender | DANGIT |
| 18:49.01 | paulc | slaps [TK]D-Fender round with a wet kipper |
| 18:57.35 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:02.09 | igcewieling | Maybe FreePBX is a better match for paule32 |
| 19:02.30 | *** join/#asterisk monsterco (40e76515@gateway/web/freenode/ip.64.231.101.21) |
| 19:03.07 | monsterco | Hi everyone - I am trying to RESET polycom soundpoint IP 335 to factory default using keys 1, 3, 5, 7 and it reboots but when I go to GUI again, I see all the same old information |
| 19:03.18 | monsterco | what could I be doing wrong? |
| 19:03.40 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 19:03.51 | igcewieling | .part #polycom-support |
| 19:05.01 | monsterco | sorry, I know this is Asterisk - I was hoping someone might have a clue |
| 19:05.08 | monsterco | I really don't like polycom phones |
| 19:05.19 | navaismo | +1000^ |
| 19:05.19 | monsterco | they are not making a register attempt even - I am setting line 1 |
| 19:06.29 | [TK]D-Fender | monsterco: When are you doing this? Also if you are provisioning your phones it could be picking up the same configs again after the flush as well |
| 19:06.55 | monsterco | [TK]D-Fender - how can I make sure provisioning is reset too? |
| 19:07.00 | paulc | monsterco: there's opens in the menu on the phone to reset user data, reformat the filesystem, a bunch of other things.. and it's easier to configure through provisioning files than using the web interface generally (more options available) but requires a bit more work to set up. |
| 19:07.13 | [TK]D-Fender | monsterco: You don't know if you're provisioning them or not? |
| 19:07.27 | paulc | You can also set provisioning to static (so it's not Option 60/66/160 etc) to make sure it's not picking anything up.. I got bit by that one before |
| 19:07.39 | monsterco | they were provisioning with Ring Central - but they told me it's open now |
| 19:08.12 | [TK]D-Fender | monsterco: Did you lok in the bootrom yourself to see that it's no longer pointed to them? |
| 19:08.16 | paulc | Plug it in when it's not connected to a network, go do all the reset options on the phone, then make sure it's not getting given pre-provisioning options via DHCP options.. |
| 19:10.32 | monsterco | paulc - Ok, so: unplug the phone, then hold keys 1,3,5,7 (soundpoint ip 335), then it restart, then reset all settings again without plugging the internet? |
| 19:10.58 | monsterco | [TK]D-Fender - I only have remote GUI access and a dumb receptionist on the other end |
| 19:11.26 | monsterco | so I am stuck with resetting or using GUI - and it seems that GUI is totally useless |
| 19:11.39 | monsterco | I mean I don't see any factory reset options there |
| 19:11.39 | igcewieling | monsterco: in 4.x firmware the GUI is slightly more useful |
| 19:11.53 | igcewieling | the factory reset menu option is specific to 4.x as well |
| 19:12.07 | monsterco | igcewieling - there is no ABOUT or STATUS page so I can't tell firmware version |
| 19:12.07 | Katty | HELLO KIDS |
| 19:12.10 | monsterco | how nice is that |
| 19:12.27 | [TK]D-Fender | [15:10]monsterco[TK]D-Fender - I only have remote GUI access and a dumb receptionist on the other end <- no, this is what you look for on the phone itself pre-boot |
| 19:12.52 | igcewieling | monsterco: I've not actually used the polycom GUI in 10 years. No reason to with the awesome provisioning features. |
| 19:13.12 | igcewieling | I looked at the polycom GUI when we updated to the 4.x firmware, but that is about it. |
| 19:13.25 | karl-s | awesome provisioining? you mean the configuration split across 4 different files in a barealy readable xml format? |
| 19:13.48 | monsterco | lol |
| 19:14.02 | [TK]D-Fender | [15:13]karl-sawesome provisioining? you mean the configuration split across 4 different files in a barealy readable xml format? <- Yes! |
| 19:14.09 | [TK]D-Fender | Keeps the idiots out@ |
| 19:14.43 | karl-s | except when they come to you and ask for polycom help... |
| 19:14.57 | monsterco | so, how can I reset the provisioning? |
| 19:15.06 | karl-s | ugh... I miss the aastra provisioning... |
| 19:15.34 | igcewieling | karl-s: you have apparently not used Cisco or Linksys provisioning. |
| 19:15.50 | monsterco | the phone is unlocked - I was told so. But I have a user on other end who is like a cave woman - what should I tell her to do to reset the phone? In advanced settings she only sees Network Reset and SIP Reset - does that mean anything? |
| 19:15.52 | karl-s | I've done Cisco (but only at a base level pre-xml) |
| 19:16.46 | igcewieling | monsterco: go read the polycom provisioning docs |
| 19:16.49 | karl-s | i've not done linksys. We usually only have a handful of SPA devices for Faxes.... The gui is fine for those |
| 19:17.06 | igcewieling | karl-s: I believe linksys uses a binary config file |
| 19:17.28 | monsterco | igcewieling - are you saying I have to un-provision is using a provision file? |
| 19:17.46 | karl-s | yea, that would deserve an "ugh..." for linksys then |
| 19:17.57 | igcewieling | monsterco: no, I'm saying read the polycom provisioning doc so you understand how polycom phones get their config files |
| 19:18.34 | monsterco | right - I would rather pass on that now. I am in an urgent situation and want to have one set at least up and running - so I am looking to reset this one for now and then do the provisioning research later |
| 19:18.55 | monsterco | is there any specific setting on phone menu or GUI that allows me to TOTALLY reset the phone? |
| 19:19.20 | igcewieling | monsterco: I guess you could just keep trying until some random thing works. |
| 19:19.29 | [TK]D-Fender | monsterco: that factory reset does that part, then remove the bootrom settings for provisioning |
| 19:20.21 | igcewieling | monsterco: you might want to connect to the provisioning server and rename the relevant files. |
| 19:21.14 | igcewieling | Though changing the boot option in the config file (don't think you can in the GUI) or change the next-server option of your DHCP server |
| 19:21.56 | igcewieling | sorry, change the boot srv option in the phone UI (not the phone GUI) as someone else mentioned 20 mins ago |
| 19:26.28 | paule32 | is it possible to write an application that act like an teletext app? also sending ton/pulse 0-9, and sending binary data to an 56kb modem? also like access point (a server that listen for incomming connections from 56kb modem, check user and communicate with them) ? |
| 19:26.55 | igcewieling | vomits |
| 19:27.32 | paule32 | vomits? what does it mean? |
| 19:28.17 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 19:29.43 | *** join/#asterisk ACiDV (18e2f1d8@gateway/web/freenode/ip.24.226.241.216) |
| 19:31.47 | monsterco | [TK]D-Fender - ok, the boot room settings is in the phone UI and can be disabled by end user? |
| 19:31.59 | paule32 | so have now, compile the new 10 version |
| 19:32.18 | paule32 | hope you support this version |
| 19:35.18 | [TK]D-Fender | monsterco: You know that is why I told you to specifically go there.... |
| 19:35.41 | monsterco | yep -just checking the settings menu place |
| 19:43.18 | paule32 | can't make a call: http://codepad.org/anJhGswS |
| 19:43.47 | paule32 | how to do? |
| 19:44.56 | ACiDV | Anyone using MulticastRTP channel ? Have a small issue with outgoing spool call file and not sure if it a "know" issue with Asterisk. I try to playback a file but no audio. If I use my SIP phone to reach the dialplan .. (Page(MulticastRTP/basic/....) it work ok. |
| 19:45.19 | [TK]D-Fender | paule32: So far I haven't seen you have any dialplan you can call at all |
| 19:45.30 | monsterco | she is has a hard time finding bootroom options - can you please be specific? thanks |
| 19:45.35 | [TK]D-Fender | paule32: Show us there is something to dial |
| 19:45.52 | [TK]D-Fender | monsterco: SERVER <- |
| 19:46.27 | navaismo | paule32, core show application originate |
| 19:46.59 | monsterco | [TK]D-Fender - sorry i am not with the phone - is that under Advance Settings |
| 19:48.15 | paule32 | navaismo, Your application(s) is (are) not registered |
| 19:48.15 | paule32 | Command 'core show application originate' failed. |
| 19:48.36 | [TK]D-Fender | monsterco: Server Menu > Server Address |
| 19:48.56 | navaismo | paule32, my advise to you is the book |
| 19:48.59 | navaismo | ~book |
| 19:48.59 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:49.04 | [TK]D-Fender | paule32: I'm better you are missing SEVERAL important Asterisk config files. |
| 19:49.18 | [TK]D-Fender | paule32: including MODULES.CONF |
| 19:49.24 | [TK]D-Fender | paule32: And nothing is loading |
| 19:49.55 | [TK]D-Fender | paule32: You should have brought over all the sample configs to at least have the basics and then started scrapping the bits out of the ones you need to personalize |
| 19:49.59 | paule32 | i do a make install |
| 19:50.19 | [TK]D-Fender | paule32: that doesn't install the samples |
| 19:50.28 | [TK]D-Fender | paulc: "make samples" <------------ |
| 19:50.35 | paule32 | ok |
| 19:50.39 | paule32 | thx |
| 19:50.53 | paule32 | any other make(s) ? |
| 19:50.58 | navaismo | hahaha again paulc |
| 19:51.38 | [TK]D-Fender | paulc: My auto-complete is going to EAT YOU :p |
| 19:52.01 | [TK]D-Fender | paule32: No, that should do it. the restart * |
| 19:54.05 | [TK]D-Fender | paule32: Asterisk 10.0.0-rc1 <- this is also a horrible version ot be running, It wasn't even a full-relase version, and is several versions behind on that branch alone... with is ***EOL*** |
| 19:54.27 | [TK]D-Fender | paule32: I highly recommend you upgrade to the latest 11 revision |
| 19:54.32 | [TK]D-Fender | which* |
| 19:55.16 | paule32 | im new in this field |
| 19:55.37 | paule32 | i can't spent time for updates |
| 19:55.43 | paule32 | and changes |
| 19:56.07 | [TK]D-Fender | paule32: You don't even have a configured Asterisk install at all. |
| 19:56.15 | [TK]D-Fender | paule32: You don't have anything to "save" there |
| 19:56.16 | paule32 | so, now, a info text is shown |
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| 20:00.27 | monsterco | [TK]D-Fender - No Server Menu in menu |
| 20:00.42 | monsterco | is it under Admin Settings Advanced |
| 20:01.04 | monsterco | I see Reset to Default |
| 20:01.10 | monsterco | is that it? |
| 20:01.12 | [TK]D-Fender | monsterco: You are mistaken... and in the WRONG PLACE |
| 20:01.38 | monsterco | where should I be in UI? |
| 20:01.39 | [TK]D-Fender | monsterco: There is nothing named "Admin Menu" .... in the *** BOOTROM*** |
| 20:02.05 | monsterco | how do I get to BOOTROOM ? is it not accessed by Menu button on phone? |
| 20:02.34 | [TK]D-Fender | monsterco: I take it you haven't so much as read the admin guides or really looked at the phones.... |
| 20:02.51 | [TK]D-Fender | monsterco: you REBOOT the phone and before the SIP APP starts you go into tht eBOOTROM confg |
| 20:03.17 | monsterco | yep not yet - sorry - is there a key I should press at reboot? |
| 20:04.16 | [TK]D-Fender | You've REALLY never payed attention at boot time.... |
| 20:04.24 | [TK]D-Fender | you have a menu with a countdown.... |
| 20:04.28 | paule32 | thanks for listen, hints and tips .. must go , till next time, have download the pdf and html site from asterisk book |
| 20:04.38 | paule32 | have a good day, bye |
| 20:04.40 | monsterco | I don't use Polycom phones - and this is remote to me |
| 20:06.07 | monsterco | she is rebooting now and reading me all she sees |
| 20:07.45 | monsterco | [TK]D-Fender - so we got to SERVER MENU - now should we just EDIT and remove the provisioning server? or is there a way to disable |
| 20:07.53 | *** join/#asterisk lukerobi (~lukerobi@rrcs-97-79-163-146.sw.biz.rr.com) |
| 20:07.55 | lukerobi | can you have a blf for 1 extension and a differen't speed dial for the same button? |
| 20:08.29 | [TK]D-Fender | monsterco: read the options. |
| 20:08.44 | monsterco | I told her to remove RingeCentral references |
| 20:09.27 | monsterco | she says server type and all the info |
| 20:09.35 | monsterco | should she remove them or is there a way to disable them? |
| 20:11.30 | [TK]D-Fender | REMOVE |
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| 20:13.19 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 20:13.24 | monsterco | [TK]D-Fender - so now server address is removed and phone said can't contact provisioning server - so now, should she go in Advanced Setting and RESET? |
| 20:13.49 | monsterco | or would RESET put phone back to provisioning mode? |
| 20:16.15 | [TK]D-Fender | no, putting the SETTINGS BACK would |
| 20:16.36 | [TK]D-Fender | And of course it can't contact the provisioning server.. we just went through all this to REMOVE IT |
| 20:16.52 | [TK]D-Fender | I don't think you're keeping coherent here... |
| 20:17.40 | [TK]D-Fender | You want to reconfigure this for yourself, then remove the external influences and go configure it yourself |
| 20:19.10 | monsterco | [TK]D-Fender- that's what I thought - so, now that the provisioning address is removed. What steps should be taken to Factory Default it? or was removing address the factory default step? |
| 20:20.50 | [TK]D-Fender | That is so that the provisioning server doesn't just walk right over your attempt to factory reset |
| 20:22.17 | monsterco | Great - so now, I can tell her to press MENU and go to ADMIN settings and then remove local configs? |
| 20:23.09 | monsterco | [TK]D-Fender - ^^^^ seems like as part of provisioning they have disabled all that is entered in GUI to be in effect |
| 20:25.02 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 20:28.06 | [TK]D-Fender | checkout time, BBL |
| 20:28.31 | monsterco | So, which one of these totally reset the Polycom 335? Reset Local Configuration - Reset Device Settings - Format File System |
| 20:29.08 | karl-s | depends on what you are trying to reset... |
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| 20:32.21 | monsterco | karl-s - factory default |
| 20:32.51 | monsterco | I have already removed the provisioning server address from BOOTROOM - so what is my next option to reset the phone to FACTORY DEFAULT? |
| 20:35.49 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 20:36.16 | monsterco | karl-s - so what is my option? |
| 20:36.44 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mwkbindvovirodry) |
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| 20:38.09 | karl-s | monsterco, there are a couple of aspect to a polycom phone: the bootrom, the application, the configuration. I dont think you want to reset all of them. If you do, you would have to do each of those one at a time... |
| 20:38.36 | karl-s | There is rarely a need to reset anything except the bootrom config though |
| 20:38.50 | karl-s | anything beyond that is either a provisioning server issue or a network issue |
| 20:38.59 | monsterco | karl-s - there is no reset in BOOTROOM - I did remove the provisioning address - is that what you mean? |
| 20:39.44 | karl-s | I "think" reset device settings clears the bootrom config but... |
| 20:39.59 | karl-s | the method I usually do is 1) power off/power on phone |
| 20:40.22 | karl-s | 2) while it does the countdown "5 seconds until boot", hold down 1,3,5,7 |
| 20:40.30 | karl-s | 3) it will ask for password, enter 456. |
| 20:40.44 | jmetro | normally i just hit the factory reset button. |
| 20:40.45 | karl-s | 4) the screen will say clearing config (or something to that effect) |
| 20:41.08 | karl-s | jmetro, could be. I dont recall if they were the same or not |
| 20:42.04 | karl-s | for me, I find its easier to walk someone through that process over the phone than having to navigate menu's |
| 20:43.51 | monsterco | so 1,3,5,7 is done at reboot and not when phone is booted? |
| 20:44.17 | monsterco | jmetro - there is no "factory reset" button on polycoms |
| 20:44.41 | karl-s | monsterco, correct. It must be held down when the phone is first booting while it does that countdown thingy |
| 20:45.09 | karl-s | For ip5xx and larger, you hold down 4,6,8,* |
| 20:45.18 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 20:45.26 | igcewieling | is it safe yet? |
| 20:45.40 | monsterco | it's ip 355 - and she tells me that holding 1,3,5,7 doesn't ask her for password |
| 20:45.51 | monsterco | she will try again |
| 20:46.13 | igcewieling | apparently not |
| 20:50.42 | monsterco | can she not press 1,3,5,7 when phone is booted? |
| 20:51.01 | karl-s | sure |
| 20:51.08 | karl-s | you can hold 1,3,5,7 whenever |
| 20:51.18 | karl-s | so long as its held during that countdown |
| 20:52.31 | karl-s | Not that I particularly appreciate Trixbox but... these guys have a guide on it: http://help.fonality.com/index.php?title=IP_Phones/Polycom/Polycom_320%2F%2F330_Flash&highlight=pound |
| 20:53.30 | igcewieling | as does the polycom support center http://support.polycom.com/global/documents/support/technical/products/voice/SoundPointIP_Resetting_Log_Files_QT18298.pdf |
| 20:55.15 | igcewieling | actually this may be better http://community.polycom.com/t5/VoIP/FAQ-How-can-I-reset-Factory-default-my-Phones-configuration/td-p/4307 |
| 20:55.58 | monsterco | What both polycom and fonality mention is that 1,3,5,7 can be held anytime - even after countdown- even when phone is booted fully |
| 20:56.10 | igcewieling | and no, a factory reset does not reset the flash perameters |
| 20:56.15 | monsterco | is that right or no it should be HELD before countdown? |
| 20:56.33 | karl-s | could be if you want |
| 20:56.42 | karl-s | as long as its held during the countdown |
| 20:57.03 | igcewieling | that is correct, when the phone is counting down during the first few seconds of the boot process you can't do the factory reset using the reser key sequences |
| 20:57.05 | karl-s | (unless its not an IP335 such as an IP550) |
| 20:57.41 | igcewieling | during the countdown? I'd have to check that,but I didn't think it worked that way |
| 20:58.12 | karl-s | igcewieling, Yea, i've always done it during the countdown |
| 20:58.26 | karl-s | but they do need to be held for like 2 seconds atleast |
| 20:59.01 | *** join/#asterisk mohadib (~mohadib@unaffiliated/mohadib) |
| 20:59.14 | igcewieling | karl-s: I tried to get monsterco to read the polycom provisioning docs, but he "just needed it done right away" that was like 2 hours ago. |
| 20:59.30 | igcewieling | nothing delays a fix like trying to do it asap |
| 20:59.31 | karl-s | yea but what can you do :) |
| 20:59.46 | mohadib | are their any editors with word completion for dialplans? |
| 20:59.50 | karl-s | Just RMA the phone, that'll be faster |
| 20:59.51 | igcewieling | karl-s: we can put him on /ignore until he gets some sense. |
| 21:00.55 | jmetro | mohadib you need word completion for "dial" ? |
| 21:01.22 | igcewieling | I'm starting to think the channel has been invaded by trolls |
| 21:01.42 | mohadib | jmetro: word completion for the commands and sound files when editing extensions.conf |
| 21:02.00 | karl-s | that would be nice |
| 21:02.07 | karl-s | where can I get that editor? |
| 21:02.11 | mohadib | hah |
| 21:02.16 | mohadib | i was hoping you could tell me |
| 21:02.41 | karl-s | we'll i'm sure theres some kind of extension for visual studio or eclipse |
| 21:02.45 | karl-s | sounds overkill though |
| 21:02.50 | mohadib | i checked for eclipse and could not find one |
| 21:02.59 | mohadib | i was suprised |
| 21:03.12 | karl-s | I created a dinky one for notepad++ but its incomplete |
| 21:03.21 | mohadib | nice |
| 21:03.36 | igcewieling | jedit has a primitive and sometimes incorrect syntax highlighter for AEL |
| 21:03.45 | karl-s | how about extending something like this: http://codiad.com/ ? |
| 21:03.49 | igcewieling | (3rd party I think) |
| 21:03.56 | mohadib | vim does a pretty decent job of syntax highlighting |
| 21:04.02 | karl-s | Have any of you actually deployed complex dialplans via IDE? |
| 21:04.09 | karl-s | err i mean... |
| 21:04.12 | karl-s | Have any of you actually deployed complex dialplans via AEL? |
| 21:04.36 | karl-s | I found it actually is still a little incomplete... |
| 21:04.50 | igcewieling | -= 48 extensions (460 priorities) in 22 contexts. =- <--much of that is AEL |
| 21:05.09 | karl-s | thats pretty good! |
| 21:05.19 | karl-s | let me see my prod ael code...\ |
| 21:05.19 | igcewieling | most of the real work is done in an AGI, but actual dialing is done via AEL script |
| 21:06.06 | igcewieling | 357lines ofAEL including commends and blank lines |
| 21:06.16 | igcewieling | comments and blank lines |
| 21:07.18 | karl-s | hey i'm close: -= 56 extensions (277 priorities) in 13 contexts. =- |
| 21:07.29 | jmetro | lelz |
| 21:07.29 | igcewieling | karl-s: how many TNs? |
| 21:07.38 | karl-s | none actually |
| 21:07.50 | karl-s | its for a small autodialer i built for a medical company |
| 21:07.53 | karl-s | appt reminders |
| 21:08.04 | karl-s | all outbound |
| 21:08.06 | igcewieling | ah. We have something around 10,000 TNs |
| 21:08.11 | jmetro | oh, like a phone tree, i hate those things |
| 21:08.40 | ChannelZ-Wk | Autodialers are evil |
| 21:09.04 | igcewieling | karl-s: "Just a reminder you have an appointment with Doctor Smith on April 1, 2012. If you miss your appt we'll bill your ass a $50 fee. thank you have have a great day!" |
| 21:09.28 | karl-s | kinda, it more like asterisk + cepstral "hello [your name], we are calling to remind you of an appt on [date/time] at [location]" |
| 21:09.35 | ChannelZ-Wk | If I answer the phone and say Hello and you don't respond in 1 second, I hang up. |
| 21:09.35 | karl-s | igcewieling, yea exactly |
| 21:09.46 | karl-s | yea, i turned of amd for that reason |
| 21:10.16 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 21:10.22 | karl-s | I wanted to do AEL for this one but I dont think I'd do AEL again |
| 21:10.32 | karl-s | the structure is great and building it is great |
| 21:10.46 | karl-s | troubleshooting sucks, and not everything is implemented 100% |
| 21:20.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:28.25 | *** join/#asterisk devand (~devand@isnet-wap1-cgy.navigata.net) |
| 21:38.48 | *** join/#asterisk monsterco (40e76515@gateway/web/freenode/ip.64.231.101.21) |
| 21:39.35 | monsterco | so, the Polycom 335 is now reset - and I have filled out the Line 1 credentials and server address in GUI but the phone is not sending any SIP packets to server - could it be that the GUI configs / changes are still disabled? |
| 21:42.11 | jmetro | how do you know its not sending anything |
| 21:42.16 | jmetro | maybe youre just not receiving anything. |
| 21:42.22 | monsterco | I am on the server and sip debug is on |
| 21:42.33 | monsterco | a cisco phone on same network as polycom has registered just fine |
| 21:43.24 | ChannelZ-Wk | Networking on the poly right? It's gotten an IP and everything? |
| 21:43.29 | monsterco | Do I have to put any config info under SIP section as well or is the "line 1 config" enough? |
| 21:43.46 | monsterco | ChannelZ-Wk - does have any IP and I am accessing it's GUI now |
| 21:44.43 | monsterco | polycom does have *an* ip |
| 21:45.02 | [TK]D-Fender | monsterco: you might want to show us something we can comment on... |
| 21:45.32 | jmetro | you need configs to tell it where to send info.. you know, like proxy, registrar, etc |
| 21:45.33 | monsterco | Is there a log for polycom I can look at to see if it is sending SIP packets or not? |
| 21:45.48 | karl-s | are you not provisioning the phones via a provisioning server? |
| 21:45.59 | monsterco | I have set Server 1 address and port |
| 21:46.06 | monsterco | no provisioning |
| 21:46.15 | monsterco | Register: 1 |
| 21:46.23 | jmetro | and have you watched the logs on the polycom / wiresharked it |
| 21:46.58 | karl-s | ouch. I know it sucks doing a provisioning server with polycom but it really is the right way to go |
| 21:47.04 | monsterco | is there logs on polycom GUI? |
| 21:47.07 | karl-s | no |
| 21:47.20 | karl-s | thats why you need to setup a provisioning server |
| 21:47.35 | monsterco | I just gotta say polycom is worst in configs |
| 21:47.41 | karl-s | and you need rw access |
| 21:47.53 | karl-s | so the polycoms can upload configs |
| 21:47.56 | monsterco | Aastra - then Linksys/Cisco - then few other chinese phone sets - then Polycom |
| 21:48.12 | karl-s | its not that bad if you use an XML editor like XML notepad |
| 21:48.40 | karl-s | I can still usually get by using nano but you need to know what options you are after |
| 21:48.54 | monsterco | i am giving up on this - will just send them Aastra phone and tell them to garbage this |
| 21:49.02 | [TK]D-Fender | He's never used Polycom and isn't even in front of this one. |
| 21:49.21 | [TK]D-Fender | monsterco: Show us the config screens |
| 21:49.32 | karl-s | yea, thats gonna be tough to do |
| 21:49.38 | [TK]D-Fender | There are plenty of ways you could have misconfigured it. |
| 21:49.54 | karl-s | A customer once paid me to build them a dedicated web interface for polycom provisioning. There may be a thousand config options but you usually only need to mess with about 5 of them |
| 21:49.58 | monsterco | [TK]D-Fender- Line config screen or SIP config screen? |
| 21:50.05 | [TK]D-Fender | ALL OF IT |
| 21:50.09 | jmetro | Polycoms seem relatively easy to manually provision assuming all the rest of your stuff is setup right..dont even need a provisioning server. |
| 21:50.20 | [TK]D-Fender | ]Yuo stare at one tiny thing and you'll miss everything else |
| 21:50.44 | [TK]D-Fender | jmetro: You only need it if you want anything more than basic functionality |
| 21:52.18 | monsterco | is there an easy way to take snapshot of Chrome browser when page is lengthy? |
| 21:52.25 | monsterco | I want to post it all |
| 21:53.01 | monsterco | [TK]D-Fender - yep the staring is what wastes time |
| 21:53.07 | WIMPy | Save the page? |
| 21:56.45 | monsterco | i will give up today - too late and it's bugging me- i will drop by tomorrow - but wish the stupid thing had a factory reset button so I wouldn't have wasted this long on it |
| 21:57.17 | *** join/#asterisk felipealmeida (~user@187-15-203-147.user.veloxzone.com.br) |
| 21:58.53 | [TK]D-Fender | It did, and you walked in telling us you'd found it |
| 21:59.11 | [TK]D-Fender | But didn't prevent your provisioning settings from walking all over it right after |
| 22:02.26 | paulc | print to PDF with non-custom page sizes? I'd be interested in the answer too.. |
| 22:03.02 | ChannelZ-Wk | I used to have a plugin for Firefox that would make an image of the entire page, however long it was |
| 22:03.29 | WIMPy | Why do you want to make it a picture? |
| 22:03.40 | monsterco | I found a web2pdf plugin but then it has to be posted to some pastebin that accepts pdf |
| 22:04.04 | jmetro | you mean like a dropbox |
| 22:04.17 | jmetro | or sendfile mediafire putfile |
| 22:04.47 | ChannelZ-Wk | I don't. Just sayin |
| 22:05.26 | ChannelZ-Wk | though vectorizing web pages usually turns out badly anyway |
| 22:07.05 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 22:11.46 | *** join/#asterisk JoeyJoeJo (~brian@pool-108-44-169-124.clppva.fios.verizon.net) |
| 22:13.11 | JoeyJoeJo | I'm setting up a sip/voip system where all clients will be behind a NAT (connecting from the internet) and I want all calls to go through my relay. What type of NAT traversal is good for this type of setup? ICE seems good but I haven't found out how to force it to use a relay every time |
| 22:16.34 | paulc | JoeyJoeJo: some documentation searching for "canreinvite" might help you there.. or is your relay not the same as your asterisk box? |
| 22:17.02 | JoeyJoeJo | it is on the same server |
| 22:17.33 | paulc | so terminating provider <--> Your Asterisk <--> End users behind NAT ? |
| 22:17.42 | JoeyJoeJo | Is canreinvite an Asterisk thing or is it part of the sip protocol? |
| 22:17.49 | JoeyJoeJo | Yeah, that's about right |
| 22:18.49 | paulc | it's an asterisk thing.. in sip.conf |
| 22:19.06 | paulc | basically says "make the media flow through me and don't let two separate end points exchange media directly" |
| 22:19.08 | JoeyJoeJo | I see, but reINVITE is part of sip |
| 22:19.16 | JoeyJoeJo | Gotcha |
| 22:19.31 | paulc | ok, sure, technically it's a SIP thing.. and I'm talking about how to control it within Asterisk ;-) |
| 22:30.49 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:32.40 | karl-s | hasn't canreinvite been deprecated as an option and replaced with directrtp? |
| 22:36.21 | file | directmedia, and "canreinvite" still exists as a valid option name |
| 22:36.26 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 22:36.28 | file | it just doesn't reflect what it means |
| 22:54.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.11) |
| 22:58.14 | hjf | hi guys |
| 22:58.20 | hjf | me and my dumb questions again |
| 22:58.30 | hjf | can i use an FXO card as a fax/ |
| 22:58.31 | hjf | ? |
| 22:58.51 | hjf | like a fax server connected to an analog line |
| 23:07.27 | hjf | er |
| 23:07.31 | hjf | not fxo card |
| 23:07.39 | hjf | i meant fxo device like an spa3000 |
| 23:08.42 | hjf | do that kind of devices support, for example, fax detection on the FXO and .. do something about it? like for example, dialing another extension and T.38 the fax to it"? |
| 23:09.54 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 23:10.42 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:15.43 | jmetro | Was there any way to make the timeout for extension matching shorter? |
| 23:15.53 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 23:16.08 | WIMPy | core show function TIMEOUT |
| 23:16.55 | WIMPy | Or juat make sure they don't overlap. |
| 23:18.04 | [TK]D-Fender | hjf: IIRC the SPA-3000 doesn't support T.38, the SPA-3102 does |
| 23:18.45 | [TK]D-Fender | hjf: If you're looking to do faxing I'd get a real card though... |
| 23:19.07 | jmetro | client has audio done with "1" as a prompt... odnt ask me why |
| 23:19.14 | jmetro | but of course their extensions are all 1xx |
| 23:20.51 | igcewieling | jmetro: the only good solution is to not have an option 1 |
| 23:21.16 | karl-s | you "could" work around it |
| 23:21.38 | karl-s | set extension '1' to go to another context |
| 23:21.42 | igcewieling | playing with timeouts will make someone unhappy. they will either be unhappy because the timeout is too long or they will be unhappy because it is too short, but one way or the other someone will be annoyed |
| 23:22.12 | WIMPy | Definitely. |
| 23:22.14 | jmetro | i'm working with a company full of old ladies, im pretty sure 5 seconds is too short. |
| 23:22.31 | WIMPy | But how is switching to another context going to help? |
| 23:22.49 | igcewieling | jmetro: so you want the caller to wait more than 5 seconds between pressing IVR option 1 and asterisk processing the request |
| 23:22.52 | jmetro | it would only help if the IVR had "press 1 to do this, or 2 to hear the rest of the menu" |
| 23:23.18 | jmetro | as it stands 1 and 1xx already go different places, so its not a legit answer. |
| 23:23.37 | igcewieling | if you know your parties extension you dial it now, for sales press 2 |
| 23:23.58 | jmetro | i always nudge my client away from 1 but in this case they...are not going to re-record. |
| 23:24.39 | igcewieling | then they will have angry and/or confused callers |
| 23:24.56 | jmetro | they are angry and confused long before they call in |
| 23:24.57 | hjf | [TK]D-Fender: yeah well i have several problems with cards. for one, i don't use Linux (i use freebsd). and also, my home server (HP Microserver) only accepts pci-express half-height cards... |
| 23:25.01 | igcewieling | ...to complain about how long it takes to read your selection press 9 |
| 23:25.20 | jmetro | their IVR is a full minute long for 5 options |
| 23:25.37 | hjf | i thought the spa3000 was the same as the 3102 sans router |
| 23:25.46 | igcewieling | if you are confused or don't know the option you need stay on the line and someone will be with you shortly. If you are calling from a rotary phone then go down to a store and buy a phone made in the past 30 years. |
| 23:26.21 | hjf | jmetro: i bet you never called WDC's IVR |
| 23:26.30 | jmetro | If you are dialing on a rotary phone, yell "Help" as loudly as possible for thirty seconds into the receiver. |
| 23:27.41 | hjf | "thank you for calling wdc support center, next you'll hear a menu with several options, which you can choose by pressing the number. if you need support for hard disk drives, press the number one. if you need support for SSD devices, please press the number 2 on your phone keyboard......" |
| 23:27.50 | hjf | it's got to be the longest menu i ever heard |
| 23:28.15 | jmetro | on the rare occasions i have to call tech support for a company, i feel like EVERY SINGLE ONE is a Voice Only IVR |
| 23:28.25 | WIMPy | You know that episode of Married With Children where Al called Dodge? |
| 23:28.34 | jmetro | like great, what am i going to do as a deaf guy, calling this line |
| 23:28.38 | igcewieling | If I was a suspicious person I'd be thinking all the new people on the channel deliberately make sure to only do stuff which Asterisk is not very good at or doesn't support at all, and what the things Asterisk is good at they try doing it the wrong way. |
| 23:28.51 | [TK]D-Fender | [19:24]hjf[TK]D-Fender: yeah well i have several problems with cards. for one, i don't use Linux (i use freebsd). and also, my home server (HP Microserver) only accepts pci-express half-height cards... <- Sangoma A200 = half height. And FreeBSD can run this fine |
| 23:28.54 | hjf | jmetro: problem is you never know if "0" will connect you to a person, or "9" will |
| 23:29.11 | jmetro | if 0 isnt the digit to reach a person, the company isnt worth talking to |
| 23:29.26 | hjf | [TK]D-Fender: what if we add VMWAre ESXi to the equation? |
| 23:29.44 | igcewieling | heh, I often see people pressing 0 (and being denied) in our customer's IVRs |
| 23:29.47 | [TK]D-Fender | hjf: Depends who's the host |
| 23:29.53 | karl-s | hjf, dont add that in. it will start a holy war |
| 23:29.58 | WIMPy | hjf: Get a modem from the skip. |
| 23:30.12 | igcewieling | hjf: go ahead, it's not like it will make stuff any more confusing for you |
| 23:30.17 | [TK]D-Fender | hjf: Then again, you're designing your system to fail. Virtualizing for fax? Sorry, we can't fix cheap, lazy, OR stupid :P |
| 23:30.47 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 23:30.47 | hjf | igcewieling: i like challenges i guess |
| 23:31.00 | hjf | WIMPy: from what |
| 23:31.26 | igcewieling | hjf: A challenge is creating a useful and stable PBX. You're just doing the geek version of jacking off. |
| 23:31.30 | hjf | [TK]D-Fender: nah i was thinking of virtualizing so i can use freebsd for zfs and have a real linux for "things" |
| 23:32.13 | hjf | igcewieling: i could just do what everyone else does and just install elastix and be done with it |
| 23:32.23 | hjf | but what's the fun in that? |
| 23:32.23 | igcewieling | nobody here installs elastix |
| 23:32.47 | igcewieling | hjf: start by reading the Asterisk book. |
| 23:32.48 | ChannelZ | except in their pants |
| 23:33.05 | hjf | igcewieling: why? i can just ask here and annoy people with noob questions |
| 23:33.28 | hjf | igcewieling: i can open SSH for you so you can configure my system too =D |
| 23:33.34 | igcewieling | hjf: eventually the few people left who still talk to you will stop. |
| 23:33.57 | igcewieling | heck, you spend most of the time on my /ignore list. |
| 23:34.45 | jmetro | a virtual fax,g729 asterisk server running a2billing with a zfs file system |
| 23:34.51 | hjf | igcewieling: i take that as a compliment |
| 23:35.00 | jmetro | initially installed as a FreePBX distro |
| 23:35.12 | igcewieling | ideally this channel is a meritocracy and if you don't read the Asterisk book you have no merit. |
| 23:35.17 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 23:35.18 | [TK]D-Fender | jmetro: On an ARM system, virtualized |
| 23:35.45 | hjf | inb4 "can i run asterisk on a raspberry pi?" |
| 23:36.13 | [TK]D-Fender | yes, and there are few prefab distros for it already |
| 23:36.15 | jmetro | [TK]D-Fender: powered by solar energy, connected to the internet via a wireless signal beamed a mile across a corn field |
| 23:36.29 | hjf | btw |
| 23:36.32 | [TK]D-Fender | jmetro: in snow 20' high, uphill, BOTH WAYS |
| 23:36.44 | hjf | i ask many hypothetical questions |
| 23:37.13 | hjf | to see what things can be done and what things can't |
| 23:37.14 | jmetro | ever tried to tether 4g cell signal into cat5 and register a phone off it? almost worked but we were on verizon 8-( |
| 23:38.06 | igcewieling | jmetro: sort of and I got the same thing |
| 23:38.10 | karl-s | jmetro, thats why you need to vpn across it |
| 23:39.47 | jmetro | anyway ive worked about 12 hours today, time to go home. |
| 23:45.37 | *** join/#asterisk viasanctus (~viasanctu@unaffiliated/viasanctus) |
| 23:46.03 | viasanctus | anyone has virtual pbxs on a cloudstack kvm env? |
| 23:54.42 | pabelanger | viasanctus, cloudstack no, kvm yes |
| 23:54.58 | viasanctus | how did that go for you? |
| 23:55.20 | viasanctus | am planning on running several pbxs in a cloud with kvm |
| 23:56.03 | pabelanger | works |
| 23:56.10 | viasanctus | scale? |
| 23:56.11 | pabelanger | all depends on what you want to do |
| 23:56.20 | pabelanger | scale outwards not up |
| 23:56.25 | viasanctus | make phone calls :) |
| 23:56.36 | pabelanger | All we do it kvm |
| 23:57.20 | viasanctus | how many users ? |
| 23:57.43 | pabelanger | couple hundred |
| 23:57.45 | viasanctus | is 150 concurrent calls reastic with 2GB RAM / 2 vcpus ? |
| 23:57.53 | viasanctus | realistic* |
| 23:57.56 | pabelanger | again, depend on what you need to do. |
| 23:58.02 | pabelanger | your best to set it up and see |
| 23:58.13 | pabelanger | if you need more, just bump up your kvm instances |
| 23:58.27 | viasanctus | that's something I don't really get |
| 23:58.45 | viasanctus | our asterisk consultant tells us that 1 vm can handle about 8k users with 150 concurrent calls |
| 23:59.01 | viasanctus | then cluster the asterisk vm with a second for more users |
| 23:59.17 | pabelanger | why would you put that many on 1 vm? |
| 23:59.27 | viasanctus | why would a 2nd vm allow more users instead of adding more resources to the initial vm |
| 23:59.40 | viasanctus | I probably will not, but that's the max he went to |
| 23:59.55 | viasanctus | in the end it's all running of the same hardware.. |