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| 00:22.39 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:22.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:19.06 | dunderproto | I need a phone number solely to use as a pager. All I need is to see the caller ID (or receive voicemails). Is there any clever trick where I can avoid signing up to a DID number? Where I live, the cost of a local DID number is expensive ($200/yr, no include minutes) plus requires much paperwork. |
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| 01:24.20 | wabbits | dunderproto just out of curiosity where do you live? |
| 01:24.40 | dunderproto | wabbits: Asia |
| 01:24.54 | wabbits | wow that was way to specific. |
| 01:25.01 | dunderproto | :) |
| 01:25.09 | wabbits | do you need the did to be in Asia? |
| 01:25.49 | dunderproto | preferably. I'm afraid that no one would want to call me if I used an int'l number |
| 01:26.23 | wabbits | The phone number is the DID its how your calls or messages get routed to you. |
| 01:26.54 | dunderproto | I think if they saw an int'l number, they'd get scared of the calling fees |
| 01:27.23 | wabbits | fair enough |
| 01:28.27 | wabbits | if you give me a country I will check to see how much the did costs. |
| 01:28.27 | dunderproto | I imagine there is a way to offer a toll-free int'l number, but I suspect I wouldn't be able to communicate that to them properly |
| 01:29.23 | dunderproto | wabbits, thanks for the offer. It's OK, no need to trouble yourself. So I guess a DID is my only option? If so, I can the necessary comparison shopping |
| 01:29.58 | wabbits | hold on one minute please. |
| 01:30.57 | wabbits | japan is $9.00USD a month |
| 01:31.03 | wabbits | includes incoming minutes |
| 01:31.23 | wabbits | 1800 minutes if memory |
| 01:31.25 | wabbits | serves |
| 01:32.31 | wabbits | malaysia is $6.50 for the same thing |
| 01:32.44 | dunderproto | what DID provider do you use? |
| 01:34.39 | Samot | dunderproto: Are there any rules about SIP/VoIP for Taiwan? |
| 01:35.49 | Samot | There are parts of the world, specially in Asia, that do not allow traffic on UDP 5060 and other ports/protocols. |
| 01:36.29 | dunderproto | Samot: I'm able to make SIP phone calls without a problem at the moment. I just can't receive any calls |
| 01:36.45 | Samot | OK. |
| 01:37.17 | Samot | So if you already have a VoIP provider to make outgoing calls with, you'll need a DID for people to call you. |
| 01:37.28 | Samot | From regular phones or other services. |
| 01:38.29 | Samot | But using a provider outside of Taiwan may prove problematic. |
| 01:44.24 | dunderproto | Samot, wabbits: Thank you for your help. |
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| 02:26.30 | craysiii | Is there a recommended book for v13? |
| 02:26.42 | Samot | No. |
| 02:26.59 | Samot | Because as far as I know, there isn't A book for v13. |
| 02:27.26 | Samot | ~book |
| 02:27.26 | 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 |
| 02:28.02 | Samot | Between that, the Wiki and the sample configs you should be pretty good. |
| 02:28.28 | craysiii | Yeah I have already started to read that one, just wanted someone more up to date in case some info is old/bad |
| 02:28.43 | Samot | Well. |
| 02:28.50 | Samot | 12 introduced PJSIP |
| 02:29.00 | Samot | Some changes to CEL |
| 02:29.19 | Samot | ARI |
| 02:29.34 | Samot | So like I said, the book covers a lot of the basics that haven't really changed. |
| 02:29.41 | Samot | dialplan, queues, etc. |
| 02:30.01 | craysiii | Ok awesome, I am just to the part of installing asterisk itself |
| 02:30.10 | Samot | Like I said, those three sources are going to get you where you need. |
| 02:30.16 | craysiii | Thank you |
| 02:31.22 | craysiii | I work with Panasonic PBXs but would like to learn asterisk to see if my company can eventually start selling asterisk cloud pbx service as well |
| 02:41.50 | drmessano | Maybe in a couple years |
| 02:53.02 | drmessano | Learn Asterisk first |
| 03:31.23 | craysiii | That's the plan. Was probably going to shoot for freepbx actually but I want to have a solid understanding of the entire system before doing so |
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| 17:00.43 | monsterco | hi - anyone familiar with IP 500 Avaya? There are 16 ports in the front of it and I see that port 1-8 are turnning phones sets on. How do I know which ports are for trunks? (i.e. ports to connect telephone provider to)? |
| 17:01.38 | Samot | I believe their documentation would have that information. |
| 17:04.06 | monsterco | Samot - I checked and with expansion modules it's confusing. I am wondering if all ports are FXO / FXS |
| 17:04.15 | [TK]D-Fender | Don't ask a Tesla tecnician about your VW deisel engine. |
| 17:04.49 | monsterco | [TK]D-Fender - lol that was deep |
| 17:05.16 | Samot | Well, you're asking questions about a specific FXO/FXS gateway. |
| 17:05.24 | Samot | In a room not about it. |
| 17:06.20 | Samot | I get it's a shatter shot question in hopes it would hit someone that actually knows about this specific gateway... |
| 17:06.28 | Samot | s/shatter/scatter/ |
| 17:08.04 | [TK]D-Fender | They have manuals. Their cards have part #'sd |
| 17:08.16 | [TK]D-Fender | And this is the "I hate commercial PBX's and don't care to touch them" channel |
| 17:08.49 | [TK]D-Fender | You also don't go expecting free service from a toyota dealer for your Ford Focus |
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| 17:20.12 | [sr] | i know what is like to have commercial product that doesn't have support (it has on paper only) |
| 17:20.39 | [sr] | specially avaya, that are only in goverment's and similar places, resuming, places that people decide with others money |
| 17:22.11 | [sr] | thats avaya and cisco, in fact! |
| 17:22.14 | [sr] | microsoft way |
| 17:22.32 | [sr] | thats how they grow, sad but very true |
| 17:22.49 | [sr] | not that the product is bad, microsoft has two beatiful things, AD and TS |
| 17:25.01 | Alblasco1702 | monsterco, https://downloads.avaya.com/css/P8/documents/101028367 |
| 17:25.39 | Alblasco1702 | monsterco, google is your friend |
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| 17:45.56 | [TK]D-Fender | [sr]> i know what is like to have commercial product that doesn't have support (it has on paper only) <- it has support. Call Avaya. |
| 17:46.08 | [TK]D-Fender | And it is not just in govenrment's |
| 17:46.19 | [TK]D-Fender | my head office uses an IP Office 50 |
| 17:46.29 | [TK]D-Fender | This is dumb consumer tech like anything else |
| 17:50.27 | xous | so, I have a provider that will send me inbound SMS to a SMSC. Are there any existing OSS tools I can use to forward that SMS as a email to an end user? |
| 17:51.00 | xous | It looks like I could kludge my own with kannel + an MTA |
| 18:05.49 | igcewieling | you'd have to ask the SMSC about API interfaces into their systems. |
| 18:08.01 | xous | That'd work for outbound |
| 18:08.13 | xous | for inbound they want to route it to my SMSC |
| 18:09.14 | xous | reading more of th documentation for kannel it seems I could easily kludge this with kannel + a php script that takes a number + sms content. |
| 18:23.36 | wabbits | I get sms from my provider via http post and send them back using their rest api |
| 18:27.14 | igcewieling | If you are REALLY using "ETSI ES 201 912" why not use app_sms? |
| 18:28.42 | wabbits | igcewieling who are you talking to? |
| 18:28.48 | igcewieling | xous: |
| 18:28.53 | wabbits | thanks |
| 18:28.59 | igcewieling | wabbits: YOU are doing what the smart people so. |
| 18:29.12 | igcewieling | s/so/do |
| 18:29.27 | xous | I'll see if my provider will deliver via http post. |
| 18:29.59 | wabbits | who is your provider? |
| 18:30.07 | xous | isptelelcom |
| 18:30.25 | xous | isptelecom |
| 18:30.35 | igcewieling | Sadly, "SMS" has become like "SIP Trunk" i.e. a term which doesn't really mean "SMS", it now means "text message, usually on a phone" |
| 18:30.46 | xous | yeah |
| 18:33.10 | wabbits | does any itsp send/recieve sip simple messages? |
| 18:33.14 | *** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-30.dsl.bell.ca) |
| 18:33.34 | wabbits | I have a vague memory that anveo might |
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| 18:34.04 | igcewieling | I don't know of any, but I don't do much messaging. |
| 18:34.36 | monsterco | if a firmware is named like this: 7942 and 7962 SIP IP Phone Firmware Files Only - Compatible CUCM Versions: 7.1(5)+ |
| 18:34.42 | monsterco | cmterm-7942_7962-sip.9-4-2SR1-1.zip <<< can I use it for SIP with any system or it must have Cisco Unified Call Manager? |
| 18:34.54 | wabbits | igcewieling fyi I blocked udp at my firewall and now I get no hack attempts 0 in 72 hours. Normally I would get 2500-5000 in this time frame. |
| 18:35.34 | igcewieling | wabbits: I used to get that many attempts to, until I allowed the registrations and jailed them, now I get 4 - 5 per day. |
| 18:35.55 | igcewieling | more than I'd like, but still not terrible. |
| 18:37.56 | monsterco | I am trying to configure a Cisco 7942 to Asterisk and people in Cisco hate this question lol |
| 18:38.09 | monsterco | but I am at loss with the firmware and configuration file - there is no proper samples |
| 18:45.41 | wabbits | mosterco are you talking remote provisioning? |
| 18:45.51 | wabbits | s/mosterco/monsterco |
| 18:46.33 | monsterco | wabbits - no, just getting the phone to regiter to asterisk sip |
| 18:47.04 | monsterco | when phone turns on it says Unprovisioned on bottom left. I think that is because it is not taking my xml or my xml config file is missing something |
| 18:47.14 | monsterco | first time working with this and it's really hard to get this working |
| 18:47.16 | wabbits | enable sip trace in context of the asterisk console log and analyse the trace |
| 18:47.41 | wabbits | this is a cisco question. |
| 18:48.09 | wabbits | looks like its looking for a provisioning server it can't find. |
| 18:48.31 | wabbits | does your asterisk server register other sip ua's? |
| 18:50.02 | wabbits | afk 120 |
| 18:51.23 | monsterco | wabbits - i haven't got there at all yet |
| 18:51.40 | monsterco | the phone doesn't even send packets to asterisk server |
| 18:51.48 | monsterco | i need a sample xml config file first |
| 18:52.03 | monsterco | Cisco channel ppl hate these questions because i am trying to get it running with asterisk |
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| 19:53.38 | wabbits | sigh I would have helped him if he would have stayed around. |
| 19:55.09 | wabbits | step1 register a sip client on pc or smartphone. |
| 20:02.37 | drmessano | He needed help with those phones, specifically |
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| 20:51.13 | xous | heh. yes you can get 79XX to register with things other than a call manager |
| 20:51.20 | xous | the phones are old as hell though |
| 20:51.42 | Samot | No one said it wasn't possible. |
| 20:51.45 | Samot | It's just a PITA. |
| 20:51.52 | xous | yeah |
| 20:51.57 | drmessano | Its not worth it |
| 20:52.02 | Samot | ^^^^ |
| 20:52.05 | xous | heh. I made by boss throw out around 600 phones |
| 20:52.22 | xous | he still doesn't know I could have made them work if I had written my own provisioning server ;P |
| 20:52.38 | drmessano | Its like masturbating with a cheese grater and arguing "It's not completely horrible" |
| 20:53.15 | xous | hhaha |
| 20:53.19 | xous | they are cheap as fuck though |
| 20:53.29 | xous | I just sold a bunch to someone |
| 20:53.37 | xous | he didn't really want the phones |
| 20:53.41 | xous | he wanted the power cubes |
| 20:53.42 | xous | :P |
| 20:54.59 | drmessano | I told him "Stop trying to be a hero and sell him new phones" |
| 20:55.05 | drmessano | Hopefully he got the message |
| 20:55.07 | xous | heh. I could sell a 7960 for $20+30 for a power cube |
| 20:55.20 | xous | or I could sell a VVX301 for $221.00 |
| 20:55.59 | xous | only thing really better with the VVX301 is support for the HD codecs |
| 20:56.05 | xous | and the stand isn't likely to breka |
| 20:56.28 | xous | provisioning is slightly better. |
| 20:56.30 | drmessano | Its horribly counterproductive to dedicate that kind of time to old phones when you can just be a good guy and sell your customer something that works |
| 20:56.41 | xous | yeah |
| 20:56.51 | xous | customer wants a sip account |
| 20:56.57 | xous | they can buy one |
| 20:57.17 | xous | but they get zero support for phones that aren't on the supported list AND bought from us. |
| 20:57.43 | xous | Boss made an exception once and I spent two days trying to get 5 VVX 300s to provision |
| 20:58.01 | xous | I dunno if it was fucked network or fucked hardware |
| 20:58.21 | xous | factory resetting them several times eventually got the phones to grab an IP |
| 21:14.39 | xous | tel /wind 3 |
| 21:14.41 | xous | balrg. |
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| 21:25.17 | bonzibuddy | hey folks |
| 21:25.37 | bonzibuddy | trying to get the command 'console dial' working |
| 21:25.41 | bonzibuddy | but I am having no luck |
| 21:25.54 | bonzibuddy | using alsa |
| 21:26.41 | bonzibuddy | but i get like "no such command" errors |
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| 21:27.46 | bonzibuddy | anything I can try? I know the alsa interface is working, I can arecord/aplay just fine |
| 21:28.17 | drmessano | You didn't compile with chan_console support |
| 21:28.38 | bonzibuddy | hmmm well, its installed from repo in this instance |
| 21:29.02 | drmessano | Okay |
| 21:29.06 | drmessano | Well |
| 21:29.12 | bonzibuddy | and it worked at one point in time... i upgraded and now it no longer works? |
| 21:29.14 | drmessano | Either find the package for it |
| 21:29.18 | drmessano | or compile your own |
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| 22:49.59 | bonzibuddy | hmm i have chan_alsa |
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| 22:50.03 | bonzibuddy | but no console.so |
| 22:50.11 | bonzibuddy | or chan_console.so |
| 22:54.20 | bonzibuddy | i can module load chan_alsa |
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| 23:24.27 | snadge | haha... 79xx.. straight in the bin |
| 23:25.01 | snadge | if someone offered me 100 of them.. in mint condition.. for free.. i wouldn't take them |
| 23:25.13 | snadge | i'd probably just punch them in the face |
| 23:25.26 | snadge | and say how dare they insult me in such a degrading way ;) |
| 23:26.23 | snadge | if they offered me $50 to dispose of them.. i'd probably still not do it |
| 23:26.52 | Maliuta_ | I use 79xx series with * just fine |
| 23:27.10 | snadge | im gonna just agree with the cheese grater masturbating comment ;) |
| 23:28.19 | snadge | they are pretty cool from a retro perspective.. dat.. 24 ring tone |
| 23:28.28 | snadge | and you can put the weather and stuff on them |
| 23:28.34 | snadge | ... but just no |
| 23:28.48 | Maliuta_ | they are cost effective for many of the groups I do stuff for |
| 23:29.13 | Maliuta_ | the sound quality in them is good, and you can make calls from them |
| 23:29.17 | snadge | they're built like a brick shithouse .. but they just suck |
| 23:29.27 | snadge | i'd say buy yealink.. or just dont bother |
| 23:29.34 | snadge | :P |
| 23:29.45 | snadge | if you cant afford $200 per phone.. then you don't need a phone |
| 23:29.53 | snadge | use zoiper like a proper cheap bastard |
| 23:31.14 | Maliuta_ | and if you have a limited budget you'd rather put budget into the calls than the kit |
| 23:32.25 | snadge | im trying to think of scenarios where the budget is so limited, that you'd use end of life phones that are a pain in the rectum |
| 23:32.37 | snadge | maybe an orphanage.. soup kitchen |
| 23:33.00 | snadge | lifeline |
| 23:33.02 | Maliuta_ | I do work with groups that are basically funded by donation |
| 23:33.45 | Maliuta_ | and the 79xx series aen't a pain at all |
| 23:34.00 | snadge | they require a provisioning server.. and they dont work properly with nat |
| 23:34.09 | snadge | asides from those two things |
| 23:35.08 | snadge | so they have to be supported by something and someone.. as opposed to a stand alone device, that can be configured easily with a web interface.. and does work behind nat by default |
| 23:35.10 | Maliuta_ | tftp is not hard to do, and I like to keep * in the middle so there is no directmedia/reinvite to deal with |
| 23:35.55 | snadge | i couldn't get the 79xx to work behind nat.. without disabling nat in the extension's settings.. and manually forwarding a unique port, for each phone |
| 23:36.29 | snadge | it would appear that the old sip firmware on them.. doesn't support respond to source port |
| 23:38.24 | snadge | ie.. they just response on port 5060.. unless you set the control port manually.. something dumb like that anyway |
| 23:50.22 | *** join/#asterisk nathani (~nathani@206.162.242.72) |
| 23:50.40 | nathani | I am having issues savinga voicemail greeting within voipms |
| 23:50.47 | nathani | keeps reverting to |
| 23:51.03 | nathani | "Person at ext ... is not available" Please record a message |
| 23:53.59 | [TK]D-Fender | If youare having trouble with a voip.ms mailbox then you should be contacting them |