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| 02:00.19 | ruben23 | hi there guys i have an asterisk server and i register extension 100-110 - now all are registed and dialing but problem, along the course of calling some extension suddenly get unreachable and then become unreachable again..all this extension are using the same gateway router and they are remote from asterisk...any idea guys how to correct this..? |
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| 05:37.40 | kevcox | I'm looking for a low cost cloud based VOIP service provider for a non-profit organization that allows calls from the Philippines to the states. The hosting service must have an auto attendant and a call queue. Any suggestions? |
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| 06:10.21 | Penguin | kevcox: I don't know if it meets all of your criteria, but VoIP.ms has an attendant and queue in the web. |
| 06:12.06 | kevcox | Penguin: Thanks checking it out now...do you have a recommendation for toll free service? |
| 06:12.39 | kevcox | If I have to I can always use 8x8 but their toll free service is kind of expensive. |
| 06:13.26 | Penguin | I typically suggest VoIP.ms and/or Flowroute for toll free and regular numbers. |
| 06:15.05 | Penguin | I don't know how your international factor fits in, though. I deal in US affairs only. |
| 06:16.14 | kevcox | Looks like they have servers in the UK so that may not be to bad. I tried using VoipO but their servers are all in the US and there is to much latency. |
| 06:20.42 | Penguin | Where will your SIP phones be located? |
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| 06:23.04 | kevcox | In the states and the call center would be in the Philippines. |
| 06:23.28 | Penguin | That... doesn't make sense to me. |
| 06:24.15 | Penguin | If your call center is in the Philippines, wouldn't your phones also be in the Philippines? |
| 06:24.23 | kevcox | So customers would call a toll free number that would be routed to a US cloud based PBX and the people answering the phones would connect to it from the Philippines. |
| 06:24.50 | kevcox | Make sense? |
| 06:25.02 | Penguin | Does my question make sense? |
| 06:25.17 | kevcox | Yes |
| 06:25.36 | kevcox | The answer to your question is "Yes". |
| 06:25.54 | Penguin | So then why are your SIP phones in the US rather than in the call center? |
| 06:26.30 | kevcox | SIP phones would be mostly in the Philippines with a few in the states but all call traffic is originating in the states. |
| 06:26.36 | Penguin | And if you have SIP phones in the US, you don't need to worry about a toll-free phone number at all. |
| 06:27.00 | kevcox | Toll free numbers is for inbound calls from customers in the US. |
| 06:27.22 | kevcox | Sorry if I'm confusing you. |
| 06:27.34 | Penguin | I can understand regular phones calling toll-free numbers. |
| 06:28.38 | Penguin | So your call center agents are not only in the call center in the Philippines, but also in the US? |
| 06:29.17 | kevcox | Correct, but mostly they will be in the Philippines. Rare that calls are answered in the US but could be. |
| 06:29.33 | Penguin | I think we're on the same page now. |
| 06:31.26 | ruben23 | high guys what is the efect of the value of timeout session in SIP are pretty high..? |
| 06:31.29 | ruben23 | any idea..? |
| 06:31.41 | kevcox | Penguin: I know know enough about VOIP to be dangerous....I'm reading over the website you provided and I'm having a difficult time understanding what they provide. |
| 06:32.09 | kevcox | Do they provide the hosted cloud PBX or just dial tone? |
| 06:32.14 | Penguin | The only thing I know to do is give it a try. VoIP.ms will refund the unused portion of your money if you don't like the service. |
| 06:32.53 | Penguin | Imagine they have asterisk at their house and you get a web interface to set up the pieces that you need. |
| 06:34.03 | kevcox | Okay, so they provide the hosted PBX which I manage via a web console and it appears they provide the dial tone as well? |
| 06:35.24 | Penguin | They provide phone numbers and termination service, of course, as well as features such as digital receptionist, call queues, call forwarding, ring groups, callback, DISA, and time conditions. |
| 06:36.16 | Penguin | For the most part I just shove everything off their system into my own asterisk box on premise, but if you don't want to manage your own hardware, they provide a lot of features. |
| 06:36.54 | kevcox | They seem fairly large but their website seems a little simplistic...I am interested and plan to call and discuss their services when they are open again. |
| 06:37.07 | kevcox | I appreciate you sharing. |
| 06:37.21 | Penguin | I've used voip.ms for many years. |
| 06:39.20 | Penguin | One thing I like to do with their web-based features is set up failover to a time condition with digital receptionist, so that if my asterisk becomes unavailable for some reason, the attendant will answer on their side before forwarding the calls to an agent's mobile phone. |
| 06:39.44 | kevcox | Nice |
| 06:40.18 | Penguin | It works good for a small office, anyway. |
| 06:44.56 | Penguin | I've got to go for the night. I've got to be up again in 5.25 hours. |
| 06:45.13 | kevcox | Goodnight |
| 06:45.22 | Penguin | Good luck with your project. |
| 06:45.28 | kevcox | Thank you |
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| 07:25.23 | Stefan27 | Hi, is there any web-page where you update the release date for next 12.x version ? |
| 07:26.16 | ChannelZ | asterisk.org |
| 07:26.50 | ChannelZ | http://www.asterisk.org/downloads/asterisk/all-asterisk-versions to be more specific |
| 07:27.33 | ChannelZ | But dates for upcomings aren't usually posted |
| 07:27.39 | Stefan27 | aha, 6-8 weeks on average |
| 07:27.43 | Stefan27 | thx |
| 07:28.21 | ChannelZ | You can get a little bit of a clue sometimes looking at the test release versions, when something hits an RC |
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| 12:24.40 | matthew-moretalk | Anyone ever heard of a situation where certain types of sip packets are not delivered by your broadband provider? |
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| 12:26.39 | [TK]D-Fender | mattlike? |
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| 12:28.04 | matthew-moretalk | well we have a situation right now where subscribe packets from devices on a certain broadband connection are not reaching our proxy |
| 12:28.22 | matthew-moretalk | I can see on wireshark the devices Handsets/Softphones are sending the subscribe packets but I can see using tcpdump that they never arrive at our proxy |
| 12:28.55 | file | I've heard of cellular providers doing funky stuff to try to tamper with VoIP |
| 12:29.20 | matthew-moretalk | I took one of the handsets home last night and as soon as I plugged it in I could see the subscribe packets arriving at our proxy |
| 12:29.43 | matthew-moretalk | SIP ALG is disabled on the router any other thoughts? |
| 12:31.04 | matthew-moretalk | Register and Invite packets are fine but PUBLISH/SUBSCRIBE is not. Very strange |
| 12:31.34 | matthew-moretalk | and it works fine on TLS too I guess because they cant mangle the packets |
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| 12:44.32 | file | nods |
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| 13:02.18 | asteriskbizua | hi |
| 13:02.26 | asteriskbizua | hi |
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| 13:03.03 | matthew-moretalk | hi |
| 13:03.43 | asteriskbizua | friendswhat does it mean Dial(DONGLE/KS-6-IN-OUTALL/${EXTEN},,S) i am about S without any parameter |
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| 13:11.17 | [TK]D-Fender | asteriskbizua: It means you are using that option improperly |
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| 14:01.56 | stevenm | is it possible to have encountered a cisco phone that has tftp disabled - and thus can't use that method to change admin password so you can do a factory reset? |
| 14:01.56 | stevenm | and if so - how else (if anyone knows) might a factory reset be done? |
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| 14:03.11 | [TK]D-Fender | depends on the model |
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| 14:03.30 | stevenm | [TK]D-Fender, SPA504G |
| 14:03.31 | [TK]D-Fender | What about the menu option on-scrren that is for this? |
| 14:03.31 | stevenm | asks for an admin password which the customer doesn't have - it's their old phones |
| 14:03.34 | [TK]D-Fender | and you can't enable provisioning on them? |
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| 14:05.09 | stevenm | [TK]D-Fender, they attempt to get anything via tftp - despite dhcp dishing it out just fine |
| 14:05.16 | stevenm | they *don't* attempt |
| 14:05.19 | stevenm | i mean ) |
| 14:05.21 | stevenm | :) |
| 14:06.47 | [TK]D-Fender | http://linux.ioerror.us/2013/05/factory-reset-cisco-spa504g-without-admin-password/ |
| 14:07.11 | stevenm | no tftp, no web interface |
| 14:07.24 | [TK]D-Fender | :/ |
| 14:07.41 | stevenm | and when it calls out to its provisioning server its https and it has a certificate installed lol |
| 14:07.54 | stevenm | (i.e. the original phone system providers provisioning server) |
| 14:08.35 | stevenm | i have notices on Cisco SPA phones though... at the back theres a little hole (with a padlock symbol printed next to it) where you can see through to the back of the circuit board and it has two little gold contact (like a jumper is meant to be there) |
| 14:09.10 | stevenm | I'm wondering if this is some kind of hardware factory reset method but the cisco SPA user guides make no mention of it (despite having a hi-res picture of the back of the phone that shows it) |
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| 14:13.05 | rrittgarn | Stevenm i think that's just the kensington lock slot |
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| 14:15.32 | crised | How can I achieve this? Having a x86 hardware running x86 I could add asterisk software. What I need is to have only 1 SIP phone in the network, that "rings" when a plain old phone number is dialed, and when I dial from this sip phone, dial from the same number. Could you guive me guidelines? Use case: Have a number abroad in an office to receive and make calls. |
| 14:16.58 | [TK]D-Fender | crised: You have not specified what your call is arriving on, or expected to go out. |
| 14:17.08 | [TK]D-Fender | crised: You can't dial out "a number" a number is not a "thing" |
| 14:17.29 | crised | [TK]D-Fender: I imagine that I would need to get a VoIP provider |
| 14:17.45 | [TK]D-Fender | crised: that is an OPTION> What you want is another matter. |
| 14:18.04 | crised | [TK]D-Fender: What would be another options? |
| 14:18.19 | [TK]D-Fender | crised: You'll need to make up your mind on the solution. As for how... you need to lear your asterisk dialplan basics, which is all in the book. |
| 14:18.21 | [TK]D-Fender | ~book |
| 14:18.21 | 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 |
| 14:19.02 | [TK]D-Fender | crised: Physical telco lines (several different kinds). GSM interfaces. |
| 14:19.20 | crised | [TK]D-Fender: yes I can imagine what are you talking about. |
| 14:19.34 | crised | ~buybook |
| 14:19.34 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
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| 14:23.54 | crised | [TK]D-Fender: Is there a website to select voip to telco line providers? |
| 14:24.03 | crised | sort, compare them, etc |
| 14:24.39 | [TK]D-Fender | Probably a few sites with comparisons. Nothing we refer anyone to though |
| 14:24.47 | [TK]D-Fender | Where are you looking to get severive for? |
| 14:24.53 | [TK]D-Fender | service* |
| 14:24.57 | crised | Chile |
| 14:25.05 | crised | [TK]D-Fender: I just want to look at rates |
| 14:26.49 | [TK]D-Fender | ~itsplist-us |
| 14:26.49 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 14:26.57 | [TK]D-Fender | start looking at these |
| 14:27.03 | crised | [TK]D-Fender: thx |
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| 14:27.23 | [TK]D-Fender | your are won't have a lot of companies directly located there I'm sure.... go look around |
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| 14:29.29 | LemensTS | Hello guys. If I wanted to start another room on freenode can I just join it, then other people can join too? |
| 14:29.41 | LemensTS | I did it to #calixapi |
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| 14:31.19 | Qwell | LemensTS: https://freenode.net/using_the_network.shtml |
| 14:35.25 | LemensTS | Thanks that worked |
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| 14:41.19 | crised | Which ITSP does google uses? |
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| 14:44.33 | sekil | isn't google one? |
| 14:45.49 | [TK]D-Fender | crised: They don't |
| 14:45.58 | [TK]D-Fender | sekil: they are. |
| 14:47.04 | crised | [TK]D-Fender: ok. here is an easy one for you. Is it possible to use Google Voice with just a SIP phone? and have a fixed number to receive calls? Google voice rates are low |
| 14:47.33 | [TK]D-Fender | crised: Google Voice is in the process of dying and being replaced by Google Hangout. FORGET about them |
| 14:47.51 | [TK]D-Fender | Hangouts does NOT have an API for 3rd party support for things like Asterisk |
| 14:48.22 | crised | [TK]D-Fender: will hangout support old telephones as well? |
| 14:48.39 | [TK]D-Fender | crised: So far no. |
| 14:48.53 | crised | [TK]D-Fender: they I don't think it will die! |
| 14:48.56 | crised | ;) |
| 14:49.22 | stevenm | rrittgarn, if its a kensington lock (I admit there is a K on the padlock) then why have the gold contacts visible through it? |
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| 14:50.50 | crised | [TK]D-Fender: "Future versions of Hangouts will integrate Google Voice more seamlessly." |
| 14:51.13 | [TK]D-Fender | ~gv |
| 14:51.14 | infobot | . URL: http://wino.physik.uni-mainz.de/~plass/gv/ |
| 14:51.22 | drmessano | Google Voice is being integrated into Hangouts |
| 14:51.37 | drmessano | Hangouts has no API to touch |
| 14:51.47 | [TK]D-Fender | hmm...https://productforums.google.com/forum/#!topic/voice/CdojHZ5_WBU |
| 14:51.57 | [TK]D-Fender | XMPP = BAI BAI |
| 14:52.03 | [TK]D-Fender | crised: FORGET IT |
| 14:52.15 | crised | ok |
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| 14:56.16 | gciotta | Hi, what would be the recommended way to load balance users between a couple of Asterisk PBXs? |
| 14:56.58 | gciotta | I'm looking into Kamailio and OpenSIPS but I'm not sure whether there's something simpler to look at |
| 15:00.12 | crised | [TK]D-Fender: voip.ms they have cheap rates |
| 15:00.55 | drmessano | Flowroute is better |
| 15:01.25 | drmessano | I freaking love Flowroute |
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| 15:15.18 | jeev | do most toll providers terminate calls on inbound when the call exceeds 20 to 30 seconds? |
| 15:15.24 | jeev | my current provider is terminating the call if it goes unanswered |
| 15:15.53 | Qwell | jeev: generally somewhere around 4 ring cycles (6 seconds per cycle) |
| 15:16.18 | jeev | that's becoming an issue for me, the marketing people are calling and saying the phone # doesn't work after hours, they dont have a voicemail. |
| 15:16.25 | Qwell | So ~24-30s isn't really uncommon |
| 15:16.28 | jeev | it goes into a hunt group that rings for around 24 seconds. |
| 15:16.36 | Qwell | so answer first |
| 15:16.38 | jeev | my guy says carrier keeps fixing it then it goes undone. |
| 15:16.57 | jeev | answer the ring group first? |
| 15:17.04 | Qwell | answer the incoming call |
| 15:17.05 | [TK]D-Fender | answer the CALL |
| 15:18.31 | jeev | since it's going into a ring group and obviously i'm using freepbx, i should be able to modify it to answer |
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| 15:26.56 | jeev | Qwell, any idea what that's called? some sort of fraud prevention? |
| 15:27.18 | pabelanger | what has become of me? |
| 15:27.24 | pabelanger | I am hacking freepbx dialplan code |
| 15:27.48 | jeev | well hack me a fix to my problem. |
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| 15:34.52 | josemslopes | i am trying to use chanspy to get the dtmf from a caller in queue ... but it is not working ... the DTMF is not ativating DYNAMIC_FEATURES enabled on channel with chanspy ... do i need to do more things to this work? |
| 15:37.47 | josemslopes | my question comes, from a sugestion to use dynamic features in queues ... that mjordan give a sugestion to use Local channels and chanspy |
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| 15:57.35 | [TK]D-Fender | josemslopes: where are you implementing another Dial to allow for it? |
| 15:57.54 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 15:59.40 | josemslopes | what i do is a originate between two local channels ... one local channel dial to a phone (so i can monitor, for now) ... another channel chanspy to the caller in the queue |
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| 16:00.14 | pjensen00 | mjordan: I notice in the python examples for ARI usage, they all begin with searching for an existing holding bridge. Do you only need one holding bridge and add as many people as you need? |
| 16:00.45 | [TK]D-Fender | josemslopes: Well before asking us why things don't work... you need to show us what you're doing. This means actual call debug, configs,e tc. |
| 16:02.35 | josemslopes | ok |
| 16:03.08 | josemslopes | [TK]D-Fender: where can i put that ? in here? |
| 16:03.19 | pjensen00 | pastebin |
| 16:03.25 | [TK]D-Fender | ^ |
| 16:03.27 | [TK]D-Fender | ~pb |
| 16:03.27 | 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. |
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| 16:09.22 | file | pjensen00, it depends but that is one way of doing things |
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| 16:36.18 | runfromnowhere | Hi all - I'm seeing a distinct difference in "Monitor" behavior between Asterisk 1.2 and Asterisk 11, namely that the "in" and "out" directions of the recording are coming out to substantially different file sizes under Asterisk 11. Does anyone know how to correct this behavior? |
| 16:36.59 | josemslopes | [TK]D-Fender: i put logs and configuration on the logs at http://pastebin.com/Tu49Lk8V |
| 16:41.58 | josemslopes | testjlopes i have updated to http://pastebin.com/KgFXFcjq ... i put some description about the scenario |
| 16:44.14 | [TK]D-Fender | josemslopes: wrong approach. |
| 16:44.30 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 16:44.42 | [TK]D-Fender | Your dial to the SIP device does not create a dial "brdige", it gets optimized away. |
| 16:45.43 | [TK]D-Fender | You need to call the device, then on answer put it into the dialplan where that will dial ANOTHER local channel (with /n at the end). This otehr local channel will call chanspy |
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| 16:54.19 | josemslopes | [TK]D-Fender: so i will use the originate to call to the SIP device and then call to a local context .. right? .. how do i put /n on the originate .. or do i not use originate? |
| 16:55.08 | [TK]D-Fender | originate call to device (either direct, or via a local channel). |
| 16:55.44 | josemslopes | i have done that before ... but don't work also ... |
| 16:55.49 | [TK]D-Fender | Then you have THAT device's channel dial ANOTHER local channel with "/n". THAT answering context will have ChanSpy in it |
| 16:58.23 | Penguin | So you're connecting a phone to the ChanSpy application? Seems pretty useless. |
| 16:58.44 | josemslopes | pegiun, for now is only for testing |
| 16:59.31 | josemslopes | but it is possible to use /n on the originate ... or there is another way to originate call ? |
| 16:59.44 | Penguin | What is the purpose of using a local channel to get to ChanSpy? You could simply execute the application in your originate. |
| 17:00.22 | Penguin | You can originate to a local channel using /n. Just add /n to the back end of the characters you type on your local channel. |
| 17:00.34 | josemslopes | the proporse is to monitor dtmf ... so i can provide dynamic features to caller's waiting on the queue |
| 17:00.42 | Penguin | Local/some_exten@a_context/n |
| 17:01.16 | josemslopes | i am using this AMI command : "Action: Originate\r\nChannel: SIP/tpt103\r\nContext: dynamic_features\r\nExten: s\r\nPriority: 1\r\nVariable: __SPYCHANNEL=$argv[1]\r\nTimeout: 30000\r\n\r\n" |
| 17:01.43 | josemslopes | i need to put /n after dynamic_features ? |
| 17:01.53 | Penguin | I don't see the local channel in that. |
| 17:02.34 | [TK]D-Fender | [12:59]josemslopesbut it is possible to use /n on the originate ... or there is another way to originate call ? <- you don't do it there |
| 17:02.37 | Penguin | [tk]d-fender did say to DIAL a local channel, right? |
| 17:02.44 | [TK]D-Fender | I did |
| 17:02.48 | Penguin | I thought so. |
| 17:03.38 | Penguin | I almost never use AMI, so I would originate it on the CLI using channel originate SIP/my_phone_123 application Dial Local/some_exten@a_context/n |
| 17:03.51 | Penguin | But the same thing can be done in AMI. |
| 17:03.59 | josemslopes | ok i understand |
| 17:05.21 | Penguin | When you use Context: Exten: and Priority:, that's the equivalent of using channel originate SIP/my_phone_123 extension blahBlah |
| 17:05.34 | Penguin | extension rather than application |
| 17:06.55 | Penguin | And when you use extension, the local channel gets optimized away, which is the opposite of dialing a local channel with /n on it. |
| 17:06.56 | josemslopes | i have tried ... still don't work |
| 17:07.28 | [TK]D-Fender | we aren't looking at your solution right now... |
| 17:07.30 | Penguin | Show me. |
| 17:07.39 | josemslopes | i need to go now... i will try latter ,,, thanks for the help |
| 17:09.12 | josemslopes | i put the information at http://pastebin.com/tXD7Ymvs |
| 17:15.55 | rrittgarn | So i have about 50 phones (Aastra 6737) on site at a customer site, brand new switches, and firewall coming over the internet to my PBX (public IP) and i'm continually getting peer $name is now lagged (2835ms/2000ms), then available shortly after It is only this one site, so i am not quite sure where to start troubleshooting... qualify=yes and qualifyfreq=30 on all the peers. |
| 17:17.01 | rrittgarn | anyone have any recommendations on next steps in troubleshooting? They are on a Level 3 10x10 fiber, my datacenter is on a 1GB L3 fiber, utilization looks ok (about 50%). Its starting to cause issues with taking calls though |
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| 17:24.32 | rrittgarn | for my own understanding, Qualify sends sip options messages to X peer @ the qualifyfreq to keep the port open when using NAT. Right? This is different from Keepalive (which i'm not sure if it is needed or should be configured with qualify?) |
| 17:26.45 | Penguin | The OPTIONS packets and their responses serve the purpose of keepalive, as well as checking to see if the device is still there. |
| 17:27.26 | Penguin | I use qualify for all phones and most other asterisks, even if NAT is not involved. |
| 17:27.38 | Penguin | It lets me know the other side still exists. |
| 17:27.42 | Penguin | (or doesn't exist) |
| 17:28.06 | WIMPy | Keepalives are empty packets that only keep connection tracking alive. And don't need any relevant amount of resources. |
| 17:29.00 | lvlinux | rrittgarn: have you actually tested the network to rule out that there is anything going on there? |
| 17:29.36 | Penguin | My thought on the high qualify times is that there's actual real latency on that circuit. |
| 17:29.45 | rrittgarn | bandwidth utilization doesn't look terrible, latency from them to me isn't bad (~30ms). We just put in the switching hardware (Juniper EX-2200 is their core switch). |
| 17:29.59 | rrittgarn | yeah i was leaning that way too Penguin, but don't know how to prove it or where to find its source if ICMP isn't showing the same |
| 17:30.20 | Penguin | Try mtr with udp datagrams rather than icmp. |
| 17:30.33 | lvlinux | yes |
| 17:30.58 | WIMPy | It could be an unresponsive application. |
| 17:32.14 | Penguin | If you don't see the latency using mtr and udp mode, it very well could be asterisk, as wimpy suggests. The next testing I might do would be to use iperf (again, udp mode). |
| 17:32.15 | mjordan | pjensen00: a holding bridge can hold a lot of participants. Each hears MoH, and their audio is not mixed (although a single Announcer channel will play to all). You can have multiple holding bridges as well however |
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| 17:33.41 | rrittgarn | asterisk is version 11.11.0 and is running a quad core with 4GB ram. server load is never > 1. Curious how to test if it is indeed asterisk (145 total devices registering at any given time on that server). |
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| 18:43.16 | crised | IAX2 or SIP phone? |
| 18:43.41 | crised | Are there any IAX phones? Or is IAX only for inter asterix comm? |
| 18:43.42 | [TK]D-Fender | Please try phrasing that as a complete sentence |
| 18:43.58 | [TK]D-Fender | Inter-Asterisk eXchange. Kinda says it all |
| 18:44.04 | crised | ok |
| 18:44.16 | [TK]D-Fender | There are a few phones. They pretty much suck |
| 18:44.23 | crised | [TK]D-Fender: ok |
| 18:44.41 | crised | [TK]D-Fender: so, it's better to route voice traffic with IAX2 and ITSP |
| 18:44.53 | crised | so that's an advantage of having astereix locally |
| 18:45.06 | crised | [TK]D-Fender: What about soft phones, which protocol do they mainly use? |
| 18:45.15 | [TK]D-Fender | SIP |
| 18:46.28 | crised | [TK]D-Fender: I see many voip resellers websites in my country, they all seem to suck, why would that be? rates are much lower |
| 18:47.39 | Penguin | It's asteriSK, not asterix. |
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| 18:49.08 | crised | Installing a SIP phone behind a regular router, which ports should open in the router in order to communicate with ITSP? |
| 18:50.22 | [TK]D-Fender | usually none |
| 18:50.31 | drmessano | Client shouldn't need any |
| 18:50.31 | [TK]D-Fender | Enpoints don't tend to need forwarding |
| 18:50.42 | drmessano | Think "Web Browser" instead of "Web Server" |
| 18:50.51 | drmessano | For lack of a better way |
| 18:50.54 | Penguin | Even asterisk as a client usually doesn't need any forwarding. |
| 18:51.03 | drmessano | ^ |
| 18:51.22 | drmessano | That's been my experience as well |
| 18:51.23 | crised | ok, but when an incoming call happens? the phone would be acting as a server |
| 18:51.36 | drmessano | No it wouldn't |
| 18:51.40 | Penguin | Not exactly. |
| 18:51.53 | crised | ? |
| 18:51.59 | crised | How does it work then? |
| 18:52.16 | webphage | Your phone will already have registered with server to receive calls? |
| 18:52.23 | crised | it should be listening at a given port, and the router should have external/internal port forwarding? |
| 18:52.24 | drmessano | It maintains a persistent connection |
| 18:52.24 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 18:52.35 | crised | UDP? |
| 18:52.48 | [TK]D-Fender | UDP keep-alive |
| 18:52.48 | Penguin | Some NAT implementations are weird and can't deal with it, but those are rare cases. |
| 18:53.03 | crised | [TK]D-Fender: UDP keep alive? |
| 18:53.14 | crised | I thought UDP was connectionless :( |
| 18:53.18 | drmessano | You don't need to forward ports to your phones |
| 18:53.19 | [TK]D-Fender | crised: You need to learn how NAT works |
| 18:53.26 | drmessano | You can google for more info lol |
| 18:53.43 | Penguin | I know, UDP is connectionless, but you'll still find UDP entries in the state table. |
| 18:54.20 | [TK]D-Fender | NAT routers have a a UDP TIMEOUT to keep a hole open for further comms. SIP devices tend to have keep-alive methods to ensure the phone is reachable |
| 18:54.38 | Penguin | hence statefulness |
| 18:54.43 | [TK]D-Fender | Or the SERVER could issue them. Either side can |
| 18:54.48 | crised | [TK]D-Fender: great, which is it that time? |
| 18:54.51 | [TK]D-Fender | State-ISH-ness ;) |
| 18:54.54 | crised | seconds? minutes? |
| 18:55.10 | [TK]D-Fender | crised: usually max 2 min IIRC |
| 18:55.30 | crised | [TK]D-Fender: so SIP sends a heartbeat message every <2 min? |
| 18:55.33 | Penguin | The default qualify frequency in asterisk is 60 seconds. |
| 18:56.05 | webphage | SIP question! What is expected behavior in meetme if client disappears without BYE? And then perhaps re-registers and joins call again? |
| 18:56.06 | [TK]D-Fender | crised: "SIP" doesn't send it .. the application does... |
| 18:56.29 | crised | [TK]D-Fender: damn interestnig |
| 18:56.46 | drmessano | You know what's even more interesting |
| 18:56.48 | crised | [TK]D-Fender: How is this heartbeat message called? |
| 18:56.55 | crised | drmessano: yes, I glanced the book |
| 18:56.58 | drmessano | I lost my coffee maker in the divorce.. and I need one with a SWITCH on it |
| 18:56.59 | [TK]D-Fender | crised: SIP is a protocol.. it up to the overall app to determine how it should act |
| 18:57.04 | drmessano | So it will work with my X10 |
| 18:57.09 | Penguin | In asterisk, we use qualify to send an OPTIONS packet to the peer every 60 seconds by default. The frequency can be tuned for weird NATs. |
| 18:57.10 | drmessano | and I am having a hard time locating one |
| 18:57.12 | file | webphage, if you aren't using an RTP timeout or session timers... then nothing, that channel will continue to be there |
| 18:57.30 | [TK]D-Fender | drmessano: So you can use the relay module? |
| 18:57.33 | drmessano | That to me is a CRISIS |
| 18:57.37 | drmessano | [TK]D-Fender, indeed |
| 18:57.53 | drmessano | I had timers and all |
| 18:57.55 | [TK]D-Fender | drmessano: Any real issue using a wall-jack or appliance module? |
| 18:58.08 | drmessano | [TK]D-Fender, not one issue. Worked great for years |
| 18:58.25 | [TK]D-Fender | drmessano: I mean instead of the relay trigger one. |
| 18:58.38 | webphage | @file: thanks. What is clients responsibility with respect to timers like that? |
| 18:58.39 | Penguin | But if you plug in the maker and it doesn't go to the ON state, it won't work. |
| 18:58.48 | drmessano | Ohh.. sorry, I misunderstood.. I am using the appliance module. the standard one |
| 18:58.49 | [TK]D-Fender | drmessano: or are you using it like a "start-only" solution, and letting the timeout on the machine deal with itself? |
| 18:58.59 | Penguin | The goofy electronic controlled ones default to OFF when powered up. |
| 18:59.10 | file | webphage, well RTP timeout is an Asterisk side thing as is session timers - if the other side disappears... then Asterisk terminates the channel |
| 18:59.10 | [TK]D-Fender | drmessano: Ah.... well el-cheapo coffee machines can be had for a pittance |
| 18:59.19 | drmessano | [TK]D-Fender, NOO!! I am shutting it off with my OWN timers so I don't burn coffee after I have my cuppa or two |
| 18:59.29 | drmessano | I love IT |
| 18:59.47 | [TK]D-Fender | drmessano: this should be like $20-30 at Walmart... |
| 19:00.03 | [TK]D-Fender | drmessano: I've got a big bag of X-10 gear I haven't touched in almost a decade |
| 19:00.09 | webphage | @file: but the client must answer the timer, no? Do these timers work if connection is UDP? |
| 19:00.11 | drmessano | [TK]D-Fender, just need to locate.. Mr Coffee has an $18 model, but out of stock at several |
| 19:00.17 | crised | [TK]D-Fender: let's assume that ITSP gave me a DID number in my country, when I do a local call, will the guy receiving the phone call see the same DID number? |
| 19:00.22 | file | webphage, the timer has to yes... and if it's still there it will |
| 19:00.29 | [TK]D-Fender | crised: Depends |
| 19:00.37 | crised | [TK]D-Fender: How is this called? |
| 19:01.00 | drmessano | [TK]D-Fender, if you decide to part with any of it, let me know first lol |
| 19:01.01 | [TK]D-Fender | crised: First... to them there is no such thing as a "local call". Seconds.. they can set the callerid to whatever they want.... and might let YOU do the same |
| 19:01.17 | Penguin | crised: An ITSP generally sends your calls to IP devices. The call can contain all sorts of data. |
| 19:01.36 | crised | ok, good stuff |
| 19:01.41 | [TK]D-Fender | drmessano: Shipping & duty wouldn't make it worth your while I'm sure. But hey, if you're ever up here, you tops on my "beer list" :) |
| 19:01.42 | drmessano | I X10 everything.. if they had a model that you plug a woman into, I would do that too |
| 19:01.43 | crised | I liked voip.ms |
| 19:01.45 | webphage | @file: ah okay, so need to find client that supports these timers... ;) is there generally a more "correct" timer? |
| 19:02.00 | file | webphage, session timers can be used even if the other side doesn't support it |
| 19:02.03 | Penguin | crised: I've been using them for years. |
| 19:02.18 | crised | Penguin: they look awesome |
| 19:02.34 | Penguin | Very few problems. |
| 19:02.52 | webphage | @file: I'm not sure I understand that, but I will look into session timers. Thank you for your help! |
| 19:02.53 | drmessano | How the hell did I not know there's a low-latency kernel for EC2 |
| 19:02.58 | drmessano | Gah.. |
| 19:03.07 | file | drmessano, you were blind but now you see? |
| 19:03.15 | drmessano | Yeah I guess so |
| 19:03.19 | Penguin | There's the occasional network outage or data center hiccup, but they ususally respond quickly. |
| 19:03.37 | drmessano | Found it on a forum post DONT USE ASTERISK OR FREESWITCH ON EC2 WITHOUT THIS |
| 19:03.41 | drmessano | FMLR |
| 19:05.09 | drmessano | Just wrote up a doc and threw it in my grimoire |
| 19:08.14 | crised | Is this sentence talking about the router? As an additional recommendation, Set "NAT enabled" to Yes, and on the "NAT address" parameter, enter the External (publicly addressable) IP address of Router or firewall, if any. |
| 19:08.21 | crised | http://wiki.voip.ms/article/Cisco_IP_Phone_7940/7960 |
| 19:08.35 | *** join/#asterisk zerick (~eocrospom@190.118.28.252) |
| 19:09.46 | Penguin | Whatever it says, forget it. |
| 19:09.59 | crised | Penguin: lolo k |
| 19:10.22 | Penguin | Don't enable NAT settings on the phone. Let asterisk do the work. |
| 19:10.43 | crised | Penguin: I wasn't planning to have asterisk |
| 19:10.49 | Penguin | erm... |
| 19:10.56 | Penguin | That's counterproductive. |
| 19:11.28 | crised | Penguin: Is it possible to have that SIP phone, configured directly to voip.ms? no asterisk? |
| 19:11.35 | Penguin | yes |
| 19:11.41 | crised | Penguin: this might be the first step of a business adopting VoIP |
| 19:11.58 | Penguin | They have good cloud-based PBX features, too. |
| 19:12.19 | Penguin | It's all stuff that one would typically configure in an on-premise asterisk system. |
| 19:12.36 | crised | Penguin: all the servers of voip.ms are at least 150 ms ping-away, is that too much latency? |
| 19:12.42 | crised | Penguin: great |
| 19:12.59 | Penguin | Latency isn't the big problem. Jitter is what kills voice calls. |
| 19:13.24 | crised | Penguin: HOw do I measure jitter? |
| 19:13.28 | Penguin | 150ms isn't that bad, really. |
| 19:13.38 | Penguin | You can measure jitter using mtr. |
| 19:14.23 | crised | Penguin: std deviation> |
| 19:14.24 | crised | ? |
| 19:14.39 | [TK]D-Fender | [15:11]crisedPenguin: this might be the first step of a business adopting VoIP <- only if their idea of a business phone... is the functional equivalent of a single dumb phone line. |
| 19:15.10 | crised | [TK]D-Fender: step by step |
| 19:15.37 | Penguin | [tk]d-fender: Nah, that's not true. You can register multiple phones directly to voip.ms and use the features they provide. They give you digital receptionist and call queues, for example. |
| 19:15.55 | [TK]D-Fender | crised: I would also specifically avoid Cisco 79XX series phones. More trouble than they're worth |
| 19:16.01 | crised | [TK]D-Fender: how to measure jitter? |
| 19:16.29 | [TK]D-Fender | Penguin: Well if they offer hosted PBX features... that's typically separate from the standard service |
| 19:16.32 | crised | [TK]D-Fender: Which SIP do you advice? |
| 19:16.34 | Penguin | The 7900 series phoens do work, but other phones would be better. |
| 19:16.42 | Penguin | [tk]d-fender: It's standard. |
| 19:17.04 | [TK]D-Fender | Penguin: a "bonus" from an ITSP POV. nifty that they do. What do they charge? |
| 19:17.16 | Penguin | It's not a hosted PBX solution. It's just standard features on their services. |
| 19:17.27 | crised | 11. 68-233-226-97.static.hvvc.us 0.0% 53 182.9 124.0 120.9 182.9 11.8 |
| 19:17.31 | crised | Penguin: is that good? |
| 19:17.38 | crised | [TK]D-Fender: Which SIP phone would you advice? |
| 19:17.44 | crised | better if sold in china :) |
| 19:17.48 | Penguin | Where's the jitter in that line? |
| 19:17.52 | crised | Penguin: last line |
| 19:17.58 | crised | StDEV |
| 19:18.04 | crised | last column |
| 19:18.07 | [TK]D-Fender | Penguin: queues, ring groups, ivr... sure says "PBX" to me... |
| 19:18.38 | Penguin | But it's all standard. There's no charge above your standard per minute rates. |
| 19:18.47 | crised | :-! |
| 19:19.38 | Penguin | That's what I'm trying to tell you. These features are some of the same things you'd do on your own on-premise asterisk box. They just shift it into their side of the cloud. |
| 19:20.11 | [TK]D-Fender | Penguin: It's just not BILLED separate ;) |
| 19:20.34 | Penguin | If you feel like you're paying for it, you'll pay for it even if you don't use it. |
| 19:20.44 | Penguin | There's one rate. Use the features or don't. The price is the same. |
| 19:20.52 | crised | :-/ |
| 19:21.35 | [TK]D-Fender | Penguin: Cool that they just give you the option if you want it.... |
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| 19:22.26 | crised | (15:15:27) [TK]D-Fender: crised: I would also specifically avoid Cisco 79XX series phones. More trouble than they're worth - damn I bought one of these, I spended 3 days trying to reflash them |
| 19:22.47 | crised | [TK]D-Fender: In my country, there are lot of those used. |
| 19:22.55 | crised | [TK]D-Fender: please advice me of SIP phones |
| 19:23.01 | crised | ~sip |
| 19:23.01 | infobot | well, sip is Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
| 19:23.11 | Penguin | crised: When you run mtr, you need to view the jitter. You can execute the program using the options for showing jitter, or you can select it after you start it up using the default fields. |
| 19:23.41 | crised | Penguin: I see it now |
| 19:23.50 | Penguin | The 7900 series works, but you'll hate them if you ever use some other type of phone. |
| 19:24.09 | crised | Penguin: why? |
| 19:24.16 | crised | Penguin: Javg 36.7 |
| 19:25.03 | Penguin | They just aren't very good SIP phones. They were built to use SCCP. |
| 19:25.20 | crised | Penguin: ok. Then could you suggest good SIP phones? |
| 19:25.34 | Penguin | Polycom makes good SIP phones. |
| 19:26.00 | crised | Penguin: $$$ |
| 19:26.11 | Penguin | Depends on what model you want. |
| 19:26.22 | crised | http://www.amazon.com/Polycom-SoundPoint-Phone-Supply-Included/dp/B0009VCH4W/ref=sr_1_10?ie=UTF8&qid=1409945148&sr=8-10&keywords=Polycom |
| 19:26.34 | Penguin | If you need five line keys and a nice screen, you'll spend up to $100 on a used one. |
| 19:26.47 | crised | Penguin: I need cheap now |
| 19:27.29 | Penguin | How cheap? What's your budget per phone? |
| 19:27.52 | crised | Penguin: I don't know, decent & cheap |
| 19:27.57 | crised | $50? |
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| 19:29.09 | Penguin | Polycom SoundPoint IP 335 |
| 19:29.46 | crised | Penguin: $120 |
| 19:29.56 | Penguin | $25 on ebay. |
| 19:30.35 | crised | Penguin: POE? |
| 19:30.55 | Penguin | They can be powered by power brick or PoE. |
| 19:31.12 | crised | Penguin: the dc input is standard? |
| 19:31.17 | crised | or is it something propietary? |
| 19:31.26 | Penguin | 24VDC I think |
| 19:31.33 | crised | but the jack? |
| 19:31.38 | crised | the male ... |
| 19:31.45 | Penguin | Regular DC power jack. I'm not sure the size. |
| 19:32.09 | Penguin | Like 5mm outside 1.5mm pin? |
| 19:32.28 | crised | ok |
| 19:32.45 | Penguin | It's a standard connector that you can get anywhere that sells that sort of part. |
| 19:33.05 | crised | Penguin: another brand besides polycom? |
| 19:33.23 | Penguin | That's really the best brand for the price. |
| 19:33.34 | Penguin | But you can get other SIP phones that still work good. |
| 19:33.45 | Penguin | Even the newer Cisco SIP phones aren't horrible. |
| 19:33.50 | crised | Penguin: what would be others? inferior |
| 19:34.15 | Penguin | Even the Linksys SPA 900 series aren't horrible. They are better than the Cisco 7900 series. |
| 19:35.09 | Penguin | Linksys SPA 942 |
| 19:35.25 | pjensen00 | I use the 942s |
| 19:35.31 | pjensen00 | They do well enough. |
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| 19:36.36 | crised | Penguin: good |
| 19:37.00 | crised | Penguin: these are really at <$40 |
| 19:37.19 | Penguin | Cisco 500 series |
| 19:39.27 | crised | thanks |
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| 19:55.06 | crised | Do ITSP providers charge on inbound call? through a DID number? |
| 19:56.01 | [TK]D-Fender | Do you normally expect free service? |
| 19:56.06 | [TK]D-Fender | SOMEBODY is paying for it./... |
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| 19:56.14 | crised | [TK]D-Fender: the caller? |
| 19:56.28 | crised | when I receive calls on my cell phone I don't pay |
| 19:56.35 | Penguin | Do you know what the P in ITSP means? |
| 19:56.47 | crised | provider |
| 19:56.56 | [TK]D-Fender | You get cell phone service for free? You never have to pay if you just want to recieve phone calls all day long forever? |
| 19:57.39 | crised | [TK]D-Fender: some guys do that in prepaid |
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| 19:57.55 | [TK]D-Fender | crised: .... prePAID |
| 19:58.09 | [TK]D-Fender | there is MONEY somewhere in there |
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| 20:00.26 | smellis_werk | TANSTAAFL |
| 20:01.05 | crised | DID numbers on the Per minute plan will cost from $0.99 to $1.99 per month with an incoming rate of $0.01 to $0.0149 per minute, the rate will depend on the Rate Center of the DID number. |
| 20:01.31 | crised | When on Flat Rate, they'll cost $4.95 to $6.95 per month with up to 3500 inbound minutes included. (Residental use). |
| 20:01.51 | crised | got to leave thanks |
| 20:01.59 | crised | [TK]D-Fender: thank you! |
| 20:02.10 | crised | Penguin: thanks |
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| 20:22.55 | coppice | [TK]D-Fender: caller pays is a strict rule in many places |
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| 21:08.28 | Geriatrix | hi all - i have a call that's coming in from a trunk - and once the call is established - asterisk executes h extension - as if that channel hung up ---- can someone assisnt me in sheding some light on this ? |
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| 21:44.32 | malachi_constant | Guys, I'm trying to connect a channel bank via MGCP and my configuration looks like its correct. If I nmap the channel bank with -PU, should the bank be listening on that port? |
| 21:55.42 | malachi_constant | (On 2727, which I guess it's supposed to.) |
| 21:59.16 | pjensen00 | I'm confused on the variables part of this https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-originate |
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| 21:59.50 | pjensen00 | I'm trying to figure out what the actual syntax I am supposed to put on the.... |
| 22:00.03 | pjensen00 | oh wait. Hold up. Nevermind. |
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| 22:37.13 | malachi_constant | Has anyone in here used an ADIT 600? I get the following in debug with regard to my gateway: [Sep 5 18:35:06] NOTICE[6811]: chan_mgcp.c:1853 find_subchannel_and_lock: Gateway '192.168.1.192' (and thus its endpoint '*') does not exist |
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| 22:58.11 | malachi_constant | tcpdump has things like this in it: 18:56:30.717687 IP 192.168.1.192.mgcp-callagent > 192.168.1.200.mgcp-gateway: UDP, length 45 |
| 22:59.01 | malachi_constant | So this suggests to me that asterisk is getting the packets but doesn't know what I want it to do with it. |
| 23:08.37 | malachi_constant | the 'mgcp audit endpoint' command doesn't seem to send anything to the channel bank. |
| 23:37.22 | mjordan | malachi_constant: there aren't a lot of active users of chan_mgcp, and probably even fewer in #asterisk on a Friday evening (North America, anyway). You may want to e-mail your questions to asterisk-users |
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