| 00:09.40 | *** join/#asterisk menstraighting (~menstraig@nc-76-5-178-34.dhcp.embarqhsd.net) |
| 00:14.45 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
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| 00:44.02 | *** join/#asterisk chare (322f51ac@gateway/web/freenode/ip.50.47.81.172) |
| 00:44.15 | chare | whats the current best way to get free voip |
| 00:49.55 | chare | anyone here? |
| 00:54.57 | gusto | he? |
| 00:55.10 | gusto | free voip? how do you mean "free"? |
| 00:56.00 | [TK]D-Fender | chare: Guess what.. same anwer as last week. And the week before. And the week to come, and probably the next several dozen..... |
| 00:56.20 | chare | [TK]D-Fender eventually one week the answer will change |
| 00:56.29 | chare | time moves forward |
| 00:56.39 | chare | enough one weeks add up to a long time |
| 00:57.07 | [TK]D-Fender | Yes time moves forward.... let me know ehn I get my flying cars, robot maid, and fully-nutritious meal-in-a-pills |
| 00:58.08 | [TK]D-Fender | So assuming your one desire actually gets realized in any practical time-frame is foolhardy at best |
| 00:59.38 | chare | [TK]D-Fender see you're always so pessimistic |
| 00:59.46 | chare | thats why you don't create the future |
| 01:00.46 | [TK]D-Fender | We all do. |
| 01:01.27 | chare | if you believe something is impossible you self rationalize its impossible |
| 01:01.39 | markmcn | Chare so long as telco's charge for an interconnect between sip and physical lines there will always be a cost. But people have this impression sip is free. SIP providers have costs (Servers bandwidth etc) so they would have to charge money even if teloco's didn't charge for interconnect it's like saying when will the internet be free |
| 01:02.06 | gusto | so long as telco's charge for an interconnect between sip and physical lines there will always be a cost. |
| 01:02.09 | gusto | yes |
| 01:02.14 | [TK]D-Fender | And I have over a decade of actual practical experience with Asterisk and real-world VoIP usages, and at traditional telcom background going back further still |
| 01:02.22 | chare | markmcn you gotta pay for internet, so why pay for both internet and phone when voip is free or WAY CHEAPER |
| 01:02.45 | markmcn | you said free, And you pay because it costs the sip provider |
| 01:03.05 | chare | markmcn ok but at least its way cheaper than paying for normal phone |
| 01:03.07 | markmcn | if you want free setup a dedicated asterisk box and become the first free sip provider |
| 01:03.25 | markmcn | agreed but you asked for free as in a free meal no such thing exists |
| 01:03.32 | [TK]D-Fender | [20:01]chareif you believe something is impossible you self rationalize its impossible <- IMPROBABLE. Not impossible. |
| 01:03.48 | gusto | that's why the most modern ISPs put VoIP lines instead of analogue |
| 01:03.55 | [TK]D-Fender | Thinking how business works is going to change week to week shows a serious detachment to reality |
| 01:03.57 | chare | markmcn: how much do sip provider pay the phone companies to route calls through them |
| 01:04.10 | [TK]D-Fender | There isn't a business gain to be made. Expect movement to be PAINFULLY slow |
| 01:04.50 | chare | [TK]D-Fender you gotta stop thinking purely about money |
| 01:05.34 | markmcn | in ireland it can't be cheap as the telo's own the lines and if a sip provider wants to be able to dial the telco's know they have the advantage, however a sip provider would be well within it's rights to charge for sip to sip as this still uses resources. ok you don't see it as resource use as it's going over an internet connection you are paying for already but they have to run servers etc |
| 01:05.54 | [TK]D-Fender | Money is what PAYS for service |
| 01:06.02 | [TK]D-Fender | Doesn't fall out of thin air |
| 01:06.06 | markmcn | chare: so long as there are costs to running a service some one ahs to pay for it |
| 01:06.20 | markmcn | sip is free as in open |
| 01:06.21 | [TK]D-Fender | SOMEONE is paying for you to get something for "Free" and there is no return on investment for them to do this really |
| 01:06.30 | markmcn | not free as it no resources needed |
| 01:07.19 | [TK]D-Fender | [20:04]chare[TK]D-Fender you gotta stop thinking purely about money <- You keep coming in here looking for FREE. Seems like you are VERY pre-occupied about money |
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| 01:08.03 | markmcn | I think chare needs to make the free future happen by running a server and sip server. I'll subscribe :) |
| 01:08.15 | [TK]D-Fender | Sounds fair |
| 01:08.35 | markmcn | great so where and when can we sign up. Talking clock anyone !!!! |
| 01:08.38 | [TK]D-Fender | chare Be the future. Provide US free service to the PSTN. Unlimited channels & usage please. |
| 01:08.42 | chare | markmcn: it doesn't have to be free just lower in price |
| 01:09.10 | markmcn | then say what you mean you asked for free and by the sounds of things have been asking the same question for a while |
| 01:09.16 | [TK]D-Fender | chare: We've already seen that what's out there isn't good enough for you. Too bad. The market doesn't move fast enough to warrant this weekly discussion. |
| 01:09.52 | chare | markmcn and [TK]D-Fender so you guys can force the market to move faster by starting your own SIP |
| 01:10.04 | [TK]D-Fender | charWrong. |
| 01:10.16 | markmcn | chare you have the vision of where you want it |
| 01:10.22 | [TK]D-Fender | There aren't enough Asterisk users in one country to make that move COLLECTIVELY |
| 01:10.29 | markmcn | far be it from us to steal your world changing idea |
| 01:10.49 | chare | markmcn its not my idea its OURS |
| 01:10.54 | chare | so don't be afraid to start the SIP |
| 01:10.57 | markmcn | TK at best that becomes a co-op model |
| 01:11.02 | [TK]D-Fender | No, it's just you. |
| 01:11.10 | [TK]D-Fender | We do not share it |
| 01:11.15 | markmcn | agreed sip is doing the bits i need |
| 01:11.34 | gusto | whatt change? did i miss something? |
| 01:11.52 | markmcn | chare: the solution to your problem is this https://www.google.com/search?q=low+cost+sip+trunk+providers&oq=low+cost+sip&aqs=chrome.3.69i57j0l3.4836j0j1&sourceid=chrome&ie=UTF-8 |
| 01:12.11 | [TK]D-Fender | gusto: He keeps expecting free services to pop up. Every week the same thing. |
| 01:12.23 | [TK]D-Fender | markmcn: He's too cheap for EVERYTHING out there |
| 01:12.52 | markmcn | well as i've suggested i'd support his free service by taking a free sub |
| 01:13.14 | chare | markmcn what voip do you use? |
| 01:13.27 | markmcn | a national telco provide it |
| 01:13.31 | markmcn | works for me |
| 01:13.34 | markmcn | no linerental |
| 01:13.37 | markmcn | just pay for calls |
| 01:13.40 | markmcn | fair |
| 01:14.03 | markmcn | no line rental as there is no lines. they price the call rates at a point that must cover their running costs |
| 01:14.14 | markmcn | still cheaper then pots |
| 01:14.36 | markmcn | or a fractional line |
| 01:14.57 | chare | usa doesn't have governement owned telco |
| 01:15.00 | chare | its all monopoly here |
| 01:15.13 | markmcn | there isn't a gov run telco here either |
| 01:15.18 | markmcn | it's all private industry now |
| 01:15.19 | [TK]D-Fender | chare: nonsense, we've linked you multiple providers |
| 01:16.00 | markmcn | there is more then one telco in usa therefore not a monopoly please check the meaning of that word. You have a few big players sure |
| 01:16.02 | gusto | well |
| 01:16.10 | [TK]D-Fender | [20:14]chareits all monopoly here <- there are HUNDREDS of ITSP's you could be using. |
| 01:16.48 | gusto | private monopoly or a government monopoly, thats just choosing among two bad things |
| 01:16.49 | chare | which are forced to use the telco's lines |
| 01:16.53 | markmcn | *feels conversation is going round in a circle* could result in a routing loop better be carful |
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| 01:17.14 | markmcn | hence the sip providers charge and here we are back at the start |
| 01:17.23 | [TK]D-Fender | chare: You're asking about ITSP's. Lose your double-standards and stop wasting our time |
| 01:17.51 | chare | [TK]D-Fender ok so what voip do you use |
| 01:18.07 | [TK]D-Fender | les.net |
| 01:18.23 | [TK]D-Fender | And stop calling it "VoIP" |
| 01:18.29 | [TK]D-Fender | VoIP is a CONCEPT |
| 01:18.33 | [TK]D-Fender | and ITSP offers SERVICE |
| 01:18.44 | [TK]D-Fender | get your terminology right at the very least |
| 01:19.39 | chare | why did you choose les.net VoIP |
| 01:20.06 | [TK]D-Fender | because they offered the kinds of services that I wanted at a decent price and with decent quality |
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| 01:20.49 | chare | what do you think of http://www.vitelity.com |
| 01:20.55 | [TK]D-Fender | They're fine |
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| 01:21.04 | chare | overpriced or fine? |
| 01:21.04 | [TK]D-Fender | and were recommended to you by several people |
| 01:21.39 | markmcn | Unlimited phone service starting at $7.95/month is bloody good |
| 01:22.01 | chare | TOO EXPENSIVE |
| 01:22.23 | [TK]D-Fender | Go find cheaper |
| 01:22.41 | markmcn | well when I can sign up to CHARE.net sip services we'll have a cheaper alternative |
| 01:23.38 | [TK]D-Fender | It should be "chare.org" for that "non-profit" feel |
| 01:23.43 | markmcn | chare you don't seem to grasp reality of business costs at all. maybe there are better alteratives out there for you really rather then fixation on SIP pricing |
| 01:23.51 | markmcn | sorry TK your right |
| 01:23.52 | chare | suppose i go with les.net how do I move my phone number to the ITSP |
| 01:24.01 | markmcn | ask them |
| 01:24.12 | markmcn | they will have a migration process i'd guess |
| 01:24.19 | markmcn | go waste their time |
| 01:24.41 | markmcn | hell it will only push up their costs answer questions of someone who won't pay for the service |
| 01:24.55 | markmcn | any one feel chare is trolling at this stage |
| 01:25.16 | chare | I'm not a troll |
| 01:25.21 | [TK]D-Fender | markmcn: No, that was pretty much from when he first started coming here. It's been several weeks now... |
| 01:25.22 | chare | just very very frugal |
| 01:25.34 | [TK]D-Fender | Cheap whiner |
| 01:25.52 | [TK]D-Fender | And lazy |
| 01:25.56 | markmcn | chare what's stopping you setting up this free sip service? |
| 01:26.04 | markmcn | please tell me |
| 01:26.08 | ChannelZ | it costs money to do! |
| 01:26.30 | markmcn | and if he's not willing to pay anyone to do it why should anyone do it for chare |
| 01:26.33 | chare | oh great now ChannelZ is awake |
| 01:26.52 | chare | I'm thinknig if we all setup a wireless cjdns distributed network |
| 01:27.00 | ChannelZ | You start |
| 01:27.01 | chare | then we can setup our own network on which to do voip |
| 01:27.12 | markmcn | *SIGH* |
| 01:27.23 | chare | WHAT? |
| 01:27.27 | [TK]D-Fender | chare: Nowhere in that picture does this lead to the PSTN |
| 01:27.27 | markmcn | and how do you plan on interconnecting these networks |
| 01:27.41 | ChannelZ | for free! |
| 01:27.48 | chare | we don't, we are starting a revolution |
| 01:28.10 | ChannelZ | Solar powered. |
| 01:28.19 | markmcn | great |
| 01:28.26 | [TK]D-Fender | chare: there is no "we" |
| 01:28.28 | markmcn | only call during day time |
| 01:28.45 | markmcn | batteries are bad for the environment |
| 01:28.59 | markmcn | who pays for the servers |
| 01:29.04 | [TK]D-Fender | chare: Again, it's just you. Our time is worth more than the spare change it costs us to go on with our lives |
| 01:29.04 | chare | ChannelZ YES SOLAR POWERED |
| 01:29.28 | markmcn | chare thank you |
| 01:30.30 | markmcn | you have provided me with a laugh at the fact you really don't understand business. It's late where I am and i'm now tired Good night all. TK many thanks for the help earlier in the week got the agent's issue sorted |
| 01:30.39 | ChannelZ | So the revolution is for a bunch of people (other than you) to make or buy a bunch of equipment to make some sort of shadow telephone network that can't actually call anyone interesting, all so you can save a few bucks a year making phone calls (and I can only imagine to whom you talk to..) |
| 01:31.05 | ChannelZ | Sounds like a solid plan. |
| 01:31.08 | [TK]D-Fender | markmcn: You're welcome |
| 01:31.13 | markmcn | NN all |
| 01:31.18 | chare | AND STOP NSA FROM LISTENING TO YOUR CALLS |
| 01:31.26 | ChannelZ | yahuh. |
| 01:31.35 | markmcn | tinfoil hats!!!!! |
| 01:31.59 | [TK]D-Fender | chare: They already have back-doors into all the telecom equipement used |
| 01:32.10 | chare | thats why we are starting our own network |
| 01:32.23 | [TK]D-Fender | chare: So now you're talking about converting the entire plant to keep the NSA out. |
| 01:32.30 | [TK]D-Fender | You, FULL Troll detected |
| 01:32.33 | [TK]D-Fender | yup* |
| 01:33.12 | chare | so you work for the NSA? |
| 01:33.23 | ChannelZ | Be fair. Trolls know what they're doing. |
| 01:33.38 | [TK]D-Fender | ChannelZ: They don't have to know the specifics :) |
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| 01:36.58 | [TK]D-Fender | blah blah blah free. blah blah blah revolution. blah blah blah "we" (everyone except me). blah blah blah create the future. |
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| 01:43.59 | [TK]D-Fender | http://www.magicjack.com/plans.html |
| 01:44.05 | [TK]D-Fender | http://magicjack.com/faq/ |
| 01:44.16 | [TK]D-Fender | There, the lowest of the low. |
| 01:44.24 | [TK]D-Fender | And not something Asterisk can readly use. |
| 01:44.31 | [TK]D-Fender | with no mention of that 911 cost |
| 01:44.40 | [TK]D-Fender | Which we've already seen how litte you whine about |
| 01:44.47 | [TK]D-Fender | (in terms of cost) |
| 01:44.57 | [TK]D-Fender | Go get this |
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| 04:22.15 | lowkie | About a week ago someone was helping me with the D Channel cycling. I finally found the solution. |
| 04:22.46 | lowkie | I moved the card from a |
| 04:23.38 | lowkie | PCI Express x1 slot to an x16. This fixed the timing issues. Any thanks for any assist I received in here. |
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| 08:17.57 | tparcina | If AMI command requires privilege "DBGet system,reportin Get DB Entry". Does it mean that user needs to have both of those privilages, or he can have either of those two? |
| 08:19.01 | WIMPy | Should be easy to find out. |
| 08:20.52 | WIMPy | My configuration user doesn't have reporting. |
| 08:23.48 | tparcina | WIMPy: GetConfig doens't mention reporting rights. :) GetConfig system,config,all |
| 08:25.12 | WIMPy | It only uses BDGet and DBPut. |
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| 08:26.27 | frooker | hi |
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| 08:26.49 | frooker | anyone here =) |
| 08:27.01 | WIMPy | ~ask |
| 08:27.01 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 08:40.13 | frooker | i have a problem |
| 08:40.33 | frooker | i wrote my problem in http://www.freepbx.org/forum/tips-and-tricks/about-ring-answer |
| 08:40.41 | frooker | can anyone help me ? |
| 08:40.52 | wdoekes | ~freepbx |
| 08:40.52 | 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 |
| 08:40.54 | wdoekes | ~ask |
| 08:40.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:42.24 | frooker | my question is about asterisk |
| 08:42.33 | frooker | not about freepbx |
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| 08:48.14 | wdoekes | well, I can't parse this "I am calling my customers with "Ring Answer" header and when my customer open my call my phone is calling and customer waiting me." |
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| 08:50.05 | wdoekes | I'm guessing your problem is that the auto-answer feature only works in case A and not case B, but that's as far as I understand your post |
| 08:50.32 | WIMPy | isn't sure what the issue is, either. |
| 08:51.01 | WIMPy | Try to rephrase qour question/description. |
| 08:51.12 | frooker | i have a call center |
| 08:51.30 | WIMPy | wonders how to hit q instead of y |
| 08:51.34 | frooker | and my agents calling customers |
| 08:52.08 | frooker | and i am using a crm and my calls starting with asterisk agi php |
| 08:52.42 | frooker | i am using ami for this |
| 08:53.37 | frooker | i need help about when my agent click button asterisk call customer and when customer open my call after my agents phone start the call |
| 08:53.48 | frooker | i heard about that a header |
| 08:53.57 | frooker | and i add it to my call header |
| 08:54.30 | WIMPy | What is opening a call? |
| 08:54.51 | WIMPy | When the customer answers your call? |
| 08:54.58 | frooker | eah |
| 08:55.00 | frooker | yes |
| 08:55.30 | WIMPy | ok |
| 08:55.40 | frooker | i am sorry for my english |
| 08:56.42 | frooker | when i use transfer context its connect my call to any agent |
| 08:56.42 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:56.45 | frooker | but no cdr record |
| 08:56.54 | frooker | but i need statistics about this calls |
| 08:57.01 | frooker | i need cdr and transfer this call |
| 08:58.12 | WIMPy | Ah, so it's only about the CDRs, not hte call itself? |
| 08:58.24 | WIMPy | s/hte/the/ |
| 08:59.32 | frooker | i want to create a context and when customer answer my call i transfer this call an agent with cdr record |
| 09:02.15 | frooker | it work only with transfer in my context list but when i use transfer context for this section asterisk hasnt record any cdr |
| 09:03.26 | frooker | because i want to check if an agent calling a customer another agent can't do this. |
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| 10:28.47 | mirela666 | yawns |
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| 10:32.35 | *** join/#asterisk Geek-Linux (~Geek-Linu@182.188.200.189) |
| 10:34.24 | Geek-Linux | Hi All, i am configuring SIP ITSP with telco and getting the following in debug. Scheduling destruction of SIP dialog 'fg.Pz0xfGWQzQOzh0Jd0@test.com' in 32000 ms (Method: OPTIONS) |
| 10:34.24 | Geek-Linux | Really destroying SIP dialog '3sN2bw2vwFmswiMGtQLZ@test.com' Method: OPTIONS |
| 10:38.36 | Chainsaw | Geek-Linux: Need the raw data, not snippets chosen by you. |
| 10:38.37 | Chainsaw | ~pb |
| 10:38.38 | 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. |
| 10:42.41 | wdoekes | Geek-Linux: that's a pure debug message, it isn't indicative of a problem |
| 10:43.11 | wdoekes | if you have a problem, describe the symptoms |
| 10:44.29 | Geek-Linux | I have pasted the debug in pastebin with the following URL: http://pastebin.com/W373885S |
| 10:45.12 | Geek-Linux | The Problem is Telco is sending URI and asterisk is not accepting the calls over sip. |
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| 10:50.04 | Geek-Linux | wdoekes: can you help ? |
| 10:52.10 | *** join/#asterisk jhlavacek (~jirka@proformatique1-gw-std.alionis.net) |
| 10:52.40 | kaldemar | Geek-Linux: there is nothing about any calls in your pastebin. |
| 10:54.23 | Geek-Linux | yes, because calls are not comming and on SIP only debug shows following.. |
| 10:56.17 | *** join/#asterisk crax (~bergamot@5.46.26.40) |
| 10:56.27 | crax | hello |
| 10:56.45 | mirela666 | ehlo |
| 10:59.18 | kaldemar | Geek-Linux: then your asterisk is not getting any messages and you need to describe your setup. |
| 11:04.33 | *** join/#asterisk Geek-Linux (~Geek-Linu@182.188.200.189) |
| 11:12.55 | frooker | hi |
| 11:13.11 | frooker | how can i transfer a call with cdr to an agent |
| 11:13.22 | frooker | i tried transfer context but it hasnt cdr record |
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| 12:10.45 | Zogot | Ahoyhoy |
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| 12:24.34 | Geek-Linux | Hi all, i have pasted my problem in Pastebin: http://pastebin.com/y5c5n3ik kindly if any one can help .. ? |
| 12:24.56 | Geek-Linux | Is there any solution to this ? |
| 12:28.49 | mirela666 | Geek-Linux: error and warning messages are telling you everything you need to know |
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| 12:30.35 | *** join/#asterisk leetwanker (~leetwanke@c-50-143-113-109.hsd1.fl.comcast.net) |
| 12:30.40 | leetwanker | morning all |
| 12:31.33 | leetwanker | I'm using Asterisk-gui 2.0 and want to send *21 to my voip provider. Is there a way to do this from the gui? |
| 12:31.39 | *** part/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 12:32.48 | leetwanker | been trying to set it up in calling rules and it won't accept the the asterisk (pun intended) |
| 12:35.31 | leetwanker | Was thinking a custom macro under local destinations might work? |
| 12:36.00 | Geek-Linux | mirela666: what could be the solution in asterisk ? |
| 12:36.18 | mirela666 | <PROTECTED> |
| 12:36.52 | mirela666 | or To: <sip:993@<IP address> |
| 12:38.13 | mirela666 | <PROTECTED> |
| 12:39.00 | Geek-Linux | thank you, let me test and get back to you.. |
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| 13:07.54 | kippi | all |
| 13:08.02 | kippi | will meetme work with g722? |
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| 13:15.20 | matthew-moretalk | Hi Anyone got any ideas how to get asterisk to generate a PUBLISH when the voicemail box state changes (for use with kamailio presence) |
| 13:17.08 | mjordan | matthew-moretalk: Asterisk doesn't support PUBLISH requests to that extent. It has some support for such things, but that is really limited to just Call Completion. |
| 13:17.48 | matthew-moretalk | Hi mjordan thanks for your response. Any ideas how to mwi to work without registering peers to asterisk then? |
| 13:18.23 | matthew-moretalk | I have realtime peers setup which point to the kamailio proxy where the devices register to. |
| 13:18.46 | [TK]D-Fender | matthew-moretalk: * doesn't broadcast anything like that to non-rgistered and non-fixed peers |
| 13:19.37 | matthew-moretalk | Hi i have realtime peers setup pointing to our kamailio proxy. any ideas then how to get MWI Working? |
| 13:20.51 | matthew-moretalk | I dont NEED to use kamailio's Presence system if there is another way anyone can think of? |
| 13:22.04 | [TK]D-Fender | matthew-moretalk: How are these peers "pointing" there? |
| 13:25.42 | leetwanker | pbx.c:4715 pbx_extension_helper: No application 'Followme' for extension (macro-stdexten-followme, s, 6) |
| 13:26.05 | leetwanker | exten = s,6,Followme(${ORIG_ARG1},${FOLLOWMEOPTIONS}) |
| 13:26.19 | leetwanker | anyone help me figure out what this is looking for that it's not finding? |
| 13:26.24 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:26.53 | leetwanker | FOLLOWMEOPTIONS = |
| 13:27.05 | matthew-moretalk | The host address of the peer is set to the proxys hostname |
| 13:27.29 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:28.24 | [TK]D-Fender | matthew-moretalk: then you should be able to just specify the mailbox like any other peer and let * throw it's NOTIFY's like normal |
| 13:28.30 | [TK]D-Fender | its* |
| 13:30.33 | leetwanker | oops, think I'm missing followme.so. |
| 13:31.31 | matthew-moretalk | Thats what I thought but looking at wireshark the Notify Packets asterisk is generating dont have the Username in the To: address. for example it just says To : <sip:sip.domain.com> rather than <sip:100@sip.domain.com> like an invite packet |
| 13:31.51 | matthew-moretalk | so kamailio responds back with Incomplete address |
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| 13:33.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:34.17 | [TK]D-Fender | matthew-moretalk: then set up a DEVICESTATE hint and create a script to poll the mailbox and flip it as needed for the presence approach |
| 13:35.10 | matthew-moretalk | okay thanks. Any ideas where I would start with that? |
| 13:35.39 | mjordan | matthew-moretalk: do you have fromuser or mwi_from set on the peer? |
| 13:36.05 | [TK]D-Fender | mjordan: He said the "To:" was the issue... |
| 13:36.31 | mjordan | rereads |
| 13:36.34 | mjordan | doh |
| 13:36.54 | matthew-moretalk | i have fromuser set to the same as the defaultuser and name. didnt even know mwi)from existed! |
| 13:37.54 | mjordan | matthew-moretalk: is Kamailio subscribing for the NOTIFY requests? |
| 13:38.27 | matthew-moretalk | come to think about it I guess I could fudge it in kamailio's dialplan to get the username from the From header and prepend it to the TO address. |
| 13:39.05 | leetwanker | I put app_followme.so from another asterisk installation into my modules folder and but asterisk isn't reading it. Says File Not Found. |
| 13:39.37 | matthew-moretalk | and if I use kamailio presence then kamailio Expects PUBLISH Rather than Notify. My prefered option though is to just get kamailio to route the subscribe/notify packets between the device and asterisk |
| 13:40.43 | [TK]D-Fender | leetwanker: And when you try to load it manually? |
| 13:41.17 | mjordan | matthew-moretalk: without something telling Asterisk what entity is subscribing for the notifications, it doesn't know who exactly wants it. So it sends it to the configured host |
| 13:41.56 | leetwanker | how do you load a module manually? |
| 13:42.07 | leetwanker | (sorry, imma newbie) |
| 13:42.15 | [TK]D-Fender | module load app_followme.so |
| 13:42.19 | matthew-moretalk | so if kamailio subscribes to asterisk then you think the notify packets will then have the username in? |
| 13:42.29 | mjordan | there's at least a chance :-) |
| 13:43.01 | mjordan | it depends on how the SUBSCRIBE is populated |
| 13:43.31 | matthew-moretalk | okays thanks ill check that out. I mean talking with you guys has given me some ideas on how to fudge it anyway if needs be |
| 13:43.35 | leetwanker | dd-wrt*CLI> module load app_followme.so |
| 13:43.35 | leetwanker | Unable to load module app_followme.so |
| 13:43.35 | leetwanker | Command 'module load app_followme.so' failed. |
| 13:43.36 | leetwanker | [Jan 27 08:43:03] WARNING[15788]: loader.c:418 load_dynamic_module: Error loading module 'app_followme.so': File not found |
| 13:43.36 | leetwanker | [Jan 27 08:43:03] WARNING[15788]: loader.c:894 load_resource: Module 'app_followme.so' could not be loaded. |
| 13:44.38 | leetwanker | the 'dynamic' makes me think that other files are required for it to work? |
| 13:45.11 | [TK]D-Fender | leetwanker: check permissions, and the owner and be VERY sure that the version and compile-time options match |
| 13:45.43 | leetwanker | permission and owners are all in good order. |
| 13:46.11 | leetwanker | I may need help with the version and compile-time. How do I check those? |
| 13:47.44 | [TK]D-Fender | version you'll see at CLI between your servers right away |
| 13:49.02 | mjordan | 'core show settings' will give you both |
| 13:49.39 | leetwanker | the other * install isn't around anymore. I wiped it out to install this one. I pulled the app from a backup that I had made |
| 13:50.41 | leetwanker | This is my current version info: Asterisk 1.8.25.0 built by ryzhovau @ ryzhov2 on a x86_64 running Linux on 2014-01-23 07:42:52 UTC |
| 13:50.55 | mjordan | why wouldn't you just install app_followme from the local machine? |
| 13:51.40 | leetwanker | there's no .ipk package for it. I'm running an embedded version of * |
| 13:52.39 | mjordan | when you did a core show settings, under the Build Options, did you see LOADABLE_MODULES? |
| 13:52.52 | leetwanker | core show settings didn't work |
| 13:52.55 | leifmadsen | WIMPy: 0031 to dial a local area code when 031 will do seems odd to me, but I defer to someone who actually lives in Europe |
| 13:53.05 | leetwanker | I did version |
| 13:53.21 | mjordan | when you say it didn't work, what did it do? |
| 13:54.28 | leetwanker | dd-wrt*CLI> core show settings |
| 13:54.28 | leetwanker | No such command 'core show settings' (type 'core show help core show settings' for other possible commands) |
| 13:55.36 | mjordan | Do you have a cli_aliases.conf? |
| 13:56.00 | leetwanker | yes |
| 13:56.12 | mjordan | Do you have an alias for 'core show settings'? |
| 13:56.30 | leetwanker | but that's probably from my old * install also. Let me check |
| 13:56.31 | leifmadsen | even with an alias, I believe the command should still work for the non-aliased version |
| 13:56.55 | leetwanker | no, no alias |
| 13:59.02 | leetwanker | this is the only build of * I could find where the gui didn't crash asterisk when accessed. |
| 13:59.08 | leetwanker | so I'm stuck with it |
| 13:59.54 | leetwanker | but if it wasn't built with app_followme, does that mean that no app_followme.so that I find will work with it? |
| 14:00.10 | [TK]D-Fender | it has to match |
| 14:00.15 | leifmadsen | correct |
| 14:01.20 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:01.26 | leetwanker | well... crap |
| 14:01.52 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 14:02.33 | leetwanker | I guess I'll have to track down ryzhovau and ask 'em |
| 14:04.16 | mjordan | hm. Apparently 'core show settings' doesn't register with LOW_MEMORY compiled. You've got a pretty specific Asterisk build there. I wouldn't try moving anything from one machine to another. |
| 14:07.01 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 14:08.36 | leetwanker | yea I think I'll stop. But atleast my gui works! ^^ |
| 14:09.31 | leetwanker | I'd love to compile my own * but I'm not sure how to do that on my router. |
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| 14:17.50 | BrokenNoze | hi all, not sure this is the right channel, maybe someone can point me in the right direction. I've written my own IAX2 stack and have queries about ISeqno and OSeqno and what they should be set to |
| 14:18.56 | [TK]D-Fender | #asterisk-dev |
| 14:19.00 | [TK]D-Fender | wold be the place |
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| 14:23.12 | BrokenNoze | thanks |
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| 14:52.06 | jkroon | hi guys, i'm sitting with a situation where on transfer some calls just drop. |
| 14:52.44 | jkroon | I've checked my call logs, and have even done a few sip debugs but cannot seem to trace where the disconnects are coming from, is there a general guide for "things to check" that I can possibly just check myself against? |
| 14:53.41 | jkroon | and then, at a different site I've got a situation with some x-Lite phones sending the # key from features.conf for blind transfer but this doesn't seem to take effect, I've confirmed the SIP DTMF mode is RFC2833 and have looked at a rtp debug trace and can see the digits there as 0xb (which I'm hoping is #?) |
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| 14:58.20 | [TK]D-Fender | Things to check is always the same : AsteriskCLI with the appropriate debugs enabled. |
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| 15:05.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:08.46 | WIMPy | leifmadsen: I'm not sure what you mean. The "when 031 will do" bit is wrong. |
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| 15:33.31 | SGjunior | Hey guys, I have been going all over the place for implementing sms. I have two different provider. One seems to have Text to Landline capabilities since it sends back comfirmation msg back to my cellphone after the txt (it also appears as an inbound call on the cli) I am trying to catch the msg in that call (which is really an sms msg) and was hop |
| 15:33.31 | SGjunior | ing it was stored in one of the channel variables (yet it seems not since I get nothing from the coreshowchannels manager command) |
| 15:34.00 | SGjunior | has anyone been here? |
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| 15:34.44 | *** join/#asterisk crax (~bergamot@5.46.44.153) |
| 15:40.36 | crax | how can we deliver outgoing calls via tdm2400p DAHDI card in extensions.conf? |
| 15:41.09 | crax | http://pastebin.com/f2vWmmEF |
| 15:41.38 | crax | here is my extension but i could not succeed |
| 15:42.12 | crax | is there any advice that you suggest? |
| 15:45.33 | crax | there are 3 pstn lines connected to dahdi card, when there is an outgoing call it goes through DAHDI/1, when there is another outgoing call i want to deliver it through DAHDI/3, because DAHDI/1 is busy/in use |
| 15:46.29 | crax | I tried to get dialstatus devicestate but not way to do it |
| 15:47.10 | [TK]D-Fender | [10:41]craxhere is my extension but i could not succeed <- You have to show us a failure typically isf you want us to tell you why it didn't succeed |
| 15:47.34 | [TK]D-Fender | crax: And for your channels.. you should be assigning them to a GROUP so * can pick a free channel to use |
| 15:47.48 | [TK]D-Fender | crax: This is documented in the sample config |
| 15:48.17 | [TK]D-Fender | crax: exten => s,n,Goto(s-${DIALSTATUS},1) <- this will do nothing since the pattern above it is not "s" |
| 15:49.30 | crax | when there is an outgoing call it succesfully use DAHDI/1, but at the same time when second outgoing call tries to connect, it doesn't use DAHDI/3, it tries DAHDI/1 again and busy/congestion state happens so i fail |
| 15:49.43 | [TK]D-Fender | crax: also if bother return "busy" it'll loop forever |
| 15:50.04 | [TK]D-Fender | both* |
| 15:50.15 | [TK]D-Fender | set a GROUP for your channels |
| 15:50.16 | crax | yes, i want to swith to another DAHDI port but how? |
| 15:50.28 | crax | hmm |
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| 15:51.09 | crax | it is the first time that i hear GROUP, thanks i will search it |
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| 16:39.44 | SGjunior | crax, here is what it should look like imo and from what I can muster of the info given => same => n,Dial(DAHDI/G0/${EXTEN},,e,) the */G0/* specifies how asterisk picks(order) the DAHDI lines to use for your calls |
| 16:40.29 | *** join/#asterisk teddy_salad (~teddy_sal@eniccpsc01-v410.mid.sta.suddenlink.net) |
| 16:42.18 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:55.27 | ChannelZ | CNAM of the day: "IMPORTANT" |
| 16:59.25 | file | hrm, I just had an amusing thought |
| 16:59.39 | file | grabbing DIDs, setting their CNAMs to song lyrics, and calling someone in order |
| 17:00.30 | crax | SGjunior, so how can i detect if DAHDI/1 in use or not? |
| 17:00.58 | crax | now i am studing on GROUPs i hope i will handle it |
| 17:01.19 | crax | normally i use in the first step |
| 17:01.19 | crax | exten => _0X.,1,Dial(DAHDI/1/${EXTEN}) |
| 17:01.58 | crax | if DAHDI/1 is busy then need to use Dial(DAHDI/3/${EXTEN}) |
| 17:02.04 | [TK]D-Fender | You don't have to detect |
| 17:02.05 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:02.12 | [TK]D-Fender | the group picks automatically |
| 17:02.17 | [TK]D-Fender | This was a 1-minute search |
| 17:02.24 | [TK]D-Fender | group=4 |
| 17:02.27 | crax | yeah i am searhing now |
| 17:02.35 | [TK]D-Fender | Dial(DAHDI/g4/1234567890) |
| 17:02.53 | [TK]D-Fender | Assign the group in your channels. Dial the group and not a specific channel. |
| 17:03.00 | [TK]D-Fender | G4 / g4 |
| 17:03.08 | [TK]D-Fender | On is ascending order, the other descending |
| 17:05.00 | crax | so i can specify a group to multiple channels when creating right? |
| 17:05.27 | ChannelZ | yes |
| 17:05.31 | [TK]D-Fender | that is the point |
| 17:05.33 | ChannelZ | group = 2 |
| 17:05.38 | [TK]D-Fender | A group of 1 channel isn't a "group" |
| 17:05.40 | ChannelZ | channels => 1-10 |
| 17:05.42 | ChannelZ | or whatever |
| 17:06.12 | crax | now i understood thanks [TK]D-Fender and ChannelZ |
| 17:08.47 | ChannelZ | FWIW using g# in your dial goes forward through the group, or G# goes backwards though it. |
| 17:16.32 | *** join/#asterisk serafie (~erin@nat/digium/x-unkgxbitfuywtnev) |
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| 17:51.14 | TSM | does it matter to * how often you re-register your endpoints? would 60s be considered too short? |
| 17:53.32 | [TK]D-Fender | TSM: What is your goal for aiming for such a short period? |
| 17:53.44 | [TK]D-Fender | Most let it stand for an hour+ |
| 18:02.41 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-bfbjhybpmzhbmfeb) |
| 18:03.04 | TSM | [ |
| 18:03.12 | TSM | [TK]D-Fender: the dreaded sonicwall |
| 18:03.29 | [TK]D-Fender | Disable ALG and they work just fine... |
| 18:04.08 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 18:05.36 | pabelanger | Anybody ever have stuck dahdi channels on a PRI? |
| 18:05.52 | [TK]D-Fender | pabelanger: Only when using "channel destroy" |
| 18:07.14 | pabelanger | ya, not using that |
| 18:08.20 | pabelanger | wow, dahdi destroy channel 13 nuked it from my pri channels |
| 18:08.21 | pabelanger | heh |
| 18:10.20 | pabelanger | so what does 'dahdi destroy' actually do? remove the channel from the active configuration? |
| 18:12.04 | eppigy | hello |
| 18:12.39 | [TK]D-Fender | I'M SORRY I CAN'T DO THAT DAVE |
| 18:13.25 | [TK]D-Fender | pabelanger: I think that is pretty much what it does. knocks it out of groups, etc... just nukes it to the point of needing to restart * |
| 18:13.29 | [TK]D-Fender | (or the module) |
| 18:14.19 | pabelanger | Ya, god knows why that command exists. Reading http://lists.digium.com/pipermail/asterisk-users/2013-April/278544.html now |
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| 18:33.09 | pabelanger | okay, How disruptive is resetinterval for chan_dahdi? |
| 18:34.21 | paulc | is that the thing that does a channel reset every so often? to make sure the telco's awake, aware, ready to send calls on each channel etc? |
| 18:35.58 | pabelanger | paulc, Ya, I think so. IIRC, if the channel hasn't been used in X seconds (whatever resetinterval is) the dahdi resets the channel |
| 18:36.06 | pabelanger | I cannot remember if I have used it before or not |
| 18:36.16 | pabelanger | but trying to think of ways to clear a hung dahdi channel |
| 18:36.39 | paulc | Yeah, I think I have one box with a T1 in it.. I should see what it's set to, if anything.. I know our old legacy IVR product (using Dialogic T1 cards) did a daily reset on all channels.. |
| 18:36.47 | paulc | @pabelanger: "core restart now" Mwuhahaha |
| 18:37.18 | pabelanger | paulc, Ya, that is a hard hack if I cannot find the issue |
| 18:45.14 | Katty | FIDDLE STICKS. |
| 18:55.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 18:57.53 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 18:57.53 | *** mode/#asterisk [+o angler] by ChanServ |
| 18:58.22 | wonderworld | did anyone use libPRI with euroISDN lately? i setup a pbx about 4 years ago with a digium 4-port card. used to be very buggy with libPRI, switched to mISDN to make it work. Would like to know if it's considered stable by now. Have see it has gotten quiet some updates over the years... |
| 18:59.43 | *** join/#asterisk farooq (~farooq@202.166.165.50) |
| 19:00.32 | farooq | Hi I am facing issue when phone is hang up and server gives hangup log but call is not disconnected |
| 19:00.52 | [TK]D-Fender | Show us the call |
| 19:00.53 | [TK]D-Fender | ~pb |
| 19:00.54 | 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. |
| 19:00.55 | [TK]D-Fender | ^^^^ |
| 19:01.37 | wonderworld | dum dee dum |
| 19:08.03 | mjordan | wonderworld: your exact scenario aside, lots of fixes have gone into chan_dahdi and libpri over the past four years. mISDN is dead. |
| 19:11.32 | farooq | We have to call from uk to rest of the world if i give dial pattern . & 00. than call take too much time proceeding (100) and we have to call 11 to 15 digit can any body tell me dial |
| 19:11.57 | farooq | pattern for quick call |
| 19:14.04 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 19:15.08 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 19:21.38 | *** join/#asterisk monsterco (~monsterco@64.231.105.94) |
| 19:22.39 | monsterco | I have a PSTN line with Bell Canada - I use it for fax - is there any way to automatically (1) when need and not to add it when not needed for long distance areas? or does Bell offer any service to simply allow all calls go through without 1 for long distance? |
| 19:23.29 | paulc | monsterco: I feel your pain. It's always been an annoyance to me that North American telcos don't support "full national" dialing with 1+10 digits for LD *and* local cals |
| 19:24.07 | paulc | It also doesn't help that you have have 1+10 LD dialing within your home areacode (604 = local, but 604 755 xxxxx needs 1+10 where I'm at) |
| 19:24.07 | [TK]D-Fender | [14:22]monstercoI have a PSTN line with Bell Canada - I use it for fax - is there any way to automatically (1) when need and not to add it when not needed for long distance areas? or does Bell offer any service to simply allow all calls go through without 1 for long distance? <- no |
| 19:24.52 | [TK]D-Fender | monsterco: For my side I took my entire phoone list, summarized by exhange, and then hand-dialed from that line to confirm the result and made my own table to correct by |
| 19:25.06 | Katty | hi fender bender |
| 19:25.18 | [TK]D-Fender | Katty: Mew. |
| 19:25.26 | Katty | [TK]D-Fender: how're you dear? did you have a good time sat night? |
| 19:25.59 | eppigy | whatttt |
| 19:26.05 | eppigy | did TK tear down the town? |
| 19:26.11 | Katty | hugs eppigy |
| 19:26.16 | eppigy | hello Katty |
| 19:26.22 | Katty | hopefully he did. |
| 19:26.27 | eppigy | yes |
| 19:26.31 | Katty | eppigy: did you? |
| 19:26.36 | eppigy | i did not |
| 19:26.40 | [TK]D-Fender | Katty: not bad. When I left it was for an "industrial espionage" bar-show run to checkout a new name listed at the venue I'm debuting my new group at in 3 weeks. Ends up it's headed by another friend who has 2 other projects going, and another who was there as a sub on bass... |
| 19:26.42 | eppigy | i am a homebody now |
| 19:26.53 | eppigy | and I had food poisoining D: |
| 19:27.00 | eppigy | poisoning |
| 19:27.11 | Katty | [TK]D-Fender: sounds like a busy evening. |
| 19:27.15 | Katty | [TK]D-Fender: and not at all relaxing. |
| 19:27.25 | Katty | eppigy: i can't seem to picture you as a homebody |
| 19:27.37 | Katty | eppigy: did you get married while my back was turned? |
| 19:27.47 | Katty | eppigy: and fail to send me an invitation |
| 19:27.47 | eppigy | haha nah just in a relationship |
| 19:28.10 | Katty | consoles eppigy |
| 19:28.22 | Katty | eppigy: they're wonderful, aren't they? except for the homebody part |
| 19:28.30 | [TK]D-Fender | Katty: It was pretty normal. All I do is write down everything they play so I know what the competition is up to :) I also hopped on vocals for a STP song, and bass for one instrumental. Plenty of fun and a great bunch of guys... |
| 19:28.56 | Katty | [TK]D-Fender: good! at least you got to do something other than Take Notes. |
| 19:29.16 | farooq | #freepbx - User support |
| 19:29.17 | Katty | [TK]D-Fender: you really do work too much. |
| 19:30.00 | [TK]D-Fender | Katty: this is fun work though... I'm a creature of habit so going out to see a new group is a good thing even though it turned out to be 2/3 people I already know ;P |
| 19:30.13 | Katty | lol |
| 19:30.21 | [TK]D-Fender | Katty: I REALLY need to branch out (and get my sorry ass to the gym) |
| 19:30.43 | Katty | [TK]D-Fender: why don't you find yourself a ladyfriend to drag you on a manditory vacation too, while your'e at it? |
| 19:30.52 | Katty | [TK]D-Fender: it would be good for you. |
| 19:31.04 | Katty | [TK]D-Fender: maybe even make your brain shutup for an hour or so. |
| 19:31.20 | Katty | [TK]D-Fender: or in eppigy's case, 6 hours |
| 19:32.16 | [TK]D-Fender | [14:30]Katty[TK]D-Fender: why don't you find yourself a ladyfriend to drag you on a manditory vacation too, while your'e at it? <- lot of work on the pre-requisites for that due... |
| 19:33.12 | Katty | [TK]D-Fender: you have plenty of charm dear. |
| 19:33.22 | Katty | [TK]D-Fender: the only pre-requisite should be that you be yourself. |
| 19:37.04 | [TK]D-Fender | Katty: Yes, but I need to be a different me. |
| 19:37.17 | Katty | [TK]D-Fender: why do you have to be different? )= |
| 19:37.45 | [TK]D-Fender | Katty: I've done nothing since my accident last year and have not been taking care of myself. It's starting to show..... |
| 19:38.36 | Katty | [TK]D-Fender: i hate tell you this, cupcake, but when you turn 50 there are no amount of pulls up that are going to save you. |
| 19:39.09 | Katty | [TK]D-Fender: but i understand wanting to be happy with yourself first. |
| 19:39.14 | *** join/#asterisk ggayan (~ggayan@200.71.197.170) |
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| 19:41.16 | [TK]D-Fender | True, but I'm not 50. And there is a difference between realizing you have let yourself go way to long, and just neurosing over nothing while purging in a bathroom. I've never been the latter, just solidly aware of the consequences of a poor lifestyle. Don't forget I did my big dive from 265->195 in 5 months in 2003. It took a while, but I've slipped up and let my ingury drag things on... |
| 19:41.17 | [TK]D-Fender | ...longer than it deserves |
| 19:42.46 | Katty | nods |
| 19:44.51 | pabelanger | Anybody running snom 320 with their asterisk boxes? Trying to deal with an issue where the phone appears to stop registering with asterisk. Trying to get log files now, but seems when you reboot the phone, it will kickstart the register again |
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| 19:58.34 | *** part/#asterisk monsterco (~monsterco@64.231.105.94) |
| 20:11.33 | eppigy | Katty: is has its ups and downs |
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| 20:43.38 | iulhk | in freeswitch we have bypass mode where rtp media by pass and there is no load on server. what we have in asterisk is it "directmedia=yes" as well by using "directmedia=yes" do i also need to handle NAT separately by using some stun or asterisk will handle itself? |
| 20:56.25 | iulhk | <PROTECTED> |
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| 21:19.30 | iulhk | nobody is there ? |
| 21:21.07 | lpmusic | iulhk: i can't remember the name off hand, or find it on google really quick, but there are a few tools to benchmark SIP calls. To find out how many calls per second, total calls, etc |
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| 21:35.45 | *** join/#asterisk gmalsack (~gmalsack@162.210.116.242) |
| 21:37.10 | gmalsack | hey all ~ upgraded from 1.8 to 11.7 last night. everything was running fine this morning now suddenly people are complaining about 1 way audio. Looking at sip show peers, for some odd reason public address are starting to show up rather than private addresses for the phone registrations... thoughts? |
| 21:38.54 | pabelanger | check your nat settings in sip.conf, the defaults changes IIRC |
| 21:38.59 | pabelanger | also, read UPGRADE.txt |
| 21:42.46 | gmalsack | gracias mi amigo!!!!! excelente!!!!!! |
| 21:43.01 | gmalsack | icesupport = no for the win!!! |
| 21:46.27 | *** join/#asterisk mrhank (~info@c-2ec321cc-74736162.cust.telenor.se) |
| 21:51.02 | mrhank | Hi! I have a problem with registering in asterisk. On my test server the registration works fine, but when i copy the configuration (sip.conf) to my production server on another network it does not register. Asterisk says "Registration time out", and its not the firewall. I can telnet to the remote server on the correct port, i can register from the servers IP via x-lite. I have also done a |
| 21:51.03 | mrhank | udp dump and checked it in wireshark, and there i can se that i get the response "401 Unauthorized" back, but asterisk does not say that, and i know i have the correct user/password and domain as its a copy from another, fully functional server. I am out of ideas, does anyone have any tips? |
| 22:07.05 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 22:14.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:21.05 | WIMPy | mjordan: Maybe mISDN is dead, but mISDN is very much alive. |
| 22:22.09 | WIMPy | Although for some reason many Asterisk "vendors" still seem to use the "old and deprecated mISDN". |
| 22:22.45 | WIMPy | wonders what a correct term would be |
| 22:27.16 | gmalsack | PLEASE!!! send me to the moon alice!!! |
| 22:39.54 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 22:40.15 | *** join/#asterisk Deshi (~deshi@174-125-252-73.dyn.centurytel.net) |
| 22:40.49 | bsdice | no |
| 22:50.06 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 22:50.19 | WIMPy | Speaking of hardware. Does anyone know if the Digiums and/or others FXS interfaces are symmetrical? |
| 22:50.41 | *** join/#asterisk dingle (dingle@162.243.8.166) |
| 22:51.14 | dingle | hello everyone. i am new to asterisk, never new. can someone point me to the direction for a newbie to set it up? |
| 22:51.29 | [TK]D-Fender | ~book |
| 22:51.29 | 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 |
| 22:51.42 | [TK]D-Fender | Installation instructions come with the tarball. |
| 22:51.44 | SGjunior | ^ |
| 22:52.48 | dingle | yeah i have the tarball and got it all installed |
| 22:53.00 | dingle | with the 100 dependencies it took.. heh |
| 22:54.10 | mjordan | dingle: following the book is a good way to get it set up. They do a good job going through the 'best practices' for configuring Asterisk. |
| 22:54.23 | Deshi | damn i can get asterisk 1.8 with the debian package system to work fine minus google voice, i compile asterisk 11 from source load it up and cant get my sip or iax clients to connect |
| 22:54.42 | dingle | let me see if the library has it, i doubt it does |
| 22:56.49 | *** part/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 22:57.05 | [TK]D-Fender | dingle: "and a version is available for reading online at http://www.asteriskdocs.org/ " |
| 22:57.19 | dingle | it's $25 used at amazon |
| 22:57.30 | Deshi | i must be missing something to get google voice to work with asterisk since 1.8 was current |
| 22:57.40 | dingle | oh, cool |
| 22:57.44 | dingle | thank you very much |
| 22:59.44 | *** part/#asterisk dingle (dingle@162.243.8.166) |
| 23:05.36 | Deshi | latest asterisk build i am getting a jabber socket read error i am using debian wheezy |
| 23:10.33 | Deshi | if anyone can help it would be appreciated i live in an area with only satellite internet no phone service so i need to get this working so i can get calls for my job |
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| 23:35.24 | SGjunior | anyone has a VoIP to GSM modem to recommend?? (for sending sms) |
| 23:39.19 | SGjunior | either as a pci card or external modem |
| 23:40.53 | JeffC_NN | SGjunior: why not use Google Voice API? |
| 23:42.23 | [TK]D-Fender | SGjunior: chan_mobile can do this with a regular cell phone + BT IIRC |
| 23:43.53 | WIMPy | SGjunior: Any "modem" or USB stick. And I'd recommend using SMS software like gnokii. |
| 23:44.08 | SGjunior | Never even thought about it honestly :S, did not expect to get much flexibility from it. Ill check it out. As for chan_mobile, I need it to be reliable. If it happens that the phone stops broadcasting or the connection simply fails ( iphone users probably can relate) |
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| 23:46.12 | SGjunior | gnokii saves the msgs locally on the phone. I need to be able to intercept them and say run a script with it's content when they arrive |
| 23:46.53 | WIMPy | They are always saved on the phone or SIM. |
| 23:47.25 | WIMPy | Unless you get display only messages. |
| 23:47.56 | WIMPy | You retrieve them from the terminal, whtever that is. |
| 23:48.42 | *** join/#asterisk luckman212_ (~luckman21@unaffiliated/luckman212) |
| 23:49.24 | SGjunior | So, TX SMS comes into asterisk. I retrieve it's content with say Message(body). Store it somewhere. Then the sms will keep running its course to the phone or modem or w/e |
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| 23:50.23 | WIMPy | I don't know how the different asterisk channels handle them exactely. |
| 23:50.46 | WIMPy | They need to be retrieved and can than be deleted. |
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| 23:54.32 | SGjunior | ummm kk, thanks! Ill tell you what I find |
| 23:58.17 | *** part/#asterisk OpenSpace (~ja@178-223-193-216.dynamic.isp.telekom.rs) |