IRC log for #asterisk on 20130526

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01:23.17danfromukGot an unusual issue. When I submit a 'dialplan reload' from the cli, I get no errors or warnings. But when I issue the same command from the AMI, i get loads of warnings about contexts already been included.
01:23.21danfromukeg... Context 'standard_service' already included in 'tbirch_phones' context on include at line 538 of /etc/asterisk/extens_peers.conf
01:23.38danfromukI dont get that warning when using the CLI.
01:24.11WIMPyHave you compared results?
01:26.36danfromukcompared results? Theres nothing to compare. I get no warnings when issueing 'dialplan reload' from the CLI. And get loads of warnings (all identical but mentioning different contexts) when issueing a 'dialplan reload' from the AMI.
01:27.03danfromukI cant seem to find any duplicates so i think the warnings generated by the AMI are incorrect.
01:27.14WIMPy'dialplan show' anythin doubled?
01:27.37[TK]D-Fenderdanfromuk: Show us
01:27.47WIMPyMight be interesting to see the result if you remove something.
01:28.30WIMPySounds like it doesn't start with an empty dialplan to me.
01:28.43danfromukActually... I've misdiagnosed it.
01:28.56danfromukIf verbose is set to 2 or below, i get loads of warnings.
01:29.10danfromukIf verbose is set to 3 or more, it appears to reload properly.
01:29.33WIMPyWhich way round?
01:30.02danfromukThe way I just stated. verbose <=2, loads of warnings.
01:30.31danfromukverbose >= 3, loads of 'added extension' messages but no warnings.
01:30.43WIMPydebug?
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01:36.18danfromukOk. Honest. I have no idea whats causing this. Sometimes it throws warnings, or times its fine.
01:40.29danfromukIts getting late. I'll try figuring it out tomorrow. Thanks anyway.
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03:05.23jayktitz
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03:59.43KobazMM
04:00.36WIMPyWW
04:04.58KobazQQ
04:05.15WIMPyebay?
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04:09.16Kobazit was more like i had tme caps lock on
04:09.32Kobazwhich i usually disable, i wonder why it's not anymore
04:09.44WIMPyGremlins
04:11.09jayktitz aren't generally sold on ebay
04:11.28WIMPyThey are no longer?
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04:21.12Kobaznap time
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10:19.20deohi all.. im in the middle of troubleshooting my asterisk server, i had to test one extension for both inbound and outbound calls.. i noticed that when i hangup the call. i can still see on the logs that the extension is still using the dahdi channel.. i know i can soft hangup the extension but i need to know whats causing this issue..
10:20.20WIMPyExtensions are the destinations defined in extensions.conf.
10:20.36WIMPySo what is that other end you have trouble hanging up?
10:20.42deomy question is how am i going to show some properties of this channel being used by the extension and see if i can found one ..
10:21.57deoWIMPy: the other end is a PSTN.. i tried to call the extension back and forth..and it seems okay but there are times that it will stucked and basing the logs its still using the channel dahdi 3/1
10:22.19deothats why i need to softhangup the channel.. but i dont know whats causing this issue..
10:22.38WIMPySo it's the dahdi channel not detecting the far end hangup?
10:22.57deoyes WIMPy i believed...
10:23.20WIMPyYou tell us what your question is.
10:23.31WIMPyIs that a POTS line?
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10:23.45deothe other end it is..
10:24.00deohttp://pastie.org/7961756
10:24.02WIMPyOn your end.
10:24.11deothis is the one ive talking about ^^
10:24.39WIMPyIs that a POTS line?
10:26.43deoon the asterisk server end, the line is from our regular PSTN provider..
10:27.17WIMPyWhat kind of dahdi interface is this?
10:27.54deowhat do you mean what kind WIMPy ?
10:28.27WIMPyWhat kind of line are we talking about?
10:29.24deoahmm not really sure what your question mean but i know its a dahdi channel being used here DAHDI/3-1
10:29.46WIMPyThat could be anything.
10:30.08deohow do i really check the line you are referring to WIMPy ?
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10:30.22deocan i do that on the command line on asterisk?
10:30.46WIMPyBy looking at what you ordered or what you pay for, or by looking at the hardware you've got.
10:31.43deoahhhhh okay its a TDM800P
10:32.27WIMPySo that is POTS.
10:32.41WIMPyThought so. In that case it's perfectely normal.
10:33.12WIMPyUnless your line provides disconnect supervision you don't know when the far end has hung up.
10:33.24WIMPyCheck what your telco can offer.
10:33.41deoahhmm so your saying that this issue is normal when using POTS?
10:33.46WIMPyIf they can't offer anythign you can experiment with busydetect settings.
10:33.53WIMPyYes.
10:34.02WIMPyAnalog is evil.
10:34.05WIMPyGet digital.
10:34.30deohmmnn what actually you mean by Digital? im sorry im pretty new to this Asterisk world..
10:34.46deoso i know that this POTS is analog..
10:34.46WIMPyISDN (BRI/PRI) or VOIP.
10:35.41WIMPyPOTS *is* analog.
10:35.59deohmmnnn im not really familiar with those terms.. and I know your busy so i may not dig this with you...
10:36.36WIMPy~pots
10:36.36infobot[~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide.  The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS).  POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco.
10:37.07deohmmnn that make sense..
10:37.52deoby the way the sip client that im using is from the remote side...
10:38.53deosip client(Xlite) is from remote user thats me..
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10:39.33WIMPyNot sure what that's supposed to tell me.
10:40.00deoWIMPy: can you advise/suggest on how can we deal with this POTS? or are you suggesting that we should go for ISDN(BRI/PRI)
10:40.09deoand leave this POTS
10:40.31WIMPyOr VOIP
10:40.35deobut i think we may need this for local numbers in our region.. i dont know if im saying this right :)
10:40.38WIMPyDepends on your location.
10:40.59WIMPyBut you won't have much fun with POTS.
10:41.03deodepends on our location? hmmmn so ISDN picks places? :)
10:41.26deohmmnn but if we used ISDN, can we still call local numbers?
10:41.31WIMPyNo, but it seems to be rare and expensive in North America.
10:41.40deoohhhh
10:42.03WIMPyISDN is still PSTN, just digital and with a lot more features.
10:42.18deoahhh okay...now im learning with you..
10:42.38deoi dont know if we have it here in our region :p
10:42.53WIMPyWhere are you?
10:42.58deoPhilippines
10:43.37WIMPyNo idea :-) Check with your telco what they can offer.
10:43.47deoyeah :)
10:44.40deoif i can remember.. you said a while ago regarding busydetect settings
10:44.58deois it implemented on the asterisk right?
10:45.05WIMPyYes.
10:45.15WIMPyBut it's not really reliable.
10:45.23deohmmnn..
10:45.42deookay so this hangup issue is really know when using POTS...
10:45.51WIMPyyes
10:46.01deowill try to look for that ISDN....
10:46.32deobut this has nothing to do with the card that im using WIMPy ? TDM800P
10:46.53deoso when switching to ISDN i can still used this card?
10:47.10WIMPyNo, you need another type of card then.
10:47.32deooooopss that will cost something :)
10:48.00WIMPyDepends on the number of channels you need.
10:48.21WIMPyMultiport cards cost money, single port cards cost next to nothing.
10:48.23deowhat card to be used when using ISDN?
10:48.44WIMPyThere are many to choose from. How many channels?
10:48.59deocurrently were using 4 lines from our telco.. so thats 4 channels
10:49.39WIMPyYou could do that with two el cheapo cards then.
10:49.55deohmnnn
10:50.09WIMPyBut better check what you can get first.
10:50.35deookay... thank you for that advise WIMPy :) ill check what can i do ..
10:50.53WIMPyMaybe the Phillippines are a region where you should go for an ITSP instead of PSTN lines.
10:51.13deo0_o ITSP?
10:51.17WIMPy~itsp
10:51.18infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
10:51.42deoohhhhh... okay let me add that too...
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10:51.47deothank you WIMPy :)
10:51.57[sr]hi WIMPy, always working :)
10:52.04WIMPyThat would go without any hardware. Just IP.
10:52.17WIMPyHi [sr]. No, not always. :-)
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10:53.17WIMPyHow are your chan_dongle experiments?
10:53.24deohmmn i dont know if this question would matter.. when using that ITSP or ISDN.. is the end part of the cable is still and rj11? :p
10:53.45deoim talking about the line for the telco :)
10:53.59deoi mean from the telco
10:54.22WIMPyNo. ISDN comes on an 8p8c or 8p4c modular plug, The ITSP comes via internet.
10:54.34[sr]if it's IP there's nothing else than your internet
10:55.07deooops so that ITSP needs a lot of bandwidth...
10:55.29deohmmnn okay
10:55.39WIMPyDepending on codec and latency up to 100kbps/call.
10:56.34deohmmnn okay noted...
10:58.29deothanks WIMPy [sr] :)
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11:02.01[sr]deo: 50€ :)
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11:04.04WIMPywonders how much he should charge at that rate.
11:04.46[sr]WIMPy: at least an extra zero :)
11:05.04WIMPyWhat am I supposed to do with Zero? ;-)
11:05.34[sr]an extra zero, on the right of 50 :)
11:11.45WIMPyThat would make a nice business.
11:11.53WIMPyHow are your chan_dongle experiments?
11:22.25[sr]WIMPy: havent made any progress... have to get a dongle, only have one and it's used for internet
11:22.46[sr]but havent found much of examples of the integration/configuration
11:22.56WIMPyhasn't got round to any experiments, either.
11:23.24WIMPyWell, I did make a call. But that was all so far.
11:23.56[sr]thats far than what i did till now
11:25.24[sr]for what i saw, the conf is a bit easy, will be, Dial(Dongle/x1/etc etc)
11:27.11[sr]project seems a bit stoped... so im a bit afraid to use it on production
11:27.55WIMPyYes, most 3rd party developers seem to be fed up with the changes in Asterisk :-(
11:28.14[sr]that :S
11:28.44[sr]anyway chan_dongle should do into asterisk tree, it's a great great ideia to use dongle's
11:28.50[sr]my opinion
11:28.59[sr]GSM manufactors won't like it
11:29.18WIMPyIt certainly makes sense. Like many things did.
11:31.07[sr]lets see what the future brings... for me a 1x SIM IP GSM  costs arround 240€+VAT
11:32.04WIMPyI'd also like to try to do it via osmocom.
11:35.25[sr]hum
11:35.33[sr]have to studdy that
11:36.31WIMPyJa, that one is a lot more complicated. But it offers a lot more features.
11:38.21[sr]ya, i like specially this one: Sending of SMS from external applications by writing to the SQL tables
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11:46.54[sr]WIMPy: be well, lunch now!
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14:53.25DexterParkAnybody else excited about Asterisk: The Definitive Guide, 4th Edition coming out soon?
14:58.45Freeaqingmehell yeah
15:00.57DexterParkI think the PDF version is already out , but I pre-ordered the Paperback already so I'm waiting for that.
15:01.34DexterParkYou can get them both for $60 - http://shop.oreilly.com/product/0636920025894.do
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15:10.39dorknetI am interested in offering an incoming context and integration to echolink
15:10.43dorknetI have googled everything
15:10.52dorknetanyone point me in the right direction pretty pretty please?
15:24.33WIMPyWhat is echolink?
15:52.13dorknetit is a ham radio voip network, usually integrated into from community radio repeater systems
15:52.37dorkneti want to offer a context that can be connected to from several of the repeaters, and join them to a silent meet me conference
15:52.50dorknetecho link is really only interesting if you are a ham radio person
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17:00.17dorknetany idea where i can start to ask these questions?  i have seen a deprecated echolink.conf for asterisk, and another article mentioning res_RTP...
17:00.26dorkneti would have thought it would use XMPP or Jabber or something...
17:00.51dorkneti have been spending every evening and weekend playing with my new asterisk server, and i just think you guys, and this project, are the best
17:00.52dorknetthank you
17:01.11dorkneti used to do a bit of phreaking as a kid, and i just can't beleive how amazing this toy is.
17:01.18dorkneti have wanted something like this since i was 11
17:01.37dorknetI'm 30 now, and a decent coder, this application has consumed my free time
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