IRC log for #asterisk on 20150212

00:05.59[TK]D-Fendersiptipping, for those basics #freepbx can help
00:06.19[TK]D-Fendersiptipping, Join their channel and ask again in there and provide actual details for what you need help with
00:08.39phixsiptipping: I wouldn't consider WIMPy's response being a "smartass".
00:09.17phixHe was just stating that we don't support Elastix in here and justifying that reason.
00:09.59siptippingahh ok ok
00:10.20WIMPyIt's ok. I guess you can overreact if you can't find help anywhere.
00:10.21phixany way, what is a Inbound/Outbound route??  I haven't heard of that before in the context of asterisk
00:11.13phixsiptipping: Is it relating to SIP channels?  or actual network / tracffic control routing?
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00:11.37WIMPyBut actually I am in smartass mode tonight. :-) But thta't not related to this channel.
00:11.53[TK]D-FenderAnd the comment is valid regardless of attitude
00:11.57phixNo that is illrelavent :)  man my spelling sucks this morning, MORE COFFEE!!!
00:12.20WIMPyIt's FreePBXs terminology for dialplan. Possibly also involves device configuration. I don't know.
00:12.25[TK]D-Fenderphix, dialpla stuff created by FreePBNX.  Those are their structure names.
00:12.37phix[TK]D-Fender: I find it difficult to convey attitude in text, I mean the attitude I want to express that is, I get misunderstood a lot :)
00:12.46phixah ok
00:13.11[TK]D-FenderHe confirmed it... but doesn't change it's validity as a direct meaningful comment
00:13.20WIMPyWell, it's not supported in text. And a well known struggle.
00:13.33phix:)
00:13.41[TK]D-FenderAnd it's been over 5 minutes since I asked him to clarify the actual problem in 2 channels and NOTHING yet.....
00:13.53phixAh, he's that guy!
00:14.17[TK]D-FenderMaybe he had an emergency bathroom break...
00:14.25[TK]D-FenderGuess we'll see how long this takes...
00:15.54WIMPyOr calling mummy because my answer didn't help.
00:16.08WIMPy(Did I mention that I'm in a mood tonight?)
00:17.58mbowieWIMPy: I got the impression he was suggesting infobot was a smartass; although I guess by proxy, you can take the credit.
00:19.25WIMPyProxy credit. Nice one.
00:19.38[TK]D-Fender~infobot
00:19.39infobotrumour has it, infobot is A program on the IRC that helps users, ask it to do something by putting a ~ and then say a command!
00:19.39phixWIMPy: You did
00:19.45[TK]D-Fender~jbot
00:19.45infoboti guess jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass, or ibot's stupid cousin
00:19.50[TK]D-FenderMY BITCH!
00:19.58[TK]D-Fender***MINE***
00:20.06WIMPyThat's rude.
00:20.09phixheh
00:20.25phixDo I have access to infobot?
00:21.40robmal~siemens
00:21.40infobot[~siemens] Siemens C675IP, C650IP, C450IP phones OK using 2 phones. C470ip -> bad firmware for * V02123: 021230000000 / 041.00
00:21.51robmal~unify
00:22.28robmalCan I add my opinion about unify to this bot somehow?
00:24.35phixhmmm, I guess my question didn't merit a response :(
00:25.07[TK]D-Fendermessage me for him :)
00:25.24phixmeh!
00:25.25phix:)[1;2D
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01:46.22KobazMeetMe(1000,DMP,1234)   shouldn't that prompt for a pin
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01:49.51Kobazi'll just replace it with my askforpin macro
01:49.52Kobazhmm
01:50.01KobazdP doesn't ask for a pin, and D doesn't ask for a pin
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03:02.03hebber<[TK]D-Fender> Your advice with answering the line after calling attempt was made in an originate to remove the unwanted CDR record worked perfectly
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03:36.06juanmapaladhi
03:36.42juanmapaladi am a biginner in this field. where can i get study mterial for asterisk?
03:37.03juanmapalad!ping
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04:16.53ChannelZ~book
04:16.53infobotAsterisk: 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
04:16.55ChannelZ~primer
04:16.56infobotNew to asterisk configuration? Check out this primer to get started.  http://burner.com/asterisk-primer
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06:38.22juanmapaladwheres bob?
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07:24.03juanmapaladhi, can anyone show me the basic netwoek diagram of a call center?
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07:48.11MaliutaLapjuanmapalad: sip_provider->internet->asterisk->phones
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07:49.17juanmapaladMaliutaLap, is there a website for that ?
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07:50.06juanmapaladMaliutaLap, can you present that in a form of diagram? i mean fro voip (sip) how do we connect from the internet?
07:54.44MaliutaLap~book
07:54.44infobotAsterisk: 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
07:54.51MaliutaLap~primer
07:54.51infobotNew to asterisk configuration? Check out this primer to get started.  http://burner.com/asterisk-primer
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08:14.11juanmapaladanyone?
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08:39.03ChannelZEh?
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08:40.08juanmapaladhi
09:01.24ChannelZAhoyhoy.
09:06.54ChannelZYou had a question I don't understand?
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09:45.17juanmapaladChannelZ, im a starter and would like to study asterisk
09:45.25juanmapaladwhere can i download study material
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12:06.07awkHi please can somebody look at this.. http://pastebin.com/cs6582Hy
12:06.48awkIt goes quite for awhile then says nobody answered
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14:49.07g-mauriziI'm not seeing any MWI:XXX@default hints in core show hints on freepbx, can anyone guess why? what conf file are mwi hints in ?
14:49.58fileMWI hints?
14:50.00WIMPyawk: And what are we looking for there?
14:50.17WIMPyg-maurizi: MWI is MWI and doesn't require a hint.
14:51.03WIMPyBut maybe it's just the good old case of
14:51.10WIMPy~freepbx
14:51.10infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
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15:11.38WIMPyawk: What are you trying to call? Or what is it you have connected to those 4 ports?
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15:12.43WIMPyawk: And it helps if you first provide a description of the issue. If we have to dig in the dark for an unknown problem, that's not too encouraging.
15:26.32sbrathWIMPy: Regarding my fast voice issue from the other day, I just built a new server. I think it's something about the computer.
15:27.56sbrathBut I have a different problem now, I have a PRI-NET connection with a Flip-T1 cable to a Toshiba-CIX 100 and the calls go thru, Asterisk says it's bring up SPEECH,   but there is no audio in either direction?   I've tried this with a Wildcard and a Sangoma card?   What else can I try?
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15:36.00WIMPysbrath: I don't even see how that should be possible.
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15:41.36sbrathMe either.
15:41.59sbrathI turned up the gain to see if that was it, but nothing.
15:42.20sbrathIt starts when I don't even get "Ring Ring" when calling an extension.
15:42.49sbrathCould it be a bad card on the Toshiba? I know the Samgoma card is working as I just pulled it from service.
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15:44.02WIMPyAre both ends talking the same protocoll? Do they agree the call was answered?
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15:45.12sbrathThey agree that is was answered and asterisk says it's briding SPEECH,
15:45.48sbrathI'm PRI_NET, and I know the Toshiba is CPI becuase if I try to be CPE it complains.  I'm also the master-clock.
15:47.10WIMPyHard to say what's going on. Any opportunity to connect that Toshiba to something else?
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15:52.52Kobazanyone here do Avaya CM consulting?
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15:58.09nfi|ermesasterisk 1.4
15:58.17nfi|ermescall forward not working
15:58.44[TK]D-Fendernfi|ermes: What "call forwarding"?
15:59.19nfi|ermeswhen i call extension A , extension B should ring
15:59.24nfi|ermesunconditionally
15:59.46[TK]D-Fendernfi|ermes: Due to what state/action?
16:00.18nfi|ermesi tried to set the DB value CFIM/extA to the extB
16:00.31WIMPynfi|ermes: It's your dialplan. Asterisk has not built-in concept of forwarding.
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16:01.00[TK]D-Fendernfi|ermes: That is indeed just dialplan... so your dilaplan isn't using it the way you hoped
16:02.03nfi|ermesi pastebin the part of my dialplan developing this feature
16:03.05nfi|ermeshttp://pastebin.com/56rxZCQs
16:03.38sbrathWIMPy: The vendor for the Toshiba is going to come in and check their T1 card with a tester
16:03.57[TK]D-Fendernfi|ermes: That sets a value... it isn't the part that does anything  with it
16:07.12nfi|ermesi was supposing that
16:07.31[TK]D-Fendernfi|ermes: Shouldn't be "supposing" any of this ... it's your dialplan....
16:07.48WIMPysbrath: I think PCs make good tester. But they have to be know good, off course.
16:10.51nfi|ermesi was supposing the setting the value was enough :|
16:11.12WIMPy(if they are as reliable as my typing, they would be useless)
16:11.25nfi|ermeswhat do you suggest to develop call forwarding in asterisk ?
16:11.38doopnfi|ermes: everything is roll your own in asterisk
16:11.42doopeverything
16:11.53WIMPynfi|ermes: I'm pretty sure google will have some examples.
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16:12.22WIMPydoop: Well, you don't have to care about the transport of audio.
16:12.55dooptrue haha
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16:13.16[TK]D-Fendernfi|ermes: YOU have to check for whatever you set in the places in your dialplan that should care.
16:13.20[TK]D-Fender~book
16:13.21infobotAsterisk: 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
16:13.22[TK]D-Fender^^^
16:13.57[TK]D-Fendernfi|ermes: What gave you the impression that the code you showed us actually did anything on its own?
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16:17.06nfi|ermesi thught the asterisk itself was going to read that value and act conseguentally
16:17.26WIMPy>>nfi|ermes: It's your dialplan. Asterisk has not built-in concept of forwarding.
16:17.53WIMPyOr as doop quite correctly stated: You have to do EVERYTHING yourself.
16:20.20[TK]D-Fendernfi|ermes: Based on what?
16:22.45doopit's browbeating time
16:22.45nfi|ermesbased on my ignorance :|
16:23.17[TK]D-FenderBlindly copying FreePBX dialplan is possibly the worst possible way to try to learn Asterisk.
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16:35.40rrittgarn1I am aware that asking questions about ITSPs here is outside the scope of the channel. I am going to ask anyway as I have a specific need... Does anybody know of / use an ITSP that does prepaid, and allows international dialing?
16:36.17WIMPyAlmost all?
16:36.57rrittgarn1I've found the opposite with my current providers, they do not want to do prepaid and international to the Caribbean
16:39.17[TK]D-Fenderrrittgarn1: Who have you looked at?
16:41.39rrittgarn1NexVortex is having issues with their international service (on of my current providers), so i need to get away from them. Bandwidth.com won't do prepaid, Waiting on emails from L3 and other ones online dont list international with their prepaid... working on examples
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16:45.38dooprrittgarn1: what kind of call volume are you looking at
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16:48.04rrittgarn1not super high. Probably 200 minutes / day during the week
16:48.19rrittgarn1would be my guess anyway... i've not tabulated how much we're doing right now
16:49.12[TK]D-Fender~itsplist-us
16:49.12infobotHere 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
16:49.21[TK]D-FenderLast 3....
16:51.41rrittgarn1Thanks. We tried vitelity once upon a time, haven't tried flowroute or voip.ms... i did just break the vitelity search looking for area code 284 rates (British Virgin Islands)
16:52.58doopive only had good experiences with vitelity
16:55.37ChannelZI use 'em
16:59.07sbrathI've also used les.net, good experiences so far, used them for a few years. Real easy to deal with. and they are canadian so no us-taxes. :)
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17:05.28lvlinuxI like Vitelity too. If you want to go cheap, I also like DIDLogic, which has been good in my experience as well (but at 30-40% cheaper than the other providers).
17:11.32rrittgarn1les.net wins @ their sip wholesale requirements on their website
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17:25.40malachi_constantHi folks!
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17:29.04malachi_constantIs there a way to set per-user caller ID on calls using asterisk as a SIP proxy? To clarify, I have a device in sip.conf that has allows users/lines on a channel bank to place calls using asterisk.
17:34.01[TK]D-Fendermalachi_constant: Asterisk is NOT a proxy.
17:34.39[TK]D-Fendermalachi_constant: As for callerID, you can set whatever you want for it in your dialplan, or leave whatever comes in from any given channel.
17:38.00malachi_constant[TK]D-Fender: Appreciate it. I might try something like putting it in the 'Description' header on the channel bank and seeing if I can take that and set it as caller ID. The channel bank seems not to have a field for caller ID.
17:41.46[TK]D-FenderProper use of the term "Channel Bank" is to describe a gateway between digital TDM to Analog TDM.
17:41.56[TK]D-FenderIf you'r referring to a SIP GATEWAY ... thta is another matter.
17:42.49malachi_constantI think you're correct. It's a SIP gateway. It has channel bank functionality but that's not what it's being used for.
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18:21.22jeffspeffIf I get a time-out (don't receive any packets from my providers media server) after sending a CFR (confirmation to receive) during a T.38 fax session, that would indeed be something on the providers end right?
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18:54.42adoodlehas the ${HANGUPCAUSE} variable changed in asterisk 11? everytime i call it (i.e. exten => h,1,NoOp(Cause: ${HANGUPCAUSE}) ), it returns a value of 0, when the inbound caller hangs up
18:55.27adoodleoo...it might only be valid for dahdi interfaces and not SIP
18:57.39doopadoodle what behavior are you expecting
18:57.52mjordanjeffspeff: I would interpret it that way
18:58.04adoodledoop: ideally, i want to know if the SIP call terminated normally, or due to some exception
18:59.05doopsuch as
18:59.36WIMPyadoodle: It should be valid for all channeltypes.
19:00.00adoodledoop: well, i was catching the outbound call attempts via checking the ${DIALSTATUS} variable
19:00.24adoodledoop: so i'm more interested in knowing if the user hung up, or if my dialplan was incorrect and terminated the call
19:01.26adoodleWIMPy: that's what i thought....but let me double check to make sure i don't have a typo or something
19:01.58jeffspeffmjordan, any ideas on what would make it intermittent? 2 out of approximately 30 faxes will succeed. The other 28 will fail as I described above. I've confirmed my QOS/TOS/COS settings on the asterisk machine, the switch and our firewall, so i know it's not due to lost packets in my network.
19:03.02adoodlehttp://pastie.org/9942602   the cli output and relevant dialplan portion
19:03.40mjordanjeffspeff: I'd confirm that what you are seeing on your end is what your provider receives
19:03.49adoodleand that's me hanging up the inbound leg of the call (haven't gone far enough in to place an outbound dial yet...still in the IVR)
19:04.29mjordanjeffspeff: for example, if the provider never receives the CFR, or if they believe (for some reason) that the CFR came out of order, etc. - things can get out of wack
19:04.45mjordanit isn't uncommon for fax stacks to be very, very optimistic about the nominal paths that they are going to follow through their states
19:05.11jeffspeffmjordan, that's been the problem. their packet captures are very high-level and don't show hardly any details except for the start and end of the call. i had to explain to one of their engineers what a CFR was and he actually challenged me that it wasn't part of the T.38 RFC. It's been a fiasco.
19:06.16jeffspeffI'm trying to do all that I can to verify that it isn't an issue on my end.
19:09.39solmstedHello, anyone using UniMRCP and speech recognition? I can set MRCP params globally in a config file. Is there any way to set them during a call? Ideally I'd like to be able to call Asterisk's SpeechBackground application with custom MRCP values.
19:09.56adoodlehmmm....it's looking like that the ${HANGUPCAUSE} variable is only set after a Dial or Transfer occurs...at least so far in my testing...
19:09.59mjordanjeffspeff: does it look like the training succeeded prior to the CFR?
19:10.22mjordansolmsted: you'll need to ask the UniMRCP folks, as they wrote a different set of modules around the speech recognition
19:10.45mjordanspeech recognition API/Engine that Asterisk typically uses
19:11.24solmstedmjordan, so there's no way to pass extra params using Asterisk's generic speech recognition APIs?
19:11.55mjordansolmsted: I don't know.
19:12.12mjordansolmsted: you'll need to look at the UniMRCP docs for what they wrote, and the modules they wrote that make use of their engine
19:13.34jeffspeffmjordan, I can show you the pcaps if you'd like.
19:13.52mjordanjeffspeff: ew?
19:14.05jeffspeffew?
19:14.16mjordanjeffspeff: I'd look at the pcap and make sure that training succeeded and that we received a TCF.
19:14.27mjordanwe should then transmit the V.21 flags, followed by a CFR.
19:14.32mjordanif we don't get anything back after that, they've screwed up.
19:14.47fileI'm sorta glad I've forgotten this stuff
19:14.53mjordanjeffspeff: digging through T.38 pcaps is probably my least favorite activity of all time.
19:14.58solmstedmjordan, okay thanks, I was also hoping there was a common solution that could span other engines instead of using a UniMRCP specific interface, but so far it doesn't look like it
19:15.47mjordanfile: There's always T.38 over datachannels
19:15.53filemjordan, hush
19:15.58mjordanfile: I'm sure we'll be violating that RFC soon enough
19:16.12filesolmsted, UniMRCP does not use the generic mechanism in Asterisk to do it - thus the generic common way is of no use
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19:23.44solmstedfile: I've been using UniMRCP with the generic APIs SpeechCreate, SpeechLoadGrammar, SpeechBackground etc
19:24.08filethey have a module for that? interesting
19:24.23jeffspeffmjordan, here's the jist an inbound fax flow... after the invites, acks, and agreeing to use SDP(t38)... i send a no-signal to the prvdr media svr, i recieve the same. i send v21-preamble, i send CSI, I send DIS DSR:ITU-TV and then a no-signal. I get back a v21-preamble, DCS DSR:14400, no-signal, v17-14400-long-training, then a no-signal. I send a v21-preamble followed by a CFR and then a no-signal. I don't get anything in return at this point so I send
19:24.24jeffspeffa SDP invite to their sip srvr they say ok which I ack. Then I get a single RTP packet from their media server at which point i send a BYE to their sip srvr.
19:24.25fileI haven't seen it but the SPEECH_ENGINE dialplan function is supposed to be use to set engine specific properties
19:24.35fileI don't know how/if they implemented it, that's in their court
19:25.24mjordanjeffspeff: you're doing it right. The issue is 99% on your provider's end.
19:25.31mjordan99% sure, anyway
19:25.50mjordangranted, if they don't get the CFR, or if they drop the V.21 preamble, then of course, things may get 'broke' on their end.
19:25.57mjordanso there may just be Network shenanigans
19:26.13mjordanbut if they can't look at the packets and confirm or understand what is going on, I'd start shopping for a new provider
19:26.36jeffspeffthey can't/won't provide me with or even confirm that they're getting the CFR or replying to it.
19:26.50solmstedfile, interesting is there any docs for SPEECH_ENGINE?  I haven't seen it before.  I'm finding that whatever speech recognition engine I use, there are more parameters I'd like to be able to set, but the API in Asterisk doesn't give me any way to set them
19:27.17filesolmsted, it's a generic opaque dialplan function which calls into the engine
19:27.23filethe parameters it takes are up to the implementation of the engine
19:27.27jeffspeffthey told me they only support RTP for t38 and don't support udptl at all, yet all of the faxes (even the ones that have succeeded) have been UDPTL
19:27.47mjordanyeah...
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19:28.40filesolmsted, https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SPEECH_ENGINE that's literally it - because the attributes are specific to the engine... there are no common ones
19:29.04jeffspeffI had read that if my t38_udptl setting doesn't match what they have then t38 won't work right. if so, is there a way to determine what they're expecting? I've tried yes; yes,fec; yes,fec,maxdatagram=400; yes,redundancy; yes,redundancy,maxdatagram=400
19:33.15solmstedfile, thanks, that gives me something to look into.  Google didn't get me there for some reason and SPEECH_ENGINE isn't listed here: https://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API
19:34.13fileah, an older page
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19:48.12nnyHey all. Asked this question many years ago and was told mediatrix is a good option, but i'll ask again as it's been a while. I am going to be putting in enough media gateways to handle ~ 850 phones from analog to FXS/SIP to an asterisk server. They aren't used that much (think land lines in a resort) so Asterisk should be fine as the SIP switch/server. Apart from Mediatrix any other hardware suggested?
19:50.12nnyCertainly staying away from Sangoma, have a Europa 8 port FXO device and was a pain to deal with
19:50.33nnyi use sangoma cards, but the Europa manufacturer they bought isn't really the same thing at all
19:54.48[TK]D-Fendernny: Mediatrix became Media5.  AudioCodes is another big player in the same range
19:55.02nny[TK]D-Fender: sounds good. I am looking around now.
19:55.13[TK]D-Fendernny: My experiece with Mediatrix 24-port FXS back ~2007 was good.  Those weere indeed very simply to configure
19:55.23nnyYeah the Vega OTOH oh boy.
19:56.17nny[TK]D-Fender: I'll shop around. Thank you
19:56.34[TK]D-Fendernny: You're welcome
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21:29.01[TK]D-Fendercheckout time, BBIAB
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21:49.39adoodleare there any troubleshooting tips for debugging odbc cdr's? my cdr's won't insert and i cant' see why it's failing....
22:02.54malachi_constanthttp://pastebin.com/AQ5cxxPE <-- should this stuff work? Asterisk seems not to add the header.
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22:24.24malachi_constantOK I fixed that -- what I'm trying to do now is take the From: header from the first packet sent to asterisk and then switch that phone number into the P-Asserted-Identity field.
22:27.32malachi_constantIs there a way to take the data in a header and use it in the dialplan?
22:28.53newtonrmalachi_constant, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
22:31.14malachi_constantnewtonr: VERY cool!
22:31.21malachi_constantThank you!
22:32.11newtonrAlso, a lot of information in headers is already extracted into various channel variables and other places
22:32.15newtonrfor example: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL
22:32.36newtonrhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
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22:38.20malachi_constantnewtonr: That's awesome. ${CALLERID(num)} was exactly what I was looking for. Thanks!
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22:40.54newtonrmalachi_constant, Cool, you are welcome
22:41.39newtonrmalachi_constant, you may be interested in this whole page: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
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22:42.57malachi_constantWow. Yes, I am. :-)
22:52.47*** join/#asterisk mafairnet (~mafairnet@187.188.176.204)
22:54.23mafairnetHello, how can I use one extension in multiple devices (not at the same time). I'm using AsteriskNow (FreePBX + Asterisk 11).
22:54.58MaliutaLapQueue?
22:55.27mafairnetnope
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22:57.41rexwin_I am able to goto my IVR when I add 1 to my DID number or else it says cannot reach. how to check this?
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22:57.43mafairnetI want that a user have one extension and can use it on multiple phones (Not at the ssame time).
22:57.45MaliutaLapNot entirely sure you have actually described the requirement in sufficient detail ... multiple end points one extension? but only one end point at a time?
22:58.21rexwin_your call cannot be completed, please check the number and dial again
22:58.27MaliutaLapASTDB+login+thing that logs out previous instance?
22:59.02MaliutaLaprexwin_: pre-pend, or append?
22:59.39rexwin_prepend 1 works but direct did doesnot
22:59.52MaliutaLaprexwin_: check the phone's dialplan
23:00.29MaliutaLaprexwin_: or ignorepat
23:01.01mafairnetHave yu ever used User/Device mode?
23:01.31MaliutaLaprexwin_: for example if you have a phone specific dialplan that means it needs to dial '1' to get a specific connection that will be the issue
23:01.44MaliutaLapmafairnet: not personally
23:02.07rexwin_no, i dont have phone specific dialplan
23:02.38rexwin_all other did works without prepending 1 except for one did number, all are similar in function
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23:08.04mafairnet:/
23:09.10mafairnethave someone tried blocking transfers from one extension to the same queue that it belongs?
23:10.24MaliutaLaphah? isn't that a function of the Dial() options?
23:13.30*** join/#asterisk tom_w_______ (~telcomind@wsip-24-120-113-148.lv.lv.cox.net)
23:14.49tom_w_______hello,  I just was forced to upgrade to centos 7 and asterisk current.  I now get this error  res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination
23:15.13tom_w_______I have googled and not found much of a response.  Can someone point me in the right direction
23:18.38MaliutaLaptom_w_______: is there an upgrade of * versions?
23:18.46MaliutaLaptom_w_______: and from what to what?
23:20.00tom_w_______1.12 to 13.2.0
23:20.14[TK]D-Fender1.12 is not an * version
23:21.14tom_w_______bad typing on my part 1.13 to 1.13.2
23:21.49tom_w_______i did go from centos 6 to 7
23:22.54MaliutaLapthat doesn't sound like enough of a version change for a dist upgrade like that
23:23.11MaliutaLapsure you're not running 13 now?
23:23.24tom_w_______I am running 13.2 now
23:23.33tom_w_______compiled from source
23:23.39MaliutaLapand previously you were running?
23:24.02tom_w_______centos 6 with 1.13
23:24.06[TK]D-Fender<tom_w_______> hello,  I just was forced to upgrade to centos 7 and asterisk current.  I now get this error  res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination <- sounds like an IP4/IP6 misimatch
23:24.36tom_w_______no ipv6 on he server.
23:24.43tom_w_______I can verify and make sure
23:24.53MaliutaLaptom_w_______: and did you bring the configs up to spec? there is a big change between those versions of *
23:26.16tom_w_______yes.  very simple *.  just 1 sip peer who sends to 8 t1's.  everything else is from samples
23:27.29[TK]D-Fendertom_w_______, place another call with SIP DEBUG enabled
23:27.36tom_w_______ok
23:27.36[TK]D-Fender"sip set debug on"
23:27.37[TK]D-Fender~pb
23:27.38infobotA "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.
23:27.40[TK]D-Fender^^^^
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23:43.33MaliutaLapnotes the silence
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