| 00:05.59 | [TK]D-Fender | siptipping, for those basics #freepbx can help |
| 00:06.19 | [TK]D-Fender | siptipping, Join their channel and ask again in there and provide actual details for what you need help with |
| 00:08.39 | phix | siptipping: I wouldn't consider WIMPy's response being a "smartass". |
| 00:09.17 | phix | He was just stating that we don't support Elastix in here and justifying that reason. |
| 00:09.59 | siptipping | ahh ok ok |
| 00:10.20 | WIMPy | It's ok. I guess you can overreact if you can't find help anywhere. |
| 00:10.21 | phix | any way, what is a Inbound/Outbound route?? I haven't heard of that before in the context of asterisk |
| 00:11.13 | phix | siptipping: Is it relating to SIP channels? or actual network / tracffic control routing? |
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| 00:11.37 | WIMPy | But actually I am in smartass mode tonight. :-) But thta't not related to this channel. |
| 00:11.53 | [TK]D-Fender | And the comment is valid regardless of attitude |
| 00:11.57 | phix | No that is illrelavent :) man my spelling sucks this morning, MORE COFFEE!!! |
| 00:12.20 | WIMPy | It's FreePBXs terminology for dialplan. Possibly also involves device configuration. I don't know. |
| 00:12.25 | [TK]D-Fender | phix, dialpla stuff created by FreePBNX. Those are their structure names. |
| 00:12.37 | phix | [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 :) |
| 00:12.46 | phix | ah ok |
| 00:13.11 | [TK]D-Fender | He confirmed it... but doesn't change it's validity as a direct meaningful comment |
| 00:13.20 | WIMPy | Well, it's not supported in text. And a well known struggle. |
| 00:13.33 | phix | :) |
| 00:13.41 | [TK]D-Fender | And it's been over 5 minutes since I asked him to clarify the actual problem in 2 channels and NOTHING yet..... |
| 00:13.53 | phix | Ah, he's that guy! |
| 00:14.17 | [TK]D-Fender | Maybe he had an emergency bathroom break... |
| 00:14.25 | [TK]D-Fender | Guess we'll see how long this takes... |
| 00:15.54 | WIMPy | Or calling mummy because my answer didn't help. |
| 00:16.08 | WIMPy | (Did I mention that I'm in a mood tonight?) |
| 00:17.58 | mbowie | WIMPy: I got the impression he was suggesting infobot was a smartass; although I guess by proxy, you can take the credit. |
| 00:19.25 | WIMPy | Proxy credit. Nice one. |
| 00:19.38 | [TK]D-Fender | ~infobot |
| 00:19.39 | infobot | rumour 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! |
| 00:19.39 | phix | WIMPy: You did |
| 00:19.45 | [TK]D-Fender | ~jbot |
| 00:19.45 | infobot | i 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-Fender | MY BITCH! |
| 00:19.58 | [TK]D-Fender | ***MINE*** |
| 00:20.06 | WIMPy | That's rude. |
| 00:20.09 | phix | heh |
| 00:20.25 | phix | Do I have access to infobot? |
| 00:21.40 | robmal | ~siemens |
| 00:21.40 | infobot | [~siemens] Siemens C675IP, C650IP, C450IP phones OK using 2 phones. C470ip -> bad firmware for * V02123: 021230000000 / 041.00 |
| 00:21.51 | robmal | ~unify |
| 00:22.28 | robmal | Can I add my opinion about unify to this bot somehow? |
| 00:24.35 | phix | hmmm, I guess my question didn't merit a response :( |
| 00:25.07 | [TK]D-Fender | message me for him :) |
| 00:25.24 | phix | meh! |
| 00:25.25 | phix | :)[1;2D |
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| 01:46.22 | Kobaz | MeetMe(1000,DMP,1234) shouldn't that prompt for a pin |
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| 01:49.51 | Kobaz | i'll just replace it with my askforpin macro |
| 01:49.52 | Kobaz | hmm |
| 01:50.01 | Kobaz | dP doesn't ask for a pin, and D doesn't ask for a pin |
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| 03:02.03 | hebber | <[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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| 03:36.06 | juanmapalad | hi |
| 03:36.42 | juanmapalad | i am a biginner in this field. where can i get study mterial for asterisk? |
| 03:37.03 | juanmapalad | !ping |
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| 04:16.53 | ChannelZ | ~book |
| 04:16.53 | 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 |
| 04:16.55 | ChannelZ | ~primer |
| 04:16.56 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
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| 06:38.22 | juanmapalad | wheres bob? |
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| 07:24.03 | juanmapalad | hi, can anyone show me the basic netwoek diagram of a call center? |
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| 07:48.11 | MaliutaLap | juanmapalad: sip_provider->internet->asterisk->phones |
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| 07:49.17 | juanmapalad | MaliutaLap, is there a website for that ? |
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| 07:50.06 | juanmapalad | MaliutaLap, can you present that in a form of diagram? i mean fro voip (sip) how do we connect from the internet? |
| 07:54.44 | MaliutaLap | ~book |
| 07:54.44 | 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 |
| 07:54.51 | MaliutaLap | ~primer |
| 07:54.51 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
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| 08:14.11 | juanmapalad | anyone? |
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| 08:39.03 | ChannelZ | Eh? |
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| 08:40.08 | juanmapalad | hi |
| 09:01.24 | ChannelZ | Ahoyhoy. |
| 09:06.54 | ChannelZ | You had a question I don't understand? |
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| 09:45.17 | juanmapalad | ChannelZ, im a starter and would like to study asterisk |
| 09:45.25 | juanmapalad | where can i download study material |
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| 12:06.07 | awk | Hi please can somebody look at this.. http://pastebin.com/cs6582Hy |
| 12:06.48 | awk | It goes quite for awhile then says nobody answered |
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| 14:49.07 | g-maurizi | I'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 ? |
| 14:49.58 | file | MWI hints? |
| 14:50.00 | WIMPy | awk: And what are we looking for there? |
| 14:50.17 | WIMPy | g-maurizi: MWI is MWI and doesn't require a hint. |
| 14:51.03 | WIMPy | But maybe it's just the good old case of |
| 14:51.10 | WIMPy | ~freepbx |
| 14:51.10 | 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 |
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| 15:11.38 | WIMPy | awk: What are you trying to call? Or what is it you have connected to those 4 ports? |
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| 15:12.43 | WIMPy | awk: 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. |
| 15:26.32 | sbrath | WIMPy: Regarding my fast voice issue from the other day, I just built a new server. I think it's something about the computer. |
| 15:27.56 | sbrath | But 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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| 15:36.00 | WIMPy | sbrath: I don't even see how that should be possible. |
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| 15:41.36 | sbrath | Me either. |
| 15:41.59 | sbrath | I turned up the gain to see if that was it, but nothing. |
| 15:42.20 | sbrath | It starts when I don't even get "Ring Ring" when calling an extension. |
| 15:42.49 | sbrath | Could 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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| 15:44.02 | WIMPy | Are both ends talking the same protocoll? Do they agree the call was answered? |
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| 15:45.12 | sbrath | They agree that is was answered and asterisk says it's briding SPEECH, |
| 15:45.48 | sbrath | I'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. |
| 15:47.10 | WIMPy | Hard to say what's going on. Any opportunity to connect that Toshiba to something else? |
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| 15:52.52 | Kobaz | anyone here do Avaya CM consulting? |
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| 15:58.09 | nfi|ermes | asterisk 1.4 |
| 15:58.17 | nfi|ermes | call forward not working |
| 15:58.44 | [TK]D-Fender | nfi|ermes: What "call forwarding"? |
| 15:59.19 | nfi|ermes | when i call extension A , extension B should ring |
| 15:59.24 | nfi|ermes | unconditionally |
| 15:59.46 | [TK]D-Fender | nfi|ermes: Due to what state/action? |
| 16:00.18 | nfi|ermes | i tried to set the DB value CFIM/extA to the extB |
| 16:00.31 | WIMPy | nfi|ermes: It's your dialplan. Asterisk has not built-in concept of forwarding. |
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| 16:01.00 | [TK]D-Fender | nfi|ermes: That is indeed just dialplan... so your dilaplan isn't using it the way you hoped |
| 16:02.03 | nfi|ermes | i pastebin the part of my dialplan developing this feature |
| 16:03.05 | nfi|ermes | http://pastebin.com/56rxZCQs |
| 16:03.38 | sbrath | WIMPy: The vendor for the Toshiba is going to come in and check their T1 card with a tester |
| 16:03.57 | [TK]D-Fender | nfi|ermes: That sets a value... it isn't the part that does anything with it |
| 16:07.12 | nfi|ermes | i was supposing that |
| 16:07.31 | [TK]D-Fender | nfi|ermes: Shouldn't be "supposing" any of this ... it's your dialplan.... |
| 16:07.48 | WIMPy | sbrath: I think PCs make good tester. But they have to be know good, off course. |
| 16:10.51 | nfi|ermes | i was supposing the setting the value was enough :| |
| 16:11.12 | WIMPy | (if they are as reliable as my typing, they would be useless) |
| 16:11.25 | nfi|ermes | what do you suggest to develop call forwarding in asterisk ? |
| 16:11.38 | doop | nfi|ermes: everything is roll your own in asterisk |
| 16:11.42 | doop | everything |
| 16:11.53 | WIMPy | nfi|ermes: I'm pretty sure google will have some examples. |
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| 16:12.22 | WIMPy | doop: Well, you don't have to care about the transport of audio. |
| 16:12.55 | doop | true haha |
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| 16:13.16 | [TK]D-Fender | nfi|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 |
| 16:13.21 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:13.22 | [TK]D-Fender | ^^^ |
| 16:13.57 | [TK]D-Fender | nfi|ermes: What gave you the impression that the code you showed us actually did anything on its own? |
| 16:15.45 | *** join/#asterisk cyford (~junkmail@76.122.73.37) |
| 16:17.06 | nfi|ermes | i thught the asterisk itself was going to read that value and act conseguentally |
| 16:17.26 | WIMPy | >>nfi|ermes: It's your dialplan. Asterisk has not built-in concept of forwarding. |
| 16:17.53 | WIMPy | Or as doop quite correctly stated: You have to do EVERYTHING yourself. |
| 16:20.20 | [TK]D-Fender | nfi|ermes: Based on what? |
| 16:22.45 | doop | it's browbeating time |
| 16:22.45 | nfi|ermes | based on my ignorance :| |
| 16:23.17 | [TK]D-Fender | Blindly copying FreePBX dialplan is possibly the worst possible way to try to learn Asterisk. |
| 16:25.48 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
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| 16:35.40 | rrittgarn1 | I 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? |
| 16:36.17 | WIMPy | Almost all? |
| 16:36.57 | rrittgarn1 | I'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-Fender | rrittgarn1: Who have you looked at? |
| 16:41.39 | rrittgarn1 | NexVortex 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 |
| 16:45.14 | *** join/#asterisk mjordan (mjordan@nat/digium/x-txxsuxepchbeuxpb) |
| 16:45.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:45.38 | doop | rrittgarn1: what kind of call volume are you looking at |
| 16:48.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:48.04 | rrittgarn1 | not super high. Probably 200 minutes / day during the week |
| 16:48.19 | rrittgarn1 | would be my guess anyway... i've not tabulated how much we're doing right now |
| 16:49.12 | [TK]D-Fender | ~itsplist-us |
| 16:49.12 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 16:49.21 | [TK]D-Fender | Last 3.... |
| 16:51.41 | rrittgarn1 | Thanks. 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) |
| 16:52.58 | doop | ive only had good experiences with vitelity |
| 16:55.37 | ChannelZ | I use 'em |
| 16:59.07 | sbrath | I'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. :) |
| 17:00.01 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:05.28 | lvlinux | I 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). |
| 17:11.32 | rrittgarn1 | les.net wins @ their sip wholesale requirements on their website |
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| 17:25.15 | *** join/#asterisk malachi_constant (~root@unaffiliated/malachiconstant/x-837457) |
| 17:25.40 | malachi_constant | Hi folks! |
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| 17:29.04 | malachi_constant | Is 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-Fender | malachi_constant: Asterisk is NOT a proxy. |
| 17:34.39 | [TK]D-Fender | malachi_constant: As for callerID, you can set whatever you want for it in your dialplan, or leave whatever comes in from any given channel. |
| 17:38.00 | malachi_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-Fender | Proper use of the term "Channel Bank" is to describe a gateway between digital TDM to Analog TDM. |
| 17:41.56 | [TK]D-Fender | If you'r referring to a SIP GATEWAY ... thta is another matter. |
| 17:42.49 | malachi_constant | I 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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| 18:21.22 | jeffspeff | If 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? |
| 18:27.52 | *** join/#asterisk Tom-M (~Tom-M@541CB5AA.cm-5-5c.dynamic.ziggo.nl) |
| 18:54.42 | adoodle | has 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 |
| 18:55.27 | adoodle | oo...it might only be valid for dahdi interfaces and not SIP |
| 18:57.39 | doop | adoodle what behavior are you expecting |
| 18:57.52 | mjordan | jeffspeff: I would interpret it that way |
| 18:58.04 | adoodle | doop: ideally, i want to know if the SIP call terminated normally, or due to some exception |
| 18:59.05 | doop | such as |
| 18:59.36 | WIMPy | adoodle: It should be valid for all channeltypes. |
| 19:00.00 | adoodle | doop: well, i was catching the outbound call attempts via checking the ${DIALSTATUS} variable |
| 19:00.24 | adoodle | doop: so i'm more interested in knowing if the user hung up, or if my dialplan was incorrect and terminated the call |
| 19:01.26 | adoodle | WIMPy: that's what i thought....but let me double check to make sure i don't have a typo or something |
| 19:01.58 | jeffspeff | mjordan, 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. |
| 19:03.02 | adoodle | http://pastie.org/9942602 the cli output and relevant dialplan portion |
| 19:03.40 | mjordan | jeffspeff: I'd confirm that what you are seeing on your end is what your provider receives |
| 19:03.49 | adoodle | and 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) |
| 19:04.29 | mjordan | jeffspeff: 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 |
| 19:04.45 | mjordan | it isn't uncommon for fax stacks to be very, very optimistic about the nominal paths that they are going to follow through their states |
| 19:05.11 | jeffspeff | mjordan, 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. |
| 19:06.16 | jeffspeff | I'm trying to do all that I can to verify that it isn't an issue on my end. |
| 19:09.39 | solmsted | Hello, 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. |
| 19:09.56 | adoodle | hmmm....it's looking like that the ${HANGUPCAUSE} variable is only set after a Dial or Transfer occurs...at least so far in my testing... |
| 19:09.59 | mjordan | jeffspeff: does it look like the training succeeded prior to the CFR? |
| 19:10.22 | mjordan | solmsted: you'll need to ask the UniMRCP folks, as they wrote a different set of modules around the speech recognition |
| 19:10.45 | mjordan | speech recognition API/Engine that Asterisk typically uses |
| 19:11.24 | solmsted | mjordan, so there's no way to pass extra params using Asterisk's generic speech recognition APIs? |
| 19:11.55 | mjordan | solmsted: I don't know. |
| 19:12.12 | mjordan | solmsted: you'll need to look at the UniMRCP docs for what they wrote, and the modules they wrote that make use of their engine |
| 19:13.34 | jeffspeff | mjordan, I can show you the pcaps if you'd like. |
| 19:13.52 | mjordan | jeffspeff: ew? |
| 19:14.05 | jeffspeff | ew? |
| 19:14.16 | mjordan | jeffspeff: I'd look at the pcap and make sure that training succeeded and that we received a TCF. |
| 19:14.27 | mjordan | we should then transmit the V.21 flags, followed by a CFR. |
| 19:14.32 | mjordan | if we don't get anything back after that, they've screwed up. |
| 19:14.47 | file | I'm sorta glad I've forgotten this stuff |
| 19:14.53 | mjordan | jeffspeff: digging through T.38 pcaps is probably my least favorite activity of all time. |
| 19:14.58 | solmsted | mjordan, 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 |
| 19:15.47 | mjordan | file: There's always T.38 over datachannels |
| 19:15.53 | file | mjordan, hush |
| 19:15.58 | mjordan | file: I'm sure we'll be violating that RFC soon enough |
| 19:16.12 | file | solmsted, 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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| 19:23.44 | solmsted | file: I've been using UniMRCP with the generic APIs SpeechCreate, SpeechLoadGrammar, SpeechBackground etc |
| 19:24.08 | file | they have a module for that? interesting |
| 19:24.23 | jeffspeff | mjordan, 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 |
| 19:24.24 | jeffspeff | a 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. |
| 19:24.25 | file | I haven't seen it but the SPEECH_ENGINE dialplan function is supposed to be use to set engine specific properties |
| 19:24.35 | file | I don't know how/if they implemented it, that's in their court |
| 19:25.24 | mjordan | jeffspeff: you're doing it right. The issue is 99% on your provider's end. |
| 19:25.31 | mjordan | 99% sure, anyway |
| 19:25.50 | mjordan | granted, 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. |
| 19:25.57 | mjordan | so there may just be Network shenanigans |
| 19:26.13 | mjordan | but if they can't look at the packets and confirm or understand what is going on, I'd start shopping for a new provider |
| 19:26.36 | jeffspeff | they can't/won't provide me with or even confirm that they're getting the CFR or replying to it. |
| 19:26.50 | solmsted | file, 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 |
| 19:27.17 | file | solmsted, it's a generic opaque dialplan function which calls into the engine |
| 19:27.23 | file | the parameters it takes are up to the implementation of the engine |
| 19:27.27 | jeffspeff | they 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 |
| 19:27.47 | mjordan | yeah... |
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| 19:28.40 | file | solmsted, 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 |
| 19:29.04 | jeffspeff | I 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 |
| 19:33.15 | solmsted | file, 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 |
| 19:34.13 | file | ah, an older page |
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| 19:48.12 | nny | Hey 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? |
| 19:50.12 | nny | Certainly staying away from Sangoma, have a Europa 8 port FXO device and was a pain to deal with |
| 19:50.33 | nny | i use sangoma cards, but the Europa manufacturer they bought isn't really the same thing at all |
| 19:54.48 | [TK]D-Fender | nny: Mediatrix became Media5. AudioCodes is another big player in the same range |
| 19:55.02 | nny | [TK]D-Fender: sounds good. I am looking around now. |
| 19:55.13 | [TK]D-Fender | nny: My experiece with Mediatrix 24-port FXS back ~2007 was good. Those weere indeed very simply to configure |
| 19:55.23 | nny | Yeah the Vega OTOH oh boy. |
| 19:56.17 | nny | [TK]D-Fender: I'll shop around. Thank you |
| 19:56.34 | [TK]D-Fender | nny: You're welcome |
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| 21:29.01 | [TK]D-Fender | checkout time, BBIAB |
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| 21:49.39 | adoodle | are there any troubleshooting tips for debugging odbc cdr's? my cdr's won't insert and i cant' see why it's failing.... |
| 22:02.54 | malachi_constant | http://pastebin.com/AQ5cxxPE <-- should this stuff work? Asterisk seems not to add the header. |
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| 22:24.24 | malachi_constant | OK 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. |
| 22:27.32 | malachi_constant | Is there a way to take the data in a header and use it in the dialplan? |
| 22:28.53 | newtonr | malachi_constant, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER |
| 22:31.14 | malachi_constant | newtonr: VERY cool! |
| 22:31.21 | malachi_constant | Thank you! |
| 22:32.11 | newtonr | Also, a lot of information in headers is already extracted into various channel variables and other places |
| 22:32.15 | newtonr | for example: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL |
| 22:32.36 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables |
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| 22:38.20 | malachi_constant | newtonr: That's awesome. ${CALLERID(num)} was exactly what I was looking for. Thanks! |
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| 22:40.54 | newtonr | malachi_constant, Cool, you are welcome |
| 22:41.39 | newtonr | malachi_constant, you may be interested in this whole page: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
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| 22:42.57 | malachi_constant | Wow. Yes, I am. :-) |
| 22:52.47 | *** join/#asterisk mafairnet (~mafairnet@187.188.176.204) |
| 22:54.23 | mafairnet | Hello, how can I use one extension in multiple devices (not at the same time). I'm using AsteriskNow (FreePBX + Asterisk 11). |
| 22:54.58 | MaliutaLap | Queue? |
| 22:55.27 | mafairnet | nope |
| 22:57.26 | *** join/#asterisk rexwin_ (~rexwin@106.208.121.141) |
| 22:57.41 | rexwin_ | 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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| 22:57.43 | mafairnet | I want that a user have one extension and can use it on multiple phones (Not at the ssame time). |
| 22:57.45 | MaliutaLap | Not entirely sure you have actually described the requirement in sufficient detail ... multiple end points one extension? but only one end point at a time? |
| 22:58.21 | rexwin_ | your call cannot be completed, please check the number and dial again |
| 22:58.27 | MaliutaLap | ASTDB+login+thing that logs out previous instance? |
| 22:59.02 | MaliutaLap | rexwin_: pre-pend, or append? |
| 22:59.39 | rexwin_ | prepend 1 works but direct did doesnot |
| 22:59.52 | MaliutaLap | rexwin_: check the phone's dialplan |
| 23:00.29 | MaliutaLap | rexwin_: or ignorepat |
| 23:01.01 | mafairnet | Have yu ever used User/Device mode? |
| 23:01.31 | MaliutaLap | rexwin_: 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 |
| 23:01.44 | MaliutaLap | mafairnet: not personally |
| 23:02.07 | rexwin_ | no, i dont have phone specific dialplan |
| 23:02.38 | rexwin_ | all other did works without prepending 1 except for one did number, all are similar in function |
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| 23:08.04 | mafairnet | :/ |
| 23:09.10 | mafairnet | have someone tried blocking transfers from one extension to the same queue that it belongs? |
| 23:10.24 | MaliutaLap | hah? 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) |
| 23:14.49 | 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 |
| 23:15.13 | tom_w_______ | I have googled and not found much of a response. Can someone point me in the right direction |
| 23:18.38 | MaliutaLap | tom_w_______: is there an upgrade of * versions? |
| 23:18.46 | MaliutaLap | tom_w_______: and from what to what? |
| 23:20.00 | tom_w_______ | 1.12 to 13.2.0 |
| 23:20.14 | [TK]D-Fender | 1.12 is not an * version |
| 23:21.14 | tom_w_______ | bad typing on my part 1.13 to 1.13.2 |
| 23:21.49 | tom_w_______ | i did go from centos 6 to 7 |
| 23:22.54 | MaliutaLap | that doesn't sound like enough of a version change for a dist upgrade like that |
| 23:23.11 | MaliutaLap | sure you're not running 13 now? |
| 23:23.24 | tom_w_______ | I am running 13.2 now |
| 23:23.33 | tom_w_______ | compiled from source |
| 23:23.39 | MaliutaLap | and previously you were running? |
| 23:24.02 | tom_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 |
| 23:24.36 | tom_w_______ | no ipv6 on he server. |
| 23:24.43 | tom_w_______ | I can verify and make sure |
| 23:24.53 | MaliutaLap | tom_w_______: and did you bring the configs up to spec? there is a big change between those versions of * |
| 23:26.16 | tom_w_______ | yes. very simple *. just 1 sip peer who sends to 8 t1's. everything else is from samples |
| 23:27.29 | [TK]D-Fender | tom_w_______, place another call with SIP DEBUG enabled |
| 23:27.36 | tom_w_______ | ok |
| 23:27.36 | [TK]D-Fender | "sip set debug on" |
| 23:27.37 | [TK]D-Fender | ~pb |
| 23:27.38 | 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. |
| 23:27.40 | [TK]D-Fender | ^^^^ |
| 23:36.32 | *** join/#asterisk Cynagen (~cynagen@ip72-208-223-60.ph.ph.cox.net) |
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| 23:43.33 | MaliutaLap | notes the silence |
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