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| 01:29.30 | aandrew | howdy howdy everyone |
| 01:30.42 | aandrew | Moving from a 1.8.22 system to a 13.x system my SIP config seems to no longer allow connections through NAT (asterisk server is not natted, but my sip phone is) - calls set up fine but no audio which is classic nat |
| 01:30.53 | aandrew | nat=rport or rport,comedia don't seem to help |
| 01:30.59 | aandrew | just dead audio once the bridge is up |
| 01:32.06 | aandrew | directmedia setting doesn't seem to help either |
| 01:54.37 | aandrew | I think I got it now... directmedia=nonat and nat=comedia (auto_comedia wasn't working) |
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| 04:33.48 | Jymmm | WIMPy: It's weird, I've been searching for GSM-FXS devices, and just can't find any domestically in North AMerica except carrier locked ones, that's really strange. FCC type acceptance (or lack there of) maybe? |
| 04:36.13 | Jymmm | Huawei B932, Super 3G Router, Simado GFX11-v5, Solwise 3G Router NET-3G-434T, ZTE WF721 (ATT), etc |
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| 06:55.57 | Jymmm | WIMPy: Ok, I think I'm getting close http://bailing88.com/products/view_36.html |
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| 10:37.11 | WIMPy | My chinese isn't that good, but that looks nice. |
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| 14:08.11 | hxczero | Hi i'm looking for some support using Asterisk as a "switch". I would like to share my sip channel between 3 different pbx. Some friends to me that it was possible, i just want know where should I start looking for some answer. I had been read about that for about a month now. thanks |
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| 14:12.38 | jrun | what are these analogue pci/pci-e modules? |
| 14:12.46 | jrun | do they replace ATA? |
| 14:18.23 | jrun | i've found from digitum and atcom with huge price difference! |
| 14:19.08 | jrun | i'm trying to get a 6-unit building (i'm a tenent here) hooked-up to voip.ms |
| 14:20.07 | jrun | we're placing an order for an amd64 server (only pci-e mb) so i was wondering if i could just use the server with analogue modules instead of using an ATA? |
| 14:21.08 | jrun | the old telephone copper wirering is still in place and i have access to the phone closet (or whatever it's called) |
| 14:22.00 | jrun | so all phone wires from units come into a single closet in the basement and i have space there to put my server. |
| 14:23.18 | jrun | it would be nice to just hook them all up to the server, run asterisk on it and talk to voip.ms's server via sip or IAX2. |
| 14:23.46 | jrun | well that's the plan but this is my first time doing this so i'm wondering how far i've got it right? |
| 14:24.26 | zekoZeko | i suppose you'll be better off with some external ATA |
| 14:24.31 | zekoZeko | and a small server |
| 14:25.01 | zekoZeko | which could even probably be a raspberry pi or someting similar if you take care to make sure you don't need transcoding on it. |
| 14:25.14 | zekoZeko | that's what I would do anyway |
| 14:25.34 | zekoZeko | running a normal server will cost you hundreds/year in electricity |
| 14:26.01 | zekoZeko | and with something embedded/cheap you can easily have 2 onsite for reliability |
| 14:27.03 | jrun | the landlord owns a resturant that's right underneath the building. server is gonna do lots of things including multimedia streaming |
| 14:27.14 | zekoZeko | that's another thing then :) |
| 14:27.24 | zekoZeko | if it was only for telephony it wouldn't make sense |
| 14:27.36 | zekoZeko | still, use a physical ATA |
| 14:28.02 | jrun | i thought about doing it with embedded stuff but we might need video streaming of camera (doorbell/security) as well and they don't seem to have enought power for video encoding |
| 14:28.31 | zekoZeko | you have analog cameras? |
| 14:28.33 | jrun | oh you would recommend a seperate ATA over the modules? |
| 14:29.00 | jrun | well, they took over a bankrupt tenent and he had analogue cameras |
| 14:29.16 | jrun | but i want to put a proper one for the doorbell/intercomm |
| 14:29.30 | zekoZeko | i don't know what's your budget, but i *think* it might be cheaper to replace all cameras with IP ones |
| 14:29.37 | jrun | hopefully using sip to push HD to units |
| 14:29.40 | zekoZeko | good quality ones are like $100/pop |
| 14:29.55 | zekoZeko | then you don't need any encoding |
| 14:30.25 | jrun | yeah, that's the plan actually... oh, camera will do it themselves? |
| 14:30.29 | zekoZeko | sure |
| 14:30.36 | jrun | we're sending all of these over gigabit ethernet btw |
| 14:30.49 | zekoZeko | i believe that's called "overprovisioning" :) |
| 14:30.50 | jrun | i wanna get rid of old coaxial cables they have |
| 14:31.09 | zekoZeko | yeah well, you've got a nice little project there to do. |
| 14:31.17 | zekoZeko | gotta go, good luck |
| 14:31.29 | jrun | you said $100, i was gonna put one together for 50-60 using raspberry-pi :) |
| 14:31.41 | zekoZeko | just.... don't |
| 14:31.47 | zekoZeko | i mean for home use that's ok |
| 14:31.51 | zekoZeko | but for business? |
| 14:32.18 | zekoZeko | i mean for cameras, they have to look somewhat ok |
| 14:32.24 | jrun | i see |
| 14:32.50 | jrun | i know you have to go but any recommendation on the ATA? |
| 14:32.57 | jrun | i have found these two: |
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| 14:32.58 | zekoZeko | i'm partial to ubnt.com myself, chinese ones have lousy software on cameras and recorders... |
| 14:33.06 | zekoZeko | that was for cameras |
| 14:33.13 | jrun | i have ubnt, ER-Pro 8 |
| 14:33.20 | zekoZeko | can't tell you much about ATAs, I'm using 2 atcom ones |
| 14:33.21 | jrun | oh, for camera |
| 14:33.33 | zekoZeko | quite happy |
| 14:33.50 | jrun | anyone with grandstream experience here? |
| 14:33.56 | zekoZeko | me |
| 14:34.00 | zekoZeko | you were asking about 2140? |
| 14:34.02 | jrun | i've been looking into their modules |
| 14:34.09 | zekoZeko | i hae a bunch of 2130, which are the same basically |
| 14:34.14 | jrun | zekoZeko: wait, let me get the links |
| 14:34.50 | zekoZeko | no time now |
| 14:35.45 | jrun | i will stick around, just have a look at these if you please: |
| 14:35.50 | jrun | http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht702_704 |
| 14:35.51 | jrun | http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-analog-gateways/gxw410x |
| 14:36.06 | jrun | i don't understand what i would get with gxw410x that makes it so expensive? |
| 14:36.09 | jrun | just the internal router? |
| 14:36.51 | zekoZeko | one is fxs, other is fxo |
| 14:37.00 | zekoZeko | not comparable really |
| 14:37.02 | jrun | i can do away with 4 FXS for now btw. 2 of the units are empty and 3 can use androind's sip stack |
| 14:37.18 | jrun | yeah, i don't understand that. why would one need fxo on ATA? |
| 14:37.30 | jrun | fxo is the phone plug on the old phone devices right? |
| 14:37.57 | zekoZeko | you use ti to connect your VoIP network to PSTN |
| 14:38.05 | jrun | and fxs is the *end or the wire* from phone service provider? |
| 14:38.14 | zekoZeko | that you plug your analog phone into, yes |
| 14:38.26 | zekoZeko | yon could very well plug one into another |
| 14:38.32 | zekoZeko | but that would make no sense :) |
| 14:38.55 | jrun | so for my case i can be happy with fxs? ht704? |
| 14:38.59 | zekoZeko | i guess |
| 14:39.26 | zekoZeko | you put it in the wiring closet, connect the phone lines to the units to it and that's it. |
| 14:39.30 | jrun | just give a single use case for fxo. if that's what gxw410x does |
| 14:39.44 | zekoZeko | well, you have an office with N IP phones |
| 14:39.54 | zekoZeko | but your outside connectivity is pstn |
| 14:40.13 | jrun | oh, i see. that's what they use at my university i guess :) |
| 14:40.58 | jrun | my uplink is going to be voip.ms (they run astersik) and they hook me up to pstn, as far as i understand it. |
| 14:41.09 | jrun | it's not going to be, it is atm. |
| 14:41.27 | zekoZeko | they don't hook you to pstn |
| 14:41.29 | jrun | so there is no benefit to me running asterisk i guess |
| 14:41.36 | zekoZeko | they hook you to the telephone network |
| 14:41.49 | jrun | oh, i thought those were the same |
| 14:42.00 | zekoZeko | pstn is just the last leg to the customer these days. |
| 14:42.03 | jrun | oh, which may or may not be pstn |
| 14:42.40 | jrun | other may well be connected to another voip provider and then we're all just using ip (the internet) |
| 14:42.45 | jrun | *other end |
| 14:42.50 | zekoZeko | i remember the days when there were overhead lines from my town to the nearest big one, that was PSTN too |
| 14:42.54 | jrun | do i have it right? |
| 14:43.02 | zekoZeko | but that was the eighties in the eastern europe :) |
| 14:43.11 | zekoZeko | you do |
| 14:43.29 | zekoZeko | bye now |
| 14:43.56 | jrun | zekoZeko: thanks |
| 14:43.58 | jrun | bye |
| 14:45.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 17:43.28 | aandrew | I see pjsip is an option now; is there a compelling reason to use it over Asterisk's native stack? |
| 17:44.27 | hxczero | Hi i'm looking for some support using Asterisk as a "switch". I would like to share my sip channel between 3 different pbx. Some friends to me that it was possible, i just want know where should I start looking for some answer. I had been read about that for about a month now. thanks |
| 17:55.10 | [TK]D-Fender | hxczero, elaborate on "share a SIP channel" |
| 17:59.17 | hxczero | TK, sorry I will try to explain myself better, using english as a secondary language. Actually I have 3 sites, every site has a asterisk pbx on-site. 1st has 1 sip trunk with 3 channel, 2nd same, 3rd has 1 sip with 2 channel for a total of 10 channels. My concurrent calls is at 5 during business hours. I readed it was possible share line beetween the 3 sites, so basicly pay 5 channel instead of 10 and share them to my 3 pbx. |
| 18:00.00 | [TK]D-Fender | That depends on your provider |
| 18:00.01 | hxczero | Some people told me I needed to use asterisk "as a switch", just trying to keep searching the right information. |
| 18:00.12 | [TK]D-Fender | As to what THEY offer as billing bundles |
| 18:00.52 | hxczero | they offered me "wholesale" package, and says it was possible to "split" them on my side? |
| 18:01.08 | [TK]D-Fender | Could be |
| 18:01.44 | [TK]D-Fender | At which point you have that central server do the resgitration and handle the incoming calls and then jsut dial each other PBX to pass the calls the need to go to them |
| 18:01.50 | [TK]D-Fender | There is nothing to look up for this |
| 18:01.56 | [TK]D-Fender | "switch" isn't a thing |
| 18:01.58 | [TK]D-Fender | Everything = dialplan |
| 18:02.13 | [TK]D-Fender | The way one * would talk to another is no different that * talking to anything else |
| 18:02.56 | [TK]D-Fender | Outbound would jsut the the reverse direction |
| 18:03.17 | hxczero | So using Dialplan I would be able to handle many calls dooing to different pbx? |
| 18:03.37 | hxczero | gooing* |
| 18:03.42 | [TK]D-Fender | All call handling = dialplan |
| 18:04.25 | hxczero | Ok I will look forward on forum to get more information. |
| 18:04.44 | [TK]D-Fender | There is nothing really to look for. |
| 18:04.50 | [TK]D-Fender | You need to actually learn * |
| 18:04.52 | [TK]D-Fender | ~book |
| 18:04.52 | 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 |
| 18:04.57 | [TK]D-Fender | Theis isn;t a setting. |
| 18:05.01 | [TK]D-Fender | Dialplan = PROGRAMMING |
| 18:05.12 | [TK]D-Fender | You determine what you do with every call your system gets |
| 18:05.32 | hxczero | ty |
| 18:06.31 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-dhgzyajrgtkbngiz) |
| 18:07.49 | [TK]D-Fender | "Call comes in from your provider via SIP targeting DID # 1234567890. That call has a CALLERID attached. You'd process that however you want. Maybe you want all calls for DID 1234567890 to go to PBX #1. So dial them. Maybe you want to distribute them evenly between your systems. Keep track of the last on you sent it to and send to the next, then loop around |
| 18:08.15 | [TK]D-Fender | Maybe you want to check if PBX #2's city is raining first and only send them calls then. |
| 18:08.32 | carrar | why not sunny? |
| 18:08.34 | [TK]D-Fender | There is no such thing as a "guide" for this. Like all programming it is 100% you. |
| 18:08.54 | [TK]D-Fender | carrar, Because it's rainy and they will feel loney stuck inside, so give them company. |
| 18:08.59 | carrar | haha |
| 18:09.32 | [TK]D-Fender | carrar, If you want to drill on this, I've already started naming these hyporthetical people and started on their back-stories ;) |
| 18:09.52 | carrar | Might be better to have a external process check every 5 mins on the weather and put that in the db |
| 18:10.00 | carrar | so as not to slow the incoming call down |
| 18:11.11 | carrar | maybe have it check 2 or 3 weather API's |
| 18:11.18 | carrar | "weather" avg |
| 18:11.32 | [TK]D-Fender | cross-referencing required with historical accuracy as a ratings modifier |
| 18:11.40 | carrar | yes yes |
| 18:12.34 | [TK]D-Fender | https://www.youtube.com/watch?v=v77SF4TFUoM |
| 18:28.49 | [TK]D-Fender | heads out for a ride |
| 18:31.30 | babak | Hi, if we have 2 Asterisk with same configuration files and use DNS SRV to distribute sip phones to register on 2 Asterisks, if a calls comes to Asterisk1 but the sip phone is registered on Asterisk2 , how we can realize and route it ? |
| 18:34.55 | *** join/#asterisk Xadof (~dade@S0106d050995743c0.vc.shawcable.net) |
| 18:37.00 | carrar | use dundi? |
| 18:39.37 | WIMPy | Or be lazy and just dial doth. |
| 18:39.39 | WIMPy | both |
| 18:42.07 | Xadof | Hi guys, I just noticed that most or all of my DTLS related crashes on 11 branch are preceded by a debug message anywhere from 2 seconds to a minute earlier, on a specific line in res_rtp_asterisk.c:dtls_srtp_check_pending (AST_SCHED_DEL_UNREF outputs "Unable to cancel schedule ID nnnnnn). I'm wondering if anybody here familiar with the asterisk code can tell me how to either 1) increase |
| 18:42.07 | Xadof | debug level and verbosity on that line and/or 2) trigger a custom AMI event (ie "ImminentCrashDetected") on that line so that our scripts that read that event can begin to prepare (phase out active calls, stop new calls) |
| 18:43.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:43.05 | Xadof | unfortunately I spent two hours writing a new issue for this (since it's more of a parent to all the other DTLS issues already open rather than a duplicate or extension of one in particular) and then I must have failed to actually submit it :s |
| 18:44.02 | Xadof | ah nevermind about the missing issue, my filters blocked me from finding it. ISSUE-25127 |
| 18:45.25 | babak | carrar: there should be a way that we can check which asterisk registered sip phone |
| 18:45.37 | WIMPy | It's uncool, but if you have a message to work on, you could always parse the log. |
| 18:46.44 | WIMPy | babak: Before I try to answer the wrong question, can you re-phrase that one? |
| 18:47.14 | carrar | babak: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-5 |
| 18:50.35 | babak | WIMPy: there is 2 Aaterisk with Unison for bidirectional sync of configuratioan files I want to use DNS SRV for loadbalancing... |
| 18:51.15 | WIMPy | Yes, got that. Just the last question. |
| 18:53.26 | babak | the problem is may be som sip phones register on first Asterisk but calls comes to second server for that sip phone , how to relize and route this call to Asterisk1 ? without DUNDI |
| 18:54.10 | WIMPy | The lazy version is to just call it on both instances. |
| 18:54.40 | carrar | he should then get voicemail on the server without a reg? no? |
| 18:55.09 | babak | asteisk saves sip registered IPs on internal sqlite ? |
| 18:55.37 | WIMPy | You need a context for the cross-connect anyw. So just don't do anything there except for calling the device. |
| 18:55.51 | WIMPy | yes |
| 18:56.32 | carrar | could use realtime and figure out where phone is before sending |
| 18:56.46 | carrar | keeping the state in a db |
| 18:56.58 | WIMPy | Or a registercontext and DUNDi. |
| 18:57.05 | carrar | thats too simple!! |
| 18:57.29 | WIMPy | More complicated than just blindly trying both. |
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| 18:57.54 | carrar | babak: Why are you not interested in using dundi to do this? |
| 18:59.29 | carrar | babak: You can also google "Using DUNDi with a Cluster" |
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| 18:59.47 | carrar | lots of info out there |
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| 19:01.13 | babak | carrar: not sure , wants to hold it simple as possible |
| 19:02.09 | carrar | your requirement is "not simple" |
| 19:02.19 | carrar | so the solution matches |
| 19:02.36 | carrar | (really is simple) |
| 19:02.39 | WIMPy | thinks it's quite simple. |
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| 19:24.38 | babak | PJSIP_ENDPOINT() ? |
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| 20:07.33 | kasanop | Xadof: I think building Asterisk with DUMP_SCHEDULER may help gather a little more information |
| 20:18.40 | madduck | how do i create an extension pattern that matches 1+ digits without using ! (due to atxfer)? |
| 20:19.21 | madduck | _X. won't match '1', and with _X!, it'll be impossible to enter a longer extension during atxfer |
| 20:20.56 | WIMPy | Either _. which shouldn't be used without extreme caution, or _X! and make suere you have anoter extension that could also match. |
| 20:25.51 | madduck | WIMPy: what do you mean with "another extension that could also match"? just anything, or an exten like _XXXXXXXXXXXXWILLNEVERMATCH |
| 20:26.02 | madduck | just to keep asterisk on its toes? |
| 20:26.37 | WIMPy | Exactely. ! matches as soon as it's the ONLY possible match. |
| 20:27.04 | madduck | ok thank you |
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| 20:27.36 | madduck | there's another weirdity here⦠I have Background() play back a file followed by a simple Read(VAR) |
| 20:27.48 | madduck | if I wait for the playback to end and enter 123#, VAR will be 123 |
| 20:28.05 | madduck | if I press '1' during playback, it stops, and eventually, VAR will be 123# :/ |
| 20:28.17 | WIMPy | o.O |
| 20:28.48 | madduck | http://slexy.org/raw/s2s489tVg0 |
| 20:28.49 | WIMPy | Is there a WaitExten as well? |
| 20:29.20 | madduck | no |
| 20:29.23 | madduck | here is the code: http://slexy.org/raw/s21lbVqADj |
| 20:29.56 | WIMPy | Why do you use Background there? |
| 20:30.05 | WIMPy | Read() can also playback. |
| 20:30.51 | madduck | right |
| 20:31.18 | madduck | ftr, if I use WaitExten(), then even after playback, entering 123# causes extensin 123# to be dialed |
| 20:31.53 | WIMPy | Yes. That will take everything literally. |
| 20:32.27 | madduck | ah ok |
| 20:32.28 | madduck | <PROTECTED> |
| 20:32.32 | madduck | this works fine |
| 20:34.37 | WIMPy | I don't think, Background() make sense with anythign other than WaitExten(). |
| 20:35.12 | madduck | ok thank you for hand-holding ;) |
| 20:55.40 | aandrew | is there any compelling reason to choose the pjsip channel driver over the native asterisk one? |
| 20:58.01 | WIMPy | New features? |
| 21:08.17 | aandrew | so not really then. :-) |
| 21:13.55 | Xadof | Don't be like me. Use pjsip. I'm virtually locked in to chan_sip off of bad design decisions in the past, written so much crap that's looking for "SIP/" in channel names & for dialing. I regret the heck out of it.. |
| 21:15.22 | Xadof | there's some compelling reasons you can find online pretty easily |
| 21:25.18 | aandrew | interesting |
| 21:25.42 | aandrew | I used to hack on asterisk (old irc name tzanger) and I am coming from a 1.8.x setup so I'm fairly behind the curve |
| 21:25.54 | aandrew | in some ways I think it's worse than coming into this fresh because I've got to unlearn a lot of things |
| 21:26.27 | WIMPy | ConfBride, pjsip, what else? |
| 21:26.30 | [TK]D-Fender | not much.... |
| 21:26.39 | [TK]D-Fender | More like add a pile of newer things |
| 21:28.13 | aandrew | yeah the conf bridge configuration is completely new, voicemail has some minor changes, there's the entire stasis message bus which looks cool as hell... |
| 21:28.22 | aandrew | xmpp looks like it's been rewritten as well |
| 21:29.10 | aandrew | sip natting seems to have been overhauled, * server is not behind NAT but 99% of my devices are and nat=comedia was the only thing that made them better again... the auto_ options didn't work at all |
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