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| 04:58.48 | jkroon | hi guys, if got a BRI (ISDN) call quality issue that I think may be related to jitter buffers. |
| 04:59.19 | jkroon | this is with asterisk 11.5.1, and I know the JB handling changed from 1.8 => 11 |
| 05:00.13 | jkroon | I don't currently use the JITTERBUFFER() function in dialplan, but have jbenable=yes; jbimpl=fixed and jbmaxsize=10 in chan_dahdi.conf |
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| 05:08.02 | [TK]D-Fender | DAHDI doesn't have jitter |
| 05:08.11 | [TK]D-Fender | DAHDI is TDM, not packet-based |
| 05:08.17 | [TK]D-Fender | there is no such thing |
| 05:08.30 | jkroon | ah yes, but it can't accept jitter |
| 05:08.57 | jkroon | therefor (at least in 1.8, and according to the code for 11.5.0 too) there is a jitter buffer on the tx side |
| 05:09.05 | [TK]D-Fender | no |
| 05:09.17 | jkroon | ok ... ? |
| 05:09.23 | [TK]D-Fender | jitter only happens on packet-based comms |
| 05:09.33 | [TK]D-Fender | that part of the call has no such thing at all |
| 05:10.04 | jkroon | yes, i know, so let's say we have RTP(SIP/whatever else uses RTP) as the other side, we need a JB going from RTP => DAHDI |
| 05:11.54 | [TK]D-Fender | yes, RTP would need it for that side |
| 05:12.23 | jkroon | ok, so you're telling me the jb* options in chan_dahdi doens't do what I think it does? |
| 05:12.35 | [TK]D-Fender | Should be no such thing. |
| 05:13.04 | jkroon | ok, now I need to pick between what you're saying and what the code is saying ... ? |
| 05:16.17 | jkroon | [TK]D-Fender, line 17451 of chan_dahdi.c calls ast_jb_read_conf which sets the options |
| 05:20.15 | jkroon | are you telling me that in spite of being set they get ignored? |
| 05:21.14 | jkroon | ast_jb_configure is also being called from dahdi_new ... ? |
| 05:23.13 | [TK]D-Fender | Sounds like it's just to help patch whatever the other channel type is you might bridge it with... |
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| 05:25.56 | jkroon | yea, well, i know enough to read the code, but I'll be honest, I don't understand all of the details, in jb_choose_impl there seems to be code that checks both directions regarding whether it can generate jitter, and whether it can accept jitter and installs jitter buffers as required ... but that's not what's been said before, nor does that match my experience |
| 05:26.43 | jkroon | but as hans reiser said ... the code never lies (our understanding of it is just wrong sometimes) |
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| 08:10.26 | BarthezZ | I just read https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching but does this differ with multiple included context? I has for example context [a] with exten _*1X. and context [b] with exten _*11X. |
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| 08:11.31 | BarthezZ | the pattern matching page would suggest *11X to get prescedence... but in practice it goes to *1X, when dialing 1111, even though context [b] is include "first" (both with one include between them) |
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| 08:15.30 | matjam | hey all, is it possible to play some tones when the other end of a call hangs up? |
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| 09:55.54 | hos7ein | hi |
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| 09:58.35 | hos7ein | i have tow server (server A & server B) |
| 09:59.10 | hos7ein | for connect this tow server ,create SIP trunk |
| 09:59.29 | hos7ein | trunk on server A : http://paste.fedoraproject.org/53412/50220138/ |
| 09:59.51 | hos7ein | trunk on server B : http://paste.fedoraproject.org/53413/42502601/ |
| 10:01.24 | hos7ein | all extensions on tow server have from-internal context |
| 10:01.49 | hos7ein | so now,what is the after step? |
| 10:02.03 | hos7ein | * next step? |
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| 10:04.29 | hos7ein | What to do in connection extensions on tow server? |
| 10:08.30 | kaldemar | 2 is two, not tow |
| 10:09.39 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291281 |
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| 12:02.38 | gen1us2k | Hello guys |
| 12:02.51 | gen1us2k | Have some trouble with inband calls |
| 12:03.40 | gen1us2k | Is there solution for https://code.google.com/p/webrtc/issues/detail?id=2347 |
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| 12:04.49 | Greenlight | One way audio is usually NAT related. |
| 12:05.16 | Greenlight | There seems to be a lot of other software involved though, and nothing clearly point to an asterisk related issue |
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| 12:10.12 | gen1us2k | So. This is NAT issue? |
| 12:10.35 | Greenlight | No idea, show us a call effected by the issue |
| 12:10.56 | gen1us2k | console log? |
| 12:11.00 | Greenlight | And sip trace |
| 12:11.09 | gen1us2k | wait |
| 12:11.33 | Greenlight | Also, explain the actual setup in more detail. |
| 12:13.29 | *** join/#asterisk darkalien (59923d90@gateway/web/freenode/ip.89.146.61.144) |
| 12:13.32 | darkalien | hi all |
| 12:14.05 | darkalien | any one got a spare sec to help out a newby :) |
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| 12:16.36 | darkalien | hi there vlad |
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| 12:16.50 | darkalien | oeh it is getting crowded?:) |
| 12:17.05 | darkalien | or automatic logons? |
| 12:17.37 | Greenlight | darkalien: Ask away ... |
| 12:17.50 | darkalien | thanks:) |
| 12:18.17 | darkalien | i am setting up a Kerio opperator |
| 12:18.35 | Greenlight | What's a "kerio operator" ? |
| 12:18.53 | darkalien | its where my soft/hard phone connect to |
| 12:19.01 | darkalien | its a local voipserver? |
| 12:19.08 | Greenlight | Okay |
| 12:19.22 | darkalien | and it uses asterisk to connect to my voip provider |
| 12:19.52 | Greenlight | So, Kerio -> Asterisk -> ITSP ? |
| 12:19.55 | Greenlight | Why does it do that ? |
| 12:20.13 | darkalien | kerio is probbably more of a webbased shell around asterisk |
| 12:20.25 | darkalien | to make it user frendly? |
| 12:20.27 | Greenlight | Ahh okay |
| 12:20.39 | darkalien | but kerio wont allow me to change what i need to change:) |
| 12:20.44 | darkalien | so i am back to asterisk |
| 12:20.51 | darkalien | wich is wy i am here |
| 12:21.16 | darkalien | after 10 hours of digging and trying i finaly got the register part fixed, |
| 12:21.26 | darkalien | now i am stuck on the outgoign call invites |
| 12:21.29 | Greenlight | I suspect that Kerio likely acts in a similar way to FreePBX, which makes it difficult to help with |
| 12:21.43 | darkalien | difficult in wich manner? |
| 12:21.49 | Greenlight | ~freepbx |
| 12:21.49 | 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 |
| 12:22.46 | darkalien | from what i see the problem is mainly how it sends its invites |
| 12:22.59 | Greenlight | So, what problem are you getting ? |
| 12:23.06 | darkalien | username@localip wich should be username@domain, |
| 12:23.21 | darkalien | when i try to call i get service unavailible |
| 12:23.29 | darkalien | and wireshark showed me the username@localip |
| 12:23.35 | darkalien | or username@proxyip |
| 12:23.50 | darkalien | but my voip provider wants username@domain |
| 12:24.18 | Greenlight | I think you can specify a "fromdomain" in sip.conf |
| 12:25.28 | darkalien | i have |
| 12:27.25 | darkalien | i tryed fromdomain=domainname |
| 12:27.39 | darkalien | still it kept sending localip adres |
| 12:27.42 | darkalien | or ext ipadres |
| 12:52.55 | *** join/#asterisk saxa (~sasa@84-255-205-164.static.t-2.net) |
| 12:53.56 | saxa | hi, I have a small problem, I bought a TE133F from Digium, but the modem I received from digivoice seems to work only on the coaxial cable connectors |
| 12:54.31 | saxa | it have one RJ45 port but it seems is not possible to configure it to work with a so called Digitronco product. |
| 12:54.43 | WIMPy | Then you need to get a balun. |
| 12:54.44 | saxa | does exists out there some kind of adapters I could use ? |
| 12:54.56 | saxa | hi WIMPy |
| 12:55.02 | saxa | what is a balun ? |
| 12:55.24 | WIMPy | The adaptor you need. |
| 12:55.46 | WIMPy | You can get them for a few quid on dx. |
| 12:55.47 | saxa | oh ok I see images on google |
| 12:56.03 | saxa | any site selling that ? |
| 12:56.24 | saxa | let me check if I find somebody here in the salvador area selling this stuff. |
| 12:57.20 | saxa | maybe i can do one myself :) |
| 12:57.53 | saxa | another question is, this will be probably completely transparent to the configuration I will use correct ? |
| 12:58.17 | saxa | and I get the info that it works on G.703 |
| 13:03.14 | cusco | howdy |
| 13:03.48 | cusco | how do I know what echo canceller modules are compatiple with a PRI card? |
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| 13:14.55 | WIMPy | saxa: yes |
| 13:15.27 | WIMPy | cusco: I don;t think there are any where you have a choice. |
| 13:15.46 | cusco | WIMPy: so a echo canceller will work on all digium cards? |
| 13:16.04 | cusco | I'm looking at http://www.voipango.com/en/ISDN-Interfaces/Digium/TDM-AEX-HX8-Modules/Digium-VPMADT032-EC-Modul.html?listtype=search&searchparam=VPMADT032 |
| 13:16.19 | WIMPy | I meant that there will only be one that works. |
| 13:16.20 | cusco | VPMADT032 |
| 13:16.26 | cusco | per card? |
| 13:16.35 | cusco | so each module for a specified card |
| 13:16.58 | saxa | WIMPy: thx |
| 13:17.14 | WIMPy | Call them. I'm sure they will tell you what options you have. |
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| 13:18.11 | ghost75 | hmmm if minrate 2400 is set in res_fax.conf then v29 cannot be used because it is incompatible |
| 13:19.25 | ghost75 | but 2400 is default as minrate and default modem order is v17,v27,v29 |
| 13:22.37 | ghost75 | [Nov 12 14:11:38] ERROR[8432]: res_fax.c:2086 sendfax_exec: 'modems' setting 'V29' is incompatible with 'minrate' setting 2400 |
| 13:32.07 | cusco | so.. if lspci shows TE410P/TE412P - how do I know which one this card is? |
| 13:32.26 | cusco | WIMPy: I did call them heh, they asked me to send a e-mail :/ |
| 13:33.00 | WIMPy | Open the case, pull out the card, look at it. |
| 13:33.25 | cusco | its in a production environment, I was looking for antoher way? |
| 13:33.27 | cusco | ow well |
| 13:33.30 | cusco | nvm |
| 13:35.44 | [TK]D-Fender | cusco: TE412P has HWEC already, and the TE410P is an ancient card and does not support any |
| 13:36.28 | cusco | so I'm sure it is the ancient one :P |
| 13:36.38 | cusco | and TE410P/TE412P ? |
| 13:36.41 | cusco | oops |
| 13:36.47 | cusco | TE210P/TE212P |
| 13:37.04 | cusco | probably 212 already has H EC as well? |
| 13:38.00 | cusco | I see that the module VPMOCT64 is comaptible with TE210P hm... |
| 13:38.34 | cusco | we also have a TE420P - should be more up to date |
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| 13:50.51 | cusco | and.... |
| 13:51.10 | cusco | I seem to forget, but I have to recompile dahdi or asterisk, after installing libpri? |
| 13:51.14 | cusco | or both? |
| 13:51.33 | WIMPy | Asterisk is last. |
| 13:51.39 | cusco | I see |
| 13:51.44 | cusco | that is my issue then :)= |
| 13:52.07 | WIMPy | I can't remember what order libpri and dahdi need now. |
| 13:52.31 | WIMPy | It used to not matter, but now it does. |
| 13:55.49 | [TK]D-Fender | libpri, DAHDI, asterisk |
| 13:58.14 | cusco | I understand, libpri is probably required by dahdi |
| 13:59.13 | cusco | required... as in used |
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| 14:16.45 | *** join/#asterisk Bkhan (ca7d90e5@gateway/web/freenode/ip.202.125.144.229) |
| 14:16.51 | Bkhan | Hi |
| 14:17.35 | Bkhan | I am using sipaddheader function in macro |
| 14:17.51 | Bkhan | but value is not assining. How can we do it |
| 14:18.05 | Greenlight | Lets see the dialplan |
| 14:19.04 | [TK]D-Fender | That isn't a function |
| 14:19.16 | Bkhan | let me paste caod |
| 14:19.22 | [TK]D-Fender | SIPAddHeader is a dialplan APPLICATION |
| 14:22.43 | Bkhan | exten => _X.,n,SIPAddHeader(X-GPCallID: ${ACDCallID}) |
| 14:22.53 | [TK]D-Fender | Bkhan: show us the call. |
| 14:23.06 | [TK]D-Fender | Bkhan: not just some randome line of supposed dialplan code |
| 14:23.09 | [TK]D-Fender | ~pb |
| 14:23.09 | 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. |
| 14:23.10 | [TK]D-Fender | ^^^^^ |
| 14:23.35 | Bkhan | it is working. But when we use this command in marco we do not set it |
| 14:23.46 | Bkhan | ok. Let me paste in macro |
| 14:24.00 | [TK]D-Fender | no. |
| 14:24.01 | Bkhan | paste in pastebin |
| 14:24.02 | [TK]D-Fender | Show us the call |
| 14:24.13 | [TK]D-Fender | We need to see it being called and then failing to work |
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| 14:27.37 | Bkhan | http://pastebin.ca/2475613 |
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| 14:30.39 | Greenlight | You have to set the header *before* Dial |
| 14:31.23 | Greenlight | And, where is the call ? |
| 14:32.22 | Greenlight | Also, the macro doesn't have access to that variable, I think it's out of scope |
| 14:32.49 | [TK]D-Fender | Next, it isn't a VARIABLE |
| 14:32.59 | [TK]D-Fender | ten => s,1,SIPAddHeader(X-GPCallID: ${ACDCallID}) |
| 14:33.05 | [TK]D-Fender | this does not read it back |
| 14:33.41 | [TK]D-Fender | M(outMarco,XGPCallID) <- I see no macro context named [macro-outMarco] anywhere either |
| 14:33.51 | [TK]D-Fender | And you are not paying attention to spelling anywhere either |
| 14:37.33 | [TK]D-Fender | Line 9 is the only one calling the SIPAddHeader application that I can see... and you haven't shown it failing to work. |
| 14:41.52 | Bkhan | TK]D-Fender: then how can we do it |
| 14:42.18 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 14:43.15 | Bkhan | <[TK]D-Fender>: Actually scenario was we have an asterisk server and it goes to forwarder ( which is also on asterisk) and then it goes to our provider |
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| 14:45.13 | Bkhan | what is going now when call come to our server to forwarder and forwarder sends it to provider |
| 14:45.28 | [TK]D-Fender | Bkhan: You have what looks like misnamed macro's and an inablility to show us the FAILURE |
| 14:45.37 | [TK]D-Fender | Bkhan: Where is the FAILED CALL? |
| 14:45.40 | Bkhan | using forwarder's UNIQUE ID |
| 14:45.44 | bulkorok | <PROTECTED> |
| 14:47.21 | Bkhan | we want to send provider the UNIQUE ID of machine from where call orginating not of forwarder |
| 14:48.06 | [TK]D-Fender | Bkhan: Show us the FAILED CALL |
| 14:48.10 | [TK]D-Fender | Bkhan: I don't see the problem. |
| 14:48.22 | [TK]D-Fender | Bkhan: Where is the proof that it is not doing what it's supposed to be doing? |
| 14:48.35 | Bkhan | Call is not failing |
| 14:48.52 | [TK]D-Fender | Bkhan: Where do we see SIPAddHeader NOT doing its job? |
| 14:49.00 | [TK]D-Fender | Bkhan: that would be considered a failure |
| 14:49.11 | [TK]D-Fender | http://www.lamatel.combkYou came here asking "why doesn't it work?" |
| 14:49.22 | [TK]D-Fender | Bkhan: Well we don't SEE it not doing its job]' |
| 14:49.31 | [TK]D-Fender | oops on the link... |
| 14:50.04 | Bkhan | exten => s,1,SIPAddHeader(X-GPCallID: ${ACDCallID}) |
| 14:50.12 | [TK]D-Fender | Bkhan: Stop showing us code |
| 14:50.25 | [TK]D-Fender | Bkhan: Show us the DIALPLAN EXECUTION where it doesn't do its job |
| 14:50.38 | Bkhan | after it if we use NoOP command it will be blanck |
| 14:50.41 | [TK]D-Fender | SHOW US |
| 14:50.59 | Bkhan | wait after execuation will show u |
| 14:51.10 | Bkhan | asterisk CLI |
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| 15:10.54 | cusco | I'm afraid there is something wrong with our PRI card, that has been set asside for some time now... |
| 15:11.04 | cusco | dmesg shows: [Tue Nov 12 12:26:43 2013] hub 4-0:1.0: over-current condition on port 7 |
| 15:11.11 | cusco | :/ |
| 15:11.56 | Chainsaw | cusco: That is a USB message. |
| 15:12.08 | Chainsaw | cusco: Does your PRI card plug into a USB port? |
| 15:12.09 | cusco | ow... ok I'm feeling better for that |
| 15:12.11 | cusco | no |
| 15:12.26 | Chainsaw | cusco: Then it is unrelated. |
| 15:12.37 | cusco | its a digium single span... but telco says he's not seeing anything attached to their equipment |
| 15:12.46 | cusco | asked if we had a loop.. |
| 15:12.52 | cusco | :/ |
| 15:13.29 | cusco | is there a dahdi tool that can tell me my hardware seems to be ok? |
| 15:13.30 | cusco | lol |
| 15:14.19 | Chainsaw | cusco: It suggests a configuration issue on your end. |
| 15:14.31 | cusco | all seems configured fine |
| 15:14.34 | cusco | really |
| 15:14.39 | Chainsaw | cusco: Your telco disagrees. |
| 15:14.52 | cusco | well thei're still looking into their equipment |
| 15:15.27 | cusco | dahdi_scan output seems ok |
| 15:15.44 | [TK]D-Fender | cusco: What does ASTERISk say? |
| 15:16.05 | cusco | asterisk's pri debug state its trying to send SABME |
| 15:16.07 | cusco | and it expires |
| 15:16.11 | cusco | I can paste |
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| 15:17.37 | cusco | here: http://paste.debian.net/hidden/cd83a44d/ |
| 15:17.45 | cusco | Status: Down, Active |
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| 15:24.55 | orn | Hello. I'm trying to make test calls via asterisk using .call files. It's working fine, but as soon as the remote end answers the call the local leg seems to hang up. The dialplan instructions are Answer(), Playback(xxx), Wait(1), Hangup(). On the remote end, everything is working fine -- audio plays and so forth -- the only problem is that the CDR that gets created in Asterisk has the duration from the local leg (0 seconds) and there's no ot |
| 15:25.32 | orn | in the call file, i have Channel: Local/[REMOTE-NUMBER]@[CONTEXT] |
| 15:25.32 | [TK]D-Fender | orn: Show us.... |
| 15:25.45 | [TK]D-Fender | cusco: I'd pass that on to your provider... |
| 15:25.54 | [TK]D-Fender | orn: Show us the call file and the call |
| 15:25.59 | orn | sec |
| 15:33.02 | orn | http://pastebin.com/ByJuctb2 |
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| 15:33.34 | [TK]D-Fender | orn: Where's the call? I asked to see the call |
| 15:33.45 | [TK]D-Fender | orn: And I don't see enough to see anything that it would actually dial |
| 15:33.47 | orn | ah from the cli, ok I misunderstood |
| 15:42.12 | cusco | so.. inserted a loop in telco's equipment and they state it is all good on their side. the problem should be in our PRI card |
| 15:42.27 | cusco | I guess I'm going to check the jumpor on E1/T1 |
| 15:42.41 | orn | http://pastebin.com/zVP4NE8D |
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| 15:45.20 | file | your dialplan does not match what is actually happening |
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| 15:46.35 | [TK]D-Fender | indeed |
| 15:47.28 | orn | ah |
| 15:48.04 | orn | the filename |
| 15:48.12 | orn | herp derp |
| 15:48.43 | orn | the testout context is from a different server with a different filename, must've copied form the wrong window |
| 15:49.41 | orn | it's exactly the same apart from the filename, let me paste it in another pastebin |
| 15:49.57 | orn | http://pastebin.com/ZfG3N7aF |
| 15:50.00 | orn | here it is from the right server |
| 15:50.12 | [TK]D-Fender | orn: You are also using 2 separate channels... because you get a CDR for that DIAL. |
| 15:50.36 | [TK]D-Fender | orn: call file a LOCAL channel .. THAT channel does a DIAL |
| 15:50.38 | cusco | can I connect a loop on the PRI card to test it? |
| 15:51.00 | [TK]D-Fender | sure |
| 15:51.57 | orn | Okay... In this particular example there's no real reason to go through the dialplan, but on the production servers I need to -- can I do that without having two separate channels? |
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| 15:52.40 | [TK]D-Fender | you might be geting 2 different CDR's, and you are focussing on just one... |
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| 15:53.10 | orn | I checked that -- only one CDR |
| 15:53.21 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 15:53.44 | [TK]D-Fender | Could be that since that's the dialplan that "channel" executed", the fact it bridged to another gets lost... |
| 15:53.55 | orn | at one point there were two, not exactly sure what's changed, but then one call had a duration of 5 and the other a duration of 0, and the call with duration 5 actually had 0 and vice versa |
| 15:55.48 | cusco | and once I connect the loop, what shoud I see different, stating it has something connected there? |
| 15:56.08 | orn | But do you know of any other way to send a call through the dialplan and then play a recording? That is, any other way other than using a local channel? |
| 16:00.34 | [TK]D-Fender | orn: the local channel is your CHANNEL: |
| 16:00.40 | [TK]D-Fender | eliminate THAT |
| 16:00.55 | [TK]D-Fender | Or find out what happened for your CDR change |
| 16:04.40 | orn | Yeah, but what I meant to ask is: Is it possible to originate a call from asterisk using a call file and have the call be processed by the dialplan without having to use a local channel, so that there will be only one channel for the call? |
| 16:04.58 | [TK]D-Fender | correct |
| 16:08.19 | orn | Do you have any hints for how I can achieve that, or locate some info that'll lead me there? |
| 16:09.11 | orn | That is, so I can locate some info :D |
| 16:09.12 | [TK]D-Fender | Dial your SIP device directly |
| 16:10.11 | orn | Ok, and that's the only way? Cause I need to figure out which device (or provider, rather) to call via the dialplan |
| 16:11.12 | [TK]D-Fender | Also try forcing the channels from bmerging with "/n" |
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| 16:14.03 | orn | hmm |
| 16:14.24 | orn | where does the /n go? |
| 16:15.37 | orn | never mind |
| 16:15.40 | orn | figured it out |
| 16:16.55 | orn | you are a demigod! |
| 16:16.56 | orn | :-) |
| 16:17.00 | orn | that works |
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| 16:17.26 | [TK]D-Fender | You're welcome |
| 16:21.19 | kenlik | suppose that a call takes to long to finish due to some problem; instead of take this call recording (saving to a file) for a long time, there's a configuration to finish it? |
| 16:22.10 | [TK]D-Fender | kenlik: It stops when you tell it to stop or it has a reason to |
| 16:22.11 | kenlik | it can be a timeout of a call or the max duration of call ... |
| 16:22.28 | [TK]D-Fender | kenlik: "something takes longer than expected" is not a decision it is going to make |
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| 16:25.55 | Chainsaw | kenlik: This isn't one of those PPI calls you're scripting up, is it? |
| 16:26.05 | kenlik | [TK]D-Fender, perhaps the best solution will be to investigate the reason of a call takes to long .... even if the user tell it to stop |
| 16:26.54 | kenlik | but i'd like to know if this could be simple to configure the asterisk to stop a call if .... |
| 16:27.49 | kenlik | Chainsaw, no it's not! |
| 16:27.51 | [TK]D-Fender | if... |
| 16:28.31 | kenlik | to stop a call if i takes to long to finish ;-) |
| 16:28.53 | [TK]D-Fender | "takes too long" is undefined |
| 16:32.22 | kenlik | for i.e: if a call duration time is more than 30 minutes |
| 16:32.45 | cusco | so the loop conneted states: [Jan 4 00:21:31] WARNING[1668]: sig_pri.c:7276 pri_dchannel: PRI Error on span 1: We think we're the CPE, but they think they're the CPE too. |
| 16:32.47 | cusco | seems normal |
| 16:34.02 | WIMPy | It is normal. |
| 16:34.24 | [TK]D-Fender | kenlik: Then you should limit the call |
| 16:35.02 | cusco | so.. I have a problem with telco :/ |
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| 16:36.45 | kenlik | [TK]D-Fender, there's a configuration for that? |
| 16:38.26 | [TK]D-Fender | "core show application dial" <- |
| 16:38.55 | [TK]D-Fender | "core show function TIMEOUT" |
| 16:40.30 | kenlik | [TK]D-Fender, Ok thanks! |
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| 16:43.52 | puzzled | evening all |
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| 16:45.02 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:46.25 | puzzled | I'm wondering about the build order. In the book/online docs the order is libpri, dahdi, asterisk but the libpri utility pridump.c requires dahdi/user.h. So what's the deal? Keep order and disable building of utilities in libpri? |
| 16:47.06 | Chainsaw | puzzled: I've had to turn that test suite off, yes. We frown upon circular dependencies in Gentoo land. |
| 16:48.14 | puzzled | Chainsaw: indeed, especially since the libpri Makefile has all: ... $(UTILITIES) so it will always fail unless dahdi was installed before libpri |
| 16:48.26 | puzzled | which is against the advice in the docs |
| 16:49.39 | Chainsaw | puzzled: I suspect Digium only test on RedHat Expensive Linux with pre-built RPM packages. |
| 16:49.53 | Chainsaw | puzzled: So from time to time, you have to supply your own initiative. Disabling the test suite is the right call. |
| 16:52.06 | puzzled | Chainsaw: ok, thanks. Even on RHEL you would still need to build the libpri package without $(UTILITIES), install libpri packages then build and install dahdi and then build libpri again. Weird |
| 16:52.07 | drmessano | puzzled: No.. https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+LibPRI |
| 16:52.22 | drmessano | Dahdi then Libpri |
| 16:52.31 | *** part/#asterisk orn (~orn@2a01:8280:101:0:9ab:e799:af20:c2ee) |
| 16:52.36 | ghost75 | jitter 11 and delay 300ms is crap or ok? |
| 16:53.05 | Chainsaw | ghost75: Depends, from where to where? |
| 16:53.06 | puzzled | drmessano: thanks, that's the actual opposite from the 3rd edition book. Nice they are in agreement there... |
| 16:53.25 | ghost75 | voip to isp |
| 16:53.38 | WIMPy | puzzled: On previous versions it didn't matter. |
| 16:53.45 | drmessano | puzzled, keep in mind the book may be oft-pointed-to, but is not official documentation. Experts can be wrong |
| 16:53.51 | Chainsaw | ghost75: *facepalm* Geographical location to geographical location. If it's not continent to continent then it is not great. |
| 16:54.03 | drmessano | puzzled, ... and often are |
| 16:54.07 | puzzled | ghost75: above 175ms delay voice quality tends to get worse |
| 16:54.19 | ghost75 | it is same country :< |
| 16:54.30 | Chainsaw | ghost75: Then it is horrendous, yes. |
| 16:54.48 | puzzled | WIMPy: got it |
| 16:54.59 | ghost75 | what jitter values are "ok" |
| 16:55.18 | puzzled | drmessano: in that case I'm wondering if I should order the 4th edition book if they can't even get the build order right |
| 16:55.50 | puzzled | changes build order |
| 16:55.56 | puzzled | thanks all |
| 16:57.36 | ghost75 | or is jitter same as delay? |
| 16:58.02 | Chainsaw | ghost75: No, jitter is the standard deviation. |
| 16:59.15 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qaupemjliqmtlmfz) |
| 16:59.46 | ghost75 | packet loss 0,15% |
| 16:59.53 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:59.54 | Chainsaw | recoils in horror |
| 17:00.27 | danfromuk | Hi, what does the AMI Command QUEUES do? It doesnt seem to return anything interesting |
| 17:00.37 | Chainsaw | Ther- re-ipe fo- -udio quali-y pr-bl-ms. |
| 17:00.54 | danfromuk | How can I retrieve the all queue stats for every queue? |
| 17:01.01 | *** join/#asterisk human39 (~human39@c-71-236-110-155.hsd1.pa.comcast.net) |
| 17:01.34 | ghost75 | welcome to germany high speed isp :< |
| 17:02.29 | drmessano | puzzled, it seems a bit pointless now, IMO |
| 17:03.14 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
| 17:03.28 | ghost75 | is it possible to disable a service from within asterisk without giving root? |
| 17:03.28 | Chainsaw | danfromuk: You probably want QueueStatus then. |
| 17:03.43 | rrittgarn | ghost75: sudo? |
| 17:03.51 | Chainsaw | danfromuk: "manager show commands". It's a bit terse, but there is a list of commands to try there. |
| 17:04.13 | human39 | I'm looking for something I'm not sure exists :-). I'm tasked to implement a chat based support system that is somehow intergrated with our asterisk system. We're mostly worried about call distribution and automatically pausing an agent when on a chat. I couldn't find any third party systems. Any suggestions? |
| 17:04.13 | ghost75 | sudo is root |
| 17:04.42 | danfromuk | Chainsaw: Thanks. Wish it provided a bit more detail :-) |
| 17:04.54 | Chainsaw | danfromuk: Well yes, that would be neat. |
| 17:05.00 | Chainsaw | danfromuk: But it's all I have to give. |
| 17:05.04 | ghost75 | and visudo works only for binaries, not for init scripts |
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| 17:08.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 17:12.05 | *** join/#asterisk porkchop85 (~porkchop@50.200.28.42) |
| 17:12.32 | porkchop85 | Hi I was wondering if someone could answer some questions about implimenting Asterisk |
| 17:12.34 | JonathanD | danfromuk: I was just fiddling with that this morning, maybe I can help. |
| 17:13.04 | [TK]D-Fender | porkchop85: ... |
| 17:13.07 | [TK]D-Fender | ~ask |
| 17:13.07 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:13.15 | *** join/#asterisk mitchrodrigues (~mitchrodr@204-16-155-82-static.ipnetworksinc.net) |
| 17:13.58 | [TK]D-Fender | BRB |
| 17:14.03 | porkchop85 | I am running a school district we have 13 physical building running a siemens ip system |
| 17:14.13 | porkchop85 | i have 3 t1s |
| 17:14.39 | porkchop85 | Is it possible to ditch the siemens |
| 17:14.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 17:14.52 | porkchop85 | I am running openstage and optipoint phones |
| 17:15.11 | porkchop85 | Hub and spoke topology with KSUs at each campus |
| 17:16.23 | porkchop85 | I also have POTS phones running on the system |
| 17:16.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:17.16 | puzzled | porkchop85: ask on the mailinglist if the Optipoint phones are supported. IIRC there was some discussion about it but I don't recall the conclusion |
| 17:17.59 | porkchop85 | If they aren't thats not a big deal since almost all of them have broken screens and I need an excuse to replace them |
| 17:20.02 | porkchop85 | Are POTS phones supported |
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| 17:22.10 | ghost75 | is this digium way that bugs are closed without solution? |
| 17:22.54 | [TK]D-Fender | porkchop85: * is software |
| 17:23.06 | file | is there a specific bug? |
| 17:23.12 | [TK]D-Fender | porkchop85: There are many HARDWARE interfaces you can use to plug analog phone into |
| 17:23.20 | file | ghost75, issue number that is |
| 17:23.20 | [TK]D-Fender | porkchop85: That * can talk to |
| 17:23.37 | porkchop85 | My concern is that I will need a box at every campus |
| 17:23.45 | porkchop85 | for the analog phones to convert |
| 17:23.58 | [TK]D-Fender | porkchop85: That sounds like a requirement.... what is the "concern" part? |
| 17:24.05 | porkchop85 | Money |
| 17:24.11 | ghost75 | https://issues.asterisk.org/jira/browse/ASTERISK-22842 <- i know that v29 doesnt do 2400 but that was not the question |
| 17:24.12 | LieutPants | [ASTERISK-22842] [Status: Closed] V29 is incompatible with minrate setting 2400 - https://issues.asterisk.org/jira/browse/ASTERISK-22842 |
| 17:24.19 | [TK]D-Fender | Well I guess you'd see that coming as it is :) |
| 17:24.19 | porkchop85 | Public schools dont get alot funding |
| 17:24.32 | [TK]D-Fender | porkchop85: Give us #'s and we'll suggest models |
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| 17:24.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:25.56 | file | ghost75, I don't understand what the issue is |
| 17:26.11 | porkchop85 | I have 250 ip phones and about 300 pots phones |
| 17:26.23 | ghost75 | it is not faxing when v29 is used with minrate 2400 |
| 17:26.48 | [TK]D-Fender | <PROTECTED> |
| 17:26.57 | file | right... because 2400 is not allowed for that rate... |
| 17:26.58 | [TK]D-Fender | (IP) |
| 17:27.25 | *** join/#asterisk u0m3 (~u0m3@109.96.175.232) |
| 17:27.59 | ghost75 | file: but then the default modem order v17,v27,v29 doesnt make sense because minrate 2400 is specified as default |
| 17:28.38 | ghost75 | and also v17 doesnt allow 2400 baud |
| 17:28.46 | porkchop85 | Well lets take the High School we have 200 pots phones and 50 Ip phones a mix of openstage 10s 40s and optipoint 410s |
| 17:29.25 | file | okay, if you had said that then I wouldn't have been confused reading the issue |
| 17:29.56 | ghost75 | english is not my mother language so thats a bit hard for me |
| 17:31.10 | file | I've reopened and copy/pasted part of the IRC log |
| 17:31.23 | ghost75 | thx a bunch |
| 17:32.03 | ghost75 | me is grammar nazi :> |
| 17:32.24 | [TK]D-Fender | porkchop85: the 410's seem to speak standard SIP which means they should woork with * as-is |
| 17:33.26 | porkchop85 | The 410s are garbage though every screen broke |
| 17:33.38 | porkchop85 | So I can't reprogram them |
| 17:34.04 | [TK]D-Fender | \porkfor the phones, that'd take 9 x 24-port gateeways like : |
| 17:34.09 | [TK]D-Fender | porkchop85: for the phones, that'd take 9 x 24-port gateeways like : |
| 17:34.17 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways/AudioCodes-MediaPack-MP-124-FXS |
| 17:34.31 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Analog-Gateways/Mediatrix-4124 |
| 17:34.40 | porkchop85 | Ouch |
| 17:35.14 | porkchop85 | That would be 10 grand just for the high school |
| 17:35.47 | [TK]D-Fender | porkchop85: Here's a cheaper one : http://www.telephonydepot.com/Catalog/Analog-Gateways/Grandstream-GXW-4224-Analog-FXS-IP-Gateway-24-Port |
| 17:36.03 | ChannelZ-Wk | Probably cheaper than they paid for their analog phone system originally |
| 17:36.24 | JonathanD | What are the issues with the current system? |
| 17:36.48 | porkchop85 | It is out of support and I am looking for alternatives |
| 17:37.40 | [TK]D-Fender | porkchop85: You're talking about supporting HUNDREDS of phones. |
| 17:37.56 | [TK]D-Fender | porkchop85: Have you seen what the price of what you have was? |
| 17:38.35 | porkchop85 | We are looking at 20 grand between buying a new KDS and upgrading the broken phones |
| 17:39.19 | porkchop85 | plus the support contracts |
| 17:41.05 | rrittgarn | besides a call to CelGenUserEvent, is there any easy way to log DTMF in CEL? I feel like i'm missing something simple |
| 17:43.02 | porkchop85 | Am I crazy for wanting to support all my phones |
| 17:43.14 | JonathanD | porkchop85: I don't envision any option being very cheap here. |
| 17:43.32 | JonathanD | porkchop85: if it helps, the place I came from last spent 25k on a new phone system, for 30 phones. |
| 17:44.23 | JonathanD | porkchop85: I don't think you're crazy for wanting support but if the district doesn't see the value in it, it's not likely to happen. |
| 17:44.38 | file | ghost75, and now that I'm done finishing some thoughts... sorry about that! we get a lot of issues and sometimes with the language issues we miss stuff/meaning |
| 17:45.03 | ghost75 | that is understandable! |
| 17:45.19 | porkchop85 | Well at least I dont need to buy alot of new phones :) |
| 17:46.36 | file | Greenlight, NUDGE |
| 17:46.39 | JonathanD | porkchop85: it might not be that much more to buy the phones, though. |
| 17:47.04 | JonathanD | eh, I guess it would, you'd still need switching too. |
| 17:47.32 | porkchop85 | I have been pushing for WIFI IP phones to replace the pots |
| 17:47.57 | JonathanD | Don't know how I'd feel about that many wifi phones. |
| 17:48.35 | porkchop85 | I have 150 Meru APs in this building |
| 17:48.40 | porkchop85 | I have about 300 district wide |
| 17:48.45 | porkchop85 | every building has it |
| 17:49.12 | ghost75 | 150 in one building oO |
| 17:49.19 | porkchop85 | We are a big district |
| 17:49.22 | porkchop85 | I have 5000 students |
| 17:49.29 | porkchop85 | 13 buildings |
| 17:49.54 | ghost75 | is voip over wifi working good? |
| 17:50.05 | porkchop85 | I currently dont have any Wifi phones |
| 17:50.27 | ghost75 | i just used some old nokia and delay was not good |
| 17:50.36 | JonathanD | well if you replaced the pots with voip over wifi, you're looking at phone replacement + the asterisk hardeware still. |
| 17:50.52 | JonathanD | You might be able to knock the price around a bit, but it's still gonna end up costing a good bit. |
| 17:51.49 | porkchop85 | I'll have to get a firm budget and see what I can do |
| 17:51.54 | [TK]D-Fender | ~33$ to adapt an analog set. |
| 17:52.04 | [TK]D-Fender | How much for AP's & the phones to match that way? |
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| 17:52.25 | porkchop85 | Cheapest wifi phone I saw was $150 |
| 17:52.36 | porkchop85 | I already have the wireless in place |
| 17:52.43 | porkchop85 | hell been that way for 5 years |
| 17:53.07 | JonathanD | porkchop85: you have the wireless, but it's presumably servicing users already. |
| 17:53.19 | porkchop85 | Not that many to be honest |
| 17:53.28 | porkchop85 | there are just a few mobile carts in each school |
| 17:53.34 | porkchop85 | and about 30 ipads |
| 17:53.59 | [TK]D-Fender | So > 4x the cost |
| 17:54.23 | porkchop85 | my current wireless throughput at the highschool is currently around 12 MBps |
| 17:54.28 | [TK]D-Fender | and hoping they don't suck, and don't get lost, broken, stolen, etc |
| 17:55.04 | porkchop85 | Well I'm replacing about 10 pots phones a month |
| 17:55.44 | porkchop85 | and I need to replace all of my optipoint phones |
| 17:55.47 | porkchop85 | because of the broken screens |
| 17:55.55 | JonathanD | which wifi phone are you considering. |
| 17:56.23 | porkchop85 | http://www.voipsupply.com/quickphones-qa-342 |
| 17:56.39 | JonathanD | I've not seen wifi phones used for much more than a 1 or 2 kinda deal. |
| 17:57.30 | porkchop85 | Well we aren't exactly a call center most of our calls are internal anyways |
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| 17:58.14 | JonathanD | You might be able to get away with it then. I'd still be worried about theft/damage etc on something like that. |
| 17:58.34 | JonathanD | porkchop85: are there ethernet runs in place? |
| 17:58.43 | porkchop85 | yes there is ethernet in every room |
| 17:58.51 | porkchop85 | but they aren't powered |
| 17:59.01 | JonathanD | thats not difficult to fix. |
| 17:59.27 | porkchop85 | I have injectors all over the place |
| 17:59.59 | porkchop85 | I cant put POE in my Catalyst because I am running fiber to all the rooms |
| 18:00.44 | JonathanD | porkchop85: you generally want your phones on their own switches/runs |
| 18:00.49 | jblack | fiber to the individual classrooms? |
| 18:00.53 | porkchop85 | yes |
| 18:01.12 | porkchop85 | They built the network infrastructre before IDFs were popular |
| 18:01.27 | porkchop85 | every building is that way |
| 18:01.41 | porkchop85 | with either MTRJs or media converters |
| 18:01.50 | porkchop85 | from the fiber to the catalysts |
| 18:02.09 | jblack | It's been a long while since I've run fiber. back then it was pretty expensive |
| 18:02.30 | porkchop85 | It was a million dollar build when it was done |
| 18:03.11 | jblack | I hate school systems |
| 18:03.23 | porkchop85 | I agree |
| 18:03.35 | porkchop85 | but its a 4 mile commute |
| 18:03.37 | jblack | It's not just that they're government institutions... |
| 18:03.46 | jblack | they're full of teachers. |
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| 18:45.52 | Kobaz | can you put multiple subnets for localnet in sip.conf (ast 1.8) |
| 18:49.30 | file | yes. |
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| 18:54.04 | Kobaz | yeah i just checked the source |
| 18:54.10 | Kobaz | it's not obvious that you use multiple localnet= in the sip.conf sample |
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| 18:55.21 | file | I pulled it up... it says multiple are allowed and then gives an example |
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| 18:56.35 | tenspeed705 | Hey guys. I was wondering if anyone can recommend some call centre type opensource projects? |
| 18:58.31 | tenspeed705 | only catch is must be strictly *nix+http(s) |
| 19:00.30 | Kobaz | it says |
| 19:00.38 | Kobaz | assigning the "localnet" parameter with a list of network addresses that are considered "inside" of the NATted network. |
| 19:01.02 | Kobaz | to me that means: localnet=10.0.0.0/255.0.0.0,192.0.0.0/255.0.0.0 |
| 19:01.04 | Kobaz | that's a list |
| 19:01.07 | file | ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY. |
| 19:01.07 | file | ; Multiple entries are allowed, e.g. a reasonable set is the following: |
| 19:01.16 | Kobaz | oh that part |
| 19:01.19 | Kobaz | heh whoops |
| 19:01.50 | Kobaz | my brain stopped at the entire line in caps |
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| 20:20.29 | Penguin | Does someone here know much about fail2ban? It looks like someone found out a way to exploit/bypass it. |
| 20:21.21 | Penguin | I'm not really wanting to put out the details in public. |
| 20:23.40 | kasanop | chan_sip doesn't log attacker's IP addresses in some cases |
| 20:23.50 | rrittgarn | i know a bit about fail2ban, but i've not heard about someone getting around it |
| 20:23.56 | rrittgarn | i use it to secure some of my windows based PBXs. |
| 20:24.37 | Penguin | The IP addresses are showing clearly, and in regular circumstances, the IPs get blocked. |
| 20:24.41 | Penguin | rrittgarn: PM? |
| 20:24.50 | rrittgarn | already did |
| 20:24.59 | Penguin | I just now unblocked it. |
| 20:25.08 | rrittgarn | you blocked me eh? |
| 20:25.17 | Penguin | No, I blocked everyone. |
| 20:25.27 | Penguin | Then I add people as I need to. |
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| 20:37.57 | ChannelZ-Wk | Is your issue delay between attempts? |
| 20:38.27 | ChannelZ-Wk | I've been seeing that a lot lately.. a person only trying once every hour (or longer) to try to avoid triggering the ban |
| 20:38.47 | Penguin | Crisis averted. It was a problem with findtime being JUST shorter than their attempts. |
| 20:39.04 | ChannelZ-Wk | exactly |
| 20:39.22 | Penguin | I was set at 3600 and they were trying at like 61 minutes. |
| 20:40.21 | Penguin | I just thought the worst before I examined closely enough. rrittgarn steered me in the right direction to get it fixed up. |
| 20:42.53 | Penguin | They'll get a surprise the next few times they try it. |
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| 20:45.09 | ChannelZ-Wk | Yea I'm up to like 6000 on my box here at work |
| 20:45.59 | ChannelZ-Wk | An hour and a half seems to be the attention span (so far) |
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| 22:11.26 | Penguin | After changing the findtime value and restarting fail2ban, the offending addresses were banned. |
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