| 00:00.56 | snadge | im not sure if its just because its a monday or whatever.. but sip.conf ? asterisk.conf ? :p |
| 00:01.34 | snadge | theres lots of language = settings.. maybe i should just change them all and hope for the best |
| 00:09.28 | *** join/#asterisk Defraz (~Defraz@gump.fuzecore.com) |
| 00:10.32 | snadge | sweet.. that worked.. i just set language=en_AU in sip.conf |
| 00:10.57 | snadge | now i get some full on ocker aussie going.. "orrr yeah!! echo test mate!" ;) |
| 00:13.06 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 00:42.36 | WIMPy | Now I wonder is someone already mada a bavarian or saxxon sound set. |
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| 00:58.04 | dmbrooking | hey, I'm brand new to asterisk and hoping someone can answer a question of mine |
| 00:58.17 | WIMPy | ~ask |
| 00:58.17 | 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. |
| 00:59.47 | cusco | short for: don't ask to ask, just ask |
| 01:00.11 | dmbrooking | I need to add the ability to have voice chat on a website. Functionality will be like AIM/Yahoo, with push to chat. Everything I've researched leads me to Asterisk. I always thought Asterisk is for integrating with actual phone systems. Is Asterisk something that can help me with that? If it's a yes, I just need a yes, I can do the digging for more info myself. Just want to make sure I'm not barking up the wrong tre |
| 01:01.09 | WIMPy | There is webrtc support, even though webrtc isn't really specified, yet. |
| 01:01.41 | WIMPy | Or in short: Use at your own risk. |
| 01:01.44 | dmbrooking | thank you, that answer helps me out a ton |
| 01:03.17 | WIMPy | The "Use at your own risk" part means webrtc in general, not the Asterisk implementation. |
| 01:03.31 | WIMPy | Until there is a final spec. |
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| 01:36.21 | *** join/#asterisk TheProf (~chatzilla@205.236.104.113) |
| 01:36.57 | TheProf | Hello. I hope you are all very well today. I have a question - at times our Internet connection cuts out for maybe an hour. When it cuts out our VOIP server (Elastix) loses its registration with our VOIP provider. Once the Internet is re-established Elastix doesn't attempt re-establish the registration with the VOIP provider. How can it be set to automatically reconnect if it disconnects? Tha |
| 01:36.58 | TheProf | nk you very much. |
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| 02:31.26 | SeRi | dijib: yo |
| 02:53.59 | dijib | im here |
| 03:06.12 | justdave | is there a way to tell dundi to always ignore a specific route if it gets it as a response? |
| 03:07.10 | justdave | like I have a bunch of offices using dundi to figure out which offices own which extensions. two of the offices act as hubs for everyone else (both of those two offices have every other office as a dundi peer, the other offices besides those two only have those two) |
| 03:07.42 | justdave | ideally I would have every office advertise the two hub offices as backup routes, so if a given office can't reach one of them directly they can relay off one of the hubs instead |
| 03:07.43 | ChannelZ | TheProf: take a look at defaultexpiry in sip.conf |
| 03:08.04 | justdave | but telling them to do that seems to cause routing loops on the two hub servers (who keep seeing each other) |
| 03:08.41 | justdave | so telling the two hubs to ignore any response that points at itself seems like the obvious fix |
| 03:09.08 | justdave | not sure if there's a way to do that |
| 03:09.17 | TheProf | ChannelZ: Thank you - I will take a look at that. |
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| 03:38.51 | whtsup | hello |
| 03:39.25 | whtsup | can i use iax2 over 500ms delay with good voice quality |
| 03:39.26 | whtsup | ? |
| 03:41.49 | whtsup | ? |
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| 03:54.43 | ChannelZ | if the packet latency is consistent and it's not actually janking up your bandwidth, it should sound fine but just be delayed. |
| 04:15.58 | whtsup | hmmmmm |
| 04:16.10 | whtsup | can i use jitter buffer ? |
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| 08:59.54 | lijingcheng | Here is so quiet. |
| 09:02.25 | heraclitus | hi lijingcheng |
| 09:05.28 | lijingcheng | Hi, heraclitus! |
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| 13:09.43 | Katty | g'morning |
| 13:10.19 | Neoti | Katty: middle of afternoon ... lol |
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| 13:32.35 | jansiva | Hi |
| 13:32.46 | jansiva | we have an issue |
| 13:33.09 | jansiva | when use log into the conference room 8501 |
| 13:33.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pklyaccgnwdiybnx) |
| 13:33.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:34.08 | jansiva | after some time every one phone starts disconnectng one by one |
| 13:34.19 | jansiva | and then they need to log into 8501 |
| 13:34.25 | jansiva | once again |
| 13:34.38 | Greenlight | Does the CLI indicate what the reason might be ? |
| 13:34.39 | jansiva | can some one help me to sove the isssue |
| 13:35.08 | jansiva | i am new to asterisk |
| 13:35.42 | jansiva | unable to analyses log need your help |
| 13:36.06 | Greenlight | Lets see the CLI or a log of when it occurs |
| 13:38.22 | jansiva | http://pastebin.com/TxCk0vAf |
| 13:38.34 | jansiva | i quried 8501 and then got the log |
| 13:39.48 | anonymouz666 | sruffell: bad news using 2.7.0.1(linux+tools). the result was the same as using dahdi-linux 2.6.0.1 with old dahdi-tools. |
| 13:40.14 | sruffell | ok…then I guess I really need to see the config and the output from dahdi_cfg. |
| 13:40.27 | Greenlight | Which version of asterisk are you using ? |
| 13:40.29 | anonymouz666 | this time I have the output |
| 13:41.07 | Greenlight | Also, I don't see any channels hanging up on that log... |
| 13:41.28 | *** join/#asterisk ickmund (uid11004@gateway/web/irccloud.com/x-rtyyurewanmldubr) |
| 13:41.48 | jansiva | Asterisk 1.8.7.0, Copyright (C) 1999 - 2011 Digium, Inc. and others. |
| 13:41.48 | jansiva | Created by Mark Spencer <markster@digium.com> |
| 13:42.49 | Greenlight | Any reason you're using meetme, as opposaed to ConfBridgE ? |
| 13:43.40 | jansiva | no idea |
| 13:43.45 | anonymouz666 | sruffell: http://pastebin.com/EE1573ph |
| 13:43.56 | sruffell | looks |
| 13:44.24 | sruffell | and the system.conf output? |
| 13:44.34 | sruffell | <PROTECTED> |
| 13:44.44 | sruffell | contents not output. :/ |
| 13:45.53 | anonymouz666 | sruffell: http://pastebin.com/emRUk6pA |
| 13:47.00 | sruffell | hmm….cas= not something I check regularly. Using r2 singalling? (and let me recreate here and see what happens) |
| 13:47.02 | jansiva | please let me know if you need more logs |
| 13:47.15 | anonymouz666 | sruffell: yeah, it is MFC/R2. |
| 13:48.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:48.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:48.32 | Greenlight | jansiva: I can't see any of the channels going hungup there, can I see the dialplan, and also a log with debug disabled |
| 13:49.02 | jansiva | please let me know how to enable it |
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| 14:32.15 | sruffell | anonymouz666: while I'm setting this cas= test up, let me ask you another question. When you had multiple providers connected to a single framer, how often did your calls get dropped due to HDLC aborts? |
| 14:33.15 | *** join/#asterisk Dovid (~Dovid@float-gw.core-fw-t.mypbxmanager.net) |
| 14:33.50 | jansiva | can i know where to get dial plan |
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| 14:40.21 | [TK]D-Fender | jansiva: extensions.conf |
| 14:41.22 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 14:42.10 | killown | which one ports do I need to open to allows remote registering for a sip client? |
| 14:43.28 | *** join/#asterisk heise2k (~rheise@static-96-245-76-194.phlapa.fios.verizon.net) |
| 14:44.28 | Greenlight | Usually 5060 |
| 14:45.25 | [TK]D-Fender | killown: You may way to consider allowing the actual voice through as well.... |
| 14:48.26 | killown | 1000:2000 |
| 14:48.41 | killown | 10000:20000 |
| 14:51.28 | Greenlight | Depends what you've set in rtp.conf |
| 14:51.39 | Greenlight | I think the default is 10000 - 20000 |
| 14:52.41 | WIMPy | Or let the conntrack find out automagically. |
| 14:52.55 | anonymouz666 | sruffell: very few. this is not a problem on most cases. |
| 14:56.02 | cusco | "While the canary is tweeting, we know Asterisk is not taking 100% CPU |
| 14:56.04 | cusco | time." |
| 14:56.17 | cusco | asterisk was already tweeting before twitter |
| 14:56.22 | cusco | twitter should change name :p |
| 14:58.05 | WIMPy | How often does it take 100%CPU for a longer time? |
| 15:07.00 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 15:09.43 | cusco | no, not in my case |
| 15:09.47 | cusco | I was only reading |
| 15:10.14 | cusco | I'm having a issue tho.. for some time now but I'm afraid to ask, as I might not gather all the required debug details |
| 15:10.23 | cusco | seems as if sip stops working |
| 15:10.36 | cusco | even issuing a 'sip reload' in console does not work |
| 15:10.41 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 15:10.47 | cusco | new extensions do not register |
| 15:10.52 | cusco | existing calls go OK |
| 15:11.01 | cusco | calls in queue go into local dialplan |
| 15:11.05 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:11.10 | cusco | and try to dial SIP/peers insanelly |
| 15:11.29 | WIMPy | Which version? |
| 15:13.02 | cusco | yea.. thats what I'm afraid |
| 15:13.03 | cusco | lol |
| 15:13.04 | cusco | 1.6.2 |
| 15:15.46 | cusco | also this seems to happen when we have quite some calls in /var/spool/asterisk/outgoing/ |
| 15:16.13 | cusco | actually, we moved, they're now placed via ami, originate |
| 15:16.38 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 15:18.59 | *** join/#asterisk boratynskikamil (5311a3f3@gateway/web/freenode/ip.83.17.163.243) |
| 15:19.02 | cusco | we made some tests with 11 already, but we have a partner connecting his cisco via sip, and some calls comming from there (always via same sip peer, via same dialplan) - some would have audio, some would only have one side |
| 15:19.03 | boratynskikamil | Hello. |
| 15:19.42 | boratynskikamil | I would like to start working with asterisk. Could anyone of you advice me some good tutorial or book? I would be very thankful. |
| 15:19.56 | [TK]D-Fender | ~book |
| 15:19.56 | 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 |
| 15:19.58 | [TK]D-Fender | ^ |
| 15:20.11 | boratynskikamil | Fender. |
| 15:20.14 | boratynskikamil | Thank you a lot. |
| 15:21.21 | boratynskikamil | One more question: http://wklej.org/hash/6787370981e/ - will this card work correctly? |
| 15:22.05 | [TK]D-Fender | Looks like your standard HFC BRI card... |
| 15:22.15 | [TK]D-Fender | seems like one of the usual chipsets |
| 15:22.21 | [TK]D-Fender | I'd say probably... |
| 15:22.28 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-rephaaqxlauybhjk) |
| 15:22.47 | boratynskikamil | [TK]D-Fender: As I see, it works right now. I mean: kernel driver is used. |
| 15:22.58 | boratynskikamil | Should I see it in: ifconfig -a? |
| 15:24.19 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:24.29 | [TK]D-Fender | boratynskikamil: it isn't an IP interface... it's BRI |
| 15:25.58 | ChannelZ | Call from '' (188.227.184.50:5070) to extension '000000011448708756821' rejected |
| 15:26.05 | ChannelZ | My, that's a number. |
| 15:28.08 | Penguin | I had one of those recently. |
| 15:28.14 | WIMPy | boratynskikamil: No, it's not an IP interface. You can see it with misdn_info. |
| 15:29.05 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:30.46 | Penguin | I guess I had another earlier this morning, too. |
| 15:30.50 | Penguin | [Oct 14 04:10:06] NOTICE[23281]: Ext. 000000011448708756821:2 @ unauthenticated: "100" <100@188.227.184.50> is attempting to make unauthorized calls |
| 15:32.06 | cusco | Call from '204' to extension '9890913519890913515855' rejected |
| 15:33.09 | cusco | lol I was grepping and found worse |
| 15:33.21 | cusco | to extension '9877Paulo253926373Espirito72686Silva' rejected |
| 15:33.44 | Penguin | I got it from the same scanner that got channelz, though. |
| 15:34.04 | cusco | hmm scanners.. |
| 15:34.04 | Greenlight | Perhaps trying to exploit some sort of known bug or overflow |
| 15:34.17 | cusco | iptables is clean atm |
| 15:34.26 | cusco | they seem to strike during night time |
| 15:34.29 | WIMPy | It's just for out entertainment. |
| 15:34.43 | Greenlight | Might be daytime where they strike from :) |
| 15:34.54 | cusco | might |
| 15:35.06 | Penguin | They stopped doing multiple attempts... presumably to avoid getting hit by fail2ban or other rate limiting mechanisms. |
| 15:40.15 | cusco | I noticed that fail2ban wasn't piking the 5 times as specified, it eventually banned but took many more times |
| 15:40.54 | cusco | for instance: |
| 15:40.55 | cusco | [2013-10-14 09:16:19] NOTICE[8106] chan_sip.c: Registration from '"15424" <sip:15424@10.100.100.5:5060>' failed for '176.31.65.140' - No matching peer found |
| 15:41.18 | [TK]D-Fender | heads out for a while |
| 15:41.38 | WIMPy | Well, it's far from realtime. So unless the atteacker waits for possible reactions he will get a lot more attempts than you want. |
| 15:41.41 | Penguin | You said it took many more times before it banned, but over what time period? minutes, hours? |
| 15:43.51 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:44.26 | Greenlight | fail2ban is far from a perfect 100% solution, but it's a hell of a lot better than not having it |
| 15:44.50 | WIMPy | And at the same time it's also worse than nothing. |
| 15:45.00 | WIMPy | Pick your favourite part. |
| 15:46.07 | *** join/#asterisk alx0 (~alx@213.138.88.177) |
| 15:51.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:54.23 | cusco | and using fail2ban-regex just noticed something is wrong :/ |
| 15:54.50 | Greenlight | This is the other danger.. if it's setup wrong, you get a false sense of security... |
| 15:55.14 | cusco | seems that thats me ! |
| 15:56.55 | WIMPy | You always get a fals sense of security. But if you configure it wrong, you also shoot yourself in the foot. |
| 16:03.42 | Greenlight | We found blocking IPs outside the UK also greatly helped, but we've the luxary that we don't need to allow non UK people to connect in |
| 16:05.07 | *** join/#asterisk HolTech (~IceChat77@pool-108-40-37-180.bltmmd.fios.verizon.net) |
| 16:06.00 | Blashyrkh | what exactly are trustrpid and sendripd doing when iam connecting my asterisk to a voip trunk |
| 16:07.03 | WIMPy | Copy caller ID to or from the Remote-Party header. |
| 16:07.08 | Blashyrkh | does it have consequences if i can initate a call or just what information is in the RPID |
| 16:07.08 | Penguin | I don't think iax2 has trustrpid and sendrpid. |
| 16:07.22 | Blashyrkh | sip voip |
| 16:07.39 | Penguin | You said "trunk" so I had to assume iax since sip doesn't trunk. |
| 16:07.48 | Blashyrkh | yeah sry my fault |
| 16:11.48 | HolTech | Is there a quicker way to test out a dialplan besiders editing extensions.conf, issuing a dialplan reload in the cli and then calling into the extension? any debugging tips/tricks ? |
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| 16:27.06 | cusco | actually seems taht fail2ban is right I was testing the wrong rule |
| 16:27.12 | cusco | testing the filter file is fine |
| 16:27.25 | cusco | thing is bantime only lasts 10 minutes altho I've set it for 72h |
| 16:27.26 | cusco | :/ |
| 16:27.47 | cusco | fail2ban log http://paste.debian.net/57476/ and jail conf http://paste.debian.net/57475/ |
| 16:27.50 | cusco | gah :/ |
| 16:28.21 | *** join/#asterisk darkley (~darkleyab@204.10.62.82) |
| 16:32.28 | Penguin | What's wrong with that? |
| 16:32.50 | Penguin | I have a maxretry of 3 and a bantime of 604800. Yours is only slightly different. |
| 16:33.02 | cusco | well I have bantime 72h but keebs unbaning after 10 minutes |
| 16:33.03 | cusco | :/ |
| 16:33.11 | cusco | Penguin: what do you use for findtime? |
| 16:33.22 | Penguin | Where is findtime set? |
| 16:33.48 | Penguin | Oh, also in jail.conf |
| 16:33.52 | Penguin | 600 |
| 16:33.58 | cusco | thats the default |
| 16:34.34 | cusco | in #fail2ban |
| 16:34.34 | Penguin | I think I'm going to increase it. |
| 16:34.35 | cusco | 17:24 < fab23> cusco: add the repository according to http://neuro.debian.net/#get-neurodebian and you will have a current version of f2b |
| 16:34.44 | cusco | Penguin: thats the time it takes to re-read the log |
| 16:34.51 | cusco | maybe not good to increase |
| 16:35.06 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:35.06 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:35.06 | Penguin | That's not what it says to me. |
| 16:35.10 | cusco | err |
| 16:35.17 | cusco | thats what I assumed I did not check |
| 16:35.31 | cusco | ahhhh |
| 16:35.38 | cusco | that needs to be decresed indeed |
| 16:35.49 | cusco | like 5 times in 2 minutes |
| 16:35.55 | cusco | seems ok |
| 16:36.03 | Penguin | It says if "maxretry" times within the time set by "findtime", then ban. |
| 16:36.12 | Penguin | So right now, it's 3 times within 600 seconds. |
| 16:36.35 | Penguin | I think I'll increase it to like 1800. |
| 16:36.58 | Penguin | Maybe 3600. |
| 16:41.05 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:42.22 | cusco | actually 3 times in 600 seconds seems ok |
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| 17:04.16 | ironoxid | Hi, I need some help. Can I ask here? I have an E1 PRI, chan_dahdi and a TE122BF card. All works well but I can't receive our last free numbers (ex. 082434XXX) when I call from a common phone. Instead it works perfectly when I call from a mobile phone :( I've tried in every way, but nothing seems to work. Sadly I can't use an IVR. |
| 17:04.58 | WIMPy | Make sure you have overlap enabled. |
| 17:05.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 17:07.58 | ironoxid | I have it, also I've tried a full shutdown/restart. But it won't work. I've overlapdial=yes in chan_dahdi.conf. Is it correct? Also I've set immediate=no |
| 17:09.41 | WIMPy | And what about your dialplan? |
| 17:11.21 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:12.08 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:12.08 | ironoxid | I have a very basic dialplan: exten => _X.,1,Answer / exten => _X.,n,SayDigits(${EXTEN}) / exten => _X.,n,HangUp |
| 17:13.01 | WIMPy | That should mean you would always have to wait for a timeout. |
| 17:14.00 | ironoxid | I have to wait for the overlap timeout in the extensions? |
| 17:14.34 | WIMPy | If you don't specify them in your extensions.conf, yes. |
| 17:17.50 | ironoxid | Should I insert exten => _X.,1,Wait(3) before the Answer? |
| 17:18.18 | WIMPy | No. |
| 17:18.25 | WIMPy | There is nothing to wait. |
| 17:18.51 | WIMPy | The dialplan matching will wait for a timeout if you have an endless pattern as extension. |
| 17:26.52 | ironoxid | From a common phone I've "Accepting call from '498732066' to '5782230' on channel 0/12, span 1" and it doesn't work. From a mobile phone I've "Accepting call from '3397337964' to '578223001' on channel 0/13, span 1" (it works). Note 001 appended correctly from mobile calls. |
| 17:27.47 | WIMPy | Because normal phones use overlap dialling and mobiles use en-block dialling. |
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| 17:34.21 | sruffell | anonymouz666: problem is with the dchan=16. Newer versions of DAHDI will not allow dchan to be set on channel 16 when the span is in cas mode, since it doesn't make sense to do any HDLC framing in that case. |
| 17:39.00 | anonymouz666 | sruffell: should I remove the dchan from system.conf? |
| 17:39.14 | sruffell | yes…that should allow it to work for you. |
| 17:39.51 | anonymouz666 | sruffell: do you know why I can't see the ioctls() being called and therefore no detailed output when loading dahdi? |
| 17:40.08 | sruffell | Because the initial chan config is failing, so the start span is not being called. |
| 17:40.28 | sruffell | You might see some extra information is you load the driver with debug=1 |
| 17:40.36 | sruffell | modprobe wct4xxp debug=1 |
| 17:41.00 | anonymouz666 | sruffell: alright, newer versions = 2.7.x.x? |
| 17:41.17 | sruffell | Let me find the exact commit for you.. (but newer like 2.4.x I believe) |
| 17:41.36 | tm1000 | sruffell: does that debug statement work for all drivers? |
| 17:41.39 | tm1000 | or will some crash |
| 17:41.56 | sruffell | it's driver specific. Different drivers do different things with it (but generally, it should be safe) |
| 17:42.02 | tm1000 | ok |
| 17:42.13 | anonymouz666 | sruffell: 2.6.1 supports cas= |
| 17:42.22 | anonymouz666 | I have lots of setups using it in that version |
| 17:42.41 | sruffell | yeah…but it was a specific commit to error when a HDLC framing is trying to be set on an invalid channel. |
| 17:43.04 | sruffell | anonymouz666: but I'm guessing you do not have a dchan= line in those configs? |
| 17:43.10 | sruffell | with 2.6.1? |
| 17:43.57 | anonymouz666 | I have cas= line and dchan= lines in 2.6.1. |
| 17:44.02 | anonymouz666 | 8 E1s |
| 17:44.08 | sruffell | hmm... |
| 17:44.13 | sruffell | well..let me find that commit. |
| 17:44.31 | anonymouz666 | tm1000: modinfo <module> gives to you the options |
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| 17:44.50 | tm1000 | anonymouz666: oohhh? |
| 17:44.53 | tm1000 | neat-o |
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| 17:54.05 | sruffell | anonymouz666: http://svnview.digium.com/svn/dahdi?view=revision&revision=9056 is the commit in question. It's specific to the wct4xxp and wcte12xp driver from 3 years ago. |
| 17:55.10 | anonymouz666 | confused. There is something wrong |
| 17:55.20 | sruffell | https://github.com/sruffell/dahdi-linux/blob/master/drivers/dahdi/wct4xxp/base.c#L1676-L1683 are the lines in the driver where if the span is set to "CAS" mode, channel 16 is declared as supporting no signalling. |
| 17:58.32 | anonymouz666 | I can't explain why this works in 2.6.1 |
| 17:58.54 | anonymouz666 | and this piece of code it's there |
| 17:59.19 | sruffell | perhaps an older version of dahdi_tools that doesn't error out on chanconfig failure? |
| 18:00.40 | sruffell | hmm..no..seems to have failed since early on. |
| 18:01.21 | anonymouz666 | sruffell: http://pastebin.com/GH1ZzsiT |
| 18:02.23 | sruffell | dchan= isn't set on any of the spans with cas configured. |
| 18:02.37 | sruffell | only the ccs spans have the dchan set it looks like. |
| 18:02.41 | sruffell | so that looks right to me. |
| 18:05.02 | anonymouz666 | Ooooopsss. You are right |
| 18:05.16 | *** join/#asterisk bchia (~Adium@nat/digium/x-zxuzgdlfppdzdlbk) |
| 18:05.29 | anonymouz666 | basic error, my bad |
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| 18:10.27 | darkley | hello all |
| 18:10.49 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:11.15 | darkley | I have a quick question conecerning getting cisco 8961 phones running properly with asterisk |
| 18:11.58 | darkley | When it misses a call currently the call never leaves the display, seems to hang the line, and does not show in the missed calls log |
| 18:12.10 | darkley | Have any of you come accross this before? |
| 18:17.01 | *** join/#asterisk serafie (~erin@nat/digium/x-vytnbgpdszrhjnus) |
| 18:18.05 | darkley | I am running 11.4 have tried 1.8.14 and 11.5 with the same behavior |
| 18:19.27 | darkley | at this point I am not sure if its a problem in my SEP file or on the backend of the Asterisk. |
| 18:20.20 | sruffell | darkley: I'm not familar with cisco phones, but does Asterisk believe the channel is still up? (core show channels) |
| 18:21.31 | *** join/#asterisk bchia (~Adium@nat/digium/x-xkwizstyiokccmdl) |
| 18:24.08 | darkley | let me check I don't think so |
| 18:24.55 | darkley | nope 0 channels active |
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| 19:31.19 | *** join/#asterisk Shaan (~quassel@unaffiliated/shaan) |
| 19:31.52 | Shaan | hey guys does anyone know of fax over ip software that is not bloated with other shit? |
| 19:31.52 | *** join/#asterisk shamilbear (~shamil@2001:41d0:1:ef31::1) |
| 19:32.17 | shamilbear | hi, has anyone here successfully setup a linksys spa3102 correctly? I seem to end up with Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)... |
| 19:36.05 | rrittgarn | its not registering then |
| 19:36.46 | rrittgarn | that or you typed the device name wrong in your extensions.conf |
| 19:37.02 | Katty | pokes sruffell |
| 19:37.24 | sruffell | bows |
| 19:37.27 | shamilbear | I see registration is failing in the FXO |
| 19:37.59 | WIMPy | Why does pri debug write empty lines? |
| 19:38.22 | sruffell | that I do not know. |
| 19:38.28 | WIMPy | I.e. just a "PRI Span: 1" as every second line it seems. |
| 19:40.21 | WIMPy | And why does it always send PROCEEDING messages? |
| 19:41.00 | sruffell | I've not spent too much time poking around libpri. |
| 19:41.13 | shamilbear | also |
| 19:41.22 | shamilbear | how do I get the FXO to register? |
| 19:43.37 | shamilbear | oh my god |
| 19:44.07 | sruffell | sounds like someone had an epiphany |
| 19:44.20 | shamilbear | I JUST EFFING DIALLED OUT :D |
| 19:44.37 | anonymouz666 | sruffell: ISDN/MFCR2 is OK. just stay away from SIP and you will be fine :-) |
| 19:44.51 | sruffell | lol. datachomper would love to hear that. |
| 19:45.11 | sruffell | I was just curious whether MFC/R2 was more / less suspectible to timing variances from different providres. |
| 19:45.14 | sruffell | providers |
| 19:46.23 | anonymouz666 | lots of telcos, differents prices, it quite common to have more than one pluggued into the same card |
| 19:46.59 | WIMPy | Yes, but no vendor seems to care :-( |
| 19:48.27 | sruffell | so sounds like MFC/R2 works better with those timing slips than PRI? |
| 19:51.17 | anonymouz666 | all the cases I have seen with timing slips (GES counter, if I remember correctly) was due problems elsewhere |
| 19:52.12 | sruffell | cool |
| 19:52.14 | sruffell | thanks |
| 19:53.08 | sruffell | WIMPy: You're in the UK right? You don't have multiple providers connected to a single card ever (/very often) do you? |
| 19:53.40 | WIMPy | No, Germany. |
| 19:54.04 | sruffell | Ah, I didn't know that. |
| 19:54.14 | anonymouz666 | sruffell: telco still insist using the old and slow MFC/R2 due costs with the interconnecting equipments |
| 19:54.21 | WIMPy | And whenever I had multiple lines, I'd make sure they are from different telcos. |
| 19:54.34 | sruffell | WIMPy: and the different telcos do not sync their clocks in Germany? |
| 19:54.46 | anonymouz666 | but PRI is taking place, right now telco are offering 1 E1 splitted. 30 channels with g729 and 1 mb of data |
| 19:55.28 | WIMPy | Du to the cheap technology, even the timing on one line can be worse than it should be. |
| 19:55.40 | WIMPy | Due.. |
| 19:56.18 | WIMPy | BRIs wil almost always come out of VOIP gateways today, so their timing is just bad and never in sync. |
| 19:56.22 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:56.47 | WIMPy | PRIs over G.SHDSL are known to have suboptimal timing as well. |
| 19:56.59 | sruffell | makes sense (unfortunately) |
| 19:58.06 | WIMPy | But even if you have real BRIs, they aren't synchronous. |
| 19:58.31 | anonymouz666 | BRIs are very rare here |
| 19:58.44 | *** join/#asterisk BriGuy (~BriGuy@74.115.41.6) |
| 19:59.11 | WIMPy | It's the most likely interface here. |
| 19:59.46 | BriGuy | what is the difference between echo cancellation settings in system.conf and chan_dahdi.conf? |
| 19:59.59 | WIMPy | It's an issue with PBXs as well. You need to define the line you will use for fax as the timing source. Otherwise you get in to trouble. |
| 20:00.21 | WIMPy | ... and don't use others. |
| 20:00.34 | anonymouz666 | I had problems with DAHDI and v.90 in the past |
| 20:01.06 | sruffell | BriGuy: system.conf just configures any software echocanceler you would like attached. The ones in chan_dahdi.conf actually control echo canceler settings in general. |
| 20:01.21 | WIMPy | V.90 is probably even even a lot more picky than fax. |
| 20:01.41 | anonymouz666 | yeap |
| 20:01.50 | sruffell | that is because historically asterisk only dealt with "the echocanceler" attached to a channel. So the system.conf settings are to set what that echocanceler is. |
| 20:02.41 | Shaan | hey guys can anyone help with a FOIP question? |
| 20:02.46 | BriGuy | ahhh got ya - so for HW based echo cancellation can I just remove the line from system.conf or do I need to specify it |
| 20:03.20 | BriGuy | also, side-note question if a SW EC was specified in the system.conf file - would double echo cancellation be performed? |
| 20:03.27 | sruffell | You can remove it…or leave it. It doesn't matter *unless* you load dahdi with the hwec_overrides_swec=0 (by default it's 1) |
| 20:04.13 | BriGuy | ok, thanks! that helps! |
| 20:05.03 | anonymouz666 | sruffell: if the hw is set, the card still moves TDM data just like if all the channels are in use? |
| 20:06.38 | *** part/#asterisk shamilbear (~shamil@2001:41d0:1:ef31::1) |
| 20:06.43 | sruffell | hmm..I'm not sure I understand the question. If hwec is set for any channel, the card takes the TDM data from the backside of the hardware echocanceler and then keeps it in passthrough if echocan is not on. If hwec is not set for any channel, or an echocan module is not loaded, the TDM data comes from the same place as the input to the hwec. |
| 20:07.00 | sruffell | keeps it in passthrough on a channel by channel basis I mean. |
| 20:07.40 | sruffell | Shaan ~ask |
| 20:07.43 | sruffell | ~ask |
| 20:07.43 | 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. |
| 20:08.43 | anonymouz666 | I asked that because in some machines even if there's no use you see the load increasing a little bit. |
| 20:08.48 | Shaan | well i am trying to figure out a way to send faxes in the simplest form available i.e lighweight software or some sort of print driver via VOIP |
| 20:08.56 | anonymouz666 | if you remove chan_dahdi, the load decreases |
| 20:09.01 | anonymouz666 | using a sw echocan |
| 20:09.26 | sruffell | ahh..yeah, because it's not on by default. It's turned on when channels go in use. |
| 20:09.42 | sruffell | same for hwec too though. |
| 20:10.18 | sruffell | Shaan: hmm…yeah, I'm not going to be much help there. I know we use fax for Asterisk here at Digium. Others have used spandsp + hylafax. |
| 20:12.06 | WIMPy | I think that's somethign I need to look in to some time, as well. |
| 20:13.05 | Shaan | stuffel how are you sending faxes |
| 20:13.07 | Shaan | sruffell: |
| 20:14.19 | sruffell | I've not personally set this up, but I know this is what is in use in our office: http://www.digium.com/en/products/software/fax-for-asterisk It's not the only solution. |
| 20:15.27 | anonymouz666 | sruffell: what's growing a lot in here, it's the appliances (like R-Series). there's a lot of benefits moving the TDM gateway outside asterisk box. |
| 20:15.44 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:15.49 | WIMPy | Does res_fax require FFA or is that an option? |
| 20:16.45 | *** join/#asterisk jsjc (~Adium@5.Red-81-34-89.dynamicIP.rima-tde.net) |
| 20:17.45 | anonymouz666 | I think it does. For spandDSP is res_fax_spandsp |
| 20:18.21 | WIMPy | It seems like you can choose from either res_fax or app_fax. |
| 20:18.35 | WIMPy | (if you want to make it with Asterisk) |
| 20:23.43 | WIMPy | Wrong unit on FAQ 23. It's bps not baud. |
| 20:25.29 | Penguin | What's the difference? |
| 20:25.44 | Penguin | I thought baud was equivalent to bits per second. |
| 20:25.54 | WIMPy | Baud is symbols/s. |
| 20:28.45 | dijib | my thoughts aswell |
| 20:29.07 | *** join/#asterisk krapper (krapper@territory.krapper.net) |
| 20:32.13 | dijib | why do u think the following line maxes out at like 15-20seconds |
| 20:32.19 | dijib | same => n,Record(/var/spool/asterisk/monitor/asterisk-recording-${record-time}.wav); |
| 20:33.01 | WIMPy | Define "max out". |
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| 21:56.02 | dijib | compiling 11.5.1 |
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