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| 01:18.00 | hypknight|Away | is now away - Reason : Auto-Away (Away from Keyboard for 60 minutes) |
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| 03:53.17 | ruben23 | guys if i may ask is cdr_mysql module included already on asterisk 11..? Thanks |
| 04:22.43 | wyoung | ruben23: if you compiled it in then yes. If you installed a binary then maybe. |
| 04:40.59 | ruben23 | <PROTECTED> |
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| 06:37.24 | ruben23 | wyoung: ..? |
| 06:37.31 | ruben23 | <PROTECTED> |
| 06:38.42 | avb | ruben23: its more healthy to use res_odbc cdr_odbc modules this days |
| 06:38.55 | avb | i think cdr_mysql is already depricated |
| 06:43.10 | ruben23 | avb: does that work with mysql database also..? |
| 06:43.35 | avb | yes |
| 06:43.42 | avb | pretty much the same |
| 06:43.57 | avb | http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc |
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| 07:10.32 | ruben23 | avb: thanks ill do this |
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| 08:02.30 | wyoung | ruben23: the next step is to see if it is loaded in modules.conf |
| 08:02.41 | wyoung | or if it is excluded in there |
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| 09:30.55 | ntt | Hi, I'm using asterisk 13.7.2 with freepbx 13.0.95 and I have many messages in /var/log/asterisk/full: " requested media update control 26, passing it to SIP". these messages cause some kind of "flood" during the conversation. It seems that happens randomly and in rare cases.... someone can help me please? |
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| 10:37.27 | vankooch | Hi does anyone know a good way to analyze call profiles, using CDRâs? Iâll need to finde out if the incoming calls are bad or if the gateway has a problem. Is there any good solution for it? |
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| 11:49.19 | wyoung | hi gang |
| 11:49.42 | wyoung | vankooch: define "bad"? |
| 11:49.56 | wyoung | vankooch: CDR doesn't log anything useful for troubleshooting or debugging |
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| 12:08.48 | tzafrir | file, mind bumping the version of DAHDI-* in the topic? |
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| 12:09.23 | file | what's the version and release date? |
| 12:17.04 | vankooch | wyoung: soory for the delay⦠I mean wrong number or dialer calls |
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| 12:32.37 | tzafrir | file,dahdi-linux-2.11.1: 01-Mar-2016 16:55, dahdi-tools-2.11.1: 01-Mar-2016 16:55 |
| 12:32.46 | file | ack |
| 12:33.11 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.0 (2016/03/29), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.0 (2016/03/01), DAHDI-tools 2.11.0 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 12:34.16 | tzafrir | file, s/2.11.0/2.11.1/ |
| 12:34.27 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.0 (2016/03/29), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 12:34.31 | file | waves hand |
| 12:35.59 | *** join/#asterisk k0mb0y (~k0mb0y@169-0-95-2.ip.afrihost.co.za) |
| 12:37.09 | k0mb0y | I have a SIP trunk and a configured outbound route. I can see the call been routed over the SIP trunk however I am getting "the number you have dialled is not in service". Do you know what can be causing this? |
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| 12:49.46 | k0mb0y | anyone know? |
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| 13:02.28 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:03.17 | [TK]D-Fender | k0mb0y, If that's their audio then either that number isn't valid as dialed, or there is an issue at the other side. |
| 13:03.40 | k0mb0y | umm... how can I tell if it's their audio? |
| 13:03.56 | k0mb0y | I don't see asterisk playing the local recordings |
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| 13:05.00 | kchehab | hi ppl , where can i ask for android pjsip stack !! any idea |
| 13:05.13 | [TK]D-Fender | k0mb0y, Show us the call |
| 13:05.33 | [TK]D-Fender | k0mb0y, it could also be that the call IS going through and thatt message is from the person you are calling. |
| 13:05.46 | k0mb0y | oh no it's not |
| 13:06.01 | [TK]D-Fender | k0mb0y, Which if they were using a FreeePBX-based system and screwed up on their side.... |
| 13:06.11 | [TK]D-Fender | What proof do we have? |
| 13:06.16 | [TK]D-Fender | other #'s being called? |
| 13:06.25 | [TK]D-Fender | calling them from another phone? |
| 13:06.33 | [TK]D-Fender | (not your server) |
| 13:06.35 | k0mb0y | Called SIP/trunk/legitnumber |
| 13:06.36 | k0mb0y | <PROTECTED> |
| 13:06.36 | k0mb0y | <PROTECTED> |
| 13:06.51 | [TK]D-Fender | PB the FULL call |
| 13:07.00 | [TK]D-Fender | I see no answer there... so nothing to give you audio |
| 13:07.58 | [TK]D-Fender | ~pb |
| 13:07.58 | 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. |
| 13:08.00 | [TK]D-Fender | ^^^ |
| 13:10.20 | k0mb0y | I do get audip |
| 13:10.24 | k0mb0y | audio* |
| 13:11.17 | [TK]D-Fender | Show us the full call |
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| 13:39.33 | [TK]D-Fender | sits back with a coffee... |
| 13:40.23 | k0mb0y | could this be something to do with the dial plan on the telephone? |
| 13:40.32 | [TK]D-Fender | No. |
| 13:40.41 | [TK]D-Fender | Telephones don't magically play audio on their own |
| 13:40.43 | [TK]D-Fender | Show us the call |
| 13:40.53 | [TK]D-Fender | This shouldn't be taking half an hour. |
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| 13:42.30 | k0mb0y | ok I will but what can these be? |
| 13:42.44 | [TK]D-Fender | Stop asking without showing tthe evidence |
| 13:43.31 | [TK]D-Fender | "Doctor, before I let you examine me, tell me what's wrong with me!" |
| 13:43.58 | [TK]D-Fender | I'm only psychic on Tuesdays |
| 13:44.56 | k0mb0y | looser |
| 13:45.51 | [TK]D-Fender | No, I win big on Tuesdays, and succeed modestly on the the rest. |
| 13:46.41 | [TK]D-Fender | And we'll have an answer for you when we see the full call. |
| 13:50.13 | jamesc | anyone know why MeetMeChannelAdmin does not find the channel sometimes app_meetme.c:3652 channel_admin_exec: Specified user (SIP/185.38.229.19-000000f0) not found |
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| 13:52.51 | [TK]D-Fender | can we see the channel dump and command call to match? |
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| 14:31.22 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 14:49.25 | adeel | question...why does this syntax fail to match? ExecIf($[ '${DIALSTATUS}' != ( 'ANSWER' | 'BUSY' | 'CANCEL') ] |
| 14:50.16 | [TK]D-Fender | Where do you see () as being legal for a set? |
| 14:51.11 | adeel | don't recall, but i've must have seen it somewhere to think it was legal |
| 14:57.54 | [TK]D-Fender | Perhaps in another language... |
| 15:00.24 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 15:08.04 | wyoung | [TK]D-Fender: hey mate |
| 15:08.09 | wyoung | [TK]D-Fender: How you been? |
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| 15:24.27 | [TK]D-Fender | Doing alright... |
| 15:35.01 | wyoung | :D |
| 15:35.09 | wyoung | I am having issues transfering calls between snom phones |
| 15:35.21 | wyoung | it just disconnects and doesn't log anything |
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| 15:37.32 | gravspeed | good morning world |
| 15:37.48 | gravspeed | anyone know a good siptapi client that works correctly in windows 10? |
| 15:37.54 | gravspeed | preferably free |
| 15:38.11 | wyoung | gravspeed: mornin' |
| 15:38.27 | gravspeed | seems to happen every day doesn't it. |
| 15:38.33 | wyoung | gravspeed: I like snom phones, I am not a fan of computer clients |
| 15:38.48 | gravspeed | looking for click to dial |
| 15:38.54 | gravspeed | we use cisco phones. |
| 15:39.14 | gravspeed | we've used this one in the past http://www.ipcom.at/en/telephony/siptapi/ |
| 15:39.23 | gravspeed | but it doesn't work well in windows 10 |
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| 15:40.00 | wyoung | gravspeed: you can do that with snom |
| 15:40.17 | wyoung | gravspeed: All snom phones have a REST like API that you can control the phone with |
| 15:40.55 | wyoung | https://phonehostname/?dial=00116124811111 |
| 15:41.08 | wyoung | (that is pizzahut in Australia if you are hungry :P) |
| 15:41.24 | wyoung | replace 0011 with your international dialing code |
| 15:41.30 | gravspeed | i didn't know snom did that, that's neat |
| 15:42.03 | wyoung | as well as sending DMTF;s too, so if you are in a call you can send it digits to type ion |
| 15:42.07 | wyoung | and transfer, etc.. |
| 15:42.12 | gravspeed | we use the snom pa1 bridges all the time, but our phones are all cisco 79x0 or spa50x |
| 15:42.17 | WIMPy | But does it work by now? In the old firmwares they completely failed at http. |
| 15:42.18 | wyoung | the vision extention modules uses it too |
| 15:42.26 | wyoung | but you can write your own rest client to control it |
| 15:43.26 | WIMPy | I made tel: URIs available to click in a browser back then, but that always cerated 3 calls because of retries because of the broken reply. |
| 15:43.56 | wyoung | cisco's are too rigid, proprietary and expensive compared with snom |
| 15:44.20 | wyoung | WIMPy: the latest firmware works fine |
| 15:44.38 | wyoung | WIMPy: you can also controll it via asterisk (with SIP headers) |
| 15:45.25 | wyoung | I have written a few web based apps for it. |
| 15:45.59 | wyoung | For provisioning as well as configuration and real time operation (transfering, dialing, etc) |
| 15:46.09 | WIMPy | Maybe I get a newer one where I can try a newer firmware. |
| 15:46.19 | wyoung | I have heard polycom is good too but i havent used them before |
| 15:46.33 | wyoung | WIMPy: I use the 710's and 715D's |
| 15:46.51 | wyoung | the D's have gigabit two port switches |
| 15:47.13 | wyoung | the 710's are 100mbit ethernet |
| 15:47.19 | WIMPy | I have only used 320/360/370. It's been some years since I did that. |
| 15:47.49 | wyoung | I own a 300, it stil supports REST like API to control it and easy provisioning via DHCP |
| 15:47.55 | wyoung | or SIP / asterisk |
| 15:48.56 | wyoung | I have heard polycom is good too but they are too expensive in Australia, snom's I can get for about $130.00 inc GST AUD |
| 15:49.12 | wyoung | polycoms are usually over $200AUD |
| 15:49.34 | wyoung | WIMPy: what country are you from again? |
| 15:49.34 | WIMPy | Except for their conference stations, Polycom doesn't seem to exist in Europe. |
| 15:49.41 | WIMPy | de |
| 15:49.46 | wyoung | DU HAST! |
| 15:49.52 | wyoung | MICH! |
| 15:50.21 | wyoung | if you are in DE then why don't you use snoms? you must beable to get them cheaper than I can |
| 15:50.50 | WIMPy | Whay do you think so? |
| 15:51.12 | wyoung | when you but phones in DE do they tell you the price of the phone and the packaging / material / outter case seperatly too? like they do with food / bottles of drink? |
| 15:51.14 | WIMPy | But yes, so far I liked them the most. But I haven't used any IP phones since back then. |
| 15:51.23 | wyoung | but - buy |
| 15:51.25 | wyoung | - = = |
| 15:51.45 | WIMPy | No. Such a thing doesn't exist here. |
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| 15:52.24 | wyoung | oh, when I visited DE last (2012) the price of the packaging was listed seperatly to the actual product |
| 15:52.37 | wyoung | so you could see that stuff packaged in plastic was cheaper than glass |
| 15:52.45 | WIMPy | Where did you see that? |
| 15:52.53 | WIMPy | has never seen that. |
| 15:52.55 | wyoung | however you got a better refund for glass than plastic |
| 15:53.29 | wyoung | WIMPy: umm, the town a few KMs south of the denmark border, I forget the name of the place |
| 15:53.36 | Kunsi | also has never seen that |
| 15:53.58 | WIMPy | Sounds like HERE. |
| 15:54.11 | Kunsi | I think WIMPy doesn't whant to know location of store, but type of product :o |
| 15:54.37 | wyoung | All I remember is the grog was cheap (cheaper than denmark, the tax is insane there), the snitzel was awesome and the sausages were great too |
| 15:54.42 | WIMPy | Yes, but that was interesting anyways :-) |
| 15:54.47 | wyoung | :P |
| 15:55.14 | wyoung | WIMPy: it was on a river :) |
| 15:55.28 | wyoung | and it used to be Denmark before world war I |
| 15:55.29 | WIMPy | Yes that a whole chain across Europe. The people from the north come to the south to buy alcohol. |
| 15:56.00 | WIMPy | Like Flensburg. |
| 15:56.07 | wyoung | sounds about right |
| 15:56.10 | Kunsi | WIMPy: not everywhere. austria beer is much more expensive than german beer |
| 15:56.27 | wyoung | Kunsi: belgium beer is the most expensive |
| 15:56.44 | wyoung | Kunsi: do you prefer wheat beer? |
| 15:56.53 | WIMPy | hasn't had any contact to Austrian beer so far. |
| 15:57.02 | wyoung | WIMPy: Wheat beer? |
| 15:57.39 | wyoung | I have never been to Austria, but I heard most people there can speak German very well :P |
| 15:57.44 | WIMPy | I'm not in to beer at all, actually. |
| 15:57.47 | Kunsi | WIMPy: i paid 1.20⬠for a 0.5l can of beer - here in germany i usually pay less than 1⬠including pfand |
| 15:58.00 | wyoung | WIMPy: I am a little bit, I prefer scotch whiskey |
| 15:58.10 | wyoung | Kunsi: :O |
| 15:58.36 | wyoung | Kunsi: one beer here costs about $5.00 AUD (3.something EURO) |
| 15:58.41 | wyoung | min |
| 15:58.55 | WIMPy | prefers Whisky as well. Any type but burbon. |
| 15:59.01 | wyoung | <3 |
| 15:59.16 | Kunsi | AUD? Australian Dollars? |
| 15:59.23 | wyoung | I went to the Elgan whisky festival last year, it was awesome! |
| 15:59.26 | wyoung | Kunsi: yes |
| 15:59.29 | Kunsi | ok |
| 16:00.06 | Kunsi | (how did we get from "sip tapi" over "snom phones" to "i prefer whisky"?) |
| 16:00.21 | WIMPy | wyoung: But I'd still go for a beer if you give a shout next time you're in town :-) |
| 16:00.44 | wyoung | I booked two whisky tasting events in the same day, 12 drinks all up, I didn't realise that the Scottish poured 100 Ml shots for each drink, I was expecting 10 - 20Ml taster |
| 16:01.18 | wyoung | Kunsi: I am great at segways |
| 16:01.26 | WIMPy | Sounds like a fun day. [cough] |
| 16:01.39 | wyoung | WIMPy: no coughing, it was smoother as |
| 16:01.42 | wyoung | smooth* |
| 16:02.01 | wyoung | I didnt drink all of it in one sitting, they had take away cups :) |
| 16:02.30 | wyoung | but I was quote inebriated at the end of it |
| 16:02.35 | wyoung | quite* |
| 16:02.58 | wyoung | 8 year and above (up to 25 years), single malt! |
| 16:03.23 | wyoung | even the 8 year single malt was better than jonnie walker blue |
| 16:03.37 | wyoung | and a 1/3 of the cost |
| 16:07.57 | wyoung | n |
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| 16:22.40 | Get_The_Fish | Anyone else having stabillity issues with 13.7.2? |
| 16:24.09 | Get_The_Fish | And before you ask, I have NO IDEA why. I have full verbose logging set up, debugging to console, still crashes without a whiff of a reason why |
| 16:24.29 | [TK]D-Fender | Upgradde |
| 16:25.33 | wyoung | nn |
| 16:43.12 | Get_The_Fish | [TK]D-Fender: upgrade? Is that a blanket statement or is there some reasoning to support that? Any particular issue that I should be looking for here? |
| 16:43.59 | [TK]D-Fender | You have issues. You are not "current". First step is to upgrade and attempt to replicate |
| 16:44.59 | Get_The_Fish | [TK]D-Fender: well actually the first step is to revert my production clients to a version that I know doesnt have issues. Then I can test 13.8. |
| 16:45.07 | wyoung | [TK]D-Fender: Lets asterisk |
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| 17:03.06 | Kunsi | Get_The_Fish: first step would be update |
| 17:03.26 | Kunsi | then check if newest version does have issue, if yes, THEN downgrade |
| 17:03.31 | WIMPy | And don't forget to pray. |
| 17:04.01 | Get_The_Fish | Kunsi: That may make sense to a tech, but not for business. I cant wait for a customer's phone system to fail to downgrade. Sorry. |
| 17:04.37 | Kunsi | that actually does make sense |
| 17:04.50 | Get_The_Fish | Kunsi: Really, how? |
| 17:05.26 | Get_The_Fish | Kunsi: I would like to know how an upset customer makes sense for my business. |
| 17:05.46 | Kunsi | uhm |
| 17:05.55 | WIMPy | You can charge for service. |
| 17:06.06 | Kunsi | i was reffering to your line of text. |
| 17:06.07 | WIMPy | That's what it's all about, isn't it? |
| 17:06.31 | Get_The_Fish | WIMPy: No. My customers are on a subscription basis. |
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| 18:03.51 | Lappy | Hey, would anyone be able to help me figure out why my asterisk installation isn't honouring summer time for the Europe/London timezone when making a call to SayUnixTime? |
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| 18:17.53 | Samot | Get_The_Fish: I have numerous 13.7.2 servers that are running without any stability issues. |
| 18:18.14 | cmendes0101 | Lappy: Which tz are you using? |
| 18:19.21 | cmendes0101 | oh sorry for some reason I didn't see you listed it |
| 18:20.05 | cmendes0101 | Are you setting the tz in the function or letting it go off system time? |
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| 18:25.14 | Lappy | in the function cmendes0101, however the timezone on the system is the same |
| 18:25.25 | Lappy | I've tried omitting it from the function and the outcome doesn't change |
| 18:26.44 | cmendes0101 | The system time is showing dst correctly? |
| 18:26.59 | Lappy | it is indeed |
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| 19:46.38 | oDesk | I reinstalled asterisk 13.6 but i found that the rtp is not loaded, and when i try to load the module i get this error message 'Error loading module 'res_rtp_asterisk.so': libsrtp.so.1: cannot open shared object file: No such file or directory' |
| 19:46.59 | oDesk | although the module is there in /usr/lib/asterisk/modules |
| 19:47.48 | oDesk | i did refresh for the lib cache by ldconfig |
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| 19:48.28 | oDesk | and by rebooting but i don't know from where else the asterisk trying to load the module |
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| 20:02.17 | oDesk | it's ok, i had to change modules dir owner to asterisk |
| 20:02.21 | oDesk | now it works |
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| 20:22.06 | Splix76 | I have an issue where my older asterisk server, 1.8, will send the rpid in the header however my newer compiled from source server 13.x will not. |
| 20:22.38 | Splix76 | example of working: From: "rpidhere" <sip:actualnumberhere@hostip> |
| 20:22.53 | Splix76 | The non=working header has no number in quotes before the <sip: data |
| 20:23.08 | Splix76 | I have confirmed both configs are the same, and sendrpid is set to yes |
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| 20:24.32 | Splix76 | I could always stop using the 13.x compiled from source version and use the CentOS 1.8 ,but I have read 1.8 is end of life now. |
| 20:25.48 | Splix76 | I would appreciate it if someone has any tips or suggestions to try. I have been googlefu-ing for the past 4 hours without success. |
| 20:50.23 | Splix76 | I may have answered my own question in capturing the SIP header packets. It looks like the older version will spoof a number before the from:, and newer versions provide the Remote-Party-ID: data in the header which the SIPTrunk must honor for it to show the proper CallerID. |
| 20:50.48 | Splix76 | Anyone with experience or know how on this, please share. I am by no means an asterisk expert and could use the help. |
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| 22:57.37 | Micc | on asterisk 1.8.9.2 I my cdr's behaved differently and I'm not sure how to get it back to the old way. It used to only log one entry for a Dial with multiple devices, it would log the answered call, now it shows the no answer devices too. Is there a way to turn that feature off? |
| 22:58.28 | Micc | I already set unanswered = no in cdr.conf |
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