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| 00:49.42 | no2pencil | quesiton : I am using a Cisco ATA 188 with both phone lines, both work with my dialplan when I am using a phone, however when I try to use a modem they get "forbidden" status is 'CHANUNAVAIL' |
| 00:49.46 | no2pencil | am I just not passing the correct codec with the ATA? |
| 00:49.48 | no2pencil | or is there a 'faxpassthrough' or 'modempassthrough' or something that needs to be configured for the extention? |
| 00:58.25 | no2pencil | nevermind... |
| 00:58.27 | no2pencil | got it |
| 01:00.29 | *** part/#asterisk no2pencil (~no2pencil@direct2.akroncdnr.com) |
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| 02:32.18 | raspberrypifan | anyone know about gsm gateways |
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| 02:58.16 | [TK]D-Fender | Yes. They act as a gateway between GSM and some other protocol |
| 02:59.47 | raspberrypifan | i mean i bought one and im trying to use it for a nice overseas project |
| 03:02.00 | [TK]D-Fender | Good to know... |
| 03:05.32 | drmessano | Are you using it for GSM? |
| 03:14.35 | raspberrypifan | yup |
| 03:14.43 | raspberrypifan | sadly the box has zero instructions |
| 03:16.26 | [TK]D-Fender | Then return it for a refund if you can't get them from the vendor or manufacturer |
| 03:17.22 | raspberrypifan | there isnt a manufacturor |
| 03:18.03 | raspberrypifan | its all alibaba |
| 03:18.09 | raspberrypifan | i mean i figured out how ot log in |
| 03:18.13 | raspberrypifan | and there are sip trunking options |
| 03:18.15 | raspberrypifan | so in theory thats it |
| 03:18.19 | [TK]D-Fender | I'm sure those molecules didn't randomly collect into that product out of thin air... |
| 03:19.18 | [TK]D-Fender | It would take an infinite number of alternate realities for such a thing to occur, and you just aren't that lucky |
| 03:21.37 | raspberrypifan | u never know |
| 03:22.36 | [TK]D-Fender | WE aren't that lucky. |
| 03:25.54 | raspberrypifan | well its one of those fly by night |
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| 03:44.19 | drmessano | If it has SIP "trunking' options then it shouldnt be hard to set up |
| 03:44.27 | drmessano | Set up a matching peer in Asterisk. Done. |
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| 03:49.27 | raspberrypifan | yea i was trying to use yate to conenct as a peer or account but that didnt really work |
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| 04:01.13 | [TK]D-Fender | Also good to know... |
| 04:03.37 | raspberrypifan | well |
| 04:03.38 | raspberrypifan | its a start |
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| 05:58.43 | DanQuinney | for those users who manage multiple SIP servers do you use any tools like; http://www.sipcapture.org for central management of SIP debugging? If so do you have any recommendations? |
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| 07:30.39 | Lope | If I can connect to a VoIP provider through 1 NAT, is there any reason why I'd have an issue connecting through 2 NATs? |
| 07:30.54 | Lope | Not that I have an issue, I'm just considering it. |
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| 07:51.40 | sl4ck | . |
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| 09:32.57 | Lope | Does anyone have much experience using VoIP over OpenVPN? |
| 09:33.48 | Lope | I needed to connect to a VoIP provider now that doesn't support roaming. So I was using openVPN to my server in that country and connecting via NAT. it worked great for about 15 mins now it's dropping every minute or two. |
| 09:33.55 | Lope | the VPN is over UDP. |
| 09:34.07 | Lope | and it's tun. |
| 09:35.24 | jkroon | Lope, not really an asterisk question direct, but ... let's see what we can do. |
| 09:35.36 | jkroon | you're presumably using private IP ranges on the tun interfaces? |
| 09:35.45 | Lope | yes 10.0.x.x |
| 09:36.12 | jkroon | how stable is the openvpn portion itself? Ie, if you run something like "ping -f -i 0.02 10.1.2.3" on the host closest to you does it show any losses? |
| 09:36.52 | Lope | I'll try that |
| 09:37.14 | jkroon | on the host performing the NAT, is there any firewall rules in place? |
| 09:38.04 | Lope | it drew some dots under the command then got stuck. |
| 09:38.15 | Lope | after about 15 dots |
| 09:38.27 | jkroon | dots are bad :). |
| 09:38.39 | Lope | 1818 packets transmitted, 1803 received, 0% packet loss, time 42897ms |
| 09:38.44 | jkroon | it outputs a dot for every ICMP echo request sent, and a ^H (backspace) for every response received. |
| 09:38.54 | Lope | looks alright? |
| 09:38.58 | jkroon | open a terminal and leave it for 10 minutes or so. |
| 09:39.01 | jkroon | indeed, it does :) |
| 09:41.11 | Lope | Laptop OpenVPN > RaspberryPi-NAT > FiberModem > a lot of hops > ADSL-Appliance-NAT-Port-Forwarding > BananaPi-OpenVPN-NAT- and then out the ADSL-Appliance-NAT again. |
| 09:41.20 | Lope | It's quite an adventure it goes on. |
| 09:41.29 | *** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 09:41.55 | Lope | The weak link is probably the appliance router. I hate appliance routers. Underpowered CPUs and buggy software. |
| 09:42.24 | Lope | No wireless involved. |
| 09:46.59 | Lope | hmm, I should actually be connecting the BananaPi with it's own IP. Router's got bridge mode enabled. That will be 1 less NAT to deal with. |
| 09:47.31 | Lope | Likely better performance. |
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| 10:03.49 | Lope | 39861 packets transmitted, 39835 received, 0% packet loss, time 905564ms |
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| 11:02.41 | pawiecki | I'm trying to make dialplan extension pattern '_uzdr61XXX', but calling 'uzdr61891' gives notice, that extension was not found. But when i set it directly to extension 'uzdr61891' - the call proceeds without a problem. What am I doing wrong here? The pattern should be ok. |
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| 11:07.01 | wdoekes | pawiecki: z is [1-9] |
| 11:07.09 | wdoekes | _u[z]dr61XXX |
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| 11:31.44 | pawiecki | wdoekes: I was thinking about it, but according to http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns it should be ok if I use lowercase letters |
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| 11:38.56 | nixnothing | morning |
| 11:39.12 | nixnothing | all the coffee |
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| 11:43.52 | Lope | Why won't XFS format? http://codepad.org/A2oPItHU |
| 11:44.08 | Lope | doh! wrong channel. |
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| 11:46.27 | WIMPy | pawiecki: Case doesn't matter, unfortunately. |
| 11:49.09 | wdoekes | pawiecki: don't. ever. rely. on. voip-info.org |
| 11:55.19 | pawiecki | wdoekes: ok, will have that in mind. |
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| 12:23.17 | [TK]D-Fender | pawiecki, its IS for any lower-case letter. You did not read the instructions correctly |
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| 12:56.59 | pawiecki | [TK]D-Fender: for now i've changes it to different number. Based on this https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching - lower-case 'z' matches 1-9 digits. |
| 12:57.04 | pawiecki | changed* |
| 12:57.17 | [TK]D-Fender | you still aren't reading that page right |
| 12:58.12 | pawiecki | [TK]D-Fender: what's 'not right' about my reading? |
| 12:58.41 | [TK]D-Fender | they are in BRACES <------------ |
| 12:58.52 | [TK]D-Fender | You have to LITERALLY put the letters IN braces to be taken literally |
| 12:59.19 | [TK]D-Fender | Just like "[1237-9] matches any digit or letter in the brackets" |
| 12:59.28 | [TK]D-Fender | ""in the brackets" is for the letters as well |
| 13:00.00 | [TK]D-Fender | <PROTECTED> |
| 13:00.29 | [TK]D-Fender | you need to [] escape out every pattter letter so as not to be treated as such |
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| 13:07.49 | pawiecki | [TK]D-Fender: now calm down, and tell me where i'm wrong writing: 'lower-case 'z' matches 1-9 digits' |
| 13:08.05 | [TK]D-Fender | look on the LEFT |
| 13:08.09 | [TK]D-Fender | the letters are in brackets |
| 13:08.13 | [TK]D-Fender | they are on purpose |
| 13:08.26 | [TK]D-Fender | see the reference for z, x, etc? Notice they are not in braces? |
| 13:08.26 | pawiecki | in my example, they were not, so what is the problem? |
| 13:08.43 | [TK]D-Fender | That's because outside of braces they treated as pattern indicators |
| 13:09.01 | pawiecki | you are trying to tell me, what I already know, so you are wasting time. I know about the usage of brackets. |
| 13:09.04 | [TK]D-Fender | in your sample the problem is that you did not put the pattern-based chars in braces |
| 13:09.18 | [TK]D-Fender | and thus got taken for digit matching and not as the letter itself |
| 13:09.34 | pawiecki | yes i know, what the problem was in my example. Why are you telling me that i do not know that? |
| 13:09.47 | [TK]D-Fender | pawiecki |
| 13:09.47 | [TK]D-Fender | I'm trying to make dialplan extension pattern '_uzdr61XXX', but calling 'uzdr61891' gives notice, that extension was not found. But when i set it directly to extension 'uzdr61891' - the call proceeds without a problem. What am I doing wrong here? The pattern should be ok. |
| 13:10.33 | [TK]D-Fender | And I do now see that wdoekes pointed out the right pattern |
| 13:10.36 | [TK]D-Fender | missed that comment |
| 13:10.41 | pawiecki | your assuming, I'm at the begginning, but I already understood what the origin of my problem was, in the mean time i've changed the extension already, so it's not a problem anymore. |
| 13:11.19 | [TK]D-Fender | Wel you could match the originally desired format. |
| 13:11.36 | [TK]D-Fender | And I missed a comment to clue in that this had been answered actually.. |
| 13:11.41 | [TK]D-Fender | So my bad on that... |
| 13:11.42 | pawiecki | [TK]D-Fender: yes, now i know :) no problem |
| 13:12.07 | pawiecki | my bad also, on not reading from source on asterisk wiki |
| 13:12.09 | [TK]D-Fender | All good. I uses selective caps for emphasis btw, not "yelling" |
| 13:12.44 | [TK]D-Fender | Not if you see full lines of all caps that doesn't look like a caps inversion ... then that'd be another matter :) |
| 13:15.53 | *** join/#asterisk obelixBE (~obelix@d5152CCD8.static.telenet.be) |
| 13:23.47 | [TK]D-Fender | Note* |
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| 13:24.58 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 13:28.57 | tparcina | To connect Asterisk and Skype, I would need Skype for Business account? |
| 13:31.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:32.02 | cresl1n | Does Skype for Business support SIP? |
| 13:32.08 | cresl1n | goes to look |
| 13:32.38 | [TK]D-Fender | Yes |
| 13:32.55 | cresl1n | tparcina: presumably? |
| 13:34.57 | tparcina | [TK]D-Fender: So, I pay few ⬠every month, and I will be able to route incoming Skype calls to my phones? |
| 13:35.14 | [TK]D-Fender | Yes |
| 13:35.41 | tparcina | cresl1n: As far as I have seen, Skype for Business supports SIP. |
| 13:36.04 | tparcina | [TK]D-Fender: Are you using it? |
| 13:36.20 | [TK]D-Fender | Nope, but I did help someone out with it once |
| 13:36.40 | [TK]D-Fender | I have no personal or business need for Skype currently |
| 13:36.52 | tparcina | And everybody who has Skype account will be able to contact me, not just other people who have Skype for Business? |
| 13:37.01 | [TK]D-Fender | The instructions are ther.... and easily google-able |
| 13:37.06 | [TK]D-Fender | Correct. |
| 13:37.21 | tparcina | Great. I think I'll go for it. |
| 13:37.22 | [TK]D-Fender | Anyone calling your Skype account.. regardless of their end |
| 13:37.22 | tparcina | TY |
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| 13:48.44 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:48.58 | cresl1n | newtonr!!!! |
| 13:49.06 | file | IT'S RAINING TACOS!!! |
| 13:49.16 | newtonr | greetings! |
| 13:49.22 | newtonr | you guys are hyper this morning |
| 13:50.22 | file | maybe |
| 13:51.16 | cresl1n | I'm usually really groggy in the morning so I try to counteract it with additional enthusiasm |
| 13:53.19 | newtonr | The grog is strong with me as well |
| 13:53.36 | cresl1n | sends newtonr some grog to help |
| 13:53.49 | file | cheer up you two |
| 13:54.03 | file | or I'll have sent you sent to the third floor |
| 13:54.08 | file | er have you sent |
| 13:54.17 | cresl1n | sends file less grog |
| 14:00.57 | newtonr | lol |
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| 14:12.56 | *** join/#asterisk st2 (~st@static.78-158-207-178.kgts.ru) |
| 14:18.24 | st2 | hello everyone! first of all sorry for my poor english. would you help me with a little misunderstanding? |
| 14:18.25 | st2 | I'd like to provide realtime dialplan via sqlite engine. but i can't get extensions from db. |
| 14:18.44 | st2 | i've got following message in debug: |
| 14:18.44 | st2 | [2016-05-24 16:21:38] DEBUG[508][C-00000001] res_config_sqlite3.c: DB: asterisk SQL: SELECT * FROM "sip_extensions" WHERE "exten" LIKE '\_%' AND "context" = 'user_107' AND "priority" = '1' |
| 14:19.08 | st2 | the same query on db file didn't return any extensions. but query like that SELECT * FROM "sip_extensions" WHERE "exten" LIKE '_%' AND "context" = 'user_107' AND "priority" = '1' return what i need |
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| 14:21.10 | [TK]D-Fender | Not clear. |
| 14:21.16 | [TK]D-Fender | Which works, and which doesn;t? |
| 14:21.42 | st2 | SELECT * FROM "sip_extensions" WHERE "exten" LIKE '_%' AND "context" = 'user_107' AND "priority" = '1' |
| 14:21.42 | st2 | work perfect on db file |
| 14:22.09 | st2 | but asterisk got query with escaped '_' |
| 14:22.22 | st2 | this query did not return anything |
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| 14:24.03 | st2 | i've found few code about escaping queries in res_config_sqlite.c |
| 14:24.03 | st2 | tmp_str = (!strcmp(vals[0], "\\_%")) ? "_%" : (char *)vals[0]; |
| 14:24.03 | st2 | maybe similar code needed in res_config_sqlite3.c ? |
| 14:24.15 | [TK]D-Fender | Show the entire call so we can also prove that the connect worked, etc |
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| 14:31.02 | jimbop | Hi all, we're seeing some strange stuff here with 1.8.30. We're seeing big pauses in SIP OPTIONs qualifying various hosts. Asterisk just stops sending the options messages. Looks similar to bug 15721 |
| 14:31.11 | jimbop | but this is apparantly resolved. |
| 14:31.24 | jimbop | Does anyone know of reasons Asterisk would do this? |
| 14:31.42 | jimbop | i checked the change logs and there doesn't seem to be anything in .31 or .32 that addresses this. |
| 14:32.03 | [TK]D-Fender | 1.8 is not supported at all anymore |
| 14:32.08 | [TK]D-Fender | 11 or 13 only |
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| 14:33.53 | st2 | part of debug log: |
| 14:33.53 | st2 | http://pastebin.com/tv7NLggC |
| 14:33.57 | jimbop | ok, this i know and understand. However is there any valid reason it would do this? |
| 14:46.13 | st2 | [TK]D-Fender, do you need some additional information? |
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| 14:59.19 | [TK]D-Fender | st2, Not sure what more I can recommend. It doesn't say that there were erros, etc... |
| 14:59.32 | [TK]D-Fender | file, newtonr ? |
| 14:59.49 | file | uno sec |
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| 15:01.53 | file | in meeting, will be out in 10 minutes or so maybe kinda |
| 15:03.51 | st2 | i think it does not return any error cause connection with db is correct. just query did not return any row. so it is not an error |
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| 15:47.00 | newtonr | st2, sounds like a bug to me. File an issue on issues.asterisk.org/jira and I'll look at it a bit closer. |
| 15:47.34 | newtonr | read the guidelines before posting on the tracker: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
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| 16:04.12 | jms77 | I have a question about configuring the system object in pjsip.conf |
| 16:04.31 | jms77 | Im trying to change timer_b (specifically to a value that is less then 32*timer1) |
| 16:04.42 | jms77 | I added these lines in pjsip.conf |
| 16:04.51 | jms77 | [system] |
| 16:04.51 | jms77 | type=system |
| 16:04.51 | jms77 | timer_b=15000 |
| 16:04.51 | jms77 | [system] |
| 16:04.51 | jms77 | type=system |
| 16:04.51 | jms77 | timer_b=15000 |
| 16:04.52 | jms77 | type=system |
| 16:04.57 | jms77 | doesnt work |
| 16:05.09 | jms77 | so I added ps_systems to extconfig/sorcery |
| 16:05.11 | jms77 | since im using odbc |
| 16:05.53 | jms77 | so in the database I added it as well |
| 16:05.55 | jms77 | [system] |
| 16:05.55 | jms77 | type=system |
| 16:05.55 | jms77 | timer_b=15000 |
| 16:06.00 | jms77 | mysql> SELECT * FROM ps_systems; |
| 16:06.01 | jms77 | +--------+----------+---------+-----------------+-------------------------+---------------------------+-------------------------+---------------------+ |
| 16:06.01 | jms77 | | id | timer_t1 | timer_b | compact_headers | threadpool_initial_size | threadpool_auto_increment | threadpool_idle_timeout | threadpool_max_size | |
| 16:06.01 | jms77 | +--------+----------+---------+-----------------+-------------------------+---------------------------+-------------------------+---------------------+ |
| 16:06.01 | jms77 | | system | NULL | 15000 | NULL | NULL | NULL | NULL | NULL | |
| 16:06.01 | jms77 | +--------+----------+---------+-----------------+-------------------------+---------------------------+-------------------------+---------------------+ |
| 16:06.02 | jms77 | 1 row in set (0.00 sec) |
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| 16:06.32 | jms77 | if I do pjsip show settings in saterisk console I still get 32000 for the value of timer_b |
| 16:07.21 | jms77 | I dont get any errors in the log when I boot asterisk, so I'm wondering what I might have done wrong |
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| 16:08.02 | jms77 | sorcery.conf has this line under [res_pjsip]: system=realtime,ps_systems |
| 16:08.22 | jms77 | extconfig has this line: ps_systems => odbc,asterisk |
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| 16:16.22 | jms77 | Well nm, I guess that timer_t1*64 thing is actually enforced |
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| 16:56.44 | nixnothing | My mom always told me not to play with the POTS and LANs |
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| 17:10.11 | lvlinux | Where in pjsip.conf do i put the pickupgroup and callgroup settings for an endpoint? |
| 17:13.33 | [TK]D-Fender | ....in the endpoint |
| 17:13.38 | [TK]D-Fender | just like the sample config tells you |
| 17:14.59 | rmudgett | also CLI "config show help res_pjsip endpoint" |
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| 17:28.35 | lvlinux | oh, i forgot to check the sample config lol. the Wiki doesn't say |
| 17:29.16 | [TK]D-Fender | Sample telsl you what parms are and where they go |
| 17:33.41 | F2Knight | I got an asterisk box that keeps doing core dumps randomly. I have created the backtrace, but really I do not know what I am looking at here. Can anyone give me a suggestion as to how to figure out what is causing the dump |
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| 17:48.10 | lvlinux | F2Knight: first thing would be to find out when it dumps---what was going on at that time. |
| 17:48.22 | lvlinux | F2Knight: if it's unpredictable then that may be a problem. |
| 17:48.36 | F2Knight | lvlinux the system was 'making calls' |
| 17:48.52 | F2Knight | https://issues.asterisk.org/jira/browse/ASTERISK-26054 |
| 17:49.57 | lvlinux | you might try asking in #asterisk-dev |
| 17:50.02 | F2Knight | Basically the system is doing a hundred to 180 concurrent calls, and is fed by a dialer. |
| 17:50.26 | F2Knight | Okay thanks |
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| 18:58.19 | jrun | ./configure seems to ignore --with-jansson=n. any idea? |
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| 19:16.44 | newtonr | jrun, What do you mean ignores? what does the config.log show regarding jansson? |
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| 19:20.51 | jrun | newtonr: in 2 minutes |
| 19:21.19 | lvlinux | If I do a Progress(), shouldn't the other end of the line here ringing before the call is actually answered? |
| 19:21.31 | lvlinux | s/here/hear/g |
| 19:23.02 | WIMPy | Progress does not imply ringing. |
| 19:24.25 | jrun | Your paste can be seen here: https://gist.github.com/a4d2bd36d6ab1445962b46c8620c0ea6 |
| 19:24.31 | jrun | newtonr: ^ |
| 19:24.56 | lvlinux | WIMPy: Well it is ringing on the other side, but I don't hear it on the calling side. I just hear when it is answered. |
| 19:26.48 | WIMPy | Sounds like incompatible assumptions about early media. |
| 19:27.28 | lvlinux | Well, it's an * server on both sides of the call, so I wouldn't think that would be an issue. ? |
| 19:27.50 | newtonr | jrun, why are you setting the path to "n" ? |
| 19:27.59 | WIMPy | Depends on your configuration. |
| 19:28.07 | newtonr | jrun, it can't find jansson, "configure:13113: error: *** JSON support not found (this typically means the libjansson development package is missing)" |
| 19:28.35 | lvlinux | WIMPy: ok, where should I look for that? |
| 19:29.17 | newtonr | jrun, --with-jansson should be set to the path where libjansson is. |
| 19:29.28 | WIMPy | Everything early media. "early", "progress", "inband" |
| 19:29.50 | jrun | newtonr: looking into ./configure script i thought if with_jansson is set to no|n then USE_JANSSON=no and PBX_JANSSON=-1 to skip the dependency |
| 19:30.25 | jrun | i am trying to avoind jansson. |
| 19:30.43 | jrun | *avoid |
| 19:31.34 | gtjoseph | --without-jansson |
| 19:31.58 | jrun | gtjoseph: same thing |
| 19:32.26 | jrun | it fails at the same line that newtonr posted. |
| 19:32.56 | gtjoseph | oh, wait. i think jansson is required now. |
| 19:33.38 | jrun | any official dependacy matrix? |
| 19:35.02 | jrun | ./configure bails out JANSSON_LIB is empty which is not touched when checking for with_jansson |
| 19:35.33 | gtjoseph | did you run contrib/scripts/install_prereq? That's probably the most up to date. |
| 19:36.09 | jrun | don't think it support slackware |
| 19:36.29 | jrun | yep, unsupported_distro |
| 19:36.50 | jrun | wait, gentoo is unsupported too? |
| 19:37.06 | jrun | anyway, gentoo doesn't matter. |
| 19:37.29 | jrun | just wanna make sure jansson is now required. |
| 19:37.31 | WIMPy | That's exactely the two distros I use. |
| 19:37.41 | gtjoseph | you can get an idea of what's required by looking at the other distros |
| 19:38.03 | jrun | WIMPy: asterisk version? |
| 19:38.04 | WIMPy | I can give you my tag files if you want them. |
| 19:38.16 | WIMPy | 11 |
| 19:38.39 | jrun | exactly, that's our production server. we're planning to move to 13 series. |
| 19:39.30 | jrun | hmm, what's this ac_mandatory_list |
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| 19:39.58 | gtjoseph | they're the things configure will fail on if not found |
| 19:42.14 | jrun | doesn't look like it. it gets built for --with-FEATURE passed by the user; it seems. |
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| 19:52.28 | gtjoseph | yes, anything specified with --with- becomed mandatory. :) |
| 19:52.35 | gtjoseph | s/becomed/becomes/ |
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| 23:42.01 | maelstorm | Is this a good place to ask for help with an asterisk 1.8 upgrade to 11? |
| 23:42.48 | WIMPy | All you need to know is in UPGRADE.TXT. And it shouldn't be much. |
| 23:47.44 | maelstorm | Having never performed an asterisk upgrade, is there any good material on it? Looking at the upgrade-11.txt and upgrade-10.txt, I dont quite know all the things to change. |
| 23:48.18 | maelstorm | I was attempting to use asterisk-version-switch |
| 23:48.58 | WIMPy | Just install the new version. There's a good chance that's all there is to do, depending on the complexity of your setup, off course. |
| 23:49.22 | WIMPy | Is that a FreePBX thing? |
| 23:49.23 | Samot | maelstorm: Is this on a FreePBX box? |
| 23:49.37 | Samot | asterisk-version-switch is a FreePBX tool. |
| 23:49.53 | *** part/#asterisk kharwell (kharwell@nat/digium/x-drcufmtnnpljjxfp) |
| 23:49.54 | maelstorm | Yes, its Freepbx 12/13. After a version switch, ext to ext stopped working. |
| 23:50.10 | WIMPy | ~freepbx |
| 23:50.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:50.27 | maelstorm | Thanks |
| 23:50.30 | maelstorm | :( |
| 23:50.33 | Samot | Yeah that. |
| 23:50.58 | Samot | No need for a frowny face. You're using a FreePBX distro tool, better to ask in there. |
| 23:51.36 | WIMPy | Especially if it goes wrong. |
| 23:53.41 | Samot | So like, are you going to ask in #freepbx? |