| 00:00.25 | *** join/#asterisk TandyUK (~admin@87.252.44.195) |
| 00:03.07 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 00:28.21 | *** part/#asterisk JAT-WR (JAT-WR@wsip-70-184-71-20.oc.oc.cox.net) |
| 00:28.27 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zdrgzjjrzhnvpqee) |
| 00:34.17 | *** join/#asterisk resist0r (uid18260@gateway/web/irccloud.com/x-qiizqnzzqbbxupxq) |
| 00:42.43 | Akari | Yeah there are a lot of assumptions involved in FreePBX |
| 00:43.08 | Akari | Not that it's a bad thing but you have to know they're being made. |
| 00:43.40 | Akari | O that's basically what the thingy said |
| 00:51.40 | *** join/#asterisk daemon (~daemon@hostname.racing) |
| 01:04.42 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:08.48 | daemon | hey all what is the default port the web administration panel opens up on |
| 01:09.49 | [TK]D-Fender | Depends on which one you're referring to |
| 01:10.04 | daemon | asterisk13-13.12.1 Open Source PBX and telephony toolkit |
| 01:10.19 | daemon | oh I thought asterisk normally came with a web panel |
| 01:10.25 | daemon | been quite a while since I had to set one up |
| 01:10.40 | daemon | not even really sure if its the right tool for the job |
| 01:10.59 | [TK]D-Fender | Asterisk is a telephony engine |
| 01:11.21 | [TK]D-Fender | Everything more is something completely else |
| 01:11.28 | daemon | I have a phone number / voip tunnel on sipgate, I want it so that I can ring that number and asterisk picks up and from there I can ask for some system information temperature etc |
| 01:11.49 | [TK]D-Fender | * is certainly capable of that |
| 01:12.08 | daemon | perfect |
| 01:14.56 | [TK]D-Fender | If you're referring to the actual server temperature then you'll need to know how to get that information, but once you have it it's easy enough to get * to read it back. |
| 01:15.06 | [TK]D-Fender | And prompt as to what you want to look up |
| 01:15.36 | daemon | ah yes already got that, I have a perl script that grabs the relevent information pumps it through a text2voice and gives you an mp3 |
| 01:18.15 | daemon | hmm |
| 01:18.16 | daemon | No ethernet interface found for seeding global EID. You will have to set it manually. |
| 01:18.27 | daemon | is that set in sip.conf or elsewhere? |
| 01:21.16 | [TK]D-Fender | no need for text 2 voice |
| 01:21.20 | [TK]D-Fender | * can just read the # back |
| 01:21.31 | [TK]D-Fender | not sure what's giving that other message |
| 01:21.35 | [TK]D-Fender | You've have to actually show it |
| 01:21.59 | daemon | https://paste.ee/r/8TyX3 |
| 01:27.34 | daemon | looks like they even tell you config you need |
| 01:27.34 | daemon | https://basichelp.sipgate.co.uk/hc/en-gb/articles/206404129-Asterisk-PBX |
| 01:27.35 | daemon | :) |
| 01:28.08 | daemon | wonder how I test if it works the panel says the line is not connected |
| 01:28.10 | daemon | hmm |
| 01:31.01 | [TK]D-Fender | How did you install your system? |
| 01:31.12 | daemon | portmaster -Bd misc/asterisk13 |
| 01:31.22 | daemon | there was also asterisk11 |
| 01:31.27 | daemon | went with the later one |
| 01:33.12 | daemon | I managed to connect with asterisk -r |
| 01:33.22 | daemon | seems happy enough, did not bring up that voip connection though |
| 01:35.04 | [TK]D-Fender | "bring up voip connection" doesn't really say anything |
| 01:35.19 | daemon | the panel on sipgate says nothing is connected |
| 01:35.35 | daemon | like if I use a softphone or so the panel normally notes its connected |
| 01:35.43 | daemon | but its not saying anything is connected to the account at the moment |
| 01:36.05 | [TK]D-Fender | to debug that we'd hav to actually look |
| 01:36.14 | [TK]D-Fender | and prove that it's trying to do something... is configure, etc |
| 01:36.17 | daemon | hostname*CLI> sip set debug on |
| 01:36.20 | daemon | I set debugging on |
| 01:48.15 | *** part/#asterisk saul (~hubert@165.98.98.114) |
| 01:52.41 | [TK]D-Fender | prove something |
| 01:52.46 | [TK]D-Fender | that you ave peers configured. |
| 01:52.52 | [TK]D-Fender | if you're trying to register. |
| 01:52.58 | [TK]D-Fender | etc |
| 01:54.23 | daemon | https://paste.ee/r/AOhvR |
| 01:54.33 | daemon | userid is the userid they gave me at the voip provider |
| 01:54.38 | daemon | just did not want to put it in public paste :) |
| 01:54.53 | [TK]D-Fender | ok, so you have a peer for them |
| 01:55.00 | [TK]D-Fender | now to prove that you're registered |
| 01:55.10 | daemon | ok dokey, how do I do that |
| 01:58.25 | daemon | if I try to call the phone number its says its not connected |
| 01:58.43 | daemon | I used this config: |
| 01:58.44 | daemon | https://basichelp.sipgate.co.uk/hc/en-gb/articles/206404129-Asterisk-PBX |
| 01:58.54 | daemon | replaced all the SIPID/SIP-ID with the userid they gave me |
| 01:59.01 | daemon | and SIPpassword with the pass |
| 02:00.53 | [TK]D-Fender | You haven't proven that you registered |
| 02:01.10 | [TK]D-Fender | Pointing to a guide doesn't proved taht it's right... or that you followed it |
| 02:03.14 | daemon | how do you prove its registered |
| 02:08.35 | [TK]D-Fender | "sip show registry" |
| 02:10.54 | daemon | 0 sip registrations |
| 02:14.21 | [TK]D-Fender | And that would be "failure to do that half of the job" |
| 02:14.41 | [TK]D-Fender | Time to go back ... and follow that guide you said you did |
| 02:14.57 | daemon | I did follow it |
| 02:15.13 | daemon | well its not really a guide its two configs you change id & password in |
| 02:16.05 | [TK]D-Fender | You didn't register |
| 02:16.08 | [TK]D-Fender | So no, you didn't |
| 02:16.27 | daemon | yes, I did. |
| 02:16.41 | daemon | the config is probably for a different version of asterisk or not been updated in years |
| 02:16.54 | daemon | how do you get asterisk to give you debug information so perhaps I can figure out whats wrong |
| 02:17.36 | [TK]D-Fender | https://teamhelp.sipgate.co.uk/hc/en-gb/articles/207414635-How-Do-I-Configure-Asterisk-for-sipgate-trunking- |
| 02:17.43 | *** join/#asterisk matrix1233 (~matrix123@197.2.140.217) |
| 02:17.45 | [TK]D-Fender | There is no debug information |
| 02:17.51 | daemon | its not for trunking |
| 02:17.51 | [TK]D-Fender | You aren't listening you SKIPPED A STEP |
| 02:18.31 | [TK]D-Fender | daemon> if I try to call the phone number its says its not connected <- You have to REGISTER or they don't knwo where to CONTACT YOU |
| 02:18.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 02:19.41 | daemon | right how do I register |
| 02:19.51 | daemon | that guide is for there trunking services |
| 02:19.58 | [TK]D-Fender | Same thing |
| 02:20.08 | daemon | trunking is a paid service |
| 02:20.10 | [TK]D-Fender | ... |
| 02:20.12 | daemon | I do not have that on my account |
| 02:20.26 | [TK]D-Fender | Where you can DIAL doesn't mean a damn bit of difference |
| 02:21.27 | *** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net) |
| 02:26.36 | drmessano | Can I use Asterisk to replace my Obamaphone? |
| 02:26.38 | drmessano | Asking for a friend |
| 02:38.50 | *** join/#asterisk ketas (~ketas@0011-0000-0000-0000-35dc-8408-07d0-2001.dyn.estpak.ee) |
| 02:42.21 | *** join/#asterisk ketas- (~ketas@200-62-46-176.dyn.estpak.ee) |
| 02:47.40 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 02:49.17 | *** part/#asterisk daemon (~daemon@hostname.racing) |
| 02:52.14 | [TK]D-Fender | ...and he's gone |
| 02:52.24 | [TK]D-Fender | So much for hearing if he even tried |
| 03:37.11 | *** join/#asterisk rpifan (~rpi@73.106.73.174) |
| 03:39.00 | Samot | Hello! I'd like to buy a registration... |
| 03:39.25 | Samot | A registration for my pet fish Eric. He isssssss a halibut. |
| 03:39.59 | drmessano | I register all my illegals |
| 03:40.02 | drmessano | Does that count? |
| 03:40.21 | Samot | Are they all named Eric? |
| 03:41.20 | drmessano | They can bee |
| 03:41.29 | Samot | Cleese + Chapman was the gold standard. However, Cleese + Palin was always a riot as well. Cheese Shop, Fish License... |
| 03:42.18 | Samot | And of course, Parrot Skit. |
| 03:44.20 | drmessano | Yes, yes |
| 03:44.56 | drmessano | Strange women, lying in ponds, distributing swords is no basic for a system of government |
| 03:45.17 | drmessano | Supreme executive power should be decided by an electoral college |
| 03:47.26 | drmessano | Clearly the "Lady of the lake" is the state of Florida |
| 03:48.09 | Samot | Lol |
| 03:48.36 | Samot | What was it called? Four Yorkshire Men or something... |
| 03:49.12 | Samot | Where they sat around one upping each other about how rough their childhoods each where. |
| 03:49.29 | drmessano | The one with the shoebox |
| 03:49.36 | Samot | Yes. |
| 03:49.37 | drmessano | Hole in the floor |
| 03:49.43 | Samot | Street |
| 03:49.49 | Samot | Hole in the street |
| 03:50.18 | drmessano | https://www.youtube.com/watch?v=Xe1a1wHxTyo |
| 03:51.07 | Samot | "I'd wake up a half hour before going to bed, work 26 hours at the mill and when we get home our father would kill us and dance on our graves" |
| 03:51.18 | drmessano | Half the floor was missing |
| 03:51.26 | drmessano | All 26 of them huddled into one corner |
| 03:51.32 | drmessano | Thats what I was thinking of |
| 03:51.45 | Samot | Sucking on rockm |
| 03:51.51 | Samot | Rocks |
| 03:52.09 | drmessano | Cardboard box? You were lucky |
| 03:53.27 | Samot | But we where happy in those days |
| 03:53.46 | Samot | Cold cup of tea.. |
| 03:53.56 | Samot | With no milk or surgar... |
| 03:54.02 | Samot | Or tea.. |
| 03:54.15 | drmessano | Had to get up in the morning, at 10 o'clock at night, a half an hour before I went to bed.. |
| 03:54.23 | drmessano | Always loved that line |
| 03:54.30 | Samot | Yup |
| 03:54.50 | drmessano | You try to tell the kids of today that |
| 03:55.07 | Samot | They dont believe you... |
| 03:55.37 | drmessano | I feel like that, actually |
| 03:55.54 | drmessano | Walking 2 miles to school, IN THE SNOW, up a hill, with no iPhone. |
| 03:56.16 | Samot | We had to use....cassettes!!! |
| 03:58.01 | Samot | Back in my day, you could record only two shows at a time.... |
| 03:58.10 | drmessano | We could only listen to ONE ALBUM at a TIME |
| 03:58.20 | Samot | And there was none of this fancy streaming.. |
| 03:58.29 | drmessano | When a band dropped a new mixtape, we called it an ALBUM and we had to go BUY IT |
| 03:58.38 | drmessano | At a STORE |
| 03:58.44 | Samot | No..... |
| 03:59.39 | drmessano | We could only listen to 8-12 songs at a time |
| 03:59.45 | Samot | We had to have phone lines... |
| 04:00.02 | Samot | Real ones |
| 04:00.15 | drmessano | You know what games we had on our phones? TALK TO PEOPLE |
| 04:00.22 | drmessano | That was it |
| 04:00.25 | Samot | And choose if we wanted calls or internet access |
| 04:00.59 | Samot | Your contact list was a notebook |
| 04:01.36 | drmessano | Social Media was the MALL |
| 04:01.55 | Samot | Editing require scratching out old info and writing in new |
| 04:03.02 | Samot | Online banking meant you balanced you check book by hand while chatting in a BBS room. |
| 04:03.22 | drmessano | We didnt show a girl our interest by sending a dick pic.. We would start a conversation |
| 04:03.33 | Samot | Well... |
| 04:03.42 | Samot | that because it took 30 minutes to get the pic. |
| 04:03.46 | drmessano | Speaking of dick pics.. You know how long it took to send one? 4 days! |
| 04:03.47 | Samot | You had to kill time during the transfer. |
| 04:04.00 | drmessano | You had to take it |
| 04:04.01 | drmessano | Get it processed |
| 04:04.06 | drmessano | Wait for it to come back |
| 04:04.09 | drmessano | Then mail it |
| 04:04.19 | drmessano | and hope she was the only one that opened her mail |
| 04:04.33 | Samot | Not make eye contact with the person working the photo booth at the drug store when you picked up your pictures. |
| 04:04.48 | drmessano | YES! lol |
| 04:04.53 | Samot | Because you know they know. |
| 04:05.17 | drmessano | Yes |
| 04:05.21 | drmessano | They know |
| 04:05.43 | Samot | It's why Polaroids made such a come back in the early ages of the Internet. |
| 04:05.52 | drmessano | I remember when getting your photos on Floppy disk was all the rage |
| 04:06.15 | Samot | I got a polaroid and a scanner... LADIES BEWARE |
| 04:06.42 | Samot | I mean sending a dick pic to dedication. |
| 04:07.11 | Samot | There were steps involved. |
| 04:07.26 | drmessano | Remember when you had to give chicks your real number/ |
| 04:07.28 | drmessano | ? |
| 04:07.51 | Samot | Remember how chicks still gave you a fake number? |
| 04:07.57 | Samot | Somethings haven't changed. |
| 04:08.20 | Samot | But instead of the corner party store you just get Lenny now. |
| 04:08.21 | drmessano | Now its like "Text me at umm.. lemme go check the number" |
| 04:09.25 | Samot | Lol. |
| 04:10.17 | Samot | Or fake emaisl. |
| 04:10.20 | Samot | er emails. |
| 04:10.29 | Samot | Because email actually cost money back then... |
| 04:10.44 | drmessano | Not if you had Juno and a floppy disk |
| 04:10.47 | drmessano | lol |
| 04:10.49 | Samot | hahaha. |
| 04:10.57 | Samot | That's right, Juno... |
| 04:11.04 | drmessano | I LOVED juno |
| 04:11.16 | Samot | That's where the rod4booty@ emails went. |
| 04:11.31 | drmessano | I wrote a launcher for Juno in VB.. it would delete all the ads then launch the EXE |
| 04:11.51 | drmessano | JunoAdKiller |
| 04:12.18 | drmessano | That was my legit first app.. and I had to read books to learn how to do it |
| 04:14.12 | drmessano | My parents still use Juno |
| 04:14.30 | drmessano | Well, whats left it.. AKA NetZero email on the juno.com domain |
| 04:15.43 | drmessano | Also wrote an NetZero launcher.. which would delete the JRE and replace java.exe with a zero length, read-only +system file |
| 04:18.28 | Samot | NetZero is still around |
| 04:18.29 | Samot | ? |
| 04:19.14 | drmessano | Its just free email now, i think |
| 04:19.28 | drmessano | I stand corrected |
| 04:19.58 | drmessano | I guess they do still have some limited internet |
| 04:20.02 | drmessano | and paid stuff |
| 04:20.03 | drmessano | wow |
| 04:21.02 | drmessano | Yeah the mobile data stuff too.. I remember that |
| 04:21.33 | drmessano | 200MB Free per month would be cool for shell access |
| 04:22.38 | drmessano | Since they're on the Sprint network, it will take you a month to use 200MB anyway |
| 04:24.08 | drmessano | Damn, imma have a new phone in 2 days |
| 04:40.10 | *** join/#asterisk daemon (~daemon@hostname.racing) |
| 05:20.09 | Akari | bbl |
| 05:20.14 | *** part/#asterisk Akari (akari@unaffiliated/akari) |
| 05:29.23 | *** join/#asterisk Pegasus_RPG (~Icedove@47.142.200.28) |
| 05:30.20 | Pegasus_RPG | Hello. Are DPMA and Digium Configuration Server the same thing? |
| 05:55.05 | [TK]D-Fender | https://www.google.ca/#q=Digium+Configuration+Server+ |
| 05:55.32 | [TK]D-Fender | I see an answer before even following any of the links from the first page of results. |
| 05:57.56 | *** join/#asterisk jkroon (~jkroon@2c0f:f720:1:0:fcb3:e235:8b0a:1f6b) |
| 06:00.08 | Rasputin3711 | Is a ring-group right thing for parallel call to 2 extensions? |
| 06:00.45 | [TK]D-Fender | Dial() can call multiple devices simultaneously |
| 06:00.57 | [TK]D-Fender | Queue() can also call more than 1 device at a time |
| 06:01.00 | [TK]D-Fender | So can Page() |
| 06:01.07 | [TK]D-Fender | Possibly others |
| 06:03.31 | Rasputin3711 | Dial() - is best for performance ? |
| 06:04.09 | [TK]D-Fender | That isn't a valid question |
| 06:04.16 | [TK]D-Fender | There is no miracle "performance". |
| 06:04.22 | [TK]D-Fender | You need to understand what each does |
| 06:07.14 | Rasputin3711 | thx, i think Dial() is a good for my case. |
| 06:18.17 | *** join/#asterisk jkroon (~jkroon@2c0f:f720:1:0:fcb3:e235:8b0a:1f6b) |
| 06:23.55 | drmessano | Rotary() is a terrible option |
| 06:24.03 | drmessano | Takes forever |
| 06:25.24 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 06:28.23 | [TK]D-Fender | Yes, my # is 9900000 |
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| 08:50.25 | UncleKiwi | hi all |
| 08:52.23 | WIMPy | lo you :-) |
| 08:53.38 | UncleKiwi | i spent a whole day troubleshooting a printer |
| 08:53.51 | UncleKiwi | ahaha |
| 08:54.41 | UncleKiwi | WIMPy have you used chan_mobile ? |
| 08:54.43 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 08:57.49 | UncleKiwi | im thing of using chan_mobile in a production environment |
| 08:59.26 | UncleKiwi | i'll get my testing started tomorrow but im just curious about the idea being a good one or not |
| 08:59.49 | UncleKiwi | bluetooth between asterisk and mobile phone |
| 08:59.52 | UncleKiwi | hmmm |
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| 10:12.12 | cmendes0101 | I don't seem to have SipAddHeader. Anyone know what module this is? Poking around menuselect but can't find it |
| 10:15.23 | *** join/#asterisk afournier (~admin@241-117.80-90.static-ip.oleane.fr) |
| 10:15.27 | afournier | hi |
| 10:22.00 | TandyUK | cmendes0101: i didnt realise that was an option |
| 10:22.27 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 10:22.31 | cmendes0101 | So I'm digging a little deeper. I guess that was just something in chan_sip |
| 10:23.16 | cmendes0101 | since I'm using pjsip I would need to set PJSIP_HEADER |
| 10:23.43 | cmendes0101 | more of a setting variable I guess. Docs just mentioned it as a function so not sure totally |
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| 11:23.24 | wyoung | hihihihijiji |
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| 12:54.45 | pawiecki | If I reload my dialplan (connected via 'asterisk -vvvr'), it shows Warnings about some duplicated extensions, but when i reload it again, even after some changes, it will not warn me again. Is that normal? How can I make sure, I'm not missing any Warnings? |
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| 14:01.51 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
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| 14:11.13 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:21.19 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-kxjddfvbfxmzhwiw) |
| 14:21.20 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:49.54 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 15:08.22 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:11.12 | *** join/#asterisk roukoswarf (root@rouk.org) |
| 15:12.17 | roukoswarf | hola, im getting a 403 according to bria, and ioerror according to linphone on a fresh asterisk build, port is open, and no iptables running, debug isnt spitting anything out, any ideas? |
| 15:13.01 | Kuunsi | roukoswarf: configuration? sip debug? (nopaste!) |
| 15:13.01 | roukoswarf | not sure how to go about debugging, tcpdump i guess? |
| 15:13.13 | roukoswarf | nothing hitting sip debug... its spooky. |
| 15:13.52 | roukoswarf | checking if im somehow not getting packets now |
| 15:14.10 | Kuunsi | if you want, tell me IP, and i'll check if port is open |
| 15:14.18 | roukoswarf | rouk.org |
| 15:14.22 | roukoswarf | v4 and v6 |
| 15:14.27 | roukoswarf | been testing on v4 |
| 15:15.28 | roukoswarf | dont seem to be getting any packets... "tcpdump -n udp dst port 5060" |
| 15:15.42 | Kuunsi | 5600/udp open|filtered unknown |
| 15:15.47 | Kuunsi | oh, wait |
| 15:15.50 | Kuunsi | typo |
| 15:15.52 | roukoswarf | yeah, not sure how to make sense of that |
| 15:16.05 | roukoswarf | 10:17:19.602642 IP 81.75.57.27.56556 > 192.95.36.179.5060: SIP |
| 15:16.07 | roukoswarf | is that you? |
| 15:16.35 | roukoswarf | if so, why is my traffic not hitting it. |
| 15:16.50 | Kuunsi | 5060/udp open|filtered sip |
| 15:16.56 | Kuunsi | and yes |
| 15:17.06 | Kuunsi | open|filtered |
| 15:17.08 | Kuunsi | Nmap places ports in this state when it is unable to determine whether a port is open or filtered. This occurs for scan types in which open ports give no response. The lack of response could also mean that a packet filter dropped the probe or any response it elicited. So Nmap does not know for sure whether the port is open or being filtered. The UDP, IP protocol, FIN, NULL, and Xmas scans classify ports |
| 15:17.10 | Kuunsi | this way. |
| 15:17.55 | roukoswarf | so, what am i doing wrong? does asterisk not respond globally by default? |
| 15:18.01 | roukoswarf | am new to asterisk |
| 15:18.43 | Kuunsi | are you _sure_ there is no iptables running? |
| 15:18.57 | Kuunsi | what does `iptables -S` give you? |
| 15:19.27 | Kuunsi | also, check `netstat -tulpe` to make sure asterisk is listening there (and not some other tool |
| 15:19.41 | roukoswarf | 3 lines of accept, the iptables service is disabled. |
| 15:20.08 | roukoswarf | udp 0 0 0.0.0.0:sip 0.0.0.0:* asterisk 11686 638/asterisk |
| 15:20.36 | Kuunsi | show me your sip.cof |
| 15:20.39 | Kuunsi | conf* |
| 15:20.47 | roukoswarf | any pefered paste server? |
| 15:21.20 | roukoswarf | i cut the config down to nothing and used some random one i found online, but its the same behavior as the default configs. |
| 15:21.25 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 15:22.16 | Kuunsi | ~pb |
| 15:22.16 | infobot | [pastebin] 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:23.17 | roukoswarf | https://ptpb.pw/Dw8U |
| 15:25.06 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:25.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:26.37 | roukoswarf | any ideas? |
| 15:27.13 | roukoswarf | https://ptpb.pw/2MrH |
| 15:27.14 | roukoswarf | this too |
| 15:27.50 | [TK]D-Fender | dialplan has nothing to do with 403 |
| 15:27.55 | [TK]D-Fender | 403 is an auth refusal |
| 15:28.11 | [TK]D-Fender | "sip set debug on" |
| 15:28.17 | [TK]D-Fender | prove the call attempt is arriving |
| 15:28.43 | roukoswarf | i think the client may be bonkers, as im getting 403 on bria on my cell, and ioerror on linphone on my desktop |
| 15:28.58 | roukoswarf | and no debug ever appears, even at core 5 sip 5 |
| 15:29.28 | roukoswarf | im not sure if my network here is crazy or something |
| 15:29.45 | roukoswarf | i can register to other sip servers fine, and this chat is relayed through that server. |
| 15:29.54 | Kuunsi | sip debug has no levels |
| 15:30.17 | roukoswarf | i mean verbose 5 core 5 |
| 15:30.19 | [TK]D-Fender | There is no "sip 5" |
| 15:30.36 | [TK]D-Fender | <roukoswarf> i mean verbose 5 core 5 <- and that is NOT what I told you to do |
| 15:30.46 | [TK]D-Fender | <[TK]D-Fender> "sip set debug on" <---------------------- |
| 15:31.13 | roukoswarf | yeah, when i tested it with verbose i also put sip debug on. |
| 15:31.31 | roukoswarf | just put it on again, got 403, no debug in shell. |
| 15:31.41 | roukoswarf | tested from another network too. |
| 15:33.42 | roukoswarf | thats why im here, i dont understand whats going on. |
| 15:34.17 | [TK]D-Fender | "sip show settings" <- |
| 15:34.24 | [TK]D-Fender | ~pb |
| 15:34.24 | infobot | well, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:34.26 | [TK]D-Fender | ^^^ |
| 15:35.30 | roukoswarf | yes, im aware of not pasting 300 lines here |
| 15:36.24 | roukoswarf | https://paste.debian.net/hidden/acfa1fc0/ |
| 15:38.06 | *** join/#asterisk miralin (~Thunderbi@194.8.128.51) |
| 15:38.57 | [TK]D-Fender | Is your * behind NAT? |
| 15:39.08 | [TK]D-Fender | Where is the cliet relative to your server? |
| 15:39.54 | roukoswarf | client is behind nat, server is public |
| 15:40.19 | roukoswarf | need to enable something for nat traversal? |
| 15:40.45 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:41.16 | [TK]D-Fender | So your server has a public IP directly on it? |
| 15:41.22 | roukoswarf | yes, of course. |
| 15:41.32 | [TK]D-Fender | "iptables --list" |
| 15:41.50 | roukoswarf | server is a vm on a shared adapter |
| 15:42.10 | roukoswarf | no rules, default accept on all. |
| 15:42.11 | [TK]D-Fender | needs to be bridged witha unique IP |
| 15:42.27 | roukoswarf | not when youre sharing an adapter with macvtap |
| 15:42.35 | [TK]D-Fender | ... |
| 15:42.38 | roukoswarf | works for the other 100 vms, including this one. |
| 15:43.00 | [TK]D-Fender | that is a BAD assumption for UDP |
| 15:43.46 | roukoswarf | really dont have anything running over udp usually, but havnt hit any weirdness yet. |
| 15:44.05 | [TK]D-Fender | SIP = UDP |
| 15:44.21 | roukoswarf | yeah, first sip server im messing with on this setup |
| 15:44.36 | roukoswarf | could tcp it, but would be kinda stupid. |
| 15:45.03 | roukoswarf | i cant find any issues with macvtap+udp. do you have any info on that? |
| 15:46.39 | *** join/#asterisk rwb (~Thunderbi@65-183-131-95-dhcp.burlingtontelecom.net) |
| 15:47.19 | igcewieling | roukoswarf: does the VM have a unique public IP? |
| 15:47.27 | roukoswarf | yes, of course. |
| 15:47.52 | igcewieling | What is the problem? |
| 15:48.22 | roukoswarf | bria is throwing me a 403, no debug is printed from asterisk, dont see any packets hitting it at all. |
| 15:48.28 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ctzqtmgbtjhkrnas) |
| 15:48.28 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:48.47 | igcewieling | I assume you are not using Verizon Wireless for internet access? |
| 15:48.48 | roukoswarf | saw other people hit 5060 with packets, but not mine, and i tried from multiple networks at this point. |
| 15:49.01 | roukoswarf | nope, bell mobility, cogeco cable, rogers. |
| 15:49.21 | roukoswarf | have sip working on another server on the exact same devices and networks |
| 15:49.24 | igcewieling | ah, you live in a civilized part of north america. |
| 15:49.44 | igcewieling | do you see the packets if using tcpdump instead? |
| 15:49.46 | roukoswarf | yeah, no trumperino here |
| 15:50.31 | roukoswarf | thats the thing... i dont. |
| 15:50.47 | roukoswarf | tcpdump -n udp dst port 5060 yields nothing |
| 15:51.03 | roukoswarf | but, i did see traffic when someone else here tested it. |
| 15:51.25 | roukoswarf | here being the channel |
| 15:51.34 | igcewieling | then the problem is NOT with asterisk or your server. |
| 15:52.15 | roukoswarf | its with 3 whole isps and only when dealing with a single sip server? |
| 15:52.39 | igcewieling | maybe the problem is your client?: |
| 15:53.21 | roukoswarf | both linphone and bria are registering to other sip servers at the time of doing this |
| 15:53.43 | roukoswarf | both are giving different errors |
| 15:53.44 | igcewieling | Well, I wish you the best of luck then. |
| 15:54.01 | roukoswarf | RIP |
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| 16:03.38 | *** part/#asterisk afournier (~admin@241-117.80-90.static-ip.oleane.fr) |
| 16:07.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:07.14 | Kuunsi | roukoswarf: maybe test if tcp is working? |
| 16:07.35 | Kuunsi | if yes, macvtap fails at udp |
| 16:09.43 | WIMPy | use tcpdump on the host. |
| 16:20.21 | roukoswarf | Kuunsi: igcewieling: WIMPy: [TK]D-Fender: I tried tcp, didnt work, figured it out though, i had SRV records on the domain pointing sip for the domain to sip.rouk.org, and the clients were silently following it |
| 16:20.50 | igcewieling | so, the packets were never getting to your server? |
| 16:20.52 | roukoswarf | sip.rouk.org was an old alias to not-the-server-i-wanted |
| 16:21.03 | roukoswarf | correct, it was a client+dns issue. |
| 16:21.05 | [TK]D-Fender | That would do it |
| 16:21.38 | roukoswarf | so thanks all, i completely forgot the dns for that was there, and didnt know they followed it. |
| 16:21.58 | roukoswarf | put in the ip address manually and it worked, thats how i figured it out |
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| 17:21.18 | igcewieling | heh, there must be some new telnet exploit. connect attempts to port 23 are exceeding the connect attempts to port 5060 |
| 17:22.04 | TandyUK | is port 23 even open? |
| 17:22.09 | TandyUK | if so, WHY?? |
| 17:22.19 | igcewieling | huh? it isn't open. |
| 17:22.40 | igcewieling | that's why I see the connect attempts...in the iptables logs. |
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| 17:35.11 | daemon | hey all can anyone give me a hand setting up a connection to a sip provider (not trunk) I want it so that my bog standard voip account (sipgate.co.uk) when called asterix answers on my server, I used there config from https://basichelp.sipgate.co.uk/hc/en-gb/articles/206404129-Asterisk-PBX added the 'register' to [global] and then the two relevent further down, I replaced SIPID and SIPassword thats marked bold in template as shown did not replace |
| 17:35.11 | daemon | SIP-ID etc further down assumed those was templates (as not bold) |
| 17:35.25 | daemon | never seems to show up in sip registers though |
| 17:37.17 | daemon | using logging level 8 never see any warnings about the connection or failed to register |
| 17:37.58 | daemon | the connection shows up in sip show peers as 'unmonitored', at the summary at the bottom it shows up as Unmonitored: 1 online |
| 17:38.05 | daemon | so perhaps its the registration that is wrong somehow? |
| 17:38.31 | [TK]D-Fender | Told you yesterday how to prove if * is even trying... |
| 17:38.35 | [TK]D-Fender | "sip show registry" <- |
| 17:39.37 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 17:41.40 | daemon | yep 0 registrations but I know register is in the right place, https://paste.ee/r/DURpi |
| 17:41.46 | daemon | checked it in docs |
| 17:43.21 | igcewieling | I see the problem. |
| 17:43.36 | igcewieling | try putting your register in sip.conf not extensions.conf. |
| 17:43.53 | daemon | ah would make more sence |
| 17:44.08 | [TK]D-Fender | yes, that would be a rather large failure |
| 17:44.12 | *** join/#asterisk jfindley (~jfindley@104-181-196-33.lightspeed.hstntx.sbcglobal.net) |
| 17:44.20 | *** join/#asterisk karelk (~karel@31.10.149.26) |
| 17:44.37 | daemon | sipgate.co.uk:5060 N MyUsername 105 Registered Wed, 16 Nov 2016 17:44:24 |
| 17:44.44 | daemon | perfect! I best send them an email |
| 17:44.59 | jfindley | Any ideas why, on a busy server, when playing back a 2 minute long audio file, Asterisk closes the AGI connection? |
| 17:45.40 | jfindley | usually within 30 to 45 seconds |
| 17:45.41 | igcewieling | jfindley: no idea, but I can give a suggestion to help you troubleshoot it. |
| 17:45.41 | *** join/#asterisk Pegasus_RPG (~Icedove@47.142.200.28) |
| 17:45.56 | jfindley | igcewieling: Sure |
| 17:46.07 | daemon | igcewieling, thank you :) |
| 17:46.19 | igcewieling | Are you using PHP for your AGI? |
| 17:46.29 | jfindley | no, Python and FastAGI |
| 17:47.00 | Pegasus_RPG | Has anyone else seen a problem where DPMA/DCS traffic is not being seen by the server when the phone is on a different subnet? I'm having the same problem as this person: http://community.freepbx.org/t/digium-d40-cant-contact-dpma-no-multicast/21935 |
| 17:47.18 | Pegasus_RPG | The crazy thing is that regular SIP works fine |
| 17:47.41 | Pegasus_RPG | by that I mean, I can manually enter the account details in the Digium phone and it registers and can make calls just fine |
| 17:49.14 | igcewieling | I know this is a hassle, but: stop Asterisk. Then start asterisk as "asterisk -cvvv" then issue an "agi set debug on" command. This will run Asterisk in the foreground and will make stderr from your AGI to be seen on the console. This means ctrl-C to exit asterisk will kill all calls. a ctrl-s to pause output will pause Asterisk. |
| 17:49.25 | igcewieling | ^^ for jfindley |
| 17:49.29 | Pegasus_RPG | A packet capture on the firewall on the server's LAN shows the DCS packets headed toward the server but a capture on the server doesn't show them at all. The only thing between the FW and the server is a switch stack |
| 17:49.59 | Pegasus_RPG | And again, regular SIP works fine, so WTF else can I check? |
| 17:50.20 | jfindley | igcewieling: Would running it myself make much of a difference as opposed to asterisk -vvvvvvvvr? |
| 17:50.50 | igcewieling | define "running it myself" |
| 17:51.23 | igcewieling | asterisk -rvvv won't let you see STDERR output from your script, which makes debugging a living hell. |
| 17:51.54 | jfindley | with asterisk -cvvvvv I start an asterisk process with CLI, whereas with asterisk -vvvvvr I connect to an already running asterisk process (managed by the OS) |
| 17:52.22 | jfindley | i see, I'll give that a try |
| 17:52.33 | igcewieling | correct. asterisk -cvvv + "agi set debug on" will show you STDERR output from your script. |
| 17:53.15 | igcewieling | There could be other non-agi related issues, but it is best to resolve any AGI issues before troubleshooting more. |
| 17:53.45 | igcewieling | 3/4 of my day is sometimes spent writing AGIs in PHP |
| 17:55.35 | jfindley | yep, same here except with Python |
| 18:05.30 | *** join/#asterisk makowals_ (~makowals@unaffiliated/makowals) |
| 18:06.27 | *** part/#asterisk makowals_ (~makowals@unaffiliated/makowals) |
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| 18:15.46 | daemon | hey all I expected to get 'demo' when I called instead of I got this notice: https://paste.ee/r/nedQR I expect I need to add 2140753e0 as an extension in the public context for it to auto answer with the demo? |
| 18:16.36 | daemon | I assume the context will always be public because its handling inbound calls, as a note the asterix server is firewalled off from the rest of the net it can only receive calls from that host |
| 18:34.21 | [TK]D-Fender | Call from 'SIP-ID' (217.10.79.23:5060) to extension '2140753e0' rejected because extension not found in context 'public'. |
| 18:34.31 | [TK]D-Fender | means what it says. |
| 18:34.43 | [TK]D-Fender | it matched a peer and was sent into a context in the dialplan |
| 18:35.00 | [TK]D-Fender | the extension it is looking for in htere based on the call does not have a match there to process it |
| 18:35.12 | daemon | ah I see |
| 18:41.21 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 18:43.23 | daemon | wow asterisk gets alot of random scans |
| 18:43.31 | daemon | 02005 11 8619 deny udp from any to me dst-port 5060,4569,2727,4520 in via re0 |
| 18:43.38 | daemon | only been really up 30 minutes |
| 18:43.53 | [TK]D-Fender | Would you like to have a few deliberate ones to balance those out? |
| 18:44.02 | daemon | 02001 47 31872 allow udp from 217.10.79.23 to me dst-port 5060,4569,2727,4520 in via re0 |
| 18:44.03 | daemon | ;) |
| 18:44.37 | [TK]D-Fender | Like an ex of mine used to joke: Got a problem? No? You want one? |
| 18:44.37 | daemon | I got it to play me the demo do not think I went about it the right way though |
| 18:44.46 | daemon | going to have to have a read through |
| 18:44.51 | igcewieling | daemon: well how else would the hack in before you have a chance to install the OS updates? 8-) |
| 18:45.30 | daemon | lol true, for some reason I never expected the asterisk connect attempts to be faster than the vnc weakpass ones though |
| 18:46.43 | daemon | speaking of versions |
| 18:47.05 | daemon | wonder if I am using the latest |
| 18:47.14 | [TK]D-Fender | <PROTECTED> |
| 18:47.17 | [TK]D-Fender | ^ |
| 18:47.18 | daemon | I appear to be using the LTS version |
| 18:47.24 | [TK]D-Fender | AN LTS version |
| 18:48.03 | daemon | hmm I cannot even see a 14.x branch in ports |
| 18:48.14 | igcewieling | daemon: on some of my servers, I allow all connections and registrations and send unwanted ones to a jail context. seems to reduce attempts quite a bit. |
| 18:48.31 | daemon | H |
| 18:49.04 | daemon | I did notice they all seem to be trying to connect to that 'test set' of accounts |
| 18:49.24 | daemon | well was |
| 18:57.56 | *** join/#asterisk pigpen (~n5yzv@216-177-181-17.block0.gvtc.com) |
| 19:02.44 | Kuunsi | daemon: i always find pleople trying to place international calls using my asterisk instance |
| 19:03.15 | daemon | I bet people try to get access to them and ring premium rate numbers they have setup |
| 19:03.29 | Kuunsi | but i've set up fail2ban to iptables -j DROP those IPs after three failed call/registration attempts |
| 19:03.45 | Kuunsi | that's what they try to do |
| 19:04.17 | Kuunsi | however, maximum damage they could to is about 4â¬, which is all i got on my POTS SIP Provider |
| 19:05.05 | daemon | luckily mine could only be called from one place ... but it gave me an idea, I do run a private vpn for 9-10 friends |
| 19:05.24 | daemon | might make them all accounts could be fun for games |
| 19:05.50 | daemon | only accessible over the vpn |
| 19:05.53 | daemon | of course :) |
| 19:06.56 | Kuunsi | you actually can place a unauthenticated call onto my asterisk, calling sip:1000@kaworu.kunbox.net - you'll get a notice about this call being recorded, and then get directed to my office phone |
| 19:07.20 | Kuunsi | set this up some time ago so my parents would be able to call me for free |
| 19:08.18 | daemon | ah you ever get any interesting calls from around the glove? |
| 19:08.20 | daemon | globe*? |
| 19:08.44 | Kuunsi | i remember someone calling me from israel |
| 19:09.13 | daemon | they have anything intersting to say or incompatible language wise to determine |
| 19:09.31 | Kunsi | i don't remember the call contents |
| 19:09.44 | Kunsi | anyways, i'll have dinner now |
| 19:09.54 | daemon | ok, have a nice meal :) |
| 19:14.58 | daemon | get all if I add the contents of [demo] into [public] and add my extension to the start of the 'exten' callout I get the demo when I call it, but if I add: exten => 2140753e0,1,Goto(demo) in [public] the error states: Priority 'demo' must be a number > 0, or valid label |
| 19:15.12 | daemon | but [demo] is the stock demo and is in extensions.conf |
| 19:15.15 | daemon | did I miss something |
| 19:15.43 | daemon | I assume it has to be a valid label |
| 19:16.46 | daemon | hmm even adding a priority does not seem to change its mind, simply states: Channel 'SIP/sipgate.co.uk-00000003' sent to invalid extension but no invalid handler: context,exten,priority=public,demo,1 |
| 19:17.13 | daemon | because the context is still public and thus cannot access demo? |
| 19:25.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 19:32.43 | jeffspeff | if max_contacts in aor is set to 0 does that allow unlimited contacts or zero contacts? |
| 19:33.15 | [TK]D-Fender | daemon, itt is telling you where it is sending the call and exactly what it is looking for |
| 19:39.18 | daemon | oops think I brok something |
| 19:39.25 | daemon | Serious Network Trouble; __sip_xmit returns error for pkt data |
| 19:42.29 | [TK]D-Fender | Sounds like a firewall issue |
| 19:42.42 | [TK]D-Fender | Usually something being blocked from transmitting |
| 19:42.47 | daemon | I thought so too so I temporary removed the rules |
| 19:45.00 | *** join/#asterisk rwb (~Thunderbi@65-183-131-95-dhcp.burlingtontelecom.net) |
| 19:47.39 | *** join/#asterisk Pegasus_RPG (~Icedove@47.142.200.28) |
| 19:50.18 | *** join/#asterisk Pegasus_RPG (~Icedove@47.142.200.28) |
| 19:51.30 | daemon | reverted to origianl config |
| 19:51.32 | daemon | works fine now |
| 19:51.34 | daemon | wonder what I caught |
| 19:52.22 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 19:54.41 | *** join/#asterisk BeachBall (~BeachBall@unaffiliated/beachball) |
| 19:57.17 | BeachBall | i have a video conference unit, connection to my asterisk box, when I dial a sip address from the video conf unit, asterisk shows as unknown extension. I'm not sure how to format the extension - i'm using a test one of 201@ideasip.com since this will change, how might I add in a dialplan for this? |
| 19:58.44 | [TK]D-Fender | Show the call |
| 20:00.07 | BeachBall | breif output, or detailed debugging? |
| 20:00.41 | [TK]D-Fender | verbose 10, SIP debug obviously |
| 20:01.52 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:01.52 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:06.21 | BeachBall | http://pastebin.com/8zM0jT0P |
| 20:06.51 | BeachBall | I don't have a dialplan in place for these types of calls, because i'm not sure how to format it. |
| 20:07.14 | [TK]D-Fender | [2016-11-16 20:03:48] NOTICE[26578]: res_pjsip_session.c:2125 new_invite: Call from '7000' (UDP:206.162.174.29:5060) to extension '201' rejected because extension not found in context 'internal'. |
| 20:07.24 | [TK]D-Fender | Thre is no such thing as "these type of calls" |
| 20:07.40 | [TK]D-Fender | SIP is SIP and it hits the dialplan where the peer tells it to with the number they are requesting |
| 20:07.47 | [TK]D-Fender | it is telling you EXACTLY what & where there |
| 20:07.53 | [TK]D-Fender | So what's the mystery? |
| 20:08.25 | BeachBall | how to format a dialplan that will be used when dialing a sip address that changes |
| 20:08.46 | [TK]D-Fender | There is no address |
| 20:08.52 | [TK]D-Fender | it is dialing 201 |
| 20:08.59 | [TK]D-Fender | and it hits your dialplan |
| 20:09.18 | BeachBall | i'm testing with 201@ideasip.com but tomorrow it might be 2112@beachball.com |
| 20:09.36 | [TK]D-Fender | Yes, but * knows NOTHING of domain there |
| 20:09.42 | [TK]D-Fender | * is not a SIP router |
| 20:10.09 | [TK]D-Fender | it will hit based on the pre-@ target in the To: |
| 20:10.10 | [TK]D-Fender | From: "7000" <sip:7000@159.203.17.27>;tag=plcm_1684764072-283312607;epid=Q60949BE3037231C |
| 20:10.10 | [TK]D-Fender | To: <sip:201@ideasip.com>;tag=9cc78d30-9f7f-4639-988c-e10679c7c182 |
| 20:10.22 | [TK]D-Fender | Now if you want something you can dial out then go chop that up |
| 20:10.35 | [TK]D-Fender | "core show functions like SIP" |
| 20:10.36 | [TK]D-Fender | ^ |
| 20:10.49 | *** join/#asterisk jkroon (~jkroon@2c0f:f720:1:0:fcb3:e235:8b0a:1f6b) |
| 20:10.49 | [TK]D-Fender | Very obvious ones for getting that header so you do this |
| 20:12.48 | BeachBall | so since 201 is not internal to me, and I have no trunk to it... how can my call go through? |
| 20:13.12 | [TK]D-Fender | that is a URI |
| 20:13.16 | [TK]D-Fender | You dial it |
| 20:13.57 | BeachBall | that gives me something to google |
| 20:21.43 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 20:28.31 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:28.31 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:30.00 | daemon | hey all can anyone take a peak at this and explain what is going on, I understand the error but I do not really understand what I can do about it: https://paste.ee/r/fGf2V |
| 20:30.31 | daemon | I tried changing the context that those calls are in but then just got: |
| 20:30.35 | daemon | Nov 16 20:22:44] NOTICE[101290][C-00000009]: chan_sip.c:26258 int handle_request_invite(struct sip_pvt *, struct sip_request *, struct ast_sockaddr *, uint32_t, int *, const char *, int *): Call from 'CompanyCall' (217.10.79.23:5060) to extension '2140753e0' rejected because extension not found in context 'demo' |
| 20:30.56 | daemon | I take it thats because there is not exten 2140753e0m.... in demo |
| 20:31.03 | daemon | but I assumed demo should respond to any extension |
| 20:38.45 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:40.29 | jfindley | I think I figured out my broken pipe issue. Is there a timer for FastAGI connections that are established that, after a certain amount of time without a response the connection will timeout? |
| 20:48.24 | [TK]D-Fender | <daemon> but I assumed demo should respond to any extension <- because? |
| 20:48.36 | daemon | because its a demo |
| 20:49.39 | [TK]D-Fender | it's a dumb name |
| 20:49.43 | [TK]D-Fender | it may as well be "fred" |
| 20:49.55 | [TK]D-Fender | it has no special properties |
| 20:50.01 | [TK]D-Fender | it has only what you put into it |
| 20:51.06 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 20:51.23 | kfife | exten => _6[34][34],1,SayAlpha(G) |
| 20:51.23 | kfife | should match 634, 633, 643, and 644 |
| 20:51.27 | kfife | BUT |
| 20:51.33 | kfife | But what if I only want to match 633 or 644? |
| 20:51.54 | [TK]D-Fender | Then you're going to have to make 2 extens |
| 20:52.08 | kfife | Is it possible to do that with one expression, not two seprate... D'oh You beat me to it. |
| 20:52.15 | [TK]D-Fender | no |
| 20:52.18 | kfife | I was hoping for an OR |
| 20:52.19 | daemon | no way to use full perl-re's ? |
| 20:52.38 | [TK]D-Fender | no |
| 20:53.15 | kfife | anyone know if 900 and 976 are an exhaustive list of premium rate prefiexes in NANP? |
| 20:53.50 | [TK]D-Fender | There are several countries in NANPA that can FUBAR you |
| 20:54.49 | kfife | e.g. our beloved Canadians to the north can have a premium prefix, or are you just talking about tolls |
| 20:55.04 | kfife | e.g. unknowingly dialing 'international' |
| 20:55.17 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 20:56.20 | Dovid | Hi. I see in https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration that there are 3 drivers. In extconfig.conf.sample I see options for ODBC, SQlite etc. If I wanted real time MOH and I wanted something lite could I just use sqlite? |
| 20:56.31 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 20:56.31 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 20:56.33 | kfife | Who knew Texas was in North America? |
| 20:57.13 | kfife | Or Vermont. Damned secessionists. |
| 20:57.18 | kfife | adjusts Vermont t-shirt |
| 20:57.44 | daemon | ah ha finally got it to answer with a menu and such :) https://paste.ee/r/r5Bio ... I do this right? |
| 20:58.05 | Kunsi | kfife: 6(33|44) |
| 20:58.18 | kfife | Woohoo! |
| 20:58.34 | kfife | tips hat to Kunsi |
| 20:58.51 | Kunsi | may work, please try |
| 20:59.07 | kfife | do you mean [33|44]? |
| 20:59.14 | kfife | or literally paren |
| 20:59.51 | Kunsi | () |
| 20:59.57 | kfife | roger |
| 21:00.08 | Kunsi | [33|44] would meam 3,4 or | |
| 21:00.15 | Kunsi | [33|44] would meam 3, 4 or | |
| 21:00.34 | Kunsi | (33|44) should do 33 or 44 |
| 21:04.07 | kfife | exten => _19(00|76)XXXXXXX,1,Playback(tt-monkeys) |
| 21:04.14 | kfife | doesn't seem to work |
| 21:04.38 | kfife | Maybe there's a better approach. |
| 21:06.38 | kfife | if it's even possible. |
| 21:07.39 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:07.39 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 21:11.47 | *** join/#asterisk rwb (~Thunderbi@65-183-131-95-dhcp.burlingtontelecom.net) |
| 21:17.13 | [TK]D-Fender | I'm really not sure what part of "no" left room for confusion. |
| 21:17.30 | [TK]D-Fender | <[TK]D-Fender> no |
| 21:17.37 | [TK]D-Fender | <[TK]D-Fender> Then you're going to have to make 2 extens |
| 21:17.56 | [TK]D-Fender | *NOT* .... wait for it ... *POSSIBLE* |
| 21:18.17 | *** join/#asterisk covalschi (~covalschi@212.28.74.160) |
| 21:18.24 | [TK]D-Fender | How are we on this concept now? |
| 21:19.39 | kfife | Any idea why it's not possible? Seems liek a reasonable request. |
| 21:20.41 | kfife | You're right though. It doesn't work. |
| 21:20.54 | [TK]D-Fender | It's nothing to do with "reasonable request" |
| 21:20.55 | kfife | I asked for the monkeys, and I got a different recording ;-) |
| 21:21.00 | [TK]D-Fender | It is NOT that smart |
| 21:21.05 | [TK]D-Fender | it was not made that way. |
| 21:21.32 | [TK]D-Fender | BECAUSE CODE |
| 21:21.52 | daemon | is there a text2speech in asterisk I can use or do I need to go find my microphon :) |
| 21:22.11 | kfife | ?? |
| 21:22.18 | kfife | Why would you need a microphone to do text to speech? |
| 21:22.43 | kfife | Just to play with while you're listening maybe. |
| 21:22.48 | daemon | well may as well go for the extra effort I just wondered if asterisk had anything internal to do it |
| 21:23.07 | [TK]D-Fender | There are 3rd party TTS's |
| 21:23.13 | [TK]D-Fender | * does not provide any itself |
| 21:23.17 | daemon | ah ok |
| 21:23.26 | [TK]D-Fender | Festival is the OSS one of choice |
| 21:23.46 | [TK]D-Fender | Cepstral is cheap (comparatively) but not free, and better |
| 21:24.10 | [TK]D-Fender | If you don't need your text converted live I'm sure there are several sites that will do it for free online |
| 21:24.19 | daemon | festival is in ports I will give it a spin, prtty sure mic is in loft |
| 21:25.13 | [TK]D-Fender | Again.. you're doing TEXT ... *to* SPEECH* |
| 21:25.29 | [TK]D-Fender | Why would you need a MICROPHONE? |
| 21:25.55 | daemon | because the choice was: use text2speech generate so menu voice or do it my self |
| 21:25.58 | [TK]D-Fender | Or were you just going to do the voice yourself? |
| 21:25.58 | daemon | with no text2speech |
| 21:26.06 | daemon | yep :) |
| 21:26.16 | [TK]D-Fender | And if you have a phone of any kind up on your *.. you can jsut record it directly |
| 21:26.21 | daemon | probably get the mrs to do that though .... women sound nicer on the phone |
| 21:26.40 | daemon | oh really |
| 21:28.02 | daemon | ah neat Record(asterisk-recording%d:ulaw) |
| 21:30.02 | kfife | Can asterisk do any ASR without a 3rd party corpus, such as basic number recog? |
| 21:30.35 | kfife | Lumenvox was the shizzy years ago, but it's a lot of overhead for "Press or Say 1" |
| 21:30.52 | [TK]D-Fender | * does no ASR or TTS |
| 21:30.57 | [TK]D-Fender | Everything is bolt-on |
| 21:31.40 | kfife | What's the recommended corpus today? |
| 21:31.44 | [TK]D-Fender | OSS ASR = CMU Sphinx (easily found if one takes 5 seconds to google) |
| 21:31.53 | [TK]D-Fender | thats for the cheapest quality. |
| 21:31.59 | [TK]D-Fender | Lumenvox is MUCH better |
| 21:32.07 | [TK]D-Fender | and on that note... |
| 21:32.11 | kfife | It does text-to-screech with playback(tt-monkeys) |
| 21:32.14 | [TK]D-Fender | heads off on his way home |
| 21:32.16 | kfife | Wah, wahhhh. |
| 21:32.36 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 21:33.37 | daemon | what do I need to change to get asterisk to record the sound sample somewhere else |
| 21:33.50 | daemon | or whre is the path set... or do I jsut enter a full path in Record() |
| 21:35.08 | daemon | ah got it |
| 21:36.44 | daemon | and it worked christ thats cool |
| 21:47.43 | *** join/#asterisk ipengineer (~zconkle@71.252.134.63) |
| 21:48.31 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:56.04 | ipengineer | Does anyone have any ideas as to why a queue call gets stuck on the first agent and never fails over to the second available member? I basically have a queue with two member, one has a penalty of 8, the other a penalty of 9. When a queue call comes in the call is delivered to the agent with a penalty of 8 but if he is unavailable the call is never sent to the second member. The ring stategy is set to ring all. In short, the call basically gets stuck wi |
| 21:56.05 | ipengineer | the first agent the queue thinks should answer the call. |
| 21:57.13 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:57.13 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 22:00.22 | Dovid | When using realtime MOH and an external script such as madplay if in MusicOnhold we have set cachertclasses=yes, after an Asterisk restart the first time a call comes in madplay is started. If ten people are using the same moh class there is only one instance of madplay. The issue is that if no one is using the MOH class once there is an instance of madplay it's upforever till Asterisk is restarted. On the other |
| 22:00.22 | Dovid | <PROTECTED> |
| 22:00.22 | Dovid | <PROTECTED> |
| 22:06.44 | DivideBy0 | Dovid: so it keeps as many instances of madplay running as your max concurrent playing moh? |
| 22:07.53 | DivideBy0 | I'm about to walk out the door, but I'll fool with it this evening |
| 22:23.05 | *** join/#asterisk Eloy (~eloy@51.37.74.42) |
| 22:23.57 | daemon | hey all I request my users to enter a pin, I then want to nip off and do a lookup (not in an asterisk related system) can I do something like call a perl script and return a code for what the ivr should do |
| 22:25.09 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 22:25.40 | daemon | ah I see I can use realtime sip and hook it to postgres |
| 22:31.59 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 22:39.51 | rrittgarn | Daemon, in asterisk do core show function SHELL |
| 22:41.46 | daemon | ah |
| 22:41.49 | daemon | thank you |
| 22:48.11 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-qeiylcydnmjgprpx) |
| 22:48.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 22:54.32 | daemon | WaitExten() seems to wait for me to enter 1 number, is there anything that will wait for me to enter 4? |
| 23:03.32 | Dovid | Divideby0: where cachertclasses=no it does |
| 23:03.45 | Dovid | if it's set to yes then when the call is over madplay stays up |
| 23:10.24 | daemon | ah ha got it, funny how fast you can be swearing at your own voice doing an IVR lol |
| 23:18.43 | covalschi | guys does it have sense to switch to Asterisk from Freeswitch to send faxes? |
| 23:19.20 | covalschi | is there any significant different between mod_spandsp and Asterisk fax? |
| 23:19.28 | covalschi | difference* |
| 23:33.33 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:35.45 | *** join/#asterisk Oatmeal (~Suzeanne@2602:306:3676:c60:5058:3d2f:50ab:8964) |
| 23:36.04 | kfife | FFA is not supported anymore |
| 23:36.39 | kfife | That's the difference |
| 23:46.09 | daemon | hey all can anyone tell me why this always goes to CompanyPBX_balance https://paste.ee/r/4e4bt |
| 23:46.15 | daemon | when it should be going to timeout |
| 23:47.50 | [TK]D-Fender | last I saw it's = not == |
| 23:48.08 | daemon | [TK]D-Fender, I changed it from = when it did not work I assumed like perl it wanted == |
| 23:48.09 | [TK]D-Fender | And you should be showing the CALL proving what's getting executed |
| 23:48.24 | [TK]D-Fender | OH... and your "" is wrong |
| 23:48.29 | [TK]D-Fender | those are taken as LITERAL chars |
| 23:48.30 | daemon | should be "1" |
| 23:48.38 | [TK]D-Fender | and they don't exist on BOTH sides of the equation |
| 23:49.59 | [TK]D-Fender | Later you'll be able to reduce this from 5 lines .. to 2 |
| 23:50.51 | daemon | https://paste.ee/r/HS51p changed it back to the original with = " |
| 23:50.58 | daemon | still showing th same behaviour |
| 23:51.01 | daemon | how do I log calls |
| 23:51.53 | [TK]D-Fender | * CLI |
| 23:51.57 | [TK]D-Fender | where you should be full-time |
| 23:52.04 | [TK]D-Fender | verbose 10 |
| 23:52.21 | [TK]D-Fender | Show the call |
| 23:52.33 | *** join/#asterisk willwh (~willwh@unaffiliated/willskills) |
| 23:54.07 | *** join/#asterisk rwb (~Thunderbi@65-183-131-95-dhcp.burlingtontelecom.net) |
| 23:54.10 | daemon | I have debug set at 10, I cannot see a menu option for call |
| 23:54.37 | [TK]D-Fender | What do you mean 'menu option" |
| 23:54.41 | [TK]D-Fender | Sit in CLI. PLACE THE CALL |
| 23:54.53 | [TK]D-Fender | and I do NOT mean "core debug" |
| 23:55.01 | [TK]D-Fender | "core set verbose 10" <---------- |
| 23:56.30 | daemon | https://paste.ee/r/D6OWl |
| 23:56.45 | daemon | is it because of the \n |
| 23:56.52 | daemon | from echo by the looks of that |
| 23:57.19 | [TK]D-Fender | Certainly does, doesn't it... |
| 23:59.54 | daemon | yep it was :) |