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| 01:00.07 | RadicalDev | I guess the best test is to try a noop and look for a broken pipe? |
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| 01:48.22 | Samot | CEL will show that information. |
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| 01:58.57 | Get_The_Fish | Does anyone have the DB schema for *13 ODBC voicemail storage using PGSQL as a backend? |
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| 04:28.36 | retentiveboy | pardon the noob question but is there a way to use a list for extension matching? i.e. _(12|23|34)XX |
| 04:29.23 | retentiveboy | where that pattern would match _12XX or _23XX or _34XX? |
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| 07:50.41 | wdoekes | retentiveboy: no |
| 07:51.02 | wdoekes | but you could use dialplan functions and goto |
| 07:51.49 | wdoekes | _XXXX,n,GotoIf($["${EXTEN}"=~"^(12|23|34).*"]?match:nomatch) |
| 07:52.11 | wdoekes | or, you could goto from the patterns |
| 07:52.43 | wdoekes | _12XX,n,Goto(match,1); _23XX,n,Goto(match,1) |
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| 10:22.39 | Guest9903 | We're having some issues with DTMF, currently running dtmfmode=rfc2833. However we use the G711 codec so it is worth a try to go for inband instead? Pros and cons? Thank you |
| 10:31.09 | Guest9903 | Is setting it to auto a good idea? |
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| 12:59.56 | retentiveboy | Thanks wdoekes. |
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| 13:21.40 | t4nk820 | hi everyone, I'm looking for some help |
| 13:22.05 | t4nk820 | need to now how to add webrtc module on asterisknow |
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| 13:37.22 | babak | Hi, did anybody install libss7_2.0 on debian/ubuntu ? |
| 13:38.36 | babak | there is no error message after compile and make install but no ss7 command on asterisk... |
| 13:39.13 | WIMPy | Did you reconfigure and rebuild it? |
| 13:39.32 | WIMPy | And dahdi as well, I guess. |
| 13:39.44 | WIMPy | In reverse order. |
| 13:45.50 | babak | yes I done many times |
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| 13:46.32 | babak | from some messages from make install I suspect it was designrd just for centos |
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| 15:18.16 | Zogot | do local dialing not match to patterns? |
| 15:18.46 | WIMPy | What? |
| 15:18.59 | Zogot | i have a local dial to a context that has _X. |
| 15:19.25 | WIMPy | What is a local dial? |
| 15:19.40 | Zogot | Local/ |
| 15:19.48 | Zogot | Local/0@7584c8d6-38dd-412a-9780-43ffcc526a75&Local/1@7584c8d6-38dd-412a-9780-43ffcc526a75&Local/2@7584c8d6-38dd-412a-9780-43ffcc526a75,q for example |
| 15:19.49 | WIMPy | Ah |
| 15:20.06 | WIMPy | It's the same as any dialplan match. |
| 15:20.20 | Zogot | [Mar 11 15:08:28] NOTICE[5947][C-00000004]: core_local.c:701 local_call: No such extension/context 0@7584c8d6-38dd-412a-9780-43ffcc526a75 while calling Local channel |
| 15:20.21 | Zogot | for all 3 |
| 15:20.27 | WIMPy | You are aware that "-" is ignored? |
| 15:21.02 | Zogot | ignored? nah, ive got my entire project right now running off those :p |
| 15:21.06 | WIMPy | And "0" does not match "_X.". |
| 15:21.14 | Zogot | https://gist.github.com/zogot/69c15e2d07c2af55f5a3 |
| 15:21.20 | Zogot | but 1 should no? |
| 15:21.23 | Zogot | i get it for all 3 |
| 15:21.44 | WIMPy | You need at least another chatacter to match ".". |
| 15:21.53 | Zogot | ahhh |
| 15:21.54 | Zogot | great |
| 15:21.54 | Zogot | thanks |
| 15:21.56 | Zogot | that'll be it |
| 15:23.58 | WIMPy | The "-" part is for the extensions ony, not for contexts, so that's fine. |
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| 15:25.42 | Zogot | WIMPy: ah, good, i keep my extensions short anyway |
| 15:25.55 | Zogot | i think at most, its a phone username of companySlugName_phoneRandomString |
| 15:26.18 | Zogot | the fix worked, added a line exten => _X |
| 15:26.34 | Zogot | a minor change into changing some of the vars to exist for other channels with _ |
| 15:26.36 | Zogot | and it works :) |
| 15:41.20 | Zogot | WIMPy: all that, for this: http://i.imgur.com/LIimOUJ.png |
| 15:41.21 | Zogot | lol |
| 15:42.22 | WIMPy | You're doing a new GUI? |
| 15:43.05 | Zogot | yeh, in a very simple way of describing it. big aspect is the abstraction of commands from what the user fills and what asterisk needs |
| 15:43.09 | Zogot | but yeh :) |
| 15:43.13 | Zogot | its almost done actually |
| 15:43.18 | Zogot | been like 2 years now lol |
| 15:44.01 | Zogot | inbuilt chat, inbuild web phone with rtc, big change to typical way of handling calls, with specifications of what we call routers, incoming and outgoing, belonging to the company, and at a user level |
| 15:44.19 | Zogot | queues, all the standard phone things |
| 15:44.31 | Zogot | i'll be happy to give you a user if you want to take a look |
| 15:44.39 | Zogot | its a bit out of the date, the current in testing version |
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| 15:45.51 | WIMPy | I don't think I have a use, but thanks anyway. |
| 15:46.39 | Zogot | probably most people here wont lol |
| 15:46.53 | Zogot | but you may still find it interesting :p, your choice though :) |
| 15:47.02 | Zogot | anyway, friday end for me today, have a good weekend |
| 15:47.49 | WIMPy | is trying to become a Gentoo user. That'll keep me bysy for a while, I guess. |
| 15:47.55 | Zogot | haha ye indeed |
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| 17:28.19 | jeffspeff | is the minmemfree option in asterisk.conf supposed to look at just the free memory or is it also supposed to consider cached memory as free? |
| 17:30.18 | WIMPy | Only free, which obviousely makes it a less usefull. |
| 17:31.27 | jeffspeff | do you know of a reason why it wouldn't consider cached mem as free? |
| 17:32.10 | jeffspeff | my understanding of it is based on the information provided here -> http://www.linuxatemyram.com/ |
| 17:52.08 | WIMPy | No, I just know that others found it useless before, |
| 17:52.13 | WIMPy | . |
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| 18:09.55 | BlueInk | Hello, I am tinkering with an asterisk instance to get my t-home voip working. outbound is perfect but inbound refuses to work. Here: http://pastebin.com/TYryYsRp is a capture of the problem. Any ideas to get inbound calls to work? Huge thanks |
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| 18:15.03 | [TK]D-Fender | Reliably Transmitting (NAT) to 217.0.23.100:5060: |
| 18:15.09 | [TK]D-Fender | Contact: <sip:03095591394@192.168.3.254:5060> |
| 18:15.23 | [TK]D-Fender | You are calling out to the public internet and passing your PRIVATE IP for the conttact |
| 18:15.32 | [TK]D-Fender | you ahve not set your system up properly to work from behind NAT |
| 18:15.48 | [TK]D-Fender | You need to fix your peers and sstup your subnet values, etc |
| 18:17.43 | BlueInk | Thanks [TK]D-Fender I'll try to do that |
| 18:30.26 | BlueInk | Now I get this sort of a mess http://pastebin.com/APyU4n0z |
| 18:32.53 | [TK]D-Fender | And you don't have basic verbose enabled in there |
| 18:34.38 | BlueInk | just set debug peer |
| 18:36.18 | [TK]D-Fender | you had basic verbsoe before, and now don't. "core set verbose 10" |
| 18:37.13 | BlueInk | I tried to make to log shorter, well I failed |
| 18:39.03 | BlueInk | This is now with verbose 10 http://pastebin.com/3h1Pas6D |
| 18:40.57 | Kevin` | today's message of fun: [Mar 11 12:38:03] ERROR[12463]: chan_sip.c:17237 register_verify: 'WS' is not a valid transport for 'webif'. we only use 'WS'! ending call. |
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| 18:53.04 | BlueInk | So I set my external IP via externhost in [general] as test and got a spooky answer: We do not know this number |
| 18:55.04 | [TK]D-Fender | show us |
| 18:56.10 | BlueInk | the sip.conf? |
| 18:56.33 | [TK]D-Fender | yes, and the debug. |
| 18:56.36 | [TK]D-Fender | mask only password |
| 18:56.39 | [TK]D-Fender | s |
| 18:58.22 | Kobaz | extension Changed 200[blf_brown-c30028] new state Hold for Notify User c30028-201 (queued) |
| 18:58.40 | Kobaz | so i'm having problems with blf notifications.. anything that gets queued never gets sent so far as i can tell |
| 18:59.19 | Kobaz | https://issues.asterisk.org/jira/browse/ASTERISK-21677 |
| 19:01.26 | BlueInk | That's the sip.conf http://pastebin.com/xQEizmWP |
| 19:02.27 | [TK]D-Fender | BlueInk: "directmedia=no" under [general] and your peers |
| 19:02.39 | [TK]D-Fender | BlueInk: make sure your router has no SIP ALG,/proxy, etc |
| 19:02.53 | [TK]D-Fender | BlueInk: And that you've forwarded the ports for RTP & SIP |
| 19:05.25 | BlueInk | I think I can't completely disable SIP in my FritzBox |
| 19:05.56 | [TK]D-Fender | Then you're likely going to have real problems. |
| 19:06.15 | [TK]D-Fender | Is this using the same ACCOUNT as you have on it? |
| 19:07.21 | BlueInk | I disabled 2 numbers of 3 and tried to route them with asterisk |
| 19:07.36 | BlueInk | and they all use the same account |
| 19:09.47 | BlueInk | I disabled sip & dect on the FritzBox now |
| 19:15.26 | BlueInk | Now I am down to this http://pastebin.com/NcXgG1HV |
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| 19:15.51 | [TK]D-Fender | that doesn't really amtter |
| 19:15.57 | [TK]D-Fender | that's just the keep-alive you set up |
| 19:16.02 | [TK]D-Fender | "qualify=yes" |
| 19:16.08 | [TK]D-Fender | set it to "no" and those will go away |
| 19:18.44 | BlueInk | lol no log in the cli but a lovely voice saying: The number you have called is unknown |
| 19:19.15 | [TK]D-Fender | If you have verbose enabled, and SIP debug as we see.. then that isn't youyr system saying that |
| 19:21.05 | BlueInk | verbose 10, debug 10, debug peer |
| 19:21.43 | BlueInk | And the voice is the lovely phone company voice |
| 19:22.12 | [TK]D-Fender | Then it's not your server |
| 19:22.16 | [TK]D-Fender | "sip show registry" <- |
| 19:23.32 | BlueInk | http://pastebin.com/rvq9zTcm |
| 19:24.15 | [TK]D-Fender | "module reload chan_sip.so" |
| 19:24.22 | [TK]D-Fender | and watch for an attempt to register |
| 19:26.30 | BlueInk | looks good to me http://pastebin.com/xzt0RV9Z |
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| 19:34.17 | BlueInk | But still no luck, only phone company voice |
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| 19:46.26 | BlueInk | How can sip be registered but unknown? |
| 19:47.20 | [TK]D-Fender | because you registered. |
| 19:47.25 | [TK]D-Fender | that has NOTHING to do with your peer |
| 19:47.30 | [TK]D-Fender | the peer is not being verified |
| 19:47.36 | [TK]D-Fender | So therefor it is "unknown" |
| 19:47.44 | [TK]D-Fender | Doesn't mean you can use it to dial out |
| 19:47.51 | [TK]D-Fender | Doesn't eman it won't match if they send you a call |
| 19:47.54 | [TK]D-Fender | So that is just fine |
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| 19:48.43 | BlueInk | So I can dial out. that means my auth data is fine, right? |
| 19:48.59 | [TK]D-Fender | pretty safe bet |
| 19:49.14 | [TK]D-Fender | odds are they'd have checked the auth with the call before allowing it |
| 19:50.21 | BlueInk | Pardon me but I am clueless right now what do try next to get everything going |
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| 19:57.53 | [TK]D-Fender | So far everything you server is trying and seeing is working.. |
| 19:58.01 | [TK]D-Fender | You'll have to check your router for the inbound call issue |
| 20:02.53 | BlueInk | i tried a different password and got unauthorized so the previous password must be fine |
| 20:04.02 | BlueInk | My port forward are UDP 5060 UDP 30000-31000 and udp 40000 -41000 |
| 20:04.52 | retentiveboy | examples for starting asterisk via systemd on Redhat/CentOS seem to use the -f (don't fork) option and we end up with the STDOUT being logged to /var/log/messages. It's nasty as well because the timestamps and colorizing escape codes end up in there as well. Why use -f? |
| 20:05.48 | retentiveboy | It I wanted all that info there, I'd add a syslog entry in logger.conf. Curious is remaining in the foreground is required for systemd. |
| 20:06.23 | [TK]D-Fender | BlueInk: Still hard to say if your router is screwing you... |
| 20:06.32 | [TK]D-Fender | You should ask your provider what they see |
| 20:13.34 | BlueInk | My home setup may be wonky the router has a 192.168.3.0/24 net behind that I have a pfsense box which hosts a 192.168.1.0/24 net with wan set to 192.168.3.249. in the pfsense box is a vpn which is on a 192.168.10.0/24 net and the asterisk which is on 192.168.1.0/24 |
| 20:15.48 | *** join/#asterisk ks3 (~ks3@cpe-24-209-22-231.cinci.res.rr.com) |
| 20:16.44 | [TK]D-Fender | that's DUAL NAT which is BAD |
| 20:16.57 | retentiveboy | NM, removed -f and set Type=forking and all good now. |
| 20:30.33 | BlueInk | [TK]D-Fender: I know I know. :D still looking for a pure vdsl modem |
| 20:30.38 | BlueInk | brb |
| 20:30.55 | [TK]D-Fender | Have you tried doing PPPoE through it? |
| 20:31.11 | [TK]D-Fender | if that's what your outer modem is doing I have one here that I can chain through |
| 20:31.28 | [TK]D-Fender | So while they offer NAT I am doing my own PPPoE login through their combo modem/router |
| 20:31.37 | [TK]D-Fender | And thus get the raw IP interface on mine |
| 20:43.20 | *** join/#asterisk BlueInk (d598a165@gateway/web/freenode/ip.213.152.161.101) |
| 20:43.38 | BlueInk | I am back |
| 20:44.20 | BlueInk | what did I miss? I am on a new device without the chat history |
| 20:46.01 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 20:57.30 | BlueInk | [TK]D-Fender: Thanks for your kind help. I might postpone the asterisk until I can get rid one NAT |
| 20:59.00 | [TK]D-Fender | [15:30][TK]D-FenderHave you tried doing PPPoE through it? |
| 20:59.01 | [TK]D-Fender | [15:31][TK]D-Fenderif that's what your outer modem is doing I have one here that I can chain through |
| 20:59.03 | [TK]D-Fender | [15:31][TK]D-FenderSo while they offer NAT I am doing my own PPPoE login through their combo modem/router |
| 20:59.04 | [TK]D-Fender | [15:31][TK]D-FenderAnd thus get the raw IP interface on mine |
| 21:01.03 | BlueInk | PPPoE through... hmm then I have to clear the login parameter in the first box, right? |
| 21:02.21 | [TK]D-Fender | not sure |
| 21:02.34 | [TK]D-Fender | it's an idea. I'll let you run with it |
| 21:04.22 | BlueInk | They removed pppoe passthrough |
| 21:04.51 | BlueInk | Why do they __always__ remove the good stuff?? |
| 21:11.05 | BlueInk | Bye everybody, i'll be back once I know more |
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| 22:17.15 | Kevin` | how can I allow stun/ice negotition to work between two clients on the same lan behind the same nat? |
| 22:17.30 | *** join/#asterisk sekil (~sekil@adsl-82-117-222-227.dynamic.sbb.rs) |
| 22:17.36 | Kevin` | also, seperately, without internet access? |
| 22:53.46 | jeffspeff | does anyone know of a reason why the minmemfree setting wouldn't consider cached mem as free? |
| 22:53.52 | jeffspeff | my understanding of it is based on the information provided here -> http://www.linuxatemyram.com/ |
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