| 00:40.48 | carrar | STRAWBERRY JAM YES PLS |
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| 05:26.22 | themrrobert | rats :( i got another crash. This time it said it died on signal 6, which seems to indicate stack corruption. I had enabled DONT_OPTIMIZE and BETTER_BACKTRACES, but unfortunately, the backtrace has virtually 0 information. It always says "in ??" and/or "no symbol table found" even though it found the symbols in the exe and doesn't ask for more debuginfo installs |
| 05:27.32 | themrrobert | it doesn't have nearly the info my previous core dumps did, no threads, no function names, no real usable info at all. Any idea how to fix that? |
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| 05:28.40 | themrrobert | i mean it does have threads, but all the symbols are ?? |
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| 05:30.13 | themrrobert | here is the backtrace for anyone interested: http://pastebin.com/9BYABRdx |
| 05:33.37 | themrrobert | should I disable BUILD_NATIVE and/or enable REBUILD_PARSERS or any other flags? |
| 05:35.18 | themrrobert | lol i love how the wiki says it will be [almost] unusable without D_O and B_T, but at least before I could see the actual backtrace not a bunch of ??'s |
| 05:49.16 | madduck | does anyone know of a resource discussing best practices relating to hosting multiple sites independently using a single asterisk? (virtual hosting, essentially?) |
| 05:49.48 | madduck | i.e. #include sip.conf and extensions.conf parts for each site, but they are still sharing a common namespace. |
| 05:50.29 | drmessano | madduck, it's pretty much up to you |
| 05:50.53 | madduck | yeah, which is why I am trying to learn from other people's mistakes⦠|
| 05:53.04 | drmessano | Honestly with virtualization being what it is now, you're better off building a container for each customer |
| 05:54.11 | madduck | yeah, except then I have to manage redundancy in the commonalities, e.g. the conferencing setup, or the stdexten function |
| 05:54.20 | madduck | i suppose i could #include those⦠|
| 05:57.25 | drmessano | Same company? |
| 06:15.37 | carrar | madduk you could use contexts to split them out, and a include of something like #include accounts/*/extensions.conf |
| 06:15.58 | carrar | same with sip.conf |
| 06:16.47 | carrar | no reason you can't host hundreds of customers |
| 06:17.45 | drmessano | What about managing the redundancy in the commonalities? |
| 06:18.20 | carrar | run it in vmware |
| 06:18.33 | carrar | fail over to another node |
| 06:19.10 | carrar | or another datacenter |
| 06:19.39 | drmessano | or another planet |
| 06:20.07 | carrar | cooling should be cheaper out there |
| 06:22.16 | drmessano | Well, im sure Pofung will have a $75 satellite data center you can buy in 2016 |
| 06:23.20 | phix | so what you reckon? |
| 06:23.30 | phix | any ideas how to get dahdi working? |
| 06:23.45 | drmessano | Install it? |
| 06:24.42 | phix | done |
| 06:24.48 | phix | but it isn't picking up the channels |
| 06:25.06 | phix | I has the pastebin link of my conf if you missed it |
| 06:25.14 | phix | I did paste it yesterday |
| 06:44.44 | madduck | carrar: yeah, something like this, but then one still needs to prefix the contexts and SIP stanzas with companyname- or somewuch |
| 06:45.23 | madduck | drmessano: yeah, maybe common files in a Git repo cloned to all asterisk instances and included from the individual files might be better |
| 07:16.21 | drmessano | Sounds like a plan |
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| 13:17.08 | WIMPy | In case someone here is in to it: asteriskdocs.org is unreachable. |
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| 13:57.48 | boch | hi all |
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| 14:00.35 | boch | is it possible to ignore flash hook? some channels are playing music on hold when hangup because asterisk gets the hangup as a flash hook and holds the caller |
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| 14:08.07 | nixz | what ports are being blocked if the one I am calling can't hear me? But I can hear him |
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| 14:31.57 | WIMPy | boch: What are your phones connected to? |
| 14:32.35 | WIMPy | nixz: The rtp ports. - Or you have a NAT issue. |
| 14:32.59 | nixz | seems like rtp ports are blocked at the other end |
| 14:32.59 | boch | WIMPy, i really dont know because this happens calling trough a carrier |
| 14:33.09 | nixz | I have sip server. |
| 14:33.27 | nixz | how to determine if what port is being blocked? |
| 14:33.46 | nixz | I tried 10000 and 20000 and they are both opened. |
| 14:34.10 | nixz | wow.. I cannot test between 10000-20000... that's a huge number of ports to test.. |
| 14:34.17 | WIMPy | boch: Without knowing what we're dealing with, you can only try the glas sheres. |
| 14:34.20 | nixz | any advice? |
| 14:34.36 | nixz | using other isp.. it works.. |
| 14:34.55 | nixz | so I want to pin down the exact port that isp is blocking.. so I can make an alternative. |
| 14:35.36 | nixz | it can register. It rings when i call.. It can make outgoing calls.. but the audio is one way. |
| 14:35.43 | nixz | sip server is at my side. |
| 14:35.48 | WIMPy | nixz: Thats just the normal SIP issues that almost everyone had to go through. Any NAT involved? |
| 14:36.13 | nixz | I do not know. |
| 14:36.17 | nixz | at his end. |
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| 14:47.54 | nixz | or perhaps the better question is.. how to check which port is blocked by isp? |
| 14:48.04 | nixz | the one that is being used for rtp |
| 14:49.38 | WIMPy | Read the SIP debug. |
| 14:49.54 | nixz | can I find an answer there? |
| 14:49.56 | WIMPy | That will also tell you if it even tried to do somehting sensible. |
| 14:50.22 | nixz | how to do it? |
| 14:50.33 | WIMPy | In the SDP |
| 14:50.43 | WIMPy | 'sip set debug on' |
| 14:50.55 | WIMPy | And then have fun reading it. |
| 14:51.24 | nixz | SIP Debugging enabled... where to check the logs. |
| 14:51.41 | nixz | asterisk logfiles? |
| 14:52.09 | WIMPy | Depends on your logger.conf. |
| 14:52.16 | WIMPy | You will see it on the *CLI usually. |
| 14:56.59 | nixz | how to see live logs? |
| 14:57.08 | nixz | I'm on cli now |
| 14:58.05 | WIMPy | Make a call. |
| 14:58.34 | nixz | how to get to sip cli? |
| 14:58.36 | nixz | using putty |
| 14:59.39 | nixz | got it.. asterisk -r... is there a way to only display logs from a particular IP? |
| 15:00.19 | WIMPy | Use your tab key. |
| 15:00.29 | WIMPy | 'sip set debug ...' |
| 15:01.44 | nixz | peer? |
| 15:02.11 | nixz | is peer for extension number? |
| 15:02.12 | WIMPy | The IP of the named peer. |
| 15:02.29 | WIMPy | No. Peer name. |
| 15:02.36 | WIMPy | Extensions don't have IPs. |
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| 15:11.22 | jzu_ | Yo |
| 15:11.36 | jzu_ | any of you had to adjust chan_dongle volume down through the dongle.conf? |
| 15:20.52 | nixz | WIMPy, Reliably Transmitting (no NAT) to IP address |
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| 15:27.24 | nixz | here's the log. http://pastebin.com/L3c31WGg |
| 15:28.15 | nixz | can you help me guys? the issue is.. I can't hear the voice of the person at the other end. He is the remote. |
| 15:33.34 | nixz | is there something in the logs that would guide me on how to solve my problem? |
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| 15:47.30 | nixz | http://pastebin.com/ZtMTW0zP (here's the new link for the logs). |
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| 16:18.14 | nixz | is there a workaround on this? |
| 16:19.12 | [TK]D-Fender | There is no "workaraound". There is only "doing it properly". |
| 16:19.49 | [TK]D-Fender | We don't see a full call from both sides and configs to match |
| 16:20.12 | [TK]D-Fender | this means unfiltered. Screw with the evidence and we will miss where mistakes are made |
| 16:30.06 | jsearle | quit |
| 16:35.22 | nixz | are you telling me to make call.. then other party to call too? |
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| 16:41.27 | nixz | [TK]D-Fender, thanks for the insights. |
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| 16:43.29 | WIMPy | nixz: We can't help if you remove the relevant information from the debug. |
| 16:43.41 | nixz | okay. |
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| 21:31.32 | TheZealous | hi all |
| 21:31.48 | TheZealous | in the sip.conf, I set the: allowguest=yes |
| 21:32.07 | TheZealous | but when I point a DID number that I purchased from voip.ms, it is not working |
| 21:32.17 | TheZealous | I got this error: |
| 21:32.33 | TheZealous | Connected to Asterisk 11.17.0 currently running on a2billingtest (pid = 911) |
| 21:32.33 | TheZealous | [May 16 11:02:23] NOTICE[978][C-00000001]: chan_sip.c:25791 handle_request_invit e: Call from '' (174.34.146.162:5060) to extension '16418475141' rejected becaus e extension not found in context 'public'. |
| 21:32.33 | TheZealous | a2billingtest*CLI> |
| 21:33.07 | TheZealous | Anyone have any idea on how to change settings in asterisk so that it can accept all calls from any provider please? |
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| 21:49.02 | ChannelZ | well, you have no extension that can match the extension 16418475141 in the context called 'public', as it says. |
| 21:50.25 | ChannelZ | But you should probably really have a sip peer configured for voip.ms that goes into an incoming call handling context rather than depending on anonymous (guest) calls |
| 21:50.56 | ChannelZ | but either way, you just don't have a proper extension defined to deal with it at the moment |
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