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| 00:24.26 | saint_ | i did all the backups i can. i m in the process of the release 13 make - wish me luck , coming from a 1.8 |
| 00:24.37 | saint_ | s/i can/i cloud/ |
| 00:25.52 | saint_ | ouch - make[1]: *** [test_ari_model.o] Error 1 |
| 00:28.58 | saint_ | Anyone can give me a hint with this compile error ? http://pastebin.com/yHDEU2W5 |
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| 00:49.27 | ruben23 | hi guys i have 2 asterisk server --> asterisk one usersdial with conference then on that conference session he is only the one and from there dials out but this call are pass on to the second asterisk where phones extensions are registered then calls are recorded here also the dialout pattern is executed here until it dialsout....the issue is on this second asterisk it only records all calls in one audio file with multiple call session..this means i lik |
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| 01:16.05 | saint_ | any idea boys ? |
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| 01:24.11 | Phlunk3 | I am trying to telnet to my asterisk server on port 5038 and running a login action, it keeps booting me because of password failure... the password is the secret from the manager right? |
| 01:24.22 | Phlunk3 | manager.conf* // manager_custom.conf |
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| 01:32.39 | Milos | What is the solution to this? http://www.voip-info.org/wiki/index.php?page_id=2883&comments_page=1 |
| 01:32.48 | Milos | "I have the same problem. When an incoming call arrives from provider, always matching to the first 'peer', what are in sip show peers output. How can I control this procedure? It's a big problem, because the phones using this accounts simultaneously, I need to know, which one is free for dialout." |
| 01:35.32 | Milos | http://www.voip-forum.com/asterisk/2006-01/multiple-sip-registrations/ this is 8 years old :/ |
| 01:41.08 | saint_ | Milos what s your issue so i don t have to read the whole page |
| 01:41.20 | Milos | saint_, I have multiple numbers with the same provider |
| 01:41.28 | Milos | and |
| 01:41.37 | Milos | so I Have |
| 01:42.00 | Milos | [blah_line1] context=context-one [blah_line2] context=another-context |
| 01:42.12 | Milos | everything from blah provider always matches context-one |
| 01:42.32 | Milos | if I paste sip.conf will that help? |
| 01:42.53 | WIMPy | Either you didn't register an extension or you used the wrong type=. |
| 01:43.18 | saint_ | if it s the same provider, why do you have multiple declaration ? |
| 01:43.36 | saint_ | is it one trunk ? |
| 01:43.49 | WIMPy | Good question. |
| 01:44.03 | Milos | here: https://bpaste.net/show/8b8306b62875 |
| 01:44.04 | Milos | is this wrong? |
| 01:44.06 | WIMPy | Extensions are the way to go. |
| 01:44.37 | saint_ | you would have only one context, but multiple exten=> in it , each one with a different number. there you tell the incoming number what to do |
| 01:44.49 | Milos | is that the best way then? |
| 01:44.58 | saint_ | holy moly. i just found out about transfercarus.com |
| 01:45.08 | saint_ | Milos having one sip connection to the provider |
| 01:45.13 | Milos | ah |
| 01:45.17 | saint_ | they can add multiple channels in it . |
| 01:45.18 | Milos | one connection but multiple numbers |
| 01:45.19 | Milos | I See |
| 01:45.22 | Milos | s/S/s/ |
| 01:45.24 | Milos | let me try that |
| 01:45.25 | saint_ | you probably come from a ISDN world |
| 01:45.30 | Milos | I come from nowhere |
| 01:45.33 | Milos | I just want my voip working |
| 01:45.35 | Milos | :P |
| 01:45.37 | saint_ | T1 = 23 chan , so if you want more, you would have multiple T1s |
| 01:45.41 | Milos | ah |
| 01:45.52 | saint_ | With sip, you can ask your provider to put everything in the same pipe |
| 01:45.57 | Milos | yup |
| 01:46.00 | Milos | that makes a lot of sense |
| 01:46.04 | Milos | would it be type=peer here? |
| 01:46.06 | saint_ | then in extension.conf , you manage your different incoming numvers |
| 01:46.07 | Milos | instead of friend |
| 01:46.12 | Milos | because that's what I had when I had one entry |
| 01:46.18 | Milos | btu when I added another type=peer of the same I got 403 |
| 01:46.22 | Milos | so I had to change to type=friend |
| 01:46.24 | saint_ | nah, you can keep your type=friend i think |
| 01:46.30 | Milos | wtf is type=friend |
| 01:46.33 | Milos | such silly names |
| 01:47.09 | Milos | if I use password in register line what is the point of password in [blah] entry? |
| 01:47.14 | Milos | I don't understand all the double-ups |
| 01:47.22 | Milos | because that's why I thought I needed multiple "connections" to sip |
| 01:47.28 | Milos | because I use a different password.. |
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| 01:48.13 | WIMPy | Well, you don't need both. |
| 01:48.24 | Milos | good |
| 01:48.29 | Milos | I will remove password form [blah] entry |
| 01:48.31 | saint_ | i'd have to look at the doc,Peers handle both inbound and outbound calls and are matched by ip/port, |
| 01:48.39 | Milos | and see what happens |
| 01:48.46 | WIMPy | You can use callbackexten in the peers and skip the register lines. |
| 01:48.55 | saint_ | friends requires that the authorization matches rather than the ip |
| 01:49.06 | WIMPy | You need them to call out. |
| 01:49.18 | Milos | which method should I use.... |
| 01:49.24 | saint_ | check out the sip.conf.sample, it explains it all |
| 01:50.14 | saint_ | Milos you might want to spend some time reading http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html too |
| 01:50.15 | WIMPy | The trouble is that ITSPs usually don't authenticate with you when sending you a call, so you can only match them by source IP. |
| 01:51.56 | saint_ | i use flowroute , and i have it setup as friend |
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| 02:28.19 | Milos | WIMPy, if I remove from [provider] entry, I get handle_response_invite: Failed to authenticate on INVITE |
| 02:28.36 | Milos | remove the username/password that is |
| 02:35.13 | WIMPy | What type? |
| 02:36.01 | Milos | type=peer |
| 02:36.27 | WIMPy | So they come from multiple IPs? |
| 02:36.31 | Milos | what? |
| 02:36.44 | WIMPy | The calls. |
| 02:36.47 | Milos | no? |
| 02:36.54 | Milos | it's one IP |
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| 02:40.34 | WIMPy | Err, that was on an outbound call? |
| 02:42.43 | Milos | yup |
| 02:43.02 | Milos | WIMPy, it was. |
| 02:43.03 | WIMPy | Then I hope you didn't expect anything else. |
| 02:43.14 | Milos | so I need do put user and pass twice? |
| 02:43.16 | Milos | s/do/to/ |
| 02:43.27 | WIMPy | No |
| 02:43.38 | Milos | so putting it in register once isn't enough for outbound |
| 02:43.50 | Milos | where do I put it so that it works both ways and I don't have to put user and pass everywhere |
| 02:43.56 | WIMPy | No, register is for registering only. Nothing else. |
| 02:44.02 | Milos | okay |
| 02:44.25 | WIMPy | Replace the register line byt callbackexten in the peer. |
| 02:45.16 | Milos | and if I have two extensions from a single peer? |
| 02:46.13 | WIMPy | Then you need a 2nd peer. |
| 02:46.22 | Milos | ........... |
| 02:47.25 | WIMPy | You need a place to put username/password anyway. |
| 02:47.57 | Milos | yes. in the register line. |
| 02:48.07 | Milos | with my one peer. |
| 02:48.15 | WIMPy | And also for calling out. |
| 02:48.20 | Milos | argh |
| 02:48.31 | Milos | such a mission |
| 02:49.01 | Milos | saint_ was just telling me it's silly to have multiple peers if they're the same provider and now you're telling me I need to |
| 02:49.01 | WIMPy | Not really if they only use one IP. |
| 02:49.37 | WIMPy | You don't need multiple to receive calls, but you want them to dial out on them. |
| 02:49.56 | WIMPy | And you can put the registration in there as well. |
| 02:50.51 | Milos | okay I will try |
| 02:50.57 | Milos | so I can get rid of register => |
| 02:51.08 | WIMPy | Incomming/outgoing/registering are 3 different things. |
| 02:51.13 | WIMPy | Yes |
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| 02:56.15 | Milos | yay WIMPy :D |
| 02:56.42 | Milos | how can I accept num1 or num2 in the same context? |
| 02:56.43 | Milos | I have: |
| 02:56.49 | Milos | exten => num1,1,Ringing |
| 02:56.51 | Milos | if I put |
| 02:56.55 | Milos | exten => num2,1,Ringing |
| 02:56.56 | Milos | under it |
| 02:56.57 | Milos | only num2 works |
| 02:57.04 | Milos | num1 says something unknown blah blah when I call in |
| 02:57.22 | WIMPy | What exactely? |
| 02:57.25 | Milos | ok |
| 02:57.33 | Milos | Auto fallthrough, channel 'SIP/2talk_line1-00000011' status is 'UNKNOWN |
| 02:57.52 | Milos | right after executing(ringing... ) |
| 02:58.08 | WIMPy | You don't have anything more than that Ringing? |
| 02:58.16 | Milos | I have heaps |
| 02:58.19 | Milos | but after I use same => n |
| 02:58.37 | WIMPy | For both extensions? |
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| 02:58.51 | Milos | I want not to rewrite it all just for another extension |
| 02:58.54 | Milos | I want the first line to be |
| 02:59.00 | Milos | accept num1 OR num2 then do { code } |
| 02:59.11 | Milos | rather than accept num1 do { code }; accept num2 do { code } |
| 02:59.30 | WIMPy | Then why did you want to seperate them in the forst place? |
| 02:59.33 | nix8n82 | Milos use gosub or macro |
| 02:59.46 | WIMPy | You can use patterns or you can use Goto(). |
| 02:59.48 | Milos | but I have to define the incoming number WIMPy |
| 02:59.56 | WIMPy | macro is deprecated. |
| 03:00.00 | Milos | if I use exten => _X.,1,Ringing |
| 03:00.03 | Milos | it will fix the issue |
| 03:00.09 | Milos | but I want to restrict it to just those to numbers (for argument's sake) |
| 03:00.10 | WIMPy | No, you don't have to. YOu can just accept all. |
| 03:00.28 | Milos | how does one "accept all" ? isn't that _X. |
| 03:00.45 | Milos | s/to numbers/two numbers/ |
| 03:00.56 | WIMPy | You can use patterns or you can use Goto() or you can use single extensions at prio 1 and then continue with a pattern from prio 2 onwards. |
| 03:01.09 | Milos | hmm ok I will try prio2 |
| 03:01.18 | nix8n82 | <PROTECTED> |
| 03:01.34 | WIMPy | _X. is anything that starts with a digit and is followed by at leas a 2nd character. |
| 03:01.56 | WIMPy | It's not dangerous, but it probably won't do what you want. |
| 03:01.59 | Milos | prio2 has the same outcome as before |
| 03:02.12 | Milos | I don't think you understood my question - I'll be more clear |
| 03:02.42 | Milos | https://bpaste.net/show/b13d2b16018b |
| 03:02.50 | Milos | for EITHER num1 OR num2 I want to continue down that chain |
| 03:02.52 | WIMPy | Did you understand the "and continue with a pattern" bit? |
| 03:03.07 | Milos | I want to match directly the number and not have to use a pattern? |
| 03:03.13 | Milos | or are you saying I need to regex out a num1|num2 pattern |
| 03:03.22 | WIMPy | You can have both. |
| 03:03.27 | WIMPy | That's not possible. |
| 03:03.46 | WIMPy | I gave you 3 different ways to do it. |
| 03:03.52 | Milos | okay, let me try figure them all out |
| 03:04.24 | Milos | "or you can use single extensions at prio 1 and then continue with a pattern from prio 2 onwards." |
| 03:04.34 | Milos | this means I have to duplicate the logic, is that right? |
| 03:04.42 | WIMPy | no |
| 03:04.51 | WIMPy | Only fro prio 1. |
| 03:05.11 | Milos | well, I tried using exten => 09num1,1,Ringing ... exten => 09num2,2,Ringing ... same => n, ... |
| 03:05.15 | Milos | and I got unknown for num1 again |
| 03:05.30 | WIMPy | Yes. |
| 03:05.58 | Milos | does that mean I'm not doing it correctly or did I not understand what you meant? |
| 03:06.10 | WIMPy | You need to continue with _X.,n,... and than you can use same |
| 03:06.29 | WIMPy | same means same, not whatever. |
| 03:06.42 | Milos | ahhhhhhh |
| 03:06.44 | Milos | I think I get it |
| 03:06.51 | Milos | same implies the latest extension |
| 03:06.55 | Milos | so same was always num2 |
| 03:06.56 | Milos | is that right? |
| 03:07.01 | WIMPy | exactely. |
| 03:07.03 | Milos | riiiiiiiight |
| 03:08.30 | WIMPy | You can do pretty nifty things that way, but you can also make your dilplan pretty unreadable the same way. |
| 03:08.38 | Milos | hmm yes |
| 03:08.44 | Milos | it works great now |
| 03:08.56 | Milos | do you think that is a good solution or bad solution? |
| 03:09.05 | Milos | I'm happy to change if you think it's a poor design |
| 03:09.27 | WIMPy | That's about personal preferences. |
| 03:09.46 | nix8n82 | https://bpaste.net/show/29acfc29706b. Milos |
| 03:09.47 | Milos | WIMPy, https://bpaste.net/show/c7885dd1bb0b |
| 03:09.54 | Milos | would you approve? |
| 03:10.01 | WIMPy | I can't open that here. |
| 03:10.04 | Milos | argh |
| 03:10.24 | Milos | WIMPy, http://dpaste.com/2WSEVW6 |
| 03:10.39 | Milos | that works - would you approve? |
| 03:10.56 | WIMPy | That looks fine. |
| 03:11.11 | Milos | so if a 3rd number somehow snuck in, it wouldn't come through, is that right? |
| 03:11.54 | WIMPy | You just also need to be aware that n always is the prio of the previous text line +1, if you plan to do more sophisticated stuff. |
| 03:12.01 | WIMPy | right. |
| 03:12.12 | Milos | nix8n82, sorry I completely didn't see you mentioned me before. having a look now |
| 03:12.32 | Milos | thanks for that wimpy. |
| 03:12.49 | Milos | nix8n82, ah yup. that works too. a bit clearer I suppose. |
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| 04:45.16 | Milos | Why doesn't asterisk retry its connection to peer? |
| 04:45.35 | Milos | "Registration for '....' timed out, trying again (Attempt #2) then nothing |
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| 04:45.39 | Milos | and it sits at unknown! |
| 04:45.44 | Milos | that is absolutely shocking |
| 04:49.09 | Milos | registerattempts=0 |
| 04:49.09 | Milos | registertimeout=20 |
| 04:49.10 | Milos | I have this |
| 04:49.17 | Milos | and yet it doesn't even retry twice |
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| 05:11.24 | Milos | https://www.mail-archive.com/asterisk-users@lists.digium.com/msg156849.html again this was 8 years ago |
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| 05:14.44 | Milos | yeah, definitely seems related to DNS timeouts only |
| 05:14.55 | Milos | so when DNS can't resolve asterisk refuses to retry a registration |
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| 05:29.00 | ChannelZ | Are you using DNS Manager? It should do periodic lookups if you ask it to |
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| 05:42.16 | Milos | ChannelZ, it is set to N in `sip show registry`. |
| 05:42.55 | Milos | What is "DNS Manager" in this case? I already have a caching name server, on the same box in fact, localhost. Why can't it just look up more often? Or is that what "DNS Manager" does? |
| 05:43.32 | ChannelZ | /etc/dnsmgr.conf |
| 05:44.21 | ChannelZ | Asterisk only resolves hostnames for things like sip peers when they get loaded. dnsmgr lets it periodically poll to refresh hostnames |
| 05:44.22 | Milos | -ENOENT |
| 05:44.48 | ChannelZ | eh? |
| 05:44.59 | Milos | -ENOENT = No such file or directory |
| 05:45.07 | Milos | I don't have that file on my box |
| 05:45.14 | Milos | and why is it in /etc/ and not /etc/asterisk/ ? |
| 05:45.24 | Milos | ah, it is in /etc/asterisk/ |
| 05:45.26 | Milos | :D |
| 05:46.17 | Milos | ChannelZ, I hope you understand my concern here. I already have a variable registertimeout, why do now need to set refreshinterval |
| 05:46.41 | ChannelZ | Because they're different. |
| 05:46.46 | Milos | It's only natural for asterisk to retry the DNS name after registertimeout seconds... because otherwise it's not retrying. |
| 05:47.20 | ChannelZ | As I said, asterisk only resolves hostnames once when it loads your sip config. |
| 05:47.23 | Milos | asterisk stops retrying after [Oct 30 17:21:24] ERROR[2659] netsock2.c: getaddrinfo("sip.2talk.co.nz", "(null)", ...): Temporary failure in name resolution |
| 05:47.41 | Milos | well... no point in arguing, I just disagree very strongly with that decision |
| 05:47.58 | Milos | I think, if there is a DNS error, it should retry (including the DNS) after registertimeout seconds |
| 05:47.59 | ChannelZ | I think there's a few cases it looks things up on demand |
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| 05:48.27 | Milos | otherwise the docs should say, "asterisk will retry forever... unless it couldn't find DNS, in that case it will give up and not tell you." |
| 05:48.40 | Milos | very bad design, I might have to submit a bug |
| 05:48.41 | ChannelZ | I don't specifically know for registers, but considering your problem... |
| 05:49.09 | Milos | (in my case, reason was because I rebooted the asterisk box which is also the router... and asterisk loaded before ppp) |
| 05:49.17 | Milos | and it just gave up. |
| 05:49.26 | Milos | despite docs saying it will retry forever |
| 05:49.44 | Milos | with registerattempts=0 |
| 05:49.52 | Milos | you see my concern now? |
| 05:51.20 | Milos | even with dnsmgr... what, you're saying I have a guaranteed downtime of 5 minutes for my 30 seconds reboot because dnsmgr won't retry for refreshinterval seconds? |
| 05:51.21 | ChannelZ | Your failure is prior to the registration. It's different. I'm not arguing it's right; it's the way it works. |
| 05:51.40 | Milos | I understand, it just seems counter-intuitive |
| 05:52.46 | Milos | I don't want to refresh my DNS every 5 seconds, but 5 seconds on top of the 30 seconds reboot is reasonable downtime. however, now I have all this stupid dns traffic for no reason |
| 05:52.57 | Milos | so yeah, I'll hit up a bug sometime soon |
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| 05:53.03 | Milos | but for now, I put the IP in /etc/hosts |
| 05:53.13 | Milos | because as mentioned, even with dnsmgr, it will be instant downtime on startup |
| 05:53.49 | Milos | (yes, I could delay asterisk startup until ppp is up, which is only a few extra seconds, but I shouldn't *have* to) |
| 05:54.04 | Milos | as asterisk is not strictly external communication |
| 06:00.08 | ChannelZ | I'm guessing pjsip probably does this better by default </canofworms> |
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| 06:14.26 | aruntomar | i'm facing a weird problem. sip users are able to make outbound calls, but can't receive incoming calls. sip show users reveal their host as unspecified. i've read that unspecified occurs when the user is not subscribed/registered. but then, i don't even get a wrong password error. |
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| 06:21.05 | ChannelZ | Is asterisk behind a firewall? |
| 06:21.30 | aruntomar | no, it's on lan and users are also on the same lan |
| 06:21.35 | ChannelZ | Or rather, are they? (You said they can't receive calls, which would be asterisk -> device) |
| 06:22.56 | ChannelZ | Do you have hostnames or IPs set for them in sip.conf or are they dynamic? Because if they are dynamic they have to be registering to be making calls.. unless you've got anonymous turned on and all the outgoing calls are being generically picked up by that |
| 06:23.54 | ChannelZ | or wait.. you said 'sip show users'.. they are configged only as users? |
| 06:24.20 | ChannelZ | (hopefully you meant 'sip show peers') |
| 06:24.30 | aruntomar | yes, i meant sip show peers |
| 06:24.31 | ChannelZ | users don't have hosts |
| 06:24.38 | ChannelZ | ok |
| 06:24.47 | aruntomar | allowguest=no |
| 06:24.55 | aruntomar | and qualify=yes |
| 06:25.01 | aruntomar | host=dynamic |
| 06:25.04 | ChannelZ | so if they don't have IPs set, then they are dynamic. But haven't registered. |
| 06:25.29 | aruntomar | sometimes they show the ip and sometimes they don't |
| 06:25.40 | aruntomar | that's the weird part |
| 06:25.43 | ChannelZ | So either they aren't even trying to register, or they are and are being rejected (but you say you don't see anything in the console), or it's network problems. |
| 06:26.37 | ChannelZ | I'm guessing it's not #2 since they can make calls, and with no guests, would have to be authenticating as themselves |
| 06:31.26 | ChannelZ | Well lacking any more info, and given that you said this is all LAN, the devices aren't configured quite right and aren't registering. |
| 06:31.27 | *** join/#asterisk zblk (~andrey@176.32.191.138) |
| 06:31.42 | ChannelZ | ...is my guess |
| 06:32.45 | aruntomar | couple of days ago, i've configured it myself and everything was working fine |
| 06:33.02 | aruntomar | then automatically, since the last 2 - 3 days, we are facing this issue |
| 06:33.24 | aruntomar | nobody has changed any config either on the server or on the client sip (xlite) phone. |
| 06:33.50 | aruntomar | that's why i mentioned it as a weird problem. |
| 06:34.19 | aruntomar | as we've been using asterisk for more than 4 yrs now, and haven't faced such an issue. |
| 06:34.43 | ChannelZ | I haven't used x-lite in ages, does it have a register/unregister button? |
| 06:34.56 | aruntomar | yes |
| 06:35.33 | aruntomar | i've tried that as well (register/unregister), even exit the program and then start it again. |
| 06:35.40 | ChannelZ | And on the asterisk console with some verbose on ( -rvvv or 'core set verbose 3') if you hit that on one of the malfunctioning x-lites, you see nothing? |
| 06:36.14 | aruntomar | we were earlier using zoiper, but then it was having sound (disturbance) issues, and xlite was working fine, so switched to xlite. |
| 06:38.36 | aruntomar | can't even set debug on that peer with the command "sip set debug peer <peer> " as it throws error, unable to get ip address of peer <peer> |
| 06:38.57 | ChannelZ | of course, it doesn't know what its IP is. |
| 06:39.35 | r00f | if you know the device ip, try if tcpdump shows anytyhing from it |
| 06:40.49 | aruntomar | ChannelZ, by the way, what sip softphone do you use on windows? |
| 06:43.16 | aruntomar | and what's the recommended softphone on linux. earlier we were using twinkle, but it's no longer supported on the latest linux distros/versions. |
| 06:43.17 | ChannelZ | zoiper classic when I do, which is rare |
| 06:44.01 | ChannelZ | Blink works in Windows, never tried it on Linux |
| 06:45.24 | ChannelZ | I just installed X-Lite and when I botch the account settings, I get a big message in the window "Account failed to enable .... Click here to retry" |
| 06:45.46 | ChannelZ | If you're not seeing that, has "Register with domain and receive calls" become unchecked in the Account settings? |
| 06:47.36 | aruntomar | i'll check that setting. |
| 06:48.03 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:49.14 | ChannelZ | (although I still don't know why your broken peers are able to make outgoing calls without Asterisk knowing their IP, that shouldn't be working if allowguest really is not on) |
| 06:50.16 | aruntomar | yes, even i wondering the same thing |
| 06:50.21 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 06:50.59 | aruntomar | i've rechecked the settings and allowguest is set to no |
| 06:51.16 | ChannelZ | unless you have those devices configured as friends and not peers |
| 06:51.44 | aruntomar | yes, the type is friend |
| 06:51.49 | ChannelZ | Oh.. well that's why |
| 06:52.18 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 06:52.39 | ChannelZ | (and generally speaking not a good idea, but that's not your problem.. but it's why they can make outgoing calls) |
| 06:55.12 | aruntomar | ok, i found that other setting in xlite that you asked me to check. |
| 06:55.34 | aruntomar | under domain proxy, there is register with domain and receive calls., which is unchecked. |
| 06:55.47 | ChannelZ | Actually it's not strictly a bad idea using friends per se. In your case it's the only reason they can even make calls :) But let's say it's more rare to use friends. |
| 06:55.54 | ChannelZ | So you need to check that to get X-Lite to register |
| 06:56.26 | aruntomar | i don't have a sip proxy, hence, unchecked that option |
| 06:56.31 | aruntomar | guess, it was a bad idea. |
| 06:56.53 | ChannelZ | Well you sort of do.. asterisk |
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| 06:57.10 | aruntomar | thanks for the help ChannelZ |
| 06:57.12 | ChannelZ | but click 'Domain' for the 'outbound via' |
| 06:57.41 | aruntomar | ok |
| 06:57.51 | aruntomar | even i don't use xlite |
| 06:58.12 | aruntomar | but as zoiper was creating problem so started using it, and was not familiar with it's settings. |
| 06:58.22 | ChannelZ | heh it doesn't look too bad actually. The last time I used it was way the hell back on XP and it looked like a silver phone pad thing. and I never could get the audio to work right. |
| 06:58.36 | ChannelZ | This one is at least simple and clean looking, sans the stupid ad flashing at the bottom |
| 06:59.01 | aruntomar | i need to find a good softphone for linux now. |
| 06:59.25 | aruntomar | i was using twinkle because i could integrate click to dial functionality with it. |
| 07:00.03 | aruntomar | with zoiper i couldn't, but if anyone has done click to dial with zoiper, i would love to know about it. |
| 07:00.27 | ChannelZ | Click to dial? Like clicking phone numbers from a webapge? |
| 07:00.40 | ChannelZ | s/webapge/webpage/ |
| 07:00.46 | aruntomar | hence, i'm not looking at sipjs etc javascript based sip clients. |
| 07:00.54 | aruntomar | yes |
| 07:01.50 | aruntomar | there is web based crm, and users need to click on the phone number to dial. |
| 07:02.51 | ChannelZ | Well you can try Blink.. "Handle TEL and SIP URLs clicked in other applications " |
| 07:03.12 | aruntomar | sure, i'll give it a try. |
| 07:03.57 | aruntomar | on the webrtc on asterisk, i've query. when a sipjs client registers and authenticates with asterisk, it shows up in sip show peers list. |
| 07:05.26 | aruntomar | assume there are 10 sip peers, and all of them have different id/pass to auth and they are in a office behind nat. the asterisk server is on the cloud. so when they login/register, sip show peers shows the host ip as the public ip. |
| 07:06.39 | aruntomar | so, sip show peers, show 1 single ip address for all the 10 peers |
| 07:07.02 | aruntomar | will that not create a problem while receiving incoming calls. |
| 07:07.12 | ChannelZ | same as vanilla sip peers |
| 07:07.19 | ChannelZ | They should all have different port numbers though |
| 07:07.46 | ChannelZ | So no it's not a problem |
| 07:08.58 | aruntomar | so, do i need to configure something specifically to handle this different port numbers, or it'll be handled by asterisk automatically. |
| 07:09.25 | ChannelZ | [Phone LAN IP:5060] -> [Firewall - Mangles to WAN IP:39483] -> [asterisk] |
| 07:10.38 | ChannelZ | No, the NAT firewall mangles the port numbers. Asterisk just sees the traffic coming from the IP:Port and sends traffic back on it. The NAT firewall maps the mangled port back to the correct IP on the LAN |
| 07:11.36 | Stefan27 | i asked this before but lost info- what's the easiest way to redirect the core dumps of asterisk upon crash to /home not /tmp |
| 07:12.00 | ChannelZ | (the 39483 in my example above is an aribtrary port number, will be different for each peer behind the firewall) |
| 07:12.56 | aruntomar | ChannelZ, ok, so basically, i don't need to touch anything on asterisk or sip client. the firewall will handle it. |
| 07:14.28 | ChannelZ | Stefan27: are you running asterisk with -g ? |
| 07:14.52 | ChannelZ | Because supposedly it dumps "in the present working directory of the Linux shell from which Asterisk was started." |
| 07:15.21 | Stefan27 | starting it from safe_asterisk called from systemctl |
| 07:15.38 | ChannelZ | aruntomar: you shouldn't need to, no. I've not used webrtc specifically but it aught work the same as sip peers or any other NAT traffic in general |
| 07:15.58 | Stefan27 | i see-- |
| 07:16.34 | aruntomar | Thanks again, ChannelZ :) |
| 07:16.46 | ChannelZ | besides NAT config options, same as for SIP peers behind NAT, to make asterisk do things like ignore the LAN IP in messages it gets from peers behind NAT.. though with webrtc I think they rely heavily on STUN and ICE and all that to sort that out |
| 07:19.50 | ChannelZ | Well good luck and have fun y'all.. I'm off to bed |
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| 07:52.38 | wasanzy | hello folks |
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| 08:03.00 | wasanzy | I need more help please, am very stuck |
| 08:04.40 | wasanzy | Am trying to setup a conference in a way that, I initiate the conference and have an audio playing, so anyone can join by dialing a code, to listen to the audio. Currently, it is not working. |
| 08:05.44 | wasanzy | my configs: http://pastebin.com/aLgkHRj9 and the output when I initiate the conference using a call file and joining by dialing 602 on my softphone http://pastebin.com/Rh9pvxFf |
| 08:07.34 | r00f | they need to talk to each other? why do you need conference? they can just dial some extension which PlayBack's to them |
| 08:11.59 | wasanzy | Yea I need conference, because I don't want each person who dial in to start listening to the audio from beginnig, the person will listen from where the audio got to before he or she join. perhaps, this is going to be a streaming audio on another server. am only using this to test the setup |
| 08:12.46 | wasanzy | the call file basically initiate the conference and get the audio going for anyone who join to listen to it. |
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| 08:53.16 | tekzilla | can someone recommand a simple tool for the linux commandline to test SIP registration ? |
| 08:54.13 | wdoekes | tekzilla: sipsak -U |
| 08:54.59 | wdoekes | or you could use sipp, it's less simple but way more powerful |
| 08:56.14 | tekzilla | wdoekes: great thanks, i'll try sipsak first |
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| 09:22.21 | mayai | Hello asterisk community ! |
| 09:25.16 | *** part/#asterisk riess82 (~riessma@194-166-84-59.adsl.highway.telekom.at) |
| 09:33.37 | mayai | i feel a bit stupide but i cant find the rigth information , are Asterisk can be a VABX ? i know Centrex are a VABX it's an material or software ?? sorry for he stupide question , thanks yo |
| 09:42.04 | wdoekes | mayai: I don't know what a "VABX" is. |
| 09:43.11 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:43.49 | wdoekes | mayai: see the promo talk here http://www.asterisk.org/get-started <-- what can you do with asterisk |
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| 09:45.32 | mayai | IPBX Virtual |
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| 09:47.20 | Stefan27 | sip show sched shows <unknown> : 48 what might those be? |
| 09:58.15 | wasanzy | any help? |
| 09:58.59 | mayai | helo Wasanzy |
| 09:59.03 | mayai | oups hello |
| 09:59.04 | wasanzy | Am trying to setup a conference in a way that, I initiate the conference and have an audio playing, so anyone can join by dialing a code, to listen to the audio. Currently, it is not working. my configs: http://pastebin.com/aLgkHRj9 and the output when I initiate the conference using a call file and joining by dialing 602 on my softphone http://pastebin.com/Rh9pvxFf |
| 09:59.31 | wasanzy | mayai: hello |
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| 10:02.25 | Zetten | Is there an equivalent to the sip.conf option "vmexten" in pjsip.conf? |
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| 11:01.05 | Zetten | I discovered the "mwi_from_user" option, but it doesn't seem to be used as I would expect - the 'From' header in the MWI NOTIFY message doesn't use it |
| 11:01.47 | Zetten | basically some of our phones are receiving a 'From' of their own sip address, so pressing the voicemail button results in them dialing themselves |
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| 11:21.08 | marceloamorim | guys, why I don`t have module app_chanisavail.so on lib after I upgraded my asterisk? |
| 11:22.46 | marceloamorim | if I copy the module from another asterisk, like I`m installed the asterisk11cert7, and I had cert4, can I copy the module from the cert4 to /usr/lib/asterisk/modules? |
| 11:24.36 | WIMPy | I assume you can, but it should refuse to load. |
| 11:25.55 | marceloamorim | should I compile again? |
| 11:26.17 | WIMPy | yes |
| 11:26.32 | WIMPy | But switch the module back on first :-) |
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| 11:27.37 | r00f | what is the best approach to save SIP debug for each call? preferably in mysql |
| 11:28.13 | r00f | is it achievable without some third party proxy in front of asterisk? |
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| 11:59.17 | marceloamorim | WIMPy: what is the best policy for recompile the system? like, I didn`t get the chan_sip for the first time, when I recompile again I make sure that I mark the chan_sip from menuselect |
| 11:59.26 | marceloamorim | and now I didn`t get the module =( |
| 12:00.15 | marceloamorim | but, If I try again, should I do something different? Like, the last time I just stop the asterisk and enter in another version I downloaded and compile from there |
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| 12:02.22 | WIMPy | You can copy menuselect.makeopts from your old version to get the same config, as far as it still applies, that is. |
| 12:03.24 | marceloamorim | but I meant, how can I make sure that this time the asterisk will download all modules? |
| 12:04.05 | WIMPy | They are all there. You just select to make them or not. |
| 12:04.52 | marceloamorim | well, I didn`t find app_chanisavail when I make menuselect |
| 12:05.23 | marceloamorim | but I`ll try. thx man |
| 12:05.53 | WIMPy | Are you sure you read the whole list? |
| 12:06.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:07.12 | WIMPy | It's only sorted alphabetically withing the support status sections. |
| 12:08.03 | marceloamorim | I`ll check again |
| 12:09.42 | marceloamorim | MENUSELECT_APPS=app_skel app_adsiprog app_alarmreceiver app_amd app_chanisavail app_dahdiras app_dictate app_externalivr app_fax app_festival app_getcpeid app_ices app_image app_ivrdemo app_jack app_meetme app_minivm app_morsecode app_mp3 app_nbscat app_osplookup app_saycounted app_sms app_talkdetect app_test app_url app_waitforring app_waitforsilence app_zapateller app_dahdibarge app_readfile app_setcallerid |
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| 12:11.25 | marceloamorim | app_chanisavail is there, I don`t know what the module isn`t there /usr/lib/asterisk/modules |
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| 12:12.17 | marceloamorim | this file is on /usr/src/certified-asterisk-11.6-cert7/menuselect.makeopts |
| 12:12.26 | marceloamorim | but, I`ll try compile again |
| 12:12.55 | WIMPy | Did you install to the same location as before? |
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| 12:13.53 | ndb | Hi, I have a pjsip python client that access a dummy alsa device, the first run it identifies on enum_snd_dev after killing the client and getting up again the device disapears (I am using snd-dummy) |
| 12:14.09 | ndb | any suggestions on how to implement a dummy device with pjsip |
| 12:14.38 | marceloamorim | what do you mean? Like, I put the cert4 and the cert7 in the /usr/src. like /usr/src/certified-asterisk-11.6-cert7/ and /usr/src/certified-asterisk-11.6-cert4/ |
| 12:15.36 | WIMPy | ndb: Where is the relation between that client and your alsa devices? Or to Asterisk for that matter? |
| 12:15.40 | marceloamorim | I downloaded the cert7 put at this /usr/src and compile and install after I stopped the asterisk-cert4 |
| 12:16.05 | WIMPy | marceloamorim: Did you configure it with the same parameters? Especially prefix? |
| 12:16.50 | ndb | WIMPy: it is a SIP client, and my pjsip is compiling with alsa support |
| 12:17.04 | ndb | no directly related |
| 12:17.38 | marceloamorim | kind of, I didn`t mark some options in this installation |
| 12:18.15 | WIMPy | was talking about parameters to ./configure. |
| 12:18.39 | marceloamorim | the basic, just ./configure without any special |
| 12:20.47 | marceloamorim | should I remove all files on /usr/lib/asterisk/modules before I recompile? to make sure all new files go there again? |
| 12:21.26 | WIMPy | 'make install' will moan if there's anything left over from an old install. |
| 12:22.12 | marceloamorim | and if the module change some script they will parse and install the new one? |
| 12:22.19 | WIMPy | So if you had modules before that haven't been built now, you should have seen that warning. |
| 12:22.31 | WIMPy | What script? |
| 12:23.35 | marceloamorim | I meant lines in that module, like any change on modules the make install will parse the modules? |
| 12:24.55 | WIMPy | I'm not sure what you'er on about. |
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| 12:26.19 | marceloamorim | don`t worry, It was a stupid question I think, they probably parse the module trying to find any change |
| 12:27.17 | WIMPy | If you build a new version, everything is new. |
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| 12:36.52 | Stefan27 | my rpm does the sequence ./configure ;;; make menuselect.makeopts ;; menuselect/menuselect --disable-category MENUSELECT_CORE_SOUNDS ;; make |
| 12:37.29 | Stefan27 | if i want to import a static makeopts i just skip the make menuselect.makeopts and menuselect step? |
| 12:38.55 | Stefan27 | does menuselect/menuselect --disable thisandthat only write to file menuselect.makeopts? and does 'make menuselect.makeopts' do something weird? |
| 12:39.48 | WIMPy | hasn't used or even seen that before. |
| 12:41.00 | Stefan27 | hard to trial and error test it when it's the middle of a long time consuming process :) |
| 12:42.10 | WIMPy | I can only tell you that an old menuselect.makeopts usually doesn;t work without make menuselect. |
| 12:42.36 | WIMPy | Long time? Are you compiling on a raspberry Pi? |
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| 13:08.14 | marceloamorim | I don`t know whats is going on =( |
| 13:08.47 | marceloamorim | I save using the make menuselect, and when I`ll check if the app is there, is not =( |
| 13:09.24 | marceloamorim | I`ll copy the menuselet.makeopts from cert4 like you said |
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| 13:17.14 | *** join/#asterisk wanna (~wanna@194.183.244.5) |
| 13:18.14 | WIMPy | Did you check with make menuselect? |
| 13:18.25 | marceloamorim | yes =( press f12 |
| 13:18.29 | WIMPy | And then mak and make install? |
| 13:18.32 | *** join/#asterisk saint_ (~saint@66.85.174.42) |
| 13:18.39 | marceloamorim | and didn`t save on menuselect.makeopts |
| 13:19.06 | marceloamorim | I tried menuselect/menuselect --enable app_chanisavail menuselect.makeopts too |
| 13:19.12 | WIMPy | It worked without saving? |
| 13:19.13 | marceloamorim | and didn`t work if I list |
| 13:19.22 | marceloamorim | using menuselect/menuselect --list-options |
| 13:20.44 | saint_ | So my upgrade to Asterisk 13 did not go too well.. No big issue, but could not compile Asterisk because of this: http://pastebin.com/XjzcEpJy Can anyone give me a hint on a fix ? |
| 13:21.06 | *** join/#asterisk ApPrOaCh (~sgadgil@122.175.33.214) |
| 13:21.19 | WIMPy | What are you experimenting there? Just make menuselcet, check that things look the way you want, save and then continue. |
| 13:22.32 | marceloamorim | yeah, there is something wrong |
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| 13:27.38 | ApPrOaCh | I am from India, I am using TDM800P card, I need to setup appropriate cadence for India, I have setup tonezone=28 (India) under chan_dahdi.conf and loadzone = in , defaultzone = in under /etc/dahdi/system.conf , not sure if I am missing anything ..thx |
| 13:29.18 | saint_ | WIMPy any idea on my make problem by any chance ? http://pastebin.com/XjzcEpJy |
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| 13:30.17 | WIMPy | saint_: No. That looks like a bug to me. Any reason why you build tests? |
| 13:31.26 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 13:31.46 | saint_ | WIMPy 1.8 is not supported by the end of 2015 anymore. We are going to try selling Asterisk -once again- in my company, so i'd like to show an "up to date" version of it, with the latest features ... |
| 13:32.19 | WIMPy | Why are you building test modules? |
| 13:32.25 | [TK]D-Fender | saint_: They changed the EOL on it? |
| 13:33.12 | WIMPy | ~asterisk-versions |
| 13:33.12 | infobot | Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:34.43 | saint_ | [TK]D-Fender i thought it was supposed to be 2018 or 2017 , but i went back again not too long and saw 2015. i don t know if it s my memory or if they changed it. |
| 13:35.34 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:35.35 | saint_ | WIMPy well.. not really.. but i did not find out where to remove anything related to "test" modules .. |
| 13:36.06 | saint_ | WIMPy in menuselect, all the test modules have XXX already. so i gfigured out it was not that |
| 13:36.31 | WIMPy | Hmm. Looked like one of those to me. |
| 13:38.35 | saint_ | WIMPy any clue on how to disable that crap ? |
| 13:39.00 | WIMPy | Disabling ari should help. |
| 13:40.28 | saint_ | WIMPy just res_ari , or res_ari_* ? |
| 13:41.12 | WIMPy | I'd assume the others need the core module. |
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| 13:43.16 | saint_ | WIMPy ooohhh yyeeeaahhh... compilation is passed the error now.. |
| 13:44.11 | saint_ | okay .. well that will be scheduled for tonight again ... thanks for the trick WIMPy |
| 13:45.26 | WIMPy | It's not a trick. It's disabling something. |
| 13:46.04 | saint_ | s/trick/hint/ |
| 13:46.29 | WIMPy | One of the features to show off missing. |
| 13:47.48 | saint_ | I'll be back later. |
| 13:47.52 | saint_ | thanks again |
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| 13:58.14 | ApPrOaCh | Need help to setup CallerID detection and Cadence setup, I tried various options not working as expected, any pointer will be useful, this is for India site |
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| 15:05.11 | Zetten | Is it possible to forcibly unregister a pjsip contact server-side, as listed in "pjsip qualify $ENDPOINT"? |
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| 15:42.46 | t4nk911 | hi+ |
| 15:42.49 | t4nk911 | hi |
| 15:43.32 | t4nk911 | i need some help for asterisk ...... |
| 15:45.42 | [TK]D-Fender | ~ask |
| 15:45.42 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 15:48.42 | t4nk911 | OK! I'm configuring SIPml5 with FreePBX distro 6.12.65 (64bit). The user can register to pbx and have a call but no audio on both ends. The version of asterisk is 11.13. Someone can help me ? |
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| 16:09.44 | ApPrOaCh | Need help to configure CallerId on asterisk , working with Digium TDM800P card, this need to be configure in India, tried various options under chan_dahdi.conf, not working as expected , will provide more details if needed |
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| 17:26.04 | kannan | hello, I have a context where i added a dead agi. i have several calls already running live. can i issue a dialplan reload , will the already established calls also execute the dead agi (version asterisk 1.8.24.0) |
| 17:28.52 | kannan | can we safely issue a dialplan reload under heavy loads generally, loadavg is over 4 on a 8 core cpu, with 140 - 150 concurrent calls |
| 17:34.48 | mjordan | generally, the answer is 'yes'. That being said, the question is: why are you having to reload the dialplan that frequently? |
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| 17:44.49 | kannan | mjordan, thanks, its not frequent at all, just to know |
| 17:49.52 | mjordan | we have several tests that hammer 'dialplan reload' in the presence of ongoing activity. So I'm not aware of any bugs you may encounter with issuing that command at any time. |
| 17:50.23 | mjordan | Do keep in mind that calls that are in flight will *not* keep the dialplan they are executing - as soon as they are able to do so, they'll use the new dialplan. So you may have some interesting behavior if you remove something that they were executing |
| 17:52.37 | WIMPy | So to be safe, always create new contexts with the new contents and keep the old ones untill all ongoing calls have gone. |
| 17:52.44 | WIMPy | Anyone scripted that already? |
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| 18:39.51 | R3nan | hello, can someone help me |
| 18:42.01 | R3nan | how to reduce noise on an iax trunk, sorry bad english |
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| 18:50.05 | [TK]D-Fender | R3nan: Noise is generated at the end that is converted to IAX2. |
| 18:50.11 | [TK]D-Fender | This is not created BY IAX2 |
| 18:53.25 | R3nan | [TK]D-Fender, what can be the problem? |
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| 19:01.16 | [TK]D-Fender | the other end of the call |
| 19:01.26 | [TK]D-Fender | if this is a provider then they probably have noisy lines. |
| 19:01.35 | [TK]D-Fender | At which point there is nothbing you can do to fix this. |
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| 19:15.31 | R3nan | [TK]D-Fender, tks |
| 19:15.38 | Frojoe | Hi all |
| 19:16.30 | Frojoe | I have a sip peer who keeps becoming unreachable 66 seconds after registering on my asterisk server. The asterisk console gives me this: chan_sip.c:24938 sip_poke_noanswer: Peer 'prophecy' is now UNREACHABLE! Last qualify: 89 |
| 19:17.04 | Frojoe | I've tried increasing the qualify timeout on the peer and experimented with different NAT settings in sip.conf,but to no avail |
| 19:17.04 | WIMPy | Frojoe: Enable keepalives on that device. |
| 19:17.51 | Frojoe | "Device" meaning the peer connecting or the the asterisk host? |
| 19:17.59 | WIMPy | yes |
| 19:24.35 | Frojoe | so does that just mean adding qualify=yes to the peer definition? |
| 19:25.02 | WIMPy | It the other end an Asterisk as well? |
| 19:25.24 | Frojoe | No. It's a Voxeo prophecy server. |
| 19:25.41 | WIMPy | Then I don't understand the question. |
| 19:25.50 | WIMPy | See if you can enable keepalives there. |
| 19:26.07 | Frojoe | Ah |
| 19:26.29 | WIMPy | If you can't you can decrease Asterisks qualifyfrequency, but that will increase load if you have many peers. |
| 19:37.16 | *** join/#asterisk saint_ (~saint@66.85.147.138) |
| 19:38.12 | saint_ | Are any of you running Asterisk on ESXI ? |
| 19:39.28 | saint_ | or any other kind of virtual machine ? |
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| 19:43.32 | Frojoe | saint_, yes |
| 19:43.40 | Frojoe | Though I'm not sure how much help I can provide |
| 19:43.45 | saint_ | Frojoe: esxi or vmware or something else ? |
| 19:44.03 | Frojoe | Currently XEN |
| 19:45.19 | saint_ | Frojoe that s in the cloud, right ? |
| 19:45.35 | [TK]D-Fender | I'm not seeing an actual request for help... |
| 19:45.38 | [TK]D-Fender | ~polls |
| 19:45.38 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 19:46.37 | saint_ | Frojoe is it a production server ? how many users, if you do not mind me asking ? |
| 19:47.05 | Frojoe | Not production, at most maybe 10 concurrent users' |
| 19:48.17 | Frojoe | And no, it's a local XEN instance |
| 19:50.05 | ChannelZ-Wk | HEY! Does anyone here like hot dogs!? |
| 19:50.45 | saint_ | Frojoe okay. I was asking because I do not have a backup here of our Asterisk. Our management is so cheap. I was telling another guy that if the box crashed, we would be screwed . I was thinking about migrating it to a esxi because at least I can do snapshots of the OS (for example before an upgrade, so I can roll back if it does not work) and I can restore it quickly to another esxi if the virtual server crashes itself. We tried in the past virtual |
| 19:50.45 | saint_ | <PROTECTED> |
| 19:52.06 | [TK]D-Fender | ESXI works well, so does KVM. |
| 19:52.21 | [TK]D-Fender | XEN has more issues than these from reports I've seen here |
| 19:52.27 | Frojoe | Xen is pretty stable |
| 19:53.04 | Frojoe | At least in my experience. It's slightly less of a pain in the ass these days as xen support is cooked into the mainline kernel now, so you don't need to compile xen kernels anymore |
| 19:53.23 | saint_ | [TK]D-Fender : I will try to migrate to esxi tonight then. The install / upgrade to release 13 did not go well. Esxi will allow me to run another instance without disturbing the prod server |
| 19:53.44 | Frojoe | But yeah, I'd go with either KVM or vmware if you have minimal exposure to virtualization |
| 19:56.14 | Frojoe | Though as an aside, in order to use snapshotting with KVM or XEN, you need to use LVM/ZFS/BTRFS for your storage |
| 19:56.56 | saint_ | Frojoe i have an esxi running already here. got it a couple of months ago, and well powered, and running about 15 VMs |
| 19:57.40 | pigpen | yeah, VMWare ESXi is our friend. We are running several instances of Asterisk connecting to sip gateways. |
| 19:58.08 | saint_ | pigpen prod or test asterisks ? how many users in each ? No issue at all ? |
| 20:00.20 | pigpen | We have only about 10 running in production. Users/phones from 20 - 200. No issue at all? Not really, cough it up to luck or good planning. |
| 20:01.10 | saint_ | pigpen good to know. |
| 20:01.29 | pigpen | This is also on customers who have a well established infrastructure. ie: 2 or more SANâs, 3 or more VHosts. |
| 20:01.47 | pigpen | Dedicated iSCSI network, large UPSâs. |
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| 20:04.13 | saint_ | pigpen I was making some researches today, and I saw some people using esxi with a SAN for the datastorage. I was thinking about putting asterisk on the same type of configuration, that would be awesome to prevent lose of data due to hd crash |
| 20:04.26 | saint_ | we have synology here .. |
| 20:05.53 | pigpen | yeah, we donât do other configs. ie: No SAN, No Virtualization from us. |
| 20:06.24 | pigpen | In fact, we got tired of using other manufacture SANâs. We do our own based on Linux. |
| 20:07.39 | saint_ | It's funny, I just realized actually Synology as an Asterisk package available ... |
| 20:07.39 | pigpen | Just did a 24 drive enterprise Intel SSD drive array, with iSCSI 10GB to a pair of Extreme Networks x670âs (I think that was the model). |
| 20:08.23 | pigpen | That box is amazing with the IO. The customer liked it so much, they are ready to buy a second. Only about $44k. |
| 20:08.50 | pigpen | for the SAN. I think the switch was ~12K. But we have others we just do direct connect 10G. |
| 20:09.00 | saint_ | how much was the hardware, if you don t mind asking ? We quoted a solution for a fire department (asterisk / san / network) - trying to SAN using our own hardware was about $2k more expensive than buying Synology directly. And synology came with more storage than they could afford if they bought hardware directly |
| 20:09.31 | pigpen | Lots of variables. Their solution would be different than others as you can imagine. |
| 20:09.45 | saint_ | AH.. Asterisk package on Synology is release 1.8 .. damn it. it s the one i m trying to get away from. |
| 20:09.54 | malcolmd | don't use the Sinology package for Asterisk |
| 20:10.02 | malcolmd | you'll be tempted to use their package for the asterisk-gui also |
| 20:10.04 | pigpen | We use Gentoo for the SAN and the Asterisk. |
| 20:10.09 | malcolmd | and then you'll post to the forums about it |
| 20:10.16 | malcolmd | and there will be sadness because no one supports the asterisk-gui |
| 20:10.23 | malcolmd | and david55 and i will frown |
| 20:10.52 | pigpen | heh. we have our own scripts to handle asterisk. Easier. And when there is a PHP update, bash isnât affected. ;-) |
| 20:11.23 | saint_ | malcolmd I heard that yesterday , no worries .. lol ... |
| 20:11.39 | malcolmd | ./s/Sinology/Synology ...bleh |
| 20:11.41 | saint_ | malcolmd do you guys work with them directly for their asterisk package, or are they on their own ? |
| 20:11.45 | malcolmd | they're on their own |
| 20:11.46 | pigpen | Oh, sorry if this sounded like a sales job. I was clicking past my irc channels and saw vmware. |
| 20:12.03 | malcolmd | and it makes me really, really sad that they package asterisk-gui |
| 20:12.17 | pigpen | yeah, asterisk-gui, yuck. |
| 20:12.59 | saint_ | malcolmd well... one of the advantage of this san box is its GUI , comparing to any other (like dropbo) . it's awesome . I can see that they had over 114,000 download of the Asterisk package.. not too bad .. |
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| 20:48.47 | bachler | can anyone tell me how to set up a voip server that would allow me to use it to connect (i have public ip, domainname and all that set up) my samsung s5 sip account to it and some other sip phone and then call between them. I am a experianced linux user but i am new to asterisk and all this voip stuff. just need to get in the right direction, I have googled for some hours now and got a freepbx set up on a raspberrypi. but all tutorials I could find |
| 20:48.47 | bachler | seems old and most of them use some kind of paid voip service. |
| 20:49.34 | WIMPy | Just ignore that part. |
| 20:49.43 | WIMPy | You only configure your devices. |
| 20:50.26 | WIMPy | If you need help with FreePBX, ask in #freepbx. If you want to conffigure Asterisk yourself try |
| 20:50.30 | bachler | ok, so I just skip the register line then |
| 20:50.31 | WIMPy | ~primer |
| 20:50.31 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 20:50.34 | WIMPy | or |
| 20:50.38 | WIMPy | ~book |
| 20:50.39 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:51.12 | bachler | im not sure that ill use freepbx, just wanted to try something really fast. |
| 20:51.52 | bachler | probably will try and get asterisk up on raspian when i have had a play with it |
| 20:53.26 | bachler | but I should be able to connect my samsun s5 as a sipphone and make calls to another smartphone (say another samsung as a sipphone) on my same server? |
| 20:54.11 | bachler | and if I then expose some ports thru my firewall in to my "server" i should be able to use it everywhere |
| 20:54.48 | WIMPy | yes |
| 20:55.37 | bachler | very nice, thanks |
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| 22:52.15 | WIMPy | Argh. Why don't quotes really work? |
| 22:57.33 | Milos | Because who knows. |
| 22:58.15 | Milos | I think there is a bug in this android native SIP clients. If you call a remove party and then hang up before they answer, it keeps ringing the other end and asterisk soon shows messages about retransmission timeouts. |
| 22:58.31 | Milos | If I paste debug output would someone be able to confirm? |
| 22:59.04 | WIMPy | Sounds more like a networking issue. |
| 22:59.25 | Milos | It's on the same subnet though. |
| 22:59.31 | Milos | and that's the only bug. |
| 22:59.41 | Milos | everything else works including talking, ringing inwards, etc. |
| 22:59.51 | Milos | and outwards if they answer |
| 23:00.09 | WIMPy | Strange. |
| 23:00.28 | Milos | I installed CSipSimple client on the same phone and it works correctly with that |
| 23:00.32 | Milos | but I want to find out what's wrong with the native client |
| 23:04.24 | Milos | is it using the wrong call ID? https://bpaste.net/show/a10fc3cab115 |
| 23:15.25 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 23:16.46 | ChannelZ-Wk | Maybe, but you never showed us the original INVITE |
| 23:18.27 | Milos | okay, I'll post the whole thing |
| 23:20.21 | Milos | ChannelZ-Wk, https://bpaste.net/show/6639be0e78b5 |
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| 23:27.47 | ChannelZ-Wk | hmm |
| 23:49.25 | Milos | hmm? |
| 23:49.34 | Milos | it's weird isn't it |
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| 23:59.43 | ChannelZ-Wk | recompiling to figure out which instance it's not matching the call ID |