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| 12:34.33 | stkoch | hi, I make a test with asterisk behind a NAT. The server is connected to an SIP provider for calls. Clients are connected over network to the asterisk server. |
| 12:35.25 | stkoch | If the clients are connected by a mobile internet connection I can't hear nothing |
| 12:36.12 | stkoch | Via: SIP/2.0/UDP 192.168.133.52:10740;received=93.218.19.241;branch=xyz;rport=10740 |
| 12:36.18 | phix | yaya |
| 12:36.21 | stkoch | Why is there a local IP? |
| 12:36.44 | phix | stkoch: cause you haven't set an external IP? |
| 12:36.56 | stkoch | phix: the external is the 93... |
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| 12:37.51 | stkoch | phix: sip.conf: externaddr=93.218.19.241:10740 |
| 12:42.52 | stkoch | Is this correct? |
| 12:43.05 | phix | no idea |
| 12:43.09 | phix | ask WIMPy |
| 12:44.51 | WIMPy | Why? So I can tell that it's perfectely normal for SIP not to work? |
| 12:46.25 | WIMPy | That's the part that enables you to earn money. |
| 12:46.25 | stkoch | phix: WIMPy: it works now |
| 12:46.49 | stkoch | after setting |
| 12:46.50 | stkoch | localnet=192.168.0.0/255.255.0.0 |
| 12:46.58 | chandoo | Packet timed out after 32000ms with no response |
| 12:47.05 | chandoo | i am getting this error |
| 12:47.17 | stkoch | phix: WIMPy: thanks |
| 12:47.27 | chandoo | call between devices is getting disconnected after 32secs, how to fix this issue |
| 12:47.30 | WIMPy | Sounds like round about the same kind of issue. |
| 12:49.40 | phix | WIMPy: hehe |
| 12:49.45 | phix | stkoch: yes, use IAZ! |
| 12:49.48 | phix | IAX! |
| 12:50.27 | WIMPy | We just had to add subscriptions to IAX and there wouldn't be a need for anythgin else. |
| 12:50.53 | stkoch | phix: is it supported by CSipSimple for android? |
| 12:51.30 | phix | WIMPy: I have never used WIMPy |
| 12:51.51 | WIMPy | What? Subscriptions? |
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| 12:53.56 | phix | WIMPy: IAX |
| 12:54.00 | phix | I don't know what that is |
| 12:54.27 | WIMPy | The best thing from Asterisk. |
| 12:56.13 | stkoch | WIMPy: what's the preffered IAX client for android? |
| 12:58.37 | phix | stkoch: I just use SIP client that comes with Android (from ginger bread and above) |
| 13:00.20 | stkoch | phix: is it possible to select codecs for low and high speed connections? |
| 13:00.37 | phix | With default client? |
| 13:00.52 | stkoch | phix: yes |
| 13:01.42 | phix | no idea |
| 13:17.23 | chandoo | phix, any idea how to fix the 32sec issue |
| 13:18.42 | WIMPy | Check your NAT settings. |
| 13:20.43 | file | does http://www.asteriskdocs.org/ work fine for others? |
| 13:21.20 | WIMPy | Works for me (tm) |
| 13:21.28 | file | nifty |
| 13:21.32 | phix | chandoo: I think they have pills for that now |
| 13:21.58 | WIMPy | *ggg* |
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| 17:19.28 | qakhan | is there any way we can save DTMF values user sent in Read() app |
| 17:20.00 | qakhan | like what digits user press, every time in loop |
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| 17:35.49 | skyroveRR | I've opened a port to my asterisk server, port 5060, and I've port forwarded 5060 from my openwrt router to the asterisk server, do I still need to use the nat option? |
| 17:36.25 | WIMPy | Do you also want audio? |
| 17:36.32 | skyroveRR | I do. |
| 17:38.46 | skyroveRR | WIMPy? |
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| 17:40.13 | Milenco | skyroveRR, WIMPy means you also need to make sure the rtp packets can get through |
| 17:40.57 | Milenco | i dont think openwrt supports sip alg through luci |
| 17:41.19 | Milenco | you'll need the install an iptables nathelper for sip and manually configure iptables to forward all the traffic |
| 17:41.31 | Milenco | or forward the whole range of udp ports for the rtp packets |
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| 17:45.04 | skyroveRR | Milenco: ok. |
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| 17:45.44 | Milenco | 5060 doesnt contain the audio packets |
| 17:45.58 | skyroveRR | Understood. |
| 17:46.00 | skyroveRR | Thank you. :) |
| 17:46.18 | Milenco | np, gl :) |
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| 18:27.25 | [TK]D-Fender | <qakhan> is there any way we can save DTMF values user sent in Read() app <- you get the variable ... go save it |
| 18:45.19 | qakhan | [TK]D-Fender here is my dialplan |
| 18:45.41 | [TK]D-Fender | qakhan, no need |
| 18:45.45 | [TK]D-Fender | You gtg a variable at the end |
| 18:45.50 | [TK]D-Fender | SAVE IT YOURSELF |
| 18:45.55 | [TK]D-Fender | Whatever way you want |
| 18:45.56 | [TK]D-Fender | to |
| 18:46.06 | qakhan | i can save lastest DTMF in variable. |
| 18:46.18 | [TK]D-Fender | What "astest"? |
| 18:46.23 | [TK]D-Fender | You get a complete variable |
| 18:46.38 | [TK]D-Fender | Do whatever you want with it |
| 18:46.53 | qakhan | but i want to save all DTMF in 1 variable user sent every time |
| 18:48.01 | [TK]D-Fender | What do you mean "every time"? |
| 18:48.16 | [TK]D-Fender | Read does what it does. You get ONE variable. What you do from then on is up to you |
| 18:49.56 | qakhan | user press 2 1st time to listen the message again, value 2 saves in ${input}, |
| 18:50.11 | qakhan | <PROTECTED> |
| 18:51.20 | [TK]D-Fender | It's YOU job to copy the result somewhere else to collect them |
| 18:51.29 | [TK]D-Fender | You know how to SET a variable? |
| 18:51.30 | [TK]D-Fender | DO IT |
| 18:51.44 | [TK]D-Fender | Get input. Take result and ADD to some other variable. CONTINUE |
| 18:53.42 | qakhan | ok but when next time value comes it will override the new variable |
| 18:53.50 | [TK]D-Fender | Says who? |
| 18:53.56 | [TK]D-Fender | I didn't say OVERWTIRE IT |
| 18:53.59 | [TK]D-Fender | I said ADD TO IT |
| 18:54.09 | [TK]D-Fender | A = A + B |
| 18:54.10 | [TK]D-Fender | ^^^ |
| 18:54.48 | [TK]D-Fender | You put ANYTHING you want after the = |
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| 18:55.53 | qakhan | you mean result = ${input} + ${input} |
| 18:55.56 | qakhan | ? |
| 18:56.21 | qakhan | everytime value will come in ${input} |
| 18:57.36 | [TK]D-Fender | What you call your variable doesn't matter |
| 18:57.41 | [TK]D-Fender | <[TK]D-Fender> A = A + B |
| 18:58.17 | [TK]D-Fender | that is the most basic logic in all of programming |
| 18:58.49 | qakhan | but this can only work 2 time. if user wants to listen message then ? |
| 19:00.46 | [TK]D-Fender | Says who? |
| 19:00.47 | [TK]D-Fender | ADD MORE |
| 19:00.53 | [TK]D-Fender | KEEP ADDING MORE |
| 19:00.59 | [TK]D-Fender | You don't seem to understand basic programming |
| 19:01.23 | [TK]D-Fender | <qakhan> you mean result = ${input} + ${input} <- of course not. this makes no sense |
| 19:01.31 | [TK]D-Fender | And it doesn;'t look like what I SHOWED YOU |
| 19:01.36 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> A = A + B |
| 19:04.31 | qakhan | so where B will get value from? |
| 19:04.54 | [TK]D-Fender | B is WHAT YOU JUST READ IN |
| 19:05.13 | [TK]D-Fender | ADD TO THE END |
| 19:08.04 | qakhan | ok [TK]D-Fender |
| 19:08.08 | qakhan | Thanks i got |
| 19:08.34 | qakhan | it |
| 19:25.35 | [TK]D-Fender | Show us what you're finished |
| 19:25.39 | [TK]D-Fender | when* |
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| 19:39.23 | tompaw | Hello |
| 19:40.33 | tompaw | A year and a half ago I managed to get some serious non-free help here from one of the ops, who claimed he was going to be available in future for similar services. |
| 19:40.53 | tompaw | Unfortunately, I forgot his name... and now my system grows and I need to pick one of your brains again. |
| 19:41.27 | WIMPy | puts on his tin foil hat to prevent brain picking. |
| 19:42.40 | tompaw | Anyway, this regards a call center solution based on Asterisk, I described my questions in detail here: http://hastebin.com/xemamelupe.txt |
| 19:44.02 | tompaw | If that person is reading this - speak up man. If anyone else wishes to help - for free or not - please do :-) |
| 19:44.23 | WIMPy | won't allow everyone to execute scrips on his computer. |
| 19:44.43 | tompaw | WIMPy: scripts? |
| 19:45.23 | WIMPy | Your URL desn't have any content, just scripts. |
| 19:45.40 | tompaw | It's a pseudocode that describes what I'm doing... |
| 19:45.57 | [TK]D-Fender | ARE doing, or WANT to be doing? |
| 19:46.15 | WIMPy | No, it's just Javascript. |
| 19:47.18 | tompaw | [TK]D-Fender: well, most bits are in place, what's missing is referred to in the questions. |
| 19:47.46 | tompaw | WIMPy: Javascript? Where do you see any JS there?.... |
| 19:48.13 | WIMPy | On the URL you gave. |
| 19:49.39 | tompaw | WIMPy: it's just a pastebin.... |
| 19:51.15 | WIMPy | Yes. One with only scripts. |
| 19:54.33 | tompaw | I really don't know where you see any script code there, these are just pseudo functions that describe what I do and where... |
| 19:57.40 | tompaw | Anyway, let me try and ask some generic non-case-specific questions then and see what happens. |
| 20:01.46 | tompaw | On 13.x, I have a system based on ConfBridges, which works together with a web-based CRM. The CRM sends instructions to Asterisk via ACM (telnet). What would be the most reliable method of informing the CRM about events such as new party joining a ConfBridge? |
| 20:02.17 | WIMPy | Listen on AMI. |
| 20:05.29 | tompaw | What about ARI and WebSockets? |
| 20:06.17 | WIMPy | Yes, but that would be instead of using ConfBridge then. |
| 20:06.29 | qakhan | [TK]D-Fender did you ask me? |
| 20:07.22 | tompaw | WIMPy: why? How are events broadcasted via AMI different from the ones sent by Asterisk to a WebSocket? |
| 20:07.41 | file | AMI is different than ARI. |
| 20:08.39 | WIMPy | With ARI you control the channel instead of using dialplan applications. AMI informs you about everythign that ahppens and also alows to take some action. |
| 20:09.57 | [TK]D-Fender | <[TK]D-Fender> Show us what you're finished |
| 20:09.57 | [TK]D-Fender | <[TK]D-Fender> when* |
| 20:11.19 | tompaw | Understood. Is it a good idea to keep one AMI session with events opened and use separate ad-hoc sessions to control Asterisk from the CRM? Or would it be better if it was all in one stream? |
| 20:11.58 | WIMPy | Whatever suits you. |
| 20:12.13 | tompaw | The latter does. |
| 20:12.29 | WIMPy | But Just keeping one connection might be more efficient if it fits to how you CRM integration works. |
| 20:13.47 | tompaw | Understood. |
| 20:14.26 | tompaw | Is there a limit to how many AMI connections asterisk can handle concurrently? |
| 20:15.05 | qakhan | [TK]D-Fender here Set(result=${result}+${input}) |
| 20:15.24 | WIMPy | Probably no intended one, unless you configure it to only allow one per user, but I haven't tried. |
| 20:15.31 | file | there is no fixed one |
| 20:15.42 | [TK]D-Fender | qakhan, I mean the full code |
| 20:17.18 | tompaw | OK, I do not expect to have more than 10 sessions at once, hopefully this will not break things. |
| 20:17.27 | file | 10 won't |
| 20:18.30 | qakhan | [TK]D-Fender here http://pastebin.com/ns7RWCQb |
| 20:19.20 | tompaw | Great. Now, from the AMI I pull parties into ConfBridges using: channel originate [PARTY] extension [CRM_CALL_ID]@cont-join, where cont-join looks more or less like this: http://hastebin.com/zobapagazo.txt |
| 20:21.04 | tompaw | Do I need a separate context for the initial ConfBridge set up, or is Asterisk smart enough not to damage the record file even if Set(CONFBRIDGE(bridge,record_file... is called multiple times on the same ConfBridge? |
| 20:21.36 | [TK]D-Fender | qzkYou are doing silly things in there |
| 20:21.53 | [TK]D-Fender | qakhan, You are doing aplayback of that filename twice back to back |
| 20:22.58 | qakhan | yes it is required |
| 20:23.36 | [TK]D-Fender | qakhan, Why would you say it twice in a row immediately? |
| 20:23.55 | [TK]D-Fender | qakhan, You also aren't doing anything with the input, and I see no point in what you're doing with ${key} |
| 20:24.10 | [TK]D-Fender | exten => s,n,Goto(s,play) <- no need to semtion the extension on this either |
| 20:25.51 | qakhan | [TK]D-Fender i am saving DTMF keys and number of attempts user listen the message in DB |
| 20:26.02 | qakhan | that is why ${key} there |
| 20:26.46 | [TK]D-Fender | Where are you saving the input anywhere? |
| 20:26.50 | [TK]D-Fender | Those are channel variables. |
| 20:26.53 | [TK]D-Fender | You are HANGUP UP{ |
| 20:26.59 | [TK]D-Fender | they DIE after that |
| 20:27.13 | tompaw | It's quite a broad question but: how stable is ConfBridge on 13.x? I remember that on 1.8 and before it required some dark kernel voodoo and still acted up from time to time. |
| 20:27.31 | WIMPy | No. It never did. |
| 20:27.42 | tompaw | Was it the MeetMe? |
| 20:27.48 | WIMPy | MeetMe required DAHDI. |
| 20:27.50 | [TK]D-Fender | Meetme required DAHDI |
| 20:27.54 | [TK]D-Fender | Confbridge never did |
| 20:28.12 | [TK]D-Fender | WIMPy, One cap off... impressive synchronization |
| 20:28.30 | [TK]D-Fender | and a period, but enough blood's been spilled already ;) |
| 20:29.27 | tompaw | So as we stand today, on a E5 (6c ht) server with 64GB ram, can I expect 100-200 ConfBridges to work reliably? (there will be a mix of g729 and g711 unless it's a problem) |
| 20:29.50 | qakhan | [TK]D-Fender i did not post last line |
| 20:29.57 | tompaw | Each ConfBridge with 2-3 participants? |
| 20:30.26 | WIMPy | Don't expect anyone to comment on expected performanece. Too many variables. |
| 20:30.27 | qakhan | exten => h,1,AGI(/script/call_statusupdate.php,${key},${input}) |
| 20:30.38 | [TK]D-Fender | qakhan, doesn't matter if there is 1 more line. What we see shows an explicit hangup |
| 20:30.47 | [TK]D-Fender | qakhan, That 100% kills the call |
| 20:31.24 | [TK]D-Fender | exten => s,n,SET(result=${result}+${input}) <- and I don't think you actually want the extra characters in there |
| 20:32.14 | qakhan | yes i am get + before the 1st digit |
| 20:32.20 | qakhan | getting* |
| 20:33.05 | [TK]D-Fender | I doubt you want ANY + in there at all |
| 20:33.22 | qakhan | ok |
| 20:33.31 | qakhan | yes |
| 20:35.22 | tompaw | WIMPy: fair enough. Let me rephrase the question then: what is an acceptable max load per core to reliably support ConfBridge? |
| 20:35.29 | [TK]D-Fender | There is therefor no point in collecting ${key} at all. |
| 20:35.57 | qakhan | [TK]D-Fender i am confuse |
| 20:36.00 | WIMPy | Load doesn't mean much, |
| 20:36.46 | tompaw | What does then? what is the single most important resourse to watch in order to assess that my server is pushed to its limits? |
| 20:36.53 | qakhan | ${key} for number of time a use listend the message |
| 20:37.13 | qakhan | 1 time, 2 time, n time |
| 20:37.14 | WIMPy | You realy have to try it out. |
| 20:37.17 | [TK]D-Fender | qakhan, Which you don't need to colelct |
| 20:37.19 | [TK]D-Fender | collect |
| 20:37.24 | qakhan | both |
| 20:37.26 | Cyford33 | is there a funtion in dialplan where i can aleert the user he has a vm outside the voicemail() app |
| 20:37.40 | qakhan | now i am able to collect both |
| 20:37.47 | [TK]D-Fender | qakhan, You don't HAVE to. |
| 20:38.00 | tompaw | WIMPy: Not the answer I was hoping for, but thanks anyway. |
| 20:38.02 | [TK]D-Fender | qakhan, You are not looking at what you're already doing in there |
| 20:38.19 | [TK]D-Fender | qakhan, There is NO point in doing what you are doing with ${key} |
| 20:38.48 | WIMPy | At least ConfBridge should be able to use multiple cores. |
| 20:39.24 | tompaw | Do you know how much performance is lost on handling multiple codecs in single room? |
| 20:39.41 | tompaw | (if any)? |
| 20:39.58 | WIMPy | 'core show translation' |
| 20:40.10 | tompaw | that's one to one, isn't it? |
| 20:40.31 | WIMPy | sure |
| 20:40.39 | qakhan | [TK]D-Fender what do you suggest |
| 20:41.07 | file | each channel has to be transcoded to signed linear in order to mix it |
| 20:41.17 | tompaw | WIMPy: but my understanding is that Asterisk needs to encode a new "merged" audio stream anyway for every single participant. So does it even matter if they use different codecs? |
| 20:41.37 | file | and transcoded back when going out |
| 20:41.57 | tompaw | file: so it doesn't make too much of a difference if there are different codecs used, does it? |
| 20:42.33 | file | each codec has a different transcoding cost |
| 20:42.46 | WIMPy | Only as far as they have different complexity in transcoding. |
| 20:43.17 | WIMPy | The fact that they are different in itself does not matter. |
| 20:43.52 | tompaw | So if you have two parties, A and B, both using - say g711. Would asterisk encode the merged audio stream A+B *separately* for each one of them? |
| 20:44.15 | [TK]D-Fender | qakhan, LENGTH OF INPUT = KEY |
| 20:44.24 | [TK]D-Fender | qakhan, You clearly don't need to COLLECT IT |
| 20:44.34 | [TK]D-Fender | qakhan, Just look at the STRING LENGTH |
| 20:44.34 | tompaw | just realized how stupid his question was. |
| 20:44.38 | qakhan | ok i got it. |
| 20:44.40 | tompaw | Please disregard it. |
| 20:44.51 | WIMPy | Not really. |
| 20:45.00 | qakhan | i did got ${key} first. |
| 20:45.14 | tompaw | Well, obviously each participant has to receive a DIFFERENT stream, with his own voice excluded. |
| 20:45.30 | tompaw | No? |
| 20:45.35 | qakhan | i was looking for ${input} which you helped me |
| 20:45.53 | WIMPy | No |
| 20:46.24 | tompaw | Seriously? |
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| 20:47.12 | WIMPy | There is only one sum. |
| 20:47.46 | file | well - yes, there's only one sum, but if the channel being provided it spoke then their audio is removed |
| 20:48.10 | file | and if silence detection is enabled then you may have multiple channels that did not speak - and they'll receive the same thing |
| 20:48.56 | Cyford33 | is there a funtion in dialplan where it can alert the user he has a vm outside the voicemail() app... in the IVR |
| 20:49.03 | tompaw | file: and will it be encoded separately for each one of them? |
| 20:50.29 | file | I do not remember |
| 20:50.46 | [TK]D-Fender | Cyford33, "core show functions like VM |
| 20:50.47 | [TK]D-Fender | " <- you should already have looked at the function list and found these |
| 20:55.26 | Cyford33 | lol didnt know core show funtions .. thnks |
| 20:55.45 | [TK]D-Fender | That's half of what there is in the dialplan........ |
| 20:55.50 | [TK]D-Fender | Applications & functions... |
| 20:56.04 | [TK]D-Fender | The list is also all over the WIKI(s) |
| 20:58.34 | Cyford33 | yes, i been using google to figuire things out up to now, and use core show help... i couldnt find my answer till now.. thanks a million |
| 20:59.37 | [TK]D-Fender | you're welcome |
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| 21:26.36 | tompaw | All right, AMI events it is. Thanks for your time, guys. |
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| 22:25.00 | przerull | exit |
| 22:25.29 | przerull | ip config |
| 22:31.43 | file | this is not the screen you are looking for |
| 22:36.20 | ChannelZ | rm -rf ./* |
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| 23:17.55 | dan_j | Hi. Is there any way to enable debug or verbose on realtime Db connections? |
| 23:18.38 | dan_j | Asterisk regularly (once every 2 days) stops responding to SIP messages for about 30 seconds. I assume it's a DB issue but want to see what Asterisk is doing internally with the mysql/ODBC calls. |
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| 23:37.18 | ChannelZ | I don't know but you could certainly do it on the datbase side |
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| 23:51.20 | andoh | Hello everyone. |
| 23:51.42 | andoh | Anyone here use pyst? :) |
| 23:52.46 | andoh | I am stuck on a rather interesting error/issue with a python script. |
| 23:52.53 | andoh | If anyone has the mind to help? :) |
| 23:57.16 | andoh | Damn agi thing.. |
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