| 00:01.04 | Kobaz | cal/s@MsgCenter_Dialer_DialPhone-da44;2!MsgCenter_Dialer_DialPhone!s!12!Ringing!Dial!SIP/240,10!918882472425!!!3!8!SIP/240-00001bde!1371427226.10742 |
| 00:01.07 | Kobaz | that's why |
| 00:01.13 | Kobaz | SIP/240 Answer()'s |
| 00:01.23 | Kobaz | but asterisk still considers it ringing |
| 00:01.47 | Kobaz | [2013-06-16 20:00:30] -- <Local/s@MsgCenter_Dialer_DialPhone-da44;2> calling SIP/240-00001bde (callee answered) |
| 00:29.31 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:35.58 | *** part/#asterisk nny (~Scott@cpe-024-088-116-144.sc.res.rr.com) |
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| 00:58.08 | pabelanger | What sort of performance different would transcode_via_sln = yes / no offer? |
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| 02:08.51 | Kobaz | oh |
| 02:08.55 | Kobaz | pabelanger: hmm what's that do? |
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| 03:01.46 | killown | is there a way to reload sip evey N seconds? |
| 03:02.36 | killown | I have a voip accounts and after some minutes it expires the register in some way and asterisk won't detect that and will keep saying it's registered |
| 03:02.44 | killown | so sip reload re-register it |
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| 03:15.51 | Kobaz | while [ 1 ]; do sleep 300; asterisk -rx "sip reload"; done |
| 03:17.20 | Kobaz | heh |
| 03:17.24 | Kobaz | or actually |
| 03:17.26 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 03:17.33 | Kobaz | it's right in the docs |
| 03:17.33 | Kobaz | ister => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 03:18.06 | Kobaz | killown: asterisk isn't going to 'detect' registration expiration |
| 03:18.36 | Kobaz | killown: it's the remote servers job to keep track of that registration until the host becomes unreachable for a time |
| 03:18.41 | Kobaz | so it sounds like your remote host is broken |
| 03:21.11 | killown | Kobaz, I am using insecure=port,invite and it seems the problem |
| 03:21.30 | killown | it register in the first time and in the seconds seems get forbidden auth |
| 03:21.46 | killown | I am trying now insecure=very |
| 03:22.02 | Kobaz | insecure has to do with calls, not registration |
| 03:22.39 | killown | so it's forbidden calls, then that explain why it still keep registered |
| 03:22.45 | killown | forbidding* |
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| 03:31.22 | killown | what does reinvite=no set up? |
| 03:33.03 | Kobaz | that's also known as directmedia, depending on the version |
| 03:33.23 | Kobaz | it will make the endpoints talk directly to each other... it'll have generally have problems if you use nat |
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| 03:33.42 | Kobaz | and it wont work if you're doing call recording |
| 03:33.57 | killown | I am recording calls |
| 03:36.27 | Kobaz | then yeah that's not going to do anything |
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| 03:55.40 | killown | Kobaz, my issue is pretty similar to this one http://forums.digium.com/viewtopic.php?f=1&t=85254&start=0 |
| 03:55.54 | killown | I am using the same voip provider |
| 03:56.59 | Kobaz | you're using originate? |
| 03:57.42 | killown | Kobaz, what is originate?/ |
| 03:58.06 | killown | oh sorry |
| 03:58.09 | Kobaz | then you're probably not using it |
| 03:58.11 | killown | not using it |
| 03:58.17 | Kobaz | it's a way to tell asterisk to spawn a call |
| 03:58.42 | killown | Kobaz, the error is this: |
| 03:58.45 | killown | <--- SIP read from UDP:201.86.87.35:5060 ---> |
| 03:58.45 | killown | SIP/2.0 403 Forbidden - Wrong domain or Username format |
| 03:58.54 | Kobaz | your provider is telling you that your auth is wrong |
| 03:59.35 | killown | Kobaz, sorry wrong error |
| 03:59.38 | killown | this one: |
| 03:59.46 | killown | <--- Reliably Transmitting (no NAT) to 201.10.87.35:5060 ---> |
| 03:59.48 | killown | SIP/2.0 401 Unauthorized |
| 04:00.01 | Kobaz | that's also saying that your provider is telling you that your auth is wrong |
| 04:01.17 | killown | Kobaz, it works sometimes, and sometimes not |
| 04:01.44 | Kobaz | then you should put in a ticket with the provider |
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| 04:48.11 | phix | Because that capital E makes all the difference |
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| 05:06.14 | killown | No matching peer for '1133224455' from '201.10.87.35:5060' then > <--- Reliably Transmitting (no NAT) to 201.10.87.35:5060 ---> |
| 05:06.14 | killown | <PROTECTED> |
| 05:06.27 | killown | seems something related to the peer settings? |
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| 05:13.36 | killown | when I enable anonymous guest support it seems to fix that SIP/2.0 403 Forbidden issue |
| 05:13.54 | killown | I am still testing if it's fixed |
| 05:19.39 | killown | still not fixed :/ |
| 05:20.34 | killown | at least it has no issues with No matching peer for... only with SIP/2.0 403 Forbidden |
| 05:21.22 | [TK]D-Fender | If you are looking for help you should be showing us your peer and the complete call debug |
| 05:23.24 | kaldemar | killown: for the registration part, set expiry in the register statement. see a sample sip.conf for the syntax. |
| 05:24.18 | killown | [TK]D-Fender, http://bpaste.net/show/9fhQe45xYrvcZKhRp3rC/ |
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| 05:24.53 | killown | kaldemar, I am using defaultexpiry=1800 |
| 05:24.53 | killown | maxexpiry=3600 |
| 05:25.04 | killown | but still have the same issue |
| 05:25.24 | kaldemar | killown: for incoming calls you need a peer that matches calls from the provider and insecure=port,invite in case they don't authenticate to you. allowguest would only hide the issue. |
| 05:25.28 | [TK]D-Fender | I don't see an error in there... |
| 05:26.02 | kaldemar | killown: as i said, see the sample config. those are for incoming registrations. |
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| 05:28.48 | killown | kaldemar, maybe it was lacking bindaddr=0.0.0.0 and srvlookup=yes |
| 05:29.15 | killown | lets test it |
| 05:31.01 | kaldemar | killown: are you looking at the register statement syntax and its expiry part? |
| 05:31.32 | killown | kaldemar, I am trying to find it |
| 05:37.20 | killown | I am using google now to find this register thing |
| 05:38.57 | kaldemar | http://svn.digium.com/svn/asterisk/tags/11.4.0/configs/sip.conf.sample |
| 05:39.07 | killown | kaldemar, thank you |
| 05:40.31 | killown | kaldemar, register => login:senha_vono@vono.net.br/vono~expiry ? how do I use expiry in that string? |
| 05:42.42 | killown | I included minexpiry=1800 |
| 05:46.04 | killown | seems to be login:senha_vono@vono.net.br/vono~3600 |
| 05:55.30 | killown | kaldemar, seem to work now, thanks a lot |
| 05:55.59 | killown | maybe my voip provider gives it as forbidden if the expiry time is very low |
| 05:56.15 | killown | so that explains why sometimes it worked and sometimes not |
| 05:56.59 | killown | registered first 60 seconds < working > 120 seconds not working > 190 seconds working and so on |
| 05:57.14 | killown | /190/180 |
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| 06:12.50 | tparcina | How to find out how many concurent calls I head yesterday? |
| 06:13.58 | tparcina | The best thing I have found so far is the report from FreePBX. |
| 06:14.25 | tparcina | Just there are some issues with it. |
| 06:14.46 | tparcina | I need to chose the day, then check for every hour seperately. |
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| 06:15.23 | v0lZy | tparcina: I dont know the answer, but your question seems to be at odds with itself. You probably want to see how many concurrent calls you had at a certain moment. 1 day constitutes a rather large interval in which you had calls ending also |
| 06:15.57 | v0lZy | tparcina: you probably want to know the maximum number of concurrent calls you experienced yesterday |
| 06:16.15 | tparcina | v0lZy: Yes, you are right. |
| 06:17.00 | v0lZy | tparcina: Other than writting your own dialplan that does some counting for you, I think what you want to know is only avialable with processing the call records report |
| 06:17.01 | tparcina | My question wasn't clear enough. |
| 06:17.55 | tparcina | v0lZy: That what I was thinking as well. |
| 06:17.58 | v0lZy | If you are in particular interested about yesterday for some reason, then I think you have to consult the call records if you keep them. |
| 06:18.33 | v0lZy | tparcina: you could export them to a txt file, then grep for the hour you are interested in |
| 06:18.48 | v0lZy | tparcina: but i suggest a different approach if you want to use this functionality in the future |
| 06:18.54 | tparcina | v0lZy: And I have search for some solution that will process my Master.csv (or mysql database), but I couldn't find anything usefull. |
| 06:20.01 | tparcina | All discussions are about calculating how many simultaneous calls Internet connection or CPU/memory can handle. |
| 06:20.52 | v0lZy | tparcina: in your dialplan, add a counter to your sections of the dialplan which involve calling outside your pbx and calls incomign from the outside to your pbx (im supposing thats the only part you are interested in). Increment the number for each outgoing call made by you and each incoming call made to you. Whenever a call ends, dont send it to hangup immidiately, send it to a macro or something and there, decrement the number by 1 for each call |
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| 06:20.57 | tparcina | v0lZy: I believe I would need to use this solution more often. |
| 06:21.21 | v0lZy | That way you will have a counter going up and down based on the number of simultanious active calls |
| 06:21.37 | v0lZy | whenever a call is in place, the counter will increment by 1, and whenever its hungup, it will decrement by 1 |
| 06:21.53 | v0lZy | At any point in between, you will see a number indicating the amount of ongoing calls |
| 06:22.22 | tparcina | Thank you for suggestion. |
| 06:22.25 | v0lZy | the only thing left to do is to add an if clause that will echo this number to a file, if it is larger then the number already in the file |
| 06:22.42 | v0lZy | you can then at any time cat that filee |
| 06:22.51 | v0lZy | And it will show you the peak. |
| 06:23.42 | v0lZy | make it a different file for each day if you want to keep history |
| 06:23.51 | v0lZy | based on what the linux date command returns. |
| 06:24.20 | v0lZy | or simply, just clear the file at midnight via cron |
| 06:24.49 | v0lZy | since you are comparing the variable values, it will always show the number of peak calls after first write... |
| 06:25.00 | v0lZy | means it could be off by 1 concurrent call. |
| 06:25.25 | v0lZy | but if you copy the file first and name it according to date, then you make it completely accurate. |
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| 06:26.21 | v0lZy | But then you have to have 2 lines... one showing peak at start and one showing peak otherwise if the start was from 0 |
| 06:26.37 | tparcina | That would be nice solution. |
| 06:26.43 | v0lZy | cause if you peak on 8 concurrent yesterday, then transfer that number to today, and don't peak it, it will continue showing 8 until you reach 9. |
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| 06:27.44 | tparcina | Just it would be nice to have one extra option, and that is ability to see how the number of concurrent calls was changing during the period of time. |
| 06:27.58 | tparcina | And this solution doesn't provide that. |
| 06:28.09 | v0lZy | well |
| 06:28.17 | v0lZy | you could modify it to do so |
| 06:28.34 | v0lZy | but thats not 'peak' information then |
| 06:28.49 | v0lZy | it makes more sense to just keep the total amount of calls then |
| 06:28.53 | v0lZy | choose your increment step |
| 06:29.11 | v0lZy | then based on that, run a cron script summerizing the number in the file |
| 06:29.17 | v0lZy | like every 10 minutes or so |
| 06:29.35 | v0lZy | u should probably use SQL for this or something |
| 06:30.27 | v0lZy | but basically select * from 'concurrent_peak' where date exactly '2013-06-16' or something |
| 06:30.43 | v0lZy | it should then give you all results in 10 minute steps. |
| 06:30.47 | v0lZy | tadaaa |
| 06:30.49 | v0lZy | :D |
| 06:32.04 | v0lZy | you could then with some php build a graph from it :) |
| 06:32.11 | tparcina | You have interesting suggestions. :) |
| 06:32.24 | tparcina | Thank you for that. :) |
| 06:32.45 | v0lZy | would love to check out your code once you are done, lolz |
| 06:33.08 | v0lZy | I dont like SQL stuff, but i guess i need to bite into it one day sooner or later. |
| 06:33.28 | tparcina | Just I wonder, maybe it's not necessary to do all those things since all information I already have in Master.csv and in mysql database. |
| 06:33.56 | v0lZy | the information in your call records is for when each individual call starts and each invidual call ends. |
| 06:34.08 | v0lZy | its up to you to calculate how many were concurrent |
| 06:34.20 | v0lZy | you could process it by something like |
| 06:34.35 | tparcina | Maybe it would be useful to other people as well, if they just could use this solution on their old data. |
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| 06:34.59 | v0lZy | grep all incoming/outgoing calls between start_of_interval and end_of_interval |
| 06:35.15 | v0lZy | but the logic gets fuzzy |
| 06:35.47 | v0lZy | since you have to count how many were ended and how many were started... you dont necesserily know. |
| 06:35.58 | tparcina | It is lots of work. And, just like everyone else, I don't have that much time to do it. :( |
| 06:36.19 | v0lZy | tparcina: if you imagine that you hav ea call log something like |
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| 06:36.28 | tparcina | That why I was hoping I'll find some solution that has everything I need. |
| 06:36.36 | v0lZy | 2013-06-16 00:00 Incoming call 1 |
| 06:36.59 | v0lZy | and 2013-06-16 00:05 ENDOF Incoming call 1 |
| 06:37.05 | v0lZy | then its easy. |
| 06:37.09 | v0lZy | but what if you have |
| 06:37.25 | v0lZy | 2013-06-15 23:55 Incoming call 1 |
| 06:37.32 | v0lZy | and 2013-06-16 00:05 ENDOF Incoming call 1 |
| 06:38.04 | v0lZy | then when you will grep for the info, you will see 'ENDOF' and will decrement by 1.... |
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| 06:38.12 | tparcina | I have call 1 incoming on channel at 12:03 end at 12:05. Basicly that is what is written in Master.csv |
| 06:38.55 | v0lZy | tparcina: if you want number of concurrent calls, then you need to grep at a certain point, not at a certain interval |
| 06:39.03 | v0lZy | as soon as you go the interval way, things are messy |
| 06:39.27 | tparcina | I'm not good in mysql nor PHP, that's why I think that someone else could have done it much better than I could. |
| 06:39.42 | v0lZy | well i think for accurate data |
| 06:39.49 | tparcina | But, if soon I don't find anything usefull, I just might try to solve the issue on my own. |
| 06:39.57 | v0lZy | you kind of have to keep track of them as they appear/disappear if you want PEAK |
| 06:40.06 | v0lZy | its 2 solutions you are asking for, not just 1 |
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| 06:40.48 | tparcina | Thank you on your insight on this topic. |
| 06:40.59 | v0lZy | solution 1) get peak in day. solution 2) Get graph of how many calls during each of N incremental snapshots. |
| 06:41.01 | tparcina | Now I'll search a bit more. |
| 06:41.36 | v0lZy | good luck |
| 06:42.21 | tparcina | v0lZy: Thank you. :) |
| 06:44.26 | v0lZy | you're welcome |
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| 07:36.45 | _omer | hello |
| 07:38.16 | _omer | A caller is in the queue and agent answers his call....I want to get the talk time of agent-caller .... I have tried variables like CDR(duration) , CDR(billsec) .. but none of them is exact talk time between agent and caller. |
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| 07:47.35 | bulkorok | hello |
| 07:48.00 | bulkorok | _omer: check queue_log |
| 07:48.46 | bulkorok | _omer: btw. if yxou don't have Answer() in your dialplan, billsec should be the exact connection-time |
| 07:52.51 | _omer | if I dont answer() then will the caller listen to MOH? |
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| 07:53.54 | _omer | I was wondering if there is any variable that provide actual talk time between agent and caller .. otherwise I have to develop my own ways to calculate talk time. |
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| 08:54.25 | gavimobile | I have a remote peer which has crappy wifi internet it connects and disconnects. in the cli I see it says retransmiting a bunch of times. can I change the time cycle asterisk checks if the peer is connected |
| 08:54.26 | gavimobile | ? |
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| 09:21.55 | Greenlight | I think you can specify in the qualify= option |
| 09:22.03 | Greenlight | As the number of milliseconds |
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| 09:23.45 | kaldemar | if you define qualify in milliseconds, you define the time asterisk waits for the peer to answer to a single qualify message. qualifyfreq defines the interval that asterisk sends them. |
| 09:24.25 | Greenlight | Ahh there you go |
| 09:28.00 | gavimobile | hrm, qualify is set to yes |
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| 09:28.23 | gavimobile | ahh ok, so qualifyfreq |
| 09:29.38 | gavimobile | another question. I just created another company which I want it to use a different country context from the indications.conf where and how do I specify which contenxt to use |
| 09:30.10 | kaldemar | "yes" uses the default value of 2000 ms. |
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| 10:19.10 | Greenlight | Any ideas what would cause: " ast_rtcp_write_rr: RTCP RR transmission error, rtcp halted: Operation not permitted |
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| 10:57.04 | wdoekes | Greenlight: operation not permitted could indicate a firewall on the OUTPUT chain |
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| 11:20.51 | Greenlight | Naa, output chain is empty |
| 11:21.06 | Greenlight | I think it may be been as I had directmedia enabled at one side and not the other |
| 11:21.18 | Greenlight | I've corrected that, and am monitoring now, fingers crossed! |
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| 13:28.06 | giany | if Asterisk sends a REGISTER request and sets Expires: 1800 in it and the provider in its 200 OK sets in Contact, expire=300, what will happen after 300 seconds? Will asterisk re-register? |
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| 13:42.18 | [TK]D-Fender | it should |
| 13:43.18 | giany | sorry? |
| 13:46.11 | [TK]D-Fender | wHAT'S THE QUESTION? |
| 13:46.27 | [TK]D-Fender | "Will asterisk re-register?" --> it should |
| 13:46.35 | [TK]D-Fender | Not sure what there is to misunderstand... |
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| 13:47.00 | WIMPy | That was easy, but not an answer. |
| 13:47.23 | [TK]D-Fender | "what will happen after 300 seconds? Will asterisk re-register?" |
| 13:47.33 | [TK]D-Fender | after 300 seconds.. it should register |
| 13:48.06 | giany | [TK]D-Fender: ok, then that is my problem, after 300 seconds it does not register.. |
| 13:48.11 | giany | most likely a bug in 1.6.1.x |
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| 13:51.58 | jmetro | i have sip autodestructs getting way out of hand now |
| 13:52.04 | jmetro | every day if im not watching they start flooding the server |
| 13:52.25 | WIMPy | Then don;t stop watching :-) |
| 13:52.35 | jmetro | ive got something like 4 repeating autodestructs right now, whenever they are rolling i get users complaining they cant call people or leave vm's |
| 13:53.13 | jmetro | and one autodestruct makes others spawn because of that dead air channel, the user hangs up and there goes another autodestruct rolling along |
| 13:54.30 | jmetro | always on VM |
| 13:55.29 | jmetro | exit |
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| 14:05.10 | [TK]D-Fender | giany: 1.6.x = unsupported |
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| 14:15.21 | fredericve | where should I start looking if the T.38 gateway in asterisk sometimes does not start? |
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| 14:17.38 | Ice_Strike | How does IVR detect where I am calling from? I mean IVR know I am calling from Landline or Mobile (Both set with same callerID) |
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| 14:26.35 | igcewieling | Ice_Strike: it doesn't. |
| 14:30.25 | igcewieling | fredericve: I asked a similar question several months ago. Every single answer was totally useless. |
| 14:31.14 | fredericve | and did you find a solution to your problem? |
| 14:32.58 | Ice_Strike | igcewieling It does :) |
| 14:33.19 | Ice_Strike | Because it say different thing on the IVR |
| 14:33.28 | Ice_Strike | even with same CallerID |
| 14:33.32 | igcewieling | fredericve: no. We decided to postpone until Asterisk's support matures. |
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| 14:36.23 | fredericve | igcewieling, which version of asterisk were you using? |
| 14:37.15 | igcewieling | fredericve: Asterisk 1.8.x and Asterisk 11. Asterisk 1.8 doesn't have gateway support, of course. |
| 14:37.34 | igcewieling | we needed passthru rather than gateway support. |
| 14:38.05 | Ice_Strike | igcewieling When I dial from Mobile - the IVR know i am calling from mobile phone. |
| 14:38.30 | igcewieling | Ice_Strike: I'm happy for you. |
| 14:38.35 | Ice_Strike | When I use sip account (changing callerID to mobile number) - the IVR said please dial 124 |
| 14:38.39 | Ice_Strike | which mean dial from mobile |
| 14:38.49 | igcewieling | Ice_Strike: However, I cannot help further. |
| 14:38.59 | Ice_Strike | Oh ok lol |
| 14:39.18 | Ice_Strike | I am trying to figure out how did IVR detect it. |
| 14:40.29 | wdoekes | giany: in handle_response_register there is code that reads the expires= parameter, even in 1.6.1(.12). if you enable sip debugging, you should see a message about the expiry |
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| 14:46.16 | MLNoah | is it possible to instruct asterisk to load & cache specific realtime SIP peers on startup? |
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| 14:49.15 | igcewieling | MLNoah: no. Why do you need this? |
| 14:49.46 | MLNoah | i have a pair of servers functioning as a cluster, using RT SIP for peer registrations. |
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| 14:50.16 | igcewieling | and? |
| 14:50.31 | MLNoah | some of my peers are other PBXes, with a static IP, that I have configured for Asterisk to call directly. If a call comes in from the remote PBX to Asterisk before the sip peer has been cached out of RT, Asterisk isn't properly matching the peer's username |
| 14:51.00 | MLNoah | and is rejecting the call from the peer, instead of trying to look up the host IP from the RT database |
| 14:51.11 | igcewieling | MLNoah: It always seemed to for for us when we did something similar. |
| 14:51.16 | MLNoah | if I do a sip show peer <peername> load from the CLI, causing RT to cache the peer, things work. |
| 14:51.24 | igcewieling | keep in mind you MUST have a port set on your static peers |
| 14:52.04 | MLNoah | >< why I didn't think to check that, I'll never know... |
| 14:52.34 | igcewieling | MLNoah: when using realtime you won't get very far if you don't enable query logging on your SQL server to see what the problems are |
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| 14:53.15 | MLNoah | didn't even thing to look and see if asterisk was doing an sql query. |
| 14:53.25 | MLNoah | guess i have my noobie question of the week out of the way early this week. |
| 14:53.34 | igcewieling | we gave up on realtime and now generate our sip.conf with a script via a #exec in sip.conf |
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| 14:54.28 | MLNoah | other than my static peers, i haven't really run into anything that seems like a problem so far. particular problems you had with RT that led you to go elsewhere? |
| 14:54.47 | igcewieling | MLNoah: All of our peers are static |
| 14:55.34 | igcewieling | MLNoah: Realtime gives us nothing other than being able to use a GUI to change the database. |
| 14:56.06 | MLNoah | ah, yeah, i can see that. i'm about 70/30 dynamic vs static |
| 14:57.07 | igcewieling | MLNoah: our main issue with realtime is you cant seperate out the read ops from the write ops (for things having the database locally replicated on the local server, etc |
| 14:58.36 | MLNoah | so far, it seems to be running reasonably well against MySQL Cluster for me *crosses fingers* |
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| 14:59.54 | igcewieling | NBD cluster? |
| 14:59.58 | MLNoah | yes |
| 15:00.18 | igcewieling | *nod* since all servers can be written to, that setup would not have the issue we had. |
| 15:03.18 | MLNoah | so, my basic structure is each Asterisk node is running an SQL node + data node for the NBD cluster, and then phpMyAdmin and the other stuff I use to interface with the MySQL is on the NBD admin node. |
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| 15:03.47 | MLNoah | frustrating how poor the "setup guides" for mysql cluster are though, and the oracle documentation is long on theory, short on implementation |
| 15:04.12 | igcewieling | MLNoah: *nod* one of the reasons we did not go with NBD |
| 15:06.39 | MLNoah | any rate. thanks again for the information. |
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| 15:22.57 | Katty | good morning asteriskables. |
| 15:23.18 | Qwell | looks at Katty |
| 15:23.35 | eirirs | looks at Qwell |
| 15:24.56 | Katty | how was your trip dear |
| 15:25.13 | Qwell | Katty: It was... *sunglasses* ...OK. |
| 15:25.35 | Katty | ^______________________^ |
| 15:25.56 | Katty | eggggggcellent. |
| 15:26.22 | Katty | i get to go to sam baker next weekend! yay! |
| 15:26.28 | Katty | well, this weekend. |
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| 16:38.34 | nricciar | when you have something like exten => number/extension in your dialplan is there a way to run a specific line if there was no matching extension line? |
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| 16:59.44 | navaismo | nricciar exten => number |
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| 17:01.41 | flapjacks | you could do extent => _X. |
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| 17:51.26 | zoid_ | elguero: ping |
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| 18:00.59 | dtx | So I'm running AsteriskNow everything fully updated and the digium phones addon enabled. It works pretty well, except there is no apparent way to set the default timezone for phones if you are using easy mode. Is this a known issue or am I missing something. |
| 18:01.16 | zoid_ | Being in the context of an agi script, if some command returns -1, how can I differenciate if it's because the command itself failed or de user hung up? |
| 18:01.44 | dtx | This is in FreePBX, btw. |
| 18:02.22 | tm1000 | dtx: #freepbx |
| 18:02.47 | dtx | tm1000: I'm in there as well. But it is the Digium Addons module. |
| 18:03.16 | dtx | Made by Digium. |
| 18:03.48 | tm1000 | dtx: http://pbxinaflash.com/community/index.php?threads/digium-d70-and-piaf-installer-for-digium-dpma-module.12722/#post-81139 |
| 18:04.05 | pabelanger | dtx: might want to also try opening a support ticket directly with them too |
| 18:04.19 | pabelanger | since DPM is a product of theirs |
| 18:04.22 | tm1000 | pabelanger: why? |
| 18:04.32 | tm1000 | malcolm's reply in that thread has the answer |
| 18:04.40 | tm1000 | http://pbxinaflash.com/community/index.php?threads/digium-d70-and-piaf-installer-for-digium-dpma-module.12722/#post-81140 |
| 18:04.45 | pabelanger | didn't read URL |
| 18:05.13 | dtx | Yes, but essentially you can't set a default or use Easy Mode |
| 18:05.21 | dtx | You have to set it per phone |
| 18:05.43 | tm1000 | dtx: sounds like your answer is a feature request |
| 18:05.46 | dtx | which is manageable but needlessly irritating |
| 18:05.48 | tm1000 | so, do as pabelanger said. |
| 18:06.01 | dtx | I guess so |
| 18:07.13 | pabelanger | dtx: The issue is, we don't have access to source code to provide a fix, if any. So, you'll need to open an issue with digium directly |
| 18:07.17 | *** join/#asterisk ariel_ (uid3533@pdpc/supporter/active/abatista) |
| 18:07.39 | pabelanger | If you own the phones, you should have support |
| 18:07.54 | dtx | okie dok |
| 18:08.22 | dtx | It's unfortunate that they would make that module closed |
| 18:08.36 | Qwell | The FreePBX module is GPL. |
| 18:08.42 | tm1000 | yeah as Qwell said |
| 18:08.48 | tm1000 | the source is open |
| 18:09.10 | tm1000 | pabelanger is confused between DPMA and the freepbx module |
| 18:09.51 | pabelanger | indeed |
| 18:10.06 | dtx | I guess I am too. DPMA just provides an API that freepbx module uses? |
| 18:10.15 | Qwell | dtx: no |
| 18:10.35 | Qwell | http://www.digium.com/en/supportcenter/custom-communications-solutions/dpma/what-is-it |
| 18:10.48 | Qwell | https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phone+Module+for+Asterisk+%28DPMA%29+v+1.6 |
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| 18:15.28 | elguero | zoid_: I would think that "result=-1" would indicate an error occurred... then you can get the channel variable AGISTATUS to see if it was because the the user hungup or not |
| 18:15.29 | dtx | thanks for the links |
| 18:16.28 | zoid_ | elguero: but isn't AGISTATUS set up after the agi script finishes? |
| 18:17.56 | zoid_ | using GET VARIABLE on agistatus during on runtime returns nothing |
| 18:18.18 | Qwell | It should return a status after trying to set the variable. |
| 18:18.23 | Qwell | or, issuing any command. |
| 18:21.20 | zoid_ | Qwell: It doesn't seems to be doing that, or maybe I'm doing something wrong, see: http://pastebin.com/EExuAuvE |
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| 18:36.59 | Cuzner | how do the big guys sniff out SIP gateway traffic on a per-trunk basis? |
| 18:37.55 | [TK]D-Fender | "zip set debug ip w.x.y.z" |
| 18:37.59 | [TK]D-Fender | sip* |
| 18:38.35 | Cuzner | sorry, i guess i should have said to generate call statistics |
| 18:39.15 | zoid_ | Cuzner: can't you do that using queries over cdr? |
| 18:39.30 | Cuzner | possibly |
| 18:39.33 | [TK]D-Fender | cdr != SIP traffic |
| 18:39.57 | [TK]D-Fender | And so far we have no details for what is actually desireed |
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| 18:41.17 | Cuzner | i'm looking to get total concurrent calls per trunk, I guess that's a good place to start. |
| 18:41.28 | Cuzner | this is purely for stats generation. |
| 18:41.31 | [TK]D-Fender | CDR is the place to go |
| 18:42.36 | Cuzner | Thanks, this is probably what I'm looking for. |
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| 19:26.17 | igcewieling1 | CEL can also be useful, it contains more info than the CDRs |
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| 19:47.44 | trystero | I'm planning to replace a proprietary PBX in my home with an Asterisk system. I don't (yet) care about switching to a SIP provider instead of PSTN. Should I replace the Cat3 phone lines with Ethernet and use something like Gigasets, or connect existing analog lines into a Digium card and use cheap DECT phones? |
| 19:48.58 | trystero | s/Cat3/2-wire/ |
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| 19:49.28 | derjanni | Good evening. |
| 19:49.41 | derjanni | My Raspbian asterisk package came without smsq - do I need to recompile entire asterisk?! |
| 19:50.14 | derjanni | app_sms.so is onboard and working |
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| 20:03.41 | [TK]D-Fender | trystero: You can use whatever you want for phones, and if you want to use your existing lines then you'll need an interface for those as well. |
| 20:04.00 | [TK]D-Fender | trystero: This could be a PC card of some sort, or an external gateway box |
| 20:04.46 | derjanni | can I compile smsq without compiling asterisk again? |
| 20:04.51 | derjanni | and where do I get the sourcE? |
| 20:04.57 | [TK]D-Fender | asterisk.org <- |
| 20:06.03 | Cuzner | trystero: http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht701 |
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| 20:07.56 | [TK]D-Fender | ~gs |
| 20:07.56 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:08.27 | Cuzner | dude, it's a fucking ata box, it works. |
| 20:08.46 | Cuzner | it's not like it's rocket appliances |
| 20:08.49 | [TK]D-Fender | And there are better manufacturers |
| 20:08.54 | Cuzner | not for the money |
| 20:09.08 | [TK]D-Fender | what is $10 in the big picture? |
| 20:09.40 | [TK]D-Fender | And that also only covers a single phone, and doesn't account for his phone line... |
| 20:10.16 | Cuzner | http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht702_704 |
| 20:10.24 | Cuzner | that covers it |
| 20:11.20 | [TK]D-Fender | trystero: So how many ports of each type are you looking at? |
| 20:12.10 | [TK]D-Fender | ooooohhh Cisco's got a new multi-line DECT gatweway... |
| 20:13.54 | derjanni | I haz HT503 |
| 20:14.04 | derjanni | it even doz teh sms |
| 20:14.42 | igcewieling1 | derjanni: what country are you located in? |
| 20:15.59 | trystero | [TK]D-Fender: we have 3 PSTN lines and about a dozen phones. Will probably reduce that to 2 outside lines and 4 DECT phones w/ extra handsets. |
| 20:17.35 | derjanni | Germany |
| 20:17.48 | derjanni | HT503 on Cisco EPC3212 EuroDOCSIS VoC modem |
| 20:18.06 | [TK]D-Fender | trystero: Once you're looking at a dozen phones you might want to pay attention to your overall deployment... |
| 20:18.49 | derjanni | ahhh.... just go for cable chaos |
| 20:19.06 | [TK]D-Fender | trystero: You could uses analog phones on FXS ports, but you won't get all the nice native features of dedicated SIP phones. It will be cheaper per-port though and you might have wiring concerns. Depends what you have, and what you want |
| 20:20.39 | derjanni | dect has a longer range than wifi - dont go for wifi sip phones. |
| 20:21.49 | trystero | [TK]D-Fender: thanks. I have a pretty extensive wired LAN already, but naturally the Ethernet ports aren't where I want the phones to go :-) |
| 20:22.31 | [TK]D-Fender | trystero: Good to keep in mind... how many are where you'd wnat them to be? |
| 20:22.40 | [TK]D-Fender | Might be worth going Hybrid |
| 20:22.57 | trystero | I'm still smarting from being locked into proprietary Panasonic phones, so the idea of almost-disposable DECT phones from Costco is pretty appealing |
| 20:23.40 | [TK]D-Fender | trystero: What common gear do you alrady possess? |
| 20:23.57 | [TK]D-Fender | trystero: Or is your current system all panasonic proprietary and non-reusable? |
| 20:25.04 | trystero | I have a PC that I'll use as the server, a couple of Digium cards that I got pretty cheaply on eBay, and also a couple of Cisco SIP phones (ditto) for the places where it's already convenient to plug them in. |
| 20:25.47 | derjanni | A PC? |
| 20:26.03 | derjanni | You mean a thing with that old x86 or x86_64 crap? |
| 20:26.20 | derjanni | just trolling :-P |
| 20:26.56 | [TK]D-Fender | trystero: Sounds like you have all the gear you need potentially... |
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| 20:28.54 | derjanni | can I just compile utils ? |
| 20:28.57 | derjanni | and how? |
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| 20:30.20 | [TK]D-Fender | trystero: You'd needs ATA's or larger gateways to account for the analog phones at most and should proabably be ready to go freom therem. get a count of hown many before deciding what base unit to start with... |
| 20:30.22 | [TK]D-Fender | heads home from the office... |
| 20:30.22 | [TK]D-Fender | BBIAB |
| 20:30.26 | trystero | [TK]D-Fender: yep, right now I'm thinking of a hybrid system, and moving towards more SIP phones if/when I get the phone wiring upgraded or start missing the SIP features on the other lines |
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| 20:35.29 | derjanni | http://www.raspberrypi.org/phpBB3/viewtopic.php?f=66&t=47380 |
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| 20:49.34 | igcewieling1 | 1) you can run 10Mbps ethernet on cat 3 2) going analog first will waste money |
| 20:51.03 | Cuzner | not if he's already wasted the money on analog stuff. |
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| 21:03.09 | eirirs | I guess 10mbps are more than enough for SIP phones, no? |
| 21:03.26 | jagster` | depends on how many |
| 21:03.34 | jagster` | rule of thumb is 64kbps per phone line |
| 21:03.38 | eirirs | one 10mbps for each phone |
| 21:03.46 | eirirs | heh |
| 21:04.06 | eirirs | I'm thinking about the old internal phone wiring in buildings |
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| 21:04.19 | jagster` | oh cabling? |
| 21:04.20 | jagster` | you're fine |
| 21:04.20 | eirirs | pull out old PBX and plug up a PoE switch |
| 21:04.31 | eirirs | and swap all phones to SIP ones |
| 21:04.35 | jagster` | if you're going to upgrade might as well get cat6 though its only a few cents more than cat5e |
| 21:04.47 | eirirs | ? |
| 21:04.59 | eirirs | I'm talking about reusing the old phone lines already existing in buildings |
| 21:05.06 | jagster` | right you should be fine |
| 21:05.09 | eirirs | buildings with own PBX's |
| 21:08.15 | Cuzner | SIP over token ring, that's my next project |
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| 21:11.16 | eirirs | hehe |
| 21:11.41 | gmalsack | is anyone here any good with T.38???? |
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| 21:13.59 | Cuzner | there might be, but you'll be hard pressed to find someone willing to admit to it ;) |
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| 21:14.10 | Cuzner | re leifmadsen |
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| 21:14.32 | leifmadsen | what am I not choosing to avoid admittance to? |
| 21:14.59 | Cuzner | just saying hi, didn't mean to imply that i was referring to you in my last sentence |
| 21:15.04 | Cuzner | (his question was about T38) |
| 21:15.44 | igcewieling1 | Someday we'll looks back on T.38 with nostalgia, like we do with smallbox or the plague. |
| 21:15.53 | gmalsack | wonderful!!! |
| 21:16.11 | Cuzner | faxing just won't die |
| 21:16.15 | gmalsack | hey leif - just a shout out! Thanks for all the good books. |
| 21:16.33 | gmalsack | *Cuzner - yea tell me about it. |
| 21:16.58 | gmalsack | 200 people in my companies call center, requests for faxing just don't stop.... |
| 21:17.03 | [TK]D-Fender | igcewieling1++ |
| 21:17.19 | [TK]D-Fender | igcewieling1: You've hit your Comedy Gold quota with that one! |
| 21:17.31 | igcewieling1 | 8-) |
| 21:17.37 | gmalsack | igcewieling!!! :-)) |
| 21:17.43 | Cuzner | gmalsack: see, everyone in here loves t38 :) |
| 21:18.16 | [TK]D-Fender | Cuzner: The cake is a lie.... |
| 21:18.17 | gmalsack | yea, as much as I. going on 16 hours of smacking my head against the wall. why am I not dead yet???? |
| 21:18.48 | dtx | So I patched the FreePBX Digium Phones addon so that each phone entry will include a new file called res_digium_phone_devices_global.conf . So you can essentially set global overrides for all your phones (like timezone support) without having to turn off easy mode. |
| 21:18.53 | igcewieling1 | gmalsack: VERY recent Asterisk 11 has some T.38 and UDPTL fixes |
| 21:19.38 | dtx | pretty simple patch, do they have a github for that module? Anybody know? |
| 21:19.48 | gmalsack | igcewieling1: thanks for the heads up. I'll build one up and see if that resolves the problem! |
| 21:22.57 | malcolmd | dtx: no github for it; it's just maintained in digium's svn. if you want to contribute, the best place would be https://issues.asterisk.org/jira/browse/ANOW |
| 21:24.30 | malcolmd | also, thanks for using it :) |
| 21:24.34 | dtx | malcolmd: alrighty |
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| 21:53.40 | kieppie | anyone know of a Mumble/Murmur integaration/bridge? |
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