| 00:00.24 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:13.53 | ruben23 | hi guys how do i setup custom cdr for asterisk 11 somehow and can be exported as CSV...thanks |
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| 00:45.14 | linuxfool | hi all |
| 00:46.55 | TazzNZ | hey all - my google fuu is failing me - any one know how I can force SendFax to t38 without the re-invite |
| 00:47.10 | linuxfool | has anyone here set up a small business on asterisk? |
| 00:47.18 | TazzNZ | linuxfool: most of us :) |
| 00:47.28 | TazzNZ | even bigger corporations :) |
| 00:48.32 | linuxfool | TazzNZ: what's the biggest number of extensions you've delpoyed for a single project? |
| 00:49.22 | TazzNZ | linuxfool: 700 |
| 00:49.24 | TazzNZ | +- |
| 00:49.44 | TazzNZ | company wide is about 1200 |
| 00:49.58 | TazzNZ | this site is 700, smaller site is running the rest |
| 00:50.16 | linuxfool | TazzNZ: Do you work for a telecom company or did you do the project for the company you were already working for? |
| 00:50.46 | TazzNZ | linuxfool: I still work for the company and we are not a telco - what are you after ? |
| 00:51.22 | linuxfool | Nothing in particular. I'm just interested in asterisk and like hearing about peoples' success stories with it |
| 00:51.55 | linuxfool | How much money do you estimate you saved by going the asterisk route? |
| 00:52.47 | TazzNZ | linuxfool: for the install we have going now around $300k/year |
| 00:53.40 | TazzNZ | but it does depend on how much your current implementation is |
| 00:53.51 | linuxfool | TazzNZ: are those savings mainly because of the support costs you avoid? |
| 00:54.13 | TazzNZ | if you have an implementation that is in-house, with little support/software fees then you will save less |
| 00:54.21 | TazzNZ | basically, yes |
| 00:54.50 | TazzNZ | there was some software saving as well - I can't recall how much of that $300k was software |
| 00:55.09 | linuxfool | What desk phones did you settle on? Or did you reuse existing hardware? |
| 00:56.45 | TazzNZ | new phones - they were due for an upgrade - we are using snom hardware |
| 00:58.27 | linuxfool | how do the users like them? |
| 01:04.22 | TazzNZ | they love them |
| 01:04.43 | TazzNZ | I have the feeling you are doing research for your own implementation ? |
| 01:05.21 | linuxfool | In the distant future, maybe. |
| 01:05.48 | linuxfool | I'd like to start a non-profit after school and am looking for any opportunities to use open source software to save on expenses. |
| 01:07.22 | TazzNZ | softphones (and I hate them for normal users) and asterisk will save a bunch of money |
| 01:07.45 | TazzNZ | even yealink (a cheap brand of hardware phone) can be $80+ per handset |
| 01:09.49 | linuxfool | I could never get used to using a softphone only. And even then you need a decent quality headset to make it worth it, right? |
| 01:10.11 | linuxfool | Used Cisco handsets can be bought pretty cheaply. |
| 01:17.08 | TazzNZ | correct - I dislike softphones too |
| 01:17.16 | TazzNZ | but yeah - second hand will work |
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| 01:31.23 | linuxfool | TazzNZ: Were you compensated for helping to implement a program that saves $300k a year? |
| 01:42.21 | TazzNZ | linuxfool: yes, this is my day job |
| 01:44.14 | linuxfool | I hope the higher-ups were happy about all the money you saved them. Did you propose the idea of using asterisk? |
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| 01:56.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:57.51 | TazzNZ | linuxfool: it took 8 months to convince them to use asterisk |
| 02:07.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:08.42 | linuxfool | TazzNZ: Why do you think it took so long? Were they afraid of change? |
| 02:09.48 | *** join/#asterisk asfp (~user@2a00:7143:102:101:d267:e5ff:feec:d3f8) |
| 02:12.00 | asfp | if I want to be able to break into an ongoing call with a message ("Fire alarm! please exit the building"), what is this called, and can someone point me in the general direction of this feature? |
| 02:14.05 | TazzNZ | linuxfool: not trusting a free product |
| 02:16.50 | TazzNZ | asfp: look into agent coaching |
| 02:16.59 | TazzNZ | it's the whisper function |
| 02:17.05 | TazzNZ | or you will need to look into the AMI |
| 02:19.43 | asfp | TazzNZ: Can I whisper to both ends of a call at the same time A gets "Fire alarm!" while B gets "The person you are talking to needs to go for a smoke!"? |
| 02:21.20 | TazzNZ | http://stackoverflow.com/questions/8099018/asterisk-how-to-whisper-music-using-chanspy-or-any-alternative <-- asfp |
| 02:21.41 | asfp | TazzNZ: heh, I was just reading it.. thanks! |
| 02:21.44 | TazzNZ | iirc, changing the w to W will whisper in the other direction |
| 02:22.21 | TazzNZ | oh no - you will have to find both channels |
| 02:22.28 | TazzNZ | and spie that way |
| 02:23.20 | TazzNZ | s/spie/spy |
| 02:23.40 | asfp | TazzNZ: thanks. I am mostly looking at the feasability and complexity of it, and it seems fairly easy. |
| 02:24.19 | TazzNZ | asfp: yeah - good idea btw |
| 02:26.49 | linuxfool | Aside from asterisk and linux dstros in general, what are some other examples of open source software out there that is almost or just as capable as a very expensive for-profit product? |
| 02:27.34 | TazzNZ | linuxfool: Apache vs IIS |
| 02:28.08 | TazzNZ | but really, Linux as a whole, both Apple and Sun/oracle have paid versions of it (in a unix format) |
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| 03:03.59 | gkwhc | anyone know where i can find tutorials on creating new extensions? for ex, dial 999 to play some kind of audio file |
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| 04:52.47 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 04:52.52 | hrolf | Hi #asterisk |
| 04:53.25 | hrolf | What does the "8" in this line mean? |
| 04:53.26 | hrolf | [Oct 18 22:40:04] VERBOSE[28207] pbx.c: -- Executing [2135360438@call_manager:8] AGI("DAHDI/12-1", "agi://10.1.100.100:4574/callmanager,1413653924.11257,2014-10-18 22:38:44,I2Q-mblrendevous_MainMenu-MeezanRendevous,None,Inbound") in new stack |
| 04:53.53 | hrolf | This thing "Executin [2135360438@call_manager:8]" <----- |
| 04:55.50 | Penguin | priority number 8 of extension 2135360438 |
| 04:56.19 | hrolf | Penguin: Does this mean that this command is written in dialplan?? |
| 04:56.24 | Penguin | yes |
| 04:56.32 | hrolf | Penguin: But there is none. |
| 04:56.58 | Penguin | On your asterisk cli, run this: dialplan show 2135360438@call_manager |
| 04:56.59 | hrolf | Problem is my last command is this same AGI command. |
| 04:57.08 | hrolf | it has priority 7 |
| 04:57.20 | hrolf | then I again I get this executed once more. |
| 04:57.23 | hrolf | Okay let me try. |
| 04:58.39 | hrolf | Got it. |
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| 04:58.45 | hrolf | Maybe. |
| 05:00.14 | hrolf | Penguin: Can you take a look at this http://pastie.org/9662243 |
| 05:00.27 | hrolf | Penguin: When I executed the command I got two extensions. |
| 05:00.47 | Penguin | yep |
| 05:00.55 | hrolf | Penguin: one 2135360438 |
| 05:00.56 | Penguin | Two extensions match. |
| 05:01.01 | hrolf | Second _X!5 |
| 05:01.12 | hrolf | Does this mean that is the problem? |
| 05:01.20 | hrolf | That it first matches 2135360438 |
| 05:01.27 | hrolf | then _X!5 ? |
| 05:01.55 | Penguin | Both of them match. Priority 8 of the second is executed after 1-7 of the more-specific match. |
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| 05:02.20 | hrolf | Penguin: I see. |
| 05:02.24 | Penguin | This is correct behavior. |
| 05:05.29 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 05:08.33 | hrolf | Penguin: How do I modify it so that it doesn't not jump to the second extension. |
| 05:09.12 | Penguin | Add priority 8 to the extension 2135360438. |
| 05:09.30 | Penguin | a NoOp() will be fine. |
| 05:09.51 | hrolf | Penguin: I see, then it won't jump right? |
| 05:09.57 | Penguin | right |
| 05:10.36 | hrolf | Penguin: Second I have more extensions similar to 2135360438 (with same commands as this one) I would have to another priority 8 to them too. right? |
| 05:10.43 | hrolf | in same context |
| 05:11.13 | Penguin | If you have a less-specific match with more priorities, it will just like this one did. |
| 05:11.29 | hrolf | Penguin: Thanks. |
| 05:11.34 | Penguin | s/just/jump/ |
| 05:12.03 | hrolf | Penguin: Right, got it. Thanks. |
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| 05:19.00 | hrolf | Penguin: This would go if both extensions are of same size? That is the I make the second one 7 instead of 8 right? (cause there is one extra/useless command in there) |
| 05:19.17 | Penguin | Yes. |
| 05:19.34 | Penguin | I saw that Set(_Callasdfasdf=Inbound) in there. |
| 05:19.38 | hrolf | Penguin: Yes. |
| 05:19.51 | hrolf | Penguin: It was just garbage that went in. Shouldn't be there. |
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| 05:22.24 | hrolf | Penguin: Thanks. |
| 05:22.33 | Penguin | No problem. |
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| 06:33.29 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 06:33.45 | Milos | I'm trying to use short codes like *1, *2, etc. Some work fine and others don't, with the same configuration. |
| 06:33.48 | Milos | Numbers, that is. |
| 06:34.01 | Milos | For exmaple *1, *2, *3 work but *4 doesn't, *9 doesn't, etc. |
| 06:34.12 | Milos | How can I override whatever these are normally used for in Asterisk so that they do work. |
| 06:34.24 | Milos | Or, is this something to do with my SPA 504G? |
| 06:34.43 | Milos | ;exten => _*1,1,GoToIf( ... |
| 06:34.55 | Milos | this line works for 1, 2, 3 etc but not others |
| 06:35.17 | Milos | the only thing that shows up in -rvvv logs is "Using SIP RTP TOS bits" |
| 06:35.21 | Milos | no actual asterisk messages |
| 06:35.34 | Milos | does this mean my 504G is overriding the shortcodes or what? |
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| 06:38.01 | kaldemar | Milos: if you don't see anything in sip debug, then it does not send anything to asterisk. |
| 06:38.35 | Milos | I see "Using SIP RTP TOS bits 184, Using SIP RTP CoS mark 5" each time I dial one of the ones that doesn't work |
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| 06:41.36 | kaldemar | Milos: have you enabled sip debug? |
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| 06:57.26 | Milos | I can enable it |
| 06:57.34 | Milos | kaldemar, when I dial *8 my phone says "Forbidden" |
| 06:57.47 | Milos | versus dialling other numbers that aren't configured, where it says InvalidNumber |
| 07:00.29 | Milos | features.conf:;pickupexten = *8 ; Configure the pickup extension. (default is *8) |
| 07:00.31 | Milos | this could be why |
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| 07:15.57 | fling | Hello. |
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| 08:01.40 | michael_work | WIMPy, downgraded to 11.6-cert and now after 12 hours 500MB of memory in 12 hours vs old 11.11 1.5GB in 16 hours |
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| 08:27.15 | Zetten | Using pjsip, is there any way to set a placeholder 'Anonymous' contact name for incoming calls using withheld numbers? |
| 08:27.55 | Zetten | Right now we get a dynamic uuid and there's a concern that some users will be confused if this appears on their phone |
| 08:28.50 | Zetten | This is with pjproject 2.3 and asterisk 12.6.0 |
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| 08:45.43 | WIMPy | michael_work: Anything special you're doing? |
| 08:46.47 | michael_work | a lot of recordings, sip devices saved in database and a lot of AGIs. |
| 08:46.53 | michael_work | no idea if there si something special |
| 08:47.02 | michael_work | what version you use of 11? |
| 08:47.26 | WIMPy | Mostly 11.13 |
| 08:47.38 | michael_work | have you had 11.11? |
| 08:47.50 | WIMPy | Yes. |
| 08:47.56 | michael_work | worked fine? |
| 08:47.57 | WIMPy | At least here. |
| 08:48.04 | michael_work | heavy usage? |
| 08:48.14 | WIMPy | Apart from transfers, yes. |
| 08:48.26 | WIMPy | No. Not much going on here. |
| 08:48.36 | WIMPy | Many minutes, but not many calls. |
| 08:48.48 | michael_work | i had other wierd problem with 11.11 i had both machines 11.11.0 and on one there was problem with ICE (webrtc) on other it worked perfectly. updating to 11.13 solved the issues |
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| 08:49.07 | WIMPy | has ICE disabled. |
| 08:49.25 | michael_work | those were devel;opment machines with webrtc |
| 08:49.36 | WIMPy | Or acutally many machines don't even have SIP enabled. |
| 08:50.01 | michael_work | iax2/. |
| 08:50.21 | WIMPy | Enabled everywhere, but not neccessarily in use. |
| 08:50.30 | WIMPy | Other than that I have the link status. |
| 08:53.17 | michael_work | :\ |
| 09:13.54 | michael_work | WIMPy, what are the timer you use |
| 09:14.22 | WIMPy | timerfd |
| 09:14.30 | michael_work | what kernel? |
| 09:14.34 | TrevorH | I asked a couple of times last week about how to disable sslv3 connections and having read the source, I've come to the conclusion that it's just not possible at the moment |
| 09:14.39 | WIMPy | Except for one that has dahdi |
| 09:14.49 | WIMPy | Various 3.x |
| 09:15.02 | michael_work | <PROTECTED> |
| 09:15.32 | WIMPy | My preferred one ATM is 3.15.10 |
| 09:15.44 | michael_work | that machine is using timerfd right now |
| 09:16.08 | michael_work | 3.15? what OS? |
| 09:16.29 | WIMPy | Linux :-) |
| 09:16.34 | michael_work | i think i have older even on my laptop |
| 09:16.39 | michael_work | i mean what distro |
| 09:16.50 | michael_work | though i have 3.17 on desktop :P |
| 09:16.59 | WIMPy | I am cured of distro kernels. Never use anything else than the ones from kernel.org. |
| 09:17.42 | michael_work | on desktop i use vanilla + few gentoo patches (gentoo-sources) but at servers default of debian |
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| 10:47.52 | Rico | hi there |
| 10:47.57 | Rico | I need help please |
| 10:48.12 | Rico | I have an 1.8 asterisk behind nat |
| 10:48.19 | Rico | sip exchanges look correct |
| 10:48.27 | Rico | except c= header in SDP |
| 10:48.36 | Rico | so I don't have voice |
| 10:48.42 | Rico | I've set externIP |
| 10:48.46 | Rico | what did I forget ? |
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| 10:57.00 | seik0 | Hi everydoby. I have a question: res_odbc.conf have option share_connection = yes (btw, by default it is yes). Comment says that it allows asterisk to share single connection for simultaneouse queries. Also, comment says, that Sybase and MS SQL won't work. We are using asterisk and I wonder if it is capable of acting in "share" mode for connection |
| 10:57.09 | seik0 | Google hardly helps on search |
| 10:57.27 | seik0 | heh |
| 10:57.32 | seik0 | *we are using Oracle |
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| 10:59.35 | seik0 | In fact it looks, that Oracle does not provide "share_connection" possibility, because when under load queries go slower (they go in queue, one-by-one, in fact) |
| 11:00.41 | seik0 | Oracle on itself provides "Shared Server Mode" - the thing that will be in google if you will try to google the issue |
| 11:01.20 | seik0 | but shared mode is not about "share_connection" in res_odbc.conf |
| 11:01.34 | seik0 | (if someone interesting, of course) |
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| 11:08.05 | seik0 | so it goes |
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| 11:41.11 | Zetten | Re-request: Using pjsip, is there any way to set a placeholder 'Anonymous' contact name for incoming calls using withheld numbers? |
| 11:42.52 | Zetten | Would the module 'res_pjsip_endpoint_identifier_anonymous' be any use? I'm not 100% sure if it's doing what I think it it's doing, or I could have misconfigured it |
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| 12:40.56 | Geek-Linux | Hi All: if i am using dial() application to connect two persons on the calls. can i play any sound during the call ? |
| 12:46.32 | [TK]D-Fender | Geek-Linux: Not via anything in the dialplan in that call. You'd have to hook in through and external process |
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| 13:04.51 | michael_work | WIMPy, false alarm, the problem with that server is hdd (though still there is less memory used compared from prev version so it might be both) |
| 13:20.01 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
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| 13:35.23 | *** join/#asterisk Muhammad (5896ae8c@gateway/web/freenode/ip.88.150.174.140) |
| 13:35.33 | Muhammad | hi |
| 13:36.03 | Muhammad | I want to run radio over IP ( live streaming in asterisk) |
| 13:36.23 | Muhammad | Is any one has experience of it? |
| 13:43.13 | orn | i did it once... worked okay while the stream was alive, but i had problems when it died... i think i had to restart something to get the stream going again |
| 13:43.22 | orn | but it was a long time ago so i don't remember any details |
| 13:47.22 | *** join/#asterisk BurmaSauce (~evansj@217.138.19.122) |
| 13:47.45 | Muhammad | orn: any idea |
| 13:49.32 | Muhammad | my link is mta.tv/live |
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| 14:46.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:17.37 | ChannelZ | So with pjsip how do we show information about an active channel? Like what codec it's using, who it's connected to, etc. 'pjsip show channel xxxx' just seems to show the same couple of lines that 'pjsip show channels' does but for the one channel. |
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| 15:26.22 | o_be_one | hi all :)) just to know can i add a beep or something to warn when i forward a call ? Cause i can tell "hey ive someone for you" but when its forwarden there is no transition sound ... |
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| 15:31.35 | tenspeed705 | Morning guys. I want to log SIP registrations to mysql. Anyone know if any easy way to do this exists? |
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| 15:45.20 | rmudgett | ChannelZ: use "core show channel" |
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| 15:56.43 | monsterco | I have few polycom phones and all users want to be able to pickup each other's calls - how can I accomplish that? |
| 15:58.06 | Chainsaw | monsterco: With the pickup feature in Asterisk. You need to ensure that the relevant phones are in the same "pickupgroup" |
| 15:58.46 | monsterco | how does pickup work? It's like blf? |
| 15:59.21 | Chainsaw | monsterco: It's a key combination like "*8" |
| 15:59.33 | Chainsaw | monsterco: I don't believe you can do useful hints for it. If you can, I must admit that I never got it working correctly. |
| 16:01.22 | monsterco | got it - thanks |
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| 16:48.04 | marceloamorim | guys, when I receive on sip debug the CSeq: 3 BYE, it means the end of call right? The problem is, I`m receiving this bye when I use the transfer on the called side |
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| 16:48.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:49.10 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: Asterisk 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 16:49.17 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gtadsjwbvhamzicd) |
| 16:49.26 | *** join/#asterisk areski (~areski@80.174.128.40.dyn.user.ono.com) |
| 16:57.32 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 17:00.30 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:01.04 | dym | Hey, im trying to establish a peer connection from my home asterisk server to an online asterisk server that holds a PSTN sip trunk. This is my configuration: http://pastebin.com/fFUVecvi Why cant the online asterisk see/contact the local asterisk? |
| 17:02.37 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 17:11.40 | [TK]D-Fender | dym: register => openroot-PASSWORD@phone.openroot.de/openroot-up <--- this is not correctly formatted, nor is it using the matching username |
| 17:16.27 | dym | sorry, it's actually register => openroot-up:PASSWORD@phone.openroot.de/openroot-up |
| 17:16.51 | dym | which obviously results in a successful peer register on phone.openroot.de |
| 17:16.58 | dym | (see sip show peers) |
| 17:18.46 | Penguin | Your home asterisk uses a dynamic public IP addr? |
| 17:20.01 | dym | affirmative. |
| 17:20.14 | dym | which is why i want it to register to the online asterisk, which has a static ip |
| 17:20.36 | dym | and from there "bridge" calls to PSTN |
| 17:21.35 | Penguin | You can use this for reference. http://pastebin.com/Ag7tknm2 |
| 17:21.50 | Penguin | It's for two static addrs, but still relevant. |
| 17:21.51 | [TK]D-Fender | [13:16]dymwhich obviously results in a successful peer register on phone.openroot.de <- no, it doesn't |
| 17:22.11 | [TK]D-Fender | openroot-up (Unspecified) D N 0 UNKNOWN <---------------- |
| 17:22.17 | [TK]D-Fender | Clearly not registered |
| 17:22.40 | dym | [TK]D-Fender: but why does the local asterisk say "OK"? |
| 17:22.52 | Penguin | The qualify for the other peer is OK. |
| 17:23.25 | [TK]D-Fender | dym: because it is using "qualify" to send OPTIONS packets to the other system, and the other system is responding to that request |
| 17:23.29 | Penguin | Home asterisk sees the online asterisk because you specified the host addr. |
| 17:23.37 | [TK]D-Fender | dym: that proves nothing about being "registered". |
| 17:23.43 | Penguin | The qualify was good, so it reports OK. |
| 17:23.59 | [TK]D-Fender | dym: Time to go look at real debug |
| 17:24.03 | Penguin | If you wanted to see if you were registered, you'd do sip show registry. |
| 17:24.34 | dym | okay, i changed settings according to Penguin's example. Where do i get the register debug? |
| 17:24.38 | dym | (core set debug is 9) |
| 17:24.53 | Penguin | I've given an example of two static peers, though. |
| 17:25.03 | dym | Penguin: i only applied some useful things |
| 17:25.04 | dym | :) |
| 17:25.11 | Penguin | Give me a minute and I'll change it to reflect one side static and the other registers. |
| 17:25.17 | dym | Penguin: thanks! |
| 17:25.35 | [TK]D-Fender | dym: core debug is mostly useless. "core set verbose 10", "sip set debug on" |
| 17:28.33 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:30.12 | Penguin | modified for one stattic, one dynamic http://pastebin.com/0Xgwatc4 |
| 17:35.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:37.28 | dym | Okay |
| 17:37.30 | dym | well |
| 17:37.38 | dym | Penguin: state on both is now "unknown" |
| 17:38.14 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 17:40.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:40.57 | dym | Penguin: http://pastebin.com/2Uwmc236 |
| 17:41.06 | dym | ended up at "unmonitored" for both |
| 17:41.07 | Penguin | In my example, Tampa has a static address and Miami registers to Tampa. I also did not enable qualify on either, but I probably should. |
| 17:41.36 | Penguin | qualify will make it monitored as opposed to unmonitored. |
| 17:41.48 | Penguin | Your register statement is still wrong. |
| 17:42.05 | Penguin | register => username:password@host |
| 17:42.15 | Penguin | colon between valid username and the password |
| 17:43.56 | dym | Penguin: sorry, the os-x autocorrection seems to make it that way |
| 17:43.58 | dym | <PROTECTED> |
| 17:43.59 | Penguin | Now your usernames are all jacked up. |
| 17:44.29 | Penguin | Pick an arbitrary name for one system and another arbitrary name for the other system. |
| 17:44.39 | dym | do i require different usernames? I thought one system registers at the others |
| 17:45.01 | Penguin | One system DOES register to the other, and the username must be correct for that to work. |
| 17:45.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:45.49 | Penguin | In my examples, the system in tampa has a peer configured by the name of miami. |
| 17:46.03 | Penguin | The system in miami has a peer configured on it for tampa. |
| 17:46.14 | dym | right. changed. |
| 17:46.15 | Penguin | The configured peer is for the other side. |
| 17:46.44 | Penguin | While the names are arbitrary, it's a good idea to have them make sense so you don't get lost of confused. |
| 17:46.58 | dym | ah! |
| 17:47.05 | dym | qualify=yes locally resulted in : [Oct 20 19:46:49] NOTICE[1189]: chan_sip.c:23554 handle_response_peerpoke: Peer 'openroot-online' is now Reachable. (39ms / 2000ms) |
| 17:47.25 | Penguin | So one is called openroot-online, and the other is called openroot-home? |
| 17:47.42 | dym | affirmative |
| 17:47.49 | Penguin | I'll adjust your paste. |
| 17:47.50 | dym | now openroot-home can "see" openroot-online |
| 17:47.54 | dym | thanks |
| 17:50.44 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 17:51.18 | dym | Thing i dont understand is: How does the reverse connection work. When openroot-home uses the register statement to register at openroot-online, how does it work vice-versa? |
| 17:52.10 | ipengineer | When using announcements in a queue, once the announcement plays the hold music starts over from the beginning. Is there a way to make this resume playing where it left off? |
| 17:53.31 | Penguin | http://pastebin.com/cwQGrj5C |
| 17:53.40 | dym | checks |
| 17:53.47 | Penguin | dym: Registration is to tell the other side how to reach you. |
| 17:54.08 | Penguin | So if your home system is dynamic, it needs to be able to tell the online system where it is when it moves. |
| 17:54.27 | Penguin | It'll send a registration to the online system as the address in the register statement. |
| 17:54.36 | Penguin | s/as/at/ |
| 17:56.09 | dym | Penguin: local asterisk now gives: openroot-online/openroot- 78.47.108.244 Auto (No) No 5060 Unmonitored |
| 17:56.14 | dym | online states unknown |
| 17:56.17 | Penguin | no qualify |
| 17:56.23 | dym | required on both? |
| 17:56.44 | Penguin | If you want them to check each other to see if they exist, you'd need it on both sides. |
| 17:56.50 | Penguin | This is only for seeing if they exist. |
| 17:57.03 | Penguin | You don't HAVE to qualify your hosts. |
| 17:57.19 | dym | okay and now i can use this peer connection seamlessly for calls? ill try. |
| 17:58.03 | Penguin | Calling is another part of configuration. You'll have to have extensions set up to call via the other asterisk. |
| 17:58.11 | dym | yupp i know |
| 17:58.13 | dym | done already |
| 17:58.15 | dym | one second |
| 17:58.43 | dym | okay, as expected |
| 17:58.44 | dym | [Oct 20 19:58:29] WARNING[11175][C-00007ec7]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) |
| 17:58.51 | Penguin | And you'll reference them as SIP/openroot-home/${EXTEN} and SIP/openroot-online/${EXTEN} |
| 17:59.01 | dym | i have a public number on openroot-online |
| 17:59.24 | [TK]D-Fender | [13:58]dym[Oct 20 19:58:29] WARNING[11175][C-00007ec7]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) <- hasn't registered |
| 17:59.26 | dym | and threw it into a context that calls SIP/openroot-home/1 for testing - i should see the call on openroot-home's CLI if it goes through |
| 17:59.27 | Penguin | Does the home system show that it is registered? sip show registry |
| 17:59.30 | dym | neg |
| 17:59.38 | Penguin | sip reload ? |
| 17:59.40 | [TK]D-Fender | Look at the actual debug |
| 17:59.45 | [TK]D-Fender | this is not a guessing game |
| 18:00.08 | Penguin | If the one is not registered, the other cannot send any calls to it. |
| 18:01.11 | *** part/#asterisk voxter (~voxter@irc.voxter.net) |
| 18:03.25 | dym | okay local sip debug gives 404 not found |
| 18:03.47 | Penguin | Maybe the host address was mistyped? |
| 18:03.49 | dym | http://pastebin.com/qnrjAhnz |
| 18:04.19 | [TK]D-Fender | dym: that is OPTIONS... not a registration eattempt |
| 18:04.30 | [TK]D-Fender | make it register again |
| 18:04.31 | dym | then no such attempt occoured. |
| 18:04.31 | Penguin | That's a... yeah, what he said. |
| 18:04.35 | Penguin | sip reload ? |
| 18:04.41 | [TK]D-Fender | that should do it |
| 18:04.49 | *** join/#asterisk voxter (~voxter@irc.voxter.net) |
| 18:04.51 | Penguin | You've got to remember to sip reload after making the change. |
| 18:04.55 | dym | did |
| 18:05.02 | dym | please see the paste |
| 18:05.04 | dym | top lines |
| 18:05.53 | dym | it's not trying to register, although the register line is present. |
| 18:05.53 | Penguin | I can't tell which system that is, so just sip reload them both. |
| 18:06.06 | Penguin | Did you put the register line in the general section of sip.conf? |
| 18:06.12 | Penguin | If it's not in general, it won't work. |
| 18:06.24 | [TK]D-Fender | dym: pastebin the actual configs masking only the password. No re-ordering, including the full [general] section, etc |
| 18:07.10 | [TK]D-Fender | And then also do "sip show registry on that system that is expected to register |
| 18:08.08 | TazzNZ | morning all - "silly" question - can you force Asterisk to make an outbound T38 call without the re-invite ? (My google fuu failed me :) ) |
| 18:09.03 | dym | okay |
| 18:09.07 | dym | i cleaned the local config |
| 18:09.15 | dym | (a lot of default stuff left in there) |
| 18:09.22 | dym | it's now registered |
| 18:09.26 | Penguin | Doesn't the call start with a regular voice codec such as ulaw/alaw, and then it has to reinvite to change to T.38? I don't use t38, so I'm just thinking aloud. |
| 18:09.50 | *** join/#asterisk riess82 (~riessma@188-22-124-139.adsl.highway.telekom.at) |
| 18:10.00 | TazzNZ | Penguin: correct - but I have an ITSP that thinks otherwise :( |
| 18:10.29 | TazzNZ | they are insisting that the invite should contain T38 |
| 18:10.35 | TazzNZ | as in the first invite |
| 18:10.45 | Penguin | I'd be asking them how to make that happen. |
| 18:11.00 | TazzNZ | they don't "support" asterisk |
| 18:11.05 | Penguin | Clearly! |
| 18:11.10 | TazzNZ | but I just wanted to triple check |
| 18:11.28 | dym | Penguin [TK]D-Fender: i have a ringing phone. Thanks! |
| 18:11.34 | Penguin | Nice. |
| 18:12.22 | *** part/#asterisk riess82 (~riessma@188-22-124-139.adsl.highway.telekom.at) |
| 18:12.52 | Penguin | I guess I should decide on what to have for lunch. Ham or chicken? |
| 18:13.00 | TazzNZ | CHICKEN !!!! |
| 18:13.03 | TazzNZ | :D |
| 18:13.31 | Penguin | I think you're probably right. |
| 18:14.55 | Penguin | chicken and cheese sandwich, garden salsa flavored Sun Chips |
| 18:17.12 | dym | Penguin: I have another one, if you're up for it. |
| 18:17.31 | Penguin | Sure. |
| 18:17.57 | dym | I have a phone registered to openroot-home. when it calls a any number i throw it into [openroot-up] context with "exten => _X.,1,Dial(SIP/openroot-online/${EXTEN},WwXx)" |
| 18:18.17 | dym | but on openroot-online i get: "[Oct 20 20:18:05] NOTICE[2437][C-00007ed5]: chan_sip.c:25300 handle_request_invite: Failed to authenticate device "Geschke" <sip:homefone@10.1.55.95>;tag=as329c06b6" |
| 18:20.07 | [TK]D-Fender | dym: You have settings on one side that you didn't copy to the other |
| 18:20.12 | [TK]D-Fender | dym: And should have... |
| 18:20.32 | dym | i used the (mostly) unmodified example. |
| 18:20.41 | [TK]D-Fender | dym: Look at qwhat you PB'd for us |
| 18:20.42 | dym | except from the qualify statement |
| 18:20.50 | [TK]D-Fender | you have a few settings taht you didn't make match |
| 18:21.07 | [TK]D-Fender | (many) |
| 18:21.23 | *** join/#asterisk nny (~Scott@174.107.201.51) |
| 18:21.52 | dym | in the peer contexts? |
| 18:22.03 | Penguin | The phone registers to the home system and you want to send the call over to the online system, yes? |
| 18:22.17 | nny | working on option 66/160 for polycoms. I have put the value "192.168.1.77" for both options in the router. The info I found says to put "tftp://192.168.1.77" under option 160. Any advice? |
| 18:22.39 | nny | If i try tftp:// the router says no bueno |
| 18:23.45 | [TK]D-Fender | just put the IP |
| 18:23.54 | [TK]D-Fender | your phone should be told the protocol. |
| 18:23.59 | [TK]D-Fender | in the bootrom |
| 18:24.00 | nny | yeah. Phone is saying no boot server. |
| 18:24.11 | [TK]D-Fender | [14:23][TK]D-Fenderyour phone should be told the protocol. <--- |
| 18:24.13 | nny | have tcpdump running on server looking for packets |
| 18:24.21 | [TK]D-Fender | hint: It is NOT tftp "out of the box" |
| 18:24.21 | nny | Yeah I am using just the IP |
| 18:25.15 | nny | So if I have the options set in the router and the phone is saying no boot server, is it not being told the protocol, IP or should I look somewhere else? |
| 18:25.55 | nny | Oh... |
| 18:25.57 | [TK]D-Fender | [14:23][TK]D-Fenderyour phone should be told the protocol. [14:23][TK]D-Fenderin the bootrom <--------------- |
| 18:26.00 | nny | yeah |
| 18:26.02 | nny | re-read it |
| 18:26.11 | nny | so I have to specify the server somewhere on boot regardless |
| 18:26.17 | nny | well in the setup |
| 18:26.33 | nny | hmmph. May as well go back to ftp as it was before through the bootrom |
| 18:26.46 | [TK]D-Fender | nny: You cans et the server explicityl... but you must specify the PROTOCOL regardless |
| 18:27.48 | nny | [TK]D-Fender: ok thanks |
| 18:27.54 | dym | Penguin: yes |
| 18:28.31 | Penguin | dym: Show me the peer entry for the phone. |
| 18:28.41 | [TK]D-Fender | Penguin: Not the phone |
| 18:29.02 | [TK]D-Fender | Penguin: it's matching CID for the username because he didn't use the appropriate settings on BOTH peers |
| 18:29.19 | dym | [TK]D-Fender: i cant find it |
| 18:29.32 | Penguin | hmm |
| 18:29.34 | [TK]D-Fender | dym: your 1st peer is TWICE the size of the other one. |
| 18:29.43 | [TK]D-Fender | dym: So what do you mean "i can't find it"? |
| 18:30.58 | [TK]D-Fender | Ah, I see variences between Penguin's sample and yours |
| 18:31.07 | Penguin | Okay, I see the online system is the one that complained about the call. |
| 18:31.24 | [TK]D-Fender | dym: You should have "fromuser=openroot-up" on BOTH sides |
| 18:31.25 | Penguin | It only sees the home asterisk. |
| 18:32.01 | [TK]D-Fender | [14:18]dymbut on openroot-online i get: "[Oct 20 20:18:05] NOTICE[2437][C-00007ed5]: chan_sip.c:25300 handle_request_invite: Failed to authenticate device "Geschke" <sip:homefone@10.1.55.95>;tag=as329c06b6" <- which is why you see the calling device's name there |
| 18:32.36 | Penguin | I don't think I used fromuser on either side. |
| 18:32.50 | dym | which is why i did not |
| 18:32.58 | [TK]D-Fender | Penguin: Saw it on one PB somewhere there IIRC. A lot of scrolling to do now... |
| 18:33.18 | dym | one second |
| 18:33.20 | dym | ill re-paste both |
| 18:33.27 | Penguin | Oh, it was probably on one of his that I edited for username configuration. |
| 18:33.33 | Penguin | I didn't touch the other parts he pasted. |
| 18:34.11 | [TK]D-Fender | dym: I scrolled to the last you showed. just add what I have given you |
| 18:34.14 | dym | http://pastebin.com/nkRvzncD |
| 18:34.15 | dym | there. |
| 18:34.25 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 18:34.34 | dym | so fromuser=openroot-up on both? that is just the context's name |
| 18:34.45 | Penguin | That'll break it for sure. |
| 18:35.06 | [TK]D-Fender | dym: that's the USER name |
| 18:35.10 | dym | negative |
| 18:35.17 | dym | the user names differ |
| 18:35.26 | [TK]D-Fender | .... |
| 18:35.28 | dym | home has openroot-home and online has openroot-online |
| 18:35.51 | [TK]D-Fender | they should not be different |
| 18:36.08 | dym | okay - let's row back. fromuser does not have to match defaultuser? |
| 18:36.09 | Penguin | oh boy |
| 18:36.22 | [TK]D-Fender | FRED on A should be the same as FRED on B |
| 18:36.39 | Penguin | That's totally not important. |
| 18:36.45 | dym | :D |
| 18:36.51 | dym | im confused. |
| 18:37.01 | Penguin | The peers are configured for the username of the OTHER SIDE. |
| 18:37.49 | dym | affirmative. |
| 18:37.55 | dym | which is the current state |
| 18:38.00 | dym | (aka defaultuser) |
| 18:38.32 | *** join/#asterisk jwr__ (~quassel@205.196.167.54) |
| 18:38.37 | nny | doing some research but maybe someone here can point me in the right direction. I am setting up dual wans for external registration, using phone.domain.com and phone2.domain.com. How do I specify this in sip.conf (i.e. extenrip)? |
| 18:38.47 | nny | externip* |
| 18:38.59 | dym | Penguin: they actually are not. they are configured according to your comments |
| 18:39.01 | Penguin | The peer I use to put calls to my ITSP has its own arbitrary name; I don't use the same name THEY have for me on THEIR side. |
| 18:39.17 | [TK]D-Fender | nny: You can't |
| 18:39.35 | Penguin | dym: In my examples, each peer is configured with the info for the other side. |
| 18:39.54 | Penguin | dym: SysA has a peer for SysB. SysB has a peer for SysA. |
| 18:40.01 | nny | [TK]D-Fender: should I just setup external round robin then? |
| 18:40.25 | [TK]D-Fender | nny: * will advertise just 1 IP... |
| 18:40.47 | nny | [TK]D-Fender: what if one WAN is on a public interface? |
| 18:40.49 | nny | er |
| 18:40.56 | *** join/#asterisk doop (~doop@2602:ffea:a::6f2f:d8d8) |
| 18:41.01 | nny | what if one WAN is connected directly to the asterisk box? |
| 18:41.14 | [TK]D-Fender | nny: "externip" is ONE value. |
| 18:41.32 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:41.34 | Penguin | deprecated. should be externaddr |
| 18:42.32 | nny | [TK]D-Fender: what is the suggested method for failover? |
| 18:42.45 | [TK]D-Fender | Same net resutl when using version-appropriate parameters |
| 18:43.06 | [TK]D-Fender | nny: probably a good SIP proxy in front |
| 18:43.28 | nny | [TK]D-Fender: what if i use a domain name under externip and have the failover in DNS? |
| 18:43.33 | nny | er externaddr |
| 18:43.53 | Penguin | externhost |
| 18:44.41 | nny | What about this method http://ericmason.net/2011/11/asterisk-automatically-update-externip-when-your-ip-address-changes/ |
| 18:45.26 | nny | or http://lists.digium.com/pipermail/asterisk-users/2012-February/270057.html |
| 18:45.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 18:45.45 | nny | I assume both will work, just looking for advice on something smart |
| 18:46.26 | nny | externhost and a dyanmic dns service may suffice |
| 18:47.15 | Penguin | Using externhost with a dynamic dns and an appropriate externrefresh value should work just fine. |
| 18:47.34 | *** part/#asterisk voxter (~voxter@irc.voxter.net) |
| 18:48.12 | nny | Penguin: thanks |
| 18:48.56 | Penguin | I'm curious why Eric Mason didn't just do that instead. |
| 18:51.25 | Penguin | By the way, in 1.8 externrefresh defaults to 10s. That value works well for me with my dynamic dns which checks/updates each five minutes. |
| 19:12.37 | *** join/#asterisk infobot (ibot@rikers.org) |
| 19:12.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: Asterisk 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 19:12.46 | [TK]D-Fender | marceloamorim: I'm strongly suspecting "user error" on the actual proper procedure to do a transfer in the first place. |
| 19:13.16 | marceloamorim | I tried transfer on the called side, isn`t on that debug, but at this asterisk is where the call is ended |
| 19:14.13 | [TK]D-Fender | marceloamorim: No. |
| 19:14.19 | [TK]D-Fender | marceloamorim: [2014-10-20 14:43:11] <--- SIP read from UDP:192.168.2.100:5062 ---> [2014-10-20 14:43:11] BYE sip:032812800@192.168.3.254:5060 SIP/2.0 |
| 19:14.29 | [TK]D-Fender | marceloamorim: the "BYE" is RECEIVED, not SENT |
| 19:14.47 | [TK]D-Fender | marceloamorim: This looks like your user screwed up the transfer |
| 19:15.29 | marceloamorim | so maybe my other system is sending the wrong sequence? |
| 19:16.04 | [TK]D-Fender | marceloamorim: "system"? [2014-10-20 14:43:11] User-Agent: Intelbras Telefone IP TIP300 65.61.75.15 <- so this isn't ajust a PHONE? |
| 19:17.14 | marceloamorim | well, so maybe is the phone |
| 19:23.43 | marceloamorim | the weird think is, when the same "user-Agent" call for my number 3281 8350, this number is on E1, and when call for my another number 3281 2800 this number is on FXO card, the transfer on my side don`t drop when is on E1, just for FXO card |
| 19:24.11 | marceloamorim | but I`ll check the user-agent, ty so much [TK]D-Fender |
| 19:28.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:32.48 | Penguin | dym: Did you try the fromuser thing? |
| 19:36.04 | *** join/#asterisk nny (~Scott@cpe-174-107-201-051.sc.res.rr.com) |
| 19:38.06 | nny | Well this is odd. I had NAT to NAT working... but have updated the router and no longer works. http://pastebin.com/WG4WYXrG relevant sip.conf (In testng I cannot hear prompts when dialing asterisk box remotely, this worked up until some recent changes) |
| 19:38.18 | nny | I have the rdp/sip auth ports forwarded |
| 19:38.32 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:42.05 | nny | I can dial through the sip provider and hear my cell though |
| 19:42.08 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:42.14 | nny | (client -> Asterisk -> Sip Provider) |
| 19:47.40 | *** join/#asterisk theron (~theron@2620:10d:c0a4:1303:2acf:e9ff:fe1c:8fb) |
| 19:49.35 | *** join/#asterisk theron (~theron@199.201.65.135) |
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| 19:59.13 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-qbdjcywmnggvvzgq) |
| 20:00.22 | nny | Also just tested no nat -> nat and no luck. Seems the rtp stream can't leave the network. I'll check the router |
| 20:09.29 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-92-159.dynamic.qsc.de) |
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| 20:22.10 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 20:22.56 | nny | ok tested further I am havig issues with NAT to NAT only. This worked behind another router, so I am kind of perplexed. |
| 20:40.35 | *** join/#asterisk Pernat (~Pernat@rtr-6-92.interkar.pl) |
| 20:42.45 | *** join/#asterisk nny (~Scott@cpe-174-107-201-051.sc.res.rr.com) |
| 20:42.59 | nny | Fixed! Damn barely labeled router settings |
| 20:43.08 | nny | Sip: Standard, Compatibility mode :\ |
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| 20:49.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:59.05 | nny | anyone know of a way to disable DNS caching on polycom phones? |
| 20:59.05 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:04.46 | nny | http://pastebin.com/C7xUA02X |
| 21:05.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:05.23 | nny | trying to discern what values to change to tell the phone to always consult the DNS server for IP |
| 21:05.44 | nny | Penguin: Have you messed with any of those settings on Polycoms? |
| 21:07.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 21:10.11 | lex2 | Hey |
| 21:10.57 | lex2 | I'm having huge issues configuring Vitelity with my pbxinaflash config |
| 21:11.13 | lex2 | I'm using the web ui, I don't know what to use for outbound routes |
| 21:12.11 | tm1000 | ~freepbx @ lex2 |
| 21:12.15 | tm1000 | ~freepbx |
| 21:12.15 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:12.19 | tm1000 | lex2: ^ |
| 21:12.39 | lex2 | sure, thanks |
| 21:15.51 | zaf | anyone recommend a decent call management / phone status app for vanilla asterisk 12? |
| 21:16.38 | nny | Ok.. so looking to register a polycom remotely via dynamic DNS |
| 21:16.50 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 21:16.52 | nny | anyone who has done this or has advice on how to configure the phone to do so let me know |
| 21:17.02 | nny | Willing to paypal if needed, I hate polycoms :D |
| 21:17.27 | zaf | i just figured out polycoms myself with http-based config files |
| 21:17.29 | tm1000 | zaf: isymphony |
| 21:17.37 | tm1000 | and no it does not require freepbx |
| 21:17.39 | zaf | tm1000, no asterisk 12 support |
| 21:25.09 | nny | zaf: also FOP2 |
| 21:25.28 | nny | brb gotta run out.. anyone willing to assist with polycom stuffs when I get back I'd appreciate it |
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