| 00:12.10 | *** join/#asterisk kaushal (~kaushal@14.97.58.235) |
| 00:12.17 | kaushal | Hi WIMPy |
| 00:12.48 | kaushal | I have upgraded the libpri and the issue still occurs |
| 00:12.58 | kaushal | leifmadsen: hi |
| 00:14.43 | kaushal | when i just use 4 E1 PRI Lines for Outbound Dialling and initiating campaign for 250 Phone Numbers, I do not see any issue. The issue occurs when i use 5 and more PRI Lines |
| 00:15.14 | kaushal | I see got hangup, cause 101 |
| 00:15.35 | kaushal | in asterisk log messages and sangoma files |
| 00:17.11 | kaushal | I am stuck very badly and unable to make it live in production |
| 00:17.54 | WIMPy | Did you rebuld Asterisk after upgrading libpri? |
| 00:18.16 | kaushal | I have not rebuild it |
| 00:18.31 | kaushal | I have installed the binary only |
| 00:19.11 | WIMPy | Hmm. Doesn't libpri go into chan_dahdi? |
| 00:19.26 | WIMPy | or sig_pri or whatever. |
| 00:19.40 | kaushal | chan_dahdi |
| 00:19.52 | WIMPy | 'pri show version' |
| 00:19.57 | kaushal | Do you want me to pastebin the configs ? |
| 00:19.59 | kaushal | sure |
| 00:20.39 | kaushal | libpri version: 1.4.12 |
| 00:21.23 | WIMPy | Ok. I thought you had to rebuild Asterisk to use that. |
| 00:23.35 | WIMPy | But it would have been a good idea to contact Sangoma support. They should know how to get their stuff going. |
| 00:24.03 | kaushal | WIMPy: Also i have noticed here is that when i use all the 8 PRI ports and just use 13 channels per PRI it works perfectly fine |
| 00:24.13 | WIMPy | wonders if it would happen with 2 4-port cards as well. |
| 00:24.22 | kaushal | so 8 * 13 channels works fine |
| 00:24.25 | WIMPy | oh |
| 00:24.43 | kaushal | but when i try to use 8 * 16 Channels i get got hangup, cause 101 |
| 00:25.04 | kaushal | WIMPy: Anything to add here ? |
| 00:25.13 | WIMPy | So it's the number of active channels, not the number of ports. |
| 00:25.21 | kaushal | yes |
| 00:25.41 | kaushal | WIMPy: does it have any relation with the OS ? |
| 00:26.00 | kaushal | or Asterisk or Sangoma PRI Card or Apps or Telco ? |
| 00:26.14 | kaushal | I just see only these possibilities |
| 00:26.16 | WIMPy | Don't think so. Maybe there's an issue with the call reference numbers wrapping around? |
| 00:26.54 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 00:27.00 | kaushal | call reference numbers wrapping around? |
| 00:27.06 | kaushal | not sure i understand that |
| 00:27.22 | WIMPy | Sounds like you have a 128 call limit. |
| 00:27.39 | kaushal | i see |
| 00:27.52 | WIMPy | Maybe for some odd reason it's using 7-but call reference values. |
| 00:27.59 | WIMPy | bit |
| 00:28.12 | kaushal | How do i set it to higher number of using 240 channels |
| 00:28.30 | WIMPy | It certainly shouldn't, but I have to guess anyway. |
| 00:29.04 | WIMPy | Or it is the number of channels dahdi will address. |
| 00:29.12 | kaushal | ok |
| 00:29.51 | kaushal | WIMPy: shall i pastebin the output of *CLI> dahdi show channels ? |
| 00:29.57 | WIMPy | Unfortunately this is going to get messy with that amount of calls, but have you tried to look at pri debug to see if it actuallt tied to place calls >128? |
| 00:30.27 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 00:30.46 | kaushal | WIMPy: pri debug ? |
| 00:30.56 | kaushal | can it be made global debug ? |
| 00:31.05 | WIMPy | as in 'pri set debug ...' |
| 00:31.13 | BenC[UK] | hi guys, I am still struggling with my bloody dialplan |
| 00:31.15 | kaushal | yeah for every span right ? |
| 00:32.16 | BenC[UK] | I am originating a call from AMI, it goes into the dialplan, calls the user as its supposed to, if the call is ended but it has been dispositioned in a way where we need to call another lead.. it does it fine twice, then on the third one, it just drops the call |
| 00:32.35 | kaushal | WIMPy: can i email you the debug log level with verbosity set to 10 ? |
| 00:32.59 | kaushal | It would be explained in precise manner |
| 00:33.11 | kaushal | instead of just making a wild guess |
| 00:33.20 | WIMPy | kaushal: It would be quicker if you can upload it somewhere. |
| 00:33.24 | kaushal | sure |
| 00:33.31 | kaushal | WIMPy: Any pastebin ? |
| 00:33.36 | kaushal | upload ? |
| 00:33.48 | WIMPy | yes, or pastebin. |
| 00:34.01 | kaushal | pastebin would not be able to handle it |
| 00:34.13 | kaushal | Any free file upload service ? |
| 00:35.10 | WIMPy | Lots, but I've never used them. I just copy it to where Apache can find it :) |
| 00:36.00 | BenC[UK] | same here :) |
| 00:36.34 | WIMPy | But the good thing is that a pri debug can be filtered. If you fill the first 4 ports, the trouble will be on the 5th port only. Might still be a little messy, tho. |
| 00:37.32 | WIMPy | kaushal: Might be a good idea to compress the log before uploading. |
| 00:38.30 | BenC[UK] | This is my dialplan - http://pastebin.com/dg6Mst6W, log is here: http://pastebin.com/h4sx1jb1 - if you look at bottom (line 171) of the log, theres "Exiting on time-out cycle"... |
| 00:39.03 | kaushal | WIMPy: http://pastebin.ubuntu.com/667806/ |
| 00:47.08 | kaushal | WIMPy: Are you able to retrieve the files ? |
| 00:47.34 | WIMPy | Got two so far. |
| 00:48.19 | kaushal | ok |
| 00:48.25 | WIMPy | Ok, got them. |
| 00:48.34 | kaushal | ok |
| 00:52.00 | kaushal | WIMPy: i suppose the file is corrupt ? |
| 00:52.21 | kaushal | I hope its not |
| 00:52.24 | WIMPy | Nope. all fine. |
| 00:52.38 | kaushal | WIMPy: how did you downloaded it ? |
| 00:52.41 | WIMPy | Trying to find anything interesting... |
| 00:52.58 | WIMPy | How? With a browser? |
| 00:53.02 | kaushal | WIMPy: did you registered to filefactory |
| 00:53.16 | WIMPy | nope. Waited to 30s each. |
| 00:53.20 | WIMPy | for |
| 00:54.18 | kaushal | WIMPy: i mean to retrieve the file |
| 00:54.33 | *** join/#asterisk coppice (~chatzilla@116.92.38.165) |
| 00:54.50 | kaushal | i did wget http://www.filefactory.com/file/cd01225/n/messages.gz |
| 00:54.55 | kaushal | it got corrupt |
| 00:55.11 | WIMPy | I clicked on 'slow download' filled the captcha, waited 30s and then got the file. |
| 00:55.18 | kaushal | oh ok |
| 00:55.20 | kaushal | great |
| 00:55.28 | WIMPy | Yes, you get a script-filled webpage. |
| 00:56.37 | kaushal | ok |
| 00:57.02 | WIMPy | Argh |
| 00:59.44 | kaushal | WIMPy: Anything weird ? |
| 00:59.51 | WIMPy | It starts before we get those 101s. There are som STATUS messages popping up. |
| 01:00.04 | WIMPy | Yes, "unknown protocol" :-( |
| 01:00.17 | kaushal | oh ok |
| 01:00.17 | WIMPy | You're in the wrong area. |
| 01:00.44 | WIMPy | I usually prefer to use coloured output. |
| 01:00.47 | kaushal | WIMPy: not sure i understand that |
| 01:01.24 | kaushal | coloured output ? |
| 01:01.34 | kaushal | I am learning new and new things |
| 01:01.35 | WIMPy | My decoder doesn't support that protocol. |
| 01:01.46 | kaushal | oh ok |
| 01:01.56 | kaushal | which decoder do you use ? |
| 01:02.19 | WIMPy | I guess I could just remove the check and (at least) most things would work. |
| 01:02.23 | WIMPy | tracI |
| 01:02.37 | kaushal | WIMPy: Also what does 101 exactly mean ? |
| 01:02.48 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 01:03.30 | WIMPy | {0x65, "Message not compatible with call state"}, |
| 01:03.56 | WIMPy | Oh. |
| 01:04.01 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:04.16 | WIMPy | This is getting interesting. |
| 01:04.20 | kaushal | WIMPy: sure |
| 01:04.25 | kaushal | what does it say ? |
| 01:06.04 | WIMPy | The "Just queued I-frame due to peer busy condition or window shut" is interesting. |
| 01:06.11 | *** join/#asterisk james_zhu (~Administr@183.16.209.216) |
| 01:06.12 | kaushal | ok |
| 01:06.18 | WIMPy | Are you causing overload at your telco? |
| 01:06.26 | kaushal | hmmmm |
| 01:06.59 | kaushal | WIMPy: does 101 mean Message not compatible with call state ? |
| 01:07.13 | WIMPy | yes |
| 01:07.27 | kaushal | not sure i still understand that |
| 01:07.59 | kaushal | WIMPy: please help me understand {0x65, "Message not compatible with call state"} |
| 01:08.32 | WIMPy | It means that your asterisk and your telcos switch don;t agree on the current call state. |
| 01:08.36 | WIMPy | Or there's a bug. |
| 01:09.19 | kaushal | WIMPy: is there a wiki which says about {0x65, "Message not compatible with call state"} |
| 01:09.44 | WIMPy | just google for "isdn cause codes" |
| 01:09.57 | kaushal | i need some sort of proof to show it to the development team |
| 01:09.59 | kaushal | ok |
| 01:11.27 | WIMPy | Having seperate interfaces with overlapping call references is quite messy to read. |
| 01:11.29 | kaushal | WIMPy: Does this site http://networking.ringofsaturn.com/Routers/isdncausecodes.php explain it correctly ? |
| 01:11.57 | kaushal | I mean is it authentic ? |
| 01:13.09 | WIMPy | Seems ok. |
| 01:13.22 | kaushal | ok |
| 01:13.38 | kaushal | call state meaning ? |
| 01:13.55 | kaushal | does it mean hungup or disconnected or ..... |
| 01:13.59 | WIMPy | dialing, ringing, proceeding, connected... |
| 01:14.16 | kaushal | ok |
| 01:14.34 | kaushal | so what does Message not compatible mean ? |
| 01:14.53 | kaushal | which Message is he referring to ? |
| 01:15.04 | WIMPy | That a message was received that makes no sens at that time. |
| 01:15.19 | kaushal | ok |
| 01:15.29 | kaushal | so Message received by telco switch ? |
| 01:15.39 | WIMPy | Like e.g. indicating ringing after the call has been connected. |
| 01:15.42 | WIMPy | yes |
| 01:15.45 | kaushal | ok |
| 01:15.49 | kaushal | WIMPy: great |
| 01:16.35 | kaushal | WIMPy: so it means i need to talk to telco and not Asterisk or Sangoma folks ? |
| 01:16.47 | kaushal | just ensuring it completely |
| 01:16.53 | WIMPy | IThat situation can be normal to a certain extent. |
| 01:17.13 | WIMPy | There can be race conditions. |
| 01:17.31 | kaushal | ok |
| 01:17.56 | WIMPy | Ok, filtered out span 8. That makes it easier. |
| 01:18.08 | kaushal | WIMPy: so do i need to tweak some config files ? |
| 01:18.16 | kaushal | in order to make it work ? |
| 01:18.35 | kaushal | I mean in order to overcome race conditions ? |
| 01:19.08 | WIMPy | Generating the calls more slowly might help. |
| 01:19.16 | WIMPy | BUt let me read on... |
| 01:19.20 | kaushal | sure |
| 01:19.22 | kaushal | WIMPy: np |
| 01:23.00 | WIMPy | I currently thin we have two issues here: 1. signalling overload and 2. a bug. |
| 01:23.09 | kaushal | ok |
| 01:25.28 | kaushal | bug in the sense ? |
| 01:25.42 | WIMPy | Not sure yet. |
| 01:26.08 | kaushal | is it on the dialer or sangoma card or asterisk or wanpipe or libpri or dahdi ? |
| 01:26.16 | WIMPy | It is Asteris that starts sending several status messages. |
| 01:26.18 | kaushal | dialer is written in Java |
| 01:26.48 | WIMPy | I'm never totally sure, what happens where, but I think that's in libpri. |
| 01:27.02 | kaushal | ok |
| 01:27.15 | kaushal | so whom do i contact for libpri ? |
| 01:27.35 | kaushal | you also said signalling overload ? |
| 01:28.28 | WIMPy | Yes, there seems to be a queue building up. That ist starting the trouble. |
| 01:29.03 | kaushal | ok |
| 01:29.04 | WIMPy | But I'm not sure why the call is cleared. |
| 01:29.30 | kaushal | WIMPy: does this explain http://www.comptek.ru/telephony/sr/isdn/0867-05-10.html about call state ? |
| 01:31.32 | WIMPy | There's more. |
| 01:31.48 | *** join/#asterisk mistergibson (~mistergib@71-36-122-200.ptld.qwest.net) |
| 01:31.51 | kaushal | WIMPy: can you please point me to the correct web page ? |
| 01:32.32 | WIMPy | Q.931 but there might be something more readable. |
| 01:34.30 | kaushal | WIMPy: Thanks |
| 01:34.51 | kaushal | Also whom do i contact for this issue ? |
| 01:34.54 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 01:34.58 | WIMPy | I can /msg you the list if you want. |
| 01:35.06 | kaushal | WIMPy: sure |
| 01:35.25 | WIMPy | I think you should ensure that the dialler doesn't generate more calls than possible. |
| 01:37.38 | kaushal | ok |
| 01:41.32 | WIMPy | Hmm. It gets more interesting on the call I'm on to now. |
| 01:41.42 | kaushal | WIMPy: sure |
| 01:43.20 | WIMPy | Span 8, call 153: You receive a progress 3s after an alerting. |
| 01:43.58 | WIMPy | With 1695 lines of debug in between. |
| 01:44.34 | WIMPy | Looks like you managed to generate packet loss on the D-channel |
| 01:45.07 | mistergibson | Shameless Newb Plea #2: I've cobbled something together for a config, but would appreciate any advice. link => https://gist.github.com/1150617 |
| 01:45.46 | WIMPy | But it lookes like the situation could be dealt with in a better way. |
| 01:46.17 | kaushal | ok |
| 01:46.36 | WIMPy | You might be able to tweak some timers. |
| 01:47.13 | WIMPy | Once the link is congested, it generates a lot of status messages which make the situation even worse. |
| 01:47.21 | kaushal | ok |
| 01:48.09 | WIMPy | That goes on for 19s. |
| 01:48.56 | WIMPy | Probably the call has been answered during that time, which couldn't be acknowledged because you never got the message. |
| 01:53.24 | WIMPy | You might be able to tweak a few bits, but that dialler should really pay attention to the the progress of the calls it generates. |
| 01:53.57 | *** join/#asterisk jetlag (jetlag@pool-71-188-2-156.cmdnnj.east.verizon.net) |
| 02:01.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:03.33 | mistergibson | anyone : @extensions.conf, how would any suggest I process a call attempt to an undefined extension? for sip calls, I see it jumps to current-priority+101, but unsure if there is another jump distance for undefined extension. |
| 02:04.33 | WIMPy | Don't use jumps. |
| 02:04.51 | WIMPy | Haven't they even been removed completely? |
| 02:04.54 | mistergibson | WIMPy: not trying to, I understand it is default behavior |
| 02:05.21 | WIMPy | It might have been with Asteisk 1.2. |
| 02:05.45 | mistergibson | I understand that a Dial(SIP/blah,xx); fail will goto >current-priority< + 101 ... right? |
| 02:06.00 | WIMPy | Only if you told it to. |
| 02:06.12 | mistergibson | hrm, well -- I didn't |
| 02:06.38 | WIMPy | you might have priorityjumping under [general] |
| 02:07.08 | mistergibson | WIMPy: you know of an example context that deals with call fails, etc? |
| 02:07.32 | WIMPy | Any of those standard extension macros. |
| 02:22.31 | mistergibson | WIMPy: I see -- so use M or U ... |
| 02:23.48 | WIMPy | For what? |
| 02:26.18 | mistergibson | WIMPy: for handling the possibilities of a call attempt |
| 02:26.36 | mistergibson | not found, not-online, voicemail, whatever |
| 02:26.43 | mistergibson | or success I suppose |
| 02:29.14 | WIMPy | Just check DIALSTATUS and HANGUPCAUSE after the Dial. |
| 02:30.51 | mistergibson | WIMPy: ok, I'll look those up |
| 02:32.22 | ChannelZ | Anyone setup a fax printer with Samba? |
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| 02:43.36 | kaushal | WIMPy: sorry was away for a while |
| 02:43.48 | kaushal | WIMPy: i looked at the scroll |
| 02:44.00 | kaushal | < WIMPy> You might be able to tweak a few bits, but that dialler should really pay attention to the the progress of the calls it generates. |
| 02:44.35 | kaushal | tweak a few bits meaning editing /etc/asterisk/*.conf files ? |
| 02:45.06 | kaushal | not sure i understand that part |
| 02:45.14 | kaushal | as i said i am still learning |
| 02:46.46 | WIMPy | No. You could save a few bits in the messages. But I doubt that's worth the effort. |
| 02:50.56 | kaushal | ok |
| 02:51.10 | kaushal | WIMPy: also when i look at http://en.wikipedia.org/wiki/Q.931 |
| 02:51.21 | kaushal | I see two fields Hex and Dec |
| 02:51.31 | kaushal | whats their significance |
| 02:51.47 | kaushal | I mean their actual use case |
| 02:56.45 | WIMPy | It's just how the number is written. |
| 02:56.49 | WIMPy | http://en.wikipedia.org/wiki/Hexadecimal |
| 02:57.24 | WIMPy | Cause codes have always been written in hex until Asterisk came. |
| 02:57.43 | WIMPy | Well at least almost always. |
| 02:58.19 | WIMPy | As they are groued in to 8 groups of 16 causes each, that makes more sense than writing them in decimal. |
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| 03:01.11 | kaushal | 8 groups of 16 causes ? |
| 03:01.23 | kaushal | i see more than 16 causes each |
| 03:02.13 | kaushal | WIMPy: never mind |
| 03:02.19 | kaushal | I will have a look at it |
| 03:02.20 | WIMPy | (0) 1-15, 16-31, 32-47, ... |
| 03:02.30 | kaushal | oh ok |
| 03:02.50 | kaushal | so it means channels ? |
| 03:02.57 | WIMPy | Or in hex 00-0f, 10-1f, 20-2f, ... |
| 03:03.03 | WIMPy | Just the numbers. |
| 03:03.19 | kaushal | ok |
| 03:03.49 | WIMPy | You can see that xF is alwyas a (group), unspecified. |
| 03:04.52 | WIMPy | And there's also a location field that belongs to the cause code. |
| 03:05.53 | kaushal | ok |
| 03:05.59 | kaushal | trying to understand |
| 03:06.36 | kaushal | location field ? |
| 03:06.39 | WIMPy | You don;t only want to know WHY a call was released, but also WHERE. |
| 03:09.20 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 03:11.34 | kaushal | WIMPy: Thanks again |
| 03:12.31 | kaushal | You have been so kind and with utmost patience for explaining me |
| 03:12.55 | kaushal | I will take it further and definetly get back to you after my findings |
| 03:13.26 | kaushal | is obliged to WIMPy |
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| 03:18.11 | kaushal | WIMPy: Also whom do i report about libpri ? |
| 03:18.33 | kaushal | if at all there is a issue |
| 03:20.50 | WIMPy | I don;t think it's a real bug. |
| 03:21.02 | *** join/#asterisk james_zhu (~Administr@183.16.209.216) |
| 03:21.05 | WIMPy | It's an extreme race condition. |
| 03:21.33 | kaushal | ok |
| 03:22.21 | WIMPy | It might be possible to hadle that in a little more graceful way. But that's not the real issue. |
| 03:22.28 | kaushal | WIMPy: I have been reading through http://www.asterisk.org/architecture |
| 03:23.09 | WIMPy | How are the calls generated? And what happens to them? |
| 03:23.47 | kaushal | Just trying to understand the vertical block of Asterisk |
| 03:24.02 | kaushal | where does libpri and dahdi and various components |
| 03:24.08 | kaushal | fit into this blocks |
| 03:24.12 | WIMPy | That doesn't have much to do with Asterisk. |
| 03:25.02 | WIMPy | Dahdi is the hardware driver. chan_dahdi interfaces Asterisk to that hardware. |
| 03:25.41 | WIMPy | And (part of) the protocol that is spoken by the hardware on the line is done by libpri. |
| 03:26.53 | kaushal | ok |
| 03:27.28 | WIMPy | How are the calls generated? And what happens to them? |
| 03:27.33 | kaushal | so where does wanpipe fit ? |
| 03:28.03 | kaushal | WIMPy: yes I would really understand that |
| 03:28.30 | WIMPy | That's the Sangoma stuff that interfaces to (chan_)dahdi. Not sure exactely where they connect. |
| 03:28.56 | WIMPy | You should concentrat on the other side of Asterisk. |
| 03:29.04 | WIMPy | e |
| 03:29.26 | WIMPy | i.e. where the calls are generated/come in, not where they go out. |
| 03:31.40 | kaushal | WIMPy: ok |
| 03:33.23 | kaushal | Cause: Message not compatible with call state (101), class = Protocol Error (e.g. unknown message) (6) |
| 03:33.52 | kaushal | is there a detailed way of understanding it ? |
| 03:34.04 | kaushal | I mean stating about class |
| 03:34.39 | kaushal | Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) |
| 03:34.40 | WIMPy | Yes, that's what I said: 101 = 0x65, class 6 code 5. |
| 03:34.42 | kaushal | < Ext: 1 Cause: Message not compatible with call state (101), class = Protocol Error (e.g. unknown message) (6) ] |
| 03:34.45 | kaushal | < Cause data 1: 7d (125, STATUS message) |
| 03:35.50 | WIMPy | And you see the location there. |
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| 03:36.24 | kaushal | ok |
| 03:36.37 | kaushal | two more questions |
| 03:36.43 | kaushal | code 5 meaning ? |
| 03:36.50 | kaushal | Location: Public network serving the local user (2) meaning ? |
| 03:37.17 | WIMPy | But the location can be wrong in case of an EWSD if both caller and callee are on the same switch. |
| 03:37.43 | WIMPy | "Message not compatible with call state" You got that list. |
| 03:38.10 | WIMPy | And the location means exactely what it saus. |
| 03:38.25 | WIMPy | you = local user |
| 03:39.15 | kaushal | code 5 meaning ? |
| 03:40.18 | WIMPy | In class 6 "Message not compatible with call state" |
| 03:41.43 | kaushal | ok |
| 03:42.15 | kaushal | so that means i need to revisit the dialer logic which is written in JAVA ? |
| 03:42.28 | kaushal | to make optimum use of the channels ? |
| 03:43.08 | kaushal | or some sort of involving timeout |
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| 03:44.31 | kaushal | WIMPy: Am i understanding you correctly ? |
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| 03:58.11 | WIMPy | yes |
| 03:58.28 | WIMPy | How do the calls get to Asterisk? |
| 03:58.54 | WIMPy | You should check how the calls proceed. |
| 03:59.28 | WIMPy | And how do you distribute the calls ofer the interfaces? |
| 04:00.45 | kaushal | ok |
| 04:01.21 | kaushal | interfaces meaning w1g1 , w2g1 ..... ? |
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| 04:02.16 | WIMPy | Yes, do you have them all in one group? |
| 04:02.23 | kaushal | nope |
| 04:02.29 | kaushal | g0 ... g7 |
| 04:02.39 | mistergibson | is NoOp(value) the easiest way to return data from a macro? |
| 04:03.51 | WIMPy | mistergibson: You don;t return anything from a macro, but you can use NoOp or Verbose to display things on your console depending on verbose level. |
| 04:04.05 | kaushal | WIMPy: its spread to different call groups |
| 04:04.10 | kaushal | does that hurt ? |
| 04:04.24 | mistergibson | WIMPy: ok, got it thanks |
| 04:04.42 | WIMPy | No, you might want to try to distribute the calls over the interfaces. |
| 04:05.02 | kaushal | WIMPy: basically i have 8 port Sangoma PRI Card with 8 PRI Lines |
| 04:05.03 | WIMPy | So that the signalling load gets distributed. |
| 04:05.38 | kaushal | WIMPy: so shall i pastebin the chan_dahdi.conf ? |
| 04:05.49 | WIMPy | Nope. |
| 04:06.02 | WIMPy | That's about how you generate the calls. |
| 04:06.19 | kaushal | < WIMPy> No, you might want to try to distribute the calls over the interfaces. |
| 04:06.31 | WIMPy | Not that you dial the first 30 calls on the first pri then the next 30 on the 2nd and so on. |
| 04:06.39 | kaushal | so you mean programmatically ? |
| 04:07.05 | WIMPy | Dial the first call on the 1st pri, the 2nd call on the 2nd pri and so on. |
| 04:07.07 | WIMPy | yes |
| 04:07.13 | kaushal | oh ok |
| 04:07.39 | kaushal | so can it be addressed concurrently |
| 04:07.52 | kaushal | since i need to have 240 calls per minute |
| 04:08.27 | kaushal | I mean if your solution work for 240 calls per minute |
| 04:08.38 | WIMPy | Hmm. One call per minute an cahnnel or one call every 2 seconds per pri istn't that much. |
| 04:08.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:08.42 | carrar | heh |
| 04:09.00 | WIMPy | There must be some bad bursts. |
| 04:09.30 | tonsofpcs | you're not going to actually connect 240 calls per minute unless they're simultaneous. 240 line latches (off-hooks) per minute, maybe... |
| 04:10.07 | WIMPy | on 240 channel. |
| 04:10.14 | WIMPy | s |
| 04:10.29 | tonsofpcs | WIMPy: right, I get that. How are you going to synchronize people dialing? I know some people that take up to a minute and a half to dial a 7 digit number! |
| 04:10.29 | kaushal | WIMPy: so basically i have 250 phone numbers with a campaign of 725K obd-demo.mp3 file |
| 04:10.41 | tonsofpcs | oh, automated calling |
| 04:10.44 | kaushal | yeah |
| 04:11.09 | kaushal | so the dialer dials out and plays this obd-demo.mp3 file |
| 04:11.12 | tonsofpcs | does obd-demo.mp3 start with your name and how I can identify you to the government for calling me illegally? |
| 04:11.26 | WIMPy | tonsofpcs: And in Asterisk land dialling doesn't count as everybody but me seems to take en-block dialing for granted. |
| 04:12.08 | kaushal | WIMPy: does that sound unreasonable ? |
| 04:12.21 | WIMPy | Err, what? |
| 04:12.28 | tonsofpcs | en-block dialing? Is that a fancy term for "we don't latch the line until the handset has dialed the required number of digits" ? |
| 04:12.41 | WIMPy | tonsofpcs: yes |
| 04:12.46 | kaushal | so what i mean was making use of 240 channels to dial 250 phone numbers and play a campaign of 725K obd-demo.mp3 file |
| 04:13.05 | tonsofpcs | yea, my current PBX latches the line the moment you dial the dial-out code (read: 9) |
| 04:13.22 | kaushal | WIMPy: you said Dial the first call on the 1st pri, the 2nd call on the 2nd pri and so on. ? |
| 04:13.29 | WIMPy | kaushal: 250>240 |
| 04:13.29 | tonsofpcs | kaushal: you can't dial 250 numbers on 240 channels. |
| 04:13.38 | WIMPy | tonsofpcs: Yes, that's the way PBXs work. |
| 04:14.02 | kaushal | WIMPy: tonsofpcs apologies |
| 04:14.06 | tonsofpcs | WIMPy: well, I can program it to not latch until the fourth (or arbitray number) digit is dialed.... |
| 04:14.07 | kaushal | the number is 239 |
| 04:14.24 | WIMPy | Unless you thin that at least 10 of those 250 numbers are invalid or busy. |
| 04:14.25 | kaushal | so what if the phone numbers are more than 239 ? |
| 04:14.43 | kaushal | if i have capacity of 240 channels |
| 04:15.09 | WIMPy | But you shouls track the calls and make sure you only dial 240 and then start the next call if one has finished. |
| 04:15.10 | kaushal | basically i am trying to understand how this can be addressed efficiently |
| 04:15.47 | tonsofpcs | also, you want to keep channels open for incoming, channels for local outgoing, channels for.... |
| 04:16.04 | kaushal | There is no incoming call here |
| 04:16.13 | kaushal | its exclusively outgoing |
| 04:16.26 | WIMPy | Get the timing right. Ich you calculate 1 minute per call, don't dial more than 5 per second unless you know you have free channels. |
| 04:16.39 | tonsofpcs | so what shows up on caller ID when you call me? Where does that number go to? What if your family has an emergency and needs to contact you? |
| 04:16.41 | WIMPy | If |
| 04:17.16 | kaushal | WIMPy: its a 30 sec campaign |
| 04:17.22 | WIMPy | You don;t do that kind of stuff on lines you also use for other things. |
| 04:17.23 | kaushal | i have 240 channels |
| 04:17.53 | tonsofpcs | also, do you actually have 240 bearer lines or are some of those 240 data lines? |
| 04:18.05 | kaushal | its only bearer channels |
| 04:18.13 | kaushal | data lines ? |
| 04:18.22 | kaushal | meaning control and signal |
| 04:18.26 | tonsofpcs | timing/control/signal |
| 04:18.31 | kaushal | nope |
| 04:18.36 | tonsofpcs | "D-channel" if you will |
| 04:18.36 | WIMPy | Aharing lines is a pure US phenomena, I think. |
| 04:18.37 | kaushal | i have total of 248 |
| 04:18.48 | WIMPy | 256 |
| 04:18.48 | kaushal | channels |
| 04:19.02 | kaushal | since its E1 ISDN |
| 04:19.16 | kaushal | out of which 8 are D channels |
| 04:19.22 | kaushal | and rest are bearer channels |
| 04:19.32 | tonsofpcs | yup. |
| 04:19.36 | WIMPy | And 8 unused / for synchronisation. |
| 04:19.38 | tonsofpcs | no, it's 248. 31 * 8 |
| 04:19.55 | WIMPy | It's 32*8 |
| 04:20.30 | tonsofpcs | I thought that there's 32 time slots, 31 are available as channels, 1 of those is the D-channel... |
| 04:20.36 | kaushal | tonsofpcs: WIMPy so how do ideally run a campaign of 30 sec using 240 available bearer channels |
| 04:20.48 | WIMPy | Yes, that's what I'm saying. |
| 04:21.32 | kaushal | WIMPy: are you referring to me ? |
| 04:21.46 | WIMPy | Either start the calls at the estimated average speed or distribute tem sequentially across the lines. |
| 04:22.12 | WIMPy | No, that was for tonsofpcs, sorry. |
| 04:22.19 | kaushal | ok |
| 04:22.45 | tonsofpcs | kaushal: can't you check for an available line and only attempt a connection if one is available? |
| 04:22.59 | kaushal | ok |
| 04:23.06 | kaushal | tonsofpcs: yes makes sense |
| 04:23.12 | WIMPy | He's overruning the switch. |
| 04:23.25 | tonsofpcs | well, you still need to stagger them |
| 04:23.34 | tonsofpcs | minimum delay between each, etc. |
| 04:23.43 | WIMPy | And yes. once you have set up 240 calls generate one if another one has finished. |
| 04:24.01 | kaushal | got it |
| 04:24.07 | kaushal | completely understood |
| 04:24.12 | WIMPy | Good |
| 04:24.26 | kaushal | so ideally E1 ISDN means 32 * 8 channels ? |
| 04:24.43 | WIMPy | E1 is 32 channels. |
| 04:25.06 | kaushal | out of which is 240 bearer channels 8 D channels and 8 channels for synchronisation ? |
| 04:25.09 | WIMPy | Or as tonsofpcs more correctly said: 32 timeslots. |
| 04:25.34 | kaushal | Am i understanding it correctly ? |
| 04:25.42 | tonsofpcs | 31 channels, 32 timeslots. The last timeslot is NOT and CANNOT be a channel. |
| 04:26.10 | WIMPy | Slot 0 only carries a sync pattern. |
| 04:26.25 | kaushal | so is it 240 bearer channels 8 D channels ? |
| 04:26.32 | tonsofpcs | we have the 24th timeslot here too, it just happens to be a single bit instead of a full 8 bits. |
| 04:26.40 | tonsofpcs | kaushal: yes. |
| 04:26.43 | WIMPy | kaushal: yes |
| 04:26.57 | kaushal | great |
| 04:27.12 | kaushal | WIMPy: tonsofpcs: would update further |
| 04:27.19 | kaushal | Thanks for all the help |
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| 05:57.42 | schmidts | good morning |
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| 06:46.37 | giany | hello |
| 06:47.18 | giany | anyone can explain what this mean : |
| 06:47.20 | giany | asterisk -rx 'g729 show licenses' |
| 06:47.20 | giany | 2/4 encoders/decoders of 100 licensed channels are currently in use |
| 06:47.29 | giany | and |
| 06:47.29 | giany | asterisk -rx 'sip show channels' |
| 06:47.35 | giany | shows : 2 active SIP dialogs |
| 06:47.43 | ChannelZ | ~pb |
| 06:47.43 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 06:47.46 | giany | what does 2/4 encoders/decoders mean ? |
| 06:48.39 | giany | http://pastebin.com/UEVKd3H5 |
| 06:49.26 | ChannelZ | What are the 2 channels actually doing? |
| 06:49.54 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:50.26 | giany | there is only one call on the system |
| 06:50.45 | giany | that why it shows 2 sip dialogs |
| 06:50.46 | grEvenX | anyone else have had issues with the SIP messages coming in on the network interface, but for some reason is not picked up by asterisk? |
| 06:51.00 | grEvenX | this has happened to a few of our servers some times |
| 06:51.05 | giany | though i don't understand what 2/4 encoders/decoders mean |
| 06:51.20 | grEvenX | the load on the server is low |
| 06:51.31 | kaldemar | 1 call through asterisk is 2 dialogs. |
| 06:51.33 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 06:52.07 | kaldemar | UA -(dialog)- UA -(dialog)- UA |
| 06:52.16 | WIMPy | grEvenX: There is a known problem when using timerfd, stalling chan_sip. See issues.asterisk.org. |
| 06:52.24 | *** join/#asterisk oej (~olle@office.ipvision.dk) |
| 06:52.33 | giany | kaldemar: yes, but 2/4 what does that mean? |
| 06:52.35 | ChannelZ | Well if your device is talking g729 and your ITSP is talking g729 but Asterisk is in the middle having to "listen" to the audio stream (IE the two endpoints are not bridged directly together) you're using 2 encoders/decoders. Not sure why it's showing 4 though which is why I asked what it's actually doing |
| 06:53.39 | grEvenX | WIMPy: in 1.4 branch as well? |
| 06:53.55 | WIMPy | grEvenX: No |
| 06:54.04 | WIMPy | 1.4 didn't know timerfd, yet. |
| 06:54.23 | grEvenX | hm, ok, this happends for me on 1.4 :/ |
| 06:56.29 | singler | grEvenX: is your server also NATing/masquerading internet for local devices? |
| 06:57.28 | grEvenX | singler: no |
| 06:59.47 | grEvenX | restarting asterisk did not help |
| 06:59.58 | grEvenX | so I am afraid that this is something happening on the OS level :( |
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| 07:04.06 | grEvenX | restarted the entire server and the problem seems to go away@ |
| 07:04.08 | ChannelZ | have you done tcpdumps or anything to see the traffic coming in? |
| 07:04.28 | grEvenX | ChannelZ: yes, and I see the traffic coming in, but no response from asterisk |
| 07:04.37 | grEvenX | so, just a lot of retries |
| 07:05.13 | ChannelZ | Does the machine lose connectivity completely? |
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| 07:07.15 | grEvenX | ChannelZ: No, not at all (and then I wouldn't see the incoming SIP packets in tcpdump) |
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| 07:07.53 | asterisk-Tester | hello, can someone telll point me to the appropriate channel to disscuss smpp (SMS) protocol ? |
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| 07:15.13 | grEvenX | both servers that were facing the issues had about 300 days of uptime |
| 07:17.24 | *** join/#asterisk lkthomas (~lkthomas@59.152.236.158) |
| 07:17.26 | lkthomas | hey guys |
| 07:17.33 | lkthomas | I am trying to add 6 in front of all extension |
| 07:17.43 | lkthomas | but asterisk still taking the old 3 digi number |
| 07:17.47 | ChannelZ | grEvenX: I more meant other services stop working |
| 07:17.54 | lkthomas | for example, it works when I use 700 ext |
| 07:17.58 | lkthomas | but not 6700 |
| 07:18.08 | lkthomas | how could I trace which config have problem ? |
| 07:18.13 | ChannelZ | extensions.conf |
| 07:18.13 | grEvenX | ChannelZ: the servers are almost entirely dedicated to asterisk, so not much else running there :/ |
| 07:18.25 | ChannelZ | hmm |
| 07:18.28 | lkthomas | ChannelZ, the problem is that extensions.conf is normal |
| 07:18.37 | lkthomas | I restart asterisk and still the same |
| 07:18.39 | ChannelZ | It might be normal but it apparently isn't doing what you want |
| 07:18.50 | lkthomas | it have 6700 in place |
| 07:18.51 | kaldemar | lkthomas: pastebin a CLI output of a call and someone will tell you exactly what you need to change. |
| 07:18.53 | lkthomas | no 700 |
| 07:18.54 | ChannelZ | extensions.conf is where all dialing is controlled though |
| 07:19.19 | ChannelZ | (well unless you're using AEL etc but..) |
| 07:24.46 | lkthomas | http://pastebin.com/3bwNDr5D |
| 07:25.21 | lkthomas | Accepting call from '35552235' to '700' on channel 0/23, span 1 <--- this one should be 6700 |
| 07:25.29 | lkthomas | but I can't understand why it still using 700 |
| 07:25.44 | kaldemar | why should it be 6700 if you dial 700? |
| 07:25.58 | kaldemar | or are you actually dialing 6700 with your phone? |
| 07:26.13 | lkthomas | well, 700 extension does not exists |
| 07:26.39 | lkthomas | only 6700 e xists |
| 07:27.06 | WIMPy | Yes. but 700 was dialled, so that didn;t work. |
| 07:27.09 | ChannelZ | you're confusing extensions with devices it sounds like |
| 07:27.51 | lkthomas | but once I chage extension to 700, it work |
| 07:28.02 | lkthomas | why is it unable to pick up "6" ? |
| 07:28.02 | ChannelZ | pastebin your extensions.conf because we have no idea what you're talking about |
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| 07:28.26 | lkthomas | ChannelZ, mind to tell me the way to trace the problem instead ? |
| 07:28.39 | kaldemar | lkthomas: where do you expect asterisk to pick up the 6? |
| 07:28.46 | ChannelZ | not really since I don't even know what the problem really is. |
| 07:29.01 | ChannelZ | You're dialing 700 as far as Asterisk is concerned |
| 07:29.16 | ChannelZ | If you're really dialing 6700 on your phone, probably the phone's dialplan is chewing it up |
| 07:30.09 | lkthomas | ChannelZ, again, 700 does not exists on dail plan |
| 07:30.27 | kaldemar | the 700 came in from DAHDI. |
| 07:30.30 | kaldemar | change Set(__FROM_DID=${EXTEN}) to Set(__FROM_DID=6${EXTEN}) |
| 07:30.33 | WIMPy | lkthomas: Yes, that's why a call to 700 doesn;t work. |
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| 07:31.42 | kaldemar | or go to #freepbx to ask for a better solution. |
| 07:31.56 | ChannelZ | "-- Accepting call from '35552235' to '700' on channel 0/23, span 1" -- Wherever it's coming from, that's what it's getting. |
| 07:32.42 | ChannelZ | And apparently it DOES exist in the dialplan or you'd be getting a different error. Plus: "-- Executing [700@from-pstn:1]" |
| 07:33.01 | *** join/#asterisk jkroon (~jkroon@197.169.62.235) |
| 07:33.20 | jkroon | hi guys, got a locked up chan_sip.so on a asterisk 1.8.5.0 setup. |
| 07:33.30 | jkroon | what info should I grab before restarting asterisk? |
| 07:33.55 | schmidts | jkroon you should have a backtrace.txt file in your asterisk source directory |
| 07:34.36 | schmidts | jkroon or you look here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 07:34.37 | jkroon | schmidts, it didn't crash - it's just locked up. |
| 07:34.57 | WIMPy | ChannelZ: That is probably a catchall. |
| 07:35.12 | schmidts | Getting Information For A Deadlock read this part in the wiki jkroon |
| 07:35.31 | oej | Run "core show locks" |
| 07:35.42 | oej | And possibly attach GDB to the running process and produce a backtrace |
| 07:35.45 | ChannelZ | WIMPy: true.. |
| 07:35.57 | oej | There was a report on deadlocks in 1.8 on asterisk-dev yesterday |
| 07:36.05 | oej | We really, really need to fix 1.8 |
| 07:36.10 | schmidts | oej core show locks is only available if you have set the compiler flag for it ;) |
| 07:36.11 | oej | So please help us debugging! |
| 07:36.19 | oej | Right. But that's what we need |
| 07:36.39 | schmidts | i know |
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| 07:41.26 | jkroon | oej, don't seem to have core show locks. |
| 07:41.48 | jkroon | will see what I need to get it, for now I had to restart asterisk but it will lock up again. |
| 07:41.50 | oej | As schmidts said above, you need to compile it with the compiler flag for it in order to be able to do that |
| 07:41.57 | jkroon | so will prepare for that. |
| 07:42.06 | jkroon | got it - will go find it. |
| 07:42.28 | WIMPy | debug_threads |
| 07:43.12 | kaldemar | and DONT_OPTIMIZE |
| 07:43.25 | jkroon | thanks guys. |
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| 07:44.33 | e-fon_patrick | Hi guys, got a problem with realtime Queue in Asterisk 1.6.1.20. 2 Servers, same Database, same Asterisk Release, same configuration files. BUT: on one Server: i'm not able to get the queue member in service "SIP/0435000491 (realtime) (Unavailable)" same account on other server is fine. Any hint is really appreciated |
| 07:45.09 | gokulnath | hello, in ubuntu 10.04 server is header packages necessary to be installed before dahdi and asterisk |
| 07:46.14 | ChannelZ | kernel headers before dahdi, yes. asterisk, no |
| 07:46.43 | ChannelZ | but dahdi before asterisk so at the end of the day, yes! |
| 07:46.53 | ChannelZ | How's that for an answer!? |
| 07:47.30 | e-fon_patrick | oh forgot to mention for my problem servers are both running on Centos 5: servers are both Centos 5. |
| 07:50.13 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 07:54.54 | Gugge | e-fon_patrick: what does "sip show peers" show abot that sip peer? |
| 07:58.37 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 08:01.52 | lkthomas | ChannelZ, Thanks, the problem is come from telco |
| 08:02.00 | lkthomas | because telco use 3 digi |
| 08:02.07 | lkthomas | while we need 4 digi |
| 08:03.28 | WIMPy | lkthomas: How does that work out? They would cut the last digit, not something in the middle. |
| 08:03.30 | ChannelZ | How do you even dial those? |
| 08:05.26 | *** join/#asterisk k3asd` (~k3asd@static-217-133-87-244.clienti.tiscali.it) |
| 08:05.35 | k3asd` | hi |
| 08:07.20 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:21.59 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
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| 08:22.50 | *** part/#asterisk mistergibson (~mistergib@71-36-122-200.ptld.qwest.net) |
| 08:24.23 | irroot | ok cool new load balancing macro works well balances calls accross n sip accounts each routed to there own DSL line to same remote ITSP |
| 08:28.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:31.47 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 08:34.20 | lkthomas | ChannelZ, I think telco did something |
| 08:34.33 | lkthomas | ChannelZ, not sure, but after they modify to 4 digi, it works |
| 08:36.35 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 08:38.03 | nunne | Using asterisk1.6 + imapstorage for voicemail. is it normal to have slow output when doing voicemail show users for default (i have 30-ish mailboxes). i first get the first 15-20.. and then a few seconds it lists the rest. |
| 08:39.32 | *** join/#asterisk james_zhu (~Administr@183.16.209.216) |
| 08:41.38 | wdoekes2 | I haven't used imap storage, but I wouldn't be surprised that it's slow |
| 08:45.05 | *** join/#asterisk carloimperia (~carloimpe@109.116.214.6) |
| 08:47.00 | *** join/#asterisk Sakuranbo (~Sakuranbo@59.152.236.158) |
| 08:47.24 | Sakuranbo | sup guys |
| 08:55.14 | ChannelZ | the sky |
| 08:58.21 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 08:58.48 | *** join/#asterisk jkroon (~jkroon@197.173.245.48) |
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| 09:16.47 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:18.10 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 09:18.47 | x1user | Hi, i have set up a2billing, but it seems that it doesnot log calls? Can you give me hints what might be wrong. |
| 09:21.10 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 09:21.16 | james_zhu | check the database, CDR |
| 09:21.54 | james_zhu | how do you config A with A2billing? |
| 09:24.58 | *** join/#asterisk gokulnath (~gokulnath@115.111.177.122) |
| 09:36.44 | x1user | What is CDR? |
| 09:37.38 | wdoekes2 | google it |
| 09:38.02 | kaldemar | x1user: what makes you think that it does not log calls? |
| 09:38.36 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:39.35 | x1user | the log files i've created when installling a2billing seems empty |
| 09:41.57 | x1user | http://pastebin.com/U7MJYqT0 |
| 09:43.46 | *** join/#asterisk Rufus` (Rufus@unaffiliated/rufus) |
| 09:53.15 | nunne | x1user: i dont know much about a2billing.. but seems like a weird setup not using CDR but only AGI.. |
| 09:54.42 | nunne | x1user: but even though (i'm guessing you want to log your outgoing call?) nothing will happen.. since your users context teleco doesnt call any of the agi-scripts |
| 09:56.15 | x1user | Which comes first DeadAGI or Dial, and they should be in one context? |
| 09:58.54 | james_zhu | are you sure the AGI worked? |
| 09:59.25 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:03.55 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 10:05.29 | nunne | x1user: normally you use DeadAGI after hangup on the channel. for example in conjunction with the h-extension |
| 10:05.52 | nunne | x1user: but when you make a call from your phone in the config you have supplied no AGI scripts will be executed |
| 10:05.58 | nunne | because they are in the wrong context |
| 10:06.33 | nunne | but i'm pretty sure a2billing is supposed to be setup with cdr. sounds more logical. but maybe someone else can give you an input on that since i have not worked with a2billing |
| 10:06.46 | x1user | The right context should be like this may be http://pastebin.com/6LnSm0KQ ? |
| 10:06.51 | x1user | if i got it |
| 10:10.55 | nunne | x1user: http://pastebin.com/nwQdLdze looks more correct in that case |
| 10:11.16 | nunne | but are you sure your not supposed to pass arguments to the AGI-script?? |
| 10:30.31 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 10:45.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 11:04.27 | *** join/#asterisk MoJo1760 (MoJo1760@ip72-208-78-110.ph.ph.cox.net) |
| 11:04.52 | MoJo1760 | Hello, is anyone awake at this hour? |
| 11:05.31 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 11:05.44 | MoJo1760 | I am having trouble with DAHDI not loading and I can't figure it out..this was a system that was functional |
| 11:07.32 | MoJo1760 | anyone? |
| 11:08.06 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 11:10.43 | MoJo1760 | yawns |
| 11:10.58 | MoJo1760 | I'll take that as a no |
| 11:14.18 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 11:15.07 | kaldemar | ~ask |
| 11:15.07 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:16.39 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 11:23.38 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:30.16 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 11:31.17 | *** join/#asterisk empiric (cb51daca@gateway/web/freenode/ip.203.81.218.202) |
| 11:31.43 | empiric | guys i want to setup a trunk on asterisk with spitfiretsp it gives me error |
| 11:31.57 | empiric | [Aug 17 12:29:50] NOTICE[2380]: chan_sip.c:12104 sip_reg_timeout: -- Registration for '441985377980@spitfiretsp.net' timed out, trying again (Attempt #20) > doing dnsmgr_lookup for 'spitfiretsp.net' > ast_get_srv: SRV lookup for '_sip._udp.spitfiretsp.net' mapped to host mproxy1.spitfiretsp.net, port 5060 |
| 11:32.01 | empiric | any idea? |
| 11:32.17 | empiric | i am able to ping it though |
| 11:32.51 | empiric | its behind firewall with port open 5060 |
| 11:33.08 | empiric | but i move in on live ip no firewall still same issue |
| 11:33.13 | empiric | any idea whats wrong |
| 11:36.22 | kaldemar | enable sip debug and see what the trace looks like. |
| 11:39.32 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 11:41.09 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 11:53.19 | *** join/#asterisk zokko (bbajorek@unaffiliated/zokko) |
| 11:53.21 | zokko | hello |
| 11:54.21 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:54.50 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 11:56.17 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 11:58.23 | empiric | debug shows |
| 11:58.25 | empiric | Retransmitting #3 (NAT) to 83.218.143.18:5060: REGISTER sip:spitfiretsp.net SIP/2.0 Via: SIP/2.0/UDP 10.101.1.181:5060;branch=z9hG4bK11e1a62c;rport Max-Forwards: 70 From: <sip:441985377980@spitfiretsp.net>;tag=as53d64584 To: <sip:441985377980@spitfiretsp.net> Call-ID: 2da39a971a930ec44e7db1e01626f7a3@spitfiretsp.net CSeq: 114 REGISTER User-Agent: Asterisk PBX 1.6.2.20 Expires: 120 Contact: <sip:s@10.101.1.181> Content-Length: 0 |
| 11:58.45 | empiric | keep trying to register |
| 11:58.52 | empiric | kaldemar |
| 11:58.53 | empiric | ? |
| 12:00.30 | zokko | hi |
| 12:00.49 | zokko | any1 can help me debugging why asterist doesnt want to start? |
| 12:00.58 | zokko | cant see any verbose info in the logs |
| 12:01.13 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:02.11 | kaldemar | empiric: no responses to the registration messages? |
| 12:02.52 | kaldemar | zokko: start asterisk with "asterisk -vvvc" and pastebin 10 last lines. |
| 12:03.02 | kaldemar | ~pb |
| 12:03.02 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:04.44 | empiric | ok |
| 12:04.45 | empiric | sorry |
| 12:04.51 | empiric | waht to do now |
| 12:04.56 | empiric | how to register |
| 12:05.06 | zokko | kaldemar: ok |
| 12:05.06 | empiric | Retransmitting #2 (NAT) to 83.218.143.18:5060: |
| 12:05.19 | empiric | what this mean this machine is behind fw |
| 12:06.08 | kaldemar | empiric: you seem to be behind a firewall and have not configured asterisk accordingly. |
| 12:06.11 | kaldemar | ~sipnat |
| 12:06.11 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 12:06.29 | zokko | kaldemar: seems like he indeed wants to start |
| 12:06.41 | kaldemar | empiric: what version of asterisk are you using? |
| 12:07.05 | zokko | 'sip show peers' shows peers, and i have signal in my phone |
| 12:07.13 | zokko | but from init scripts it doesnt want to start.. |
| 12:07.42 | empiric | 6 |
| 12:07.59 | kaldemar | zokko: is the init script right for your system? where did it come from? |
| 12:09.44 | zokko | kaldemar: dunno, im new here, new job, new opportunities |
| 12:09.59 | zokko | kaldemar: is there any way to launch asterisk in daemon mode from cli? |
| 12:10.16 | zokko | eg, i dont want to launch it in screen with '&' .. |
| 12:10.17 | kaldemar | did you install it yourself? |
| 12:10.51 | zokko | nope, i must use existing installation |
| 12:10.54 | zokko | :/ |
| 12:11.31 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:11.47 | kaldemar | no absolutely no output appears in /var/log/asterisk/messages when you try to start it with the init script? |
| 12:12.31 | zokko | kaldemar: nope |
| 12:12.44 | zokko | even when i've launched it from scratch no logs are appearing there |
| 12:13.05 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 12:13.07 | kaldemar | what does the messages line say in /etc/asterisk/logger.conf ? |
| 12:13.14 | kaldemar | messages => ... |
| 12:13.27 | zokko | nothing |
| 12:13.33 | zokko | only 'full' is defined |
| 12:13.49 | kaldemar | is it commented out with ;? |
| 12:13.56 | zokko | kaldemar: yes |
| 12:15.02 | kaldemar | add "messages => notice,warning,error" in logger.conf and try to start it with the init script. if you get no output at all, take a look at the init script. it may try to call the wrong binary or something. |
| 12:15.30 | zokko | this script looks like a drunk monkey wrote it. |
| 12:21.53 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.5.0/contrib/init.d/ |
| 12:24.09 | zokko | thank you, kaldemar |
| 12:27.18 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 12:32.11 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:32.42 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 12:42.13 | *** join/#asterisk AL13N_work1 (~alien@91.183.52.232) |
| 12:44.11 | AL13N_work1 | i want to have some kind of call logs (i think it's called CDR?), but i want it to be instantanious on every change, so, my question is: is there something that does what i want? if not, should i make a module or an app? what is kind of the difference? |
| 12:46.53 | *** join/#asterisk sdh (~foo@steve.st) |
| 12:50.04 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 12:51.53 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:53.56 | kaldemar | AL13N_work1: CDR does that already. you'll find call logs by default in /var/log/asterisk/cdr-csv/Master.csv |
| 12:58.13 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 12:58.18 | BenC[UK] | hi guys, I have a weird one.. |
| 12:58.22 | BenC[UK] | I have a "bad number" |
| 12:58.39 | BenC[UK] | asterisk is calling it.... I get no dial status, and it doesnt get answeered, but I see the hangup |
| 13:01.37 | BenC[UK] | I need to be able to try another carrier on it |
| 13:03.14 | BenC[UK] | http://pastebin.com/fnzvhnnF |
| 13:03.22 | BenC[UK] | is the log for it |
| 13:07.06 | AL13N_work1 | kaldemar: yes, but only AFTER the call has ended, i have an app where i plan to have immediate notifications for who is calling, which extensions are on the phone at the moment and how long they have been calling atm, even though they are still calling |
| 13:08.10 | AL13N_work1 | kaldemar: i kind of need a module or app that sees most of the changes in calls at the moment it's happening |
| 13:08.20 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:09.01 | kaldemar | AL13N_work1: AMI events are usually used for that. |
| 13:09.17 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 13:09.18 | kaldemar | AL13N_work1: AMI aka Asterisk Manager Interface |
| 13:09.32 | AL13N_work1 | hmm |
| 13:09.51 | *** join/#asterisk jkroon (~jkroon@196-215-109-237.dynamic.isadsl.co.za) |
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| 13:16.02 | p3nguin | This is not good. Two times now, asterisk crashes when I hang up the phone after taking a call. It happened from on two different phones. I guess it's time to troubleshoot the channel driver for the phones, since that's the only thing I knowingly changed recently. |
| 13:20.57 | Merlin | does anyone know how Asterisk's support for TTY/TDD works on the Zaptel driver? The Wiki has information as of 2006, which is like asterisk 1.4, but the code doesn't appear to have changed as of 1.8. i'm wondering how asterisk passes the TDD data to a SIP channel, where does it appear in the SIP packet, if anywhere? |
| 13:24.26 | *** join/#asterisk _omer (~omer@182.185.246.215) |
| 13:24.27 | _omer | hello |
| 13:24.53 | _omer | how to dial a call and play an IVR at the same time until the call is not answered. |
| 13:25.22 | _omer | any suggestions? |
| 13:26.18 | Merlin | use a queue |
| 13:28.09 | _omer | this is what came into my mind but the main picture is still not clear..... |
| 13:28.32 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 13:28.54 | Merlin | setup a local extension to dial an outboud number of your choosing, and make it a member of the queue |
| 13:29.04 | Merlin | send your calls to the queue and configure a context for the queue |
| 13:29.27 | Merlin | in the context you can configure keypresses that you want to make available while on hold |
| 13:29.43 | Merlin | and naturally you can setup hold music and a number of voice prompts that play at different times |
| 13:29.57 | _omer | hmmmmm.....yeah I can call an AGI script too.. |
| 13:30.19 | Merlin | i don't believe the keypress context for queues allows calling of an AGI script |
| 13:30.35 | Merlin | one a keypress is made though, the call leaves the queue and you can send it to an AGI |
| 13:30.45 | Merlin | but at that point you cannot reconnect it to the number that you were dialing |
| 13:30.55 | _omer | I dont need to keypress .... actually call will be dialed from a software. |
| 13:31.09 | Merlin | i have no idea what that means |
| 13:31.13 | _omer | and i need to play User Balance + Dial the call at the same time |
| 13:32.00 | Merlin | ok, forget the queue then |
| 13:32.09 | Merlin | you can't do that with a queue, unless all callers have the same balance :) |
| 13:32.12 | _omer | ok... |
| 13:32.28 | Merlin | you will need to have a threaded AGI open a new channel with the dialing number |
| 13:32.44 | Merlin | then loop through dialplan while the other channel connects |
| 13:32.50 | Merlin | then when it connects, you can join the channels |
| 13:34.22 | _omer | Never heard of threaded agi before :) |
| 13:36.10 | kaldemar | _omer: do you just want to play an announcement? |
| 13:36.36 | merlin8282 | Does someone have a good 3-way or N-way conferencing example ? |
| 13:37.04 | Merlin | _omer: a multi threaded perl script :) |
| 13:37.07 | merlin8282 | I'm trying this one http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO but ChannelRedirect does not redirect the channel immediately, and this is *not* good. |
| 13:37.07 | _omer | kaldemar: I want to play a dynamic announcement (User Balance) |
| 13:37.39 | _omer | Merlin: Do you have any example of multi threaded perl script? or Async Agi |
| 13:37.43 | kaldemar | _omer: dial option m() could do it. |
| 13:39.32 | _omer | m() executes when call is answered. |
| 13:40.01 | _omer | I think I can answer the call in context then play the IVR :) |
| 13:40.03 | kaldemar | _omer: no, it plays the musiconhold until the requested channel answers. |
| 13:40.54 | _omer | ohh .. I read M(x) |
| 13:41.01 | _omer | M(x): Executes the macro (x) upon connect of the call |
| 13:42.45 | kaldemar | merlin8282: how do you want to use it? |
| 13:43.51 | kaldemar | merlin8282: if by calling a number, the simplest way is Page(Technology/Resource[&Technology2/Resource2[&...]],d) |
| 13:44.40 | merlin8282 | kaldemar: like it's described. 1 and 2 are in a call. 1 presses *0, 2 is redirected in a free conference and 1 gets a dialtone. 1 dials 3 so 1 and 3 are connected. By pressing ** it takes 1 and 3 into the conference where 2 already is. |
| 13:45.24 | merlin8282 | So here my problem is that when 1 presses *0, 2 has no audio and is not immediately sent to the conference room. |
| 13:45.46 | merlin8282 | 2 only goes into conference only when 1 and 3 do. |
| 13:46.10 | merlin8282 | In fact it's stranger than that, because sometimes it works correctly. |
| 13:47.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mwjvfriadpvciiaj) |
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| 13:51.21 | Merlin | _omer: music on hold isn't going to be able to play a balance number that is different for every caller |
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| 13:56.16 | kaldemar | Merlin: why not? |
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| 14:07.28 | _omer | Merlin: well queue is not a good idea..in this case Each call needs to have it's own queue .... |
| 14:07.35 | _omer | 100 calls mean, 100 queues |
| 14:07.40 | _omer | :-/ |
| 14:08.09 | _omer | I can use AGI parameter with Queue cmd ... in a tricky way |
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| 14:13.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:46.03 | merlin8282 | kaldemar: I think i've solved the problem. While the called party is in the macro [macro-nway-start], ChannelRedirect() does not do anything, until the called party goes out of the macro. |
| 14:47.06 | merlin8282 | So juste after the ChannelRedirect() that redirects the caller to the conference, I did another ChannelRedirect() that redirects the called party to [dynamic-nway-invite]. |
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| 15:11.35 | *** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru) |
| 15:11.51 | catphish | i'm having trouble with func_odbc, the only error i get is "Unable to execute query [...]" |
| 15:11.59 | catphish | is there any way to debug what is happening? |
| 15:12.08 | OrNix | tcpdump |
| 15:12.53 | catphish | good thought |
| 15:13.02 | catphish | it does appear to be hitting the db server |
| 15:16.17 | catphish | no |
| 15:16.20 | catphish | i take it back |
| 15:16.27 | catphish | it's not actually reaching mysql |
| 15:17.28 | OrNix | check, if odbc.ini and res_odbc.conf are configured properly |
| 15:17.42 | catphish | http://paste.codebasehq.com/pastes/585 |
| 15:17.55 | catphish | have i some something stupid? |
| 15:18.30 | catphish | that dsn works through "isql -v asterisk" |
| 15:18.56 | OrNix | catphish, show me piece of res_odbc.conf |
| 15:19.33 | OrNix | have you described your dsn in res_odbc.conf ? |
| 15:19.36 | catphish | no |
| 15:19.38 | catphish | i haven't :) |
| 15:19.43 | OrNix | do it ;) |
| 15:20.19 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 15:20.53 | catphish | i was wondering where to define pre-connect etc. |
| 15:20.57 | Katty | hello |
| 15:21.03 | Katty | my asterisk does not work at all, how to fix plz?? |
| 15:21.05 | catphish | seems i fail at finding sensible docs :) |
| 15:21.15 | catphish | lol Katty |
| 15:21.34 | OrNix | does anybody know, how to fix WARNING Asked to transmit frame type slin, while native formats is 0x8 (alaw) ? I get it on Music On Hold when chan_local is used |
| 15:21.36 | beek | waves to Katty |
| 15:21.44 | Katty | hugs on beekers |
| 15:21.56 | Katty | what's new with you, beeks |
| 15:22.08 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 15:23.24 | beek | Not much Katty -- how about you? My big project is moving a 1.6.0 instance to 1.8.5 on another machine. |
| 15:23.34 | Katty | sounds exciting |
| 15:23.38 | beek | I live for it. |
| 15:23.50 | Katty | i'm doing a scan job/bates number thing for one of our clients |
| 15:24.04 | beek | sounds exciting |
| 15:24.08 | Katty | apparently no one else is smart enough to use adobe pro :| |
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| 15:25.02 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:26.08 | Katty | :> |
| 15:26.10 | Katty | glomps Naikrovek |
| 15:26.20 | Naikrovek | hey, you :) |
| 15:26.23 | Naikrovek | how have you been |
| 15:26.36 | Katty | nothing short of fabulious |
| 15:26.39 | catphish | OrNix: thanks for the pointer, sadly still failing with the same rather non-specific error |
| 15:26.40 | catphish | http://paste.codebasehq.com/pastes/587 |
| 15:26.42 | Naikrovek | most excellent |
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| 15:30.00 | OrNix | catphish, compare with my working config: http://paste.codebasehq.com/pastes/588 |
| 15:31.04 | catphish | OrNix: thanks, i'll give that a go |
| 15:31.25 | OrNix | and may be you should not use `` in your queries ? |
| 15:32.33 | Merlin | does anyone know anything about TTY devices and business phone systems? |
| 15:32.41 | Naikrovek | in what respect |
| 15:32.59 | Merlin | i'm trying to figure out how best to prepare an Asterisk PBX to send and receive TTY calls |
| 15:33.23 | Merlin | it appears that TTY data can be decoded in the zaptel driver, but that is buggy and old code |
| 15:33.31 | catphish | Option=3 |
| 15:33.34 | catphish | what does that do? |
| 15:33.41 | Merlin | it also appears that most tty devices are acoustic couplers |
| 15:33.45 | OrNix | catphish, i can't remember :) |
| 15:33.52 | Merlin | while ridiculous, i don't see why you couldn't use that |
| 15:33.58 | OrNix | i wrote this cfg 2 years ago |
| 15:35.02 | beek | Merlin: The last time I implemented that was about five years ago. Just used an analog card and looked for the TTY tones, then dialed a TTY as an extension. No big deal. |
| 15:35.20 | beek | The TTY had an RJ11, so on acoustic coupling required. |
| 15:35.48 | Merlin | beek: so the tty tones are trasmitted like any dtmf tones? |
| 15:35.58 | Merlin | my understanding is that they are not the same |
| 15:36.16 | Merlin | how does asterisk know how to convert the tones from analog to SIP? |
| 15:36.21 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:36.29 | beek | Yep. In fact, IIRC TTY uses a 5-level code at some gawd-awfully slow speed (I don't remember what) so accuracy wasn't a problem. |
| 15:36.35 | Merlin | 45.5K |
| 15:36.57 | beek | Those tones are inband. |
| 15:37.02 | Merlin | ah ha |
| 15:37.29 | Merlin | so what is the point of the tdd.c component of the zaptel driver? |
| 15:37.29 | beek | In my case it was analog from the POTS line to the TTY |
| 15:37.32 | catphish | OrNix: what's with the random swing from => to = in your res_odbc.conf |
| 15:37.35 | beek | Never used it. |
| 15:37.44 | Merlin | beek: it doesn't seem like anyone has haha |
| 15:37.53 | Merlin | beek: the code hasn't been touched in more than 6 years |
| 15:38.06 | beek | I wasn't interested in it doing any decoding. I was just looking for Asterisk to recognize that the inbound call was TTY and then direct it to the TTY device. |
| 15:38.12 | catphish | but if it works you're one step ahead of me! |
| 15:38.12 | beek | Kind of checking for fax tones. |
| 15:38.18 | Merlin | beek: ok, so how did you do that? |
| 15:38.45 | beek | I'd have to look again. The place where I did that is no longer in business and I don't have the code. The phone system was sold off. |
| 15:38.57 | Linuturk | WIMPy: glad you are here. I updated to 1.8.5 last night, and I'm getting the same congestion problem as last time |
| 15:39.02 | beek | Hang on and let me see if I can revive the ol' brain cells. |
| 15:39.31 | Merlin | haha thanks :) |
| 15:40.16 | catphish | OrNix: thanks, working now based on using your config, no idea what i changed though |
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| 15:41.13 | catphish | thanks |
| 15:41.27 | pabelanger | ~collectdebug |
| 15:41.27 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 15:41.36 | pabelanger | ~deadlock |
| 15:41.48 | pabelanger | ~backtrace |
| 15:41.49 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 15:42.02 | beek | Merlin: There was a function that recognized TTY... I'm looking for it. |
| 15:42.29 | Merlin | beek: k |
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| 15:44.30 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:45.10 | Merlin | i'm looking up polycom phones in the meantime and every single one has a 2.5mm headset jack that works with commercially available TTY devices |
| 15:45.12 | beek | Sorry Merlin -- I'm coming up blank. If you hang out in the channel I'll let you know what I did... damn I wish I had kept copies of that system. |
| 15:45.13 | Merlin | how cool is that |
| 15:45.23 | Merlin | i'm hanging out here, not going anywhere, but no worries |
| 15:45.27 | Merlin | i'm not implementing anything today |
| 15:45.39 | Merlin | just trying to figure out if Asterisk can meet all the Section 508 guidelines |
| 15:45.49 | Naikrovek | most polycoms have the rj6 headset connector, but the 32x's have 2.5mm |
| 15:46.03 | Merlin | true |
| 15:46.10 | Merlin | i'm looking at the 331s, which have the 2.5mm |
| 15:46.42 | Merlin | the 560 is RJ9 too |
| 15:55.44 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 15:58.20 | Linuturk | https://issues.asterisk.org/jira/browse/ASTERISK-18154 << can anyone give me some help with this bug? |
| 15:58.31 | Linuturk | I've basically got two choices |
| 15:58.50 | Linuturk | 1.8.4.4 and not being able to call certain 1800 numbers, and 1.8.5 and not having full use of my PRI. |
| 16:02.16 | pabelanger | Linuturk: option 3, find the issue that fixed your 1800 problem and backport the patch into your 1.8.4.4 version |
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| 16:45.47 | Linuturk | pabelanger: that's an option, but I have no idea where to start. I think the 1-800 issue is related to early media |
| 16:50.31 | *** join/#asterisk SantiamTech (alex@173-164-96-37-Oregon.hfc.comcastbusiness.net) |
| 16:52.01 | Merlin | beek: any idea if TTY tones can be used to respond to auto attendent prompts? |
| 16:52.09 | Merlin | i.e. be recognized as keypresses within Asterisk |
| 16:53.31 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 16:55.52 | Merlin | "Voice mail, messaging, auto-attendant, and interactive voice response telecommunications systems that require a response from a user within a time interval, shall give an alert when the time interval is about to run out, and shall provide sufficient time for the user to indicate more time is required." |
| 16:55.57 | Merlin | can you do that in Asterisk? |
| 16:56.04 | Merlin | i don't think you can |
| 16:58.37 | p3nguin | Is there a setting in asterisk to control the permissions of newly created voicemail messages? One system is creating them with -rw-r--r-- but another is creating with -rwxr-xr-x. The umask on both systems is 0022. |
| 16:58.37 | Naikrovek | yeah you can |
| 16:59.30 | Merlin | Naikrovek: was that to me? |
| 16:59.53 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:00.05 | Naikrovek | yeah, sorry. pretty sure you can, anyway. actually nevermind me, i'm not enough of an authority to tell you for sure it's possible |
| 17:00.20 | Merlin | even if you have an idea on how to do it |
| 17:00.55 | Naikrovek | well when i said it was possible, i was thinking of dialplan; anything is possible in there, but if you're talking voicemail application, you'd have to rewrite the app a bit to add functionality that isn't there |
| 17:01.12 | Naikrovek | and i don't know if the functionality you're looking for IS there |
| 17:02.02 | Merlin | gotcha |
| 17:02.20 | Merlin | for the dialplan, how could you alert a caller that a timeout was nearing the end though? |
| 17:02.50 | Merlin | change a normal Wait() to a Wait(), Beep, Wait() > |
| 17:02.50 | Merlin | ? |
| 17:03.27 | Naikrovek | i'm not super versed in how to do those things, but i know they can be done. i've heard tale of such things before |
| 17:03.40 | Merlin | ok |
| 17:04.01 | Naikrovek | if it's not super simple with unmodified asterisk i wouldn't really spend a lot of time looking at asterisk if i were you, though |
| 17:04.06 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 17:04.16 | Merlin | oh i already have asterisk :) |
| 17:04.20 | Merlin | i need to make it work :) |
| 17:04.35 | Naikrovek | gotcha |
| 17:05.08 | Naikrovek | you could always hire a consultant. leif (a mod in here) is very knowledgeable. also [tk]d-fender who hangs out in #freepbx is very knowledgeable |
| 17:05.26 | Merlin | that's a good idea |
| 17:05.43 | p3nguin | What are you waiting for when you want to beep at the caller? |
| 17:05.53 | Naikrovek | input |
| 17:06.04 | p3nguin | WHAT KIND?! |
| 17:06.05 | Naikrovek | like a menu choice or a password or something |
| 17:06.10 | p3nguin | an extension, a number for something else, a passwd, etc? |
| 17:06.26 | p3nguin | Wait() does not accept input. |
| 17:06.32 | Naikrovek | yeah |
| 17:06.44 | Merlin | p3nguin: this is a generic requirement for hearing impaired mandated by the US government |
| 17:07.03 | p3nguin | BackGround() and WaitExten() will accept input, though. |
| 17:07.06 | Merlin | p3nguin: so essentially, anywhere a keypress is required, i need to add beeps to let the caller know that the timeout is nearing and end |
| 17:07.13 | Merlin | yeah, that's what I meant |
| 17:07.26 | Merlin | WaitExten(), Beep(), WaitExten() |
| 17:07.47 | Merlin | but i would want the 2nd WaitExten() to be skipped if the first WaitExten received input |
| 17:07.52 | Merlin | that can be done |
| 17:10.34 | p3nguin | more like WaitExten(10); BackGroup(beep); WaitExten(2) |
| 17:11.29 | Merlin | there we go |
| 17:11.31 | Merlin | thanks |
| 17:11.32 | p3nguin | If you enter a valid extension during the first WaitExten(), the beep would never be heard. |
| 17:11.37 | Merlin | perfect |
| 17:13.11 | Naikrovek | see i knew it could be done |
| 17:13.14 | Naikrovek | i just had no idea how to do it |
| 17:13.15 | p3nguin | There, no hiring of consultant required. |
| 17:13.18 | Naikrovek | like most things |
| 17:13.27 | Merlin | well we still have the issue of voicemail |
| 17:14.00 | Merlin | i don't have a problem making a change similar to that at the C source code level within the voicemail app though |
| 17:14.00 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:14.11 | Linuturk | pabelanger: I think I tracked down the revision that fixes the early media issues in 1.8.5 |
| 17:14.17 | anonymouz666 | stress testing the 1.8.6-rc1 |
| 17:14.19 | Linuturk | how do i backport this to 1.8.4.4? |
| 17:14.47 | anonymouz666 | waiting for rc2 |
| 17:15.39 | anonymouz666 | 10.000 calls processed in a distributed callcenter setup, running fine so far |
| 17:15.54 | anonymouz666 | will increase sipp calls, and see how things will behave |
| 17:16.26 | p3nguin | merlin: I missed your concern about the voicemail. Care to explain it to me? |
| 17:16.26 | anonymouz666 | it's nice to see the output 'queue show' with device state in sync each other |
| 17:17.06 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:17.35 | Merlin | p3nguin: well basically, i need the same kind of tone towards the end of the keypress wait time within the voicemail system |
| 17:17.55 | anonymouz666 | leifmadsen: is there a chance that 1.8.6 rc2 will be released this week? |
| 17:18.07 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
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| 17:38.46 | Katty | grumbles. |
| 17:39.30 | rbd | hey guys...is it possible to connect a cisco IP phone with a SIP load directly to a SIP trunk service like flowroute, or do I need to go through asterisk first? I can register to a sip proxy with the phone, but I'm not sure if it can dial out of the line properly |
| 17:39.33 | *** join/#asterisk netrat (~netrat@va-69-34-33-233.sta.embarqhsd.net) |
| 17:41.30 | netrat | hello i'm using asterisk 1.6.2.20 with mostly unidata wpu-7700 wifi phones and one wpu-7800 phone. all of the wpu-7700 phones work flawlessly, but the wpu-7800 has intermittent one way audio only on outgoing calls. all of the phones are on the same network and have identical configurations. any ideas? |
| 17:41.36 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 17:41.42 | *** join/#asterisk jetlag (jetlag@pool-71-188-2-156.cmdnnj.east.verizon.net) |
| 17:43.24 | penguin | check the settongs on that particualar phone |
| 17:43.56 | penguin | if its sip check ur sip.conf for peers details |
| 17:44.21 | netrat | penguin, the sip entries are identical |
| 17:44.39 | p3nguin | rbd: If you want to use Cisco with SIP, I don't see any reason you have to have asterisk in the middle. You can set it up and point it to the ITSP directly. But without having the phone config files on a tftpd, it will take a while longer to boot up each time. |
| 17:44.49 | netrat | except for different sip extenstions |
| 17:46.02 | penguin | netrat: can you pastebin ur sip.conf |
| 17:46.14 | netrat | penguin, sure |
| 17:47.53 | beek | Merlin: I'm sure that they couldn't. I |
| 17:48.20 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 17:48.21 | penguin | netrat: which extension is the one that is having that problem |
| 17:48.38 | netrat | http://pastebin.com/DUFwjYxA |
| 17:48.50 | netrat | penguin, extension 160 is having the problems |
| 17:49.05 | beek | Merlin: I'd have to say absolutely not. The two TTY tones (mark and space) occur only one at a time. DTMF has a column and row tone that sounds at the same time. And I doubt that they're even related frequencies. |
| 17:50.12 | netrat | penguin, the problem only happens intermittently on outgoing calls. the outgoing audio works, but i cannot hear the incoming audio |
| 17:50.52 | netrat | all of the other phones work flawlessly and all phones are on the same network with no firewall between the asterisk server |
| 17:52.23 | penguin | netrat: some phones require and restrict certain codescs. That particulater phone on ext 160 might not accepts g729 so set disallow=all and replace with allow=ulaw allow=alaw |
| 17:52.48 | penguin | also set under context for ext 160 nat=yes |
| 17:53.13 | netrat | penguin, i've already tried different codecs. at first i was using ulaw then changed to g729 |
| 17:53.35 | penguin | and if the phone is behind that same network please open those ports on your router to accomadate that phone |
| 17:53.37 | netrat | penguin, i will try the nat setting, but there is no nat between it and the asterisk server. they are on the same network |
| 17:54.12 | penguin | since they are on the same network are you connecting those phones using localhost? or 192.168.X.X |
| 17:55.07 | netrat | 192.168.42.0/24 |
| 17:55.42 | cerienjean | hello - is there a trick to create templates to speed up users creation process in sip.conf - ie, having all common parameters defined once, and then instanciating username & secret ? |
| 17:56.00 | netrat | penguin, actually it's 172.16.9.0/24 |
| 17:56.50 | *** join/#asterisk eject_ck (d59ff011@gateway/web/freenode/ip.213.159.240.17) |
| 17:56.53 | eject_ck | Hi all |
| 17:57.18 | penguin | netrat: have you played around with the actual hardware phones settings? |
| 17:57.21 | eject_ck | I'm getting message - Call from 'f3W6uHeH' to extension 'f3W6uHeH' rejected because extension not found in context 'inbound'. |
| 17:57.57 | eject_ck | I have context [inbound] exten => s,1,MusicOnHold() |
| 17:58.10 | eject_ck | why it not match ? |
| 17:58.14 | penguin | eject_ck: then you have create an extension locally in that context |
| 17:58.33 | netrat | penguin, somewhat but there isn't much to really change. i've gone as far as capturing packets with wireshark, but i cannot see a difference between call with two-way audio and calls with one-way audio. even when audio is only working one way it should a rtp stream in each direction |
| 17:58.45 | eject_ck | penguin: check above |
| 17:59.04 | netrat | should a rtp stream in each direction=shows a rtp stream in each direction* |
| 17:59.43 | netrat | is there any recommended wifi voip phones for use with asterisk? |
| 18:00.18 | netrat | the wpu-7700 work great, but there were discontinued in favor of the wpu-7800 |
| 18:01.25 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 18:01.29 | jaytee | does the wpu-7800 work great too? |
| 18:02.00 | penguin | netrat: i'd try a different phone it may not be compatible with that version of asterisk |
| 18:02.06 | netrat | jaytee, i only have one wpu-7800 but am having problems with one way audio |
| 18:02.24 | netrat | penguin, they tout the phone as being SIP compliant... oh well |
| 18:03.02 | penguin | eject: have you set ur calls in the [global] section in sip.conf to point to conext=inbound? |
| 18:03.05 | netrat | penguin, i've also tried asterisk 1.8 with the same results. i think i may have a bad phone |
| 18:03.55 | penguin | netrat: thats whay im starting to think. because if all configuration is the same same and there is no firewall restricting or blocking access it must be the options or configurations within the phone itself |
| 18:04.36 | netrat | penguin, thanks for your help. do you have any wifi phone to recommend? |
| 18:04.37 | penguin | also zaptel might be missing drivers for that particular phone or tryin updating the zaptel package to see if it offer that phone. |
| 18:05.12 | penguin | netrat: unfortunatley i don't i use all hard wired PSTN phones connecting into ata adapters or channel banks |
| 18:05.29 | penguin | netrat: works the best plus audio quality is a major increase |
| 18:05.52 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 18:06.00 | penguin | netrat: then your able to config everything for each phone or line with same config details and you will get same results on all lines or phone |
| 18:06.12 | penguin | I must run guys i will be back later to help if you need it |
| 18:06.24 | eject_ck | any clue? It works when I've replaced s with _. |
| 18:06.35 | eject_ck | what's wrong ? |
| 18:07.04 | Qwell | 'f3W6uHeH' has no 's' in it. |
| 18:07.07 | Qwell | Why would it match? |
| 18:08.30 | eject_ck | Qwell: s match all, no ? |
| 18:08.34 | Qwell | no |
| 18:08.42 | eject_ck | then ok, thank you! |
| 18:09.32 | eject_ck | btw, how can I make exten pattern to match f3W6uHeH ? |
| 18:10.39 | Qwell | exten => f3W6uHeH,1,NoOp |
| 18:10.45 | ChannelZ | depends on what else you don't want to match |
| 18:11.52 | eject_ck | ChannelZ: I wat only f3W6uHeH |
| 18:12.52 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 18:13.42 | *** join/#asterisk Guest80572 (~fisted@unaffiliated/fisted) |
| 18:13.58 | trumee | i want to secure iax2 between two servers running 1.8. Is it possible to secure using TLS/SRTP without resorting to vpn? |
| 18:15.52 | Qwell | trumee: encryption=yes, done |
| 18:16.42 | trumee | Qwell: so i should generate certificates for both the servers first and copy them over like https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 18:17.17 | Qwell | That is for SIP |
| 18:17.58 | trumee | Qwell: so certificates are not needed for iax? |
| 18:18.40 | ChannelZ | eject_ck: then what Qwell said |
| 18:22.32 | eject_ck | ChannelZ: yes it works for me now |
| 18:22.45 | eject_ck | I figured that what's wrong now! Thank you al! |
| 18:22.48 | eject_ck | all |
| 18:22.59 | p3nguin | Where do people get the idea that s means everything? |
| 18:23.06 | Qwell | NFI |
| 18:24.05 | ChannelZ | I don't get 'everything' a lot, but that they think it means 'start' like the dialplan should always start there |
| 18:24.42 | Qwell | it does mean start |
| 18:24.50 | trumee | Qwell: no certificates are needed for iax? sorry if this is a stupid question, i am a newbie. |
| 18:24.51 | ChannelZ | well sort of |
| 18:25.18 | ChannelZ | more like 'start here if there is no extension' |
| 18:25.41 | p3nguin | I guess there needs to be a notation that in SIP there is always an extension. |
| 18:25.49 | p3nguin | even if it is 's' |
| 18:27.35 | ChannelZ | Yeah 's' makes more sense defined as "start" when talking about ye olde analog phones |
| 18:45.52 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 18:47.11 | trumee | Qwell: wow, i didnt realise iax2 encryption was so simple, http://www.voip-info.org/wiki/view/IAX+encryption |
| 18:47.35 | trumee | So all the fuss about TLS/SRTP in 1.8 is only for sip calls. |
| 18:50.54 | anonymouz666 | yes, IAX2 does not use RTP. |
| 18:51.04 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 18:54.37 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 18:56.11 | p3nguin | magic! |
| 18:56.11 | pzn | I'm a sip newbie... can someone help me in this concept: I need to communicate my asterisk with a siemens pabx. they told me that I should use "sip trunking" for this. I don't know what is sip trunk... can someone point me to an URL for me to learn something about it? thanks |
| 18:56.27 | Qwell | SIP trunks don't exist. |
| 18:56.44 | Qwell | Perhaps they just mean "use SIP" (hint: that's precisely what they mean) |
| 18:57.50 | pzn | Qwell, ok! thanks! a call will arrive in my asterisk, the destination extension is "800", so I should "forward" the call thro asterisk to reach siemens pabx, because ip phone extension 800 is behind siemens. |
| 18:58.12 | pzn | so what should I read/understand to be able to do this? |
| 18:58.23 | Qwell | ~book |
| 18:58.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 18:59.20 | pzn | network topology: my asterisk server has two network interfaces, one with external net, one with internal net. siemens only has internal net, so asterisk should "act as a router" to the call. |
| 19:01.33 | anonymouz666 | this book is pretty good |
| 19:01.40 | anonymouz666 | i bought it |
| 19:04.28 | p3nguin | pzn: Phones are not extensions. |
| 19:04.41 | p3nguin | pzn: Asterisk it not a proxy. |
| 19:05.23 | p3nguin | pzn: Asterisk is a Back-to-Back User Agent. Asterisk will receive a call from somewhere, and then it will call another device. Once there are two legs of the call, they will be bridged together. |
| 19:08.18 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 19:23.49 | pzn | p3nguin, sorry for the newbie questions. a sip-IP-phone is a "sip-client" and asterisk is a "sip-server". can I use this terms to explain my next question? |
| 19:24.04 | Qwell | no |
| 19:24.29 | Qwell | a phone is both. as is Asterisk. |
| 19:24.59 | pzn | Qwell, which is the right term to use in explanations of a "single-sip-call", the 2 endpoins are what? "sip-manager" and "sip-phone"? |
| 19:25.16 | pzn | I'd use FXS and FXO if it were analog phones |
| 19:25.30 | Qwell | how about "phone" and "Asterisk"? |
| 19:26.46 | carrar | woah |
| 19:27.02 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:27.05 | pzn | Qwell, ok. let me try to ask this way then... phone behaves as "phone" and asterisk behave as "asterisk", this I know how to configure. I want to ask if I can make a call between 2 asterisks, one behaving as phone and other behaving as astesisk |
| 19:27.45 | carrar | Does the handset plug into the power port? |
| 19:27.54 | p3nguin | pzn: Asterisk behaves as both client and server. |
| 19:28.07 | Qwell | pzn: SIP has no concept of that. It just doesn't matter. |
| 19:28.10 | Qwell | The question is invalid. |
| 19:28.52 | p3nguin | pzn: Many people say "asterisk server" and they aren't really using the correct term because this isn't a client-server model such as a web server and a web browser. |
| 19:29.33 | pzn | let me try again. phone1(extension700)<->network<->asterisk1<->asterisk2<->phone2(extension800) how would I configure asterisk1 to accept a call from 700->800 and "re-send" it to asterisk2? |
| 19:29.44 | trumee | Qwell, anonymouz666: is iax encryption not robust? http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.html |
| 19:29.45 | p3nguin | pzn: One asterisk can certainly call another asterisk, but neither is a phone. Let's just call them "peers" or "devices." |
| 19:29.48 | Qwell | pzn: Start by reading the book. |
| 19:29.54 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 19:29.55 | Qwell | pzn: It will all be clear. |
| 19:30.02 | atheos | pan - through a IAX2 or SIP trunk between your asterisk boxes |
| 19:30.11 | atheos | pretty easy to do |
| 19:30.19 | atheos | oops, pzn not pan |
| 19:30.26 | p3nguin | atheos: He has already been told that SIP trunks are non-existent. |
| 19:30.29 | p3nguin | Don't confuse him. |
| 19:30.34 | pzn | atheos, before someone said me that "sip trunk" does not exist |
| 19:30.44 | p3nguin | There is no SIP trunking. |
| 19:30.50 | p3nguin | ~siptrunk\ |
| 19:30.51 | p3nguin | ~siptrunk |
| 19:30.51 | infobot | extra, extra, read all about it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 19:31.37 | pzn | ~iaxtrunk |
| 19:31.38 | infobot | hmm... iaxtrunk is IAX2 trunking enables Asterisk to send media (as mini-frames) from multiple channels using a single header. The reduction in overhead makes the IAX2 protocol more efficient when sending multiple streams to the same endpoint (usually another Asterisk server). The option is 'trunk=<yes|no>' in iax.conf, and is peer configurable. |
| 19:31.38 | atheos | fine, an iaxtrunk then. |
| 19:31.48 | p3nguin | If someone says "use a sip trunk," they probably just mean "use sip," but they don't understand VoIP. |
| 19:32.16 | carrar | a collection of sip channels across a single path is now deamed as a trunk! |
| 19:32.26 | p3nguin | There goes the world. |
| 19:32.28 | pzn | p3nguin, atheos, Qwell: thanks for your answers, I'll read asterisk book and think better about all of this. |
| 19:33.02 | p3nguin | After you read the sections relevant to making and receiving calls, if there is something you don't understand, just ask here. |
| 19:33.20 | pzn | p3nguin, tks! |
| 19:33.28 | p3nguin | scowls |
| 19:34.38 | p3nguin | You try to help someone, and he or she can't even type out the other three letters in "thanks." |
| 19:35.22 | atheos | he's conserving packets |
| 19:37.02 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 19:38.03 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:38.21 | anonymouz666 | 10x |
| 19:39.45 | p3nguin | ten x |
| 19:39.51 | p3nguin | one zero x |
| 19:40.36 | BenC[UK] | Hi Guys - I have a "bad number" I can't trap the DIALSTATUS of.. |
| 19:40.36 | BenC[UK] | QAOQ |
| 19:40.41 | BenC[UK] | oops, let me get the right paste! |
| 19:40.48 | BenC[UK] | 8067984890 |
| 19:41.44 | BenC[UK] | if you call that, it gets answered saying the number has been disconnectef |
| 19:43.13 | BenC[UK] | but asterisk doesnt seem to think its connected, as it does go to the bit that normal answered calls goto (I am using AMI to originate a call,and send it to an extension) |
| 19:44.44 | p3nguin | Where does the DID come from? |
| 19:45.35 | BenC[UK] | how do you mean sorry? |
| 19:45.46 | BenC[UK] | thats is a number one of my customers has submitted to be called |
| 19:45.59 | p3nguin | If I dial that number, how does it get to Asterisk? |
| 19:46.08 | *** join/#asterisk doug (doug@breakout.telerama.com) |
| 19:46.22 | BenC[UK] | It doesnt, sorry - its an outbound call |
| 19:46.28 | BenC[UK] | my asterisk server is trying to call it |
| 19:46.55 | p3nguin | Is this a USA number? |
| 19:46.56 | doug | request '(@localphones' does not exist |
| 19:47.02 | doug | anyone know how to match parens in the dialplan? |
| 19:47.12 | BenC[UK] | yes p3nguin |
| 19:47.18 | p3nguin | doug: Did you try [(]? |
| 19:47.24 | doug | so i can paste stuff like "(800) 321-3212" and have it actually dial |
| 19:47.58 | doug | hm |
| 19:48.03 | doug | exten => _[(],1,Set(digits=${FILTER(0123456789,${digits})}) |
| 19:49.20 | p3nguin | benc[uk]: It took a long time for it to start ringing, but now it just rings and rings and rings without ever getting an answer. |
| 19:50.12 | doug | no go |
| 19:50.42 | doug | tried _[(], _[(]., [(] and a few other variants |
| 19:50.51 | Qwell | \( ? |
| 19:50.53 | doug | they show up in the dialplan, but don't appear to match. |
| 19:50.55 | doug | yeah, tried that. |
| 19:51.44 | p3nguin | Where will the pasting of such crap be performed? |
| 19:52.34 | BenC[UK] | p3nguin: hmm, when I tried in skype, It answered saying it was unavailabe |
| 19:52.45 | p3nguin | I'm calling again. |
| 19:53.21 | p3nguin | I get progress immediately, then silence for 23 seconds, now it's ringing. |
| 19:54.16 | BenC[UK] | but I dont get a NOANSWER or anything |
| 19:54.22 | BenC[UK] | at the end, just a hangup |
| 19:54.42 | p3nguin | now it stopped. |
| 19:55.12 | BenC[UK] | any dialstatus? |
| 19:55.16 | p3nguin | If you got a message saying it was unavailable, that sounds to me like you got an answer, so when it is done, a normal hangup is to be expected. |
| 19:55.32 | p3nguin | I'll do it again and watch for status. |
| 19:58.46 | BenC[UK] | Hmm, getting a CANCEL some times |
| 20:00.30 | p3nguin | It just keeps ringing. I think it is actually starting over, based on the messages I see on the CLI. |
| 20:00.36 | p3nguin | There it quit. |
| 20:00.49 | p3nguin | Status is CONGESTION |
| 20:01.17 | BenC[UK] | thats odd |
| 20:01.20 | BenC[UK] | I dont get that |
| 20:01.23 | BenC[UK] | let me try another carrier |
| 20:01.30 | p3nguin | Strange. I thought every time I got congestion, I should get the tones for it. |
| 20:02.05 | BenC[UK] | im finding a lot of things strange during this part of my testing :) |
| 20:02.19 | p3nguin | I'll call again and again to watch the status each time. |
| 20:02.25 | *** join/#asterisk oej (~olle@195.41.130.3) |
| 20:03.31 | p3nguin | It's stopping and starting over. |
| 20:03.37 | p3nguin | <PROTECTED> |
| 20:03.41 | p3nguin | <PROTECTED> |
| 20:03.51 | p3nguin | then restarts at "making progress" |
| 20:06.49 | BenC[UK] | http://pastebin.com/h8Be9icx |
| 20:07.00 | p3nguin | CONGESTION again |
| 20:07.27 | BenC[UK] | I am not getting congestion on any of my calls to it |
| 20:07.44 | BenC[UK] | but.. |
| 20:07.44 | BenC[UK] | <PROTECTED> |
| 20:07.49 | BenC[UK] | I am seeing that |
| 20:07.56 | BenC[UK] | says it exited non-zero |
| 20:08.01 | BenC[UK] | could that be my issue? |
| 20:12.40 | *** part/#asterisk mykhyggz (~col@evolone.org) |
| 20:15.56 | jaytee | every normal call I've ever seen exits non-zero |
| 20:18.22 | BenC[UK] | im so confused |
| 20:20.06 | carrar | Those damn none-zero status indicators!! |
| 20:20.11 | carrar | Who did that! |
| 20:20.51 | carrar | especially the ANSWER status |
| 20:21.38 | BenC[UK] | Ok, so thats not my issue then |
| 20:21.49 | BenC[UK] | I dont see why p3nguin is getting congestion every time |
| 20:21.50 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 20:21.54 | BenC[UK] | and I dont see any dial status |
| 20:21.59 | BenC[UK] | p3nguin: what version are you running? |
| 20:22.07 | p3nguin | version of asterisk? |
| 20:22.16 | BenC[UK] | yeah |
| 20:22.18 | p3nguin | 1.4.39.2 I think |
| 20:22.25 | BenC[UK] | Asterisk 1.8.4.4 |
| 20:22.27 | BenC[UK] | im on that one |
| 20:22.33 | BenC[UK] | maybe theres an issue |
| 20:22.47 | p3nguin | upgrade to 1.8.5.0 if you think it's possible. |
| 20:22.50 | BenC[UK] | Can I run a new Dial() from a hangup extension? |
| 20:23.00 | p3nguin | not to my knowledge, no. |
| 20:23.09 | BenC[UK] | didnt think so, that was my work around |
| 20:23.51 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 20:28.44 | BenC[UK] | nope wasnt that |
| 20:30.57 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:49.23 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:55.40 | *** join/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 20:55.57 | wizbit | is it a good idea to upgrade from 1.6 to 1.8 ? |
| 20:56.44 | p3nguin | I suppose that depends on all of your factors. |
| 20:57.02 | wizbit | its only for home use |
| 20:57.06 | wizbit | im not a power user |
| 20:57.09 | p3nguin | I'd go for it! |
| 20:57.17 | p3nguin | 1.8.5.0 is waiting for you. |
| 20:57.20 | wizbit | will 1.6 configs work with 1.8 ? |
| 20:57.24 | p3nguin | mostly |
| 20:57.39 | wizbit | or will i have a large headache and no telephony |
| 20:57.50 | p3nguin | That depends on your skill level. |
| 20:57.56 | wizbit | aye ok |
| 20:58.11 | BenC[UK] | setup a test server first |
| 20:58.17 | p3nguin | If you know what you're doing in Asterisk, I'd guess you'd have everything converted in under an hour. |
| 20:58.18 | wizbit | ok |
| 20:58.19 | BenC[UK] | get all the dialplans/config working on that |
| 20:58.32 | wizbit | aye ok |
| 20:58.49 | wizbit | is it a security risk running 1.6.1 ? |
| 20:59.36 | jaytee | wizbit, one thing to do is read the CHANGES file and the UPGRADE file for 1.8.5 That will show you what's been deprecated and what problems you might encounter |
| 20:59.49 | jaytee | the files are in the tarball |
| 21:00.34 | wizbit | ace :D |
| 21:00.47 | wizbit | feels scared and excited at the same time |
| 21:00.50 | p3nguin | wizbit: 1.6.1 isn't a version, so I can't answer you accurately. |
| 21:01.13 | wizbit | im running 1.6.1.6 |
| 21:01.18 | p3nguin | If you're asking if there were any security problems in 1.6.1 branch, sure there were. |
| 21:01.29 | wizbit | http://slackbuilds.org/repository/13.37/network/asterisk/ |
| 21:01.30 | wizbit | this |
| 21:01.33 | p3nguin | Let me see if I can find a good report on 1.6.1.6. |
| 21:02.38 | wizbit | 1.8.5.0 to 10.0.0 is a big jump? |
| 21:03.43 | p3nguin | 1.6.1 branch went EOL 2011-04-27. There were 19 fixes after 1.6.1.6. |
| 21:04.00 | wizbit | eeek i better upgrade |
| 21:04.32 | wizbit | thanks for checking :D |
| 21:04.35 | p3nguin | 1.6.1.6 was released 2009-09-03. |
| 21:04.46 | wizbit | jeeze |
| 21:04.50 | p3nguin | That's nearly an antique. |
| 21:04.54 | wizbit | :-S |
| 21:05.21 | wizbit | i will modify the slackbuild |
| 21:05.32 | p3nguin | I think 1.8 branch will serve you well, as long as you don't require any reasonable sccp support. |
| 21:05.50 | wizbit | i have a polycom phone |
| 21:05.57 | p3nguin | You'll be fine. |
| 21:05.57 | wizbit | in my room |
| 21:06.20 | wizbit | ill upgrade this weekend :D |
| 21:06.24 | p3nguin | There aren't too many serious changes in configs' syntax. |
| 21:06.30 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 21:06.57 | wizbit | p3nguin: do you use special ./configure options? |
| 21:07.16 | p3nguin | I do, yes... because I package my asterisk and want the files where the belong. |
| 21:07.37 | wizbit | i will edit the slackbuild to my needs |
| 21:07.40 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:07.43 | p3nguin | ./configure --prefix=/usr --sysconfdir=/etc --localstatedir=/var |
| 21:08.04 | p3nguin | I don't know where things will end up if I don't use those values. |
| 21:08.18 | p3nguin | I know where they SHOULD go, but I don't know if they would. |
| 21:08.19 | wizbit | sure |
| 21:08.29 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 21:09.22 | p3nguin | If you're into source install (without packagings), you could always use --prefix=/usr/local --sysconfdir=/usr/local/etc |
| 21:09.34 | wizbit | yep keeping it local is a good idea |
| 21:09.43 | p3nguin | not if you package it, though. |
| 21:09.49 | p3nguin | (some people don't understand this) |
| 21:09.51 | wizbit | the slackbuild does this |
| 21:09.52 | wizbit | http://slackbuilds.org/slackbuilds/13.37/network/asterisk/asterisk.SlackBuild |
| 21:10.24 | wizbit | installs the rc.d, man pages etc |
| 21:10.25 | p3nguin | At quick glance, it looks sane. |
| 21:10.31 | wizbit | yep |
| 21:10.34 | wizbit | ill just change version |
| 21:10.37 | Qwell | What ever happened to the good old days of slackpkgs? |
| 21:10.45 | Qwell | What's all this script crap? |
| 21:10.47 | p3nguin | It has been years since I used Slackware. |
| 21:11.12 | wizbit | Qwell: asterisk does not come with slackware, thats why i use a slackbuild |
| 21:11.16 | p3nguin | Now I do everything with makepkg and/or pacman. |
| 21:11.21 | wizbit | it keeps everything tidy |
| 21:11.48 | wizbit | slackpkg still exists |
| 21:12.00 | wizbit | slackpkg - version 2.81.1 |
| 21:12.15 | p3nguin | I'd go ahead and roll my own with slackbuild if it's easier. |
| 21:13.05 | p3nguin | Well, *I* might not, but I'm saying it's probably fine for you. |
| 21:14.01 | p3nguin | wonders if checkinstall is able to build a slackware pkg. |
| 21:16.01 | *** join/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 21:17.21 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 21:17.30 | pdtpatrick | Question .. i'm seeing this. |
| 21:17.31 | pdtpatrick | http://pastebin.com/NfeJ0fYB |
| 21:17.46 | pdtpatrick | unable to transfer.. is there anywhere I can look to see what is going on ? |
| 21:21.37 | wizbit | 1.8 source code is not here |
| 21:21.38 | wizbit | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 21:22.13 | phix | hihi |
| 21:22.18 | wizbit | oh yes it is |
| 21:22.26 | ChannelZ | http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.8.5.0.tar.gz looks there to me |
| 21:22.49 | wizbit | asterisk-1.8.5.0.tar.gz |
| 21:23.01 | wizbit | ace :D |
| 21:25.42 | wizbit | compiling :D |
| 21:31.59 | wizbit | oh dear |
| 21:32.00 | wizbit | http://pastebin.com/cb5mvRTv |
| 21:32.05 | wizbit | i think i better save this for the weekend |
| 21:34.07 | p3nguin | Do you need to make samples? |
| 21:34.15 | wizbit | nope |
| 21:34.17 | p3nguin | If not, skip it -- edit your slackbuild. |
| 21:34.19 | wizbit | ok |
| 21:34.44 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 21:41.08 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 21:42.05 | wizbit | hmm it failed again |
| 21:42.11 | wizbit | i might just install directly from source |
| 21:43.05 | *** part/#asterisk doug (doug@breakout.telerama.com) |
| 21:43.19 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tmtogtrijpabheuw) |
| 21:47.09 | p3nguin | At least check if it builds correctly. Then you'll know where the problem isn't. |
| 21:48.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:49.02 | wizbit | p3nguin: i needed to edit some more sample config stuff from the slackbuild, it might work now |
| 21:50.14 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 21:56.52 | anonymouz666 | any particular reason that the book uses the corosync 1.2.8 (the current is 1.4.1 and is giving me trouble)... but there's a chance too the 1.2.8 was the latest when the book was written |
| 21:58.52 | *** join/#asterisk jeffg (~jeffg@pdpc/supporter/professional/jeffg) |
| 22:00.02 | jeffg | slightly off-topic question: can anybody recommend an asterisk-friendly hosting provider with generally good connectivity to the UK and western europe and without gold-plated pricing? |
| 22:04.06 | wizbit | Slackware package /tmp/asterisk-1.8.5.0-x86_64-1_SBo.tgz created. |
| 22:05.43 | wizbit | Package asterisk-1.8.5.0-x86_64-1_SBo.tgz installed. |
| 22:06.03 | wizbit | Connected to Asterisk 1.8.5.0 currently running |
| 22:06.04 | wizbit | :D |
| 22:06.23 | wizbit | my voip still rings :D |
| 22:06.28 | wizbit | yay :D |
| 22:06.34 | wizbit | WOW |
| 22:06.41 | wizbit | multi colours in the cli |
| 22:08.01 | wizbit | it all seems to be working fine :D |
| 22:08.03 | wizbit | thanks :D |
| 22:08.58 | pdtpatrick | If i'm seeing a flood saying Registration failed ... TONS of from the same IP ? that means someone trying to mess with the system ? |
| 22:09.11 | jeffg | kiddies |
| 22:09.25 | jeffg | slash botnets |
| 22:09.34 | jeffg | welcome to the intardwebz |
| 22:10.47 | Merlin | can anyone recommend optimal tcp keepalive settings (timeout, interval, num. misses) for an AMI connection between two servers on the same gigabit LAN? |
| 22:11.11 | phix | pdtpatrick: tweek your firewall to stop that |
| 22:11.30 | jeffg | look into something like fail2ban or, if you don't expect to accept SIP calls from the internet at large, use a local or discrete firewall as phix suggests. |
| 22:12.10 | Merlin | pdtpatrick: or you could DoS the IP back |
| 22:12.21 | Merlin | that'll teach them |
| 22:12.30 | *** join/#asterisk WebSprocket (~WebSprock@93-97-23-210.zone5.bethere.co.uk) |
| 22:19.15 | pdtpatrick | haha Merlin |
| 22:19.20 | pdtpatrick | that'll be fun |
| 22:19.50 | wizbit | somebody is DoSing my server :( |
| 22:21.31 | phix | :( |
| 22:21.39 | phix | ring up your ISP and get them to solve it :) |
| 22:22.22 | Merlin | it's not me |
| 22:23.40 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 22:24.03 | leifmadsen | anonymouz666: yes this is a chance, but I don't be doing it, and there is likely going to be an RC2. I'm out until August 29th for my wedding |
| 22:34.09 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 22:54.54 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 22:55.08 | joesuffceren | anyone know of a good TAPI dialer extension for FireFox? |
| 23:03.30 | *** join/#asterisk vjfromgt (~vjfromgt@pool-96-224-232-165.nycmny.fios.verizon.net) |
| 23:03.52 | vjfromgt | been out of game for a long time, would like to be reminded how to check why a sip ext is not registering |
| 23:03.57 | vjfromgt | i have sip debug on |
| 23:04.02 | vjfromgt | have verbose at 15 |
| 23:04.13 | vjfromgt | still cant tell reason not registering |
| 23:04.44 | vjfromgt | <--- SIP read from UDP:192.168.10.11:5060 ---> |
| 23:04.44 | vjfromgt | REGISTER sip:192.168.10.50 SIP/2.0 |
| 23:04.44 | vjfromgt | Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK8ioa25vwxbzd64255cg1 |
| 23:04.44 | vjfromgt | From: <sip:100@192.168.10.50> |
| 23:04.44 | vjfromgt | To: <sip:100@192.168.10.50> |
| 23:04.44 | vjfromgt | Call-ID: 9b06yrs9st14dvi24u8f-0@192.168.10.50 |
| 23:04.44 | vjfromgt | CSeq: 325 REGISTER |
| 23:04.45 | vjfromgt | MAX-Forwards: 70 |
| 23:04.45 | vjfromgt | Contact: <sip:100@192.168.10.11:5060> |
| 23:04.46 | vjfromgt | Expires: 60 |
| 23:04.46 | vjfromgt | Content-Length: 0 |
| 23:04.55 | WIMPy | ~pb |
| 23:04.56 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 23:06.26 | vjfromgt | sorry |
| 23:06.59 | vjfromgt | http://pastebin.com/hJ06Z1z8 |
| 23:09.09 | *** join/#asterisk SantiamTech (alex@173-164-96-37-Oregon.hfc.comcastbusiness.net) |
| 23:12.20 | *** join/#asterisk korch (~hogemoge@ec2-204-236-228-187.compute-1.amazonaws.com) |
| 23:14.26 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:16.38 | korch | question: anyone ever see this error: "ast_read() called with no recorded file descriptor." |
| 23:18.21 | korch | i'm on 1.6.2.17, centos, using a call file, the channel is Local/s@outbound, all [outbound] does is Dial(SIP/<number>@flowroute), the extension is a custom C module i made |
| 23:18.38 | korch | i'm trying to capture the incoming audio so i can do stuff with it outside of asterisk |
| 23:19.53 | korch | there are a few parts i'm not 100% clear on in my C module, mostly i'm following the C source examples in a few of the apps |
| 23:20.22 | korch | i'm calling ast_waitfor(), and when it returns, the channel type is "null" |
| 23:20.46 | WIMPy | Maybe you should take that to #asterisk-dev |
| 23:20.51 | korch | so something in the channel setup is not working. i've been digging through main/channel.c but it's pretty complex |
| 23:20.57 | korch | ahh thx WIMPy |
| 23:21.05 | *** part/#asterisk korch (~hogemoge@ec2-204-236-228-187.compute-1.amazonaws.com) |
| 23:23.19 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:25.28 | *** join/#asterisk SantiamTech (alex@173-164-96-37-Oregon.hfc.comcastbusiness.net) |
| 23:27.25 | *** join/#asterisk DrDigi (~mmurphy@c-67-181-106-23.hsd1.ca.comcast.net) |
| 23:30.41 | pdtpatrick | Question .. if i was to do dial using dundi .. so say my box picks up 6674 and i was it to dial the ${EXTEN} using dundi .. how should i put that? Can someone give me an example ? |
| 23:31.28 | WIMPy | The easy way is to use a switch. Unless you want to do it manually. |
| 23:31.50 | pdtpatrick | in the context i've added the switch |
| 23:32.15 | WIMPy | That's all you need. The rest is just magic :-) |
| 23:33.30 | pdtpatrick | i tried this |
| 23:33.30 | pdtpatrick | http://pastebin.com/SD3yh9uZ |
| 23:34.00 | pdtpatrick | however if i take out exten => 6676 .. it fails |
| 23:34.10 | pdtpatrick | but then using this .. it says circuit is busy/congestion |
| 23:35.17 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 23:39.31 | WIMPy | If you have a matchin extension that will always take precedence |
| 23:40.03 | WIMPy | If it doesn't work withot, there must be something wrong in your dundi setuo. |
| 23:40.22 | WIMPy | ip |
| 23:40.27 | WIMPy | dmn |
| 23:40.32 | WIMPy | up |
| 23:41.13 | WIMPy | Have you tried 'dundi lookup 6674@fh_int' etc. to see if you get the right result. |
| 23:41.18 | WIMPy | ? |
| 23:41.56 | pdtpatrick | i will try that now |
| 23:42.34 | WIMPy | Err. BTW: [FROM_dundi] ? |
| 23:42.50 | pdtpatrick | ? |
| 23:43.53 | WIMPy | That looks like the context you'd include in your outgoing context. If it is the context where you put dundi calls, you will have a loop. |
| 23:45.25 | pdtpatrick | btw dundi returns no results |
| 23:45.41 | WIMPy | That's your problem then. |
| 23:45.47 | WIMPy | i.e. at the other end. |
| 23:46.52 | WIMPy | Or they aren't peered correctly. |
| 23:47.15 | WIMPy | 'dundi show peers' |
| 23:47.53 | gladier | hey guys - have you ever seen voicemail constantly think that a user needs to be setup? ie a user sets up their vm, dials the vm extension to retreive vm and get prompted to resetup their vm? |
| 23:48.45 | WIMPy | That's when the PIN equals the mailbox number, IIRC. |
| 23:50.09 | *** join/#asterisk DrDigi (~mmurphy@173-162-2-137-Stockton.hfc.comcastbusiness.net) |
| 23:50.50 | gladier | now i feel like a noob |
| 23:50.53 | gladier | thanks WIMPy |
| 23:51.25 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |