| 00:11.17 | *** join/#asterisk KavanS (~quassel@LINBIT/KavanS) |
| 00:11.27 | KavanS | any suggestions for software echo cancellation for an openvox card w/dahdi? - tried mg2, and kb1 - there's no built in hardware echo cancellation on this card |
| 00:12.35 | pabelanger | KavanS: oslec |
| 00:18.47 | KavanS | any further words on oslec? |
| 00:18.52 | KavanS | it's basically the best? |
| 00:21.36 | [TK]D-Fender | yup |
| 00:35.27 | *** join/#asterisk oatha (~oatha.inf@unaffiliated/athayde) |
| 00:43.00 | KavanS | is it hackish to get it working on a modern os? - debian 7? - I will google, just wanted opinions before I dive down the rabbit hole :) |
| 00:43.36 | [TK]D-Fender | jump now |
| 01:09.52 | Greek-Boy | if my analog channels are FXO am I supposed to configure dahdi to use FXO or FXS signalling? |
| 01:13.21 | KavanS | lol thanks TK |
| 01:20.19 | [TK]D-Fender | Greek-Boy: fxsks |
| 01:22.02 | Greek-Boy | thanks |
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| 01:35.37 | Greek-Boy | if I haven't defined the incoming context in extensions.conf, the asterisk cli should still show incoming calls on the analog card right? |
| 01:49.54 | [TK]D-Fender | with proper verbose, yes |
| 01:50.15 | [TK]D-Fender | Thought nt having configured anything is not a smart idea |
| 02:06.31 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
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| 02:13.27 | pigeonflight | anyone have asterisk working with plivo as a trunk? |
| 02:13.37 | pigeonflight | looking for some guidance |
| 02:15.10 | Greek-Boy | i wish this was working http://www.itslenny.com/ |
| 02:44.19 | snadge | this ones going to be from left field.. because I love you guys. ;) |
| 02:45.39 | snadge | an example of the kind of junk that comes this way sometimes.. uniden analog handset DSS 8955, plugged into a linksys SPA112, registering to an asterisk server |
| 02:46.15 | snadge | up until 4 or 5 weeks ago, the incoming number was displayed on the uniden handset.. now says "incomplete data" |
| 02:52.45 | ChannelZ | maybe it's the government |
| 02:56.47 | floren | ChannelZ: bob i have a quick question. in sip.cong i there a way to combine 2 devices into one? for example i have [100] and [voipms], 100 is my local phone and the other is voip provider |
| 02:57.26 | floren | the thing is, i don't think is possible as i deal with 2 diff contexts |
| 02:57.54 | floren | my goal would be to have one device that hadnles both local and voipms calls |
| 02:59.46 | floren | this is how i have it set now: http://pastie.org/private/yotzrqbycoadjccmkwoxw |
| 03:00.14 | ChannelZ | well no, 100 and voipms presumably have completely different IP addresses. |
| 03:00.21 | floren | ya they do |
| 03:00.33 | floren | so i have to stick with a separate setup |
| 03:00.34 | ChannelZ | what sense would it make to combine them? IE what do you think you'd be gaining |
| 03:00.46 | ChannelZ | They're two different things. |
| 03:00.57 | floren | i'm trying to think how large setups are managed with 10-15 lines |
| 03:01.10 | floren | i created some templates that avoid the repetitive stuff |
| 03:01.22 | floren | so it is pretty compact everything in extensions.conf |
| 03:01.57 | ChannelZ | Well config is config.. it is what it is |
| 03:02.04 | floren | ya |
| 03:02.09 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 03:02.13 | ChannelZ | I have 10+ devices but it's not like I'm changing them every 5 minutes |
| 03:02.40 | floren | ineed |
| 03:02.47 | floren | indeed* |
| 03:03.01 | floren | thanks bob |
| 03:04.45 | floren | btw, is there a variable that returns the device name in extensions.conf? for example [60201] |
| 03:05.07 | floren | right now i use setvar to do that |
| 03:06.01 | floren | i think is the simplest way, [TK]D-Fender suggested this instead of using functions |
| 03:06.13 | floren | i sk this for learning purposes |
| 03:06.18 | floren | ask* |
| 03:10.14 | ChannelZ | Well I think ${CHANNEL(peername)} tells you |
| 03:10.24 | raspberrypifan | irc to sip? |
| 03:10.40 | floren | ChannelZ: i think ya |
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| 03:11.40 | ChannelZ | raspberrypifan: what does that mean? |
| 03:12.05 | raspberrypifan | idk |
| 03:12.54 | ChannelZ | floren: dunno in what context you're needing to use it, why asking the channel would be good or bad |
| 03:14.23 | KavanS | oh shit |
| 03:14.37 | ChannelZ | shits |
| 03:14.38 | KavanS | you've got irc to sip capabilities? |
| 03:14.50 | raspberrypifan | i do |
| 03:21.17 | floren | ChannelZ: bob here it is an example of a conpacted dialplan: http://pastie.org/private/4hjegkcgjfebb7krb7udea |
| 03:21.39 | floren | all i have to do is add new numbers into [inbound] |
| 03:23.29 | floren | the ${EXTENSION} is set with setvar into device |
| 03:23.40 | floren | i think is a nice workaround |
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| 04:35.50 | raspberrypifan | anyone know the primus lingo? |
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| 07:45.41 | Zogot | Ahoyhoy |
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| 07:49.43 | areay | Hi all :) I have a large list of telephone numbers and I'd like to use Asterisk to validate them (attempt a call, if it fails, mark the telephone number as invalid, otherwise hang up as soon as possible).. Does anyone know of any software capable of this? If not what would be the best way to go about it? Thanks in advance! |
| 07:51.16 | MaliutaLap | ~agi |
| 07:51.16 | infobot | [~agi] AGI is the Asterisk Gateway Interface. Similar to CGI for web applications, AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), PERL (astperl?), and other languages. See also: https://wiki.asterisk.org/wiki/display/AST/Application_AGI or http://www.voip-info.org/wiki-Asterisk+AGI |
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| 09:40.29 | hrolf | Hi #asterisk |
| 09:41.34 | hrolf | I'm getting an issue, that is a call is offered to a SIP device in queue, but as soon as it changes to ringing, the call is dropped. Why does this happens. |
| 09:42.25 | hrolf | http://pastie.org/9189051 |
| 09:42.41 | hrolf | ^^ are the logs. I had enabled DEBUG and VERBOSE. |
| 09:44.54 | hrolf | You can see in the first line, Queue application is executed, then it tries to give the call to SIP users and SIP/1003 is matched. The state for SIP/1003 changes to RINGING. Then we get message "DEBUG[28233] app_queue.c: SIP/mppl-pri-00000540: Nobody answered." |
| 09:45.07 | hrolf | and it hangs up |
| 09:45.29 | hrolf | if you see in logs it remained in RINGING state for < 1sec |
| 09:48.18 | hrolf | Any idea why this is happening? |
| 09:49.42 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 09:56.38 | hrolf | Issue is that calls only ring for 0-3 seconds |
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| 10:41.50 | *** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad) |
| 10:42.24 | fahmad | can someone tell me why calls stuck in asterisk for more then 12 hours or so ... |
| 10:42.57 | fahmad | even if someone disconnected the call but i can see the call when typing "core show channels verbose" |
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| 10:59.54 | fahmad | anyone ? |
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| 11:23.42 | pigeonflight | I’ve added an extension that points directly to a sip address … Dial(SIP/myaddress@myserver.com) |
| 11:24.05 | pigeonflight | It works (rings) but when the call is answered we can’t hear each other |
| 11:24.20 | pigeonflight | I think I’ve come across the “newbie firewall/port” issue |
| 11:24.58 | pigeonflight | I tried opening ports starting from 5060, through to 10000 using iptables |
| 11:24.59 | fahmad | it might be codec issue :) |
| 11:25.04 | pigeonflight | but that made no difference |
| 11:25.32 | pigeonflight | fahmad: oh, there’s a “newbie codec issue” too |
| 11:25.44 | pigeonflight | is having loads of fun with asterisk/sip newbie issues |
| 11:27.28 | fahmad | :) |
| 11:28.27 | pigeonflight | fahmad: what’s the fastest way to fix a codec issue? |
| 11:28.35 | pigeonflight | fahmad: just switch out different codecs? |
| 11:29.10 | pigeonflight | I think it currently uses ulaw |
| 11:31.38 | pigeonflight | it could be codec related |
| 11:33.12 | pigeonflight | my incoming sip providers seems to require ulaw (though I have only tried g729 as an alternative) |
| 11:33.58 | pigeonflight | it turns out that my target sip address only supports PCMU, PCMA, and G.722 |
| 11:35.45 | fahmad | pigeonflight: check `core show transalation` |
| 11:36.09 | fahmad | then you must have enabled ulaw & alaw into your asterisk server ... |
| 11:36.35 | pigeonflight | just enabled it |
| 11:36.36 | fahmad | g729 is proprietary |
| 11:36.51 | fahmad | and for that you need to purchase license from Digium |
| 11:37.03 | pigeonflight | output of my core show translation http://pastie.org/9189306 |
| 11:37.48 | fahmad | you do not have g729 codec installed :) |
| 11:38.10 | fahmad | nore speex, ilbc and g723 |
| 11:38.25 | fahmad | nor* |
| 11:39.03 | pigeonflight | understood |
| 11:39.25 | pigeonflight | but I do have g722 it seems |
| 11:39.43 | fahmad | yes |
| 11:40.09 | pigeonflight | but when I set my incoming provider to g722 I get “no compatible codec” |
| 11:40.15 | pigeonflight | set it back to ulaw and at least it rings |
| 11:40.20 | fahmad | yes |
| 11:40.25 | pigeonflight | but when they pick up the sip address I get silence |
| 11:40.25 | fahmad | use ulaw or alar |
| 11:40.28 | fahmad | use ulaw or alaw |
| 11:40.36 | pigeonflight | so I’m using ulaw |
| 11:40.44 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 11:40.46 | pigeonflight | but my endpoint doesn’t support ulaw |
| 11:41.06 | pigeonflight | may have some conceptual gaps with this codec thing |
| 11:42.04 | pigeonflight | fahmad: how do I know whether I’m connecting to that sip address using g722? |
| 11:43.45 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 11:43.58 | pigeonflight | can I register a single sip address as a “sip provider”? |
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| 11:44.59 | pigeonflight | fahmad: would something like this be weird? http://pastie.org/9189348 |
| 11:45.07 | pigeonflight | ^^^^ |
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| 12:25.14 | znf | Hello. |
| 12:25.39 | znf | With .call files, can I just use Application: Wait and Data: X, where X is the number of seconds I need the call to be opened? |
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| 12:25.41 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 12:51.24 | WIMPy | znf: yes |
| 12:51.47 | znf | thanks |
| 13:00.37 | pigeonflight | would something like this be weird? Is it possible to register a single sip address as a sip provider/trunk http://pastie.org/9189348 |
| 13:00.55 | bmurt | anyone have suggestions on monitoring asterisk + mysql performance |
| 13:01.10 | bmurt | basically, checking responsiveness & overall health between the two services |
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| 13:03.42 | WIMPy | pigeonflight: I don't understand your question. But it looks like your host contained an @. |
| 13:04.07 | pigeonflight | WIMPy: I may be using the wrong terminology (let me try to rephrase) |
| 13:04.23 | pigeonflight | I want to send calls to a single sip address |
| 13:04.29 | pigeonflight | e.g. myaddress@blah.com |
| 13:04.38 | WIMPy | That's what you usually do. |
| 13:04.58 | pigeonflight | but I have several sip trunks |
| 13:05.07 | pigeonflight | not including anything related to “blah.com" |
| 13:05.12 | WIMPy | But the username is the username (or extension) and not part of the host. |
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| 13:06.28 | pigeonflight | I’ve attempted the following exten => 9,1,Dial(SIP/25474152@blah.com) |
| 13:06.32 | pigeonflight | and it rings |
| 13:06.59 | WIMPy | Yes, you can do it without definig a peer. |
| 13:07.09 | pigeonflight | but when they pick up I get silence (I’ve opened the ports on my firewall) |
| 13:07.29 | pigeonflight | someone suggested that it is a codec issue |
| 13:07.38 | pigeonflight | the incoming call is treated as ulaw |
| 13:07.52 | WIMPy | Possible. You can see that if you turn on sip debug. |
| 13:07.55 | pigeonflight | my target sip address uses g722 |
| 13:08.07 | WIMPy | But if it works in the other direction it's more likely to be a network issue. |
| 13:13.24 | pigeonflight | wow… too much sip traffic, this feels like needle in a haystack |
| 13:13.37 | pigeonflight | WIMPy: anyway to filter this with something like grep? |
| 13:14.38 | WIMPy | You can filter by ip. |
| 13:15.16 | WIMPy | sip set debug ip <ip> |
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| 13:18.30 | jmls | afternoon all |
| 13:19.29 | jmls | I want to be able to queue .call files to an extension . when call #1 is done (ie the extension is no longer busy) then initiate call #2 etc |
| 13:19.31 | jmls | is this possible using .call files ? |
| 13:20.16 | WIMPy | No |
| 13:20.49 | jmls | bugger. |
| 13:20.51 | jmls | thanks |
| 13:20.55 | WIMPy | Depending on what you need exactely, you could perhaps do it in the dialplan. |
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| 13:22.21 | jmls | yeah, I was kinda thinking of having a set of .call files in a folder, and when the extension hangs up, seeing if there are any files left. If so, move the latest one to the spool folder |
| 13:23.45 | WIMPy | Call files are processed immediately. |
| 13:24.05 | jmls | not if you touch them 2-3s into the future |
| 13:24.11 | WIMPy | Unless they have a timestamp in the futur, which would then be the time they are used. |
| 13:24.31 | jmls | heh |
| 13:24.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 13:24.47 | WIMPy | But a fixed time is the only option. |
| 13:24.53 | jmls | yeah |
| 13:25.11 | jmls | however, a couple of seconds would be enough for my purpose |
| 13:25.16 | jmls | thanks |
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| 13:36.33 | fahmad | pigeonflight: sorry got dc |
| 13:36.37 | fahmad | pigeonflight: yes use it |
| 13:36.52 | fahmad | add alaw |
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| 13:38.51 | pigeonflight | fahmad: added alaw |
| 13:39.00 | pigeonflight | if I list it first it gets chosen? |
| 13:39.35 | pigeonflight | but it still doe the same thing…. transfers to my sip address and then silence when the other party answers |
| 13:41.40 | [TK]D-Fender | lack of audio is not a codec issue |
| 13:43.12 | pigeonflight | [TK]D-Fender: so it’s NEVER a codec issue? ALWAYS a routing issue? |
| 13:43.48 | [TK]D-Fender | If the 2 sides can't agree on a codec the call drops like a rock instantly |
| 13:44.18 | pigeonflight | in other words they agreed and then had a NAT/routing issue |
| 13:44.33 | [TK]D-Fender | or similar |
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| 14:00.45 | pigeonflight | when configuring a sip trunk what does the line “nat=yes” do? |
| 14:01.00 | [TK]D-Fender | in the peer itself? |
| 14:01.17 | [TK]D-Fender | because it can go there.. and it could be a [general] setting |
| 14:01.33 | [TK]D-Fender | And they seve different purposes |
| 14:01.48 | newtonr | pigeonflight, check out the sip.conf sample, there is a whole section for that option |
| 14:01.50 | WIMPy | It will give you a warning that your syntax is outdated. |
| 14:01.53 | Zogot | hey, can you trigger an ivr from the asterisk -r? |
| 14:01.55 | newtonr | WIMPy, lol |
| 14:02.17 | pigeonflight | WIMPy: the box I inherited is asterisk 1.6.2… |
| 14:02.34 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mpzcqdgstbupjhnr) |
| 14:02.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:02.37 | pigeonflight | I’m very “newbie” so some of my questions may reveal gaps and ignorance |
| 14:02.38 | WIMPy | That might be a bigger issue. |
| 14:02.40 | pigeonflight | but I’m trying to learn |
| 14:03.02 | newtonr | Zogot, that question doesn't make any sense. Do you want to originate calls from the Asterisk console? |
| 14:03.37 | pigeonflight | the sip.conf sample is hugely useful… reading through it now |
| 14:04.35 | newtonr | pigeonflight, make sure you are reading the one that came with your Asterisk version |
| 14:04.48 | pigeonflight | newtonr: yup, directly on the server |
| 14:05.17 | pigeonflight | realizes that may not guarantee that it shipped with the version of Asterisk being used now |
| 14:06.36 | newtonr | pigeonflight, you can always look at the one on SVN for that branch. http://svnview.digium.com/svn/asterisk/branches/1.6.2/configs/sip.conf.sample |
| 14:06.55 | pigeonflight | wow… subversion |
| 14:08.02 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:08.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:08.15 | pigeonflight | used to like subversion |
| 14:09.25 | pigeonflight | WIMPy: just reread your last comment. I know you’re talking about my version of asterisk but the context could be understood as my “newbieness” being the “bigger issue” :) |
| 14:10.00 | pigeonflight | how difficult is it to upgrade to the latest asterisk from 1.6.2.x? |
| 14:12.57 | [TK]D-Fender | Depends on the dialplan mostly |
| 14:15.39 | pigeonflight | [TK]D-Fender: here’s the output of my sip debug (altered to protect the innocent) http://pastie.org/9189742 |
| 14:15.53 | [TK]D-Fender | That is not SIP debug |
| 14:15.54 | WIMPy | pigeonflight: It was indeed about the Asterisk version. |
| 14:15.58 | [TK]D-Fender | We do not see a single packet |
| 14:16.20 | [TK]D-Fender | It shows literally nothing at all thanks to the masking as well |
| 14:16.27 | pigeonflight | that’s funny |
| 14:16.31 | [TK]D-Fender | "sip set debug on" <- |
| 14:16.48 | pigeonflight | [TK]D-Fender: sorry about that |
| 14:16.49 | [TK]D-Fender | and show the ENTIRE call from beginning to end |
| 14:17.01 | [TK]D-Fender | And masking is like asking for an autopsy and screwing with the evidence |
| 14:17.08 | pigeonflight | I’m claiming “newbie” privilege… I hear I can do that for at least another hour |
| 14:17.09 | [TK]D-Fender | the thing you mask *IS* what is wrong. |
| 14:17.55 | fahmad | there is one issue which i am stuck is that i can page using softphone its work but from hard phone it does not |
| 14:18.23 | [TK]D-Fender | fahmad: that description doesn't mean anything to us. Show us the 2 calls so we can compare |
| 14:18.32 | [TK]D-Fender | ~pb |
| 14:18.40 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:18.41 | [TK]D-Fender | ^^^^^ |
| 14:20.16 | pigeonflight | [TK]D-Fender: still working out how to show just that call, I though the sip set debug ip X.X.X.X would have done it |
| 14:20.34 | [TK]D-Fender | There are 2 sides to every call |
| 14:21.03 | [TK]D-Fender | Don't make assumptions about where the problem is. That's the fastest way to start running blind. |
| 14:25.12 | fahmad | [TK]D-Fender: got it working now i used only one digit :P |
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| 14:25.32 | Zogot | newtonr: yep, wrong window :p . was testing a doorbell into an ivr |
| 14:25.42 | Zogot | newtonr: didn't wanna keep asking the client to go press the doorbell |
| 14:25.59 | Zogot | newtonr: turns out it was a picnic error, the dial extensions where wrong :p |
| 14:28.13 | Zogot | if anyone is using php and has aastra phones btw, im working on our own version of the xml support for aastra, its on our github at http://github.com/clearvox |
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| 14:33.32 | pigeonflight | [TK]D-Fender: hopefully this is more helpful http://pastie.org/9189778 |
| 14:35.55 | newtonr | Zogot, cool |
| 14:37.20 | Zogot | newtonr: yeh having a lot of fun with this. Also got to see the 3 new aastra 6800 series phones, added provisioning support for them in our system last week |
| 14:37.22 | pigeonflight | [TK]D-Fender: line 86 looks important “Got SIP response 603 "Decline" back from 54.241.2.206” http://pastie.org/9189778#86-87 |
| 14:38.07 | [TK]D-Fender | that is not an entire call |
| 14:38.48 | [TK]D-Fender | And you're still masking the important bits |
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| 14:44.19 | [TK]D-Fender | mind you the only bit we see is a call that isn't getting answered at all. |
| 14:44.31 | [TK]D-Fender | And doesn't match the previous description of a call with no audio |
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| 16:22.19 | Kobaz | anyone have any issues where you pick up a ringing polycom phone, and the polycom drops he call and sends asterisk a DECLINE |
| 16:22.47 | bmurt | anyone have suggestions on monitoring (and graphing) asterisk + mysql performance? basically, checking responsiveness & overall health between the two services.. |
| 16:23.12 | Kobaz | bmurt: munin |
| 16:23.25 | anonymouz666 | munin? |
| 16:23.32 | Kobaz | munin, yeap |
| 16:23.34 | bmurt | that'll provide details between asterisk <=> mysql? |
| 16:23.45 | Kobaz | depends what you write code for |
| 16:23.54 | Kobaz | it can graph and track anything |
| 16:24.03 | Kobaz | it has plugins for asterisk and mysql |
| 16:24.17 | anonymouz666 | this sounds GREAT |
| 16:24.20 | litn | hey guys, if I reload my config/dialplan/extensions, will that drop any calls that are currently going on? |
| 16:24.24 | Kobaz | so with a little glue you can write something that tracks the two working together |
| 16:24.30 | pabelanger | Kobaz, not here |
| 16:25.31 | Kobaz | pabelanger: on this same customer, people are complaining about calling in like 5 times and getting disconnected |
| 16:25.36 | bmurt | yeah, im trying to research & monitor performance between the two |
| 16:25.39 | bmurt | queries, etc. |
| 16:26.11 | Kobaz | bmurt: bmurt: you'll have to take a timestamp before and after the query in asterisk and then write it in a log in munin format and then it can graph it for you |
| 16:26.17 | [TK]D-Fender | litn: no |
| 16:26.20 | Kobaz | (or any other rrdtool style graphing app) |
| 16:26.24 | bmurt | ya |
| 16:26.27 | bmurt | ok |
| 16:26.39 | bmurt | didnt know if something already existed |
| 16:26.50 | Kobaz | you want to store the time differences, basically query duration from asterisk's point of view |
| 16:27.13 | Kobaz | bmurt: it's really instance-specific |
| 16:27.17 | pabelanger | Kobaz, DND or something like that? |
| 16:27.26 | Kobaz | pabelanger: doesn't appear to be |
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| 16:27.39 | Kobaz | like, i called an extension, she went to pick it up, hit line 1 |
| 16:27.49 | bmurt | asterisk by default doesn't log a separate time entry |
| 16:27.53 | bmurt | basically just the timestamp |
| 16:28.01 | Kobaz | my call went to voicemail and i saw in asterisk that sip/203 sent a decline |
| 16:28.14 | pabelanger | sure they are hitting line 1 and not something else? |
| 16:28.25 | pabelanger | any external headsets attached? |
| 16:28.29 | Kobaz | i've seen this happen with old old polycom sip firmware |
| 16:28.37 | Kobaz | but this phone's running 3.3.4 |
| 16:28.39 | anonymouz666 | I have been tracking issues with AMI just because got slower due the nature of mysql and sync operations if your database got slow, AMI events could happen with delay |
| 16:28.41 | pabelanger | Hmm |
| 16:28.45 | pabelanger | we are 3.3.4 too |
| 16:28.47 | pabelanger | never heard of it |
| 16:28.59 | Kobaz | like in 2.2.x polycoms would routinely sent decline or otherwise reject a perfectly good call |
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| 16:32.18 | Kobaz | so anyway |
| 16:32.20 | Kobaz | the other issue |
| 16:32.28 | Kobaz | fun monday issues |
| 16:32.37 | Kobaz | i love when customers say "this has been happening for 2 months!!!" |
| 16:32.42 | Kobaz | and this is the first time i hear about it |
| 16:32.50 | anonymouz666 | heh. |
| 16:33.03 | Kobaz | the other issue is asterisk picks up the call, rings some phones, and then 2-3 seconds later i get a hangup (carrier-side) |
| 16:33.20 | anonymouz666 | that happens all the time |
| 16:33.21 | Kobaz | people calling in... who do call back in and complain... say they got hung up on |
| 16:33.47 | Kobaz | the carrier says it's us |
| 16:33.51 | Kobaz | and i say it's the carrier |
| 16:34.29 | anonymouz666 | you have to prove that your system works. |
| 16:34.39 | Kobaz | i just checked my pri dump |
| 16:34.40 | anonymouz666 | even if it's telco fault |
| 16:34.51 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 16:34.52 | Kobaz | 100% of the calls since i turned on debug are 16 normal clearing |
| 16:35.02 | Kobaz | anonymouz666: well yeah |
| 16:35.11 | Kobaz | anonymouz666: i have a long history of proving our system works |
| 16:50.31 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:51.02 | anonymouz666 | imagine a country that has more mobile phones than people. it's total chaos. We have to prove what is happening with the calls that even reach our system! |
| 16:51.48 | pigeonflight | what does the 40 in this line mean? exten => 7270,1,Dial(SIP/7270,40) |
| 16:51.56 | pigeonflight | I’ve also seen exten => 7270,1,Dial(SIP/7270,45) |
| 16:52.08 | anonymouz666 | ring time |
| 16:52.16 | pigeonflight | ah… seconds? |
| 16:52.19 | anonymouz666 | yes |
| 16:52.27 | pigeonflight | anonymouz666: thanks |
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| 16:56.44 | [TK]D-Fender | pigeonflight: "core show application APPLICATION_NAME", "core show function FUNCTION_NAME" |
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| 17:07.31 | btracht | where are call logs stored? |
| 17:09.31 | [TK]D-Fender | CDR |
| 17:09.42 | [TK]D-Fender | well explained in the book... |
| 17:09.43 | [TK]D-Fender | ~book |
| 17:09.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 17:21.14 | btracht | where can i find the log through the cli |
| 17:23.22 | *** join/#asterisk pigeonflight (~pigeonfli@72.252.224.80) |
| 17:24.10 | newtonr | btracht, https://wiki.asterisk.org/wiki/display/AST/Directory+and+File+Structure most logging or reporting will be by default going to /var/log/asterisk/ or a sub-directory |
| 17:24.22 | newtonr | depends on your configuration of course |
| 17:24.41 | [TK]D-Fender | btracht: "cdr.conf <- |
| 17:25.02 | [TK]D-Fender | If you're looking for the "call list" |
| 17:25.18 | [TK]D-Fender | If you want actual call-flow, then that's defined by logger.conf, and asterisk.conf |
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| 17:27.52 | btracht | [TK]D-Fender, @newtonr: thanks |
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| 17:40.28 | Vardan | Hi |
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| 17:45.34 | pabelanger | hello? yes, dog here |
| 17:46.25 | whaletern | hello dog this is whale |
| 17:48.46 | matthew-moretalk | hello all |
| 17:49.13 | file | hi |
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| 18:30.56 | litn | hey guys... when I'm on the phone with someone and they hang up, my leg doesn't hang up and I hear a busy tone |
| 18:31.04 | litn | anyone know what's going on off the top of their head? |
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| 18:50.43 | Vardan | hi all |
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| 18:51.20 | [sr] | ai |
| 18:54.30 | Vardan | people is that possible to create asterisk application which will allow to have many clients connected with one admin, but clients only send video/voice and don't see admin, so one way video/voice connection to one user(admin)? |
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| 19:02.52 | Vardan | hello??? |
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| 19:15.20 | ChannelZ-Wk | OH HAI |
| 19:16.24 | Vardan | hi |
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| 19:19.45 | ipalmer | Hi all, does anyone know if there's a limit to how long a variable can be, I'm passing a variable with a csv of column names and a variable with a csv of values into a GoSub but seem to be having problems which I can't quite pinpoint |
| 19:26.05 | Vardan | how can I have conference where all speak and one person see them all? |
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| 20:03.55 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:03.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 21:03.47 | jameswf | ipalmer: NOOP your variable after you set it to see what it is |
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| 22:57.03 | devwork_ | Hi I got a soundstation ip 6000 cheap @ auction. I want to put the latest software on it and mess around with it on *. I see firmware releases saying "Polycom UC Software" and other sets ( look older ) with SIP 3.X.X, which do i want? |
| 22:58.37 | WIMPy | wiki.polycom.com? |
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| 22:59.08 | devwork_ | 404? I've been reading documents all day on it. |
| 23:00.20 | devwork_ | I think the polycom unified communications stuff is a proprietary system or turnkey or whatever. |
| 23:00.37 | devwork_ | idk if its in the phone only, or you need the UC part if you have their stuff. |
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| 23:30.24 | pabelanger | devwork_: http://voipt2.polycom.com/ |
| 23:30.29 | pabelanger | you want 3.3.4 |
| 23:31.42 | devwork_ | Awesome, so I can get this boot files and load them from http using bootp parameters. |
| 23:31.57 | pabelanger | yes |
| 23:32.04 | devwork_ | sweet, so that UC stuff, basically, is what I thought it was, a polycom solution which i don't have/want |
| 23:32.06 | pabelanger | site explains it |
| 23:32.16 | pabelanger | well, they changed the way the did firmwares |
| 23:32.23 | pabelanger | cannot remember what UCS did |
| 23:32.27 | pabelanger | but 3.3.x |
| 23:32.27 | devwork_ | yea i have a provisioning server setup already. |
| 23:32.36 | pabelanger | which http://voipt2.polycom.com/335/ is the latest I guess |
| 23:32.41 | pabelanger | we still run 3.3.4 every where |
| 23:32.47 | devwork_ | i don't know what version these are, but I can't get into the web interface. |
| 23:32.56 | devwork_ | I just did a reset using *68 |
| 23:33.39 | pabelanger | you might need a 2 stage upgrade |
| 23:33.45 | pabelanger | but, format FS |
| 23:33.47 | pabelanger | then use 3.3.4 |
| 23:33.52 | pabelanger | if that does not work |
| 23:33.56 | pabelanger | you'll need 2.2.x or something to step up |
| 23:34.20 | devwork_ | Alright. I read the docs re: the web interface, I get a web login popup, all the docs seems to have like a "select user/admin button then enter password" |
| 23:34.37 | pabelanger | 456 is default password |
| 23:34.40 | pabelanger | Polycom is user |
| 23:35.18 | devwork_ | hmm yea im just not getting in with 456 |
| 23:35.39 | pabelanger | you don't need the web |
| 23:35.48 | pabelanger | you should be able to do everything from phone |
| 23:36.12 | devwork_ | ok. ( not physically there atm ) |
| 23:36.23 | devwork_ | but I got the xml setup |
| 23:36.40 | devwork_ | wasn't sure what it wanted for sip address, I just put sipuser@sipserverip |
| 23:37.12 | devwork_ | going to add a sip.conf entry now and see if it picks up the config files |
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