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| 09:07.50 | michael_work | hey |
| 09:11.23 | michael_work | sorry for stupid Q as i do not have too much experience with PRI and most cases dahdi works out of box :) If i have 2 cards and as i assume the dialstring of one would be standard DAHDI/i1/XXXX what is the next letter it would assign for send card? and is there way to find it out? |
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| 11:20.46 | Stefan27 | how come 'devstate change Custom:MY_DS UNKNOWN' is invalid command but 'devstate change Custom:MY_DS BUSY' works? according to https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_DEVICE_STATE UNKNOWN is a valid state? |
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| 11:41.00 | MarkS- | Hello, I have upgraded from asterisk 11.7.0 to 11.11.0 and now I have a problem with music on hold. It doesn't seem to read musiconhold.conf for classes and files. |
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| 11:42.42 | MarkS- | res_musiconhold.so is loaded |
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| 11:52.36 | tuxx- | question for the asterisk developers. when i have an ast_channel struct, is there a way to know what parent/child channels are related to the current one? |
| 11:55.13 | file | define "parent/child channels" |
| 12:02.28 | tuxx- | ok, for instance, SIP channel comes in, dials a local channel for a queue, queue dials a local channel for the agent, agent does a transfer on the local channel.. When having the local agent channel, i'd like to know if it is related to the incoming SIP channel. |
| 12:02.43 | tuxx- | hm, that is somewhat cryptic |
| 12:03.16 | youjelly | Hi, anyone worked on IVVR |
| 12:04.13 | file | then the answer is not really... you can infer a relationship for Local channels because they come in pairs and have naming as such, for the others you can only know the full path when bridged |
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| 12:04.42 | file | or in 12 and up you can use the stasis message bus to receive evnts to know what is going on |
| 12:05.12 | WHiZZi | did anyone already tried to install Asterisk 12 on CentOS 7 ? |
| 12:05.21 | WHiZZi | using the RPMs |
| 12:07.02 | MarkS- | "moh show classes" << that stays empty and musiconhold doesn't work |
| 12:07.34 | tuxx- | file: what exactly do you mean by 'full path' ? |
| 12:08.46 | MarkS- | after an asterisk restart it contains some content and seems to work |
| 12:09.16 | file | tuxx-, the path of what is connected and the relationships |
| 12:09.19 | file | tuxx-, until bridged there is none you can deduce from code |
| 12:13.31 | tuxx- | tnx file |
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| 12:53.31 | Stefan27 | Sometimes when I call a queue (about every 15th attempt). Asterisk (ver 12.3) freezes, the last thing i see is Executing [101@queue2:20] Queue("SIP/test4_phone_stefan-00000099", "queue-01,tTw") in new stack followed by a spam of warnings: [Aug 18 14:37:11] WARNING[6088]: chan_sip.c:4520 __sip_autodestruct: Autodestruct on dialog '3c2beb76b239-nqs7kgdpw38l@snom320-000413279585' with owner |
| 12:53.32 | Stefan27 | SIP/test4_phone_stefan-00000098 in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 12:53.52 | Stefan27 | is this part of a known bug? I tried googling it and found similar issues for older versions of asterisk |
| 12:56.12 | [TK]D-Fender | Stefan27: You're already a release behind, test with that... then show the full call debug |
| 12:56.20 | [TK]D-Fender | (if it continues) |
| 12:56.46 | file | and if there's not a current JIRA issue open then it's not a known bug, everything goes there |
| 12:57.51 | [TK]D-Fender | yup |
| 12:58.29 | Stefan27 | oki, full call debug means i have to do cli "set core debug 3" and paste console logs? |
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| 13:00.10 | [TK]D-Fender | Stefan27: yes |
| 13:02.28 | file | chan_sip, you are a thing |
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| 13:06.17 | Stefan27 | i see -- im browsing the jira page for the first time... Can new issues still be reported to older versions even after later versions have been released? |
| 13:08.01 | [TK]D-Fender | No. You are required to prove it still exists |
| 13:08.22 | [TK]D-Fender | Sometimes other changes inadvertently fix things people didn't notice was broken |
| 13:08.50 | Stefan27 | oki cool -- ill try localizing the problem a bit more first |
| 13:11.04 | file | yeah, we prefer not to waste time (it has happened in the past) |
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| 15:40.37 | Phunter | I have a group of users complaining of calls from certain area codes being choppy or 'digitalized' and dropping. Yet, its intermittent. Any ideas or thoughts on where to look? |
| 15:41.09 | [TK]D-Fender | Look at your calls. |
| 15:41.40 | Phunter | Ive made countless calls from my cell phone into the system and each time its fine. |
| 15:42.44 | [TK]D-Fender | Go replicate the problem and look at them |
| 15:43.15 | Phunter | Thats the point. I can't replicate it. Ive been monitoring and playing with comedia settings thinking there is a bottleneck somewhere. |
| 15:43.19 | Phunter | But I cannot spot it. |
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| 16:12.19 | ChannelZ | Maybe it's _their_ bandwidth. Or are the complainers local to the asterisk box? |
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| 16:49.06 | tenspeed705 | Hey guys, anyone know of a script (PHP preferably) to park calls? |
| 16:49.52 | [TK]D-Fender | You need to be a lot more specific |
| 16:50.38 | tenspeed705 | I want to make a button on my cisco 50XG phones, to envoke a php script that will park a call, and display the slot number on the LCD |
| 16:51.52 | [TK]D-Fender | How will you make this button on your phone? What will it actually call? |
| 16:53.37 | tenspeed705 | the Cisco 50X phones have a PSK (programmable soft key function) I want it to call the PHP script using HTTP. the PHP script (that I am looking for) would then call the AMI interface and park the call |
| 16:54.43 | WIMPy | Do you also know a way to send the information back to the phone? |
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| 16:54.48 | [TK]D-Fender | Pretty sure nothing exists off-hand specifically for that, but it wouldn't be too hard to make |
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| 16:55.32 | [TK]D-Fender | tenspeed705: And you've confirmed that the PSK is capable of making that kind of request? |
| 16:55.59 | tenspeed705 | WIMPy: the phone should read the HTTP headers coming back to it |
| 16:56.07 | [TK]D-Fender | WIMPy: If it can I presume the web-server script it'd call would respond with the result of the attempt in HTML |
| 16:56.28 | tenspeed705 | [TK]D-Fender: I have no issues doing it, I am just not sure where to start. I can read code, just not create it |
| 16:56.30 | WIMPy | It could. But I wouldn't assume anything. |
| 16:58.14 | [TK]D-Fender | tenspeed705: AMI is well documented on the WIKI and int eh book.... |
| 16:58.17 | [TK]D-Fender | ~asteriskwiki |
| 16:58.18 | infobot | asteriskwiki is, like, http://wiki.asterisk.org |
| 16:58.19 | [TK]D-Fender | ~book |
| 16:58.20 | 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 |
| 16:58.58 | tenspeed705 | im familiar with the DOCS, its the API I am not |
| 16:59.04 | WIMPy | Might be hard to find out what call to park if you have more than one. |
| 16:59.11 | [TK]D-Fender | tenspeed705: either pass it the username to lookup to find which channel they intend to part, or look it up by IP. |
| 16:59.33 | [TK]D-Fender | temAPI is documented in both and both have code samples for PHP |
| 17:00.09 | tenspeed705 | thanks alot WIMPy and [TK]D-Fender |
| 17:00.24 | [TK]D-Fender | WIMPy: more challenging perhaps... the channel state of "up" is trackable. Would have issues with 3-way calls of course. |
| 17:01.28 | WIMPy | All your calls will be "up". |
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| 17:02.10 | tenspeed705 | WIMPy: That is a good point. |
| 17:02.11 | WIMPy | I don't think there's a way to find out what call(s) might be on hold, other than tracking the state yourself. |
| 17:02.51 | [TK]D-Fender | WIMPy: Several ways |
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| 17:06.52 | tenspeed705 | why would the call be on hold in the first place. the phone is active when the PARK function is called, and then it hits the parking lot. where I can set the channel name to something unique |
| 17:09.14 | [TK]D-Fender | tenspeed705: Lets say that dealing with multiple calls at a time makes things a lot harder |
| 17:09.28 | [TK]D-Fender | tenspeed705: In terms of identifying which channel to "park" |
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| 17:10.17 | tenspeed705 | [TK]D-Fender: Sorry, I wasnt thinking that far yet. that may pose some issues |
| 17:10.40 | [TK]D-Fender | tenspeed705: You may need to think farther still, but this is an early consideration |
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| 17:13.09 | tenspeed705 | I think Ward Mundy might have done something similar for PBX in a flash. Might be worth checking out how he does it |
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| 17:15.05 | [TK]D-Fender | s/how/if |
| 17:17.06 | tenspeed705 | yea I was wrong about PIAF, its a FOP thing |
| 17:17.14 | [TK]D-Fender | WIMPy: between "core show channel X", and "sip show channels" you can pin down which are actually active.. |
| 17:17.35 | WIMPy | In this specific case, yes. |
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| 17:18.11 | WIMPy | Athhough I don't know if I wanted to parse that. |
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| 17:19.02 | [TK]D-Fender | I'm pretty sure I wouldn't want to :) |
| 17:19.07 | [TK]D-Fender | WIMPy: But ... it is there... |
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| 17:20.00 | WIMPy | But it only works because we know it's coming from a SIP phone. |
| 17:21.07 | [TK]D-Fender | WIMPy: See, sometimes the forces of the universe are not perfectly aligned against us! |
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| 17:21.50 | WIMPy | :-) |
| 17:21.58 | [TK]D-Fender | WIMPy: When he finds an analog phone with a web-enabled interface on it like that while on calls... we'll burn that bridge when we get to it. |
| 17:22.59 | WIMPy | I'm sure they exist, but I'm also sure that you couln't use it while you have a voice call. |
| 17:23.28 | WIMPy | But it should be a lot easier to find a H.323 phone that could do http. |
| 17:23.38 | WIMPy | ... while on a call. |
| 17:23.54 | [TK]D-Fender | WIMPy: I'll let you know after I'm done searching for unicorns :) |
| 17:24.42 | WIMPy | Oh, that's easy. Just drive to Hollywood. |
| 17:29.28 | [TK]D-Fender | Not likely as I don't have a passport and there's good odds I'm on several wide-sweeping lists already :) |
| 17:30.01 | [TK]D-Fender | And I could do without the involuntary colonoscopy |
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| 17:33.53 | tenspeed705 | I think im just going to use BLF for now. use the HTTP calls when I have more time to screw around |
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| 17:46.26 | [TK]D-Fender | tenspeed705: Low ROI |
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| 18:07.49 | lorenzo | hi! is there something like variables for the dialplan? |
| 18:08.00 | [TK]D-Fender | Dialplan has variables... |
| 18:08.02 | [TK]D-Fender | ~book |
| 18:08.02 | 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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| 18:08.05 | [TK]D-Fender | ^^^^ |
| 18:08.06 | lorenzo | e.g. $transfer = "Please hold on while we transfer your call" |
| 18:08.22 | lorenzo | and then pass it to a called application |
| 18:08.28 | [TK]D-Fender | "core show application set" |
| 18:10.31 | [TK]D-Fender | "called" how? |
| 18:10.57 | [TK]D-Fender | anything called via System, SHELL, etc you can pass in-line parameters to already.... |
| 18:12.04 | lorenzo | [TK]D-Fender: e.g. I have exten => 104,n,agi(googletts.agi,"Please hold on while we transfer your call",en) |
| 18:12.18 | lorenzo | since I have this repeatedly in my dialplan I'd like to have a single variable from which I can edit the text if needed |
| 18:12.50 | [TK]D-Fender | So go set a dialplan variable and reference it there |
| 18:15.54 | lorenzo | works ^_^ thanks |
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| 20:01.05 | *** join/#asterisk saxa (1000@84-255-205-164.static.t-2.net) |
| 20:01.10 | saxa | hi |
| 20:01.27 | saxa | is this a hardware problem ? |
| 20:01.43 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 20:01.50 | saxa | http://pastebin.com/89GQsw0A |
| 20:02.08 | saxa | i tried to upgrade to dahdi 2.10.0 |
| 20:02.45 | saxa | but then also downgrading and using 2.9.1 which worked , at leastup to last week |
| 20:02.57 | saxa | but now issues the same error message |
| 20:03.25 | saxa | i suppose it is due to a tdm card problem , is my assumption correct ? |
| 20:03.47 | saxa | call in / call out doesnt work at all |
| 20:04.03 | saxa | but the tdm card has the leds lit up. |
| 20:04.35 | [TK]D-Fender | What error? |
| 20:04.41 | [TK]D-Fender | Where's the failed call? |
| 20:12.08 | saxa | [TK]D-Fender: the console in asterisk gives me the following http://pastebin.com/CFZLFXjJ |
| 20:12.28 | saxa | this is an attempt to call my cellphone |
| 20:12.36 | saxa | it gives me congestion |
| 20:12.40 | [TK]D-Fender | saxa: Show us your dahdi configs and channel dumps. |
| 20:13.01 | [TK]D-Fender | saxa: We don't see that there are channels defined at all... |
| 20:16.05 | saxa | http://pastebin.com/4pyzFvnx |
| 20:16.36 | saxa | [TK]D-Fender: here ^^^ are chan_dahdi.conf dahdi-channels.conf and system.conf |
| 20:17.11 | [TK]D-Fender | Where's the channel dump from CLI? |
| 20:19.37 | saxa | http://pastebin.com/dyZ0476Q <- this is what dahdi show channels show. |
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| 20:20.30 | saxa | i did many times upgrades with this same config |
| 20:20.44 | saxa | and up to now has worked out everytime. |
| 20:21.18 | [TK]D-Fender | hrm |
| 20:21.23 | saxa | but this time the upgrade from 2.9.1.1 to 2.10.0 has not worked out |
| 20:21.55 | saxa | i have rebuilt dahdi on this machine using my usual build script. |
| 20:22.07 | saxa | same for asterisk. |
| 20:22.43 | saxa | but to me it seems like its complaining about the power on the tdm card, i never saw this message before. |
| 20:22.49 | [TK]D-Fender | Syntax looks right, channels seem to be in a legit group, * sees them.... |
| 20:22.54 | [TK]D-Fender | really not sure on this |
| 20:23.30 | saxa | yep it was working, last call in on this line i remember was last week |
| 20:23.46 | saxa | about thursday or friday |
| 20:23.58 | [TK]D-Fender | Have you restarted * since? |
| 20:24.14 | saxa | since we do use this line only for people be able to call us in |
| 20:24.20 | [TK]D-Fender | Only thing I can think of off-hand is things like "channel destroy" which kill it off... |
| 20:24.26 | saxa | i do not remember when i lastly called out. |
| 20:24.35 | [TK]D-Fender | when did you last restart *? |
| 20:24.43 | saxa | few minutes ago |
| 20:24.54 | saxa | i mean, the server was up about 30 days |
| 20:25.13 | saxa | but when i saw it doesnt work i tried to reboot the machine. |
| 20:25.29 | [TK]D-Fender | Ok, I'm out of ideas on this one so far.. |
| 20:25.33 | saxa | usually i do not reboot the machine for the upgrade, it was always worked out well. |
| 20:25.43 | saxa | thx anyway |
| 20:26.12 | saxa | I will try to see if I disassemble this machine and make a good clean up and upgrade of the OS. |
| 20:26.43 | saxa | its a quite old version of Slackware on this machine, but i have not touched it since it was working well up to now. |
| 20:27.17 | [TK]D-Fender | saxa: I wouldn't think it's the OS since chan_dahdi itslef has loaded and sees them... |
| 20:27.50 | saxa | yep, but those lines dahdi_handle_device'[7771]: add: /devices/pci0000:00/0000:00:04.0/0000:01:06.0/pci:0000:01:06.0 |
| 20:28.02 | saxa | i never saw that before |
| 20:28.11 | saxa | seems like some dbus erros |
| 20:28.46 | saxa | and what is also strange is this line |
| 20:28.52 | saxa | [A[A[A[Awctdm24xxp 0000:01:06.0: Port 4: FAILED FXS (FCC) |
| 20:29.12 | saxa | afair i never saw that |
| 20:30.18 | saxa | will try to look on the logs in more detail to see if there is something what can tell me more. |
| 20:30.24 | saxa | thanks for now. |
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| 20:44.17 | dan_j | Hi. It is possible to send an event to the AMI from a macro or dialplan applicatioon/function? |
| 20:45.06 | dan_j | Never mind. Just found it. |
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| 21:00.48 | Phunter | Sorry, had to go deploy a server.. |
| 21:00.56 | Phunter | ChannelZ: Maybe it's _their_ bandwidth. Or are the complainers local to the asterisk box? |
| 21:01.36 | Phunter | I suspect its because their internet is crap. they have 80mbps down and 15mbps up but occasionally their connection is spotty. Starts dropping packets. |
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| 21:01.46 | Phunter | They are not local to the asterisk box. |
| 21:02.10 | Phunter | Hard to troubleshoot because they don't call me and tell me, they call everyone else. |
| 21:02.19 | Phunter | So by the time it reaches me, its too late. |
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| 21:35.52 | bibz2 | guys |
| 21:36.16 | bibz2 | all my outbound calls are dropping after a few seconds |
| 21:36.22 | bibz2 | here is my sip.conf http://pastebin.com/VeR0eaFg |
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| 21:37.15 | DynamicFail | If I can ping both sides of an attempted phone call, and the signaling is getting through for a call and the call has one way audio... what could be the issue? |
| 21:37.33 | bibz2 | NAT issue probably |
| 21:37.40 | bibz2 | forwarded all ports correctly? |
| 21:37.59 | DynamicFail | all on a private network |
| 21:38.00 | DynamicFail | no NAT |
| 21:38.05 | bibz2 | oh |
| 21:45.35 | [TK]D-Fender | Describe both ends in detail |
| 21:47.09 | bibz2 | what are the prices for commercial support for asterisk? |
| 21:48.02 | navaismo | depends... country, skills, typoe of issue and the value of the person involved |
| 21:48.46 | DynamicFail | one end is a zultys phone, the other is a softphone |
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| 21:49.12 | DynamicFail | none of the audio or even dtmf events seem to be making it when one way is dialed |
| 21:49.47 | [TK]D-Fender | Check firewalls, and pastebin a complete call with SIP DEBUG enabled. |
| 21:49.50 | [TK]D-Fender | ~pb |
| 21:49.50 | 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. |
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| 22:18.42 | DynamicFail | [TK]D-Fender, What does the number after m=audio [number] RTP/AVP 0 101 mean? |
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| 22:18.57 | DynamicFail | right after there is the SIP OK message |
| 22:19.46 | DynamicFail | the port correct? |
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| 22:49.04 | ccabrera | Hello all. I was hoping any of you could guide me regarding a non-support question. |
| 22:49.14 | ccabrera | Could you point me out the contact information of the person responsible for integrating new GPL sound files into the Asterisk source code? Our company is willing to sponsor new professionally recorded spanish sounds for Asterisk with neutral latin american accent (since the ones included with Allison Smith are definitely not recorded by a native spanish speaking person). |
| 22:49.28 | ccabrera | Does anyone here know who should I contact regarding this matter? |
| 22:51.24 | navaismo | wink wink |
| 22:52.23 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Soundfiles+Submission+Process is the process for contributing sound files |
| 22:52.42 | zerohalo | ccabrera: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Soundfiles+Submission+Process |
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| 22:55.44 | ccabrera | Nice, that was exactly what I was looking for. Thank you. |
| 22:56.09 | navaismo | Hi ex-dad o/ |
| 22:56.16 | zerohalo | ccabrera: anytime - glad to see people wanting to contribute |
| 23:01.25 | ccabrera | Hello navaismo. Glad to see you keep enojoying your retirement. |
| 23:02.14 | DynamicFail | What does it mewan if you see a SIP Request Options sip:<addr> followed by a SIP Status: 404 Not Found |
| 23:02.35 | DynamicFail | packet capturing in wireshark |
| 23:04.08 | navaismo | ccabrera: only as watcher and when i have no job to do here |
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| 23:16.09 | DynamicFail | are dtmf tones not relayed by default? |
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| 23:28.08 | DynamicFail | this is so weird |
| 23:28.16 | DynamicFail | one way I can make a call and I get two way audio |
| 23:28.21 | DynamicFail | the other direction I make a call |
| 23:28.23 | DynamicFail | and I get one way audio |
| 23:28.30 | DynamicFail | things like key presses dont' seem to work |
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