| 00:00.26 | sectech | Everyone who messages me on office messenger is going to get a SIP error message. |
| 00:00.36 | sectech | Why? because it's Friday... |
| 00:00.37 | sectech | and.. |
| 00:00.49 | sectech | I'm trying to remember what all the errors are. |
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| 07:36.12 | ChannelZ | So silent. It's like a Saturday in here. |
| 07:37.11 | WIMPy | Friday was cancelled. |
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| 08:21.02 | wdoekes | haha |
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| 08:41.49 | ChannelZ | I'm considering it |
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| 09:47.39 | bsdice | Morning |
| 09:48.17 | bsdice | Question: I need a channel to dial that does nothing, just ring and that's it. No answer, nothing. How to do in 11? |
| 09:50.07 | whizzi | bsdice: make a dial plan entry which only has ringing() ? |
| 09:50.28 | whizzi | in extensions.conf |
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| 11:04.46 | wasanzy | hello |
| 11:05.31 | wasanzy | does IVONA supports asterisk 11.5.0? |
| 11:05.52 | skrusty | via unimrcp |
| 11:06.02 | skrusty | im doing a setup at the moment with ivona and asterisk |
| 11:06.37 | skrusty | not got it working 100% yet mind you, but yeah, the generally accepted way from ivona is mrcp, and unimrcp has a nice app for asterisk to make it work |
| 11:06.50 | skrusty | in fact, im trying unimrcp for both tts and asr |
| 11:06.52 | wasanzy | can u guide me please? am looking at http://www.voip-info.org/wiki/view/IVONA+Asterisk+Module bu not helping |
| 11:08.41 | wasanzy | skrusty: Where do I get those to install? |
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| 11:09.28 | bsdice | whizzi thanks, I will try that |
| 11:12.55 | wasanzy | hello |
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| 11:17.10 | skrusty | wasanzy: google "unimrcp asterisk" |
| 11:17.19 | skrusty | there's a few guides on the unimrcp website |
| 11:17.35 | skrusty | plus ivona should give you some help getting the unimrcp config right for use with their software |
| 11:17.51 | skrusty | i assume you have the unimrcp version of ivona tts? |
| 11:17.55 | wasanzy | skrusty: but did you see the link I sent you? |
| 11:17.56 | skrusty | mrcp sorry |
| 11:18.11 | skrusty | they dont support it any more |
| 11:18.35 | skrusty | it's not the way they would support asterisk integration |
| 11:18.59 | wasanzy | skrusty: you mean ivona doesn |
| 11:19.12 | wasanzy | 't support it anymore? |
| 11:19.58 | skrusty | that's correct |
| 11:20.27 | wasanzy | ok let me search for unimrcp asterisk and see |
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| 11:22.31 | Farkie | Hey - I'm using Asterisk 1.6.1.18 - but i'm getting one way audio - it looks like it's a NAT issue - but I'm failing to see where. We are using nat=yes, qualify=5000 and using extip/host |
| 11:22.53 | Farkie | firewall is open on 10000-20000 udp |
| 11:22.59 | whizzi | Farkie: tcpdump for the win |
| 11:23.13 | Farkie | but it's udp? |
| 11:23.29 | whizzi | unless you configured different, yes |
| 11:24.23 | Farkie | what am I grepping for? |
| 11:24.32 | Farkie | it's quite noisy :P |
| 11:24.36 | kaldemar | Farkie: what is configured as nat=yes? |
| 11:24.59 | Farkie | well it's in sip_nat.conf and sip.conf |
| 11:25.18 | kaldemar | not file name, but as in [general] or the peer? |
| 11:25.44 | kaldemar | and which way does the audio go? |
| 11:25.55 | Farkie | they can hear us |
| 11:25.58 | Farkie | lemme check |
| 11:26.08 | Farkie | it's in general |
| 11:26.19 | Farkie | we are using gradwell as trunk |
| 11:26.29 | Farkie | probably doesn't help, but more info etc |
| 11:26.34 | kaldemar | what is "us"? a phone in the same subnet as asterisk? what is "they"? |
| 11:26.57 | Farkie | sorry |
| 11:27.16 | kaldemar | "they" is a provider and you don't get audio in? sounds like an issue in your setup. sip debug will tell. |
| 11:27.53 | Farkie | our setup is: gradwell trunk -> router -> smoothwall -> our asterisk server -> 10 phones internal |
| 11:28.19 | Farkie | one of our lines is a BT line, if that's busy it goes to the SIP line |
| 11:28.33 | Farkie | (we have a DAHDI card) |
| 11:28.50 | whizzi | make sure in your firewall you have UDP 10000-20000 INCOMING open to your Asterisk machine |
| 11:28.52 | Farkie | it's only silent on the second call |
| 11:28.57 | Farkie | yep it does |
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| 11:30.56 | whizzi | only second time? |
| 11:31.03 | whizzi | what kind of router are you using? |
| 11:31.05 | Farkie | we always hear sound on the first call |
| 11:31.20 | whizzi | and is there a SIP Helper or SIP_ALG active? |
| 11:31.22 | wasanzy | skrusty: I don't have the unimrcp version of ivona tts can you please help me out? |
| 11:31.35 | Farkie | technicolor one 582n |
| 11:32.36 | Farkie | I don't believe so |
| 11:32.56 | Farkie | DMZ is to smoothwall |
| 11:34.05 | whizzi | tcpdump combined with wireshark is your friend I guess |
| 11:35.14 | whizzi | or set the DMZ temporary to the Asterisk server |
| 11:35.50 | whizzi | perhaps the technicolor is sending the RTP to the DMZ |
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| 11:37.46 | skrusty | wasanzy: you'll need to get it from ivona |
| 11:37.47 | skrusty | sorry |
| 11:38.37 | wasanzy | is this the best way to get it working? http://www.lumenvox.com/knowledgebase/index.php?/article/AA-01129/0/2.-Installing-and-Configuring-UniMRCP-Modules.html |
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| 11:56.44 | skrusty | wasanzy: it's the best way in terms of getting support from ivona |
| 11:58.11 | wasanzy | skrusty: am a little confused, why will I need LumenVox server? |
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| 12:35.21 | Farkie | there we go |
| 12:35.27 | Farkie | whizzi, It was TCP 5060 was blocked |
| 12:36.07 | whizzi | strange ... tcp5060 is only needed for signalling |
| 12:36.38 | whizzi | beware if you have opened tcp5060 from outside to your server, you need very strong passwords |
| 12:37.55 | eirirs | I have no strong passwords on my port 5060 outside :D |
| 12:38.11 | eirirs | but guess that ACL restricting to only ISP's pbx are allowed to connect to my port 5060 does that trick :P |
| 12:38.54 | Farkie | yup |
| 12:38.57 | Farkie | we have that |
| 12:39.12 | Farkie | I thought 5060 tcp was a weird one to open though |
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| 12:59.05 | skrusty | skrusty: i don't know, why would you? |
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| 13:46.41 | ddickenson | hello all |
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| 13:50.00 | ddickenson | I've been struggling with an issue where we need to have 3 phones outpulse a different callerid than the rest of the facility. Any ideas how to do that? My first thought, that worked pretty well was to put them in their own context and then have pattern matching and set the callerid there. This, however, for some reason makes my Polycom 650s not allow call waiting where when I change only the context to the main users context I |
| 13:50.00 | ddickenson | <PROTECTED> |
| 13:52.13 | ddickenson | scratch that. it will allow call waiting but not in the queue. As I said, I |
| 13:52.31 | [TK]D-Fender | Nothing in the dialplan has any impact of a SIP phone allowing multiple calls or not |
| 13:52.33 | ddickenson | I've only changed the context in sip.conf and it changes |
| 13:52.42 | ddickenson | That's waht I thought |
| 13:53.03 | ddickenson | At first I thought it was "ringinuse" in queues.conf |
| 13:53.49 | [TK]D-Fender | As for callerid, either put them in different contexts, or check who the caller is in the extensions you use to dial out, or set a variable in your peer as to the # to present and check it in your dialout extensions |
| 13:54.08 | [TK]D-Fender | Or make a lookup of some kind (DB, etc) |
| 13:54.18 | [TK]D-Fender | In short.. any way you want |
| 13:55.15 | ddickenson | ok, I guess that's more what I need to learn. I was looking into some sort of "if" statement to check the callerid of the set doing the dialing but wasn't able to make that work. |
| 13:56.51 | ddickenson | I have two contexts, longdistance, and local. They both set the outgoing callerid to the main facility clid as they match for cases |
| 13:56.56 | [TK]D-Fender | maybe work looking at that and fixing it then |
| 13:58.43 | ddickenson | IF(<expr>?[<true>][:<false>]) This is what I've found on voipinfo but is there a way to just check everything that enters that context and if it matches the clid I want send it onto the other context? |
| 13:59.02 | rrittgarn | when in a A -> B call, If i use the AMI to redirect to a third party (C) I don't get any CDR entries. Is this a known bug, or would more information be helpful to ye almighty devs? |
| 13:59.31 | ddickenson | I know there's a way to pattern match everything _X. but is that a good idea, would it screw up other stuff? I'm on a production system so I can't be trying too much without knowing the consequences |
| 13:59.47 | eric_hill | ddickenson: GotoIf($[${CALLERID(num)}!=123]?regularcall,specialcall) |
| 13:59.55 | [TK]D-Fender | " just check everything that enters that context" <- very vague |
| 14:00.12 | [TK]D-Fender | You have to do this in each pattern you want to check... which is pretty much all of them |
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| 14:00.23 | ddickenson | dang... that's what I was afraid of |
| 14:00.24 | [TK]D-Fender | Sometimes this means you should reconsider how you are dialing out... |
| 14:00.32 | [TK]D-Fender | this is a tiny job really. |
| 14:00.37 | ddickenson | that's also what I was thinking about |
| 14:01.16 | ddickenson | I've been building systems for a while now but sometimes small issues make me wonder how close to "best practices" I am... |
| 14:01.18 | [TK]D-Fender | I gave you 3 ways. One of them requires 1 line per "exception phone"'s entry, and 1 line of dialplan per pattern |
| 14:02.00 | [TK]D-Fender | There is no such thing as "best practices" really. This is "coding" effectively. |
| 14:02.16 | [TK]D-Fender | Its a question of how efficient you can make it to your needs. |
| 14:02.26 | [TK]D-Fender | Sometimes you can use lazier approaches than others |
| 14:02.54 | ddickenson | I know there's no one way to do things just like in writing software but most would also agree that using tons of recursion or something isn't exactly the best way to do something |
| 14:03.09 | [TK]D-Fender | If it works, it works, there may simply be more efficient, cleaner and readable ways of doing it. It's up to you |
| 14:03.23 | eric_hill | Coding is the art of using the least code possible to accomplish a task. |
| 14:03.52 | [TK]D-Fender | ddickenson: If you've got any programming chops at all... just read the syntax * offers you... I already described what would be the smallest way. |
| 14:04.23 | [TK]D-Fender | eric_hill: While keeping an open eye for future requirements |
| 14:04.31 | ddickenson | thanks |
| 14:04.49 | eric_hill | Indeed. |
| 14:05.00 | ddickenson | If i weren't struggling with it I wouldn't be here. I have a few ideas to run with i think now though. |
| 14:05.10 | [TK]D-Fender | [09:01][TK]D-FenderI gave you 3 ways. One of them requires 1 line per "exception phone"'s entry, and 1 line of dialplan per pattern |
| 14:05.20 | [TK]D-Fender | ddickenson: your clue : SetVar |
| 14:05.30 | [TK]D-Fender | ddickenson: run with it a little |
| 14:05.35 | ddickenson | will do, thanks |
| 14:10.18 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qkqajbkiflobiuju) |
| 14:10.19 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:10.55 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 14:13.03 | ddickenson | ok, I think I see it. sip.conf I can setvar=except=true then in the pattern IF(except?Set${CALLERID(num)=11111111:Set${CALLERID(num)=2222222) |
| 14:13.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:13.31 | ddickenson | *set the variable only on those phones I need that is |
| 14:13.34 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 14:14.26 | ddickenson | of course I didn't close my braces... ha |
| 14:16.50 | eric_hill | ddickenson: ${FOO} expands foo into a value |
| 14:17.15 | eric_hill | ddickenson: Set(FOO=bar) puts "bar" into the variable FOO |
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| 14:17.59 | eric_hill | ddickenson: $[${FOO}==bar] expands to $[bar==bar] which evaluates to 1 |
| 14:18.17 | eric_hill | ddickenson: $[] is for evaluation, ${} is for expansion. |
| 14:18.28 | eric_hill | ddickenson: It can get ugly without syntax highlighting. |
| 14:19.54 | ddickenson | eric_hill: I see so ( ) is like = and { } is like == |
| 14:20.25 | ddickenson | so if I'm using Set(CALLERID(num)=1111) |
| 14:21.00 | ddickenson | sorry, that's not right, {} isn't a comparision just for viewing the value inside a var? |
| 14:21.12 | eric_hill | Set(CALLERID(num)=1111) is correct. |
| 14:21.36 | eric_hill | Set(${CALLERID(num)}=1111) expands to Set(1234=1111) which is *not* correct. |
| 14:23.40 | ddickenson | yeah when you put it like that it makes sense |
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| 14:24.27 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 14:25.21 | ddickenson | ok. but otherwise that should work? setting the variable as part of the sip.conf entry and then checking against it and setting the callerid accordingly? If I do that comparision do I need to list an option for false or can it just fall through to the next priority? |
| 14:25.39 | *** join/#asterisk Ice_Strike (~Ice_Strik@cpc1-oldh7-0-0-cust772.10-1.cable.virginm.net) |
| 14:26.16 | eric_hill | Sip.conf is for defining your peers and phones, not the dialplan. Extensions.conf is for your dialplan. |
| 14:26.17 | ddickenson | like IF(except?Set(CALLERID(num)=1111111) |
| 14:26.28 | wdoekes | ddickenson: yes |
| 14:26.37 | Ice_Strike | I suspect there is a problem with the network.. the calls keep muting for like a second a few time during a call. It happen to other PBX boxes as well |
| 14:26.44 | wdoekes | ExecIf(except?Set(CALLERID(num)=1111111) |
| 14:26.52 | wdoekes | ExecIf(except?Set(CALLERID(num)=1111111)) |
| 14:26.56 | ddickenson | ahhh |
| 14:27.16 | Ice_Strike | Will Broadcast Storm affect the call? |
| 14:27.29 | eric_hill | Ice_Strike: Muted audio may mean voice packets are getting dropped. |
| 14:27.53 | wdoekes | where I'm not sure which values of "except" are legal: 0 means false and any other number means true |
| 14:28.06 | wdoekes | what happens with "true" and/or an empty except, I don't know |
| 14:28.14 | Ice_Strike | eric_hill I need to find out what causing it.. It happen to other PBX boxes as well. |
| 14:28.19 | Ice_Strike | Different ISDN and SIP |
| 14:29.07 | ddickenson | wdoekes: right, that's how I normally do bool is 0 or 1 I'm not sure why I spelled out true. |
| 14:29.27 | eric_hill | Ice_Strike: Best way I know how to do that is open a voice call, then flood the pipe with a traffic generator. WanKiller works great. If the audio drops, your QoS rules aren't right. |
| 14:29.35 | *** join/#asterisk vlad_starkov (~vlad_star@212.34.52.66) |
| 14:29.49 | wdoekes | ddickenson: fyi: $["${except}"="true"] would evaluate to 1 |
| 14:30.04 | Ice_Strike | eric_hill QoS is not enabled |
| 14:30.20 | eric_hill | Ice_Strike: I found your problem. |
| 14:30.25 | ddickenson | eric_hill: if not in the sip.conf entry for the endpoint where would I assign that variable so that it is associated with those three phones? |
| 14:30.47 | ddickenson | wdoekes: " " gotcha |
| 14:31.14 | Ice_Strike | eric_hill The thing is I havent had this problem for while.. Starting having problem from yesterday |
| 14:31.27 | wdoekes | (oh, and I meant ExecIf(${except}? obviously that first time) |
| 14:32.12 | eric_hill | Ice_Strike: That doesn't mean it was right before... |
| 14:33.00 | eric_hill | ddickenson: I was just saying the ExecIf, GotoIf, Set, etc stuff is done in the extensions.conf. You can set whatever you want in sip.conf, just not dialplan stuff :) |
| 14:33.30 | Ice_Strike | eric_hill Ok, I will look into Qos. I have managed switches which include QOS features but a internet router include QOS too. |
| 14:33.33 | Ice_Strike | which one to use? |
| 14:33.45 | ddickenson | eric_hill: yeah I know all the applications are used in the dialplan, I guess what I'm asking is if I set that variable in sip.conf can I use it in the dialplan |
| 14:34.27 | ddickenson | other slight issue is I'm using realtime and I don't have a column in the db for setvar... I guess I'll have to put these three in the flat file |
| 14:34.40 | eric_hill | Ice_Strike: Both: http://en.wikipedia.org/wiki/Quality_of_service |
| 14:35.43 | Ice_Strike | eric_hill Cool. I have two pbx boxes use ISDN cards. and other pbx use Sip trink internet |
| 14:36.02 | eric_hill | ddickenson: Why not use the phone extension? I'm not sure what you're trying to put in sip.conf. |
| 14:36.34 | ddickenson | I guess I could. That would probably be easier |
| 14:36.53 | ddickenson | can you match based on a range? 460[1-3] sort of thing |
| 14:37.04 | ddickenson | _460[1-3] I mean |
| 14:37.22 | [TK]D-Fender | Not by pattern, but by high-low, yes, that is an expression |
| 14:37.34 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:39.31 | eric_hill | Ice_Strike: Every installation I've done in the last decade has used 4 QoS buckets: |
| 14:39.47 | eric_hill | Ice_Strike: Realtime for voip and video |
| 14:40.11 | eric_hill | Ice_Strike: Interactive for things that are visually criticial such as RDP or VNC |
| 14:40.20 | eric_hill | Ice_Strike: Business for things important to the business |
| 14:40.27 | eric_hill | Ice_Strike: And Other for everything else. |
| 14:40.52 | eric_hill | Ice_Strike: Most stuff should fall into Other, and be dropped if there's ANYTHING else going on. |
| 14:41.16 | eric_hill | Ice_Strike: Realtime goes at priority 1 above everything else to eliminate stuttering audio. |
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| 14:42.13 | ddickenson | sweet, thanks for the help. i think I've got it |
| 14:48.00 | Ice_Strike | eric_hill I did more testing and I notice there is a problem with internal calls too |
| 14:48.27 | Ice_Strike | It like a bit of mute during the calls, and also volume like too low randomly. |
| 14:49.31 | Ice_Strike | I am deffo going to look into QoS. |
| 14:49.50 | wdoekes | ddickenson: you can simply add the setvar column |
| 14:49.58 | eric_hill | Ice_Strike: "Volume too low" is not a QoS issue. Missing audio is. Dropped packets don't make the volume change, you just don't have audio. |
| 14:50.31 | [TK]D-Fender | ddickenson: Let us know if it works, and if it does and you'd like opinions, PB away |
| 14:50.42 | ddickenson | Thanks |
| 14:50.54 | Ice_Strike | eric_hill There is like a buzzing noise in the background - what is causing that? |
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| 14:51.21 | eric_hill | Ice_Strike: Analog issues. Digital stuff doesn't buzz. Are you a beekeeper? |
| 14:51.31 | Ice_Strike | lol no. |
| 14:53.02 | eric_hill | Ice_Strike: In all seriousness, you should probably grab an electrician and make sure you don't have a grounding issue in the building. |
| 14:53.20 | eric_hill | Ice_Strike: 50 or 60V on your ground circuit could cause all sorts of issues. |
| 14:54.38 | coppice | true. its important to ensure your building is on the ground |
| 14:55.15 | Ice_Strike | eric_hill We dont even have UPS |
| 14:55.17 | [TK]D-Fender | ~book |
| 14:55.17 | 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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| 14:55.40 | Ice_Strike | Will UPS help to stablize the power? |
| 14:56.26 | [TK]D-Fender | Ice_Strike: They usually just make my sign for my package and leave. I've never gotten tthem to do much more... |
| 14:58.49 | eric_hill | TKD: lol |
| 14:59.01 | ddickenson | ok, It seems to be working although i just got a troubling call from my boss to my number that had the "special" callerid coming through from him where it shouldn't be. I haven't been able to replicate it though. This is an example of one of my outgoing pattern matches. http://pastebin.com/t31ceMZx |
| 14:59.37 | eric_hill | Ice_Strike: That's *VERY* dependent on the UPS. And most good UPS systems will throw a warning with grounding issues. |
| 14:59.56 | Ice_Strike | Oh I see |
| 15:00.28 | Ice_Strike | We hardly get power cut here. |
| 15:00.36 | eric_hill | ddickenson: ExecIf(CALLERID(num)=460[1-3]...) is incorrect. |
| 15:00.38 | Ice_Strike | I will look into UPS |
| 15:00.57 | eric_hill | ddickenson: ExecIf($[${CALLERID(num)}=460[1-3]]...) is better |
| 15:01.14 | ddickenson | oh... |
| 15:01.27 | WIMPy | Since when can you use patterns in an If? |
| 15:03.28 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:03.39 | eric_hill | WIMPy: Good point. I was just cleaning up the expansions. |
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| 15:04.05 | ddickenson | meant to change that to high-low like TKD said, I wonder how it worked... |
| 15:04.06 | *** join/#asterisk navaismo (~navaismo@201.124.114.217) |
| 15:04.39 | eric_hill | ddickenson: I'm betting it didn't work like you thought. Since you had no expansion, it was probably matching *everything*. |
| 15:05.55 | ddickenson | yeah, I've already commented it out |
| 15:07.10 | eric_hill | ddickenson: Use same => n,ExecIf($[4601=${CALLERID(num)}],Set(CALLERID(num)=12345)) |
| 15:07.33 | eric_hill | That changes caller ID to 12345 if the calling number is 4601. |
| 15:08.13 | ddickenson | i guess i jsut need to do three lines then... for each phone? |
| 15:08.36 | eric_hill | ddickenson: Yes. |
| 15:08.54 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 15:09.05 | *** join/#asterisk lroe (~roe@unaffiliated/roe) |
| 15:10.30 | lroe | I realize it doesn't really matter, but is there an article explaining the change in version numbering? |
| 15:11.25 | *** join/#asterisk iggi (~iggi@ip69.50-31-21.static.steadfastdns.net) |
| 15:11.38 | Qwell | ~12 |
| 15:11.38 | infobot | from memory, 12 is 1 |
| 15:11.42 | Qwell | ~asterisk 12 |
| 15:11.47 | Qwell | infobot: stupid bot |
| 15:11.48 | infobot | Stupid human. |
| 15:11.49 | Qwell | ~asterisk12 |
| 15:11.56 | lroe | haha |
| 15:11.58 | WIMPy | eric_hill: Or use a setvar? |
| 15:12.07 | Qwell | There's a factoid for it... |
| 15:12.20 | Qwell | wait, no |
| 15:12.22 | Qwell | ~asterisk 10 |
| 15:12.25 | Qwell | ~asterisk10 |
| 15:12.25 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15. |
| 15:12.43 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:12.51 | lroe | thanks |
| 15:13.43 | navaismo | the bot here is amazing that stupid frog from freepbx should learn something |
| 15:14.47 | iggi | Hi, can anyone help me with an issue I'm having with the IVR and hold music. The IVR and Hold music both are choppy, but the calls are just fine. This happened after an AsteriskNow update a while back and I haven't been able to figure it out via google yet. |
| 15:14.55 | [TK]D-Fender | ~asterisk-versioning |
| 15:14.56 | infobot | somebody said asterisk-versioning was http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 15:15.08 | iggi | I tried 3 different timing modules and each seem to do it |
| 15:15.11 | [TK]D-Fender | hrm |
| 15:15.51 | navaismo | iggi pop? Maybe a issue with the bandwidth or cpu usage |
| 15:16.44 | iggi | navaismo, CPU usage is <3% at all times. Network is 1gbit with < 10% usage |
| 15:17.03 | iggi | worked just fine before my coworker did an update a while back |
| 15:17.54 | iggi | Tried dahdi, pthread and timerfd. They all seem to have the same issue. |
| 15:18.00 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 15:18.15 | iggi | It is a virtual machine, but nothing has changed on the host side |
| 15:19.06 | Ice_Strike | http://www.adcapnet.com/blog/why-online-ups-systems-are-necessary-for-voip-phones/ |
| 15:19.10 | MaliutaLap | iggi: if it's a virtual machine then why would you use DAHDI timing? it obviously doesn't have a DAHDI device |
| 15:19.13 | Ice_Strike | Just reading this eric_hill |
| 15:19.23 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:19.58 | MaliutaLap | Ice_Strike: you can buy me an online UPS, they're a tad exxy :P |
| 15:20.15 | iggi | MaliutaLap, honestly, not sure. I inherited this system, I'll admit I am a beginner when it comes to asterisk. What timing should I use? |
| 15:20.37 | navaismo | timmerfd |
| 15:22.48 | iggi | put pthread and dahdi as noload in modules.conf, timerfd is running as of module show like timing, but still have choppyness. |
| 15:23.44 | Greenlight | Usually I'd start looking at what timing source your kernel is using and what freq its set to; but you say it used to work, and only an Asterisk update was done ? |
| 15:24.01 | Greenlight | Ideally you want to be using HPET if you can |
| 15:25.07 | iggi | himm |
| 15:25.49 | iggi | Doesn't seem to have HPET, I'll have to figure out how to install/compile it |
| 15:26.11 | Greenlight | I beleive older kernels don't support it |
| 15:26.49 | Greenlight | Have you tried "timing test" from the CLI |
| 15:26.51 | iggi | It looks like a distro based on rhel/centos 6 |
| 15:27.00 | iggi | 2.6.32 |
| 15:27.30 | Ice_Strike | Greenlight Do you know any good UPS? |
| 15:27.40 | Greenlight | Ice_Strike: Sorry, not my area of expertise |
| 15:27.55 | ddickenson | eric_hill: got a little smarter, I think anyway and set the callerid for these three phones the same (since I need them to show up as that clid internal as well) and came up with this. exten => _NXXXXXX,1,ExecIf($[18552223333=${CALLERID(num)}],Set(CALLERID(num)=18552223333) |
| 15:28.05 | iggi | <PROTECTED> |
| 15:28.10 | Greenlight | I know centos 6 kerenls do support it, centos5 didn't iirc |
| 15:28.26 | Greenlight | Ahh, ok, so modify gruf.conf to set clocksource=hpet |
| 15:28.51 | Greenlight | In fact, might be worth a look there anyway; perhaps the kernel was updated and some options missed off in there ? |
| 15:29.05 | iggi | where's gruf.conf? |
| 15:29.17 | Greenlight | grub.conf sorry |
| 15:29.30 | iggi | ahh ok, just making sure |
| 15:29.32 | MaliutaLap | Ice_Strike: I have had good experiences with Solar and APC in the past |
| 15:29.39 | ddickenson | eric_hill: however, can I append an "else" on there like this: exten => _NXXXXXX,1,ExecIf($[18552223333=${CALLERID(num)}],Set(CALLERID(num)=18552223333:Set(CALLERID(num)=2225556666) |
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| 15:30.47 | MaliutaLap | iggi: kernels for virtual machines can be tricky. Do you know what hypervisor the VM is running under? |
| 15:30.56 | iggi | MaliutaLap, KVM |
| 15:31.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-86-84.ip.telfort.nl) |
| 15:31.11 | Greenlight | +1 virtual machines can make timing issues a nighmate |
| 15:31.16 | Greenlight | nightmare |
| 15:31.54 | MaliutaLap | iggi: hmmm, that should pass on underlying timing to the guest kernels (i.e. if the host kernel supports RTC and HPET then it should be no problem for the guests) |
| 15:32.31 | Greenlight | I'm hoping forcing HPET will solve it, since he seems to have it available |
| 15:32.31 | iggi | MaliutaLap, Well the host is Ubuntu 12.04 which is 3.2 iirc, so it should uspport it |
| 15:32.42 | MaliutaLap | Greenlight: oh, I've seen some shitey behaviour from them ... it's like hyperthreading on high i/o systems |
| 15:33.04 | MaliutaLap | iggi: I wouldn't count on it |
| 15:33.15 | ddickenson | eric_hill: my problem is that prior to this priority I'm setting the callerID for the normal "facility" callerid so it's already overridden the one set up for each specific phone when it's matched with an outgoing context so I thought maybe I could jump that priority or something, or possibly just eliminate it and have it change the clid to one thing if it matches the extension I'm calling from or if it doesn't match to just send |
| 15:33.15 | ddickenson | out the normal one |
| 15:33.21 | MaliutaLap | iggi: it might be worth while rebuilding the host kernel |
| 15:34.00 | Greenlight | iggi: Try cat /sys/devices/system/clocksource/clocksource0/available_clocksource |
| 15:34.09 | Greenlight | And cat /sys/devices/system/clocksource/clocksource0/current_clocksource |
| 15:34.14 | iggi | Host supports hpet |
| 15:35.58 | iggi | waiting for pbx to boot now with hpet=enable |
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| 15:36.53 | iggi | ok vm has hpet available as a clocksource |
| 15:37.01 | Greenlight | Is it now using it ? |
| 15:37.01 | iggi | but current clocksource is kvm-clock |
| 15:37.15 | MaliutaLap | which is probably correct |
| 15:37.17 | Greenlight | Yea |
| 15:37.22 | Greenlight | Sounds like best option |
| 15:37.32 | Greenlight | whats the results from "timing test" |
| 15:37.37 | Greenlight | Are we *sure* this is a timing issue ? |
| 15:37.39 | MaliutaLap | this is one reason I refuse to run * on a VM |
| 15:37.58 | iggi | I'm honestly not sure this is a timing issue. I know it's not CPU/bandwidth related. |
| 15:38.04 | MaliutaLap | even a low powered, or old machine is more reliable for things like this |
| 15:38.09 | Greenlight | I go with bare bones hypervisors, ESXi for example |
| 15:38.29 | Greenlight | iggi: Do a few "timing test" from the CLI then |
| 15:38.35 | iggi | "It has been 1000 milliseconds, and we got 50 timer ticks" |
| 15:38.38 | iggi | using timerfd |
| 15:38.39 | MaliutaLap | iggi: what volume of calls is the * system doing? |
| 15:38.42 | Greenlight | Thats perfect |
| 15:38.48 | Greenlight | Repeat a handful of times |
| 15:38.58 | iggi | MaliutaLap, incredibly light. about a dozen a day. |
| 15:39.17 | MaliutaLap | iggi: have you considered something like a Pi for this? |
| 15:39.21 | eric_hill | ddickenson: yes. |
| 15:39.27 | iggi | I could |
| 15:39.54 | iggi | ran timing test about 10-15 times and got the same response every time |
| 15:40.14 | Greenlight | Ok, so i'd say timing looks okay. |
| 15:40.27 | Greenlight | When we got timing related issues, that would show slips |
| 15:40.38 | iggi | I don't want to celebrate, but I just called it |
| 15:40.39 | Greenlight | Like 52/1000 |
| 15:40.40 | iggi | seems good |
| 15:40.45 | Greenlight | lol |
| 15:40.50 | Greenlight | SO a reboot did the trick ? |
| 15:41.24 | iggi | idk if it's a reboot or enabling hpet, disabling dadhi, something did it |
| 15:41.56 | *** join/#asterisk zapphir (~zapphir@46.195.16.114) |
| 15:41.58 | Greenlight | Had the server been up a long time ? |
| 15:42.05 | iggi | 1 week |
| 15:42.20 | iggi | we updated a week ago to try to see if it would go away |
| 15:42.27 | Greenlight | Gotcha |
| 15:42.38 | Greenlight | Oh, well, at least it's sorted, even if you don't know exactly what fixed it .. |
| 15:42.49 | MaliutaLap | because updates don't make things break :) |
| 15:42.58 | Greenlight | Never! |
| 15:43.03 | iggi | Yeah, well if it happens again I'll be back |
| 15:43.14 | MaliutaLap | especially when things like major * version change during them |
| 15:43.16 | iggi | Thanks Greenlight, MaliutaLap, navaismo |
| 15:44.03 | MaliutaLap | just noticed the rather large number of people with ops |
| 15:44.45 | Greenlight | No more than usual |
| 15:44.56 | iggi | My outlook is generally if it is broken, an update might fix it, but if it doesn't I'll still have to go in and fix it, might as well have the latest security updates if I have to fix it already. |
| 15:45.32 | iggi | but then again, trying to keep current with security updates caused the problem |
| 15:46.02 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 15:46.06 | Greenlight | I'd generally been rather cautious with updates, but got cought out recently where an update between asterisk 11.0 and 11.7 had fixed incorrect billsecs in CDRs, which meant I was billing customers wrongly |
| 15:46.24 | MaliutaLap | iggi: risky, if something gets point version updated you might have to start the configuration from scratch |
| 15:46.29 | Greenlight | So no I'm taking a more active approach with updates, when it comes to * at least |
| 15:46.33 | zapphir | I have problems with background noise when using queues. When I make external calls (using x-lite) the sound quality is good, and no background noise. When use the same x-lite, joins a queue and recieves a call, there is background noise which continues even after the recieved call is ended and I am waiting for the next call. Have tried all I can think of, changed asterisk versions (now using |
| 15:46.33 | zapphir | 11.5), tried denoise (which helps a bit in the call, but the background noise continues after the call is ended and I am back in the queue), I have also tried changing codecs with no success. I use alaw in all ends now. |
| 15:46.37 | MaliutaLap | iggi: you risk breaking more than what's already gone |
| 15:47.08 | iggi | I like to live dangerously |
| 15:47.59 | MaliutaLap | iggi: in a professional environment I'd call it "living stupidly" : |
| 15:48.01 | MaliutaLap | :) |
| 15:48.13 | zapphir | Does anyone know if there is a way to get complete silence between the calls in a queue? |
| 15:48.36 | Greenlight | zapphir: Not sure I understand what you mean ? |
| 15:48.41 | Greenlight | "between the calsl" ? |
| 15:50.23 | MaliutaLap | zapphir: what are you playing in the background? |
| 15:50.37 | zapphir | Greenlight: I use x-light for connecting to asterisk and login my agent to a queue that recieves incoming calls, when a call comes in to the queue it is connected to my x-lite. When that call is ended, there is an ongoing noise in the background, like a static white noise. Nothing is playing in the background.. |
| 15:51.08 | Greenlight | From memory I think chan_agent plays hold music |
| 15:51.25 | MaliutaLap | which would be the issue |
| 15:51.41 | iggi | MaliutaLap, Well, if it's already in a maintenance window, it's alright. I honestly don't know how this one slipped past, but it took a couple weeks before someone noticed it was actually an pbx issue. They all thought it was their phones and never told anyone. |
| 15:51.47 | MaliutaLap | zapphir: have you configure hold music? |
| 15:51.50 | Greenlight | I'm not sure you can "disable" that, but you *can* play a silent file |
| 15:52.10 | zapphir | Greenlight: I see, could be worth a try actually... |
| 15:52.29 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 15:52.31 | Greenlight | iggi: I wish our users were more like that; we could have a user with a phone based and in pieces not plugged in, and it's still a PBX issue... |
| 15:52.39 | Greenlight | *bashed |
| 15:53.12 | MaliutaLap | reason to hate users 1065 |
| 15:53.14 | Greenlight | zapphir: I beleive you can configure a music on hold class for chan_agent .. but it's been a while since i used |
| 15:54.47 | Greenlight | Although if you're getting white noise, it sounds like you have an underlying problem anyway |
| 15:54.52 | MaliutaLap | I prefer it when users are just a concept |
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| 15:55.16 | Greenlight | MaliutaLap: If only! :) |
| 15:56.17 | iggi | Greenlight, helps to have mostly users who are used to handling support calls and tend not to sweat the small stuff. |
| 15:56.43 | MaliutaLap | Greenlight: those postions do exist |
| 15:58.08 | Greenlight | No, your phone isn't broken, and it isn't a system problem. You've put it on DND. You know, just like you did last week. And the week before. Would you like me to expalin how to take it off DND? No. Are you sure?. Okay... |
| 15:59.58 | MaliutaLap | reminds me of a customer at the first ISP I ran. He used to call every day with exactly the same dialup configuration problem. It didn't matter how many times you talked him through fixing it |
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| 16:28.34 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 16:37.06 | linagee_ | Is there a 2013 better alternative to festival? :) |
| 16:37.38 | coppice | mardi gras will be along soon |
| 16:40.02 | Greenlight | chuckles |
| 16:41.43 | linagee_ | coppice: Swift would be like a one time cost of $140? ($100 for one channel, $40 for one voice) |
| 16:42.04 | Greenlight | chuckles more |
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| 16:49.19 | [TK]D-Fender | [11:41]linagee_coppice: Swift would be like a one time cost of $140? ($100 for one channel, $40 for one voice) <- it COULD be |
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| 16:54.06 | linagee_ | n/m regarding swift. I am not making a dime off of this project. :) |
| 16:54.27 | linagee_ | reverses Greenlight's chuckles |
| 16:54.54 | linagee_ | git revert chuckles |
| 16:55.52 | eric_hill | If I add an attendant on a Polycom phone, it can subscribe directly to the other phone and show state, but when going through an asterisk hint, phone state is ignored. Thoughts? |
| 16:56.15 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:56.20 | [TK]D-Fender | eric_hill: You set up your hints or peers wrong |
| 16:57.29 | eric_hill | The hints are simply exten => 123,hint,SIP/123, so that's not it... |
| 16:57.55 | coppice | linagee: flite is free |
| 16:58.52 | [TK]D-Fender | eric_hill: wrong place, or your PEERS are not setup properly |
| 17:00.04 | eric_hill | TKD: I have the hints in a "subscriptions" context, and I have subscribecontext=subscriptions in the peer definitions (via macro). |
| 17:00.18 | eric_hill | TKD: core show hints shows an active subscription to the target device. |
| 17:01.26 | eric_hill | I'll force notifyringing = yes even though the default is yes to see if that helps. |
| 17:02.52 | [TK]D-Fender | You are missing the required basics... |
| 17:02.56 | [TK]D-Fender | callcounter=yes <- |
| 17:06.39 | eric_hill | TKD: Fantastic. I hadn't set callcounter. What is its purpose? |
| 17:08.12 | zapphir | Greenlight: Thank you so much for the tip about an empty music on hold-file. That did the trick and the weekend is saved! cheers! |
| 17:08.35 | Greenlight | Heh, no problems, glad it worked :) |
| 17:09.25 | [TK]D-Fender | eric_hill: If you're not counting how many calls they are on... then you don't know if they have ANY |
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| 18:56.58 | cheasee | hi |
| 18:57.47 | cheasee | what could be wrong when read() gets no response, tough dtmf tones go through? |
| 18:58.16 | [TK]D-Fender | If DTMF is functional then read should work. |
| 18:58.19 | [TK]D-Fender | Show us both |
| 18:58.42 | [TK]D-Fender | ~pb |
| 18:58.42 | 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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| 18:58.46 | [TK]D-Fender | ^^^^ |
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| 19:03.44 | cheasee | [TK]D-Fender: thx, i know what a pastebin is ;) |
| 19:03.48 | cheasee | http://pastebin.com/JnkfVU53 |
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| 19:04.31 | cheasee | the two commented exten lines do work (when i pickup on 1021, i hear the dtmf tones) |
| 19:05.00 | [TK]D-Fender | You hearing audio, and * USING audio for DTMF are 2 entirely different things |
| 19:05.16 | [TK]D-Fender | And does not constitute prrof that * acks it in the first place |
| 19:05.41 | cheasee | ok? |
| 19:06.07 | [TK]D-Fender | You should alsoe do a playback to ensure that audio is established first |
| 19:06.17 | [TK]D-Fender | and you have 2 priority #1 there. |
| 19:06.20 | [TK]D-Fender | very bad |
| 19:06.46 | [TK]D-Fender | And no timeout set on the read |
| 19:06.48 | cheasee | oh, paste-error gg |
| 19:07.11 | [TK]D-Fender | indeed the dialplan does not match the execution at all |
| 19:07.15 | cheasee | i pasted only relevant information |
| 19:07.53 | [TK]D-Fender | Apparently not |
| 19:08.02 | [TK]D-Fender | the dialplan does not match the output |
| 19:08.23 | [TK]D-Fender | make the requested fixes and retest |
| 19:08.49 | cheasee | sec, ill try it again and repaste .. |
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| 19:17.05 | cheasee | [TK]D-Fender: http://pastebin.com/wHxvCg9N |
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| 19:20.45 | [TK]D-Fender | You have not made the fixes I suggested |
| 19:21.22 | cheasee | could it really be because of not setting a timeout on read? |
| 19:21.40 | cheasee | it seems to default to 10s |
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| 19:24.46 | cheasee | setting timeout didnt change anything (just that i dont have to wait 10s for timeouting the read heh) |
| 19:25.06 | cheasee | exten => s,n,Read(ffext,,4,,,3) |
| 19:25.36 | eric_hill | cheasee: How does the call come in? Through a T1? SIP? From a tin can on a string? |
| 19:25.50 | cheasee | do a playback? ok then ill play some audio first.. |
| 19:26.23 | cheasee | eric_hill: from a huawei 3g usb dongle with a sim card :) |
| 19:27.40 | eric_hill | Directly on the asterisk box? |
| 19:28.06 | eric_hill | Assuming so, what are your values in sip.conf for mindtmfgap, mindtmfduration, mindtmfdinterval, and do you have dtmf=relax set? |
| 19:28.15 | cheasee | ive got a raspberry pi with that dongle to accept gsm calls, this setup should forward the calls on my asterisk box |
| 19:28.58 | cheasee | im playing with dtmf setting, changed from (default) "relax" to "off" but no change |
| 19:29.27 | cheasee | so i let it on relax |
| 19:29.48 | cheasee | mindtmfgap=45 ; minimal interval from end of previews DTMF from begining of next in ms |
| 19:29.51 | cheasee | mindtmfduration=80 ; minimal DTMF tone duration in ms |
| 19:30.03 | cheasee | mindtmfinterval=200 ; minimal interval between ends of DTMF of same digits in ms |
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| 19:32.57 | [TK]D-Fender | cheasee: I told you to do a PLAYBACK first, and then explicitly set a timeout |
| 19:37.01 | cheasee | <PROTECTED> |
| 19:37.01 | cheasee | <PROTECTED> |
| 19:37.01 | cheasee | <PROTECTED> |
| 19:37.02 | cheasee | exten => s,n,Read(ffext,vm-extension,4,,,3) |
| 19:37.36 | cheasee | same with 20s timeout |
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| 19:38.13 | cheasee | fl49gsm*CLI> core show version |
| 19:38.13 | cheasee | Asterisk 1.8.13.1~dfsg1-3+deb7u3 built by buildd @ build06.raspbian.lan on a armv7l running Linux on 2014-01-05 23:16:59 UTC |
| 19:38.16 | cheasee | if it helps anyhow |
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| 19:43.20 | Pegasus_RPG | Hello. Here's a strange one: When I place a call from my Snom 300 phone using SRTP, the called party can hear me but I can't hear them. If I put the call on hold, then pick it back up, then I can hear them. This happens every time. Any ideas on how I can fix this? |
| 19:44.10 | Pegasus_RPG | Or where I should start looking? The verbose * log shows nothing different from when it works fine using another SIP phone |
| 19:47.16 | Pegasus_RPG | (Turning off SRTP works, but I'd like not to have to resort to that.) |
| 19:48.27 | cheasee | im dialing: +436603578649,1021 |
| 19:48.29 | cheasee | "," seems to be a special thing for a two-second pause |
| 19:48.31 | WIMPy | Where? |
| 19:48.31 | [TK]D-Fender | http://pastebin.com/wHxvCg9N <- is your latest pastebin. This doesn''t show it |
| 19:48.31 | cheasee | (this part works, cause when i send the call to an extension directly and pickup there, i hear the dtmf tones) |
| 19:48.32 | cheasee | [TK]D-Fender: sorry, i pasted latest output here in channel :) |
| 19:48.32 | cheasee | WIMPy: on any gsm phone? |
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| 19:49.38 | navaismo | maybe in the sdp via sip debug |
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| 20:29.14 | WIMPy | Pegasus_RPG: With 12 I had that even without srtp. |
| 20:36.13 | WIMPy | Has probing numeric usernames gone out of fashion? I see multi, pbx, pots, pstn, tech and user. |
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| 20:41.21 | Katty | HI LADS |
| 20:41.45 | WIMPy | Hi Katty |
| 20:42.08 | Katty | how're you WIMPy? |
| 20:42.52 | WIMPy | Ok |
| 20:43.19 | Katty | just ok? |
| 20:43.24 | Katty | not super duper? or peachy keen? |
| 20:43.32 | j4jackj | I hate life. |
| 20:43.33 | WIMPy | Somewhoat in the winter sleep mood, but I reinstalld a server today, so not too bad. |
| 20:43.53 | WIMPy | j4jackj: Don't worry. It will end. |
| 20:44.00 | j4jackj | Lel. |
| 20:44.08 | Katty | WIMPy: yes the weather will do that too. but yay for a little productivity |
| 20:45.07 | Pegasus_RPG | WIMPy: With 12 what? |
| 20:45.17 | WIMPy | Now I can finally use my config scripts on the central Asterisk server. |
| 20:45.19 | *** part/#asterisk drjfreeze (~Jim@207.191.114.82) |
| 20:45.38 | WIMPy | Pegasus_RPG: one-way audio until I put the call on hold and retrieve it again. |
| 20:45.57 | WIMPy | Or wait. I think I had to transfer it to myself. |
| 20:46.12 | Pegasus_RPG | How did you resolve it? |
| 20:46.26 | WIMPy | Downgrade. |
| 20:46.32 | Pegasus_RPG | suspects a firmware bug but this is also his first time with SRTP |
| 20:46.35 | Pegasus_RPG | oh ick |
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| 20:49.26 | WIMPy | Are you using 12? |
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| 21:18.22 | LemensTS | I have Polycom 321's, if one of them is on the line and I call them they hear a call waiting beep but I just hear ringing. I need to hear a busy tone to know they are on a call. I got it to work setting call-limit=1, but then I could not do attended transfer and another thing....any idea? |
| 21:20.40 | LemensTS | One other thing I heard...If I set RTP ports 10000-10025, it will only use the even ports 10000, 10002, 10004, etc? Opening ports in Windows Azure and I can't do a range without using powershell and dont want to open odd ports if there is no reason too... |
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| 21:34.51 | [TK]D-Fender | heads off home... |
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| 21:45.44 | newtonr | LemensTS, what version of Asterisk are you using? You should probably be using call-counter in sip.conf, plus GROUP and GROUP_COUNT functions instead of the deprecated call-limit. |
| 21:46.35 | newtonr | "callcounter" |
| 21:46.47 | LemensTS | Asterisk 12 |
| 21:47.14 | newtonr | LemensTS, yeah so check out like http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html |
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| 21:47.32 | Katty | is it time for tea yet |
| 21:47.56 | Katty | CAN IT BE ROYAL TEA |
| 21:48.48 | LemensTS | newtonr: Nice that will help me, looks like exactly what I need. Thanks! |
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| 22:09.26 | *** join/#asterisk flyingbull (~Flyingbul@cpe-098-122-151-068.nc.res.rr.com) |
| 22:10.22 | flyingbull | Hi everyone, Anyone around today? |
| 22:12.40 | paulc | ~ask |
| 22:12.40 | 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. |
| 22:12.56 | paulc | flyingbull: what's up? |
| 22:13.44 | j4jackj | I think he's looking for quality chats. |
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| 22:18.55 | flyingbull | Sorry, I'm on a mac and the IRC I'm using is a memory hog, and bogging it down. I'm trying to figure out how build a dialer program, I have ten users, what I want to happen, when the call file is moved into the directory, it creates the call, then when it connects, automatically connects to one of the SIP devices already logged in. I don't want to use a Queue, because I don't want to make people who pick up the phone, listen to |
| 22:18.56 | flyingbull | prompt or music. I've been trying to figure out if there is a way to open up a connection automatically to the SIP device, like a login, then when a call is answered, it checks who is logged in, and automatically connects the call. I've looked around, but I can't seem to phrase google right to get the answers I'm looking for, can any one point me to either an example or at least tell me what Dialplan apps I would need to use? |
| 22:18.57 | flyingbull | Thanks. |
| 22:25.12 | paulc | flyingbull: 2 things spring to mind.. 1: Look at ViciDial - it might do what you want. 2: Use a queue, because it deals with the whole logged in/available/not thing really well, and gives the best distribution of calls.. and either set the phones to auto answer (a setting on the phone) or use a SIP header (like Alert-Info) to make the phones auto answer.. This is assuming your logged in people are wearing headsets, and can be trusted to logout/change st |
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| 22:27.33 | paulc | Generally, I'd go with ringing phones, because then you can timeout/fall over etc with no weird empty air calls for the people you're calling |
| 22:30.11 | flyingbull | I think I must not understand Asterisk as well I thought, my thought was that when it connects, that means it is ringing the line, and goes to the next step of the dial plan. |
| 22:31.42 | flyingbull | I've looked at ViciDial before, and it is an interesting solution, but since my computers are in one state, and they are in another AND I'm using CENTOS, it caused me some headaches -- the directions for a manual install are pretty painful, and recompiling it from the console was impossible. |
| 22:32.39 | flyingbull | Not from the console is what I meant. I have ssh access but to recompile, CentOS just wouldn't do it. |
| 22:34.17 | paulc | Gotcha.. I've had issues with Centos before.. but generally always go with compiling Asterisk from source rather than using a package etc. |
| 22:34.59 | paulc | And no.. if you've got a Dial in the dialplan, it blocks at that point.. Dialplan can continue if there's busy, no answer etc.. or if the other end DOES answer, then hangs up (there's an option for this.. maybe "g"? can't remember) |
| 22:36.07 | flyingbull | I like the idea of auto answer, I'll have to play with that, it just determining who is logged in is the difficult part, trying to determine who was logged in with the sip server and their status is set to available/not in use. |
| 22:37.05 | mjordan | WIMPy: Are you getting one way audio right now with 12? |
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| 22:39.52 | flyingbull | Asterisk is pretty easy to install from source code, and I get the version I'm interested in, not what ever happens to be in the repository. |
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| 22:41.31 | paulc | flyingbull: Queues work great for the logging in/out thing.. or you can roll your own, using some scripting (either in the dialplan, or external via CURL etc). Custom device states can be useful too, depending on what phones you have.. We have Cisco SPA508Gs and they're great for that.. Keys to toggle features, lights change, etc. |
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| 22:45.00 | flyingbull | Everyone I'm dealing with is using Softphones, so it isn't too difficult, and right now we use Asterisk with FOP and with that, they all are able to pick up phone calls etc, right now my boss wants to be able to do campaign dialing, so I was looking at that, and how to group them in Asterisk. The trick is to prevent dead air while an agent is being chosen to pick up the call. |
| 22:46.14 | paulc | Gotcha.. And making sure there are any agents available before starting dial attempts.. |
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| 22:46.24 | paulc | it's a science, that whole predictive dialing thing.. or can be.. |
| 22:50.33 | flyingbull | Yes, that is a big one, What I was considering doing is playing with OBDC with asterisk, and have a website, that you would log into, this would log you into the campaign (then with a soft phone) once you logged in, you could control your status. I'll have to play with it, I'm sure if I can send the information to the website using a web socket to bring up the script that they need to use, I can probably capture the end event of |
| 22:50.33 | flyingbull | phone call, and force a disposition from the agent, then mark them available. The database would be updated with their current status, so it would just go down the line of who was available by making a sql call to the server to build a dial plan. |
| 22:52.14 | flyingbull | I'll just have to dig through the documentation a bit more, and do some experimentation on my local server and see what I can come up with. Thanks Paulc for your advice. |
| 22:52.43 | JeffC_NN | I've written a custom queue that does essentially what you're talking about with agent status. |
| 22:56.42 | flyingbull | I thought that Agents were being phased out by Asterisk? |
| 22:58.32 | JeffC_NN | I'm not sure, I use a custom desktop app. I wasn't referring to any Asterisk "agent" feature. |
| 23:05.43 | flyingbull | Are you saying that you used an app that created a dial plan that handled it, or you created a program that managed the queue's using like AGI or something like that? |
| 23:07.42 | mjordan | flyingbull: agents haven't been phased out. chan_agent was removed in 12, as it no longer fit with the modified architecture (and had a design that was brittle in the first place). That channel driver has been replaced with dialplan applications that achieve the same functionality (perma-logged in agent in a pool, etc.) |
| 23:08.50 | mjordan | flyingbull: see https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_AgentLogin |
| 23:09.13 | flyingbull | mjordan: OIC, I was reading the Asterisk-docs.org site, and came across to me that the authors didn't think that agent setup was the way to go. |
| 23:09.45 | mjordan | that's because the Asterisk authors are smart and wise people, who knew that chan_agent is a ticking time bomb |
| 23:10.32 | mjordan | if, however, you *really* want your agents always logged in on their phones, and you like pretty hold music entertaining them, then we wrote the dialplan applications in 12 that let you do that. You still use a Local channel member in the Queue to do the actual strategy of picking someone |
| 23:10.40 | mjordan | but you can, from the Local channel, request which agent you want |
| 23:10.49 | mjordan | you sort of have a 'two-pass' ringing strategy in that case |
| 23:11.44 | mjordan | except for that leifmadsen guy. Don't trust his advice. Srsly. |
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| 23:15.32 | flyingbull | mjordan: What I want is no dead air, when a call is dialed out. If the call connects and pushed into a queue, what I'm worried about is that there will be music or dead air, until an agent is connected. |
| 23:16.33 | JeffC_NN | could have recordings explaining the reason for the call while the agent connects (assuming this happens quickly enough) |
| 23:17.27 | JeffC_NN | you could time how quickly the agents get connected, and adjust the rate which your campaign dials out if it's taking too long. |
| 23:20.45 | JeffC_NN | you could set a timeout on the Dial() to the agent to match the length of the greeting, so that after the greeting, they either hear an agent, or control is back in the dialplan so you can handle it with additional audio (not dead air). |
| 23:21.24 | flyingbull | I guess I could do that, I would prefer that when it starts ringing that an agent is connected. |
| 23:22.32 | flyingbull | there doesn't appear to be a method for that, I wonder if I use a local channel with the bridge app in the call file, if that will do it. I'll have to just dig through the documentation and do some experimentation, and see what I get. |
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| 23:25.38 | flyingbull | Ok, need AFK for a little bit, catch you all later. |
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