| 00:00.28 | WIMPy | Depending on technologies you can set it on the calling device, its peer configuration and/or the outgoing peer as well as fron the dialplan. |
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| 00:02.10 | Micc_ | Looks like you can pass a parameter to the Dial option o(callerid) |
| 00:03.20 | dtrainor_ | Awesome, exactly what I was looking for. Thanks, newtonr |
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| 00:05.35 | newtonr | np |
| 00:10.41 | dtrainor_ | Should I use DeadAGI if I wanted to do some AGI hackery after the channel has been hung up? Is that the idea of DeadAGI? |
| 00:11.45 | WIMPy | Yes, but do you really need AGI or just a system call? |
| 00:12.36 | dtrainor_ | Hmm I'd like to use AGI. I just did a quick test, it may be erroneous but the result in the log was "DeadAGI has been deprecated, please use AGI in all cases!" |
| 00:13.03 | dtrainor_ | Ultimately I'd like to extract CDR data with AGI, that's the goal |
| 00:13.20 | dtrainor_ | Figured the most accurate data would be that of after the call had hung up? |
| 00:13.23 | WIMPy | That must have passed me by. |
| 00:13.35 | dtrainor_ | I remember reading it somewhere, I believe in the Asterisk book. |
| 00:13.55 | dtrainor_ | The heading was "DeadAGI is dead!", but I didn't know when/where it applied? idk. |
| 00:14.09 | WIMPy | Does not sound liek you need AGI for that. You only need AGI if you want to send somethign back to Asterisk. |
| 00:15.11 | dtrainor_ | oh, i was going to use a bunch of GET VARIABLE CDR(n) calls from my AGI script. are you hinting at a better way of doing this? |
| 00:15.47 | WIMPy | Put the Variables as parameter to the call to your script. |
| 00:16.33 | WIMPy | And if you care about efficiency, you can listen on AMI instead, |
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| 00:19.04 | dtrainor_ | Oh good call. Thank you. |
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| 00:57.56 | plut0 | i hear no ringing tone when making a phone call via google voice, what could be the problem? I'm using 'r' in my Dial command |
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| 02:11.50 | dtrainor_ | Do I need any special considerations when passing data as arguments to an AGI command? In particular, I'm passing CDR data which contains slashes, spaces etc. |
| 02:12.37 | [TK]D-Fender | You shouldn't be passing anything at all |
| 02:12.45 | [TK]D-Fender | That is against the entire idea of AGI |
| 02:12.54 | [TK]D-Fender | AGI can call those values DIRECTLY. |
| 02:15.27 | dtrainor_ | Right, that's what I was doing earlier, then I changed it up a bit to be in line with what WIMPy had suggested. |
| 02:18.32 | [TK]D-Fender | Using AGI ... drive it like AGI |
| 02:19.03 | dtrainor_ | ok. |
| 02:19.05 | dtrainor_ | thank |
| 02:19.06 | dtrainor_ | s |
| 02:19.49 | dtrainor_ | because, AGI |
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| 07:05.26 | Rico | hi |
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| 07:49.48 | barbariandude | Hello! I've recently been put in charge of my company's Asterisk server, and though I'm very confident in Linux stuff, I'm completely new to telephone systems. Could anybody possibly point me to a good manual, wiki or other type of guide specifically for telephone system stuff? All the ones I've seen so far focus on the Linux admin side of things, which I'm comfortable with |
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| 07:58.08 | barbariandude | :( I guess nobody is here at the moment |
| 07:58.21 | barbariandude | Oh well, hopefully somebody sees it at some point |
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| 08:02.52 | doop | barbariandude: i hear the asterisk book is pretty ok |
| 08:05.20 | barbariandude | doop: Just googled it. You're talking about the one at www.asteriskdocs.org? |
| 08:05.53 | doop | "asterisk: the future of telephony" |
| 08:07.04 | barbariandude | doop: Thanks, I'll go through it |
| 08:07.37 | doop | barbariandude otherwise it might just be a matter of learning telephony stuff |
| 08:08.20 | barbariandude | doop: Yeah, learning telephony stuff is exactly what I'm after, but I'm finding it a little difficult figuring out where to start |
| 08:09.37 | doop | maybe start with some specific questions you have regarding the new system you're inheriting |
| 08:12.34 | barbariandude | Well, my questions might sound a little silly, and they're exactly the kind of thing you would respond with "rtfm". What's a trunk, exactly? What are the different phases of a phone call? Why am I seeing "skip locking" all over the logs, is that bad? How does DAHDI work? and on and on. Basically, I fully expect this to be a bit overly much to ask someone to type out in a chat, so I'm looking for go |
| 08:12.40 | barbariandude | od reading materials to get to grips with the basics (thanks again for the recommendation, by the way) |
| 08:13.13 | doop | well, i'll toss you some freebies |
| 08:13.24 | doop | a trunk is a connection between two phone switches |
| 08:14.30 | doop | different phases of a phone call are generally setup, talk, and teardown |
| 08:14.45 | doop | no idea re skip locking |
| 08:20.04 | barbariandude | doop: I've just gone through the contents and introduction to the pdf (the future of telephony), and it's literally exactly what I wanted. Looks like 3/4 of it is focusing purely on the telephony side of it. Cool. |
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| 08:20.40 | doop | :) |
| 08:23.49 | ChannelZ | ~book |
| 08:23.49 | 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 |
| 08:23.53 | ChannelZ | ~primer |
| 08:23.54 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 08:30.04 | barbariandude | ChannelZ: Useful bot is useful :) |
| 08:31.14 | barbariandude | ~buybook |
| 08:31.14 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
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| 09:22.23 | CustosL1men | hi |
| 09:22.44 | CustosL1men | I want to setup a sip network, and then allow sip clients to dial PSTN routable numbers |
| 09:23.01 | CustosL1men | I would imagine there are providers that give you some SIP -> PSTN option |
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| 09:29.45 | mirela666 | <PROTECTED> |
| 09:31.16 | CustosL1men | mirela666, well yes - that is basically what I want |
| 09:31.26 | CustosL1men | but at some point it should be able to dial out |
| 09:31.34 | CustosL1men | to PSTN |
| 09:31.53 | CustosL1men | so I just want some provider that gives me a service which I can somehow intrgrate with SIP to achive PSTN dialout |
| 09:33.00 | cheche | I have an strange problem with asterisk 1.8.13.1 . I am able to received sip calls to the Thomson 2030, but when dial same number, I have got this error "app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)". This is just a local network setup. |
| 09:33.35 | cheche | I guess. that username/password/codeds and sip config is right. because I am able to received. |
| 09:33.49 | cheche | but I do not konw why I am not able to make calls from that phone. |
| 09:34.11 | cheche | how can I increase debug info? |
| 09:36.46 | mirela666 | CustosL1men, well you can google arround for sip terminition providers, and you can compare prices |
| 09:37.06 | CustosL1men | mirela666, cool - that is the term I was looking for |
| 09:37.32 | mirela666 | cheche, core set verbose 3 |
| 09:37.43 | mirela666 | cheche, core set debug 10 |
| 09:38.04 | mirela666 | if you have debug flag enabled in any log file in logger.conf |
| 09:39.48 | mirela666 | cheche, that usually means you are dialing a non existing sip user/peer, for example you dialed 1243 which in dialplan did Dial(SIP/1234) and 1234 is not defined in sip.conf or not registered (with status UNKNOWN/UNREACHABLE) |
| 09:50.34 | cheche | mirela666: -- Executing [2006@default:1] Dial("SIP/2001-00000015", "SIP/2006") in new stack |
| 09:50.35 | cheche | -- Executing [2001@default:1] Dial("SIP/2006-00000017", "SIP/2001") in new stack |
| 09:50.35 | cheche | [Jul 15 11:36:42] WARNING[25508]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 09:53.51 | mirela666 | cheche, run in CLI> sip show peer 2001 |
| 09:56.06 | cheche | mirela666: you are right. |
| 09:56.14 | mirela666 | unreachable? |
| 09:56.31 | cheche | Name/username Host Dyn Forcerport ACL Port Status |
| 09:56.31 | cheche | 2001/2001 (Unspecified) D N 0 Unmonitored |
| 09:56.31 | cheche | 2006/jaime 192.168.1.85 D N 5060 Unmonitored |
| 09:56.50 | cheche | somehow 2001 was not setup |
| 09:57.19 | mirela666 | yes, recheck the config and networking |
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| 10:01.03 | cheche | mirela666: I will double check, but I feel that I am allow guest users to make phone calls. |
| 10:04.49 | mirela666 | cheche, yes but if they don't register , you can't call them, at least not with Dial application |
| 10:05.35 | mirela666 | so 2001 > 2006 will work, and 2006 > 2001 will fail cause asterisk won't know the IP of destination |
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| 12:07.44 | Rico | hello |
| 12:07.55 | Rico | I have a problem when compiling asterisk 13 pjsip module : http://pastebin.com/GY2NYb38 |
| 12:08.12 | Rico | pjsip release get from github |
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| 12:26.13 | Eric-K | I've been reading about MOS scores and specifically G.729A. This codec should have a MOS score of 3.7. But what does that mean? Is 3.7 the max or is 3.7 is an average for good quality? |
| 12:27.01 | WIMPy | IIRC 4 would be perfect audio. |
| 12:28.07 | Eric-K | But what does 3.7 mean? Is that the max score for a G.729A call? |
| 12:28.27 | WIMPy | What max? |
| 12:28.55 | Barbariandude | I think he's asking if 3.7 is the maximum MOS score attainable for a G.728A-encoded call |
| 12:28.59 | Barbariandude | Right? |
| 12:29.08 | Eric-K | Exactly |
| 12:29.35 | WIMPy | Well, by the nature of the thing it should be average. |
| 12:30.06 | WIMPy | But that is surely measured on a perfect transmission. |
| 12:30.34 | Eric-K | So if I monitor my calls with VoIPmonitor a MOS score of 4.4 should be possible for a G.729A call? |
| 12:31.35 | WIMPy | Oh, 5 was max, not 4. |
| 12:31.40 | Eric-K | Yes |
| 12:32.11 | WIMPy | And if G.711 is listed with a MOS of 4.1, G.729* can't be better. |
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| 12:58.38 | Barbariandude | This isn't nearly as bad as I thought it'd be... I'm 100 pages through the first of 3 books I want to get through before I start messing with my company's Asterisk server, and I'm already starting to get stuff. In practice I'll probably fall flat on my face the first time around, but getting there :P |
| 12:59.04 | [TK]D-Fender | \o/ |
| 13:00.01 | WIMPy | /-o |
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| 13:09.21 | Barbariandude | Quick question: the book I'm reading is relatively old (but very informative on the telephony front) and says "there are now several phones that use IAX protocol, and work is progressing on hard phone support" |
| 13:09.38 | Barbariandude | Is IAX widespread now, or still a minority compared to SIP? |
| 13:09.46 | Barbariandude | Or dropped entirely? |
| 13:09.56 | WIMPy | It has become less not more. |
| 13:10.13 | WIMPy | I don't think there are any IAX appliances any more. |
| 13:10.25 | Barbariandude | Gotcha. Good to know. |
| 13:11.32 | WIMPy | You can earn more with stuff that doesn't work :-) |
| 13:11.41 | Barbariandude | Hehehe :P |
| 13:12.04 | Barbariandude | Maybe later, but for the time being I'd like to focus on what can get their Asterisk server not malfunctioning |
| 13:12.37 | Barbariandude | It hasn't been updated since 2009ish, and has started malfunctioning |
| 13:13.10 | Barbariandude | Basically, I want to set up more-or-less a new install, move the config over, make any changes from 1.6 to whatever the latest stable version is |
| 13:13.30 | Barbariandude | And put that HDD with the install on the old hardware (which still works fine) |
| 13:14.27 | WIMPy | Sounds like a plan. |
| 13:16.34 | Barbariandude | WIMPy: A rather nice side-affect of all of this is I can then put Asterisk on my CV. Good for both salary negotiations and for when I'm back looking for a job |
| 13:16.45 | Barbariandude | I'm getting paid to read docs atm, more or less xD |
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| 13:20.44 | CustosL1men | is there like general voip related channel ? |
| 13:21.37 | WIMPy | I think there's #vuc. Possibly oters. |
| 13:23.34 | CustosL1men | is nobody there ;( |
| 13:23.49 | Barbariandude | watches tumbleweed blow through #vuc |
| 13:24.17 | CustosL1men | so I want a SIP/VoIP termination service which allows aribtrary Caller ID |
| 13:24.26 | CustosL1men | anybody know of something like this ? |
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| 13:24.33 | CustosL1men | I would guess skype works with people like this |
| 13:24.56 | CustosL1men | actually does not have to be SIP |
| 13:25.10 | CustosL1men | I just want to make a call from software to a PSTN number with any Caller ID |
| 13:25.43 | CustosL1men | What is difference between SIP trunking and VoIP/SIP termination ? |
| 13:26.32 | WIMPy | Most ITSPs seem to allow any Caller ID. |
| 13:26.59 | WIMPy | Those are just marketing terms with little to no specific meaning. |
| 13:27.11 | CustosL1men | WIMPy, but I want to use like my cellphone number as Caller ID |
| 13:27.18 | CustosL1men | and actually, not just mine, but other peoples |
| 13:27.37 | WIMPy | Most ITSPs don't give a damn. |
| 13:27.38 | CustosL1men | which, is actually all dandy and cool, cos I will ask them to confirm with SMS from their number |
| 13:27.41 | CustosL1men | I see |
| 13:27.46 | CustosL1men | WIMPy, can you give me the name of one |
| 13:27.49 | CustosL1men | or where to find a list |
| 13:27.54 | CustosL1men | I want one where I can signup quickly |
| 13:27.56 | CustosL1men | and cheaply |
| 13:28.06 | WIMPy | Country? |
| 13:28.30 | CustosL1men | WIMPy, I'm in .za - but I don't care where they are |
| 13:28.39 | CustosL1men | WIMPy, as long as I can call most places in world |
| 13:28.47 | CustosL1men | and have Caller ID that they don't care about |
| 13:28.59 | CustosL1men | I care more for decent support than country |
| 13:29.17 | WIMPy | Then ask google for the best rates. |
| 13:29.27 | CustosL1men | don't care for rates really, just one one ;) |
| 13:29.32 | CustosL1men | but yeah, will google some more |
| 13:29.58 | WIMPy | Wow. Don't care about rates is a new one. |
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| 13:40.46 | Rico | is there realtime hints in asterisk 13 ? |
| 13:45.38 | newtonr | Rico, not that I am aware of |
| 13:45.46 | Rico | thanks newtonr |
| 13:46.01 | Rico | and is pjsip REQUIRED or is there a way to not use it ? |
| 13:46.36 | WIMPy | You can still use chan_sip. |
| 13:48.30 | CustosL1men | WIMPy, its not really for mass use that I want this, just testing |
| 13:48.40 | CustosL1men | WIMPy, so I want something that works then I will look for something cheap |
| 13:48.52 | newtonr | Rico, like WIMPy said, you can still use chan_sip. That being said, current and future development effort is going into pjsip and chan_sip is now extended support (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States) |
| 13:49.01 | CustosL1men | WIMPy, if I search for "ITSP caller-ID OR CLI" I find a bunch of arbitrary pages |
| 13:49.01 | *** join/#asterisk johnny_|_ (~johnny@unaffiliated/johnny-/x-2623418) |
| 13:49.02 | Barbariandude | Right, I'm off. Thanks to the people that answered my nooby questions! |
| 13:49.05 | CustosL1men | which doesnt seem like something I want |
| 13:49.07 | Barbariandude | o/ |
| 13:49.23 | Rico | thanks winpy and newtonr |
| 13:49.37 | Rico | s/winpy/WIMPy |
| 13:50.29 | WIMPy | CustosL1men: Just search vor voip rates. |
| 13:50.34 | newtonr | Rico, This article https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip includes notes on disabling pjsip if you want to use chan_sip for a while still until you are able to migrate to using pjsip. |
| 13:51.37 | [TK]D-Fender | CustosL1men: Very few will actually put it in writing like that. This search will be almost worthless |
| 13:51.46 | WIMPy | Yes, the SIP situation is going to become interesting. chan_sip is on extended support and chan_pjsip requires external support. Does not sound like a situation you want to find yourself in. |
| 13:52.25 | [TK]D-Fender | WIMPy: Where was that TTl/support part documented? |
| 13:53.25 | WIMPy | The versions? |
| 13:54.08 | WIMPy | I guess if you want sip support, you might have more luck with 11 than 13. At least in theory. |
| 13:55.15 | [TK]D-Fender | I meant about chan_sip as "extended" support. |
| 13:55.36 | [TK]D-Fender | Was that meant as "ongoing", or "going to die soon"? |
| 13:56.01 | WIMPy | I can't remember, but I have seen it and I also trust newtonr. |
| 13:56.18 | [TK]D-Fender | newtonr: Care to chime in? |
| 13:56.32 | WIMPy | I guess it's more like you have to find someone who cares. |
| 13:57.03 | WIMPy | Just scroll up some 20 lines. |
| 13:57.23 | [TK]D-Fender | I care to know where things are heading... and then after those few seconds net me my answer I can quietly continue on with everything else I have to do :) |
| 13:57.50 | newtonr | By extended support I meant the literal support state noted in menuselect, I believe we moved chan_sip to extended support starting in 12 |
| 13:58.08 | file | it was 12 or 13 |
| 13:59.25 | file | from a Digium perspective we're focusing on PJSIP |
| 14:00.35 | file | and we have a good relationship with Teluu, the makers of PJSIP |
| 14:00.58 | file | I've (and other people) have contributed fixes back and they are responsive if we come across stuff |
| 14:02.05 | WIMPy | I'd be a lot more confident if Asterisk would check out a specific version of PJSIP to use instead of relying on some preinstalled one. |
| 14:02.36 | file | the install_prereq script pulls from our github and compiles/installs |
| 14:02.55 | file | otherwise the packagers wanted it completely separate and not part of things |
| 14:04.12 | WIMPy | Ok, that way sounds better. |
| 14:04.57 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-rtauymqzzvdgabsb) |
| 14:05.01 | file | I also pull pjsip from mainline trunk nightly and build against it during my normal work, so if something immediate breaks I notice |
| 14:05.31 | newtonr | [TK]D-Fender, Regarding your mention of "ongoing" or "going to die soon". I'd say chan_sip is likely to be indefinitely in extended support. I doubt we would ever drop support completely as too many people use it. |
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| 14:08.47 | Cyford33_p_l_y | enteresting... i stopped using pjsip becuase asterisk dialplan couldnt retrieve a sip header message when using pjsip for moble push... and couldnt even register a device that had extra message in the sip header https://issues.asterisk.org/jira/browse/ASTERISK-24654 |
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| 14:16.19 | newtonr | Cyford33_p_l_y, Patches accepted! |
| 14:16.51 | newtonr | but really, it is a group effort. If you want to improve the behavior, go for it! |
| 14:18.05 | Cyford33_p_l_y | if i could do it, it would of been done already... my only option was to move back to chan_sip which is capable.. but if pjsip will be replacing chan_sip then it would be nice for pjsip to be able to do everythinng chan_sip can do |
| 14:19.21 | newtonr | It would be nice. chan_sip was developed with the help of thousands of contributors. It'll take contributors like you to help make res_pjsip/chan_pjsip do everything that you want it to do. |
| 14:20.26 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 14:21.23 | Cyford33_p_l_y | i can help by testing its limits and reporting it :) i am no programmer though... by the time i learn asterisk/pjsip and there inter hooks,.. and programming patches, it probably would be done by then. |
| 14:23.37 | newtonr | The community will appreciate whatever help you can provide of course. :) If you are unable to provide a patch for behavior that you want, then you can only be patient and wait for others to do it, hire someone, or maybe post a bounty for others. |
| 14:23.50 | newtonr | testing the limits and reporting issues is always a huge help. |
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| 14:33.59 | Cyford33_p_l_y | what i dont get though is your goal is to move forward with pjsip and you guys have great communication with pjsip.. but refuse to patch something that may seem to be easy between asterisk programmers and pjsip programmers, and important to common users. using messages in sip headers opens up many doors and adds alot more power to our dialplan when used. |
| 14:34.48 | Cyford33_p_l_y | i also use sip messages to send text to cell phones, i wonder if pjsip is able to do that in asterisk as well |
| 14:35.52 | Cyford33_p_l_y | SMS text messages i mean |
| 14:41.27 | mjordan | where did anyone refuse to patch something? |
| 14:42.26 | mjordan | the issue was closed because it was an improvement without a patch. No one submitted a patch for it. |
| 14:42.41 | mjordan | If someone submits a patch, I'd be happy to have it reopened, and help that person through the review process. |
| 14:43.01 | mjordan | But there's no maliciousness here. As a project, we don't leave improvement requests lying around with no patches. |
| 14:43.46 | newtonr | We even have documentation! https://wiki.asterisk.org/wiki/display/AST/New+Feature+Guidelines |
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| 14:51.45 | Cyford33_p_l_y | ok so pretty much i need to find a fix, or pay someone to fix it, then show u how i did it? is this pretty much what ya mean by i need to give u a patch |
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| 14:53.58 | newtonr | Cyford33_p_l_y, yes, Asterisk is open source. If you are unable to provide a patch for a feature that you want, then you need to persuade or pay someone to do it for you. https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process |
| 14:54.15 | file | or wait for someone to come along and be interested |
| 14:54.25 | newtonr | Yup, patience |
| 14:54.54 | newtonr | If the issue is as important as you feel it is then others will work on it when they are able. You've done what you can by reporting it. |
| 14:55.08 | *** join/#asterisk DrDamnit (~michael@2601:cf:4500:ab00:a62:66ff:fe82:a1f7) |
| 14:58.14 | DrDamnit | Have Asterisk 1.8.32 with 17 phones that work great. Server has two nics 192.168.10.0/24 and a public IP 173.165.187.157. Routes all look good. Remote polycom phone cannot register. CLI doesn't even show an attempt. Syslog server on asterisk box DOES receive syslog info from remote phone. Setup is correct. Remote phone can download (via http) the configs successfully. Seems like Asterisk not listening on that adapter. Do I need |
| 14:58.14 | DrDamnit | to enable ipv4 masquerading? |
| 15:01.34 | DrDamnit | Eventually, Polycom registration times out, and I get this reported to syslog (on the asterisk box via the public IP): Registration failed User: 199, Error Code:480 Temporarily not available |
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| 15:06.17 | bmurt | so this is probably a dumb question, but what is the difference between dahdi-linux and tools vs dahdi-linux-complete |
| 15:06.35 | drmessano | 1, 2, or 1+2 |
| 15:06.39 | WIMPy | One is part of the other. |
| 15:06.41 | drmessano | The complete package contains both |
| 15:06.49 | WIMPy | The other part are the drivers. |
| 15:07.15 | WIMPy | The mathematical version was shorter. |
| 15:07.19 | bmurt | both accomplish the same steps just depending on how many steps you want to take |
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| 15:13.55 | drmessano | If you don't need the tools and only the drivers, no one is forcing you to download them |
| 15:14.03 | drmessano | Thats about it |
| 15:14.09 | drmessano | CHOICE |
| 15:14.14 | drmessano | FREEDOM |
| 15:14.17 | drmessano | #ASTERISK |
| 15:14.35 | drmessano | #OPENZORSE |
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| 15:22.10 | newtonr | DrDamnit, " CLI doesn't even show an attempt." Do you have "sip set debug on" enabled? |
| 15:22.27 | newtonr | DrDamnit, I'd also grab a tcpdump on the machine, check it out in wireshark to make sure the register is actually hitting the box |
| 15:22.42 | bmurt | drmessano: 'merica |
| 15:22.48 | bmurt | ty guys |
| 15:25.24 | DrDamnit | newtonr, sip set debug on piped via tee to /tmp/asterisk.log. Grepping log does not show a registration attempt from that peer or from the remote WAN or remote LAN address. I'll run a tshark, and advise. |
| 15:27.04 | newtonr | DrDamnit, if you see it hitting the right port in the dump then Asterisk should be getting it, especially if it is receiving traffic from other phones on that port |
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| 15:31.30 | DrDamnit | newtonr, "other phones" are all on the local subnet (192.168.10.0/24), which is coming in on eth0. running tshark, will advise with results. |
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| 15:50.45 | DrDamnit | newtonr, Thank you for the reminder to check with wireshark. Registrations attempts never arrived, which prompted me to check the Polycom configs (carefully). DNS was incorrect for voIpProt.server.1.address. |
| 15:52.42 | stefan27 | I made 3 different calls at different times to 3 different softphones on my PC (firefox webrtc, chrome webrtc, zoiper) and captured the udp traffic on wireshark, where I can see that the jitter distribution is very similar for all 3 calls, but the perceived sound quality is very different (zoiper is superior). it looks like even though they all 3 supposedly use adaptive jitter buffers, zoiper |
| 15:52.42 | stefan27 | does more buffering. |
| 15:53.50 | mjordan | wireshark sees all; wireshark knows all |
| 15:54.37 | stefan27 | i want my webrtc clients to always buffer more (add delay), but the browsers doesnt seem to support configuration for fixed delay jitter buffers |
| 15:55.25 | stefan27 | anyone got an idea for a work-around? somehow manipulating the adaptive jitter buffer of the browsers? |
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| 15:56.37 | WIMPy | Add artificial jitter? |
| 15:56.45 | stefan27 | my jitter distribution is pretty much 'perfect for 30 seconds, (no delay); then a random 200 ms void out of nowhere; then no jitter again...' |
| 15:57.13 | stefan27 | I don't know how that could be done conviniently? |
| 15:57.54 | WIMPy | There is software available to do that. |
| 15:58.20 | WIMPy | But it won't help you if the artifical delay coincides with a real delay elsewehere. |
| 15:58.42 | WIMPy | VOIP needs a perfect network. |
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| 16:02.31 | newtonr | DrDamnit, cool I'm glad you found it |
| 16:03.51 | stefan27 | The network is not the problem, in this case the asterisk which is in the media path is run in a bad environment. |
| 16:04.27 | WIMPy | Wow. That must be seriousely overloaded then. |
| 16:04.42 | WIMPy | But at least that's easy to fix. |
| 16:05.14 | stefan27 | the environment is not in my control at the moment, someone else is managing that |
| 16:05.42 | WIMPy | SEP. Even better :-) |
| 16:05.59 | stefan27 | but it seems complicated, because on average it behaves fine, it just gets random big spikes at odd times |
| 16:06.36 | stefan27 | SEP=? |
| 16:07.08 | WIMPy | Somebody Elses Problem. |
| 16:08.19 | stefan27 | it's my problem too because the users are unhappy with the sound quality :) |
| 16:11.15 | stefan27 | Maybe i can try play with the artificial delay idea, thanks |
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| 16:24.57 | CustosL1men | hi |
| 16:25.17 | CustosL1men | I want to apply Diameter Credit Control Application (DCCA/Ro) charging to SIP |
| 16:25.24 | CustosL1men | what is best option for this ? |
| 16:27.13 | [TK]D-Fender | Go read up on CDR & CEL. That's what we;ve got |
| 16:29.58 | CustosL1men | [TK]D-Fender, but that won't help for prepaid charging though |
| 16:30.00 | CustosL1men | I think |
| 16:30.23 | [TK]D-Fender | ok. So go check it yrouself. * has nothing for this |
| 16:30.41 | [TK]D-Fender | It's up to you to read in whatever account/limiting info you want and apply that to your calls |
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| 18:10.21 | Bouldr | Ok guys so I have 4 extensions, three of which are cordless DP715's in my office and one remote in another city running the same DP715, and they randomly go "UNREACHABLE" every so often and then come back randomly. |
| 18:10.45 | Bouldr | When they go unreachable, it just spams a 102 OPTIONS request over and over |
| 18:11.10 | Bouldr | Or, The server sends that to the extensions rather |
| 18:12.06 | Bouldr | The remote location I thought I had an idea with it being perhaps a sip ALG issue with the modem and I'm getting a router there to bridge from the modem but the ones in my office stump me. |
| 18:13.24 | Bouldr | This is what it looks like when I debug the extension in asterisk when its unreachable:m http://pastebin.com/Sy7ceHRV |
| 18:16.56 | Bouldr | It's also worth mentioning I have ~20 other extensions all with the same config that work just fine |
| 18:17.06 | Bouldr | both remotely and in this building |
| 18:20.18 | Chainsaw | Bouldr: They could just be badly behaved SIP stacks. Are they all the same hardware & firmware? |
| 18:22.25 | Bouldr | Same hardware, Different firmwares |
| 18:22.39 | Bouldr | But, I've got working versions on the same firmware as ones that are broken |
| 18:23.01 | Chainsaw | Bouldr: I agree, that is odd. |
| 18:23.06 | Chainsaw | Bouldr: Is this Asterisk 11 or 13? |
| 18:23.38 | WIMPy | Does it ever come back online before it re-registers? |
| 18:23.59 | Bouldr | 11.17.1 |
| 18:24.15 | Chainsaw | Bouldr: Two things to consider with Asterisk 11 and the chan_sip stack... |
| 18:24.28 | Chainsaw | Bouldr: A large astdb will slow register handling, by a lot. |
| 18:24.49 | Chainsaw | Bouldr: The DNS handling is single-threaded and prone to odd delays if you reference things that do not resolve. |
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| 18:30.00 | Bouldr | WIMPy: There is no rhyme or reason when it comes back, except every single time I reboot the phone it will always come back then drop after a bit, then back again and gone. |
| 18:30.42 | Bouldr | Chainsaw: I see, it always goes unreachable after the "poke no answer" from chan_sip |
| 18:30.50 | WIMPy | Off course it will come back when it registers. That's why I asked. |
| 18:31.14 | Bouldr | WIMPy: I've tried manually qualifying in asterisk once it goes unreachable and that just sends an endless spam that it never responds to |
| 18:31.45 | Bouldr | Similar to pastebin above except it just goes forever |
| 18:31.47 | WIMPy | Sure. And that was not my question. |
| 18:32.12 | Bouldr | Apologies, the answer is probably no. |
| 18:33.04 | WIMPy | Ok. So that just looks like a router forgetting the "connection". Perfectely normal. |
| 18:33.11 | WIMPy | It might have run out or RAM. |
| 18:34.13 | Bouldr | The server? |
| 18:34.29 | Bouldr | Just checked, it's running at ~14% |
| 18:34.33 | WIMPy | The router. |
| 18:34.50 | Bouldr | Ah - Got it. |
| 18:38.51 | Bouldr | WIMPy: Does that make sense if it's only this one base station / 3 handsets having the issue in the building? We have 10 extensions just in this office, none of the others have issues like this |
| 18:44.05 | Chainsaw | Bouldr: Are they all behind NAT? Or only the malfunctioning ones? |
| 18:52.52 | Bouldr | Chainsaw: Yes they're all behind NAT |
| 18:53.31 | WIMPy | The same router? |
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| 20:08.03 | pramsky | anyone run into any issues converting Cisco6901s to SIP ? |
| 20:08.41 | WIMPy | yes |
| 20:09.41 | pramsky | the one I am running into is that the phone never gets the sip FW from tftp |
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| 22:00.06 | Dick-Tracy | Afternoon #asterisk -- Im looking for a way to tranfer a channels var's to another channel, however, its only after the 1st has hungup that i can do the transfer. is there a way to do this without using a global variable ? |
| 22:00.39 | Dick-Tracy | or should i be looking in another place? |
| 22:01.38 | [TK]D-Fender | You'll need to be more specific about the exact channels involved and their relationship to each other |
| 22:04.12 | Dick-Tracy | both are sip channels, and at hangup of the 1st it is creating the 2nd |
| 22:06.02 | *** part/#asterisk mjordan (mjordan@nat/digium/x-wlsdeymygjxddslf) |
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| 22:08.41 | [TK]D-Fender | Creating how? |
| 22:09.04 | Dick-Tracy | Originate currently -- |
| 22:09.37 | [TK]D-Fender | Which kind of Originate? |
| 22:09.58 | Dick-Tracy | cli based w/ asterisk v11 |
| 22:10.14 | robmal | You could try Dial() earlier and ChannelRedirect() |
| 22:10.42 | [TK]D-Fender | Wouldn't advise either |
| 22:10.52 | [TK]D-Fender | Use a CallFile or AMI Originate for this |
| 22:10.59 | [TK]D-Fender | You have the option to set channel variables with those |
| 22:11.04 | [TK]D-Fender | And you can pass on what you want |
| 22:13.04 | Dick-Tracy | the .call is whats being used now, was just hoping i could do it all in the dialplan, instead of on disk, as the "failed" ext doesnt seems to want to move to a goto priority |
| 22:14.26 | Dick-Tracy | which doesnt allow me to findout the final status of the call |
| 22:14.39 | [TK]D-Fender | It should work fine. |
| 22:14.54 | [TK]D-Fender | You could also dial a local channel and let that dialplan do all the logic including handling the failure |
| 22:26.46 | Dick-Tracy | Alright [TK]D-Fender, I'll go about rewriting the dialplan to make .call files -- just to clairify you shouldnt edit them while in the outgoing directory right... in which case i dont know how im going to account for redials |
| 22:27.22 | [TK]D-Fender | It has it's own redial count in there |
| 22:27.22 | Penguin | [TK]D-Fender: its |
| 22:27.34 | [TK]D-Fender | Or do it yourself in a local channel instead of a direct device dial |
| 22:28.08 | [TK]D-Fender | And you nver edit these. Make the file outside of the spool DIR and you are required to MOVE them in from the same filesystem |
| 22:28.59 | Dick-Tracy | thats where the problem lies, i want to send an email if it failed... at the end of all the redials |
| 22:29.41 | Dick-Tracy | i'll lookinto the local channel .. never done it that way before |
| 22:31.02 | [TK]D-Fender | That'll let you do whatever you want |
| 22:31.10 | [TK]D-Fender | you do the redials yourself direct int he dialplan. |
| 22:31.12 | [TK]D-Fender | total control |
| 22:32.25 | Micc_ | Is there any way to avoid this error when I'm intentionally using spaces? Please avoid unnecessary spaces on variables as it may lead to unexpected results |
| 22:32.46 | Micc_ | I've tried using quotes, but that doesn't seem to help. |
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| 23:45.34 | Dick-Tracy | Hey [TK]D-Fender doen't look like i can make Local work either as it stops processing at hangup() |
| 23:45.56 | Dick-Tracy | unless theres another flag i need to be passing other then /n |
| 23:46.11 | WIMPy | Off course. |
| 23:46.19 | WIMPy | Don't hangup. |
| 23:46.36 | WIMPy | (unless you're finished) |
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| 23:49.59 | Dick-Tracy | its the remote end hanging up, not asterisk |
| 23:50.16 | WIMPy | That's not what you said. |
| 23:50.44 | Dick-Tracy | yup i see that now sorry |
| 23:50.50 | WIMPy | The remote and hanging up will onlt end the Dial() applicatiom. There's no Hangup() involved. |
| 23:51.11 | WIMPy | And there are options for Dial() to continue after the call has ended. |
| 23:53.29 | Dick-Tracy | more then just the h ext WIMPy ? |
| 23:54.16 | Dick-Tracy | no.. stupid question... was the next letter on the page <_< |
| 23:56.27 | Dick-Tracy | i keep forgetting its all running in stub contexts <_< |