| 00:01.26 | *** join/#asterisk RickCogley (~anonymous@EM114-48-67-66.pool.e-mobile.ne.jp) |
| 00:05.07 | carrar | ohayoo!! |
| 00:14.54 | *** join/#asterisk coppice (~coppice@m121-203-206-237.smartone.com) |
| 00:23.51 | drudge` | on asterisk 1.8.9.3, i get "Command 'module load chan_sip' failed." when trying to load chan_sip |
| 00:24.01 | *** join/#asterisk BJD10 (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 00:26.10 | BJD10 | Q: I have a string SIP/101&/SIP/102&SIP/203 and I want I want to do is split it in to each part and then do something with it. like dial. Other languages have the idea of using this as an array or list, and 'popping' off variables when your done with them.. anything in asterisk or AEL that will let me do something like that? |
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| 00:34.28 | phix | :D |
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| 00:39.47 | paulc | carrar: well will you take a look at that! Temps in C as well as F! that's sweet.. |
| 00:39.51 | paulc | delayed reply, sorry - had people at my desk |
| 00:40.11 | paulc | That DS3611xs is niiiiiice (and I bet the price tag is niiiiiice to match?) |
| 00:42.08 | carrar | yeah a good NAS with good drives (5 year warrenty) are not cheap |
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| 02:04.59 | Docfxit | Hi, I'm am having a hard time getting a polycom 320 phone to update it's time. If I look at the status it says SNTP pool.ntp.org GMT Øffset -28800 I'm on the west coast. I reboot the phone and it doesn't change the time. |
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| 02:10.28 | nny | i have a system here I have stripped of most modules due to them not being needed, but somehow I broke CLI output of messages. Can someone tell me which module and/or log file is specifically reponsible for CLI output? Thanks, I know it seems silly |
| 02:10.38 | nny | er log file = conf file |
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| 03:49.17 | alexoc | Asterisk ignores changes to realtime queue member table after initial startup. How can I do to solve this? |
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| 04:05.41 | alexoc | I already resolved, thanks. |
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| 05:48.48 | joobie | hey guys.. anyone know much about building redundancy into a 100-indial isdn? |
| 05:49.26 | joobie | we have a 100-indial setup with a provider and the numbers terminate on our ISDN.. but the other day the datacenter where the ISDN is had an outage and i want to keep these numbers functioning duringa ny outage |
| 05:49.40 | joobie | im thinking about setting up another asterisk box at another site.. but not sure how to achieve the 100 number redundancy |
| 05:50.13 | joobie | i could fire up another isdn circuit at the other site and ask the provider to set it up so that if there's an alarm on our main isdn, it automatically reroutes to our secondary.. but if there's a problem with their ISDN switching then this wont happen |
| 05:50.32 | joobie | is there a standard way to do this? kinda like how BGP is handled i guess to give this cross-provider redundancy |
| 05:51.36 | [TK]D-Fender | joobie: If you can't trust your upstream carrier you'er screwed |
| 05:52.21 | joobie | TK |
| 05:52.25 | joobie | u are back |
| 05:52.58 | joobie | when did the ban get lifted? |
| 05:53.12 | joobie | TK, the fukers on average have about 2 outages a year |
| 05:53.36 | joobie | it's painful.. I was thinking about putting in another asterisk box at another site and switching over the phones via DNS to the secondary site in the event of downtime |
| 05:54.01 | joobie | for the price we pay, we are keen to stick with the carrier.. |
| 06:00.38 | [TK]D-Fender | joobie: Your having a second box doesn't sound great when its your carrier that is the issue |
| 06:01.06 | [TK]D-Fender | joobie: See with an ITSP you can have your internet connection and server redundant, but if the carrier itself flakes out then you're toast either way |
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| 06:21.26 | joobie | TK, i hear ya.. |
| 06:21.40 | joobie | last outage I gave them a call and asked them to divert my numbers |
| 06:21.54 | *** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru) |
| 06:21.56 | joobie | and their response was that all their engineers are tied up trying to resolve the outage so there's no one to do it |
| 06:22.14 | joobie | the sales guy before that told me I could call and ask them to divert the numbers in an outage.. wanker |
| 06:59.48 | phix | joobie: Yeah sales and marketing == bunch of liers and deceivers |
| 06:59.59 | phix | I don't know how they can sleep at night |
| 07:05.40 | phix | any way, back to asteriskl |
| 07:06.16 | phix | I want to dial a bunch of extensions, any way to group them? or do I need to specifiy all of the extensions I want to dial delimitered by & ? |
| 07:07.57 | kaldemar | phix: do you want to dial them all at once every time or balance the calls between a group of them? |
| 07:10.17 | phix | All at once on this particular setup, but I do have another setup where I want all but a couple of phones to ring, then the couple of phones to ring if no one has picked up within x secs |
| 07:11.03 | phix | I am guessing I do this with queues? |
| 07:12.24 | kaldemar | or with dialplan directly. exten => s,1,Dial(SIP/adsf&SIP/qwer,20) exten => s,n,Dial(SIP/zxcv) |
| 07:14.08 | kaldemar | you'll get better control over the members (destinations) and calls if you use queues, local channels can also be of value when done directly. |
| 07:14.19 | phix | I dont want it to stop dialing the first lot, I want it to ring some first, then all after x, I guess I can just nclude all of them in thre in the next call to diak |
| 07:14.34 | phix | but to make things more complicated I will be using queues |
| 07:23.42 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:23.44 | schmidts | good morning |
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| 07:34.06 | wdoekes2 | good morning |
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| 07:57.23 | ChannelZ | is it? |
| 08:01.20 | wdoekes2 | why wouldn't it be? |
| 08:03.42 | ChannelZ | well it's only Tuesday |
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| 08:20.45 | tzafrir | jkroon, here |
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| 08:31.22 | ChannelZ | OK, random question |
| 08:32.08 | ChannelZ | Anyone know of any cli utilities which I can stream audio through and have it act as a level compressor (with make-up gain)? |
| 08:33.35 | jkroon | tzafrir, hope you're doing well. |
| 08:34.32 | tzafrir | I hope so too. What's up? |
| 08:34.47 | jkroon | we cooked a patch based on the sf branch that you maintain to split the oct612x stuff into a separate .ko file which is then depended on by both wct4xxp and the opvxd511 (iirc) .ko modules. |
| 08:35.09 | jkroon | this avoids a parallel build issue in the kernel sources and allows for object sharing between the two drivers. |
| 08:35.46 | jkroon | is this something that can potentially be (partially) merged into dahdi-linux and then the remainder (opvxd511 specific stuff) into the sf branch? |
| 08:35.59 | ChannelZ | Oooh NM, looks like sox can probably do it. |
| 08:36.33 | jgowdy | Cool, I think I figured this issue out |
| 08:36.35 | jgowdy | https://issues.asterisk.org/jira/browse/ASTERISK-19303 |
| 08:37.10 | jgowdy | chan_sip.c is quite the tangled web |
| 08:37.13 | jkroon | tzafrir, https://bugs.gentoo.org/404407 and then specifically: https://404407.bugs.gentoo.org/attachment.cgi?id=303645 |
| 08:40.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:42.00 | Chainsaw | leifmadsen: Any chance of the patch on https://issues.asterisk.org/view.php?id=18010 being applied now that there is a test report and detailed bug linked in? |
| 08:42.28 | tzafrir | jkroon, generally looks good. I'll look it up more closely |
| 08:43.18 | jkroon | thanks tzafrir |
| 08:43.55 | Chainsaw | Ah, Leif isn't here. I shall wait. |
| 08:44.13 | jkroon | tzafrir, you want me to mail the patch split into dahdi-linux portion and non-digium portions to you? |
| 08:44.18 | Chainsaw | (Reason I ask is because the patch looks obviously correct with a capital O. I have had it in the distro patchset for years.) |
| 08:45.27 | Dovid | morning |
| 08:45.28 | tzafrir | jkroon, yes, please do |
| 08:46.27 | jkroon | tzafrir, any specific MODULE_AUTHOR() that I should use? |
| 08:52.00 | tzafrir | octasic's. I don't remember exactly what it is |
| 08:59.15 | jkroon | tzafrir, should be in your email. |
| 09:03.06 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 09:03.09 | jkroon | tzafrir, i suspect that it has something to do with the hardware echo canceller present on the wct4xxp module. |
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| 09:12.54 | Dovid | is there anyone in Japan here? |
| 09:16.55 | phix | hey the globals section in extensions.conf. are they global to that file alone or can I use the same vairables in queues.conf or other configuration files? |
| 09:18.15 | *** join/#asterisk krotos (~d3v1l@157.27.178.26) |
| 09:18.21 | krotos | hi all |
| 09:19.07 | phix | hhihi |
| 09:19.30 | kaldemar | phix: they are global inside dialplan. variables are not tied to files, but dialplan and channels. |
| 09:19.46 | phix | ah |
| 09:19.57 | phix | so does that mean I can access them within queues.conf or not? |
| 09:20.08 | phix | that is part of the dial plan right? |
| 09:20.45 | kaldemar | queues.conf is not dialplan. |
| 09:20.59 | phix | ok |
| 09:21.24 | kaldemar | if you want to use dialplan variables in queue config, use them as application arguments. |
| 09:24.20 | phix | any where I can define global variables that are actaully global within all conf files? |
| 09:25.51 | kaldemar | there is no such concept. what exactly are you trying to set? |
| 09:26.27 | phix | location to custom sounds / prompt / messages |
| 09:26.42 | phix | I guess I can just use a symlink in /var/share/asterisk/sounds or whatever it is kept |
| 09:27.04 | kaldemar | or set language in dialplan to what you want. |
| 09:27.26 | phix | yeah language is set |
| 09:27.34 | phix | I Just want to put in custom sounds |
| 09:27.36 | kaldemar | then it will look for sound files in /var/lib/asterisk/sounds/<lang>/ |
| 09:28.26 | phix | ln -s /srv/asterisk/my/custom/sound/location /var/lib/asterisk/sounds/$LANG/MyCustomSoundLocation |
| 09:28.32 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 09:28.33 | phix | solved |
| 09:33.11 | anny__ | when 2 sip endpoints (not behind a nat) are doing video/audio chat, the media doesn't pass in asterisk right? |
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| 09:45.41 | kaldemar | anny__: depends on configurations. |
| 09:46.25 | anny__ | kaldemar: yes i know, but i read that normally asterisk shouldn't relay any media unless nating is involved or there is other scenarios? |
| 09:46.56 | *** join/#asterisk IronMania (~IronMania@mail.it-ag.de) |
| 09:47.16 | nny | i am researching building a gateway between my asterisk box and multiple providers using LCR. The idea would be to only handle the session negotiation and not the media stream. Any suggestions on software or using Asterisk? I am looking outside of asterisk for this as well |
| 09:51.40 | kaldemar | anny__: the default setting for directmedia (or the old config parameter canreinvite) is yes, which causes asterisk to setup the media directly between caller and callee. there are also other options that affect the behavior, such as app Dial options t and T. |
| 09:52.31 | kaldemar | anny__: if asterisk needs to listen for inband DTMF for example, it forces media to go through asterisk even if directmedia is set to yes. |
| 09:53.37 | IronMania | I want to add a functionality to asterisk. So I want an app nearly the same like app_echo.c. When I created it, do I only need to compile or do I need to do something else? |
| 09:53.45 | nny | kaldemar: how does asterisk determine if it needs to listen for DTMF? |
| 10:01.22 | kaldemar | nny: if some option that causes it is enabled. |
| 10:01.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:02.04 | nny | kaldemar: ahh, assuming this is if the Dial statement calls a feature in features.conf, for example |
| 10:03.10 | kaldemar | Dial doesn't call any features. features may be enabled by setting them to the DYNAMIC_FEATURES channel variable. |
| 10:05.06 | nny | kaldemar: yeah that's what I meant, was looking at how it was implemented. If you have a DYNAMIC_FEATURE set in your dialplan, does that cause asterisk to listen still? I am setting up a system that is very simple, but has a dynamic_Feature. Wondering what the lowest overhead way is to handle this |
| 10:08.25 | nny | kaldemar: I should add I already intend to use the same codec on both ends of the channel |
| 10:09.39 | kaldemar | dynamic features most likely cause asterisk to stay on the media path. if DTMF is sent with SIP, then maybe not. this depends on many factors. |
| 10:10.04 | nny | kaldemar: assuming you mena inband vs rcf2833 |
| 10:10.45 | nny | kaldemar: good to know though, i'll do some more testing. Been working on some setups that really only need to act as the sip session negtioator/ interpreter and do as little translation/overhead as possible |
| 10:11.07 | kaldemar | rfc2833 uses RTP/RTCP, so it should really behave the same as inband in this case. |
| 10:12.58 | nny | kaldemar: hmm yeah that makes sense actually, the dmtf exists in the rtp stream, does info also require asterisk to sit in between? |
| 10:14.45 | kaldemar | yes. |
| 10:15.28 | nny | hmm guess that means there's no way to interpret dtmf without asterisk sitting in stream |
| 10:16.54 | nny | reading the viop info entry on it now as well |
| 10:16.58 | kaldemar | nny: i did mention using SIP for DTMF earlier. you might want to try that. |
| 10:17.58 | nny | kaldemar: this may be negated in at least one scenario anyways, the sip clients using the system are all behind NAT, not sure if they can successfully connect to the provider without asterisk in between, something else i need to research |
| 10:18.08 | kaldemar | but i'm not sure how the decision is implemented. the decision might be done on feature existence alone, despite the used dtmfmode. |
| 10:18.54 | nny | kaldemar: i'm also gonna look into how much overhead asterisk uses per channel with no translation, I may be ok with just having no codec translation but asterisk still sitting in stream |
| 10:20.22 | nny | With native bridging, the audio flows outside of Asterisk between the endpoints. With P2P bridging, the audio comes into the RTP layer of Asterisk but does not pass through the Asterisk core. This allows for Asterisk to intercept DTMF or play warning files to the bridged parties. |
| 10:20.27 | nny | probably what I am looking for |
| 10:21.05 | nny | assuming it will work where the asterisk server and termination provider are both on ap ublic interface but the sip client is behind NAT |
| 10:22.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:33.17 | nny | kaldemar: one last question, maybe you can answer, unrelated to my previous topic. I am working on a system that I configured from scratch (no example configs) and have many modules under noload=. I don't have any CLI output, only logging to messages. What did I miss? |
| 10:35.11 | wdoekes2 | logger.conf console=> settings? |
| 10:35.30 | nny | wdoekes2: ha.. missed that indeed. thanks |
| 10:35.40 | nny | wdoekes2: never touched that aspect before :\ |
| 10:35.57 | nny | or rather, only to isolate log messages, no console. Crap that's handy though |
| 10:47.11 | nny | what's a good way to see what kind of media handling is happening on a channel? core show channel and sip show channel i can't really determine from the vars presented, i may be missing it |
| 10:51.43 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 10:55.37 | nny | i guess what i am asking is what is the best way to determine from cli how asterisk is handling the rtp streams? |
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| 11:10.24 | IsUp | hi |
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| 12:28.01 | gestahlt | Hi |
| 12:28.16 | gestahlt | Anyone had experience with Asterisk -> remote capi to bintec? |
| 12:28.23 | gestahlt | im a bit stuck at the config |
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| 12:53.17 | WIMPy | ~ask |
| 12:53.17 | 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. |
| 12:55.41 | gestahlt | Okay |
| 12:56.48 | gestahlt | Iḿ Trying to use a Bintec Bingo ISDN Gateway as ISDN Trunk. I use Asterisk 1.6.2 and id like to know how to configure a remote capi to the gateway. |
| 12:56.55 | gestahlt | better? |
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| 13:02.38 | kaldemar | gestahlt: gateway between ISDN and what? |
| 13:03.21 | gestahlt | kaldemar: LAN -> ISDN |
| 13:03.28 | gestahlt | kaldemar: ISDN -> PTSN |
| 13:04.02 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 13:04.12 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:04.40 | gestahlt | kaldemar: I basically want to use it as a LAN ISDN interface to create a ISDN Trunk and us that for telephony |
| 13:07.48 | kaldemar | what is the device on the LAN side? |
| 13:08.11 | gestahlt | kaldemar: ?? |
| 13:08.58 | gestahlt | Kaldemar: Asterisk Server - LAN - > Bintec Bingo - ISDN - > PTSN |
| 13:10.27 | kaldemar | asterisk and the device need a common protocol on the LAN side. what is it? |
| 13:11.09 | gestahlt | TCP? |
| 13:11.44 | gestahlt | Or are you talking about the G 721? |
| 13:11.57 | kaldemar | no and no. |
| 13:13.29 | WIMPy | gestahlt: Well you first need the CAPI connection, but that is device specific. |
| 13:13.58 | gestahlt | Wimpy: and thats my problem. I dont know what to use |
| 13:14.10 | gestahlt | Wimpy: Im trying to get chan_capi running |
| 13:14.13 | WIMPy | You need to ask Bintek. |
| 13:14.22 | WIMPy | They are the only ones who can answer that. |
| 13:14.31 | kaldemar | seems it is speaking CAPI on the LAN side. |
| 13:14.35 | gestahlt | Wimpy: well, i read that chan_capi is using the bintec protocoll for remote capi |
| 13:14.52 | gestahlt | wimpy: but i cant get it configured |
| 13:15.27 | WIMPy | chan_capi includes a remote capi for Bintek? |
| 13:15.44 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:16.34 | gestahlt | wimpy: well, some say so |
| 13:16.44 | gestahlt | wimpy: im not sure myself |
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| 13:38.21 | IronMania | my asterisk modules were not compiled with the same compile-time otpions as in this aserisk. I already recompiled, but that didn't help. what can I do? |
| 13:38.40 | kaldemar | IronMania: install the newly compiled modules. |
| 13:40.05 | IronMania | how? I have ubuntu |
| 13:40.11 | kaldemar | IronMania: actually, you might have some module in there that you don't even have selected with your current install. delete all modules from /usr/lib/asterisk/modules or where you have installed them to get rid of such and re-install. |
| 13:40.23 | IronMania | ok |
| 13:40.24 | IronMania | thx |
| 13:40.28 | kaldemar | IronMania: did you install from source? what version do you have? |
| 13:40.35 | IronMania | i installed from source |
| 13:40.41 | IronMania | 10.1.3 |
| 13:41.04 | kaldemar | rm all modules and do a "make install". |
| 13:41.11 | IronMania | ok |
| 13:41.49 | IronMania | thank you |
| 13:43.55 | IronMania | that worked |
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| 13:58.25 | gestahlt | grrrrr |
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| 14:17.14 | IronMania | if I want to call my asterisk from my sip phone, what do I need to do? I installed, added extension.ael one extension I want to call. so when I call the asterisk I didn't even get the Log(NOTICE,"test") in my asterisk |
| 14:17.22 | IronMania | what did i forget? |
| 14:20.18 | leifmadsen | IronMania: is sip.conf configured to point to the correct context? does the phone authenticate? is the phone pointed at the right server? do you see an INVITE come into Asterisk when you place a call (with sip debug enabled)? -- there are many reasons why it might not be working. |
| 14:20.23 | leifmadsen | I suggest narrowing it down |
| 14:21.43 | IronMania | sip debug could help ... i try thanks |
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| 14:23.58 | IronMania | how can I turn that on? |
| 14:24.17 | bulkorok | in cli type: sip set debug on |
| 14:24.18 | *** join/#asterisk Greenlight (~wluke@cpc3-dund11-2-0-cust462.sgyl.cable.virginmedia.com) |
| 14:24.23 | bulkorok | or use tcpdump |
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| 14:24.42 | *** mode/#asterisk [+o file] by ChanServ |
| 14:25.22 | IronMania | so when I don't have that command, obvisouly my sip module is not correctly loaded? |
| 14:26.47 | leifmadsen | IronMania: module show like sip |
| 14:26.57 | Greenlight | Hi folks. I've a TE410P with 4 E1's connected. One of the E1's has gone down and it's showing RED alarm, however Asterisk is still trying to place calls over it, and it's obviously failiing. Is there a way to prevent it trying to use this E1 which is out of service, without resarting Asterisk/DAHDI? |
| 14:27.33 | IronMania | 0 |
| 14:27.38 | bulkorok | Greenlight: Did you unplug the cable?! |
| 14:28.03 | Greenlight | No, however now you mention it, I feel sily for not having tried it - will that work ? |
| 14:28.07 | Greenlight | *silly |
| 14:29.01 | bulkorok | Greenlight: I had that situation a long time ago... it worked... but I don't know why ^^ |
| 14:29.07 | Greenlight | "pri show spans" lists: |
| 14:29.12 | Greenlight | PRI span 1/0: Up, Active |
| 14:29.12 | Greenlight | PRI span 2/0: In Alarm, Up, Active |
| 14:29.12 | Greenlight | PRI span 3/0: Up, Active |
| 14:29.12 | Greenlight | PRI span 4/0: Up, Active |
| 14:29.24 | Greenlight | SO you reckon unplug and it'll go Down ? |
| 14:29.26 | IronMania | my chan_sip.so module is missing |
| 14:29.43 | [TK]D-Fender | IronMania, as in not in the modules folder at all? Or jsut not loaded? |
| 14:29.51 | bulkorok | I think the Up, Active makes Asterisk using the span... |
| 14:29.55 | [TK]D-Fender | IronMania, "module load chan_sip.so |
| 14:30.08 | bulkorok | Greenlight: unplugging caple should make it down ^^ |
| 14:30.12 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 14:30.18 | Greenlight | Yea seems like sound logic, I'll try unplugging it. Many thanks! |
| 14:30.32 | bulkorok | :) |
| 14:31.07 | IronMania | it is not loaded |
| 14:31.34 | IronMania | second, let me try |
| 14:31.57 | IronMania | its gone |
| 14:32.04 | IronMania | I don't have it in the folder |
| 14:33.12 | IronMania | i installed from source. did I forgot to install a package? |
| 14:35.13 | kaldemar | IronMania: do a "make menuselect" and see if chan_sip is selected under "Channel Drivers". |
| 14:36.12 | IronMania | there are XXX |
| 14:36.14 | IronMania | on this one |
| 14:36.54 | bulkorok | then you don't have the desired packeges installed |
| 14:37.07 | IronMania | ok |
| 14:37.12 | IronMania | i check the dependencies |
| 14:37.13 | kaldemar | interesting. chan_sip should only really depend on chan_local. |
| 14:37.28 | kaldemar | IronMania: have you run the configure script? |
| 14:37.56 | IronMania | yes |
| 14:38.07 | IronMania | it also depends on res_crypto, which depends on openssl |
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| 14:38.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:41.18 | bulkorok | check the steps 1 and 2 if installing from source: https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source |
| 14:42.27 | IronMania | i didn't install DAHDI and libpri |
| 14:42.32 | bulkorok | take espacially care of the -dev packages! |
| 14:43.01 | bulkorok | (dahdi and libpri is not that necessary) |
| 14:43.46 | IronMania | jippi, i could select it |
| 14:43.49 | IronMania | now it should install |
| 14:43.51 | IronMania | thank you! |
| 14:44.31 | IronMania | I forgot openssl-dev |
| 14:45.08 | bulkorok | great! |
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| 14:51.59 | Chainsaw | leifmadsen: Any chance of the patch on https://issues.asterisk.org/view.php?id=18010 being applied now that there is a test report and detailed bug linked in? |
| 14:52.53 | leifmadsen | Chainsaw: did you check your memo I sent the other day? |
| 14:57.19 | Chainsaw | leifmadsen: I did not receive any memos from you I'm afraid. |
| 14:57.40 | Chainsaw | leifmadsen: Occasionally I get a "you have a new memo" notifier, which when I read it results in a memo from steev that I received in 2009. |
| 14:57.56 | leifmadsen | there you go |
| 14:58.01 | leifmadsen | just notified that you read it |
| 14:58.03 | Chainsaw | leifmadsen: Ah, it is 3, not 1. |
| 14:58.29 | Chainsaw | leifmadsen: What a shame. You were a good bug marshal. I shall be bothering others instead. |
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| 15:04.17 | IronMania | can someone help me with the apps/app_echo.c ? |
| 15:05.00 | leifmadsen | what is your real question? |
| 15:05.13 | IronMania | i want to add a delay to the echo. so I would buffer the packets in a list and than play them delayed |
| 15:05.14 | leifmadsen | it's implied that someone will help if they can |
| 15:05.46 | leifmadsen | on Asterisk 10, there is a JITTERBUFFER() function, or you can potentially use a Local channel on earlier versions with the /j flag |
| 15:05.47 | IronMania | i assume i must send empty packages first and than send the delayed correct |
| 15:06.07 | IronMania | i try jitterbuffer first |
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| 15:07.45 | Twitchnln | morning |
| 15:09.28 | Twitchnln | has anyone got any experience with setting up cisco 7960 sccp handsets using chan_sccp? I have one setup and working but am having some difficulty getting the line keys mapped correctly currently it will only take advantage of line 1, I would like to be able to use all 6 |
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| 15:20.35 | Greenlight | I tried unplugging that E1 line, but it didn't seem to work, Asterisk still seen it as "Active". I've now swapped it with the 4th E1 so that it doesn't effect things. Im thinking though I must be doing something wrong, is there no way to tell Asterisk to not attempt to use a PRI span which is in RED alarm ? |
| 15:22.44 | kaldemar | Greenlight: that needs to be a dialplan change. if you try to dial a channel, asterisk will try to use it, what ever its state is. |
| 15:24.28 | IronMania | leifmadsen: worked perfect. just what I wanted. |
| 15:24.30 | IronMania | thanks |
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| 15:25.11 | leifmadsen | IronMania: np |
| 15:25.15 | IronMania | when upgrading from 1.6.2 to 10.1.3, do I need to considere something or will a upgrade work. |
| 15:26.18 | kaldemar | IronMania: take a look at UPGRADE*.txt in the 10.1.3 source package for config and/or syntax changes. |
| 15:26.37 | IronMania | thx |
| 15:26.45 | Greenlight | So, I split the E1's into seperate groups and have the dialplan failover? |
| 15:29.58 | kaldemar | Greenlight: app ChanIsAvail before a Dial or checking DIALSTATUS variable after a Dial will help with that. |
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| 15:45.26 | Dovid | i have two asterisk boxes. they both have G729 and G711A. trying to figure out why one of them keeps flipping between G729 and G711A |
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| 15:51.49 | k3asd` | hi |
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| 16:16.06 | bbourdage | I am getting a mmetme application not available when I try and use page in the dialplan. I have a meetme.conf, do I need to complile something extra , or load a special module ? |
| 16:20.23 | malcolmd | sounds like app_meetme.so isn't loaded or doesn't exist |
| 16:20.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:22.10 | bbourdage | You are correct, it does not exist !. I tried to search apt-get for a meetme option but could not find one ? |
| 16:26.23 | *** join/#asterisk juninhogr (~juninhogr@187.59.183.30) |
| 16:27.22 | leifmadsen | app_meetme requires asterisk to compiled against dahdi |
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| 16:29.29 | bbourdage | Thanks Leif, I am currently using the debian base package, are you reccomending that I add asterisk-dahdi to the machine ?, can I do that live ?, |
| 16:30.25 | juninhogr | hello everyone... does someone have experience with use of a linksys voice gateway spa3102? |
| 16:30.51 | leifmadsen | bbourdage: I don't know how the packages work sorry |
| 16:31.00 | bbourdage | Leif, |
| 16:31.30 | leifmadsen | but meetme requires dahdi |
| 16:31.32 | bbourdage | <PROTECTED> |
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| 17:13.30 | Chainsaw | mjordan: Any chance of the patch on https://issues.asterisk.org/view.php?id=18010 being applied now that there is a test report and detailed bug linked in? Obviously correct comes to mind when I see it... |
| 17:14.14 | mjordan | Is there an associated Asterisk issue? |
| 17:14.18 | mjordan | in JIRA |
| 17:15.59 | Chainsaw | mjordan: Two in Mantis, but no, none in JIRA. |
| 17:16.10 | russellb | well everything from mantis was migrated. |
| 17:16.26 | Chainsaw | Mantis 17982 & 18010. |
| 17:16.41 | russellb | 18010 -> https://issues.asterisk.org/jira/browse/18010 |
| 17:17.17 | Chainsaw | russellb: Neat, thanks. The associated report is linked as well. |
| 17:18.41 | *** join/#asterisk gusto (~gusto@nrbg-4dbe33d9.pool.mediaWays.net) |
| 17:19.04 | mjordan | Chainsaw: so, app_alarmreceiver is an extended support module, which means that development effort for it typically comes from the open source community contributors |
| 17:19.19 | Chainsaw | mjordan: This effort has been put in, a patch is attached. |
| 17:19.40 | Chainsaw | mjordan: Another, independent community member has given the patch the okay. |
| 17:19.42 | gusto | so |
| 17:19.46 | gusto | i have a problem |
| 17:19.46 | mjordan | correct. The next step would be to either have a developer with commit access review the patch and commit it, or preferably, put it up on Review Board |
| 17:20.02 | gusto | i have recieved a VoIP account from my provider, but it does not resolve |
| 17:20.02 | Chainsaw | mjordan: I was hoping that your control panel had a button that did that, yes. |
| 17:20.07 | mjordan | nope |
| 17:20.35 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States |
| 17:20.48 | gusto | seems to be somehow referring to a SOA and giving no answers to A records |
| 17:21.01 | Chainsaw | sighs briefly and will continue to apply this patch downstream then |
| 17:21.26 | mjordan | well, its not game over yet :-) If you contact some folks in #asterisk-dev, there's a chance someone would be willing to assist you with getting the patch in |
| 17:21.59 | Chainsaw | mjordan: It is not the first time that I have drawn attention to this low hanging fruit. |
| 17:23.26 | gusto | my sipproxy is sipproxy.endesha.be-converged.com and registar endesha.be-converged.com ... it works from every internet connection, also from different providers and so on ... but was only reported to run on AVM FritzBox, so there is something the FritzBox must do differently with the SIP to get a connection |
| 17:25.22 | mjordan | Chainsaw: you may get some additional traction if you post the patch to Review Board |
| 17:25.40 | mjordan | that's typically where patches are looked over and get vetted. |
| 17:26.03 | mjordan | if you'd like, I can set you up an account |
| 17:28.42 | Chainsaw | mjordan: My existing JIRA account seems to work there. |
| 17:29.00 | Chainsaw | mjordan: It's on my todo list, thank you. |
| 17:33.34 | Twitchnln | anyone played with chan_sccp? would love some pointers…. |
| 17:35.15 | gusto | does anyone have a VoIP provider who uses domain names that do not resolve to IP addresses? |
| 17:38.42 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
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| 17:39.02 | chuckf | gusto: all valid domain names resolve to ip addresses |
| 17:41.47 | Qwell | chuckf: not true |
| 17:43.43 | chuckf | Qwell: really? I've not run into any that I'm aware of |
| 17:44.08 | gusto | chuckf: try sipproxy.endesha.be-converged.com |
| 17:44.58 | chuckf | gusto: I get a host not found |
| 17:45.19 | gusto | chuckf: yes |
| 17:45.25 | *** join/#asterisk twodogs (~twodogs@74.200.253.228) |
| 17:45.34 | gusto | chuckf: that's what i get when i want to set up my VoIP account |
| 17:46.00 | *** part/#asterisk twodogs (~twodogs@74.200.253.228) |
| 17:48.05 | chuckf | gusto: it may be that there is something in the code on the FritzBox that translates the sipproxy.endesha.be-converged.com to an IP address. |
| 17:49.05 | gusto | chuckf: that idea came to me as well |
| 17:49.09 | chuckf | Qwell: and when I said that about valid domain names I was talking in terms of those accessable via the intertubes |
| 17:50.03 | Qwell | chuckf: still not true |
| 17:50.29 | chuckf | Qwell: do you have an example? |
| 17:50.45 | chuckf | is genuinely interested |
| 17:51.00 | Qwell | http://en.wikipedia.org/wiki/Reverse_DNS_lookup#Records_other_than_PTR_records |
| 17:51.10 | _Corey_ | chuckf: The common practice of resolving a domain name pretty began when people became lazy and stopped typing www. to reach websites |
| 17:51.20 | Qwell | also you don't need an A or AAAA record for PTR |
| 17:51.54 | Qwell | _Corey_: if you ever feel like being a guinea pig, I've got something for you to try. :p |
| 17:52.04 | gusto | but we are not talking about PTR, who cares about that .. |
| 17:52.30 | Qwell | gusto: I wasn't saying we were - I was just correcting an incorrect statement. |
| 17:52.35 | _Corey_ | Qwell: As long as it doesn't involve DNS ;) |
| 17:52.52 | Qwell | _Corey_: nope, AsteriskNOW |
| 17:53.14 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:53.29 | gusto | but no ptr record does not make a machine inaccessible |
| 17:53.46 | gusto | i have several domain names without ptr, but they all show to an IP |
| 17:53.56 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:54.01 | _Corey_ | Qwell: malcolmd sent me an up-to-date one recently I've got running... let me know what you need |
| 17:54.03 | gusto | now we have the opposite ... we have a domain that shows to nothing |
| 17:58.28 | *** join/#asterisk tokozedg (bca97ef4@gateway/web/freenode/ip.188.169.126.244) |
| 17:59.20 | tokozedg | hello, anyone can suggest normal application for monitoring asterisk live channels in realtime. Tried all I found, but nothing useful. Thanks |
| 18:02.21 | chuckf | Qwell: ultimatly you need to have an ip address at the end of the DNS translations to get anywhere. |
| 18:02.30 | Qwell | chuckf: untrue |
| 18:03.08 | drudge` | whos using that new fpbx gui |
| 18:03.14 | Qwell | new? |
| 18:04.05 | drudge` | the 10rc3 or whatever they call it |
| 18:04.21 | chuckf | Qwell: so if my server has no ip address, there is a way to type in chuckf.com<assuming that's a valid name> and reach it? |
| 18:05.58 | Qwell | chuckf: You are now saying 2 different things. |
| 18:06.50 | Qwell | Fine - you want an example? dig will never give you an address for qwell.local |
| 18:07.28 | Qwell | However, you could send that through something, and be able to reach it. |
| 18:08.31 | chuckf | Qwell: no, I'm not. In order to get from a dns name to a box somewhere ultimately it is translated from the name to the ip. Maybe not a 1:1 relationship, meaning it may go from name-name-ip but ultimatly it is translated |
| 18:09.17 | Qwell | That is not the same statement you made earlier. :) |
| 18:09.36 | *** part/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 18:10.38 | chuckf | I made a simple statement at first and just didn't clairfiy it enough apparently with the followup |
| 18:13.28 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:15.56 | *** join/#asterisk twodogs (~twodogs@hendra.biohazard.seattle.wa.us) |
| 18:25.45 | jgowdy | I'm unfamiliar with how quickly patches can be merged in Asterisk, is there any possibility that my patch for https://issues.asterisk.org/jira/browse/ASTERISK-17929 and https://issues.asterisk.org/jira/browse/ASTERISK-19303 will make it into 10.2 before it releases? |
| 18:26.11 | Qwell | jgowdy: not unless it's a regression from 10.1 |
| 18:26.31 | jgowdy | It's a regression from before that |
| 18:26.41 | jgowdy | 1.8 and 10 |
| 18:27.36 | jgowdy | Several of the INVITE failure scenarios call transmit_response_reliable without setting the state in such a way that ACK won't be ignored. Thus you get retransmit until timeout in all those scenarios. |
| 18:27.40 | Qwell | I think that was committed already? |
| 18:27.51 | Qwell | or a very similar issue.. see 10.2.0-rc3 |
| 18:27.54 | jgowdy | It's not the same as what was fixed in rc3 |
| 18:28.00 | mjordan | Qwell: nope. Its targeted for 1.8.12 / 10.3.0 |
| 18:28.29 | jgowdy | mjordan: the fix I'm talking about is targeted for 1.8.12 and 10.3? |
| 18:28.30 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 18:28.51 | mjordan | Asterisk sending a 481 for an ACK? |
| 18:29.23 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 18:29.25 | jgowdy | No, in several of the 481 scenarios, it uses transmit_response_reliable but then ignores the ACKs, so it retransmits 481 over and over until timeout |
| 18:29.47 | jgowdy | Because in the 481 scenarios, it doesn't setup the state such that handle_incoming won't disregard the ACK |
| 18:29.48 | mjordan | which is because it doesn't recognize the call-id in the ACK |
| 18:30.02 | jgowdy | no, it's because pendinginvite isn't set |
| 18:30.04 | mjordan | regardless of the root cause, that issue should be worked this month, which puts it on schedule for 1.8.12 |
| 18:30.06 | jgowdy | the call ID matches |
| 18:30.08 | mjordan | and 10.3 |
| 18:30.19 | jgowdy | handle_incoming won't honor the ACK if pendinginvite is 0 |
| 18:31.19 | jgowdy | and SIP pendantic bitches and derails us because initreq isn't set either, if you have pedantic on, which obv is the 1.8+ default |
| 18:32.19 | jgowdy | by setting initreq and pendinginvite in the error block of the first half of handle_request_invite, a whole class of scenarios of retransmit-until-timeout is fixed |
| 18:33.11 | jgowdy | really anywhere that transmit_response_reliable is called in handle_request_invite, we should ensure that initreq and pendinginvite are set properly |
| 18:36.40 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:37.21 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-92-161.lsanca.fios.verizon.net) |
| 18:38.19 | _Corey_ | Qwell: All is normal after that firmwared update, btw |
| 18:38.24 | Qwell | yay! |
| 18:38.34 | _Corey_ | Thanks |
| 18:46.59 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:52.23 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:00.57 | *** join/#asterisk ajg (~inaki@wah.dreamchaos.net) |
| 19:02.07 | jgowdy | Are there any plans still active to replace chan_sip with a "chan_sofia" ? |
| 19:02.45 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:04.58 | ajg | hah.. here's a fun one.. I'm setting up a new asterisk 1.8 server.. I can get a softphone (3cx) to register just fine IF the SIP account name is set to the extension. Otherwise it says no peer found |
| 19:05.02 | ajg | any ideas? |
| 19:08.44 | leifmadsen | jgowdy: I'm not sure there were any "plans" to do it |
| 19:09.21 | leifmadsen | ajg: look at the sip trace and determine what the problem is |
| 19:09.23 | p3nguin | ajg: You misconfigured your device in sip.conf. |
| 19:09.28 | leifmadsen | and that |
| 19:09.57 | p3nguin | You called it [your-extension-number] instead of something more appropriate. |
| 19:09.59 | jgowdy | leifmadsen: cool, just curious |
| 19:10.55 | ajg | p3nguin: I've tried calling it other things and it breaks when I do that (changing the name in sip.conf, extensions.conf, and the ID on the softphone) |
| 19:11.58 | leifmadsen | jgowdy: the Asterisk SCF project is using pjsip, so my guess is, if anything were to replace chan_sip eventually, it would be that |
| 19:12.07 | leifmadsen | common code bases across common projects is a good thing (tm) |
| 19:12.35 | leifmadsen | the other thing to consider is licensing costs since Asterisk is a dual-licensed project |
| 19:12.37 | ajg | hmm correction.. using users.conf instead of sip.conf |
| 19:12.44 | p3nguin | ~users.conf |
| 19:12.44 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 19:12.53 | p3nguin | There's your mistake. |
| 19:13.10 | leifmadsen | wow, harsh :) |
| 19:13.10 | *** part/#asterisk sysdef (sysdef@debiancenter/founder.developer/pdpc.professional.sysdef) |
| 19:13.14 | ajg | haha |
| 19:13.15 | ajg | okay |
| 19:13.43 | p3nguin | Do people who DON'T use FreePBX or Asterisk GUI ever use users.conf? |
| 19:14.06 | leifmadsen | no :) |
| 19:14.20 | leifmadsen | harsh, but not necessarily wrong |
| 19:14.30 | ajg | aah.. that'd explain where users.conf came from |
| 19:14.55 | leifmadsen | it's basically an experiment gone wrong |
| 19:15.05 | ajg | good to know |
| 19:15.18 | jgowdy | oh wow, I just discovered Asterisk SCF |
| 19:15.20 | jgowdy | glorious |
| 19:16.00 | ajg | alright.. let's see if this behaves a little less dumb if I remove asterisk gui and just do a stock server |
| 19:16.55 | p3nguin | jgowdy: Discovered... like the white man discovered North America? ;) |
| 19:17.21 | leifmadsen | jgowdy: check out www.astricon.net in the archives for the presentation(s) around Asterisk SCF. There is an introduction to it in the developers track on day one (video) you can watch. If the whole discussoin is there, I suspect it is at least a couple hours of data |
| 19:17.37 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.168) |
| 19:17.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:18.00 | b0ot | Is it possible to transode g711 call to g729 when going over a trunk |
| 19:18.12 | leifmadsen | jgowdy: there is also a lot of documentation being written on the wiki at http://wiki.asterisk.org |
| 19:18.18 | leifmadsen | note the project is in the early stages |
| 19:18.30 | p3nguin | b0ot: If you mean is it possible to transcode g711 to g729, the answer is yes. |
| 19:18.42 | leifmadsen | you will need a license to transcode g729 |
| 19:18.59 | leifmadsen | (as an elaboration on p3nguin's response) |
| 19:19.18 | p3nguin | You don't transcode in the middle of a channel's path... asterisk performs the transcoding. |
| 19:19.35 | b0ot | how much is a liscense |
| 19:19.41 | leifmadsen | $10 per channel |
| 19:19.54 | b0ot | interesting |
| 19:19.54 | leifmadsen | simultaneous channel |
| 19:19.57 | p3nguin | Asterisk does not care what is on the other end of a channel -- if the end point supports the codec, it will speak that codec to it. |
| 19:20.13 | jgowdy | Yes, I realize I'm timeline.jpg to this. I'm new to Asterisk development, but I think I could be a fairly strong contributor. |
| 19:20.33 | leifmadsen | jgowdy: that's good news! Just wanted to set expectations and point you at the best information |
| 19:20.37 | jgowdy | Nice that SCF is in C++, clean up a lot of these goto error_path things |
| 19:20.44 | jgowdy | much appreciated |
| 19:20.52 | leifmadsen | jgowdy: there is #asterisk-scf and #asterisk-scf-dev fyi |
| 19:21.12 | jgowdy | thx |
| 19:22.59 | b0ot | !thebook |
| 19:23.01 | b0ot | !book |
| 19:23.33 | _Corey_ | ~thebook |
| 19:23.33 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:23.44 | _Corey_ | b0ot: ~ is what you wanted |
| 19:23.52 | b0ot | yep |
| 19:24.28 | *** join/#asterisk Meaulnes (~kirk@69.64.7.98) |
| 19:26.03 | b0ot | do you need to install zaptal seperatly for conf or is it included normally with sudo apt-get install asterisk |
| 19:28.55 | tzafrir | you don't need to install zaptel. You may need to install dahdi |
| 19:32.57 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 19:33.36 | *** join/#asterisk Gestahlt (~chatzilla@p5DCD197F.dip0.t-ipconnect.de) |
| 19:34.02 | Gestahlt | Hi |
| 19:34.13 | Gestahlt | Whats the difference between asterisk 10.xxx and 1.8.xxx=+# |
| 19:34.15 | Gestahlt | ? |
| 19:34.20 | Gestahlt | i dont get the version jump |
| 19:34.34 | wdoekes2 | ~asterisk10 |
| 19:34.34 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 19:36.36 | Gestahlt | Ah i get it |
| 19:36.54 | leifmadsen | Asterisk 1.10 |
| 19:36.57 | leifmadsen | s/1.// |
| 19:36.57 | Gestahlt | is it my imagination or have devs lately something weird in their head regarding version numbering? |
| 19:37.09 | Gestahlt | Just like Firefox |
| 19:37.16 | leifmadsen | just your imagination |
| 19:37.19 | Gestahlt | suddently they started to jump major versions |
| 19:37.27 | leifmadsen | numbers mean nothing really |
| 19:38.43 | Gestahlt | you are right |
| 19:38.47 | Gestahlt | doesnt really matter |
| 19:38.54 | Gestahlt | i think i try tomorrow version x then |
| 19:41.44 | *** join/#asterisk albertoandrade (~albertoan@187.59.65.22) |
| 19:45.12 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 19:46.19 | Gestahlt | whats the "best" distro to use with Asterisk? Where is the least pain? |
| 19:46.46 | Qwell | AsteriskNOW is easy |
| 19:46.53 | Qwell | but, use whatever you like |
| 19:49.06 | p3nguin | I just did an emergency deployment last night using AsteriskNOW. I had the guy at the site drop in the CD and boot from it, pressing 6 when prompted. |
| 19:49.19 | p3nguin | 20 minutes later, I was configuring phones and asterisk. |
| 19:49.59 | Qwell | p3nguin: man, somebody should buy the maintainer of AsteriskNOW a beer or something. |
| 19:50.05 | Qwell | I agree with Qwell! |
| 19:50.08 | p3nguin | haha |
| 19:50.22 | *** join/#asterisk k3asd` (~k3asd@host168-86-dynamic.21-87-r.retail.telecomitalia.it) |
| 19:50.38 | p3nguin | Is there a plan to put 1.8 in AsteriskNOW soon? |
| 19:51.10 | Qwell | p3nguin: 2.0.0 is available as of this morning. It has 1.8-digiumphones (1.8 + support for the stuff our new phones can use) |
| 19:51.18 | Qwell | well, 2.0.1-beta1 |
| 19:52.22 | Qwell | p3nguin: but, it's very very easy to move to 1.8 in existing installs |
| 19:55.42 | *** join/#asterisk xavia (~xavia@38.65.32.130) |
| 19:57.43 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 20:00.30 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 20:03.28 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 20:05.19 | Qwell | p3nguin: you probably missed my answers |
| 20:05.55 | leifmadsen | p3nguin: don't worry, you didn't miss much |
| 20:05.59 | leifmadsen | Qwell: oh burn! |
| 20:06.07 | Qwell | twss |
| 20:08.31 | p3nguin | The last thing I saw was that it is very easy to upgrade. |
| 20:10.45 | Qwell | p3nguin: that is all |
| 20:10.55 | Qwell | (see topic in #asterisknow - it's literally 1 command) |
| 20:14.58 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 20:15.21 | asilva | Hi, does anyone nows how to calculate the simultaneous calls based on CDR data ? |
| 20:16.00 | asilva | knows* |
| 20:31.55 | lanning | sure, the CDR has the start time and length |
| 20:32.01 | *** join/#asterisk sweat (~mike@c-50-138-44-89.hsd1.fl.comcast.net) |
| 20:32.59 | *** join/#asterisk aberrios (~Lucia@psiclik.plus.com) |
| 20:34.25 | aberrios | Hi, I have a question about Pacemaker configuration with an rseries device. In the provided Pacemaker.cfg file we should change primitive GatewayStatus ocf:pacemaker:ping \ |
| 20:34.25 | aberrios | 14 params host_list="10.0.0.1" multiplier="100" \ to reflect the ip of the Gateway, I wondered what "Gateway" referred to? The network gateway? The floating IP? |
| 20:35.39 | aberrios | Also how would I go about removing the pacemaker.cfg that I installed with "crm configuration update pacemaker.cfg" and replacing it with a newly edited pacemaker.cfg one? |
| 20:36.41 | Qwell | aberrios: The network gateway. |
| 20:36.48 | Qwell | You can also just do crm configure edit |
| 20:37.33 | aberrios | Qwell, thanks! I think i've buggered up the cluster config since everytime I try a "crm configure" i get something like "failed CIB signon" |
| 20:37.50 | Qwell | restart corosync on both boxes |
| 20:38.50 | sweat | aberrios: gateway to me always means the precise ip of my router at home.. |
| 20:39.37 | aberrios | sweat, mmmmm but with little current knowledge of pacemaker syntax "gateway" could mean a few things to me. Anyway I have the answer, thanks Qwell |
| 20:40.07 | aberrios | its been bugging me since earlier today. I'll get shell access back tomorrow and sort it out. Thanks Qwell. |
| 20:40.07 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 20:41.15 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 20:41.39 | Qwell | aberrios: I thought that was in the user guide |
| 20:41.53 | *** join/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 20:41.57 | *** part/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 20:42.35 | aberrios | Qwell, The user guide was very thorough but I don't remember reading what Gateway refered to, it just mentions something like "The gateway to ping"... iirc |
| 20:43.33 | Qwell | aberrios: ahh |
| 20:43.45 | Qwell | aberrios: it would usually just be the first address in a traceroute |
| 20:44.17 | Qwell | it really just depends what you want to check. it could very well be your Internet gateway address too, or just some random address on the Internet. |
| 20:44.24 | aberrios | Qwell, So that param tells Pacemaker to ping the network gateway so that it can tell if its the gateway at issue rather than a node network issue? |
| 20:44.56 | Qwell | aberrios: basically |
| 20:44.59 | sweat | I think I need to be set straight on some concepts, namely I'm here in #asterisk because I *think* I need it installed on my router to eliminate a latency i'm experiencing with SIP on my android phone.. If I had 3G on the phone and SIP still had a major latency or lag I could say precisely the lag is coming off of the router and I need some type of home voip catalyst like asterisk (i'm assuming.) I've never used asterisk or a PBX server.. |
| 20:46.23 | sweat | and hello everyone, sorry for not saying hi earlier |
| 20:46.31 | aberrios | Qwell, Would me putting the incorrect IP address in (I actually put node1's ip in there) for the gateway cause drbd to not mount /dev/drbd0 to /mnt/asterisk ? |
| 20:47.21 | Qwell | aberrios: no. but, you can't use the IP of a node, since the config is shared between servers. Node B would be pinging itself. |
| 20:49.18 | *** join/#asterisk wonderworld (~ww@dsdf-4db5143f.pool.mediaWays.net) |
| 20:49.34 | aberrios | Qwell, hmm, well I'm sure we did everything according to the manual except for the wrong gateway IP. Also a small problem with GRUB and the USB keys for drbd (solved in BIOS). But when the nodes booted asterisk wouldn't start on either because of brbd not mounting /dev/brbd0 |
| 20:50.15 | Qwell | unfortunately I haven't looked at any of that stuff for a few months |
| 20:50.56 | aberrios | Qwell, should /etc/init.d/drbd status show Secondary/Secondary or Primary/Secondary on first boot if everything is normal? |
| 20:51.06 | sweat | would asterisk eliminate a delay in calls I make with my SIP device? |
| 20:51.09 | aberrios | Qwell, I think more reading and playing is in order. |
| 20:51.23 | Qwell | aberrios: secondary/secondary means they're talking to each other, but neither is taking lead |
| 20:51.38 | Qwell | There may be something preventing it from trying to become master - possibly the gateway |
| 20:52.09 | Qwell | sweat: How could it? If there's a delay getting the packets... |
| 20:52.41 | aberrios | Qwell, if that is the case then that would explain why it wouldnt mount. When I force node1 to primary I can mount /dev/drbd0 |
| 20:54.21 | Qwell | aberrios: Pastebin the output of `crm status` |
| 20:54.42 | sweat | Qwell: how could I kill the delay in packets and would a 3G connection eliminate such a problem |
| 20:55.22 | Qwell | sweat: The delay would be caused by network latency. A faster connection would reduce the latency, yes. |
| 20:55.27 | aberrios | Qwell, I would if I had shell access. I left the site a few hours ago, but I've requested remote shell access. I've got more ammo to throw at the problem now. It just been bugging me all the drive home. Thanks for some answers! |
| 20:55.45 | Qwell | aberrios: ahh, k |
| 20:56.01 | sweat | Qwell: I've been scouring my router's configuration for a way to set RTP since I read something about packets having a long latency but I've hit a dead end because I don't see anything in it about RTP |
| 20:56.43 | sweat | Qwell: that's another thing, my internet at home is Cable, 1.5 megabits per second down and 256kb up |
| 20:57.04 | Qwell | 256k is sufficient for a call or two |
| 20:57.24 | sweat | Qwell: that's what I thought |
| 20:58.57 | sweat | Qwell: i'm using pbxes.org as a SIP provider and google voice as a trunk to dial out to landlines but here's the thing, the latency happens both on calls outside and to calls to the machine hosting the service, and it happens with sip2sip--a free provider I tried in addition to pbxes |
| 20:59.31 | Qwell | Get a less latent connection. That is your only choice. |
| 20:59.51 | sweat | Qwell: can you say that in a layman's way |
| 20:59.54 | sweat | :) |
| 21:00.11 | Qwell | Yell at your ISP to unclog the tubes. |
| 21:01.09 | sweat | Qwell: I thought that my connection is really great becuase it downloads nearly two megabytes a second and uploads over 200 kilobytes per second |
| 21:01.48 | sweat | Qwell: from what I see on the phone HD audio says something like 64kb |
| 21:01.58 | Qwell | 1.5mbit is fantastic for 1999 |
| 21:02.06 | sweat | seriouslY? |
| 21:03.08 | sweat | i use a docsis 2.0 modem on comcast |
| 21:03.20 | sweat | i think 20 megabits is the highest i can get at this time |
| 21:03.25 | p3nguin | 30+ |
| 21:03.49 | sweat | hmm |
| 21:04.17 | p3nguin | DOCSIS 2.0 is limited to 38 Mbits/s. |
| 21:04.32 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 21:04.34 | p3nguin | DOCSIS 2.0 is limited to 38 Mbits/s downstream. |
| 21:04.48 | p3nguin | DOCSIS 2.0 is limited to 27 Mbits/s upstream. |
| 21:05.21 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:05.39 | sweat | thank you that's interesting to me |
| 21:06.25 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 21:07.24 | p3nguin | If you go to DOCSIS 3.0, you get to start bonding channels together for that speed multiplied by as many channels as the cable provider gives you. |
| 21:07.51 | sweat | I think what I really need is money, if only I had a job |
| 21:07.59 | sweat | :3 |
| 21:08.00 | p3nguin | We currently have four channels on the downstream, and we have packages for 100 Mbit services. |
| 21:08.14 | sweat | are you shitting me |
| 21:08.19 | sweat | I can combine them |
| 21:08.25 | sweat | that's insane |
| 21:13.34 | asilva | Hi, does anyone knows how to calculate the simultaneous calls based on CDR data ? or know a tool that does that? |
| 21:15.24 | p3nguin | (1431.55) <lanning> sure, the CDR has the start time and length |
| 21:16.43 | asilva | that i know.. i meant like the sql code or a tool that would do without me having to struggle with SQL statements |
| 21:19.09 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:19.13 | [sr] | hwllow my friends |
| 21:19.35 | WIMPy | Hi [sr] |
| 21:19.37 | [sr] | need a small help, on which ports should I open so that the comunication works OK, the RTP ports |
| 21:19.39 | [sr] | hi WIMPy! |
| 21:21.47 | [sr] | WIMPy: don't know whats wrong with my test machine, but GIT LCR doesn't compile with asterisk trunk, only 10 ou 1.8 branch |
| 21:22.42 | WIMPy | It should work at least unil last mondays TRUNK. |
| 21:23.20 | Twitchnln | anyone in here ever played with chan_sccp? |
| 21:23.37 | WIMPy | The current version works for me, but I haven;t made a patch, yet. |
| 21:27.15 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 21:27.32 | [sr] | WIMPy: let me test again with last trunk |
| 21:28.34 | Qwell | [sr]: Whatever ports you've set in rtp.conf |
| 21:28.56 | [sr] | Qwell: great, good start, going to check that |
| 21:29.10 | [sr] | thats simples then... will give feedback in a few secs |
| 21:30.13 | WIMPy | As I said: Use SVN from at least 9 ddays ago for the moment. |
| 21:32.54 | p3nguin | twitchnln: I use chan_sccp. |
| 21:34.35 | Twitchnln | p3nguin: I've been able to get sccp extension registered on 7960 with 7914 sidecar, but I would like the 6 line keys on the 7960 to be line keys, it wants to make the first a line key then start the BLFs with second line key, instead of first button on 7914, have you experienced this? |
| 21:35.41 | *** join/#asterisk Rewris (~Rewris@201008081252.user.veloxzone.com.br) |
| 21:35.54 | p3nguin | I don't use a 7914 expansion. |
| 21:36.33 | Twitchnln | everything that i've read says that you define the line keys in the sccp.conf with the button= variable, but I have been unable to get this to work as yet |
| 21:37.42 | Twitchnln | it seems like the only way to get all 6 line keys working is to assign them individual extensions, am I correct about this? |
| 21:37.49 | p3nguin | Maybe I don't understand what you're doing. |
| 21:38.04 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 21:38.13 | p3nguin | Phones is completely extension agnostic. |
| 21:38.19 | p3nguin | s/is/are/ |
| 21:38.28 | p3nguin | Phones do not care about extensions. |
| 21:38.33 | p3nguin | Do not assume that they do. |
| 21:38.59 | p3nguin | What do you want on the second line button? |
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| 21:40.49 | Twitchnln | I would like the second line button to show the same extension as the first, in SIP<mac>.cfg file this would be done with line1_name/line1_password, line2_name/line2_password, but with sccp firmware I cannot figure out how to do this |
| 21:41.43 | p3nguin | Using chan_sccp, all the configuration is done in sccp.conf. |
| 21:42.01 | Twitchnln | correct, but how do i assign 1 line to multiple keys? |
| 21:42.33 | Twitchnln | so that when second call comes in it shows on line 2, third call shows on line 3, etc |
| 21:43.28 | p3nguin | That's not how VoIP works. |
| 21:44.12 | p3nguin | You only need to have one line key active to receive and make multiple calls. |
| 21:45.02 | Twitchnln | currently when second call comes in, it shows on line 1 key, so i put line 1 on hold, to answer line 2, now a 3rd call comes in, i put the second call on hold, answer the 3rd put them on hold and have 3 lines on hold on same button, how do I pickup the first line? |
| 21:45.13 | p3nguin | The dial plan *could* be configured to send subsequent calls to other lines, but I don't know why anyone would want to do that. |
| 21:46.17 | p3nguin | Put any active call on hold, use the arrow up/down buttons on the phone to move up and down the screen to highlight the call you want to resume, then press the resume softkey. |
| 21:46.55 | p3nguin | And you don't have "3 lines on hold." You have three CALLS on hold. |
| 21:48.07 | Twitchnln | correct, and i can work with that, i guess i just need to figure out how to tell the BLFs not to start on the line keys, but to skip them |
| 21:48.12 | [sr] | simples (and maybe stupid question), is there a switch in asterisk so that i can see the options that were passed on the build? |
| 21:48.17 | [sr] | simples=simple |
| 21:48.37 | p3nguin | You have the second line key configured as a speed dial and BLF for another phone? |
| 21:48.55 | p3nguin | What do you want to show up on line keys 2-6? |
| 21:49.30 | Twitchnln | nothing would work fine if I don't need them to show as line appearances as you said |
| 21:49.46 | p3nguin | You can configure 2-6 to be empty. |
| 21:49.55 | p3nguin | Is that what you want? |
| 21:50.07 | Twitchnln | sure, that would work great |
| 21:50.35 | p3nguin | In your sccp.conf, find the section where your phone is configured. [SEP<MAC>] |
| 21:50.48 | Twitchnln | got it |
| 21:51.05 | p3nguin | You probably have description, button ... |
| 21:51.37 | p3nguin | List one button as a line and then use five as empty: button = empty |
| 21:52.59 | p3nguin | http://pastebin.com/7UG7eLcG |
| 21:53.30 | Twitchnln | i've tried placing button=empty, but it just populates the first BLF there |
| 21:53.52 | Twitchnln | been beating my head against a wall on this one all day |
| 21:54.07 | p3nguin | That's pretty odd. Empty line keys work fine for me without a 7914 attached. |
| 21:54.08 | Twitchnln | and for once google has not been my friend |
| 21:55.04 | p3nguin | I copied that right out of my sccp.conf, so the syntax has to be correct. |
| 21:55.34 | Twitchnln | strange |
| 21:55.37 | p3nguin | Which version are you using? |
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| 21:56.18 | Twitchnln | SCCPv2 |
| 21:56.23 | p3nguin | I'm using Release: 4.0.0 DEV r3206, but empty line keys have worked since a long time ago. |
| 21:56.23 | Twitchnln | asterisk 1.4.21 |
| 21:56.31 | p3nguin | Upgrade to 3. |
| 21:56.34 | p3nguin | It is much better. |
| 21:56.47 | Twitchnln | i will give that a shot, thanks for your time |
| 21:57.01 | p3nguin | I used to use 2 on 1.4, then moved to 3 on 1.4 and it was much better. |
| 21:57.31 | p3nguin | As soon as they added working support for asterisk 1.8, then I moved to 1.8 with version 4 dev. |
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| 21:59.28 | Twitchnln | I will give v3 a shot, hopefully it will work better for me |
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| 22:06.13 | [sr] | RTP ports are UDP only am i right? |
| 22:06.28 | p3nguin | yes |
| 22:07.34 | [sr] | tks |
| 22:08.29 | p3nguin | atk! |
| 22:10.15 | [sr] | atk = ? |
| 22:10.16 | [sr] | :p |
| 22:10.32 | p3nguin | tnoi! |
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| 22:14.04 | [sr] | not getting it.. :p |
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| 22:15.11 | p3nguin | Random letters are like that. |
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| 22:15.35 | p3nguin | That's why we use special combinations of letters called "words." |
| 22:16.05 | [sr] | i see, 'cause of my "tks" |
| 22:16.18 | [sr] | but i dont know the meaning of "atk" |
| 22:16.50 | p3nguin | atk is like tks... not a word and not an acronym -- it's random. |
| 22:18.24 | p3nguin | In asterisk 1.4, is there any way to show the number of calls processed since startup? |
| 22:18.37 | [sr] | p3nguin: i get you know |
| 22:18.38 | p3nguin | In 1.8, core show channels or calls will report that number. |
| 22:22.35 | [sr] | what's the minimum range for RTP ports? 100 is enought? |
| 22:22.59 | p3nguin | Since you shouldn't be running asterisk as root, 1024 is the least number that the asterisk process can use. |
| 22:23.28 | p3nguin | Err... you want to know the smallest number of ports within the range? |
| 22:23.40 | [sr] | not the smallest port |
| 22:23.42 | p3nguin | Figure on four ports per call for calls going through a NAT. |
| 22:23.43 | [sr] | i mean the reange |
| 22:24.00 | p3nguin | I just misinterpreted your question. |
| 22:24.01 | [sr] | like, 10000:10010 |
| 22:24.09 | p3nguin | Good for 2 calls! |
| 22:24.19 | [sr] | ok so each call used 4 ports |
| 22:24.30 | p3nguin | It can. |
| 22:24.37 | [sr] | that the max per call? |
| 22:24.40 | [sr] | sorry the question |
| 22:25.09 | p3nguin | I think that will be the max per call when there are only two end points in the call. |
| 22:25.17 | p3nguin | Figure on 2 ports per call leg. |
| 22:26.22 | p3nguin | For a call where one person is on hold and one person is talking to another person, that would be 6 ports. |
| 22:26.40 | p3nguin | How many calls do you want to support? |
| 22:29.54 | [sr] | i got it, its just for my knowlage |
| 22:30.08 | [sr] | not much for now, about 10 the max |
| 22:31.10 | p3nguin | I'd probably allocate 100 ports or so. |
| 22:31.59 | [sr] | that |
| 22:32.40 | [sr] | allow me to ask other thing, i have an extra problem on the linux machine (as router), nf_nat_sip is loaded, i have sip_direct_signalling=0 sip_direct_media=0, but how should this options be? |
| 22:33.47 | WIMPy | Do not load nf_nat_sip. |
| 22:33.55 | p3nguin | I have it loaded. |
| 22:33.55 | [sr] | hum i see the nf_conntrack_sip params, i think i can specify ports there directly |
| 22:34.00 | [sr] | WIMPy: really? hum |
| 22:34.01 | p3nguin | nf_nat_sip 4432 0 |
| 22:34.01 | p3nguin | nf_conntrack_sip 13916 1 nf_nat_sip |
| 22:34.02 | WIMPy | nf_conntrack_sip can be helpful. |
| 22:34.20 | p3nguin | Looks like nf_conntrack_sip uses nf_nat_sip? |
| 22:34.35 | WIMPy | nf_nat_sip is there to mangle your sip packets. That is most likely not a good thing. |
| 22:34.48 | p3nguin | I don't have any problems with calls. |
| 22:34.52 | WIMPy | The other way round. |
| 22:36.07 | *** join/#asterisk fofware (~fabian@host25.190-30-47.telecom.net.ar) |
| 22:36.10 | [sr] | WIMPy: it does the trick! |
| 22:36.27 | p3nguin | What trick would that be? |
| 22:36.28 | WIMPy | You only need nf_conntrack_sip if you don't want to open rtp ports. |
| 22:36.36 | [sr] | p3nguin: removing the sip modules, |
| 22:36.56 | p3nguin | I really do not understand what you're trying to say. |
| 22:37.00 | [sr] | but i have to test some sip clients from my phone network that I use inside the lan |
| 22:37.19 | p3nguin | If it is on the lan, why would it be passing through a firewall? |
| 22:37.40 | [sr] | p3nguin: right now i'm outside of that network, |
| 22:38.17 | p3nguin | I still don't know what the problem is. I have phones outside of the network and I have those modules loaded in my firewall. |
| 22:38.38 | p3nguin | I didn't choose to load them; they are just loaded. |
| 22:39.16 | [sr] | ok but i think i have it solved |
| 22:39.30 | p3nguin | What was the problem you were trying to fix? |
| 22:39.53 | [sr] | p3nguin: question is, i have to see if the sip client of my phone network works inside of the lan, to connect the network SIP server (to make calls from the PC) |
| 22:40.14 | [sr] | p3nguin: right now was just the RTP ports to the outside, to connect outside of the lan |
| 22:40.31 | [sr] | you were helpful |
| 22:40.33 | [sr] | and WIMPy |
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| 22:43.20 | [sr] | thanks, i can handle it from now :) |
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| 23:24.26 | [sr] | gotta go |
| 23:24.29 | [sr] | see ya |
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