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| 01:25.41 | Katty | hello my asterisk does not work at all how to fix pls |
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| 02:00.54 | carrar | change careers |
| 02:01.20 | carrar | I recommend health plan consulting |
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| 02:37.33 | Lars_G_ | Hello. |
| 02:38.18 | Lars_G_ | If I share my case succinctly (I'm on my tablet) will you give me advice on gateways? |
| 02:39.50 | Lars_G_ | I want to migrate the phone system to voip this year. Using asterisk as the pbx. And I need to choose my gateways. |
| 02:40.03 | Lars_G_ | Here are my restrictions, |
| 02:40.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.183) |
| 02:42.10 | Lars_G_ | I have around 36 analog fxs coming in. they're on the main fxb, a small cabinet in the hall, it's un ventilated but I'll try to add some ventilation. Can't be anything advanced still, and space is small so I'm discarding the idea of a pc with cards |
| 02:43.20 | Lars_G_ | And budget is low and I can ONLY purchase locally, so if you could suggest two or three brands and/or models I'll try to find local distributors in order |
| 02:43.28 | Lars_G_ | Any suggestions? |
| 03:04.05 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 03:06.33 | leifmadsen | Lars_G_: perhaps the Cisco SPA112 |
| 03:06.44 | *** join/#asterisk ggayan (~ggayan@200.71.197.170) |
| 03:06.47 | leifmadsen | get a few of those, plug them into a switch, then connect that to asterisk |
| 03:07.01 | Lars_G_ | Is it asterisk compatible? |
| 03:07.33 | Lars_G_ | Are the fxo on those fixed or modular? |
| 03:08.25 | Lars_G_ | .... They're two port |
| 03:08.42 | Lars_G_ | So I'd need 18... |
| 03:09.45 | Lars_G_ | Hmmm |
| 03:12.09 | leifmadsen | Lars_G_: there is likely a higher port count |
| 03:12.15 | leifmadsen | alternatively, look at the xorcom stuff |
| 03:13.36 | Lars_G_ | I'll look for xorcom too. Thanks |
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| 03:14.22 | Lars_G_ | In general, Can micro pbx in a box like the spa9000 be used only as a gateway? |
| 03:16.43 | ChrisInSydney | Yes. Set is as a static SIP host. Forward calls direct to the Asterisk box IP |
| 03:17.28 | ChrisInSydney | Outbound calls SIP/Gateway-name/number |
| 03:17.46 | Lars_G_ | Nod |
| 03:18.10 | Lars_G_ | Well thanks. I need a ballpark budget tomorrow morning but I have enough |
| 03:19.59 | Lars_G_ | How i wish I could get an e1 |
| 03:21.23 | Lars_G_ | Thanks and good night |
| 03:21.40 | ChrisInSydney | night night Lars_G_ |
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| 05:09.15 | boratynskikamil | Good morgning |
| 05:09.58 | boratynskikamil | My question is, what does it mean if I type dahdi_hardware and I have '+' next to one device and '-' next to second one? |
| 05:19.45 | LiuYan | boratynskikamil: man dahdi_hardware |
| 05:35.52 | boratynskikamil | LiuYan: Ok, see it. |
| 05:37.00 | boratynskikamil | So my question is. I installed DAHDI from module-assistant and sources. Debian, by the way. Then I saw my HFC-S ISDN BRI Card worked properly. I hadn't got any problem with loading zaphfc module; then I installed OpenVox G400E driver from chan_extra. And now, I'm not able to load zaphfc. Suggestions? |
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| 05:40.37 | linocisco | HI all |
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| 05:42.21 | linocisco | should we have separate cabling for voice and data in the LAN. or should we combine this two? If we combine voice and data in the same Cat 6 LAN, to avoid jitter and low quality, what could we use economically? |
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| 05:47.01 | LiuYan | boratynskikamil: sorry, I don't have experience on mixing difference cards and/or mixing dahdi & zaptel drivers. |
| 05:48.05 | boratynskikamil | LiuYan: Is it possible it is impossible to mix them both? |
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| 05:50.21 | LiuYan | boratynskikamil: sorry, i don't know that. i'm currently using up-to-date dahdi-linux driver for an OpenVox Single E1 card |
| 05:50.53 | boratynskikamil | Understand, I will ask later, mayby someone will advice me, thank you for help. |
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| 06:33.12 | WIMPy | boratynskikamil: Why can't you load zaphfc any more? And shouldnt that be dahdi-hfcs anyway? |
| 06:34.06 | boratynskikamil | WIMPy: As far as I see, there is no module like zaphfc. But I'm sure it was. |
| 06:34.37 | boratynskikamil | WIMPy: I created snapshot on this point I started installing OpenVox G400E, yep |
| 06:34.39 | boratynskikamil | And now I restored that. |
| 06:34.46 | WIMPy | It has always been a 3rd party addon and still is. |
| 06:34.51 | boratynskikamil | My GFC-S ISDN BRI card works well. |
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| 06:36.29 | WIMPy | I haven't looked at it lately, but as far as I know the current version is dahdi-hfcs if you want to use the card via DAHDI. |
| 06:36.34 | boratynskikamil | WIMPy: dahdi-hfcs -what do you mean? |
| 06:36.50 | WIMPy | As replacement for zaphfc. |
| 06:36.54 | boratynskikamil | WIMPy: I would like to use both of thm at same time. |
| 06:37.09 | boratynskikamil | WIMPy: Is it a module? |
| 06:37.18 | WIMPy | Well, you need drivers for both, obviousely. |
| 06:37.23 | WIMPy | yes |
| 06:37.27 | boratynskikamil | WIMPy: Right. |
| 06:37.29 | boratynskikamil | Ok, wait. |
| 06:39.05 | WIMPy | And you have to make sure, the Linux drivers aren't loaded. |
| 06:39.19 | boratynskikamil | WIMPy: Yes, yes. |
| 06:39.22 | boratynskikamil | WIMPy: blacklist. |
| 06:39.46 | boratynskikamil | WIMPy: My dahdi-source package version is: 2.6.1+dfsg2-1 |
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| 06:40.18 | boratynskikamil | Hmmm... what is the newest one? As far as I remember, chan_extra wants to download the newest one and installs it... |
| 06:41.05 | WIMPy | 2.8.0.1 or 2.9.0-rc1 |
| 06:42.04 | WIMPy | Err, no. 2.9.0 (without rc1) |
| 06:45.49 | WIMPy | But if you need - what should I call it - 3rd party stuff for 3rd party stuff, it's quite likely it won't like the latest version(s). |
| 06:46.52 | WIMPy | Do we count and call it 4th party stuff or do we add and call it 6th party stuff, or do we multiply (as the issues might do) and call it 9th party stuff? |
| 06:47.33 | boratynskikamil | What is "party stuff"? :-) |
| 06:47.36 | boratynskikamil | My main question. |
| 06:48.04 | WIMPy | Stuff from other parties. |
| 06:48.44 | WIMPy | parties as in vendors |
| 06:49.18 | WIMPy | or supplier |
| 06:49.32 | snadge | ok.. i have a call thats being redirected to another server.. and for some reason its chopping the 0 off the front |
| 06:50.04 | WIMPy | snadge: You must have told it to do so. |
| 06:50.13 | snadge | extensions.conf im looking at.. yeah im trying to find out exactly where |
| 06:50.53 | snadge | exten => 0388035251,1,goto(soul-incoming-old,${EXTEN}) |
| 06:50.59 | snadge | that works.. then when I look at soul-incoming-old |
| 06:51.19 | snadge | exten => _03XXXXXXXX,n,Dial(SIP/gw2/${EXTEN}) |
| 06:51.27 | snadge | its meant to do that.. (after a bunch of other things).. but it isnt |
| 06:51.43 | snadge | what does the _ mean? |
| 06:52.14 | WIMPy | It means that it's a pattern match. |
| 06:52.42 | WIMPy | Look for dialplan basics in the |
| 06:52.43 | WIMPy | ~book |
| 06:52.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 06:53.03 | snadge | <PROTECTED> |
| 06:53.36 | snadge | thats why its failing anyway.. because the 0 is being dropped somewhere.. :| |
| 06:53.59 | WIMPy | And all numbers that match that pattern are replaced by that single one? |
| 06:54.35 | boratynskikamil | WIMPy: So, mainly, what should I do? Download some drivers for my DAHDI-sources version, compile it and check if it works? |
| 06:55.02 | boratynskikamil | WIMPy: http://wiki.openvox.cn/index.php/OpenVox_G400E_Installation_Manual - i tried to do it according to this manual. |
| 06:55.47 | WIMPy | boratynskikamil: Well, I don't know how that chan_extra thing works. Havent had the `pleasure'. But if it's automatic, use that and then add dahdi_hfcs afterwards. |
| 06:56.33 | boratynskikamil | WIMPy: Should I be sincere? "pleasure" is a great word... Stupid tool, that's it. |
| 06:57.05 | WIMPy | If it isn't cooperative, you can still use the Linux driver :-) |
| 06:58.01 | WIMPy | prefers to buy hardware that's supported by Linux. Far less trouble, but not allways possible, unfortunatly. |
| 06:59.21 | WIMPy | That manual sure talks of rater old versions. I hope it's just the manual that's out of date. |
| 07:02.22 | *** join/#asterisk liquidamber (1817a92b@gateway/web/freenode/ip.24.23.169.43) |
| 07:03.25 | liquidamber | i have a very small java program i want to execute inside a dialplan, the program just opens a TCP connection to another device and sends a hex string. it works with sudo -u asterisk but not from inside the dialplan |
| 07:04.07 | liquidamber | similarly i have an extension that executes /usr/bin/wol and that works fine... |
| 07:05.52 | WIMPy | Check your paths. |
| 07:06.23 | WIMPy | Do you wait for any response from your program? |
| 07:06.24 | liquidamber | i put the entire path. i'm using some switches, do i need to quote or backtick anything? |
| 07:06.29 | liquidamber | no |
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| 07:06.53 | liquidamber | for reference its an iBoot dataprobe. it just takes a packet to turn a switch on or off, and my java application works fine to do that on the CLI |
| 07:07.38 | liquidamber | exten => 802,1,System(/usr/bin/java -classpath /usr/bin/iboot on) |
| 07:07.38 | WIMPy | Chek teh syntax. You probably have to seperate parameters as another parameter in the dialplan, i.e. with a comma. |
| 07:08.22 | WIMPy | No, system is fin with just one string. |
| 07:08.32 | WIMPy | fine |
| 07:08.42 | liquidamber | hmm ok. so i also have one that tries to run /usr/bin/eject for testing, that doesn't run either |
| 07:09.01 | liquidamber | but the wake on lan one works fine, which is kindof odd. |
| 07:09.18 | WIMPy | I assue that asterisk doesn't have permissions to access your drive. |
| 07:09.30 | WIMPy | +m |
| 07:09.32 | liquidamber | sudo -c asterisk /usr/bin/eject works though |
| 07:09.47 | liquidamber | er |
| 07:09.58 | liquidamber | -u |
| 07:10.30 | WIMPy | You might have additional groups when logged in via a terminal. |
| 07:13.42 | liquidamber | any possibility it's selinux related? |
| 07:14.11 | WIMPy | Sure |
| 07:17.56 | liquidamber | Durrr yeah it was selinux |
| 07:18.03 | liquidamber | now my sweet toy works |
| 07:19.48 | liquidamber | thank you |
| 07:21.51 | WIMPy | boratynskikamil: Did you get both drivers working? What versions did you end up with? |
| 07:24.21 | linocisco | should we have separate cabling for voice and data in the LAN. or should we combine this two? If we combine voice and data in the same Cat 6 LAN, to avoid jitter and low quality, what could we use economically? |
| 07:25.10 | snadge | ok another dumb question.. why does this work: exten => 0388035251,1,goto(soul-incoming-old,${EXTEN},1) .. but this doesnt |
| 07:25.18 | WIMPy | Either use seperate cabling or good switches configured to prioritize RTP streams. |
| 07:25.24 | snadge | exten => 038803525[1-9],1,goto(soul-incoming-old,${EXTEN},1) |
| 07:25.44 | boratynskikamil | WIMPy: Nope. |
| 07:25.51 | WIMPy | I assume that a 2nd switch is usually cheaper than a good switch and someone who knows how to configure it propperly. |
| 07:25.51 | snadge | in other words, i can match the single number.. but if I put a range in like that, it wont match |
| 07:26.13 | WIMPy | snadge: Because th _ is missing. |
| 07:26.26 | snadge | thats what I thought actually.. i should have learned that from before :) .. thanks |
| 07:27.17 | WIMPy | And you can use Z instead of [1-9]. |
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| 07:31.26 | liquidamber | snadge: auto qos voip cisco-phone :P |
| 07:32.12 | liquidamber | but i think vlans and daisy chaining is best for a retrofit where you have to stuff voip somewhere that didn't have 2 drops per cube / workstation |
| 07:32.47 | WIMPy | You definitely need qos or whatever. |
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| 07:46.11 | liquidamber | any thoughts on how to do click to dial in windows -> asterisk? |
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| 08:14.13 | wizbit | what do i option do i need to configure to make my phone number show on the phone its ringing? at the moment, if i ring a phone it shows as private |
| 08:14.42 | wizbit | do i need to edit outgoing extensions or can this be done inside sip.conf? |
| 08:15.49 | WIMPy | Either you explicitely configured that in your dialplan, or it is set by your phone. |
| 08:16.13 | WIMPy | Or you caller ID is set to "private". |
| 08:16.24 | wizbit | WIMPy: can i do this inside sip.conf? |
| 08:16.49 | WIMPy | AFAIK, no. |
| 08:17.11 | wizbit | so i need something like this inside extensions.conf |
| 08:17.11 | wizbit | exten => _X.,1,Set(CALLERID(num)=) |
| 08:17.44 | WIMPy | (num-pres) |
| 08:18.11 | WIMPy | Unless you don;t have any caller ID at all, but AFAIk Asterisk then replaces it with "Asterisk". |
| 08:19.06 | WIMPy | The caller ID itself can be set in sip.conf, which is probably a good idea for phones unless you want the users to be able to change their caller ID. |
| 08:19.37 | WIMPy | That also requires trustrpid=no. |
| 08:22.08 | wizbit | WIMPy: i will edit the [outgoing] context in my iax.conf |
| 08:23.29 | wizbit | WIMPy: is this all what is required: |
| 08:23.30 | wizbit | trustrpid=no |
| 08:23.30 | wizbit | callerid=0560XXXXXXXX |
| 08:23.53 | WIMPy | For your sip phone, yes. |
| 08:23.57 | wizbit | acve |
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| 08:25.31 | wizbit | WIMPy: when i ring my cellphone, it still shows as private number |
| 08:26.05 | WIMPy | So how do you call your cell phone? |
| 08:26.22 | wizbit | my asterisk is setup to use a voip provider |
| 08:26.34 | wizbit | and i have a polycom phone connected to it |
| 08:26.59 | WIMPy | Then chek the configuration of your account at that provider. Also check what format they expect caller IDs to have. |
| 08:27.07 | wizbit | ok |
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| 08:28.47 | WIMPy | Then chek the configuration of your account at that provider. <== that includes the question if you're allowed to send caler ID at all. |
| 08:31.21 | skrusty | morning |
| 08:31.34 | WIMPy | Good morning skrusty |
| 08:35.06 | skrusty | how are well all today? :) |
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| 08:40.17 | WIMPy | should be sleeping but is updating his asterisk pages instead. |
| 08:40.39 | wizbit | WIMPy: the voip provider does support caller id, and mine is turned on |
| 08:42.31 | wizbit | maybe there is a polycom setting to turn it on |
| 08:43.07 | WIMPy | Is the format correct? |
| 08:43.27 | wizbit | i will paste my configs |
| 08:43.41 | WIMPy | That doesn't help. |
| 08:43.53 | WIMPy | Unless we know what your provider wants. |
| 08:44.51 | WIMPy | Try to send your full unformatted number, i.e. including country and area codes, but no prefixes. |
| 08:45.29 | WIMPy | That's what most seem to expect. |
| 08:45.41 | WIMPy | So either experiment or ask them. |
| 08:46.26 | wizbit | WIMPy: http://dpaste.com/1586580/plain/ |
| 08:46.38 | wizbit | ok |
| 08:47.53 | wizbit | http://dpaste.com/1586580/plain/ |
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| 08:57.33 | wizbit | hmmm this was not set correctly |
| 08:57.34 | wizbit | reg.1.displayName |
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| 09:06.21 | AsteriskUser | Is there a way to show the heap use in asterisk? |
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| 09:27.09 | bpietro | hi all, somebody with portech mv-370 setting experiences ? If so, can he/she pastebin example of sip.conf and mv-370 settings? tnx in advance |
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| 10:15.59 | vedic | While using FastAGI over where the application sits on a remote server, what are the ways to secure the communication between Asterisk server and the AGI application? |
| 10:16.22 | vedic | As I read, AGI doesn't provide SSL. It can only use TCP without SSL |
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| 10:23.21 | skrusty | vedic: you could try some sort of fastagi proxy |
| 10:24.28 | skrusty | https://github.com/pdeschen/nodast - not sure if it supports ssl though, it says it's on the todo... but looks like it may not have been developed for some time |
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| 10:42.30 | LiuYan | is any an application model guide about how to use AMI api to implement a web client for small/medium call center? i mean do i need to to create a server application which sit between web client and asterisk server, or it's better to let web client communicate with asterisk directly? |
| 10:45.01 | WIMPy | You most probably need an application. |
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| 10:51.56 | *** join/#asterisk wizbit (~wizbit@unaffiliated/wizbit) |
| 10:52.03 | wizbit | grrr still cant get caller id working |
| 10:52.26 | wizbit | ive tried adding callerid to extensions and putting in callerid= into sip.conf and iax.conf |
| 10:52.38 | wizbit | my voip provider callerid is turned on and i had this working once |
| 10:53.29 | LiuYan | WIMPy: that's what i prefered, but the cons is complicated. is a guide about it? |
| 10:54.12 | WIMPy | LiuYan: Well, it's about what YOU want it to do. |
| 10:56.03 | LiuYan | WIMPy: currently i just want to basic function: (1) received incoming call event and popup a web page (2) originate a call from web pages |
| 10:56.53 | WIMPy | Do it :-) |
| 10:57.15 | WIMPy | You can do a popup by calling some command from the dialplan. |
| 10:57.39 | WIMPy | And to originate you don't need a persistant connection (daemon), either. |
| 10:58.15 | WIMPy | It's much more efficient to listen on AMI to do the popups, off course. |
| 10:58.27 | LiuYan | WIMPy: I've wrote a simple test page according astman.js provides by asterisk, but this model is the second one: web client communicate to asterisk directly |
| 10:59.54 | WIMPy | For simple functions, it can be good enough. |
| 11:01.23 | LiuYan | WIMPy: is there a performance issue in such model? for example, a small call center with less 100 agents/web-clients |
| 11:03.43 | skrusty | LiuYan: what os? |
| 11:04.10 | skrusty | oh sorry, web application |
| 11:04.20 | WIMPy | It is less effective, but if it's an issue is another question. |
| 11:04.22 | LiuYan | skrusty: server OS will be CentOS 6, |
| 11:04.33 | skrusty | there's a few desktop clients that use AMI to output call centre style info, ike wallboard etc |
| 11:06.44 | *** join/#asterisk ggayan (~ggayan@200.71.197.170) |
| 11:07.19 | LiuYan | skrusty: desktop client or ActiveX in IE is also a solution, such solution with a CTIServer may not need to poll events from asterisk |
| 11:09.42 | vedic | While using FastAGI where the application sits on a remote server, what are the ways to secure the communication between Asterisk server and the AGI application? Has anybody attempted that? |
| 11:10.26 | WIMPy | Any VPN, stunnel, server configuration, all the usual stuff. |
| 11:10.50 | skrusty | vedic: i've already said, not native to fastagi |
| 11:10.59 | skrusty | but you could use a proxy, or as WIMPy says, a tunnel |
| 11:11.29 | skrusty | LiuYan: if you have any dev skills, i suggest you check out the AMI frameworks out there (asternet, asterisk-java etc) |
| 11:11.45 | vedic | skrusty: Proxy like what? Do you know any such write up or out of the box proxy which will support this? |
| 11:11.56 | LiuYan | WIMPy: ok thanks. i'll try the second model first for a small project, if performance is an issue, then consider the first model. |
| 11:12.41 | skrusty | vedic: i posted a link further up to nodast, might be worth looking at, otherwise try using an ssh tunnel? |
| 11:15.32 | LiuYan | skrusty: yeah, i'm considering use asterisk-java instead of writing raw code to communicate with asterisk, |
| 11:15.43 | bulkorok | how can I listen to g722 streams in wireshark!? |
| 11:16.40 | skrusty | LiuYan: i recommend asternet if you know c#, only because i wrote it ;) hehe |
| 11:17.48 | skrusty | bulkorok: not sure you can |
| 11:20.36 | *** join/#asterisk Cynagen (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
| 11:22.49 | *** join/#asterisk wonderworld (~ww@62.143.158.113) |
| 11:23.13 | LiuYan | skrusty: cool! i need to learn c# first ~ |
| 11:23.46 | *** join/#asterisk Cynagen (~cynagen@70.190.135.8) |
| 11:26.52 | skrusty | hehe :D |
| 11:27.06 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 11:27.11 | skrusty | there is a sample desktop client written in asternet c#, but it doesn't do any queue related stuff |
| 11:28.30 | skrusty | vedic: http://www.revsys.com/writings/quicktips/ssh-tunnel.html |
| 11:31.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:39.23 | LiuYan | skrusty: just take a quick look your sample applications, it seems they're all using FastAGI. do you have samples using AMI? |
| 11:39.34 | skrusty | one moment |
| 11:39.43 | skrusty | https://asttray.codeplex.com/ |
| 11:40.14 | skrusty | it's basic, and just meant as a starting point for other peoples projects |
| 11:42.38 | LiuYan | skrusty: basic is what all i needed ~ |
| 11:43.41 | skrusty | well, that'll get you started! |
| 11:43.56 | skrusty | it's quite easy to build up some simple call centre functionality |
| 11:44.00 | skrusty | like wallboard etc |
| 11:44.14 | skrusty | just watch for the queue events |
| 11:53.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
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| 12:22.16 | mirela666 | does anyone know where I can find the asterisk ring sound file? |
| 12:22.49 | *** join/#asterisk roramirez (~rodrigo@pc-34-179-160-190.cm.vtr.net) |
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| 12:31.05 | WIMPy | What? |
| 12:31.58 | mirela666 | I need to add ringing sound at the begging of the MOH file |
| 12:32.23 | mirela666 | I don't want to Queue(queue,r) and then Queue(queue) |
| 12:32.50 | mirela666 | so to awoid that I just need 10 sec of ringing effect on moh to play to callers |
| 12:33.03 | mirela666 | but can't find any normal ringing sound on web |
| 12:33.05 | WIMPy | That's not a file. It's generated. |
| 12:33.14 | mirela666 | need to improve my googling skills :D |
| 12:34.12 | mirela666 | hmmm |
| 12:41.57 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 12:49.40 | mirela666 | does anyone knw where to find some good ringing sounds? |
| 12:54.29 | WIMPy | Start Audacity. Select Generate -> Tone. |
| 12:57.37 | *** join/#asterisk Cynagen (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
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| 13:03.58 | mirela666 | WIMPy: thx got it :) |
| 13:17.00 | *** join/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 13:17.07 | StaRetji | howdy folks |
| 13:17.51 | StaRetji | I have a problem and I am sure some of you will have at least some advices |
| 13:18.03 | StaRetji | I am sending ivr over early media |
| 13:18.18 | StaRetji | and ivr say to call here press 1, to call here press 2 |
| 13:18.41 | StaRetji | no, session is not answered cause I don't want to charge clients for early media |
| 13:19.30 | StaRetji | problem is, I have many clients complaining that their switches can't send dtmf over non answered session |
| 13:19.40 | StaRetji | so, I am kind of stuck here |
| 13:20.10 | StaRetji | can't answer session before they enter 1 or 2 and calls really gets answered |
| 13:20.28 | StaRetji | but half of them can't pass ivr as dtmf is not being sent to me |
| 13:20.42 | StaRetji | any idea, hint, advice is highly appreciated |
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| 13:46.30 | *** join/#asterisk Lars_G (~Lars@unaffiliated/lars-g/x-000001) |
| 13:46.32 | Lars_G | Hello. |
| 13:46.41 | Lars_G | I came yesterday. :) not that it matters at all. |
| 13:46.56 | Lars_G | Two questions. One from yesterday. Are xorcom builds sturdy in your experience? well built? |
| 13:47.30 | Lars_G | And one unrelated question, is there something like t.38 that would let me somehow hook and use a pots modem based credit card terminal onto the voip loop and have it work? |
| 13:47.35 | tzafrir | Lars_G, what specific xorcom builds? |
| 13:47.46 | Lars_G | xr2000 |
| 13:48.02 | Lars_G | Will it whitstand some abuse? |
| 13:48.31 | tzafrir | (disclaimer: I work at Xorcom) |
| 13:48.36 | tzafrir | The case is metal |
| 13:49.05 | Lars_G | Ventilation? |
| 13:49.54 | tzafrir | It's reasonably sturdy. Not designed for heavy-duty industrial-use |
| 13:50.10 | tzafrir | It does require ventilation, as you would expect |
| 13:50.28 | tzafrir | (as anything that fits in a rack) |
| 13:50.49 | *** join/#asterisk simonmox (~simonmox@5.133.168.42) |
| 13:50.57 | Lars_G | hmmm |
| 13:51.11 | Lars_G | Can you see any trouble from it not being horizontal, but vertical? |
| 13:51.40 | tzafrir | I don't think so. Asking... |
| 13:51.53 | *** join/#asterisk CeBe (~CeBe@port-92-206-93-127.dynamic.qsc.de) |
| 13:51.54 | Lars_G | And, last question, would it be doable, with the asterisk in it, and the ui, to trunk to a remote (over internet/vpn) pure asterisk setup to connect two sites? |
| 13:52.19 | simonmox | Hey, is there a way to access global variables via AGI? |
| 13:53.37 | Lars_G | Oh yes I have an extra question, and sorry that I'm using you as a sort of sales rep.... I have a nice ldap directory where I store lots of stuff, using fusiondirectory which is a plugin based ui to handle ldap directories, and it has an asterisk voip extension that would let me provision some asterisk stuff, specifically provision extensions/handsets... will the xorcom system be able to hook into an external ldap tree? or is it limited in its |
| 13:55.29 | tzafrir | (I'm not a sales rep) |
| 13:56.13 | tzafrir | Regarding vertically: we haven't tested it. I can't see any reason why it won't work except, maybe, the hard disk |
| 13:56.25 | Lars_G | hmmm shouldn't hurt the hdd much, in theory |
| 13:56.45 | Lars_G | And sorry my intention is just to ask a couple technical questions |
| 13:56.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:57.47 | tzafrir | The dialplan is basically freepbx. |
| 13:58.18 | Lars_G | I'll investigate freepbx |
| 13:58.41 | tzafrir | I'm not sure FreePBX can take data directly from LDAP. Maybe this can be done using an external convertion script. Not really sure |
| 13:59.39 | Lars_G | Hmm I'll investigate more. |
| 14:00.29 | simonmox | It's real easy to set a global variable via AGI using 'set GLOBAL(MYVAR)=somevalue' |
| 14:00.35 | Lars_G | tzafrir: Can I ask one more? and pardon me. |
| 14:00.45 | simonmox | but there seems no obvious way to retrieve it? |
| 14:01.09 | simonmox | Anybody know of one? Previously there was get_variable and set_variable, but these seem to be deprecated now |
| 14:01.53 | tzafrir | Lars_G, well, one more shouldn't hurt |
| 14:03.32 | Lars_G | tzafrir: When they (xorcom) mentions "just hook your fax, we use pll" (and I guess this applies to credit card terminals too), does that mean hook directly to an FXS module on the xorcom? is there a protocol, like t.38 that would let me use a compatible gateway for the fax? |
| 14:05.27 | tzafrir | directly to an FXS port (an FXS port serves the same roll as the analog port you get from a telephony company) |
| 14:06.24 | tzafrir | s/roll/role/ |
| 14:06.27 | mjordan | simonmox: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AGICommand_get+variable |
| 14:08.23 | [TK]D-Fender | FreePBX doesn't touch LDAP. |
| 14:08.33 | [TK]D-Fender | It's its own set of MySQL tables |
| 14:09.47 | Lars_G | [TK]D-Fender: ok. thanks. |
| 14:09.52 | Lars_G | tzafrir: sigh. thanks. |
| 14:11.04 | Lars_G | There exists in general then, no such protocol that would let to hook an external modem to the voip loop.... for faxes, I could use t.38 (specially if the system will let me hook with an external fax server, or it'll act as a sender appart from receiver), but I'm screwed with the credit card terminals. |
| 14:11.43 | Lars_G | tzafrir: And I don't know PLL well, but I'd bet it works only with an on-xr2000 fxs and not with an fxs module in astribanks |
| 14:12.02 | simonmox | mjordan: THanks, that worked a treat! |
| 14:12.06 | Lars_G | Since I'm betting the pll means the xr tristates a direct connection between fxs and fxo inside the device for the transmission to work |
| 14:12.51 | tzafrir | Lars_G, the XRs use Astribank modules that are the same as those that are inside Astribanks. |
| 14:13.15 | tzafrir | So if something works with one and not with the other, it's probably a software bug |
| 14:13.43 | tzafrir | PLL refers to adjusting the different clocks involved |
| 14:14.09 | tzafrir | !wikipedia PLL |
| 14:14.17 | tzafrir | hmm |
| 14:14.20 | tzafrir | https://en.wikipedia.org/wiki/PLL |
| 14:17.19 | Lars_G | Yes I kinda know a little about pll from radio (I want to be a ham). |
| 14:18.03 | Lars_G | odd. still that means the internal modules use usb bus then. And then the adc and dac process of passing data between the modules inside the xr or between xr and astribank does nto affect the modem transmision. which... well it's doable really |
| 14:19.33 | Lars_G | Well it makes sense |
| 14:19.38 | Lars_G | I worked for an ISP for fucks sake |
| 14:20.06 | Lars_G | I know digital quantization when passing between digital and analog phone centrals in the city coudl cause a little degradation, but the modems would still work.... |
| 14:20.26 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 14:20.27 | Lars_G | so what hinders modem/fax on voip is not the digital quantization, but the compression protocol. |
| 14:20.31 | Lars_G | Makes sense |
| 14:20.36 | Lars_G | hmmmm |
| 14:21.21 | [TK]D-Fender | And losing 20ms in a single packet.. or worse... or jitter/delay |
| 14:21.35 | Lars_G | yeah |
| 14:21.37 | Lars_G | signaling |
| 14:21.39 | Lars_G | you're right |
| 14:21.48 | [TK]D-Fender | the serious problem is packetization |
| 14:22.33 | Lars_G | Because the stream breaks into blocks |
| 14:22.54 | Lars_G | This is all solvable... but not easily. |
| 14:23.41 | Lars_G | for packetization alone, syncing packet rate to encodign rate would be the only way, and it'd be hell. I think a different transport alternative to udp would need to be developed. |
| 14:24.51 | Lars_G | then you'd still have delay, and jitter to play with, which would require tuning the whole end to end, program to network stack to os to hardware |
| 14:24.54 | Lars_G | yeah |
| 14:25.17 | Lars_G | I think I'll keep a couple couple of pots cabling lying around for the credit card terminals |
| 14:25.36 | Lars_G | it's the cablign that worries me, more than fxs module |
| 14:29.14 | [TK]D-Fender | [09:23]Lars_Gfor packetization alone, syncing packet rate to encodign rate would be the only way, and it'd be hell. I think a different transport alternative to udp would need to be developed. <- you mean ... like T.38? ;) |
| 14:30.22 | Lars_G | People who use xorcom (to leave tzafrir alone), I know this is mostly asterisk and freepbx based... my question, how possible, and if possible easy is it to touch code/binaries on the device? (beyound maybe warranty troubles from it) ? |
| 14:30.53 | *** join/#asterisk chazzam (~chazz@donutokyo.info) |
| 14:30.54 | Lars_G | [TK]D-Fender: But there must be a reason why I've not seen any small gateway with t.38 that would let me hook a fax/modem to it |
| 14:31.39 | justdave | is there not a way to get the number that's being looked up in the dialplan function you can use in the weight field in a DUNDI mapping? |
| 14:31.50 | justdave | ${NUMBER} seems to be empty |
| 14:32.21 | justdave | (I was guessing it would be that since that's what you use in the response part to get it) |
| 14:32.39 | justdave | the docs for it don't seem to say one way or the other |
| 14:32.55 | [TK]D-Fender | Lars_G: "modem" no, the protocol isn't for generic modem use. As far as T.38 .... there are small gateways with it |
| 14:34.06 | Lars_G | [TK]D-Fender: but a fax is a modem. |
| 14:34.57 | [TK]D-Fender | Lars_G: just adding for your comment about CC terminals |
| 14:35.26 | [TK]D-Fender | Lars_G: And there are lots of ATA's supporting T.38 |
| 14:35.56 | justdave | actually, the docs do say so in the current version, and specifically gives what I'm actually trying to do as an example. |
| 14:35.57 | Lars_G | ok |
| 14:36.12 | justdave | wonder if that was added in a newer version than what I have |
| 14:36.19 | justdave | (on 1.8 branch) |
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| 14:37.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:46.48 | Lars_G | tzafrir: it might be my proxy (haven't tested) but if not, tell your guys to check as I'm getting a few 503 errors from my proxy when trying to open your page. comes and goes. Other sites respond well |
| 14:46.58 | Lars_G | tzafrir: So, IT might want to check web server logs. |
| 14:47.16 | Lars_G | goddam |
| 14:47.22 | Lars_G | I have 42 FXS. sigh |
| 14:47.28 | Lars_G | that means no cheap solution for me :'( |
| 14:50.50 | tzafrir | Lars_G, what page? |
| 14:51.15 | tzafrir | hmm... same here.. |
| 14:51.20 | boom^time | I'm doing an AMI originate to local to a dial app so I can have a more detailed call status via the dialplan. But asterisk is only generated a CDR value for the "call" to the context/ext, which isn't really the call. The DIal app isn't generating one and the other leg of the originate command isn't either. Is there something simple I'm missing or should I be adding like a ForkCDR before the Dial app and in the other |
| 14:51.20 | boom^time | leg of the originate? |
| 14:51.20 | Lars_G | tzafrir: Anything ok your domain |
| 14:51.27 | Lars_G | s/ok/on/ |
| 14:51.53 | tzafrir | Not http://updates.xorcom.com/ :-) |
| 14:52.26 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-32-191.pn.at.cox.net) |
| 14:53.15 | Lars_G | :P |
| 14:53.33 | Lars_G | Probably different server. the one serving www.xorcom.com seems down or bouncy though |
| 14:54.03 | igcewieling | Does anyone know of a way to access dialplan functions like SYSINFO via Manager without any calls (and hence no channel name which is required by Manager GetVar |
| 14:54.21 | tzafrir | anyway, thanks for the report. Working with the hosting provider |
| 14:55.49 | boom^time | igcewieling, could you originate a call to a context/ext that submitted a userevent for that? |
| 14:56.16 | igcewieling | boom^time: that is exactly what I was trying to avoid |
| 14:56.36 | boom^time | I understand trust me. It's hacky. I've been trying to avoid a LOT of things I'm having to do to get asterisk to work with AMI |
| 14:57.15 | igcewieling | I'll use SNMP to get the info I needd. |
| 14:58.19 | boom^time | I finally conceded to using an originate to a local context to dial out just so I can get a more detailed response on the success of the call. |
| 14:59.05 | boom^time | I originally hacked asterisk to return different codes for the state reason but the guys in #asterisk-dev said it wasn't the proper way, and that I should be originating to a local ext. But now asterisk won't generate cdrs for me. |
| 14:59.26 | boom^time | one step forward two back |
| 15:00.47 | Lars_G | tzafrir: it seems the device offers fax to email, but not email to fax? |
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| 15:25.42 | tenspeed705 | just out of curiosity, anyone here have a metaswitch? |
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| 15:49.07 | ghost75 | getting randomly SIP 480 temporarily not available from itsp (busy/congested) when making new calls, but peer status shows always ok in * |
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| 15:58.46 | wizbit | how can i setup music on hold so it only works when i press the 'hold' button on my polycom? |
| 15:58.57 | ghost75 | can this be other problems than provider? |
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| 16:17.32 | ghost75 | somebody doing testcalls and monitoring the result? |
| 16:18.36 | Penguin | wizbit: Would moh be on if a call wasn't put on hold? |
| 16:19.03 | Katty | PENGUIN |
| 16:19.06 | Katty | YOU |
| 16:19.08 | Katty | are in trouble. |
| 16:19.21 | Katty | hugs Penguin |
| 16:19.29 | Penguin | That's my punishment? |
| 16:19.56 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-32-191.pn.at.cox.net) |
| 16:20.05 | Penguin | I'll have to get into trouble more often! |
| 16:21.15 | wizbit | Penguin: should moh be set to a extension, so for example, i call somebody |
| 16:21.33 | wizbit | then to put them on hold i could press 1 what is configured in extensions.conf to play moh music ? |
| 16:21.38 | Penguin | wizbit: What you're saying does not make any sense to me. |
| 16:21.55 | wizbit | Penguin: im not sure how it works |
| 16:22.03 | wizbit | should moh be always be playing? |
| 16:22.16 | Penguin | wizbit: If moh is loaded, when you put a call on hold, moh plays. |
| 16:22.16 | wizbit | unless you are speaking |
| 16:22.42 | wizbit | ok so i need to find out how to define that in my extensions.conf |
| 16:22.55 | Penguin | wizbit: Who said anything about extensions.conf? |
| 16:23.08 | wizbit | oh |
| 16:23.17 | wizbit | so it should happen anyhow? |
| 16:23.33 | Penguin | module show like music |
| 16:23.39 | Penguin | Run that on your asterisk CLI. |
| 16:24.01 | Penguin | Does it show the moh module? |
| 16:24.17 | wizbit | Penguin: i can get moh music to play by doing this: http://dpaste.com/1587067/plain/ |
| 16:24.19 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:24.25 | wizbit | and pressing 602, that plays the music fine |
| 16:24.45 | Penguin | Is that a yes or a no? |
| 16:25.00 | wizbit | ok ill test |
| 16:25.24 | wizbit | http://dpaste.com/1587068/plain/ |
| 16:25.36 | Penguin | I don't need a pastebin. I just need yes or no. |
| 16:25.39 | wizbit | yes |
| 16:25.49 | wizbit | res_musiconhold.so 1 modules loaded |
| 16:25.57 | Penguin | Good. Now make a call and put the person on hold. |
| 16:26.07 | Penguin | That's all there is to it. |
| 16:26.12 | wizbit | ok ill test |
| 16:27.52 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:27.59 | wizbit | Penguin: just did a test, rang my cellphone and pressed the hold button on my polycom, no music played, just blank |
| 16:28.00 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:28.21 | Penguin | wizbit: Did you answer the cell phone? |
| 16:28.44 | Penguin | wizbit: Did you put the speaker to your ear or enable speakerphone? |
| 16:29.04 | wizbit | speaker to ear |
| 16:29.13 | wizbit | yes answered it, heard my own voice |
| 16:29.16 | wizbit | then pressed hold |
| 16:29.32 | wizbit | do i need to specify musiconhold = mp3 in my iax.conf ? |
| 16:30.17 | Penguin | No. But you should configure the default moh class to use the correct files in musiconhold.conf. |
| 16:31.33 | wizbit | http://dpaste.com/1587076/plain/ |
| 16:31.43 | wizbit | ohh default |
| 16:33.01 | wizbit | Penguin: IT WORKS!!!! |
| 16:33.08 | wizbit | thanks :-) |
| 16:37.08 | Penguin | If you do not configure the default class to use your files, then you have to set the different music class in the dial plan. |
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| 17:50.43 | jeffspeff | anybody here familiar with sonicwall firewalls? |
| 17:51.34 | jeffspeff | i'm setting up a new remote office, the phones are going to be connecting through the vpn tunnel to the * server in another office. |
| 17:52.10 | jeffspeff | the phones are registering with sip, yet they show as "unreachable" and running sip qualify peer has no results |
| 17:52.49 | jeffspeff | i'm wondering what the SIP consistent nat setting needs to be on the sonicwall |
| 17:57.17 | tenspeed705 | exit |
| 18:00.10 | [TK]D-Fender | jeffspeff: OFF. Keep it out of the way |
| 18:09.20 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
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| 18:12.29 | jeffspeff | [TK]D-Fender, the consistant nat is turned off, in the iptables on my * box, i'm allowing udp from that /24 network. yet the sip packets are getting dropped because they're coming from the public ip of that office |
| 18:13.09 | jeffspeff | when i turn the firewall off it registers and the ADDR->IP shown in sip show peer shows the phones local ip |
| 18:13.14 | jeffspeff | not the public |
| 18:13.17 | [TK]D-Fender | You shouldnt' be NAT'ing |
| 18:13.45 | [TK]D-Fender | And you aren't showing any actual debug |
| 18:14.02 | jeffspeff | I know that. but what i just described implies that there's a nat setting on the remote office firewall right? |
| 18:14.43 | [TK]D-Fender | nothing implies anything until we see actual debug. |
| 18:14.49 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 18:14.58 | jeffspeff | debug of what? iptables dropping packets? |
| 18:15.18 | [TK]D-Fender | ASTERISK SIP DEBUG |
| 18:15.40 | [TK]D-Fender | iptables shouldn't be dropping packets... that's your server firewall which you should also have accounted for |
| 18:17.12 | jeffspeff | http://pastebin.com/9gDvDTms |
| 18:17.53 | jeffspeff | the iptables is dropping packets because the traffic is coming from the offices public ip and not the internal |
| 18:18.27 | *** join/#asterisk BartVE (d57dd8c2@gateway/web/freenode/ip.213.125.216.194) |
| 18:18.57 | jeffspeff | the office lan ip is 192.168.168.0/24 the public ip is 96.254.22.218 the extension is 2101 |
| 18:19.52 | BartVE | Hello, can somebody help me? I have a problem and I don't know if I can do something about it in my asterisk configuration or that my provider changed something |
| 18:19.59 | [TK]D-Fender | [Feb 5 12:15:59] Reliably Transmitting (NAT) to 192.168.168.52:5060: <---- |
| 18:20.12 | [TK]D-Fender | Contact: <sip:asterisk@12.49.160.154:5060> <------ you set that peer as NAT=yes. |
| 18:20.21 | [TK]D-Fender | they are NOT NAT'd if you are VPN routing the subnet |
| 18:20.32 | [TK]D-Fender | and you are passing them your PUBLIC IP as the return address because of it |
| 18:21.22 | jeffspeff | i have the nat settings for the peers set up as force_rport,comedia |
| 18:21.35 | [TK]D-Fender | NO <- |
| 18:21.42 | [TK]D-Fender | And make sure to specify your local subnets |
| 18:22.06 | [TK]D-Fender | Which you seem to have skipped |
| 18:22.31 | jeffspeff | that's what i forgot to do |
| 18:23.06 | [TK]D-Fender | Critical failure |
| 18:23.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:27.54 | BartVE | I will just tell my problem. I have the following setup: incoming call -> sip phones ringing -> (not answered) cell phone ringing -> (not answered) voicemail . This worked before but now the voicemail is never reached, asterisk thinks the call is instantly answered after dialing the cell phone. |
| 18:28.55 | [TK]D-Fender | Go look at that call. |
| 18:31.33 | BartVE | Logging says: "Called SIP/xxxx@provider" and then instantly "SIP/provider-000000b9 answered SIP/provider-000000b7" |
| 18:31.51 | BartVE | and then : Locally bridging etc |
| 18:32.32 | [TK]D-Fender | Then that's the end of it.... |
| 18:32.48 | [TK]D-Fender | The provider is ack-ing your call the moment it allows it |
| 18:35.19 | BartVE | Ok.. I will ask them why they changed something because this was working before... |
| 18:35.48 | BartVE | I can't think of a reason why they want to change it to behave like this |
| 18:36.40 | jeffspeff | [TK]D-Fender, yes, critical failure. i added that net to the localnets and it worked like magic |
| 18:36.49 | jeffspeff | thank you |
| 18:38.00 | ghost75 | anyone doing testcalls and monitoring? |
| 18:38.12 | [TK]D-Fender | jeffspeff: You're welcome |
| 18:38.31 | [TK]D-Fender | ghost75: Rephrase a little less vague please... |
| 18:38.33 | BartVE | [TK]D-Fender: Thanks for you help :) |
| 18:38.59 | ghost75 | call something to see if the system works and create alarm on problem |
| 18:44.16 | [TK]D-Fender | ghost75: still way too vague |
| 18:44.47 | ghost75 | calling test numbers |
| 18:44.50 | ghost75 | by script |
| 18:45.02 | ghost75 | monitoring? |
| 18:45.23 | [TK]D-Fender | So what is a "problem"? |
| 18:45.28 | [TK]D-Fender | Dial. |
| 18:45.30 | navaismo | sounds like icinga plugin for asterisk+devices |
| 18:45.32 | [TK]D-Fender | Check the result. |
| 18:45.34 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:45.39 | [TK]D-Fender | Do something if you don't like it |
| 18:45.43 | ghost75 | was asking if somebody doing it |
| 18:45.45 | navaismo | and see that in the graph section |
| 18:45.52 | [TK]D-Fender | I'm sure SOMEBODY is. |
| 18:46.01 | [TK]D-Fender | ~survey |
| 18:46.02 | infobot | List all your config. URL: ftp://metalab.unc.edu/pub/Linux/utils/scripts/survey-0.95.tar.gz |
| 18:46.05 | [TK]D-Fender | ~surveys |
| 18:46.10 | [TK]D-Fender | hrm... |
| 18:46.28 | [TK]D-Fender | ~polls |
| 18:46.28 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 18:46.32 | [TK]D-Fender | THATS the one... |
| 18:46.57 | *** join/#asterisk dillbilly (~dillbilly@69.61.135.139) |
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| 18:47.30 | ghost75 | make a call file and then just parse the * message log ? |
| 18:47.42 | file | is called |
| 18:49.18 | [TK]D-Fender | ghost75: DIALSTATUS <- |
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| 18:49.53 | ghost75 | hmm |
| 18:50.06 | navaismo | try icinga/nagios |
| 18:51.14 | mjordan | ghost75: I don't think anyone knows what you're trying to measure. Asterisk provides all sorts of ways to monitor ... stuff. But without knowing specifics of what you're looking to monitor, there's not much help anyone can give you. |
| 18:51.47 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 18:52.27 | [TK]D-Fender | mjordan: How do I do STUFF? And THINGS?!?!?! |
| 18:52.30 | ghost75 | just to test if i can call out, because my itsp gives me sometimes sip 480 temp not available messages |
| 18:52.40 | *** join/#asterisk wizbit (~wizbit@unaffiliated/wizbit) |
| 18:52.47 | [TK]D-Fender | ghost75: Go look at the dialstatus |
| 18:53.33 | navaismo | at the end you can have a graph like this but with "extensions"(devices) green for up red for down--->http://3.bp.blogspot.com/-uxhLWjlrh7M/T8Ayqn79hvI/AAAAAAAABX8/kW993HUVwK0/s1600/IcingaMap.tiff |
| 18:53.37 | navaismo | aaah nevermind |
| 18:53.45 | mjordan | [TK]D-Fender: with style? |
| 18:54.09 | ghost75 | homies style |
| 18:54.17 | [TK]D-Fender | navaismo: Which doesn't address what he's asking |
| 18:54.32 | navaismo | ah you miss my nevermind |
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| 19:21.35 | Kobaz | something i haven't gotten into just yet... what's a good way to debug packet loss? |
| 19:21.53 | Kobaz | or like where rtp frames might be getting dropped |
| 19:22.01 | Kobaz | whether it's inside asterisk or network related |
| 19:45.14 | boom^time | Does anyone have a problem with trying to change directories in asterisk 11.5 in asterisk.conf? |
| 19:45.31 | boom^time | I'm changing the astagidir and it ignores it and attempts to look in /var/lib/asterisk/agi-bin |
| 19:46.09 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-64.dllstx.fios.verizon.net) |
| 19:48.11 | [TK]D-Fender | boom^time: show us your config |
| 19:49.23 | boom^time | http://pastebin.com/QvsPSj7B |
| 19:49.37 | [TK]D-Fender | (!) = IGNORE this entire section |
| 19:49.43 | [TK]D-Fender | remove it |
| 19:49.56 | boom^time | Thank you |
| 19:52.19 | Kobaz | sooo |
| 19:52.22 | Kobaz | i did some testing |
| 19:52.29 | Kobaz | i know there's lots of variables here.... but |
| 19:53.21 | Kobaz | i have an asterisk 1.6.0.28 box in a colo, and on the same lan i have another box with asterisk 1.8.25 (I also tried some other 1.8's, earlier like 1.8,12 and etc) |
| 19:53.52 | Kobaz | so on the asterisk 1.6 box my rtp is almost flawless, like 99.9% of the rtp is streaming out |
| 19:54.27 | Kobaz | and on the 1.8 box the rtp is a bit choppy... not crazy choppy, but i would say like 95% |
| 19:55.02 | Kobaz | so just doing some really rudimentary testing like playing MusicOnHold with my local phone muted.. i can hear dropouts every 3-5 seconds |
| 19:55.11 | Kobaz | but I don't get that on the 1.6 box |
| 19:55.12 | Kobaz | sooooooo |
| 19:55.22 | Kobaz | what's a good place to start debugging this? |
| 19:57.48 | Kobaz | both using dahdi timing... and the local dahdi timing testing is coming back like 99.9 through 99.8 accuracy |
| 20:00.31 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 20:05.26 | aness | hi, I'm looking for rpms of asterisk 1.8 for centos 6 |
| 20:05.26 | aness | according to docs they should be available at http://packages.asterisk.org/centos/6/asterisk-18/%E2%80%A6, but latest there seems to be 1.4 |
| 20:06.35 | [TK]D-Fender | I jsut went and looked... and the OLDEST they have is * 11 |
| 20:06.49 | [TK]D-Fender | Go retrace the path back and look |
| 20:07.19 | Kobaz | waves in the general direction of [TK]D-Fender |
| 20:07.36 | aness | kinda done that, note the Centos _6_ |
| 20:08.38 | aness | and yes, I've seen the http://packages.asterisk.org/centos/centos-asterisk-1.8.repo file |
| 20:09.25 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-64.dllstx.fios.verizon.net) |
| 20:12.28 | aness | ok, I'm slow today.. 11 is later than 1.8? |
| 20:14.00 | Kobaz | yeah |
| 20:14.07 | Kobaz | 1.11 became 11 |
| 20:14.28 | aness | ah, I was reading it as 1.1 |
| 20:15.15 | aness | then I need to kick some developer and ask if his code works on later versions :) |
| 20:15.38 | aness | thanks! |
| 20:18.34 | file | Asterisk 11, Asterisk 12, and in the future... Asterisk 13! |
| 20:20.32 | aness | got it, but I also see that the 1.8 is a LTS |
| 20:20.40 | file | yup |
| 20:22.04 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 20:23.12 | aness | trying to upgrade a system from centos5 to centos6, using official packages.. guess I'll aim for 11 then |
| 20:26.20 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-052-155.pools.arcor-ip.net) |
| 20:26.58 | ghost75 | [TK]D-Fender: any idea what could cause the sip 480 message on outgoing calls? http://pastebin.com/sHziyVRa |
| 20:28.15 | [TK]D-Fender | Ask your provider |
| 20:28.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:28.50 | [TK]D-Fender | [15:07]anesskinda done that, note the Centos _6_ ----- > http://packages.asterisk.org/centos/6/ |
| 20:29.06 | [TK]D-Fender | I did note centos 6. Note that I drilled right back to that poitn... and * 11 is the lowest |
| 20:31.58 | aness | my mistake, I expected 11 to be 1.1 |
| 20:32.18 | [TK]D-Fender | \o/ |
| 20:33.53 | aness | [TK]D-Fender: btw, if you look at the "current" dir you'll find 1.4 |
| 20:34.51 | [TK]D-Fender | Where? |
| 20:35.16 | *** join/#asterisk kleszcz (~tick@linuxmafia.pl) |
| 20:35.17 | aness | http://packages.asterisk.org/rhel/6/current/x86_64/RPMS/ |
| 20:35.50 | [TK]D-Fender | Nope.. misreading again.... |
| 20:36.02 | aness | yeah, I'll shut up now :) |
| 20:36.06 | [TK]D-Fender | asterisk-sounds != asterisk |
| 20:36.34 | [TK]D-Fender | we flat-lined a while back :) |
| 20:37.08 | aness | clearly I did too.. |
| 20:37.17 | *** join/#asterisk jsjc (~Adium@217.Red-79-153-28.dynamicIP.rima-tde.net) |
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| 21:00.17 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
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| 21:10.09 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.192) |
| 21:13.55 | boom^time | Hey guys, does anybody know why when I use an AMI orginate from leg A to a local extension which then uses Dial, and then Leg B goes to the dial plan, I only receive a CDR for the originate to the Dial command, but not from the dial command nor the call to Leg B? |
| 21:16.28 | [TK]D-Fender | Three is no "call to leg B". |
| 21:16.37 | [TK]D-Fender | it is a single channel |
| 21:17.15 | [TK]D-Fender | using originate and landing in the dialplan is no different that using a SIP phone and placing a call in yoruself for example |
| 21:17.28 | [TK]D-Fender | The fact of which side started it does not matter |
| 21:18.07 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.136) |
| 21:19.27 | boom^time | The second the dial plan is hit and it goes to the Dial app it generates a CDR and then nothing is generated for the actual Dial app |
| 21:20.59 | boom^time | Where as before when I was dialing a phone number instead of using 'local/' I would have the disposition of the call and all of the cdr variables I set on the other side of the originate |
| 21:22.54 | boom^time | Instead just a single 1 second long CDR is generated |
| 21:28.05 | mjordan | so, here's the thing. Asterisk doesn't know about 'local' extensions. When your perform an origination, you've essentially created a new inbound channel - or in this case, two of them. That origination is completely independent of the channel that created them. |
| 21:28.21 | mjordan | I would suspect that you are / are not seeing certain things based on two criteria: |
| 21:28.39 | mjordan | (1) CDRs, by default, only show Answered channels. You probably aren't answering everything - and in your scenario, you have at least four channels at play |
| 21:29.40 | mjordan | (2) You may also be running afoul of "what is a dialed channel". Without seeing the dialplan, however, that's a bit hard to predict. In general, dialed channels don't get a CDR. However, what is a "dialed" channel can change if that dialed channel lives past the Dial application. |
| 21:30.34 | mjordan | At the point where you've involved Originate and multiple channels to construct your "call", you're leaving behind the kinds of scenarios that CDRs will show you in a simple set of records. |
| 21:31.30 | boom^time | I have unanswered=yes and congestion=yes in cdr.conf |
| 21:32.15 | boom^time | hmm. trying to digest what you've said |
| 21:34.17 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:34.21 | boom^time | This is all that exists in the local context that I'm dialing exten => _X.,1,Dial(SIP/trunk1/${EXTEN}) |
| 21:38.50 | boom^time | mjordan, so if I should basically throw out the default asterisk CDR concept at this point. How would you recommend logging CDRs? |
| 21:39.00 | mjordan | there isn't a way to configure this |
| 21:39.22 | mjordan | If, what it gives you, isn't what you want, you have to construct your own billing records using CEL |
| 21:39.27 | mjordan | or some other event driven based technology |
| 21:40.02 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:41.17 | *** join/#asterisk ggayan (~ggayan@200.71.197.170) |
| 21:41.37 | boom^time | I would be fine with 4 different CDR values for all of the channels if I could get them. But I only see one that is created immediately after the local context starts a Dial app |
| 21:43.42 | mjordan | So, you originated a Local channel to that extension, correct? |
| 21:44.03 | mjordan | just to double check, what version of Asterisk are you using? |
| 21:44.17 | mjordan | If you originated a Local channel to that extension, where did you put the other half? |
| 21:45.28 | boom^time | 11.5 |
| 21:45.47 | boom^time | the other half goes to a bit of dial plan that plays an audio stream |
| 21:45.56 | mjordan | k, do you Answer() that other half? |
| 21:46.31 | boom^time | Yes |
| 21:47.12 | boom^time | Which was required when I simply originated to my ITSP |
| 21:47.20 | boom^time | going to take it out now |
| 21:47.28 | mjordan | That's probably not a good idea. |
| 21:47.57 | mjordan | So you would have to Local channel halves, one issuing a Playback; the other performing a Dial. The one that performs the Dial should get a CDR, if the other end answers. |
| 21:50.46 | boom^time | Still didn't make a difference, not sure what's breaking it then. |
| 21:50.56 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 21:53.15 | boom^time | I'm using AMD on the second half, do you think that could be answering it? |
| 21:53.23 | boom^time | the record seems to generate right after |
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| 22:20.19 | vomit | Hello. I am experiencing some difficulties with res_fax session. Looks like it cannot reserve a fax session, anyone knows how to address this? How to increase the limit? This happens when a fax arrives. Logs and software versions here: http://pastebin.com/Ani4C3XU |
| 22:21.31 | [TK]D-Fender | [2014-02-05 15:34:09] WARNING[22743] res_fax_digium.c: Cannot reserve FAX session - session limit exceeded (max: 0). |
| 22:21.40 | [TK]D-Fender | I'd be guessing you didn't set up your licenses |
| 22:25.13 | boom^time | mjordan, nothing I do seems to help. It looks like the second both sides are bridged the cdr is created. I can't seem to get any additional ones to be created whether I run Answer or not. |
| 22:25.56 | boom^time | I know a lot of people use this method for call origination. Do they all parse CEL events or am I just missing something you think? |
| 22:26.36 | [TK]D-Fender | boom^time: You still on Originate / CDR issues? |
| 22:26.43 | boom^time | Yes :\ |
| 22:26.49 | boom^time | It's really bothering me |
| 22:27.18 | [TK]D-Fender | pastebin the full call |
| 22:27.26 | boom^time | sure |
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| 22:28.55 | vomit | [TK]D-Fender Thank you kindly |
| 22:28.56 | boom^time | Do you want it with debug on? |
| 22:28.59 | boom^time | It is really nasty |
| 22:29.12 | [TK]D-Fender | jsut verbose |
| 22:29.15 | boom^time | okay |
| 22:30.20 | boom^time | http://pastebin.com/cwQwxQay |
| 22:31.51 | [TK]D-Fender | Yup, single call |
| 22:32.12 | [TK]D-Fender | You originated, processed dialplan, and ended |
| 22:32.18 | [TK]D-Fender | no way you should have 2 CDR there |
| 22:33.33 | boom^time | So there isn't anything I can do |
| 22:33.55 | [TK]D-Fender | Its 1 call channel |
| 22:33.55 | boom^time | If I dial just a direct sip trunk to my ITSP everything works fine |
| 22:34.01 | [TK]D-Fender | There is nothing in need of doing |
| 22:34.09 | boom^time | just when I call the local channel and then have that Dial the number |
| 22:34.32 | boom^time | The CDR is generated before the call is over with? |
| 22:34.39 | [TK]D-Fender | Obviously not |
| 22:34.52 | [TK]D-Fender | Otherwise where would it get the DURATION from? |
| 22:35.33 | boom^time | What I mean is, I'm still on the phone, listening to the recording after the CDR has been created |
| 22:36.32 | boom^time | If I don't do a local/number@context and instead do a sip/trunk/number the CDR encapsulates the entire duration of time I'm on the phone |
| 22:36.48 | [TK]D-Fender | One thing you seem to have forgotten is how Local Channels get optimized backwards unless you tell it to do otherwise |
| 22:37.01 | [TK]D-Fender | <PROTECTED> |
| 22:37.15 | [TK]D-Fender | -- Executing [machine_check@binsons_CS-01-ScheduleOPM:7] Set("SIP/501-00000014", "CDR(amd_status)=HUMAN") in new stack |
| 22:37.27 | [TK]D-Fender | you should notice a clear renaming of the channel |
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| 22:38.26 | boom^time | How can I tell it to do otherwise? |
| 22:39.33 | [TK]D-Fender | <PROTECTED> |
| 22:43.46 | boom^time | [TK]D-Fender, thank you very much. Still a little confused on what it did need to compare logs |
| 22:43.56 | boom^time | But I have two CDR values now |
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