| 00:00.41 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 00:06.58 | *** join/#asterisk K3rmit (~asdf@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 00:07.10 | K3rmit | what does exten => a,1,VoicemailMain |
| 00:07.10 | K3rmit | <PROTECTED> |
| 00:07.14 | K3rmit | do? |
| 00:07.46 | pdtpatrick_ | Question: besides extensions.conf .. where else is ivr related information set? im seeing something like this: exten => s,n(ivr),Background(ivr-farheap-hello) |
| 00:10.35 | pdtpatrick_ | what does ${ARG1} refer to? I understand it is argument 1 but is it an argument from user input or something else? |
| 00:12.04 | paulc | K3rmit: It says if you press * during voicemail greeting, go run the voicemail login/msg retrieval app |
| 00:12.40 | paulc | pdtpatrick_: ${ARG1} is usually used if you're in a macro and passing variable data in |
| 00:13.54 | pdtpatrick_ | i c |
| 00:14.12 | pdtpatrick_ | im trying to find out where i can information related to the ivr |
| 00:16.27 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 00:17.30 | kuku | How do I forward caller ID number when forwarding calls int he dialplan via a trunk ? |
| 00:23.09 | WIMPy | kuku: By not changing it. |
| 00:23.19 | WIMPy | i.e. it happens by default. |
| 00:23.47 | kuku | but this dialplan somehow changes it :( What are my options |
| 00:24.05 | WIMPy | Remove the part that changes it. |
| 00:24.36 | WIMPy | Or it gets set in the peer definition. |
| 00:25.42 | kuku | Obvious answeres - but they helped. I need to do more tests - thanks ! |
| 00:25.42 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 00:25.43 | *** join/#asterisk docid (~PISSS@s76-9-57-75.nt.northwestel.net) |
| 00:27.14 | docid | anybody got any ideas what would be causing calls between extension on the same system (same switch even, no natting, local server, etc) to have no audio for the first 20 seconds and then work just fine, and yes, every call between internal extentions, calls to and from the outside work fine, all phones and server locked down to ulaw, * ver 1.6 |
| 00:27.47 | paulc | pdtpatrick_: You're going to need to be a bit more specific, because none of us know what you're doing with IVR. The dialplan defines a sequence of steps, which can play prompts, collect digits etc.. |
| 00:27.57 | kuku | canreinvite = ? |
| 00:28.14 | docid | kuku, that diested at me? |
| 00:28.17 | docid | directed |
| 00:28.55 | kuku | docid: yes |
| 00:29.00 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 00:29.12 | docid | set to no, trying it with yes |
| 00:29.25 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 00:30.09 | docid | no change |
| 00:31.08 | docid | it would at least make sense if it was audio, or no audio, but the big delay is confusing.... nothing else on that lan other than phones and asterisk |
| 00:31.37 | kuku | what phones |
| 00:32.13 | *** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) |
| 00:32.49 | docid | aastra 6731i's and 6757i's |
| 00:33.15 | docid | unfortunantly we dont have any other brands available for testing onsite |
| 00:33.23 | kuku | what about x-lite |
| 00:34.00 | kuku | can you show your sip.conf ? ( or at least the definition for you extension(S)) |
| 00:34.21 | pdtpatrick_ | paulc: here's what i have so far |
| 00:34.21 | pdtpatrick_ | http://pastebin.com/mhLQZh2D |
| 00:34.36 | docid | yep, lemme go a diggin |
| 00:34.48 | pdtpatrick_ | im trying to follow it to figure out what file is it accessing to play the greeting |
| 00:35.40 | docid | well, would be in a standard centos install in /var/spool/asterisk/sounds/ |
| 00:36.02 | docid | and the name of the playback file with the extention matching the codec of the channel |
| 00:36.03 | paulc | pdtpatrick_: Line 7, (ivr) is just a label. It's the Background app that's playing audio. The file is ivr-farheap-hello. Then you're waiting for 8 seconds for a digit. Which is weird, cos line 6 sets first digit timer to 4 seconds, and line 5 sets a 3 second interdigit timer |
| 00:36.15 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:37.13 | paulc | pdtpatrick_: And if I was going to be anal, I'd say it's shit to answer the line, then simulate ringing for X seconds. Because the customer is paying. I'd do Ringing, then Wait, then Answer - it's more correct from a telco billing perspective. |
| 00:39.38 | WIMPy | Wouldn't a Wait() without Answer() imply Ringing()? |
| 00:39.47 | pdtpatrick_ | Thanks.. im trying to pick up where someone left off really |
| 00:40.10 | pdtpatrick_ | @paulc: how then can i follow where ivr-farheap-hello is set? just search around the same file ? |
| 00:40.31 | WIMPy | It is the file name. |
| 00:40.59 | paulc | pdtpatrick_: Anything that looks ${LIKETHIS} is a variable. Things like ivr-farheap-hello are probably filenames. |
| 00:41.42 | paulc | so in your example, ${WAITTIME} is probably set somewhere else, either via Set(WAITTIME=5), or maybe just WAITTIME=5 in your [globals] context/block. |
| 00:41.48 | WIMPy | The name of the sound file that's played. |
| 00:42.11 | pdtpatrick_ | i c .. nice it is all starting to make more sense |
| 00:42.28 | WIMPy | Have you tried the book? |
| 00:42.33 | WIMPy | ~newbook |
| 00:42.34 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 00:43.05 | WIMPy | That's what happens if you haven't been here for a while. |
| 00:43.12 | WIMPy | ~thebook |
| 00:43.12 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 00:43.59 | pdtpatrick_ | getting that book now |
| 00:44.02 | pdtpatrick_ | thanks guys! |
| 00:46.18 | pdtpatrick_ | while i wait for book to arrive. that ivr-farheap-hello file is probably in /var/lib/asterisk/sounds ?? |
| 00:46.38 | WIMPy | yes |
| 00:46.47 | docid | kuku, heres sip conf with a couple of the extensions ... http://pastebin.com/TS9MrXwA |
| 00:48.11 | pdtpatrick_ | ha! thanks guys |
| 01:05.41 | docid | kuku, any ideas? im open to random suggestion that might have some effect, or lead me down the path towards possible solutions |
| 01:09.54 | *** join/#asterisk De_Mon (de_mon@fl-71-49-12-102.dhcp.embarqhsd.net) |
| 01:11.17 | De_Mon | i'm looking to build a system that can handle 200 concurrent SIP calls using ulaw for a codec. What sort of minimum should I start testing with? |
| 01:12.02 | WIMPy | Impossible to say. It depends on what you want to do with the calls. |
| 01:12.28 | De_Mon | just passthrough |
| 01:12.46 | De_Mon | i figure a P166 is too slow to start with but I really have no idea how high to go |
| 01:13.32 | De_Mon | like, is a a dual quad core xeon over kill? or a good place to start |
| 01:13.34 | WIMPy | If you use directmedia, you probably don't need much more. |
| 01:14.11 | WIMPy | If you really don't want to process voice in any way, that overkill. |
| 01:14.40 | De_Mon | really? I was looking at a switchvox system and it said a maximum of 75 concurrent calls so I was a bit puzzled |
| 01:14.56 | De_Mon | i guess they assume you're going to use all those features... |
| 01:15.16 | WIMPy | You often do. |
| 01:15.30 | WIMPy | Like VoiceMail e.g. |
| 01:16.58 | De_Mon | in the case where you are using a feature such as whisper or queues those a low cpu hit while things that do transcoding are where the cycles go? |
| 01:17.24 | De_Mon | this is a callcenter scenario where there's not going to be an IVR and just people making outbound calls for the most part. |
| 01:18.02 | WIMPy | If you want to spy or whisper, you need access to the media stream. |
| 01:18.33 | WIMPy | So you need to shift the data through the box, even if it needs no processing. |
| 01:19.33 | WIMPy | So that's not much more than forwarding of network traffic. |
| 01:19.57 | De_Mon | which is relitivly light on processing power |
| 01:20.07 | WIMPy | yes |
| 01:20.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 01:20.31 | WIMPy | Transcoding is the worst CPU wise. |
| 01:20.37 | De_Mon | well I feel a lot better. a switchvox appliance would probably work just fine in that case. |
| 01:21.12 | De_Mon | now to decide if we need to spend $18,000 for a GUI =) |
| 01:21.21 | WIMPy | I don't know what they are like. |
| 01:21.31 | WIMPy | Err? |
| 01:21.59 | De_Mon | wut? |
| 01:22.04 | WIMPy | Take the book and skip the GUI idea. |
| 01:22.26 | docid | ive been throughly impressed with the atom 525 boxes weve been building, but never tested them with a large callload |
| 01:22.48 | De_Mon | well, it's between a gui that any idiot can use or taking ownership of the system and managing the configs myself. |
| 01:23.07 | WIMPy | 525 is dual core, isn't it? |
| 01:23.15 | docid | tip: any idiot cant take the reins of asterisk and expect it to keep running even with a gui |
| 01:23.19 | docid | yeah wimpy |
| 01:23.33 | WIMPy | indeed |
| 01:24.01 | docid | and when it breaks, youll be very thankfull you know how to run it without one |
| 01:24.11 | De_Mon | well, let me rephrase that and say any windows admin can use it |
| 01:24.15 | WIMPy | yes |
| 01:24.28 | docid | De_Mon, im not seeing the difference |
| 01:24.31 | De_Mon | hahaha |
| 01:24.40 | De_Mon | my windows admins would take offense to that |
| 01:25.10 | docid | that being said, i do keep freepbx on most my boxes so people can use the web interface for their voicemail, and csr's can make small chanes |
| 01:25.33 | De_Mon | is virtualization worth considering? i haven't done any research on that topic in a good 3 years |
| 01:25.58 | Freeaqingme | De_Mon, yes, but you cant virtualize fxo/fxs cards so if you're planning on using those it isnt worth it |
| 01:26.03 | WIMPy | It works for some... |
| 01:26.41 | docid | well, qemu with the device directly assigned to the vm i would think ya would have a good shot |
| 01:26.42 | De_Mon | nope, pure SIP. But last time I tried it there were timing issues |
| 01:26.50 | docid | but i could see timing issues popping up |
| 01:27.09 | docid | De_Mon, you can buy timing dongles |
| 01:27.16 | docid | dunno if that would help though |
| 01:27.23 | Freeaqingme | you can probably overcome those by assigning dedicated cpu cores to the vm |
| 01:27.48 | De_Mon | i doubt it, mapping local hardware from Virtual host to a VM is kinda spotty |
| 01:28.28 | De_Mon | we decided it was problems with ztdummy and a virtualized usb port or somesuch |
| 01:28.28 | docid | depends what hypervisor ya got behind it |
| 01:28.43 | Freeaqingme | docid, name me one with which you can do that with pci-e devices? |
| 01:28.45 | De_Mon | latest vmware |
| 01:28.59 | docid | wouldnt know, havent moved to any pci-e servers |
| 01:29.02 | docid | heheh |
| 01:29.07 | docid | we use used hardware |
| 01:29.11 | docid | mostly |
| 01:32.49 | docid | havent really found a good reason to virt an * box |
| 01:34.15 | Freeaqingme | a reason would be to put multiple boxes on the same hardware |
| 01:34.21 | Freeaqingme | to reduce costs |
| 01:34.28 | De_Mon | or just a lower datacenter footprint |
| 01:34.42 | Freeaqingme | that'd be a result |
| 01:34.50 | De_Mon | we condenced about 3 racks into 10U this year with virtualization |
| 01:34.59 | docid | aye, i wasnt saying there werent good reasons... i just said i hadent found one in my experence |
| 01:35.05 | docid | nice |
| 01:35.07 | De_Mon | with the added benifit of not having to worry about equipment becoming obsolete |
| 01:35.12 | docid | been meaning to dig in again |
| 01:35.17 | docid | been away for a while... |
| 01:35.35 | Freeaqingme | De_Mon, yeah, same for us (approx), but the most important thing is that we're way faster in recovery with hardware issues |
| 01:35.45 | De_Mon | it sucks when you have 10 servers with hardware warranties about to expire that you can't renew much less find parts for on ebay =( |
| 01:39.05 | docid | is fantasizing longingly about hardware with warranties |
| 01:40.07 | De_Mon | you have no idea what you're missing! |
| 01:45.35 | *** join/#asterisk kaushal (~kaushal@115.246.154.6) |
| 01:45.41 | docid | hrmm, so any ideas on why today my phones started doing this odd thing where when ya call another extention on the same system (no nat, seperate network/switch/etc) will not pass audio for the first 12-20 seconds of the call, calls from outside trunks are fine |
| 01:46.08 | kaushal | what programming language is being used to develop Asterisk ? |
| 01:47.17 | De_Mon | C# ! |
| 01:47.26 | De_Mon | wohoo i have a status line in vim again |
| 01:47.42 | seraphie | lol |
| 01:47.48 | seraphie | kaushal: C |
| 01:48.11 | kaushal | C or C++ ? |
| 01:48.16 | kaushal | C# ? |
| 01:48.49 | seraphie | C |
| 01:48.53 | kaushal | ok |
| 01:49.00 | De_Mon | C# was a joke |
| 01:49.04 | kaushal | :) |
| 01:49.11 | seraphie | I believe there are some small pieces in C++, however, I have no experience with those. |
| 01:49.21 | seraphie | the vast majority at least is C. |
| 01:49.22 | kaushal | ok |
| 01:50.26 | De_Mon | all you gotta do is look at the source to see if they are c or cpp |
| 01:50.37 | kaushal | ok |
| 01:51.25 | seraphie | kaushal: http://svnview.digium.com/svn/asterisk/trunk/main/ |
| 01:54.55 | docid | ok, so ive traced it down i believe, but it seems to indicate another issue... (issue being calls between extensions have no audio for first 12 seconds) ... apperently the audio does not start untill the rtpkeepalive timer is hit, but im not natting, so this shouldnt have an effect right? |
| 02:01.04 | kaushal | seraphie: Thanks a lot |
| 02:01.09 | kaushal | De_Mon: Thanks |
| 02:06.43 | *** part/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |
| 02:07.28 | kaushal | seraphie: is it easy to learn to C :) |
| 02:07.46 | kaushal | C is so powerful still its being used |
| 02:09.04 | seraphie | Sure, it's easy to learn C. It's not easy to learn C on the Asterisk codebase. |
| 02:09.18 | florz | no, it's not easy to learn C, quite to the contrary |
| 02:09.30 | florz | as long as you think it is you probably don't know C |
| 02:11.06 | docid | hides his battered dusty scrolls of pascal |
| 02:11.20 | florz | *g* |
| 02:11.59 | seraphie | OK, yeah, so "easy" is a bad word. |
| 02:12.24 | seraphie | nevertheless, do not try to learn C by developing Asterisk. |
| 02:16.21 | seraphie | Learning C is a worthwhile pursuit. |
| 02:21.00 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 02:25.03 | kaushal | seraphie: yes |
| 02:25.12 | kaushal | I agree to it |
| 02:25.30 | kaushal | florz: I also feel the same |
| 02:25.40 | kaushal | C is very hard to learn |
| 02:25.45 | De_Mon | i learned C while fixing bugs and developing small featuers in asterisk.. alas thats as far as I got. |
| 02:25.48 | kaushal | atleast for me |
| 02:27.53 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 02:33.13 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
| 02:45.06 | russellb | seraphie: why not learn C on Asterisk? :-) |
| 02:48.02 | Maxus2 | hi people is this valid for a sip connection string in asterisk? SIP/[username]:[password]@[host]:[port]/[extension] |
| 02:48.53 | russellb | check the top of sip.conf.sample |
| 02:49.00 | russellb | it documents the valid SIP dial strings |
| 02:49.25 | Maxus2 | i have doen that, none appear to include the extension |
| 02:49.34 | Maxus2 | is putting the extensiont here valid/ |
| 02:49.35 | Maxus2 | ? |
| 02:50.15 | Maxus2 | the only ones that have extension, dont include username, password and host |
| 02:51.35 | russellb | try it and see what happens |
| 02:51.55 | Maxus2 | did, didn't work |
| 02:52.05 | russellb | k, then I guess it's not valid |
| 02:52.09 | Maxus2 | but im not sure if im missing somthing else |
| 02:52.23 | Maxus2 | well then can you tell me what would be valid? |
| 02:52.34 | russellb | what you see in sip.conf is the documentation of what's valid |
| 02:53.07 | Maxus2 | then you telling me it is impossible to pass an extension when using a username, password and host? |
| 02:53.43 | russellb | I guess so. |
| 02:53.59 | Maxus2 | yeah, im not buying it |
| 02:54.04 | russellb | ok. |
| 02:54.08 | Maxus2 | if you dont know, just say so. |
| 02:54.50 | russellb | i pointed you to where the documentation was for what is supported. |
| 02:55.04 | Maxus2 | thanks. |
| 02:56.54 | WiretapWork | Maxus2, putting the extension there is valid, so long as the other end expects it |
| 02:57.07 | Maxus2 | thanks WiretapWork |
| 02:57.16 | WiretapWork | sipgate use exactly that format |
| 02:57.19 | Maxus2 | that is a useful answer. |
| 02:57.32 | russellb | what is your problem? |
| 02:58.01 | Maxus2 | im dial that string and the other asterisk box is simple returning: |
| 02:58.43 | Maxus2 | Got SIP response 503 "Service Unavailable" back from *.*.*.* (ipaddress removed) |
| 02:59.10 | Maxus2 | i managed to get it to work once, but not sure if the structure of my dial command is correct |
| 02:59.23 | WiretapWork | hang on a tick |
| 02:59.31 | WiretapWork | the syntax you posted above is for a register string |
| 02:59.35 | WiretapWork | it is NOT for a dial string |
| 02:59.37 | Maxus2 | oh |
| 02:59.49 | Maxus2 | that could be my problem :) |
| 02:59.49 | WiretapWork | the dial string is SIP/<EXTEN>@<HOST>:<PORT> |
| 03:00.02 | WiretapWork | you can't use a username and password in a dial string |
| 03:00.04 | WiretapWork | no such thing |
| 03:00.17 | Maxus2 | what if im dialing to a remote box? |
| 03:00.22 | russellb | you can, but it doesn't support also including the extension ..... as the documentation says |
| 03:00.23 | *** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) |
| 03:00.31 | WiretapWork | Maxus2, one that requires registration? |
| 03:00.48 | Maxus2 | im having to do it without registration |
| 03:00.56 | WiretapWork | Maxus2, the SIP DIAL format never errs from the syntax I supplied |
| 03:01.01 | Maxus2 | just basically dialling and ip with a password and user name |
| 03:01.06 | WiretapWork | you can't do that |
| 03:01.34 | Maxus2 | so two boxes must be register in order to pass calls? |
| 03:01.38 | Maxus2 | registered |
| 03:01.59 | WiretapWork | nope |
| 03:02.05 | *** join/#asterisk jizzzum6 (jizzzum6@peacekeeperv6.darksideresearch.com) |
| 03:02.16 | WiretapWork | if they have peer definitions for each other that are _Static_ then no |
| 03:02.21 | WiretapWork | if theyre _dynamic_ then yes |
| 03:02.28 | WiretapWork | but obviously one must be a static host |
| 03:02.35 | WiretapWork | otherwise they won't be able to find each other at all |
| 03:02.54 | Maxus2 | hmmm okay |
| 03:03.12 | Maxus2 | will have a play and see how i go, thanks for the help. |
| 03:04.19 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 03:04.41 | russellb | the syntax says ... SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 03:04.55 | russellb | the username part is slightly misleading. that's the user part of the URI, not the authentication username. |
| 03:05.25 | russellb | so SIP/EXTEN:mypassword::myauthenticationusername@ipaddress |
| 03:05.47 | russellb | that should work. |
| 03:05.50 | Maxus2 | cool but still no extension |
| 03:05.56 | Maxus2 | oh wait |
| 03:06.01 | Maxus2 | sorry missed that at the start |
| 03:06.16 | russellb | yeah, it's the user you are dialing |
| 03:06.23 | russellb | or extension, whatever you want to call it |
| 03:07.19 | WiretapWork | russellb, bear in mind that you'll probably be treating that dial as from external, so you'll have to assign a 'VDDI' for the call |
| 03:07.59 | russellb | is off to bed |
| 03:08.01 | russellb | good luck. |
| 03:08.42 | Maxus2 | thanks! |
| 03:14.14 | Maxus2 | yeah no luck with that |
| 03:14.40 | Maxus2 | im using realtime and have been told i cant do registrations in the database |
| 03:14.55 | Maxus2 | so im trying to do it in the dial string, so i dont define things in the sip.conf |
| 03:15.03 | Maxus2 | no luck so far. |
| 03:22.01 | stope | I'm having fax issues: I have a linksys 2102, * 1.8.4 with the digium FFA module and keep getting this: T.38 re-INVITE detected but no fax extension |
| 03:22.32 | stope | faxing in from the pstn to * works over 711, but faxing from the linksys 2102 to * if iffy |
| 03:22.47 | stope | anything obvious that I'm missing? |
| 03:25.56 | WiretapWork | it seems to be looking for a fax extension to send the call to |
| 03:26.00 | *** join/#asterisk OldGrumpy (Whacko@p5B312AEB.dip.t-dialin.net) |
| 03:35.57 | stope | stupid question coming..... whats a fax extension? |
| 03:36.19 | stope | I just want to get it from my faxing machine to the server and land as a tif |
| 03:40.21 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 03:49.39 | *** join/#asterisk dinesh___ (~dinesh@46-126-192-144.dynamic.hispeed.ch) |
| 03:50.13 | dinesh___ | hey all, I'm setting my asterisk server up again (and migrating from 2.4 to 2.6), everything is working as before, expect 1 little thing |
| 03:50.49 | dinesh___ | playback() claims that it is currently playing my .gsm file, but i don't hear anything on the line (SIP) |
| 03:51.07 | dinesh___ | voice does go through |
| 03:51.20 | dinesh___ | any idea where this could come from? |
| 03:55.04 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cmqmjiixsapxteym) |
| 03:56.44 | WiretapWork | dinesh___, asterisk hasn't even reached version 2.0 yet??? |
| 03:56.46 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 04:01.05 | dinesh___ | oh yeah right, i meant 1.6 |
| 04:03.08 | dinesh___ | well well i don't know what's going on, but the "reload" command in the asterisk -r console is not found anymore |
| 04:05.07 | WiretapWork | dinesh___, core reload |
| 04:05.13 | WiretapWork | its not just one reload now |
| 04:05.16 | WiretapWork | there are several |
| 04:10.26 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 04:11.20 | dinesh___ | actually |
| 04:11.23 | dinesh___ | i fixed both issues |
| 04:11.45 | dinesh___ | the thing is that i deleted all the files in /etc/asterisk, and left only sip.conf, extensions.conf and modules.conf |
| 04:11.52 | dinesh___ | because the other ones i have no clue what they are doing |
| 04:12.12 | dinesh___ | this approach was working fine with asterisk 1.4, but it doesn't look like it's similar with 1.6 |
| 04:12.34 | dinesh___ | now that i put the default files back, and overwrote only sip.conf and extension.conf it's all ok |
| 04:14.22 | dinesh___ | hm well no, that was just for the "reload" part |
| 04:14.32 | dinesh___ | removing the "noanswer" from my playback fixed the playback issue |
| 04:14.35 | dinesh___ | but that's not optimal |
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| 06:01.28 | ectospasm | I have a quick question: suppose I'm a SIP server (not necessarily Asterisk), and in the normal INVITE process I send a 183 Session Progress message. Shouldn't I send a 200 OK that should be ACK'd before I send out RTP? My customer is using Asterisk to send to a remote SIP provider, and I see <183 followed by <RTP, with no <200 OK or >ACK, then the far end sends <403 Forbidden |
| 06:02.21 | ectospasm | '<' meaning coming from the SIP provider, and '>' meaning coming from Asterisk (my customer's machine) |
| 06:06.50 | jizzzum6 | my brain hurts |
| 06:11.24 | kaldemar | ectospasm: in a case of early media, RTP might begin before the 200 OK. |
| 06:11.54 | ectospasm | kaldemar: yeah, I was thinking that might be the case. |
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| 06:29.01 | schmidts | good morning |
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| 06:36.50 | oej | Good morning |
| 06:37.03 | X-Rob | Heya oej |
| 06:38.02 | X-Rob | I need to have a good look at applerasin in the not too distant future. |
| 06:38.10 | oej | Great |
| 06:38.30 | oej | The FreePBX people have done a lot of testing of it lately |
| 06:39.08 | X-Rob | Yeah, I got them onto it. |
| 06:39.26 | X-Rob | I'm not MEANT to be doing any work until 1st July |
| 06:39.34 | X-Rob | because I'm not getting paid for it |
| 06:39.38 | schmidts | Morning olle |
| 06:39.43 | X-Rob | but I've been playing with a pile of astribanks |
| 06:40.07 | X-Rob | A literal pile: http://hipbx.org/sites/default/files/styles/large/public/field/image/IMAG0258_0.jpg |
| 06:40.18 | schmidts | oej have you seen the reply from siemens to the problem we had yesterday? issue 19281 |
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| 06:40.40 | oej | schmidts: no |
| 06:41.03 | oej | x-rob: Yeah, that's a huge pile |
| 06:41.42 | X-Rob | It's (un)surprising how many things break when you have that many. |
| 06:41.50 | X-Rob | like I can't turn them all on at the same time - half of them crash. |
| 06:41.59 | schmidts | maybe you can take a look, i am not sure if they didnt understand the rfc wrong or if maybe asterisk does something wrong ;) |
| 06:42.02 | X-Rob | I'm mildly irritated. |
| 06:42.11 | X-Rob | but I'm being nice and have emailed support. |
| 06:44.42 | X-Rob | schmidts, I may be stupid, but what's the point of a device sending a media line and then saying 'DON'T USE THIS'. |
| 06:45.08 | *** join/#asterisk coppice (~chatzilla@m121-203-207-242.smartone-vodafone.com) |
| 06:45.15 | X-Rob | I've alway ssaid that SIP is a protocol written by a committee. |
| 06:45.17 | X-Rob | heya coppice |
| 06:45.18 | oej | schmidts: One always learn. I need to check if they're right in the assumption of order of preference though. Have never heard that. |
| 06:45.25 | schmidts | X-Rob dont ask me, thats siemens style ;) |
| 06:45.37 | oej | The a=sendrecv is definitely wrong, they don't comment on that |
| 06:45.49 | coppice | SIP is a protocol written by people who had never seen a telephone |
| 06:46.02 | oej | coppice: What's a "telephone" ? |
| 06:46.17 | schmidts | coppice so you mean h323 is better than? |
| 06:47.02 | coppice | H.323 builds on 100 years of learning what works well in telephony. The IETF wanted non of that perversity in their network |
| 06:51.11 | coppice | SIP is so horrible, they had to write the MGCP specs just to make it look good by comparison |
| 06:54.51 | X-Rob | you know, I'd never actually LOOKED at the MGCP protocol until just then, coppice |
| 06:54.54 | X-Rob | now I need eye bleach. |
| 06:55.02 | wdoekes2 | good morning |
| 06:56.04 | schmidts | morning walter |
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| 07:06.37 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 07:06.43 | jacc0 | gooooood morning all!! |
| 07:06.45 | jacc0 | :) |
| 07:08.58 | schmidts | morning jacc0 |
| 07:09.19 | jacc0 | I've been reading some RFCs last night and i hate to admit that siemens is right |
| 07:09.26 | jacc0 | :P |
| 07:09.52 | jacc0 | there responce was a copy/past from that rfc |
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| 07:13.04 | jacc0 | so if the port is ZERO we MUST NOT use it |
| 07:15.02 | schmidts | jacc0 which RFC? |
| 07:15.46 | jacc0 | RFC3264 |
| 07:16.44 | jacc0 | end beside that; asterisk is doing something else wrong; the first one should be the preferd |
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| 07:19.44 | jacc0 | and ONLY when streams of DIFFERENT types are present it means that it wants both; not when they arew the same type |
| 07:20.50 | jacc0 | So I'll try and make a patch that does exactly that |
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| 07:30.17 | schmidts | jacc0 i dont think this will be solved by a simple patch cause its a deep infrastructure thing |
| 07:30.35 | schmidts | only using the first media descriptor will not work, you have first check if asterisk could even handle this |
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| 07:34.42 | Ineluctable | I want to install asterisk 1.8 on a CentOS 5.6 fresh install. I would like to use chan_gtalk.so, and res_jabber.so. My question is can i install these modules if I install asterisk from the asterisk/digum repo? Or is the only way to use these modules is to compile from source? |
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| 07:35.26 | ChannelZ | they really need to be compiled from the same source as asterisk was |
| 07:36.07 | jacc0 | @Ineluctable: I'm not sure. but what I usualy do do is : install from repo (to get things like the astcanary and dependencies and stuf installed) |
| 07:36.20 | jacc0 | then download source en compile: |
| 07:36.30 | jacc0 | ./configure |
| 07:36.42 | jacc0 | make menuselect |
| 07:36.44 | jacc0 | make |
| 07:36.46 | jacc0 | make install |
| 07:36.57 | ChannelZ | there's not much point in using the packages at that point |
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| 07:38.00 | jacc0 | why not? it creates the user/group and installs astcanary en runs asteriks in realtime prio; do you want to do this all manualy? |
| 07:38.04 | Ineluctable | What about the dep factor as jacc0 pointed out? |
| 07:40.00 | wdoekes2 | why realtime/canary? and useradd is not too hard to do.. I'd advise against having an asterisk installed by package next to a manually installed one |
| 07:40.12 | ChannelZ | I don't like mixing a package and then building from source yourself, as when the package gets updated it could stomp on or break your current setup |
| 07:40.21 | wdoekes2 | exactly |
| 07:40.31 | ChannelZ | Do one or the other. |
| 07:41.50 | ChannelZ | If you want to install the asterisk *source* package to let it fetch other development dependencies you might need out of laziness, fine.. |
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| 07:48.24 | Ineluctable | ChannelZ: asterisk18-devel ? |
| 07:48.59 | ChannelZ | maybe, I'm not familiar with CentOS and it's package system and what they might call things |
| 07:49.48 | ChannelZ | Usually 'devel' on other distros are headers/libs needed for developers to use whatever it is, and there is a 'src' or 'source' version which is the actual source code for a package. |
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| 07:51.58 | Ineluctable | OK. If that is the case why cant I just install asterisk18 from the repo with asterisk18-devel do the ./configure make menuselect deal to add the modules? Would this not work, or would it recompile everything? |
| 07:53.53 | jacc0 | after make menuselect you should recompile: make |
| 07:53.59 | ChannelZ | Let's back up for a second - have you looked, are other channel modules separate packages? |
| 07:54.00 | jacc0 | and install : make install |
| 07:54.46 | ChannelZ | and/or are you sure that particular package doesn't already have the gtalk channel driver? |
| 07:56.39 | Ineluctable | asterisk18 does not cone with it afaik |
| 07:56.42 | kaldemar | at least the .deb packages from digium already have chan_gtalk and res_jabber. |
| 07:56.44 | Ineluctable | *come |
| 07:57.45 | Ineluctable | I will give it a try and see. |
| 07:58.45 | ChannelZ | They are included in the base source but aren't built by default - however most people who build binary packages will build separate packages for 'optional' modules like this, if they didn't just build it into the main package (since it's a loadable module in Asterisk anyway, there's no harm in supplying it with the package..) |
| 08:00.18 | Ineluctable | how? |
| 08:00.39 | ChannelZ | how what? |
| 08:02.13 | Ineluctable | were you saying if it is not built into the package then build it myself? |
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| 08:04.50 | ChannelZ | I'm saying do you even know that it isn't before we go down this road? |
| 08:05.11 | ChannelZ | I'm browsing an RPM searcher thing on the net and it looks like it already is. |
| 08:05.59 | ChannelZ | though I don't know exactly what package from what repository you'd actually be getting |
| 08:06.28 | Ineluctable | asterisk / digum repo |
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| 08:11.58 | ChannelZ | it looks like those modules are already built in the main package |
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| 08:14.30 | Ineluctable | Well after installing asterisk 1.8 asterisk -rx "module show" only shows res_adsi ADSI Resource 0 |
| 08:14.54 | ChannelZ | that only means it's not loaded probably because it's not configured |
| 08:15.24 | Ineluctable | what res_adsi or chan_gtalk? |
| 08:15.49 | ChannelZ | gtalk. Look in /usr/lib/asterisk/modules (hopefully) |
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| 08:17.11 | Ineluctable | no chan_gtalk, or res_jabber |
| 08:17.31 | ChannelZ | hmm then I don't know where that package came from |
| 08:17.55 | ChannelZ | I just looked at one of the .deb files from digium and it's in there |
| 08:18.56 | Ineluctable | Yeah nothing for centos. That is why I am not using asterisknow |
| 08:19.51 | ChannelZ | then I'd just remove the package, download the source and build the whole thing. |
| 08:20.33 | ChannelZ | There aren't many dependencies for the base of asterisk besides the development tools (compiler et al). For gtalk/jabber in particular you need libiksemel-dev |
| 08:21.37 | Ineluctable | alright I will give that a try. |
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| 09:01.02 | basti1101 | hello, has anybody a valid configuration for asterisk with sip over tls with client certificate validation? |
| 09:02.26 | kaldemar | basti1101: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 09:04.45 | basti1101 | thx, but i know this page, there the client certification isn't enforced by the server. it's possible to connect without client certificate too |
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| 09:06.51 | basti1101 | there is the undocumented option tlsverifyclient for sip.conf. but it doesn't seen to work. with this option my server automaticly restarts on connecting |
| 09:10.52 | jacc0 | where can I find the License v3.0 papers to sign if I want to upload patches? |
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| 09:16.12 | sigmounte | hi ! |
| 09:18.57 | jacc0 | okay, I found this: https://issues.asterisk.org/view_license_agreement.php |
| 09:19.09 | jacc0 | should I print it and sign it? where do I send it? |
| 09:19.38 | *** join/#asterisk StaRetji (~BigAll@80.93.240.172) |
| 09:19.48 | StaRetji | hello good people |
| 09:20.52 | jacc0 | hi |
| 09:21.10 | StaRetji | I'm testing webphone on a website, so I've made a sip user and it works great. Now, my problem is want to block incoming call to that number |
| 09:21.41 | StaRetji | at least until i prepare sip users for each account |
| 09:21.59 | StaRetji | I've looked on google, but I must be blind lol |
| 09:23.25 | jacc0 | exten = 200,1,hangup() ; if the number is 200 |
| 09:24.50 | StaRetji | jacc0: thx so much, I found blocking anon calls etc, but couldn't find how to block to a specific sip user, in your example user is 200 |
| 09:24.55 | StaRetji | thx :) |
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| 09:31.49 | jacc0 | @StaRetji: be carefull if you use: _X. in your dialplan |
| 09:32.06 | jacc0 | it might still be possible to call 200 |
| 09:32.42 | jacc0 | if some sip phone dials: 150&sip/200 |
| 09:32.44 | StaRetji | jacc0: oh, I used it |
| 09:33.25 | StaRetji | [a2billing] exten => _X.,1,Answer |
| 09:33.43 | jacc0 | if the dialplan looks like dial(sip/${EXTEN}) |
| 09:34.18 | jacc0 | if ${EXTEN} = 150&sip/200 it will result in : dial(sip/150&sip/200) |
| 09:34.26 | jacc0 | ;) |
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| 09:35.20 | StaRetji | jacc0: thx for this, must admit I will have to figure out first lol |
| 09:35.39 | StaRetji | I have exten => _00.,1,Dial(SIP/${EXTEN},60) |
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| 09:37.41 | jacc0 | so if a sipphone calls : 00231424234&sip/200 |
| 09:37.50 | jacc0 | phone 200 will ring |
| 09:42.24 | jacc0 | http://www.securiteam.com/securitynews/5BP380K19G.html |
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| 09:45.38 | jacc0 | thinks everybody is checking his dialplans now |
| 09:46.10 | jacc0 | :P |
| 09:46.36 | kaldemar | func FILTER is good for extensions that allow characters. |
| 09:49.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:51.38 | StaRetji | jacc0: will take a risk ;) |
| 09:51.57 | StaRetji | thanks once again, I really appreciate your help |
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| 11:34.36 | QuantumSchema | Good moring all! |
| 11:34.50 | Lantizia | Lo, anyone here use a2billing (or could point me to a more appropriate IRC channel for it?) |
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| 11:35.51 | Chainsaw | leifmadsen: Got a patch for you on https://issues.asterisk.org/view.php?id=19192 that allows me to use 1.8.4 now. |
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| 11:42.27 | jacc0 | is it my connection or is issues.asterisk.org very slow? |
| 11:42.41 | QuantumSchema | I'm having a little problem using mpg123 to stream music from ShoutCast... the process starts (viewable from "ps -ax" and the dialplan output in the CLI shows that it is playing the right class... just no music.... |
| 11:42.53 | QuantumSchema | Does this musiconhold.conf look right? http://pastebin.com/60nRVayw |
| 11:43.47 | jacc0 | it's not my connection; I guess issues.asterisk.org is down |
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| 11:49.04 | coppice | well, it gets depressed being full of all those issues that sit there unfixed for so long |
| 11:53.10 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:53.23 | jacc0 | thats no reason for DDOSing issues.asterisk.org |
| 11:53.30 | jacc0 | :p |
| 11:54.07 | jacc0 | O, it's working again |
| 11:54.10 | jacc0 | :) |
| 11:56.27 | jacc0 | @coppice: for that reason I joint asterisk dev. today; maybe you should do the same |
| 11:56.27 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:56.41 | jacc0 | I’ve created a patch that will make asterisk ignore all media streams with port 0 and therefore will fix interoperability between asterisk and the HiPath 3000 V8 M5T SIP Stack/4.0.26.26 and will make asterisk more compliant with RFC3264 |
| 11:56.58 | jacc0 | you can find it here : https://issues.asterisk.org/view.php?id=19281 |
| 11:57.00 | *** join/#asterisk frawd (~francois@242.Red-81-38-28.dynamicIP.rima-tde.net) |
| 11:58.14 | QuantumSchema | Any one? Please? :-) |
| 12:00.58 | Chainsaw | QuantumSchema: With Asterisk, I found the best way to find out is to try... |
| 12:01.29 | Chainsaw | QuantumSchema: It seems daring to me to use a remote MP3 feed as MOH, as it could skip or fail entirely for reasons outside of your control. |
| 12:02.40 | puzzled | QuantumSchema: those " look a bit odd. are you sure that is correct |
| 12:03.01 | QuantumSchema | Thanks Chainsaw... at the moment we're just seeing how feasable using an outside MOH source is. |
| 12:03.24 | puzzled | bad idea if you ask me |
| 12:03.59 | QuantumSchema | Puzzled, from what I've seen, when you use the -@ switch the prompt is expecting quotes but I'll give that whirl with out them... |
| 12:04.58 | QuantumSchema | Same effect without quotes. |
| 12:05.38 | QuantumSchema | I am noticing that if I list processes, it lists /usr/local/bin mpg123 -q -s --mono -r 8000 -f 8192 -b 2048 http://184.107.159.100:8100... notice the space between bin and mpg123. |
| 12:05.41 | *** join/#asterisk adnc (~akif@unaffiliated/adnc) |
| 12:06.12 | QuantumSchema | it's the only process listed that seperates the executable from the path. |
| 12:06.17 | *** join/#asterisk frawd (~francois@242.Red-81-38-28.dynamicIP.rima-tde.net) |
| 12:06.24 | adnc | hello, could someone tell me if there is anything wrong with this peace of config http://pastebin.com/d5uWDvin |
| 12:06.36 | puzzled | QuantumSchema: buffering takes too long. there are seconds of silence and then MoH starts. my clients would not find that acceptable |
| 12:08.04 | QuantumSchema | Puzzled, from what I've read in the docs, the moments of silence are only existant the first time the process starts to buffer. Once the process is up and running, buffering continues and further MOH instances tap into the existing buffer. Is that not true? |
| 12:08.46 | kaldemar | adnc: what do you expect it to do and what is the unexpected behavior? |
| 12:09.15 | puzzled | QuantumSchema: afaik MoH starts every time it is requested. but as Chainsaw said you just need to test it and see how it works |
| 12:09.32 | QuantumSchema | I'm with you on the testing part... |
| 12:09.38 | QuantumSchema | I just can't get the stream to ever start. |
| 12:09.41 | adnc | kaldemar, I've written a tool that generates this sort of configs. it should behave like an IVR. a friend just sait this wouldnt work. for my knowledge it looks good, but I wanted to ask some experts here |
| 12:09.58 | puzzled | QuantumSchema: so afaik it is not start once, buffer and play but start, buffer play to client #1, start buffer, play to client#2 etc. |
| 12:10.11 | QuantumSchema | Hmmm... |
| 12:10.40 | QuantumSchema | I do know the switches are correct for mpg123, if I execute it from the command line it actually pulls down the stream. |
| 12:11.01 | kaldemar | adnc: "won't work" is not enough to work on. there's nothing wrong with it as is, but there can be many issues depending on the surroundings. you need a CLI output of a call. |
| 12:11.05 | puzzled | yes I tried that too and it works but it takes too long for the MoH to kick in. at least from .nl where I am |
| 12:11.15 | leifmadsen | Chainsaw: thanks! marked as a blocker for 1.8.5 |
| 12:11.48 | puzzled | QuantumSchema: just use local MoH and if you are worried about I/O just put the files on tmpfs |
| 12:11.58 | puzzled | hi leifmadsen |
| 12:12.02 | leifmadsen | ohai |
| 12:12.06 | jacc0 | hi :) |
| 12:12.08 | leifmadsen | ramdisks ftw! :) |
| 12:12.13 | puzzled | indeed :) |
| 12:12.22 | Chainsaw | puzzled: Can you not mitigate that with cachertclasses=yes in the config though? |
| 12:12.49 | QuantumSchema | What I'm confused about though is the process list.. |
| 12:13.01 | jacc0 | just signed the v3.0 License but my patch still states: license pending; can you help me with that leif? |
| 12:13.10 | puzzled | Chainsaw: not familiar with that option. is that a 1.8 option? |
| 12:13.27 | leifmadsen | jacc0: you just asked in another room and I answered you there |
| 12:13.32 | adnc | kaldemar, thank you very much |
| 12:13.35 | QuantumSchema | There's a space between /usr/local/bin and mpg123... does that look right? |
| 12:13.45 | puzzled | no |
| 12:13.46 | Chainsaw | puzzled: I thought that worked in 1.6 already; but I might be mistaken. |
| 12:14.14 | puzzled | Chainsaw: heh I am not familiar with 1.6 either. have planned to move to 1.8 or trunk in the next two weeks or so |
| 12:14.18 | Chainsaw | puzzled: Basically it only has 1 MOH stream, always keeps it going, and just uses that for clients. So they get patched in halfway somewhere. |
| 12:14.19 | QuantumSchema | disregard that last one about the process. |
| 12:14.38 | Chainsaw | puzzled: My MOH is a 12 minute or so loop that sounds great even from a random point. So that saves resources. |
| 12:14.39 | asterisk-learner | Hey guys, how do u think the acquisition of Skype by Microsoft will affect the world of VOIP and asterisk ? |
| 12:14.54 | puzzled | Chainsaw: sounds good. thanks for the info |
| 12:14.56 | Chainsaw | asterisk-learner: Less income for Digium if Skype for Asterisk disapears. |
| 12:15.06 | Chainsaw | asterisk-learner: But more business if Windows users finally get the hang of SIP. |
| 12:15.06 | leifmadsen | I'm sure they'll be fine :) |
| 12:15.22 | *** join/#asterisk Leitaox (~Leitaox@189.20.94.66) |
| 12:15.57 | Chainsaw | asterisk-learner: Remember that Skype isn't open; it is not using the SIP protocol. So it's an outsider. |
| 12:16.04 | asterisk-learner | what do u mean by "get the hang of SIP." ? |
| 12:16.28 | asterisk-learner | yeah but i heard u could connect a skype account to asterisk or smthg like that .... |
| 12:16.34 | Chainsaw | asterisk-learner: That Windows users might finally use the standard like the rest of us, instead of using some weirdo closed protocol client. |
| 12:16.35 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-umhotnzouocjnpml) |
| 12:16.39 | asterisk-learner | there was effort to make them compatible or so ... |
| 12:16.41 | puzzled | asterisk-learner: Skype will die but not before it is renamed Microsoft Unified Office Communication Super Duper Thingy 2011, is integrated into Lync and they chuck it full of security holes |
| 12:16.59 | asterisk-learner | puzzled: hahaha |
| 12:17.01 | leifmadsen | Chainsaw: https://issues.asterisk.org/view.php?id=19289 <-- same issue? |
| 12:17.19 | jacc0 | lol |
| 12:17.29 | leifmadsen | puzzled: I don't know.... Skype is a pretty good brand name to use |
| 12:17.44 | Chainsaw | leifmadsen: I think that's different. "My" bug is a segfault, very definitive. |
| 12:18.09 | Chainsaw | leifmadsen: It's likely that I am affected by what they've reported, because on a bad day my SIP stack does go unresponsive and I have to kill -9 asterisk to get things going again. |
| 12:18.09 | puzzled | problem is that J. Rosenberg of SIP fame (thanks for the NAT handling by the way) is now a Microsoft fellow so it will be interesting to see how that pans out |
| 12:18.13 | asterisk-learner | Chainsaw: so ur saying either Windows users will hate skype and look for an alternative (SIP) or in case Microsoft does it right (probably not) , then asterisk will be weakened ? |
| 12:18.13 | leifmadsen | Chainsaw: ok sounds good -- just didn't want you two working on the same thing independently |
| 12:18.39 | Chainsaw | asterisk-learner: Asterisk will not be weakened, because Skype has very different goals. |
| 12:18.50 | leifmadsen | asterisk is just another end point |
| 12:19.02 | Chainsaw | asterisk-learner: Asterisk is a PBX that speaks SIP. Skype does not speak SIP. It speaks some weirdo super-encrypted secret protocol. |
| 12:19.11 | puzzled | leifmadsen: it's a good brand name but with little traction in the Enterprise, hence the integration into Lync etc. |
| 12:19.30 | leifmadsen | puzzled: I could see Skype being the consumer division and integration with Lync being Enterprise |
| 12:19.39 | puzzled | yup me too |
| 12:19.43 | Chainsaw | puzzled: The paid video conferencing was actually quite good. We were considering using it instead of Marratech, which was embraced & extinguished by Google. |
| 12:20.00 | Chainsaw | puzzled: Now Microsoft will be embracing & extinguishing Skype... so that makes me sad. |
| 12:20.02 | asterisk-learner | Chainsaw: true but skype is so widespread, I even have it on Android phone |
| 12:22.14 | Chainsaw | asterisk-learner: It's still a closed application that a single company has control over. Even if Microsoft does not kill the Android client off immediately, it will die the next upgrade. |
| 12:23.13 | Chainsaw | asterisk-learner: This is how we slowly but surely lost our Marratech when Google killed it. |
| 12:23.51 | coppice | sure, skype is widespread today, but ms has the power to change that |
| 12:24.03 | asterisk-learner | Chainsaw: mmm, does this situation compare to mysql / oracle or is this a diffrent story ?> |
| 12:24.25 | Chainsaw | asterisk-learner: Very different. MySQL is open, you can fork it if you don't like what the company in charge is doing. |
| 12:24.42 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:24.55 | *** join/#asterisk Oliver1 (~oliver@manz-590ef161.pool.mediaWays.net) |
| 12:25.09 | asterisk-learner | ok but do u feel mysql is sitll being update like it used to be before oracle's acquisition, or is the process slower now ? |
| 12:25.09 | Chainsaw | asterisk-learner: Same with Asterisk. If Digium were to be bought out by someone malicious, you could fork the code and run off. |
| 12:25.33 | Chainsaw | asterisk-learner: MySQL got forked into MariaDB and now has a competitor. If MySQL slows down, people will jump ship. |
| 12:25.37 | Chainsaw | asterisk-learner: This helps to keep Oracle honest. |
| 12:26.02 | Chainsaw | asterisk-learner: Same thing happened with OpenOffice. Got forked, a competitor appeared. I think they've already given up on that one actually. |
| 12:26.17 | file | gotta make my mind up... which seat can I take? it's Friday Friday gotta get down on Friday |
| 12:26.32 | asterisk-learner | MariaDB never heard of it before... |
| 12:27.02 | asterisk-learner | OpenOffice is not being developed anymore ?? |
| 12:27.11 | jacc0 | hardly |
| 12:27.34 | jacc0 | Ubuntu now includes libereoffice by default |
| 12:27.36 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 12:28.00 | Chainsaw | As I predicted, OpenOffice -> LibreOffice was as quick and dramatic as XFree86 -> X.Org |
| 12:28.29 | asterisk-learner | mmmm |
| 12:29.21 | coppice | x.org gave us more development. libreoffice is starved of rsources |
| 12:29.49 | Chainsaw | coppice: Libreoffice dropped the copyright assignment weirdness. This gets them a lot more developers. |
| 12:29.50 | wdoekes2 | leifmadsen: my 19289 is not about kill -9'ing or about the deadlock. it's about not being able to bind() after it fails once |
| 12:30.02 | Chainsaw | coppice: Not to mention an immediate merge of "OpenOffice Go". |
| 12:30.38 | asterisk-learner | Since we are on the subject of Oracle and Open Source .... what is the suit that Oracle have against Google concerning Android / Dalvik ... ? |
| 12:31.16 | Chainsaw | asterisk-learner: They feel that Dalvik is an unauthorised Java platform, due to changes that Google have made. |
| 12:31.31 | coppice | openoffice needs some hard core development of its spreadsheet, or its going nowhere |
| 12:31.54 | Chainsaw | asterisk-learner: More of a software patent issue then anything else. It might kill java. That'd be fun. Better performance everywhere as students are forced to use C++ instead! |
| 12:32.18 | Chainsaw | coppice: A lot is happening on that front, see the commit logs. |
| 12:32.37 | asterisk-learner | Chainsaw: u mean anyone is allowed to write a JVM , but they felt Google made a bad implementation that could hurt java... ? like Microsoft did once ? |
| 12:32.49 | asterisk-learner | Chainsaw: Amen to C++ !!! |
| 12:32.52 | asterisk-learner | more work to us |
| 12:32.54 | asterisk-learner | :-P |
| 12:33.15 | eXcAliBuR | Hey guys I be backs |
| 12:33.20 | Chainsaw | asterisk-learner: There are a few "levels" of java that you can implement, which mandate operands that you must implement. |
| 12:33.39 | Chainsaw | asterisk-learner: Google picked J2ME, but did not implement a few of the mandatory operands. |
| 12:33.51 | asterisk-learner | could you elaborate plz ? |
| 12:34.15 | Chainsaw | asterisk-learner: Sure. http://en.wikipedia.org/wiki/Dalvik_(software)#Lawsuit |
| 12:34.24 | coppice | if the openoffice spreadsheet performed like excel 2003, but without the crashes, they'd have a winner |
| 12:35.45 | asterisk-learner | you mean some parts of the java specifications oblige you to implements some parts in a very specific way ? |
| 12:35.50 | asterisk-learner | and Google bypassed it ? |
| 12:35.55 | asterisk-learner | (according to Oracle) |
| 12:36.43 | leifmadsen | I've never been able to get past how 1999 OpenOffice looks |
| 12:36.51 | Chainsaw | asterisk-learner: Pretty much, yeah. |
| 12:37.29 | Chainsaw | asterisk-learner: And if they dislike what you have done, they can sue you for any frivolous software patent they have. |
| 12:38.27 | asterisk-learner | Chainsaw: thx for the clarification |
| 12:38.59 | asterisk-learner | and if Oracle wins the case, will Andoroid be removed from all cell phones accross the world ? |
| 12:39.01 | asterisk-learner | :-P |
| 12:39.05 | Chainsaw | (Then again, I feel any software patent is frivolous, the US courts disagree quite strongly with that. Take my opinion with a suitable amount of salt) |
| 12:39.19 | *** part/#asterisk oelewapperke (wapper@eth1.mine.nu) |
| 12:39.20 | leifmadsen | I like salt |
| 12:39.52 | coppice | I like salt, so I pepper my meals with it |
| 12:40.41 | leifmadsen | classy! |
| 12:40.52 | leifmadsen | actually when I have food I don't put extra salt on it, just pepper :) |
| 12:41.20 | coppice | actually I hate salt, but that doesn't make a one liner |
| 12:46.34 | leifmadsen | :) |
| 12:49.34 | Chainsaw | Very secretive bugtracker. 19272 is forbidden. |
| 12:50.24 | jacc0 | I like salt aspecialy when i digest my messages |
| 12:50.50 | leifmadsen | Chainsaw: they thought it might be a security issue |
| 12:51.08 | Chainsaw | leifmadsen: Oh, right. |
| 12:51.27 | Chainsaw | is not convinced one could remotely trigger this |
| 12:51.56 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 12:54.04 | eXcAliBuR | hey leifmadsen, |
| 12:54.08 | eXcAliBuR | congrats |
| 12:54.18 | leifmadsen | yes, I'm still alive, and it's glorious |
| 12:54.27 | eXcAliBuR | I saw your pic yesterday |
| 12:54.35 | eXcAliBuR | your one fine looking dude |
| 12:54.42 | leifmadsen | o.O |
| 12:54.45 | eXcAliBuR | O.o |
| 12:54.54 | eXcAliBuR | :} |
| 12:55.04 | leifmadsen | did you just congratulate me on being good looking.... ? |
| 12:55.05 | eXcAliBuR | isn't it nice to get a compliment? |
| 12:55.11 | leifmadsen | yes I suppose so :) |
| 12:55.47 | eXcAliBuR | have you had much experience with getting ruby on rails to play nice with asterisk? |
| 12:56.06 | leifmadsen | I've never used RoR but perhaps someone else here has :) |
| 12:56.39 | *** join/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 12:57.10 | eXcAliBuR | Perhaps you would know a better solution than what i'm trying to make... I want to have asterisk call of list of students to alert them of school being cancled. |
| 12:57.38 | leifmadsen | easy enough to do with the language of choice and AMI |
| 12:57.45 | eXcAliBuR | I was gonna have RoR create a .call file and put it in the outgoing folder |
| 12:58.04 | leifmadsen | use AMI for that kind of thing |
| 12:58.11 | leifmadsen | .call files are not really designed for what you want to do |
| 12:58.21 | eXcAliBuR | oh |
| 13:00.24 | eXcAliBuR | do u know of any ready made apps that might do it? |
| 13:00.34 | leifmadsen | not off the top of my head, but they may exist |
| 13:01.19 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 13:01.37 | *** join/#asterisk Starscreamer (~Starscrea@mailgate.mackellar.co.uk) |
| 13:01.56 | eXcAliBuR | i don't think my ami is working... i enabled it, but it won't connect with telnet |
| 13:01.56 | Nugget | telnet is eeeeeeevil! |
| 13:02.02 | eXcAliBuR | telnet |
| 13:02.07 | eXcAliBuR | hmmm |
| 13:02.18 | eXcAliBuR | kicks nuggets |
| 13:02.23 | eXcAliBuR | is that a bot? |
| 13:03.27 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 13:03.43 | *** part/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 13:04.21 | leifmadsen | nugget isn't, but i think that response is :) |
| 13:04.25 | leifmadsen | I <3 telnet!@ |
| 13:07.25 | *** join/#asterisk wonderworld (~ww@port-92-201-34-220.dynamic.qsc.de) |
| 13:07.31 | eXcAliBuR | telnet is refusing my connections |
| 13:07.40 | leifmadsen | did you enable AMI? |
| 13:08.01 | eXcAliBuR | yes, I followed page 457 and page 458 |
| 13:08.25 | leifmadsen | eXcAliBuR: http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html |
| 13:09.08 | eXcAliBuR | hmmm it seems to work if i telnet from the same box |
| 13:09.13 | eXcAliBuR | why not external machine |
| 13:12.59 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:13.50 | ruyo | eXcAliBuR, maybe you have bindaddr=127.0.0.1. |
| 13:13.52 | kaldemar | eXcAliBuR: it's probably listening to 127.0.0.1 only. |
| 13:14.19 | eXcAliBuR | i have all 0's |
| 13:14.29 | mocker | Woo, got on the Google Music beta. |
| 13:14.40 | eXcAliBuR | so it should let all ip's work |
| 13:14.44 | eXcAliBuR | i would think |
| 13:14.58 | leifmadsen | We're sorry. Music Beta is currently only available in the United States |
| 13:15.07 | leifmadsen | shakes his Canadian fist wildly |
| 13:15.40 | ruyo | eXcAliBuR, then you may have deny/permit configured to 127.0.0.1 only. |
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| 13:43.13 | eXcAliBuR | leifmadsen where abouts do you live? |
| 13:44.25 | leifmadsen | Canada |
| 13:45.06 | eXcAliBuR | ;D |
| 13:45.08 | eXcAliBuR | me too |
| 13:45.11 | eXcAliBuR | i'm in quebec |
| 13:45.23 | mocker | leifmadsen: Land of no distribution rights. :( |
| 13:45.51 | leifmadsen | mocker: eh? |
| 13:46.51 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 13:48.25 | mocker | leifmadsen: I'm assuming that's why you can't get music.google.com |
| 13:48.37 | leifmadsen | mocker: oh ya |
| 13:50.05 | eXcAliBuR | leifmadsen: how did you become so proficient in asterisk? |
| 13:50.25 | eXcAliBuR | did u just sit for hours using it? |
| 13:50.29 | leifmadsen | eXcAliBuR: many hours of using it :) |
| 13:50.34 | leifmadsen | I've been using it for 7+ years |
| 13:50.49 | eXcAliBuR | i have trouble completing stuff... i get excited at first... then it fades |
| 13:50.54 | mocker | leifmadsen: Teaching classes probably helped. |
| 13:51.00 | mocker | learns best when teaching others. |
| 13:51.18 | eXcAliBuR | how do u stay motivated? |
| 13:51.33 | leifmadsen | mocker: indeed, I know all I know from hanging out with people like oej, ssokol, jsmith, russellb, jvanmeggelen, etc.... |
| 13:51.38 | leifmadsen | eXcAliBuR: it's hard sometimes ;) |
| 13:51.53 | leifmadsen | I have a reasonably short attention span, so I do lots of things |
| 13:52.03 | leifmadsen | sometimes not to the benefit of getting things done as quickly as I should |
| 13:52.27 | jkroon | leifmadsen, similar problem here. task switching/distraction is a major issue. |
| 13:52.48 | jkroon | and yea, using the software is probably the best way to learn it. 3+ years myself, and still discovering new things every other day. |
| 13:53.19 | leifmadsen | jkroon: ya and I'm not good about going back to issues after switching which is the problem -- I'm in the process of offloading customer support elsewhere so I don't get random, "OMG IT'S ON FIRE" calls |
| 13:58.29 | Lantizia | tzafrir, hey you about? |
| 14:02.31 | QuantumSchema | Alrighty... I ditched streaming MoH. |
| 14:02.40 | mocker | leifmadsen: Guh, I hate that. |
| 14:02.49 | mocker | I have some legacy systems I setup and those calls suuuuuck. |
| 14:02.59 | leifmadsen | mocker: I have one running ABE 2.4.2 :) |
| 14:03.10 | QuantumSchema | Here's a good one.... is there a way to triger an action if an agent misses a call (ring no answer) from a queue? |
| 14:03.35 | QuantumSchema | I'd like the agent to be paused if the miss a call (ring no answer) but I'm not quite sure how to sneak my way into that event. |
| 14:04.24 | leifmadsen | QuantumSchema: you could be using Local channels and then evaluate it after the Dial() inside the Local channel to see if it just rang through, then do PauseQueueMember() |
| 14:04.41 | mocker | ; Autopause will pause a queue member if they fail to answer a call |
| 14:04.49 | QuantumSchema | Ooooo autopause!!! :-D |
| 14:04.49 | mocker | ;autopause=yes |
| 14:04.55 | QuantumSchema | time to look... :-D |
| 14:05.58 | QuantumSchema | Thanks a ton mocker! |
| 14:13.23 | mocker | QuantumSchema: np. |
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| 14:28.26 | Jcook_5xData | I have a problem i have this context http://pastebin.com/2YARrB9t I when I place a call to 6169888392 instead of going to [voice-custom-5] it is going to the catch all. how do I make respect the number |
| 14:30.05 | jacc0 | there is no goto(voice-custom-5) |
| 14:30.07 | kaldemar | Jcook_5xData: don't have exten = 6169889392 in the same context. |
| 14:30.09 | *** join/#asterisk cerberus_za (~coert@196.215.151.194) |
| 14:30.26 | jacc0 | so it will never goto [voice-custom-5] |
| 14:30.31 | kaldemar | Jcook_5xData: a better match always wins |
| 14:31.58 | Jcook_5xData | OK thanks for the help I will try that |
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| 14:37.38 | Jcook_5xData | Sorry it was my own fault typo:P |
| 14:39.19 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 15:03.31 | jacc0 | have a nice weekend all! bye!! |
| 15:04.36 | Deeewayne | eXcAliBuR, see pm |
| 15:12.24 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
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| 15:20.47 | *** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 15:20.58 | DerkKo | Hello guys |
| 15:21.22 | leifmadsen | ohai |
| 15:24.25 | *** part/#asterisk AndyML (~alauppe@unaffiliated/andymli7) |
| 15:24.51 | DerkKo | I have a weird problem and looking for troubleshooting pointers.... I have an asterisk box that i use to record calls, my calls are sip to sip. At random asterisk introduces a garble sound in the call. I have not been able to put my finger on why it is happening, i turned up as much verbosity and trying to collect data now. But has this happen to anyone? It is weird.. BTW my codec is g711u on both channels no codec changes |
| 15:24.51 | DerkKo | by asterisk.... any pointers here |
| 15:28.24 | _Corey_ | DerkKo: Check your box for unusual activity when you're hearing these effects. Sometimes excessive load or things like this will impact quality |
| 15:28.39 | DerkKo | low is minimal |
| 15:28.53 | DerkKo | i tried both asterisk 1.8 and 1.6.2 |
| 15:29.10 | DerkKo | no hardware level errors either |
| 15:29.51 | DerkKo | oh and i can reproduce in different hardware |
| 15:29.55 | _Corey_ | I had one once where a customer was complaining of bad quality on a box with normal load averages, but ever hour for ONE minute at exactly :44 mins past the hour sendmail was choking on 40k messages it couldn't deliver |
| 15:31.05 | DerkKo | mmmm ... how did you figure it out? |
| 15:31.19 | DerkKo | I been thinking mayber its because o dont have zaptel or ztdummy |
| 15:31.47 | DerkKo | but i dont use meetme or anything else that would require clock source, the only thing i use is monitor and chan_sip |
| 15:32.00 | leifmadsen | shouldn't be necessary then |
| 15:32.59 | _Corey_ | DerkKo: We had a script that monitored process load every 15s and after a few hours it was obvious. Some things are hard to see with 5m averages |
| 15:35.01 | *** join/#asterisk phoenixsampras (~phoenixsa@static-190-181-38-121.acelerate.net) |
| 15:35.19 | phoenixsampras | help!! IVR menu doesnt respond to keys? just rebooting fixes the problems, and happens every 10 mins |
| 15:35.45 | leifmadsen | ~31337 |
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| 15:36.30 | *** join/#asterisk Janos (~Janos@186.32.53.52) |
| 15:36.56 | leifmadsen | infobot: 31337 is <reply> 31337 is the public conference room for #asterisk people to hang out. It's not a support room however. You can connect to the conference room at sip:31337@shifteight.org or ISN: 31337*1273 |
| 15:36.56 | infobot | okay, leifmadsen |
| 15:37.01 | leifmadsen | ~31337 |
| 15:37.01 | infobot | 31337 is the public conference room for #asterisk people to hang out. It's not a support room however. You can connect to the conference room at sip:31337@shifteight.org or ISN: 31337*1273 |
| 15:37.18 | Janos | hello, simple question, is the Dial command always suppose to return after the timeout ? or is there circumstances that could make it "break" ? |
| 15:37.28 | phoenixsampras | help! |
| 15:38.03 | _Corey_ | Janos: "break"? |
| 15:38.21 | Janos | _Corey_, does not continue and exit |
| 15:38.55 | _Corey_ | I've never seen it not return some kind of DIALSTATUS |
| 15:39.33 | _Corey_ | it may stop before the ring timeout because of a telco response or whatever |
| 15:39.40 | leifmadsen | agreed |
| 15:39.58 | leifmadsen | if the other end "answered" the call and immediately hung up or dropped, that would appear as an answered call |
| 15:40.02 | leifmadsen | and would not continue in the dialplan |
| 15:40.26 | phoenixsampras | how to reset all asterisk by command line? |
| 15:40.35 | leifmadsen | you could check the status of ${DIALSTATUS} in the 'h' extension |
| 15:40.40 | leifmadsen | phoenixsampras: core restart now |
| 15:40.48 | leifmadsen | it'll drop all active calls though |
| 15:41.30 | JonathanRose | There's also 'gracefully' and 'when convenient' |
| 15:41.42 | JonathanRose | Those won't drop your calls. |
| 15:41.52 | JonathanRose | But they'll make you wait. |
| 15:41.52 | *** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net) |
| 15:42.05 | JonathanRose | gracefully doesn't accept new calls until it's restarted |
| 15:42.20 | JonathanRose | when convenient just restarts once there are no more calls. |
| 15:42.24 | leifmadsen | when convenient -- just waits until there are no active calls (could be a long time) |
| 15:42.36 | leifmadsen | I don't necessarily like using those because you never know "when" the restart is going to happen |
| 15:42.46 | _Corey_ | use gracefully carefully, it won't place calls either so if you're on a busy system with people navigating IVRs, ugly things can happen :) |
| 15:42.47 | leifmadsen | I usually just keep monitoring and/or schedule a particular time to restart |
| 15:42.55 | *** join/#asterisk m_tadeu (~quassel@89-180-97-202.net.novis.pt) |
| 15:43.02 | m4xx | can you nest expressions? ie: GotoIf($[ $[${x} < 26| ${x} > 28] | ${x} = 32]?PRIORITY_LABEL) |
| 15:43.31 | leifmadsen | m4xx: yes |
| 15:43.38 | pabelanger | Anybody ever used res_http_post.so? |
| 15:43.43 | leifmadsen | but surround each one |
| 15:43.55 | leifmadsen | you didn't quite do it right in your example |
| 15:44.43 | leifmadsen | GotoIf($[$[20 < 21] | $[foo = bar] | $[${paul} = awesome]]?somewhere |
| 15:44.45 | leifmadsen | ) |
| 15:44.56 | phoenixsampras | leifmadsen: that command will reset everything? for example for some reason the IVR menu doesnt respond to the keys, and incoming calls are halted |
| 15:45.04 | leifmadsen | phoenixsampras: it restarts asterisk |
| 15:45.22 | phoenixsampras | leifmadsen: how can i fix those 2 problems? |
| 15:46.01 | m4xx | leifmadsen: so i'm only missing the $[] around "${x} = 32" correct? |
| 15:46.11 | leifmadsen | m4xx: not correct |
| 15:46.26 | leifmadsen | you're kinda doing it funny.... |
| 15:46.40 | leifmadsen | GotoIf($[ $[${x} < 26 | ${x} > 28] | $[${x} = 32]]?PRIORITY_LABEL) |
| 15:46.45 | leifmadsen | ya I guess that's valid |
| 15:46.59 | m_tadeu | hi, I'm using memberdelay in queues.conf to warn the caller that the call is going to be answered. I was expecting that the caller would receive the regular calling tone instead of MOH within that time. How can I make this happen? |
| 15:47.31 | leifmadsen | m_tadeu: make a MoH class that plays ringing? |
| 15:47.35 | *** join/#asterisk JimDickenson_ (~dickenson@c-71-193-138-117.hsd1.or.comcast.net) |
| 15:47.39 | JonathanRose | I was about to make that joke :< |
| 15:48.20 | leifmadsen | I wasn't joking :) |
| 15:48.33 | JonathanRose | Well, funny solution. |
| 15:48.36 | m_tadeu | leifmadsen: and how does it change from the regular MOH for the ringing one? |
| 15:48.49 | m4xx | actually i'm looking for != 32 |
| 15:48.52 | leifmadsen | m_tadeu: you set it in the dialplan? |
| 15:49.10 | JonathanRose | Is there a ringing MoH profile in the default configs? |
| 15:49.16 | JonathanRose | I never bothered to look at MoH. |
| 15:49.58 | JonathanRose | I'm guessing no. |
| 15:51.57 | m_tadeu | leifmadsen: please give me some direction. when the call goes into a queue, the dialplan is not calling the shots. how can the dialplan say that when the agent picks the call, to change to the ringing moh? |
| 15:53.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:53.24 | *** join/#asterisk Tim_Toady (~fuzzy@athedsl-165855.home.otenet.gr) |
| 15:53.26 | leifmadsen | m_tadeu: I misunderstood, you can't |
| 15:53.34 | DerkKo | UGH! |
| 15:53.37 | leifmadsen | I've never run into that issue as I don't use memberdelay |
| 15:53.42 | leifmadsen | DerkKo: ARGH! |
| 15:54.03 | DerkKo | there is not evidence of anything causing this garble on these calls |
| 15:54.37 | DerkKo | trying to paste the output of my full verbosity debug but its too large for pastebin |
| 15:54.38 | DerkKo | :-) |
| 15:56.11 | m_tadeu | :) so my problem is this. since the caller may be waiting for several minutes listening the moh, I want to warn the caller for 5 secs that the call is going to be picked by an agent. I'm figuring that a regular calling tone is the best choice. Any sugestions for such an aproach? |
| 15:57.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:59.02 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003507.dsl.bell.ca) |
| 16:02.08 | DerkKo | just for a pease of mind, ztdummy is not needed for monitor and mixmonitor correct? |
| 16:07.44 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:08.52 | kaii | hm. how can i disable SRTP in SIP entirely? when asterisk is compiled with SRTP support, it automatically offers SRTP. encryption=no in the global section of sip.conf seems to not do the job. |
| 16:09.14 | kaii | addition: i use mysql realtime peers, the table structure does not contain a "encryption" field. |
| 16:09.29 | kaii | i am using 1.8.4-rc3 |
| 16:13.59 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 16:14.29 | kaii | i had three segfaults in res_srtp yesterday |
| 16:16.37 | leifmadsen | unload res_srtp? |
| 16:18.58 | Janos | mm, ok this is what i'm trying to achieve, i'm receiving calls on my DAHDI interface, all this calls are sent to my operator '2000', my operator will then transfer calls to the right extensions, some of these extensions are sip channels on the same asterisk and other are sip channels on another asterisk |
| 16:19.38 | Janos | now this is my operator context and dialing macro, http://pastebin.com/f7qhrkWY |
| 16:20.30 | m_tadeu | how "heavy" are AM scripts? I mean, does it make sence to run a script for each call(say about 10 simultaneous calls)? or will this really affect asterisk's job? |
| 16:20.58 | Janos | as you can see the macro is very simple, call the extension and then call the operator again after the timeout |
| 16:21.49 | Janos | this works when the extension dialed by the operator is a local sip channel, channel rings and if not picked up after the timeout the call return the my operator |
| 16:22.51 | Janos | but when the same is done on an extension on the remote asterisk the first dial exits with a non-zero error and the second one never gets executed, now this used to work in 1.4 |
| 16:23.06 | Janos | i just recently upgraded to 1.6 and it's not working anymore |
| 16:23.18 | _Corey_ | Janos: is the other end "answering" the channel somehow? |
| 16:23.25 | Janos | yes |
| 16:23.31 | Janos | and it rings |
| 16:23.32 | leifmadsen | then that's the way it should work |
| 16:23.44 | leifmadsen | what does the SIP trace say? |
| 16:23.49 | leifmadsen | is it 180 Ringing? |
| 16:23.55 | _Corey_ | Well, it needs to NOT answer... |
| 16:24.02 | leifmadsen | ringing != indicate answer |
| 16:24.18 | leifmadsen | unless it is answering then sending back audio that sounds like ringing or something |
| 16:24.20 | _Corey_ | If it gets 180 or 183 all will work |
| 16:24.24 | Janos | ok ok, not the extension on the other asterisk never answer |
| 16:24.38 | Janos | i mean, the end user never picks up |
| 16:24.40 | leifmadsen | the other asterisk is probably answering the call then |
| 16:24.43 | _Corey_ | Yeah, but does the dial plan on the other asterisk "answer" even though the caller doesn't pick up |
| 16:24.48 | leifmadsen | I bet it does |
| 16:24.49 | Janos | yes |
| 16:24.51 | Janos | it does |
| 16:25.07 | Janos | verbose does show the call as answer |
| 16:25.12 | _Corey_ | yeah, so your local asterisk can't distinguish between that and a person answering so... |
| 16:25.26 | Janos | kk that makes sense |
| 16:25.32 | Janos | how to avoid it ? :P |
| 16:25.48 | leifmadsen | if the other end answers, there is nothing you can do about it |
| 16:25.59 | _Corey_ | the other asterisk needs to just "Dial()" |
| 16:26.00 | leifmadsen | you could look into the flags in the Dial() app that tells it to continue in the dialplan |
| 16:26.11 | leifmadsen | but that's more of a hack -- asterisk is doing as it should |
| 16:26.28 | Janos | i control both asterisk so i can change anything on the remote asterisk as well |
| 16:26.44 | Janos | let me get some info on what the other asterisk is doing |
| 16:28.49 | Janos | it just does 'exten => _3XXX,1,Dial(SIP/${EXTEN},20,tTrdwW)' and 'exten => _3XXX,n,Hangup', may that r flag is causing it to answer ? |
| 16:29.36 | leifmadsen | I'd guess the tT and wW flags first actually |
| 16:29.42 | _Corey_ | r shouldn't |
| 16:29.48 | Janos | kk let me try removing them |
| 16:29.51 | leifmadsen | it causes the audio path to be required to go through asterisk and might require asterisk to answer the call |
| 16:30.34 | _Corey_ | my money is on the 'Ww' |
| 16:31.11 | kaii | leifmadsen: unloading / noload'ing res_srtp sounds pragmatic |
| 16:37.23 | Janos | well, removed tTwW from both sides, but still had to remove d from the second site, r flag does seem to work, so right now it only has r flag on both sides, might just had been the d flag, will test on that later |
| 16:37.49 | kaii | i do not understand this part of the documentation regarding SIP and SRTP encryption: "Due to limitations of SDP, encryption=yes in sip.conf results in a call with only a secure media offer, therefor forceencrypt=yes would be redundant in sip.conf." |
| 16:38.10 | Janos | thanks a lot guys, really appreciate the help, this had been driving me crazy for a while now |
| 16:38.39 | kaii | i do not understand which limitations are refered to .. other vendors can of course offer AVP _and_ SAVP in the same SDP |
| 16:44.29 | kaldemar | kaii: maybe a poor choice of words. "limitations of asterisk's implementation of SDP" maybe? |
| 16:45.43 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:46.02 | m_tadeu | where can I check the full list of AM commands and events for asterisk1.8? |
| 16:48.12 | serafie | m_tadeu: https://wiki.asterisk.org/wiki/display/AST/AMI+Actions <-- Is this what you mean? |
| 16:48.36 | kaldemar | m_tadeu: in CLI, "manager show commands", or https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI) |
| 16:52.47 | m_tadeu | serafie, kaldemar, thanx...that is exactly what I needed |
| 17:03.02 | QuantumSchema | Alrighty.... I'm think this one through... Instead of queue having a timeout for a call and then failing over, is there away that if all the agents are busy it rolls over to the fail over destination? |
| 17:03.23 | QuantumSchema | (wow I'm tired. my grammar just took a nose dive) |
| 17:03.54 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:03.54 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:04.45 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:08.05 | DerkKo | im a little confuse... for asterisk 18 i can still use ztdummy correct ? |
| 17:08.12 | paulc | QuantumSchema: isn't that just a timeout of zero? |
| 17:08.27 | paulc | DerkKo: Wouldn't you rather switch to DAHDI? |
| 17:08.47 | DerkKo | yea i downloaded dahdi-firmware |
| 17:08.53 | QuantumSchema | Well, that would make sure that the call doesn't time out and stays in the queue right? |
| 17:09.07 | DerkKo | but modprobe ztdummy fails |
| 17:09.14 | QuantumSchema | I'd like it to stay in the queue but fail over if all agents are busy/unavailable. |
| 17:09.32 | paulc | ztdummy=zaptel, you're now using DAHDI? it's one or the other? |
| 17:10.11 | paulc | QuantumSchema: So you send the call into the queue, but if all agents are busy and the call can't be serviced, you want to send the call somewhere else? Trying to understand what you're trying to accomplish? |
| 17:11.48 | QuantumSchema | paulc: You're right. |
| 17:11.58 | QuantumSchema | That's what I'm looking to do. |
| 17:13.10 | DerkKo | so whats the equivalent of ztdummy for dahdi |
| 17:13.30 | DerkKo | sorry for the stupid question and delay, trying to read trough documentation at the same time |
| 17:14.28 | paulc | DerkKo: a dummy timing device equivalent is built into DAHDI I think |
| 17:14.33 | kaldemar | DerkKo: there is no separate module for the dummy functionality anymore. the core dahdi module includes it. there was dahdi_dummy in older dahdi versions. |
| 17:14.52 | m_tadeu | serafie, kaldemar: sorry, I'm missing the events as in here http://www.voip-info.org/wiki/view/asterisk+manager+events but this is from 2004. where can I find an up to date version of this? |
| 17:16.46 | paulc | QuantumSchema: Not sure how to do what you want (ie "immediately"). You may have to balance between a timeout that is acceptable to callers versus allows agents to answer |
| 17:16.49 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 17:21.20 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 17:26.30 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:27.06 | _Corey_ | m_tadeu: wiki.asterisk.org |
| 17:29.59 | *** join/#asterisk MrNemus (~MeNemus@firewall.drgutah.com) |
| 17:32.00 | m_tadeu | _Corey_: thanx...but I can't find the AMI events there....only AGI commands and AMI actions |
| 17:33.26 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:36.47 | *** join/#asterisk serafie (~erin@nat/digium/x-cplllmewbcadcxmd) |
| 17:36.55 | _Corey_ | hmm good point |
| 17:37.22 | _Corey_ | seems like this is close: https://wiki.asterisk.org/wiki/display/AST/Some+Standard+AMI+Headers |
| 17:37.28 | _Corey_ | but not exactly what you want |
| 17:37.39 | _Corey_ | they're spread around in the sources too |
| 17:38.26 | _Corey_ | you could always do a "grep manager_event -r ." in your source folder i guess ;) |
| 17:38.28 | kaldemar | m_tadeu: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/doc/manager_1_1.txt |
| 17:40.02 | kaldemar | might be the same as this: https://wiki.asterisk.org/wiki/display/AST/AMI+1.1+Changes |
| 17:42.07 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:44.02 | *** join/#asterisk gerhard7 (~quassel@82-171-103-215.ip.telfort.nl) |
| 17:45.38 | m_tadeu | _Corey_, kaldemar, thanx guys....I think I'll make a small manager and dump the events for the flow I'm using...might be simpler, 'cos I'll find out which ones I'm supposed to care about :) |
| 17:47.13 | QuantumSchema | paulc: sorry, I was thinking it through and had work get in the way. |
| 17:47.42 | paulc | QuantumSchema: No worries. I'm the same at work :-) |
| 17:48.06 | QuantumSchema | paulc: LoL If only asterisk was the primary concern... |
| 17:48.14 | *** join/#asterisk drdru (~Adium@76.77.182.168) |
| 17:48.22 | QuantumSchema | paulc: What I was thinking was kind of like having the distant end being a kind of like an agent but not able to be hit untill all other agents had been tried. |
| 17:48.32 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:48.47 | ruben23 | hi guys where i can find teh voicemail directory..? |
| 17:48.54 | paulc | QuantumSchema: I hear ya! Asterisk all day would be sweet, compared to some of the other stuff I do. |
| 17:49.25 | paulc | QuantumSchema: So.. hmm.. what about if you had a local channel as an agent, with a priority lower than all others? Hmm.. but.. not sure - maybe that would only handle one call at a time |
| 17:49.29 | QuantumSchema | ruben23:/var/spool/asterisk/voicemail/ |
| 17:49.30 | paulc | kludgey workaround |
| 17:51.07 | QuantumSchema | paulc: Hmmm... how would that work if the distant end had the highest priority... |
| 17:51.17 | drdru | I'm looking for a SIP host that will allow me to connect my users directly to their service from their mobile devices, allow them to receive incoming and make outgoing telephone calls - can someone tell me how to find such a provider? |
| 17:51.49 | QuantumSchema | paulc: by the way, it the big idea about this is that the distant end is an outsourced call center to catch over flow for a sales queue we have. |
| 17:52.30 | paulc | QuantumSchema: I was thinking give your real agents a higher priority, and a dummy/local agent a lower priority.. calls would always go to real agents if available, or hit the lower priority agent if not.. where you could "do something" with the call. |
| 17:52.49 | QuantumSchema | hmmmm.... |
| 17:53.04 | paulc | drdru: I would have thought any providers would be fine with this - the real issue is getting a SIP client running on your users' devices, no? |
| 17:53.37 | paulc | QuantumSchema: depending on how the far end is set up, you could have their agents log in to your system.. or jsut have a bunch of virtual agents that send calls their way |
| 17:53.40 | QuantumSchema | drdru & paulc: the iPhone and Android devices have sip clients... |
| 17:53.53 | QuantumSchema | add a hosted SIP/IPPBX provider and you should be good to go. |
| 17:54.04 | QuantumSchema | I've played with both clients on the different devices.... |
| 17:54.29 | drdru | QuantumSchema: yeah, I am looking for a SIP host - I already have my own SIP app :) |
| 17:54.37 | drdru | right now I'm using sipgate.com |
| 17:54.43 | drdru | but want to know of alternatives |
| 17:55.03 | drdru | I thought I would be able to make it work with Vonage, but had some technical problem with my app making calls on their network |
| 17:55.07 | drdru | probably had the wrong encoding |
| 17:55.15 | paulc | ~istp-us |
| 17:55.20 | paulc | ~itsp-us |
| 17:55.31 | paulc | frowns - what am I missing? |
| 17:55.36 | paulc | infobot: itsp-us |
| 17:56.11 | paulc | drdru: Yeah, probably not Vonage.. but there's a bunch of others.. vitelity.com, voip.ms, flowroute.com |
| 17:56.13 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-185-miami.txt.hfc.comcastbusiness.net) |
| 17:56.26 | drdru | thank you |
| 17:56.30 | drdru | is there a way to search for them? |
| 17:56.38 | drdru | I have tried googling a lot, but not found anything |
| 17:57.31 | paulc | drdru: seriously? ;-) http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 17:57.55 | paulc | or http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 17:58.36 | drdru | awesome |
| 17:58.37 | drdru | thanks |
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| 17:59.07 | *** part/#asterisk drdru (~Adium@76.77.182.168) |
| 18:03.59 | docid | lol, jackasses... http://i.imgur.com/RgHXv.jpg |
| 18:04.42 | carrar | ahah 15 years ago |
| 18:04.58 | docid | nice, new holes in the fukishima reactors, fuel leaking into ocean... no sushi for a while peoplez... http://blog.alexanderhiggins.com/2011/05/13/tepco-holes-fukushima-nuclear-reactor-1-3-discovered-radiation-leaking-ocean-22704/ |
| 18:05.04 | docid | ohh geeze |
| 18:05.05 | docid | sorry |
| 18:05.13 | docid | whrong channel, apologies |
| 18:08.16 | *** join/#asterisk ack_syn (~Jedi@unaffiliated/ackz0r) |
| 18:08.39 | Qwell | docid: That letter is not passive aggressive enough. |
| 18:09.12 | docid | haha, lol, good point... but it was ment for another network/channel |
| 18:09.56 | docid | ahh, yer a good source of info, gunna repost a question if ya got a min to think about it.... |
| 18:10.02 | docid | <PROTECTED> |
| 18:10.22 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-82.dsl.stlsmo.sbcglobal.net) |
| 18:10.27 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-163-82.dsl.stlsmo.sbcglobal.net) |
| 18:10.35 | docid | but audio was immediate from stuff coming in on the analog card |
| 18:16.58 | ack_syn | hey do I need to install oss modules to make a auto dial out call ?? |
| 18:17.59 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
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| 18:36.50 | fish-bulb | ack_syn: Nope. You do mean like, with a call file right? |
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| 18:40.29 | ack_syn | fish-bulb: yes I made it |
| 18:40.30 | ack_syn | thank you |
| 18:40.43 | fish-bulb | np |
| 18:40.45 | ack_syn | I just moved a formated file to my spool/asterisk/outgoing |
| 18:41.36 | fish-bulb | the oss module is for playing audio through an OSS device, like your sound card |
| 18:41.44 | fish-bulb | groovy |
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| 18:57.52 | ack_syn | got it, thank u fish-bulb |
| 18:58.26 | fish-bulb | no problem |
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| 19:08.03 | m4xx | leifmadsen, i've found a few mistakes typeos if you're interested |
| 19:08.17 | m4xx | mistakes/typeos |
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| 19:41.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 20:06.39 | pdtpatrick_ | Question -- in Asterisk, tonezone, there's bunch of numbers, im guessing those are all various sounds? how can one tell what a number does or how it relates to a sound |
| 20:09.04 | Qwell | They are frequency pairs. |
| 20:09.59 | Qwell | 1000+1200/2000,0/4000 = 1000Hz and 1200Hz played for 2000 milliseconds, then silence for 4000 milliseconds. |
| 20:17.09 | russellb | syntax should be discussed somewhere ... tonezone.conf.sample maybe |
| 20:23.02 | pdtpatrick_ | oh i c |
| 20:23.04 | pdtpatrick_ | thanks again |
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| 21:00.01 | Syrex | Anyone here usin Polycom? |
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| 21:10.07 | pabelanger | yes |
| 21:10.10 | pabelanger | and |
| 21:10.11 | pabelanger | ~ask |
| 21:10.11 | 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. |
| 21:15.19 | Syrex | Having a problem with SIP 3.3.1... BLF directed call pickup doesn't work as the phone doesn't append extension number after pickup code... Anyone here using 3.3.1? |
| 21:20.00 | *** part/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 21:23.40 | atan | Does Plantronics offer a headset with two buttons, one for a bluetooth cellphone and the other to activate a lifter for a desk phone? |
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