IRC log for #asterisk on 20151221

00:10.05ArcticFoxHave no such experience, but whole config file is hard to navigate. Splitting is also not good. Wizard is easy because of type/params, is any same way?
00:10.40[TK]D-FenderThere are alrady samples about how this all works
00:10.44[TK]D-FenderActually follow them
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00:12.22ArcticFoxok, I'll try to get it. One more issue with TCP transport - second tcp transport is unable to be created
00:13.57ArcticFox[Dec 21 03:12:49] ERROR[6899] res_pjsip/config_transport.c: Transport 'ipv4-tcp-external' could not be started: Object with the same type exists (PJSIP_ETYPEEXISTS)
00:13.57ArcticFox[Dec 21 03:12:49] ERROR[6899] res_sorcery_config.c: Could not create an object of type 'transport' with id 'ipv4-tcp-external' from configuration file 'pjsip.conf'
00:14.53[TK]D-FenderYou are clearly making duplicates you shouldn't
00:15.25ArcticFoxhttp://pastebin.com/HWf94LQ3
00:16.34ArcticFoxhere clearly no dupes
00:17.04ArcticFoxI've tryed a lot of UDP transports - it's ok, but with TCP - NOK
00:17.14[TK]D-FenderYou are assuming that you can do MULTIPLE per type
00:17.25[TK]D-FenderAnd protocol
00:18.45[TK]D-FenderYou can't
00:18.53[TK]D-FenderYou get a single bind.
00:18.56[TK]D-FenderThat is all
00:19.18[TK]D-FenderAs far as I've seen
00:24.47ArcticFoxnot clear for me, sry, but found same issue in PJPROJECT JIRA
00:26.48ArcticFoxany posibilities for dynamic routing in asterisk? w/o external tools
00:39.32[TK]D-Fenderbind=ex.ter.nal.ip
00:39.41[TK]D-FenderIs this an actual IP address?
00:39.50[TK]D-Fenderon the line?
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01:28.04TagorIs there something like chan_dongle for Qualcomm modems?
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01:54.10mawkeeIs there anything similar to "sip show channelstats" on pjsip? Just want realtime statistics about jitter, latency and packet loss
01:54.43MawkeeCan also be a Manager command
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05:34.17pcAngelHi guys.  If I want to submit a patch for a problem in Asterisk, and there's not already an issue for it in Jira, do I have to submit an issue in Jira, or can I send it straight to Gerrit?
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11:43.42nttHi, I'm trying to configure asterisk with voip credentials. Actually I have a "temporary" number with username and password from my ISP and today I received a comunication that tells me that number portability is ok and my ISP created an "alias" binded to my temporary number. Now, when I call the temporary number all works well, but when I try to call the alias call is interrupted. This could be related to CID and DID in the inbound route of the temporary
11:43.43nttnumber?? (actually I'm using CID and DID = ANY).
11:44.50WIMPyYour description is extremely vague.
11:45.07WIMPyWhat does interrupoted mean?
11:45.14WIMPyAnd what's that about CID and DID?
11:45.26WIMPyAnd what is that "alias"?
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11:47.48nttWIMPy: I'm sorry... i'm not an expert. Please be patient. Actually I configured a trunk with the temporary number and an inbound route and this works well
11:48.26nttNow that my provider created some kind of alias, when I try to call this alias call is rejected
11:53.55Guggelook at your debug log while you call the number, and see why asterisk rejects it
11:58.54WIMPyIs it even reaching Asterisk?
12:09.29nttWIMPy: thank you for you time. Problem solved: all works well, the problem is related to my mobile operator and number portability. Not all operators have already updated the configuration for the number (I'm sorry if I use wrong terms.... )
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12:13.12WIMPyDon't they use redirects?
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13:28.36mawkeeIs there anything similar to "sip show channelstats" on pjsip? Just want realtime statistics about jitter, latency and packet loss. Can also be a Manager command or event
13:28.40mawkeeAny help is appreciated :-)
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13:41.23filethere is not
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13:58.14mawkeefile, thank you
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15:32.35onebreeCan someone help me through this tutorial? I have questions about creating the client certificates: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#SecureCallingTutorial-Keys
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15:38.13onebreeWe are using WebRTC for conference calls with Asterisk. However the tutorial expects you to use desktop sip software, in which the end user must manually select the certificate. Can I, in sip.conf, choose the certificate I want per user, since the web UI won't provide the option to choose?
15:40.45filein WebRTC the browser will generate a certificate automatically, but you still have to provide a certificate for the Asterisk side and it depends on the version of Asterisk and what channel driver you are using whether it can be specified per user...
15:42.03onebreefile: We are using Asterisk 13.6.0 and chan_sip.
15:42.59filethen the various dtls options are available in the peer
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15:45.18onebreefile: What do you mean that the options are available in the peer? Do you mean I can specify them in sip.conf [some-user] ?
15:45.29fileyes
15:45.42filebut I don't know why you'd have one per user in the first place
15:47.23onebreefile: One certificate per user? I am simply following the tutorial I linked before. But thank you for the help
15:47.48onebreeThat means all users, with access to the webrtc conference, would have the same dtls config?
15:47.52filethe only part applicable in WebRTC on that tutorial is certificate creation
15:48.01fileyes
15:49.03onebreeThank you. Does that mean I can skip the client certificate section? (-m client command)
15:49.32fileyour Asterisk still needs a client certificate
15:49.45onebreeOkay, but I only need to generate one, right?
15:49.54filethe CA process should create one
15:50.02fileso, you shouldn't need to create any additional client certificates
15:52.03onebreeWhat is "CA process" ?
15:52.37filecertificate authority creation process
15:54.20onebreeWhich LOC would that be in the tutorial?  The first or second time you execute ./ast_tls_cert
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15:55.20filethe first part
15:55.35onebreeOkay, thank you for all the help, file!
15:55.49fileand make sure you follow the WebRTC guide...
15:56.43onebreeYes, I am doing that as well. The issue lies in problems with VIDEO conferencing on WebRTC... But I have talked about that enough on this channel to know that few, if any, could help with that
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16:11.36eagles0513875heyo :)
16:16.04lvlinuxhiyo!
16:16.11eagles0513875hey lvlinux  :)
16:16.22eagles0513875<PROTECTED>
16:16.49lvlinuxgo for it! have you read the Definitive Guide book yet?
16:17.32eagles0513875no
16:17.41eagles0513875pplus the link on the website takes you to an advertising site
16:18.01onebreeYes, it is a great book. The 3rd edition (v 1.8) is free online, but I recommend edition 4 (v 11).
16:18.10eagles0513875im just starting to read asterisk the future of telephony the 2nd edition
16:18.18eagles0513875onebree: how much is the book
16:19.15lvlinuxyou can find the 4th edition online in various places. Google helps you with that. But I highly recommend purchasing it as well.
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16:19.35lvlinuxBut the 3rd edition has all the info you need to get started.
16:19.39lvlinux~book
16:19.39infobotAsterisk: 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
16:19.42onebreeeagles0513875: You may want a used copy, if listed in good condition: http://www.amazon.com/gp/product/1449332420
16:19.55eagles0513875right now im broke as can be lol
16:20.17lvlinuxthen just read the free one 3rd edition---it's very good.
16:20.19eagles0513875it will be a long term project here at the office
16:20.29onebreeWell, like someone else said, google can help you look for other means of getting the book. But you will likely want a physical copy if you plan to use it at work
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16:20.44eagles0513875ya now i will see one step at a time
16:20.53eagles0513875im busy beyond belief im tying up alot of lose ends
16:21.03eagles0513875that previous IT staff did not take care of
16:21.06lvlinuxtell me about it lol...
16:21.21eagles0513875lvlinux: and im preparing stock to have on hand over the holidays in case some things go wrong
16:21.28eagles0513875i work for a 24x7x365 taxi company
16:21.31eagles0513875as IT admin
16:21.34lvlinuxinteresting
16:21.37eagles0513875so im slowly taking things on my shoulders
16:21.50eagles0513875we are the most advanced company on the island in regards to taxi transportation
16:21.57lvlinuxisland?
16:22.08eagles0513875island of malta
16:22.13lvlinuxah cool!
16:22.21lvlinuxweather is nice there i bet
16:22.37eagles0513875cold in winter and wet
16:22.40eagles0513875hot and dry in summer
16:22.44eagles0513875i prefer the summer heat
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16:22.56eagles0513875at work right now we are using 3cx :X
16:23.02lvlinuxewww
16:23.06eagles0513875and we have some plans that would fit perfectly for asterisk
16:23.12eagles0513875and would probably super easy to implement
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16:23.29eagles0513875what worries me a bit is the hardware we already have in place would we need to replace it or would it work with asterisk as well
16:23.35eagles0513875like we have various models of SNOM voip phones
16:23.50lvlinuxSNOM works great with Asterisk
16:24.01eagles0513875what about patton gateways
16:24.15eagles0513875and what about gsm gateways
16:24.23eagles0513875are there pitfalls with those devices i should be aware of?
16:24.32lvlinuxPatton? Hmmm, not familiar with them. But Asterisk does work with many GSM gateways.
16:24.49stratayour taxi company runs their own gsm network? that's cool
16:24.54A1F4is it possible to make call on mobile network using sip ?
16:25.00eagles0513875lvlinux: http://www.patton.com/products/product_detail.asp?id=385
16:25.01lvlinuxPretty much any hardware that supports SIP will work
16:25.04eagles0513875strata: no we dont
16:25.05strataA1F4: not without hitting a gateway
16:25.21eagles0513875we have things set when calling a mobile number a sim on the gsm gateway is used
16:25.31eagles0513875calls are alot cheaper here from mobile to mobile instead of landline to mobile
16:25.41lvlinuxthat page says they have full SIP support so should be piece of cake.
16:25.49eagles0513875good to know
16:25.49A1F4what is gateway ? is is possible to create your own ?
16:26.07eagles0513875need to get a lab setup going here at the office to test a number of things out
16:26.12lvlinuxA1F4: yes it is possible, but probably not something you'd want to do.
16:26.14eagles0513875fyi lvlinux pattons are linux based
16:26.30strataeagles0513875: you can set up asterisk to route calls like that
16:26.48eagles0513875strata: i know but with 3cx currently in place we have a super complex setup
16:26.53eagles0513875does asterisk have its own softphone
16:26.57lvlinuxnope
16:26.58eagles0513875in otherwords pc based phone interface?
16:27.01strataeagles0513875: your dialplan could consist of prefixes known to be mobile numbers and any call to them gets routed to the gsm gateway by asterisk.
16:27.08eagles0513875that might be the biggest issue for us there
16:27.10WIMPyBuilding your own gateway give you a chance to get it going a little smoother.
16:27.11lvlinuxwell i take that back, sortof (via webrtc)
16:27.21eagles0513875all our call center agents dont have physical phones
16:27.25eagles0513875they use the 3cx softphone at the moment
16:27.36lvlinuxyou can use any decent softphone that supports SIP or IAX
16:27.58lvlinuxBria, Zoiper, etc. etc.
16:28.22lvlinuxor web based like SIPml5 works too
16:28.34eagles0513875interesting
16:29.20eagles0513875again i need to get some more hardware to setup a test environment
16:29.25eagles0513875so i can setup break repeat
16:30.20eagles0513875lvlinux: would it be easy to write ones own softphone?
16:30.27eagles0513875custom softphone to interface with asterisk
16:31.15lvlinuxeasy? depends on your VoIP and coding skills and knowlege. But I would highly not recommend it as there are so many good ones out there.
16:31.37eagles0513875lvlinux: what coding know how would one need
16:31.51eagles0513875to be able to work with asterisk
16:32.28lvlinuxgood knowledge of SIP or IAX, RTP, and ability to implement it in your language of choice.
16:32.42lvlinuxAsterisk itself doesn't need any coding skills, just if you want to make a soft phone.
16:32.49eagles0513875ohh very nice :) would pose a great challenge so im happy :)
16:33.01eagles0513875and excited for sure :)
16:33.47lvlinuxyeah, don't worry about that. Start reading the book and it will clear up lots of stuff for you. It's not really difficult if you already understand VoIP and telephony in general.
16:34.37eagles0513875ya i do as i have administered a number of 3cx installations
16:35.38lvlinuxThen you shouldn't have any problem as long as you aren't scared of commandline style stuff. (and if you are, get over it lol.)
16:35.57onebreeThis may be a silly question. I am installing packages asked for in a tutorial. However, yum search returns nothing related. I think I am using Centos 6? https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
16:36.12onebreeThe packages are: build-essential libncurses5-dev libxml2-dev libsqlite3-dev libssl-dev libsrtp0-dev uuid-dev
16:36.22eagles0513875lvlinux: i love command line
16:36.40eagles0513875onebree: FACEPALLM :P why 6
16:37.23eagles0513875centos at times brings back some scary memories of 5 and around 5 years ago when i used to work part time at a DC
16:37.31eagles0513875i had to deploy a vm template with ISPCP control panel
16:37.43eagles0513875that uses perl and the minimum perl version 5 had was too old for what the panel required.
16:37.53eagles0513875i learned a neat hack and work around though cuz of it
16:38.00onebreeeagles: Why 6? The server has been in use before I was employed, and I am just using what they gave me.
16:38.08eagles0513875fair enough
16:38.17eagles0513875lvlinux: how well does asterisk play with SEL
16:38.29onebreeDo those packages not exist for centos 6?
16:38.47eagles0513875onebree: i dont know but im green to asterisk so you can kinda ignore me
16:38.50lvlinuxonebree: they probably do somewhere but the names are different
16:39.06eagles0513875lvlinux: that and probably would need to be compiled from source as they would probably be greatly outdated
16:39.22lvlinuxonebree: look on google for libncurses5-dev centos rpm and you'll see the appropriate name
16:39.34onebreeHow do I know if the package already exists (compiled from source)?
16:39.42lvlinuxlook in /usr/src
16:40.06lvlinuxBut the names are different in a lot of distros.
16:40.07onebreeyeah... nothing but asterisk is in there. And dahdi linux
16:40.25lvlinuxi think rpm based distros often use -devel instead of -dev and such
16:40.38onebreeIs it better to compile from source, or yum install through a repo?
16:40.47eagles0513875does asterisk play nicely in a virtualized environment
16:41.10lvlinuxeagles0513875: SEL pretty sure it works fine but havne't ever tried it myself.
16:41.28lvlinuxeagles0513875: oh yes does fine in VMs, i use it that way sometimes
16:41.49eagles0513875ya im looking at getting us off hyper-v
16:42.02lvlinuxonebree: you mean asterisk itself or those other packages? Asterisk is generally better from source since any packages are usually out of date.
16:42.03eagles0513875but that will be a super long term project.
16:42.06eagles0513875anyway
16:42.14eagles0513875im out been up for over 12 hours
16:42.31lvlinuxk cya. Don't forget to start reading the book tomorrow! :-D
16:42.39eagles0513875wwont be before thursday
16:42.42onebreeI am talking about other packages in general, like the ones I listed before.
16:42.44eagles0513875next two days are goign to be insane
16:42.57lvlinux:-)
16:43.16lvlinuxonebree: i would go with the packages if possible.
16:43.41lvlinuxif you have a problem, then you can install from source, but why mess with it if it isn't broken> :-)
16:44.14onebreeThis is interesting: build essentials is names MUCH diff in centos: http://www.asim.pk/2010/05/28/build-essentials-in-centos/
16:44.36eagles0513875onebree: another example
16:44.42eagles0513875apache on ubuntu httpd on centos
16:44.53eagles0513875in terms of package name
16:45.42eagles0513875centos 8 is going to have some major changes
16:45.48eagles0513875they have deprecated yum in fedora 23 for dnf
16:46.33eagles0513875im out for now
16:46.35onebreeno more yum?? Do you have a link to the changes for centos8? IDK if it is my machine, but centos 7 is extremely buggy
16:46.52eagles0513875onebree: take a look at fedora 23
16:47.06eagles0513875i found out those are what RHEL base their changes off of and those filter down to centos
16:47.15eagles0513875anyway ill ttyl im beat
16:54.18igcewielingThat's why we won't be upgrading past CentOS 6.x anytime soon.
16:54.30igcewielingwe = my employer
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17:38.35IPalmerhello all, I'm trying to send DTMF tones via ami, they work fine from the handset but when I send via AMI I get a successful response but the tone doesn't appear to be played.  Anyone have any ideas?
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17:58.50igcewielinghow, exactly, are you doing this?
17:59.08igcewielingoh, ans what version of Asterisk and what SIP stack?
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18:56.19onebreeI am following the webrtc/sipml5 tutorial. It says to execute "./install_prereq. By default, I assume it installs for debian, but I am on CentOS. Are there flags I can set to tell it to run the centos commands?
18:58.38onebreeLooking at the script, I do not want lines with PACKAGE_DEBIAN to execute, but rather PACKAGE_RH
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19:21.45onebreeI want to run the script ./install_prereq on CentOS 6. However, it is trying to run the PACKAGE_DEBIAN commands inside the file. Does this executable take any flags/args so I can only run the RH related things?
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19:25.18igcewielingMaybe nobody here has used ./install_prereq?
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19:29.07onebreeI guess... -
19:30.02newtonronebree, it attempts to detect what you OS you are on and install the most appropriate packages
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19:31.30onebreenewtonr: That is what I gathered. How can I override what it assumes I have?
19:32.38newtonrIt isn't too fancy of a script, you'll likely  have to open up the script and copy the package names you want. Then paste them into your install command
19:33.23newtonrI don't remember having any issues with install_prereq in CentOS 6 so I'm not sure off-hand why it isn't working for you specifically.
19:33.55newtonrFeel free to file a bug on the issue tracker. Of course even better if you can hunt down the issue and provide a fix for the script. :)
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19:37.19onebreeI would not know where to even start for fixing the sript
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19:39.08newtonrIn that case you could still file a bug and provide the information a developer would need to fix it.
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19:47.01spicyramenIs there a module in Asterisk for protection against sipvicious?
19:47.20onebreenewtonr: Ah-ha! I found the issue. It checks `uname -s`, which is set to Linux for whatever reason.
19:51.09[TK]D-Fenderspicyramen: No, firewalling is your job
19:51.27[TK]D-Fenderspicyramen: * spits out logs & AMI.  You can use that to feed other general things like fail2ban, etc
19:59.42spicyramenok, thanks, we are setting up Kamailio in front of our Asterisk using realtime integration
19:59.52spicyramenbut wondering if they have such module
19:59.58spicyramenthanks
20:00.57onebreenewtonr: the script works fine, but is just trying to install packages that are not found in the repos I have. Possibly not even available anymore for centos
20:01.18[TK]D-Fenderspicyramen: * does do "modules"
20:01.34[TK]D-Fenderspicyramen: This, like any other bit of interaction with outside resources is your job
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20:13.10acidfoomoo!
20:13.23acidfoohow can I make sure a channel is transcoding or not ?
20:13.38acidfoovia the asterisk CLI i mean
20:16.07[TK]D-FenderLook at the call
20:16.17[TK]D-Fender"sip show channel THECHANNEL"
20:16.21[TK]D-Fenderand the one it is bridged to
20:16.36[TK]D-Fenderget them from "core show channels concise"
20:21.02acidfoogood I didn't know about "channels concise"
20:25.53acidfoo(phoneA support ulaw/g722, phoneB support ulaw, then phoneA call phoneB I set SIP_CODEC to ulaw) phoneA will indeed use ulaw, but asterisk will do ulaw->slin->g722 then g722->slin->ulaw
20:25.55acidfoofreak
20:28.17igcewielinghave you considered accepting the call and routing it to and endless IVR or message?   Seems to stop many attacks.
20:32.48igcewielinghttp://pastebin.com/BGaJvZ7X has iptables rules to block many sip bots
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20:39.19onebreeIs menuselect required for asterisk? The WebRTC/SipML5 says to `make menuselect`, but I am wondering if it is really needed
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21:01.57WangDangI've rebooted my asterisk box because of kernel security update.  After reboot, * is not answering calls.   dahdi show channels shows an empty list.  lsmod doesn't show the kernel module for my digium pots card.  modprobe wctdm says it can't find the module.  Any suggestions on what I should look at next?
21:02.19igcewielingWhen the kernel is upgraded you MUST rebuild DAHDI.
21:03.05WangDangigcewieling: it's the same kernel, just a security fix
21:03.32igcewielingWangDang: try copying over the modules from the old kernel to the new kernel.....or. rebuild DAHDI.
21:03.54igcewielingI find "rebuild DAHDI" is so trivial to do, it doesn't make sense to do anything else.
21:05.01igcewielingsomething like make clean; make install; make config; service dahdi restart; asterisk -rx "module unload chan_dahdi.so"; sleep 1; asterisk -rx "module load chan_dahdi.so"
21:05.38WangDangThis is a debian system installed from debian packages.  I never built dahdi in the first place, that I remember (I set this box up a long time ago)
21:05.52igcewielingWangDang: Then I cannot help you.  Good luck though.
21:06.03igcewielingyou could boot into your old kernel until you figure thijgs out.
21:06.54WangDangLike I said, it wasn't kernel upgrade, just a security fix, so there is no old kernel
21:07.36[TK]D-Fenderupgrade = no longer the same kernel
21:07.37igcewielingWangDang: try this:  find / -name "wctdm.so" -print  to find you r module.
21:07.45igcewielingthat will tell you where the file it.
21:08.17igcewieling[TK]D-Fender: first he needs to accept there is a different kernel.  I'm still working on that.
21:08.29igcewielingthough only for the next 5 mins.
21:09.58igcewielingsorry, not wctdm.so, but wctdm.ko
21:10.32igcewielingOn my centos it is under /lib/modules/<kernelversion>
21:11.14WangDangfind / -name wctdm.ko turns up nothing.  The module appears to have disappeared
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21:12.45WangDangI understand that I don't have the same kernel as before the security fix, but upgrading a debian kernel of the same version overwrites the previous one.  So there is no old kernel
21:13.37WangDangI tried wctdm.so, wctdm.ko, "wctdm.so" and "wctdm.ko".  None returned any results
21:13.45WangDangin the find command that is
21:14.28acidfoowhat debug knob I need to turn to see the dialplan execution in the asterisk log ?
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21:18.58ophukI'm trying to use two PBX servers to generate SIP traffic. Is there a way to generate a specific number of calls per second and monitor the calls per second? The SIP traffic is working fine. I"m just looking for a way to generate say 200 calls per second and guarantee I have 200 calls other than copying 200 files into /var/spool/asterisk/outgout. I've searched google but most seems to target measuring max calls per second a server can handle
21:19.12onebreeI was told last week (in a DM) that I should include DTLS options (dtlscertfile, etc) in sip.conf, while TLS options in http.conf. Is this true?
21:24.39[TK]D-FenderNo, there is no "calls per second" thing with Asterisk
21:24.51[TK]D-FenderEverything it does is triggered by you one at a time.
21:25.02[TK]D-Fenderlike AMI originate, call files, etc
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21:25.11[TK]D-Fender* is not a "testing suite" for this
21:25.16[TK]D-FenderThat's what things like SIPP are for
21:25.25ophukGotcha, thanks
21:25.38ophukWe were using SIPp but it kept crashing
21:26.46[TK]D-Fenderpacks up to head home
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21:31.22WangDangigcewieling:  I just poked around the package system.  I remember now, there is a module assisant in debian to prepare kernel modules.  I'll try running that.  I've done security updates in the past without having to rebuile modules.  But I guess that doesn't mean anything.
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22:05.04mjordanophuk: whenever I use Asterisk to simulate call traffic, I usually drive it from a small dedicated external app, and using something like ARI/AMI to originate the calls
22:05.13mjordanThat way it can keep track of how many calls I have, how long they're up, etc.
22:17.05WangDangigcewieling: I'm in business.  Thanks for making me think of rebuilding.  The debian way was "module-assistant a-i dahdi".  Strange that I've not had to do that after every security update.  Anyway, have a nice day
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23:02.18Tagordoes anyone know how to replace new lines in a var?
23:05.45igcewielingTagor: core show function FILTER
23:06.52Tagorigcewieling: I actually want to replace them, not filter
23:07.49igcewielingTagor: try "core show functions"
23:09.45Tagorigcewieling: don't you think I already read the manual before asking here? there's a REPLACE function but it doesn't say how to replace new lines
23:10.31igcewielingI don't recall you telling us what you tried.    Try using \n like with FILTER
23:19.30Tagorigcewieling: I already tried that but \n just escapes the letter N
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