00:10.05 | ArcticFox | Have 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-Fender | There are alrady samples about how this all works |
00:10.44 | [TK]D-Fender | Actually follow them |
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00:12.22 | ArcticFox | ok, I'll try to get it. One more issue with TCP transport - second tcp transport is unable to be created |
00:13.57 | ArcticFox | [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) |
00:13.57 | ArcticFox | [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-Fender | You are clearly making duplicates you shouldn't |
00:15.25 | ArcticFox | http://pastebin.com/HWf94LQ3 |
00:16.34 | ArcticFox | here clearly no dupes |
00:17.04 | ArcticFox | I've tryed a lot of UDP transports - it's ok, but with TCP - NOK |
00:17.14 | [TK]D-Fender | You are assuming that you can do MULTIPLE per type |
00:17.25 | [TK]D-Fender | And protocol |
00:18.45 | [TK]D-Fender | You can't |
00:18.53 | [TK]D-Fender | You get a single bind. |
00:18.56 | [TK]D-Fender | That is all |
00:19.18 | [TK]D-Fender | As far as I've seen |
00:24.47 | ArcticFox | not clear for me, sry, but found same issue in PJPROJECT JIRA |
00:26.48 | ArcticFox | any posibilities for dynamic routing in asterisk? w/o external tools |
00:39.32 | [TK]D-Fender | bind=ex.ter.nal.ip |
00:39.41 | [TK]D-Fender | Is this an actual IP address? |
00:39.50 | [TK]D-Fender | on the line? |
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01:28.04 | Tagor | Is there something like chan_dongle for Qualcomm modems? |
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01:54.10 | mawkee | Is there anything similar to "sip show channelstats" on pjsip? Just want realtime statistics about jitter, latency and packet loss |
01:54.43 | Mawkee | Can also be a Manager command |
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05:34.17 | pcAngel | Hi 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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06:43.45 | Ellenor | fk |
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11:43.42 | ntt | Hi, 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 |
11:43.43 | ntt | number?? (actually I'm using CID and DID = ANY). |
11:44.50 | WIMPy | Your description is extremely vague. |
11:45.07 | WIMPy | What does interrupoted mean? |
11:45.14 | WIMPy | And what's that about CID and DID? |
11:45.26 | WIMPy | And what is that "alias"? |
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11:47.48 | ntt | WIMPy: 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 |
11:48.26 | ntt | Now that my provider created some kind of alias, when I try to call this alias call is rejected |
11:53.55 | Gugge | look at your debug log while you call the number, and see why asterisk rejects it |
11:58.54 | WIMPy | Is it even reaching Asterisk? |
12:09.29 | ntt | WIMPy: 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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12:13.12 | WIMPy | Don't they use redirects? |
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13:28.36 | mawkee | Is 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 |
13:28.40 | mawkee | Any help is appreciated :-) |
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13:41.23 | file | there is not |
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13:58.14 | mawkee | file, thank you |
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15:32.35 | onebree | Can 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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15:38.13 | onebree | We 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? |
15:40.45 | file | in 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... |
15:42.03 | onebree | file: We are using Asterisk 13.6.0 and chan_sip. |
15:42.59 | file | then the various dtls options are available in the peer |
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15:45.18 | onebree | file: What do you mean that the options are available in the peer? Do you mean I can specify them in sip.conf [some-user] ? |
15:45.29 | file | yes |
15:45.42 | file | but I don't know why you'd have one per user in the first place |
15:47.23 | onebree | file: One certificate per user? I am simply following the tutorial I linked before. But thank you for the help |
15:47.48 | onebree | That means all users, with access to the webrtc conference, would have the same dtls config? |
15:47.52 | file | the only part applicable in WebRTC on that tutorial is certificate creation |
15:48.01 | file | yes |
15:49.03 | onebree | Thank you. Does that mean I can skip the client certificate section? (-m client command) |
15:49.32 | file | your Asterisk still needs a client certificate |
15:49.45 | onebree | Okay, but I only need to generate one, right? |
15:49.54 | file | the CA process should create one |
15:50.02 | file | so, you shouldn't need to create any additional client certificates |
15:52.03 | onebree | What is "CA process" ? |
15:52.37 | file | certificate authority creation process |
15:54.20 | onebree | Which LOC would that be in the tutorial? The first or second time you execute ./ast_tls_cert |
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15:55.20 | file | the first part |
15:55.35 | onebree | Okay, thank you for all the help, file! |
15:55.49 | file | and make sure you follow the WebRTC guide... |
15:56.43 | onebree | Yes, 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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16:11.36 | eagles0513875 | heyo :) |
16:16.04 | lvlinux | hiyo! |
16:16.11 | eagles0513875 | hey lvlinux :) |
16:16.22 | eagles0513875 | <PROTECTED> |
16:16.49 | lvlinux | go for it! have you read the Definitive Guide book yet? |
16:17.32 | eagles0513875 | no |
16:17.41 | eagles0513875 | pplus the link on the website takes you to an advertising site |
16:18.01 | onebree | Yes, it is a great book. The 3rd edition (v 1.8) is free online, but I recommend edition 4 (v 11). |
16:18.10 | eagles0513875 | im just starting to read asterisk the future of telephony the 2nd edition |
16:18.18 | eagles0513875 | onebree: how much is the book |
16:19.15 | lvlinux | you 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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16:19.35 | lvlinux | But the 3rd edition has all the info you need to get started. |
16:19.39 | lvlinux | ~book |
16:19.39 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
16:19.42 | onebree | eagles0513875: You may want a used copy, if listed in good condition: http://www.amazon.com/gp/product/1449332420 |
16:19.55 | eagles0513875 | right now im broke as can be lol |
16:20.17 | lvlinux | then just read the free one 3rd edition---it's very good. |
16:20.19 | eagles0513875 | it will be a long term project here at the office |
16:20.29 | onebree | Well, 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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16:20.44 | eagles0513875 | ya now i will see one step at a time |
16:20.53 | eagles0513875 | im busy beyond belief im tying up alot of lose ends |
16:21.03 | eagles0513875 | that previous IT staff did not take care of |
16:21.06 | lvlinux | tell me about it lol... |
16:21.21 | eagles0513875 | lvlinux: and im preparing stock to have on hand over the holidays in case some things go wrong |
16:21.28 | eagles0513875 | i work for a 24x7x365 taxi company |
16:21.31 | eagles0513875 | as IT admin |
16:21.34 | lvlinux | interesting |
16:21.37 | eagles0513875 | so im slowly taking things on my shoulders |
16:21.50 | eagles0513875 | we are the most advanced company on the island in regards to taxi transportation |
16:21.57 | lvlinux | island? |
16:22.08 | eagles0513875 | island of malta |
16:22.13 | lvlinux | ah cool! |
16:22.21 | lvlinux | weather is nice there i bet |
16:22.37 | eagles0513875 | cold in winter and wet |
16:22.40 | eagles0513875 | hot and dry in summer |
16:22.44 | eagles0513875 | i prefer the summer heat |
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16:22.56 | eagles0513875 | at work right now we are using 3cx :X |
16:23.02 | lvlinux | ewww |
16:23.06 | eagles0513875 | and we have some plans that would fit perfectly for asterisk |
16:23.12 | eagles0513875 | and would probably super easy to implement |
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16:23.29 | eagles0513875 | what 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 |
16:23.35 | eagles0513875 | like we have various models of SNOM voip phones |
16:23.50 | lvlinux | SNOM works great with Asterisk |
16:24.01 | eagles0513875 | what about patton gateways |
16:24.15 | eagles0513875 | and what about gsm gateways |
16:24.23 | eagles0513875 | are there pitfalls with those devices i should be aware of? |
16:24.32 | lvlinux | Patton? Hmmm, not familiar with them. But Asterisk does work with many GSM gateways. |
16:24.49 | strata | your taxi company runs their own gsm network? that's cool |
16:24.54 | A1F4 | is it possible to make call on mobile network using sip ? |
16:25.00 | eagles0513875 | lvlinux: http://www.patton.com/products/product_detail.asp?id=385 |
16:25.01 | lvlinux | Pretty much any hardware that supports SIP will work |
16:25.04 | eagles0513875 | strata: no we dont |
16:25.05 | strata | A1F4: not without hitting a gateway |
16:25.21 | eagles0513875 | we have things set when calling a mobile number a sim on the gsm gateway is used |
16:25.31 | eagles0513875 | calls are alot cheaper here from mobile to mobile instead of landline to mobile |
16:25.41 | lvlinux | that page says they have full SIP support so should be piece of cake. |
16:25.49 | eagles0513875 | good to know |
16:25.49 | A1F4 | what is gateway ? is is possible to create your own ? |
16:26.07 | eagles0513875 | need to get a lab setup going here at the office to test a number of things out |
16:26.12 | lvlinux | A1F4: yes it is possible, but probably not something you'd want to do. |
16:26.14 | eagles0513875 | fyi lvlinux pattons are linux based |
16:26.30 | strata | eagles0513875: you can set up asterisk to route calls like that |
16:26.48 | eagles0513875 | strata: i know but with 3cx currently in place we have a super complex setup |
16:26.53 | eagles0513875 | does asterisk have its own softphone |
16:26.57 | lvlinux | nope |
16:26.58 | eagles0513875 | in otherwords pc based phone interface? |
16:27.01 | strata | eagles0513875: your dialplan could consist of prefixes known to be mobile numbers and any call to them gets routed to the gsm gateway by asterisk. |
16:27.08 | eagles0513875 | that might be the biggest issue for us there |
16:27.10 | WIMPy | Building your own gateway give you a chance to get it going a little smoother. |
16:27.11 | lvlinux | well i take that back, sortof (via webrtc) |
16:27.21 | eagles0513875 | all our call center agents dont have physical phones |
16:27.25 | eagles0513875 | they use the 3cx softphone at the moment |
16:27.36 | lvlinux | you can use any decent softphone that supports SIP or IAX |
16:27.58 | lvlinux | Bria, Zoiper, etc. etc. |
16:28.22 | lvlinux | or web based like SIPml5 works too |
16:28.34 | eagles0513875 | interesting |
16:29.20 | eagles0513875 | again i need to get some more hardware to setup a test environment |
16:29.25 | eagles0513875 | so i can setup break repeat |
16:30.20 | eagles0513875 | lvlinux: would it be easy to write ones own softphone? |
16:30.27 | eagles0513875 | custom softphone to interface with asterisk |
16:31.15 | lvlinux | easy? 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. |
16:31.37 | eagles0513875 | lvlinux: what coding know how would one need |
16:31.51 | eagles0513875 | to be able to work with asterisk |
16:32.28 | lvlinux | good knowledge of SIP or IAX, RTP, and ability to implement it in your language of choice. |
16:32.42 | lvlinux | Asterisk itself doesn't need any coding skills, just if you want to make a soft phone. |
16:32.49 | eagles0513875 | ohh very nice :) would pose a great challenge so im happy :) |
16:33.01 | eagles0513875 | and excited for sure :) |
16:33.47 | lvlinux | yeah, 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. |
16:34.37 | eagles0513875 | ya i do as i have administered a number of 3cx installations |
16:35.38 | lvlinux | Then 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.) |
16:35.57 | onebree | This 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 |
16:36.12 | onebree | The packages are: build-essential libncurses5-dev libxml2-dev libsqlite3-dev libssl-dev libsrtp0-dev uuid-dev |
16:36.22 | eagles0513875 | lvlinux: i love command line |
16:36.40 | eagles0513875 | onebree: FACEPALLM :P why 6 |
16:37.23 | eagles0513875 | centos at times brings back some scary memories of 5 and around 5 years ago when i used to work part time at a DC |
16:37.31 | eagles0513875 | i had to deploy a vm template with ISPCP control panel |
16:37.43 | eagles0513875 | that uses perl and the minimum perl version 5 had was too old for what the panel required. |
16:37.53 | eagles0513875 | i learned a neat hack and work around though cuz of it |
16:38.00 | onebree | eagles: Why 6? The server has been in use before I was employed, and I am just using what they gave me. |
16:38.08 | eagles0513875 | fair enough |
16:38.17 | eagles0513875 | lvlinux: how well does asterisk play with SEL |
16:38.29 | onebree | Do those packages not exist for centos 6? |
16:38.47 | eagles0513875 | onebree: i dont know but im green to asterisk so you can kinda ignore me |
16:38.50 | lvlinux | onebree: they probably do somewhere but the names are different |
16:39.06 | eagles0513875 | lvlinux: that and probably would need to be compiled from source as they would probably be greatly outdated |
16:39.22 | lvlinux | onebree: look on google for libncurses5-dev centos rpm and you'll see the appropriate name |
16:39.34 | onebree | How do I know if the package already exists (compiled from source)? |
16:39.42 | lvlinux | look in /usr/src |
16:40.06 | lvlinux | But the names are different in a lot of distros. |
16:40.07 | onebree | yeah... nothing but asterisk is in there. And dahdi linux |
16:40.25 | lvlinux | i think rpm based distros often use -devel instead of -dev and such |
16:40.38 | onebree | Is it better to compile from source, or yum install through a repo? |
16:40.47 | eagles0513875 | does asterisk play nicely in a virtualized environment |
16:41.10 | lvlinux | eagles0513875: SEL pretty sure it works fine but havne't ever tried it myself. |
16:41.28 | lvlinux | eagles0513875: oh yes does fine in VMs, i use it that way sometimes |
16:41.49 | eagles0513875 | ya im looking at getting us off hyper-v |
16:42.02 | lvlinux | onebree: you mean asterisk itself or those other packages? Asterisk is generally better from source since any packages are usually out of date. |
16:42.03 | eagles0513875 | but that will be a super long term project. |
16:42.06 | eagles0513875 | anyway |
16:42.14 | eagles0513875 | im out been up for over 12 hours |
16:42.31 | lvlinux | k cya. Don't forget to start reading the book tomorrow! :-D |
16:42.39 | eagles0513875 | wwont be before thursday |
16:42.42 | onebree | I am talking about other packages in general, like the ones I listed before. |
16:42.44 | eagles0513875 | next two days are goign to be insane |
16:42.57 | lvlinux | :-) |
16:43.16 | lvlinux | onebree: i would go with the packages if possible. |
16:43.41 | lvlinux | if you have a problem, then you can install from source, but why mess with it if it isn't broken> :-) |
16:44.14 | onebree | This is interesting: build essentials is names MUCH diff in centos: http://www.asim.pk/2010/05/28/build-essentials-in-centos/ |
16:44.36 | eagles0513875 | onebree: another example |
16:44.42 | eagles0513875 | apache on ubuntu httpd on centos |
16:44.53 | eagles0513875 | in terms of package name |
16:45.42 | eagles0513875 | centos 8 is going to have some major changes |
16:45.48 | eagles0513875 | they have deprecated yum in fedora 23 for dnf |
16:46.33 | eagles0513875 | im out for now |
16:46.35 | onebree | no more yum?? Do you have a link to the changes for centos8? IDK if it is my machine, but centos 7 is extremely buggy |
16:46.52 | eagles0513875 | onebree: take a look at fedora 23 |
16:47.06 | eagles0513875 | i found out those are what RHEL base their changes off of and those filter down to centos |
16:47.15 | eagles0513875 | anyway ill ttyl im beat |
16:54.18 | igcewieling | That's why we won't be upgrading past CentOS 6.x anytime soon. |
16:54.30 | igcewieling | we = my employer |
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17:38.35 | IPalmer | hello 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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17:58.50 | igcewieling | how, exactly, are you doing this? |
17:59.08 | igcewieling | oh, ans what version of Asterisk and what SIP stack? |
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18:56.19 | onebree | I 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? |
18:58.38 | onebree | Looking at the script, I do not want lines with PACKAGE_DEBIAN to execute, but rather PACKAGE_RH |
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19:21.45 | onebree | I 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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19:25.18 | igcewieling | Maybe nobody here has used ./install_prereq? |
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19:29.07 | onebree | I guess... - |
19:30.02 | newtonr | onebree, it attempts to detect what you OS you are on and install the most appropriate packages |
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19:31.30 | onebree | newtonr: That is what I gathered. How can I override what it assumes I have? |
19:32.38 | newtonr | It 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 |
19:33.23 | newtonr | I 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. |
19:33.55 | newtonr | Feel 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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19:37.19 | onebree | I would not know where to even start for fixing the sript |
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19:39.08 | newtonr | In that case you could still file a bug and provide the information a developer would need to fix it. |
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19:47.01 | spicyramen | Is there a module in Asterisk for protection against sipvicious? |
19:47.20 | onebree | newtonr: Ah-ha! I found the issue. It checks `uname -s`, which is set to Linux for whatever reason. |
19:51.09 | [TK]D-Fender | spicyramen: No, firewalling is your job |
19:51.27 | [TK]D-Fender | spicyramen: * spits out logs & AMI. You can use that to feed other general things like fail2ban, etc |
19:59.42 | spicyramen | ok, thanks, we are setting up Kamailio in front of our Asterisk using realtime integration |
19:59.52 | spicyramen | but wondering if they have such module |
19:59.58 | spicyramen | thanks |
20:00.57 | onebree | newtonr: 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-Fender | spicyramen: * does do "modules" |
20:01.34 | [TK]D-Fender | spicyramen: This, like any other bit of interaction with outside resources is your job |
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20:13.10 | acidfoo | moo! |
20:13.23 | acidfoo | how can I make sure a channel is transcoding or not ? |
20:13.38 | acidfoo | via the asterisk CLI i mean |
20:16.07 | [TK]D-Fender | Look at the call |
20:16.17 | [TK]D-Fender | "sip show channel THECHANNEL" |
20:16.21 | [TK]D-Fender | and the one it is bridged to |
20:16.36 | [TK]D-Fender | get them from "core show channels concise" |
20:21.02 | acidfoo | good I didn't know about "channels concise" |
20:25.53 | acidfoo | (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 |
20:25.55 | acidfoo | freak |
20:28.17 | igcewieling | have you considered accepting the call and routing it to and endless IVR or message? Seems to stop many attacks. |
20:32.48 | igcewieling | http://pastebin.com/BGaJvZ7X has iptables rules to block many sip bots |
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20:39.19 | onebree | Is menuselect required for asterisk? The WebRTC/SipML5 says to `make menuselect`, but I am wondering if it is really needed |
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21:01.57 | WangDang | I'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? |
21:02.19 | igcewieling | When the kernel is upgraded you MUST rebuild DAHDI. |
21:03.05 | WangDang | igcewieling: it's the same kernel, just a security fix |
21:03.32 | igcewieling | WangDang: try copying over the modules from the old kernel to the new kernel.....or. rebuild DAHDI. |
21:03.54 | igcewieling | I find "rebuild DAHDI" is so trivial to do, it doesn't make sense to do anything else. |
21:05.01 | igcewieling | something 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" |
21:05.38 | WangDang | This 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) |
21:05.52 | igcewieling | WangDang: Then I cannot help you. Good luck though. |
21:06.03 | igcewieling | you could boot into your old kernel until you figure thijgs out. |
21:06.54 | WangDang | Like I said, it wasn't kernel upgrade, just a security fix, so there is no old kernel |
21:07.36 | [TK]D-Fender | upgrade = no longer the same kernel |
21:07.37 | igcewieling | WangDang: try this: find / -name "wctdm.so" -print to find you r module. |
21:07.45 | igcewieling | that will tell you where the file it. |
21:08.17 | igcewieling | [TK]D-Fender: first he needs to accept there is a different kernel. I'm still working on that. |
21:08.29 | igcewieling | though only for the next 5 mins. |
21:09.58 | igcewieling | sorry, not wctdm.so, but wctdm.ko |
21:10.32 | igcewieling | On my centos it is under /lib/modules/<kernelversion> |
21:11.14 | WangDang | find / -name wctdm.ko turns up nothing. The module appears to have disappeared |
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21:12.45 | WangDang | I 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 |
21:13.37 | WangDang | I tried wctdm.so, wctdm.ko, "wctdm.so" and "wctdm.ko". None returned any results |
21:13.45 | WangDang | in the find command that is |
21:14.28 | acidfoo | what debug knob I need to turn to see the dialplan execution in the asterisk log ? |
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21:18.58 | ophuk | I'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 |
21:19.12 | onebree | I 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-Fender | No, there is no "calls per second" thing with Asterisk |
21:24.51 | [TK]D-Fender | Everything it does is triggered by you one at a time. |
21:25.02 | [TK]D-Fender | like 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-Fender | That's what things like SIPP are for |
21:25.25 | ophuk | Gotcha, thanks |
21:25.38 | ophuk | We were using SIPp but it kept crashing |
21:26.46 | [TK]D-Fender | packs up to head home |
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21:31.22 | WangDang | igcewieling: 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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22:05.04 | mjordan | ophuk: 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 |
22:05.13 | mjordan | That way it can keep track of how many calls I have, how long they're up, etc. |
22:17.05 | WangDang | igcewieling: 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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23:02.18 | Tagor | does anyone know how to replace new lines in a var? |
23:05.45 | igcewieling | Tagor: core show function FILTER |
23:06.52 | Tagor | igcewieling: I actually want to replace them, not filter |
23:07.49 | igcewieling | Tagor: try "core show functions" |
23:09.45 | Tagor | igcewieling: 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 |
23:10.31 | igcewieling | I don't recall you telling us what you tried. Try using \n like with FILTER |
23:19.30 | Tagor | igcewieling: I already tried that but \n just escapes the letter N |
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