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| 01:05.24 | ruben231 | hi guys anyone help im installing asterisk 1.8 on ubuntu server and i get this error --> http://pastebin.com/Jgi9n85E <---------- during make...any idea guys..? |
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| 01:11.49 | [TK]D-Fender | http://forums.asterisk.org/viewtopic.php?f=1&t=88083 |
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| 07:14.05 | liquidamber | too quiet in here... what weird crap are you doing with your asterisk servers? |
| 07:15.56 | liquidamber | i have some extensions that call wake on LAN for some workstations, another that toggles a 120VAC lamp on and off, and working on a weather script... |
| 07:16.52 | liquidamber | was thinking about making something to email or SMS a text summary of incoming calls |
| 07:18.57 | drmessano | What are you using for WOL? |
| 07:19.07 | liquidamber | wol from repos |
| 07:19.14 | liquidamber | centos |
| 07:19.22 | liquidamber | system call from dialplan |
| 07:19.27 | drmessano | Then System(wol blah) |
| 07:19.29 | drmessano | ok |
| 07:19.41 | liquidamber | bedphone! |
| 07:20.24 | liquidamber | anyone doing screen pop or click to call? |
| 07:20.59 | drmessano | I've got a list of things i've tried.. some practical, some not. Tweeting, X10, remote controlling broadcast gear, SMS |
| 07:21.16 | liquidamber | been considering x10 |
| 07:21.32 | drmessano | Are you stateside? |
| 07:21.36 | liquidamber | was going to do a thing where if i call from my cell phone it runs an expect script to open up a port on my firewall for remote access |
| 07:21.38 | liquidamber | yeah, california |
| 07:22.44 | drmessano | I use an app called heyu that handles all my X10. So I call heyu with System() from a menu I built |
| 07:23.19 | liquidamber | i just got app_swift and allison working. pretty happy with it so far |
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| 07:24.45 | drmessano | I've used that combo. Cepstral pretty much broke app_swift for us with 6.0 and now you have to pay out the ass for port licenses for multiple calls |
| 07:25.01 | drmessano | I use google TTS for applications |
| 07:25.07 | liquidamber | the cepstral stuff is easily reversible |
| 07:25.18 | liquidamber | the license check stuff is like, boolean check |
| 07:25.32 | liquidamber | you put a nor or a nop or some crap somewhere and it doesnt care any more :P |
| 07:25.55 | liquidamber | but, i havent done that with this setup |
| 07:26.06 | liquidamber | i actually run a hotline that gets multiple callers sometimes, so that is good to know |
| 07:26.14 | liquidamber | albeit a little frustrating because i wasnt aware of the limitation |
| 07:26.23 | liquidamber | i hate the google voices |
| 07:26.37 | liquidamber | i wish nuance was easier to steal |
| 07:27.42 | liquidamber | ;check this out, 1800-427-7623. it's a highway information service. one time i called it and it says "a serious error has occured, exiting." by this i determined that they were running openvxi / voiceglue |
| 07:27.46 | drmessano | Google is at least better than the robotic free ones. I ditched Cepstral when Asterisk 10 came out, and there was 6 months or so before it was updated. |
| 07:28.10 | drmessano | Interesting |
| 07:28.13 | liquidamber | yeah it wasa piece of crap last time i checked but working OK now. cepstral should probably at least attempt to get their shit together |
| 07:28.17 | liquidamber | but their support is pretty helpful |
| 07:28.39 | drmessano | I was offered a bunch of free TTS when I came up with my Asterisk 8-Ball |
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| 07:29.12 | drmessano | Never did anything with it, because Ceptral, then Google TTS pretty much worked within my coding limitations |
| 07:29.14 | liquidamber | ive heard of people using the google voice recognition with asterisk |
| 07:29.26 | liquidamber | was the recommended solution last time i checked |
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| 07:29.47 | liquidamber | kindof interesting i used to work with someone who sold dialogic cards, they were hardware voice recognition and text to speech ISA cards |
| 07:30.01 | liquidamber | super proprietary though and only ran with like 1 application |
| 07:31.16 | drmessano | I have an app that returns a random Magic 8-Ball response, and had it tweeting and return TTS for a while. I don't think the tweeting works because my access token expired |
| 07:31.21 | drmessano | Haven't added it back |
| 07:31.43 | drmessano | I still use it for decision making |
| 07:31.52 | liquidamber | haha |
| 07:32.11 | drmessano | All built in Asterisk dialplan |
| 07:32.13 | liquidamber | what handsets are you using |
| 07:32.25 | drmessano | Polycoms and Linksys SPA series |
| 07:32.39 | liquidamber | i bought 15 ip550's for 900 bucks |
| 07:32.59 | liquidamber | sold 12 of them for 1300 |
| 07:33.08 | liquidamber | kept the other 3. pretty happy with them |
| 07:33.18 | drmessano | Good handset. Have a number of them |
| 07:33.26 | liquidamber | wish polycom would offer a GUI config utility |
| 07:33.47 | drmessano | FreePBX provisioning is the way to go |
| 07:34.11 | liquidamber | do you just do a minimal centos install and build freepbx / asterisk or use a distro |
| 07:34.27 | drmessano | Ubuntu + Asterisk + FreePBX |
| 07:34.38 | liquidamber | thru repos? |
| 07:34.43 | drmessano | No, source |
| 07:35.17 | liquidamber | ive been using the digium repo and no freepbx |
| 07:35.21 | liquidamber | pain sometimes but meh |
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| 07:36.26 | drmessano | I was using AsteriskNOW and ditched it shortly after the first time a new kernel was released and there was no DAHDI waiting |
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| 07:37.09 | drmessano | I would hope whoever took over development when the lead dev left Digium got on the ball after that |
| 07:37.15 | drmessano | But that was a disaster |
| 07:37.37 | drmessano | It wasnt just that, but that was the last straw for that install |
| 07:38.17 | drmessano | I'm not really a CentOS or Red Hat fan in general anymore |
| 07:39.53 | liquidamber | kinda feel like centos takes longer to change than debian / ubuntu, which i like |
| 07:40.38 | liquidamber | got pissed at ubuntu when major things changed eery update |
| 07:40.44 | drmessano | You can't really group those together. CentOS and Debian are on relatively long release cycles. Ubuntu, by design, changes often |
| 07:41.07 | liquidamber | debian testing :P |
| 07:41.18 | liquidamber | im not sure about centos either really |
| 07:41.23 | drmessano | Ubuntu does have LTS, and I stick with it for critical installs |
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| 07:41.25 | liquidamber | i just tend to like the redhat way of doing things |
| 07:41.45 | liquidamber | most of this stuff isnt public facing so i dont really care anyway |
| 07:41.53 | drmessano | Well, Debian Testing is damn near Ubuntu. That's like complaining that CentOS + EPEL has too much cutting edge crap |
| 07:42.20 | liquidamber | i would be using redhat if it were free |
| 07:42.32 | drmessano | Debian with standard repos should have the same stability as CentOS or Ubuntu LTS |
| 07:42.45 | drmessano | I go with Ubuntu LTS for mission critical installs |
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| 07:43.40 | liquidamber | the main thing is i prefer the conventions in rhel |
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| 07:44.20 | drmessano | I used CentOS for a number of projects for the first 5 years or so that I was involved in Linux |
| 07:44.48 | liquidamber | its all the same crap anyway |
| 07:44.57 | drmessano | But I got tired of a few things, like having to rebuild a box from major release <> major release, and RPMs in general |
| 07:45.08 | liquidamber | yeah ive had yum flip out a few times |
| 07:45.18 | liquidamber | leaving the system in an unstable state |
| 07:45.24 | liquidamber | never had aptitude do that |
| 07:45.33 | drmessano | yum - The sound your box makes while it eats itself during an update |
| 07:45.40 | liquidamber | lol |
| 07:45.49 | liquidamber | it hasn't shit the bed yet on this build |
| 07:46.07 | liquidamber | oh so |
| 07:46.15 | liquidamber | for your customers what kinda box are you building this stuff on ? |
| 07:46.24 | liquidamber | ive seen people use the supermicros with some success |
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| 07:47.16 | drmessano | Depends. I have used the SuperMicro boxes a few times. I have a couple deployed on a low volume, but highly critical install and they've been great. Great value and stable |
| 07:47.37 | drmessano | But i've built on everything from routers to HP DL380/DL385s |
| 07:47.45 | drmessano | and VPS |
| 07:47.50 | liquidamber | are you ever using a PRI cards? |
| 07:48.03 | drmessano | I try not to.. I have ONE in use in one install |
| 07:48.10 | drmessano | Connecting to a channel bank |
| 07:48.13 | liquidamber | SIP then? |
| 07:48.22 | drmessano | Mostly |
| 07:48.35 | liquidamber | dedicated internet connection? |
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| 07:49.22 | drmessano | Not necessarily. QoS on the router with a shared connection, have a provider that drops fiber in and a dedicated VLAN for others |
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| 07:50.17 | liquidamber | ive had pretty good success with ASA and shape average |
| 07:51.01 | drmessano | Most of the time a new PBX install will coincide with a move from AT&T to an ITSP, which helps justify the expense. SIP is king. |
| 07:51.33 | liquidamber | what firewalls you using? |
| 07:51.59 | liquidamber | this helps too: class voip-rtp priority |
| 07:52.26 | liquidamber | i like the fact that the ASA can keep track of SIP connections also.. |
| 07:52.44 | drmessano | It varies. I've put boxes behind Linksys/Netgear consumer boxes, I have a Juniper deployed at a couple sites as well |
| 07:53.08 | drmessano | I've heard the ASA's are decent |
| 07:53.11 | liquidamber | sonicwall works pretty well too |
| 07:53.12 | drmessano | Just don't use a PIX |
| 07:53.23 | liquidamber | haha PIX are doorstops now |
| 07:53.27 | liquidamber | but many still in production use |
| 07:53.28 | drmessano | I never liked a Sonicwall with SIP |
| 07:53.34 | drmessano | Maybe its gotten better |
| 07:54.00 | drmessano | Sonicwall ALG was terrible |
| 07:54.11 | liquidamber | it has some sip inspect crap that you ahve to turn off |
| 07:54.22 | liquidamber | after that, the traffic shaping works pretty well |
| 07:55.15 | *** join/#asterisk vlad_starkov (~vlad_star@94.228.240.93) |
| 07:55.23 | liquidamber | so any suggestion on click to call / screen pop? |
| 07:55.35 | liquidamber | im no longer really working with this stuff so this is just for my home system |
| 07:55.49 | liquidamber | been playing with this UC520 |
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| 07:56.41 | drmessano | No, not really.. There's some windows apps, but I haven't kept up. |
| 07:56.43 | liquidamber | trying to find a copy of CUCM |
| 07:56.45 | drmessano | UC520 = evil |
| 07:56.48 | liquidamber | haha why |
| 07:56.55 | drmessano | I hated those fucking things |
| 07:57.02 | liquidamber | the configuration syntax makes me want to jump off a bridge |
| 07:57.10 | drmessano | Yeah |
| 07:57.19 | liquidamber | the router parts fine but the telephony stuff is such an obvious afterthought |
| 07:57.27 | drmessano | I also found them to be unstable. I couldnt stand supporting them |
| 07:57.32 | drmessano | Agreed |
| 07:57.43 | drmessano | The telephony stuff doesn't even follow a common syntax |
| 07:57.45 | liquidamber | hmm havent heard that before but it wouldnt surprise me |
| 07:57.49 | liquidamber | about the reliability |
| 07:58.06 | liquidamber | it seemed once it was obtusely configured it just sat there, as the fat piece of shit it is, and did its job |
| 07:58.15 | liquidamber | oh if you really want to hate something |
| 07:58.19 | liquidamber | avaya |
| 07:58.24 | drmessano | Router --> Shut/No Shut, Phone ---> set daytime mode/unset unnighttime mode |
| 07:58.24 | liquidamber | S8800 |
| 07:58.33 | drmessano | Cisco was STONED when they did that |
| 07:58.48 | liquidamber | for a "big iron" company some of their products are totally half baked |
| 07:58.58 | liquidamber | apparently the entire nexus platform is kindof a piece of shirt |
| 07:59.32 | drmessano | I think Mitel takes the cake, at least in my experience |
| 07:59.33 | liquidamber | but really avaya is like wtf? |
| 07:59.54 | drmessano | Mitel understands as much about VoIP as NEC |
| 08:00.00 | liquidamber | lol |
| 08:00.15 | liquidamber | NEC as a standalone PBX actually works pretty well |
| 08:00.22 | liquidamber | everyone ive met who has it is very happy with it |
| 08:00.27 | liquidamber | oh no wait though |
| 08:00.35 | liquidamber | i have the ultimate shit show |
| 08:00.36 | liquidamber | ESI |
| 08:00.41 | liquidamber | what. the. fuck. |
| 08:01.42 | drmessano | I didn't hate our NEC Electra Elite. But throwing an "IP" card in there was like gluing a Raspberry Pi to a 20 yr old fridge and calling it a "Smart Appliance" |
| 08:01.54 | *** join/#asterisk lok-jpr-india (lok-jpr-in@202.191.157.74) |
| 08:02.12 | liquidamber | i kinda like the design of the SV handsets |
| 08:02.27 | liquidamber | and users like the way paging works |
| 08:02.48 | drmessano | Our vendor told me "For $1500 I can throw a couple SIP cards in there and turn it into an IP PBX".. I told him "For $3000 I can replace it and not pay for service calls ever again" |
| 08:03.10 | drmessano | That wasnt completely true, but |
| 08:03.41 | liquidamber | lol |
| 08:03.54 | drmessano | Lipstick on a pig and all |
| 08:03.56 | liquidamber | vendors of nec, esi, panasonic, ip office are usually clueless |
| 08:04.10 | liquidamber | not saying they are bad systems, but they kinda have their heads in the sand |
| 08:04.34 | liquidamber | its something that was once a monopoly and times are rapidly changing |
| 08:04.49 | liquidamber | my old job paid 350 thousand dollars for their avaya phone system |
| 08:04.54 | drmessano | After doing years of work on the side with Asterisk, I got to build a PBX for my office.. which was great, except the server I chose |
| 08:04.54 | liquidamber | they had 300 employees |
| 08:05.00 | drmessano | Wow |
| 08:05.11 | liquidamber | and, the handsets are nice |
| 08:05.20 | liquidamber | but modular messaging is a FUCKING PIECE OF SHIT |
| 08:05.28 | liquidamber | i mean really, what the fuck |
| 08:05.36 | liquidamber | what a joke |
| 08:05.41 | liquidamber | same with the CMS |
| 08:05.41 | drmessano | Word of advice, HP DL380 series servers make for shitty Linux servers |
| 08:06.18 | liquidamber | they spent 10000 dollars on an EOL rack mount sparcstation to run some stupid call center monitoring application |
| 08:06.29 | liquidamber | for a call center of <15 people |
| 08:06.38 | drmessano | wow |
| 08:06.49 | liquidamber | the backend only ran on solaris, the frontend only ran on windows |
| 08:06.50 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:80ba:2bf4:d6bd:1c08) |
| 08:06.51 | liquidamber | thanks avaya |
| 08:07.09 | liquidamber | oh and then OneX. another gigantic piece of shit |
| 08:08.53 | *** join/#asterisk vlad_starkov (~vlad_star@94.228.240.93) |
| 08:09.20 | liquidamber | i did like the avaya media gateways though |
| 08:09.20 | drmessano | 3Com apparently adopted Asterisk.. but the old 3Com stuff |
| 08:09.26 | drmessano | Goes on the shame list |
| 08:09.27 | liquidamber | hmm |
| 08:09.37 | liquidamber | they made a pbx product? |
| 08:09.46 | drmessano | Yeah |
| 08:10.25 | drmessano | 3COM NBX |
| 08:10.41 | liquidamber | one of my customers has like, the oldest NEC PBX ever |
| 08:10.49 | drmessano | lol |
| 08:11.04 | liquidamber | i told him, you need to get patch panels and new wiring to every location on your 3 floors |
| 08:11.17 | liquidamber | for when your phone system explodes and you have to switch to IP handsets |
| 08:11.46 | drmessano | I retired an NEC Electra system about 2 years ago that had been in service since the mid 90s |
| 08:13.00 | liquidamber | like 616 type crap? |
| 08:13.02 | liquidamber | thats what he has |
| 08:13.10 | *** join/#asterisk vlad_starkov (~vlad_star@94.228.240.93) |
| 08:14.56 | drmessano | No, a bit newer.. Wow |
| 08:15.10 | drmessano | A 616 is kinda scary |
| 08:15.17 | liquidamber | dude. |
| 08:15.26 | liquidamber | let me see if i can find a pic of this thing |
| 08:15.58 | drmessano | You can upgrade him to an Electra Elite 48 for $55 |
| 08:16.01 | drmessano | eBay lol |
| 08:20.55 | liquidamber | he has this janky operator panel too |
| 08:21.02 | liquidamber | looks like some star trek crap |
| 08:21.14 | liquidamber | which is actually older than the other handsets |
| 08:21.46 | liquidamber | i have a pic somewhere cant find it but it's a gigantic metal enclosure with exposed line cards |
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| 08:53.04 | eirirs_ | hehe |
| 08:53.14 | eirirs_ | therer are lots of old venus phone systems here around |
| 08:53.15 | eirirs_ | :/ |
| 08:53.28 | liquidamber | my tax person has a merlin |
| 08:53.38 | liquidamber | might sell them the UC520 i had for my lab :P |
| 08:53.52 | liquidamber | ... so they can have caller ID |
| 09:16.41 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 09:16.53 | linocisco | hello all |
| 09:17.00 | liquidamber | . |
| 09:18.36 | linocisco | http://sipml5.org/call.htm?svn=179 doesn't work |
| 09:20.10 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:25.23 | *** join/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 09:25.59 | whizzi | G'day all :) |
| 09:27.16 | liquidamber | looks like a clusterfuck, i'm not surprised that it doesnt work |
| 09:29.23 | *** join/#asterisk DataWraith (~s1gny@188-194-90-27-dynip.superkabel.de) |
| 09:30.20 | mirela666 | <PROTECTED> |
| 09:30.33 | linocisco | can't register |
| 09:31.09 | mirela666 | to your PBX? |
| 09:31.30 | linocisco | mirela666, it always said unauthorized. how can I get mandatory items(*)? I try to use as per http://code.google.com/p/sipml5/wiki/Public_SIP_Servers. but not ok. |
| 09:31.42 | linocisco | I use on my firefox desktop |
| 09:32.39 | mirela666 | linocisco: register publi account on sip2sip (https://mdns.sipthor.net/register_sip_account.phtml) and try it out |
| 09:33.01 | linocisco | mirela666, let me try. thanks |
| 09:35.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:46.30 | linocisco | mirela666, Request Timeout always |
| 09:52.10 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 09:53.37 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 10:01.30 | linocisco | mirela666, Failed to connet to the server |
| 10:04.00 | DataWraith | how to i transfer a call to an internal sip phone? |
| 10:07.48 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 10:10.39 | mirela666 | linocisco: yes, i get timeouts too now |
| 10:12.21 | *** join/#asterisk lok-jpr-india (lok-jpr-in@202.191.157.74) |
| 10:12.41 | *** join/#asterisk vlad_sta_ (~vlad_star@195.218.175.34) |
| 10:13.00 | lok-jpr-india | anyone handy with using openssl with softphone and asterisk..! |
| 10:13.19 | *** join/#asterisk vlad_st__ (~vlad_star@195.218.175.34) |
| 10:18.04 | mirela666 | linocisco: try now |
| 10:18.11 | mirela666 | I'm connected |
| 10:18.25 | mirela666 | I guess they had some down time |
| 10:18.56 | linocisco | mirela666, yes. i m connected. so happy. what is next? which number should I call to test? |
| 10:19.20 | mirela666 | they had some echo test number if remeber |
| 10:19.58 | mirela666 | 3333 |
| 10:20.12 | mirela666 | and 4444 for microphone |
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| 10:30.17 | mirela666 | hm getting Not Acceptable Here msg |
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| 10:32.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:45.03 | linocisco | mirela666, Media stream permission denied |
| 10:45.03 | linocisco | <PROTECTED> |
| 10:47.57 | mirela666 | hm |
| 10:55.37 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
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| 11:06.49 | asilva | Hi all.. |
| 11:06.58 | cusco | yellow |
| 11:07.56 | asilva | I'm Using Asterisk 11.7.0 with MySQL realtime with this configuration - http://pastebin.com/D1qRRjFg , from time to time after the peer is registered I get "Wrong Password" message on /var/log/asterisk/messages from a Peer that registered before, If I run "sip prune peer XX" the message stop, but again will show up after |
| 11:09.13 | cusco | I would suggest looking at a sip debug trying to figure why you see wrong password |
| 11:10.32 | whizzi | <PROTECTED> |
| 11:12.33 | asilva | no eerrors on mysql. Let me debug sip to see if anything odd shows |
| 11:14.52 | asilva | Its a SIP REGISTER followed from a Sip UNAUTHORIZED then another REGISTER with UNAUTHORIZED again |
| 11:15.03 | asilva | no |
| 11:15.14 | asilva | First register = UNAUTHORIZED , Second attempt = FORBIDDEN |
| 11:15.22 | asilva | http://pastebin.com/PKJbSxT2 |
| 11:15.43 | whizzi | that's pretty common |
| 11:16.25 | asilva | yup, nothing odd, If I sip prune that specific extension, it will register again |
| 11:16.33 | whizzi | first register is without password, second one has "nonce" code |
| 11:17.34 | whizzi | hmm |
| 11:18.10 | whizzi | you're trying to link a Yealink to Asterisk |
| 11:18.16 | asilva | yes |
| 11:18.25 | asilva | it's an Yealink Phone |
| 11:18.39 | whizzi | I've had issues in the past with Yealinks, but I'm trying to recall what I've done to fix that |
| 11:19.30 | asilva | we have like 10000 yealinks here, this is the first problem, there are 2 changes, upgrade to new firmware and using asterisk 11.7(was 1.8 before but firmware wasn't the one using) |
| 11:20.52 | asilva | I have other servers running asterisk 1.8 with the up-to-date firmware and hasn't had any issues like this one using 11.7 |
| 11:21.00 | whizzi | hm? you are using Asterisk 11.8.0-rc1 according to the trace |
| 11:21.06 | asilva | yes |
| 11:21.25 | asilva | i mentioned 11.7 because is where I first had the problem(never used before 11.7) |
| 11:21.35 | whizzi | you said earlier you were using 11.7.0 |
| 11:21.43 | asilva | then I saw some sip register patches on changelog and updated to 11.8.0-rc1 to see if anything changes |
| 11:21.49 | asilva | but nothing |
| 11:23.18 | whizzi | any other reason why Asterisk gives a 403 Forbidden? |
| 11:23.28 | whizzi | what does the verbose-log say? |
| 11:23.34 | whizzi | any warnings/errors on the CLI ? |
| 11:24.36 | asilva | [Feb 10 09:23:37] NOTICE[1550] chan_sip.c: Registration from '"SUPORTE - Thiago Fonseca" <sip:1192@186.217.91.254>' failed for '186.217.89.116:5062' - Wrong password |
| 11:24.40 | asilva | this is the only message on log |
| 11:25.29 | whizzi | but it does register eventually ? |
| 11:25.35 | asilva | even re-setting the password on Yealinks Management interface doesn't change anything. |
| 11:26.03 | asilva | Only will register again if I execute sip prune peer |
| 11:26.40 | whizzi | it seems to me that somehow Asterisk lost it's connection to MySQL or is unable to get the password |
| 11:27.01 | whizzi | max-connections on MySQL or something like that |
| 11:27.03 | asilva | make sense, but I can't see mysql errors anywhere |
| 11:31.54 | asilva | all I see from asterisk realtime mysql is - connection okay |
| 11:32.14 | whizzi | any indication in mysql.log ? |
| 11:36.14 | asilva | no |
| 11:36.30 | asilva | no errors at all or message saying connection timed out or anything related |
| 11:36.53 | asilva | checking asterisk log, for mysql SELECT's and UPDATEs, at each of these lines there is one first saying - connection okay |
| 11:40.20 | *** join/#asterisk vlad_starkov (~vlad_star@195.218.175.34) |
| 11:54.11 | asilva | whizzi: check this out if makes sense to you |
| 11:54.42 | asilva | just realized the field type on mysql table was null, I set them to friend as it supposed to be, and reloaded, seems to fixed |
| 11:55.54 | whizzi | could be the solution indeed :) |
| 11:56.51 | asilva | it was |
| 11:57.03 | asilva | no problems so far afther that change |
| 11:57.39 | whizzi | good :). Another thing I've learned today \o/ |
| 11:57.47 | asilva | so do I |
| 11:57.50 | asilva | :D |
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| 16:42.30 | salz212 | Hello guys, would someone please mention the link of codec_opus.so file for Asterisk 12. I tried to use the once from v.11. But "Module 'codec_opus.so' was not compiled with the same compile-time options as this version of Asterisk". |
| 16:49.27 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.118) |
| 16:53.01 | navaismo | As far i know you need toinstall the opus library then it will be available to compile in the make menuselect |
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| 17:12.39 | polysics | hello! we oddly have calls that drop after 30s |
| 17:13.06 | polysics | calls are placed thorugh Originate, they drop after 30s Asterisk has not played any audio |
| 17:13.21 | skrusty | are the calls answered? |
| 17:13.23 | navaismo | maybe rtptimeout or the retransmission errors... shooting in the dark without more info |
| 17:13.25 | polysics | ie. if I play nothing it drops after 30s, if I play monkeys it drops after I think about 52s |
| 17:13.35 | [TK]D-Fender | Go look at the call |
| 17:13.40 | polysics | calls are answered |
| 17:14.53 | *** join/#asterisk Solution-X (~Derp@unaffiliated/solution-x) |
| 17:15.04 | Solution-X | anyone have a specific softphone that they prefer? |
| 17:16.09 | [TK]D-Fender | ~soft-phone |
| 17:16.14 | [TK]D-Fender | ~softphone |
| 17:16.14 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 17:16.27 | [TK]D-Fender | All suck. Some les than others |
| 17:16.31 | [TK]D-Fender | Avoid where possible |
| 17:17.48 | *** join/#asterisk serafie (~erin@nat/digium/x-iqmivlthrtybapva) |
| 17:17.52 | Chainsaw | Soft phones and NAT. Add some restrictive 3G networks and iPads to the mix and you're really in trouble. |
| 17:18.44 | Chainsaw | (G729, ICE and encrypt what you can. If your religion allows... pray.) |
| 17:21.43 | *** join/#asterisk mjordan (~matt@nat/digium/x-daiwqknnhprckbtc) |
| 17:21.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:27.00 | polysics | [TK]D-Fender: what do you mean with "let's see the call" please? |
| 17:27.27 | [TK]D-Fender | polysics: You're asking why it drops... LOOK AT THE CALL. |
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| 17:31.01 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:31.18 | polysics | a-ha! Reason: SIP;description="RTP stream closed due to inactivity; mediaType=AUDIO" |
| 17:31.22 | *** topic/#asterisk by leifmadsen -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
| 17:31.28 | polysics | can this be turned off? |
| 17:31.44 | navaismo | rtptimeout |
| 17:32.51 | [TK]D-Fender | Why wouldn't you have RTP? |
| 17:33.37 | polysics | that's what I am wondering too |
| 17:35.13 | polysics | rtptimeout is on but is at 60, not 30 |
| 17:35.43 | polysics | if I do play something, I still get the timeout |
| 17:36.09 | Solution-X | [TK]D-Fender - thank you sir...yeah ive kinda noticed they pretty well suck. ive used zoiper and xlite, havent used the others |
| 17:36.30 | Solution-X | not really happy with either but zoiper seems to be what my subordinates like |
| 17:37.08 | [TK]D-Fender | pc sound-cards = meh, softphone interfaces & distraction value = meh |
| 17:37.20 | [TK]D-Fender | PC headsets = meh |
| 17:37.40 | [TK]D-Fender | Don't want my haed chained to a PC.... |
| 17:39.53 | navaismo | Solution-X, linphone, jitsi and blink |
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| 17:49.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:01.37 | *** join/#asterisk boratynski (~kamilbora@aqd243.internetdsl.tpnet.pl) |
| 18:01.39 | boratynski | Hello. |
| 18:02.05 | boratynski | Question. Is it possible I may see: dahdi channels and I'm not able to see pri show spans? |
| 18:03.14 | [TK]D-Fender | yes |
| 18:03.33 | [TK]D-Fender | by having allocated DAHDI channels.. and NOT having a PRI defined |
| 18:04.00 | [TK]D-Fender | pastebin your configs and status dumps. |
| 18:04.01 | [TK]D-Fender | ~pb |
| 18:04.02 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:04.03 | [TK]D-Fender | ^^^ |
| 18:04.12 | boratynski | [TK]D-Fender: Ok, minute. |
| 18:10.03 | boratynski | [TK]D-Fender: http://wklej.org/id/1268863/ |
| 18:10.35 | [TK]D-Fender | boratynski: permanently remove all the commented out junk from your configs. |
| 18:11.03 | [TK]D-Fender | signalling = bri_cpe_ptmp |
| 18:11.09 | [TK]D-Fender | This alsodoesn't look like PRI |
| 18:11.10 | *** join/#asterisk eirirs_ (eirirs@dalvik.ping.uio.no) |
| 18:11.42 | boratynski | [TK]D-Fender: Mainly, BRI doesn't work too. |
| 18:11.43 | boratynski | I mean. |
| 18:11.44 | [TK]D-Fender | We also need ot see your system.conf |
| 18:11.53 | boratynski | Ok. |
| 18:12.18 | boratynski | [TK]D-Fender: /etc/asterisk/system.conf? |
| 18:13.27 | boratynski | I do not see such as file. |
| 18:13.37 | [TK]D-Fender | DAHDI configs.... |
| 18:14.01 | [TK]D-Fender | <PROTECTED> |
| 18:15.17 | boratynski | [TK]D-Fender: system.conf - http://wklej.org/id/1268873/ |
| 18:16.18 | [TK]D-Fender | Well this isn't PRI.... |
| 18:16.34 | [TK]D-Fender | So is there something that is actually wrong right now? |
| 18:16.35 | boratynski | [TK]D-Fender: BRI, I guess. |
| 18:17.10 | boratynski | [TK]D-Fender: Mainly. I show get some notification, there is no such as context [from-ptns] or something, when I call. |
| 18:17.24 | boratynski | And I should be able to write gsm command and call from GSM card. |
| 18:17.31 | [TK]D-Fender | That error is pretty clear about its meaning |
| 18:17.48 | [TK]D-Fender | You told you channel config to send calls into the dialplan and there is no match there for it |
| 18:18.18 | boratynski | [TK]D-Fender: But... I don't see that notification. |
| 18:18.23 | boratynski | And I guess. I should, right? |
| 18:18.37 | boratynski | I'm connecting to asterisk with lots of 'v' as verbose. |
| 18:18.44 | [TK]D-Fender | [13:17]boratynski[TK]D-Fender: Mainly. I show get some notification, there is no such as context [from-ptns] or something, when I call. <- just just said you see it |
| 18:18.53 | [TK]D-Fender | you* |
| 18:19.19 | boratynski | Sorry. I don't get that notification |
| 18:19.37 | boratynski | And I think... I should see them. |
| 18:19.56 | [TK]D-Fender | "core set verbose 10" |
| 18:20.03 | [TK]D-Fender | show the call |
| 18:21.08 | boratynski | [TK]D-Fender: Still nope. |
| 18:21.37 | [TK]D-Fender | show the channel dump and that verbose is set |
| 18:22.01 | boratynski | Ok. Se. |
| 18:22.05 | boratynski | from-gsm works well. |
| 18:22.12 | boratynski | So the problem may be... the connection is wrong. |
| 18:23.09 | boratynski | But... what's strange. |
| 18:23.17 | boratynski | dahdi channels don't presents from-gsm. |
| 18:23.18 | boratynski | ;-/ |
| 18:23.30 | [TK]D-Fender | why should it? |
| 18:23.40 | [TK]D-Fender | I see you put "context=from-pstn" in there |
| 18:24.06 | [TK]D-Fender | http://wklej.org/id/1268863/ <- 2197 |
| 18:25.16 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 18:25.25 | boratynski | [TK]D-Fender: http://wklej.org/id/1268890/ - actual channels. |
| 18:25.35 | boratynski | And... when I call on GSM card. It works well... |
| 18:27.04 | [TK]D-Fender | gsm CARD DOESN'T LOOKS LIKE THE CHANNEL IS THERE |
| 18:27.13 | [TK]D-Fender | I see 3/4 from a BRi card |
| 18:27.25 | [TK]D-Fender | and I see a different context |
| 18:27.34 | [TK]D-Fender | you also aren't showing this GSM call you say is "good" |
| 18:27.45 | [TK]D-Fender | We have nothing to compare for what you are saying |
| 18:28.41 | boratynski | [TK]D-Fender: That's right. Shouldn't GSM card be a next one DAHDI-channel? |
| 18:29.26 | [TK]D-Fender | It should. But you're talking about successful calls and haven't shown us anything |
| 18:30.18 | boratynski | [TK]D-Fender: Wait, I will do nextone call and show Asterisk logs, ok? |
| 18:33.43 | boratynski | [TK]D-Fender: http://wklej.org/id/1268899/ |
| 18:33.48 | boratynski | Strange... |
| 18:34.41 | [TK]D-Fender | boratynski: Is there some other config for that GSM card? |
| 18:34.55 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:35.14 | [TK]D-Fender | Because chan_dahdi has NO reference to "gsm" anywhere |
| 18:35.34 | [TK]D-Fender | #include extra-channels.conf <--------------- |
| 18:37.15 | boratynski | Damn... |
| 18:38.12 | boratynski | [TK]D-Fender: I don't see that too... |
| 18:38.49 | [TK]D-Fender | Did you find that file? |
| 18:38.54 | [TK]D-Fender | Didi you look for others? |
| 18:39.03 | [TK]D-Fender | Did you grp that context name in your config folder? |
| 18:39.05 | [TK]D-Fender | grep* |
| 18:39.07 | boratynski | See it! |
| 18:40.16 | boratynski | [TK]D-Fender: extra-channels.conf http://wklej.org/id/1268908/ |
| 18:40.39 | boratynski | [TK]D-Fender: WTF... ;-/ |
| 18:41.03 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:41.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:41.34 | [TK]D-Fender | that looks like system.conf & chan_dahdi.conf settings mashed together. |
| 18:41.41 | [TK]D-Fender | go rebuild this mess you've got. |
| 18:41.50 | boratynski | [TK]D-Fender: See it. |
| 18:41.53 | boratynski | What do you suggest? |
| 18:41.56 | boratynski | dahdi_genconf? |
| 18:45.04 | boratynski | [TK]D-Fender: Isn't it strange, when I don't see dahdi-channel? |
| 18:45.15 | *** join/#asterisk Milarepa (~Milarepa@wsip-24-234-72-245.lv.lv.cox.net) |
| 18:48.24 | [TK]D-Fender | Clean up both files yourself to the bare minimum |
| 18:54.19 | *** join/#asterisk wtfitsme (~WTFitsME@static-72-89-169-149.nycmny.fios.verizon.net) |
| 18:56.00 | *** join/#asterisk YannikS (~Miranda@31-18-193-221-dynip.superkabel.de) |
| 18:59.50 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 19:04.11 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:05.16 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:06.37 | boratynski | [TK]D-Fender: Is it enought to specify spans in [channels] group in chan_dahdi.conf file? |
| 19:06.40 | boratynski | enough* |
| 19:06.47 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:06.59 | [TK]D-Fender | You don't specify spans in that file at all |
| 19:07.02 | [TK]D-Fender | that is only system.conf |
| 19:07.13 | boratynski | [TK]D-Fender: Damn. |
| 19:07.19 | [TK]D-Fender | you define individual channels in chan_dahdi |
| 19:07.20 | boratynski | So this file was corrupted at all. |
| 19:07.26 | boratynski | Minute. |
| 19:07.32 | [TK]D-Fender | as I said, you have stuff mashed together that shouldn't be |
| 19:08.41 | *** join/#asterisk monsterco (~monsterco@64.231.105.94) |
| 19:09.26 | monsterco | I have a SIP trunk with a provider that works if I have "username, secret, fromuser" variables in trunk but it wont work with just "username, secret" variables. What is it with "fromuser" that makes it required? |
| 19:10.00 | boratynski | [TK]D-Fender: chan_dahdi.conf http://wklej.org/id/1268953/ and system.conf - http://wklej.org/id/1268956/ |
| 19:10.05 | boratynski | Now it looks quite well, right? |
| 19:11.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.214) |
| 19:11.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.214) |
| 19:16.51 | [TK]D-Fender | no, very backwards |
| 19:17.26 | [TK]D-Fender | http://wklej.org/id/1268953/ <-- why do I see the words system.conf in here? |
| 19:17.33 | [TK]D-Fender | And half of it is still commented out.. why? |
| 19:18.23 | [TK]D-Fender | http://wklej.org/id/1268956/ <- maybe OK.... check dmesg to be sure |
| 19:18.48 | [TK]D-Fender | looking again the files might be OK, but you've commented out your GSm channel... |
| 19:19.09 | boratynski | [TK]D-Fender: Yes. |
| 19:19.14 | boratynski | I did it. |
| 19:19.23 | boratynski | And now I would like to see only from-pstn. |
| 19:19.41 | [TK]D-Fender | not the same thing.. |
| 19:19.45 | [TK]D-Fender | you disabled the DEVICE. |
| 19:19.49 | monsterco | anyone on the "fromuser" ? |
| 19:19.56 | [TK]D-Fender | Where its channels point to is another matter |
| 19:20.17 | [TK]D-Fender | monsterco: Please rephrase that as a complete question... |
| 19:20.50 | boratynski | [TK]D-Fender: Hm? I doesn't matter, I think. I just don't want to turn on GSM card. |
| 19:21.20 | [TK]D-Fender | monsterco: Ok, rading back I think I see what you';re getting at. When you call out typically the CALLERID of your initiating channel gets put in the "from" and that's what they'll take as the "user" it is from. That's why |
| 19:21.36 | boratynski | [TK]D-Fender: Now, dahdi command is unavailable. <3 :-) |
| 19:21.38 | [TK]D-Fender | boratynski: Ok, but You didn't say that =to us clearly |
| 19:21.59 | [TK]D-Fender | boratynski: For those changes you'll have to stop *, and re-initi dahdi |
| 19:22.08 | boratynski | [TK]D-Fender: I did it. |
| 19:22.19 | boratynski | Restarted DAHDI and after that Asterisk. |
| 19:22.21 | [TK]D-Fender | dahdi_cfg -vvvv |
| 19:23.49 | boratynski | [TK]D-Fender: http://wklej.org/id/1268975/ |
| 19:24.03 | salz212 | navaismo: as far as I know I need codec_opus.so file for Asterisk 12. Typical installation of Asterisk 12 does not compile opus module as it is not supported other than passthrough. However is it really that hard to find the appropriate .so file to test with Asterisk 12. |
| 19:25.17 | [TK]D-Fender | go contact the autor. |
| 19:25.25 | salz212 | searching it over the internet didn't help much, would have been better if it was in http://downloads.digium.com/pub/telephony/ |
| 19:26.32 | [TK]D-Fender | salz212: It isn't supported in *12 ... you WON'T find it. |
| 19:26.43 | monsterco | So, why does the other side ignore the "username" field? can I not go without fromuser? that means changes to a lot of systems for me |
| 19:27.09 | [TK]D-Fender | monsterco: Sounds like one file to change so far... what is this "lot of changes" you're talking about? |
| 19:27.21 | monsterco | [TK]D-Fender - also, the CALLERID is becoming an issue as I want the caller ID to be different from username and when I send "fromuser" they pass it as my callerid |
| 19:27.22 | [TK]D-Fender | boratynski: what GSm card is that exactly? |
| 19:27.35 | [TK]D-Fender | monsterco: So set your peers to use RPID for it |
| 19:27.36 | boratynski | [TK]D-Fender: OpenVox G400E. |
| 19:27.39 | monsterco | there are many instances I have to make the changes |
| 19:27.53 | monsterco | [TK]D-Fender - please elaborate on RPID? |
| 19:28.21 | salz212 | [TK]D-Fender: are you sure about that, as it is clearly mentioned opus and vp8 are supported in pass through mode. |
| 19:28.52 | [TK]D-Fender | monsterco: "sendrpid=yes", "trustrpid=yes" |
| 19:29.09 | [TK]D-Fender | [14:27]monstercothere are many instances I have to make the changes <- sounds horrible... |
| 19:29.13 | salz212 | [TK]D-Fender: I know about asterisk-opus source on git but it is for trans-coding and more specifically for v.11.x |
| 19:29.45 | [TK]D-Fender | salz212: Well there isn't a code_opus.so from Asterisk for this. |
| 19:29.53 | [TK]D-Fender | salz212: So whatever you're looking for is 3rd party |
| 19:30.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.24) |
| 19:31.12 | [TK]D-Fender | boratynski: http://wiki.openvox.cn/index.php/OpenVox_G400E_Installation_Manual#3.3_chan_extra.2C_DAHDI_and_asterisk_installation |
| 19:31.28 | [TK]D-Fender | boratynski: that is a very unusual channel driver it's using ther... not sure what to make of it |
| 19:31.29 | boratynski | [TK]D-Fender: It works! |
| 19:31.39 | boratynski | I mean. The GSM works well. |
| 19:31.44 | boratynski | I am able to make calls and go on. |
| 19:31.44 | salz212 | [TK]D-Fender: of course its not Asterisk's for sure, all I need is to test opus with Asterisk 12 in pass-through, and after searching all over could not find it. |
| 19:31.50 | boratynski | I sent SMS to myself. |
| 19:32.03 | [TK]D-Fender | salz212: Searching for what? |
| 19:32.59 | salz212 | [TK]D-Fender: to make opus work in passthrough... .so file perhaps |
| 19:33.05 | [TK]D-Fender | no |
| 19:33.08 | [TK]D-Fender | there IS NO FILE |
| 19:33.11 | [TK]D-Fender | you don't seem to get it. |
| 19:33.24 | [TK]D-Fender | "code_opes" would be PROVIDING actual transcoding support. |
| 19:33.32 | [TK]D-Fender | Passthrough does not NEED a module. |
| 19:33.50 | [TK]D-Fender | It takes raw payload, rewraps it and shoves it out the door |
| 19:35.27 | monsterco | [TK]D-Fender - Setting "sendrpid=yes, trustrpid=yes" in conjunction with "fromuser" still gives me the username as CALLERID on the other side |
| 19:35.34 | monsterco | What are those parameters supposed to do? |
| 19:37.30 | boratynski | [TK]D-Fender: Suggestion, if so? |
| 19:37.47 | [TK]D-Fender | monsterco: and did you set it on BOTH sides? |
| 19:38.37 | [TK]D-Fender | boratynski: Not sure what you're asking at this point. You just finished telling me all sort of thigns work. |
| 19:39.31 | boratynski | I mean... Both cards are working... but something is not like it should be. I should see channel, but I don't see it. BUT I am able to see the context defined to channel. |
| 19:40.40 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 19:41.28 | *** join/#asterisk Big_E (~Big_E@50.240.55.81) |
| 19:41.56 | Big_E | Hello everyone |
| 19:42.13 | Big_E | anyone ever used DID2Call ? |
| 19:44.38 | *** join/#asterisk JeffC_NN (~jcollell@mail.nownerd.com) |
| 19:44.47 | JeffC_NN | Having issues with caller ID |
| 19:45.08 | [TK]D-Fender | boratynski: which channel? You are being VERY vague and you have 2 cards and I can never tell what you are talking about. |
| 19:45.13 | Big_E | having an issue setup trunk |
| 19:45.44 | [TK]D-Fender | Big_E: such as? |
| 19:45.51 | JeffC_NN | Whenever a user calls over our PRIs with a blocked caller ID (Ex: NoOp("DAHDI/i3/-325", "Incoming Call From "" <>")) I can't set the caller ID in asterisk, and it always sends From headers like this: From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as20ea0375 |
| 19:46.00 | ipengineer | When trying to install Asterisk 12 does mysql-server have to be installed? I have "mysql" installed but am getting a bunch of errors on make like mysql/mysql.h: No such file or directory |
| 19:46.31 | WIMPy | JeffC_NN: Set (pres) as well. |
| 19:47.27 | wdoekes | ipengineer: you don't need mysql, unless you're compiling mysql-addons. you can make menuselect and uncheck all mysql-using modules |
| 19:47.46 | ipengineer | wdoekes: I am using several mysql add-ons but the sql server is on a different box |
| 19:47.48 | wdoekes | or, re-run ./configure and the module won't even be selectable |
| 19:47.50 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:48.04 | [TK]D-Fender | ipengineer: You shouldn't be using those anyway. ODBC <--------- |
| 19:48.20 | wdoekes | if you want to compile mysql modules, you'll need the devel packages |
| 19:48.39 | wdoekes | mysql-devel or libmysqlclient-dev or whatever it's called on your system |
| 19:48.50 | ipengineer | I have ODBC as well and am primarily using that.. I thought I might end up needing res_mysql so I wanted to install with that as well |
| 19:49.00 | wdoekes | just use odbc |
| 19:49.14 | ipengineer | Ok |
| 19:49.16 | [TK]D-Fender | JeffC_NN: "core show function CALLERPRES" |
| 19:50.10 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 19:50.40 | *** part/#asterisk monsterco (~monsterco@64.231.105.94) |
| 19:51.39 | *** join/#asterisk Milarepa (~Milarepa@wsip-24-234-72-245.lv.lv.cox.net) |
| 19:52.59 | Weezey | having a weird issue when I access a new menu I created the playback of the audio starts playing about 2-3 seconds after the call is answered. using background() to play it via realtime dialplan. |
| 19:53.14 | Weezey | Other menus play instantly. |
| 19:53.39 | Weezey | Thought it was the sound file so set this menu to play a beep, or one working elsewhere, same deal. |
| 19:55.40 | JeffC_NN | Thanks!! That did it. |
| 19:58.21 | salz212 | [TK]D-Fender: In that case a call between 2 SIP extensions having codec can communicate right? |
| 20:00.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.24) |
| 20:00.42 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
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| 20:02.40 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:02.40 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:03.00 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 20:04.45 | JeffC_NN | Can you think of any reason I shouldn't Set(CALLERPRES()=allowed) on all incoming calls? |
| 20:05.17 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-wmslanlybribyriz) |
| 20:09.12 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-zypsoujgpdmmdkyt) |
| 20:09.38 | ipengineer | To use odbc instead of res_mysql I just need to add the connection to obdc.ini and change the mysql in extconfig here: "queue_log => mysql,asterisk" to odbc? Does res_odbc.conf not matter? |
| 20:12.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.24) |
| 20:13.09 | [TK]D-Fender | [14:58]salz212[TK]D-Fender: In that case a call between 2 SIP extensions having codec can communicate right? <- assuming * doesn't have to process anything in the midle |
| 20:13.26 | [TK]D-Fender | salz212: You should have a proper understanding about what "pass-though" actually means |
| 20:13.55 | [TK]D-Fender | ipengineer: it does |
| 20:13.56 | salz212 | [TK]D-Fender: sure do now. thanks |
| 20:15.02 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-pnimaxtqeppztrcw) |
| 20:15.10 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:15.22 | ipengineer | [TK]D-Fender: Ok so I am using odbc for cdd and it is not configured there.. So if I have the connection in odbc.ini setup what needs to be added to res_odbc? |
| 20:16.15 | ipengineer | [TK]D-Fender: nvm. I think I see what I need to do I will come back if it doesn't work |
| 20:16.23 | [TK]D-Fender | ok |
| 20:18.49 | WIMPy | JeffC_NN: Well, if you set te CallerID, ok, but if you want to keep it, you should keep the other fields as well. |
| 20:25.49 | *** join/#asterisk arakai (~Thunderbi@94.139.47.64) |
| 20:35.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.24) |
| 20:53.23 | ipengineer | [TK]D-Fender: I was able to configure my res_odbc but when I do a pjsip show endpoints I get a bunch of SQL Execute errors.. If I do "odbc show" it says it is connected: Yes |
| 20:53.58 | [TK]D-Fender | go check your tables & mappings, etc |
| 20:56.07 | ipengineer | [TK]D-Fender: this is what I have in extconfig: ps_endpoints => odbc,asterisk where asterisk is the table name and DSN name |
| 20:56.36 | [TK]D-Fender | look at the execute errors, verify table structure, etc.... |
| 20:57.36 | ipengineer | Ok. If I change it over to use res_config_mysql it works though? So it makes me think I am doing something wrong with the odbc connection |
| 20:57.53 | *** part/#asterisk sereal (~sereal@unaffiliated/sereal) |
| 20:58.22 | arakai | some know if the cisco 9971 phone working with asterisk 12 |
| 21:00.59 | [TK]D-Fender | ipengineer: You're not showing anything. I have nothing more to add at this point |
| 21:01.57 | ipengineer | [TK]D-Fender: Ok let me show you the error |
| 21:03.24 | *** join/#asterisk wtfitsme (~WTFitsME@cpe-67-244-92-207.nyc.res.rr.com) |
| 21:03.55 | *** part/#asterisk arakai (~Thunderbi@94.139.47.64) |
| 21:04.00 | [TK]D-Fender | arakai: it speaks SIP. Asterisk version is pretty much irrelevant |
| 21:04.19 | [TK]D-Fender | very patient... |
| 21:04.21 | ipengineer | [TK]D-Fender: https://gist.github.com/zconkle/9cdf2b90bb13690684c5 |
| 21:06.02 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:07.31 | [TK]D-Fender | not seeing verification of tables, etc... |
| 21:08.40 | ipengineer | [TK]D-Fender: I updated the gist with the db tables |
| 21:09.08 | *** join/#asterisk vader- (~me@204.183.88.101) |
| 21:09.18 | *** join/#asterisk CeBe (~CeBe@port-92-206-15-23.dynamic.qsc.de) |
| 21:09.28 | vader- | Do any of you guys have experience with the: Konftel 300W Wireless Conference Phone |
| 21:09.55 | vader- | I have gotten a request from my management (who currently have a Polycom 7000 Conference phone) to get something wireless. |
| 21:10.15 | vader- | So im looking to see if anyone has any good/bad things to say about this Konftel device. |
| 21:10.56 | WIMPy | At least it's not wifi. |
| 21:11.03 | WIMPy | So it could be ok. |
| 21:11.17 | ipengineer | [TK]D-Fender: One thing I do notice is I don't have a res_config_odbc just a res_odbc which has all of my connections in it |
| 21:11.32 | [TK]D-Fender | You should probably look at that... |
| 21:12.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.183) |
| 21:12.49 | *** join/#asterisk ldardini (~leandro@ip-224-67.sn2.eutelia.it) |
| 21:13.01 | ldardini | hello all... |
| 21:13.20 | vader- | Unless you guys know of a decent Wireless SIP conference phone... |
| 21:13.26 | vader- | I was told NO WIRES! |
| 21:13.27 | vader- | hehe |
| 21:14.05 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:14.34 | [TK]D-Fender | Who says the phoe itself has to be SIP? |
| 21:14.58 | WIMPy | Noone, and luckily that one isn't. |
| 21:15.13 | WIMPy | But it's also only old DECT. |
| 21:15.18 | [TK]D-Fender | I us a Polycom wireless conf phone... |
| 21:15.35 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:15.53 | vader- | which model? |
| 21:15.58 | WIMPy | 2W? |
| 21:16.30 | [TK]D-Fender | yup |
| 21:17.03 | WIMPy | The ttechnical spoecs seem to be similar. |
| 21:17.29 | ldardini | a little help... when a voicemail message is left on my server, I always see this warning message: retrieve_file: SQL Get Data error! coltitle=category [SELECT * FROM voicemail_messages WHERE dir=? AND msgnum=?] ... how can I get rid of it? |
| 21:18.57 | vader- | are their any wifi sip conference phones you can think of? |
| 21:19.25 | WIMPy | You don't want to use wifi for voice. |
| 21:22.48 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 21:30.26 | *** join/#asterisk u0m3_ (~u0m3@92.80.106.31) |
| 21:31.31 | vader- | ever see this guy? |
| 21:31.31 | vader- | http://www.revolabs.com/flx2 |
| 21:33.51 | WIMPy | Some unknown kind of wireless? |
| 21:36.22 | *** part/#asterisk polysics (~Adium@host190-64-dynamic.248-95-r.retail.telecomitalia.it) |
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| 22:19.52 | *** join/#asterisk Neozonz (~arajakul@unaffiliated/neozonz) |
| 22:19.55 | *** join/#asterisk xaristax (~xaristax@148.240.65.35) |
| 22:20.24 | xaristax | Hi can someone help me is there a way two transmit a playback during a dial comman two both channels |
| 22:20.28 | Neozonz | Can anyone help me with an Asterisk + X-Lite issue? ( no audio recieved or sent during calls, I see payload packet in logs though) |
| 22:20.57 | xaristax | im using the U parameter into the Dial command to make a Playback |
| 22:21.11 | xaristax | but only plays for one channel |
| 22:21.21 | Neozonz | Client (X-Lite) (behind nat) + Asterisk (Public) + Client (X-Lite) (Behind NAT) |
| 22:21.28 | Neozonz | any help would be appreciated |
| 22:21.57 | xaristax | neozon do you already use the nat=force_rport,comedia |
| 22:24.02 | Neozonz | no, set nat=no |
| 22:24.09 | Neozonz | as asterisk is not behind the fw |
| 22:24.17 | Neozonz | it's on a public interface |
| 22:24.32 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 22:26.16 | rrittgarn | Playing with a Polycom 301. Having issues originating calls via AMI to it. Specifically when its to an Audiocodes FXS118 device. Getting SIP status 481/500 Internal server error in the pcap. If i dial the extension manually, the call goes through. Anyone have an idea of an obvious thing I'm probably missing? I can put the pcap somewhere if anyone wants to take a minute to look at it. |
| 22:28.36 | Neozonz | xaristax, yes just tried that |
| 22:28.39 | Neozonz | still no good |
| 22:31.09 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 22:35.15 | *** join/#asterisk jsjc (~Adium@61.Red-79-150-66.dynamicIP.rima-tde.net) |
| 22:40.50 | *** join/#asterisk lachesis (~lachesis@unaffiliated/lachesis) |
| 22:41.23 | lachesis | what's the best way (at a high level) to create a call forwarding box? e.g. call comes in to asterisk and gets sent out to another phone number (with caller ID matching inbound call)? |
| 22:42.17 | WIMPy | Just do it |
| 22:42.36 | WIMPy | But make sure you're using a service that allows you to send the original caller ID. |
| 22:44.10 | Neozonz | anyone have any other suggestions for me? |
| 22:44.22 | *** join/#asterisk Lars_G_ (~Lars@unaffiliated/lars-g/x-000001) |
| 22:44.30 | Lars_G_ | Hello. |
| 22:44.35 | Lars_G_ | Quick question. |
| 22:44.37 | lachesis | WIMPy: do i need to define a custom extension? |
| 22:44.53 | Lars_G_ | From what I see, digium pci/e analog FXO (modular) cards do not support T.38, is that right? |
| 22:45.00 | WIMPy | All extensions are custom. What do you mean? |
| 22:45.52 | WIMPy | T.38 is fax *over IP*. So how could an analog interface support it? |
| 22:46.23 | WIMPy | But Asterisk can do T.38. |
| 22:47.17 | Lars_G_ | Yes, but having an asterisk with t.38, will it be able to send faxes over a digium analog card? |
| 22:47.35 | WIMPy | yes |
| 22:47.38 | Lars_G_ | ok |
| 22:49.34 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 22:50.00 | Lars_G_ | Will fax sending/receiving work of course also with e1 gateways..... right? |
| 22:51.08 | WIMPy | Only if the gateway supports it. |
| 22:51.14 | Neozonz | Anyone know what the nat config for asterisk should be for Client (X-Lite) (behind nat) + Asterisk (Public) + Client (X-Lite) (Behind NAT) |
| 22:51.31 | WIMPy | In the case of a card, Asterisk is the gateway. |
| 22:52.56 | Lars_G_ | WIMPy: SHould in theory a g200 digium gateway support it? what do I need to check for, since t.38 not what to check for in the gateway? |
| 22:53.20 | Lars_G_ | Btw do you guys know of any "cheap" ata with V.150.1 support? |
| 22:53.23 | WIMPy | For a gateway you should check for T.38 support. |
| 22:53.30 | Lars_G_ | Ok, |
| 22:53.47 | WIMPy | googles |
| 22:53.49 | Lars_G_ | WIMPy: Of course, the digium card doesn't count as a gateway per se... it's just an interface.... makes sense |
| 22:54.00 | Lars_G_ | WIMPy: seems it's like t.38 for modems |
| 22:54.02 | WIMPy | Oh. that sounds evil. |
| 22:54.32 | file | I know of nothing that supports that. |
| 22:54.36 | [TK]D-Fender | Neozonz: set both to nat=yes, directmedia=no, qualify=yes |
| 22:54.47 | WIMPy | That would obviousely require modems at both ends. |
| 22:55.04 | Lars_G_ | Evil, but sometimes it's less evil that cabling thorough 10 floors for a single credit card terminal. |
| 22:55.04 | WIMPy | Like you had with mobile CSD. |
| 22:55.13 | Lars_G_ | hmmm |
| 22:55.25 | file | it's not less evil. |
| 22:55.26 | WIMPy | Get an IP enabled credit card terminal. |
| 22:55.34 | file | yeah. |
| 22:55.34 | Lars_G_ | heh |
| 22:55.36 | Lars_G_ | yeah |
| 22:55.37 | Lars_G_ | I wish |
| 22:55.54 | WIMPy | Don't wish. Do it. |
| 22:56.13 | Lars_G_ | it took 14 months of begging to get an extra terminal, phone based. they had troubles at the setting level, and since they couldn't fix the trouble in a week, they took the terminal away because we weren't using it |
| 22:56.26 | WIMPy | And pulling a cable through 10 floors is probably still easier. |
| 22:56.59 | Neozonz | [TK]D-Fender, set both to nat=yes? |
| 22:57.05 | Neozonz | what did you mean by both? |
| 22:57.10 | Neozonz | there is only one asterisk server |
| 22:57.14 | Lars_G_ | I wonder if I could pick the FXS from a digium card and it'd work, so the line could doble as a call line too hmmm |
| 22:57.33 | [TK]D-Fender | [17:51]NeozonzAnyone know what the nat config for asterisk should be for Client (X-Lite) (behind nat) + Asterisk (Public) + Client (X-Lite) (Behind NAT) |
| 22:57.35 | [TK]D-Fender | ^ |
| 22:57.47 | [TK]D-Fender | both of the PEERS your softphones are using |
| 22:58.01 | Lars_G_ | See you latter, thanks |
| 22:58.11 | Neozonz | there is no config options on the softphones for nat |
| 22:58.27 | Neozonz | only asterisk has that option |
| 22:59.28 | [TK]D-Fender | ... |
| 22:59.37 | [TK]D-Fender | in the PEER entry for each of them |
| 23:00.01 | Neozonz | I set it under general, so it should apply to the peers as well no? |
| 23:01.32 | [TK]D-Fender | you shouldn't set it as general normall, and you should always configure your peers explicitly |
| 23:01.45 | Neozonz | yeah just for testing doing it thisway |
| 23:01.56 | Neozonz | but didn't work... no audio still :( |
| 23:02.13 | Neozonz | driving me nuts this thing |
| 23:03.11 | Neozonz | would icesuppport=yes / stunaddr=stun.counterpath.net |
| 23:03.11 | Neozonz | <PROTECTED> |
| 23:03.28 | WIMPy | yes and yes |
| 23:03.37 | Neozonz | turn them both off and give it a try |
| 23:04.13 | WIMPy | Don't use stun unless your Asterisk is behind NAT and you don't know its external address. |
| 23:04.26 | Neozonz | roger that |
| 23:04.31 | Neozonz | what about icesupport? |
| 23:04.48 | WIMPy | Random outcome. |
| 23:05.07 | WIMPy | It can help but for me it only made things worse and certainly less predictable. |
| 23:05.22 | Neozonz | ok turned both off |
| 23:05.24 | Neozonz | but still no good |
| 23:05.38 | WIMPy | Did you restart Asterisk? |
| 23:06.02 | WIMPy | 'sip reload' is unsafe. |
| 23:06.11 | Neozonz | service asterisk retart |
| 23:06.26 | WIMPy | ok |
| 23:06.34 | Neozonz | man this is frustrating |
| 23:06.43 | WIMPy | Check the firewalls. |
| 23:07.10 | Neozonz | nothing is done on the 2 clients |
| 23:07.26 | WIMPy | Nah, you don't even suspect about the frustrating parts, yet. |
| 23:08.50 | Neozonz | the public astrix has https://gist.github.com/arajakul/14a2631c9249bbb6c4bc open |
| 23:09.29 | Neozonz | clients behind the nat with x-lite are doing no port forwards or anything |
| 23:09.55 | WIMPy | But they need to be able to use ports for RTP. |
| 23:10.40 | Neozonz | how can I go about using asterisk without having users behind nats configure the ports? |
| 23:11.03 | WIMPy | Depends on the firewall in use. |
| 23:11.11 | Neozonz | the usual home routers |
| 23:12.25 | WIMPy | Most should be clever enough to open the ports when used, some are too clever and mess it up and some don't allow SIP traffic because they direct it to their internal gateway. |
| 23:13.21 | Neozonz | anything I can do on asterisk to make things smoother/working? |
| 23:13.22 | Neozonz | lol |
| 23:13.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.169) |
| 23:13.37 | WIMPy | Only nat= |
| 23:14.11 | [TK]D-Fender | Neozonz: What is that FW dump from that you showed us? |
| 23:14.20 | [TK]D-Fender | Neozonz: that doesn't look like iptables. |
| 23:14.21 | Neozonz | no that is the ruleset |
| 23:14.27 | [TK]D-Fender | Neozonz: and it's intentions are vague |
| 23:14.30 | Neozonz | from an aws server |
| 23:14.44 | Neozonz | it opens the ports for all interfaces |
| 23:14.49 | [TK]D-Fender | "Actio: delete" looks negative... |
| 23:14.57 | Neozonz | delete is just a button to remove the rule |
| 23:15.14 | [TK]D-Fender | "sip set debug on" |
| 23:15.20 | [TK]D-Fender | "core set verbose 10" |
| 23:15.25 | [TK]D-Fender | PASTEBIN a complete failed call. |
| 23:15.28 | [TK]D-Fender | ~pb |
| 23:15.28 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:15.29 | [TK]D-Fender | ^^^ |
| 23:15.30 | WIMPy | You probably shouldn't have any tule for SIP. |
| 23:15.55 | WIMPy | rule |
| 23:16.24 | *** join/#asterisk DevWork_ (~DevWork@38.96.32.242) |
| 23:16.27 | WIMPy | As long as you allow any outbouind traffic. |
| 23:16.37 | DevWork_ | Hi what cards will work with centos, asterisk, and a PRI? |
| 23:16.46 | WIMPy | If you don't you surely won't get RTP running. |
| 23:16.59 | WIMPy | DevWork_: Quite a lot. |
| 23:17.47 | DevWork_ | I wasn't aware there would be a lot, but I still don't know which and where I would find those. |
| 23:18.18 | DevWork_ | What gets you a dialtone that works with old school normal phone? |
| 23:18.21 | WIMPy | There are many vendors and quite a few drivers to choose from. |
| 23:18.52 | WIMPy | That's a new question, I hope. |
| 23:19.04 | [TK]D-Fender | DevWork_: normal phones get plugged into FXS interfaces |
| 23:19.06 | DevWork_ | Yea. |
| 23:19.13 | DevWork_ | So those are the FXS, alright. |
| 23:19.27 | [TK]D-Fender | DevWork_: For which there are PCI(e) cards, SIP gateways (ATA), etc |
| 23:19.42 | DevWork_ | Yea well, I saw some PRI stuff, like the TE411P. it says 4 span T1. |
| 23:19.44 | [TK]D-Fender | DevWork_: What precisely do you want to connect? |
| 23:19.54 | [TK]D-Fender | Indeed that is a 4-port T1 card |
| 23:19.55 | DevWork_ | I wasn't sure what a span was, or what voltage pci I wanted. |
| 23:20.08 | [TK]D-Fender | span = port |
| 23:20.15 | [TK]D-Fender | that handles FOUR T1 connections. |
| 23:20.18 | DevWork_ | So I can plug in 4 Pri's |
| 23:20.19 | DevWork_ | Ok. |
| 23:20.20 | [TK]D-Fender | yes |
| 23:20.35 | DevWork_ | I have a pri interface served over the internet. |
| 23:20.37 | WIMPy | So you need T1? |
| 23:20.38 | DevWork_ | Like a virtual PRI. |
| 23:20.51 | DevWork_ | But yea it works with the T1 interface cards. |
| 23:21.12 | [TK]D-Fender | DevWork_: Sounds like you have an IAD |
| 23:21.29 | DevWork_ | Care to eleaborate, I am not familiar with that term. |
| 23:21.39 | [TK]D-Fender | DevWork_: your router does VoIP to a telco and spits our PRI for legacy systems begind it? |
| 23:21.48 | [TK]D-Fender | behind* |
| 23:21.59 | WIMPy | The only sensible thing to do with IADs it to hack them to get your account data out of them. |
| 23:22.00 | Neozonz | [TK]D-Fender, full log https://gist.github.com/arajakul/260284b545353e5197eb |
| 23:22.24 | DevWork_ | We have a WISP in the area. They do microwave hookup to their tower. They only do data over it, so they do, yea I guess a IAD over it |
| 23:22.26 | [TK]D-Fender | WIMPy: that would the logical thing |
| 23:23.17 | WIMPy | It could be a simple gateway. |
| 23:23.28 | DevWork_ | What do you mean? |
| 23:23.28 | Neozonz | any idea? |
| 23:24.06 | [TK]D-Fender | DevWork_: Normally it's wasteful to spend money on an interfaqce like that |
| 23:24.12 | DevWork_ | I have no idea how it works on their end, but we get a PRI out. |
| 23:24.23 | [TK]D-Fender | DevWork_: It's already coming in VoIP. Best option is to take it DIRECT into yourserver if you can. |
| 23:24.32 | [TK]D-Fender | DevWork_: And save the $ you'd put on that card |
| 23:25.22 | DevWork_ | We get a PRI currently, we'd have to work a different deal out under contact with them if that was the case. Existing system takes the PRI. |
| 23:25.35 | DevWork_ | I've talked to them about getting SIP credentials. |
| 23:25.38 | DevWork_ | They seemed open to it. |
| 23:25.42 | [TK]D-Fender | Neozonz: that is some odd level of CORE debug. I specified sip + verbose 10. |
| 23:25.48 | DevWork_ | They've never actually had a SIP only account. |
| 23:25.49 | [TK]D-Fender | Neozonz: and is not a complete call. |
| 23:25.53 | [TK]D-Fender | Neozonz: place another call |
| 23:26.12 | Neozonz | ok |
| 23:26.19 | Neozonz | how do i know its a complete call? |
| 23:26.25 | Neozonz | just drop the call? |
| 23:26.31 | [TK]D-Fender | DevWork_: Could be their client base is mostly legacy.. for which it'ss great they have an IAD they can drop in for you. But it isn't ideal for YOU. |
| 23:26.45 | [TK]D-Fender | Neozonz: When I see full SIP debug, you'll know. |
| 23:26.49 | DevWork_ | No. They aren't a phone provider. They provide internet + phone normally. |
| 23:26.56 | DevWork_ | No one has ever gotten JUST phone. |
| 23:27.07 | [TK]D-Fender | Neozonz: from the first INVITE coming in, the the final notice of * tearing down the last leg |
| 23:27.09 | DevWork_ | ( They are a wisp, most clients have no other connectivity, or slow other connectivity ) |
| 23:27.27 | WIMPy | DevWork_: Get the credentials and forget about the hardware they gave you. |
| 23:27.55 | DevWork_ | Yea probably will. |
| 23:28.10 | DevWork_ | Whats the deal with 3.3V pci or 5v PCI? |
| 23:28.14 | [TK]D-Fender | excellent idea. If they're open to it, take it |
| 23:28.22 | WIMPy | Your mainboard. |
| 23:28.31 | [TK]D-Fender | DevWork_: Decrepit concept for dated motherboards. |
| 23:28.41 | DevWork_ | Well, whats modern? |
| 23:28.45 | [TK]D-Fender | DevWork_: totally wothless to buy a card locked to just one. |
| 23:28.56 | [TK]D-Fender | DevWork_: get a better card that takes both :) |
| 23:28.58 | DevWork_ | Ah, so they can be either/ |
| 23:29.10 | [TK]D-Fender | a card can be... a motherboard... is fixed |
| 23:29.24 | DevWork_ | I have bought many PCI cards. I've never been seen that option in them. Not a telephony guy though. |
| 23:29.31 | *** part/#asterisk mjordan (~matt@nat/digium/x-daiwqknnhprckbtc) |
| 23:29.43 | DevWork_ | Just step down converter if you detect 5v and design for both imo. |
| 23:29.49 | WIMPy | Most cards can do both, |
| 23:29.51 | DevWork_ | buck converter. |
| 23:30.37 | Neozonz | [TK]D-Fender, what are the config options for those settings? |
| 23:31.01 | Neozonz | verbose = 10 |
| 23:31.01 | Neozonz | <PROTECTED> |
| 23:31.03 | [TK]D-Fender | I just gave all of them to you |
| 23:31.03 | Neozonz | ? |
| 23:31.05 | [TK]D-Fender | scroll up |
| 23:31.15 | Neozonz | If I do it form the CLI, it just spams to the console |
| 23:31.22 | Neozonz | can't really pastebin it |
| 23:31.31 | [TK]D-Fender | yes you can. |
| 23:31.54 | newtonr | Neozonz, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information is helpful |
| 23:32.27 | [TK]D-Fender | [18:15][TK]D-Fender"sip set debug on" |
| 23:32.28 | [TK]D-Fender | [18:15][TK]D-Fender"core set verbose 10" |
| 23:32.30 | [TK]D-Fender | [18:15][TK]D-FenderPASTEBIN a complete failed call. |
| 23:33.34 | DevWork_ | FXO is the receiving end, FXS is the sending end. I would plug my old school phone into the FXS. |
| 23:34.56 | [TK]D-Fender | ~fxsfxs |
| 23:35.01 | [TK]D-Fender | ~fxofxs |
| 23:35.01 | infobot | hmm... fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 23:37.10 | DevWork_ | yea we got a alarm system here, needs two phone jacks. |
| 23:37.19 | Neozonz | [TK]D-Fender, http://paste.ubuntu.com/6912000/ |
| 23:37.20 | DevWork_ | Was thinking of just having verizon send in two plain lines. |
| 23:37.48 | Neozonz | grr, config reset verbose 15 |
| 23:37.48 | Neozonz | bah |
| 23:38.54 | DevWork_ | So, what is echo cancellation, and how does it work on the digital cards? |
| 23:39.05 | Neozonz | ah nm, its set correctly |
| 23:39.06 | DevWork_ | Wouldn't you do that kind of thing in software? its confusing. |
| 23:39.16 | Neozonz | [TK]D-Fender, http://paste.ubuntu.com/6912000/ |
| 23:42.05 | [TK]D-Fender | Neozonz: "core set debug 0" <- |
| 23:42.07 | [TK]D-Fender | another call. |
| 23:43.15 | Neozonz | [TK]D-Fender, http://paste.ubuntu.com/6912022/ |
| 23:43.34 | [TK]D-Fender | DevWork_: For an alarm system that'd be just their system on the line... * wouldn't be using it for anything that I can see... |
| 23:44.09 | DevWork_ | Im not sure I follow what you are saying. |
| 23:45.44 | [TK]D-Fender | Neozonz: Contact: <sip:7001@10.245.205.129:5060> <-- your server has a NAT'd IP and is showing a private contact for return. |
| 23:46.01 | [TK]D-Fender | Neozonz: even though they map a public to you it's still technically a private. |
| 23:46.32 | [TK]D-Fender | localnet=10.0.0.0/255.0.0.0 |
| 23:46.53 | [TK]D-Fender | externaddr=54.196.144.109 |
| 23:46.55 | [TK]D-Fender | nat=yes |
| 23:46.59 | [TK]D-Fender | directmedia=no |
| 23:47.11 | [TK]D-Fender | Add these to the bottom of your [general] section of sip.conf |
| 23:47.17 | [TK]D-Fender | reload, retest |
| 23:48.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.230) |
| 23:57.45 | DevWork_ | How does video conferencing work with asterisk and sip type setups? |
| 23:57.59 | Neozonz | [TK]D-Fender, thank you so very much |
| 23:58.09 | [TK]D-Fender | Neozonz: All good now? |
| 23:58.11 | Neozonz | I spent pretty much the entire day on this |
| 23:58.12 | Neozonz | yeah |
| 23:58.27 | Neozonz | would have never figured that out |
| 23:58.42 | Neozonz | maybe a small note on the config might help future users |
| 23:58.56 | Neozonz | Public IPs may still be NAT'd :p |
| 23:59.04 | [TK]D-Fender | Neozonz: What I gave you were what * needs to know about it's own environment. As * talls the other side "hey audio is as 10.1.2.3" that will end in falilure, hence the need to tell it the WAN IP |
| 23:59.05 | Penguin | [TK]D-Fender: its |
| 23:59.28 | [TK]D-Fender | Neozonz: You may have a public IP directed to your server... but it's still NAT'd to get there (DMZ + FW" |
| 23:59.59 | DevWork_ | Yea thats how our DMZ is. |