| 00:31.49 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 00:35.57 | devwork_ | pabelanger, thanks, with that 335/ directory, and some googling on "asterisk voIpProt.server.1.address" I was able to get the phone updated and it connecting to asterisk mighty quick |
| 00:36.11 | devwork_ | the reg-basic and sip-basic wasn't enough |
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| 01:25.14 | *** mode/#asterisk [+o file] by ChanServ |
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| 01:33.10 | MrVamp | hi, can someone guide me how to add avaya phones to an asterisk...im having a hard time here :( |
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| 03:35.37 | linocisco | where could I possibly download previous astricon Presentation powerpoints? |
| 03:46.41 | hellinterim | Hi I'm using SFLPhone to test my asterisk server locally. I'm able to successfully dial hello world dialplan using SRTP DES. When I enable TLS, I get the following error: http://dpaste.com/3R4CSHM/ I can run a sip reload and my self signed cert loads OK. |
| 03:48.45 | hellinterim | My sip.conf: http://dpaste.com/27XAGX7/ |
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| 04:02.21 | hellinterim | When I make the call without TLS enabled I get this in rasterisk: http://dpaste.com/1D1RV4S/ So I guess SRTP is working but tls is not. |
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| 04:28.21 | linocisco | where could I possibly download previous astricon Presentation powerpoints? |
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| 04:58.18 | hellinterim | Ok I spun up a windows VM, used the "blink" softphone and I was able to connect using TLS. I guess SFLphone was just giving me trouble. Disregard my previous questions. |
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| 06:34.53 | hrolf | Hi #asterisk, I'm facing an issue. Issue is that sometimes (randomly) it happens that the calls are ringing at agent end only for 1-3 secs (or less) and then dropped and offfered to another. |
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| 06:39.53 | hrolf | http://forums.asterisk.org/viewtopic.php?f=1&t=90359 |
| 06:40.17 | hrolf | above ^^ I have shared the log file. |
| 06:40.27 | hrolf | here <<< http://pastie.org/9189186 |
| 06:41.37 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 06:42.33 | ageis | any way to increase the delay allowed between DTMF inputs before considering it an extension input?.. i.e. someone could enter ext "100" really slowly and it wont go to ext 1 or 10 |
| 06:44.59 | ageis | with background() |
| 06:54.09 | *** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad) |
| 06:56.49 | fahmad | hello, i frequenlty see that call stuck in asterisk and i need to disconnect them from asterisk console. Can someone give me an idea how can i over come from this issue. Here is my output for `core show channels verbose` http://pastie.org/private/3earhb4l7ju0jngijqfw6a |
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| 07:01.38 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:02.26 | Zogot | ahoyhoy |
| 07:16.32 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 07:20.21 | ChannelZ | ageis: see the TIMEOUT function |
| 10:44.20 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 10:44.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:01.09 | alami | i have setup a confbridge like that http://pastebin.com/e9Z1ZuDp, i can use this conference room with Zoiper as i expected but xlite and blink told me that the 234 pin is wrong |
| 13:01.48 | alami | what's wrong with my script? |
| 13:02.23 | [TK]D-Fender | who says the dialplan is wrong? |
| 13:02.28 | [TK]D-Fender | Clearly the CALL is different |
| 13:02.34 | [TK]D-Fender | You should look at the call in proper detail... |
| 13:02.39 | *** join/#asterisk saxa (1000@84-255-205-164.static.t-2.net) |
| 13:03.10 | Katty | hello my asterisk does not work at all how to fix pls is urgent thx |
| 13:03.25 | Zogot | +1 for amazing question |
| 13:03.27 | file | Katty, I'm gonna be an optimist about this... |
| 13:03.39 | Katty | waits patiently |
| 13:04.14 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 13:06.41 | Gugge | Is it possible to get the request-uri from a dialplan function? |
| 13:06.56 | Gugge | the request-uri on an incoming call that is :) |
| 13:10.16 | [TK]D-Fender | IIRC you can't get the INVITE line directly.... I take it that it is not copied in the "To:" header? |
| 13:12.24 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:14.04 | Katty | invites [TK]D-Fender |
| 13:15.24 | alami | [TK]D-Fender: Sorry i don't understand what did you mean with the call |
| 13:15.56 | alami | i've worked all the time with Zoiper and this dialplan work with Zoiper |
| 13:16.21 | alami | so i don't see why xlite or other softphone don't accexpt the pin |
| 13:16.38 | *** part/#asterisk StaRetji (~vencuga@178.79.6.17) |
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| 13:17.57 | bitwize | Hi! I'm working with an migration project from v1.8 to v11.3.0. It seems that my 11.3.0 setup is way more cpu intentive for some reason. 180 concurrent calls the 1.8 version yeilds ~60% avarge cpu usage, 180 concurrent calls on 11.3.0 yields ~85%. I'm confused, anybody who can point me in the right direction, can I profile * to locate the hotspots? |
| 13:21.27 | Gugge | [TK]D-Fender: correct, the INVITE line is different from the To: header in my case :) |
| 13:22.51 | [TK]D-Fender | alami: * CLI. Show us the actual call debug for both calls. |
| 13:28.23 | Katty | infobot: is factoid a real word |
| 13:28.36 | Katty | infobot: why u no answer?! |
| 13:28.36 | infobot | why not? |
| 13:28.45 | Katty | infobot: yes. |
| 13:28.46 | infobot | it has been said that yes is the opposite of no |
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| 14:17.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:38.28 | saxa | hello |
| 14:38.40 | saxa | I'm getting lots of warnings in the console |
| 14:38.46 | saxa | http://pastebin.com/QVMYpMmN |
| 14:38.53 | saxa | how can I get rid of it ? |
| 14:39.23 | saxa | It seems is due to it cant find the ip6 address, which is the one of my box |
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| 14:50.16 | bmurt | is it possible to utilize memcache or something similar to add a layer of caching to an asterisk+mysql configuration? |
| 14:52.50 | bmurt | basically, we're experiencing a good amount of table fragmentation and trying to reduce that |
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| 14:58.54 | Zogot | bmurt: using realtime? |
| 14:59.51 | bmurt | i believe so |
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| 15:05.33 | *** join/#asterisk talnti (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 15:05.50 | talnti | anyone here have an alaska DID? (907) |
| 15:05.55 | talnti | if so, who's your provider? |
| 15:17.15 | jameswf | talnti: just looked at ours ( no availibility ) for 907. This just means there are non in the pool. If you contact the provider they may be able to order did's for the area code. |
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| 15:57.50 | anonymouz666 | anyone in here using NodeJS with AMI? |
| 15:58.32 | navaismo | not me |
| 15:59.16 | anonymouz666 | what do you use? |
| 16:00.09 | mjordan | danjenkins wrote a library that used NodeJS with AMI: https://github.com/holidayextras/node-asterisk-ami |
| 16:00.44 | anonymouz666 | mjordan: web, i am watching right now his astricon presentation |
| 16:00.47 | anonymouz666 | yeap |
| 16:00.50 | anonymouz666 | heh. |
| 16:00.59 | mjordan | tis a good presentation :-) |
| 16:01.02 | danjenkins | hi mjordan & anonymouz666 |
| 16:01.24 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 16:01.37 | anonymouz666 | hi danjenkins |
| 16:02.45 | anonymouz666 | i just mentioned the NodeJS because after watching your presentation and listen some friends, I am conviced that is the right way to go. Replacing the old approach that uses the python+twisted+starpy etc. |
| 16:02.58 | dan_j | Hi, I'm looking at high availability solutions for use with asterisk. So far I've tried out opensips, but didnt like that the asterisk CLI only showed the IP address of the opensips server. I was wondering what other solutions people have seen mentioned. Has anyone seen anything other than opensips? |
| 16:04.09 | mjordan | anonymouz666: well, NodeJS is pretty nifty, but really those are two similar paradigms: asynchronous, event driven frameworks that focus on scaling out using a non-multithreaded approach |
| 16:04.48 | litn | hey guys- if I do a supervised transfer to someone who is on DND or busy, I get to their voicemail.. if I hangup the call then of course I lose the transfer and I can't hang it up from the other person's line.. |
| 16:04.52 | litn | not sure what to do |
| 16:05.27 | anonymouz666 | NodeJS is much simpler than Twisted as far I can see, it can reduce a lot of lines of code |
| 16:06.07 | anonymouz666 | dan_j: didn't understand your question |
| 16:06.53 | anonymouz666 | i have to lunch |
| 16:06.55 | anonymouz666 | brb |
| 16:08.40 | danjenkins | anonymouz666: yeah, I prersonally prefer Node.js but thats because I’ve been using it for years now |
| 16:09.20 | danjenkins | anonymouz666: I actually have a new version of that code as a beta branch but I left holiday extras before I had a chance to complete it, and just haven’t had time to go back to it |
| 16:09.29 | danjenkins | i will though! |
| 16:09.57 | danjenkins | there are also a few other nodejs libraries out there but mine gives you the most flexibility in my view |
| 16:10.15 | danjenkins | https://www.npmjs.org/search?q=asterisk |
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| 16:23.38 | navaismo | dan_j, You can cretae a peer between asterisk and opensips, what are you trying to do? to see? |
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| 16:40.00 | dan_j | navaismo: when I submit sip show peers, I'd like to see the IP addresses of all peers, as you see when asterisk runs on its own. |
| 16:40.06 | *** join/#asterisk troyt (~troyt@2601:7:6200:1362:44dd:acff:fe85:9c8e) |
| 16:42.36 | navaismo | hmm i think you are kind confused, if you are using opensips its will act as a provider for you like voip.ms flowroute and others do, or what is your goal the description seems messi(GOAL) |
| 16:42.41 | *** join/#asterisk troyt (~troyt@2601:7:6200:1362:44dd:acff:fe85:9c8e) |
| 16:43.24 | dan_j | the goal is to end up with a primary and a backup asterisk server with opensips in front of them. if the primary fails, then opensips switches and sends all peers to the backup server. |
| 16:44.03 | dan_j | it works nicely, but asterisk shows all peers as coming from the opensips server (which i expected) |
| 16:44.14 | dan_j | so i dont really like that solution as its harder to debug issues |
| 16:44.50 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 16:46.07 | navaismo | so why you dont use the heartbeat-pacemaker-drbd solution between 2 asterisk servers with a floating ip? |
| 16:47.39 | dan_j | How fast is the failover from when the first server fails? In the event that asterisk crashes but the server is fine, can it still failover? |
| 16:50.20 | navaismo | i cant tell you exactly how fast, but i saw test with less than 3 minutes |
| 16:51.49 | navaismo | dan_j, https://www.youtube.com/watch?v=kIlnUBho2n4 |
| 16:53.02 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:54.40 | dan_j | whats the software in the top right? |
| 16:57.57 | *** join/#asterisk nabblet (~deep@unaffiliated/benkinooby) |
| 17:00.37 | nabblet | hi, what's the cleanest way to run Asterisk (versionwise) on Debian? 1.8. is in the repositories but 11 is the recommended stable. shoudl i install from source? is there 11 as additional repo available? any version conflicts (tools and asterisk) known? |
| 17:01.18 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:04.47 | navaismo | dan_j, putty |
| 17:06.31 | navaismo | dan_j, or you mean the monitor tool? its the crm_mon utility |
| 17:07.28 | navaismo | that use drbd-pacemaker-heartbeat with asterisk & mysql |
| 17:08.15 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 17:12.26 | pigeonflight | what might cause these in the logs? Registration from '"7274" <sip:7274@72.25x.x13.5>' failed for '5.15x.5x.2x7' - Device does not match ACL |
| 17:14.19 | *** join/#asterisk Ian_AU (~ian_au@203.23.208.92) |
| 17:14.48 | navaismo | that device doesnt match the permit IP/range in the peer settings |
| 17:15.09 | pigeonflight | so this is someone “brute forcing” our system? |
| 17:15.26 | pigeonflight | because they seem to be incrementing the extensions they are testing |
| 17:15.33 | ChannelZ | uh, yeah |
| 17:15.43 | navaismo | yea bruteforce attack |
| 17:15.51 | navaismo | change password and close ports NOW! |
| 17:16.17 | pigeonflight | I inherited this box |
| 17:16.21 | pigeonflight | which password |
| 17:16.24 | pigeonflight | root? |
| 17:16.38 | pigeonflight | can I just block the ip? |
| 17:16.40 | ChannelZ | your passwords are probably fine, it's just some moron probing |
| 17:16.56 | ChannelZ | Just block them |
| 17:17.03 | navaismo | hehe secret=1000 |
| 17:17.18 | navaismo | or secret=7274 |
| 17:17.38 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:20.36 | marceloamorim | Guys, I wish to know if I can set dtmfcidlevel value for each channel I have on my fxo |
| 17:20.45 | *** part/#asterisk Ian_AU (~ian_au@203.23.208.92) |
| 17:20.59 | pigeonflight | thought something like this would be enough “iptables -A INPUT -s 5.153.59.227 -j DROP" |
| 17:22.02 | navaismo | do you need sip/rtp ports to be public? |
| 17:22.15 | ChannelZ-Wk | you're kind, I'd do the whole netblock |
| 17:22.30 | pigeonflight | navaismo: not sure, we want to dial out to sip endpoints |
| 17:23.19 | ChannelZ-Wk | unless you're expecting registrations from the philippines |
| 17:23.24 | pigeonflight | ChannelZ-Wk: I did the drop but still seeing the probe from the same ip address |
| 17:24.01 | ChannelZ-Wk | You might look into fail2ban to take care of this for you |
| 17:25.05 | pigeonflight | yeah, but just wanted to get something in place now |
| 17:25.17 | pigeonflight | then spend 30 minutes on fail2ban |
| 17:35.00 | dan_j | navaismo: ok thanks. i'll take a look. |
| 17:37.51 | dan_j | When using a queue, is it possible to use a Local channel as one of the agents? |
| 17:38.13 | dan_j | The reason being, I want the agent to be able to press 1 to accept the call. |
| 17:40.07 | mjordan | Yes. |
| 17:40.42 | dan_j | Ok. Perfect. Thanks |
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| 17:42.20 | mjordan | dan_j: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id288932.html#ACD_id288989 |
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| 17:47.18 | dan_j | thanks |
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| 19:19.05 | pigeonflight | how do I create an extension that when dials dials a sip address (e.g. mysip@myserver.com:5060)? |
| 19:20.21 | [TK]D-Fender | Dial(SIP/mysip@myserver.com) |
| 19:21.41 | pigeonflight | [TK]D-Fender: okay that’s what I have. Would setting nat=yes help with a call that goes silent when answered? |
| 19:21.48 | pigeonflight | if so, where would I set that? |
| 19:22.05 | [TK]D-Fender | in extensions.conf clearly... |
| 19:22.14 | [TK]D-Fender | extension = extensions.conf |
| 19:22.26 | [TK]D-Fender | oops... |
| 19:22.29 | [TK]D-Fender | missed a line there |
| 19:23.19 | [TK]D-Fender | Your NAT settings should be representative for your server's operating environment |
| 19:23.39 | [TK]D-Fender | You just say "nat=yes" just because.... |
| 19:23.53 | [TK]D-Fender | And if your server is NAT'd then there are SEVERAL settings that have to be made |
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| 19:48.05 | cusco | hi [TK]D-Fender, meanwhile I have setup hints with dialplan that is not used just to have "core show hints" display the sip peers state |
| 19:49.16 | cusco | can Hi get hinted state via dialplan? |
| 19:49.41 | cusco | to show that a queue tries to ring a member that is either paused or inuse |
| 19:55.35 | newtonr | cusco, you might be looking for the HINT function |
| 19:55.38 | newtonr | let me check |
| 19:56.10 | cusco | well.. in 1.6 :/ |
| 19:57.07 | *** part/#asterisk talnti (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 19:57.14 | cusco | ${DEVICE_STATE(SIP/mypeer)} ? |
| 19:57.28 | newtonr | yeah that is the one |
| 19:57.37 | newtonr | HINT grabs a list of devices associated to a particular hint |
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| 19:59.01 | cusco | ok .. well I'm having a weird issue on 1.6 ... where queue is dialing to members that are in use, in dialplan I have |
| 19:59.10 | cusco | if("${DEVICE_STATE(SIP/${dstExt})}" != "NOT_INUSE"){ ... Busy(); ... } |
| 19:59.20 | [TK]D-Fender | cusco: you are still doing this wrong... |
| 19:59.28 | newtonr | cusco, oh but you can also use EXTENSION_STATE to get the extension state (hint) https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_EXTENSION_STATE |
| 19:59.42 | [TK]D-Fender | cusco: the point was to SEE if * was tracking the state properly or not. Not for you to use it from the dialplan |
| 19:59.50 | cusco | ok... |
| 20:00.02 | [TK]D-Fender | this should have been a 5 minute job..... |
| 20:00.11 | cusco | so... I see peers in use and unavailable and idle ... |
| 20:00.24 | cusco | yea, I've been kind of busy with other stuff.. |
| 20:01.15 | cusco | and I have to go now... sorry, I'll be back tomorrow :p |
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| 20:08.01 | fprior | Hi all ! I'm testing hangup_handlers following this page http://goo.gl/SbPHqY ; I need to retrieve SIP Code on Hangup Call. Problem is that I cannot retrieve a variable set in hangup_handler; SHARED function is not functioning. Anyone can help me ? |
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| 20:26.57 | fprior | Anyone can share syntax or example to use MASTER_CHANNEL function ? |
| 20:27.46 | fprior | core show function MASTER_CHANNEL show description but no syntax |
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| 20:28.38 | cmendes0101 | I'm having some issues with Queue() in a dialplan. I'm using the c flag for the dialplan to continue after queue but there are some calls that end right at queue and I cannot grab the MEMBERINTERFACE. Should it be safe to perform this in a hangup context? |
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| 20:30.01 | newtonr | fprior, I don't have time to look at it. but if you end up thinking there is bug with the SHARED function, then file an issue on the tracker and I'll look at it when I can. |
| 20:30.47 | bmurt | dumb question... but does anyone know the format for v1.4 for displaying milliseconds? |
| 20:31.04 | bmurt | i've tried: dateformat=%B %d %H:%M:%S:%3q to no avail |
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| 20:33.04 | Varazir | hi, can you have a e-mailadress as username/Gatewayname ? |
| 20:33.49 | Varazir | to connect to a sip system |
| 20:34.31 | Varazir | please use my nick when you reply |
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| 20:34.46 | fprior | ok newtonr, I don't think is a bug, only I cannot understand how pass variable from on side (hangup_handler) to master channel |
| 20:35.53 | newtonr | fprior, at a glance, the dialplan in your linked example looks correct |
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| 20:54.13 | fprior | newtonr, but ${MASTERCHANNEL} is empty |
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| 20:54.43 | newtonr | fprior, MASTERCHANNEL OR MASTER_CHANNEL ? |
| 20:54.53 | newtonr | isn't one a variable and the other a function? |
| 21:00.06 | newtonr | newtonr, i don't see that MASTERCHANNEL variable being set anywhere and I don't think it is a built-in... |
| 21:00.33 | newtonr | newtonr, i'm about to leave, but you might go through these wiki pages, they might help: https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause, https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
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| 21:08.27 | talnti | if I wanted to make an extension that just rings forever, anyone have any advice on how to do that? besides registering a phone and making it silent and putting it in the corner |
| 21:09.16 | MaliutaLap | talnti: answer it an then playback(ringing) |
| 21:09.17 | [TK]D-Fender | talnti: "core show application ringing" |
| 21:09.30 | MaliutaLap | talnti: depending on how mean you want to be |
| 21:09.52 | talnti | ah ha |
| 21:09.57 | MaliutaLap | answering it to an extension that just plays the ringing tone back to them will cost them money |
| 21:10.04 | navaismo | aja* |
| 21:10.49 | MaliutaLap | sometimes you just need to be a prick about it :) |
| 21:11.53 | bmurt | hmm, [TK]D-Fender quick question.. do you know off-hand what to set dateformat= in v1.4 for displaying milliseconds in logger.conf? |
| 21:13.04 | [TK]D-Fender | bmurt: nope.... |
| 21:13.13 | [TK]D-Fender | 1.4 is decrepit and shouldn't be used..... |
| 21:13.25 | bmurt | legacy shit |
| 21:13.35 | bmurt | its a WIP, but a long process |
| 21:13.47 | bmurt | for now, we're stuck with v1.4 |
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