| 00:05.46 | navaismo | which hardware do you recommend to handle 18 E1s in one server |
| 00:06.57 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.201) |
| 00:08.25 | zopsi | seriously [TK]D-Fender: you don't have to be such an a#@hole ALL the time. Literally just put together all that stuff and you basically tell me that I wasn't fast enough because times up. You give me and my coworkers a bad impression of Digium and Digium's opensource community. Guess some people just want to die alone. |
| 00:08.50 | *** part/#asterisk zopsi (sid22708@gateway/web/irccloud.com/x-hdtcrssxdqehamfy) |
| 00:09.10 | navaismo | too late ^ |
| 00:10.40 | outtolunc | you know fender made a backup of himself that will attend his funeral, right. ;) |
| 00:29.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:29.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:39.06 | jesselang | Hello. I've got Asterisk 1.4 running behind NAT. I'm sending calls to a gateway outside of NAT. How can I get RTP from that gateway back to the Asterisk? Is there a way to traverse NAT? |
| 00:42.30 | DougsTech | forward ports |
| 00:43.00 | DougsTech | asterisk normally works behind NAT though, when connecting to trunks |
| 00:45.07 | jesselang | DougsTech, I would, but I have no control over the firewall. Does Asterisk do any UDP hole punching? |
| 00:45.27 | DougsTech | no |
| 00:45.35 | DougsTech | you mean like upnp? |
| 00:46.27 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 00:47.36 | jesselang | No. I mean just sending a UDP packet, sort of how Skype does it. |
| 00:47.44 | leifmadsen | you can still send a UDP packet |
| 00:48.06 | jesselang | leifmadsen, do tell. |
| 00:48.06 | leifmadsen | typically the router is going to hold open a connection so that the other side can return packets over the port you opened when you established the connection |
| 00:48.45 | leifmadsen | for inbound, the way that happens is with OPTIONS packets via the qualify=yes option for the peer |
| 00:49.15 | jesselang | leifmadsen, I have SIP messages making it back and forth. My issue is one way audio. |
| 00:49.17 | leifmadsen | there are several articles on getting asterisk to work behind NAT though at this point |
| 00:49.26 | jesselang | So RTP is the problem, I think. |
| 00:49.28 | ChannelZ-Wk | use symmetric RTP |
| 00:49.42 | jesselang | Specifically, I need DTMF. |
| 00:49.54 | leifmadsen | well the dtmf is probably carried in the rtp stream |
| 00:50.15 | jesselang | ChannelZ-Wk, how do I use symmetric RTP with Asterisk? I have no control over the firewill, unfortunately. |
| 00:50.20 | jesselang | firewall, rather. |
| 00:51.27 | ChannelZ-Wk | see the 'comedia' option for nat in sip.conf |
| 00:52.10 | jesselang | Is that an option in 1.4? |
| 00:52.26 | ChannelZ-Wk | hmm. I don't think so. |
| 00:53.03 | ChannelZ-Wk | Which way does your audio _not_ work? |
| 00:53.21 | jesselang | From the external gateway into the Asterisk which is NAT'd. |
| 00:54.37 | ChannelZ-Wk | Oh. Well comedia would help anyway. And you have no control over the gateway to port forward yes? |
| 00:54.51 | ChannelZ-Wk | s/would/wouldn't/ |
| 00:55.47 | ChannelZ-Wk | (and I too wonder why there was never an option to do Skype-style NAT traversal) |
| 00:57.02 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.66) |
| 00:57.35 | ruben231 | hi guys good morning |
| 00:58.16 | ChannelZ-Wk | ahoyhoy |
| 01:00.39 | ruben231 | does my dream possible, anyone who have a good heart and can sponsor asterisk basic to advanced training and after learning it, ill be oblige to share adn trained it to others also...i been dreaming this eversince..:-( |
| 01:00.59 | jesselang | ChannelZ-Wk, correct. I have no control over the firewall to do port forwarding. |
| 01:02.09 | DougsTech | ruben231, ask CBT nuggets |
| 01:02.54 | ChannelZ-Wk | jesselang: well you might be a little stuck then particularly with 1.4 |
| 01:03.21 | jesselang | ChannelZ-Wk, would I have to go to 1.8? |
| 01:03.23 | ChannelZ-Wk | ruben231: we're all free whores here. Do you have a specific problem? |
| 01:03.49 | ChannelZ-Wk | well if you were going to go to the trouble to upgrade, might as well go to 11 |
| 01:04.13 | ChannelZ-Wk | However I'm still not sure if it would help your case.. I don't know if any of the ICE and STUN crap will assist when asterisk is behind the firewall. |
| 01:05.38 | ChannelZ-Wk | and comedia is for assisting clients behind NAT |
| 01:06.30 | ruben231 | ChannelZ-Wk: not really, just wanted to be traineed with asterisk from basic to advance- coz i see on our place there so many poor IT graduates that dont have the money to afford expensive training, empowering helpless IT somehow gives oppurtunity for charity. |
| 01:07.03 | *** join/#asterisk sectech (~sectech@stjhnbsu0ww-142134150058.dhcp-dynamic.FibreOP.nb.bellaliant.net) |
| 01:08.22 | ChannelZ-Wk | Mmm. Well I'm certifiable, but not certified in anything. |
| 01:09.15 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 01:10.43 | ChannelZ-Wk | time to go home |
| 01:11.40 | ruben231 | ChannelZ-Wk: nice |
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| 02:11.18 | Nugget | eyes DougsTech |
| 02:11.27 | DougsTech | Nugget, ? |
| 02:12.24 | pabelanger | ~lastseen iroot |
| 02:12.30 | pabelanger | ~help |
| 02:12.46 | pabelanger | ~seen iroot |
| 02:12.50 | infobot | pabelanger: i haven't seen 'iroot' |
| 02:12.51 | DougsTech | Nugget, ah haha, yea I assume you know what CBT is |
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| 03:45.26 | ChannelZ | Cock & Ball Torture |
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| 04:45.39 | bsdice | Americans... |
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| 05:44.38 | jrsharp | hey all |
| 05:44.56 | jrsharp | anyone here particularly familiar with RTSP/RTP? |
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| 05:49.20 | ChannelZ | low attention span |
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| 06:22.00 | *** join/#asterisk hanuman (31cc5610@gateway/web/freenode/ip.49.204.86.16) |
| 06:22.17 | hanuman | hi |
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| 06:31.04 | hanuman | my sip phone is not working with remote asterisk server |
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| 06:49.23 | hanuman | please give reply any one |
| 06:49.52 | [TK]D-Fender | outtolunc: Lovely.. I run out of time and have to go... and suddenly I'm an "asshole"? I had to leave.. not just ditching him.... |
| 06:57.46 | hanuman | Hello |
| 07:08.32 | *** part/#asterisk hanuman (31cc5610@gateway/web/freenode/ip.49.204.86.16) |
| 07:11.51 | *** join/#asterisk hanuman (31cc5610@gateway/web/freenode/ip.49.204.86.16) |
| 07:12.24 | hanuman | sip phone not working with remote asterisk server |
| 07:12.26 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 07:12.39 | fling | Is there a module for voice changing? |
| 07:13.25 | [TK]D-Fender | There used to be. JunK-Y was the author... |
| 07:13.27 | hanuman | sip phone is ringing only |
| 07:13.33 | [TK]D-Fender | Not sure if it's ported and current |
| 07:13.33 | hanuman | not answering |
| 07:14.16 | hanuman | but that sip phone is registered |
| 07:14.53 | hanuman | and agi set debug on that IVR flow working fine when i am calling from my sip phone |
| 07:20.14 | [TK]D-Fender | heads to bed.... |
| 07:22.21 | *** join/#asterisk qoottaa (~ololo@83.69.225.78) |
| 07:24.11 | fling | https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT |
| 07:26.29 | *** join/#asterisk Ian_AU (~ian_au@203.23.208.92) |
| 07:29.09 | hanuman | sip phone is registered but not responding with remote asterisk server |
| 07:32.28 | *** join/#asterisk moshin (~mo@udp001682uds.hawaiiantel.net) |
| 07:32.41 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:35.35 | ChannelZ | Sleepy NAT |
| 07:38.06 | hanuman | ChannelZ@i am not configured NAT |
| 07:38.22 | hanuman | ChannelZ: i am not configured NAT |
| 07:38.42 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:39.00 | ChannelZ | then define "not responding" |
| 07:40.42 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:40.46 | fling | exten => hanuman,1,Answer() |
| 07:42.45 | hanuman | fling i written that in dialplan |
| 07:43.28 | fling | ChannelZ: :> |
| 07:45.08 | hanuman | fling: sip registered with my asterisk server and IVR flow also working when i am calling to that extention but my sip phone not responding only ringing |
| 07:51.37 | *** join/#asterisk hanuman (31cc5610@gateway/web/freenode/ip.49.204.86.16) |
| 07:52.58 | ChannelZ | What do you mean it's "not responding only ringing" - the phone is ringing? Capture a sip debug of the call and pastebin it |
| 07:52.59 | ChannelZ | ~pb |
| 07:53.00 | 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. |
| 08:00.21 | hanuman | ChannelZ: sip not working with remote server. this is pastebin name for sip debug |
| 08:01.38 | hanuman | ChannelZ: http://pastebin.com/u7kzCVvr |
| 08:02.27 | ChannelZ | ok that's not enough, that shows me nothing |
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| 08:05.23 | hanuman | ChannelZ: http://pastebin.com/yxkpY8da |
| 08:06.24 | ChannelZ | ok these are still only a couple of OPTIONS packets, not a call attempt showing anything happening |
| 08:09.26 | hanuman | ChannelZ: http://pastebin.com/00jaZU77 |
| 08:10.05 | *** join/#asterisk _omer (~omer@116.71.182.246) |
| 08:10.46 | ChannelZ | Retransmitting #1 (NAT) to 49.204.86.16:5061 |
| 08:11.53 | _omer | I installed Asterisk 11.7.0 yesterday and it was working fine but today it gives error "illegal instruction" ... Tried to find solution on google but could not find a proper solution..... I have uninstalled asterisk completely and going to install it again..... Is there anything that I should care about before installing asterisk 11.7.0 again to avoid "illegal instruction" error again ? |
| 08:12.28 | ChannelZ | There's some sort of communication issue with that peer.. interestingly it responds to the lack of auth on the initial INVITE but then after sending the followup auth INVITE it seems to go braindead |
| 08:12.57 | ChannelZ | _omer: you get that when, trying to start it or it crashes somewhere while running? |
| 08:13.14 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 08:13.28 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:13.49 | _omer | yesterday, asterisk installed fine and it was running .. even I connected to Asterisk CLI .... but today when I tried to connect to CLI (asterisk -r) then I got error "illegal instruction" |
| 08:14.01 | hanuman | ChannelZ: what is the problem, how can i rectify that |
| 08:14.01 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:14.47 | _omer | But ... I have removed asterisk completely and going to install it again.....so I need help to avoid such issue this time. |
| 08:14.48 | kaldemar | hanuman: your phone is behind a nat. either the nat is causing disruptions in the dialog or twinkle isn't working. |
| 08:14.50 | ChannelZ | hanuman: I don't know off hand.. I don't know if Twinkle is broken or if the packets suddenly stop making it off your network and never make it to that machine, or what. |
| 08:16.12 | ChannelZ | _omer: not sure. What distro are you on |
| 08:16.43 | _omer | centos 6 |
| 08:16.56 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3459:e5e0:b0ef:d3ae) |
| 08:17.33 | _omer | by the way BUILD_NATIVE compiler flag is XXX in this system but it is active in my another system where I have already installed asterisk 11.3.0 successfully. |
| 08:17.51 | _omer | What package do I need to activate BUILD_NATIVE for centos 6 |
| 08:17.52 | _omer | ? |
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| 08:26.34 | *** join/#asterisk _omer (~omer@116.71.182.246) |
| 08:26.37 | _omer | hello |
| 08:26.59 | _omer | what package do I need to activate BUILD_NATIVE compiler flag in centos 6 |
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| 10:24.45 | Ice_Strike | I am seeing this in CLI |
| 10:24.48 | Ice_Strike | <PROTECTED> |
| 10:24.55 | Ice_Strike | and flooding with that |
| 10:24.59 | Ice_Strike | Where is that coming from? |
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| 10:38.13 | mirela666 | from WaitForSilence app |
| 10:39.24 | mirela666 | Ice_Strike: is it increasing for 20ms |
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| 10:48.35 | Ice_Strike | I am getting told that customers cant hear the agents properly but agents can hear them well |
| 10:48.40 | Ice_Strike | not sure what might be causing it |
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| 11:08.36 | k3asd` | hi |
| 11:16.51 | mirela666 | hi |
| 11:17.19 | mirela666 | damn, how is encripted RTP suported (read configured) on asterisk 11.7 ? |
| 11:30.38 | mirela666 | should only be encryption=yes |
| 11:31.04 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 11:33.58 | mirela666 | thanks kal |
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| 12:32.27 | cusco | hey folks |
| 12:32.42 | cusco | I'm looking for the first time at lockcdr() and resetcdr() |
| 12:32.58 | cusco | I tried both, seems I'm not getting what I expected.. |
| 12:33.10 | cusco | in a dialplan I have several Dial() |
| 12:33.26 | cusco | and if(${DIALSTATUS}!=ANSWER){ DIAL() } |
| 12:33.37 | cusco | so it will try 4 destinations |
| 12:34.04 | cusco | and before each Dial() I tried ForcCDR(wv) |
| 12:34.11 | cusco | sorry |
| 12:34.16 | cusco | ResetCDR(wv) |
| 12:34.26 | cusco | and was expecting a cdr entry per each dial |
| 12:34.36 | cusco | but I only get one... for the whole call... |
| 12:43.38 | mirela666 | cusco: I don't think it is ment to store multiple cdrs, but to reset CDR() values and cloecct new ones after the app is called |
| 12:43.45 | mirela666 | collect* |
| 12:46.51 | mirela666 | Asterisk UNKNOWN__and_probably_unsupported built by root @ .. |
| 12:49.36 | cusco | mirela666: right, I also tried ForkCDR(aDev) |
| 12:53.59 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 12:54.09 | mirela666 | cusco: make some custom cdr store DB with odbc or REALTIME functions |
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| 13:14.14 | cusco | mirela666: nope, I rather not |
| 13:14.30 | cusco | used sort of that.. in the past, not good |
| 13:15.01 | cusco | if you need to change something in the table, add a field, change name, change structure, it changes in next asterisk version etc... |
| 13:15.05 | cusco | breaks functionality |
| 13:16.21 | cusco | can't I just force current CDR to write, and have a new one ready? |
| 13:16.29 | cusco | with ForkCDR() ? |
| 13:17.00 | cusco | "e - End the original CDR. Do this after all the necessary data is copied from the original CDR to the new forked CDR." |
| 13:17.03 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 13:17.11 | cusco | shouldn't this option do just that? |
| 13:33.45 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:36.08 | *** join/#asterisk hanuman (31cc5610@gateway/web/freenode/ip.49.204.86.16) |
| 13:36.56 | hanuman | my linux sip phones are not making sound but windows sip phone is working what is the problem |
| 13:37.41 | *** part/#asterisk qoottaa (~ololo@83.69.225.78) |
| 13:38.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:40.19 | cusco | [TK]D-Fender: I was asking, and now you joined, so I'm asking again.. |
| 13:40.41 | cusco | I have a dialplan that performs several Dial(); and before each I have ForkCDR(e) |
| 13:40.52 | cusco | shouldn't this create a cdr entry for each dial? |
| 13:41.00 | cusco | "e - End the original CDR. Do this after all the necessary data is copied from the original CDR to the new forked CDR." |
| 13:41.30 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:43.31 | [TK]D-Fender | cusco: There is a general CDR option that says if CDR is actually recorded if the call is actually answered or not. |
| 13:43.33 | [TK]D-Fender | cusco: Check that |
| 13:44.21 | cusco | [TK]D-Fender: you mean the logunansweredcalls=yes ? |
| 13:44.33 | cusco | actually: unanswered=yes |
| 13:45.16 | cusco | in the dialplan I am actually dialign 4 different destinations, having if(${DIALSTATUS} != ANSWER) to dial the next one |
| 13:45.33 | cusco | and the goal is to have a cdr entry for each dial... |
| 13:45.50 | cusco | so I could see: busy, busy, answer ... for instance |
| 13:46.50 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:48.00 | [TK]D-Fender | shouldn't need to check the dialstatus... only reason to be continuing on is because it was not answered |
| 13:48.49 | Ice_Strike | [TK]D-Fender I am having a bit of problem. *Sometime* when agents making outbund calls, the caller can't the agents properly but agents can hear the caller well. |
| 13:49.06 | Ice_Strike | What the issue might be? |
| 13:49.17 | cusco | normally I have the g option in Dial, but in this case I don't need it.. right.. I don't need to evaluate the dialstatus ... |
| 13:49.19 | [TK]D-Fender | provider |
| 13:49.44 | cusco | but still, to have a cdr record for each attempt... I'm right by looking at ForkCDR() ? |
| 13:49.50 | [TK]D-Fender | (in cases of volume, line noise, etc) |
| 13:50.05 | Ice_Strike | Yes its like a low volume |
| 13:50.10 | [TK]D-Fender | cusco: I *think* so, but I've never used it personally |
| 13:50.15 | cusco | ow, ok |
| 13:50.17 | [TK]D-Fender | Ice_Strike: Provider |
| 13:50.52 | cusco | those sometimes.. always go trough the same dialplan? all the callers have the same nat options, and are alwyas in the localnet or external? |
| 13:51.34 | [TK]D-Fender | the only thing that makes volume low is conversion at the point of termination |
| 13:51.48 | [TK]D-Fender | NAT != quiet unless your "quiet" is NO sound |
| 13:51.56 | cusco | aw.. sorry |
| 13:54.45 | Ice_Strike | [TK]D-Fender this is what it look like http://pastebin.com/kmKf8KpP |
| 13:55.01 | [TK]D-Fender | Ice_Strike: your peer has nothing to do with this. |
| 13:55.13 | [TK]D-Fender | Ice_Strike: Low volume is THEIR issue |
| 13:55.17 | Ice_Strike | Ah |
| 14:21.14 | Katty | morning |
| 14:22.53 | [TK]D-Fender | Good local low solar azimuth to you too... |
| 14:24.11 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:24.11 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:25.04 | Ice_Strike | How well does asterisk run on a VM via Vmware ESXi? |
| 14:25.15 | Ice_Strike | Let say 80 concurrents call |
| 14:25.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.161) |
| 14:25.35 | mirela666 | Ice_Strike: wel enough |
| 14:25.37 | eric_hill | Ice_Strike: Depends on your VMware infrastructure. If you reserve CPU and RAM, just fine. |
| 14:26.07 | Ice_Strike | eric_hill Using Vmware ESXi is a great advantage for failover |
| 14:26.10 | mirela666 | if you give enough procesors and memory, but not needed much |
| 14:26.14 | Ice_Strike | so asterisk could still run. |
| 14:26.28 | *** join/#asterisk mjordan (~matt@nat/digium/x-gkflrqmgsjbergxj) |
| 14:26.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:27.09 | Ice_Strike | I was told there is a bit of network latency issue using asterisk with ESXi |
| 14:27.17 | eric_hill | I'm installing a system this weekend using asterisk on VMware. |
| 14:27.23 | eric_hill | DRs works great |
| 14:27.43 | Ice_Strike | Did you setup or plan to setup fail over? |
| 14:27.44 | Greenlight | We run a load of boxes under ESXi |
| 14:27.49 | Ice_Strike | like what if asterisk goes dwon |
| 14:27.50 | Ice_Strike | down |
| 14:27.57 | Ice_Strike | Greenlight Thats nice. |
| 14:28.10 | Greenlight | No issues. |
| 14:28.34 | mirela666 | run it with safe_asterisk |
| 14:28.35 | eric_hill | I have the VM set for monitoring by VMware and if it goes offline, another host will bring it up, yes. That's built in to VMware advanced licenses. |
| 14:28.38 | Ice_Strike | I currently using ESXi at work for Windows Server 2012 with two DC's |
| 14:28.46 | Ice_Strike | Might install Asterisk and use second NIC |
| 14:28.59 | Greenlight | Two DC's on the same ESXi box ? |
| 14:29.03 | Ice_Strike | Yep |
| 14:29.06 | Greenlight | lol |
| 14:29.09 | Ice_Strike | I know lol |
| 14:29.26 | Greenlight | Like, why |
| 14:29.33 | Ice_Strike | Second DC for files shares and AD. |
| 14:29.53 | Greenlight | Usually the idea of a second DC is for redundancy |
| 14:29.59 | Ice_Strike | Yep. |
| 14:30.04 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 14:30.23 | Ice_Strike | If I get more resources, then I will install on second ESXi |
| 14:31.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.161) |
| 14:31.23 | Ice_Strike | Greenlight What are you using to backup VM's on ESXi |
| 14:36.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.161) |
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| 14:57.14 | *** join/#asterisk sjs205 (~sjs205@host86-186-243-61.range86-186.btcentralplus.com) |
| 14:58.29 | sjs205 | Hello all, I've got a problem with my server, it seems the server gets a particular failed registration two of my lines ring... I'm trying to debug now, but does this sound familar to anyone? |
| 14:58.48 | sjs205 | *when the server |
| 15:01.21 | WIMPy | Sounds very familiar. |
| 15:01.40 | WIMPy | It's the usual situation of not enough information to give any sensible comment. |
| 15:02.08 | *** join/#asterisk Chotaire (chotaire@host-089-207-249-134.vipri.net) |
| 15:02.16 | sjs205 | WIMPy, haha... sorry... |
| 15:03.04 | sjs205 | Basically, I have two phones connected to a remote server, every now and then the phones will ring but there will be no-one on the other side of the call. |
| 15:03.26 | Katty | sjs205: that means he wants logs. |
| 15:03.35 | Katty | sjs205: cause the proof is in the pudding, buttercup |
| 15:03.41 | sjs205 | I have checked the server with sip debug off and it seems that occasionally this happens when there is a failed registration attempt... |
| 15:04.10 | sjs205 | I have now set sip debug on, and am watching the local network with a sniffer... |
| 15:04.31 | sjs205 | I was just wondering whether this ould be some sort of hacking attempt??? |
| 15:04.53 | Katty | where's fender bender this morning? |
| 15:04.55 | sjs205 | or anything sinister... or ideally, even just something I've setup incorrectly! |
| 15:05.03 | Katty | pokes at [TK]D-Fender |
| 15:05.30 | mirela666 | was here hour ago |
| 15:06.37 | [TK]D-Fender | prods katty |
| 15:06.48 | Katty | he lives! |
| 15:07.13 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:07.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:07.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:08.30 | navaismo | Morning people! Anyone can recommend hardware or tips to configure 18 E1's in one PBX, 9 are from one provider and the other 9 from another provider, or I should use Xorcom Channels banks |
| 15:08.51 | Katty | good morning navaismo |
| 15:09.16 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-jjjyobxntmqxnctm) |
| 15:09.16 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:09.44 | *** join/#asterisk sp00kz_ (~ilubj00@unaffiliated/sp00kz) |
| 15:09.45 | WIMPy | navaismo: Not much choice there. AFAIK 8 PRI cards are only available form Sangoma and Digium. |
| 15:09.46 | [TK]D-Fender | navaismo: The term channel bank tends to be reserved for converting multiplexed digital to individual analog (POTS) |
| 15:10.04 | WIMPy | Unless Sangoma has made their E3 card usable for voice. |
| 15:10.14 | [TK]D-Fender | WIMPy: Nope, still not channelized |
| 15:11.02 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 15:11.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:11.22 | Katty | morning russellb |
| 15:11.23 | WIMPy | And you shouldn't put lines from different telcos to the same card. so that makes a minimum of 4 cards. |
| 15:11.27 | russellb | o/ |
| 15:11.31 | russellb | haven't been in here in ages :) |
| 15:11.40 | WIMPy | Are you sure you want them all on the same server? |
| 15:11.46 | Katty | navaismo: i think i'd be kinda tempted to register a bunch of other asterisk boxes with the main one |
| 15:11.52 | Katty | navaismo: i then forward them out |
| 15:12.14 | Katty | navaismo: but that's just me |
| 15:12.31 | navaismo | so ill need 3 cards, do i need to consider any in the timing settings when the two providers mix in the same card? |
| 15:13.36 | navaismo | Katty, hmm asterisk acting as a PRI gateway? |
| 15:14.36 | Katty | navaismo: yeah |
| 15:14.53 | Katty | navaismo: just based on the DIDs, send it somewhere else |
| 15:14.57 | Katty | navaismo: to some other registered server |
| 15:15.11 | Katty | navaismo: like multi-tenant |
| 15:15.25 | Katty | there's probably some fancy word or phrase for it |
| 15:16.02 | navaismo | yeah i can consider that scenario, but my concern is mixing two different providers in the same card, you know the timing stuff |
| 15:16.26 | Katty | well you could always seperate them out if you have problems |
| 15:16.41 | Katty | do the same thing on each box |
| 15:16.57 | WIMPy | navaismo: Did you read what I wrote on that topic? |
| 15:16.59 | Katty | break it down by DID and forward it to some other server |
| 15:17.04 | WIMPy | Don't |
| 15:17.41 | navaismo | Yep but we are poor lol |
| 15:18.07 | WIMPy | Not yet. |
| 15:18.09 | mirela666 | less fortunate :D |
| 15:18.14 | WIMPy | But maybe after you order the hardware :-) |
| 15:18.43 | WIMPy | Or downgrade from 9 to 8 lines per telco. |
| 15:19.11 | Katty | navaismo: i hear ya |
| 15:19.35 | navaismo | :'( |
| 15:20.34 | sjs205 | Okay, a little more information: I am monitoring my local network where the phone is located with wireshark. My phone starts ringing and the only packet I see is one incoming INVITE packet, the wireshark info for the packet say unknown RTP version 1... |
| 15:20.46 | Katty | navaismo: have you found that pris are cheaper than "SIP" trunks? |
| 15:20.58 | sjs205 | Does that mean anything? sounds like someone is trying to ring my phone but I have no idea why! |
| 15:21.42 | [TK]D-Fender | INVITE != RTP |
| 15:23.35 | sjs205 | [TK]D-Fender, sorry, I know that... I'll pastebin the packet... one sec... |
| 15:24.22 | navaismo | Katty, talking about the network requirements and the cost of the internet here, yes is better a PRI than a SIP provider |
| 15:26.04 | Katty | nods |
| 15:28.57 | cusco | We had troubles using 2 telcos in one PRI card... :( |
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| 15:32.23 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
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| 15:33.54 | *** join/#asterisk wizbit (~wizbit@unaffiliated/wizbit) |
| 15:34.10 | wizbit | is this the best way to manage asterisk over http: https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API |
| 15:34.28 | [TK]D-Fender | wizbit: "manage" is a very vague term |
| 15:34.44 | wizbit | do basic stuff like check voicemail ,etc |
| 15:34.46 | wizbit | nothing too special |
| 15:35.08 | [TK]D-Fender | wizbit: clarify "check voicemail" |
| 15:35.24 | [TK]D-Fender | wizbit: You could use AMI to see if there IS voicemail waiting for a specific box. |
| 15:35.26 | wizbit | if somebody leaves a message, i could log on remotely via http and check it |
| 15:35.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.161) |
| 15:35.45 | wizbit | *listen to it |
| 15:35.46 | [TK]D-Fender | wizbit: it is not a means of configuring anything though. Or retreiving VM, etc |
| 15:35.55 | wizbit | ok |
| 15:36.01 | [TK]D-Fender | wizbit: No, AMI is not a generic interfact to do "whatever". |
| 15:36.26 | [TK]D-Fender | wizbit: it is for general call flow and status polling, live events, etc. |
| 15:36.28 | [TK]D-Fender | ~book |
| 15:36.28 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:36.30 | [TK]D-Fender | ^^^^ |
| 15:36.43 | [TK]D-Fender | And look on the WIKI for the list of commands and & events |
| 15:36.51 | [TK]D-Fender | What it does is quite evident |
| 15:36.54 | wizbit | ok but it will allow me to listen to a voicemail message or view missed calls? |
| 15:36.58 | [TK]D-Fender | no |
| 15:37.02 | wizbit | aye ok |
| 15:37.14 | [TK]D-Fender | calls are in logs |
| 15:37.22 | [TK]D-Fender | You want some sort of managing GUI. |
| 15:37.29 | wizbit | [TK]D-Fender: are there any decent web interfaces out there for asterisk or should i avoid? |
| 15:38.11 | [TK]D-Fender | wizbit: I think there is one for voicemails, and others for CDR but that might not be the best way to isolate missed calls |
| 15:38.21 | [TK]D-Fender | wizbit: You're likely going to have to make your own/. |
| 15:38.31 | wizbit | ok i will avoid then! |
| 15:38.37 | wizbit | will get asterisk to email me instead :) |
| 15:39.06 | wizbit | if there is a missed call i think asterisk could be setup to email me what call i missed? |
| 15:39.12 | wizbit | or if i recieve a new voice mail message |
| 15:42.51 | mirela666 | wizbit: for VM mailing read voicemail.conf email part |
| 15:43.19 | [TK]D-Fender | [10:39]wizbitif there is a missed call i think asterisk could be setup to email me what call i missed? <- you could do this. that flow is for you to invent. It isn't part of * |
| 15:43.27 | mirela666 | wizbit: ofcours if you configured mail deamon |
| 15:44.11 | wizbit | ace |
| 15:44.39 | *** join/#asterisk drjfreeze (~Jim@207.191.114.82) |
| 15:45.08 | drjfreeze | I am trying to forward a call inbound from a PRI to an external number |
| 15:46.12 | drjfreeze | I have a simple Dial(Dahdi/g1/15125551111) command, but keep getting the message that circuits are busy |
| 15:46.39 | drjfreeze | I do the same for the internal extensions. This should work for an inbound call, shouldn't it? |
| 15:46.53 | [TK]D-Fender | All calls are inbound |
| 15:47.01 | mirela666 | wizbit: there are plenty of examples , cross platform, for call notification http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 15:47.03 | [TK]D-Fender | Go look at the actual call |
| 15:47.10 | wizbit | ace |
| 15:47.28 | mirela666 | wizbit: modify and on no aswer send a notification |
| 15:47.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.161) |
| 15:47.58 | wizbit | wow |
| 15:47.59 | wizbit | https://code.google.com/p/xbmc-pbx-addon/ |
| 15:48.51 | mirela666 | Works with both Asterisk 1.4 and Asterisk 1.6; |
| 15:48.56 | mirela666 | hmm |
| 15:48.57 | WIMPy | drjfreeze: Do you have some service enabled in your configuration that isn't enabled on your line? |
| 15:49.13 | wizbit | is there a stable release of 1.6? |
| 15:49.29 | [TK]D-Fender | No |
| 15:49.33 | [TK]D-Fender | 1.6.x = dead |
| 15:49.33 | wizbit | eeek |
| 15:49.39 | mirela666 | lol |
| 15:49.44 | wizbit | so it would be insecure for me to use it? |
| 15:49.47 | [TK]D-Fender | yes |
| 15:49.53 | wizbit | im sure that will work on 1.8 |
| 15:50.29 | drjfreeze | -- Channel 0/5, span 1 got hangup request, cause 41 |
| 15:50.57 | [TK]D-Fender | drjfreeze: pastebin the entire call |
| 15:50.59 | [TK]D-Fender | ~pb |
| 15:50.59 | 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. |
| 15:51.01 | [TK]D-Fender | ^^^ |
| 15:51.45 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.182.252.tbinet.bm) |
| 15:51.52 | [TK]D-Fender | Cause No. 41 - temporary failure. This cause indicates that the network is not functioning correctly and that the condition is no likely to last a long period of time; e.g., the user may wish to try another call attempt almost immediately. |
| 15:53.08 | drjfreeze | [TK]D-Fender: http://pastebin.com/fYqiZ1C8 |
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| 15:53.38 | *** join/#asterisk merbanan (~benjamin@h217-27-188-82.cust.tyfon.se) |
| 15:54.13 | drjfreeze | WIMPy: what type of service are you thinking? |
| 15:54.17 | merbanan | how do you generate a dialplan extension trigger based on an event in a channel |
| 15:54.39 | [TK]D-Fender | merbanan: as in? |
| 15:55.37 | merbanan | some kind of trigger based on chan_sip register status |
| 15:55.43 | WIMPy | drjfreeze: Transfer/CD |
| 15:56.13 | merbanan | there is supposed to be functionality for it |
| 15:56.32 | merbanan | just not hooked up to chan_sip registry events |
| 15:56.50 | [TK]D-Fender | merbanan: that is more like a status check than a "trigger" |
| 15:57.46 | [TK]D-Fender | Merthe difference in the approach to handling each would be very different |
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| 16:00.39 | merbanan | [TK]D-Fender: well call it what you like, when something happens to the chan_sip registry state, I want an extension in the dialplan called |
| 16:00.54 | [TK]D-Fender | merbanan: the difference is critical... |
| 16:01.05 | [TK]D-Fender | merbanan: Now that you've clarified... |
| 16:02.01 | [TK]D-Fender | merbanan: You could poll on a continuing basis (AMI) and on change Originate one yourself |
| 16:02.36 | merbanan | there is an api for it |
| 16:02.37 | [TK]D-Fender | merbanan: Or use REGEXTEN in your peers and look for the dynamic dialplan addition AMI Event |
| 16:02.50 | merbanan | that sounds more like it |
| 16:03.07 | [TK]D-Fender | merbanan: Then again RE-REGISTRATION may not flag the way you hope... |
| 16:03.21 | *** join/#asterisk JeffC_NN (~jcollell@mail.nownerd.com) |
| 16:03.26 | [TK]D-Fender | merbanan: and could trigger tons of "not new" events |
| 16:03.51 | Katty | tangles [TK]D-Fender up in a ball of yarn |
| 16:06.22 | [TK]D-Fender | knits a blanket |
| 16:06.38 | Katty | yay! now i don't have to :P |
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| 17:42.04 | gmalsack | does anyone have any idea as to why asterisk 11.6 using odbc sipppeers may repeatedly report wrong password for a sip peer, then after a sip prune realtime all the peer is able to register, make a few calls, then start getting wrong password again |
| 17:46.58 | gmalsack | nevermind I think I may have figured it out. astersik is actually sending a 403 forbidden, which I'm thinking is because the phone is trying to re-register prior to the asterisk minimal registration period.... sound right? |
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| 17:49.54 | newtonr | gmalsack, sounds plausible. I've been running realtime sippeers with ODBC/MySQL for a few days and haven't run into that. Though I haven't messed with registration min/max times |
| 17:50.36 | [TK]D-Fender | can't imagine you should ever have an auth on re-registration like that |
| 17:51.08 | rrittgarn | gmalsack, from my experience it loads the peer into memory, if you load in a wrong password or something the phone thinks is a comment, then change the DB after it tries to register, it will cause issues |
| 17:51.09 | [TK]D-Fender | AFAIK every reg is just a reg and * should accept it the way regardless of the previous not being expired |
| 17:51.26 | gmalsack | newtonr: just set the register expiration time back to default on the phones.... ;-) |
| 17:51.49 | gmalsack | rrittgarn: no db changes during this process.... |
| 17:52.59 | gmalsack | [TK]D-Fender: kind of what I was thinking initially. especially since the phone not making calls seems to be sending registration requests just as often but not having a problem, the problem seems to follow the phone that was making the calls..... :-( |
| 17:53.42 | gmalsack | I suppose it could be an issue with the $39.00 test phones I'm using... lol ;-) |
| 17:53.51 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:54.28 | igcewieling | Has anyone gotten HANGUPCAUSE_KEYS not return an empty string? |
| 17:57.08 | gmalsack | nope. doesn't seem to be the phone. just factory reset the phone and asterisk still throwing a wrong password response |
| 17:57.52 | gmalsack | is there anyway to see what asterisk is comparing that's causing it to send wrong password? |
| 18:02.00 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 18:02.40 | gmalsack | ok just ran sip prune peer 7006 and phone registered just fine.... |
| 18:03.08 | gmalsack | '/SIP/Registry/7006 : 172.21.2.84:5060:3600:7006:sip:7006@172.21.2.84:5060' |
| 18:03.27 | gmalsack | can anyone tell me what the 3600 in the database value string? |
| 18:04.34 | newtonr | max registration expire I think |
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| 19:03.55 | igcewieling | Has anyone gotten HANGUPCAUSE_KEYS to work? |
| 19:08.45 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 19:11.54 | anonymouz666 | it seems to work here |
| 19:11.56 | anonymouz666 | wiki example |
| 19:11.57 | anonymouz666 | asterisk 11 |
| 19:12.02 | anonymouz666 | did you try it? |
| 19:21.07 | igcewieling | anonymouz666: yes. verbose HANGUPCAUSE_KEYS='${HANGUPCAUSE_KEYS}' displays "HANGUPCAUSE_KEYS=''" |
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| 19:34.20 | igcewieling | anonymouz666: do you have a link to the "wiki example" voip-info does not have a page covering it and wiki.asterisk.org does not have an example on the page for hangupcause_keys. |
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| 19:55.30 | mjordan | igcewieling: https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
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| 19:57.04 | igcewieling | mjordan: that info might be useful on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_HANGUPCAUSE_KEYS too. 8-| |
| 19:58.18 | mjordan | igcewieling: that page is part of the reference information. They describe what the functions/applications are and the correct syntax for calling them. Other pages in the wiki describe how to go use that particular item. |
| 19:58.52 | mjordan | and, if I search for HANGUPCAUSE_KEYS, the first return from the wiki after reference pages is the page I linked |
| 20:01.21 | igcewieling | mjordan: I must have assumed the reference would have examples (like show application X or core show function X sometimes does). Thanks for the link |
| 20:04.56 | Katty | do i want chicken nuggets or lumpia for dinner |
| 20:05.13 | *** join/#asterisk eric_hill (eric_hill@wsip-184-180-163-58.ks.ks.cox.net) |
| 20:05.45 | eric_hill | Does anyone on here do freelance asterisk C coding? I'd like to fund a feature. |
| 20:06.02 | Katty | watches everyone disappear into the shrubberies |
| 20:06.09 | eric_hill | exactly. |
| 20:06.13 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:06.17 | WIMPy | eric_hill: Describe what you want, maybe someone is interested. |
| 20:06.26 | Katty | keyword being Maybe |
| 20:06.40 | eric_hill | Polycom phones have a native Call Park feature that uses a special SIP INVITE to park an active call. |
| 20:07.22 | eric_hill | I'd like to fund asterisk to receive that SIP message, park the call, and reply with a SIP message including the 399 Warning header to indicate the parking spot to the phone. |
| 20:08.05 | WIMPy | never understood what that parking stuff could be good for. |
| 20:08.27 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 20:08.38 | eric_hill | thinks it's very common in manufacturing companies where not everyone even HAS a phone. |
| 20:08.55 | eric_hill | "Bob Smith, you have a call holding on 701." |
| 20:09.08 | eric_hill | Bob goes to some nearby phone and dials 701. |
| 20:09.16 | chuckf | wonders what lumpia is |
| 20:09.51 | WIMPy | Seriousely? People are cheaper than phones in your area? |
| 20:10.10 | eric_hill | I can't have phones in certain explosion-prone areas. |
| 20:10.39 | eric_hill | Audio can come in from another area through a pipe. |
| 20:10.54 | eric_hill | Think speaker on the end of a PVC pipe with positive airflow. |
| 20:11.10 | WIMPy | Ok, that's finally a use that makes sense. |
| 20:11.38 | eric_hill | We have three paint booths where this is an issue for us. |
| 20:11.44 | chuckf | Katty: if its takeout, either. If the boy is cooking one of the two, its that one. |
| 20:13.31 | WIMPy | (or can make) |
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| 20:17.16 | *** join/#asterisk mirela666 (~mirko.bra@95.180.59.116) |
| 20:17.59 | chuckf | WIMPy: parking is great for factory floors, warehouses, or similar areas where you have 50 employees and 10 calls a week between them |
| 20:20.20 | anonymouz666 | is it possible in AMI the flow of events (due the nature of threads) sometimes could arrive in bigger chunks? normal flow: event -> event -> event event -> event event event -> event -> event |
| 20:20.30 | anonymouz666 | hope someone understand what I tried to say |
| 20:21.10 | WIMPy | They arrive when they arrive. I guess that means I didn;t understand the question. |
| 20:21.38 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 20:22.15 | anonymouz666 | let's say the normal flow you got 1 event per packet. is it possible due some delay of threads i got 15 events per packet? somethink like that |
| 20:22.43 | WIMPy | Off course. |
| 20:23.15 | anonymouz666 | due the load, the behaviour could change and the client listening to 1 event per packet is not so robust that can't handle 15 events per packet. |
| 20:23.27 | anonymouz666 | the got disconnected from AMI |
| 20:23.55 | anonymouz666 | of course the numbers 1 and 15 are examples, but I expected something like that happening with a PHP client. |
| 20:24.10 | WIMPy | I think you can get disconnected if the queue becomes too large. |
| 20:24.32 | anonymouz666 | you mean eventq? |
| 20:25.00 | WIMPy | The one on the socket. |
| 20:25.19 | WIMPy | I.e. if it's that congested that Asterisk fails to send. |
| 20:29.13 | *** join/#asterisk Beanie (Beanie@2.223.134.254) |
| 20:29.50 | Katty | chuckf: i plan on cooking. |
| 20:30.08 | anonymouz666 | nice and that could be watched looking into Recv-Q and Send-Q using netstat |
| 20:30.22 | WIMPy | yes |
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| 20:44.31 | monsterco | Hi everyone - wondering if anyone knows why FolloMe feature announces "Please hold while I try to get your party...." before even dialing the physical extension? |
| 20:45.18 | j-fish | Does anyone have any recommendations for a cheap VOIP wholesaler in the US ? |
| 20:45.53 | *** join/#asterisk polysics (~Adium@95.233.75.53) |
| 20:49.19 | [TK]D-Fender | monsterco: Show us |
| 20:49.49 | eric_hill | j-fish: inphonex |
| 20:50.42 | polysics | hi everyone |
| 20:50.50 | polysics | huge memory leak on a box used as a PBX |
| 20:50.54 | j-fish | Looking for something around 25 cents per DID . |
| 20:51.13 | polysics | 11.7 CentOS package. Stuff works, crashes mid-to-late work day, works fine after restart until next day |
| 20:51.23 | polysics | occasionally crashes fast, like it just happened |
| 20:51.39 | polysics | memory usage goes up to25Gb then the kernel does the mercy kill thing |
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| 20:53.01 | Katty | j-fish: do you plan on calling 911 over a .25 DID? |
| 20:54.08 | polysics | I have next to no idea on what to poke at, there's no logging |
| 20:54.17 | drmessano | polysics, did you check what I asked? |
| 20:54.23 | Qwell | Katty: exten => 911,1,Dial(SIP/0118999881999119725www3@provider) |
| 20:54.26 | polysics | trying to find out |
| 20:55.28 | drmessano | polysics, asterisk -rx "xmpp show connections" |
| 20:56.31 | Katty | grins at Qwell |
| 20:57.08 | Katty | Qwell: now with better looking drivers! |
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| 20:58.02 | Katty | Qwell: also, can i fb that? |
| 20:58.33 | Qwell | Katty: sure! |
| 20:58.39 | Katty | ^_____________________^ |
| 20:58.44 | polysics | No such command 'xmpp show connections' |
| 20:58.52 | polysics | think that means there is no Google Voice |
| 20:59.30 | drmessano | I thought you said... |
| 21:00.31 | Katty | what the drmessano say?! |
| 21:00.39 | Katty | troll troll trolltroll trolololol |
| 21:01.01 | polysics | I think I should just communicate by mo-o-o-o-orse instead of using this PBX. |
| 21:01.15 | drmessano | polysics, are you using Google Voice accounts on your PBX? |
| 21:01.21 | drmessano | As in, with Asterisk |
| 21:01.25 | drmessano | as in, on this system |
| 21:01.27 | Katty | polysics: maybe a couple tins with string? |
| 21:01.41 | polysics | I can't tell you because Asterisk is writing an humongous core dump atm, one sec :) |
| 21:01.54 | drmessano | You dont know? |
| 21:02.01 | Katty | i'd enjoy getting my asterisk box working with a couple google voice accounts |
| 21:02.05 | Katty | but i've not had time to tinker with it |
| 21:02.12 | Katty | maybe next year. |
| 21:02.17 | polysics | I happen to not know, yes -_-' |
| 21:02.24 | drmessano | ehrm ok |
| 21:02.31 | drmessano | This is YOUR box? |
| 21:02.34 | Katty | drmessano: have you done incoming google voice to asterisk? |
| 21:02.48 | Katty | drmessano: assuming outbound calls cost Cash(tm) |
| 21:02.53 | drmessano | Katty, I have. It works well. But we'll only have it for a few more months |
| 21:03.01 | drmessano | Katty, outbound is free too |
| 21:03.06 | Katty | drmessano: elaborate |
| 21:03.13 | Katty | drmessano: what happens in a few more months? |
| 21:03.17 | drmessano | Google is killing off XMPP in a few months |
| 21:03.23 | Katty | :< :< :< |
| 21:03.25 | drmessano | So we have no conduit |
| 21:03.45 | monsterco | [TK]D-Fender> - here is a paste from CLI showing "pls-hold-while-try" is played before ringing the physical extension http://paste.debian.net/76490/ |
| 21:04.20 | Katty | drmessano: google kind of reminds me of fairchild semiconductor |
| 21:04.28 | *** join/#asterisk dongola7 (~blair_@unaffiliated/blair/x-0911782) |
| 21:04.30 | [TK]D-Fender | monsterco: First... that isn't an ASTERISK Followme.. |
| 21:04.32 | Qwell | ~freepbx |
| 21:04.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:04.37 | Katty | drmessano: let's make cool new things for the sake of making them! |
| 21:04.48 | Katty | drmessano: and then just...kind of forget about it |
| 21:04.49 | Qwell | glares at infobot |
| 21:05.17 | drmessano | Katty, you don't need this IC anymore, so we're discontinuing it.. even though its hugely popular? |
| 21:05.33 | polysics | I would remove "by people who aren't deeply involved" in the phrase. It just can;t be supported. :D |
| 21:06.09 | drmessano | I would be careful about bashing FreePBX |
| 21:06.29 | polysics | it's a great system but you really have to just learn it yourself |
| 21:06.36 | Katty | drmessano: mhmm. like what they did with the silicon and germanium transistors |
| 21:06.51 | drmessano | polysics, hows that? Its highly supported? |
| 21:07.12 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 21:07.19 | polysics | it is so complex there is actually a low chance of someone else having seen your issue |
| 21:07.26 | [TK]D-Fender | monsterco: So if it's flow is inappropriate.. they'll have to fix it |
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| 21:07.32 | polysics | I noticed I was getting results when I actually started learning the internals myself :) |
| 21:07.33 | [TK]D-Fender | its* |
| 21:07.48 | drmessano | polysics, I don't think thats true at all. |
| 21:08.01 | drmessano | polysics, i've seen very few unresolved issues |
| 21:08.04 | [TK]D-Fender | polysics: I've seen their devs be quite responsive in-channel with regards to issues |
| 21:08.07 | Katty | drmessano: that's not true |
| 21:08.14 | Katty | drmessano: i've had an unresolved issue for 7 years |
| 21:08.31 | drmessano | Katty, you're using underperforming batteries |
| 21:08.44 | Katty | drmessano: everyone keeps giving me solutions, but none of them work! |
| 21:09.03 | drmessano | Katty, dare I ask? |
| 21:09.10 | Katty | can't say i've heard underperforming batteries before. |
| 21:09.12 | Katty | that's a new one! |
| 21:09.45 | Katty | fender suggested a chicken broth server bath yesterday. |
| 21:10.52 | [TK]D-Fender | [16:03]monsterco[TK]D-Fender> - here is a paste from CLI showing "pls-hold-while-try" is played before ringing the physical extension http://paste.debian.net/76490/ <-- And.... |
| 21:10.52 | drmessano | I've had the same issue for 30+ years. The only solutions I have received are people directing me to new places of greater warmth |
| 21:10.59 | [TK]D-Fender | monsterco: Ther is never any magical playback that is simultaneous to a dial. |
| 21:11.07 | [TK]D-Fender | monsterco: There is no way for this to even happen. |
| 21:11.10 | Katty | speaking of problems. |
| 21:11.25 | Katty | i wonder what juipter looks like |
| 21:11.38 | [TK]D-Fender | monsterco: Playing a file is one step. Dialing is another. there is no "dial, then if it starts ringing play the file" |
| 21:12.08 | [TK]D-Fender | monsterco: Your expectation of these being synchronous is faulty |
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| 21:14.32 | polysics | anyways, no, we do not have Google Voice on this box |
| 21:14.41 | polysics | xmpp is not even a command |
| 21:14.53 | drmessano | ok |
| 21:15.13 | Katty | oh boy. |
| 21:15.13 | polysics | this is a stock package install on CentOS of 11.7 |
| 21:15.23 | polysics | we were having crashes on 11.6 too |
| 21:15.52 | polysics | but less frequently, though I can't vouch on the frequence having increased because of the updates or something else |
| 21:21.19 | j-fish | Katty: no 911 calls,only inbound |
| 21:21.34 | Katty | j-fish: well i guess that it some relief |
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| 21:34.02 | [TK]D-Fender | checkout time, BBIAB |
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| 22:06.28 | darkdrgn2k | From a "one vpn per branch at the gateway" model whats the better vpn protocol to use when it comes to VOIP, openvpn, ipsec, somethign else? |
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| 22:34.53 | polysics | drmessano: for the record, it might have been an ulimit issue |
| 22:35.10 | polysics | raising those seems to have stabilized the box |
| 22:37.39 | bsdice | so Katty |
| 22:38.07 | bsdice | I hear you are a walking bundle of issues :) |
| 22:40.04 | Beanie | hello, i'm thinking about getting a voip phone for my mum that is nearly blind - she feels for the numbers - recently got the internet and I want her to be able to dial three digit numbers to get through to me and others - what equipment is needed - doing this on a budget? |
| 22:40.15 | Beanie | it would connect to a remote pbx |
| 22:40.18 | Beanie | over the net |
| 22:41.54 | drmessano | bsdice, that's supposed to be a secret |
| 22:42.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:48.52 | navaismo | Beanie, if the phone is analog you can use an ata pap2t is cheap on ebay |
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| 23:10.44 | bsdice | drmessano are you two married? :) |
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| 23:30.04 | edwin_quijada | Hi! |
| 23:30.26 | edwin_quijada | There is any issue with AGI and virtual machines with asterisk? |
| 23:31.24 | navaismo | dont know |
| 23:31.31 | navaismo | try to be more specific |
| 23:31.41 | edwin_quijada | I have 3 days trying to do a simple script that just read a few digist and it is imposible because any command to read DTMF works. Asterisk doesnt stop to read DTMF |
| 23:32.41 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 23:32.46 | edwin_quijada | I did the script perl, php and java and always is the same |
| 23:33.00 | edwin_quijada | for example getData() doesnt work |
| 23:34.45 | [TK]D-Fender | [18:30]edwin_quijadaThere is any issue with AGI and virtual machines with asterisk? <- no |
| 23:35.09 | navaismo | edwin_quijada, show us the call |
| 23:35.12 | edwin_quijada | I have done everything and always is the same |
| 23:35.30 | navaismo | edwin_quijada, in the past i have used get_data from phpagi and it works |
| 23:35.31 | edwin_quijada | navaismo |
| 23:35.38 | edwin_quijada | me 2 |
| 23:35.52 | edwin_quijada | Really I dont know what is the problem |
| 23:36.04 | edwin_quijada | I tried 3 language |
| 23:36.29 | edwin_quijada | everything that can read doesnt work |
| 23:37.04 | [TK]D-Fender | [18:35]edwin_quijadaReally I dont know what is the problem <- niether do we |
| 23:38.09 | edwin_quijada | http://pastebin.com/cHf2qbjA |
| 23:38.25 | edwin_quijada | this is my script but I dont think that has problem |
| 23:39.55 | [TK]D-Fender | Show an actual call with AGI debug |
| 23:42.28 | edwin_quijada | OK |
| 23:44.08 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:44.08 | *** mode/#asterisk [+o file] by ChanServ |
| 23:48.14 | edwin_quijada | [TK]D-Fender |
| 23:48.27 | edwin_quijada | This is the debug of the script http://pastebin.com/67Dm0Ngp |
| 23:48.52 | edwin_quijada | the script and their debug |
| 23:51.07 | navaismo | now enable the dtmf debug too |
| 23:51.14 | edwin_quijada | ok |
| 23:52.35 | edwin_quijada | what is the command for dtmf debug ? |
| 23:58.25 | [TK]D-Fender | edwin_quijada: So you hit timeout |
| 23:58.31 | edwin_quijada | nop |
| 23:58.42 | edwin_quijada | that is so weird |
| 23:58.52 | edwin_quijada | I am using the SJphone |
| 23:59.24 | edwin_quijada | I dont know why this timeout |