00:00.10 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
00:02.06 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
00:19.54 | *** join/#asterisk theron (~theron@216.51.209.130) |
00:54.48 | *** join/#asterisk theron (~theron@216.51.209.130) |
01:16.32 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
01:16.32 | *** mode/#asterisk [+o Qwell] by ChanServ |
01:35.29 | *** join/#asterisk makubi (~makubi@xdsl-89-0-124-140.netcologne.de) |
01:40.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
01:44.47 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
01:46.26 | *** join/#asterisk D30 (~deo@222.127.13.226) |
02:34.21 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
02:46.45 | *** join/#asterisk makubi_ (~makubi@xdsl-89-0-124-84.netcologne.de) |
02:47.12 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
03:09.29 | *** join/#asterisk theron (~theron@216.51.209.130) |
03:33.52 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
03:35.02 | linocisco | anybody who has tried asterisk external paging with grandstream phones? I tried according to Asterisk definitive guide 4th edition. ended with problem. |
03:40.06 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
03:40.19 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-nadlnieyxkdqvump) |
03:53.14 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
04:00.26 | *** part/#asterisk zopsi (~zopsi@zopsi.com) |
04:02.38 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
04:18.13 | *** join/#asterisk tulga (cb5b710a@gateway/web/freenode/ip.203.91.113.10) |
04:18.25 | tulga | multiple asterisk can use 1 E1 trunk? |
04:21.01 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
04:25.39 | WIMPy | tulga: You can't connect multiple devices to an E1. |
04:26.56 | tulga | WIMPy: install 10 asterisks on 1 server with 1 E1, then share 30 concurrent calls? |
04:27.39 | WIMPy | What kind of sense does it make to install multiple Asterisks on one server? |
04:31.21 | tulga | WIMPy: I have 1 server with 1 E1, then I want service for install 3-4 companies |
04:31.32 | tulga | can I use 1 asterisk for multi tenant? |
04:31.41 | WIMPy | You don't need multiple Asterisks for that. |
04:36.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
04:38.37 | tulga | WIMPy: great. I was checking thirdlane multi-tenant PBX and MIRTaPBX. is it what I want? |
04:43.14 | linocisco | anybody has done calling cards solution using asterisk? |
04:44.18 | WIMPy | tulga: I don't know what YOU want. |
05:15.31 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
05:18.52 | [TK]D-Fender | linocisco: Tons of people |
05:20.43 | linocisco | [TK]D-Fender, If it is the call between only two countries and i am ok only with one originating countryand the other destination country has no easy negotiation |
05:21.26 | [TK]D-Fender | linocisco: That is a mangled question. |
05:22.33 | linocisco | [TK]D-Fender, thanks |
05:22.56 | [TK]D-Fender | Rephrase it. |
05:23.06 | [TK]D-Fender | I can't make sense of what you're trying to ask there. |
05:33.13 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
05:34.15 | *** join/#asterisk r00f (~r00f@bba134140.alshamil.net.ae) |
06:05.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
06:48.08 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
06:49.54 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
06:54.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
06:54.16 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
07:01.59 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
08:10.22 | *** join/#asterisk simpleTon` (~ck@109.161.147.251) |
08:11.05 | simpleTon` | hi i am building an asterisk based ip-pbx system with 10 incioming lines and 20 extensions.. how many ports trunk card should i buy |
08:18.28 | linocisco | simpleTon`, what 10 lines? landlines? |
08:38.41 | linocisco | hi all |
08:39.46 | linocisco | hi all, I tried to put this line exten => *723,1,Page(MulticastRTP/basic/239.0.0.1:1234) under "internal" dialplan and calling from one extension to another doesn't work anymore though that line was removed and asterisk and phones are restarted |
08:40.02 | linocisco | I tested with grandstream GXP1405 and zoiper |
08:42.10 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
08:49.32 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
08:50.30 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
09:17.25 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
09:17.33 | gusto | hi |
09:25.32 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
09:34.36 | *** join/#asterisk Neoti (~Thunderbi@86.26.181.34) |
09:58.20 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
10:16.55 | *** join/#asterisk Geek-Linux (~mubbashir@101.50.76.211) |
10:17.11 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
10:17.40 | Geek-Linux | Hi All, i want to know about asterisk video support and what would i need to implement the setup. ? |
10:19.15 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
10:32.34 | *** join/#asterisk Draecos (~Draecos@58-7-129-45.dyn.iinet.net.au) |
10:34.28 | *** join/#asterisk boratynskikamil (5032e462@gateway/web/freenode/ip.80.50.228.98) |
10:35.18 | gusto | so |
10:35.34 | boratynskikamil | Hello. I am preparing cluster based Call Center. How would you recommend. Divide 4 ISDN lines to 2 independent Asterisk PBX and each one is checking the state of devices on another one? |
10:35.56 | gusto | i just made asterisk 12 run with my custom set of modules |
10:36.38 | gusto | ha? |
10:38.08 | gusto | seems that all this pjsip replaced the modules which were asterisk-own ones and do not exist any more like callerid seems to be a part of pjsip now along with some SIP RFCs |
10:38.11 | gusto | so |
10:38.21 | gusto | then the bridges are extra |
10:38.24 | gusto | and now it works |
10:38.43 | gusto | but there are a lot more modules now |
10:39.35 | gusto | will chan_sip disappear completely now, or what? |
10:39.59 | gusto | i still have it in, and it does not seem to work without it, but it is somehow redundant now |
10:42.38 | file | chan_pjsip and chan_sip do not have feature parity, and can co-exist on the same system if they bind to different ports |
10:42.43 | file | so both will continue to exist |
10:43.08 | gusto | there is no chan_pjsip to me |
10:43.16 | gusto | i have only res_pjsip in there |
10:43.24 | file | it exists. |
10:43.37 | file | the PJSIP stuff is broken up into many many many modules |
10:43.55 | gusto | ah, i can load it, i just noticed it |
10:44.11 | gusto | however, i have pjsip not loaded now |
10:44.20 | gusto | you say that i can get rid of that pjsip res stuff? |
10:44.31 | file | if you aren't using PJSIP support you don't need to load any of the modules. |
10:44.37 | gusto | ah |
10:44.54 | gusto | and will my callerid work? because the old asterisk 11 callerid module is not there any more |
10:45.08 | file | what "old asterisk 11 callerid module"? |
10:45.12 | gusto | w8 |
10:45.41 | gusto | 'app_setcallerid.so' |
10:46.19 | file | that's been deprecated for, oh, 5 or more years now? |
10:46.24 | gusto | ok |
10:46.36 | file | it's still in the tree, though |
10:46.40 | gusto | so all this functions are now implemented into chan_sip, or where? |
10:46.45 | file | it's just not built by default |
10:46.47 | file | no |
10:46.48 | linocisco | hi all, for asterisk to asterisk over ADSL, what is the maximum bandwidth requirement for uplink speed /downlink speed for how many users |
10:46.49 | linocisco | ? |
10:47.08 | linocisco | hi all, for asterisk to asterisk over ADSL, what is the minimum bandwidth requirement for uplink speed /downlink speed for how many users |
10:47.14 | file | if you need that module then go into menuselect and enable it, it is not enabled by default because everyone has moved on to using func_callerid |
10:47.28 | gusto | you can count ... like 64kbps per user x 2 (up , down) |
10:47.54 | gusto | file: aha, so func_callerid is it, ok |
10:48.05 | file | func_callerid REPLACED it but it is not used in the same way |
10:48.30 | gusto | however, i have it in, so it must be right |
10:49.13 | gusto | i never checked that modules one by one, i moved from 1.8 with a list that i found on openwrt router package, because that one was good and fully functional |
10:50.07 | gusto | so basically the only difference now to the asterisk 12 is that they moved the bridge functions out |
10:50.25 | gusto | without that modules i was not able to make a call :D |
10:50.33 | file | we completely rewrote bridging. |
10:51.00 | file | and yes, you wouldn't be able to bridge calls without the modules to bridge them |
10:52.32 | gusto | ok, i kicked out that pjsip and it works |
10:52.57 | gusto | now i have only 55 modules |
10:53.01 | gusto | that is better readable |
10:53.46 | *** join/#asterisk Draecos (~Draecos@58-7-129-45.dyn.iinet.net.au) |
10:56.08 | *** join/#asterisk r00f (~r00f@bba134140.alshamil.net.ae) |
11:23.06 | Geek-Linux | Hi All, i want to know about asterisk video support and what would i need to implement the setup. ? |
11:33.14 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
12:06.48 | gusto | btw |
12:07.43 | gusto | is there a possibility to tell him some kind of priority for the codecs so that he does not go and transcode something while there is a codec that could be used instead that would not need transcoding for making a connection |
12:07.45 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
12:20.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
12:30.52 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginm.net) |
12:36.54 | davlefou | hi, somes thing is strange, i need to put canreinvite=no and nat=force_rport,comedia on my asterisk who is under server host. |
12:41.38 | *** join/#asterisk brendan`_ (~textual@107-1-118-122-ip-static.hfc.comcastbusiness.net) |
12:47.08 | *** join/#asterisk brendan` (~textual@107-1-118-122-ip-static.hfc.comcastbusiness.net) |
12:50.00 | *** join/#asterisk OneNarrowWay (~OneNarrow@86.92.252.236) |
12:57.36 | gusto | a very brutal way, but i proved it to be functional is to just remove every codec module that you do not want to be reencoded (into,out-to) and he is then taking the next best one |
12:58.11 | gusto | but i wonder if this causes some problems, for example because of retransmiting some messages through sip and so on, i ll have to debug that as well |
12:58.59 | *** join/#asterisk brendan` (~textual@107-1-118-122-ip-static.hfc.comcastbusiness.net) |
12:59.58 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
13:10.43 | gusto | eh, as of before asterisk 12 he told me if he is locally bridging or remotely bridging, i even looked now at the bridge show command and not even there i can read if it is a local or remote bridge, the only way i see it is either over rtp set debug on or sip set debug on |
13:10.52 | gusto | there must be some other way to look into it |
13:11.46 | gusto | or local show channels, that may indicate something, i ll have to try a local bridge, maybe it will show up there |
13:13.42 | gusto | local show shows nothing |
13:14.09 | gusto | but there is a difference in the technology in native bridge being directmedia and simple bridge locally bridged one |
13:15.14 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
13:28.18 | *** join/#asterisk kannan (~chatzilla@42.104.62.70) |
13:31.46 | gusto | is the asterisk wiki down or something? |
13:32.12 | gusto | it does not seem to work, however |
13:41.53 | Qwell | gusto: It's working fine. |
13:42.15 | gusto | what? |
13:42.29 | gusto | not to me |
13:42.39 | gusto | is it somehow ipv6 enables w/o ipv6 delivering? |
13:45.30 | gusto | ok, i just installed w3m, because on graphical browsers you can not put an option which ipv protocol he should use and really, it loads under -4 but does not with -6 |
13:45.36 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
13:45.48 | gusto | w3m says that he is stuck while opening socket using ipv6 |
13:46.03 | gusto | and somehow that crashed my w3m :-D |
13:46.30 | gusto | i can put it in a list of ipv4 only websites |
13:46.34 | gusto | on firefox |
13:47.30 | gusto | ok, i put it in ipv4 only domains, now it works, however, that is a quirk, one should fix that ipv6 connectivity |
13:47.50 | gusto | it looks like a firewall issue there |
13:48.41 | gusto | 2001:470:e0d4::ee answers to icmp, so that can not be the problem, maybe misconfiguration on a webserver? |
13:52.13 | gusto | a ssl problem maybe ... that he wants to redirect on ssl port and recieves no answer there, that's my unproven theory now, lol |
13:52.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
13:52.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
13:54.38 | gusto | ... https://[2001:470:e0d4::ee]/ here, for those who are curious |
13:55.28 | gusto | port is open and responds, but fails somewhere at ssl |
13:55.40 | gusto | 443/tcp open https |
13:56.38 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-lrgdempgcskujqjv) |
13:56.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
13:58.56 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
13:58.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
14:00.57 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
14:01.38 | *** join/#asterisk sliske_- (~sliske_@2001:4800:780e:510:6c33:9987:ff05:245f) |
14:01.46 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-fhilwbinbpcxdzru) |
14:01.47 | *** mode/#asterisk [+o newtonr] by ChanServ |
14:02.54 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
14:13.15 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
14:13.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
14:13.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:28.32 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
14:30.09 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
14:43.23 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
14:47.03 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
14:47.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
14:49.07 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
14:49.09 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
15:19.51 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
15:24.29 | *** join/#asterisk u0m3 (~u0m3@109.96.138.238) |
15:27.39 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
15:31.14 | ipengineer | file: Are you around? |
15:31.23 | file | yes |
15:32.47 | *** join/#asterisk aness (~aness@2a02:fe0:c311:180:5c19:5727:26e:8813) |
15:33.14 | ipengineer | I was talking to one of my developers about this dialog-info issue and I am not sure we can easily do this without spending a LOT of time getting them up to speed with the asterisk core itself. That is where the breakdown will be.. We got to talking and we have Kamailio as a proxy between Asterisk and the UA. Could we just have kamailio look for a certain Notify event and add the dialog-info payload to that? If so that ma |
15:33.15 | ipengineer | a simple solution for us until this can get done correctly and implemented the correct way |
15:33.43 | file | I don't know Kamailio to that extent and can't answer that question. |
15:34.05 | *** join/#asterisk navaismo (~navaismo@189.146.0.53) |
15:34.06 | file | Probably not, though. |
15:37.36 | mjordan | Asterisk wouldn't know to raise the NOTIFY event |
15:38.18 | ipengineer | Can kamailio not subscribe for notify events? Maybe not dialog-info but some other format? |
15:42.36 | *** join/#asterisk makubi (~makubi@xdsl-87-79-220-138.netcologne.de) |
15:51.17 | ipengineer | file: I can post a bounty for this.. I know we discussed that initially. If someone picks it up that will save us both some work. I am more than willing to pay to get it done.. Do you have “rough” time estimates so I can come up with an offer? |
15:51.34 | file | that's a complicated question |
15:51.58 | ipengineer | I tend to be on a role with those here lately.. |
15:52.27 | file | you should ask mjordan! he's good at making up fictional numerical values |
15:52.35 | ipengineer | Haha ok.. |
15:53.52 | ipengineer | mjordan: Do you have any thoughts on how long it may take someone that is familiar with asterisk to build out this dialog-info functionality? |
15:55.35 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
16:04.32 | *** join/#asterisk Arsenick (~arsenick@fedora/Arsenick) |
16:12.48 | *** join/#asterisk KavanS (~quassel@LINBIT/KavanS) |
16:18.18 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginm.net) |
16:24.49 | *** join/#asterisk kannan (~chatzilla@123.238.235.11) |
16:33.00 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
16:41.08 | *** join/#asterisk CeBe (~CeBe@port-92-206-10-196.dynamic.qsc.de) |
16:55.09 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
16:56.27 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
16:59.43 | jeffspeff | i'm getting the following sip debug output when a call is sent to a new sip trunk. http://pastebin.com/jLRctWc5 |
16:59.52 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
17:01.06 | mjordan | ipengineer: familiar with the pjsip stack, not long. Not familiar with the pjsip stack, a bit longer. Particularly if they don't ask for help :-) |
17:01.35 | [TK]D-Fender | jeffspeff: that has nothing to do with a call. |
17:01.42 | mjordan | as for fictional numbers... let me see if I can't find a corollary |
17:02.39 | mjordan | file: does this only need a body generator? |
17:08.27 | mjordan | looks at file |
17:10.05 | mjordan | ipengineer: so my guess is (since file has gone 'dark') is that this is two modules, both approximately a few hundred lines of code. It's not super hard, as there are (a) examples of modules that do this already and (b) a review with a rejected approach available |
17:10.42 | mjordan | ipengineer: you'd need a module that subscribed for the dialog state in Asterisk (or otherwise monitored the state) and hooked itself onto res_pjsip_pubsub |
17:10.43 | file | my IRC disconnected for a sec :P |
17:10.57 | mjordan | ipengineer: and you'd need a module that was a body generator for the NOTIFY requests and file can correct me if I'm wrong |
17:10.59 | file | mjordan, no - it requires slight modification to allow body generators to store state |
17:11.09 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
17:11.13 | mjordan | ah, so a bit of work in res_pjsip_pubsub then too |
17:11.31 | file | and a body generator |
17:11.35 | mjordan | file: can it use res_pjsip_extenstate? |
17:11.38 | file | and preferably not duplicating the res_pjsip_exten_state module |
17:11.42 | mjordan | right |
17:12.34 | mjordan | file: so you'd want to base the dialog state off of the device state subscription? |
17:13.48 | file | yes |
17:14.09 | file | dialog-info+xml is unlike other contents because it has an incrementing numerical value |
17:14.28 | file | (per subscription, must start at 1 if I remember right) |
17:14.37 | mjordan | ew |
17:14.53 | mjordan | but if it is per subscription, that's not terrible. |
17:15.12 | mjordan | Store a void * to some user defined ao2 object that they can pass via the body generator on subscription |
17:15.34 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
17:15.51 | mjordan | although that could get a bit messy, since you'd have to have the body generators consume subscription state so that they could interpret the current state of the counter |
17:16.16 | mjordan | ipengineer: SO. Revision #2: it's one module (not a very large one), with some tweaks to some existing APIs in res_pjsip_exten_state and res_pjsip_pubsub |
17:17.09 | file | mjordan, the slightly annoying part is that subscriptions have datastore support for this already |
17:17.16 | file | (that's what I used last time) |
17:17.19 | file | but body generators are generic |
17:17.24 | mjordan | file: but we didn't expose them up through the API? |
17:17.54 | file | putnopvut kept body generators generic - they aren't sub specific |
17:21.25 | putnopvut | Correct. Body generators need to be usable for requests outside of subscriptions, such as PUBLISH and unsolicited MWI. |
17:23.17 | davlefou | hi, i have that error : chan_sip.c:4343 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
17:23.25 | davlefou | i try to use sql |
17:23.26 | file | walks home |
17:27.01 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
17:33.09 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
17:37.15 | file | awww I missed nothing |
17:44.35 | ipengineer | file: mjordan Sorry.. I just got back from a meeting. Ok I think that clears it up a little bit |
17:45.33 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
17:58.37 | tm1000 | can feature codes be detected in app_meetme or app_confbridge |
17:58.46 | davlefou | it seems an problem with sip.conf put un static |
17:58.48 | tm1000 | such as *<dtmf> |
18:15.55 | *** join/#asterisk willwh (~willwh@unaffiliated/willskills) |
18:17.14 | file | tm1000, no |
18:17.21 | file | tm1000, not unless you front end your call with a Local channel |
18:27.48 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
18:32.27 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
18:35.47 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
18:37.36 | moy | willwh: did you get wss working? |
18:38.42 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
18:44.51 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
18:45.00 | willwh | moy: ! zomg |
18:45.02 | willwh | no :( |
18:45.07 | willwh | I didn't have time to file a bug |
18:45.23 | WIMPy | You get used to that. |
18:45.52 | willwh | it's interesting, in a js console; conn = new WebSocket('http://blabla:10060/ws','sip') |
18:45.59 | willwh | conn.readyState = 1 |
18:46.17 | willwh | but if I go, https://blabla:10060/ws - no dice |
18:46.28 | willwh | my cert is ok, I enabled the AMI http interface with tls in manager.conf |
18:46.33 | willwh | and it is a kosher cert |
18:47.01 | willwh | and oops, the WebSockets are like (ws://server:10060/ws) and (wss://) respectively |
18:47.13 | willwh | when trying wss I just get "3" which is closed |
18:47.49 | willwh | moy: what timezone are you on? I'm at work atm, and I don't have much time to chat |
18:48.04 | willwh | I'm on Pacific Standard Time |
18:48.28 | willwh | I'm running the very latest 12.2.0-RC-2 from svn |
18:48.32 | willwh | which I am pretty sure has your changes |
18:48.43 | willwh | I didn't bring my laptop to work or I could fire up my dev env |
18:49.00 | willwh | moy: ^ :) |
18:52.38 | willwh | moy: when issuing 'http show status' - should I expect to see anything about Secure ws? |
19:22.11 | *** join/#asterisk jemidy (~jemidy@nat/digium/x-qaxfgmrmmiayqdyx) |
19:23.00 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
19:27.27 | *** join/#asterisk lachesis (~lachesis@unaffiliated/lachesis) |
19:39.52 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
19:42.54 | willwh | moy: give me a shout when you can, I'll idle in here :) (woot irssi on a vps) |
19:45.09 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginm.net) |
19:47.10 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.182.172.tbinet.bm) |
19:48.30 | *** join/#asterisk benasse (~benasse@cicogna.fr) |
19:52.08 | davlefou | i find the problem, ip error |
19:59.13 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
20:20.45 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
20:27.23 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
20:39.07 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
20:39.27 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
20:39.34 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
20:41.59 | *** join/#asterisk lordvadr (~lordvadr@jose-tc.ctc.biz) |
20:43.31 | lordvadr | Is there a version of "core show channels" that lists the full channel names? Longer channel names gets truncated by the output formatting. From a different perspective, I'm writing a script and I need to find all the channels that are up and have called some number. "asterisk -rx 'core show channels' | grep <number>" kinda gets there, but often times part of the channel names are truncated. |
20:44.54 | WIMPy | core show channels <tab> |
20:45.20 | lordvadr | WIMPy: Can't script that. |
20:45.54 | WIMPy | Have you tried doing it yopurself? |
20:46.30 | lordvadr | doing what myself? Actually, "core show channels concise" appears to be exactly what I want. I've used the verbose version but didn't know that "concise" existed. |
20:46.52 | WIMPy | See, that's why you should press tab. |
20:46.59 | willwh | yay tab completion >.> |
20:47.29 | eirirs | tab symbol on my tab are gone. |
20:47.33 | eirirs | tab abuse <3 |
20:47.35 | lordvadr | I've hit tab after that several times, but when a server responds with a couple hundred channels, the "concise" part always got lost. |
20:48.06 | eirirs | core show channels c<tab> works too |
20:48.08 | WIMPy | No. |
20:48.23 | WIMPy | That would have been core show channel <tab> |
20:48.41 | lordvadr | WIMPy: Yeah, that's exactly what happened. Thanks for that. |
20:52.53 | *** join/#asterisk saint__ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
20:59.20 | moy | willwh: you should at least see HTTPS server enabled |
21:00.14 | moy | willwh: you should read the troubleshooting hints on this presentation: http://goo.gl/iJyaXZ |
21:00.48 | moy | starting from slide 16 particularly |
21:02.38 | moy | willwh: regarding tz, I am on EDT |
21:10.13 | *** join/#asterisk shuggans (~shuggans@ucomnetworks-209-203-168-138.ucom.net) |
21:10.48 | shuggans | Hi all - Wondering if anyone can help me out with getting nortel 1110 phones working with asterisk/freepbx |
21:11.47 | *** join/#asterisk johndropper (~justin@cpe-065-190-162-078.nc.res.rr.com) |
21:13.10 | *** join/#asterisk johndropper (~justin@cpe-065-190-162-078.nc.res.rr.com) |
21:19.52 | *** join/#asterisk navaismo (~navaismo@201.161.2.54) |
21:23.22 | newtonr | shuggans, depending on the issue, yes, someone can likely help. If you are using FreePBX. You may get a better answer in the #freepbx channel |
21:23.30 | newtonr | ~ask |
21:23.30 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
21:30.47 | *** join/#asterisk sectech (~sectech@stjhnbsu0ww-142134150058.dhcp-dynamic.FibreOP.nb.bellaliant.net) |
21:31.25 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
21:44.49 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
22:09.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
22:19.06 | willwh | moy: cool, well, I will crank on this tonight, you'll probably be sleeping :) |
22:19.27 | willwh | moy I should say, I have gone through that :) |
22:19.42 | willwh | If I don't get it, should I bug report, or just mailing list? |
22:20.35 | willwh | moy: thanks so much for reaching out to me though :) |
22:20.47 | willwh | I'll write up my configs / deps installed etc |
22:21.17 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
22:22.10 | willwh | ah, moy I see my prob |
22:22.38 | willwh | I guess it helps to read, re-read, then come back another day and read it again, haha, I don't have tlsenable/tlsbindaddr set in the http.conf |
22:22.42 | willwh | shoots himself |
22:22.56 | willwh | I'd imagine this will work when I get home, but I'll ping you and let you know |
22:23.06 | willwh | thank you for all your work, just awesome! :) |
22:23.54 | willwh | moy++ |
22:26.13 | willwh | this needs an update: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support :) |
22:26.16 | willwh | I'll try and do that too |
22:27.33 | navaismo | that is ok for WS only |
22:27.47 | willwh | navaismo: yeah, but it should have a section for wss |
22:27.58 | willwh | 'cause I think all that I need is tls details in http.conf, and 12.2.0-rc2 will work |
22:28.16 | navaismo | and just to be curious did you know soon chrome will not work with asterisk because the recent change with chrome |
22:28.22 | willwh | I did not\ |
22:28.28 | willwh | which is what? |
22:28.39 | willwh | navaismo: got a link? |
22:28.51 | navaismo | 1 sec |
22:28.58 | willwh | appreciate it thx |
22:29.05 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
22:30.00 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
22:30.12 | navaismo | https://code.google.com/p/webrtc/issues/detail?id=2774 |
22:31.35 | navaismo | so if you are "investing" time creating an app you need to consider or a MediaGateway or wait for an official RFC for Webrtc this days the hype is mess |
22:32.38 | navaismo | the Guys from Jssip wants to rule the world with their RFC, in the meantime others are stuck in a battle between SDP or not SDP and finally others with VP8 and open-H264 |
22:32.57 | navaismo | so basically webrtc turns in a Lego kit for recreation |
22:33.28 | navaismo | ok let me rephrase that... WebRTC apis |
22:35.17 | navaismo | https://issues.asterisk.org/jira/browse/ASTERISK-23425?filter=-3 |
22:48.23 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
22:57.09 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
22:57.25 | joobie | hey guys, what variable can i use to identify the dialling (src) extension ? |
23:01.38 | [TK]D-Fender | joobie: "core show function CDR" <- |
23:02.17 | joobie | ta [TK]D-Fender |
23:08.15 | marceloamorim | guys, I'm installing the asterisk on the debian, and I'm following the wiki.asterisk.og, I'm on the install_prereq unpackeg and I don't have subversion installed, so I installed and do the command "./install_prereq isntall-unpackeged", but now it takes so long |
23:08.19 | marceloamorim | this is normal? |
23:09.09 | marceloamorim | I was checking the the install_prereq and this script is so simple, just svn the nbs-trunk, make & make isntall |
23:15.10 | marceloamorim | never mind, I killed the process and run again, now it did faster |
23:15.12 | sectech | Not interested in starting a flamewar, but is there really that much different between Asterisk and Freeswitch? |
23:15.35 | sectech | I kinda need to dedicate my efforts towards one... not both |
23:15.51 | sectech | I know asterisk fairly well, not much about freeswitch though |
23:17.32 | sectech | If you try and google the differences, you just find the pissing matches that have gone on in the past... Not very much useful info |
23:23.55 | *** join/#asterisk jhirley (~chatzilla@c-66-176-182-203.hsd1.fl.comcast.net) |
23:24.07 | file | falls into existence |
23:25.38 | *** join/#asterisk valeech (~valeech@pool-71-171-123-210.clppva.fios.verizon.net) |
23:28.53 | *** join/#asterisk navaismo (~navaismo@189.146.0.53) |
23:49.54 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |