| 00:00.02 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:00.03 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:02.16 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 00:10.42 | *** join/#asterisk petris (znc@ip-50-63-24-91.ip.secureserver.net) |
| 00:10.52 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 00:23.57 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 00:33.32 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:37.33 | *** join/#asterisk atcmmi (~atcmmi@14.127.177.19) |
| 00:53.13 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.191) |
| 00:53.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.239) |
| 00:55.09 | *** join/#asterisk ketas (~ketas@135.216.191.90.dyn.estpak.ee) |
| 01:25.59 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:25.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:39.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.144) |
| 01:44.30 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 01:55.15 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 02:18.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:20.35 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 02:20.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:27.36 | *** join/#asterisk youjelly (~youjelly@39.47.106.18) |
| 02:31.14 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 02:53.00 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:00.12 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 03:01.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cktbzqfbuyqjyclp) |
| 03:01.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:24.51 | *** join/#asterisk jackdeth_ (~jackdeth_@2001:1938:232:0:74e9:8fe4:1294:1ced) |
| 03:25.21 | jackdeth_ | is there a "show" command that will show all the audio codec transcoder sessions happening on the asterisk server? |
| 03:25.40 | jackdeth_ | anyone know? |
| 03:26.53 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:27.41 | linocisco | hi all , i want analog FXO 4 port gateway for asterisk (cisco brand). can anyone tell me? |
| 03:39.02 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 03:45.33 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 03:59.28 | igcewieling | jackdeth_: sip show channels will show that for sip channels |
| 04:00.45 | jackdeth_ | hi igcewieling thanks for the response |
| 04:01.31 | jackdeth_ | i know that sip show channels will show the codec the sip call is using, and if i look at the "show" command for the device on the the other end (in this case a sccp device), I can see it's using a different codec, so I can infer there is a transcode happening |
| 04:01.46 | jackdeth_ | but i wasn't sure if there was a "show" command (maybe part of "core") that would show all the transcoder sessions |
| 04:04.35 | igcewieling | if you run dumpchan after Dial hangs up you can see more codec info about the call |
| 04:09.37 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 04:13.36 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 04:13.38 | Rahail | hi |
| 04:13.51 | Rahail | which version of asterisk you can say is more stable and less bugs |
| 04:13.54 | Rahail | on iax2 |
| 04:17.06 | Rahail | WIMPy |
| 04:17.08 | Rahail | you around |
| 04:18.31 | linocisco | hi all , i want analog FXO 4 port gateway for asterisk (cisco brand). can anyone tell me? |
| 04:32.55 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:45.30 | *** join/#asterisk FonJockey (~FonJockey@c-24-30-113-162.hsd1.ca.comcast.net) |
| 04:47.52 | linocisco | hi all , i want analog FXO 4 port gateway for asterisk (cisco brand). can anyone tell me? |
| 04:55.58 | Maliuta | linocisco: why cisco? sounds like a digium card would be a better fit |
| 04:56.31 | linocisco | Maliuta, i wish i could use digium. but management from HQ set Cisco is our standard |
| 04:56.51 | igcewieling | linocisco: is Cisco Linksys an option? |
| 04:57.24 | linocisco | igcewieling, whatever. it should have cisco original label and should be product of cisco |
| 04:57.58 | linocisco | hope cisoco is better than grandstream ? |
| 04:58.23 | Maliuta | linocisco: but they're using Asterisk and not Cisco Call Manager? |
| 04:59.07 | igcewieling | The software of cisco phones for SIP is not all that great |
| 04:59.08 | linocisco | Maliuta, no budget for that. I have to use asterisk on QNAP which is non-cisco exsiting standard |
| 04:59.24 | linocisco | igcewieling, SPA phones are not good? |
| 04:59.42 | Maliuta | linocisco: I'd argue that the Digium TDM400 is what you need, and that since it was designed with Asterisk in mind it would be a superior solution. |
| 05:00.10 | igcewieling | linocisco: The SPA series are far more interoperable than the non-SPA series. |
| 05:00.17 | Maliuta | igcewieling: I have no problem with SIP on my Cisco handset |
| 05:00.25 | *** join/#asterisk FonJockey (~FonJockey@c-24-30-113-162.hsd1.ca.comcast.net) |
| 05:00.32 | igcewieling | Maliuta: do you do BLF? |
| 05:00.38 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:00.52 | linocisco | igcewieling, I am using SPA502G and 7942G with sip firmware. BLF is diffent issue |
| 05:01.16 | Maliuta | igcewieling: no, I don't |
| 05:07.14 | Maliuta | linocisco: what are you looking to put this card in? I can't seem to find a cisco product to stick in a PC for this |
| 05:07.32 | linocisco | Maliuta, no |
| 05:07.49 | linocisco | Maliuta, i would rather use cisco FXO gateway as external device |
| 05:09.20 | Maliuta | are the SPA products going to stay cisco? Since they offloaded Linksys |
| 05:19.12 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 05:33.07 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 05:58.26 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 05:58.27 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:05.39 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:06.57 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:17.10 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:22.41 | linocisco | Maliuta, linksys Sinpura , whatever. can be still found under cisco site |
| 06:33.54 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 06:50.34 | *** join/#asterisk jayk (jayk@centauri.slackware.net) |
| 06:59.53 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:29c7:206c:b0d7:f245) |
| 07:08.09 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 07:13.26 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 07:25.29 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 07:28.50 | *** join/#asterisk areski (~areski@80.174.128.95.dyn.user.ono.com) |
| 07:30.17 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:33.56 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 07:34.19 | linocisco | hi all , i want analog FXO 4 port gateway for asterisk (cisco brand). can anyone tell me? |
| 07:36.12 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:41.13 | *** join/#asterisk iamamoron (~Miranda@210.238.181.187) |
| 07:41.36 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 07:45.57 | iamamoron | hi there i have a panasonic pabx i want to purchase fxo card to bridge the ip phones on it |
| 07:46.04 | iamamoron | would there be issues? |
| 07:49.47 | linocisco | iamamoron, i dont think u can integrate with asterisk |
| 07:50.07 | iamamoron | i have KX-TDE200 |
| 07:50.11 | iamamoron | ic |
| 07:50.12 | iamamoron | so sad |
| 07:50.26 | iamamoron | for what reason? |
| 07:50.40 | linocisco | iamamoron, unless u use Digium gateway devices |
| 07:51.07 | linocisco | iamamoron, which are very new products from digium and interoperability with legacy non-IP pbx |
| 07:51.30 | linocisco | iamamoron, i have KX-TDA824, TDA100 |
| 07:51.39 | kaldemar | if the panasonic pbx has an interface that is compatible with something that asterisk can use, then it's no problem. |
| 07:53.47 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:58.17 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:02.58 | iamamoron | even if i purchae fxo card? |
| 08:05.26 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:07.19 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:10.13 | kaldemar | iamamoron: what interface does your panasonic pbx have? |
| 08:11.12 | iamamoron | what do you mean? |
| 08:11.26 | iamamoron | for now it was cabled with pots |
| 08:11.35 | iamamoron | all extension runs on it |
| 08:11.42 | iamamoron | i want to integreate ip phones |
| 08:12.02 | iamamoron | thats what i want to achieve |
| 08:12.21 | iamamoron | because in my view if there is lan cable its easy for me to put telephone |
| 08:12.39 | iamamoron | i dont have to wire up all just for 1 telephone |
| 08:12.44 | kaldemar | i mean what interface does it have that you could use to connect to an asterisk box? |
| 08:12.45 | iamamoron | extention |
| 08:12.49 | iamamoron | extension |
| 08:13.33 | kaldemar | if you have a "LAN" connector on a device it doesn't mean it speaks every VoIP protocol. |
| 08:14.24 | kaldemar | so presumably you have an FXS, if it only has analog interfaces that are for phones. so you need an FXO device. |
| 08:15.16 | iamamoron | i know i can use UTP for phone |
| 08:15.24 | iamamoron | but how about wireless |
| 08:15.40 | iamamoron | in that case i need VOIP |
| 08:15.44 | iamamoron | it is LAn only |
| 08:15.51 | iamamoron | any recommendations? |
| 08:15.58 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-047-089.pools.arcor-ip.net) |
| 08:16.12 | kaldemar | for what? you're a bit unclear. |
| 08:16.25 | *** join/#asterisk [sr] (~kvirc@smtp.decimal.pt) |
| 08:16.51 | iamamoron | i want to mix VOIP on my KX-TDE200 |
| 08:17.07 | iamamoron | i want to use IP phone |
| 08:17.15 | iamamoron | so that fast install for me |
| 08:17.42 | iamamoron | but that IP phone should have an extension from my KX-TDE200 |
| 08:17.52 | iamamoron | any ideas? |
| 08:18.38 | kaldemar | are you setting up an asterisk box or are you just looking to connect a VoIP phone directly to your panasonic pbx? |
| 08:23.34 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 08:24.07 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:25.43 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:30.09 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 08:30.32 | linocisco | iamamoron, check out new digium gateway |
| 08:37.06 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 08:41.26 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 08:44.43 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 08:49.48 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 08:50.06 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 08:51.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 08:51.58 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:54.29 | iamamoron | if fxo only |
| 08:54.38 | iamamoron | would it be problem? |
| 08:55.29 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.129) |
| 08:57.15 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.147) |
| 08:58.31 | kaldemar | linocisco: the digium gateways are PRI-SIP. they will not do. |
| 08:58.51 | kaldemar | iamamoron: if you have FXO in the pbx, get an FXS device. if you have FXS, get an FXO device. |
| 09:04.37 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 09:14.59 | *** join/#asterisk areski (~areski@165.Red-95-120-9.dynamicIP.rima-tde.net) |
| 09:23.20 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 09:23.35 | MarKsaitis | hey |
| 09:23.47 | MarKsaitis | asterisk 11.... how do I compile it with srtp and zrtp? |
| 09:23.51 | MarKsaitis | all these guides online are dull |
| 09:37.25 | linocisco | kaldemar, how can asterisk detect panasonic PBX as FXO or FXS ports? any driver to patch or any tweak? |
| 09:40.39 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 09:48.42 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:53.16 | MarKsaitis | how on earth do i tell asterisk 11 to compile with srtp support? |
| 09:56.41 | *** join/#asterisk F-JJTH (~clement@car84-1-78-229-89-36.fbx.proxad.net) |
| 09:57.44 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:58.40 | F-JJTH | Hi all, a simple question : For now I use an IAX client in order to communicate with my Asterisk server. If I want to change my client for a SIP client, all my [macro] and more globally dialplan still compatible ? The protocol doesn't affect it, right ? |
| 10:06.07 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 10:08.00 | linocisco | what is the best device for CDMA800Mhz sim card and GSM or WCDMA as FXO gateway? |
| 10:09.25 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 10:12.48 | kaldemar | linocisco: the fact that it is panasonic is irrelevant. if it is an FXO or FXS and asterisk has a compatible interface, there's nothing to recognize. you configure it and use it. |
| 10:13.00 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 10:15.42 | linocisco | kaldemar, thanks . i m not into that question. my very last question is "what is the best device for CDMA800Mhz sim card and GSM or WCDMA as FXO gateway?" |
| 10:16.17 | linocisco | kaldemar, as very hard to get PSTN or internet lines for me |
| 10:16.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 10:16.32 | kaldemar | linocisco: if you're not interested in it, then don't highlight me with it. |
| 10:17.25 | linocisco | kaldemar, sorry bro. just letting u know |
| 10:18.07 | MarKsaitis | help? how do i compile asterisk 11 with srtp? ubuntu 12.04 lts here |
| 10:18.19 | MarKsaitis | there are no good topics online on this |
| 10:19.11 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 10:19.55 | kaldemar | MarKsaitis: enable res_srtp in make menuselect under "Resource Modules" |
| 10:20.09 | MarKsaitis | I know this for a fact kaldemar :))) I cant enable it |
| 10:20.27 | MarKsaitis | I did install srtp from sources - still cant enable it |
| 10:21.45 | kaldemar | MarKsaitis: install libsrtp-dev, re-run the configure script and try again. |
| 10:22.29 | kaldemar | or if insist on installing libsrtp from source, run the configure script with --with-srtp=/parth/to/srtp/headers and then enable it in make menuselect |
| 10:22.49 | MarKsaitis | kaldemar, I found the thing now.... reconfig told me to recompile srtp with cflags=-fPIC .... what a stupid thing :D |
| 10:23.40 | MarKsaitis | whats the diff between libsrtp-dev and libsrtp--dev ? |
| 10:23.44 | MarKsaitis | I will try some packages |
| 10:23.56 | MarKsaitis | woops, libsrtp0-dev |
| 10:25.44 | kaldemar | libsrtp-dev is a virtual package, just a convenience to install a package that provides the development package for libsrtp. |
| 10:25.48 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:26.02 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 10:26.29 | kaldemar | libsrtp0-dev or libsrtp1-dev for example is the package that provides the headers. |
| 10:26.54 | MarKsaitis | okay, in that case what is libsrtp0 then? |
| 10:27.01 | MarKsaitis | there isnt one without 0 like libsrtp... |
| 10:27.12 | MarKsaitis | which ones would I need in my scenario? |
| 10:27.16 | MarKsaitis | I suppose them all :) |
| 10:27.37 | kaldemar | "aptitude show libsrtp0" will tell you. |
| 10:28.21 | MarKsaitis | thanx :) |
| 10:28.40 | MarKsaitis | I bet I only need headers and library |
| 10:29.10 | MarKsaitis | YES. Packages worked. Love it :) less mess with source |
| 10:32.00 | linocisco | <PROTECTED> |
| 10:39.01 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 10:39.09 | linocisco | <PROTECTED> |
| 10:43.09 | tparcina | I need to enable appropriate logging so that I gather enough information why some calls get busy signal. |
| 10:43.45 | tparcina | We are using several GSM gateways for outbound calls to mobile phones. |
| 10:44.56 | tparcina | Sometimes, in peek hours, we experience that, more than usual, we can't establish the phone call. |
| 10:45.23 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 10:45.36 | tparcina | In CDR I see those phonecalls with status BUSY. |
| 10:45.57 | Greenlight | Enable logging and check the logs of the calls that failed. |
| 10:46.04 | Greenlight | BUt GSM gateways ewww |
| 10:47.02 | tparcina | Thank you, is there any special options that I should use when I enable logging? |
| 10:47.16 | tparcina | And I don't understand this one: "BUt GSM gateways ewww". |
| 10:48.02 | Greenlight | I'm not a fan of the technology |
| 10:48.20 | Greenlight | And find in practice the cons outweigh the pros |
| 10:49.33 | kaldemar | tparcina: you could find out what happens in your dialplan and use app Log to output the reason in asterisk logs. |
| 10:49.51 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:50.05 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:51.57 | tparcina | kaldemar: Thank you for suggestion. I'll search for solution in that direction. |
| 10:53.17 | tparcina | Greenlight: What do you find as a cons for using gateways? |
| 10:54.36 | Greenlight | In my experience, on top of problems such as these, presenting a single number when calling out (the number of the SIM used gets presented) meaning that inbound calls back are hard to control. Often carriers apply "fair usage" policies which means that I have to keep track of usage and route to other sources accordingly |
| 10:56.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 10:59.52 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 11:01.48 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 11:03.59 | tparcina | Greenlight: Thank you. |
| 11:04.44 | Greenlight | Your mileage may vary though. I guess a lot depends on where in the world you are as well |
| 11:04.59 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:13.13 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:32.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.39 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 11:37.49 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 11:41.50 | *** join/#asterisk blackbag (~blackbag@gateway/tor-sasl/blackbag) |
| 11:42.38 | *** join/#asterisk MxG (~quassel@sysadmin1.lon.atpla.net) |
| 11:49.11 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:49.45 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 11:55.59 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 12:01.23 | transfinite | I'm having problems with DTMF recognition from analog phones during the "echo test" application. |
| 12:07.22 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 12:08.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:08.30 | mirela666 | transfinite: are they not detected at all or selectivly detecte? |
| 12:10.29 | *** join/#asterisk msaraiva (~msaraiva@81.148.8.239) |
| 12:16.25 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 12:18.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 12:19.25 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:24.13 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:27.17 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 12:27.55 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 12:29.17 | *** join/#asterisk CommaCrazy (1000@87.250.37.53) |
| 12:32.33 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cktbzqfbuyqjyclp) |
| 12:35.09 | CommaCrazy | hi all, question, I made a trunk form Topex to Trixbox and in the asterisk full log I get this on a call http://pastebin.com/ufbfGgrS and I can't figure out why it does not let me dial the extension in Trixbox |
| 12:36.47 | WIMPy | ~trixbox |
| 12:36.48 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 12:36.51 | Greenlight | Trixbox isn't asterisk and isn't supported here |
| 12:36.56 | Greenlight | Or that ^^ |
| 12:37.11 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 12:37.32 | [TK]D-Fender | CommaCrazy: Because it isn't going to the right context and you haven't permitted anonymous calls |
| 12:37.35 | WIMPy | It should also tell to try in #trixbox. |
| 12:37.48 | [TK]D-Fender | CommaCrazy: is "5060" indeed the peer name you gave it? |
| 12:37.54 | CommaCrazy | no |
| 12:37.59 | CommaCrazy | it's not |
| 12:38.10 | [TK]D-Fender | CommaCrazy: enable SIP debug and paste a new call |
| 12:38.21 | CommaCrazy | WIMPy, I did aski in #trixbox |
| 12:38.27 | CommaCrazy | ok [TK]D-Fender will do |
| 12:38.31 | CommaCrazy | thank you |
| 12:38.43 | [TK]D-Fender | #trixbox is dead in just about every way... |
| 12:39.02 | [TK]D-Fender | expect for the poor schmusck somehow still using it or thinking that its relevant |
| 12:40.21 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 12:40.22 | file | remembers the Trixbox booth babes at a Phoenix Astricon |
| 12:41.59 | MLNoah | i'm working on setting up an Asterisk 11.4.0 server using dynamic Realtime, and getting a little confused by the documentation/samples. The sample file references a 'sippeers' table but no sipusers or sipfriends. the Wiki article (from dec 2011) references 'sippeers' and 'sipusers' in the main article, then 'sipfriends' in the sample table structure. which one is right? is there a better... |
| 12:42.01 | MLNoah | ...place I should be looking at for Realtime configuration help? |
| 12:43.44 | CommaCrazy | hehehe, I know trixbox is dead but I still have some users who are connected to this PBX that I manage who still use it... still have to give them service :), thank you [TK]D-Fender it was the context... my head kind stopped working after I managed to get Topex part of the trunk working.... dont ever use Topex it's the worst. |
| 12:43.52 | CommaCrazy | so thanx [TK]D-Fender :) |
| 12:45.36 | *** part/#asterisk Matthias (~Matthias@bnc1.ip6.dirlinger.at) |
| 12:50.09 | tparcina | How to update /DEVICE/350/user value in database? |
| 12:50.15 | *** join/#asterisk Matthias (~Matthias@bnc1.ip6.dirlinger.at) |
| 12:51.01 | tparcina | I have tried database put /DEVICE/350/user 760 but I receive the error that I should enter database put <family> <key> <value> |
| 12:51.22 | MLNoah | should be database put DEVICE/350/user 760 |
| 12:51.23 | tparcina | Do I have two family or two keys here? |
| 12:52.03 | tparcina | MLNoah: It doens't work. |
| 12:52.26 | tparcina | I still get message: "Usage: database put <family> <key> <value>" |
| 12:52.58 | MLNoah | or, in dialplan, Set(DB(DEVICE/350/user)=760) |
| 12:53.11 | kaldemar | tparcina: you should divide /DEVICE/350/user into family and key. |
| 12:53.15 | MLNoah | er, sorry, DEVICE 350/user 760 |
| 12:53.23 | MLNoah | database put DEVICE 350/user 760 |
| 12:53.55 | tparcina | That does it. ;) |
| 12:54.03 | tparcina | Thank you! :) |
| 12:54.54 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 12:55.12 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 12:56.05 | *** join/#asterisk serafie (~erin@nat/digium/x-celgbekuabytuljf) |
| 12:59.11 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 12:59.32 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:01.07 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 13:01.43 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:02.26 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:03.40 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:07.30 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:10.31 | tparcina | If in one context I include two other context, and both of included context have s in it, which one will be executed? |
| 13:11.25 | jmetro | whichever one you goto |
| 13:12.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 13:12.35 | WIMPy | the one included first. |
| 13:12.36 | jmetro | [a] includes [b] and [c] and if you goto(a,s,1) it will goto context a, if A has a goto(b,s,1) it will go to b |
| 13:14.28 | tparcina | jmetro: Then what is the point of including the context? |
| 13:14.49 | jmetro | code organization, and what WIMPy said, if your [a] doesnt have [s] it will drill down. |
| 13:14.56 | jmetro | s, not [s] |
| 13:16.19 | kaldemar | jmetro: [a] does not have to have any goto's. |
| 13:16.25 | WIMPy | Included context are searched in the order listed and only if there was no match so far. |
| 13:18.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 13:19.19 | jmetro | kaldemar: also true, i just mean to drill down manually, to go to [a] then [b] and maybe execute code between them |
| 13:19.40 | [TK]D-Fender | [09:14]tparcinajmetro: Then what is the point of including the context? <- you didn't say if there was anything else in there |
| 13:20.11 | [TK]D-Fender | tparcina: and the one that gets executed is the FIRST one you included |
| 13:22.46 | tparcina | OK guys, thank you all for help. |
| 13:35.51 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ngmypakqwkqskgeo) |
| 13:35.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:37.54 | Greenlight | I've an odd issue occuring. I use a Local channel to "whisper" audio to a channel; so I originate a call between an extension which uses Playback, and the application ChanSpy. The audio gets whispered fine. Most of the time once playback ends the local channel connected to ChanSpy also ends, however, in some cases the ";1" part of the Local channel hangs around still in ChanSpy, even though |
| 13:37.54 | Greenlight | the ";2" of the Local channel has hungup. Any ideas what might cause this? |
| 13:40.45 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:43.25 | *** join/#asterisk navaismo (~navaismo@189.191.248.18) |
| 13:51.06 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:51.35 | *** join/#asterisk zoid_ (~awainer@181.29.125.3) |
| 13:52.49 | *** join/#asterisk itgrl (~itgrl@72.242.229.7) |
| 13:53.39 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 13:56.14 | *** part/#asterisk mihamina (~mihamina@41.188.46.121) |
| 13:56.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.147) |
| 14:02.57 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 14:03.42 | Captain_Proton | Anyone know a good replacement for astassistant? |
| 14:04.18 | Captain_Proton | I know hud and isymphony |
| 14:05.16 | [TK]D-Fender | Depends what you want out of it. |
| 14:05.46 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:05.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.48 | jmetro | i would wait until isymphony 3 comes out and go with that probably |
| 14:06.09 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:08.34 | Captain_Proton | Mostly queue support able to pause agents and possibly transfer via drag and drop |
| 14:09.18 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 14:09.24 | jmetro | isymphony is good at that. |
| 14:10.07 | navaismo | Monast maybe |
| 14:10.47 | *** part/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 14:10.51 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 14:12.45 | Captain_Proton | Monast that a new one I have not heard of that |
| 14:13.24 | protocolus | howdy everyone |
| 14:14.49 | transfinite | mirela666: they are detected fine during dialing, but when the echo test application starts, it only does "DTMF begin passthrough '#' on DAHDI/1-1". There's no end passthrough, and no further DTMF tones get recognized. But the echo test continues to run until hangup. |
| 14:16.56 | protocolus | anyone knkow of a way to detect when a call file is dialing a disconnted number? |
| 14:17.25 | protocolus | it just keeps redialing until it expires |
| 14:18.03 | igcewieling | protocolus: on any interface except analog FXO that is the default |
| 14:20.02 | [TK]D-Fender | Well ... DAHDI common FXO practice anyway... |
| 14:20.04 | protocolus | igcewieling: so no way to catch it? Maybe trying to process the voice on the channel that gives that message? |
| 14:20.30 | igcewieling | protocolus: use a different interface like PRI or SIP. |
| 14:20.56 | protocolus | igcewieling: i'm using google sip actually |
| 14:21.23 | Qwell | ...Google SIP? |
| 14:21.42 | protocolus | yeah the google voice sip interface |
| 14:21.45 | igcewieling | I didn't know google used SIP, I thought they used XMPP. Sorry, I can't help you with google. |
| 14:21.53 | Qwell | umm |
| 14:22.01 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-144-33.user.veloxzone.com.br) |
| 14:22.18 | protocolus | igcewieling: ok gotcha i have a PRI i'll try routing the calls thru that |
| 14:22.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:22.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:22.45 | protocolus | igcewieling: will asterisk update my call file and kick it out? |
| 14:22.47 | igcewieling | protocolus: Dial will set the variable HANGUPCAUSE |
| 14:23.05 | protocolus | igcewieling: oh perfect that's what i was looking for |
| 14:23.25 | Qwell | protocolus: I'm pretty sure Google Voice doesn't support SIP. |
| 14:23.33 | igcewieling | protocolus: this should be documented in the docs for Dial as well as the Asterisk book. It is a basic question. |
| 14:23.50 | protocolus | Qwell: hmm i don't know it can with support in PIAF |
| 14:24.15 | protocolus | Qwell: maybe it is xmpp begind the scenes |
| 14:24.21 | igcewieling | PIAF? That isn't even really "asterisk" |
| 14:24.25 | file | Google Voice does XMPP, Asterisk does the protocol conversion |
| 14:24.39 | igcewieling | At least for another few weeks |
| 14:24.54 | protocolus | igcewieling: what's bad about PIAF? |
| 14:25.12 | Qwell | ~freepbx |
| 14:25.12 | 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 |
| 14:25.15 | Qwell | protocolus: that |
| 14:25.43 | [TK]D-Fender | GV is dumping XMPP, no? |
| 14:25.47 | igcewieling | Qwell: the freepbx people don't want PIAF users either |
| 14:26.15 | igcewieling | [TK]D-Fender: that is what was announced, yes. 8-| |
| 14:26.23 | protocolus | aww well i used to run Druid :D |
| 14:27.00 | [TK]D-Fender | PIAF does still pretty much run base-line FreePBX IIRC |
| 14:27.14 | [TK]D-Fender | Elastix was the one that diced everything up |
| 14:33.26 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:33.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:36.20 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 14:39.24 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 14:40.18 | *** mode/#asterisk [+b caterwaul!*@*] by Qwell |
| 14:40.23 | igcewieling | That is LOW. Spam offering to sell me the list of users subscribed to the cisco users mailing list. |
| 14:40.33 | Qwell | igcewieling: awesome |
| 14:40.46 | msaraiva | Indeed, PIAF is basically FreePBX. |
| 14:40.52 | igcewieling | Qwell: not from the Crazy One, in an e-mail I got |
| 14:40.58 | *** mode/#asterisk [-b caterwaul!*@*] by Qwell |
| 14:41.01 | msaraiva | protocolus: but unfortunatelly you won't find support for it here. |
| 14:41.23 | Qwell | igcewieling: yeah I figured. unrelated. :p |
| 14:41.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:41.58 | msaraiva | protocolus: But Google Voice is supported through XMPP. That's what PIAF/FreePBX/Asterisk uses. |
| 14:42.09 | protocolus | msaraiva: gotcha |
| 14:42.24 | msaraiva | protocolus: Using chan_motif |
| 14:42.58 | igcewieling | msaraiva: for a few more weeks at least |
| 14:43.25 | msaraiva | igcewieling: Do you have any idea of what protocol they're going to use next? |
| 14:43.57 | igcewieling | msaraiva: I do not and I do not care. |
| 14:44.26 | msaraiva | It's a shame...ppl spent countless hours developing chan_motif, and now this. |
| 14:44.40 | igcewieling | I've had the same ITSP since 2003, they don't change their protocols and cause trouble -- unlike Google and Microsoft (skype) |
| 14:45.01 | msaraiva | You may not care, but there's a lot of ppl interconecting Asterisk with GV. |
| 14:45.04 | msaraiva | Me included. |
| 14:45.08 | igcewieling | msaraiva: I feel sorry file, not for the users. |
| 14:45.09 | msaraiva | It's cheap. |
| 14:45.30 | Qwell | msaraiva: Feel free to yell at Google for us. |
| 14:45.33 | igcewieling | msaraiva: so is my ITSP. My ITSP is not free, but it is cheap |
| 14:46.07 | msaraiva | Good for you. Haven't found anything that suited my needs better than GV, yet. Time to start searching again, i guess. |
| 14:46.27 | msaraiva | Qwell: i actually did... |
| 14:46.54 | Qwell | "Please don't stop me from using this service I don't pay for." unfortunately doesn't go over very well. |
| 14:47.15 | yano | unless you are using Google Reader :-p |
| 14:47.21 | igcewieling | Under $3/month for the DID and e911 service + 1 cent/min usage |
| 14:47.31 | Qwell | yano: Did they reverse that decision? |
| 14:47.40 | yano | Qwell: sadly, nope :-( |
| 14:47.48 | yano | i don't use it, but i know a lot of people who did |
| 14:47.56 | Qwell | Then, it doesn't go over very well there either then. :P |
| 14:48.03 | Qwell | s/then// |
| 14:48.05 | yano | i use irc bots that read rss feeds and spit out in specific channels on freenode, for me |
| 14:48.21 | yano | i may have an irc addiction, :-p |
| 14:48.45 | Qwell | * yano (yano@freenode/staff/yano) has joined #asterisk |
| 14:48.45 | Qwell | nah! |
| 14:48.54 | msaraiva | Qwell: i do buy GV credit pretty often. |
| 14:49.02 | yano | Qwell: :-p |
| 14:49.05 | msaraiva | So in a sense, i pay for it. |
| 14:49.54 | igcewieling | yano: can you kick caterwaul into the next dimension (or maybe the afterlife) for us? |
| 14:51.10 | jmetro | hes back again? |
| 14:52.33 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 14:52.38 | yano | we are taking a look at it |
| 14:53.21 | igcewieling | jmetro: I saw a ban. |
| 14:53.28 | eirirs | ... coming? |
| 14:53.28 | eirirs | :P |
| 14:53.59 | igcewieling | (10:40:18 AM) mode (+b caterwaul!*@*) by Qwell |
| 14:54.13 | Qwell | igcewieling: That one turned out to be unnecessary. |
| 14:54.19 | drmessano | I've been reading up on the Google XMPP developments, but I don't see where they're dropping XMPP altogether. What am I missing? |
| 14:54.42 | drmessano | Everything I see is related to Hangouts and dropping XMPP support to that part of the Borg |
| 14:54.47 | jmetro | googles release saying theyre dropping xmpp |
| 14:55.02 | drmessano | Link? |
| 14:55.07 | kaldemar | they announced that hangouts does not support XMPP. that's it. |
| 14:55.23 | drmessano | kaldemar, that's all I see too |
| 14:55.25 | mmlj4 | me no understand dropping things people rely upon |
| 14:55.33 | mmlj4 | solution: stop relying upon google |
| 14:55.47 | jmetro | embrace, extend, extinguish. apple has been doing it for years which is why they are an awful company. |
| 14:56.03 | kaldemar | that's probably a step towards dropping XMPP support since they encourage people to move to hangouts. |
| 14:56.16 | drmessano | It looks to me like they're making Hangouts a proprietary island.. I don't see where they are killing off XMPP anywhere else. Specifically no announcement of Google Talk's death |
| 14:56.32 | mmlj4 | I use their caldav, which I hear is also to be axed |
| 14:56.45 | Qwell | drmessano: well, there was some conversion of Talk > Hangouts, which apparently caused problems talking to Talk users. I never had that issue though. |
| 14:56.47 | drmessano | They're ditching caldav? |
| 14:57.04 | mmlj4 | to my knowledge, yes |
| 14:57.06 | zoid_ | Hi, I'm trying to run an AGI script using the full path, but get "File does not exist" error. The file exists, running manually works, where should I look? |
| 14:57.31 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 14:57.35 | zoid_ | asterisk version is 1.8.10, script is in python |
| 14:57.46 | drmessano | Qwell, yeah.. That sounded like a kludge anyway.. and it looks like they decided to not waste time making XMPP work with *Hangouts*. |
| 14:58.01 | drmessano | But I don't see anything between the lines |
| 14:58.04 | mmlj4 | copy the path and file from your script, then paste it into an ls |
| 14:58.10 | mmlj4 | ls <paste> |
| 14:58.25 | mmlj4 | if it's there, well... |
| 14:59.04 | zoid_ | mmlj4: I forgot to mention something important: it has a space in it, it goes like AGI(/path/to/the/interpreter /path/to/script) |
| 14:59.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:59.18 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:59.19 | mmlj4 | oh |
| 14:59.26 | mmlj4 | escape the spaces |
| 14:59.26 | zoid_ | exten => 1234,n,AGI("/opt/voicesms-env/bin/python /usr/share/asterisk/agi-bin/voicesms/bin/voicesms_service.py,18") |
| 14:59.34 | zoid_ | backslash? |
| 14:59.35 | mmlj4 | but honestly, lose the spaces |
| 14:59.49 | mmlj4 | wait,... what? |
| 15:00.04 | *** join/#asterisk FonJockey (~FonJockey@c-24-30-113-162.hsd1.ca.comcast.net) |
| 15:00.29 | mmlj4 | why are you telling it the interpreter? do that in the first line of your script: #!/path/to/interpreter |
| 15:01.07 | mmlj4 | and what's ",18"? |
| 15:01.08 | [TK]D-Fender | Lose the quotes... |
| 15:01.12 | zoid_ | mmlj4: I'm doing that because I use virtualenv and on the developers machine the virtualenv is in a different path, probably a bad idea anyway |
| 15:01.18 | zoid_ | 18 is an argument my script needs |
| 15:01.23 | mmlj4 | it's bad idea |
| 15:01.49 | msaraiva | Ok, i just read somewhere that Google is dropping XMPP *only* for server to server connections (federation) |
| 15:01.55 | msaraiva | Client to server will continue working. |
| 15:02.24 | zoid_ | mmlj4: it's kind of a trade off, I wouldn't want to have to touch the script if the path to the interpreter changes, but it's not completly unreasonable |
| 15:02.27 | *** join/#asterisk windback (~windback@190.123.122.162) |
| 15:02.50 | mmlj4 | if you have to change one thing occasionally, change the right thing |
| 15:03.33 | zoid_ | the other option is to wrap it in a shell script |
| 15:03.43 | mmlj4 | yep |
| 15:03.45 | zoid_ | but that would be a waste of resources |
| 15:04.09 | igcewieling | zoid_: I've never called an AGI like that, I don't know if it is even supported |
| 15:04.12 | windback | I'm looking for a good python API to asterisk manager interface. I was reading pyst and py-asterisk. Both seems to be unmantained. Any recomendation? |
| 15:04.12 | mmlj4 | not much, if it's run infrequently |
| 15:05.10 | mmlj4 | the ,18 thing still bugs me, but that's because I do Perl, and would do " 18", then a shift in my script |
| 15:05.38 | zoid_ | igcewieling: it seems is not, should I report a bug about this? |
| 15:05.54 | zoid_ | windback: I'm using pystrix, I like it |
| 15:06.04 | mmlj4 | but I would think that your /path/to/your/script,18 would indeed not be found on the filesystem |
| 15:06.07 | zoid_ | sorry, I'm using it for AGI, but it has AMI too |
| 15:06.37 | zoid_ | forget about the quotes, it was a test I did, it still doesn't work without them |
| 15:07.28 | *** join/#asterisk ClockworkAnt (~Clockwork@www.4comgroup.co.uk) |
| 15:07.51 | zoid_ | mmlj4: I'll follow your advice and just change the interpreter in the #! line |
| 15:08.28 | ClockworkAnt | Good afternoon all, I'm looking into sip subscriptions and publishing, does asterisk support these features? (excuse my ignorance I'm developing a client for an asterisk pbx and dont have control over the pbx myself) |
| 15:08.42 | mmlj4 | igcewieling: funny story... I got called by a staffing company, local dev shop needs 3 weeks' asterisk development for a project... the staffing dude googles new orleans asterisk, finds 2 hits, calls me and sends me to the dev shop... apparently the devs have never heard of google |
| 15:09.45 | jmetro | oh i love staffing companies |
| 15:09.56 | jmetro | trying to explain your qualifications to them XD |
| 15:10.27 | windback | zoid_, Thanks. I'm going to see it |
| 15:11.12 | file | is qualified in nothing, nope, nothing |
| 15:11.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:11.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:16.06 | msaraiva | drmessano: Google's CalDAV API will still be available, but just for whitelisted developers... |
| 15:16.58 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 15:17.17 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 15:19.26 | mmlj4 | msaraiva: URL? |
| 15:20.59 | mmlj4 | grr... I'm going to have to put up a caldav server, apparently |
| 15:21.54 | msaraiva | mmlj4: http://googleblog.blogspot.co.uk/2013/03/a-second-spring-of-cleaning.html |
| 15:22.40 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 15:25.03 | mmlj4 | danke |
| 15:25.25 | msaraiva | You're welcome. |
| 15:26.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 15:27.06 | *** join/#asterisk gorkish (~gork@fw.energynet.com) |
| 15:29.56 | gorkish | Hello; I recently upgraded an installation from 1.8.11 to 11.4.0 and experienced a regression with Polycom phones and SRTP (SRTP auth failure after call resumes from hold).. Before digging into it to determine the scope of the problem I wanted to ensure I am using the 'definitive' version of libsrtp as there appear to be many variants. |
| 15:30.26 | drmessano | We remain committed to supporting open protocols like CalDAV. <-- Obviously not |
| 15:30.46 | drmessano | Most developers’ use cases are handled well by Google Calendar API, which we recommend using instead. <-- YUUUUP |
| 15:31.59 | gorkish | The 1.8.11 install was working with 1.4.4+20100615 as packaged by Debian/Ubuntu. I have tried this version as well as 1.4.5 from https://github.com/cisco/libsrtp and both experience the same regression with 11.4.0 |
| 15:32.11 | drmessano | "We support open standards. In other words, we sit around and drink beers and talk about how open standards are neato. However, we don't believe in actually implementing them or retaining their use at Google" |
| 15:33.41 | drmessano | What happened is that all the cool "Lets take all this money and create a utopia" crap of their youth has gone and now they're slowly becoming the next Microsloth |
| 15:34.13 | coppice | they're not bad. they're just chaotic |
| 15:36.52 | drmessano | I think they've grown out of being young and hip and are interested in maximizing earnings and protecting assets. |
| 15:37.40 | coppice | well, if they were less chaotic more of the things they do would be effective earners |
| 15:37.41 | drmessano | They've very much gone from "We're implementing this protocol and this API because then we can hook into everything", and now you can hook whatever you want into google as long as it's a google product |
| 15:38.41 | gorkish | that all works fine and dandy until they cancel the product; then whatever you have implemented against their proprietary system has to be completely redone |
| 15:39.08 | *** join/#asterisk Xaviertoor (~Xaviertoo@189.52.87.66) |
| 15:39.10 | drmessano | I'm not so much impressed by this whole XMPP/hangout thing as I was them ditching ActiveSync for ALL Gmail accounts, where they had implemented it for ALL Gmail accounts as a "feature". |
| 15:39.37 | jmetro | Is there any way to turn asterisk hold music into the "call waiting beeps" |
| 15:40.05 | pabelanger | jmetro: just update the moh audio, with audio of beeps |
| 15:40.06 | drmessano | Now if you want ActiveSync you need a PAID Google Apps account. Otherwise you're stuck with the GMail app or Fscking IMAP |
| 15:40.11 | gorkish | At least with AS they were paying Microsoft a license fee so I could see how that would be a drain on the free service |
| 15:40.15 | jmetro | pabelanger: oh god @.@ |
| 15:40.37 | drmessano | gorkish, that's understandable.. but a fee they were once happy to pay |
| 15:40.43 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 15:40.44 | pabelanger | jmetro: Asterisk will not generate them, so you need an actual recording |
| 15:41.02 | drmessano | gorkish, so they ditch it, and then tell everyone "Use our shitty apps.. or hey, IMAP." |
| 15:41.10 | drmessano | IMAP on a mobile device is hell on a battery |
| 15:41.25 | jmetro | imap is also crap for syncing |
| 15:41.33 | msaraiva | As is AS. |
| 15:41.48 | gorkish | AS is https long polling. s/IMAP IDLE is about equivalent if the device supports it (most do not) |
| 15:41.54 | msaraiva | (batterywise, i mean) |
| 15:42.27 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:43.05 | drmessano | Well, the way it stands now, you can continue to use it with your CURRENT device. You're next device won't work |
| 15:43.06 | Greenlight | From looking at CDR's, how can I tell if a call which entered a queue was actually answered or not? |
| 15:43.26 | drmessano | I tried switching a couple accounts to IMAP, and my battery life was cut by 80% |
| 15:43.35 | drmessano | Your* |
| 15:44.43 | drmessano | So that's been boiling under the surface for me for months. If I drop my phone just right, I am done using my GMail and my Apps accounts on my mobile device |
| 15:45.10 | drmessano | Hopefully this XMPP/Hangouts announcement doesnt turn into ANYTHING XMPP. |
| 15:45.12 | [TK]D-Fender | Greenlight: You don't.... that's what the queue log is for |
| 15:45.29 | jmetro | i dont know what the problem was, the default droid app on my account does exchange just fine |
| 15:45.57 | drmessano | How long have you had it? |
| 15:46.13 | jmetro | <PROTECTED> |
| 15:46.37 | drmessano | [11:43:05] <drmessano> Well, the way it stands now, you can continue to use it with your CURRENT device. Your next device won't work |
| 15:46.59 | jmetro | from what i remember the activesync drop was done about a year ago, so my 1 month phone wouldnt work? |
| 15:47.48 | drmessano | The announcement was in Dec 2012. |
| 15:48.20 | jmetro | so my april phone shouldnt be getting all my emails essentially? |
| 15:48.22 | drmessano | http://support.google.com/a/bin/answer.py?hl=en&answer=2716936 |
| 15:48.28 | drmessano | Thats the announcement |
| 15:48.31 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 15:49.28 | Greenlight | [TK]D-Fender: Ahh okay I need to check the queue log as well. Thanks |
| 15:49.52 | msaraiva | drmessano: Are you on Android? I've used AquaMail (supports IMAP IDLE) for quite a long time, and battery life was just fine on my GS2 and GS3. |
| 15:50.01 | drmessano | Nope, IOS |
| 15:50.03 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 15:50.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:50.38 | msaraiva | Oh, i see. |
| 15:51.04 | jmetro | IOS is silly, youre silly. |
| 15:51.05 | msaraiva | Well, i have no experience with IMAP on iOS. But i can assure you that it worked just fine for me on Android. |
| 15:51.24 | msaraiva | Didn't kill my battery at all... |
| 15:51.28 | jmetro | android-fives msaraiva |
| 15:51.40 | msaraiva | android-fives jmetro back |
| 15:52.20 | msaraiva | Anyone here using a did from DIDWW? |
| 15:53.06 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 15:53.08 | jmetro | msaraiva: assuming we both have near-field devices you just got shared all my awesome pics |
| 15:53.20 | msaraiva | lol |
| 15:53.23 | msaraiva | Yeah, GS3 here. |
| 15:53.48 | jmetro | razr maxx hd but im pondering the HTC One / S4 / Nexus |
| 15:53.59 | jmetro | [unlocked beauty] |
| 15:53.59 | drmessano | iOS doesn't support IMAP IDLE. But even still.. Why do I need to set up IMAP, Google CalDAV, and CardDAV for ONE ACCOUNT? |
| 15:54.11 | jmetro | because youre on ios |
| 15:54.24 | drmessano | What about on my desktop? |
| 15:54.32 | jmetro | exchange ? |
| 15:54.38 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 15:54.45 | msaraiva | "cloud apps" |
| 15:54.46 | msaraiva | lol |
| 15:55.08 | drmessano | exchange what? |
| 15:55.11 | jmetro | i like how people stopped using the term "server" and started using the term "cloud". like its different. |
| 15:55.14 | msaraiva | jmetro: Go for S4. And i9505 variant, if you care for AOSP. |
| 15:55.19 | drmessano | If there's no AS, I can't use AS |
| 15:55.31 | jmetro | desktop -> outlook |
| 15:55.44 | msaraiva | jmetro: that's the reason for the "" and the lol. ;) |
| 15:55.57 | SuperNull | Hey guys, is there any special ownership permissions that need to be set when pushing custom msg001.txt files into a voicemail directory ? |
| 15:56.03 | drmessano | jmetro, and how do I magically connect Outlook to my google account with AS if THERE IS NO AS? |
| 15:56.18 | drmessano | outlook -> derp |
| 15:56.26 | jmetro | why are you using google account ? exchange -> outlook |
| 15:56.40 | drmessano | I am so lost here |
| 15:56.48 | drmessano | We're talking about GOOGLE ACCOUNTS |
| 15:57.04 | drmessano | What the heck does Exchange have to do with anything here |
| 15:57.30 | jmetro | maybe im lost too, i never thought of gmail or google accounts as business accounts |
| 15:57.41 | msaraiva | SuperNull: the user that asterisk process is running as has to have access to the files. |
| 15:57.43 | drmessano | Google Apps? |
| 15:57.53 | drmessano | You're never heard of Google Apps for Business? |
| 15:58.37 | jmetro | hm. nope |
| 15:58.40 | *** join/#asterisk gtTuna (Tuna@d149-67-253-75.try.wideopenwest.com) |
| 15:58.42 | drmessano | wow |
| 15:59.26 | jmetro | so basically paying google to do all your file server stuff it looks like |
| 15:59.36 | drmessano | No.. |
| 15:59.42 | drmessano | Mail, calendar, contacts, etc |
| 15:59.55 | jmetro | thats all exchange stuff |
| 16:00.25 | drmessano | I don't consider Exchange to be a file server |
| 16:01.02 | jmetro | so youre offloading exchange and file server then |
| 16:01.51 | drmessano | I don't use Google Apps/Drive as a file server |
| 16:02.21 | jmetro | "docs" is certainly a file server |
| 16:02.33 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 16:02.45 | drmessano | Whatever |
| 16:03.56 | igcewieling | the google thing I use is search and gmail, but only for the throwaway e-mail address needed by Android stuff |
| 16:04.07 | jmetro | point being you offloaded your IT into a third-party company that is focused on providing these services for the world, not to you explicity. |
| 16:04.30 | igcewieling | jmetro: you still need local people |
| 16:05.11 | jmetro | correct, so now you have two IT companies, one of which is willing to do anything for you, but cannot access your mail or file server. |
| 16:05.16 | igcewieling | At least half of our new customers try to draft us (their PHONE provider) as their IT staff, since they have none of their own. |
| 16:06.14 | drmessano | jmetro, I don't know of many small businesses, clubs, and small non-profits that can afford to maintain an Exchange environement. This is really getting silly |
| 16:06.19 | igcewieling | jmetro: yup! two IT organizations -- one which can do local stuff and might care, and the other which handles your e-mail and fileservers and doesn't give a rats ass about you as a customer. |
| 16:07.04 | jmetro | drmessano: correct, they should not have local servers, they should be going hosted on one of those local IT companies. Which most likely use exchange. |
| 16:07.44 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:908e:df4a:185f:1f13) |
| 16:07.49 | jmetro | rather than a globally based hosting company who lets you rent space on a giant magnetic spindle and leaves the rest up to you |
| 16:08.04 | jmetro | [as much as i like google] |
| 16:08.05 | drmessano | I realize the only two correct answers here are "IOS Sucks" and "Use Exchange", but Google Apps and GMail are products that many people rely on and have heavily invested in. Cutting off something like ActiveSync and forcing many users to jump to buggy IMAP implementations or Google's shitty Apps is really crap |
| 16:08.09 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:09.57 | igcewieling | drmessano: nah, the only two correct answers are "IOS is OK" and "exchange is the problem, not the answer" |
| 16:11.25 | drmessano | igcewieling, well, every conversation about mobile devices seems to degrade down to Android being the bomb, and everything else sucking, including IOS. I wouldn't be caught dead with an Android phone, though if I had an Android phone and had an emergency, I probably would be. |
| 16:11.48 | *** join/#asterisk Scapal (~Scapal@host-85-201-20-117.brutele.be) |
| 16:11.53 | jmetro | drmessano: or if you had an iphone and were relying on it for directions. |
| 16:11.58 | drmessano | So the argument "Well, IMAP with Gmail works great on Android" is really beyond useless |
| 16:12.18 | drmessano | jmetro, do you actually have an iPhone? |
| 16:12.30 | jmetro | i did for about 3 months, replaced it with the razr maxx |
| 16:12.37 | jmetro | kept getting lost when going on site. |
| 16:12.41 | SuperNull | asterisk -rv |
| 16:12.44 | SuperNull | wrong window. |
| 16:12.48 | igcewieling | drmessano: Apple's phones lack a few things I require, but I don't think they "suck". My android phone is what I expected from a smart phone. Occasionally laggy interface, occasionaly crashes, but mosty works, |
| 16:12.59 | drmessano | The maps app in iOS works fine. Most of the bullshit about the maps app was overblown for media consumption... I rely on it heavily, and it works better than google maps |
| 16:13.12 | jmetro | ^ now i know we are being trolled |
| 16:13.14 | Scapal | Houston, I have a problem... calls are being added in front of the queue: |
| 16:13.17 | Scapal | <PROTECTED> |
| 16:13.17 | Scapal | <PROTECTED> |
| 16:13.34 | drmessano | jmetro, you're the only one here doing the trolling |
| 16:13.35 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 16:13.45 | drmessano | and quite frankly, i'm bored of it |
| 16:13.48 | Scapal | Any idea what could cause that? |
| 16:14.12 | pabelanger | Scapal: First call in, is not always first call out |
| 16:14.21 | pabelanger | found that out the hardware |
| 16:14.27 | pabelanger | hardway* |
| 16:16.02 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:16.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:16.15 | jmetro | so probably rewrite the queue app. |
| 16:16.22 | pabelanger | Scapal: not really. You can drop announcements, to help minimize it, but we are re-writing app_queue for our needs |
| 16:16.41 | pabelanger | Plan on talking about it at astricon, if the talk gets accepted |
| 16:17.51 | jmetro | if you re-write queue, please add a top-down functionalty :) |
| 16:18.11 | pabelanger | top-down? |
| 16:18.24 | Qwell | if you re-write app_queue, I'll help pay your asylum fees. |
| 16:18.29 | jmetro | rings one member until timeout, rings next member to timeout, etc.. always in a set order, not relying on penalties. |
| 16:18.50 | Qwell | jmetro: you mean like rrmemory? |
| 16:18.54 | pabelanger | Oh, ya. Version 1 is FIFO |
| 16:18.56 | Scapal | may the autofill have something to do with the behavior? |
| 16:18.59 | jmetro | for example, Front desk, secretary, boss. should always be in that order. |
| 16:19.06 | pabelanger | Scapal: yes, you can also disable that too |
| 16:19.28 | jmetro | Qwell: rr memory is a circle, i want a repeating line. |
| 16:19.30 | pabelanger | Scapal: but may increase your wait times |
| 16:19.44 | Qwell | circles are lines |
| 16:19.46 | Scapal | I'm using ringall strategy since it's more a 4 people team than a real call center |
| 16:19.49 | *** join/#asterisk vfabi (~fabi@host-static-92-115-178-18.moldtelecom.md) |
| 16:20.06 | Scapal | Qwell: blue is the new red :-/ |
| 16:20.35 | pabelanger | Qwell: not as much a re-write, but fresh start. Only doing very basic stuff right now |
| 16:20.41 | pabelanger | will see how it goes |
| 16:21.04 | jmetro | Its called a top-down strategy by a couple different pbx systems but asterisk cant duplicate it without severe headsmashing |
| 16:21.53 | pabelanger | Ideally, the business logic (or strategies) should be drop in plug ins, allow anybody to write there own method of matching |
| 16:26.38 | jmetro | like queue_topdown.qsm or something |
| 16:26.42 | jmetro | [queue strategy module] |
| 16:27.34 | pabelanger | something like that |
| 16:28.16 | *** join/#asterisk dr4c4n (dr4c4n@milliways.ramdrive.org) |
| 16:30.04 | SuperNull | hey ugys anyone use a linksys type ATA with multi-server asterisk and a dedicated voicemail server ? |
| 16:30.22 | SuperNull | im looking for people that have used the voicemail server options specifically of an ATA. (for notify subscription) |
| 16:34.14 | [TK]D-Fender | SuperNull: .. huh? |
| 16:34.53 | SuperNull | trying to fix this nat voicemail notification problem.. i might attempt to use the 'voicemail subscription' but im looking for anyone who has had success with this |
| 16:37.37 | igcewieling | SuperNull: that won't help |
| 16:38.03 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 16:38.14 | igcewieling | SuperNull: you're not using FreePBX and experiencing the FreePBX voicemail bug, are you? |
| 16:40.51 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 16:41.45 | SuperNull | nahhh this is ast 1.8.21 |
| 16:42.14 | SuperNull | igcewieling.. the problem i have with nat is we use a custom MWI generator.. that doesnt use asterisks port (5060) as the source. |
| 16:42.35 | SuperNull | so it wont pass nat correctly. everything else works. |
| 16:43.24 | igcewieling | SuperNull: I wish you the best of luck, but you should be using Asterisk's MWI support, not writing your own |
| 16:44.56 | SuperNull | i am attempting that but its not recognizing the msg0001.txt type files.. and it always sends out no mwi 0/0 .. |
| 16:46.38 | SuperNull | im just looking for options, my only working option right now is to provide these butt plugs 70 public ips for linksys handsets. that isnt much of an option |
| 16:46.52 | msaraiva | drmessano: Err, no way Apple Maps is better than Google Maps. And it's not Google's fault if iOS does not support such a basic feature as IMAP IDLE. |
| 16:47.07 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:47.21 | msaraiva | Ok, i bit the bullet and compiled asterisk again with MALLOC_DEBUG enabled. |
| 16:47.30 | msaraiva | Time to track down this memory usage problem... |
| 16:47.49 | SuperNull | sounds like fun msaraiva. which debugger ya using ? |
| 16:49.01 | msaraiva | Valgrind |
| 16:50.53 | SuperNull | hmm. i will remember that .. never heard of it before. most of my 'crazy' debugging was done on winderz. |
| 16:51.33 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 16:53.13 | igcewieling | SuperNull: then you are doing something wrong if Asterisk is not picking up the new messages. |
| 16:54.38 | igcewieling | why are you using an external voicemail system for asterisk anyway? |
| 16:55.13 | SuperNull | we have 10 asterisk servers. |
| 16:55.17 | SuperNull | for end users. |
| 16:55.51 | igcewieling | we have three. so what? |
| 16:56.41 | SuperNull | we dont store voicemail on the box they register to. they are not designed to hold storage like that. |
| 16:57.48 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 16:58.03 | igcewieling | SuperNull: the OUTBOUND MWI SUBSCRIPTIONS options won't work for you? |
| 16:59.31 | SuperNull | which release did that get added to ? |
| 17:00.04 | igcewieling | SuperNull: I saw it in the sip.conf.sample in asterisk-1.8.21.0 |
| 17:00.45 | SuperNull | our vm server is running 1.4 i believe. (yes i know upgrade.. blah blah) |
| 17:00.52 | igcewieling | Looks like 1.6 |
| 17:01.06 | SuperNull | damn. |
| 17:01.09 | SuperNull | alright. |
| 17:01.10 | SuperNull | well |
| 17:01.16 | SuperNull | i just wish i knew wtf i was doing wrong with the files. |
| 17:01.19 | igcewieling | read the info CAREFULLY, doesn't look like the MWI server even has to be asterisk |
| 17:02.08 | SuperNull | i want to setup a local vm on this just to see where it makes the msgtxt files |
| 17:03.51 | igcewieling | Are you SURE Asterisk only cares about the msgxxxx.txt file? |
| 17:04.04 | SuperNull | no. |
| 17:05.53 | SuperNull | based on scripts for mwi in a multi-server single vm envirnonment that does appear the only thing they are doign but.. there may be back end options that are different. |
| 17:05.54 | igcewieling | SuperNull: http://pastebin.ca/2380176 |
| 17:06.07 | igcewieling | SuperNull: maybe you should be looking at app_voicemail.c to see what it expects |
| 17:06.34 | *** join/#asterisk msaraiva (~msaraiva@host81-148-8-239.in-addr.btopenworld.com) |
| 17:06.50 | SuperNull | i am now, figured it would be known by someone in here. |
| 17:07.10 | igcewieling | SuperNull: no. what you are doing is crazy and nobody does it that way |
| 17:07.18 | igcewieling | (ok, ALMOST nobody) |
| 17:07.48 | SuperNull | sadly at the time some of these things were developed it was the only way. |
| 17:08.33 | igcewieling | and now there are better options |
| 17:12.17 | SuperNull | is there a mass method of this outbound mwi ? its obviously not feasable to set this up for 12,000 extensions manually. |
| 17:14.09 | jmetro | asterisk realtime does it automatically.. or are you in some crazy system |
| 17:14.33 | SuperNull | no, using realtime. |
| 17:14.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:14.57 | igcewieling | SuperNull: let me introduce you to the world of one-off-scripting..... |
| 17:21.02 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:23.32 | MLNoah | does anyone have a suggestion on a sane max_allowed_packet size for ODBC/MySQL voicemail storage? |
| 17:23.57 | jmetro | where is that setting? i can grab you one |
| 17:24.15 | MLNoah | etc/my.cnf, in cent-6 |
| 17:25.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 17:25.13 | jmetro | ive got 16M |
| 17:25.13 | *** join/#asterisk itgrl (~itgrl@72.242.229.7) |
| 17:25.56 | jmetro | itgrl: congratulations on geting your name back to normal |
| 17:26.20 | MLNoah | hm, the follow up question may be... has anyone gotten ODBC VM working with MySQL Cluster... |
| 17:26.25 | itgrl | jmetro: thanks. I just logged out on my other computer. |
| 17:26.43 | jmetro | itgrl: oh, i was going to say you could do the /deregister username secret i think |
| 17:27.12 | itgrl | well I needed to get on my work PC for something anyway. So I just did it that way. |
| 17:32.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.186) |
| 17:33.14 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 17:34.03 | SuperNull | igcewieling, based on the app_voicemail.c code it uses a function (when not odbc/imap) called __has_voicemail(...) which ONLY looks at the file count of the msg files and not the contents. |
| 17:36.37 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 17:38.28 | igcewieling | SuperNull: maybe the file your script creates has the wrong permissions? |
| 17:39.27 | SuperNull | thats what i was thinking but.. sadly no output from asterisk to indicate ... plus asterisk is running as root temporarily and it doesnt seem to work either. |
| 17:40.27 | SuperNull | is there a way from cli to see who is subscribed ? |
| 17:45.26 | SuperNull | part of me wants to stick additional debug code into this. |
| 17:47.15 | igcewieling | you mean something like "sip show subscriptions" |
| 17:48.43 | SuperNull | probably works. |
| 17:49.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.186) |
| 17:53.22 | jmetro | good luck weeding through sip show subscriptions.. |
| 17:53.39 | SuperNull | i have nothing. could be a realtime thing. |
| 17:53.43 | SuperNull | so that was easy. |
| 17:54.01 | jmetro | i have a majillion lines in mine and i run realtime.. |
| 17:54.06 | SuperNull | weird. |
| 17:54.12 | SuperNull | 1.8 ? |
| 17:54.14 | jmetro | 11 |
| 17:54.29 | jmetro | wait i think the one i just checked was a 1.8 |
| 17:55.06 | SuperNull | what the fuck. |
| 17:55.20 | SuperNull | the previous guy has no subscribe allowed in general config. |
| 17:55.27 | jmetro | fixed. |
| 17:55.53 | SuperNull | derp. |
| 17:57.38 | igcewieling | SuperNull: subscriptions have very little to do with voicemail in Asterisk |
| 17:57.46 | igcewieling | at least for the clients. |
| 17:58.00 | SuperNull | testing something. |
| 17:58.05 | igcewieling | asterisk sends UNSOLICITED notifys for VM, they are not dependent on the device subscribing to anything |
| 17:58.50 | SuperNull | in the code it references a subscribe requirement if using 'pollmailboxes=yes' |
| 18:02.22 | igcewieling | the only time I've seen clients sending a SUBSCRIBE is for BLF |
| 18:03.17 | SuperNull | yeah. just checking honestly. im still drilling into the source. |
| 18:20.03 | MLNoah | I'm setting up Asterisk 11.4 with ODBC Realtime and ODBC Voicemail. The SQL server is MySQL Cluster (5.5.30-ndb-7.2.12), and the ODBC connector installed on the asterisk boxes is the MySQL 5.2.5 from the MySQL website. Realtime is working just fine for both SIP and Voicemail users. When I try to leave a voicemail, i'm getting a "SQL Direct Execute failed!" error when it tries to store the... |
| 18:20.05 | MLNoah | ...voicemail. Any suggestions on what I'm missing? |
| 18:20.36 | *** join/#asterisk dr4c4n (dr4c4n@milliways.ramdrive.org) |
| 18:30.04 | SuperNull | do you allow update/delete/insert on the odbc user for voicemail ? |
| 18:30.35 | MLNoah | the odbc user should have full access, and i've been using that user via mysql to create/edit the tables |
| 18:31.51 | SuperNull | are you storing the audio file into mysql ? |
| 18:32.14 | MLNoah | trying, but it's failing. |
| 18:32.48 | SuperNull | see if you can turn on mysql debugging perhaps and get the query .. from that we can probably glean what is wrong (missing field or something ?) |
| 18:36.23 | MLNoah | hrm. it doesn't look like my google-fu is finding the right way to log the query |
| 18:36.43 | SuperNull | yeahhhh i fubar that too |
| 18:36.47 | SuperNull | <PROTECTED> |
| 18:37.05 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 18:37.51 | igcewieling | you can log queries, ask on #mysql |
| 18:43.28 | *** join/#asterisk navaismo (~navaismo@189.191.248.18) |
| 18:48.23 | MLNoah | hm. ok, queries are logging, but I only see the SELECT queries related to the voicemail box. the INSERT isn't logging |
| 18:49.57 | MLNoah | my ODBC user has GRANT ALL PRIVILEGES ON `asterisk`.* TO 'asterisk-user'@'%' |
| 18:50.10 | MLNoah | and the voicemail storage table is in the asterisk DB |
| 18:52.10 | SuperNull | hmm |
| 18:52.20 | SuperNull | googling provide any insight ? |
| 18:52.59 | MLNoah | the closest i've come to insight is a single digium forum post with no responses other than the OP, who indicated that max packet size might be causing the issue |
| 18:53.01 | MLNoah | but i've changed that |
| 18:53.17 | SuperNull | !!!!!!!!!! |
| 18:53.21 | SuperNull | igcewieling. |
| 18:53.23 | SuperNull | i got it. |
| 18:53.28 | SuperNull | damn thing is using the wrong context to find it. |
| 18:54.04 | SuperNull | i modded my app_voicemail to output the path its looking in, and its totally using default not the correct context from realtime. |
| 18:54.30 | igcewieling | welcome to realtime |
| 18:54.40 | SuperNull | yeah. |
| 18:54.41 | SuperNull | i know. |
| 18:54.55 | igcewieling | I STILL have get realtime VM to store greetings in ODBC, it will take the config and store the messages there, but not the greetins |
| 18:54.57 | SuperNull | i want to go opensips/freeswitch. |
| 18:55.06 | igcewieling | I still have NOT gotten... |
| 18:55.37 | SuperNull | lol now i have all these __HAS_VOICEMAIL_:: PATH /dfasd/asd/sadasd/asdas/ logs going every few seconds lol |
| 18:57.15 | MLNoah | well, FWIW, my ODBC voicemail storage doesn't work even if I turn off realtime for VM users. |
| 18:57.35 | MLNoah | oh. |
| 18:58.53 | MLNoah | asterisk is expecting column name `msg_id`. Whatever walkthrough I found this morning said I would need `msgid` |
| 18:58.55 | MLNoah | *sigh* |
| 19:00.15 | Qwell | https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage |
| 19:00.43 | SuperNull | NLNoah.. common realtime problem |
| 19:01.02 | SuperNull | and its different between 1.4->1.6->1.8 |
| 19:01.14 | SuperNull | going to 10,11 is .. also |
| 19:03.17 | MLNoah | so, i guess we chalk that up as an ID_10_T error |
| 19:03.23 | MLNoah | reading is hard >_> |
| 19:03.31 | SuperNull | its def expecting user@context for this to translate right.. how ever.. since we use phone #s for all things.. (bad i know) .. i can use default and be done with this. |
| 19:03.39 | SuperNull | damn words. |
| 19:03.43 | SuperNull | heh. |
| 19:08.39 | *** join/#asterisk Sha`Bren (~Sha`Bren@pdpc/supporter/active/shabren) |
| 19:13.11 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-99-246.dhcp.embarqhsd.net) |
| 19:15.21 | SuperNull | OHHHHHLYYYYYY SHIZZZZZ it works. |
| 19:15.44 | SuperNull | that only took 2 days + of tracking down the issue. (which was thought to be a software bug at first) |
| 19:25.56 | gorkish | anyone new around that can answer my SRTP question from earlier: what version of which libsrtp library should I be using w/ Asterisk 11? I am currently using https://github.com/cisco/libsrtp 1.4.5 which appears to be the current lineage of the original library but i want to rule that out before I debug my polycom SRTP hold issue |
| 19:26.30 | Qwell | gorkish: what, specifically, is the problem? |
| 19:27.40 | *** join/#asterisk miztic (~gerard@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 19:28.03 | gorkish | Qwell, after taking a call off of hold, I no longer have audio to or from the polycom (* is an RTP endpoint here) and I receive a message on the console of SRTP auth failure. |
| 19:28.22 | *** join/#asterisk areski (~areski@80.174.128.95.dyn.user.ono.com) |
| 19:28.29 | Qwell | and what version of Asterisk? |
| 19:29.17 | gorkish | It worked correctly with 1.8.11; does not work with 11.4.0; same libsrtp 1.4.4+20100615 as packaged in debian/ubuntu with both * versions |
| 19:29.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.186) |
| 19:30.45 | gorkish | however i did not want to start poking around in asterisk code until i determined which of the 3 or 4 different libsrtp codebases and which version was considered to be the de-facto standard to use |
| 19:31.13 | Qwell | The packaged version is fine. |
| 19:32.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:32.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:32.51 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:34.56 | *** join/#asterisk rocksfrow (~kyle@50-197-9-21-static.hfc.comcastbusiness.net) |
| 19:35.16 | rocksfrow | hey, i recently replaced an old asterisk 1.6 box with a newer box with 11 on it |
| 19:35.38 | gorkish | Qwell, OK I will start at that version and debug further from there |
| 19:35.40 | rocksfrow | is there any documentation on the way recordings are now stored? i had some scripts written which was relying on the old file structure |
| 19:35.57 | rocksfrow | now i see it looks like the recordings are stored in ../monitor/YYYY/MM/XX/. ? |
| 19:36.05 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 19:36.11 | rocksfrow | anyway to revert it to simply storing them all in the monitor folder? |
| 19:36.21 | rocksfrow | or if i can just find good documentation so I can update my scripts, thats fine too |
| 19:36.25 | rocksfrow | any help is MUCH appreciated |
| 19:37.12 | Qwell | rocksfrow: What did it say in the UPGRADE.txt file? |
| 19:38.07 | rocksfrow | Qwell, searching my box for that file I find nothing |
| 19:38.26 | rocksfrow | any chance you can point me in the right direction... i wouldn't be here if I knew the answer |
| 19:38.56 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 19:39.35 | Chainsaw | rocksfrow: http://svnview.digium.com/svn/asterisk/branches/11/UPGRADE-1.6.txt?revision=371121&view=markup |
| 19:39.45 | Chainsaw | rocksfrow: http://svnview.digium.com/svn/asterisk/branches/11/UPGRADE-1.8.txt?revision=372473&view=markup |
| 19:39.55 | Chainsaw | rocksfrow: http://svnview.digium.com/svn/asterisk/branches/11/UPGRADE-10.txt?revision=371121&view=markup |
| 19:40.18 | Chainsaw | rocksfrow: I'm afraid I can't decode Qwell's suggestion into something more specific, but the "upgrade" file that he is referring to will definitely be that one. |
| 19:40.59 | Chainsaw | hopes he hasn't broken an unwritten channel rule by actually giving a vaguely helpful answer |
| 19:41.12 | igcewieling | or you could just look in the SOURCE CODE DIRECTORY you downloaded |
| 19:41.42 | *** join/#asterisk kurupti0n (kuruption@laffs.at.the.lol.lawlz.lulz.liberlawls.com) |
| 19:41.50 | Chainsaw | igcewieling: Many use the magic of a package system to make Asterisk binaries appear. |
| 19:41.58 | *** join/#asterisk infernixx (nix@unaffiliated/infernix) |
| 19:42.17 | gtTuna | i love calling other companies and hearing the standard Asterisk MOH |
| 19:42.34 | jmetro | i love hearing the CCM hold music. EVERYWHERE |
| 19:42.35 | rocksfrow | lol |
| 19:42.56 | rocksfrow | yeah bc i compiled from source, siiike |
| 19:42.57 | rocksfrow | you know that cool thing called a package manager |
| 19:43.08 | rocksfrow | it unfortunately leaves you less aware of the source :) |
| 19:43.24 | rocksfrow | and, we just moved into a new office setting up this new PBX is one of a hunred things i'm doing this week so just a small detail |
| 19:43.27 | rocksfrow | thanks for the links! |
| 19:43.36 | Chainsaw | tips hat to rocksfrow |
| 19:44.36 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:45.59 | *** join/#asterisk kresp0 (~kresp0@0.Red-88-3-207.dynamicIP.rima-tde.net) |
| 19:50.59 | SuperNull | rocksfrow .. asterisk is so riddled with 'features' its best to build from latest bug release er source release. ;) |
| 19:51.30 | rocksfrow | i can't find anything about the recording structure in any of these upgrade files :/ |
| 19:51.40 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.131) |
| 19:51.42 | igcewieling | *censored* *censored* Adtran *censored* bricks router when rebooting with *censored* ntp servers configured and this *censorted* *censored* *censored* firmware </rant> |
| 19:51.54 | rocksfrow | lol |
| 19:51.56 | rocksfrow | nice ^ |
| 19:51.57 | Chainsaw | igcewieling: DD-WRT. |
| 19:52.09 | igcewieling | Chainsaw: those people are called idiots |
| 19:52.15 | SuperNull | 900 series igcewieling ? |
| 19:52.34 | igcewieling | SuperNull: I'll let you know when our tech gets there with a console cable. |
| 19:53.05 | igcewieling | I THINK it was a 924e with 10.mumble firmware |
| 19:53.18 | igcewieling | but it seems to happen on NetVantas too |
| 19:53.26 | SuperNull | hm |
| 19:53.36 | rocksfrow | ANYWAY, that was easy enough |
| 19:53.40 | SuperNull | we deploy probably 5 -6 Adtran TA900 series a year. so. |
| 19:53.43 | SuperNull | i know your problems. |
| 19:53.48 | SuperNull | pats igcewieling |
| 19:53.49 | rocksfrow | so now it stores in ../monitor/{year}/{month}/{day}/. |
| 19:53.51 | rocksfrow | easy enough :) |
| 19:54.08 | rocksfrow | goes to update his scripts |
| 19:54.36 | SuperNull | seriously tho they are good mass ATAs. i love the quality monitoring. |
| 19:54.47 | SuperNull | customer all 'stuff sucked' we reply 'it doesnt say it sucked?' lol |
| 19:55.43 | igcewieling | SuperNull: we deploy 2 - 3 per week |
| 19:56.38 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 19:56.52 | igcewieling | This is one of these days were, at 5:pm, you say to yourself "I wish I had some good drugs." |
| 20:02.44 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 20:18.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:19.48 | *** join/#asterisk Addisk (~WhyYouLoo@pool-173-51-117-162.lsanca.fios.verizon.net) |
| 20:21.41 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:21.42 | *** join/#asterisk kresp0 (~kresp0@0.Red-88-3-207.dynamicIP.rima-tde.net) |
| 20:25.13 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:27.31 | SuperNull | igcewieling .. define good. Marijuana is nice and its mostly legal (in ny). |
| 20:28.06 | igcewieling | I've done most of them but prefer psychedelics |
| 20:28.18 | igcewieling | Ah, the glory days of my lost youth |
| 20:28.53 | SuperNull | mushrooms are good if you have long term mild headaches too (or intense ones even) |
| 20:29.08 | SuperNull | i had a week long like.. very mild head ache.. took a small amount of mushrooms.. it helped. |
| 20:29.19 | SuperNull | didnt get trippy.. cause .. yeah. last time i did it wooza |
| 20:30.28 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 20:40.29 | drmessano | Ok, so I have been reading over all this Google XMPP business |
| 20:40.37 | drmessano | and I think I see between the lines now |
| 20:41.27 | drmessano | Google voice is going to be nothing more than a phone number tied into Hangouts.. much like Skype's SkypeIn numbers |
| 20:41.38 | drmessano | So bam, dead |
| 20:42.03 | igcewieling | time to move to skype! |
| 20:42.08 | igcewieling | .er..um... 8-0 |
| 20:42.12 | drmessano | There's no point in keeping it alive.. and even further.. if the hangouts app had a keypad, it would handle the entire task |
| 20:42.39 | drmessano | So I guess I can see the death of it coming |
| 20:43.31 | igcewieling | drmessano: do they mention anything about federated SIP? |
| 20:43.33 | jmetro | then again, more obtuse propietary software and hardware has existed before, like the dock connector, itunes, etc |
| 20:43.54 | igcewieling | that was the ONLY thing I was even remotely interested in |
| 20:44.14 | drmessano | Nah, I don't see SIP EVER happening |
| 20:45.18 | igcewieling | one of these days I'll enable federation for SIP on our servers. |
| 20:45.39 | drmessano | What I found interesting was that Gmail users that upgraded from Talk to Hangouts could no longer make calls from GMail. How do you fix that? You make phone calls a feature of hangouts |
| 20:45.56 | drmessano | No need for a completely different "app" in Google voice, and no need at all for XMPP |
| 20:46.29 | drmessano | So, time for chan_hangout lol |
| 20:46.45 | igcewieling | I'm not familiar with any of those. I have one gmail account and that is only for my android phone the only other google service I use is search and maps |
| 20:47.05 | igcewieling | drmessano: somehow I don't think Digium will be writing that. |
| 20:47.15 | drmessano | Yeah, I agree lol |
| 20:47.42 | igcewieling | chan_motif supports other XMPP stuff, doesn't it? |
| 20:47.48 | drmessano | Yes it does |
| 20:48.26 | drmessano | If you chop off the Google XMPP implementations, there's still Jingle |
| 20:49.17 | drmessano | I could see using XMPP/Jingle clients as softphones with some support on the client side and some work in Asterisk |
| 20:49.28 | drmessano | That would be NEATO |
| 20:52.52 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 21:00.39 | drmessano | and there we go --> https://plus.google.com/106636280351174936240/posts/DG6h32BWaQW |
| 21:01.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.227) |
| 21:04.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:09.53 | jmetro | sigh i wish there was just a standard converter |
| 21:10.01 | jmetro | getting something asterisk-friendly can be... annoying |
| 21:12.33 | Qwell | Anything SIP will work. What's the problem? |
| 21:13.59 | *** join/#asterisk kresp0 (~kresp0@0.Red-88-3-207.dynamicIP.rima-tde.net) |
| 21:14.02 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:27.45 | *** join/#asterisk kresp0 (~kresp0@0.Red-88-3-207.dynamicIP.rima-tde.net) |
| 21:31.48 | drmessano | It's not plug-n-play.. |
| 21:34.04 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 21:35.57 | *** join/#asterisk rocksfrow (~kyle@50-197-9-21-static.hfc.comcastbusiness.net) |
| 21:36.26 | rocksfrow | hey, is there a way to configure MixMonitor to keep all three (in, out, and mixed) recording files after mixing? |
| 21:40.08 | rocksfrow | looks like mixmonitor automatically records to one file? is 'monitor' still safe to use? |
| 21:44.40 | igcewieling | We have a customer who plugged in one of their polycoms like this: office ethernet switch <-> Phone LAN Port <-> Phone PC Port <-> their most important server |
| 21:45.16 | navaismo | 0_o |
| 21:45.18 | jmetro | igcewieling: i had a client who plugged their computer in like this. |
| 21:45.18 | jmetro | Entire network - > switch - > switchport 1 -> switchport 24 |
| 21:48.11 | *** join/#asterisk gatty (ajg@2a01:348:11b:beef:d53:529b:fe93:2d00) |
| 21:50.00 | rocksfrow | lol wow, running a server through the phones internal switch? yuck |
| 21:50.34 | WIMPy | At least it has a switch. |
| 21:52.05 | Chainsaw | Certainly a good theft/fiddle protection. |
| 21:53.14 | *** join/#asterisk DBordello (~DBordello@unaffiliated/dbordello) |
| 21:53.22 | gatty | quick (hopefully) question... I'm running Asterisk 11.3 on an Ubuntu 12.04 HA pair using corosync. Is there a way to specify the TCP *source* address for SIP sessions from Asterisk towards my PBX gateway card and Exchange 2010 UM nodes, or will I just have to add each node's IP plus the HA VIP to the peers? |
| 21:53.51 | DBordello | Anybody ahve any good suggestions for ~$50/each used IP phones that work well with Asterisk? The cisco 7960 seems readily available |
| 21:55.43 | Chainsaw | DBordello: The 7960 isn't great at SIP. |
| 21:56.07 | Chainsaw | DBordello: It's basically an SCCP handset with a SIP retrofit. |
| 21:56.10 | Qwell | gatty: That address is all that should be visible on that network. You should have a separate network for the corosync communication. |
| 21:56.22 | DBordello | Chainsaw, so I should stay away from it? |
| 21:57.01 | Chainsaw | DBordello: As long as you understand you're getting a cheap & nasty phone that'll only do SIP over UDP... it may be a bargain. |
| 21:57.18 | DBordello | Chainsaw, fair enough. Other suggestions for a good "value"? |
| 21:57.26 | Chainsaw | DBordello: The internet is full of examples of how to work around the awkwardness. |
| 21:57.34 | Chainsaw | DBordello: (Some work better than others, but still) |
| 21:57.39 | DBordello | I'd prefer something that works right. |
| 21:57.43 | Chainsaw | DBordello: Polycom. |
| 21:58.03 | Chainsaw | DBordello: See if you can score some old second-hand 300 series. |
| 21:58.15 | gatty | Qwell: hmm... ok, so separate VLAN and set kernel route entries towards the SIP peers? Otherwise how do the corosync nodes get patches etc? |
| 21:58.29 | DBordello | The IP 331 seems reasonably priced |
| 21:58.49 | Chainsaw | DBordello: You'll have documentation that actually matches what the product does. |
| 21:59.11 | DBordello | Chainsaw, that sounds better :) |
| 21:59.22 | Chainsaw | DBordello: And it'll implement the SIP protocol better than Asterisk. |
| 22:01.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 22:01.51 | DBordello | I am just planning on setting up a simple setup at home, ~6 Phones, 2 x Google Voice. The most important thing to me is the speakerphone quality. Do I really care about the screen, # of lines, etc? |
| 22:01.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.159) |
| 22:02.37 | Chainsaw | DBordello: Speakerphone on a Polycom is implemented well, yes. Early IP series (300/500 etc) will have a rather minimal screen, but it sounds like you can cope with that. |
| 22:02.41 | gatty | DBordello: polycom speakerphones are generally really good quality, after all they do produce near top line enterprise conferencing phones and video conference systems. :) |
| 22:03.04 | [TK]D-Fender | 300/301 have no speakerphone |
| 22:03.21 | DBordello | Ah, great, I'll browse the Polycoms on ebay then |
| 22:03.23 | [TK]D-Fender | I'd avoid 3XX period for that. 5XX+ has a better fram for that |
| 22:03.31 | [TK]D-Fender | frame* |
| 22:03.49 | [TK]D-Fender | Also avoid 300/500 specifically. they are the oldest of the series |
| 22:03.58 | igcewieling | the VVX series are supposedly pretty cheap and they have a real browser |
| 22:04.47 | Chainsaw | [TK]D-Fender: If we're bargain-basement shopping though, I can't rule out 500. |
| 22:04.53 | [TK]D-Fender | You can. |
| 22:05.03 | Chainsaw | [TK]D-Fender: We're up against $50 7960s. |
| 22:05.05 | [TK]D-Fender | Because it's worht the extra 1$ the higher model will cost |
| 22:05.23 | DBordello | I am currently looking for a summary of polycom models |
| 22:05.36 | [TK]D-Fender | Do NOT get that lower model, there is a BIG differnce in the FIRMWARE available for them. |
| 22:06.13 | Chainsaw | Yes, if the budget was unlimited I'd just get DBordello IP670 handsets. |
| 22:06.21 | Chainsaw | Or IP7000, if speakerphone is more important. |
| 22:06.28 | Chainsaw | But I don't get the sense that it is. |
| 22:06.40 | DBordello | Does "HD Voice" Matter if using Google Voice for the backhaul? |
| 22:06.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ngmypakqwkqskgeo) |
| 22:06.59 | drmessano | Nope |
| 22:07.02 | Chainsaw | (I don't know what Google Voice supports, that's a US-specific product) |
| 22:07.14 | [TK]D-Fender | DBordello: To where? |
| 22:07.21 | drmessano | They don't support G.722 |
| 22:07.27 | DBordello | [TK]D-Fender, general US calls |
| 22:07.47 | DBordello | Is using Google Voice a bad idea in general? Would it be better to get a real SIP provider and forward? |
| 22:07.48 | [TK]D-Fender | DBordello: Correct answer -> PSTN |
| 22:07.53 | DBordello | [TK]D-Fender, god it. |
| 22:07.54 | DBordello | got* |
| 22:07.56 | [TK]D-Fender | DBordello: And no, that will not matter |
| 22:08.20 | drmessano | Google Voice may not work much longer with Asterisk |
| 22:08.23 | drmessano | So it |
| 22:08.27 | drmessano | it's not a safe bet |
| 22:08.49 | Chainsaw | drmessano: You mean chan_hangout isn't done yet? How disappointing! |
| 22:09.05 | drmessano | I know, right! |
| 22:09.22 | DBordello | drmessano, ah, okay. |
| 22:09.54 | drmessano | Which kills half of my use for Google Voice |
| 22:10.00 | drmessano | GAH |
| 22:10.54 | DBordello | I like google voice, it integrates well with my android device, and using it with Asterisk seemed like a good way to setup a nice phone network at home |
| 22:11.11 | DBordello | But it can forward to any PSTN number, so a good SIP provider makes sense too |
| 22:11.50 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 22:11.52 | drmessano | I had it set up to ring my office phone, my cell phone, and ring into my home PBX via chan_motif |
| 22:12.06 | drmessano | Worked really well... A misplaced phone wouldn |
| 22:12.13 | drmessano | Worked really well... A misplaced phone wouldn't prevent me from getting calls |
| 22:12.35 | drmessano | and I could answer calls on my desk phone and not have to worry about them running up my cell bill |
| 22:13.05 | DBordello | drmessano, why do you think Google Voice will stop working with Asterisk? |
| 22:13.23 | Chainsaw | DBordello: Because they're abandoning open standards. |
| 22:13.31 | DBordello | Chainsaw, that is unfortunate |
| 22:13.51 | drmessano | Because Google is killing XMPP support for Hangouts.. and it looks like Google Voice is going to move to being a "feature" of hangouts |
| 22:14.00 | drmessano | So they're breaking the vital XMPP link |
| 22:14.35 | DBordello | aaha, lame |
| 22:14.49 | przerull | hello. So i've been playing around with the hangupcause function in asterisk 11, but how can I get the hangup cause for the outbound leg of a dial to an invalid number? |
| 22:15.11 | DBordello | Do you guys recommend any VOIP providers then for use with Asterisk? |
| 22:15.19 | drmessano | Flowroute |
| 22:15.29 | WIMPy | przerull: The same way as for any call. Check that variable. |
| 22:15.49 | drmessano | Flowroute first and foremost, then everyone else lumped together |
| 22:16.13 | przerull | yeah but I need to access it on the inbound leg of the call |
| 22:16.33 | DBordello | drmessano, excellent, i'll check them out |
| 22:16.46 | WIMPy | przerull: That doesn;t make sense. |
| 22:17.01 | DBordello | drmessano, they look cheap, $0.01/minute |
| 22:17.23 | drmessano | They are very inexpensive |
| 22:17.27 | drmessano | and reliable as hell |
| 22:17.51 | drmessano | Total rockstar of a provider, and I don't mean just in the context of being an ITSP |
| 22:18.13 | drmessano | Every provider should be as inexpensive, friendly, and reliable. |
| 22:18.52 | przerull | WIMPy: so let's say that priority 1 of the s extension in my context is Dial(SIP/blahblah/+15555555555); since that is an invalid number i should recieve a sip 404 message. What I am trying to do is essentially get the sip status code in priority 2 of the s extension |
| 22:19.19 | DBordello | drmessano, It looks great. $1.25/DID, ~$0.01/minute. Thanks for the recommendation. That should be a good tie-in with Google Voice |
| 22:19.37 | WIMPy | przerull: You Only need to look at ${HANGUPCAUSE}. |
| 22:20.24 | przerull | ah thanks. I'll give that a shot. I was trying to use the hangupcause function but BRIDGEPEER wasn't set |
| 22:21.25 | WIMPy | przerull: Variable, not function. The function will give you channel specific results. |
| 22:23.18 | przerull | WIMPy: thanks, is there a function to convert the HANGUPCAUSE number into one of the Q.931 cause texts? |
| 22:23.49 | WIMPy | przerull: No. And it's Q.850. |
| 22:24.44 | przerull | WIMPy: hmmm the wiki says Q.931 https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
| 22:26.14 | WIMPy | Cause codes are in Q.850. Q.931 is a LOT more. |
| 22:27.38 | igcewieling | przerull: for the most part asterisk's internal hangupcauses mirror Q.850 hangup causes |
| 22:27.38 | przerull | ah thanks |
| 22:28.15 | igcewieling | you might be better served with the HANGUPCAUSE function if you want protocol specific cause information |
| 22:28.51 | przerull | so I notice that when I dial the 5555555555 u get a HANGUPCAUSE of 1 and when my dial answers I get a HANGUPCAUSE value of 16 |
| 22:29.13 | przerull | is there a table somewhere that maps those to something more meaningful? |
| 22:29.34 | WIMPy | There are tons of those tables on the web. |
| 22:29.48 | przerull | igcewieling: thanks I was trying that at first but was unable to get the information for the outbound leg of the call) |
| 22:30.00 | WIMPy | Hmm. Would make sense to add the numbers to the wiki page. |
| 22:30.04 | przerull | it's just a Q.850 table? |
| 22:30.21 | WIMPy | You can read Q.850 or just google for cause codes. |
| 22:30.55 | przerull | great thanks |
| 22:30.56 | WIMPy | And yes, 1 is unallocated number. |
| 22:30.58 | igcewieling | (6:27:38 PM) igcewieling: przerull: for the most part asterisk's internal hangupcauses mirror Q.850 hangup causes |
| 22:32.52 | przerull | sweet thanks a lot |
| 22:41.08 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 22:41.35 | j-fish | Is there any way to tell if a customer is calling to me from skype? |
| 22:42.41 | navaismo | modifiyng the CID |
| 22:50.56 | przerull | do you guys know of any trunking providers that offer something like a $10/month unlimited plan? |
| 22:55.13 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 22:55.16 | saint_ | hi all |
| 22:55.53 | saint_ | what is the best way if I have 2 different SIP providers, with 2 incoming numbers , for both numbers to go through the same process I wrote (blacklist, vip, vm, etc..) |
| 22:56.12 | WIMPy | GoSub() |
| 22:56.22 | saint_ | GoSub or Goto ? |
| 22:56.43 | saint_ | I'm all excited. I got ride of my vonage to go with VoIP.ms |
| 22:56.48 | saint_ | I'm saving about $80 / month ! |
| 22:56.55 | WIMPy | You can use Goto if you save the destination somewhere. |
| 22:59.35 | saint_ | ok, thank you |
| 23:03.50 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 23:06.41 | j-fish | navaismo , Modifying the CID to what? |
| 23:08.40 | Katty | ASTERSIK |
| 23:08.56 | *** join/#asterisk Khronos (~Khronos@c-76-110-126-209.hsd1.fl.comcast.net) |
| 23:09.03 | WIMPy | Katty: Is that something edible? |
| 23:09.14 | Katty | i wouldn't recommend it |
| 23:09.34 | navaismo | j-fish: the name --> skype+originalcid |
| 23:09.40 | WIMPy | So it's unhealthy? |
| 23:10.17 | *** join/#asterisk JackStorm (no@ip72-204-153-71.no.no.cox.net) |
| 23:11.48 | igcewieling | WIMPy: Asterisk can get quite bitter at times |
| 23:12.16 | WIMPy | Int's that the default? |
| 23:18.32 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 23:24.42 | j-fish | navaismo: can i pm? |
| 23:29.26 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 23:30.59 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-fnfzhlmnfkxnclrp) |
| 23:34.38 | *** part/#asterisk DBordello (~DBordello@unaffiliated/dbordello) |
| 23:34.41 | *** join/#asterisk DBordello (~DBordello@unaffiliated/dbordello) |
| 23:34.52 | DBordello | I am looking at buying ~5 Polycom 501s with AC adapters. Thoughts? |
| 23:38.49 | igcewieling | DBordello: "pre-owned"? |
| 23:41.30 | *** join/#asterisk navaismo (~navaismo@189.191.248.18) |
| 23:51.22 | DBordello | igcewieling, yes |
| 23:56.09 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |