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| 00:22.26 | [TK]D-Fender | Changing software won't change that |
| 00:22.39 | [TK]D-Fender | So fix whoat's wrong |
| 00:22.58 | [TK]D-Fender | start by proving the contents of all your firewalls |
| 00:25.10 | [TK]D-Fender | I'd also be looking at how you're addressing it, forwards, etc |
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| 01:33.22 | cmendes0101 | With pjsip is there a way to gave a catchall type of context for inbound calls? |
| 01:33.29 | cmendes0101 | have* |
| 01:35.52 | cmendes0101 | ahh. Ok need to create an endpoint named anonymous w/ res_pjsip_endpoint_identifier_anonymous.so |
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| 05:32.03 | poing | Anyone know how to read() and record() at the SAME TIME. Both stop with # I want to⦠IF empty(read()) THEN speech_to_text(record()) |
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| 06:19.16 | voipmonk | I don't understand the question |
| 06:19.58 | voipmonk | stick the channel in a conference and do what you want? |
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| 06:46.49 | RudyValencia | Does anyone know of a dialplan I could use to mimic Skype's call test service? |
| 06:48.55 | wdoekes | RudyValencia: answer, playback, record, playback, playback($recording) |
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| 08:17.55 | stefan27 | If I want to force a sync of real time members of a queue called queue-01, I can do this with AMI Command: queue show queue-01, but is there a less expensive way of doing it? |
| 08:19.01 | stefan27 | queue reload members queue-01 ? |
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| 08:23.04 | Resnik | is there any way to force incomming call what trunk to use? I have multiple trunks configured, and in inbound routes I also configured proper DID's, yet all the calls are comming from the first trunk |
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| 09:31.43 | shuttleduck | Is it enough to uncomment chan_pjsip and add chan_sip in modules.conf to change to chan_sip? |
| 09:31.59 | shuttleduck | or is there a little more to it? |
| 09:49.39 | jkroon | significantly more. |
| 09:49.49 | jkroon | there is some wizzard to help (which I haven't tried yet) |
| 09:52.05 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.1 (2016/09/09), 11.23.1 (2016/09/09), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 09:52.36 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.1 (2016/09/08), 11.23.1 (2016/09/08), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 09:58.51 | shuttleduck | Well can i compile Asterisk 13 with chan_sip instead of pjsip then? |
| 09:59.44 | shuttleduck | i'm unable to unselect chan_pjsip in menuconfig it seems.. |
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| 10:10.19 | jkroon | does anybody know of any tools that can take a pcap file and look at the timing etc of rtp streams for me? |
| 10:10.37 | jkroon | yes you can. i've used it like that before for a test bench. |
| 10:10.49 | jkroon | but we're not quite ready to make the jump from 11 to 13. |
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| 10:31.01 | KValchev | jkroon: wireshark |
| 10:31.23 | jkroon | will it perform jitter analysis and check for missing packets in the rtp streams? |
| 10:31.37 | jkroon | or do I still need to do that manually? |
| 10:32.20 | jkroon | nm, quick google found some potentially useful data thanks. |
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| 11:09.55 | stefan27 | wireshark has good tools for that; but it's trickier if the stream is encrypted and wireshark cant identify it as rtp |
| 11:11.30 | Samot | You need to capture the ports for RTP. |
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| 11:40.55 | bounceman | Hi guys, perhaps this is not an asterisk question but I will ask anyway hoping any of you know the PSTN. In a Call from A to B, which part generates the ringing tone in A's phone? |
| 11:41.07 | bounceman | We use Asterisk as a media gw B2BUA setup btw |
| 11:42.07 | bounceman | I would believe it is the operator at the far end at B who generates it |
| 11:43.40 | bounceman | Yeah it looks like the RTP stream containing the ringing tones are indeed from our upstream operator. Anyone want to confirm my thoughts? |
| 11:44.41 | WIMPy | It can be either way. |
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| 12:17.46 | jkroon | Samot, stefan27 - (fortunately) we've not been able to flip the switch on srtp+sip/tls just yes. |
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| 12:23.46 | polysics | hey! question for y'all: is CCXML still relevant? Would someone use a CCXML browser? |
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| 14:05.48 | crisderock | hi |
| 14:08.12 | crisderock | a little question: i use Set(CHANNEL(language)=foo) before Dial(......A(welcome-file)) but $welcome-file will be searched in 'sounds/een' not in 'sounds/foo'. whats wrong? |
| 14:08.29 | crisderock | s/een/en/ |
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| 14:14.16 | [TK]D-Fender | Show us the full call attempt and the folder dump |
| 14:14.19 | [TK]D-Fender | ~pb |
| 14:14.25 | infobot | hmm... pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:17.44 | crisderock | in know pastebin..... |
| 14:21.39 | crisderock | http://pastebin.com/Abz0wFNa |
| 14:26.26 | crisderock | i also created a macro which is called from Dial: http://pastebin.com/SP2dUFS8 |
| 14:27.12 | crisderock | (the macro contains only NoOp(${CHANNEL(language)}); ) |
| 14:28.18 | [TK]D-Fender | <PROTECTED> |
| 14:28.24 | [TK]D-Fender | and "core show settings" |
| 14:28.28 | [TK]D-Fender | PB them |
| 14:28.55 | crisderock | ls is bottom at the first page |
| 14:29.57 | crisderock | s/page/paste/ |
| 14:30.08 | crisderock | settings: http://pastebin.com/58mPaibh |
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| 14:30.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:33.06 | [TK]D-Fender | <PROTECTED> |
| 14:33.18 | [TK]D-Fender | * does not look for sounds in share |
| 14:33.24 | [TK]D-Fender | it's loking in lib |
| 14:35.06 | crisderock | nope. when copyed to /usr/share/sounds/en the file will be found |
| 14:36.06 | crisderock | i mean /usr/share/asterisk/sounds/en |
| 14:37.04 | [TK]D-Fender | move them where * is telling you it is looking |
| 14:37.24 | [TK]D-Fender | and I asked for that other command for a reason |
| 14:38.11 | KValchev | crisderock: Are you sure that file welcome.wav in u-law codec ? |
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| 14:38.46 | crisderock | yepp |
| 14:39.43 | KValchev | show file info : sox welcome.wav -n stat |
| 14:40.09 | [TK]D-Fender | prove the PERMISSIONS on the files and moce them where I told you * is looking |
| 14:40.19 | crisderock | [TK]D-Fender: you mean the ls -la? as i writek, the command is bottom at the first paste. or what for a command? |
| 14:40.24 | WIMPy | wav in µ-law??? |
| 14:40.36 | [TK]D-Fender | # ls /usr/share/asterisk/sounds/foo/ |
| 14:40.37 | [TK]D-Fender | welcome.gsm welcome.ogg welcome.wav |
| 14:40.50 | [TK]D-Fender | <[TK]D-Fender> ls -la /usr/share/asterisk/sounds/foo/ <---------------- |
| 14:41.52 | crisderock | im not a n00b. all directorys are executable and all files are readable by asterisk |
| 14:42.14 | *** join/#asterisk rwb (~Thunderbi@65-183-151-87-dhcp.burlingtontelecom.net) |
| 14:43.00 | rwb | Does anyone know of any analog FXS/FXO cards that are USB? |
| 14:43.07 | [TK]D-Fender | <PROTECTED> |
| 14:43.13 | [TK]D-Fender | Follow what it is telling you |
| 14:43.38 | [TK]D-Fender | rwb, Sangoma makes some, xorcom makes larger units |
| 14:44.57 | tzafrir | (technically: sounds are under astdatadir, which doesn't have to be astvarlibdir) |
| 14:45.46 | [TK]D-Fender | news to me.... |
| 14:46.03 | [TK]D-Fender | Only time I've seen share even referenced is on odd package installs |
| 14:46.28 | crisderock | no. /var/lib/asterisk/sounds/foo/welcome will not be found. as i say |
| 14:46.54 | crisderock | look at the second paste. it shows to you, dial resets the language to en |
| 14:47.08 | crisderock | http://pastebin.com/SP2dUFS8 |
| 14:49.53 | [TK]D-Fender | <PROTECTED> |
| 14:50.00 | [TK]D-Fender | Version: 11.13.1~dfsg-2 |
| 14:50.15 | [TK]D-Fender | I'd recommend updating and attempting to replicate the issuegrading to |
| 14:50.40 | crisderock | this is the current version on debian stable |
| 14:51.33 | crisderock | you cannot expect to have every time a bleeding edge version in your repository |
| 14:52.13 | crisderock | and the version you talking bout is from yesterday. so no time for packagers and maintainers |
| 14:52.17 | [TK]D-Fender | that was almost half of the 11 branch history ago |
| 14:52.28 | [TK]D-Fender | 23 vs 13 |
| 14:53.08 | file | Date: Thu Mar 28 14:22:41 2013 +0000 |
| 14:54.12 | file | ha, that's 3.0 |
| 14:54.17 | file | Date: Wed Sep 24 19:00:38 2014 +0000 |
| 14:54.18 | file | there we go |
| 14:54.39 | file | or for the point release, Date: Mon Oct 20 15:40:56 2014 +0000 |
| 14:54.51 | igcewieling | I think you'll find few people want to help with older Asterisk releases, especially one part of a distro. |
| 14:55.34 | crisderock | this is a company-site. no. selfcompiled. versions. ever. you know the problem? |
| 14:55.48 | crisderock | is asterisk not for professional use? |
| 14:56.10 | WIMPy | igcewieling: Which is a good indication for something being unstable. |
| 14:57.23 | igcewieling | I feel it is an indication lots of bugs get fixed. |
| 14:58.45 | igcewieling | nope. Asterisk is the only software I compile from scratch. Phones are far, far too important to be taken down by a bug fixed in a later release. |
| 15:03.44 | crisderock | [TK]D-Fender: just for your interest. i made a diff from my current version to the 11.23.1 and there is nothing to see bout my problem..... |
| 15:04.36 | igcewieling | crisderock: did you fix the sounds directory problem? |
| 15:04.47 | crisderock | nope |
| 15:04.49 | igcewieling | (10:43:07 AM) [TK]D-Fender: VarLib directory: /var/lib/asterisk <- this is where "sounds" is supposed to be |
| 15:04.53 | igcewieling | then you'll never fix your problem. |
| 15:05.20 | crisderock | have tried this but it fixes nothing |
| 15:05.49 | crisderock | as you can see in my paste, after Dial, dthe language is reset to 'en' |
| 15:06.08 | igcewieling | Try __CHANNEL(language) |
| 15:06.13 | crisderock | and _this_ is the problem. not somne directory |
| 15:06.20 | [TK]D-Fender | you should be doing proper sanity checks like taking an existing stock recording, renaming it, and putting it in the folder to prove your file isn't actually inappropriatte, etc |
| 15:06.33 | crisderock | igcewieling: look at http://pastebin.com/SP2dUFS8 |
| 15:06.35 | igcewieling | crisderock: you've put the relevant sound file in the en/ directory and confirmed that it works. |
| 15:06.52 | [TK]D-Fender | prove the file is ok by doing a direct playback of it |
| 15:07.02 | crisderock | i_use_ CHANNEL(language) |
| 15:07.55 | crisderock | but you can use is often as you want as long as the language will be reset during dial |
| 15:08.16 | igcewieling | I assumed, so that's why I suggested the two-underscores prefix to Channel() |
| 15:08.38 | igcewieling | Set(__CHANNEL(language)=foo) |
| 15:08.42 | igcewieling | clear enough? |
| 15:09.43 | crisderock | ast_func_write: Function __CHANNEL not registered |
| 15:10.41 | [TK]D-Fender | <[TK]D-Fender> prove the file is ok by doing a direct playback of it <--- |
| 15:10.47 | igcewieling | then your asterisk is SERIOUSLY screwed. |
| 15:10.48 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance |
| 15:10.52 | igcewieling | see above. |
| 15:10.57 | [TK]D-Fender | and another proper sanity check of the channel languge right after doing so |
| 15:11.09 | [TK]D-Fender | igcewieling, Functions don't work like that |
| 15:11.40 | igcewieling | Good point. I guess he'll have to send in a bug report. 8-| |
| 15:11.59 | KValchev | crisderock: Dial command create new channle and alway set language variable on new channel to global . to change this use dial option b to change language on new remote channel |
| 15:13.29 | crisderock | KValchev: you say, just create a context that sets the language to the right value? |
| 15:14.21 | crisderock | igcewieling: dont send bugreports. i have fightet for a patch for weeks...... |
| 15:15.09 | igcewieling | crisderock: I was being somewhat sarcastic. since you don't upgrade your Asterisk you'd never be able to take advantage of the fix. |
| 15:16.09 | crisderock | igcewieling: if i patch my asterisk versions, the versions of the distro are always newer and with every upgrade my patches are gone |
| 15:16.19 | KValchev | crisderock: yes |
| 15:16.35 | igcewieling | BTW, when I say "upgrade" I am referring to upgrading within a branch. |
| 15:16.45 | crisderock | this is why company sites avoid self packages or self compiled versions |
| 15:17.12 | Penguin | Uniformity? |
| 15:17.56 | igcewieling | crisderock: what makes you think your distro will upgrade asterisk any time soon? Asterisk has had 10 bug fix releases in the past year and your distro still has an old version. |
| 15:18.56 | igcewieling | KValchev' |
| 15:19.07 | igcewieling | KValchev's suggestion seems to be the "right" one. |
| 15:19.39 | crisderock | igcewieling: maybe a communication problem between digium and the debian maintainers? I was able to enjoy myself even to the great kindness of digium developers...... |
| 15:19.40 | igcewieling | my suggestion for using a double underscore prefix was trying to accomplish the same thing |
| 15:30.52 | file | crisderock, we don't have any input or feedback into distro packaging except for maybe Fedora - it's their choices |
| 15:31.27 | file | I hope that one day we (the project) will do packages but I can't guarantee that, and it would merely be a package of the current version probably - same as if you had built it yourself and made a package |
| 15:33.47 | crisderock | file: just contact (for debian) pkg-voip-maintainers@lists.alioth.debian.org to say that there is a new version (pls with link and changes if possible) |
| 15:34.09 | file | that in and of itself does not mean that they will make a new version |
| 15:34.15 | file | they have their own policies as well |
| 15:37.46 | KValchev | hum is very cool bugs, asterisk alway play files with same name from en directory . Cool |
| 15:45.56 | KValchev | crisderock: rename file in foo directory to other name. If asterisk have two file wih same names alway play file from default global sounds directory . I tested it. Very cool bugs |
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| 16:19.01 | *** join/#asterisk eduzimrs (~eduzimrs@mail.aytycrm.com.br) |
| 16:20.53 | eduzimrs | Hi Guys, im running Aster 1.4.6, and having a issue with importing queues, i have a routine that import queues from a Mysql, sometimes one queue is not imported to Ast and all logged in user are kicked off, it happens randonly |
| 16:21.24 | eduzimrs | sorry Asterisk Version 1.6.2.13 |
| 16:23.00 | eduzimrs | just to make myself clear, all the queues imported from Mysql are saved into a file that is readen by Asterisk |
| 16:24.09 | eduzimrs | i have evaluated all the queues syntaxes, it all seems nice |
| 16:26.49 | eduzimrs | my Ast has 390 queues, some of then over 100 agents logged in |
| 16:27.30 | [TK]D-Fender | eduzimrs, 1.6.x, 1.8, 10, 12 are no longer supported |
| 16:29.06 | eduzimrs | [TK]D-Fender, only 13 ? |
| 16:29.25 | [TK]D-Fender | and 11 |
| 16:29.29 | [TK]D-Fender | 1.6 is ANCIENT |
| 16:29.55 | eduzimrs | yah i know, but i've got an enterly sistem over it |
| 16:30.15 | [TK]D-Fender | Should have gotten off of it 5 years ago |
| 16:30.22 | eduzimrs | nightmare upgrade it |
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| 16:31.03 | [TK]D-Fender | It has not been supported in a long time. |
| 16:31.32 | eduzimrs | yah, im planning to update to 13 |
| 16:31.45 | eduzimrs | take too long |
| 16:32.01 | eduzimrs | thanks anyway |
| 16:36.10 | igcewieling | eduzimrs: upgrading to Asterisk 11 will be reasonably easy |
| 16:39.18 | file | 11 goes security only end of October |
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| 16:41.37 | eduzimrs | humm |
| 16:41.44 | eduzimrs | better 13 so |
| 16:43.02 | eduzimrs | too much apps syntaxes has been changed and deprecated |
| 16:44.28 | igcewieling | eduzimrs: Asterisk 13 has significant changes from Asterisk 11, so much of the docs available won't apply. That's why I stick to Asterisk 11 -- we don't have to redesign all of our internal scripts and dialplan when upgrading from 1.6 |
| 16:45.15 | igcewieling | I plan on waiting for Asterisk 15 before upgrading from Asterisk 11. |
| 16:46.07 | eduzimrs | igcewieling, will 15 be a LTS ? |
| 16:46.24 | igcewieling | eduzimrs: Last time I checked it will. |
| 16:46.41 | eduzimrs | ok, tks brother |
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| 18:10.03 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/08), 11.23.1 (2016/09/08), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:10.27 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:11.23 | gtjoseph | ok, it still seems like the 8th. :) |
| 18:13.13 | SeiGGy | how would I go about configuring asterisk "make menuselect" to instead not use the menuselct tool and use a config file or something instead so I can build using an automated build process without someone interacting with it? |
| 18:14.09 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 18:16.32 | voipmonk | SeiGGy - start here https://wiki.asterisk.org/wiki/display/AST/Build+System+Architecture |
| 18:16.42 | SeiGGy | thanks! |
| 18:41.16 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 18:41.52 | jrun | is endpoint's permit= roughly equivalent to host= in chan_sip settings? |
| 18:44.17 | igcewieling | jrun: chan_sip or chan_pjsip? |
| 18:45.11 | igcewieling | for chan_sip no, it isn't the same as the host= setting But there are situations where permit does nothing at all. |
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| 18:58.35 | jrun | i have host=<ip> setup in chan_sip.conf; moving to res_pjsip.conf should i just do permit=<ip> under endpoint or have an aor with contact=sip:<ip> ? |
| 19:00.18 | file | host=<ip> is equivalent to an aor with contact=sip:<ip> in PJSIP |
| 19:00.28 | file | provided the endpoint references the aor |
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| 19:00.39 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 19:04.47 | jrun | file: what's permit then? |
| 19:05.00 | file | it's an ACL. |
| 19:05.21 | file | it permits an IP address, or range, from using the endpoint or denies a range, etc |
| 19:05.49 | jrun | sounds like host=<ip> to me |
| 19:05.53 | file | nope |
| 19:06.19 | file | host=<ip> controls where SIP traffic is sent if you contact that peer, and it can also be used for matching |
| 19:06.22 | jrun | does the aor you mentioned have to be labeled like [<ip>] |
| 19:06.32 | file | an AOR is arbitrary |
| 19:07.03 | file | generally if a device isn't registering to it it can be named anything |
| 19:07.12 | file | if registering for maximum compatibility it is best to use the same name as the endpoint |
| 19:07.31 | jrun | these are two pbxes trying to talk to each other |
| 19:08.15 | jrun | btw, all my other pbxes are chan_sip, the *dev* pbx is using chan_pjsip |
| 19:08.21 | jrun | would that cause any problem? |
| 19:08.25 | file | no, SIP is SIP |
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| 19:13.01 | jrun | hmm, identify_by=ip would be nice too :) |
| 19:13.10 | moose_ | Howdy all! - Does anyone have a good recommendation for Wireless SIP Phones? |
| 19:13.45 | jrun | file: what's the equivalent to insecure=port,invite? |
| 19:13.52 | file | a type=identify section |
| 19:14.18 | file | if you want to simplify you can probably use wizards, http://blogs.asterisk.org/2016/05/04/pjsip-configuration-wizard/ |
| 19:14.22 | igcewieling | moose_: We used to use Polycom ones, but I think they discontinued the wireless products. Might want to check with Polycom anyway, in case they have a different product line for wireless. |
| 19:15.35 | robmal | Lies! |
| 19:15.37 | robmal | D60 |
| 19:16.32 | robmal | It's a DECT addon for VVX but it works. |
| 19:16.35 | igcewieling | moose_: yup, they did http://www.polycom.com/voice-conferencing-solutions/desktop-ip-phones/vvx-d60-wireless-handset.html |
| 19:16.45 | igcewieling | robmal: heh. |
| 19:16.57 | igcewieling | robmal: I was thinking of their Kirk line of coreless phones. |
| 19:17.17 | robmal | That was a dead end, yes. |
| 19:17.45 | SeiGGy | has anyone tried to get TLS working in Asterisk in a docker container? Not sure how I'm supposed to deal with the fact the hostname and ip address aren't static...so not sure what I should use for the ast_tls_cert script |
| 19:18.12 | moose_ | cool - I have a hotel that uses a mix of Polycom IP650s and Panasonic TGP500 but the maintenance travels throughout floors need something the can leave on their carts or sides so a base station wouldnt work |
| 19:18.18 | moose_ | I will check the D60s |
| 19:18.31 | jrun | file: where is type=identify documented? |
| 19:19.26 | jrun | file: never mind. it's pjsip.conf. thanks |
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| 19:20.13 | moose_ | although the D60 is dect back to the base station |
| 19:22.02 | robmal | You could use softphones for mobiles if that's any help. |
| 19:23.47 | moose_ | yeah - I have tried them - Haven't really found a reliable app yet. |
| 19:23.55 | moose_ | Well - I should say I have tried some |
| 19:24.46 | robmal | Android has voip integrated since hmm v2? |
| 19:24.54 | robmal | And it works nice. |
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| 19:34.12 | jrun | file: can endpoint under type=identify be a comma-seperated list of endpoints? |
| 19:34.18 | file | no. |
| 19:34.28 | file | you can only match to a single endpoint. |
| 19:35.21 | jrun | no qualify_timeout either i guess. then aor with contact= seems to be a better option for us. |
| 19:35.39 | file | they serve different purposes |
| 19:35.55 | file | type=identify merely matches inbound traffic based on addresses and identifies it as coming from an endpoint |
| 19:36.07 | file | type=aor is for sending traffic to something, and has qualify |
| 19:39.57 | jrun | i see. thanks |
| 19:40.05 | jrun | do templates support nesting? |
| 19:43.05 | igcewieling | robmal: as I understand it android built in voip is only available on devices with cellular access. My nexus 7 wifi-only doesn't have such an app. |
| 19:44.37 | moose_ | I also would have to assume all employees have an android which one be the case. |
| 19:44.50 | moose_ | I am only finding a few models out there that do WiFi SIP |
| 19:46.54 | igcewieling | hmm? I was referring only to the included apps. I tried a few of them before deciding it wasn't any more secure than a cellphone and stopped. |
| 19:47.12 | igcewieling | <-- no cellphone for more than 2 years now. |
| 19:49.14 | moose_ | yeah I am not really worried about secure - I am looking for reliability within a multifloor hotel - they do not want to put up cell towers but have wifi and they want it to work with the current asterisk PBX |
| 19:49.45 | moose_ | they dont want to put up cell or radio repeaters in the building* |
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| 20:34.07 | robmal | igcewieling: Lies! http://www.voipvoip.com/android/sip.html |
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| 20:39.48 | igcewieling | robmal: I don't see nexus 7 tablet on that list, only nexus phone. Also, the wifi only nexus tablet doesn't have any kind of dialer, so it is hard to go into the call settings. |
| 20:42.19 | igcewieling | Granted, that is a small minority of the nexus tablets |
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| 20:52.42 | robmal | I've used csipsimple in the past, also integrates nicely with android. |
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| 21:33.35 | igcewieling | robmal: *nod* IIRC that is the one which is nice and simple and not a swiss army knife of a sip client. (hello counterpath!) |
| 21:36.49 | robmal | And also is very hard to configure in most cases so the myth that voip over wifi doesn't work lives. |
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