| 01:20.10 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 01:20.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 01:34.47 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-gnbnaavuxlwsymmy) |
| 02:14.50 | *** join/#asterisk knightr (~Nicolas@mythtv/developer/knightr) |
| 02:15.14 | knightr | Hi! I noticed a problem with ast_say_date_with_format_fr in say.c but I am not sure if it's simply a little packaging problem of the sound files or if the source code should be modified... |
| 02:15.59 | knightr | So if I try to fix it or at least report it I am not fully how it should be addressed... |
| 02:16.11 | knightr | the source code asks for a sound file which is not in the digits directory but the root language folder and it's not exactly the same one if the names are supposed to match exactly.. |
| 02:16.24 | knightr | so when that code is called, it aborts... |
| 02:16.33 | knightr | just so I make sure I am getting the appropriate sound files, where is the official repository for them? |
| 02:31.46 | *** part/#asterisk t4nk997 (4b953aad@gateway/web/freenode/ip.75.149.58.173) |
| 02:44.23 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:49.35 | *** join/#asterisk brookshire (~mbrooks@user-24-214-210-134.knology.net) |
| 02:52.08 | *** join/#asterisk brookshire (~mbrooks@user-24-214-210-134.knology.net) |
| 03:23.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:35.06 | wyoung | mmm, asterisk doesn't use namespaces? |
| 03:35.23 | wyoung | oh right, C |
| 04:02.00 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 04:05.03 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 04:10.46 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-lpfckcltmvhvgbzc) |
| 04:12.28 | *** join/#asterisk troyt (~troyt@c-24-10-220-108.hsd1.ut.comcast.net) |
| 04:13.45 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 04:15.37 | *** join/#asterisk troyt (~troyt@c-24-10-220-108.hsd1.ut.comcast.net) |
| 04:27.16 | *** join/#asterisk troyt (~troyt@c-24-10-220-108.hsd1.ut.comcast.net) |
| 05:13.01 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 05:25.07 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:38.58 | *** join/#asterisk troyt (~troyt@c-24-10-220-108.hsd1.ut.comcast.net) |
| 05:59.25 | *** join/#asterisk mokmeister (~quassel@86-44-212-179-dynamic.agg2.shn.lmk-pgs.eircom.net) |
| 06:17.23 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:09.10 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:13.00 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 07:52.54 | *** join/#asterisk eschmidbauer__ (~eschmidba@unaffiliated/eschmidbauer) |
| 07:55.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 07:56.52 | *** join/#asterisk markusl (~markus@hodor.lindenberg.io) |
| 08:06.30 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:14.01 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 08:15.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:35.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 08:35.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 08:35.51 | *** join/#asterisk mzdravkov (~me@89.25.103.182) |
| 08:59.34 | WIMPy | knightr: Sound files including a text file with their descriptions are at http://downloads.asterisk.org/pub/telephony/sounds/ |
| 09:00.38 | *** join/#asterisk chendy (~alexc@183.39.136.93) |
| 09:08.57 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 09:10.59 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 09:17.37 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 09:24.40 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:32.48 | mzdravkov | I need to get from ARI whenever a Dialing is made, but the only way I can get events is when I call Stasis(appname) in the rule at extensions. Is there a way to get all dialings without making my users change all their rules? |
| 09:34.15 | *** join/#asterisk xpheres (~bodhi64@dslb-088-073-011-220.088.073.pools.vodafone-ip.de) |
| 09:34.28 | xpheres | hello everybody, |
| 09:34.49 | xpheres | I managed to set up SRTP encryption, however, TLS seems not be working |
| 09:35.06 | xpheres | I wonder if srtp is enough for privacy |
| 09:46.22 | WIMPy | STRP envrypts the audio. TLS would encrypt who you're calling. |
| 09:46.34 | xpheres | yes that's what I've been reading |
| 09:46.50 | xpheres | so |
| 09:46.55 | xpheres | I guess audio is more important |
| 09:47.01 | xpheres | at least till I find how to make tls work |
| 09:47.34 | WIMPy | Depends on who/what you're calling. |
| 09:47.54 | WIMPy | If you call via your ITSP it's both pretty pointless. |
| 09:50.06 | xpheres | what is ITSP? do you mean my ISP? |
| 09:50.20 | WIMPy | ~itsp |
| 09:50.20 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 09:50.34 | WIMPy | Or any way to the PSTN, really. |
| 09:53.17 | *** join/#asterisk davlefouAMD (~davlefouA@41.225.183.4) |
| 09:54.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 09:54.28 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 10:07.35 | *** join/#asterisk Puck` (~puck@unaffiliated/go) |
| 10:07.38 | Puck` | hi everyone |
| 10:07.52 | Puck` | is anyone monitoring asterisk through zabbix? If so, where can I find the template for it? |
| 10:37.57 | *** join/#asterisk akiims (~a@95.130.35.101) |
| 10:41.19 | akiims | hello, incoming call in Queue get disposition Busy if one of the queue members is busy. Shouldn't disposition be NO ANSWER? (asterisk 1.8.10.0) |
| 10:44.51 | ketas | oh... messages work now! |
| 10:44.59 | ketas | night well spent... |
| 10:51.04 | xpheres | hehe |
| 10:51.21 | xpheres | it took me time to make messages works too |
| 11:23.05 | *** join/#asterisk chendy (~alexc@113.116.60.225) |
| 11:26.19 | *** part/#asterisk xpheres (~bodhi64@dslb-088-073-011-220.088.073.pools.vodafone-ip.de) |
| 11:28.37 | carrar | Puck`: http://bfy.tw/239L |
| 11:32.18 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 11:34.49 | Puck` | carrar: wow, you're .. wow, okay. Already fixed it (: |
| 11:34.59 | Puck` | fun to "unsocial" people on IRC |
| 11:35.58 | carrar | World of Warcraft? |
| 11:36.18 | ketas | ulaw is best codec, right? |
| 11:36.37 | carrar | g722 sounds better |
| 11:36.42 | ketas | hmm |
| 11:36.51 | ketas | does it? |
| 11:37.17 | carrar | between phone and phone yes |
| 11:37.24 | carrar | phones that support g722 yes |
| 11:38.08 | WIMPy | There are even better ones. |
| 11:38.28 | ketas | well, are they supported... :P |
| 11:39.15 | WIMPy | But it's already twice as good, so not a bad choice. |
| 11:43.24 | ketas | stop, it doesn't fit |
| 11:43.42 | ketas | that's sometimes problem |
| 11:43.55 | ketas | (in voip too) |
| 11:44.18 | WIMPy | What doesn't fit? |
| 11:46.44 | carrar | Costco hotdogs in those cheap small buns |
| 11:47.51 | WIMPy | Are the buns are more important to you than the saussages? |
| 11:48.25 | carrar | They go together, one without the other is a let down |
| 11:48.28 | ketas | they go together |
| 11:48.35 | ketas | oh! |
| 11:48.58 | WIMPy | alwas thought of the bun as a sort of handle for the saussage. |
| 11:49.00 | carrar | Alex I'll take burgers for 400 |
| 11:50.21 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:50.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:56.34 | ketas | weird, codecs change midstream? |
| 11:57.30 | ketas | hmm |
| 11:58.15 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:09.40 | ketas | at least i didn't cross the streams |
| 12:10.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:12.01 | ketas | thanks to my isp which now only does sip... i have very good asterisk knowledge :P |
| 12:12.31 | ketas | 2 connections, 2 numbers each... fun to play |
| 12:24.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:44.20 | *** join/#asterisk davlefouAMD (~davlefouA@197.15.12.10) |
| 12:54.47 | *** join/#asterisk wonderworld (~ww@89.204.154.33) |
| 12:57.07 | *** join/#asterisk twanny796 (d4388db5@gateway/web/freenode/ip.212.56.141.181) |
| 12:57.47 | twanny796 | I need full fledged cdr software on linux. |
| 12:58.47 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 12:59.52 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 13:03.00 | catphish | got another segfault today after upgrading to 13.5 :( |
| 13:03.10 | [TK]D-Fender | releases a Fledge Expansion pack to devalidate all existing software |
| 13:11.01 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:13.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:14.14 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:14.44 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:17.43 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:19.33 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 13:19.36 | wasanzy | hello |
| 13:20.01 | wasanzy | is it possible to specify more than one host for sip in sip.conf |
| 13:21.18 | catphish | more than one remote SIP host to talk to? |
| 13:21.33 | *** join/#asterisk astpain (~manon@46.10.238.204) |
| 13:21.40 | catphish | if so, then yes, asterisk is designed to talk to lots of SIP devices |
| 13:23.24 | Phil-Work | wasanzy, I assume you mean by way of a failover? |
| 13:23.46 | wasanzy | <catphish: yes |
| 13:24.11 | [TK]D-Fender | wasanzy: No |
| 13:24.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:24.44 | [TK]D-Fender | wasanzy: Every entry = 1 host line only |
| 13:27.19 | wasanzy | so I can't do host=IP host=IP host=IP per line? |
| 13:28.23 | file | chan_sip does not allow that in a single peer |
| 13:29.43 | catphish | wasanzy: you have to create a separate entry per IP |
| 13:30.07 | astpain | has anyone here used asterisk to make a lot of simultanious outgoing calls |
| 13:30.12 | astpain | like 100-200 |
| 13:30.24 | astpain | concurent |
| 13:30.40 | [TK]D-Fender | That isn't a "lot" |
| 13:30.45 | wasanzy | ok |
| 13:33.53 | astpain | well we are playing audio files from the dial plan and after about an hour of working fine the communication with our SBC breaks |
| 13:34.18 | astpain | and it starts calling numbers without playing the audio files |
| 13:34.48 | [TK]D-Fender | And you've proven that * isn't sending the RTP? |
| 13:36.12 | astpain | we have debuged the SIP communication and found out that * is not sending back ACK to the SBC |
| 13:36.55 | [TK]D-Fender | What versiona re you running? |
| 13:36.56 | astpain | so we think the problem happens before any RTP is sent |
| 13:37.11 | astpain | 13, 13.4, 13.5 |
| 13:37.15 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 13:37.29 | astpain | and we have tried a few of the 12 flavors |
| 13:37.31 | [TK]D-Fender | Barring a bug, the load itself is nothing |
| 13:37.37 | [TK]D-Fender | 12=dead |
| 13:37.41 | [TK]D-Fender | 11 or 13 only |
| 13:37.45 | astpain | yes |
| 13:37.53 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+dimensioning <- for reference |
| 13:38.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:38.11 | astpain | we also tried 11 and it worked fine |
| 13:38.21 | astpain | but it is missing some vital features that we need |
| 13:38.36 | [TK]D-Fender | Could be a bug at that point. post it up and see. |
| 13:38.48 | astpain | ok, thanks |
| 13:39.20 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:42.03 | catphish | does anyone have any idea what could be causing this? http://paste.codebasehq.com/pastes/hys0ngkntdml2lx1j0 "json.c: Error building JSON from '{s: s, s: s, s: s, s: i, s: i, s: i, s: o}': Invalid UTF-8 string" |
| 13:42.16 | catphish | it seems to ultimately result in a segfault |
| 13:42.55 | catphish | i'll go ahead and raise a ticket, since i have a couple of things worth reporting about it now |
| 13:46.39 | *** join/#asterisk mub (~jub@50-205-171-170-static.hfc.comcastbusiness.net) |
| 13:50.21 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-agmlkzilpejnmrtv) |
| 13:50.21 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:54.35 | *** part/#asterisk chendy (~alexc@113.116.60.225) |
| 13:58.09 | *** join/#asterisk cw1972 (~cw1973@host81-136-221-45.in-addr.btopenworld.com) |
| 13:58.38 | wasanzy | am confuse as in how I can add the separate entery for them |
| 13:59.25 | wasanzy | please any example? because I keep loosing connection to the Host, they have more than one host so am sure using all of them will be good. |
| 14:01.44 | [TK]D-Fender | No such thing as a "sample" |
| 14:01.46 | [TK]D-Fender | MAKE MORE PEERS |
| 14:02.00 | [TK]D-Fender | This isn't a "thing". |
| 14:02.32 | wasanzy | I should make more peers ok, but how will that failover? |
| 14:02.55 | catphish | raised a bug in jira for my ongoing asterisk 13 segfaults https://issues.asterisk.org/jira/browse/ASTERISK-25436 |
| 14:03.50 | *** join/#asterisk kharwell (kharwell@nat/digium/x-jjiznphgdhezebdp) |
| 14:04.36 | [TK]D-Fender | WaYour dialplan. |
| 14:04.41 | [TK]D-Fender | wasanzy: Your dialplan. |
| 14:06.23 | wasanzy | Can I have multiple entery like this? http://pastebin.com/ZQ1vwtT8 |
| 14:06.38 | [TK]D-Fender | No |
| 14:07.29 | wasanzy | ok |
| 14:07.39 | catphish | wasanzy: they need to have different names |
| 14:08.00 | wasanzy | You mean I have to use my dialplan to control failover |
| 14:08.12 | [TK]D-Fender | [10:04][TK]D-Fenderwasanzy: Your dialplan. <------------------ |
| 14:08.45 | catphish | wasanzy: yes, the simplest way is just to dial all the peers simultaneously, or sequentially |
| 14:09.02 | [TK]D-Fender | catphish: Former = horribly bad idea |
| 14:09.13 | catphish | [TK]D-Fender: why? |
| 14:09.28 | catphish | [TK]D-Fender: because i do this a lot, and it works well |
| 14:09.31 | [TK]D-Fender | You going to bombard every callee with maybe 5 SIMULTANEOUS calls? |
| 14:09.55 | wasanzy | ok |
| 14:10.07 | [TK]D-Fender | And they watch their "missed calls" counters go through the roof and some providers start billing from the moment it's ringing. |
| 14:10.07 | catphish | [TK]D-Fender: i suppose it depends whether the end user wants all their devices to ring or not |
| 14:10.12 | [TK]D-Fender | TON of reasons not to |
| 14:10.32 | [TK]D-Fender | catphish: We aren't talking user phones here... ITSP CALLS <- |
| 14:10.49 | [TK]D-Fender | the PROVIDER hosts are not reliable |
| 14:11.11 | catphish | oh i see, i missed that part, my suggestion was ridiculous then, sorry |
| 14:12.09 | [TK]D-Fender | yup, gotta read back... |
| 14:12.27 | catphish | not sure if i should downgrade to 1.8, or risk 11, which i've never tested |
| 14:13.36 | catphish | 1.8 was super stable |
| 14:14.11 | *** join/#asterisk hjf (~hjf@unaffiliated/hjf) |
| 14:14.16 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:14.44 | [TK]D-Fender | 1.8 = dead |
| 14:14.50 | [TK]D-Fender | 11 getting older now. |
| 14:15.46 | [TK]D-Fender | soo 11 if you're "slow", and 13 if you're actually not planning on going through all this again in another 2 years |
| 14:16.30 | catphish | i'm using 13 right now, daily segfaults are proving problematic to that plan :) |
| 14:18.16 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 14:21.05 | [TK]D-Fender | Yup, that would be an issue.... |
| 14:21.30 | catphish | sadly i'll have to go back to 1.8 until i can test a new upgrade path |
| 14:22.12 | catphish | i fear i'm using some less supported modules, and may need to move away from those |
| 14:22.32 | catphish | most likely i'll go back to 1.8 until i can re-engineer my solution about the new stuff in 13 |
| 14:22.39 | catphish | *around |
| 14:22.45 | wdoekes | catphish: I suspect far_ident contains some invalid utf8 on line 266 in app_fax.c |
| 14:26.12 | *** join/#asterisk Dovid (~dovid@ool-45738930.dyn.optonline.net) |
| 14:29.41 | catphish | wdoekes: thanks for looking, is there any chance i can do something trivial to work around this? |
| 14:31.15 | catphish | my 13.5.0 also appears to be leaking memory :( |
| 14:31.31 | catphish | the core dump was nearly 10GB |
| 14:35.26 | mub | On our vicidial server we have inbound calls sent to our voicemail extension if they've been waiting for more than 2 minutes. However, in OSDial this feature only seems to work on outbound calls. |
| 14:36.00 | mub | aaand our mail server stopped working which is more important kthxbai |
| 14:36.12 | [TK]D-Fender | #chainfail |
| 14:38.20 | *** part/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 14:38.27 | wdoekes | catphish: best thing would be if you could figure out what is wrong exactly and then fix that: if you patch asterisk to write out the values before putting them into the ast_json_pack, you can look at those and determ... |
| 14:38.31 | wdoekes | thanks for leaving mid-sentence |
| 14:39.19 | *** join/#asterisk VultureZ (~vulture@185.22.183.196) |
| 14:39.47 | file | wdoekes, I'll never leave you! |
| 14:39.55 | wdoekes | thank you file! |
| 14:39.58 | wdoekes | I needed that |
| 14:40.04 | file | ...unless my internet goes down, or my IRC proxy dies |
| 14:40.42 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 14:40.43 | wdoekes | don't say that, now you have me worried again... |
| 14:41.33 | wdoekes | catphish: best thing would be if you could figure out what is wrong exactly and then fix that: if you patch asterisk to write out the values before putting them into the ast_json_pack, |
| 14:41.39 | wdoekes | you can look at those and determ... |
| 14:41.48 | wdoekes | ...ine why they're invalid |
| 14:42.36 | catphish | that makes sense |
| 14:42.42 | wdoekes | it would involve crashing asterisk another time; although you could possibly patch around that too, at once |
| 14:43.30 | *** join/#asterisk hjf (~hjf@unaffiliated/hjf) |
| 14:44.44 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 14:45.09 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 14:45.28 | catphish | i had another segfault the day before under 13.2.0 - i'm not sure if it's related http://pastebin.com/8dG915tc |
| 14:47.09 | catphish | wdoekes: thank you for your assistance, as much as i wish i could assist further, it needs to be my priority to stop the crashing, i will therefore need to downgrade my production systems, but i will try to reproduce this in a controlled environment |
| 14:47.30 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 14:48.01 | catphish | i'm not sure where the non-ascii data might be coming from, obvious sources are my database, or incoming callerid |
| 14:48.03 | wdoekes | catphish: something like this http://fpaste.org/273170/raw/ |
| 14:48.58 | wdoekes | correction: http://fpaste.org/273172/raw/ |
| 14:50.44 | catphish | so ast_json_pack is failing correctly? but something later on is trying to access the non-existent json object? |
| 14:51.06 | wdoekes | that's the quick conclusion I draw from your debug statement + traceback |
| 14:52.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:55.34 | catphish | thank you |
| 15:03.10 | *** join/#asterisk glNito (6ca1d582@gateway/web/freenode/ip.108.161.213.130) |
| 15:04.38 | *** join/#asterisk aness (~aness@cm-84.215.91.31.getinternet.no) |
| 15:05.04 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 15:06.15 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 15:06.59 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:08.37 | *** join/#asterisk wonderworld (~ww@82.113.99.142) |
| 15:08.49 | *** join/#asterisk nunopereira (~nuno.pere@alpes.nortenet.pt) |
| 15:09.59 | hjf | ok... finally managed to establish a SIP call |
| 15:10.08 | hjf | the problem was the ITSP |
| 15:10.16 | hjf | they were "whitelisting" the wrong IP |
| 15:10.18 | hjf | idiots |
| 15:10.52 | hjf | they made me lose one day for being "there is no problem on our end" |
| 15:14.17 | *** join/#asterisk u0m3 (~u0m3@92.80.92.123) |
| 15:14.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:15.09 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:15.25 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-trczhfpjicewbhkn) |
| 15:19.58 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:28.19 | *** join/#asterisk glNito (6ca1d582@gateway/web/freenode/ip.108.161.213.130) |
| 15:33.01 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 15:34.02 | hjf | i have this sip "trunk" with several lines, i have to set the callerid to the proper number so my provider will allow the call, |
| 15:34.17 | hjf | i set the callerid on each peer |
| 15:34.37 | hjf | how do i "forward" this? on the dialplan |
| 15:36.09 | [TK]D-Fender | What does "forward" mean here |
| 15:36.43 | [TK]D-Fender | When a call is placed that is what it will use |
| 15:36.52 | [TK]D-Fender | or "fromuser" if set |
| 15:38.32 | [TK]D-Fender | BRB |
| 15:44.34 | *** join/#asterisk BBone_ (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 15:45.45 | catphish | hjf: the local peer's callerid should pass through |
| 15:56.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:59.04 | hjf | sorry... i was setting it wrong. it now works with callerid |
| 15:59.45 | hjf | is it possible to make the trunk return a busy tone to a caller when the extension is offline? |
| 16:00.06 | igcewieling | hf yes. |
| 16:00.15 | igcewieling | ..er.. hjf: yes |
| 16:00.24 | hjf | what happens now is that i get a "invalid number" message when i call an offline extension |
| 16:00.27 | hjf | igcewieling: how? |
| 16:09.26 | [TK]D-Fender | "core show application playtones" <- |
| 16:09.31 | [TK]D-Fender | "core show application playtones" <- |
| 16:09.36 | [TK]D-Fender | "core show application busy" <- |
| 16:09.44 | [TK]D-Fender | "core show application congestion" <- |
| 16:09.46 | [TK]D-Fender | take your pick |
| 16:10.53 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:10.53 | igcewieling | hjf: something like this after the dial ExecIf($["${HANGUPCAUSE}" == "20"]?Busy); |
| 16:10.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:12.19 | *** join/#asterisk davlefouAMD (~davlefouA@197.15.12.10) |
| 16:13.53 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:14.20 | igcewieling | This might be more what you are looking for. ExecIf($["${DIALSTATUS}" == "ANSWER"]?Busy); |
| 16:19.17 | *** join/#asterisk fergus (~sergei@178.120.216.126) |
| 16:23.24 | *** join/#asterisk jrrose (jon@nat/digium/x-kgwgajajrsfrvxgx) |
| 16:28.41 | *** join/#asterisk BBone_ (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 16:37.12 | hjf | sigh. i wonder why they make it say "phone number does not belong to a client in service" instead of "phone number is unavailable" or just a busy tone |
| 16:38.00 | hjf | at least where i live, if you leave your phone off the hook, if someone calls it gives a busy tone. but after some time (hours?) it just says "the call could not be established" |
| 16:38.05 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 16:38.45 | hjf | that would be a better message. i mean, my SIP provider's numbers are in service. they're just offline at the moment |
| 16:39.34 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 16:40.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:48.06 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:59.34 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 17:00.10 | *** join/#asterisk aness (~aness@cm-84.215.91.31.getinternet.no) |
| 17:09.49 | *** join/#asterisk bluemerlin (~bluemerli@109.236.171.1) |
| 17:14.31 | hjf | hah ... nice. my client is MAD a the itsp now |
| 17:14.48 | hjf | they sold him this "trunk" and told him "sure we can give you as many numbers as you want" |
| 17:15.01 | hjf | turns out they only provide 10 SIP channels |
| 17:15.13 | hjf | so they can give him 1000 numbers but only 10 concurrent calls |
| 17:17.03 | [TK]D-Fender | Which is very normal |
| 17:17.29 | [TK]D-Fender | And they won't sell him MORE than 10 concurrent? |
| 17:17.39 | [TK]D-Fender | Maybe that's all he's paying for NOW, but you saying they won't sell him more? |
| 17:18.55 | hjf | no.. he's mad because they told him "10 extensions" |
| 17:19.10 | hjf | and now it turns out it's 10 channels, no extensions |
| 17:19.28 | hjf | i told him yeah doesn't make sense that they sell you "extensions", since extensions are something of your PBX |
| 17:20.13 | hjf | [TK]D-Fender: btw they're charging him $100/mo for 3 numbers and 10 channels now. dunno what they charge for more numbers |
| 17:21.08 | [TK]D-Fender | Sounds normal so far |
| 17:21.43 | [TK]D-Fender | DIDs tend to vary depending where you are |
| 17:22.01 | [TK]D-Fender | US ~ $1-$1.50 each at many places |
| 17:22.23 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 17:26.29 | hjf | still, it's a big mess |
| 17:26.34 | hjf | he-said-she-said |
| 17:26.40 | hjf | sales vs tech |
| 17:26.54 | *** join/#asterisk eofster (~eofster@host86-189-4-80.range86-189.btcentralplus.com) |
| 17:26.59 | hjf | tech support who refuses to check stuff when you tell them it's not working |
| 17:28.08 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-gmhebvjupwslgpjm) |
| 17:30.54 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:37.22 | *** join/#asterisk eofster (~eofster@tmo-096-98.customers.d1-online.com) |
| 17:38.23 | igcewieling | or: customers who don't provide enough information for tech support to check stuff |
| 17:39.23 | robmal | But you know it's broken, figure it out yourself. |
| 17:41.01 | hjf | igcewieling: i called them, the guy just said "everything is ok on our end, i don't even see you reaching our host" |
| 17:41.16 | hjf | so i got a little aggressive |
| 17:41.23 | igcewieling | hjf: I was speaking generally. |
| 17:41.33 | hjf | they put me to a more senior person... and turns out they whitelisted the wrong ip |
| 17:41.41 | hjf | mine was .134 and they whitelisted .133 |
| 17:41.55 | igcewieling | *nod* |
| 17:42.09 | hjf | once they sorted that, everything worked fine. pretty impressive call quality too |
| 17:42.43 | hjf | they aren't that kind of "virtual" itsp, low cost stuff |
| 17:43.27 | *** join/#asterisk ipso (~ipso@184.68.39.238) |
| 17:43.32 | hjf | i noticed because i tried several providers and in all cases i got "private number" at the caller id |
| 17:43.41 | hjf | with them i get a caller id display |
| 17:43.44 | ipso | Is there a way from the CLI to get the time of the last processed call? |
| 17:44.15 | hjf | private number sucks... myself,i don't pick up calls from private numbers cause it's ALWAYS someone trying to sell something |
| 17:45.12 | robmal | hjf: It's usually because of bad number format on your side, maybe some P-Asserted-Identity and lots of other variables. |
| 17:45.32 | hjf | robmal: they don't accept caller id settings |
| 17:45.43 | hjf | you can't "spoof" a caller id |
| 17:45.49 | robmal | Of course i can. |
| 17:46.03 | hjf | and honestly i dont know why in the US you can do that |
| 17:46.45 | robmal | You can do that everywhere but it's also illegal ;-) |
| 17:47.05 | hjf | you can't do that here in Argentina |
| 17:47.21 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 17:47.32 | hjf | this company uses callerid to identify your call. if i try another number not associated to my account, the call is just refused |
| 17:48.04 | robmal | Well, that's how it's supposed to be done so good choice. |
| 17:48.18 | hjf | and with the low-cost ITSPs, shadier companies, they didn't even have the option to set caller id outbound themselves |
| 17:48.37 | mub | hjf: I work at a really sketchy call center. We have to spoof as much as possible so that our outbound/inbound dialing campaigns aren't linked to the company in any way |
| 17:48.55 | mub | None of our numbers appear as ours |
| 17:50.01 | hjf | best you can do here is just "private number" |
| 17:50.07 | hjf | you can't fake someone else's number |
| 17:50.34 | robmal | You weren't trying hard enough ;-) |
| 17:50.47 | mub | I don't think it's right, but I also don't have any say in the matter. |
| 17:50.55 | mub | I just do what I'm told :( |
| 17:50.57 | hjf | not myself, i'mt alking about people like call centers |
| 17:51.04 | hjf | and they are big VOIP users |
| 17:51.14 | hjf | but all call center calls i received were just "private number" |
| 17:51.42 | hjf | must be a reason why they do that |
| 17:51.46 | hjf | probably: they can't |
| 17:52.36 | hjf | maybe there is a list of phone number blocks assigned to each phone company? |
| 17:52.54 | hjf | so when one phone company exchanges calls with another, they check if the number is allowed |
| 17:53.42 | hjf | you could lie about caller ID within your own numbers, but not with other company's numbers |
| 17:54.29 | *** join/#asterisk bluemerlin (~bluemerli@cpc65602-trow6-2-0-cust1020.18-1.cable.virginm.net) |
| 17:55.46 | hjf | i researched that a couple of years ago and found that the phone company that owns the wires (monopoly) had some info on their website about how to interface with them |
| 17:55.51 | hjf | SS7 and other stuff |
| 18:02.16 | hjf | i took a 1-day course on fiber optics one day... some dude teaching how to splice fiber, |
| 18:02.45 | hjf | he mentioned that argentina had the first "optic fiber ring" installed for phone calls |
| 18:02.50 | hjf | dunno if that's just bullshit? |
| 18:03.10 | hjf | i googled around and all i could find is that it was installed in 1982 by NEC |
| 18:03.59 | *** join/#asterisk kroms (~textual@s087n070.csun.edu) |
| 18:07.27 | igcewieling | hjf: Providers not providing outgoing CID info? What country do you live in? |
| 18:07.58 | igcewieling | ah, sorry, do not assist people at telemarketing or collections agencies. |
| 18:10.01 | *** join/#asterisk zaf (~zaf@76.72.92.37) |
| 18:10.05 | hjf | igcewieling: "cheap" voip providers. i think they just have a bunch of analog lines and just dial *31# (hide caller ID) before calling |
| 18:13.07 | igcewieling | that is not a provider, that is a hobbiest. |
| 18:15.20 | *** join/#asterisk bulkorok (~Adium@92.206.232.166) |
| 18:17.41 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 18:18.17 | *** join/#asterisk bulkorok (~Adium@92.206.232.166) |
| 18:19.48 | *** join/#asterisk monsterco (~monsterco@207.35.222.226) |
| 18:22.43 | *** part/#asterisk kroms (~textual@s087n070.csun.edu) |
| 18:25.28 | *** join/#asterisk ibercom (b0551c97@gateway/web/freenode/ip.176.85.28.151) |
| 18:26.39 | WIMPy | hjf: But you don't have to use a local telco. Just get a SIP account somewhere in the world and fake whatever number you want from any country you want. |
| 18:27.09 | hjf | WIMPy: international calls dont have caller ID here |
| 18:27.20 | WIMPy | Huh? |
| 18:27.37 | WIMPy | Do you only have a single analog line in to the country? |
| 18:28.18 | hjf | no, international calls just don't show ID |
| 18:28.49 | hjf | and calls from pay phones show caller id 111111111 |
| 18:28.58 | WIMPy | Well, I'd get my telephony from another country then, so I can get caller ID. |
| 18:29.20 | ibercom | asterisk 11.20.0-rc1 -> If I haven't enabled ICE, why this error occurs? |
| 18:29.30 | ibercom | res_rtp_asterisk.c: Set role failed; no ice instance |
| 18:32.16 | ibercom | bug or config |
| 18:32.58 | WIMPy | What are you doing? Are you trying to do webrtc or something? |
| 18:33.25 | file | it occurs because the check is too broad |
| 18:33.34 | file | you can ignore it |
| 18:35.31 | ibercom | My log file starts to fill. |
| 18:38.35 | file | you can modify the source code and remove the log message, otherwise you can follow https://issues.asterisk.org/jira/browse/ASTERISK-25438 for an eventual patch - a community member has one up but it is incomplete |
| 18:38.55 | file | if they do not respond then it'll get done by someone else |
| 18:38.56 | ibercom | @file: As you say too broad. I think it is inappropriate. |
| 18:39.25 | *** part/#asterisk monsterco (~monsterco@207.35.222.226) |
| 18:40.39 | WIMPy | Is configure fixed in rc1? |
| 18:40.55 | file | I have no idea what you're talking about |
| 18:41.13 | file | the configure script itself hasn't been touched in a bit |
| 18:41.49 | WIMPy | If you don't have libxm2-dev, it will tell you that you can configure with --disable-xmldoc, but in 11.19 it won't compile that way any more. |
| 18:42.07 | ibercom | @file: One sip call, one log line. |
| 18:42.21 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 18:42.23 | file | unless there's been a JIRA issue created with information then nope, noone has looked at it |
| 18:42.33 | *** part/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 18:44.04 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 18:44.57 | file | https://gerrit.asterisk.org/#/c/1346/ there I just fixed it now |
| 18:47.37 | ibercom | @file: Dropping it to debug is ok for me. |
| 18:49.09 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:49.46 | WIMPy | The SIP TOS/COS marks are also annoying. Can't remember if they apper at verbose 2 by default or what. I've been patching them to debug for a long time. |
| 18:50.10 | WIMPy | That information is really not that interesting. It's always the same anyway. |
| 18:58.42 | *** join/#asterisk eofster (~eofster@ip5b4063a6.dynamic.kabel-deutschland.de) |
| 18:59.22 | shudon | hi all. noob here :) i can include a context in another context, but can i write an extension that will invoke commands from another extension or pass a channel off to another extension? |
| 18:59.53 | WIMPy | Sot sure what you want to do. |
| 19:00.15 | WIMPy | Sounds a little like Goto(). |
| 19:00.34 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:00.34 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:01.18 | shudon | hmmm |
| 19:01.21 | shudon | i think i just need to keep reading |
| 19:01.26 | shudon | thanks WIMPy :) |
| 19:16.29 | *** join/#asterisk bulkorok (~Adium@92.206.232.166) |
| 19:33.51 | *** join/#asterisk glNito (6ca1d582@gateway/web/freenode/ip.108.161.213.130) |
| 19:35.06 | *** part/#asterisk OmSai (~quassel@li510-44.members.linode.com) |
| 19:47.22 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 20:07.10 | *** join/#asterisk bluemerlin (~bluemerli@cpc65602-trow6-2-0-cust1020.18-1.cable.virginm.net) |
| 20:08.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:08.18 | *** join/#asterisk bluemerlin (~bluemerli@cpc65602-trow6-2-0-cust1020.18-1.cable.virginm.net) |
| 20:40.23 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:41.26 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 20:45.27 | *** join/#asterisk ibercom (53327c0b@gateway/web/freenode/ip.83.50.124.11) |
| 20:47.52 | Cuzner | same => n,Set(LOCAL(this_prefix)=${IF($[${RAND(1,2)} = 1])}?prefix1:prefix2) |
| 20:48.05 | Cuzner | I can't see anything wrong with that, where did i go wrong? |
| 20:48.09 | Cuzner | :) |
| 20:50.02 | shudon | o dm |
| 20:50.11 | shudon | whoops |
| 20:50.13 | Cuzner | oh i think i see it |
| 20:51.03 | ibercom | ) near 1 |
| 20:51.53 | ibercom | ) to the end |
| 20:52.10 | Cuzner | same => n,Set(LOCAL(this_prefix)=${IF($[${RAND(1,2)} = 1]?prefix1:prefix2)}) |
| 20:52.18 | Cuzner | i think that's what i wanted... |
| 20:53.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:11.35 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 21:12.40 | *** join/#asterisk xpheres (~bodhi64@dslb-088-073-011-220.088.073.pools.vodafone-ip.de) |
| 21:12.45 | *** join/#asterisk bluemerlin (~bluemerli@cpc65602-trow6-2-0-cust1020.18-1.cable.virginm.net) |
| 21:13.53 | xpheres | hello |
| 21:14.01 | xpheres | I managed to make SRTP works but not TLS |
| 21:14.04 | xpheres | can anyone help me? |
| 21:14.32 | xpheres | I've read that SRTP is useless without TLS |
| 21:17.47 | *** join/#asterisk t4nk997 (4b953aad@gateway/web/freenode/ip.75.149.58.173) |
| 21:31.47 | mub | Oh boy.. I've been asked to get all of our agents to log into 3 different inbound campaigns at once... I'm pretty sure that's not possible |
| 21:32.00 | mub | He's going to be an unhappy camper |
| 21:33.17 | *** join/#asterisk bluemerlin (~bluemerli@cpc65602-trow6-2-0-cust1020.18-1.cable.virginm.net) |
| 21:33.55 | [TK]D-Fender | It's your dialplan, do wahtever you want.... |
| 21:34.24 | mub | There's no way they can handle that many calls though.. |
| 21:34.33 | mub | They'll get hammered |
| 21:38.34 | glNito | just have them drink some water between calls |
| 21:41.37 | mub | glNito: lol they aren't allowed to have anything at their desks. No drinks, snacks, pencils, paper, and their cellphones MUST be "checked in" to the cellphone lockers. |
| 21:42.59 | mub | It's a dictatorship |
| 22:00.37 | *** join/#asterisk eofster (~eofster@ip5b4063a6.dynamic.kabel-deutschland.de) |
| 22:09.04 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 22:11.59 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:19.54 | *** join/#asterisk trdillon1 (d055a664@gateway/web/freenode/ip.208.85.166.100) |
| 22:20.51 | trdillon1 | Hello |
| 22:21.09 | trdillon1 | is it possible to have multiple outbound contexts? |
| 22:21.34 | trdillon1 | like have 1 dial plan go out 1 way and another dial plan go out another |
| 22:21.53 | WIMPy | That doesn't make sense. |
| 22:22.06 | trdillon1 | ok |
| 22:22.10 | WIMPy | Contexts are for inbound calls. |
| 22:22.16 | trdillon1 | i have 2 customers |
| 22:22.27 | trdillon1 | they both have separate numbers that the dial to enter the pbx |
| 22:22.40 | trdillon1 | when i transfer them out i want to sent them out 2 different sip providers |
| 22:22.41 | WIMPy | And what you do in them to make outbound calls with Dial() is entirely up to you. |
| 22:22.43 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:23.17 | WIMPy | So you put them in different contexts. |
| 22:24.17 | trdillon1 | great |
| 22:24.17 | trdillon1 | thanks |
| 22:24.21 | trdillon1 | that is what i thought |
| 22:31.57 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 22:35.44 | xpheres | <PROTECTED> |
| 22:35.48 | xpheres | anyone knows why? |
| 22:46.00 | *** part/#asterisk kharwell (kharwell@nat/digium/x-jjiznphgdhezebdp) |
| 22:55.39 | *** join/#asterisk WangDang (~wangdang@24.140.224.98) |
| 23:02.40 | *** join/#asterisk MarcoZink (~MarcoZink@189-210-78-151.static.axtel.net) |
| 23:05.52 | *** join/#asterisk bkruse (~Adium@162.17.78.98) |
| 23:08.41 | drab | hi, yesterday I was asking about managing different outoging ITSP for different departments in a company while allowing extensions in both departments to talk to each other |
| 23:09.13 | drab | it seemed that contexts were the right way to implement this so I took a stab but I'm unsure I'm doing it correctly and would love another pair of eyes before making changes |
| 23:09.26 | drab | here it is: http://pastebin.com/haZdzsXL does this make sense? |
| 23:10.23 | drab | as far as I understand if the eng phones call outside they will match itsp2 first and if they call biz the include will trigger since 1xx isn't found in the 2xx context and the internal extension will match and the call will be connected |
| 23:10.39 | *** join/#asterisk robmal (r@wporzo.pl) |
| 23:10.44 | drab | so there's no danger that biz will ever use itsp2 and eng itsp1, which is the real worry for people here |
| 23:11.00 | WIMPy | You should make 3 contexts. |
| 23:11.34 | WIMPy | 2 for the groups to call out and a 3rd one with all your local extensions that is included in the other two. |
| 23:11.58 | WIMPy | But it would also work the way you have it now. |
| 23:25.02 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 23:34.48 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 23:35.05 | drab | WIMPy: and the devices's context (in sip.conf) would be the outgoing group, correct? |
| 23:35.20 | drab | the respective outgoing group* |
| 23:36.43 | drab | also I'm interested in doing it the way that's most common, thinking if somebody else one day has to pick this up or we're gonna need to pay somebody to get help |
| 23:37.04 | drab | so if what you mentioned is what you've seen a lot around I'd rather go that way even if what I had come up with works too |
| 23:37.45 | drab | but I've no idea what's common in asterisk land, I tried to google for best practices to do this kind of thing and pretty much came back empty handed except for a recommendation to use contexts |
| 23:39.17 | WIMPy | It becomes interesting when it gets more complex. Say if only the biz people are allowed intenational calls. |
| 23:48.55 | gruetzkopf | i really need to ask the guys at the office how they manage to max out a E1 line with 8 people |
| 23:49.14 | *** join/#asterisk eofster (~eofster@ip5b4063a6.dynamic.kabel-deutschland.de) |
| 23:50.12 | WIMPy | Not bad. |
| 23:50.46 | WIMPy | A full E1? Are you sure everything is working as it should? |
| 23:51.02 | gruetzkopf | a full E1 |
| 23:52.20 | gruetzkopf | its all that conferencing |
| 23:52.40 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 23:52.43 | robmal | You know that it means all 8 people are on 3.75 calls in the same time? |
| 23:54.05 | WIMPy | 0.75 calls only works with VOIP, not with E1. |
| 23:55.28 | gruetzkopf | no, that means there are two people in two phone conferences, 3 simultaneous group 4 fax transmissions, some individual calls and two always-on dial connections for equipment monitoring |
| 23:56.19 | WIMPy | Yeah, Modern times. For a 3-way call you should only need one channel. |
| 23:57.23 | gruetzkopf | yeah, the PABX pushes 3 way calls to the telco, but thats far more than 3 participants |
| 23:58.10 | robmal | The fax transmissions bother me. Nobody uses fax this often. |
| 23:59.00 | WIMPy | What about all the fax spam? |
| 23:59.05 | gruetzkopf | i'm in germany. have you tried emailing a german government agency? |
| 23:59.26 | WIMPy | Yes. Didn't work. |
| 23:59.41 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:59.41 | robmal | gruetzkopf: Seriously? 4 at a time? All the time? |