| 00:02.50 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 00:19.56 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 00:23.09 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.90.41) |
| 00:25.39 | *** join/#asterisk theron (~theron@173.252.71.129) |
| 00:42.40 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:47.48 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 00:48.04 | ShapeShifter499 | hi |
| 00:49.02 | ShapeShifter499 | has anyone here unlocked a Linksys SPA-2102-R that has "Customization: acndigital" ? |
| 00:49.26 | ShapeShifter499 | I'm wondering how I can unlock or who could help me unlock it |
| 00:51.23 | *** join/#asterisk JTL (JTL@c-98-212-52-94.hsd1.in.comcast.net) |
| 01:05.47 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:43.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:52.25 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 02:00.46 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 02:17.53 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 02:39.23 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:44.54 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 02:53.31 | *** join/#asterisk bmurt (~brendan@64-121-18-195.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 02:54.22 | *** join/#asterisk Ispira (~Ispira@unaffiliated/ispira) |
| 02:55.09 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 03:01.39 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 03:05.24 | *** join/#asterisk theron (~theron@208.90.174.164) |
| 03:06.20 | *** part/#asterisk Ispira (~Ispira@unaffiliated/ispira) |
| 03:08.38 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:23.11 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:35.50 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 03:50.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 03:53.15 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.90.41) |
| 03:53.27 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:54.04 | *** join/#asterisk wonderworld (~ww@194-166-33-163.adsl.highway.telekom.at) |
| 03:56.55 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 03:57.57 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 04:04.25 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 04:09.20 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 04:13.27 | *** join/#asterisk asteriskmonkey1 (~Tardis@74-51-38-204.telnetcommunications.com) |
| 04:15.25 | asteriskmonkey1 | do x-headers pass transparently from one leg to another or do they have to be set again prior to forming second leg? |
| 04:16.05 | [TK]D-Fender | * isn't a proxy. |
| 04:16.09 | [TK]D-Fender | That says it all |
| 04:19.19 | asteriskmonkey1 | good point thanks |
| 04:20.07 | TazzNZ | you can make your dialplan "proxy" them over |
| 04:20.40 | [TK]D-Fender | If so inclined.... |
| 04:24.42 | asteriskmonkey1 | yep done it in dialplan |
| 04:24.44 | asteriskmonkey1 | :) |
| 04:30.44 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 04:32.24 | raspberrypifan | anyone playd with chan_dongle |
| 04:36.22 | *** join/#asterisk rnovotny22 (~rnovotny2@184-97-215-55.mpls.qwest.net) |
| 04:41.43 | [TK]D-Fender | Meta questions are poor conversation starters... |
| 04:42.31 | raspberrypifan | well i want a general view of it not much info it seems |
| 04:42.54 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 04:43.55 | [TK]D-Fender | I've seen plenty |
| 04:44.03 | raspberrypifan | well can u feed it to me |
| 04:44.05 | raspberrypifan | via google |
| 04:44.15 | [TK]D-Fender | Google gave me plenty of results. |
| 04:44.34 | [TK]D-Fender | can place & receive calls and SMS depending on the model |
| 04:44.38 | [TK]D-Fender | that's it. |
| 04:45.18 | [TK]D-Fender | Oh, and USSD |
| 04:50.57 | *** join/#asterisk maani (~babak@2.178.132.188) |
| 04:52.48 | *** join/#asterisk NoobSaibot_ (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 04:55.09 | raspberrypifan | whats ussd |
| 04:55.15 | raspberrypifan | but there are no real tutorials on settingi t up and shit |
| 04:55.26 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.90.41) |
| 04:55.43 | coppice | http://en.wikipedia.org/wiki/Unstructured_Supplementary_Service_Data |
| 04:58.03 | raspberrypifan | i read that but it didnt necesarily make that much sense |
| 04:59.34 | coppice | its a variant of SMS packets to and from the phone for application purposes. often things accessed by *<something> |
| 05:00.09 | raspberrypifan | hmm |
| 05:00.11 | raspberrypifan | interesting |
| 05:02.45 | [TK]D-Fender | https://www.google.ca/#q=asterisk+chan_dongle <- I found plenty of guides. |
| 05:05.45 | raspberrypifan | i wonder if it works with freepbx |
| 05:07.08 | [TK]D-Fender | SIXTH link from Google results |
| 05:07.21 | [TK]D-Fender | And that's with even looking for FreePBX specifically. |
| 05:07.23 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 05:08.14 | raspberrypifan | how do u install asterisk-addons, asterisk-devel, |
| 05:10.28 | [TK]D-Fender | <PROTECTED> |
| 05:12.25 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:14.03 | [TK]D-Fender | And the other is some distro-based meta-package. |
| 05:18.19 | raspberrypifan | Chan_dongle works with freepbx, you will need install asterisk-addons, asterisk-devel, get source of current freepbx asterisk release (1.8.15.0) and compile chan_dongle, |
| 05:20.59 | [TK]D-Fender | half of that is distro-based and you are vague on what you're installing on. |
| 05:21.30 | raspberrypifan | thats what it says on the forum |
| 05:21.45 | [TK]D-Fender | Depends on what YOU are installing on. |
| 05:22.19 | raspberrypifan | the pi/beaglebone now |
| 05:22.30 | [TK]D-Fender | Clearly not applicable |
| 05:22.41 | raspberrypifan | huh? |
| 05:22.42 | [TK]D-Fender | "as-is" |
| 05:23.11 | raspberrypifan | im stupid, i promise i have no idea what ur talking about |
| 05:23.14 | [TK]D-Fender | those are FREEPBX ISO DISTRO extra packages. |
| 05:23.34 | [TK]D-Fender | You don't seem to be conscious of the the concepts of distro-specific packages. |
| 05:24.04 | raspberrypifan | i figure freepbx is a whole thing by itself |
| 05:24.40 | [TK]D-Fender | it is a bolt-on GUI for Asterisk. |
| 05:24.51 | [TK]D-Fender | They also happen to make an ISO.... that is NOT for that arch |
| 05:26.00 | raspberrypifan | ur saying i cant do it or just that i need different instructions |
| 05:27.00 | [TK]D-Fender | different steps obviously |
| 05:27.08 | [TK]D-Fender | And you are complicating the whole process. |
| 05:27.12 | [TK]D-Fender | Go install *. |
| 05:27.33 | [TK]D-Fender | Then go install chan_dongle based on THEIR instructions adapting whatever you require for the pre-reqs |
| 05:28.00 | raspberrypifan | dont i already have * if im using freepbx |
| 05:28.38 | [TK]D-Fender | [01:27][TK]D-FenderThen go install chan_dongle based on THEIR instructions adapting whatever you require for the pre-reqs |
| 05:29.15 | raspberrypifan | whose instructions? |
| 05:30.05 | [TK]D-Fender | chan_dongle |
| 05:31.06 | raspberrypifan | do not follow the forum |
| 05:40.40 | *** join/#asterisk outerkom (~apache@41.164.20.123) |
| 05:40.53 | outerkom | anyknow know how to fix Got SIP response 480 "No Routes Found" |
| 05:41.10 | outerkom | I can initiate outbound calls over the trunk |
| 05:41.17 | outerkom | this just happens with follow me |
| 05:48.08 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 06:00.55 | outerkom | anyone know? |
| 06:07.17 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:15.44 | *** join/#asterisk bhavikpatel6842 (~root@122.169.5.68) |
| 06:17.28 | *** join/#asterisk Fwny (~potato@192-0-250-51.cpe.teksavvy.com) |
| 06:20.37 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:35.13 | outerkom | anyone know? |
| 06:35.18 | outerkom | anyknow know how to fix Got SIP response 480 "No Routes Found" |
| 06:35.21 | outerkom | I can initiate outbound calls over the trunk |
| 06:35.27 | outerkom | this just happens with follow me |
| 06:36.23 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 06:38.29 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 06:43.11 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 06:44.38 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 06:55.27 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.180.116) |
| 06:58.03 | *** join/#asterisk vlad_sta_ (~vlad_star@195.68.180.116) |
| 06:59.43 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:00.07 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:04.36 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:07.38 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:40be:6b5a:968d:32e1) |
| 07:37.55 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:51.22 | *** join/#asterisk CeBe (~CeBe@port-92-206-245-27.dynamic.qsc.de) |
| 07:59.16 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 08:03.04 | *** join/#asterisk maani (~babak@2.178.132.188) |
| 08:07.41 | *** join/#asterisk lucido (4e32cf81@gateway/web/freenode/ip.78.50.207.129) |
| 08:09.18 | lucido | hello, can asterisk use a linsys spa3102 as an fxs and an fxo interface? I#m building a fax detecting, fax sendin and receiving and answering machine server |
| 08:17.44 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 08:27.00 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:36.10 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 08:39.34 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 08:45.34 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 09:03.29 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 09:04.52 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:11.40 | *** join/#asterisk lucido (~chris@f050207129.adsl.alicedsl.de) |
| 09:15.55 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 09:22.47 | *** join/#asterisk DataWraith (~s1gny@188-195-8-4-dynip.superkabel.de) |
| 09:23.41 | snadge | one of our carriers has shut us off for doing 150 calls/sec .. rude :p |
| 09:24.18 | outerkom | I have an issue with an external number been treated as a local call |
| 09:24.26 | outerkom | I need this number to be put on the SIP trunnk |
| 09:24.34 | outerkom | I'm appending the # |
| 09:24.41 | snadge | apparently asterisk can break "carrier grade" voip equipment |
| 09:25.20 | lucido | can I use pap2t as fxs interface to asterisk? |
| 09:25.38 | outerkom | snadge, lucido |
| 09:25.46 | outerkom | HELP |
| 09:26.38 | lucido | whats an external number? |
| 09:26.49 | *** join/#asterisk Ice_Strike (~Ice_Strik@cpc1-oldh7-0-0-cust772.10-1.cable.virginm.net) |
| 09:27.12 | phpboy | snadge: that is a ridiculous statement |
| 09:29.43 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 09:30.01 | coppice | plenty of "carrier grade" voip equipment is very easy to break. |
| 09:33.25 | phpboy | I have an asterisk system dialing anything from 100,000 to 500,000 numbers a day |
| 09:33.41 | phpboy | I'd say the asterisk itself is carried grade in this instance :P |
| 09:35.15 | coppice | When its handling calls on the USS Bonhomme Richard its carrier grade |
| 09:35.27 | outerkom | I figured it out |
| 09:35.31 | outerkom | thanks guys! |
| 09:35.35 | phpboy | no problem |
| 09:46.59 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:49.29 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 09:49.37 | linocisco | hi all |
| 09:49.56 | linocisco | is there any free open asterisk server on cloud so that everyone can access and test? |
| 09:54.58 | *** part/#asterisk JTL (JTL@c-98-212-52-94.hsd1.in.comcast.net) |
| 10:09.00 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 10:24.11 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 10:27.38 | *** join/#asterisk yottanami (~yottanami@130.255.250.31) |
| 10:28.35 | yottanami | I can send a call to a queue with Goto(ext-queues,503,1) now how can I send a call to a extension directly? |
| 10:30.25 | *** join/#asterisk CeBe (~CeBe@port-92-206-245-27.dynamic.qsc.de) |
| 10:31.06 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 10:43.37 | yottanami | Can I use : Dial(SIP/300,1) ? |
| 10:57.48 | *** join/#asterisk slackie (~x@unaffiliated/slackie) |
| 10:59.50 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 11:20.14 | *** join/#asterisk matthew-moretalk (~matthew@88.96.27.150) |
| 11:20.42 | matthew-moretalk | Hi I know this is the Asterisk IRC but has anyone got any idea how to handle Daylight Saving Times in A2billing |
| 11:21.03 | matthew-moretalk | my Asterisk CDRs Show the correct time but the A2billing CDR is an hour behind! |
| 11:39.33 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 11:56.30 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 12:00.42 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 12:03.47 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:07.49 | *** join/#asterisk user258467 (~user25846@cue83-1-88-163-202-186.fbx.proxad.net) |
| 12:09.52 | user258467 | Hi sometimes my Peer 'xxx' is UNREACHABLE how do I fix it? |
| 12:12.14 | matthew-moretalk | well the device cant connect to the server either because of a connectivity issue or NAT Issue |
| 12:12.37 | matthew-moretalk | is this a sip trunk to a provider or a device like a softphone or SIP Handset? |
| 12:14.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:14.41 | workingcats | hi, is there some reasonably clean way to pass some information along to asterisk when making a call from a SIP client? |
| 12:14.52 | workingcats | all i really need is 6 numbers |
| 12:15.15 | [TK]D-Fender | make it part of your dial. |
| 12:15.22 | workingcats | oh wait i think i have an idea.. i could prepend that to the phone number, then split it up in asterisk and use the phone number to dial and the rest for my purposes |
| 12:15.27 | workingcats | i think thats what you mean? |
| 12:15.33 | [TK]D-Fender | yes |
| 12:15.39 | workingcats | great, ty |
| 12:17.16 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 12:19.34 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:25.46 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:27.47 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:29.27 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:36.23 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 12:45.06 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 12:58.36 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 13:03.17 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 13:03.50 | *** join/#asterisk wonderworld (~ww@194-166-33-163.adsl.highway.telekom.at) |
| 13:06.46 | *** join/#asterisk lucido (~chris@e179204095.adsl.alicedsl.de) |
| 13:10.55 | *** join/#asterisk mae_tae (~AndChat70@125.212.121.80) |
| 13:11.22 | mae_tae | hello people |
| 13:13.26 | mae_tae | how do i resolve this no audio problem it registers and ring but cannot i cannot hear anything |
| 13:16.41 | mae_tae | its behind nat |
| 13:17.24 | newtonr | using chan_sip? |
| 13:18.15 | snmp | telepathy mode on |
| 13:18.27 | newtonr | read the whole NAT SUPPORT section in the sip.conf.sample |
| 13:19.05 | snmp | directmedia=no |
| 13:19.06 | newtonr | then configure localnet, and externaddr or externhost appropriately |
| 13:19.07 | snmp | :) |
| 13:19.11 | newtonr | yeah and that |
| 13:19.34 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:19.44 | newtonr | also the nat= options |
| 13:20.06 | newtonr | and maybe a dozen other options depending on your very specific scenario |
| 13:21.04 | newtonr | but it is usually those few. Be aware of a SIP ALG mangling your packets however, and check your firewall to let the appropriate traffic through. |
| 13:26.19 | *** part/#asterisk yottanami (~yottanami@130.255.250.31) |
| 13:27.40 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 13:27.56 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 13:30.27 | mae_tae | ok |
| 13:30.33 | *** join/#asterisk lucido (~chris@e179204095.adsl.alicedsl.de) |
| 13:30.46 | lucido | what do you think about the sp3102 can I use it with asterisk? |
| 13:30.56 | lucido | for faxing and answering machine |
| 13:33.19 | *** join/#asterisk joesmoe (~joesmoe@c-73-181-72-237.hsd1.co.comcast.net) |
| 13:34.00 | *** join/#asterisk calum_ (~calum_@host81-154-219-160.range81-154.btcentralplus.com) |
| 13:34.21 | *** join/#asterisk dkakoti (~dkakoti88@120.59.229.92) |
| 13:34.41 | [TK]D-Fender | lucido: Should work. |
| 13:34.59 | dkakoti | hello everyone how to redirect a call |
| 13:35.01 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wbtvoagftdazdrhm) |
| 13:35.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:35.13 | lucido | [TK]D-Fender, yay! |
| 13:35.15 | [TK]D-Fender | dkakoti: When? How? What was the call doing prior? |
| 13:35.39 | WIMPy | And define "redirecting". |
| 13:35.49 | WIMPy | What technologies? |
| 13:36.10 | dkakoti | call is in bridge mode between A AND B . I need to use ami to transfer B to another call |
| 13:36.23 | [TK]D-Fender | .... Then use AMI.... |
| 13:36.29 | dkakoti | there is a function redirect() in ami |
| 13:36.32 | WIMPy | "Redirect" |
| 13:36.37 | [TK]D-Fender | The command is right there... what's the trouble you're having with it? |
| 13:36.42 | WIMPy | use it |
| 13:37.04 | dkakoti | I need the channel for B |
| 13:37.31 | dkakoti | How to get the channel info for B. |
| 13:37.50 | dkakoti | so that i can use the redirect() |
| 13:38.01 | WIMPy | Well, you need to know who you want to redirect, obviousely. |
| 13:38.49 | dkakoti | but channel is of form SIP/abcd12121 random digit |
| 13:38.54 | [TK]D-Fender | dkakoti: Get the channel dump for A... it will give you the bridged channel. |
| 13:39.23 | dkakoti | ok |
| 13:39.23 | WIMPy | We don't know how you want to decive whot to redirect. |
| 13:39.35 | WIMPy | decide |
| 13:41.26 | dkakoti | @WIMPy I jus want to know suppose A and B is in bridged mode and is there any way to force B to redirect from panel using ami redirect function |
| 13:41.44 | WIMPy | What panel? |
| 13:41.56 | WIMPy | How does the call get there? |
| 13:42.39 | [TK]D-Fender | dkakoti: Yes AMI will do the job. YOU have to look at the channel list to get the one you want. |
| 13:42.39 | dkakoti | type of panel freepbx offer |
| 13:42.52 | dkakoti | ok D-Fender |
| 13:44.05 | WIMPy | If you are able to display a call somewhere, you will know the channels involved. |
| 13:44.51 | [TK]D-Fender | I suspect he hadn't gotten that far... |
| 13:45.21 | [TK]D-Fender | dkakoti: Get the channel list. Display it for the user. Then act on the one they pick. |
| 13:45.48 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 13:49.30 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 13:53.15 | dkakoti | Ok @D-Fender and @WIMpy thank you for your help . I ll get the channel list from the current context A and B is in bridged mode and use redirect() to redirect B forcefully . Thanks a lot |
| 14:11.17 | snmp | itts a knights name: Thanksalot |
| 14:15.01 | *** join/#asterisk marcoAndres (~marco.dom@200-127-146-186.net.prima.net.ar) |
| 14:18.02 | D-Boy | Hello, My last asterisk 'touch' was 1.2 & 1.6, now i'm back to the voip server and see a lot of changes, there is asterisk 1.8 and 11.9 ? are those serving different purposes for asterisk ? what should I use ? |
| 14:19.00 | snmp | it depends |
| 14:19.02 | snmp | http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 14:19.52 | D-Boy | snmp: i am on that page, that's confusing, is asterisk 11.9 a continuation of asterisk 1.8 ? |
| 14:20.16 | D-Boy | (there is also an asterisk 12) |
| 14:20.33 | D-Boy | I see the LTS 'option' but is that only the main difference ? |
| 14:21.05 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 14:22.14 | file | ummm they are each newer than the last, with new features/changes |
| 14:22.26 | snmp | googling for a comparison chart |
| 14:23.00 | D-Boy | file: i see that the 1.8 branch still 'active' so presuming the 11.9 break things with the 1.8 ? |
| 14:23.32 | file | it's different. |
| 14:23.36 | [TK]D-Fender | Just like every branch, there are changes. |
| 14:23.39 | D-Boy | so for a new start, i should use the 'higher' version ? |
| 14:23.43 | [TK]D-Fender | Some may require tweaking |
| 14:23.50 | file | you should check when each thing will be supported until and choose accordingly |
| 14:23.51 | [TK]D-Fender | You should be on 11 or 12. |
| 14:24.15 | [TK]D-Fender | 11 = LTS, 12 = current with some substantial new features you may want to start with planning forward |
| 14:24.26 | D-Boy | okey, thx guys |
| 14:24.28 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions?src=search |
| 14:24.41 | file | 1.8 goes security fix only October 21st of this year |
| 14:25.13 | anonymouz666 | [TK]D-Fender: just curious, are you using the asterisk 12? |
| 14:25.13 | D-Boy | (of next year) |
| 14:25.21 | D-Boy | but yes i see clearly now \o/ |
| 14:25.32 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 14:25.34 | file | no, of this year. |
| 14:25.46 | file | security fix only this year, end of life next year |
| 14:25.54 | D-Boy | file: you're right :) |
| 14:26.07 | D-Boy | so why the jump from 1.8 to 11 ? |
| 14:26.13 | D-Boy | (i mean in versioning) |
| 14:27.00 | [TK]D-Fender | marketing |
| 14:27.59 | D-Boy | Okey, wish me luck ^^ |
| 14:28.12 | D-Boy | file: the wiki is a great start, thx ! |
| 14:29.24 | snmp | Tomlinson Holman's eXperiment... |
| 14:31.25 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:39.20 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 14:40.00 | mjordan | I think we should change up the version numbering again. Maybe start going backwards. |
| 14:40.18 | mjordan | Instead of Asterisk 13, the next version will just be Asterisk 11-2. |
| 14:40.59 | sgriepentrog | I like Asterisk9. Kind of like plan9... |
| 14:41.09 | mjordan | from outer space! |
| 14:45.18 | *** join/#asterisk wonderworld (~ww@194-166-33-163.adsl.highway.telekom.at) |
| 14:45.23 | jameswf | Asterisk ONE |
| 14:45.43 | newtonr | we should quit using the word Asterisk and start using the character. So the next version is *11-2 |
| 14:46.48 | *** join/#asterisk theron (~theron@66.220.145.145) |
| 14:47.13 | *** join/#asterisk navaismo (~navaismo@189.191.215.28) |
| 14:56.19 | Qwell | newtonr: While we're at it... Versions are overrated. What version of Chrome are you running? I don't know either. Let's just do auto-upgrades to the latest release. |
| 14:58.11 | newtonr | Qwell, yeah, that would be so fast and easy we could tack on the word "express" *Express |
| 14:59.25 | jameswf | wonders if coffee should be under "cool stuff" |
| 14:59.41 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-hbcbarjlturojdax) |
| 15:02.07 | D-Boy | suggests the 'SOA' versionning :p YYYYMMDDXX |
| 15:03.45 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:40be:6b5a:968d:32e1) |
| 15:07.55 | jameswf | could do.... Asterisk puprle, asterisk gold, asterisk beige |
| 15:08.37 | [TK]D-Fender | Well ABE already started running through the alphabet.... |
| 15:08.50 | [TK]D-Fender | And colour-coding is already done by Elastix. |
| 15:09.00 | [TK]D-Fender | You need a new paradigm |
| 15:09.02 | Qwell | jameswf: and then if you need a special color for somebody you're mad at, you can use brown |
| 15:09.57 | jameswf | never use the brown versions.... they are complete crap |
| 15:10.08 | Qwell | That's the joke... |
| 15:10.35 | navaismo | what about Nordic Gods |
| 15:10.46 | Qwell | (PIAF actually did use brown though, because they were mad about Certified Asterisk) |
| 15:11.17 | navaismo | or Anime series like Asterisk-Fullmetall |
| 15:12.35 | Qwell | jameswf: We could do that with FreePBX (and use color he already has...). Ward would lose his mind. "Introducing, PIAF yellow, with FreePBX silver!" |
| 15:13.15 | jameswf | and Asterisk blue |
| 15:15.02 | Qwell | I still like the versioning scheme for TeX. "Since version 3, TeX has used an idiosyncratic version numbering system, where updates have been indicated by adding an extra digit at the end of the decimal, so that the version number asymptotically approaches pi." |
| 15:15.05 | newtonr | only if you use crayon color names http://www.colourlovers.com/web/blog/2008/04/22/all-120-crayon-names-color-codes-and-fun-facts |
| 15:15.32 | newtonr | Asterisk Piggy Pink |
| 15:15.43 | Qwell | newtonr: it's all about Banana Mania |
| 15:16.26 | newtonr | Laser Lemon! There is many good ones |
| 15:18.12 | newtonr | There are? |
| 15:18.18 | newtonr | My words are fail |
| 15:19.05 | user258467 | How do I display name instead of number in asterisk log? |
| 15:19.19 | [TK]D-Fender | What log? |
| 15:19.23 | user258467 | verbose |
| 15:19.30 | [TK]D-Fender | NoOP it. |
| 15:19.35 | [TK]D-Fender | Or Verbose |
| 15:19.55 | WIMPy | doesn't know anywhere where Asterisk logs a number without the name. |
| 15:20.36 | user258467 | perhaps i have to specify an alias |
| 15:21.05 | [TK]D-Fender | nope |
| 15:21.55 | matthew-moretalk | anyone here good with a2billing? |
| 15:23.02 | *** join/#asterisk FuriousGeorge_ (45736701@gateway/web/freenode/ip.69.115.103.1) |
| 15:25.52 | FuriousGeorge | greetings all |
| 15:26.48 | *** part/#asterisk FuriousGeorge (45736701@gateway/web/freenode/ip.69.115.103.1) |
| 15:26.55 | *** join/#asterisk FuriousGeorge (45736701@gateway/web/freenode/ip.69.115.103.1) |
| 15:31.02 | FuriousGeorge | quit today |
| 15:31.08 | FuriousGeorge | quiet* |
| 15:31.30 | navaismo | ~ask |
| 15:31.30 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:31.42 | navaismo | matthew-moretalk & FuriousGeorge ^ |
| 15:32.47 | FuriousGeorge | on inbound calls, "if you know your party's extension" the IVR handles inbound calls differently depending on the time of day |
| 15:32.59 | FuriousGeorge | http://pastebin.com/2CG07Mtj |
| 15:33.36 | FuriousGeorge | users just taught themselves to do a blind xfer, and it turns out that the callerid(num) is not updated in the case of blind x-fer |
| 15:34.01 | FuriousGeorge | so a blind xfer to another extension will go to the transferring party's vm on NOANSWER |
| 15:34.34 | [TK]D-Fender | proper blind x-fer shows the original callerid. |
| 15:34.41 | [TK]D-Fender | And I don't see your code caring at all. |
| 15:37.08 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:37.54 | user258467 | Does the conf is in mysql database or in sip.conf? because I have some users in the database and some sip in sip.conf? |
| 15:38.31 | [TK]D-Fender | Depends how you configured your system |
| 15:38.59 | user258467 | [TK]D-Fender, Where do I know if my conf prefer mysql or conf files? |
| 15:39.17 | [TK]D-Fender | in your conf files. |
| 15:39.19 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 15:39.27 | [TK]D-Fender | It's your system... how do you not know what you configured? |
| 15:39.42 | [TK]D-Fender | Do you see the DB setup for Realtime? |
| 15:39.56 | *** join/#asterisk JJDeveloper (~joedevelo@104.0.141.33) |
| 15:40.04 | user258467 | [TK]D-Fender, I have access to the db right |
| 15:40.34 | [TK]D-Fender | user258467: How do I know you even have a DB? You have not told us much of anything about your system.... |
| 15:41.07 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:41.11 | *** join/#asterisk Draecos (~Draecos@106-69-50-73.dyn.iinet.net.au) |
| 15:41.21 | JJDeveloper | I am trying to figure out how to get Asterisk to answer the phone for 2 different SIP providers. One of my providers is IP-based, and doesn't require a password. The other provider is a domain with a username/password. Is there some simple example out there how to do this? |
| 15:41.23 | [TK]D-Fender | user258467: Asterisk does not require a DB for configuration. That is simply an option. You need to configure it to use one. |
| 15:42.09 | [TK]D-Fender | JJDeveloper: make a peer for each. Set the one not requiring auth as "insecure=port,invite", and "no" on the other |
| 15:42.45 | *** join/#asterisk lucido (~chris@e179204095.adsl.alicedsl.de) |
| 15:43.12 | Qwell | ~book |
| 15:43.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:43.13 | JJDeveloper | [TK]D-Fender - so if one is [general], just put the other in [general-sip2]? |
| 15:44.20 | [TK]D-Fender | JJDeveloper: [general] is NOT a peer entry |
| 15:44.30 | [TK]D-Fender | JJDeveloper: that is for global settings. Make a proper peer for each |
| 15:44.53 | jameswf | wonders if he should make his arduino an over complicated BLF |
| 15:45.31 | [TK]D-Fender | jameswf: res_goldberg.so |
| 15:46.18 | *** part/#asterisk user258467 (~user25846@cue83-1-88-163-202-186.fbx.proxad.net) |
| 15:46.48 | FuriousGeorge | I updated the PB to show the line that calls the macro |
| 15:47.14 | [TK]D-Fender | FuriousGeorge: You can't "update" it. It creates a NEW link |
| 15:47.43 | FuriousGeorge | [TK]D-Fender: good to see you again. i meant that I updated the pb, not that I changed the dialplan |
| 15:48.01 | [TK]D-Fender | FuriousGeorge: it gives you a NEW link. The old one won't change.... |
| 15:48.24 | FuriousGeorge | [TK]D-Fender: lol, my bad, just realized i didn't paste link http://pastebin.com/66Ysu3T9 |
| 15:49.03 | [TK]D-Fender | FuriousGeorge: This code still doesn't care about callerid at all.... you should be looking at the call.... |
| 15:49.07 | FuriousGeorge | i guess i expected the blind xfer to use the dialplan that the extension is registered in, defined in sip conf |
| 15:49.46 | FuriousGeorge | yeah, i realize it's caused by the line calling the function. when the user blind xfers ARG1 is still the xferring users extension, not the destination extension |
| 15:49.55 | FuriousGeorge | but I'm not sure how to work around it |
| 15:50.13 | [TK]D-Fender | exten => _3[7-8],1,macro(external_extension,${EXTEN},sip/${EXTEN}) <- arg1 is what you DIALED |
| 15:50.25 | [TK]D-Fender | this is not who you are, it's what you dialed |
| 15:53.36 | FuriousGeorge | [TK]D-Fender: What seems to be happening is that the IVR hits that line you just quoted if I dial 38 (in context "open_inbound") and everything will work fine, unless the user who receives the call blind xfers (let's say to extension 37), in which case ARG1 is still 38.... |
| 15:53.39 | FuriousGeorge | im confused |
| 15:54.00 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 15:54.05 | FuriousGeorge | should not be the case afaict |
| 15:54.11 | [TK]D-Fender | You're not looking at the call.... |
| 15:54.37 | *** join/#asterisk saint-ron (~saint-ron@bb42-60-83-246.singnet.com.sg) |
| 15:54.49 | FuriousGeorge | i was looking at it in the console... is that what you mean by "looking at" |
| 15:54.53 | FuriousGeorge | ? |
| 15:55.21 | [TK]D-Fender | yes. |
| 15:55.30 | FuriousGeorge | one sec |
| 15:56.25 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 16:01.52 | FuriousGeorge | [TK]D-Fender: appreciate your help. http://pastebin.com/BECQJng7 |
| 16:02.04 | FuriousGeorge | xferring user's dialplan is a different context than I thought initally |
| 16:02.30 | FuriousGeorge | but i still don't understand why exten is 38 when it should be 37 |
| 16:04.37 | [TK]D-Fender | 37 is what was DIALED |
| 16:04.58 | [TK]D-Fender | that is the exten |
| 16:05.03 | FuriousGeorge | right, so shiouldn't it be ARG1 too! |
| 16:05.16 | [TK]D-Fender | Macro("SIP/38-00001cd7", "internal_extension,37,sip/37") |
| 16:05.22 | [TK]D-Fender | arg1 8is 37 |
| 16:05.23 | FuriousGeorge | iwhy is it 38? |
| 16:05.32 | FuriousGeorge | oops |
| 16:06.35 | FuriousGeorge | [TK]D-Fender: -- Executing [s-NOANSWER@macro-internal_extension:1] VoiceMail("SIP/wlg-gateway-00001cd5", "38@default,u") in new stack |
| 16:06.43 | boom^time | What is the best free/open source text to speech tool for asterisk available? |
| 16:06.51 | FuriousGeorge | so if arg1 is in fact 37, why is it going to mailbox 38? |
| 16:08.06 | FuriousGeorge | [TK]D-Fender: i fubared the explanation initially, but see what i mean now? it makes no sense |
| 16:08.26 | [TK]D-Fender | -- Executing [s@macro-internal_extension:1] SIPAddHeader("SIP/38-00001cd7", "Alert-Info: ") in new stack |
| 16:08.34 | [TK]D-Fender | this is NOT a blind transfer |
| 16:08.45 | FuriousGeorge | lemme ask user how they doing it |
| 16:08.46 | [TK]D-Fender | -- Executing [s@macro-internal_extension:3] Goto("SIP/wlg-gateway-00001cd5", "s-NOANSWER,1") in new stack |
| 16:08.51 | [TK]D-Fender | Look at your channels |
| 16:08.57 | [TK]D-Fender | Obviously wrong |
| 16:09.58 | FuriousGeorge | it's wlg gateway because it's an inbound call from trunk that hit IVR, but when user x-fers they are using their "Toll" context to make the call (toll incluides local extens) |
| 16:10.12 | FuriousGeorge | i'm not seeing what is wrong there |
| 16:11.01 | navaismo | boom^time, breaking the TOS the google one, then festival, |
| 16:11.48 | [TK]D-Fender | FuriousGeorge: I'm not sure how this is not sinking in. That is NOT a blind transfer, and WLG's channel STILL has its channel variables intact. |
| 16:12.02 | boom^time | navaismo, the google one? And if you were to use this in a commercial environment you mean it wouldn't be a viable option I'm assuming due to breaking TOS |
| 16:12.11 | boom^time | navaismo, what about flite vs festival? |
| 16:12.28 | navaismo | for a commercial one I used cepstral |
| 16:13.29 | boom^time | I'm going to try them but I wanted to see what the free alternatives were like. I attempted to contact them yesterday but just a ring no answer |
| 16:14.43 | boom^time | Does the software reside locally on your machine and do TTS in near real-time? Or do you offload it to a cepstral server preemptively and hope it responds in time? |
| 16:15.13 | FuriousGeorge | [TK]D-Fender: didn't mean to frustrate you, let me call them and verify what they are doing |
| 16:17.19 | navaismo | boom^time, it use app_swift |
| 16:17.43 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-vtkqdzbxvktgmvkz) |
| 16:17.48 | boom^time | I see that but I don't see where you specify the ports, how the licensing works, the app itself could offload the processing to a cepstral server |
| 16:17.50 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-vtkqdzbxvktgmvkz) |
| 16:17.50 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:18.29 | boom^time | these were the questions I had for them yesterday but they didn't answer and no voicemail which I found odd. at around 4:00PM their time. |
| 16:20.30 | navaismo | http://www.digium.com/en/products/software/cepstral-text-to-speech |
| 16:21.17 | *** join/#asterisk MrMuffin (~MrMuffin@97.107.98.164) |
| 16:21.17 | *** join/#asterisk theron (~theron@66.220.145.145) |
| 16:23.16 | MrMuffin | we use voip.ms with asterisk for about 2 weeks now. user noticed that when calling a certain number we got indefinite silence. calling from my cell phone the number is busy. i found some other "always busy" test numbers and can duplicate the issue. my desk phone is silent for as long as i will let it sit. x-lite going directly to voip.ms and bypassing asterisk gives me busy within 10 seconds. |
| 16:23.17 | MrMuffin | i checked my asterisk log and about 10 seconds after the call connects, this appears in the log (but no busy signal) |
| 16:23.18 | MrMuffin | app_dial.c: -- SIP/voip.ms-000047a5 requested media update control 26, passing it to SIP/303-000047a4 |
| 16:23.18 | MrMuffin | app_dial.c: -- SIP/voip.ms-000047a5 answered SIP/303-000047a4 |
| 16:23.54 | MrMuffin | it looks like its trying to give me the busy signal i hear in the softphone but i continue to get silence. voip.ms told me to seek asterisk help since the softphone worked fine |
| 16:24.57 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 16:25.02 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:26.11 | FuriousGeorge | [TK]D-Fender: http://www.surevoip.co.uk/sites/default/files/imagecache/product_full/images/provu/SPA-303G.jpg Users hit the right-most soft button (xFer). To do an attended transfer they simply wait for called part to answer, then hit same button again. HOWEVER, if they don't wait for destination to answer and hit the xFer button again, then instead of going to destination party VM it goes to source party VM |
| 16:27.20 | FuriousGeorge | i know there is another way to do a blind x-fer, and I'll instruct them now on how to do it, and I assume that will work as advertised |
| 16:27.34 | [TK]D-Fender | that is not "blind" |
| 16:27.46 | [TK]D-Fender | that is "not waiting for answer, handing them off anyway" |
| 16:28.09 | [TK]D-Fender | and there is a specific feature on the phone for blind transfer |
| 16:28.26 | [TK]D-Fender | What they were following is misrepresented |
| 16:30.56 | FuriousGeorge | [TK]D-Fender: yeah, this is not how i taught them to blinf xfer and therin ies the confusion |
| 16:38.05 | FuriousGeorge | [TK]D-Fender: you nailed it. That's exactly what it was. Instead of using phone's bXfer soft button they were doing AtXfer and not waiting for destination to pick up |
| 16:39.04 | FuriousGeorge | Now they want to know if there is some way to "not-waiting-for-answer-handing-them-off-anyway" AtXfer, and I gave them a tentative "no" |
| 16:39.41 | FuriousGeorge | (well, obviously they can do it, but they wanna do it without the call then hitting their VM box) |
| 16:39.51 | FuriousGeorge | (on NOANSWER) |
| 16:40.01 | ipengineer | Would someone mind helping me find out where I am going wrong with my findme implementation? Everything works well but if I hang up during the playback of the options and dont allow the findme macro to timeout the call starts ringing again. How can I make the macro hangup if the Dialed party hangs up? Here is my macro and diaplan: https://gist.github.com/zconkle/272b9dc75e5b24fa2946 |
| 16:40.31 | *** join/#asterisk BCS-Satori (~BCS-Sator@2001:470:e169:860:6e3b:e5ff:fe38:da53) |
| 16:40.58 | [TK]D-Fender | exten=>s,n,Set(TIMEOUT(response=5)) <- fail |
| 16:41.17 | [TK]D-Fender | exten=>s,n,WaitExten(5) <- Never EVER do IVR's inside of a MACRO |
| 16:41.36 | [TK]D-Fender | Macro's get merged with the calling context. |
| 16:41.54 | [TK]D-Fender | If you want user input, use READ |
| 16:43.01 | ipengineer | Ok.. Let me try changing that |
| 16:44.26 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 16:45.45 | BCS-Satori | Question: With call queues using static members when a user puts their phone on DND and when Asterisk tries to call that phone Asterisk receives a 486 "Busy Here" and moves on to the next phone as we want. The problem is when no phone is available (lets say they are all inuse) and one is in DND; Asterisk keeps attempting to send the call to the DND phone until timeout for the queue occurs (exten => 1,n,Queue(estimate-queue |
| 16:45.45 | BCS-Satori | ,t,,,30)). How can I make it stop attempting to send to the DND and just leave the queue so it can move on to the next step? |
| 16:48.29 | [TK]D-Fender | use Local channels for your memebers |
| 16:53.39 | ipengineer | [TK]D-Fender: I updated that and it is working as expected except if I hand up without putting an option it it just hangs up on the caller. Can I setup a timeout option and a user disconnected option? https://gist.github.com/zconkle/272b9dc75e5b24fa2946 |
| 16:55.10 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 16:55.16 | [TK]D-Fender | ipengineer: exten=>i,1,Set(MACRO_RESULT=CONTINUE) <- I only see you seeting this var in 1 instance... what do you think the default is? |
| 16:57.19 | ipengineer | I have no clue.. Thats what I had the wait for earlier if it expired then it would goto that mailbox |
| 17:00.10 | [TK]D-Fender | I'd go look at the list of possible values..... |
| 17:00.36 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:00.48 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:02.22 | ipengineer | Alright. I guess what is getting me is I dont see what is being triggered to cause the hangup. I saw where there is a readstatus.. I will have to dig in and play with it a little more |
| 17:10.12 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:14.00 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:15.45 | BCS-Satori | [TK]D-Fender: would you have an example of this with call queues? |
| 17:15.58 | [TK]D-Fender | member=Local/...................... |
| 17:24.05 | BCS-Satori | I understand that I am just don't know how to take member => Local/6133 in queues.conf to call phone SIP/6133 |
| 17:24.33 | [TK]D-Fender | dialplan.... |
| 17:24.47 | [TK]D-Fender | What did you think chan_local implied? |
| 17:27.07 | *** join/#asterisk jasonwert (~w3rt@75-134-81-102.static.aldl.mi.charter.com) |
| 17:32.58 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 17:35.26 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 17:44.23 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 17:51.01 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 17:51.08 | skirmisha | guys, i need some help |
| 17:51.25 | skirmisha | just upgraded tigase server to latest one and asterisk can't get auth |
| 17:51.41 | skirmisha | on the old server i added accounts using pidgin |
| 17:52.06 | skirmisha | on this new version, i am able to create accounts with pidgin and connect, but asterisk is not able to auth |
| 17:52.14 | skirmisha | any ideas where to look? |
| 17:53.34 | skirmisha | do i need to create resource as well on this new version? |
| 17:54.50 | skirmisha | ??? |
| 17:55.07 | smkelly | file: hello |
| 17:55.13 | file | smkelly, hello |
| 17:55.25 | file | smkelly, why is vish still in the DMZ? |
| 17:55.27 | smkelly | You look so different with a @ prefix |
| 17:55.41 | smkelly | He's on the wrong wifi ssid |
| 17:55.53 | file | can he EVEN wifi? |
| 17:58.46 | smkelly | about as good as he can timezone |
| 17:59.41 | hrnt | so.. i am curious what is your opinion regarding SIP and UDP. Do you think that UDP was the best transport protocol for SIP? (or should they have just gone with TCP) |
| 18:00.27 | file | they went with both. |
| 18:01.09 | hrnt | file: true, but they did design their protocol to be "transport-agnostic" by including functionality that usually belongs to the transport layer |
| 18:01.55 | smkelly | It seems best suited for UDP |
| 18:02.03 | hrnt | smkelly: why is that? |
| 18:02.20 | hrnt | (and let me emphasize i am just talking about SIP, not RTP) |
| 18:02.37 | smkelly | oh |
| 18:02.51 | [TK]D-Fender | Especially as RTP does not go over TCP |
| 18:02.59 | [TK]D-Fender | And he asked about SIP.... |
| 18:06.03 | hrnt | RTP and UDP are a great match, but i never understood why SIP was designed to work over UDP |
| 18:06.38 | hrnt | and the more i work with VOIP, the less i understand that original decision |
| 18:07.52 | file | many things are subject to regret, and thus why additional RFCs exist |
| 18:08.12 | hrnt | if i google that stuff, i find some stuff about stateless proxies (but wouldn't those be vulnerable to reflection attacks?) and C10k issues (granted, but are those real problems?) |
| 18:08.34 | hrnt | file: heh, and there definitely are additional RFCs available for SIP :) |
| 18:12.17 | skirmisha | anyone here with tigase experience |
| 18:16.02 | *** join/#asterisk volga629 (~Thunderbi@CPE085b0e07d3f2-CM788df74f6260.cpe.net.cable.rogers.com) |
| 18:16.10 | *** part/#asterisk ageis (kevin@ageispolis.net) |
| 18:18.37 | navaismo | not me |
| 18:24.03 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:30.51 | JJDeveloper | I am trying to do a outbound call using: Action: originate Channel: SIP/5555555555@sip-outbound etc... But I get the error "chan_sip.c: No audio format found to offer. Cancelling call to..." What does this error mean? |
| 18:31.44 | [TK]D-Fender | No codecs selected so you can't talk |
| 18:32.59 | JJDeveloper | How do I specify the codecs? |
| 18:34.29 | *** join/#asterisk Marquis42 (~mrbmw@d149-67-95-208.try.wideopenwest.com) |
| 18:35.07 | Marquis42 | skirmisha: ping |
| 18:35.30 | skirmisha | ok i did an upgrade of tigase from ver 5.0 beta to latest 5.2.1 |
| 18:35.44 | Marquis42 | Was it working before and now doesn't? |
| 18:36.14 | [TK]D-Fender | JJDeveloper: in your peer.... |
| 18:36.16 | [TK]D-Fender | ~book |
| 18:36.16 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:36.16 | skirmisha | the problem i face now is that ast account is not auth with tigase server, i got 401 back from tigase. However when i try to connect with pidgin i am able to connect. Account was created with pidgin... |
| 18:36.18 | [TK]D-Fender | ^^^^ |
| 18:36.32 | skirmisha | i am using plain text passwords |
| 18:36.47 | skirmisha | i think that new tigase expect sasl type of conn |
| 18:37.04 | Marquis42 | Hrm, could be. Can you enable debugging output and pastebin it? |
| 18:37.51 | Marquis42 | By that I mean setting "debug=yes" in jabber.conf. |
| 18:39.20 | skirmisha | yes its done |
| 18:40.17 | skirmisha | i am trying to see what is in the log |
| 18:40.20 | skirmisha | too many data in it |
| 18:40.21 | Marquis42 | Can you get the console output when res_jabber tries to connect and pastebin it somewhere? |
| 18:41.29 | skirmisha | yes, this is the answer....type="error" id="aabhx" xmlns="jabber:client"><error type="auth" code="401"><not-authorized xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/><text xml:lang="en" xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">Authentication failed</text></error></iq> |
| 18:41.49 | skirmisha | let me try to compare login from pidgin and asterisk |
| 18:42.44 | volga629 | Hello Everyone, idle state TLS start generate this error |
| 18:42.46 | volga629 | [2014-05-23 14:29:46] ERROR[25124]: tcptls.c:96 ssl_close: SSL_shutdown() failed: 5 |
| 18:42.48 | volga629 | SSL certificate ok |
| 18:44.27 | skirmisha | there is no ssl installed |
| 18:44.45 | skirmisha | ahh that is another thing |
| 18:46.32 | skirmisha | Marquis42, FINEST: Packet processed by: [jabber:iq:auth] |
| 18:46.47 | skirmisha | WARNING: Can't authenticate with given CallbackHandler java.io.IOException: Password verification problem. |
| 18:47.45 | skirmisha | Caused by: tigase.db.AuthorizationException: Protocol is not supported |
| 18:47.56 | skirmisha | how to enable that protocol |
| 18:48.30 | Marquis42 | Good question. I'd have to look it up in the docs really, so I suggest you have a look there. |
| 18:49.00 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:51.12 | skirmisha | do you know what res_jabber is using |
| 18:51.18 | skirmisha | probably it is a plugin i have to call |
| 19:04.19 | skirmisha | Marquis42, any ideas |
| 19:06.42 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-245-27.dynamic.qsc.de) |
| 19:07.56 | D-Boy | any softphone which work great with asterisk ? |
| 19:08.34 | [TK]D-Fender | All |
| 19:08.43 | [TK]D-Fender | Protocols are protocols. |
| 19:08.53 | [TK]D-Fender | Some have more features than others. Others cost more |
| 19:09.23 | [TK]D-Fender | They work fine. Except all soft-phones suck |
| 19:09.34 | D-Boy | cool, thx, any suggestions for linux distros based on debian/ubuntu ? |
| 19:09.43 | [TK]D-Fender | Having your head chained to a computer and distracting from work-flow blows |
| 19:09.46 | *** join/#asterisk adeeln (~adeeln@fw1.ridgeway.scc-zip.net) |
| 19:09.51 | [TK]D-Fender | Jitsi |
| 19:14.11 | adeeln | it appears that in asterisk 11.9.0, that the following RURI format: sip:user;tag1=foo@domain:5060;tag2=bar SIP/2.0 isn't properly separated...that is, the user field ends up including the user tag (tag1 in my example)....unless that's the expected behavior |
| 19:14.52 | adeeln | i was able to resolve it via dialplan, but shouldn't that be handled by the sip parser? |
| 19:15.36 | Varazir | Can you use a e-mail adress as username ? |
| 19:15.59 | Marquis42 | skirmisha: Sorry, I had to step away for a few. I'd like to see the whole negotiation between Tigase and Asterisk. I'll have a better idea of why it's failing that way. |
| 19:16.19 | WIMPy | Varazir: Whatever you want. |
| 19:18.20 | Varazir | How do I then registrate towards a SIP server. e-mail@foo.sepassword@server.se ? |
| 19:18.48 | WIMPy | You register with the name THEY give you. |
| 19:19.36 | WIMPy | Same as with e-mail. |
| 19:19.50 | WIMPy | Do it yourself and you have the full choice. |
| 19:19.59 | navaismo | D-Boy, Blink |
| 19:20.11 | Varazir | WIMPy: Okay thanks |
| 19:20.40 | navaismo | i need blink for fedora :'( |
| 19:32.29 | D-Boy | [TK]D-Fender & navaismo : thx for your suggestions |
| 19:52.18 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:55.07 | *** part/#asterisk JJDeveloper (~joedevelo@104.0.141.33) |
| 20:10.54 | ipengineer | [TK]D-Fender: I am still having issues with this find me thing.. I tried setting macro_result like you suggested and still must not have done it correctly.. I have actually got it now where if I press 2 to decline the call it connects anyways. My initial issue was with it not going to voicemail on hangup.. https://gist.github.com/zconkle/72f7924a47944e4f39bc |
| 20:11.42 | [TK]D-Fender | exten=>s,n,GoToIf($[ "${action}" = "2" ]?decline,1) |
| 20:11.47 | [TK]D-Fender | Don't go jumping OUT of "s" |
| 20:11.48 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:12.01 | [TK]D-Fender | exten=>i,1,Set(MACRO_RESULT=CONTINUE) <- NOT APPLICABLE |
| 20:12.05 | *** join/#asterisk maani (~babak@2.178.132.188) |
| 20:12.08 | [TK]D-Fender | You are not doing an IVR any more... |
| 20:12.34 | ipengineer | Ok so the i and h extensions dont matter here |
| 20:12.40 | [TK]D-Fender | trash them |
| 20:13.37 | [TK]D-Fender | And you are still only setting 1 value for that variable.... |
| 20:14.27 | ipengineer | What do you mean dont go jumping out of “s”? dont have an accept/decline exten do like: exten=>s,n,GoToIf($[ "${action}" = "1" ]?s,5) |
| 20:14.49 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 20:15.19 | [TK]D-Fender | yes, something like that. |
| 20:15.39 | skirmisha | Marquis42, fixed it |
| 20:17.15 | ipengineer | Same thing.. In the console I see where it does a GoTo (main_mailbox,u,1) then immediately jumping out of macro 'blast' |
| 20:19.51 | [TK]D-Fender | Stop jumping out of the macro.... |
| 20:20.02 | [TK]D-Fender | [16:11][TK]D-FenderDon't go jumping OUT of "s" |
| 20:21.50 | ipengineer | I dont understand? I changed that to: exten=>s,4,GoToIf($[ "${action}" = "2" ]?6) I dont see where I am jumping out of “s”. There is no extension but “s” in this macro. Unless you are talking about my GoTo |
| 20:23.39 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:25.03 | *** join/#asterisk lucido (~chris@e179204095.adsl.alicedsl.de) |
| 20:25.31 | ipengineer | The only place I see it jumping out of the macro is after the Goto.. Which doesnt it have to leave the macro to goto? https://gist.github.com/zconkle/72f7924a47944e4f39bc |
| 20:30.50 | [TK]D-Fender | exten=>s,6,Goto(${ARG1},u,1) <- NO |
| 20:33.09 | ipengineer | [TK]D-Fender: What would be the best way then for me to redirect to that context? I cant hard set that in this macro that context is set in the realtime dialplan and updated frequently. Exit the macro and go back to a specific extension on eric_findme? |
| 20:33.23 | [TK]D-Fender | "core show application dial" |
| 20:33.45 | [TK]D-Fender | You seem to have failed to read the basic instructions for use of macros in Dial() |
| 20:33.52 | [TK]D-Fender | You do NOT jump around in there |
| 20:34.15 | [TK]D-Fender | You ABORT THE CALL if you don't want to bridge and you check for the RESULT afterwards <- |
| 20:48.54 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:55.31 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:55.41 | lucido | Are there extensions implementing these tasks: 1. asterisk answers the phone, determines if it's a fax call, if it is then receives the fax and emails the images, if it isn't then it rings the house phone and connects the call. 2. if noone answers the house call then asterisk takes a voice message and emails it to an address in mp3 format. 3. asterisk receives an email with a pdf document or an image or text in it and a phone number in subjec |
| 20:55.43 | lucido | t or body and then faxes it to the number, replies with the result of the faxing. |
| 20:56.20 | [TK]D-Fender | 3. asterisk receives an email with a pdf document or an image or text in it and a phone number in subjec <- no, Asterisk is not a mail server |
| 20:56.44 | [TK]D-Fender | 1. asterisk answers the phone, determines if it's a fax call, if it is then receives the fax and emails the images, <- yes.... in as much as YOU call the e-mail script to do the actual sending |
| 20:57.41 | [TK]D-Fender | 2. if noone answers the house call then asterisk takes a voice message and emails it to an address in mp3 format <- Yes, it can call an e-mail script and send you the recording, but * cannot record to MP3. You can configure a cleanup script of your own to do this conversion and e-mailing if you want. |
| 20:58.24 | [TK]D-Fender | For #3, once you have a TIF converted for a faxout you can trigger * to place the call and send it |
| 20:58.32 | lucido | [TK]D-Fender, what are the keywords (extension names or other) that I should search for documentation for these tasks? |
| 20:58.46 | [TK]D-Fender | ~book |
| 20:58.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:58.47 | lucido | I heard of fax2email |
| 20:58.53 | ipengineer | [TK]D-Fender: can exten=>s,6,Set(MACRO_RESULT=ABORT) be accessed on the calling context? |
| 20:58.56 | [TK]D-Fender | You need to learn all the basics first for the dialplan |
| 20:59.05 | lucido | thanks |
| 20:59.38 | [TK]D-Fender | ipengineer: Yes, and you can spend your time in your macro manipulating the PARENT CHANNELS' variables, etc |
| 21:01.43 | ipengineer | Ok that makes sense.. ${MACRO_RESULT} returns nothing do you know what the variable name is? I dont see it listed in the macro docs |
| 21:02.11 | [TK]D-Fender | Show us |
| 21:05.19 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:06.13 | ipengineer | [TK]D-Fender: https://gist.github.com/zconkle/7fdd69aed1613a5f6088 |
| 21:06.43 | [TK]D-Fender | ... |
| 21:07.54 | ipengineer | I cant figure out how to access the macro result in eric_find me. NoOp("SIP/provider-000001ee", "Result = ") |
| 21:09.03 | [TK]D-Fender | IT NEVER GOT SET |
| 21:09.44 | ipengineer | exten=>s,n,Set(MACRO_RESULT=ABORT) Didnt set it? |
| 21:09.58 | [TK]D-Fender | Where? |
| 21:10.12 | ipengineer | In the macro.. 3rd line from bottom |
| 21:10.23 | [TK]D-Fender | No, in your CALL |
| 21:10.32 | [TK]D-Fender | You aren't paying attention at all or following instructions. |
| 21:11.21 | ipengineer | I think it is a communication exchange issue. I am nto grasping the concept is the root problem I think. |
| 21:16.31 | *** join/#asterisk volga629 (~Thunderbi@CPE085b0e07d3f2-CM788df74f6260.cpe.net.cable.rogers.com) |
| 21:17.42 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 21:22.38 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 21:30.27 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 21:31.25 | ageis | Are there any configuration GUIs that one can add on top of Asterisk 1.8 without reinstalling? |
| 21:32.51 | WIMPy | Possibly, but certainly not without losing your configuration. |
| 21:34.29 | ageis | Wimpy: what are the options? |
| 21:35.32 | WIMPy | I think the only one that has support worth mentioning (even if not in here) is FreePBX. |
| 21:36.32 | WIMPy | But don't use it if you want to do more interesting things or if you want to learn Asterisk. |
| 21:37.52 | ageis | Wimpy: thought so, thanks. If I have a lot of custom stuff you think FreePBX will end up being limiting? |
| 21:38.10 | WIMPy | Yes |
| 21:39.20 | ageis | K |
| 21:40.09 | [TK]D-Fender | Checkout time, BBIAB |
| 22:08.36 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269581.dsl.bell.ca) |
| 22:13.02 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ntjnuqhnmdmsfkwz) |
| 22:13.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:16.07 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 22:16.23 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 22:21.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:11.26 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-169.ph.ph.cox.net) |
| 23:14.57 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:29.35 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-03-1242451165.dsl.bell.ca) |
| 23:34.54 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 23:52.23 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
