| 00:04.18 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 00:04.46 | mjordan | CHANNEL function |
| 00:05.16 | mjordan | technically, the variables you are referring to aren't recommended even for chan_sip |
| 00:05.29 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL |
| 00:05.40 | mjordan | the formatting on the page is still a bit wonky, but: |
| 00:06.00 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 00:06.32 | mjordan | CHANNEL(rtcp, ... ) should get the various stats |
| 00:11.13 | ChannelZ | I'll have to try again.. I swore I tried a bunch of those from the CHANNEL function but never found anything. |
| 00:16.12 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:54.49 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:58.06 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 00:58.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:01.39 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:35.38 | *** join/#asterisk Graiden (~Matt@65.35.34.94) |
| 01:51.21 | *** join/#asterisk ttyUSB3 (~o@gateway/tor-sasl/omlib) |
| 01:52.59 | *** join/#asterisk vinhdizzo (~vinh@cpe-76-173-171-205.socal.res.rr.com) |
| 01:56.21 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 02:26.59 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 02:27.14 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-67-107-37.lsanca.fios.verizon.net) |
| 02:47.14 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 02:53.28 | *** join/#asterisk ryan_turner (~ryan_turn@eth0.irc-bouncer.ret.memhamwan.net) |
| 02:54.34 | ryan_turner | Can I create a new channel using the asterisk rest api? |
| 02:55.01 | ryan_turner | I've found the ARI hooks for creating a local channel, but instead Im interested in manipulating what's normally in sip.conf |
| 02:55.18 | krapper | Dial wildcards possible for dialing multiple SIP registrations simultaneously... for example... DIAL(SIP/station*) as opposed to DIAL(SIP/station01&SIP/station02&SIP/station03..... |
| 02:55.22 | WIMPy | Now it's beginning to become unclear. |
| 02:55.41 | WIMPy | krapper: No |
| 03:05.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:11.52 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-67-107-37.lsanca.fios.verizon.net) |
| 03:14.43 | *** join/#asterisk jetlag (~jetlag@pool-71-168-194-254.cmdnnj.east.verizon.net) |
| 03:45.54 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hfjfireitogpumhl) |
| 03:45.54 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:55.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hfjfireitogpumhl) |
| 04:13.36 | *** join/#asterisk saint_ (~saint@66.85.176.98) |
| 04:14.24 | saint_ | hi all - in Asterisk 13, how do you reload pjsip ? |
| 04:19.32 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-ihamjoivdlmmoupt) |
| 04:30.19 | *** join/#asterisk marlinc (~marlinc@ip1.weert.li.nl.cvo-technologies.com) |
| 04:33.42 | ChannelZ | pjsip has a ton of components |
| 04:35.58 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 04:36.06 | ChannelZ | You can try module reload res_pjsip |
| 04:50.02 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:13.19 | *** join/#asterisk corretico (~luis@186.96.91.228) |
| 05:38.28 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:52.55 | *** join/#asterisk corretico (~luis@186.96.91.235) |
| 05:54.42 | *** join/#asterisk corretico (~luis@186.96.91.235) |
| 05:58.50 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 06:06.31 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 06:06.31 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 06:10.43 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-stkiurqgtgowgmzy) |
| 06:11.01 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 06:16.33 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 06:24.50 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 06:27.44 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 07:05.24 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:12.35 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-67-107-37.lsanca.fios.verizon.net) |
| 07:12.40 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.205) |
| 07:24.03 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 07:25.29 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 07:25.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:26.13 | D30 | hi all,, when i entered sip show channels i had some entries into... |
| 07:26.36 | D30 | but when i hit core show channels verbose theres nothing to it.. |
| 07:26.45 | D30 | so what does sip show channels actually do? |
| 07:28.04 | ChannelZ | sip show channels shows SIP channels specifically, whereas core show channels will show channels of all types. |
| 07:28.34 | ChannelZ | If you had active SIP channels that sip show channels showed, core show channels should have as well.. |
| 07:28.35 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:30.38 | D30 | ChannelZ: in my case sip show channels had an entries.. while core show channels verbose doesnt |
| 07:32.07 | ChannelZ | Well if it was active in the dialplan, it should have. What version of asterisk? |
| 07:33.20 | D30 | its 1.11 ChannelZ |
| 07:33.28 | D30 | oopss sorry |
| 07:33.35 | D30 | its 11 |
| 07:35.09 | D30 | ChannelZ: entries in sip show channels can be soft hangup? |
| 07:37.31 | ChannelZ | Well it doesn't really show you the channel ID there |
| 07:38.16 | D30 | wait ill pastebin the entries |
| 07:40.08 | D30 | http://codepad.org/I1m08cuQ |
| 07:41.06 | D30 | what does the peer actually means? its weird it has the same ip where it uses different sip extension |
| 07:41.50 | ChannelZ | Well the peer is whatever peer in sip.conf it belongs to |
| 07:42.16 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:42.39 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c4fe:2da6:5602:ef) |
| 07:43.22 | ChannelZ | And what you have there aren't active channels. Like they don't belong to an active call which is why 'core show channels' shows nothing |
| 07:43.59 | ChannelZ | If you make an extension that does MusicOnHold or something, or call another device and answer it, you should see something |
| 07:45.20 | D30 | ahhhh okay ChannelZ make sense |
| 07:50.40 | D30 | thanks |
| 07:51.11 | ChannelZ | no prob |
| 07:57.42 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:58.14 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.205) |
| 07:59.02 | *** join/#asterisk jhlavacek (~jirka@195.70.143.8.adsl.nextra.cz) |
| 08:00.19 | *** join/#asterisk Vutral (K4PV9UUl0P@mirbsd/special/Vutral) |
| 08:03.59 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 08:18.21 | *** join/#asterisk Iamnach0 (~Iamnacho@ip72-213-61-117.om.om.cox.net) |
| 08:39.55 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 08:40.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:41.48 | *** part/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 08:45.06 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:48.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:53.06 | *** join/#asterisk CeBe (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 08:57.17 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 09:05.41 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 09:05.43 | wasanzy | hello |
| 09:05.52 | wasanzy | please am getting this error |
| 09:05.54 | wasanzy | ARNING[28931][C-00145987]: pbx.c:6524 increase_call_count: Maximum loadavg limit of 5.000000 load exceeded by 'SIP/outgoing-001438b9' (currently 5.010000)! |
| 09:06.07 | wasanzy | what am I not doing right? |
| 09:14.23 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:20.20 | *** join/#asterisk xytis (~xytis@77.241.193.82) |
| 09:25.03 | wdoekes | wasanzy: `maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit` |
| 09:25.06 | wdoekes | in asterisk.conf |
| 09:25.11 | *** join/#asterisk xytis (~xytis@77.241.193.82) |
| 09:25.52 | wdoekes | check the output of `w` and/or `top` to see your load and who is causing the load |
| 09:29.50 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 09:31.09 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 09:34.15 | wasanzy | wdoekes: is this configured in asterisk.conf? |
| 09:35.05 | wdoekes | was it really faster to ask than to open up asterisk.conf and check? |
| 09:40.28 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 09:43.20 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 09:45.21 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 09:45.53 | wasanzy | thanks, I have seen it |
| 09:46.04 | wasanzy | what is the advisable settings to use? |
| 09:47.15 | wasanzy | I currently have this: maxcalls = 500 and maxload = 5.0 , I want to change it |
| 09:49.48 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 09:51.59 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 09:59.34 | *** join/#asterisk areski (~areski@80.174.128.55.dyn.user.ono.com) |
| 10:00.59 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 10:02.39 | *** join/#asterisk Ast001 (~uros@82.117.198.142) |
| 10:04.49 | Ast001 | Hello, I have problem with Asterisk 1.8.29 cli command queue show returns just 1 from many queues defined in queue_table (mysql realtime table). Restart of Asterisk or reload module did not help. When I do queue show campaign1 it lists campaign and after that queue show can list it too. Do you know what might be the problem ? Database/table is not. It is the same and now queue show works (after doing queue show campaign for every campaign). |
| 10:05.41 | Ast001 | I never had similar situation in the past. |
| 10:07.25 | Ast001 | Sorry it is 1.8.27 version. Should I recompile it ? Perhaps to newer version from 1.8 branch ? |
| 10:22.48 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:25.41 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:27.05 | *** join/#asterisk iulhk (~iulhk@host-14-net-119-160-119.mobilinkinfinity.net.pk) |
| 10:30.23 | *** join/#asterisk zpotoloom (~quassel@tom.data.ee) |
| 10:30.39 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
| 10:31.32 | zpotoloom | hi, anyone using T.38 gateway ? https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway |
| 10:31.32 | zpotoloom | with 11.4 and res_fax_digium there's no gateway supported |
| 10:31.32 | zpotoloom | Capabilities : SEND RECEIVE T.38 G.711 MULTI-DOC |
| 10:32.18 | zpotoloom | 11.14.0 i meant |
| 10:38.11 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:40.41 | *** join/#asterisk areski (~areski@80.174.128.55.dyn.user.ono.com) |
| 10:44.31 | *** join/#asterisk michael_work (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
| 10:47.29 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 10:48.47 | *** part/#asterisk Ast001 (~uros@82.117.198.142) |
| 10:53.17 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:57.59 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 10:59.10 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 10:59.17 | *** join/#asterisk CeBe (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 11:15.06 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 11:23.49 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 11:28.01 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:35.51 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 11:41.20 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 11:47.08 | pa | are usb gsm dongles and chan_dongle the cheapest way to attach a mobile number to asterisk? |
| 11:51.30 | wdoekes | that depends on your region and your definition of mobile |
| 11:52.04 | pa | well what i want essentially is to route calls to my mobile number (sim) to a voip channel |
| 11:52.35 | pa | because it's much cheaper to be called that way when abroad than directly on the mobile number while abroad |
| 11:52.56 | pa | and it also allows to have only one sim card while abroad instead of 2 |
| 12:08.22 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 12:19.37 | zpotoloom | to answer my previous question then to me seems like that the res_fax_digium module does not support gateway mode |
| 12:20.08 | zpotoloom | with res_fax_spandsp Capabilities : SEND RECEIVE T.38 G.711 GATEWAY |
| 12:23.32 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:24.06 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:25.38 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 12:39.08 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:39.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:40.33 | *** join/#asterisk Pernat (~Pernat@host15-89-206-24.limes.com.pl) |
| 13:00.03 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 13:05.42 | *** join/#asterisk Draecos (~Draecos@106-69-27-44.dyn.iinet.net.au) |
| 13:10.29 | *** join/#asterisk Graiden (~Matt@static-173-65-4-2.tampfl.fios.verizon.net) |
| 13:20.39 | *** join/#asterisk MarkSX (~MarkSX@unaffiliated/marksx) |
| 13:24.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:26.09 | *** join/#asterisk MadHatter42 (~tuwid@109.69.2.253) |
| 13:34.27 | *** join/#asterisk calum_ (~calum_@host86-141-198-121.range86-141.btcentralplus.com) |
| 13:40.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-plcxfkozgsplmvap) |
| 13:40.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:44.23 | *** join/#asterisk mbowie (~mbowie@162-212-36-2.sfo3.rocket-space.com) |
| 13:55.42 | *** join/#asterisk areski (~areski@80.174.128.55.dyn.user.ono.com) |
| 13:58.30 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:00.10 | *** join/#asterisk theron (~theron@199.201.65.135) |
| 14:05.45 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:15.51 | *** join/#asterisk obrut (~obrut@static.88-198-178-60.clients.your-server.de) |
| 14:22.15 | *** join/#asterisk Klemorali (~androirc@mobile-166-173-187-254.mycingular.net) |
| 14:22.19 | *** join/#asterisk cunningpike (~cunningpi@166.170.24.209) |
| 14:23.24 | *** join/#asterisk aruntomar (~aruntomar@123.252.240.106) |
| 14:31.22 | *** join/#asterisk iggi (~iggi@ip69.50-31-21.static.steadfastdns.net) |
| 14:41.44 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:57.50 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kesnfrksyedwunpx) |
| 14:57.50 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:07.40 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 15:12.52 | *** join/#asterisk linetrace (~linetrace@65.19.81.126) |
| 15:27.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 15:33.53 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-vkudkbmbftcllukz) |
| 15:46.22 | *** join/#asterisk u0m3 (~u0m3@92.80.86.86) |
| 15:50.25 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 15:52.41 | *** join/#asterisk CeBe1 (~CeBe@dhcp-215-199.vpn.tu-berlin.de) |
| 15:53.53 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 16:03.06 | *** join/#asterisk corretico (~luis@190.241.180.58) |
| 16:03.40 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 16:24.50 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:26.55 | *** join/#asterisk bulkorok (~Benjamin@i5E879A29.versanet.de) |
| 16:28.03 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:30.07 | *** join/#asterisk bulkorok (~Benjamin@i5E879A29.versanet.de) |
| 16:32.38 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 16:46.47 | *** join/#asterisk cw1972 (~cw1972@213.123.58.248) |
| 16:55.53 | Graiden | There aren't, by chance, any Audiocodes ninjas hanging about here are there? |
| 16:56.29 | WIMPy | ~polls |
| 16:56.29 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 16:58.17 | Graiden | Well, that's a handy piece of information, thank you. I'd prefer to avoid rabid weasels at virtually any cost. To elaborate, however, we have an MP-124d that won't factory reset, and the console simply spits out gibberish. We are entirely unsure how to proceed at this stage, and hoping someone might know of a way to flash one to a factory ROM without a proper interface. |
| 16:59.59 | Chainsaw | Graiden: That's generally going to end with TFTP. |
| 17:00.14 | Chainsaw | Graiden: Try a google with TFTP and your model number and see what comes up. |
| 17:02.22 | Chainsaw | Graiden: It looks like a BootP program is provided that you could try. You haven't upgraded straight from 5.6 to 6.0A have you? |
| 17:02.36 | Graiden | Chainsaw: I appreciate that. I'll start hunting that ground. Quite the investment to lose for something we could potentially resolve with TFTP. |
| 17:03.22 | Graiden | Chainsaw: Unfortunately, management purchased these on an auction site pre-owned, so God only knows what sort of ridiculousness has been done to them at one point or another. |
| 17:04.11 | Chainsaw | Graiden: Very nice. |
| 17:04.31 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 17:04.51 | WIMPy | Sound fun |
| 17:05.02 | Graiden | Chainsaw: Alas, not all decisions come from the phone nerds that make things work after the fact. |
| 17:05.51 | Chainsaw | Graiden: I would start here: http://www.audiocodes.com/downloads |
| 17:06.19 | Chainsaw | Graiden: And see if you can get yourself registered with the unit details. They make explicit reference to a BootP program they offer that has ".cmp" firmware files. |
| 17:06.51 | Chainsaw | Graiden: I would expect some documentation with it that explains how to do a deeper reset and push some firmware in over TFTP. A low-level rescue, if you like. |
| 17:08.18 | Graiden | Chainsaw: Brilliant. They offer no software, but now that I'm looking in the 'documentation' section, I see a boatload of guides. With any luck, one of them will cover my issue. Once I've uncovered a solution, does the channel have anywhere they keep documentation on these sorts of things? Just in case someone else were to run across it in the future, for example. |
| 17:09.07 | Chainsaw | Graiden: I would say blog about it. |
| 17:09.12 | Chainsaw | Graiden: Google will do the rest. |
| 17:09.34 | Chainsaw | WIMPy: Is this appropriate for the Asterisk wiki at all? |
| 17:10.08 | WIMPy | Not really. But it would surely fit on voip-info. |
| 17:10.10 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 17:10.37 | Chainsaw | WIMPy: True. That site is so full of outdated stuff though, I wouldn't want it sniffed at due to its location. |
| 17:10.38 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 17:10.56 | WIMPy | It's not useless, though. |
| 17:11.00 | Chainsaw | Graiden: Alternatively, there is a serial port on the device. Have you tried whether that gives more intelligible information as you connect the power? |
| 17:11.41 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 17:12.13 | Graiden | Chainsaw: Indeed. We are connected via serial with the parameters in their general user guide, and the output is unreadable at best. Squares and random characters. It's almost as though the ROM is just purely wrong and the hardware has no idea what to make of it. |
| 17:12.40 | Chainsaw | Graiden: Try other baud rates. 9600, 19200, 115200 are all common. |
| 17:13.20 | Chainsaw | Graiden: Boot loaders sometimes use different parameters than the main code, and from the sound of it the main code is definitely unusable. |
| 17:13.51 | WIMPy | Inded. Garbage is very likely due to wrong speed. |
| 17:20.57 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 17:25.09 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 17:26.58 | *** join/#asterisk riess82 (~riessma@80-121-0-170.adsl.highway.telekom.at) |
| 17:30.38 | *** join/#asterisk zekoZeko (~b@77.234.148.119) |
| 17:32.52 | zekoZeko | hello everyone. I'm having trouble dialing out my sip trunk (managed to get incoming calls working). I must be doing something dumb, but I'm really new at this. |
| 17:33.41 | WIMPy | Are you sure you send the number in the format they expect? |
| 17:33.54 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:33.58 | WIMPy | Turn on sip debug, maybe they will give you an sensible message. |
| 17:34.15 | [TK]D-Fender | You need to elaborate more on what you mean by "having trouble dialing". |
| 17:34.32 | [TK]D-Fender | What part of this process is actually failing? |
| 17:34.45 | zekoZeko | i created an outbound route, but it doesn't seems to get used. any number I dial (except local extension) returns "this call can't be completed as dialed" |
| 17:34.51 | [TK]D-Fender | ~freepbx |
| 17:34.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:34.53 | [TK]D-Fender | ^^^ |
| 17:34.58 | WIMPy | Well, as dialing out via sip isn't possible... |
| 17:35.01 | [TK]D-Fender | You need to use their support channel for call flow issues |
| 17:35.11 | zekoZeko | it doesn't come to the part where my Asterisk contacts the provider... no outgoing SIP |
| 17:35.24 | [TK]D-Fender | zekoZeko: #freepbx <---- |
| 17:35.50 | zekoZeko | ok thanks, will go there. |
| 17:37.59 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 17:41.43 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 18:00.24 | *** join/#asterisk danjenkins__ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:03.41 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 18:09.45 | *** join/#asterisk ttyUSB3 (~o@gateway/tor-sasl/omlib) |
| 18:15.17 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:17.46 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:27.29 | Graiden | Hey WIMPy: As an update, the BootP solution seems to have worked. I'm factory resetting the device now, I believe we successfully unbricked an MP-124 today. |
| 18:28.05 | WIMPy | Nice |
| 18:28.42 | Graiden | Yeah, I'm quite happy to see it booting. I'll write up an article on it tonight. No sense in letting the knowledge die. |
| 18:29.31 | WIMPy | reminds me that I have an Parley gateway to unbrick. |
| 18:33.27 | *** join/#asterisk linuxluke (~linuxluke@66-208-244-53.ubr01a.maysld01.nj.hfc.comcastbusiness.net) |
| 18:34.19 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 18:35.42 | ChannelZ | mjordan: thanks for the hangup handler/pre-dial help yesterday, got it implemented and working a few mins ago. |
| 18:38.42 | mjordan | np |
| 18:39.40 | ChannelZ | Comcast appears to have fixed my random packet loss, so that's lovely. |
| 18:41.41 | *** join/#asterisk riess82 (~riessma@80-121-0-170.adsl.highway.telekom.at) |
| 18:45.18 | *** join/#asterisk ttyUSB3 (~o@gateway/tor-sasl/omlib) |
| 18:47.30 | Graiden | WIMPy: Have you ever messed with Vega's? Those things nearly made me throw my laptop on numerous occasions. |
| 18:47.51 | WIMPy | Graiden: nope |
| 18:48.21 | Graiden | WIMPy: Do yourself a favor, if ever offered a gig working on them, just run. |
| 18:48.37 | WIMPy | ok |
| 19:01.43 | *** join/#asterisk blistov (~bens@S0106e03f490494e9.vs.shawcable.net) |
| 19:03.33 | blistov | hey, if I have 4 extensions (200-204) in a queue (2000), and call routing is setup so that if 200 doesn't answer, the call goes to queue 2000, asterisk tries to ring everyone in the queue, including 200, but because 200 is unavailable, the new request goes into the queue as well, and creates an infinite loop. |
| 19:04.25 | WIMPy | You don't put extensions in to a queue, you put devices there. |
| 19:04.37 | WIMPy | Usually, that is. |
| 19:05.19 | [TK]D-Fender | Or check if you're already in a queue before following that extra step afterwards |
| 19:05.26 | blistov | So I remove 200 from the queue, and all is well. but 200 is reception, so if they cant' answer the call, we want the call to go into the queue, but if 200 is then ready to take the call, the call is no longer rinign them. |
| 19:05.48 | [TK]D-Fender | Or create another section of dialplan code for actually dialing these members and don't use the one that has that redirect to Queue |
| 19:06.08 | [TK]D-Fender | It's your dialplan, do whatever you want with it |
| 19:06.31 | blistov | I'm not following. |
| 19:06.34 | WIMPy | Is there a reason you used extensions instead of devices? |
| 19:06.40 | blistov | What's the difference between an ext and a device? |
| 19:07.01 | WIMPy | An extension is a piece of dialplan. |
| 19:12.55 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 19:14.42 | blistov | okay, sorry, 200-204 are devices. |
| 19:14.52 | blistov | 200 is reception phone. |
| 19:15.11 | WIMPy | Then how does the call get in to the queue again? |
| 19:15.18 | blistov | all inbound routes land on 200. if 200 doesn't answer after N seconds, the call goes into the queue 2000 |
| 19:15.34 | blistov | queue 2000 then employs a ringall strategy for 200-204 |
| 19:16.33 | blistov | but that of course, rings 200 as well. There's the options for "ringinuse" but setting that to no, doesn't work if 200 is simply not connected to the PBX. |
| 19:17.04 | blistov | so when a call hits the queue, it rings everyone in the queue, but 200's ring gets pushed back into the queue as if it were a uniq incoming call. |
| 19:17.21 | blistov | So everyone in the queue sees a hundred incoming calls in about 4 seconds. |
| 19:17.25 | WIMPy | Only if you made it do so. |
| 19:17.26 | blistov | and they can't answer any of them. |
| 19:17.40 | blistov | Sure, and clearly I DID make it do so, which is why I'm asking now. |
| 19:17.47 | blistov | This is with FreePBX. |
| 19:17.54 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 19:17.59 | [TK]D-Fender | ~freepbx |
| 19:17.59 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:18.03 | WIMPy | You need to find out what an extension is and what a device. Even if you called them the same names, they are not the same. |
| 19:18.05 | [TK]D-Fender | blistov: Not supported here |
| 19:18.20 | blistov | deal. |
| 19:18.30 | [TK]D-Fender | blistov: If you want assistance in setting it up to try to work the way you like then you'll have to take it up in their channel |
| 19:20.47 | blistov | Doing so. |
| 19:20.56 | blistov | thanks. |
| 19:21.00 | *** part/#asterisk blistov (~bens@S0106e03f490494e9.vs.shawcable.net) |
| 19:21.38 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 19:22.09 | Milenco | Hey Guys! Perhaps you can help me with a little issue I'm having. :) |
| 19:22.46 | *** join/#asterisk ayrjola (~ayrjola@80.248.109.192) |
| 19:23.11 | Milenco | I've got Asterisk setup on a PRI card and can receive/place calls just fine. Caller-id also works for incoming calls, but I can't seem to modify the caller-id when I'm calling out over the ISDN card. |
| 19:23.54 | WIMPy | US or ROW? |
| 19:23.59 | Milenco | It always shows the default number while I have a whole range of numbers in use, both numbers that route back throught the same ISDN line and VoIP numbers I want to send |
| 19:24.15 | Milenco | hey WIMPy :) I'm using Dahdi and using default config |
| 19:24.18 | WIMPy | Did you set the TON correctly? What dahdi calls the plan. |
| 19:25.04 | WIMPy | If you want to send numbers that aren't routed to that line, you need special permission. |
| 19:25.41 | Milenco | default is pridialplan=unknown and prilocaldialplan=national (I believe), but I've set pridialplan to dynamic |
| 19:25.49 | *** join/#asterisk MadHatter42 (~tuwid@109.236.46.33) |
| 19:26.09 | Milenco | The caller-id is set correctly, when I place the same call over voip the correct caller-id shows |
| 19:26.30 | WIMPy | IIRC prilocaldialplan is the one for sending caller IDs. |
| 19:26.33 | Milenco | I would figure that much WIMPy, but even my 'assigned' numbers wont show, just the default XXXX0-number |
| 19:26.42 | WIMPy | That means absolutely nothing. |
| 19:27.47 | WIMPy | So make sure you send the numbers in the format that fits the configured TON (plan). |
| 19:28.42 | Milenco | Right..so if I set it to international and specify callerid(num) as 0031534806860 that should work |
| 19:28.44 | WIMPy | And you haven't told us which country. |
| 19:28.47 | Milenco | where 0031 is the nation code |
| 19:28.51 | Milenco | its in germany |
| 19:28.56 | WIMPy | No |
| 19:29.32 | WIMPy | The leading 0s are not part of the number. |
| 19:29.47 | WIMPy | Sending formatted numbers only works with unknown. |
| 19:30.24 | Milenco | in that case the pridialplan matched my send callerid's by default, i would assume? |
| 19:30.40 | Milenco | since i send them in the 0031 or 0049 format with pridialplan=unknown |
| 19:30.52 | WIMPy | ??? |
| 19:30.56 | WIMPy | Yes. |
| 19:32.12 | Milenco | alright, got it working now :) |
| 19:32.37 | Milenco | Or so it seems |
| 19:32.50 | Milenco | I've set both values to unknown |
| 19:33.16 | *** join/#asterisk pigpen (~pigpen@216-177-181-17.block0.gvtc.com) |
| 19:33.18 | WIMPy | That's usually the safe choice. |
| 19:34.11 | pigpen | So, I have a few polycom SP 550âs that when I lift up the receiver, the speaker phone engages. The other 90% of the phones have no issues. Ideas? |
| 19:34.54 | WIMPy | Interesting feature. |
| 19:35.25 | Milenco | indeed :) |
| 19:35.37 | Milenco | some SP 550's behave like that pigpen? or all? |
| 19:35.50 | pigpen | about 5 of 60 |
| 19:36.10 | pigpen | All same firmware, script generated config |
| 19:36.17 | Milenco | then there are becoming unstable because of possible hardware issues |
| 19:36.28 | Milenco | no differences in configuration or network i assume? |
| 19:36.45 | pigpen | yeah, all nice 1 GB POE. |
| 19:36.57 | pigpen | Good point though, some are 3COM, some are Extreme. |
| 19:37.01 | pigpen | switches that is. |
| 19:37.05 | Milenco | possible broken capacitors |
| 19:37.18 | pigpen | hmm, new phones, but possible. |
| 19:37.49 | Milenco | if the situation is exactly the same |
| 19:37.51 | pigpen | The users have noted, that if thy unplug the handset from the back of the phone, and plug it back in, it does ok for awhile. Certainly seems like hardware. |
| 19:38.07 | Milenco | like firmware/configuration/environment, there must be something wrong with the phones |
| 19:39.00 | pigpen | k, that was my take too. Just wanted to put it out there for second opinion. |
| 19:39.21 | Milenco | the switches causing it seem unlikely |
| 19:39.50 | Milenco | but its worth to try to create as less as difference in the situation as possible |
| 19:39.54 | Milenco | between a working/non-working phone |
| 19:44.10 | *** join/#asterisk cunningpike (~cunningpi@166.170.24.209) |
| 19:47.23 | pjensen00 | mjordan: good job on the webinar! |
| 19:48.45 | *** join/#asterisk newtonr (~newtonr@173-21-90-203.client.mchsi.com) |
| 19:48.45 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:49.42 | *** join/#asterisk PhirePhly (~PhirePhly@99-46-142-3.lightspeed.sntcca.sbcglobal.net) |
| 19:52.35 | *** join/#asterisk riess82 (~riessma@80-121-0-170.adsl.highway.telekom.at) |
| 19:58.58 | Milenco | WIMPy, could you perhaps point me in the right direction with a follow-up question? |
| 20:00.11 | Milenco | It appears Asterisk sets ${CALLERID(num)} only correct (like the cid_number i've setup in my sip.conf) the first time I call |
| 20:00.33 | Milenco | the next times it sets it as the SIP-account number, which is 3 digits |
| 20:01.40 | WIMPy | What first time? After what? |
| 20:02.09 | WIMPy | Turn up verbose and look at your calls. |
| 20:03.59 | Katty | pokes eppigy |
| 20:04.05 | Katty | pokes [TK]D-Fender |
| 20:04.43 | Milenco | Strange, restarted Asterisk now everything is running smooth |
| 20:05.44 | Milenco | It sets the callerid properly the first time i call out (after a sip reload), after that it sets the callerid-num as the 3-digit internal number. But it seemed fixed now by restarting |
| 20:05.54 | Milenco | not modifying the callerid-num variable in any way |
| 20:06.12 | *** join/#asterisk cunningpike (~cunningpi@166.170.24.209) |
| 20:07.12 | Milenco | Or not fixed.. very strange |
| 20:14.15 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 20:35.27 | Milenco | Now it's working again for quite some time, while I didnt make any changes |
| 20:35.46 | Milenco | I'll keep monitoring it to see if I can spot what happens when it goes wrong |
| 20:36.08 | WIMPy | Doesn;t sound very trustworthy. |
| 20:36.23 | Milenco | Indeed, but I can't figure out now where it goes wrong |
| 20:36.43 | Milenco | And I don't want to annoy others with broad-ranged problems |
| 20:36.47 | WIMPy | Make sure you have enought details in your logs to find out. |
| 20:37.27 | Milenco | I have maximum verbose on in console and print the callerid when executing my dial macro |
| 20:38.26 | Milenco | at some point it switches from the cid_number to the sip-account name |
| 20:43.54 | Milenco | Thanks again for your help WIMPy, I'll keep monitoring the callerid-possible-issue i'm having and let you know if i figure it out |
| 20:47.15 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 20:47.15 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 20:58.06 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 21:06.59 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 21:13.49 | *** join/#asterisk saint_ (~saint@198.15.70.2) |
| 21:14.37 | saint_ | hi all - so i am trying to install and configure Asterisk 13. pjsip is a piece of work.... what could be the reason for not having ring back tone when someone calls in from a SIP trunk ? |
| 21:14.56 | saint_ | Voice is full duplex. It's just that the caller does not have ringback tone.. |
| 21:19.59 | saint_ | mmhh.. Also when I hang up from the sip endpoint , it does not hangup on the caller's side .. |
| 21:22.37 | *** join/#asterisk antiochIst (~taylorhaw@168.244.48.230) |
| 21:25.55 | file | without a log, console output, SIP traffic - those are things that can not be answered |
| 21:28.05 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 21:28.31 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-105-142.dynamic.qsc.de) |
| 21:33.49 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:46.50 | Milenco | saint_, are you using Answer()? |
| 21:50.41 | Milenco | Answer() causing the ringing tone to stop, and Hangup() should be done after the Dial command |
| 21:51.18 | Milenco | see this snippet i am using: http://pastebin.com/dkgYLrHA |
| 21:54.46 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:00.59 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 22:01.14 | saint_ | Milenco yes I do.. Answer() , Dial(), Hangup() |
| 22:01.35 | saint_ | my asterisk 1.8 has the same dialplan settings, and works |
| 22:01.49 | Milenco | Ah..sorry then I'm still on 1.8 myself |
| 22:05.31 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:05.54 | saint_ | i ll be back laterz |
| 22:05.58 | Milenco | gl :) |
| 22:10.57 | *** join/#asterisk Weezey (~ohno@206.210.111.135) |
| 22:20.15 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 22:27.13 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 22:30.29 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
| 22:42.13 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 22:55.58 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 22:57.06 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 22:58.17 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 23:31.37 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 23:34.31 | *** join/#asterisk antiochIst (~taylorhaw@168.244.48.230) |
| 23:34.40 | antiochIst | should asterisk 13 be responding to SIP invite with codecs not offered in the INVITE SDP? |
| 23:35.41 | antiochIst | version 12.7 does not do this |
| 23:52.22 | newtonr | antiochIst, huh... I don't think so.. |
| 23:52.41 | newtonr | antiochIst, can you file an issue on issues.asterisk.org/jira and include your SIP trace and Asterisk log? |
| 23:52.48 | newtonr | I've got to run |
| 23:52.55 | antiochIst | yea I will |
| 23:53.39 | newtonr | antiochIst, thanks a bunch. |