| 00:00.06 | darkdrgn2k | :-P |
| 00:00.21 | darkdrgn2k | so 70 is saying "i dont speak 38" right? |
| 00:01.30 | igcewieling | darkdrgn2k it appears so, yes. What is .70 your SBCor asterisk or what? |
| 00:01.43 | darkdrgn2k | ip pbx |
| 00:01.55 | darkdrgn2k | with t.38 in every freaken codec list i coudl find! |
| 00:01.59 | igcewieling | could you be more vague? |
| 00:02.17 | darkdrgn2k | there is a reason to my vaugness |
| 00:02.17 | igcewieling | I assume it is not an Asterisk "IP PBX"? |
| 00:02.26 | darkdrgn2k | looks at the name of the channel :-P |
| 00:02.27 | Maliuta | igcewieling: Those Buddhists are and odd lot, for a people with a peaceful manifesto they put up a heap of resistance everytime they meditate |
| 00:02.34 | darkdrgn2k | ShoreTel |
| 00:02.42 | igcewieling | darkdrgn2k: In ASTERISK t.38 support is not configured as part of codecs, it is a seperate config |
| 00:02.49 | darkdrgn2k | no asterisk |
| 00:03.29 | igcewieling | perhaps in shoretel T.38 is configured somewhere other than the codec list as well? |
| 00:03.46 | darkdrgn2k | its very confusing |
| 00:03.53 | igcewieling | look for UDPTL settings |
| 00:03.55 | darkdrgn2k | they have a "intrasite" "intersite" and a "fax" codec list |
| 00:04.03 | darkdrgn2k | all 3 have t.38 in the mix |
| 00:05.04 | igcewieling | doing a asking the google.brain about shoretel udptl returns lots of hits |
| 00:05.12 | igcewieling | s/doing a// |
| 00:05.53 | darkdrgn2k | hmm sbc does not show any evidence of t.38 |
| 00:06.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:06.56 | Maliuta | People on facebook aren't giving me the instant gratification I need. So I'm gonna do this here, it's igcewieling's fault anyhow ... |
| 00:07.22 | Maliuta | How many meditating Buddhists does it take to terminate a BNC network? |
| 00:07.30 | darkdrgn2k | 1? |
| 00:07.45 | igcewieling | Maliuta: pagans and buddhists are the only two religions I do not have a strong dislike for. |
| 00:07.50 | Maliuta | 100, 50ohms at each end :P |
| 00:08.02 | darkdrgn2k | lol |
| 00:08.48 | Maliuta | igcewieling: and they make for such good resistor jokes! ;) |
| 00:09.11 | darkdrgn2k | Did you gfuys know computers run on smoke? |
| 00:09.35 | darkdrgn2k | gatat keep it all in, if it escapes comptuer stops workin! |
| 00:13.04 | Maliuta | darkdrgn2k: everyone knows it's steam. |
| 00:13.21 | darkdrgn2k | $@#$@$ SBC doesnt even mention t38 anywhre |
| 00:14.33 | *** join/#asterisk cali (~cali@unaffiliated/cali) |
| 00:15.09 | igcewieling | "T.38 has been added to the default Codec Lists (Fax Codecs – High Bandwidth and Fax Codecs – Low Bandwidth) after upgrading to ShoreTel 10. To remove the T.38 Fax codec support, please create a custom Codec List without T.38. Existing custom Codec Lists will not be affected." |
| 00:18.48 | darkdrgn2k | yep im on 13 |
| 00:19.08 | darkdrgn2k | liek i said its There but its being rejected |
| 00:21.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cxltynbyhbyrmiii) |
| 00:30.11 | igcewieling | if you want to learn real patience try training a cat to sit, stay, etc. |
| 00:30.27 | igcewieling | so far mine sits when I say stay and stays when I say come here. |
| 00:30.42 | newtonr | igcewieling: that skill should be a requirement for call center support techs |
| 00:30.52 | igcewieling | newtonr: 8-) |
| 00:31.41 | igcewieling | mine is almost at the point now where he will sit and stay while put food in his supper dish. He is a little unclear on "eat" though. |
| 00:32.32 | igcewieling | he does know "move!" pretty well. |
| 00:38.15 | phix | Cats are dicks |
| 00:39.23 | igcewieling | Cats are like Linux, they are not user unfriendly, they are just picky about who their friends are. |
| 00:40.10 | phix | what are you on about? Linux is very user friendly |
| 00:40.31 | phix | You just need to read a few books first :P |
| 00:40.41 | [TK]D-Fender | phix: No, cats are PUSSIES. |
| 00:40.42 | phix | most users are too lazy to do that |
| 00:40.50 | phix | hai [TK]D-Fender |
| 00:43.51 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 00:54.45 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.155) |
| 00:56.43 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 01:32.08 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 02:06.33 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 02:06.34 | saint_ | hi all |
| 02:10.39 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:27.33 | phix | hai hai |
| 02:27.51 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 02:27.58 | phix | I am looking for a simple application to connect to an asterisk server and popup with the caller id when someone calls |
| 02:29.46 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 02:29.58 | phix | any ideas or should I write my own? |
| 02:30.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:30.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:30.37 | phix | if I need to write one, is there a simple api for java or python to connect to an asterisk server via SIP or something that doesn't require dependencies on RTP stuff? |
| 02:35.04 | *** join/#asterisk nicknam1232 (d9494764@gateway/web/freenode/ip.217.73.71.100) |
| 02:42.21 | fling | what does this mean? > [Feb 8 09:41:34] WARNING[17488]: chan_sip.c:3895 __sip_xmit: sip_xmit of 0x7fd250033ee0 (len 527) to 188.187.245.6:5060 returned -1: Operation not permitted |
| 02:54.20 | saint_ | so i'm trying to change a file name using the command system: n(saveit),System(mv -f ${CHANNEL(uniqueid)}.wav ${promptnumber}.wav) |
| 02:54.38 | saint_ | I can see in the logs: System("SIP/100-00000027", "mv -f 1360291977.46.wav 100.wav") in new stack |
| 02:55.01 | saint_ | but when i list the folder, the file 1360291977.46.wav still exists ... |
| 02:55.20 | saint_ | anyway to trace that or understand what's going on ? |
| 03:04.04 | igcewieling | try specifying the full path |
| 03:14.26 | lorsungcu | wtf |
| 03:14.26 | lorsungcu | making my sounds go nuts |
| 03:17.14 | *** join/#asterisk nicknam1232 (d9494764@gateway/web/freenode/ip.217.73.71.100) |
| 03:34.56 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-45-142.lsanca.fios.verizon.net) |
| 03:46.44 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 03:46.56 | studybot_ | hi everyone |
| 03:47.11 | studybot_ | I'm building asterisk 11.2 on macosx |
| 03:47.29 | studybot_ | the configure failed with message uuid support not found |
| 03:47.34 | studybot_ | how can I fix this issue? |
| 03:54.18 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:01.27 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 04:01.29 | kuku | What would cause an AGI to run 10 seconds, but if I ran it from the command line with the same parameter it takes a second ? |
| 04:03.43 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 04:12.09 | lorsungcu | kuku: guess id need to see the parameter :p |
| 04:12.23 | igcewieling | kuku: then it isn't an AGI unless you piped stuff to it to give it the stdin agis are required to read |
| 04:15.40 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:23.25 | kuku | [2013-02-07 18:34:20] VERBOSE[1363] pbx.c: -- Executing [s@macro-dialout-trunk:38] AGI("SIP/97-00007806", "phones_counter.pl,55555555") in new stack |
| 04:25.04 | kuku | Part of the day it will work fine, another part it slows down |
| 04:25.21 | kuku | it calls a db, show processlist has nothing there |
| 04:25.41 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 04:25.41 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 04:25.41 | *** join/#asterisk Who-m3 (john@my.virt-host.com) |
| 04:25.42 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 04:25.42 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 04:25.42 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 04:25.42 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 04:25.42 | *** join/#asterisk dym (~patrick@intrepid.openroot.de) |
| 04:25.42 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 04:25.47 | kuku | mysql slow logs have nothing ( and if I run the query then on the db, it works fine as well ) |
| 04:25.50 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-gayvxxkashhgzrkt) |
| 04:25.50 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 04:25.50 | *** join/#asterisk zamba (marius@flage.org) |
| 04:25.57 | kuku | iostat looks good as well - I can't find the issue |
| 04:26.01 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 04:26.01 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 04:26.01 | *** join/#asterisk alsuren (alsuren@60e-007.static.bnc4free.com) |
| 04:26.01 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 04:26.01 | *** join/#asterisk zpotoloom (~toomas@2001:ad0:1:1:3ed9:2bff:fe5c:e027) |
| 04:26.01 | *** join/#asterisk hwt (hwt@curb.thorkildssen.com) |
| 04:26.01 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 04:26.01 | *** join/#asterisk Sprocks (~Sprocks@bas1-brampton37-845523112.dsl.bell.ca) |
| 04:26.01 | *** join/#asterisk derpzilla (~gid@you.can.neverw.in) |
| 04:26.01 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 04:26.01 | *** join/#asterisk JonMR (JonMR@2600:3c03::f03c:91ff:feae:9b1d) |
| 04:26.01 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-pydzvamgrckyfftw) |
| 04:26.02 | *** join/#asterisk razu_ (~razu@2001:ad0:1:1:202:b3ff:fe36:921c) |
| 04:26.15 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 04:26.15 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 04:26.15 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 04:26.15 | *** join/#asterisk prgmrchris (~chris@c-83-233-34-145.cust.bredband2.com) |
| 04:26.15 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 04:26.15 | *** join/#asterisk paweu (~root@li502-253.members.linode.com) |
| 04:26.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 04:26.28 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 04:26.28 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 04:26.34 | *** join/#asterisk ponyofdeath (~vladi@cpe-75-80-173-129.san.res.rr.com) |
| 04:26.34 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 04:26.34 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 04:26.34 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 04:26.34 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 04:26.34 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 04:26.35 | *** join/#asterisk hbee (~hbee@87-100-155-185.bb.dnainternet.fi) |
| 04:26.35 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 04:26.35 | *** join/#asterisk freeedrich| (friedrich@176.9.118.18) |
| 04:26.35 | *** join/#asterisk DarwinSurvivor (~DarwinSur@S0106687f74a1273d.vs.shawcable.net) |
| 04:26.35 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 04:26.35 | *** join/#asterisk carrar (tim@osburn.com) |
| 04:26.35 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 04:26.35 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 04:26.35 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 04:26.35 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:26.48 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 04:26.48 | *** join/#asterisk curfont (~q@ytmd.ath.cx) |
| 04:26.48 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 04:26.48 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 04:26.48 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 04:26.48 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 04:26.48 | *** mode/#asterisk [+o newtonr] by moorcock.freenode.net |
| 04:27.09 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 04:27.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:27.09 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:44c9:c90b:a530:9b49) |
| 04:27.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 04:27.09 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 04:27.10 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 04:27.10 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 04:27.10 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 04:27.10 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 04:27.10 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 04:27.10 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 04:27.10 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 04:27.10 | *** mode/#asterisk [+o leifmadsen] by moorcock.freenode.net |
| 04:27.15 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 04:27.15 | *** join/#asterisk resist0r (~resist0r@c-174-57-0-74.hsd1.nj.comcast.net) |
| 04:27.15 | *** join/#asterisk doctorray (~ray@72.26.99.19) |
| 04:27.15 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 04:27.15 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 04:27.15 | *** join/#asterisk dkostousov (~dkostouso@78.159.236.154) |
| 04:27.15 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.31.8) |
| 04:27.15 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 04:27.15 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 04:27.26 | *** join/#asterisk mattsl (~user@c-76-123-231-159.hsd1.tn.comcast.net) |
| 04:27.26 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 04:27.26 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 04:27.26 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 04:27.26 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 04:27.26 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 04:27.26 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 04:27.26 | *** join/#asterisk Dibbler (~Dibbler@host86-156-142-101.range86-156.btcentralplus.com) |
| 04:27.38 | *** join/#asterisk nc11235 (~textual@pool-71-97-75-217.dllstx.fios.verizon.net) |
| 04:27.38 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 04:27.38 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 04:27.38 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 04:27.38 | *** join/#asterisk Tim_Toady (~fuzzy@194.50.55.245) |
| 04:27.38 | *** join/#asterisk indra (~weechat@2a02:2308::216:3eff:feeb:9481) |
| 04:27.38 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 04:27.38 | *** join/#asterisk rogers (rogers@bling.bling.org) |
| 04:27.38 | *** join/#asterisk tzanger (tzanger@wallace.mixdown.ca) |
| 04:27.38 | *** mode/#asterisk [+o file] by moorcock.freenode.net |
| 04:27.56 | *** join/#asterisk cali (~cali@unaffiliated/cali) |
| 04:27.56 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 04:27.56 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 04:27.56 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 04:27.56 | *** join/#asterisk dijib (~dijib@67-204-199-13.eastlink.ca) |
| 04:27.57 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 04:27.57 | *** join/#asterisk Kyosh (~whoa@pool-72-89-93-13.nycmny.fios.verizon.net) |
| 04:27.57 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 04:27.57 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 04:27.57 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-ejxbpipookvtgbun) |
| 04:27.57 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 04:27.57 | *** join/#asterisk stian (~Stian@svr3.st0len.co.za) |
| 04:27.57 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 04:27.57 | *** mode/#asterisk [+o Qwell] by moorcock.freenode.net |
| 04:27.59 | *** join/#asterisk dimbulb (~totallyre@indra.vis.nu) |
| 04:27.59 | *** join/#asterisk bytemaster (~erw@host81-150-217-168.in-addr.btopenworld.com) |
| 04:27.59 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 04:27.59 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 04:28.00 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 04:28.00 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 04:28.00 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 04:28.00 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 04:28.00 | *** join/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |
| 04:28.00 | *** join/#asterisk reisi (koivunej@mustatilhi.cs.tut.fi) |
| 04:28.00 | *** join/#asterisk Sicelo (Sicelo@unaffiliated/sicelo) |
| 04:28.10 | *** join/#asterisk scubes13 (scubes13@3ee-007.static.bnc4free.com) |
| 04:28.29 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-xiacyymlenfhpxhk) |
| 04:29.16 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 04:31.55 | fling | [Feb 8 11:31:10] WARNING[18888]: chan_sip.c:3895 __sip_xmit: sip_xmit of 0x7f5240063b20 (len 527) to 188.187.245.6:5060 returned -1: Operation not permitted |
| 04:32.24 | fling | [3766851.368414] nf_ct_sip: dropping packetIN= OUT=ppp0 SRC=10.80.230.79 DST=188.187.245.6 LEN=555 TOS=0x00 PREC=0x00 TTL=64 ID=2633 PROTO=UDP SPT=5060 DPT=5060 LEN=535 UID=101 GID=120 |
| 04:32.37 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 04:33.57 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-45-142.lsanca.fios.verizon.net) |
| 04:34.34 | *** join/#asterisk deo_ (~deo@122.53.72.218) |
| 04:40.25 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 04:44.58 | igcewieling | that might be the reason |
| 04:45.26 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 05:04.12 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.155) |
| 05:16.01 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 05:30.30 | ruben231 | hi guys how do i create custom voicemail greeting or unavailable grreting..any idea guys.? |
| 05:31.02 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 05:31.42 | igcewieling | ruben231: perhaps, login to voicemail and explore the options? |
| 05:35.30 | *** join/#asterisk andy09usa (~Andrey@81.211.36.129) |
| 05:36.44 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 05:51.11 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 05:58.51 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 06:08.25 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 06:09.36 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 06:14.06 | *** join/#asterisk andy09usa (~Andrey@audotov.com) |
| 06:14.14 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 06:31.55 | *** join/#asterisk din3sh (~din3sh@41.136.253.215) |
| 06:32.13 | din3sh | gd mrning all |
| 06:44.46 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 06:57.53 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.181.145) |
| 07:00.17 | jmls | anyone got astmanproxy running on a raspberry pi ? |
| 07:00.29 | *** join/#asterisk bytemaster (~erw@host81-150-217-168.in-addr.btopenworld.com) |
| 07:01.30 | ChannelZ | No but I just ate a bowl of fruit loops |
| 07:05.03 | din3sh | :D |
| 07:05.46 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:06.07 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:14.20 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:16.07 | ectospasm | I haven't had froot loops in a while. |
| 07:16.17 | ectospasm | that or apple jacks |
| 07:18.31 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 07:24.45 | *** join/#asterisk bogrd__ (~sandeep@115.119.115.26) |
| 07:25.32 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 07:25.44 | ChannelZ | I never liked apple jacks |
| 07:27.51 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:27.59 | bogrd__ | Hi, I'm working on automating dialscripts using FastAGI for my application. My prefered language of choice is python. I found a couple of libraries but did not know which one to use. I found starpy on the asterisk github repo. It looks good, but there is very very less documentation. Is there a good tutorial / documentation or a different library recommended? Any help would be great. Thanks! :) |
| 07:28.12 | bulkorok | friday :-) |
| 07:28.29 | ChannelZ | YAY! |
| 07:30.34 | ChannelZ | bogrd__: dunno about python specifically but the AGI interface is very simple. http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 07:31.03 | ChannelZ | You read a bit from STDIN and write a bit to STDOUT. |
| 07:35.10 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 07:37.04 | bogrd__ | ChannelZ: yes. I saw that. But as people have already implemented these, i thought why reinvent the wheel by writing everything from scratch instead use a library written by someone.. |
| 07:37.13 | ChannelZ | try pyst |
| 07:37.50 | bogrd__ | ChannelZ: on it.. thanks :) |
| 07:43.44 | *** join/#asterisk din3sh (~din3sh@41.136.255.14) |
| 07:44.20 | din3sh | having an issue with cisco phones running on asterisk 1.8.14 , pickup drops call after 30sec |
| 07:47.36 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.155) |
| 08:03.08 | wdoekes | din3sh: sounds like an ACK that doesn't get through |
| 08:03.09 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 08:08.58 | din3sh | how to fix this? |
| 08:08.58 | din3sh | :s |
| 08:09.27 | wdoekes | you should start by capturing what happens |
| 08:09.42 | wdoekes | on the asterisk side: sip set debug on |
| 08:25.47 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:f0d3:32d1:ff4d:3c18) |
| 08:26.14 | din3sh | ok |
| 08:26.45 | din3sh | anybody here using Gareth's patch for presence/BLF with Cisco phones? |
| 08:27.26 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 08:53.33 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 08:54.11 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:54.36 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 09:03.32 | din3sh | anybody here using Gareth's patch for presence/BLF with Cisco phones? ---> https://issues.asterisk.org/jira/browse/ASTERISK-13145#comment-202714 |
| 09:14.49 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 09:29.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 09:42.41 | [sr] | hi |
| 09:46.52 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 09:50.26 | madhatt3r | hello |
| 09:50.44 | madhatt3r | anybody installed an atcom ax4g gsm card on elastix? |
| 09:51.34 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 09:52.34 | madhatt3r | i tried following installing on 2.0.3 as per atcoms instructions and driver, but no joy |
| 09:53.01 | madhatt3r | thinking that maybe elastix 2.4 would abide on this regard, but not really sure |
| 10:10.01 | *** join/#asterisk jmls (~somefake@217.144.151.2) |
| 10:17.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 10:26.00 | stian | Does anyone know where I can find a dial plan for Norway? :( I'm having a hard time finding any information on this. |
| 10:28.58 | ectospasm | can you tell the difference between local, long distance (national), or international numbers? << stian |
| 10:31.00 | stian | uhm |
| 10:31.32 | stian | Well, in my case our phone numbers are for example +47 72881585 |
| 10:31.56 | stian | so, a dial pattern which is like this: XXXXXXX |
| 10:31.58 | stian | should be correct |
| 10:32.11 | stian | and I let the provider handle the rest |
| 10:32.13 | stian | afaik |
| 10:32.48 | ectospasm | make sure you set that pattern to be _XXXXXX... there's probably an area prefix, so if you really wanted to be pedantic you could figure out the format for those... |
| 10:33.06 | stian | and I should set this in the trunk or the outbound route? |
| 10:33.09 | stian | That's always confused me :P |
| 10:33.12 | stian | where to put the dial rules. |
| 10:33.37 | kaldemar | stian: which GUI are you using? |
| 10:33.49 | stian | Incrediblepbx |
| 10:34.21 | kaldemar | they will know better where to put your patterns. do they have their own support channel? |
| 10:34.26 | stian | dunno |
| 10:34.32 | stian | hmm, might it be a codec issue perhaps? |
| 10:34.43 | ectospasm | where did you get that idea? |
| 10:34.51 | stian | not sure |
| 10:34.57 | stian | stupid idea, since calling my cell phone works |
| 10:35.03 | stian | and it's the same amount of numbers |
| 10:35.08 | ectospasm | You'll need to find the IncrediblePBX support forum or IRC channel. |
| 10:35.15 | stian | -_- |
| 10:35.30 | ectospasm | You might have to have a special prefix, that your PBX isn't dialing. |
| 10:35.31 | kaldemar | and be clear on what does not work and what you observe. |
| 10:36.02 | stian | well, there is not difference really, since my cell 99361296 works, but dialing for example 72881585 does not. |
| 10:36.12 | din3sh | atcom? |
| 10:36.16 | din3sh | gd luck with that |
| 10:36.47 | stian | hah what? |
| 10:37.14 | din3sh | not you stian |
| 10:37.15 | din3sh | :) |
| 10:37.40 | stian | <PROTECTED> |
| 10:37.40 | stian | <PROTECTED> |
| 10:37.44 | stian | This is just weird to me. |
| 10:37.46 | stian | It doesn't make any sense. |
| 10:39.24 | din3sh | using E1? |
| 10:39.31 | stian | E1? |
| 10:39.48 | din3sh | what kind of trunk are you using to dial out |
| 10:39.52 | stian | Phonzo |
| 10:39.55 | stian | same as at my office. |
| 10:41.08 | stian | Let me try to use the dial patterns I have at work |
| 10:41.10 | stian | should work |
| 10:42.25 | stian | :) |
| 10:42.34 | stian | somehow the dial patterns I have set up at work are insane |
| 10:42.39 | stian | can't even recall doing it :P |
| 10:42.49 | stian | gah, same problem |
| 10:46.39 | stian | ay! |
| 10:46.40 | stian | got it |
| 10:47.07 | stian | works now |
| 10:50.23 | stian | think it might be because asterisk didn't have a local ip set... |
| 10:50.30 | stian | even though it is set to static |
| 10:50.31 | stian | :) |
| 10:50.32 | stian | got it working though |
| 10:57.58 | bogrd__ | ChannelZ: I've configured Asterisk-11 to connect to a MySQL database in the same network. I have a table containig all the fields required for a SIP client to register.When I try to register from Ekiga to asterisk, I get the message 'Forbidden please check if the username and password are correct'. |
| 10:58.22 | bogrd__ | ChannelZ: I checked it with the database but its correct/ Please help |
| 11:03.03 | bogrd__ | WIMPy: ^^ |
| 11:03.36 | ectospasm | don't target individual users, we're all volunteers here. << bogrd__ |
| 11:04.26 | jacekowski | bogrd__: enable querylog in your sql and check if asterisk is querying the database |
| 11:04.37 | jacekowski | and it's getting the right results |
| 11:04.58 | bogrd__ | ectospasm: sorry. I had asked them before so i asked again.. |
| 11:05.05 | bogrd__ | jacekowski: one sec.. will check that.. |
| 11:08.34 | din3sh | anybody tried Call Completion Request on 1.8? |
| 11:11.15 | WIMPy | bogrd__: What? Where? *look around* |
| 11:13.25 | bogrd__ | WIMPy: sorry. had some problem with connecting mySQL with asterisk.. jacekowski i checked it.. asterisk is not querying the database.. but odbc show in CLI shows its connected... |
| 11:14.51 | WIMPy | has never used sql in any application. |
| 11:16.22 | jacekowski | bogrd__: check your logs |
| 11:17.16 | bogrd__ | jacekowski: asterisk logs or mysql? i checked mysql logs and it does not show any interaction with asterisk... |
| 11:19.09 | jacekowski | asterisk |
| 11:19.26 | din3sh | I get SIP/2.0 603 Declined |
| 11:19.26 | din3sh | when using exten => _30,1,CallCompletionRequest |
| 11:21.00 | bogrd__ | jacekowski: this is what i get when i try to register a sip client with asterisk ( http://paste.kde.org/667364 ) |
| 11:21.09 | bogrd__ | jacekowski: in the asterisk log |
| 11:21.56 | jacekowski | not then |
| 11:22.04 | jacekowski | when you start it up and it tries to connect to sql |
| 11:22.24 | bogrd__ | jacekowski: one sec.. on it.. |
| 11:27.31 | madhatt3r | din3sh: what would you recommend then? |
| 11:28.58 | bogrd__ | jacekowski: here is the asterisk log from the point when it starts... ( http://paste.kde.org/667370 ) |
| 11:41.15 | din3sh | madhatt3r: have many bad experiences with atcom hardware |
| 11:41.58 | madhatt3r | yeah, im having them too |
| 11:42.00 | din3sh | buggy at best |
| 11:42.07 | din3sh | which hardware |
| 11:42.08 | madhatt3r | thinking on buying sangoma or digium, but not sure |
| 11:42.09 | din3sh | ? |
| 11:42.19 | madhatt3r | fxs and gsm |
| 11:42.27 | madhatt3r | sorry |
| 11:42.31 | madhatt3r | fxo and gsm |
| 11:42.32 | din3sh | i bought few IP04/08s |
| 11:42.46 | din3sh | and phones |
| 11:42.58 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:43.34 | madhatt3r | chekin it out now |
| 11:43.36 | madhatt3r | looks good |
| 11:43.55 | din3sh | for provisioning the phones, you have to log in web-gui and select dhcp option 66 as yes, what kind of provisioning is that |
| 11:43.56 | din3sh | lol |
| 11:44.12 | din3sh | you have to provision the phone manually for automatic provisioning |
| 11:46.49 | madhatt3r | can i put whatever asterisk distro i please on these though? |
| 11:47.23 | din3sh | have to ask atcom |
| 11:47.59 | din3sh | good luck trying to make them understand english |
| 11:48.54 | *** join/#asterisk alsuren (~dlaban@80.169.133.251) |
| 11:49.21 | madhatt3r | ok i thought you was making a recommendation |
| 11:49.51 | madhatt3r | no no, i think ive had enough of atcom, we bought some phones too and they aint very good either |
| 11:50.00 | madhatt3r | tried to contact support to no avail too |
| 11:51.12 | din3sh | i have like 50-60 phones dead |
| 11:51.15 | din3sh | or even more |
| 11:51.21 | din3sh | in less than 3years |
| 11:51.30 | madhatt3r | right |
| 11:51.36 | madhatt3r | we are sendin em back tomorrow already |
| 11:56.18 | bogrd__ | jacekowski: we analysed the logs and found a message which says, "Failed to obtain database object for 'table_name_here'!" . |
| 12:01.36 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 12:08.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:09.33 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 12:10.23 | izx | Is it possible hide caller id in asterisk for a particular SIP ??? |
| 12:12.00 | ectospasm | izx: sure. put that SIP peer/friend/whatever into its own dialplan context, and hide the outbound CID with the CALLERID() function... |
| 12:13.21 | izx | ectospasm: Could you pls share here an example?? Because i dont want to break my current setup.. |
| 12:13.59 | ectospasm | ...Set(CALLERID(name)="${WHATEVER}") |
| 12:14.26 | ectospasm | izx: Set(CALLERID(num)=99999999) |
| 12:14.37 | ectospasm | you'd have to duplicate it |
| 12:14.57 | ectospasm | Or use dialplan logic to determine if a given caller should hide caller ID |
| 12:14.58 | WIMPy | Or to hide set(CALLERID(pres)=restricted). |
| 12:15.26 | izx | I dont want to display any numbers.. Instead i want to display text like "Private Number" or some other text.. |
| 12:15.32 | WIMPy | has anyone ever tried to use that with setvar in the peer? I think that might work. |
| 12:16.42 | ectospasm | I dunno if you can call dialplan functions in setvar.... it'd be cool (and logical) to do so... |
| 12:18.22 | Greenlight | Does any part of the queueing in asterisk cause a new thread to be spawned? I'm trying to track down an issue with performance where it seems caused by a large number (50-100/s) of threads being spawned by asterisk, and this is causing problems. I'm using a lot of queues and am wondering if this could be the cause, as I've servers with more calls and calls per second that don't have issues |
| 12:18.23 | Greenlight | but aren't using queues. |
| 12:18.47 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 12:19.12 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:20.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:24.57 | kaldemar | functions can be used in sip.conf setvar. |
| 12:29.56 | *** join/#asterisk DarwinSurvivor (~DarwinSur@S0106687f74a1273d.vs.shawcable.net) |
| 12:31.36 | din3sh | CallCompletionRequest() does not seem to work |
| 12:32.30 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 12:40.00 | izx | ectospasm: Can you pastebin the sample dialplan here with caller id function.. |
| 12:40.22 | ectospasm | izx: I'll give you one line |
| 12:40.32 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 12:40.37 | *** part/#asterisk bakermd (~bakermd@38.104.0.142) |
| 12:40.44 | ectospasm | I'll even follow WIMPy's suggestion |
| 12:41.12 | WIMPy | was just about to ask if we didn't provide enough examples. |
| 12:41.26 | ectospasm | izx: exten => _NXXNXXXXXX,n,Set(CALLERID(pres)=restricted) |
| 12:42.18 | izx | ectospasm: What i am asking you is, can i set this for particular SIP??? |
| 12:42.52 | ectospasm | izx: like I said, use a different dialplan context for each SIP peer/user/friend that needs to hide CID. |
| 12:43.07 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 12:43.16 | ectospasm | ...but that's just one way to do it, there may be others that escape me now. |
| 12:44.55 | WIMPy | You might be able to do it on the phone, but I'm not sure that invormation survives in Asterisk. |
| 12:45.30 | Greenlight | Would dialling out using a Local/ channel which enters the dialplan to determine which SIP trunk to use, spawn an extra tread in comparison to dialling out using the SIP/ trunk "directly"? |
| 12:45.55 | izx | I am afraid to customize the default dial plan?? Anything can be done from SIP settings?? |
| 12:46.46 | WIMPy | izx: Try the setvar thing. |
| 12:49.25 | ectospasm | yeah, instead of Set(..., try in sip.conf: setvar=CALLERID(pres)=restricted |
| 12:49.37 | ectospasm | it is not guaranteed to work, just a try |
| 12:50.03 | kaldemar | din3sh: it does. |
| 12:51.22 | kaldemar | izx: "for particular SIP" needs to be elaborated. do you mean incoming calls that match a particular peer or outgoing calls that use a peer? |
| 12:53.22 | izx | kaldemar: For example we have configured a SIP starting from 4001 to 4499.. I want to hide caller id when an outgoing call is made from a particular sip say 4010.. |
| 12:54.02 | kaldemar | "a SIP" is nothing in asterisk. are you talking about devices/peers in sip.conf or about extensions (=numbers) in dialplan? |
| 12:54.32 | izx | I am talking about the extensions.. |
| 12:54.38 | kaldemar | sounds like you mean [4010] in sip.conf (i.e. a device/peer). |
| 12:54.54 | izx | Yes you are right.. |
| 12:55.04 | kaldemar | something you define in sip.conf is not an extension. |
| 12:55.24 | kaldemar | extensions are the "exten =>" lines you have in extensions.conf, which is the dialplan. |
| 12:56.00 | izx | Sorry i mean the device/peer.. |
| 12:56.24 | kaldemar | the setvar solution is what you want then. |
| 12:57.00 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 12:58.18 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:58.28 | din3sh | kaldemar: I see call request |
| 12:58.42 | din3sh | when i do "cc report status" |
| 12:58.55 | din3sh | i can see the request |
| 12:58.59 | din3sh | but then nthg happens |
| 12:59.17 | din3sh | it is deleted without the call completion |
| 12:59.44 | kaldemar | maybe you misconfigured it. |
| 12:59.52 | din3sh | i mean the phone does not ring. all the phones in this test are in same context |
| 13:00.20 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:02.56 | din3sh | my css.conf ---> http://pastebin.com/tQz3KkHv |
| 13:03.53 | din3sh | sip peer setting --> http://pastebin.com/jpa4h5YH |
| 13:04.04 | din3sh | i have set limitonpeer=yes in general section |
| 13:05.32 | kaldemar | you may need to set busylevel on the peer and use other policies. |
| 13:06.01 | din3sh | busylevel? |
| 13:06.05 | din3sh | where to set that? |
| 13:06.56 | ghost75 | is this normal that cisco phone displays "upgrading" when no lan cable is connected? |
| 13:07.36 | din3sh | the phone is stuck in a previously started upgrade |
| 13:07.52 | ghost75 | great .... |
| 13:07.58 | ghost75 | is it then bricked? |
| 13:08.11 | din3sh | connect it to the lan |
| 13:08.19 | din3sh | with proper dhcp option and firmware |
| 13:08.27 | din3sh | let it upgrade |
| 13:08.33 | ghost75 | i bought it used |
| 13:08.55 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 13:10.13 | din3sh | set up the the tftp environment, tail the tftp log to see if it does request firmware files |
| 13:10.43 | ghost75 | will do, just wanted to make sure its not abnormal behaviour |
| 13:10.56 | din3sh | or if you can access admin settings |
| 13:11.02 | din3sh | set the tftp server manually |
| 13:11.17 | din3sh | but i bet since its stuck you wont be able to do that |
| 13:11.49 | din3sh | once the phone has proper ip address with option 66& 150, it should normally download the firmware |
| 13:11.51 | ghost75 | correct |
| 13:12.08 | din3sh | bricked cisco's dont even light up |
| 13:12.13 | din3sh | the screen stays dull |
| 13:12.49 | ghost75 | do you recommend specific sip firmware version? |
| 13:13.10 | din3sh | my frnd plugged a SPA 5v to a 48V cisco power cube, doesnt turn on |
| 13:13.13 | din3sh | try the latest |
| 13:13.33 | ghost75 | 9-3-1sr1 is latest i found from various sources |
| 13:13.38 | din3sh | thought you have to change proxy setting for firmware 9.x to register with asterisk |
| 13:14.03 | din3sh | yeah you'll have to change <proxy> to USECALLMANAGER |
| 13:14.11 | din3sh | else it wont register on ASterisk |
| 13:14.39 | ghost75 | where is that? |
| 13:15.41 | din3sh | while you at it, try this patch for BLF/prsence and tell me if it works |
| 13:15.42 | din3sh | :p |
| 13:15.43 | din3sh | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 13:16.01 | din3sh | <PROTECTED> |
| 13:16.18 | din3sh | earlier versions you jst had to put the asterisk ip address in there |
| 13:16.26 | ghost75 | in sep.cnf.xml ? |
| 13:16.31 | din3sh | yup |
| 13:16.34 | ghost75 | k |
| 13:16.55 | din3sh | <transportLayerProtocol>4</transportLayerProtocol> change value to 2 |
| 13:17.06 | din3sh | new firmware now supports tcp |
| 13:17.11 | din3sh | ok am off |
| 13:17.17 | din3sh | bbl |
| 13:17.36 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 13:17.53 | madhatt3r | bye din |
| 13:19.14 | bogrd__ | hi, i'm trying to debug database connection to asterisk-11. someone please help me.. the asterisk logs say "Failed to obtain database object" |
| 13:19.58 | bogrd__ | i've been tryin from past 4 hours without any leads.. |
| 13:20.22 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:20.39 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:21.46 | *** join/#asterisk polysics (~Adium@c-71-56-124-110.hsd1.ga.comcast.net) |
| 13:25.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-lmruqazuvgakokgg) |
| 13:27.02 | bulkorok | short question: ist this a vaild from-header: From: "A+B Cdefg" <sip:A+B Cdefg@192.168.1.1>;tag=bla |
| 13:31.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6d78:65d3:d26b:d8ea) |
| 13:37.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 13:38.24 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 13:47.26 | leifmadsen | bulkorok: the space might need to be escaped to %23 |
| 13:48.20 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:52.37 | *** part/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 13:55.19 | *** join/#asterisk serafie (~erin@nat/digium/x-hpbaejcbiswfeocv) |
| 13:57.11 | bulkorok | yeah... I suppose there is an error with the space... I have an openser that sends 500 server error with space in from |
| 13:57.16 | *** part/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 13:58.51 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:59.07 | bulkorok | and two softclients that never register with a space in username... |
| 14:06.42 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 14:10.57 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:11.14 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:11.45 | *** join/#asterisk kuruption (kuruption@laffs.at.the.lol.lawlz.lulz.liberlawls.com) |
| 14:14.31 | wdoekes | leifmadsen/bulkorok: %20 |
| 14:14.42 | Changos | Hi guys, is my first time configuring one E1 card, and team from PSTN said me that I need configure "the number A" or Identification number the E1 Card, I searching on google, and don't find nothing useful :s |
| 14:15.13 | leifmadsen | wdoekes: oops you're right, %23 is # |
| 14:15.39 | Changos | signaling, frame, kernel module, work fine |
| 14:16.48 | bulkorok | wdoekes: so a "real" space in displayname is not a good idea... |
| 14:16.58 | wdoekes | bulkorok: in display name it's fine |
| 14:17.06 | wdoekes | it isn't in the user-addr |
| 14:17.29 | wdoekes | From: "A+B Cdefg" <sip:A+B%20Cdefg@192.168.1.1>;tag=bla would be valid |
| 14:18.13 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 14:19.23 | bulkorok | ok... then I'm happy that Freeswitch is taking the call so that I can try to change the space to %20 in addr |
| 14:25.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:25.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:29.47 | kaldemar | Changos: the A number is also known as caller id number. that's not really configured on the card. |
| 14:30.49 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:32.22 | kuku | What would cause an AGI to run 10 seconds, but if I ran it from the command line with the same parameter it takes a second ? |
| 14:32.58 | kuku | I have 7 script that I run in my dialplan before it hits the trunk, and they run fine one moment, then they slow down |
| 14:33.34 | wdoekes | kuku: add logging to your AGI |
| 14:33.47 | kuku | what type of logging ? |
| 14:33.49 | wdoekes | or "tracing" rather |
| 14:34.08 | wdoekes | so you can find out where the slowness is |
| 14:34.12 | kuku | I did an strace, it takes less than a second to run via command line, same parameter |
| 14:34.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:34.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:34.45 | wdoekes | we heard that |
| 14:35.01 | kuku | write(1, "SET VARIABLE CALL_COUNTER \"2\"\n", 30SET VARIABLE CALL_COUNTER "2" |
| 14:35.40 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 14:35.44 | kuku | "how" do I add this trace it then ? |
| 14:35.55 | kuku | Just need someone to point me in the right direction |
| 14:36.28 | wdoekes | it's your agi script.. add a couple of fwrite's to a log with timestamps |
| 14:36.56 | wdoekes | until you know where it stalls, we cannot help you |
| 14:39.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qsxcykevmavqqhvl) |
| 14:39.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:42.16 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 14:43.29 | Greenlight | Asterisk is spawning 50-90 new threads per second (!); somethings not right here surely... |
| 14:44.38 | pabelanger | chan_iax2? |
| 14:45.07 | igcewieling | Why is it people insist on not using AGI libraries? |
| 14:45.32 | WIMPy | What are they good for? |
| 14:45.35 | igcewieling | kuku in the cli agi<tab> |
| 14:45.47 | igcewieling | WIMPy: handing all the protocol crap. |
| 14:46.02 | WIMPy | It's just text. |
| 14:46.15 | kuku | WHy ? I'm running a perl script and writing variables back |
| 14:46.37 | igcewieling | like reading STDIN on startup (REQUIRED) and parsing the data asterisk hands it. for example AGI won't work right if you don't clear out stdin |
| 14:47.12 | WIMPy | What languages do you know that can't handle text on stdin/stdout? |
| 14:47.49 | igcewieling | I did not say it can't handle it. not using an AGI lib makes writing AGIs more error prone and harder to write. |
| 14:48.43 | WIMPy | Searching for a library that does simple text i/o seems to me like overcomplicating things. |
| 14:50.07 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-245-234.pools.arcor-ip.net) |
| 14:50.38 | igcewieling | WIMPy: searching? there is a standard AGI library for the common languages. They are lightweight. |
| 14:51.00 | igcewieling | kuku: when your AGI starts up do you read the data on STDIN which asterisk sends. |
| 14:51.17 | igcewieling | WIMPy: over and over again I see people with AGI problems because they didn't follow the spec. |
| 14:51.59 | WIMPy | If you can't read, a library won't help, either. |
| 14:53.13 | wdoekes | the nazi's used AGI libs! |
| 14:53.33 | igcewieling | wdoekes: they also used toilets |
| 14:54.03 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:54.11 | coppice | shitty people do that |
| 14:56.44 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 15:01.39 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 15:02.03 | igcewieling | WIMPy: also if you are doing AGI you are going to end up writing functions to handle all the details anyway so why use an AGI library which already does that? |
| 15:04.48 | WIMPy | Because it most probably won;t do it the way I need it. |
| 15:05.02 | WIMPy | And no. For AGI I din;t need anyhing so far. |
| 15:05.15 | WIMPy | AMI might be more interesting. |
| 15:06.56 | WIMPy | In my experience, libraries are often like GUIs. Tehy claim to make things easier, but unless you do exactely the same thing the creator needed, they actually make things a lot more complicated. |
| 15:07.09 | igcewieling | WIMPy: why? It is just a socket connection? 8-) |
| 15:07.19 | pabelanger | Who else is getting HDLC Abort (6) errors? |
| 15:07.33 | igcewieling | pabelanger: only people with errors on their circuit |
| 15:07.41 | igcewieling | or using really old Digium cards |
| 15:07.48 | WIMPy | Yes, it is, but there is moe recycling potential with event parsing. |
| 15:08.28 | WIMPy | pabelanger: A few times a month. |
| 15:08.56 | kuku | http://pastebin.ca/2311084 << any ideas why one side of the call gets dropped right away after I bridge ? |
| 15:08.59 | kuku | brb |
| 15:12.03 | pabelanger | igcewieling: Circuit clean, and gen 5 TE220 |
| 15:12.26 | pabelanger | Digium wants we to upgrade to 2.6, but not convinced it will fix it. |
| 15:12.32 | pabelanger | Next step is RMA the card |
| 15:12.38 | pabelanger | but was curious what others did |
| 15:12.42 | igcewieling | pabelanger: IIRC that error means "dahdi driver received corrupted data" |
| 15:12.50 | pabelanger | Right |
| 15:12.59 | bulkorok | wdoekes: problem solved with changing spaces to underscore.. dirty, but working... |
| 15:13.09 | wdoekes | kuku: I thought we were debugging slowness. and you pastebin seems invalid |
| 15:13.12 | wdoekes | s/you/your |
| 15:13.20 | pabelanger | but data could be getting corrupt via IRQs, if circuit is clean |
| 15:13.26 | pabelanger | but here is the kickers |
| 15:13.29 | wdoekes | bulkorok: who was sending the space? asterisk? |
| 15:13.29 | pabelanger | kicker* |
| 15:13.55 | pabelanger | If I slave timing from the network, I get thousands of HDLC errors... if I change timing to internal, they go away |
| 15:14.05 | bulkorok | wdoekes: user-agent: FPBX-2.8.1(1.8.12.0) |
| 15:14.32 | igcewieling | Until I stopped using Digium cards years ago that error was most commonly caused by sharing interrupts or crappy motherboards interacting with Digium cards, and occasionally circuit issues. When I switched to Sangoma the only time I ever had that error was circuit issues. I'm told Digium solved the card issues long ago and have no reason not to believe that. |
| 15:14.34 | wdoekes | pedantic=no? |
| 15:15.11 | WIMPy | pabelanger: Interesting. |
| 15:15.23 | bulkorok | can't say.. it's not my machine... we have many customers sending spaces in From... so I have to hack them out like I did :-/ |
| 15:15.30 | igcewieling | pabelanger: sounds clock slippy. What is on the other end of the line? |
| 15:15.47 | pabelanger | igcewieling: Hmm, I should give Sangoma I try |
| 15:15.55 | pabelanger | igcewieling: Bell Canada |
| 15:15.58 | pabelanger | the CO |
| 15:16.12 | igcewieling | pabelanger: you have a modern Digium card, switching to Sangoma is unlikely to help. |
| 15:16.40 | WIMPy | Too modern for your dahdi? |
| 15:16.47 | igcewieling | can't hurt to give it a try if you have one on hand, but I'd not buy one to try it. |
| 15:17.05 | pabelanger | Agree |
| 15:17.12 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 15:17.14 | pabelanger | my understanding is 2.5 support gen 5 cards |
| 15:17.21 | pabelanger | but will go to 2.6 like they asked |
| 15:17.26 | pabelanger | just packaging it now |
| 15:17.44 | wdoekes | bulkorok: with pedantic=yes (the default since 1.8) it should properly encode spaces in usernames (which are a bad idea anyway), as long as the asterisk is recent enough |
| 15:18.25 | igcewieling | wdoekes: for some reason I seem to remember at least 1.8+ defaults to pedantic=yes, but I don't recall where I saw that info |
| 15:19.15 | lorsungcu | sup fools |
| 15:19.25 | igcewieling | pabelanger: if you try Sangoma and it fixes the issue I very much want to hear about it. |
| 15:20.16 | bulkorok | wdoekes: good to know... but for the future, I will recommend not using spaces... |
| 15:20.39 | pabelanger | igcewieling: Ya, won't be able to convince this client to change vendors at this point, but might try them out for a new deployment |
| 15:20.51 | pabelanger | but honestly, TDM installs for us is getting smaller and smaller |
| 15:21.03 | igcewieling | I love PRI installs. |
| 15:22.54 | jmetro | TDM..tedium. |
| 15:23.11 | WIMPy | I prefer Linux supported hardware. |
| 15:23.11 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:23.11 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:25.25 | pabelanger | I think this site is a dell r310, or something |
| 15:27.48 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 15:32.33 | *** join/#asterisk evilman_home (kvirc@78-106-161-7.broadband.corbina.ru) |
| 15:36.15 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 15:38.14 | kuku | wdoekes: the pastebin is regarding a different issue im having. |
| 15:46.19 | *** join/#asterisk deo (~deo@112.198.90.165) |
| 15:59.46 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 15:59.50 | saint_ | hi all |
| 16:11.14 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.248.136) |
| 16:14.47 | Kyosh | AMI UpdateConfig is not working under 1.4.21 when asterisk is in realtime http://pastebin.ca/2311653 is this is an effect of realtime? is there any other way to programmatically create and edit contexts in extensions.conf while asterisk is in realtime? i can create new extensions in the database without a problem but asterisk doesnt know the context exists without extensions.conf being updated. thanks. |
| 16:15.35 | igcewieling | Kyosh: does asterisk see new contexts if you do a "dialplan reload" after updating the database? |
| 16:15.50 | Kyosh | no. i wish it did |
| 16:15.58 | Kyosh | actually.. hold on |
| 16:16.48 | Kyosh | yes in the CLI it shows the dialplan, but only for existing contexts. however the new extension added to the context is not added to the database or the extensions.conf. its only in temporary memory |
| 16:17.08 | Kyosh | all i really want to do is add a new context to extensions.conf programmatically. |
| 16:17.14 | Kyosh | not want...need. |
| 16:18.34 | leifmadsen | use includes and then programmatically add information to a file |
| 16:19.19 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.248.136) |
| 16:19.22 | Kyosh | leif, and i would still need to programmatically add the includes. then after that, i would have to see if i can edit the includes via AMI. |
| 16:19.35 | leifmadsen | I think either way you need to do dialplan reloads to make the changes active |
| 16:19.53 | Kyosh | ok at least i can try that. let me go bang my head on the wall for a bit with that. it was an idea but i was hoping for a more direct route to get the job done. thanks. |
| 16:20.13 | Kyosh | leif, of course. i was going to reload regardless. |
| 16:20.30 | Kyosh | is the problem i am having because i am using realtime, you think? |
| 16:20.38 | Kyosh | or maybe just an old version of asterisk |
| 16:20.41 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 16:20.44 | WIMPy | Why do you want to add contexts dynamically anyway? |
| 16:21.42 | Kyosh | i have a few reasons. create new inbound contexts for ivr, new outbound contexts for ivr with an ami autodialer |
| 16:22.38 | WIMPy | And you do that all the time? Sounds like a situation whe the usual 'dialplan reload' wouldn't hurt. |
| 16:22.45 | kuku | This slow AGI is killing me - I cna't find the reason why asterisk take so loong to run them |
| 16:23.22 | [TK]D-Fender | Kyosh, Use AMI to launch a script to regenerate your dialplan based on the DB |
| 16:23.46 | [TK]D-Fender | Kyosh, Or a DB stored procedure, etc. |
| 16:23.53 | [TK]D-Fender | Kyosh, Both of which I mentioned yesterday |
| 16:24.03 | Kyosh | wimpy, we need to have our users have the ability to create contexts for their needs. |
| 16:24.27 | Kyosh | fender, and both in this case are not working. |
| 16:24.41 | [TK]D-Fender | Kyosh, Why not? |
| 16:24.59 | Kyosh | thats why i was here, to ask why. |
| 16:25.05 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 16:25.17 | [TK]D-Fender | Kyosh, Those 2 methods do work... |
| 16:25.50 | *** join/#asterisk vlad_sta_ (~vlad_star@178.176.248.136) |
| 16:26.13 | Kyosh | the AMI method keeps giving me the missing action in request error http://pastebin.ca/2311653 |
| 16:26.34 | Kyosh | and the CLI method has no way to create a context, just add extensions within an existing context |
| 16:27.10 | [TK]D-Fender | Kyosh, Those are not the 2 things i told you last yesterday and again jsut now. |
| 16:27.22 | [TK]D-Fender | Kyosh, I did not just say "add direct via AMI" |
| 16:27.30 | [TK]D-Fender | Kyosh, I said "trigger a rebuild script" |
| 16:27.32 | WIMPy | Missing action probably means tyour Asterisk don't know that acton (yet). |
| 16:27.48 | [TK]D-Fender | 1.4 = ancient |
| 16:27.53 | Kyosh | fender, then i am sorry to say i am not fully grasping what you are hinting that i do. i do not believe that you have directly giving me any direction, only suggested that i look into things that require more research. |
| 16:28.11 | Kyosh | wimpy, thats what i was afraid of, thanks. |
| 16:28.22 | Kyosh | i know 1.4 is ancient, sadly. |
| 16:28.24 | [TK]D-Fender | Kyosh, Trigger a script that will mannually recreate a dialplan file based on your DB,. |
| 16:28.36 | Kyosh | i was going to recompile for 64 bit anyways, i should just make the jump to 1.8 |
| 16:29.08 | Kyosh | unfortunately my server admin is not giving me the VM space required for test building |
| 16:29.25 | WIMPy | Why do you want to upgrade to an old version? |
| 16:29.34 | Kyosh | wimpy, you read wrong |
| 16:29.42 | [TK]D-Fender | Kyosh, 1.8 IS old |
| 16:29.44 | Kyosh | oh wait |
| 16:29.46 | Kyosh | 1.8 is old now? |
| 16:29.46 | [TK]D-Fender | Kyosh, 1.4 is WORSE |
| 16:29.53 | Kyosh | sheesh |
| 16:29.54 | [TK]D-Fender | Kyosh, Read the channel topic. |
| 16:30.02 | [TK]D-Fender | Kyosh, Asterisk 11 is the latest LTS |
| 16:30.11 | Kyosh | from v1 to v11 ? |
| 16:30.17 | [TK]D-Fender | Kyosh, Correct |
| 16:30.22 | Kyosh | big jump |
| 16:30.31 | [TK]D-Fender | Technically 10 was in between |
| 16:30.35 | Kyosh | heheh |
| 16:30.52 | Kyosh | i should read about the new features |
| 16:30.55 | [TK]D-Fender | Kyosh, And still not needed. make the rebuid script like I mentioned. this is a 5 minute job |
| 16:31.10 | Kyosh | if i only knew how |
| 16:31.33 | Kyosh | but here is an interesting question then... |
| 16:31.52 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 16:32.26 | Kyosh | what if i had the extensions.conf located on a UNC instead of being local? how would i tell asterisk (which file to modify the location) to load the extensions.conf from a UNC located resource? |
| 16:35.39 | ChannelZ | I'd just #include it assuming that even works (on Windows) |
| 16:36.44 | kuku | How do I diagnose why AGI scripts within Asterisk are slow ( they are not slow via command line ) |
| 16:37.18 | kuku | [TK]D-Fender: Are you saying I should be running v11 instead of 1.8 ? |
| 16:37.30 | Kyosh | Channelz, hmmm.... THAT is a nice idea. #include the UNC extensions file? hmmm |
| 16:41.08 | Kyosh | bbl, thanks. |
| 16:41.13 | *** join/#asterisk serafie (~erin@nat/digium/x-hogbtcgidajnffvf) |
| 16:41.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:41.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:45.58 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 16:47.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:48.46 | Greenlight | I think I'm missing something simple here. I'm originating a call via the AMI, to say SIP/trunkname/number. I can't see a way to uniquely identify the channel which is created so that I can then track it. WHat I used to do was originate to the Local channel and fire a userevent with a variable, but I want to dial it directly now. Whats the established method to track the call? |
| 16:50.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:51.22 | kuku | http://pastebin.ca/2311876 << please help me - please - |
| 16:52.07 | Greenlight | kuku: try "agi set debug on" |
| 16:52.12 | Greenlight | See if that helps any |
| 16:52.52 | *** join/#asterisk Dovid (~Dovid@ool-1826d69c.dyn.optonline.net) |
| 16:53.22 | Greenlight | Surely there must be an "established" method for tracking calls originated via the AMI, that other folks use? Do I have to do something like set a variable in the Originate and watch for the SetVar event ? |
| 16:58.52 | Greenlight | Or use and abuse the callerid/accountcode fields? Wow it really does look like there's no proper way to do this ... |
| 16:59.24 | kuku | Greenlight: I did that |
| 17:02.43 | kuku | Greenlight: here is the debug http://pastebin.ca/2311878 |
| 17:03.30 | Greenlight | Right, and what were you expecting to happen that isnt ? |
| 17:03.40 | kuku | It took too long |
| 17:03.45 | Greenlight | It appears to set two variables. |
| 17:03.47 | kuku | yes |
| 17:03.50 | Greenlight | Too long ? |
| 17:03.51 | kuku | IT took too long |
| 17:03.52 | kuku | yes |
| 17:03.59 | Greenlight | Too long... till what? |
| 17:04.01 | kuku | 12 seconds |
| 17:04.11 | Greenlight | Oh I see |
| 17:04.18 | kuku | When I run that via cli, its less than 1 second |
| 17:04.57 | ghost75 | i hooked up cisco phone to lan and tftp server doesnt list any request :< |
| 17:05.05 | kuku | Also, when I restart the system, it takes less than 1 second in asterisk |
| 17:05.10 | Greenlight | I'd look at your script, check it's not timing out waiting on stdin input or something |
| 17:05.28 | kuku | but after a few minutes, asterisk slows down at that. |
| 17:05.40 | Greenlight | Hmm how very odd |
| 17:06.25 | Greenlight | I know we execute 20 or so AGI scripts a second, all day, and never have any such issues |
| 17:06.49 | Greenlight | Although we use php not perl, I can't see that making a difference |
| 17:09.02 | *** join/#asterisk Uthark (~Uthark@190.0.58.186) |
| 17:09.29 | Uthark | Any opinions about Atcom equipment? |
| 17:09.52 | Greenlight | I'm sure I saw folks earlier saying it was crap, but no personal experiance |
| 17:11.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:12.51 | Uthark | Greenlight: Seems they were right. A company I'm working for purchased about 30 phones, and they slowed the whole network to a crawl |
| 17:14.49 | *** join/#asterisk 16SAAXE8T (~jburns@75-171-154-172.hlrn.qwest.net) |
| 17:19.19 | Greenlight | Uthark: You could capture the traffic with wireshark, and check what's actually clogging the network up |
| 17:19.24 | *** join/#asterisk [TK]D-Fender (~Joe@v5208878.static2.cidc.net) |
| 17:19.34 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:24.21 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:24.31 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 17:25.51 | Uthark | Greenlight: Those phones are supposed to support vlan tagging. I've configured them to send voice over one VLAN and let data traffic untagged |
| 17:26.53 | Uthark | I think that the phone's firmware is buggy in its vlan implementation. It even required at least 3 power cycles to deliver traffic to the correct VLAN! |
| 17:32.51 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:41.01 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 17:42.02 | kuku | I have an aserisk server, where the call gets hanged up after 25 minutes |
| 17:50.07 | ghost75 | is tftp server in dhcp option 66 or 150 ? |
| 17:51.05 | Greenlight | http://www.networksorcery.com/enp/protocol/bootp/options.htm |
| 17:52.58 | Greenlight | 66 is the common one, 150 is less common, maybe even Cisco proprietry |
| 17:53.32 | slav3_kitten | option 150 is cisco |
| 17:53.49 | ghost75 | i set both to ip address |
| 17:54.04 | slav3_kitten | what phones are you using again? |
| 17:54.11 | ghost75 | 7941 |
| 17:54.19 | slav3_kitten | ah |
| 17:54.20 | ghost75 | i dont see any response on tftp server |
| 17:54.33 | slav3_kitten | i know that tagging an everything works on 79x0s |
| 17:54.59 | slav3_kitten | i've got a working 7911 phone though |
| 17:55.03 | ghost75 | if i power on the phone its just sayin "upgrade" |
| 17:55.08 | ghost75 | thats how i got it |
| 17:55.25 | slav3_kitten | are you using a cisco switch? |
| 17:55.36 | ghost75 | no i have 48 psu |
| 17:55.40 | ghost75 | 48v |
| 17:55.43 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 17:56.16 | slav3_kitten | yea but for the vlan crap to work iirc the cisco phones ask the switch what the voice vlan an all that jazz is |
| 17:56.47 | ghost75 | no idea |
| 18:00.51 | [TK]D-Fender | ghost75, Have you tested and confirmed your DHCP server is handing out the address and that your server is offering up the files? |
| 18:01.30 | ghost75 | on linux client i checked the lease file |
| 18:01.43 | ghost75 | there is only "server-name" |
| 18:03.57 | ghost75 | http://pastebin.com/D0V10qwm |
| 18:04.10 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:04.13 | ghost75 | not sure if this option really works |
| 18:04.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:05.45 | ghost75 | the tftpd itself works |
| 18:07.43 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 18:10.41 | [TK]D-Fender | ghost75, I don't see you setting any in there. |
| 18:18.56 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-159-194.pools.arcor-ip.net) |
| 18:20.04 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 18:28.51 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 18:31.20 | *** join/#asterisk nightrid3r (~kvirc@62.205.64.59) |
| 18:33.39 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:35.50 | ghost75 | somebody here has linux client connected to dhcp server with tftp option? |
| 18:38.19 | drmessano | uh what |
| 18:39.32 | ChannelZ | sure! |
| 18:40.07 | ghost75 | does it list tftp anywhere? |
| 18:40.46 | tzanger | Katty: I saw these and immediately thought of you: http://feedproxy.google.com/~r/failblog/~3/lUgaBi-Lvyo/7039111680 |
| 18:41.39 | Kobaz | anyone use pure ftpd |
| 18:42.00 | ChannelZ | that's crazy talk! |
| 18:45.39 | [TK]D-Fender | ghost75, option tftp-server-name "192.168.55.55"; |
| 18:46.14 | ghost75 | u have in linux? |
| 18:47.12 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.41) |
| 18:47.34 | b0ot | what tools do people recommend for testing network capacity for calls and for testing QoS policies? |
| 18:48.46 | ChannelZ | sipsak is a tester thingymajig |
| 18:49.20 | ChannelZ | or wait that's not the one I'm thinking of... |
| 18:49.56 | b0ot | SIPp? |
| 18:51.38 | b0ot | i was wondering what peopple thought of that versus something like IXchariot |
| 18:53.16 | kuku | [TK]D-Fender: so 1.8 is outdated, I shoudl get v11/ |
| 18:53.55 | [TK]D-Fender | kuku, Unless you have a special reason not to. |
| 18:54.12 | [TK]D-Fender | ghost75, /etc/dhcpd.conf (ISC) |
| 18:55.06 | ghost75 | i dont have dhcp on linux, only the client i asked |
| 18:58.32 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 19:03.30 | *** join/#asterisk igcewieling (~igcewieli@68.sub-70-193-68.myvzw.com) |
| 19:05.06 | ChannelZ | doesn't follow |
| 19:11.55 | kuku | [TK]D-Fender: are there upgrades to 11 vs 1.8 ? |
| 19:12.12 | igcewieling | doesn't follow |
| 19:15.46 | navaismo | switchvox use asterisk? |
| 19:17.03 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:17.44 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 19:22.57 | igcewieling | navaismo: http://lmgtfy.com/?q=does+switchvox+use+asterisk |
| 19:23.11 | igcewieling | 8-) |
| 19:25.18 | navaismo | nope i dont go to the let me google that for you anymore |
| 19:26.05 | ChannelZ | clearly |
| 19:26.17 | ChannelZ | or just google.. |
| 19:26.36 | navaismo | nah if someone can tell me why bother into type words and words |
| 19:27.54 | ChannelZ | like the words you typed into this window? |
| 19:30.05 | navaismo | ok let me rephrase: " nah if someone can tell me why bother into type words and words and looking around without exact results" |
| 19:31.25 | navaismo | but wait, people ChannelZ preffer to be a troll than write a yes or no, Ladies and Gentlemans: "the helping people" these days |
| 19:31.57 | ChannelZ | Well the second result on "does switchvox use asterisk" is Digium... so take a guess if it uses Asterisk |
| 19:32.35 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 19:32.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:33.09 | igcewieling | navaismo: usually that is only when the question is easy to find on your own. |
| 19:33.28 | *** join/#asterisk [TK]D-Fender (~Joe@v5208878.static2.cidc.net) |
| 19:34.06 | igcewieling | I think think of you as especially needy so I was surprised by your question. |
| 19:35.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:35.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:36.19 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 19:36.30 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 19:36.54 | navaismo | i was looking for a simply yes or no from people that have used it, not to check 3 or 4 result in google, my laziness my bad |
| 19:37.26 | ChannelZ | Well, now you've learned 2 new things today. |
| 19:37.31 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-159-194.pools.arcor-ip.net) |
| 19:37.37 | ChannelZ | NEXT! |
| 19:41.06 | *** join/#asterisk jsjc (~Adium@164.Red-2-136-102.dynamicIP.rima-tde.net) |
| 19:41.13 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 19:42.42 | ghost75 | i can haz gets crazy? |
| 19:50.02 | ChannelZ | Yes. Let's party like it's 1999 |
| 19:54.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 19:54.50 | *** join/#asterisk igcewieling1 (~igcewieli@104.sub-70-193-64.myvzw.com) |
| 20:04.12 | ghost75 | so looks like cisco is using option 150 for phones |
| 20:04.29 | slav3_kitten | i said that already :P |
| 20:04.43 | ghost75 | but still not works |
| 20:05.13 | ghost75 | even changed dhcp server |
| 20:05.25 | slav3_kitten | so looking at the sys log you don't see any tftp attempts? |
| 20:05.34 | ghost75 | exact |
| 20:05.34 | ChannelZ | Can't you just tell it the tftp server yourself? |
| 20:06.01 | ghost75 | the phone is just saying "upgrade in process" |
| 20:07.05 | slav3_kitten | ghost75, with my cisco router and switch i have my 7911 working perfectly. it sounds like it may have had a failure upgrading the firmware. have you initiated a factory reset? |
| 20:07.27 | ghost75 | no, they have buttons for that? |
| 20:08.11 | slav3_kitten | http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml#ipphone |
| 20:08.15 | ghost75 | i dont have any menu |
| 20:08.29 | slav3_kitten | but you'll need firmware for it to pull down if you factory reset it |
| 20:08.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:08.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:09.30 | ghost75 | If you enter this key sequence correctly, the phone displays this prompt:upgrading |
| 20:09.32 | ghost75 | HAHA |
| 20:09.41 | ghost75 | thats whats display now |
| 20:10.51 | slav3_kitten | also on my 7911 it took forever to reload the firmware. so long in fact i thought i bricked it |
| 20:11.11 | ghost75 | but not like hours or? |
| 20:11.55 | slav3_kitten | https://supportforums.cisco.com/thread/2044177 |
| 20:12.00 | Katty | tzanger: well wouldya look at that. kitty knees |
| 20:12.09 | Katty | tzanger: can't say i've ever seen anything like that before |
| 20:13.18 | slav3_kitten | it was a nightmare to get my cisco phones upgraded to the proper firmware and playing nicely with asterisk ghost75 |
| 20:13.51 | slav3_kitten | if you're not heavily invested in cisco gear i'd highly recommend going polycom or something that's actually documented worth a shit |
| 20:14.32 | ghost75 | well it was cheap |
| 20:15.16 | ChannelZ | The questions practically answer themselves! |
| 20:15.19 | slav3_kitten | it tends to be, cisco gives zero support for anything outside of call manager, and their sip firmware is more of a kludge then anything else |
| 20:17.14 | slav3_kitten | best bet is to take a laptop. run dhcp server on it, tftp server on it, then connect the phone directly to the laptop using a hub or such |
| 20:17.24 | slav3_kitten | factory reset it and hope it works well |
| 20:20.51 | ghost75 | reset doesnt change, nothing doing on tftp |
| 20:21.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nmldjvcdlieeqcku) |
| 20:21.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:21.30 | slav3_kitten | is it just connected to a single computer running the dhcp and all that crap? |
| 20:22.01 | ghost75 | no |
| 20:22.04 | slav3_kitten | it takes a good bit of time for it to update |
| 20:22.55 | ghost75 | i see log on tftp server |
| 20:24.33 | ghost75 | in dhcp: option voip-tftp-server code 150 = ip-address; and option voip-tftp-server 192.168.2.2; |
| 20:25.16 | drmessano | I used the dhcp in dnsmasq |
| 20:25.20 | drmessano | Works like a charm |
| 20:25.34 | ghost75 | i tried openwrt and isc dhcp server |
| 20:26.37 | ghost75 | if i connect straight to pc i require crossover cable? |
| 20:26.49 | slav3_kitten | yea, or a hub |
| 20:27.02 | ghost75 | i only know if all is gigabit then no crossover is needed |
| 20:27.11 | ghost75 | but no clue when gigabit + 100 |
| 20:27.52 | ghost75 | what did you use as dhcp server? |
| 20:28.19 | slav3_kitten | whatever i got from the debian repository |
| 20:29.56 | jmetro | or if the ports are smart they automatically knwo if the cable is cross or straight and adjust to it. |
| 20:30.53 | slav3_kitten | now back to looking for a 486 33mhz computer |
| 20:30.57 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:31.16 | jmetro | and my god slav3_kitten who even has hubs anymore |
| 20:31.25 | slav3_kitten | i do |
| 20:31.46 | slav3_kitten | i got like 3 or 4 |
| 20:32.29 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.164) |
| 20:32.38 | ruben231 | hi guys |
| 20:32.46 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 20:33.14 | ruben231 | i installed asterisk 1.8...but i get only white Console please help..been a week troubleshooting |
| 20:34.44 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 20:34.55 | jmetro | so your linux terminal is white? |
| 20:35.29 | leifmadsen | use a dark console |
| 20:36.26 | ruben231 | <PROTECTED> |
| 20:36.39 | ruben231 | so logs can be idetified easly |
| 20:36.52 | jmetro | revert any changes you made then. |
| 20:37.17 | leifmadsen | -W Adjust terminal colors to compensate for a light background |
| 20:37.24 | ruben231 | <PROTECTED> |
| 20:37.37 | leifmadsen | -n Disable console colorization |
| 20:37.52 | ruben231 | <PROTECTED> |
| 20:37.58 | leifmadsen | asterisk -h |
| 20:40.12 | ruben231 | <PROTECTED> |
| 20:41.31 | ruben231 | when i run asterisk -W.....nothing happens |
| 20:42.08 | jmetro | im guessing asterisk -rW |
| 20:42.21 | igcewieling | ruben231: how are you starting Asterisk and EXACTLY what version of Asterisk are you using? |
| 20:43.34 | igcewieling | Oddly we don't get colorization when using asterisk 1.8.20.1 either and we start it using the centos init script installed using make config |
| 20:43.37 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
| 20:43.51 | igcewieling | heh, that is actually wrong. |
| 20:44.22 | ruben231 | igcewieling: im using asterisk 1.8 and for asterisk i do this start_asterisk_boot.pl |
| 20:44.24 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 20:45.10 | igcewieling | ruben231: never heard of that script. What distro are you using. |
| 20:46.20 | igcewieling | ruben231: you are not using asterisk 1.8. you might be using asterisk 1.8.15 or 1.8.20 or something like that. Now EXACTLY what verison of Asterisk are you using. |
| 20:46.42 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 20:48.22 | ruben231 | Asterisk 1.8.19.0 |
| 20:49.27 | igcewieling | If you are using CentOS, then in the Asterisk source directory do a "make config" and then "chkconfig asterisk on" then stop Asterisk and run "service asterisk start" do you still not have color when you do an "asterisk -rvvv" ? |
| 20:50.19 | ruben231 | no color when i do with asterisk -rvvvvvvvvvvvvvvv im suing Ubuntu Server.....but when i used asterisk 1.4 i get the full color of console |
| 20:51.14 | igcewieling | since you are using Ubuntu nothing I just said applies. That is why I asked what distro you are using. |
| 20:51.32 | ghost75 | slav3_kitten: i didnt check the most obvious thing |
| 20:51.46 | igcewieling | ruben231: where did you get the "start_asterisk_boot.pl" script from? |
| 20:52.25 | slav3_kitten | oh? |
| 20:52.55 | ghost75 | if it gets ip |
| 20:53.10 | slav3_kitten | was it not? |
| 20:53.29 | ghost75 | didnt check |
| 20:53.49 | slav3_kitten | working now |
| 20:53.51 | slav3_kitten | ? |
| 20:53.51 | ruben231 | igcewieling: i try to ignore that script and instead used the make config <---------------still tsame console i have_ no color |
| 20:54.35 | jmetro | did you try asterisk -rW |
| 20:54.40 | ghost75 | not yet |
| 20:54.45 | igcewieling | ruben231: stop asterisk. verify it is stopped with a "ps --ax | grep asterisk". Then run "asterisk" (no options) wait 2 seconds for it to start up, then "asterisk -rvvv". |
| 20:55.06 | igcewieling | check if you have color now |
| 20:57.46 | ruben231 | http://pastebin.com/1b0nJ0kP <-----------------after i stop it and run this ps ax | grep asterisk |
| 20:58.10 | ruben231 | should i start now by typing asterisk then ---> asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 20:59.16 | igcewieling | ruben231: your pastebin indicates asterisk might not be stopped. |
| 20:59.24 | igcewieling | not sure about that screen line |
| 20:59.38 | igcewieling | oh! and stop running asterisk under screen |
| 20:59.47 | igcewieling | that may very well be your issue. |
| 20:59.50 | ghost75 | slav3_kitten: nope doesnt get address |
| 21:00.52 | slav3_kitten | is it looking on vlans for that ip address? |
| 21:01.18 | ghost75 | its connected straight to laptop |
| 21:01.27 | ghost75 | i do not see any lease |
| 21:01.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 21:02.19 | slav3_kitten | did you factory reset it? |
| 21:02.33 | ghost75 | yes |
| 21:02.34 | slav3_kitten | also, tried using a crossover cable? |
| 21:02.37 | ghost75 | yes |
| 21:02.49 | slav3_kitten | got a hub? |
| 21:02.50 | ghost75 | other pc gets ip on this laptop |
| 21:03.08 | slav3_kitten | is the other pc a gig or 10/100 nic? |
| 21:03.17 | ghost75 | giga |
| 21:03.31 | ghost75 | no |
| 21:03.36 | ghost75 | laptop has giga, pc has 100 |
| 21:04.10 | slav3_kitten | could be a difference. i'd suggest using hub and something with wireshark to see if the 7941 is even asking for an ip |
| 21:04.20 | igcewieling | ghost75: have you looked at the output of ethtool to see if there is anything weird? |
| 21:04.34 | slav3_kitten | i'm also going to ask, are you sure you're plugged into the right port on the phone |
| 21:04.58 | ChannelZ | mmheh |
| 21:04.59 | slav3_kitten | it should be in the one labeled "10/100 SW" |
| 21:05.10 | ghost75 | wireshark would be an idea |
| 21:05.17 | ghost75 | i connected to sw port on phone |
| 21:05.23 | ChannelZ | or a hammer |
| 21:05.36 | slav3_kitten | i'd use wiresharkto see if it's even asking |
| 21:05.42 | ChannelZ | (how cheap was it exactly?) |
| 21:05.51 | slav3_kitten | and make sure you give it a good bit of time to ask |
| 21:05.55 | ghost75 | too cheap to spend so much time lol |
| 21:06.05 | slav3_kitten | good learning experience |
| 21:06.13 | slav3_kitten | also cheap and easy don't go well |
| 21:06.24 | ChannelZ | I guess if you're learning to hate Cisco, yes :) |
| 21:06.38 | ghost75 | ebay ... |
| 21:07.33 | igcewieling | In my opinion it is only a Cisco if it runs Cisco IOS. All others are not really cisco regardless of what it says. |
| 21:07.34 | ghost75 | i bought this phone for 20 and another guy for 10 (his is working lol) |
| 21:09.28 | slav3_kitten | igcewieling, i think you missed the current discussion boat |
| 21:15.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 21:15.16 | igcewieling | slav3_kitten: *nod* I assumed Cisco router. Only crazy people try using Cisco phones when there are so many much better ones out there so it didn't occur to me. |
| 21:15.41 | slav3_kitten | hey now |
| 21:15.47 | slav3_kitten | i run 6 cisco phones quite happily |
| 21:16.15 | slav3_kitten | it was a frustrating nightmare though |
| 21:16.29 | slav3_kitten | but i have a cisco router, switch with inline power, an phones |
| 21:16.39 | ghost75 | i see cdp and lldp from phone |
| 21:17.13 | slav3_kitten | well cdp is the phone trying to ask the switch wht it should configure it's internal switch module for |
| 21:17.30 | slav3_kitten | then lldp is trying to do the same |
| 21:17.38 | ghost75 | and then dhcp discover |
| 21:17.41 | ghost75 | dhcp offer |
| 21:18.07 | jmetro | cisco is nice, if they would make a better effort to be easier. i guess you cant charge people to learn your product and get certified if you dont make it obfuscated and confusing though. |
| 21:19.01 | slav3_kitten | cisco is good gear, but they rake in money with the certification shit and i don't see them getting away from that anytime soon |
| 21:19.08 | igcewieling | jmetro: The thing is Cisco doesn't WANT your business for hardphones unless you are running Call Manager. |
| 21:19.29 | ghost75 | it does it again and again, lldp,cdp,dhcp |
| 21:19.34 | slav3_kitten | ^^ that as well |
| 21:19.47 | igcewieling | I like Cisco routers, switches, etc, but their phones are really designed for call manager. |
| 21:20.11 | slav3_kitten | ghost75, give it like 15 |
| 21:20.29 | ghost75 | what? |
| 21:20.35 | slav3_kitten | minutes |
| 21:20.35 | jmetro | to me, proprietary software is as bad an idea as lotus notes. |
| 21:20.53 | ghost75 | it tells even the power consumption lol |
| 21:21.53 | slav3_kitten | ghost75, well yea, switch needs to know how much it's taking |
| 21:22.14 | ghost75 | 802.3af and stuff |
| 21:23.03 | slav3_kitten | yea mine don't do 802.3af, just cisco pre poe, poe |
| 21:23.43 | ghost75 | i think i have a sign somewhere "sell be your broken stuff" |
| 21:23.51 | ghost75 | -be +me |
| 21:24.04 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 21:24.35 | ghost75 | or does it require original cisco psu lol |
| 21:24.36 | *** join/#asterisk lorsungcu_ (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 21:26.47 | slav3_kitten | it shouldn't |
| 21:27.00 | slav3_kitten | but you're going to have tuck with it to get it working |
| 21:27.17 | slav3_kitten | fuck* |
| 21:27.17 | ghost75 | i just used regular 48v 0,35A psu |
| 21:27.47 | slav3_kitten | i'm pretty sure the 78x1's supported 802.3af |
| 21:27.57 | ghost75 | yes they do |
| 21:30.19 | kikohnl | ghost75: are you sure the 7941 has the SIP software or does it have the SCCP, I had lots of trouble with a 7941G-GE converting it to SIP |
| 21:31.13 | ghost75 | i dont know what it has |
| 21:31.23 | kikohnl | Ah |
| 21:32.12 | ghost75 | i read something you need to first flash 7, then 8 etc |
| 21:32.40 | kikohnl | 7 then 8 then 9 |
| 21:34.11 | kikohnl | looks like the oldest firmware I have is 8.2-1S |
| 21:35.12 | ghost75 | me too |
| 21:36.37 | kikohnl | I had to wire the phone directly to the tftp boot server to get the reset to default to work, it took about 30 minutes. |
| 21:37.23 | ghost75 | mine doesnt get ip |
| 21:41.22 | lvlinux | you can convert POE to Cisco inline power with an ethernet cable wired in a non-standard way. |
| 21:41.44 | lvlinux | I do it with Cisco 79xx phones all the time. |
| 21:43.14 | ghost75 | http://www.adrianandgenese.com/blogger/2011/02/16/how-to-upgrade-or-convert-a-cisco-ip-79xx-7940-7960-794x-796x-797x-phone-to-sip-or-sccp/ |
| 21:43.28 | ghost75 | hold the # key till the extension lights blink amber…then press 3491672850*#. <- huh? |
| 21:45.09 | lvlinux | wow |
| 21:45.53 | ghost75 | cisco writes: hold the # key till the extension lights blink amber…then press 3491672850*#. |
| 21:46.24 | kikohnl | unplug the power or power old the # key while plugging in the power/poe then when amber do the code |
| 21:46.41 | kikohnl | ~/old/poe/ |
| 21:47.24 | kikohnl | i think the 7941G-GE code is 123456789*0# |
| 21:47.53 | ketas | that subst made no sense |
| 21:47.56 | igcewieling | Why could they not use something simple like 322867973738# |
| 21:48.43 | kikohnl | lunch time
. back in a few |
| 21:48.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:50.49 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 21:52.09 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:74be:802b:7763:2127) |
| 21:55.22 | *** join/#asterisk bchia (~Adium@nat/digium/x-dbbapbtjsejjxsjz) |
| 21:55.59 | dpilon | ghost75 are you still trying to convert the firmware to the latest 9.3 firmware ? |
| 21:56.57 | ChannelZ | I think he's trying to get it to do anything at this point |
| 21:57.40 | dpilon | reading the info from cisco..it will tell you that i think you need 8.5.2 before trying anything higher |
| 21:58.08 | ghost75 | wishes he was at this point |
| 21:58.22 | dpilon | what version are you now ? |
| 21:58.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:58.39 | ghost75 | version 1.brick |
| 21:58.42 | dpilon | i just did a 7961G 2 weeks ago |
| 22:00.57 | *** join/#asterisk macroevolve (~macroevol@c-67-169-148-235.hsd1.ca.comcast.net) |
| 22:01.04 | dpilon | at this point i go from SIP to SCCP without problems...just by changing the SEP<mac>.xml file |
| 22:01.05 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 22:01.19 | dpilon | crap...even chan-sccp works :) |
| 22:01.28 | macroevolve | hi - i've hired someone in the Philippines to make some outbound calls for me to the US. Unfortunately his connection isn |
| 22:02.07 | jmetro | i country blocked the philippines |
| 22:02.14 | dpilon | nice |
| 22:02.35 | dpilon | ghost75 pm me if you want |
| 22:02.55 | macroevolve | hi - i've hired someone in the Philippines to make some outbound calls for me to the US. Unfortunately his connection isn't so great. I was trying to figure out the best codec to use for poor internet connections. I came across SILK, G729A, Speex - was wondering what you guys thought would be the best codec to use (sounds quality isn't as important as being able to make an intelligible |
| 22:02.56 | macroevolve | conversation). |
| 22:05.04 | ghost75 | now i have other dhcp server like in this article |
| 22:05.21 | ghost75 | its doing dhcp discover but doesnt take the address |
| 22:07.06 | dpilon | i used my * box for dchp..i could then monitor the logs |
| 22:07.30 | ghost75 | had also wireshark on |
| 22:08.02 | dpilon | ghost75 what version of the firmware do you haev available ? |
| 22:08.13 | ketas | [Feb 8 20:19:38] NOTICE[52675] chan_sip.c: Call from '' (96.8.118.203:5060) to extension '672407875' rejected because extension not found in context 'sip-default-incoming'. |
| 22:08.16 | ketas | here we go :) |
| 22:08.23 | ketas | now i need to handle those for fun |
| 22:08.56 | ghost75 | ftp://ftp.itl.ua/pub/cisco/ip-7900ser/ |
| 22:09.12 | dpilon | send me pm...it will be faster |
| 22:09.40 | ghost75 | is more problem now to get ip address |
| 22:09.57 | ghost75 | no ip, no tftp, no firmware |
| 22:10.24 | dpilon | do you see it in logs that it is trying ? |
| 22:10.47 | ghost75 | yeah it sends discover |
| 22:11.38 | ghost75 | would it be possible that it doesnt get enough power? |
| 22:12.59 | dpilon | how much is it getting ? |
| 22:13.48 | ghost75 | 48v 0,3A psu but the plug is not maybe not having full contact (hard to see) |
| 22:14.05 | dpilon | should be fine |
| 22:14.29 | ghost75 | i measured its pulling 3.2w |
| 22:15.38 | kikohnl | My cisco psu is 0.38a 48v |
| 22:16.12 | jmetro | My queue currently dials one person all the time, and then another set member if that times out, and then a third , etc... would this be "Ringall" with penalties established, or Round Robin? |
| 22:16.25 | *** part/#asterisk macroevolve (~macroevol@c-67-169-148-235.hsd1.ca.comcast.net) |
| 22:17.14 | jmetro | Based on the wiki, it looks like ringall with penalties, but i'm sanity checking |
| 22:17.19 | jmetro | myself. |
| 22:21.17 | ghost75 | i tried now: 3491672850*# |
| 22:21.22 | ghost75 | now blinkyblinky red |
| 22:23.50 | ghost75 | display empty now |
| 22:24.00 | dpilon | from what i see...the 349 kindof erases the memory :) |
| 22:24.34 | ghost75 | it tries to flash |
| 22:25.15 | dpilon | ok..when you reboot...use the 123456789*0# |
| 22:25.32 | ghost75 | i think now its dead |
| 22:25.40 | dpilon | nope |
| 22:25.48 | dpilon | i had same problem |
| 22:25.57 | ghost75 | no other lights now on lol |
| 22:25.57 | kikohnl | so have I |
| 22:26.04 | ghost75 | something its still doing |
| 22:26.22 | lvlinux | wow i've never seen one get that bad lol... |
| 22:26.49 | ghost75 | hey |
| 22:26.59 | ghost75 | now i think it took the address |
| 22:27.16 | dpilon | ghost75 again....you will need firmware 8.5.2 to get 9.3 working |
| 22:27.35 | ghost75 | yesyes later |
| 22:28.20 | ghost75 | i can ping the phone now but its still doing stuff, lets wait |
| 22:28.56 | dpilon | is it pulling anything from tftp ? |
| 22:29.49 | ghost75 | no clue, thats just a all in one windows dhcp/tftp tool |
| 22:30.51 | ghost75 | just frightens a bit that the display is all black |
| 22:31.16 | dpilon | try to see if you can get logs..this way you can see if it is pulling files |
| 22:32.23 | dpilon | so where is your * server setup ? |
| 22:32.24 | *** join/#asterisk mattsl (~user@74.223.143.244.nw.nuvox.net) |
| 22:33.55 | ghost75 | no tftp log |
| 22:34.01 | ghost75 | maybe xml is wrong |
| 22:34.38 | dpilon | you have problem tftp-server option pointing to you server |
| 22:34.45 | dpilon | problem/proper |
| 22:35.17 | ghost75 | its like here http://www.adrianandgenese.com/blogger/2011/02/16/how-to-upgrade-or-convert-a-cisco-ip-79xx-7940-7960-794x-796x-797x-phone-to-sip-or-sccp/ |
| 22:35.38 | dpilon | ghost75 do you not have an asteisk server with linux setup ? |
| 22:36.00 | ghost75 | why do you ask about asterisk now? |
| 22:36.23 | dpilon | because...you would have linux..and dhco/tftp server |
| 22:38.50 | dpilon | also the xml has the proper <loadInformation> for the firmware you are trying to install (not sure it matters at this point as you just told it to flash manually) |
| 22:40.55 | ghost75 | manual? |
| 22:41.57 | dpilon | doing the code will try to forces a flash |
| 22:42.08 | *** join/#asterisk MarKsaitis (~MarKsaiti@5e0533a1.bb.sky.com) |
| 22:42.10 | dpilon | so...anything on the screen yet |
| 22:42.16 | ghost75 | no |
| 22:42.46 | ghost75 | maybe it doesnt like the fw |
| 22:42.58 | dpilon | so..back to linux...do you it installed ? |
| 22:43.42 | ghost75 | it didnt get even ip from linux |
| 22:43.42 | dpilon | having tftpd server from linux box..you can see if it tries to download |
| 22:43.57 | mattsl | Do you already have an asterisk box? |
| 22:46.06 | ghost75 | XMLDefault.cnf.xml is this exact name? |
| 22:47.02 | ghost75 | maybe should try older fw first |
| 22:48.37 | *** join/#asterisk mintos (~mvaliyav@114.143.45.62) |
| 22:49.26 | dpilon | is the 7941G entry listed in your XMLDefault file ? |
| 22:50.48 | ghost75 | i had before |
| 22:51.10 | dpilon | pastebin your file somewhere |
| 22:51.54 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:51.55 | ghost75 | <loadInformation>SIP41.9-3-1-1SR1-1S</loadInformation> |
| 22:52.01 | ghost75 | the rest is like in article above |
| 22:52.33 | dpilon | again...unless you have 8.5.2 already on..you will NOT get 9.3 to flash |
| 22:53.20 | ghost75 | which one to take then |
| 22:54.09 | dpilon | you will need to flash phone to 8.5.2 first |
| 22:54.30 | ghost75 | i dont have |
| 22:54.37 | ghost75 | only older |
| 22:55.03 | dpilon | google...try to find it |
| 22:55.17 | dpilon | but..that is if you need the 9.3 firmware |
| 22:56.21 | ghost75 | maybe you need something more older to put 8.5.2 |
| 22:57.56 | ghost75 | need 8.3(3) to 8.5(2) to get to 8.5(3)+ |
| 22:58.41 | dpilon | good luck :) dinner time |
| 23:00.56 | Kobaz | mm |
| 23:01.08 | Kobaz | anyone know if there's a way you can track call forwarding on a polycom |
| 23:01.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.36 | Kobaz | ie: when someone sets up a call forward locally on the polycom, can you have the phone post the request to the server saying 'hey im forwarded' |
| 23:02.33 | Kobaz | or i guess even better, write a microbrowser app that does all the forwarding and dont use the phone |
| 23:03.53 | WIMPy | Or just use a speeddial? |
| 23:07.27 | dpilon | ghost75 yes....8,3 to 8.5.2 to 9.3 should be fine |
| 23:08.21 | ghost75 | its in a kind of loop with blank display |
| 23:08.35 | ghost75 | do you think i can power off? |
| 23:11.42 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 23:12.44 | ketas | hahaha... now i handle those autodialers with MixMonitor -> Ringing -> Wait |
| 23:12.47 | ketas | we'll see |
| 23:18.09 | ketas | i hope Wait(31557600) will be enough |
| 23:23.29 | ghost75 | -Additional option, enter 150 in the first box, then 0x0101A8C0 in the second box (this number is 192.168.1.1 in Hexadecimal). |
| 23:23.31 | ghost75 | not really is it? |
| 23:24.48 | WIMPy | What? The conversion? |
| 23:25.19 | ghost75 | yeah |
| 23:25.26 | WIMPy | sure |
| 23:25.35 | ghost75 | wrong or true |
| 23:26.05 | WIMPy | The first statement is correct. |
| 23:26.34 | ghost75 | 0x0101A8C0 equals 192.168.1.1 ? |
| 23:26.46 | WIMPy | yes |
| 23:27.14 | WIMPy | Depending on your hosts byte sex, off course. |
| 23:28.09 | *** join/#asterisk din3sh (~din3sh@41.136.81.209) |
| 23:28.24 | din3sh | hello all |
| 23:30.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nmldjvcdlieeqcku) |
| 23:31.08 | ghost75 | Feb 09 00:30:23 server atftpd[9234.-1218823312]: Serving term41.default.loads to 192.168.2.241:50647 |
| 23:31.11 | ghost75 | fuck yeah! |
| 23:31.20 | ghost75 | (sorry) |
| 23:33.13 | ketas | have you shouted it over the office? |
| 23:33.20 | ketas | ever |
| 23:33.23 | ketas | :) |
| 23:33.37 | ghost75 | not yet |
| 23:34.09 | ghost75 | display is back |
| 23:36.35 | ghost75 | now see only clock and "unprovisioned" |
| 23:39.37 | din3sh | create the SEPmac.cnf.xml |
| 23:39.40 | din3sh | should be ok |
| 23:39.49 | din3sh | you been at it all day? |
| 23:39.50 | din3sh | :p |
| 23:39.58 | ghost75 | no comment lol |
| 23:40.35 | din3sh | I forgot |
| 23:40.41 | din3sh | to tell you |
| 23:41.12 | din3sh | with the new firmware, callfwd softkey does not seem no work |
| 23:41.14 | din3sh | :P |
| 23:41.52 | din3sh | I guess the new firmware is optimized for UCM |
| 23:42.05 | ghost75 | <processNodeName>10.0.0.9</processNodeName> this is * ? |
| 23:43.32 | *** join/#asterisk mintos (~mvaliyav@114.143.43.15) |
| 23:43.46 | kikohnl | yes |
| 23:45.33 | Kobaz | how would i turn off sending the diversion header when you get a sip 402 moved temporarily |
| 23:45.59 | kikohnl | ghost75: here is my working SEP file for 9.3.1 http://pastebin.com/tWeGA7RP |
| 23:46.16 | WIMPy | Kobaz: core show function CALLERID |
| 23:48.44 | Kobaz | i don't see diversion as an option for callerid |
| 23:49.06 | Kobaz | is that in 10 or 11? |
| 23:49.12 | Kobaz | this is 1.8 |
| 23:49.31 | Kobaz | i can use SIPRemoveHeader |
| 23:50.14 | WIMPy | Kobaz: Sorry. It's REDIRECTING. |
| 23:52.18 | Kobaz | that's not in there either |
| 23:52.57 | WIMPy | Maybe it was new in 10. |
| 23:53.15 | Kobaz | i'll just sip remove header on Diversion: |
| 23:53.50 | WIMPy | Time to upgrade. |
| 23:53.55 | Kobaz | yeah |
| 23:53.56 | Kobaz | not quiet yet |
| 23:54.00 | Kobaz | need to port all my crap to 11 |
| 23:54.28 | Kobaz | patch -p1 < kobazcrap.patch |
| 23:54.42 | Kobaz | and then watch everything break |
| 23:54.43 | ChannelZ | crap fail at offset 303 |
| 23:54.49 | Kobaz | heh yeah exactly |
| 23:55.10 | Kobaz | i think half the stuff i can safely not port though |
| 23:55.12 | WIMPy | Don't forget to flush. |
| 23:55.16 | Kobaz | a lot of it is bug fixes and whatnot |
| 23:55.18 | Kobaz | workarounds |
| 23:55.29 | Kobaz | stuff that's been already fixed in the newer versions |
| 23:55.41 | Kobaz | and i have features that are now in 11, so i dont have to port that |
| 23:58.23 | Kobaz | oh noes |
| 23:58.38 | Kobaz | SIPRemoveHeader can only remove headers added by SIPAddHeader |
| 23:59.02 | ghost75 | the phone monitors its xml files all the time? |
| 23:59.28 | WIMPy | Didn't 1.8 already have CALLERID(rdnis)? |
| 23:59.45 | Kobaz | rdnis? is that what diversion uses |
| 23:59.57 | Kobaz | RDNIS |