| 00:02.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.24.251) |
| 00:20.28 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 00:21.00 | carrar | http://i.imgur.com/yCK53At.jpg |
| 00:21.09 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:21.09 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:22.39 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 00:22.45 | *** join/#asterisk vastersk (~vinscentp@124.6.136.142) |
| 00:23.59 | vastersk | are newbees to asterisk welcome here? :) |
| 00:25.07 | vastersk | thank it's yes... ^_^ |
| 00:27.16 | ChannelZ | Sure |
| 00:28.23 | vastersk | great... thanks! |
| 00:29.50 | vastersk | i'll be here most of the time then... ned to go atm... |
| 00:30.19 | *** join/#asterisk gown (~phonetic@gateway/tor-sasl/martinphone) |
| 00:43.55 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 01:07.54 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 01:12.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:19.49 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:23.39 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:38.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:38.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:33.37 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 03:02.48 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 03:20.33 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 03:22.49 | *** join/#asterisk dijib (~dijib@67-204-199-13.eastlink.ca) |
| 03:24.20 | dijib | hows * doing this evening?! |
| 03:24.56 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 03:29.42 | *** join/#asterisk nightrid3r (~kvirc@62.205.85.188) |
| 03:31.52 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:42.36 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 03:44.54 | *** join/#asterisk gajini (~gajini@61.12.12.132) |
| 03:55.28 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 04:04.12 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.159) |
| 04:04.23 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 04:09.26 | *** join/#asterisk jsjc (~Adium@149.Red-2-136-113.dynamicIP.rima-tde.net) |
| 04:26.55 | *** join/#asterisk studybo__ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 04:59.38 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.61) |
| 05:00.04 | ChannelZ | dijib: just ! |
| 05:00.15 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 05:00.42 | ruben231 | hi guys how do i correct call delay on asterisk like 2.5 seconds |
| 05:05.02 | ruben231 | any idea guys..? |
| 05:05.15 | ChannelZ | Define call delay |
| 05:05.31 | ChannelZ | Audio latency, or something about dialing? |
| 05:06.52 | ruben231 | ChannelZ: when i call someone using asterisk server the one who answers when i speak he detects delay on my voice before it arrives to him 2-3 seconds |
| 05:09.01 | ChannelZ | What is the full call path (SIP phone -> Asterisk -> ?? or whatever) and what codec is being used? |
| 05:09.07 | igcewieling | ruben231: a 2-3 second delay is unusual, you should expect the delay of about the same as a cell phone. Call someone (on a different carrier) who is physically close enough you can hear them outside of the phone and notice the delay. |
| 05:09.47 | igcewieling | that is about the amount of delay to expect. on voIP. If you have more delay than a cell then you have an interesting task as a 2-3 second delay is unusual. |
| 05:10.03 | igcewieling | there is more delay on cell to cell calls then you might think |
| 05:10.18 | ChannelZ | Congradulations, you have a time machine! Why can't you just be happy? |
| 05:16.39 | apb1963 | Hey gang... I can't seem to get any recordings using feature codes (*1). The logs seem to indicate recording has been triggered, but I don't see any files in /var/spool/asterisk/monitor/ Any clues? Here's a complete call: http://ix.io/4o4 |
| 05:19.04 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 05:23.32 | *** join/#asterisk Wiretap_ (~wiretap@unaffiliated/wiretap) |
| 05:23.44 | ruben231 | ChannelZ: yes..sip calls ---> asterisk -----> phone line ( using voip) |
| 05:29.52 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 05:44.40 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 05:53.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:56.06 | lorsungcu | I've also got a delay question |
| 05:56.10 | lorsungcu | although with recordings |
| 05:56.21 | lorsungcu | ~5-15s delay between channels on recordings |
| 05:56.29 | lorsungcu | using mixmonitor |
| 06:03.44 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 06:08.00 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 06:08.13 | ChannelZ | A meteor crashes in Russia? http://t.co/xpfW2XFQ |
| 06:15.26 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 06:35.54 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.61) |
| 06:36.25 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:45.30 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 06:53.31 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:55.53 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 07:03.28 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:05.28 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 07:06.07 | Changos | [TK]D-Fender: really easy configure dial plan on E1 card, xD |
| 07:08.39 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:22.15 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-tjhvbjazcgobrknj) |
| 07:23.10 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:30.11 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:44.20 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 07:58.03 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 08:11.26 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:11.32 | bombev | hi all |
| 08:26.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:52.27 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:55.06 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 08:56.47 | *** join/#asterisk aurs (~aurs@84.49.69.110) |
| 09:06.29 | *** join/#asterisk dgeary2 (~david@120.21.79.246) |
| 09:08.59 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.162.48) |
| 09:12.25 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.162.48) |
| 09:16.53 | *** join/#asterisk vlad_sta_ (~vlad_star@178.176.162.48) |
| 09:19.53 | *** join/#asterisk ghost75 (~ghost75@dslb-088-064-208-056.pools.arcor-ip.net) |
| 09:23.24 | *** join/#asterisk Vince-0 (~vincent@105-236-84-112.access.mtnbusiness.co.za) |
| 09:27.09 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 09:27.35 | *** part/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 09:28.00 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 09:37.31 | *** join/#asterisk James87 (~wiv@unaffiliated/james87) |
| 09:38.01 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 09:41.36 | linocisco | how/and what to study for dCAA exam at home without taking course? I have only GXW410x Series IP Analog Gateway, a wifi router WRT54G, and analog desktop phones. |
| 09:42.38 | linocisco | HandyTone 702/704 (HT702/704) ATA is also with me |
| 10:03.47 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 10:13.43 | linocisco | how/and what to study for dCAA exam at home without taking course? I have only GXW410x Series IP Analog Gateway, a wifi router WRT54G, and analog desktop phones. |
| 10:13.46 | linocisco | HandyTone 702/704 (HT702/704) ATA is also with me |
| 10:13.48 | *** join/#asterisk dxtr (~dxtr@unaffiliated/dxtr) |
| 10:13.53 | dxtr | Ohai guys |
| 10:14.54 | dxtr | I was wondering, I've set up a PBX and it works with two of my phones. WIth the third one (Telephone.app on OSX) I only get <--- SIP read from UDP:83.241.162.99:52411 ---> jaK <-------------> when I want to call internally and the call just hangs up |
| 10:15.30 | dxtr | Why is that? |
| 10:18.08 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 10:23.30 | k610 | i got constant doing dnsmgr_lookup for ... is there something with my dns ? |
| 10:30.00 | *** part/#asterisk James87 (~wiv@unaffiliated/james87) |
| 10:40.24 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 10:42.27 | *** join/#asterisk willcooke (~will@canonical/willcooke) |
| 10:44.05 | willcooke | Hi folks, I'd like to change the dial tone frequency on a dahdi analogue channel. It sounds exactly like the normal UK dialtone, but I want people to know they are behind a PBX by using a different tone |
| 10:44.27 | willcooke | I've found the place in tonezone.c in the dahdi code, so I could change the source and compile up a new version |
| 10:44.55 | willcooke | but I've also seen hints that I can specify the dial tone at run time. |
| 10:45.03 | willcooke | Any pointers on how to do that? |
| 10:48.09 | kaldemar | willcooke: see indications.conf |
| 10:49.01 | willcooke | kaldemar: perfect, thanks a lot |
| 10:54.20 | *** join/#asterisk v0lZy (~chatzilla@84-255-194-41.static.t-2.net) |
| 10:55.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 10:57.48 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 10:57.54 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 11:01.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:08.32 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 11:10.50 | danfromuk | Does anyone know how to disable SIP ALG on a BT Business Hub? |
| 11:11.08 | linocisco | where can I find official asterisk videos on youtube? |
| 11:21.19 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 11:27.24 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 11:41.46 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 12:00.16 | Greenlight | I guess this is what happens why an admin has too much spare time: traceroute 216.81.59.173 |
| 12:05.05 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 12:07.12 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 12:22.31 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 12:33.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:37.39 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:46.35 | *** join/#asterisk FLeiXiuS` (~FLeiXiuS@unaffiliated/fleixius) |
| 13:03.27 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:03.55 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:12.26 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 13:30.53 | dxtr | Bah |
| 13:30.56 | dxtr | I'm giving up on this soon |
| 13:31.06 | dxtr | I can get either inbound or outbound calls to work |
| 13:31.08 | dxtr | Not both |
| 13:31.13 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 13:33.26 | ectospasm | dxtr: they should be independent |
| 13:33.37 | ectospasm | ...but, alas, I've run out of time. |
| 13:37.01 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:37.34 | dxtr | ectospasm: I think so too :P |
| 13:47.24 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 13:55.07 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:55.21 | *** join/#asterisk serafie (~erin@nat/digium/x-bfohgmeqyflcnfam) |
| 13:58.23 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 14:04.46 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:14.30 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:16.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:16.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:23.37 | Katty | hello my asterisk does not work at all how to fix?? plz answer is urgent thx. |
| 14:24.05 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 14:24.51 | jmetro | katty: make sure your toaster is enabled and set to "very dark" or the bagel daemons wont initialize |
| 14:26.25 | *** join/#asterisk k612 (~K610@cred.epid.ucl.ac.be) |
| 14:27.36 | Katty | :> |
| 14:27.45 | Katty | awards 5 points to house jmetro |
| 14:28.21 | Greenlight | I want points too |
| 14:34.29 | glaz | I want a cupcake. |
| 14:40.48 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:40.49 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 14:41.10 | saint_ | hi all |
| 14:41.24 | saint_ | anyone using google voice around here ? :D |
| 14:42.58 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 14:44.05 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 14:49.44 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 14:50.37 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 14:51.08 | rolandow | hi! i am having issues with an internet provider who's probably blocking traffic on 5060 .. (KPN, the netherlands) .. can i change this port for a specific user?? |
| 14:51.22 | rolandow | in my asterisk box?? so could i define this for one sip account only? |
| 14:51.29 | igcewieling | rolandow: no. |
| 14:51.58 | rolandow | hm.. to bad.. |
| 14:52.02 | igcewieling | rolandow: you'll need to do complicated iptables hackery or install a sip proxy in front of Asterisk. |
| 14:52.07 | rolandow | so the only way is a proxy then i suppose? |
| 14:52.29 | rolandow | right |
| 14:52.46 | rolandow | i started with forwarding 5060 to the phone.. maybe that'll help .. |
| 14:52.54 | igcewieling | rolandow: the port= option in peers is the remote port to connect to, not the local port to listen on (just mentioning it so you don't get confused later) |
| 14:53.05 | rolandow | but i've had more issues with this modem/router before |
| 14:53.51 | rolandow | yes.. so the port option would only make sense for a peer. |
| 14:56.07 | rolandow | so would a proxy be very complicated? |
| 14:56.21 | rolandow | have you done that igcewieling ? |
| 14:56.32 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 14:57.45 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-okccstbnlemvepno) |
| 14:57.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:11.28 | saint_ | ok, so i can totally reproduce my issue. |
| 15:15.44 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:f8f0:a5fb:cc76:8999) |
| 15:16.10 | saint_ | If I make 10 calls non-answered to my digium phone, the whole asterisk box goes down |
| 15:16.15 | saint_ | nobody can call each other anymore |
| 15:16.23 | *** part/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 15:16.24 | saint_ | what the heck ? |
| 15:19.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 15:23.07 | saint_ | can someone tell me why i can t even call my own voicemail..? http://pastebin.com/q1KQ60fH |
| 15:23.40 | saint_ | if I start asterisk, it works perfectly a zillion time |
| 15:23.46 | saint_ | my asterisk is connected to google voice |
| 15:23.59 | saint_ | if I call from my cell to my google voice, my digium D70 rings |
| 15:24.09 | saint_ | i hang up before it answers |
| 15:24.17 | saint_ | i do this 10 times. it works 9 times. |
| 15:24.25 | saint_ | at the 10th time, i can't do anything anymore with my asterisk |
| 15:24.36 | saint_ | internal phones can't call each other, nor the voicemail |
| 15:24.41 | saint_ | can't make outgonig calls anymore |
| 15:24.44 | saint_ | can't receive incoming calls |
| 15:25.06 | lorsungcu | pb sip.conf |
| 15:25.50 | *** join/#asterisk serafie (~erin@nat/digium/x-qbgvlbtyoooxakqi) |
| 15:26.18 | jsmith | saint_: Anything weird in the Asterisk logs? Any warnings or errors on the Asterisk CLI? |
| 15:26.36 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:26.36 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:26.53 | *** join/#asterisk lorsungcu (~anonymous@23.30.228.69) |
| 15:28.04 | malcolmd | saint_: i can't reproduce what you reported to mjordan. please contact our Support department directly so that they can see what's going on. they can be telephoned at +1 256 428 6000 or you can open a ticket online - http://www.digium.com/en/users/support-create-a-case/ |
| 15:28.53 | mjordan | well, I just noticed the statements, sanit_ didn't really report it to me :-) |
| 15:28.57 | mjordan | saint_ |
| 15:29.01 | saint_ | jsmith: nop |
| 15:29.42 | saint_ | malcolmd / mjordan what I just wrote ? or are you guys talking about something else ? |
| 15:30.00 | malcolmd | mjordan: ahh |
| 15:30.33 | malcolmd | saint_: i'm referring to the comment that calling a digium phone 10 times and not answering it breaks asterisk. |
| 15:34.50 | saint_ | lorsungcu: http://pastebin.com/SXNkLDNg |
| 15:35.20 | Qwell | saint_: Can you reproduce it without using chan_gtalk? |
| 15:37.30 | saint_ | Qwell: hold on |
| 15:45.12 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 15:46.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:46.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:46.54 | *** join/#asterisk lorsungcu_ (~anonymous@12.40.176.42) |
| 15:48.30 | rolandow | so can somebody recommend a sip proxy or anything else to make sure that an user at home can use an other port than 5060? |
| 15:49.40 | saint_ | malcolmd: is your asterisk connected to google voice ? |
| 15:49.56 | malcolmd | nope |
| 15:49.56 | saint_ | Qwell: i will try that right now. but i reproduce when i call my google voice from outside. |
| 15:50.03 | saint_ | malcolmd: that is where the issue is |
| 15:50.16 | saint_ | malcolmd: i call my google voice, my D70 rings, I hang up instead of picking up |
| 15:50.24 | saint_ | malcolmd: it works 9 times |
| 15:50.28 | malcolmd | if you replace the d70 with something else does the same thing happen? |
| 15:50.29 | saint_ | 10 times, nothing works anymore |
| 15:51.09 | saint_ | malcolmd: i am not sure the D70 is really important in the picture.. i think it s the fact that i call GV , asterisk picks it up , whatever rings i do not pickup , then start again |
| 15:51.13 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 15:51.33 | malcolmd | if the d70's not important in the picture then it's not a d70 problem. |
| 15:51.34 | saint_ | malcolmd: i can try to divert the GV calls to another phone if you want |
| 15:52.07 | saint_ | malcolmd: agreed. it's an asterisk problem. i will try right now redirecting calls to another phone |
| 15:59.43 | jmetro | anyone familiar with queues that can answer a logic question? |
| 16:00.12 | saint_ | malcolmd: cant reproduce with another phone. let me try again putting the D70 in the picture |
| 16:00.47 | navaismo | jmetro, ask, someone maybe has the answer |
| 16:01.06 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:01.07 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:01.58 | jmetro | I'm trying to do a top-down queue. I figured that Ringall strategy with penalties with work, but i cant get the queue to timeout and move on to the next phone. |
| 16:02.24 | jmetro | will work* |
| 16:03.53 | saint_ | malcolmd: if I put the D70 in the picture, I can reproduce. |
| 16:03.58 | saint_ | that might be totally random |
| 16:04.03 | saint_ | i catched a couple of error message |
| 16:04.05 | saint_ | let me pastebin |
| 16:05.04 | malcolmd | what did you use instead of the d70 to not reproduce? is your configuration identical for the other device, including dial plan? |
| 16:05.36 | saint_ | malcolmd: yes, all I did is replace the Dial(xxx) were xxx was the D70 to another SIP phone from another brand |
| 16:05.37 | saint_ | http://pastebin.com/uJGbQUgg |
| 16:05.44 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:05.47 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 16:05.58 | malcolmd | okay, back to my first statement. please contact our support department. |
| 16:06.03 | apb1963 | I can't seem to get any recordings using feature codes (*1). The logs seem to indicate recording has been triggered, but I don't see any files in /var/spool/asterisk/monitor/ Any clues? Here's a complete call: http://ix.io/4o4 |
| 16:06.09 | saint_ | now, I have a rtp.conf with rtpstart=10000 rtpend=10040 , would this be the issue ..? |
| 16:06.20 | navaismo | jmetro did you configure the timeout? |
| 16:06.51 | jmetro | Set timeout to 5 for testing, just kept ringing the first member, tho 2 were in the queue. |
| 16:07.11 | jmetro | it actually said "call timed out" [or similar] in the console, but just kept ringing the first member |
| 16:07.26 | saint_ | ha.. when the issue happens, I can see this on my console: Oh dear... we couldn't allocate a port for RTP instance '0x7f7f44004e08' |
| 16:07.37 | navaismo | jmetro, when you run: queue show <yourqueue> how looks the members status? |
| 16:07.38 | saint_ | is it because I do not have enought RTP ports ? |
| 16:07.49 | igcewieling | saint_: that would be the logical conclusion |
| 16:08.36 | jmetro | Navaismo: Both were fine.. let me setup the test again from scratch and i can post real output |
| 16:09.49 | navaismo | apb1963, this file exist and run without errors? /var/lib/asterisk/bin/one_touch_record.php |
| 16:09.50 | saint_ | igcewieling: i don t know how to read netstat , but does not this http://pastebin.com/wxUAaPC8 shows that ports 10k-10k40 are free to listen ? |
| 16:09.55 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 16:10.31 | apb1963 | navaismo: it definitely exists. |
| 16:10.32 | igcewieling | saint_: the ports are not allocated until a call happens so they would not show up as in use |
| 16:10.34 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 16:10.54 | apb1963 | navaismo: as for errors... all I know is what's in the log file |
| 16:11.02 | igcewieling | saint_: why don't you increase your port range? |
| 16:11.03 | apb1963 | navaismo: and I don't see any there so... |
| 16:11.20 | navaismo | apb1963, if you run with php filename? |
| 16:11.29 | apb1963 | navaismo: I don't follow |
| 16:11.32 | saint_ | igcewieling: because I want to understand first ... if I see foreign address being *:* doesn't it mean they are not in use and free ? in which case why would asterisk cry about it ? |
| 16:12.17 | navaismo | apb1963, run from linux shell: php /var/lib/asterisk/bin/one_touch_record.php and check for errors |
| 16:12.24 | apb1963 | k |
| 16:12.26 | saint_ | igcewieling: so if it's the issue, i should be able to reduce from 10k-10k40 to 10k-10k10 , and be able to reproduce in a shorter time.. agreed ? |
| 16:14.58 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 16:15.57 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 16:16.37 | apb1963 | PHP Fatal error: Call to a member function GetVar() on a non-object in /var/www/html/admin/modules/callrecording/bin/one_touch_record.php on line 262 |
| 16:18.42 | navaismo | apb1963, if you check the /var/log/httpd/error_log there is also the error when you call that script via asterisk? |
| 16:19.23 | apb1963 | no such path |
| 16:20.00 | apb1963 | it's apache2 here |
| 16:20.01 | apb1963 | checking |
| 16:20.08 | navaismo | apb1963, k, that is for centos apache |
| 16:20.13 | jmetro | Navaismo: It looks like it successfully hits the second member, but only if the first member is "unavailable" |
| 16:20.31 | saint_ | whoa it's getting bad...: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 16:20.57 | navaismo | jmetro, can you set both members available, made a call, run queue show and pb all? |
| 16:22.23 | teloniusz | hi |
| 16:22.41 | teloniusz | is there a way to check whether my trunk is registered in the dialplan? |
| 16:22.49 | jmetro | I've got three physical phones for testing, an outside caller, and an inside 126 and 127. 127 is set to Penalty 1 and 126 is set to Penalty 2, if i queue myself up, 127 rings indefinately regardless of my timeout, unless i hit the "ignore" or "reject" button |
| 16:23.12 | saint_ | malcolmd: igcewieling so i see the issue.. when i put a digium phone as destination, if i do not pickup, and hang up before the voicemail picks up , it keeps the udp ports in use after i hang up |
| 16:23.17 | jmetro | let me get a queue show. |
| 16:23.41 | teloniusz | I mean, my trunk for outbound calls registers to the outside and I want to check if it's registered already before attempting to dial. |
| 16:24.50 | *** join/#asterisk gtTuna (~Tuna@d149-67-253-75.try.wideopenwest.com) |
| 16:25.24 | apb1963 | navaismo: there are a few errors that appear like they would be related, but none from today... so I'm not sure, and I would guess they're prolly GUI related, so they're not the issue I don't think. http://ix.io/4pG |
| 16:26.15 | jmetro | http://pastebin.com/set5UQST |
| 16:26.27 | jmetro | the call history is from back when i was testing |
| 16:26.59 | jmetro | if [I] make 127 unavailable, the queue strategy works, but it never actually times out on its own from ringing an available member |
| 16:27.10 | navaismo | apb1963, i see, well basically you are triggering a script to monitor, not using asterisk native way so you may check that script first or use asterisk native way, if its a freepbx issue maybe check in #freepbx |
| 16:27.10 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:27.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:27.44 | apb1963 | navaismo: I don't believe it's a freepbx issue. |
| 16:28.11 | apb1963 | how do i go native? |
| 16:28.29 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 16:28.39 | navaismo | apb1963, using mixmonitor or monitor |
| 16:29.01 | apb1963 | i'm listening |
| 16:29.11 | apb1963 | how do i do that? |
| 16:29.55 | navaismo | apb1963, look at the features.conf, |
| 16:30.06 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 16:30.23 | navaismo | you are looking for automon and automixmon features |
| 16:30.53 | navaismo | but i suggest to check with the fpbx guys, since they are handling via script |
| 16:31.04 | apb1963 | ok |
| 16:31.18 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 16:31.32 | apb1963 | yeah those features aren't in the conf files. Just the apprecord |
| 16:31.41 | gtTuna | I know I'll probably just get suggested to upgrade to a new version of Asterksik (1.6.2.22), but has anyone seen an issue with channels not being cleared if the cdr_addon_mysql.so can't connect to a DB? |
| 16:31.42 | apb1963 | the one touch script |
| 16:31.57 | navaismo | jmetro, if you use linear or rrmemory without penalties the ring start tu jump? |
| 16:32.07 | apb1963 | let me ask over in #freepbx ... thanks |
| 16:32.19 | navaismo | apb1963, ;automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call! |
| 16:32.33 | navaismo | ;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call! |
| 16:33.11 | jmetro | Navaismo: I will test linear. Does linear always start in the same spot? [i require the first agent to always be first essentially] |
| 16:33.34 | navaismo | jmetro, linear: Rings interfaces in the order they are listed in the configuration file. Dynamic members will be rung in the order in which they were added. (new in 1.6) |
| 16:39.44 | AkkerKid | BZAO! |
| 16:40.10 | AkkerKid | anyone know if there's a date/time variable I can call within the dialplan? |
| 16:40.36 | WIMPy | STRFTIME |
| 16:42.31 | AkkerKid | That's not in the book... I wanna do something like Set(DeliveryDateRequested=${Current_Date}) |
| 16:42.47 | AkkerKid | how do I use STRFTIME? |
| 16:43.22 | WIMPy | core show function STRFTIME |
| 16:43.42 | WIMPy | Or check http://wiki.asterisk.org/ |
| 16:44.01 | jmetro | Linear seems to do the same thing, on timeout it rings 127 over and over |
| 16:44.12 | jmetro | if i ignore/reject/become unavailable it jumps to 126 |
| 16:45.12 | AkkerKid | thanks WIMPy |
| 16:45.19 | jmetro | 126 seems to jump back to 127 but 127 cant go back up to 126 |
| 16:47.48 | navaismo | jmetro, with linear and same penalties on all members? |
| 16:47.54 | jmetro | Yes |
| 16:48.21 | jmetro | I cant use RR memory unless the penalties make it behave similarly [top-down] |
| 16:48.42 | navaismo | jmetro, can you pb the call when enter in the queue with the sip debug? |
| 16:48.48 | jmetro | oh you mean penalty 1 and penalty 1 instead of penalty 1 and penalty 2? |
| 16:49.32 | navaismo | yes |
| 16:49.39 | navaismo | also try penalty 0 for all |
| 16:50.41 | jmetro | Linear umps properly when there is the same penalty, but i cant force it to always do the same member first if the agents login out of order? |
| 16:51.19 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 16:52.34 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-llgttbjomdavguzf) |
| 16:54.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:55.01 | saint_ | igcewieling: so isn't asterisk supposed to release the udp ports once they are not in use ? |
| 16:56.00 | navaismo | jmetro, you need dinamic members? |
| 16:57.06 | jmetro | My members need to be able to log in / out of the queue, but its always the same 3/4 people every day. |
| 16:57.46 | [TK]D-Fender | jmetro: "core show application AddQueueMember" |
| 16:58.37 | jmetro | i've just been doing queue add member from the CLI |
| 16:58.50 | eirirs | upgrading from asterisk 1.6 till 11 |
| 16:58.55 | eirirs | hope I don't break anything |
| 16:59.33 | gtTuna | eirirs, let me know how it goes. currently have a decent amount of 1.6 servers id like to upgrade |
| 16:59.34 | gtTuna | what OS? |
| 16:59.58 | saint_ | [TK]D-Fender: is asterisk supposed to release UDP ports once a call is done ? |
| 17:00.04 | jmetro | Ringall + Penalties was going to be the perfect solution since one member should always be rang first if she is present, and if not , second member if present, etc. |
| 17:00.14 | *** join/#asterisk Frojoe (~Frojoe@CPEbc1401dea6d3-CMbc1401dea6d0.cpe.net.cable.rogers.com) |
| 17:00.19 | jmetro | but there needs to be a timeout |
| 17:02.04 | igcewieling | eirirs: reading the UPGRADE*.txt files will reduce the chances of messing something up |
| 17:04.04 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:04.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:04.44 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 17:04.55 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 17:05.43 | navaismo | jmetro, i think the call is not passing to the next member because the penalty, if you set memeber with penalty 1 the second with penalty 2, call come in, reach member 1 timeout and call again member 1 because member still free |
| 17:06.01 | navaismo | i guess the call only reach the second member if member 1 is busy |
| 17:07.39 | igcewieling | saint_: I suspect recent Asterisk 1.8.x have an RTP port leak, but I've not had time to trace it down |
| 17:07.50 | saint_ | that is perfect for a DOS attack. |
| 17:08.00 | saint_ | igcewieling: i just traced it , and this is what i get |
| 17:08.05 | saint_ | igcewieling: the UDP ports are not freed |
| 17:08.16 | igcewieling | saint_: report it as a bug |
| 17:08.26 | saint_ | igcewieling: i can t loogin to issues.asterisk.org |
| 17:08.33 | WIMPy | It can be DOSed easily anyway.# |
| 17:08.50 | saint_ | WIMPy: well.. that would be the easiest way. dosing it with a phone. |
| 17:08.58 | saint_ | you don t even need a computer behing it. |
| 17:09.00 | jmetro | navaismo: can you think of a strategy or setting to configure that would be able to do a true top-down with timeout? |
| 17:09.01 | saint_ | just call and hang up |
| 17:09.02 | WIMPy | The current type of SIP scanners just congest the line. |
| 17:09.03 | saint_ | poof |
| 17:09.24 | igcewieling | no need to DDOs, just wait. |
| 17:09.34 | igcewieling | WIMPy: like "friendly scanner" |
| 17:09.38 | saint_ | igcewieling: yeah, i'll wait for someone to call , lol |
| 17:10.02 | WIMPy | I haven't seen "friendly scanner" for a looong time. |
| 17:10.21 | igcewieling | calling the SIP scanner "friendly scanner" is like calling Genghis Kahn a "friendly Mongol" |
| 17:10.26 | WIMPy | This year they use random strings for the UA. |
| 17:10.28 | navaismo | jmetro, linear but for static agents |
| 17:10.38 | saint_ | WIMPy: igcewieling I have to say though... something is weird... if I call asterisk from outside, and it rings a phone which is not a digium or it does not go to voicemail , then the port is freed |
| 17:10.40 | igcewieling | WIMPy: I saw it a few days ago though not against Asterisk |
| 17:10.42 | navaismo | jmetro, all same penalty |
| 17:10.48 | saint_ | it's only if the call goes to a digium phone or voicemail |
| 17:10.53 | WIMPy | It was freindly in that it didn't set up RTP and was rather harmless that way. |
| 17:11.27 | Greenlight | What might cause: [Feb 15 17:09:49] ERROR[25290][C-000000a8]: astobj2.c:155 INTERNAL_OBJ: bad magic number for 0x7fbcf402a268. Object is likely destroyed. |
| 17:11.37 | Greenlight | And should I be worries |
| 17:11.41 | Greenlight | *worried |
| 17:11.55 | Qwell | Greenlight: many things could cause it |
| 17:12.04 | jmetro | Navaismo: Now, i thought static agents were deprecated? |
| 17:12.28 | Qwell | jmetro: false |
| 17:12.34 | Greenlight | I was told they were too |
| 17:12.54 | jmetro | Oh ho, i see clearly now. |
| 17:13.08 | jmetro | Static agents defined in queues.conf rather than in agents.conf since agents.conf is what was deprecated. |
| 17:13.16 | Qwell | also false |
| 17:13.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 17:13.22 | Qwell | and those are not called agents. Those are queue members. |
| 17:13.29 | Qwell | Agents are very specifically one thing. |
| 17:13.43 | Greenlight | I asked about a problem with Agent/ channels in here, and was told they were depreached ;/ |
| 17:14.35 | Qwell | You were misinformed. |
| 17:15.09 | Qwell | That doesn't mean you should use them though. There is only one use-case where it is useful, and it's a silly one. |
| 17:15.15 | igcewieling | blames Canada |
| 17:15.21 | jmetro | The wiki is suggesting that i define a static queue member as Member => Agent/peername, is this correct, or do i use Sip/peername |
| 17:15.38 | Greenlight | Qwell: Which case is that ? |
| 17:15.47 | Qwell | The one that chan_agent exists for,. |
| 17:16.02 | Greenlight | I needed to keep users connected between calls |
| 17:16.10 | Qwell | Then you need to use agents. |
| 17:16.49 | Greenlight | They did seem quite buggy, and then I was told they were depreached, so have rewritten all my code to not use them |
| 17:16.56 | Greenlight | :( |
| 17:16.58 | Qwell | They are quite buggy. |
| 17:17.03 | Greenlight | Heh |
| 17:17.08 | Greenlight | Maybe not all in vain then :) |
| 17:17.09 | puzzled | hi |
| 17:17.34 | Greenlight | What I do no is put "agents" into a loop of hold music, and then use the AMI to Bridge them to customers as required |
| 17:24.07 | eirirs | igcewieling: I remember I had to edit an asterisk conf file to workaround a timing issue, dunno whether I need to do that same in v11 |
| 17:24.19 | eirirs | and I don't even remember what config file that was lol |
| 17:24.35 | Greenlight | internal_timing in asterisk.conf ? |
| 17:28.06 | eirirs | Greenlight: looks like it was not in asterisk.conf |
| 17:29.20 | jmetro | it even looks like isymphony supports the static queue members...freakin awesome |
| 17:31.32 | jmetro | hm.. you cannot log off a static queue member? |
| 17:32.03 | igcewieling | People often have one problem which they try to solve with queues, then they have two problems |
| 17:32.43 | jmetro | unfortunately as a call center, the problem was "we have a queue" and the solution was "we still have a queue" |
| 17:33.29 | Greenlight | What are you trying to acheive? |
| 17:34.16 | jmetro | top-down strategy, agents that login and logout |
| 17:34.44 | Greenlight | Why not use dynamic queue members ? |
| 17:34.51 | eirirs | Greenlight: ah that was in sip_general_custom.conf , session_timers=refuse and session_timeout=180 |
| 17:35.23 | jmetro | dynamic queue members apparently cannot support the top-down approach, the first queue member should always be rang first if she is present, same with second, etc. |
| 17:35.35 | eirirs | now I should be all set for upgrade I hope |
| 17:35.43 | jmetro | the linear strategy works, however, login/out of queue members breaks the order of queue members unless they are static. |
| 17:36.18 | Greenlight | jmetro: NOt sure why that wouldn't work, but you could just use two queues instead. Queue1 falls into Queue2 if no one available to answer |
| 17:37.06 | jmetro | That runs into a licensing issue, where my front end [isymphony] charges per queue. It is a totally legitimate strategy that should be natively supported without queue dumping =/ |
| 17:37.26 | Greenlight | I see |
| 17:43.16 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:48.15 | igcewieling | jmetro: there are many ways to get "queue-like" features without using queues or using queues in non-traditional ways |
| 17:50.22 | jmetro | guess its up to me to figure out how to get asterisk to work top-down then |
| 17:51.49 | igcewieling | for example, we set up a "queue" using dedicated lines on the phones and rolling the call over when a dest was busy. That is just one example of queue-like stuff but not using queues |
| 17:54.29 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 17:57.17 | navaismo | igcewieling, but you cant pull statistics right? In a call center statistics are essential. Many operators works by hours against productivity |
| 17:57.44 | igcewieling | navaismo: queue stats were not required in that case. |
| 17:57.54 | navaismo | i see |
| 18:08.18 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:13.34 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 18:14.03 | *** join/#asterisk job (job@instituut.net) |
| 18:14.50 | job | hi, i have a freepbx installation with asterisk in it, and i somehow want to make a small (php?) script that when with something like http://derp/connect.php?source=$my_mobile_number&destination=$my_friends_phonenumber |
| 18:15.17 | job | that the box first calls my cellphone, and after i pick up the call, calls my friends phonenumber and connects the two calls together |
| 18:16.00 | navaismo | php+callfile / php+ami |
| 18:16.04 | navaismo | can do that |
| 18:19.12 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 18:21.19 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 18:25.15 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:31.01 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 18:32.36 | *** join/#asterisk Sevet (steveayre@149.241.69.221) |
| 18:34.59 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
| 18:37.21 | *** join/#asterisk k612 (~K610@cable-78.29.241.186.coditel.net) |
| 18:47.50 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 18:49.22 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
| 18:51.04 | *** join/#asterisk din3sh (~din3sh@41.136.87.157) |
| 18:54.41 | din3sh | hello all |
| 18:55.03 | din3sh | I tried to set crc4 on my dahdi/system.conf |
| 18:55.11 | din3sh | here is what i get once i start dahdi |
| 18:55.33 | WIMPy | What did the telco say? |
| 18:55.49 | din3sh | kernel: Uhhuh. NMI received for unknown reason a1 on CPU 0. |
| 18:55.50 | din3sh | kernel: You have some hardware problem, likely on the PCI bus. |
| 18:55.50 | din3sh | kernel: Dazed and confused, but trying to continue |
| 18:56.35 | din3sh | i cant request telco intervention directly, asked client to contact em |
| 18:56.38 | WIMPy | That's a hardware thing. And I wouldn't relate it to dahdi. |
| 18:56.59 | din3sh | telco said they tested the isdn line , its ok |
| 18:57.23 | din3sh | they don't wanna understand the term monitor the line rather than test |
| 18:57.33 | WIMPy | If dmesg doesn't provide anything better, maybe the BIOS error log will. |
| 18:58.08 | din3sh | i have contacted digium support |
| 18:58.17 | din3sh | reviewing the logs |
| 18:58.21 | WIMPy | Whaterver test means. Last week BT thought that calling your number and hear ringing is an adequate test. |
| 18:59.03 | din3sh | client wants me to set up another * box |
| 18:59.16 | din3sh | he seems to think the box is the problem |
| 18:59.26 | WIMPy | Might be worth to take that server apart and put it back together again. If you're lucky it's a bad connection somewhere. |
| 18:59.46 | WIMPy | It might well be. |
| 19:00.08 | din3sh | is lenght of cable an issue? |
| 19:00.21 | WIMPy | Which cable? |
| 19:00.35 | din3sh | both the staright cable from modem to * |
| 19:00.42 | din3sh | or cross over from * to nortel |
| 19:01.05 | WIMPy | The S2m can have 250m. |
| 19:01.26 | din3sh | both are 10mts long |
| 19:01.58 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 19:02.54 | din3sh | badly seated card in the pci bus? |
| 19:03.09 | WIMPy | >>Might be worth to take that server apart and put it back together again. If you're lucky it's a bad connection somewhere. |
| 19:03.33 | din3sh | yeah |
| 19:03.39 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 19:03.45 | din3sh | installing another server on monday |
| 19:03.49 | din3sh | hopefully |
| 19:04.57 | WIMPy | Probably the best thing to do at this stage. |
| 19:05.16 | din3sh | system getting crazier, nortel extension could not hear pstn caller talking |
| 19:05.57 | WIMPy | Only one way? |
| 19:06.22 | din3sh | apparently |
| 19:06.57 | WIMPy | Scary. |
| 19:07.03 | din3sh | lol |
| 19:07.26 | din3sh | i ran the autosupport script from digium and sent them |
| 19:08.02 | WIMPy | At leat it's clear that it happens inside the box. |
| 19:08.29 | WIMPy | Unless the call was never connected. |
| 19:09.37 | din3sh | going to try a different brand of server with another centos version |
| 19:11.52 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 19:12.49 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
| 19:30.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:31.27 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 19:34.08 | navaismo | ~itsp-en |
| 19:34.17 | navaismo | ~itsp |
| 19:34.17 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:34.26 | navaismo | ~itsplist-us |
| 19:34.26 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:36.40 | igcewieling | din3sh: does "ifconfig" show any dropped packets? |
| 19:41.14 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:46.39 | *** join/#asterisk lorsungcu_ (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 19:59.21 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 20:01.20 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 20:01.29 | slicknick5181 | Hello all |
| 20:02.55 | slicknick5181 | I'm trying to set up a dialplan that allows the north american NXXNXXXXXX and 1NXXNXXXXXX and also the same with a 9 in front--each send a different caller ID out |
| 20:03.49 | igcewieling | slicknick5181: set up exten lines for each pattern and set the callerid there. |
| 20:07.10 | slicknick5181 | igcewieling: I did hold on I will pastebin it to ya |
| 20:09.40 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat12.beelinegprs.ru) |
| 20:10.49 | slicknick5181 | igcewieling: http://pastebin.com/A9pPBcKS |
| 20:11.24 | [TK]D-Fender | slicknick5181: "|" is not part of a dialplan pattern |
| 20:11.27 | igcewieling | not exten => 9|1XXXXXXXXXX but exten => _91XXXXXXXXXX |
| 20:11.34 | [TK]D-Fender | sliWhere di you get this invented syntax from>? |
| 20:11.54 | slicknick5181 | igcewieling: I want to remove the 9 though |
| 20:12.00 | [TK]D-Fender | slicknick5181: And you were missing the "_" to indicate that even what follows was in fact a pattern |
| 20:12.20 | [TK]D-Fender | slicknick5181: exten => 9|1XXXXXXXXXX,n,Dial(SIP/${EXTEN}@callcentric,,tTkK) <-- then knock it out of the VARIABLE you are using there |
| 20:12.33 | [TK]D-Fender | slicknick5181: THAT is what you're dialing. So trim the variable |
| 20:12.34 | igcewieling | exten => _91XXXXXXXXXX,n,Dial(SIP/${EXTEN:1}@callcentric,,tTkK) |
| 20:12.36 | igcewieling | this is basic dialplan |
| 20:12.55 | igcewieling | You understand that freepbx causes impotence, right? |
| 20:13.49 | slicknick5181 | [TK]D-Fender: Ok thats what I was looking for |
| 20:14.36 | slicknick5181 | igcewieling: Thank you, I'm not using freepbx. I want to try things the right way first |
| 20:15.03 | igcewieling | slicknick5181: it is also a good idea to add a Hangup at the end of each as well just to be sure the dialplan doesn't continue somewhere else |
| 20:15.39 | [TK]D-Fender | slicknick5181: Wherever you were looking that even suggested "|"was part of a dialplan pattern highlights them as a terrible resource. Please refer to the official Asterisk WIKI and TFOT |
| 20:15.44 | [TK]D-Fender | wiki.asterisk.org |
| 20:15.46 | [TK]D-Fender | ~book |
| 20:15.46 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:16.17 | slicknick5181 | Thank you guys! |
| 20:17.10 | [TK]D-Fender | XXXXXXXXXX <- also not great to use such a pattern where NANPA is clearly more specific |
| 20:23.22 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:23.45 | *** part/#asterisk AmbassadorPants (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:23.52 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 20:27.43 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:40.50 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat12.beelinegprs.ru) |
| 20:44.45 | [sr] | what's the sintax in configure to enable DEBUG_THREADS and DONT_OPTIMIZE ? |
| 20:45.01 | seanbright | make menuselect |
| 20:45.12 | seanbright | ./configure --enable-dev-mode |
| 20:45.19 | seanbright | then 'make menuselect' |
| 20:48.40 | [sr] | dev-mode will enable this two flags? ok nice |
| 20:48.42 | [sr] | merci :) |
| 20:50.07 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 20:50.24 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:51.31 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 20:51.52 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 20:51.57 | polysics | hi there |
| 20:52.06 | polysics | does anybody use MRCPRecog here? |
| 20:52.24 | polysics | I have it working and connecting to LumenVox, but it seems to never actually return |
| 20:53.29 | slicknick5181 | Hello again, I am looking how to route calls based on DID. I have 2 DIDs 1st one 12345 I would like to ring SIP/101&SIP/102 where as I want the 2nd DID 22334455 to ring SIP/102 |
| 20:53.57 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 20:54.17 | [TK]D-Fender | slicknick5181: So go do it. Match the first DID. Do what you want. Make a different match for the other. Do what you want there. |
| 20:54.42 | igcewieling | there are multiple ways, the easiest is the "ex-girlfriend option" google it. |
| 20:54.55 | [TK]D-Fender | igcewieling: Not quite |
| 20:54.59 | [TK]D-Fender | igcewieling: that's CALLERID. |
| 20:55.12 | [TK]D-Fender | igcewieling: not inbound targeted number |
| 20:55.16 | igcewieling | oh, I suppose it would help if I actually READ the question. |
| 20:55.22 | [TK]D-Fender | igcewieling: Possibly ;) |
| 20:55.53 | slicknick5181 | [TK]D-Fender: Well they are both from same carrier I don't have different settings is sip.conf. And thats where I get confused my carrier says they send the DID in the sip header |
| 20:56.39 | polysics | does anyone else use MRCPRecog? I am starting to think no one does :-9 |
| 20:56.51 | [TK]D-Fender | slicknick5181: Do you tell sip.conf every number that you are going to dial to tht epeer from the phone you made it for? |
| 20:56.54 | polysics | at this point I don't just need help, I also feel lonely :-) |
| 20:57.21 | [TK]D-Fender | slicknick5181: the peer just identifies that they ARE your provider.... the CALL from them targets the NUMBEr (DID0 that was actually called by the outside caller. |
| 20:57.28 | [TK]D-Fender | slicknick5181: You don't define that in sip.conf |
| 20:57.45 | [TK]D-Fender | slicknick5181: that is your dialplan's job to look at the number that the peer calls |
| 20:57.49 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:58.14 | *** join/#asterisk kchehab (~ck@bba425084.alshamil.net.ae) |
| 20:58.38 | slicknick5181 | [TK]D-Fender: I know that but I have never done DID routing I always had different carriers. And I found an article on this but I wasn't sure I understood where to put the DID in the incoming string |
| 21:00.08 | slicknick5181 | [TK]D-Fender: http://pastebin.com/AMKXYmHC |
| 21:00.10 | kchehab | i enable srtp + tls but i face a problem while two extensions are behind one router ,calls fails and send 503 service unavailable and if remove the srtp the call pass normally any idea on how to fix it |
| 21:00.52 | igcewieling | slicknick5181: "s" is not a "catch all" |
| 21:01.19 | igcewieling | in fact "s" only matches if the provider does NOT send the DID with the call. |
| 21:01.30 | kchehab | igcewieling hi |
| 21:01.43 | igcewieling | "s" means "provider or technology is too 's'tupid to send the information" |
| 21:04.00 | slicknick5181 | igcewieling: I understand that but where would I put the DID |
| 21:04.09 | [TK]D-Fender | slicknick5181: You are showing dialplan that assumes what the call from the carrier looks like and that you even HAVE to split of the "To" header to get the targeted bumber which for the vast majority you do not. |
| 21:04.44 | [TK]D-Fender | slicknick5181: And you have not told us who your provider is so we don't have the to use in judging what you did just show us |
| 21:04.48 | igcewieling | slicknick5181: exten => 12125551212,1,whatever where "12125551212" is the DID |
| 21:04.51 | kchehab | [TK]D-Fender good evening |
| 21:04.59 | [TK]D-Fender | slicknick5181: If you want to know how to deal with the incoming call.. then LOOK at the incopming call |
| 21:06.17 | slicknick5181 | [TK]D-Fender: So I can basically put the DID in like I would an outgoing string |
| 21:06.24 | slicknick5181 | <PROTECTED> |
| 21:06.40 | kchehab | [TK]D-Fender last time you help me in TLS and i am stuck now in SRTP :) please can you advise ? |
| 21:06.41 | [TK]D-Fender | slicknick5181: outgoing string? a DID is an INBOUND number |
| 21:07.13 | slicknick5181 | <PROTECTED> |
| 21:07.26 | [TK]D-Fender | slicknick5181: Lets also look at the CALL |
| 21:08.25 | igcewieling | slicknick5181: what [TK]D-Fender is saying is the CLI will provide valuable information about what you need to do by just trying an inbound call and seeing how it fails. |
| 21:09.40 | slicknick5181 | http://pastebin.com/aeW2xBPp |
| 21:09.59 | slicknick5181 | [TK]D-Fender: http://pastebin.com/aeW2xBPp |
| 21:10.51 | slicknick5181 | igcewieling: Calls do not fail as I have an incoming setup already in place from when I had only 1 number |
| 21:10.58 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 21:11.05 | [TK]D-Fender | slicknick5181: you have OUTGOING stuff in the same context as INCOMING stuff. Horrible layout |
| 21:11.09 | [TK]D-Fender | slicknick5181: Split that up first |
| 21:11.24 | [TK]D-Fender | slicknick5181: Next... if this is your dfialplan, then what was that previous pastebin? |
| 21:12.14 | slicknick5181 | [TK]D-Fender: I have it split up I mixed it for you so I did not have 2 Pastebins, the previous was an example I found online |
| 21:12.41 | [TK]D-Fender | slicknick5181: That does feel about right for CC. |
| 21:14.20 | slicknick5181 | [TK]D-Fender: That previous pastebin is what kinda confused me as I was not really sure where exactly to put the DID |
| 21:15.42 | slicknick5181 | [TK]D-Fender: But if I understand correct I can add to my incoming context exten => (incoming did) followed by who to call |
| 21:15.57 | [TK]D-Fender | slicknick5181: http://pastebin.com/V37kBF8Q |
| 21:16.09 | [TK]D-Fender | slicknick5181: uses that sample to split the DID and then go to a context that jsut has te basic in it. |
| 21:16.18 | [TK]D-Fender | slicknick5181: And separates the OUTBOUND from the INBOUND parts. |
| 21:17.39 | [TK]D-Fender | slicknick5181: Fine tune this to your actual #'s |
| 21:17.44 | [TK]D-Fender | slicknick5181: And learn from it |
| 21:18.14 | slicknick5181 | [TK]D-Fender: Thank you very much guys!!!! |
| 21:19.18 | polysics | does anyone know if there is an IRC channel for UniMRCP, please? |
| 21:19.38 | kchehab | i enable srtp + tls but i face a problem while two extensions are behind one router ,when exten 100 call 101 the call fails and have a reason 503 service unavailable and if remove the srtp the call pass normally any idea on how to fix it |
| 21:21.01 | [sr] | navaismo: from the yesterday matter, there's no deadlock when i cant make/receive call's |
| 21:26.54 | *** join/#asterisk kchehab (~ck@bba425084.alshamil.net.ae) |
| 21:27.00 | kchehab | sorry i got disconnected |
| 21:28.38 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:29.20 | kchehab | any idea if there is a a config in sip.conf i should add? |
| 21:32.20 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 21:35.53 | *** join/#asterisk Marco-123 (d1a2ff8b@gateway/web/freenode/ip.209.162.255.139) |
| 21:37.40 | Marco-123 | hi all. im trying to some sip video calling. it works! i wanted h263p codec for video and i just got that to work. however, asterisk transcodes this at a lower frame-size than i'd like. is there any way to configure this? my clients specify cif as a frame size. and peer-to-peer, they work. but when i call through asterisk, i only get qcif. any ideas? |
| 21:37.46 | Marco-123 | plz and thanks |
| 21:39.11 | [TK]D-Fender | Asterisk does not transcode video at all |
| 21:40.25 | kchehab | i am facing a call drop when two client are using srtp behind one router ,is there any thing i can do from sip.config |
| 21:42.15 | Marco-123 | [TK]D-Fender: oh! thats what i interpreted from my googling. so it only does that for audio codecs |
| 21:42.16 | Marco-123 | ? |
| 21:42.28 | [TK]D-Fender | kchehab: Look correct |
| 21:42.33 | [TK]D-Fender | Marco-123: correct |
| 21:42.44 | [TK]D-Fender | kchehab: Look at your actual call |
| 21:42.58 | Marco-123 | hmm. any idea why my sip video turns into qcif over asterisk but cif (as specified) when peer-to-peer? |
| 21:43.34 | [TK]D-Fender | Marco-123: no... especially since I don't SEE your configs and call |
| 21:44.34 | Marco-123 | [TK]D-Fender: fair. ok googling where to find such logs |
| 21:44.34 | igcewieling | Marco-123: maybe you are not allowing the right video codecs? |
| 21:45.00 | [TK]D-Fender | Marco-123: Asterisk CLI |
| 21:45.02 | [TK]D-Fender | not logs. |
| 21:45.05 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 21:45.06 | [TK]D-Fender | Live with SIP DEBUG enabeld |
| 21:45.07 | Marco-123 | igcewieling: dont think so. ive specified h263 and my client reports that its using h263. lower level logs from the client indicate so too. |
| 21:45.46 | [TK]D-Fender | " i wanted h263p codec for video" |
| 21:45.53 | [TK]D-Fender | "ive specified h263" |
| 21:46.09 | kchehab | [TK]D-Fender please what do you mean by Look at your actual call |
| 21:46.28 | kchehab | [TK]D-Fender you mean to get a capture ? |
| 21:46.36 | [TK]D-Fender | look at the call. |
| 21:46.39 | [TK]D-Fender | What is unclear here? |
| 21:46.46 | [TK]D-Fender | Where do we see asterisk processing your call? |
| 21:46.50 | [TK]D-Fender | Actual details. |
| 21:47.10 | [TK]D-Fender | YES, including SIP DEBUG to prove what is negotiated |
| 21:47.13 | kchehab | [TK]D-Fender using core set verbose or sip debug so you preffer |
| 21:47.19 | kchehab | ok |
| 21:47.38 | [TK]D-Fender | BOTH |
| 21:47.47 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 21:47.48 | kchehab | [TK]D-Fender thanks i will replicate the scenario and get back to you |
| 21:49.05 | navaismo | [sr], hmmm, and you need to restart dahdi from linux console to get it work again right? |
| 21:49.32 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 21:50.13 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
| 21:50.17 | [sr] | navaismo: ya |
| 21:50.55 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:50.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:55.01 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 21:56.35 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 21:57.15 | navaismo | dmesg or syslog show something about dahdi |
| 21:57.31 | navaismo | [sr], ^, for some reason its dying |
| 21:57.42 | [sr] | navaismo: zero zero zero :( |
| 21:58.55 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:01.36 | Marco-123 | looking at a log from a test call i just made with sip set debug on. one of the INVITE requests has qcif=3 in one of the "a" options(?) in the message. i think this is my culprit |
| 22:02.04 | Marco-123 | oops its an OK request |
| 22:02.23 | Marco-123 | SIP/2.0 200 OK |
| 22:03.49 | navaismo | [sr], can you turn on the debug on the console make a reload and pb all the output? |
| 22:04.42 | jmetro | navaismo: did anyone come up with a fix for the top-down queue? i had to run |
| 22:05.09 | Marco-123 | if anyone can help. the TRYING request from the caller is here. it looks like it is NOT specifying a frame-size. http://pastebin.com/c47ZWmjb |
| 22:05.14 | polysics | does any way exist to do AST + TTS prompt with barge in on Asterisk? |
| 22:05.17 | Marco-123 | can anyone confirm this? |
| 22:05.52 | navaismo | jmetro, igcewieling comment a solution without using a queue, that's the only i saw |
| 22:05.53 | job | navaismo, do you maybe have an example how to do that? |
| 22:06.05 | job | (connecting two calls) |
| 22:06.52 | navaismo | job: originate local/mycelnumber@mycontext extension mylocalnumber@mycontext |
| 22:07.11 | job | is that with callfile or the other option? |
| 22:07.33 | igcewieling | you could also so something like making static queue members (always logged in) and dedicating a line appearance to the queue calls, set DND on the phone to "log out" of the queue. There are dozens of things you can do. |
| 22:07.44 | igcewieling | that way you get some queue stats for people who needs them |
| 22:09.01 | navaismo | job, using the cli, but yes you can adapt that for both: a call file or using AMI |
| 22:09.29 | navaismo | jmetro, do you need the stats? if not the igcewieling solution will work without issues |
| 22:10.41 | igcewieling | the biggest issues with queues in my experience is that employees are lazy sods and dodge queue calls like they are the plague |
| 22:11.17 | igcewieling | "I'm sorry all your queue callers were dumped into voicemail because all your agents logged out of the queue, but that IS NOT OUR PROBLEM." |
| 22:11.24 | jmetro | navaismo did he actually type out a dialplan for it? cause i could use that... otherwise i'm guessing its just a hard-coded loop between whoever I want. |
| 22:12.13 | [TK]D-Fender | jmetro: what is this "top down" deal you're talking about? |
| 22:12.22 | jmetro | My queue agents are reliable because I only have 2-4 of them. Its just the strategy that i need to nail down. I cant have the president of the company getting a queue call because the other guys logged out and in and he is left as the first member |
| 22:12.22 | [TK]D-Fender | jmetro: Before it was just "add memebers on the fly". |
| 22:12.54 | jmetro | Agent 1 should always be dialed first if present, Agent 2 should always be dialed second if present [or first if agent 1 is gone], etc. For 4 people. |
| 22:13.02 | navaismo | jmetro, nope he didn't but just use dial & dial & dial and in the last step jump again to the first dial. |
| 22:14.23 | navaismo | jmetro, if you want to exit add some variables to count loops, or use a while to exit and do other stuff, or just leave your client forever and ever there jumping and listening moh |
| 22:14.34 | navaismo | like my bank does when i call for support |
| 22:14.36 | jmetro | right |
| 22:14.58 | jmetro | i know coding logic, i just figured it would be built in |
| 22:16.30 | [TK]D-Fender | jmetro: that's why queue members have "priority" levels |
| 22:19.37 | jmetro | you mean Penalties |
| 22:20.21 | jmetro | unfortunately penalties do not escalate the call when the Timeout happens. |
| 22:20.37 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 22:21.02 | [TK]D-Fender | They do. |
| 22:21.14 | [TK]D-Fender | There is even magical CONFIG file that determines... EXACTLY THAT |
| 22:21.22 | [TK]D-Fender | You should probably take a peekl |
| 22:21.59 | jmetro | with strategy "ringall", timeout set to 5, two members in the queue with penalty 1 and 2, i can watch the timeout happen and it rings the first member again |
| 22:23.00 | [TK]D-Fender | I'd go look at that magical config file that has been overlooked.... |
| 22:23.11 | jmetro | queues.conf? |
| 22:23.15 | [TK]D-Fender | No. |
| 22:23.33 | [TK]D-Fender | Time to go rip open the tarball and look at the sample configs again.... |
| 22:23.46 | [TK]D-Fender | It'll be more than a little obvious even... |
| 22:24.05 | jmetro | unfortunately ls /etc/asterisk/ | grep "magical" returns no results. |
| 22:24.34 | [TK]D-Fender | So magical that "magical" isn't even VISIBLE in its name.. |
| 22:27.25 | jmetro | hm. I dont recall setting the maxpenalty values if thats what you mean. especially since if the max penalty was "1" then even if the first member was busy, the max is too low for penalty "2" to be allowed |
| 22:28.32 | [TK]D-Fender | I'm not seeing you clue in to what I've directed you to find yet... |
| 22:28.53 | jmetro | i was guessing queuerules |
| 22:30.09 | [TK]D-Fender | OMG HOPE! |
| 22:32.02 | [TK]D-Fender | Now I don't have much for reference as to what you've actually done so far to suggest where you might have gone wrong.... |
| 22:32.06 | [TK]D-Fender | So... |
| 22:37.38 | job | navaismo, when i enter such a command, my phone rings, after i pick up my friends phone rings, but then there is silence |
| 22:37.45 | job | (after he picks up) |
| 22:41.28 | navaismo | job, weird, what technology you are using to call. SIP Analog or E1/t1 |
| 22:43.10 | job | sip i guess |
| 22:44.12 | jmetro | yeah queuerules didnt change anything, penalty incrememnt every 5 seconds, still hits the first member. |
| 22:45.20 | jmetro | as long as that first member is available it always rings him with penalties in place. |
| 22:47.04 | navaismo | job, check your cli output |
| 22:48.25 | job | it's a lot of output :) |
| 22:49.20 | navaismo | pb someone here can help you |
| 22:49.27 | navaismo | ~pb |
| 22:49.28 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:50.52 | job | https://paste.as5580.net/p/z65dwwjq6yzbwfl4 |
| 22:56.25 | navaismo | job, i don't see the bridging between SIP/u8200620-000006a8 & SIP/u8200620-000006a9 |
| 22:56.32 | navaismo | but dont sure using freepbx |
| 22:56.41 | [TK]D-Fender | -- SIP/u8200620-000006a9 answered Local/0654942365@outbound-allroutes-1bbb;1 -- Executing [h@macro-dialout-trunk:1] Macro("Local/0654942365@outbound-allroutes-1bbb;1", "hangupcall,") in new stack |
| 22:56.47 | [TK]D-Fender | Looks like the call died instantly |
| 22:56.56 | [TK]D-Fender | You should probably be DEBUGGING that call |
| 22:57.17 | job | i'm sorry, i am not familiar with asterisk or pbx's at all |
| 22:58.43 | navaismo | [TK]D-Fender, the local channels hangup when the other party answer but i don't see the bridging between both lines |
| 22:58.59 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 22:59.35 | job | feels like a tool :) |
| 23:01.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.45 | navaismo | [TK]D-Fender, job this is how i see it from my cli without fpbx http://pastebin.com/rtEBUGsY |
| 23:06.43 | job | could it be i am using the wrong context? |
| 23:07.31 | navaismo | don't think so, the calls seems to work, but as a fender said try debugging, with sip set debug on |
| 23:08.25 | navaismo | and pb again |
| 23:09.50 | lorsungcu | anyone worked with the third lane developer? |
| 23:09.59 | job | guess all the phones in the office are turned on |
| 23:10.02 | job | so many subscribes |
| 23:11.42 | navaismo | brb |
| 23:13.25 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 23:13.48 | job | seems like same sympthoms http://marc.info/?l=asterisk-dev&m=126466732027853 |
| 23:14.44 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat12.beelinegprs.ru) |
| 23:18.23 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 23:22.41 | *** join/#asterisk lorsungcu_ (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 23:25.28 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 23:26.12 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 23:34.03 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:37.05 | navaismo | job, the sip debug can tell you more |
| 23:52.01 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
