| 00:02.31 | *** join/#asterisk Buuyo (~electrum@ieee1003.1-1988.posi.xxx) |
| 00:03.51 | Buuyo | nickfennell: Thanks for the hint earlier about IRQ sharing. I rebooted teh box and killed the (unused) firewire controller sharing that irq and magically the FXS ports are silent. :) |
| 00:04.15 | Buuyo | So much for cheap counterfeit cards on ebay. :) It turned out ok in the end. |
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| 00:27.05 | jmordica_mac | are you guys working on a scalable cloud solution with asterisk? |
| 00:27.16 | jmordica_mac | heard this from someone but couldn't find anything on it |
| 00:36.38 | *** join/#asterisk its_jeremy_ (~omghax@24-119-28-208.cpe.cableone.net) |
| 00:36.39 | its_jeremy_ | node |
| 00:39.40 | WIMPy | ~scf |
| 00:39.57 | WIMPy | jmordica_mac: It's called SCF. |
| 00:42.49 | jmordica_mac | are there API's that can be hooked into so developers can build their own GUI's or is there a GUI being working on? |
| 00:44.49 | *** join/#asterisk moy (~moy@173.239.155.74) |
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| 01:17.21 | picard276 | anyone very familiar with USSD and HLR's? |
| 01:17.31 | p3nguin | jmordica_mac: GUIs suck. Use vim. |
| 01:23.32 | *** part/#asterisk mahables (~Jason@c-76-101-131-88.hsd1.fl.comcast.net) |
| 01:27.38 | phix | paulc: That sounds pretty neat |
| 01:28.08 | phix | jmordica_mac: If I hear the word cloud one more time!>>R*(#*#&#!!!11 |
| 01:38.37 | Buuyo | What could cause asterisk's Hangup() to not really hangup an FXO? For some reason, if I Hangup() as a callee, but the caller doesn't hang up, the line stays busy. Actually, when I get Asterisk to dial using that FXO, the caller that didnt hang up can hear the numbers dialed. |
| 02:03.58 | *** join/#asterisk rhamnett (~rick@5e094092.bb.sky.com) |
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| 02:15.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:16.48 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 02:16.55 | Micc | anyone ever use voipmonitor? |
| 02:25.50 | WIMPy | Buuyo: Sounds like your telco hasn't updated their equipment for at least 30 years. |
| 02:25.58 | Buuyo | heh |
| 02:26.09 | Buuyo | I just realized it's not asterisk's fault. |
| 02:26.16 | Buuyo | It does it with a plain ol' telephone on the line too |
| 02:26.37 | Buuyo | It must be something about the unit they have on premises :| |
| 02:26.52 | WIMPy | What country is it? |
| 02:26.55 | Buuyo | canada |
| 02:27.09 | WIMPy | hmm |
| 02:27.15 | Buuyo | should be fxs_ks yeah? |
| 02:27.47 | WIMPy | Probably. |
| 02:28.12 | WIMPy | I'm not that good at history :-) |
| 02:28.16 | Buuyo | heh |
| 02:28.43 | Buuyo | Asterisk notices right away if the caller hangs up at least. |
| 02:29.12 | Buuyo | I almost wonder if there's a way to keep the line as occupied until at least like 15 seconds after asterisk hangs up. :| |
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| 03:16.47 | SeRi | p3nguin: I got a new job :D |
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| 03:21.28 | SeRi | I been busy thats why I havent been online much. |
| 03:21.34 | SeRi | But yes finally it happend |
| 03:48.18 | WIMPy | Is there some configuration option for chan_dahdi to make it accept keypad? |
| 03:50.29 | p3nguin | seri: What happened to the old job, and what is the new job? |
| 03:50.38 | *** part/#asterisk Buuyo (~electrum@ieee1003.1-1988.posi.xxx) |
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| 04:42.39 | ChannelZ | WIMPy: what do you mean? |
| 04:45.08 | WIMPy | I tried to dial a number starting with # and had to find out that it doesn't work. |
| 04:45.28 | ChannelZ | hmm.. misinterpreted as a feature code? |
| 04:45.37 | WIMPy | Many phones automatically switch to keypad mode when you dial something starting with * or #/ |
| 04:45.51 | WIMPy | And chan_dahdi seems to ignore those. |
| 04:49.02 | *** join/#asterisk keepar (ayane@quality.webhosting.at.sunsaturn.com) |
| 04:49.55 | [TK]D-Fender | WIMPy: If you're referring to FXS yes, by default it steals # as an end-of-dial char |
| 04:50.07 | [TK]D-Fender | WIMPy: For first-dial, not IVRs, etc |
| 04:50.09 | WIMPy | No. BRI. |
| 04:51.00 | WIMPy | POTS doesn't differ between called number and keypads. |
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| 06:03.25 | BeeBuu | i got a problem |
| 06:04.43 | BeeBuu | i had a queue,the agent off line,but when i type "queue show" command in CLI, it still show agent is "BUSY" why? |
| 06:05.06 | BeeBuu | i had a queue,the agent off line,but when i type "queue show" command in CLI, it still show agent is "BUSY"! why? how can i fix this? |
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| 07:14.02 | luke0512 | hello |
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| 07:33.41 | luke0512 | need some help for configuring hfc card on centos and asterisk 1.6...her are some infos of my installation http://fpaste.org/QIZT/ |
| 07:35.11 | luke0512 | the asterisk box is connected to an internal S0 of an telefon box with 1 external S0, one internal S0 an 4 Analog connections |
| 07:35.19 | WIMPy | That resource allocation conflict doesn't look good. |
| 07:35.31 | WIMPy | But I'd recommend at least Asterisk 1.8 anyway. |
| 07:36.14 | luke0512 | therefore i have to use a beta, cause its only in testing area |
| 07:36.48 | WIMPy | For what? |
| 07:36.53 | WIMPy | And what's your issue? |
| 07:37.14 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:37.15 | WIMPy | The resource conflict seems to happen elsewhere. |
| 07:37.16 | schmidts | good morning |
| 07:37.17 | luke0512 | asterisk 1.8, cause i want to use rpm source |
| 07:37.58 | WIMPy | I think you will find rpms for all official versions, but I don't care about packets. |
| 07:38.05 | WIMPy | Moin schmidts |
| 07:40.27 | WIMPy | But you still haven't told us what isn't working. The card seems fine from your paste. |
| 07:46.06 | luke0512 | what isn't working is not correct sense or i have used wrong sentence...i have get asterisk working with some internal phones (Gigaset 610 and Ekiga Softphone)...but what do i have to configure for outgoing calls? |
| 07:46.18 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 07:46.39 | luke0512 | the asterisk box is connected via ISDN to the phone box |
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| 07:48.08 | luke0512 | i just tried to find information in asterisk book from stan wintermeyer but there is not much info inside for isdn connection |
| 07:48.47 | luke0512 | i just have found information that i have to use the misdn module |
| 07:48.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 07:48.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 07:49.06 | luke0512 | but what about configuration? |
| 07:49.08 | WIMPy | If I get it right, you're using misdn2. |
| 07:49.26 | WIMPy | That works with LCR which then connects to Asterisk via chan_lcr. |
| 07:49.54 | WIMPy | http://linux-call-router.de/ |
| 07:50.07 | luke0512 | what i have installed is mISDN and not misdn, take care of Capital letter |
| 07:50.27 | WIMPy | Or you downgrade to misdn1 and use chan_misdn or you try a dahdi version with hfc support. |
| 07:50.51 | WIMPy | Both are written mISDN. I'm just too lazy to press Shift. |
| 07:51.08 | luke0512 | ok i think i try the way by using 1.8 from testing |
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| 08:04.12 | luke0512 | i now have installed 1.8.3 but the asterisk misdn will not be installed --> http://fpaste.org/kSBV/ |
| 08:04.54 | WIMPy | That's what I said above. The versions don't fit. |
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| 08:54.00 | myschyk | Does anyone has fix for astmonproxy getting bufferoverlow issue? |
| 08:58.29 | *** join/#asterisk moneer (~moneer@adsl-109-74-39-198.dynamic.yemennet.ye) |
| 08:58.45 | moneer | hello any one here |
| 08:58.49 | moneer | I nned help |
| 08:58.54 | moneer | i need help |
| 08:59.16 | schmidts | ~ask |
| 08:59.16 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:59.57 | jacc0 | wazup moneer? |
| 09:00.29 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 09:00.56 | moneer | i use say number in asterisk 10 dialplan but no sound is heard, when i use plapack(digit/${number}) i hear the digit sound, but when i use SayNumber(${number}) i didn;t hear any thing |
| 09:02.57 | moneer | jacc8 , can u tell me why, SayNumber is not working in asterisk 10 |
| 09:03.00 | jacc0 | could you pastebin your dialplan ? |
| 09:03.06 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 09:03.08 | jacc0 | ~pb |
| 09:03.08 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 09:03.28 | jacc0 | I haven't tried * 10 |
| 09:03.41 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:03.48 | moneer | yes jacc8, i'm newbie and trying to figure out the dialplan concept |
| 09:04.02 | moneer | i use this dialplan code from one of the books |
| 09:04.04 | moneer | exten => 123,1,Set(COUNT=10) |
| 09:04.04 | moneer | same => n(start),GotoIf($[${COUNT} > 0]?:goodbye) |
| 09:04.05 | moneer | same => n,SayNumber(${COUNT}) |
| 09:04.05 | moneer | same => n,Set(COUNT=$[${COUNT} - 1]) |
| 09:04.05 | moneer | same => n,Goto(start) |
| 09:04.05 | moneer | same => n(goodbye),Hangup() |
| 09:04.07 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:04.10 | jacc0 | the dialplan is in your extensions.conf; I would like to check it for typ0s |
| 09:04.32 | schmidts | moneer i am not sure but you might use an answer before saynumber |
| 09:04.37 | jacc0 | I'm femiliar with `same` |
| 09:05.58 | moneer | if i put Playback(digits/${COUNT}) instead of SayNumber(${COUNT}) I can hear the voice |
| 09:06.40 | moneer | but with sayNumber(${COUNT}), it is not working and the voice didn't work |
| 09:07.46 | jacc0 | could you try to replace : 'same => n' with 'exten => 123,n' |
| 09:07.53 | moneer | schmidts, does sayNumber only work with Answer |
| 09:08.45 | schmidts | moneer normally most functions which playback any sound answer the channel first, maybe sayNumber doesnt do this |
| 09:09.09 | moneer | schmidts, yes it works when i use Answer before saynumber, that is good |
| 09:09.14 | moneer | thanks for help |
| 09:09.36 | moneer | one more question |
| 09:10.57 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 09:11.51 | moneer | i have the asterisk in linuxmint 10, and my softphone is in the same machine, i have defined one sip user, and one iax user, i need to comunicate with them in the same softphone, as my softphone support iax and sip, when i trying to register iax user after sip user it is not registered, and the server give me this |
| 09:11.54 | moneer | Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.2 in the calltokenoptional list or setting user 0000FFFF0005 requirecalltoken=no |
| 09:12.28 | WIMPy | So do as it told you. |
| 09:12.36 | ollii | indeed |
| 09:12.43 | moneer | when i put the requirecalltoken=no in the iax it is registered, but the iax user can not call any extension |
| 09:12.47 | ollii | i assume your using * > 1.2 ? |
| 09:13.07 | moneer | what is going on, what the problem with that? |
| 09:13.09 | ollii | you're using the right context? |
| 09:13.35 | ollii | put requriecalltoken to iax peer conf and start a call, then paste this output to |
| 09:13.36 | ollii | ~pb |
| 09:13.36 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 09:14.01 | ollii | too much needle in a haystack |
| 09:14.02 | ollii | :< |
| 09:14.47 | moneer | odes it because i have to users defined in the same machine? |
| 09:14.48 | *** join/#asterisk Autoeth (~nguyendp0@ip98-165-34-222.ph.ph.cox.net) |
| 09:14.54 | luke0512 | hmm WIMPy what do you think shall i do now? |
| 09:15.06 | moneer | does it because i have two users defined in the same machine? |
| 09:16.08 | WIMPy | luke0512: I gave you the three options you've got. Yu can also read the summary on http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 09:19.43 | jacc0 | anyone interested in host my brand new "Cisco SPA remote controle without authentication" |
| 09:21.26 | jacc0 | ? |
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| 09:40.35 | nickfennell | 40% wait... |
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| 09:48.42 | Cadey | Hi guys, does anyone know how to get a cisco (LinkSys) 5XX range to allow a * mid dial string? when ever a * is part of the dial string and NOT the first charater of the dial string it simply fails the call and sends nothing to the server |
| 09:48.49 | *** join/#asterisk jzaw (~jzaw@2001:8b0:7:0:216:cbff:febe:f5dd) |
| 09:48.55 | jzaw | morniing peeps |
| 09:48.58 | Cadey | so *43 for ech works, but 111*111 would cause it to do nothing |
| 09:48.58 | moneer | when i try to register iax user using softphone i got this from asterisk server "Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.2 in the calltokenoptional list or setting user 0000FFFF0005 requirecalltoken=no |
| 09:48.59 | moneer | ", what does that means?? |
| 09:50.12 | luke0512 | bye for now i have to go to work |
| 09:51.04 | moneer | hello guys any one can help here?, please |
| 09:52.37 | wdoekes2 | ~ask |
| 09:52.37 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:54.14 | moneer | why i try to register an iax user using a software phone which support iax, i got this error from the asterisk server "Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.2 in the calltokenoptional list or setting user 0000FFFF0005 requirecalltoken=no |
| 09:54.15 | moneer | " what does that means |
| 09:55.40 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 09:57.22 | moneer | when i try to register an iax user using asterisk 10 by a softphone, i get this error from the server "Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.1.2 in the calltokenoptional list or setting user 0000FFFF0005 requirecalltoken=no" what does this mean?? please help |
| 09:58.01 | WIMPy | hands moneer some glasses. |
| 10:02.01 | nickfennell | Just updated Chrome |
| 10:02.07 | nickfennell | Let's see what happens to my system now |
| 10:02.22 | *** join/#asterisk RZero (~RZero@gateway-hq-uk.oxygen8.com) |
| 10:02.48 | RZero | Hi guys is it possible to run SS7 and E1 connections together on the same Server ? |
| 10:03.30 | WIMPy | RZero: No sure what you're asking for. You can run SS7 on E1. |
| 10:04.43 | *** join/#asterisk DaneoShiga (~dshiga@kraz.dreamhost.com) |
| 10:04.43 | RZero | Sorry I mean we have to suppliers one uses ss7 (c7) signalling and the other uses EuroISDN |
| 10:05.22 | RZero | is it possible to run both types of connections on a single server ? |
| 10:05.46 | WIMPy | So you want to run DSS1 on one port and SS7 on another port? |
| 10:06.05 | WIMPy | Should work. But maybe someone can definitely confirm that. |
| 10:06.10 | RZero | yes |
| 10:07.14 | RZero | we have two digium cards installed so 8 ports are available |
| 10:07.27 | DaneoShiga | there's a common used cdr fields combination to have real unique reference? since uniqueid aren't unique between CDRs? |
| 10:11.32 | moneer | using asterisk 10 iax2 can not register, and can not receive, make call, please help |
| 10:13.35 | moneer | what is IAX provisioning, when i try to reload the module chan_iax2.so, I got IAX provisioning disabled |
| 10:15.22 | WIMPy | That is to configure IAXys. |
| 10:17.11 | Diffen | Wimpy hello. Just wanted to say that all worked out well after a reboot and a disk check. The only problem I had was to start iSymphony but that turned out to be an error in an xml file :) |
| 10:17.58 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 10:18.13 | moneer | WIMPy, i use iax.conf to configure iax, what else should i do to make iax work? |
| 10:19.17 | WIMPy | Read the messages you have been posting here several times. |
| 10:21.01 | moneer | WIMPy i put that requiretokencall=no, the device is registered, but no calles |
| 10:21.36 | moneer | this message appear "Restricting registration for peer '0000FFFF0005' to 60 seconds (requested 120)" |
| 10:21.44 | WIMPy | moneer: You have already been asked if you've got the right context. |
| 10:22.08 | WIMPy | You can safely ignore that one so far. |
| 10:22.16 | moneer | WIMPy, yes it right context |
| 10:22.46 | WIMPy | Then show us the output of a failed call attempt. |
| 10:23.44 | moneer | when i the iax device is registered it give s this "Restricting registration for peer '0000FFFF0005' to 60 seconds (requested 120)" |
| 10:23.57 | moneer | when i try to call , nothing happen |
| 10:24.14 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 10:24.17 | WIMPy | Turn up debug and verbose. |
| 10:24.32 | WIMPy | If you still see nothing, the problem is on the phone. |
| 10:24.40 | *** join/#asterisk master_of_master (~master_of@p57B52184.dip.t-dialin.net) |
| 10:28.04 | moneer | but the phone is working fine with sip, and it is supporting iax as well |
| 10:32.29 | moneer | when i call i got this error "Determining if address 192.168.1.2 with username 0000FFFF0005 requires calltoken validation. Optional = 0 calltoken_required = 2 |
| 10:32.30 | moneer | ip callno count incremented to 2 for 192.168.1.2 |
| 10:32.30 | moneer | Checking device state for device 0000FFFF0005 |
| 10:32.30 | moneer | iax2_devicestate: Found peer. What's device state of 0000FFFF0005? addr=-1062731518, defaddr=0 maxms=0, lastms=0 |
| 10:32.31 | moneer | Changing state for IAX2/0000FFFF0005 - state 0 (Unknown) |
| 10:32.32 | moneer | device 'IAX2/0000FFFF0005' state '0' |
| 10:32.34 | moneer | Restricting registration for peer '0000FFFF0005' to 60 seconds (requested 120) |
| 10:32.36 | moneer | Checking device state for device 0000FFFF0005 |
| 10:32.38 | moneer | iax2_devicestate: Found peer. What's device state of 0000FFFF0005? addr=-1062731518, defaddr=0 maxms=0, lastms=0 |
| 10:32.41 | moneer | Changing state for IAX2/0000FFFF0005 - state 0 (Unknown) |
| 10:32.45 | moneer | device 'IAX2/0000FFFF0005' state '0' |
| 10:32.47 | moneer | schedule decrement of callno used for 192.168.1.2 in 60 seconds |
| 10:32.49 | moneer | ip callno count decremented to 1 for 192.168.1.2" |
| 10:39.22 | *** join/#asterisk moneer (~moneer@adsl-109-74-39-198.dynamic.yemennet.ye) |
| 10:42.21 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 10:54.34 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 10:55.01 | pif | hi, should I use dahdi-channels.conf or chan_dadhi.conf ? |
| 10:55.05 | pif | are they the same? |
| 10:55.50 | kaldemar | chan_dahdi.conf is the config file for the DAHDI channel driver. anything else is something included in chan_dahdi.conf or invalid. |
| 10:56.18 | pif | so there is no dahdi-channels.conf ? |
| 10:56.24 | pif | ok I see! |
| 10:58.27 | kaldemar | you can use what ever names for custom config files, as long as they are included with an "#include" statement in the real config file. |
| 10:59.04 | pif | dahdi is not loaded by asterisk, where should I look? |
| 10:59.31 | kaldemar | "module load chan_dahdi.so" in CLI with verbosity enabled |
| 10:59.47 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 11:00.58 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 11:01.14 | pif | hmm, /usr/lib/asterisk/modules/chan_dahdi.so is not there |
| 11:03.49 | pif | must install asterisk-dahdi |
| 11:13.31 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:13.38 | pif | ok works now, thanks |
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| 11:39.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 11:53.44 | moneer | people the voice email have bug in asterisk 10 |
| 11:53.59 | moneer | voicemail has some bug, any one agree |
| 11:55.32 | moneer | any one try voicemail in asterisk 10 |
| 12:01.17 | wdoekes2 | moneer: you're not being real specific ;) would you care to explain the symptoms of the observed bug? |
| 12:02.45 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:02.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:05.36 | *** join/#asterisk zarnick (~minterci@unaffiliated/zarnick) |
| 12:05.41 | zarnick | hello |
| 12:06.06 | zarnick | one question, is there anyway to know what codec has a physical telefone connect on a specific call? |
| 12:06.52 | jacc0 | from CLI? |
| 12:06.56 | jacc0 | or in dialplan? |
| 12:07.04 | zarnick | from cli |
| 12:07.16 | jacc0 | sip show channels |
| 12:07.24 | zarnick | I'm having some audio problems, and I think it may be codec related |
| 12:07.31 | moneer | wdoekes2, ok, i'm trying to login toa test voice mail, what happen is that when i click 1, asterisk duplicate 1 to be 11, so each digit i press it is duplicated so for example i have a mailbox 100, it gives me 110000 for password is the same my password is 1234 but it give it 11223344 or duplicate it deffent order based on the speed of pressing for example 12123344 |
| 12:07.36 | jacc0 | in the 'format' collumn |
| 12:08.22 | zarnick | thanks |
| 12:08.23 | schmidts | moneer then you have a DTMF problem and not a voicemail problem, check your DTMF settings |
| 12:08.28 | *** join/#asterisk moneer (~moneer@adsl-109-74-39-198.dynamic.yemennet.ye) |
| 12:09.03 | zarnick | jacc0: is there any way to leave this on the logs so I can check lather when the problem occurs? |
| 12:10.00 | schmidts | zarnick core show function CHANNEL |
| 12:10.09 | moneer | ok, i'm trying to login to a voicemail test account using asterisk 10, when the server prompt me to enter the mailbox i put 100, but unfortnatly the digit when pressed are duplicated for example 100 become 110000 or 101000 based on the digit press speed, the same is applied to the password, is that a bug? |
| 12:10.26 | schmidts | zarnick specially the audioreadformat, audionativeformat and audiowriteformat is interesting for you |
| 12:10.46 | schmidts | moneer no it is not a bug, your DTMF settings are wrong |
| 12:10.52 | zarnick | schmidts: I c |
| 12:11.16 | moneer | schmidts, so how can i correct it? |
| 12:11.58 | schmidts | moneer do you call in to asterisk using a sip client or something else? |
| 12:12.08 | moneer | sip client , yes |
| 12:12.42 | schmidts | moneer then take a look at your dtmf settings in sip.conf and also what settings you are using in your client |
| 12:12.58 | schmidts | moneer sorry its dtmfmode |
| 12:13.02 | moneer | in sip.cof i put this dtmfmode=auto |
| 12:13.44 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:13.55 | zarnick | schmidts: how can I query to audioreadformat for instance? |
| 12:14.58 | schmidts | zarnick i dont know if this will work before the outgoing call leg is initiated but normally you use it like this: exten => 123,n,Noop(asterisk reads audio in format: ${CHANNEL(audioreadformat)} !) |
| 12:15.00 | moneer | my sitting for dtmfmode is auto in sip.conf, and i use SFLphone softphone |
| 12:15.34 | schmidts | moneer and whats your setting in your softphone for dtmf? |
| 12:16.03 | moneer | no option for that in my softphone |
| 12:16.19 | zarnick | schmidts: I c, there's no way I can use this to see, without actually changing the dialplans then right? only the 'sip show channels' commands right? |
| 12:16.57 | schmidts | zarnick core show channel SIP/phoneA should also show you these 3 formats |
| 12:17.04 | zarnick | let me see |
| 12:17.40 | schmidts | moneer then try to add this option to your sip.conf file: relaxdtmf=yes |
| 12:17.51 | moneer | i delete that dtmfmode=auto, and it work now, but it will be make problem in other softphone or phone devices |
| 12:18.06 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 12:18.24 | schmidts | moneer then you should set dtmfmode=rfc2833 most phones work well with this |
| 12:18.49 | moneer | schmidts, ok thanks, my friend, i try it |
| 12:18.58 | schmidts | moneer your welcome |
| 12:21.24 | zarnick | thanks...it helps |
| 12:21.43 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 12:21.51 | qakhan | hi all |
| 12:22.13 | zarnick | let's see if I can catch that god damn error...some calls (outside) get muted only on some end (either caller or receiver), and it's completelly random, and both for incoming and outgoing connections... |
| 12:23.30 | qakhan | i want to setup play message. if caller dial an ext which is not exist in asterisk then message play "ext not exist". |
| 12:27.02 | *** part/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 12:27.29 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 12:41.09 | schmidts | zarnick do you use snom360 and maybe this calls were transfered from this phones? |
| 12:52.53 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:07.20 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:13.11 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:17.21 | *** join/#asterisk moneer (~moneer@adsl-89-189-77-200.dynamic.yemennet.ye) |
| 13:19.57 | moneer | i have this error appear when i register iax user in asterisk 10 "Restricting registration for peer '0000FFFF0005' to 60 seconds (requested 120)", what is the error, even the user can't make calls, but can receive calls |
| 13:20.30 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:21.11 | [TK]D-Fender | That isn't an error |
| 13:21.44 | moneer | so why user can't make call?? |
| 13:21.46 | [TK]D-Fender | As for failing to make calls, you should enable IAX2 debug and actually look at the failed attempt and show us. |
| 13:23.08 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 13:26.16 | nickfennell | Hey [TK]D-Fender |
| 13:26.18 | moneer | there is no any message appear when i'm calling using the iax user |
| 13:26.42 | nickfennell | Here's one for your awesome brain |
| 13:26.59 | nickfennell | I've got a few handsets that repeatedly fail to clear their inuse status |
| 13:27.13 | nickfennell | i have to core restart to clear it |
| 13:27.16 | nickfennell | seen it before? |
| 13:27.53 | [TK]D-Fender | Rarely, and solution was the same. No clue why and nothing I've bothered to dig into myself... wasn't critical |
| 13:28.27 | [TK]D-Fender | moneer, If you're in * CLI at verbose 10 and IAX2 debug and you see nothing then packets aren't even reaching your server. |
| 13:28.29 | nickfennell | hmmm. darn. |
| 13:28.36 | [TK]D-Fender | moneer, Check all your firewalls, forwarding, etc |
| 13:29.01 | moneer | how to turn IAX2 debug |
| 13:29.52 | [TK]D-Fender | "help iax2" <- * CLI |
| 13:30.08 | [TK]D-Fender | Time to learn about *'s CLI help options. |
| 13:30.16 | [TK]D-Fender | first place to look for things |
| 13:30.41 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 13:32.30 | jzaw | depends on vers but 1.6 1.8 and 10 prob use |
| 13:32.30 | jzaw | core show help iax2 |
| 13:32.56 | jzaw | moneer ^^^^ |
| 13:33.27 | [TK]D-Fender | Every error points you better help :) |
| 13:34.30 | moneer | ok i have enabled it |
| 13:34.40 | moneer | using iax set debug on command |
| 13:35.08 | moneer | but seems the packet is not received |
| 13:35.27 | [TK]D-Fender | Time to look at all your networking. |
| 13:35.45 | moneer | do u think it is a problem in the phone, but the phon working fine with sip |
| 13:36.43 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:37.14 | [TK]D-Fender | IAX2 should generally work every place SIP does and more. |
| 13:37.22 | [TK]D-Fender | Check your forwarding and firewalls |
| 13:38.26 | jzaw | moneer: is your phone registering with your * ? |
| 13:38.39 | moneer | yes it is |
| 13:38.48 | jzaw | and you see that in the console ? |
| 13:38.54 | moneer | the soft phone is in the same machine right now with * |
| 13:39.13 | moneer | and the phone support the to protocols, sip and iax |
| 13:39.32 | moneer | with sip it is working fine, but with iax it is not , it can make call |
| 13:40.15 | [TK]D-Fender | If it's on the same machine I'd have to wonder if it's fighting with * to bind the IAX2 port |
| 13:40.46 | [TK]D-Fender | and so forth. This is a networking issues straight up and since its local you should have no issue tracking it to tht point of failure |
| 13:41.13 | zarnick | schmidts: (sorry for the "small" delay), but no, I don't use, they are simple outgoing calls that use a E1 line (with a digivoice board) |
| 13:41.23 | moneer | yes and the peer is registered as show here "0000FFFF0005 192.168.1.2 (D) 255.255.255.255 42259 (E) Unmonitored " |
| 13:42.19 | [TK]D-Fender | moneer, packet dump time.... |
| 13:42.41 | moneer | using what command |
| 13:44.30 | [TK]D-Fender | man tcpdump |
| 13:44.35 | [TK]D-Fender | man netstat |
| 13:45.29 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 13:46.00 | moneer | D-fender, please check this "Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ |
| 13:46.01 | moneer | <PROTECTED> |
| 13:46.01 | moneer | <PROTECTED> |
| 13:46.01 | moneer | <PROTECTED> |
| 13:46.01 | moneer | <PROTECTED> |
| 13:46.02 | moneer | Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK |
| 13:46.04 | moneer | <PROTECTED> |
| 13:46.06 | moneer | <PROTECTED> |
| 13:46.08 | moneer | <PROTECTED> |
| 13:46.10 | moneer | <PROTECTED> |
| 13:46.14 | moneer | <PROTECTED> |
| 13:46.16 | moneer | Received packet 2, (6, 4) |
| 13:46.18 | moneer | Cancelling transmission of packet 1 |
| 13:46.20 | moneer | Really destroying 430, having been acked on final message |
| 13:46.22 | moneer | schedule decrement of callno used for 192.168.1.2 in 60 seconds |
| 13:46.24 | moneer | Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK |
| 13:46.26 | moneer | <PROTECTED> |
| 13:46.28 | moneer | ip callno count decremented to 1 for 192.168.1.2 |
| 13:46.30 | moneer | " |
| 13:48.16 | [TK]D-Fender | ... |
| 13:48.24 | [TK]D-Fender | ~pb |
| 13:48.24 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:48.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^6 |
| 13:48.33 | [TK]D-Fender | Do not flood in here |
| 13:48.38 | *** join/#asterisk b3nt_pin (~b3ntpin@142.162.36.51) |
| 13:49.05 | moneer | ok , but did u check those messages |
| 13:50.30 | [TK]D-Fender | moneer, Doesn't tell me nuch. You also haven't described your working environment. |
| 13:51.24 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 13:52.21 | moneer | D-Fender, ok i'm installing the asterisk 10 , with all the dependecies, in a linux mint 10os, and i configure the sip and iax.conf files, and i have a softphone wich is supporting sip and iax, i'm testing and learning asterisk |
| 13:53.13 | moneer | so asterisk 10 with the softphone are in the same machine |
| 13:53.37 | [TK]D-Fender | moneer, Not the pertinent details : what PC&OS is your softphone running? What IP? What routing to your server. LOOK at your firewalls. What about your sever? Check them there as well. |
| 13:54.01 | [TK]D-Fender | You need to change the port that the softphone uses |
| 13:54.07 | [TK]D-Fender | so it doesn't conflict with * |
| 13:55.01 | moneer | the asterisk and softphone are in the same machine, so the ip of the machine is 192.168.1.2 and i use that in the doftphone configuration to connect with asterisk which is also in the same machine |
| 13:55.12 | moneer | so no firewall |
| 13:55.39 | moneer | the OS is ubuntu desktop |
| 13:56.29 | [TK]D-Fender | prove the FW is empty, and verify that the softphone is running on a different port |
| 13:58.04 | moneer | ok, here si the sip registration peer i have |
| 13:58.05 | moneer | 0000FFFF0001/moneer 192.168.1.2 D N 36190 Unmonitored |
| 13:58.25 | moneer | and here is the iax2 peer i have |
| 13:58.26 | moneer | Name/Username Host Mask Port Status Description |
| 13:58.27 | moneer | 0000FFFF0005 192.168.1.2 (D) 255.255.255.255 42259 (E) Unmonitored |
| 13:58.32 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 13:58.34 | moneer | they are in defferent port |
| 13:58.37 | jzaw | moneer: add |
| 13:58.44 | jzaw | qualify=2000 |
| 13:58.54 | jzaw | in your iax.conf for that phone context |
| 13:59.14 | moneer | jzaw ok let me try |
| 13:59.26 | jzaw | can also tie things down |
| 13:59.27 | jzaw | bindport = 4569 ; Port to bind to (IAX is 4569) |
| 14:00.33 | moneer | ok |
| 14:01.56 | [TK]D-Fender | moneer, If you can't see packets for a call attempt then you have a netowrking SNAFU. |
| 14:03.02 | moneer | D-Fender, do u think that because the two account are registered in the same machine |
| 14:03.22 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pdpofgzzlhtpgssi) |
| 14:03.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:03.23 | [TK]D-Fender | You're running multiple softphones AND * on the same machine? |
| 14:03.30 | moneer | jzaw i did that but unfortunatly no luck, not working the same problem |
| 14:03.38 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 14:03.56 | moneer | D-Fender, yes |
| 14:05.15 | *** join/#asterisk serafie (~erin@nat/digium/x-hfgsvmuqtxltlchu) |
| 14:06.06 | moneer | D-Fender but the Sip i have tow accounts work and they are works fine, the calling each other, they also call the iax user., but the iax user can't call them or even call the voice mail |
| 14:06.48 | moneer | that is make me cray |
| 14:07.23 | kaldemar | "call the iax user" = the phone rings and you actually have a call? |
| 14:07.35 | moneer | Yes |
| 14:07.44 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
| 14:07.46 | [TK]D-Fender | Show us all of your settings in * and on the softphone. show us the firewall on the PC. |
| 14:08.17 | moneer | let show you the sip file, but is it ok to fload it here |
| 14:08.20 | moneer | ? |
| 14:08.45 | [TK]D-Fender | NO |
| 14:08.47 | ayrjola | pastebin |
| 14:08.47 | [TK]D-Fender | PASTEBIN |
| 14:08.51 | [TK]D-Fender | ~pb |
| 14:08.51 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:08.54 | [TK]D-Fender | ^6 |
| 14:09.14 | [TK]D-Fender | tinypic.com for screenshot for phone config, etc |
| 14:09.34 | lhfnet | Hi, I have asterisk 10 configured with ODBC and some times when there is no activity the asterisk daemon goes down. Here the log http://pastebin.com/rdGEBrEb |
| 14:10.21 | moneer | D-fender, ok let me find about them and once i ready i will talk with you |
| 14:18.06 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 14:19.24 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:19.48 | *** join/#asterisk coppice (~coppice@m121-202-30-141.smartone.com) |
| 14:21.06 | moneer | D-Fender, Check this the iax file http://pastebin.com/RwEW9aD6 |
| 14:22.22 | kaldemar | moneer: did you at any point dump incoming packets to port 4569 when you try to call using the soft phone? |
| 14:22.48 | moneer | no i didn't because i don't know how |
| 14:25.26 | moneer | this is the sip.conf http://pastebin.com/sqJXqpja |
| 14:26.08 | moneer | d-fender: this is the extensions.conf http://pastebin.com/ZstY2m1d |
| 14:26.36 | moneer | d-fender: is that ok? |
| 14:28.10 | jzaw | nat=yes ; assume device is behind NAT |
| 14:28.21 | jzaw | is the client behind nat compared to the server? |
| 14:28.34 | jzaw | or vica versa |
| 14:28.34 | [TK]D-Fender | moneer, Ok, I've said this several times and it doesn't seem to be sinking in... |
| 14:28.41 | moneer | jzaw: no it is not |
| 14:28.44 | [TK]D-Fender | moneer, If * gets no packets then it's not *'s fault |
| 14:29.01 | jzaw | what [TK]D-Fender said ^^^^ |
| 14:29.23 | moneer | D-fender: ok then u think the softphone has a problem |
| 14:29.31 | [TK]D-Fender | moneer, You don't seem to be checking your networking, you seem to say you DIDN'T change the port your softphone uses (you said you couldn't). I expect failure. |
| 14:29.39 | [TK]D-Fender | your setup has issues. |
| 14:29.54 | [TK]D-Fender | Doesn't matter which end changes but you are not making them fit. |
| 14:30.29 | [TK]D-Fender | And I'm tired of going over the same point repeatedly on this. |
| 14:30.38 | moneer | D-Fender: what is the default port for iax server? |
| 14:30.40 | [TK]D-Fender | goes to work on other matters |
| 14:30.51 | jzaw | i posted that above |
| 14:31.28 | moneer | ok guys i knew i'm newbei in this |
| 14:31.36 | moneer | i have to check it further |
| 14:31.47 | moneer | any way thnaks for your help and your time |
| 14:31.52 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 14:31.56 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 14:32.12 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:41.01 | *** join/#asterisk lcat (~lcat@187.45.253.153) |
| 14:41.06 | kaldemar | moneer: asterisk listens on port 4569 for IAX2 by default. |
| 14:42.01 | moneer | kaldemar: thanks for that, and 5060 for sip right? |
| 14:42.08 | kaldemar | yes. |
| 14:42.55 | moneer | what is the best ubuntu iax2 softphone |
| 14:51.59 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
| 14:52.50 | p3nguin | Fucking storms... blew up a bunch of important things. |
| 14:55.31 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 14:57.19 | plundra | Hmm, what's exit code 135? Grepping through the source I can't really find anything. |
| 14:57.58 | *** join/#asterisk Cadey (5a9804ea@gateway/web/freenode/ip.90.152.4.234) |
| 14:58.24 | plundra | Because asterisk just died using/with that. (And imeditaly got started again by upstart, thankfully). But I'd like to know the cause, because it came out of nowhere. |
| 14:59.04 | Cadey | HI guys, does asterisk use a SIP notify to update the callerID when a call is transfered (Attended). I have seen this issue with Aastra 57i phones but it also present in using Cisco SPA200 range SIP phones |
| 15:00.30 | r0m|u | p3nguin: that sucks |
| 15:00.50 | r0m|u | a storm came by here two weeks ago and did some crazy damage |
| 15:01.16 | r0m|u | p3nguin: I still have my old job just moving up to bigger and better things |
| 15:01.49 | r0m|u | soing the same thing with very little pay race just really does not cut it |
| 15:01.57 | r0m|u | s/soing/doing/ |
| 15:02.19 | p3nguin | Fatality count so far: two switches, one NIC, one phone, one TV, one light bulb, and possibly one UPS battery. |
| 15:02.53 | r0m|u | p3nguin: damn :( |
| 15:03.19 | r0m|u | p3nguin: something similar happen to me. the lightning came threw the comcast line and zapped everything connected to the network. |
| 15:03.41 | WIMPy | That's the good thing about being in the middle of the city. |
| 15:03.48 | r0m|u | from TV's to systems and such. |
| 15:05.28 | p3nguin | The good thing about being in the city is that I get lightning? |
| 15:06.25 | r0m|u | wonders... |
| 15:06.58 | *** join/#asterisk timahvo1 (~rogue@41.81.154.225) |
| 15:07.04 | p3nguin | The lightning hit and it sounded like my big rifle. |
| 15:07.22 | p3nguin | The crack was so sharp. |
| 15:07.37 | r0m|u | damn. |
| 15:10.04 | p3nguin | I assume you are familiar with the crack from a supersonic round. |
| 15:10.14 | r0m|u | Yes. |
| 15:10.45 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 15:11.29 | WIMPy | Here on a block in another part of the town, people had nice sound effects a few weeks back. |
| 15:11.38 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:12.10 | WIMPy | When engeneers for the electricity provider tried to find a faul by using 3kv test voltage and forgot to disconnect the households first. |
| 15:13.14 | p3nguin | ewww |
| 15:22.57 | r0m|u | nasty |
| 15:25.58 | *** join/#asterisk jetlag (~jetlag@pool-71-168-199-149.cmdnnj.east.verizon.net) |
| 15:26.03 | p3nguin | I'm still not sure how it got into my stuff with surge protectors on everything. |
| 15:26.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:26.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:27.02 | *** join/#asterisk Roadblock_RVA (~Roadblock@office.neteasyinc.com) |
| 15:39.48 | *** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com) |
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| 15:48.36 | beek | p3nguin: You need to have multiple surge protectors. Start at the service with a meterbase mounted one. Add one for the panel. Then put individuals on the more sensitive devices. |
| 15:50.14 | p3nguin | Would a surge protector at the meter loop be something the power company would need to provide? |
| 15:53.09 | p3nguin | It's very irritating that the surge ran across the Ethernet cable. It killed two switches, the jack on a Cisco phone, and a NIC. |
| 15:53.34 | p3nguin | And that's just what I have found so far. I haven't tested all connected equipment yet. |
| 15:54.15 | ketas | surge protectors are fscking expensive |
| 15:54.35 | ketas | WIMPy: wtf?! |
| 15:54.39 | WIMPy | I'd rather ask how it cam in to the ethernet cable. |
| 15:54.55 | ketas | lightning? |
| 15:55.07 | WIMPy | yes |
| 15:55.15 | ketas | then don't be suprised |
| 15:55.29 | p3nguin | It could have gotten onto the ethernet cable through any of the electronics connected to either switch. |
| 15:55.41 | p3nguin | computers, phones, print servers |
| 15:55.48 | beek | p3nguin: They come in two flavors... one that the power company installs via pulling the meter, plugging it in and then seating the meter into it. Another option is to have an electrician install one on the load side of the meter. |
| 15:56.11 | beek | p3nguin: You probably already know this but surge protectors are one-shot devices. |
| 15:56.26 | p3nguin | In a building that I do not own, I'd guess the power company's protector would be the way to go. |
| 15:56.44 | ketas | beek: some have changeable elements |
| 15:56.49 | beek | And they only are capable of taking so much power. |
| 15:56.51 | ketas | beek: but basically yes |
| 15:56.53 | beek | ketas: True. |
| 15:56.58 | beek | works for a power company. |
| 15:57.17 | p3nguin | If it just has a varistor in it, I can change those if they blow. |
| 15:57.33 | p3nguin | So far, I haven't found any blown out surge protectors. |
| 15:57.37 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 15:57.39 | ketas | lucky |
| 15:57.52 | beek | ketas: Not really. His equipment got fried instead. |
| 15:58.06 | ketas | with surge protectors? |
| 15:58.11 | p3nguin | I'd rather replace a surge protector or the components in them as opposed to hundreds of dollars worth of equipment. |
| 15:58.17 | beek | Yep. |
| 15:58.25 | [TK]D-Fender | Thank God the bullet-proof vest is unmarred! Johnny's face however.... |
| 15:58.31 | ketas | you had surge protector on every ethernet port? |
| 15:58.40 | ketas | on every device on EVERY port |
| 15:58.46 | beek | [TK]D-Fender is in good form today. |
| 15:59.05 | p3nguin | I didn't have a surge protector on any ethernet port, but I did have it on all electrical equipment. |
| 15:59.17 | ketas | nice |
| 15:59.27 | WIMPy | And what about telco lines? |
| 15:59.32 | p3nguin | I have none. |
| 15:59.33 | beek | Depending on where the lightning hit you also have the potential for high induced voltages. |
| 15:59.45 | ketas | wait, what are your external lines |
| 15:59.49 | p3nguin | It struck very close by. |
| 15:59.51 | ketas | besides power |
| 15:59.58 | p3nguin | Just power. |
| 16:00.06 | beek | And cable? |
| 16:00.09 | p3nguin | Wait, power and cable. |
| 16:00.10 | p3nguin | Yes. |
| 16:00.13 | ketas | hahaha! |
| 16:00.20 | ketas | did cable had surge protector |
| 16:00.27 | p3nguin | But the modems are not harmed. |
| 16:00.40 | ketas | but did it have one? |
| 16:00.41 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 16:00.50 | p3nguin | I didn't have suppressors on the coax. I kind of left that up to the cable company. |
| 16:01.01 | ketas | surge may as well go through the devices |
| 16:01.12 | ketas | they can fail later, mysteriously |
| 16:04.05 | ketas | if you have somewhat expensive equipment, maybe you can find way to bind (i mean virtually) at least all external lines together |
| 16:04.11 | ketas | might keep damage on edge |
| 16:04.51 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 16:05.08 | ketas | lightning once strike close to my home, i heard some snap around house and adsl link went down |
| 16:05.21 | ketas | it had one of those fine weak protectors on |
| 16:05.36 | beek | To top it off, if your grounds aren't bonded together like ketas said then you can get ground loops. They can be really nasty when you are dealing with the power of a lightning strike. |
| 16:05.40 | ketas | maybe it helped and no damage was done to anything |
| 16:06.09 | ketas | i mean, everything still works |
| 16:07.22 | ketas | p3nguin: is coax ground connected to power ground? |
| 16:07.33 | p3nguin | I have no idea. |
| 16:07.55 | ketas | it's often connected to ground and sometimes even required |
| 16:07.58 | p3nguin | I would imagine that the coax ground is the earth outside. |
| 16:08.28 | ketas | is power and coax overhead? |
| 16:08.38 | p3nguin | no |
| 16:08.58 | p3nguin | Everything is underground. |
| 16:09.47 | ketas | that can help but is no guarantee on lightning cases |
| 16:12.57 | coppice | sometimes you find your protectors intact.... but they've landed two streets away :-) |
| 16:13.35 | p3nguin | :) |
| 16:18.22 | *** join/#asterisk w9sh (~chatzilla@64-238-96-125.corp.cbeyond.net) |
| 16:20.58 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 16:21.45 | *** join/#asterisk clintc (~clintc@n128-227-215-193.xlate.ufl.edu) |
| 16:21.52 | *** part/#asterisk clintc (~clintc@n128-227-215-193.xlate.ufl.edu) |
| 16:22.13 | akrohn | I have a cell phone that calls a DID on my asterisk box. the auto attendant answers and forwards the call to another cell phone. is there any way I can get asterisk to drop out of the middle of that call once it's connected? |
| 16:23.09 | beek | I'm thinking not. |
| 16:24.16 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 16:24.23 | ollii | reboot |
| 16:24.24 | beek | waves to Katty |
| 16:24.29 | ollii | repeat this step until it works |
| 16:24.38 | Katty | hugs beek |
| 16:24.44 | Katty | ollii: what is reboot?!?? |
| 16:24.52 | WIMPy | Katty: Use a bigger hammer. |
| 16:24.53 | ollii | plug off the ac cord |
| 16:25.01 | Katty | what is ac cord?! |
| 16:25.04 | Katty | and how to plug off??? |
| 16:25.05 | *** join/#asterisk irroot (~gregory@197.169.187.99) |
| 16:25.09 | WIMPy | akrohn: How do you get the call and how do you send it out again? |
| 16:26.09 | ollii | grab your hand around the black cord and pull so hard till its out |
| 16:26.11 | Katty | telnet miku.acm.uiuc.edu ^___________________^ |
| 16:26.11 | Nugget | telnet is eeeeeeevil! |
| 16:26.15 | akrohn | WIMPy, i send it out using our macros that eventually do a Dial()... not sure what you mean by 'how do i get it' |
| 16:26.16 | Katty | Nugget: i love you. |
| 16:26.32 | fenrus | Katty, neat. :D |
| 16:26.34 | Katty | or telnet -t if you're on winders. |
| 16:26.40 | [TK]D-Fender | Katty, I was about to copy that to your wall post :) |
| 16:26.58 | WIMPy | akrohn: What technology are you using? |
| 16:27.06 | Katty | i have to figure out where that came from |
| 16:27.08 | Katty | i bet it was from reddit |
| 16:27.16 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 16:27.20 | p3nguin | Maybe I should take this opportunity to upgrade my 7960 to a 7970, and then replace the blown up 7940 with my retired 7960. |
| 16:27.50 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 16:28.10 | akrohn | WIMPy, the * box does everything in SIP. our core voip router handles our PRIs |
| 16:28.19 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 16:29.02 | p3nguin | Anyone here using a Cisco 7970? |
| 16:29.17 | WIMPy | akrohn: So if that core voip router knows reinvites, you can at least get rid of the media stream. |
| 16:31.37 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 16:31.48 | akrohn | fascinating WIMPy. I'm going to check that out |
| 16:32.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:33.00 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-203.mobile.uci.edu) |
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| 16:36.04 | *** join/#asterisk bipul (~root@unaffiliated/bipul/x-4918593) |
| 16:36.57 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:38.00 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 16:38.00 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:39.39 | Cadey | Hi guys, with asteriskNow how do you setup a digium PRI card, I have ran dahdi_test, dahdi_scan etc and i can see teh card |
| 16:40.00 | Cadey | but I have no idea how to actualy get the links up? |
| 16:40.44 | ollii | dahdi_genconf could be a start |
| 16:41.18 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-203.mobile.uci.edu) |
| 16:41.51 | angler | Cadey, you may want to ask in the #asterisknow channel. |
| 16:43.23 | [TK]D-Fender | I don't believe genconf has a clue on how to set up digital cards.... |
| 16:48.05 | *** join/#asterisk vipkilla (~t_dot_zil@unaffiliated/t-dot-zilla/x-2830497) |
| 16:48.10 | Cadey | dahdi_genconf does nothing |
| 16:48.22 | Cadey | just drops me back to a command line |
| 16:48.27 | Cadey | with no errors |
| 16:49.46 | WIMPy | Then try dahdi_cfg -v |
| 16:50.57 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:50.57 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 16:51.11 | angler | malcolmd, howdy |
| 16:52.42 | Cadey | ok |
| 16:52.46 | Cadey | that output the channels |
| 16:57.05 | WIMPy | Is there any config option I didn;t find for chan_dahdi that enables it to recogize keypad infos? |
| 16:57.09 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
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| 17:28.56 | jrbaldwin | what is the best /cheapest way to add a sms gateway to a local server, so users can send / receive text updates to the server (via python + sql) |
| 17:32.50 | [TK]D-Fender | jrbaldwin, Something that has nothing to do with * |
| 17:34.59 | jrbaldwin | [TK]D-Fender: ah sorry, but i didn't know where to ask...do you think a shell based sms gateway be achieved with a gsm usb modem and kannel software or? |
| 17:36.23 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 17:36.32 | WindBack | Hi, I have a spa942 phone. When I leave the call forwarded from this phone, I'm receiving a 302 Moved Temporaly message in Asterisk. Then I see a local channel doing the call to the forwarded destination. Is there any way to know from the dial plan the channel who originates the call forwading (in this case the SPA942)? |
| 17:39.34 | [TK]D-Fender | WindBack, Highly doubt it, but take a look at the ${BLINDTRANSFER} var. They might have tacked it on in there... |
| 17:42.41 | WindBack | [TK]D-Fender: thanks, I will look it |
| 17:44.38 | Nugget | huggles Katty |
| 17:58.41 | Qwell | Nugget: hey, guess what |
| 18:10.27 | *** join/#asterisk oliver1 (~oliver@manz-590f1ac1.pool.mediaWays.net) |
| 18:12.15 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:12.52 | WindBack | [TK]D-Fender: The ${BLINDTRANSFER} variable doesnt contain that information |
| 18:13.15 | WindBack | [TK]D-Fender: There is no varible to get that information. |
| 18:15.06 | WindBack | [TK]D-Fender: I want that information. because some people leave the phone forwareded to the mobile Phone and they want the forwareded call with the CID of their extensions |
| 18:15.48 | [TK]D-Fender | WindBack, Then tell them to stop doig it at the phone level and do it in the dialplan instead. |
| 18:16.32 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:16.37 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:16.56 | WindBack | [TK]D-Fender: Do you mean to stop doing the call forwarding in the phone a to do in the dialplan? |
| 18:17.19 | [TK]D-Fender | Funny that looks exactly like what I just said :) |
| 18:17.35 | Nugget | Qwell: what's up? |
| 18:17.44 | Qwell | nothing, just playing with telnet |
| 18:17.48 | Qwell | aww |
| 18:17.55 | WindBack | [TK]D-Fender: I |
| 18:18.12 | WindBack | [TK]D-Fender: Yes, I thought that posibility.. Thanks |
| 18:18.17 | WIMPy | Telnet? Wasn't that evil? |
| 18:18.50 | WIMPy | Doing diversions on a terminal is a pretty stupid idea anyway. |
| 18:21.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:21.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:22.13 | *** join/#asterisk elperepat (~elperepat@76-10-149-61.dsl.teksavvy.com) |
| 18:24.04 | elperepat | Hi! I'm new here... Just reinstalled asterisk in my home server and have a problem with the "system" command in my extensions.conf. Anybody wanna help? |
| 18:26.49 | p3nguin | core show application System |
| 18:27.00 | p3nguin | (from the Asterisk CLI) |
| 18:28.14 | elperepat | Yes, I know the basics, and copied/pasted a few examples fron the net. Here's what I have in my extensions: |
| 18:28.24 | elperepat | System(/usr/bin/wget -O /dev/null "http://galaxy/asterisk/inboundCall.php?number=${CALLERID(num)}&name=${CALLERID(name)}") |
| 18:29.15 | p3nguin | And the problem with that is what? |
| 18:29.22 | elperepat | The problem I have is that the dialplan hangs at that line and wget sits in my processes. |
| 18:29.48 | elperepat | it does not return. However, if I execute this command from the shell, it runs smoothly |
| 18:31.38 | p3nguin | Would it help to background it and continue on, even if the command did not succeed? |
| 18:31.53 | *** join/#asterisk DanFromUK (~IceChat77@2.30.231.89) |
| 18:32.32 | elperepat | Yes and no: the command should succeed, but indeed, if I use TrySystem, the dialplan continues, but wget never run successfully |
| 18:32.49 | DanFromUK | Hi all, I'm currently running two separate asterisk boxes (one main, one backup). Is there any way to run them live, together, and still allow call transfer, internal calls etc...? |
| 18:33.10 | DanFromUK | they are in different datacentres. |
| 18:34.37 | p3nguin | Assuming you are running asterisk as the user 'asterisk', can the asterisk user execute /usr/bin/wget? |
| 18:34.54 | elperepat | that's a good question. How could I test that? |
| 18:34.58 | elperepat | su asterisk ? |
| 18:35.09 | p3nguin | And can it successfully resolve the host 'galaxy'? |
| 18:35.52 | elperepat | it can resolve correctly because I can run this command myslelf... |
| 18:35.57 | elperepat | su asterisk |
| 18:35.57 | elperepat | This account is currently not available. |
| 18:36.12 | elperepat | How can I run a command from an other user from shell? |
| 18:37.10 | p3nguin | asterisk user should not have a good shell, so that's not going to work. |
| 18:37.31 | elperepat | sudo -u seems to work. give my 30 seconds to test... |
| 18:37.35 | p3nguin | You could set a shell for asterisk and then su asterisk. |
| 18:38.57 | jzaw | DanFromUK: thats the whole point of IAX .... inter asterisk exchange |
| 18:39.05 | elperepat | sudo -u asterisk wget -O /dev/null "http://galaxy/asterisk/inboundCall.php?number=1234567890&name=asdfasdf" |
| 18:39.05 | elperepat | returned correctly |
| 18:39.20 | elperepat | 2012-01-17 13:38:37 (2.55 MB/s) - `/dev/null' saved [229/229] |
| 18:39.44 | jzaw | DanFromUK: and your dial plans will take care of routing |
| 18:40.07 | jzaw | if busy ... route to other box etc |
| 18:41.17 | DanFromUK | jzaw: So to call a user with sip peer name of 'SIPJohn', I would do a Dial to SIP/SIPJohn,SIP/SIPJohn@Server2 ? |
| 18:42.37 | [TK]D-Fender | DanFromUK, No, you dial a number at the other server. You can't dial it's devices directly. |
| 18:42.46 | [TK]D-Fender | DanFromUK, Dialplan = all |
| 18:42.51 | DanFromUK | Ah, ok |
| 18:43.00 | jzaw | ive got two boxes |
| 18:43.14 | jzaw | all the numbers of the extensions on one are 2xxx |
| 18:43.21 | jzaw | and all the numbers on the other are 3xxxx |
| 18:43.38 | DanFromUK | Is there any point implementing OpenSER? |
| 18:45.05 | [TK]D-Fender | DanFromUK, For you? Who knows. |
| 18:45.21 | [TK]D-Fender | DanFromUK, State a need and we'll stats the means. |
| 18:45.48 | DanFromUK | Its a small hosted pbx platform for a handful of companies |
| 18:46.19 | DanFromUK | I need it to be highly available. |
| 18:46.23 | jzaw | also DanFromUK look up how to use dundi ... nice way to call each other on different boxes |
| 18:46.34 | [TK]D-Fender | DUNDi != HA |
| 18:47.00 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
| 18:47.09 | DanFromUK | How do you load balance the remote sipphones between the boxes? DNS entries? |
| 18:47.15 | [TK]D-Fender | Food for thought |
| 18:47.36 | DanFromUK | Saw that page, but its not changed for a long time. |
| 18:47.36 | [TK]D-Fender | dnfNow you'er entering into OpenSER territory... |
| 18:48.07 | [TK]D-Fender | Round-robin DNS, SIP proxy for enpoint, limiting what roles * has in your routing & processing, etc. |
| 18:49.40 | Nugget | hee |
| 18:50.10 | *** join/#asterisk irroot (~gregory@197.174.135.33) |
| 18:52.17 | DanFromUK | Thanks for your help. I'll try it out. |
| 18:55.02 | *** part/#asterisk irroot (~gregory@197.174.135.33) |
| 19:03.49 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 19:04.12 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:04.12 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 19:10.25 | *** join/#asterisk krotos (~d3v1l@host26-34-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:10.27 | krotos | hi all :) |
| 19:11.10 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 19:12.00 | krotos | i've got two account on the same provider ( same ip, same port) the configuration is register + peer (type= friend, insecure=port,invite) and incoming call fall into the last peer declared in sip.conf. I'know that if i change port ( and my provider support another port for signaling) i can solve the problem |
| 19:12.33 | krotos | but in this case, my provider offer only port 5060 for signaling, so i dont see a good solution for this |
| 19:20.40 | p3nguin | "UNKNOWN NAME" <unavailable> |
| 19:20.45 | p3nguin | Gee, that's useful. |
| 19:21.22 | p3nguin | What is the file and the location of the settings configured by make menuselect? |
| 19:23.32 | p3nguin | I guess menuselect.makeopts in the main src dir, but what about menuselect.makedeps, menuselect-tree, and the menuselect directory? |
| 19:27.55 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 19:30.51 | p3nguin | Do I just need to copy over menuselect.makedeps and menuselect.makeopts? |
| 19:31.45 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 19:32.47 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 19:33.16 | *** join/#asterisk edwin_quijada (~edwin_qui@186.120.66.86) |
| 19:33.36 | hudony | hi there : using asterisk with cisco spa504G and when speaking, there is no "comfort noise", just like VAD (and CNG was deactivated) was activated but it is not |
| 19:33.40 | sereal-work | Hello, i'm having a strange problem with trunking to a broadsoft system. I can dial certain numbers, like what appears to be extensions inside the PBX there, but dialing numbers outside the PBX simply hang. I had thought it was a issue with broadsoft but registering a softphone works fine. |
| 19:34.31 | sereal-work | Has anyone had similar issues with broadsoft systems? I figured out I need to set callerid - which allowed me to dial what I think are extensions inside the broadsoft pbx. |
| 19:34.54 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:36.31 | p3nguin | Oh, I see something about this in the book. Maybe that will help me. |
| 19:38.55 | edwin_quijada | i have a T1 working but now it card dowsnt sync with my telco. I have a Red alarm , change the cable and check everything but can not sync with telco. I am using dahdi and 1.4.38 |
| 19:39.20 | edwin_quijada | there is an app to run a diagnostic ? |
| 19:39.41 | edwin_quijada | I use lspci -vv and the card is here and it blink in red |
| 19:41.15 | p3nguin | It indicates that I would only need to copy menuselect.makeopts. I guess I should forget about menuselect.makedeps. |
| 19:41.42 | edwin_quijada | any idea where continue ? |
| 19:44.21 | Katty | naps |
| 19:44.37 | Katty | why do i always turn into a zombie around 2 |
| 19:45.21 | WIMPy | edwin_quijada: Either you've got a configuration issue at a very low level, i.e. dahdi/system.conf or you don't have connectivity. |
| 19:45.40 | p3nguin | What's the deal with .asterisk.makeopts that I found in a mailing list thread? "you now should have a" |
| 19:45.44 | p3nguin | file called menuselect.makeopts. Copy this file to your $HOME but |
| 19:45.49 | p3nguin | grr... |
| 19:46.21 | p3nguin | "you now should have a file called menuselect.makeopts. Copy this file to your $HOME but make sure it's called .asterisk.makeopts" |
| 19:46.43 | p3nguin | Whassupwiddat? |
| 19:47.17 | Katty | maybe it requires a nap |
| 19:47.21 | Katty | it is 2 afterall |
| 19:47.24 | Katty | yes? |
| 19:47.43 | p3nguin | no |
| 19:48.01 | p3nguin | Still have 12 minutes until 2pm. |
| 19:48.14 | edwin_quijada | WIMPy: Conectivity I have |
| 19:49.06 | edwin_quijada | respect a dahdi/system.conf i check again and I didnt see anything weird |
| 19:49.18 | edwin_quijada | I can post dahdi/system.conf |
| 19:49.47 | *** join/#asterisk timahvo1 (~rogue@41.81.243.23) |
| 19:49.58 | WIMPy | You have to find out if it fits your line. |
| 19:56.15 | [TK]D-Fender | sereal-work, Provide actual configs & call debug for us to look at. |
| 19:59.46 | sereal-work | [TK]D-Fender, i'm trying to find what is actually useful. my configs are very similar to http://www.commpartners.us/jht2/cpbs.html |
| 20:00.16 | [TK]D-Fender | sereal-work, "similar" is of no help. We need to see actual configs and actual debug/ |
| 20:00.59 | sereal-work | well for sip.conf mine is exactly that except I have outboundproxy and my own sip data. |
| 20:01.14 | sereal-work | "failed to extend from 1024 to 1321" is the weird thing i'm getting in the logs. |
| 20:01.53 | Katty | p3nguin: :< |
| 20:01.55 | sereal-work | it hangs on the call, doing nothing, the phone i'm calling doesn't ring, and I finally get "No one is available to answer at this time (1:0/0/0)" |
| 20:02.49 | p3nguin | 14:02 ... Now you may nap. |
| 20:03.15 | sereal-work | - Executing [9999@from-sip:1] Set("SIP/test-000000df", "CALLERID(all)=1234567890") in new stack |
| 20:03.15 | sereal-work | <PROTECTED> |
| 20:07.04 | Katty | infobot: nap? |
| 20:07.04 | infobot | nap is probably a command line napster client, at http://www.gis.net/~nite , also check out http://opennap.sourceforge.net/, or is now open source finally, or you may want to try TekNap instead |
| 20:07.20 | Katty | infobot: why u no have proper entry for nap?! |
| 20:07.20 | infobot | why not? |
| 20:07.36 | Katty | facepalms |
| 20:07.40 | Katty | infobot: fail. |
| 20:07.40 | infobot | FAIL. |
| 20:07.44 | Katty | infobot: yes. |
| 20:07.45 | infobot | rumour has it, yes is the opposite of no |
| 20:07.47 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:09.01 | Katty | o/~ rumor has it! o/~ |
| 20:09.53 | [TK]D-Fender | sereal-work, You need to look at complete calls WITH SIP debug. |
| 20:10.05 | [TK]D-Fender | sereal-work, Just dialplan apps won';t prove what's going from A to B |
| 20:10.35 | Katty | so i have this 87 page pdf i have to read |
| 20:10.41 | Katty | documentation to a new product.. |
| 20:11.14 | edwin_quijada | thyis is my dahdi/system.conf http://pastebin.com/re3ni5EV |
| 20:11.44 | Katty | puts document on head, hopes for osmosis |
| 20:14.29 | edwin_quijada | this is the system.conf and dahdi_scan http://pastebin.com/LJ3gGhXd |
| 20:14.29 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 20:14.35 | [TK]D-Fender | edwin_quijada, If you're connecting to the telco, then they should be providing timing. should be span=1,1,0 not span 1,0,0 |
| 20:15.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:15.29 | edwin_quijada | [TK]D-Fender: Ok, let me do this change even I had it before :-/ |
| 20:15.40 | edwin_quijada | I need restart the dahdi ? |
| 20:15.49 | edwin_quijada | or asterisk >? |
| 20:18.13 | Katty | infobot: blacklist |
| 20:18.13 | infobot | In etch, lenny and sid, create/edit /etc/modprobe.d/blacklist.local and add a line similar to this (without quotes): \"blacklist module_name\". IMPORTANT: ask about <blacklist-initramfs>. For sarge ask me about <blacklist sarge>. To blacklist a module at installation time, ask me about <installer blacklist>. |
| 20:18.32 | Katty | that's not what i mean |
| 20:19.44 | Katty | who can i bribe into pastebinning me a blacklist example |
| 20:19.49 | Katty | how to add one, from the CLI |
| 20:20.01 | Katty | for i am lazy |
| 20:20.06 | edwin_quijada | [TK]D-Fender: I change the line but nothing happens. I need to restart something? |
| 20:20.08 | Katty | and have the 2 oclock zombies |
| 20:20.43 | WIMPy | edwin_quijada: dahdi_cfg |
| 20:21.00 | ChannelZ | restart it all |
| 20:21.30 | ChannelZ | stop asterisk, restart DAHDI, start asterisk |
| 20:21.42 | [TK]D-Fender | ediYes, * & dahdi completely |
| 20:21.46 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 20:22.38 | p3nguin | Why do these IDIOTS sell IP phones on ebay without handsets? Like handsets aren't a crucial part of the phone or something? |
| 20:23.02 | WIMPy | Not if you have headsets. |
| 20:23.05 | WIMPy | a |
| 20:23.12 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 20:24.10 | p3nguin | So you use a phone base with an empty handset cradle when you use a headset? |
| 20:24.38 | p3nguin | They came from the factory with handsets. |
| 20:24.38 | edwin_quijada | WIMPy: this is http://pastebin.com/DeLguW7E |
| 20:24.41 | WIMPy | No, but it wouldn't really make a difference. |
| 20:24.44 | jzaw | which ip phone maybe id like one ? |
| 20:25.53 | WIMPy | edwin_quijada: It's loaded ok. Now do dahdi and your telco like each other? |
| 20:26.54 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:33.45 | p3nguin | _corey_: Do you use any CP-7970G phones? |
| 20:34.18 | _Corey_ | p3nguin: I think I have a customer with a few live ones. i have a few in a cabinet down in our lab |
| 20:34.43 | p3nguin | I was just wondering if there is any issue with them and Asterisk. I'd be using SCCP. |
| 20:34.54 | edwin_quijada | WIMPy: Now, I have alarm from my telco DS1 and ALM |
| 20:35.00 | _Corey_ | p3nguin: I think we've used both SCCP and SIP without incident |
| 20:35.08 | edwin_quijada | I dont have alarm from asterisk |
| 20:35.15 | edwin_quijada | now |
| 20:36.19 | p3nguin | Lightning wiped out the Ethernet port on a 7940 I have, so I was thinking of retiring my 7960 to replace that burned phone and getting myself a 7970. |
| 20:36.45 | p3nguin | I can't decide. It's an extra expense, so I may wait... I still have two switches and a NIC to replace, too. |
| 20:37.28 | _Corey_ | p3nguin: I'm told they weren't the easiest to image from SCCP to SIP and that there were some MWI quirks with SIP. If you're still running SCCP that'd probably be smoother |
| 20:37.55 | _Corey_ | p3nguin: If you want something colorful that's nice, I've had one of the new Polycom VVX500s on my desk for a couple weeks |
| 20:38.10 | _Corey_ | They're like $250 or something like that |
| 20:39.45 | p3nguin | As much as a want a fancy phone, I should probably save the money and replace the 7940 with a spare 7940. |
| 20:40.02 | p3nguin | s/a /I / |
| 20:40.14 | _Corey_ | :) |
| 20:43.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 21:03.01 | *** join/#asterisk SupYoshi (~SupYoshi@ip51cc8577.speed.planet.nl) |
| 21:03.14 | SupYoshi | Hey can someone help me with my error in CLI? I get http://pastebin.com/qaDs33b6 this when I try to make a call |
| 21:05.52 | [TK]D-Fender | https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/887998 |
| 21:06.58 | SupYoshi | sweeet |
| 21:07.03 | SupYoshi | So how do I fix it? NOT? :P |
| 21:07.39 | SupYoshi | or should I provide the outfit there??? |
| 21:09.04 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:10.08 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
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| 21:22.16 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:25.45 | pigpen | I have a taboo question: Using asterisk 1.8.7 and an Audiocodes FXO: If the telco has call waiting setup on the line, can it be utilized by the asterisk/audiocdes worth a dam. |
| 21:26.03 | pigpen | I have always avoided call waiting as it never really had a place in business IMHO. |
| 21:26.27 | pigpen | My gut tells me, disable call waiting on the line, then pickup a second line. |
| 21:26.37 | WIMPy | Asterisk can't do such things. |
| 21:26.44 | pigpen | But I would like to hear from your gut. ;-) |
| 21:26.53 | pigpen | WIMPy, yeah, I have never heard of it. |
| 21:26.58 | WIMPy | Which is one of the biggest issues IMHO. |
| 21:27.13 | pigpen | so the Audiocodes -might-, but Asterisk, no. |
| 21:27.41 | WIMPy | Don't know about the Audiocodes, but probably not. |
| 21:28.15 | WIMPy | That's not easily done on POTS. |
| 21:28.49 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-151-136.w83-192.abo.wanadoo.fr) |
| 21:29.23 | pigpen | yeah, I am on the same line of thought. But before I said, "no" I figured I would at least get some opinions. |
| 21:31.20 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:33.48 | _Corey_ | pigpen: Sounds like a mess... :) I think you'd be able to rig something on DAHDI trunks but the implementation would involve users doing something to trigger the Flash() application when they hear the beep |
| 21:34.31 | pigpen | Na, not a big deal. It is one location out of 380 that has this. Easy enough to say, "ah, no." |
| 21:35.01 | _Corey_ | the least ugly solution would probably involve features.conf and some kind of inband-dtmf triggered thing but, imho a mess |
| 21:35.06 | pigpen | If the cookie can't be cut the same, then it doesn't need to be cut. |
| 21:35.11 | _Corey_ | lol |
| 21:35.15 | pigpen | yeah, not going there. |
| 21:39.42 | SupYoshi | ugh -.- |
| 21:39.46 | SupYoshi | sutpid ubuntu bug |
| 21:39.54 | pigpen | gentoo. |
| 21:40.02 | SupYoshi | ;p |
| 21:40.06 | SupYoshi | lolz whatever i hate it |
| 21:40.15 | SupYoshi | =/ i cant call now makes everything totally useless |
| 21:40.26 | navaismo | hate the distro packages |
| 21:41.08 | navaismo | use the sources/"may the source be with you" |
| 21:42.57 | davlefou | hi, |
| 21:43.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:43.46 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.1 (2011/12/30), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:44.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:53.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 22:10.25 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 22:10.29 | cusco | hi |
| 22:10.36 | cusco | found something weird today |
| 22:10.38 | cusco | http://pastebin.com/v6VvZFFN |
| 22:11.10 | cusco | a bit from full log file |
| 22:12.01 | eZz | manager set debug off and forget about this |
| 22:12.26 | eZz | looks like this is a regular attack to ami |
| 22:12.52 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 22:58.38 | *** join/#asterisk doug (doug@breakout.horph.com) |
| 22:58.58 | doug | hm, since ugprading to a newer asterisk, i don't get caller id on my phone no mores |
| 22:59.46 | doug | that is, i don't see callerid when i get an incoming call |
| 23:00.44 | navaismo | pstn callerid or internal callerid? |
| 23:00.48 | doug | at least not on the iax smartphone i'm using |
| 23:00.53 | doug | navaismo: either |
| 23:01.04 | *** join/#asterisk rhamnett (~rick@5e094092.bb.sky.com) |
| 23:01.55 | navaismo | weird in the internal case, what do you see? "Asterisk" only |
| 23:02.46 | doug | (unknown) |
| 23:03.32 | doug | which may be what the softphone fills in |
| 23:03.43 | navaismo | do you use conf files and set the callerid in the sip.conf? |
| 23:03.59 | doug | yup (or iax.conf) |
| 23:04.48 | navaismo | iax2 show peer <peer> show the right callerid? |
| 23:05.21 | doug | totally |
| 23:05.37 | doug | <PROTECTED> |
| 23:06.15 | Nugget | pfft that can't be real. Surely "Your Mama" is a 900 number. |
| 23:06.27 | navaismo | LOL |
| 23:07.43 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 23:10.33 | navaismo | <PROTECTED> |
| 23:12.23 | doug | i run a nonprofit sex line |
| 23:12.44 | WIMPy | lol |
| 23:13.00 | doug | hm, how's the iax part make calls? |
| 23:17.10 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
| 23:20.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pdpofgzzlhtpgssi) |
| 23:21.02 | navaismo | doug: http://imageshack.us/photo/my-images/715/screenshot1hd.png/ |
| 23:23.18 | *** part/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
| 23:23.49 | doug | hm, a little lower level than i was thinking. |
| 23:24.23 | doug | i was wondering more how the generic asterisk call switching stuff interfaced with the iax routines to place the calls to my softphone |
| 23:25.40 | navaismo | your phone has an IP addres the asterisk contact that IP addres asking for your peer and send the call to it |
| 23:26.19 | doug | i'm really wondering what the call stack looks like when a call is placed |
| 23:26.28 | doug | cuz i figure caller id will flow through the arguments there |
| 23:26.58 | doug | although actually, navaismo, i'd guess that asterisk doesn't contact the ip address, but rather triggers a channel. |
| 23:27.17 | navaismo | iax2 set debug peer <peer> |
| 23:27.21 | doug | it doesn't have inbound access to my softphone, so it'll have to rely on an existing tcp session that was initiated from the smartphone side earlier |
| 23:27.22 | navaismo | to see the call details |
| 23:27.34 | doug | cool, i'll try that. |
| 23:28.05 | doug | looks like i was about 6 months behind the current tip of svn, so i'm recompiling now. |
| 23:31.35 | *** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net) |
| 23:41.51 | *** join/#asterisk devcoder (~leemelnyk@216.18.244.34) |
| 23:43.34 | devcoder | hows it going everyone |
| 23:43.44 | doug | shitty |
| 23:44.04 | devcoder | why is that? |
| 23:44.27 | *** join/#asterisk jkroon (~jkroon@dsl-244-35-76.telkomadsl.co.za) |
| 23:44.27 | WIMPy | It's called "life". |
| 23:44.36 | devcoder | yeah i hear ya there |
| 23:45.15 | devcoder | so anyone deal with registration of a soft phone running on a cellphone over verizon to a natter asterisk box? |
| 23:45.53 | doug | that's two natted endpoints? |
| 23:46.18 | devcoder | i get it to work actually, but keep getting a error about a critical packet then hangs up, even thought i have two way audio communication working |
| 23:46.37 | devcoder | after about 30 or so seconds that is |
| 23:46.55 | devcoder | trying to figure out if i need to open up a port on the firewall or what |
| 23:49.19 | doug | could try it. that's the first thing i'd do. |
| 23:49.46 | doug | urgh, ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name); |
| 23:49.57 | doug | i guess struct ast_channel doesn't have a name field any longer |
| 23:50.26 | devcoder | [Jan 17 19:04:14] WARNING[26886]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 7V-wMBrs39Rk4DSE65AheBGYcaXQlDk--S for seqno 5985 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 23:50.26 | devcoder | Packet timed out after 63937ms with no response |
| 23:50.26 | devcoder | [Jan 17 19:04:14] WARNING[26886]: chan_sip.c:3651 retrans_pkt: Hanging up call 7V-wMBrs39Rk4DSE65AheBGYcaXQlDk--S - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 23:50.56 | doug | chan->__do_not_use_name, nice. |
| 23:51.18 | navaismo | devcoder: do you set the: externhost/externip, localnet, and the peers involved as NAT=yes? |
| 23:51.36 | devcoder | yes i did |
| 23:53.13 | navaismo | sip debug will be usefull |
| 23:53.42 | doug | so i guess we're up to 1.8.9.0-rc2 now? |
| 23:55.13 | devcoder | i will try that, always fun looking threw that |
