| 00:42.41 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 01:24.18 | *** join/#asterisk ZenKen (~ZenKen@99-62-61-118.lightspeed.toldoh.sbcglobal.net) |
| 01:30.23 | ChannelZ | I don't need it but am curious about TDMoE, is there like a generic driver that comes in DAHDI to do TDM over ethernet? |
| 01:32.42 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-221-190.sc.res.rr.com) |
| 01:35.36 | ChannelZ | nevermind. dahdi_dynamic |
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| 01:58.43 | *** part/#asterisk ZenKen (~ZenKen@99-62-61-118.lightspeed.toldoh.sbcglobal.net) |
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| 02:44.55 | volga629 | Unable to create channel of type 'SIP' (cause 20 - Unknown) what can cause |
| 02:47.30 | p3nguin | The device is not available. |
| 02:48.28 | volga629 | hmmm, follow me can be a problem ? |
| 02:48.36 | volga629 | Extension exist ? |
| 02:48.42 | volga629 | sorry |
| 02:49.22 | volga629 | extension is exist and has follow me with hunt for 2 cell phones |
| 02:51.33 | volga629 | <PROTECTED> |
| 03:15.51 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
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| 06:01.43 | jgowdy | Is there a reason why there's no stop command for AGI? I know you can work around it using ExternalIVR, but I'm just curious why that seems to be missing, like is there a technical reason it's difficult |
| 06:02.52 | *** join/#asterisk datruth (datruth@gurutechie.org) |
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| 06:34.31 | dijib | any HTB experts here? |
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| 06:35.40 | ChannelZ | Humm. OK I am behind a firewall at home and my remote Asterisk is behind a firewall, which is configured with externaddr and localnet. However the localnet is the same on both networks (mine at home and Asterisk's) so even though Asterisk knows my peer should be NAT, it sends RTP to 192.168.1.5 |
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| 06:36.08 | ChannelZ | HTB? |
| 06:38.25 | dijib | bit shaping |
| 06:38.28 | dijib | traffic shaping |
| 06:39.06 | dijib | ive been building a new box and im getting to the point where if i can get this dont i just need to install asterisk and then put it into production |
| 06:39.10 | dijib | but i need shaping |
| 06:47.30 | *** join/#asterisk srbh (~srbh@124.123.173.211) |
| 06:48.49 | dijib | anybdy have shorewall experience? |
| 06:49.32 | srbh | I am developing a app/plugin for asterisk which needs to allow transfer from inside of the app.I am calling app_exec("transfer","sip:abcd@mydomain.org)").REFER is being sent but the sip phone responds with 100 trying and it keeps on going this way.can someone help? |
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| 07:22.14 | iulhk | hello all, testing on asterisk-1.1.2, dahdi-linux-complete installed, trying video conference with confbridge, for this do i need dedicated machine or it can be handle at cloud hosting type machine too ? becuase in asterisk-1.4 with zaptel1. |
| 07:23.00 | iulhk | dedicated machine was required for call conference type services |
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| 08:00.51 | ChannelZ | Asterisk 1.1!? |
| 08:01.49 | ChannelZ | I hope you mean 10.1 |
| 08:04.16 | iulhk | yes |
| 08:04.18 | vlt | Hello. I have two Asterisk boxes with the same iax peer (type friend). One registers at the other and then can put calls through. But the other way round I get "chan_iax2.c:10276 socket_process: Call rejected by 10.20.0.18: No authority found". Any idea why? |
| 08:05.58 | iulhk | <ChannelZ>: yes i mean if i am using asterisk 10.1 and using confbridge, do i need dedicated machine for video conferencing or it can be setup easily on some good cloud server, bcoz by using asterisk1.4 with zaptel1.4 (ztdummy for conferences) we need dedicated machines ? |
| 08:08.37 | ChannelZ | vlt: IP for the peer on the other box wrong? not matching the right peer.. |
| 08:12.42 | ChannelZ | iulhk: if you're asking about timing, there is no longer a dependency on DAHDI (was zaptel) as either a timing source or for mixing |
| 08:13.01 | ChannelZ | so in that sense yes it can be run 'anywhere' |
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| 08:47.56 | iulhk | <ChannelZ>: do we have any calculation criteria that, by usig asterisk-10.1 in quad core or xeon server with 4GB ram can handle more than 10 simaltaneous video conferences where no transcoding involove? |
| 08:48.34 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 08:48.38 | qakhan | hi all |
| 08:49.31 | qakhan | i have a queue and 5 agents in that queue. i want to setup an alarm after 3 mins if no agent attend the call |
| 08:49.36 | qakhan | please help me |
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| 08:57.34 | qakhan | ... |
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| 09:01.06 | k3asd` | hi |
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| 09:10.27 | PBXman | hello |
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| 09:17.35 | schmidts | good morning |
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| 10:12.20 | vlt | ChannelZ: Ok, I solved it. I didn't have each other's user names in the "user=" line. Works now. Thanks. |
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| 10:20.54 | jacc0 | does anyone know if sangoma wanpipe is supporting asterisk 1.8 by now? or shoul dI use SMGv3 (as they told me months ago)? |
| 10:25.01 | jacc0 | i guess so :) http://support.sangoma.com/index.php?/News/NewsItem/View/3/dahdi-26-is-now-supported-in-the-sangoma-wanpipe-driver |
| 10:30.04 | schmidts | jacc0 why should it not support i? |
| 10:30.14 | schmidts | ah dahdi 2.6 ok ;) |
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| 10:32.38 | ijpalmer | Morning all, can anyone tell me if the relative-periodic-announce setting is available in realtime for asterisk 1.8.5 |
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| 11:49.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:04.00 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 12:06.42 | teloniusz | I'm trying to install AX-4E ISDN card according to manual shown here: http://www.inteligentneprodukty.pl/pliki/Asterisk_Card_Install_Guide20090505.pdf - debian, asterisk 1.6.1, dahdi 2.2.0.2, libpri 1.4.11 |
| 12:07.51 | teloniusz | I've successfully installed kernel module tor2, inserted it, got message in dmesg: [ 470.084958] dahdi: Telephony Interface Registered on major 196 |
| 12:07.55 | teloniusz | [ 470.084961] dahdi: Version: 2.2.0.2 |
| 12:07.57 | teloniusz | [ 484.932850] Registered Tormenta2 PCI |
| 12:08.10 | teloniusz | so all looks pretty good. Then I'm trying: |
| 12:08.21 | teloniusz | dahdi_cfg -vvvvvvv |
| 12:08.49 | teloniusz | which ends up with: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) |
| 12:09.11 | teloniusz | is there anything to try before I go asking vendor for help? |
| 12:10.53 | [sr] | isdn card will work with isdn drivers, not dahdi |
| 12:12.19 | teloniusz | [sr]: sounds logical, but manual tells me to configure it with zaptel.conf or system.conf if I use dahdi |
| 12:12.29 | teloniusz | I'm pretty confused... |
| 12:12.48 | [sr] | i may be wrong but you'll never get a isdn card to work with dahdi |
| 12:13.14 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
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| 12:24.18 | teloniusz | [sr]: this: http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html tells otherwise |
| 12:24.34 | teloniusz | [sr]: that's how my card is detected by lspci |
| 12:25.10 | [sr] | 410p ok |
| 12:25.16 | [sr] | when u say isdn |
| 12:25.24 | [sr] | you meant BRI ? |
| 12:25.30 | [sr] | isdn card's are different |
| 12:25.33 | teloniusz | PRI |
| 12:25.36 | [sr] | ah |
| 12:25.38 | [sr] | ok |
| 12:25.51 | [sr] | isdn cards, are those old ISDN modems to connect to the internet |
| 12:25.53 | teloniusz | still, hisax got loaded |
| 12:26.03 | teloniusz | do I need to balcklist it? |
| 12:26.04 | [sr] | define BRI or PRI cards |
| 12:26.14 | [sr] | your problem may be dahdi version, issssssssssss old |
| 12:27.34 | teloniusz | I'm afraid it'd be hard to upgrade it, I have a custom modified version to support another card |
| 12:27.56 | teloniusz | I could try a fresh installation |
| 12:28.15 | teloniusz | Still, kernel seems to load tor2 without any problems... |
| 12:30.48 | teloniusz | waaait |
| 12:31.27 | teloniusz | I tried to ignore all vendor's advices and used wct4xxp |
| 12:31.33 | teloniusz | looks promising... |
| 12:33.53 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 12:36.58 | ijpalmer | Afternoon all, can anyone tell me if the relative-periodic-announce setting is available in realtime for asterisk 1.8.5 |
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| 13:22.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:01.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:35.18 | jacc0 | hi all, I'm about to make my first asterisk 10 install from source; anything I should know before I start? |
| 14:37.10 | [TK]D-Fender | Read the instructions |
| 14:37.13 | [TK]D-Fender | Then follow them |
| 14:39.32 | ijpalmer | Afternoon all, can anyone tell me if the relative-periodic-announce setting is available in realtime for asterisk 1.8.5 |
| 14:40.37 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 14:48.11 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 14:51.58 | Rufus | hello. anyone ahs any idea how to reset admin password for web interface on a linksys spa921 ip phone please? |
| 14:53.31 | [TK]D-Fender | Rufus, https://www.google.ca/#hl=en&sclient=psy-ab&q=SPA921+factory+reset&oq=SPA921+factory+reset&aq=f&aqi=g-v1&aql=&gs_sm=3&gs_upl=112l7844l0l8011l20l12l0l8l8l1l708l5740l5-9.1l10l0&gs_l=hp.3..0i15.112l7844l0l8012l20l12l0l8l8l1l708l5740l5-9j1l10l0&pbx=1&bav=on.2,or.r_gc.r_pw.r_cp.r_qf.,cf.osb&fp=69fa65b8621582b5&biw=1600&bih=883 |
| 14:54.17 | Rufus | so factory reset would be only solution [TK]D-Fender? was hopping I could avod that ( doing things remotely ) |
| 14:54.30 | *** join/#asterisk mutex7c (~mutex7c@HSI-KBW-095-208-202-191.hsi5.kabel-badenwuerttemberg.de) |
| 14:55.23 | mutex7c | Hello everybody. Is it possible to perform a "register" command for SIP trunk registration on the CLI ? I could not find anything like it ... |
| 14:57.10 | [TK]D-Fender | Rufus, The whole point of passwords to LOCK OUT remote access |
| 14:57.34 | [TK]D-Fender | mutex7c, No |
| 14:58.02 | mutex7c | thx :) |
| 15:03.49 | mandla | Gentlemen, i need help, when i make a call from my analog phone, DAHDI/22-1, or any other analog phone, i cant hear the ring tone although the phone i am calling is ringing. |
| 15:04.22 | mandla | I just updated dahdi to 2.6.0 |
| 15:07.01 | *** join/#asterisk libryder (~Adium@209.33.214.243) |
| 15:08.12 | libryder | can someone recommend me the best way to manage dtmf tones? we have problem with losing tones and i'm trying to figure out where to start to troubleshoot it |
| 15:08.55 | libryder | voip/sip |
| 15:10.12 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 15:10.56 | ChannelZ | rfc is usually what you want 99% of the time |
| 15:11.06 | jacc0 | @libryder; what type of dtmf mode do you use? inband,info or rfc26833? |
| 15:11.12 | ChannelZ | but it depends on your ITSP really |
| 15:11.40 | jacc0 | *rfc2833 |
| 15:11.59 | libryder | ChannelZ: everything we've seen points at the itsp but i'm not sure how to watch it from the console when we have 8 concurrent calls |
| 15:12.19 | *** join/#asterisk bbourdage (~bbourdage@75-146-129-225-SouthBend.hfc.comcastbusiness.net) |
| 15:12.43 | mandla | Gentlemen, i need help, when i make a call from my analog phone, DAHDI/22-1, or any other analog phone, i cant hear the ring tone although the phone i am calling is ringing. |
| 15:13.10 | libryder | jacc0: we are using rfc2833 |
| 15:13.12 | jacc0 | check the ring-volume |
| 15:13.26 | jacc0 | okay, you can debug rfc2833 with tshark |
| 15:13.36 | jacc0 | tshark -R rtpevent |
| 15:14.06 | tzafrir_laptop | mandla, what do you see in the output of: dahdi show channels |
| 15:14.12 | tzafrir_laptop | in the asterisk CLI |
| 15:14.27 | libryder | thanks jacc0 i'll try that |
| 15:14.32 | [TK]D-Fender | mandla, Don't ask the same question every 5 minutes. We can all see the original... You should be whoing us a complete call attempt at CLI to show where this is happening with debug on the channel you Dial out as well. |
| 15:15.27 | ChannelZ | mandla: you're using an analog phone, connected to what? And where does the call go from there? (SIP?) |
| 15:16.22 | jacc0 | you could also trun on dtmf debug in asterisk cli. but I always use tshark |
| 15:16.30 | mandla | tzafrir_laptop, http://pastebin.com/Uw5J05B2 |
| 15:17.13 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 15:17.35 | mandla | ChannelZ, Im using an analog phone connected to the astribank, the call comes from the DAHDI Ports. |
| 15:17.47 | tzafrir_laptop | mandla, what do you call through? ISDN? |
| 15:18.02 | ChannelZ | you said "when I make a call from my analog phone" |
| 15:18.03 | mandla | tzafrir_laptop, yes ISDN |
| 15:18.37 | mandla | ChannelZ, yes from the analog phone, the phone is connected to the DAHDI port. |
| 15:31.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:31.54 | libryder | ok i just got a failure but tshark detected the tone |
| 15:33.06 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:34.38 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:35.03 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:35.24 | jacc0 | did you configure the sip account to use rfc2833? |
| 15:35.29 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:35.39 | jacc0 | tr4y to enable dtmf debug in cli |
| 15:35.59 | *** join/#asterisk dxd828 (~dxd828@dxdservers-3-pt.tunnel.tserv10.par1.ipv6.he.net) |
| 15:38.16 | jacc0 | can someone tell him how to enable dtmf debug in cli? |
| 15:38.36 | mandla | tzafrir_laptop, http://pastebin.com/5bxFD7im |
| 15:38.38 | mandla | http://pastebin.com/5bxFD7im |
| 15:39.09 | mandla | Im getting some Null Frame when i place a call on debug mode. |
| 15:41.24 | tzafrir_laptop | mandla, what version of asterisk and of libpri do you have? |
| 15:41.55 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 15:42.27 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:42.41 | libryder | where could i find an asterisk consultant to spend a couple hours fixing this issue? i'm not really much of an asterisk system |
| 15:43.12 | Qwell | libryder: most of us aren't Asterisk systems |
| 15:43.19 | dym | :D |
| 15:43.25 | dym | i was waiting for that reply :D |
| 15:43.34 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:43.35 | libryder | hahah too early |
| 15:43.35 | jacc0 | asterisk consultants can be found here |
| 15:43.38 | dym | anyways - i always thought you were, Qwell |
| 15:44.44 | jacc0 | you are already enjoying the best asterisk consultantsy there is :) I'l give you my bank account number to transfer the money to |
| 15:44.45 | jacc0 | :P |
| 15:44.52 | libryder | lool |
| 15:45.28 | jacc0 | someone, tel him how to enable dtmf debug in asterisk |
| 15:46.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:46.37 | libryder | jacc0: is it in logger.conf? |
| 15:46.54 | libryder | errr i don't have that file |
| 15:48.19 | libryder | what's the command to debug a channel in cli? |
| 15:50.35 | jacc0 | core set debug channel Zap/1-1 |
| 15:50.43 | jacc0 | for example |
| 15:51.11 | libryder | i know you can get a list of channels, using core show channels; but how do you stop it from truncating the name of the channel? |
| 15:53.59 | jacc0 | what asterik version are you using? |
| 15:54.59 | libryder | 1.6 |
| 15:55.39 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 15:56.26 | jacc0 | in asterisk 1.6 /etc/asterisk/logger.conf should be there else create one |
| 15:58.59 | jacc0 | put this in /etc/asterisk/logger.conf : http://pastebin.com/PWKBYYfJ |
| 15:59.01 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:59.19 | jacc0 | Working hours are over |
| 15:59.27 | jacc0 | I'll be gone in a sec |
| 15:59.39 | libryder | jacc0: i don't have logger.conf, create it? |
| 15:59.46 | jacc0 | yes create it |
| 15:59.51 | libryder | ok thanks |
| 16:00.15 | libryder | thanks for you all your help |
| 16:00.44 | jacc0 | restart asterisk to enable the new settings in logger.conf |
| 16:01.30 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 16:01.57 | jacc0 | I'll be back here tomorrow. If you are lucky I'll have some time to help you |
| 16:02.06 | libryder | cool thanks |
| 16:02.07 | jacc0 | bye for now |
| 16:02.09 | libryder | cya |
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| 16:49.02 | libryder | i have the following log: http://pastebin.com/ftvUG7am is there any reason why the first digit never received DTMF BEGIN? |
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| 18:22.35 | jgowdy | Is there a reason why there's no stop command for AGI? I know you can work around it using ExternalIVR, but I'm just curious why that seems to be missing, like is there a technical reason it's difficult? |
| 18:22.41 | *** join/#asterisk serafie1 (~erin@nat/digium/x-smjmmluaddkghftk) |
| 18:25.04 | *** join/#asterisk ThiagoCMC (~martinx@187.75.230.175) |
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| 18:29.13 | ThiagoCMC | Hi! I need some help here! My asterisk stops working... at asterisk console I see: "Span 1: Channel 0/1 got hangup, cause 27" "DAHDI/i1/XXXXXXXX-2 is circuit-busy"... Can somesone help me?! I have a sangoma A104 card, no alarms ON. |
| 18:29.49 | ThiagoCMC | all was working smoothly 3 hours ago... no server changes was made. |
| 18:30.24 | ThiagoCMC | in outgoing calls: status is 'CONGESTION' |
| 18:30.30 | ThiagoCMC | but is all free! |
| 18:32.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:39.43 | navaismo | define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27 |
| 18:40.29 | ThiagoCMC | where?! |
| 18:41.02 | ThiagoCMC | let me google it... |
| 18:41.03 | ThiagoCMC | :) |
| 18:42.00 | *** join/#asterisk Joel (~chatzilla@unaffiliated/joel) |
| 18:44.37 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 18:45.35 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:45.51 | *** join/#asterisk dtrich (~qwe@athedsl-4365051.home.otenet.gr) |
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| 18:50.13 | ThiagoCMC | that page is not very helpful... I can not understand why it stops working from nothing... no alarms, pri okay, dahid status ok... :( VoIP complete offline... my boss is about to kill me... lol |
| 18:50.19 | ThiagoCMC | :/ |
| 18:51.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:51.51 | ThiagoCMC | all wanpipemon alarms are OFF... |
| 18:57.38 | navaismo | asterisk is ge3tting the destination out of order |
| 18:57.53 | navaismo | like if the dialed numbers are wrong |
| 18:57.57 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 18:57.59 | navaismo | so check your dial |
| 19:01.40 | ThiagoCMC | I'm dialing as usual. |
| 19:02.29 | ThiagoCMC | only internal calls (SIP <-> SIP) are working... |
| 19:03.35 | navaismo | so your issue is in your E1/PRI |
| 19:04.00 | autofsckk | i have a pap2 behing pfsense, i used siproxd but it crashed my pfsense box, now im trying to do it with firewall rules, so i only have to open and send ports 5060 and 10k-30k to my pap2's ip? is that enough? |
| 19:05.08 | ThiagoCMC | pri show spans -> PRI span 1/0: Up, Active |
| 19:05.12 | ThiagoCMC | I'll call my telco... |
| 19:10.00 | *** join/#asterisk dtrich (~qwe@athedsl-4365051.home.otenet.gr) |
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| 19:31.52 | [TK]D-Fender | autofsckk, You're not supposed to have to open port on phone-side |
| 19:33.13 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:46.16 | *** part/#asterisk libryder (~Adium@209.33.214.243) |
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| 19:51.18 | Cadey | Hi guys, any hints on how to optimise h263p - I have it working but the quality is shocking :( |
| 19:51.41 | Cadey | lots and lots of artifacts all over and sometimes its just bad data |
| 19:52.04 | Cadey | i realis over the net its anyones game but are there any "you should really do this" type settings to help |
| 19:53.03 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
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| 19:59.35 | p3nguin | Heck, I don't even have ports forwarded TO asterisk with the ITSP sitting on the outside. |
| 20:02.53 | *** join/#asterisk Fubard (~brawr@vpn.bctconsulting.com) |
| 20:04.14 | Fubard | if the .wav & .txt files in a voicemail box's folder have been deleted (not through voicemailmail for example) how can you force the mwi status to update? |
| 20:06.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:08.32 | mjordan | Fubard: pollmailboxes=yes in voicemail.conf, combined with some pollfreq value |
| 20:13.04 | ChrisInSydney | Quick question, whats the "best" ,way to manage multi row data from func_odbc call ? The query will return between 3 and 30 rows. Looping in the dial plan seems cumbersome |
| 20:14.15 | [TK]D-Fender | that's what you'v got. Otherwise use an external script |
| 20:15.59 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 20:16.26 | Dovid | OT: does anyone know if sipsak as a time out setting? say if i get no response in x amount of seconds to cancel the request? |
| 20:19.55 | ChrisInSydney | [TK]D-Fender: As I thought. So the winner is AGI() |
| 20:19.59 | ChrisInSydney | :-/ |
| 20:20.23 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 20:21.36 | ChrisInSydney | off to take Felix to Daycare. Thanks for the input |
| 20:29.16 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:30.38 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 20:30.43 | [sr] | evening mates |
| 20:30.51 | mcrownover | hi everyone |
| 20:32.10 | kaldemar | ChrisInSydney: you could maybe use func HASH. |
| 20:33.09 | *** join/#asterisk gundy (~gabe@69.169.150.25.provo.static.broadweavenetworks.net) |
| 20:34.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:35.23 | mcrownover | i am considering setting up a test pbx along side our production one - can they both use the same trunk info in and out since they are behind the same firewall? in essence can the test one make and receive live calls for DIDs and extensions that won't live on the production one? |
| 20:36.11 | gundy | mcrownover, are you registering with that GW? |
| 20:36.34 | gundy | mcrownover, or is it all IP based? |
| 20:36.38 | mcrownover | yes - the outside of the firewall is the registered IP |
| 20:36.49 | *** join/#asterisk bmg505 (~leon@196-209-72-124.dynamic.isadsl.co.za) |
| 20:37.18 | gundy | Assuming your NATing doesn't jack with you, you should be OK. Right? |
| 20:37.52 | mcrownover | i guess my FW rules would cause me some grief since they point to the one production box inside |
| 20:37.53 | ajg | hmm.. is there a way to have asterisk forward a call to an external number, like a cellphone.. then if it doesn't pick up, continue on with the dialplan? |
| 20:38.30 | gundy | mcrownover, yeah, if you're using your FW to handle NAT, that's going to have to be reviewed. |
| 20:38.36 | kaldemar | ajg: see option "timeout" in "core show application Dial" |
| 20:38.51 | ajg | kaldemar: okay thanks :) |
| 20:39.15 | ajg | it definitely just rings forever with Trixbox.. hoping stock asterisk will be a bit better about that |
| 20:39.20 | gundy | You can just have Server B use server A as a trunk and server A hits the GW outside of your network. |
| 20:40.02 | mcrownover | gundy: where can i read up on that? |
| 20:40.26 | gundy | mcrownover, I don't know, haven't done Asterisk in years. |
| 20:40.31 | mcrownover | aah |
| 20:40.45 | gundy | But, the general idea can be simple.... |
| 20:41.13 | mcrownover | i am basically looking for now to be able to dick with the phones and firmwares without screwing myself on the production box |
| 20:41.17 | gundy | Register server B to server A like you might with any other phone. |
| 20:41.38 | gundy | (look in sip.conf in each server) |
| 20:41.56 | gundy | Then setup the dial plan on A to send calls to B when it matches the DID. |
| 20:42.09 | gundy | Then tell B to send all calls to A. |
| 20:42.55 | gundy | mcrownover, that way you could have as many dev setups as you like. |
| 20:43.19 | mcrownover | thanks gundy - i will read up on it |
| 20:43.28 | gundy | And as long at the NATing and FW works for A, everyone is happy. |
| 20:43.30 | gundy | good luck. |
| 20:47.00 | Joel | ajg: you also need to be careful with voicemail 'answering'. |
| 20:47.17 | Joel | mcrownover: the best way to do what you're asking is peer the development box to the production box. |
| 20:54.01 | mcrownover | joel: can you point me where to do that? |
| 20:54.39 | Joel | mcrownover: point? |
| 20:54.45 | mcrownover | also been wondering about how to cluster asterisk servers |
| 20:54.55 | Joel | mcrownover: you need to setup peer connections using either sip or iax. |
| 20:55.12 | mcrownover | joel: bad choice of words |
| 20:55.15 | mcrownover | on my end |
| 20:55.21 | Joel | and all the necessary dialplan to route calls appropriately. |
| 20:55.32 | mcrownover | i'll search for peering two servers |
| 20:56.06 | mcrownover | i am sip to my provider |
| 20:56.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:56.20 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:58.15 | mcrownover | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 20:58.26 | mcrownover | think this is current info? |
| 20:58.40 | mcrownover | i am using asterisk 1.6 |
| 20:59.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:59.58 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 21:01.58 | leifmadsen | 1.6 isn't a version |
| 21:02.14 | [TK]D-Fender | or even a branch... |
| 21:02.16 | leifmadsen | you're either on branch 1.6.0, 1.6.1, or 1.6.2, and versions are the 4 space |
| 21:07.26 | mcrownover | please forgive - my version is 1.6.2.20 |
| 21:35.51 | brian98 | That information will work fine for you. |
| 21:36.14 | navaismo | why im seeing this msg in the cli? "Sending fake auth rejection for device" |
| 21:36.16 | *** join/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 21:36.31 | navaismo | i have alwaysauthreject=yes |
| 21:37.50 | brian98 | So it's working. |
| 21:37.56 | navaismo | asterisk 1.6.2.16 |
| 21:38.58 | brian98 | alwaysauthreject=yes prevents Asterisk from telling a sip scanner which are valid extension numbers. |
| 21:39.22 | brian98 | your box is being scanned and asterisk is sending fake auth rejections. |
| 21:39.58 | navaismo | thx, how can i do something? |
| 21:40.31 | brian98 | do what? |
| 21:40.46 | kl4m | Hi, I'm trying to put cell phones in a Queue (for on-calls), but I don't want the cell voicemail to answer. My first idea was to call the phones via Local channel + FollowMe, but it doesn't jump to the next member if the call is not accepted/denied, Even if I make sure to not Answer() the call. Does anyone have ideas for a better method? |
| 21:40.49 | brian98 | why have you got SIP open to the world? |
| 21:40.58 | brian98 | have you got strong passwords |
| 21:41.08 | navaismo | dont send fakeauth, |
| 21:41.18 | navaismo | nope its closed |
| 21:41.38 | brian98 | you don't want to send fakeauth? |
| 21:41.50 | brian98 | have you tried alwaysauthreject=no ? |
| 21:41.51 | brian98 | .. |
| 21:42.01 | navaismo | i dont want to see that msg in the cli every minute |
| 21:42.20 | brian98 | if you are seeing that message in the cli every minute you have something more going on |
| 21:42.29 | brian98 | what is causing the message |
| 21:42.40 | navaismo | sure |
| 21:42.45 | brian98 | have you looked at sip traffic on the box to see why it may be happening/ |
| 21:42.46 | *** join/#asterisk k3asd` (~k3asd@host193-87-dynamic.6-87-r.retail.telecomitalia.it) |
| 21:42.49 | brian98 | ? |
| 21:42.55 | navaismo | not yet |
| 21:42.57 | brian98 | ok |
| 21:43.00 | brian98 | good place to start |
| 21:43.15 | navaismo | ok thx |
| 21:46.01 | navaismo | i think something is wrong now after enable the sip debug i see: Correct auth, but based on stale nonce received from |
| 21:46.18 | navaismo | isnt normal right? |
| 21:46.25 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 21:50.40 | brian98 | How many sip devices? |
| 21:50.44 | brian98 | It means that a SIP device is re-using an old authentication challenge. |
| 21:50.45 | brian98 | If it still registers and can place calls, there's no problem to worry |
| 21:50.45 | brian98 | about. It's just a warning. |
| 21:51.07 | navaismo | 4 devices, again thx |
| 21:56.17 | *** join/#asterisk serafie1 (~erin@nat/digium/x-knfjlxuhetlvofbw) |
| 22:04.17 | Joel | kl4m: press to accept. |
| 22:09.56 | autofsckk | [TK]D-Fender: sorry i wasnt here, the thing is that my phone with pap2 isnt working, i dont get sound |
| 22:14.30 | *** join/#asterisk neurosys (~neurosys@c-98-254-216-32.hsd1.fl.comcast.net) |
| 22:23.34 | volga629 | mjordan: I file bug report about IMAP in free pbx, but they suspect that is asterisk problem http://www.freepbx.org/trac/ticket/5673#preview |
| 22:27.28 | mjordan | volga629: kind of. The error message is from Asterisk, but the way in which FreePBX handles its voicemail message boxes is.... weird. |
| 22:27.50 | mjordan | Disregarding the error message, is there actually a problem with the IMAP/VM functionality? |
| 22:29.34 | volga629 | no IMAP vm delivery into assigned folder and I can check and it update the status only those kind of warnings I see [2012-03-12 18:21:41] WARNING[16234] app_voicemail_imapstorage.c: IMAP Warning: Bad quota root list for VoiceMail |
| 22:29.55 | mjordan | volga629: I'm about to head out for the day, but if you want to create an ASTERISK issue referencing the FreePBX issue, that'd be fine. I imagine it will end up being a "we can tone down that error message" sort of resolution; however, FreePBX is making some pretty hefty assumptions that things working with a File backend is going to behave exactly the same as an IMAP backend. |
| 22:30.15 | mjordan | as far as the "bad quota root list" - I'll need to confirm, but I suspect that's coming from the IMAP server itself |
| 22:30.33 | mjordan | so that's probably legitimate, even if it isn't necessarily harming anything. |
| 22:31.16 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xbcjglsbrpoykooj) |
| 22:31.34 | volga629 | yes I think too those message related only to IMAP server configuration. I will open this issue in asterisk |
| 22:42.03 | autofsckk | what do i have to do to make a pap2 work behing pfsense? i already opened port 5060 and 10k-30k udp, can somebody help me please? |
| 22:45.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 23:16.56 | p3nguin | First, you don't open ports for a client device. |
| 23:18.15 | p3nguin | Second, configure Asterisk to work with phones behind a NAT. nat=yes in the general section and no other nat= settings in any peer entries.` |
| 23:19.37 | p3nguin | Ensure that you set directmedia=no so that asterisk has to stay in the media stream. |
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| 23:44.34 | linuxishawt | Hello All, I have an issue with a the following error flooding my asterisk cli and logs >>>> ERROR[14648]: res_timing_pthread.c:422 write_byte: Error writing to timing pipe: Broken pipe, I've tried googling around but couldn't find any information, i've tried to reload and restart asterisk to no avail, any advice? |
| 23:45.30 | p3nguin | Try not using pthread for timing. Try using dahdi or timerfd. |
| 23:51.08 | linuxishawt | p3nguin: Thanks...any ideas on how to get pthread working again? |
| 23:54.53 | p3nguin | Since dahdi provides a much better timing source, I wouldn't waste my time on pthread. |
| 23:55.12 | pabelanger | Yes, don't use pthreads |
| 23:55.27 | pabelanger | either dahdi or timerfd |
| 23:55.47 | p3nguin | pthread timing is bottom-of-the-barrel. You were only using it because you didn't install the right thing to begin with. |
| 23:56.54 | linuxishawt | Thanks! |