| 00:11.34 | jaytee | FYI...as of Jan.1st, 2011, YouTube, Twitter and Facebook will merge. The new program will be known as YouTwitFace. |
| 00:27.24 | Micc | I keep seeing a message to turn off comfort noise on one of my servers that is just used as a dialout from another asterisk server. And the clients connected to the other server all have VAD turned off. |
| 00:28.05 | [TK]D-Fender | Micc: Clearly one is not disabled |
| 00:28.11 | Micc | I've never seen that error on the server that the customers are actually connected to. So I'm not sure why this other server is saying it now. |
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| 01:47.08 | Micc | I think I know what device is using VAD. I don't remember enabling it and I think the default is off for handytone 502s. |
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| 01:48.46 | Puma1337 | can someone help me reconfigure my asterisk installation with curl without completely destroying my server? I am running version 1.4.21.2 |
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| 02:42.15 | cosmicwombat | cosmo1 |
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| 02:51.15 | aceio | hi all |
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| 02:51.40 | aceio | i am looking for a good off dialplan |
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| 02:56.12 | aceio | can anyone help |
| 03:00.59 | ChannelZ | exten => s,1,Hangup |
| 03:01.42 | Puma1337 | can someone help me reconfigure my asterisk installation with curl without completely destroying my server? I am running version 1.4.21.2 |
| 03:08.07 | aceio | Puma1337: what help do you need |
| 03:10.34 | Puma1337 | i need curl installed |
| 03:10.39 | Puma1337 | but it is showing up as not a registered function |
| 03:10.52 | Puma1337 | and i have no clue how to recompile asterisk without destroying everything |
| 03:12.14 | aceio | Puma1337: so you what too move to a newer verison ? |
| 03:12.48 | Puma1337 | no, i just want to have curl registered as a function |
| 03:13.05 | aceio | Puma1337: I SEE |
| 03:13.50 | aceio | Puma1337: sorry not sure how |
| 03:14.23 | WIMPy | Puma1337: Nothing special. Just don't 'make samples'. |
| 03:15.10 | aceio | that will wipe you cofig |
| 03:15.27 | aceio | and it will replace it with defaults one |
| 03:17.28 | aceio | does anyone have a good dialplan i can borrow |
| 03:18.11 | WIMPy | I have several, but I don't see how other people's dial plans could be of any use for you. |
| 03:20.16 | aceio | WIMP: i need some thing to kick start me |
| 03:24.39 | aceio | i am look for a dialplan that will take all calls and allow caller to select the different departments |
| 03:25.26 | aceio | WIMPy: if you can pastebin one or two plz |
| 03:26.03 | WIMPy | There should be plenty of examples for menus on the webs. |
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| 03:50.29 | TJNII | Jesus google is getting bad. I did a search for "Cold Cranking Amps" and it was smart enough to know the acronym for cold cranking amps is CCA, so the number one link it returned is the California College of the Arts (CCA) |
| 03:50.38 | ChannelZ | Other people's dialplans are almost of no use to anyone else since everyone has different setups and uses |
| 03:53.04 | WIMPy | And definitely different numbers. |
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| 04:28.16 | edIII | Hello |
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| 04:43.35 | carrar | HARRO |
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| 04:50.25 | edIII | Hi |
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| 05:11.09 | nwks | hello, i have a problem configuring dahdi. When i change the genconf_parameters " pri_termtype SPAN/1 NT SPAN/2 TE" and executing dahdi_genconf all spans are in te mode |
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| 05:25.23 | sprite-- | Moving my development environment over to my production environment. Running into a weird issue that I do not have on my development server. When two parties are conferenced with app_conference asterisk does not detect hangup. The channels for both parties show up under sip show channels. Asterisk and app_conference is same version in both environments. |
| 05:25.45 | sprite-- | Was wondering if anyone has run into this before? Perhaps some sort of configuration error? Any ideas? |
| 05:31.04 | *** join/#asterisk Brack10 (~Travis@97.90.65.23) |
| 05:31.06 | Brack10 | Hey |
| 05:31.18 | Brack10 | A sip phone number is username@sipproxy.com right? |
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| 06:32.50 | Janos | hey there, got a small problem, trying to setup an asterisk with a TE405P card, everything seems to be fine except that cat /proc/zatpel/1 tells me that all my lines are in use, is there any way to hang up those lines manually ? |
| 06:34.41 | ChannelZ | It means they're in use by asterisk - not in use as in calls in use |
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| 06:40.43 | Janos | ChannelZ: ahh ok kool, how can i know which lines are in use ? |
| 06:41.07 | ChannelZ | within asterisk you can do 'core show channels' |
| 06:41.28 | ChannelZ | or 'dahdi show channels' for a hardware point of view |
| 06:42.16 | ChannelZ | (although that doesn't really tell you a lot besides status0 |
| 06:43.02 | Janos | ok, it says 0 calls, so i guess i have something wrong in my config cuz, when i try to dial out i get "Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)" |
| 06:44.01 | ChannelZ | Zap != DAHDI |
| 06:44.14 | ChannelZ | oh wait you are using Zaptel |
| 06:44.26 | Janos | yes |
| 06:44.50 | ChannelZ | well it's hard to say without knowing if you configged your asterisk right or what you're even trying to Dial() |
| 06:47.24 | Janos | well everything on my asterisk is working, SIP phones, IAX to an outside provider etc but can't get this TE405P to work (first time using it) |
| 06:47.36 | Janos | it's a 4 port E1 card |
| 06:48.15 | Janos | /etc/zaptel.conf seems to be fine since i'm getting ok in the /proc/zaptel/1 header |
| 06:48.29 | Janos | trying to dial Zap/G1/<NUMBER> |
| 06:48.48 | ChannelZ | And you configged /etc/asterisk/zapata.conf for asterisk to consume the channels? |
| 06:50.29 | Janos | i have group=1 and channels => 1-15,17-31 in zapata.conf, those are the only channels i have available in since channel 16 is control and port 1 is the only one active |
| 06:51.17 | ChannelZ | I don't remember most of the zaptel commands - does 'zap show channels' tell you anything in Asterisk? |
| 06:51.50 | Janos | yes i have all the channels there, with their respective context |
| 06:52.23 | Janos | just one question, what is zapata-channels.conf for ? |
| 06:52.52 | ChannelZ | I dunno that's not a standard config. You're not using FreePBX are you? |
| 06:53.13 | Janos | no, debian here |
| 06:53.30 | Janos | let me remove that file, cuz it was generated by zapconf |
| 06:53.34 | ChannelZ | zapata.conf is what is read by asterisk - you might have 'includes' in your config that include other files. |
| 06:53.53 | ChannelZ | well yeah they intend that you include that file |
| 06:54.33 | ChannelZ | but if you've already configured the channels yourself in zapata.conf then you don't need to or you're just doubling up |
| 06:55.47 | Janos | yeah nvm that file is there, but no included by anything |
| 06:57.17 | ChannelZ | well not sure what to tell you, perhaps you have an issue with your telco - do you know the line is active? |
| 06:59.11 | Janos | yes, it's, let me get some more info the on the zapata channel, i'm sure there is something wrong in my config |
| 06:59.13 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 06:59.19 | ChannelZ | or you've got the thing plugged into the right port? |
| 07:00.02 | ChannelZ | Do you have any red alarms showing up? (again I can't remember about zaptel, there are some commands to show hardware states..) |
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| 07:00.44 | WIMPy | Same memory problem here, but I think the 'OK' in /proc means the line is active. |
| 07:01.21 | ChannelZ | ztscan maybe |
| 07:02.04 | WIMPy | Janos: You told just about B-channels. Have you also configured the D-channel? Is it the right protocol? |
| 07:02.27 | WIMPy | What does 'pri intense debug' come up with? |
| 07:02.29 | ChannelZ | Yah you should probably pastebin your zapata.conf and zaptel.conf |
| 07:03.03 | Janos | ok one sec |
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| 07:04.52 | Janos | zaptel.conf http://asterisk.pastebin.com/jtAb5fGv |
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| 07:11.14 | Janos | on sec on the zapata |
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| 07:21.52 | Gopal | cant able to hear the ring back tone from other end... |
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| 07:22.23 | Janos | zapata.conf http://asterisk.pastebin.com/WXkUrUs9 |
| 07:22.36 | Janos | ok that's the zapata.conf file |
| 07:25.47 | Janos | i have TRUNK=Zap/G1 in extensions.conf and trying to dial 'exten => _9.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,t)' where TRUNKMSD = 1 |
| 07:26.05 | Janos | and i get 'dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)' |
| 07:26.57 | Janos | so WIMPy, ChannelZ, if you have any idea, it would be great |
| 07:27.04 | *** join/#asterisk TimeRider (steve@5ac7b36e.bb.sky.com) |
| 07:27.05 | WIMPy | So do you have traffic on the interface? |
| 07:27.09 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 07:27.24 | Janos | WIMPy: checking |
| 07:27.28 | WIMPy | _thinks_ that message looks like yes, but I'm not sure. |
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| 07:29.41 | Janos | WIMPy: can i check this with zttool ? or what should i use to check traffic ? |
| 07:30.02 | WIMPy | pri intense debug |
| 07:31.47 | Janos | i get no such a command with that, but pri debug span 1, is giving me '1 -- Restarting T203 counter' that, over and over |
| 07:32.26 | WIMPy | ok, so you've got a signalling problem. |
| 07:33.13 | Janos | pri show spans says 'PRI span 1/0: Provisioned, Down, Active' |
| 07:33.27 | Janos | i guess down means it's not working :P |
| 07:33.50 | WIMPy | wonders how 'down' and 'active' fot together. |
| 07:33.55 | WIMPy | fit |
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| 07:34.15 | Janos | lol good question |
| 07:34.33 | WIMPy | Is the portProbably configured and actial status. |
| 07:34.38 | WIMPy | Is the port green? |
| 07:34.48 | Janos | checking |
| 07:35.16 | Janos | yes |
| 07:35.26 | WIMPy | Or maybe you need to activate crc4? |
| 07:35.31 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:35.39 | Janos | no, with crc4 it doesn't work |
| 07:35.46 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 07:36.19 | WIMPy | Ok, what does it say then? |
| 07:36.36 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 07:36.42 | Janos | timing source auto card 0! i get that on syslog and the card never comes up |
| 07:37.34 | WIMPy | The part about timing source sonds ok, what does 'never comes up' mean? |
| 07:38.37 | Janos | zttool never list the span as 'OK' |
| 07:39.11 | WIMPy | Ok, so that should be the 'OK' that is identical to the green LED :-) |
| 07:39.50 | Janos | ahh ok i guess |
| 07:40.50 | WIMPy | So you get nothing but timeouts? |
| 07:41.32 | Janos | well when i try to dial through it, it says it's busy |
| 07:41.42 | Janos | i don't know what else should i try |
| 07:41.55 | WIMPy | in pri debug |
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| 07:42.49 | Janos | mm i just restarted asterisk and it's working now <.< |
| 07:43.09 | Janos | let me try to call |
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| 07:45.36 | Janos | nvm it's not, i just got a lot of debug this time and PRI span 1/0: Provisioned, Up, Active |
| 07:45.41 | Janos | but still can't call |
| 07:45.52 | Janos | let me try to read the debug code |
| 07:46.11 | WIMPy | Are you receiving anything? |
| 07:48.29 | Janos | yes let me paste it |
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| 07:50.19 | Janos | pri intense debug span 1, gives me this http://asterisk.pastebin.com/w4B2gpQ7 |
| 07:53.29 | WIMPy | Ok, Doesn't look bad. |
| 07:53.53 | WIMPy | As long as it stops once it has restarted all channels. |
| 07:54.48 | WIMPy | has never understood, why one would want to restart channels, BTW. |
| 07:55.00 | ehb | i have a problem setting up dahdi. i cant telephone on the internal isdn. i use two Digium B410P one conecctet to the ntba´s an one connectet to the pbx. |
| 07:55.02 | ehb | When i do a dahdi_scan alarmes=ok and the type is also right (4 ports are digital-TE and 4 are digital-NT). |
| 07:55.03 | ehb | i ve editet genconf_parameters to |
| 07:55.05 | ehb | pri_termtype SPAN/1 NT SPAN/2 NT SPAN/3 NT SPAN/4 NT SPAN/5 TE SPAN/6 TE SPAN/7 TE SPAN/8 TE |
| 07:55.06 | ehb | and do a dahdi_genconf but in dahdi-channels.conf ther are all ports marked wit signaling signalling = bri_cpe_ptmp. |
| 07:55.08 | ehb | i can phone form external to the asterisk but i cat get a dialtone fron internal isdn. |
| 07:56.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 07:56.17 | Janos | WIMPy: now i keep getting this http://asterisk.pastebin.com/FechwAVS |
| 07:56.26 | WIMPy | ehb: Isn't TE/NT switching done with jumpers on that card? |
| 07:57.05 | WIMPy | Janos: Looks perfectely normal, when idle. |
| 07:57.22 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:57.54 | ehb | WIMPy: yes they are but i have to define signaling ind dahdi-cahnnels.conf hadent i |
| 07:58.34 | WIMPy | ehb: I thought the idea was that it gets auto-detected. |
| 07:58.53 | WIMPy | But why don't you just change it to bri_net then? |
| 08:00.13 | ehb | WIMPy i did it but. ther is still no dialtone available |
| 08:02.07 | WIMPy | I never tried dahdi with bri-nt as it doesn't seem to work with HFC cards. |
| 08:02.36 | WIMPy | The is a parameter 'immediate' or something. have you played with that? |
| 08:02.41 | WIMPy | there |
| 08:03.31 | WIMPy | Otherwise take a look at what's going one on the D-channels. |
| 08:03.39 | WIMPy | on on |
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| 08:11.46 | ehb | WIMPy: i dont know how to look up d-channel activity. but you ve written bri-nt and i have bri_net ? |
| 08:12.39 | WIMPy | ehb: So sis I further up. I was just referring to BRI in NT mode, not some parameter. |
| 08:12.42 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 08:13.24 | WIMPy | And I don't know if you can also debug BRIs with 'pri [intense] debug' or if there's another syntax. |
| 08:14.36 | WIMPy | I have changed to misdn2/lcr, which seems to have been a good idea. |
| 08:18.46 | ehb | WIMPy: I´ve just editet the dahdi-channels.conf by hand and restart asterisk an dahdi but the Dahdi-channels was not present. i´ve tried reload chan_dahdi.so an get |
| 08:18.49 | ehb | <PROTECTED> |
| 08:19.39 | ehb | new is signalling = bri_net_ptmp old was signalling = bri_cpe_ptmp |
| 08:23.46 | WIMPy | ehb: IIRC it's only bri_net, as ptp isn't implemented. So it's always ptmp. |
| 08:29.23 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-cplioxlibzcwjmja) |
| 08:29.44 | ehb | WIMPy: it is a ptmp port but one is listening bri_cpe konnectet to the NTBA and one is serving the pbx bri_net. |
| 08:30.22 | WIMPy | Yes, got that |
| 08:31.42 | ehb | WIMPy: but why dose dahdi refuses to start when i change the signaling to bri_net_ptmp |
| 08:32.13 | WIMPy | ehb: IIRC it's only bri_net, as ptp isn't implemented. So it's always ptmp. |
| 08:32.25 | WIMPy | bri_net, not bri_net_ptmp |
| 08:32.32 | WIMPy | (still IIRC) |
| 08:32.49 | ehb | WIMPy: what is IIRC |
| 08:33.11 | WIMPy | If I Remember Correctely |
| 08:33.42 | WIMPy | Are you new to the internet? ;-) |
| 08:34.46 | ehb | What is Internet ;-) ? |
| 08:35.45 | ehb | WINPy: i´ve changed ist to bri_net and dahdi channels are back in asterisk. but no dialtone on the int ISDN Ports |
| 08:35.54 | *** part/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 08:36.27 | WIMPy | Ok, so can you dial anyway? |
| 08:36.40 | ehb | WIMPy: do i have to set up users for int ISDN? |
| 08:37.08 | WIMPy | users? |
| 08:37.14 | *** join/#asterisk vutamhoan (~chatzilla@113.22.67.103) |
| 08:37.54 | Wimme | hey, i have a setup where i can accept or decline incoming calls that used to work with ast 1.4 but now in 1.6.0.10 it no longer works. |
| 08:37.58 | Wimme | i use Dial(sip/${EXTEN},15,mM(confirmcust)) |
| 08:38.04 | Wimme | in my [macro-confirmcust] is ask for input to accept or decline the call: |
| 08:38.05 | Wimme | Background(incoming-call-1-accept-2-decline&silence/5,,,macro-confirmcust) |
| 08:38.06 | ehb | WIMPy: I can dial in from pstn to hello world and dial out from asterisk console but i cant dial from my pbx connectet to ports bri_net |
| 08:38.11 | *** part/#asterisk vutamhoan (~chatzilla@113.22.67.103) |
| 08:38.12 | Wimme | ,,,macro-confirmcust) i use the context in the background so i can have exten => 1 and exten => 2 inside the macro instead of the context that called the macro. |
| 08:38.18 | Wimme | When i try this is see that the DTMF is being detected, but nothing happens. |
| 08:39.15 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 08:39.16 | WIMPy | ehb: Ok, have you tried the other way? But, I guess it's time to look at the signalling. |
| 08:39.34 | Wimme | when i added exten => 1,1,NoOp(test) in the context that calls the macro it also does not execute |
| 08:41.20 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:44.03 | ehb | WIMPy: extensions.conf |
| 08:44.04 | ehb | [Int-ISDN] |
| 08:44.06 | ehb | exten => _X.,1,Answer() |
| 08:44.07 | ehb | exten => _X.,2,Playback(hello-world) |
| 08:44.09 | ehb | exten => _X.,3,Dial(DAHDI/g15/${EXTEN}) |
| 08:44.10 | ehb | exten => _X.,4,Hangup() |
| 08:44.12 | ehb | dahdi-channels.conf |
| 08:44.14 | ehb | ; Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS |
| 08:44.15 | ehb | group=0,11 |
| 08:44.17 | ehb | context=Int-ISDN |
| 08:44.18 | ehb | switchtype = euroisdn |
| 08:44.20 | ehb | signalling = bri_net |
| 08:44.21 | ehb | channel => 1-2 |
| 08:44.23 | ehb | context = default |
| 08:44.24 | ehb | group = 63 |
| 08:45.03 | *** join/#asterisk vutamhoan (~chatzilla@113.22.67.103) |
| 08:46.06 | WIMPy | ehb: Can you call you old PBX from Asterisk? |
| 08:48.32 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 08:50.30 | *** join/#asterisk fibres (~no@cpc1-nfds1-0-0-cust582.lei3.cable.ntl.com) |
| 08:50.33 | fibres | Morning all |
| 08:51.34 | fibres | Got a major problem. Upgraded a server from asterisk 1.2 to asterisk 1.4.31 and now I get a lot of calls disconnect after 30 secs. |
| 08:53.07 | ehb | WIMPy: no, and i´ve just used my ISDN phone on the port. it says wrong nomber . so i´ve to defin the msn for the int isdn port an the asterisk. but i dont know wher i can do it. |
| 08:53.59 | WIMPy | Immediately or after you dialled a number? |
| 08:54.42 | WIMPy | Ports dont't have MSNs with Asterisk. |
| 08:54.45 | *** join/#asterisk lhz (~shrekz@c-b9aa72d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 08:56.24 | vutamhoan | I got this issue - https://issues.asterisk.org/view.php?id=15915 - Any suggestion is appreciated. Thank in advanced |
| 08:58.00 | ehb | WIMPy: i´ve had that message when i try to connect to an PBX internal ISDN port whith the wrong numer set in my isdn phone |
| 08:58.51 | WIMPy | ehb: Immediately or after you dialled a number? |
| 08:59.47 | WIMPy | Noone cares, what MSN you use on your phone unless you build something into your dialplan to check it. |
| 09:01.13 | WIMPy | If you want screening, you have to change to using LCR. |
| 09:01.21 | *** join/#asterisk checkerboard123 (~lakesh@202.70.87.202) |
| 09:02.56 | fibres | Any ideas anyone this is causing me a major problem |
| 09:04.32 | WIMPy | fibres: define 'calls'. And why didn't you upgrade to something recent. Same trouble, double benefit. |
| 09:07.54 | Janos | WIMPy: i feel terrible, it was an error in my dialplan, my E1 was just fine :P, thanks for the help |
| 09:08.53 | ehb | WIMPy: i´ve tried with a blank dialplan, only [Int-ISDN] and [default] but i cant get a dialtone. |
| 09:09.18 | *** join/#asterisk Tim_Toady (~moi@77.49.63.231.dsl.dyn.forthnet.gr) |
| 09:10.06 | WIMPy | Ok, last try: |
| 09:10.25 | WIMPy | ehb: did you get the invalid number message immediately or after you dialled a number? |
| 09:11.52 | ehb | WIMPy: no i ve got that message when i connects my ISDN phone immediately, before i pickt up. |
| 09:13.06 | WIMPy | wonders what the phone did. But you probably should ignore it then. |
| 09:16.53 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 09:17.18 | ehb | WIMPy: The phone is a ISDN Messuerment Phone. and i hve those messages when the phone sends other numers (ISDN) than the one are set in the PBX (own MSN) . This is a normal ISDN feature its normaly used for call acounting. one can only dial if the right msn is set. |
| 09:18.29 | WIMPy | ehb: Asterisk ist _not_ an ISDN PBX! It doesn't care. |
| 09:18.51 | WIMPy | You probably get the message, because the phon sends some facility message Asterisk doesn't understand. |
| 09:19.10 | WIMPy | Just ignore it. |
| 09:20.01 | *** join/#asterisk nitram (foo@superblob.com) |
| 09:20.43 | *** join/#asterisk davix (~abutbul@81.218.229.226) |
| 09:21.10 | ehb | WIMPy: but I can´t dial out from that port not with the phone and not with the pbx |
| 09:21.23 | davix | if I want to place a IP-phone in my house and connect it to some remote pbx machine, do I need to have a dedicated IP for that phone or can it work on the same IP as my computer? |
| 09:21.34 | *** join/#asterisk Xt0f_ (~d54c8bc2@gateway/web/freenode/x-bbpgkxwjdexpcwvf) |
| 09:21.37 | WIMPy | What does "can't dial" mean? |
| 09:22.05 | Trixboxer | <PROTECTED> |
| 09:22.25 | Xt0f_ | is any one here can help me to solve my problem to connect to AMI using OpenSSL ? |
| 09:23.05 | davix | and if my extension rings |
| 09:23.15 | davix | the pbx knows how to tell the phone to ring? and not my computer? |
| 09:23.42 | Trixboxer | davix: The phone knows what to do if ring is send by pbx |
| 09:23.49 | WIMPy | davix: You have a router? |
| 09:23.55 | davix | yes |
| 09:24.20 | davix | do i need some nat rules? |
| 09:24.25 | WIMPy | Great. It will take care of finding the right destination. |
| 09:24.31 | davix | what about if i want to have two phones on the same IP? |
| 09:25.13 | WIMPy | You will probably need NAT, yes. And that usually meand your phone needs to support STUN and some form of keep-alive. |
| 09:25.26 | ehb | WIMPy: do i have to set up type=friend or so on dahdi-channels.conf? |
| 09:25.34 | WIMPy | The number doesn't matter. |
| 09:25.49 | davix | what do I need to configure the NAT as? I don't have a STUN server anyway on the other side, I think |
| 09:26.02 | davix | the phone im getting is snom360 or 300 |
| 09:26.05 | WIMPy | ehb: no, but you're relly not being especially helpfull in helping yourself. |
| 09:26.41 | Wimme | davix, does your phone register? |
| 09:26.51 | WIMPy | davix: The usual ITSPs have STUN servers. And there's also stunserver.org. |
| 09:27.25 | Xt0f_ | Sorry to interupt the chat, but nobody to help me on the Open SSL + AMI issue..? Thank you in advance. |
| 09:27.44 | WIMPy | The Snom 360 will do about everytrhing you tell it to. No idea about the 300, that's a different series, AFAIK. |
| 09:28.57 | ehb | WIMPy: thank you anyway |
| 09:30.37 | Trixboxer | davix: hi |
| 09:30.49 | Trixboxer | davix: Your remote pbx needs a public ip |
| 09:31.09 | davix | oh |
| 09:31.11 | davix | ok |
| 09:31.16 | Trixboxer | davix: Your phones need an internet connectivity to connect to remote pbx |
| 09:31.19 | davix | and then when it registers it opens a tunnel? |
| 09:31.29 | Trixboxer | davix: dont worry if they are behind nat |
| 09:31.39 | Trixboxer | davix: yes it takes care of it |
| 09:32.00 | WIMPy | There's no tunnel involved. |
| 09:32.01 | davix | ok I will try it, thanks for clarifying this issue for me trix, wimme and wimpy |
| 09:32.22 | WIMPy | Unless you go for the 370 which supports openvpn. |
| 09:32.46 | Wimme | davix, on your snom phone, thers nat settings, set up a keepalive interval of 20 seconds |
| 09:33.01 | davix | ok |
| 09:33.06 | davix | I will remember to set this |
| 09:33.09 | Trixboxer | Hey there, do u know a good video phone ? |
| 09:33.11 | Wimme | your phone will register but the connection will timeout, causing incoming calls to fail |
| 09:33.22 | Wimme | the keepalive keeps the connection alive :) |
| 09:33.49 | Trixboxer | Im looking for a video phone compatible with asterik |
| 09:34.10 | Wimme | if you notice incoming calls sometimes fail, maybe you need to lower the keepalive interval |
| 09:35.32 | Xt0f_ | nobody for OpenSSL + AMI ? :-( Thank you..... |
| 09:36.03 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:36.28 | angryuser | for the europe it's ulaw or alaw ? |
| 09:36.42 | Trixboxer | Xt0f_: what issue u have ?? |
| 09:37.54 | Xt0f_ | well I have created a certificate with openSSL, put it on /var/lib/asterisk/keys, configure manager.conf with port on 5039....but cannot connect => got the warning : tcptls.c:218 handle_tcptls_connection: FILE * open failed! |
| 09:42.55 | Xt0f_ | any ideas ? |
| 09:43.37 | Trixboxer | Xt0f_: im checking |
| 09:43.56 | Xt0f_ | Trixboxer : thanks |
| 09:43.59 | Trixboxer | Xt0f_: did u google it ? |
| 09:44.35 | Xt0f_ | yep but only information about the tcptls.co... |
| 09:44.41 | Xt0f_ | the code |
| 09:44.57 | Trixboxer | Xt0f_: whr did u encounter this error ? |
| 09:45.02 | Trixboxer | Xt0f_: amportal start ? |
| 09:45.43 | Xt0f_ | Trixboxer : no when i am connected on console...and trying to connect on port 5039 with a SSL client |
| 09:46.10 | *** join/#asterisk BrokenNoze (~BrokenNoz@host81-137-172-233.in-addr.btopenworld.com) |
| 09:46.57 | Trixboxer | Xt0f_: to be frank i dnt have its idea.. but a log m8 help.. if u can.. just paste it in pastebin |
| 09:47.48 | BrokenNoze | Hi All, does anyone know if there is a way to identify the end of a call within the dialplan? Not a hangup event but the actual termination of the channels? Internally MixMonitor seems to be able to do this however I need to perform a cleanup at the same time |
| 09:47.57 | Wimme | Xt0f_, have you checked the obvious? read permission on that file, and is it acually there? maybe a type? |
| 09:48.06 | Wimme | typo* |
| 09:50.01 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:50.47 | Trixboxer | BrokenNoze: but anyways the call is going to hangup.. so u can have a look at its context |
| 09:51.45 | Xt0f_ | Trixboxer : http://pastebin.com/tHTZUY8W |
| 09:54.34 | Trixboxer | Xt0f_: not getting any idea from logs |
| 09:55.00 | Trixboxer | Xt0f_: try checking file perms |
| 09:55.04 | Trixboxer | ok key |
| 09:55.07 | Trixboxer | of* key |
| 09:57.10 | Xt0f_ | Trixboxer : -rw-r--r-- |
| 09:58.48 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 09:59.57 | carrar | speaking of centos, 5.5 released |
| 10:00.19 | Xt0f_ | Wimme : what do you mean by type ? |
| 10:00.47 | Jumpie | ugh im having issues as well |
| 10:02.14 | *** join/#asterisk af_ (~getsmart@88-149-241-148.dynamic.ngi.it) |
| 10:03.10 | Wimme | Xt0f_, i coorected on the next line, it should have been typo |
| 10:03.23 | BrokenNoze | TrixBoxer : the problem is the h extension is run after a transfer which is causing us serious problems ending the recording and then copying a call to a network location when complete. |
| 10:05.44 | Wimme | BrokenNoze, you can set MONITOR_EXEC to a custom script, when monitor ends you can do whatever you want with the files |
| 10:06.13 | BrokenNoze | Wimme: Awesome. I'll give that a go |
| 10:06.51 | Wimme | BrokenNoze, asterisk will call the script with 3 variables, 2 audio paths and a filename |
| 10:07.11 | Trixboxer | BrokenNoze: Do u want to make a backup of calls as soon as they gets completed ? |
| 10:07.37 | Xt0f_ | Wimme: I have checked everything....:-(...pems, path, typo...not a lot of docs on this kind of issue....don't know if Open SSL + AMI is widely used.... |
| 10:08.05 | BrokenNoze | Trioxboxer: Yes thats it. problem is if the call is transferred. |
| 10:08.22 | Wimme | Xt0f_, i know nothing about using ami with ssl, just wanted to make sure some typo or file permission didnt cause it |
| 10:10.48 | Wimme | BrokenNoze, what happens with the call when it gets transfered? does monitor keep recording it? |
| 10:12.05 | BrokenNoze | Wimme: MixMonitor does yes. However we are using the h extension to try and copy the call at the end of recording. problem with this is the h extension is fired upon transferring the call, which means only half of the call is recorded |
| 10:12.14 | BrokenNoze | before it's then copied |
| 10:12.32 | Trixboxer | BrokenNoze: I can tell u a solution which is OS based n not at all related to asterisk.... Check for inotifytools on google |
| 10:12.54 | Trixboxer | Its a kernel notify thing |
| 10:13.35 | Wimme | BrokenNoze, look into using monitor and setting monitor_exec to a custom script |
| 10:13.43 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 10:13.59 | Wimme | http://pastebin.org/221179 check this one for example |
| 10:15.02 | Wimme | it gets the 2 audio files and the output filename as arguments, mixes the 2 files to 1, converts it to an MP3, mails it and cleans up the files |
| 10:15.13 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 10:15.14 | BrokenNoze | Wimme: but i ideally want to use MixMonitor.(empty pastebin) |
| 10:15.38 | BrokenNoze | Wimme: OK. sounds good but empty bin? |
| 10:16.12 | Wimme | my bad check again? |
| 10:16.34 | BrokenNoze | no luck |
| 10:17.22 | Wimme | http://pastebin.org/243755 |
| 10:17.37 | BrokenNoze | lovely. thanks! |
| 10:19.04 | Wimme | only diffrence between mixmonitor and monitor is that mixmonitor ends up with a sinlge file with both audio legs, but you can use soxmix to merge them anyway |
| 10:25.40 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 10:26.08 | BrokenNoze | Wimme: yes, I'm assuming then that this will still allow INHERIT_HOOK features. |
| 10:31.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:32.27 | Wimme | BrokenNoze, no idea what that is :) |
| 10:32.38 | Wimme | try and find out i guess |
| 10:32.56 | BrokenNoze | Wimme: :) OK, have a play |
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| 10:48.54 | *** join/#asterisk garymc (~chatzilla@host86-174-140-25.range86-174.btcentralplus.com) |
| 10:51.18 | garymc | Hi, |
| 10:52.12 | garymc | Can anyone tell me ..... just to clear this up in my own head.... Im using a sip phone through a PPTP VPN im connecting to my server from abroad. Will that still be using port 5060? or just the VPN port 1723 |
| 10:53.04 | garymc | and how could I check it was just using that port? Could I just close port 5060 and see if it is still working |
| 10:53.21 | joobie | garymc, it'll tunnel 5060 over your VPN |
| 10:53.55 | garymc | so say the wifi abroad blocks 5060 it will still work? |
| 10:54.41 | joobie | should do.. so long as you are blocking 5060 before the tunnel terminates |
| 10:54.56 | garymc | what do you mean? |
| 10:55.38 | garymc | blocking 5060 where? |
| 10:55.38 | *** join/#asterisk sourcode (~code@ppp-61-90-15-9.revip.asianet.co.th) |
| 10:55.39 | garymc | at home? |
| 10:56.06 | joobie | if you had for example... SIP CLIENT => VPN ENDPOINT => FIREWALL BLOCKING 5060 => VPN ENDPOINT => SIP SERVER, it would let the traffic through.. if you had SIP CLIENT => VPN END POINT => VPN END POINT => FIREWALL BLOCKING 5060 => SIP SERVER, it wouldn't.. |
| 10:57.08 | *** join/#asterisk simple64 (~Administr@66-227-172-142.dhcp.aldl.mi.charter.com) |
| 10:57.16 | garymc | so as long as port 5060 is open on the server at home/office I will be ok |
| 10:57.47 | joobie | understand the above |
| 10:59.36 | joobie | guys, I have a ISDN-10 hooked up to my "Sangoma A101DX" .. looking at upgrading the ISDN-10 to an ISDN-30 .. Once my telco does the upgrade, is there anything I need to do on my end to increase from the 10 to 30? |
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| 11:00.07 | garymc | ok let me write it out...... SIP CLIENT(Me Abroad) --> VPN (on phone) --> HOTEL WIFI FIREWALL blocking 5060 --> VPN on Server ---- SIP SERVER (at office) |
| 11:00.22 | garymc | joobie I have isdn30 with Sangoma 101D |
| 11:00.40 | garymc | I think you might have to just run sangoma setup to find the channels |
| 11:00.42 | garymc | thats it |
| 11:00.51 | garymc | it wont be hard |
| 11:01.29 | joobie | can I just change the 'channe =>' variable in chan_dahdi.conf to 1-30 instead of 1-10 ? |
| 11:01.44 | *** join/#asterisk visioconf (~FSC@196.203.15.214) |
| 11:01.47 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 11:01.55 | visioconf | hola guy's |
| 11:01.57 | garymc | yeah probably, but if you run sangoma setup it does it for you .. if I remeber correctly |
| 11:02.02 | gregd | hi guys, a stupid question... what's milkfish irc channel? ;) |
| 11:02.12 | WIMPy | joobie: No, 1-15,17-31 |
| 11:02.16 | garymc | its for people who milk fish |
| 11:02.33 | gregd | :D |
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| 11:02.44 | joobie | WIMPy, I think you saw this one coming - why is the 16th channel excluded? |
| 11:02.49 | garymc | I like to milk cows and woman not fish, I bet it tastes like ................fish |
| 11:02.52 | visioconf | as a newbie in the subject , could anyone help me in having an idea about how to become a Voip minutes seller |
| 11:03.12 | visioconf | i would like to sell a minute pack |
| 11:03.23 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 11:03.24 | WIMPy | joobie: It's the signalling channel |
| 11:03.26 | garymc | yeah, buy voip minutes off a provider for one price and sell them for another hopefully for a profit |
| 11:04.03 | visioconf | <garymc>, yes, in fact I want to be a provider |
| 11:04.10 | joobie | WIMPy, what is the signalling channel in an ISDN-10? given i have 1-10 specified in my chan_dahdi.conf ? |
| 11:04.17 | garymc | well me too |
| 11:04.26 | garymc | one day lol |
| 11:04.58 | WIMPy | joobie: There's no difference to ISDN-30, you're just not allowed to use 20 of the B-channels. |
| 11:05.12 | Wimme | joobie, you have a partial pri |
| 11:05.28 | garymc | I think youll need BIG servers and BIG data pipe coming in. I know you can get a BIG data pipe in UK for 80k per anum |
| 11:05.46 | garymc | ohhh and Technical people doing all the work for you |
| 11:05.53 | garymc | BIG marketing budget |
| 11:05.53 | joobie | ahh |
| 11:06.25 | joobie | WIMPy, and upgrading from my ISDN-10 - is it just a case of changing that one variable in chan_dahdi and reloading the chan_dahdi module? |
| 11:06.37 | garymc | The correct licenses etc etc ....... good luck visioconf |
| 11:07.08 | Wimme | visioconf, you're better of becoming a reseller for an existing voip rovider :) |
| 11:07.24 | WIMPy | joobie: Should be it. |
| 11:07.35 | garymc | joobie honestly just run sangoma setup its automatic and does it in like 10 seconds |
| 11:08.03 | joobie | WIMPy, cheers.. also, anything different I should be asking my provider to do when they upgrade the link from ISDN-10 to ISDN-30 ? I have the settings I gave them last time - was just going to resubmit these |
| 11:08.03 | garymc | it takes existing setup and rechecks the channels |
| 11:08.04 | visioconf | yeah, but is there a documentation for that |
| 11:08.11 | joobie | things like provider dialtone, etc etc.. |
| 11:08.15 | visioconf | even if it looks so hard |
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| 11:08.32 | garymc | what country you in joobie |
| 11:08.40 | joobie | garymc, au |
| 11:08.51 | WIMPy | joobie: The only difference should be the number of channels you're allowed to use. |
| 11:08.54 | garymc | ok there is a pstn setting you may need to change |
| 11:09.02 | joobie | WIMPy, cheers |
| 11:09.06 | garymc | let me look, I had to fiddle about for ages |
| 11:09.48 | joobie | garymc, would this not be set already on my isdn-10? |
| 11:10.05 | garymc | possibly but its a differnt signal so maybe not |
| 11:10.58 | garymc | joobie found it |
| 11:11.12 | garymc | chan_dahdi.conf |
| 11:11.40 | garymc | there is a line that I had to change untill my PRI (isdn30) worked for incoming and outgoin calls |
| 11:11.55 | garymc | mine is set to |
| 11:11.56 | garymc | pridialplan=unknown |
| 11:12.14 | garymc | under [Channels] |
| 11:12.57 | garymc | you may need to set it to pridialplan=local or something else. Your telco should beable to tell you. But mine couldnt cos they are shit. They are called BT |
| 11:13.59 | garymc | also in same file joobie. My channels are set like this below for the Sangoma A101D wanpipe |
| 11:14.00 | garymc | channel =>1-15,17-31 |
| 11:14.08 | joobie | thanks |
| 11:14.08 | garymc | like WIMPy said |
| 11:14.17 | joobie | im not sure what this pridialplan is for |
| 11:14.21 | joobie | will need to read into it more |
| 11:14.31 | garymc | Do you not have one in your file now? |
| 11:14.36 | joobie | <PROTECTED> |
| 11:14.36 | joobie | <PROTECTED> |
| 11:14.39 | WIMPy | That's about number types. |
| 11:14.48 | joobie | garymc, i dont |
| 11:15.02 | joobie | WIMPy, what about it? |
| 11:15.07 | garymc | ok well you may need to add that line at the top just under [channels] |
| 11:15.35 | garymc | so joobei is my VPN gonna let me make calls if 5060 is blocked :) |
| 11:15.43 | garymc | * joobie |
| 11:16.05 | WIMPy | Instead of using prefixes, you can set the 'type of number'. |
| 11:16.16 | joobie | u'll be fine garymc |
| 11:16.22 | garymc | cheers :) |
| 11:16.27 | joobie | so long as your hotel wifi doesnt block 1723 also |
| 11:16.31 | garymc | ok well good luck with your ISDN 30 switch over |
| 11:16.38 | garymc | im sure you wont have much trouble |
| 11:16.43 | joobie | your port 5060 traffic is tunneled over the pptp vpn |
| 11:16.51 | joobie | so the hotel firewall will only see 1723 traffic |
| 11:16.57 | garymc | ok so they dont know im using 5060 |
| 11:17.00 | joobie | within that is your 5060, which is encrypted and hidden from the hotel firewall |
| 11:17.00 | garymc | cool :) |
| 11:17.01 | joobie | werd |
| 11:17.39 | garymc | ok im off have a good day |
| 11:17.42 | garymc | bye bye |
| 11:18.50 | joobie | WIMPy, i dug up my old 3G USB modem |
| 11:18.56 | joobie | turns out it's the E169 :P |
| 11:18.58 | joobie | wrapped |
| 11:19.36 | WIMPy | joobie: I just read that chan_datacard is supposed to receive text as well. |
| 11:19.55 | WIMPy | But still didn't have any luck with it so far. |
| 11:20.29 | WIMPy | In fact the whole Asterisk on the netbook behaves pretty strangely. Not sure, what has happened there. |
| 11:21.10 | joobie | ahh |
| 11:21.40 | joobie | ill let you know how i go with it |
| 11:21.44 | joobie | gotta build up another box for * |
| 11:21.53 | joobie | only got one in the datacenter that i cant afford to crash |
| 11:24.34 | *** join/#asterisk FRbsd (~FRbsd@zone0.gcu-squad.org) |
| 11:24.42 | FRbsd | hello |
| 11:25.11 | FRbsd | i did an upgrade to asterisk 1.6 and my phones cannot make outgoing calls with this error "Failed to authenticate on INVITE to " |
| 11:25.20 | FRbsd | does it ring a bell to anyone? |
| 11:25.58 | ijpalmer | Morning, I'm having a problem whereby a call transferred by a sip device is hanging up the first part of a recording when the second party finishes the transfer to the thrid party. http://www.pastebin.org/243941 |
| 11:27.06 | *** part/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 11:33.24 | joobie | FRbsd, presuming that bit after the "to" is your ITSP .. if so, make sure your sip settings for your ITSP are setup properly |
| 11:33.58 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 11:35.24 | FRbsd | joobie: i believe it is, here is a default client config (which used to work on 1.4): http://pastie.org/963601 |
| 11:39.08 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 11:41.13 | MAbbas | Hi All, how Can I transfer my incoming call to Queue(,tT), from one agent to another agent. Transfer() does not work, as incoming call is sip and Agent/ is different technology than sip? |
| 11:41.54 | Gopal | what need to check if there is no ringback tone |
| 11:42.03 | Gopal | do we need to see RTP packet flow? |
| 11:46.34 | joobie | FRbsd, try specify defaultuser=166 instead of defaultusername=166 |
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| 12:02.48 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:03.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:09.25 | *** join/#asterisk Wildy (~simba@94.25.161.197) |
| 12:11.13 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 12:11.30 | MAbbas | [TK]D-Fender: how Can I transfer my incoming call to Queue(,tT), from one agent to another agent. Transfer() does not work, as incoming call is sip and Agent/ is different technology than sip? |
| 12:12.25 | [TK]D-Fender | MAbbas: Yes. It is a channel type, just like its position in a dial string indicates. |
| 12:12.51 | [TK]D-Fender | MAbbas: And it looks like you also didn't read what Transfer() is for... |
| 12:13.34 | MAbbas | <[TK]D-Fender> More precisly, how can I blind transfer calls in queues to some other extension? |
| 12:15.14 | [TK]D-Fender | MAbbas: make an extension that dials the agent. The End. |
| 12:16.40 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 12:17.32 | [TK]D-Fender | MAbbas: "core show application dial" |
| 12:19.08 | MAbbas | <[TK]D-Fender> I tried exten=>_80X,1,Answer() exten=>_80X,n,Queue(testQ,tT) exten=>0123,1,Dial(Agent/202) |
| 12:19.24 | MAbbas | But it logs off agent that transfered the call? |
| 12:19.37 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 12:19.47 | [TK]D-Fender | MAbbas: What gives you that impression? |
| 12:20.26 | *** join/#asterisk neurosys (~neurosys@166.192.76.123) |
| 12:21.24 | MAbbas | My call was originally xfered to agent/201 when I dialed extension 0123. It connected to Agent/202. Now when I CLI> agent show online. Only agent 202 is online. |
| 12:21.49 | MAbbas | <[TK]D-Fender> ? |
| 12:22.19 | [TK]D-Fender | MAbbas: Maybe you should actually show a complete debug for a call the shows the before & after sides of this |
| 12:22.30 | [TK]D-Fender | MAbbas: You are asking empty questions and not collecting backup. |
| 12:23.51 | MAbbas | <[TK]D-Fender> ok, let me |
| 12:27.28 | MAbbas | <[TK]D-Fender> http://pastebin.com/bjaCT7Nh |
| 12:27.55 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:28.26 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:29.53 | MAbbas | <[TK]D-Fender> ?? |
| 12:31.00 | [TK]D-Fender | MAbbas: Whats the QUESTION? |
| 12:31.07 | [TK]D-Fender | MAbbas: What am I supposed to be seeing in there? |
| 12:32.26 | MAbbas | <[TK]D-Fender> This is call log. I am asking you, why Agent/201 loged-off when tranfered call? |
| 12:32.56 | [TK]D-Fender | MAbbas: Where do I see "logged off" in there? |
| 12:33.20 | *** join/#asterisk shader (~user@janustw.tavve.com) |
| 12:33.44 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:33.52 | MAbbas | <[TK]D-Fender> After -- Executing [0123@test:1] Dial("SIP/1003-0000000f", "Agent/202") in new stack |
| 12:34.22 | MAbbas | <[TK]D-Fender> Although, I did not explicitly log-off that agent |
| 12:35.50 | [TK]D-Fender | MAbbas: How did you transfer the call? |
| 12:37.12 | joobie | hmm.. what's the standard way to do failover for ISDN ? |
| 12:37.50 | MAbbas | <[TK]D-Fender> by pressing tranfer button on eyeBeam softphone |
| 12:37.58 | joobie | my telco has dropped my ISDN once.. just thinking if i approach another telco, how can i get failover going, where a 100-indial and DID's assigned to one circuit on one ISP can failover to another providers ISDN? |
| 12:38.12 | [TK]D-Fender | [08:37]<MAbbas><[TK]D-Fender> by pressing tranfer button on eyeBeam softphone <--- clearly you didn't think that through. |
| 12:38.55 | [TK]D-Fender | MAbbas: Your phone EJECTS the entire damn call. |
| 12:39.30 | [TK]D-Fender | MAbbas: -- Executing [800@test:2] Queue("SIP/1003-0000000f", "testQ,tT") in new stack <--- You're supposed to be doing ***ASTERISK DTMF TRANSFERS** |
| 12:39.48 | florz | yeah, you have to be careful to not point the phone at yourself when it's EJECTing calls |
| 12:40.56 | MAbbas | <[TK]D-Fender> Pardon me, how we do ***ASTERISK DTMF TRANSFERS** ? |
| 12:41.36 | [TK]D-Fender | MAbbas: "core show application queue". |
| 12:44.46 | *** join/#asterisk neurosys (~neurosys@166.194.249.108) |
| 12:45.23 | *** join/#asterisk shader (~user@janustw.tavve.com) |
| 12:51.18 | *** join/#asterisk scalex000 (~chatzilla@190.166.170.137) |
| 12:51.19 | *** join/#asterisk Baylink-work (~jra@cerberus.vicimarketing.com) |
| 12:51.24 | scalex000 | hello |
| 12:51.31 | shader | hi |
| 12:52.24 | scalex000 | I have problem I unable to connect to remote asterisk(does /var/run/asterisk,ctl exits?) what it's mean? |
| 12:52.29 | scalex000 | what can I do? |
| 12:53.03 | [TK]D-Fender | scalex000: Try starting * or connecting with a suer that has the rights to conenct |
| 12:53.08 | [TK]D-Fender | user* |
| 12:53.20 | scalex000 | I use the root |
| 12:53.33 | [TK]D-Fender | scalex000: Then maybe * isn't running |
| 12:55.25 | scalex000 | unable to connect |
| 12:55.56 | *** join/#asterisk Janos (~cramos@186.4.1.246) |
| 12:57.23 | scalex000 | I reinstall asterisk again, its the same |
| 12:58.24 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 12:58.27 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 12:58.56 | tuxx- | hey guys, is there an asterisk 1.4.31 application or feature which allows me to see if there is someone in a parking lot from the dialplan? |
| 13:00.24 | lirakis | tuxx-, show parkedcalls Lists parked calls |
| 13:00.47 | tuxx- | tnx. think i'll have to go AGI on this one :) |
| 13:01.07 | lirakis | ? |
| 13:01.32 | lirakis | oh you want to do it from the dialplan |
| 13:01.36 | tuxx- | ye :) |
| 13:01.37 | lirakis | .. i thought you wanted cli |
| 13:04.53 | [TK]D-Fender | [08:53]<[TK]D-Fender>scalex000: Then maybe * isn't running |
| 13:05.13 | *** join/#asterisk Z_God (~julius@wlan227084.mobiel.utwente.nl) |
| 13:05.19 | scalex000 | tk, thanks was not running dead lock |
| 13:05.26 | scalex000 | its okey now |
| 13:05.41 | [TK]D-Fender | tuxx-: "core show application chanisavail" |
| 13:06.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:06.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:07.42 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:09.25 | tuxx- | [TK]D-Fender: how would you use chanisavail to check if someone is in the parking lot? Dont quite get it. |
| 13:10.01 | [TK]D-Fender | tuxx-: Because when you park a call an EXTENSION is created for that lot. Do the math.... |
| 13:10.19 | tuxx- | ofc. thnx :) |
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| 13:56.57 | mpe | Hi I have a problem with 1.6 and features.conf |
| 13:56.57 | mpe | this line work fine in 1.4, but it wont work in 1.6 |
| 13:56.57 | mpe | features.conf |
| 13:56.57 | mpe | [applicationmap] |
| 13:56.57 | mpe | test => *3,callee,Goto,test|s|1 |
| 13:57.31 | leifmadsen | mpe: you didn't read UPGRADE.txt |
| 13:57.37 | leifmadsen | mpe: you can't use pipes anymore |
| 13:58.04 | leifmadsen | read CHANGES and UPGRADE.txt before going any further. |
| 13:58.08 | mpe | yes I know but I cant either use comma as I will then define MOH |
| 13:58.34 | Naikrovek | would you just read it |
| 13:58.53 | leifmadsen | moves onto more pressing issues |
| 13:59.31 | Naikrovek | it's probably test => *3,callee,Goto(test,s,1) now. |
| 13:59.48 | mpe | damm http://svn.digium.com/view/asterisk/tags/1.6.0/UPGRADE.txt?view=markup is slow |
| 14:00.03 | Naikrovek | the file is in the source you downloaded |
| 14:00.11 | leifmadsen | mpe: it's just you -- loaded fine for me |
| 14:01.24 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
| 14:02.34 | shader | so, what does it mean for a phone to be registered with asterisk? |
| 14:02.40 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 14:03.03 | shader | is it required for sending calls, or just receiving them? |
| 14:03.05 | leifmadsen | shader: it means the phone has told asterisk what IP address it is available at if it needs to place a call to it |
| 14:03.08 | leifmadsen | shader: only sending |
| 14:03.09 | shader | ok |
| 14:03.10 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 14:03.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:03.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.35 | leifmadsen | shader: sorry -- only receiving calls to the phone -- sending calls FROM the phone does not require registration |
| 14:03.52 | leifmadsen | "sending" and "receiving" are relative to what end point you are talking about |
| 14:03.57 | leifmadsen | putnopvut: welcome to monday |
| 14:05.07 | *** join/#asterisk d00gster (~dt@87.109.238.2) |
| 14:05.55 | putnopvut | leifmadsen: thanks, good to be here. |
| 14:06.33 | shader | leifmadsen: thanks, I think I sorted it out |
| 14:07.14 | mpe | have cheked UPGRADE.TXT it only mention that pipe have been replaced with comma |
| 14:07.14 | mpe | but even escapint the commas wont work |
| 14:07.15 | mpe | test => *3,calle,Goto,test\,s\,1 |
| 14:07.15 | mpe | Mapping Feature 'test' to app 'Goto(test\)' with code '*3' |
| 14:07.33 | leifmadsen | why are you escaping the commas? |
| 14:08.20 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:09.01 | mpe | as aplication map will take the next are as MOH |
| 14:09.01 | mpe | <FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]] |
| 14:09.49 | leifmadsen | hmmmm |
| 14:09.56 | mpe | I can only provide one argument to AppArguments, but goto take 3 |
| 14:09.58 | leifmadsen | perhaps you will need to file a bug then |
| 14:10.14 | shader | can you use macros in sip.conf? |
| 14:10.46 | shader | or can you assign any attribute in the general section to make it global? |
| 14:11.14 | mpe | just homet that there was some magic charcter that I ditten know about like test^s^1 |
| 14:11.18 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 14:11.29 | mpe | hoped |
| 14:12.09 | leifmadsen | shader: you desperately have some reading to do it seems like |
| 14:12.13 | leifmadsen | ~book |
| 14:12.14 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:12.44 | anonymouz666 | it's time for a new edition |
| 14:12.47 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 14:13.02 | shader | cool, I didn't know it was online for free |
| 14:13.03 | shader | thanks |
| 14:13.04 | leifmadsen | anonymouz666: that's what I've been working on every morning from 7:30am -> 9:00am |
| 14:13.05 | rubbs | btw. thanks for those that wrote the book. I'm still reading it now, and I feel i"m finally starting to just barely grasp this whole thing. |
| 14:13.20 | rubbs | leifmadsen: we thank you for it, us newbs. |
| 14:13.35 | shader | indeed |
| 14:14.08 | rubbs | I had my boss buy it. It's easier to read in book form, that and we wanted to support documentation ;) |
| 14:14.21 | leifmadsen | places another dime into his piggy bank |
| 14:14.45 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 14:14.51 | rubbs | that said... I should go read more... |
| 14:15.02 | shader | so, what do you work on from 9 to 5? |
| 14:15.20 | rubbs | probably work. |
| 14:16.04 | leifmadsen | shader: asterisk consulting, bug tracking, and asterisk releases |
| 14:16.13 | leifmadsen | you know -- things that pay me enough money to pay rent :) |
| 14:16.23 | leifmadsen | sadly enough documentation does not pay the bills |
| 14:17.54 | rubbs | it's too bad. It's actually helped me quite a bit. |
| 14:19.43 | leifmadsen | well we'll still get it done, will just take longer than 1-2 months :) |
| 14:19.50 | shader | maybe you should add more wizards or vampires. They seem to make lots of money :P |
| 14:19.53 | leifmadsen | it'll be based on 1.8 anyways, and 1.8 isn't even branched yet |
| 14:20.02 | leifmadsen | shader: I will take that into consideration :) |
| 14:20.45 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:20.49 | leifmadsen | shader: I just passed it along to our editor :) |
| 14:20.55 | shader | lol |
| 14:21.00 | rubbs | heh |
| 14:21.16 | beefpastry | leifmadsen: I asked this question to no avail last week...perhaps you have an answer -- considering learning Ruby to get into Adhearsion...the book seems to portray Adhearsion positively, but I'm curious as to what the working relationship is three years later. |
| 14:21.29 | rubbs | and make sure the vamps sparkle in the sun to really add in the pre-teen girl demographic. |
| 14:21.44 | leifmadsen | beefpastry: well, I don't think Adhearsion has taken off quite like we thought |
| 14:21.56 | beefpastry | The adhearsion website left me curious. |
| 14:22.16 | leifmadsen | I think there are other methods of doing the same type of thing but through online APIs |
| 14:22.52 | leifmadsen | beefpastry: i.e. tropo.com and cloudvox.com |
| 14:23.51 | Naikrovek | omigosh |
| 14:23.58 | Naikrovek | a siren started working here |
| 14:24.05 | beefpastry | I'm debating how I want to go about integrating my PBX with our customer DB...always tended to Perl, but the writeup on Adhearsion in the book made me think twice. |
| 14:24.12 | Naikrovek | demon in the form of a woman who intices you into the lake and drowns you |
| 14:24.21 | leifmadsen | beefpastry: ODBC integration is what you want for DB integration |
| 14:25.04 | shader | Naikrovek: really? |
| 14:25.13 | Naikrovek | she is WAY too hot to be human |
| 14:25.31 | *** join/#asterisk neurosys (~neurosys@mobile-166-193-147-074.mycingular.net) |
| 14:25.33 | Naikrovek | heh |
| 14:25.33 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 14:25.35 | shader | how many people have drowned so far? |
| 14:25.49 | Naikrovek | she's got one person already training her on the time entry system |
| 14:25.51 | Naikrovek | he's a goner |
| 14:27.06 | beefpastry | I'm talking about simple lookups to get secretaries service info to their desktops based on caller-id...nothing terribly fancy. |
| 14:27.17 | beefpastry | at least yet |
| 14:27.25 | *** join/#asterisk af_ (~getsmart@88-149-241-148.dynamic.ngi.it) |
| 14:27.42 | beefpastry | Thanks for the response. |
| 14:27.59 | leifmadsen | beefpastry: right, it would be trivial to do that with func_odbc |
| 14:28.21 | leifmadsen | Set(RESULT=${ODBC_LOOKUP_CID(${CALLERID(num)})}) |
| 14:28.38 | leifmadsen | or push it to a DB the other way |
| 14:28.55 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 14:29.08 | leifmadsen | Set(ODBC_PUSH_CID(${SECRETARY_NAME})=${CALLERID(num)}) |
| 14:29.31 | leifmadsen | (you of course create the functions in func_odbc.conf and define the SQL) |
| 14:29.51 | *** join/#asterisk jetlag (jetlag@pool-173-61-213-121.cmdnnj.east.verizon.net) |
| 14:30.21 | *** join/#asterisk pawpro (~spam@213.166.12.34) |
| 14:31.04 | pawpro | hello everybody. How does one reject a call in PHP AGI with hangupcause 34? |
| 14:33.34 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
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| 14:33.50 | ManxPower | pawpro, "core show application hangup" and "core show application congestion" |
| 14:34.19 | *** join/#asterisk WimD (~Wim@d54C0234B.access.telenet.be) |
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| 14:34.53 | WimD | hey all, has anyone ever had any problems with a snom320 + 42key sidecar with all 42 keys programmed? |
| 14:34.55 | pawpro | ManxPower: Did you mean $agi->exec("Congestion"); ? |
| 14:34.56 | beefpastry | Thanks leif...I'm just laying groundwork in my head right now. |
| 14:35.50 | frk2 | hey guys |
| 14:36.01 | frk2 | whats a good (professional) operator panel that you guys use with Asterisk? |
| 14:36.03 | frk2 | other than FOP |
| 14:36.11 | ManxPower | pawpro, however you would run the application in your AGI library |
| 14:36.27 | pawpro | ManxPower: I'll try it then. Thanks |
| 14:36.29 | WimD | i have a snom320 here that has a sidecar, all 42 buttons are programmed but the 12 buttons on the phone itself are free. the sidecar buttons work, but the phones buttons do not |
| 14:36.39 | WimD | its also not possible to take a second line |
| 14:37.49 | ManxPower | pawpro, actually I meant "core show application hangup" or "core show application congestion" or you won't know how to send specifically hangupcause 34 or your PRI |
| 14:39.26 | ManxPower | WimD, sounds to me like a phone config issue, but you can confirm that with a SIP debug and then try it. |
| 14:40.23 | *** join/#asterisk Arsenick (~y@69.70.231.230) |
| 14:40.31 | *** join/#asterisk jrz (~jrz@c63047.upc-c.chello.nl) |
| 14:40.44 | WimD | well, i think so too, just noticed freeing some buttons on the sidecar allows a second incoming line; i know this isnt a snom chat room, but figured i could try here anyways :) |
| 14:42.57 | WimD | how nice, when you program all the keys on the extender, the keys on the phone no longer work. gonna try a fw upgrade |
| 14:46.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:46.46 | *** join/#asterisk Mango (~iMango@S0106001bfc77a834.vc.shawcable.net) |
| 14:47.40 | Mango | I have a VoIP provider claiming that the NIC on his Asterisk server randomly dies. He switched users to the backup server, but then...the exact same issue happened there, too. |
| 14:48.06 | Mango | He has a script to "restore the connection once it goes out" which limits the outages to about 15 minutes. |
| 14:48.34 | ChannelZ | Which does what? |
| 14:48.39 | Mango | He doesn't say. |
| 14:48.51 | Mango | The fact that it's happening on two servers with different hardware makes me very curious. |
| 14:48.55 | Mango | Anyone seen this before? |
| 14:49.02 | Mango | Ostensibly there's nothing exciting in the logs. |
| 14:49.22 | drmessano | Check the thing it's connected to |
| 14:50.01 | drmessano | There's two ends to that cable |
| 14:50.05 | Mango | looks like they're in different DCs too. |
| 14:50.16 | drmessano | O.o |
| 14:50.24 | Mango | completely different IPs, and they're 10ms away from each other. |
| 14:51.35 | drmessano | Same hardware? Maybe the Realtek NIC he's using isn't cutting it :) |
| 14:51.59 | Mango | nope. he says different NICs. |
| 14:52.54 | drmessano | Dunno, but doesn't sound like much of an Asterisk issue |
| 14:53.20 | Mango | I cannot for the life of me think what would cause that. |
| 14:55.01 | Chainsaw | Intermittent packet loss. Remember that VoIP is mostly UDP. |
| 14:55.15 | Chainsaw | Most TCP sessions will keep on trucking until the loss gets outrageous. |
| 14:55.24 | Chainsaw | You'll see failures in VoIP, DNS... |
| 14:55.38 | *** join/#asterisk the_weard (~arthur@41-132-110-62.dsl.mweb.co.za) |
| 14:57.39 | *** join/#asterisk jetlag (jetlag@pool-173-61-204-20.cmdnnj.east.verizon.net) |
| 14:59.17 | ChannelZ | Well without knowing what he's "resetting" its hard to say.. might be treating the symptom and not the cause |
| 14:59.32 | Mango | Aye. |
| 15:00.15 | Chainsaw | Mango: Work DNS problems into the conversation. |
| 15:00.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 15:00.35 | Chainsaw | Mango: If he has those too... I can give you the "intermittent loss" diagnosis in writing. |
| 15:00.43 | Mango | <grins!> |
| 15:01.02 | Mango | wonder who he uses for dns |
| 15:01.03 | Mango | checks |
| 15:01.35 | Mango | DNS Made Easy |
| 15:02.42 | *** join/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org) |
| 15:04.55 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 15:05.05 | proute | Hello, |
| 15:05.49 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 15:06.17 | Trixboxer | hey there, anybody configured gtalk video with asterisk ? |
| 15:07.16 | proute | I use * 1.4.29 with aastra phones 55i and 57i. Sometimes blf display question mark instead of the phone status. when I reboot the phone, this problem can be corrected or we can see again question mark. do you have any ideas? |
| 15:07.22 | proute | thanks for your help :) |
| 15:09.07 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
| 15:09.23 | ijpalmer | hi, does anyone know how i can copy a recording just made to a location other than the monitor directory, i'm using monitor to make the recording |
| 15:09.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:10.29 | luke-jr | now that I have like 20 termination routes, is it possible to have Asterisk sort them by latency and try the lowest latency first? ;) |
| 15:10.35 | luke-jr | or will I need to manually do that |
| 15:10.39 | elred_ | Hi, is there possibility to do a REFER (transfer) between two asterisk box ? Like, if it doesn't find the matching callid/fromtag/totag it will seek on another asterisk ? |
| 15:11.01 | elred_ | i am trying to do loadbalancing between multiple asterisk box but i need transfer to work |
| 15:14.01 | [TK]D-Fender | ijpalmer: Tell it to make the file somewhere else, or tell it to execute a script after to do the move for you. |
| 15:14.09 | [TK]D-Fender | ijpalmer: "core show application monitor" |
| 15:16.58 | ijpalmer | D-Fender: thanks, i've been trying to use monitor_exec but am unable to get it working, i don't really know linux that weel i'm afraid |
| 15:18.37 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 15:19.09 | [TK]D-Fender | ijpalmer: Or.. you could jsut tell it to save it precisely where yuo want it right from teh start <- |
| 15:20.06 | *** join/#asterisk neurosys (~neurosys@mobile-166-193-147-074.mycingular.net) |
| 15:20.20 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:1007:4f63:4352:51fd) |
| 15:21.13 | pabelanger | http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 15:21.27 | pabelanger | Anybody have other ways to accomplish this? ^^^ |
| 15:21.38 | Qwell | meetme? |
| 15:22.18 | Qwell | that looks to be a decent way to do it |
| 15:22.19 | [TK]D-Fender | Qwell: That's what it's doing... |
| 15:22.28 | pabelanger | Yar, looking for other dialplan suggestion to invite an outside person to an established conference |
| 15:23.12 | Qwell | maybe a feature to spit out the conf # you're in somehow |
| 15:23.48 | [TK]D-Fender | paWhat's wrong with this one? |
| 15:23.53 | [TK]D-Fender | pabelanger: What's wrong with this one? |
| 15:25.12 | pabelanger | [TK]D-Fender: Nothing. Was curious to see if anybody had a different solution. |
| 15:25.42 | [TK]D-Fender | pabelanger: Get a phone that supports more than 3-way calling |
| 15:25.53 | [TK]D-Fender | pabelanger: Otherwise ... that's what *'s got |
| 15:26.00 | pabelanger | nods |
| 15:34.58 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:35.30 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 15:37.23 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
| 15:44.04 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:47.56 | sprite-- | Moving my development environment over to my production environment. Running into a weird issue that I do not have on my development server. When two parties are conferenced with app_conference asterisk does not detect hangup. The channels for both parties show up under sip show channels. Asterisk and app_conference is same version in both environments. |
| 15:49.30 | ManxPower | app_conference must be new in 1.6.x |
| 15:49.30 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 15:49.30 | ManxPower | sprite--, does meetme work? |
| 15:49.30 | sprite-- | I'm running 1.4.22 |
| 15:49.30 | ManxPower | sprite--, app_conference is not part of 1.4.x |
| 15:49.31 | *** part/#asterisk pawpro (~spam@213.166.12.34) |
| 15:50.00 | sprite-- | Not sure. My whole application is written around app_conference. Everything is the same in both environments except my development environment is Ubuntu while running Gentoo in production. |
| 15:50.34 | sprite-- | I'm running Astersik 1.4.22.2 with app_conference 2.0.1 |
| 15:50.35 | ManxPower | very few people seem to use app_conference. I wish you the BEST of luck. |
| 15:50.39 | [TK]D-Fender | ManxPower: No, its a 3rd-party addon, which we don't support, running on a ancient release of a dead branch of 8 thrice removed :) |
| 15:50.46 | [TK]D-Fender | * |
| 15:50.57 | ManxPower | [TK]D-Fender, But his application is written around app_conference! |
| 15:51.09 | idespinner | 1.4 is a dead branch? |
| 15:51.23 | ManxPower | idespinner, sort of. |
| 15:51.32 | [TK]D-Fender | idespinner: 1.6.1 is about to hit "security-only" fix level itself. Do the math |
| 15:51.36 | ManxPower | I don't remember when ALL updates stop to 1.4, but it is soon. |
| 15:51.53 | russellb | 1.4 will be maintained until at least December of this year |
| 15:51.54 | leifmadsen | ~asterisk-versions |
| 15:51.55 | infobot | [asterisk-versions] Information about the maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at http://www.asterisk.org/asterisk-versions |
| 15:51.57 | russellb | probably a little bit longer |
| 15:52.15 | *** join/#asterisk jonny330 (~jon@12.222.63.34) |
| 15:52.32 | sprite-- | I suppose I could upgrade. I was always told meetme was unreliable and that app_conference handled a large number of conferences much better. |
| 15:52.59 | ManxPower | sprite--, you should contact the author of app_conference |
| 15:53.02 | [TK]D-Fender | sprite--: When was this? |
| 15:53.13 | sprite-- | 1.5 years ago. |
| 15:54.15 | idespinner | maybe its that app_conference doesnt need dahdi/zaptel timing devices |
| 15:54.33 | [TK]D-Fender | idespinner: that is a factor |
| 15:54.41 | idespinner | somepeople with shoddy hardware have issues with that... |
| 15:56.30 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 15:58.38 | *** join/#asterisk BANSAL (~BANSAL@117.199.124.135) |
| 15:58.47 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 16:01.12 | sprite-- | I think it might be a timing issue in my production environment. Getting monmp3thread: Request to schedule in the past?!?!? and app_conference giving the following warning conference.c:161 conference_exec: processed frame frequency variation, name => 39, tf_count => 50, tf_diff => 797, tf_frequency => 15.9400 |
| 16:01.42 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:02.38 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 16:03.21 | *** join/#asterisk jrz (~jrz@a190165.upc-a.chello.nl) |
| 16:04.47 | luke-jr | is it possible to send a callerid like 1NPANXXXXXX"TTT0000#"1NPANXXXXXX ? |
| 16:04.58 | Qwell | if your provider accepts it and passes it on, sure |
| 16:06.01 | luke-jr | the two I just tried (Flowroute and voip.ms) just hang :( |
| 16:06.13 | Mango | ...why? |
| 16:06.29 | luke-jr | as in, no call at all, not even with truncated or wrong CID |
| 16:06.39 | luke-jr | Mango: so if I return a call, it goes through DISA :) |
| 16:07.09 | luke-jr | my cell phone uses "T" for a delay |
| 16:07.35 | ManxPower | luke-jr, I doubt any provider would pass that |
| 16:07.51 | luke-jr | oh well :( |
| 16:08.12 | luke-jr | when will stupid cell phone carriers start offering simple SIP? :/ |
| 16:08.16 | ManxPower | the quotes alone would be a problem |
| 16:08.26 | *** join/#asterisk hluesea (~hluesea@ip-72-55-148-6.static.privatedns.com) |
| 16:08.34 | luke-jr | the quotes were for clarity... the actual CID wouldn't have them |
| 16:08.51 | hluesea | hello, is any one main problem of the chan_sip.c:3782 retrans_pkt warning ? |
| 16:09.01 | ManxPower | kinda hard to get the right answer when you ask the wrong question |
| 16:09.12 | ManxPower | hluesea, NAT/Firewall issues |
| 16:09.19 | hluesea | hımm ok thank you |
| 16:09.54 | hluesea | <PROTECTED> |
| 16:10.08 | ManxPower | I doubt it |
| 16:10.09 | hluesea | or how can i fixed this problem for the trunk |
| 16:10.25 | ManxPower | hluesea, fix your NAT or Firewall issue. |
| 16:10.32 | ManxPower | ~sipnat |
| 16:10.32 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:14.08 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.225.159) |
| 16:17.37 | Micc | Anyone have experience with AudioCodes ATAs and T38? Do they work well? |
| 16:18.02 | idespinner | they work but theyre a pain to configure |
| 16:18.28 | Micc | Is there a web admin? |
| 16:18.31 | idespinner | yes |
| 16:18.52 | pabelanger | idespinner: +1 for that, audiocodecs are a pain to configure |
| 16:19.05 | idespinner | for FXS its a bit easier but still not a simple task |
| 16:19.19 | idespinner | i do another sip PBX that uses primarily audiocodes |
| 16:19.32 | idespinner | takes a while to get used to them but they do T38 just fine |
| 16:20.08 | Micc | Any sites I can read up on how to configure or is it just going to take some playing with to get used to it? |
| 16:20.39 | idespinner | ive googled before but there isnt a whole lot out there on them... |
| 16:21.08 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 16:22.13 | exothermc | Hi I have an odd issue where asterisk 1.6.2.7 is sending out invites with what seems to be mal formed headers. Specifically the To: header To: "Unavailable" <sip:extension @host>;tag=as10dbc7e4 Notice the space between the extension and @ sign. |
| 16:24.19 | exothermc | Actually sorry it is the From and Contact header that it is doing that too. |
| 16:24.32 | exothermc | the To is fine. |
| 16:25.48 | pabelanger | exothermc: can you pastebin your sip.conf? |
| 16:26.18 | exothermc | pabelanger: which part general or the extension or both? |
| 16:26.43 | pabelanger | exothermc: extension should be good for now |
| 16:27.21 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 16:28.49 | exothermc | pabelanger: http://pastebin.com/5N7dQ3Ln |
| 16:29.44 | exothermc | pabelanger: btw both the contact and from coming in from external show proper formatting, but then when asterisk passes the call along is where they get mangled. |
| 16:30.00 | [TK]D-Fender | exothermc: Show us the failed call with SIP DEBUG |
| 16:30.07 | pabelanger | exothermc: looks good. Can you capture a debug log of an incoming call? |
| 16:30.13 | exothermc | [TK]D-Fender: it doesn't fail. |
| 16:30.29 | [TK]D-Fender | exothermc: Malformed, whatever. Show us the PROBLEM |
| 16:30.48 | exothermc | I just noticed I have the limitonpeers in there twice. doubt that effects anything. |
| 16:33.05 | exothermc | How do I turn on sip debug in asterisk, or do I just need to show you the packet texts? |
| 16:33.48 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-exukqaevwphvnapq) |
| 16:34.28 | pabelanger | ~collectdebug |
| 16:34.29 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:36.35 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:36.37 | cusco | hi |
| 16:37.05 | cusco | is there any problem with connecting 4 different PRI lines from 4 different TELCO in the same PRI card? |
| 16:37.22 | Qwell | cusco: nope, should be fine |
| 16:37.28 | cusco | ok |
| 16:37.30 | cusco | another quetion |
| 16:37.31 | pabelanger | cusco: shouldn't be. Just make sure you have your timing setup properly |
| 16:37.36 | cusco | err |
| 16:37.46 | cusco | the timings should go each with theyr telco, right? |
| 16:37.48 | cusco | or else what? |
| 16:37.58 | cusco | what happens if I align timings with different telcos? |
| 16:43.26 | ManxPower | Digium cards only support one timing source per card, so if you have multiple providers going into one card, you should set the timing source to one of the spans and hope for the best. |
| 16:43.31 | ManxPower | (It usually works just fine) |
| 16:45.06 | *** join/#asterisk imox1234 (~imox1234@p4FC5C424.dip0.t-ipconnect.de) |
| 16:45.33 | *** part/#asterisk kth5 (~kth5@archlinux/developer/kth5) |
| 16:45.41 | cusco | is it? |
| 16:45.43 | cusco | :o |
| 16:49.57 | imox1234 | hello how can i change the device name from asterisk. as default it is Asterisk PBX 1.6.2.6 i think. i dont know how called this but i hope you know what i mean :-) ? |
| 16:51.38 | pabelanger | imox1234: if you are referring to useragent, yes. sip.conf. Why though? |
| 16:52.19 | imox1234 | sip conf ? |
| 16:53.30 | pabelanger | ~book |
| 16:53.30 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:53.34 | pabelanger | imox1234: ^^^ |
| 16:54.04 | imox1234 | what should i write in the sip.conf ? :-) ? |
| 16:54.08 | *** part/#asterisk vlt (~dm@suez.activ-job.com) |
| 16:54.50 | *** join/#asterisk hfb (~hfb@pool-98-112-146-142.lsanca.dsl-w.verizon.net) |
| 16:55.02 | pabelanger | imox1234: Have you read it? It tells you in the sample sip.conf file. |
| 16:55.31 | pabelanger | imox1234: http://asterisk.pastebin.com/0aKWpSjR |
| 16:55.51 | imox1234 | ok thanks |
| 17:07.23 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 17:08.14 | imox1234 | when i registred a sip account from my asterisk how can i found out the useragent from the other serveR ? |
| 17:08.27 | cusco | how do I know if the PCI card I need must be 3.5 volts or 5volts (digium card) |
| 17:09.18 | cusco | dmidecode |grep PCI -C10 tells me the current one is 3.3v but there is a free slot with 5v ? |
| 17:10.49 | cusco | and another question, today two clients called and once they were answered while queuing, the operators were talking to each other instead of talking with clients |
| 17:10.54 | cusco | how can I debug that? |
| 17:11.20 | *** join/#asterisk Z_God (~julius@wlan227084.mobiel.utwente.nl) |
| 17:12.10 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 17:12.19 | pabelanger | imox1234: sip set debug on |
| 17:12.57 | pabelanger | cusco: read your motherboard manual |
| 17:13.27 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:14.11 | p3nguin_ | imox1234: sip show peer <that peer name here> |
| 17:18.21 | imox1234 | i get a attack from this IP adress |
| 17:18.22 | imox1234 | 91.121.147.130 |
| 17:18.25 | imox1234 | every time |
| 17:18.43 | imox1234 | can i set this ip on a black list by asterisk ? |
| 17:19.13 | imox1234 | 41.131.135.108 |
| 17:19.17 | imox1234 | and this here |
| 17:21.23 | pabelanger | imox1234: http://blogs.digium.com/2009/03/28/sip-security/ |
| 17:21.33 | pabelanger | imox1234: Or setup a firewall |
| 17:21.47 | imox1234 | i use iptable :D |
| 17:22.48 | pabelanger | imox1234: then setup better firewall rulesets. Only accept SIP packets from specific IPs |
| 17:23.03 | imox1234 | hmm ok thanks |
| 17:24.34 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 17:25.26 | Naikrovek | yes iptables is your answer |
| 17:25.30 | Naikrovek | perhaps your router |
| 17:26.14 | imox1234 | and better passwords :-9 |
| 17:26.16 | imox1234 | :-) |
| 17:27.06 | bent_screwdriver | does anyone know a good NPA/NXX source for determining how a number should be dialed for a given area? |
| 17:28.23 | dohd | well, well, improvement... dahdi for x100p working on freebsd a little |
| 17:28.35 | dohd | can do incoming calls, but something wrong with outgoing :-( |
| 17:31.08 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:34.10 | leifmadsen | I'm surprised x100p stuff works on anything anymore :) |
| 17:35.30 | Qwell | I'm surprised people still try to use them |
| 17:35.36 | Qwell | I'm surprised people ever tried to use them |
| 17:36.00 | dohd | :) well, and what would a reasonable alternative be? |
| 17:37.51 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 17:38.19 | [TK]D-Fender | dohd: TDM410P. A200d, SPA-3102 |
| 17:38.49 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 17:40.26 | *** part/#asterisk minaguib (~mina@modemcable115.49-57-74.mc.videotron.ca) |
| 17:41.34 | paulc | @leifmadsen: ping! You get my email the other week? |
| 17:41.48 | leifmadsen | paulc: yes, it is still marked as unread as I haven't had a chance to read it :( |
| 17:41.54 | leifmadsen | but I did get the email |
| 17:41.57 | leifmadsen | thanks for the response! |
| 17:42.02 | leifmadsen | sorry for the delay |
| 17:42.17 | paulc | leifmadsen: No worries - I have a bunch of those in my inbox too.. "I'll get to this one, uh.. later.." |
| 17:42.31 | leifmadsen | I just followed up on an email from 2009 today :) |
| 17:42.36 | paulc | leifmadsen: Skim it when you get a chance and me know if it gels well or not |
| 17:42.43 | leifmadsen | will do sir! |
| 17:42.47 | paulc | LOL.. delayed reply much?! 2009.. wow |
| 17:43.00 | leifmadsen | haha well it was a request for some linkedin review |
| 17:43.03 | leifmadsen | I finally wrote it :) |
| 17:43.08 | leifmadsen | I think it was Dec 2009 so not SOOOOO bad |
| 17:43.11 | leifmadsen | only 5 months :) |
| 17:43.52 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 17:45.00 | *** join/#asterisk AndyML (~AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
| 17:46.35 | AndyML | I've just read through voicemail.conf.sample and couldn't find what i'm looking for - is there a way to disable the post-greeting greeting from Allison that says "please leave a message after the one" etc? |
| 17:47.02 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 17:47.28 | leifmadsen | AndyML: I think you want something like the 's' flag to Voicemail() |
| 17:48.17 | *** join/#asterisk neurosys (~neurosys@32.160.227.216) |
| 17:48.31 | AndyML | nice leifmadsen, thanks. |
| 17:52.58 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
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| 18:07.39 | tzafrir_laptop | Qwell, as for your question of "who needs them": the customers of x100p.eu, who need quality products |
| 18:16.21 | leifmadsen | tzafrir_laptop: I detect sarcasm :) |
| 18:25.28 | Baylink-work | leifmadsen: You can be replaced by a bot. :-) |
| 18:25.41 | leifmadsen | really? awesome! |
| 18:25.49 | Baylink-work | You didn't see the /. piece? |
| 18:25.56 | leifmadsen | bots can spend money, so I'll still collect money! |
| 18:26.04 | leifmadsen | I haven't read slashdot since about 2007 |
| 18:26.30 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 18:28.24 | Baylink-work | http://www.freshnews.org/article/slashdot/software-recognizes-sarcastic-tweets |
| 18:28.28 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:29.54 | *** join/#asterisk ksn (~ksn@93-35-4-129.ip52.fastwebnet.it) |
| 18:31.30 | ksn | any idea of why a fxo card doesn't tick? |
| 18:32.38 | tzafrir_laptop | hardware issue? |
| 18:32.49 | tzafrir_laptop | do you see any error from the driver? |
| 18:32.54 | ksn | it works on a trixbox server, not in asterisk |
| 18:32.57 | ksn | no errors, ever |
| 18:33.05 | tzafrir_laptop | fxo card: x100p or something similar? |
| 18:33.11 | ksn | tdm400 |
| 18:34.13 | tzafrir_laptop | do you see a span for it under /proc/dahdi ? (or /proc/zaptel ) |
| 18:34.36 | ksn | y |
| 18:34.37 | *** join/#asterisk sa000 (fressh@unaffiliated/sa000) |
| 18:34.44 | ksn | i can see the channels in asterisk |
| 18:34.47 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:35.00 | ksn | the problem is that seems all fine, but when i call i get the free line tone |
| 18:35.10 | ksn | but asterisk doesn't recognize the cazz |
| 18:35.12 | ksn | call |
| 18:35.26 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 18:35.42 | sa000 | hi! i am doing some jitter calculation using asterisk pbx. Now i need asterisk compititor just to see the results difference. Can anyone tell me asterisk compititor product ? |
| 18:35.48 | sa000 | is there any ? |
| 18:37.33 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 18:39.10 | p3nguin_ | You think the software used on the server causes jitter? |
| 18:40.18 | [TK]D-Fender | sa000: FreeSWITCH, Yate, SIPX, 3CX |
| 18:40.37 | sa000 | p3nguin: i did simple experiment computer a - Asterisk server - computer b and than run wireshark on both computer |
| 18:40.55 | sa000 | [TK]D-Fender: are these pbx ? |
| 18:41.03 | [TK]D-Fender | sa000: Clearly |
| 18:41.11 | sa000 | yes thanks |
| 18:41.16 | sa000 | just check google |
| 18:41.46 | sa000 | actually asterisk is the well know :) . never heard of these :P |
| 18:42.45 | ksn | do you need a particular kernel config for the fxo card to tick? |
| 18:42.53 | ksn | like timers or something like that |
| 18:43.15 | sa000 | [TK]D-Fender: I only need one. which should i take to compare with asterisk pbx ? |
| 18:43.21 | sa000 | just a suggestion ;) |
| 18:44.03 | [TK]D-Fender | sa000: I suggest you get off your ass and just PICK ONE. You could use a single soft phone as far as I can see for this test. |
| 18:44.05 | AndyML | sa000: definitely yate heh |
| 18:44.43 | sa000 | nice |
| 18:45.07 | voxter | If my asterisk 1.4 box has the time/date set incorrectly, will changing the date have any adverse affects on calls in progress? |
| 18:45.41 | sa000 | single softphone .. hmm .. how .. if i ask more than i guess you will be more rude |
| 18:46.22 | AndyML | well, there are other people in the channel... |
| 18:46.29 | AndyML | download x-lite |
| 18:46.36 | Baylink-work | voxter: I can't speak to asterisk specifically, but non-monotonic time jumps are rarely tolerated well. I would expect that CDRs would be wrong, at the very least. |
| 18:46.38 | AndyML | it's a sip softphone and should work fine. |
| 18:47.00 | sa000 | AndyML: i am using xlite |
| 18:47.18 | AndyML | then you have what you need. good luck! |
| 18:47.42 | sa000 | AndyML: thanks |
| 18:47.55 | *** part/#asterisk AndyML (~AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
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| 18:51.51 | leifmadsen | voxter: I don't think so... how far out is it? |
| 18:52.13 | leifmadsen | voxter: generally it is best to restart after a date update though I think |
| 18:52.27 | voxter | leifmadsen: yeah... it thinks its 2008 |
| 18:52.34 | leifmadsen | lol |
| 18:52.36 | leifmadsen | good vintage |
| 18:52.50 | leifmadsen | I'd wait till after hours then |
| 18:53.01 | voxter | I think i will, yeah. |
| 18:53.09 | leifmadsen | just in case :) |
| 18:54.01 | ksn | anyone can give me a clue for my problem? things to check etc? :( i'm desperate |
| 18:54.11 | ksn | working on it since 14 hours and counting |
| 18:56.13 | [TK]D-Fender | ksn: I don't see you mentioning what OS< kernel, * version, DAHDI version, providing debug output, or much else yet.... |
| 18:57.12 | ksn | yeah because i debugged so many different things, i'll start with the basics sec |
| 18:58.23 | ksn | dahdi 2.2.0.2, kernel 2.6.27-gentoo-r8, Asterisk 1.6.2.6. The debug seems fine, as it's like asterisk doesn't recognize the call in the first place |
| 18:58.49 | [TK]D-Fender | ksn: Pastebin card configs, CLI tests, etc |
| 18:58.54 | [TK]D-Fender | ~pb |
| 18:58.55 | 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.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. |
| 18:59.19 | ksn | oki |
| 19:04.41 | ksn | this is the first things who got into my mind http://pastebin.com/EwiBTp7A |
| 19:05.31 | ksn | if you have any idea of where i can get more debug i'll do it :p |
| 19:07.08 | ksn | actually i pasted a line more of the dmesg, that was me reloading the module ofc |
| 19:07.36 | voxter | does asterisk do its own log rotation or does it rely on logrotate to generate the 'log.#' files ? |
| 19:09.22 | p3nguin_ | You'll need to use another rotation mechanism. |
| 19:09.54 | tzafrir_laptop | ksn, how exactly do you see that the card does not tick? |
| 19:10.01 | voxter | weird, my logrotate.conf says only keep 4 weeks, but my asterisk logs will fill up the drive if i dont police it |
| 19:10.10 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 19:10.18 | ksn | dahdi_test hangs, and no interrupts on /proc/interrupts |
| 19:10.43 | WIMPy | Asterisk will auto-logrotate if the log gets >2GiB. |
| 19:10.44 | tzafrir_laptop | voxter, you can tell logrotate to also rotate that specific log by size |
| 19:11.14 | tzafrir_laptop | Note that logrotate runs daily |
| 19:12.06 | voxter | tzafrir_laptop: sure yeah, my point though, is that logrotate seems to be instructed to keep only 4 weeks worth of logs, when in actuality, there were more like 3 years of logs kept |
| 19:12.09 | tzafrir_laptop | ksn, for starters, asterisk is not relevant at this point. This is something to do with the kernel-level stuff |
| 19:12.31 | *** join/#asterisk jtodd (qyflrdc847@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 19:12.31 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:12.49 | p3nguin_ | logrotate can run as often as you want it to run. |
| 19:13.16 | p3nguin_ | logrotate can also keep as many or as few rotated logs as you configure. |
| 19:14.09 | ksn | tzafrir_laptop, yeah that's what i thought, still i'm not sure if this is connected to the fact asterisk doesn't pick up calls, or if even if a card is not ticking asterisk can work |
| 19:14.14 | tzafrir_laptop | Did it look at the right file? |
| 19:14.31 | voxter | Yes, im familiar with logrotate and how it works - maybe i will rephrase my question. Have any of you experienced logrotate not following the configuration its been told to, specifically with asterisk logfiles? |
| 19:15.22 | voxter | logrotate.conf says to keep a maximum of 4 weeks worth of logs. In /var/log/asterisk there were logs such as full, queue_log, etc, from .0, .1, .2 all the way to .26 |
| 19:15.31 | voxter | indicating to me that it kept .26 rotations |
| 19:16.31 | voxter | hmm, i suppose if each of those is one day, that makes sense, but unless they were each 2gb and asterisk rotated them on its own |
| 19:17.08 | p3nguin_ | I have the current file and 4 rotated and gzipped files. No more. |
| 19:18.04 | p3nguin_ | Check to see if the size of all your rotated files is the same. If so, then it was likely rotated based on the size instead of the day. |
| 19:19.04 | voxter | yeah, unfortunately i deleted all the logs before I could check that. duhh. I'll look on another similar box. |
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| 19:37.55 | *** join/#asterisk eddyo (~Edd@91.144.31.136) |
| 19:38.17 | eddyo | hi people, how can i edit the default port of iax2 from 4569 to 5555 ? |
| 19:40.08 | pabelanger | eddyo: iax.conf bindport |
| 19:40.49 | p3nguin_ | Why would you want to do that? |
| 19:41.06 | eddyo | they have blocked the default port in my country |
| 19:41.47 | eddyo | what is iax.conf located? |
| 19:42.18 | eddyo | well |
| 19:42.23 | eddyo | sorry this is not my question |
| 19:42.28 | eddyo | there are so many iax.conf |
| 19:42.31 | eddyo | which one i should edit |
| 19:42.42 | [TK]D-Fender | eddyo: Go back to #freepbx |
| 19:42.59 | p3nguin_ | lol |
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| 19:44.27 | Defraz | Anyone have any experience with a bitboss ISDN pri and connecting a cisco router? |
| 19:50.07 | ManxPower | have you tried setting it up as a non-bitboss PRI? |
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| 19:59.27 | *** join/#asterisk imox1234 (~imox1234@p4FC5C424.dip0.t-ipconnect.de) |
| 19:59.40 | KavanS | how long is the average "ring" for a telephone? |
| 19:59.51 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:59.51 | leifmadsen | 24 seconds (4 rings) |
| 19:59.55 | p3nguin_ | In the US, 2 on, 4 off. |
| 20:00.03 | p3nguin_ | seconds, that is. |
| 20:00.11 | KavanS | ok cool, I'll make note of this, thank you |
| 20:00.19 | leifmadsen | 6 seconds per ring cycle typically |
| 20:00.24 | leifmadsen | (per p3nguin_ ) |
| 20:00.31 | imox1234 | hello, how can i import the cdr_csv log from asterisk in Excel ? |
| 20:00.44 | imox1234 | it write it only the first cell |
| 20:01.31 | [TK]D-Fender | imox1234: #excel <------------- |
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| 20:40.30 | sat-man | who wants to help the new asterisk user |
| 20:40.46 | rubbs | ~book |
| 20:40.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:40.57 | sat-man | who's the cheapest sip provider these days |
| 20:41.08 | Qwell | Do you really want the cheapest? |
| 20:41.16 | sat-man | just to get started |
| 20:41.22 | sat-man | not for production, personal practice |
| 20:41.35 | sat-man | something i can byod too of course |
| 20:43.24 | ManxPower | the most providers will be cheap enough for you |
| 20:45.54 | p3nguin_ | ~itsp |
| 20:45.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:46.05 | p3nguin_ | ~itsplist-us |
| 20:46.06 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:46.22 | p3nguin_ | sat-man: ^^^^^^^^^^^^^ |
| 20:48.30 | sat-man | thanks! |
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| 21:05.05 | *** join/#asterisk edIII (~dont@75-140-45-90.static.reno.nv.charter.com) |
| 21:06.34 | edIII | Hello |
| 21:06.40 | *** join/#asterisk Logicwrath (~no@c-68-43-112-9.hsd1.mi.comcast.net) |
| 21:07.36 | Logicwrath | there is one particular phone number that when i call their IVR does not recognize my DTMF sounds and when i get defaulted to the operator they cannot hear me. i have no other audio issues and I am not behind a firewall. What can I look at? |
| 21:12.15 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 21:13.14 | ruben23 | hi there i have setup DID number on my asterisk server adn when i cal the DID# i got this on the CLI logs--------> http://pastebin.com/8Pu6veWj ----> and incoming calls are not getting through.. |
| 21:13.20 | ruben23 | any idea on this.. |
| 21:14.07 | edIII | Can anyone help me with development information for realtime asterisk using CURL? |
| 21:17.46 | tzafrir_laptop | edIII, for starters, have you looked at pbx_config_curl (or whatever it is called?) |
| 21:18.04 | tzafrir_laptop | ruben23, something is wrong with sip.conf? |
| 21:18.44 | tzafrir_laptop | edIII, btw: you don't have to wait for so long before asking |
| 21:19.36 | ruben23 | <PROTECTED> |
| 21:20.16 | Logicwrath | ruben: who is the voip trunk with? |
| 21:20.43 | Logicwrath | oh i see its DID nvm |
| 21:21.21 | Logicwrath | i only ask because sometimes you need to have a special register statement with diff providers |
| 21:21.34 | Logicwrath | broadvoice for example is real picky |
| 21:22.25 | ruben23 | <PROTECTED> |
| 21:22.43 | tzafrir_laptop | you have "did_new" as the username, and you expected it to be 7037637134 ? |
| 21:23.29 | ruben23 | tzafrir_laptop: "did_new" is the context... |
| 21:23.58 | Logicwrath | ruben you might want to change your password too on that DID |
| 21:24.19 | Logicwrath | you left it in the register line |
| 21:24.40 | Logicwrath | who is the provider? |
| 21:24.51 | pabelanger | ruben23: context=did_new |
| 21:24.53 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 21:25.45 | muiro | Does anyone know of any good audio annotation software? I'm looking at a pile of recorded calls that I need to go through and tag for potential audio problems and I want to see if there's a good audio annotation program I could use to assist me |
| 21:25.47 | ruben23 | pabelanger:i mean im suing the default for the dial plan.. sorry did_new - is the name of my trunk |
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| 21:39.16 | *** join/#asterisk shader (~user@janustw.tavve.com) |
| 21:39.26 | shader | can you indent the .conf files? |
| 21:40.07 | *** join/#asterisk TheDracle (~jthomas@209.169.244.29) |
| 21:40.32 | TheDracle | Where would I go to ask questions about OPAL/OpenH323? |
| 21:41.01 | Slugs_ | TheDracle, what's the question? |
| 21:41.15 | TheDracle | Slugs_, Heh... Well ;p Ptlib keeps blowing up... |
| 21:41.42 | TheDracle | Slugs_, I'm porting it to a embedded device, and I'm getting "Catastrophic failure, PProcess::Current() = NULL!!" |
| 21:42.22 | Slugs_ | your building ptlib src and that's ur error on the device? |
| 21:42.28 | TheDracle | Slugs_, I GDB traced it, and tracked it to have something to do with object creation with PTlib. I.E: PProcessInstance is set to a value, an object (even one with an empty constructor) is initialized, and it ends up NULL afterwards. |
| 21:42.45 | TheDracle | Slugs_, Exactly. |
| 21:43.03 | TheDracle | Slugs_, It's like, for no apparent reason, PProcessInstance is getting clobbered... |
| 21:43.09 | *** join/#asterisk mducharme-work (~nothing@4-121-188-206.rev.knet.ca) |
| 21:43.18 | mducharme-work | good afternoon |
| 21:43.25 | TheDracle | Slugs_, It works all the way up until creating a new PThread, and trying to initiate a 323 call. |
| 21:43.31 | Slugs_ | TheDracle, where did you get the src? |
| 21:43.47 | mducharme-work | we have an asterisk phone system.. the gentleman managing it says it's not possible to change the directory recording |
| 21:43.47 | shader | is there any reason to require dialing 9 before an external number? A lot of people seem to do that. |
| 21:43.54 | mducharme-work | is that true? |
| 21:44.11 | shader | mducharme-work: probably not |
| 21:44.13 | TheDracle | Slugs_, http://sourceforge.net/projects/opalvoip/files/ |
| 21:44.14 | Qwell | you have the sourcecode. all the prompts are available. |
| 21:44.21 | Qwell | there's nothing you can't change. |
| 21:44.39 | Slugs_ | TheDracle, one sec |
| 21:44.41 | [TK]D-Fender | shader: It was never "required". It was common in PBX dating back 20 years and some morons are incapable of change |
| 21:44.48 | mducharme-work | it says "enter the first three letters of your party's first name" instead of "enter the first three letters of your party's first or last name" |
| 21:45.04 | mducharme-work | it used to say "first or last name", he says it's impossible to change back to that |
| 21:45.06 | Qwell | mducharme-work: then the first and last name option isn't set |
| 21:45.14 | mducharme-work | he says it is set |
| 21:45.17 | shader | [TK]D-Fender: so, the only reason to have it in the dialplan is to let people dial it anyway and still reach their number? |
| 21:45.32 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 21:45.50 | [TK]D-Fender | shader: It allows you faster dialing of common patterns that DON'T start with 9... |
| 21:46.07 | shader | how much faster? |
| 21:46.16 | shader | by default, that is |
| 21:47.14 | Slugs_ | TheDracle, did you get the latest trunk? |
| 21:47.16 | Slugs_ | svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk ptlib |
| 21:47.20 | Slugs_ | this works for me... |
| 21:47.22 | TheDracle | Slugs_, Basically in h323.cxx, H323Connection::SetUpConnection() creates a new PThread, which attaches to StartOutgoing. |
| 21:47.45 | TheDracle | Slugs_, Hm, I'll try that. |
| 21:47.57 | TheDracle | Slugs_, Hopefully my patches will apply :) |
| 21:47.57 | shader | it probably would have been better if the first person to make a phone had used a "number completed" signal, instead of just waiting for more numbers |
| 21:47.59 | [TK]D-Fender | shader: When the system knows that you are dialing a 4 digit # because it starts with a 5 instead.... well.. you don't have to sit around at the 4th digit |
| 21:48.09 | Slugs_ | TheDracle, second of all i didnt think opal worked for *? |
| 21:48.21 | Slugs_ | im using opal w/ fs |
| 21:48.32 | TheDracle | Slugs_, Hm, I think it used OpenH323. |
| 21:48.45 | TheDracle | Slugs_, But, they are related- I'm not sure where to ask about Opal :) |
| 21:48.47 | Slugs_ | well that's diff |
| 21:49.03 | Slugs_ | simiar but definitly not the same |
| 21:49.38 | TheDracle | Yes, I know, I figured people who knew about OpenH323 may know about Opal :) |
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| 21:57.20 | shido6 | what do you guys use to measure voice quality between phones? |
| 21:57.38 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 21:57.51 | [TK]D-Fender | uses his ears |
| 21:58.01 | shido6 | :) |
| 21:58.03 | shido6 | hey TK |
| 21:58.05 | shido6 | long time |
| 21:59.03 | Qwell | I don't know about you guys, but I don't want a mean opinion about something like voice quality. |
| 21:59.18 | Qwell | I'd much rather have a nicer opinion. |
| 21:59.47 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 22:01.10 | shido6 | if I have phones in another part of the world that I need to monitor and know if there are issues before people find me and shoot me |
| 22:01.24 | shido6 | that came out wrong |
| 22:02.05 | shido6 | I have phones all over and I need to monitor the quality of conversations betfore the users of these phones find me, track me down and .... do very bad things |
| 22:02.16 | shido6 | to me |
| 22:03.30 | *** join/#asterisk pwell (~pwell@pool-72-80-196-178.nycmny.east.verizon.net) |
| 22:14.06 | *** join/#asterisk AdamTwelve (~Adam@adsl-71-131-134-131.dsl.sntc01.pacbell.net) |
| 22:15.42 | shader | should I have three separate entries for 1-areacode-number, areacode-number, and number? Or is having one entry for just the number sufficient? |
| 22:16.12 | shader | i.e., do dialplan patterns match on the end of a series of numbers, or the beginning? |
| 22:18.17 | AdamTwelve | Hi, is this a place I could get some help connecting a remote softphone to my new trixbox install? |
| 22:21.17 | [TK]D-Fender | shader: matches on positions within the string |
| 22:23.51 | shader | so, is there one pattern that can match all of the different versions of the same number(1234567,NXX1234567,1NXX1234567), or do I need more than one pattern? |
| 22:24.34 | [TK]D-Fender | shader: Clearly multiple patterns |
| 22:24.36 | shader | ok |
| 22:26.56 | TheDracle | Slugs_, Yeah, ported it, exact same behavior :( |
| 22:27.06 | AdamTwelve | I am on Trixbox 2.8.0.3 and am just not sure where to configure the nat settings. should it be under sip_general_custom.conf or sip_net.conf |
| 22:27.22 | AdamTwelve | sip_nat.conf that is |
| 22:27.31 | [TK]D-Fender | AdamTwelve: the former, plus specific settings in your peers |
| 22:27.49 | [TK]D-Fender | AdamTwelve: split this guide's advice into the right places for FreePBX : |
| 22:27.51 | [TK]D-Fender | ~sipnat |
| 22:27.52 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:29.04 | AdamTwelve | Thanks |
| 22:30.06 | edIII | <tzafrir_laptop> edIII, for starters, have you looked at pbx_config_curl (or whatever it is called?) >> I can't find that file |
| 22:30.28 | tzafrir_laptop | edIII, what version of asterisk do you use? |
| 22:30.35 | edIII | However, if you are referring to the CGI configuration file dbsep.conf, then I have looked at it |
| 22:30.44 | edIII | It won't help me |
| 22:31.10 | edIII | I can't use CGI, don't want to use CGI either. Additionally, it is difficult to get Firebird supported |
| 22:31.50 | edIII | I have my own classes in PHP for Firebird and I am trying to construct an API for Asterisk Realtime with CURL |
| 22:32.23 | edIII | What I cannot seem to find out anywhere, Doxygen included, is the data structures being passed by Asterisk |
| 22:32.43 | edIII | I am using 1.6.1.13 I think |
| 22:32.44 | edIII | hold on |
| 22:33.02 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 22:33.02 | edIII | 1.6.1.12 |
| 22:33.27 | edIII | We were going to schedule an upgrade to the latest 1.6.2.x this weekend |
| 22:34.27 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 22:36.18 | tzafrir_laptop | edIII, I meant res_config_curl |
| 22:36.34 | edIII | Would I find that in the source? |
| 22:36.53 | tzafrir_laptop | it is a real-time engine that fetches configuration at real-time through cural rather than from a "real" database |
| 22:37.02 | tzafrir_laptop | res/res_config_curl.c |
| 22:39.21 | edIII | I found it.... |
| 22:40.13 | edIII | Do I really have to reverse engineer that? Can anybody just give me examples of what Asterisk posts? The data structure it expects to receive? |
| 22:40.52 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:46.12 | tzafrir_laptop | edIII, I can't find documentation, but you can see that CGI (contrib/scripts/dbsep.cgi ) to see how the server side should behave |
| 22:46.29 | edIII | I know, it just is not helping at all |
| 22:47.01 | edIII | Reverse engineering it is a major pain in the butt when it is not your language in the first place |
| 22:47.21 | edIII | I was hoping somebody could just give me examples of the post and response documents |
| 22:48.00 | edIII | Realtime without CURL would not be so bad if Asterisk didn't *hate* Firebird so much |
| 23:01.15 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 23:02.58 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 23:03.43 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:04.12 | joako | Using setcallingpres I can get the SIP header to transmitt privacy=full. How can I make it transmit Privacy=id? |
| 23:05.51 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:11.20 | tzafrir_laptop | edIII, maybe you can use firebird through odbc? |
| 23:15.07 | edIII | 1) That is really hard. Getting reliable working UnixODBC drivers is not easy to do and it seems like it is not being supported or developed very well |
| 23:15.42 | edIII | 2) Asterisk itself is incredibly Firebird unfriendly. Could not be more so if it tried. It uses reserved words all over the place |
| 23:16.31 | edIII | I got lucky replacing "password" with "secret" in the source for the voicemail system, but other than that... still have plenty of issues relating to reserved words |
| 23:17.46 | edIII | Asterisk using CURL allows me to create an API (willing to share that) that would allow any number of database types to be supported, load balancing, failover, multi-company, security, etc. |
| 23:18.06 | edIII | It's API for the win. |
| 23:20.37 | edIII | Additionally, I am looking into making API calls directly from the dial plan to accomplish other things too. So there are a lot of reasons why I want to concentrate development on the API |
| 23:27.56 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:37.26 | tzafrir_laptop | edIII, there's the function CURL |
| 23:37.33 | tzafrir_laptop | core show function CURL |
| 23:39.31 | tzafrir_laptop | e.g.: Set(VAR=${CURL(http://server/path)}) |
| 23:43.04 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:44.52 | edIII | Interestingly enough CURL does not show up in the list of functions |
| 23:45.14 | edIII | I know I had it compiled because I have been testing CURL in Realtime |
| 23:46.18 | SaiSoma|AtHome | Hey Polycom guys, I suspect I'm having an issue with NAT timeout to my * box (331's behind NAT tp blic ip'). Will the setting nat.keepalive.interval be the right fix perhaps? |
| 23:48.09 | WIMPy | SaiSoma|AtHome: If the phone becomes unreachable after some amount of inactivity, yes. |
| 23:50.23 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:50.24 | SaiSoma|AtHome | WIMPy: I can see * sending to the phone (ringing SIP/<ext>) but the phone does not ring until a reboot, and then only until idle for the first time. sip notify commands also do not go through, but you can pick up the phone and dial out, so I suspect that is the case. Any other way to tell? |
| 23:50.50 | SaiSoma|AtHome | I show "unmonitored" on all my sip lines in "sip show peers" Should that be otherwise? |
| 23:50.59 | WIMPy | Yes, that sounds like the router forgetting about that "connection". |
| 23:51.28 | WIMPy | Depends on what you want :-) |
| 23:51.45 | SaiSoma|AtHome | WIMPy: thanks. phones all reloading (via web int), now I would give my right %D$AS to have someone IN that office to listen for a ring, heh. |
| 23:51.57 | WIMPy | But the qualify messages are usually not frequent enought to keep the netted route alive. |
| 23:52.14 | WIMPy | natted |
| 23:52.30 | SaiSoma|AtHome | WIMPy: is qualify=yes a good thing anyway? Should I have that in place? It seemss like I should. |
| 23:52.33 | WIMPy | . o O ( why do I double the t? ) |
| 23:53.04 | WIMPy | If you have, Asterisk will regularly check if your peers are reachable. |
| 23:53.12 | SaiSoma|AtHome | WIMPy: no worries. My brain is half fried from 10 hours of work already, I never even noticed (and have to ignore my lovely typos). |
| 23:53.27 | SaiSoma|AtHome | WIMPy: thanks. this is perfect and exactly what I needed. :) |
| 23:58.11 | *** join/#asterisk puckett_jw (~puckett_j@70.166.66.188) |
| 23:58.46 | puckett_jw | Can someone help me out with the following⦠exten => s,n,Set(LOCAL(nextSet)=${IF($["${CDR(dcontext)}" != "${CDR(accountcode)}"])?${CUT(ARG1,:,2-)}}) |
| 23:58.48 | paulc | Alternative is to set a low refresh/reregister time, the NAT timer thing doesn't work out |
| 23:59.33 | SaiSoma|AtHome | paulc: *nod* i've seen that too. hoping the NAT piece works so I don't have to worry with the vm chip. |
| 23:59.43 | puckett_jw | If is giving the following error⦠[May 17 13:53:19] WARNING[4699]: func_logic.c:188 acf_if: Syntax IF(<expr>?[<true>][:<false>]) (expr must be non-null, and either <true> or <false> must be non-null) |