| 00:04.36 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
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| 00:06.36 | jimi_ | !trunks |
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| 00:27.52 | nsgn | goodevening. this stupid grandstream handset can't seem to get DTMF to control asterisk IVR menus and such. what the heck? the dtmf palyload type is set to 101 in the phone..which is what asterisk supports is it not? |
| 00:28.47 | nsgn | i know the IVRs are responding because calling from an outside phone works when sending tones to control the IVR |
| 00:32.01 | jimi_ | !gs |
| 00:32.04 | jimi_ | ~gs |
| 00:32.05 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 00:32.35 | nsgn | jimi_, i'm aware, but there's one existing here that would be wonderful to use for this simple purpose. polycom is hands down my preference that i use everywhere i choose |
| 00:35.05 | nsgn | bah, it looks like this moronic phone defaults to dtmf in audio only. apparently asterisk doesnt listen for audio dtmf on local extensions. lemme see if this resolves it. didnt realize the phone wasnt sending this the way the polys do by default -_- |
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| 00:38.15 | nsgn | bah, that did it. that is one annoying default |
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| 00:51.48 | ruben23 | hi guys what is the main function of pap2..? is it for anlaog phones being used for VOIP..? |
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| 01:03.06 | itiliti | I know this isnt an a2billing channel, but there is no one in the a2billing channel ever....Anyone have any experience with A2billing? |
| 01:03.16 | itiliti | I am try ing to configure it to use multiple config files. |
| 01:03.46 | itiliti | I have them setup, I just cant figure out where to reference them in the dialplan, and how to associate a customer with it. |
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| 01:18.55 | Mish- | I'm stumped. :( - I had the SPA3000 sending inbound PSTN calls to Asterisk, now when a PSTN call comes in it shows as "Ringing" on the SPA3000, but never goes to Asterisk, no events in Asterisk, not even any UDP packets from the SPA3000, it's definitely an SPA3000 issue, but it all looks correct and I've reconfigured it from scratch, same issue |
| 01:22.40 | jimi_ | get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems |
| 01:22.43 | jimi_ | what does this mean? |
| 01:40.27 | jimi_ | or this Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. |
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| 01:48.36 | nsgn | goodevening. i'm having a frustrating issue where voicemail sent out via email never arrive. what the heck gives? i'm not extremely familiar with sendmail but have been banging at this for a while |
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| 02:03.29 | DiligaF | Does anyone have experience with peering Switchvox with Asterisk? I need some help |
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| 02:07.16 | p3nguin | mish-: Make sure you've set the pstn-to-voip gateway to enabled. |
| 02:08.09 | pabelanger-lap | nsgn: Work on sending an email outside of Asterisk first |
| 02:10.01 | p3nguin | nsgn: Often, default installed sendmail settings are for local relay only. You have to change a few lines in a couple files, run a command or two afterward, and then it should work. |
| 02:24.14 | ChannelZ | death to sendmail |
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| 02:27.24 | WIMPy | Good for what's printed on the package. |
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| 02:53.10 | nsgn | alright, so I can send email from my * box using sendmail at the command line but voicemail email from asterisk never makes it through. what logs can i check to see where this email is failing? it doesnt seem to log in the same spot that command line usage of sendmail does |
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| 03:15.11 | ChannelZ | nsgn: is your mailcmd in voicemail.conf calling 'sendmail', in the right place? |
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| 03:50.15 | Shaaan | Hi, has anyone in here ever been able to successfuly do call broadcasting with Asterisk? |
| 03:50.38 | Kyosh | call broadcasting? |
| 03:50.57 | Kyosh | PA system? |
| 03:51.52 | Shaaan | no Kyosh, i guess your not familiar with Call broadcasting... |
| 03:52.12 | Kyosh | PA system? |
| 03:52.15 | Shaaan | its Basicly where your able to upload a list of numbers of your customers and have asterisk call them all and play a recorded message and if there interested they press 1. |
| 03:52.22 | Shaaan | Kyosh, GOOGLE! |
| 03:52.49 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 03:53.16 | Kyosh | we usually call those "dialers" |
| 03:53.29 | Kyosh | and it's not specific to your "customers" |
| 03:53.50 | Shaaan | Well its called a dialer yes, but its more of a Call broadcasting is the technical term, or Blasting. |
| 03:54.21 | Kyosh | blasting, yes |
| 03:54.37 | Kyosh | autodialer, outbound ivr, call blasting is the act of commencing the dialer |
| 03:55.39 | Shaaan | Yes, do you know how it can be done with asterisk or a similar Application? |
| 03:55.42 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 03:55.43 | Kyosh | read that |
| 03:56.34 | ChannelZ | I call it "fucking obnoxious political calls at 8pm" |
| 03:56.38 | Kyosh | haha |
| 03:56.45 | Kyosh | i get those bitches quite often |
| 03:56.49 | ChannelZ | You know what I really hate, is the ATTENDED autodialers. |
| 03:57.10 | ChannelZ | You know where you pick up the phone and say hello, and no one replies for like 3 seconds because they weren't actually there to begin wtih |
| 03:57.28 | Kyosh | ring ring "hello?" .click click . "hello?" click "hi hello yes hello" . . "buh bye" |
| 03:57.53 | ChannelZ | Today I get one on behalf of my phone company (which I recently dumped) which is just irony. You want to know why I got rid of your service, lady? You do the math. |
| 03:59.33 | Shaaan | so i guess here we are starting a conversation then trying to help :P |
| 03:59.54 | ChannelZ | I have nothing constructive to add. |
| 03:59.59 | ChannelZ | Write a script. |
| 04:00.04 | Kyosh | heh |
| 04:00.07 | ChannelZ | It's not really rocket science |
| 04:00.14 | Kyosh | shaaan, did that page help? |
| 04:00.42 | ChannelZ | Puke a bunch of call files into the spool directory and let Asterisk deal with it. Though there is the pesky business of answering machine detection, etc |
| 04:01.15 | Kyosh | bah leave obscene messages |
| 04:03.01 | Shaaan | Kyosh, yes it helps a bit i guess its more complex then i thought time to find a developer :) |
| 04:03.12 | Kyosh | hmm |
| 04:03.29 | Kyosh | for those i've referred out to that page, few have said that |
| 04:03.55 | Shaaan | well i would have someone do it so its done right the first time around rather then fiddling with it to be honest :) |
| 04:04.20 | Kyosh | i dont think its too hard |
| 04:04.23 | Kyosh | ive never done it |
| 04:04.24 | Kyosh | haha |
| 04:04.45 | Kyosh | it requires making the voice prompts, thats always fun |
| 04:04.56 | ChannelZ | Spend some time to do it and then sell it to others for dozens of dollars |
| 04:05.07 | Kyosh | right, dozens |
| 04:06.39 | boodu | bye |
| 04:09.00 | Shaaan | thats where the developer comes in :0 |
| 04:09.10 | Shaaan | why do it yourself when you can pay someone to do it for you and take the headache |
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| 04:30.25 | Alton35 | they sure have some stupid shows on tv |
| 04:30.45 | Alton35 | syfy channel: fact or faked, paranormal files |
| 04:31.17 | ChannelZ | Dancing With The Stars.. The Bachelor/ette |
| 04:31.33 | ChannelZ | don't get me started |
| 04:31.51 | Alton35 | hah |
| 04:35.10 | ChannelZ | *anything* on MTV or E! (except The Soup) |
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| 04:39.36 | nsgn | ChannelZ, thanks for your reply. my voicemail.conf doesn't have a mailcmd line..? |
| 04:40.10 | nsgn | is this something with my version or can i declare the line and override the problem default? |
| 04:40.21 | nsgn | because i fiddled at the command line and got a sendmail string that works |
| 04:40.29 | nsgn | i just couldnt get * to play with it |
| 04:41.27 | ChannelZ | default is just '/usr/sbin/sendmail -t |
| 04:41.34 | ChannelZ | it should be in the example |
| 04:42.20 | ChannelZ | mailcmd=/usr/sbin/sendmail -t |
| 04:44.13 | nsgn | ChannelZ, isn't in the example file for voicemail.conf either |
| 04:44.29 | nsgn | but can i go ahead and throw it in voicemail.conf? i'm gonna try |
| 04:45.40 | ChannelZ | yes, and yes it is, what version of asterisk are you using |
| 04:46.15 | nsgn | Asterisk 1.4.24 |
| 04:47.46 | ChannelZ | well it's in my 1.4 supplied sample configs.. |
| 04:48.31 | nsgn | weird. just left myself a VM with the mailcmd set and we'll see if anything happens |
| 04:48.56 | nsgn | ! lo and behold |
| 04:49.03 | nsgn | first VM email i've gotten this * box to send me |
| 04:49.09 | nsgn | lemme make sure the attachment plays |
| 04:49.42 | nsgn | beautiful! it plays |
| 04:50.18 | nsgn | ChannelZ, thank you for steering me toward that command. god knows why it's non existant on my system but being able to set one manual flag on sendmail that my SMTP server requires is what did the trick |
| 04:51.22 | nsgn | and bonus: it plays on the iphone with no hesitation. this rocks |
| 04:51.49 | ChannelZ | crap your pants! |
| 04:52.04 | nsgn | out of curiosity..if i specify two email addresses in voicemail.conf for that extension will it address to both? |
| 04:54.54 | ChannelZ | I don't think you can specify multiple addresses |
| 04:55.03 | ChannelZ | probably have to do that with a sendmail alias or something instead |
| 04:55.55 | ChannelZ | needs a shower and to watch tonight's Futurama |
| 04:55.57 | nsgn | yeah, alias was what i was just workin on. something like cheese@localhost : email1@domain.com, email2@domain.com |
| 04:56.07 | nsgn | should be as simple as that, i'd think |
| 04:56.20 | nsgn | but i'm seriously rough on sendmail so we'll see how i can screw it up |
| 05:11.00 | nsgn | hey, that works. i'll call it a night while things are working ;) |
| 05:11.16 | nsgn | night! thanks again channelz |
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| 08:33.56 | Alton35 | haha, go to www.keitholbermann.com |
| 08:36.32 | gr0mit | Faustov, ping |
| 08:36.41 | Faustov | pong |
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| 09:27.22 | FILLVAIO3 | Hi guys. |
| 09:28.13 | iamy_china | FILLVAIO3: Hi |
| 09:28.38 | FILLVAIO3 | Is there possible to send vocemails only in email, and do not collect into voceimail dir? |
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| 10:04.37 | Brismark | hi |
| 10:04.40 | Brismark | anyone alive? |
| 10:05.06 | Brismark | can some one help me with streamplayer? |
| 10:05.30 | Brismark | russellb .. r u there? |
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| 10:33.32 | Brismark | ? |
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| 11:09.32 | Mish- | I'm getting nowhere with my SPA3000 inbound, it was working, now not. I'm starting to wonder if I have an intermittant issue caused by a bad config. Does Asterisk need to be told if I want it to listen on multiple interfaces? |
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| 12:18.11 | phretor | hello, is there any non-official list of supported FXO cards (to be used in EU)? http://www.asterisk.org/hardware are not easy to get. |
| 12:21.46 | [TK]D-Fender | phretor: http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 12:21.56 | [TK]D-Fender | phretor: What are your precise needs? |
| 12:23.04 | phretor | [TK]D-Fender: I'm preparing my first Asterisk box and it has to be able to receive calls from either the VoIP network and PSTN |
| 12:23.39 | phretor | I've been told to use Sangoma USB FXO cards but they are not easy to find here in EU. So I thought of using something equivalent. |
| 12:24.37 | [TK]D-Fender | phretor: I wouldn't look for the USB one first.... I'd go for a card. And please specify what kin of line you're using for PSTN <- |
| 12:25.00 | phretor | [TK]D-Fender: mh, all I know is that they are PSTN |
| 12:25.28 | [TK]D-Fender | phretor: How do you not know the kind of line you have? |
| 12:26.29 | [TK]D-Fender | phretor: Is it indeed an ANALOG line? (not BRI, etc)? |
| 12:26.32 | russellb | is it a straight line or a curvy one? |
| 12:26.40 | WIMPy | didn't even know there were different kinds so far, but then I'm not a historian :-) |
| 12:26.53 | phretor | [TK]D-Fender: this is a huge building and I would have to figure out who is charge of the telephone system. In any case, which details I need to know specifically? I am pretty sure it's analog |
| 12:27.13 | [TK]D-Fender | phretor: that s what you have to be completely sure of. |
| 12:27.31 | phretor | [TK]D-Fender: ok, so first analog vs. digital. |
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| 12:28.07 | [TK]D-Fender | phretor: if its analog then you need an analog FXO interface. A lot of EU uses ISDN-BRI for smaller usage as larger PBXs use E1 (typically ISDN-PRI signalling) |
| 12:28.32 | phretor | ok, thanks. |
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| 12:55.56 | nonlinearly8 | Hi to everybody... |
| 12:56.13 | markfeatherston_ | Howdy |
| 12:56.37 | nonlinearly8 | I have a problem with the zaptel drivers installation |
| 12:57.32 | markfeatherston_ | can you elaborate? |
| 12:58.51 | nonlinearly8 | when i type make it responses that: You do not appear to have the sources for the 2.6.18-128.el5 kernel installed |
| 12:59.09 | markfeatherston_ | what distribution is this? |
| 12:59.25 | markfeatherston_ | it needs the headers package for whatever distribution's kernel you are running |
| 13:00.02 | nonlinearly8 | I have centos 5.3 with kernel 2.6.18-194.8.1.el5 |
| 13:01.00 | drmessano | yum install kernel-devel |
| 13:01.47 | nonlinearly8 | I have already installed kernel-devel |
| 13:02.16 | drmessano | Does it match the output of uname -r |
| 13:02.23 | nonlinearly8 | no |
| 13:02.56 | markfeatherston_ | did you build your own kernel? |
| 13:02.58 | drmessano | Sounds like you need to reboot |
| 13:03.27 | nonlinearly8 | uname -r output No I did not built my own kernel |
| 13:03.33 | drmessano | You're probably behind on a kernel update. Reboot and your kernel will match the installed headers |
| 13:03.42 | drmessano | Then you can install |
| 13:03.48 | nonlinearly8 | sorry I mean No I did not built my own kernel |
| 13:03.53 | drmessano | ^^^^ |
| 13:04.03 | markfeatherston_ | nonlinearly8: heh, we get that, but you need to reboot |
| 13:04.22 | drmessano | Reboot and run make again. it will work |
| 13:04.23 | nonlinearly8 | why |
| 13:04.28 | markfeatherston_ | centos will push out updates occasionally, and there must have been a kernel releas you haven't updated to yet |
| 13:04.30 | drmessano | I explained that |
| 13:04.35 | markfeatherston_ | yea |
| 13:04.37 | *** join/#asterisk mort_gib (~mjensen@adsl-2-234.gibnet.gi) |
| 13:04.39 | drmessano | [09:03] <drmessano> You're probably behind on a kernel update. Reboot and your kernel will match the installed headers |
| 13:04.43 | drmessano | ^^^^ |
| 13:05.08 | nonlinearly8 | I will try I will come back... |
| 13:05.11 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:05.16 | mort_gib | I need some info on recording calls using MixMonitor |
| 13:05.33 | mort_gib | I have issues when calls are transferred |
| 13:06.08 | mort_gib | Does anyone have any input?? |
| 13:08.03 | drmessano | nonlinearly8: Unless you are planning to reboot, you should run yum update with --exclude=kernel, otherwise your -devel package is going to match the kernel you just installed (but are not running) and anything that requires those to match (like zaptel and dahdi) will fail until you've rebooted |
| 13:09.45 | markfeatherston_ | mort_gib: issues could mean a lot of things. you have to explain this better or nobody can help you |
| 13:10.37 | markfeatherston_ | mort_gib: is it disconnecting when it's transferred? |
| 13:11.42 | Faustov | can i send register() to a certain context or does it always have to be an extension? |
| 13:12.20 | mort_gib | markfeatherston_: OK, if an incoming call is transferred I can get the entire call recorded using AUDIOHOOK_INHERIT |
| 13:12.59 | mort_gib | markfeatherston_: But if the call is made from, say a receptionist and later passed on to another phone internally, the recording stops |
| 13:13.13 | Faustov | I could of course use extension with a goto to the right context - but I guess that's a workaround |
| 13:15.18 | markfeatherston_ | mort_gib: https://issues.asterisk.org/view.php?id=7717 It seems this was an attemped fix, but this may have not been corrected |
| 13:15.51 | nonlinearly8 | I come back... |
| 13:16.51 | nonlinearly8 | I have to correct something that I said... |
| 13:17.00 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:17.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:17.13 | Katty | morning |
| 13:17.15 | nonlinearly8 | the uname -r output the same |
| 13:17.46 | nonlinearly8 | but I can not find the source |
| 13:18.30 | nonlinearly8 | It appears a /etc/src/kernels with newer version |
| 13:18.52 | [TK]D-Fender | nonlinearly8: It isn't jsut the source <- You need headers, etc and there is another package or so that can throw off a misleading error like that |
| 13:18.58 | [TK]D-Fender | nonlinearly8: READ ->> http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 13:19.06 | [TK]D-Fender | nonlinearly8: For the list of dependencies |
| 13:19.18 | mort_gib | markfeatherston_ I haven't tried the /n "fix" but I suppose I can try |
| 13:19.49 | *** join/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 13:19.58 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:20.01 | Zeeek | Hay now |
| 13:20.27 | nonlinearly8 | But the book says that must have the source or a link to that in the /etc/src |
| 13:20.38 | [TK]D-Fender | nonlinearly8: ALSO. |
| 13:20.43 | mort_gib | markfeatherston_ Thanks anyway |
| 13:20.48 | [TK]D-Fender | nonlinearly8: there are a lot of other dependencies. Now go read |
| 13:20.50 | nonlinearly8 | ? |
| 13:20.51 | markfeatherston_ | mort_gib: np |
| 13:22.02 | Zeeek | [TK]D-Fender still going strong, cool |
| 13:24.09 | [TK]D-Fender | [09:18]<[TK]D-Fender>nonlinearly8: READ ->> http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 13:24.32 | drmessano | Install KERNEL-DEVEL and it will take care of the needed dependencies |
| 13:24.34 | nonlinearly8 | But my problem has not to do with all the insallation |
| 13:24.39 | Zeeek | what is an E-SBC? |
| 13:24.41 | drmessano | YUM INSTALL KERNEL-DEVEL |
| 13:24.48 | nonlinearly8 | I did |
| 13:24.53 | drmessano | Then REBOOT |
| 13:24.57 | nonlinearly8 | I did |
| 13:25.10 | *** join/#asterisk ttwhy (~tekkno@p4FECF5DE.dip.t-dialin.net) |
| 13:26.31 | [TK]D-Fender | nonlinearly8: Install ALL of the pakcages that page lists |
| 13:27.05 | nonlinearly8 | How can I have the kernel 2.6.18-128 and the source in /etc/src/kernels has a directory to 2.6.18-194? |
| 13:27.53 | [TK]D-Fender | nonlinearly8: Because you have outdated stuff. Go yum update and/or re-install your packages |
| 13:29.20 | nonlinearly8 | I will try... |
| 13:29.23 | nonlinearly8 | thanks |
| 13:30.14 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 13:30.26 | phretor | [TK]D-Fender: apparently on this building everything is VoIP and old PSTN terminals use PSTN-over-VoIP |
| 13:30.54 | [TK]D-Fender | phretor: Then you don't need any kind of interface card for your "lines" |
| 13:31.16 | phretor | [TK]D-Fender: I'd just need an ethernet card and the IP of the VoIP concentrator |
| 13:31.24 | chazzam | nonlinearly8: if you installed an updated kernel and rebooted, your grub config may be booting the wrong kernel |
| 13:31.41 | chazzam | err, and you are still not running the right kernel version |
| 13:31.43 | [TK]D-Fender | phretor: ITSP that is, yes |
| 13:33.53 | *** join/#asterisk ttwhy (~tekkno@p4FECF5DE.dip.t-dialin.net) |
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| 13:44.21 | ickmund | I'm having some problems with Asterisk 1.6.9 not receiving an ACK to finalize an initial invite and call is cut after 20 seconds of re-sending the 200 OK. I notice that Asterisk is not adding a record-route with its IP, which as I understand things it should. Is this a bug, a configuration issue or a misunderstanding on my part? |
| 13:46.01 | [TK]D-Fender | ickmund: that isn't a real * version, and I'm not seeing a pastebin of the SIP DEBUG to trace |
| 13:47.47 | ickmund | If I re-phrase things as: In theory, should * always add itself to record-route in the 200 OK? |
| 13:48.49 | [TK]D-Fender | ickmund: Not sure. * isn't a SIP router so I'm not sure that'd be the behaviour you'd see |
| 13:52.17 | *** join/#asterisk lost_soul (shackett@devio.us) |
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| 13:55.41 | Zeeek | Session Border Controllers and their relation to SIP in about 2 hours on VUC - http://vuc.Me and #vuc right here on Freenode |
| 13:58.56 | Naikrovek | is thinking of setting up a TF2 server at work. |
| 13:58.58 | Naikrovek | hrmm. |
| 13:59.27 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:00.20 | [TK]D-Fender | Naikrovek: I miss the old office FPS lunch times we used to have. |
| 14:00.55 | Naikrovek | yeah |
| 14:00.57 | Naikrovek | same here |
| 14:01.19 | Naikrovek | when i worked at Verio part of the interview process was Quake II deathmatch skills |
| 14:01.25 | Naikrovek | we wound up playing tons of starcraft though |
| 14:01.50 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 14:01.54 | drmessano | I used to play Command and Conquer with a former General Manager |
| 14:02.15 | drmessano | Made it a necessity to keep the network running up to par |
| 14:02.26 | Naikrovek | yeah |
| 14:02.51 | Naikrovek | when quake 1 came out we'd have deathmatches (we were an ISP so there were always people coming in to play) and we'd MURDER the network. |
| 14:02.58 | Naikrovek | that game used all available bandwidth |
| 14:03.03 | Naikrovek | all available |
| 14:03.09 | Naikrovek | even things like DNS lookups would fail |
| 14:03.26 | drmessano | nice |
| 14:03.36 | Naikrovek | yeah |
| 14:03.38 | Naikrovek | it was |
| 14:03.40 | Naikrovek | :) |
| 14:04.06 | phretor | what's the difference between Asterisk and AsteriskNOW? Just user friendly-ness? |
| 14:04.26 | Faustov | yes, asterisk is more user friendly |
| 14:04.29 | Naikrovek | asterisk is software that must be installed on a linux OS |
| 14:04.39 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 14:04.46 | Naikrovek | AsteriskNOW is a linux distribution with asterisk included, and a web GUI as well |
| 14:05.49 | SaiSoma | hey guys, I have a digium TDM800P hooked to analog 1fbs and I'm getting serious . .I hesitate to call it echo, more like strong sidetone. Only on my side, the remote end works fine. any ideas? |
| 14:06.31 | SaiSoma | 1.6.2.7 asterisk, dahdi 2.3.0 |
| 14:06.43 | chazzam | go to 2.3.0.1 first then |
| 14:06.53 | chazzam | there were issues in 2.3.0 with analog cards |
| 14:06.56 | SaiSoma | *nod* |
| 14:06.58 | SaiSoma | thanks |
| 14:07.00 | SaiSoma | doing so now |
| 14:09.06 | phretor | ah I See |
| 14:10.09 | *** join/#asterisk MrJones (~jonas@p5B13DC20.dip.t-dialin.net) |
| 14:10.09 | *** join/#asterisk wpbrown (~wpbrown@wh-gtw-0001.woolfharris.com) |
| 14:10.39 | MrJones | how exactly would I do a phone extension to my capi calls coming from a single number so I can access multiple, different phones through it? |
| 14:11.01 | MrJones | some sort of tutorial or so would be nice, the ones I found deal with administration GUIs and other things to configure this |
| 14:11.04 | wpbrown | Do you guys recommend any call center reporting packages that work well with Asterisk? |
| 14:11.32 | MrJones | and simply adding the extension to the incoming exten number command doesn't work since the additional digits don't get passed on it seems |
| 14:11.48 | MrJones | (it gets immediately routed to the default number without any additional digits I typed into my phone) |
| 14:11.57 | nonlinearly8 | thanks to everybody specially [TK]D-Fender...I updated and ok... |
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| 14:13.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.14.237) |
| 14:17.07 | SaiSoma | chazzam: updated. still sidetone/echo locally |
| 14:17.21 | chazzam | hmm, and you've run fxotune ? |
| 14:17.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:18.17 | SaiSoma | nope. all new to the analog/asterisk thing. typically have used sip trunks before. i'll find docs on it. I don't mean to be ignorant, just point me in the right direction and I'll work to correct it:). |
| 14:18.31 | chazzam | heh, aiigt |
| 14:18.47 | chazzam | if the system had zaptel before, then make sure to run the dahdi one from /usr/sbin/fxotune |
| 14:19.01 | SaiSoma | nah, new build. dahdi only, but thanks |
| 14:19.06 | MrJones | so how can I make asterisk record additional dialed numbers before immediately diving into the incoming context and pick the exten line there? |
| 14:19.23 | MrJones | to make up longer numbers than the original base number of my capi device |
| 14:19.26 | chazzam | something like <stop asterisk> fxotune -i 4; fxotune -s; <start asterisk> |
| 14:19.53 | chazzam | is generally enough for most people |
| 14:20.11 | chazzam | can take a while depending on how many fxo ports you have though |
| 14:20.29 | SaiSoma | *nod*. only 8. it's running now |
| 14:20.39 | SaiSoma | low usage lines, so it's not a big deal |
| 14:20.55 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:21.55 | chazzam | kay, it'll run through all the ones you don't have up to 255, so don't get scared |
| 14:21.58 | *** part/#asterisk minaguib (~mina@modemcable109.56-20-96.mc.videotron.ca) |
| 14:22.00 | chazzam | =p |
| 14:22.19 | chazzam | but it'll fly through those |
| 14:22.50 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:24.32 | Naikrovek | hmm TF2 server is like 3GB to download. that will choke the office T1 for some time |
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| 14:32.00 | SaiSoma | chazzam: ahh .the wonders of fxotune. not perfect, but the sidetone is more than manageable now. thanks! |
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| 14:32.51 | chazzam | heh, yay! |
| 14:32.58 | chazzam | it might be further tweakable |
| 14:33.09 | chazzam | what got written out to /etc/fxotune.conf? |
| 14:33.39 | chazzam | they may all be the same, so an example of a line would suffice in that case |
| 14:33.48 | SaiSoma | 1=13,0,0,0,0,0,0,0,0 |
| 14:33.50 | SaiSoma | except 8 |
| 14:33.54 | SaiSoma | which is all 0s |
| 14:34.05 | SaiSoma | it had almost no "echo" before" |
| 14:35.06 | SaiSoma | bah. spoke too soon. echo back. |
| 14:35.18 | chazzam | hmm, yeah all 0s. if you run fxotune again with -v -p, it will print out the parameters it is going through, and you can pick one of the ones from the top five list made by -p |
| 14:35.23 | SaiSoma | except on 8 |
| 14:35.24 | SaiSoma | rgr |
| 14:35.26 | chazzam | then just fill in those values |
| 14:35.30 | chazzam | in the file |
| 14:35.36 | chazzam | removing spaces and stuff |
| 14:35.48 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:35.58 | chazzam | and what is the number of taps shown in the cli when you run 'dahdi show channel 1' ? |
| 14:36.03 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 14:36.11 | SaiSoma | 1 tap |
| 14:36.30 | chazzam | that means echo cancellation isn't enabled |
| 14:36.49 | chazzam | try turning it on and see if it changes at all? |
| 14:36.56 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:37.14 | chazzam | it may not, but why not try kicking it on and off eh? |
| 14:38.40 | SaiSoma | mmm. i must be misunderstanding. in /etc/dahdi/system.conf i have echocanceller=mg2,1-8 |
| 14:38.59 | SaiSoma | that doesn't do it then. something have to go in /etc/asterisk/chan_dahdi.conf too maybe? |
| 14:39.14 | chazzam | yup |
| 14:39.29 | chazzam | echocancel={yes,128,256,512,1024} type stuff |
| 14:39.34 | chazzam | but yes==128 |
| 14:39.44 | SaiSoma | *nod* reading now |
| 14:39.58 | chazzam | and for hardware echocan, any (positive) number == yes |
| 14:40.02 | chazzam | I believe |
| 14:41.17 | *** part/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 14:41.31 | SaiSoma | testing now. channel 1 has 128 taps |
| 14:41.36 | chazzam | kk |
| 14:42.02 | SaiSoma | still echoing. running the fxotune -v -p |
| 14:42.09 | *** part/#asterisk rafael-ec (~rafael@200.110.234.162) |
| 14:44.18 | SaiSoma | as a side note, a butt set on the lines directly gets 0 echo/sidetone problems. probably doesn't matter, but just an FYI |
| 14:48.22 | chazzam | hmm, ok. does this card have the purple vpmadt032 module attached? |
| 14:52.24 | SaiSoma | mm. i don't know. it's in the server in a rack |
| 14:52.41 | SaiSoma | is that the hardware echo cancellor? |
| 14:53.19 | [TK]D-Fender | yes |
| 14:54.15 | SaiSoma | any way to tell without physically looking? |
| 14:54.16 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:54.19 | wcselby | o/ |
| 14:55.14 | *** join/#asterisk Slator (~Slator@80.71.13.1) |
| 14:55.27 | chazzam | dahdi_scan or dmesg or cat /proc/dahdi/1 might tell you |
| 14:56.16 | Slator | afternoon |
| 14:56.19 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:56.30 | chazzam | what values did you pick for fxotune from the -p? |
| 14:57.33 | SaiSoma | #3 on here: http://pastebin.com/Mt3N1yHa |
| 14:57.43 | SaiSoma | only using one port at a time for testing |
| 14:58.21 | SaiSoma | to make those setttings take effect, i edit /etc/fxotune.conf, stop asterisk, restart dahdi, start asterisk, right? am i missing anything? |
| 15:00.36 | Slator | I'm trying to set up a way of logging some or all users out from their devices at the end of the day. |
| 15:00.57 | Slator | I've written a macro for this, but would ideally find a way of doing it from the CLI |
| 15:01.28 | SaiSoma | ok, this is interesting |
| 15:01.46 | SaiSoma | http://pastebin.com/GURVv1WD it appears that hardware ec is on port 8 (which sounds good) but not on the others. |
| 15:01.55 | SaiSoma | somehow my config must be messed up? |
| 15:02.03 | [TK]D-Fender | Slator: Please elaborate on this "logging out".... |
| 15:02.23 | Slator | system is in device and user mode. Users log in to a device in the morning using *11 |
| 15:02.33 | Slator | then should log out at the end of the day with *12 |
| 15:02.39 | Slator | but often forget |
| 15:03.23 | Slator | I already have a system to clear all the call users from the queues but there doesn't seem to be a command to manipulate or even show logins at the CLI |
| 15:03.32 | MrJones | hmm. I dare toask again, hoping someone has an idea (and it isn't a too newbish question): when I get a call from my CAPI device, asterisk goes to the respective context and fetches the "exten" line with that number. that's fine, but if I dial additional numbers asterisk will still fetch the "exten" line with the original number, not one of the additional numbers |
| 15:03.49 | MrJones | so how can I make "subnumbers" then and have asterisk wait for those additional numbers and get the respective exten line with those numbers being appended? |
| 15:04.30 | [TK]D-Fender | Slator: Go look at the dialplan code that uses and make your own dummy form and dump a bunch of call-files or AMI/CLI Originates pointed to that dialplan in a scripted manner |
| 15:05.25 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 15:05.53 | [TK]D-Fender | MrJones: Make an exten for that part of the pattern * grabs in the manner you consider "premature", and run a silent IVR to collect the remaining digits and include the timeout. |
| 15:06.34 | Slator | I knocked up the macro earlier, but I need a way to do this from the asterisk CLI. |
| 15:06.47 | Slator | Wouldn't originate need to have someone pick up a phone |
| 15:07.00 | [TK]D-Fender | Slator: Told you already : call files, or AMI/CLI Originate <- |
| 15:07.07 | [TK]D-Fender | Slator: No. |
| 15:14.41 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:15.36 | Slator | TK: What do you suggest that I use for the other channel in the originate command? |
| 15:16.13 | [TK]D-Fender | Slator: chan_local naturally...... |
| 15:16.15 | Slator | or should I define another macro to just tit there for a bit then hang up? |
| 15:16.28 | [TK]D-Fender | Slator: Congratulations, you seem to be catching on... |
| 15:16.36 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 15:16.37 | Slator | s/tit/sit |
| 15:17.20 | *** join/#asterisk coppice (~chatzilla@210.17.200.188) |
| 15:17.44 | MrJones | [TK]D-Fender: do you know a good tutorial for that? or an example config how this would look like - IVR seems to be a wide field to learn |
| 15:18.17 | [TK]D-Fender | MrJones: It isn't. Its about 6 lines of dilaplan for your complete solution |
| 15:18.35 | [TK]D-Fender | MrJones: WIKI / Book time. |
| 15:21.20 | SaiSoma | chazzam: bah! I think I found it. line 8, no echo. move physical wiring on telco demarc (swap 7/8), echo moves to 8, 7 clear. change back and then reverse 7/8 at the card, same result, so it's not the port, it's the wiring |
| 15:21.22 | chazzam | SaiSoma: hmm, I don't think hwec can be turned on/off for particular ports on a card |
| 15:21.35 | chazzam | ahh |
| 15:21.51 | SaiSoma | chazzam: yea, sorry on that one, the port was active when i looked. that indicator appears on active ports so it seems |
| 15:21.55 | chazzam | heh, but to ask an earlier question, fxotune -s is what applies the parameters from /etc/fxotune.conf |
| 15:22.00 | chazzam | yup |
| 15:22.07 | SaiSoma | chazzam: *nod* and thanks |
| 15:22.13 | Slator | saisoma: waterlogged pair maybe. Try yelling at BT |
| 15:22.29 | MrJones | [TK]D-Fender: is there a specific name for the directive that will fetch additional numbers for me? so I can jump to the right section in the docs |
| 15:22.32 | SaiSoma | Slator: nah. just running across server room from the physical card to the demarc |
| 15:22.52 | Slator | if it's that short a run, I'd just replace the cable |
| 15:23.42 | [TK]D-Fender | mrWaitExten |
| 15:23.48 | [TK]D-Fender | MrJones: WaitExten |
| 15:24.05 | MrJones | thanks :) |
| 15:24.14 | Kobaz | every time i see the name MrJones i keep thinking of the counting crows song |
| 15:25.01 | SaiSoma | Slator: yup. omw. bbiab |
| 15:25.23 | [TK]D-Fender | Kobaz: I play it... |
| 15:25.51 | [TK]D-Fender | Kobaz: And tonight is another paying gig. |
| 15:26.32 | Kobaz | oh, nice |
| 15:26.48 | Kobaz | i've been playing guitar for about a year |
| 15:26.57 | Kobaz | haven't gotten into singing yet |
| 15:28.00 | chazzam | SaiSoma: you're welcome! glad you got it figured out |
| 15:29.34 | *** join/#asterisk Entulho (~foo@201.67.212.124) |
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| 15:38.24 | xheliox | dammit to hell, now I'm singing that friggin song. |
| 15:39.03 | Kobaz | haha |
| 15:42.01 | Naikrovek | :) |
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| 15:47.25 | DogBoy | what song? "stairway to gilligan's island"? |
| 15:49.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:49.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:50.15 | coppice | stairway to devon |
| 15:51.34 | telnettech | `Mr Jones and me...wanna be a big staaaaaarrr' |
| 15:53.44 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 15:55.21 | *** join/#asterisk MrJones (~jonas@p5B13DD6B.dip.t-dialin.net) |
| 15:58.28 | MrJones | hi. I tried using waitexten now and it looks like this: http://pastebin.com/J6WRfR8Y expected result: I dial xxx 914880102 and get immediately into the 02 extension. actual result: the call is still handled by the default extension and as soon as asterisk is taking/answering the call, I need to enter 02 manually and additionally |
| 15:58.43 | MrJones | so I cannot simply append it to the original phone number when doing the initial call which was what I intended to achieve |
| 15:59.27 | MrJones | someone having an idea what I might do so adding it to the original number works aswell? (given that's possible at all) |
| 15:59.36 | MrJones | if that helps ,it's a capi connection and not some voip thing |
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| 16:29.50 | kuku | Anyone know of an issue with recording calls on transfers ? |
| 16:33.29 | *** join/#asterisk jtodd (r91g3nodnj@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:33.29 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:36.40 | telnettech | Mr Jones: have you tried a GotoIf statement in your current dialplan? |
| 16:37.10 | MrJones | telnettech: I read about Direct Inward Dial (DID) now |
| 16:37.10 | Nugget | telnet is eeeeeeevil! |
| 16:37.22 | MrJones | is it possible that the phone company will simply not pass on those additional digits at the end of the dialed number? |
| 16:37.28 | MrJones | if it's not "activated" |
| 16:37.45 | MrJones | so users will always need to dial them AFTER actually dialing the base number to connect to the asterisk if that's not activated |
| 16:37.56 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:38.20 | MrJones | which is a bit odd and only useful for playing an audio file saying "Welcome to blah. If you want to speak to section a, press 1. IF you want to..." but not for direct dials |
| 16:38.30 | telnettech | so you need to setup an simple IVR that once the called dials the DID number the caller then needs to decide which person they want |
| 16:38.58 | MrJones | it seems I don't get the full DID number but only the truncated base number |
| 16:39.07 | MrJones | WaitForExten() will also just fetch additionally pressed numbers AFTER the actual dial |
| 16:39.13 | MrJones | of which additional digits are simply omitted |
| 16:39.16 | MrJones | no idea how to fetch them |
| 16:39.24 | [TK]D-Fender | MrJones: So whats wrong with the code you've shown? Also you need to ANSWER first |
| 16:39.31 | MrJones | yea I added that for now |
| 16:39.38 | telnettech | that is where a GotoIf statement can come in |
| 16:39.47 | MrJones | [TK]D-Fender: I don't know. nothing maybe? it might be possible the phone company needs to support DID if I get that feature right |
| 16:39.58 | [TK]D-Fender | MrJones: Show us the failed call |
| 16:40.34 | jtodd | poof. |
| 16:40.45 | telnettech | TK .....couldnt he do a GotoIf statement like GotoIf($[${EXTEN}=01]?01,1:) |
| 16:41.02 | MrJones | so I dialed actually 9148801-01 |
| 16:41.08 | MrJones | and that's how it looks like in the asterisk output: |
| 16:41.17 | MrJones | http://pastebin.com/yTySg4rj |
| 16:41.42 | MrJones | it doesn't seem to get that additional -01 of the original dial and WaitForExten() only waits for digits entered manually by the caller after the initial call |
| 16:41.58 | [TK]D-Fender | mrHow long did it actualy wait for the timeout? |
| 16:42.15 | MrJones | 5 seconds, you can hear the silence prompted by Answer() |
| 16:42.21 | MrJones | and if you enter 01 during that time it works |
| 16:42.36 | [TK]D-Fender | [12:40]<telnettech>TK .....couldnt he do a GotoIf statement like GotoIf($[${EXTEN}=01]?01,1:) <--- umm... wow.. caffeine time bro :) |
| 16:42.40 | MrJones | well, 1 actually... switched to one digit since for two I probably need two WaitForExten(), haven't tried that yet |
| 16:42.57 | MrJones | the exten is 0148801 |
| 16:43.02 | MrJones | even if I dialed 91488011 |
| 16:43.05 | telnettech | correct cause the telco is ignoring that...you have told the telco that you want 7 digits for your DID |
| 16:43.12 | [TK]D-Fender | MrJones: use _X and see if you get ANYTHING |
| 16:43.27 | drmessano | You can't pass additional digits until asterisk is in the loop |
| 16:43.39 | telnettech | TK.....whats wrong with my GotoIf statement? :0) |
| 16:43.47 | drmessano | Dialing two extra digits before the call is even switched will just send them to the bitbucket |
| 16:44.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:44.23 | MrJones | drmessano: how is it done then? |
| 16:44.36 | MrJones | do telcos normally fetch those and add them to the original call? |
| 16:44.50 | drmessano | You can't arbitrarily add extra digits to a number |
| 16:45.05 | MrJones | ok |
| 16:45.43 | drmessano | Once Asterisk answers the call, your waitexten can wait for the 01, 02, etc |
| 16:45.55 | drmessano | (no, you dont need a WaitExten for each digit) |
| 16:46.44 | MrJones | ah ok |
| 16:47.03 | MrJones | drmessano: but the user will always need to dial 01 manually? and any appended stuff to the original, dialed number gets thrown off |
| 16:47.16 | drmessano | Yes! |
| 16:47.18 | MrJones | ok! |
| 16:47.25 | MrJones | thanks for making that clear |
| 16:47.25 | telnettech | Mr Jones.....the point is that the Asterisk box has to answer the call and then you can dial the digits for the callers you want.....there is no way around that part |
| 16:47.26 | drmessano | Your lucky what you're dialing is even being passed at ALL |
| 16:47.28 | [TK]D-Fender | telnettech: LOTS |
| 16:48.06 | drmessano | MrJones: I've never seen a case where dialing 55512129999 was passed as 5551212 and the 9999 was ignored |
| 16:48.07 | [TK]D-Fender | telnettech: GotoIf($[${EXTEN}=01]?01,1:) <-- first if the ${EXTEN} is 01 you then jump to.. THE SAME EXTEN. Loop |
| 16:48.40 | MrJones | drmessano: huh? |
| 16:48.46 | drmessano | MrJones: Normally, most telco switched would stab you in the throat and dump the call |
| 16:48.50 | telnettech | that is the exten => that he wanted it to go to if someone pressed 01 |
| 16:48.52 | drmessano | switches |
| 16:49.01 | MrJones | drmessano: that happens here though |
| 16:49.04 | [TK]D-Fender | MrJones: Look at the channel debug for your call. |
| 16:49.10 | [TK]D-Fender | MrJones: See what BRI passes on. |
| 16:49.32 | MrJones | drmessano: but I see, we will simply need to register those subnumbers with the telco then |
| 16:49.40 | drmessano | Ok, great.. It's actually IGNORING the extra digits and completing the call... However, don't expect those extra digits to get passed |
| 16:50.21 | [TK]D-Fender | telnettech: NO DCC |
| 16:50.34 | [TK]D-Fender | telnettech: std /msg |
| 16:51.06 | telnettech | ok sorry |
| 16:52.05 | drmessano | hacker |
| 16:52.57 | [TK]D-Fender | [12:48]<[TK]D-Fender>telnettech: GotoIf($[${EXTEN}=01]?01,1:) <-- first if the ${EXTEN} is 01 you then jump to.. THE SAME EXTEN. Loop |
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| 17:19.15 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 17:20.22 | *** join/#asterisk Teknickal (~nick@rrcs-71-43-16-130.se.biz.rr.com) |
| 17:23.15 | *** join/#asterisk rbd (~rbd@rrcs-98-101-33-14.midsouth.biz.rr.com) |
| 17:23.22 | rbd | anyone have a good, cheap, per minute outbound SIP trunk provider they can recommend? |
| 17:23.36 | p3nguin | ~itsplist-us |
| 17:23.37 | 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 |
| 17:23.52 | p3nguin | also... |
| 17:23.56 | p3nguin | ~trunk |
| 17:23.57 | infobot | methinks trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 17:24.21 | rbd | SIP termination provider, then :) |
| 17:24.30 | rbd | thanks |
| 17:24.30 | p3nguin | ~itsp |
| 17:24.31 | 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. |
| 17:24.40 | p3nguin | ITSP is a better term. |
| 17:24.49 | rbd | ok |
| 17:24.51 | Teknickal | i've had very good success so far with sipstation, but that is unlimited minutes not per minute |
| 17:25.30 | rbd | yeah...I'm using didforsale for inbound...outbound would harly be used, so I was looking for something that didn't cost per month...was just a per minute thing |
| 17:25.50 | p3nguin | If you want metered channels (and you said you do), VoIP.ms will be my recommendation. |
| 17:26.29 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 17:26.31 | Teknickal | I'm kinda new here, so I don't want to seem like a jerk just jumping in with my problems, but hoping someone here can help me debug an asterisk 1.2 system |
| 17:26.41 | Teknickal | I did not build this system so I know very little about it |
| 17:27.06 | Teknickal | looking at the asterisk -r output I see a lot of warnings |
| 17:27.54 | mmlj4 | so pastebin them |
| 17:28.01 | p3nguin | ~pb |
| 17:28.02 | 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. |
| 17:28.47 | Teknickal | I don't want to flood it so I'll just abbreviate stuff |
| 17:28.58 | p3nguin | PASTEBIN |
| 17:28.58 | Teknickal | heres one that comes up a lot, just unsure what it means: |
| 17:29.00 | [TK]D-Fender | Teknickal: PASTEBIN <--------------- |
| 17:29.12 | Teknickal | alright, sheesh |
| 17:29.25 | [TK]D-Fender | Teknickal: And don't be too conservative in what you show us. You'll likely end up cutting off the import parts |
| 17:29.43 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 17:30.21 | Teknickal | http://pastebin.com/ynBD6p8J |
| 17:30.35 | Teknickal | thats just one of many, i'm culling through the logs now to find more |
| 17:30.57 | [TK]D-Fender | Teknickal: *1.2 is full of deadlock issues, and theya re NOT getting fixed at this point. |
| 17:31.04 | [TK]D-Fender | telnettech: What precise version are you on? |
| 17:31.19 | *** part/#asterisk drumkilla (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:31.59 | Teknickal | 1.2.30.4 |
| 17:32.02 | DogBoy | heh, got asterisk 1.6 install on seagate freeagent dockstar |
| 17:32.13 | DogBoy | working |
| 17:32.24 | Teknickal | i know its outdated, but they've got some much junk running on this system its going to be a nightmare to upgrade |
| 17:32.46 | Teknickal | I'm suggesting getting a second box, running it in parallel until we get everything transitioned, and then dumping the current |
| 17:33.02 | [TK]D-Fender | Teknickal: We're on 1.2.40 At a MINIMUM upgrade to the latest in your branch |
| 17:33.04 | p3nguin | You could just upgrade if the hardware is still good. |
| 17:33.11 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:33.25 | Teknickal | I plan on it, but I have no idea what it will break |
| 17:33.38 | Teknickal | they are using some kind of autodialer but I dont know anything about it |
| 17:34.05 | [TK]D-Fender | Teknickal: Fear of the random unknown is SAID. You have problems NOW. If your unjustified fear worth more than solving the problem you are showing us? |
| 17:34.12 | Teknickal | its a mess...ive done plenty of new installs, but don't have much experience fixing asterisk because my boxes always work so beautifully |
| 17:34.38 | [TK]D-Fender | SAD* |
| 17:34.56 | Teknickal | i know what you're saying fender, trust me, but I only started working on this an hour ago |
| 17:35.07 | Teknickal | I'm not in a rush |
| 17:36.39 | [TK]D-Fender | Teknickal: Well 1.2 as a branch has all sorts of issues in core design. these things happen. |
| 17:36.52 | [TK]D-Fender | Teknickal: You are currently 4 branches behind |
| 17:37.05 | [TK]D-Fender | Teknickal: And on one that has no more bug-fixes coming. |
| 17:37.13 | Teknickal | Fully agree...I started using Asterisk on 1.4 and any time I see a 1.2 system its always buggy |
| 17:37.45 | Teknickal | stupid question here, but can I upgrade to the latest 1.2 branch without taking down the system? |
| 17:38.16 | Kobaz | no |
| 17:38.22 | p3nguin | You'll have to restart asterisk, but other than that, probably. |
| 17:38.22 | [TK]D-Fender | Teknickal: Of course you have to restart * |
| 17:38.25 | Teknickal | Didn't think so |
| 17:38.37 | p3nguin | Restarting asterisk isn't hard. |
| 17:38.53 | p3nguin | and only takes a few seconds, if there are no calls at the time. |
| 17:39.20 | p3nguin | DId 1.2 have restart gracefully or restart when convenient? |
| 17:40.05 | Teknickal | whats the easiest way to see how many calls are in progress in 1.2 |
| 17:40.16 | Kobaz | does restart when convenient still block all new incoming calls until it restarts? |
| 17:40.26 | Kobaz | Teknickal: show channels |
| 17:40.28 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 17:40.35 | Teknickal | when convenient still takes calls, just restarts when there is no activity |
| 17:40.37 | p3nguin | gracefully does that, when convenient restarts when there are no calls. |
| 17:40.51 | Kobaz | ah okay |
| 17:41.11 | Kobaz | i once used gracefully on a high volume production system, and i'm like... oh shit,, it's blocking calls now |
| 17:41.19 | p3nguin | yep |
| 17:41.21 | Kobaz | took me a little while to figure out how to unlock it |
| 17:41.30 | p3nguin | You can undo it? |
| 17:41.37 | [TK]D-Fender | ONEWAY |
| 17:41.40 | Kobaz | yeah it's like cancel restart, or something like that |
| 17:42.08 | Teknickal | Right now there are about 40 calls in progress...I think I'll wait until tonight |
| 17:42.28 | Kobaz | p3nguin: ever since then i always restart manually |
| 17:42.38 | telnettech | TK....sorry had to take care of something...... using 1.4.22 |
| 17:42.47 | Teknickal | I truly appreciate all the help you guys are offering, btw |
| 17:42.50 | p3nguin | I hate waiting, so I don't like "when convenient" nearly as much. |
| 17:42.50 | Kobaz | p3nguin: "abort shutdown" |
| 17:44.01 | Kobaz | i gotta upgrade this 1.6.0.19 system at some point |
| 17:44.15 | Kobaz | i have to restart it every few days because it leaves open file handles in sip |
| 17:44.18 | [TK]D-Fender | telnettech: Umm... I never asked you :) |
| 17:44.35 | [TK]D-Fender | oop |
| 17:44.36 | [TK]D-Fender | s |
| 17:44.41 | [TK]D-Fender | telnettech: Never NEAMT to :) |
| 17:44.44 | [TK]D-Fender | MEANT |
| 17:45.02 | b14ck | Hey everyone, I've got a very finnicky problem here with a new install. I've been trying to figure it out for ~2 days now, to no avail. |
| 17:45.17 | b14ck | Here's a pastebin which contains the problem description at the top, and all the relevant configuration files: http://pastie.org/private/mtur57ty8v0bzj9kes49g |
| 17:45.27 | *** join/#asterisk Alagar (~Administr@122.164.36.164) |
| 17:45.30 | Kobaz | impressiver |
| 17:45.31 | [TK]D-Fender | telnettech: Your 2 nicks are the same length and same first 2 letters... completely missed that one... |
| 17:45.42 | b14ck | Basically, to summarize: I cannot load chan_sip.so because Asterisk thinks that my sip.conf contains #include directives to files which do not exist. |
| 17:46.06 | [TK]D-Fender | b14ck: Yup, pretty clear error |
| 17:46.14 | *** join/#asterisk m_c_le (~marcello@2001:470:1f0b:d4b:2e0:4dff:fe6c:9372) |
| 17:46.17 | b14ck | [TK]D-Fender, did you check out my pastebin? |
| 17:46.21 | [TK]D-Fender | b14ck: Yes |
| 17:46.22 | b14ck | It's super weird. |
| 17:46.25 | b14ck | What's the problem? |
| 17:46.27 | [TK]D-Fender | b14ck: Nope |
| 17:46.36 | Kobaz | looks pretty obvious to me: ---> /nfs/aserisk/maain/sip/sip_main.conf |
| 17:46.40 | [TK]D-Fender | b14ck: Seriously? |
| 17:46.44 | [TK]D-Fender | b14ck: Are you even reading? |
| 17:46.46 | Kobaz | do you always spell main with two a's? |
| 17:46.50 | [TK]D-Fender | b14ck: [2010-07-16 12:43:40] ERROR[9982]: config.c:1098 process_text_line: The file '/nfs/aserisk/maain/sip/sip_main.conf' was listed as a #include but it does not exist. <--- 2 x "a" |
| 17:46.55 | b14ck | Kobaz, [TK]D-Fender, look down one section |
| 17:47.00 | *** join/#asterisk jpmcallister (~ec06113@200.242.28.231) |
| 17:47.02 | b14ck | In the part where I pastebin my /etc/asterisk/sip.conf |
| 17:47.10 | b14ck | That file does NOT contain the spelling that asterisk is outputting. |
| 17:47.11 | [TK]D-Fender | b14ck: /nfs/asterisk/main/sip/sip_main.conf |
| 17:47.16 | p3nguin | aserisk/maain |
| 17:47.28 | b14ck | My sip.conf says: |
| 17:47.31 | Kobaz | b14ck: something tells me you're not showing us everything |
| 17:47.33 | p3nguin | two probablems. |
| 17:47.35 | b14ck | #include "/nfs/asterisk/main/sip/sip_main.conf" |
| 17:47.41 | b14ck | Asterisk says: |
| 17:47.49 | Kobaz | b14ck: cd /nfs/asterisk; grep maain * -R |
| 17:47.49 | b14ck | he file '/nfs/aserisk/maain/sip/sip_main.conf' was listed as a #include but it does not exist." |
| 17:47.59 | p3nguin | Did you reload the files? |
| 17:48.09 | *** join/#asterisk mbowie (~mbowie@99-7-126-96.lightspeed.simica.sbcglobal.net) |
| 17:48.23 | p3nguin | Oh, you're trying to load the module, nevermind. |
| 17:48.30 | b14ck | Kobaz, http://pastie.org/1047591 |
| 17:48.30 | [TK]D-Fender | b14ck: You aren't looking at the right file in the right place or similar |
| 17:48.39 | Kobaz | b14ck: grep grep grep |
| 17:48.51 | b14ck | [TK]D-Fender, I've grepped every single *conf file in my /nfs/asterisk directory, none of them contain 'maain' |
| 17:48.59 | b14ck | Ok, I'll paste the greps. |
| 17:49.14 | [TK]D-Fender | b14ck: Why do you have sip.conf in /nfs AND in /etc ? |
| 17:49.23 | p3nguin | grep -r aserisk / |
| 17:49.30 | Kobaz | b14ck: and grep /etc/asterisk for maain too |
| 17:49.36 | [TK]D-Fender | http://pastie.org/1047591 <-- this shows configs in /nfs |
| 17:49.40 | b14ck | The files are all /nfs/asterisk/blah, but it is symlinked to /etc/asterisk |
| 17:49.48 | b14ck | So that Asterisk can use the same config files on multiple servers. |
| 17:49.59 | Kobaz | b14ck: sounds like a bad design... but... continue |
| 17:50.16 | [TK]D-Fender | http://pastie.org/private/mtur57ty8v0bzj9kes49g <-- here you are shoing us ***ETC* sip.conf |
| 17:50.37 | b14ck | Here's my grep output: http://pastie.org/1047602 |
| 17:50.49 | b14ck | [TK]D-Fender, /etc/asterisk/sip.conf is a symlink to /nfs/asterisk/sip.conf |
| 17:50.54 | mbowie | Good day folks. I've had someone suggest Asterisk as the solution to a problem and I'm not sure if it's possible based on what I've read. It was basically suggested to use to TA's with Asterisk in the middle to simulate a local extension to a branch PBX. Is that doable and if so, what's the terminology I should be googling? |
| 17:50.56 | Kobaz | b14ck: no |
| 17:51.04 | p3nguin | (1249.23) <p3nguin> grep -r aserisk / |
| 17:51.07 | Kobaz | b14ck: no no no |
| 17:51.07 | [TK]D-Fender | b14ck: I see 2 places and nothing that backs it up |
| 17:51.21 | Kobaz | b14ck: cd /etc/asterisk; grep maain * -R; cd /nfs/asterisk; grep maain * -R |
| 17:51.22 | b14ck | [TK]D-Fender, gotcha, let me do another pastie. |
| 17:51.30 | p3nguin | (1251.03) <p3nguin> (1249.23) <p3nguin> grep -r aserisk / |
| 17:51.39 | Naikrovek | uhoh |
| 17:51.43 | Naikrovek | p3nguin: is recursing again |
| 17:52.01 | pabelanger-lap | mbowie: TA's? |
| 17:52.13 | Kobaz | p3nguin: why would you want to do that? |
| 17:52.33 | b14ck | Kobaz, this will take a few minutes, there's a lot of files in here |
| 17:52.33 | p3nguin | You know what grep does, so you already know why. |
| 17:52.42 | Kobaz | b14ck: don't do anything fancy... just do those |
| 17:52.43 | mbowie | pabelanger-lap: Telephony adapters... although I suppose that's irrelevant really. |
| 17:52.45 | Kobaz | minutes? |
| 17:52.49 | Kobaz | what the hell do you have in there |
| 17:52.55 | b14ck | Kobaz, there's a *lot* of code in there, and |
| 17:53.03 | b14ck | Also soundfiles for version control purposes |
| 17:53.05 | Naikrovek | it's probably something like non-printable characters in the text files. delete the include line and retype it by hand |
| 17:53.08 | b14ck | It's complex. |
| 17:53.15 | b14ck | Naikrovek, already did that. |
| 17:53.16 | b14ck | =/ |
| 17:53.17 | *** part/#asterisk m_c_le (~marcello@2001:470:1f0b:d4b:2e0:4dff:fe6c:9372) |
| 17:53.24 | Naikrovek | b14ck: k |
| 17:53.26 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 17:53.48 | Kobaz | b14ck: obviously you have maain somewhere in your config... and so far you haven't found it by just looking at where you're thinking to look... so you might as well look at everything |
| 17:53.51 | pabelanger-lap | b14ck: try: #include '/nfs/asterisk/main/sip/sip_main.conf' (in your sip.conf) |
| 17:54.06 | pabelanger-lap | b14ck: single quotes |
| 17:54.10 | b14ck | pabelanger, ok |
| 17:54.15 | Kobaz | b14ck: either that, or it's some crazy-weird asterisk bug... but it sounds very much configuration |
| 17:54.29 | pabelanger-lap | Kobaz: yes, it is |
| 17:54.30 | b14ck | Kobaz, I've had a lot of experience setting up / using asterisk, never seen this before. |
| 17:54.38 | b14ck | This happened after an asterisk upgarde |
| 17:54.44 | b14ck | The config files have not been modified. |
| 17:54.45 | b14ck | :x |
| 17:54.55 | b14ck | But I don't doubt it could be a config issue |
| 17:54.58 | b14ck | Which is why I'm asking =p |
| 17:55.12 | Kobaz | just find your maain... if you can't... well... time to start digging through the asterisk code |
| 17:55.13 | Naikrovek | time for some sanity checks |
| 17:55.13 | [TK]D-Fender | it is. Files aren't where you think they are, etc. |
| 17:55.21 | [TK]D-Fender | b14ck: You are overlooking something basic |
| 17:55.36 | [TK]D-Fender | Naikrovek: No... he is quite clearly insane... no need to check ;) |
| 17:55.37 | b14ck | [TK]D-Fender, I'm sure you're right. That's why I'm here. Thanks for all the help so far :) |
| 17:55.42 | Naikrovek | lol |
| 17:55.52 | pabelanger-lap | b14ck: https://issues.asterisk.org/view.php?id=17472 |
| 17:56.10 | b14ck | hey pabelanger if i use single quotes I get a new error messsage oO |
| 17:56.11 | b14ck | I get: [2010-07-16 12:55:45] ERROR[10181]: config.c:1098 process_text_line: The file ''/nfs/asterisk/main/sip/sip_main.conf'' |
| 17:56.16 | b14ck | Which also isn't correct. |
| 17:56.27 | pabelanger-lap | b14ck: upgrade to latest 1.6.2 branch, it should be fixed |
| 17:56.51 | b14ck | pabelanger, I am using the latest. |
| 17:56.54 | Kobaz | The parser appears to be dropping the 7th character and doubling the 16th character: |
| 17:56.57 | Kobaz | haha |
| 17:57.01 | Kobaz | like i said |
| 17:57.06 | pabelanger-lap | b14ck: 1.6.2.9 is not latest |
| 17:57.10 | b14ck | So wait, is this a confirmed bug then? Or am I crazy? |
| 17:57.12 | Kobaz | crazy-weird asterisk bug |
| 17:57.14 | jpmcallister | do not use any quotes |
| 17:57.39 | pabelanger-lap | b14ck: $ svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 |
| 17:57.39 | b14ck | Oh shit |
| 17:57.42 | b14ck | If I remove quotes it works |
| 17:57.44 | b14ck | god damnet |
| 17:57.45 | b14ck | =/ |
| 17:57.47 | Kobaz | b14ck: it was one or the other... configs or crazy-weird bug |
| 17:57.48 | b14ck | I knew I wasn't crazy@! |
| 17:57.50 | Naikrovek | lol |
| 17:57.53 | Naikrovek | why would you quote it |
| 17:57.57 | Naikrovek | i dunno |
| 17:58.00 | b14ck | Because that's the standard syntax. |
| 17:58.05 | Naikrovek | is it |
| 17:58.09 | Kobaz | Naikrovek: because most config syntax allows quotes |
| 17:58.13 | Kobaz | but asterisk is special |
| 17:58.19 | b14ck | Oh wow. |
| 17:58.21 | Naikrovek | short bus special? |
| 17:58.21 | b14ck | It not works. |
| 17:58.28 | b14ck | *now |
| 17:58.29 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:58.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:58.34 | Naikrovek | yay b14ck |
| 17:58.36 | Kobaz | Naikrovek: asterisk handles quotes very poorly in general |
| 17:58.43 | b14ck | [TK]D-Fender, woa that's crazy =p |
| 17:58.51 | Naikrovek | seems like that's something that's been figured out in other bits of software... the quoting |
| 17:59.10 | b14ck | pabelanger, thanks so much! |
| 17:59.22 | b14ck | pabelanger-lap, not sure if I could have gotten that one without your help |
| 17:59.30 | b14ck | Will update to svn release, and try that |
| 17:59.42 | b14ck | I spent an entire 2 days debugging that. |
| 17:59.43 | Naikrovek | if only somenoe could invent something to match patterns in some regular way |
| 17:59.46 | b14ck | I grepped every file in there. |
| 17:59.49 | pabelanger-lap | b14ck: np |
| 17:59.52 | b14ck | :x |
| 18:01.41 | b14ck | Someone should add some unit tests to asterisk. |
| 18:01.54 | b14ck | I bet it would be a huge project though, to unit test the entire asterisk core. |
| 18:01.59 | b14ck | Since there are none (that I'm aware of) atm. |
| 18:02.00 | pabelanger-lap | b14ck: patches welcome |
| 18:02.06 | Naikrovek | if only there were a technology to automate that. ah well. maybe my grandparents will live in such a new-fangled time period |
| 18:02.17 | Naikrovek | man i manged that |
| 18:02.18 | *** join/#asterisk exelnet (~exelnet@i59F7EF04.versanet.de) |
| 18:02.22 | b14ck | Hrm. |
| 18:02.24 | jpmcallister | Anyone linked asterisk with a SIEMENS HIPATH 4000? |
| 18:02.33 | Naikrovek | "linked" |
| 18:02.34 | Naikrovek | ? |
| 18:02.40 | Naikrovek | trunked? |
| 18:02.43 | jpmcallister | yes |
| 18:02.45 | florz | ... in particular you'd have to deal with all the bugs you'd find ... ;-) |
| 18:02.50 | Naikrovek | does the siemens speak SIP |
| 18:03.10 | Naikrovek | florz: this is why there aren't any good C/C++ static analysis tools |
| 18:03.12 | jpmcallister | no, It is trunked through E1 |
| 18:03.22 | florz | Naikrovek: hmm? |
| 18:03.24 | Naikrovek | jpmcallister: CAN the siemens speak SIP |
| 18:03.37 | jpmcallister | Naikrovek: no. it is trunked via E1 |
| 18:04.16 | Naikrovek | data e1 or voice |
| 18:04.32 | exelnet | heya. my asterisk installation with capi isdn trunk is working fine, except a small problem. it doesnt play any ringtone when dialing out. so it stays silent till the other side picks up |
| 18:04.43 | jpmcallister | Naikrovek: it is working except for one facility. When a try to config a "follow me" from an analog extension to an asterisk one I get an error in asterisk: WARNING[3417]: chan_dahdi.c:12784 pri_dchannel: Ring requested on unconfigured channel 0/0 span 1 |
| 18:05.13 | Naikrovek | jpmcallister: you should have started with that. :) someone can help you but it isn't me |
| 18:05.46 | jpmcallister | Naikrovek: tank you anyway :) |
| 18:06.18 | jpmcallister | Naikrovek: my main problem is that siemens is like a black box to me |
| 18:07.59 | pabelanger-lap | exelnet: There is usually no ringback on ISDN |
| 18:08.11 | telnettech | jpm......what version of software do you have on the HIPATH |
| 18:08.23 | pabelanger-lap | exelnet: Your telco maybe able to provide inband ringback, call them |
| 18:08.53 | pabelanger-lap | exelnet: or use 'r' flag in dial command |
| 18:09.01 | exelnet | pabelanger-lap: well it works when i use it with any of my isdn phones... |
| 18:09.23 | jpmcallister | telnettech: I don't really know. Like I said it is a black box to me. I was hopping to discover something on the asterisk side |
| 18:12.08 | exelnet | pabelanger-lap: hmm how can i add this flag? i might have to do this in freepbx. this is my dialstring: CAPI/ISDN1/$OUTNUM$ where should i add the flag there? |
| 18:13.42 | Naikrovek | exelnet: Dial(CAPI/ISDN1/$OUTNUM$,r) I believe |
| 18:14.13 | Naikrovek | but that means it will sound like ringing no matter what. bad phone number? ringing. busy? ringing. out of service? ringing. |
| 18:15.36 | b14ck | Ah, everything is back to normal operations now. |
| 18:15.37 | b14ck | woot |
| 18:15.41 | exelnet | Naikrovek: ok, thats not what i want. i want a similar ringing to the one used by my phone |
| 18:15.48 | Naikrovek | yes, i know |
| 18:15.57 | Naikrovek | but you'll have to speak with your ISDN provider probably |
| 18:16.16 | Naikrovek | normally they don't ring because phones aren't usually put on the ends of ISDN circuits |
| 18:16.26 | exelnet | Naikrovek: well it works with all phones... i doubt its the providers fault |
| 18:16.42 | Naikrovek | well i don't know then |
| 18:16.48 | Naikrovek | if it works with all phones what's the problem |
| 18:16.59 | exelnet | Naikrovek: in germany we use isdn phones directly behind the ntba, which is common. |
| 18:17.25 | exelnet | Naikrovek: well not with the sip ones connected to the asterisk server |
| 18:18.11 | Naikrovek | hmm. are you sure the source of the ring sound is the provider and not the ISDN phone itself |
| 18:18.50 | tzafrir | Naikrovek, what do you need static analysis tools for? |
| 18:19.12 | Naikrovek | tzafrir: i'm spoiled on them in Java, I don't need them for C. I don't know C |
| 18:19.35 | exelnet | Naikrovek: well if its the phone, then it does know a way how to play different sounds for busy free, ... |
| 18:19.40 | tzafrir | Naikrovek, you have a code base and you want to get around? |
| 18:19.54 | tzafrir | Is that code base Asterisk? |
| 18:20.34 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:20.36 | Naikrovek | earlier there was a conversation about a bug in Asterisk that funges up #include lines. I was saying "if only there were automated computing tools to catch bugs like this" knowing that there are lots of them |
| 18:21.01 | Naikrovek | then someone said "C programmers don't use them because they don't want to fix bugs" |
| 18:21.13 | tzafrir | Naikrovek, IIRC there is |
| 18:21.16 | Naikrovek | paraphrasing |
| 18:21.29 | Naikrovek | i know there are, there are lots of static analysis tools |
| 18:21.40 | Naikrovek | and testing suites |
| 18:22.02 | pabelanger-lap | exelnet: either way, call your provider. They will tell you very easily if they are providing ringback or not. |
| 18:22.31 | [TK]D-Fender | BRI should send progress. |
| 18:22.37 | Naikrovek | yes |
| 18:22.43 | Naikrovek | but will it send the progress as audio |
| 18:22.47 | Naikrovek | or as a signal |
| 18:23.00 | *** join/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 18:23.01 | Naikrovek | for something to interpret then play a sound on its own |
| 18:23.52 | *** part/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 18:23.59 | pabelanger-lap | Naikrovek: Usually not, that is the whole purpose of the call progress message. Without a debug from exelnet we cannot see what is happening |
| 18:24.11 | ChannelZ | OT I know but is anyone here running Snow Leopard and Firefox? |
| 18:24.24 | tzafrir | cppcheck looks potentially interesting |
| 18:26.43 | exelnet | pabelanger-lap: after some more googling and reading the readme of the asterisk capi channel plugin i found an option: /bo |
| 18:26.54 | exelnet | does what it should :=) |
| 18:27.03 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 18:29.14 | tzafrir | hmm... here's one report from cppcheck: [main/ast_expr2.c:3517]: (error) Memory leak: vs |
| 18:29.37 | tzafrir | branches/1.6.2@273145 |
| 18:29.49 | tzafrir | fg |
| 18:29.52 | Kobaz | df |
| 18:31.40 | tzafrir | function op_tildetitle() |
| 18:36.55 | *** join/#asterisk yonahw (~user@75.99.93.178) |
| 18:38.32 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 18:40.56 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 18:41.45 | Kobaz | tzafrir: post to the asterisk-dev mailing list with your findings |
| 18:42.12 | ZeXr0 | I have an application that receive a call and capture the digits pressed (it's a device that is sending a code). I'd like to be able to record the digits pressed but in a wav file for debugging purpose. Monitor() doesn't seems to work well with the default DTMF. Is there another way to record what is pressed in an audio file ? |
| 18:42.34 | Naikrovek | sounds like you need in-band DTMF |
| 18:42.34 | yonahw | any suggestions for a door phone setup with a pure voip installation? I've checked out the wiki but can't really decide what my best option is and not sure how dated the information there is. |
| 18:42.53 | Naikrovek | door phone? like "hey can i get in" kind of a thing? |
| 18:43.18 | yonahw | Naikrovek: yes, with capability to open striker |
| 18:43.20 | Naikrovek | pretty easy to find phones that hot-dial a number when picked up |
| 18:43.31 | Naikrovek | well the phone will never be able to open a striker |
| 18:43.43 | Naikrovek | you'll have to have the phone connect to something that can control a striker |
| 18:43.46 | ZeXr0 | Naikrovek : Like before answering the call I should set SIPDtmfMode(inband) ? |
| 18:44.18 | Naikrovek | ZeXr0: i dunno if you can do it per-call. the provider has to know to send the DTMF to you in-band I think |
| 18:44.19 | yonahw | Naikrovek: I don't really mean a phone but rather an intercom system, we currently have one setup with our Toshiba system but it seems to be proprietary |
| 18:44.28 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:44.32 | [sr] | howdy friends |
| 18:44.54 | beardy | Howdy. |
| 18:44.59 | Naikrovek | yonahw: well you can easily have a phone dial a receptionist or something when picked up. receptionist can push button controlling the striker |
| 18:46.05 | Naikrovek | how does it work currently |
| 18:46.17 | Naikrovek | person picks up phone, dials passcode, door unlocks? |
| 18:46.48 | yonahw | No, there is an intercom panel which you would press the button on, extensions ring and then by the press of a button signal is sent to striker |
| 18:46.54 | Naikrovek | ah |
| 18:47.02 | Naikrovek | easily done with voip. |
| 18:47.16 | Naikrovek | easily replicated |
| 18:47.21 | yonahw | there is one device which controls all this called the MDFB Door Phone which in turn connects to a Toshiba controller |
| 18:48.02 | yonahw | I can figure out the dial plan for this but don't know what it would take to control the MDFB which sends the signal to the striker by closing the circuit |
| 18:48.04 | Naikrovek | so does a person push a button on a computer to open the door or is it something else |
| 18:48.16 | yonahw | A button on their phone |
| 18:48.20 | Naikrovek | nice |
| 18:48.27 | Naikrovek | ABCD button column? |
| 18:48.31 | Naikrovek | not enough phones have those |
| 18:48.43 | Naikrovek | oh lol |
| 18:48.50 | Naikrovek | "a button" not "the A button" |
| 18:48.56 | Naikrovek | nevermind me |
| 18:48.58 | yonahw | correct |
| 18:49.17 | yonahw | I was wondering what you meant by the ABCD button column, got you now |
| 18:49.44 | Naikrovek | some older phones (and the DTMF standard) allow an extra column of buttons to the right of the three columns commonly seen |
| 18:49.48 | Naikrovek | A B C D |
| 18:49.59 | Naikrovek | they were originally intended for menus, I believe |
| 18:50.10 | Naikrovek | phones should have a hexadecimal keypad |
| 18:50.17 | Naikrovek | 0-9, A-F |
| 18:50.19 | yonahw | yeah, I do recall having seen one of them back in the day and am familiar with the dtmf standard specifying them |
| 18:50.25 | Naikrovek | okay cool |
| 18:50.49 | Naikrovek | anyway, asterisk can call system applications |
| 18:51.11 | Naikrovek | so if you want, you can have a phone auto dial an extension (which could then ring a group of phones) |
| 18:51.13 | Naikrovek | someone picks up |
| 18:51.32 | Naikrovek | if person is authorized to enter, person who answered pushes a button. this would then launch a system command within linux to do what you needed |
| 18:51.54 | Naikrovek | so now it's a matter of finding a controller that can interface with linux. door striker controllers are not expensive |
| 18:52.10 | yonahw | right what i was asking about is the controller part |
| 18:52.12 | Naikrovek | or you could just go the analog route and put a button on everyone's desk |
| 18:52.29 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 18:52.32 | Naikrovek | pinball machine servo, and a 24v power supply or whatever the striker requires |
| 18:52.43 | Naikrovek | i dunno |
| 18:52.47 | Naikrovek | i'm loopy on medicine today |
| 18:52.51 | Naikrovek | but i'm close to a solution |
| 18:52.56 | yonahw | there seem to be so many options and what I would really like is to use the existing wiring |
| 18:53.13 | Naikrovek | what's the wiring to the door striker look like |
| 18:53.16 | Naikrovek | couple of wires? |
| 18:53.31 | yonahw | yeah there are two pairs one carrying power apparently |
| 18:53.46 | yonahw | they go to the intercom which must in turn be wired to the striker |
| 18:54.14 | Naikrovek | hmm |
| 18:54.47 | yonahw | the controller has a wire to power, a pair for voice, and then another wire to the striker with a wire joining the power to the striker |
| 18:55.02 | Naikrovek | there has to be an asterisk solution for this for sale by someone |
| 18:55.32 | Naikrovek | weird that the controller is in the phone path |
| 18:55.53 | yonahw | that's what I was hoping, I do see a page about it on the wiki but some of the products don't seem to be around anymore and nothing really presents a complete solutions |
| 18:55.55 | Naikrovek | it must just listen for a DTMF tone and when it hears it the door opens |
| 18:56.02 | yonahw | s/solutions/solution |
| 18:56.10 | yonahw | exactly what it does |
| 18:56.32 | Naikrovek | couldn't the caller outside just play the DTMF himself to open the door? |
| 18:56.41 | yonahw | Toshiba tends to do things in the most difficult and esoteric fashion possible |
| 18:56.47 | Naikrovek | phone probably has no buttons but the sound could be recorded |
| 18:57.08 | Naikrovek | and you're getting rid of the phone system this interacts with |
| 18:57.13 | yonahw | precisely, you could probably play it back or just provide the voltage |
| 18:57.53 | yonahw | Phone system is on it's way out the door, along with it's hefty price and lack of features |
| 18:58.01 | ZeXr0 | Naikrovek : Seems like I won't be able to record the digits pressed. If I change the mode, I can't use the Read function. So I'm out of luck on that one. I'll have to rely on the other side of the conversation to find out why it's not working properly. |
| 18:58.03 | Naikrovek | good |
| 18:58.33 | Naikrovek | ZeXr0: if the DTMF is inband you'll be able to record it, but you'll have to set the entire trunk to in-band |
| 18:59.07 | Naikrovek | yonahw: maybe, if the door phone is working well enough as it is, you could leave that bit analog and plug it into asterisk via an FSX card |
| 18:59.42 | yonahw | So I originally was hoping to do that but it seems that this particular one requires the Toshiba door controller to work with |
| 18:59.52 | wpbrown | Do you guys recommend any call center reporting packages that work well with Asterisk? |
| 19:00.01 | yonahw | not really a surprise based on my experience with the rest of the Toshiba system |
| 19:00.25 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 19:00.55 | yonahw | I also would rather use an ATA instead of actually getting an FSX card since we don't have any other analog devices for the system and all our lines are already sip |
| 19:01.56 | *** join/#asterisk emora (~emora@62.83.68.127.dyn.user.ono.com) |
| 19:02.59 | Naikrovek | I would think that an ATA would work just as well |
| 19:03.21 | Naikrovek | just a matter of interfacing that analog line for that one phone with your voip system |
| 19:03.35 | Naikrovek | or |
| 19:03.43 | Naikrovek | this might not be a bad idea actually |
| 19:03.48 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 19:03.56 | Naikrovek | find an EE major online or at the local college |
| 19:04.16 | Naikrovek | ask them to design/build you a controller that integrates perfectly with your asterisk system, and uses existing wiring |
| 19:04.24 | mbowie | Good day folks. I've had someone suggest Asterisk as the solution to a problem and I'm not sure if it's possible based on what I've read. It was basically suggested to use to ATA's with Asterisk in the middle to simulate a local extension to a branch PBX. Is that doable and if so, what's the terminology I should be googling? |
| 19:04.34 | yonahw | hmm I never thought of that but it might actually be a nice route to go |
| 19:04.53 | yonahw | brb |
| 19:05.33 | *** join/#asterisk exelnet (~exelnet@i59F7EF04.versanet.de) |
| 19:06.06 | Naikrovek | mbowie: yes it's possible. you want asterisk to be a PBX for a bunch of analog phones |
| 19:07.06 | mbowie | Naikrovek: In a sense. The idea is to maintain the legacy PBX (for "company" reasons) but add SIP functionality via ATA's connected to extensions. |
| 19:07.53 | exelnet | hmm still got a problem. my sip clients should always use the capi trunk, so i set a dial rule: "." but how can i handle outgoing calls in the +49 format? e.g.: 01234567 works, +491234567 not. |
| 19:08.13 | Naikrovek | exelnet: +49|. |
| 19:08.24 | Naikrovek | hmm that may be a FreePBX thing though |
| 19:09.29 | exelnet | Naikrovek: well asterisk initiates the call, but i get the busy tone. |
| 19:09.30 | Naikrovek | mbowie: set up an Asterisk system. Trunk it to the legacy system someone is in love with. connect your voip phones to asterisk, leave the legacy phones on the legacy system |
| 19:10.07 | exelnet | bah laptop is dying... need to restart again... brb |
| 19:10.12 | Naikrovek | exelnet: you want to strip the +49, right |
| 19:10.22 | exelnet | Naikrovek: yeah just a second |
| 19:10.29 | Naikrovek | k |
| 19:12.02 | mbowie | Naikrovek: That's pretty much what's been done. What we're looking for now is for those extensions to be usable via SIP... so if someone on the legacy system dials 2002 (which one of the ATA's is connected to) the call goes to the SIP extension. By the same token, calls from the SIP extension presumably need to have the DTMF kicked down to the legacy PBX. |
| 19:12.20 | Naikrovek | yeah that's doable |
| 19:12.34 | mbowie | (I may well be missing the elementary... if that's the case, don't hesitate to kick me in the teeth. ;-) ) |
| 19:12.45 | Naikrovek | it's just going to be a setting in Asterisk and the legacy PBX to direct calls to the appropriate extensions to the appropriate place |
| 19:13.16 | mbowie | Is there a term I should be googling for this, or is it really just a matter of getting my head around it? |
| 19:13.27 | *** join/#asterisk demiv (~demiv___@190.144.133.98) |
| 19:13.41 | Naikrovek | so on Asterisk, you'd do something like: exten _4XXX,1,Dial(SIP/trunk_to_legacy/${EXTEN}) |
| 19:13.53 | Naikrovek | if your legacy extensions are 4000-4999 |
| 19:14.17 | Naikrovek | the setup on the legacy side will depend entirely on the product manual |
| 19:14.25 | Naikrovek | i can't tell you what to do there |
| 19:14.34 | *** join/#asterisk exelnet (~exelnet@i59F7EF04.versanet.de) |
| 19:14.40 | exelnet | re... |
| 19:15.19 | mbowie | Ok.. I think you've described it in a way that makes more sense to me. I'll run with that and see what I come up with. Sincerest thanks for taking the time (and keystrokes) to humor me. ;-) |
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| 19:16.38 | Naikrovek | exelnet: i think the Asterisk dial command may be something like exten => _49XXXXXXXX,1,Dial(SIP/whatever/${EXTEN:2}) |
| 19:16.42 | Naikrovek | to strip the 49 off |
| 19:16.58 | Naikrovek | the _ means to pattern match |
| 19:17.02 | Naikrovek | mbowie: glad i could help |
| 19:18.27 | exelnet | Naikrovek: well the problem is: its +491234567 or 01234567 |
| 19:18.51 | Naikrovek | okay |
| 19:19.39 | Naikrovek | exten => _49XXXXXXX,1,Dial(SIP/whatever/0${EXTEN:2}) |
| 19:19.46 | Naikrovek | that strips the 49 and adds a 0 |
| 19:19.50 | Naikrovek | i think |
| 19:19.53 | Naikrovek | my dialplan-fu is weak |
| 19:20.17 | Naikrovek | something like that though |
| 19:20.22 | Naikrovek | [TK]D-Fender: wake up |
| 19:20.26 | Naikrovek | need dial plan help |
| 19:20.29 | yonahw | I think you really need _+49XXXXXXX |
| 19:20.35 | Naikrovek | how do you dial a + |
| 19:20.58 | Naikrovek | US numbers are actually +1 (321) 321-3213 but we dont' dial a + |
| 19:21.09 | *** part/#asterisk joeflyde (~user@97.104.194.220) |
| 19:21.19 | Naikrovek | he's indicating that the 49 is a country code |
| 19:21.21 | [TK]D-Fender | + = 00 |
| 19:21.29 | exelnet | Naikrovek: + = 00... yeah |
| 19:21.35 | Naikrovek | okay |
| 19:21.51 | Naikrovek | exten => _0049XXXXXXX,1,Dial(SIP/whatever/0${EXTEN:4}) |
| 19:22.41 | Naikrovek | thanks for that 00 info though |
| 19:22.41 | exelnet | ok so what i need would be a general replace + with 00 |
| 19:22.45 | Naikrovek | that answers a few questions |
| 19:23.03 | Naikrovek | afk a moment |
| 19:25.14 | exelnet | Naikrovek: your solution would always add a 0 |
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| 19:25.57 | yonahw | exelnet: you need to have different matches for the different options |
| 19:26.09 | yonahw | that solution handles the +49 case |
| 19:26.24 | yonahw | you need another line in the dialplan to handle the 0. case |
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| 19:27.51 | [TK]D-Fender | exelnet: Where do you actually gt a "+" sign literally? |
| 19:29.27 | exelnet | [TK]D-Fender: mobile phones show incoming connections as +491... so if any of the users adds a number to his/her addressbook, it is highly possible it starts with + |
| 19:30.27 | exelnet | hmm i wonder how i integrate those rules with freepbx... in freepbx, the rules and the dial string get seperated |
| 19:30.49 | p3nguin | Strip the + from the CALLERID() as it comes in. |
| 19:31.36 | exelnet | p3nguin: well i cant do this since those calls will not go over the pbx |
| 19:31.38 | Naikrovek | exelnet: that would add a 0 only when the number starts with 0049, and it would strip the 0049 before adding the 0 |
| 19:31.53 | Naikrovek | if you have an actual + then just remove it in the same manner |
| 19:32.03 | Naikrovek | exten => _+49XXXXXXX,1,Dial(SIP/whatever/0${EXTEN:3}) |
| 19:32.09 | [TK]D-Fender | exelnet: So the call comes in via SIP with a literal + in front? Feel free to manipulat the number however you want before passing the call on so that its formatted in a way that a phone can just it back and go out the way you'd like |
| 19:32.30 | Naikrovek | caller id manipulation does kinda seem like the answer |
| 19:32.34 | Naikrovek | a little |
| 19:32.42 | [TK]D-Fender | exelnet: And as to how to do that within the scope of FreePBX.. that isn't supported here, ask in their channel. As for dilapln raw... well... this is variables 101 |
| 19:32.52 | p3nguin | I have one DID that the CID would have +1 on the front of the 10-digit number when it came in. I ended up using this to get rid of it (and standardize it with my other DIDs): ExecIf($["${CALLERID(num):0:2}" = "+1"]|Set|CALLERID(num)=${CALLERID(num):2}) |
| 19:38.47 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 19:39.55 | exelnet | ok i hacked it a bit... i added a second trunk with CAPI/ISDN1/00$OUTNUM$/bo and added two outbound routes X. and +|. |
| 19:40.20 | Naikrovek | freepbx |
| 19:40.30 | p3nguin | wtf |
| 19:40.40 | p3nguin | Lame people and their stupid GUIs. |
| 19:40.46 | Naikrovek | eh freepbx alone isn't stupid |
| 19:40.52 | Naikrovek | it's just not #asterisk |
| 19:41.25 | Naikrovek | anyone know if chan_sccp supports video |
| 19:41.31 | Naikrovek | http://www.ipphone-warehouse.com/Cisco-Unified-VoIP-Phone-CP-9971-CL-K9-p/cisco-cp-9971-cl-k9=.htm |
| 19:41.40 | Naikrovek | boss is gonna want me to get one of those it hink |
| 19:41.42 | Naikrovek | i think |
| 19:41.54 | exelnet | yeah... hrhr p3nguin :=) i would prefer doing it by hand if i have to do more complicated things... but freepbx helps me staying noob :=) |
| 19:42.11 | exelnet | p3nguin: your solution is way better though... |
| 19:42.12 | Naikrovek | nothing wrong with freepbx, if it gets you what you need |
| 19:42.13 | p3nguin | Rather than normalizing the Caller*ID to not include a non-standard dial character, let's just add a new "route" to compensate. |
| 19:42.29 | Naikrovek | p3nguin: WHOA |
| 19:42.35 | exelnet | p3nguin: thats why i added *hack* |
| 19:42.36 | Naikrovek | no need to add a new route man |
| 19:42.43 | Naikrovek | just add a new rule to the EXISTING route |
| 19:42.44 | Naikrovek | GOSH |
| 19:43.03 | exelnet | Naikrovek: i cant do this, since the gui wont let me :=) |
| 19:43.26 | Naikrovek | it won't? |
| 19:43.34 | Naikrovek | koay |
| 19:43.59 | p3nguin | FreePBX doesn't allow manipulation of Caller*ID? Add another reason to the list for why not to use it. |
| 19:44.30 | Naikrovek | i think you can to a degree |
| 19:44.38 | Naikrovek | but not sure; never tried |
| 19:44.46 | exelnet | p3nguin: well it might, havent looked into manipulation yet |
| 19:45.34 | p3nguin | Sanitizing the CID seems like a simple task. I find it hard to believe it can't be done, and done easily. |
| 19:47.01 | Naikrovek | there is a module to do it, it appears |
| 19:47.11 | Naikrovek | again, haven't tried it |
| 19:47.32 | Naikrovek | looks like it will do what you pasted in earlier though |
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| 19:49.06 | Skeeter- | Does anyone happen to have some Old Spectralink 8030 in his hands?? |
| 19:49.21 | exelnet | a stupid question. is the cid the number the client tries to dial? e.g +491234567 ? |
| 19:49.52 | yonahw | the cid is the caller id |
| 19:50.16 | yonahw | how are your users trying to dial with a +? what device are they using? |
| 19:50.39 | exelnet | yonahw: mobile |
| 19:50.58 | exelnet | yonahw: long press the 0 |
| 19:51.32 | yonahw | exelnet: these are users dialing into your system or out from? |
| 19:52.13 | exelnet | yonahw: using my system to dial out. |
| 19:52.27 | *** join/#asterisk adam1 (~adam1@adam.niagara.com) |
| 19:52.52 | yonahw | how are they connected to your system? |
| 19:53.03 | exelnet | sip |
| 19:53.36 | adam1 | How can I get my PBX system to dial a 310-1010 (Pizza hut) Les.net does not support local dialing of these such numbers is there a main number that 3101010 works (canada). |
| 19:53.57 | Naikrovek | adam1: why is 310 special |
| 19:54.00 | Naikrovek | eggs him on |
| 19:54.21 | adam1 | because I want to order my pizza god dammit! :P |
| 19:54.31 | Naikrovek | because 310 will find the local pizza place even if you dunno the local number |
| 19:54.35 | Naikrovek | lol |
| 19:54.42 | [TK]D-Fender | adam1: Because les.net wants you to dial more than 7 digits <----- |
| 19:54.57 | yonahw | so you need to strip then the way Naikrovek was telling you before |
| 19:55.12 | *** part/#asterisk joeflyde (~user@97.104.194.220) |
| 19:55.15 | Naikrovek | i told my wife to strip. she hit me. |
| 19:55.25 | yonahw | you can manipulate the number being dialed so that it is acceptable to your provider |
| 19:55.50 | [TK]D-Fender | adam1: And you made a very sweeping claim that you can grab ANY phone in Canada and just dial 310-1010 and it will work. Lots of places require 10 digits regardless |
| 19:56.01 | [TK]D-Fender | adam1: And yes I'm talking analog Bell lines |
| 19:58.01 | ZeXr0 | [TK]D-Fender : Even here in montreal, there are some number that can still be dialed with only 7 numbers |
| 19:58.02 | yonahw | adam1: what happens if you prepend your area code to the number? |
| 19:58.26 | ZeXr0 | But these are exception |
| 19:59.11 | ZeXr0 | But I can confirm that in the 514 with my cellphone, dialing 310-1010 does in fact reach pizza hut |
| 19:59.15 | [TK]D-Fender | ZeXr0: I'd love to know where.. |
| 19:59.48 | adam1 | thank you ZeXr0 |
| 19:59.49 | [TK]D-Fender | ZeXr0: most cell companies required 10 digi LONG before many exchange areas like Montreal, GTa, etc required it |
| 19:59.57 | ZeXr0 | Let me try with a Bell analog line |
| 20:00.16 | adam1 | ZeXr0: I'm in St. Catharines it works on my Rogers iphone. |
| 20:00.55 | ZeXr0 | [TK]D-Fender : And the bell analog line from Montreal (514) also allows 310-1010 |
| 20:01.18 | adam1 | <---me thinks I started something hehe |
| 20:01.50 | ZeXr0 | But dialing 514-310-1010 works too. But 450-310-1010 doesn't work |
| 20:01.53 | adam1 | but seriously now when customers switch to us they have to use the local pizza hut # form the phone book. I wish there was a way to route them all. |
| 20:02.03 | ZeXr0 | I'll see if a friend of mine that dial it from a 450 line to see if it's working |
| 20:04.25 | ZeXr0 | It works from a 450 number, you can dial 310-1010 but not 450-310-1010 |
| 20:04.45 | adam1 | well alternative is to order form pizza hut website hrmzz |
| 20:06.22 | adam1 | I suppose an alternative is to get once big enough to get it is a Bell PRI with multiple outgoing channels and use a digium card to do the 310-1010 |
| 20:11.32 | ZeXr0 | There's a weird ringtone when calling 310-10-10 |
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| 20:22.52 | [TK]D-Fender | cehckout time, later all |
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| 20:39.19 | [sr] | it looks winter here!!!!!!! |
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| 21:04.12 | *** join/#asterisk xxNickxx (~chatzilla@dyn-227-161.wireless.concordia.ca) |
| 21:07.33 | xxNickxx | How can I trigger/run a script (python, php or any other kind of script) on an incoming call. I would like to pass the caller ID information to the script. |
| 21:08.10 | [TK]D-Fender | xxNickxx: "core show application system" |
| 21:08.19 | xxNickxx | thank you |
| 21:08.31 | p3nguin | You can also use AGI. |
| 21:08.44 | [TK]D-Fender | p3nguin: lets not complicate things just yet |
| 21:08.54 | p3nguin | :/ |
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| 22:21.35 | digitalc2 | Any bug marshall's around? I'd like https://issues.asterisk.org/view.php?id=14244 reopened |
| 22:23.06 | digitalc2 | The problem is confirmed in 1.6.2.9, although I can't update the bug with such details. |
| 22:25.31 | pabelanger-lap | digitalc2: simple create a new issue, we can reference the original. Besure to also upload a complete debug log. |
| 22:25.35 | pabelanger-lap | ~collectdebug |
| 22:25.36 | infobot | collectdebug is, like, 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 |
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| 23:35.39 | xxNickxx | total newbe here: i am in asterisk CLI (issued asterisk -r) and now want to try out some commands. Whatever i type does not work.... |
| 23:36.03 | bkruse | xxNickxx: asterisk -r |
| 23:36.07 | bkruse | then type 'core show channels' |
| 23:36.13 | xxNickxx | core show application system |
| 23:36.16 | bkruse | it would help to know the version |
| 23:36.17 | xxNickxx | works and shows me some help |
| 23:37.11 | xxNickxx | it says the syntax for System command is: System(command) |
| 23:37.45 | xxNickxx | I try System(ls) and i get an error message |
| 23:37.55 | bkruse | facepalms |
| 23:38.06 | bkruse | That is for use in the Asterisk dialplan |
| 23:38.12 | bkruse | /etc/asterisk/extensions.conf |
| 23:38.14 | xxNickxx | i just installed the latest asterisk now version on a VM to test somoe stuff |
| 23:38.26 | bkruse | exten => 1,1,System("ls") ; this has no point, whatsoever |
| 23:39.48 | xxNickxx | humm |
| 23:40.35 | xxNickxx | what i am trying to do is: 1. detect an incoming call 2. run a python script passing the caller ID info to it. |
| 23:40.41 | xxNickxx | is this too complicated? |
| 23:40.53 | xxNickxx | or i can do it easily |
| 23:41.44 | bkruse | that's relatively simple, look on voip-info.org for asterisk agi |
| 23:41.46 | bkruse | there are python examples |
| 23:42.43 | beardy | For it to have any meaning you'd want speak the text. |
| 23:42.49 | beardy | +to |
| 23:47.32 | xxNickxx | so to call AGi commands, i have to add the command to extensions.conf? |
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| 23:49.23 | csd-199 | hi. I'm using the command "Record" in order to record voice but it does now work. I read that I must press "#" key to stop record, but seems not to work, any idea? |
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