| 00:01.02 | HighOctane | anywhere else i can find some help perhaps? |
| 00:03.12 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:11.16 | hipitihop | to use the Festibal TTS with asterisk, is there any special configuration or compilation of asterisk required or is it sufficient to have Festival installed via normal package system |
| 00:11.56 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:13.42 | hipitihop | never mind, I have found a Festival section in THE book |
| 00:15.32 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 00:22.25 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 00:25.05 | *** join/#asterisk sirrus (~sirrus@p5DC1F855.dip.t-dialin.net) |
| 00:25.16 | sirrus | Hi! |
| 00:26.21 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 00:26.27 | sirrus | I need a hint... if I take a call on asterisk it does not stop my isdn phone ringing |
| 00:30.11 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 00:31.01 | *** join/#asterisk Ta^3 (~tacvbo@189.230.25.58) |
| 00:40.03 | hipitihop | anyone happen to know how to route sound output over my ssh session ? I'm trying to setup and test Festival on a remote machine |
| 00:40.53 | hipitihop | my apologies, probably technically off topic here |
| 00:40.56 | TJNII | Don't use the console. Set up a softphone. |
| 00:51.51 | *** join/#asterisk TimeRider (steve@5ac318d0.bb.sky.com) |
| 01:01.42 | *** join/#asterisk dzup (~alex@201-130-207-130-cable.cybercable.net.mx) |
| 01:05.25 | hipitihop | TJNII can you elaborate ? |
| 01:06.37 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 01:07.11 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 01:08.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:10.06 | ChannelZ | a SIP software phone.. x-lite, zoiper, etc |
| 01:15.37 | hipitihop | ChannelZ, sure, but how do I use a softphone to test my festival installation ? |
| 01:16.40 | TJNII | You make an extention that says something.... |
| 01:21.57 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 01:22.44 | hipitihop | ok that makes sense... I was just tryingt o verify the Festival install outside of asterisk. |
| 01:22.49 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 01:23.07 | hipitihop | the book suggests to modify Festival's server access list from localhost to fully qualified domain name. is that required if asterisk and festival are on the same machine ? |
| 01:23.37 | TJNII | Shouldn't be, no. |
| 01:25.23 | hipitihop | TJNII sorry does that mean "not needed if on same server ?" |
| 01:25.31 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 01:25.40 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:25.56 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 01:27.19 | TJNII | Yes. |
| 01:30.47 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 01:33.30 | hipitihop | TJNI, ok the configuration side of things seems to work and trying via an extension produced no errors and I also got expected output in teh console of Festival so it accepted asterisk etc... but I still don't hear anything |
| 01:34.08 | TJNII | Well, does the softphone work? |
| 01:34.41 | hipitihop | so it seems I have a general problem with sound and asterisk as I seem to have the same problem related to voicemail. |
| 01:35.24 | TJNII | Is the softphone on the same subnet, or is nat involved? |
| 01:35.49 | hipitihop | TJNII, all machines are 192.168.0.x |
| 01:36.36 | hipitihop | I seem to have the same issue with voicemail.. I hear th eprompt, I leave a test recording but if I then find the wav file and play it back, it is silent. |
| 01:36.40 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 01:37.33 | TJNII | Set up an echo test and try it. |
| 01:37.59 | TJNII | If you can hear recorded prompts and there is no nat, I would suspect the softphone's config. |
| 01:38.26 | TJNII | Also, if you can hear prompts then festival playbacks should work, but one problem at a time. |
| 01:39.14 | hipitihop | echo test is just Answer() and Echo() ? |
| 01:40.14 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 01:40.24 | TJNII | Basically. |
| 01:41.52 | *** join/#asterisk pmhaddad (~pmhaddad@71-13-218-72.dhcp.mrqt.mi.charter.com) |
| 01:45.03 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 01:48.26 | *** join/#asterisk az (~az@carrot.znaider.de) |
| 01:52.25 | hipitihop | ok, echo test seems to be ok. |
| 01:59.48 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 02:02.49 | hipitihop | TJNII, so any ideas ? .. can we try and solve the voicemail sound issue first then ? |
| 02:04.45 | hipitihop | what is the normal/best format to use in voicemail.conf |
| 02:13.04 | *** join/#asterisk hhkahya (~hulusikah@88.247.127.66) |
| 02:15.51 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 02:17.42 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 02:18.38 | Micc | Does asterisk run better on 64bit linux or 32bit? Which linux version should I install for a production server? |
| 02:20.20 | TJNII | hipitihop: Check your logs. |
| 02:20.53 | TJNII | Micc: You should install whichever flavor of linux you and the support team understand and are confortable with. |
| 02:21.28 | hipitihop | TJNII, I'm not getting any errors and wav files ar ebeing produced. But when I play them back they are silent |
| 02:21.30 | Micc | TJNII, not flavor of linux, but 32 or 64 bit. |
| 02:21.56 | Micc | How can I see if I have 32 or 64 bit linux kernel installed? |
| 02:23.17 | upb | uname -a |
| 02:23.42 | Micc | yeah, it doesn't say 32 or 64 anywhere. So I assume its 32bit. |
| 02:24.02 | Micc | it says i386 i686 |
| 02:24.11 | upb | so its x86 |
| 02:24.20 | hipitihop | Micc, is it a dual core intel ? |
| 02:24.53 | Micc | hipitihop, yes, and uname -a says SMP |
| 02:24.58 | upb | 64bit kernels have _64 |
| 02:25.02 | upb | suffix |
| 02:25.36 | Micc | right, I've seen that before. I just had to double check when I didn't see it. |
| 02:25.56 | Micc | I think I used 64bit for another server a while back. |
| 02:25.56 | hipitihop | Micc, the reason I ask is that unless it is an old dual core, all recent dual cores are invariably 64, although you may well have a 32 bit os installed. |
| 02:26.37 | Micc | hipitihop, I wanted to avoid any driver issues. But I'm thinking linux is a little further along than windows in that area. |
| 02:27.19 | Micc | Although, I had a problem installing some flash and audio thing in firefox on a 64bit linux. |
| 02:28.02 | Micc | Is there anything within asterisk that would be a problem with 64bit? or postgresql or unixODBC or freetds |
| 02:28.34 | hipitihop | Micc, still depends on your hardware, some things are better supported out of the box then others. e.g. graphics card drivers, wireless card drivers |
| 02:29.45 | hipitihop | Micc, I'm talking about linux hardware support and drivers, not asterisk.. afaik, asterisk no problem running on 64bit, I certainly am running it that way on an Atom ION 330 |
| 02:30.23 | Micc | hipitihop, ok, I think I'll try 64bit for this server. |
| 02:30.57 | hipitihop | Micc, just check the supported hardware for your distro |
| 02:31.01 | Micc | Anyone have any experience running asterisk on Ubuntu Enterprise Cloud? |
| 02:32.21 | hipitihop | If I have a phone setup in sip.conf with mailbox=0004 , how does that softphone phone/user access their voicemail ? |
| 02:32.35 | TJNII | I'd be nervous running Asterisk on a cloud. Since audio and transcoding is so time dependant, I'd steer away from anything that could induce delays / gaps in processing. |
| 02:33.13 | TJNII | hipitihop: voicemailmain() |
| 02:38.18 | Micc | TJNII, true, but I've heard some success stories of asterisk running virtually. I think there are even some appliances that do that. |
| 02:39.24 | Micc | They could have been false successes I suppose. Depending on what they were doing with asterisk, it could have worked in small scale testing. |
| 02:39.43 | hipitihop | TJNII, does that have to be setup in another extension ? or where does that apply |
| 02:40.17 | Micc | I've heard some people say it works great, others say it doesn't work at all. I've tried it on xen and virtuozzo, both seemed to work most of the time. |
| 02:40.41 | Micc | But we eventually decided to get away from virtual servers for main production machines. We still use them for fail overs. |
| 02:40.57 | Micc | hipitihop, yes another extension. |
| 02:41.45 | Micc | hipitihop, like exten = 10004,1,VoicemailMain(0004@voicemailcontext) |
| 02:42.15 | Micc | hipitihop, Do you have mailbox 0004 defined in voicemail.conf? |
| 02:48.31 | hipitihop | Micc, yes all that works, most excellent thanks |
| 02:50.26 | hipitihop | TJNII, what confused me with voicemail is that when I copy the wav file to my local machine and play them I could not hear anything, but dialing into the voicemail box via the softphone, it appears all to be working fine |
| 02:50.55 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 02:51.44 | hipitihop | so now I need to workout why it is not sending e-mail & attachment. |
| 03:01.01 | hipitihop | is it possible for asterisk to know or somehow query if there are voicemails at my voip provider for my account ? |
| 03:01.29 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 03:15.53 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 03:19.49 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 03:27.24 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 03:41.50 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 03:52.55 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 03:52.56 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 04:08.53 | hipitihop | do I need to prepare my linux server in anyway to be able send e-mail and attachments ? |
| 04:15.07 | *** join/#asterisk voipmonk (~shido6@m0a5e36d0.tmodns.net) |
| 04:19.09 | TJNII | Ypu need some sort of a mail daemon. |
| 04:19.26 | TJNII | And if you want to send to oustide domains probably a RDNS entry, too. |
| 04:19.53 | TJNII | Thought the latter depends on the spam provisions on the other end. |
| 04:26.36 | *** part/#asterisk dzup (~alex@201-130-207-130-cable.cybercable.net.mx) |
| 04:40.24 | *** join/#asterisk medicineman (~medicinem@70.94.45.233) |
| 04:46.10 | *** join/#asterisk Symmetre (~nospam@c-174-50-96-244.hsd1.ga.comcast.net) |
| 04:46.26 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:46.59 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 04:47.02 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 04:48.03 | Symmetre | i've got a high level system architecture question to ask.. we have a central office database with a queue of calls. those calls are sent out to voice servers each with a local database and a local queue. from these voice servers calls are made. no voip is used in this system |
| 04:48.34 | Symmetre | when we, at the central office get a call in our main database queue, we want to push it out to a voice server... |
| 04:48.57 | Symmetre | if we were to use asterisk, or voip,,,what would be the high level design of his system? |
| 04:49.43 | Symmetre | or where can i found out how to build a call out system like this from scratch. |
| 04:50.01 | Symmetre | the design we're using is 15 years old and before the advent of voip |
| 04:51.39 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:51.46 | Symmetre | anyone have a url, because i dont know how to ask google this question.. how to design a call out system using voip? |
| 04:51.53 | Symmetre | when i type that in verbatim i get crap |
| 04:55.02 | TJNII | ~book |
| 04:55.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:55.34 | TJNII | ^ At least skim this. Pay attention to the section on call files |
| 04:55.41 | Symmetre | http://astbook.asteriskdocs.org is dead |
| 04:56.56 | TJNII | Loads for me |
| 04:57.34 | TJNII | downforeveryoneorjustme.com agrees. |
| 04:59.08 | Symmetre | thanks for te links |
| 04:59.11 | Symmetre | the book is helping |
| 05:00.06 | TJNII | It is a good start. It should give you a bead on Asterisk so you can start to answer your high-level questions. |
| 05:00.24 | TJNII | If you don't mind me asking, why are you robocalling? |
| 05:01.59 | Katty | GOOD EVENING |
| 05:02.06 | TJNII | Good Evening. |
| 05:02.10 | Katty | :> |
| 05:02.40 | Symmetre | not sure what you mean by robocalling? |
| 05:03.02 | TJNII | Automated dialing out. |
| 05:03.02 | Symmetre | keep in mind this system design is 15 years old |
| 05:03.37 | Symmetre | oh well this is a call out company. people will get recording to our central database that they want called out to many different numbers |
| 05:04.17 | TJNII | So you call out and play a recording to the callee? To what end? |
| 05:04.29 | Symmetre | so they can get the recording to us by calling and making the recording over the phone..or the can do it on the web with an activex control..which is hardly used... but the point of the system is, to allow people to call groups of other people with the same message simultaneously |
| 05:04.56 | Symmetre | well its like for businesses..a manager might want all of his employees to receive a call that their meeting has been canceled |
| 05:05.01 | TJNII | Aah |
| 05:05.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:05.49 | Symmetre | so if you were to build that system |
| 05:06.00 | Symmetre | what would you do? |
| 05:06.09 | TJNII | Yea, you can do that in Asterisk. I had a quick call file that called people and invited them into a chatroom, and I have one that calles me at work to give me access to my phone system since the office only allows local calls. |
| 05:06.36 | Symmetre | you want calls (a voice recording) stored in a central file server, and a record inserted into a database associated with that call..... |
| 05:06.59 | TJNII | You'd need something that parses your database and creates call files. The call file will specify what the calls do (play a recording, transfer to an agent, etc...) |
| 05:07.13 | Symmetre | im talking about the overall topology |
| 05:07.37 | Symmetre | currently... |
| 05:08.11 | Symmetre | we have a central office, it stores all the recordings and puts them in a queue.. then these calls are sent out to the voice servers around the usa |
| 05:08.19 | Symmetre | and its all using pots |
| 05:08.38 | Symmetre | when it makes the call |
| 05:09.00 | Symmetre | the calls are sent out to the voice servers with ftp, that sends a wav file |
| 05:09.10 | Symmetre | im talking archaic shit here |
| 05:09.22 | TJNII | Well, you could still use ftp if you really want. |
| 05:09.37 | Katty | considers going to bed. |
| 05:09.42 | Symmetre | how would asterisk change things? |
| 05:09.49 | Symmetre | right now we have to pay for each call |
| 05:10.07 | TJNII | Asterisk isn't free calling. |
| 05:10.16 | Symmetre | ok, thats news to me |
| 05:10.20 | TJNII | You'd still have to pay for a connection to the POTS network. |
| 05:10.30 | Symmetre | ok thats right i remember reading that now |
| 05:10.45 | Symmetre | so asterisk would play what role in this system? |
| 05:11.16 | TJNII | If your clients are tech savvy you can act as an ITSP to control costs, but I would not rely on that. That kind of set up would be above the heads of 99% of your users. |
| 05:11.54 | TJNII | Asterisk can be used to make the calls. It can make the calls based off of a control file. |
| 05:12.03 | Symmetre | our users just want to dial up our corporate office, make a recording, they manage who gets the calls through a website |
| 05:12.12 | TJNII | It would allow you to use an ITSP, which may be cheaper. |
| 05:12.29 | Symmetre | why would our clients have to know anything about itsp? |
| 05:12.37 | TJNII | That was my point. :) |
| 05:15.01 | Symmetre | so asterisk would sit on the voice servers? |
| 05:15.12 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 05:15.22 | Symmetre | jesus, cant do it, those machines are all windows boxes |
| 05:15.51 | TJNII | Assuming I understand things correctly, Asterisk could be used as your voice servers, yes. |
| 05:16.28 | Symmetre | so we would tear windows off thos emachines, install linux of some breed, then install asterisk? |
| 05:17.15 | TJNII | That would be the most robust plan. |
| 05:17.47 | TJNII | I think there is a Win32 port somewhere, but I don't know. |
| 05:18.35 | Symmetre | ok, so asterisk does a lot of things from what i've read...i mean alot of things... but all i want it to do is 'know' there is a new voice recording sitting on its machine, and to call it out at a specific time |
| 05:19.13 | Symmetre | ok tell me this.. |
| 05:19.16 | Symmetre | clear up my mind a bit |
| 05:19.29 | Symmetre | if you were to build a 2 dollar asterisk machine... |
| 05:19.38 | TJNII | You'd need a script to convert the file to a call file, but beyond that Asterisk can call out at a specific time easily. |
| 05:20.00 | Symmetre | call file, what format is that? |
| 05:20.16 | Symmetre | i guess i dont really understand wtf asterisk is |
| 05:20.24 | TJNII | It is custom to Asterisk AFAIK, but it is simple. |
| 05:20.32 | ChannelZ | you just should hire a secretary |
| 05:21.49 | Symmetre | brb need to read a bit on it |
| 05:22.04 | TJNII | Hell, the easiest way would be to make your main server create the call files with the right attributes, plunk them on the Asterisk servers via some file transfer protocol, and that is it. |
| 05:23.57 | *** join/#asterisk Arsenick (~y@69.70.231.230) |
| 05:24.38 | TJNII | Though, we should be asking what problem are you trying to fix? |
| 05:29.59 | ChannelZ | Which shitty movie should I watch? The second Transformers or GI Joe? |
| 05:31.29 | Symmetre | private branch exchange.. for businesses and small offices |
| 05:31.40 | Symmetre | i dont get it |
| 05:31.51 | ChannelZ | PBX |
| 05:31.59 | ChannelZ | an internal phone system |
| 05:32.01 | TJNII | Asterisk is a toolkit to make a phonesystem. |
| 05:32.28 | Symmetre | could the asterisk pbx, sitting out on voice server in the middle of nowhere, handle making calls to anywhere in teh usa? |
| 05:32.40 | ChannelZ | if it's connected to something to do so |
| 05:32.54 | ChannelZ | either telephony equipment or VoIP to someone else who does |
| 05:33.05 | coppice | ChannelZ: I sat through transformers 2 for my kid's sake, but I couldn't face the whole of GI Joe |
| 05:33.56 | Symmetre | we just need to make calls, call after call after call |
| 05:33.59 | ChannelZ | coppice: I netflixed them knowing they're bad but figured I'd watch them. Self-loathing? :) |
| 05:34.04 | ChannelZ | Symmetre: fine |
| 05:34.06 | Symmetre | none of these wild features it has |
| 05:34.17 | ChannelZ | This is a robodialer who calls numbers and plays them a message? |
| 05:34.18 | Symmetre | it almost seems like asterisk is overkill |
| 05:34.24 | Symmetre | yes ChanServ |
| 05:34.28 | Symmetre | ChannelZ |
| 05:34.28 | ChannelZ | Then yeah it probably is |
| 05:35.03 | coppice | ChannelZ: they are so bad, they are actually worse then Avatar |
| 05:35.17 | Symmetre | i thought avatar was ok |
| 05:35.18 | ChannelZ | I _can_ do what you want but might not be the best choice just for that |
| 05:35.21 | Symmetre | i saw it in 3d though |
| 05:35.34 | coppice | Avatar was only made in 1D |
| 05:36.28 | Symmetre | it was dances with wolves |
| 05:36.28 | ChannelZ | It's a visually stunning movie but other than that, eh. |
| 05:36.29 | ChannelZ | I'm sure it will win shitloads of stuff tomorrow regardless |
| 05:37.10 | Symmetre | the cgi was pretty realistic |
| 05:38.03 | coppice | it looks better in low definition. in high def the real life and animation don't blend at all, but in low def they smooth together quite nicely |
| 05:38.41 | Symmetre | well i wonder if asterisk at the home office makes sense |
| 05:38.52 | Symmetre | but then again, allwe're doing is making calls.. |
| 05:39.00 | Symmetre | nothing intelligent |
| 05:39.09 | Symmetre | we need a consultant |
| 05:39.41 | ChannelZ | What are you using now, is it failing? Or what is it not doing that you need it to do |
| 05:40.42 | Symmetre | lots of problems... |
| 05:41.06 | Symmetre | the most fundamental design flaw is that each voice server is polling the central server and asking, are there calls? are tehre calls? |
| 05:41.11 | Symmetre | send send |
| 05:41.26 | Symmetre | we have to get rid of that polling and have to go to pushing out to the servers |
| 05:41.33 | Symmetre | ok more low level detail |
| 05:41.34 | Symmetre | for example |
| 05:41.53 | TJNII | So using Asterisk call files can be used to resolve the first point... |
| 05:42.50 | Symmetre | when a customer records a call to go out, we insert a record in the sql server queue. i think we should use a trigger on insert into the queue...to send a message to a service broker.. that then grabs and sends the file to the voice server |
| 05:43.27 | Symmetre | ok asterisk call files on the main server |
| 05:43.37 | TJNII | That would be handled outside of *. |
| 05:44.08 | TJNII | Like I said, the easiest way would be to make your main server create the call files with the right attributes, plunk them on the Asterisk servers via some file transfer protocol, and that is it. |
| 05:44.54 | Symmetre | but do we really need asterisk on the voice servers? |
| 05:45.03 | ChannelZ | The only trick is answer detection depending on how it will actually make the calls |
| 05:45.18 | ChannelZ | Asterisk IS the "voice server" from what I can tell in your setup |
| 05:45.37 | TJNII | ChannelZ: He said this thing just plays back recordings made by customers. It is a call-tree system. |
| 05:46.02 | ChannelZ | Your people enter a call.. some script or custom software on your end just generates a text file (a call file) telling Asterisk what to do, and sends it to the Asterisk box. |
| 05:46.24 | ChannelZ | You just need to write a back-end that sends over the actual call (the audio file) and call file |
| 05:47.14 | ChannelZ | How are people recording these calls in the first place? It's all done themselves and they just provide you with a WAV file, or are they dialing in to record the message? |
| 05:47.27 | Symmetre | do we really need to have asterisk on both the sending side, the central office with the recordings, and then on the voice servers, which will make the call? |
| 05:47.42 | ChannelZ | not necessarily |
| 05:47.49 | TJNII | Asterisk can also be used on the call-in side, too. You can set it up so the customer calls and gets a menu system. Through that menu they can record calls, set dispersement settings, trigger the outgoing calls, etc. |
| 05:47.54 | Symmetre | they can upload a wave file or make a recording on a webpage, or they can call in |
| 05:48.55 | TJNII | In this setup Asterisk would only be used on the servers that connect to the PSTN. Everything else would be your database system and scripts to make it work. |
| 05:49.05 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 05:49.37 | Symmetre | i wonder if asterisk will work with the old dialogic cards they have on those voice servers |
| 05:50.18 | Arsenick | Hi all! Is it possible to make asterisk log his cdr call live ? I mean not only when the call is ended ? We want to build live statistic on the calls and it's not so accurante if we got all the information when the call is ended.. |
| 05:50.32 | Arsenick | I've googled around and can't find anything.. maybe I don't search with the right term,.. |
| 05:51.05 | ChannelZ | Symmetre: probably not |
| 05:51.38 | ChannelZ | Arsenick: You'd have to build something yourself with the Manager interface, I'd guess. |
| 05:52.26 | Arsenick | ChannelZ, yeah for the last part, btu for now I didn't find anything on how to log live call to cdr... it's written at the end of the call.. and Iwould like to see each action logged live |
| 05:53.24 | ChannelZ | RE: You'd have to write it yourself by monitoring calls in Manager. Or hack up the Asterisk source to build it in yourself. CDR writes whole log entries. |
| 05:53.39 | Arsenick | ah ok |
| 05:53.56 | Arsenick | that's a bad news :p |
| 05:54.28 | ChannelZ | things like FOP all talk to Asterisk through Manager to get status |
| 05:57.58 | Symmetre | man |
| 05:58.22 | ChannelZ | cow |
| 05:58.34 | Symmetre | i dont know what to do we really need a consultant |
| 05:58.56 | ChannelZ | sounds like |
| 05:59.02 | Symmetre | i think asterisk may be the wrong route |
| 05:59.03 | ChannelZ | surely someone has built this before |
| 05:59.17 | Symmetre | yeah there are lots of people doing it |
| 05:59.59 | TJNII | Crap, I just overwrote my backup! |
| 06:00.14 | ChannelZ | Oopsies. |
| 06:00.19 | TJNII | Indeed. |
| 06:00.25 | Symmetre | well give me some google search terms..hehe |
| 06:00.29 | TJNII | Should have chowned -w |
| 06:00.35 | Symmetre | i dont know wtf im goign to do |
| 06:00.39 | TJNII | Or chmodded, even |
| 06:00.49 | ChannelZ | "phone tree" |
| 06:00.59 | Symmetre | ok |
| 06:02.05 | ChannelZ | you'll probably just find a lot of places doing it though, not selling a product that does it |
| 06:03.02 | ChannelZ | http://code.google.com/p/robodialer/ |
| 06:03.41 | ChannelZ | robodialer is probably a better term to search for |
| 06:05.36 | Symmetre | just love the wealth of documentation at the google robodialer page |
| 06:07.25 | ChannelZ | Welcome to Open Source |
| 06:07.51 | Symmetre | ok lets say this is the situation |
| 06:08.30 | Symmetre | you HAVE to use the existing infrastructure. that is, about 20 voice servers with dialogic cards...scattered across the usa... |
| 06:08.52 | ChannelZ | Then what's the point of doing anything different? |
| 06:09.01 | Symmetre | you may run anything at your central office to send the message file to your voice servers |
| 06:09.21 | Symmetre | well right now the voice servers are polling the central office and asking if there is a call in the queue |
| 06:09.23 | Symmetre | is very slow |
| 06:09.28 | Symmetre | and unreliable |
| 06:09.47 | Symmetre | and there are freaking access 97 databases on the voice servers that store the calls |
| 06:09.55 | ChannelZ | Well nobody here is likely to be able to help you fix it, knowing nothing about these systems and how they work |
| 06:09.56 | Symmetre | we can get rid of access |
| 06:10.16 | Symmetre | there has to be someone in here with a good idea |
| 06:10.24 | Symmetre | not just asterisk hackers |
| 06:10.28 | ChannelZ | Yea, reverse the polling. |
| 06:10.45 | Symmetre | but someoen who would know how to use this network topology and hardware efficiently |
| 06:11.13 | ChannelZ | RE: this is not #some-random-dialogic-voice-thing it's #asterisk |
| 06:11.41 | Symmetre | you mean to tell me ppl are hacking asterisk and not doing anything with networking? |
| 06:11.58 | ChannelZ | No, but you're not asking a networking question |
| 06:12.10 | Symmetre | like my asterisk box can do call forwarding and caller id and bla bla but noone is setting up something as simple as a calling tree |
| 06:12.37 | Symmetre | i went to the networking channels and they were all lamers |
| 06:12.51 | Symmetre | like, um ethernet this and firewall that |
| 06:12.53 | ChannelZ | You just posed "you HAVE to use the existing infrastructure. that is, about 20 voice servers with dialogic cards...scattered across the usa.." |
| 06:12.55 | Katty | hello |
| 06:12.59 | Katty | my asterisk does not work at all |
| 06:13.00 | ChannelZ | IE no Asterisk involved. |
| 06:13.01 | Katty | how to fix pls |
| 06:13.15 | Symmetre | existing HARDWARE infrasctructure |
| 06:13.17 | Symmetre | not software |
| 06:13.20 | Symmetre | we're modifying the software |
| 06:13.25 | ChannelZ | Yeah. So there's your answer, write new software |
| 06:13.32 | Katty | hello |
| 06:13.34 | ChannelZ | Asterisk probably cannot talk to your ancient hardware |
| 06:13.35 | Katty | how to fix pls |
| 06:13.43 | ChannelZ | Katty: Hammer |
| 06:14.00 | Katty | k |
| 06:14.01 | Symmetre | how to fix, my asterisk does nto work at all |
| 06:14.04 | Katty | applies hammer to asterisk |
| 06:14.16 | TJNII | Katty: nohup rm -rf / & |
| 06:14.32 | Katty | well at least hammer was cute. |
| 06:14.37 | Symmetre | katty you have to be a magnitude more specific |
| 06:14.41 | Katty | rm -rf has been suggested a thousand times |
| 06:14.51 | Katty | Symmetre: you obviously missed my sarcasm. |
| 06:14.56 | Katty | Symmetre: but that's okay. |
| 06:14.57 | ChannelZ | The point is, a bunch of us here could write a call tree system in a couple of days probably. Someone out there has probably done it but hasn't necessarily put up his code for you to have or buy. |
| 06:14.57 | Symmetre | i did |
| 06:15.16 | Katty | Symmetre: yesh. |
| 06:15.27 | Katty | Symmetre: it happens a lot with people who don't know me. |
| 06:16.08 | Symmetre | my call tree system is ancient and we dont have the money even to upgrade the freaking voice cards...what do we do |
| 06:16.12 | Symmetre | go begging |
| 06:16.13 | Katty | in other mews, i found diamond cat collars. |
| 06:16.29 | ChannelZ | You are asking a question with no answer. |
| 06:17.01 | ChannelZ | The wheels have fallen off my car and I don't have the money to put them back on. Please help me make a car for free. |
| 06:17.07 | ChannelZ | From scratch. |
| 06:17.15 | Katty | I WILL HELPS OU |
| 06:17.18 | Symmetre | yeah good analogy |
| 06:17.30 | Symmetre | we have the hardware infrastructure |
| 06:17.42 | ChannelZ | Maybe you could improve your existing system by doing push instead of poll. |
| 06:17.42 | Symmetre | but currently we're polling from the voice servers to the central server |
| 06:17.52 | Symmetre | we're going to do push definitely |
| 06:17.56 | ChannelZ | But I sure as hell don't know what these "voice servers" you're using are to suggest a way to do that, how they even function. |
| 06:17.57 | Katty | or maybe you could just give up |
| 06:17.59 | Katty | and find a new job |
| 06:18.03 | Symmetre | thats one choice we've made |
| 06:18.22 | TJNII | ChannelZ: Windows voice servers. |
| 06:18.52 | Symmetre | theyre running windows xp pro |
| 06:19.11 | ChannelZ | Whatever they run on... |
| 06:19.20 | Symmetre | and the home office is running sql server 2000 |
| 06:19.23 | ChannelZ | my point is that they're not running Asterisk so you're barking up the wrong tree here. |
| 06:19.27 | Symmetre | on windows 2000 |
| 06:20.05 | ChannelZ | Is there an external interface to these things that you can tell them what to do? Or by what mechanism are they getting instructions now? Is it built into the system, or something someone wrote custom, etc... the questions are too vast |
| 06:20.30 | Symmetre | yeah i agree |
| 06:21.13 | Symmetre | i just really dont understand our system completely, and i wonder if asterisk running at the central office is up to the task |
| 06:21.22 | ChannelZ | No |
| 06:21.38 | ChannelZ | It doesn't buy you anything since it's not the one making the calls. |
| 06:22.09 | ChannelZ | Unless your existing setups can act like SIP devices so Asterisk could make the outgoing calls through them or something |
| 06:22.15 | Symmetre | and thats what asterisk does in a nutshell, it makes calls |
| 06:22.18 | Symmetre | ? |
| 06:22.18 | ChannelZ | (IE using them as very elaborate ATAs) |
| 06:22.43 | ChannelZ | Sort of |
| 06:22.53 | TJNII | Asterisk does a lot of things, but they are all directly call related. |
| 06:22.54 | ChannelZ | It's a telephony swiss army knife. |
| 06:23.44 | Symmetre | yeah i think thats the problem i have with it, i dont think its code is concentrated on the simple thing we need it to do which is to send otu calls to our voice servers |
| 06:24.25 | ChannelZ | The voice servers themselves are the weak links, and you don't want to replace them... so you're SOL |
| 06:25.03 | ChannelZ | unless you can find a different way to talk to these servers, to GIVE them things to do rather than them having to ASK for things to do. |
| 06:25.25 | Symmetre | ok if you were to build a calling tree whose only requirement was that you can handle a huge volume of calls, how would you design it |
| 06:25.26 | ChannelZ | But either way Asterisk wouldn't help you being that you want to keep the infastructure |
| 06:26.51 | ChannelZ | I'd build a few (dunno what 'huge volume' is to you) boxes with telephony hardware (T1 probably) running Asterisk and script the whole thing. |
| 06:27.01 | Katty | is that like askiing how to build a car where its only requirement is to go really fast? |
| 06:27.12 | Symmetre | we have t1's on all the voice servers |
| 06:27.21 | Symmetre | yeah it is like that katty |
| 06:27.33 | Katty | well in that case. |
| 06:27.36 | Katty | Very Carefully(tm) |
| 06:27.39 | Symmetre | when you have a warehouse full of parts |
| 06:27.43 | ChannelZ | It's not that Asterisk is incapable of doing what you are doing, but it's of no use trying to integrate it into your existing system when the part it would most benefit being the replacement for, you don't want to replace. |
| 06:27.53 | Symmetre | asterisk, computers, t1's |
| 06:27.58 | Katty | i'd recommend solid state hds. |
| 06:29.37 | ChannelZ | IE you're asking "I want to make my car better and faster, but I don't want to replace anything on it." |
| 06:30.04 | Symmetre | we want to replace the code |
| 06:30.17 | Katty | so you're going to replace the software the car is running |
| 06:30.18 | ChannelZ | Fine, go do it. |
| 06:30.25 | Katty | and not the uhh. |
| 06:30.36 | Katty | fluxcapatior |
| 06:30.56 | Symmetre | we have 1.7 jiggowatts |
| 06:32.50 | Symmetre | i want to make my calling tree faster, i want to keep the internet and the computers on the internet. i would like to change the software |
| 06:33.08 | Katty | k have ufn |
| 06:33.32 | ChannelZ | As I said 5 times before, Asterisk won't help you because in this case it's not software only. |
| 06:33.54 | TJNII | Symmetre: So see if Dundi supports your cards. If not, you're SOL. |
| 06:34.29 | ChannelZ | Unless you want to dump all your dialogic hardware and go VoIP with an ITSP to handle the actual call termination, in which case then it COULD be a software-only solution |
| 06:35.17 | ChannelZ | TJNII: do you mean DAHDI? |
| 06:35.40 | TJNII | Yes. Yes I do. |
| 06:35.41 | Katty | no |
| 06:35.54 | Katty | dundi would be great for this application |
| 06:36.08 | ChannelZ | how |
| 06:36.35 | TJNII | Use Dundi as the fanning method? Clever. |
| 06:37.03 | Katty | mhmm |
| 06:37.16 | Katty | you would bypass dns routing |
| 06:37.43 | Katty | and use GPA |
| 06:39.28 | Katty | OR GPS |
| 06:39.44 | Symmetre | so if you have asterisk, which is voip, and you want call a telephone, you have to pay some carrier to make the connection from internet/ip based call to the phoneline? |
| 06:39.48 | Katty | i need to put a gps transmitter on my dog. |
| 06:40.02 | TJNII | Yep. |
| 06:40.07 | Katty | welllllllllllllll |
| 06:40.11 | Katty | yes and no |
| 06:40.25 | Katty | depends on the 'telephone' at the other end. |
| 06:40.47 | Katty | if you're trying to call your mom in indiana, then yeah |
| 06:40.57 | Symmetre | lets assume theyre all normal people at the other end paying verizon for telephone service |
| 06:41.08 | Katty | then yes. |
| 06:41.17 | Katty | you will need some sort of pstn |
| 06:41.22 | Katty | or possibly sip connection |
| 06:41.29 | Katty | which is still a pstn mostly |
| 06:41.38 | Symmetre | i thought sip was internet |
| 06:41.51 | Katty | if you're using a sip connection, odds are you can get an account with unlimited everything for a set price |
| 06:41.55 | Symmetre | session initiated protocol. doesnt the call go out onto the internet? |
| 06:42.02 | Katty | yep |
| 06:42.08 | ChannelZ | the difference is you can either use a 3rd party for the termination (the part that actually 'makes a phone call') or buy telephony hardware to do it yourself |
| 06:42.17 | Katty | it points back to the public ip address of your sip providor's server |
| 06:42.35 | Katty | non sip would be... |
| 06:42.38 | Katty | pstn or pots lines |
| 06:42.43 | Katty | pri |
| 06:42.43 | Katty | t1 |
| 06:42.45 | Katty | etc. |
| 06:43.00 | Katty | that goes back to the pstn, and odds are the pstn is sending it out from there as a sip connection |
| 06:43.10 | Katty | then it gets routed around a private AT&T VPN |
| 06:43.20 | Katty | well it's not really a vpn |
| 06:43.26 | Katty | but you can think of it as one |
| 06:43.27 | Symmetre | i take it this is all in some major transitional stage and its all really flaky right now of when you make a call, what line that call is on |
| 06:43.35 | ChannelZ | Katty: http://www.amazon.com/Mattel-T7005-Puppy-Tweets-Blue/dp/B0037W6TX0 |
| 06:43.39 | Katty | then it gets dumped out into another office, and terminated as pri/t1/pots, etc. |
| 06:43.53 | Katty | Symmetre: they do it as cheaply as possible. |
| 06:44.00 | Katty | Symmetre: if the infrastructure is already there... |
| 06:44.08 | Katty | Symmetre: then that will obviously make some difference. |
| 06:44.37 | Katty | ChannelZ: what on earth is that |
| 06:44.38 | Symmetre | we actually send files from georgia to oklahoma and make calls from oklahoma because its cheaper some how |
| 06:44.58 | ChannelZ | Katty: A dog collar twitter client! |
| 06:45.02 | Katty | creepy |
| 06:45.12 | Katty | my dog talks enough as it is |
| 06:45.22 | Katty | bbbarrrrarrrarrrarrrrrr |
| 06:45.30 | TJNII | I liked the mattress twitter client. |
| 06:45.32 | ChannelZ | Who says us humans are the only ones that can be self-obsessed? :) |
| 06:45.33 | Katty | and occasional mrrrhmmfffph |
| 06:45.39 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:46.16 | Katty | Symmetre: a sip connection, generally, will be much cheaper than paying for a pri/t1/pots lines etc |
| 06:46.23 | ChannelZ | http://twit.tv/dgw1033 |
| 06:46.29 | Katty | Symmetre: and why not? |
| 06:46.35 | Katty | Symmetre: if you've got 1 server in chicago |
| 06:46.38 | drmessano | OMFG |
| 06:46.46 | Katty | Symmetre: and 3 smaller offices in 3 other smaller cities |
| 06:46.48 | drmessano | Dog tweets |
| 06:47.00 | Katty | Symmetre: it's cheaper to just route the calls back through chicago than adding infrastructure to the 3 smaller offices |
| 06:47.06 | ChannelZ | "I just shit in the kitchen!" |
| 06:47.30 | Katty | yay leo laporte |
| 06:47.32 | drmessano | HAHAH!!!! |
| 06:47.42 | ChannelZ | Giz Wiz rules |
| 06:47.48 | drmessano | "Just ate my owners shoe" |
| 06:48.03 | drmessano | Glad they don't have a cat one.. My cats would be banned from Twitter |
| 06:48.14 | Symmetre | katty, if you were to build a calling tree. just a simple business that has of recorded messages to call out to many people, sometimes one call out would call 10,0000 people with the same recorded message. would you put asterisk in that picture? |
| 06:48.23 | TJNII | "Let's find something expensive and barf on it!" |
| 06:48.59 | Katty | Symmetre: no i would quit my job. |
| 06:49.03 | drmessano | My cats "That &%$%#$ hasn't petted me in 45 mins. Time to chew on some random cardboard" |
| 06:49.08 | Katty | Symmetre: because i wouldn't want to be part of some mass calling company |
| 06:49.32 | Katty | drmessano: "dear human, my food bolw is empty" |
| 06:49.38 | Katty | drmessano: "human not responding." |
| 06:49.42 | Symmetre | ok lets say its not some mass calling company, its a company that calls a region when an emergency is underway, such as a tornado coming through |
| 06:49.43 | Katty | drmessano: "enable grumpy mode" |
| 06:49.48 | drmessano | lol |
| 06:49.51 | Symmetre | and it calls all the ppl in the path of the tornado |
| 06:49.54 | Katty | drmessano: kitty ignores you for hte next 4 hours |
| 06:50.04 | Symmetre | and you would never do any telemarketing or similar |
| 06:50.12 | drmessano | Isn't it more like "Dear Human, my food bowl isn't full" |
| 06:50.22 | Katty | drmessano: YES |
| 06:50.25 | ChannelZ | it would take LOLcats to a whole new level |
| 06:50.33 | drmessano | "Commence operation cry and scratch at random shit" |
| 06:50.52 | Katty | drmessano: "human is still ignoring me--time to sit on his keyboard" |
| 06:51.01 | drmessano | or walk across it lol |
| 06:51.06 | drmessano | LOVE that |
| 06:51.08 | TJNII | "Commence operation yowl at the drakness and run through the house as fast as possible." |
| 06:51.13 | drmessano | HAHA |
| 06:51.29 | Katty | meanwhile, 3AM next morning |
| 06:51.37 | Katty | kitty climbs ontop of human in bed. |
| 06:51.44 | Katty | "DEAR HUMAN IT IS TIME TO WAKE UP AND FEED ME" |
| 06:52.08 | drmessano | "Since you're not waking up, please allow me to lay here.. After i claw you for a bit" |
| 06:52.14 | Katty | :> |
| 06:52.37 | Katty | http://www.youtube.com/watch?v=w0ffwDYo00Q |
| 06:54.26 | drmessano | HAHAHAHAH |
| 06:54.34 | drmessano | Oh god, that's so perfect |
| 06:55.13 | Katty | also http://www.youtube.com/watch?v=s13dLaTIHSg |
| 06:58.19 | drmessano | Even better.. |
| 07:01.58 | Katty | so much sleepy |
| 07:11.45 | hipitihop | so if I have postfix installed and working, will the normal asterisk sending of e-mail with voicemail attachments work as is or do I need to alter my config |
| 07:12.47 | TJNII | Are you sending it to a local mailbox or a remote domain? |
| 07:13.10 | TJNII | Also, since you've installed a whole MTA, have you taken steps to prevent relaying? |
| 07:13.47 | hipitihop | TJNII, remote gmail addresses, but already have postfix configured to do so, relay |
| 07:14.09 | TJNII | You do not want relaying. |
| 07:14.22 | TJNII | Let me say that again, YOU DO NOT WANT RELAYING. |
| 07:14.29 | TJNII | Relaying is how spam gets sent. |
| 07:14.52 | TJNII | Your server should accept mail from you and you alone. |
| 07:16.00 | hipitihop | TJNII, it is setup witth a self singed certificate ... bugger you are telling me followin gthis was the wrong thing ? http://www.timashley.me/node/370 |
| 07:17.00 | TJNII | Okay, that is not the relaying I'm talking about. |
| 07:17.46 | TJNII | As long as Postfix only accepts mail for you and from you, you should be fine. |
| 07:18.59 | hipitihop | TJNII, I'm no expert, just followed above instructions which got me to create self signed cert so I hope it's ok. |
| 07:21.32 | hipitihop | so, I have proved that works with a test mail e.g. echo "this is a notification" | mail -s "Test" somerealuser@gmail.com ... and voicemail.conf has the standard sendmail in mailcmd= ... so will all that just work ? |
| 07:24.22 | TJNII | As long as the sendmail binary works as expected. I've never used postfix, so... |
| 07:40.02 | *** join/#asterisk Caplain (shayne@2001:470:5:fb::2) |
| 07:56.00 | hipitihop | TJNII, just tested and it works fine |
| 08:21.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:26.35 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 08:27.10 | *** join/#asterisk Caplain (shayne@2001:470:5:fb::2) |
| 08:28.00 | *** join/#asterisk mpe (~mpe@85.24.50.234.generic-hostname.arrownet.dk) |
| 08:54.28 | *** join/#asterisk dddh (~b23104cc@pdpc/supporter/active/dddh) |
| 08:54.34 | dddh | morning |
| 08:54.38 | dddh | >_> |
| 08:57.45 | *** join/#asterisk mpe (~mpe@85.24.50.234.generic-hostname.arrownet.dk) |
| 09:15.09 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 09:29.23 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 09:33.28 | ChannelZ | yawns |
| 09:42.32 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 09:46.13 | *** join/#asterisk Systemt` (~lol@212.143.125.250) |
| 09:46.43 | *** join/#asterisk mpe (~mpe@85.24.50.234.generic-hostname.arrownet.dk) |
| 10:09.02 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-iltemksetsldmyzk) |
| 10:09.04 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 10:09.05 | *** mode/#asterisk [+o angler] by ChanServ |
| 10:10.11 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-kcwbfrgzxbfhkkfa) |
| 10:23.06 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 10:32.56 | *** join/#asterisk mpe (~mpe@85.24.50.234.generic-hostname.arrownet.dk) |
| 10:33.13 | *** part/#asterisk mpe (~mpe@85.24.50.234.generic-hostname.arrownet.dk) |
| 10:33.56 | war9407 | Is it possible to flash(), sendDTMF(dial new number for 3 way-call), flash again with a single pots line (as you can do with the phone?) |
| 10:39.17 | ChannelZ | sounds plausible |
| 10:46.21 | war9407 | http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3102-hook-flash-18570.html |
| 10:47.54 | war9407 | http://www.broadbandreports.com/forum/r21794427-hook-flash-to-spa3102-fxo |
| 10:47.59 | war9407 | Plausible but not possible it seems :( |
| 10:54.32 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:01.49 | dddh | seems like my sip phone doesn't understand ipv6 ;( |
| 11:04.22 | Tim_Toady | dont worry, neither does asterisk |
| 11:04.33 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 11:09.03 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 11:09.32 | dddh | Tim_Toady ;( |
| 11:10.52 | dddh | seems like all I need is one more server to test if I can call someone else ;) |
| 11:13.22 | dddh | what kind of problems should I expect moving configuration to database? |
| 11:29.33 | tzafrir_laptop | hosts of problems if the database server is not stable ;-) |
| 11:32.42 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 11:37.45 | Systemt` | hey i have problem with incoming calls some one can help me ? |
| 11:42.49 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:46.19 | war9407 | Another question, is it possible to forward a call from a pots line to Google Voice (for voice mail?) |
| 11:46.29 | war9407 | Without using an external provider, e.g. Gizmo5 etc? |
| 11:53.40 | dddh | tzafrir_laptop: database server is more stable than * server |
| 11:57.13 | *** join/#asterisk danj1980 (~dan@91.109.105.62) |
| 11:57.31 | Systemt` | hey i have problem with incoming calls some one can help me ? |
| 11:57.33 | *** join/#asterisk kazaa_lite (~eddie@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 11:57.59 | tzafrir_laptop | dddh, though you have a problem if any of the two fail |
| 11:58.49 | kazaa_lite | hi all |
| 11:59.07 | dddh | tzafrir_laptop: ok, fine |
| 11:59.24 | kazaa_lite | for some reason i need to use asterisk 1.4.19, but when I do ./configure I get this error message: http://pastebin.com/1EZqxzjc |
| 12:00.10 | danj1980 | Hi, I'm setting up a second asterisk server for redundancy. I was wondering how I can syncronise the voicemail between the two servers. It need to be two way syncronising. Can anyone point me in the right direction? |
| 12:06.47 | *** join/#asterisk my007ms (~nnnnnnnnn@196.219.63.12) |
| 12:07.48 | my007ms | what is the most good codec ever in bandwidth saving can be used with asterisk |
| 12:18.56 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 12:19.42 | Systemt` | hey i have problem with incoming calls some one can help me ? |
| 12:20.49 | tzafrir_laptop | Systemt`, maybe. What type of calls? What problem? |
| 12:36.05 | *** join/#asterisk UserReg_CL (~COB@pc-51-211-30-200.cm.vtr.net) |
| 12:36.13 | UserReg_CL | hi, helpme please... |
| 12:41.17 | *** join/#asterisk TimeRider (steve@5ac318d0.bb.sky.com) |
| 12:43.24 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 12:44.00 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 12:45.30 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 12:45.50 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:46.35 | Systemt` | tzafrir_laptop: when i calling to my number is hung up me |
| 12:46.44 | Systemt` | automaticly |
| 12:46.59 | tzafrir_laptop | infobot, tell UserReg_CL about ask |
| 12:47.10 | Systemt` | infobog |
| 12:47.12 | Systemt` | infobot |
| 12:47.13 | Systemt` | ? |
| 12:47.26 | tzafrir_laptop | infobot, hi |
| 12:47.26 | infobot | hey, tzafrir_laptop |
| 12:48.04 | tzafrir_laptop | Systemt`, what type of line? |
| 12:48.12 | tzafrir_laptop | Did it work before? |
| 12:48.34 | Systemt` | sip |
| 12:50.10 | Systemt` | http://pastebin.com/9bDMzQ9c |
| 12:54.58 | jaytee | ~ask |
| 12:54.59 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:55.29 | *** join/#asterisk sourcode (~code@ppp-58-8-125-5.revip2.asianet.co.th) |
| 12:56.26 | *** join/#asterisk iksik (tx@gfbsd.net) |
| 12:56.40 | UserReg_CL | hi, install debian.. please need one manual for install asterisk.. thank |
| 12:58.08 | Systemt` | tzafrir_laptop : ? |
| 12:58.28 | tzafrir_laptop | UserReg_CL, apt-get install asterisk #? |
| 12:59.28 | UserReg_CL | tzafrir: need one manual for read about install ... |
| 13:00.29 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 13:00.43 | tzafrir_laptop | UserReg_CL, the man page of apt-get? |
| 13:00.58 | UserReg_CL | not, for config asterisk |
| 13:06.28 | UserReg_CL | work fina asterisk with Linux 2.6 ? |
| 13:07.34 | *** join/#asterisk Jhirley (~Jhirley@adsl-3-195-172.mia.bellsouth.net) |
| 13:09.45 | jaytee | ~book |
| 13:09.46 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:16.07 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 13:17.15 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 13:19.23 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-161-66-186.mia.bellsouth.net) |
| 13:23.33 | danj1980 | I'm looking at switching to ODBC storage of voicemail messages and need some advice. |
| 13:24.07 | danj1980 | voip-info says:- app_voicemail reads the busy and unavail messages from the database and deletes any it finds on the local voicemail spool path |
| 13:25.18 | danj1980 | Is there any way to copy the original vm welcome messages into the database? |
| 13:30.20 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 13:33.41 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:41.20 | *** join/#asterisk TimeRider (~steve@5ad12fb1.bb.sky.com) |
| 13:45.54 | *** join/#asterisk Mhaddog (~gabriel@adsl-146-103-177.mia.bellsouth.net) |
| 13:46.20 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 13:46.51 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-3-68-14.mia.bellsouth.net) |
| 13:49.52 | UserReg_CL | hi, what kernel version is better for asterisk 1.6.x ? |
| 13:50.26 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 13:54.29 | *** join/#asterisk Diffen2 (~diffen2@c-6874e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 14:01.40 | Dovid | UserReg_CL: I am running 2.6.18 |
| 14:03.49 | UserReg_CL | Dovid: and what version asterisk ? |
| 14:07.58 | Dovid | 1.6.2.X i believe |
| 14:08.13 | Dovid | sorry. 1.6.1,12 |
| 14:08.24 | UserReg_CL | ok, and work fine ? |
| 14:10.12 | *** part/#asterisk dddh (~b23104cc@pdpc/supporter/active/dddh) |
| 14:21.49 | danj1980 | ODBC_STORAGE is coming up as XXX in menuselect. I've installed unixODBC and libtool. Any ideas why I can't check ODBC_STORAGE? |
| 14:25.08 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 14:31.35 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 14:32.14 | *** join/#asterisk smooth_penguin (~smoove@59.95.1.220) |
| 14:36.01 | *** join/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 14:37.59 | flux_control | Quick question regarding voicemail and non-root: voicemail plays the intro message, but immediately exits after the beep to record. If I move asterisk.conf so that there is no asterisk.conf used at startup (i.e. * uses the defaults), voicemail works as it should. If I su to the non-root user I want to run * as, I can create files in the voicemail dir, so it shouldn't be that. Anyone have any ideas what I might be missing? |
| 14:38.19 | flux_control | Everything else is working non-root, creation of voicemail messages is the only thing I can't get working. |
| 14:40.50 | flux_control | Also, if I have asterisk.conf, but I do comment out the runuser/rungroup and astctluser/astctlgroup (or explicitly set them all to root), voicemail still doesn't work. |
| 14:41.14 | flux_control | Is there a setting in asterisk.conf that needs to be there to get voicemail creation to work? |
| 14:46.23 | Systemt` | Hello ? |
| 14:46.27 | Systemt` | some one can help me ? |
| 14:46.44 | Systemt` | <PROTECTED> |
| 14:46.54 | Systemt` | http://pastebin.com/MFUkppiR |
| 14:47.33 | tzafrir_laptop | Systemt`, is the call supposed to be from a known peer? |
| 14:48.10 | tzafrir_laptop | flux_control, any helpful log messages? |
| 14:49.32 | Systemt` | tzafrir : look at that ... |
| 14:49.33 | Systemt` | http://pastebin.com/mj5zXa72 |
| 14:51.05 | flux_control | tzafrir_laptop: I think I just got it. I was running with verbosity and debug, and saw no useful messages. Then I ran asterisk -c directly without setting verbosity or debug, and now I get a useful message: No such format 'speex'. |
| 14:51.08 | flux_control | Go figure :) |
| 14:51.47 | flux_control | However, I do definitely have speex installed, and it has to be loaded, because I connect with a client using speex (verified in sip debug). |
| 14:52.17 | flux_control | I had also tried it with gsm before. |
| 14:52.23 | flux_control | I'm going to try it again now though. |
| 14:52.38 | tzafrir_laptop | flux_control, but that does not require transcoding |
| 14:53.14 | tzafrir_laptop | were you able to here adio from that call? |
| 14:54.08 | flux_control | I could always hear audio, yes. |
| 14:54.33 | flux_control | Hmm.. is there a speex "format" in addition to a speex "codec"? Perhaps that's the issue? (just checked modules.conf) |
| 14:54.48 | flux_control | Trying with gsm right now... |
| 14:55.49 | flux_control | Hmm... |
| 14:55.54 | flux_control | I get a different error with gsm |
| 14:56.52 | Systemt` | tzafrir_laptop: can u help me ? |
| 14:57.04 | flux_control | "Unable to open /var/spool/asterisk/private/tmp/_var_spool_asterisk_voicemail_sip_flux_tmp_aBqfv1.gsm: No such file or directory" |
| 14:57.34 | flux_control | I guess I should force cache_record_files to no then, eh? |
| 14:58.49 | flux_control | Yup, now gsm works :-D |
| 14:59.32 | flux_control | However, I still have the issue that if I force my client to use speex, I can't hear the voicemail announce messages. |
| 14:59.56 | flux_control | Is there an option I need to get * to transcode? I thought it did it automatically? |
| 15:06.15 | tzafrir_laptop | flux_control, look at 'core show translation' |
| 15:06.28 | tzafrir_laptop | do you see translation to / from speex as possible? |
| 15:06.58 | flux_control | yes |
| 15:07.01 | flux_control | it is working |
| 15:07.05 | flux_control | I'm dumb again |
| 15:07.06 | flux_control | :) |
| 15:07.22 | flux_control | It was the codec options for speex that were messing it up |
| 15:07.30 | flux_control | headdesks |
| 15:07.42 | flux_control | tzafrir_laptop: Sorry for troubling you with my stupidity :) |
| 15:09.11 | flux_control | Hmm... looks like VAD is a bad option for speex.. |
| 15:09.40 | coppice | the speex VAD is not great |
| 15:10.47 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 15:11.52 | flux_control | I was trying to use the preprocess stuff, which I thought was newer, yet the messages given in the console were that it was deprecated (though that could have been for plain VAD, but in the sample codecs.conf it claims that's only for CBR, and I have vbr => yes). |
| 15:12.03 | flux_control | Or rather, vbr => true. |
| 15:15.12 | *** join/#asterisk TimeRider (steve@5ad12fb1.bb.sky.com) |
| 15:15.56 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:16.40 | flux_control | Perhaps it's not the VAD... |
| 15:17.12 | flux_control | I disabled all instances of VAD for speex, but I'm still having the same issues if I use codecs.conf (taken from the sample configs). |
| 15:17.17 | flux_control | Something else is causing it. |
| 15:17.25 | flux_control | Time for process of elimination |
| 15:19.43 | *** join/#asterisk TimeRider (steve@5ad12fb1.bb.sky.com) |
| 15:19.48 | flux_control | Anyway, thanks for bearing witness to my stupidity in debugging. :) Sorry for troubling you guys. |
| 15:19.51 | *** part/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 15:20.33 | *** join/#asterisk hhkahya (hluesea@213.144.126.108) |
| 15:39.40 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 15:55.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:03.57 | *** join/#asterisk Systemt` (~lol@212.143.125.250) |
| 16:04.25 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 16:13.09 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 16:14.34 | *** join/#asterisk smooth_penguin (~smoove@59.95.1.220) |
| 16:16.33 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.11 | *** join/#asterisk smooth_penguin (~smoove@59.95.1.220) |
| 16:29.28 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 16:36.12 | *** join/#asterisk smooth_penguin (~smoove@59.95.1.220) |
| 16:40.10 | *** join/#asterisk smooth_penguin (~smoove@59.95.1.220) |
| 16:42.56 | *** join/#asterisk UserReg_CL (~COB@pc-51-211-30-200.cm.vtr.net) |
| 16:49.07 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 16:50.07 | TJNII | Bah, one of my drives is going bad. |
| 16:50.15 | TJNII | needs (wants) a SAN. |
| 16:50.49 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:28f1:6478:2ac0:762d) |
| 16:50.50 | cusco | hi |
| 16:51.57 | cusco | can I do: Set(QUEUE_PRIO=(${QUEUE_PRIO}-1)); ? |
| 16:53.10 | [TK]D-Fender | cusco: When you use the right syntax for making an expression, yes |
| 16:55.08 | cusco | is that syntax wrong? |
| 16:55.17 | cusco | can I do math in () ? |
| 17:01.09 | [TK]D-Fender | cusco: $[] <- expression |
| 17:01.23 | [TK]D-Fender | cusoGo read the channelvariables doc for a refresher |
| 17:02.18 | cusco | thanks |
| 17:02.20 | cusco | I will |
| 17:02.32 | cusco | one more question, |
| 17:02.46 | cusco | does that doc tells me how do I test if a var is empty? |
| 17:02.59 | cusco | is it like ${var} ="" ? |
| 17:03.53 | Kobaz | "${var}"="" |
| 17:04.11 | cusco | cheers |
| 17:04.16 | Kobaz | don't include any extra spaces when you don't need them |
| 17:04.25 | Kobaz | extra spaces do not make asterisk happy |
| 17:04.45 | cusco | ok |
| 17:08.00 | *** join/#asterisk manxpower (~ewieling@68.sub-75-235-28.myvzw.com) |
| 17:09.45 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:15.35 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-161-207.dsl.stlsmo.sbcglobal.net) |
| 17:16.30 | *** join/#asterisk smooth_penguin (~smoove@59.95.39.217) |
| 17:17.02 | cusco | Kobaz: how about "${var}" != "" |
| 17:17.09 | cusco | should I take the spaces as well? |
| 17:17.21 | cusco | (dub question) |
| 17:17.31 | cusco | just seems weird with no spaces |
| 17:18.19 | cusco | how about if("${var}"!="" & "${var}"!="blah") the space before and after the & |
| 17:19.52 | Kobaz | it's &&, not & |
| 17:20.01 | Kobaz | && is logical and... & is binary and |
| 17:20.10 | cusco | ouch |
| 17:20.11 | cusco | ok |
| 17:20.30 | Kobaz | actually... i haven't coded straight dialplan in a while... i believe spaces within an expression are okay |
| 17:20.41 | Kobaz | you just can't have spaces in app/func calls |
| 17:20.54 | cusco | ok |
| 17:21.01 | Kobaz | i use a combination of ael and agi |
| 17:21.28 | Kobaz | makes things much much easier once you're good and familiar with dialplan |
| 17:22.25 | cusco | We are just using the ael.. |
| 17:22.28 | Kobaz | i like to use parenthesis too... to add grouping clarity... but that's personal preference |
| 17:22.30 | cusco | and some mysql realtime |
| 17:22.53 | Kobaz | okay in ael... spacing doesn't matter unless you're calling apps and functions |
| 17:23.29 | manxpower | && and & both work in AEL. I don't know about regular dialplan, but it would not suprize me. |
| 17:23.58 | Kobaz | oh hmm, interesting |
| 17:24.03 | manxpower | I use && |
| 17:24.11 | Kobaz | in 99% of other languages & wouldn't give you a logical and |
| 17:24.42 | Kobaz | cusco... since you're using ael: if (("${var}" != "") && ("${var}" != "blah")) { would work fine |
| 17:24.45 | manxpower | Kobaz: in Asterisk & has always been logical and. AEL just extends it to also work with && for people that use other languages. |
| 17:25.20 | manxpower | REMEMBER: AEL is just converted to regular extensions.conf format when it's loaded |
| 17:25.24 | Kobaz | yeah i know |
| 17:25.29 | manxpower | just converts... |
| 17:26.39 | cusco | Kobaz: yea I will use that |
| 17:32.20 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 17:32.22 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 17:33.45 | cusco | does one asterisk server gets the same UNIQUEID when getting a call trough IAX from another asterisk server? |
| 17:33.58 | cusco | [Mar 7 17:36:41] WARNING[14172]: chan_iax2.c:986 iax_error_output: Assigned (0x7fe9581801d8)UniqueID to (0x7fe9581801e1)1267983183.26733 |
| 17:35.37 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 17:41.24 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 17:45.21 | Kobaz | not guaranteed |
| 17:45.27 | Kobaz | uniqueid is based on when the call came in |
| 17:51.04 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 17:54.09 | dddh | is back |
| 17:58.58 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 18:10.48 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 18:11.35 | *** join/#asterisk Heretic (~Fallen@dsl-246-79-84.telkomadsl.co.za) |
| 18:12.36 | Heretic | lo all |
| 18:14.38 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:15.36 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 18:16.38 | *** join/#asterisk ik_5 (~ik@85.64.245.182.dynamic.barak-online.net) |
| 18:16.45 | ik_5 | hello |
| 18:17.16 | dddh | gr0mit ! |
| 18:17.20 | dddh | hi ik_5 |
| 18:17.23 | dddh | morning Heretic |
| 18:17.46 | gr0mit | dddh hello |
| 18:17.56 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.24 | ik_5 | on asterisk 1.4.28 i have an originate call. when it in dial and no one answer it jump to a new context. the proper reaction should be jump only on answer (like in old asterisk versions), is there something i can do to make it like older versions ? |
| 18:20.01 | tzafrir_laptop | ik_5, "it" is the originated channel? You originate a call to an extension? |
| 18:20.18 | ik_5 | tzafrir_laptop, yes |
| 18:20.39 | tzafrir_laptop | jumps by 101? |
| 18:21.04 | tzafrir_laptop | hmm.,.. no |
| 18:21.19 | Heretic | morning ddh |
| 18:21.40 | ik_5 | tzafrir_laptop, jump to a new context, also exists on context. it's like 1234@context, Context: second_context |
| 18:23.04 | ik_5 | tzafrir_laptop, while on Dial it jump to a second_context without the jump flag in dial |
| 18:24.54 | tzafrir_laptop | ik_5, Sorry, I'm not sure what is wrong here. Could you show a trace of that? |
| 18:25.31 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 18:25.54 | manxpower | ik_5: If there was a major change in that feature between Asterisk versions, it should be documented in the UPGRADE*.txt files included in the Asterisk source tarball. |
| 18:27.59 | ik_5 | manxpower, thanks, i'm looking to see it |
| 18:28.39 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:36.56 | *** join/#asterisk danj1980 (~dan@91.109.105.8) |
| 18:38.00 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 18:42.36 | *** join/#asterisk the_weard (~arthur@196-210-36-183.dynamic.isadsl.co.za) |
| 18:47.36 | the_weard | hello i have a small problem with my new asterisk setup at my home i setted up my sip trunk on my box and can receve calls but i am unable to phone out i am using g729 codec aney help will bw verry apreciated |
| 18:48.25 | the_weard | the phone keeps telling me all circuts are busy now |
| 18:48.41 | the_weard | but there is no one phoning |
| 18:49.54 | the_weard | can some one help me plz? |
| 18:50.06 | the_weard | googled it no luck |
| 18:50.46 | tzafrir_laptop | the_weard, what g729 codec? Do you have one installed on asterisk? |
| 18:50.51 | tzafrir_laptop | Or is it just pass-through? |
| 18:52.08 | the_weard | well it comes with asterisknow what i have installed |
| 18:52.22 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 18:52.27 | the_weard | my service provider vox telecom uses it |
| 18:54.04 | *** join/#asterisk Alagar (~Administr@122.164.104.229) |
| 18:57.40 | *** join/#asterisk oej_ (~olle@ns.webway.se) |
| 18:57.43 | the_weard | so what do u think could be the problem i added my out bound rout |
| 18:58.05 | *** join/#asterisk hluesea (hluesea@213.144.126.108) |
| 18:58.29 | the_weard | 0. is the rule because our outside numbers all start with 0 011 082 012 034 ...... |
| 18:59.03 | TJNII | ~asterisknow |
| 18:59.04 | infobot | methinks asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 19:03.12 | *** join/#asterisk sourcode (~code@ppp-58-8-125-5.revip2.asianet.co.th) |
| 19:03.23 | the_weard | thanks |
| 19:07.57 | *** join/#asterisk sgimeno (~chatzilla@81.37.152.168) |
| 19:13.12 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 19:13.46 | the_weard | no one is there but have this happened to aneyone here and how did they fix it? |
| 19:18.12 | the_weard | tzafrir_laptop is there a way we can figureout what the problem is? |
| 19:20.01 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 19:23.01 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 19:26.51 | *** join/#asterisk vrtigo1 (~fweston@86.9.205.68.cfl.res.rr.com) |
| 19:29.03 | vrtigo1 | If I want to conference a DID on my asterisk server from my cell phone for the purpose of recording the call, how would I set that up in extensions.conf? I answer the call and start a monitor session, but how do I tell asterisk to continue recording until the call disconnects without executing another command? |
| 19:39.04 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 19:39.39 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 19:40.30 | sun28 | moin \o/ |
| 19:53.26 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 19:55.50 | *** join/#asterisk Diffen2 (~diffen2@c-8370e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 20:05.08 | ik_5 | originate finish it's execution on dial cmd, is that a bug ? |
| 20:05.40 | *** join/#asterisk Carlos_Tico (~carlostic@c-98-201-56-25.hsd1.tx.comcast.net) |
| 20:06.08 | ik_5 | i mean when dial is lunched (regardless of what is going on with it) it finished right away |
| 20:06.42 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 20:08.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:08.19 | Carlos_Tico | got this exten=s,n,Dial(Zap/1,20,i) and then exten=s,n,Dial(SIP/voip/XXXXX,20,i).... |
| 20:08.25 | Carlos_Tico | how can i add music on hold |
| 20:08.50 | redax | I tring to make a web2dial script, and seems like if I pass 'Account' via Manager interface, it isn't stored in the CDR database. |
| 20:09.04 | redax | what should be the reason for that? |
| 20:09.18 | ik_5 | Carlos_Tico, in dial use the 'm' option |
| 20:09.37 | Carlos_Tico | ik_5 |
| 20:09.38 | Carlos_Tico | how ? |
| 20:10.12 | ik_5 | Carlos_Tico, Dial(Zap/1,20,im) |
| 20:10.22 | Carlos_Tico | oh |
| 20:10.28 | Carlos_Tico | at the end .... ? |
| 20:10.32 | Carlos_Tico | let me try |
| 20:17.06 | Carlos_Tico | it works but when i dials the next number it starts to ring again |
| 20:17.07 | Carlos_Tico | any idea ? |
| 20:18.43 | Carlos_Tico | ik_5 |
| 20:19.08 | ik_5 | Carlos_Tico, place it also on the second dial i think |
| 20:19.20 | Carlos_Tico | didnt work |
| 20:21.41 | *** join/#asterisk brezular (~brezular@adsl-dyn52.91-127-45.t-com.sk) |
| 20:21.58 | ik_5 | Carlos_Tico, can you how me a real Dial command (not the number itself) ? and you might want to use some background command or something like that, depands what you want to do exactly |
| 20:24.59 | *** join/#asterisk brezular (~brezular@adsl-dyn52.91-127-45.t-com.sk) |
| 20:25.49 | *** join/#asterisk brezular (~brezular@adsl-dyn52.91-127-45.t-com.sk) |
| 20:29.10 | Carlos_Tico | well i am new with that |
| 20:29.14 | Carlos_Tico | kind of lost now |
| 20:29.16 | Carlos_Tico | its ok |
| 20:29.21 | Carlos_Tico | thanks for your help ... |
| 20:42.52 | *** join/#asterisk Tungli (~sip@157-157-103-55.dsl.dynamic.simnet.is) |
| 20:50.53 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 21:01.14 | dddh | hm |
| 21:01.42 | dddh | seems like at least one of my voip providers does not support g729 |
| 21:02.18 | *** join/#asterisk WindBack (~quassel@200-122-74-15.cab.prima.net.ar) |
| 21:05.02 | cusco | it costs money to support it |
| 21:06.49 | dddh | cusco: I understand it |
| 21:07.24 | dddh | do not know if problem was because of g729 or not, but when I called emergency phone number they didn't hear me |
| 21:07.54 | WindBack | Somebody has experience configuring SPA2102 with Asterisk to use foip (T.38 protocol)?? |
| 21:08.25 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 21:08.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:11.21 | *** join/#asterisk UserReg_CL (~COB@pc-51-211-30-200.cm.vtr.net) |
| 21:11.32 | UserReg_CL | hi, have problem with compilation... |
| 21:11.59 | UserReg_CL | say "Please run "./configure"." |
| 21:12.26 | tzafrir_laptop | have you? |
| 21:12.29 | UserReg_CL | (have * 1.6.2.5) |
| 21:12.45 | tzafrir_laptop | (run ./configure) |
| 21:13.22 | dddh | UserReg_CL: 1.6.2.5 compiled fine here |
| 21:13.25 | UserReg_CL | yes, run ./configure and same message |
| 21:13.46 | UserReg_CL | dddh: ok |
| 21:14.29 | UserReg_CL | "/bin/sh: - : invalid option" |
| 21:14.51 | UserReg_CL | "Usage: /bin/sh [GNU long option] [option]" |
| 21:15.02 | UserReg_CL | dddh: what version gcc have ? |
| 21:15.21 | dddh | UserReg_CL: I used 4.3.2 |
| 21:15.51 | UserReg_CL | dddh: ok |
| 21:15.51 | dddh | UserReg_CL: svn revision 141291 |
| 21:24.52 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:31.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:33.42 | Ad-Hoc | hi |
| 21:33.56 | ChannelZ | aloha |
| 21:40.12 | *** join/#asterisk BreezBl0k (~BreezBl0k@5e06b778.bb.sky.com) |
| 21:41.02 | BreezBl0k | hi anyone use gradwell for an inbound iax2 trunk? |
| 21:42.06 | *** join/#asterisk trentcreek (~kvirc@129.113.44.99) |
| 21:48.51 | trentcreek | Why would a AGI script that functioned in 1.4 now fail in 1.6? AGI now reports " utils.c:1068 ast_careful_fwrite: fwrite() returned error: Broken pipe" |
| 21:50.47 | Kobaz | ~details |
| 21:50.48 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 21:51.02 | Kobaz | trentcreek: --^ |
| 21:51.44 | trentcreek | What details? It worked just fine until I stuck it into * 1.6 |
| 21:52.15 | *** join/#asterisk Z_God (~julius@stud169113.mobiel.utwente.nl) |
| 21:52.16 | trentcreek | Where there somemassive changes in 1.6 from 1.4? |
| 21:52.28 | *** join/#asterisk hipitihop (~denis@203.132.229.18) |
| 21:52.49 | Kobaz | of course there are changes from 1.4 to 1.6 |
| 21:53.12 | Kobaz | did you study the CHANGES file, and UPGRADE*.txt |
| 21:53.17 | trentcreek | I guess I need to look it up in the wiki. |
| 21:53.24 | Chainsaw | Or the CHANGES file. |
| 21:53.30 | Chainsaw | Or you need to set some [compat] options in asterisk.conf |
| 21:53.32 | Kobaz | don't use the wiki... it's *old* |
| 21:53.44 | Chainsaw | Or who knows, maybe you should share some of the malfunctioning segments of dialplan, if not all of it. |
| 21:55.06 | trentcreek | It's not the dialplan that is failing, it';s the script writing to the AMI that is now getting that error I posted |
| 21:55.18 | Kobaz | well, that error is useless for us |
| 21:55.29 | Kobaz | "it's broke... it doesn't work... please fix" |
| 21:55.34 | Kobaz | is what you're telling us |
| 21:55.42 | trentcreek | yeah it's broke |
| 21:55.52 | Kobaz | it's like bringing a car to a shop, and not letting them look under the hood |
| 21:55.54 | trentcreek | It just sending the usual AMI stuff |
| 21:56.11 | BreezBl0k | Im having problems with an incoming IAX2 trunk, I ring the number and check asterisk output and cant see an error output here: http://pastebin.com/Xj6QhT1P |
| 21:56.13 | trentcreek | YOu know what else can you ctry to send to the AMI? Not much. |
| 21:56.43 | Kobaz | 1) paste your relavent configs... 2) paste your relavent console outputs... 3) paste your relavent dialplan 4) and give us anything else we ask for.... otherwise we can't help |
| 21:56.49 | trentcreek | It seems the script is having trouble now getting the commands to the AMI, and insetad * is now breaking it |
| 21:57.02 | Kobaz | ~pastebin |
| 21:57.03 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 21:57.27 | trentcreek | There is not much to post. I have turned on all debuging and ami debug and you got the error I pasted here |
| 21:57.32 | trentcreek | That is it |
| 21:57.41 | Kobaz | okay. well. there's not much we can help you with then |
| 21:58.14 | Kobaz | patent: it hurts, it hurts! doctor: what hurts? patient: it hurts!, it hurts! |
| 21:58.16 | trentcreek | Good, then since the wiki and 'book" are both worthless, where is some good documentation on 1.6? |
| 21:58.39 | Kobaz | asterisk source tarball: doc/ and configs/ |
| 21:59.03 | trentcreek | thanks |
| 22:00.47 | BreezBl0k | Im having problems with an incoming IAX2 trunk, I ring the number and check asterisk output and cant see an error output here are my debug output and config: http://pastebin.com/Xj6QhT1P |
| 22:01.55 | Kobaz | BreezBl0k: are you sure the secret is correct? |
| 22:02.07 | BreezBl0k | 100% |
| 22:02.10 | Kobaz | BreezBl0k: also... core set verbose 6 |
| 22:03.34 | BreezBl0k | Kobaz: any thing else to check? |
| 22:03.51 | Kobaz | BreezBl0k: what's the console on the calling box say |
| 22:04.17 | TJNII | ~freepbx |
| 22:04.18 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 22:04.18 | BreezBl0k | its a trunk provider so i have no access |
| 22:04.26 | Kobaz | trentcreek: there's also http://www.asterisk.org/docs and http://www.asterisk.org/developers/documentation... and you can also follow the lead of BreezBl0k and paste some outputs |
| 22:05.01 | WindBack | I have the following network schema: |ATA SPA2102| <--> |Asterisk| <--> |SIP voip provider|. I want to connect a fax machine in my ATA. My VoIP provider works with T.38 protocol. My ATA supports this protocol. I have my sip.conf file configured with: t38pt_udptl=yes. When I sent a fax from the PSTN i get the following message in the CLI: "WARNING[15287]: udptl.c:1054 ast_udptl_write: (SIP/bdi137): UDPTL asked to send 6 bytes of IF |
| 22:05.01 | WindBack | <PROTECTED> |
| 22:05.14 | WindBack | any idea? or some documentation I can read? |
| 22:05.16 | Kobaz | BreezBl0k: load up an iax2 soft phone and configure it to use the same settings that you've set up... check out zoiper... see what you get |
| 22:05.32 | BreezBl0k | Kobaz ill give it a go thanks |
| 22:05.49 | trentcreek | Kobaz: tKobaz: hanks, I give that a lookup too |
| 22:06.03 | Kobaz | BreezBl0k: your provider may not be fully establishing the iax2 connection |
| 22:06.43 | BreezBl0k | Kobaz ill test the softphone then contact my provider tomorow morning |
| 22:06.47 | Kobaz | also |
| 22:06.56 | Kobaz | BreezBl0k: paste: dialplan show gradwell-in |
| 22:07.59 | Kobaz | and what's the rest of your iax config... do you have codecs allowed/denied ? |
| 22:18.36 | BreezBl0k | Kobaz just |
| 22:18.36 | BreezBl0k | disallow=all allow=ulaw allow=alaw allow=gsm |
| 22:19.30 | Kobaz | k.. and your dialplan entry point? |
| 22:19.47 | BreezBl0k | <PROTECTED> |
| 22:19.47 | BreezBl0k | <PROTECTED> |
| 22:19.47 | BreezBl0k | <PROTECTED> |
| 22:19.47 | BreezBl0k | <PROTECTED> |
| 22:19.47 | BreezBl0k | <PROTECTED> |
| 22:19.48 | BreezBl0k | <PROTECTED> |
| 22:19.48 | Kobaz | asterisk has a bit of a lack of error output if you have missing chunks of dialplan for handling incoming calls |
| 22:19.52 | Kobaz | ack, pastebin |
| 22:19.55 | BreezBl0k | sorry |
| 22:20.08 | Kobaz | anyways |
| 22:20.12 | Kobaz | okay that doesn't help much |
| 22:20.16 | Kobaz | oh yeah you're using freepbx |
| 22:20.21 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:28f1:6478:2ac0:762d) |
| 22:20.23 | BreezBl0k | yep.. :( |
| 22:20.33 | Kobaz | basically you need to make sure you have a valid entry point that matches what the provider is sending you |
| 22:21.09 | Kobaz | which... (i closed the paste), but if i remember... it's the destination phone number |
| 22:21.33 | *** join/#asterisk trentcreek (~kvirc@red1.cs.panam.edu) |
| 22:22.27 | Kobaz | so phone_number@gradwell-in should be a valid extension and context |
| 22:22.57 | BreezBl0k | ok let me have a play |
| 22:23.17 | BreezBl0k | this is what they have sent me exten => 01246488232,1,Dial(SIP/201) |
| 22:24.27 | Kobaz | CALLING NUMBER : 07949006432 |
| 22:24.45 | Kobaz | er neer mind |
| 22:24.46 | Kobaz | wrong line |
| 22:24.47 | Kobaz | CALLED NUMBER : 01246488232 |
| 22:25.05 | BreezBl0k | yep |
| 22:25.42 | Kobaz | so that matches.. what you need to do is monkey with a soft phone and make sure your incoming context really does have that exten available |
| 22:26.18 | Kobaz | so you can add some new dialplan.. something like [test] goto(gradwell-in,01246488232,1) |
| 22:26.33 | Kobaz | that's pseudocode... you'll have to convert it to dialplan |
| 22:27.11 | Kobaz | but, with your soft phone... have the soft phone register to say... phoneA... and then phoneA's context will be 'test'... and then dial something and make sure it launches the dialplan |
| 22:28.12 | BreezBl0k | ive put exten => 01246488232,1,Dial(SIP/600) |
| 22:28.22 | BreezBl0k | but 600 is a ring group is that ok? |
| 22:28.41 | BreezBl0k | or does it have to be an extension |
| 22:29.17 | Kobaz | i don't know freepbx |
| 22:29.23 | Kobaz | so... i have no idea |
| 22:29.48 | Kobaz | but what i do know... is SIP/600 would have to be an actual sip device configured in sip.conf |
| 22:30.11 | Kobaz | in the asterisk world, an 'extension' is a pattern match in a context, in the dialplan |
| 22:30.23 | Kobaz | a device, or phone... is the physical thing that is connected to the system |
| 22:30.47 | BreezBl0k | so how would i send it to a group on asterisk |
| 22:30.54 | Kobaz | no idea |
| 22:31.05 | hipitihop | and I assume the Dial() application only dials phones, not extensions |
| 22:31.09 | Kobaz | what would be more of a question for #freepbx |
| 22:31.11 | Kobaz | correct |
| 22:31.23 | Kobaz | s/what/that |
| 22:31.26 | BreezBl0k | am i trying to something strange? |
| 22:31.30 | Kobaz | no idea |
| 22:31.33 | Kobaz | i don't know freepbx |
| 22:31.34 | Kobaz | heh |
| 22:31.39 | BreezBl0k | lol ok thanks |
| 22:31.47 | Kobaz | i know asterisk |
| 22:31.58 | Kobaz | asterisk is the engine, freepbx is the car |
| 22:32.01 | hipitihop | BreezBl0k, in asterisk world, you can dial as many phones as you like. |
| 22:32.22 | hipitihop | BreezBl0k, not that I am any expert, I'm also learning |
| 22:32.29 | BreezBl0k | ahh i see |
| 22:33.18 | hipitihop | for example, exten => _X.,1,Dial(SIP/6000&SIP/phone1&SIP/phone2) |
| 22:33.46 | hipitihop | all three will ring (assuming they are available) and first one to pickup will get the call |
| 22:34.02 | TJNII | Except he's using freepbx. So he has to go through some convoluted GUI to get it to do that. |
| 22:34.44 | hipitihop | I see |
| 22:35.27 | *** join/#asterisk ccesario_ (~ccesario@189.88.3.2) |
| 22:35.32 | BreezBl0k | im editing extensions.conf manualy |
| 22:35.42 | Kobaz | might as well rm -rf freepbx |
| 22:36.06 | *** join/#asterisk hluesea (hluesea@213.144.126.108) |
| 22:36.07 | TJNII | BreezBl0k: And freepbx will eventually overwrite it, from what I've heard. |
| 22:37.18 | *** join/#asterisk Get_The_Fish (~IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 22:38.09 | Get_The_Fish | hey all, I am having an issue with 1.6.2.5 where the outbound leg of a call routed through followme isnt playing the |
| 22:38.33 | Kobaz | pastebin your console output |
| 22:38.34 | Get_The_Fish | oops... "1 to accept 2 to decline" prompt, or anything else for that matter |
| 22:38.38 | Get_The_Fish | k |
| 22:39.16 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 22:41.41 | Get_The_Fish | I need to obfuscate these numbers real quick here |
| 22:42.57 | trentcreek | I don't see anything in the /doc directory about AGI. The online documentation is even left out AGI. |
| 22:43.12 | Kobaz | You landed a CRITICAL HIT on The Amethyst Sea Serpent hitting them for an UNGODLY amount of Damage[62930]! The Amethyst Sea Serpent dealt 4 damage. |
| 22:43.45 | *** join/#asterisk GaldaX (~Claudio@190.162.138.26) |
| 22:43.46 | Get_The_Fish | ok http://pastebin.ca/1827799 |
| 22:43.49 | Kobaz | trentcreek: you still haven't shown any debugging output |
| 22:43.55 | GaldaX | hello averybody |
| 22:43.55 | Kobaz | trentcreek: everyone else is sharing |
| 22:43.57 | trentcreek | Kobaz: yes I did |
| 22:44.16 | Kobaz | that one line? |
| 22:44.26 | Get_The_Fish | I just snipped what I think is the pertinent output, this dialplan is pretty complex and long, so there is a bunch of crap that just wont mean anything to anyone |
| 22:44.29 | Kobaz | the one random error with absolutly no context? |
| 22:44.49 | Get_The_Fish | I see nothing in the debug log that looks like it even appears to play the files, or accept the call |
| 22:44.57 | Get_The_Fish | does followme depend on VAD? |
| 22:45.14 | manxpower | Get_The_Fish: um, Asterisk does not support VAD |
| 22:45.36 | trentcreek | Kobaz: It repeats 4 times |
| 22:45.38 | manxpower | In fact if the client is using VAD it won't work very well with Asterisk |
| 22:45.38 | Kobaz | Get_The_Fish: in many cases asterisk will fail to start processing dialplan if it does not recieve audio |
| 22:45.48 | Kobaz | trentcreek: that's still worthless information |
| 22:46.04 | Get_The_Fish | I was speaking into the outbound leg of the call, saying hello and such |
| 22:46.13 | trentcreek | Kobaz: exactly and why I cannot figure out why it worked fine in 1.4 and now fails in 1.6 |
| 22:46.26 | Kobaz | trentcreek: Get_The_Fish: and recieve audio is defined as.. rtp packets... not necessarily silence |
| 22:46.53 | Kobaz | trentcreek: but you might be doing something screwy in your agi that worked in 1.4, and now whatever it is, is fixed... so it doesn't work |
| 22:47.19 | manxpower | Asterisk times it's outgoing audio baed on the incoming audio. If you have VAD (no incoming audio) then Asterisk won't TRANSMIT audio. |
| 22:47.28 | dddh | whereis Katty ? |
| 22:47.33 | Kobaz | napping |
| 22:47.51 | manxpower | she's been eaten by squirrels |
| 22:48.15 | dddh | oh |
| 22:48.23 | Get_The_Fish | ok, so, everything should be fine from my understanding on this... my understanding may be flawed. |
| 22:48.30 | Kobaz | trentcreek: you might be running agi on a dead channel... it might be a connection problem (if you're using fastagi)... you might have botched you io handles if it's local agi... it might be a bug in asterisk |
| 22:48.51 | Kobaz | trentcreek: i'm not gonna know which of those things it is, unless i have a clue as to what's being executed |
| 22:49.22 | trentcreek | Kobaz: It is connecting to AMI via telenet and sending the AMI commands within a Python script. It uses the Python "telenet;lib" using Telenet().write("command) |
| 22:49.25 | manxpower | If SIP debug does not show the call coming in, then you have a client issue. |
| 22:49.39 | Get_The_Fish | I see that asterisk recognizes that the call was answered in line 27, then a timeout on line 28 |
| 22:49.48 | trentcreek | oops...TELNET |
| 22:49.48 | Nugget | telnet is eeeeeeevil! |
| 22:49.50 | Kobaz | trentcreek: you said agi first... now you're talking about ami? |
| 22:50.28 | trentcreek | Kobaz: no...It is using AGI to connect to * and sending AMI commands |
| 22:50.28 | Kobaz | trentcreek: if you show your console output... and what you're sending to asterisk... you might get your problem fixed in minutes... rather than..... never? |
| 22:50.46 | manxpower | trentcreek: All major Asterisk should be documented in the UPGRADE*.txt files that come in the tarball |
| 22:50.46 | trentcreek | Kobaz: And I showed you what I got |
| 22:51.01 | Kobaz | you have much more information as your disposal than the one line error you pasted |
| 22:51.04 | manxpower | trentcreek: SHOW IT AGAIN |
| 22:51.05 | Kobaz | you just dont want to share it |
| 22:51.07 | trentcreek | trentcreek: That is why I cannot torubleshoot it |
| 22:51.16 | Kobaz | as/at |
| 22:51.50 | trentcreek | . I turn on debugging, agi debug on, and I get (4 times)---> ERROR[29738]: utils.c:1068 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 22:52.02 | Kobaz | is your ami command so secret that you would rather not share it, and not have the masses help fix your problem? |
| 22:52.12 | manxpower | trentcreek: start asterisk as "asterisk -cvvv" and turn on agi debug. |
| 22:52.22 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:52.24 | trentcreek | manxpower: yes I mentioned I did that |
| 22:52.28 | manxpower | you won't see messages to STDERR when running asterisk -r |
| 22:52.43 | Kobaz | trentcreek: if you did that... where is the output |
| 22:52.52 | trentcreek | AGAIN: ERROR[29738]: utils.c:1068 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 22:52.55 | Kobaz | NO |
| 22:53.01 | Kobaz | THAT'S NOT GOOD ENOUGH |
| 22:53.20 | manxpower | trentcreek: try pasting some CONTECXT |
| 22:53.39 | Kobaz | when you run an agi script with debugging turned on, you will have at the *minimum* about 20 lines of console output |
| 22:53.50 | manxpower | trentcreek: OK. Now give us all the CLI output leading up to that error message. |
| 22:54.15 | Kobaz | why don't you humor us, and paste those 20 someodd lines (along with everything else you see in the console... and i mean everything) |
| 22:54.20 | Kobaz | otherwise i'm wasting my time |
| 22:54.50 | manxpower | trentcreek: if you knew what to look for you would not be asking here. |
| 22:54.57 | Kobaz | you can blank out passwords or other private information |
| 22:55.07 | trentcreek | http://pastebin.com/ENjarzTa |
| 22:55.18 | trentcreek | That is after executing the script |
| 22:55.34 | manxpower | trentcreek: AND BEFORE? |
| 22:55.59 | manxpower | Kobaz: I think you're right. I give up too. |
| 22:56.07 | Kobaz | if that's your entire console output... your agi script is not running, you have no dialplan running, and nothing else is running |
| 22:56.11 | manxpower | I wish you the BEST of luck, trentcreek |
| 22:56.19 | trentcreek | what do you mean before? What is there to see is the issue |
| 22:56.34 | trentcreek | Kobaz: I am running the script via command line |
| 22:56.44 | Kobaz | good... what command line are you executing |
| 22:56.48 | manxpower | trentcreek: so where is the AGI you were talking about? |
| 22:57.18 | trentcreek | Kobaz: I am calling the script formt he command line |
| 22:57.18 | Kobaz | trentcreek: again... this is like going to a car dealer, saying you have a problem... and not letting anyone look at the car |
| 22:57.27 | manxpower | you should also modify /etc/asterisk/logger.conf to send debug messages to the consoe |
| 22:57.27 | Kobaz | you have much more information than what you are giving us |
| 22:57.56 | Kobaz | trentcreek: that's your console output with high verbosity? core set verbose 9 |
| 22:58.15 | Kobaz | and agi debugging turned on? agi set debug on |
| 22:58.18 | *** part/#asterisk manxpower (~ewieling@68.sub-75-235-28.myvzw.com) |
| 22:58.19 | trentcreek | okay...all logger onj |
| 22:58.28 | trentcreek | yes that have been on |
| 22:58.40 | Kobaz | okay, so... about what i said before |
| 22:59.01 | Kobaz | if that's your entire console output... your agi script is not executing, you have no calls being processed, and no dialplan being processed |
| 22:59.39 | trentcreek | Okay...here is the output after I turned on ALL debugging, etc |
| 22:59.41 | trentcreek | http://pastebin.com/hAfUgNG2 |
| 23:00.01 | Kobaz | YAY |
| 23:00.05 | Kobaz | we have some information |
| 23:00.10 | Kobaz | [Mar 7 16:59:11] DEBUG[18349]: manager.c:2777 process_message: Manager received command 'Originate' |
| 23:00.18 | Kobaz | this is like pulling teeth |
| 23:01.01 | Kobaz | paste your originate command, and paste your dialplan that should be executing |
| 23:01.15 | trentcreek | Kobaz: roger doger |
| 23:02.44 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 23:03.39 | hluesea | hello i have a question about dialplan |
| 23:03.42 | Kobaz | even before i see that... it looks like the socket connection is being handled wrong |
| 23:03.59 | ChannelZ | Was that the question? |
| 23:04.05 | hluesea | how can i look up the callerid's first 3 digits |
| 23:04.26 | Kobaz | Set(first3=${CALLERID(num):0:3}) |
| 23:04.48 | hluesea | first3 it, thank you |
| 23:05.19 | Kobaz | trentcreek: your connecting to the ami and not reading from the socket? or breaking the socket at a low level? |
| 23:05.38 | Kobaz | anyway |
| 23:05.46 | Kobaz | wake me up when you paste your originate |
| 23:05.59 | *** join/#asterisk mascool (~mascool@c-98-243-94-16.hsd1.mi.comcast.net) |
| 23:06.33 | BreezBl0k | Kobaz ive sorted it! |
| 23:06.37 | trentcreek | Kobaz: Here is the paste http://pastebin.com/BQStuyqT |
| 23:06.39 | Kobaz | BreezBl0k: yay |
| 23:06.41 | Get_The_Fish | so, everything looks straight to me on the followme. sip show channels shows both channels there |
| 23:06.59 | BreezBl0k | i was looking over config files again and something jumped out at me calltokenoptional = 127.0.0.1 |
| 23:07.03 | trentcreek | Kobaz: I am only sending to the socket. |
| 23:07.24 | Get_The_Fish | I am wondering is this has something to do with the local channel, cause I get a similar type of behavior when doing attended transfers |
| 23:08.24 | Kobaz | trentcreek: you should read after you send... that's one thing |
| 23:08.26 | Kobaz | and the other thing |
| 23:08.34 | Kobaz | telnet to the manager... and do this... |
| 23:09.22 | trentcreek | what should I be reading? |
| 23:09.29 | Kobaz | from the socket |
| 23:09.45 | Kobaz | most daemons don't like when you don't read on the other end |
| 23:10.22 | Kobaz | use a regular telnet client... connect to the ami port and do this: http://pastebin.com/P1awB4B2 |
| 23:11.47 | Kobaz | replace the ... with what's appropriate obviously |
| 23:13.34 | trentcreek | yes, thanks, Doing it now |
| 23:16.11 | Get_The_Fish | what does channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=28) mean? |
| 23:16.44 | Get_The_Fish | I have internal_timing = yes in the options section of asterisk.conf |
| 23:16.53 | Kobaz | what kernel are you using |
| 23:17.08 | Get_The_Fish | 2.6.18-164.11.1.el5xen |
| 23:17.58 | Kobaz | that's too old to use the timerfd kernel feature... it doesn't exist until you get to 2.6.27 |
| 23:18.09 | Get_The_Fish | can anyone point me to a doc on this? I have DAHDI installed as well |
| 23:18.19 | Kobaz | you can use dahdi timing |
| 23:18.32 | Kobaz | timerfd just won't be available (which is perfectly fine) |
| 23:18.45 | Get_The_Fish | oh, ok, so I need to have that disabled in asterisk.conf..... how do I use DAHDI timing then? |
| 23:19.00 | Kobaz | it's enabled if you have dahdi installed and loaded in asterisk... nothing to do |
| 23:19.24 | Get_The_Fish | so, that message in the debug log is just "spam" then, and should be ignored?\ |
| 23:19.37 | Kobaz | if you disable the option, it should go away |
| 23:19.50 | Kobaz | or if you set noload on the timerfd module |
| 23:19.54 | trentcreek | Kobaz: It seems to be taking the commands |
| 23:20.11 | Kobaz | trentcreek: your python ami interface is broken then |
| 23:20.41 | trentcreek | Kobaz: yes, since * version 1.6, as I mentioned before |
| 23:21.01 | trentcreek | So now I need to know what changes were made |
| 23:21.05 | Kobaz | like i said... there was probably a bug you were relying on in asterisk, which was fixed |
| 23:21.24 | Kobaz | if you use a proper client/server tcp library, you should be fine |
| 23:21.34 | Kobaz | connect... read... login... read... send command... read |
| 23:22.03 | trentcreek | okay, let me look up the Python telnet library commands |
| 23:22.07 | Kobaz | trentcreek: we could have reached this conclusion an hour ago |
| 23:22.13 | Kobaz | don't use telnet |
| 23:22.17 | Kobaz | use a client/server tcp library |
| 23:23.02 | Get_The_Fish | ok, well, the only res_timing_* that I see is res_timing_dahdi.so in /usr/lib/asterisk |
| 23:23.17 | Kobaz | it's timerfd is what i'm thinking of |
| 23:23.26 | Kobaz | search for *time* |
| 23:23.48 | Get_The_Fish | nada |
| 23:24.02 | Kobaz | and this is 1.6.2 ? |
| 23:24.10 | Get_The_Fish | 1.6.2.5 |
| 23:24.10 | trentcreek | Kobaz: why tcp lib over telnet? |
| 23:24.29 | Kobaz | trentcreek: because telnet uses extra control codes that a standard tcp stream does not |
| 23:25.12 | Get_The_Fish | accoding to timing.txt, there should be a res_timing_timerfd.so if that module was compiled with asterisk, which I specifically didnt include in the menuselect |
| 23:25.14 | Kobaz | even though a console telnet client will work to connect, it's not the proper thing to use |
| 23:25.20 | Kobaz | Get_The_Fish: that'll do it |
| 23:25.41 | Kobaz | so you're still seeing the warning/error in your console with internal timing turned off? if so... this is a bug... issues.asterisk.org |
| 23:26.16 | Get_The_Fish | disabled internal timing in asterisk.conf |
| 23:26.29 | Get_The_Fish | trying again |
| 23:28.36 | trentcreek | Kobaz: So then you know of some examples online? All I see are using telnet |
| 23:28.57 | Kobaz | trentcreek: #python... i'm a c/perl guy |
| 23:29.22 | trentcreek | or PHP |
| 23:30.09 | Kobaz | http://us3.php.net/manual/en/ref.sockets.php |
| 23:32.38 | Get_The_Fish | yep, still see it.... I even set internal_timing= no in asterisk.conf.... and I know that it's reading that section of the file as well. I suppose that I will post this to the mailing list first... I hate submitting bug reports where the only bug is user error |
| 23:33.28 | Kobaz | sure |
| 23:35.57 | *** join/#asterisk GaldaX (~Claudio@190.162.138.26) |
| 23:36.38 | Get_The_Fish | well thanks anyway.... I'm sure that this isnt that big of a deal, it's just spamming my debug log, and perhaps I am a little OCD |
| 23:37.21 | Get_The_Fish | I still cant get followme to work. I think that it has to do with the same bug for attended transfers, issue #16816 |
| 23:37.54 | trentcreek | Kobaz: thanks a LOT for the assist |
| 23:38.07 | Kobaz | trentcreek: your welcome |
| 23:38.19 | Get_The_Fish | it appears that some magic that I dont understand completely isnt happening a local channel when the application that is using it depends on the answer state, ie attended transfers or followme |
| 23:38.25 | Kobaz | trentcreek: next time paste your outputs... there's lots of things that you may not think are useful... but are glaring to us |
| 23:38.45 | trentcreek | yes, thanks for the adviced |
| 23:39.08 | Kobaz | now that'll be six goats, a rack of lamb, and a cow pelase |
| 23:39.31 | Kobaz | er... pleast |
| 23:39.35 | Kobaz | sdfsadfsdf |
| 23:39.38 | Get_The_Fish | lol |
| 23:39.41 | Kobaz | i give up |
| 23:40.39 | Kobaz | typing 80wpm is only good if you can hit the proper keys |
| 23:41.20 | TJNII | I can type 150gspm (gibberish strings per minute) |
| 23:41.29 | Kobaz | sounds good |
| 23:44.15 | trentcreek | yumy |
| 23:46.43 | Get_The_Fish | cant wait for asterisk 1.8 |
| 23:47.00 | Get_The_Fish | LTS on a version that has native support for LDAP, YAY |
| 23:48.35 | Kobaz | well |
| 23:48.38 | Kobaz | 1.8 will be out |
| 23:48.46 | Kobaz | and then it'll be another year before it's nice and stable |
| 23:48.48 | Get_The_Fish | "when it's ready" I know |
| 23:48.48 | Kobaz | and then you can be happy |
| 23:49.31 | Kobaz | i wish 1.6 was the lts along with 1.8... 1.4 is pretty solid already |
| 23:50.36 | Kobaz | i have a box that's awww it crashed |
| 23:50.41 | Kobaz | it was up for 300+ days |
| 23:50.49 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:50.54 | Kobaz | now it's got |
| 23:50.58 | Kobaz | oh wait... that's weeks not days |
| 23:51.05 | Kobaz | System uptime: 43 weeks, 3 days, 5 hours, 7 minutes, 49 seconds |
| 23:51.08 | Kobaz | so it didn't crash, yay |
| 23:51.19 | TJNII | That sucks. I have a box that had been up for months, crashed an hour ago. No idea why. |
| 23:51.24 | Get_The_Fish | I think that 1.8 will be pretty solid actually... the community has gone to a more test driven development model |
| 23:51.32 | Kobaz | TJNII: no i tricked myself... it didn't die |
| 23:51.36 | Kobaz | it's still running good |
| 23:51.39 | Get_The_Fish | yeah, my 1.4 instance has been up for over a year |
| 23:51.56 | Kobaz | Get_The_Fish: yes, and that makes me very happy |
| 23:52.03 | Get_The_Fish | exactly |
| 23:52.16 | leifmadsen | Get_The_Fish: pretty sure LDAP exists in the current 1.6 stuff too |
| 23:52.17 | Kobaz | Get_The_Fish: i have a bunch of unit testing of my own.. to make sure that each asterisk release behaves as my apps expect it to |
| 23:52.32 | Kobaz | i've found bunches of regressions with my own test suites |
| 23:52.32 | Get_The_Fish | leif, it does, and I am using it happily :) |
| 23:52.55 | Get_The_Fish | I wanted to get Asterisk Business Edition and the supported goodness it has with 1.8 |
| 23:53.01 | leifmadsen | Kobaz: once I learn how to use the external testing framework I'll be pinging you for some tests :) |
| 23:53.14 | leifmadsen | ya that won't be for a while |
| 23:53.15 | Kobaz | okay |
| 23:53.20 | Kobaz | my stuff is all in perl |
| 23:53.25 | Kobaz | i do application level testing |
| 23:53.43 | Kobaz | like... i originate from this local channel to that local channel... and well what do you know... it didn't pick up |
| 23:53.46 | Kobaz | heh |
| 23:53.50 | Get_The_Fish | I know, but it doesnt mean I am not excited!! (I fall into the "dog" category, not the "cat") |
| 23:54.05 | *** join/#asterisk Flashtek (~neil@flashtek-uk.com) |
| 23:54.12 | *** part/#asterisk Flashtek (~neil@flashtek-uk.com) |
| 23:54.29 | Kobaz | i have a really sexy testing framework... i can test almost everything |
| 23:54.43 | Kobaz | i'm missing stuff like sip/iax call hold, and some other things |
| 23:54.49 | *** join/#asterisk manxpower (~ewieling@68.sub-75-235-28.myvzw.com) |
| 23:54.55 | Kobaz | i'll probably have to add some stuff to chan_sip in order to do that |
| 23:55.13 | Get_The_Fish | Kobaz, are you using PERL and SIPp to do that? |
| 23:55.24 | Kobaz | no... sipp isn't good for what i'm trying to test |
| 23:55.29 | Get_The_Fish | ah ok |
| 23:55.57 | Get_The_Fish | I am just starting to test call flows with SIPp, works pretty good |
| 23:56.02 | Kobaz | i'm testing call flow logic and asterisk core functionality/behavior (ie: did the call i just make... actually make it through... did it hit the right queue... did asterisk crash?) |
| 23:56.47 | Kobaz | i have a list of about 30 items for things i need to add for testing if asterisk crashed |
| 23:57.07 | Get_The_Fish | ok, yeah, thats pretty hot |
| 23:57.28 | Get_The_Fish | perhaps that would be helpful to the community, what do you think Lief? |
| 23:57.29 | Kobaz | i have some boxes that handle 100k+ calls a day... i really don't trust any new asterisk release until i run my army of tests on it |
| 23:58.13 | Kobaz | sure it's helpful, but we might turn our testing suite into a product |
| 23:58.34 | Kobaz | i'm not exactly sure what we're gonna do with it at the moment though |
| 23:58.41 | Get_The_Fish | well, depending on how much you are talking about there, I'd buy it |
| 23:58.53 | leifmadsen | Kobaz: I'd like to see what you're doing, because we're building an external testing framework too (in additional to the unit testing stuff) |
| 23:59.11 | Kobaz | sure i can draw you some diagrams and stuff |
| 23:59.24 | Get_The_Fish | would be quite helpful in vetting a new release I would think. |
| 23:59.43 | Kobaz | yeah... before i started writing tests, it was playing russian roulette with every upgrade |
| 23:59.55 | leifmadsen | awesome, anything you can provide to help out the better. I've got this software that will compare audio files too, so I might be using that to test call quality and make sure there is audio where it should be |