| 00:14.26 | salz212 | I am unable to connect by "telnet localhost 5038" even though I have allowed localhost in manager.conf .. what could be the problem? |
| 00:19.02 | LemensTS | are you connecting locally |
| 00:19.17 | salz212 | yes |
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| 00:25.38 | salz212 | got that sorted. |
| 00:32.52 | salz212 | how do I make a socket with 5038, it says this address is already in use, and when I check it is being used by asterisk.. so dont I have any other mean? other than telnet? |
| 00:34.52 | p3nguin | If you do not want asterisk to listen on 5038, disable that part. Then you can do whatever you want to create a listening process on that unused port. |
| 00:34.56 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:36.37 | salz212 | after disabling the port, Am I suppose to get the response of any ami command? |
| 00:37.00 | p3nguin | No. If you turn off AMI, why would AMI still be working? |
| 00:37.57 | salz212 | lol .. but I want that to work.. I need to communicate with AMI with a script within localhost |
| 00:38.10 | salz212 | I do not want to use telnet in script. |
| 00:38.44 | p3nguin | If you want to use AMI, enable it and stop complaining that it is working. |
| 00:40.16 | salz212 | nah.. I just want to know who I am do simple socket communication with AMI |
| 00:40.23 | salz212 | how* |
| 00:40.36 | p3nguin | netcat or telnet |
| 00:40.56 | salz212 | so there is no other way. like port mapping/ forwarding etc? |
| 00:41.15 | salz212 | to replicate the port for other user? |
| 00:41.15 | p3nguin | You're not making any sense. |
| 00:42.25 | salz212 | it was just an idea. ... to have a socket established all the time at one end.. .. meanwhile AMI works... I don't even know it exists.. |
| 00:42.48 | salz212 | i guess I have to use telnet in a script.. |
| 00:42.55 | p3nguin | or netcat. |
| 00:43.32 | salz212 | yup thanks for guidance. |
| 00:44.35 | WIMPy | You don't want to do that in a shell script. |
| 00:48.08 | salz212 | I would most probably be using Perl |
| 00:50.09 | WIMPy | Which means no telnet or whatever. |
| 00:51.21 | salz212 | no there are libraries for telnet .. |
| 00:52.20 | WIMPy | Maybe you should find out what it's all about first. |
| 00:54.12 | salz212 | about AMI or perl? |
| 00:54.27 | WIMPy | And basics. |
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| 00:57.00 | salz212 | I know asterisk to some extent .but.. linux is a big fish and I am just 5 months old with asterisk so... yes I read .. try to read a lot.. but implementation makes me happy . |
| 00:59.31 | WIMPy | Telnet is not just the well known client. There's the proticoll as well, an you don't want that for anything else. |
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| 01:07.25 | salz212 | what do you suggest, what the better way and common practice.. for AMI communication |
| 01:07.59 | WIMPy | Just a socket. |
| 01:09.19 | salz212 | on a same localhost, how? |
| 01:09.40 | salz212 | just give me idea or name, i will search it. |
| 01:10.19 | WIMPy | That first question dosn't make ant sense to me. |
| 01:10.28 | WIMPy | And te keyword is "socket". |
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| 05:00.21 | p3nguin | When musiconhold does not play any music, what is there to check for troubleshooting it? |
| 05:03.48 | cstachris | p3nguin, musiconhold.conf and which directory your MoH should be in. |
| 05:04.07 | cstachris | then the format of the file vs what asterisk will play (e.g. wav, mp3, sln) |
| 05:04.27 | cstachris | permissions of the /var/lib/asterisk/ directories that hold the MoH files |
| 05:07.50 | p3nguin | I've been using 1.4 branch for a long time and I'm trying to migrate to 1.8.6.0. MoH is the only thing that I notice that isn't working for me. |
| 05:08.36 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 05:11.33 | p3nguin | It doesn't play mp3 files, an mp3 stream, nor the wav and ulaw files in the moh directory. |
| 05:11.48 | p3nguin | The verbose output shows that it runs, but there is no sound. |
| 05:19.53 | p3nguin | Debug doesn't show anything regarding moh. |
| 05:21.41 | p3nguin | module unload res_musiconhold.so, then module load res_musiconhold.so ... now it's working. |
| 05:22.12 | p3nguin | I reloaded it several times with moh reload and also module reload res_musiconhold.so, but never did unload and load. |
| 05:22.39 | cstachris | cool |
| 05:24.42 | p3nguin | Not quite... |
| 05:24.54 | p3nguin | Every class I use is playing the same sound file. |
| 05:25.31 | irroot | p3nguin hi there |
| 05:25.53 | p3nguin | hi |
| 05:25.58 | irroot | just popped in you got moh woes |
| 05:30.42 | p3nguin | I replaced my old config with the sample, and now one of the included sound files plays when I use default class. Now I'll try adding another class. |
| 05:40.45 | p3nguin | It seems like it's an mp3 problem. |
| 05:41.16 | p3nguin | mpg123 shows that moh files are being played, but the sound isn't making it out to the phone channels. |
| 05:41.24 | p3nguin | Falling back to wav only, for some reason. |
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| 05:56.16 | sshock | p3nguin: I had that problem before, |
| 05:56.39 | sshock | and it was because I didn't have an Answer() before calling MusicOnHold() |
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| 06:21.10 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:21.12 | schmidts | good morning |
| 06:21.59 | wdoekes2 | morning |
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| 06:37.49 | schmidts | does anyone of you knows inum.net? sounds like enum but worldwide ;) |
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| 07:05.44 | schmidts | stupid question i know, but a pointer has allways the size of an int, right? |
| 07:06.37 | schmidts | forget it, its better i use a void * instead of an int ;) |
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| 07:07.56 | *** join/#asterisk qakhan (~qakhan@182.185.192.71) |
| 07:08.16 | qakhan | hi all |
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| 07:09.08 | qakhan | i have 2 asterisk servers and both config with company directory |
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| 07:11.40 | qakhan | i want shared company directory on both servers, if someone call in server A caller can access both company directory and if someone call on Server B call can access both company directory. |
| 07:11.49 | phoenixmage | Hi guys, can anyone recommend a cheap FXO card? Preferably PCIe. Just playing around and dont want to spend $300+ on a digium |
| 07:11.59 | qakhan | can anybody help in this? |
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| 07:13.06 | schmidts | qakhan take a look at DUNDI thats maybe what you are looking for |
| 07:13.20 | qakhan | DUNDI? |
| 07:13.27 | qakhan | what about that? |
| 07:15.53 | qakhan | can you describe me? |
| 07:18.06 | schmidts | qakhan https://wiki.asterisk.org/wiki/display/AST/Introduction+to+DUNDi ;) |
| 07:18.55 | jkroon | schmidts, can DUNDi now be used for full directory services too? last I checked it was mostly distributing numbers only ... |
| 07:19.56 | schmidts | jkroon i thought thats what he was asking for ;) distributed numbers, sorry if i was wrong |
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| 07:21.48 | qakhan | so what i suppose to look at? |
| 07:23.24 | jkroon | schmidts, directory - ie, name => number lookups ? |
| 07:23.37 | jkroon | qakhan, you can possibly misuse DUNDi for that. |
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| 07:24.27 | jkroon | there is nothing stopping you from (a) hashing names to numbers for directory lookup purposes, or (b) use names as numbers ... not sure if either strategy will actually work properly but you can try. |
| 07:25.01 | schmidts | qakhan i guess it would be the easiest way to use your own database for this |
| 07:25.04 | schmidts | jkroon thx ;) |
| 07:25.14 | jkroon | I know I use DUNDi to locate specific usernames, so I'll do a DUNDILOOKUP() on *trunk*${username} which will let me know to which server in my cluster ${username} has registered to. |
| 07:25.47 | jkroon | then I transfer the call to that server which will then pass it to the appropriate peer ... :p |
| 07:26.34 | qakhan | schmidts plz tell me how can i use database for shared company directory |
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| 07:34.11 | kaldemar | qakhan: perhaps you should clarify what you want to do. |
| 07:34.31 | schmidts | qakhan for example stored in mysql, you can directly access your mysql db from your asterisk dialplan |
| 07:35.56 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:36.04 | qakhan | that mysql will be shared DB |
| 07:36.09 | qakhan | right? |
| 07:43.13 | schmidts | qakhan if you make it reachable from both servers you can use it as a shared db |
| 07:49.52 | phoenixmage | No takers on a cheap FXO card? |
| 07:50.07 | ChannelZ | define cheap |
| 07:50.15 | phoenixmage | Sub $100 |
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| 07:53.31 | ChannelZ | Linksys SPA3102 (not a card but ethernet) or maybe one of those cheap X100P things from fleabay |
| 07:53.37 | ChannelZ | though dunno how well they actually function |
| 07:54.40 | ChannelZ | http://www.voip-info.org/wiki/view/X100P+clone |
| 07:54.44 | ChannelZ | ~x100p |
| 07:54.44 | infobot | well, x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 07:54.46 | ChannelZ | FWIW |
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| 08:03.44 | kaldemar | TDM01B being a 1TDM410PLF or 1TDM410PELF. |
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| 08:07.34 | grEvenX | hi guys |
| 08:08.11 | grEvenX | anyone know of a way to tell asterisk to mark a certain voicemail message as read, even though the user hasn't read it through the VoicemailMain app? |
| 08:08.20 | *** part/#asterisk phoenixmage (~d370170@ncao.vtcif.telstra.com.au) |
| 08:08.34 | grEvenX | e.g. through the Manager |
| 08:08.48 | grEvenX | (I'm using file-based voicemail message storage) |
| 08:09.07 | irroot | grEvenX or the file sustem ?? in the folder there are new / cur folders like imap |
| 08:09.09 | grEvenX | haven't found a way, so my thought was that I would need to move the message in the filesystem and re-index/rename all the files |
| 08:09.36 | grEvenX | but then I'm not sure how the message-number index in asterisk works right now, how it is sorted |
| 08:09.57 | grEvenX | it would be easier to make use of asterisk's internal routine on it rather than coding that re-index part again |
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| 08:29.27 | aberrios | I'm getting a very strange issue where my *logs will stop on a Sunday at 00:00 and not resume until 04:00 that day |
| 08:33.00 | aberrios | scratch that |
| 08:33.28 | aberrios | but it is weird the full logs have been written in this order: full.41 > full.1 > full.42 |
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| 08:42.28 | *** join/#asterisk Beltechs (~Beltechs@rrcs-76-79-247-40.west.biz.rr.com) |
| 08:43.28 | Beltechs | hello Im usind the dahdi module to configure my pri but I still get ALARM Not configured. Do I have to edit some files manually? |
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| 08:50.07 | kaldemar | Beltechs: what exactly did you do until now? |
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| 08:55.09 | Beltechs | the card is installed I havent run commands on cli |
| 08:55.48 | Beltechs | I set in dahdi mod name wcte11xp |
| 08:56.38 | kaldemar | are you using a GUI to configure your system? |
| 08:56.45 | Beltechs | yes |
| 08:56.50 | Beltechs | freepbx |
| 08:57.06 | kaldemar | ask in #freepbx how to configure the card. |
| 08:57.35 | Beltechs | no response 3 hours |
| 08:57.41 | kaldemar | and yes, you must configure the card. loading a module is not enough. |
| 08:57.47 | Beltechs | its an asterisk module... |
| 08:58.45 | kaldemar | no it's not. wcte11xp is a DAHDI module and you're using freepbx to generate configuration for it. |
| 08:59.08 | Beltechs | made by asterisk... |
| 08:59.33 | kaldemar | does not mean anything |
| 08:59.35 | Beltechs | ah |
| 09:00.23 | Beltechs | I got this .... [root@sip ~]# dahdi_genconf /usr/sbin/dahdi_genconf: missing default group (termtype=UNKNOWN) |
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| 09:19.31 | Beltechs | kaldemar: My alarms are OK now but I still not receiving calls, I have inbound route and ext ready |
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| 09:24.40 | hetii | Q: what i should use under linux to have virtual printer that allow send fax to my hylafax server ? (something like winprinthylafax functionality) |
| 09:25.14 | irroot | hetii check the hylafax home page there many options listed |
| 09:25.23 | schmidts | hetii on the hylafax homepage is a well documented list of known working drivers for all Oses |
| 09:25.30 | schmidts | irroot :P |
| 09:25.32 | Korolev | Beltechs, dahdi_genconf seems to be broken |
| 09:25.53 | hetii | yep but i read them and i`m more confused then before ;( |
| 09:26.13 | Beltechs | should I look for another dahdi module? |
| 09:26.16 | Korolev | I dont do a lot of dahdi hardware, but happened to me a few days ago trying to install two quad spans T1 cards too |
| 09:26.30 | Beltechs | how u correct it? |
| 09:26.36 | Korolev | no, just copy a working system.conf and chan_dahdi.conf from someone |
| 09:26.44 | Beltechs | ouch |
| 09:27.00 | Korolev | and modfy it to meet what you need |
| 09:27.06 | schmidts | hetii just search for a driver which works on linux and try it out, its very easy normally |
| 09:27.15 | hetii | there are many howto`s but non of them said on clear way how it will works :) and for my purpose will be the best when i will have poscrtipt printer that ask me about desctination number after printing and thats it :> |
| 09:27.17 | Korolev | what card is it? |
| 09:27.31 | Beltechs | Digium Wildcard TE110P T1/E1 Card |
| 09:28.13 | Korolev | ah ok, I have 2 TE410P, let me get you my configs |
| 09:28.21 | Beltechs | kool |
| 09:28.23 | Korolev | just change the spans and the d-channels |
| 09:28.26 | Korolev | you'll be ok |
| 09:28.30 | Beltechs | 10 4 |
| 09:29.03 | schmidts | hetii which OS do you have? |
| 09:29.09 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.182.117) |
| 09:29.10 | hetii | ubuntu |
| 09:29.33 | hetii | 11.04 |
| 09:29.47 | schmidts | hetii try this one: http://wiki.xtronics.com/index.php/Faxfrontend its for debian but it should work for ubuntu too |
| 09:31.51 | devil_evoxxx | hi, someone use asterisk 1.8.6 interfaced in sip with a Quescom Q401 ? |
| 09:33.09 | hetii | ok i will try but need find valid download page of this project :( |
| 09:34.24 | Beltechs | do I have to uncomment from modules conf? |
| 09:34.32 | irroot | devil_evoxxx post a sip debug + verb output ?? |
| 09:34.34 | Beltechs | wct1xxp |
| 09:34.48 | Beltechs | # Digium TE110P: PCI single-port T1/E1/J1 |
| 09:34.56 | Beltechs | wct1xxp |
| 09:34.57 | Beltechs | ? |
| 09:36.49 | Korolev | Beltechs, here: http://pastebin.com/h6vY9GK5 |
| 09:38.36 | Korolev | run dahdi_cfg, it will load only the module you need |
| 09:39.09 | Korolev | just take out the extra channels and d-channels |
| 09:45.48 | Beltechs | dahdi_cfg = http://pastebin.com/98vt9vGW |
| 09:51.03 | devil_evoxxx | irroot: iv'e got a strange problem , with asterisk 1.4.x work fine, the incoming call from PRI (trought quescom) come in asterisk and i can see it on asterisk cli |
| 09:51.24 | irroot | devil_evoxxx yeah id like to see why |
| 09:51.42 | devil_evoxxx | with asterisk 1.8.6, the peer is up, but on incoming call nothing appear on asterisk 1.8.6 |
| 09:51.48 | irroot | need to run debug and see whats what |
| 09:53.03 | irroot | "sip set debug peer ...." |
| 09:53.18 | devil_evoxxx | actually i've got client using this machine ( 1.4) , |
| 09:53.27 | devil_evoxxx | i can do this, after 18 :( |
| 09:53.33 | devil_evoxxx | after 6 P.M. |
| 09:54.00 | devil_evoxxx | yesterday i do some sip dump |
| 09:54.27 | irroot | ok post it lets see |
| 09:55.16 | devil_evoxxx | and on the " To " parts of sip, is only the ip and not To:<123456@10.0.0.66>, but only To: <10.0.0.66) |
| 09:55.46 | devil_evoxxx | this appear in both version ( your svn trunk , and tha version i can download from asterisk.org) |
| 09:56.24 | devil_evoxxx | i'm really thinking to buy a digium pri card.. |
| 09:57.21 | irroot | devil_evoxxx this could be something in the dialplan too |
| 09:57.59 | irroot | the gateway sends the call we need to collect it in the dialplan |
| 09:58.13 | irroot | and get some idea why it does not route |
| 09:59.11 | devil_evoxxx | the dialplan is the same, migrating from 1.4 to 1.8 ( syntax replace) |
| 09:59.40 | devil_evoxxx | because i use the quescom gw for faxing , and for some call |
| 09:59.47 | irroot | devil_evoxxx yes but there may be something we need to look at |
| 10:00.46 | devil_evoxxx | i know..( the diagram flux is ast 1.4 -> ast 1.8 -> quescom ) |
| 10:01.20 | devil_evoxxx | irroot: thankyou for your support, as soon as possible i post a sip debug |
| 10:01.30 | devil_evoxxx | when there are no call trought this E1.. |
| 10:01.47 | devil_evoxxx | now, i'm collecting sip debug of 1.4 |
| 10:01.50 | devil_evoxxx | for comparing |
| 10:03.25 | irroot | great |
| 10:03.55 | devil_evoxxx | at 13 pm i switch the machine and i post a debug :) |
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| 10:04.00 | devil_evoxxx | thankyu again |
| 10:04.44 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 10:05.57 | irroot | devil_evoxxx no problem its a pleasure 12pm here now so all good |
| 10:14.34 | Beltechs | Korolev: that didnt work should I rid the dahdi module since is broken? |
| 10:15.18 | Korolev | its not the module, its the genconf that seems broken |
| 10:15.32 | Korolev | did you restart dahdi? |
| 10:15.49 | Beltechs | even the machine |
| 10:16.37 | Beltechs | when I asterisk -vvvvr I dont see any activity when the call comes in |
| 10:16.52 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:24.35 | Korolev | what does dahdi_tool say? |
| 10:25.43 | Beltechs | http://pastebin.com/A4sMWa5V |
| 10:25.58 | Beltechs | its unconfigured again |
| 10:27.31 | Beltechs | dahdi_scan http://pastebin.com/Ybda2337 |
| 10:27.54 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:36.25 | *** join/#asterisk mintos (~mvaliyav@117.206.23.8) |
| 10:42.55 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 10:51.39 | devil_evoxxx | irroot: i' think that cloning ast machine for fax test ..the lic file is not valid because vcenter change the hw address |
| 10:51.43 | devil_evoxxx | of ethernet interface |
| 10:52.12 | devil_evoxxx | the lic file for res_fax_digium.so |
| 10:52.51 | irroot | bummer |
| 10:53.15 | devil_evoxxx | i've manually changed :) now its working.. |
| 10:54.47 | devil_evoxxx | now i make a sip debug on old machine |
| 10:54.53 | devil_evoxxx | and next, swap into ast 1.8 |
| 10:59.31 | devil_evoxxx | irroot: this is the sip debug of working ast 1.4. The call diagram is (office pbx (ast 1.4) -> main switch ( ast 1.4) -> border_fax_machine(ast 1.4)-> quescom gw (interfaced with a pri e1) |
| 10:59.35 | devil_evoxxx | http://pastebin.com/d1v3YmmK |
| 10:59.49 | devil_evoxxx | what i'm trying to change is border_fax_machine |
| 10:59.52 | devil_evoxxx | from 1.4 to 1.8 |
| 10:59.56 | devil_evoxxx | now i swap and debug.. |
| 11:00.31 | irroot | yeah when you ready |
| 11:03.42 | devil_evoxxx | irroot: |
| 11:03.46 | devil_evoxxx | i'm ready |
| 11:03.49 | devil_evoxxx | this is the sip debug |
| 11:03.49 | devil_evoxxx | of |
| 11:03.54 | devil_evoxxx | non working machine |
| 11:03.58 | devil_evoxxx | http://pastebin.com/hBVnmRV7 |
| 11:04.10 | irroot | also we can see "core set debug 1" in trace |
| 11:04.17 | irroot | may give hints |
| 11:04.59 | devil_evoxxx | ok |
| 11:06.16 | devil_evoxxx | http://pastebin.com/c4NN3DmS |
| 11:06.20 | devil_evoxxx | here the new one |
| 11:06.33 | irroot | problem seems that the quescom is rejecting it |
| 11:07.28 | devil_evoxxx | asterisk says "invalid sip message - no callid" |
| 11:07.30 | devil_evoxxx | why?!!?!?!? |
| 11:09.19 | devil_evoxxx | irroot: the stranghe is that in ast 1.4 quescom does not reject |
| 11:11.27 | irroot | one uses rport one does not |
| 11:12.04 | devil_evoxxx | sorry? |
| 11:13.31 | Beltechs | dahdi restart http://pastebin.com/8na0pbD9 give and error can someone please help trouble shoot/ Thanks |
| 11:15.12 | irroot | INVITE sip:3477365087@94.230.64.45 SIP/2.0 |
| 11:15.14 | irroot | Via: SIP/2.0/UDP 94.230.64.46:5060;branch=z9hG4bK750048e4 |
| 11:15.18 | irroot | INVITE sip:3477365087@94.230.64.45 SIP/2.0 |
| 11:15.20 | irroot | Via: SIP/2.0/UDP 94.230.64.46:5060;branch=z9hG4bK70bb4884;rport |
| 11:15.22 | irroot | devil_evoxxx |
| 11:15.25 | devil_evoxxx | yes? |
| 11:15.45 | irroot | subtle diff |
| 11:15.54 | devil_evoxxx | oh shit.. |
| 11:16.43 | devil_evoxxx | it's a nat problem |
| 11:16.46 | devil_evoxxx | ...? |
| 11:16.49 | devil_evoxxx | right? |
| 11:17.58 | irroot | seems like it may be thats the only diff |
| 11:18.15 | irroot | the quescom rejects it may need rport ?? |
| 11:18.58 | devil_evoxxx | you think the right way is to enable rport on quesco |
| 11:19.01 | devil_evoxxx | quescom |
| 11:20.16 | irroot | yeah try it put the rport option on for that peer |
| 11:21.29 | devil_evoxxx | and, on asterisk peer,i've to modify something? |
| 11:21.53 | devil_evoxxx | i try to search rport options on quescom..(quescom have a management interface non user friendly) |
| 11:22.27 | irroot | put nat=rport in the sip.conf for the quescom |
| 11:22.46 | devil_evoxxx | ok |
| 11:23.03 | irroot | nat = force_rport |
| 11:23.15 | irroot | or nat = yes |
| 11:23.34 | devil_evoxxx | ok |
| 11:23.40 | devil_evoxxx | i'll try |
| 11:24.20 | devil_evoxxx | swapping machine.. |
| 11:25.08 | Beltechs | Is there any one who can help me set up TE110P ? |
| 11:26.38 | devil_evoxxx | irroot: nat=rport same problem |
| 11:26.43 | devil_evoxxx | now try with force_port |
| 11:26.56 | irroot | its force_rport not rport sorry |
| 11:27.06 | devil_evoxxx | reload |
| 11:27.11 | devil_evoxxx | ops, wrong shell |
| 11:27.13 | devil_evoxxx | ok |
| 11:27.23 | schmidts | :D |
| 11:28.13 | devil_evoxxx | irroot: , same problem |
| 11:28.32 | irroot | its got the rport showing now |
| 11:28.36 | devil_evoxxx | the outgoing call ( to quescom) |
| 11:28.38 | devil_evoxxx | ok |
| 11:28.39 | irroot | any logs on the quescom ? |
| 11:28.53 | devil_evoxxx | quescom have no verbose log.. |
| 11:28.56 | *** join/#asterisk stix__ (~stix@193.89.191.209) |
| 11:28.59 | devil_evoxxx | i'm rellly thinkg to elimiante it |
| 11:29.05 | devil_evoxxx | with quescom i've got only problem.. |
| 11:29.06 | irroot | firmware update ?? |
| 11:29.16 | devil_evoxxx | i've lost quescom access..due to warrany |
| 11:29.24 | devil_evoxxx | it's based on win2000!!!! |
| 11:29.27 | irroot | ah ok |
| 11:29.38 | irroot | petrol and matches ... |
| 11:31.41 | devil_evoxxx | irroot: this is a new debug with nat=force_port |
| 11:31.50 | devil_evoxxx | http://pastebin.com/Smf65miF |
| 11:32.06 | irroot | force_rport <- not force_port |
| 11:32.16 | devil_evoxxx | yes |
| 11:32.32 | devil_evoxxx | i've made a mistake writing here.. |
| 11:32.39 | devil_evoxxx | i've placed force_rport |
| 11:32.40 | irroot | its showing |
| 11:32.42 | Beltechs | an someone help configre TE110P? |
| 11:33.33 | irroot | devil_evoxxx you here in ZA ?? see its a voda num you calling |
| 11:35.33 | devil_evoxxx | i'm in Italy irroot , you? |
| 11:36.09 | irroot | devil_evoxxx ah ok im in south africa ... without an idea of why a 5XX error is coming back its impossible to know what it is |
| 11:37.10 | devil_evoxxx | quescom..if i can have log on quescom, i can debug |
| 11:37.18 | devil_evoxxx | but no log, no work.. |
| 11:37.28 | devil_evoxxx | i'll buy a pri card now.. |
| 11:37.35 | devil_evoxxx | and a new 1U server with raisercard |
| 11:37.42 | devil_evoxxx | eliminating the problem on the root.. |
| 11:37.59 | irroot | yeah we use paton gateways here and they work fine |
| 11:38.02 | *** join/#asterisk ocx (5ebb3951@gateway/web/freenode/ip.94.187.57.81) |
| 11:38.16 | irroot | but prefer card |
| 11:38.25 | devil_evoxxx | i use patton gw for isdn, always work in any conditions! |
| 11:38.46 | devil_evoxxx | the quescom was bought 3 or 4 years ago from another people.. |
| 11:38.46 | ocx | if i want to connect my traditional pbx to the asterisk (voip gateway) do i need to connect the traditional pbx via an fxo port into the asterisk? or fxs |
| 11:40.34 | devil_evoxxx | irroot: you prefer to use PciE card or Pci? |
| 11:40.44 | devil_evoxxx | actually i've used only pci BRI card |
| 11:40.46 | devil_evoxxx | never PCIe |
| 11:41.16 | *** join/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 11:41.31 | *** part/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 11:42.18 | irroot | ocx is there way to put pri in trad pbx and use asterisk as gateway to it |
| 11:42.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:42.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:42.47 | irroot | devil_evoxxx cheapest :P |
| 11:43.09 | irroot | devil_evoxxx given choice pciE |
| 11:43.46 | ocx | no |
| 11:44.03 | ocx | i want to connect external trunk of trad pbx into asterisk |
| 11:44.07 | irroot | ocx pri then bri then fxs/o |
| 11:44.12 | *** join/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 11:45.00 | ocx | what do you mean by pri and bri |
| 11:49.05 | *** join/#asterisk jkroon (~jkroon@dsl-241-239-182.telkomadsl.co.za) |
| 11:49.19 | irroot | ocx ISDN line types |
| 11:49.26 | jkroon | hey, on chan_dahdi, it seems to take forever for fxo channels to realize there is an inbound call. |
| 11:49.48 | jkroon | it'll say starting simple switch but then only about three rings later will it enter the dialplan - is there anything I can do to speed this up? |
| 11:50.08 | ocx | i am confused with understanding the fxs and fxo ports |
| 11:50.10 | *** join/#asterisk bmg505 (~leon@196-209-71-70.dynamic.isadsl.co.za) |
| 11:50.29 | ocx | the port is named to what it connects to but not what it does |
| 11:50.30 | ocx | :/ |
| 11:51.42 | devil_evoxxx | ahahahha :) same for me |
| 11:52.54 | ocx | FXS ports provide power |
| 11:53.02 | ocx | FXO receives signal |
| 11:53.06 | ocx | this is what i understood |
| 11:53.23 | ocx | so you connect analog phones to FXS ports |
| 11:53.46 | ocx | and Central offices to FXO since they will provide the power to the lines coming over asterik |
| 11:53.54 | ocx | correct me if i am wrong |
| 11:54.54 | catphish | p3nguin: you around? |
| 11:54.59 | irroot | jkroon helkom is your friend .... |
| 11:55.25 | irroot | jkroon check your caller id ring collect and settings |
| 11:56.31 | ocx | irroot: in my case where my traditional pbx connects to the asterisk for voip services, i need to connect the pbx to the asterisk via FXO port right? |
| 11:57.14 | irroot | Phone Handset <-> FXS <-> Asterisk <-> FXO <-> Telco |
| 11:57.20 | jkroon | irroot, you mean to say those things makes it take even longer? |
| 11:58.11 | irroot | jkroon it appears too be the cause i have no emperical debug info on hand |
| 11:58.25 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:59.01 | catphish | i imagine your traditional PBX would want to connect to a FXS port if its making and receiving calls to IP-land |
| 11:59.09 | ocx | irroot: Phone Handset <-> analog PBX <-> FXO <-> Asterisk <-> VOIP service provider |
| 11:59.14 | ocx | is that connect? |
| 11:59.27 | catphish | no, i'd use FXS there |
| 11:59.43 | catphish | FXS simulates a telco |
| 11:59.44 | Beltechs | Can some on help with a TE110P Config? |
| 11:59.53 | irroot | ^^^ yes you need to be the telco so FXS to the PBX |
| 11:59.55 | catphish | FXO simulates a telephone |
| 12:00.39 | ocx | Phone Handset <-> analog PBX <-> FXS <-> Asterisk <-> VOIP service provider |
| 12:00.55 | ocx | in this case i can use phone handset to dial into ovip |
| 12:01.03 | ocx | and also accept incoming calls from voip gateway into the handsets right? |
| 12:01.07 | ocx | 2 ways |
| 12:01.23 | ocx | this should be done in the asterisk dialplan |
| 12:01.52 | catphish | i have a problem where incoming calls Dial(Local/xxx) a local extension, but if the call is subsequently forwarded to another number, billsec on the second leg is always zero |
| 12:02.00 | ocx | but the physical connect above is only needed for allowing outbound calls to the internet and accepting incoming calls from the internet |
| 12:02.20 | jkroon | irroot, what we need is for "updating cli" :p ie, get the call into the dialplan and just update CALLERID(num) at a later stage :p |
| 12:02.22 | catphish | ocx: correct |
| 12:02.38 | ocx | thanks |
| 12:02.47 | catphish | with an FXS port, your asterisk bill become an analog telco |
| 12:02.56 | catphish | plug in your PBX like you would into a real phone line |
| 12:03.07 | catphish | it will send and receive calls in the same way |
| 12:03.18 | catphish | you will need an appropriate dialplan of course |
| 12:05.00 | ocx | so telephone is equal to an FXO device and i need an FXS port to connect that telephon to asterisk |
| 12:05.18 | ocx | and telco is an FXS device so i use an FXO device to connect to it |
| 12:05.53 | catphish | yes |
| 12:17.52 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:17.53 | catphish | an the solution to my issue is /n |
| 12:17.56 | catphish | i understand why now |
| 12:33.36 | *** join/#asterisk orioni (~chatzilla@46.183.121.39) |
| 12:33.58 | orioni | hi there , my calls drop after 1 sec using g729 but i dont have any problem using alaw.ulaw |
| 12:34.17 | orioni | im using the precompiled modules and the load/unload is OK |
| 12:34.23 | catphish | proposed solution: use alaw/ulaw |
| 12:34.37 | catphish | :) |
| 12:34.53 | orioni | :) |
| 12:35.10 | orioni | my gateway provider says it will allow only g729 inf the future |
| 12:35.46 | orioni | so it will drop the a/ulaw in the future |
| 12:35.53 | catphish | that's sucky :( |
| 12:36.02 | catphish | any errors worth pasting? |
| 12:36.04 | catphish | bbl |
| 12:36.07 | orioni | nothing at all |
| 12:36.10 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:36.53 | orioni | on the console i get only : Spawn extension (PLANET, 0594060185, 3) exited non-zero on 'SIP/35542255553-00000008' |
| 12:37.07 | p3nguin | catphish: I am now. |
| 12:41.13 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 12:44.58 | devil_evoxxx | irroot: i think i've got a news.. |
| 12:45.00 | Faustov | is there a soft phone for windows you could recommend? |
| 12:45.10 | devil_evoxxx | use nat=never was added around June 29, 2004 to solve a problem where some SIP UAs can't handle the addition of support for "rport" in the header (see RFC3581 ) |
| 12:45.14 | Faustov | freeware if possible |
| 12:45.25 | irroot | devil_evoxxx cool you pregnanr ?? ooo ok not lol |
| 12:45.48 | irroot | devil_evoxxx it works ? |
| 12:45.57 | devil_evoxxx | i try now.. |
| 12:46.16 | *** join/#asterisk Fritz09 (~Fritz09@2.211.67.188) |
| 12:46.18 | irroot | devil_evoxxx in the 1.4 trace you have rport ?? |
| 12:47.36 | devil_evoxxx | is set to nat=no |
| 12:47.55 | devil_evoxxx | maybe in new 1.8.6 nat = no is different of nat=never? something here? |
| 12:50.26 | p3nguin | faustov: zoiper classic |
| 12:50.51 | Faustov | p3nguin: thx! |
| 12:52.16 | devil_evoxxx | irroot: ..with nat = never |
| 12:52.21 | devil_evoxxx | worl-- |
| 12:52.24 | devil_evoxxx | work... |
| 12:52.32 | irroot | lovely |
| 12:53.01 | devil_evoxxx | ..it's a bogus? |
| 12:53.49 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zgvxtqggqunbuabw) |
| 12:56.27 | ocx | analog phone <> analog PBX <> FXS port Asterisk <ASTERISK> FXO PORT <> TELCO |
| 12:56.29 | ocx | correct? |
| 12:56.49 | ocx | do i need echo cancellation on the pci board in this case? |
| 12:56.59 | ocx | if 1 port fxs and 1 fxo used on the same pci |
| 12:59.46 | devil_evoxxx | god..save me |
| 12:59.53 | devil_evoxxx | now is making same problem |
| 13:01.57 | devil_evoxxx | for the first two call does not make problem |
| 13:01.58 | devil_evoxxx | after that |
| 13:02.54 | devil_evoxxx | the problems come again |
| 13:05.21 | *** join/#asterisk serafie (~erin@nat/digium/x-ctwagbpaogvkystn) |
| 13:09.11 | catphish | p3nguin: no worries, i figured out what i needed |
| 13:09.17 | catphish | i just knew that you knew the answer |
| 13:09.19 | *** join/#asterisk theridge (~kvirc@adsl-108-198-97-47.dsl.chi2ca.sbcglobal.net) |
| 13:10.15 | *** join/#asterisk bchia (~Adium@nat/digium/x-jdgvbqciewknjwdj) |
| 13:17.17 | p3nguin | Glad I could help. :) |
| 13:17.51 | p3nguin | Now maybe you can tell me why moh can't play mp3s. |
| 13:22.35 | ocx | does asterisk with 1TDM410BF support incoming caller ID from telco? |
| 13:22.47 | ocx | or another hardware chip is required? |
| 13:28.13 | Chainsaw | ocx: If the telco sends the appropriate signalling, you can generally get it to work. |
| 13:28.37 | Chainsaw | ocx: It is vital that you set the card to the correct country though, as there are very different standards for this in different parts of the world. |
| 13:29.05 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:29.05 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:29.07 | ocx | Chainsaw: to set the card to the correct country, i only need to do some software configuration right? no hardware |
| 13:29.15 | kaldemar | ocx: an FXS interface can't receive caller id and is not used for connecting to a telco, but for connecting a phone. you need a card with an FXO interface. |
| 13:29.23 | Chainsaw | ocx: Just software config, indeed. DAHDI-specific. |
| 13:29.29 | ocx | great |
| 13:29.30 | ocx | thanks |
| 13:29.37 | Chainsaw | ocx: But note kaldemar's comment. |
| 13:29.53 | kaldemar | and 1TDM410BF is a card with a single FXS. |
| 13:30.10 | Chainsaw | kaldemar: Right. I just latched on to TDM410. |
| 13:30.13 | ocx | well i received this quotation |
| 13:30.14 | ocx | 1TDM410BF with 4 port modular analog PCI 3.3/5.0V card |
| 13:30.24 | ocx | and each of these ports can be an FXS or FXO |
| 13:30.45 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 13:30.47 | Chainsaw | ocx: Assuming you slap an FXO module on, it can work. But as kaldemar says, FXS is the wrong direction. That's to attach analog phones of your own, and play telco. |
| 13:31.01 | Katty | yarrrr it bes talk like a pirate day |
| 13:31.13 | Katty | we'll be 3 sheets to the wind tnight!!! |
| 13:31.34 | ocx | yea i would use the FXO port to receive caller id from telco |
| 13:31.44 | ocx | but works on the 1TDM410BF no? |
| 13:31.52 | ocx | [16:29] <kaldemar> and 1TDM410BF is a card with a single FXS. |
| 13:31.56 | malcolmd | ocx: each of the ports can be populated by either an fxs or fxo module; you need the right module for the task you're trying to accomplish with whatever port you're wanting to populate |
| 13:32.18 | ocx | ok great |
| 13:32.24 | malcolmd | if you're talking to the telco, you want an FXO interface on your side. if you're talking to a telephone set, you'll use an FXS interface to drive that telephone set. |
| 13:32.29 | ocx | that applies on the 1TDM410BF right? |
| 13:32.36 | malcolmd | so, if you want a single connection to the telco, you want 1TDM401BF |
| 13:32.36 | kaldemar | ocx: 1TDM410BF is a name for a TDM410 with a single FXS module. so no, you can't connect to a telco with that. |
| 13:32.55 | malcolmd | if you want a single connection to an analog telephone set that you're driving inside your office (and you're not connected to the telco) then you want 1TDM410BF |
| 13:33.12 | kaldemar | 1TDM410PELF is a card with a single FXO module. |
| 13:33.27 | kaldemar | http://store.digium.com/telephony_card_selector.php |
| 13:33.36 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-awohxlvmyjosdavq) |
| 13:34.07 | ocx | malcolmd: i will use the 1TDM401BF with an FXO port to connect to telco and have caller id option |
| 13:34.14 | ocx | asuming my telco provides this service |
| 13:36.59 | *** join/#asterisk Imox (~Imox@p54BE108C.dip0.t-ipconnect.de) |
| 13:39.35 | Imox | hello, I use freepbx and I want disbale FOP how can I do this? when I disable the FOB in the settings but dont help |
| 13:44.54 | Imox | in the FreePBX System Status is the Op Panel Disabled but it works? whats that. I think its a bug |
| 13:45.16 | kaldemar | Imox: #freepbx will know better |
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| 13:48.28 | Katty | dear monday...please...piss off... |
| 13:48.39 | Chainsaw | Katty: I second that. |
| 13:48.46 | Katty | bangs head on desk a few dozen times |
| 13:48.47 | Chainsaw | Katty: Never liked Mondays. Today is no exception. |
| 13:49.05 | Katty | always those pesky mondays runing my happy weekend! |
| 13:49.16 | Chainsaw | Katty: At least you had a weekend :) |
| 13:49.27 | Chainsaw | Katty: I worked from noon to midnight on both Saturday & Sunday. Worked from home, but still. |
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| 14:01.29 | catphish | p3nguin: my asterisk can play mp3s, but it doesn't i transcode everything to gsm and mormalize it |
| 14:01.46 | Chainsaw | 's hold music is an S3M |
| 14:02.03 | catphish | not sure why i make everything gsm |
| 14:02.15 | catphish | seems like a dumb idea since i don't actually use gsm for anything else |
| 14:06.07 | *** join/#asterisk master_of_master (~master_of@p57B542A8.dip.t-dialin.net) |
| 14:10.40 | malcolmd | might as well make everything slin, since asterisk's going to take it to that in many (conferencing) cases, anyway, for mixing purposes |
| 14:16.10 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 14:19.10 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:22.36 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:29.06 | *** join/#asterisk gxdssoft (~gxdssoft@190.236.37.136) |
| 14:30.16 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 14:32.38 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:33.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:34.58 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:39.09 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 14:40.32 | *** join/#asterisk Fritz09 (~Fritz09@pop1-2551.catv.wtnet.de) |
| 14:42.44 | r33dtard | question about iax2 registration refused |
| 14:44.04 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 14:45.16 | kaldemar | r33dtard: go ahead and ask |
| 14:45.23 | *** part/#asterisk Beltechs (~Beltechs@rrcs-76-79-247-40.west.biz.rr.com) |
| 14:46.01 | r33dtard | kaldemar: I had iax2 registration working...then I rebooted to apply some system patches |
| 14:46.05 | r33dtard | and now I get registration refused |
| 14:48.54 | r33dtard | where should I check...I have debug turned way up |
| 14:50.07 | kaldemar | iax2 set debug on |
| 14:50.40 | r33dtard | yep |
| 14:52.08 | kaldemar | what are you registering to? |
| 14:52.20 | r33dtard | myself |
| 14:52.34 | kaldemar | also see what the receiving side says. |
| 14:56.44 | r33dtard | also the user from the IAX client that is attempting to conect to the astrisk server is getting registration refused |
| 14:58.07 | kaldemar | feel free to pastebin debug output. |
| 14:58.10 | kaldemar | ~pb |
| 14:58.10 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:01.10 | r33dtard | http://pastebin.com/hSZUa5SV |
| 15:01.15 | r33dtard | was getting there :) |
| 15:06.00 | kaldemar | r33dtard: Peer 'warvox' is not dynamic |
| 15:06.13 | r33dtard | yeah I saw that |
| 15:06.17 | r33dtard | fixed it |
| 15:06.38 | r33dtard | do I need the calltokenoptional stuff? |
| 15:08.56 | kaldemar | r33dtard: only if the client does not support calltokens. |
| 15:09.03 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:09.26 | *** join/#asterisk gxdssoft (~gxdssoft@201.230.220.101) |
| 15:09.26 | r33dtard | ok I am back to normal |
| 15:09.40 | r33dtard | figured it out |
| 15:10.14 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:13.32 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
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| 15:34.15 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 15:35.52 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
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| 15:43.01 | *** join/#asterisk Yxa (~Yxa@bb116-14-79-5.singnet.com.sg) |
| 15:44.55 | Yxa | i'm using 1.8.6.0 when I try to Dial(SIP/249), which is an invalid extension, it continues to dial with a ringing tone with many WARNINGS in the CLI. is it a bug? |
| 15:46.37 | Qwell | Yxa: sort of. it's trying to dial 249 as an IP. |
| 15:46.54 | Yxa | huh? |
| 15:47.38 | Yxa | any workarounds? you want a screen dump? |
| 15:47.53 | Qwell | Don't dial invalid extensions? |
| 15:48.35 | kaldemar | it shouldn't give a ringing tone though. |
| 15:48.41 | Yxa | i thought dialling invalid extensions is supposed to give a busy tone. |
| 15:48.43 | Qwell | sure it should |
| 15:49.01 | Qwell | it isn't an invalid extension (it isn't an extension at all) |
| 15:51.19 | Yxa | is that what happens when the actual extension number is also the username? |
| 15:52.59 | Yxa | shit u're right. it's trying to dial an IP. chan_sip.c:3349 __sip_xmit: sip_xmit of 0x9e6cf30 (len 808) to 0.0.0.249:5060 returned -1: Invalid argument |
| 15:54.21 | *** join/#asterisk jhoppe (~jhoppe@68-188-9-110.static.stls.mo.charter.com) |
| 15:54.51 | *** part/#asterisk jhoppe (~jhoppe@68-188-9-110.static.stls.mo.charter.com) |
| 15:55.57 | Yxa | any way to force it to treat the 249 as an extension, not ip? |
| 15:56.23 | Qwell | again, it isn't an extension. |
| 15:56.27 | Qwell | you mean peer, perhaps? |
| 15:56.41 | Yxa | yes peer |
| 15:56.46 | Qwell | define that peer... |
| 15:57.29 | Yxa | they are all defined in sip.conf |
| 15:58.00 | Yxa | 249 isnt though, it's just an example of a invalid number |
| 15:58.15 | leifmadsen | so check if something is valid before you dial it |
| 15:58.17 | Qwell | obviously you can't dial a peer that doesn't exist... |
| 15:58.31 | leifmadsen | there are dialplan functions to check for that kind of thing |
| 15:58.37 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:02.11 | leifmadsen | ~devicenames |
| 16:02.11 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 16:02.54 | Qwell | malcolmd: ^^ heh |
| 16:04.00 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:04.33 | Yxa | if i cant change that, would going back to 1.6 help? |
| 16:04.44 | leifmadsen | I don't know.... does it? |
| 16:04.54 | leifmadsen | like I said -- you can check for the validity of a peer before you dial it |
| 16:05.01 | leifmadsen | 'core show function SIPPEER' |
| 16:12.53 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 16:16.36 | Yxa | leifmadsen i tried looking up docs but i'm still not sure how to use it |
| 16:16.54 | leifmadsen | Really |
| 16:17.16 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 16:17.20 | [sr] | howdy frinds |
| 16:17.20 | leifmadsen | GotoIf($[${ISNULL(${SIPPEER(${EXTEN})})}]?skip_dial) |
| 16:17.32 | [sr] | im back after 2 weeks in the us! |
| 16:18.15 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:20.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-vjrmnveuhjovutvj) |
| 16:21.07 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:23.55 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 16:26.09 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 16:29.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:30.12 | Yxa | leifmadsen nice i got it. thanks a lot |
| 16:34.24 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:34.42 | Naikrovek | [sr]: what did you think of the US |
| 16:34.48 | Naikrovek | also, who are you and where are you now? |
| 16:35.10 | [sr] | Naikrovek: the part "who am i" its dificult to answer, i'm me!!! |
| 16:35.20 | Naikrovek | heh okay |
| 16:35.45 | [sr] | the other part, from what i knew it's great, bettear to live there than here |
| 16:36.22 | drift- | linksys phone are they decent? |
| 16:36.34 | drift- | spa942 how much of pain in ass is it to configure get it working on asterisk |
| 16:36.35 | [sr] | i was with family so i saw the reallity, and not the "fancy" US when people go on a holiday trip |
| 16:38.21 | Yxa | drift- 942s are decent. very easy to configure. |
| 16:39.07 | Katty | OMGGUESSWHOISHIGHLYCAFFEINATED *boingboingboing* |
| 16:39.31 | Yxa | drift- the newer phones are the Cisco 500 series, which is the sucessor to the 900 series. |
| 16:40.14 | *** join/#asterisk MRH2_ (~chatzilla@62.49.242.3) |
| 16:40.36 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 16:41.28 | MRH2_ | Hi Am ai correct in thinking that in 1.8 I can add extra cdr fields and set their values in the dialplan |
| 16:45.22 | *** join/#asterisk Yxa (~Yxa@bb116-14-79-5.singnet.com.sg) |
| 16:48.40 | MRH2_ | that wasn;t available in 1.2 was it? |
| 16:48.55 | *** join/#asterisk irroot (~irroot@41.55.17.106) |
| 16:50.04 | malcolmd | Qwell: i see your humor :) |
| 16:53.35 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:57.21 | Katty | ahoy me hearties! avast ye! it's bucko day, you landlubbers! heave ho! weigh anchor and hoist the mizzen, we'll be three sheets to the wind with our rum a fizzin! |
| 17:01.40 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:03.39 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:13.35 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:14.23 | MRH2_ | and can i use custom fields with cdr_pgsql? |
| 17:17.31 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 17:17.33 | *** join/#asterisk cyford (~cyford@c-24-30-48-255.hsd1.ga.comcast.net) |
| 17:19.12 | irroot | MRH2_ not so sure i use adaptive odbc now might want to look at it |
| 17:19.36 | Katty | irroot: are you on the christmas card exchange yet? |
| 17:19.53 | MRH2_ | thanks |
| 17:19.59 | irroot | bah humbug .. sara keen though |
| 17:22.55 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 17:23.21 | irroot | devil_evoxxx hi there |
| 17:25.17 | Kobaz | is it possible to do a hint to someone on park |
| 17:25.27 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 17:25.31 | Kobaz | ie: have a blf button that shows if a parking lot has someone |
| 17:26.52 | *** join/#asterisk Fritz09 (~Fritz09@pop1-2551.catv.wtnet.de) |
| 17:26.58 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 17:27.31 | devil_evoxxx | hi irroot |
| 17:27.58 | devil_evoxxx | irroot: tomorrow an italian quescom reseller |
| 17:28.02 | Katty | Kobaz: you want to join the christmas card exchange? |
| 17:28.03 | devil_evoxxx | give me the new rom |
| 17:28.14 | devil_evoxxx | for quescom device.. |
| 17:28.15 | irroot | awesome !!!! |
| 17:28.47 | devil_evoxxx | i've spent two hour from the italian reseller where i spent 2000 euro / month in Ata/ digium card / patton |
| 17:28.49 | Katty | Qwell: ping |
| 17:28.54 | devil_evoxxx | if not help me.. |
| 17:28.57 | Katty | Qwell: i want you on the xmas card exhcnage |
| 17:29.13 | irroot | ah money talks |
| 17:29.23 | devil_evoxxx | always money talk!! |
| 17:30.20 | Katty | leifmadsen: and you need to be on the xmas card exchange list too! |
| 17:30.21 | irroot | hehe the folks from digium had a reseller here i have been almost always top customer but they had no support at first so they got me to teach there guys how to use the basics |
| 17:30.32 | leifmadsen | I don't need to be on anything! |
| 17:30.38 | Katty | yes you do. |
| 17:30.41 | Katty | i want to send you xmas cards |
| 17:30.56 | irroot | is thinking since its summer here may not do traditional xmas cards have some ideas |
| 17:30.59 | Katty | you can query it to me, rather than put it on the list, if you want |
| 17:35.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:40.28 | devil_evoxxx | irroot: i'm searching for a 1U Case for mounting the pri card ( bought today..) |
| 17:42.02 | irroot | devil_evoxxx not easy to find here |
| 17:42.06 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:42.18 | devil_evoxxx | i'm was thinking in supermicro.. |
| 17:43.00 | devil_evoxxx | something like this http://www.supermicro.com/products/system/1U/ |
| 17:48.00 | irroot | nice have no supply here that is reasonable price |
| 17:48.57 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qwsmxigqdelfynui) |
| 17:52.37 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 17:52.37 | *** mode/#asterisk [+o angler] by ChanServ |
| 17:55.28 | *** join/#asterisk x1user (~chatzilla@84.40.127.166) |
| 17:56.10 | x1user | Hi my asterisk doesnot detect dtmf tones, how to understand why? |
| 17:56.55 | irroot | x1user what channel type / setting you using have you debuged it |
| 17:57.32 | x1user | Sorry irrot i cant understand what do you mean? |
| 17:57.48 | citywok | x1user: sip, pri, etc.? |
| 17:58.08 | x1user | Mobile |
| 17:58.43 | irroot | x1user chan_mobile ?? |
| 18:01.35 | x1user | yep |
| 18:07.08 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:07.54 | *** join/#asterisk ocx (5ebb3951@gateway/web/freenode/ip.94.187.57.81) |
| 18:08.31 | ocx | are fxo/fxs ports dependent on the country they are used in or they are all the same and compatible with any analog telco? |
| 18:09.04 | drzed | hi there |
| 18:09.09 | *** join/#asterisk DanFromUK (DanFromUK@2.30.229.177) |
| 18:09.45 | drzed | little diaplan issue: i try to call "23", my phone tells me "adress incomplete", but "dialplan show 23@" show me the correct channel |
| 18:09.50 | DanFromUK | hi, if a user has set their own voicemail prompt, how can I remove alison's message that says "when you have finished recording you may hang up"? |
| 18:09.51 | irroot | ocx yes/no no they not manufactured for a country but do require country settingd |
| 18:10.16 | ocx | i mean from a hardware perspective |
| 18:10.25 | ocx | do i have to worry in which country i have to run that fxo/fxs port? |
| 18:10.29 | irroot | hardware is uniform |
| 18:10.51 | drzed | i'm calling from context "lokal" => "isdn-out" which is a included context; also "dialplan show 23@lokal" show me the right channel |
| 18:10.57 | irroot | you will need to pass the correct paramaters on load ocx |
| 18:11.12 | ocx | like ring tones and voltages etc? |
| 18:12.04 | irroot | yes ring voltage and line impedance |
| 18:17.51 | navaismo | drzed and what show the cli when you dial 23? |
| 18:18.20 | *** join/#asterisk grEvenX (~even@84-52-254.130.3p.ntebredband.no) |
| 18:20.47 | drzed | navaismo: with "core set verbose 4" just some notify stuff - nothing else |
| 18:21.21 | drzed | navaismo: http://nopaste.voric.com/paste.php?f=woxy8u |
| 18:24.05 | navaismo | the cli doesnt show the call flow |
| 18:24.23 | drzed | navaismo: how do i tell the cli to show the call flow? |
| 18:24.25 | ocx | from where can i get these parameters? |
| 18:24.31 | ocx | do i have to ask the telco provider? |
| 18:24.41 | ocx | i am sure they will be like wtf |
| 18:25.15 | *** join/#asterisk bchia (~Adium@nat/digium/x-dhnadsnrkvgoshzo) |
| 18:27.31 | *** join/#asterisk Fritz09 (~Fritz09@pop1-2551.catv.wtnet.de) |
| 18:28.46 | irroot | ocx what country you in ... and then find the settings i would check source not sure if there is a list ... digium support will help you also |
| 18:29.06 | p3nguin | Okay, so I'd really like to rectify this issue... I've obviously been running an asterisk 1.4 system for a while, and everything has been great with it, but I decided to come into this decade. I installed 1.8.6.0, and musiconhold refuses to play my mp3s that I used to use in my 1.4 system. lsof shows that mpg123 is in fact playing the files, but the sound never makes it to the phones. What gives? |
| 18:29.33 | irroot | p3nguin format_mp3 works fine |
| 18:29.59 | *** join/#asterisk Fritz09 (~Fritz09@pop1-2551.catv.wtnet.de) |
| 18:30.04 | irroot | need to build it from addons with a script from contrib/scripts |
| 18:30.28 | p3nguin | format_mp3.so MP3 format [Any rate but 8000hz mono is 0 |
| 18:30.32 | p3nguin | Check. |
| 18:30.53 | p3nguin | Next? |
| 18:31.15 | navaismo | drzed just dial again |
| 18:31.17 | irroot | ah we were using it this morning so fresh ... but alas |
| 18:31.18 | *** join/#asterisk Korolev (~Korolev@204.88.28.115) |
| 18:31.34 | p3nguin | I think I'll see if MP3Player() can play the files and stream. |
| 18:31.46 | irroot | p3nguin i use a raw file and hack of a script otherwise |
| 18:31.56 | irroot | ill pb for you |
| 18:32.08 | p3nguin | It will do no good for you to do that. |
| 18:32.19 | p3nguin | I just need to know what might prevent the sound from coming out the phones. |
| 18:33.31 | p3nguin | If MusicOnHold() is not able to play the files in my moh class, which operated fine under 1.4 versions, I have to assume something is wrong. |
| 18:33.38 | Katty | p3nguin: do you want on the xmas card exchange list? |
| 18:33.56 | drzed | navaismo: did so, nothing changed - even restarted * |
| 18:34.26 | navaismo | asterisk version |
| 18:34.35 | navaismo | ? |
| 18:34.39 | p3nguin | katty: Oh, I s'pose. |
| 18:35.01 | irroot | p3nguin cant remember what site it was to double check |
| 18:35.39 | p3nguin | When I call the moh class, mpg123 is actually playing the files, but the sound never reaches the phone. |
| 18:35.49 | *** join/#asterisk MiserySoft (~LND@host81-148-14-51.in-addr.btopenworld.com) |
| 18:36.44 | irroot | p3nguin all it should be doing is reading the output as audio frames |
| 18:37.11 | irroot | so odd one you got |
| 19:43.22 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:43.22 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 19:47.09 | p3nguin | leifmadsen: I found a new verbose level that adds something to the output. |
| 19:47.38 | leifmadsen | oh? |
| 19:47.41 | p3nguin | verbose 11 adds the CDR writes. |
| 19:47.44 | leifmadsen | ah |
| 19:47.52 | leifmadsen | so dub |
| 19:47.54 | leifmadsen | dumb... :) |
| 19:47.59 | p3nguin | haha |
| 19:48.17 | p3nguin | I couldn't find any other secret ones. |
| 19:57.22 | *** join/#asterisk Fritz09 (~Fritz09@pop1-2551.catv.wtnet.de) |
| 19:58.18 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:00.57 | *** join/#asterisk Felix1125 (~root@nat/digium/x-gxarfdrwznnuovkr) |
| 20:17.06 | nny | does asterisk 1.8 handle calls to a unregistered extension (or non existent) differently than prior? |
| 20:17.17 | nny | (using qualify=yes) |
| 20:17.27 | nny | and * 1.8.6 |
| 20:17.37 | p3nguin | If the extension does not exist, you'll get a temp fail. |
| 20:18.00 | p3nguin | aka congestion |
| 20:18.25 | nny | http://pastebin.com/dstLvRGQ |
| 20:18.36 | nny | seeing sip_xmit to odd addresses |
| 20:18.55 | nny | wait |
| 20:18.58 | nny | urgh |
| 20:19.08 | nny | er no |
| 20:19.16 | nny | that's right, it should congest |
| 20:19.31 | p3nguin | I'm not sure what that has to do with extensions being non-existent. |
| 20:19.39 | Qwell | s/extension/peer/ |
| 20:19.52 | p3nguin | That changes things. |
| 20:20.12 | nny | ahh yes |
| 20:20.13 | p3nguin | If the peer doesn't exist, the call will do whatever the extension tells it to do. |
| 20:20.14 | nny | sorry |
| 20:20.21 | nny | hmm welll.. crap |
| 20:21.19 | nny | i was using http://pastebin.com/qpe13qif to try and eliminate typing the exten => (person1,1,) etc for 100 extensions |
| 20:21.37 | nny | so it would use a pattern match, the extension range is not conistent though |
| 20:22.05 | nny | i could make a script, but was trying to make dialplans more clean/simple |
| 20:22.56 | nny | I was surprised to hear ringing to the non existent peer |
| 20:22.58 | p3nguin | What a weird dial plan. You've named the phones by the same name as the extension used to call it? |
| 20:23.03 | nny | the i extension is wrong |
| 20:23.16 | dr0ck | tsk tsk |
| 20:23.22 | p3nguin | And you want both callers and callees to be able to park and record calls? |
| 20:23.35 | nny | the dial statement is wrong |
| 20:23.41 | nny | er wrong options |
| 20:24.00 | nny | but alas, that's not what I asked about, as far as extension name/peer name, I have always seen it done that way |
| 20:24.46 | p3nguin | ~devicenames |
| 20:24.46 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 20:24.57 | nny | um |
| 20:25.10 | nny | ffs, did I ask about best p[ractices with naming conventions? |
| 20:25.32 | nny | afaik that is up to the implmentation, not some universal constant |
| 20:26.46 | p3nguin | The pattern should probably be _[256][57]XX, too, if you want people to be able to call 2500-2599, 2700-2799, 5500-5599, 5700-5799, 6500-6599, and 6700-6799. |
| 20:27.21 | Qwell | p3nguin: What, you mean ,,00 shouldn't be a mapped extension? |
| 20:27.24 | Qwell | nonsense! |
| 20:27.31 | nny | are you fucking kidding me? |
| 20:27.47 | p3nguin | I guess if you wanted to dial ,,00 - ,,99 |
| 20:28.19 | nny | you guys just gonna be dicks this afternoon? |
| 20:28.28 | p3nguin | But that'll be up to you, since it's obviously not my call. |
| 20:28.54 | *** mode/#asterisk [+q nny!*@*] by leifmadsen |
| 20:29.14 | leifmadsen | nny: play nice |
| 20:30.15 | Qwell | You're dialing a peer that doesn't exist, so it tries it as an IP address. |
| 20:30.33 | Qwell | If you think that's incorrect, open an issue on JIRA. Yelling at people here isn't going to fix anything. |
| 20:31.02 | Qwell | If you don't want advice from people in this channel, don't come here looking for it. |
| 20:31.32 | leifmadsen | ASTERISK-17663 |
| 20:31.48 | leifmadsen | oh right, no DocAwesome here |
| 20:33.36 | leifmadsen | anyways, if it can't find a peer, it tries to find it via the hostname. If you want to determine if the peer exists first, use something like the SIPPEER() function |
| 20:36.34 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:37.40 | *** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 20:37.55 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 20:38.08 | p3nguin | /: |
| 20:38.55 | leifmadsen | I muted him |
| 20:39.03 | leifmadsen | he couldn't talk anymore, so I put him on a timeout |
| 20:39.16 | p3nguin | I noticed that. |
| 20:39.17 | *** mode/#asterisk [-q nny!*@*] by leifmadsen |
| 20:39.19 | Qwell | Sometimes I wish I could take a timeout from #asterisk. |
| 20:39.32 | *** mode/#asterisk [-o+b Qwell *!*north@*pdpc/sponsor/digium/Qwell] by leifmadsen |
| 20:39.32 | *** kick/#asterisk [Qwell!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (TIMEOUT!) |
| 20:39.49 | *** mode/#asterisk [-b qwell!*@*] by leifmadsen |
| 20:39.55 | *** mode/#asterisk [-b *!*north@*pdpc/sponsor/digium/Qwell] by ChanServ |
| 20:40.11 | leifmadsen | heh |
| 20:40.11 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 20:40.11 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:40.19 | leifmadsen | Qwell: <3 |
| 20:41.45 | p3nguin | Any reason not to noload pbx_ael.so if I'll never use ael dial plan? |
| 20:42.46 | *** join/#asterisk nikolaidis (~Adium@c-24-63-100-26.hsd1.ma.comcast.net) |
| 20:43.21 | umay | AEL dial plans are a lot easier to read & debug, why not upgrade ? |
| 20:43.31 | p3nguin | It's not an upgrade. |
| 20:43.38 | *** join/#asterisk Felix1125 (~john@nat/digium/x-etgepbfdauuqusaa) |
| 20:44.05 | p3nguin | It's also not what I asked about. |
| 20:44.20 | navaismo | umay really ael it easier? Do you have a link to learn it? |
| 20:44.32 | p3nguin | AEL is pretty darn easy. |
| 20:44.44 | p3nguin | Just take a look at the sample file. |
| 20:44.56 | leifmadsen | p3nguin: no, I never computer AEL |
| 20:45.07 | leifmadsen | s/computer/compile/ |
| 20:45.09 | p3nguin | You never computer it? |
| 20:45.10 | p3nguin | oh |
| 20:45.12 | leifmadsen | so you can safely noload it |
| 20:45.17 | p3nguin | perfect. |
| 20:45.18 | leifmadsen | is not a fan of AEL |
| 20:45.36 | p3nguin | It doesn't seem to have the same capability as the standard .conf. |
| 20:45.36 | umay | looks cleaner than extensions.conf |
| 20:45.36 | leifmadsen | but that's personal opinion -- some people find it easier to use |
| 20:45.43 | p3nguin | But it is easy to write and read. |
| 20:45.54 | umay | even for non-coders |
| 20:46.02 | WIMPy | Things that can do less tend to be easier. |
| 20:46.04 | p3nguin | Since I'm not going to use it, I didn't see any reason to leave it loaded. |
| 20:46.06 | leifmadsen | depends on how used to regular asterisk dialplan you are ;) I find issues with it changing things on my every time I use it |
| 20:46.10 | p3nguin | wimpy: exactly |
| 20:46.36 | leifmadsen | anyways, my intention was not to start a debate on whether AEL is better or what... it's different, and I don't prefer it, and that's about it ;) |
| 20:46.53 | WIMPy | And I disabled it in menuselect as well. |
| 20:47.11 | p3nguin | And my intention was to find out if it was needed in the case where I'm not using it... and you told me with certainty. |
| 20:47.49 | p3nguin | I'm going to leave it compiled in, just in case I decide some day I want to drop a couple extensions in it. |
| 20:48.07 | leifmadsen | p3nguin: don't forget to also noload the interpreter |
| 20:48.09 | Qwell | p3nguin: way to commit |
| 20:48.42 | voxter | Baraghhtrhg? The patch i need for chan_unistim.so is in the review board, but is against trunk. Guess its time to check out and try running from svn! |
| 20:49.01 | p3nguin | leifmadsen: That's not by chance res_ael_share.so, is it? |
| 20:49.29 | p3nguin | That's the only other file I have matching \*ael\* |
| 20:50.05 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:50.41 | p3nguin | If that's not it, I don't know what module the interpreter would be in. |
| 20:50.50 | pabelanger | leifmadsen: stop being a dick |
| 20:50.57 | pabelanger | just saying |
| 20:51.00 | JasonL | has anyone seen RTP packets being transmitted multiple times (based on tcpdump trace) ? According to my trace, some calls are showing -50% packet, or more |
| 20:51.32 | leifmadsen | p3nguin: ya I think so |
| 20:51.37 | leifmadsen | pabelanger: sorry boss |
| 20:51.46 | pabelanger | right you are |
| 20:51.47 | p3nguin | If it blows up, I'll know that wasn't it! |
| 20:51.48 | pabelanger | :D |
| 20:52.01 | Katty | peeks in |
| 20:53.25 | Kobaz | what failure rate do you guys usually see with polycom phones |
| 20:53.49 | Qwell | tackles Katty |
| 20:54.04 | Kobaz | we have a deployment with about 80 phones, every 3 months or so we're replacing about 3 phones |
| 20:54.46 | citywok | Kobaz: we have 150 or so aastra phones and i've never had to RMA one... 2 years |
| 20:55.04 | Katty | wahhh! *oof* |
| 20:55.10 | citywok | that said, the cordless phones by aastra are horrrible (the CT line) |
| 20:55.11 | Kobaz | in about 2 years I think I've rma'd about 15 phones |
| 20:55.32 | citywok | that's brutal |
| 20:55.35 | Kobaz | yeah |
| 20:58.37 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 20:58.48 | nny | SIPPEER was perfect, thanks leifmadsen |
| 21:00.35 | Katty | LEIF |
| 21:03.02 | leifmadsen | Katty: ohai |
| 21:03.21 | Katty | leifmadsen: ohaider <3 |
| 21:04.37 | nny | I am curious as to why device names shouldn't be named after the extension, ( i mean I can see it in a situation where corporate phones are passed around) but it shouldn't be the defacto standard for all implementations. If anything, it would make this dialplan more complex in exchange for a passive ability that would never get used. |
| 21:05.28 | nny | matter of fact today is the first time I have heard of it used outside of a specific need. To each his own, the beauty of asterisk. |
| 21:05.52 | WIMPy | nny: Just put a letter in front of the extension to avoid trouble. |
| 21:05.52 | Katty | who's representin digium |
| 21:06.09 | nny | WIMPy: interesting, I assume you mean for security reasons |
| 21:06.09 | Katty | and i don't mean put your hands up, representin |
| 21:06.22 | p3nguin | Then you'll have to figure out how to dial letters from your keypad. |
| 21:06.25 | Qwell | nny: Using numeric peers is a security issue, and it makes it stupid when you have to renumber a device. |
| 21:06.34 | p3nguin | We don't typically have letters on US phones. |
| 21:06.34 | nny | flashes a D and makes a phone handset gesture to his ear |
| 21:06.40 | WIMPy | nny: Yes, and to make it a unique name. |
| 21:07.14 | nny | WIMPy: understandable, I run fail2ban and use pretty decent passwords (not mac etc) so I guess it's security through obsurity if anything. Noted |
| 21:07.16 | WIMPy | p3nguin: "in front of the extension" to form the peer name. |
| 21:07.21 | p3nguin | The MAC address, for example, will forever be the same for that phone. You can give it a new extension and never have to rename it. |
| 21:07.37 | nny | p3nguin: what about the phone config? |
| 21:07.50 | p3nguin | The phone doesn't care about the extension used to dial it. |
| 21:07.58 | nny | just seems like one way to skin a cat if you ask me, other than the additional security layer |
| 21:08.06 | WIMPy | wonders how often you reoute an extension to a single device only. |
| 21:08.14 | Qwell | WIMPy: very often |
| 21:08.29 | Katty | i'd say as often as your mom... |
| 21:08.32 | nny | p3nguin: but if you change the phone extension, you need to configure the phone over.. no matter. it's not a perfect situation |
| 21:08.33 | Katty | ..makes cookies. |
| 21:08.42 | Qwell | nny: no you don't. |
| 21:08.46 | p3nguin | nny: That's incorrect. |
| 21:08.57 | p3nguin | The phone can't care less about what extension is used to reach it. |
| 21:09.04 | nny | p3nguin: Qwell my phone has fields that are used to display the extension number tothe user |
| 21:09.11 | nny | ^^^ |
| 21:09.36 | p3nguin | It only cares about the device name in the channel driver config file, and since you use a unique ID that never changes, nothing in teh phone needs to be changed. |
| 21:10.00 | Gugge | i would change the "displayname" field in my snoms :) |
| 21:10.21 | nny | Gugge: or a linksys will display it's registration number for EXTEN on the line buttons |
| 21:10.29 | p3nguin | There goes an S. |
| 21:10.44 | nny | p3nguin: har har you're hilarious |
| 21:10.57 | p3nguin | I wouldn't have seen it if it weren't for that apostrophe. |
| 21:11.38 | nny | where would be without the grammar nazis of the world? |
| 21:11.59 | nny | forever typing LOL OMG WTF MBTTF P3nguin, that's where |
| 21:12.04 | WIMPy | Should I fetch the [opcorn? |
| 21:12.21 | nny | WIMPy: nah i have an ignore option. It's passed. |
| 21:12.58 | nny | WIMPy: thanks for the security option, i like the simplicity of adding the letter, makes dialplan adjustments simple |
| 21:13.46 | WIMPy | That's not much security, really. But it will avoid numbers being read as IPs. |
| 21:14.05 | nny | WIMPy: explain? |
| 21:14.53 | WIMPy | I missed it, but wasn't it about undefined peers getting interpreted as IPS, earlier on? |
| 21:15.34 | p3nguin | That may have been someone else. |
| 21:15.47 | Katty | infobot: playtones |
| 21:15.52 | nny | WIMPy: oh, indeed. I missed that, thanks. I actually added a SIPPEER line to check to see if the host qualifies first, anything else gets sent to the proper message/resent to IVR etc |
| 21:16.30 | nny | something using exten => s,n,Set(status=${SIPPEER(${ARG1}:status):0:2}) |
| 21:16.51 | nny | if it's OK it moves on, anything else get's sorted and a message played based on what crappened |
| 21:16.56 | nny | er |
| 21:16.56 | nny | gets |
| 21:17.05 | nny | raises right hand in salute |
| 21:17.33 | Qwell | That isn't valid dialplan syntax. O.o |
| 21:17.58 | Qwell | wait, nevermind |
| 21:18.01 | nny | Qwell : or , it's deprecated |
| 21:18.12 | nny | Qwell that's what I get for using the wiki |
| 21:18.23 | Qwell | That's some serious overloading of : |
| 21:18.42 | nny | s,n,Set(status=${SIPPEER(${ARG1},status):0:2}) |
| 21:18.44 | nny | ;) |
| 21:19.14 | nny | i noticed the deprecated notice when testing, thank god for asterisk console feedback being actually useful |
| 21:20.25 | nny | and my goal is accomplished. A simpler (in my case) [people] context and less per extension definition |
| 21:27.04 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:32.46 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:36.14 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 21:40.17 | nny | anyone know of a recent script for checking against voicemail.conf for auth passwords? The perl one or the php one on the wiki haven't been too kind. |
| 21:41.12 | citywok | write a regex to match for it? |
| 21:41.22 | nny | ooh |
| 21:41.25 | citywok | isn't it displayname,mailbox,password ? |
| 21:41.30 | nny | VMWauthenticate? |
| 21:41.32 | nny | er |
| 21:41.37 | nny | VMNauthenticate |
| 21:41.45 | nny | man i'm all thumbs today |
| 21:41.58 | nny | but you get the idea, testing now |
| 21:45.49 | nny | wondering if it support "j" like Authenticate does. seems damn useful though. |
| 21:48.26 | nny | hmm it doesn't, but it says goodbye like vm does, that works. Cool, good to know |
| 22:00.03 | nny | ${CALLERID(num)} is still valid, no? Using the same provider on two systems, one is showing as unknown. Just looking around before I call them |
| 22:00.15 | nny | er setting callerid on two systems* |
| 22:00.39 | nny | exten => _xxxxxxx,n,Set(CALLERID(num)=${7145551212}) |
| 22:01.16 | nny | one system is 1.6, just wondering if anything changed :) |
| 22:01.21 | WIMPy | 7145551212 is not a variable name, is it? |
| 22:01.33 | nny | er sorry wimpy, i edited it to exclude my variable |
| 22:01.35 | nny | was exten => _xxxxxxx,n,Set(CALLERID(num)=${COMPANYNUMBER}) |
| 22:01.50 | nny | so replace companynumber with a number, my bad |
| 22:03.37 | citywok | yea that shold work |
| 22:03.44 | citywok | or CALLERID(all) |
| 22:03.51 | nny | no errors etc, assume this is carrier related (voip provider) |
| 22:04.23 | nny | ok just checking for gravity, i'll call them tomorrow. Oddly I remembering that this worked fine last week. |
| 22:06.40 | nny | looks like it's my cell carrier odd, maybe the expect 1+10 digits, or maybe i'm just crazy |
| 22:08.04 | nny | nope and nope. Odd. |
| 22:08.59 | nny | send voip provider a ticket, meh not much more I can do here |
| 22:09.36 | citywok | most will let you send it however and figure it out, although qwest is stupid and breaks if you send anything other than 10digits |
| 22:09.55 | citywok | with qwest it just rings indefinitely and doesn't actually DO anything if you feed it 11 digits. fail. |
| 22:10.10 | nny | yeah, this is odd. Using vitelity, my main pbx does just fine, this one shows up as unknown, but only on (so far tested) my cell. Maybe be upstream or something of Vitelity |
| 22:10.27 | nny | my cell and my business partner's cell that is |
| 22:10.40 | dr0ck | do you have sendrpid set on the one thats not working |
| 22:11.11 | nny | dr0ck: not that I know of |
| 22:11.22 | nny | dr0ck: i assume it's at the default, I haven't defined it anywhere on either |
| 22:12.16 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:12.47 | nny | testing other destinations |
| 22:14.03 | nny | looks like tmo only, sprint works |
| 22:20.42 | p3nguin | loader.c:387 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister |
| 22:20.46 | p3nguin | Is this a problem? |
| 22:21.37 | p3nguin | It's a WARNING, not a NOTICE. I believe that indicates increased severity. |
| 22:22.22 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zgvxtqggqunbuabw) |
| 22:30.53 | *** join/#asterisk CrossWired (~CrossWire@pool-71-101-94-85.tampfl.dsl-w.verizon.net) |
| 22:31.03 | CrossWired | afternoon everyone |
| 22:31.38 | CrossWired | anyone have a pointer to any links discussing database based autodialer setup? |
| 22:41.39 | citywok | CrossWired: lol what? |
| 22:43.03 | blizzow | Anyone here running queuemetrics on top of their asterisk install? |
| 22:43.24 | navaismo | on top? |
| 22:43.40 | p3nguin | Below? Along side? |
| 22:43.43 | p3nguin | At all? |
| 22:43.48 | blizzow | yes, I just placed the DVD on top of my server, it doesn't do anything. |
| 22:44.02 | citywok | you have to use the 5.25 floppy |
| 22:44.04 | citywok | duh |
| 22:44.50 | nny | blizzow: i have a server using it |
| 22:45.01 | blizzow | Oh, crap. how do I fit the floppy in the cd drive? Do I need to take the cover off? Or can I get a floppy to DVD adapter? |
| 22:45.42 | p3nguin | You have to disassemble the floppy to get the disc out. |
| 22:46.01 | blizzow | Screwdriver or hammer? |
| 22:46.09 | p3nguin | scissors |
| 22:46.20 | blizzow | I have a razor blade, will that do? |
| 22:46.25 | p3nguin | yes |
| 22:46.39 | p3nguin | Have someone else do it so you don't cut yourself. |
| 22:48.00 | blizzow | okay, so once I have queuemetrics inside my asterisk server. I was wondering if it's possible to integrate with some kind of click to dial plugin? I want my sales team to be able to click on a lead's phone number from salesforce and have it automagically routed through queuemetrics. |
| 22:49.20 | nny | blizzow: you'd want to find a click to dial plugin that work with either your broswer or salesforce. After that you can (i believe) tell it what context to send the call to. |
| 22:49.25 | nny | works* |
| 22:49.40 | nny | blizzow: i have tested some here and there, mostly from the voip-info wiki |
| 22:49.59 | nny | i also believe there is a salesforce asterisk integration thing somewhere |
| 22:50.47 | blizzow | nny: thanks, I'll take a gander. I was poking around briefly for salesforce/asterisk integration and only found vaporware. |
| 22:51.38 | nny | has more to do with lead pop up in inbound etc though, if you just want click to dial you'll need an plugin for the broswer your're using. It will probably use the AMI (asterisk manager interface) to tell the system to dial a number and send it to extension XXX (configured in your plugin on that desktop) |
| 22:52.15 | nny | a hackish way also would be to use Google Voice and send all calls to a DID specific to your agent |
| 22:52.27 | nny | G voice has a click to dial thing as well |
| 22:52.34 | nny | blizzow: sorry ^^^ |
| 22:53.18 | nny | blizzow: some more info from the wiki http://www.voip-info.org/wiki/view/Asterisk+click+to+call |
| 22:54.03 | nny | blizzow: my testing has always lead to abandonware etc, the g voice hack would be an other option |
| 22:54.10 | nny | another* |
| 22:54.54 | nny | blizzow: crap that link is wrong one sec |
| 22:57.30 | nny | blizzow: http://www.voip-info.org/wiki/view/Asterisk+call+notification is better, and projects like http://www.tttelecom.nl/index.php/en/products/adat have browser integration. Like i said though my experience has been the projects are abandoned, not finished etc |
| 23:01.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:06.05 | CrossWired | citywok: auto-dialer from a database, dial this number play this message, rather than putting it in a file as i have seen else where one that utilizes a database instead |
| 23:09.39 | nny | CrossWired: have constructed one of those using scripting and mysql |
| 23:10.12 | CrossWired | are there any open projects or writeups to get me started anywhere? |
| 23:10.46 | nny | CrossWired: look into asterisk auto dial and you can use bash scripting, php or C to do the import from mysql to a call file. |
| 23:11.05 | nny | haev to run, been fun gents, and those other people |
| 23:11.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:11.09 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 23:11.14 | CrossWired | thank you |
| 23:19.11 | p3nguin | leifmadsen: There's a mistake in README-SERIOUSLY.bestpractices.txt |
| 23:19.49 | p3nguin | Actually there is more than one, but I'm only pointing out one for now. |
| 23:20.19 | p3nguin | In the secure passwords section, it suggests that extensions have passwords rather than devices having passwords. |
| 23:26.20 | pabelanger | p3nguin: write a patch and submit it to the tracker |
| 23:26.57 | rotten777 | p3nguin: password protect the README-SERIOUSLY.bestpractices.txt to show just how important it is to read it |
| 23:27.19 | p3nguin | If you password protect it, that will discourage the reading of it. |
| 23:53.50 | *** join/#asterisk pdtpatrick1 (~pdtpdt@mainstwan.farheap.com) |
| 23:54.21 | pdtpatrick1 | Question .. i've enabled web on asterisk 1.8 and restart asterisk.. but i keep getting access denied via apache. Is there something else that I need to do? |
| 23:55.54 | p3nguin | What do you mean "via apache" in your statement? |
| 23:56.02 | p3nguin | That makes me think you're proxying through apache. |
| 23:56.20 | pdtpatrick1 | in manager.conf i turned web to enabled |
| 23:56.33 | pdtpatrick1 | in http.conf i've always turned on enabled and bindport = 8088 |
| 23:56.52 | pdtpatrick1 | restart asterisk and then i go to localhost:8088 |
| 23:57.16 | pdtpatrick1 | which points to /static/config/index.html |
| 23:57.22 | pdtpatrick1 | and it says Access Denied |
| 23:57.50 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 23:58.11 | *** join/#asterisk adeel (~adeel@184.175.36.92) |