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| 01:32.31 | Sophira | I managed to resolve my previous question, but I have a new one. Is it possible to have Asterisk ignore invalid extensions and just react when the numbers typed so far could lead to (or is) a valid extension? My reasoning is that I want a Background() file to play, but I want the # key to be able to skip it. However, I don't want any other key to interfere with the playback of the file. |
| 01:32.37 | Sophira | Is this possible? |
| 01:32.50 | Sophira | I tried 'ignorepats', but it doesn't seem to do what I want it to do. |
| 01:33.15 | Sophira | Well, |
| 01:33.17 | Sophira | 'ignorepat'. |
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| 02:51.18 | [TK]D-Fender | Sophira: ControlPlayback instead of Background |
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| 03:28.07 | doogienz | Hi to all - hunted high and low and my googlefu has let me down. Does anyone know of an easy way in a dial plan to match X number of digits? i.e. I want to match when the callerid is only 3 digits long and replace it. Just need to know how I match X amount of digits. |
| 03:28.32 | [TK]D-Fender | make your pattern that long |
| 03:28.54 | doogienz | Probably need a better example |
| 03:29.06 | doogienz | Client has presented callerid wrong - a 3 digit extension. |
| 03:29.18 | [TK]D-Fender | exten => 12345,1,NoOp(this pattern is clearly 5 long ... and a single value) |
| 03:29.21 | doogienz | I want to match anything less than 7 digits and overwrite it |
| 03:29.41 | [TK]D-Fender | You said you wanted to match X .... not a RANGE |
| 03:29.54 | doogienz | No I want to match a count of digits. |
| 03:30.10 | doogienz | If CALLERID(num) is less than 7 digits I'm going to rewrite it. |
| 03:30.13 | [TK]D-Fender | There is no length range... you'll need multiple patterns, or you'll have to accept a wider range and make a logical check in the dialplan |
| 03:30.27 | [TK]D-Fender | You can process it that way... but it isn't a "match". |
| 03:31.03 | doogienz | No wonder my googlefu let me down. |
| 03:31.18 | doogienz | So no way to count the numbers of digits and return that as a value? |
| 03:31.53 | doogienz | Wihtout breaking to an agi I suppose |
| 03:32.07 | [TK]D-Fender | doogienz: "core show function LEN" |
| 03:32.18 | [TK]D-Fender | doogienz: "core application gotoif" |
| 03:32.27 | [TK]D-Fender | doogienz: "core application execif" |
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| 03:37.19 | doogienz | danke @[TK]D-Fender the LEN function was the one. |
| 03:37.28 | doogienz | Much appreciated. |
| 03:37.46 | [TK]D-Fender | MeanderingCode: Those are the pieces for you work your way around it |
| 03:41.15 | doogienz | See my googlefu really let me down.. I was searching count when I should have searched for length :D |
| 03:49.16 | doogienz | NoOp(Length: ${LEN(${CALLERID(num)})}) |
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| 03:58.44 | [TK]D-Fender | getting places |
| 04:02.11 | doogienz | Cool and macro finished. thanks again. |
| 04:03.56 | [TK]D-Fender | Excellent |
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| 04:37.19 | phix | hai [TK]D-Fender!! how ya been? |
| 04:42.15 | blizzow | I enabled the built in mini http server in /etc/asterisk/http.conf, I also have my bind port set to 38088. The prefix is set to: prefix=asterisk. I restarted the amportal but whenever I try and visit http://sipserver:38088/asterisk I get: "Not Found The requested URL was not found on this server. Asterisk Server" |
| 04:42.41 | blizzow | I don't see anything in /var/log/asterisk/full that explains why the http server isn't seeing anything. |
| 04:43.38 | blizzow | Can anyone explain where I can look for a little more information why this is failing or what I might be doing wrong? |
| 04:49.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:54.37 | kaldemar | blizzow: amportal suggests that you use something that might have overrun you config changes. if you use freepbx, ask in #freepbx. |
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| 06:15.23 | The_Ball | Hi this might not be something Asterisk related, is it possible to setup a voip device so it is represented as a soundcard to the os running Astreisk? |
| 06:21.52 | ChannelZ | Besides just a softphone? |
| 06:26.15 | The_Ball | Well a softphone doesn't act like a soundcard, I want the exact opposite of a softphone in a way |
| 06:26.53 | The_Ball | A softphone takes a real sound card -> voip, I want voip -> virtual sound card |
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| 06:38.01 | schmidts | good morning |
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| 06:42.06 | blizzow | okay, so I am trying to convert a piece "show dialplan" output back into a dialplan .conf file. Do I have this right? http://pastebin.com/jMDZUFSV |
| 06:45.26 | schmidts | blizzow instead of numbering each row you should use n instead of numers like exten => _XXX.,n |
| 06:45.39 | schmidts | s/numers/numbers |
| 06:45.43 | wdoekes2 | [ Context 'queuedial-ctd' created by 'pbx_config' ] => [queuedial-ctd] |
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| 06:46.39 | wdoekes2 | and good morning |
| 06:47.24 | blizzow | good morning, w00t! |
| 06:50.26 | schmidts | good morning wdoekes2 ;) |
| 07:01.23 | kaldemar | blizzow: there's also a command to write dialplan to a file, "dialplan save". requires writeprotect=no under [general] in dialplan config. if you have some stuff in extensions.conf, take a backup before the command or it might screw up your labeled priorities etc. |
| 07:01.37 | schmidts | can someone help me to get rid of these warnings: warning: implicit declaration of function 'ldap_simple_bind_s' i have written an app for asterisk which checks an ldap database and it compiles and it also works but i allways get these warnings :( |
| 07:06.54 | schmidts | ok i maybe found it, depening to ldap.h these functions are "depricated" but still working |
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| 07:18.29 | blizzow | kaldemar: thanks. |
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| 08:12.56 | jww_ | Hello ! |
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| 08:16.25 | qakhan | hi i have a queue. it has defualt music. i want to setup simple ring which caller wait in queue |
| 08:21.40 | Rico29 | hello |
| 08:22.22 | Rico29 | I have questions about voicemail. First one, I can't manage to set the maximum message length to more than 40 seconds |
| 08:22.53 | Rico29 | second one : i'm trying to announce the caller number as "zero siz sixty-two thirty-three ..." instead of 'zero six six two three three..." |
| 08:28.20 | Rico29 | can anyone help me with these problems ? |
| 08:30.07 | kaldemar | 1. maxsecs= under [general] in voicemail.conf 2. no-go |
| 08:32.16 | Rico29 | kaldemar > I tried, but it doesn't seem to be understood by asterisk |
| 08:32.16 | qakhan | kaldemar can u help me |
| 08:32.30 | Rico29 | maxsecs=360 |
| 08:33.27 | kaldemar | qakhan: "core show application Queue" |
| 08:33.34 | Rico29 | but the recording of the message stops after 40 secs |
| 08:33.48 | Rico29 | my voicemail users are in DB |
| 08:34.01 | Rico29 | do I have tu add a field "maxsec" for each ? |
| 08:34.10 | Rico29 | i don't think.. |
| 08:34.12 | Rico29 | so |
| 08:37.32 | kaldemar | Rico29: oh well, works here. maybe you have some conflicting configuration in your voicemail.conf. |
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| 08:39.43 | StaRetji | urgv4 mate, ping |
| 08:40.02 | StaRetji | urvg4: let me know when you are available, thx |
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| 08:41.34 | N1x_ | hi guys , i have extension like this http://pastie.org/3815456 |
| 08:41.53 | N1x_ | when call is coming from sip provider background is played and |
| 08:42.16 | N1x_ | if i press 7777 on my phone |
| 08:42.26 | N1x_ | it should work in IVR_OPTIONS context yes? |
| 08:44.56 | wdoekes2 | Rico29: in constrast with most other modules, app_voicemail uses the first parameter in voicemail.conf instead of the last one |
| 08:45.23 | Rico29 | wdoekes2 > so why shouldn't it work ? |
| 08:45.38 | wdoekes2 | Rico29: like kaldemar said, you probably have the option defined multiple times |
| 08:45.52 | Rico29 | i don't |
| 08:46.19 | Rico29 | [root@SRV-IPOP1 dialplan]# cat /etc/asterisk/voicemail.conf | grep maxsecs |
| 08:46.19 | Rico29 | maxsecs=360 |
| 08:46.30 | Rico29 | asterisk 1.8.8.0 |
| 08:46.33 | N1x_ | any ideas? :-( |
| 08:46.43 | kaldemar | Rico29: where you do have that in the file? |
| 08:47.41 | wdoekes2 | maxmessage=40 ? |
| 08:48.54 | doogienz | Here's a question, if i 4 variables to check for exec if, is there a better way to do it rather than 4 exec if lines in the dial plan? |
| 08:49.14 | Rico29 | kaldemar > wdoekes2 : http://pastebin.com/n2DEVF6L |
| 08:49.18 | doogienz | Basically just checking the first two digits of the dnid and setting a variable |
| 08:50.14 | wdoekes2 | Rico29: you did reload the voicemail module? |
| 08:50.44 | wdoekes2 | do the voicemail users in the db define a maxsecs (or maxmessage)? |
| 08:51.15 | wdoekes2 | doogienz: parse error on your question |
| 08:51.18 | Rico29 | wdoekes2 > I did a "voicemail reload", not a "module reload app_voicemail.so" |
| 08:51.31 | Rico29 | wdoekes2 > no maxsec voir voicemail users in BD |
| 08:51.44 | Rico29 | neither maxmessage |
| 08:52.28 | doogienz | Sorry @wdoekes2 parse error? |
| 08:53.41 | wdoekes2 | doogienz: did you read your question before hitting enter? |
| 08:53.45 | N1x_ | kaldemar: can you help me , i'll dont waste your time for a long time |
| 08:53.52 | Rico29 | kaldemar > wdoekes2 > maxmessage parameter doesn't change anything |
| 08:54.53 | doogienz | @wdoekes2 Think it makes sense maybe I'll paraphrase a bit more |
| 08:56.42 | kaldemar | doogienz: IF($["${var1:0:1}" = "0" | "${var2}" = "123"]?true) |
| 08:57.10 | kaldemar | N1x_: do you have an issue to show? |
| 08:57.14 | doogienz | Can you do the same piping with an ExecIf, or only IF? |
| 08:57.30 | kaldemar | doogienz: that's not piping, it's an OR |
| 08:58.05 | kaldemar | and it works in all expressions, as in what is inside $[]. |
| 08:58.22 | N1x_ | kaldemar: yes , http://pastie.org/3815456 is it correct? when call is coming from sip provider background is played and if i pres 7777 on my phone , it should work in IVR_OPTIONS ? |
| 08:59.18 | N1x_ | but when i press 7777 , asterisk deleting last 3 digits and said number "7" is not valid number and not found in context |
| 08:59.45 | N1x_ | but i have context => _7XXXX in my IVR_OPTIONS |
| 09:01.47 | kaldemar | N1x_: _7XXXX is five digits, 7777 is four. they don't match. |
| 09:02.03 | N1x_ | sorry i have _7XXX |
| 09:03.05 | kaldemar | N1x_: turn on DTMF debugging in logger.conf by adding "dtmf" on the console line and do a "logger reload". you'll see incoming DTMF digits to know if all is received. then pastebin a CLI output of a failed call and the both contexts from extensions.conf. |
| 09:03.22 | wdoekes2 | Rico29: # gdb -p `pidof asterisk` -ex 'print vmmaxsecs' -ex detach -ex quit | grep '^\$1' |
| 09:05.31 | Rico29 | wdoekes2 > i'll do it in about 1 hour, and give you a feedback |
| 09:08.58 | N1x_ | kaldemar: http://pastie.org/3815548 |
| 09:09.05 | N1x_ | this is output |
| 09:10.46 | N1x_ | kaldemar: http://pastie.org/3815557 this is IVR_OPTIONS |
| 09:11.33 | N1x_ | kaldemar: http://pastie.org/3815564 and this is incoming context from sip provider |
| 09:11.38 | doogienz | @kaldemar thanks muchly - winnah! exten => s-settrunk,1,ExecIf($[${CALLERID(dnid):0:2}=03|${CALLERID(num):0:2}=03],Set,(trunk=chch)) |
| 09:15.10 | kaldemar | N1x_: that's a single digit you're getting in. your problem is not with the contexts but with getting the whole extension. a short digit timeout (TIMEOUT(digit)) might be one reason. |
| 09:15.39 | din3sh | on fxo card, phone continues to ring 5,6secs after fxo caller has hang up |
| 09:15.45 | din3sh | how to lower this time? |
| 09:16.06 | N1x_ | kaldemar: hm i need to add timeout after my background extension? |
| 09:17.48 | kaldemar | N1x_: no. |
| 09:18.17 | kaldemar | N1x_: "core show function TIMEOUT" |
| 09:19.29 | kaldemar | N1x_: add a NoOp(${TIMEOUT(digit)}) before the BackGround to find our your current setting. if it is ridiculously short for some reason, increase it. before BackGround. |
| 09:20.12 | N1x_ | ok sec |
| 09:21.40 | kaldemar | din3sh: by using disconnect supervision or by decreasing busycount if you use busydetect. |
| 09:24.52 | N1x_ | kaldemar: i added it but same output |
| 09:26.26 | kaldemar | what's the timeout value? |
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| 09:28.52 | din3sh | what does increasing busycount do? |
| 09:29.01 | din3sh | actually what does busycount do? |
| 09:30.42 | kaldemar | din3sh: the sample config will tell you |
| 09:31.46 | din3sh | :) |
| 09:31.47 | din3sh | oki |
| 09:32.03 | kaldemar | din3sh: but that is not a good option if you have disconnect supervision available from the other end. see option hanguponpolarityswitch. |
| 09:32.44 | din3sh | if there is surpervision, busycount must be disabled? |
| 09:33.33 | kaldemar | busydetect should be disabled, yes. |
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| 09:35.34 | doogienz | Asterisk 1.4 syntax for exec if, i thought it was ],Set,(blah=foo)) . Is that not correct? |
| 09:36.28 | doogienz | derp nevermind |
| 09:36.37 | doogienz | ],Set,blah=foo) |
| 09:37.29 | din3sh | anyone have experience with rowotel's embedded * IP04 |
| 09:37.31 | din3sh | ? |
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| 09:59.06 | doogienz | Okay last question I hope! On Execif can you set something and THEN issue a goto, or is it one exec only? |
| 09:59.13 | doogienz | If true |
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| 10:44.56 | aberrios | just reading about app_milliamp is this the same a "core show application milliwatt"? I'm wanting to generate a tone at 2970Hz |
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| 10:46.25 | kareena | hi |
| 10:46.51 | kareena | i want to use mailtofax and faxtomail with asterisk how can i do that? |
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| 10:53.20 | din3sh | kareena kapoor? |
| 10:53.22 | din3sh | :p |
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| 10:54.51 | kareena | no |
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| 11:00.38 | RZero | Hi, has any one seen a problem where a sip trunk between asterisk 1.6.2.13 and asterisk 1.6.2.20 plays no audio, yet IAX works fine, both servers have external IP's |
| 11:00.53 | RZero | so its not a nat issue |
| 11:01.10 | RZero | firewall are open |
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| 11:11.33 | aberrios | ahhh PlayTones() |
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| 11:49.25 | ipconfeng | any know what this means: rc_avpair_new: unknown attribute 1490026597 |
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| 11:58.52 | Jasnejac | ipconfeng: http://www.voip-info.org/boards/index.php?t=17493 |
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| 12:53.58 | ozzy_osbourne | hi, just wanted to know if theres any call-forward feature in asterisk |
| 12:55.32 | [TK]D-Fender | ozzy_osbourne, It's your dialplan, you can do whatever you want |
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| 13:20.07 | mgarces | hi there |
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| 13:20.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:20.43 | mgarces | has anyone been successful on making a trunk using voipyo.com ? |
| 13:21.39 | RZero | Hi, has any one seen a problem where a sip trunk between asterisk 1.6.2.13 and asterisk 1.6.2.20 plays no audio, yet IAX works fine, both servers have external IP', so its not a nat issue, firewall are open |
| 13:22.51 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 13:23.06 | [TK]D-Fender | RZero, directmedia=no <---- make sure this is set for all of your sip.conf entries |
| 13:24.55 | RZero | is that not the same as canreinvite ? |
| 13:27.52 | RZero | ah I see it change in .20 thanks TK |
| 13:33.07 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-185.nayatel.pk) |
| 13:33.58 | mechbangirc | how can I change user and group of /dev/dahdi from asterisk.asterisk to something else? |
| 13:34.14 | mechbangirc | any option in init.conf or somewhere? |
| 13:44.55 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e5fa.pool.mediaWays.net) |
| 13:45.07 | jaytee | In which version of asterisk 1.6.2.x did canreinvite change to directmedia? |
| 13:45.27 | *** join/#asterisk rjvvliet (~rjvvliet@2001:610:600:86a5:8dab:94e8:edb2:1e16) |
| 13:45.44 | schmidts | jaytee IMHO it was 1.6 itself or atleast 1.6.2.0 but not somewhere in between |
| 13:46.21 | din3sh | [TK]D-Fender: i have cisco phones which do not work with directmedia=no |
| 13:47.19 | din3sh | why is it so? |
| 13:47.28 | jaytee | I have a server running 1.6.2.18 but I've been using canreinvite=no |
| 13:47.38 | schmidts | din3sh sounds like a problem with codecs, have you checked if the cisco phone use the same codecs as asterisk? |
| 13:47.58 | din3sh | alaw evrywhere |
| 13:48.16 | schmidts | din3sh which cisco phones? |
| 13:48.17 | din3sh | anyway i have them working with directmedia=yes |
| 13:48.22 | din3sh | 79xx series |
| 13:48.27 | schmidts | ah ok |
| 13:48.55 | din3sh | how can directmedia=yes affect the calls? |
| 13:49.10 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:49.46 | din3sh | anyway gtg |
| 13:49.50 | din3sh | later |
| 13:49.51 | [TK]D-Fender | absolutely no reason they should NOT work with directmedia disabled... that just tells * to not offer a re-invite. the phone should have no problem staying on * and letting * brdige internally. this is the norm when non-lan endpoints are involved.... |
| 13:50.06 | din3sh | ok |
| 13:53.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:53.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:53.48 | *** join/#asterisk DanFromUK (~IceChat77@2.26.253.199) |
| 13:54.49 | DanFromUK | Hi all. I was just wondering, what is the benefit of asterisk realtime compared to generating sip.conf files automatically and reloading them? |
| 13:56.52 | [TK]D-Fender | You don't have to manually reload. They can be located on another server. Being in a DB you can do DB things to them.... all of the blatently obvious things that a DB implies... |
| 13:56.55 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 14:00.13 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:04.35 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 14:07.07 | DanFromUK | ok, i hear that. |
| 14:07.26 | DanFromUK | thanks |
| 14:08.35 | *** join/#asterisk error27 (~dcarpente@41.139.221.94) |
| 14:10.45 | error27 | sorry if this is a faq, but does asterisk use the telephony stuff in the kernel that got moved to staging? |
| 14:12.46 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 14:21.56 | leifmadsen | DanFromUK: reloading constantly is not really a very good practice. Basically with realtime the information is immediately available without having to generate the configs and reloading the module that reads the configuration file. |
| 14:22.09 | leifmadsen | DanFromUK: depending on your implementation and requirements, both options are valid |
| 14:22.35 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 14:28.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:32.34 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.140.113) |
| 14:51.12 | *** join/#asterisk din3sh (din3sh@41.136.247.246) |
| 14:55.31 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 14:56.30 | jaytee | is any DID number in the USA portable from one provider to another? |
| 15:00.46 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:04.33 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:06.06 | p3nguin | usually |
| 15:08.25 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 15:08.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:16.33 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
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| 15:32.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-202.mobile.uci.edu) |
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| 15:43.26 | *** part/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 15:44.17 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 15:47.18 | ketas | someone i know wants statistics about call duration vs. time, probably with rrd |
| 15:47.29 | ketas | anyone has any ideas on that? |
| 15:47.50 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:47.57 | ketas | call duration graphs |
| 15:50.33 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:52.43 | *** join/#asterisk TheMan (~TheMan@66.237.29.132.ptr.us.xo.net) |
| 15:59.01 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 16:01.13 | *** join/#asterisk teratoma (~teratoma@i.dont.get.mad.i.get.stabby.net) |
| 16:01.59 | RZero | ketas just google asterisk cdr stats lots of good free stuff out there |
| 16:03.25 | *** join/#asterisk mroe (~mroe@unaffiliated/roe) |
| 16:04.30 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 16:05.06 | ketas | RZero: already managed to find something |
| 16:06.05 | RZero | ;) |
| 16:10.54 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:11.56 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 16:25.06 | keith4 | how do I confirm the existence/absence of a hwec module on a Digium TE121, from 20 miles away? |
| 16:27.03 | coppice | keith4: a really good telescope? |
| 16:27.08 | anonymouz666 | haha |
| 16:27.32 | anonymouz666 | keith4: cat /proc/dahdi/<channel> |
| 16:28.42 | anonymouz666 | keith4: 8 TE4/0/1/8 CAS (In use) (SWEC: OSLEC) (EC: VPM450M) |
| 16:28.54 | anonymouz666 | output of channel 8 |
| 16:28.59 | anonymouz666 | oops |
| 16:29.03 | anonymouz666 | it's not channel, it is SPAN |
| 16:29.20 | anonymouz666 | cat /proc/dahdi/<SPAN> |
| 16:29.26 | keith4 | well, that shows me what the driver is doing/using (OSLEC), yes? |
| 16:29.49 | anonymouz666 | there is no way to disable HWEC |
| 16:29.55 | anonymouz666 | it is using the vpm450m |
| 16:30.05 | anonymouz666 | but the swec is OSLEC |
| 16:30.50 | anonymouz666 | we don't use miles we use kilometers instead :-P |
| 16:30.54 | keith4 | okay. let me ask the guy who "installed it" |
| 16:31.44 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 16:31.50 | coppice | anonymouz66: why do you use OSLEC when you have a hardware EC module? |
| 16:32.09 | keith4 | alright... it's allegedly a VPMOCT032 |
| 16:32.27 | anonymouz666 | coppice: is it in use? |
| 16:32.35 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 16:34.23 | anonymouz666 | or it is just listed? |
| 16:34.45 | anonymouz666 | I really don't know, does not make any sense using two EC |
| 16:35.49 | anonymouz666 | SWEC in this machine would consuming a few GHz of CPU I would guess for 8 spans... |
| 16:36.03 | *** join/#asterisk din3sh (din3sh@41.136.243.157) |
| 16:44.14 | *** join/#asterisk vfabi (~fabi@178.76.64.166) |
| 16:46.14 | keith4 | hmm. the dude sent me a picture of the module installed on the card.... but I see no evidence of it in software |
| 16:47.55 | coppice | maybe its a cardboard cutout |
| 16:48.43 | anonymouz666 | coppice: I just get the confirmation that DAHDI will choose by default HW over SW, so I don't need to worry about using two EC. |
| 16:50.36 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 16:50.52 | keith4 | maybe this thing needs a firmware update, or something? I'm starting to think I'm crazy |
| 16:53.05 | keith4 | do these firmware version numbers correspond to any particular dahdi versions? http://downloads.digium.com/pub/telephony/firmware/releases/ |
| 16:53.12 | keith4 | or should I just grab the latest, and not worry about it? |
| 16:58.28 | anonymouz666 | keith4: why don't you call Digium? |
| 16:59.13 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 17:02.40 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:04.21 | keith4 | anonymouz666: I'm going to make him do it, I think ;-) |
| 17:04.58 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 17:05.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:07.06 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.dsl.siol.net) |
| 17:07.38 | jozza | hi i have aweird problem that started to happen with sip rtp |
| 17:08.05 | jozza | extension is on outside ip but it gives its local ip in the sdp message |
| 17:08.34 | jozza | there has never been a problem with audio before so i wasn paying attention as to where rtp packets are going |
| 17:09.22 | jozza | but now that there is a problem with audio i looked at sdp and i see the asterisk sends rtp to the ip that extesnion specified |
| 17:09.34 | jozza | which is a local ip |
| 17:09.56 | jozza | what can be done to fix this? |
| 17:13.07 | IsUp | jozza: localnet= externip= nat= |
| 17:15.38 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:16.04 | jozza | IsUp: yes, i have that, but doesnt help |
| 17:20.49 | jozza | sip messages are going correctly between extension and ast |
| 17:21.22 | jozza | but ast accepts the rtp ip address specified in sdp of the extension which is local |
| 17:21.25 | jozza | is that normal? |
| 17:22.49 | IsUp | jozza: pastebin your sip.conf and enable sip debugging |
| 17:23.06 | jozza | i m running freepbx |
| 17:23.17 | jozza | so too much info for sip |
| 17:23.28 | jozza | but i can post a debug |
| 17:24.02 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:27.50 | jozza | http://pastebin.com/MwH3xVTk |
| 17:28.02 | jozza | IsUp: sent the link |
| 17:28.15 | jozza | it makes no sense to me |
| 17:28.32 | jozza | its a call flow of the situation |
| 17:29.08 | jozza | this happened suddenly out of unknown reason that ast sends rtp to local ip |
| 17:39.48 | IsUp | jozza: as i understand, everything looks okay. c=IN IP4 178.254.132.162 <-> c=IN IP4 192.168.1.2 |
| 17:42.05 | IsUp | jozza: you have an extension 116, which in your local network. and your provider is 178.254.132.162, which on public IP. sending call to your extension. right? |
| 17:42.26 | jozza | 116 is on 172....249 |
| 17:43.59 | jozza | IsUp: 116 is on 178.222.58.249 |
| 17:44.53 | IsUp | jozza: set nat=no under 116. and wheres your sip.conf? pastebin it too. mask passwords |
| 17:48.05 | jozza | IsUp: i am running freepbx so sip.conf is complicated |
| 17:48.25 | IsUp | jozza: #freepbx is the right place. anyways, set nat=no for extension 116 |
| 17:48.26 | jozza | IsUp: i tried nat=no, no change |
| 17:48.38 | jozza | but it cant be freepbx problem |
| 17:48.41 | jozza | can it? |
| 17:50.06 | IsUp | jozza: it doesnt matter. freepbx managed by GUI. so i cant point to right direction |
| 17:50.13 | IsUp | jozza: did you do 'reload' ? |
| 17:50.18 | jozza | yes i did |
| 17:50.28 | jozza | i know how freepbx works |
| 17:50.40 | jozza | so you cant try to point me the standard ast directions |
| 17:50.43 | jozza | i can manage |
| 17:51.17 | jozza | if you have any ideas, i'd love to try them out |
| 17:54.31 | jozza | IsUp: http://pastebin.com/JF3Qm6hu |
| 17:55.16 | jozza | IsUp: http://pastebin.com/1W3fTZ9B |
| 17:56.15 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 17:58.30 | *** join/#asterisk Valken (~valken@gw.ptr-80-238-181-50.customer.ch.netstream.com) |
| 17:58.34 | Valken | Hi. |
| 18:10.30 | Valken | Is there a way to set the sounds files path from the extensions.conf file or from the cli? |
| 18:10.31 | *** join/#asterisk DennisG (~dennisg@541AFD1E.cm-5-3d.dynamic.ziggo.nl) |
| 18:11.17 | jozza | IsUp: i have figured out the problem |
| 18:11.23 | IsUp | jozza: yes, what is it |
| 18:11.27 | [TK]D-Fender | Valken, You can specify the full path by starting with a "/". Everything else is relative |
| 18:11.36 | [TK]D-Fender | (to "varlibdir" in asterisk.conf |
| 18:12.12 | Valken | [TK]D-Fender: Thank but I really need something similar to a path. Since I don't know where the file is. |
| 18:12.18 | jozza | IsUp: ast waits for first rtp packet from extension. If that packet is not received in atime that rtp data exchange should start, ast takes the rtp ip address specified in the sdp message |
| 18:12.31 | Valken | The path is told by an AGI script and then I need to concat the file name. |
| 18:12.58 | [TK]D-Fender | Valken, there is no "path". |
| 18:13.01 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:13.19 | [TK]D-Fender | Valken, You tell it what & where |
| 18:13.23 | jozza | IsUp: but as soon as rtp packet magically appears from extension, ast changes the rtp ip address to correct one |
| 18:14.22 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 18:14.30 | jozza | IsUp: so the problem is not in ast but with rtp packets not being received from extension |
| 18:14.41 | jozza | its interesting though |
| 18:15.12 | jozza | now we'll be looking at router/firewall if packets are being received at all or dropped |
| 18:15.31 | Valken | [TK]D-Fender: I finally manage to got it this way: |
| 18:15.33 | Valken | exten => 1,1,Set(SOUNDTOPLAY=${SOUNDFOLDER}hello) |
| 18:15.34 | Valken | exten => 1,n,Playback(${SOUNDTOPLAY}) |
| 18:16.02 | [TK]D-Fender | Valken, that isn't a "search path". |
| 18:16.19 | Valken | It would be nice if we could have a global variable we could set to tell where asterisk must look for sound file |
| 18:16.23 | [TK]D-Fender | Valken, that is just a variable or constant reference |
| 18:16.30 | Valken | [TK]D-Fender: I know. I do what I can to perform the job. ;) |
| 18:16.33 | [TK]D-Fender | Valken, asterisk.conf <- |
| 18:16.44 | [TK]D-Fender | varildir + sounds |
| 18:16.49 | [TK]D-Fender | varlibdir |
| 18:17.53 | Valken | anyway, thank for your help. |
| 18:18.43 | [TK]D-Fender | Valken, Don't really see a point to doing a Set there first... |
| 18:19.02 | jozza | IsUp: do you understand? |
| 18:19.41 | IsUp | jozza: no, not at all |
| 18:20.25 | jozza | IsUp: which part is confusing you? |
| 18:20.51 | IsUp | jozza: your SDP is wrong. |
| 18:21.17 | jozza | IsUp: i cant do anything about sdp |
| 18:21.35 | jozza | because the phone is in a private lan behind a non sip aware router |
| 18:22.16 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.42) |
| 18:22.51 | jozza | IsUp: should the phone be aware of its public ip through sip msg exchange? |
| 18:23.26 | IsUp | jozza: your scenerio is complicated (for me at least), maybe you should use a STUN server. |
| 18:23.35 | Qwell | ~nat |
| 18:23.35 | infobot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:24.38 | jozza | IsUp: ok, i identified the problem and i can fix it now by start looking elsewhere for a problem |
| 18:24.51 | IsUp | jozza: good luck |
| 18:25.01 | jozza | IsUp: thx for help :) |
| 18:31.00 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:31.04 | *** join/#asterisk keiths_ (~keiths@goatfarm.vianet.ca) |
| 18:31.49 | *** join/#asterisk ashd (~ashleyd@212.241.164.132) |
| 18:43.20 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:44.05 | *** join/#asterisk error27 (~dcarpente@41.139.221.94) |
| 18:46.03 | *** join/#asterisk m_tadeu (~quassel@78-130-56-17.net.novis.pt) |
| 18:46.13 | m_tadeu | hi everyone |
| 18:47.06 | m_tadeu | I've been looking for docs about the status of jingle in asterisk. can anyone point me out? |
| 18:51.45 | m_tadeu | brb |
| 19:07.13 | ChannelZ | (the status?) |
| 19:09.53 | *** join/#asterisk edguy3 (~edguy3@office.cleverspoke.com) |
| 19:11.45 | edguy3 | anyone seen this: Error loading module 'res_digium_phone.so': /usr/lib/asterisk/modules/res_digium_phone.so: undefined symbol: ast_msg_tech_unregister |
| 19:11.49 | edguy3 | ? |
| 19:12.33 | pabelanger | edguy3: contact digium support |
| 19:14.05 | pabelanger | edguy3: which version of asterisk? |
| 19:19.49 | *** join/#asterisk m_tadeu (~quassel@89.180.85.170) |
| 19:21.02 | Qwell | edguy3: You need to use the 1.8-digiumphones branch of Asterisk, for DPMA |
| 19:22.31 | m_tadeu | does jingle work good on asterisk? |
| 19:26.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:28.57 | pabelanger | m_tadeu: meh, it isn't actively tested. So your mileage will vary |
| 19:29.01 | pabelanger | what are you looking to do? |
| 19:30.24 | m_tadeu | I have asterisk in my office and from now on, I have people getting calls using a 3G connection....problem is that providers are blocking SIP packages |
| 19:30.32 | m_tadeu | so I was willing to try xmpp |
| 19:32.03 | m_tadeu | and since I'm already using jabberd2 for messaging, looked like the way to go |
| 19:32.27 | edguy3 | Sorry called away |
| 19:32.32 | edguy3 | Connected to Asterisk 1.8.11.0-1digium1~squeeze currently running on softlyraining (pid = 22699) |
| 19:33.01 | Qwell | edguy3: You'll have to build the abovementioned branch from source, to use DPMA. |
| 19:33.11 | pabelanger | m_tadeu: try chan_jingle |
| 19:33.33 | pabelanger | but I wouldn't use it for a business case |
| 19:33.33 | edguy3 | got it. love the new hardware! |
| 19:33.50 | Qwell | edguy3: we like them too :D |
| 19:34.41 | m_tadeu | pabelanger: thanx for the advice |
| 19:34.48 | pabelanger | Qwell: Can you not add logic to detect which version of asterisk res_digium_phone is being loaded into and just fail and output a warning if the wrong branch? |
| 19:35.13 | pabelanger | rather then undefined symbol: ast_msg_tech_unregister |
| 19:35.26 | edguy3 | qwell: what is latest firmware? ( only reason I am installing DPMA is for firmware upgrades. ) |
| 19:35.28 | Qwell | pabelanger: no, it can't load the .so to output said warning |
| 19:35.49 | Qwell | edguy3: 1_0_0 is the latest released firmware, tested with DPMA. |
| 19:36.27 | pabelanger | Qwell: Ya, guess you'd need to add the logic into branches/1.8 and branches/10 |
| 19:37.33 | edguy3 | qwell: are there other reasons to use the 1.8-digiumphones? |
| 19:37.49 | Qwell | edguy3: all the fancy stuff DPMA offers, which is quite a bit |
| 19:38.09 | edguy3 | qwell.. OK. sold. |
| 19:39.20 | *** join/#asterisk timahvo1 (~rogue@197.178.42.129) |
| 19:42.58 | CyfordTechnologi | i need some help with my asterisk running in realtime mode.. it seems asterk can see my extenstions in the database but it will not allow these extenstions too register |
| 19:46.05 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:46.29 | *** join/#asterisk timahvo1 (~rogue@197.177.107.43) |
| 19:46.37 | *** part/#asterisk asterisk-Tester (~ramy@80.79.159.42) |
| 19:48.38 | *** join/#asterisk twanny796 (~twanny@85.232.219.136) |
| 19:52.01 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:54.44 | Greenlight | howdy - just wondering if it's normal for asterisk to use 3.5gb of memory right after starting (no calls etc made yet) |
| 19:55.19 | Greenlight | 3764 asterisk -11 0 3408m 39m 10m S 2.3 0.2 0:01.94 asterisk |
| 19:55.53 | pabelanger | Oh ya, something is wrong there |
| 19:56.32 | Greenlight | I've not really checked what it's been using before, but 3.5 gigs seems a lot. The system has plenty memory like, but you think there's a prob theere, yea? |
| 19:57.02 | Greenlight | The reason I've been investigating is that after about 10 days uptime the timing starts to "slip" and I need to reboot |
| 19:57.18 | Greenlight | Eg conferences audio all choppy and "timing test" results bad results |
| 19:58.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:59.42 | Greenlight | It's really strange and I've no other boxes where I'm seeing this on |
| 20:03.52 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 20:04.20 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 20:04.30 | glaz | Is Dial() supposed to keep going in the context if call answered ? |
| 20:04.53 | Greenlight | Nope |
| 20:05.00 | glaz | any option that can do it? |
| 20:05.12 | glaz | I'm trying to do TBCT with Dial() and Transfer() |
| 20:05.14 | Greenlight | I think there's a paramenter you can give in to run on answer iirc |
| 20:05.36 | glaz | Greenlight: That's what I think too but I can't find it |
| 20:05.43 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 20:06.22 | [TK]D-Fender | glaz, normally you don't use that app. You just set "transfer=yes" in chan_dahdi.conf in your channel definitions |
| 20:06.31 | Greenlight | M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers) |
| 20:06.47 | glaz | [TK]D-Fender: that option will enable TBCT ? |
| 20:06.51 | [TK]D-Fender | yes |
| 20:07.07 | [TK]D-Fender | ~2bct |
| 20:07.07 | infobot | [~2BCT] 2BCT (2 B-Channel Transfer) allows a call coming in over DAHDi and back out again to the same telco to be handed off freeing the channels from your circuit. To enable this (if your carrier supports it) add "transfer=yes" to your channel configurations. |
| 20:07.12 | glaz | so basicaly, I Anwser() Dial() and it will transfer? |
| 20:07.15 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 20:07.18 | [TK]D-Fender | yum |
| 20:07.21 | [TK]D-Fender | yup even |
| 20:07.29 | glaz | Let me try :-) |
| 20:07.45 | glaz | We'll see if you lie! :p |
| 20:08.13 | glaz | dahdi-channels.conf rather? |
| 20:08.19 | glaz | oh, my bad |
| 20:12.39 | CyfordTechnologi | i need some help with my asterisk running in realtime mode.. it seems asterk can see my extenstions in the database but it will not allow these extenstions too register |
| 20:13.25 | WIMPy | glaz: You probably have to enable facilities as well. |
| 20:13.27 | glaz | [TK]D-Fender: by channel configuration it means to my "group" ? |
| 20:13.34 | glaz | facilities? |
| 20:13.50 | WIMPy | facilityenable or something. |
| 20:14.08 | [TK]D-Fender | glaz, it is a channel parm like any other |
| 20:17.29 | *** join/#asterisk jsjc (~Adium@103.Red-2-136-100.dynamicIP.rima-tde.net) |
| 20:21.07 | glaz | I have transfer = yes and facilityenable = yes |
| 20:23.53 | glaz | I get the -- Native bridging DAHDI/1-1 and DAHDI/2-1 |
| 20:24.01 | glaz | but the calls aren't released on the switch |
| 20:24.05 | glaz | eh, PRI |
| 20:24.13 | glaz | any idea? |
| 20:27.15 | glaz | I scared D-Fender... |
| 20:28.01 | WIMPy | You can't scare the fender, you can only piss him off. |
| 20:33.53 | biberao | hi |
| 20:33.54 | biberao | h4 |
| 20:33.56 | biberao | arr |
| 20:35.00 | glaz | WIMPy: it works now, it was on DMS100 |
| 20:35.01 | WIMPy | Are you missing a finger? |
| 20:35.10 | glaz | I switched it to ne2 |
| 20:35.14 | glaz | ni2 |
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| 21:13.08 | *** join/#asterisk Valken (~valken@gw.ptr-80-238-189-250.customer.ch.netstream.com) |
| 21:15.28 | Valken | Hi Back. |
| 21:15.33 | Valken | oups, again. |
| 21:15.51 | Valken | I'm in front of a very strange (to me) problem. |
| 21:16.09 | Valken | I made an AGI Script which behave perfectly when run from the shell (bash) |
| 21:16.19 | talntid | permissions. |
| 21:16.26 | Valken | no |
| 21:16.34 | talntid | chown asterisk:asterisk agiscriopt |
| 21:16.35 | talntid | :) |
| 21:16.36 | Valken | however, when the AGI is called from asterisk, it works nice but always skip the IF statements. |
| 21:16.41 | talntid | oh. odd. |
| 21:16.49 | talntid | I'm guessing, that... |
| 21:17.03 | talntid | it's trying to use a csh shell or something, instead of bash |
| 21:17.13 | Valken | it's set to use python |
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| 21:17.18 | talntid | ah, hm. |
| 21:17.29 | Qwell | skips IF statements? |
| 21:17.41 | Valken | lets say, always consider them false. |
| 21:17.59 | Qwell | Then your condition is false. |
| 21:18.07 | Valken | if I have something like if ("OK == "OK) |
| 21:18.16 | Valken | it doesnt go into the if stuff. |
| 21:18.33 | Valken | Qwell: but if I run it from bash it works fine. |
| 21:18.42 | Valken | I'm getting crazy. :/ |
| 21:18.48 | Qwell | pastebin an example that fails |
| 21:21.27 | [TK]D-Fender | If an AGI runs right from the shell.. then it isn't an AGI |
| 21:21.53 | Qwell | [TK]D-Fender: not necessarily |
| 21:22.08 | Qwell | It could be super simple and require no Asterisk IO |
| 21:22.24 | Qwell | no vars, prompts, etc |
| 21:22.31 | [TK]D-Fender | Qwell: No in, no out = just a dumb script that doesn't qualify as "AGI" |
| 21:23.05 | [TK]D-Fender | Yeah.. my Mazda 3 is a plane ... except that it doesn't fly, have wings, oh and it's actualy a peach basket... |
| 21:24.03 | [TK]D-Fender | Just sayin' ;) |
| 21:24.23 | Valken | ok, I prepared an example and found out it works. So it's time to find out why this hell doesn't work out. ;) |
| 21:30.14 | Valken | hum, I partially found out. |
| 21:40.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:41.36 | Valken | I found out that the sys.stdout.write is not welcome after some operations in the if stuff. |
| 21:41.45 | Valken | some "specific" |
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| 21:49.53 | m_tadeu | I'm trying to use gtalk to receive calls from SIP users. but when I press the answer button(in gtalk), the call is not answered (Auto fallthrough, channel .... status is NOANSWER) |
| 22:09.19 | talntid | Does anyone here know... |
| 22:09.30 | talntid | how to send SMS messages to people in USA, without sending it to the carrier's email? |
| 22:10.55 | [TK]D-Fender | By asomething else that also isn't related to *.... |
| 22:12.17 | talntid | I thought asterisk supports SMS... ? |
| 22:12.32 | [TK]D-Fender | Asterisk is not an SMS platform |
| 22:13.07 | talntid | no more than it is a SIP platform |
| 22:13.54 | [TK]D-Fender | No, Asterisk is a perfectly vialbe SIP user agent. |
| 22:14.06 | [TK]D-Fender | Asterisk does not do SMS. |
| 22:14.12 | [TK]D-Fender | viable* |
| 22:15.00 | talntid | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 22:15.01 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:16.24 | [TK]D-Fender | Which leads to "not in the USE |
| 22:16.26 | [TK]D-Fender | USA |
| 22:16.51 | [TK]D-Fender | "n research the use of this functionality in the US I have been unable to concretely identify a method to mirror the use shown for Europe" |
| 22:17.23 | [TK]D-Fender | Other companies offer API's for this. Use them. |
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| 23:47.39 | sh0ckkk | somebody wanna do me a huge favour? u can make 1$ out of it. setup a buy it now listing on ebay (which is accessible from ebay uk) and label it "ebook: how to tell people who stalk you online and dont even mention it to you to go and die" |
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