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| 00:46.27 | ex-parrot | hi all! where is the dialplan used for the blindxfer (on DTMF # detect) configured? it seems to think every number I dial at the moment is an invalid extension :) |
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| 03:03.05 | ex-parrot | do macros execute in the context they're called from? I am having trouble getting the blindxfer DTMF sequence to accept any extensions |
| 03:05.26 | ex-parrot | ah nevermind, I finally worked out how to enable verbose logging |
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| 05:56.20 | v0lZy | lo |
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| 06:01.36 | v0lZy | hei guys |
| 06:02.17 | v0lZy | 10.5.1 here and i just noticed that adding sound_place_into_conference=none in my confbridge.conf breaks confbridge. |
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| 06:08.09 | diab | Morning. |
| 06:15.54 | v0lZy | moin |
| 06:16.01 | v0lZy | or what was it again? :D |
| 06:17.10 | diab | Matin ^^ |
| 06:20.01 | v0lZy | Matin! :D |
| 06:20.14 | v0lZy | I had a year of french when i was a kid |
| 06:21.10 | v0lZy | obviously it didnt stick .d |
| 06:21.13 | v0lZy | merde.... |
| 06:21.14 | v0lZy | :D |
| 06:21.18 | v0lZy | sacre blu! |
| 06:21.28 | v0lZy | boulongerie! |
| 06:21.36 | v0lZy | just a few from the top of my head .D |
| 06:21.37 | diab | you can stop at merde, with this word you are fluent :)) |
| 06:21.46 | v0lZy | hehe :D |
| 06:22.06 | v0lZy | sexy language though. |
| 06:24.42 | diab | :)) |
| 06:29.32 | v0lZy | btw |
| 06:29.51 | fling | wth is going on here :p |
| 06:30.10 | v0lZy | IF i use 2 extensions in 1 context and use originate... never mind |
| 06:33.06 | v0lZy | Is there a way to originate to more then 1 phone at the same time? |
| 06:34.03 | kaldemar | v0lZy: use more than one Originate. |
| 06:34.26 | v0lZy | then i guess i have to loop |
| 06:34.37 | v0lZy | btw |
| 06:34.51 | v0lZy | right now i have it set up like this |
| 06:35.33 | v0lZy | http://bpaste.net/show/vlQ9XPjm0Sz3v4uBe2dd/ |
| 06:35.42 | kaldemar | why would you have to loop? |
| 06:36.04 | v0lZy | kaldemar: i dont want to think about it in the future when i add/remove phones |
| 06:36.31 | kaldemar | ... |
| 06:36.36 | v0lZy | (btw, sound_place_into_conference=none) breaks confbridge apparently |
| 06:37.01 | v0lZy | anyway... bpaste thingy.. thats what I have now. I'm trying to pass callerid name and num |
| 06:37.25 | v0lZy | when im originating, the phones keep displaying anonymous |
| 06:37.49 | kaldemar | with app Originate, you can't. |
| 06:38.03 | kaldemar | AMI originates and call files allow that. |
| 06:38.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:38.34 | v0lZy | I can if i type it in by hand |
| 06:38.35 | v0lZy | like |
| 06:38.41 | v0lZy | just before Dial(SIP/44) |
| 06:38.52 | v0lZy | I can set callerid |
| 06:39.07 | v0lZy | but I want to set it based on who is calling |
| 06:39.14 | v0lZy | i guess i can use __ |
| 06:39.17 | v0lZy | on variables |
| 06:39.19 | kaldemar | Dial != Originate |
| 06:39.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:41.05 | v0lZy | yeah but if i set for example |
| 06:41.23 | v0lZy | Set(__PAGINGUSER=${ARG2}) |
| 06:41.45 | v0lZy | and then in AutoAnswer extension i use NoOp(${PAGINGUSER}) |
| 06:41.49 | v0lZy | it doesnt return anything :| |
| 06:42.04 | kaldemar | did you read what i just said? |
| 06:42.19 | v0lZy | Yeah...you said that Dial != originate |
| 06:42.25 | kaldemar | and before that? |
| 06:42.35 | v0lZy | ami originates and call files allow that. |
| 06:42.41 | v0lZy | and before that |
| 06:43.17 | v0lZy | with app Originate, you can't. |
| 06:43.36 | kaldemar | ^ |
| 06:43.47 | v0lZy | (but most of the times I dont understand completely what you refer to :|) |
| 06:44.10 | v0lZy | Ok... so when using originate, you cant pickup any variables... at all? |
| 06:44.24 | v0lZy | except globals i reckon? |
| 06:45.22 | kaldemar | an Originate makes a completely new channel that is not associated to the one that executes the application. hence channel variables are not inherited. |
| 06:45.36 | kaldemar | you can use SHARED or GLOBAL variables. |
| 06:47.28 | v0lZy | ok |
| 06:47.47 | v0lZy | Shared sounds an option |
| 06:48.02 | v0lZy | ${SHARED(varname)} retrieves it? |
| 06:49.36 | v0lZy | ah, found the description |
| 06:50.23 | v0lZy | The primary reason for having this space is writing out to another channel, since you can already import variables (and functions) from another channel, with the IMPORT function |
| 06:50.37 | v0lZy | hm, apparently i can import? |
| 06:51.48 | v0lZy | but since i dont know who the caller is in that extension, i probably have to use shared i gather |
| 06:52.44 | v0lZy | but dont i get the same issue? |
| 06:52.55 | v0lZy | I dont know which channels shared variable to refer to. |
| 06:53.07 | v0lZy | if i do |
| 06:54.31 | v0lZy | Set(SHARED(foo)=${ARG1}) for example |
| 06:54.58 | v0lZy | how do i know what channel to reference when i want to pull it back? |
| 06:55.33 | v0lZy | the command says its used for writing out |
| 06:55.44 | v0lZy | so i gather I can specify the chan i want to write to? |
| 06:57.12 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:57.14 | schmidts | good morning |
| 06:58.20 | v0lZy | morning |
| 07:01.54 | v0lZy | kaldemar: How do i fetch the channel originate created? |
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| 07:02.25 | kaldemar | v0lZy: seems that a channel has to exist before you can write to its shared variable space. so that won't do the trick either since the channel is created after setting the variable. |
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| 07:02.44 | v0lZy | so globals it is then |
| 07:06.33 | v0lZy | SetGlobalVar ? |
| 07:06.40 | v0lZy | or Set(GLOBAL(...)) ? |
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| 07:08.46 | bulkorok | hi |
| 07:09.07 | v0lZy | ok, figured it out |
| 07:09.09 | v0lZy | hello |
| 07:09.38 | v0lZy | Now just to loop through all the phones... is there a way to do something like 'sip show peer' in dialplan and filter out all the phones? |
| 07:11.37 | schmidts | v01Zy why you dont use something like Set(__ORIGCHAN=${CHANNEL(name)}) |
| 07:11.56 | schmidts | vars with two _ in front get copied to every channel created from this one |
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| 07:12.32 | v0lZy | even if im originating? |
| 07:13.11 | v0lZy | i tried using __FOO=${ARG1} and then refering back to it |
| 07:13.28 | v0lZy | it didnt carry the value |
| 07:13.43 | kaldemar | schmidts: the channel that Originate creates is not created by the current channel. no inheritance there. |
| 07:14.02 | wdoekes | what are you trying to do v0lZy? |
| 07:15.31 | v0lZy | well, at the moment, im trying to get extensions of all the phones so that I can originate calls to them |
| 07:15.41 | v0lZy | and not worry about it later on when i add/remove phones |
| 07:18.47 | v0lZy | I could keep a variable |
| 07:18.57 | v0lZy | but if theres a way to keep that variable automatically |
| 07:19.00 | v0lZy | that would be better. |
| 07:19.21 | v0lZy | i can do sip show peers in CLI |
| 07:19.26 | v0lZy | and i get all the phones + providers |
| 07:19.37 | v0lZy | phones are all prefixed with SIP |
| 07:19.54 | v0lZy | or actaully |
| 07:19.55 | v0lZy | no, they arent |
| 07:20.09 | v0lZy | but the two others that arent numbers are providers. |
| 07:22.47 | wdoekes | v0lZy: db storage (astdb or func_odbc/realtime) |
| 07:23.04 | wdoekes | I don't think you can enumerate the connected peer without resorting to ugly hacks |
| 07:23.11 | wdoekes | s/peer/peers |
| 07:23.46 | wdoekes | (as for the global thing you're trying to do.. using globals for state is never the right thing to do) |
| 07:24.06 | wdoekes | (unless you have locks) |
| 07:24.52 | v0lZy | im using globals just to fill in some information |
| 07:25.01 | v0lZy | i unset them afterwards |
| 07:25.05 | v0lZy | btw |
| 07:25.09 | v0lZy | is theere an unset command |
| 07:25.11 | v0lZy | or just |
| 07:25.15 | wdoekes | yes.. and if to calls do that simultaneously? |
| 07:25.16 | v0lZy | Set(GLOBAL(name)=) |
| 07:25.21 | wdoekes | s/to/two |
| 07:25.39 | v0lZy | wdoekes: that i plan to put into the dialplan so that it doesnt happen |
| 07:25.58 | wdoekes | using more globals? ;) |
| 07:26.15 | v0lZy | no |
| 07:26.24 | v0lZy | its an intercom application in making |
| 07:26.29 | v0lZy | with confbridge |
| 07:26.42 | v0lZy | logically, if the confbridge im settingup already exists |
| 07:26.53 | v0lZy | then dont continue with execution. |
| 07:27.31 | v0lZy | ill worry about that when i get the basics working |
| 07:27.46 | v0lZy | 1) a list of all phones. 2 a way to loop through it |
| 07:28.10 | v0lZy | if i cant get a list of all phones i can still maintain it manually |
| 07:33.53 | v0lZy | asterisk doesnt have named arrays does it? |
| 07:34.01 | v0lZy | just an option to set several variables at the same time. |
| 07:35.32 | Gugge | look at HASH |
| 07:44.08 | v0lZy | reading |
| 07:44.09 | v0lZy | not much to read |
| 07:44.15 | v0lZy | some stuff is really underexplained. |
| 07:46.10 | v0lZy | si HASH(name,key) |
| 07:46.12 | v0lZy | and HASHKEY(name) to return all of them |
| 07:46.35 | v0lZy | how to return just number 2 for example? |
| 07:47.43 | v0lZy | how to use a comma as a delimiter in fieldqty and cut etc? |
| 07:53.43 | kaldemar | v0lZy: ${HASH(yourhash,${CUT(HASHKEYS(yourhash),\,,2)})} |
| 07:57.03 | v0lZy | ugly award :D |
| 08:01.25 | v0lZy | kaldemar: so its Set(HASH(name,key)=value) or ? |
| 08:03.49 | v0lZy | seems to checkout |
| 08:07.13 | v0lZy | one more questions |
| 08:07.20 | v0lZy | is HASH also local to the channel? |
| 08:07.37 | v0lZy | or can it be made global or something? |
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| 08:42.33 | v0lZy | hm |
| 08:42.37 | v0lZy | i just created an infinite loop |
| 08:42.46 | v0lZy | but i have no idea why its infinite |
| 08:43.46 | v0lZy | http://bpaste.net/show/rCXAbXbWEpVRs0nHLQwH/ |
| 08:43.49 | v0lZy | anyone care to examine this? |
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| 08:47.30 | v0lZy | my bet is on the gotoif |
| 08:47.35 | v0lZy | but i dont know whats wrong |
| 08:51.36 | v0lZy | never mind |
| 08:51.37 | v0lZy | got it |
| 08:51.39 | v0lZy | was using a global |
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| 09:27.21 | vizzy | hello :) just a small question i couldnt dig out a solution, i have exten => s-dosomething,1,Dial(sip/someid1&sip/someid2 mobile|20|owW) |
| 09:27.56 | vizzy | which is working great for someid1 but the identity for the other phone contains a whitespace. |
| 09:28.44 | vizzy | i tried sip/someid1&"sip/someid2 mobile" but didnt help |
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| 09:29.26 | vizzy | so, how can i dial a sip identity with whitespaces? (yeah, i should rename the phone, but thats not possible for some reason at this moment) |
| 09:29.40 | vizzy | the boss cries me a river :)) |
| 09:30.17 | sebastianpersic | HELP, using isymphony to create extensions, user profiles, qeues etc. But it doesn't appear in Freepbx |
| 09:31.12 | kaldemar | sebastianpersic: ask in #freepbx |
| 09:31.19 | sebastianpersic | No one there |
| 09:31.29 | kaldemar | ~freepbx |
| 09:31.30 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 09:33.30 | v0lZy | kaldemar: is there a way for conference bridge to send audio only to one user? |
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| 09:34.00 | v0lZy | im checking options now...i want to do soemthign like press * to talk to admin and press *1 to talk to everyone |
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| 09:36.01 | v0lZy | Its possible to exit and do something in the dialplan |
| 09:36.53 | v0lZy | but is it possible to bridge multiple channels to 1 channel? |
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| 09:57.03 | yang | Who assigns Telephone numbers to providers in European region ? The ENUM provider or ? |
| 09:58.27 | WIMPy | Some local govenment agency. |
| 09:59.13 | yang | right |
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| 10:01.20 | v0lZy | hey ang |
| 10:01.26 | v0lZy | yang* |
| 10:01.46 | v0lZy | hi WIMPy |
| 10:02.20 | WIMPy | hi v0lZy |
| 10:02.49 | v0lZy | doing well i trust? |
| 10:04.01 | WIMPy | Nah. Good times are over. |
| 10:04.07 | v0lZy | :| |
| 10:04.57 | v0lZy | I got my paging app working |
| 10:05.03 | v0lZy | the one where i use conference bridge |
| 10:05.37 | v0lZy | its pretty neat but i need to secure it a bit... |
| 10:06.07 | yang | hello v0lZy |
| 10:06.20 | WIMPy | hasn't tried Page on Confbridge, yet. |
| 10:06.44 | v0lZy | its set so that all the phones originate to the confbridge and then the initiator joins and talks to them... they start muted, but if they press *, they can talk on the intercom... problem though, they talk to everyone |
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| 10:08.17 | WIMPy | Today I'd probably try push-to-talk with milticast. |
| 10:09.07 | v0lZy | walkie-talkie like ? |
| 10:09.27 | WIMPy | yes |
| 10:09.45 | v0lZy | i think all phone conversations should be 1 way like that |
| 10:09.57 | v0lZy | its terrible when people start speaking over eachother on the phone |
| 10:09.58 | v0lZy | i hate it |
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| 10:10.27 | WIMPy | That's only because of the annoying delay you get from using voip. |
| 10:11.23 | v0lZy | no i dont mean that |
| 10:11.26 | v0lZy | i dont get delay |
| 10:11.36 | v0lZy | i mean in general |
| 10:11.38 | v0lZy | phone culture |
| 10:12.03 | WIMPy | Like calling peaple when sitting on the toilet? |
| 10:12.13 | v0lZy | yeah |
| 10:12.20 | v0lZy | or munching stuff |
| 10:12.30 | v0lZy | btwž |
| 10:13.02 | v0lZy | is it possible to check if theres a confbridge active and if it is, stop the dialplan? |
| 10:13.44 | WIMPy | ConfBridges have a device_state. |
| 10:14.15 | WIMPy | But if you ask for names it will not be active during that time, yet. |
| 10:16.45 | v0lZy | I have a Confbridge named AppAnnounce |
| 10:17.09 | v0lZy | just want to check if its active or not so that I dont get 2 users dialing inito the intercom at the same time |
| 10:18.19 | WIMPy | DEVICE_STATE(confbridge:AppAnnounce) |
| 10:20.02 | kaldemar | v0lZy: GotoIf($[${CONFBRIDGE_INFO(parties,1234)} > 0]?hangup) |
| 10:21.15 | kaldemar | ${CONFBRIDGE_INFO(parties,1234)} will reference to the number of participants in the conference. "core show function CONFBRIDGE_INFO" |
| 10:24.05 | v0lZy | nifty |
| 10:24.07 | v0lZy | thanks |
| 10:24.13 | v0lZy | thats even better than what i had in mind |
| 10:27.21 | dax_roc | When I try and place a call I get From: "Anonymous" <sip:393@anonymous.invalid>;tag=as3565a22e in the header. |
| 10:27.48 | WIMPy | had forgotten about that one. |
| 10:28.56 | v0lZy | dax_roc: before u call, do Set(CALLERID(num)=your number) and Set(CALLERID(name)=your name) |
| 10:29.35 | dax_roc | im using originate from the cli, |
| 10:29.44 | v0lZy | WIMPy, kaldemar: Is it possible to have 2 channels to one device? |
| 10:30.30 | v0lZy | dax_roc: then you're in the same situation i was. |
| 10:30.30 | WIMPy | v0lZy: Sure, but you will only have one active channel, off course. |
| 10:30.48 | WIMPy | dax_roc: What are you calling from? |
| 10:31.30 | v0lZy | WIMPy: what im looking to do is to send audio to confbridge, but only to 1 participant (admin) |
| 10:31.41 | dax_roc | I'm trying to place a call to a siptrunk/ext and pass it to a queue in asterisk |
| 10:32.10 | dax_roc | channel originate SIP/telrad/339 application queue DriverQueue |
| 10:32.52 | bulkorok | with an AGI you can call the manager originate wich takes callerid as vars... |
| 10:34.09 | kaldemar | dax_roc: you can't set caller id in the Originate app. only way is to originate to an extension that sets the caller id and then uses Dial. |
| 10:34.17 | WIMPy | dax_roc: Use a local channel and the CALLERID function. Or use another interface. |
| 10:34.42 | kaldemar | why mix AGI in the soup if AMI is all that is needed? |
| 10:34.58 | bulkorok | just came up my mind... |
| 10:41.41 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 10:43.23 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:56.37 | *** join/#asterisk pbxMan (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 10:56.40 | pbxMan | hello |
| 10:56.45 | v0lZy | Executing [callforward@AppInternalCalls:3] Dial("SIP/50-0000016d", "Local/16@AppInternalCalls,,Tt") in new stack |
| 10:56.46 | v0lZy | <PROTECTED> |
| 10:56.47 | v0lZy | <PROTECTED> |
| 10:56.49 | v0lZy | <PROTECTED> |
| 10:56.51 | v0lZy | humph... |
| 10:56.53 | v0lZy | why am i getting congested |
| 10:56.56 | v0lZy | the line's free |
| 11:02.10 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 11:02.35 | kaldemar | there is no "line". |
| 11:04.47 | kaldemar | you don't have an extension in [AppInternalCalls] that would match "16". |
| 11:11.27 | v0lZy | but i do |
| 11:11.43 | v0lZy | well.. no i dont not specified |
| 11:12.35 | v0lZy | but this used to work kaldemar |
| 11:17.35 | *** part/#asterisk jsjc (~Adium@49.Red-81-43-165.staticIP.rima-tde.net) |
| 11:17.39 | v0lZy | though i was probably using somthing allong the catchall... |
| 11:18.42 | v0lZy | ah got it |
| 11:27.20 | *** join/#asterisk riwarren (4645282b@gateway/web/freenode/ip.70.69.40.43) |
| 11:27.41 | riwarren | Anyone have much experience working with MVTS II and routing faxes? |
| 11:35.23 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 11:55.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:57.54 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 12:03.13 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:07.42 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:15.44 | rgsteele | Is there any way to get Asterisk to use a different time zone than what's in /etc/localtime (the system clock)? We just switched to using UTC on all our boxes. Our dialplan makes decisions about what to do with callers depending on what time it is locally at the call center, which used to be easy when using EST (America/NewYork). |
| 12:16.14 | rgsteele | But now that the server uses UTC, I can't just shift the hours in the dialplan forward a couple of hours, because it changes twice a year with daylight savings time |
| 12:16.31 | rgsteele | (We're currently using the GotoIfTime function, which doesn't appear to accept a time zone) |
| 12:18.00 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 12:19.46 | kaldemar | rgsteele: GotoIfTime(times,weekdays,mdays,months[,timezone]?[labeliftrue][:labeliffalse]) |
| 12:20.25 | rgsteele | I've been looking at the Asterisk docs, but didn't see timezone as an available option (http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime) |
| 12:20.42 | leifmadsen | voip-info is woefully out of date |
| 12:20.44 | leifmadsen | use the asterisk wiki |
| 12:20.48 | kaldemar | that's not asterisk docs. |
| 12:20.55 | leifmadsen | since it generates the application docs directly from the source |
| 12:21.02 | leifmadsen | ya, voip-info is just... ugh |
| 12:21.03 | WIMPy | or core show ... |
| 12:21.10 | leifmadsen | use wiki.asterisk.org or www.asteriskdocs.org |
| 12:21.13 | leifmadsen | WIMPy: that |
| 12:21.17 | kaldemar | or use your asterisk to get the docuemntation for your version. "core show application GotoIfTime". |
| 12:22.00 | rgsteele | kaldemar: Yeah, I guess I should've just asked Asterisk... :) |
| 12:22.17 | WIMPy | The bad thing about auto generated doc is however, that they are no more clear than what's in the usually rather brief built-in documentation. |
| 12:22.53 | rgsteele | Fantastic though, just what I needed. Doesn't say what syntax the tz has to be in, but can't be too hard to figure out. |
| 12:24.37 | kaldemar | the timezone option was added in 1.6.2.0 (dec 2009). |
| 12:25.01 | rgsteele | Bleh, someone's gotta have this documented. |
| 12:26.34 | kaldemar | rgsteele: US/Eastern |
| 12:27.08 | kaldemar | /usr/share/zoneinfo/... |
| 12:27.38 | rgsteele | kaldemar: Thanks. Yeah, wasn't sure if that was what it used, or if it wanted something like EST/EDT/etc. |
| 12:29.34 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:31.34 | WIMPy | wonders how many phones (especially phone books) will have limints on the length of numbers that cause issues with the upcomming enlerged number space. |
| 12:31.51 | *** join/#asterisk flujan (~flujan@189.111.106.4) |
| 12:32.44 | flujan | hello guys, I am running asterisk 1.4 and now I wanna try res_fax. Did Digium open sourced it? I saw downloads of two modules res_fax.so an res_fax_digium.so. Do I need both to get fax running? |
| 12:33.26 | [TK]D-Fender | No, one is OOS w/ SpanDSP, the other is closed |
| 12:33.48 | [TK]D-Fender | latter = FFA |
| 12:33.53 | [TK]D-Fender | ~ffa |
| 12:33.53 | infobot | Free For All. Fax For Asterisk |
| 12:34.03 | [TK]D-Fender | Again... latter |
| 12:35.29 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 12:42.05 | flujan | hi [TK]D-Fender thanks for the help. :) |
| 12:55.08 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 12:57.57 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 13:00.38 | eXcAliBuR | my asterisk box had a digium card in it that has a phone cable plugged into it. the phone cable goes to a toshiba strata cix40 call system. when a call is placed from anything using the strata cix40 -- when hungup, asterisk hangs up the channel between it and the cix40, but does not disconnect the channel between asterisk and the sip phone that was called. |
| 13:01.58 | *** join/#asterisk bchia (~Adium@nat/digium/x-bimuemtuqykpfhku) |
| 13:03.07 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 13:04.42 | [TK]D-Fender | ~pb |
| 13:04.43 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:04.44 | [TK]D-Fender | ^^^ |
| 13:05.55 | *** join/#asterisk serafie (~erin@nat/digium/x-xjualsejfgnhmgvp) |
| 13:07.37 | *** join/#asterisk flujan (~flujan@189.111.106.4) |
| 13:12.08 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cytbcruwyeeewyfj) |
| 13:12.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:13.26 | eXcAliBuR | is there an easy way to copy my chan_dahdi.conf? |
| 13:13.30 | eXcAliBuR | it's really long |
| 13:13.37 | eXcAliBuR | to put in pastebin |
| 13:13.53 | WIMPy | Remove the comments first. |
| 13:14.01 | eXcAliBuR | oh god that will take me days |
| 13:14.34 | WIMPy | grep -v ^# |
| 13:15.10 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 13:17.09 | eXcAliBuR | i typed that an nothing happened |
| 13:17.11 | *** join/#asterisk gbit (~no@unaffiliated/gbit) |
| 13:17.15 | eXcAliBuR | do i need a file name? |
| 13:17.51 | gbit | Hello, I have a TDM400 that seems I got 2 channels broken, how can I debug dahdi to figure this out? |
| 13:18.32 | beardy | eXcAliBuR: Yes, you do. Or a pipe to it. |
| 13:19.14 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:19.20 | WIMPy | Should be a ; anyway. |
| 13:19.25 | bitglue | what's a reasonable thing to signal if a Dial() timed out but wasn't anwered? |
| 13:19.34 | WIMPy | grep -v "^;" chan_dahdi.conf |
| 13:19.50 | beardy | > copyofit |
| 13:20.15 | WIMPy | bitglue: 19 |
| 13:20.36 | bitglue | WIMPy: what's 19? |
| 13:20.51 | WIMPy | bitglue: Hangup(19) |
| 13:21.04 | eXcAliBuR | that worked very nicely thanks WIMPy |
| 13:21.10 | WIMPy | "No answer from user (user alerted)" |
| 13:22.00 | bitglue | WIMPy: where are those documented? |
| 13:22.20 | eXcAliBuR | http://pastebin.com/2LGypMpz |
| 13:22.24 | eXcAliBuR | that is to help with my problem |
| 13:22.25 | eXcAliBuR | :) |
| 13:22.34 | WIMPy | bitglue: Q.850 |
| 13:23.01 | wdoekes | you can use NO_ANSWER in the dialplan |
| 13:24.05 | bitglue | gotcha |
| 13:24.52 | WIMPy | eXcAliBuR: But from your description you have a SIP issue. pb a call with debug and verbose enabled. |
| 13:25.25 | eXcAliBuR | that won't show much, when i hangup the phone from cix40, asterisk doesn't show any type of debugging info |
| 13:25.33 | eXcAliBuR | it just shows that the call was answered |
| 13:25.56 | WIMPy | But you said it hangs up the one side of the call. |
| 13:26.08 | eXcAliBuR | yes |
| 13:26.08 | WIMPy | Which didn't sound likely, I have to admit. |
| 13:26.27 | eXcAliBuR | command to set proper debugging? |
| 13:26.44 | WIMPy | core set debug 9 |
| 13:26.53 | WIMPy | core set verbose 9 |
| 13:27.12 | WIMPy | Or just do a 'core show channels' while in the situation. |
| 13:28.37 | eXcAliBuR | http://pastebin.com/Maeqz9H0 |
| 13:30.30 | WIMPy | So both channels stay up. |
| 13:30.40 | WIMPy | That makes a lot more sense. |
| 13:30.53 | eXcAliBuR | ok, i didn't realize both stayed up |
| 13:32.56 | WIMPy | Well, you have to dive in to the config of that pbx to see if there's anything you can enable that will let dahdi detect the hangup. |
| 13:34.39 | eXcAliBuR | so asterisk doesn't know there is a hangup |
| 13:34.43 | eXcAliBuR | thats what it comes down too |
| 13:34.52 | WIMPy | yes |
| 13:35.04 | gbit | There is any way to disable only one dahdi channel? |
| 13:35.09 | WIMPy | ... which is perfectely normal on analog. |
| 13:35.15 | WIMPy | ~cds |
| 13:35.15 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 13:35.30 | WIMPy | gbit: Don't configure it. |
| 13:36.00 | gbit | WIMPy, thanks. There is any way to debug a channel to see if the channel is broken? |
| 13:36.34 | WIMPy | What kind of channel? |
| 13:36.51 | gbit | dahdi |
| 13:37.03 | WIMPy | Can be anything. |
| 13:37.16 | gbit | I have a TDM400, and channel 4 stop working, now channel 3 stop working too. |
| 13:37.36 | *** join/#asterisk MaDiNfO_ (~madinfo@81.92.217.159) |
| 13:37.38 | MaDiNfO_ | hi |
| 13:37.53 | WIMPy | doesn't see any other test than try to use it. |
| 13:38.15 | gbit | So I don't know if I should change the FXO module or buy another card. |
| 13:38.34 | *** join/#asterisk Sean-Der (~sean@diala-bu-078-005.wcnet.org) |
| 13:38.34 | WIMPy | Swap things around. |
| 13:38.38 | MaDiNfO_ | i am connecting a shindler elevator on a FXS port, everything fine but when shindler support guys try to connect to elevator and send DTMF codes asterisk puts call on old.... how can i change this ? and let asterisk pass all dtmf on that extencion in and out ? |
| 13:38.53 | gbit | WIMPy, I will try that, thanks alot. |
| 13:39.27 | WIMPy | MaDiNfO_: Don't enable those features. |
| 13:39.41 | WIMPy | core show application dial |
| 13:40.04 | Sean-Der | Is there a way I could have read feel like an interuptable Playback() ? |
| 13:40.26 | MaDiNfO_ | WIMPy: ? |
| 13:40.38 | MaDiNfO_ | WIMPy: where can i see that configuration |
| 13:40.43 | MaDiNfO_ | and disable all those features |
| 13:40.48 | MaDiNfO_ | for that extencion only ? |
| 13:41.13 | mirela666 | Sean-Der |
| 13:41.15 | WIMPy | Sean-Der: Just use it? |
| 13:41.27 | WIMPy | MaDiNfO_: In your dialplan. |
| 13:41.30 | mirela666 | Sean-Der: Read() |
| 13:41.43 | Sean-Der | I am trying to get rid of the timeout at the end of the Read(), but If I set timeout to 0.1 it will make it impossible for the user to enter the full string if they do interupt it |
| 13:42.06 | MaDiNfO_ | WIMPy: ? |
| 13:42.22 | Sean-Der | Sorry I have been using Read(), but just trying to bend it a little |
| 13:42.56 | WIMPy | MaDiNfO_: Have you ever had contact with Asterisk before? |
| 13:43.22 | MaDiNfO_ | yes |
| 13:43.26 | MaDiNfO_ | no anwser yet |
| 13:43.27 | MaDiNfO_ | :| |
| 13:43.27 | mirela666 | Sean-Der: maybe try Background(sound) + WaitExten() |
| 13:43.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f13e:d553:f26b:7bac) |
| 13:44.27 | WIMPy | Sean-Der: What happens if you don't specify a timeout? |
| 13:44.33 | mirela666 | or just Background lol |
| 13:45.02 | Sean-Der | WIMPy: Won't it just be the standard timeout? let me try it real quick |
| 13:45.08 | MaDiNfO_ | WIMPy: can you help me plz? allready googled and can't find anyone connecting a damn elevator on a asterisk... |
| 13:45.13 | Sean-Der | I am gonna feel like an idiot if it does what I want hah |
| 13:45.14 | mirela666 | Sean-Der: [Synopsis] Play an audio file while waiting for digits of an extension to go to. |
| 13:45.55 | Sean-Der | mirela666: I need to set a variable though. Its a user login that then I Shell() out to a PHP script |
| 13:46.24 | WIMPy | I don't think I ever specified a timeout, but I didn't run in to extra pauses, either AFAIR. |
| 13:47.17 | WIMPy | Do these PINs have a fixed length? |
| 13:47.38 | Sean-Der | WIMPy: I am getting pauses :/ Asterisk 1.8.13 if it makes a difference |
| 13:47.50 | Katty | someone traceroute 64.82.199.210 |
| 13:47.53 | Katty | and tell me if it breaks |
| 13:47.59 | mirela666 | Sean-Der: well while in Background, user enters digits which are found in pattern in same context than dilaplan jumps to that extension and you have a ${EXTEN} value of what user entered :P |
| 13:48.05 | WIMPy | Or make the users press #. |
| 13:48.36 | WIMPy | Yes, but if you don't know the length, WaitExten will also add a timeout. |
| 13:49.13 | Sean-Der | Nope its a variable length login :/ |
| 13:49.32 | Sean-Der | mirela666: You might be on to something though. |
| 13:50.10 | WIMPy | If you don't know the lengt, you can only press #, wait, or read the PIN one digit at a time and call your script after each digit. |
| 13:50.13 | mirela666 | Sean-Der: type in CLI> core show application BackGround |
| 13:50.28 | mirela666 | or BackgroundDetect, and see what fits :) |
| 13:51.07 | bitglue | someone on asterisk-users told me that if i Set(FORWARD_CONTEXT=foo), then any 302 responses my handsets return will be interpreted in "foo", but after testing it, FORWARD_CONTEXT has no effect. Is this accurate? Is there some other way I can handle a 302 response differently? |
| 13:51.36 | WIMPy | With WaitExten you can't even press #. |
| 13:51.47 | Sean-Der | I am gonna pastebin my code, will make it a bit easier |
| 13:52.44 | WIMPy | bitglue: TRANSFER_CONTEXT |
| 13:52.54 | carrar | Katty |
| 13:53.03 | carrar | it LOOPs |
| 13:53.29 | carrar | I expecting nothing less from cogentco |
| 13:54.16 | carrar | You might try using MTR |
| 13:54.56 | carrar | err it doesn't loop sorry |
| 13:55.00 | carrar | heh |
| 13:55.06 | carrar | !@#$where is my coffee |
| 13:55.33 | eXcAliBuR | looks at the coffee in his had |
| 13:55.36 | eXcAliBuR | hand* |
| 13:55.39 | eXcAliBuR | sorry :[ |
| 13:55.43 | carrar | yeah |
| 13:55.44 | carrar | see |
| 13:55.56 | carrar | it's a friday morning reading issue |
| 13:56.05 | carrar | or spelling |
| 13:56.21 | carrar | I should call it a day |
| 13:56.25 | carrar | and go back to bed |
| 13:56.32 | eXcAliBuR | yuh |
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| 13:59.10 | [TK]D-Fender | <Sean-Der> Is there a way I could have read feel like an interuptable Playback() ? <- core show application controlplayback |
| 13:59.54 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:59.59 | Sean-Der | http://pastebin.com/3bRchAcx |
| 14:00.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.08 | dax_roc | I'm wanting to write an automated line check script for testing if an extension can make a call to a fixed number. What would be the best way to check. Dial tone isn't a guarentee, would ringing work ? |
| 14:00.15 | Sean-Der | [TK]D-Fender: Ok will look at that now |
| 14:01.01 | dax_roc | *the extensions are actual gsm routers |
| 14:01.19 | [TK]D-Fender | dax_roc, "can make it" requires "DID make it", which means you have to actually CALL. |
| 14:01.37 | [TK]D-Fender | dax_roc, So place your call and deal witht he result of the attemp. |
| 14:01.54 | Sean-Der | [TK]D-Fender: ${CPLAYBACKSTOPKEY} will only be one key :( |
| 14:02.12 | [TK]D-Fender | read closer |
| 14:02.17 | [TK]D-Fender | heads off for a few... |
| 14:02.53 | MaDiNfO_ | WIMPy: do you know hoe to disable Enable or disable Feature Codes |
| 14:02.57 | MaDiNfO_ | in only one extencion ? |
| 14:03.04 | MaDiNfO_ | or disable all for one extencion ? |
| 14:04.31 | eXcAliBuR | welcome back [TK]D-Fender |
| 14:09.56 | dax_roc | [TK]D-Fender: A little lost, I need to dial a sip/extension and then on dialtone try the fixed number. is that dial(sip/ext&fixednumber)? |
| 14:11.57 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:12.04 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 14:12.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:12.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:12.37 | Hive | Hey [TK]D-Fender, I had to leave after i sent you my pastebin last night |
| 14:12.48 | Hive | about my music on hold predicament |
| 14:12.53 | eXcAliBuR | how does busydetect work? |
| 14:13.06 | bitglue | WIMPy: is a forward not distinguishable from a transfer? I'm hoping to use this to require the called party to accept the call if it's a queue call and the handset is forwarded, because I can't know if voicemail or whatever answered. Probably don't need to take such confirmations for transfers. |
| 14:13.16 | eXcAliBuR | because I added it to my chan_dahdi.conf and it doesn't hangup when it hears the beep beep beep |
| 14:13.20 | Hive | did you have any insignt about that issue, or should ijust do the hack I proposed (change channel music class before the caller is queued, and after they leave the queue) |
| 14:13.50 | newtonr | eXcAliBuR: you may have to configure busycount= to get it working |
| 14:13.58 | eXcAliBuR | i set it to 3 |
| 14:14.48 | bitglue | WIMPy: although, TRANSFER_CONTEXT doesn't seem to work, either :( |
| 14:17.30 | eXcAliBuR | i can't find where to change settings in the strata cix40 to make asterisk detect the hangup, is there a way to debug the busydetect? |
| 14:20.16 | newtonr | with dahdi_monitor you can record the audio stream, otherwise a high DEBUG level on asterisk CLI or logs to see if Asterisk is throwing out any errors or indication of whats happening when you see the busy tone |
| 14:20.45 | newtonr | verbose,debug,warning,error, etc |
| 14:21.51 | dax_roc | After I issue a dial is there a way to check for dial tone? and place a call if so |
| 14:22.29 | dax_roc | I'm dialing a sip/extension and then if there is dial tone dial a pstn/gsm number for testing |
| 14:23.03 | WIMPy | MaDiNfO_: I told you: In your dialplan. Look at the documentation for Dial as stated above. |
| 14:23.48 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:23.58 | WIMPy | bitglue: Only by checking if the call has been answered. |
| 14:24.17 | Hive | Hello, I'm trying to get my musicclass setting of queues.conf to work, however it is apparently overridden by Set(CHANNEL(musicclass)=<themusic>). Is there a way to set music on hold without setting CHANNEL(musicclass)? |
| 14:24.29 | WIMPy | bitglue: It's not really a forward anyway. It's deflection, i.e. a thing in between forwarding and transferring. |
| 14:24.51 | bitglue | WIMPy: is that why TRANSFER_CONTEXT and FORWARD_CONTEXT seem to have no effect? |
| 14:25.01 | eXcAliBuR | newtonr: does my chan_dahdi.conf look right for the busy detect stuff? http://pastebin.com/2LGypMpz |
| 14:25.37 | WIMPy | bitglue: No. |
| 14:25.47 | bitglue | WIMPy: ok, some other idea why, then? :) |
| 14:26.47 | newtonr | eXcAliBuR: no it doesn't actually, the options at the bottom are not being set on any channel |
| 14:26.59 | eXcAliBuR | oh, how do i fix that |
| 14:27.15 | newtonr | eXcAliBuR: the channel definition inherits options set above it |
| 14:27.24 | WIMPy | bitglue: Nope. I only ever tried with tranfer and that works. |
| 14:27.33 | bitglue | WIMPy: http://paste.debian.net/184196/ note that though TRANSFER_CONTEXT was set to confirmation-required, it went to context default anyway. |
| 14:27.50 | newtonr | eXcAliBuR: move the options you want to set on a particular channel, above that channel |
| 14:27.57 | MaDiNfO_ | WIMPy: its on a call that i recieve... |
| 14:28.05 | MaDiNfO_ | elevator sends ## |
| 14:28.18 | MaDiNfO_ | and puts shindler computer on hold... |
| 14:28.21 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:28.51 | WIMPy | MaDiNfO_: Yes, that's what I answered to. |
| 14:29.12 | MaDiNfO_ | i don't know were to start |
| 14:29.14 | MaDiNfO_ | ... |
| 14:29.14 | eXcAliBuR | IT WORKED |
| 14:29.16 | eXcAliBuR | !!! |
| 14:29.19 | eXcAliBuR | everyone see?? |
| 14:29.19 | MaDiNfO_ | i told u i am new to |
| 14:29.21 | eXcAliBuR | ^^ |
| 14:29.22 | MaDiNfO_ | asterisk |
| 14:29.32 | newtonr | eXcAliBuR: yay |
| 14:29.35 | eXcAliBuR | (~._.~) |
| 14:29.38 | eXcAliBuR | ()-*-() |
| 14:29.40 | eXcAliBuR | (_)-(_) |
| 14:29.41 | WIMPy | ~book |
| 14:29.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:29.48 | WIMPy | MaDiNfO_: Start there ^^ |
| 14:29.51 | eXcAliBuR | gives a teddy to newtonr |
| 14:30.04 | eXcAliBuR | yay 3 days later i'm all done |
| 14:30.05 | eXcAliBuR | :} |
| 14:30.23 | MaDiNfO_ | WIMPy: many tanks... |
| 14:30.37 | newtonr | eXcAliBuR: go through the book linked above, it really helps |
| 14:30.45 | eXcAliBuR | i have that book |
| 14:30.49 | eXcAliBuR | i read it last night |
| 14:31.01 | newtonr | read it slower |
| 14:31.46 | WIMPy | You probably have to read it twice anyway. |
| 14:32.31 | *** join/#asterisk CrazyTux[m] (~Brandon@108-64-56-113.lightspeed.irvnca.sbcglobal.net) |
| 14:34.44 | newtonr | i could probably stand to read it a few more times myself |
| 14:37.58 | *** join/#asterisk McBoingBo (~McBoingBo@206.191.52.231) |
| 14:38.58 | McBoingBo | we had a power outage, and now time has been reset, boot log just shows me "|Could not load time from 192.168.129.254(192.168.129.254)" nothing should have changed, so where should I be looking? thanks! |
| 14:40.34 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 14:44.09 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:44.09 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:45.01 | jeffspeff | so, followme... i've got mine set to record the callers name. i've found that it saves all the recordings in /var/spool/asterisk instead of /var/spool/asterisk/followme and the recordings don't get deleted. they just accumulate. is this a bug or a settings issue? |
| 14:45.16 | Sean-Der | Ok much closer! Now I just need to do some regex and sanitize out # if the user enters it |
| 14:46.10 | [TK]D-Fender | <dax_roc> [TK]D-Fender: A little lost, I need to dial a sip/extension and then on dialtone try the fixed number. is that dial(sip/ext&fixednumber)? <- no, you pass the number immediately and there is no such thing as "tone" |
| 14:46.11 | jeffspeff | scratch the /var/spool/asterisk/followme i meant /var/spool/asterisk/tmp |
| 14:46.30 | [TK]D-Fender | dax_roc, SIP is a fully signalled protocol. the OTHER side detects tones and retuns states, etc as applicable |
| 14:46.52 | [TK]D-Fender | McBoingBo, Asterisk is not a time server |
| 14:47.25 | jeffspeff | [TK]D-Fender, what do you mean it's not ntp server? lol |
| 14:47.56 | McBoingBo | [TK]D-Fender, well the IP for the time server IS the Asterisk server, only borked after a bad power outage, but only error I get is Could not load time, so still digging... |
| 14:48.19 | Sean-Der | I am just going to use replace. Any gurus know if this is going to be a stupid idea |
| 14:48.54 | [TK]D-Fender | McBoingBo, If you're running ntpd, etc on there... that is not Asterisk |
| 14:49.13 | *** part/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 14:49.41 | [TK]D-Fender | seanWhat are you trying to get again? |
| 14:49.49 | [TK]D-Fender | Sean-Der, What are you trying to get again? |
| 14:51.06 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 14:51.09 | Assid | hi |
| 14:51.24 | Assid | i am trying to setup my pri card.. i have managed to connect to the vendor.. and am receiving calls |
| 14:51.43 | Assid | however, i am stuck with a situation where i am unable to give a dialtone to the NEC pbx system previously connected |
| 14:52.08 | Sean-Der | [TK]D-Fender: I am just trying to sanitize a variable, I want to remove all '#'s from a var |
| 14:52.18 | [TK]D-Fender | PRI doesn't have dialtone. It is DIGITAL. I think you missed the point entirely |
| 14:52.41 | [TK]D-Fender | Sean-Der, Don't allow it in the first place. |
| 14:52.48 | Assid | i have tried outsignalling=featdtmf |
| 14:52.59 | Assid | [TK]D-Fender: the nec isnt picking up anything.. its like a dead line.. |
| 14:53.01 | [TK]D-Fender | Sean-Der, and Read() allows you to like valid chars or use # as a terminator |
| 14:53.05 | WIMPy | PRI does provide dialtone anyway. |
| 14:53.19 | Assid | im trying to keep the voip box as a "proxy" |
| 14:53.32 | Assid | to receive the connection fromt he modem.. im using pri_cpe .. |
| 14:53.47 | Assid | im just trying to figure out how to interconnect to the NEC if thats my input |
| 14:54.08 | WIMPy | Assid: Do you have a connect at all? |
| 14:54.20 | WIMPy | i.e. is the span geen? |
| 14:54.22 | Sean-Der | [TK]D-Fender: I am allowing people to jump to an extension during a background. Some people are appending # because of old ways |
| 14:54.23 | Assid | WIMPy: its not showing a connection.. thats my problem |
| 14:54.43 | WIMPy | Assid: Did you use a crossover cable? |
| 14:54.46 | Sean-Der | I am just trying to clean it up |
| 14:54.57 | [TK]D-Fender | Sean-Der, Just include your context with the interal extens in it and it will grab before they hit # |
| 14:55.02 | WIMPy | Or junper the port that way if possible. |
| 14:55.04 | Assid | WIMPy: its always been a straight cable.. methings |
| 14:56.09 | WIMPy | Assid: Only if you have the port on your PRI card jumpered for NT mode. |
| 14:56.30 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 14:56.33 | Sean-Der | [TK]D-Fender: Ahh ok thank you |
| 14:57.15 | Assid | WIMPy: i used a straight from the modem -> nec it worked fine |
| 14:57.44 | WIMPy | Assid: Yes, because the NT has a NT interface. Read the link I posted. |
| 14:59.27 | Assid | so i need a crossover from the card to the nec ? |
| 15:01.06 | Assid | also i have 1 more issue.. i am unable to set my own caller id .. |
| 15:01.16 | Assid | err i am in india |
| 15:01.56 | Assid | i am using : cidstart=polarity_in cidsignalling=dtmf |
| 15:01.57 | jeffspeff | Assid, where at in India? |
| 15:02.05 | Assid | jeffspeff: mumbai .. airtel |
| 15:02.18 | Assid | err the pri is airtel's |
| 15:02.31 | Assid | kind of stuck here.. |
| 15:02.32 | WIMPy | Assid: Callerid is transmitted out of band on pri. So these settings have no effect. |
| 15:02.39 | jeffspeff | Assid, oh. we don't have any offices in mumbai |
| 15:02.55 | Assid | the nec is able to set the caller id.. so im trying to find it why |
| 15:03.33 | WIMPy | Assid: You might have a type of number issue as described on page 5. |
| 15:04.42 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:07.11 | [TK]D-Fender | Someone isn't looking at the actual calls... |
| 15:08.44 | Assid | actual calls? |
| 15:09.33 | *** part/#asterisk MaDiNfO_ (~madinfo@81.92.217.159) |
| 15:10.02 | [TK]D-Fender | wonders why people keep repeating back simple actions as questions |
| 15:10.33 | Assid | which actual calls.. |
| 15:11.09 | Assid | WIMPy: i did set prilocaldialplan=unknown didnt help |
| 15:11.23 | [TK]D-Fender | facepalms |
| 15:12.43 | Assid | what did i miss ? |
| 15:14.09 | WIMPy | Assid: Then check that the numbers you set are in the correct format. |
| 15:14.25 | Assid | yes its the same as what the NEC does |
| 15:14.40 | WIMPy | Obviousely not. |
| 15:14.53 | jeffspeff | so, followme... i've got mine set to record the callers name. i've found that it saves all the recordings in /var/spool/asterisk instead of /var/spool/asterisk/tmp and the recordings don't get deleted. they just accumulate. is this a bug or a settings issue? |
| 15:15.19 | WIMPy | Do you have the NEC connected to your Asterisk now? Then go and compare what they send. |
| 15:15.49 | Assid | making the cross over as you suggested |
| 15:15.52 | Assid | as we speak |
| 15:18.05 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 15:20.12 | Assid | WIMPy: okay .. i put the cross over.. but nothing |
| 15:20.29 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 15:21.57 | WIMPy | Assid: Did you configure the port for providing timing? |
| 15:22.13 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 15:22.16 | Assid | nope.. didt know how |
| 15:22.24 | WIMPy | What kind of card it is, BTW? |
| 15:22.39 | Assid | Tormenta 3 (PCI) Quad E1 Card |
| 15:23.02 | WIMPy | Oh. |
| 15:23.17 | Assid | its a temp till the vendor gets me another one |
| 15:23.28 | WIMPy | /etc/dahdi/system.conf |
| 15:23.42 | WIMPy | Set timing to 0 for NT ports. |
| 15:24.36 | Assid | ok.. nothing yet. no dialtone.. busy signal |
| 15:24.48 | Assid | and yes i did put immediate=yes |
| 15:24.50 | WIMPy | Got a link? |
| 15:25.19 | WIMPy | That shouldn't influence the dialtone. |
| 15:25.40 | WIMPy | It rather decides if you call goes to dialplan or if the driver collects digits. |
| 15:25.53 | WIMPy | s/ou/our/ |
| 15:27.57 | [TK]D-Fender | PRI doesn't give dialtone... people are still CRAZY here... |
| 15:28.16 | WIMPy | [TK]D-Fender: Have you ever used one? |
| 15:28.22 | [TK]D-Fender | Yes. |
| 15:28.33 | [TK]D-Fender | Like the one 10 feet behind me for the past 7 years. |
| 15:28.38 | WIMPy | Then maybe you should try again. |
| 15:28.39 | [TK]D-Fender | And all my clients whose cards I set up |
| 15:29.40 | WIMPy | I haven't heard of any telco lines that don't provide dialtine, yet. |
| 15:31.32 | [TK]D-Fender | PRI is digital. |
| 15:31.38 | [TK]D-Fender | CAS is something else |
| 15:31.42 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 15:31.50 | WIMPy | Unrelated |
| 15:31.50 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 15:31.53 | Assid | sory |
| 15:31.57 | [TK]D-Fender | this is not a CAS /RBS line as described. Dialing is OOB. |
| 15:31.59 | Assid | so i got the link |
| 15:32.07 | Assid | and the caller id is the same as im getting |
| 15:32.12 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 15:32.17 | Assid | Span 4: Extension s@from-pstn does not exist. Rejecting call from ' ... |
| 15:32.31 | WIMPy | OOB doesn't mean without audio. |
| 15:32.56 | [TK]D-Fender | You don't sit on dialtone generated by the CARD |
| 15:33.04 | WIMPy | Assid: That's the waitexten thing. I think I wrote than on page 5 as well. |
| 15:33.33 | WIMPy | prefers to sit on a chair |
| 15:34.02 | WIMPy | And yes, I listen to dialtone generated from my PC a lot. |
| 15:35.17 | sruffell | I have to admit, I'm confused as well….I'm pretty sure dialtone doesn't have any part of call setup on PRI lines. Granted, once the call is established something can inject dialtone into it…but it's never needed. |
| 15:35.18 | Assid | yes.. okay.. so once i do waitexten.. i should disa ? |
| 15:35.58 | WIMPy | Assid: No. WaitExten is all you need. |
| 15:36.02 | Sean-Der | I am spolied from PHP... is there anyway to change Timezone or set the time per channel? |
| 15:36.27 | WIMPy | sruffell: Sure, but it's still there. |
| 15:37.05 | sruffell | odd…when I run dahdi-monitor on idle b-channels, I've never heard dialtone. |
| 15:37.42 | Katty | drmessano: wow, not even soda is safe anymore :< |
| 15:37.43 | WIMPy | sruffell: Probably because you always send a called party number immediately. |
| 15:38.10 | Assid | WIMPy: its only taking thefirst extension |
| 15:38.19 | WIMPy | If you just Dial(dahdi/g1/) you can listten to the COs dialtone and dial then. |
| 15:38.35 | WIMPy | Assid: Please explain. |
| 15:38.53 | sruffell | hmm….interesting. I didn't know that was standard. |
| 15:38.57 | WIMPy | [TK]D-Fender: You can try that as well. |
| 15:39.11 | WIMPy | How do you think normal phones work? |
| 15:39.57 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:40.33 | [TK]D-Fender | WIMPy, * won't give a dialtone ot his NEC on inbound like that out of thin air |
| 15:40.38 | Assid | WIMPy: if i do 021222222 .. it gets 0 .. and hence hits my asterisk server.. by it tries to dial extension 2 |
| 15:40.38 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:40.41 | [TK]D-Fender | WIMPy, Call arrives to * with a DID <- |
| 15:40.59 | WIMPy | [TK]D-Fender: It will. |
| 15:41.07 | [TK]D-Fender | If that is blank for having dialed nothin then it will look for a match for "nothing" and if it fails, then call = DEAD |
| 15:41.19 | [TK]D-Fender | And noone is looking at the CALL. |
| 15:41.35 | WIMPy | Yes, but that's still not related to the question of dialtone or not. |
| 15:41.48 | [TK]D-Fender | * won't accept the call and ask for dialtone. |
| 15:41.51 | [TK]D-Fender | That's the poitn. |
| 15:41.52 | WIMPy | Assid: That's your dialplan then. |
| 15:42.02 | [TK]D-Fender | It will try to match... in this case nothing = nothing. |
| 15:42.11 | WIMPy | Assid: You can do immediate=no of fix your dialplan. |
| 15:42.15 | [TK]D-Fender | And a FAILURE to match "nothing" = refues call |
| 15:42.43 | WIMPy | What direction are you on now? |
| 15:42.49 | [TK]D-Fender | Right now we have no proof the card is even seeing a call. |
| 15:42.58 | [TK]D-Fender | Or whats in there now if there does happen to be one. |
| 15:43.09 | [TK]D-Fender | refuse* |
| 15:43.16 | WIMPy | He gets one digit. That's proof enough to me. |
| 15:43.31 | [TK]D-Fender | I don't see anything yes |
| 15:43.40 | [TK]D-Fender | yet* |
| 15:43.52 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:44.28 | WIMPy | I'm not sure if dahdi itself provides a dialtone. But I think even that works. |
| 15:44.57 | [TK]D-Fender | INDICATIONS.CONF does that.... not Dahdi |
| 15:45.11 | [TK]D-Fender | it's dumb audio |
| 15:45.12 | Assid | err will have to do the rest later.. im now being kicked out |
| 15:45.18 | Assid | had to do this while no one here |
| 15:45.21 | [TK]D-Fender | Assid, Next time actually LOOK at something |
| 15:45.39 | Assid | [TK]D-Fender: im able to send signals to the asterisk server |
| 15:45.48 | [TK]D-Fender | Assid, means nothing. |
| 15:45.54 | Assid | i am able to place calls to the phone provider |
| 15:45.55 | [TK]D-Fender | Assid, You aren't lokoing at the call |
| 15:46.02 | WIMPy | If you have immediate=no it's dahdi itself, if you have immediate=yes, it's Asterisk. |
| 15:46.13 | [TK]D-Fender | Assid, You are giving us a fraction of a story and nothing to debug. You have wasted all this time |
| 15:46.25 | Assid | sorry.. not sure what i need to play with here.. |
| 15:46.33 | Assid | will try and get more debugging tomorrow.. once they let me |
| 15:46.56 | Assid | apologies.. and thanks for your patience |
| 15:49.59 | [TK]D-Fender | An hour of never actually looking at the call. Ridiculous. |
| 15:50.23 | [TK]D-Fender | No wonder some people keep coming back in circles here. |
| 15:51.54 | sruffell | WIMPy: You have me curious now (and I don't pretend to be a PRI expert) but looking at the Q.931 spec isn't called party number required in user -> network setup messages? I'm looking at section 3.1.4 of http://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-Q.931-199805-I!!PDF-E&type=items |
| 15:53.32 | sruffell | what gets filled in there when doing standard dtmf based call setup? Wouldn't that be all provider specific, however they have their networks setup? Or is this something that's standard in the Q.931 spec? |
| 15:53.53 | WIMPy | Let me check. |
| 15:54.35 | WIMPy | But I can tell you that a phone won;t send a clled party IE if you just lift the handset. |
| 15:55.12 | WIMPy | Nope, my version ideed say optional. |
| 15:55.38 | WIMPy | says |
| 15:56.12 | sruffell | that one I linked says optional for network -> user….but mandatory from user -> network |
| 15:56.50 | WIMPy | It can still be empty :-) |
| 15:57.29 | sruffell | true. |
| 15:57.45 | WIMPy | just confirmed with 3 brands pf phones. None sends a clled party IE. |
| 15:57.54 | WIMPy | of |
| 15:58.08 | sruffell | phones? are you talking about BRI and not PRI? |
| 15:58.28 | WIMPy | Where's the difference? Apart from the number of channels? |
| 15:58.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 15:58.33 | sruffell | the protocol is different. |
| 15:58.56 | sruffell | for just this sort of thing. :) |
| 15:58.57 | WIMPy | The coding of the channel identification, yes. |
| 15:59.04 | WIMPy | And PRIs are always PTP. |
| 15:59.16 | WIMPy | Anything else? |
| 15:59.37 | sruffell | goes to look |
| 16:00.33 | WIMPy | Might be fun to do a PTMP pri impementation :-) |
| 16:01.26 | WIMPy | But PRIs may lack the E channel required for that to work, I think. |
| 16:01.30 | *** join/#asterisk albertoandrade (~albertoan@200.195.146.34) |
| 16:01.33 | WIMPy | But that's a different layer. |
| 16:06.09 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 16:06.22 | *** join/#asterisk Consolas (~Consolas@108-231-29-20.lightspeed.miamfl.sbcglobal.net) |
| 16:06.24 | Consolas | hey guys |
| 16:08.54 | Consolas | having this exten => s,1,Dial(SIP/${SOME_VARIABLE:2}@${ANOTHER_VARIABLE},40) |
| 16:09.20 | Consolas | I want the another variable to ceom from a direct url request but in case it is not define, I want to have a standard valkue |
| 16:09.52 | Consolas | should I do something like exten => 100,1,SetGlobalVar(ANOTHER_VALUE=the_value) |
| 16:11.02 | Consolas | and then the other ext will call this variable, if it is set the value will change, if not, it would get eh previously set value |
| 16:12.19 | WIMPy | That would be one way of doing it. |
| 16:13.30 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:14.20 | Consolas | is there a better way? |
| 16:15.14 | [TK]D-Fender | Consolas, variables do not have a context scope in the dialplan |
| 16:15.46 | [TK]D-Fender | "global" variables are across multiple CALLS on the system that may not be at all related to one another. |
| 16:16.01 | Consolas | ok so using global would be an overkill here |
| 16:16.20 | Consolas | can't I just do somthing like if variable not set, use this value, else use variable |
| 16:16.22 | Consolas | something like this |
| 16:16.28 | Consolas | Im fairly new to this |
| 16:17.15 | [TK]D-Fender | You can check if a var is set.... that's the point |
| 16:17.23 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:18.13 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:18.13 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 16:20.54 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
| 16:21.14 | geek0091 | Got a quick question with asterisk and extensions with pots. |
| 16:21.17 | Consolas | yes that is what I want |
| 16:22.12 | [TK]D-Fender | Consolas, The go do it... "core show function IF". |
| 16:22.36 | [TK]D-Fender | geek0091, Ask away |
| 16:22.48 | Consolas | cool, wil check it |
| 16:22.52 | Consolas | thank you very much Fender |
| 16:23.21 | geek0091 | When dialing into a pots line from an external phone, sip and analog phones on the line ring. After this, if answering an analog phone, the fxo connected sip phones continue to ring for about 7 seconds. Anyway to fix this? I have an idea it might be busy detection but its puzzling me. |
| 16:23.59 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 16:25.51 | [TK]D-Fender | geek0091, First.. you need to properly describe what is in the chain of this call... |
| 16:26.41 | geek0091 | By chain what do you exactly mean? Sorry im just now getting into pbx and fxo based systems. |
| 16:27.13 | [TK]D-Fender | You haev not properly described what precise phones, or what yuo are using to conenct to PTS with. |
| 16:27.17 | [TK]D-Fender | POTS* |
| 16:27.52 | Consolas | ${IF($[ ${VARIABLE} = ""]?STANDARD_VALUE:VALUE_FROM_URL)} |
| 16:28.04 | Consolas | what do you think Fender? |
| 16:28.22 | geek0091 | x100p to zaptel/asterisk to polycom sounpoint IP phones. |
| 16:29.04 | [TK]D-Fender | Consolas, the "" in your expression are LITERAL. they need to exist on both sides. |
| 16:29.15 | [TK]D-Fender | geek0091, Ok, now pastebin a call for us to look at. |
| 16:29.16 | [TK]D-Fender | ~pb |
| 16:29.16 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:29.17 | [TK]D-Fender | ^^^ |
| 16:30.04 | Consolas | sorry, didnt understand, Fender |
| 16:30.27 | Consolas | doesn't this states if variable is empty set it to standard else use the value from url ? |
| 16:31.49 | [TK]D-Fender | Consolas, you need "" on the LEFT side of the = as well |
| 16:32.07 | [TK]D-Fender | ${IF($["${VARIABLE}" = ""]?STANDARD_VALUE:VALUE_FROM_URL)} |
| 16:32.08 | [TK]D-Fender | ^ |
| 16:32.19 | Consolas | oh.. |
| 16:32.22 | Consolas | wow, strange |
| 16:32.53 | Consolas | not used to this hehe have a strong background in developing but this is somewhat strange |
| 16:32.54 | Consolas | thanks |
| 16:34.00 | [TK]D-Fender | This is an Asterisk quirk... it isn't "logical" to those working with "real" languages... |
| 16:34.11 | [TK]D-Fender | For teh fact of not having typed variables, etc |
| 16:34.44 | geek0091 | http://pastebin.com/ZxFc27PP |
| 16:40.18 | Consolas | fender: exten => s,1,Dial(SIP/${something:2}@${IF($["${provider}" = ""]?IP:${provider})}, |
| 16:40.22 | Consolas | this should work, right |
| 16:41.19 | [TK]D-Fender | Depends on the variables |
| 16:41.23 | [TK]D-Fender | Go try it.. |
| 16:42.08 | [TK]D-Fender | geek0091, dial without the "r" dial option.... I see no confirmation of answer. pastebin your zapata.conf and everything INCLUDE-d in it |
| 16:42.33 | Consolas | yes, disregarding the values, the logic is correct right? |
| 16:42.50 | [TK]D-Fender | Consolas, looks about right |
| 16:43.17 | Consolas | thank you very much |
| 16:43.38 | [TK]D-Fender | geek0091, Also... I don't see that phone answer anywhere there |
| 16:44.43 | sruffell | WIMPy: always learning. Not BRI specific…but what you described is in the spec as part of the overlapped dialing mechanism. Section 5.1.3 specifically calls for dialtone to be provided while waiting for more information elements that contain digits which are sent on the dchannel. |
| 16:45.11 | sruffell | <— always learning |
| 16:45.30 | WIMPy | Ok, now we need to convince [TK]D-Fender :-) |
| 16:45.54 | WIMPy | Or sa said above: What did you think how a normal phone works? |
| 16:46.33 | WIMPy | And the whole thing reminds me that I need to debug exately that. |
| 16:47.25 | cresl1n | <—— Always learning too :-) |
| 16:47.51 | [TK]D-Fender | <--- still never seeing debug for people who claim to want to get things working. |
| 16:47.52 | sruffell | I'm still don't know how common overlapped dialing on PRI is in the U.S….so you both may be right. |
| 16:48.01 | cresl1n | It's not used in the USA |
| 16:48.11 | geek0091 | Fender: It does not, an analog phone answers. But the sip phone keeps ringing past answered for some time. |
| 16:48.11 | sruffell | hey cresl1n! |
| 16:48.15 | cresl1n | Overlap isn't used at all |
| 16:48.21 | cresl1n | hey sruffell!!! |
| 16:48.51 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 16:48.57 | WIMPy | cresl1n: So you always have to have a send key or wait for a timeout? |
| 16:49.04 | cresl1n | WIMPy |
| 16:49.19 | cresl1n | WIMPy: We have fixed digit width numbering plans |
| 16:49.40 | WIMPy | What if you call abroad? |
| 16:49.50 | [TK]D-Fender | WIMPy, BRI doesn't realistically exist in North America and many other places. And PRI dialing is done with the entire number already sent |
| 16:49.52 | cresl1n | I think it uses a timeout |
| 16:50.21 | WIMPy | [TK]D-Fender: Forget about there being a difference. |
| 16:50.35 | WIMPy | There isn't really. |
| 16:50.46 | [TK]D-Fender | Yeah, 'cause it's not like that is a point that is being debated or anything .... |
| 16:50.54 | WIMPy | And timeouts are really annoying. |
| 16:51.28 | cresl1n | Well, in the USA, I think for the average person it's rare to dial outside of the USA |
| 16:51.32 | [TK]D-Fender | And all of that is beisdes the point. We never saw a call. For al we know the call never even hit the channel driver level. |
| 16:51.36 | [TK]D-Fender | We saw NOTHING |
| 16:51.43 | [TK]D-Fender | so arguing his case is a waste of time |
| 16:51.52 | WIMPy | I did. |
| 16:52.27 | [TK]D-Fender | WIMPy, I sus didn't and he offered only a "?" in response to being told he wasn't showing the call. |
| 16:52.32 | [TK]D-Fender | sure* |
| 16:52.38 | WIMPy | Maybe you overlooked the fact that he made progress several times while arguing that PRIs don;t provide dialtone. |
| 16:53.42 | [TK]D-Fender | WIMPy, "no debug" <- I didn't ask for "Story time". I don't want someone's mangled interpretation of things. i want hard proof. |
| 16:54.58 | [TK]D-Fender | I don;t care what you think you hear on a line. want debug data that proves that something made it to * and how * reacted to it. |
| 16:56.03 | [TK]D-Fender | Everything else is a subjective mess who's actual reason for having happened cannot be proven. |
| 16:56.14 | [TK]D-Fender | PRI would have PRI DEBUG. |
| 16:56.43 | WIMPy | That would be the next step for him to compare caller ID settings. |
| 16:56.57 | [TK]D-Fender | If you can't enable that and show us then you are pretty much useless. Without it I don't even trust that the call ARRIVED or was accepted in any capacity |
| 16:57.06 | [TK]D-Fender | And Tormenta cards aer BEYOND ancient. |
| 16:57.25 | [TK]D-Fender | He is using decrepit crap and not backing a single bit of it up. |
| 16:57.38 | WIMPy | How does the age of the hardware come in to play? |
| 16:57.44 | [TK]D-Fender | First step is proving somethine arrived at all. |
| 16:58.05 | [TK]D-Fender | Do I trust the card is working? How about drivers? What is he running? |
| 16:58.31 | [TK]D-Fender | Basically you can waste all the time you want but he didn't do the very first most basic steps in proving something even arrived |
| 16:58.44 | [TK]D-Fender | So guessing settings, etc is a complete waste of time. |
| 16:58.53 | [TK]D-Fender | Sure, keep on telling stories.. that's going to get you somewhere.... |
| 16:59.26 | WIMPy | Seems to have worked out for him. |
| 17:00.32 | [TK]D-Fender | His situation is still not solved and continues to debug ... without debugging. |
| 17:01.08 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 17:02.27 | [TK]D-Fender | ~wglwat |
| 17:02.27 | infobot | hmm... wglwat is well, good luck with all that |
| 17:03.00 | [TK]D-Fender | Maybe he'll start being thorough on his next pass.... |
| 17:05.02 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 17:08.35 | dhuckaby | anyone got asterisk work'n with googlevoice? |
| 17:09.02 | *** join/#asterisk Sidrov (~sid@85.186.159.201) |
| 17:09.38 | Sidrov | is there any way to disable beeps in asterisk 1.8 RECORD command ? |q has no effect |
| 17:10.05 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 17:14.39 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:18.22 | Katty | oh god |
| 17:18.23 | Katty | so full |
| 17:18.30 | Katty | sprawls on floor |
| 17:18.42 | pabelanger | Sidrov: pipe (|) is not a valid. Try comma (,) |
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| 17:21.57 | [TK]D-Fender | I've never heard of Record beeping by default... |
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| 17:24.50 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:25.28 | WIMPy | Ok, mixed dialling fixed. |
| 17:26.06 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
| 17:28.11 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
| 17:32.19 | anonymouz666 | oh my, the new range rover is... ugly. |
| 17:33.24 | [TK]D-Fender | anonymouz666, You clearly want a Nissan Cube, Pontiac Aztec, or Chevy PT Cruiser ;) |
| 17:34.10 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
| 17:36.04 | anonymouz666 | haha |
| 17:36.09 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
| 17:37.33 | Katty | ohhhhhh where is my hair brush! |
| 17:37.58 | anonymouz666 | http://www.autogaleria.pl/fotografie/index.php?id=53374&news=4954 -> this is the new |
| 17:38.07 | *** join/#asterisk geek0091 (~chatzilla@unaffiliated/geek0091) |
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| 17:46.14 | anonymouz666 | sangoma released a 16-span card |
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| 17:49.01 | dhuckaby | anyone using the googlevoice module? |
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| 18:10.57 | navaismo | o/ |
| 18:16.10 | paulc | Anyone got any favourite providers of dedicated hosting for Asterisk boxes? |
| 18:16.45 | paulc | c |
| 18:17.22 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:22.03 | jeffspeff | me |
| 18:22.13 | jeffspeff | i love hosting and providing my own stuff |
| 18:22.16 | jeffspeff | lol |
| 18:22.43 | WIMPy | aol |
| 18:22.53 | jeffspeff | i like aol too |
| 18:23.19 | WIMPy | Do they still exist? |
| 18:23.26 | jeffspeff | there's a good joke there... somewhere |
| 18:23.26 | jeffspeff | lol & aol |
| 18:23.31 | jeffspeff | i think they do |
| 18:23.41 | jeffspeff | yep |
| 18:23.43 | WIMPy | wow |
| 18:23.46 | jeffspeff | http://aol.com |
| 18:23.53 | jeffspeff | i think we just did the same thing |
| 18:24.55 | WIMPy | Oh, actually I think I'm on AOL economical wise. |
| 18:25.51 | jeffspeff | the last thing i heard about them is that they determined something like 70% of their paid subscription base doesn't actually use their internet anymore, that they actually have cable or dsl and only continue to pay their AOL bill because they think AOL is the magic that makes the internet work... like their cable connection wouldn't give them internet with also having an AOl account |
| 18:26.31 | WIMPy | Cool |
| 18:26.33 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:26.45 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:26.53 | jeffspeff | Cool for aol but not cool the US |
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| 18:27.02 | drmessano | They even charged like $10 a month to continue using the GUI over DSL/Cable |
| 18:27.09 | WIMPy | That surely explains why they still exist. |
| 18:27.13 | drmessano | Bring-Your-Own kinda thing |
| 18:27.14 | jeffspeff | yep |
| 18:28.54 | drmessano | I thought it was extremely pompous of them to think people would spend all day in one application so they could share content and chat with their friends, etc. |
| 18:29.03 | drmessano | Oh, wait |
| 18:29.13 | jeffspeff | lol |
| 18:29.26 | jeffspeff | fb say what? |
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| 18:33.19 | paulc | Is Compuserve still around too? |
| 18:34.07 | paulc | And yes, I love hosting and providing my own stuff too.. but for this particular project, we don't fancy forking out for a server and term/dedicated bandwidth.. (blame TPTB and the PHBs) |
| 18:37.50 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:37.50 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 18:38.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:38.21 | jeffspeff | so, followme... i've got mine set to record the callers name. i've found that it saves all the recordings in /var/spool/asterisk instead of /var/spool/asterisk/tmp and the recordings don't get deleted. they just accumulate. is this a bug or a settings issue? |
| 18:39.07 | Sidrov | how can I disable RECORD beeps in asterisk 1.8 ? |q has no effect |
| 18:39.21 | pabelanger | Sidrov: I already told you, | is not valid |
| 18:39.24 | pabelanger | switch to , |
| 18:39.51 | Sidrov | pabelanger switch to what ? q also is not valid |
| 18:39.58 | Sidrov | pabelanger: u mean ,q ?? |
| 18:40.05 | drmessano | ,q |
| 18:40.09 | drmessano | Not |q |
| 18:40.14 | drmessano | Pipes are gone |
| 18:40.15 | Sidrov | got it |
| 18:40.17 | pabelanger | ^this |
| 18:40.17 | Sidrov | thanks |
| 18:40.27 | jeffspeff | Sidrov, the pipe | was deprecated quite some time ago... it was replaced by the comma , |
| 18:40.41 | Sidrov | thanks jeffspeff |
| 18:40.52 | WIMPy | About 5 years. |
| 18:42.36 | Sidrov | ,q no effect also :(((( |
| 18:42.42 | Sidrov | it gives me two beeps |
| 18:42.50 | Sidrov | one starting, one stop i guess |
| 18:44.45 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 18:45.25 | jpsharp | Anyone know the magic incantation to load the right Asterisk-making-happy SQLite3 package on Ubuntu 11.10 |
| 18:45.37 | drmessano | jpsharp: Hang on |
| 18:46.14 | drmessano | libsqlite3-dev |
| 18:46.51 | jeffspeff | jpsharp, nice, thats funny |
| 18:47.46 | jeffspeff | if you rub your belly, pat your head, hop on one foot while turning counter-clockwise and singing the hokey-pokey your odds of success are dramatically increased |
| 18:48.23 | jpsharp | My wife bought me a rubber chicken to wave over the top of recalcitrant hardware. |
| 18:49.07 | jeffspeff | lol |
| 18:49.44 | jeffspeff | i still need to figure out the voodoo magic behind scripting the make menuselect command |
| 18:49.44 | jpsharp | drmessano: And that was exactly it. Thanks muchly. |
| 18:49.51 | jeffspeff | and subsequent selections |
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| 18:55.23 | Sidrov | I was wrong. My question was about: AGICommand_record file |
| 18:55.28 | nafg_ | Hi, can someone help me, I just installed asterisk on a new (dev) system, and the sip phone is not connecting to it. |
| 18:55.31 | Sidrov | RECORD FILE FILENAME FORMAT ESCAPE_DIGITS TIMEOUT OFFSET SAMPLES BEEP S=SILENCE |
| 18:55.36 | Sidrov | can't disable beep |
| 18:56.10 | nafg_ | I have [1000] / type=friend /context=phones / host=dynamic |
| 18:56.13 | nafg_ | in sip.conf |
| 18:56.24 | nafg_ | <PROTECTED> |
| 18:57.09 | WIMPy | No secret? |
| 18:58.15 | nafg_ | Never mind, it's working |
| 18:58.25 | nafg_ | WIMPy: no, it's my developing laptop |
| 19:00.31 | *** join/#asterisk onixx (1000@bas1-stetherese38-2925308036.dsl.bell.ca) |
| 19:01.01 | Sidrov | AGICommand_record file -> BEEP NOBEEP Can't diable damm beep :(( |
| 19:01.19 | Sidrov | is it a bug ? |
| 19:01.23 | onixx | hello, I checked out a few version of svn trunk in the past few days and I am getting lots of these failed to extend from 1024 to 1298 on the console |
| 19:01.32 | onixx | anybody gets these too ? |
| 19:02.48 | WIMPy | onixx: Try icesupport=false in rtp.conf |
| 19:05.29 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:06.10 | Sidrov | pabelanger are you still here ? |
| 19:06.29 | Sidrov | is there a bug in AGICommand_record -> BEEP |
| 19:06.30 | Sidrov | ? |
| 19:06.52 | nafg_ | Now I'm getting an error that the sip phone failed to open port 5060 , already in use |
| 19:07.20 | jpsharp | Are you trying to run a SIP phone and Asterisk on the same box? |
| 19:07.47 | nafg_ | jpsharp: YEs |
| 19:07.53 | nafg_ | my dev laptop |
| 19:08.03 | jpsharp | Not gonna work. They're both trying to bind to port 5060 to receive SIP messages. |
| 19:08.15 | nafg_ | What do you mean, I do it all the time. It worked a few minutes ago. |
| 19:08.21 | jpsharp | Unless you change the bind port on one of them. |
| 19:08.30 | onixx | WIMPy: will don |
| 19:08.32 | nafg_ | Well then how do they talk to each other? |
| 19:09.03 | nafg_ | Asterisk has to listen on 5060 as a socket server, and the phone has to open the connection, no? |
| 19:09.29 | WIMPy | Yes, but from another port, as jpsharp explained. Basic networking rules. |
| 19:10.04 | jpsharp | the SIP client is also trying to listen on port 5060 for messages from Asterisk or another SIP server. |
| 19:10.17 | nafg_ | K I changed the phone port, it's working |
| 19:11.03 | onixx | WIMPy: works !! no more issue with icesupport=false. any clue why that is ? I seems my ipad with bria was triggering these. It also could only make calls and not receive any before I changed icesupport=false |
| 19:13.06 | dhuckaby | anyone know what firewall ports need forwarded to the Asterisk box for incoming calls over a NAT'd firewall? |
| 19:14.02 | WIMPy | onixx: The errors you were seeing are probably caused by your server havong too many interfaces. ICE will list all your local IPs. |
| 19:15.13 | onixx | WIMPy: 7 interfaces. should i report an issue on this or just not bother ? |
| 19:15.48 | WIMPy | You can comment on mine :-) |
| 19:16.48 | WIMPy | https://issues.asterisk.org/jira/browse/ASTERISK-20088 |
| 19:16.57 | WIMPy | Or re-open it. |
| 19:17.29 | WIMPy | I guess we need a bug marshall to re-open it. |
| 19:19.29 | onixx | WIMPy: I was getting "Bad SDP" sip errors when calling the IPAD Bria and these failed to extend |
| 19:19.49 | nafg_ | What's the best way to give asterisk write access to a subdirectory of $HOME? |
| 19:19.50 | WIMPy | Because the message was truncated. |
| 19:19.52 | onixx | WINPy: seems similar to your issue but not quite exactly the same |
| 19:19.55 | WIMPy | That makes sense. |
| 19:20.09 | WIMPy | It's part of it. |
| 19:20.35 | onixx | ok ! I will comment on you issue and hopyfully it gets somebody's attention |
| 19:20.36 | WIMPy | It two parts: Duplicate IPs and truncated sdp. |
| 19:21.56 | onixx | do you know by hear the command to gether my revision from my current svn directory ? |
| 19:21.59 | onixx | hearth |
| 19:22.21 | WIMPy | core show version from *CLI |
| 19:22.39 | onixx | thanks |
| 19:25.05 | onixx | WIMPy: thanks for all the help !! I just posted on your issue. |
| 19:39.59 | *** join/#asterisk joshaidan (~brianj@S010698fc113e438d.tb.shawcable.net) |
| 19:41.55 | Sidrov | is there a bug in AGICommand_record -> BEEP ? |
| 19:42.37 | dhuckaby | is there a FREEPBX channel? |
| 19:42.50 | [TK]D-Fender | #freepbx <- |
| 19:42.55 | WIMPy | dhuckaby: Guess the name. |
| 19:54.06 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:54.48 | dhuckaby | I'm having a @#!%! of a time trying to get google voice working with it |
| 19:55.00 | dhuckaby | some googling says maybe I'm missing modules of jabber and gtalk |
| 19:57.17 | jpsharp | dhuckaby: IF you're using < Asterisk 11, then you need those two chanels. |
| 19:57.19 | jpsharp | channels, too. |
| 19:58.02 | jpsharp | They have some dependencies, so they won't build unless you satisfy those dependencies before running ./configure for asterisk. |
| 19:58.20 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
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| 20:40.22 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:43.44 | eduzimrs | Hi, i have my sip usernames set to Mac Addr, how could i set in each peer to e.g: Callerid and at cdr logs show me a number like 400 instead the Mac ? is it possible? |
| 20:44.24 | WIMPy | Can you re-phrase that? |
| 20:46.12 | eduzimrs | ok, i want to put all my sip username different from the extensions, to put more security |
| 20:46.43 | WIMPy | Makes sense. |
| 20:46.44 | eduzimrs | i`d like to rename all of them to Mac Add from the client |
| 20:46.48 | WIMPy | What's the question? |
| 20:47.46 | WIMPy | A lot harder to guess than some 2 or 3 digit number, but not *that* hard. |
| 20:47.56 | eduzimrs | so, how could i mantain the numbers in the cdr logs e.g: to appear the old usernames like 400 instead of the Mac Addr |
| 20:48.10 | jpsharp | Set the callerid= in the sip.conf |
| 20:48.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:48.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:48.43 | WIMPy | Or just append them to the mac. |
| 20:48.49 | eduzimrs | will be the same to Dial ? will apear 400 instead Mac ? |
| 20:49.06 | eduzimrs | will be the same to Dial ? will apear 400 instead Mac Adr ? |
| 20:49.17 | WIMPy | You dial an axtension and that Dial()s a name. |
| 20:49.33 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 20:49.47 | eduzimrs | oww |
| 20:49.49 | eduzimrs | yeap |
| 20:50.23 | eduzimrs | callerid=Bla <400> worked fine! |
| 20:50.29 | eduzimrs | tks guys |
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| 21:44.41 | dijib | anybody know anything about PAP2 SPA2102-R and WRT54GP2 |
| 21:46.25 | dijib | i kindof got them for $20 so its worth a shot getting them going to me. |
| 21:46.40 | dijib | is there a hardware channel i should be in? |
| 21:47.42 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:55.02 | jpsharp | I've seen lots of people use the PAP2 with Asterisk. |
| 21:58.43 | *** part/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 22:01.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ykaedgrfiovtwcxo) |
| 22:14.40 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-lvshdfkduecnqbmz) |
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| 22:49.26 | dijib | if their vonage devices are they locked ? |
| 22:53.54 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:53.54 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:57.32 | jpsharp | Most likely, yes. |
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| 23:23.04 | jpsharp | you might be able to unlock them. |