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| 00:07.59 | ectospasm | tim____: they're not part of the mainline kernel... you'd have to use make menuconfig from the DAHDI source. |
| 00:08.16 | ectospasm | I honestly don't remember if menuconfig even exists for DAHDI, I haven't used it in so long. |
| 00:10.18 | tim____ | it doesn't seem to; there's a Kconfig in /drivers/dahdi, but |
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| 00:36.20 | Neptu | - Got SIP response 486 "Busy Here" back from 5.77.36.136:5060 -> Any idea what this mean? |
| 00:36.57 | ectospasm | Neptu: that usually means the endpoint is tied up for some reason |
| 00:37.18 | Neptu | ectospasm, so no more lines? |
| 00:37.36 | ectospasm | so the busy message gets played |
| 00:37.42 | ectospasm | er |
| 00:37.49 | Neptu | ok |
| 00:37.56 | slav3_kitten | 486 Busy Here mean the callee is busy. as per wiki |
| 00:38.04 | slav3_kitten | means* |
| 00:46.38 | Neptu | just want to understand if the trunk was busy or the phone was busy |
| 00:46.40 | Neptu | that its all |
| 00:47.24 | WIMPy | Yes. I really miss some meaningfull feedback. |
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| 01:22.24 | igcewieling | many phones send back a 486 Busy Here when they have DND enabled. |
| 01:22.35 | igcewieling | ah, he's gone already |
| 01:25.28 | robert_ | so RudyValencia and I are writing a basic IVR system, our problem when it tries to transfer to either of us and we decline it (or are busy), it falls through. we're wondering how to transfer you to voicemail instead |
| 01:26.41 | WIMPy | See DIALSTATUS and for additional information HANGUPCAUSE. |
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| 01:44.25 | robert_ | it says "Nobody picked up in 1000 ms" |
| 01:44.33 | robert_ | and then just dumps the call |
| 01:44.38 | robert_ | (different issue) |
| 01:44.55 | WIMPy | Same answer. |
| 01:45.13 | robert_ | I mean how do I get it to ring longer? |
| 01:45.33 | robert_ | for some reason, Dial() doesn't want to ring for longer than 1ms |
| 01:45.39 | WIMPy | You specify the timeout. |
| 01:45.39 | robert_ | er 1000ms |
| 01:45.48 | robert_ | yeah but it seems to be ignoring the timeout. |
| 01:46.08 | robert_ | is it different for macros? |
| 01:46.31 | WIMPy | What do macros have to do with it? |
| 01:46.49 | robert_ | half of the dialplan is in two different macros. |
| 01:47.01 | WIMPy | If Dial stops there are either no phones left to ring or the timeout is up. |
| 01:49.16 | [TK]D-Fender | robert_: Your dial is wrong. |
| 01:51.18 | robert_ | [TK]D-Fender: exten => s,1,Dial(${MACRO_EXTEN},40,m) |
| 01:51.33 | robert_ | [TK]D-Fender: I can also post our dialplan if that helps. |
| 01:53.05 | [TK]D-Fender | robshow us the call |
| 01:53.10 | [TK]D-Fender | ~pb |
| 01:53.10 | 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. |
| 01:53.13 | [TK]D-Fender | ^^^ |
| 01:56.57 | robert_ | yes |
| 01:57.10 | robert_ | the call detail or the dialplan? (sorry) |
| 02:10.03 | [TK]D-Fender | the CALL |
| 02:20.24 | [TK]D-Fender | Time's up, I'm out for a bit. |
| 02:22.39 | robert_ | okay. |
| 02:26.41 | robert_ | http://pastebin.com/TfvTwKDS |
| 02:32.17 | WIMPy | Looks like your dialplan just ended there. |
| 02:35.45 | robert_ | hm |
| 02:36.18 | robert_ | it sort of "ends", even though we have (or think we do, anyway) it forwarding to voicemail. |
| 02:37.07 | WIMPy | Show us. |
| 02:37.20 | robert_ | One moment. |
| 02:39.16 | robert_ | WIMPy: http://pastebin.com/kK3XTEyn |
| 02:40.44 | robert_ | the context that's screwing up is macro-phone-ivr |
| 02:40.56 | WIMPy | I see exacely that. Your dialplan (in fact the whole file) ends after that Dial. |
| 02:41.45 | WIMPy | The call you showed useed the Dial at ivr,t,5. |
| 02:41.56 | robert_ | but shouldn't phone-ivr be kicking in? |
| 02:43.12 | WIMPy | Seems the caller didn;t enter anything and ended up at the t extension. |
| 02:43.39 | robert_ | yeah |
| 02:44.02 | WIMPy | There's nothing going on after the Dial there. |
| 02:44.22 | robert_ | we're testing the "Call the IVR, fall through to the operator and then fall through again to the operator's voicemail (if busy/NA)." part of the menu. |
| 02:45.17 | WIMPy | You need to check the DIALSTATUS *after* the Dial. |
| 02:46.17 | WIMPy | I don't know at which time M was to be executed. It doesn't exist in my help. But I doubt it makes sense. |
| 02:46.48 | WIMPy | Just rename that context and do a Goto() there after the Dial. |
| 02:47.55 | WIMPy | e.g. add an Goto(postdial,${DIALSTATUS},1) after each dial and rename macro-phone-ivr to postdial. |
| 02:48.17 | WIMPy | And remove the M(). |
| 02:48.39 | robert_ | M appears to be "Call macro" or something. Perhaps it's from an old Asterisk version? |
| 02:49.01 | WIMPy | Macros have been deprecated. |
| 02:49.12 | robert_ | they have? |
| 02:49.17 | WIMPy | But I think they were called upon answer. |
| 02:49.28 | robert_ | hm |
| 02:49.41 | WIMPy | You're supposed to use Gousb() instead. |
| 02:50.01 | robert_ | Gousb()? |
| 02:50.06 | robert_ | you mean Gosub()? |
| 02:50.22 | WIMPy | Right |
| 02:50.35 | robert_ | that makes it sound like BASIC, lol |
| 02:51.22 | WIMPy | Or like dying? |
| 02:51.52 | robert_ | anyway, I'm going to note that for Rudy. |
| 02:53.24 | robert_ | thanks. |
| 02:57.13 | igcewieling | Go USB! |
| 02:57.34 | robert_ | igcewieling: +1. :p |
| 02:57.44 | igcewieling | robert_: did you read the INSTALL-*.txt files which come with the Asterisk source? |
| 03:04.41 | robert_ | igcewieling: I'm assuming you get people who don't pay attention to packaged material very much, then? |
| 03:15.05 | igcewieling | robert_: if everyone read the most common documentation included with Asterisk we would have 1/4 fewer questions. |
| 03:15.28 | robert_ | oh, wow. |
| 03:16.55 | igcewieling | The Asterisk Book, the sample configs, the INSTALL-*.txt files, and all the applications or functions they find interesting in the output of "core show applications" and "core show functions" |
| 03:17.50 | igcewieling | maybe voip-info.org, but you have to verify what they say applies to YOUR version of Asterisk |
| 03:19.34 | robert_ | yeah, we've been using voip-info.org |
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| 05:33.28 | b0ot | What is the best codec for high latency lossy links |
| 05:33.41 | b0ot | latency of around 1000 ms |
| 05:46.10 | WIMPy | IRC |
| 05:57.02 | ChannelZ | LOL |
| 05:58.05 | ChannelZ | The lower the bitrate the less latency you'll add.. so g729 is probably desireable |
| 05:58.36 | WIMPy | As if that would make a difference at that latency. |
| 05:59.35 | ChannelZ | yeah. It's like talking to the space shuttle vs the moon. What's another half a second? |
| 05:59.41 | WIMPy | Unless someone knows a codec with built-in time machine, that's just a plain no go. |
| 06:06.46 | WIMPy | And the bitrate doesn't matter much. The transmission delay will always be less than the packetization delay. |
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| 06:52.29 | work | hello - please how collect CDRs I did cdr custom and CDR.CONFIG file (I use asterisk now my signaling is PRI_PCE) |
| 06:52.49 | work | I did not success |
| 06:53.54 | work | I only could see some phone numbers in FULL log file |
| 06:58.23 | work | ? |
| 07:08.26 | ChannelZ | What are you saying, it's not logging at ALL or just some? |
| 07:09.39 | ChannelZ | and/or what does 'cdr show status' say on the console? |
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| 07:45.51 | work | channelZ - thankx -chow cdr status (log enabled ) type (simple) backend (cdr costum)) |
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| 07:58.08 | work | channelz I could not got any CDR |
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| 08:39.45 | ChannelZ | Are you sure? Is the log empty or not being written? Does it have permission? |
| 08:40.47 | ChannelZ | /var/log/asterisk/cdr-csv/Master.csv usually |
| 08:43.06 | Rhomber | hmm, i am trying to originate a call via the manager api (through asterisk java).. I get "Originate failed".. the same works when i create a spool file though |
| 08:43.17 | Rhomber | do i need to configure the manager connection some how? |
| 08:46.06 | Rhomber | turning on manager debug doesn't seem to show anything useful |
| 08:53.43 | work | ChannelZ Master.cvs file is not found I only configured back end cdr-custom |
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| 08:56.22 | work | both cdr-cvs and cdr-costum are empty |
| 09:00.06 | work | is permission should be asterisk:asterisk ? |
| 09:00.26 | Neptu | w8 i can check |
| 09:01.34 | Neptu | root |
| 09:03.01 | Neptu | meaning root:root |
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| 09:25.23 | Rhomber | hmm, i get the same using a very simple perl script |
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| 09:35.50 | ChannelZ | well it depends on who your asterisk is run by. If you run asterisk as a user called asterisk, you don't want root owning the logs. |
| 09:36.25 | Rhomber | still running as root |
| 09:36.31 | Rhomber | it's on my todo list to lock it down |
| 09:36.42 | Rhomber | it seemed to work when I set the callerid |
| 09:37.16 | Rhomber | looks as though the upstream trunk was being given anonymous@ .. not sure why it would do that |
| 09:37.16 | ChannelZ | Rhomber: Does your Manager user have originate access? |
| 09:37.52 | Rhomber | well i guess so, since it worked when i set the caller id |
| 09:38.14 | Rhomber | i did look for such permissions, but couldn't see where to assign them |
| 09:38.18 | ChannelZ | I was referring to your question an hour ago sorry |
| 09:38.30 | Rhomber | ah :) |
| 09:38.43 | ChannelZ | what are you talking about now? |
| 09:38.50 | Rhomber | same thing |
| 09:38.56 | Rhomber | originating calls via AMI |
| 09:39.27 | Rhomber | without a "CallerId" i see this error in the console.. |
| 09:39.27 | Rhomber | chan_sip.c:22718 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:anonymous@anonymous.invalid>;tag=as78233244' |
| 09:45.30 | ChannelZ | Where is the call going? |
| 09:45.44 | Rhomber | still doesn't seem to work in java land.. which isn't asterisks fault but still.. |
| 09:45.52 | Rhomber | over a SIP trunk |
| 09:46.03 | Rhomber | channel: SIP/alltel-1/613XXXXXXX |
| 09:46.10 | Rhomber | (to protect my number..) |
| 09:46.51 | ChannelZ | Well they probably want your from to be at least semi-valid so it makes sense |
| 09:47.01 | Rhomber | it could very well be alltel rejecting it without the callerid.. but, i don't really understand the error message |
| 09:47.05 | Rhomber | yeah |
| 09:47.25 | Rhomber | so asterisk.. being in a completely bare state.. is assigning the callerid of anonymous? |
| 09:47.37 | Rhomber | usually it would be like 9011 or something from my SIP phone |
| 09:47.42 | RudyValencia | Is it possible to have a warning play to the callee on an inbound call before they are connected to the caller (e.g., "this is not a secure line")? |
| 09:47.51 | ChannelZ | but you're using the Originate command which isn't really related to any peer. |
| 09:48.06 | Rhomber | RudyValencia: sure |
| 09:48.15 | ChannelZ | You can look at the SIP debug and see precisely what it's putting in there |
| 09:48.26 | RudyValencia | Rhomber: How do I configure that in extensions.conf? |
| 09:48.38 | ChannelZ | RudyValencia: do a Playback before your Dial |
| 09:48.52 | RudyValencia | This is for calls coming in on a DID |
| 09:48.52 | robert_ | we tried |
| 09:48.57 | robert_ | I heard it. |
| 09:49.02 | Rhomber | exactly |
| 09:49.10 | robert_ | If I'm calling RudyValencia, we want HIM to hear it. |
| 09:49.21 | Rhomber | ChannelZ: oh, i didn't know there was a separate sip debug.. will have a look |
| 09:49.24 | RudyValencia | we don't want the person calling to hear it, we want the person *being called* to hear it. |
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| 09:49.31 | robert_ | yes. |
| 09:49.31 | ChannelZ | Oh.. there's a Dial option for that I think |
| 09:49.35 | Rhomber | the originator should hear it |
| 09:49.56 | ChannelZ | I'm confused, are all you working together? |
| 09:49.58 | robert_ | Rhomber: as in, the CALLING party, not the party BEING CALLED. |
| 09:49.59 | robert_ | yes. |
| 09:50.03 | ChannelZ | RudyValencia, Rhomber, robert_ ? |
| 09:50.08 | robert_ | oh no |
| 09:50.17 | RudyValencia | OK I have a better explanation |
| 09:50.19 | robert_ | ChannelZ: just RudyValencia and me. |
| 09:50.21 | Rhomber | CALLING=Originator.. BEING CALLED=Termination |
| 09:50.23 | Rhomber | usually |
| 09:50.41 | ChannelZ | you mean callER |
| 09:50.42 | RudyValencia | robert_ and I have DIDs assigned to ourselvs. |
| 09:50.54 | Rhomber | caller/calling party |
| 09:51.04 | ChannelZ | A caller is the originator of the call. The callee is the one being called. |
| 09:51.10 | RudyValencia | we're the callee in this case |
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| 09:51.58 | RudyValencia | we want asterisk to tell us "this call is coming from your DID so it's insecure" without the caller hearing that |
| 09:52.01 | Rhomber | then Playback/Background should play to the other party.. if Dial is the means to connect to you after the Playback |
| 09:52.09 | Rhomber | oh |
| 09:52.14 | Rhomber | I did that the other day |
| 09:52.20 | RudyValencia | No, Answer() is the statement that connects us to the caller |
| 09:52.38 | Rhomber | You can use a M() as a Dial option |
| 09:52.48 | Rhomber | then create a macro to do the Playback |
| 09:53.00 | Rhomber | or a Gosub.. which is another option |
| 09:53.09 | robert_ | how does that help? |
| 09:53.13 | RudyValencia | it's not for when we call out to someone |
| 09:53.24 | RudyValencia | it's when someone calls us from a regular phone |
| 09:53.38 | ChannelZ | I think you want U and then do a Playback in your gosub routine |
| 09:53.50 | Rhomber | Caller calls in.. Dial cmd issued to call you... Macro fires on Dial cmd.. you play yourself a message |
| 09:53.57 | robert_ | ChannelZ: what does U() do? |
| 09:54.14 | ChannelZ | calls a subroutine on the called channel |
| 09:54.23 | robert_ | oh, sweet. that's exactly what we want. |
| 09:54.26 | robert_ | Rudy |
| 09:54.27 | robert_ | try it. |
| 09:54.28 | RudyValencia | Flowroute says we need to use Answer() on our inbound context |
| 09:54.38 | RudyValencia | (according to their configurator) |
| 09:54.40 | ChannelZ | You'll have to try it though |
| 09:54.40 | Rhomber | lol, when he says it it's right.. i said exactly that |
| 09:55.02 | Rhomber | you'll need a Macro if you want to do anything useful other than an announcement though.. that's what i found anyway |
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| 09:55.45 | ChannelZ | oh yeah if you just want an announce the A arg is probably easier |
| 09:57.03 | Rhomber | I did something simmilar to implement the queue delivery.. press 1 to accept, 2 to reject or 3 to logout |
| 09:57.20 | Rhomber | but you need a macro for that.. otherwise it just hangs up on the queued caller.. lol |
| 09:57.26 | Rhomber | (if you reject) |
| 09:59.02 | Rhomber | sip debug looks very useful |
| 09:59.04 | Rhomber | thanks :) |
| 09:59.16 | Rhomber | can probably figure out whats happening now |
| 10:06.19 | Rhomber | in terms of permissions |
| 10:06.20 | Rhomber | read=all |
| 10:06.22 | Rhomber | write=all |
| 10:09.03 | ChannelZ | What are you sending to AMI? |
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| 11:19.43 | Rhomber | funny you should ask, just trying to debug that now |
| 11:20.07 | Rhomber | asterisk-java is completely impossible.. everything is private/protected |
| 11:21.03 | Rhomber | as far as I can tell what im doing should work, it seems to work in perl.. so just trying to see precisely what is generated |
| 11:34.04 | Rhomber | Hmmm |
| 11:34.17 | Rhomber | I got the exact string and tried it in perl |
| 11:34.21 | Rhomber | originate failed.. |
| 11:34.28 | Rhomber | I removed the "Timeout: 30" and it worked |
| 11:34.30 | Rhomber | odd? |
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| 11:40.15 | Rhomber | 100% sure now |
| 11:40.30 | Rhomber | with Timeout: 30.. "Orginate failed" |
| 11:40.45 | Rhomber | and also, if you don't set a caller id when using my SIP trunk.. lol |
| 11:40.54 | Rhomber | I will have to google up on it after dinner |
| 11:44.05 | Rhomber | ROFL |
| 11:44.12 | Rhomber | Timeout: needs to be in milliseconds! |
| 11:44.27 | Rhomber | of course!.. why be consistent? |
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| 11:45.49 | Rhomber | hmm, well that is what the docs say... guess im a stupid face. |
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| 15:50.45 | Dynamicfail | if you define a siptrunk as a peer and set a specific ip as the host, will the system only accept calls from that ip address? |
| 15:50.57 | Dynamicfail | what if I want to specify a range or just accept all incomming calls |
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| 15:56.48 | igcewieling | Dynamicfail: that question is more complicated when you might thing. |
| 15:57.29 | igcewieling | I think permit/deny is the ACL if you use username/password and if you don't use username/password then only the host= used for ACL. |
| 15:57.54 | igcewieling | I think it also depends on if the type is frield or peer or user |
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| 16:11.39 | igcewieling | If you think Asterisk docs are bad, try rooting an android phone sometime. |
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| 16:32.36 | q3k | so my crap SIP Phone is answering "400 Syntax Error" to Asterisk's INVITEs. ANy idea where to start debugging? |
| 16:34.06 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
| 16:35.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:53.37 | kresp0 | Hi all, I'm trying to use moh but it doesn't work (no sound) |
| 16:53.47 | kresp0 | On the logs seems that everything is OK (http://pastebin.com/raw.php?i=gYakAS7h) but I cannot hear the music. I know that this wav file is OK as it worked before on older asterisk. |
| 16:53.51 | kresp0 | What should I do to further debug (or fix) this? |
| 16:55.55 | kresp0 | also, the phone is a hardphone and echo test works ok |
| 16:59.53 | [TK]D-Fender | kresp0: maybe ..... just maybe ... you should ANSWER first. |
| 17:01.00 | kresp0 | ouch xD |
| 17:01.29 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 17:01.31 | [TK]D-Fender | kresp0: ... and now? |
| 17:02.23 | kresp0 | now.. it works :D |
| 17:02.36 | kresp0 | thank you so much [TK]D-Fender |
| 17:16.47 | *** join/#asterisk StaRetji (~LittleAll@80.93.240.171) |
| 17:16.48 | *** join/#asterisk ra21vi (~ravi@122.177.221.9) |
| 17:17.30 | ra21vi | is it possible to record message from caller, and get the filename recorded? |
| 17:19.17 | *** join/#asterisk areski (~areski@80.174.128.6.dyn.user.ono.com) |
| 17:21.34 | [TK]D-Fender | ra21vi: What does "filename recorded" mean? |
| 17:22.26 | eZz | I think he's talking about something like mixmonitor |
| 17:23.30 | [TK]D-Fender | "record message" does not sound like a "call" |
| 17:23.41 | eZz | yes, you're correct |
| 17:24.02 | [TK]D-Fender | I'd like to hear it direct..... |
| 17:24.16 | [TK]D-Fender | Lord knows what some people come up with... |
| 17:25.21 | ra21vi | [TK]D-Fender: oh, I mean, If caller is waiting too long in queue, I want to have an option for him, that he can get off the queue, record a message and email will be sent. |
| 17:25.47 | ra21vi | [TK]D-Fender: I probably found one CMD, record. That records to given filename |
| 17:26.22 | [TK]D-Fender | ra21vi: Have you considered just using a voicemail box? |
| 17:27.51 | ra21vi | [TK]D-Fender: not yet. Since I am writing AGI, I thought using cmd in program would be better, as I can send them to mail and DB as required. |
| 17:28.01 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 17:28.16 | [TK]D-Fender | ra21vi: well if you have other things to do sure... |
| 17:28.25 | ra21vi | [TK]D-Fender: is there any benefit using VoiceMail, or any cons using direct Record? |
| 17:28.36 | [TK]D-Fender | ra21vi: So go make an exten for them to quit to. |
| 17:28.57 | ra21vi | [TK]D-Fender: do you mean BreakOut Context? |
| 17:29.18 | [TK]D-Fender | ra21vi: Voicemail alreday has a prompting system built in. Already has an e-mailing feature built-it. Already has options for them to REVIEW their recording and start over built-in...... |
| 17:29.32 | [TK]D-Fender | ra21vi: Cleary that is the option meant for them to break out with |
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| 18:23.13 | RudyValencia | OK, so I'm having problems with converting a dialplan from using macros to using Gosub. I can't seem to get the call status to send the caller to VM if the callee does not answer, is busy, or is unreachable. |
| 18:23.53 | WIMPy | How are those two facts linked to another? |
| 18:28.16 | RudyValencia | The macro used to do this without a problem. |
| 18:28.26 | RudyValencia | I can't seem to get the Gosub to work the same. |
| 18:28.57 | [TK]D-Fender | RudyValencia: you've done something wrong and we need to see it to point it out to you |
| 18:29.33 | [TK]D-Fender | ~pb |
| 18:29.33 | 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. |
| 18:29.35 | [TK]D-Fender | ^^^ |
| 18:29.39 | RudyValencia | Oh, here's a paste of extensions.conf: http://pastebin.com/kQTTyAhg (note, some slight redaction has been done to remove our DIDs, but not affecting the text |
| 18:30.34 | [TK]D-Fender | RudyValencia: Your Goto is bad |
| 18:30.45 | RudyValencia | Which line? |
| 18:30.46 | [TK]D-Fender | RudyValencia: same => n,Goto(${DIALSTATUS},1) <------------ |
| 18:30.54 | RudyValencia | oh |
| 18:31.04 | [TK]D-Fender | RudyValencia: ${DIALSTATUS} <-- is in the EXTENSION paramater |
| 18:31.16 | RudyValencia | OH |
| 18:31.41 | WIMPy | What's wrong about that? |
| 18:31.56 | WIMPy | There are matching extensions. |
| 18:31.56 | [TK]D-Fender | WIMPy: because those are what he uses for PRIORITIES in that "s" exten |
| 18:32.12 | [TK]D-Fender | WIMPy: 21-27 |
| 18:32.42 | RudyValencia | should it be Goto(s,${EXTEN},${CALLSTATUS}) then? |
| 18:32.46 | WIMPy | Oh, teh first one. Yes, [phones] should probably look lile [postdial]. |
| 18:32.54 | [TK]D-Fender | RudyValencia: No, only PRIORITY. |
| 18:33.11 | RudyValencia | Oh |
| 18:33.15 | igcewieling | RudyValencia: or Goto(${DIALSTATUS}) which will go to that priority in the same extension. |
| 18:33.22 | [TK]D-Fender | RudyValencia: ...... you are jumping WITHIN your exten.... |
| 18:33.27 | WIMPy | The syntax of the extensions isn;t correct. |
| 18:33.41 | [TK]D-Fender | WIMPy: just the Goto is bad. The rest is sane |
| 18:33.51 | RudyValencia | We made [phone] based off [postdial] but it didn't seem to work, I'll change the Goto |
| 18:34.01 | igcewieling | RudyValencia: looks like you tried adapting the sample in extensions.conf.sample 8-) |
| 18:34.05 | WIMPy | No, they have a ,1 after the lable. |
| 18:34.12 | RudyValencia | right |
| 18:34.25 | WIMPy | So I'd remove the s(), but you can also remove the ,1 and modify the Goto. |
| 18:34.27 | [TK]D-Fender | RudyValencia: [phone] , not [postdial] |
| 18:34.52 | [TK]D-Fender | RudyValencia: WIMPy His Gosub's are to [phone] not [postdial] |
| 18:35.06 | RudyValencia | OK, I'm reloading the dialplan |
| 18:35.15 | [TK]D-Fender | WIMPy: Which is not to say that that other context isn't a mess |
| 18:35.23 | RudyValencia | I removed the ,1 from Goto |
| 18:35.33 | RudyValencia | It's just Goto(${CALLSTATUS}) now |
| 18:35.46 | RudyValencia | Er, DIALSTATUS |
| 18:35.48 | WIMPy | And the extensions in [phones] are syntactically bad. |
| 18:35.53 | igcewieling | I don't really see how it can work at all exten => 2000,1,Gosub(phone,${EXTEN},1) would go to 2000 in [phone], but all I see is exten => s in [phone] |
| 18:36.39 | [TK]D-Fender | RudyValencia: exten => s(ANSWER),1,Hangup() <- and these... should be like exten => s,n(ANSWER),Hangup() |
| 18:36.41 | WIMPy | Yes, these Gosubs should probably be Gotos and s could be _X. . |
| 18:36.55 | [TK]D-Fender | RudyValencia: You put the label on the EXTENSION portion instead of the PRIORITY portion |
| 18:37.02 | [TK]D-Fender | RudyValencia: our syntax is very wrong.... |
| 18:37.04 | [TK]D-Fender | your* |
| 18:37.34 | RudyValencia | lemme bring my tools to this side (they're on a VM right now) so I can have an easier time editing |
| 18:37.49 | [TK]D-Fender | RudyValencia: exten => s(BUSY),1,Voicemail(${EXTEN}@employee,b) <-- and your use of ${EXTEN} here is very wrong look at the EXTEN you are in on that line |
| 18:37.57 | [TK]D-Fender | RudyValencia: You need to clean that whole mess up |
| 18:38.03 | RudyValencia | I do |
| 18:38.17 | [TK]D-Fender | just about every piece of that context is wrong |
| 18:38.35 | WIMPy | That's wy I sugested to make [phone] like [postdial], which looks sane already, except for the s extension. |
| 18:38.40 | RudyValencia | has only been doing this for a week |
| 18:39.21 | [TK]D-Fender | WIMPy: "Goto" into another exten is the old pre-label concept and should not be used unless needed.... and it certainly isn't in this case. |
| 18:40.26 | WIMPy | That's a matter of personal preferences. |
| 18:40.47 | WIMPy | But I'd keep the two contexts as identical as possible. |
| 18:41.02 | [TK]D-Fender | WIMPy: Certainly less spaghetti keeping it within the same exten |
| 18:41.05 | WIMPy | Or maybe there's only the need for one of them. |
| 18:41.16 | igcewieling | Goto: http://xkcd.com/292/ |
| 18:41.21 | [TK]D-Fender | WIMPy: Bring the worse one up to par then, don't drag a potentially better one down. |
| 18:41.42 | [TK]D-Fender | igcewieling: FTMFWY |
| 18:41.55 | WIMPy | I prefer the other for readability. |
| 18:42.11 | RudyValencia | OK, so I am trying this |
| 18:42.29 | RudyValencia | (I've fixed the phone context to match postdial) |
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| 18:42.55 | [TK]D-Fender | WIMPy: A single blank line between priority labels breaks it off plenty... Also breaks other things in cases where you might be using ${EXTEN} |
| 18:43.36 | WIMPy | Personal preferences. No argument. |
| 18:43.58 | WIMPy | RudyValencia: Yu also need to have that Dial on a pettern to ever match. |
| 18:44.34 | [TK]D-Fender | RudyValencia: Which was for your not using the ARG's you are passing as you're supposed to. |
| 18:44.38 | RudyValencia | e.g., exten => _NNNN,1,Dial(... ? |
| 18:45.27 | RudyValencia | (postdial is for the IVR to dial us, phone is for us to dial within the system) |
| 18:46.09 | RudyValencia | basically, we want the IVR to place the caller on hold while it tries to reach out to us. |
| 18:47.01 | WIMPy | Putting the list as Argument would make more sense there I guess. Certainly looks cleaner. |
| 18:47.46 | WIMPy | However there's no return. I don't like to Gosub without (the intention to) return. |
| 18:48.09 | [TK]D-Fender | No point in Goto-ing a pattern-ed exten VS passing it as an arg... just makes more room for error and more work to adapt to other uses |
| 18:49.06 | RudyValencia | OK, I'll use ARG1 to pass the extension |
| 18:49.14 | RudyValencia | Idea |
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| 18:49.46 | RudyValencia | I could use another argument to pass choice of ring/hold |
| 18:53.04 | [TK]D-Fender | RudyValencia: http://pastebin.com/i86zkdG4 |
| 18:53.10 | [TK]D-Fender | RudyValencia: Merry Christmas |
| 18:53.15 | RudyValencia | Thanks |
| 18:56.08 | RudyValencia | I presume this could also apply to the IVR dialing out to us? |
| 18:56.29 | RudyValencia | (instead of postdial) |
| 18:56.47 | [TK]D-Fender | RudyValencia: I fiex your GOSUB one. Learn from it to fix the others |
| 18:56.50 | [TK]D-Fender | fixed* |
| 18:56.53 | RudyValencia | OK |
| 18:57.50 | [TK]D-Fender | RudyValencia: Note important changes like implicit hangup's after Voicemail.. because it can exit to CONTINUE processing dialplan and jsut leave more VM's |
| 18:58.43 | RudyValencia | ah |
| 18:59.47 | RudyValencia | It is still giving an error |
| 18:59.57 | RudyValencia | NOTICE[18751][C-00000055]: pbx.c:4698 pbx_extension_helper: No such label 'phone' in extension '2000' in context 'employee' |
| 19:00.17 | RudyValencia | Followed by WARNING[18751][C-00000055]: pbx.c:12072 pbx_parseable_goto: Priority 'phone' must be a number > 0, or valid label |
| 19:00.31 | RudyValencia | and ERROR[18751][C-00000055]: app_stack.c:547 gosub_exec: Gosub address is invalid: 'phone(s,1,2000)' |
| 19:01.12 | [TK]D-Fender | RudyValencia: You did not look at the fact I changed your GOSUB lines as well |
| 19:01.30 | RudyValencia | I copied those over |
| 19:01.51 | RudyValencia | exactly as you put them on that paste |
| 19:01.52 | [TK]D-Fender | RudyValencia: Show us the call and your new dialplan |
| 19:03.12 | RudyValencia | http://pastebin.com/VA4wUFu5 |
| 19:03.34 | [TK]D-Fender | RudyValencia: http://pastebin.com/VA4wUFu5< - you should not have this in [default] |
| 19:03.44 | RudyValencia | ah |
| 19:03.51 | [TK]D-Fender | RudyValencia: exten => 2000,1,Gosub(phone(s,1,${EXTEN}))<- this is NOT the line I gave you in my pastebin at all |
| 19:04.11 | [TK]D-Fender | RudyValencia: exten => 2000,1,Gosub(phone,s,1,${EXTEN}) <--- THIS is what I gave you |
| 19:04.25 | RudyValencia | It complained about the missing parenthesis |
| 19:04.45 | RudyValencia | WARNING[18711][C-00000054]: app_stack.c:533 gosub_exec: Ouch. No closing paren: 'phone(s,1,2000'? |
| 19:04.53 | [TK]D-Fender | RudyValencia: Becase you started by putting your FIRST "extra" one that you shouldn't have |
| 19:05.05 | [TK]D-Fender | RudyValencia: My syntax was correct as provided |
| 19:05.19 | RudyValencia | oh |
| 19:05.50 | [TK]D-Fender | RudyValencia: Mine never had a "(" after "phone" in the first place |
| 19:06.43 | RudyValencia | whacks himself using a 2x4 with a rusty nail in it |
| 19:06.48 | RudyValencia | :P |
| 19:07.30 | RudyValencia | Now I get an error "Dial argument takes format (technology/resource)" |
| 19:07.45 | [TK]D-Fender | RudyValencia: Very apt .. i am partial to the rusty-nail enhanced ClueBat (tm) |
| 19:08.13 | [TK]D-Fender | RudyValencia: Show us the NEW new dialplan and call |
| 19:08.16 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 19:09.19 | RudyValencia | http://pastebin.com/QFc3UhqD |
| 19:09.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 19:09.23 | [TK]D-Fender | oops |
| 19:09.27 | [TK]D-Fender | dangit |
| 19:09.30 | RudyValencia | reposting: http://pastebin.com/QFc3UhqD |
| 19:09.45 | RudyValencia | (you missed it a sec ago) |
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| 19:12.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:12.51 | [TK]D-Fender | RudyValencia: exten => 2000,1,Gosub(phone,s,1(${EXTEN})) |
| 19:12.54 | [TK]D-Fender | RudyValencia: My bad |
| 19:13.17 | [TK]D-Fender | RudyValencia: And I think I copy-pasted 2 x 2000's there and killed your original other extension # |
| 19:13.34 | RudyValencia | no biggy |
| 19:13.50 | RudyValencia | it's working |
| 19:13.59 | [TK]D-Fender | RudyValencia: There we go... |
| 19:14.47 | RudyValencia | Thanks btw :) |
| 19:15.06 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:15.13 | RudyValencia | I'd send you some good beer but the logistics of delivery over IRC... |
| 19:15.32 | RudyValencia | hm... |
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| 20:30.00 | Neptu | RudyValencia, spain? |
| 20:30.35 | RudyValencia | nope, Colorado, USA |
| 20:42.15 | ChannelZ | Get off my lawn! |
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| 21:00.31 | Neptu | plants & zombies?? |
| 21:01.33 | RudyValencia | hm? |
| 21:03.02 | ChannelZ | is in CO too so I was just yelling. |
| 21:03.50 | RudyValencia | ah cool |
| 21:04.15 | RudyValencia | ChannelZ: I'm in Greeley, to be more specific |
| 21:05.10 | ChannelZ | Golden |
| 21:05.30 | Neptu | Stockholm |
| 21:06.17 | RudyValencia | I go to the casinos in Black Hawk and Central City and the bus passes through Golden near the Coors beer factory |
| 21:06.48 | RudyValencia | mmm... Coors, "the banquet beer"... |
| 21:07.22 | ChannelZ | yeah I'm a few minutes away from them |
| 21:08.08 | RudyValencia | I went to Denver last night to hang with friends |
| 21:09.00 | ChannelZ | I sat at home watching someone play Simcity on Youtube |
| 21:09.07 | RudyValencia | ah |
| 21:09.19 | ChannelZ | I live la vida loca |
| 21:09.30 | RudyValencia | I live an adventurous life |
| 21:09.35 | RudyValencia | I go all over to find fun |
| 21:10.10 | RudyValencia | if you ever see a silver Saturn coupe zoom by you at speeds exceeding posted limits it's probably me :P |
| 21:11.32 | RudyValencia | btw did your nickname come from the B-52s song of the same name? |
| 21:12.05 | RudyValencia | doesn't use a nickname because it's silly |
| 21:12.08 | ChannelZ | Getting nothing but static |
| 21:12.42 | RudyValencia | Static in my attic from ChannelZ :P |
| 21:13.14 | RudyValencia | All I know we've got to change what's happening... |
| 21:13.15 | ChannelZ | I ususally go by ManChicken but that nick was already regged here |
| 21:13.17 | RudyValencia | oh |
| 21:14.13 | RudyValencia | I'm starting a company with robert_ and we are using Asterisk as our phone system |
| 21:14.20 | RudyValencia | (on a Linode in Texas, no less) |
| 21:14.34 | RudyValencia | We're using Flowroute for DIDs and outgoing trunks |
| 21:15.09 | RudyValencia | I think Flowroute is the least expensive provider out there |
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| 21:15.29 | ChannelZ | Yeah that's what I did, sans running in texas. Using 4 POTS though |
| 21:16.25 | RudyValencia | ah, utilizing the PSTN as your backing. CenturyLink, I presume? |
| 21:18.52 | ChannelZ | Started out on Qwest but I dumped those bitches a long time ago and am with a local place called Integra |
| 21:19.10 | ChannelZ | I do VoIP at home though through Vitelity |
| 21:19.46 | RudyValencia | ah Integra, I drive by them on I-25 heading to Micro Center at times. |
| 21:21.05 | RudyValencia | IIRC one of those old buildings used to be a Ford assembly center decades ago |
| 21:21.06 | ChannelZ | Yah they seem great, paying probably half what we were through Qwaste and they are actually competent. Qwaste couldn't figure out how to make all 4 lines show up with as the same CallerID number |
| 21:21.45 | RudyValencia | I call CenturyLink a different name by changing the first E to a U |
| 21:21.50 | RudyValencia | :P |
| 21:22.22 | RudyValencia | I dropped their DSL because it was dog-slow and I wanted faster but the only service here that is faster is Comcast |
| 21:22.31 | RudyValencia | So I went to Comcast. |
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| 21:22.41 | ChannelZ | Yeah I think I could still only get 1mb at my house |
| 21:23.08 | ChannelZ | Comcast sucks too but their business class is OK. They at least answer the phone in english |
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| 21:23.33 | RudyValencia | Do I have to have a legal business to get their business class service or can I get it with the claim of "working from home"? |
| 21:23.52 | ChannelZ | Integra has a bonded DSL that a friend of mine uses |
| 21:23.56 | RudyValencia | ooh |
| 21:23.59 | ChannelZ | No, I have it at home |
| 21:24.22 | RudyValencia | I can just tell them I have a home-office and get it then? |
| 21:24.32 | ChannelZ | yeah |
| 21:24.39 | RudyValencia | checks to see what he pays for his current service |
| 21:24.48 | ChannelZ | they don't give a shit.. its more money |
| 21:24.57 | RudyValencia | ah |
| 21:25.03 | RudyValencia | probably not much more |
| 21:25.42 | RudyValencia | I pay about $81 a month for Internet + digital phone |
| 21:26.18 | RudyValencia | I get Blast-level speed |
| 21:26.30 | ChannelZ | My bill is like $82 for 16mbit business with a static IP |
| 21:26.43 | RudyValencia | nice |
| 21:27.00 | RudyValencia | I could turn the digital phone off and use VoIP for calls |
| 21:27.02 | ChannelZ | but they don't block ports or whine at you for running services |
| 21:27.25 | RudyValencia | I've never had a problem with blocked ports or them whining about services from my line |
| 21:27.33 | ChannelZ | Although I had home service for awhile years ago and they didn't ever bother me |
| 21:28.09 | ChannelZ | except when they initially started interfering with bit torrent which is when I dumped them the first time |
| 21:28.15 | RudyValencia | I guess if you aren't running the next facebook on the home service then they are ok |
| 21:28.41 | RudyValencia | I kinda think they've softened up a bit wrt services on home lines |
| 21:28.46 | ChannelZ | yeah I was just running my own SMTPd and little personal website. |
| 21:29.02 | ChannelZ | I seem to remember they blocked outgoing SMTP except to their own server which is what really pissed me off |
| 21:29.10 | RudyValencia | (since I have a test VM providing a basic Website) |
| 21:29.42 | RudyValencia | virtualization is a bit of snakeoil but it's an interesting concept |
| 21:30.19 | RudyValencia | I use Proxmox VE locally |
| 21:31.43 | ChannelZ | so what kind of business are you making? |
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| 21:35.01 | *** join/#asterisk StaRetji (~LittleAll@80.93.240.171) |
| 21:35.44 | menti | Hi guys, i'm looking for a C# API To connect with Asterisk v1.8 i saw "Asterisk .NET" but i only can use it with Asterisk v1.4. i also looked here: http://www.voip-info.org/wiki/view/Asterisk+AGI#NET but all of them are too old, my question is: Is there any C# API to connect with Asterisk? a modern one with some documentation and support? Which one do you use? |
| 21:36.57 | ChannelZ | Write your own, get rich |
| 21:37.46 | ChannelZ | AGI or AMI aren't that complicated |
| 21:38.09 | menti | well, i'm still at the university, i'm not sure if i can write my own.... :/ |
| 21:38.53 | ChannelZ | well AGI is just stdin/stdout, AMI is sockets. But they're different uses |
| 21:40.13 | ChannelZ | AGI is for running scripts in your dialplan; AMI is more asyncronous and unrelated to a particular call, gets used mainly for writing monitoring/management applications |
| 21:40.18 | menti | well, that could be the last option.... but is there any API? |
| 21:40.45 | ChannelZ | AGI and AMI are documented |
| 21:41.37 | ChannelZ | as for specifically a C# one I dunno. They're all just wrappers with varying amounts of actual usefulness. |
| 21:43.46 | [TK]D-Fender | menti: If you're in university ... and aren't sure you can write a basic plain-text socket set of procedures... do you think this requires post-university genius? |
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| 22:04.57 | carrar | menti, do note that if you are going to write anything (like docs) then you know as a university person you will need to write those in LaTeX!! :) |
| 22:05.22 | RudyValencia | ChannelZ: Software/systems/interactive media |
| 22:05.37 | [TK]D-Fender | Last time I did anything in latex it was illegal in at least 17 states.... |
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| 22:05.43 | carrar | haha |
| 22:05.57 | RudyValencia | yeah |
| 22:06.45 | carrar | actaully I'm a fan of LaTeX |
| 22:15.49 | ChannelZ | you probably love vi too |
| 22:16.56 | carrar | kinda goes along with that yeah |
| 22:19.03 | jeev | does digium provide g729 demo or do you guys suggest just buying one channel and trying it |
| 22:19.24 | WIMPy | What do you want to try about it? |
| 22:19.40 | carrar | they used to give 1 or 2 our for free at convention meetings |
| 22:19.42 | carrar | out |
| 22:19.49 | jeev | want to try the quality, my friend is telling me to use it, the one or two places i do voip at are complainint recently.. when i check their graphs, their bw is pretty much maxed. |
| 22:20.12 | carrar | like astricon |
| 22:20.46 | ChannelZ | When do the patents expire? |
| 22:21.09 | [TK]D-Fender | jeev: What phones & PSTN service+tech do you use? |
| 22:21.19 | carrar | 729 works great |
| 22:21.56 | jeev | fender, they're polycom 330's, no pstn. they're voice, i mean their pstn is dedicated to hylafax. |
| 22:22.31 | jeev | fender, one of the monkey's there.. is torrenting, always killing bandwidth, i firewalled his ip. the other place, they have a mail server on the t1, i've asked the uplink to prioritize sip packets.. but they're still maxing out the bandwidth. |
| 22:22.48 | [TK]D-Fender | jeev: What do youmaen no PSTN? Their PBX is basically only an INTERCOM system? |
| 22:23.05 | jeev | fender, it's itsp only.. |
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| 22:23.13 | [TK]D-Fender | ITSP = PSTN |
| 22:23.18 | [TK]D-Fender | I asked how they get to it |
| 22:23.28 | jeev | i've tried 4 or 5 different carriers. |
| 22:23.33 | [TK]D-Fender | So go set a phone to G.729 and try direct. |
| 22:23.56 | [TK]D-Fender | Do yoy know how many people use G.729? TONS. |
| 22:24.01 | jeev | i bet. |
| 22:24.02 | carrar | 3 |
| 22:24.12 | [TK]D-Fender | Yes it works, and you only need the license to transcode. |
| 22:24.26 | [TK]D-Fender | Which is basically your prompts and recording. |
| 22:24.43 | carrar | or if you carrier doesn't support it, in which case drop them |
| 22:24.44 | jeev | the bandwidth savings is definitely worth the cost |
| 22:24.53 | jeev | carrar, they do support it. |
| 22:25.02 | jeev | i just always looked down on it because i'm ignorant mostly about telephony |
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| 22:26.15 | jeev | anyway, the one with the t1 also has cable, i'll figure out a way to connect it to the main server in the colo. maybe tunnel it so i dont have port issues |
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| 22:26.54 | cusco | hi |
| 22:27.07 | cusco | can asterisk mixmonitor a call except for silence? |
| 22:27.59 | [TK]D-Fender | cusco: No, but you CAN clean it up yourself in post-processing |
| 22:28.24 | jeev | fender, i got some sangoma t1 cards the other day to weed out the b.s. cards i've been using that i'm having faxing issues with. |
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| 22:28.37 | cusco | hmm.. what tool does that.. sox? |
| 22:28.49 | [TK]D-Fender | cusco: Excellent first choice |
| 22:29.04 | cusco | ok I'll look into it. Thanks |
| 22:29.34 | [TK]D-Fender | cusco: tons of audio tools out there. Audacity is scriptable and can be called from CLI w/ X |
| 22:29.39 | [TK]D-Fender | w/o |
| 22:29.58 | cusco | if sox does it, I'll use sox as it is already installed |
| 22:30.24 | cusco | im already calling a sh script in mixmonitor() so, adding a line or two, is ok |
| 22:50.16 | kresp0 | I'm getting trouble trying to understand pattern matching order on asterisk. I have put "include => from-pln" at the beginning inside of [to-pln] context. But, if I dial 0000800, asterisk match the more general pattern. I want asterisk to match the more concrete patterns on from-pln first. |
| 22:50.20 | kresp0 | my dialplan: http://pastebin.com/SEiWxvNF |
| 22:50.27 | kresp0 | CLI debug: http://pastebin.com/HBGKBTnN |
| 22:50.54 | WIMPy | Includes are only searched if no local extension match. |
| 22:51.21 | kresp0 | ohh, ok then |
| 22:51.25 | kresp0 | thank you WIMPy |
| 22:51.29 | WIMPy | So it doesn't matter when you put the include. Only the order of multiple includes matter. |
| 22:51.41 | kresp0 | now I get it |
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| 22:53.01 | WIMPy | s/when/where |
| 22:53.08 | WIMPy | off course |
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