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| 04:11.36 | opterons | anyone here? |
| 04:12.50 | [TK]D-Fender | Possibly |
| 04:12.57 | [TK]D-Fender | I see 185 in the list... |
| 04:13.22 | opterons | yeah the chances are good |
| 04:14.49 | opterons | think something might be wrong with the mailing script on the forum |
| 04:15.16 | opterons | never got notified about a reply and checked to get them on the post |
| 04:15.34 | opterons | figure someone might want to know |
| 04:15.39 | opterons | check spam no mail |
| 04:15.42 | [TK]D-Fender | Could be delays I suppose. I never use the forums. |
| 04:16.09 | opterons | jan. 9th was the reply |
| 04:16.18 | opterons | hope it's not a delay lol |
| 04:16.20 | [TK]D-Fender | You could of course..... start a thread ON the forum so it's stated right there wher it's admins will see it |
| 04:16.52 | opterons | ya I did my dumbass was waiting for an email instead of checking the forum |
| 04:17.15 | opterons | lazyness at its best |
| 04:19.45 | [TK]D-Fender | \o/ |
| 04:20.42 | *** join/#asterisk ipn (4a6c86f1@gateway/web/freenode/ip.74.108.134.241) |
| 04:21.46 | ipn | Hey all - I was hoping to get some help with outbound routes. I want the system to be able to call a hunt group if 911 was dialed. Any ideas how to accomplish this in the simplest form would be greatly appreciated |
| 04:23.23 | [TK]D-Fender | ~freepbx |
| 04:23.24 | 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 |
| 04:23.25 | [TK]D-Fender | ^^^ |
| 04:23.39 | ipn | its not free pbx |
| 04:23.53 | ipn | i am just spit balling ideas |
| 04:24.33 | [TK]D-Fender | It's your dialplan, it does whatever that extension tells it to do |
| 04:24.55 | [TK]D-Fender | "hunt group" and "outbound route" are not Asterisk things |
| 04:25.03 | [TK]D-Fender | You Dial() whatever you dial.\ |
| 04:25.13 | [TK]D-Fender | You check whatever you want to check before. Or after |
| 04:25.56 | ipn | does the dial plan wait for anything regarding system commands to return or if i do system ( write the file ), the dial plan will keep executing or wait to see if the write was successful? |
| 04:26.41 | [TK]D-Fender | Dialplan apps are almost all "blocking". System() is no exception |
| 04:27.07 | [TK]D-Fender | Now for System() you could have the shell background it the same way you would anything else |
| 04:27.19 | [TK]D-Fender | & <- |
| 04:29.03 | ipn | Yeah - I didnt want anything to hang the dialplan up which would prevent the 911 call to go through so I was thinking more of a CDR after the fact approach |
| 04:29.50 | ipn | I am thinking about setting a userfield in mysql from the dialplan on the dial() and then having a cron checking for that field, setting a call file and changing that field so the next cron search didnt send the same alert |
| 04:30.38 | [TK]D-Fender | Userfield is a run-time CDR field and takes no time |
| 04:31.14 | [TK]D-Fender | So you could set it immediately |
| 04:32.02 | [TK]D-Fender | As for the call-file, that process alone should be practically instantaneous unless you are doing something odd. |
| 04:32.29 | [TK]D-Fender | And at worst ... background it and it'll still clear the System() pretty instantly |
| 04:32.31 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 04:32.42 | [TK]D-Fender | No reason to waste a background daemon for this |
| 04:32.55 | ipn | yeah - I have the call file under my belt. I was trying to figure out the best way to alert when 911 was dialed and eliminating any possibility of a resource block or something |
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| 04:34.43 | [TK]D-Fender | Your notification pattern sounds fixed so it should be a single file copy, then move. Virtually instant. |
| 04:34.53 | ipn | so I could system( echo "syntax for call file on one line\n &") and then system( mv the file to outgoing & ) and those should run and the dialplan continue regardless of the system commands and any resource or odd things happening to those commands |
| 04:35.44 | [TK]D-Fender | unless you are really screwing with what you're passing on a system it should never hang |
| 04:36.55 | ipn | makes sense. You are full of knowledge, my friend! I will script something up real quick and test it out. |
| 04:41.11 | [TK]D-Fender | now if your FS hangs somehow... that would be bad, but odds are EVERYTHING would be in serious trouble in all likelyhood. |
| 04:47.04 | ipn | true |
| 04:47.50 | ipn | Just ran into a bump - with call files, there is no way to tell the manager which extension dialed 911; as far as I can tell |
| 04:51.01 | *** join/#asterisk nighty^ (~nighty@s229123.ppp.asahi-net.or.jp) |
| 04:53.35 | [TK]D-Fender | Sure there is |
| 04:53.54 | [TK]D-Fender | MANY different ways |
| 04:54.56 | [TK]D-Fender | If you are using your call file(s) to announce this... you could have it SayDigits or SayNumber it |
| 04:55.06 | [TK]D-Fender | Or have that encoded in the CALLERID |
| 04:56.34 | [TK]D-Fender | You could have pre-recorded names and make that part of your playback. You could send it in MORSE CODE if you felt like it |
| 04:57.23 | ipn | I think I will add the exten to the the caller ID for sure. I like that. I could pass in a setVar on a channel variable for playback to. |
| 04:57.48 | [TK]D-Fender | Exactly |
| 04:57.54 | [TK]D-Fender | You can pass whatever you want |
| 04:57.59 | ipn | The simple 911 alert worked. Manager got call as the call went through to the destination |
| 05:31.04 | opterons | hey when I see the arg ${EXTEN:1} that means the current extension dialed with the leading 0 stripped from it correct? |
| 05:35.48 | [TK]D-Fender | no, that strips the CURRENT EXTEN you're executing (which chan change as you Goto/Gosub/etc) of its first character. It is not implicitly a 0 |
| 05:36.07 | [TK]D-Fender | That would depend on what was actually being executed |
| 05:42.07 | opterons | so for this |
| 05:42.13 | opterons | exten => _0XXX,3,Voicemail(${EXTEN:1}@default,u) ; go to voicemail for extension |
| 05:42.42 | opterons | that would strip the 0 that is offset 0 of the exten array |
| 05:42.49 | opterons | _0XXX |
| 05:43.02 | opterons | XXX dialed number |
| 05:43.08 | opterons | means i take it |
| 05:44.42 | [TK]D-Fender | That would indeed return the XXX part of the match there |
| 05:45.35 | opterons | not sure if my term is right when I called that an array |
| 05:46.00 | opterons | seems like it would be with the knowledge in programming I have |
| 05:46.05 | opterons | lil knowledge |
| 05:48.12 | opterons | for this that follows that line |
| 05:48.18 | opterons | exten => 1,1,NoOp(${CELL${EXTEN}}) |
| 05:48.40 | opterons | would I still need to apply the fir char strip or no? |
| 05:48.46 | opterons | first |
| 05:51.47 | [TK]D-Fender | Strip it for what? |
| 05:51.55 | [TK]D-Fender | Also that is a NESTED reference you have going on there. |
| 05:52.48 | opterons | very new to asterisk |
| 05:52.56 | [TK]D-Fender | It doesn't change the EXTEN BTW.... it outputs a substring of it. |
| 05:55.11 | opterons | think you could look at a pastebin for me I am trying to make it possible for the caller once they get the extension voicemail to be able to press 1 to dial that extension cell and 2 to dump them into the extensions voicemail? |
| 05:56.23 | opterons | I aquired a asterisk 1.8 box from a friend. |
| 05:56.54 | opterons | If you are busy it's cool though. |
| 05:57.59 | [TK]D-Fender | 1.8 is dead |
| 05:58.03 | [TK]D-Fender | FYI |
| 05:58.13 | opterons | I figured. |
| 05:58.23 | [TK]D-Fender | "core show application voicemail" <- |
| 05:58.30 | opterons | but it still works :D |
| 05:59.36 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions |
| 06:00.26 | Lope | Hello my favorite telephony friends! |
| 06:00.49 | Lope | I'm getting warmed up for some VoIP action! |
| 06:04.30 | opterons | thanks! |
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| 06:51.32 | opterons | when I see a line like this in extensions.conf |
| 06:51.35 | opterons | exten => n,Dial(DAHDI/g2/${CELL${EXTEN}},20) |
| 06:52.14 | opterons | in the string: DAHDI/g2 the g2 means group 2 correct? |
| 06:53.06 | opterons | based in the chan_dahdi.conf correct? |
| 06:56.44 | opterons | nm found it |
| 06:56.51 | [TK]D-Fender | that line is wrong |
| 06:57.03 | [TK]D-Fender | And yes for the other questions |
| 06:57.20 | opterons | what is wrong with that line |
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| 06:59.22 | [TK]D-Fender | The complete lack of an extension |
| 06:59.41 | [TK]D-Fender | exten => extension,priority,application(data) |
| 07:01.18 | opterons | can you send me a link explaining when I see n,s,i ect in that first offset? |
| 07:01.29 | opterons | what they actually mean |
| 07:01.48 | opterons | exten => s,3,Wait(1) |
| 07:02.12 | [TK]D-Fender | ~book |
| 07:02.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:02.12 | opterons | <PROTECTED> |
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| 07:02.35 | opterons | ~buybook |
| 07:02.35 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
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| 07:22.48 | opterons | should have been: same => n,Dial(DAHDI/g2/${CELL${EXTEN}},20) |
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| 07:24.07 | opterons | correct ? |
| 07:24.48 | opterons | with the n meaning the previous prority +1 |
| 07:24.57 | opterons | priority |
| 07:30.08 | [TK]D-Fender | that makes more sense |
| 07:30.18 | [TK]D-Fender | Assuming there was a first priorty that matches to start things with |
| 07:30.30 | [TK]D-Fender | And still dealing with the nested variable quesiotn. |
| 07:30.33 | [TK]D-Fender | Things to read... |
| 07:30.36 | [TK]D-Fender | and on that note .... |
| 07:30.38 | [TK]D-Fender | bed time |
| 07:30.39 | opterons | yeah |
| 07:30.45 | opterons | thanks for the help |
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| 08:10.38 | PoWeRKiLL | Hi |
| 08:10.59 | PoWeRKiLL | Is it possible to use cdr_adaptive_odbc to log to two differente mysql server ? |
| 08:20.21 | opterons | http://lists.digium.com/pipermail/asterisk-users/2008-September/218238.html |
| 08:21.05 | opterons | i'm no expert but do understand mysql replication |
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| 13:38.09 | dan_j | Is there any way to override the early audio received when the number dialled is ringing? |
| 13:38.15 | dan_j | This is a SIP query |
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| 13:52.33 | Penguin | dan_j: What do you want to happen by overriding it? |
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| 13:54.06 | dan_j | I'm based in the UK and clients expect a UK ringtone when they make an outbound call. I've got a outbound call provider who provides a high level of service but occasionally, the ring tone you hear when dialing through them, is an foreign ring tone. |
| 13:54.13 | dan_j | Clients complain, so i cant use them. |
| 13:54.53 | dan_j | I want to override the foreign ring tone with a UK ring tone. |
| 13:55.31 | Penguin | dan_j: If you set your ring sound and then provide ring when dialing, does it change when the provider makes progress? |
| 13:56.12 | dan_j | I believe it starts ringing instantly, even if the provider hasnt made progress yet. |
| 13:56.31 | Penguin | Right, and it will be the ring tone you want... |
| 13:56.38 | dan_j | Shame there isnt an option to play the ringtone only when progress is made. |
| 13:56.45 | Penguin | But does it change to the ring sound from the provider after that? |
| 13:57.03 | dan_j | Yes because the provider hasnt answered yet |
| 13:57.22 | Penguin | The provider shouldn't answer ever. |
| 13:57.53 | dan_j | The provider sends the answered message when the other side answers. |
| 13:58.07 | Penguin | And then the ringing stops. |
| 13:58.23 | Penguin | I'm talking about the time between your ring playing and the answer. |
| 13:59.08 | dan_j | If I play a ringing sound, it starts early (before the provider has made progress) and stops at the expected time (when the other side answers) |
| 13:59.22 | Penguin | So then it sounds like your problem is solved. |
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| 14:00.04 | dan_j | But, it doesnt really work because if I start playing a ringing sound, and then the provider comes back a few moments later saying BUSY, some SIP phones get confused, and non-sip phones just lose the plot. |
| 14:00.36 | Penguin | Starts out ringing and then changes to busy tone? |
| 14:00.41 | Penguin | I hate that. |
| 14:00.54 | dan_j | Hence the problem. |
| 14:01.25 | dan_j | I dont think there is a solution to this. |
| 14:01.53 | Penguin | You may be right. Now that we've talked about it, maybe others will get interested and participate in the discussion. |
| 14:02.12 | dan_j | And if I do it, I'm sure i'll get a client asking why a foreign ring tone isn't played when the callee was travelling abroad. |
| 14:02.24 | WIMPy | Should be easy enough with a little programming. |
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| 14:04.15 | dan_j | I'll only spend time programming it if there is a chance to get it accepted into the asterisk code. So far, I've not seen anyone mention this issue. |
| 14:06.28 | WIMPy | In theorey it should be good enough to just configure Asterisk to never use inband progress, but you'd miss out on other things, then. |
| 14:07.02 | WIMPy | I haven't found a correct solution for progress signalling when calling out via SIP, yet. |
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| 15:02.26 | nny | I am bashing my head against the wall on this issue. The provider states "a diversion header is sending any incoming call forward calls to the 4 digit extension instead of the full 10 digit extension" WIMPy gave me some advice last night on how it works but I can't visualize where the issue originates or howcome it only affects this setup when I have numerous out there. Can I see this header with a sip debug? |
| 15:03.09 | WIMPy | I can't even parse that statement form your provider. |
| 15:03.28 | WIMPy | A diversion header doesn't send calls anywhere. |
| 15:03.54 | [TK]D-Fender | nny: These are your calls... of course you can see the debug. |
| 15:03.59 | WIMPy | So you better describe what's actually happening. |
| 15:04.24 | [TK]D-Fender | Screw descriptions... show the actual call so we have proof :) |
| 15:05.20 | nny | Doing so now |
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| 15:08.01 | nny | ok since I have to scrub some data to post the whole dialog here is the relevant packet in sip debug http://pastebin.com/TK2hwbw1 |
| 15:08.06 | pawiecki | Hello! Did you ever encounter Cisco IP 6901 Phones? I need to load SIP firmware instead od SCCP one. Can't find anything useful on the internet. Any ideas? |
| 15:08.17 | pawiecki | of* |
| 15:11.36 | nny | WIMPy: [TK]D-Fender check my post above, working on a full dialoge pastebin, the test person decided to check their vm after the test call >< |
| 15:12.31 | WIMPy | I don't even know what to look for. |
| 15:13.23 | WIMPy | Is teh diversion header used for different things in different messages? |
| 15:13.31 | nny | WIMPy: [TK]D-Fender Full pastebin here. http://pastebin.com/f03mH4qP |
| 15:14.42 | nny | WIMPy: I don't know ha. I am not clear why this is happening with just Level 3. From what I have read it pertains to certain provider switches (i.e. Broadsoft) and it should read something else. I assume they want it to read the full 10 digit of the account |
| 15:14.43 | [TK]D-Fender | pawiecki: Go download the SIP firmware for it |
| 15:15.05 | nny | I can set it in my dialplan |
| 15:15.27 | nny | via sip add header. Or there is this redirecting function thing I know nothing about and documentation is rather slim |
| 15:15.55 | nny | this seems like one of those situations where I shouldn't have to f*** with it but since I do, I do |
| 15:16.36 | pawiecki | [TK]D-Fender: I've got the firmware, but have no idea how to load it on the phone :) |
| 15:16.59 | [TK]D-Fender | nny: -- Executing [4047401627@sip:100] Dial("Local/4047401627@sip-91cb;2", "SIP/14047401627@voipstreet,,r") in new stack |
| 15:17.00 | [TK]D-Fender | [2016-01-13 10:04:50] WARNING[22479]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 15:17.23 | WIMPy | nny: Looks ok to me. But off course "6775" is not a valid caller ID, so that's why they won't like it. |
| 15:17.47 | nny | [TK]D-Fender: one sec, that's intentional. |
| 15:18.14 | [TK]D-Fender | nny: I see a redirect, but you're calling out another peer |
| 15:18.16 | nny | [TK]D-Fender: http://pastebin.com/h34s1pnQ |
| 15:18.43 | WIMPy | Yes, why do you call 6775 again, just via you provider? |
| 15:18.58 | [TK]D-Fender | pawiecki: They have an admin guide you know.... and like a million WIKI pages, blogs, etc for Cisco provisioning... |
| 15:19.14 | [TK]D-Fender | nny: Show the actual full call. |
| 15:19.16 | nny | [TK]D-Fender: the voipstreet part I can retest without but it's failover. Obviously they didn't pay the voipstreet bill heh |
| 15:19.30 | [TK]D-Fender | nny: This is crappy piecemeal.... |
| 15:19.31 | pawiecki | [TK]D-Fender: yes, for most of cisco phones, yes. |
| 15:19.33 | WIMPy | Hrm. Somehow the dial command doesn't match the debug or what am I missing? |
| 15:19.46 | [TK]D-Fender | pawiecki: This phone isn't that different |
| 15:20.02 | [TK]D-Fender | pawiecki: And I'm sure the guide for that model says what's still relevant to it |
| 15:20.31 | WIMPy | nny: Anyway, you probably just need to fix CALLERID(rdnis) to be a valid number. |
| 15:20.38 | nny | [TK]D-Fender: I believe the full call (based on the dialplan I showed you) is at 141 in this pastebin I posted earlier http://pastebin.com/f03mH4qP |
| 15:20.50 | nny | I can isolate it from the noise if need be in a new pastebin |
| 15:21.04 | nny | rdnis? |
| 15:21.26 | nny | interesting, can test |
| 15:21.43 | [TK]D-Fender | nny: 141 says nothing of value to me yet. It just gets thrown elsewhere in the dialplan. |
| 15:21.52 | [TK]D-Fender | nny: What are you DOING with it afterwards? |
| 15:22.27 | [TK]D-Fender | nny: We VERY clearly don't have the debug for the call it places after that |
| 15:22.35 | nny | [TK]D-Fender: Oh |
| 15:22.43 | nny | [TK]D-Fender: I don't have debugging enabled for the provider |
| 15:22.52 | nny | just that peer (6775) |
| 15:23.00 | [TK]D-Fender | Think you're just going to debug the phone and not the peer you are calling OUT ON? nonsense |
| 15:23.09 | nny | [TK]D-Fender: hahaa correct. |
| 15:23.12 | [TK]D-Fender | :p |
| 15:23.38 | [TK]D-Fender | You DO know better... and must be dreadfully low on caffeine |
| 15:23.54 | nny | [TK]D-Fender: also correct |
| 15:24.04 | nny | fixing both stand by |
| 15:29.46 | nny | [TK]D-Fender: may be a moot point, seems rdnis made it work although I am still testing. I had her test but trying to isolate the debug info from the cruft may be akward. i will try a clean debug if an issue persists now |
| 15:29.55 | nny | awkward too |
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| 16:57.43 | DisasteR | hi |
| 16:58.08 | DisasteR | got an error trying to match some text in a var containing double quote " |
| 16:59.59 | mastahh|work | Hi all, what HA tools do you guys use for your primary and backup call processing servers? |
| 17:03.16 | DisasteR | http://pastebin.com/Xi8jSMzY |
| 17:03.21 | DisasteR | here is my setup |
| 17:03.42 | DisasteR | trying the ExecIf on ARG3 |
| 17:04.19 | DisasteR | where ARG3 = "name" <sip:anonymous@172.24.1.24:5063>;....... |
| 17:05.15 | DisasteR | and got a syntax error cause when ARG3 is evaluated i got : $[""name" <sip:anonymous@172.24.1.24:5063>;......." =~ "anonymous"] |
| 17:05.30 | [TK]D-Fender | show the full debug |
| 17:05.34 | DisasteR | note the double "" before name |
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| 17:06.42 | DisasteR | [TK]D-Fender: last line on the pastebin |
| 17:06.57 | [TK]D-Fender | I have no proofFULL DEBUG |
| 17:07.03 | [TK]D-Fender | I can't prove what that line belongs to |
| 17:07.21 | DisasteR | ok |
| 17:07.23 | [TK]D-Fender | that means the previous lines before it executing, etc |
| 17:11.32 | DisasteR | http://pastebin.com/xhgvvFCp |
| 17:11.35 | DisasteR | is that ok ? |
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| 17:14.21 | DisasteR | [TK]D-Fender: do you need more ? |
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| 17:26.11 | [TK]D-Fender | Yup, nice big parsing error. |
| 17:26.26 | [TK]D-Fender | First use CUT() to chop the name part off of that |
| 17:26.53 | [TK]D-Fender | because you are comparing to tthe single word "anonymous". That var holds a massive string including other garbage. So that'll never work in the firstt palce |
| 17:32.13 | DisasteR | [TK]D-Fender: mayb to bve more accurate i can match =~ "sip:anonymous" |
| 17:32.38 | DisasteR | to avoid false match on user named anonymous |
| 17:32.45 | DisasteR | but how to extract that part ? |
| 17:33.04 | [TK]D-Fender | [12:26][TK]D-FenderFirst use CUT() to chop the name part off of that |
| 17:33.09 | DisasteR | cut @ get 1 and cut < get 2 ? |
| 17:36.07 | [TK]D-Fender | give it a try |
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| 17:55.03 | DisasteR | it works thank for your help [TK]D-Fender |
| 17:55.23 | DisasteR | exten => s,n,Set(MySipDest=${CUT(CUT(ARG3,@,1),<,2)}) |
| 17:55.30 | [TK]D-Fender | You're welcome. *'s variable handling is honestly kinda flakey |
| 17:55.34 | DisasteR | ExecIf($["${MySipDest}" = "sip:anonymous"]? |
| 17:56.01 | DisasteR | yeah |
| 17:56.05 | DisasteR | tim to go for me |
| 17:56.08 | DisasteR | time* |
| 17:56.16 | DisasteR | really big thanks |
| 17:56.21 | DisasteR | got day to you |
| 17:56.24 | [TK]D-Fender | as well |
| 17:56.27 | DisasteR | see you |
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| 18:11.38 | craigify | K lined! |
| 18:11.41 | craigify | bam |
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| 23:52.22 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |