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| 02:13.12 | Mango45 | Hi. |
| 02:13.30 | Mango45 | Is it possible, in one line, to increment a variable or create it with a value of 1 if it does not yet exist? |
| 02:13.38 | Mango45 | The best I've done is: exten => talk,1,Set(i=${IF($[${EXISTS(${i})}]?$[${i}+1]:1)}) |
| 02:13.52 | Mango45 | But that errors the first time because $[${i}+1] is invalid. (Though the code runs.) |
| 02:18.03 | Mango45 | HAH! exten => talk,1,Set(i=$[0${i}+1]) |
| 02:22.14 | [TK]D-Fender | yup. Old trick |
| 02:22.39 | [TK]D-Fender | Why wouldn't it be set though? |
| 02:23.55 | Mango45 | First line of that extension. |
| 02:24.45 | [TK]D-Fender | I'm asking why it WASN'T set prior in the first palce... |
| 02:25.16 | Mango45 | It wasn't set because nothing has happened yet. |
| 02:25.34 | [TK]D-Fender | It is standard procedure in programming to initialize your variables before testing them. This is usually indicative of some poor choices earlier. |
| 02:26.26 | Mango45 | I could initialize it; was trying to see how small I could make the code and still have it run. |
| 02:26.50 | [TK]D-Fender | Not worth the risks just to be sloppy |
| 02:27.16 | Mango45 | I don't think it risks anything; if it's the first line of the extension it's not possible for it to be already set, is it? |
| 02:27.37 | [TK]D-Fender | Don't make that the first line then |
| 02:27.50 | [TK]D-Fender | Set(i=0) |
| 02:28.01 | *** part/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 02:28.10 | [TK]D-Fender | Consider it the same as declaring your variables at all. |
| 02:30.24 | [TK]D-Fender | And on that note... checkout time, BBL |
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| 02:47.40 | StaRetji | howdy |
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| 02:47.57 | StaRetji | just a quick question, can I set asterisk 1.8 to listen on 2 ports? |
| 02:48.09 | StaRetji | without using iptables... |
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| 04:13.39 | ChannelZ | because it be old |
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| 04:42.50 | Penguin | What was the question? |
| 04:45.55 | ChannelZ | listen to multiple ports. Though in my brain he asked "why can't it" not "how can I" |
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| 08:32.38 | sekil | is there a way to get aoc vars in cdrs nowadays? |
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| 09:16.22 | WIMPy | sekil: Nothing new. |
| 09:21.16 | sekil | the whole infrastructure is there...the way to store that info is apparently to use ami |
| 09:22.44 | WIMPy | I'm not even sure they appear there. I went SIP in September, where the AOC messages aren't decoded any more. |
| 09:24.17 | sekil | hm...I'm using 11.6...I can see aoc-d being decoded |
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| 09:25.27 | WIMPy | Yu could copy the information to a CDR field via AMI then. |
| 09:26.18 | WIMPy | The issue that you'd miss the last message, would remain, I guess. |
| 09:26.29 | sekil | the aoc-e? |
| 09:26.42 | WIMPy | Yes. |
| 09:26.50 | michael_work | if anyone knows the way to get rtp ports used in specific channel/call ? |
| 09:27.00 | WIMPy | But if you have AOC-D that should be ok. |
| 09:27.13 | michael_work | without tcpdump of course and only using asterisk |
| 09:27.21 | sekil | I think delayed hangup takes care of that now |
| 09:27.45 | WIMPy | And I guess, you wouldn't get AOC information for calls without an Asterik channel. |
| 09:28.09 | WIMPy | But it might still be too late to change CDRs manually. |
| 09:28.38 | WIMPy | Let us know if you tried it :-) |
| 09:29.02 | sekil | I talked to a coleague recently ...I think he did some simple patches to set vars |
| 09:29.13 | sekil | he told me he has both aoc-e and aoc-d data |
| 09:29.17 | sekil | in the cdrs |
| 09:30.15 | WIMPy | A patch would be in execution order. If you do it via AMI, you might be too slow. |
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| 10:45.10 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 10:45.12 | [sr] | hi WIMPy |
| 10:45.48 | WIMPy | Hi [sr] |
| 10:47.08 | [sr] | ur up early |
| 10:47.23 | WIMPy | Kinda, yes. |
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| 11:30.49 | lmorgan69a | Morning all - has anyone had any joy with upgrading AsteriskNOW 3 to use Asterisk 12? Any pointers? |
| 11:31.26 | lmorgan69a | Really want to try out the new AMI updates |
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| 15:24.22 | funky1 | trying to register a trunk but get this back: "VoIP SIP register fail (481)" anyone knows what 481 is? |
| 15:27.25 | wdoekes | 481 is dialog not found, you shouldn't get that as response to a register |
| 15:27.59 | wdoekes | unless perhaps you're sending a to-tag in the register request (which would be wrong) |
| 15:30.04 | funky1 | i thought it is related to the useragent |
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| 15:51.37 | funky1 | i'm having the exact same problem as this user, is also same modem/router http://forums.digium.com/viewtopic.php?f=1&t=84744&start=0 |
| 15:52.04 | funky1 | going nuts with it, hope someone has an idea what is going wrong with registration here... |
| 15:57.43 | funky1 | anyone knows what does means? This status indicates that the UAS received a request that does not match any existing dialog or transaction. |
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| 17:24.16 | linjan | hello! is there any way to change default 's' extension to something else? maybe some option in sip.conf? |
| 17:24.35 | [TK]D-Fender | What's the point? |
| 17:24.48 | [TK]D-Fender | And what's the assumption of "default"? |
| 17:25.31 | linjan | point is provide correct data to CDR. now its like "A call from s to 100". And i would like to change it to smthing like "A call from 90973744 to 100" |
| 17:25.46 | linjan | we are talking about incoming calls |
| 17:26.08 | [TK]D-Fender | If a call comes in targeting "s" .. that's because that's what the caller is asking for |
| 17:26.13 | [TK]D-Fender | It is not a "fallback" |
| 17:26.54 | [TK]D-Fender | Typically providers would never ask for "s" for any reason other than your * TELLING to send calls to that. |
| 17:27.19 | [TK]D-Fender | Which would be because you didn't specify an extension to dial in your registration to them |
| 17:28.02 | [TK]D-Fender | register => user:pass@host/extension-or-if-you-leave-this-off-Asterisk-will-say-"s" |
| 17:28.03 | linjan | oh, solution maybe in correct registration string? |
| 17:28.19 | linjan | thanks! ill try it now! |
| 17:28.30 | linjan | double thanks for theory) |
| 17:28.39 | [TK]D-Fender | Go re-read the sip;.conf sample to see what else you've missed |
| 17:34.15 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:34.21 | Kobaz | linjan: did you set trustrpid=yes in your sip.conf for the user? |
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| 17:34.57 | linjan | nope |
| 17:36.26 | Kobaz | that'll help with your cdr. by default asterisk doesn't pass through the callerid if the remote end uses remote-party-id for the caller info |
| 17:39.39 | [TK]D-Fender | Kobaz: not related |
| 17:40.12 | [TK]D-Fender | Kobaz: Its the src extension |
| 17:40.26 | [TK]D-Fender | Kobaz: Which is the INVITE line itself and is due to registration |
| 17:41.05 | Kobaz | yeah, that part is |
| 17:41.19 | Kobaz | i'm just saying in general it's good to turn on trustrpid if working with an itsp |
| 17:42.18 | Kobaz | i think trustrpid no is a silly default |
| 17:42.51 | Kobaz | there's many more situations where you would want it versus when you would specifically not want ti |
| 17:43.07 | bsdice | Asterisk, we need to talk |
| 17:45.13 | bsdice | Where is it documented that when you touch a ConfBridge's settings anywhere in a dialplan, it just shoots any defaults set in confbridge.conf to hell? |
| 17:45.24 | bsdice | Setting like Set(CONFBRIDGE(user,admin)=yes) |
| 17:46.17 | Kobaz | that's persistent across channels after that's set? |
| 17:46.41 | bsdice | I can't answer that |
| 17:47.37 | Kobaz | so what do you mean it shoots the defaults? |
| 17:47.47 | Kobaz | like, at what point are the defaults changed? |
| 17:47.47 | bsdice | but if you call ConfBridge in dialplan with reference to user/bridge set in the configuration file, those are all reset to defaults |
| 17:48.01 | bsdice | example: |
| 17:48.18 | bsdice | I define extensions in dial plan to be conferences |
| 17:48.38 | [TK]D-Fender | Show your specific sample |
| 17:50.40 | bsdice | same => n,Set(CONFBRIDGE(user,admin)=yes) |
| 17:50.40 | bsdice | same => n,ConfBridge(${CONF_ID},default_bridge,default_user,default_menu) |
| 17:52.31 | Kobaz | why are you passing all those extra options? |
| 17:52.58 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 17:53.21 | bsdice | This will reset setting in confbridge.conf [default_bridge] sound_join=custom/join to default, out of the blue imho |
| 17:53.30 | file | 'tis documented |
| 17:53.38 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ConfBridge |
| 17:53.58 | file | basically... if it's blank it uses the dynamically created one in the dialplan, if it's not blank then it uses the one you provided |
| 17:54.20 | file | if you want to use a profile as a template you can use CONFBRIDGE(user,template)=default_user |
| 17:54.39 | bsdice | duh :) |
| 17:58.06 | bsdice | file are you certain it is documented if I set a user_profile feature that it changes a setting within bridge_profile? |
| 17:58.22 | bsdice | resets bridge_profile rather |
| 17:58.47 | file | I don't understand the question. The example above the CONFBRIDGE dialplan function usage wouldn't get used |
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| 17:59.38 | bsdice | let me check my example |
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| 18:03.24 | file | I don't see anything in the issue tracker either |
| 18:03.44 | bsdice | must be me :) |
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| 18:04.19 | bsdice | ok I just checked: if I do same => n,Set(CONFBRIDGE(user,admin)=yes) in dialplan, then in ConfBridge(${CONF_ID},default_bridge,default_admin,default_menu) the user profile is ignored |
| 18:04.30 | bsdice | which matched what you said |
| 18:05.07 | bsdice | ConfBridge checks if anything was set in CONFBRIDGE and if there was, it uses that instead of profile parameter |
| 18:05.19 | bsdice | CONFBRIDGE beats conf file |
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| 18:13.20 | bsdice | ok leave it at that |
| 18:13.27 | bsdice | "quirk" |
| 18:13.56 | bsdice | next up: [default_bridge] sound_join=/embinet/conf-join will produce: [2014-02-02 19:12:28] WARNING[4521][C-00000001]: file.c:701 ast_openstream_full: File confbridge-join does not exist in any format |
| 18:14.24 | bsdice | while gateway*CLI> confbridge show profile bridge default_bridge gives: |
| 18:14.33 | bsdice | sound_join: /embinet/conf-join |
| 18:15.32 | bsdice | it's ignoring my setting imho |
| 18:15.49 | bsdice | leading forward slash added to provoke file error |
| 18:16.07 | bsdice | and third, <Bridge/0x7fea6000cd78-input> Playing '/home/asterisk/spool/asterisk/confbridge/confbridge-name-8000-1391364724.3.slin' (language 'de') |
| 18:16.46 | bsdice | those slin files used to play introductory/exit messages are never deleted |
| 18:17.06 | bsdice | not sure if this is intended |
| 18:21.57 | bsdice | on the flip side I never used ConfBridge until yesterday |
| 18:22.18 | bsdice | and got a conferencing system with pin entry from external trunk working in under an hour |
| 18:22.25 | bsdice | using db and dialplan foo |
| 18:22.48 | bsdice | for free |
| 18:30.37 | bsdice | argh found another dialplan bug here |
| 18:30.54 | bsdice | <PROTECTED> |
| 18:31.19 | bsdice | same => n,Set(CONFBRIDGE(bridge,language)=de) -> messed up join/leave |
| 18:31.42 | bsdice | <PROTECTED> |
| 18:34.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:34.52 | bsdice | last issue: announce_join_leave=yes .slin file remains auto clearing when nobody is left in conference |
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| 19:31.54 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:31.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:33.19 | bsdice | . |
| 19:33.24 | bsdice | http://pastebin.com/AcbdDp5n |
| 19:33.58 | bsdice | made a cronjob to clean old ohai sln files |
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| 19:39.32 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 19:40.12 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 19:41.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 19:43.06 | wonderworld | bought the book. again. hail me. |
| 19:43.15 | bsdice | what book |
| 19:43.36 | bsdice | and what happened to to old book |
| 19:46.50 | [TK]D-Fender | sends a hailstorm to wonderworld |
| 19:47.00 | bsdice | ~book |
| 19:47.00 | 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 |
| 19:48.18 | wonderworld | ha! |
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| 20:08.39 | bsdice | hu |
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| 20:32.49 | skrusty | evening |
| 20:40.26 | sekil | WIMPy: I guess you're right...on 11.6 certified looks like aoc-e not fully decoded |
| 20:40.37 | sekil | is not |
| 20:52.59 | *** join/#asterisk vitovt (~vitovt@81.162.249.241) |
| 20:53.08 | vitovt | hi all! |
| 20:53.53 | vitovt | what does it mean ? |
| 20:53.55 | vitovt | [2014-02-02 22:53:36] WARNING[4560][C-00000031]: pbx.c:6390 __ast_pbx_run: Channel 'Dongle/sim3-010000000c' sent to invalid extension but no invalid handler: context,exten,priority=from-gsm,+1234567890,1 |
| 20:54.53 | [TK]D-Fender | exactly what it says. |
| 20:55.16 | [TK]D-Fender | The call is coming in... and it was told to look for an extension in your dialplan and there is NO MATCH for it just like it says |
| 20:55.35 | [TK]D-Fender | context,exten,priority=from-gsm,+1234567890,1 <------ |
| 20:56.26 | vitovt | thats my config: http://pastebin.com/Vnk4tkXC |
| 20:57.19 | [TK]D-Fender | vitovt: And just like the error says, there is no match in there. |
| 20:57.32 | vitovt | such configuration worked on asterisk 1.8 but want not on 11 |
| 20:57.49 | [TK]D-Fender | This has nothing to do with Asterisk version. |
| 20:58.05 | [TK]D-Fender | That dialplan does not match what is being send from chan_dongle |
| 20:58.17 | [TK]D-Fender | Fix your pattern |
| 21:00.30 | vitovt | if I know how ... |
| 21:00.51 | [TK]D-Fender | It is telling you what it is looking for.... |
| 21:00.53 | vitovt | I use freepbx |
| 21:01.34 | [TK]D-Fender | that is manual dialplan, that is not part of FreePBX |
| 21:02.19 | [TK]D-Fender | and that pattern is the same format as is used in an inbound route |
| 21:02.40 | [TK]D-Fender | ~book |
| 21:02.41 | 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 |
| 21:02.45 | [TK]D-Fender | Dialplan basics ^ |
| 21:03.35 | [TK]D-Fender | [15:55][TK]D-Fendercontext,exten,priority=from-gsm,+1234567890,1 <------ |
| 21:04.02 | [TK]D-Fender | It's looking for context [from-gsm]. I see you have one (supposedly). |
| 21:04.37 | [TK]D-Fender | It's looking for an EXTEN to match "+1234567890". You do NOT have one in there that matches it. |
| 21:05.14 | vitovt | ok, i've got the problem. The extension should be modem imei |
| 21:05.25 | vitovt | but it returns the default one ... |
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| 21:12.33 | vitovt | [TK]D-Fender Thankyou very much! |
| 21:12.40 | vitovt | It works now |
| 21:12.59 | [TK]D-Fender | vitovt: It's very important to look at precisely what is being targeted... |
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