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| 01:59.04 | saint_ | greetings all - is the PDF book Asterisk / The future of telephony / 2nd edition , the best way to start with Asterisk ..? |
| 01:59.29 | navaismo | ~book |
| 01:59.29 | 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 |
| 02:01.19 | saint_ | okay, so is this 3rd edition the best way to start with asterisk ? |
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| 02:01.55 | qakhan | hi all |
| 02:01.55 | navaismo | yes |
| 02:02.05 | qakhan | what is minimum hardware requirment of asterisk |
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| 02:03.13 | saint_ | qakhan: seriously ? |
| 02:03.18 | saint_ | qakhan: http://bit.ly/Wuf5p4 |
| 02:03.18 | navaismo | people have asterisn on routers like linksys wrt, on raspberry pi so depends on your needs I guess |
| 02:03.45 | navaismo | s/asterisn/asterisk |
| 02:04.02 | navaismo | s/asterisn/asterisk/ |
| 02:04.06 | navaismo | ¬¬ |
| 02:10.49 | saint_ | navaismo: i m curious . i me reading the asterisk book 3rd edition, and they talk about dialplan applications. |
| 02:10.54 | saint_ | where are those applications located ? |
| 02:11.25 | saint_ | never mind, i found it |
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| 02:12.36 | SeRi | saint_: seriously? |
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| 02:12.46 | SeRi | lol jk ;P |
| 02:12.51 | saint_ | <Oo> |
| 02:15.37 | SeRi | data migrations. They are so much fun :) best part is when we tell the users not to try to mount and still try and complaint that the "server" is broken :) |
| 02:16.14 | SeRi | all in all it was a success. |
| 02:16.23 | saint_ | what's a channel bridging (from bridging modules) ? Is it bridging in the sens of a conference bridge, where many channels are connected together ? Or is it bridging in the sens of connected 2 channels one to the other only ? |
| 02:16.27 | SeRi | now time to rest. |
| 02:17.17 | saint_ | root@moonlight:~> uptime |
| 02:17.18 | saint_ | <PROTECTED> |
| 02:17.23 | saint_ | can't do that with windows , lol |
| 02:18.40 | SeRi | saint_: depending on which moduels... bridge_blah.so Thats for channel it self. |
| 02:18.51 | saint_ | ok. |
| 02:19.01 | saint_ | i ll keep reading the book, i m sure i ll have the occasion to test it later on . |
| 02:19.02 | saint_ | thanks |
| 02:19.08 | SeRi | Thats an example... |
| 02:19.13 | SeRi | your welcome |
| 02:20.00 | SeRi | saint_: what type of services you run on that server? just asterisk? |
| 02:21.39 | SeRi | ok off to bed. |
| 02:21.42 | SeRi | g/n all! |
| 02:22.31 | p3nguin | saint_: Bridging is connecting of multiple channels, usually just two in the case of a "call." |
| 02:23.39 | saint_ | p3nguin: thanks |
| 02:23.51 | saint_ | SeRi: that server is not for asterisk, that's just one that i monitor for something else.. |
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| 03:25.32 | qakhan | hi i am install asterisk 11.0.1 on centos 6.3 |
| 03:25.54 | qakhan | when i install asterisk after ./configure |
| 03:26.00 | qakhan | make menuselect |
| 03:26.07 | qakhan | i got this message |
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| 03:27.58 | qakhan | The configure script must be executed before running 'make'. |
| 03:27.59 | qakhan | **** Please run "./configure". |
| 03:28.03 | echo777 | hey guys im back with a new problem |
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| 03:32.42 | echo777 | this is my current dialplan, i can make calls but cant recieve http://pastebin.com/bCA3ksLW |
| 03:33.44 | navaismo | qakhan, the configure script end wthout error |
| 03:34.05 | qakhan | navaismo what i do now |
| 03:34.31 | navaismo | im asking if the cofigue script ends without errors?? |
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| 03:37.09 | navaismo | s/configue/configure/ |
| 03:37.11 | qakhan | navaismo here is my error |
| 03:37.13 | qakhan | http://pastebin.com/6HgqVt56 |
| 03:37.34 | navaismo | yum install make |
| 03:37.49 | navaismo | then try again |
| 03:38.27 | qakhan | Package 1:make-3.81-20.el6.x86_64 already installed and latest version |
| 03:40.59 | navaismo | hmmm |
| 03:41.10 | navaismo | what architechture are you trying to install |
| 03:41.23 | qakhan | i am using centOS 6.3 |
| 03:41.33 | qakhan | asterisk 11.0.1 |
| 03:41.59 | echo777 | can i call one of you guys using my current settings and let me know whats wrong with the echo and delay im getting |
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| 03:45.27 | navaismo | qakhan, hardware? |
| 03:48.01 | echo777 | this is my current dialplan, i can make calls but cant recieve http://pastebin.com/bCA3ksLW |
| 03:48.06 | qakhan | 512MB ram, 1 processor, 20GB HD. its Virtual machine |
| 03:49.03 | navaismo | please re run the configure script and pb the output |
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| 03:49.36 | qakhan | ok |
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| 03:51.11 | qakhan | here http://pastebin.com/1ubJwVHa |
| 03:52.28 | echo777 | thanks for ignoring me :/ |
| 03:53.20 | navaismo | sorry echo777 im not ignoring you just cant help you |
| 03:54.35 | echo777 | why not |
| 03:55.11 | navaismo | qakhan, ---> configure: WARNING: *** Asterisk now uses SQLite3 for the internal Asterisk database. |
| 03:55.11 | navaismo | configure: WARNING: *** Please install the SQLite3 development package |
| 03:55.22 | navaismo | qakhan, install sqlite3 |
| 03:55.30 | navaismo | echo777, you cant call me |
| 03:55.41 | echo777 | no i meant with the dialplan |
| 03:55.41 | qakhan | yum install sqlite3? |
| 03:55.43 | echo777 | this is my current dialplan, i can make calls but cant recieve http://pastebin.com/bCA3ksLW |
| 03:55.59 | navaismo | let me see |
| 03:56.05 | navaismo | qakhan, yes try that |
| 03:56.21 | qakhan | yum install sqlite3 not working |
| 03:57.33 | navaismo | echo777, what show the cli when you try to call in |
| 03:58.38 | echo777 | hang on |
| 03:59.22 | echo777 | oh. it works fine,, odd it didnt last night |
| 04:00.04 | tonyclewis | echo777: this is common with google voice |
| 04:00.13 | tonyclewis | seems randomly calls do not come in |
| 04:00.37 | navaismo | qakhan, try yum install sqlite |
| 04:00.39 | tonyclewis | and outbound seems there gateways also just loose calls |
| 04:01.36 | echo777 | sometimes its choppy, is that my connection orr? |
| 04:02.25 | tonyclewis | could be anything |
| 04:02.40 | tonyclewis | but I would never use google voice as primary line |
| 04:02.43 | tonyclewis | or rely on it |
| 04:03.33 | qakhan | i installed sqlite but same message |
| 04:05.54 | qakhan | navaismo i got it |
| 04:06.10 | qakhan | yum install sqlite-devel* |
| 04:06.28 | qakhan | sqlite-devel was not installed :) |
| 04:07.42 | navaismo | great |
| 04:07.57 | navaismo | re run the configure and try again |
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| 04:14.31 | qakhan | yes its working now |
| 04:16.25 | navaismo | fine |
| 04:19.04 | qakhan | navaismo now i m getting this error when i run make |
| 04:19.23 | qakhan | http://pastebin.com/55kVb3kJ |
| 04:20.21 | navaismo | its openssl installed |
| 04:20.34 | navaismo | and openssl-devel? |
| 04:23.53 | qakhan | now i installed |
| 04:26.13 | qakhan | same result |
| 04:28.10 | navaismo | are you compiling with -j<n> where n is the number of cpus? |
| 04:28.50 | navaismo | qakhan, try the last response from here http://forums.digium.com/viewtopic.php?f=1&t=84775&start=0 |
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| 04:36.12 | qakhan | navaismo i did this |
| 04:36.13 | qakhan | cd res/pjproject |
| 04:36.13 | qakhan | ./configure |
| 04:36.13 | qakhan | cd ../.. |
| 04:36.26 | qakhan | and now i m getting this error |
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| 04:36.58 | qakhan | http://pastebin.com/UhM0QyjQ |
| 04:44.14 | navaismo | hmm weird in that thread someone reported to JIRA check if there is a solution |
| 04:44.20 | navaismo | got to go now good luck |
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| 04:48.28 | echo777 | hey qakhan maybe i can help |
| 04:48.46 | qakhan | yes plz |
| 04:49.16 | qakhan | here i want to let you know that i am installing on virtual machine |
| 04:49.17 | echo777 | what seems to be the problem |
| 04:50.05 | echo777 | ok |
| 04:50.50 | qakhan | echo777 did u see my pb? |
| 04:51.02 | echo777 | \not yet hang on |
| 04:53.06 | echo777 | did you read that thread navaismo posted? |
| 04:54.51 | qakhan | yes |
| 04:55.09 | echo777 | and solutions? |
| 04:56.52 | qakhan | it fix one error and 2nd came up |
| 04:57.01 | echo777 | which is? |
| 04:58.44 | qakhan | the post on pb |
| 04:58.54 | qakhan | posted* |
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| 04:59.05 | echo777 | hmm. |
| 04:59.16 | echo777 | anyone got any help for him? |
| 05:01.25 | asr33 | Hello folks, how would I disable SRTP SIPS in Asterisk 1.8.14.1? |
| 05:09.45 | asr33 | I've place "encryption=no" in "[general]" section of sip.conf without success |
| 05:12.42 | qakhan | guys any update |
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| 05:34.11 | ChannelZ | qakhan: Are you trying to cross-compile or something? |
| 05:38.29 | ChannelZ | actually.. what the hell is pjproject anyway |
| 05:43.41 | ectospasm | pjsip is a SIP implementation |
| 05:44.19 | ectospasm | ...I know the Asterisk devs were thinking of replacing chan_sip with the pjsip stack. Don't know where that went, unfortunately. |
| 05:55.11 | ChannelZ | hmm |
| 05:57.15 | ChannelZ | I see it in my build directory but it doesn't look like it's really done anything with it |
| 05:57.40 | ectospasm | yeah, I don't know how far they got with it |
| 05:57.50 | ectospasm | I just remember hearing about it at work |
| 05:58.00 | ectospasm | (I work for Digium) |
| 05:58.01 | ChannelZ | So I'm wondering why qakhan's build is fiddling in there |
| 05:58.12 | ectospasm | I didn't see his problem statement |
| 05:59.49 | ectospasm | hmmm... I don't know what the state of pjsip is... qakhan, unless you need some specific feature provided by it, you should probably avoid pjsip for production. |
| 06:00.25 | ChannelZ | He was getting this on a make - http://pastebin.com/55kVb3kJ |
| 06:02.03 | ChannelZ | then he fiddled with something and got a different error which is the one I initially saw.. |
| 06:02.53 | ChannelZ | but my guess is it's a red herring caused by whatever failed prior |
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| 06:46.19 | echo777 | waffles |
| 06:46.33 | ChannelZ | mmmmmm |
| 06:46.37 | ChannelZ | with ice cream |
| 06:46.47 | echo777 | ýes |
| 06:48.20 | echo777 | and hot sauce :P |
| 06:59.28 | ectospasm | hot caramel sauce |
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| 07:15.48 | ChannelZ | with no pants on |
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| 07:24.24 | ectospasm | I dunno, I don't want hot caramel going anywhere near my nether bits |
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| 07:24.40 | bombev | Good Morning to all |
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| 10:24.20 | metiu_ | Hi, I'm asking here before filing a bug report: Asterisk 11.0.1, Confbridge with only two participants, ulaw 8000Hz on both sides, one muted, one admin, sound is stuttering badly. I checked with tcpdump, and wireshark shows packets as "skewed" more and more. CPU load is around 50% and same setup works properly with A* 1.6.2 and MeetMe |
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| 10:35.18 | kaldemar | metiu_: what timing module are you using? |
| 10:39.52 | nunne | metiu_: using debian? because I have had various strange bugs because of the default timing module in debian |
| 10:42.57 | metiu_ | kaldemar: well, good question. in 1.6.2 I have dahdi compiled in |
| 10:43.27 | metiu_ | nunne: using Angstrom from openembedded and linux 3.0.x |
| 10:44.15 | metiu_ | how do I check which timing module I'm using? I currently don't have the system here, but I'm going to check very soon and get back to you. Thank you for now |
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| 10:50.33 | nunne | metiu_: I have no experience with Ångström, so don't know how kernel timing works there. You can do "timing test" in console and it will say what module your using |
| 10:50.40 | kaldemar | metiu_: "timing test" in CLI will tell you. |
| 10:51.47 | nunne | metiu_: and to see what you can avaible write "module show like res_timing_" and then simply disable / enable with noload / load in /etc/asterisk/modules.conf |
| 10:51.51 | kaldemar | metiu_: "module show like timing" lists timing modules if you happen to have more than one loaded. disable those you don't want to use with noloads in modules.conf. |
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| 10:55.27 | metiu_ | ok, very good |
| 10:55.32 | metiu_ | thank you again |
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| 12:53.40 | zafu | hi, are mgcp phones hard to use with asterisk? |
| 12:56.21 | ectospasm | zafu: I don't think the mgcp stack is well maintained |
| 12:56.28 | ectospasm | they can work, but it's a pain to set up |
| 12:56.35 | zafu | ok |
| 12:56.37 | ectospasm | ...the documentation is stale at best |
| 12:56.42 | zafu | ouch |
| 12:56.58 | ectospasm | ...I've helped a customer set it up, but there were some undocumented things I don't remember |
| 12:57.06 | ectospasm | it's been like a year since I helped that guy out |
| 12:57.19 | ectospasm | ...if you've got the budget for it, get new phones. |
| 12:57.34 | zafu | a potential customer has ShoreTel's 230 (MGCP apparently), should I tell him to dump them? |
| 12:57.55 | zafu | new phones, got that :) |
| 12:59.48 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 13:00.32 | Ice_Strike | I am setting up the call center configuration |
| 13:00.46 | Ice_Strike | Do you have any idea what that varibles is reffering to http://pastebin.com/4pUnznWT |
| 13:01.35 | Ice_Strike | What does external_line_type and internal_line_type reffering to? |
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| 13:20.00 | qakhan | hi all |
| 13:20.54 | qakhan | i m going to install asterisk 10 on centos 6.0 please let me know which package is required for asterisk 10 on centos 6 |
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| 14:10.45 | *** join/#asterisk whtsup (~whtsup@WimaxUser38143-21.wateen.net) |
| 14:10.48 | whtsup | hello0 |
| 14:11.04 | whtsup | can i set default hangup cause for everycall |
| 14:11.08 | whtsup | ? |
| 14:11.24 | Rokfan | Hi guys. Is it possible to register a sip trunk using realtime? |
| 14:11.48 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 14:12.41 | AkkerKid | so I can originate a call to an internal extension from within the CLI but getting it to make an outbound call isn't working yet. what am I missing? |
| 14:14.07 | kaldemar | originate usage |
| 14:14.46 | AkkerKid | i'm typing channel originate SIP/1064 extension s@MyContext |
| 14:15.01 | AkkerKid | but if I replace 1064 with a POTS number, i get nothing |
| 14:15.30 | kaldemar | whtsup: by default in a single parameter? no. depends on your whole dialplan how to achieve that. i wouldn't recommend such forgery though. |
| 14:16.02 | kaldemar | AkkerKid: how could you? |
| 14:16.38 | kaldemar | AkkerKid: how do you connect to PSTN (note, PSTN, not POTS)? |
| 14:16.46 | AkkerKid | sip trunk |
| 14:17.04 | kaldemar | do you have it configured sip sip.conf? |
| 14:17.11 | kaldemar | do you have it configured in extensions.conf? |
| 14:17.27 | AkkerKid | i would imagine. I've been using it for years... |
| 14:17.44 | AkkerKid | for normal, human initiated calls, anyway... |
| 14:17.56 | kaldemar | then use the same dialstring that you already have in extensions.conf. |
| 14:18.44 | kaldemar | if you give asterisk SIP/123456789 it will first try to look for [123456789] in sip.conf and use that. if it is not found, it tries to use 123456789 as a host. |
| 14:19.21 | kaldemar | SIP/peer/123456789 would dial 123456789 via "peer". |
| 14:19.27 | AkkerKid | hmmm... |
| 14:19.30 | AkkerKid | AHA! |
| 14:19.33 | AkkerKid | let me try that |
| 14:20.11 | AkkerKid | great it worked! |
| 14:20.39 | AkkerKid | now I have to figure out how to get it to go through my usual outbound routes and use my callerID |
| 14:20.59 | p3nguin | I told you. Set up an extension just for setting things before the Dial(). |
| 14:21.09 | p3nguin | Then dial that extension through the local channel. |
| 14:22.28 | p3nguin | If you're only dialing North American numbers, your extension pattern is _NXXNXXXXXX |
| 14:22.38 | AkkerKid | is there any way I could force that call to go through the [from-internal] context on it's way out? |
| 14:22.42 | p3nguin | its |
| 14:22.48 | kaldemar | AkkerKid: you can also use what you already have in dialplan with the Local channel. instead of SIP/... you use Local/exten@context, where exten is the number you dial and... |
| 14:23.05 | AkkerKid | aha |
| 14:24.06 | AkkerKid | we have a winner! |
| 14:24.44 | AkkerKid | Good thing I've already built extens that augment CID info as calls go through them |
| 14:24.59 | AkkerKid | thanks guys! |
| 14:25.34 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:30.55 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
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| 14:40.05 | asr33 | hello folks, what would be the official way of disable SRTP SIPS in Asterisk asterisk-1.8.14.1? |
| 14:45.31 | Ice_Strike | Why did this happen |
| 14:45.32 | Ice_Strike | No such command 'show agents' (type 'core show help show agents' for other possible commands) |
| 14:47.57 | [TK]D-Fender | Ice_Strike, because that command isn't valid |
| 14:48.09 | [TK]D-Fender | Ice_Strike, Syntax changes between version. |
| 14:48.17 | Ice_Strike | Oh damn |
| 14:48.22 | [TK]D-Fender | "help" and tab auto-complete are your friends... |
| 14:48.52 | Ice_Strike | Call Center script is executing show agents |
| 14:49.11 | Ice_Strike | Might downgrade old version of asterisk |
| 14:49.15 | [TK]D-Fender | Then go change it. |
| 14:53.08 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 14:53.51 | Ice_Strike | does asterisk 1.4 support show agents |
| 14:54.18 | [TK]D-Fender | <[TK]D-Fender> "help" and tab auto-complete are your friends... |
| 14:54.27 | qakhan | i m install asterisk 10.10.0 on centOS 6 |
| 14:54.27 | [TK]D-Fender | Asterisk 1.4 = ancient |
| 14:54.50 | qakhan | and getting this error |
| 14:54.51 | qakhan | make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. |
| 14:55.26 | asr33 | I've put "encryption=no" in "[general]" section of sip.conf, and it doesn't work? |
| 14:56.55 | Ice_Strike | What the correct command for "show agents" in the new version? |
| 14:58.38 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:58.38 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:58.43 | Ice_Strike | Ah agent show |
| 15:03.15 | [TK]D-Fender | asr33, Is that .... a question? |
| 15:03.24 | [TK]D-Fender | asr33, Does your PEER allow it? |
| 15:03.29 | [TK]D-Fender | asr33, Got a call to show us? |
| 15:03.33 | [TK]D-Fender | ~pb |
| 15:03.33 | 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. |
| 15:03.40 | [TK]D-Fender | ^ your friend.... |
| 15:04.41 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:05.41 | asr33 | [TK]D-Fender: I can't get my snom300 to work with it all my phones are on a local LAN and if I could disable SRTP SIPS it just would simplify my life? |
| 15:06.05 | *** join/#asterisk bchia (~Adium@nat/digium/x-faiapcxbodbexxxb) |
| 15:06.16 | [TK]D-Fender | asr33, We don't see that SRTp is being attempted.... or refused. You aren't actually showing us the problem. |
| 15:06.50 | asr33 | plus my ITSP doesn't support SRTP or SIPS or TLS |
| 15:07.50 | [TK]D-Fender | asr33, You are giving us partial facts and no prrof. |
| 15:07.56 | asr33 | the problem is I need to disable a quasi feature that just over complicates my setup |
| 15:07.58 | [TK]D-Fender | You are not actually looking at the call and the problem. |
| 15:09.58 | asr33 | the solution could be to remove Asterisk and just register my Snom with my ITSP? |
| 15:11.02 | qakhan | [TK]D-Fender can u help me in this |
| 15:11.20 | qakhan | make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. |
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| 15:19.38 | [TK]D-Fender | qakhan, http://forums.digium.com/viewtopic.php?f=1&t=80729 |
| 15:28.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 15:32.20 | slav3_kitten | ok what am i doing wrong... this line same => n,GotoIf($[DB(${CALLERID(num)}/blockid)=1]?blocked); |
| 15:32.51 | [TK]D-Fender | slav3_kitten, You aren't referencing your function. |
| 15:33.07 | slav3_kitten | ... whoops |
| 15:33.23 | slav3_kitten | so it should be $[$DB |
| 15:33.30 | [TK]D-Fender | no. |
| 15:33.39 | [TK]D-Fender | that is not how you reference a function |
| 15:33.49 | slav3_kitten | $[${DB * |
| 15:34.05 | [TK]D-Fender | ${FUNCTION(parameters...)} |
| 15:34.28 | [TK]D-Fender | $[] is an expression. that is a completely separate layer |
| 15:34.41 | slav3_kitten | odd, that breaks syntax hilighting in vi lol |
| 15:34.47 | [TK]D-Fender | Do not intertwine them. Each has to be right by itself |
| 15:35.50 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
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| 15:36.43 | slav3_kitten | wonder why that breaks syntax hilighting ... |
| 15:39.18 | p3nguin | For me, it only breaks the hilight on the ] |
| 15:39.22 | p3nguin | Everything else is right. |
| 15:40.04 | slav3_kitten | p3nguin, mind pastebin of your vi syntax.vim? |
| 15:42.44 | p3nguin | It is the default that came with vim-runtime 7.3.138: http://pastebin.com/fsphzkw7 |
| 15:43.10 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:46.05 | slav3_kitten | odd.. mine is much more full of stuff |
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| 15:51.38 | slav3_kitten | yea i'm using the ones that came with the svn for 11 and 1.8 an both break syntax hilighting |
| 15:51.40 | slav3_kitten | oh well |
| 15:51.45 | p3nguin | I also only use vim and never use vi. Maybe that makes a difference. |
| 15:52.22 | p3nguin | There's also an asterisk.vim and an asteriskvm.vim syntax file. |
| 15:53.01 | p3nguin | " Updated for 1.2 by Tilghman Lesher (Corydon76) |
| 15:53.01 | p3nguin | " Last Change: 2006 Mar 20 |
| 15:53.01 | p3nguin | " version 0.4 |
| 15:53.05 | *** join/#asterisk Alex25 (~kvirc@bzq-79-183-209-222.red.bezeqint.net) |
| 15:53.30 | p3nguin | Need to see that? |
| 15:54.35 | Alex25 | How to check if a variable is numeric-only? |
| 15:55.23 | *** join/#asterisk felipealmeida (~user@139.82.86.2) |
| 15:55.42 | leifmadsen | the variable name, or contents of the variable? |
| 15:55.49 | leifmadsen | you could probably use the REGEX() function |
| 15:56.13 | Alex25 | content of variable |
| 15:56.31 | Alex25 | AKA value |
| 15:57.43 | slav3_kitten | well removing vim-nox and installing vim solved that |
| 15:58.24 | Alex25 | thank you that seems right :) |
| 15:58.25 | Alex25 | exten => 123,1,Set(foo=${REGEX("[abc][0-9]" b3)}) ; returns 1 |
| 15:59.02 | Alex25 | That's an example i just found |
| 16:00.49 | Alex25 | does that mean that 'foo' will be 1, and will be an integer? |
| 16:01.09 | p3nguin | 1 is always an integer. |
| 16:01.52 | Alex25 | it can also be a string, no? |
| 16:02.23 | Alex25 | at least on other programming language it works that way.. |
| 16:02.28 | p3nguin | 1, by itself, is an integer. Your function can utilize it as a number or a string. |
| 16:03.55 | p3nguin | For example comparing 2 > 1, 1 is a number. In "1" < "2" it is a string. |
| 16:04.04 | p3nguin | It's all how you use it. |
| 16:04.21 | Alex25 | so i guess i need to add quotes to get it considered as a string |
| 16:04.28 | Alex25 | i mean |
| 16:04.30 | p3nguin | Usually, yes. |
| 16:04.36 | Alex25 | if foo=1 |
| 16:05.05 | p3nguin | In that, foo is just going to equal the number. It is neither a number nor a string until used in an expression. |
| 16:05.30 | p3nguin | s/number/numeral/ |
| 16:05.31 | Alex25 | to refer to it on another priority as a string i need to call it "${foo}" |
| 16:05.32 | slav3_kitten | kind of like perl iirc |
| 16:05.36 | Alex25 | right? |
| 16:05.45 | slav3_kitten | but it's been years since i've done perl so i could be mistaken |
| 16:06.04 | p3nguin | You're going to have to use ${foo} to dereference it, no matter what. |
| 16:06.20 | p3nguin | It isn't used in an expression yet, so it is still just a 1. |
| 16:06.47 | p3nguin | If you did $[${foo} < 3], then it is an integer. |
| 16:07.25 | p3nguin | If you did $["${foo}" != "bar"}, it is a string value. |
| 16:07.53 | p3nguin | Without using it in an expression, it merely exists. |
| 16:08.30 | Alex25 | so i mean, if i want to refer it later as an expression, and i want existing foo considered as a string |
| 16:08.40 | Alex25 | can I just use "${foo}" ? |
| 16:08.46 | p3nguin | Just quote it and compare it to something that isn't a number. |
| 16:09.40 | qakhan | [TK]D-Fender i didnt get solution in this link http://forums.asterisk.org/viewtopic.php?p=168545 |
| 16:09.56 | Alex25 | thank you. going to test |
| 16:10.15 | [TK]D-Fender | qakhan, it tells you to delete a line |
| 16:10.29 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 16:10.30 | [TK]D-Fender | Alex25, * dialplan has not data types |
| 16:10.33 | [TK]D-Fender | no* |
| 16:10.49 | [TK]D-Fender | Alex25, Quotes are literal only. |
| 16:11.07 | qakhan | i m confuse which line |
| 16:11.13 | qakhan | plz help me |
| 16:11.38 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.194) |
| 16:12.02 | Alex25 | I just want to learn how to convert an integer-var into a string-var in dialplan xpression |
| 16:12.21 | p3nguin | It's all about how you use it in the expression. Other than that, the value simply exists. |
| 16:12.44 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 16:13.09 | p3nguin | What I mean by just existing is that it is neither a string nor an integer. It just is what it is. It is the value. |
| 16:13.10 | WIMPy | Alex25@ There's no conversion other than implicit. |
| 16:13.40 | p3nguin | Until you do something with the value or the variable, it is just a value in memory. |
| 16:13.40 | Alex25 | so as a thumb rule - i just need to add quotes "". that's what i understand |
| 16:13.52 | p3nguin | It depends on the expression. |
| 16:13.58 | p3nguin | I don't know how many ways I can say this. |
| 16:14.20 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-208-220.red.bezeqint.net) |
| 16:14.33 | p3nguin | Outside of an expression, it is just a value in RAM. |
| 16:14.42 | WIMPy | It doesn't really matter, what you add, but quotes are an obvious choice. |
| 16:15.06 | p3nguin | I often add a letter to my variable when it is a number to turn it into a string. |
| 16:15.21 | p3nguin | if x${foo} = x${bar}, then ... |
| 16:17.30 | slav3_kitten | p3nguin, with ${CHANNEL} how do i strip the -000000b2 of kittenroom-000000b2 and will it always be the same length? |
| 16:17.37 | Alex25 | thanks for the tip |
| 16:17.55 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 16:18.53 | slav3_kitten | i was thinking like ::9 |
| 16:18.58 | WIMPy | slav3_kitten: If you know it's a SIP cahnnel you can use CHANNEL(peername). Otherwise use CUT. |
| 16:19.14 | slav3_kitten | ohh |
| 16:19.20 | Alex25 | what is best practice to check if a var is not empty? |
| 16:19.32 | WIMPy | Alex25: ISNULL() |
| 16:19.46 | WIMPy | Or add garbage like in p3nguin example. |
| 16:20.21 | slav3_kitten | it /should/ always be a sip channel.. wait i'll have an SCCP too |
| 16:20.24 | slav3_kitten | looks up cut |
| 16:20.29 | WIMPy | slav3_kitten: Not all channels create names line that, BTW. |
| 16:20.44 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:20.55 | Alex25 | thanks again! |
| 16:21.01 | WIMPy | So the need not be a "-" in the name. |
| 16:21.03 | WIMPy | there |
| 16:21.29 | slav3_kitten | well that's going to complicate things |
| 16:21.42 | WIMPy | It does. |
| 16:21.48 | slav3_kitten | maybe i'll just not have the sccp be able to use wakeup call |
| 16:21.53 | WIMPy | What are you trying to do? |
| 16:22.10 | WIMPy | User the caller ID? |
| 16:22.15 | WIMPy | -r |
| 16:22.40 | slav3_kitten | yea i've got my wakeup call creating a .call in the future with the SIP/ |
| 16:22.56 | slav3_kitten | i had it setup for a variable i created in each peer entry in my sip.conf |
| 16:23.06 | slav3_kitten | but i thought there had to be a better way. guess not though |
| 16:23.12 | *** part/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 16:23.22 | WIMPy | I use the caller ID via a local channel. |
| 16:23.23 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 16:23.44 | WIMPy | That way I don;t need to know the peer. |
| 16:23.51 | slav3_kitten | well i have the caller ID set to be Kitten <100> an such |
| 16:24.11 | slav3_kitten | that way it displays correctly who's calling from what phone on internal stuff |
| 16:24.25 | WIMPy | And you can't call 100? |
| 16:25.18 | slav3_kitten | i can from the phones... maybe i have the context wrong in my .call |
| 16:25.34 | p3nguin | what good is a cid num if you cant dial it? |
| 16:25.44 | WIMPy | Just use the right one :-) |
| 16:26.11 | slav3_kitten | right now i have same => n,System(echo "Channel: SIP/${CHANNEL(peername)}\nMaxRetries: 2\nRetryTime: 60\nWaitTime: 30\nContext: Wake-Up\nExtension: 23\nCallerID: ${CALLERID(all)}" > /tmp/${UNIQUEID}.call); |
| 16:26.54 | *** join/#asterisk festr_ (~festr@voipmonitor.org) |
| 16:26.55 | festr_ | hi |
| 16:27.05 | WIMPy | So that definitely only works for sip. |
| 16:27.17 | slav3_kitten | so that should be echo "Context: Phones\nExtension: ${CALLERID(num)}\nMaxRetries: ? |
| 16:27.31 | festr_ | I need to parse in SIP channel INVITE sip:123@bla;group_id=3 |
| 16:27.35 | festr_ | which is in RURI |
| 16:27.43 | festr_ | I need to get either the RURI in some variable |
| 16:27.56 | festr_ | or get the group_id (better option but can parse with CUT or so) |
| 16:28.09 | slav3_kitten | WIMPy, i had it previously Channe: ${CALLERCHAN} which i defined in my sip.conf for each peer |
| 16:28.11 | festr_ | is that possible? I'm trying hard to find out how to get the RURI but cannot see it anywhere in google |
| 16:28.20 | p3nguin | alex25: ISNULL() to check if null, EXISTS() to check if nonnull. |
| 16:28.25 | WIMPy | slav3_kitten: Something like channel: local/${CALLERID(num)}@internal |
| 16:28.39 | slav3_kitten | oh |
| 16:30.00 | Alex25 | cool. just found this page on Google. Very helpful: http://the-asterisk-book.com/1.6/funktionen-regex.html |
| 16:31.07 | p3nguin | slav3_kitten When I am not on my phone with a small screen and kb, I'll show you how I trim channel names. |
| 16:31.25 | slav3_kitten | p3nguin, thanks |
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| 16:39.33 | *** join/#asterisk deo (~deo@112.198.90.214) |
| 16:39.48 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:39.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:41.10 | p3nguin | slav3_kitten: ${CHANNEL:0:-9} |
| 16:41.58 | WIMPy | For the current implementation of the sip channel. |
| 16:43.21 | slav3_kitten | now i feel kinda dumb for not thinking negative length |
| 16:44.18 | WIMPy | It's a gamble anyway. |
| 16:44.47 | slav3_kitten | yea, WIMPy has the best solution with the local/calleridnum@localsets |
| 16:46.15 | WIMPy | Asterisk 12 will get a new sip channel. It might not even work for sip then. |
| 16:49.50 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.208) |
| 16:54.46 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:00.04 | *** join/#asterisk bkw_ (~Adium@freeswitch/developer/bkw) |
| 17:00.22 | bkw_ | WHAT HAPS! |
| 17:01.27 | [TK]D-Fender | bkw_, NEXT@!@!@@!ONE@!@!AT!!!ELEVEN |
| 17:08.18 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:09.35 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:dc5:7a98:f50d:2086) |
| 17:14.38 | qakhan | [TK]D-Fender i m getting this now |
| 17:14.39 | qakhan | res_rtp_asterisk.c:(.text+0x1d26): undefined reference to `pj_ice_sess_change_role' |
| 17:14.39 | qakhan | collect2: ld returned 1 exit status |
| 17:14.55 | qakhan | after remove the line EMBED_LDSCRIPTS+=../main/modules.link |
| 17:15.54 | Qwell | qakhan: Why are you changing things? |
| 17:17.11 | qakhan | @Qwell i am installing Asterisk 11 on centOS 6 |
| 17:17.50 | qakhan | when i was installing Asterisk i was getting |
| 17:17.51 | qakhan | make[1]: *** No rule to make target '../main/modules.link' , needed by 'asterisk'. Sto |
| 17:17.51 | Qwell | Don't embed modules. |
| 17:18.16 | qakhan | [TK]D-Fender send me this link http://forums.asterisk.org/viewtopic.php?p=168545 |
| 17:18.22 | qakhan | and i followed it |
| 17:18.34 | qakhan | now i m getting above error |
| 17:18.53 | Qwell | Disable embedding. |
| 17:19.27 | qakhan | how? |
| 17:19.32 | Qwell | The same way you enabled it. |
| 17:21.27 | qakhan | in make menuselect? |
| 17:21.31 | Qwell | yes |
| 17:21.58 | qakhan | what is the perpose of module embedding |
| 17:23.31 | Alex25 | could you tell why the following is not working? |
| 17:23.33 | bkw_ | Qwell: walk over and bridges lately ? |
| 17:23.33 | Alex25 | Set(str="466") ; Verbose(2,output is ${REGEX("^[0-9]+$" ${str})}) |
| 17:23.49 | Alex25 | it returns 0 instead of 1 |
| 17:28.23 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 17:36.39 | [TK]D-Fender | Alex25, Quotes are literal and is PART of the string. There are no data types in * dialplan. It IS that dumb. Stop using quotes like that. |
| 17:36.52 | [TK]D-Fender | Alex25, Are we clear now? |
| 17:38.37 | Alex25 | i'll test without quotes. moment |
| 17:43.16 | Alex25 | ok it's working now |
| 17:44.22 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 17:51.30 | *** join/#asterisk mjordan (~mjordan@50.201.8.0) |
| 17:51.30 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:52.13 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:53.23 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 17:53.24 | p3nguin | I guess I need to be more blunt sometimes. |
| 18:03.40 | *** part/#asterisk bkw_ (~Adium@freeswitch/developer/bkw) |
| 18:06.47 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:06.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:14.47 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:15.22 | *** join/#asterisk jsjc (~Adium@75.157.78.188.dynamic.jazztel.es) |
| 18:20.35 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 18:26.16 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:28.05 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:28.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:32.22 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 18:45.52 | *** join/#asterisk jsjc (~Adium@75.157.78.188.dynamic.jazztel.es) |
| 18:48.11 | *** join/#asterisk apb1963 (~apb1963@174.134.102.14) |
| 18:51.34 | cusco | hi folks |
| 18:51.51 | cusco | how can I know if asterisk 1.6.2 supports eventfilter in manager.conf ? |
| 18:52.47 | malcolmd | i believe event filter was introduced in 1.8 |
| 18:53.13 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 18:53.29 | Qwell | Why bother with 1.6.2? It's dead. |
| 18:53.44 | cusco | because queue_log structure is being used |
| 18:53.53 | cusco | real time queue_log db structure |
| 18:54.04 | cusco | and 1.8 introduces strucuture changes |
| 18:54.56 | cusco | we do use 1.8 in simple gateways or such |
| 18:55.10 | cusco | but our main, db writting/reading stuff, is 1.6.2 |
| 18:56.16 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:56.28 | cusco | :/ |
| 18:57.27 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:57.36 | JasonL | I’m using a Digium TE420F (no echo can.) to passthrough (and record) a telco PRI to a Nortel PBX. We are getting echo and not sure how to get rid of it. I have echocancel=yes in chan_dahdi.conf and echocanceller=mg2 in system.conf. Can anyone help? |
| 18:58.17 | cusco | do the recorded calls have echo? |
| 18:58.29 | JasonL | no |
| 18:58.49 | cusco | thought so, its not a dahdi issue |
| 18:58.55 | cusco | feedback from microphone? |
| 18:59.20 | JasonL | but there is no echo when the telco is connect direct to the PBX |
| 19:01.47 | [TK]D-Fender | JasonL, disable EC and let it pass through |
| 19:02.56 | JasonL | [TK]D-Fender: You helped me with this a couple weeks ago, and recommended that.. which I did and we're still experiencing echo |
| 19:03.13 | [TK]D-Fender | show us what you've got set up |
| 19:03.23 | JasonL | ok give me a sec |
| 19:07.27 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.164) |
| 19:07.45 | JasonL | http://pastebin.ca/2257577 |
| 19:09.22 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 19:09.33 | [TK]D-Fender | echocancel=no |
| 19:09.38 | [TK]D-Fender | echocancelwhenbridged=no |
| 19:09.43 | [TK]D-Fender | apply, restart * |
| 19:10.37 | JasonL | ok, can you tell me what that'll do? |
| 19:11.04 | JasonL | i didn't think the first echocancel=yes even did anything because i was defining below? |
| 19:11.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:14.58 | JasonL | [TK]D-Fender: should i be setting all echocancel instances to no in chan_dahdi.conf ? |
| 19:17.33 | [TK]D-Fender | chan_dahdi is what really controls it |
| 19:22.31 | *** join/#asterisk adeeln (~adeel@216.183.80.220) |
| 19:24.35 | adeeln | anyone happen to have a mysql concurrent call query via the cdr's they wouldn't mind sharing? |
| 19:27.19 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 19:39.21 | SeRi | p3nguin: pkg dropped at usps. |
| 19:39.43 | Qwell | How come he gets a package? I want a package. |
| 19:40.13 | SeRi | Qwell: Sure. I am not responssible for the content :P |
| 19:40.27 | SeRi | :) |
| 19:40.44 | SeRi | man the whole enterprise network got infected with some efup worm. |
| 19:41.23 | SeRi | from UK to US. is very bad. I feel bad for the windows guys... |
| 19:41.29 | SeRi | I helped as much as I could. |
| 19:48.51 | *** join/#asterisk wonderworld (~w@dsdf-4db5143b.pool.mediaWays.net) |
| 19:49.53 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:57.35 | qakhan | codec_g729.so is not loading in asterisk |
| 20:01.31 | SeRi | qakhan: you have a license for g729? |
| 20:03.35 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:05.05 | qakhan | no i m using free |
| 20:05.12 | qakhan | its free one |
| 20:05.59 | Qwell | Nobody will help you with that here. |
| 20:06.02 | Qwell | It is not legal to use. |
| 20:06.18 | SeRi | Qwell: +1 |
| 20:08.02 | qakhan | ohhh |
| 20:08.05 | qakhan | really |
| 20:08.20 | qakhan | i did not know that |
| 20:09.49 | dijib | * is now a certified dCAA |
| 20:10.38 | jmetro | * is now pondering how to make a railgun out of harddrive magnets |
| 20:11.22 | dijib | you will need to split the 4 quadrants of them first |
| 20:12.02 | SeRi | dijib: conf |
| 20:12.06 | dijib | hey k |
| 20:12.11 | dijib | see that |
| 20:12.24 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 20:12.54 | qakhan | @Qwell can u tell me where to by g729 licence |
| 20:13.02 | *** join/#asterisk plantseeker (~Plantseek@77.240.56.100) |
| 20:13.23 | Qwell | http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
| 20:15.34 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 20:16.05 | *** join/#asterisk Galen (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 20:16.10 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:16.10 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:17.29 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:21.03 | qakhan | @Qwell what i have understand it. each Asterisk is required 1 g729 licence |
| 20:21.05 | feeshon | Currently have an issue where someone joins a conference bridge (from an external #) gets a music on hold but doesn't not join the person in |
| 20:21.14 | qakhan | am i correct? |
| 20:21.40 | feeshon | The people in the room hear that someone joined but doesn't actually hear the person |
| 20:21.43 | Qwell | qakhan: No. |
| 20:23.21 | feeshon | Any ideas=? |
| 20:23.41 | qakhan | then |
| 20:25.08 | *** join/#asterisk plantseeker (~Plantseek@77.240.56.100) |
| 20:29.17 | SeRi | qakhan: You need a license per channel. |
| 20:33.16 | [TK]D-Fender | qakhan, If you have 10 calls that require G.729 translation, then that's 10 licenses. |
| 20:33.20 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 20:34.02 | qakhan | 10 calls per min? |
| 20:34.12 | kikohnl | concurrent calls |
| 20:34.38 | qakhan | its not per ext right? |
| 20:36.53 | JasonL | hehehe |
| 20:37.32 | [TK]D-Fender | qakhan, You know what a call is? a call that is IN PROGRESS? If 10 people are talking NOW, then that's 10 calls. |
| 20:38.18 | [TK]D-Fender | qakhan, So if you have 10 calls that each need to transcode then you need 10 licences to support them. If you run our, your channel attempts will all fail |
| 20:41.24 | qakhan | [TK]D-Fender thanks |
| 20:41.32 | qakhan | you always help me :) |
| 20:45.18 | qakhan | if i buy 5 licences for now and after then can i add more licences in it? |
| 20:45.29 | qakhan | in same asterisk |
| 20:45.46 | SeRi | yes |
| 20:49.07 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:50.24 | qakhan | ok |
| 20:52.02 | WIMPy | 4 |
| 20:52.30 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:03.08 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 21:03.08 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:10.41 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 21:14.32 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 21:19.18 | *** join/#asterisk Ice_Strike (~Ice_Black@94-192-112-241.zone6.bethere.co.uk) |
| 21:20.40 | Ice_Strike | Why I am getting "Permission denied" when I try to originate a call? |
| 21:20.46 | Ice_Strike | See http://pastebin.com/ekYijsyn |
| 21:24.36 | p3nguin | seri: I'll track it later. |
| 21:24.46 | SeRi | p3nguin: ok. |
| 21:24.52 | *** join/#asterisk festr_ (~festr@voipmonitor.org) |
| 21:24.59 | SeRi | ugh... feeling like crap today. |
| 21:25.06 | p3nguin | (1524.54) Irssi uptime: 190d 17h 22m 40s |
| 21:25.14 | SeRi | nice! |
| 21:25.18 | festr_ | hi. is it possible to read RURI (whole line in INVITE ...;params) in asterisk? it seems it is not possible which seems to be curious |
| 21:25.22 | SeRi | today is comcast install day |
| 21:25.24 | p3nguin | I'm having an irrsi issus, and I'm considering restarting it. |
| 21:25.29 | SeRi | :/ |
| 21:25.30 | p3nguin | irssi, even |
| 21:25.32 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 21:30.55 | gusto | i never had issues with irssi |
| 21:31.11 | gusto | as long as there are no scripts running on it it should be stable however what the uptime is |
| 21:31.11 | p3nguin | This all started when I was logged on from my phone and forgot to detach from screen before disabling 3G. Maybe it's screen and not irssi. |
| 21:31.25 | p3nguin | I've got dozens of scripts. |
| 21:31.42 | gusto | so it's either screen or the scripts |
| 21:31.59 | p3nguin | But irssi takes the heat, even if screen is at fault. |
| 21:32.03 | gusto | what kind of issues? |
| 21:32.14 | p3nguin | It kills irssi temporarily and drops the network connections. |
| 21:32.27 | p3nguin | It never actually destroys the irssi process. |
| 21:32.32 | gusto | aha |
| 21:32.37 | p3nguin | Just freezes it up for LONG times. |
| 21:32.42 | gusto | so that does not seem to be screens fault |
| 21:32.48 | gusto | because i know some issues with screen |
| 21:33.09 | p3nguin | It is a result of screen not being detached, but irssi still gets the problems from it. |
| 21:33.15 | p3nguin | Maybe it is the fault of both together. |
| 21:33.18 | gusto | but they are more a problem with representing |
| 21:33.42 | gusto | for example when you switch screens it does not clear the text there was before and so on |
| 21:33.50 | gusto | that kind of problems with screen |
| 21:34.20 | Ice_Strike | Why I am getting "Permission denied" when I try to originate a call? See http://pastebin.com/ekYijsyn |
| 21:35.58 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 21:37.12 | kaldemar | Ice_Strike: your manager user has no privileges to originate |
| 21:38.25 | *** join/#asterisk nir (~smuxi@192.117.240.253) |
| 21:39.11 | nir | Hello , can i get help to "AMI" Related issues ? |
| 21:39.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:39.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:40.10 | SeRi | ~ask |
| 21:40.10 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:40.19 | SeRi | cya guys. comcast is here |
| 21:40.28 | dijib | w00t |
| 21:41.43 | Ice_Strike | kaldemar There is.. |
| 21:41.53 | Ice_Strike | write = system,call,log,verbose,command,agent,user,originate |
| 21:43.30 | dijib | still able to ping seri. |
| 21:45.11 | dijib | seri is down. |
| 21:45.15 | dijib | repeat seri si down |
| 21:45.21 | dijib | seri is down |
| 21:49.33 | *** join/#asterisk carrar (tim@osburn.com) |
| 21:49.46 | Ice_Strike | What does that mean |
| 21:49.49 | Ice_Strike | <PROTECTED> |
| 21:50.01 | Ice_Strike | Sip user 6100 is connected |
| 21:50.02 | *** join/#asterisk vimreaper (~vimreaper@rrcs-70-62-43-252.central.biz.rr.com) |
| 21:50.46 | vimreaper | hey guys, is there a command that will allow me to get the ip of the sip trunk? |
| 21:51.35 | kaldemar | vimreaper: func CHANNEL |
| 21:51.47 | vimreaper | would it be in the asterisk -rx "show all peers" command |
| 21:52.08 | p3nguin | No such command 'show all peers' (type 'core show help show all' for other possible commands) |
| 21:52.11 | p3nguin | So, no. |
| 21:52.22 | kaldemar | sip show peers |
| 21:52.30 | vimreaper | ^ |
| 21:53.31 | Ice_Strike | p3nguin` What does that mean http://pastebin.com/in0qTQ2B |
| 21:53.32 | vimreaper | sorry im not well versed in asterisk.. i just have to write a automated script to setup iptables rules for asterisk to whitelist all peers and the trunk |
| 21:53.46 | p3nguin | ice_strike: Is it the same thing you've pasted at least two times before? |
| 21:53.47 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:53.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:54.00 | Ice_Strike | no |
| 21:54.07 | Ice_Strike | its different question |
| 21:54.34 | p3nguin | What's the question? |
| 21:54.36 | vimreaper | so kaldemar do you know if the trunk ip would be in "sip show peers" with the rest of the ips |
| 21:55.06 | Ice_Strike | I am getting channel.c:5394 __ast_request_and_dial: Unable to request channel SIP/6100 |
| 21:55.08 | p3nguin | vimreaper: You realize there is no such thing as a sip trunk, right? |
| 21:55.15 | Ice_Strike | when I try to orginate a call |
| 21:55.16 | p3nguin | vimreaper: Your ITSP is just another peer, the same as a phone. |
| 21:55.23 | vimreaper | ok cool |
| 21:55.35 | Ice_Strike | However user 6100 is connected on softphone |
| 21:56.13 | p3nguin | *shrug* |
| 21:56.15 | kaldemar | vimreaper: it will be there. it's up to you to know which one it is. |
| 21:56.30 | p3nguin | I didn't answer you the first time you asked, so what makes you think I'm going to answer now that you have singled me out? |
| 21:56.31 | vimreaper | thats alright, im whitelisting all of them |
| 21:57.01 | p3nguin | vimreaper: If you know the host name, it would be easy enough to do a DNS lookup. |
| 21:58.44 | vimreaper | p3nguin we have 60+ phone servers and I have to write a script to setup iptables for all of them.. I know how to find the ips manually but i'm coding a script for techs to deploy upon installation of new phone systems so it has to pull ips from asterisk commands |
| 21:59.20 | p3nguin | Interesting. |
| 21:59.52 | Qwell | You're using static IP addresses for phones? |
| 22:00.44 | vimreaper | nope we have another script that runs every 5 mins so when a new device is registered it whitelists the ip |
| 22:00.49 | dijib | SeRi is taking forever |
| 22:03.45 | Qwell | How is it going to register, if it's not whitelisted? |
| 22:04.00 | vimreaper | we live 5060-5062 open |
| 22:04.05 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:04.18 | Qwell | So then why are you whitelisting them? |
| 22:05.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:05.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:05.59 | vimreaper | this will allow them access to the web interface we have for checking emails and such |
| 22:06.22 | Qwell | So their desktop is going to share the same IP as the phone? |
| 22:07.29 | vimreaper | nope im confused ;) |
| 22:07.56 | vimreaper | basically isnt 5060-5062 used for registration but in order to make calls other ports need opened correct? |
| 22:08.01 | Qwell | ...no |
| 22:08.16 | vimreaper | everything is done through 5060-5062? |
| 22:08.18 | Qwell | No. |
| 22:08.54 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:08.55 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:09.06 | Qwell | Everything SIP is on whatever port (singular) you specify. |
| 22:09.08 | *** join/#asterisk thehar_ (~thehar@diddlebox.thehar.com) |
| 22:09.42 | vimreaper | what i was told is that with 5060-5062 open new devices can be registered but they wont be able to make phone calls till they are whitelisted |
| 22:09.47 | Qwell | You were told wrong. |
| 22:10.29 | vimreaper | i guess so.. i was just given a list of iptables rules and told to pull from the peers list and code a script |
| 22:10.46 | vimreaper | ill have to talk to the customer |
| 22:11.24 | vimreaper | so port 10000:20000 what goes on in that range? |
| 22:11.29 | Qwell | media |
| 22:12.53 | vimreaper | wow so all thing sip related is going to go on at 5060-5062 in our case therefore defeating the purpose of this script |
| 22:13.08 | Qwell | Again, no. But yes. Your script is pointless. |
| 22:15.29 | p3nguin | What do you do on ports 5061 and 5062? |
| 22:16.13 | Ice_Strike | I have been reading about AgentLogin() at http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentLogin |
| 22:16.22 | Ice_Strike | "The agent can dump the call by pressing the star key." |
| 22:16.29 | Ice_Strike | Which softphone allow me to do that? |
| 22:16.41 | Qwell | Which softphone allows you to dial numbers? |
| 22:16.56 | Ice_Strike | no |
| 22:17.14 | Ice_Strike | From what I understand when the call is ringing, i can press a star key right? |
| 22:17.52 | Ice_Strike | but does call have to be answered first? |
| 22:18.03 | Ice_Strike | It say "The agent can dump the call by pressing the star key." |
| 22:18.18 | Ice_Strike | but i cant press they key before answering the call |
| 22:18.23 | Ice_Strike | the* |
| 22:18.51 | Qwell | You aren't answering a call like normal. You're already on a call. |
| 22:19.14 | Ice_Strike | Yea |
| 22:19.16 | Ice_Strike | true |
| 22:20.29 | Ice_Strike | but from what i understand I can press a star key while its ringing? |
| 22:20.58 | Qwell | Correct. You're thinking about it wrong though. |
| 22:21.19 | *** join/#asterisk qakhan (~qakhan@pool-71-163-79-89.washdc.fios.verizon.net) |
| 22:21.54 | Ice_Strike | Problem with softphone zoiper i cant press any key while its ringing (incoming) |
| 22:22.16 | Qwell | You're thinking about it wrong. The phone doesn't ring. |
| 22:22.38 | Ice_Strike | oh |
| 22:23.10 | p3nguin | Agents are sitting on the line listening to music, waiting for a call to be thrown into their lap. |
| 22:24.41 | Ice_Strike | yea i can hear music when i answer |
| 22:24.53 | p3nguin | There is no "answer." |
| 22:25.05 | gusto | who does not answer? how does he/she dare? |
| 22:25.10 | p3nguin | You call the login, you log in, you wait. |
| 22:25.12 | p3nguin | That is all. |
| 22:25.27 | Ice_Strike | Yep that what I meant :) |
| 22:25.31 | gusto | ah, an asterisk script |
| 22:25.35 | p3nguin | nope |
| 22:25.43 | p3nguin | Just regular AgentLogin(). |
| 22:25.55 | p3nguin | When you call and log in, you get music. Right? |
| 22:27.21 | Ice_Strike | Yes thats right |
| 22:27.29 | p3nguin | That's it. Do not hang up. |
| 22:27.35 | p3nguin | Now you sit and wait. |
| 22:27.35 | Ice_Strike | I understand now :) |
| 22:27.57 | p3nguin | When a call goes to your Agent channel, the call will "appear" on your phone and you can talk. |
| 22:27.59 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:28.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:28.14 | p3nguin | When the caller is done, you do not hang up. |
| 22:28.27 | Ice_Strike | Yep I see |
| 22:28.35 | p3nguin | You will go back to music while you wait. |
| 22:28.41 | Ice_Strike | I understood it but what the purpose of AgentLogin |
| 22:28.49 | Ice_Strike | Why not just Dial and wait? |
| 22:28.53 | Ice_Strike | Dial() |
| 22:29.00 | p3nguin | It is to turn your SIP phone into an Agent channel. |
| 22:29.25 | p3nguin | If you login to AgentLogin() as agent 1234, then the queue needs to call you via Agent/1234. |
| 22:29.55 | p3nguin | Your phone will only be available via Agent/1234 while logged in. |
| 22:30.10 | p3nguin | This is NOT RELATED to SIP channels. |
| 22:31.00 | Ice_Strike | I understand now, need to play around with that :) |
| 22:31.06 | Ice_Strike | Thanks for your time bro |
| 22:31.33 | p3nguin | If you want people to take calls from the queue as regular phone calls where they have to answer a ringing phone, consider using the Local channel to call your phone the same as another person would call your phone. For example, If Jan calls your phone at extension 555 in context phones, assign the queue to member Local/555@phones |
| 22:32.38 | p3nguin | I do not recommand using SIP channels directly from the queue. Do not use SIP/black as the member. |
| 22:37.31 | Ice_Strike | I see what you mean |
| 22:37.58 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 22:38.17 | p3nguin | In my office, since we have pretty low call volume, there is no reason to have call agents sitting on their phones all day waiting for calls. |
| 22:38.51 | p3nguin | In a call center where all they do all day long is take calls, AgentLogin() is a good idea. |
| 22:40.04 | p3nguin | (or something that mimics the behavior of AgentLogin, anyway) |
| 22:40.08 | qakhan | can anyone tell which asterisk 10.x version is full working |
| 22:40.15 | p3nguin | Not everyone likes to use the agent apps. |
| 22:40.31 | p3nguin | qakhan: 10.10.0 |
| 22:40.48 | Qwell | qakhan: What's wrong with the one in the topic? |
| 22:42.10 | qakhan | @Qwell i didnt get you |
| 22:42.26 | Qwell | Use the latest version. |
| 22:42.28 | *** join/#asterisk myyrdin (~myyrdin@gateway/tor-sasl/myyrdin) |
| 22:42.39 | p3nguin | <p3nguin> qakhan: 10.10.0 <------------------------- |
| 22:42.47 | p3nguin | taps the microphone ... |
| 22:42.47 | p3nguin | is this thing on? |
| 22:42.57 | Qwell | p3nguin: You must be new here. |
| 22:43.24 | p3nguin | Thinking that people read the answers to their questions? Yeah, that was my bad. |
| 22:44.50 | *** join/#asterisk Dovid (ad3f69d2@gateway/web/freenode/ip.173.63.105.210) |
| 22:45.21 | Qwell | rookie mistake |
| 22:46.12 | *** join/#asterisk myyrdin (~myyrdin@gateway/tor-sasl/myyrdin) |
| 22:49.40 | Ice_Strike | p3nguin Thats very cool. If I want an agent to login on the website in order to active AgentLogin() I am thinking adding 3 fields. Username, Password and Softphone extention Number |
| 22:50.00 | Ice_Strike | Once logged in, the script will then execute AgentLogin() |
| 22:56.48 | dijib | just passed another cert at brainbench |
| 22:57.29 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:02.24 | Ice_Strike | p3nguin Can 2 agents login with same AgentNoAgentLogin() |
| 23:02.32 | Ice_Strike | same AgentNo to AgentLogin() |
| 23:06.18 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:11.37 | p3nguin | dijib: What do you mean by "another?" |
| 23:11.38 | *** join/#asterisk deo (~deo@112.198.79.156) |
| 23:12.21 | p3nguin | ice_strike: I wouldn't think it would work the way you want it to work. |
| 23:12.34 | p3nguin | It would be like two different phones registering to one SIP account. |
| 23:12.46 | p3nguin | Only the last one gets the calls. |
| 23:13.57 | Ice_Strike | Ahh thats right |
| 23:14.42 | apb1963 | So my user/extension in Bangladesh can't register his softphone. After much discussion here, I was told to go try openVPN. So, I now have openVPN installed and working. My question is... what do I have to do to get asterisk to use openVPN? |
| 23:15.11 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-043-011.pools.arcor-ip.net) |
| 23:15.22 | Ice_Strike | p3nguin I am thinking each softphone - the username(number) can be the same as AgentNo for AgentLogin() |
| 23:15.39 | Ice_Strike | Is that good method? |
| 23:15.43 | WIMPy | apb1963: Nothing. That's a networking thing. |
| 23:16.10 | ghost75 | got that during db_put over AMI: http://pastebin.com/hQqmAfLb <- this is something to worry about? |
| 23:16.14 | apb1963 | <looks for the #networking group> |
| 23:16.24 | apb1963 | err... channel |
| 23:16.45 | WIMPy | ghost75: Are you net reading the result? |
| 23:17.12 | ghost75 | net reading? |
| 23:17.48 | WIMPy | apb1963: Once you have it running it's pretty basic networking. IPs and routing. |
| 23:17.55 | ghost75 | everything is localhost |
| 23:18.28 | WIMPy | ghost75: You just should shove stuff in to AMI without listening. |
| 23:18.58 | ghost75 | wihtout running cli |
| 23:22.05 | WIMPy | What? |
| 23:22.56 | ghost75 | then what do you mean with "listening" |
| 23:23.45 | WIMPy | Listen for the replies. |
| 23:24.02 | WIMPy | And preferably logoff when done. |
| 23:26.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:31.20 | apb1963 | so you're saying there's nothing I need to do specific to asterisk to make this work? It's purely a routing issue? |
| 23:32.17 | apb1963 | Based on your comment I would think ok, I guess I know which ports asterisk uses... so I guess I can simply route all traffic from those ports to my tunnel |
| 23:33.04 | WIMPy | apb1963: Yes, but routig has nothing to do with ports (usually). |
| 23:33.27 | apb1963 | forwarding? |
| 23:33.33 | ghost75 | i think this logs off automatically from ami: http://search.cpan.org/~greenbean/Asterisk-AMI/lib/Asterisk/AMI/Common.pm |
| 23:34.32 | WIMPy | apb1963: Theer should be no need for forwarding unless you set up the tunnel in a way that it doesn't reach the intended end. |
| 23:34.34 | *** join/#asterisk ag4ve (~ag4ve@96.26.67.194) |
| 23:35.02 | WIMPy | ghost75: NFI |
| 23:35.15 | ghost75 | what? |
| 23:35.33 | WIMPy | No fabulous idea. |
| 23:35.35 | apb1963 | errrm |
| 23:35.38 | WIMPy | Or some other f-word :-) |
| 23:35.58 | apb1963 | so... it's not routing, and it's not forwarding. |
| 23:36.06 | ghost75 | well, at least it works |
| 23:36.12 | WIMPy | It is routing. |
| 23:37.12 | apb1963 | Yup but if it has nothing to do with the ports, how do I know which traffic is VOIP packets? |
| 23:37.38 | WIMPy | Why do you want to know? |
| 23:38.27 | apb1963 | errrmmm... so you're saying to send everything down the tunnel? |
| 23:38.52 | WIMPy | You route IPs via that tunnel. Use them. |
| 23:38.56 | apb1963 | <scratch scratch> but... what if the packets aren't for that particular tunnel? |
| 23:39.12 | apb1963 | I don't follow |
| 23:39.33 | WIMPy | That's about routing. |
| 23:40.04 | WIMPy | Usually only traffic for the remote IP should go via the tunnel. |
| 23:40.38 | WIMPy | Bog standard routing table. |
| 23:40.42 | apb1963 | I have a client <---> server VPN relationship... when the phone rings, if the call is destined for that extension, then I have to route those packets to that extension through that VPN tunnel. |
| 23:41.15 | apb1963 | If it's for another extension, it may or may not be routed through another tunnel (or no tunnel at all). |
| 23:41.21 | WIMPy | Forget about the tunnel. You just use the remote IP of that tunnel. |
| 23:41.33 | WIMPy | Everything else should do by itself. |
| 23:41.55 | apb1963 | That's easy for you to say |
| 23:42.07 | apb1963 | Me... I don't know what the heck you're talkin' about :) |
| 23:42.12 | WIMPy | Not only to say :-) |
| 23:42.36 | WIMPy | But if you have trouble with that, #networking might indeed be a better place. |
| 23:43.22 | apb1963 | And you maintain that I don't need to do anything in asterisk to accomplish this? |
| 23:44.14 | WIMPy | No, either you specofy the host or the phone has to register. Just as usual. |
| 23:44.31 | WIMPy | However the usual NAT issues apply here as well. |
| 23:45.02 | WIMPy | As the NAT issues are mainly just routing issues. |
| 23:45.50 | apb1963 | ok... normally the softphone specifies the server as host using it's public IP. A tunnel uses private IPs. So you're saying I should configure the softphone to use the private IP. |
| 23:46.07 | WIMPy | Exactely |
| 23:46.16 | apb1963 | So just change the config on the softphone |
| 23:47.07 | WIMPy | Yes. And then beware of that "NAT" issue. |
| 23:48.04 | WIMPy | But that depends on the behaviour of the client and the configuration on both sides. The joy of SDP. |
| 23:48.08 | apb1963 | NAT is the whole reason I've installed openVPN |
| 23:49.24 | WIMPy | The so called NAT isse comes up whenever (at least) one end uses more than one IP. |
| 23:49.59 | apb1963 | My user is behind a NAT |
| 23:50.17 | apb1963 | He can't register his softphone |
| 23:50.28 | apb1963 | openVPN was suggested as the solution |
| 23:50.55 | WIMPy | So there is a LAN IP, a public IP and now you've added a tunnel IP. |
| 23:51.09 | apb1963 | Pretty much |
| 23:51.33 | WIMPy | NAT doesn't keep ypu from registering. |
| 23:51.58 | apb1963 | something is. Assuming his ISP is blocking. |
| 23:52.07 | WIMPy | But if is a firewall thing, a tunnel will help. |
| 23:52.41 | apb1963 | At the moment I'm just trying to get it working on MY machines... where I have full time access. |
| 23:53.13 | apb1963 | so I tried changing the softphone to use the private IP... no joy. |
| 23:53.28 | apb1963 | Perhaps I'm using the wrong private IP? I want the server's private IP, yes? |
| 23:53.48 | WIMPy | sure |
| 23:54.37 | apb1963 | oops.. .dyslexia |
| 23:54.45 | apb1963 | It's back on hook :) |
| 23:55.15 | apb1963 | See Wimpy? You ARE helpful :) |
| 23:55.40 | apb1963 | Now why am I getting echo/feedback? |
| 23:56.03 | qakhan | i am installing Asterisk 11 on centOS 6 |
| 23:56.10 | WIMPy | Softphone? |
| 23:56.14 | qakhan | getting this message |
| 23:56.16 | qakhan | http://pastebin.com/bgwsweAy |
| 23:57.45 | WIMPy | Looks like something couldn't cope with your system. |
| 23:58.04 | WIMPy | But it does contain a suggestion. |
| 23:58.18 | qakhan | what suggestion? |
| 23:58.55 | WIMPy | Use PIC |
| 23:59.04 | *** join/#asterisk cyborg-one (~cyborg-on@79-140-5-100.broadband.tenet.odessa.ua) |
| 23:59.10 | qakhan | what is this? |
| 23:59.19 | WIMPy | configure --with-pic or someting. See --help |
| 23:59.35 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |