| 00:02.25 | MajesticFudgie | On another note, |
| 00:02.42 | MajesticFudgie | I'm starting to wonder if its the power supply being dodgy |
| 00:02.57 | MajesticFudgie | I've measured it and its outputting just under 7 volts |
| 00:03.01 | MajesticFudgie | When it shouldnt be |
| 00:03.09 | MajesticFudgie | It's set to 5v |
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| 00:23.02 | syadnom | anyone having issues with ilbc on asterisk 13.7? |
| 00:23.12 | syadnom | I'm getting one-way audio. |
| 00:23.21 | syadnom | just on ilbc |
| 00:28.24 | syadnom | getting this whacky message |
| 00:28.26 | syadnom | codec_ilbc.c:118 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? |
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| 00:56.19 | scv | whats on the other end of the channel syadnom |
| 00:57.30 | syadnom | scv, Ive tried IVRs and other in-asterisk things, A yealink phone from a polycome and reverse, zoiper to both. The only things that works is when ilbc is passed through so polycom to polycom works, and yealink to yealink works. |
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| 01:01.18 | scv | whats your frame size? |
| 01:01.31 | syadnom | scv, no idea |
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| 01:22.19 | syadnom | scv, well, it looks like ilbc is expecting the 13.33kbps 50byte but the phones are sending the 15.2kbps version |
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| 01:23.47 | scv | so reconfigure your phone(s) |
| 01:23.55 | scv | or switch to that type of framing |
| 01:23.58 | scv | in asterisk if possible |
| 01:23.59 | scv | not sure |
| 01:23.59 | syadnom | I see no option to set this in the phone... |
| 01:24.16 | syadnom | and I cant figure out how to configure ilbc to use the different frame size :/ |
| 01:24.22 | syadnom | and google hates me today |
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| 01:36.32 | syadnom | yeah, so the yealink is sending 38 byte ilbc frames but I cant figure out how to get asterisk to accept those. |
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| 01:55.25 | warewolf | syadnom: oh, I have one of those phones |
| 01:55.35 | syadnom | warewolf, yealink? |
| 01:55.35 | warewolf | syadnom: go into the phone web interafce and change from 13 to 15, and it works |
| 01:55.38 | warewolf | yes |
| 01:55.41 | warewolf | SIP-T22P |
| 01:55.41 | syadnom | where? |
| 01:55.59 | warewolf | the phones have a web interface, sec |
| 01:56.41 | warewolf | browse to the IP of the phone, login w/ default creds, admin:admin, then clic the 'account' tab |
| 01:56.48 | warewolf | on the left, click 'codec' |
| 01:57.15 | warewolf | er wait, I have my phone set to iBLC_13_3 as the first codec, then iBLC. |
| 01:57.24 | warewolf | that setup makes Asterisk happy. |
| 01:57.58 | syadnom | warewolf, what asterisk version? |
| 01:57.59 | warewolf | anyway, hope that helps/works for you, I'm off to bed. Good luck! |
| 01:58.12 | warewolf | syn13.7.1 |
| 01:58.36 | syadnom | warewolf, any chance you can send me the codec parts of your config file first?? |
| 01:58.51 | warewolf | I don't have a config file, I set it through the web UI |
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| 01:59.14 | syadnom | how did you create the ilbc_13_3? |
| 01:59.20 | syadnom | that's not there by default... |
| 01:59.22 | warewolf | syadnom: in the phone or in asterisk? |
| 01:59.28 | syadnom | in the phone |
| 01:59.33 | warewolf | which phone od you have |
| 01:59.38 | syadnom | T48G |
| 01:59.48 | warewolf | that should be similar enough |
| 02:00.10 | syadnom | which model is yours |
| 02:00.11 | syadnom | ? |
| 02:00.16 | warewolf | SIp-T22P |
| 02:00.31 | warewolf | yours has the same guts as mine, just slightly better external features |
| 02:01.37 | warewolf | anyway, poke aroudn the web interface. |
| 02:02.01 | warewolf | the config backup from the phone isn't like the config file you put on a webserver or tftp server for the phone to provision itself |
| 02:02.05 | warewolf | I forgot that. |
| 02:02.13 | warewolf | I really do have to head to bed though. |
| 02:02.29 | warewolf | if you still have problems friday, send me a /msg? I'm in training at work all day tomorrow. |
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| 02:03.07 | syadnom | ok, thanks warewolf , goodnight |
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| 09:52.50 | Haris | where is the no answer with short time setting in fpbx or asterisk ? |
| 09:52.52 | Haris | hello all |
| 09:54.11 | Rasputin3711 | fpbx dialplan |
| 10:06.00 | Haris | is this global or specific to an extention |
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| 10:47.17 | Haris | I'm getting this msg alot ---> [2016-Jun-09 15:45:50] [CRITICAL] (admin/bootstrap.php:258) - Connection attmempt to AMI failed <--- how do I make AMI to be up ? Is there a way to check or troubleshoot it going/being down ? |
| 10:52.07 | Greenlight | manager.conf |
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| 11:18.55 | Haris | does it require reboot to fix connection with ami ? Is there a way to bring AMI back without reboot ? |
| 11:21.36 | Haris | ok, in manager.conf I have deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 |
| 11:22.48 | Haris | how do I restore connectivity to AMI ? |
| 11:22.58 | Haris | freepbx has the correct u/p for AMI |
| 11:23.23 | Haris | still I'm getting msgs ---> [2016-Jun-09 16:23:01] [CRITICAL] (admin/bootstrap.php:258) - Connection attmempt to AMI failed |
| 11:32.24 | p1ke | Hello! Does anyone know a good guide/blog/whatever for VoIP sysctl tuning? |
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| 11:56.23 | Haris | in what shape or form does ami run ? |
| 11:56.44 | Haris | freepbx says it cant connect to it. I'v rebooted this box 3 times. but in vain. I still am unable to connect freepbx with AMI |
| 11:56.54 | Haris | how can I make sure AMI is running or accessible ? |
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| 12:16.22 | proute | Hello |
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| 12:19.09 | proute | I use Asterisk 1.8.32. This Asterisk is on a datacenter (centrex mode IP address 172.16.5.30). I have phone in a customer company (IP address 192.168.170.0/24) . I try to do directmedia=yes on peers to redirect RTP flow between phone, but it doesn't work. There is no nat between phones and Asterisk. Someone has any ideas about this problem in order to enable directmedia? Thanks for your |
| 12:19.09 | proute | help.... :) |
| 12:31.04 | Haris | when I make call, it goes to receiving end. but when I click to pick the call, it gets dropped and the caller gets msg from asterisk pbx to give a msg for VM |
| 12:31.18 | Haris | why is the destination not able to pick call ? |
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| 12:35.58 | [TK]D-Fender | Haris, We don't know because you haven't shown us the call. |
| 12:37.28 | Haris | hold please |
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| 13:11.16 | proute | So, to answer to my question, the problem come from mixmonitor function that was activated on peers.... |
| 13:15.15 | [TK]D-Fender | Not going to reinvite if * has to stay in the path.... |
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| 14:30.14 | _AxS_ | hey all .. so i'm looking into what's involved in adding g729 support to my asterisk install. I've seen the http://asterisk.hosting.lv stuff, and i've seen the page about software g729 on digium's site. Are they the same? do all asterisk software g729 modules use/need a license from digium to be used? |
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| 14:38.40 | [TK]D-Fender | G.729 is a patented & licensed codec. If you are using it anywhere where the patent holds and are doing so without a license then you are legally liable and can be sued, etc |
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| 14:50.11 | _AxS_ | Yep i follow that, and I don't plan to use it without a license. But, is digium the de-facto clearing house for the license for any asterisk-based implementation? or.. |
| 14:50.32 | drmessano | Well |
| 14:51.32 | drmessano | I would consider Digiums implementation the one most likely to work with Asterisk, and it's supported |
| 14:52.39 | [TK]D-Fender | Yes, it is the only standard legimate option. |
| 14:53.04 | [TK]D-Fender | Because they pay off the patent-owner properly. |
| 14:53.19 | drmessano | "Pay off" is a great way to phrase it |
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| 14:57.14 | NirS | morning everybody |
| 14:57.16 | NirS | anybody home ? |
| 14:57.35 | [TK]D-Fender | No |
| 14:57.44 | drmessano | Im not at home |
| 14:57.54 | [TK]D-Fender | Roll calls are are just another poll |
| 14:57.55 | [TK]D-Fender | ~poll |
| 14:58.00 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 14:58.02 | [TK]D-Fender | ~polls |
| 14:58.02 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 14:58.12 | [TK]D-Fender | RABID WEASELS |
| 14:58.19 | [TK]D-Fender | Not just regular weasels |
| 14:58.35 | NirS | hey all, anyone has experience with SRTP ? |
| 14:58.42 | [TK]D-Fender | POLL ^ |
| 14:58.47 | drmessano | ROFL |
| 14:58.49 | drmessano | Another poll |
| 14:59.14 | drmessano | is not feeling the Bern |
| 14:59.32 | [TK]D-Fender | "I'm getting error XYZ trying to do 123, here's the DEBUG AND CONFIGS" <- how to get places. |
| 15:00.00 | NirS | Not a poll, got a very interesting issue |
| 15:00.06 | drmessano | NirS: Sorry, English is not my first language.. Do you have an actual issue, or are you just taking a poll? |
| 15:00.18 | NirS | here's the issue |
| 15:00.28 | [TK]D-Fender | drmessano, I'm not sure sarcasm is a language... more like an accent or dialect |
| 15:00.28 | drmessano | #suspense |
| 15:00.37 | NirS | I have a client (PJSUA) running a call into Asterisk using TLS |
| 15:00.48 | NirS | When I do TLS+RTP, no problem, all works just fine |
| 15:00.57 | NirS | when I do TLS+SRTP, the call is rejected with a 488 cause |
| 15:01.05 | NirS | 'core set debug 4' shows the following: |
| 15:01.26 | drmessano | I thought this would be a hard one |
| 15:01.40 | NirS | [2016-06-09 10:50:55] DEBUG[967][C-0000000e]: sdp_srtp.c:384 ast_sdp_crypto_build_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:baNm0bvzsRut7jnzuD9DJjn/ml3y8DNrnG4Om34n |
| 15:01.40 | NirS | [2016-06-09 10:50:55] DEBUG[967][C-0000000e]: chan_sip.c:10503 process_sdp: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uX8d0OXRdHp1c0UiFldS1yEgya6aNzVV8yQnI6sB... OK. |
| 15:01.40 | NirS | [2016-06-09 10:50:55] DEBUG[967][C-0000000e]: chan_sip.c:10503 process_sdp: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YZbQEzFlOS6tpAwT/2sdf1NjLamlw+9e6gxntVXa... UNSUPPORTED OR FAILED. |
| 15:01.40 | NirS | <--- Reliably Transmitting (NAT) to 5.29.6.132:55015 ---> |
| 15:01.41 | NirS | SIP/2.0 488 Not acceptable here |
| 15:01.55 | drmessano | ~pb |
| 15:01.55 | 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. |
| 15:01.55 | NirS | btw, codecs match and work just fine with RTP |
| 15:02.03 | NirS | right sorry |
| 15:02.07 | drmessano | TLS+RTP = Just a normal call.. the TLS part is not impressive.. so you're saying NORMAL calls work |
| 15:02.08 | NirS | thought this would be faster |
| 15:02.12 | drmessano | But SRTP doesn't |
| 15:02.20 | [TK]D-Fender | PASTEBIN <- |
| 15:02.21 | NirS | yup |
| 15:02.30 | drmessano | So Pastebin |
| 15:02.54 | drmessano | Sounds like you don't have SRTP baked in or configured or both |
| 15:03.12 | drmessano | Because TLS working is basically a useless variable |
| 15:03.13 | NirS | actually, the same server also runs WebRTC no problem |
| 15:03.32 | NirS | so, SIP over WebSocket requires dtls-srtp to work |
| 15:03.41 | NirS | so, SRTP is in there for sure |
| 15:04.09 | drmessano | Pastebin the SIP debug on the device config |
| 15:04.14 | drmessano | Pastebin the SIP debug AND the device config |
| 15:04.20 | NirS | no problem |
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| 15:07.29 | NirS | http://pastebin.com/f728sMrb |
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| 15:22.06 | drmessano | NirS: What is the client? |
| 15:24.28 | file | we sorted it out in #asterisk-dev |
| 15:24.36 | file | the offer is for optional SRTP, which is not supported by chan_sip |
| 15:24.37 | drmessano | ok |
| 15:27.39 | NirS | The client is our own mobile application, using our own SDK wrapper around PJSIP |
| 15:35.46 | jrun | if i have a variable set in a template would re-defining its value in the *child* section override it? |
| 15:36.24 | jrun | say i have 'context = Longdist' in template and i'm resetting context in a seciton somewhere... |
| 15:37.41 | jrun | nevermind, just found the answer on the wiki. sorry for the noise. |
| 15:42.28 | [TK]D-Fender | It should |
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| 16:02.18 | i0x71 | Hey, i noticed interesting behaviour with blind transfer and incoming calls that use Answer() app, i have an incoming did number setup with a Goto statement to execute a feature which starts with an Answer(). Basically when you dial the number from outside, asterisk picks up and waits for dtmf input. There is no Dial application involved anywhere in the config. However when i call in and press the blind tr |
| 16:02.24 | i0x71 | ansfer key i get a "Transfer prompt" my guess is that somehow the Answer() application sets the transfer flag to allowed. Anyone have any clues on this ? |
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| 16:04.48 | lvlinux | i0x71: by "blind transfer key" do you mean the DTMF feature code for it, or the key on your phone? |
| 16:04.54 | i0x71 | errr, disregard this, the ## was being picked up from the Dial statement on test sip client... *face palm* |
| 16:05.19 | i0x71 | lvlinux: i meant the dtmf feature defined in features.conf |
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| 16:46.32 | jrun | does endpoint now encompasses friend/peer ? |
| 16:46.44 | jrun | s/now/in pjsip/ |
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| 16:47.24 | jrun | infobot: thanks |
| 16:47.24 | infobot | jrun: de nada |
| 16:52.56 | [TK]D-Fender | It always did |
| 16:53.01 | lvlinux | jrun yes, endpoint is everything |
| 16:54.27 | jrun | [TK]D-Fender: do we mean that all peers are endpoints but now all endpoints are peers? (since they could be friend?) i mean does the disinction matter? |
| 16:54.32 | file | I like to describe an endpoint as a set of configuration for talking to something |
| 16:55.22 | jrun | file: and when would you use peer or friend? |
| 16:55.28 | jrun | file: or would you at all? |
| 16:55.36 | file | chan_pjsip has no concept of a peer or friend |
| 16:55.38 | lvlinux | jrun: there is no "friend" or "peer" with PJSIP |
| 16:55.41 | lvlinux | just endpoints. |
| 16:56.01 | lvlinux | the equivalent of a chan_sip "peer" would be a PJSIP endpoint that matches on IP address |
| 16:56.11 | jrun | uh, i thought [TK]D-Fender meant that endpoint has always been there. |
| 16:56.36 | file | no, he meant that within chan_pjsip an endpoint is everything |
| 16:56.38 | jrun | lvlinux: and a fiend? |
| 16:56.49 | jrun | file: i see. |
| 16:56.59 | lvlinux | chan_sip "friend" in PJSIP = endpoint that matches on user/pass |
| 16:57.04 | jrun | lvlinux: would you use registration for that? |
| 16:57.15 | lvlinux | you can |
| 17:00.36 | lvlinux | well let me correct that, IIRC a "friend" matches on IP+user/pass in chan_sip. A "user" is just user/pass I think. |
| 17:01.20 | lvlinux | But anyway in PJSIP all of those are endpoints, just with different parameters specified. |
| 17:03.58 | lvlinux | jrun: PJSIP is great once you get past the initial hurdle of the differences. To me, the config is simpler than sip.conf, even though it has more lines sometimes. And it's much more powerful. |
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| 17:06.11 | lvlinux | I just recently forced myself to learn to use it, and I will never go back to chan_sip! |
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| 17:28.05 | lvlinux | How can I (in dialplan) get the endpoint that is making a call? |
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| 17:30.44 | lvlinux | And also, how can I get the endpoint that is being called? |
| 17:31.32 | lvlinux | ${BRIDGEPEER} ?? Or will that only be active after the call is connected? |
| 17:32.10 | _AxS_ | define endpoint in this case -- the sip channel? |
| 17:32.31 | lvlinux | no the peer/phone being called and calling |
| 17:33.09 | lvlinux | What I'm trying to do is setup call recording for all calls coming to or from a few certain phones. |
| 17:33.42 | [TK]D-Fender | You should know before you dial. |
| 17:33.58 | lvlinux | Yes |
| 17:34.32 | lvlinux | I thought about putting the special phones in their own context, but then I'd have to give them all duplicate dial pattern strings. |
| 17:35.03 | _AxS_ | lvlinux: if your dial patterns are abstracted out as a gosub then it wouldn't be that big of a deal...? |
| 17:35.42 | lvlinux | They are, but there are a LOT of them. (and they are macros instead of gosubs---i know i know...) |
| 17:36.40 | lvlinux | in other words i have multiple Macros for individual providers, special caller ID stuff, etc. |
| 17:36.44 | _AxS_ | still, this recording-call context could just jump to the regular one after setting an environment variable, no? |
| 17:37.18 | lvlinux | That could work I guess... |
| 17:37.34 | lvlinux | but that would only apply to calls FROM the phones. What about calls TO them? |
| 17:38.04 | _AxS_ | ..for the caller case at least. for the callee case, not sure yet; it depends on how complicated the dialplans are that trigger the dial to the callee |
| 17:38.26 | _AxS_ | local-channels maybe? |
| 17:38.34 | lvlinux | ugh |
| 17:38.37 | lvlinux | lol |
| 17:39.04 | _AxS_ | note i'm only a week into toying with asterisk, so i'm by no means an expert in here :) |
| 17:39.37 | lvlinux | I know there are half a dozen ways to do this, I'm just trying to find the simplest and least intrusive to the existing dialplan. |
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| 17:43.37 | jennie | hello guys |
| 17:43.41 | jennie | how you doing today? |
| 17:43.41 | lvlinux | hello! |
| 17:43.57 | lvlinux | fine and dandy |
| 17:44.11 | jennie | :D |
| 17:44.59 | jennie | I am saving call recording and was wondering if Asterisk can Asterisk can write to SQL Server instead of MySQL databases |
| 17:45.10 | jennie | call recordings* |
| 17:45.21 | jennie | Please tell me that if its doable |
| 17:45.40 | lvlinux | I believe so via ODBC |
| 17:46.00 | jennie | can you please give me a little more detail, I appreciate your help |
| 17:47.06 | jennie | also is it asterisk side workaround or mysql workaround? |
| 17:47.16 | lvlinux | well, the asterisk mysql driver has been deprecated in favor of ODBC, which can interface with all kinds of databases, including I believe SQL. |
| 17:47.56 | lvlinux | So Asterisk just works with ODBC, and then you get an ODBC driver for SQL. |
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| 17:48.19 | lvlinux | ODBC is a DB abstraction layer, so Asterisk doesn't have to speak the particular database you want to use. |
| 17:48.58 | _AxS_ | ...though sometimes the SQL still does.. hopefully asterisk's use of sql is generic enough that it'll work with anything tho |
| 17:49.46 | lvlinux | It probably is for the most part. |
| 17:50.36 | jennie | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html is what I should do? or any updated link you are aware of? |
| 17:51.13 | lvlinux | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_odbc |
| 17:51.28 | Kobaz | [2016-06-09 13:51:02.868] WARNING[6877]: chan_sip.c:3762 __sip_xmit: sip_xmit of 0x15bec40 (len 616) to 192.168.175.130:37045 returned -2: No such file or directory |
| 17:51.30 | Kobaz | what's that about? |
| 17:51.39 | Kobaz | i can reach 192.168.175.130 no problem |
| 17:51.48 | Kobaz | i can run pings on it all day without a single drop |
| 17:52.00 | Kobaz | (it's a vpn subnet) |
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| 17:53.03 | lvlinux | jennie: Also this may help too (but is for MySQL, but you can see how the ODBC part works) https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC |
| 17:54.32 | jennie | last question, can you please tell me Pros and cons if I go with this ^, is it worth doing it? I asked quetsion because I am having issues running queries in database so I am thinking to do this |
| 17:54.36 | _AxS_ | jennie: the 4th edition might have a newer/better description, since its for asterisk 11 instead of 8 , dunno if its changed much tho |
| 17:54.58 | _AxS_ | jennie: ... in database = what database? |
| 17:55.08 | jennie | MySQL databases |
| 17:55.50 | _AxS_ | Ah, so you're switching from mysql to sql-server .. well, that may depend on what the problem is you're facing with the queries.. |
| 17:56.09 | jennie | wait I show you screenshot |
| 17:56.24 | [TK]D-Fender | If you're having trouble running queries.... then you should reconsider using DB's |
| 17:57.02 | [TK]D-Fender | And last I checked, * does not Record/Monitor to db at all. |
| 17:57.24 | [TK]D-Fender | Voicemail, CDR, configs, etc it can |
| 17:57.43 | lvlinux | Yes I'm sure she meant records of the recordings through a DB function or something. |
| 17:57.45 | [TK]D-Fender | If you want recordings in there you'll have to run your own scripts to do the adds |
| 17:58.08 | [TK]D-Fender | Which would be a binary blob or some such element |
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| 17:58.37 | jennie | http://i.imgur.com/LakI6ZQ.png |
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| 17:58.58 | jennie | Please have a look at screenshot |
| 17:59.56 | mgriffin | i'll help jennie with this screenshot on #mysql |
| 18:00.14 | mgriffin | while screenshots are the devil, there are several obvious problems visible |
| 18:00.15 | lvlinux | jennie: it looks like you've got some problems with your DB, not Asterisk |
| 18:00.34 | jennie | yes thats why I want to swtich to writing directly to sql server |
| 18:01.43 | lvlinux | jennie: Either way I'd use ODBC with Asterisk. |
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| 18:02.40 | jennie | Understood |
| 18:04.09 | _AxS_ | jennie: do you have more experience with sql server than mysql? if so, it will likely be a better for for you and could be worth the switch. |
| 18:04.42 | lvlinux | ...otherwise it might be a mess :) |
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| 18:05.00 | _AxS_ | or at least not any better than using mysql already |
| 18:05.02 | jennie | mess can be messy sometimes yeah |
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| 18:12.21 | lvlinux | Aha! CHANNEL function tells me the endpoint when using PJSIP. Yay |
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| 18:45.12 | darkdev | Hello gyus! Please help, I'm really stuck. I have script for AGI in php, and variable $result has no value, but if I run same script from bash, it gives me value. First "initial" script - http://pastebin.com/CBVM4NNf |
| 18:47.00 | darkdev | Sphinx.php - http://pastebin.com/zcXag8eF |
| 18:47.36 | darkdev | Why can be that trouble? I have another class Yandex, and it is working well |
| 18:48.58 | darkdev | This class working well from both AGI and bash - http://pastebin.com/iQ3zkz42 |
| 19:00.52 | darkdev | Hello? Can anyone help me? |
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| 19:14.08 | jennie | Thank you lvlinux _AxS_ appreciate it :D |
| 19:14.21 | _AxS_ | np, good luck! |
| 19:16.37 | lvlinux | :) |
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| 19:54.47 | lvlinux | how can I check an expression for a number range. I want a check to be true if a variable contains for example 501-506. |
| 19:54.52 | lvlinux | ? |
| 19:55.04 | Kobaz | check an expression? |
| 19:55.08 | Kobaz | mathmatica? |
| 19:55.10 | Kobaz | matlab? |
| 19:55.22 | lvlinux | I mean within asterisk dialplan lol |
| 19:55.39 | lvlinux | within a GotoIf specifically. |
| 19:55.42 | Kobaz | load matlab as a plugin? |
| 19:55.55 | lvlinux | oh yeah thanks! lol |
| 19:55.57 | Kobaz | can you explain further what you mean by 'check an expression' |
| 19:56.23 | lvlinux | sorry i guess I meant "evaluate a variable" |
| 19:56.36 | lvlinux | not an expression |
| 19:58.00 | lvlinux | GotoIf($["${MYVAR}" = "501"]... |
| 19:58.14 | lvlinux | but instead of 501 I want a range of numbers. |
| 19:59.53 | lvlinux | without doing ${MYVAR}" = "501" | ${MYVAR}" = "502" | ${MYVAR}" = "503"... |
| 20:00.33 | Kobaz | use > and < |
| 20:01.04 | Kobaz | look up boolean logic |
| 20:01.37 | Kobaz | https://wiki.asterisk.org/wiki/display/AST/Operators |
| 20:01.40 | lvlinux | Well what if I have a few in the range I want excluded? |
| 20:02.02 | Kobaz | same, it's all boolean operators |
| 20:02.04 | Kobaz | use != |
| 20:02.33 | lvlinux | so I still have to have another expression for that though. No way to put a list I guess? |
| 20:03.24 | Kobaz | you can use a hash |
| 20:03.38 | lvlinux | hash? |
| 20:03.51 | Kobaz | hashtable |
| 20:03.56 | Kobaz | a lookup table |
| 20:04.13 | Kobaz | store all the numbers you want to work with in a hash so you can do direct access lookups |
| 20:04.21 | Kobaz | and if it's in the hash, it's excluded |
| 20:04.34 | [TK]D-Fender | "core show function REGEX" <- |
| 20:04.41 | Kobaz | that too |
| 20:04.49 | Kobaz | but regex doesn't do well with numeric ranges |
| 20:04.56 | Kobaz | ...depends |
| 20:04.58 | lvlinux | ah, REGEX yes that's what I needed. didn't know about that one. |
| 20:05.14 | Kobaz | simple stuff, or just keep making patterns for each specific case |
| 20:05.37 | Kobaz | sure you can do like 1[0-9] or whatever |
| 20:05.57 | lvlinux | Really there are only 5 or 6 numbers I'm needing to match, and I had it working fine, just wondering about the ways to compress my dialplan code a bit (and learn for future reference). |
| 20:05.58 | Kobaz | but if you wanted a range of 1000 to 9999 and then exclude 123 456 789 345 1124 etc etc etc |
| 20:06.19 | Kobaz | you'll wind up with a silly-complex regex where you could have used a table to lookup |
| 20:06.49 | lvlinux | where can I read about using a hash table in *? |
| 20:07.07 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+func+hash |
| 20:07.19 | lvlinux | yeah just saw it. |
| 20:08.05 | lvlinux | thanks for all the tips |
| 20:08.12 | Kobaz | np, good luck |
| 20:41.49 | lvlinux | If I'm reading a variable with inheritance set with __ do I refer to the variable with the __ or just the variable name? In other words, is the __ prefix actually part of the variable, or is it just something that tells asterisk to make the variable have inheritance? |
| 20:44.12 | lvlinux | The wiki isn't totally clear on the distinction and doesn't give an example for reading. |
| 20:47.29 | rmudgett | You refer to the variable without the underbars. |
| 20:48.01 | lvlinux | k thanks. I was in the process of testing it :) |
| 20:51.01 | jennie | see ya later boys take care and stay safe |
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