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| 00:06.57 | PHunter | WIMPy: Thank you a million times! |
| 00:07.36 | litn | hi |
| 00:07.43 | litn | oh wrong channel |
| 00:08.08 | WIMPy | "hi" is wrong? |
| 00:08.43 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.90.144) |
| 00:08.56 | ruben23 | hi guys |
| 00:09.39 | ruben23 | any iidea dialing a number that even calkl is already engage- no one is answering but the calls is actually disconnected already..no sound or audio.. |
| 00:10.44 | WIMPy | Can you try that again as a sentence? |
| 00:12.21 | ChkDigit | lol |
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| 00:13.30 | PHunter | Thanks again WIMPy! time go soak my brain in distilled water. |
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| 00:21.35 | litn | I changed the music on hold for a queue in my config, but after a reload and a restart, it's still trying to play the old music |
| 00:24.08 | litn | it's definitely not anywhere in my configs anymore and reloads aren't changing it. any idea? |
| 00:24.16 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 00:24.48 | WIMPy | MOH doesn't change on reload. You have to 'module reload res_musiconhold'. |
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| 05:02.13 | *** join/#asterisk child (b37efeca@gateway/web/freenode/ip.179.126.254.202) |
| 05:04.07 | child | is it possible to link two different calls in a same call ? |
| 05:04.30 | [TK]D-Fender | child: Yes. "AMI Bridge" |
| 05:09.33 | child | thx |
| 05:11.03 | ChkDigit | I have an extension: GotoIf($[${MYCONN}=PAGE]?dopageme) |
| 05:11.21 | ChkDigit | Except for most users ${MYCONN} is something like SIP/123 |
| 05:11.52 | ChkDigit | Which $[ ] tries to divide... Any way around that? |
| 05:14.42 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 05:14.51 | [TK]D-Fender | Tried putting it in quotes? |
| 05:16.03 | ChkDigit | $["${MYCONN}"="PAGE"]?dopageme) like that? |
| 05:17.13 | ChkDigit | Thanks, worked! |
| 05:17.46 | [TK]D-Fender | You're welcome |
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| 05:56.52 | jeev | so i get an HP DL360 G5, basic machine.. put the pri card in it and it locks up... a few times, CPU errors. |
| 05:57.03 | jeev | i take out the card, it's sitting idle, works fine. i've seen when googling that error, some pri complaitns |
| 05:57.05 | jeev | any ideas? |
| 05:57.09 | jeev | using the same card |
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| 06:40.17 | uosiu | Hi all, stupid but deadly urgent question. We have a fuckup in asterisk, none of our SIP phones can register in server so we want to forward all incoming calls to backup cell phone |
| 06:40.31 | uosiu | How to forward them? |
| 06:41.27 | uosiu | same => n,call(sip/upstream/123456) ? |
| 06:41.57 | MaliutaLap | put all incoming calls into an extension that matches all patterns and does a DIAL() |
| 06:42.11 | MaliutaLap | problem solvered |
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| 07:03.25 | uosiu | MaliutaLap: thanks |
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| 07:25.41 | yottanami | Here is my extension.conf http://dpaste.com/2CKN9WS/ and here my sip.conf http://dpaste.com/1SZZFG4/ [0] is a regular phone on phone port of my SIP gateway device, |
| 07:25.42 | yottanami | I can call from [0] to [100] (100 is a softphone) But when I am trying to call from [100] to [0] I have this error http://dpaste.com/02N9SHY/ |
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| 07:43.13 | ruben23 | hi guys im using asterisk11and when i call an extension im having this issue---->http://pastebin.com/wsdbXU3u <----call hangups |
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| 08:05.59 | yottanami | snmp: ping |
| 08:06.48 | yottanami | snmp: 0 can call all extensions but no extentions can call 0 |
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| 08:16.24 | *** join/#asterisk heliar (~heliar@unaffiliated/heliar) |
| 08:16.44 | heliar | Hi all, |
| 08:16.53 | heliar | I've got old asterisk instance |
| 08:17.02 | heliar | and im wondering |
| 08:17.09 | heliar | is there a lot of work |
| 08:17.15 | heliar | to change that instance |
| 08:17.23 | heliar | to new one machine |
| 08:17.46 | heliar | with freepbx on board? |
| 08:18.30 | heliar | Have You guys any article to help me with this. |
| 08:18.43 | heliar | I don't looking for a manual |
| 08:18.57 | heliar | to install anythink from previous i said |
| 08:20.44 | heliar | but to replace configuration |
| 08:20.51 | heliar | from text files (old instance) |
| 08:20.58 | heliar | to web gui (freepbx). |
| 08:21.09 | heliar | Im little confuzed |
| 08:21.26 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 08:21.48 | heliar | of size configurable a web gui. |
| 08:21.48 | heliar | ;p |
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| 08:35.20 | cargill | hi, I have a voicemail issue that I'm trying to simulate and don't have a HW phone handy, is there a linux softphone or a short script using some SIP library that can do the subscribe and let me see the mwi status/notify? |
| 08:36.55 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 08:39.52 | heliar | cargill: Switchvox is the easy-to-use phone system. |
| 08:42.05 | cargill | heliar: how can it help me with that if I already have an asterisk system? Only software I can find on their web is asterisk itself |
| 08:43.15 | cargill | just trying to talk to asterisk and see if/when a phone would see there is a voicemail waiting |
| 08:44.38 | cargill | but all hw phones are taken and people are starting to object to me fiddling with theirs |
| 08:47.25 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 10:00.04 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 10:00.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.0 (2014/05/29), 1.8.28.0 (2014/05/29); Standard: Asterisk 12.3.0 (2014/05/29); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 10:19.27 | heliar | Policjant pod Palacem Prezydenckim rozmawia przez telefon: "to ode mnie zalezy zycie prezydenta Stanow Zjednoczonych, a ty mi mowisz, ze mam lozko nie poslane?!" |
| 10:19.34 | heliar | Sry, |
| 10:19.37 | heliar | not that chan. |
| 10:25.06 | yottanami | TazzNZ: I Can you open http://pastebin.com/ ? I will put them there if you can |
| 10:26.18 | WIMPy | I don't see any error there. |
| 10:26.50 | WIMPy | But I suspect (hope) you didn't want to call SIP/0. |
| 10:27.17 | *** join/#asterisk sted (~sted@ppp-94-68-80-60.home.otenet.gr) |
| 10:27.40 | sted | hello again |
| 10:27.52 | phpboy | hello |
| 10:28.06 | yottanami | WIMPy: Why? Is 0 reserved ? |
| 10:28.23 | sted | Can somehow enable t38 gateway? |
| 10:28.31 | WIMPy | No. But a pretty bad username. |
| 10:28.34 | phpboy | sted: faxing? |
| 10:28.48 | sted | I have a fax connected to a t38 ata, and it's always using g711 |
| 10:29.02 | yottanami | WIMPy: Why? should I change it ? |
| 10:29.06 | phpboy | compile asterisk with app_fax |
| 10:29.21 | phpboy | Google: ReceiveFax and SendFax |
| 10:29.49 | sted | I think that it for received and sending directly from asterisk |
| 10:29.52 | WIMPy | yottanami: Read the security best practices thing. |
| 10:30.01 | sted | I just want to send and receive using plain fax machine |
| 10:30.01 | phpboy | sted: app_fax then |
| 10:30.05 | sted | ok |
| 10:31.20 | sted | phpboy, again appfax is for send/receive from asteriks, it can't help |
| 10:32.22 | *** join/#asterisk snmp (~snmp@95.47.164.2) |
| 10:34.46 | phpboy | sted: ok |
| 10:37.47 | phpboy | I misread |
| 10:37.58 | phpboy | are you having problems sending faxes? |
| 10:38.53 | sted | yes, while using the fax machine, I have about 50-60% failure rate.. |
| 10:39.02 | sted | guess it's because g711 |
| 10:39.14 | sted | the ata supports t38 |
| 10:46.06 | phpboy | That I cannot help you with |
| 10:46.08 | phpboy | sorry :( |
| 10:49.48 | sted | no problem |
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| 10:51.06 | hindi | sted, t38pt_udptl=yes, t38pt_rtp=yes |
| 10:51.36 | hindi | a long time ago we used faxmachines this working fine for us |
| 10:51.43 | sted | hindi, at extensions.conf ? |
| 10:51.48 | hindi | sip.conf general |
| 10:52.49 | hindi | what kind of ata= |
| 10:53.24 | sted | i'm using a fritz!box 7170 |
| 10:53.32 | sted | i can handle t38 |
| 10:53.35 | sted | it* |
| 10:53.55 | WIMPy | Why don't you sen directly from it? |
| 10:54.33 | sted | WIMPy, fax machine is the way older people were sending faxes more than 10 years ago |
| 10:54.44 | sted | I can't change this now, you know what I mean |
| 10:54.55 | WIMPy | ... leaving out Asterisk. |
| 10:55.26 | yottanami | As I see there is some options for Fax what do you prefer for send/receive fax on Asterisk? |
| 10:56.16 | sted | When I do the job, I'm using hylafax |
| 10:56.38 | sted | but when others do it, they want the "old plain fax" machine |
| 10:57.30 | WIMPy | Again: Why don't you sen directly from the FB instead of going through Asterisk? |
| 10:58.58 | sted | FB registers on asterisk.. |
| 10:59.07 | sted | I'm using this as an ata |
| 10:59.11 | sted | nothing more/less |
| 10:59.32 | sted | I'm using 2 fxs ... 1 phone 1 fax |
| 11:00.12 | hindi | and why using asterisk? Fritz can handle this :) |
| 11:01.41 | sted | asterisk make the whole call handling |
| 11:02.50 | hindi | ok. but using a asterisk for 1 phone and 1 fax ??!?! |
| 11:03.30 | hindi | is t38pt_udptl working? |
| 11:03.36 | hindi | how do you recive? |
| 11:03.48 | sted | no.. of course not... I have more than 10 extensions, 1 sip trunk and an isdn bri |
| 11:03.57 | hindi | sted, ok :) |
| 11:04.01 | sted | I receive in hylafax! |
| 11:04.11 | sted | always |
| 11:04.18 | hindi | hylafax is working? |
| 11:04.21 | sted | yes |
| 11:04.27 | hindi | ok |
| 11:04.29 | WIMPy | sted: You can still bypass Asterisk for Fax. |
| 11:04.33 | sted | send/receive works great in hylafax |
| 11:04.45 | hindi | show me your extensions.conf the fax sending part |
| 11:05.04 | sted | WIMPy, fritz is registering in asterisk, which manages the isdn/trunk parts |
| 11:05.11 | sted | wait |
| 11:05.45 | WIMPy | You use Asterisk to send via BRI? |
| 11:05.53 | sted | yep |
| 11:05.57 | WIMPy | ptp or ptmp? |
| 11:06.39 | sted | we use ptmp in GR |
| 11:07.31 | hindi | GR? |
| 11:07.33 | WIMPy | Great. Then just connect the FB to that BRI as well. |
| 11:08.07 | hindi | thats the best way. bybass asterisk |
| 11:08.18 | hindi | no more fax problems left :) |
| 11:09.35 | hindi | sted, show me the fax sending part in your extensions.conf |
| 11:10.45 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 11:11.04 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 11:15.17 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 11:16.06 | sted | WIMPy, asterisk & bri is in another site than fax machine |
| 11:16.21 | sted | otherwise I will simply get the POTS output from the ISDN NT |
| 11:16.39 | sted | hindi, there's no sending/receiving part |
| 11:16.48 | sted | it's an extension.... |
| 11:17.03 | sted | number/name/secret/allow/dissalow/etc |
| 11:17.04 | hindi | ok. the extension is dialing directly? |
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| 11:17.37 | hindi | i want to see the dialing part in the extensions.conf ... not your sip.conf |
| 11:18.44 | sted | wait |
| 11:18.47 | hindi | the part which is looking like: exten => _X.,1,Dial(${EXTEN}@sip.provider.de,,) |
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| 11:40.06 | sted | hindi, it's a new setup, i don't have any outbound route, from my previous was like this |
| 11:40.12 | sted | dial(dahdi/1/${EXTEN:1},,Tt) |
| 11:41.28 | hindi | sted, how u know its not working if there is no outbound route ? |
| 11:41.48 | sted | yes, now im doing an internal send for test |
| 11:42.06 | sted | i'm sending from the fax-machine/ata (ext: 399) to the hylafax (400) |
| 11:42.19 | sted | just to see codec usage etc |
| 11:42.24 | hindi | ok |
| 11:42.26 | hindi | and? |
| 11:42.30 | sted | g711u |
| 11:42.41 | sted | fritz show in decription that it's a fax call |
| 11:42.43 | sted | but nothing |
| 11:43.40 | hindi | so t38 is activated? |
| 11:43.45 | hindi | on your fb? |
| 11:46.13 | sted | yes of course |
| 11:46.26 | *** join/#asterisk Draecos (~Draecos@106-69-31-78.dyn.iinet.net.au) |
| 11:50.19 | bulkorok | usually the receiving side sends a T38 RE-INVITE to the sender... |
| 11:50.22 | WIMPy | Just to make sure: You're not usign some ancient Asterisk version, are you? |
| 11:51.20 | sted | bulkorok, this is for a passthrough I guess |
| 11:51.26 | sted | WIMPy, asterisk 11.7.0 |
| 11:51.39 | WIMPy | ok |
| 11:54.18 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 11:54.31 | bulkorok | fax should go where the lost socks are… |
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| 12:01.10 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:01.25 | puzzled | hi |
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| 12:10.36 | sted | bulkorok, what? |
| 12:11.02 | bulkorok | nobody should use fax anymore... |
| 12:11.45 | sted | Yes but these are out philosophies |
| 12:12.01 | bulkorok | yeah |
| 12:15.20 | aurs | Sometimes, if I have Dial(number1&number2,30,ot), I don't get a progress tone (depends on the (external) numbers dialed) |
| 12:15.30 | aurs | are there any downsides to adding a r to the dial options? |
| 12:15.37 | sted | so none knows/uses t38 gateway... |
| 12:15.48 | sted | or none sends fax! |
| 12:15.54 | aurs | sted: I've moved on to email ;) |
| 12:15.55 | hindi | sted, t38 aktive on FB, t38 support in sip.conf activated, |
| 12:16.17 | sted | hindi, tell me about sip.conf |
| 12:16.28 | sted | http://www.asterisk-pbx.hk/2012/08/working-with-asterisk-10-t38-gateway.html |
| 12:16.45 | hindi | sted, t38pt_udptl=yes, t38pt_rtp=yes |
| 12:17.03 | bulkorok | open ports... |
| 12:17.07 | hindi | put it into sip.conf [gerneral] section... as adviced about 1 hour ago |
| 12:17.19 | sted | i had only the updtl |
| 12:17.26 | sted | will insert rtp now |
| 12:17.31 | hindi | try both |
| 12:19.08 | sted | ok will try now |
| 12:20.30 | *** join/#asterisk retentiveboy (~retentive@c-50-150-62-29.hsd1.ga.comcast.net) |
| 12:21.38 | sted | still g711u |
| 12:22.41 | hindi | show me your sip.conf please |
| 12:22.49 | hindi | ~pb |
| 12:22.50 | 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. |
| 12:23.42 | sted | http://pastebin.com/peChhy85 |
| 12:23.47 | sted | here it is from the extension |
| 12:25.04 | sted | you need something else? |
| 12:25.16 | sted | (it is in seperate files - freepbx..) |
| 12:25.33 | sted | the rest |
| 12:25.33 | sted | http://pastebin.com/zmF0PtU7 |
| 12:26.21 | bulkorok | faxdetect=yes |
| 12:27.05 | hindi | deny=all permit= all :D |
| 12:27.12 | hindi | sry |
| 12:27.58 | hindi | what is the [general] section of your sip.conf |
| 12:27.59 | hindi | ? |
| 12:28.03 | hindi | the second link= |
| 12:32.06 | sted | ? |
| 12:32.17 | hindi | how are you configure your asterisk? |
| 12:32.26 | hindi | how do your * |
| 12:32.28 | hindi | sry :D |
| 12:32.33 | sted | freepbx/elastix |
| 12:33.10 | hindi | is there a configfile with the name sip.conf? |
| 12:33.33 | hindi | (never worked with freebpx or elastix) |
| 12:33.51 | hindi | and in this configfile is there a section called "[general]" |
| 12:39.33 | [TK]D-Fender | You should not be touching sip.conf on a freepbx system |
| 12:40.18 | bulkorok | wait… wasn't here... |
| 12:40.22 | bulkorok | ~freepbx |
| 12:40.22 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:40.27 | bulkorok | :-D |
| 12:42.23 | sted | This is system made for testing, just want to find what trigger makes the t38 enabled |
| 12:42.31 | sted | wait until find general |
| 12:42.35 | bulkorok | these two vars in sip.conf |
| 12:42.45 | [TK]D-Fender | sted: what are you using on extension 305, and what is on the other end of the call? |
| 12:42.52 | bulkorok | nothing else… there could be udptl.conf I think |
| 12:44.05 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-ftyxfkztrpptvjnh) |
| 12:44.25 | sted | [TK]D-Fender, the other end is an iax extension, hylafax, 399 |
| 12:44.50 | [TK]D-Fender | sted: what is 399? |
| 12:44.59 | [TK]D-Fender | 305 rather |
| 12:44.59 | sted | extension number |
| 12:45.08 | sted | http://pastebin.com/zmF0PtU7 the general I guess |
| 12:45.28 | [TK]D-Fender | sted: ..... you've just mentioned one end fo the call... what's on the other? |
| 12:46.16 | sted | 1st end: Fax machine connected to ATA (number 305) which calls to the number 399, hylafax |
| 12:46.53 | sted | Repeat the question, my english are such bad I guess |
| 12:47.00 | [TK]D-Fender | try being specific..... |
| 12:47.23 | sted | let's go again |
| 12:47.33 | sted | what you need to know |
| 12:48.48 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 12:49.15 | [TK]D-Fender | ATA <- details |
| 12:49.52 | sted | ATA -> Fritz!box fon 7170, T38 enabled |
| 12:49.53 | sted | http://pastebin.com/peChhy85 |
| 12:52.14 | aurs | sted: does sip.conf matter if you're using iax? |
| 12:52.33 | hindi | nope |
| 12:52.37 | hindi | sip.conf --> sip |
| 12:52.38 | sted | fritzbox is connected via sip |
| 12:52.50 | [TK]D-Fender | sted: You're running an old version of FreePBX. Regardless of the fact that the * 11 under there supports termination the DIALPLAN that your GUI generates will not have the GATEWAY command called to allow it to terminate T.38 |
| 12:53.03 | [TK]D-Fender | Because SIP settings alone isn't everything for termination |
| 12:53.33 | [TK]D-Fender | <PROTECTED> |
| 12:53.49 | sted | where I must insert this? |
| 12:54.19 | hindi | sted, please establish a call to your hylafax. than connect so asterisk an type sip show channelstats. get the call id of the call and than sip show channel <call-id> |
| 12:54.46 | [TK]D-Fender | sted: if your diaplan. This isa serious FreePBX question and I'm betting you don't have much understanding about how it works. |
| 12:54.59 | hindi | than there you can find Their Codec Capability: and Joint Codec Capability |
| 12:55.08 | [TK]D-Fender | hindi: No point as his dialplan isn't calling what's needed |
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| 12:56.02 | sarelon | \part |
| 12:56.33 | *** part/#asterisk sarelon (~sarelon@24-116-133-114.cpe.cableone.net) |
| 12:56.45 | sted | [TK]D-Fender, yes I'm begginer in these, now I'm learning |
| 12:57.06 | sted | When you say about diaplan, you mean, context or sth? |
| 12:57.25 | [TK]D-Fender | yes |
| 12:57.30 | [TK]D-Fender | The actual call flow |
| 12:57.34 | hindi | [TK]D-Fender, faxopt is needed? |
| 12:57.38 | [TK]D-Fender | YES |
| 12:57.47 | [TK]D-Fender | hindi: You missed the big-print |
| 12:58.04 | hindi | for passthrough? |
| 12:58.10 | [TK]D-Fender | .... |
| 12:58.15 | [TK]D-Fender | facepalms |
| 12:58.23 | hindi | :P |
| 13:00.32 | hindi | [TK]D-Fender, ok. |
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| 13:01.04 | sted | this is bad and I should feel bad |
| 13:01.25 | sted | It's so complicated the contexts thing in fpbx so I don't know nothing |
| 13:05.40 | sted | I need to insert this line with faxopt? |
| 13:05.43 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:05.44 | sted | that's the thing? |
| 13:06.37 | [TK]D-Fender | yes |
| 13:08.11 | hindi | [TK]D-Fender, I've checked the wiki.. never used asterisk 10... so didn't know the faxopt gateway option... long time ago we used fax machines it worked fine with the sip settings... |
| 13:08.36 | [TK]D-Fender | hindi: And you're still not reading the big print. |
| 13:12.50 | hindi | sry.. what I'am missing? cant follow |
| 13:13.27 | [TK]D-Fender | [08:58]hindifor passthrough? <- not happening |
| 13:13.37 | hindi | ok |
| 13:14.36 | hindi | thx |
| 13:16.00 | sted | I will try something else |
| 13:16.05 | sted | will add only this |
| 13:16.38 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 13:17.04 | sted | exten => _XXX./_305,1Set(FAXOPT(gateway)=yes) |
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| 13:17.16 | jaflong | hi, how is a channel removed from a group? |
| 13:17.20 | sted | sorry |
| 13:17.22 | sted | mean this |
| 13:17.27 | sted | exten => _XXX./_305,1,Set(FAXOPT(gateway)=yes) |
| 13:17.51 | [TK]D-Fender | jaWhat kind fo channel? What kind of group? |
| 13:17.55 | [TK]D-Fender | jaflong: What kind fo channel? What kind of group? |
| 13:18.34 | jaflong | sip channel . exten=>_X.,n,Set(GROUP()=${EXTEN}) |
| 13:18.55 | [TK]D-Fender | It gets removed when the call ends |
| 13:20.17 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:20.42 | jaflong | short and sweet. thanks |
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| 13:39.11 | sted | added in sip.conf |
| 13:39.20 | sted | for the specific extension |
| 13:39.46 | sted | setvar=FAXOPT(gateway)=yes,10 |
| 13:39.52 | sted | and worked, t38 |
| 13:42.51 | [TK]D-Fender | How did you set it? |
| 13:43.03 | sted | I added this in the extension |
| 13:43.12 | [TK]D-Fender | Because if you did that manually then the moment you make any change to your system that will get blown away |
| 13:43.19 | sted | in the same place it says about secrets |
| 13:43.30 | sted | it's ok I will reinsert this |
| 13:43.48 | sted | the thing is that with elastix or freepbx |
| 13:43.59 | sted | If it doesnt exist in web interface, you cannot enable |
| 13:44.13 | [TK]D-Fender | No, you just need to think of other ways |
| 13:44.14 | sted | so if I added this in contexts, will be tha same |
| 13:44.24 | sted | when fpbx regenerates contexts |
| 13:44.27 | sted | will be gone |
| 13:44.31 | [TK]D-Fender | there are CUSTOM files for you to add your stuff to |
| 13:44.41 | [TK]D-Fender | You need to learn the system you actually have |
| 13:44.53 | sted | will see it also |
| 13:45.01 | [TK]D-Fender | http://pastebin.com/JLHGuPEh <--- extensions_custom.conf |
| 13:45.02 | sted | if I can enter the specific extension to custom |
| 13:45.12 | [TK]D-Fender | Add it and make a custom destination pointed to it. |
| 13:45.41 | sted | so, you say, can I just insert these and then just use this context |
| 13:45.42 | sted | good |
| 13:45.54 | sted | that's what I wanted from the first time |
| 13:45.56 | [TK]D-Fender | And dial that instead of the hylafax IAXmodem extension directly |
| 13:46.12 | sted | :) |
| 13:47.17 | sted | hmmm tryied the first fax |
| 13:47.27 | sted | nothing, got an error |
| 13:47.28 | sted | nice |
| 13:47.45 | sted | will have to play with timeouts and error correction |
| 13:49.22 | alami | any one know something like that http://www.voip-info.org/wiki/view/MeetMe-Web-Control for confbridge |
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| 14:01.17 | sted | after adding t38pt_udptl = yes,redundancy,maxdatagram=400 |
| 14:01.35 | sted | all faxes are going smoothly |
| 14:01.38 | sted | :) |
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| 14:16.15 | sted | I have other question |
| 14:16.27 | sted | If I buy an 3line IP phone |
| 14:16.45 | sted | Can each line phone call through specific trunk? |
| 14:17.28 | [TK]D-Fender | You'd have to set them up as separate extensions |
| 14:17.41 | [TK]D-Fender | * is NOT a "key-system" |
| 14:17.53 | sted | Hmm that's what I thought |
| 14:17.57 | WIMPy | Use contexts |
| 14:18.00 | [TK]D-Fender | And FreePBX complicates this further |
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| 14:18.27 | sted | The phone will make and receive calls from 2 trunks |
| 14:19.00 | WIMPy | Use two accounts or use prefixes. |
| 14:19.05 | sted | It's ok these but when It gets an internal call? |
| 14:19.10 | sted | from other extension? |
| 14:19.32 | sted | tell me about prefixes |
| 14:19.52 | WIMPy | Use the dialplan/ |
| 14:20.02 | [TK]D-Fender | sted: use different dial patterns to go out different trunks |
| 14:20.25 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:20.26 | sted | It's ok with this but each time, I have to enter a prefix manually? |
| 14:20.42 | sted | or thes 2/3 lines in the phone will be registered with the same credentials? |
| 14:20.44 | WIMPy | If I need to select trunks I use standard routing codes with *, like 010*1 for the first route. |
| 14:21.11 | WIMPy | You can use different accounts in diferent contexts. |
| 14:21.19 | WIMPy | How you switch accounts depends on the phone then. |
| 14:22.29 | sted | ok lets assume we have for 1 phone 3 extensions |
| 14:22.34 | sted | 301/302/303 |
| 14:23.14 | sted | ok with contexts and inbound routes |
| 14:23.18 | sted | for the internal calls? |
| 14:23.22 | [TK]D-Fender | sted: You'd make separate routes for each matching on the CALLERID of that extension. |
| 14:23.35 | sted | can you tell me more about this? |
| 14:23.39 | sted | brb |
| 14:23.43 | [TK]D-Fender | sted: That phone will have have 3 separate extensions associated to it.... one per key. |
| 14:23.52 | WIMPy | For internal calls you can use whatever you want or even use yet another account. |
| 14:23.56 | [TK]D-Fender | The rules work exactly as one should think they do. |
| 14:24.02 | [TK]D-Fender | they are INDEPENDENT of each other |
| 14:24.52 | jeev | does switchvox smb 4.0 support a method to ping from the webgui? ping or trace |
| 14:31.56 | puzzled | I want to setup a receptionist phone where 2 DIDs each get their own linekey on a Polycom IP670. Afaict a Polycom config with separate per linekey SIP registrations for each DID should work. Or is there a better way to do this? I looked at SLA/BLA and BLF but not sure if they are better suited |
| 14:32.53 | [TK]D-Fender | * doesn't support SLA, and BLF is not something to ID a DID by really.. |
| 14:33.16 | [TK]D-Fender | So they will be separate regs entirely |
| 14:33.41 | puzzled | [TK]D-Fender: hi and thanks. Separate regs it is |
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| 14:38.43 | bulkorok | what was csv rotating!? |
| 14:40.08 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:40.15 | bulkorok | logger rotate... |
| 14:43.19 | *** join/#asterisk Roland- (~Roland-@unaffiliated/roland-) |
| 14:43.58 | jaflong | from the dialplan how can the channel names in a group be retrived? |
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| 14:45.31 | Roland- | hhi guys I need some help, I have an busy asterisk server and for some reasons calls are dropping... I know is a very very wide issue and hard to troubleshoot but ccan you please give me a hint where to look for? |
| 14:47.52 | [TK]D-Fender | Roland-: You'll have to look at your channels.... |
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| 14:48.40 | WIMPy | First off all, tell us how they drop. |
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| 14:52.25 | [TK]D-Fender | "for some reasons" <- Apparently this is how. |
| 14:53.56 | Roland- | I am still trying to get an idea of what is going wrong |
| 14:56.02 | WIMPy | "Make it so!" |
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| 15:04.37 | sted | I think for the phones, will be good if I have like this... 1st line) trunk1, 2nd line) trunk 2, 3rd line) only for internal calls |
| 15:04.54 | sted | just to set separate ringtones and have them always good |
| 15:05.01 | sted | Am I guessing right? |
| 15:05.34 | WIMPy | It all depends on the phones you use. |
| 15:05.48 | sted | will get minimum 3line phones |
| 15:05.55 | WIMPy | You might be able to set the ringtone or some sort of category from the server as well. |
| 15:06.01 | sted | really? |
| 15:06.23 | sted | then I have to make 2 extensions only? |
| 15:06.25 | Roland- | very weird, in the middle of the conversation asterisk sends a BYE message |
| 15:06.50 | WIMPy | And just because a phone supports multiple accounts doesn't mean it has to support different ringtones per account. |
| 15:06.58 | sted | with the same extension number? |
| 15:07.18 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 15:07.19 | WIMPy | What? |
| 15:07.43 | sted | Will I have for the specific phone only one account in asterisk? |
| 15:07.50 | sted | f.e. number 300? |
| 15:07.57 | sted | an register with this? |
| 15:07.59 | sted | and* |
| 15:08.10 | WIMPy | There are many possibilities. |
| 15:08.22 | sted | any suggestions? |
| 15:08.24 | WIMPy | Take a look at what your phones can do. |
| 15:08.27 | sted | ok |
| 15:08.29 | sted | will see |
| 15:08.30 | sted | thanks |
| 15:09.15 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:10.20 | sted | also, to get HD voice (if I have capable phones) do I have to enable sth in *? |
| 15:10.41 | WIMPy | The CODEC, yes. |
| 15:10.56 | sted | ok and then will be the default? |
| 15:11.12 | WIMPy | Depends on the CODEC priorities. |
| 15:11.19 | WIMPy | ... on both sides. |
| 15:11.26 | sted | ok |
| 15:11.32 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-qkhggrxwxswyayul) |
| 15:15.08 | sted | If I have 3 extensions for each phone |
| 15:15.18 | sted | we have a conflict in BLF |
| 15:15.28 | sted | which one to registe |
| 15:15.29 | sted | r |
| 15:15.36 | WIMPy | All? |
| 15:16.05 | WIMPy | One hint can monitor multiple devices (accounts). |
| 15:16.11 | sted | really? |
| 15:16.24 | sted | can you show me how? |
| 15:16.26 | sted | with one hint? |
| 15:17.07 | WIMPy | With &, just like you'd dial them. |
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| 15:22.35 | newtonr | sted, i'm literally editing this page right now, but look at the Defining Hints section https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints |
| 15:25.20 | [TK]D-Fender | newtonr: FREEPBX <- |
| 15:26.12 | newtonr | [TK]D-Fender, Ah, I didn't scroll up far enough. |
| 15:27.10 | [TK]D-Fender | sted: You should not be asking in here for these solutions. Your GUI defines the rules by which you have to work. Join #freepbx and continue in there for support |
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| 15:41.13 | sted | newtonr, thanks for the info |
| 15:41.32 | sted | [TK]D-Fender, i like to know the very first thing, from the source |
| 15:41.39 | sted | thank you |
| 15:41.56 | [TK]D-Fender | sted it wastes peoples time when you can't use the solution because of what you're running |
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| 15:43.21 | jaflong | any know about getting the channel names in a group |
| 15:49.15 | [TK]D-Fender | jaflong: "core show functions like GROUP" |
| 15:53.55 | marceloamorim | guys, I'm trying to use diferent contexts do transfer |
| 15:54.03 | marceloamorim | ike I use context phones and set the number like 600 Dial(SIP/person), this Dial is in the macro and this macro I use Set(__TRANSFER_CONTEXT) |
| 15:54.11 | marceloamorim | but isn´t work so far, I wish to set two diferents contexts one for call the number and check if this phone is available, and other context for transfer, on this transfer context I don't want to check anything |
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| 15:59.27 | WIMPy | marceloamorim: How do you transfer? |
| 15:59.48 | marceloamorim | I think I did another mistake |
| 16:00.18 | marceloamorim | I forgot to set on features.conf |
| 16:01.46 | marceloamorim | now I'm using atxfer and this variable __TRANSFER_CONTEXT is working well |
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| 16:08.51 | marceloamorim | actually, didn't work with this variable |
| 16:09.06 | marceloamorim | I'll do more tests |
| 16:14.38 | Katty | OH WHERE IS MY HAIRBRUSH |
| 16:16.18 | WIMPy | Katty: At the end of your arm. |
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| 17:43.39 | litn | hi |
| 17:43.52 | litn | when the other line hangs up my phone rings a fast busy instead of also hanging up |
| 17:43.57 | litn | any idea what the problem might be? |
| 17:46.07 | navaismo | IF(analog){check your disconnect supervision or try the busydetect and busycount settings} |
| 17:48.47 | litn | not analog |
| 17:53.39 | ChannelZ | So this is SIP? |
| 17:54.37 | litn | yes, completely sip |
| 17:55.15 | ChannelZ | Weird behavior (maybe configurable) of the device. What is it? |
| 17:55.57 | ChannelZ | Unless the fast busy is coming from the remote end and it's the one not realizing the call has been terminated. You'd have to look at a SIP debug and see if you get a BYE from them or not |
| 17:56.28 | ChannelZ | (IE the fast-busy might be inband) |
| 18:00.25 | litn | I don't think it's inband because I get it even after I transfer someone to another phone |
| 18:00.29 | litn | after the transfer finished I get the beeping |
| 18:00.48 | litn | probably device related then I guess .. It's a Grandstream GXP1450 |
| 18:02.07 | pabelanger | SIP trace will show you what is happening |
| 18:04.38 | ChannelZ | "Instead, it goes into a fast busy state with the display screen showing a 'Line Disconnected' message. This means that you have to manually hang up the phone each time. We easily found a setting (the "turn off speaker on remote disconnect" setting) in the web interface that resolves this issue." |
| 18:04.50 | ChannelZ | That's for a different phone but sounds like probably a Grandstream-ism |
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| 18:09.11 | marceloamorim | guys |
| 18:10.02 | marceloamorim | I tested about the transfer and the weird thing is, I can use the context that I want |
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| 18:11.18 | marceloamorim | but when I have a phone that has 2 sip accounts my verbose about that context shows on the CLI but the asterisk keep using another context |
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| 18:16.36 | marceloamorim | I debugged the situation and the problem is the phone, he send me SIP response 486 - Busy Here |
| 18:17.34 | marceloamorim | but he isn't busy, the sip account 2 has a call, the sip account 1 is free |
| 18:17.42 | marceloamorim | is this normal? |
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| 19:23.35 | ChannelZ | Are you sure DND isn't turned on or something? |
| 19:31.55 | sted | is any tutorial for SLA available? |
| 19:32.30 | sted | I think that switching between trunks will be better with SLA in ip phones than using multiple sip accounts... what do you think |
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| 19:38.19 | marceloamorim | ChannelZ: sorry about that man, I was afk |
| 19:40.36 | marceloamorim | I'm sure about that, there is a visual DND on display, and is turned off |
| 19:47.35 | jameswf | why do all ami commands lead to: ['Event: FullyBooted', 'Privilege: system,all', 'Status: Fully Booted', '', ''] |
| 19:49.08 | navaismo | are you login everytime you send a command? |
| 19:50.54 | jameswf | no |
| 19:52.30 | jameswf | kicks his laptop |
| 19:52.33 | tm1000 | jacobkiers: at least we know it’s fully booted |
| 19:55.30 | jameswf | tm1000: I wrote this class a while back and I think past assumptions/methods are no longer valid.... |
| 19:55.50 | tm1000 | jameswf: what did you write it in |
| 19:56.04 | jameswf | tm1000: twi guesses :) |
| 19:56.11 | tm1000 | php or pythong |
| 19:56.15 | tm1000 | im going with python |
| 19:57.05 | jameswf | I would process the stream back til i saw \r\n\r\n and I am thinking this event blast happens before the real data now. |
| 19:57.32 | jameswf | so it eats up my \r\n\r\n never allowing me to see the data |
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| 23:34.08 | JonathanD | Evening folks. |
| 23:34.32 | SpeedEvil | Evening |
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