| 00:00.07 | DelphiWorld | IsUp: http://dpaste.de/As2b/ dialplan |
| 00:02.26 | IsUp | DelphiWorld: go to your outbound, remove exten => 883510009900637,1,Answer() and exten => 883510009900637,1,Echo |
| 00:02.44 | IsUp | go to your iax peer, change context=default to context=voipms_incoming |
| 00:02.46 | DelphiWorld | IsUp: removed |
| 00:03.06 | IsUp | to your dialplan, add a context [voipms_incoming] and add exten => 883510009900637,1,Answer - exten => 883510009900637,n,Echo |
| 00:03.50 | IsUp | when you finish, type 'reload' on your CLI and place a call to your DID |
| 00:04.51 | DelphiWorld | IsUp: same hell :-) |
| 00:05.47 | IsUp | DelphiWorld: paste CLI output to pastebin please |
| 00:07.01 | DelphiWorld | IsUp: http://dpaste.de/8sAg/ |
| 00:07.12 | DelphiWorld | IsUp: sory if i missed some lines, i'm blind :-) |
| 00:07.45 | IsUp | DelphiWorld: you probably missed context= on your iax peer |
| 00:08.00 | IsUp | go to your peer context, set context=voipms_incoming |
| 00:08.10 | IsUp | and add voipms_incoming context to your dialplan |
| 00:08.12 | DelphiWorld | IsUp: but i do the in our out context |
| 00:08.26 | IsUp | lets try on different context now |
| 00:08.55 | IsUp | your current context is little bit confusing |
| 00:10.39 | DelphiWorld | IsUp: now is changed |
| 00:11.04 | IsUp | ok do 'reload' on your CLI |
| 00:11.08 | IsUp | and place a call to your DID |
| 00:11.40 | DelphiWorld | IsUp: can you give me how to dial a sip peer registered to me? |
| 00:11.46 | DelphiWorld | IsUp: let say peer 101 |
| 00:12.45 | IsUp | Dial(SIP/101) |
| 00:12.49 | DelphiWorld | IsUp: call is answered |
| 00:12.54 | DelphiWorld | now i need to try out echo |
| 00:13.30 | *** join/#asterisk sourcode (~code@ppp-115-87-234-237.revip4.asianet.co.th) |
| 00:17.17 | MikeH | hrm |
| 00:18.11 | MikeH | Has anyone seen an issue with Dahdi where an analogue card stops working and requires a power off and power on to start again |
| 00:19.13 | IsUp | MikeH: i had this problem. it stops ringing, not even a dialtone |
| 00:19.24 | MikeH | Well |
| 00:19.32 | MikeH | the card has actually disappeared from Dahdi |
| 00:19.33 | IsUp | /etc/init.d/dahdi stop & start was my solution :p |
| 00:19.42 | MikeH | I had this the other day |
| 00:19.45 | DelphiWorld | MikeH: what card you have? |
| 00:19.51 | MikeH | turn server off and back on again, it's fine |
| 00:20.00 | MikeH | but a reboot does not bring it back up |
| 00:20.15 | MikeH | stopping and starting dahdi only errors that the device is missing |
| 00:21.18 | MikeH | DelphiWorld, AX4G |
| 00:21.44 | DelphiWorld | MikeH: i use openvox |
| 00:23.02 | MikeH | hrm, lspci shows the card |
| 00:23.08 | MikeH | but /proc/interupts does not |
| 00:23.43 | MikeH | DelphiWorld, Openvox GSM card? |
| 00:23.57 | DelphiWorld | no, MikeH... FXO |
| 00:24.32 | MikeH | I'm thinking of sending this ax4g card back as faulty given the issues I'm having |
| 00:24.36 | MikeH | and switching to an openvox one |
| 00:24.46 | *** join/#asterisk philippel_mac_ (~p_lindhei@pool-71-188-251-53.sttlwa.fios.verizon.net) |
| 00:24.47 | MikeH | I'd like to find someone who uses one first though really |
| 00:24.58 | MikeH | For example the AX4G does not support answer detection |
| 00:25.01 | MikeH | it's ridiculous |
| 00:28.49 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 01:01.01 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 01:01.29 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.16.55) |
| 01:24.33 | *** join/#asterisk shapr (~shapr@nat/digium/x-omxhynobvgldssbh) |
| 01:31.05 | *** join/#asterisk philippel_mac (~p_lindhei@pool-71-188-251-53.sttlwa.fios.verizon.net) |
| 01:38.57 | *** join/#asterisk philippel_mac (~p_lindhei@pool-71-188-251-53.sttlwa.fios.verizon.net) |
| 01:48.25 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 01:58.27 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 02:20.15 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 02:25.33 | *** part/#asterisk RichardLynch (~RichardLy@c-67-162-103-133.hsd1.il.comcast.net) |
| 02:28.35 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:19.37 | shapr | Good Morning alla! |
| 03:45.55 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 03:53.20 | *** join/#asterisk edibrac (~buz@adsl-99-159-46-157.dsl.pltn13.sbcglobal.net) |
| 04:17.52 | *** part/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 04:18.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:47.13 | *** join/#asterisk luckman212_ (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 04:50.09 | *** join/#asterisk sourcode (~code@ppp-115-87-213-18.revip4.asianet.co.th) |
| 05:02.04 | *** join/#asterisk solokkhz (~solokkhz@bzq-79-178-203-129.red.bezeqint.net) |
| 05:02.13 | solokkhz | hello |
| 05:03.04 | solokkhz | can anyone help identify the problem with asterisk not calling a pstn trunk with the x100p card? |
| 05:09.44 | *** join/#asterisk b0gatyr (~b0gatyr@unaffiliated/b0gatyr) |
| 05:12.51 | solokkhz | someone? |
| 05:17.37 | *** join/#asterisk solokkhz (~solokkhz@bzq-79-179-198-191.red.bezeqint.net) |
| 05:32.48 | *** join/#asterisk coppice (~chatzilla@m121-202-79-4.smartone-vodafone.com) |
| 05:56.25 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 05:56.37 | *** join/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 05:56.45 | *** part/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 05:56.49 | *** join/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 05:58.37 | rushowr | hello all! I'm wondering if anyone might be willing to chat with me about some strange behavior I'm seeing with Local Channels.. I have a SIP channel entering a Queue which uses a Local to lookup info about the agent and then uses SIP to call the agent.... (SIP1 -> Local -> SIP2)... |
| 05:58.59 | shapr | what version of Asterisk? |
| 05:59.02 | rushowr | SIP2 is able to hear and speak to SIP1 |
| 05:59.06 | rushowr | 1.8. (one sec |
| 05:59.32 | rushowr | 1.8.2.2 |
| 05:59.32 | shapr | I've seen some fixes for Local channels in the 1.8.2 changelog, tried the latest release of 1.8.2? |
| 05:59.55 | rushowr | doublechecking that 1.8.2.2 is latest |
| 05:59.59 | rushowr | ahhh crap |
| 06:00.02 | rushowr | 1.8.2.3 |
| 06:00.11 | rushowr | lemme go download that and run a quick test call |
| 06:00.21 | rushowr | I'll brb :D |
| 06:00.37 | rushowr | swears he had JUST updated not 4 days ago |
| 06:01.30 | rushowr | methinks I should just resume using SVN checkouts again |
| 06:03.17 | rushowr | @shapr hey, quick question (it's been a loooonng day)..IIRC, if I want to always have the latest 1.8 vesrsion from SVN I just checkout the 1.8 branch right? |
| 06:03.50 | shapr | trunk |
| 06:04.06 | shapr | but you probably want the latest *release* instead of the latest svn |
| 06:04.10 | shapr | That's just my opinion, though |
| 06:04.18 | rushowr | er... well, yes and no |
| 06:04.29 | rushowr | bah, let me go doublecheck something |
| 06:04.51 | rushowr | the releases are tags, which means I would have to constantly go download the latest tarball or switch the source URL from SVN |
| 06:05.47 | rushowr | but, I could have sworn that using the major release branch would result in you having the latest version within that major version...because they merge trunk back into that branch as well as make a new tag whenever they push a new release within that version |
| 06:05.52 | rushowr | let me buzz a dev |
| 06:05.54 | rushowr | :) |
| 06:07.38 | shapr | I'm not a dev.... |
| 06:09.41 | rushowr | is that sarcasm? Sorry mate, I really am very very tired and not sure what you're trying to say...I don't know many of the actual core devs on sight in IRC so if you are, please say so, so that I don't embarrass myself further lol |
| 06:10.00 | shapr | No sarcasm meant. I just don't know what conventions the asterisk devs use for svn |
| 06:10.20 | rushowr | ah ok, good then :D that's what I thought. but I had to check :D |
| 06:11.03 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rswlwfjztserhiut) |
| 06:11.54 | rushowr | I've put a question out to the devs |
| 06:13.30 | rushowr | I've been working with a few clients who were on Asterisk version update locks for so long that when this client finally moved up, I found myself unclear about the conventions used as well |
| 06:13.51 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 06:13.55 | rushowr | doing a bit o' googling here too, maybe I'll finally find the reference(s) I had before :D |
| 06:16.17 | rushowr | ok here we go |
| 06:16.24 | rushowr | we were both kinda right shapr :D |
| 06:17.10 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 06:17.43 | rushowr | branches/1.8 contains pre-release (ALMOST bleeding edge) code for the 1.8.x code |
| 06:17.52 | rushowr | the stable code is always contained in tags :P |
| 06:22.56 | *** join/#asterisk astro__ (astro@silvia.drft.net) |
| 06:23.12 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:23.39 | astro__ | has zaptel.conf and zapada been depreciated for along time? |
| 06:23.46 | rushowr | yessir |
| 06:25.49 | luckman212 | hi guys, anybody here using a streaming source for MOH? |
| 06:27.16 | astro__ | anyone have an idea of what i could develop voip.ca into |
| 06:28.15 | *** part/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
| 06:34.53 | *** join/#asterisk jkroon (~jkroon@dsl-242-3-152.telkomadsl.co.za) |
| 06:37.20 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:55.23 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 06:58.47 | *** join/#asterisk sourcode (~code@ppp-115-87-239-138.revip4.asianet.co.th) |
| 07:11.11 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 07:15.41 | *** join/#asterisk jkroon (~jkroon@dsl-242-3-152.telkomadsl.co.za) |
| 07:17.52 | *** join/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 07:20.19 | *** join/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 07:21.18 | rgagnon | any ops online or is everyone sleeping? |
| 07:26.03 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:27.54 | *** join/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 07:28.20 | *** part/#asterisk rgagnon (~rgagnon@99-185-128-25.lightspeed.austtx.sbcglobal.net) |
| 07:28.55 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:29.03 | solokkhz | can anyone help identify the problem with asterisk not calling a pstn trunk with the x100p card? |
| 07:35.37 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:35.39 | schmidts | good morning |
| 07:50.36 | wdoekes2 | morning |
| 07:52.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:52.56 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:53.49 | solokkhz | no one want to help me :( |
| 07:54.16 | schmidts | solokkhz sorry didnt see whats your problem ;) |
| 07:54.45 | solokkhz | problem with asterisk not calling a pstn trunk with the x100p card |
| 07:55.00 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 07:55.19 | kaldemar | ~ask |
| 07:55.20 | 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. |
| 07:56.00 | shapr | solokkhz: Any particular error? |
| 07:56.29 | solokkhz | when i call it looks like its calling but there is no line. when i use tha passthrough phone it works. |
| 07:56.43 | solokkhz | no error |
| 07:56.52 | solokkhz | thats the strange thing about it |
| 07:56.53 | schmidts | solokkhz what does you see if you start a pri debug on the span? |
| 07:57.44 | drmessano | pri debug? |
| 07:58.08 | kaldemar | x100p is an FXO interface, you won't see anything with pri debug. |
| 07:58.27 | schmidts | :P sorry never used FXO lines with asterisk, only isdn ;) |
| 07:58.47 | drmessano | and even calling it an "FXO interface" is a stretch :) |
| 07:59.02 | schmidts | POTS interface? |
| 07:59.03 | shapr | I've never even seen an X100P |
| 07:59.42 | solokkhz | lol, i am a noob, and want to test it before i upgrade to a more expensive card |
| 07:59.47 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 08:00.23 | solokkhz | the sip extension work great |
| 08:00.34 | solokkhz | but can go out to pstn |
| 08:00.56 | shapr | Is the X100P supported by DAHDI? |
| 08:01.17 | solokkhz | yeah it sees it |
| 08:01.23 | solokkhz | see it |
| 08:01.29 | solokkhz | i think |
| 08:01.53 | shapr | Are you using DAHDI 2.4.0.0 ? |
| 08:02.32 | solokkhz | [trixbox ~]# dahdi_hardware |
| 08:02.32 | solokkhz | pci:0000:04:00.0 wcfxo+ 1057:5608 Wildcard X100P |
| 08:02.40 | drmessano | *gulp* |
| 08:03.09 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1499:98c:646c:c8f9) |
| 08:03.18 | schmidts | solokkhz is your hostname only trixbox or are you using trixbox? |
| 08:03.47 | solokkhz | hmmm.. i use trixbox, shouldn't i? |
| 08:04.35 | schmidts | solokkhz LOL thats like joing #linux and ask if you should use windows :D |
| 08:04.45 | schmidts | not that bad but like this |
| 08:05.27 | solokkhz | they both use the same freepbx and dahdi |
| 08:05.41 | solokkhz | its more like win2k and winxp |
| 08:05.42 | solokkhz | lol |
| 08:06.01 | solokkhz | both don't work with my card |
| 08:06.03 | solokkhz | :( |
| 08:06.23 | drmessano | Trixbox doesnt use FreePBX |
| 08:06.53 | drmessano | They use a fork of FreePBX 2.4 they call "PBXConfig", which includes crap changes to the code that many cringe at, at best |
| 08:07.29 | solokkhz | what sould i use then? |
| 08:08.01 | drmessano | AsteriskNOW |
| 08:08.34 | solokkhz | i installed it, and couldn't get used to the gui after using trixbox |
| 08:09.13 | drmessano | Because it wasnt green? |
| 08:09.22 | drmessano | for the most part, the GUI should be the same |
| 08:09.50 | solokkhz | lol, i guess there are more crap on the trixbox toclick on |
| 08:09.51 | solokkhz | lol |
| 08:11.51 | solokkhz | but anyway, the ssh commands are the same |
| 08:12.08 | drmessano | ssh is ssh |
| 08:13.09 | drmessano | AsteriskNOW has a nice module for configuring DAHDI cards and it's also supported on IRC in several places |
| 08:13.51 | solokkhz | <PROTECTED> |
| 08:13.51 | solokkhz | [1] |
| 08:13.51 | solokkhz | active=yes |
| 08:15.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:16.59 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:20.06 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 08:21.50 | *** join/#asterisk mchou_ (~quassel@unaffiliated/mchou) |
| 08:30.35 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 08:37.21 | solokkhz | no dial tone |
| 08:38.10 | kaldemar | where do you expect to hear a dial tone? |
| 08:44.52 | solokkhz | not a dial tone, but i excpect it to dial |
| 08:45.12 | kaldemar | can you receive calls? |
| 08:45.32 | kaldemar | what do you see in CLI when you make a call? |
| 08:46.29 | tzafrir_laptop | What exactly do you dial? |
| 08:46.37 | tzafrir_laptop | When do you expect to get a dial tone? |
| 08:48.00 | solokkhz | can't receive calls as well through pstn |
| 08:48.38 | solokkhz | i set the outbounf route to call to XXXXXXX and NXXXXX |
| 08:51.33 | solokkhz | last message say: |
| 08:51.36 | solokkhz | <PROTECTED> |
| 08:51.36 | solokkhz | <PROTECTED> |
| 08:51.55 | solokkhz | and the call still active |
| 08:52.01 | solokkhz | it does not hang off |
| 08:52.05 | solokkhz | just quiet |
| 08:52.11 | solokkhz | no idaling or something |
| 08:52.17 | solokkhz | dialing* |
| 08:54.32 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:55.34 | solokkhz | i am starting to think that maybe my card does not work |
| 09:02.06 | tzafrir_laptop | If you plug a standard phone instead, can you dial? |
| 09:07.14 | solokkhz | yes, standard phone works and the card passthrough plug also work with a regular phone |
| 09:18.18 | *** join/#asterisk donatas (~donatas@office.kis.lt) |
| 09:18.51 | donatas | hey. Can I do the same thing excluding Answer(); ? http://p.defau.lt/?0_RLRxStB88EB_MOBidJyg |
| 09:19.18 | donatas | I want, that user won't be charged if I don't answer realy. |
| 09:24.27 | kaldemar | donatas: the Playback app will answer the call anyway if you don't give it the noanswer option. the used channel has to support early media or the caller won't hear your announcement. |
| 09:25.43 | donatas | aha, I see... Is it possible to do that? |
| 09:25.47 | donatas | without answering |
| 09:26.47 | kaldemar | remove the Answer line and PlayBack(sveiki) -> Playback(sveiki,noanswer) |
| 09:27.13 | donatas | a |
| 09:27.16 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:27.20 | donatas | thank youi |
| 09:28.01 | donatas | realy.. noanswer: Play the sound file, but don't answer the channel first (if hasn't been answered already). Not all channels support playing messages while still on hook. :) |
| 09:36.24 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:48.48 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:54.18 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:56.29 | *** join/#asterisk chasing`Sol (~rc4@82.201.135.150) |
| 09:57.35 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 10:15.51 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 10:17.35 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 10:28.19 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:34.36 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:36.13 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 10:44.02 | *** join/#asterisk Guifort (~Guifort@ks34643.kimsufi.com) |
| 10:54.29 | *** join/#asterisk darkskiez_ (~dz@62.50.248.119) |
| 11:00.17 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1499:98c:646c:c8f9) |
| 11:11.39 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:14.06 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 11:21.31 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:23.03 | *** join/#asterisk Guifort (~Guifort@ks367821.kimsufi.com) |
| 11:36.20 | *** join/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org) |
| 11:52.03 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:56.44 | *** join/#asterisk darkskiez_ (~dz@62.50.248.119) |
| 12:03.12 | *** join/#asterisk chasing`Sol (~rc4@82.201.135.150) |
| 12:03.47 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 12:07.11 | *** join/#asterisk As001 (~IPN05@194.106.175.113) |
| 12:07.40 | As001 | Hi is it possible to change music oh hold for logged in agents and that change take effect emidietly without logging out agents ? |
| 12:07.44 | *** join/#asterisk bjhaid (~abejide@41.155.112.180) |
| 12:08.22 | As001 | I tried with moh reload and reload module chan_agents.so |
| 12:08.35 | As001 | but it keeps playing old moh until they log out and log in again |
| 12:10.12 | *** join/#asterisk darkskiez_ (~dz@62.50.248.119) |
| 12:12.33 | Jasnejac | As001: can you achieve this with the CHANNEL function by changing the MOH class? |
| 12:24.19 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 12:29.56 | As001 | ok thanks I will see that |
| 12:32.30 | *** part/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 12:46.47 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:53.21 | *** join/#asterisk bjhaid (~abejide@41.155.114.11) |
| 12:58.48 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 13:01.36 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 13:02.18 | verywiseman | how can i hide caller id and sip's extension number when i make call? |
| 13:04.13 | kaldemar | verywiseman: what version of asterisk are you using? |
| 13:04.29 | verywiseman | kaldemar, 1.4 |
| 13:05.08 | *** join/#asterisk ariel_ (~chatzilla@216.7.159.250) |
| 13:05.09 | *** join/#asterisk porche (~kursad@91.188.208.204) |
| 13:05.19 | porche | Hi all |
| 13:06.02 | porche | I want to ask which linux distribution you use for asterisk? |
| 13:06.26 | Tozz_ | does it matter? |
| 13:06.31 | Tozz_ | we're probably all using something else ;) |
| 13:06.40 | kaldemar | verywiseman: app SetCallerPres |
| 13:06.50 | porche | Tozz, I am used to debian |
| 13:07.08 | Tozz_ | i'm using ubuntu |
| 13:07.10 | porche | now I want an upgrade with hardware raid, hosting guys push me for Centos |
| 13:07.13 | kaldemar | porche: then use debian for your install. |
| 13:07.28 | Tozz_ | there is no reason not to use debian for hardware raid |
| 13:07.36 | kaldemar | porche: use what suits your needs and you're comfortable with. |
| 13:07.52 | Tozz_ | hosting guys probably want to use centos because thats their fav. flavour |
| 13:07.58 | porche | yeah |
| 13:08.12 | porche | but issue is, I ordered a debian box from them with software raid |
| 13:08.45 | porche | somehow, under heavy traffic, its load spikes, when I check the details, it tries to resync swap raid |
| 13:09.24 | *** join/#asterisk bjhaid (~abejide@41.155.114.11) |
| 13:12.02 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 13:12.04 | OlafsenM | hi guys |
| 13:12.30 | OlafsenM | is it possible for Sierra W Aircard to work with Asterisk? |
| 13:13.05 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:13.05 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:14.10 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 13:15.42 | *** join/#asterisk bjhaid (~abejide@41.155.114.11) |
| 13:16.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:26.32 | *** join/#asterisk superbeef (~Lane.Jenn@74-120-209-234-rev.redanvil.net) |
| 13:26.45 | verywiseman | kaldemar, i try SetCallerPres , but also sip extension is appearing |
| 13:28.58 | superbeef | I've got a realtime meetme setup, which works great, minus the fact that it breaks out to transfer mode when somebody hits '#' it breaks out into wanting to transfer |
| 13:29.04 | Tozz_ | porche: swraid sucks |
| 13:29.24 | Tozz_ | and afaik its not recommended to run your swap on a sw raid device |
| 13:29.28 | kaldemar | verywiseman: sip extension as in sip user? |
| 13:29.45 | verywiseman | kaldemar, yes |
| 13:29.48 | porche | yeah I figured it out with 400 load on asterisk |
| 13:29.53 | *** join/#asterisk Surrealist (~carlos@87.111.157.25) |
| 13:29.54 | porche | server |
| 13:30.14 | Tozz_ | that sounds like IO issues |
| 13:30.14 | porche | under high traffic, if swap resync starts asterisk goes out of control |
| 13:30.24 | kaldemar | verywiseman: try username and fromuser settings for the peer in sip.conf. |
| 13:30.27 | porche | a possible problem with disks? |
| 13:30.30 | Tozz_ | why would it resync? |
| 13:30.41 | Tozz_ | it would only do that if there is something wrong with the array |
| 13:30.45 | porche | only clue Tozz, system's ram usage is high |
| 13:30.52 | porche | was high, |
| 13:31.05 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:31.15 | Tozz_ | i think there is a hardware issue. A load as high as that is almost always lots of IO wait |
| 13:31.17 | porche | it may be possible that, out of ram, and it started huge swap |
| 13:31.33 | Tozz_ | yes but huge swap should not cause the raid array to resync |
| 13:31.42 | porche | oh is it |
| 13:31.48 | porche | I am not used to raid at all |
| 13:31.59 | porche | just they told me that raid option will be good |
| 13:32.05 | Tozz_ | hardware raid is always better ;) |
| 13:32.07 | porche | but I forced them to install debian |
| 13:32.14 | porche | they wanted centos |
| 13:32.23 | mzahariev | Tozz_: you are so wrong about the sw raid :) |
| 13:32.24 | porche | now I am holding a bomb :) |
| 13:32.42 | Tozz_ | mzahariev: I hear that alot, but I'm not wrong ;) |
| 13:32.51 | blitzrage | sw raid has burned me more than once |
| 13:33.20 | Tozz_ | me too |
| 13:33.26 | porche | Tozz, resync started on swap raid btw, not the main disks |
| 13:33.39 | Tozz_ | it should not resync at all if there is nothing wrong with the array |
| 13:33.41 | porche | they never resync other than auto-check, once a month, |
| 13:33.46 | Tozz_ | aaah |
| 13:33.54 | mzahariev | for example take one sw RAID1 and hw RAID1 and see the diference your self |
| 13:34.30 | Tozz_ | mzahariev: have a sw raid disk fail (with IO commands hanging) and your box dies |
| 13:34.38 | Surrealist | Hi, can anyone ask me if there is a way to perform some actions through Asterisk CLI? I need manage accounts and meetme conferences for my clients(add, remove, modify). The only thing i need to know is if it's possible on that way, 'cause if not i'm plannig on develop a custom app that use freepbx php functions. Thank you! |
| 13:35.20 | porche | Surrealist, it can be done from manager API |
| 13:35.27 | porche | CLI is not the right place |
| 13:35.37 | blitzrage | agreed |
| 13:36.08 | blitzrage | you can do some basic administrative tasks, but not the type of tasks you've given as an example |
| 13:36.18 | porche | true |
| 13:36.38 | Surrealist | porche, blitzrage, and it can be scripted with manager API? |
| 13:37.02 | blitzrage | Surrealist, that's the purpose of the AMI -- to allow external applications to control asterisk |
| 13:37.17 | porche | Surrealist, yes, check AMI or Manager API Asterisk |
| 13:37.21 | porche | it's event based |
| 13:37.43 | Surrealist | blitzrage, ok, so i'll put my efforts on this. Many thanks, very appreciated :-) |
| 13:38.10 | porche | so guys, hardware raid |
| 13:38.19 | porche | that's the right decision |
| 13:38.25 | porche | centos or debian ? :) |
| 13:38.43 | Surrealist | porche, thanks to toy too ;-) |
| 13:38.51 | Tozz_ | porche: whatever you prefer |
| 13:38.54 | Tozz_ | so if u like debian, use debian |
| 13:39.02 | porche | I like debian a lot, hosting guys like centos, if I push them to debian, I know they will screw up the things |
| 13:39.13 | Tozz_ | then use centos ;) |
| 13:39.17 | porche | hehe |
| 13:39.34 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:39.40 | Tozz_ | for asterisk it doesnt matter |
| 13:39.45 | porche | centos 32 bit or 64 bit? |
| 13:40.05 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:40.13 | Tozz_ | 64-bit |
| 13:40.27 | porche | Tozz, I was a centos guy once upon a time, but after I start using debian, what ever box I put, I use debian. |
| 13:40.29 | Tozz_ | otherwise you will not be able to use >4G ram |
| 13:40.39 | porche | yep, time to switch back to centos |
| 13:40.43 | porche | thank you Tozz |
| 13:40.55 | Tozz_ | np |
| 13:43.26 | blitzrage | porche, I use both CentOS and Ubuntu (server versions) |
| 13:43.59 | porche | blitz, so you are happy with centos |
| 13:44.16 | blitzrage | porche, yes I've been using it for years and we documented it here: |
| 13:44.23 | blitzrage | grrr |
| 13:44.28 | blitzrage | I can't type tilde for some reason |
| 13:44.36 | blitzrage | infobot, tell porche about newbook |
| 13:44.42 | blitzrage | there, a work around :) |
| 13:45.02 | blitzrage | the answer is always, "use whatever you're comfortable with" because Asterisk doesn't care at all what distribution you use |
| 13:45.04 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:50.18 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:51.53 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 13:56.25 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 13:58.38 | *** part/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 14:03.06 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.10.238) |
| 14:03.08 | DelphiWorld | hi guys |
| 14:03.53 | porche | blitz, I totally agree, but when I pushed the DC guys, they ruined everything |
| 14:04.51 | Surrealist | mmm I can't find any related commands in AMI to create sip extensions(for example). So i'll have to develop in the hard way :_( |
| 14:05.25 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 14:06.38 | porche | Surrealist, what are you trying to do? |
| 14:06.59 | blitzrage | porche, I don't understand what you mean by that |
| 14:07.39 | porche | blitz, I meant, I like debian, DC guys like Centos, I pushed the to debian, and I have a bad box in my hands |
| 14:07.41 | blitzrage | Surrealist, where are they being stored? You either need to modify the sip.conf file directly (or the included file, which I usually do), or use something like realtime to load your peers from a database, and then just modify the database |
| 14:07.45 | porche | which is a production server |
| 14:07.46 | porche | now |
| 14:08.02 | blitzrage | porche, then you should have used CentOS because that's what THEY are comfortable with (the people managing the box) |
| 14:08.28 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 14:09.14 | Surrealist | blitzrage, so, there is no way to say for example: add exten 6000 bla bla bla, scripted in a php? Or use a function that you can pass some parameters? |
| 14:09.17 | porche | true, blitz, I learnt that |
| 14:10.12 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 14:10.16 | Surrealist | porche, i want to add users to my asterisk box through drupal. So, when someone registers a username i'll give him a number call(Ex.6000) |
| 14:10.34 | blitzrage | Surrealist, sure but you have to write the function to do that -- asterisk doesn't write that information to the sip.conf file -- you do. So your web application has to be able to store the information about your devices, and write the information to the sip.conf file. It also has to be able to handle the ability to write to the dialplan to enable the routing of calls via extensions. |
| 14:10.43 | porche | check realtime |
| 14:10.52 | porche | Surrealist, this is not AMI |
| 14:10.59 | porche | this goes to realtime asterisk |
| 14:11.01 | porche | basically |
| 14:11.04 | blitzrage | it is MUCH easier to use realtime and just write to the database |
| 14:11.06 | porche | you add to mysql |
| 14:11.07 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 14:11.19 | Surrealist | yes, but i want to use freepbx too! |
| 14:11.24 | porche | and asterisks loads from there |
| 14:11.26 | blitzrage | good luck to you then |
| 14:11.47 | porche | I dont know much about freepbx |
| 14:12.48 | Surrealist | mmmmmmmmm, i think i will emulate what freepbx does. In a nutshell, it loads the values in a mysql db and then parse all data to the config files. |
| 14:12.59 | blitzrage | yuck |
| 14:13.03 | blitzrage | but yes, that's what you need to do |
| 14:13.14 | blitzrage | asterisk won't write to the configuration files for you |
| 14:13.37 | blitzrage | Surrealist, I suggest you read the "Relational Database Integration" chapter at http://ofps.oreilly.com |
| 14:14.10 | Surrealist | i have this book, i'll take a look |
| 14:14.27 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 14:15.38 | blitzrage | actually you don't have that book because it hasn't been printed yet :) |
| 14:15.44 | blitzrage | what I pointed you to was the third edition |
| 14:16.02 | blitzrage | (which has an updated chapter on database integration) |
| 14:16.18 | Surrealist | but i have a module that loads some data in the freepbx db, so i think it will not be much harder to do the rest. |
| 14:16.26 | *** join/#asterisk ruben23 (~Administr@121.97.63.210) |
| 14:16.35 | Surrealist | ouch, yes i have second edition :) |
| 14:17.19 | ruben23 | hi guys |
| 14:17.46 | Surrealist | thank you for your time ;-) |
| 14:18.31 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 14:21.20 | *** part/#asterisk ruben23 (~Administr@121.97.63.210) |
| 14:21.40 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 14:22.29 | BlackBishop | any ideas why I can't send sms message > 75 chars ( using chan_datacard ) ? |
| 14:23.01 | wdoekes2 | that's an odd limit, isn't it 70? |
| 14:23.11 | *** join/#asterisk Akiraa (~Akira@92.81.204.170) |
| 14:23.49 | BlackBishop | hm .. lemme check |
| 14:24.19 | OlafsenM | s it possible for Sierra W Aircard to work with Asterisk? |
| 14:24.26 | BlackBishop | 123456789012345678901234567890123456789012345678901234567890 ( 60 chars ) fails ... |
| 14:25.34 | BlackBishop | same for 12345678901234567890123456789012345678901234567890 ( 50 ) |
| 14:25.37 | BlackBishop | weird |
| 14:25.47 | wdoekes2 | ok.. 70 is how many utf16 chars you would get in a single sms, vs. 160 of gsm03.38 7bit chars.. but it's something else then |
| 14:26.29 | BlackBishop | wdoekes2: ok, where do I set what type of encoding it should use ? |
| 14:26.35 | BlackBishop | I'd like it to use 7bit PDU |
| 14:26.49 | wdoekes2 | I have no idea.. I haven't used asterisk for anything else than sip |
| 14:27.52 | BlackBishop | ok, I had smsaspdu=yes .. "SMS message too long, 70 symbols max" |
| 14:27.56 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 14:27.59 | BlackBishop | set it to no ... and I got that message |
| 14:28.02 | BlackBishop | which isn't good :/ |
| 14:28.22 | angryuser | Hello, can someone recall if asterisk 1.2 support late SDP negociation ? |
| 14:28.34 | *** join/#asterisk erinspice1 (~erin@nat/digium/x-byvhfvcdflneffub) |
| 14:29.35 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:31.08 | wdoekes2 | ok BlackBishop, so you either get utf16 (max. 70 chars) or you have to send a binary blob (as pdu), right? http://www.dreamfabric.com/sms/ |
| 14:31.55 | BlackBishop | yeah, I know the site .. I made a php to convert texts into 7bit pdu messages .. |
| 14:32.06 | BlackBishop | I was thinking about fully using asterisk though .. |
| 14:35.57 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 14:42.46 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 14:52.38 | Katty | drags in |
| 14:55.07 | beardy | looks at what Katty dragged in |
| 14:56.55 | Katty | beardy: flu |
| 14:57.12 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-kwjfiqrshuxedvgc) |
| 14:57.51 | beardy | Katty: Eep |
| 14:58.19 | Katty | beardy: eep indeed :< i hate people sharing their germs with me |
| 14:59.05 | beardy | Katty: Only time irchugs is better than real ones. |
| 14:59.15 | Katty | yesr |
| 14:59.39 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 14:59.41 | beardy | feeds Katty soup |
| 15:04.19 | *** part/#asterisk porche (~kursad@91.188.208.204) |
| 15:07.27 | *** join/#asterisk odin917 (~Gavin_Sil@64.61.105.226) |
| 15:07.43 | odin917 | hey everyone |
| 15:09.01 | odin917 | i have an issue where i have configured a sip trunk in my asterisk box, when i send out the registration I get denied by the carrier. I spoke to a tech on the phone and he said that my box was trying to identify as carrierSuppliedUsername@myIPaddress instead of carrierSuppliedusername@myCarrierHost |
| 15:09.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:09.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:09.27 | odin917 | not sure if its a setting, or if the carrier doesnt know what hes talking about |
| 15:10.45 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 15:12.01 | drmessano | Anyone using 1.8 branch SVN? |
| 15:12.22 | tobi- | what might be the problem if the calls on my asterisk seem to be only half-duplex? |
| 15:16.05 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:16.59 | schmidts | tobi- it depends on your bandwith and how many calls you want to have at once |
| 15:17.40 | schmidts | half-duplex means you can only send or receive at once, think what this would mean to a stream in both directions |
| 15:17.46 | tobi- | bandwith is 50/10mbit down/upstream and call number is 1 atm |
| 15:18.26 | schmidts | tobi- you are not talking about half duplex, you mean asynchron, right ;) |
| 15:18.31 | beardy | tobi-: Sound in only one direction, or what do you mean? |
| 15:18.35 | tobi- | wont be that many concurrent calls |
| 15:18.45 | tobi- | its like walky-talky |
| 15:19.25 | tobi- | moh stops whenever i transmit |
| 15:19.29 | jaytee | we have 50/10 with Comcast and use Flowroute and it works great. We usually never get more than 6 to 8 concurrent calls at peak but we've got bandwidth to spare. |
| 15:19.30 | tobi- | for example |
| 15:20.11 | schmidts | tobi- then you might have another problem which have nothing to do with half-duplex |
| 15:20.39 | jaytee | tobi check the phones for silence suppression or "comfort noise" as some manufacturers call it. if it's enabled disable it. |
| 15:21.37 | tobi- | oh that might be it, ill check, thanks for the hint |
| 15:22.06 | schmidts | tobi- or VAD = Voice Activation detection ;) |
| 15:23.09 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:26.09 | i_heart_asterisk | hi |
| 15:26.42 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 15:29.32 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:30.27 | *** join/#asterisk felimwhiteley_ (~quassel@109.255.104.145) |
| 15:32.44 | *** join/#asterisk moy_ (~moy@h96-61-34-34.mdsnwi.tisp.static.tds.net) |
| 15:38.33 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:40.41 | *** join/#asterisk Infin1ty|work (~erez@pdpc/supporter/active/infin1ty) |
| 15:41.04 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:46.15 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 15:47.19 | *** join/#asterisk lost_soul (~noymfb@cpe-74-78-191-114.twcny.res.rr.com) |
| 15:48.10 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:52.14 | luckman212 | does anyone have any idea why a 'moh reload' would cause my MOH to die? (Ast 1.8.2.3 + mpg123 stream) I am tearing out what little hair I have left on this one |
| 15:53.20 | *** join/#asterisk logicwrath (~no@mail.vistitude.com) |
| 15:53.31 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-40.nayatel.pk) |
| 15:54.22 | mechbangirc | how can i use one SIP peer account for multiple registration? |
| 15:54.41 | odin917 | i have an issue where i have configured a sip trunk in my asterisk box, when i send out the registration I get denied by the carrier. I spoke to a tech on the phone and he said that my box was trying to identify as carrierSuppliedUsername@myIPaddress instead of carrierSuppliedusername@myCarrierHost |
| 15:55.23 | *** join/#asterisk andyoutside (cec00d82@gateway/web/freenode/ip.206.192.13.130) |
| 15:55.25 | schmidts | odin917 take a look at the sip.conf.sample file i think you can set the peer@host by yourself in the registration string |
| 15:55.30 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 15:56.16 | logicwrath | Can someone recommend a metered carrier that will burst unlimited channels and let me specify CID? |
| 15:56.24 | luckman212 | odin917: check the 'fromdomain=' and 'host=' lines in your trunk contexts |
| 15:56.25 | andyoutside | is there an easy way to upgrade asterisk that I did from source with yum? |
| 15:57.55 | odin917 | thanks guys illl take a look now |
| 16:01.56 | ClintGoudie-Nice | andyoutside, you could make uninstall in your source and then yum install the packages you want. I'd make sure to backup all your config first though. It's probably not going to be pretty. |
| 16:04.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:04.41 | andyoutside | would it be easer to use the source code going from 1.8.0 to 1.8.2 |
| 16:04.41 | odin917 | luckman212: thanks, i think the fromdoamin did the trick |
| 16:05.01 | luckman212 | cool |
| 16:06.20 | *** join/#asterisk darkskiez_ (~dz@62.50.248.119) |
| 16:10.02 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:10.52 | Katty | driveby hugskwishes Qwell |
| 16:11.16 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-41.dsl.rcsntx.sbcglobal.net) |
| 16:12.03 | jaytee | waves at Katty as she does her driveby |
| 16:12.47 | Katty | hugs jaytee |
| 16:13.04 | *** join/#asterisk Thorn (~Thorn@unaffiliated/thorn) |
| 16:13.13 | DelphiWorld | kick out Katty from the sea |
| 16:13.16 | Tozz_ | normally u get shot during a driveby |
| 16:13.19 | andyoutside | ClintGoudie-Nice,? |
| 16:13.21 | Thorn | hello |
| 16:13.22 | Tozz_ | hugs is something new for me |
| 16:13.51 | *** join/#asterisk lost_soul (~noymfb@cpe-74-78-191-114.twcny.res.rr.com) |
| 16:14.15 | Thorn | is it possible to place a call into a queue without answering it and only answer it when an agent answers? |
| 16:14.37 | ClintGoudie-Nice | andyoutside: dunno. I've really only run from RPM, and occasionally slammed source compiles over the top of the rpm (and then later wiped the box) |
| 16:15.21 | Katty | DelphiWorld: :< |
| 16:15.37 | DelphiWorld | Katty: :D |
| 16:15.51 | Katty | kicking isn't nice. |
| 16:16.05 | i_heart_asterisk | anybody using polycoms and AGI as dialplan having issues after transferring calls? |
| 16:16.35 | leifmadsen | i_heart_asterisk: what version? |
| 16:16.41 | i_heart_asterisk | 1.6 |
| 16:16.44 | leifmadsen | oh wait, that was AMI I'm thinking of |
| 16:16.48 | leifmadsen | 1.6 is not a version (and it's not even a branch) |
| 16:17.04 | leifmadsen | branches are 1.6.0, 1.6.1, and 1.6.2. Version examples would be 1.6.2.15 |
| 16:17.06 | i_heart_asterisk | 1.6.1.20 |
| 16:17.27 | i_heart_asterisk | seems to be a problem that asterisk does not notify AGI script of transfer when using polycom's softkeys |
| 16:17.47 | i_heart_asterisk | i opened a ticket on mantis but nobody has responded... |
| 16:17.49 | i_heart_asterisk | https://issues.asterisk.org/view.php?id=18745 |
| 16:17.56 | leifmadsen | that's because I haven't triaged it yet |
| 16:18.51 | leifmadsen | why does your ticket say a different version? |
| 16:19.05 | leifmadsen | you just told me 1.6.1.20, which isn't a supported branch of Asterisk |
| 16:19.18 | jaytee | little did Leif know when he was a boy that someday his job would resemble an emergency room attendant. |
| 16:19.27 | leifmadsen | jaytee: indeed :) |
| 16:19.31 | chazzam | heh |
| 16:19.43 | i_heart_asterisk | leifmadsen...i don't think this bug is version specific |
| 16:19.56 | leifmadsen | I don't think you have proven it isn't ;) |
| 16:20.16 | i_heart_asterisk | it happens on version 1.6.2.13 |
| 16:20.18 | i_heart_asterisk | also |
| 16:20.19 | leifmadsen | i_heart_asterisk: this is not a bog |
| 16:20.21 | leifmadsen | bug* |
| 16:20.30 | leifmadsen | how is asterisk supposed to know your call is a transfer or just another call? |
| 16:20.43 | leifmadsen | the answer is, it can't -- if you need asterisk to know this is a transfer, you have to use the built in transfers |
| 16:20.59 | i_heart_asterisk | what about the REFER ? |
| 16:21.16 | leifmadsen | that is after dialplan has already happened |
| 16:21.17 | i_heart_asterisk | how can a phone make a transfer and asterisk not know about it? |
| 16:21.30 | leifmadsen | asterisk does know about it, but dialplan functionality has already happened |
| 16:21.39 | leifmadsen | there is nothing else to do dialplan/AGI wise |
| 16:21.59 | i_heart_asterisk | so, you are saying there is no fix? only built-in transfer work w/ AGI ? |
| 16:22.48 | leifmadsen | the only way to inform asterisk that a transfer is ABOUT to happen is with a built in transfer |
| 16:23.26 | leifmadsen | runs off for lunch |
| 16:23.29 | i_heart_asterisk | yes but why can't AGI be notified after transfer? |
| 16:23.37 | i_heart_asterisk | or after REFER ? |
| 16:23.48 | leifmadsen | because that's how asterisk was built -- there is no dialplan being triggered after the REFER, just briding |
| 16:23.51 | DelphiWorld | how come iPhone don't have flash? :-) |
| 16:23.52 | leifmadsen | bridging* |
| 16:24.33 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:24.37 | jaytee | why do they call them buildings when they're already finished? they should call them builts. |
| 16:24.44 | i_heart_asterisk | can there be something triggered after REFER ? |
| 16:25.06 | i_heart_asterisk | it seems to me that this should be addressed as many phones don't use asterisk's built-in transfer, and many pple use AGI w/ asterisk |
| 16:25.22 | jaytee | a Glock 9mm? maybe an avalanche in Vail? |
| 16:28.53 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:29.43 | *** part/#asterisk shortcircuit (~shortcirc@rosettacode.org) |
| 16:32.34 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 16:36.30 | DelphiWorld | what i need to install to make my gtalk channel support encription? |
| 16:39.39 | *** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 16:41.16 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 16:41.24 | Thorn | it looks like if I Queue() a call without Answer()ing it first it is only answered when a queue member answers it, is this intended behavior that can be relied upon? (1.8.2.3) |
| 16:43.50 | leifmadsen | DelphiWorld: you can't |
| 16:44.10 | leifmadsen | DelphiWorld: I don't know that Google Talk has encryption support at all |
| 16:44.12 | DelphiWorld | leifmadsen: so how do i connect to google talk |
| 16:44.21 | leifmadsen | use chan_gtalk |
| 16:44.24 | DelphiWorld | leifmadsen: gtalk support tls |
| 16:44.37 | leifmadsen | I don't understand your question then |
| 16:44.47 | leifmadsen | you use chan_gtalk to connect to Google Talk. |
| 16:45.00 | DelphiWorld | leifmadsen: i'm getting encription error in the log. |
| 16:45.27 | leifmadsen | shrugs |
| 16:45.30 | leifmadsen | you haven't shown anything |
| 16:45.37 | DelphiWorld | leifmadsen: while show. |
| 16:45.38 | leifmadsen | vague questions get vague answers |
| 16:46.38 | DelphiWorld | leifmadsen: :) |
| 16:47.10 | i_heart_asterisk | leifmadsen, there is no way to patch asterisk in order to notify AGI after a REFER? |
| 16:47.43 | leifmadsen | there are probably ways to patch asterisk, sure -- you just have to figure it out. It's probably easier to just monitor AMI for the transfers and then perform an action based on that |
| 16:47.55 | leifmadsen | i_heart_asterisk: if there is a way to patch it, that would be a feature request |
| 16:51.06 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 16:51.18 | DelphiWorld | ohhhh drmessano-lt |
| 16:51.38 | i_heart_asterisk | why can't digium accept that this is a bug in asterisk |
| 16:51.58 | i_heart_asterisk | many phones don't use asterisk's built-in transfer, and many pple use AGI w/ asterisk |
| 16:52.54 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 16:53.51 | chazzam | i_heart_asterisk: he said it would be a feature request, not a bug |
| 16:55.59 | *** join/#asterisk guilhermebr (~Guilherme@187.114.232.71) |
| 16:57.48 | leifmadsen | i_heart_asterisk: what you're describing is core asterisk functionality and just how things work. There is no dialplan or other functionality after the REFER and channels are bridged, so there is nothing else that can happen there to notify the AGI that the transfer happened. if you need that information you need to monitor for the transfers via the AMI just like all other applications. |
| 16:57.58 | leifmadsen | you saying it is a bug doesn't make it so |
| 16:58.49 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 17:00.24 | i_heart_asterisk | this may not be a core problem in asterisk but it is a problem that should be addressed, making it a feature request will just put it on the back burner and probably never be implemented. |
| 17:01.15 | Thorn | what would be the best way to log things like queue waiting time, talking time etc into a database? |
| 17:01.17 | leifmadsen | this is not a bug |
| 17:01.41 | leifmadsen | Thorn: you could use cdr_adaptive_odbc and write information that way into the DB via CDRs |
| 17:01.51 | leifmadsen | alternatively you could take a look at CEL (included in 1.8) |
| 17:02.10 | i_heart_asterisk | leifmadsen, you believe this is not a problem w/ asterisk's functionality? |
| 17:02.15 | leifmadsen | Thorn: also check out queue_log to see if there is enough information being logged there for your use. CHeck out the chapter on ACD in the new book |
| 17:02.16 | leifmadsen | ~newbook |
| 17:02.16 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 17:02.32 | leifmadsen | i_heart_asterisk: I believe this is "just how it works" and is not a bug |
| 17:03.06 | Thorn | btw, queue_log doesn't exist here, does it need to be enabled? |
| 17:03.11 | i_heart_asterisk | i understand it is "just how it works" but it is a serious limitation and "how it works" should be fixed |
| 17:05.02 | Thorn | (I have master.csv and messages but not queue_log, 1.8.2.3) |
| 17:05.04 | leifmadsen | i_heart_asterisk: that is the definition of feature request that you're saying, which means you're welcome to provide a patch that addresses the deficiency. If you are unable to provide a patch for said functionality you are welcome to hire a consultant. Beyond that, there is nothing else to be done here. |
| 17:05.33 | leifmadsen | Thorn: http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id289009 |
| 17:08.58 | i_heart_asterisk | liefmadsen, can you at least point me in the direction of where the change in code might occur? |
| 17:09.40 | leifmadsen | I'm not a programmer |
| 17:09.45 | leifmadsen | so I can't help you there |
| 17:10.26 | i_heart_asterisk | i see... well thanks for nothing then |
| 17:11.08 | leifmadsen | glad I could help |
| 17:11.13 | Thorn | no luck with queue_log :( it was not disabled in logger.conf, I set it queue_log=yes, queue_log_to_file=yes explicitly but still no log file |
| 17:11.34 | i_heart_asterisk | you mean couldn't help |
| 17:12.19 | leifmadsen | i_heart_asterisk: I guess my sarcasm wasn't thick enough |
| 17:12.39 | jaytee | I wonder if his first name is "Richard"? |
| 17:12.47 | drmessano-lt | Not as thick as his "thanks for nothing" |
| 17:12.50 | drmessano-lt | But close |
| 17:13.13 | drmessano-lt | Damn you for not being a programmer, you unhelpful SOB! (or something) |
| 17:13.13 | jaytee | because he sure comes across as a "Dick" |
| 17:13.21 | nestAr | lol |
| 17:13.27 | i_heart_asterisk | this is precisely why asterisk is going to fall by the wayside to freeswitch |
| 17:13.35 | drmessano-lt | lol |
| 17:13.43 | nestAr | perhaps it's time for a change of nick |
| 17:13.47 | drmessano-lt | Because Leif isn't coding? |
| 17:13.47 | nestAr | just sayin' |
| 17:14.01 | jaytee | so are you going to change your nick to i_heart_freeswitch ? |
| 17:14.05 | i_heart_asterisk | what does leif do? salesmen for digium? |
| 17:14.22 | drmessano-lt | No, he wrote the books you should have read |
| 17:14.37 | jaytee | and probably does more in a morning than you do all week |
| 17:14.43 | i_heart_asterisk | i don't need to read about asterisk, i already know it's broken |
| 17:14.46 | leifmadsen | I'm the release manager and primary bug marshal for Asterisk, consultant, and author |
| 17:15.19 | jaytee | where is that highway sign when you need it :-) |
| 17:16.36 | nestAr | lotta hate |
| 17:16.56 | *** join/#asterisk Lensman (~Lensman@169.130.124.38) |
| 17:17.02 | i_heart_asterisk | remember, i heart asterisk...it hurts me to see thses flaws and nothing done bout it |
| 17:17.07 | drmessano-lt | i_heart_asterisk, I would say if Asterisk were to fall, it would be due to the lack of community involvement, such as when someone finds a missing/incomplete/poorly implemented feature in Asterisk and rather than attempting to be part of the solution and contribute to fixing it, they just complaint and troll |
| 17:17.31 | i_heart_asterisk | drmessano-lt, i am more than willing to put in time for this fix |
| 17:17.36 | drmessano-lt | So do it |
| 17:17.56 | i_heart_asterisk | i already asked leif where i could look, but somehow he doesn't code |
| 17:17.57 | drmessano-lt | Leif is not a coder.. don't bash him for not being helpful.. Find someone who can help and work towards a solution |
| 17:18.02 | drmessano-lt | He doesn't |
| 17:18.11 | drmessano-lt | He told you he doesn't |
| 17:18.23 | i_heart_asterisk | ok, does anybody code in here? |
| 17:18.33 | drmessano-lt | So find someone who does.. Put up a bounty somewhere. Hire a consultant |
| 17:18.44 | leifmadsen | i_heart_asterisk: I'm sorry I can't tell you it's like 1369 of chan_sip.c that you need to add if (im_awesome == 1) { do_awesome_stuff(1); } |
| 17:19.01 | drmessano-lt | Honestly, I think you owe him an apology |
| 17:19.18 | leifmadsen | I've already moved on and have a new girlfriend, so it's not necessary |
| 17:19.20 | i_heart_asterisk | i'm not going to hire a consultant... where is no development community |
| 17:19.24 | jaytee | lol |
| 17:19.45 | drmessano-lt | If you check the bug tracker, the development community is alive and well |
| 17:19.49 | i_heart_asterisk | digium just wants me to pay for a fix |
| 17:19.55 | drmessano-lt | lol |
| 17:20.04 | i_heart_asterisk | bugs = job security |
| 17:20.15 | drmessano-lt | Have you ever looked at the bug tracker? |
| 17:20.16 | n3hxs | BS |
| 17:20.39 | leifmadsen | i_heart_asterisk: no they don't, and you can't hire Digium to program your feature request either. That's what other companies do though. |
| 17:20.49 | n3hxs | thinks heart is misspelled... hate. |
| 17:20.51 | leifmadsen | I'm sorry your issue is not as important to resolve as the other issues. |
| 17:21.04 | leifmadsen | s/issue/feature request/ |
| 17:21.06 | jaytee | there's always barber college |
| 17:22.10 | drmessano-lt | You could have already posted a bounty somewhere |
| 17:22.17 | leifmadsen | i_heart_asterisk: you're welcome to describe your problem on asterisk-users to get a second opinion on your issue. It's entirely possible I'm wrong, or misunderstand your situation and what you're trying to do, and it really is a bug, in which case your issue could be reopened. But continuing to pout about it in here is not all that productive. |
| 17:23.20 | i_heart_asterisk | liefmadsen you are not wrong, you are correct that the dialplan is not executed after the REFER, the problem is it should |
| 17:24.05 | leifmadsen | then you're welcome to contribute that particular functionality, or find another way to work around it (such as monitoring for transfers via the AMI as I've suggested previously) |
| 17:24.55 | i_heart_asterisk | indeed, i will see about using AMI, just saddens me nobody agrees this is a seriously flaw |
| 17:25.34 | p3nguin | If no one else ever had a problem with it, maybe the severity isn't as great as you think it is. |
| 17:25.40 | jaytee | not a serious flaw as most installations don't require that functionality. yours is probably a rare case. |
| 17:25.58 | leifmadsen | I'm not sure what else I can say, so I'm not going to say anything else |
| 17:26.01 | i_heart_asterisk | jaytee, anybody using AGI for dialplan and polycoms will experience this problem |
| 17:26.04 | leifmadsen | goes back to triaging issues |
| 17:26.56 | jaytee | i_heart_asterisk, you could disable the softkeys for transfer on the Polycoms and use feature keys but I'm not sure the AGI would accomodate that either. |
| 17:27.16 | *** join/#asterisk cVsup (~cVsup@189.83.218.188) |
| 17:27.21 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 17:27.28 | cVsup | i have xfo interface as X100P |
| 17:27.35 | drift- | can you have multiple g729 cards in same server? |
| 17:27.46 | cVsup | in my country the caller ID signal as is DTMF |
| 17:28.02 | cVsup | i can use Caller ID? |
| 17:28.15 | cVsup | with X100P: |
| 17:28.17 | cVsup | ? |
| 17:28.29 | i_heart_asterisk | jaytee, i'm considered that but i'm not sure i want to disable the xfer softeys on the polycoms as our customers would not like adjusting to differences |
| 17:29.03 | leifmadsen | drift-: I think so, but that is probably a better question for Digium sales/support |
| 17:30.11 | JerJer | i_heart_asterisk: hopefully when you mention AGI you really mean FastAGI |
| 17:30.23 | JerJer | because, you know, regular AGI does not scale |
| 17:30.27 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 17:30.46 | drmessano-lt | drift-, multiple transcoder cards? You've managed to impress me. |
| 17:30.54 | drift- | yes |
| 17:30.56 | drmessano-lt | That just happened |
| 17:31.04 | drift- | lol :) |
| 17:31.23 | drift- | instead of making 4 servers id rather have 2 servers with 2 cards |
| 17:31.32 | drift- | trying to beable to get 300 concurent calls |
| 17:31.38 | jaytee | why the -lt in the nick now? |
| 17:31.48 | drmessano-lt | I am on my laptop |
| 17:31.50 | drmessano-lt | At work |
| 17:31.57 | drmessano-lt | My ILLEGAL LINUX MACHINE |
| 17:32.00 | jaytee | aha! makes perfect sense now that I think about it |
| 17:32.09 | drmessano-lt | stares at the Windows boxes all around |
| 17:32.18 | leifmadsen | drift-: like I said, I'd just ask Digium since they would know for sure if that scenario is supported |
| 17:32.19 | jaytee | hahaha, illegal linux machine? |
| 17:32.30 | drmessano-lt | We don't speak of the Penguin around here |
| 17:32.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:32.53 | drmessano-lt | Linux = Evil, or so says the Domain Controller in the basement |
| 17:33.05 | leifmadsen | the comptroller? |
| 17:33.17 | leifmadsen | goes to make another batch of coffee as his right eye stopped twitching |
| 17:33.22 | drmessano-lt | lol |
| 17:33.55 | jaytee | hahaha, sounds like here. We have an Asterisk phone system I setup, Polycom phones and I've replaced our POS(not point of sale) Netgear Firewall/Router with Pentium D box running Vyatta 6.1 and yet all my coworkers are Windows developers who still bash linux. |
| 17:34.17 | file | ugh Vyatta |
| 17:34.28 | file | I was evaluating solutions for my home network router and signed up for documentation access |
| 17:34.28 | drmessano-lt | "Why does your screen look all cool and 3D-ish?" "I am running Linux" "Oh, brb, getting the torches and pointy sticks" |
| 17:34.33 | file | they won't stop emailing and calling me |
| 17:34.45 | jaytee | :-) |
| 17:34.53 | jaytee | persistent buggers |
| 17:34.53 | file | ended up going with pfSense |
| 17:34.54 | *** join/#asterisk Alborracho (~chatzilla@186.6.152.45) |
| 17:35.12 | drmessano-lt | I can't use anything BSD based |
| 17:35.56 | JerJer | i like to roll my own. PC Engines ALIX board+Voyage Linux |
| 17:36.25 | JerJer | i have also used routerboards |
| 17:37.46 | file | I use a dual core Atom 1U server |
| 17:37.49 | file | it works well. |
| 17:38.54 | jaytee | I built a mini-itx box for under 300 bucks with an Intel Atom D510MO board |
| 17:39.25 | drmessano-lt | I am building a couple systems with SuperMicro 1U boxes. Right around $300 each |
| 17:39.27 | jaytee | bit overkill but I've used it to test both Vyatta and Asterisk on it. |
| 17:40.08 | jaytee | and drmessano-lt, I'm really satisfied with Flowroute and most importantly, so is my boss. |
| 17:41.50 | drmessano-lt | Getting ready to overhaul a $1.25 Million Mobile Command vehicle and install Atom based Ubuntu desktops, an Asterisk system, and moving from POTS/Cell phones to Flowroute |
| 17:42.23 | paulc | Mobile Command sounds kinda.. sexy? what's it for? |
| 17:42.33 | Alborracho | hi is there any other way to change the volumen in asterisk, i have a te420b connected through T1s and the audio is really high, ive changed the ss7.conf rxgain and txgain but everything keeps same |
| 17:43.17 | leifmadsen | Alborracho: you could try the VOLUME() function |
| 17:43.21 | Alborracho | hmm |
| 17:43.31 | Alborracho | in extension.conf right? |
| 17:43.35 | leifmadsen | right |
| 17:43.38 | leifmadsen | core show function VOLUME |
| 17:43.45 | Alborracho | ok let me do a test |
| 17:43.57 | jaytee | and changes to gain settings won't take effect until you restart DAHDI |
| 17:44.00 | drmessano-lt | It's a mobile interoperability device. Houses a dispatch center, conference room, a dozen or so radios we can interconnect, multiple forms of connectivity, etc |
| 17:44.08 | leifmadsen | (VOLUME() is probably in Asterisk 1.6.2 and above) |
| 17:44.23 | drmessano-lt | paulc, it's a big radio/telephony wirenut |
| 17:44.47 | leifmadsen | I have an awful router from Rogers which is made by SMC... I should probably do something like what file did and use something embedded to be a router. |
| 17:44.56 | leifmadsen | some day... |
| 17:45.08 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:45.15 | jaytee | I priced out nano-itx and pico-itx systems but the costs are higher for parts than mini-itx systems. |
| 17:45.41 | paulc | drmessano-lt: Sounds cool.. any pics? |
| 17:45.50 | Alborracho | do you know what is the maximun and minimum rx and tx gain? |
| 17:46.52 | drmessano-lt | paulc, not sure what we have online. Let me see |
| 17:48.01 | drmessano-lt | paulc: http://www.columbiacountyga.gov/Index.aspx?page=3205 |
| 17:48.28 | drmessano-lt | A lot of that info the OLD/Current truck configuration. 90% of the comm equipment is being ripped out thanks to our most recent grant |
| 17:49.36 | paulc | That's pretty cool.. so you drive it to wherever and then... satellite uplink for data? |
| 17:50.05 | jaytee | Alborracho, with zaptel you could use -100 to 100 but it was always recommended not to go over -11 to 11. |
| 17:50.21 | jaytee | not sure if that's changed in DAHDI |
| 17:51.07 | Alborracho | thx |
| 17:52.26 | drmessano-lt | Well, the current Satellite data system is b0rk3d. It never really worked. I implemented EVDO, as the truck is generally outside of the hot zone where it's safe and some chance of a cellular network existing.. but that wont always be the case. We're getting a new system with this grant. We also have some Wifi clients to grab Wifi from wherever.. even free Wifi from a McDonalds if that's all that's around lol |
| 17:53.55 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-40.nayatel.pk) |
| 17:54.27 | paulc | imagines the truck parked outside Starbucks.. coffees for everyone and a shot of free wifi please :) |
| 17:54.38 | drmessano-lt | lol |
| 17:54.47 | mechbangirc | my AMD() does not work no matter what i do. (2500|1500|300|5000|120|50|2|256). it always consider me machine |
| 17:55.11 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 17:55.27 | paulc | mechbangirc: You're a machine. And you're becoming self aware? ;-) |
| 17:55.51 | mechbangirc | paulc: thanks for leaving me this msg |
| 17:56.10 | drmessano-lt | We've used Wifi from a restaurant during a test deployment. It was a nice test of "when all else fails". |
| 17:57.42 | paulc | drmessano-lt: So are you doing away with satellite totally? or still have it as the ultimate fall-back position? |
| 17:58.48 | drmessano-lt | With this new grant, satellite will again be the primary method of connectivity. EVDO follows that, and throwing out the pelican case with the Wifi client will be a distant third. |
| 17:59.18 | *** join/#asterisk MikeH (~mike@86.63.17.141) |
| 18:01.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 18:01.26 | paulc | Nice. Cool project to work on. And slightly more exciting than my current project, perhaps.. although it has its moments too.. |
| 18:02.35 | drmessano-lt | Which is? |
| 18:07.27 | p3nguin | Secret. Apparently. |
| 18:12.14 | Alborracho | jaytee: i changed txgain to -10 , then restart dahdi, but it had no effect |
| 18:12.31 | Alborracho | its really high |
| 18:13.31 | paulc | Currently working on Asterisk as a mechanism for connecting people who want to pay to talk, with people who get paid to talk.. in a sort of.. "ooh yeah baby, that's right, mmm yes" kind of way |
| 18:13.36 | Alborracho | i think its a hardware problem with my card... |
| 18:14.37 | drmessano-lt | ohhh |
| 18:15.27 | drmessano-lt | The village people had a song about that |
| 18:18.23 | Thorn | leifmadsen: thanks for the new book, it's excellent |
| 18:23.46 | *** join/#asterisk Ool (~Ool@unaffiliated/ool) |
| 18:24.10 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 18:24.56 | wwalker | strange voicemail behavior: https://gist.github.com/814886 asterisk says its recorded, then says it's saved, then there is no message.... |
| 18:25.16 | citywok | wwalker: make sure asterisk has write permissions to /var/spool/asterisk? |
| 18:25.21 | fenrus | does asterisk have permissions to save the files? |
| 18:25.22 | fenrus | :D |
| 18:26.00 | wwalker | asterisk is running as root (taking over an existing installation, I'll fix perms when they have basic phone functions) |
| 18:27.59 | *** join/#asterisk OpenSourceWay (~OpenSourc@AToulouse-257-1-54-48.w90-5.abo.wanadoo.fr) |
| 18:28.34 | *** join/#asterisk ccesario (~ccesario@189-29-59-116-ac.cpe.vivax.com.br) |
| 18:37.09 | *** join/#asterisk andrei66 (~chatzilla@AMontpellier-151-1-8-211.w92-143.abo.wanadoo.fr) |
| 18:37.26 | andrei66 | Hi all |
| 18:40.53 | *** join/#asterisk andrei66 (~andrei66@AMontpellier-151-1-8-211.w92-143.abo.wanadoo.fr) |
| 18:40.59 | andrei66 | hi all |
| 18:41.07 | *** part/#asterisk OpenSourceWay (~OpenSourc@AToulouse-257-1-54-48.w90-5.abo.wanadoo.fr) |
| 18:42.19 | fenrus | hi |
| 18:43.03 | wwalker | I've run strace on asterisk, and I see asterisk rename() the recording into INBOX and the msg00000.txt file into INBOX. then they are gone :( |
| 18:45.43 | *** join/#asterisk stefmtl (~stef@stef.istop.com) |
| 18:46.45 | stefmtl | Hi. I have a question: is it possible to use some Digium Hardware to replace a ADSL modem) ? |
| 18:47.01 | leifmadsen | no sir |
| 18:47.14 | leifmadsen | sangoma makes an ADSL modem |
| 18:47.27 | andrei66 | i think you can only use a digium card for 56k or fax |
| 18:47.30 | leifmadsen | (it's currently sitting in a box in my drawer if you're interested in buying it) |
| 18:47.34 | stefmtl | ok thanks leif |
| 18:50.50 | *** part/#asterisk Ool (~Ool@unaffiliated/ool) |
| 18:52.49 | *** join/#asterisk ixx (~ixx@cpe-173-174-60-240.austin.res.rr.com) |
| 19:04.31 | *** join/#asterisk moy (~moy@h96-61-34-34.mdsnwi.tisp.static.tds.net) |
| 19:04.52 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 19:05.56 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 19:06.00 | d_preston215 | Is there a way to use a speed dial to transfer a call to a queue? |
| 19:06.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v011-239.mobile.uci.edu) |
| 19:07.02 | Thorn | what does queue_log in extconfig.conf do? |
| 19:08.18 | citywok | Thorn: I've never done it, but i'm guessing it's so you can write queue_log events to a database instead (or in addition to?) the flat file in /var/log/asterisk |
| 19:08.50 | andrei66 | I have a problem : i use asterisk on a ubuntu pc for 3 SIP phones |
| 19:09.15 | Thorn | all the other things in extconfig.conf are for reading from databases, aren't they? |
| 19:09.16 | andrei66 | and i use a external SIp provider (asterisk registers as client) to dial outside |
| 19:09.50 | andrei66 | but i can hear, but who i'm calling can't hear me |
| 19:10.57 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 19:11.13 | paulc | d_preston215: You mean like have a single key on the phone that transfers the call to a queue/extension? |
| 19:13.54 | d_preston215 | Yes. |
| 19:13.57 | d_preston215 | Exactly that. |
| 19:14.31 | citywok | Thorn: i know sippeers reads & writes. lol. |
| 19:15.39 | Thorn | yes looks like it, thanks |
| 19:18.23 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 19:19.26 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:20.04 | Dovid | hello all |
| 19:21.12 | luckman212 | is the correct place to submit a bug to Digium https://issues.asterisk.org/main_page.php ? |
| 19:21.29 | d_preston215 | Any ideas @paulc |
| 19:22.25 | rgagnon | luckman: yes, but please search to see if it may have already been submitted |
| 19:24.27 | luckman212 | rgagnon: the bug I want to file is about mpg123 dying after a moh reload when moh is using res_timing_dahdi .. how would I even search for that? |
| 19:25.08 | rgagnon | very carefully? |
| 19:25.55 | luckman212 | slaps himself on the forehead |
| 19:25.59 | luckman212 | of course |
| 19:26.07 | rgagnon | I think they just care that people don't duplicate things without at least checking |
| 19:27.11 | rgagnon | you can search from here https://issues.asterisk.org/view_all_bug_page.php maybe for "mpg123" |
| 19:28.49 | *** join/#asterisk _Sam-- (~sam@unaffiliated/sam--/x-573746) |
| 19:28.51 | *** join/#asterisk jkroon (~jkroon@dsl-242-3-152.telkomadsl.co.za) |
| 19:31.16 | leifmadsen | luckman212: or look through the list of issues filed against Resources/res_musiconhold etc. |
| 19:31.31 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 19:33.14 | luckman212 | leifmadsen: thanks, I'm in there |
| 19:34.43 | luckman212 | is it really possible that there are only 5 open issues for res_musiconhold ? |
| 19:34.43 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 19:35.18 | raden_work | anyone seen naikorvek ? |
| 19:35.56 | leifmadsen | luckman212: yes that is certainly possible -- it'd be ideal to have zero open issues against res_musiconhold |
| 19:36.05 | leifmadsen | ~seen naikorvek |
| 19:36.10 | infobot | i haven't seen 'naikorvek', leifmadsen |
| 19:41.13 | leifmadsen | M18737 |
| 19:41.18 | leifmadsen | seanbright: that might be something you can look at if you're bored :) |
| 19:41.22 | leifmadsen | bah! |
| 19:41.24 | leifmadsen | wrong room |
| 19:44.48 | paulc | d_preston215: delayed reply, sorry... I think it depends on the phones you're using as to whether it's possible or not. What phones are you using? |
| 19:45.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:46.07 | *** join/#asterisk ks3_ (~ksandy@74.203.195.1) |
| 19:46.58 | d_preston215 | Cisco 7960 |
| 19:47.32 | d_preston215 | I'm going to try to just do the transfer feature code and extension to see if that works. |
| 19:50.25 | paulc | d_preston215: Not sure it'll work but see how you go.. I think a guy in here did something similar with Aastra phones, they seem to be a bit more flexible and nice for Asterisk integration. Let me know if you get it working though :) |
| 19:50.33 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 19:50.52 | Micc | Why does asterisk think another asterisk server has VAD enabled? |
| 19:51.03 | *** join/#asterisk corygstuart (~corygstua@mysterymachine.dis.anl.gov) |
| 19:51.42 | corygstuart | Hello. I'm going to build an asterisk server and was looking for hardware recommendations. What's worked for you? |
| 19:54.08 | Micc | I'm guessing the comfort noise message in asterisk is showing the wrong IP or getting it confused with one of the phones. They only phones on the system are SPA942 but I can't find CNG or VAD anywhere in the settings of the phone to disable it. |
| 19:54.09 | paulc | corygstuart: I've done stuff with DL-385s and a few proof of concepts using HP desktop PCs.. I hear Dells are pretty good.. there's a hardware page on the wiki too |
| 19:54.14 | paulc | ~hardware |
| 19:54.14 | infobot | extra, extra, read all about it, hardware is http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module. |
| 19:54.36 | paulc | that wasn't quite the link I was thinking of... |
| 19:54.39 | Micc | I think I remember asking this question about comfort noise before and the answer was its broken and ignore it I think. |
| 19:55.41 | d_preston215 | Yeah, feature code + extension on a speed dial didnt work. |
| 19:56.12 | corygstuart | Thanks, paulc. I'll jump on the wiki and take a look. |
| 19:58.36 | d_preston215 | Its probably because the speed dial has no input space between the feature code and extension, so asterisk doesn't have enough time to process it. |
| 20:00.41 | corygstuart | OK. I'm seeing interface cards, has anyone put together any inexpensive servers (e.g. comparable to a SwitchVox)? |
| 20:05.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 20:06.21 | *** join/#asterisk jkroon (~jkroon@dsl-242-3-152.telkomadsl.co.za) |
| 20:11.35 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 20:16.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:19.22 | devmod | when was 'same =>' added? |
| 20:27.21 | leifmadsen | devmod: 1.6.2 |
| 20:27.35 | leifmadsen | <3 same => |
| 20:27.51 | devmod | leifmadsen, yup its awesome |
| 20:27.53 | *** join/#asterisk superbeef (~Lane.Jenn@74-120-209-234-rev.redanvil.net) |
| 20:28.05 | leifmadsen | I use it all the time |
| 20:28.11 | leifmadsen | I think it came from a suggestion I had years ago :) |
| 20:28.21 | WIMPy | always forgets te n, when using it. |
| 20:28.29 | WIMPy | the |
| 20:28.36 | leifmadsen | I've done that sometimes, but most of the time I'm pretty good |
| 20:28.52 | drmessano-lt | loves same => |
| 20:29.04 | leifmadsen | loves drmessano-lt |
| 20:29.11 | drmessano-lt | Ohhh <3 |
| 20:29.35 | leifmadsen | unfortunately it's 'not the truth' monday |
| 20:29.55 | drmessano-lt | Yes, and I could love you too because that would create a goto loop |
| 20:29.57 | devmod | I'm testing someone's repo that supposedly fixes an issue and it seems its based on 1.6.0 . Where are patches committed to once approved? |
| 20:29.59 | drmessano-lt | Sorry :( |
| 20:30.06 | drmessano-lt | couldnt |
| 20:30.15 | drmessano-lt | as in, I couldnt mess that joke up and just did |
| 20:30.21 | _Corey_ | hmmm, that same thing will save a few keystrokes... :) |
| 20:30.58 | leifmadsen | devmod: just the branches that are being supported |
| 20:31.04 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:31.06 | leifmadsen | (1.4, 1.6.2, 1.8, trunk at the present time) |
| 20:31.07 | drmessano-lt | same is cool because it reminds me of having someone ask you the same question over and over and over again and getting annoyed |
| 20:31.36 | drmessano-lt | "What extension? SAME" "what extension? SAME" "ok, what about this one? SAME" |
| 20:31.39 | leifmadsen | devmod: the branch is being done against 1.6.0 just because that developers customer is using 1.6.0, and not because 1.6.0 (or 1.6.1) is supported |
| 20:31.54 | leifmadsen | drmessano-lt: what's even more confusing is the 's' priority :) |
| 20:32.04 | leifmadsen | same => s,NoOp() |
| 20:32.18 | devmod | leifmadsen, ohh ok. That is what I thought but then this ticket is set to ready for testing while the patch proposed is against 1.6.0 so i got confused |
| 20:32.20 | drmessano-lt | assplodes |
| 20:32.26 | leifmadsen | (although 's' is pretty much NEVER used, and would just be for something where you want to do s+101) |
| 20:32.37 | leifmadsen | devmod: ya well oej is confusing sometime s:) |
| 20:32.53 | leifmadsen | devmod: you'd have to request the developer perhaps provide patches for 1.6.2 or whatever branch you're using |
| 20:33.33 | *** join/#asterisk Mw3 (~mw3@mw3.hu) |
| 20:33.49 | voxter | Anyone know of an ADSL Modem + Router combo that supports handing out DHCP Option 66? |
| 20:34.08 | devmod | leifmadsen, ohh ok haha it's fine I will just test that - Just wondering |
| 20:34.22 | leifmadsen | no worries |
| 20:34.37 | leifmadsen | voxter: Sangoma ADSL modem on Linux? :) |
| 20:35.01 | voxter | leifmadsen: haha :) Looking for a small all in one unit, like a linksys-sized device |
| 20:35.16 | leifmadsen | voxter: ADSL modem in a Soekris? :) |
| 20:35.37 | voxter | leifmadsen: oh! also something that costs less than a linksys wrt + thompson adsl modem already does. |
| 20:35.39 | leifmadsen | voxter: PS: ping me about a beginner DJ set list |
| 20:35.54 | voxter | leifmadsen: sure! i've got a good list for ya. |
| 20:36.20 | leifmadsen | voxter: learning Ableton Live with an X-Session Pro and Launchpad and need to practice mixing but would love a set list that I know already "goes together" |
| 20:36.22 | p3nguin | When a call to my Google Voice phone number never reaches Asterisk, what should I look at to fix it? Outbound calling via Google Voice works fine. |
| 20:37.10 | voxter | leifmadsen: you know ive been meaning to learn how to use ableton to dj. I normally use torq. I'll hit ya up someplace else and we can chat about it some more. |
| 20:37.45 | adyn | ~pastbin |
| 20:39.02 | leifmadsen | ~pb |
| 20:39.02 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:39.05 | adyn | thank you |
| 20:42.08 | *** join/#asterisk Lensman (~Lensman@169.130.124.38) |
| 20:59.25 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 21:10.11 | adyn | I'm having an issue where I have a several pri's terminate into 2 asterisk boxes which then send the calls to their individual destination pbx's on one pbx randomly as far as I can tell it just stops taking calls I'm thinking its some sort of time out but I can't figure out where, none of the searches I've tried have turned up anything meaningfull, here logs and my iax.conf snippets: http://asterisk.pastey.net/145822 |
| 21:11.28 | adyn | also when I try core stop now asterisk just locks up and I have to kill the process manually in order to restart it |
| 21:11.51 | *** join/#asterisk solokkhz (solokkhz@bzq-79-179-198-191.red.bezeqint.net) |
| 21:11.51 | adyn | Asterisk version is 1.6.2.16.1 |
| 21:12.42 | GTXComm | If I want to dail 7 diget dialing and put in 913 without dialing it, how would I go about doing that, if anyone knows off the top of their head, otherwise, I will continue to look IE. _XXXXXX => dail(SIP/provider/${EXTEN}); |
| 21:13.18 | GTXComm | Area code |
| 21:14.06 | GTXComm | AEL |
| 21:14.20 | WIMPy | My most favourite dialplan issue. |
| 21:14.35 | leifmadsen | GTXComm: exten => _NXXXXXX,1,Dial(SIP/provider/913${EXTEN}) |
| 21:14.51 | leifmadsen | you'll have to convert to AEL yourself because I don't use it (but it's obvious what I just did) |
| 21:14.56 | GTXComm | Found it. |
| 21:15.13 | GTXComm | Thanks Lief. |
| 21:15.42 | WIMPy | And now do it fo a number with unknown length. |
| 21:16.52 | nestAr | it's just _X. isn't it? |
| 21:17.27 | GTXComm | worked. |
| 21:17.42 | WIMPy | Ok, now the final one: Do it without a timeout. |
| 21:18.33 | leifmadsen | progressinband=yes I think |
| 21:18.47 | leifmadsen | unless I misunderstand the question :) |
| 21:18.59 | WIMPy | I think you did :-) |
| 21:20.01 | leifmadsen | I'm also trying to figure out why I can't get my VMs to start in KVM |
| 21:20.04 | WIMPy | I'd like to add the area code for interactively dialled numbers of unknown length in order to be able to use enum and to avoid hacks in dundi. |
| 21:20.35 | WIMPy | But as far as I can see that's not possible without timeout. |
| 21:21.02 | leifmadsen | you mean pattern matching as you dial the digits? (I'm trying to remember what that is called) |
| 21:21.02 | WIMPy | And it makes it impossible to interactively match dundi entries. |
| 21:21.13 | leifmadsen | where each DTMF tone is sent as an INFO msg |
| 21:21.13 | WIMPy | overlap dialling |
| 21:21.17 | leifmadsen | that's the one |
| 21:21.18 | WIMPy | yes |
| 21:21.27 | leifmadsen | is that what you mean? |
| 21:21.45 | leifmadsen | you'd have to send a DUNDi request for each overlap dialed number.... |
| 21:21.52 | WIMPy | As far as I can see that would require the possibility to modify EXTEN and re-enter the dialplan. |
| 21:22.06 | leifmadsen | otherwise you need to use something like precaching in dundi |
| 21:22.20 | WIMPy | Yes, that works very well. As lond as all your numbers are within the same area code. |
| 21:22.28 | WIMPy | long |
| 21:23.18 | WIMPy | i.e. if I define my extensions both with and without area code. |
| 21:23.46 | WIMPy | So I need to have one dundi context per area code to avoid false matches. |
| 21:24.21 | WIMPy | But that's not possible for public ENUM. |
| 21:40.05 | _Sam-- | anyone looking for a project? i could use a hand trying to move our 1.4 asterisk from a physical server here at our location, to a rackspace cloud server |
| 21:40.43 | _Sam-- | should be like 1-3 hours max. but not for me, apparently. |
| 21:40.51 | p3nguin | _sam--: Do you mean a VPS? |
| 21:40.58 | _Sam-- | p3nguin: yeap |
| 21:41.18 | _Sam-- | both machines (physical and VPS) are debian |
| 21:41.46 | p3nguin | _sam--: How soon do you need to have it completed? |
| 21:42.03 | _Sam-- | no huge rush. just a longer term goal i'd like to accomplish |
| 21:43.10 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:43.21 | p3nguin | _sam--: I'll PM you if it's okay. |
| 21:43.30 | _Sam-- | pls do |
| 21:45.52 | *** join/#asterisk Da-Geek (~Da-Geek@90.155.74.11) |
| 21:46.32 | *** part/#asterisk Da-Geek (~Da-Geek@90.155.74.11) |
| 21:51.52 | *** join/#asterisk mawhii (~mawhii@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 21:52.24 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:56.26 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 21:58.47 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 21:59.04 | WIMPy | wonders if WaitExten could be modified and abused... |
| 21:59.20 | *** join/#asterisk darkskiez_ (~dz@62-50-199-254.client.stsn.net) |
| 21:59.47 | kuku | Any reason why an inbound call would go to fast busy after two rings when connected to a TDM400p, even when asterisk is not running ? |
| 22:00.42 | *** part/#asterisk joeyjones (~joeyjones@93.186.171.52) |
| 22:01.31 | Kobaz | System uptime: 30 weeks, 7 hours, 19 minutes, 12 seconds |
| 22:01.32 | Kobaz | mmm |
| 22:02.54 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:03.43 | killown | what's the best asterisk gui? |
| 22:04.00 | kuku | freepbx in my opinion |
| 22:04.45 | killown | kuku ok thank you |
| 22:05.06 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 22:06.04 | killown | kuku, it has an advanced call recording and true hot desk support? |
| 22:09.26 | leifmadsen | doubt it :) |
| 22:09.36 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 22:10.39 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:10.40 | killown | I'd like to know a complete IP PBX solution based on Asterisk |
| 22:11.08 | _Sam-- | killown: switchvox? |
| 22:13.20 | p3nguin | Your definition of complete is probably different from that of many others here. |
| 22:16.45 | Katty | hi leif |
| 22:17.45 | leifmadsen | yo |
| 22:17.57 | *** join/#asterisk powerunits (~as@116.71.187.93) |
| 22:19.08 | killown | I need an asterisk gui prefessional and for end users |
| 22:22.21 | powerunits | hi every one... |
| 22:22.34 | powerunits | im facing a issue on inbound |
| 22:22.37 | powerunits | http://pastebin.com/PCt4GrG7 |
| 22:22.48 | powerunits | this is my config file and asterisk CLI result |
| 22:22.55 | p3nguin | killown: Tried FreePBX? |
| 22:23.01 | p3nguin | powerunits: pastebin |
| 22:23.08 | p3nguin | powerunits: oops, you already did. |
| 22:23.09 | killown | p3nguin not yet |
| 22:23.17 | p3nguin | powerunits: sorry :/ |
| 22:23.22 | powerunits | :) no pro |
| 22:24.34 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 22:24.39 | p3nguin | powerunits: What is the exact problem you're having? |
| 22:25.02 | p3nguin | powerunits: I see a couple problems, but I doubt they are what you were meaning. |
| 22:25.16 | powerunits | i am unable to recive inbound calls |
| 22:25.28 | powerunits | when i dial my inbound number |
| 22:25.38 | powerunits | on asterisk it give me error msgs |
| 22:26.00 | p3nguin | powerunits: There is more to the problem than what I see in the pastebin. |
| 22:26.24 | powerunits | humm |
| 22:26.35 | powerunits | p3nguin: wht is that? |
| 22:27.09 | p3nguin | powerunits: Hold on, I'm trying to put the puzzle pieces together on my own. |
| 22:27.18 | powerunits | ok |
| 22:27.40 | p3nguin | powerunits: What is the "sipaccount" peer? |
| 22:27.53 | p3nguin | That's your ITSP? |
| 22:28.35 | powerunits | sorry wht is ITSP? |
| 22:29.06 | powerunits | this sip account from where i am doing outbound and want to recive inbound calls as well |
| 22:30.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:30.34 | p3nguin | ITSP means your provider. |
| 22:30.43 | powerunits | yes |
| 22:30.46 | powerunits | it is |
| 22:32.33 | p3nguin | powerunits: I would prefer to see your configuration more like this: http://pastebin.com/LFS58tFT |
| 22:32.39 | p3nguin | It is a much better idea. |
| 22:33.12 | citywok | powerunits: do you register your peer? |
| 22:33.31 | p3nguin | powerunits: After you make it like that, make another call to your phone number and show me the error from the CLI. |
| 22:34.00 | p3nguin | "core set verbose 4" before making the call |
| 22:34.32 | powerunits | let me see plz |
| 22:35.41 | powerunits | citywok: yes my peer is getting register. |
| 22:36.14 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 22:36.35 | adeel | has there been any major modifications in sip.conf from 1.4 to 1.8? |
| 22:36.41 | Tozz_ | yes |
| 22:37.16 | adeel | Tozz_, mind giving me a quick overview? |
| 22:37.17 | *** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net) |
| 22:37.22 | Tozz_ | or, the way SIP peers are identified has changed a little. So u might need fromuser= directives |
| 22:38.29 | adeel | that's nothing too major |
| 22:38.50 | Tozz_ | well, we've had issues with SIP peers not getting registered |
| 22:39.29 | adeel | seems like TCP/TLS peers are a bit touchy...i'm still using good old UDP peers |
| 22:39.47 | citywok | i use TCP but only for integration with OCS/Lync |
| 22:39.51 | p3nguin | powerunits: There's no indication of you registering to the ITSP based on the sip.conf you showed me. |
| 22:40.04 | p3nguin | And of course, there's no reason for the ITSP to register to you. |
| 22:43.16 | Thorn | where is the queue_log table format documented? I'm looking at https://issues.asterisk.org/view.php?id=17082 but res_odbc complains about column "time" which isn't in that example |
| 22:51.17 | powerunits | p3nguin: sip is registerd. i can show it to |
| 22:51.28 | powerunits | i have tried your example... but still same |
| 22:51.31 | powerunits | results |
| 22:51.35 | powerunits | on asterisk CLI |
| 22:52.30 | p3nguin | I understand that the results will be the same. That had very little to do with the changes I made for you. |
| 22:52.46 | p3nguin | I still need to see the CLI output from the failed call. |
| 22:53.01 | powerunits | ok let me show you.. |
| 22:53.27 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 22:54.31 | powerunits | http://pastebin.com/pPh4iPt7 |
| 22:58.42 | Thorn | how unique is uniqueid? can I use it as a database key and as a filename for call recordings? |
| 23:02.24 | citywok | Thorn: I do. It's Epoch time + an incremental call counter i think |
| 23:04.24 | Thorn | okay, thanks |
| 23:04.33 | powerunits | p3nduin: did u get any idea? |
| 23:07.29 | adeel | how is the support for the directrtpsetup option in sip.conf amongst * 1.4 - 1.8? is it reliable? |
| 23:07.59 | powerunits | http://pastebin.com/NHEFhbR4 |
| 23:08.40 | powerunits | i don't understand why on inbound call astierks CLI says username mismatch"??4 |
| 23:13.07 | Thorn | how do I get ${DIALSTATUS} into CDR? it seems like no dialplan commands are executed after Dial() (I have a NoOp there too) |
| 23:14.48 | p3nguin | powerunits: Where does this username exetech-verification come from? |
| 23:15.19 | powerunits | i have to SIP account register with ITSP |
| 23:15.27 | powerunits | this is 2nd SIP account |
| 23:16.28 | powerunits | http://pastebin.com/wfMdaSW8 |
| 23:16.41 | powerunits | this is my sip.conf general part setting |
| 23:16.51 | powerunits | plz check if any thing wrong here? |
| 23:18.54 | *** join/#asterisk dr00d (~rtp.aster@b27A5.static.pacific.net.au) |
| 23:19.35 | dr00d | hi - anyone in here able to answer a question about skype for asterisk ? |
| 23:20.00 | Thorn | looks like Fg is the answer |
| 23:20.18 | dr00d | whats fg ? |
| 23:20.33 | Thorn | it's not for you |
| 23:21.16 | dr00d | skype for asterisk isnt for me ? |
| 23:21.25 | Thorn | my message was not for you |
| 23:21.29 | dr00d | oh sorry |
| 23:22.27 | powerunits | p3nguin: any hint or idea? |
| 23:22.32 | powerunits | :( |
| 23:24.14 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:25.16 | killown | what is a better choice, freebpx or trixbox? |
| 23:25.25 | citywok | neither |
| 23:25.39 | dr00d | lol |
| 23:26.18 | dr00d | can anyone tell me where i have to put Dial(Skype/[<originator>@]<destination>) ? |
| 23:26.46 | citywok | your dial plan, where you want it to dial? |
| 23:27.06 | killown | citywok why neither? |
| 23:27.10 | mzb | killown: isn't that like asking the difference between Ubuntu and Firefox? |
| 23:27.43 | citywok | b/c you haven't given a list of clear requirements, which is something you need when making a decision :) |
| 23:28.01 | dr00d | yes - i dont understand where the dial plan is |
| 23:28.11 | mzb | one is an 'application', the other is a 'distro' (and I use both those terms loosely;)) |
| 23:28.24 | killown | mzb I need a gui asterisk tool for end users, easy to manager to do things like listen recorded calls and another things |
| 23:28.28 | citywok | yea, trixbox is a full OS+Asterisk+freepbx |
| 23:28.55 | mzb | killown: your preferred distro is??? |
| 23:29.00 | p3nguin | killown: Go with AsteriskNOW if you need a full OS with the tool included. Do not consider Trixbox if you value support. |
| 23:29.37 | killown | mzb, I have no preferred distro |
| 23:29.37 | p3nguin | powerunits: Your Asterisk is behind a NAT? |
| 23:29.39 | killown | p3nguin, ok |
| 23:30.06 | powerunits | no.. |
| 23:30.23 | powerunits | p3nguin: no its not |
| 23:30.23 | mzb | then AsteriskNOW is definitely for you ... personally I prefer building it all on Debian |
| 23:30.33 | citywok | debian ftw |
| 23:30.53 | p3nguin | powerunits: Then reconfigure your system for use without NAT. |
| 23:31.08 | mzb | I've written a script that installs from scratch on a Debian (based) distro |
| 23:31.31 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 23:31.34 | powerunits | humm let me try |
| 23:31.44 | powerunits | meah while can you look this form |
| 23:31.46 | powerunits | http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11169370.html |
| 23:31.57 | killown | mzb p3nguin thank you for the recommendation, I am downloading it right now |
| 23:32.01 | powerunits | this user has same issue llike my one |
| 23:32.15 | p3nguin | killown: AsteriskNOW is a really nice OS. |
| 23:32.21 | citywok | mzb, yep, same here. fully configures a vanilla debian install with asterisk, freepbx, my custom freepbx scripts, all the tools to autodiscover aastra phones |
| 23:32.36 | killown | p3nguin, this is a linux based os? |
| 23:32.48 | citywok | killown: no, it's windows. |
| 23:33.11 | citywok | okay... mabye it isn't :P |
| 23:33.36 | powerunits | p3nguin: i have disable nat from sip.conf genral part |
| 23:33.38 | killown | :( |
| 23:33.42 | powerunits | but still :( |
| 23:35.17 | p3nguin | killown: AsteriskNOW is built on CentOS and includes Asterisk, as well as your choice of no GUI, Asterisk GUI, or FreePBX. |
| 23:36.04 | p3nguin | I realize you don't know me, but if I like it, it can't be crap. I'm very picky with how things work. |
| 23:36.42 | dr00d | oooo extensions.conf ... |
| 23:36.47 | powerunits | :'( |
| 23:36.50 | dr00d | im in love |
| 23:37.10 | killown | p3nguin, thanks man |
| 23:37.30 | citywok | dr00d: wait... you didn't know about extensions.conf before? lol |
| 23:37.35 | p3nguin | hmm |
| 23:38.50 | dr00d | no ive been using freepbx to do everything |
| 23:39.13 | citywok | oh, freepbx rewrites extensions.conf when you hit apply |
| 23:39.14 | p3nguin | Time to wake up and smell the configs. |
| 23:39.30 | p3nguin | FreePBX is not an admin's friend, really. |
| 23:39.33 | dr00d | yes i couldnt see the trees for the forest ... |
| 23:39.37 | p3nguin | It's more like a manager's friend. |
| 23:39.57 | citywok | yea, it's a pain to work with and around |
| 23:39.57 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:41.04 | dr00d | yer im a bit new to asterisk still - its pretty interesting |
| 23:42.52 | *** join/#asterisk wolvenar (~wolv@sip.wolvenar.com) |
| 23:43.54 | dr00d | another question - extensions.con gets overwritten by freepbx - it says in that file not to make any manual changes there - can i configure dial plans from freepbx ? |
| 23:44.31 | p3nguin | That's kind of the point of FreePBX. |
| 23:45.10 | dr00d | ok - so where do i put Dial(Skype/[<originator>@]<destination>) in freepbx ? |
| 23:45.17 | dr00d | in the dial rules ? |
| 23:45.28 | powerunits | p3nguin: thanks for the help |
| 23:45.36 | powerunits | i find the solution. |
| 23:45.49 | powerunits | i simply added insecure=port,invite on sip trunk |
| 23:45.56 | powerunits | and it start working |
| 23:46.03 | powerunits | http://forums.whirlpool.net.au/archive/1133054 |
| 23:46.13 | powerunits | here is that link from wher i get help |
| 23:46.15 | powerunits | thanks |
| 23:46.33 | wolvenar | dr00d, put it in extensions_custom.conf |
| 23:46.46 | dr00d | ok ive seen that thanks |
| 23:47.08 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 23:48.31 | *** join/#asterisk jkroon (~jkroon@dsl-241-227-29.telkomadsl.co.za) |
| 23:48.39 | wolvenar | if you look into extensions_additional each has an include for a custom .. say like [macro-user-callerid] will follow with include => macro-user-calleris-custom |
| 23:49.35 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 23:49.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:50.54 | dr00d | thnx wolvenar - so it would be normal to put asterisk commands for skypeforasterisk in extensions_custom.conf or extensions_additional ? |
| 23:51.05 | wolvenar | <PROTECTED> |
| 23:51.21 | Thorn | so is there a way to get DIALSTATUS into CDR? it seems that either execution stops after Dial() or CDR is written immediately after Dial() (if it fails) |
| 23:51.24 | wolvenar | dr00d, in the custom |
| 23:51.48 | wolvenar | additional its over written each change to the webgui |
| 23:53.49 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 23:55.31 | wolvenar | I am having trouble with * attempting to dial out multiple times per call , failing the first one.. continuing later attempts . ( pbx-in-a-flash ) |
| 23:55.41 | wolvenar | anyone seen this before ? |