| 00:01.29 | fubada | Goto default,${EXTEN:-4},1 |
| 00:01.33 | fubada | is not dsipaying that behior |
| 00:01.39 | fubada | behavio |
| 00:01.39 | fubada | r |
| 00:02.25 | p3nguin | Show me the relevant dial plan that you wrote for it. |
| 00:02.40 | p3nguin | Not the whole file, just the parts in question. |
| 00:02.58 | fubada | exten = _1NXXNXXXXXX,1,Goto(default,${EXTEN:-4},1) |
| 00:03.17 | WIMPy | default is evil |
| 00:03.25 | p3nguin | Using the default context is not recommended. |
| 00:03.29 | fubada | i have no other options iwth gui |
| 00:04.04 | p3nguin | But still, a call matching that extension pattern would cause the call to go to the last four digits of the number, in the default context. |
| 00:07.18 | fubada | not working as expected :( |
| 00:07.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 00:08.42 | p3nguin | Show me. |
| 00:08.48 | p3nguin | core set verbose 3 |
| 00:08.53 | p3nguin | Make a call to it. |
| 00:23.08 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-9-126.bois.qwest.net) |
| 00:28.31 | fubada | p3nguin: another problem |
| 00:28.40 | fubada | separate from inbound dialplan |
| 00:28.45 | fubada | ill fix that later ;) |
| 00:29.09 | fubada | when i make an outgoing call to my cell, it works, but i cant hear anything on th asterisk side |
| 00:29.10 | fubada | rtp not working |
| 00:29.14 | fubada | firewall is off |
| 00:29.19 | fubada | routing is set up properly |
| 00:32.15 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 01:06.43 | p3nguin | NAT settings are probably wrong if you have one-way audio. |
| 01:13.18 | prgmrchris | yea |
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| 01:23.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:17.25 | dijib | hello all |
| 02:18.26 | dijib | p3nguin: zoneminder is slick, and my uvc camera needed spedial drivers |
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| 03:52.15 | usam | sruffell: I have the output of the pri debug at http://pastebin.com/nCGMk0Yt ... I dont understand much of it, can you or someone else help me to understand why Im getting the D-channel problem? |
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| 04:59.17 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 05:00.41 | linocisco | does asterisk recommend openvox cards (http://www.openvox.cn/en/products/fxofxs-cards.html?page=shop.browse&category_id=1) rather than digium cards? |
| 05:03.59 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 05:04.17 | ChannelZ | Why would they recommend someone else's hardware than their own? :) |
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| 05:31.00 | *** join/#asterisk din3sh (~din3sh@41.212.200.54) |
| 05:32.28 | din3sh | what are differences between asterisk and cisco call manager in terms of features? |
| 06:03.48 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 06:03.51 | qakhan | hi all |
| 06:04.12 | qakhan | i have connected asterisk with MSSQL 2008 DB, |
| 06:04.56 | qakhan | i want to get table columns in variables in dialplan |
| 06:05.38 | qakhan | like name->name variable address-> address veriable etc... |
| 06:06.06 | drmessano | din3sh, CCM probably runs 60% to 70% of the features Asterisk has.. Asterisk has a much richer feature set. |
| 06:08.38 | MrTelephone | You guys are still awake |
| 06:08.45 | MrTelephone | It's 2am |
| 06:09.04 | *** join/#asterisk pbxMan (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 06:11.10 | qakhan | i have connected asterisk with MSSQL 2008 DB, |
| 06:11.12 | qakhan | i want to get table columns in variables in dialplan |
| 06:11.14 | qakhan | like name->name variable address-> address veriable etc... |
| 06:14.36 | p3nguin | Repeating yourself every six minutes isn't going to get your answer any faster. |
| 06:19.35 | ChannelZ | At least we've moved on from the hanging up thing. |
| 06:20.52 | qakhan | ChannelZ lol i have still hangup issue |
| 06:21.16 | din3sh | still not hanging up properly? |
| 06:21.16 | din3sh | :o |
| 06:21.27 | qakhan | nop |
| 06:21.34 | din3sh | drmessano:thnx |
| 06:21.51 | qakhan | but now i m now i m working on DB thing |
| 06:22.00 | qakhan | anyone can help me in this? |
| 06:22.05 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:23.56 | din3sh | asterisk loses connection to mysql db through odbc evry now and then |
| 06:23.56 | din3sh | WARNING[25511]: res_odbc.c:105 ast_odbc_prepare_and_execute: S QL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77] MySQL server has gone away (66) |
| 06:24.02 | din3sh | any idea why so? |
| 06:25.57 | din3sh | WARNING[25511]: res_odbc.c:105 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77] MySQL server has gone away (66) |
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| 06:46.55 | *** join/#asterisk paumar (~chatzilla@77.42.241.114) |
| 06:48.41 | paumar | hello i'm a c++ dev, with not much experience in Ubuntu, we have a softphone based on iaxclient, i have sound problems on Ubuntu using ALSA / portaudio |
| 06:50.41 | paumar | i changed the sample_rate in audio_portaudio.c and iaxclient_lib.c to 44100, and started hearing something but still not good sound |
| 06:51.21 | paumar | hello i'm a c++ dev, with not much experience in Ubuntu, we have a softphone based on iaxclient, i have sound problems on Ubuntu using ALSA / portaudio. |
| 06:51.34 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.20.215) |
| 06:57.01 | pbxMan | paumar: what softphone is that? |
| 06:59.21 | paumar | a home made |
| 07:03.37 | *** join/#asterisk ndemir (~ndemir@85.105.28.77) |
| 07:04.04 | ndemir | anyone using elastix here? I have a question about call center module. I am using call center module for automated outgoing calls. But when there is no agent available, the user/callee waits in queue. Is there a option like "do not call someone when the agents are busy/unavailable?" |
| 07:04.59 | din3sh | ndemir: try #elastix |
| 07:05.30 | ndemir | I tried, but no answer there. |
| 07:05.47 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 07:08.58 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 07:09.09 | qakhan | i have connected asterisk with MSSQL 2008 DB, |
| 07:09.17 | qakhan | i want to get table columns in variables in dialplan |
| 07:09.21 | qakhan | like name->name variable address-> address veriable etc... |
| 07:10.43 | phix | so any ideas to my question? |
| 07:11.02 | phix | [TK]D-Fender would know! |
| 07:11.13 | phix | If he was here |
| 07:12.30 | phix | Any one else here just as awesome or awesomer than [TK]D-Fender ? |
| 07:16.02 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 07:21.27 | paumar | phix what's ur question ? |
| 07:21.39 | paumar | maybe i'm as awesome as Dfender |
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| 07:36.15 | zamba | WIMPy: i have two spare pci slots.. but what do you mean by "off the shelf cards"? can you give an example? |
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| 08:14.28 | phix | paumar: :D I hope so |
| 08:14.42 | phix | I have an issue with analog phones connected to my TDM400, it seems I need to press each key for longer in order for asterisk it pick it up as being a press, any way to reduce this so it picks up a shorter press? |
| 08:15.11 | phix | It does the same thing with the hook button, I need to hold it down for at least a second or two otherwise it is picked up as a flash |
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| 08:16.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:16.49 | phix | It wasnt doing it before I upgraded from 1.6 to 1.8 and changed the mobo, could it be a timing issue? or issue with asterisk? is the key press threshold configurable? those are basically my questions :) |
| 08:17.01 | phix | Now be awesome paumar! |
| 08:17.45 | drmessano | Is this a 400 card without the power molex connector? |
| 08:18.05 | paumar | phix: sorry i still get the chance to work on 1.8 |
| 08:20.00 | drmessano | The changing of the mobo is your significant change. Sounds like an older card that's not getting enough power from the bus, or maybe an interrupt conflict. |
| 08:22.25 | paumar | phix: usually events start on key up, not key down |
| 08:24.12 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 08:24.21 | paumar | phix: maybe you should change the progzone in ur dahdi config |
| 08:31.57 | paumar | phix: i'm wrong about the event start on key up, since ur talking about analog phones. i had some issues detecting busy/answer/hangup on TDM400, solved by changing the progzone. but never tried the DTMF to be sure about ur issue |
| 08:33.02 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 08:33.09 | paumar | phix: i'm in lebanon, so i had to change some code to detect the busy/answer/hangup. i should share them some day with the community |
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| 08:35.20 | elliot98 | how does one retrieve SIP packet variables after a dial command? |
| 08:37.34 | *** join/#asterisk arekm (matrix157@pld-linux/arekm) |
| 08:45.47 | din3sh | paumar: do you calls hang up properly? |
| 08:46.02 | din3sh | your* |
| 08:46.07 | paumar | yes |
| 08:46.11 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 08:46.20 | phix | drmessano: it has a power molex connector |
| 08:46.23 | phix | and it is plugged in |
| 08:46.28 | din3sh | what did you change exactly? |
| 08:46.33 | din3sh | indications.conf? |
| 08:47.27 | phix | /var/log/dmesg.3.gz:[ 14.679594] wctdm24xxp 0000:03:00.0: Found a Wildcard TDM: Wildcard TDM410P (0 digital modules, 2 analog modules) |
| 08:48.17 | phix | paumar: yeah it is set to au |
| 08:48.36 | phix | I also have the following in /etc/modules: wctdm24xxp opermode=AUSTRALIA |
| 08:49.23 | phix | I didnt change indications.conf to my knowledge, but updating from 1.6 to 1.8 (from digium;'s debian repo) may have change it |
| 08:50.32 | *** part/#asterisk ndemir (~ndemir@85.105.28.77) |
| 08:51.14 | phix | din3sh: ah! indications.conf is set to us as the default not au, I wonder if that would make a difference, I did tell dahdi I am in au though |
| 08:51.25 | paumar | din3sh: in chan_dahdi.conf i first tested the progzone:cr (costa rica), and callprogress=yes, it solved some problems, but still couldn't get answer. ASterisk was launching the event answer as a timeout |
| 08:51.27 | zamba | can someone recommend a simple ISDN bri card? |
| 08:51.34 | zamba | i need to use it for voice |
| 08:51.49 | phix | zamba: openvox or digium |
| 08:52.21 | phix | zamba: openvox is the cheapest but make sure it has the correct certifaction if you are connecting it to outside lines |
| 08:52.23 | zamba | someone here yesterday suggested an "off the shelf isdn card"? |
| 08:52.29 | paumar | din3sh: to solve all my problems i have to make some changes in the asterisk code itself |
| 08:53.01 | zamba | could this be one of those ISDN cards we used in the 90s? |
| 08:53.11 | zamba | for data? |
| 08:54.26 | zamba | http://www.novavox.co.uk/asterisk-cards-isdn-cards-bri-(isdn2)-openvox-b100p-port-isdn-bri-pci-p-40.html |
| 08:54.29 | zamba | i need two of those? |
| 08:56.03 | paumar | din3sh: in main/dsp.c i had to change the threshold, so i made a progzone lb (for lebanon) and set dsp->threshold = 6000 |
| 08:57.52 | paumar | din3sh: by default DEFAULT_THRESHOLD = 512 |
| 08:58.57 | phix | hmmm |
| 09:00.36 | paumar | phix: hope you read the things i wrote for din3sh |
| 09:01.47 | din3sh | what does this threshold do actually? |
| 09:03.19 | paumar | din3sh: no idea, i just figured it out by testing and changing all the variables. you can find this text in dsp.c : /*!\brief This value is the minimum threshold, calculated by averaging all |
| 09:03.21 | paumar | <PROTECTED> |
| 09:03.23 | paumar | <PROTECTED> |
| 09:03.25 | paumar | <PROTECTED> |
| 09:03.27 | paumar | <PROTECTED> |
| 09:03.28 | paumar | <PROTECTED> |
| 09:03.29 | paumar | <PROTECTED> |
| 09:04.12 | din3sh | how did you find the threshold for lebanon? |
| 09:04.38 | paumar | just testing, it took lots of time |
| 09:04.55 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 09:05.26 | paumar | change the value, compile, make a call, test answer/hangup, and redo it, till i found 6000 |
| 09:05.33 | paumar | my magic number |
| 09:06.28 | din3sh | woo |
| 09:06.34 | din3sh | trial and error! |
| 09:08.32 | paumar | unfortunately yes |
| 09:11.10 | paumar | i'm sure telecom people know how to find it, but i'm just a software engineer |
| 09:12.12 | din3sh | changing tones according to http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf didnt help? |
| 09:12.17 | din3sh | i see lebanon in there |
| 09:14.21 | din3sh | What if u edit zonedata.c in dahdi tools for a new zone for lebanon |
| 09:14.38 | din3sh | and set loadzone=lb in dahdi? |
| 09:14.40 | *** join/#asterisk ks3 (ks3@tatooine.digitallotus.com) |
| 09:14.40 | din3sh | would that work? |
| 09:16.06 | paumar | din3sh: 425 is the frequency, i already knew this, it the same for CostaRica, but still had no answer until asterisk timed out ringing, neither had hangup until i changed the threshold |
| 09:16.45 | din3sh | ohh |
| 09:16.51 | din3sh | good to know |
| 09:17.11 | din3sh | so changing the frequency only wouldnt solve the problem |
| 09:17.12 | din3sh | ? |
| 09:17.36 | paumar | din3sh: no, the frequency didn't solve it |
| 09:19.37 | paumar | din3sh: if u have such issues, i would make the effort to share the code, i wanted to be sure no client complains before sharing on asterisk |
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| 09:22.56 | paumar | din3sh: there are some other changes i made in the code |
| 09:23.21 | paumar | din3sh: minor changes, the main being the threshold |
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| 09:31.03 | din3sh | what other files did you change? |
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| 09:31.53 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 09:32.15 | hariom | What is the status of GTalk integration with Asterisk? I recently read that Digium is not providing now any support for this as Google has changed connections for GTalk. |
| 09:40.47 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
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| 09:44.57 | [ProB]CrazyMan | WIMPy: Hi how are you? I still have some problems with lcr, I always get some DTMF digits transmitted but I have no idear why... |
| 09:45.17 | *** join/#asterisk polysics (~polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 09:45.20 | polysics | hello |
| 09:45.39 | polysics | my SIP user registers correctly, but when calling the server i get bad auth |
| 09:45.50 | polysics | user is host=dynamic and nat=yes |
| 09:46.02 | polysics | context is just monkeys |
| 09:46.10 | *** join/#asterisk din3sh (~din3sh@41.212.200.249) |
| 09:46.12 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:46.27 | polysics | and randomly, it works |
| 09:46.27 | qakhan | any one have installed espeak? |
| 09:46.36 | polysics | from time to time calls go through |
| 09:51.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:00.57 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
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| 10:39.49 | elliot98 | how does one retreive the call-id for a dial command? |
| 10:46.34 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 10:46.39 | qakhan | any one have installed espeak? |
| 10:47.10 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1e8e.pool.mediaWays.net) |
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| 11:04.37 | qakhan | any one have installed espeak? |
| 11:06.41 | din3sh | what is espeak? |
| 11:08.24 | WIMPy | usam: Looks like your GW suddenly stops to respond completely. |
| 11:08.55 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:09.51 | WIMPy | zamba: I was to referring to the el cheapo things. Just make sure they use a CCD HFC chip. |
| 11:10.27 | WIMPy | [ProB]CrazyMan: Do you mean receive? |
| 11:11.34 | [ProB]CrazyMan | [call=1008 ast=lcr/5 line=1512] Recognised DTMF digit '2'. |
| 11:11.35 | [ProB]CrazyMan | [call=1008 ast=lcr/5 line=1988] Sending queued digit '2' to Asterisk. |
| 11:12.07 | [ProB]CrazyMan | things like that |
| 11:12.53 | WIMPy | It mey be over sensitive. But I personally have a feeling that it's actually a bug in the DTMF detection that gives false positives sometimes. |
| 11:13.18 | din3sh | is it possible to use this in queues.conf ---> member=SIP/[siptrunk]/extension |
| 11:13.19 | din3sh | ? |
| 11:13.48 | WIMPy | But playing with dtmfthreshold (module parameter to mISDN_dsp) does help. |
| 11:14.12 | *** join/#asterisk LooserOuting (~smuxi@ip-62-143-235-156.unitymediagroup.de) |
| 11:16.23 | [ProB]CrazyMan | thx I will try it |
| 11:16.37 | qakhan | dijib its TTS |
| 11:20.51 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 11:23.18 | din3sh | can i ring the queue of a second * box trunked to first * from which the call will originate? |
| 11:28.18 | kaldemar | din3sh: of course. make the call go to the queue in a matching extension. |
| 11:29.18 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 11:32.36 | hariom | What is the status of GTalk integration with Asterisk? I recently read that Digium is not providing now any support for this as Google has changed connections for GTalk. |
| 11:32.52 | hariom | Is it working for anybody now? |
| 11:32.53 | *** join/#asterisk pokus (~focus@158.193.202.91) |
| 11:33.29 | din3sh | kaldemar: testing it, should work ;) |
| 11:34.02 | din3sh | am having another issue with cisco 79xx phones on ast 1.8 |
| 11:34.18 | din3sh | critical response on sip retransmit |
| 11:36.08 | din3sh | [May 17 12:55:58] VERBOSE[8027] chan_sip.c: Retransmitting #10 (no NAT) to 192.168.6.65:5060:SIP/2.0 487 Request Terminated |
| 11:38.28 | din3sh | it happens when an extension calls another one but cancels the call, the phone it seems cannot ack the 487 |
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| 11:59.34 | usam | WIMPy: ok .. then its the gw... i will contact portech about this ,... thx a lot. |
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| 13:31.28 | iulhk | can anybody tell me, do we still have asterisk+skype integration available or its finished as i heard that asterisk has finished contract with skype etc? |
| 13:32.52 | [TK]D-Fender | gone over a year now |
| 13:34.38 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 13:34.39 | *** join/#asterisk iulhk (~iulhk@116.71.183.143) |
| 13:34.42 | iulhk | can anybody tell me, do we still have asterisk+skype integration available or its finished as i heard that asterisk has finished contract with skype etc? |
| 13:35.04 | [TK]D-Fender | gone over a year now |
| 13:38.09 | jeffspeff | i remember back then |
| 13:38.18 | jeffspeff | i didn't use it, but was still sad. |
| 13:39.43 | *** join/#asterisk JacobIsre (~JacobIsre@99-190-176-190.lightspeed.hstntx.sbcglobal.net) |
| 13:39.55 | JacobIsre | HI all. I need a little help please. |
| 13:40.34 | JacobIsre | I just installed Asterisk on Ubuntu 11.04, I have a couple of extensions and they can call each other but having hard times with outbound dialing through google voice. |
| 13:54.19 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:55.18 | [TK]D-Fender | JacobIsre, Please elaborate on "hard times" |
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| 14:01.24 | pokus | hi...i have alias for interface eth0 192.168.10.10 and eth0:1 192.168.20.10. when asterisk make invite to peer it set SDP address from interface c=IN IP4 192.168.10.10. Is possible change ip for SDP from alias to c=IN IP4 192.168.20.10? |
| 14:02.04 | pokus | i can;t find this setting..its possible? |
| 14:02.43 | pokus | asterisk version is 1.6.0.26 |
| 14:02.56 | [TK]D-Fender | pokus, there isn't a setting. packets are generated based on the kernel routing rules. |
| 14:03.19 | [TK]D-Fender | pokus, * plays no role in the selection of the interface |
| 14:04.10 | pokus | so.. i have to change route rules? |
| 14:04.30 | *** join/#asterisk krunal (73735a32@gateway/web/freenode/ip.115.115.90.50) |
| 14:04.39 | krunal | hi |
| 14:04.46 | [TK]D-Fender | Not sure how you might fix that.... |
| 14:04.57 | krunal | I am facing an issue with voicemail |
| 14:05.20 | krunal | sometimes voicemail messages gets some part ommited |
| 14:05.34 | pokus | [TK]D-Fender: i hope the exist some setting like externip for localip |
| 14:06.01 | krunal | please help me to figure out what could be the cause |
| 14:06.18 | krunal | messages are not continuous and have omitted parts. |
| 14:07.10 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-tgtvebrzkapfsgfz) |
| 14:07.24 | krunal | Like if the caller says his contact number in Voicemail message, sometimes few digits gets ommited |
| 14:09.14 | [TK]D-Fender | pokus, actually... you might be able to fix the SDP, but the packet itself is kernal-tied. jsut set the bindaddr to the IP on the interface you want... but that means it won't listen on the other(s) |
| 14:10.04 | krunal | hi there |
| 14:10.12 | [TK]D-Fender | krunal, if the playback shows lost voice data then that is due to jitter on the lines which might get corrected in a voice call, but not on a recording. |
| 14:10.23 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 14:10.33 | jaytee | totally awesome scientific breakthrough! http://www.wimp.com/perpetualmotion/ |
| 14:10.49 | krunal | [TK]D-Fender: So how to resolve that? |
| 14:11.09 | pokus | [TK]D-Fender: hmm ok thx |
| 14:11.22 | [TK]D-Fender | krunal, sip.conf and iax.conf each have jitterbuffer options to tweak. Go read the sample configs and try them. |
| 14:11.29 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:11.38 | krunal | [TK]D-Fender: ok thanks :) |
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| 14:33.04 | iulhk | can anybody tell me that skype asterisk license is still available and can we still use skype asterisk integration for calling, etc? |
| 14:33.29 | WIMPy | no, no |
| 14:34.01 | iulhk | <WIMPy>: is this no no for me ? |
| 14:34.26 | WIMPy | yes |
| 14:34.44 | mbrit | is there any alternative? |
| 14:36.11 | iulhk | <WIMPy>: so we can't even use as testing purposes too, as i just read in asterisk-now have skype-gateway feature available, will it work or not ? |
| 14:37.14 | WIMPy | ~sfa |
| 14:37.43 | WIMPy | Hmm. Anyway, it has been discontinued a year ago. |
| 14:37.50 | iulhk | <mbrit>: if you are asking about skype integration then there is one un-stable and pathetic alternate is available which is (freeswitch-skypopen) |
| 14:39.08 | mbrit | iulhk, ok |
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| 15:05.42 | danfromuk | Hi, im trying to set up a queue, however wrapuptime seems to be ignored. It takes at least 10 to 15 seconds for the next queues call to start ringing, even though wrapuptime=0 |
| 15:10.05 | RubyRails | HABTM question - I have a CallDetail model and a Keyword model that have to and belong to many each other |
| 15:10.25 | RubyRails | if I create a new @call and a new @keyword, |
| 15:10.37 | RubyRails | how do I assign the @keyword to the @call ? |
| 15:11.07 | RubyRails | sorry, wrong forum |
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| 15:27.24 | ZenMaster | Good morning gentlemen. |
| 15:32.46 | [TK]D-Fender | Take off every zig!!! |
| 15:34.52 | zamba | WIMPy: but still.. you got any examples? |
| 15:35.10 | zamba | WIMPy: could this be one of the ISDN card we used for getting internet access in the 90s? |
| 15:35.15 | zamba | cards* |
| 15:35.32 | WIMPy | zamba: It could. |
| 15:35.54 | WIMPy | But for the ones with Siemens chips we have no drivers. |
| 15:36.13 | zamba | but can you please provide me with a concrete example? |
| 15:36.18 | zamba | el cheapo is perfect for this project |
| 15:36.34 | zamba | since i don't know wheather it'll work or not |
| 15:37.35 | WIMPy | As I said: Make sure it has a HFC-S chip by CCD. They can be supported by all drivers. |
| 15:38.28 | *** join/#asterisk kenhunt (~kenhunt@nat/digium/x-otqlpltmchukbqex) |
| 15:38.32 | WIMPy | That's the best about them: You have the coice, which driver to use. |
| 15:45.23 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
| 15:49.30 | WIMPy | zamba: There are four on ebay in 17h, ony one has a bid so far. |
| 15:50.49 | *** join/#asterisk dorphalsig (be9399cc@gateway/web/freenode/ip.190.147.153.204) |
| 15:51.36 | zamba | link? |
| 15:53.35 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:53.48 | dorphalsig | Hello, I have two SPA3102 connected to two PSTN phone lines, however from time to time I get a strange tone istead of recieving the call |
| 15:53.56 | WIMPy | http://www.ebay.de/sch/jaguthalt/m.html?item=200758145667&viewitem=&rt=nc&_trksid=p4340.l2562 |
| 15:54.11 | WIMPy | Just as an example. There should be plenty of them around. |
| 15:56.05 | ZenMaster | Boss comes in and intrupts me from my task like three times. |
| 15:56.27 | ZenMaster | Boss: "Whats the matter man? You seem serious this morning.". |
| 15:56.45 | ZenMaster | Me, "I am completeing the tasks that you set out for me this morning." |
| 15:56.54 | ChannelZ | SAY HELLO TO MY LIL FRIEND *rattatatatatatt* |
| 15:57.27 | ChannelZ | Oh. I misjudged the direction the story was going to take. |
| 15:57.31 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:57.37 | dorphalsig | Hello, I have two SPA3102 connected to two PSTN phone lines, however from time to time I get a strange tone istead of recieving the call |
| 15:57.37 | WIMPy | also has a bunch of them catching dust. |
| 16:00.45 | dorphalsig | I need somebody to lend me a hand with a remote asterisk configuration... |
| 16:00.52 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-224.mobile.uci.edu) |
| 16:01.22 | ZenMaster | ChannelZ: Ehehehe. He is very compulsive. |
| 16:01.41 | ZenMaster | A lot of times gives me a monumental task then comes in 1-3 minutes later to ask if it is done. |
| 16:01.55 | dorphalsig | anybody willing to help me out? Your fee will be wired via paypal |
| 16:01.56 | dorphalsig | :) |
| 16:01.56 | ZenMaster | Yesterday I was timing him. |
| 16:02.06 | dorphalsig | I need somebody to lend me a hand with a remote asterisk configuration... anybody willing to help me out? Your fee will be wired via paypal |
| 16:02.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:02.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:02.52 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-135-101.dsl.stlsmo.sbcglobal.net) |
| 16:03.16 | LemensTS | can you define how many times you want a moh file to play? |
| 16:04.07 | LemensTS | I have someone who doesnt want moh to play, but is wondering what it will do if i disable it, how will someone know they are on hold still? |
| 16:05.35 | WIMPy | Tehir phone may tell them if they are lucky. |
| 16:05.47 | WIMPy | But that will probably be confusing. |
| 16:06.09 | WIMPy | Err, not hearing anything, that is. |
| 16:08.19 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 16:12.28 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 16:14.09 | dorphalsig | I need somebody to lend me a hand with a remote asterisk configuration... anybody willing to help me out? Your fee will be paid via Paypal |
| 16:15.11 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:17.06 | dorphalsig | I need somebody to lend me a hand with a remote asterisk configuration... anybody willing to help me out? Your fee will be paid via Paypal |
| 16:19.55 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:20.54 | *** join/#asterisk kenhunt (~kenhunt@nat/digium/x-knqspgxgooawdmop) |
| 16:22.52 | ChannelZ | dorphalsig: ask an actual question |
| 16:25.04 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:33.28 | [TK]D-Fender | Why ask questions when you can post the exact same message 6 times in 20 minutes. |
| 16:33.56 | [TK]D-Fender | All of which I can see on-screen without having to scroll even. |
| 16:34.11 | [TK]D-Fender | 7 if I roll up 1 click.. |
| 16:34.58 | ZenMaster | :d |
| 16:35.52 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:36.08 | ChannelZ | Yeah it's kind of ironic |
| 16:36.13 | *** join/#asterisk saxa (~brastrak@201-50-67-77.user.veloxzone.com.br) |
| 16:36.29 | [TK]D-Fender | Don't you think? |
| 16:36.40 | [TK]D-Fender | looks for a knife |
| 16:37.10 | ChannelZ | To stab Alanis Morriset? |
| 16:37.45 | ZenMaster | Yes. |
| 16:37.59 | ChannelZ | YAY! |
| 16:39.00 | [TK]D-Fender | Actually I just picked up "You Oughta Know" for one of my bands.... |
| 16:39.30 | ZenMaster | Now that song is playing in my head. |
| 16:39.43 | ZenMaster | Dude that music was totally my 97. |
| 16:39.46 | ZenMaster | Hahaha. |
| 16:39.53 | ZenMaster | Not really, but I can recall. :D |
| 16:41.02 | ChannelZ | Maybe go listen to Hamster Dance, that aught to wash that song out of your head |
| 16:41.42 | ZenMaster | I have sound garden playing here in the office. |
| 16:44.01 | iulhk | can i install asterisknow at some remote linux machine, as i normally install and configre asterisk, addons etc ? |
| 16:44.34 | *** join/#asterisk viaSanctus (~viaSanctu@unaffiliated/viasanctus) |
| 16:45.42 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:54.03 | ChannelZ | sure why not |
| 16:55.21 | [TK]D-Fender | iulhk, that comparison does not make sense |
| 16:55.31 | [TK]D-Fender | iulhk, AsteriskNOW is a complete distribution. |
| 16:55.47 | [TK]D-Fender | iulhk, not an "add-on". |
| 16:57.50 | viaSanctus | what is diigium? |
| 16:57.55 | viaSanctus | digium |
| 16:58.29 | [TK]D-Fender | A company that sells softwaer and hardware, and support for both |
| 16:59.27 | viaSanctus | does asterisknow work with it's hardware only? or does it provide automatic detection rules for it only? |
| 16:59.50 | viaSanctus | "Automatic detection and configuration of Digium analog and digitial telephony hardware" (http://www.asterisk.org/asterisknow) |
| 16:59.57 | [TK]D-Fender | viaSanctus, Many other hardware makers need a driver interface layer that it doesn't come included with |
| 17:00.02 | [TK]D-Fender | doesn't mean you can't add it |
| 17:00.18 | viaSanctus | I'm thinking of working with snoms |
| 17:00.27 | viaSanctus | would that be the right choise? |
| 17:01.05 | [TK]D-Fender | Depends on what your other choices are. |
| 17:01.30 | viaSanctus | pbx with ivr and voicemail |
| 17:01.30 | [TK]D-Fender | And SIP phone model isn't related to your other quesiotn. SIP is SIP.. there is no "support" issue. |
| 17:01.53 | viaSanctus | digium doesn't provide sip models? |
| 17:02.57 | [TK]D-Fender | ? |
| 17:03.07 | [TK]D-Fender | Digium just started making SIP phones of their own... |
| 17:03.19 | viaSanctus | why your answer then? |
| 17:03.20 | [TK]D-Fender | And pretty much every other one out there works jsut fine with Asterisk as well |
| 17:03.20 | iulhk | <[TK]D-Fender>: so means i have to talk with my hosting provider that they will install on some linux machine and then give access to me. actually i was looking at asterisknow it's features list still showing SKYPE GATEWAY, i just want to check which skype gateway integration and feature asterisknow supported. is there any other way to check SKYPE GATEWAY ? |
| 17:03.26 | viaSanctus | never mind |
| 17:03.39 | viaSanctus | I actually had another q |
| 17:03.42 | viaSanctus | regarding bandwith |
| 17:03.49 | [TK]D-Fender | viaSanctus, When you asked about supporting other hardware we take that as meaning PCI interface, etc... |
| 17:04.15 | [TK]D-Fender | viaSanctus, Because there is no challenge in interfacing with SIP, so manufacturer is practically irrelevant. |
| 17:04.16 | viaSanctus | lets say I have a deal with an isp for sip truncs and also for the carrier (dsl) to the clients office |
| 17:04.37 | viaSanctus | is it better to have a dedicated asterisk server @ the client |
| 17:04.49 | viaSanctus | or 1 at the backend of the isp? |
| 17:05.19 | [TK]D-Fender | depends on your confort zone, backup requirements, connectivity, etc. |
| 17:05.23 | [TK]D-Fender | DSL wise possibly not. |
| 17:05.36 | [TK]D-Fender | As calling the guy next to you would wast bandwidth... |
| 17:05.59 | viaSanctus | good point |
| 17:06.22 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:06.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:06.38 | *** join/#asterisk mattjfox (~Crakore@ip72-219-183-40.oc.oc.cox.net) |
| 17:06.53 | viaSanctus | does it matter bandwith wise for outgoing calls? |
| 17:07.08 | viaSanctus | does the tel -> pbx protocol give overhead to the sip trunc? |
| 17:08.11 | [TK]D-Fender | yes it matters. |
| 17:08.20 | [TK]D-Fender | calls take the same amount in as out |
| 17:08.49 | viaSanctus | I used the argument for non interoffice calls |
| 17:09.08 | viaSanctus | in which it would matter to have the pbx locally |
| 17:09.16 | viaSanctus | I'm concerend about the quality |
| 17:09.45 | viaSanctus | on a simple vdsl I can handle 10 simultaneous calls perfectly when the pbx is on the backend |
| 17:10.02 | viaSanctus | I wonder if I have 10 sip truncs locally if that would make a difference |
| 17:10.54 | [TK]D-Fender | count your bandwidth for external calls... |
| 17:11.00 | [TK]D-Fender | depends on codec and protocol. |
| 17:11.06 | viaSanctus | and qos |
| 17:11.19 | [TK]D-Fender | SIP+RTP has cumulative overhead where IAX2 can save you to an ITSP |
| 17:11.36 | viaSanctus | ok I need to google those terms |
| 17:11.56 | viaSanctus | but it seemed the answer to my q |
| 17:12.51 | viaSanctus | gotto run, be back later |
| 17:15.57 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 17:43.30 | wtf911 | caller id spoofing is so fun :o |
| 17:44.44 | MrTelephone | What do I need to diagnose a 488 not acceptable here when dealing with t38 reinvites? Can I put my sip dialog in the pastebin? |
| 17:45.35 | MrTelephone | It was working in asterisk 1.6 but seems to reject my t38 requests in asterisk 10.1.3. Was doing the same with 1.8.4. I have the newest res_fax.so module |
| 17:49.18 | *** join/#asterisk AsteriskT500 (~AsteriskT@hq.clearcable.ca) |
| 17:49.45 | AsteriskT500 | Hello Group first time in |
| 17:49.52 | MrTelephone | http://pastebin.com/mrpn8gHf |
| 17:50.28 | AsteriskT500 | Can anyone help me with a quick question. |
| 17:50.38 | AsteriskT500 | regarding the console scroll speed of module reload |
| 17:51.08 | MrTelephone | do an external command run and redirect to file if you want to look at it |
| 17:51.15 | paulc | AsteriskT500: what about the scroll speed? |
| 17:51.36 | *** join/#asterisk eye-scuzzy (~light@bellfast-1-pt.tunnel.tserv6.fra1.ipv6.he.net) |
| 17:52.26 | pabelanger | MrTelephone: could be a codec issue. Enable debugs and see what asterisk is doing |
| 17:52.27 | AsteriskT500 | Is the scroll speed of the asterisk cli a variable speed. |
| 17:52.30 | pabelanger | ~collectdebug |
| 17:52.30 | infobot | i guess collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:53.04 | AsteriskT500 | For instance when I used to do a module reload on an older version of 1.6.2 it would output very fast |
| 17:53.12 | AsteriskT500 | now it outputs line by line and is readable. |
| 17:54.04 | pabelanger | sounds like a buffering issue with the terminal |
| 17:57.27 | AsteriskT500 | Interesting - |
| 17:57.46 | AsteriskT500 | Calls display at regular speed it is just the reload speed |
| 18:00.11 | *** join/#asterisk eye-scuzzy (~light@bellfast-1-pt.tunnel.tserv6.fra1.ipv6.he.net) |
| 18:02.44 | MrTelephone | pabelanger, I did enable the debugs but debug fax nothing shows up. The call continues as normal but without t38. The only debug that shows what is going on is sip debug and I pasted it to http://pastebin.com/mrpn8gHf |
| 18:03.51 | pabelanger | MrTelephone: that is because you didn't read the wiki page. It explains how to properly enable debug logs for people to help |
| 18:04.41 | dorphalsig | Hey all. I got a weird behaviour in one of my boxes... its wired like this PSTN--->SPA3102-->ASTERISK+QUEUE--->RT31P2. When I dial in, the SPA3102 answers ok and forwards the call ok to *, HOWEVER IF I hangup before the call gets picked up, it just keeps going and enters the queue and rings the extensions... |
| 18:04.48 | dorphalsig | how can I address that? |
| 18:04.56 | MrTelephone | ok i'll check it out |
| 18:05.41 | *** join/#asterisk din3sh (din3sh@41.136.246.241) |
| 18:06.24 | *** join/#asterisk _pll (~anonymous@200.110.25.33) |
| 18:06.36 | _pll | Hello, I am having a strange issue after update. |
| 18:07.14 | _pll | I am being spammed by codec mismatch though call proceeds as normal. Version is 10.3.1 |
| 18:07.36 | _pll | Sometimes there is voice and sometimes there isn't. |
| 18:07.56 | dorphalsig | Hey all. I got a weird behaviour in one of my boxes... its wired like this PSTN--->SPA3102-->ASTERISK+QUEUE--->RT31P2. When I dial in, the SPA3102 answers ok and forwards the call ok to *, HOWEVER IF I hangup before the call gets picked up, it just keeps going and enters the queue and rings the extensions... |
| 18:07.58 | _pll | I tried loading all the codecs and formats available. |
| 18:08.19 | _pll | though the issue is still there, am I missing a res_? file. |
| 18:08.31 | _pll | I am using autoload = no in modules.conf |
| 18:08.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:09.57 | *** join/#asterisk felimwhiteley_ (~quassel@089-101-203026.ntlworld.ie) |
| 18:10.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 18:10.41 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:10.41 | *** join/#asterisk 16WAALI5Y (~chatzilla@mx40.roterschnee.com) |
| 18:10.41 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 18:10.41 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 18:10.41 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:10.56 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 18:12.31 | MrTelephone | pabelanger, what do I do with all the debug info after? |
| 18:16.04 | p3nguin | _pll: Use autoload=yes and be sure to set the correct codec for all peers involved. |
| 18:17.55 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 18:18.22 | Hive | is there a way to strip ' character from a string in asterisk, or is this something i should push off to a script? |
| 18:18.30 | _pll | I doubt it's the autoload directive that is causing the codec mismatch warnings. |
| 18:18.39 | _pll | I am probably missing a res_ file |
| 18:18.51 | _pll | I tried loading all those that apparented to be important. |
| 18:18.58 | _pll | I will try it with autoloading I guess. |
| 18:19.22 | pabelanger | MrTelephone: pastebin so other people can see it? |
| 18:19.24 | [TK]D-Fender | Hive, "core show application FILTER" |
| 18:21.05 | Hive | Thank you, D-Fender, just what i needed :D. Also it was under functions for future refrence |
| 18:21.21 | jaytee | on the phone with AT&T working on LNP order problem. They're rejecting the port to Flowroute because "it will drop the number of Centrex lines below 11". I'd rather be at the dentist getting a root canal without anesthesia. |
| 18:22.04 | Hive | hahaha |
| 18:22.17 | Hive | Jaytee i have to do the same thing today, dont make me scared ;{ |
| 18:22.43 | [TK]D-Fender | Hive, I meant to say "function", but flipped it wrong at the last moment.. |
| 18:23.35 | jaytee | so now the account rep is going to convert the lines to POTS, drop the unneeded lines and move the two fax lines they're going to keep to a different account number so Flowroute can resubmit the port order. |
| 18:23.36 | p3nguin | _pll: No, autoload isn't causing the mismatch, but it probably is causing a lack of loaded codecs. |
| 18:23.41 | jaytee | it will only take a week! |
| 18:24.40 | jaytee | I should have gone to tractor-trailer training school instead of pursuing a career in IT. |
| 18:25.21 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 18:25.43 | p3nguin | I just finished a port recently from AT&T. It took one business day to process it and get an FOC date (four days out). |
| 18:29.07 | _pll | I tried with autoload=yes, I keep getting codec mismatch warnings. |
| 18:29.17 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 18:29.21 | _pll | [2012-05-17 13:29:17] WARNING[16248]: channel.c:4989 ast_write: Codec mismatch on channel DAHDI/i1/4429304-1 setting write format to g729 from alaw native formats (alaw) |
| 18:29.35 | _pll | I think it happens when there is transcoding to do. |
| 18:29.41 | _pll | Also appears with gsm and alaw. |
| 18:29.56 | p3nguin | What happens when you set it to alaw only? |
| 18:30.21 | _pll | Doesn't happen. |
| 18:30.41 | _pll | I tried earlier. The warning seems to come when there is a need to transcode. |
| 18:30.59 | p3nguin | If you are using the correct codecs, no transcoding would be required. |
| 18:31.25 | _pll | [2012-05-17 13:31:04] WARNING[16273]: channel.c:4989 ast_write: Codec mismatch on channel DAHDI/i1/4429304-4 setting write format to gsm from alaw native formats (alaw) |
| 18:31.33 | _pll | That one is for the pri card. |
| 18:31.35 | MrTelephone | If you send a fax between 2 sip endpoints do both sides have to support t38? Will one endpoint terminate at asterisk with t38 and if the other endpoint doesn't support it, it will go without t38? |
| 18:31.41 | p3nguin | If you use dissimilar codecs, that is when transcoding is done. And if one of those call legs is using g.729, you'll need the license for transcoding. |
| 18:31.55 | _pll | Yeah, I am aware. |
| 18:32.06 | _pll | The warning also comes when there is gsm too. |
| 18:32.35 | p3nguin | Are you trying to use gsm on the dahdi channel? |
| 18:33.50 | ZenMaster | Oh god the big boss is coming in today. And he is going to see that I have just been setting up computer moniters to look importent, but in reality I have just been here watching netflix for weeks. |
| 18:33.54 | _pll | No? |
| 18:34.11 | _pll | I don't think that's possible. |
| 18:34.27 | _pll | I haven't set any allow disallow for the dahdi channel. |
| 18:34.49 | _pll | so it should require to transcode from whatever to alaw |
| 18:36.32 | _pll | That warning is from an incoming call. |
| 18:36.38 | p3nguin | I don't know if it is even possible to change the codec on your dahdi channels. Since the line connected to it would be g.711, I would think if there is a way to set it, you must set it to ulaw or alaw (depending on region). |
| 18:36.51 | _pll | The sip destination only allows gsm. |
| 18:36.56 | p3nguin | I generally stay clear of analog telephony. |
| 18:37.16 | _pll | It also happens against my voip provider. |
| 18:37.25 | _pll | Which requires g729 |
| 18:37.53 | _pll | There is voice, they talk just fine, but sometimes there is no audio. |
| 18:37.55 | p3nguin | I'd have to look at a sip debug. |
| 18:38.00 | _pll | I made sure it wasn't a rtp port issue. |
| 18:39.31 | _pll | Could be two issues, my voip provider is having problems and those warnings are bogus. |
| 18:39.38 | _pll | And totally unrelated. |
| 18:40.40 | _pll | Because I am having no issues with the pri yet the warning keeps showing there too. |
| 18:45.06 | MrTelephone | I'm calling a fax machine from my polycom phone. The polycom phone is replying with 488 not supported here. I have nat=yes and directmedia=no. I wonder why it cares if the other endpoint supports it or not. |
| 18:45.25 | *** join/#asterisk mattjfox (~Crakore@ip72-219-183-40.oc.oc.cox.net) |
| 18:46.17 | MrTelephone | asterisk sees that 1 endpoint supports t38 but if the second doesn't then t38 fails. I can understand why but at the same time t38 on one leg of the call seems like it would be better than nothing. |
| 18:46.20 | [TK]D-Fender | MrTelephone, Perhaps you should be showing the actual call... |
| 18:46.42 | MrTelephone | I'm looking at the debug and discovered that asterisk wants both legs to have t38 support |
| 18:47.37 | MrTelephone | And I can understand the extra work involved in transcoding t38 back to normal rtp. |
| 18:51.44 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 18:51.44 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 18:51.44 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 18:51.44 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:53.41 | MrTelephone | I think I'm missing the concept of fax for asterisk. Good from pstn to sip (t38), sip to sip (t38) if both endpoints agree |
| 18:55.04 | p3nguin | You can't have t38 on only half the fax call. |
| 18:55.13 | MrTelephone | Thanks for verifying |
| 18:55.30 | p3nguin | T.38 needs to be negotiated between the two end points. |
| 18:55.42 | MrTelephone | pabelanger, thanks for guiding me to the debug info bud |
| 18:56.00 | MrTelephone | p3nguin, I thought that was what passthrough is called |
| 18:56.32 | p3nguin | But if you are using ffa, if your provider supports T.38, you support T.38. So now you just have to make sure the other fax machine also supports it. |
| 18:56.58 | MrTelephone | makes sense |
| 18:57.16 | MrTelephone | It would be easy to transcode that though. |
| 18:57.22 | MrTelephone | cut out all the duplicate packets |
| 18:57.29 | p3nguin | Transcode what? |
| 18:57.40 | MrTelephone | The udptl |
| 18:57.56 | p3nguin | Transcode t.38 to something that isn't t.38? |
| 18:58.17 | p3nguin | I don't think that's going to happen. |
| 18:58.21 | MrTelephone | me neither |
| 18:58.45 | MrTelephone | slap in another sp2102 and call it a day |
| 18:59.20 | p3nguin | I can't support T.38 because my provider doesn't... but surely there is some way to have t.38 for calls that can negotiate it and fall back to g.711u for calls that can't. |
| 18:59.36 | MrTelephone | it the fallback works ok |
| 18:59.40 | MrTelephone | -it |
| 19:00.16 | p3nguin | If I set up things to use t.38 and then get a call that does not support it, the thing fails totally. |
| 19:00.31 | MrTelephone | hmm |
| 19:00.38 | MrTelephone | it drops the call? |
| 19:01.07 | p3nguin | It has been a while since I tried it, but I think the call just dies at that point. |
| 19:01.08 | MrTelephone | I can still here the modem tones until the fax machine hangs up because I'm doing a voice call to it to test |
| 19:01.14 | MrTelephone | hear |
| 19:01.35 | MrTelephone | Is there an option where you set it to force t38? |
| 19:01.59 | p3nguin | Let me take a look at my configs. I'm sure I just commented out the t38 settings since I can't use them. |
| 19:02.21 | MrTelephone | I setup a new server yesterday and im getting 0 errors on my t1 card now. THe pci bus must have been just wrecked on it |
| 19:02.54 | MrTelephone | what version of asterisk are you running? |
| 19:02.59 | p3nguin | I was using t38pt_udptl=yes for the SIP peer. |
| 19:03.03 | p3nguin | 1.8.something |
| 19:08.15 | wwalker | what does a SIP 480 status mean from the PSTN? In a PBX environment, I think it comes from DND status, but what is it in PSTN through a SIP VOIP provider? |
| 19:11.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:15.10 | p3nguin | Isn't a 480 a Busy response? |
| 19:19.39 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 19:24.17 | wwalker | p3nguin: I think that is 486, and this was returned during a high volume emergency notification. The 480 messages came back on 39% of the calls. No way that many lines were busy in this age of call waiting and voice mail. |
| 19:26.01 | MrTelephone | an endpoint will reply with busy while ringing |
| 19:26.56 | din3sh | i get a few 486 response, with no DND on phone, no call on phone |
| 19:27.01 | din3sh | why would it be so? |
| 19:31.46 | talntid | http://pastebin.com/Jv3nce8j |
| 19:32.33 | talntid | this is giving "MACHINE" as the response to everything.... any ideas? It is pretty much exactly the same as the wiki example |
| 19:34.32 | Dorphalsig | Hey all. I got a weird behaviour in one of my boxes... its wired like this PSTN--->SPA3102-->ASTERISK+QUEUE--->RT31P2. When I dial in, the SPA3102 answers ok and forwards the call ok to *, HOWEVER IF I hangup before the call gets picked up, it just keeps going and enters the queue and rings the extensions... |
| 19:39.30 | fubada | hi, where is global ring timeout seconds set |
| 19:39.36 | fubada | which config file |
| 19:41.25 | talntid | the exact wiki example identifies my answering the phone "hello" as a Machine, too. |
| 19:44.32 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 19:45.19 | *** join/#asterisk logicwrath (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 19:48.06 | MrTelephone | 486? |
| 19:49.02 | akrohn | i have an attendant menu. press 3 to Dial() multiple phones. I need to update the CDR custom field once I know which phone answers the call. Where in the dialplan can I put this code? Will the line immediately after Dial() know which extension answered the call once it hangs up? |
| 19:51.50 | *** join/#asterisk ccesario (~ccesario@189.29.37.189) |
| 19:54.22 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:55.17 | eduzimrs | Hi, anyone knows where to set up SRTP media on polycom IP550 ? |
| 19:59.13 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:59.17 | talntid | also, on the AMD... it says MACHINE because INITIALSILENCE-2500-2500 |
| 20:04.00 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 20:04.19 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 20:04.58 | *** join/#asterisk BlackMaria (~dickery@modemcable019.198-160-184.mc.videotron.ca) |
| 20:11.23 | p3nguin | 480 Temporarily Unavailable |
| 20:16.07 | *** join/#asterisk r3zon8 (~r3zon8@csc.interimhealthcare.com) |
| 20:16.59 | r3zon8 | whats a pretty good/inexpensive sip provider i can use at home with an obi110? |
| 20:20.36 | *** join/#asterisk galba (~ldm@3-254.197-178.cust.bluewin.ch) |
| 20:20.49 | keith4 | ~itsp |
| 20:20.49 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:21.22 | r3zon8 | ~itsplist-us |
| 20:21.22 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:21.57 | keith4 | I have no idea with an obi110 is, but voicepulse is a pretty solid recommendation |
| 20:26.23 | akrohn | using Dial(SIP/4444,5,M(blah)) to call a macro on a Dial() command... is it not possible to modify CDR fields within that macro? I'm trying to do what this link is talking about: http://lists.digium.com/pipermail/asterisk-users/2010-February/244170.html |
| 20:31.55 | talntid | p3nguin, do you have any advice in regarde to my AMD() problem? |
| 20:44.09 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 21:05.28 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:05.49 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bafpymawihqmkmsn) |
| 21:06.04 | *** join/#asterisk rhg412 (~harrisong@68-73-223-152.ded.ameritech.net) |
| 21:08.47 | *** join/#asterisk gtTuna (~Tuna@d118-75-91-174.try.wideopenwest.com) |
| 21:09.14 | rhg412 | Hey I'm getting a wrong ELF class error when trying to load the res_digium.so module any ideas ? |
| 21:09.24 | Kobaz | [2012-05-17 17:08:58] -- Got SIP response 500 "Server Internal Error - re-Invite inside a re-Invite" back from 192.168.51.20:5060 |
| 21:09.24 | Kobaz | weird |
| 21:09.54 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:10.03 | Qwell | rhg412: You're using the wrong architecture. |
| 21:10.22 | TechSmurf | yawns |
| 21:11.04 | TechSmurf | blames Qwell while he installs *now on his new pbx |
| 21:12.33 | rhg412 | thanks qwell, thought i was using 64bit... |
| 21:13.00 | BlackMaria | hi all, I have spent a lot of time on ggl, doing the suggested to no avail... |
| 21:13.07 | BlackMaria | CallToken Support required. If unexpected, resolve by placing address in the calltokenoptional <snip> requirecalltoken=no |
| 21:13.13 | BlackMaria | AsteriskNow CentOS 2.0.2 x86_64 CD |
| 21:13.22 | BlackMaria | Any ideas? |
| 21:19.10 | fubada | could someone please help me figure out why my outgoing callerid is not working |
| 21:19.15 | fubada | showing up as unknown |
| 21:21.08 | fubada | http://pastebin.ca/2149741 |
| 21:21.58 | talntid | does it work in a simple test case, of just setting the callerid(num) ? |
| 21:22.17 | talntid | setting it manually to something, instead of the big if, len statements etc |
| 21:22.27 | fubada | let me see how can I override that |
| 21:22.45 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-cwahpeadvvdhhyil) |
| 21:23.40 | fubada | exten = _91NXXNXXXXXX!,1,Set(CALLERID(all)=5555555555,${NexVortex}/${FILTER(0123456789,${EXTEN:1})},,NexVortex,) |
| 21:23.48 | fubada | will that work as you suggest? |
| 21:24.20 | p3nguin | That isn't the format for CALLERID(all). |
| 21:24.44 | *** join/#asterisk gusto (~gusto@88.128.177.106) |
| 21:25.06 | Tim_Toady | I m trying to set a channel variable from AGI, i have this: print "EXEC SET sprec_format=\${CHANNEL(audionativeformat)}\n"; |
| 21:25.13 | p3nguin | Set(CALLERID(all)=A Name <3145551212>) |
| 21:25.23 | Tim_Toady | the problem is that ${CHANNEL(audionativeformat)} is not evaluated and the variable is not set right |
| 21:25.26 | Tim_Toady | any ideas? |
| 21:25.58 | p3nguin | talntid: I didn't see your problem with AMD(). |
| 21:26.06 | TechSmurf | *now doesn't want to install from USB flash.. can't see files :( |
| 21:26.55 | talntid | p3nguin, thanks for responding. :) I have AMD in my dialplan, but it is detecting me as a machine, for the reason of: INITIALSILENCE-2500-2500 |
| 21:27.15 | talntid | http://pastebin.com/LpgmA9WP |
| 21:27.25 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 21:27.53 | gusto | hi p3nguin |
| 21:28.44 | p3nguin | talntid: Does your googletts.agi do different things for machine and for human? |
| 21:29.21 | talntid | googletts.agi just says "machine" and "human" in a text to speech way. |
| 21:29.37 | fubada | p3nguin: if you have a minute, could you take a look at my extensions.conf snip, regarding outbound caller id not working http://pastebin.ca/2149741 |
| 21:29.44 | fubada | please |
| 21:29.55 | talntid | fubada, he did ,and gave you advice... |
| 21:29.58 | p3nguin | If your initial_silence is set to 2500 and your silence before audio is 2.5s or more, it is a considered a machine. |
| 21:30.09 | talntid | p3nguin, for testing, I have set it up to 4000 |
| 21:30.28 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 21:31.18 | p3nguin | How long is the silence before audio is heard? |
| 21:31.55 | talntid | well, I set monitor on, but the first thing I hear is "Machine" |
| 21:31.58 | p3nguin | fubada: I don't do FreePBX. |
| 21:32.07 | talntid | I feel like I should hear a 2 second delay before that |
| 21:32.10 | fubada | not using freepbx :) |
| 21:32.20 | p3nguin | I doubt you wrote this dial plan by hand. |
| 21:32.26 | fubada | asterisk-guo |
| 21:32.27 | p3nguin | Too much macro crap. |
| 21:32.31 | fubada | right |
| 21:32.33 | p3nguin | I don't do GUIs. |
| 21:32.34 | fubada | its hard to understand |
| 21:32.35 | gusto | why does someone want silence being played over telephone? |
| 21:33.19 | p3nguin | gusto: Sometimes, TADs have some silence before the outgoing message is played. |
| 21:33.36 | p3nguin | ~tad |
| 21:33.36 | infobot | ACTION yells at tad for support! |
| 21:33.41 | gusto | i like writing dialplan by hand, and i found out from the book that there are even more languages one could write that in lua and so on |
| 21:33.42 | p3nguin | :( |
| 21:34.45 | p3nguin | ~tad |
| 21:34.46 | infobot | Telephone Answering Device, or just an answering machine |
| 21:36.14 | p3nguin | fubada: Set(CALLERID(all)=Some Name <3145551212>) |
| 21:36.28 | fubada | i see that, im trying to figure out where to place that |
| 21:36.36 | fubada | in the macro [macro-trunkdial-failover-0.3] |
| 21:36.46 | p3nguin | Write normal dial plan and then I'll show you. |
| 21:36.52 | fubada | heh |
| 21:36.59 | talntid | is there a better way to measure the time, p3nguin? |
| 21:37.58 | p3nguin | Can you call the number manually to determine it? |
| 21:38.18 | p3nguin | I don't know anyone with more than 4 seconds of silent lead time on their OGM. |
| 21:38.22 | fubada | unfortunately im responsible for a huge amount of other infrastructure besides this * config and dont necesarily have the time to learn * again, its been about 7 years ;( |
| 21:38.51 | fubada | which is why i chose the gui, which doesnt seem to work well and is not very well supported ;/ |
| 21:39.10 | p3nguin | Everyone here could have told you that before you chose it... if you would have just asked. |
| 21:39.15 | talntid | p3nguin, this is an outgoing call... |
| 21:39.34 | talntid | it calls out, then when I answer my cell phone, and say "hello" .. it responds "machine" even though it should say "human" |
| 21:39.35 | p3nguin | I don't understand. |
| 21:40.04 | p3nguin | Asterisk makes a call out... to a phone number... which answers. |
| 21:40.17 | p3nguin | AMD() is used to determine if the answeree is a human or a machine. |
| 21:40.19 | gusto | is listening to 'sunshine live' through dab+ terrestrial |
| 21:40.22 | p3nguin | This is typical usage for AMD(). |
| 21:40.42 | talntid | right, and when I answer, i say "hello", but it doesn't seem to register that.. as it registers me as a machine |
| 21:40.45 | p3nguin | I should have said callee. |
| 21:41.34 | p3nguin | The verbose output of the app running should show why you are detected as a machine. |
| 21:42.04 | p3nguin | I doubt it is because of the initial silence unless you really are waiting more than 4s. |
| 21:42.27 | talntid | <PROTECTED> |
| 21:42.45 | talntid | that's the ${AMDCAUSE} :( |
| 21:42.59 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 21:43.49 | talntid | also, Monitor() just records "machine" -- it doesn't hear my "hello? hello?" |
| 21:43.58 | p3nguin | If you wait less than the initial_silence value before saying hello, AND if you say less than maximum_number_of_words, AND if you wait more than after_greeting_silence after your greeting... it should think you are a human. |
| 21:44.58 | p3nguin | For example: answer, wait 1s, say hello, wait 2 seconds before saying anything else. You should be deemed a human. |
| 21:45.11 | talntid | http://pastebin.com/TH9U4eeZ |
| 21:45.17 | talntid | ok, will try |
| 21:47.02 | talntid | still called me a machine. |
| 21:47.36 | talntid | I really think it has to have something to do with the fact it's not recording from the start.. |
| 21:47.51 | talntid | i answer, wait a second, say hello, and all it records in "machine" |
| 21:47.53 | talntid | is |
| 21:48.38 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 21:49.27 | p3nguin | ANSWERING MACHINE: silenceDuration:400 0 initialSilence:4000 |
| 21:49.39 | p3nguin | I don't know if that means silenceDuration:4000 or not |
| 21:50.00 | p3nguin | I also don't know what the duration value is expressing or measuring. |
| 21:50.07 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:50.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:50.20 | talntid | latest test: http://pastebin.com/t0iPemSr |
| 21:50.23 | p3nguin | I see the problem. |
| 21:50.38 | p3nguin | You have an erroneous Answer(). |
| 21:50.51 | talntid | I didn't used to have that.. before, it was NoCDR |
| 21:51.18 | talntid | I tried that to see if it would record from the start |
| 21:51.44 | p3nguin | I would think it would start the Monitor() as soon as it was executed. |
| 21:52.16 | p3nguin | Without that Answer(), the recording should not start until there is an actual answer. And that is also the point where the intial silence begins counting. |
| 21:52.26 | talntid | ah |
| 21:52.29 | talntid | ok, i changed it to.. |
| 21:52.58 | talntid | http://pastebin.com/yJmWgN2t |
| 21:52.59 | p3nguin | The channel should not go to state "Up" until the far end answers. |
| 21:53.30 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:54.40 | talntid | does that latest paste seem more sane? it's still just showing "machine" |
| 21:54.52 | p3nguin | How do you execute this particular extension? |
| 21:55.00 | talntid | via AMI |
| 21:55.12 | talntid | i'm connecting the context to another # |
| 21:55.12 | p3nguin | with originate? |
| 21:55.40 | talntid | yes |
| 21:56.49 | talntid | http://pastebin.com/kchjuKYR |
| 21:56.51 | talntid | like this ^ |
| 21:57.47 | p3nguin | and Channel will equal something like SIP/itsp/13145551212, right? |
| 21:58.09 | talntid | yes |
| 21:58.50 | p3nguin | I suggest dropping the Wait() and running AMD() immediately after Monitor(). |
| 21:59.43 | talntid | http://pastebin.com/nutbxNec |
| 21:59.47 | p3nguin | In mine, I do NoCDR(), Set(TIMEOUT(absolute)=NNN), AMD(), GotoIf()... |
| 22:00.34 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 22:01.01 | p3nguin | That's weird. It really seems to be an issue with the initial silence. Crank it up to 20 seconds and see how it reacts. |
| 22:01.08 | talntid | k |
| 22:02.07 | p3nguin | It is showing that the duration of silence was 4000 and because if the initial silence value of 4000, you're a machine. |
| 22:02.25 | p3nguin | There is no way you were waiting 4 seconds to say hello, I'm sure. |
| 22:02.28 | p3nguin | You'd know! |
| 22:02.29 | talntid | <PROTECTED> |
| 22:02.42 | talntid | seems it maxes out at 5 seconds |
| 22:02.51 | p3nguin | I did not know that. |
| 22:02.59 | talntid | me either |
| 22:03.17 | p3nguin | Try it at 5000. |
| 22:03.20 | talntid | testing at 5000 |
| 22:03.21 | talntid | yup |
| 22:03.53 | talntid | same. hmm |
| 22:03.58 | talntid | gonna go to 4999 ;) |
| 22:04.14 | p3nguin | I'm going to do the same test against my cell number originating from the CLI. |
| 22:04.17 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 22:05.37 | talntid | yeah, no matter what... it seems to start right at the "machine" part... |
| 22:05.49 | talntid | it never records me answering and saying hello....... |
| 22:05.52 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:09.24 | p3nguin | HUMAN: silenceDuration:800 afterGreetingSilence:800 |
| 22:09.32 | p3nguin | HUMAN because HUMAN-800-800 |
| 22:09.39 | p3nguin | It works normally here. |
| 22:09.45 | talntid | what the........ |
| 22:09.55 | talntid | any way I can make mine call you? |
| 22:10.15 | talntid | could you PM me your cell#, and i'll just make it call you once, to see if it's something -else- ? |
| 22:10.32 | p3nguin | Give me just a moment and I will. |
| 22:10.35 | talntid | k |
| 22:10.47 | talntid | this thing has been kickin' my ass all day |
| 22:16.10 | rhg412 | I'm having an issue loading the res_digium_phone.so module? Getting an "undefined symbol: ast_msg_tech_unregister" error, would appreciate any tips |
| 22:17.19 | p3nguin | It called me a machine. |
| 22:17.37 | talntid | yeah, it's a jerk like that,. |
| 22:18.00 | talntid | i pasted the pastebin in your PM, so your # wasn't made public here, but so that you could review it |
| 22:18.03 | zamba | WIMPy: well, none of those sends out of germany, so those won't really help me much |
| 22:19.07 | p3nguin | When I am testing it for myself, it deems me human because of waiting at least 800ms after my greeting. |
| 22:19.21 | talntid | can you PB your code exactly? |
| 22:19.22 | p3nguin | I'll try to trick it into thinking I am a machine. |
| 22:20.03 | p3nguin | MACHINE because MAXWORDS-3-3 |
| 22:20.30 | talntid | is it recording you from the start? |
| 22:20.33 | talntid | monitor(), that is |
| 22:21.58 | p3nguin | I don't use Monitor(). I only use NoCDR(), then Set the timeout for the routine, then AMD(). |
| 22:22.50 | talntid | oh, you just added my Verbose(1,################# ${AMDSTATUS} because ${AMDCAUSE} ####################), not the whole thing? |
| 22:23.23 | p3nguin | Correct. I only added it so I could see the same info you were getting from yours. |
| 22:23.27 | talntid | gotcgha |
| 22:23.35 | talntid | can you give me your timeout line? |
| 22:23.44 | p3nguin | You want me to use yours verbatim? |
| 22:23.53 | talntid | for testing, i'd like that |
| 22:24.04 | talntid | except modify where the monitor() stores |
| 22:24.48 | talntid | maybe monitor() breaks it. I don't know |
| 22:24.51 | JacobIsre | Hi all. |
| 22:25.02 | JacobIsre | I need a little help getting Google Voice to work on asterisk. |
| 22:28.51 | p3nguin | talntid: http://pastebin.com/nipmXBxP |
| 22:28.58 | p3nguin | Now I'll test yours. |
| 22:29.12 | talntid | roger that |
| 22:29.16 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 22:32.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:32.43 | paulc | Where would I find the code for the SayNumber application? (context: I want to figure out how it strings together speech segments for numbers in different languages). I thought app_saynumber.c or similar but am drawing a blank |
| 22:33.03 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 22:36.43 | p3nguin | ANSWERING MACHINE: silenceDuration:2500 initialSilence:2500 |
| 22:36.50 | p3nguin | MACHINE because INITIALSILENCE-2500-2500 |
| 22:36.59 | p3nguin | That is with Monitor(). |
| 22:36.59 | talntid | thats what I get ,too |
| 22:37.02 | JacobIsre | so I need Freeswitch AND Asterisk to sucessfully do outbound dials with Google Voice? |
| 22:38.13 | p3nguin | Also I left in the Wait(). I'm removing the wait time now. |
| 22:38.27 | p3nguin | jacobisre: No. Just asterisk and correct configuration. |
| 22:40.03 | p3nguin | talntid: Even with Monitor(), it still works correctly for me. With the Wait(2.75) in it, it didn't work because that was longer than my initial_silence. |
| 22:40.11 | JacobIsre | ok, would u happen to know that conf? |
| 22:40.15 | talntid | :( |
| 22:40.24 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 22:40.31 | JacobIsre | Cuz I've tried making a trunk for it and it doesn't seem to want to work |
| 22:40.53 | paulc | Got it: pbx_builtin_saynumber - thanks :-) |
| 22:40.54 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 22:41.04 | talntid | JacobIsre, what have you tried? there are many tutorials online about that... |
| 22:41.17 | p3nguin | jacobisre: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 22:41.22 | gusto | TriJetScud: hi, do you have a native ipv6? |
| 22:41.24 | JacobIsre | http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add_google.html |
| 22:41.28 | p3nguin | This should be the only one needed. |
| 22:42.21 | talntid | p3nguin, the monitor sound... does it record the whole "hello" and everything? |
| 22:42.33 | JacobIsre | the one u just sent I looked at and made the changes, but I don |
| 22:42.39 | JacobIsre | DONT know where to put dialplans |
| 22:42.54 | talntid | I kinda wonder if it's because I am sending the call out via AMI, that... it waits for the person to answer, then connects the call... and there is a delay there, maybe? |
| 22:43.03 | p3nguin | talntid: It should. Let me go pull the recording and listen. |
| 22:43.04 | JacobIsre | And when I try to use the GUI it isn't clear what to put in and where to put the dialplans |
| 22:43.06 | talntid | I wonder if I can connect the call, and hear the ringing, and everything... |
| 22:43.17 | talntid | how did you originate the call via the CLI? |
| 22:43.21 | p3nguin | I'm connecting the call through channel originate. |
| 22:43.45 | p3nguin | channel originate SIP/provider/my-number extension s@fanout_call |
| 22:43.52 | JacobIsre | I do have a couple extensions that can login and ring. |
| 22:44.04 | p3nguin | Extensions don't login. |
| 22:44.27 | JacobIsre | then what is it called when you login with your extension and password? |
| 22:44.35 | gusto | well, extension is a dialplan |
| 22:44.48 | gusto | he does mean the callback extension |
| 22:44.48 | p3nguin | jacobisre: a misconception |
| 22:44.56 | gusto | the number that is matched to a peer |
| 22:45.16 | JacobIsre | so what file do dialplans go in? |
| 22:45.21 | p3nguin | extensions.conf |
| 22:45.22 | JacobIsre | extensions.conf? |
| 22:45.22 | JacobIsre | o |
| 22:45.23 | JacobIsre | ok |
| 22:45.31 | *** join/#asterisk MCarracela (~MCarracel@140.8.23.190.res.adsl.dyn.click.com.py) |
| 22:45.36 | gusto | but that is not what he is looking for |
| 22:45.36 | p3nguin | Configuration for SIP devices will go in sip.conf. |
| 22:45.48 | JacobIsre | so from this tutorial I can just go in and paste the stuff at the bottom of extensions for the dialplans |
| 22:45.56 | JacobIsre | I just have softphones. |
| 22:46.03 | p3nguin | Those are fine to use. |
| 22:46.14 | MCarracela | hello and regards to everybody from Paraguay! |
| 22:46.18 | p3nguin | You'll still have to configure a peer for them in sip.conf. |
| 22:46.21 | JacobIsre | I have 20MB DSL so it doesn't run too slow |
| 22:46.23 | gusto | MCarracela: hi |
| 22:46.29 | gusto | MCarracela: what's up there? |
| 22:46.33 | JacobIsre | ok p3nguin how do you configure a peer? |
| 22:46.37 | gusto | MCarracela: how is your retirement plan? |
| 22:46.40 | talntid | !@# :( |
| 22:46.46 | p3nguin | jacobisre: Have you read the book? |
| 22:46.48 | p3nguin | ~book |
| 22:46.48 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:46.51 | MCarracela | chaos! everywhere :D |
| 22:47.12 | MCarracela | hi gusto! |
| 22:47.13 | gusto | MCarracela: what currency do you use? |
| 22:47.25 | MCarracela | guaranies |
| 22:47.32 | gusto | that does not sound good |
| 22:47.38 | MCarracela | simbol is GS |
| 22:47.53 | gusto | GoodSource |
| 22:47.54 | MCarracela | first of all sorry for my english guys |
| 22:47.55 | gusto | :D |
| 22:48.06 | p3nguin | talntid: On the file-in.wav, I hear myself say hello. |
| 22:48.10 | gusto | MCarracela: you speak spanish? |
| 22:48.16 | p3nguin | out.wav is empty, though. |
| 22:48.23 | MCarracela | learnit alone, it work, but its lame |
| 22:48.27 | talntid | p3nguin, I don't know what is different... :( |
| 22:48.29 | talntid | this sucks. |
| 22:48.52 | p3nguin | Did you try mine? |
| 22:48.52 | MCarracela | gusto, yeah! |
| 22:49.02 | MCarracela | spanish and guarani |
| 22:49.02 | gusto | MCarracela: and some other languages? |
| 22:49.12 | MCarracela | portugues too.. |
| 22:49.29 | gusto | well, that is not in my collection |
| 22:49.59 | MCarracela | here its easy, are all languajes mixed up.. |
| 22:50.12 | gusto | of course |
| 22:50.23 | talntid | MACHINE because INITIALSILENCE-2500-2500 |
| 22:50.28 | gusto | how did you came to asterisk? |
| 22:50.41 | gusto | <PROTECTED> |
| 22:51.44 | p3nguin | talntid: Is that using mine? |
| 22:51.48 | talntid | yes |
| 22:51.59 | p3nguin | Let me pastebin my amd.conf. |
| 22:52.29 | MCarracela | gusto im trying to make run something in asterisk but i dont really know if its even possible, thats how i came here |
| 22:52.41 | p3nguin | http://pastebin.com/g06md5J5 |
| 22:53.01 | MCarracela | first time on irc chat ;d |
| 22:53.20 | talntid | identical |
| 22:53.25 | p3nguin | If yours works for me using these settings, it should work for you using the same settings. :/ |
| 22:53.38 | talntid | yeah, mine is identical |
| 22:53.50 | talntid | Asterisk 1.8.4.4~dfsg-2ubuntu1 built by buildd @ allspice on a x86_64 running Linux on 2011-09-23 10:55:02 UTC |
| 22:55.01 | p3nguin | This is the dial plan I ended up with after it always detected me as MACHINE: http://pastebin.com/CVnVzsFf |
| 22:55.10 | phix | gang! |
| 22:55.42 | phix | p3nguin: Does that cover Asterisk 1.8? |
| 22:55.47 | p3nguin | Which? |
| 22:56.46 | phix | the book |
| 22:57.00 | p3nguin | The 3rd edition is geared toward 1.8. |
| 22:57.06 | MCarracela | guys can i post an Q? i dont really know if that ok, its mi first time in this and i dont know if there are rules or something. |
| 22:57.11 | phix | ah nice |
| 22:57.12 | p3nguin | ~ask |
| 22:57.12 | 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. |
| 22:57.21 | phix | There a 4th out soon? |
| 22:57.30 | p3nguin | leifmadsen: ^ |
| 22:57.42 | phix | MCarracela: The rule is don't ask to ask a question, just ask it :P |
| 22:58.08 | p3nguin | If you only get one question, and you ask if it is okay to ask, you just blew your only hope of getting help. |
| 22:58.20 | p3nguin | Lucky for you, there isn't a one-question rule in effect today. |
| 22:58.26 | phix | Not today |
| 22:59.14 | phix | The one-question rule definitly applies to Mondays |
| 23:00.12 | MCarracela | jajajaa thanks guay, il post whats i have allready post in the help forum of elastix |
| 23:00.25 | p3nguin | ~pb |
| 23:00.25 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:00.31 | [TK]D-Fender | ^^^^^^^ |
| 23:01.12 | MCarracela | ok.. |
| 23:01.27 | MCarracela | so, im really new at this, and i have search the past 3 days on the web about it, so i dont come here to botter with some stuff that its allready somewere, but my problem is that i could find anything aboit it, im trying to make work elastix in the way that i dont have to buy anything without try it firt what i want to do is the next |
| 23:01.52 | p3nguin | We can't/don't support elastix. |
| 23:01.52 | MCarracela | 1-auto-dial, 2 play voice message, 3- finish call |
| 23:01.59 | MCarracela | please sorry |
| 23:02.02 | phix | hai [TK]D-Fender!! |
| 23:02.04 | p3nguin | If you need help with asterisk, we can help with asterisk. |
| 23:02.05 | MCarracela | asterisk |
| 23:02.42 | *** part/#asterisk rhg412 (~harrisong@68-73-223-152.ded.ameritech.net) |
| 23:02.51 | MCarracela | its confusing, the names are almost the same for me. |
| 23:03.03 | MCarracela | i mean asterisk |
| 23:03.09 | p3nguin | "elastix" vs. "asterisk" ... |
| 23:03.18 | p3nguin | Yeah, the keys are like right next to each other. |
| 23:03.34 | gusto | well, the first lesson you need to learn is, that asterisk is not elastic at all :-D |
| 23:03.36 | phix | p3nguin: yeah English isn't his/her first language |
| 23:04.12 | talntid | i want to scream. |
| 23:04.27 | gusto | it's like the opposite of a woman's vagina, there you do not know how you got there and it works, with asterisk you do not know how to get out of there when it does not |
| 23:04.39 | p3nguin | heh |
| 23:04.51 | gusto | heh |
| 23:05.05 | talntid | i just installed it on another PBX, another voip provider, in a call center I manage... |
| 23:05.11 | talntid | and, I'm a machine there too. |
| 23:05.14 | talntid | w.t.f... |
| 23:05.18 | talntid | maybe I just sound like a machine. |
| 23:05.32 | p3nguin | I guess I need to try it using AMI just to eliminate that part. |
| 23:05.40 | talntid | I'm uising originate now |
| 23:05.43 | p3nguin | oh |
| 23:05.51 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1e8e.pool.mediaWays.net) |
| 23:05.56 | talntid | thats why I asked you before... so I could eliminate that part too. |
| 23:06.00 | talntid | im freaking confused. |
| 23:06.04 | phix | BBL |
| 23:06.22 | talntid | p3nguin, mind meeting me on my PBX.. maybe a second set of eyes? |
| 23:06.31 | talntid | we can screen session |
| 23:07.12 | p3nguin | I suppose I can take a quick look. |
| 23:07.17 | talntid | do you have a SSH key? |
| 23:07.50 | talntid | if not, I can put a temp pass on |
| 23:08.04 | p3nguin | A pass would be quicker. |
| 23:09.09 | *** join/#asterisk BlackMaria (~dickery@modemcable019.198-160-184.mc.videotron.ca) |
| 23:13.49 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:14.27 | JacobIsre | im using the gui and I have ssl setup, but when I go to the site with SSL it only shows me System Status. When I go with the non-SSL url, it shows me everything. What gives? |
| 23:19.19 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 23:20.03 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 23:21.09 | *** join/#asterisk BlackMaria (~dickery@modemcable019.198-160-184.mc.videotron.ca) |
| 23:22.40 | leifmadsen | phix: p3nguin: barely started 4th edition -- will be based on Asterisk 11, so not available until near the end of the year. |
| 23:22.57 | leifmadsen | we only write ATDG books based on LTS releases. |
| 23:23.10 | leifmadsen | (that's why no 1.6.x versions or Asterisk 10 versions) |
| 23:36.31 | *** join/#asterisk Bullmoose (~Bullmoose@71-37-168-220.bois.qwest.net) |
| 23:46.55 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
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| 23:58.46 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:7963:e723:72cf:7819) |