| 00:00.30 | aut | _corey_: im technically in north miami beach |
| 00:00.47 | aut | _corey_I have something like 35/9 mbps cable for $100 |
| 00:00.57 | _Corey_ | Did you make it to AsteriskWorld last month? |
| 00:00.58 | aut | _corey_: seems reasonable... we haven't had any issues that i know of |
| 00:01.19 | p3nguin | Using PiaF is like trying to eat a plate full of vomit. |
| 00:01.25 | _Corey_ | They're hard to beat on cheap bandwidth. I'd have a backup though... :) |
| 00:01.54 | aut | _corey_: yeah, the plan is to get a t1 and split the bw for voice and data backup |
| 00:02.12 | AirGuitarist | p3nguin, i was afraid of that :( what would you use instaed for a small system with about 10 SIP phones |
| 00:02.34 | aut | _corey_: missed asteriskworld |
| 00:02.45 | AirGuitarist | i was told how great it was, and it was easy to get going. but now i'm stuck when i need to make any changes :( |
| 00:02.49 | p3nguin | airguitarist: *I* would use Asterisk. Just Asterisk, on any Linux distro I felt like using. |
| 00:03.02 | AirGuitarist | any web panel at all? |
| 00:03.20 | p3nguin | airguitarist: You obviously can't handle using vanilla Asterisk, so my recommendation would be to use AsteriskNOW. |
| 00:03.27 | _Corey_ | aut: I'd definitely go with another carrier then... You're going to have almost zero redundancy with a "PRI" from Comcast, as it's going to be delivered over the same infrastructure as your other bandwidth |
| 00:03.31 | p3nguin | No, why would i need a "web panel" for Asterisk? |
| 00:04.36 | ChannelZ | To be able to relate to Mac people |
| 00:04.41 | AirGuitarist | for now i'm too much of a n00b to type all the config files by hand. one day i'll get there! |
| 00:05.07 | p3nguin | airguitarist: AsteriskNOW is the way to go. At least you can get reasonable support for it. |
| 00:05.07 | ChannelZ | Do you just HAVE to run Asterisk or do you actually want to learn about it? |
| 00:05.39 | AirGuitarist | it's actually in production.... |
| 00:06.22 | p3nguin | airguitarist: It's WAY easier than fiddling will all the bullshit in PiaF. AsteriskNOW consists of dropping in the CDROM, booting up, selecting your installation option, waiting about 15 minutes, and rebooting into a FULLY OPERATIONAL Asterisk PBX. |
| 00:06.26 | ChannelZ | this is an existing setup you've inherited then? |
| 00:07.10 | p3nguin | airguitarist: AsteriskNOW doesn't have any scripts that you have to run when you want to change something, and then run another script to fix what the last script couldn't do right, et cetera. PiaF is a joke. |
| 00:07.13 | AirGuitarist | p3nguin, thanks, i'll give it a try. the last asterisknow install i did was about 6 years ago and then someone else who "knew more" switched our company to PBIAF |
| 00:07.20 | AirGuitarist | and now i get to clean it up lol |
| 00:07.51 | p3nguin | For 10 SIP endpoints, I'd right down the information on a sticky note and blow away the old crap. |
| 00:07.54 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 00:08.09 | p3nguin | s/right/write/ |
| 00:08.21 | joobie | hey guys.. got a polycom 320 which is always saying "in use" when it's not.. using 1.4 - any idea how to debug this? |
| 00:08.21 | AirGuitarist | lol, now that's a cool script |
| 00:08.29 | joobie | only happening for 1 phone.. previously working ok |
| 00:08.31 | AirGuitarist | never seen a regex script like that! |
| 00:09.31 | p3nguin | airguitarist: I've done a couple AsteriskNOW installs just because I was interested. It took only once to gain my approval. It's a good distro. |
| 00:10.16 | _Corey_ | +1 |
| 00:10.21 | _Corey_ | Digium keeps the junk out |
| 00:12.16 | AirGuitarist | thanks, i appreciate the advice and will install it |
| 00:12.41 | p3nguin | There's a quick start guide on asterisk.org to help you along. |
| 00:12.59 | p3nguin | It has info like the FreePBX user/password that you'll want. |
| 00:13.47 | drivefast_ | hello ppl... can any one of you offer some help building the makefile options for custom asterisk modules? |
| 00:14.22 | p3nguin | Okay, I was able to upgrade that silly phone from MGCP to SIP without any trouble at all. |
| 00:14.25 | p3nguin | This is good news. |
| 00:14.40 | drivefast_ | theres a couple of modules that i wrote, they work nicely, but to get them compiled and loaded i had to make a mess in the makefiles |
| 00:15.24 | p3nguin | Now I bet I can upgrade it from SIP to SCCP. |
| 00:21.18 | p3nguin | Sure enough. That was weird. |
| 00:21.52 | p3nguin | I guess Cisco could have documented that I need to go from MGCP to SIP and then to SCCP. |
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| 00:28.25 | bmoraca_work | has anyone ever used genband switches? |
| 00:34.24 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 00:40.42 | joobie | guys any ideas how to debug an issue where a particular SIP user is coming up as in use when i do "sip show inuse" ? Only one user, they are not in use.. ive tried reloading the phone / reloading the chan_sip module - no joy |
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| 02:17.08 | LemensTS | Are there any polycom guru's here? I was curious if in the sip.cfg file, when it has settings for IP_330, if the IP 321 falls under that or not...? |
| 02:22.20 | carrar | I would think not |
| 02:23.05 | carrar | Look in the log files the phone generates |
| 02:24.07 | LemensTS | Alot of the docs group the 321/330 together, so I wasn't sure. Ill check the logs for some info. Thanks. |
| 02:26.20 | carrar | IP_320 I would think cover that |
| 02:26.26 | carrar | But I don't know since I don't have one |
| 02:27.05 | carrar | call polycom support! |
| 02:27.16 | *** join/#asterisk philippel_mac (~p_lindhei@pool-71-188-251-53.sttlwa.fios.verizon.net) |
| 02:27.29 | carrar | Lookin their admin doc |
| 02:28.08 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
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| 02:40.03 | Blackthorn | just returned to asterisk from freepbx. got everything up and running nice and smooth. what can i do with the call logs so i can display them up nice in a web browser? |
| 02:43.18 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 02:43.32 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 02:45.51 | carrar | write a php interface to the database tables |
| 02:47.00 | Blackthorn | do i need to recomple asterisk in order to use mysql? if i've pulled from the yum reposit? |
| 02:49.20 | carrar | who knows |
| 02:49.37 | carrar | compile from scratch and use PostgreSQL! |
| 02:49.52 | carrar | err |
| 02:49.56 | carrar | compile from source and use PostgreSQL! |
| 03:09.35 | *** join/#asterisk shapr (~shapr@nat/digium/x-sviuqybgrclquhau) |
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| 03:31.10 | sbrath | If I'mnot getting the CallerID-NAME sent to a phone, and when I do a pri debug on span 1 and I see the Name come in after I ring the extension. Do i mearly have to just pad the ring time for a bit to make sure I get the callerid-name first? |
| 03:32.47 | p3nguin | It's the time before an answer that causes the problem. |
| 03:34.11 | p3nguin | The callerID is sent between the first and second rings. |
| 03:34.37 | sbrath | I've converted this site to PIAF so I guess I have to find where to pad that. |
| 03:35.13 | p3nguin | You've converted it from something that probably worked fine to something that I wouldn't run on a toaster. |
| 03:36.54 | sbrath | while I can edit files, it's hard to get someone else to do routine stuff.... I am going from a source only hand configured install to piaf... Do you have another distro that you'd prefer.. |
| 03:39.35 | p3nguin | AsteriskNOW |
| 03:39.59 | p3nguin | That's a quality piece of engineering. I would run AsteriskNOW on my own equipment. |
| 03:41.53 | sbrath | did fax detection change in 1.8 |
| 03:42.02 | p3nguin | AsteriskNOW gives you the choice at installation time of running FreePBX, the Asterisk GUI, or no GUI. |
| 03:42.22 | sbrath | I'll have to try it. |
| 03:43.01 | sbrath | I've likes piaf as it includes the source so if you want to change something, your not downloading crap tons of install tars |
| 03:43.59 | p3nguin | AsteriskNOW is built like a normal Linux distro, and Qwell compiles the RPMs for it. |
| 03:45.19 | p3nguin | PiaF just plain sucks. Stupid scripts lying around to do tasks... run one to do one thing, then run another to fix what the first one messed up. |
| 03:45.46 | sbrath | cool, do you know if fax detection changed in 1.8? I'm used to having my channel moved to the [fax] context when it detects fax signal.. |
| 03:45.59 | p3nguin | No clue. |
| 03:49.44 | *** join/#asterisk drivefast (~radu@cpe-67-49-43-2.socal.res.rr.com) |
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| 04:19.23 | sbrath | Cool, just needed a Wait(3) after Answer() and now I have caller ID again. |
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| 06:30.50 | Severian | Howdy. I have setup a ConBridge to test. It seems to work nicely. core show application ConfBridge shows an option for admin, but I have not found anything the admin can do. Can someone point me to documentation on that? |
| 06:31.11 | Severian | ConBridge -> ConfBridge |
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| 06:50.30 | gajini | how to install ARM codec on asterisk 1.4.29 |
| 06:52.39 | *** join/#asterisk ankur_6997 (~gaurav@110.224.38.58) |
| 06:53.57 | ankur_6997 | hello i would like to buy a gsm card for asterisk (4 cannel is good ) please suggest me some i need them for a small company and demonstration as a college project |
| 06:54.07 | ankur_6997 | *channel |
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| 06:56.21 | ankur_6997 | please suggest me some gsm cards for asterisk |
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| 07:01.26 | shapr | gajini: which codec? |
| 07:05.52 | gajini | AMR codec for 3 G Mobile communication |
| 07:06.06 | ankur_6997 | shapr: dude can you suggest me a gsm card |
| 07:09.31 | shapr | ankur_6997: I've never used a GSM card with asterisk, I don't have any advice. What did you turn up on Google? |
| 07:10.43 | gajini | I have used Hypermedia Device with asterisk to connect GSM network |
| 07:11.39 | ankur_6997 | lots of results |
| 07:11.51 | gajini | I haven't used GSM card with asterisk |
| 07:12.25 | ankur_6997 | shapr: dude i just want to buy a gsm card for demonstration perpose not for production so a cheaper one will be preffered |
| 07:14.21 | shapr | ankur_6997: I really know nothing at all about gsm cards. Perhaps Google can give you some info? |
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| 07:16.44 | kaldemar | ankur_6997: maybe a gateway is the way to go if you're looking for something cheap. |
| 07:17.22 | coppice | the cheapest option is probably a USB modem |
| 07:18.53 | p3nguin | I demand that you tell me every possible GSM card option that exists RIGHT EFFING NOW because I have a project that I must demonstrate. |
| 07:19.46 | coppice | well, there's the one being used to send this |
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| 07:32.33 | kleszcz | hi |
| 07:33.37 | *** join/#asterisk jkroon (~jkroon@41.54.237.106) |
| 07:33.53 | ankur_6997 | can you name some model of usb based GMS modems ? |
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| 07:38.39 | coppice | wouldn't it be faster to type that into google? |
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| 07:42.39 | ChannelZ | goo-what? what are these strange words you use? |
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| 08:00.30 | gajini | how to install AMR codec on asterisk 1.4.29? |
| 08:01.09 | coppice | with your pirate hat on |
| 08:02.06 | |TEX| | Hi. I need some assistance on a problem that is doing my head in. |
| 08:02.19 | |TEX| | Cant get any external softphones to register |
| 08:02.42 | |TEX| | and all softphones are hopeless in giving decent log reports |
| 08:02.45 | gajini | pls tell me how can I add support for AMR codec in asterisk 1.4?? |
| 08:02.55 | |TEX| | UDP port opens and checked with Nmap |
| 08:03.11 | gajini | I have been googling but without much success |
| 08:03.31 | p3nguin | I can't use AstDB lookups in hints? |
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| 08:17.00 | ChannelZ | |TEX|: do you get errors on the console? anything on sip debug? are the packets making it to Asterisk at all? |
| 08:17.05 | p3nguin | So how come variables or functions employed as variables cannot be used in hints? |
| 08:21.02 | |TEX| | Chanz: no errors on the console asterisk side |
| 08:21.17 | |TEX| | its like the packets not getting ot the box at all |
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| 08:21.34 | schmidts | good morning |
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| 08:21.36 | |TEX| | we tried the box on the DMZ also but didnt work |
| 08:21.50 | |TEX| | but did work sort of with another router |
| 08:21.52 | |TEX| | on dmz |
| 08:22.02 | |TEX| | currently using a draytek router |
| 08:24.29 | |TEX| | swapped to another draytek with same results. |
| 08:24.43 | |TEX| | Only have got it working with a cheap dlink router on DMZ |
| 08:24.55 | |TEX| | which is odd for a recommened VOIP router |
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| 09:03.49 | dijik | hi all! |
| 09:03.54 | kleszcz | hi |
| 09:04.38 | dijik | I have one question... |
| 09:05.00 | p3nguin | ~ask |
| 09:05.00 | 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. |
| 09:09.21 | dijik | ok, im install asterisk on CentOS and connect with Samsung OS 7200, then i install asterisk18-res_fax_digium, and send fax from OS 7200 to Aasterisk in RecieveFax, result - FAX session '0' is complete, result: 'FAILED' (FAX_FAILURE_PROTOCOL_ERROR), error: 'T2_TIMEOUT', pages: 0, resolution: '204x98', transfer rate: '4800', remoteSID: |
| 09:09.32 | dijik | sorry for my english |
| 09:09.49 | *** join/#asterisk chasing`Sol (~chasingSo@82.201.135.150) |
| 09:11.48 | shapr | dijik: That says the ReceiveFAX timed out because it didn't get anything from the incoming fax. |
| 09:14.13 | dijik | why? fax 100% work and recieve/send fax from/to over VoIP provider... |
| 09:14.53 | shapr | That message does not give any reason for the failure. |
| 09:15.43 | shapr | dijik: I have seen fax timeout errors with the large dialplans that are produced from FreePBX, sometimes a large Wait() is added for reasons I do not know. |
| 09:16.37 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 09:18.02 | dijik | hmmm dialplan dont have Wait() and very small, from any fax rusult identical |
| 09:18.40 | dijik | asterisk can enlarge timeout? |
| 09:21.13 | dijik | fax Panasonic KX-FL403 (analog) send fax to asterisk over OS 7200 over sip. |
| 09:26.04 | *** join/#asterisk RamaskeZ (~RamaskeZ@ip-87-86-154-74.easynet.co.uk) |
| 09:26.57 | dijik | any ideas? |
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| 09:35.06 | wdoekes2 | dijik: do you answer? I do: |
| 09:35.07 | wdoekes2 | exten => _IDX!,n,Answer() |
| 09:35.07 | wdoekes2 | exten => _IDX!,n,Playback(silence/1) ; ensure that an audio stream is started |
| 09:35.07 | wdoekes2 | ;; Fetch the actual fax (re-INVITEs with T38 if enabled) |
| 09:35.08 | wdoekes2 | exten => _IDX!,n,ReceiveFAX(${filename}.tiff) |
| 09:37.03 | dijik | exten => 498,1,Answer() exten => 498,n,ReceiveFAX(/var/spool/asterisk/tmp/${UNIQUEID}.tif,dfs) exten => 498,n,Hangup() |
| 09:39.10 | dijik | asterisk18-res_fax_digium.i386 1:1.8.0_1.3.0-1_centos5 installed |
| 09:40.12 | dijik | all programs actual.... |
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| 09:42.04 | dijik | <PROTECTED> |
| 09:47.50 | dijik | sometimes writes > Channel 'SIP/192.168.8.200:5060-00000000' fax session '0', [ 000.419469 ], channel sent 21 frames (420 ms) of silence. |
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| 10:01.04 | dijik | any ideas? |
| 10:04.06 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
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| 10:30.03 | gajini | can you tell me , how to install AMR codec on asterisk 1.4.29? |
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| 10:40.48 | *** join/#asterisk dijik (5856c365@gateway/web/freenode/ip.88.86.195.101) |
| 10:43.01 | dijik | i solved the problem recieving faxes, and i have question: "how to increase timeout in recievefax()?" |
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| 11:11.12 | *** part/#asterisk gajini (~gajini@117.196.152.136) |
| 11:11.38 | *** join/#asterisk gajini (~gajini@117.196.152.136) |
| 11:11.56 | gajini | hi |
| 11:12.48 | gajini | can u tell me , how to install AMR codec on asterisk 1.4? |
| 11:14.47 | dijik | http://www.howtonix.com/amr-codec-for-asterisk-1-4-and-1-6/ how to install AMR codec |
| 11:16.13 | gajini | Thank you, I would like to know, is it licensed codec? |
| 11:16.31 | dijik | i dont know(( |
| 11:17.19 | coppice | of course its not a licenced codec |
| 11:17.33 | gajini | sorry this link is down,http://sip.fontventa.com/svn/asterisk/ fontventa |
| 11:18.22 | dijik | see http://sip.fontventa.com/ |
| 11:19.13 | gajini | they said, Due to a server failuire the old web is currently down. |
| 11:19.38 | gajini | so i cant download the codec from there |
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| 11:20.54 | dijik | go to http://www.google.ru/search?hl=ru&newwindow=1&client=firefox&hs=Xn3&rls=org.mozilla:ru:official&&sa=X&ei=ctZsTeT2JcegOrWcmdkH&ved=0CCkQBSgA&q=amr+codec+asterisk+1.4&spell=1 |
| 11:22.15 | gajini | i already google it, but no luck |
| 11:22.39 | coppice | what do you want amr? |
| 11:22.59 | dijik | see asterisk video, link on fontventa |
| 11:23.05 | coppice | why do you want amr? |
| 11:23.09 | gajini | this codec is available for asterisk 1.6 and 1.4 |
| 11:24.01 | gajini | <coppice>: connect with 3G mobile network with asterisk |
| 11:24.22 | coppice | how does amr help with that? |
| 11:26.32 | dijik | how to increase timeout in recievefax()? |
| 11:26.52 | gajini | <dijik>asterisk video link also not working |
| 11:27.47 | gajini | <coppice> we have mobile dialer software, which is communicating with asterisk server |
| 11:27.59 | dijik | work... link on left https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/ |
| 11:29.49 | gajini | <dijik> i got it |
| 11:30.35 | coppice | gajini: the usual interfaces to 2G and 3G networks take linear data and compress it themselves. why do you need amr in asterisk? |
| 11:31.05 | coppice | well, to amr, fr, efr, or hr as appropriate |
| 11:31.05 | gajini | <dijik>Thank you, Lets try with this patch |
| 11:33.29 | gajini | <coppice>: it can use less bandwidth than the traditional GSM codec (AMR is variable bit rate), you tend to save 50% of bandwidth |
| 11:35.20 | coppice | what is your interface to the 3G network? |
| 11:39.34 | gajini | we dont have interface, we just trunk with ISTP |
| 11:39.53 | gajini | <coppice>:we dont have interface, we just trunk with ISTP |
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| 11:55.18 | dijik | how to increase timeout in recievefax() without recompiling? |
| 12:05.22 | wdoekes2 | dijik: care to share how you resolved the issue? |
| 12:06.38 | dijik | scan fax in fax memory and after send to asterisk |
| 12:07.40 | dijik | This method does work once((( |
| 12:09.54 | dijik | the real solution - is to increase waiting time |
| 12:17.37 | Dovid | anyone from Germany here ? |
| 12:18.00 | WIMPy | #asterisk.de |
| 12:18.23 | Dovid | thanks |
| 12:18.24 | schmidts | Dovid only austria ;) |
| 12:22.01 | coppice | dijik: try using spandsp instead of the digium fax module |
| 12:22.58 | Dovid | WIMPy: Did you get my PM ? |
| 12:23.24 | WIMPy | Yes, but I didn't see a question so far, anywhere. |
| 12:24.21 | dijik | <coppice> this is what application? |
| 12:25.06 | coppice | dijik:install spandsp, and remove the digium fax module |
| 12:25.35 | dijik | <coppice> spandsp installed |
| 12:26.17 | coppice | then get rid of digium fax, and use spandsp |
| 12:27.01 | coppice | make sure its a recent spandsp. not some last century version from a distro |
| 12:28.39 | dijik | spandsp.i386 0.0.6-0.1.pre12.el5 installed |
| 12:29.07 | coppice | that should be ok |
| 12:31.28 | dijik | ok i try it |
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| 12:34.43 | *** join/#asterisk giuseppe`` (~giuseppe@157.27.141.26) |
| 12:34.50 | giuseppe`` | hello guys |
| 12:35.14 | giuseppe`` | anyone have experience with ASTDB on Mysql ? |
| 12:44.49 | schmidts | giuseppe by default there is no way to use astdb with mysql, Olle Johanson has a branch which make this possible but its not in the normal source tree of asterisk |
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| 12:55.48 | GreatSUN | re |
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| 13:11.00 | giuseppe`` | schmidts: thank schmidts, do you know any way to perform a SQL query when an ATA make a register? i need it to perform routing in my dev-cluster.. |
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| 13:14.46 | russellb | giuseppe``: the only way to do that would be to watch for registration events on the manager interface. |
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| 13:18.38 | giuseppe`` | russellb: well i know.. or look messages into a pipe.. and parse it.. ok.. Thanks! |
| 13:21.10 | russellb | np |
| 13:21.17 | russellb | no better way, sorry |
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| 13:24.23 | jkroon | hi all, is there via the AMI a way to determine whether an FXS DAHDI line is onhook/offhook? The closest I can get is the DAHDIShowChannels function which gives alarms but doesn't seem to reflect anything with respect to hook state? |
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| 13:27.28 | tnt | Hi. Just wondering if anyone tried asterisk with the new SIP client in android gingerbread ? For some reason I have audio when placing a call _to_ the phone (and I get bidir audio, all fine), but when placing a call _from_ the phone then I have no audio at all in either direction. |
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| 13:29.29 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 13:31.52 | dmz | morning y'all! i am trying to trunk 2 * boxes together; i have iax setup and they are seeing each other; i have a dialplan sending the call to the 2nd pbx; both are on the same switch but when I call into the 2nd system all i hear is jittery, if any sound at all; the codec is ulaw and i can use each * system seperately; any suggestions? |
| 13:32.32 | leifmadsen | dmz: do you have a jitterbuffer enabled at all? |
| 13:32.58 | leifmadsen | there was a report yesterday on the bug tracker that using the jitterbuffer may cause audio to break up (which is exactly what it's intended to resolve) |
| 13:33.19 | leifmadsen | you may want to disable the jitterbuffer on both sides and see if that is the problem (which helps to verify the issue that was reported as well)( |
| 13:34.53 | jkroon | leifmadsen, very strange. i haven't had that, i have however gotten that effect when using trunking without timestamps! |
| 13:35.32 | leifmadsen | jkroon: I don't use iax2 so I don't run into any of those issues :) |
| 13:35.50 | jkroon | and to answer my own guestion, DAHDIShowChannels will have a Channel set if the channels is "offhook" |
| 13:36.01 | leifmadsen | dmz: jkroon made me thing of something as well -- might want to make sure internal_timing=yes is enabled in asterisk.conf (will require restart) |
| 13:36.14 | jkroon | leifmadsen, only way to go when you need to get more than 8 concurrent calls on low-BW link. |
| 13:36.28 | jkroon | leifmadsen, that is an absolute MUST yes. |
| 13:36.38 | jkroon | dmz, you reading all that? |
| 13:37.21 | jkroon | biggest issue i've had recently was a flaky network card that dropped approximately 1% of RX packets. |
| 13:37.47 | jkroon | immensively difficult to diagnose, extremely annoying effect on sound. |
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| 13:42.41 | leifmadsen | jkroon: luckily I don't have the situation :) but ya the network card thing would be hard to diagnose |
| 13:49.00 | jkroon | grumbles something nasty about internet providers that only provides 4Mb/512Kbps links at reasonable prices. |
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| 13:52.35 | dmz | no i dont have jitterbuffer |
| 13:52.56 | dmz | let me try the internal timing |
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| 13:54.28 | *** join/#asterisk bratner (~bratner@95.211.21.37) |
| 13:54.45 | dmz | i'm trying to change from iax to sip since all my phones are sip but it's not sending inbound call to the configured context |
| 13:54.57 | dmz | so it isn't finding the extension i'm telling it to go to |
| 13:55.30 | bratner | Hi! When i'm recording queue calls with MixMonitor , where do the recordings go? which directory? |
| 13:56.53 | leifmadsen | bratner: /var/spool/asterisk/monitor perhaps |
| 13:58.30 | bratner | yep! thanks leifmadsen! |
| 13:58.41 | jkroon | dmz, if you're working on a LAN I (and keep in mind I am a supporter of IAX/2) would recommend you stick with SIP. |
| 13:59.16 | jkroon | IAX/2 has it's uses - unless you're struggling with network throughput it is probably not the tool for the job. |
| 13:59.17 | dmz | i'm trying to get it working but for some reason it's not connecting to the context; when i enable sip debugging it doesn't show the context it is trying to use or anything about context |
| 13:59.33 | jkroon | pastebin your configs. |
| 14:00.09 | dmz | ahh it was using the default context; not the peer context |
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| 14:00.15 | dmz | ok it's connecting now but not hearing any osund |
| 14:00.37 | dmz | sound |
| 14:01.44 | dmz | so close... |
| 14:01.54 | tnt | mmm, what could cause no audio even tough a tcp dumps shows RTP data in both direction (and dumping / playing the rtp payload of those dumps shows there is good voice data in them) ? |
| 14:02.51 | dmz | i'll get a tcpdump |
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| 14:09.54 | *** join/#asterisk flashdeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 14:13.35 | flashdeluxe | Hi! I care for several asterisk servers of different customers and i wanted to ask, if there is any tool to administrate mulitple asterisk servers? |
| 14:14.51 | iprouteth0 | asterisk centralized management? no idea |
| 14:16.52 | dmz | well at least I have calls coming into 2nd box via sip; but it appears that no voice is going either way; there is no voice data in the tcpdump |
| 14:17.40 | flashdeluxe | mhhh :( |
| 14:18.49 | iprouteth0 | dmz: are both on same subnet? |
| 14:18.58 | dmz | yes |
| 14:20.21 | iprouteth0 | where is call originating from? |
| 14:21.55 | dmz | i've got 2 * boxes; the 1st is working great and been using it for years; the 2nd is a project called bigbluebutton, it is a presentation system (kinda cool); i'm trying to get it so my voice users can call into the bbb conference system to join meetings; a local java / flash client integrated w/bbb is working good so i know internally it works; but when i call in from outside system it connects and says it is working but i don't he |
| 14:21.55 | dmz | ar anything; btw inbound calls are w/polycom phones |
| 14:23.43 | iprouteth0 | are both * systems on public IPs? |
| 14:24.41 | dmz | yes |
| 14:24.54 | dmz | but that shouldn't matter as they are only connecting to "each other" |
| 14:25.08 | dmz | they are on the same switch; same subnet; same vlan |
| 14:25.43 | dmz | when i connected via iax i heard about 2 sec of voice (scratchy voice) then nothing; now i'm using sip and i get no audio |
| 14:25.49 | dmz | it is all ulaw so no codec problems |
| 14:25.53 | iprouteth0 | so are voice users calling from on-net or off-net? |
| 14:26.04 | dmz | both |
| 14:26.10 | dmz | but my test is on-net |
| 14:26.32 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:26.42 | dmz | i dial into my existing system; works great; i call to the bbb system and it connects; hmm let me check where it thinks it is communicating with |
| 14:26.57 | iprouteth0 | FYI I've no idea if i can help at all. I'm really just learning asterisk myself, but I do like to work my way to voice problems as I do support for Telco/ISP |
| 14:27.35 | dmz | :) |
| 14:28.04 | dmz | i've been playing w/* for about 5 years now; i like to use irc to bounce things off people as I have noone around here to bounce things off off (besides just hitting them) :) |
| 14:28.31 | iprouteth0 | Call from off-net into BBB system connects and streams audio successfully? |
| 14:28.59 | jkroon | dmz, check the ast CLI very carefully. |
| 14:29.14 | dmz | about to try that next; haven't setup direct connectoin to bbb* from phone (next step) |
| 14:29.25 | dmz | the cli plays a few gsm files when it connects |
| 14:29.29 | iprouteth0 | We have a number of voip based configuration we provide to customers, so it's nice to see how different people implement different things |
| 14:29.33 | dmz | so the cli says: playing 'conf-getconfno.gsm' |
| 14:29.37 | jkroon | no, checking the CLI should be the _first_ step. |
| 14:29.47 | dmz | i'm in the cli debugging it |
| 14:29.50 | jkroon | but no sound? |
| 14:29.53 | dmz | correct |
| 14:30.18 | jkroon | ok, during that stage - what does "sip show channel" show you? does it actually have a codec assigned to that channel? |
| 14:30.28 | dmz | let me check |
| 14:30.34 | jkroon | and rtp debug? |
| 14:31.19 | dmz | it has ulaw assigned |
| 14:32.41 | dmz | hmm the rtp debug shows me what is wrong |
| 14:32.43 | dmz | thanks :) |
| 14:33.08 | dmz | it is tryign to transfer the call and the 2nd box doesn't know the path to it; i had thought it would stay connected through the 1st system and route the call |
| 14:34.24 | dmz | hmm |
| 14:34.40 | dmz | ok so inbound call via voip works great |
| 14:34.54 | dmz | but when using a phone connected to main * box it tries to rtp bac to the phone ip |
| 14:35.13 | iprouteth0 | reinvites? |
| 14:35.46 | dmz | what does the reinvite enable/do? |
| 14:36.35 | iprouteth0 | not sure if it would completely apply |
| 14:38.03 | iprouteth0 | I believe it's mostly for NAT, but I think you said you're trying right now with call originating from 1st * box..? |
| 14:39.00 | iprouteth0 | whether the endpoints connect directly without * in between or not |
| 14:40.25 | iprouteth0 | again I am learning much of this myself, but am half split between voip and tdm/pots |
| 14:40.42 | dmz | yeah that worked |
| 14:40.57 | iprouteth0 | it did? sweet |
| 14:41.22 | dmz | yeah it had the call coming from corp* to bbb* not reinvite so it didn't try to connect to client ip |
| 14:41.30 | dmz | very cool; now i don't need webex anymore :) |
| 14:42.35 | iprouteth0 | lol. happy to help. Proud of myself for putting correct critical thinking to it XD. am surprised with myself as well |
| 14:42.43 | dmz | :) |
| 14:42.52 | dmz | it's always the 2nd set of eyes that helps |
| 14:43.37 | dmz | i had a wierd apache proxy problem yesterday and it turned out it had proxyajp & proxyhttp enabled causing conflicts; didn't see it until a 2nd admin turned debugging on (he didn't know what output meant but i saw it immediately) teamwork :) |
| 14:44.16 | iprouteth0 | indeed it is. My jobs main responsibility is network monitoring during the nightshift, and my teammates and I rely on each other as we all have a varied skill matrix |
| 14:44.19 | dmz | http://bigbluebutton.org/ this is the project i'm using; simple ubuntu install (like 10 min) and we are able to replace webex w/our own system |
| 14:44.41 | dmz | we're getting ready to start hiring some nightshift people :) |
| 14:45.06 | dmz | i'm tired of diving out of bed w/alerts (luckily there aren't that many) :) |
| 14:45.25 | iprouteth0 | we've got many many |
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| 14:45.51 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:45.55 | dmz | we're a small company, less than 100 customers but > 100 segmented networks (highly customized/secured networks) |
| 14:46.12 | dmz | ok off to finish paperwork for pci audit |
| 14:47.14 | wao | dmz: nice |
| 14:47.15 | iprouteth0 | Thats still challanging I'm sure. My company serves 29 us states I think it is, so our network is quite large as it is both alot of data and TDM voice stuff, so we have alot of alarms... gets kinda messy since maint work is done on nightshift all over |
| 14:47.55 | dmz | ah we're more data-center based; have one in dc & another in sfo but we have customers world wide |
| 14:49.49 | dmz | wao your in jp? i'd love to visit again; where in jp are you? |
| 14:50.40 | iprouteth0 | Data centers is kinda big right now I guess. I know we just acquired a data center company in MN and another one awhile ago before that |
| 14:52.25 | iprouteth0 | dmz: do you do much VPN stuff then? |
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| 14:55.16 | jkroon | dmz, you can set directmedia=no on the peer/user/friend definition. |
| 14:55.46 | jkroon | i get confused with those three in SIP world - it makes _perfect_ sense for me in iax land, but sip has me confused. |
| 14:57.32 | iprouteth0 | heh. just fixed my outbound rule |
| 14:58.06 | jkroon | he, i've got one retarted client that insists on not buying two SIP phones. |
| 14:58.08 | jkroon | *sigh* |
| 14:58.34 | jkroon | what a pain to add FXS channels into the mix on my highly focused around SIP extensions system. |
| 14:58.59 | dmz | iprouteth0 every one of our customers has a vpn; most of our customers are pci/hipaa/etc and require 2fa to even access their systems |
| 14:58.59 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:59.33 | neurosys | in 1.4, can you transfer outside callerID when xfering to different extensions? |
| 14:59.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:59.57 | iprouteth0 | I sorta figured. Just started working with cisco VPN stuff a few months ago |
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| 15:01.02 | jkroon | neurosys, blind xfers yes. |
| 15:01.08 | iprouteth0 | get them some ATAs instead jkroon |
| 15:01.19 | iprouteth0 | got my linksys ATA for $20 |
| 15:01.27 | jkroon | iprouteth0, suggested that. client is full of shit. |
| 15:01.32 | neurosys | jkroon: Right. for some reason these SPA-942's wont do it even with blind xfer |
| 15:01.34 | iprouteth0 | aw, damn |
| 15:01.34 | jkroon | but pays me enough to go to the effort. |
| 15:01.57 | jkroon | neurosys, probably because they still do an attended xfer even if you don't. |
| 15:02.20 | jkroon | ie, creates a seperate SIP channel and then requests * to bridge them using REFER or something. |
| 15:03.02 | bratner | what is the meaning of this? WARNING[19470]: translate.c:155 framein: no samples for g729tolin |
| 15:03.05 | bratner | am i missing a file? |
| 15:03.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:05.06 | jkroon | yes, a G.729 codec license. |
| 15:05.41 | jkroon | no wait, that just doesn't get any samples in, hmm, getting voice frames not containing any actual g729 data? |
| 15:06.30 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:07.02 | iprouteth0 | I have a directory full of g729 sounds |
| 15:09.36 | *** join/#asterisk rene (~Rene@109.75.224.121) |
| 15:09.46 | rene | Hi folks! |
| 15:10.32 | rene | is it possible to have asterisk identify the caller to the voicemail, and skip the password-prompt and mailbox-prompt? |
| 15:11.02 | rene | i could not find this anywhere (or i just did not know what to look for :-) |
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| 15:13.51 | jkroon | iprouteth0, then perhaps it needs to convert one of those g729 to something else for some reason (eg, incoming channel is gsm?)? |
| 15:14.26 | iprouteth0 | transcoding can be a headache.... :P |
| 15:15.13 | *** join/#asterisk n3hxs (~ed@96.234.181.58) |
| 15:15.38 | iprouteth0 | oh and I used 729 use flag when building, so I know why I have the g729 files |
| 15:15.42 | iprouteth0 | lol |
| 15:18.32 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 15:22.31 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6) |
| 15:26.42 | Amnesia | hm, anyone here who has any experience with a portech voip gateway> |
| 15:27.55 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:28.49 | Amnesia | http://pastie.org/1621000 is what I keep getting |
| 15:28.59 | benngard | rene: yes it is |
| 15:30.05 | aberrios_ | I'm going mad, trying to use REPLACE function to replace all commas in a string with \, and failing |
| 15:30.33 | aberrios_ | exten => 777,n(tobook),Set(properStreet=${REPLACE(properStreet,\,,\\\,)}) |
| 15:31.30 | benngard | rene: http://pastebin.com/LDpVMejn |
| 15:31.48 | aberrios_ | ah replace only replaces characters and not subtrings |
| 15:31.49 | aberrios_ | duh! |
| 15:38.11 | jkroon | how can I obtain the accountcode for a DAHDI channel? |
| 15:47.17 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:49.37 | rene | benngard: how does this callerid-thing work with the voicemail? |
| 15:50.39 | rene | how does the voicemail know that eg. [1234] is calling to it's voicemail? |
| 15:53.03 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 15:53.26 | omani | how do I realize sms messaging sending and receiving with gsm mobiles to asterisk box in and out? |
| 15:54.42 | Gugge | omani: what do you want to do? |
| 15:55.11 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 15:58.52 | rene | benngard: yeah.. i think i got it :-) |
| 15:59.48 | rene | how do i catch the current [caller] in the extensions.conf? |
| 16:00.59 | *** join/#asterisk hairyraven (~nobody@95.72.153.170) |
| 16:02.47 | benngard | rene: sorry was at the phone, do u understand it? |
| 16:04.19 | Dovid | http://pastebin.com/8UKBreBt |
| 16:10.46 | *** join/#asterisk wonderworld (~ww@port-92-201-111-225.dynamic.qsc.de) |
| 16:11.58 | rene | benngard: i use a cisco 7960 |
| 16:12.08 | rene | now i managed it .-) |
| 16:12.25 | rene | i created a separate extension to all phones :-) |
| 16:13.32 | *** part/#asterisk rene (~Rene@109.75.224.121) |
| 16:13.49 | *** join/#asterisk ubuspire (~ubuspire@41.190.129.207) |
| 16:15.20 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:16.33 | neurosys | Asterisk: The Definitive Guide 3rd edition... delayed :( |
| 16:21.30 | *** join/#asterisk screenn (~screenn@users-nat.more.com.ua) |
| 16:21.58 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 16:22.56 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:23.39 | bmoraca_work | has anyone here dealt with VoIP Logic before? |
| 16:35.55 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:45.00 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 16:53.13 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-147.mobile.uci.edu) |
| 17:07.21 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 17:07.43 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qiiwegovzwqosyil) |
| 17:07.49 | LemensTS | hello, do any of you know if you modify the polycom dial strings, if it will dial out without having to press the Dial key? |
| 17:11.50 | _Corey_ | LemensTS: That's what they're for |
| 17:12.13 | _Corey_ | If you have to press 'Dial' it means the phone isn't configured correctly |
| 17:17.09 | LemensTS | _Corey_: thanks |
| 17:18.55 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 17:27.12 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 17:30.31 | *** join/#asterisk umay (~chris@71-212-199-100.hlrn.qwest.net) |
| 17:32.26 | bmoraca_work | so no one here has done anything with VoIP Logic for managed broadsoft? has anyone worked with any other managed broadsoft provider besides Commpartners? |
| 17:33.38 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:47.27 | *** join/#asterisk drivefast (~radu@99.92.126.154) |
| 17:55.20 | drivefast | hello ppl, does any one of you have any experience with building asterisk modules? i have a couple of them that need to use external libraries, and i'm having a hard time making decent settings in the makefiles, any help would be great |
| 17:59.00 | voxter | any of you seen an instance in which polycom phones will request DHCP but then ignore the OFFER? |
| 18:03.09 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 18:03.25 | drivefast | !help |
| 18:24.40 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:40.36 | Benwa | is it possible to use a dynamic ip adress with a dyndns on the asterisk server for sip calls ? |
| 18:41.00 | p3nguin | Yes. |
| 18:41.12 | Benwa | ok thanks |
| 18:41.22 | Benwa | i continue my serch then ... |
| 18:41.35 | Benwa | *search |
| 18:42.00 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 18:42.22 | drift- | how can i do the verbose 5 thing?/ |
| 18:42.24 | drift- | set verbose 5 ? |
| 18:42.42 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:45.18 | p3nguin | Nothing changes above verbose level 4, so there's no reason to go to level 5. |
| 18:45.25 | p3nguin | core set verbose 4 |
| 18:46.08 | p3nguin | 3 is better for standard monitoring. |
| 18:46.29 | p3nguin | It doesn't spew the dnsmgr_lookup crap continuously. |
| 18:46.31 | Qwell | Benwa: in sip.conf, look at externhost/externrefresh |
| 18:46.44 | Benwa | <PROTECTED> |
| 18:47.18 | p3nguin | You forwarded UDP on those ports and not TCP, right? |
| 18:47.21 | Benwa | Qwell: i use freepbx so i cannot edit sip.conf |
| 18:47.27 | Benwa | right |
| 18:47.33 | Qwell | Benwa: #freepbx then |
| 18:47.37 | Benwa | p3nguin: right |
| 19:01.16 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 19:01.40 | *** join/#asterisk manji (~manjiki@adsl-141-175.adsl.ntua.gr) |
| 19:01.48 | candrews | It appears that CURLOPTS doesn't have a "followlocation" option: http://www.asterisk.org/astdocs/api/func__curl_8c-source.html |
| 19:01.50 | *** join/#asterisk wonderworld (~ww@port-92-201-111-225.dynamic.qsc.de) |
| 19:02.05 | candrews | So I'm confused - if there's no followlocation, then how is "maxredirs" ever used? |
| 19:05.33 | wdoekes2 | hmm :) |
| 19:12.33 | p3nguin | With an off-site PBX, having 50 phones in the office and all the phones connect outbound through one IP address... is there any possible way to do any sip debug for only a single device in the office? |
| 19:15.35 | candrews | reported https://issues.asterisk.org/view.php?id=18902 |
| 19:28.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:30.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:31.20 | *** join/#asterisk alecdavis (~alec@202.36.76.21) |
| 19:31.52 | *** part/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 19:34.53 | *** join/#asterisk kriegerod (~krieger@user-204.45.infomir.com.ua) |
| 19:36.10 | *** join/#asterisk kribyl (~kribyl@mail.attendit.se) |
| 19:37.59 | kribyl | Anyone aware if it is RFC-valid to begin a call-id with whitespace? |
| 19:40.32 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 19:45.25 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:45.53 | *** join/#asterisk mintee (1000@static-64-115-220-213.isp.broadviewnet.net) |
| 19:46.42 | mintee | Hum... After dissecting a few of my dial plans, I've determined that Ringing() causes FaxDetection to fail. |
| 19:46.49 | mintee | anyone else notice this? |
| 19:47.09 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 19:47.25 | *** join/#asterisk af_ (~getsmart@78.134.22.22) |
| 19:48.56 | bmoraca_work | use nvfaxdetect |
| 19:49.07 | bmoraca_work | or nvbackgrounddetect |
| 19:49.15 | mintee | i'm using hardware detection |
| 19:49.35 | mintee | this all worked fine until just recently... upgrading hylafax and * |
| 19:49.44 | bmoraca_work | right...and nvfaxdetect will allow you to play ringing tones and properly detect fax tones |
| 19:50.26 | mintee | are you saying that nvfaxdetect is more acceptable than hardware detection in the community? |
| 19:50.41 | bmoraca_work | i can't speak for the entire community |
| 19:50.49 | bmoraca_work | but in my experience, nvfaxdetect has never let me down |
| 19:51.02 | mintee | hum... |
| 19:51.11 | mintee | interesting... |
| 19:51.12 | bmoraca_work | i use it in virtual environments with SIP trunks and it works just as well as when used with TDM trunks in a dedicated environment |
| 19:51.16 | bmoraca_work | that's good enough for me |
| 19:51.26 | mintee | wonder if I'd have to turn off the hw detection. |
| 19:51.59 | bmoraca_work | the one thing about nvfaxdetect that i don't necessarily like is that the channel needs to be answered first...so it can throw off CDRs |
| 19:52.18 | mintee | ya, i do that anyway |
| 19:52.24 | bmoraca_work | for me, it's not that big a deal, though, because I don't bill origination usage for my hosted PBXes |
| 19:52.36 | mintee | i Answer(), Ringing(), WaitExten(4) |
| 19:52.58 | mintee | but the faxes would only come in about 10% of the time |
| 19:53.06 | *** part/#asterisk sorressean (~ty@host-98-127-117-55.gdj-co.client.bresnan.net) |
| 19:53.07 | mintee | i removed Ringing() and I get 100% |
| 19:53.12 | bmoraca_work | are you redirecting to an IAXModem? |
| 19:53.17 | mintee | ya |
| 19:54.59 | bmoraca_work | http://www.voip-info.org/wiki/view/NVFaxDetect |
| 19:55.02 | bmoraca_work | works pretty well |
| 19:55.21 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 19:55.23 | bmoraca_work | you can also use nvbackgrounddetect in place of background() in IVRs to detect faxes during IVRs |
| 19:58.25 | mintee | :/ |
| 19:58.31 | mintee | I didn't build it in |
| 19:58.42 | bmoraca_work | it's easy enough to add |
| 19:59.19 | bmoraca_work | you can also get a statically linked module |
| 20:03.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:04.08 | mintee | meh... I don't wanna put a 3rd party module in |
| 20:04.18 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k892.webspeed.dk) |
| 20:04.20 | mintee | I'll keep hunting down a fix for my problem |
| 20:04.28 | mintee | maybe playtones(ring) |
| 20:04.34 | E-bola | w00t finally got a dump file and core show locks from my crashing asterisk |
| 20:09.10 | mintee | damnit, playtones doesn't work either |
| 20:09.13 | mintee | weird |
| 20:13.46 | ubuspire | Hi all...In libss7,how do i determine which channel is being used for a particular call?? |
| 20:21.21 | E-bola | Can you inhclude stuff in iax.conf? |
| 20:27.47 | ubuspire | hello |
| 20:29.34 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 20:33.33 | _Corey_ | E-bola: Sure |
| 20:34.49 | E-bola | _Corey_: thanks found it |
| 20:34.55 | E-bola | Im reconfigureing my box |
| 20:35.04 | E-bola | and autokill / qualify in iax.conf got me a bit pussled |
| 20:35.22 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 20:35.36 | E-bola | the short explanation is written as if: If you have autokill=yes in ur general part, then thats the same as having qualify=yes for all peers |
| 20:35.40 | E-bola | is that correct understood? |
| 20:44.01 | *** join/#asterisk espiceland (~erin@207.98.195.107) |
| 20:44.42 | *** join/#asterisk hogejo (~hogejo@catv-89-134-117-13.catv.broadband.hu) |
| 20:45.28 | hogejo | Hello guys! Just installed AsteriskNOW, but I got stuck on the web gui. It asks for a user:pass, but nothing works. I only set the system root password on CentOS, but that one does not work :S |
| 20:46.32 | *** join/#asterisk crakrjak (~merc@rrcs-70-62-156-154.central.biz.rr.com) |
| 20:47.20 | Qwell | freepbx/fpbx |
| 20:48.14 | Katty | peeks in |
| 20:49.14 | hogejo | Qwell: nope, does not work. I am on the asterisk GUI, not the freepbx one. |
| 20:49.25 | Qwell | admin/password |
| 20:51.02 | hogejo | Cool, thanks! |
| 20:51.11 | leifmadsen | Qwell: you hacked the gibson! |
| 20:54.45 | nestAr | don't forget god.. |
| 20:54.47 | nestAr | :) |
| 20:56.02 | leifmadsen | does anyone actually use (or is allowed to use by the software) 3 character passwords? |
| 20:57.41 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 20:58.14 | Dovid | hello all |
| 21:02.16 | jaytee | hi |
| 21:03.53 | _Corey_ | leifmadsen: I'm sure nobody's going to raise their hand to that one, guilty or not... ;) |
| 21:15.10 | dmz | the closest i've ever seen is atm card pin passwords which are generally accepted at 4 digit pin |
| 21:15.17 | dmz | but never a system/user password |
| 21:15.24 | dmz | voicemail maybe |
| 21:26.00 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 21:26.13 | nestAr | 6 is about my limit.. on non-crucial accounts |
| 21:27.56 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:29.54 | *** join/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 21:42.36 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 21:42.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:44.26 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:58.43 | citywok | anybody got an easy command to copy all files of type php, recursively, from one location to another, preserving folder structure? unix of course. |
| 21:59.04 | WIMPy | find |
| 21:59.24 | citywok | yea, playing with it |
| 22:00.51 | E-bola | copy all files , then remove all files not of type php |
| 22:01.00 | E-bola | if its not big files thats one dirty way of doing it :) |
| 22:01.04 | WIMPy | 'd still like a structure only option for cp. |
| 22:06.54 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:10.09 | _Corey_ | I've got a Perl AGI question... maybe someone here can answer this definitively. I'm grabbing SIG{INT}, SIG{HUP}, and SIG{PIPE} so that I can continue executing AGI's ala the old DeadAGI... |
| 22:10.36 | _Corey_ | On occasion, I'm getting "stuck" AGI's and have a feeling one of those three is redundant/problematic |
| 22:11.13 | leifmadsen | _Corey_: what version of asterisk? |
| 22:11.36 | _Corey_ | I've seen it since switching from DeadAGI() back with 1.6, but currently with 1.8. I |
| 22:11.38 | _Corey_ | er |
| 22:11.44 | _Corey_ | I'm sure it's my method |
| 22:11.50 | leifmadsen | _Corey_: because at least in 1.8 you should not be using DeadAGI() -- just AGI() in all places |
| 22:11.56 | _Corey_ | I was using DeadAGI() everywhere before 1.6 and it was fine |
| 22:12.09 | _Corey_ | Yeah, that's why I switched... |
| 22:12.20 | _Corey_ | I was using an $AGI->setcallback before |
| 22:12.41 | _Corey_ | that worked reliably, but with: $SIG{INT} = \&onHangup; |
| 22:13.25 | _Corey_ | (same for SIG{HUP}, SIG{PIPE}) it continues to process the call OK on hangup, but doesn't actually release the call on occasion |
| 22:14.39 | _Corey_ | So, anyhow, with AGI() and a $SIG{INT} = \&onHangup; in Perl on Asterisk 1.6+ = occasional stuck AGIs |
| 22:15.07 | _Corey_ | <=1.4 with a DeadAGI() and $AGI->setcallback(\&onHangup); it worked flawlessly with never a stuck AGI |
| 22:16.57 | *** part/#asterisk jro (~jro@66.252.74.2) |
| 22:17.25 | _Corey_ | leifmadsen: I guess my question really boils down to whether you can tell me if I'm going about this right... ? |
| 22:21.02 | *** join/#asterisk talntid (talntid@c-76-104-157-191.hsd1.wa.comcast.net) |
| 22:22.15 | talntid | I'd like someone to install a system I bought from someone who isn't responding to support email, or write me a function for my asterisk PBX. I just need to change my caller ID to a different number based on area code. I have phone numbers in every area code and when I make calls to the area code, I want to use the local number. Anyone interested? I'd like it done ASAP. :) |
| 22:22.22 | talntid | Paying Job, of course. |
| 22:22.34 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 22:23.24 | Qwell | Set(CALLERID(num)=${DB(callerids,${EXTEN:0:3})}) |
| 22:23.31 | Qwell | That'll be $1k |
| 22:26.20 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 22:28.27 | Qwell | Guess not. |
| 22:29.09 | Qwell | leifmadsen: this consulting thing is hard! I give up. |
| 22:31.21 | talntid | hehe |
| 22:31.29 | Qwell | No, but seriously. That's trivial. |
| 22:31.35 | talntid | although I am sure that code would work, I don't know how to implement it. |
| 22:31.51 | Qwell | you just add the areacodes to the astdb as the key, and make the value the callerid. |
| 22:32.03 | talntid | Well, if you are willing to help me out, it'd be greatly appreicated. I'd be willing to send you some $$, or donate some to the charity of your choice. |
| 22:32.47 | Qwell | database put callerids 256 2565551212 |
| 22:32.51 | Qwell | pretty easy.. |
| 22:32.53 | talntid | i have never done anything in asterisk outside basic dialplan stuff. I am currently not in a position to dedicate a lot of time to learn it |
| 22:32.56 | talntid | oh, really. |
| 22:33.03 | talntid | that seems WAY easy. |
| 22:34.23 | Qwell | the dialplan line would need slight modifications (it's pretty close though, minus the DB syntax being wrong), then a bunch of database puts from the CLI.. |
| 22:36.12 | talntid | I could definately do the databae puts. I just did a few and pasted that into my dial-longdistance where it sets callerid |
| 22:36.19 | talntid | i'll google the db syntax |
| 22:36.28 | Qwell | it's just family/key |
| 22:36.30 | Qwell | I had family,key |
| 22:37.46 | leifmadsen | Qwell: too hard :) |
| 22:39.26 | Qwell | talntid: the ${EXTEN:0:3} also assumed 10-digit dialing |
| 22:40.10 | pabelanger | What echo canceller settings does a VPMOCT064 use? mg2? |
| 22:40.22 | Qwell | VPMOCT064 |
| 22:40.36 | talntid | hmm, testing it, it set to nothing.. |
| 22:40.45 | talntid | can I echo it for debugging ? |
| 22:40.55 | pabelanger | okay, but how is that defined in /etc/dahdi/system.conf? |
| 22:41.06 | pabelanger | help a friend confirm he is using it |
| 22:41.18 | Qwell | pabelanger: if it's present, it's used, I believe |
| 22:41.29 | Qwell | talntid: Verbose(${DB(callerids/${EXTEN:0:3})}) |
| 22:41.43 | pabelanger | Qwell: Okay, so nothing needs to be defined in system.conf? |
| 22:41.58 | Qwell | maybe echocancel=yes. I don't know |
| 22:42.17 | pabelanger | Okay, documentation about it is limited |
| 22:42.21 | pabelanger | off to google again |
| 22:43.51 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 22:43.51 | talntid | so, i do 11 digit dialing |
| 22:43.56 | talntid | 1+10dig |
| 22:44.08 | talntid | i could just change the database to 1509 |
| 22:44.19 | lauris | what kind of encryption or hashing is used to hide password in REGISTER data packet ? |
| 22:44.49 | talntid | or is the cleaner way to just change the :3 part? to pull the first 4 numbers, then the last 3? |
| 22:45.20 | Qwell | talntid: :1:3 |
| 22:47.33 | talntid | oh ;) |
| 22:47.50 | Qwell | :position:length |
| 22:47.56 | Qwell | position starts at 0 |
| 22:48.48 | *** join/#asterisk linuxviewer (test@ip70-162-145-222.ph.ph.cox.net) |
| 22:50.44 | *** part/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 22:51.04 | talntid | Where should that verbose show up? I am looking at my asterisk, with verbose = 99, and I am seeing the "CALLERID(NUM)=" |
| 22:51.42 | talntid | there is a decent amount of traffic going through the pbx. difficult to isolate the certain instance quickly, but thats what it shows currently |
| 22:52.04 | talntid | does verbose() show, despite verbosity level? |
| 22:54.41 | talntid | ah, the verbose command allows the level to be set, and when it was trying to echo the number, it technically wanted a very high verbosity level, if that makes sense. I set it to 0. :) |
| 22:55.38 | talntid | it doesn't appear to be echoing anything. |
| 22:58.44 | talntid | got it. verbose() rocks. |
| 23:00.40 | talntid | haha. that's WAY easier than my first attempt like 5 months ago. |
| 23:00.49 | talntid | Want me to donate or pay ya Qwell? |
| 23:02.03 | *** join/#asterisk Charrit (~Zairus@226.109.165.83.dynamic.mundo-r.com) |
| 23:02.12 | Charrit | hi |
| 23:21.10 | *** join/#asterisk dailylinux (~test@188.148.187.208) |
| 23:31.32 | *** join/#asterisk hipitihop (~denis@202.153.71.113) |
| 23:36.59 | Qwell | talntid: Nope, I can't take money for Asterisk stuff. |
| 23:38.50 | *** join/#asterisk leroybuckingham (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 23:39.23 | saxa | hi, anybody know what should i look at if I get my phone registered as a sip peer, but when dial 233 for music on hold, i see the asterisk console showing the play of the music, but i do not hear anything. Is this maybe codec related ? |
| 23:41.20 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:46.00 | *** join/#asterisk zerohalo (~zerohalo@cambridge.zerohalo.com) |
| 23:47.50 | |TEX| | sounds like codec |
| 23:48.12 | |TEX| | did you makes the clip yourself |
| 23:48.18 | |TEX| | or is it part of the install |
| 23:50.52 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:51.05 | *** join/#asterisk LemensTS (~matthew@adsl-70-130-171-141.dsl.stlsmo.swbell.net) |
| 23:51.20 | saxa | |TEX|: you mean the clip of the cable ? The plug ? |
| 23:51.37 | LemensTS | <PROTECTED> |
| 23:51.46 | p3nguin | saxa: "core set verbose 4" and then make the call that you're having trouble with. After you get the verbose output, pastebin it for me to see. |
| 23:51.58 | saxa | i see on the console actions going on, like moh playing |
| 23:52.26 | saxa | p3nguin: ok, let me try |
| 23:52.44 | LemensTS | oh nm it is a speakerphone issue |
| 23:54.27 | saxa | http://pastebin.com/jDxK0qxz |
| 23:54.48 | saxa | oh i set debug instead of verbose |
| 23:56.57 | saxa | http://pastebin.com/LtDvE6W0 |
| 23:57.01 | saxa | here we go |
| 23:57.10 | saxa | but I had verbose 25 |
| 23:57.25 | saxa | so i would see if something was really wrong |
| 23:57.39 | saxa | i have tried with ulaw and gsm |
| 23:58.08 | p3nguin | Verbosity does not increase above 4. 4 will show you the same as 5, and 5 will show you the same as 10000. |
| 23:59.47 | |TEX| | saxa i mean the audio clip |