00:01.00 | *** join/#asterisk zaltar (~zaltar@c-67-163-24-181.hsd1.il.comcast.net) |
00:12.54 | a1fa | grep -c "Wrong password" /var/log/asterisk/messages |
00:12.54 | a1fa | 365445 |
00:12.59 | a1fa | :) |
00:19.03 | ChannelZ | I have no idea what you're trying to do |
00:21.57 | a1fa | alwaysauthreject=yes |
00:22.16 | a1fa | scumbags can still guess extensions |
00:22.26 | a1fa | or sip accounts |
00:39.39 | niekie | The internet is great. |
00:39.58 | niekie | I Google "Free Russian DID" and I get free Russian wife. |
00:40.01 | niekie | :| |
00:41.27 | a1fa | maybe you did his wife? |
00:41.53 | niekie | Heh. |
00:50.23 | a1fa | i am finally getting some time to fix my dialplan |
00:50.36 | a1fa | and i finally cracked the broadvoice alternative number problem |
00:51.07 | a1fa | http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice+distinctive+ring+support |
00:53.42 | a1fa | is there anyway to change the server header information? |
00:54.01 | a1fa | and hide realm for 401 requests |
00:55.10 | a1fa | useragent = "" in sip.conf? |
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00:58.01 | a1fa | god.. asterisk should come default with all security features enabled |
00:58.40 | a1fa | <PROTECTED> |
00:58.49 | IsUp | blocksocialengineeringattacks=yes |
00:59.43 | a1fa | haha |
00:59.47 | a1fa | no.. not that one |
00:59.56 | a1fa | that one is enabled by layer8 routing |
01:13.05 | a1fa | god, this is the first time i think timing is messing me up.. can barely understand voicemialmain from all jittering |
01:21.22 | *** join/#asterisk tekati (~robert@cpe-75-85-246-201.bak.res.rr.com) |
01:25.09 | tekati | Looking into Faxing. What would be the best way to implement faxing with Asterisk 1.8.2.2? I would like to do faxing to email and be able to have clients who call into asterisk be able to auto send them a fax etc. |
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02:24.21 | *** join/#asterisk capitan (~captain@76.91.206.32) |
02:31.48 | capitan | is there a big overhead to being registered to too many trunks? for example, is there a big difference between A) being registered to one trunk with 99 concurrent calls or B) registering to ~33 trunks and distributing the same concurrent calls to ~3/trunk ? please forgive any poor semantics... |
02:37.49 | drmessano | capitan, if you're using IAX2 there's some potential bandwidth savings with call concurrency using the same peer. SIP is entirely different, and it doesn't matter if it's the same peer or a 99 different peers. |
02:49.03 | capitan | drmessano, thanks! :D |
02:50.20 | capitan | my case is SIP, and it'll be more managable for me with the lotsa-peers approach, so that's very good to know... |
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04:16.34 | zachsis | holy moly. any asterisk ninjas familliar with this error ? |
04:16.36 | zachsis | TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 |
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04:43.53 | zachsis | anyone active? |
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05:01.05 | carrar | moof |
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09:06.35 | Babak__ | hi is anyone using 79xx cisco ipphones? |
09:12.42 | fenrus | ayea. |
09:12.49 | fenrus | several 7940 |
09:13.41 | fenrus | if you state your real question you might get an answer. |
09:17.18 | Babak__ | fenrus: I need to show unicode characters on the display but it shows squares. it seems I need the locale files. but I do not find them |
09:17.53 | fenrus | the locale file for the phone ? |
09:18.00 | Babak__ | yep |
09:18.17 | fenrus | ill check the cisco website and see if i can find anything |
09:18.24 | Babak__ | thanx |
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09:25.54 | fenrus | cant seem to find that much, |
09:26.00 | fenrus | might be worth a shot with https://supportforums.cisco.com/ |
09:28.05 | fenrus | seems like it should work with CCM 5.0 on 7940G/7960G |
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09:39.48 | Babak__ | fenrus: I have no ccm. |
09:42.31 | fenrus | me neither :/ |
09:43.52 | Babak__ | I can find xml config files and all other things but this |
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09:58.17 | fenrus | have any filename to search for ? |
10:00.55 | ChannelZ | autoexec.bat |
10:08.11 | Babak__ | fenrus: I need to someone to have ccm. cause it seems cisco locale installer works on ccm servers only |
10:10.39 | Babak__ | for example the phone is requesting english_united_states\ipc-sip.jar for cisco ip communicator |
10:10.45 | Babak__ | on tftp |
10:16.52 | *** join/#asterisk bbear (~bbear@217.0.98-84.rev.gaoland.net) |
10:16.57 | bbear | hello. |
10:21.41 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
10:24.58 | bbear | I would like to have telephony through internet. |
10:25.22 | bbear | Is it possible to have a PCI card which I can connect with my standard RJ11 phone ? |
10:25.30 | fenrus | yes. |
10:25.33 | fenrus | or a ata box. |
10:25.37 | fenrus | or skype |
10:25.37 | bbear | (a cheap one preferably, not a fancy device) |
10:25.44 | bbear | what ata box ? |
10:26.16 | fenrus | http://lmgtfy.com/?q=ata+box |
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10:27.01 | bbear | what would you advice ? |
10:27.09 | bbear | I would like to have standard phone |
10:27.28 | bbear | Actually i wonder if you know SFR (french ISP). |
10:28.05 | bbear | I would like to use their phone service with a personal equipment (not this one provided as standard for the majority of people) |
10:28.22 | bbear | I am going to have a server. |
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10:28.30 | bbear | What is the cheapest, ata box or pci card ? |
10:28.34 | bbear | Is it so different ? |
10:29.23 | fenrus | if you have an ata box you need no server |
10:29.41 | bbear | Yes but I have a server. |
10:29.47 | bbear | with free pci slots. |
10:29.51 | fenrus | if you get a sip provider |
10:30.09 | fenrus | my guess is that a pci card is more expensive than an ata box |
10:30.29 | bbear | I already have one, but it does not support incoming calls. I wonder if either pci card or atabox will work with incoming calls. |
10:30.42 | bbear | (talking about the SIP account) |
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10:31.58 | fenrus | your sip provider does not support incoming calls? |
10:32.09 | fenrus | or your ata box ?, what ? |
10:33.20 | bbear | Actually if I directly a sip software phone (ekiga or twinkle) , I cannot have incoming calls. I can just make outgoing calls, and it's restricted to certain numbers, that becomes true when I use the ADSL box provided by the ISP. |
10:33.26 | bbear | becomes not true*. |
10:33.52 | bbear | with the box I have incoming calls and not restricted calls to special numbers. |
10:34.40 | bbear | I wonder if I can do the same with a pci card/ata box, or if the result will be the same than with the SIP softphone. |
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10:55.47 | *** join/#asterisk com3 (~com3@78.154.216.68) |
10:55.51 | com3 | Hello,, |
10:56.36 | bbear | hello |
10:56.50 | com3 | Just a quick question, are there any special hardware requirements to build a new Asterisk server? |
10:59.10 | com3 | anyone? |
11:01.50 | fenrus | that depends. |
11:01.59 | fenrus | only for sip signalling - no |
11:02.20 | fenrus | if you want to be able to use some kind of T1/E1 cards you need one of those |
11:02.26 | fenrus | and make sure it fits in the machine |
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11:06.28 | com3 | great thanks,, |
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11:15.31 | com3 | fenrus, so to be able to use a landline and voip together i need a T1/E1/J1 card? |
11:16.16 | fenrus | for a traditional landline yes, |
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11:16.29 | fenrus | but if you get a sip connection from a sip provider instead you need none |
11:16.36 | fenrus | if your phones are sip too. |
11:21.34 | com3 | I see, |
11:22.51 | com3 | do you happen to know if there are any USB based E1 or T1 cards? i'm trying to run the server on my laptop to start |
11:23.37 | coppice | see xorcom |
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17:35.47 | y7n | Hi, I have an Atcom IP01 running Asterisk 1.4. When I call out from a SIP client over PSTN I get some odd blips during the call. Anyone able to advise? I've recorded a sample: www.simplygenius.co.uk/file/misc/Atcom-IP01-PSTN-Sample-Recording.mp3 |
17:44.34 | *** join/#asterisk scrooge-mcduck (~scrooge@80-219-24-66.dclient.hispeed.ch) |
17:44.37 | scrooge-mcduck | hello |
17:44.45 | leifmadsen | yo |
17:45.07 | scrooge-mcduck | i am trying to get asterisk working |
17:45.15 | scrooge-mcduck | only it fails to connect with voipbuster |
17:45.31 | scrooge-mcduck | and i am getting a bit confused regarding the settings to use for voipbuster |
17:45.34 | scrooge-mcduck | after googling |
17:48.48 | scrooge-mcduck | does anyone know what the correct settings are? |
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18:03.27 | scrooge-mcduck | binbash_: are you good with asterisk? |
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18:26.50 | scrooge-mcduck | Call from '' to extension 's' rejected because extension not found. |
18:26.54 | scrooge-mcduck | what does this mean? |
18:28.39 | Tim_Toady | you dont have the s extension in that context |
18:29.01 | scrooge-mcduck | is that a special one from asterisk? |
18:29.22 | Tim_Toady | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
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18:33.03 | scrooge-mcduck | how can i enable the debug logging so it says in which context its looking for s? |
18:33.06 | scrooge-mcduck | i tried |
18:33.10 | scrooge-mcduck | asterisk -rvvvvv |
18:33.18 | scrooge-mcduck | and then set sip debug |
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18:34.32 | Tim_Toady | asterisk -rvvv is enough to see the dialplan code runing |
18:35.03 | Tim_Toady | as for the context its the one you have set on ur sip.conf for that peer/user/whatever |
18:36.33 | scrooge-mcduck | yeah but still ignores it |
18:36.41 | scrooge-mcduck | so i wanna know in which context it is looking for s... |
18:36.47 | scrooge-mcduck | should be from-voipbuster |
18:39.18 | scrooge-mcduck | grr |
18:41.07 | scrooge-mcduck | [Jan 23 18:40:31] WARNING[2253]: channel.c:2599 ast_indicate_data: Unable to handle indication 3 for 'SIP/5060-005568d8' |
18:41.11 | scrooge-mcduck | ah |
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18:48.54 | pabelanger | ~collectdebug |
18:48.55 | infobot | i heard 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 |
18:48.58 | pabelanger | scrooge-mcduck: ^ |
18:49.08 | scrooge-mcduck | thanks |
19:03.24 | scrooge-mcduck | i increase the logging |
19:03.25 | scrooge-mcduck | stuff |
19:03.33 | scrooge-mcduck | but still doesnt tell me which context it |
19:03.37 | scrooge-mcduck | expects it to be in |
19:05.38 | scrooge-mcduck | hmm |
19:05.43 | scrooge-mcduck | i need to do |
19:05.46 | scrooge-mcduck | [default] |
19:05.51 | scrooge-mcduck | include => from-voipbuster |
19:05.54 | scrooge-mcduck | to make it work :/ |
19:06.21 | scrooge-mcduck | while the context of the voipbuster peer of friend type is set to from-voipbuster |
19:08.03 | scrooge-mcduck | this is my config: http://pastebin.com/ihaRvGQR |
19:08.06 | scrooge-mcduck | pabelanger: http://pastebin.com/ihaRvGQR |
19:08.34 | scrooge-mcduck | from line 28 is extensions.conf |
19:19.32 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
19:26.54 | scrooge-mcduck | sip_write: Asked to transmit frame type 8, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
19:26.57 | scrooge-mcduck | :/ |
19:33.49 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
19:35.03 | ixti | Hello everybody. Probably somebody experienced (and solved) same problem as I have: ReceiveFax() works, while SendFax() not - produces lots of warnings about T38FaxMaxDatagram |
19:41.58 | pabelanger | scrooge-mcduck: remove the whitespaces in your extensions. EG: exten => s,1,Anser() |
19:42.00 | pabelanger | erro |
19:42.20 | pabelanger | s/Anser/Answer |
19:42.23 | pabelanger | s/Anser/Answer/ |
19:42.32 | pabelanger | blah |
19:42.45 | pabelanger | exten => s,1,Answer() |
19:42.49 | pabelanger | scrooge-mcduck: ^ |
19:43.48 | scrooge-mcduck | hm, does the whitespace matter? |
19:43.54 | pabelanger | scrooge-mcduck: yes |
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19:45.38 | scrooge-mcduck | :/ |
19:45.50 | pabelanger | ~book |
19:45.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
19:46.06 | pabelanger | scrooge-mcduck: ^ A good resource to learn |
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19:47.52 | scrooge-mcduck | good |
19:47.56 | scrooge-mcduck | working now sweet! |
19:47.59 | scrooge-mcduck | only sound sucks |
19:48.04 | scrooge-mcduck | but thats something for next time |
19:49.34 | scrooge-mcduck | already happy its working :) |
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19:56.01 | scrooge-mcduck | ah its usng ulaw |
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21:14.44 | *** part/#asterisk SparFux (~raoul@g224209097.adsl.alicedsl.de) |
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21:55.51 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
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23:00.27 | ixti | Can somebody tell me where to look at, when SendFax() produce lots of warnings about T38FaxMaxDatagram into console? |
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23:42.33 | *** mode/#asterisk [+o russellb] by ChanServ |
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