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| 00:31.34 | [JayMan] | Could someone tell me where I can download |
| 00:31.34 | [JayMan] | AsteriskNOW 1.7.0 32-bit |
| 00:31.35 | [JayMan] | ISO Image |
| 00:31.39 | [JayMan] | the link on the website does not work |
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| 01:24.57 | [T]ank | Im looking for a little help with call files... I am successfully placing calls using call files. What I am running into however is retries when i get a voicemail. I am using AMD to detect live body or voicemail... but the issue I am running into is when leaving a message it gets cut off because it does not wait for the vm greeting to end. Or if i hangup on the voicemail, the call is completed successfully meaning that it will not ret |
| 01:37.14 | [T]ank | wow its dead in here tonight. |
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| 01:38.37 | Whtsup | <PROTECTED> |
| 01:38.40 | Whtsup | wht does it mean |
| 01:39.32 | WIMPy | Maybe you ran out of licences? |
| 01:39.49 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 01:40.10 | [T]ank | Whtsup: do you have the g729 codec installed and registered? |
| 01:41.26 | Whtsup | i have installed but not registered |
| 01:42.38 | drmessano | Then it no work |
| 01:43.21 | [T]ank | you need to download the registration utility, use it, register the codec, then restart asterisk. Then all should be well assuming you have the config files set up correctly. |
| 01:47.29 | sawgood | If, when using X-Lite as a softphone remote extension of an Asterisk server ... there is always a 1 to 2 second delay for the called party to 'hear' the Xlite person's voice ... is this most always a network issue related to the ISP? |
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| 01:58.09 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 02:01.35 | [T]ank | when trying to use waitforsilence after AMD detects a machine, it never really detects any silence. i have it waiting for 2000 ms, ends up just timing out after a while and no message is left. is waitforsilence the be application for what I am trying to do? Or is there something more appropriate? |
| 02:03.01 | *** join/#asterisk coppice (~chatzilla@m121-202-83-52.smartone-vodafone.com) |
| 02:05.30 | [T]ank | here is the extensions.conf and resulting output once machine is detected: http://pastebin.ca/1904113 |
| 02:05.50 | [T]ank | any ideas on why it never detects silence? |
| 02:06.09 | *** join/#asterisk iamy_china (~iamy_chin@202.46.231.121) |
| 02:07.39 | coppice | because it has kids? |
| 02:08.41 | [T]ank | wish that were true... easy fix... but this one has been stressing me for weeks. I cannot figure it out. |
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| 02:15.11 | [T]ank | looks like there may be something to do with the amazon ec2 server this server is running on... but i am not finding anything more than the fact that this could be the cause..... |
| 02:15.27 | [T]ank | are there other things that do not work on the Amazon EC2 servers? |
| 02:15.50 | [T]ank | They have been so good and inexpensive to use.... this wrecks everything if its true |
| 02:18.55 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 02:23.08 | [T]ank | is there an alternative for waitforsilence? |
| 02:30.21 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 02:30.49 | boodu | hello |
| 02:31.33 | ChannelZ | oh hai |
| 02:35.16 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 02:49.55 | [T]ank | trying to understand how to add a line in the makefile in asterisk/apps. the instructions are: Then modify the Makefile in the /usr/src/asterisk/apps director to include app_machinedetect.so 'APPS=' |
| 02:50.09 | [T]ank | im not understanding where that should go or what should be typed. |
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| 02:51.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:57.37 | [T]ank | anyone able to help me add an app to the makefile? |
| 02:57.46 | [T]ank | im just not understanding how this is supposed to go in |
| 02:59.10 | WIMPy | Sounds like there should be a line starting APPS= where you should append it to. |
| 02:59.38 | [T]ank | let me pastebin it... hang tight |
| 03:00.54 | [T]ank | http://pastebin.ca/1904150 |
| 03:01.57 | [T]ank | it says just add it to the makefile in the apps directory |
| 03:02.34 | WIMPy | Doesn't look compatible. |
| 03:02.49 | WIMPy | Was that for an older version, perhaps? |
| 03:03.06 | [T]ank | i have no clue ;-) |
| 03:03.42 | [T]ank | my delema is that I cannot get waitforsilence to work and it is seemingly (according to google) because my server is running on the amazon ec2 service. So, i was looking for an alternative. |
| 03:04.15 | WIMPy | Interesting |
| 03:04.34 | [T]ank | im in need of a solution that detects human or voicemail, (AMD) and then when a machine, it waits long enough for the greeting to end then leave the prerecorded message (using call files) |
| 03:05.12 | [T]ank | waitforsilence waits for ever... irregardless of detect time... timeout works fine, but it never does detect silence. |
| 03:06.20 | [T]ank | any ideas? |
| 03:07.59 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cicfermnephgmvbd) |
| 03:08.03 | WIMPy | Yes, but it's unlikely to be of any help: I'm going to sleep. |
| 03:08.16 | [T]ank | haha... thats the idea? |
| 03:08.39 | WIMPy | That's my idea for now. |
| 03:08.45 | [T]ank | works for me :-D |
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| 04:12.53 | boodu | bye |
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| 05:40.47 | KingDavidNYC | Hello Everybody |
| 05:41.02 | Kyosh | king david |
| 05:41.06 | Kyosh | interesting name |
| 05:44.25 | Corydon76-dig | Whip your flurds, you'll figure it out |
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| 06:08.22 | sawgood | "where" in a nutshell would dial commands for an outside route fit 'best' in extensions.conf? |
| 06:08.35 | sawgood | Vs. where would dial rules for a trunk fit at best in extensions.conf |
| 06:11.09 | sawgood | exten => _X.,n,Dial(${OUT_1}/${EXTEN}) |
| 06:11.39 | sawgood | say this is in my extensions.conf as my outbound dial command ... where would dial rules go for the entire trunk? |
| 06:12.18 | sawgood | "where" in a nutshell would dial commands for an outside route fit 'best' in extensions.conf? |
| 06:17.30 | dexteruk | Basically you wouldnt want to put it under default |
| 06:17.56 | dexteruk | as this could create you a problem |
| 06:18.27 | dexteruk | so the best things to do is place your outbound route in the context of your users |
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| 06:23.29 | sawgood | dexteruk: thank you |
| 06:23.44 | sawgood | but wouldn't the users be in sip.conf? |
| 06:34.51 | sawgood | ok cool .. I have it set in my extensions.conf now to use NXX type dialing |
| 06:35.11 | sawgood | But, where do I go to tell the outside trunk dial commands to 'drop' the leading 9 |
| 06:35.29 | sawgood | I suppose I could do this in the dial rules, but I thought the purpose of dial commands was set to do these things |
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| 06:45.59 | sawgood | How can I 'dial' voicemail in a standard Asterisk 1.6.x build with the standard included voicemail.conf file? |
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| 07:33.01 | ChannelZ | you make an extension that calls VoiceMailMain |
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| 07:43.24 | sawgood | xten => 299,1,Dial(SIP/worker.two) |
| 07:43.25 | sawgood | exten => 299,2,VoiceMail(299@default) |
| 07:43.25 | sawgood | exten => 299,3,HangUp() |
| 07:43.49 | sawgood | How come ... with these lines ... voicemail does not 'answer' when I call x299 until after I hit END call? |
| 07:44.27 | sawgood | shouldn't Dial time out at some point and go to step #2? |
| 07:45.28 | tuxx- | use the timeout parameter? :P |
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| 07:45.37 | tuxx- | Dial(SIP/worker.two,10) |
| 07:45.40 | sawgood | oh! |
| 07:45.43 | sawgood | nice! |
| 07:45.48 | sawgood | thank you!!! |
| 07:46.04 | tuxx- | sawgood: for more options, go to asterisk -r and type: 'core show application dial' |
| 07:46.10 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 07:46.13 | ChannelZ | grrph |
| 07:46.17 | sawgood | What is the difference between (SIP/worker.two,10) and (SIP/299,10) |
| 07:46.18 | tuxx- | it will give you a list of all parameters and such |
| 07:46.26 | sawgood | tuxx: thanks for that tip ... I am doing it now |
| 07:46.29 | tuxx- | the sip account obviously :) |
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| 07:46.59 | sawgood | my point is ... how did Asterisk 'know' who worker.two was/is (the only reference to worker.two is in my voicemail.conf file |
| 07:47.16 | ChannelZ | it has to be in sip.conf |
| 07:47.25 | sawgood | its not in sip.conf |
| 07:47.36 | sawgood | I just made voicemail.conf ... tonight 2 minutes ago |
| 07:47.36 | ChannelZ | then it's not going to do jack but give an error |
| 07:47.40 | sawgood | oh! |
| 07:47.48 | sawgood | let me double check sip.conf then |
| 07:47.54 | ChannelZ | voicemail boxes can really only be numbers |
| 07:48.13 | ChannelZ | and they are totally unrelated to SIP peer names |
| 07:49.08 | ChannelZ | the dialplan you posted a minute ago for extension 299 (before my client pinged out) is also probably not what you intend to do |
| 07:52.46 | sawgood | even with (SIP/299,5) ... the call never goes to #2 ... it just keeps ringing the phone |
| 07:52.57 | sawgood | I have to hit decline on the softphone for the call to go to voicemail |
| 07:53.20 | ChannelZ | which is why I said that's probably not what you intend to do. |
| 07:53.30 | ChannelZ | Why do you want to call someone else? |
| 07:53.58 | sawgood | This is a practice scratch box, and I wanted to learn how to have a call ring an extension, and if no answer go to VM |
| 07:54.30 | sawgood | Here is my three lines of stuff from extensions.conf |
| 07:54.50 | ChannelZ | oh.. well then that is right. It should go to the next priority after ringing for 5 seconds, in your example. (Sorry I thought you were trying to dial into voicemail to retrieve messages) |
| 07:55.22 | sawgood | exten => 299,1,Dial(SIP/299,5) |
| 07:55.22 | sawgood | exten => 299,2,VoiceMail(299@default) |
| 07:55.22 | sawgood | exten => 299,3,HangUp() |
| 07:55.51 | sawgood | I have these entries to call into vm |
| 07:55.53 | sawgood | exten => 7299,1,Answer() |
| 07:55.53 | sawgood | exten => 7299,2,VoiceMailMain(299@default) |
| 07:56.43 | sawgood | If I call x299, it just keeps ringing and ringing (unless I hit decline the call) with the softphone |
| 07:57.06 | ChannelZ | then perhaps you failed to reload your dialplan and that's not really what is happening. |
| 07:57.26 | sawgood | oh more than just 'reload' at the CLI? |
| 07:57.31 | ChannelZ | either that or possibly there is something broken with short timeouts on Dial but otherwise what you have should be fine |
| 07:58.56 | sawgood | Jul 20 07:57:41] NOTICE[3572]: chan_sip.c:21470 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 299 |
| 07:59.07 | sawgood | I get this message when I do a dialplan reload |
| 08:00.07 | ChannelZ | put mailbox=299 for that peer in sip.conf |
| 08:00.22 | sawgood | mailbox=299@default |
| 08:00.27 | sawgood | or mailbox=299 |
| 08:01.29 | ChannelZ | the way you're setup either is probably fine but you can do 299@default to be implicit about it |
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| 08:05.25 | sawgood | for some set of reasons, (SIP/299,5) does not timeout |
| 08:05.43 | sawgood | its cool ... I've come a long ways for today, tomorrow is another set of 24 hours to play with Asterisk |
| 08:05.51 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:05.53 | ChannelZ | pastebin the console output of the attempt |
| 08:07.11 | sawgood | http://pastebin.ca/1904276 |
| 08:07.23 | sawgood | not really much to it ... it is stuck on ringing (ha) |
| 08:07.45 | ChannelZ | -- Executing [299@local:1] Dial("SIP/299-0000001f", "SIP/299") |
| 08:07.52 | ChannelZ | There is no ,5 argument |
| 08:07.59 | ChannelZ | thus no timeout. |
| 08:08.23 | sawgood | oh ... I have the setting in extensions.conf? |
| 08:08.37 | ChannelZ | so whatever extension 299 is in the local context is not doing Dial(SIP/299,5) as you say it is |
| 08:08.45 | ChannelZ | it's just doing Dial(SIP/299) |
| 08:08.50 | ChannelZ | It will ring for 139 years. |
| 08:08.54 | ChannelZ | by default |
| 08:09.02 | sawgood | hA! |
| 08:09.13 | ChannelZ | sorry. 136 years. |
| 08:09.37 | sawgood | exten => 299,3,HangUp() |
| 08:09.48 | sawgood | If #3 hangs up, what is () for? |
| 08:10.13 | ChannelZ | HangUp with no arguments |
| 08:10.20 | ChannelZ | same as saying just HangUp |
| 08:10.27 | sawgood | Lawyers wouldn't like that! |
| 08:10.39 | ChannelZ | Lawyers don't like anything. |
| 08:11.02 | sawgood | there's an agrument for that! |
| 08:11.31 | sawgood | Should I have something in sip.conf to make the (SIP/299,5) process work? |
| 08:11.43 | ChannelZ | no |
| 08:11.55 | ChannelZ | but you don't have SIP/299,5 in your extensions.conf |
| 08:12.02 | ChannelZ | dialing |
| 08:12.11 | sawgood | [local] |
| 08:12.11 | sawgood | ; Theoretically you shouldn't edit this |
| 08:12.11 | sawgood | ; local users - should be 4 character long |
| 08:12.11 | sawgood | include=voicemail |
| 08:12.11 | sawgood | include=internal |
| 08:12.12 | sawgood | exten => _XXX,1,Dial(SIP/${EXTEN} |
| 08:12.27 | sawgood | could this statement be the reason it does not make it to (SIP/299,5) |
| 08:12.57 | ChannelZ | without seeing the whole thing, who can say. There's a syntax error so your dialplan probably isn't reloading all the way. |
| 08:13.38 | sawgood | it worked when I commented out the above line I just pasted |
| 08:13.42 | sawgood | so, that is a breakthrough! |
| 08:13.49 | ChannelZ | you're missing the closing ) |
| 08:14.09 | sawgood | Thanks ... I fixed that just now and did a reload |
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| 09:47.50 | shamelessn00b | Hi I'm trying to do this: I have 2 IAX peers registered with my asterisk box, I have 2 dial statements in my dialplan calling each of the IAX peers lon different priorities, what I want is taht if peer 1 is down only then I should dial peer 2 and if peer 1 answers then the call should hangup after whatevers written in the peer 1 extension reaches its end |
| 09:48.16 | shamelessn00b | something like dial(IAX2/IAXtrunk1/111) |
| 09:48.31 | shamelessn00b | dial(IAX2/IAXtrink2/111) |
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| 10:00.34 | shamelessn00b | ok I figured it out need to use the DIALSTATUS variable to check status of peer before I dial the second trunk |
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| 10:43.22 | ranjan | hi all..is it possible to integrate Winmodems with asterisk to accept calls?? |
| 10:47.05 | tzafrir | ranjan, you'll have to write channel drivers for them |
| 10:47.40 | tzafrir | The "X100P" card with the wcfxo DAHDI driver is essentially a winmodem |
| 10:47.43 | ranjan | tzafrir, actually i have the drivers for the modem device |
| 10:48.56 | tzafrir | right. But Asterisk does not really use them as modems (device that encode data stream over analog audio stream), |
| 10:49.12 | tzafrir | It only needs to get / send the audio stream to that device |
| 10:49.33 | ranjan | tzafrir, so is there any solution for me to use that modem with asterisk?? |
| 10:49.41 | tzafrir | Solution: no |
| 10:49.52 | ranjan | tzafrir, :( |
| 10:50.00 | tzafrir | It would take writing (and debugging) code. |
| 10:50.13 | ranjan | tzafrir, i have a doubt regarding the same |
| 10:50.16 | tzafrir | There are existing adapters, of various types |
| 10:51.10 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 10:51.23 | tzafrir | If you're looking for a product that works now, look for either an ATA, or a telephony card |
| 10:51.23 | ranjan | tzafrir, my modem device is listed as /dev/ttySAGR , and when i used "mgetty /dev/ttySAGR" and when i called to that number the mgetty answered it after one ring |
| 10:51.26 | tzafrir | Another keyword: |
| 10:51.29 | tzafrir | ~fxsfxo |
| 10:51.30 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 10:52.46 | tzafrir | ranjan, a modem establishes a data connection. Do you want Asterisk to use that data connection? |
| 10:52.56 | ranjan | tzafrir, no |
| 10:52.58 | tzafrir | Or do you want Asterisk to relay it? |
| 10:53.19 | ranjan | tzafrir, that i dont know :( |
| 10:54.00 | tzafrir | Again: if you want Asterisk to accept voice calls using that modem: answer is: very unlikely |
| 10:54.54 | ranjan | tzafrir, ok i understand...thank you. but i will be able to send a receive fax using this modem ...isnt it?? |
| 10:55.17 | tzafrir | use hylafax |
| 10:55.46 | tzafrir | It's a fax server |
| 10:56.00 | ranjan | tzafrir, yes now i am implementing a fax server :) |
| 10:56.39 | ranjan | and at that time i just used the wvdial and in its configuration i gave my phone number and used wvdial |
| 10:56.51 | ranjan | and i was so happy that my mobile was ringing :) |
| 10:57.44 | *** join/#asterisk kurkowski (~felipe@189.62.199.201) |
| 10:57.44 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 10:58.23 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:58.41 | ranjan | tzafrir, have you ever implemented hylafax server?? |
| 10:58.58 | tzafrir | no |
| 10:59.42 | tzafrir | ranjan, though I suspect #hylafax (or whereever they reside) would be more useful... |
| 11:00.12 | ranjan | tzafrir, only i am there in that channel :) |
| 11:03.38 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 11:03.38 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 11:05.09 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 11:05.18 | kruemeltee | hello all together ... :-) |
| 11:05.40 | asteriskATmarmuD | <PROTECTED> |
| 11:05.50 | kruemeltee | *gg* yes |
| 11:06.05 | tzafrir | asteriskATmarmuD, what does it mean? |
| 11:07.42 | kruemeltee | this is tea ... the not the real one ... the one you can drink with cold water ... sort of little crumbs |
| 11:08.25 | kruemeltee | ha ... instant tea ... that's the word I've searched for |
| 11:09.27 | asteriskATmarmuD | <PROTECTED> |
| 11:09.39 | asteriskATmarmuD | ok, gotta get some code working |
| 11:10.16 | kruemeltee | I#ve got a simple question ... I want to forward a dialled number into a context that only has patterns. I don't remember the variable's name to set the number I want to dial |
| 11:10.20 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 11:10.41 | kruemeltee | asteriskATmarmuD, you studied in Cottbus too? You're welcome ... :-) |
| 11:12.01 | asteriskATmarmuD | <PROTECTED> |
| 11:12.13 | asteriskATmarmuD | <PROTECTED> |
| 11:13.30 | ruyo | Anyone knows where I can get more information on a segfault message? |
| 11:13.36 | ruyo | [51685.311696] asterisk[29697]: segfault at 38333236 ip 0808448e sp b5be0f80 error 4 in asterisk[8048000+103000] |
| 11:14.39 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:14.41 | kruemeltee | asteriskATmarmuD, from 2001 up to 2007 ... Informations und Medientechnik |
| 11:14.48 | kurkowski | Which extension should a call created with AMI's Originate start? The incoming extension of the calling channel or the exten you set when issuing the command? |
| 11:15.22 | shamelessn00b | anyone tried julius ASR for asterisk? |
| 11:15.35 | kruemeltee | asteriskATmarmuD, after that I changed to FH Lausitz (Informatik) and now I'm working in Dresden |
| 11:16.55 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:18.25 | asteriskATmarmuD | <PROTECTED> |
| 11:19.04 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 11:19.04 | [sr] | hi |
| 11:23.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:23.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:59.59 | kurkowski | is it possible to authenticate a peer to a sip provider without using the register => xxx:xx@sipprovider at the sip.conf? |
| 12:06.04 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 12:07.14 | spiceycurry | Having a strange problem here... numbers are showing up with shifted characters on the end in asterisk... i.e. 90529837022 is showing up as '9052983702@' with message like 'chan_iax2.c:7625 socket_process: Rejected connect attempt from 127.0.0.1, who was trying to reach '9052983702@'' |
| 12:07.20 | spiceycurry | Anyone see this before? |
| 12:07.28 | russellb | bizarre |
| 12:07.40 | spiceycurry | yea, its really wierd. |
| 12:08.35 | spiceycurry | scratch that |
| 12:08.42 | spiceycurry | everynumber has a trailing |
| 12:08.43 | spiceycurry | @ |
| 12:09.01 | spiceycurry | Everynumber has the last digit overwritten with an explicit '@' |
| 12:10.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:11.15 | Gugge | kurkowski: yes, but not to register with the provider |
| 12:11.34 | Gugge | kurkowski: you authenticate each time you call out through the provider |
| 12:12.02 | Gugge | kurkowski: but you either need to register with the provider, or tell them where to forward calls, to get incoming calls working |
| 12:12.10 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
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| 12:20.31 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:21.17 | kurkowski | Gugge, I wont get any incoming calls. I just want to originate calls from a SIP peer (currently using Skype For SIP). However, it treats all calls I originate as incoming calls first and then go the extension I choose. |
| 12:21.59 | kurkowski | When I originate from another SIP Channel, it will ring the channel first but wont treat is as a incoming call (I.E. wont go to the incoming context) |
| 12:22.25 | kurkowski | The same doesnt happen using SFS, as it will first go to the incoming context. |
| 12:23.01 | kurkowski | So I thought I can work around this if I dont register to the sip provider. |
| 12:23.11 | Gugge | kurkowski: just try without the register => stuff in sip.conf, and see if it works |
| 12:23.17 | [TK]D-Fender | kurkowski: This says either your auth is wrong or your peers are wrong. And yes, the call is incoming. all calls towards your * are "incoming" |
| 12:24.20 | [TK]D-Fender | kurkowski: If you want this this fixed you should be looking at the actual call debug |
| 12:24.27 | kurkowski | I'll check the documentation. |
| 12:24.33 | kurkowski | Yeah, I am. |
| 12:24.50 | kurkowski | I actually spend the last 24 hours looking into this. It has been fun. |
| 12:25.04 | kurkowski | if you remember, I did show the call to you yesterday. |
| 12:25.17 | kurkowski | I'll look around more and see if I can get this fixed |
| 12:25.30 | kurkowski | Thanks for you help guys |
| 12:26.13 | [TK]D-Fender | kurkowski: No, I wasn't here for that phase of things. |
| 12:26.25 | [TK]D-Fender | kurkowski: And this doesn't take reading. This takes looking. |
| 12:27.03 | [TK]D-Fender | kurkowski: Stare at your configs all you want, but they might be irrelevant when you look at the circumstances of the actual call they send you. |
| 12:29.51 | kurkowski | Yeah, I guess I already spent too much time starting at the configs. However, I did try to debug the call and I just cant understand why it will call the incoming context when I originate from SFS and wont when I originate from another channel. |
| 12:30.42 | [TK]D-Fender | kurkowski: perhaps you should show US. |
| 12:30.48 | [TK]D-Fender | kurkowski: PASTEBIN is your friend. |
| 12:30.50 | [TK]D-Fender | ~pb |
| 12:30.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:30.51 | WIMPy | What does CCSS stand for? |
| 12:31.33 | kurkowski | US you mean united states or another telephony related term? (I'm no native speaker) |
| 12:31.38 | kurkowski | :-) |
| 12:31.54 | kurkowski | I'll use pastebin |
| 12:32.55 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 12:32.59 | [TK]D-Fender | kurkowski: US as in NOT YOU |
| 12:33.05 | [TK]D-Fender | :p |
| 12:33.26 | kurkowski | oh, I read "you should know US", not "show US" hahaha |
| 12:33.49 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 12:34.26 | [Outcast] | can some one paste bin a working broadvoice config....the one thier web site is not working for me |
| 12:38.25 | [TK]D-Fender | [perhaps you should show us your FAILED CALL so we can see what's wrong |
| 12:38.44 | [Outcast] | ok |
| 12:38.46 | *** join/#asterisk lost_soul (~shawn@cpe-67-241-66-112.twcny.res.rr.com) |
| 12:38.53 | [Outcast] | I will paste bin the sip debug |
| 12:39.07 | *** join/#asterisk ms110 (~root@122.169.86.88) |
| 12:39.41 | [Outcast] | http://pastebin.ca/1904431 |
| 12:39.55 | [Outcast] | i know I have just missed something really stupid |
| 12:40.20 | ms110 | hellow everybody |
| 12:40.59 | *** join/#asterisk edguy3 (~edguy@80.169.133.251) |
| 12:41.00 | kurkowski | http://pastebin.com/1XWWjQUn |
| 12:41.38 | kurkowski | here it is, I pasted the call, extension.conf sip.conf and the java-asterisk code used to originate the call |
| 12:41.43 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-advwyxecnkqnroxn) |
| 12:42.02 | [TK]D-Fender | [Outcast]: Transmitting (NAT) to 147.135.20.221:5060: <- they are not behind NAT. should be nat=no |
| 12:42.19 | [sr] | people, the recording that cames from base with asterisk are only EN, it exists translations? or i have to make them myself? |
| 12:43.01 | [TK]D-Fender | kurkowski: ------------ WHY IT GOES TO 99051XXXX EXTENSION FIRST? <- because that's where CHANNEL sends it |
| 12:43.25 | [TK]D-Fender | kurkowski: I think you don't really understand what Originate is.... |
| 12:43.35 | kurkowski | it calls the channel |
| 12:43.43 | [TK]D-Fender | kurkowski: action.setChannel("SIP/99051000107744"); <- Calls this |
| 12:44.19 | [TK]D-Fender | kurkowski: When it answers it gets dumped into the dialplan --- > action.setContext("autodialer"); -- action.setExten("1"); -- action.setPriority(new Integer(1)); |
| 12:44.27 | kurkowski | ok, and will if I originate from, say SIP/Bob (softphone), it wont go to the incoming extension first? |
| 12:44.48 | [Outcast] | changed, still forbidden |
| 12:44.50 | kurkowski | and what* I meant |
| 12:45.07 | [TK]D-Fender | kurkowski: So it calls SIP/99051000107744 the dumps them in "autodialer,1,1" |
| 12:45.29 | [TK]D-Fender | kurkowski: No, when you call SiIP/Bob, it calls... THAT PHONE! |
| 12:45.46 | kurkowski | you're right. |
| 12:45.47 | [TK]D-Fender | kurkowski: Calling TO the deveice has nothing to do with the context in the peer. |
| 12:46.07 | kurkowski | but shouldnt the G option on the Dial works? |
| 12:46.11 | [TK]D-Fender | kurkowski: Calls FROM the device match the peer and get dumped into the context specified |
| 12:46.28 | [TK]D-Fender | kurkowski: What "G" options? the CHANNEL gets called with NO optiosn. |
| 12:46.42 | kurkowski | Dial("SIP/99051000107744-0000000a", "SIP/555192521975@99051000107744,30,G(agiscript^s^1)") |
| 12:46.54 | kurkowski | that G option I meant |
| 12:47.07 | [TK]D-Fender | kurkowski: action.setChannel("SIP/99051000107744"); <--- THIS gets called the same as if you did it as Dial(SIP/99051000107744) in the dialplan. No macro. the peer gets called. The end |
| 12:47.43 | kurkowski | Hmm, I guess I looked to the problem with the wrong perspective. |
| 12:47.56 | [TK]D-Fender | kurkowski: I also don;t see the other side ANSWER in your pastebin <- |
| 12:48.38 | kurkowski | yeah, but I did answered the call (it calls my mobile) |
| 12:48.55 | [TK]D-Fender | kurkowski: http://pastebin.com/1XWWjQUn <-- line 53 we see they get called. in the last 2 lines there I don't see it say they ANSWERED. If you don't answer, then what good is "G"? |
| 12:49.04 | [TK]D-Fender | kurkowski: * didn't seem to get it <- |
| 12:49.15 | [TK]D-Fender | kurkowski: I see no ANSER in there. |
| 12:49.39 | kurkowski | yeah, that's what I'm trying to understand. |
| 12:49.41 | [TK]D-Fender | kurkowski: Enable SIP DEBUG and look at the actual communication. |
| 12:49.52 | kurkowski | okay |
| 12:49.55 | kurkowski | will do that now |
| 12:50.04 | kurkowski | thanks for helping me out |
| 12:50.13 | [TK]D-Fender | [Outcast]: pastebin yoru configs masking only passwords |
| 12:50.19 | [TK]D-Fender | (sip.conf) |
| 12:50.29 | [Outcast] | sure thing |
| 12:51.57 | kurkowski | oh, another thing, there's any other way that I should be looking into to originate calls? All I want to do is call a number and run an AGI script to read DMTF. As you pointed out, my approach is not optimal. |
| 12:52.14 | ms110 | i am facing the destorted voice in MOH, can anybody tell me what would be cause for that? |
| 12:52.39 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:52.43 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:52.50 | [TK]D-Fender | kurkowski: Well you are calling 2 different things. |
| 12:52.57 | [Outcast] | [TK]D-Fender: http://pastebin.ca/1904440 |
| 12:53.55 | [TK]D-Fender | [Outcast]: "sip show registry" |
| 12:56.11 | kurkowski | yeah, but can I call just one using my SFS channel? Or, more precisely, can I originate the call without calling the SFS first? |
| 12:56.48 | [TK]D-Fender | kurkowski: What should your SFS have to do with this? |
| 12:57.05 | [Outcast] | [TK]D-Fender: I am registered |
| 12:57.05 | [TK]D-Fender | kurkowski: Again you are call 2 different places. What do you REALLY want to do? |
| 12:57.16 | [TK]D-Fender | [Outcast]: Ok, let me look. |
| 12:57.27 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 12:57.40 | [Outcast] | Host dnsmgr Username Refresh State Reg.Time |
| 12:57.41 | [Outcast] | sip.broadvoice.com:5060 N 4137711403@s 23 Registered Tue, 20 Jul 2010 05:07:45 |
| 12:57.41 | [Outcast] | 1 SIP registrations. |
| 12:58.01 | [TK]D-Fender | [Outcast]: No, I trusted that at face value... I meant look up comparative configs |
| 12:58.06 | [Outcast] | ok |
| 12:58.45 | [Outcast] | [TK]D-Fender: I am double checking my password in the peer settings and making sure that it is the same as the register line |
| 13:00.20 | [Outcast] | looks good |
| 13:00.40 | kurkowski | Well, what I really want to do is dial to a number and run an agi script. However, I want do it automatically, from a click-2-dial paradigm. My only way to reach the PSTN is using the SIP provider (SFS). (btw, you're really making me think haha) |
| 13:01.21 | [TK]D-Fender | kurkowski: Then you should never have made any reference to your SFS peer in there. |
| 13:01.53 | [TK]D-Fender | kurkowski: hold on.... |
| 13:01.56 | kurkowski | yeah, I get it now. I don't want to call the SFS as I'm doing right now. |
| 13:02.04 | [TK]D-Fender | kurkowski: Describe EXACTLY what you want this to do IN ORDER |
| 13:02.20 | kurkowski | OK, I will use pastebin. |
| 13:02.25 | kurkowski | One second.. |
| 13:03.18 | [Outcast] | found same issue here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg206863.html |
| 13:03.51 | [TK]D-Fender | [Outcast]: Junk post. Ignore it. What ver of * are you on? |
| 13:03.54 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:04.02 | [Outcast] | svn trunk |
| 13:04.25 | [Outcast] | should i try a release |
| 13:06.36 | [TK]D-Fender | [Outcast]: change "usereqphone=yes" to "user=phone" and retest |
| 13:07.30 | [TK]D-Fender | [Outcast]: http://broadvoice.com/support_install_asterisk.html |
| 13:07.50 | [TK]D-Fender | [Outcast]: I suggest you rename your peer as they did and adjust accordingly. |
| 13:08.06 | [TK]D-Fender | [Outcast]: Your settings are substantially different than their docs here |
| 13:08.18 | [TK]D-Fender | [Outcast]: take note of the hosts file as well |
| 13:08.20 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 13:08.33 | kurkowski | http://pastebin.com/Yfby9hYr |
| 13:08.50 | kurkowski | I couldnt branch it into more steps. As its just that simple. |
| 13:09.44 | *** join/#asterisk edguy3 (~edguy@80.169.133.251) |
| 13:11.07 | [Outcast] | [TK]D-Fender: I have already done that the user option does nothing is you want user=phone in the invite you need usereqphone |
| 13:11.13 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:11.53 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:12.17 | [TK]D-Fender | [Outcast]: change the rest including the peer name and dial string method to match their instructions. |
| 13:12.30 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:16.36 | kurkowski | Can I 'originate' a dial from the console channel? I mean, as if I was issuing a console dial command. |
| 13:17.08 | *** join/#asterisk m_c_le (~marcello@2001:470:1f0b:d4b:2e0:4dff:fe6c:9372) |
| 13:17.39 | [TK]D-Fender | kurkowski: 1. Call a PSTN number from SFS <- is sfs is calling it... then you are not doing ORIGINATE. |
| 13:17.54 | [TK]D-Fender | kurkowski: Originate Calls "A" first and then dumps them into the dialplan. |
| 13:18.10 | [TK]D-Fender | kurkowski: this is not SFS calling the PSTN number". |
| 13:18.29 | [TK]D-Fender | kurkowski: in Originate, ASTERISK calls the device listed in CHANNEL and then dumps them into the dialplan. |
| 13:18.42 | [TK]D-Fender | kurkowski: Is that clear and acceptable? |
| 13:19.05 | kurkowski | now it is. |
| 13:19.15 | [TK]D-Fender | kurkowski: When the call gets answered, run an AGI script <-- what does SFS have to do with this? |
| 13:19.20 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.89.20) |
| 13:19.32 | [TK]D-Fender | kurkowski: It sounds like you really only want ASTERISK to call out and not have SFS involved at all |
| 13:19.39 | kurkowski | well, nothing, I'm just describing what I'm trying to do. |
| 13:20.06 | kurkowski | to be honest, I guess this was my approach till now. |
| 13:20.12 | [TK]D-Fender | kurkowski: Your description has your PSTN number doing all the interaction. So what is the SFS channel you call being involved in this for? |
| 13:20.15 | FILLVAIO3 | Hi Guys! Does anybody know how to fix wrong time in my Master.cdr file in asterisk? |
| 13:20.38 | [TK]D-Fender | FILLVAIO3: How far off is it? |
| 13:21.58 | [TK]D-Fender | kurkowski: I also still don't see that call you place to that PSTN number being answered <- |
| 13:21.59 | kurkowski | From where I see it, SFS is involved becase it's the only way I can reach the pstn. |
| 13:22.06 | [TK]D-Fender | kurkowski: .... NO. |
| 13:22.19 | [TK]D-Fender | kurkowski: You just dragged *2* devices into this call. |
| 13:22.25 | FILLVAIO3 | [TK]D-Fender: in .cdr file time is 13:21 but in my timezone now is 17:22 |
| 13:22.25 | m_c_le | i have some questions using srtp and snom (asterisk trunk) |
| 13:22.26 | [TK]D-Fender | kurkowski: WHY *2*? |
| 13:22.35 | m_c_le | can sombody help? |
| 13:22.39 | [TK]D-Fender | FILLVAIO3: Becasuse CDR is in UTC, not local time <- |
| 13:22.53 | beek | ~ask |
| 13:22.54 | 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. |
| 13:23.00 | beek | m_c_le: ^^^^^^^^^^ |
| 13:23.23 | FILLVAIO3 | [TK]D-Fender: is it normal? |
| 13:23.50 | kurkowski | I actually dumped that code. I realize its wrong and I'm looking for a new approach. I DO NOT want to use 2 devices. |
| 13:23.53 | m_c_le | if i try to make a call form snom 360 |
| 13:24.08 | m_c_le | i get: WARNING[28177]: res_rtp_asterisk.c:1628 ast_rtcp_read: RTCP Read error: Success. Hanging up. |
| 13:26.18 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 13:27.16 | [TK]D-Fender | kurkowski: So yo just want to call "SIP/${numero}@99051000107744" and have them go to your AGI? |
| 13:27.23 | FILLVAIO3 | Thanx! |
| 13:27.26 | kurkowski | exactly! |
| 13:27.41 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 13:27.42 | [TK]D-Fender | kurkowski: Yeah, you made a complicated mess of this for nothing... |
| 13:27.46 | kurkowski | saying that way looks like an easy thing |
| 13:27.47 | [TK]D-Fender | kurkowski: 1 sec |
| 13:27.47 | kurkowski | ahhaha |
| 13:28.19 | ms110 | <PROTECTED> |
| 13:28.39 | Gugge | ms110. why do you have voice in your MoH ? |
| 13:29.40 | ms110 | yes, it plays some music, and at some interval voice & music simultaneously, but the distortion doesn't comes all the time, only sometime (can't predict) |
| 13:29.45 | beek | Gugge: so you can tell the caller how important their call is to you as demonstrated by the long hold times listening to that message. |
| 13:30.25 | Gugge | beek: so its really queue announcements + |
| 13:30.26 | Gugge | + |
| 13:30.26 | ms110 | Gugge: the timing device is ztdummy, |
| 13:30.44 | [TK]D-Fender | kurkowski: http://pastebin.com/1ApfLbyf |
| 13:30.49 | beek | Gugge: Yup, couched in soothing music. |
| 13:31.24 | Gugge | ms110: in some kind of virtual machine, or on real hardware? |
| 13:32.46 | ms110 | Gugge: they are dedicated dell servers |
| 13:32.48 | kurkowski | [TK]D-Fender: WOW, thanks a lot. I really mean it. Gonna test it out. |
| 13:34.19 | Gugge | ms110: where does the music come from, native files, or something else? |
| 13:35.27 | ms110 | at first they were mp3 files, but later i removed mp3 files, and placed the native files, but random occurance of distortion is still persist |
| 13:35.37 | kurkowski | [TK]D-Fender: HOLY SHIT I LOVE YOU |
| 13:35.38 | kurkowski | HAHAHAHAHA |
| 13:36.05 | beek | [TK]D-Fender frequently elicits that response. |
| 13:36.30 | kurkowski | man, thats so unfair. |
| 13:36.53 | kurkowski | I spent like 48 hours deeply looking into that and could see. |
| 13:37.09 | kurkowski | couldnt* |
| 13:37.45 | [TK]D-Fender | kurkowski: You kept dragging in aother device into the picture for no good reason. It had no place in what you wanted to do and also confused people trying to help you. |
| 13:38.02 | [TK]D-Fender | kurkowski: You wanted : * calls OUT. Dumpts them into AGI. |
| 13:38.21 | [TK]D-Fender | kurkowski: No "macro", no 2nd device. Just dial out, dump them into AGI |
| 13:40.12 | kurkowski | exactly. I thought this was a complicated thing. I never knew you could just set the channel as "SIP/NUMBER@99051000107744". I thought I had to start with an inbound channel or something like that. Plus my english doesnt help to clarify the explanation. |
| 13:40.55 | [TK]D-Fender | kurkowski: It is exactly as you put in a Dial() command you call in the dialplan |
| 13:41.14 | ms110 | Gugge: any clue? :) |
| 13:41.20 | kurkowski | many thanks, dude. many many thanks :D |
| 13:42.05 | *** join/#asterisk igorg (~igorg@net182.255.92-116.dynamic.omsk.ertelecom.ru) |
| 13:42.08 | [TK]D-Fender | ms110: if its random, its likely either poor network conditions (if VoIP), CPU tuilization (if virtualized), etc |
| 13:42.18 | [Outcast] | [TK]D-Fender: still no dice |
| 13:42.42 | [TK]D-Fender | kurkowski: Nothing says "I love you" quite like a Paypal donation to the "support [TK]D-Fender's internet connection" fund ;) |
| 13:42.52 | [TK]D-Fender | [Outcast]: Ok, show me the current incarnation. |
| 13:43.10 | [Outcast] | [TK]D-Fender: don't worry about it for now |
| 13:43.26 | [Outcast] | going to register a different device and then pack trace |
| 13:43.28 | [TK]D-Fender | [[ouIf you say so... |
| 13:43.42 | [Outcast] | then compare the two and find out what is different |
| 13:43.49 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:43.49 | [TK]D-Fender | [Outcast]: If you say so... |
| 13:43.56 | [TK]D-Fender | [Outcast]: Not a terrible idea |
| 13:44.09 | [Outcast] | exit |
| 13:44.11 | [Outcast] | oops |
| 13:44.16 | [Outcast] | wrong window |
| 13:44.17 | *** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net) |
| 13:44.49 | [TK]D-Fender | [Outcast]: Is this a metered account or something by any chance? Maybe a though if your account was out of funds |
| 13:45.06 | [Outcast] | no |
| 13:45.50 | [Outcast] | [TK]D-Fender: the stupid thing is that I use to work for them and wrote the original asterisk configuration stuff. |
| 13:46.05 | [Outcast] | I just have not had an asterisk box connected to them for six months |
| 13:47.08 | [TK]D-Fender | [Outcast]: Deactivated account maybe? |
| 13:47.37 | kurkowski | [TK]D-Fender: if this really exists, I'd happily donate |
| 13:47.37 | [TK]D-Fender | [Outcast]: Yeah, I'd say set up X-Lite direct, etc as your tes and wireshark it for sanity check |
| 13:47.43 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 13:47.46 | [TK]D-Fender | kurkowski: it does. |
| 13:47.51 | kruemeltee | says goodbye ... |
| 13:47.58 | [Outcast] | no I can make calls with other devices, I have notice there was patch that need to be applied make broadvoice work |
| 13:47.59 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-zucmhmsrbzzakvvn) |
| 13:48.17 | [TK]D-Fender | kurkowski: PM |
| 13:48.38 | [TK]D-Fender | [Outcast]: A Patch? That's crazy... no way should you ever have to go that far. |
| 13:49.17 | [Outcast] | yea, it was in the early days |
| 13:50.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:50.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.15 | [TK]D-Fender | [Outcast]: Not since 2004 at least... |
| 13:51.23 | [Outcast] | yea |
| 13:51.40 | [Outcast] | just following the trail |
| 13:52.49 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 13:53.20 | *** join/#asterisk Rajmohan (~raj@122.164.250.85) |
| 13:54.08 | Rajmohan | hi can we use asterisk as voip gateway to make calls to pstn |
| 13:56.15 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:58.05 | [TK]D-Fender | Rajmohan: * can process calls using anything you set up with it |
| 14:02.23 | Rajmohan | [TK]D-Fender: i have switch, need voip gateway, planning to use asterisk as gateway |
| 14:02.39 | [Outcast] | [TK]D-Fender: I can't see anything right off the bat....it would be nice to see someone's working config |
| 14:03.08 | [TK]D-Fender | Rajmohan: what kind of "switch" are you talking about? |
| 14:03.19 | [TK]D-Fender | Rajmohan: And as a gateway to what? |
| 14:03.33 | *** join/#asterisk boomerang (~boomerang@unaffiliated/boomerang) |
| 14:03.53 | Rajmohan | * as gateway to direct calls to pstn |
| 14:03.58 | Rajmohan | is that possible |
| 14:03.59 | Rajmohan | ? |
| 14:04.01 | [Outcast] | [TK]D-Fender: I think I found it |
| 14:04.22 | [Outcast] | [TK]D-Fender: INVITE sip:4132412554@147.135.32.221 SIP/2.0 = bad |
| 14:04.40 | [Outcast] | [TK]D-Fender: INVITE sip:4132412554@sip.broadvoice.com SIP/2.0 = good |
| 14:05.19 | [TK]D-Fender | [Outcast]: Did you also set that host entry, and change your peer to [sip.broadvoice.com] like it showed? |
| 14:05.36 | [TK]D-Fender | Rajmohan: * is jsut software. HOW should it get to the PSTN? |
| 14:05.49 | [Outcast] | [TK]D-Fender: so how do you make asterisk use the domain name in the invite instead of the ip? |
| 14:06.04 | *** join/#asterisk traxx (~traxx@91-64-218-219-dynip.superkabel.de) |
| 14:06.22 | Rajmohan | we need additional hardware for it right? digium hardwares? |
| 14:07.33 | [TK]D-Fender | Rajmohan: You would.. if you intend on access a physical line you alerady have <- |
| 14:07.55 | [TK]D-Fender | Rajmohan: You are still not being specific on HOW you plan on getting to the PSTN |
| 14:08.05 | [TK]D-Fender | Rajmohan: Or what you meant by "switch" before |
| 14:08.18 | [TK]D-Fender | [Outcast]: Didi you follow the guide completely? Including the hosts file? |
| 14:09.41 | [Outcast] | yes |
| 14:09.49 | [Outcast] | i wrote it |
| 14:10.33 | Rajmohan | i mean a sip server like openser and using asterisk as gateway |
| 14:10.50 | Rajmohan | asterisk with hardwares and dial plan |
| 14:11.10 | Rajmohan | will act as a gateway to pstn right? |
| 14:12.07 | Gugge | asterisk can send and recieve calls from/to a PSTN line with the right hardware yes. |
| 14:12.15 | Gugge | if that is what you are asking. |
| 14:12.33 | Rajmohan | yes exactly that is what iam looking for |
| 14:14.10 | [TK]D-Fender | Rajmohan: Then yes |
| 14:14.26 | [TK]D-Fender | Rajmohan: How large a deployment are you looking at? |
| 14:14.41 | Rajmohan | like around 200 concurrent calls |
| 14:14.59 | Rajmohan | * will support that and will be stable? |
| 14:15.55 | [TK]D-Fender | Rajmohan: it can. I'd recommend that you be doing G.711 from SER to * of course |
| 14:15.55 | [TK]D-Fender | Rajmohan: otherwise you'll be putting a fairly large transcoding load on it |
| 14:15.56 | Rajmohan | we cant go for g723? |
| 14:16.05 | [TK]D-Fender | Rajmohan: I might recommend using a dedicated SIP gateway at this point |
| 14:16.32 | Katty | OHAI |
| 14:16.38 | Katty | i has come to bug you all. |
| 14:16.40 | [TK]D-Fender | Rajmohan: * does not support G.723 except with the Digium TC400 transcoder card. That will be another large expense |
| 14:17.18 | Rajmohan | oh ok, does * support g729 codec? |
| 14:17.29 | [TK]D-Fender | Rajmohan: I now recommend you either change codecs or prepare for adding an expensive trancoder card |
| 14:17.40 | [TK]D-Fender | Rajmohan: Yes, but keep an eye on CPU load <_ |
| 14:17.46 | Rajmohan | ok i understand |
| 14:17.51 | [TK]D-Fender | raj200 simultaneous transocdes are going to be HEAVY |
| 14:18.20 | Rajmohan | how many concurrent calls it can support pls |
| 14:18.29 | [TK]D-Fender | Rajmohan: I'd recommend checking with Digium as to what reports they have on scalability. Also this would have to be a large dedicated server. Don't even dream ofvirtualising it |
| 14:18.33 | Rajmohan | iam planning to use * as dedicated Sip gateway |
| 14:18.40 | [TK]D-Fender | Rajmohan: TRANSCODING is the issue. |
| 14:19.15 | leifmadsen | I'm just trying to debug a wctdm2400p card in a machine to determine if a recent storm may have fried the card. I get the card loaded without errors, but when calling the number causes the lines to not be answered. With 'zap show status' I get IRQ 0 which seems odd... |
| 14:19.35 | leifmadsen | zttool shows everything as no alarms, and zttest shows accuracy OK, so the drivers are working at the least |
| 14:19.57 | leifmadsen | anything else I can try other than just swapping out the hardware? The oddest thing is that they seem to be able to make outbound calls... |
| 14:20.39 | tzafrir | leifmadsen, do you actually get interrupts from the card? |
| 14:20.50 | leifmadsen | tzafrir: how do I check that? |
| 14:20.50 | tzafrir | watch -n1 -d cat /proc/interrupts |
| 14:20.52 | *** join/#asterisk cuco (~elcuco@109.253.179.225) |
| 14:20.53 | leifmadsen | thanks, checking |
| 14:21.21 | tzafrir | oh, and contact Digium's support :-p |
| 14:21.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:21.53 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:23.22 | leifmadsen | tzafrir: on CPU0 I get a rather large number that seems to be changing.. |
| 14:23.30 | leifmadsen | but no outbound calls seem to be working now either |
| 14:24.21 | tzafrir | no outbound calls even after an inbound call? |
| 14:24.35 | leifmadsen | no calls working in either direction |
| 14:24.45 | leifmadsen | outbound calls seemed to be working sporadically yesterday |
| 14:25.15 | leifmadsen | I get somethign like this: 160: 2986581425 0 0 0 IO-APIC-fasteoi Wildcard TDM2400P |
| 14:25.22 | [TK]D-Fender | leifmadsen: plugged a phone in parallele to listen in on it? |
| 14:25.35 | [TK]D-Fender | leifmadsen: Are all ports ont he card bad? Tried swapping the modules around? |
| 14:25.44 | leifmadsen | no, I'm not on-site -- the AT&T phone guy says the line works up to the card |
| 14:26.13 | [Outcast] | does anyone know how to force asterisk use the domain name in invite instead of the ip address? |
| 14:26.16 | leifmadsen | Missouri is a bit too far for my arms |
| 14:26.25 | [TK]D-Fender | leifmadsen: Of COURSE they do... their jobs depend on "looking innocent" :p |
| 14:26.38 | [TK]D-Fender | leifmadsen: Sub-contract Katty then! |
| 14:26.58 | Gugge | [Outcast] shouldnt host=<host> in the peer section in sip.conf cover that? |
| 14:27.00 | leifmadsen | Katty: are you near Moss Point? :) |
| 14:27.17 | Katty | moss point? |
| 14:27.26 | [Outcast] | Gugge: it does not |
| 14:27.31 | Katty | leifmadsen: is that a....city? |
| 14:27.33 | leifmadsen | Gugge: kinda -- it'll resolve the IP address when the module loads |
| 14:27.54 | *** join/#asterisk dailylinux (~fedora@s21-00210.dsl.no.powertech.net) |
| 14:28.06 | Katty | leifmadsen: google claims Moss Point is in mississippi |
| 14:28.19 | leifmadsen | Katty: oops... MS != Missouri :) |
| 14:28.23 | Katty | leifmadsen: 561 Miles @ 9hrs 17min according to google |
| 14:28.27 | leifmadsen | heh |
| 14:28.31 | leifmadsen | closer than me :) |
| 14:28.53 | [TK]D-Fender | leifmadsen: Geography FAIL :) |
| 14:29.10 | [Outcast] | Gugge: currently sending INVITE sip:4132412554@147.135.32.221 SIP/2.0 |
| 14:29.32 | [Outcast] | Gugge: needs to be INVITE sip:4132412554@sip.broadvoice.com SIP/2.0 |
| 14:29.38 | tzafrir | leifmadsen, quality of the audio may also be significant. Any chance DTMFs get distorted? |
| 14:29.46 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:29.53 | [Outcast] | Gugge: i have host=sip.broadvoice.com |
| 14:29.53 | Gugge | [Outcast]: maybe Dial(SIP/username:password;number@host) would do the trick ... i cant remember if that is correct syntax though |
| 14:30.08 | leifmadsen | tzafrir: we don't even get that far -- the call doesn't get answered |
| 14:30.20 | [TK]D-Fender | [Outcast]: Did you change your Dialas per the guide as well? |
| 14:30.35 | [Outcast] | yes |
| 14:30.57 | leifmadsen | tzafrir: oh you mean no connected to the asterisk |
| 14:31.03 | Gugge | [Outcast]: are you sure the host is needed in the INVITE uri ... and not just in the from uri? |
| 14:31.37 | Gugge | [Outcast]: fromdomain=sip.xxx.com is needed on a couple of the itsp's i use. |
| 14:31.39 | [TK]D-Fender | [Outcast]: was the INVITE line the only difference in the header (except for the necessarily unique stuff)? |
| 14:32.04 | [Outcast] | [TK]D-Fender: if you name your peer [sip.broadvoice.com] it make asterisk core dump btw |
| 14:32.20 | [Outcast] | [TK]D-Fender: the invite is the only difference |
| 14:32.22 | [TK]D-Fender | [Outcast]: Shouldn't |
| 14:32.35 | [Outcast] | [TK]D-Fender: current SVN it does |
| 14:32.54 | [Outcast] | shouldn't matter |
| 14:36.51 | *** join/#asterisk WimD (~Wim_ctx@62-213-207-59.colo.kangaroot.net) |
| 14:37.59 | leifmadsen | [Outcast]: please file that as an issue with the backtrace |
| 14:38.01 | WimD | hey, does anyone know of a sip phone that also has a pstn connection? i know of the siemens gigaset phones, but i need a fixed phone |
| 14:38.03 | leifmadsen | then we can get that looked at |
| 14:38.15 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 14:39.48 | Gugge | WimD: i only know of the dualphone 3081, but that is wireless too |
| 14:43.26 | WimD | Gugge, k, looks like the only make wireless devices too |
| 14:43.30 | [Outcast] | leifmadsen: will do in a bit |
| 14:44.04 | [Outcast] | leifmadsen: do you know of a way to get asterisk to use the domain in the invite instead of the ip address? |
| 14:47.18 | leifmadsen | [Outcast]: sorry not sure -- I've never had to specifically make sure that was set or not set |
| 14:47.47 | [Outcast] | leifmadsen: ok |
| 14:48.09 | [Outcast] | it uses the hostname in the registion |
| 14:48.22 | [TK]D-Fender | WimD: SPA-3102 <---- |
| 14:51.17 | WimD | [TK]D-Fender, can you route incoming calls on the FXO directly to a sip device? |
| 14:51.40 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:52.33 | [TK]D-Fender | Winkie: Yes |
| 14:53.19 | Winkie | hi |
| 14:53.19 | Winkie | :D |
| 14:53.47 | WimD | Winkie, i think he was talking to me :-) |
| 14:55.26 | [TK]D-Fender | WimD: Correct |
| 14:55.46 | Winkie | :) |
| 14:57.16 | WimD | [TK]D-Fender, i have a grandstream HT503 here, it also has a fxo and fxs port on it, i tried having incmoning calls forwarded to a sip phone, but didnt get it to work. Do you know if it can be done with a HT503? |
| 14:57.19 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-zngccftdelqfsede) |
| 14:57.48 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:00.26 | [TK]D-Fender | Winkie: I don't recall if the 503 was a PSTN FAILOVER only or if it was really usable |
| 15:04.35 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:04.43 | *** join/#asterisk jShaf (~jShaf@c-69-250-230-10.hsd1.md.comcast.net) |
| 15:07.21 | Faustov | guys, about the call-limit option being replaced by the groupcount function in dialplans - is there any way to keep call-limit? It seems much more suited to what I need (maximum n simultaneous calls to provider X), than 3-4 extra lines to each dialplan outbound rule |
| 15:08.31 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:09.56 | *** part/#asterisk jShaf (~jShaf@c-69-250-230-10.hsd1.md.comcast.net) |
| 15:11.43 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 15:11.50 | Gugge | Faustov: dont upgrade |
| 15:12.15 | Faustov | but there are security and performance fixes... |
| 15:12.36 | Faustov | I guess this is a question mainly to the developers |
| 15:12.42 | Gugge | security issues are fixed for a long time in current versions |
| 15:12.42 | Faustov | pokes leifmadsen as he was active |
| 15:12.51 | leifmadsen | eh? |
| 15:12.58 | leifmadsen | doesn't mean I'm paying attention now |
| 15:13.04 | leifmadsen | :) |
| 15:13.32 | leifmadsen | Faustov: that is a question you should pose to the asterisk-dev mailing list |
| 15:13.38 | leifmadsen | asking here will get you nowhere |
| 15:13.44 | Faustov | right |
| 15:14.13 | Faustov | btw, does the changelog message about this mean 1.8 as "next release"? |
| 15:14.20 | leifmadsen | yes |
| 15:14.27 | leifmadsen | ~asterisk-versions |
| 15:14.28 | infobot | rumour has it, asterisk-versions is Information about the maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at http://www.asterisk.org/asterisk-versions |
| 15:14.53 | Faustov | thanks |
| 15:15.28 | Faustov | is there perhaps a document highlighting the main goals of 1.8 branch? |
| 15:16.58 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:17.09 | asteriskATmarmuD | hi all |
| 15:17.32 | Chainsaw | Faustov: "core restart gracefully" will no longer be interpreted as "core fall over and die sideways" |
| 15:17.49 | asteriskATmarmuD | has anyone fax detection for outgoing calls working and is there even a source code for that (i don't mean NVFaxDetect, this is for incoming calls) |
| 15:17.52 | Faustov | Chainsaw: that was jkroon's problem :> |
| 15:17.59 | Chainsaw | Faustov: And mine. |
| 15:18.36 | Faustov | Chainsaw: right, i always do the thing he posted in a loop while later |
| 15:18.42 | Faustov | much more efficient |
| 15:20.12 | Chainsaw | Faustov: Annoyingly there was a point at which it worked. So it was unfixed. |
| 15:20.26 | Faustov | argh |
| 15:22.21 | *** join/#asterisk zxvff (~shahid@216.245.214.147) |
| 15:22.34 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:23.14 | zxvff | hi, i know this isn't asterisk specific (i'm using pbiaf) but all of a sudden I come in this morning to work and all of the messages when people call into queues are just looping and it's impossible to get through to anyone |
| 15:23.23 | zxvff | really critical problem i need to fix asap and i was hoping someone could help |
| 15:24.21 | [TK]D-Fender | zxvff: It isn't an * problem. |
| 15:24.29 | Rajmohan | [TK]D-Fender: what is the hardware you recommend that using * as sip gateway |
| 15:24.40 | zxvff | well it's an * system |
| 15:24.58 | [TK]D-Fender | Rajmohan: "as a SIP gateway" is not something you can say "Oh you need exactly this for it" kind of case |
| 15:25.19 | [TK]D-Fender | Rajmohan: I don't know exacly what kind of system you'd need for 200 transcoded & terminated calls. |
| 15:25.34 | Rajmohan | ok please tell me for smaller ones |
| 15:25.44 | Rajmohan | like around 20 -30 calls |
| 15:25.53 | [TK]D-Fender | zxvff: Go look at the queue memeber status' |
| 15:25.55 | Rajmohan | so that i can splitup in to different servers |
| 15:26.27 | [TK]D-Fender | Rajmohan: I'd say 2 servers then. Decent dual-core modern system or better, 4 gig ram. |
| 15:26.41 | [TK]D-Fender | rajThat should do for straight termination. |
| 15:26.54 | Rajmohan | ok |
| 15:26.59 | [TK]D-Fender | Rajmohan: Then again, you might be better off elaving * out of this and jsut getting a direct SIP>PRI gateway |
| 15:27.51 | [TK]D-Fender | Rajmohan: Considering $2000 in G.729 licenses, 2 servers, and 2 x 4-port PRI ports with HWEC |
| 15:28.10 | Rajmohan | ok pl |
| 15:30.13 | *** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com) |
| 15:30.38 | asteriskATmarmuD | and again: is there fax detection for outgoing calls? |
| 15:31.00 | Chainsaw | Afternoon Gary. |
| 15:31.58 | p3nguin | Wouldn't you know if you're sending a fax? Why would it need to be detected? |
| 15:32.26 | garymc | afternoon all |
| 15:33.26 | asteriskATmarmuD | p3nguin: we don't want to get an outgoing call put through to an agent, if it is a fax machine + we want to mark the lead in our database as "fax" |
| 15:33.47 | asteriskATmarmuD | p3nguin: we don't want to fax documents or receive faxes |
| 15:33.51 | p3nguin | If it's an outgoing call, why would it get connected to one of your agents? |
| 15:34.10 | p3nguin | Do you have any concept of processing calls? |
| 15:34.19 | [TK]D-Fender | asteriskATmarmuD: What are you calling out using? |
| 15:34.56 | asteriskATmarmuD | p3nguin: yes, agents are waiting in conference rooms for calls... and get the call if someone picks up (this could also be a fx machine) |
| 15:35.07 | p3nguin | That's half of the problem. |
| 15:35.28 | asteriskATmarmuD | <PROTECTED> |
| 15:35.43 | [TK]D-Fender | asteriskATmarmuD: then you have no option for this |
| 15:36.07 | p3nguin | First, agents should not be in conference rooms, but should instead be in queues. Second, outgoing calls should not end up in the same "location" as incoming calls which are intended to reach agents. |
| 15:36.21 | asteriskATmarmuD | <PROTECTED> |
| 15:36.28 | asteriskATmarmuD | <PROTECTED> |
| 15:36.53 | [TK]D-Fender | asterBecause the only thing that can detect an OUTGOING fax is on a DAHDI channel |
| 15:36.54 | asteriskATmarmuD | <PROTECTED> |
| 15:37.42 | p3nguin | Is this an internal helpdesk type of phone system? |
| 15:37.57 | [TK]D-Fender | p3nguin: No. a CALL OUT system. |
| 15:38.03 | asteriskATmarmuD | <PROTECTED> |
| 15:38.25 | [TK]D-Fender | asteriskATmarmuD: There is no way for this information to be apssed back to *. |
| 15:38.28 | [TK]D-Fender | passed* |
| 15:38.47 | asteriskATmarmuD | <PROTECTED> |
| 15:39.07 | p3nguin | Sounds like a stupid design, so I'm not going to continue trying to "fix" it. |
| 15:39.30 | [TK]D-Fender | asteriskATmarmuD: Not unless the call is not passed immediately to * at which point a normal caller will get dead air for a while <- |
| 15:40.17 | p3nguin | I know how I would do it if it were mine: Put agents in queues. Don't put outgoing fax in queues. Done. |
| 15:41.43 | [TK]D-Fender | p3nguin: You misunderstand |
| 15:41.57 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 15:42.01 | [TK]D-Fender | p3nguin: His system dials OUT and he wants to flag tat the person he calls OUT to is not a fax machine <- |
| 15:43.20 | [TK]D-Fender | p3nguin: he isn't using Queues, he's using an auto-dialer and Mettme as a place to have his agent wait around for the call (originate, etc) until the call gets dumped in |
| 15:43.58 | Kobaz | fax detection? |
| 15:44.08 | p3nguin | Each agent uses his own MeetMe conference per call? |
| 15:44.33 | [TK]D-Fender | p3nguin: I'd bet on "Yes" |
| 15:45.07 | [TK]D-Fender | p3nguin: He isn't queued because these are outgoing calls. Outbound tele-annoyers, not "inbound call-center" |
| 15:46.14 | Kobaz | asteriskATmarmuD: you're not selling auto warrentys or debt reduction are you? |
| 15:46.33 | Chainsaw | Kobaz: Likely whatever the market will pay for. |
| 15:47.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:48.05 | xheliox | lol |
| 15:48.24 | Kobaz | press 1 to lower your credit card interest rate |
| 15:49.25 | Kobaz | 1... hi welcome to credit card services, do you want to lower your interest rate today? yeah, hi... what company are you with? credit card services sir..... and what's your website? click..... |
| 15:50.00 | asteriskATmarmuD | <PROTECTED> |
| 15:50.04 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:50.08 | Kobaz | asteriskATmarmuD: that's not you is it? |
| 15:50.08 | asteriskATmarmuD | we don't sell anything |
| 15:50.29 | asteriskATmarmuD | we are using an auto-dialer as [TK]D-Fender said |
| 15:50.40 | [TK]D-Fender | ... but do you have 20-30 minutes to fill out this quick survey? |
| 15:50.56 | asteriskATmarmuD | ;) |
| 15:50.59 | asteriskATmarmuD | <PROTECTED> |
| 15:51.17 | Kobaz | asteriskATmarmuD: if that's what your doing, i will hunt you down |
| 15:51.25 | asteriskATmarmuD | <PROTECTED> |
| 15:52.28 | asteriskATmarmuD | ok, so no fax detection for outgoing calls unless our SIP-ISDN-Interface connected to the telephone network gives us some info about it |
| 15:52.40 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:52.50 | asteriskATmarmuD | nice chat, as always :) |
| 15:55.23 | *** join/#asterisk Number5 (~nordin@dhcp-077-250-180-189.chello.nl) |
| 15:55.28 | Number5 | hello |
| 15:56.25 | Number5 | Just being curious, is it possible to have encrypted voip using Asterisk? |
| 15:57.09 | raden_work | YES w/ iax |
| 15:57.25 | Number5 | aaah thnx my friend |
| 15:57.55 | WIMPy | heard rumors that srtp is finally working in 1.8. |
| 15:58.27 | WIMPy | And if it's to a known destination, openvpn might be a good option. |
| 15:59.11 | Number5 | is srtp safe enough to use, I mean as secure as ssh? Does it use strong encryption like AES/Rijndael? |
| 16:01.30 | Naikrovek | ooh srtp |
| 16:02.00 | Naikrovek | Number5: it can probably use anything, but symmetric encryption like AES would not be my choice |
| 16:02.35 | Naikrovek | lol |
| 16:02.37 | Naikrovek | it does use AEX |
| 16:02.39 | Naikrovek | AES |
| 16:02.48 | Kobaz | Naikrovek: what would you prefer |
| 16:02.53 | Kobaz | rot13? |
| 16:02.55 | Naikrovek | private/public |
| 16:03.02 | Kobaz | mm |
| 16:03.07 | Kobaz | yeah that would be nice too |
| 16:03.14 | tzafrir | Number5, you're mixing terms here |
| 16:03.14 | *** join/#asterisk mrchrisadams (~Adium@87-194-125-43.bethere.co.uk) |
| 16:03.16 | Number5 | From security point of view AES is the strongest... |
| 16:03.26 | Naikrovek | Number5: not if your key gets out it isn't |
| 16:03.34 | tzafrir | AES is a building block used by ssh and used by asterisk |
| 16:03.35 | Naikrovek | public/private you can let your public key go anywhere it wants |
| 16:03.40 | Kobaz | Number5: you're mixing security with cryptography |
| 16:04.11 | Number5 | ok ok, I mean encryption not the secured handshaking stuff |
| 16:04.22 | Number5 | like ssl or tls |
| 16:04.31 | Kobaz | i could have the strongest crypo in the world, but if it's easy to get the key... the security of it is useless |
| 16:04.31 | tzafrir | Asterisk's encryption method is way simpler. Asterisk does not attempt to multiplex multiple channels accross a single connection like ssh |
| 16:05.02 | Number5 | Kobaz: yes I agree |
| 16:05.18 | tzafrir | Kobaz, hmm... how exactly do you get the key? |
| 16:05.30 | Number5 | Anyway, I think I have to do some more readings about security in Asterisk. |
| 16:05.47 | Number5 | buy! |
| 16:05.50 | *** part/#asterisk Number5 (~nordin@dhcp-077-250-180-189.chello.nl) |
| 16:05.52 | Kobaz | tzafrir: become a haxor |
| 16:05.56 | [TK]D-Fender | SELL |
| 16:06.45 | *** join/#asterisk BANSAL (~bansal@117.199.127.221) |
| 16:07.52 | tzafrir | The point of having a key is to have a small secret that is easy to replace, in case TK sold it to our rival Kobaz. |
| 16:08.26 | Kobaz | how does the size of the secret make it easier/harder to replace |
| 16:08.59 | tzafrir | If it's small, it's easier to send it around in a different (otherwise secure) channel |
| 16:09.19 | tzafrir | Look up one-time pad for something that is not :-) |
| 16:09.29 | Naikrovek | lol yeah |
| 16:09.53 | Naikrovek | private/public would be better here methinks. but if AES is the standard, and AES is implemented, we'll just have to find a secure way to distribute keys |
| 16:10.02 | Kobaz | how is it any harder to send 10 bytes than 1000 bytes |
| 16:10.04 | Naikrovek | probably by encrypting them with a public/private cipher |
| 16:10.22 | tzafrir | Naikrovek, using a private/public key is very expensive (computing wise). |
| 16:10.47 | Kobaz | tzafrir: you're talking about the human element... it's easier for *people* to look at a 10 byte key |
| 16:10.48 | Naikrovek | not so much that it isn't feasible |
| 16:11.55 | tzafrir | This is why public keys are used in communication protocols (like ssh, pgp, and Asterisk's IAX encryption) for the initial handshaking |
| 16:12.05 | tzafrir | And not for the actual data exchange |
| 16:12.57 | Naikrovek | right |
| 16:13.24 | Naikrovek | once the connection is secured, the data is sent and received symmetrically encrypted |
| 16:13.27 | Naikrovek | because it's fast |
| 16:13.34 | Naikrovek | and because you exchanged keys securely, it's secure |
| 16:13.55 | *** join/#asterisk cuco (~elcuco@95.35.227.51) |
| 16:14.00 | tzafrir | Naikrovek, key distribution is one issue. How do you invalidate a key if Kobaz got a hold of your private key? |
| 16:14.12 | Kobaz | change your key on the server |
| 16:14.38 | tzafrir | And change the public key on anyh server communicating with it. |
| 16:15.04 | tzafrir | Until then they have no way to tell you could actually be connecting instead of Naikrovek |
| 16:15.27 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:15.37 | Kobaz | how is that any different than a compromise in the public/private system |
| 16:15.56 | Kobaz | a breach is a breach |
| 16:16.27 | Kobaz | it's equally bad if the baddies get something that's secret, no matter what the approach is |
| 16:16.57 | [TK]D-Fender | Kobaz: I disagree... |
| 16:17.27 | Naikrovek | all i'll have to do is change my private/public key |
| 16:17.35 | Naikrovek | anyone who has my old key will no longer be able to connect |
| 16:17.41 | Naikrovek | they will ask for new |
| 16:17.46 | Naikrovek | i will give them new |
| 16:17.46 | Kobaz | Naikrovek: same thing with a symetric key system |
| 16:17.48 | [TK]D-Fender | Kobaz: If the approach is a but that only leaves a window open from 10:00am to 10:02am on Feb 29th that is far better than a conveniently re-exploitable hole :) |
| 16:18.04 | Kobaz | change your key on the server, and no one will be able to connect until they get the new one |
| 16:18.26 | Kobaz | [TK]D-Fender: how would anything be re-exploitable? |
| 16:18.26 | Naikrovek | Kobaz: yes, but i won't have to rely on your ability to keep the public key to yourself, because it isn't a requirement. with symmetric keys you need to not only rely on your own security but the security of everyone who is in possession of the key |
| 16:18.31 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:18.40 | Kobaz | Naikrovek: yeah, true |
| 16:18.45 | drmessano | Kobaz: It secures YOUR connection to them, but leaves the wide open to MiM attacks |
| 16:18.56 | drmessano | them* |
| 16:18.59 | [TK]D-Fender | Kobaz: A breach that can be easily repeated is worse than one that can't :) |
| 16:20.06 | Kobaz | you're also equally susceptible to mim unless everyone also has a copy of the servers public key to validate |
| 16:20.41 | *** join/#asterisk cuco (~elcuco@95.35.227.51) |
| 16:20.44 | Kobaz | i totally agree that public/private key is a better system overall... but i'm saying the risks associated with a breach are exactly the same |
| 16:21.23 | Kobaz | and you have to be equally careful no matter what security approach you use... that secret keys do not get out in the wild, and that you verify who you are communitating with to avoid mim |
| 16:21.31 | drmessano | Key distribution makes it entirely different |
| 16:22.49 | Naikrovek | Kobaz: this is why things like SSL and SSH work so well. per-session key generation and encryption |
| 16:23.05 | Kobaz | yes, i agree that it's a good approach |
| 16:23.11 | Naikrovek | even if someone gets a session key, it's invalid as soon as the session is over |
| 16:23.22 | Kobaz | yes |
| 16:23.26 | drmessano | Mass distribution of a public key makes a presumption of a secure point of authentication with you, and once the secret key is compromised, you may secure YOURSELF again by switching to a new key, but there's still that public key out there just waiting to be exploited |
| 16:23.29 | Kobaz | but if someone gets the server key, you're fscked anyway |
| 16:23.32 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:26.16 | *** join/#asterisk cuco (~elcuco@95.35.227.51) |
| 16:27.15 | bent_screwdriver | anyone here use the polycom soundstation IP 7000? |
| 16:27.28 | [TK]D-Fender | bent_screwdriver: What about them? |
| 16:28.37 | bent_screwdriver | I have some users wanting a new conference phone (currently they just have a analog one) and i was thinking about getting this one. They have IP650's in their offices and wanted to know if they were similar in quality and configuration (on the phone side) |
| 16:28.58 | Naikrovek | yes, they are |
| 16:29.01 | Naikrovek | to both |
| 16:29.05 | Naikrovek | quality and configuration |
| 16:29.12 | Naikrovek | i had an IP 6000 that was most excellent |
| 16:29.28 | [TK]D-Fender | bent_screwdriver: Even better clearly |
| 16:29.51 | bent_screwdriver | Naikrovek , [TK]D-Fender: thx! sold! |
| 16:32.20 | [Outcast] | I am going to ask again: Does anyone have a working broadvoice config? |
| 16:36.26 | *** join/#asterisk s0lid (~s0lid@112.201.240.34) |
| 16:38.30 | *** join/#asterisk s0lid (~s0lid@112.201.240.34) |
| 16:38.40 | *** join/#asterisk cuco (~elcuco@95.35.178.11) |
| 16:41.18 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:47.36 | *** join/#asterisk patrick^ (~patrick_@2001:470:1d:349:219:21ff:fe4e:f5de) |
| 17:00.18 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 17:07.03 | Gugge | [Outcast]: one would think the config on their page should work. :) |
| 17:07.16 | [Outcast] | Gugge: it does not |
| 17:07.17 | Gugge | and if it doesnt, they should provide a working one |
| 17:08.14 | Gugge | so you actually got you asterisk to accept the [sip.broadvoice.com] now? |
| 17:08.44 | [Outcast] | Gugge: true, but the question is, is anyone else using broadvoice |
| 17:08.53 | [Outcast] | Gugge: the name of peer does not matter |
| 17:08.53 | [Outcast] | \ |
| 17:09.03 | Gugge | [Outcast]: it shouldnt :) |
| 17:09.10 | Gugge | but it shouldnt crash either |
| 17:09.14 | [Outcast] | true |
| 17:09.45 | [Outcast] | I had this working 6 months ago, but can't get to the box that has the config |
| 17:09.58 | [TK]D-Fender | [Outcast]: So what you're saying is you didn't do it exactly like the guide showed... |
| 17:10.20 | [Outcast] | [TK]D-Fender: it did at first.......also I wrote the guide |
| 17:10.31 | [TK]D-Fender | [Outcast]: How about "at last"? |
| 17:10.40 | Gugge | shame on you, writing a non working guide :) |
| 17:10.49 | [Outcast] | Gugge: it worked when I wrote it |
| 17:10.55 | [TK]D-Fender | [Outcast]: For someone who wrote the guide.. you don't seem to good at FOLLOWING it ;) |
| 17:11.05 | [Outcast] | [TK]D-Fender: blah |
| 17:11.27 | [TK]D-Fender | [Outcast]: "Do as I say, not as I do" is a very poor philosphy |
| 17:11.41 | Katty | ATTENTION |
| 17:11.48 | Katty | I HAVE GRILLED CHEESE, USING COLBY AND RYE BREAD |
| 17:11.50 | Katty | that is all. |
| 17:12.10 | Katty | hi Naikrovek (= |
| 17:12.37 | [TK]D-Fender | Katty: Grilled cheese ... you had me at grilled cheese ... |
| 17:12.46 | Naikrovek | hi Katty |
| 17:13.32 | Katty | hi fender (= |
| 17:14.16 | Katty | FYI, i use mayo to make grilled cheese instead of butter. |
| 17:14.28 | Katty | real mayo, none of the light or fat free stuff |
| 17:15.12 | [TK]D-Fender | :/ |
| 17:15.27 | Katty | what. |
| 17:15.44 | Katty | explain your reasoning to me there. |
| 17:16.40 | Kobaz | the fat free stuff is pretty bad for you |
| 17:17.12 | Kobaz | it's mostly hydroginated oils... which are the ones that stick to your arteries and veins |
| 17:17.13 | [TK]D-Fender | Katty: Normally you need butter (or margarine) for the fried outer bread effect... I can't really imagine how mayo acts as a substitute |
| 17:18.00 | Katty | i guess it's more of an egg coating |
| 17:18.02 | Gugge | [Outcast]: did you try this one, it doesnt require editing the hosts file. http://edvina.net/broadvoice/ |
| 17:18.05 | Katty | egg and oil |
| 17:18.07 | Katty | rather than butter |
| 17:18.09 | Qwell | Katty: so it's french toast |
| 17:18.11 | [TK]D-Fender | Katty: I'd have to try it... |
| 17:18.24 | Katty | Qwell: a bit, yeah, but with more oil |
| 17:18.30 | Katty | Qwell: and cheese. |
| 17:18.58 | Katty | checks ingredients |
| 17:19.30 | Katty | oil, water, eggs, vinegar, salt, sugar, lemon juice |
| 17:19.42 | Katty | not exactly french toast., |
| 17:20.04 | Katty | i suddenly wonder why mayo is white |
| 17:20.11 | Qwell | food coloring |
| 17:20.19 | Katty | food coloring isn't listed on the ingredient list |
| 17:20.21 | leifmadsen | white is not a colour -- it is a shade :) |
| 17:20.28 | Qwell | leifmadsen: your face is a shade |
| 17:20.36 | leifmadsen | Qwell: awwww how sweet! |
| 17:20.39 | Katty | i'll shade both your faces in a minute. |
| 17:20.52 | Katty | mayo should be yellow |
| 17:22.01 | Katty | oooh |
| 17:22.09 | Katty | lemon juice and oil emulsify |
| 17:22.52 | Katty | ain't that fancy |
| 17:22.57 | WIMPy | Katty: If you leave it standing long enough, I'm sure it will develop all colours of the rainbow :-) |
| 17:23.06 | Katty | WIMPy: eww :P |
| 17:24.34 | WIMPy | I once had the cover from a yoghurt that was acualyy quite pretty. |
| 17:24.40 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
| 17:25.26 | Naikrovek | cover? |
| 17:25.37 | Naikrovek | yoghurt? |
| 17:26.07 | Katty | i presume the lid to the container |
| 17:26.11 | Naikrovek | huh guess i been spelling that wrong |
| 17:26.19 | philipp64|laptop | howdy. dumb question, but what is the protocol for conveying SMS/MMS on an SS7 network, and what does that typically get converted into when hitting an IP (SIP) network? |
| 17:26.24 | Naikrovek | i presume the dried crap on top when he said cover |
| 17:26.46 | Katty | philipp64|laptop: as far as i know, it's just an email |
| 17:27.30 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 17:29.18 | [Outcast] | I need to get asterisk put the FQDN in the INVITE and not the ip |
| 17:31.14 | philipp64|laptop | Katty: seems that SNPP would be used on the IP leg. |
| 17:31.37 | Naikrovek | i'm going to ask for some software to prevent these people who I issue computers to from being profoundly stupid |
| 17:32.04 | Naikrovek | really need to lock these people down |
| 17:34.12 | [TK]D-Fender | Naikrovek: #bdsm |
| 17:34.18 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:34.29 | Naikrovek | not tie them down |
| 17:34.39 | Naikrovek | actually i don't even want to know about them |
| 17:34.46 | Naikrovek | #throwusersoffthecliff |
| 17:34.46 | [TK]D-Fender | Naikrovek: they support locks, cuffs, etc... |
| 17:36.34 | [Outcast] | so no here uses broadvoice any more? |
| 17:36.45 | Naikrovek | guess not |
| 17:36.50 | leifmadsen | Gugge: wow, that link you send on edvina.net is from 2004... |
| 17:36.54 | tzafrir | Naikrovek, http://darwinawards.com shows several methods of preventing users from being stupid |
| 17:38.05 | Naikrovek | eh i don't want to hurt them |
| 17:38.12 | Naikrovek | i just want to prevent them from hurting themselves |
| 17:38.17 | Naikrovek | or my equipment |
| 17:38.20 | Naikrovek | or my deadlines |
| 17:39.06 | Naikrovek | or my budget |
| 17:39.51 | tzafrir | Naikrovek, the point of that site is that it won't be you who hurt them. Ahem |
| 17:40.04 | Naikrovek | my point is that i wish them no ill will |
| 17:40.07 | Naikrovek | but yeah i got ya |
| 17:40.14 | tzafrir | But then again you raise a valid point regarding the equipment |
| 17:40.33 | [TK]D-Fender | tzafrir: Actually, those are samples of users preventing themselves being stupid in the future... |
| 17:40.46 | Naikrovek | not all darwin award winners die |
| 17:40.49 | Gugge | [Outcast]: did you try the dial() with SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 17:40.51 | carrar | Naikrovek, train them or fire them |
| 17:40.52 | Naikrovek | certainly the majority though |
| 17:41.27 | [Outcast] | Gugge: yes |
| 17:41.29 | [TK]D-Fender | [Outcast]: Did you also set the realm? I don't recall seeing that |
| 17:41.32 | Naikrovek | carrar: not in a position to train them or fire them, unfortunately. i can give rudimentary instructions but i've been told that they're really busy and I shouldn't be bothering them with anything |
| 17:41.35 | Gugge | [Outcast]: and that does not work? |
| 17:41.51 | [Outcast] | setting the realm won't matter |
| 17:41.56 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:42.01 | carrar | Naikrovek, work for a different company that has smrt people |
| 17:42.04 | Naikrovek | http://www.geeky-gadgets.com/custom-nes-mouse-pads-20-07-2010/?utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+geeky-gadgets+%28Geeky+Gadgets%29&utm_content=Google+Reader |
| 17:42.14 | Naikrovek | carrar: would rather whip this lot into shape |
| 17:42.22 | Naikrovek | i just have to start with management |
| 17:42.27 | Naikrovek | rather than the users |
| 17:42.34 | carrar | make it happen |
| 17:42.38 | [Outcast] | [TK]D-Fender: it never gets the challege |
| 17:42.43 | Naikrovek | carrar: getting there |
| 17:42.44 | bmoraca_work | anyone know how well ODBC voicemail users/storage and CDRs might work over a WAN link (probably no less than 10mbps full duplex)? |
| 17:42.49 | [Outcast] | it just invite and 403 is returned |
| 17:42.51 | Naikrovek | great strides so far, |
| 17:43.05 | [TK]D-Fender | [Outcast]: Oh yeah... this shouldn't affect that line of the invite. |
| 17:43.11 | Naikrovek | bmoraca_work: i imagine it would be fine if the link never goes down |
| 17:43.28 | bmoraca_work | yeah...that's what i'm afraid of |
| 17:43.37 | [TK]D-Fender | [Outcast]: Highly recommend hitting the mailing list and contacting OEJ. |
| 17:43.50 | bmoraca_work | i wonder what would happen if the source was unavailable temporarily... |
| 17:43.50 | [Outcast] | already sent him an email |
| 17:43.55 | Naikrovek | if you can find some local redundancy or some way to seamlessly take over if/when the link goes down |
| 17:44.15 | Naikrovek | otherwise anything attempted to be stored while the link is down would likely be lost |
| 17:44.42 | Gugge | [Outcast]: when i use Dial(SIP/${EXTEN}@sbc0${RAND(1,2)}.xxx.com); i see sbc0X.xxx.com in the invite uri |
| 17:45.04 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:45.10 | Naikrovek | bmoraca_work: maybe in the dialplan you could account for failure and log CDR and save voicemail elsewhere in the event of failure |
| 17:45.11 | bmoraca_work | i plan on syncronizing sip tables and various other information, but i would like to minimize how much data is trucking across my links...if I have to replicate voicemails to 10 different asterisk boxes, that could be a lot of traffic |
| 17:46.04 | [Outcast] | Gugge: hmmm |
| 17:46.06 | bmoraca_work | wonder if unixodbc has built-in failover if a DSN becomes unavailable... |
| 17:46.43 | bmoraca_work | [Outcast]: is one of the devices behind a NAT or traversing any kind of firewall (VPN, etc)? |
| 17:47.18 | [Outcast] | bmoraca_work: the asterisk box is on the DMZ |
| 17:48.00 | bmoraca_work | [Outcast]: typically when auth fails in that manner (challenge issued, but no response seen from the UA), in my experience, it's because the UA isn't getting the 403 failure. |
| 17:48.25 | bmoraca_work | usually that's a result of a bad NAT translation |
| 17:48.29 | [Outcast] | no challenge is being issued |
| 17:48.36 | [Outcast] | nat is fine |
| 17:48.43 | [Outcast] | did a trace with DV |
| 17:48.48 | [Outcast] | *BV |
| 17:48.49 | bmoraca_work | [Outcast]> it just invite and 403 is returned <===== the 403 is the challenge |
| 17:49.02 | [Outcast] | 403 is not the challege |
| 17:49.09 | [Outcast] | 401 would be a challenge |
| 17:49.47 | Gugge | [Outcast]: can you paste a full sip debug session somewhere? |
| 17:50.32 | bmoraca_work | you're correct...403 is forbidden. however, could still be a NAT issue. i've had some funky shit go on with broken NAT (ASA 8.22 firmware, for instance) |
| 17:50.36 | [Outcast] | already did: http://pastebin.ca/1904431 |
| 17:52.01 | Gugge | [Outcast]: and one where you use SIP/number:pass::username@sip.broadvoice.com |
| 17:52.13 | Gugge | in Dial |
| 17:53.17 | bmoraca_work | Gugge: that's still not going to provide credentials to BV until BV asks for them |
| 17:53.42 | bmoraca_work | (can't salt the MD5 hash until BV responds with some parameters) |
| 17:53.48 | Gugge | but that may put sip.broadvoice.com in the URI |
| 17:54.05 | Gugge | and that is what [Outcast] seems to think is the issue |
| 17:55.36 | [Outcast] | Gugge: the trace is exactly the same |
| 17:56.02 | Gugge | do you have some kinde of dnsmgr stuff loaded ? |
| 17:56.49 | *** part/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:58.38 | [Outcast] | for hahas going to try an older version of asterisk |
| 17:58.57 | bmoraca_work | i would try it outside of your NAT first |
| 17:59.14 | Gugge | what NAT ? |
| 17:59.33 | bmoraca_work | it's clear that NAT isn't affecting the FLOW of the packets, but it could be affecting the CONTENT of the packets, and that's not something you can see from an Asterisk SIP debug |
| 18:01.01 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 18:01.02 | Gugge | but is there any NAT? |
| 18:01.51 | bmoraca_work | Gugge: "Reliably Transmitting (NAT) to 147.135.20.221:5060:" |
| 18:02.32 | *** join/#asterisk hopper75 (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:03.17 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 18:22.46 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:22.51 | [TK]D-Fender | NAT should have no impact on the fact he has an IP and not the hostname in the INVITE line as he shows |
| 18:29.05 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:31.52 | [sr] | where dahdi stores it's logs by default? |
| 18:32.02 | [TK]D-Fender | [sr]: What "logs"? |
| 18:32.22 | [sr] | incoming call's for example, well, if it saves anything |
| 18:32.28 | [sr] | i'm suposing it does |
| 18:32.43 | [TK]D-Fender | [sr]: that isn't DAHDI, that is ASTERISK. Go read the chapter on CDR in the BOOK. |
| 18:32.50 | [Outcast] | that is total BS |
| 18:33.08 | [Outcast] | I make call using 1.2 I have not change my config and it goes through |
| 18:33.22 | [sr] | [TK]D-Fender: i'm not talking about the subject of the afternoon (night here now) |
| 18:33.27 | [TK]D-Fender | [Outcast]: How about using "not trunk"? |
| 18:33.35 | [Outcast] | something has gotten lost along the way |
| 18:33.35 | [Outcast] | possible |
| 18:33.39 | [Outcast] | will try that nex |
| 18:33.44 | [TK]D-Fender | [14:32]<[sr]>incoming call's for example, well, if it saves anything <-- call logging = CDR |
| 18:33.59 | [TK]D-Fender | [cthat is it. there is nothing more. Each channel driver doesn't keep its own logs |
| 18:34.59 | [Outcast] | 1.2 invite = INVITE sip:4132412554@sip.broadvoice.com SIP/2.0 |
| 18:35.04 | [Outcast] | grr |
| 18:35.10 | [sr] | [TK]D-Fender: let me try to explain, for example incoming call's before it's sent to asterisk, for example LCR has its own log which is nice to debug and easy, dahdi doesn't have something like it? |
| 18:36.52 | *** join/#asterisk robertkeizer (~robertkei@host-208-81-1-163.les.net) |
| 18:37.32 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 18:38.36 | pabelanger | Did somebody say NT PTMP mode will be added in 1.8? |
| 18:38.50 | pabelanger | for BRI |
| 18:39.01 | tzafrir | pabelanger, it's already working in trunk |
| 18:39.43 | pabelanger | tzafrir: Thanks, is there a specific version of DAHDI needed? |
| 18:39.53 | tzafrir | dahdi? no |
| 18:40.02 | tzafrir | libpri: >= 1.4.11 |
| 18:40.11 | pabelanger | tzafrir: Ya, yes. Sorry |
| 18:40.49 | pabelanger | tzafrir: do you happen to know which commit added it? Or was it a series of patches? |
| 18:41.35 | pabelanger | Think I found it |
| 18:41.37 | pabelanger | http://svnview.digium.com/svn/libpri?revision=1242&view=revision |
| 18:42.07 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 18:43.19 | tzafrir | in chan_dahdi it is related to other major changes |
| 18:43.30 | tzafrir | So it's not a quick fix to backport to 1.6.2 |
| 18:43.49 | pabelanger | <PROTECTED> |
| 18:43.52 | bmoraca_work | fixitfixitfixit |
| 18:44.06 | pabelanger | as long as trunk supports it :) |
| 18:45.25 | [Outcast] | [TK]D-Fender: still compiling |
| 18:45.46 | [Outcast] | [TK]D-Fender: if this doesn't work, I have found two bugs today |
| 18:48.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:51.08 | *** join/#asterisk cracoucas (cracoucas@83.159.13.95) |
| 18:52.43 | [Outcast] | [TK]D-Fender: ok 1.6.2.9 works so trunk is broken |
| 18:53.14 | [Outcast] | thank God I thought I was going nuts |
| 18:54.48 | Gugge | and thats why i dont use trunk :) |
| 18:54.55 | cracoucas | D-fender is is possible with asterisk to see from what network the incoming call is coming from ? |
| 18:55.09 | cracoucas | Gugge any idea? |
| 18:55.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 18:55.33 | Gugge | depends what you mean by network |
| 18:56.02 | cracoucas | Gugge , I want to ban the only few networks that allow callerID spoofing in the US |
| 18:56.27 | p3nguin | Good luck with that. |
| 18:56.47 | cracoucas | Gugge in france it is illegal to do so , so I want to check what is the networks that is used for the incoming call |
| 18:56.57 | Gugge | you cant |
| 18:57.08 | Gugge | i can call you from any number i want |
| 18:57.35 | cracoucas | I have no clue of what network you are using? |
| 18:57.42 | Gugge | none |
| 18:57.46 | [Outcast] | I guess I need to submit some bugs |
| 18:57.47 | p3nguin | You'll have absolutely no idea. |
| 18:58.31 | cracoucas | ok thanks |
| 18:58.47 | cracoucas | bad news |
| 18:59.16 | Gugge | cracoucas: why do you care? |
| 18:59.40 | cracoucas | gugge because if I can ban foreign networks I can use callerID authentication |
| 18:59.54 | [TK]D-Fender | [Outcast]: Oh, you're still going crazy .. you just happento be RIGHT on this one point ;) |
| 19:00.14 | [Outcast] | [TK]D-Fender: ROFL |
| 19:00.22 | Gugge | cracoucas: dont ever trust callerID :) |
| 19:00.26 | *** join/#asterisk radic (~radic@dslb-094-216-245-206.pools.arcor-ip.net) |
| 19:00.37 | [TK]D-Fender | Gugge: We've been trying to beat this idea into his head for DAYS |
| 19:00.44 | p3nguin | You can't accurately authenticate anything using caller id. |
| 19:01.06 | radic | how can I set a filename for Monitor() without starting the monetoring in the dialplan? |
| 19:01.07 | p3nguin | authentirification |
| 19:01.32 | cracoucas | in france you can not fake callerID my idea is to ban all foreign networks. there must be a way in the logs to exclude all calls from outside french network |
| 19:01.46 | [TK]D-Fender | radic: You set it when you call it. So make sure you have the name in a VARIABLE and pass that on to Monitor |
| 19:01.59 | Corydon76-dig | cracoucas: so because it's illegal, it doesn't happen? |
| 19:02.26 | Corydon76-dig | cracoucas: Are murder and theft illegal in France? Do they happen? |
| 19:02.42 | Gugge | cracoucas: so noone has access to pass on callerid on forwards in france? |
| 19:02.49 | radic | [TK]D-Fender: monetoring is started if someone here presses #9 |
| 19:03.09 | cracoucas | Gugge not any network in france can fake callerID |
| 19:03.18 | Naikrovek | uh oh |
| 19:03.19 | Gugge | cracoucas: is it faking to pass on a callerid ? |
| 19:03.23 | WIMPy | Gugge: Normally these thing happen automatically. |
| 19:03.25 | Naikrovek | is allergic to something in that mixed nuts container |
| 19:03.26 | Corydon76-dig | cracoucas: complete and utter baloney |
| 19:03.30 | p3nguin | That's what they tell you, anyway. |
| 19:03.41 | Naikrovek | but which nut is it that i'm allergic to.... |
| 19:03.49 | Naikrovek | sounds like the time for SCIENCE |
| 19:04.11 | p3nguin | Bill Nye to the rescue! |
| 19:04.14 | Gugge | WIMPy: what happens automatically? |
| 19:04.17 | radic | [TK]D-Fender: the problem ist how could I set options and filename vor monetoring If Monitor() isn't in the dialplan |
| 19:04.34 | *** part/#asterisk cracoucas (cracoucas@83.159.13.95) |
| 19:05.18 | [TK]D-Fender | radic: ... The where the hell is it? |
| 19:05.43 | WIMPy | Gugge: Caller ID transmissin when forwarding calls. |
| 19:06.01 | radic | [TK]D-Fender: in features.conf |
| 19:06.05 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 19:09.28 | [TK]D-Fender | radic: go read CHANNELVARIABLES.TEX |
| 19:12.16 | Naikrovek | OR ELSE |
| 19:12.22 | Naikrovek | ? |
| 19:12.31 | Qwell | or ELSE |
| 19:12.34 | Qwell | much different |
| 19:13.36 | Naikrovek | if (radic.hasRead("channelvariables.tex")) { /* reward */ } else { /* punishment */ } |
| 19:17.10 | Corydon76-dig | 30 lashes with a wet noodle? |
| 19:17.35 | Qwell | the read() function doesn't set hasRead properly. I've been meaning to fix that. |
| 19:17.36 | [TK]D-Fender | Corydon76-dig: Satb in the eye with a rusty spork. |
| 19:17.40 | [TK]D-Fender | stab* |
| 19:19.34 | Corydon76-dig | Oddly enough, the only metal spork I've ever seen is titanium, which doesn't rust |
| 19:20.05 | Naikrovek | perfect example of where knowledge ruins the spirit of the statement |
| 19:20.22 | Nugget | heh |
| 19:20.31 | radic | I couldn't find channelvariables.tex |
| 19:20.33 | [TK]D-Fender | Corydon76-dig: I'm quite sure ti will, over time. Remember that "stainless steel" is just resistant. |
| 19:20.39 | [TK]D-Fender | radic: LOOK HAREDER |
| 19:20.50 | Naikrovek | kinda like where people say "it smells like strawberry" - - well strawberries don't have a smell, according to my nose |
| 19:20.52 | [TK]D-Fender | radic: its in your SOURCE TARBALL |
| 19:21.08 | radic | I dosn't build it from source |
| 19:21.12 | Corydon76-dig | Naikrovek: I was actually all hyped when I heard they were going to be holding public teabagging parties |
| 19:21.16 | [TK]D-Fender | radic: Well go DOWNLAOD IT |
| 19:21.40 | Corydon76-dig | I totally want to see a bunch of windbags get teabagged |
| 19:21.40 | Qwell | Naikrovek: tip; gas leaks smell like strawberries. you'd better learn the smell. |
| 19:22.03 | [TK]D-Fender | Qwell: ... thats supposed to be "rotten eggs" |
| 19:22.38 | Naikrovek | Corydon76-dig: lol |
| 19:22.38 | Corydon76-dig | Rotten eggs only smell in the presence of oxygen |
| 19:22.41 | WIMPy | No gas anymore here. They recycled the old gas pipes to pull fibes through, wich I find much more usefull actually. |
| 19:22.59 | Qwell | [TK]D-Fender: growing up, we always bought our strawberries from the guy on the freeway exit |
| 19:23.19 | Naikrovek | Qwell: sounds legit. how were those LSD trips? |
| 19:25.11 | Corydon76-dig | Naikrovek: he grew up in the 80s. They were made of crack |
| 19:25.56 | [TK]D-Fender | I would never do a drug named after a part of my ass... |
| 19:26.49 | Corydon76-dig | And yet, you have no problem using a colon |
| 19:28.51 | Kobaz | [TK]D-Fender: classic dennis leary |
| 19:29.30 | [TK]D-Fender | Kobaz: Yup |
| 19:30.39 | *** join/#asterisk redax (redax@r6.hu) |
| 19:30.43 | redax | hi |
| 19:30.57 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 19:30.58 | Corydon76-dig | I prefer where he confesses to a nun that the profane song she caught his son singing at Catholic school was something he wrote |
| 19:32.17 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 19:32.32 | [sr] | [TK]D-Fender: i give up, i don't find nothing in dahdi that can give me what comes in into the ports... |
| 19:32.56 | [TK]D-Fender | [sr]: this has nothign to do with DAHDI. |
| 19:33.18 | [sr] | [TK]D-Fender: but there's nothing in asterisk log also, with verbose 15 |
| 19:33.41 | [TK]D-Fender | [sr]: last time... *** CDR *** |
| 19:34.49 | [sr] | [TK]D-Fender: ok... but i'm not talking the same thing we discuss some hours ago... |
| 19:35.27 | [TK]D-Fender | [sr]: And your ability to clearly introduce new topics sucks something fierce. |
| 19:35.38 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 19:35.50 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 19:36.21 | [sr] | [TK]D-Fender: i'm just testing dahdi to see how it works..nothing else |
| 19:36.41 | [TK]D-Fender | [sr]: Your description is still incredibly weak |
| 19:36.53 | [sr] | ok... let me try |
| 19:37.34 | [sr] | same setup as with LCR, ports configured as TE mode, signalling=euroisdn |
| 19:37.46 | [sr] | port status OK by dahdi |
| 19:37.49 | *** join/#asterisk markfeatherston_ (~mark@71.4.41.98.ptr.us.xo.net) |
| 19:38.38 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 19:38.40 | redax | hi D-Fender |
| 19:38.43 | [sr] | with LCR, i can see on it's logs/console incoming call's, even that is not configured on asterisk |
| 19:38.49 | Naikrovek | anyone try alien swarm on steam? i really dig that game |
| 19:38.50 | WIMPy | Hi [sr]. You're already switching to dahdi? |
| 19:39.17 | redax | uh.. lcr and mISDN sucks |
| 19:39.25 | [sr] | WIMPy: no! i just wanna give it a try to know how it works.. i'll stay with LCR |
| 19:39.48 | WIMPy | [sr]: AFAIK Sangomas wanpipe stuff can do logging/debugging. Dahdi won't except via Asterisk. |
| 19:39.54 | redax | last time I tried lcr it worked for 1hour after it crashed my kernel :D |
| 19:40.01 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 19:40.02 | redax | last time ~2weeks |
| 19:40.35 | [sr] | WIMPy: that's why i don't find any option/log of dahdi then.. |
| 19:40.50 | WIMPy | redax: Well, let's see, what 1.8 comes up with, but so far they have been the only halfway (misdn) stable options for me. |
| 19:43.22 | WIMPy | A bit of a debugging / status option for dahdi could be very helpfull, I think. |
| 19:43.30 | redax | well... I tried to use misdn for years. I always sucked that way. even when I pissed off and bought a digium B410P card :) |
| 19:43.42 | redax | because b410p has misdn driver as well not zaptel |
| 19:43.48 | [sr] | WIMPy: i vote for it too :P |
| 19:44.03 | redax | now I use only dahdi / sip |
| 19:44.57 | [sr] | WIMPy: well... lcr development doesn't seem to many fast... and the options that are going to be introduzed in 1.8x may be useful in the future, but for sure LCR will have then by that time also(i hope) |
| 19:45.00 | WIMPy | Well, I still have a box using misdn, but only in TE mode. That is stable, but NT mode gave some unexpected behaviour sometimes. LCR/misdn2 has been rock solid. |
| 19:45.48 | WIMPy | Not sure. Andreas is working on far too many projects, I think. |
| 19:45.52 | redax | WIMPy: must agree.. if you have only one PCI/USB TE card, there's no problem at all with misdn |
| 19:46.05 | WIMPy | But nevertheless I submitted a path yesterday :-) |
| 19:46.36 | WIMPy | But misdn is gone anyway. |
| 19:46.49 | redax | but lcr is the same |
| 19:46.58 | WIMPy | No |
| 19:47.13 | redax | the mISDN core locks up... |
| 19:47.14 | WIMPy | It's misdn2. Thats a new design. |
| 19:47.18 | redax | whats then? |
| 19:47.37 | redax | v2 or not it locks up. and blocks your io subsys |
| 19:48.22 | WIMPy | I haven't had any issues since changing from misdn to lcr. |
| 19:49.13 | redax | maybe you didn't had any prob with misdn then. |
| 19:49.34 | WIMPy | I've been staying with misdn for far too long as I thought I dodn't want another application that only translates APIs, but I should have done that move earlier. |
| 19:49.50 | WIMPy | I only had problems with misdn in NT mode. |
| 19:50.18 | redax | main problem is the _instability_ |
| 19:50.19 | WIMPy | I use both here. the bos still running misdn only uses TE. |
| 19:50.39 | redax | it works for hours or days but someday it stop working |
| 19:50.57 | WIMPy | Yes, but as I said: Only when using NT mode. At least that was my experience. |
| 19:51.05 | redax | and if you unload and loadd misdn modules, the whole system can hangs/locks up |
| 19:51.15 | Qwell | redax: don't do that then |
| 19:51.16 | redax | and you have to ride to the box |
| 19:51.31 | WIMPy | That sounds like the wrong version for your kernel. |
| 19:51.41 | *** join/#asterisk andres833 (~andres833@200.119.41.207) |
| 19:51.57 | redax | normally I can unload/load misdn modules as many times I want |
| 19:52.12 | WIMPy | But finding the right misdn for your kernel could be a real PITA. |
| 19:52.19 | redax | after it get an irq relateed oops. you don't want to unload the modules |
| 19:52.51 | redax | Qwell: I don't use misdn by now |
| 19:55.06 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:55.38 | WIMPy | Actually now I think that two applications have their advantage: I can still make and receive calls when Asterisk dies. :-) |
| 19:58.24 | *** join/#asterisk bn-7bc (bjarne@mac.wlan.noare-1.holmedal.net) |
| 20:01.39 | *** join/#asterisk hesco (~hesco@c-24-99-160-121.hsd1.ga.comcast.net) |
| 20:03.08 | hesco | In my console I'm seeing a brute force attack rendering noise reading: [2010-07-20 15:57:21] NOTICE[3606]: chan_sip.c:18390 handle_request_register: Registration from '"9492"<sip:9492@<the_ip_at_edge_of_network>>' failed for '128.134.51.168' - No matching peer found |
| 20:04.19 | Gugge | hesco: and? :) |
| 20:05.27 | hesco | any advice on best practices for blocking this sort of thing? |
| 20:05.50 | florz | hesco: not |
| 20:05.57 | hesco | any list of IP ranges one might consider blocking at the firewall? |
| 20:06.08 | florz | hesco: none |
| 20:06.19 | Gugge | when it happens, block the ip |
| 20:06.32 | florz | Gugge: no |
| 20:06.37 | [TK]D-Fender | hesco: Sure. Block china, etc |
| 20:06.57 | [TK]D-Fender | hesco: Or jsut be very restrictive as to what you do allow |
| 20:07.11 | Gugge | florz: and why sholdnt he block ip's trying to bruteforce ? |
| 20:07.32 | florz | Gugge: better question: why should he? |
| 20:07.48 | Gugge | to save resources |
| 20:07.59 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:08.14 | Gugge | to get rid of the registraion failed logs |
| 20:08.19 | Gugge | to feal safer |
| 20:08.22 | Gugge | a lot of reasons :P |
| 20:08.40 | florz | well, "feeling safer" is kindof pointless |
| 20:08.43 | Gugge | sure |
| 20:09.01 | florz | saving resources is pointless, too, unless those resources cost you anything |
| 20:09.34 | florz | the log stuff ... well, _maybe_, but only if you are sure that just filtering the logs isn't easier and less prone to undesired side effects |
| 20:10.36 | *** join/#asterisk _pepo_ (c837ea09@gateway/web/freenode/ip.200.55.234.9) |
| 20:11.36 | hesco | always a concern as I learn new technology, those undesired side effects |
| 20:11.42 | florz | so, all in all, yeah, there _may_ be resons for blocking - but usually it's somewhere between pointless and dangerous |
| 20:11.53 | Gugge | i agree |
| 20:11.59 | Gugge | but its still the only thing he _can_ do |
| 20:13.07 | Gugge | unless you count "doing nothing" as a thing :) |
| 20:13.31 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 20:13.46 | redax | what is the easiest way to send out sms if user has new voicemail message? |
| 20:14.02 | redax | read the voicemail/..../msg*.txt files? |
| 20:14.43 | florz | Gugge: well, he just as well can start drawing stick figures onto his desk, that's "doing something" as well, and probably less dangerous than any of that blocking business ... |
| 20:15.16 | p3nguin | app_voicemail can send email saying you have no voicemail. Configure it to send the email to your app that does SMS. |
| 20:15.28 | Gugge | florz: i really dont see the danger in blocking an ip thats bruteforcing (he needs to be sure it is though) |
| 20:15.53 | Gugge | redax: i use the mailcmd option in voicemail.conf to call a script that sends mail and sms |
| 20:15.57 | florz | Gugge: "IPs" don't brute force, so ... |
| 20:16.59 | Gugge | florz: what is the danger in blocking an ip someone is using to try to bruteforce your system? |
| 20:17.24 | florz | Gugge: the danger is that you may start blocking legitimate requests |
| 20:18.13 | florz | Gugge: (while the benefit is none) |
| 20:18.25 | redax | Gugge: aaa. not bad. but I have to batch sms'es.. like send out sms only 1 per hour |
| 20:18.48 | redax | Gugge: like you have 3 messages in the last hour... |
| 20:18.49 | Gugge | redax: then add them to the batch from the mailcmd script |
| 20:18.59 | Gugge | and use another script to send every hour |
| 20:19.01 | p3nguin | s/no/new/ |
| 20:19.18 | redax | Gugge: yep. that's why I said not bad :) |
| 20:19.51 | redax | mailcmd sound good |
| 20:20.43 | Gugge | florz: and that would be the part where he should be sure the ip isnt used for anything else |
| 20:21.31 | Gugge | but hey, blocking doenst give you much |
| 20:21.50 | Gugge | it just makes sure that ip stops hammering asterisk |
| 20:21.52 | florz | Gugge: which is pretty much impossible to do unless you only serve specific networks, in which case you'd probably just use a whitelist, and wouldn't be asking questions about it here |
| 20:24.12 | [Outcast] | pabelanger: I will get the bt redone later tonight busy at work ATM |
| 20:25.14 | pabelanger | [Outcast]: np. what version of Asterisk you upgrading from? |
| 20:25.25 | [Outcast] | new install |
| 20:25.52 | [Outcast] | that config was an example provided by the provider |
| 20:26.12 | [Outcast] | it works fine in 1.6.2.9 but not in SVN |
| 20:26.40 | pabelanger | [Outcast]: Ok, that is what I wanted to know. |
| 20:26.53 | pabelanger | you'll need to update your sip.conf too, you have some out dated settings |
| 20:27.30 | [Outcast] | pabelanger: not a problem, but that config is used by a lot of people for connecting broadvoice. |
| 20:27.45 | *** join/#asterisk itiliti (~itiliti@74.222.60.226) |
| 20:27.53 | [Outcast] | or I think is used by alot of people |
| 20:27.56 | pabelanger | [Outcast]: Agree, but canreinvite does not exist anymore |
| 20:28.03 | [Outcast] | pabelanger: ok |
| 20:28.23 | [Outcast] | pabelanger: what is the option I should be using now? |
| 20:28.35 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 20:28.50 | pabelanger | directmedia |
| 20:28.56 | pabelanger | it is documented in UPGRADE.txt |
| 20:31.30 | [TK]D-Fender | checkout time, BBIAB |
| 20:31.49 | leifmadsen | ~alot |
| 20:31.50 | infobot | [alot] raping the English language |
| 20:31.54 | leifmadsen | heh |
| 20:31.59 | *** join/#asterisk edguy3 (~edguy@212.159.32.123.broadband.plus.dyn.plus.net) |
| 20:32.13 | Qwell | looks at leifmadsen |
| 20:32.25 | Qwell | looks at timestamps of when things were said in different places |
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| 20:38.48 | jhirley | http://xkcd.com/172/ |
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| 21:05.36 | tzafrir | WIMPy, what are you missing? |
| 21:05.44 | ChannelZ | courage |
| 21:06.06 | WIMPy | Err, from what? |
| 21:06.11 | tzafrir | In terms of debugging / status for DAHDI |
| 21:06.20 | WIMPy | Or should I just say: Money! |
| 21:06.31 | ChannelZ | there you go |
| 21:06.35 | WIMPy | ah |
| 21:07.11 | WIMPy | Well, a possibility to check if the line is up and running without Asterisk. And a usage dispay would be nice as well. |
| 21:07.38 | WIMPy | With dahdi-tool you can see what channels are configured, but not which are actually in use. |
| 21:07.47 | tzafrir | well, /proc/dahdi/* shows if the line is opened by anybody |
| 21:08.06 | tzafrir | Which typically translates to: configured in Asterisk |
| 21:08.22 | tzafrir | As for "in an active call" - |
| 21:08.32 | WIMPy | Line, yes, no info about channels. |
| 21:09.28 | WIMPy | And the terminology for the clocking seems to bee easy to misunderstand. |
| 21:09.45 | tzafrir | Check the config options OPTIMIZE_CHANMUTE and DAHDI_AUDIO_NOTIFY |
| 21:10.06 | WIMPy | The highest priority TE i/f shows as "Master", when it's actually slave, just the local source. |
| 21:10.19 | tzafrir | With those two I believe at least for ISDN (BRI, E1, T1) you should clearly see when a line is in use |
| 21:10.37 | tzafrir | And if not, it will only require a minor tweak to the driver |
| 21:11.08 | WIMPy | makes a note for when I use my test box next time. It's the only one using dahi ATM. |
| 21:11.31 | tzafrir | "Master" does not refer to master/slave on the line. |
| 21:11.48 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 21:11.50 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 21:12.18 | WIMPy | Yes, I know. That's why I said it easy to misunderstand. |
| 21:15.34 | radic | WIMPy: du verfolgst mich! |
| 21:16.30 | WIMPy | Nah |
| 21:17.32 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 21:19.54 | WIMPy | tzafrir: And it would be nice to be able to see if there is a working D-Channel. A simple up/down would be easier than having to enable intensive debug. |
| 21:20.41 | tzafrir | WIMPy, this is something you can't report in DAHDI. DAHDI can only report layer 1 |
| 21:21.22 | tzafrir | (actually our device fakes that: it checks when there's traffic in the D channel and reports it as "live", |
| 21:21.56 | tzafrir | Which breaks badly if you have traffic but no active D channel. e.g. connect a CPE to a CPE) |
| 21:22.18 | tzafrir | So this is something you should look for in Asterisk |
| 21:22.41 | tzafrir | WIMPy, so essentially, 'pri show spans' is not clear enough? |
| 21:22.47 | WIMPy | Ja, just like the wrong cable that makes your link receive-only. |
| 21:25.14 | [sr] | in that part of log, and i'm new at it, LCR show's about everything that is comming on the line |
| 21:26.42 | WIMPy | I'd have to check again. Not using dahdi often. But I didn;t feel too well informed. But that might be a lack of dahdi experience. |
| 21:28.19 | WIMPy | It makes sense, off course that the protocol is handled in Asterisk instead of different channel drivers, but it feels rather odd to debug/get status information. |
| 21:28.37 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:30.07 | WIMPy | Or hardware drivers rather. |
| 21:33.47 | *** join/#asterisk keli (asgaroth@unixstorm.org) |
| 21:47.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:48.55 | [sr] | guys |
| 21:49.36 | [sr] | i see that dahdi has a "pciradio" driver, is this for what i'm thinking? use an FM radio to music on hold, etc? |
| 21:54.40 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:55.10 | boodu | hello |
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| 22:05.02 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:10.26 | keli | hi there. |
| 22:11.53 | keli | I tried call through using this instruction http://www.voip-info.org/wiki/view/Asterisk+tips+call+through |
| 22:12.06 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 22:12.26 | keli | but I always get something like "lag" |
| 22:13.11 | keli | I sometines need to wait 10 seconds before asterisk recognizes the input |
| 22:13.51 | keli | if I don't wait asterisk just quits the call (with error) |
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| 22:22.26 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 22:23.06 | [TK]D-Fender | keith4: PASTEBIN is your friend... you should be showing us the failed call attempt at * CLI |
| 22:23.08 | [TK]D-Fender | ~pb |
| 22:23.09 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 22:23.11 | [TK]D-Fender | ^^^ |
| 22:23.14 | [TK]D-Fender | keli: rather |
| 22:23.57 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 22:27.04 | keli | moment please |
| 22:28.21 | keli | [Jul 21 00:28:01] WARNING[3212]: pbx.c:4415 __ast_pbx_run: Invalid extension '23', but no rule 'i' in context 'sonstige' |
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| 22:32.01 | keli | [TK]D-Fender: this is my extensions.conf http://asterisk.pastey.net/138753 |
| 22:33.24 | [TK]D-Fender | keli: I don't see a context heading so I have no idea what I'm looking at, and I don't see the failed call's CLI output either |
| 22:33.28 | [TK]D-Fender | keli: Try again... |
| 22:35.07 | keli | there is only one context header in that file [sonstige] |
| 22:35.37 | keli | its just to try the callthrough... and it works if i type slowly and wait for the digit to be said by astersik |
| 22:35.40 | [TK]D-Fender | is waiting to see the CALL. |
| 22:36.18 | keli | wait a sec |
| 22:38.54 | keli | http://asterisk.pastey.net/138755 |
| 22:39.04 | keli | there is the cli output |
| 22:39.17 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:39.23 | keli | started asterisk with vvvvvc |
| 22:39.57 | [TK]D-Fender | keli: Problem is that you have this in the same context -- exten => 49XXXXXXXX,1,Goto(sonstige,s,1) |
| 22:40.52 | keli | ok, thx i'll try to fix it |
| 22:40.57 | [TK]D-Fender | keli: _X was meant to collect the number digit by digit... UNFORTUNATELY you started with 4 as the first digit and because of that _X doesn't match immediately. this is because as far as Asterisk knows, you could be typing in that 49XXXXXXXX number |
| 22:41.34 | [TK]D-Fender | keBy the time you hit the 2nd digit it realizes that this exten isn't going to match anything and dies. |
| 22:41.52 | keli | but there's lag too if i type a '2' in the beginning |
| 22:41.56 | [TK]D-Fender | keli: You need to make this IVR in a COMPLETELY SEPARATE context. |
| 22:42.10 | [TK]D-Fender | keli: Because you have OTHER SHIT in this context <- |
| 22:42.22 | [TK]D-Fender | keli: This IVR needs to be in its own context |
| 22:42.31 | keli | oki thank you |
| 22:43.52 | MatBoy | [TK]D-Fender: shit ? bah ! |
| 22:44.07 | [TK]D-Fender | keli: Now I can also see there are pieces from that WIKI sample you didn't copy over, nor do I see you substituting in a way that is functional. |
| 22:45.36 | [TK]D-Fender | keli: Also considering even their sample dies if you wait 4 seconds without pressing # I consider this a poor implementation and actually too complex for nothing |
| 22:45.50 | keli | [TK]D-Fender: so wait one second again |
| 22:46.19 | [TK]D-Fender | Actually... no... it takes 10 seconds... but still ugly to even need that long or any special key to "end" the input |
| 22:49.25 | keli | next attempt: http://asterisk.pastey.net/138758 |
| 22:50.04 | [TK]D-Fender | [18:39]<[TK]D-Fender>keli: Problem is that you have this in the same context -- exten => 49XXXXXXXX,1,Goto(sonstige,s,1) |
| 22:50.20 | [TK]D-Fender | keli: You are STILL putting other things in this SAME CONTEXT! |
| 22:50.21 | *** join/#asterisk edguy3 (~edguy@195.166.51.2) |
| 22:50.23 | [TK]D-Fender | keli: NO! |
| 22:50.34 | keli | no i didnt fix it |
| 22:50.57 | [TK]D-Fender | keli: Then what's the point? |
| 22:51.03 | keli | it was just the whole conf because you told me that i didn't copy the whole wiki sample |
| 22:51.13 | [TK]D-Fender | .... |
| 22:51.32 | [TK]D-Fender | keli: Don't waste time with a partial fix after we've handed you specific details to fix |
| 22:51.49 | Kyosh | how do i tell asterisk to rotate log files instead of piling everything into the messages. file ? |
| 22:51.59 | [TK]D-Fender | keli: Do it ALL and then try it out and share. You are just showing something else that won't work. |
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| 22:57.11 | keli | [TK]D-Fender: thanks it works now |
| 22:57.21 | *** join/#asterisk guilhermebr (~Guilherme@189.63.78.130) |
| 22:58.53 | [TK]D-Fender | keli: Good to hear |
| 23:00.08 | Kyosh | nm, got it |
| 23:00.11 | Kyosh | gotta cron it |
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| 23:07.33 | redax | is there a way to get some channelvar from the connected pair channel ? |
| 23:11.47 | redax | like caller connected from queue to the agent... |
| 23:12.10 | redax | and I'd like to get some var from the caller's channelvars, in an AgentConnected macro |
| 23:13.04 | [TK]D-Fender | redax: Standard inheritance rules should apply. You can also use AMI to track which channels are bridged |
| 23:14.01 | redax | D-Fender, is there a way to dump _ALL_ channelvars for debugging purposes using AMI for a specified channel? |
| 23:14.49 | redax | I'm pretty sure I can get the channel uniqueid of the connected channelpair |
| 23:16.40 | *** join/#asterisk anthm][ (~anthm@173-99-155-225.pools.spcsdns.net) |
| 23:18.15 | redax | wow. DumpChan app |
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| 23:19.14 | leifmadsen | so old :) |
| 23:19.44 | redax | yah. found in a mailinglist from 2007 |
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| 23:22.56 | *** join/#asterisk fofware (fabian@190.225.12.163) |
| 23:23.49 | redax | hm. no info about the connected channelpair |
| 23:30.23 | redax | anyway why does membermacro called with the member's channelpair and not with the calling pair? |
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| 23:39.44 | redax | with a small app_queue.c patch I can have `ANSWERER' channelvar |
| 23:40.15 | redax | where I have the connecting channelname |
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| 23:51.24 | redax | q |
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| 23:51.46 | TehRabbitt | p3nguin: ya here? |
| 23:57.33 | p3nguin | I suppose I could be, if you've got something interesting to tell me. |