| 00:00.56 | ANurmi | the only files in the folder were for a new message that was msg0016.txt I have also attempted to simply rename some of the existing to match the scheme, to see if i could delete it then but to no avail |
| 00:01.30 | p3nguin | msg0016.txt is the only file? |
| 00:01.54 | ANurmi | no there are all the regular files for msg0016.wav .gsm |
| 00:02.57 | p3nguin | Is msg0016 the only one? Are there any other files? |
| 00:03.39 | ANurmi | yeah |
| 00:04.00 | ANurmi | for some reason msg0016.wav and msg0016.WAV |
| 00:04.12 | p3nguin | Those are two different formats. |
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| 00:08.58 | p3nguin | If the msg0016 files are the only ones in the directory, what I personally would do is mmv "msg0016.*" msg0000.#1 and then reload the voicemail module. |
| 00:09.31 | p3nguin | That should turn them all into msg0000 files. Then I'd call my voicemail to check my message. |
| 00:10.51 | ANurmi | I will give that a shot, thanks for your help. |
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| 00:27.21 | p3nguin | anurmi: Let me know if my idea works out. |
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| 01:37.11 | Dryanta | and it director dude told me for all my hard work and his perfect performance review that i could have a class taught just for me as long as i could get 14 friends to sign up for html/unix, ccna class/cisco lab, autocad/high end workstations and plotter |
| 01:38.47 | Dryanta | i picked cisco because i already knew at age 14 i wanted to be an internetworking engineer and systems/software engineering held no interest |
| 01:40.07 | Dryanta | what i DID NOT know is one of the kids i got to sign up in exchange for tutoring him in ap calc had a dad that was a telco engineer at qwest, and when he got the letter with his class registration pulled strings with his boss to get us REAL carrier equipment in our lab |
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| 02:42.34 | *** join/#asterisk Andrew__M (62712e05@gateway/web/freenode/ip.98.113.46.5) |
| 02:44.33 | Andrew__M | How do I make Comedian Mail send email notifications to my ISP? My ISP SMTP requires port 80 and username + password. |
| 02:46.13 | ectospasm | Andrew__M: get sendmail (man 1 sendmail) working, then Asterisk should work just fine |
| 02:46.23 | Andrew__M | ComedianMail uses sendmail. I either need to bypass sendmail, or somehow configure it. |
| 02:46.26 | ectospasm | (or whatever your MTA is) |
| 02:46.42 | ectospasm | Andrew__M: no, you need to configure it for SMTP authentication |
| 02:46.52 | ectospasm | Which is outside the scope of #asterisk |
| 02:47.23 | Andrew__M | ectospasm: Is there a way to bypass sendmail? It seems complex. |
| 02:47.39 | pushpop | Andrew__M, follow this how to |
| 02:47.40 | pushpop | http://www.xm5design.com/?p=219 |
| 02:47.45 | pushpop | if you have a gmail account |
| 02:48.15 | ectospasm | Andrew__M: I mean sendmail the console utility (provided by your chosen MTA), not THE SendMail service |
| 02:48.56 | Andrew__M | What is an MTA? |
| 02:49.39 | Andrew__M | ectospasm: I can probably make the changes on the link you sent me for my provider. |
| 02:51.29 | ectospasm | Andrew__M: MTA==Mail Transport Agent |
| 02:51.46 | ectospasm | Andrew__M: i.e., Postfix, exim4, SendMail, qmail, etc. |
| 02:52.08 | ectospasm | Andrew__M: if you can send a regular e-mail using the "mail..." command from the shell, then that's set up correctly |
| 02:52.09 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:52.17 | ectospasm | You just need to tell Asterisk the command to use |
| 02:53.09 | Andrew__M | ectospasm: I have sendmail installed, but after a full day of reading different websites, it seems as complex as Asterisk itself. |
| 02:53.37 | Andrew__M | ectospasm: is "mail |
| 02:53.39 | ectospasm | Andrew__M: which is why you should go with something simpler, like Postfix or exim4 |
| 02:54.41 | Andrew__M | ectospasm: The funny thing is: I have all the ports, credentioals, etc, and have no way of sending a message after reading and tweaking for a whole day. |
| 02:55.02 | ectospasm | Andrew__M: yeah, don't use SendMail proper |
| 02:55.25 | ectospasm | don't use QMail either, the author and maintainer isn't interested in a userbase (or wasn't last I'd checked) |
| 02:55.44 | Andrew__M | ectospasm: I guess I basically need a client, like Mozilla, except CLI. |
| 02:55.45 | ectospasm | I have the most experience with Postfix, it's the most straightforward to setup |
| 02:55.52 | ectospasm | Andrew__M: bingo |
| 02:55.59 | ectospasm | Andrew__M: and an MTA is how to do it |
| 02:56.33 | Andrew__M | ectospasm: So postfix is an MTA you would recommend for simplicity...? |
| 02:57.26 | ectospasm | yes. I hear exim is just as simple, I just have no experience with it |
| 02:57.45 | ectospasm | you may want to google for "postfix gmail" for hints on how to get it working |
| 02:59.54 | Andrew__M | ectospasm: I have a godaddy account, but gmail must be similar. Only my port 25 is blocked by FIOS. |
| 03:00.37 | ectospasm | what does godaddy have to do with this? |
| 03:01.05 | ectospasm | what port do you have to connect to your ISP's SMTP service? |
| 03:05.46 | Andrew__M | ectospasm: My email is hosted by godaddy. I need ComedianMail to send through godaddy. I can connect with port 80, or preferrably 465 with SSL/TLS. |
| 03:07.23 | ectospasm | Andrew__M: so set your MTA to send mail through Godaddy's SMTP service (SSL/TLS) on port 465 |
| 03:08.17 | Andrew__M | ectospasm: Yes, that was the goal of today. I guess I got fooled, being 4/1. |
| 03:09.58 | ectospasm | don't ever give up! |
| 03:10.50 | ectospasm | actually, if you follow the postfix guides on getting SSL/TLS SMTP working with GMail, only replacing the gmail port, username, hostname, and certs with Godaddy's junk, it should... just... work... |
| 03:11.21 | ectospasm | I figure getting the Godaddy certs would be the most nontrivial thing about that. |
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| 03:12.57 | gpled | has anyone been able to use google.com/voice with asterisk? |
| 03:13.27 | ectospasm | I haven't tried |
| 03:14.17 | ectospasm | Andrew__M: it looks like Godaddy self-signed their cert (which is of course OK 'cuz they're a recognized CA) |
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| 03:16.29 | Andrew__M | ectospasm: Yes, i will try my luck with postfix. Thanks a lot! |
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| 04:51.35 | p3nguin | Always a day late and a dollar short. I could have had andrew__m emailing voicemail notifications in under 5 minutes. |
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| 05:11.39 | daxt | guys i have to build an IVR on asterisk and struggling with it due to having no documentation , can somebody point me the right direction ? |
| 05:12.15 | p3nguin | daxt: What's the problem you're encountering? |
| 05:12.43 | daxt | hi p3nguin i have no clue of how to build an ivr |
| 05:13.00 | p3nguin | daxt: In your own words, what is an IVR? |
| 05:13.12 | daxt | Interactive Voice Response |
| 05:13.12 | kaldemar | ~newbook |
| 05:13.12 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
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| 05:13.30 | kaldemar | daxt: read on dialplans. |
| 05:13.52 | p3nguin | For an actual IVR, you're going to need more than dial plan. |
| 05:13.52 | daxt | kaldemar in that book ? |
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| 05:34.32 | benngard | hmm. i am looking for a link, that descriebes hoto emulate/simulate the "R" button of a dect phone in the dialplane, but i cant remember where i saw it |
| 05:35.01 | benngard | and i cant remember how a googled it when i found it :( |
| 05:37.35 | ChannelZ | What does that button do? |
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| 05:47.05 | drdru | anyone ever done voice activity detection? |
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| 07:06.48 | benngard | ChannelZ: u answer n the dect, pressing [R] means u put caller on hold, dial a number, either just hang up an do a transfer or talk to the guy y called, verify that he accept the call hang up and do the transfer |
| 07:07.30 | ChannelZ | that's an attended transfer |
| 07:07.46 | kaldemar | benngard: there's really nothing special needed in the dialplan for that. |
| 07:08.10 | ChannelZ | Should be a function of the device if it's SIP for instance, or a # code in features.conf for dumb phones |
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| 07:59.02 | jkroon | kaldemar, benngard ChannelZ - some of the phones send the [R] button as a "flash" event via sip. |
| 07:59.08 | jkroon | asterisk doesn't know how to handle that. |
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| 08:05.49 | kaldemar | via SIP or via RTP? |
| 08:08.36 | kaldemar | rfc 4733 deprecated hook flash via RTP, which was defined in rfc 2833. |
| 08:11.26 | kaldemar | doing it in a SIP INFO message was defined in a draft that expired in 2004. |
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| 09:06.18 | benngard | yay BLINDTRANSFER over OOH323 to my cell phone worked perfect |
| 09:06.27 | benngard | sorry, wrióng window |
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| 10:02.13 | voila | i have put my php agi in /var/lib/asterisk/agi-bin , but still i m getting . no susch a file or directory .. |
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| 11:24.57 | antoasla | hello i have a small problem with my asterisk configuration. I want to create a network with a pc that runs asterisk server and 2 other pc that are connected to the server. Should i post the sip.conf and extensions.conf and tell me what have i done wrong plz? |
| 11:25.43 | kaldemar | what problems are you experiencing? |
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| 11:27.31 | antoasla | the pc that runs asterisk has a softphone named twinkle and i managed to connect it to the asterisk. But when i try to register a softphone in my laptop (windows xp) i cant |
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| 11:30.13 | kaldemar | connect to CLI with asterisk -vvvr to find a reason why. |
| 11:31.12 | antoasla | it says "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" |
| 11:38.22 | antoasla | so is there a problem with my asterisk? |
| 11:45.00 | antoasla | here are the sip.conf and extensions.conf http://pastebin.ubuntu.com/588587/ , they are quite simple but i cant figure why i cant make it work |
| 11:52.32 | antoasla | can anyone plz try to connect to my asterisk (192.168.1.7) user tel2 pass tel2? plz i need some help |
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| 12:21.28 | voila | hello |
| 12:22.13 | voila | well ... like in internet we have unique ip address , what is unique in mobile network ??? caller id ??? |
| 12:29.18 | voila | guys . want to extract caller id of the user using phpagi , hwo can i do it |
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| 12:35.58 | voila | guys . want to extract caller id of the user using phpagi , hwo can i do it |
| 12:39.32 | voila | will " $agi['callerid'] " , will thsi give me caller id of the client ?? using php agi |
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| 12:52.49 | voila | any one here |
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| 13:11.58 | voila | can i extract callerid of softphone using " $agi[callerid] " in php agi ??? |
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| 13:31.59 | voila | i want to get the callerid of the client using php agi , how can i do it ??? |
| 13:32.03 | voila | anybody here |
| 13:39.06 | jkroon | voila, why not just test it? |
| 13:39.27 | jkroon | and yea, for the entire pstn the caller id should be unique. |
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| 13:46.48 | voila | jkroon: there ? |
| 13:47.42 | voila | jkroon: point is ,i m using phpagi2.14 , i didnt' get anything using "$agi[callerid]" |
| 13:48.49 | jkroon | voila, i'm not a phpagi user. |
| 13:48.58 | voila | oh okies |
| 13:49.01 | jkroon | can you do a print_r($agi) and get the output somewhere? |
| 13:52.04 | voila | jkroon: onething more , i m doing $agi = new AGI() .. then $agi[calllerid] .. |
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| 14:23.07 | voila | syntax to record a file ??? $agi->record_file("record",".wav",3000); ... is it correct ?? |
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| 16:23.44 | kaldemar | voila: http://phpagi.sourceforge.net/phpagi2/docs/ |
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| 17:12.48 | koffel | i have a question on this if i add custom ext to ext it should load custom ones? right? |
| 17:15.05 | kaldemar | koffel: what are you talking about? |
| 17:15.47 | koffel | extensions |
| 17:15.52 | koffel | in asterisk |
| 17:16.59 | kaldemar | if you add something to extensions.conf and make asterisk read it... it will read it. |
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| 17:49.38 | arnotixe | hi all I'm having a difficulty with the -CHANUNAVAIL suffix. When used with a fixed number, say "exten => 100-CHANUNAVAIL", it behaves ok, but when used like "exten => _85.-CHANUNAVAIL" then dialplan sorting is weird, see http://pastebin.com/CGStYxPh for "dialplan show" output... |
| 17:50.01 | arnotixe | maybe the -CHANUNAVAIL suffix is no good with wildcard extensions? |
| 17:54.23 | arnotixe | this one http://pastebin.com/AfnfKiRt has the extensions.conf sections listed, too |
| 17:59.07 | *** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net) |
| 18:00.09 | Nivex | I'm trying to get GTalk working with Asterisk 1.8. With jabber debug on, I see an incoming call arrive, but asterisk doesn't seem to be doing anything with it. |
| 18:01.34 | *** part/#asterisk doolph (~doolph@unaffiliated/doolph) |
| 18:08.15 | p3nguin | Paste your confs in the pastebin. |
| 18:14.09 | Nivex | http://dpaste.org/8AuP/ |
| 18:15.22 | p3nguin | Paste your phones-kj context from extensions.conf. |
| 18:16.21 | Nivex | http://dpaste.org/ZwQZ/ |
| 18:17.09 | p3nguin | What is the value of ${KJDEVS}? |
| 18:17.43 | Nivex | KJDEVS=SIP/nlata1&SIP/nlata2&SIP/cisco&SIP/nivex&IAX2/nivex |
| 18:17.51 | p3nguin | also, "exten => nivex6@gmail.com, 1, Dial(${KJDEVS})" won't work with those spaces present. |
| 18:18.42 | p3nguin | Is at least one of those five phones online? |
| 18:18.56 | Nivex | yep |
| 18:19.14 | Nivex | I use that context for routing other calls, hence the 's' line there |
| 18:19.39 | p3nguin | Calls from google will go to the s extension. |
| 18:21.34 | Nivex | I've got this sinking feeling there's some sort of disconnect between jabber.conf and gtalk.conf |
| 18:24.51 | p3nguin | I'll show you my google voice dial plan. Maybe it'll be of some use to you. |
| 18:26.29 | p3nguin | http://pastebin.com/wbpuMZqe |
| 18:27.14 | p3nguin | The Goto() takes it to a phone, if that wasn't obvious. |
| 18:28.16 | Nivex | problem is, I'm seeing the incoming XML frame on the Jabber debug, but I'm not seeing any dialplan execution at all |
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| 18:52.37 | CatLynx | having issues with extention"h" |
| 18:53.38 | p3nguin | What issues? |
| 18:53.44 | CatLynx | I am trying to make it play a wav file to the person who recived the call |
| 18:53.53 | p3nguin | That's not going to happen. |
| 18:54.00 | CatLynx | arg |
| 18:54.09 | CatLynx | any ideas how to do that? |
| 18:54.12 | p3nguin | h is the hangup extension. It runs after the call has ended. |
| 18:54.48 | CatLynx | if the person who made the call gets hanged up by the other end it works fine |
| 18:54.51 | p3nguin | I'm not sure how to go about that. |
| 18:55.09 | CatLynx | thanks |
| 19:00.34 | voila | hi .. is there any way to find out , where the problem in agi php script ??? some type of verbosity |
| 19:00.35 | voila | ?? |
| 19:12.50 | Nivex | sweet! got it working! |
| 19:13.01 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 19:15.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 19:16.26 | Nivex | p3nguin: missed you in the pingout. got it working. |
| 19:28.26 | Nivex | has anyone here had a gtalk configuration with two different gtalk accounts routing calls to seperate paths? |
| 19:34.21 | *** join/#asterisk drdru (~Adium@76.77.182.145) |
| 19:34.29 | drdru | does asterisk support VAD? |
| 19:38.43 | p3nguin | Uh, what? |
| 19:38.53 | drdru | voice activity detection |
| 19:39.14 | p3nguin | silence suppression? no. |
| 19:39.21 | drdru | dang |
| 19:39.23 | p3nguin | Not unless it was added in 1.8. |
| 19:39.33 | p3nguin | You could check the new features for 1.8. |
| 19:39.36 | drdru | so we're always shipping bits even if nobody's talking? |
| 19:40.15 | p3nguin | I think silence suppression does just the opposite. |
| 19:40.52 | p3nguin | When no one is talking, the media stream goes silent. Silence suppression adds comfort noise to make it seem like the line is still "open." |
| 19:41.51 | p3nguin | Prior to 1.8, Asterisk did not support silence suppression/comfort noise. I don't know if 1.8 has it or not because I don't yet run my own 1.8 system and I have never looked for it in the new features. |
| 19:42.59 | jkroon | two different things. |
| 19:43.20 | jkroon | asterisk will never support VAD as I understand it, that is the work of an end-point device. |
| 19:43.37 | jkroon | CNG (comfort noise generation) is a different matter though. |
| 19:44.00 | jkroon | then again, VAD on dahdi/console channels might be something to look at. |
| 19:44.25 | drdru | sometimes the end point device isn't smart enough to do VAD |
| 19:44.42 | p3nguin | When running an end point with VAD enabled, Asterisk does gripe about not supporting comfort noise, so while they may be different, there's certainly a relationship. |
| 19:44.50 | jkroon | albeit, my experience with VAD/CNG was that even directly between two SIP phones it's better to just switch it off, causes really nasty audio artifacts in general. |
| 19:45.22 | drdru | know where I might be able to find a VAD algorithm? |
| 19:45.28 | drdru | I don't need it for asterisk |
| 19:45.42 | drdru | I'm trying to automate voice recording |
| 19:45.54 | jkroon | p3nguin, definitely there is a relationship. but asterisk could (theoritcally I think) just pass the CNG parameters on to the next hop in the signalling path anyway. |
| 19:47.00 | jkroon | vad itself just stops sending rtp data, cng on the other hand sends noise parameters from the source to the destination so that the recipient can reproduce some whitenoise to hide the vad artifacts. |
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| 20:10.36 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 20:12.26 | Schreiber1337 | Anyone else have issues where a caller on 1.8.3.2 server doesn't hear rings on outgoing calls? |
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| 20:14.39 | p3nguin | schreiber1337: What channel tech is your phone using, and which channel tech is used for the outgoing leg (between Asterisk and the phone company or ITSP)? |
| 20:16.30 | Schreiber1337 | P3nguin: The T1 with Verizon is :switchtype=national signalling=em_w is that what you are asking? |
| 20:17.34 | p3nguin | That answers the second half. |
| 20:17.49 | Schreiber1337 | P3nguin: Phones are using SIP w/ G711u codec |
| 20:18.02 | p3nguin | Okay, so SIP and PRI. |
| 20:18.47 | Schreiber1337 | P3nguin: The T uses all 24 channels as BChannels... I think they refer to it locally as DTWink |
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| 20:53.23 | Schreiber1337 | P3nguin: Did you have any ideas what's going wrong? |
| 20:55.35 | p3nguin | schreiber1337: Not sure... When you place a call from a SIP phone, the ringing sound is produced by the other end of the call. In your case, the T1 probably isn't going to provide the ringing, so it would have to be produced by Dahdi. I don't use Dahdi and I don't have any analog devices, so I'm not totally sure what setting you need to use to make it ring. |
| 20:56.21 | Schreiber1337 | P3nguin: OK... thanks... Anyone else have any input? |
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| 21:37.33 | minaguib | Hey folks. I'm eyeing a Yealink phone as a main house phone. No need for fancy business stuff. Just a large easy to read display and good audio quality. YAY/NAY/other recommendations ? |
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| 21:52.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:53.47 | m0e | is it normal for sip show peers to show the extension's private ip if hte extension is behind nat? |
| 21:56.51 | p3nguin | m0e: sip show peers doesn't show extensions at all, it shows sip peers. |
| 21:57.06 | m0e | it shows my extensions though *shrugs* |
| 21:57.13 | p3nguin | If you want to see extensions, you'll be using dialplan show. |
| 21:57.37 | p3nguin | sip show peers does *not* show extension information. Period. |
| 21:58.33 | m0e | well.. my end shows name/username host dyn ... |
| 21:58.41 | p3nguin | If you were so inclined to name your sip devices with numbers that correspond with the extension number used to dial such sip devices, that might be why you think it shows extension information. |
| 21:59.07 | m0e | aaah.. that could be it :$ |
| 21:59.10 | p3nguin | name/username, correct. |
| 21:59.38 | m0e | well.. in that case.. it still shows with the extension's private IP.. is that correct? |
| 21:59.45 | p3nguin | So your question was if sip show peers shows the private IP address of the peer when it is behind a NAT on a remote network? |
| 21:59.53 | m0e | exactly |
| 21:59.59 | p3nguin | Extensions do not have IP addresses. |
| 22:00.07 | p3nguin | I just went over this. |
| 22:00.16 | p3nguin | Please pay attention next time. |
| 22:00.45 | m0e | I read what you wrote.. its just that, it shows IP addresses here.. anyways.. assuming we are talking peers |
| 22:00.54 | m0e | a peer that is behind NAT on a remote network |
| 22:01.01 | p3nguin | sip show peers should not show the private IP address of a phone on a remote network. |
| 22:01.20 | p3nguin | It should show the public IP address of that device. |
| 22:01.51 | m0e | thanks for the info.. I need to dig in to find out what's the problem then.. thanks.. just wanted to make sure this was odd behavior before going around in circles |
| 22:02.03 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:02.37 | p3nguin | Check your NAT settings for your own Asterisk system if it is behind NAT as well as the NAT setting for the peer. |
| 22:03.14 | m0e | Asterisk is on a public IP.. I might need to check the other end.. |
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| 22:21.50 | m0e | defining a stun server on the NATed side seems to have fixed it.. though I thought it was only the asterisk server that needed to worry about NAT |
| 22:25.56 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 22:27.01 | p3nguin | Under normal circumstances, you need only configure your peer for NAT and Asterisk will take care of the rest. |
| 22:29.03 | m0e | hmm.. ok.. quick question (thought this might be a bit of a networking one.. not sure how this is handled by asterisk/sip in general).. If i use iptable's redirect.. though its listening on the public IP.. is this considered NAT (as it really is NAT) |
| 22:29.30 | m0e | redirect as in.. port redirection.. to be able to listen to port 12345 for instance |
| 22:29.48 | m0e | and then redirect it to 5060 to keep the "defaults" |
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| 22:43.00 | juliocesarlhg | help with mgcp gateway |
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