| 00:03.45 | *** join/#asterisk i_am_good (45727b73@gateway/web/freenode/ip.69.114.123.115) |
| 00:06.12 | i_am_good | Can you guys take a look at my extensions.conf real quick please? My call are congested |
| 00:06.14 | i_am_good | https://gist.github.com/anonymous/6736900 |
| 00:06.26 | i_am_good | What am I doing wrong there? |
| 00:12.02 | phix | i_am_good: ok, let me look. |
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| 00:16.30 | i_am_good | thanks |
| 00:16.41 | i_am_good | problem is different now |
| 00:16.41 | i_am_good | https://gist.github.com/anonymous/6736963 |
| 00:16.52 | i_am_good | I hear single tick and hangup |
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| 00:29.38 | raden | Anyone have any idea why when my phone registers its saying the IP of the router at the Asterisk server not the one at my house ? |
| 00:29.44 | raden | im very baffled to say the least |
| 00:31.07 | navaismo | maybe your router is masquerading the ip |
| 00:39.57 | raden | it should not be !!!!!! |
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| 01:08.56 | Penguin | Does it work okay? |
| 01:15.45 | raden | no |
| 01:15.50 | raden | i think its vitelity |
| 01:15.57 | raden | i cannot get it to work via registration |
| 01:16.05 | raden | but if i route DID via IP it works |
| 01:16.20 | raden | and the IP im registering with is the one that the box uses |
| 01:16.26 | raden | so there is no mix up I just dont get it |
| 01:17.05 | Penguin | Sounds like a nat setting to me. |
| 01:17.32 | raden | why can i route the DID from vitelity to my IP and it work |
| 01:17.40 | raden | and then when I do it via register it dont ? |
| 01:18.00 | Penguin | Vitelity is not behind a NAT, so set the vitel peer in asterisk to nat=no. |
| 01:19.18 | raden | let me just add this has been up 2 years until today when charter fucked our provisioning on our modems |
| 01:19.45 | Penguin | What could the provisioning have to do with anything? |
| 01:20.27 | Penguin | Is the modem operating in the proper bridge mode, or did it go into a NAT mode when it reset? |
| 01:21.04 | raden | i can ssh to all our other stuff just fine |
| 01:21.06 | raden | RDP in etc... |
| 01:21.10 | raden | all the networking seems fine |
| 01:21.19 | raden | but this did not come backup |
| 01:23.10 | raden | wtf now i cannot ssh in :( |
| 01:23.13 | raden | UGH !!!!!!!!!!!!!!!!!!!!!!!!!! |
| 01:23.15 | raden | ill be back |
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| 02:10.09 | ChannelZ | mofo damnit wth shit! |
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| 03:23.24 | redotis | How do you configure voicemail to email on asterisk? |
| 03:24.00 | Penguin | See voicemail.conf |
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| 03:24.35 | redotis | Do you have to have sendmail or another email service installed or does asterisk deliver directly? |
| 03:25.11 | Penguin | You'll need some type of service to send out the mails. |
| 03:25.20 | Penguin | I use ssmtp. |
| 03:26.01 | redotis | I can set an spf record in dns so it is valid to send for my domain. Is it easier that way? |
| 03:26.14 | ChannelZ | SPF is unrelated |
| 03:26.54 | ChannelZ | In terms of asterisk sending mail that is |
| 03:27.19 | redotis | So I must have it login to another account to send? |
| 03:27.26 | Penguin | msmtp would be another choice. I used to use it before I changed to ssmtp. |
| 03:27.51 | Penguin | Feel free to set up a basic relay using postfix if you prefer that. |
| 03:27.55 | ChannelZ | It basically runs a program. That program has to do something with the mail, whether its forwarding it on to a smarterhost or sending it directly |
| 03:28.08 | redotis | Ok |
| 03:28.09 | redotis | thanks |
| 03:28.59 | ChannelZ | msmtp is a sort of "standalone" dumb mailer that will SMTP the message elsewhere, without having to have an entire mail system running on your asterisk machine |
| 03:29.54 | redotis | Anyone know of a good example based tutorial for this somewhere. I've looked around but mostly just see examples for configure it to auth to a existing account |
| 03:30.15 | Penguin | I used it before, and it worked just fine. It's real easy to set up. I'd recommend it. |
| 03:32.15 | ChannelZ | http://pastebin.com/iQpgA8nX |
| 03:33.34 | Penguin | That'll use another relay, though. |
| 03:33.37 | ChannelZ | you can actually do it all from the commandline without the msmtprc file but I also use it for other things than asterisk so it's easier to just set there |
| 03:34.10 | ChannelZ | yes |
| 03:34.12 | redotis | thanks guys |
| 03:34.18 | redotis | oh wait |
| 03:34.23 | redotis | this example uses another relay? |
| 03:34.48 | Penguin | Your ISP's, for example. |
| 03:35.00 | ChannelZ | Yes. It doesn't have a queue or anything, it's not a mail server. |
| 03:35.09 | redotis | ok thanks |
| 03:36.38 | ChannelZ | you can setup options for auth if you need that. The man page is pretty self-explanatory |
| 03:40.54 | ChannelZ | Hmm. I should go find some ice cream |
| 03:42.08 | Penguin | Blue Bunny Premium |
| 03:42.17 | Penguin | chocolate AND vanilla |
| 03:47.22 | ChannelZ | Their pistacio almond is good |
| 03:48.04 | ChannelZ | (well maybe not w/almond, I'm thinking of baskin robins') |
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| 03:49.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 04:50.31 | redotis | What do I have to set in voicemail.conf to tell it to send emails? |
| 04:50.50 | redotis | to send the voicemail as an attachment and just to send in general? |
| 04:50.52 | WIMPy | The address of the reciepient. |
| 04:52.16 | Penguin | Look at the sample section where they specify email addresses for mailboxes. |
| 04:52.29 | redotis | 603 => 603,Michael Hamsworth,michaelham@gmail.com |
| 04:52.30 | ChannelZ | dejavu |
| 04:53.07 | redotis | So that's it. I don't need the mailcmd if it used the default mail program? |
| 04:53.44 | redotis | For instance, I tested postfix from the command line and it is sending. |
| 04:53.48 | Penguin | By default, it's going to use sendmail. |
| 04:54.09 | redotis | Hmm...sendmail is not installed. |
| 04:54.14 | Penguin | Some smtp services will have links to the sendmail name. |
| 04:54.19 | redotis | ok |
| 04:54.40 | Penguin | # ls -l /usr/sbin/sendmail |
| 04:54.40 | Penguin | lrwxrwxrwx 1 root root 5 Apr 1 2010 /usr/sbin/sendmail -> ssmtp |
| 04:54.44 | Penguin | (for example) |
| 04:55.04 | Penguin | /usr/sbin/sendmail is owned by ssmtp 2.64-2 |
| 04:55.39 | Penguin | I think when I used msmtp, I had to create the symlink myself if I didn't want to specify a mailcmd pointing to the real msmtp. |
| 04:55.52 | Penguin | It's been a while. |
| 04:56.11 | redotis | so as soon as you put the email address in the line as above, it tries to deliver when a new message comes in? |
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| 04:56.27 | redotis | after a core reload? |
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| 04:56.54 | redotis | there is no enable_email_delivery=yes kind of thing? |
| 04:56.57 | Penguin | I've never done "core reload" ever, so I don't know what will happen in that case. |
| 04:58.43 | Penguin | If you've got an emailsubject and an emailbody, specifying the email address for a mailbox should be good enough. |
| 04:59.02 | redotis | those go in the general section? |
| 04:59.19 | Penguin | yes |
| 04:59.31 | Penguin | See the sample voicemail.conf |
| 05:00.27 | WIMPy | You don't need them. They have defaults. |
| 05:00.45 | Penguin | That's even less work, then. |
| 05:01.02 | Penguin | I thought you had to specify something (anything) for them. |
| 05:01.18 | WIMPy | I haven't and it works. |
| 05:01.42 | WIMPy | Or at least it used to work. Not using it ATM. |
| 05:02.02 | Penguin | It seems like the sample has string values defined. In many cases, where default values exist, the setting will be commented out. |
| 05:05.21 | redotis | Thanks guys |
| 05:05.27 | redotis | It's working! |
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| 09:09.45 | kemmler | Is there any way to get the asterisk manager interface to give a cause code on an error like this "create_addr: No such host: 23" |
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| 18:26.41 | i_am_good | How can I convert my mp3 sound files to gsm without making them sound like a person who is coughing? |
| 18:26.55 | i_am_good | Sorry wav |
| 18:27.05 | i_am_good | I am converting them with these settings |
| 18:27.06 | i_am_good | for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm" resample -ql; done |
| 18:27.56 | [TK]D-Fender | Why would you be converting to wav? |
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| 18:28.37 | i_am_good | To gms |
| 18:28.39 | i_am_good | gsm |
| 18:29.16 | [TK]D-Fender | First... what kind of "wav"? |
| 18:30.34 | i_am_good | Um, I converted them with mpg123 from mp3 to wav, not sure which wav |
| 18:30.38 | i_am_good | for a in *.mp3; do mpg123 -w "`echo $a|sed -e s/mp3//`wav" "$a"; done |
| 18:31.10 | [TK]D-Fender | You are doing a 2 step conversion which is wasteful. |
| 18:31.21 | [TK]D-Fender | And "not sure is not good. |
| 18:32.08 | i_am_good | you're right, but what's the setting for gsm that will make it sound natural? |
| 18:32.20 | [TK]D-Fender | Firs GSM is not great to begin with. |
| 18:32.41 | [TK]D-Fender | What are your calls going to be actually using BTW? |
| 18:32.52 | i_am_good | ulaw alaw |
| 18:33.05 | [TK]D-Fender | Then you are making a tragic mistake converting them to GSM |
| 18:33.16 | [TK]D-Fender | Yuo are making them sound like shit for no good reason |
| 18:33.23 | i_am_good | That's right |
| 18:33.24 | [TK]D-Fender | And still leaving a transcoding load on * |
| 18:34.07 | [TK]D-Fender | The process you have been using is the audio equivalent to making a photocopy of a photocopy of a photocopy |
| 18:34.22 | i_am_good | Ugh |
| 18:34.30 | [TK]D-Fender | And then that last one... still gets photocpied back one last time "live" |
| 18:36.26 | i_am_good | I converted all gsm's to ulaw and now it sound OK |
| 18:36.44 | [TK]D-Fender | What GSM's? |
| 18:36.50 | i_am_good | I mean mp3 |
| 18:36.52 | i_am_good | sorry |
| 18:37.11 | i_am_good | replaced gsm files with ulaw |
| 18:37.26 | i_am_good | thanks for the explanation |
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| 18:45.06 | Blashyrkh | how do i show running calls in the asterisk commandline |
| 18:45.26 | Blashyrkh | found it |
| 18:45.37 | Blashyrkh | didnt see that it was show channels |
| 18:46.10 | Blashyrkh | is there a way to see how long a call is going? |
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| 18:50.13 | Blashyrkh | Is there a way to see how long a call is already going in the asterisk cli? |
| 18:50.43 | ChannelZ | core show channel xxxxx |
| 18:50.55 | ChannelZ | xxxxx can be gotten from 'core show channels concise' |
| 18:52.22 | Blashyrkh | thanks |
| 18:52.33 | Blashyrkh | i tried the same with dahdi show channel number |
| 18:52.45 | Blashyrkh | but couldnt find the elapsed time |
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| 19:44.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 20:14.34 | foamz | Does anyone have hardware spec recommendations for handling around 200 concurrent VoIP-SIP calls (which are ultimately sent to a VoIP-SIP -> GSM gateway) in G.711 codec? |
| 20:15.24 | foamz | i was looking at this page |
| 20:15.25 | foamz | http://totalticketsystem.com/blog/technical-articles/what-server-should-i-use-for-my-asterisk-pbx-how-to-size-your-asterisk-server/ |
| 20:15.34 | foamz | and they say 4xQuad Core Xeons |
| 20:15.40 | foamz | which seems like overkill.... |
| 20:16.56 | [TK]D-Fender | it is |
| 20:19.28 | foamz | yeah |
| 20:19.28 | foamz | i thought so |
| 20:19.28 | foamz | a single quad core should be fine right? |
| 20:19.28 | [TK]D-Fender | sure |
| 20:21.27 | foamz | ty |
| 20:21.28 | foamz | i got scared for a moment |
| 20:21.28 | foamz | wondering if i was about to have to invest in some sort of 4cpu monstrosity |
| 20:23.15 | foamz | anyone have hardware recommendations? |
| 20:23.24 | foamz | im building my first Asterisk box |
| 20:23.33 | foamz | total noob here |
| 20:24.52 | Tim_Toady | foamz: if there is no transcoding in your scenario, or recording, or sound mixing (conference calls managed by the asterisk box) then the cpu demands are not really high |
| 20:25.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.8) |
| 20:26.52 | Tim_Toady | its hard to estimate tho, replicate your asterisk setup and call scenario in a text box and do stress test on it (using sipp maybe) to get an estimate of how much power u will really need |
| 20:30.07 | [TK]D-Fender | Transcoding, recording, mixing, etc. |
| 20:30.44 | [TK]D-Fender | these are they key factors |
| 20:33.10 | foamz | i dont believe there is any transcoding, im just taking SIP calls from the call originator and routing them to the VoIP-SIP port of the GSM gateway |
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| 23:16.51 | cusco | "Let's take a look at what happened when the compact cassette arrived. It was sort of an analog removable hard drive with music, that you plugged into an analog music player - the new thing at the time being that you could also write to it" |
| 23:16.55 | cusco | lol |
| 23:17.06 | cusco | todays people don't know what was k7 |
| 23:17.28 | cusco | is feeling old :( |
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