| 00:00.06 | dimitry7 | it is for E1 links |
| 00:00.18 | dimitry7 | but I don't know nothing else |
| 00:01.24 | WIMPy | only knows PRI on E1. |
| 00:06.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.221) |
| 00:24.33 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 00:24.45 | *** part/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 00:28.26 | *** join/#asterisk slicknick5181 (ccc3835e@gateway/web/freenode/ip.204.195.131.94) |
| 00:28.48 | slicknick5181 | Mitel 5220 speaker phone works fine but handset with not transmit |
| 00:29.00 | slicknick5181 | I have tried new handset and cord what else could be my issue |
| 00:30.08 | WIMPy | A broken phone? |
| 00:30.40 | slicknick5181 | but the speakerphone transmits fine |
| 00:33.08 | slicknick5181 | thats the real kicker i thought maybe some sound module isnt working but the speakerphone works perfect |
| 00:37.00 | dimitry7 | called failed :O |
| 00:37.15 | dimitry7 | entering silent mode... 3.2.1 |
| 00:47.34 | igcewieling | sounds like you need to do a "make distclean" and then reinstall Asterisk |
| 00:47.44 | igcewieling | (the make distclean in the Asterisk source dir) |
| 00:48.01 | igcewieling | this will make Asteirsk pick up any new libraries you installed |
| 00:48.45 | igcewieling | Almost nobody uses R2, where are you, Mexico? |
| 00:49.15 | WIMPy | A configure should do it. |
| 00:49.27 | igcewieling | WIMPy: you'd think so, wouldn't you. |
| 00:49.35 | WIMPy | Mexico uses E1? |
| 00:49.44 | igcewieling | but it doesn't, at least until 1.4, I've not tested it using 1.8+ |
| 00:49.48 | igcewieling | WIMPy: Yeah. |
| 00:50.34 | igcewieling | WIMPy: Almost every R2 question I've seen on the mailing lsits are for MX E1s |
| 00:52.03 | igcewieling | speaking of mailing lists, dimitry7 you should search the mailing lists archive by adding site:lists.digium.com in your google search |
| 01:00.08 | igcewieling | https://supportforums.cisco.com/thread/2126393 http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a008024903b.shtml#mexico |
| 01:02.03 | slicknick5181 | Anyone here use callcentric |
| 01:02.24 | igcewieling | ~ask |
| 01:02.25 | 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. |
| 01:03.16 | slicknick5181 | I am having issues securing my system using asterisk |
| 01:03.51 | igcewieling | iptables and fail2ban. next! |
| 01:04.19 | WIMPy | Just pull the plug. Much less work and the only secure option. |
| 01:05.12 | *** part/#asterisk putney (~putney@203.190.232.150) |
| 01:06.26 | slicknick5181 | the problem is they send me calls accross two different ranges of 255 address |
| 01:07.25 | igcewieling | my recommend stands |
| 01:08.00 | slicknick5181 | do you mind elaborating or directing me to a site that will |
| 01:08.53 | slicknick5181 | WIMPy: if that was the only secure option then nobody would use this |
| 01:10.22 | WIMPy | Do you really think so? |
| 01:12.14 | igcewieling | slicknick5181: I don't see the issue. Block all access from IPs not in CallCentric's ranges using iptables. To handle requests which get through iptables with fail2ban to block the specific IPs automatically. |
| 01:12.35 | igcewieling | this is networking 101 |
| 01:13.53 | slicknick5181 | igcewieling: I am not knowlageable on iptables or fail2ban I have used an asterisk server on a private network but now I need to open it to the internet for production |
| 01:14.11 | igcewieling | slicknick5181: Google welcomes you. |
| 01:14.37 | igcewieling | If you expect to secure your server you need to learn some networking. |
| 01:16.06 | slicknick5181 | igcewieling: As I plan to do. I needed some direction as to where to look for securing asterisk as you have suggested I will look for information on iptables and fail2ban |
| 01:22.32 | *** join/#asterisk bpmedley (~bpmedley@host-67-102.arfahpl.clients.pavlovmedia.com) |
| 01:24.06 | *** join/#asterisk andrewyager (~andrewyag@1.146.77.184) |
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| 01:27.09 | *** part/#asterisk slicknick5181 (ccc3835e@gateway/web/freenode/ip.204.195.131.94) |
| 01:35.11 | WIMPy | wonders how asterisk resolves hostnames. |
| 01:36.55 | igcewieling | WIMPy: poorly. |
| 01:37.26 | *** join/#asterisk Maliuta_ (nikolai@donetsk.lusan.id.au) |
| 01:37.27 | igcewieling | I believe it uses the systems resolver library. |
| 01:38.12 | igcewieling | i.e. DNS requests are blocking. dnsmgr (dnsmgr.conf) can set up "cacjhing" for DNS requests, but it is often better to simply install bind on the server |
| 01:38.35 | WIMPy | has bind running. |
| 01:38.39 | igcewieling | We use dnsmgr AND a local caching only nameserver on our Asterisk boxes |
| 01:39.33 | WIMPy | But I have an issue with a sip peer, where sip show peer <peer> shows the correct ip under Addr->IP, but when trying to send a call there, it uses another IP. |
| 01:40.54 | igcewieling | you didn't do something stupid like enable srclookup? |
| 01:41.00 | igcewieling | srvlookup |
| 01:41.29 | igcewieling | do you specify your peer by hostname? what does a ping of that hostname from the asterisk server show? |
| 01:41.53 | WIMPy | Interesting. It even happens if I specify the IP in the host= line. |
| 01:42.03 | WIMPy | No srvlookup=no. |
| 01:43.07 | igcewieling | what is the actual peer name? |
| 01:43.29 | WIMPy | "alice-39" |
| 01:43.53 | WIMPy | The INVITE contains the correct IP (host). |
| 01:44.05 | igcewieling | where is the incorrect IP? |
| 01:44.43 | WIMPy | The INVITE header contains the IP that was configured as host=. |
| 01:44.58 | WIMPy | But the message isn't sent there. |
| 01:45.23 | WIMPy | I will try to put that host in to a local zonefile. |
| 01:45.51 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:46.21 | igcewieling | make sure your /etc/nsswitch.conf lists hosts first |
| 01:46.41 | WIMPy | In bind. |
| 01:47.02 | igcewieling | could you have an alice-39.your.domain.com |
| 01:47.14 | igcewieling | silly me, you did say zonefile |
| 01:47.41 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qcffdvndvnmozkak) |
| 01:47.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:48.45 | igcewieling | could the endpoint be replying with the wrong IP in the SDP or something like that? |
| 01:49.02 | WIMPy | It never gets there. |
| 01:49.49 | WIMPy | does a full restart. |
| 01:52.19 | WIMPy | Didn't help, either. |
| 01:52.39 | WIMPy | wonders if it's the "-" in the peer name. |
| 01:54.15 | WIMPy | No go :-( |
| 01:56.23 | WIMPy | http://wimpy.yeti.dk/pastebin |
| 01:56.35 | WIMPy | If anyone has any idea what's going on... |
| 01:59.34 | *** part/#asterisk bpmedley (~bpmedley@host-67-102.arfahpl.clients.pavlovmedia.com) |
| 01:59.47 | WIMPy | wonders if it could be the fromdomain which isn't resolvable. |
| 02:03.30 | igcewieling | can you try removing it? |
| 02:04.18 | WIMPy | Yes, no change :-( |
| 02:07.16 | igcewieling | I bet it is just the govt intercepting the data and screwing up |
| 02:07.24 | igcewieling | 8-) |
| 02:07.32 | WIMPy | In the Asterisk source? |
| 02:09.00 | igcewieling | I was joking. |
| 02:09.40 | WIMPy | http://i.imgur.com/XLQIsnH.gif |
| 02:10.43 | igcewieling | That is awesome! |
| 02:14.30 | WIMPy | So is what Asterisk is doing :-( |
| 02:22.37 | *** join/#asterisk andrewyager (~andrewyag@180-94-235-178.blue.net.au) |
| 02:23.05 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 02:23.05 | *** mode/#asterisk [+o file] by ChanServ |
| 02:29.42 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:31.42 | *** join/#asterisk andrewyager (~andrewyag@180-94-235-178.blue.net.au) |
| 02:32.18 | *** join/#asterisk dimitry7 (~antonello@189.164.154.243) |
| 02:32.22 | dimitry7 | Registration of 'iaxpacifico' rejected: 'Registration Refused' from: '189.164.154.243' |
| 02:32.26 | dimitry7 | what can be that? |
| 02:32.30 | dimitry7 | thank you guys! |
| 02:38.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.221) |
| 02:51.01 | WIMPy | Oh, the OPTIONS messages go to the correct IP. |
| 02:51.27 | WIMPy | So maybe it doesn't recognize the destination as peer name? |
| 02:52.01 | WIMPy | fsck |
| 02:52.53 | WIMPy | it was the dial string. |
| 03:03.14 | WIMPy | And finally a nice little sendrpid=no and it works. |
| 03:12.27 | igcewieling | typoe in the dial string? |
| 03:12.48 | WIMPy | No a left over option from another channeltype. |
| 03:16.17 | WIMPy | So it used the correct part of the dial string as destination, but got the host wrong. Seems rather artistic to me, tho I have to admit that I confused it. |
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| 04:24.36 | volga629 | I got extension register, but still have some issues with rtpproxy |
| 04:25.19 | volga629 | is this is normal that all extension are register with kamailio scr on local lan |
| 04:25.24 | volga629 | ? |
| 04:32.48 | volga629 | this debug of fail rtp during the call https://fpaste.networklab.ca/WWic/ |
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| 06:23.13 | hrolf | Hi #asterisk, I'm troubleshooting an issue, need your help. In SIP debug log, the To and From fields are like ext1@myIp and ext2@myIp respectively. Does that mean that both my extensions are on the same Asterisk server? (Although the Via field is different.) |
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| 07:30.18 | hrolf | Hi #asterisk, I'm troubleshooting an issue, need your help. In SIP debug log, the To and From fields are like ext1@myIp and ext2@myIp respectively. Does that mean that both my extensions are on the same Asterisk server? (Although the Via field is different.) |
| 07:30.45 | hrolf | Why is it that the To: and From: headers have the IP Addresses same even though both extensions are on different servers? |
| 07:35.25 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
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| 08:28.33 | linocisco | hi all |
| 08:28.51 | linocisco | what is the best conference phones other than polycom phone? |
| 08:29.05 | *** join/#asterisk hanuman (31ce3ed8@gateway/web/freenode/ip.49.206.62.216) |
| 08:31.15 | hanuman | logout |
| 08:31.55 | linocisco | polycom phones without display starts with 455 USD. so expensive |
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| 08:46.38 | linocisco | what is the best conference phones other than polycom phone? |
| 08:57.25 | cneb3000 | linocisco: what do you want out of your conference phone? |
| 08:57.49 | linocisco | cneb3000, cheap, good sound quality, must work well with asterisk |
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| 09:05.53 | mic_ | can someone share his/hers experiences when running asterisk as a virtual machine? |
| 09:06.02 | mic_ | I am running myself one instance in KVM - but it's a low-traffic site |
| 09:06.17 | mic_ | and zero issues to be honest. But I am not sure how it's going to look like |
| 09:06.35 | mic_ | once many concurrent connections show up. Google does have something, but I am not sure how much that info is outdated |
| 09:06.54 | mic_ | (and yes, I am aware of things like RTC fun under VMWare etc., therefore a bit of initial distrust in the subject) |
| 09:09.36 | cneb3000 | I run a production system (Ast 11) under proxmox |
| 09:10.07 | mic_ | cneb3000: it's KVM based, right? |
| 09:10.13 | cneb3000 | that particular system has about 500-550 concurrent calls 24/7. and generates CDRs. other than that, no voicemail, or 'special' features |
| 09:10.14 | cneb3000 | correct |
| 09:10.51 | cneb3000 | I don't think I've ever had a problem specifically related to the fact it's proxmox/KVM |
| 09:11.07 | mic_ | that was also more or less my idea |
| 09:12.08 | mic_ | the test guest I am running here |
| 09:12.24 | mic_ | has plenty of features enabled - voice mails, queues, call recording - and for the low-traffic - zero issues |
| 09:12.49 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:12.50 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.192) |
| 09:12.52 | cneb3000 | yea, I think a lot of people run it under a KVM :) |
| 09:13.21 | mic_ | cneb3000: thanks :) because right now we have some people |
| 09:13.33 | mic_ | that want us to run production systems under Hyper-V |
| 09:13.37 | cneb3000 | I actually think in the asterisk definitive guide, they have some instruction spefically mentioning VM |
| 09:14.01 | mic_ | o, have that book |
| 09:14.14 | cneb3000 | yea, i mean.. it mentions it in passing |
| 09:14.15 | cneb3000 | lol |
| 09:14.33 | mic_ | hehe |
| 09:14.36 | mic_ | first hit in that book |
| 09:14.50 | mic_ | "If you're installing into a virtual machine (which we don't recommend for a production use [...] |
| 09:15.11 | mic_ | but then they also claim that a lot of people have success when using it. |
| 09:15.18 | eirirs_ | and thats the first thing I ignored |
| 09:15.19 | eirirs_ | :) |
| 09:16.11 | cneb3000 | yea, but the author also mentions a few times on his twitter stuff about VMs |
| 09:16.21 | cneb3000 | conflicting info lol |
| 09:16.56 | cneb3000 | I've run asterisk in VMs since 1.4. I can't remember any issues there. I went from 1.4, skipped 1.6, 1.8 and then straight to 11. there may have been issues with other version I don't know about |
| 09:17.15 | cneb3000 | that is I used 1.8, but skipped 1.6 |
| 09:18.15 | mic_ | that confirms pretty much the little information that's available online |
| 09:18.19 | mic_ | (+KVM, +ESXi) |
| 09:18.48 | eirirs_ | I had to change some configs for it to work |
| 09:19.01 | cneb3000 | eirirs: like what? out of curiosity :) |
| 09:19.10 | mic_ | (seconding that ;) |
| 09:19.44 | eirirs_ | uhh good quesition, thats ages ago, but ill try to find |
| 09:20.40 | mic_ | cneb3000: which codecs are you using in your VM instance? |
| 09:21.28 | cneb3000 | for the most part, I use G729. note, with no transcoding. So both legs are in g729. However, I do transcode between g711 and g729. but only for like.. maybe 20-30 of those ~500 concurrent calls |
| 09:22.07 | mic_ | so it's pretty much same here - both legs are in g711 in my case |
| 09:22.21 | cneb3000 | yea I'd say that's the same :) |
| 09:24.54 | eirirs_ | hmm cant find |
| 09:27.12 | mic_ | I cannot really think of anything that should be treated in a different way |
| 09:27.29 | mic_ | with the exception of hardware cards to do ISDN/PSTN |
| 09:28.27 | cneb3000 | that's true, although 'hardware passthrough' as it's sometimes known, is normally quite well documented |
| 09:29.02 | cneb3000 | although not for asterisk, I've used hardware passthrough before on a KVM to utilise a 4 port network card without any issues |
| 09:32.43 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 09:33.41 | *** join/#asterisk Schreda (~Schreda@80-110-33-15.static.surfer.at) |
| 09:33.46 | Schreda | hello |
| 09:33.48 | Schreda | all |
| 09:33.58 | mic_ | thanks to you both for your comments |
| 09:34.04 | Schreda | I would have an question regarding asterisk 1.6 and fax |
| 09:34.11 | cneb3000 | its ok mic_ |
| 09:34.15 | cneb3000 | good luck ;) |
| 09:34.17 | mic_ | I will know after Thursday whether we are going to start testing it under Hyper-V |
| 09:34.26 | mic_ | (which I am not happy about) |
| 09:34.29 | mic_ | (Linux fan here ;( |
| 09:34.34 | cneb3000 | lol |
| 09:34.55 | Schreda | Someone experience with spanDSP? |
| 09:35.12 | cneb3000 | Schreda: Are you aware asterisk 1.6 is no longer formally supported? |
| 09:35.25 | cneb3000 | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions <<< |
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| 09:43.58 | Schreda | is there still some fax solution available for asterisk 1.6 |
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| 10:50.55 | k610 | how can i fully disable t38 udptl.conf ? |
| 10:50.56 | k610 | <PROTECTED> |
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| 11:33.27 | Busstech | Hi all! |
| 11:34.01 | Busstech | To do successful backups of asterisk for disaster recovery, which files / folders are critical? |
| 11:35.03 | Greenlight | Don't plan a backup. Plan a recovery. Then the answer will present itself. |
| 11:36.48 | Busstech | Ok, fair enough. I would just like a community dissuasion so that I can check myself |
| 11:37.33 | Busstech | 1) /etc/asterisk/cdr_mysql.conf |
| 11:37.48 | Greenlight | Well a lot depends on your own setup. Do you have a boatload if IVR recordigns that you couldn't function without for example ? |
| 11:38.02 | cneb3000 | Greenlight raises a good point, it depends on whether your recovery includes a full system restore.. some sort of failover.. or a from scratch re-install where you apply config you've saved |
| 11:38.05 | Busstech | 2) /var/lib/mysql/astreriskcdrdb/ |
| 11:38.23 | Busstech | I have failover covered. |
| 11:38.50 | Greenlight | Plan it from a recovery point of view. Grab a VM, and start recovering. What do you need? |
| 11:40.01 | Greenlight | I would imagine that your CDR's are backed up elsewhere if they're important, so would treat them separately. |
| 11:40.26 | Greenlight | So, you'll prob want most your config files. Any custom sound files. And custom scripts. |
| 11:40.58 | Greenlight | But a backup is only good once it's been tested to work - don't try and guess what you'll need. Actually do a recovery and find out. |
| 11:41.10 | Greenlight | Get a recovered system working like the production machine. |
| 11:42.21 | Greenlight | I would imagine that your recovery plan would be to rebuild the system from scratch - but maybe not. Maybe you take an image. Maybe you run a VM. It really depends. |
| 11:43.56 | Chainsaw | Saw the light, I suppose. |
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| 12:12.08 | Greenlight | Chainsaw: Or decided it was better in the dark! :) |
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| 12:22.27 | igcewieling | http://i.imgur.com/XLQIsnH.gif (from WIMPy last night) |
| 12:24.36 | WIMPy | Good morning :-) |
| 12:26.38 | bulkorok | hi |
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| 13:02.18 | WIMPy | just got a new IAD. |
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| 13:04.43 | igcewieling | I think there is a pill for that. |
| 13:05.08 | WIMPy | I used Asterisk. |
| 13:05.20 | WIMPy | I'm not even sure I want to use it. |
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| 13:40.14 | antonioX | hi all. I dont know why but when calling to some short numbers via dahdi, asterisk hung up after dahdi answered. If I dial some other number via dahdi, it works ok. |
| 13:40.23 | antonioX | how should I debug that dahdi calls? |
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| 13:41.34 | igcewieling | put the cli output of a failed call on your non-FreePBX server on a pastebin |
| 13:41.35 | igcewieling | ~pb |
| 13:41.35 | infobot | 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/; or install pastebinit with yum or aptitude. |
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| 13:47.14 | antonioX | failed call via dahdi -> http://pastebin.com/raw.php?i=vViDqJQd |
| 13:48.54 | Greenlight | That looks very freepbx like to me |
| 13:48.54 | Greenlight | Can we see the dialplan |
| 13:49.26 | antonioX | no freepbx here, just asterisk |
| 13:49.33 | Greenlight | Ok, can we see the dialplan |
| 13:49.37 | antonioX | yes of course |
| 13:49.41 | antonioX | one minute please |
| 13:51.48 | antonioX | extensions.conf: http://pastebin.com/V7Ji1Gp5 |
| 13:52.49 | igcewieling | antonioX: What type of DAHDI interface? |
| 13:53.03 | antonioX | macros.conf: http://pastebin.com/RGgdV8Jr |
| 13:53.24 | antonioX | igcewieling: Wildcard TDM400P REV I Board 5 |
| 13:53.51 | igcewieling | antonioX: FXO or FXS? |
| 13:54.07 | antonioX | FXO |
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| 13:54.38 | igcewieling | Also unless your services.conf, macros.conf, etc are supposed to be in a [global] section, you have them in the wrong place |
| 13:54.47 | igcewieling | NOTHING should come before global and general |
| 13:54.57 | Greenlight | You're dialling 1004 on the PSTN network? |
| 13:54.59 | [TK]D-Fender | Order doesn't matter for that last I checked |
| 13:55.00 | antonioX | yes |
| 13:55.03 | Greenlight | Or is this going to another box ? |
| 13:55.05 | [TK]D-Fender | And has no impact on his DAHDI issue |
| 13:55.06 | igcewieling | in fact I can't imagine how that would work in your current setting |
| 13:55.07 | antonioX | 1004 on the PSTN |
| 13:55.30 | [TK]D-Fender | antonioX: enable SIP debug so we can confirm which end is killing the call. |
| 13:55.41 | igcewieling | antonioX: I can't believe 1004 is a valid telephone number. |
| 13:55.44 | antonioX | thank you igcewieling, I'll move that includes |
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| 13:55.54 | [TK]D-Fender | antonioX: Also, is that number actually valid? |
| 13:55.57 | antonioX | ok [TK]D-Fender |
| 13:56.00 | antonioX | yep, it is :) |
| 13:56.14 | antonioX | is a valid number on the PSTN inside spain |
| 13:56.16 | igcewieling | antonioX: try adding the city code and see |
| 13:56.27 | antonioX | not valid then igcewieling |
| 13:56.43 | antonioX | it is supossed to be dialed as is: 1004 |
| 13:56.56 | antonioX | it is the information number for movistar/telefonica here in spain |
| 13:57.07 | igcewieling | ah! a SPECIAL number. |
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| 13:57.20 | Greenlight | Are you *sure* it can be accessed via ISDN ? |
| 13:57.23 | [TK]D-Fender | antonioX: one more thought, add "ww" before your number in your dial command to have * wait 1 second before sending just to ensure the first digit doesn't get cut off |
| 13:57.25 | [TK]D-Fender | ^ |
| 13:57.32 | igcewieling | I wonder if the line is doing something weird because of the premium number |
| 13:58.01 | igcewieling | like, I dunno, reversing line polarity to indicate a premuim number or something like that |
| 13:58.19 | igcewieling | Greenlight: are you in the USA? |
| 13:58.25 | WIMPy | ISDN? |
| 13:58.26 | antonioX | sip debug: http://pastebin.com/SW8LYWuE |
| 13:58.44 | Greenlight | Sorry, didn't see that this wasn't ISDN |
| 13:58.47 | Greenlight | Scratch that :) |
| 13:58.51 | Greenlight | And, no, UK. |
| 13:59.18 | antonioX | [TK]D-Fender, let me try that "ww" thing |
| 13:59.24 | Greenlight | Don't work with analogue much, have a habit of assuming digital :) |
| 13:59.25 | antonioX | btw if I connect a phone and dial 1004 then it works. |
| 13:59.29 | [TK]D-Fender | antonioX: Yup, PSTN side did indeed die off.. |
| 13:59.51 | *** part/#asterisk volga629 (~bendersky@CPE00090f1b215c-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 14:00.14 | [TK]D-Fender | [09:59]GreenlightDon't work with analogue much, have a habit of assuming digital <- perhaps the repeated statements of "analog" , "FXO", and even the specific card he was using wasn't quite enough for you :) |
| 14:00.20 | WIMPy | Greenlight: Not compatible with #asterisk. |
| 14:01.03 | antonioX | [TK]D-Fender: adding "ww" before the number made it * wait 1 sec, but then hang up the same |
| 14:01.17 | Greenlight | ? |
| 14:01.22 | antonioX | also, if I call another short number (1415) it works |
| 14:01.43 | [TK]D-Fender | antonioX: get a splitter and plug your phone in parallel and listen in as soon as you dial. Expand it to "wwww" to give yourself enough time to grab it... |
| 14:02.22 | antonioX | ok, I have to go downstairs to the basement, brb in 2 min |
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| 14:07.37 | antonioX | If I connect a phone on the splitter and listen after dial, I hear dialtone, all 4 DTMF digits and then the call works only on the phone |
| 14:07.53 | antonioX | asterisk still hung up the call |
| 14:10.13 | antonioX | I have 3 FXO interfaces on that card. Maybe should I try on another one? |
| 14:11.58 | [TK]D-Fender | antonioX: if the call continues on the phone then I'm betting that the telco is passing some signalling mistaken as a hangup request. |
| 14:12.08 | [TK]D-Fender | antonioX: Check your zone settings |
| 14:13.02 | antonioX | [TK]D-Fender: let me check that settings |
| 14:13.37 | antonioX | zones seems to be correct: |
| 14:13.38 | antonioX | loadzone = es |
| 14:13.38 | antonioX | defaultzone = es |
| 14:16.59 | antonioX | now a fun fact: I'm connecting to my provider via a router with ATA integrated, as they dont give me the SIP details to connect on my own |
| 14:17.50 | antonioX | * > TDM400P FXO > provider ATA > provider SIP server > PSTN |
| 14:19.01 | igcewieling | Well, that is it for my assistance. Stop withholding important information. |
| 14:19.26 | antonioX | sorry igcewieling :( |
| 14:19.29 | antonioX | thank you all |
| 14:20.02 | antonioX | I didnt notice that this would be important until [TK]D-Fender tell me to check the zone settings |
| 14:20.06 | igcewieling | Now I know why you didn't need and special dahdi settings for Span. because from an Astersik standpoint you are NOT IN SPAIN. |
| 14:20.16 | WIMPy | That's how "PSTN" works in Europe. |
| 14:20.38 | antonioX | but that ATA is supossed to be configured to work as the PSTN in spain |
| 14:20.50 | antonioX | maybe I should try another zones? |
| 14:21.01 | igcewieling | antonioX: I've never seen an ATA work correctly when connected to an analog port |
| 14:21.22 | [TK]D-Fender | antonioX: It COULD be using your signalling... or not. Why not give "us" a try... |
| 14:21.27 | igcewieling | they don't normally signal disconnects in a way Asterisk understands, as just one example. |
| 14:21.33 | antonioX | it works ok, except for some short numbers |
| 14:21.54 | antonioX | [TK]D-Fender: ok, trying "us"... |
| 14:22.18 | [TK]D-Fender | antonioX: I'd restart DAHDI & * completely for this of course.. |
| 14:22.45 | antonioX | yes, ofc |
| 14:22.58 | antonioX | restarted, now trying... |
| 14:23.28 | antonioX | same hung up :( |
| 14:23.56 | [TK]D-Fender | antonioX: I'd check with your provider for information on this... |
| 14:24.14 | antonioX | I'll try wwww1004wwwwwww |
| 14:25.01 | antonioX | IT WORKS!!! |
| 14:25.08 | WIMPy | antonioX: Doesn't your IAD have an S0 port you can use instead? |
| 14:25.14 | *** part/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 14:26.27 | antonioX | WIMPy: I dont have a IAD |
| 14:26.44 | WIMPy | Or better: Go to google to find out how to hack it to get the sip credentials out of it and configure them directly in Asterisk. |
| 14:27.06 | WIMPy | Didn't you say so above? |
| 14:27.33 | antonioX | no! It was you WIMPy |
| 14:27.37 | antonioX | :) |
| 14:28.15 | antonioX | WIMPy: nice idea |
| 14:28.16 | WIMPy | So what do you have there? |
| 14:28.31 | antonioX | * |
| 14:30.23 | antonioX | I already have all the credentials needed and have done tests, but still have some problems to solve before connecting via SIP to my provider |
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| 14:35.52 | antonioX | so, I "fixed" it by adding "www" after any short number call via DAHDI |
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| 14:37.18 | antonioX | good karma to igcewieling, WIMPy, Greenlight and [TK]D-Fender! thank you! <3 |
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| 14:44.33 | [TK]D-Fender | antonioX: You're welcome, though I am curious on what added wait does... |
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| 14:51.09 | Greenlight | Perhaps the added pause causes it to ignore whatever signalling is sent right after the dial completes |
| 14:51.54 | [TK]D-Fender | maybe... |
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| 15:56.11 | *** join/#asterisk Synx|hm_ (~Synx@unaffiliated/synx-hm/x-1623004) |
| 15:57.46 | Synx|hm_ | Anyone using TLS? I just ran into an issue where i tried to register 500 peers with asterisk SIP TLS and as soon as i get to registration 335 asterisk hard faults with tcptls.c:299 ast_tcptls_server_root: Accept failed: Too many open files |
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| 15:58.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:58.56 | WIMPy | Well, if you use TCP, you need one FD per connection. |
| 15:59.00 | blitzrage | Synx|hm: sounds like you don't have enough file descriptors on your system |
| 15:59.01 | WIMPy | So check your limits. |
| 15:59.07 | blitzrage | OS level issue |
| 15:59.15 | Synx|hm | k |
| 15:59.21 | WIMPy | or ulimits rather. |
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| 16:02.47 | Synx|hm | awesome thanks for the fast reponses had 1024 ulimit set ill read how to up that |
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| 16:09.49 | Cuzner | ze problem is ze kernel |
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| 16:26.20 | nilsec | Hey, is anyone familiar with an issue where downloading a voicemail/fax archive of a user causes a crash? |
| 16:27.27 | igcewieling | no. |
| 16:27.38 | igcewieling | Thankfully Asterisk doesn't have that feature. |
| 16:28.02 | igcewieling | could you be looking for #freepbx? |
| 16:30.13 | *** join/#asterisk navaismo (~navaismo@189.191.224.201) |
| 16:31.10 | nilsec | igcewieling, hmm... well working with this product in particular: http://www.digium.com/en/products/business-phone-systems/switchvox-355 |
| 16:31.26 | igcewieling | nilsec: that is not supported here. |
| 16:31.36 | igcewieling | ~switchvox |
| 16:31.37 | nilsec | I don't know if it was the cause or just a big coincidence, but right after downloading a user's voicemail and fax (before deleting extension), I got a crash. |
| 16:31.41 | nilsec | Well... alright, thanks :) |
| 16:32.01 | igcewieling | Contact Digium support for your commercial Digium product. |
| 16:32.44 | *** part/#asterisk nilsec (~nilsec__@unaffiliated/nilsec) |
| 16:35.11 | Cuzner | they're paying for support, why not use it? |
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| 17:03.18 | igcewieling | I'm starting to get sick of customers reporting ring-no-answer as a problem, when their people don't actually answer the damn calls. |
| 17:03.37 | igcewieling | goes off to change their ringtone to "Howler" |
| 17:05.38 | WIMPy | Unfortunatly hering a ringing sound no longer means the remote end is really ringing in the VOIP world. |
| 17:06.13 | igcewieling | WIMPy: " -- SIP/320-000248c5 is ringing" is good enough for me. |
| 17:06.13 | WIMPy | Either you hear someone pick up the phone or you have a general call failure. |
| 17:06.37 | WIMPy | I have hade many calls saying that without ringing. |
| 17:07.05 | igcewieling | WIMPy: this customer had us disable voicemail because their users were not answering calls. These are local polycom phones connected to a local Asterisk box. |
| 17:07.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.163) |
| 17:07.52 | WIMPy | Ok, on local phones I could believe it. |
| 17:10.19 | igcewieling | WIMPy: this is the same customer who reported "internet down!" even though I used their internet to connect to the PBX. |
| 17:10.27 | Cuzner | WIMPy: dump 'em into a reception queue or something if their call fails then...? |
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| 17:10.32 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:10.54 | Cuzner | er i guess that was for igcewieling |
| 17:10.58 | WIMPy | How do you know if a call fails? |
| 17:11.14 | igcewieling | WIMPy: when the Dial finally times out. |
| 17:11.18 | Cuzner | define "fails" |
| 17:11.20 | WIMPy | Yes, he had an example where he probably knows. |
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| 17:38.22 | blergh- | Hello, I'm looking for some general advice in regards to decent phones that have a VPN-capability, I've found the Snom 370 and it seems good, but what are the options? |
| 17:39.15 | WIMPy | Auerswald can also do OpenVPN, but I don;t know how well they play with plain SIP. |
| 17:41.01 | blergh- | Ah yeah, I'm not very good with whats new and such in regards to "real" phones, i've only been doing softphones + openvpn, but i wish to replace my cellphone with a decent and somewhat secure real physical phone |
| 17:41.32 | WIMPy | The Snom is quite nice. |
| 17:42.03 | igcewieling | you will find your selection very limited. consider using doing VPN the traditional way and let the phone do phone stuff. |
| 17:42.05 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), 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 |
| 17:42.57 | blergh- | igcewieling: it is? that's a bummer, it'd be nice to have native support for it |
| 17:43.07 | igcewieling | ~phones |
| 17:43.07 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 17:43.24 | igcewieling | I don't think any of them except SNOM have VPN on the phone. |
| 17:43.40 | WIMPy | It has |
| 17:44.50 | igcewieling | puts on a phone company uniform, grabs a clipboard and installs wiretaps on all blergh-'s analog lines in the telecom closet |
| 17:45.05 | blergh- | From what i can read, the Snom 370, 820 and 870 too |
| 17:45.11 | igcewieling | "I'm the phone guy". "OK, let me show you where the phone closet it". |
| 17:45.18 | blergh- | :D |
| 17:45.29 | blergh- | I don't have a hardline |
| 17:46.00 | igcewieling | blergh-: A month ago I'd have said "nobody is interested in your phone calls", but I was obviously wrong. |
| 17:46.20 | igcewieling | blergh-: My point is that VoIP is not that much less secure than analog |
| 17:46.28 | blergh- | http://wiki.snom.com/Networking/VPN |
| 17:46.28 | *** join/#asterisk peetaur2 (~peter@x2f16a04.dyn.telefonica.de) |
| 17:47.35 | blergh- | igcewieling: true true, but adding the layer of point-to-point encryption couldn't be all that bad |
| 17:47.54 | igcewieling | blergh-: correct, but I personally feel doing it ON the phone is not the right place. |
| 17:48.23 | WIMPy | It will only prevent them from listening realtime. A few minutes later they will have cracked the key anyway. |
| 17:48.25 | igcewieling | blergh-: how do you connect to the PSTN? |
| 17:48.43 | igcewieling | does your ITSP support VPN connections? |
| 17:49.00 | igcewieling | not that it matters, your ITSP's upstream is where the govt would be capturing your calls. |
| 17:49.49 | WIMPy | that's why they pay the bandwidth for ITSPs so they won't do directmedia. |
| 17:50.06 | WIMPy | So if you need a sponsor for bandwidth.... |
| 17:52.11 | blergh- | igcewieling: correct, unless you had full access all the way to the person(s) having access to the PRI |
| 17:52.38 | WIMPy | ? |
| 17:52.52 | igcewieling | We don't do direct media because we are contractually (not the govt) |
| 17:53.10 | Cuzner | igcewieling: is this to be a " |
| 17:53.13 | igcewieling | prohibited. 8-( |
| 17:53.15 | Cuzner | oops "hotel phone"? |
| 17:53.42 | Cuzner | will it be moving around a lot, or is the ip going to be static for the most part, or usually from a particular subnet? |
| 17:53.57 | Cuzner | you could always just iptables in front of it, allow the range you're coming from. |
| 17:54.30 | Cuzner | but if you're really concerned about someone sniffing your packets in realtime, maybe that's not such a good solution. |
| 17:55.50 | blergh- | nah, i'm not *that* paranoid, it's just more of a fun thing that would be cool to do, learning by doing and all that |
| 17:55.52 | Cuzner | oh |
| 17:56.03 | Cuzner | how about an ATA that does VPN? |
| 17:56.17 | Cuzner | instead of getting a full voip phone, get an ATA/bridge? |
| 17:56.18 | Cuzner | http://www.realtonetech.com/product/voip-ata/55-wss2110.html |
| 17:56.44 | WIMPy | Cuzner: You're into S&M? |
| 17:56.47 | blergh- | ATM my setup is more or less; elastix + trunk + local-did provider |
| 17:56.55 | Cuzner | WIMPy: why do you ask? |
| 17:57.03 | *** join/#asterisk Starpoint09 (~Starpoint@static-71-244-26-77.dllstx.fios.verizon.net) |
| 17:57.15 | WIMPy | Because you suggest using an analog phone. |
| 17:57.40 | Cuzner | you think someone's going to be snorting between his analog phone and the ATA? |
| 17:58.40 | igcewieling | WIMPy: Most people don't have as severe a case of Analog Phobia as you. |
| 17:59.02 | WIMPy | Because they never tried? |
| 17:59.04 | igcewieling | Cuzner: you snort cocaine, you sniff traffic. |
| 17:59.11 | blergh- | The only real difference this would make, is the connection between the PBX and me, so somewhat pointless in that sense |
| 17:59.14 | blergh- | but yeah |
| 18:01.01 | WIMPy | actually finds it really annoying how sip phones try to emulate analog stuff. |
| 18:01.57 | Cuzner | igcewieling: the tool I use to do it sometimes is called snort, you say tomotoes I say tomatoes. |
| 18:03.00 | blergh- | You'll have to excuse my lack of knowledge on this subject, I am still trying to get the hang of things, so |
| 18:03.37 | igcewieling | You'd think the packets would stick in your nose. |
| 18:03.50 | igcewieling | Maybe just snort the 1's. |
| 18:04.43 | WIMPy | Better snort the highs than sniff the lows? |
| 18:06.12 | blergh- | Is ZRTP a viable option? |
| 18:06.32 | WIMPy | Not supported by Asterisk. |
| 18:06.37 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 18:10.50 | blergh- | WIMPy: ouch, no options tho? Or would that be to just VPN it? |
| 18:11.26 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:11.39 | WIMPy | SRTP |
| 18:16.44 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 18:17.14 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
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| 18:53.37 | *** join/#asterisk kchr (kchr@forskningsavd.se) |
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| 18:54.17 | smkelly | pokes file |
| 18:54.25 | kchr | hello. i'm trying to get started with the voicemail feature, and have put the template files for a voice in place. however, asterisk seems to look for ulaw (.au) format, while the files provided by my dist packager is in gsm (.gsm). how do i set the format to look for? |
| 18:55.05 | kchr | (i assume this is not the "format" option under [general] in voicemail.conf, since that says "writing voicemail" in the comments) |
| 18:55.16 | kchr | (also it makes no difference in behavior) |
| 18:55.32 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ac78:d36a:e6dc:5da2) |
| 18:56.50 | [TK]D-Fender | ".au" in not the valid extension format for ulaw. that would be ".ulaw" |
| 18:57.23 | [TK]D-Fender | Also you don't tell it what format to play. That is automatic based on the most efficient transcoding between your files & the actual channel |
| 18:58.09 | igcewieling | kchr: Put the CLI output of a failed call on pastebin (not this channel) |
| 18:58.10 | igcewieling | ~pob |
| 18:58.12 | igcewieling | ~pb |
| 18:58.13 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 19:00.43 | kchr | igcewieling: http://pastebin.com/KK2dfX0D |
| 19:01.28 | igcewieling | kchr: the ulaw message means nothing. |
| 19:01.37 | igcewieling | what directory are your sound files located? |
| 19:01.42 | igcewieling | What verison of Asterisk are you using? |
| 19:01.58 | [TK]D-Fender | kchr: You seem to be missing the basic stock sound files... |
| 19:02.07 | igcewieling | Oh! VERY sorry. you are using a packaged Asterisk. |
| 19:02.16 | [TK]D-Fender | kchr: And you did not tell Voicemail to play one of the standard announcement files at all |
| 19:02.20 | igcewieling | You are dead to me now. |
| 19:02.27 | kchr | haha |
| 19:02.47 | kchr | i'm jusing locally compiled packages but yeah, theyre vanilla as the debian sources come |
| 19:02.53 | kchr | *theyre as vanilla as |
| 19:03.15 | kchr | i put the files under /var/lib/asterisk/sounds/ as the package created that one |
| 19:03.16 | igcewieling | Download Asterisk. Install it. Profit. |
| 19:03.48 | igcewieling | kchr: that is wrong, depending on your version of Asterisk and the asterisk.conf settings |
| 19:03.56 | kchr | that doesn't really answer my question... should i convert the files to make them match my caller(s) codec? |
| 19:04.10 | igcewieling | kchr: you can if you want, but it won't do a bit of good. |
| 19:04.12 | kchr | igcewieling: ok, what setting in asterisk.conf am i looking at? |
| 19:04.25 | *** join/#asterisk Mon|A|rch (~sbean@72.29.180.35) |
| 19:04.27 | [TK]D-Fender | kchr: pastebin the whole thing |
| 19:04.51 | [TK]D-Fender | kchr: And you shouldn't need to convert. You do not have the right files in the right place.. or ar missing the format modules to play them back. |
| 19:04.57 | Mon|A|rch | for some reason I'm having trouble finding this on the internet; how do you set the autofallthrough context? |
| 19:05.01 | Mon|A|rch | is it in sip.conf? |
| 19:05.16 | igcewieling | kchr: the option about sound file prefixes in asterisk.conf.sample included in the Asterisk source code. |
| 19:05.17 | [TK]D-Fender | Mon|A|rch: "autofallthrough context" <- no such thing |
| 19:05.18 | kchr | [TK]D-Fender: i checked the available formats on CLI and gsm is in there |
| 19:05.32 | *** join/#asterisk cjm_ (~cjm@tclc.org) |
| 19:05.34 | [TK]D-Fender | kchr: then the files are in the wrong place, wrong permissions, etc |
| 19:05.35 | kchr | so my bet is on wrong directory |
| 19:05.39 | kchr | ok |
| 19:05.39 | igcewieling | kchr: you keep looking in the wrong places |
| 19:05.44 | [TK]D-Fender | kchr: PB your asterisk.conf |
| 19:05.47 | kchr | ok ok! |
| 19:05.55 | [TK]D-Fender | kchr: Actuall... "core show settings" <- |
| 19:05.58 | [TK]D-Fender | from * CLI |
| 19:06.14 | Mon|A|rch | i see |
| 19:06.33 | [TK]D-Fender | Mon|A|rch: Perhaps you meant something else. |
| 19:06.37 | Mon|A|rch | is the failed extension only for auto-dialout? |
| 19:06.45 | Mon|A|rch | [TK]D-Fender, probably |
| 19:07.18 | cjm_ | Hi Folks, Do I need OpenSER with Asterisk to host myself to receive calls from outside my network, or will Asterisk server requests for sip: details? |
| 19:07.24 | Mon|A|rch | on a failed dialout, how can i catch that and execute some dialplan stuff |
| 19:07.38 | kchr | [TK]D-Fender: ok here is asterisk.conf - http://pastebin.com/UAAk0YYR |
| 19:07.56 | [TK]D-Fender | Mon|A|rch: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out#Thefailedextension <-- seems to be documented just fine here |
| 19:08.23 | [TK]D-Fender | [15:05][TK]D-Fenderkchr: Actuall... "core show settings" <- |
| 19:08.28 | WIMPy | cjm_: Asterisk will do |
| 19:08.42 | kchr | [TK]D-Fender: http://pastebin.com/td7FG0MT |
| 19:08.56 | cjm_ | WIMPy, Thanks much. |
| 19:10.51 | [TK]D-Fender | kchr: ls -la /var/lib/asterisk/sounds/en/* |
| 19:14.35 | kchr | [TK]D-Fender: http://pastebin.com/GPK0Mwya |
| 19:14.58 | [TK]D-Fender | kchr: You should generally never be running asterisk as ROOT ... |
| 19:15.31 | [TK]D-Fender | kchr: Which if you did sanely should probably be as "asterisk" and you'll be wanting to fix your permissions on all of that |
| 19:16.15 | kchr | [TK]D-Fender: i'm not, the files are however (on purpose) made not writable by asterisk user |
| 19:16.18 | kchr | since they are static |
| 19:16.30 | kchr | but they are world readable, as you can see |
| 19:16.55 | [TK]D-Fender | kchr: Fix them |
| 19:17.34 | cjm_ | Hi Folks, is call recording a feature of Asterisk or the softphone client? |
| 19:17.42 | kchr | asterisk is running as the user "asterisk", which can read those files (and access the directory theyre in) just fine |
| 19:17.56 | kchr | whom can read* |
| 19:18.11 | [TK]D-Fender | Do not assume the mode * will be opening those under |
| 19:18.28 | [TK]D-Fender | cjm_: YES |
| 19:19.33 | kchr | [TK]D-Fender: http://pastebin.com/xjCY09nT |
| 19:20.00 | cjm_ | [TK]D-Fender, I assume you mean the both can do this independently of each other. |
| 19:20.40 | [TK]D-Fender | cjm_: * can record calls. Your client may offer such a feature. You'd have to check your client to be sure. |
| 19:21.41 | cjm_ | [TK]D-Fender, Thanks. That helps. |
| 19:22.43 | [TK]D-Fender | kchr: Since you seem intent on disregarding sane advice I wish you the best of luck. |
| 19:22.55 | cjm_ | Hi Folks, Can someone with a Skype client contact me on my Asterisk network? I think that means, is the skype client captive to the Skype SIP server? |
| 19:23.54 | [TK]D-Fender | cjm_: Skype offers a SIP interconnect service if you have a business account. * doesn't not natively support Skype |
| 19:24.41 | *** join/#asterisk peetaur2 (~peter@x2f16a04.dyn.telefonica.de) |
| 19:24.42 | cjm_ | [TK]D-Fender, So, my friends will have to get some other random softphone client to call me? |
| 19:24.59 | [TK]D-Fender | cjm_: Or what I said |
| 19:25.28 | igcewieling | cjm_: Skype actively prevents Asterisk from connecting. In fact, they refused to renew Digium's contract for Skype support. |
| 19:26.20 | igcewieling | So, you either purchase business version of their service or you don't connect to them with Asterisk. your option. |
| 19:26.58 | cjm_ | igcewieling, And that sort of pettiness is part of why I dropped Skype in the first place. I'll simply try to live without them. |
| 19:27.04 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ac78:d36a:e6dc:5da2) |
| 19:27.22 | kchr | [TK]D-Fender: look, i'm not questioning your advice. i'm merely trying to understand the concepts of them -- changing permissions on the files (from world readable to owned by "asterisk") did not change any behavior on the server side. |
| 19:27.38 | igcewieling | cjm_: A good policy. I try not to do business with companies which do not want my business. |
| 19:27.51 | cjm_ | igcewieling, Hang on a second... Do I buy the business service, which allows all my frineds to call me, or do each of them do so? |
| 19:27.52 | igcewieling | kchr: did you change your languageprefix? |
| 19:28.08 | igcewieling | cjm_: whoever has the asterisk server |
| 19:28.52 | cjm_ | igcewieling, O.K., that would be me. Thanks. That helps me. I'll probably just pass on Skype and get all my friends to use something else. |
| 19:29.04 | cjm_ | igcewieling, Will Google Voice work? |
| 19:29.25 | igcewieling | cjm_: Google voice dropped support on July 1 |
| 19:29.39 | kchr | igcewieling: languageprefix = yes; defaultlanguage = en (asterisk.conf) |
| 19:29.57 | igcewieling | kchr: and your files are in /var/lib/asterisk/sounds/en ? |
| 19:30.20 | kchr | igcewieling: correct; all in *.gsm file extensions |
| 19:30.37 | igcewieling | NOT in /var/lib/asterisk/sounds, correct? |
| 19:30.40 | cjm_ | igcewieling, So, my option will be to get my friends to download a softphone. There are no popular public services that will allow my friends to find me? |
| 19:31.12 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 19:31.25 | igcewieling | cjm_: if you want your friends to call you using SIP then get them softphones. I have no idea what the second part means |
| 19:31.47 | boichev_ | can someone tell me if I have an OpenVox B800P card and I want to use it with an analog line that comes into my house ... everything is ok in asterisk I see the card and dahdi show channels gives me 23 channels but pri show spans gives me only " In Alarm, Down, Active" ... how should I connect the line that is currently an RG11 jack pluged in one of the cards jacks ? |
| 19:31.54 | kchr | igcewieling: yes, they actually were when i first came in here but i moved them when [TK]D-Fender asked me to verify the directory contents |
| 19:32.07 | kchr | and i restarted asterisk; no dice.. same pesky message |
| 19:32.24 | [TK]D-Fender | kchr: what about all of the FOLDER permissions & owners? |
| 19:32.37 | igcewieling | kchr: Seems like maybe your packages are not so great. |
| 19:32.43 | kchr | [TK]D-Fender: entire subtree of /var/lib/asterisk is owned by asterisk |
| 19:32.45 | WIMPy | boichev_: RG11? |
| 19:33.43 | kchr | igcewieling: yeah... :( |
| 19:33.52 | kchr | how can i increaase log verbosity? |
| 19:33.58 | kchr | i would like to see which path it tries to read |
| 19:34.18 | boichev_ | WIMPy: RJ11* |
| 19:34.35 | WIMPy | boichev_: Not much better. |
| 19:35.04 | WIMPy | boichev_: And how do you get 23 cahnnels? You should have 16. |
| 19:35.52 | WIMPy | boichev_: You connect that card with a normal patch cable to your NT. |
| 19:35.57 | boichev_ | http://pastebin.com/y5JuuBR5 |
| 19:36.30 | WIMPy | Ok, so 24. |
| 19:37.12 | boichev_ | WIMPy: I have only a blue and a black wire that are in the middle of the jack .... if that means anything.... |
| 19:37.13 | [TK]D-Fender | BRI uses RJ11? News to me... |
| 19:37.36 | boichev_ | [TK]D-Fender: I have no idea how to connect it or is it possible at all |
| 19:37.40 | WIMPy | boichev_: That won't fit. Where do they come from? |
| 19:37.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.65) |
| 19:38.27 | igcewieling | [TK]D-Fender: yes, it does. |
| 19:39.15 | boichev_ | WIMPy: If I have to give the signaling cables to the two middle pins of the RJ45 the RG11 jack connects to them with no problems.... :? |
| 19:39.45 | WIMPy | boichev_: It's a 4 wire port. So no go. Where does that cable come from? |
| 19:40.02 | boichev_ | WIMPy: my telephone company |
| 19:40.03 | cjm_ | igcewieling, "Public Services", like Skype or Google Voice. Meaning something my friends might already have running on their system, as opposed to something special they would have to add to keep in touch with me, and ask the question, is it worth it... |
| 19:40.17 | WIMPy | boichev_: then you're missing the NT. |
| 19:40.48 | igcewieling | boscage: what country are you in? |
| 19:41.04 | WIMPy | boichev_: Let's hope you didn't fry it. |
| 19:41.35 | boichev_ | WIMPy: Bulgaria |
| 19:41.39 | boichev_ | I still see it |
| 19:41.54 | boichev_ | So I don't think so |
| 19:42.04 | boichev_ | what should I do to make it work |
| 19:42.06 | WIMPy | That doesn;t mean much. |
| 19:42.28 | WIMPy | Anyway: Your telco should have given you an NT. That converts to the correct interface. |
| 19:43.20 | WIMPy | You cannot directly connect CPEs to the line. |
| 19:43.42 | boichev_ | WIMPy: I'm sorry I don't know that is an NT :) Can I have a link to read about it :) |
| 19:44.01 | WIMPy | Network Terminator. |
| 19:44.03 | *** join/#asterisk tenspeed705 (~keiths@goatfarm.vianet.ca) |
| 19:44.20 | WIMPy | It converts from the U interface to the S interface. |
| 19:46.20 | WIMPy | Ask your Telco. |
| 19:46.43 | igcewieling | cjm_: Skype and Google Voice may be public, but they are closed systems. |
| 19:47.46 | igcewieling | If someone wants to talk to me, they can call my telephone number and get ignored just like everyone else. |
| 19:48.03 | *** join/#asterisk josePHPagoda (~Thunderbi@69-92-77-41.cpe.cableone.net) |
| 19:48.04 | *** join/#asterisk guitarHester (~hester@nat/digium/x-iqvdgrexenitbuxi) |
| 19:48.26 | josePHPagoda | Hello everyone! I'm using .call files and they are working great. What I'm having trouble with is making a call file that rings a ringgroup |
| 19:48.28 | josePHPagoda | any ideas? |
| 19:49.19 | boichev_ | Just for information are there normal pci-e cards that I can plug normal phone in it without this .... ? |
| 19:49.40 | WIMPy | boichev_: You've got one. |
| 19:49.59 | WIMPy | (depending on your definition of "normal phone") |
| 19:50.23 | igcewieling | josePHPagoda: that won't work if you are using analog FXO |
| 19:51.08 | boichev_ | WIMPy: normal phone is an RJ11 analog phone with two wires :) |
| 19:51.25 | josePHPagoda | igcewieling: ??? |
| 19:51.30 | igcewieling | boichev_: WIMPy has this mostly unreasonable hatred of ANALOG |
| 19:51.35 | WIMPy | Then you need an Terminal Adaptor to connect them to your card. |
| 19:51.36 | igcewieling | josePHPagoda: Was I unclear? |
| 19:52.16 | igcewieling | josePHPagoda: Analog FXO ports are considered ANSWERED once dialing is completed, because the telco does not signal when the far end picks up. |
| 19:52.46 | josePHPagoda | igcewieling: I don't think you are answering my question... I asked about .call files. |
| 19:53.04 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.65) |
| 19:53.05 | igcewieling | josePHPagoda: you asked about ring groups not working with .call files. |
| 19:53.27 | igcewieling | the most common issue with "ring groups" not working is you are trying to dial PSTN numbers over analog FXO. |
| 19:53.31 | srl295 | Hello.. shouldn't I be able to use ${DB} inside an applicationmap? Such as: xdial => *9,peer/callee,SendDTMF,${DB(XDIAL/201)}# |
| 19:53.50 | josePHPagoda | yeah, I'm having trouble getting it to even attempt the call |
| 19:53.52 | srl295 | ast 11.3.0 |
| 19:54.12 | [TK]D-Fender | [15:48]josePHPagodaHello everyone! I'm using .call files and they are working great. What I'm having trouble with is making a call file that rings a ringgroup <- ringgroup is not a proper term for an * thing. |
| 19:54.29 | josePHPagoda | yeah, i'm using the freepbx distro |
| 19:54.52 | [TK]D-Fender | josePHPagoda: Show us what you;r actually doing in a PASTEBIN. |
| 19:54.55 | [TK]D-Fender | ~pb |
| 19:54.56 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 19:54.57 | [TK]D-Fender | ^^^ |
| 19:55.54 | cjm_ | igcewieling, "Skype and Google Voice may be public, but they are closed systems." Yes... And that's my problem, for which there is few solutions. I can pay Skype to let me "register" my Asterisk system, and that is only moderately objectionable. Can I pay Google to "register" as a Google Voice participant, meaning GV will forward to my Asterisk system? |
| 19:56.14 | [TK]D-Fender | cjm_: No |
| 19:56.27 | cjm_ | [TK]D-Fender, Damn.... Thanks. |
| 19:56.43 | [TK]D-Fender | cjm_: Google Voice is being shut down and ported to Google Hanguots which will no longer use XMPP thus no more connectivity for * |
| 19:56.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:57.22 | josePHPagoda | http://hastebin.com/yucovateki.avrasm |
| 19:57.27 | srl295 | [TK]D-Fender cjm_ figures, hours after I finally get Motif/XMPP working with * :P |
| 19:57.47 | [TK]D-Fender | josePHPagoda: Your call file is wrong |
| 19:57.49 | igcewieling | josePHPagoda: Local/201@somecontext/n |
| 19:57.56 | josePHPagoda | that's my .call file. It works fine as is (201 is the internal extension). I've set up a "ring group" inside of freepbx, but when I put in it's extension, it fails |
| 19:57.57 | josePHPagoda | ah |
| 19:58.04 | [TK]D-Fender | joYou are not specifying the CONTEXT in your local channel |
| 19:58.10 | josePHPagoda | so something like Local/201@local-context/n should work? |
| 19:58.25 | srl295 | [TK]D-Fender appreciate the headsup on GV |
| 19:58.25 | [TK]D-Fender | josePHPagoda: Where you actually picka VALID context... yes |
| 19:58.34 | josePHPagoda | okee dokee |
| 19:58.35 | guitarHester | [TK]D-Fender: so what happens when someone calls my GV number, ... will it buzz my hangouts login? |
| 19:58.35 | josePHPagoda | :) |
| 19:58.37 | josePHPagoda | i'll try it out |
| 19:59.26 | cjm_ | Hi Folks, Is there a "phone book" for SIP, meaning is there someplace I can register with my demographic data and someone can look me up to find my sip:<user>@... string? |
| 20:00.16 | igcewieling | cjm_: yes. hundreds of them |
| 20:00.33 | cjm_ | igcewieling, Ha! That's almost as bad as zero! |
| 20:00.43 | igcewieling | cjm_: exactly |
| 20:01.20 | josePHPagoda | [TK]D-Fender: thanks a ton! That worked wonderfully. :) |
| 20:01.44 | cjm_ | Hmm... Business opportunity. sip phone book search! How would that turn into money? |
| 20:02.16 | WIMPy | For more spam? |
| 20:02.23 | WIMPy | Err, spit? |
| 20:02.24 | carrar | is there anything else? |
| 20:02.55 | carrar | maybe some politic voting calls |
| 20:03.03 | cjm_ | WIMPy, Geeze, that would not be my first choice... Can't be a very good business opportunity if it can't generate income... |
| 20:03.45 | cjm_ | Subscription to the search results? |
| 20:03.46 | carrar | and for $50 we can put you on the SIP Do Not Call List |
| 20:04.06 | cjm_ | carrar, Ha! Brilliant! A reverse income model! |
| 20:04.20 | cjm_ | Pay me and I WON'T list you! |
| 20:04.28 | Cuzner | carrar: do you take bitcoins? |
| 20:04.35 | srl295 | hrm - wondering if I should use an AGI for my applicationmap entry if this won't work, SendDTMF,${DB(...)} |
| 20:04.37 | carrar | Only bitcoins |
| 20:04.41 | Cuzner | nice |
| 20:05.23 | [TK]D-Fender | [16:04]cjm_Pay me and I WON'T list you! <- the mafia doesn't like people cutting in on their business model... |
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| 20:11.06 | srl295 | answered my own question. Moved the SendDTMF into a Macro |
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| 20:33.17 | Elleni | pardon me if this was already asked a zillion times but does asterisk 11 support zrtp encryption? |
| 20:35.03 | WIMPy | no |
| 20:36.13 | Elleni | ok, thanks |
| 20:37.06 | WIMPy | People trying to implement it always vanish under mysterious circumstances. |
| 20:39.41 | Elleni | hm, ok... i see.. just asked because I am evaluation a free softphone on iphone and some of them do not support srtp over tls but do support zrtp for example linphone... |
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| 20:41.59 | boichev_ | If I get an fxo/fxs card will I be able to connect an analog line directly without an Analog to ISDN convertor (terminal adapter) |
| 20:42.35 | WIMPy | boichev_: So you have an analog line? |
| 20:42.53 | boichev_ | WIMPy: yes |
| 20:43.15 | WIMPy | What do you do with that ISDN card then? |
| 20:44.06 | WIMPy | Either you can go an get a FXO card and the line you've got or you get an ISDN line and the card you've got. |
| 20:47.24 | boichev_ | WIMPy: like I sad I have no idea about this :) and now I know :) Thx :) |
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| 21:00.46 | Elleni | WIMPy another question, does srtp over tls with key verification on the softphone side establish the same level of security as zrtp? |
| 21:01.13 | WIMPy | Security is a feeling, not a fact. |
| 21:01.25 | WIMPy | zrtp is end-to-end. srtp is only to the server. |
| 21:01.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:03.23 | Elleni | hm, ok, i see. well as asterisk does not support zrtp I will stay with srtp/tls for the moment :) |
| 21:05.03 | boichev_ | WIMPy: from what I read a Linksys/Sipura SPA-3000 will do the job of converting my analog line to ISDN right..... just for learning experience |
| 21:05.30 | Elleni | so that also means that a call from two srtp enabled extensions / softphones will never be end-to-end but will always have to be routed through the server, right? |
| 21:05.46 | WIMPy | boichev_: no |
| 21:06.05 | boichev_ | WIMPy: :( |
| 21:06.09 | WIMPy | such converters are not normally available. |
| 21:06.09 | igcewieling | boichev_: Where in the world could you possibly have read that? |
| 21:06.44 | WIMPy | You can get converters from ISDN to FXS in the next electronics supermarket. |
| 21:06.54 | boichev_ | igcewieling: just a sec I will dig it out ... probably I misunderstood something |
| 21:07.02 | igcewieling | boichev_: Start Googling. Most of us here have little interest in tutoring you in basic telecom stuff |
| 21:07.39 | WIMPy | Someone does manufacture FXO to ISDN converters. They have been used to enable people to connect their old phones to cable modems with FXS ports. |
| 21:10.18 | WIMPy | Well, most probably "did" not "does" |
| 21:10.43 | boichev_ | igcewieling, WIMPy http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5 (5.3.1 ..... and can also act as an EXTENSION to connect an ANALOG telephone via its FXS interface (aka ATA - Analog Telephone Adapter).) |
| 21:11.11 | WIMPy | And where do you see ISSDN mentioned? |
| 21:11.13 | igcewieling | boichev_: where does it say ISDN or BRI there? |
| 21:11.16 | igcewieling | looks at WIMPy |
| 21:12.00 | boichev_ | igcewieling: ok my bad ISDN is at 5.5 |
| 21:12.34 | [TK]D-Fender | boichev_: ANALOG to VOIP |
| 21:12.46 | [TK]D-Fender | boichev_: And 5.5 says nothing about ANALOG |
| 21:13.07 | boichev_ | [TK]D-Fender: I think I got it now :) |
| 21:14.27 | [TK]D-Fender | boichev_: Now you can have a BRI card ... AND an analog card in your server and * will sit in the middle and handle calls between them... but that isn't a dedicated devices... it's a whole computer with multiple interface cards |
| 21:15.09 | WIMPy | Doesn't Sangoma do cards with POTS and BRI mixed? |
| 21:15.38 | [TK]D-Fender | yup |
| 21:15.58 | [TK]D-Fender | Dependsing on FXS vs FXO |
| 21:15.58 | cjm_ | I've downloaded Ekiga, an SIP softphone. I need to register with Asterisk. I can't find any such activity on the GUI. Any advice? |
| 21:16.18 | [TK]D-Fender | cjm_: What GUI? |
| 21:16.25 | WIMPy | cjm_: Go to the support channel of that GUI. |
| 21:19.11 | cjm_ | [TK]D-Fender, http://asterisk.tclc.org |
| 21:19.40 | [TK]D-Fender | link no good |
| 21:20.10 | cjm_ | WIMPy, I have "Admin", "Applications", "Connectivity", "Reports", "Settings". No, "Support". |
| 21:20.49 | WIMPy | Support channel= Website/forum or other IRC channel of the vendor. |
| 21:24.00 | *** join/#asterisk _N1X_ (~ahmed@unaffiliated/n1x/x-2632350) |
| 21:24.26 | _N1X_ | hello every one |
| 21:25.02 | _N1X_ | i have 3 accounts from 1 sip / trunk provider |
| 21:25.15 | _N1X_ | every account have 2 channels |
| 21:25.44 | _N1X_ | i want to bind them or make failover |
| 21:26.05 | [TK]D-Fender | _N1X_: It's your dialplan... do whatever you want |
| 21:26.43 | _N1X_ | yes i miss the dialplan |
| 21:27.40 | _N1X_ | [TK]D-Fender, can u please give me a simple |
| 21:27.55 | [TK]D-Fender | _N1X_: Just dial them back to back |
| 21:28.08 | [TK]D-Fender | _N1X_: If the first one doesn't get answered.... it'll just continue on. |
| 21:28.10 | [TK]D-Fender | ~book |
| 21:28.11 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:28.12 | [TK]D-Fender | ^^^ |
| 21:36.07 | cjm_ | [TK]D-Fender, Please try again. I have updated Public DNS and you should be able to see the GUI in question. |
| 21:36.34 | [TK]D-Fender | cjm_: I get an apache test page |
| 21:36.35 | cjm_ | WIMPy, Umm... This is that channel... The GUI is for Asterisk. |
| 21:36.50 | [TK]D-Fender | No, asterisk is the ENGINE |
| 21:37.06 | [TK]D-Fender | GUI's are all bolt-on's that have nothing to do with direct asterisk development |
| 21:37.07 | cjm_ | [TK]D-Fender, Well, that's inconvenient... and wrong. I'm inside so I can't see from the public IP side. |
| 21:37.11 | WIMPy | No, it is FOR Asterisk, not by or from Asterisk. |
| 21:37.24 | cjm_ | [TK]D-Fender, Yes, but this is AsteriskNOW, so it came with. |
| 21:37.47 | WIMPy | This is #asterisk, not #asterisknow. |
| 21:37.52 | [TK]D-Fender | And this isn't #asterisknow the channel |
| 21:37.59 | WIMPy | We don;t support any GUIs in here. |
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| 21:38.03 | cjm_ | FreeBBX? Is that waht I am missing? |
| 21:38.07 | [TK]D-Fender | cjm_: And AsteriskNOW used to come with MULTIPLE GUI's |
| 21:38.18 | WIMPy | Then go to #freepbx |
| 21:38.21 | [TK]D-Fender | So perhaps you shouold learn to get SPECIFIC when we ask you "which GUI". |
| 21:38.28 | cjm_ | Whoa... O.K., my mistake. |
| 21:38.43 | [TK]D-Fender | Yes, FreePBX |
| 21:38.53 | [TK]D-Fender | As for setting up your softphone... it is an EXTENSION <- |
| 21:39.02 | Elleni | Asterisk: The Definitive Guide is great! |
| 21:40.08 | cjm_ | [TK]D-Fender, Yes. Thanks. Having never done this before, I am guessing about how some of this works. Don't I need to say somewhere, "sip:cjm@tclc.org" is extension <extension>" ? |
| 21:40.56 | [TK]D-Fender | cjm_: Where are you talking about? |
| 21:41.09 | [TK]D-Fender | cjm_: You softphone REGISTERS to *. That's all it needs to know. |
| 21:41.10 | cjm_ | Elleni, Yep... Got. Reading it. On page 72 |
| 21:42.02 | cjm_ | [TK]D-Fender, I downloaded Ekiga, and it want to register to sip.ekiga.net by default. But, it can be configured. I'm just unclear on how to do it. |
| 21:42.27 | [TK]D-Fender | cjm_: username , passwrod, host/IP. It ain't Raw-Cat Sigh Hence. |
| 21:42.36 | [TK]D-Fender | cjm_: And now this is an EKIGA SOFT-PHONE client. |
| 21:42.40 | [TK]D-Fender | (question) |
| 21:43.03 | cjm_ | [TK]D-Fender, as nearly as I know. I've made mistakes before. |
| 21:43.51 | cjm_ | [TK]D-Fender, Do I need to have a username. password on a SIP proxy, meaning Asterisk? And that is my current mystery. |
| 21:44.15 | cjm_ | I'm not sure how I do that. I need to tell Asterisk that I exist. |
| 21:44.49 | [TK]D-Fender | Yes.. did you set up an account on it yet? |
| 21:45.10 | cjm_ | [TK]D-Fender, No. I'm quite sure of that. I thought the GUI would let me do this. |
| 21:45.27 | *** join/#asterisk imox (~imox@24-134-17-195-dynip.superkabel.de) |
| 21:45.29 | cjm_ | [TK]D-Fender, So far, I can't figure out how to do it. That is why I asked for help. |
| 21:45.30 | [TK]D-Fender | No, Your server has nothing to do with you setting up your Ekiga client |
| 21:45.47 | [TK]D-Fender | There are THOUSANDS of VoIP clients out there. it is your job to fill in the 3 silly blanks |
| 21:46.21 | cjm_ | [TK]D-Fender, I agree. But after telling the Ekiga client the credentails, Ekiga is going to ask Asterisk on port 5060, and Asterisk need to know about me. Right? |
| 21:46.51 | [TK]D-Fender | Correct... the same credentials you told the phone should match those of the extension you defined in FreePBX |
| 21:47.29 | cjm_ | And now we are back to how I tell FreePBX.... I think I heard that I need to ask that question somewhere else. Right? |
| 21:47.46 | [TK]D-Fender | I already answered that .... |
| 21:47.55 | [TK]D-Fender | [17:38][TK]D-FenderAs for setting up your softphone... it is an EXTENSION <- |
| 21:48.46 | cjm_ | [TK]D-Fender, Geeze! <SLAP> Yes... Now I see it. Under "Application" => Extensions!!! Thanks. Sorry for the density. |
| 21:49.27 | cjm_ | [TK]D-Fender, I really was not trying to be obtuse; it just came out that way. Thanks for the help. |
| 21:49.58 | [TK]D-Fender | cjm_: Slowly but (hopefully) surely |
| 21:50.54 | Elleni | cjm beeing a noob myself I can recommend you the following two aproaches: 1) read on Asterisk: The Definitive Guide and do step by step what they suggest (dialplan, extension...) or 2) follow http://www.freepbx.org/support/documentation/administration-guide/adding-extensions |
| 21:52.28 | Elleni | oh.. I come from seeing that you found it :) |
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| 22:16.54 | floren | good afternoon |
| 22:17.06 | WIMPy | Good morning |
| 22:17.23 | floren | i was wondering what could cause the retransmissions listed in this thread: http://forums.asterisk.org/viewtopic.php?f=1&t=87333 |
| 22:17.34 | floren | hi WIMPy |
| 22:18.28 | floren | the device reports: sip_sm_ccb_match_branch_cseq : Method index not found |
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