| 00:00.37 | p3nguin | The phones were all configured by hand, and now one phone never loads. It says Error loading <MAC>.cfg and goes into a reboot loop. What I have read indicates that this is because of a corruption in the flash, and the solution is to give it a BootROM from a provisioning server. |
| 00:00.49 | p3nguin | If you've got a better idea, I'm all ears. |
| 00:00.57 | tm1000 | p3nguin: what phone |
| 00:01.05 | p3nguin | Polycom 500 |
| 00:01.16 | tm1000 | wow...super old |
| 00:01.17 | p3nguin | unknown BootROM version, SIP app 2.1.2 |
| 00:01.17 | tm1000 | :-\ |
| 00:01.44 | tm1000 | p3nguin: I agree with the bootrom |
| 00:01.47 | tm1000 | solutio |
| 00:01.48 | tm1000 | n |
| 00:02.39 | p3nguin | From what I can tell, there is only one choice for BootROM version: 3.2.3 Rev B. |
| 00:03.00 | p3nguin | But since I can't put my hands on the phone and I've never checked it before, I have no clue if that's what was running on it. |
| 00:05.03 | [TK]D-Fender | p3nguin: if it loops then the bootrom is fine. The SIP app may be corrupted or its configs (esp if mismatched), but the BootROM is not the issue |
| 00:05.24 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 00:06.14 | p3nguin | Interesting. I'll put in the sip files and leave out the bootrom file, then. I've got the tftpd running on that LAN right now; just waiting on someone to figure out how to point the phone to the server from the phone menu. |
| 00:07.09 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-bsnhtsmyqbqdrplw) |
| 00:08.37 | [TK]D-Fender | SIP versions you can yank around. first step to fixing a phone is to unpack a stock firmware into a separate user folder and point the phone there via FTP force it on it. This can be an up or down-grade |
| 00:08.45 | pzn | Please someone validate my thinking: two asterisk servers. server1: cityA, 10lines (e1 link), internet 10Mbps (fullduplex). server2: cityB, 10 lines (e1 link) + internet 10Mbps (fullduplex). If I install asterisk in both servers, can I "save money" when an extension in cityA needs to call extension in cityB (free call via internet) and when an extension in cityA needs to call someone-else in cityB (cheaper call using internet from cityA to cityB and |
| 00:08.45 | pzn | <PROTECTED> |
| 00:09.06 | [TK]D-Fender | You should be able to get it flushed to the boring OOB condition with an unlabeled softkey this way |
| 00:09.31 | [TK]D-Fender | pzn: Of course |
| 00:10.16 | p3nguin | If at all possible, I'd prefer to retain all settings in the phone, but I do not have a backup of the settings on the boot server. |
| 00:10.33 | p3nguin | If it's not possible, however, I'm not going to cry over it. |
| 00:10.44 | p3nguin | I'll just treat it like a new phone. |
| 00:11.23 | pzn | [TK]D-Fender, nice! good answer :-) now other detail: cityA and cityB will use same internet provider, so the infrastructure will have lower latency. Since internet may be used also to navigate, do you think installing some kind of bandwitdh priority router will have best results? |
| 00:11.26 | [TK]D-Fender | p3nguin: New Phone Smell. It's worth it.... |
| 00:12.03 | [TK]D-Fender | pzn: 10mbit is a pretty big lin. Not sure what kind of users or voluem you'd have that you would have to show concern for. |
| 00:12.34 | [TK]D-Fender | pzn: if 10 lines per site, how many calls would be in the cross-site at any given time? I don't thikn you should have to blink at this.... |
| 00:13.56 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 00:14.03 | pzn | maybe 2 or 3 cross-site calls. about 20 users. youtube videos may be a problem with internet traffic. I'll try without bandwidth control, and if it is not good, I can add later |
| 00:15.42 | p3nguin | Do I need to move the localization file of my choice into the root, or leave it in the SoundPointIPLocalizastion sub directory? |
| 00:16.10 | [TK]D-Fender | p3nguin: Dump it whole... it needs to be placed as it is and ensure the user can choose on the phone |
| 00:16.21 | p3nguin | okay |
| 00:16.48 | p3nguin | I simply extracted the contents as-is, retaining hierarchy. |
| 00:17.26 | pzn | for calls that use IVR menus that are cross-site and depend on dtmf tones, should I use G711u? |
| 00:17.33 | p3nguin | And then I need to copy 000000000000.cfg to myMAC.cfg? |
| 00:17.56 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:19.53 | [TK]D-Fender | pzn: Start with G.711 just beause for the quality. DTMF won't matter. It will be OOB anyway (you should using rfc2833 regardless) |
| 00:20.13 | [TK]D-Fender | p3nguin: No, leave it completely stock and jsut get it booted even if "unconfigured" |
| 00:20.39 | [TK]D-Fender | p3nguin: Once it plain boots to the app at all then you can go ahead and start personalizing. |
| 00:22.02 | p3nguin | Okay. |
| 00:22.28 | pzn | [TK]D-Fender, imagine this: analog phone -> linksys-PAP2 -> local network -> asterisk1 -> internet -> asterisk2 -> E1 link -> (whoknows what is here) -> IVR with dtmf detection. dtmf be transmitted as inband, isn't? |
| 00:22.53 | p3nguin | I'd like to control all the phones from the asterisk box, but it's outside of the LAN, so I guess that's probably not a great idea. |
| 00:25.59 | [TK]D-Fender | pzn: analog phone (inband) -> linksys-PAP2 (rfc2833) -> local network -> asterisk1 -> internet -> asterisk2 -> E1 link (inband)-> (whoknows what is here) -> IVR with dtmf detection |
| 00:26.05 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 00:26.31 | [TK]D-Fender | pzn: As of your ATA it is already OOB. stays that way until you terminate to another PSTN link like your E1 |
| 00:26.53 | pzn | [TK]D-Fender, nice! asterisk does rfc->inband conversion automatically? will I need to configure anything in asterisk? |
| 00:27.39 | [TK]D-Fender | pzn: Its native. remember there are 2 channels in a "call". your E1 is just one end, and just like the audio, DTMF is converted to what they need |
| 00:31.58 | pzn | [TK]D-Fender, ok, that will solve my scenario! |
| 00:34.22 | *** join/#asterisk carrar (tim@osburn.com) |
| 00:34.28 | carrar | woot book update |
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| 00:45.21 | pzn | how can I force asterisk to use inband-dtmf for a specific sip device (testing purposes)? |
| 00:46.26 | p3nguin | Set the dtmfmode in the peer entry to inband. |
| 00:48.55 | pzn | p3nguin, worked as expected, thanks! |
| 00:50.03 | michael-i | anyone have a pre-paid sip provider recommendation which has a free DID? sipgate.com is out of numbers :) |
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| 00:52.51 | p3nguin | michael-i: ipcomms |
| 00:53.35 | p3nguin | ipkall also has free DIDs, but there's no paying for anything because all the offer is a free DID. |
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| 00:53.52 | p3nguin | What service are you wanting to pay for once you have a free DID? |
| 00:54.19 | michael-i | p3nguin: gotcha, thanks! I only need this for basic testing and will kill it in a few days |
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| 00:54.44 | p3nguin | I'd probably use ipkall for a quick test. |
| 00:54.54 | p3nguin | It's easy for you to create and destroy the account. |
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| 00:55.55 | *** mode/#asterisk [+o file] by ChanServ |
| 00:56.38 | michael-i | p3nguin: but I need registration, not just direct sip uri, incomms looks good |
| 00:58.00 | p3nguin | You don't have a dynamic DNS host name? |
| 00:58.26 | [TK]D-Fender | not even needed |
| 00:58.46 | [TK]D-Fender | sign up with an open SIP proxy like Ekiga.net and you can point Ipkall to them |
| 00:58.48 | p3nguin | Uh, how will calls get to him if his IP address changes? |
| 00:58.53 | [TK]D-Fender | :) |
| 00:58.56 | [TK]D-Fender | that's how :) |
| 00:59.02 | p3nguin | Aren't we talking about asterisk here? |
| 01:00.33 | [TK]D-Fender | yes. reg * to Ekiga.net./ Point Ipkall to send to your Ipkall URI. |
| 01:00.37 | p3nguin | Unless asterisk will register to ekiga.net, then that would solve the problem. |
| 01:00.54 | p3nguin | Fancy. |
| 01:00.59 | [TK]D-Fender | You may now grab a plushie off the middle shelf :) |
| 01:01.03 | p3nguin | haha |
| 01:02.14 | lkthomas | guys, anyone using cisco ip phone now ? |
| 01:02.35 | p3nguin | Yes. |
| 01:02.43 | michael-i | lovelyâ¦any place do this without requiring me to call customer service and request my did? they're already closed |
| 01:02.44 | lkthomas | 7942 ? |
| 01:02.52 | p3nguin | not me. |
| 01:03.02 | p3nguin | michael-i: Yes, ipkall. |
| 01:03.08 | lkthomas | p3nguin, so you are using another model or cisco phone ? |
| 01:03.22 | p3nguin | I use 7960, 7940, and 7912. |
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| 01:03.38 | lkthomas | I have a big problem on directories button |
| 01:03.48 | lkthomas | I pointed directoryurl and serviesurl to same xml |
| 01:03.56 | lkthomas | directories does not pick up |
| 01:04.00 | lkthomas | while servies could |
| 01:04.06 | michael-i | p3nguin: as I mentioned aboveâ¦I need registration, don't want to mess with dyndns at this stage |
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| 01:04.18 | lkthomas | p3nguin, is it because my phone is not -ccme licensed ? |
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| 01:04.26 | pzn | client1 is dtmf mode = sip info, asterisk config for client1 is dtmfmode=info; client2 asterisk config is dtmfmode=inband. I dial from client1 to client2, all ok. if I press "1" on client1 then I see UDP package for sip info, but I heard no tone in client2. why? |
| 01:04.30 | p3nguin | michael-i: You should have been reading what [tk]d-fender said. |
| 01:04.34 | michael-i | errrâ¦.wait, |
| 01:04.47 | michael-i | p3nguin: let me humbly back out of that argument |
| 01:04.50 | p3nguin | I'll recap. |
| 01:04.55 | lkthomas | p3nguin, hey ? |
| 01:05.03 | michael-i | let me grab my free sip addresses and ipkall and not bother anyone ;) sorry |
| 01:05.15 | p3nguin | Register against ekiga.net, then direct ipkall to your ekiga SIP URI. |
| 01:05.57 | p3nguin | I haven't tried it, but it sounds like it would work just fine. |
| 01:06.15 | lkthomas | p3nguin, so you don't use directory service ? |
| 01:06.29 | michael-i | I've done it before, just with a local serverâ¦monday dumbnesssssss |
| 01:07.10 | p3nguin | I have a dynamic DNS host name for my SIP URI that I put into ipkall, so I never needed to proxy it like that. |
| 01:07.29 | p3nguin | lkthomas: I do not use a 7942, nor do I use directory services on a 7942. |
| 01:07.43 | lkthomas | p3nguin, any other model you use directory ? |
| 01:08.02 | p3nguin | When I ran SIP on my phones, I used a directory. |
| 01:08.18 | lkthomas | and then how you implement it on directory button ? |
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| 01:11.41 | tyman | I am being DESTROYED today by the linode fremont, ca datacenter DDOS attack (http://status.linode.com/). I need fault tolerance (multi datacenter, multi itsp, multi server) |
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| 01:13.59 | tyman | I need recommendations for redundant ITSP, and hosting providers |
| 01:13.59 | tyman | I use flowroute and linode now respectively. |
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| 01:18.21 | [TK]D-Fender | voip.ms or les.net |
| 01:19.35 | tyman | thx TK |
| 01:29.53 | tyman | it's pathetic...any providers with california presence you can recommend? |
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| 01:30.14 | tyman | rackspace DFW location was very latent and jitter for me |
| 01:30.48 | tyman | oops...sorry about "it's pathetic"...private chat remnant |
| 01:31.15 | p3nguin | I think VoIP.ms has a PoP in LA. |
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| 01:31.28 | tyman | hosting providers...sorry. Linode competitor |
| 01:31.28 | p3nguin | losangeles.voip.ms, most likely. |
| 01:31.31 | p3nguin | Oh. |
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| 01:41.39 | Naikrovek | cripes. it took me 1/10th the time to write a java program to pull out the data I wanted from Asterisk than I spent trying to get Asterisk SNMP working. |
| 01:41.50 | Naikrovek | 1/100th maybe even. |
| 01:42.06 | Naikrovek | this is indicative of a huge problem |
| 01:42.14 | Naikrovek | SNMP is supposed to make this easy. |
| 01:43.00 | Naikrovek | I spent nearly two days trying trying trying to get Asterisk SNMP working, and maybe 30 minutes writing this java app. |
| 01:43.48 | Naikrovek | Linux is such a time sink. |
| 01:45.52 | tyman | 1/10th the time w/java no less, eh? I was planning on working on snmp with asterisk really soon...what were your troubles |
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| 01:50.05 | Naikrovek | getting SNMP on linux working is easy. Getting Asterisk SNMP is not easy. |
| 01:50.35 | Naikrovek | the Asterisk MIBs just will not show up when I walk; it's as if they aren't there. |
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| 01:50.52 | Naikrovek | According to everything I can find, i've done it correctly. |
| 01:51.10 | Naikrovek | module is loaded, config is correct, no errors in the logs; just doesn't work. |
| 01:51.16 | Naikrovek | giant waste of time. |
| 01:52.40 | tyman | interesting...that sucks... |
| 01:53.33 | Naikrovek | well |
| 01:53.48 | Naikrovek | the worst part of it is, that I MUST be doing something wrong. Somewhere. |
| 01:53.49 | russellb | we were able to get it working when documenting it for the asterisk book ... *shrugs* |
| 01:53.57 | russellb | sorry it didn't work for you |
| 01:53.59 | mocker | Naikrovek: I've gotten it wokring. |
| 01:53.59 | tyman | did u verify the mib? |
| 01:54.20 | Naikrovek | russellb: that book is the last thing I tried; I created a new virtual machine and did everything the book said; no workie workie. |
| 01:54.30 | russellb | :-/ |
| 01:54.37 | Naikrovek | no kidding |
| 01:54.45 | russellb | kicks SNMP |
| 01:54.47 | Naikrovek | lol |
| 01:54.53 | russellb | never trust something with "Simple" in the name |
| 01:55.13 | Naikrovek | so, i played to my strengths and wrote a java app that uses the AMI interface. took 30 minutes and I got what I wanted |
| 01:55.18 | tyman | no doubt... |
| 01:55.23 | Naikrovek | now Cacti is monitoring everything I wanted it to monitor. |
| 01:55.24 | russellb | excellent |
| 01:55.43 | Naikrovek | well, cacti is GRAPHING everything I wanted it to graph. |
| 01:56.05 | Naikrovek | Cacti is not a monitoring system. CINAMS |
| 01:56.16 | mocker | hehe, switched to Zabbix so I get monitoring and graphing in one. |
| 01:56.24 | mocker | Tired of setting cacti + nagios all the time. |
| 01:56.57 | Naikrovek | I would switch, but cacti and I have a history. There's LOTS of data in there I need, so I'm going to keep using it. And I'm happy to notice today that it's not dead anymore. new version a few days back and updated roadmap |
| 01:57.03 | Naikrovek | let's see if it holds. |
| 01:57.54 | *** join/#asterisk coppice (~chatzilla@m121-202-79-47.smartone-vodafone.com) |
| 01:57.56 | tyman | mocker: you're having good success with zabbix? |
| 01:57.59 | Naikrovek | the asterisk snmp thing should be far easier than it is. i knwo if i let someone look they'd find whatever thing I've been doing wrong but there should at least be an error something telling me |
| 01:58.08 | mocker | tyman: Yup. |
| 01:58.27 | tyman | I'll check it out |
| 01:58.51 | mocker | tyman: Getting used to how it does things takes a bit, but then it just makes sense. |
| 01:59.14 | tyman | as with most things *nix |
| 01:59.25 | mocker | tyman: Indeed. :) |
| 01:59.52 | mocker | tyman: But getting free graphing for any monitoring I setup is awesome. |
| 02:00.18 | tyman | <PROTECTED> |
| 02:00.34 | mocker | No, all built in Zabbix stuff. |
| 02:00.48 | mocker | Which means you can pan to different times, zoom in on sections of time, etc.. |
| 02:02.02 | tyman | sounds interesting...a positive note to an otherwise ***tty day |
| 02:07.08 | mocker | sad that I see that and first think of computer ttys |
| 02:07.21 | russellb | me too :-) |
| 02:07.48 | tyman | i noticed that too after typing...funny |
| 02:08.00 | tyman | russellb: how's RH? |
| 02:08.04 | russellb | excellent |
| 02:08.15 | p3nguin | And I tried to think of how many six-letter words there are that end in tty. I guess I'm the weirdo. |
| 02:08.48 | tyman | did you shift away from asterisk development entirely or are you working on something with RH around Asterisk? Or can u say? |
| 02:09.05 | mocker | checks if any new release candidates are out. |
| 02:09.16 | russellb | I'm not working on Asterisk for Red Hat, but I still participate as a community developer |
| 02:09.45 | russellb | I'm working on systems management APIs and related infrastructure at the moment. |
| 02:09.55 | tyman | Did you have to relocate to RTP? |
| 02:10.34 | russellb | though lots of people in my department work on clustering and high availability stuff, which is somewhat related to asterisk, in that people use it with asterisk |
| 02:10.38 | russellb | nope, I work from home |
| 02:10.49 | tyman | perfect... |
| 02:10.55 | mocker | russellb: corosync stuff? |
| 02:11.16 | russellb | Yeah, corosync is owned by some people in my department |
| 02:11.19 | russellb | i have worked on it a little bit |
| 02:11.52 | mocker | stonith may be my favorite acronym. |
| 02:11.59 | russellb | yes, it's quite amusing |
| 02:12.12 | tyman | googling that now...:-) |
| 02:12.22 | russellb | Shoot The Other Node In The Head |
| 02:12.28 | tyman | got it... |
| 02:12.55 | russellb | in clustering, when something fails, and you want to make sure it's _really_ dead, and not a half-alive zombie node screwing up your cluster |
| 02:13.18 | mocker | russellb: You work with Jared at all? |
| 02:13.22 | russellb | not really |
| 02:13.49 | russellb | we talk now and then but we're in pretty different areas |
| 02:14.24 | russellb | I talk to him more in Asterisk channels on IRC than anywhere else, heh |
| 02:14.49 | mocker | russellb: As it should be. |
| 02:15.04 | mocker | real life interaction should be avoided if possible. |
| 02:15.15 | russellb | well I mean, as opposed to something through Red Hat |
| 02:15.16 | mocker | (just saw contagion over the weekend) |
| 02:16.03 | tyman | mocker: and only fist bumps when necessary |
| 02:17.35 | mocker | reads up on the pomodoro technique. |
| 02:19.21 | tyman | mocker: what os do u use? |
| 02:19.46 | mocker | os x on my laptop |
| 02:20.12 | mocker | cent generally on my servers. |
| 02:20.14 | tyman | me too... |
| 02:20.19 | tyman | ubuntu on servs for me |
| 02:20.23 | tyman | use vitamin-R |
| 02:20.27 | tyman | tool for pomodoro |
| 02:20.38 | tyman | It works really really well |
| 02:20.45 | mocker | tyman: Cool, I'll check it out. |
| 02:20.47 | tyman | tried a bunch of others...crap |
| 02:21.15 | autofsckk | ubuntu + asterisk? is it good? |
| 02:22.06 | tyman | works great for me....asterisk 1.8.5.0 has been running on 10.04lts with zero issues since the first day 1.8.5.0 |
| 02:22.50 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:23.11 | russellb | do you use the asterisk.org repos for ubuntu? |
| 02:24.02 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 02:24.04 | mocker | guh, I had to patch app_queue to get rid of crashing every so often on 1.8.5-7 |
| 02:26.50 | autofsckk | anybody using asterisk on opensolaris or freebsd? is it really a lot much better performance than on any linux box? |
| 02:27.06 | tyman | slow-laris |
| 02:27.18 | russellb | i can tell you that virtually zero of the asterisk developers do any development or testing on anything but Linux |
| 02:27.29 | russellb | so for the most reliable install, I would _definitely_ go with Linux |
| 02:27.35 | russellb | otherwise ... YMMV, I suppose |
| 02:27.55 | russellb | not to say those platforms are bad, it's just not really targeted that much for Asterisk |
| 02:28.18 | autofsckk | russellb: what distro do you recommend? im testing right now asterisknow with centos, i normally use arch any recomendations? |
| 02:28.35 | p3nguin | I run Asterisk on Arch. |
| 02:28.37 | russellb | whatever you're comfortable with |
| 02:28.53 | russellb | at least if you don't mind installing from source |
| 02:29.06 | russellb | You can get good up to date packages in the major distros ... |
| 02:29.26 | russellb | asterisk.org has repos for CentOS / RHEL 5, Debian, and Ubuntu |
| 02:29.43 | russellb | Fedora has good asterisk packages both for Fedora and in EPEL for RHEL 6 |
| 02:30.20 | autofsckk | well im really new to asterisk, trying to understand it and reading a lot, i think that i will make a lot of testing on asterisknow and then when i fell more confortable with it ill change to arch |
| 02:30.33 | russellb | sounds good. |
| 02:30.43 | *** join/#asterisk bmg505 (~leon@196-209-163-53.dynamic.isadsl.co.za) |
| 02:30.44 | p3nguin | I'll give you my PKGBUILD to make life easier. |
| 02:30.48 | russellb | if you really want to learn, disable the gui on asterisknow :-) |
| 02:30.56 | autofsckk | so i think ill be around a lot :D |
| 02:31.15 | russellb | autofsckk: find the asterisk book online? |
| 02:31.45 | autofsckk | russellb: yes i do want to learn, i first installe asteriks on an arch box but nothin but text config files, then i saw the gui and it looked interesting ha |
| 02:31.57 | russellb | ~book |
| 02:31.57 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 02:31.58 | autofsckk | p3nguin: it would be great, thanks |
| 02:32.04 | russellb | that will help you get through the config files |
| 02:32.39 | *** join/#asterisk d1j0n (~Adium@pool-96-233-28-207.bstnma.east.verizon.net) |
| 02:33.19 | tyman | I love asterisk...it's a blast. Wish I could spend a lot more time with it...planning to |
| 02:33.31 | autofsckk | russellb: i've been reading for the last 2 days at the wiki from http://www.voip-info.org |
| 02:33.35 | tyman | lol |
| 02:33.47 | tyman | use the book |
| 02:33.52 | russellb | some good info there, but just be careful, _lots_ of outdated info there too |
| 02:33.55 | p3nguin | You'll need to forgot half or more of everything you've read there. |
| 02:34.19 | tyman | yeah...once you know what's new/old or just wrong then it's a good resource... |
| 02:34.40 | autofsckk | yes i am aware that some info is from 3 o 4 years ago |
| 02:34.44 | tyman | The book has structure...much easier to learn...that and the conf samples |
| 02:36.30 | autofsckk | for what i have read about asterisk it really kicks ass, i mean, you can do very interesting stuff with it, im amazed |
| 02:37.35 | autofsckk | anybody using asterisk with drupal ? |
| 02:38.51 | tyman | Anyone know of the best web switchboard for multi-tenant installs (oss or commercial) that's NOT fop2? |
| 02:39.48 | tyman | that works via AMI on top of vanilla asterisk I mean |
| 02:39.59 | [TK]D-Fender | autofsckk: Drupal is a CMS... I'm not sure what relationship you're asking about there... |
| 02:43.39 | autofsckk | [TK]D-Fender: maybe connecting customers trough your website to ppl receiving calls ? im trying to know all the things that could be done with asterisk, i dont know, maybe this could be done, or maybe im just telling nonesense |
| 02:44.00 | p3nguin | If I run one patch and need to recompile a single channel driver, how should I go about doing that? I doubt make channels/chan_sip.c would work. |
| 02:44.26 | russellb | p3nguin: if you've already compiled, then just "make" will only build what has changed |
| 02:44.32 | russellb | or whatever depends on what has changed |
| 02:44.41 | tyman | make yes...that's what make does for u |
| 02:44.52 | russellb | otherwise, there's not really an easy way to do that in the asterisk build system |
| 02:45.01 | p3nguin | okay |
| 02:45.08 | [TK]D-Fender | autofsckk: All the uses? I use mine as a jukebox. It also used to make me coffee |
| 02:45.21 | russellb | you can run menuselect and disable everything, but that'll take as long as just letting it all build |
| 02:45.38 | [TK]D-Fender | autofsckk: I'm not sure just how crazy you might want to get, but Asterisk may have beaten you there... |
| 02:46.05 | mocker | autofsckk: I used Asterisk w/ Adhearson to play simon over the phone a couple weeks ago. |
| 02:46.25 | mocker | adhearsion. |
| 02:46.32 | tyman | v10 will let you run a webcam girl call center |
| 02:46.38 | russellb | lol.. |
| 02:46.40 | [TK]D-Fender | mocker: I was thinking of adapting Zork to it :) I already heard of another RPG that did.... |
| 02:46.56 | russellb | tyman: that's one potential use case I suppose |
| 02:47.06 | tyman | adhearsion is really cool...need to get way into that |
| 02:47.13 | russellb | the trick of course is what are people using to call in to your call center ... |
| 02:47.14 | tyman | that's a money maker |
| 02:47.37 | tyman | yeah...zoiper doesnt have video yet right? |
| 02:47.38 | russellb | need better/easier clients |
| 02:47.43 | russellb | Jitsi |
| 02:47.55 | russellb | is what I use for video with Asterisk mostly |
| 02:48.19 | mocker | "Thank you for holding, you are now third in line." ⦠"You were eaten by a grue, please call again later." |
| 02:48.21 | tyman | wow...is that new? |
| 02:48.32 | tyman | lol |
| 02:48.40 | russellb | Jitsi? no ... has been around for a really long time, recently changed names from SIP Communicator |
| 02:48.47 | tyman | oh...ok |
| 02:48.55 | tyman | heard of that... |
| 02:49.24 | tyman | v10 asterisk only right? |
| 02:49.32 | russellb | for conferencing anyway, yeah |
| 02:49.55 | tyman | that'll be huge...looking to get right MSkype |
| 02:50.03 | tyman | s/right/rid of/ |
| 02:50.16 | p3nguin | I was patching chan_sip on a production system to add the description field, and I didn't want to have to rebuild everything. Using make rebuilt only chan_sip.so for me and everything worked fine. |
| 02:50.35 | russellb | yay. |
| 02:50.52 | p3nguin | quite |
| 02:51.04 | mocker | p3nguin: We need some release candidates for 1.8.8 :P |
| 02:52.24 | tyman | been googling a bit on redundant itsp and multiple servers....I want to make sure I understand this after today's devasting linode outages |
| 02:52.32 | russellb | I bet that won't happen until after astricon |
| 02:53.57 | tyman | so, how do inbound calls load balance when did's are provided by itsps? In BGP land we'd need to own the IP addrs and peer upstream. What's the DID equiv of this? |
| 02:55.28 | tyman | After this (http://status.linode.com/) continuing disaster, I need to migrate tonight...no end to this |
| 02:55.46 | russellb | TO THE CLOUD |
| 02:56.02 | atan | Anyone here try Bria 3? Is it worth the cost? :-) |
| 02:56.23 | russellb | atan: all softphones suck, it's just which one sucks least |
| 02:56.29 | tyman | i run bria 3 on osx and secreataries on win... |
| 02:56.33 | mocker | ~softphone |
| 02:56.34 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 02:56.36 | tyman | russellb: exactly |
| 02:56.46 | russellb | finds the free ones sufficient ... |
| 02:56.56 | russellb | Jitsi not on that list? sad |
| 02:57.10 | russellb | Jitsi and blink! |
| 02:57.16 | russellb | my two favorites aren't on there |
| 02:57.48 | tyman | bot update... |
| 02:58.11 | russellb | i figured i'd just complain and maybe someone else would update it |
| 02:58.28 | mocker | ~managermaterial |
| 02:58.37 | russellb | :-) |
| 02:58.48 | russellb | i was a manager for a little over 2 years ... |
| 02:59.14 | mocker | You probably managed programmers? |
| 02:59.18 | russellb | yup. |
| 02:59.27 | mocker | I could never do that. |
| 02:59.42 | tyman | russellb: I met you at digium early this year |
| 02:59.44 | russellb | the team was awesome |
| 02:59.50 | russellb | tyman: oh, cool :-) |
| 03:00.09 | tyman | pretty laid back office |
| 03:00.14 | tyman | seemingly |
| 03:00.20 | tyman | good for dev |
| 03:00.52 | tyman | although home is like the only place I can code well |
| 03:02.04 | russellb | airplanes used to be where i'd get the best code written ... until they added wifi |
| 03:02.10 | tyman | anyone use svwh.net for hosting? |
| 03:02.11 | tyman | lol |
| 03:02.33 | russellb | and i don't really travel that much anymore anyway |
| 03:03.21 | tyman | russellb: we talked about asterisk still using svn, when we met, and I mentioned that was my favorite part about git |
| 03:03.40 | russellb | i love git. |
| 03:03.52 | russellb | someone at digium actually has all the repo conversions working |
| 03:03.53 | tyman | by able to branch, commit, merge, etc...all offline...it's beautiful |
| 03:04.13 | russellb | now it's a matter of spending time figuring out all the tools and docs that the migration will break |
| 03:04.35 | tyman | i used to love svn (over cvs) but now it just seems...clunky and slow |
| 03:04.42 | russellb | yup |
| 03:05.03 | russellb | do you use github? |
| 03:05.17 | russellb | github is the hotness |
| 03:05.24 | mocker | My problem is I love github, but want to host an internal git repository. |
| 03:05.34 | mocker | And all the web interfaces to git stink compared to github. :P |
| 03:05.46 | russellb | well you can get a paid plan and have private repos on there |
| 03:05.56 | russellb | it's not _that_ expensive if you really have a reason to keep something private |
| 03:05.57 | tyman | yes...for client shared things...but host own over ssh for the tons of small repos that I have |
| 03:06.11 | russellb | ah, for tons of repos, it gets expensive |
| 03:06.13 | mocker | russellb: "cloud" private |
| 03:06.24 | russellb | they'll install github on your own infrastructure for some insane amount of cash, heh |
| 03:06.48 | tyman | they're good peeps, github |
| 03:07.00 | tyman | talk to them at railsconf each year |
| 03:07.07 | mocker | russellb: Most problems can be overcome with insane amounts of cash! |
| 03:07.12 | russellb | indeed |
| 03:07.48 | tyman | any good links to redundant itsp info? |
| 03:08.48 | p3nguin | autofsckk: Do you want that now or do you want to wait until you're ready to try Asterisk on Arch? |
| 03:11.17 | autofsckk | p3nguin: i changed plans, i think i will try in arch from start, i have it installed version 1.8.7.0 |
| 03:11.30 | russellb | tyman: for inbound? depends on what your ITSP supports. |
| 03:11.35 | p3nguin | Hmm. Mine's better. |
| 03:11.52 | p3nguin | But that's fine. I'll keep it for myself. |
| 03:12.04 | autofsckk | p3nguin: ok great, cn you give it to me please? |
| 03:12.09 | mocker | p3nguin: Yours in AUR? |
| 03:12.22 | p3nguin | No, someone else maintains one in AUR. |
| 03:12.26 | tyman | inbound is my big question...not sure how that would work with DIDs... |
| 03:12.40 | russellb | tyman: if only pacemaker-cloud.org was ready to be used :-) |
| 03:12.46 | p3nguin | autofsckk: Check your notices. |
| 03:12.52 | p3nguin | mocker: You want mine, too? |
| 03:13.03 | mocker | Sure. |
| 03:13.11 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.81.14) |
| 03:13.11 | p3nguin | mocker: Check your notices. |
| 03:13.12 | autofsckk | ok thanks |
| 03:13.14 | mocker | I looked at the main AUR one and couldn't see myself using it. |
| 03:13.46 | p3nguin | I've made a few changes that I feel are important. |
| 03:14.00 | p3nguin | Including the patch for sip peer descriptions. |
| 03:14.08 | russellb | bedtime. g'night folks |
| 03:14.09 | p3nguin | (which is only available in asterisk 10) |
| 03:14.23 | tyman | nite russellb |
| 03:14.31 | autofsckk | night russellb thanks for the tips |
| 03:14.39 | russellb | good luck to ya |
| 03:14.46 | mocker | p3nguin: Thanks. |
| 03:15.40 | p3nguin | I admit that I haven't looked at the one in AUR for a while. |
| 03:15.49 | mocker | tyman: When you say redundant itsp are you talking about one ITSP with redundancy, or trying to use multiple ITSPs for one DID? |
| 03:16.04 | tyman | the latter |
| 03:16.15 | mocker | tyman: Good luck with that. :) |
| 03:16.21 | tyman | yeah...thought so |
| 03:16.41 | tyman | ok...that was quick...makes sense though... |
| 03:17.20 | tyman | for outbound, just mod the dialplan? ... checking a channel var of some sort for connection status? |
| 03:17.59 | mocker | Yeah, or fall through for dial status.. |
| 03:18.28 | tyman | can u elaborate just a touch? What status in particular do i look for? |
| 03:18.38 | p3nguin | Oh, I see something I should have commented in my PKGBUILD. I added a line in it to provide ilbc support, which is not available by default. |
| 03:20.47 | mocker | tyman: Hmm, I would probably just try flowing in the dialplan from provider A to provider B |
| 03:20.56 | mocker | Not sure if it would work but it sounds good in my head. |
| 03:21.16 | mocker | if A doesn't succeed the next dialplan command would be to use B |
| 03:21.38 | tyman | ok...didnt know if there was official best practice...that had me worried much less than inbound |
| 03:21.41 | p3nguin | You can use ChanIsAvail() or just dial sequencially. |
| 03:21.56 | p3nguin | s/sequencially/sequentially/ |
| 03:22.08 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:22.50 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-129-148.chyn.qwest.net) |
| 03:22.50 | mocker | tyman: All PRIs and DS3s in my world, no SIP trunks. |
| 03:22.59 | mocker | So just guessing. |
| 03:23.43 | tyman | never thought i'd say this, but wishing for PRIs today... |
| 03:24.14 | tyman | clients DID inbound literally falling off the planet today was NOT making people happy |
| 03:25.21 | tyman | what about config sync....just have "slave" server accept rsync of all /etc/asterisk? excluding, of course, the uniq IP related info? |
| 03:25.40 | tyman | and register the phones against both servers? |
| 03:26.09 | mocker | Don't know what you are trying to accomplish.. |
| 03:26.10 | p3nguin | How do you propose that would work? |
| 03:26.50 | mocker | If it's just HA I'd look at a cluster w/ DRBD or something probably. |
| 03:26.50 | tyman | I'm trying to have two different servers in two different datacenters with two different itsps |
| 03:27.22 | tyman | client handsets registering to both servers |
| 03:27.34 | tyman | today the entire Datacenter went offline |
| 03:27.35 | p3nguin | How do you propose THAT would work? |
| 03:27.37 | tyman | the whole thing |
| 03:27.42 | tyman | dunno |
| 03:27.49 | tyman | what are my options? |
| 03:28.00 | mocker | tyman: The registration thing could be done w/ DNS and a short TTL, the multiple ITSP thing is another issue. |
| 03:28.23 | tyman | will the phone not allow configuration of multiple servers |
| 03:28.25 | p3nguin | If your ITSP is offline, you're done. |
| 03:28.48 | mocker | I would instead choose a *better* ITSP that will let you have a failover server. |
| 03:29.11 | p3nguin | Some phones will allow "backup server" configuration, but I wouldn't rely on that always being the case. |
| 03:29.17 | tyman | what protocol does the failover server use? |
| 03:29.36 | tyman | polycoms do, I believe...that's what we're using exclusively |
| 03:30.44 | mocker | Yeah, the phone shouldn't be picking which server is primary imho. |
| 03:30.59 | p3nguin | I'd rather have working SRV. |
| 03:32.05 | mocker | Ok, bedtime. |
| 03:32.22 | mocker | g'night everyone. |
| 03:33.17 | tyman | nite mocker |
| 03:33.19 | tyman | thx |
| 03:35.07 | tyman | p3nguin: so, have a server with effectively same config and work to automate dns record update so phones register unknowingly is best practice? |
| 03:35.26 | tyman | inbound being a lost cause of course |
| 03:35.49 | p3nguin | I'd prefer that the phone has no idea that anything changed. |
| 03:35.59 | tyman | y, understandable |
| 03:36.19 | tyman | but that's the gist of what you're driving at right? |
| 03:36.43 | p3nguin | If you're setting up multiple asterisks, maybe dundi is something that could be useful to you. |
| 03:36.46 | p3nguin | ~dundi |
| 03:36.47 | infobot | i guess dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 03:44.15 | *** join/#asterisk rafiyr (~rafi@hex.seas.UPENN.EDU) |
| 03:50.35 | Kobaz | [2011-10-03 23:49:59] WARNING[14276]: channel.c:1451 ast_best_codec: Don't know any of 0x0 (nothing) formats |
| 03:50.51 | Kobaz | anyone know what that means? |
| 03:50.52 | p3nguin | Yeah, chan_sccp is a bother. |
| 03:51.19 | Kobaz | is sccp-b getting stable yet against ast 1.8 |
| 03:51.57 | Kobaz | last time i tried it it was a crashy disaster |
| 03:52.00 | p3nguin | Maybe you can get Marcello going on this. https://sourceforge.net/tracker/?func=detail&aid=3417248&group_id=186378&atid=917045 |
| 03:52.12 | p3nguin | He's being a dick to me over it. |
| 03:52.29 | Kobaz | mm |
| 03:52.34 | Kobaz | it's not just chan_sccp |
| 03:52.36 | Kobaz | i get that with sip too |
| 03:52.41 | p3nguin | Oh? |
| 03:52.50 | Kobaz | yeah it's an asterisk problm |
| 03:52.54 | Kobaz | not sccp |
| 03:52.55 | p3nguin | Now that I did not know. |
| 03:53.17 | p3nguin | For me, it didn't start showing up until I upgraded my sccp. |
| 03:53.19 | Kobaz | got introduced somewhere in 1.8 it seems |
| 03:53.31 | p3nguin | I run 1.8.7.0. |
| 03:53.41 | Kobaz | yeah |
| 03:53.44 | Kobaz | i didn't see it in early 1.8 |
| 03:53.55 | p3nguin | It wasn't there on one sccp build, then was on another. |
| 03:54.01 | p3nguin | not changing asterisk. |
| 03:54.28 | Kobaz | then it's a change in sccp that triggered the bug in asterisk |
| 03:56.32 | p3nguin | Regardless, he was still being a dick to me over it. I filed the bug, and he simply closed it. No comments, no messages, nothing; just closed it. |
| 03:57.07 | p3nguin | I had to send him a message to tell him to open it back up. Then he starts off with the "Not enough information!" statement. |
| 03:57.11 | p3nguin | Like that's helpful. |
| 03:57.31 | Kobaz | well, it is really difficult to debug something like that |
| 03:57.35 | Kobaz | it is really not enough information |
| 03:58.01 | p3nguin | But does that really warrant closing it without at least trying? |
| 03:58.16 | p3nguin | I certainly would not have done that to someone. |
| 03:58.43 | Kobaz | if i was marshaling it, i would mark it needs feedback but not close it |
| 03:58.51 | Kobaz | since it's legitimately a bug in something |
| 03:59.06 | p3nguin | With it closed, I couldn't even comment on it to give more info if I wanted to. |
| 04:02.40 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 04:04.41 | coppice | "not enough information" is the standard response from people who can't be bothered improving stuff. you get it all the time from a number of places |
| 04:07.41 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.81.14) |
| 04:08.11 | p3nguin | If I don't provide enough information, it's because I either don't have the information or can get it but don't know where to get it from. |
| 04:08.48 | Kobaz | yeah |
| 04:09.18 | Kobaz | i don't know what causes it yet otherwise i would post a bug |
| 04:09.25 | Kobaz | it's not very consistent |
| 04:10.04 | p3nguin | Can you tell me the precise conditions in which is does happen? Is there any debug or verbose info to go along with it? |
| 04:10.47 | Kobaz | i have no idea. all i know is I see it happen *sometimes* when i run my unit tests |
| 04:12.19 | p3nguin | For me, pressing a line key, lifting the handset, or dialing a number and pressing the dial key makes it appear. |
| 04:14.42 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.81.14) |
| 04:15.45 | p3nguin | I'd guess that may not be the case for SIP. |
| 04:17.59 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 04:21.27 | Kobaz | it's upon making a new call but not every new call |
| 04:22.03 | Kobaz | seems to be after an Answer() |
| 04:26.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:28.30 | SeRi | p3nguin, did it come in? |
| 04:28.37 | p3nguin | Negative. |
| 04:29.46 | SeRi | fuck. ok ill be patience :) |
| 04:30.32 | p3nguin | I would have expected arrival on the 3rd day. |
| 04:30.55 | p3nguin | USPS must be really backed up this week. |
| 04:30.57 | SeRi | me too |
| 04:31.07 | SeRi | that would have been monday |
| 04:31.23 | p3nguin | Maybe tomorrow! |
| 04:31.33 | SeRi | p3nguin, can you give me a quick hand with a blacklist? |
| 04:31.34 | p3nguin | I'll check again tomorrow. |
| 04:31.44 | SeRi | ok cool |
| 04:31.52 | p3nguin | Probably... although I have no idea what we are blacklisting. |
| 04:32.20 | SeRi | ok I am blacklisting callers that I dont want from my cdr |
| 04:32.38 | SeRi | so I created a black list db called blacklist |
| 04:33.20 | SeRi | I will do a pastebin |
| 04:34.38 | p3nguin | Are you talking about using BLACKLIST()? |
| 04:34.45 | SeRi | yes sr |
| 04:34.53 | p3nguin | Oh, that's simple, then. |
| 04:35.15 | SeRi | :D cool I am sort of lost. here is what I got one sec |
| 04:36.49 | [TK]D-Fender | Pretty terrible app... nothing you can't do better yourself with your own dialplan |
| 04:38.31 | p3nguin | I find it's easier to manage things from the db when you have a static extensions.conf. |
| 04:38.50 | p3nguin | It's easier to add and remove things on the fly, too. |
| 04:39.06 | p3nguin | For example, I have an extension to add the last caller to the blacklist. |
| 04:39.36 | p3nguin | It creates the key in the database. I couldn't do that easily in extensions.conf. |
| 04:41.29 | SeRi | p3nguin, http://pastebin.com/tQm4DRaG |
| 04:41.39 | SeRi | I know you are going to scream at me :) |
| 04:42.53 | p3nguin | I'll just say that you've used BLACKLIST() correctly. |
| 04:43.49 | SeRi | ok. can you say if it will work? |
| 04:44.01 | p3nguin | There are several broken aspects of that paste, so no, that dial plan will not work. |
| 04:44.35 | SeRi | ok. can you give me a hand? :) |
| 04:44.51 | p3nguin | Do you want me to make it the way I would have written it? |
| 04:45.10 | SeRi | sure. |
| 04:46.13 | SeRi | I am trying to use zapteller if that make sense... |
| 04:47.02 | [TK]D-Fender | It is broken.... |
| 04:47.12 | [TK]D-Fender | exten => 8325855209,1,GotoIf($[${BLACKLIST()]?blocked,1) |
| 04:47.21 | [TK]D-Fender | exten => 8325855209,1,Dial(SIP/1003,20) ;your DID |
| 04:47.30 | [TK]D-Fender | 2 x priority #1 |
| 04:47.52 | [TK]D-Fender | You just overrode your attempt to cal Blacklist |
| 04:47.57 | p3nguin | http://pastebin.com/BLLxTanU |
| 04:48.56 | p3nguin | I didn't stray too far from your design. |
| 04:49.13 | SeRi | [TK]D-Fender, Yeap I saw my n00b mistake. Thanks for the point out. :) |
| 04:49.22 | SeRi | p3nguin, well thats more elegant. |
| 04:50.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 04:51.50 | SeRi | p3nguin, ok so include => fax-in does not need the detect for fax correct? because the fax module is listening for a fax tone anyways correct? |
| 04:52.03 | *** join/#asterisk brdude (~brdude@12.200.180.147) |
| 04:52.19 | p3nguin | Your question doesn't make sense to me. |
| 04:52.29 | SeRi | lol. I hear that a lot. |
| 04:52.40 | p3nguin | All I'm doing is moving your fax extension to what I feel is a more appropriate context. |
| 04:53.05 | SeRi | you took out exten => fax,1,Gosub(fax-rx,s,1) |
| 04:53.05 | SeRi | exten => fax,2,Hangup() |
| 04:53.15 | p3nguin | You still have to have your faxdetect enabled in sip.conf, and you still have to have the fax extension in fax-in. |
| 04:53.26 | p3nguin | And what's with the numbered priorities? |
| 04:53.55 | p3nguin | Hang on. I'll change it again. |
| 04:54.48 | SeRi | my 713 number is the number where all faxes come in |
| 04:56.54 | p3nguin | http://pastebin.com/2cvXv82f |
| 04:57.01 | p3nguin | Then you don't need fax detect at all. |
| 04:57.15 | p3nguin | And that needs changed again. |
| 04:57.33 | p3nguin | Will you EVER have voice calls on your fax number? |
| 04:57.54 | SeRi | no sr |
| 04:58.17 | p3nguin | Why do you have it Dial()ing a phone, then? |
| 04:58.33 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 04:59.21 | SeRi | I thought I had to have it ring before fax detect will pick it up |
| 04:59.57 | p3nguin | Do you ever get faxes on the other number, ever? |
| 05:00.10 | SeRi | no sr. |
| 05:00.12 | p3nguin | Then you do not need fax detect at all. |
| 05:00.20 | SeRi | well its an email base |
| 05:00.26 | SeRi | there is no fax machine |
| 05:00.34 | SeRi | it emails me the faxes |
| 05:01.10 | p3nguin | If the 713 number is only for fax, here's how I'd do it: http://pastebin.com/spbC6V8J |
| 05:02.02 | p3nguin | Well, sorta... there's a stale bit in there that I needed to take out. |
| 05:02.45 | SeRi | well I have the fax in working ok fir me so Ill just use your plan |
| 05:03.11 | p3nguin | Here's a bug fix: http://pastebin.com/typExZ1f |
| 05:03.54 | p3nguin | I had it attaching a recording file that didn't exist. |
| 05:03.57 | p3nguin | But I removed it. |
| 05:06.45 | SeRi | p3nguin, do I need to define my database blacklist? |
| 05:07.10 | p3nguin | You have to add stuff to the db to make it work. |
| 05:07.25 | SeRi | ok got it. working on it now. testing. |
| 05:07.28 | p3nguin | database add blacklist <number to blacklist> 1 |
| 05:07.55 | p3nguin | You just have to have the key existing to make it work. |
| 05:13.51 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 05:16.52 | SeRi | damn dns server |
| 05:20.31 | p3nguin | I'm having some issues, too. Damn cable companies like to work on shit right after midnight. |
| 05:20.51 | p3nguin | They think people don't get affected at that time of day or something. |
| 05:21.10 | SeRi | no shit! |
| 05:21.19 | p3nguin | Maybe the number of affected people is less. I don't know. |
| 05:21.45 | SeRi | more like "nobody will notice" |
| 05:21.50 | SeRi | at least they think* |
| 05:21.57 | p3nguin | 4 AM would probably be better. |
| 05:22.06 | SeRi | well I have to say it works! |
| 05:22.11 | SeRi | yeap same here |
| 05:22.34 | p3nguin | By 4 AM, people who stay up late are usually in bed, and people who get up early aren't up yet. |
| 05:22.34 | SeRi | now I have to disable voip.ms vmail. |
| 05:22.35 | SeRi | lol |
| 05:22.45 | SeRi | voip.ms sent the call to the internal vmail |
| 05:23.20 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:23.33 | SeRi | p3nguin, I found some error by the dial plan |
| 05:23.38 | p3nguin | okay |
| 05:24.01 | p3nguin | I sometimes create bugs. |
| 05:24.07 | SeRi | http://pastebin.com/zZvqJHjK |
| 05:24.11 | p3nguin | It keeps people on their toes. |
| 05:24.37 | SeRi | :) |
| 05:24.38 | SeRi | lol |
| 05:25.34 | p3nguin | GotoIf($[${BLACKLIST()]?blacklist,s,1) |
| 05:25.40 | p3nguin | GotoIf($[${BLACKLIST()}]?blacklist,s,1) |
| 05:25.58 | p3nguin | That's your bug, though, not mine. |
| 05:26.22 | SeRi | ah! |
| 05:26.23 | SeRi | LOL |
| 05:26.34 | p3nguin | You also forgot to create your extension s in context blacklist. |
| 05:27.21 | SeRi | ok thats where I got lost last time. defining "s" |
| 05:27.59 | p3nguin | exten => s,1,Stuff() |
| 05:28.34 | p3nguin | I actually send my blacklisted numbers to misc,blocked,1 rather than blacklist,s,1 |
| 05:28.38 | p3nguin | exten => blocked,1,Verbose(Blocked call); |
| 05:28.40 | p3nguin | same => n,Playback(silence/1&privacy-you-are-blacklisted&vm-goodbye); |
| 05:28.42 | p3nguin | same => n,Hangup(); |
| 05:29.25 | SeRi | ok my intention is to use zapteller |
| 05:29.49 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:30.22 | p3nguin | I think the cableco is jacking my connection around. |
| 05:30.29 | SeRi | same here |
| 05:30.37 | SeRi | I keep dropping dns |
| 05:30.47 | SeRi | exten => s,1,Zapateller |
| 05:30.55 | SeRi | is that correct? |
| 05:31.14 | p3nguin | I Might work, but what are you going to do after that? |
| 05:31.25 | p3nguin | s/I M/It m/ |
| 05:31.39 | SeRi | exten => s,n,hangup |
| 05:31.48 | SeRi | () |
| 05:31.50 | SeRi | ops :P |
| 05:32.03 | p3nguin | I don't quite understand that concept, but give it a try. |
| 05:32.21 | p3nguin | This is really pissing me off. I don't think it's a DNS problem for me. |
| 05:32.40 | p3nguin | I think they are literally doing something to my connection (such as a tap or something). |
| 05:32.56 | SeRi | damn. |
| 05:33.08 | SeRi | fire up wireshark |
| 05:33.17 | p3nguin | Trying to send an email, progress shoots to 93% and then hangs until it times out. |
| 05:33.35 | p3nguin | It says it couldn't reach the server. But nc can reach it just fine. |
| 05:34.23 | p3nguin | drift just got killed, you're having issues, and I'm having issues... what do comcast and charter share that's fucked up? |
| 05:34.35 | p3nguin | And I guess firestart is on comcast, too. |
| 05:35.12 | [TK]D-Fender | Checkout time, later all |
| 05:35.29 | p3nguin | Trying to edit my pastebin, pastebin.com keeps trying to get me to save edit.php to my computer. |
| 05:36.23 | *** join/#asterisk Foxi352_work (~quassel@213.135.228.202) |
| 05:36.58 | SeRi | p3nguin, http://pastebin.com/rvdVWP45 |
| 05:37.25 | SeRi | I have that issue some times with chrome. It does not render php properly |
| 05:38.00 | p3nguin | It might load for me in a while. |
| 05:38.54 | SeRi | I am going to paste it here hold on |
| 05:38.57 | SeRi | one by one |
| 05:39.12 | SeRi | exten => s,1,Zapteller |
| 05:39.22 | SeRi | exten => s,n,Playback(sit) |
| 05:39.29 | SeRi | exten => s,n,Playback(ss-noservice) |
| 05:39.38 | SeRi | exten => s,n,Hangup |
| 05:39.43 | *** join/#asterisk dirkD (~dirk@84-245-20-6.dsl.cambrium.nl) |
| 05:41.08 | SeRi | I have sit.wav |
| 05:45.02 | Foxi352_work | Hi fellow dev's. I am a dev of another large OSS project where we rely on asterisk for our telecom needs. We choosed to go over to realtime db and face the apparently well known (> 1 year) problem of asterisk-addons in Lucid 10.04LTS repo's not in sync with asterisk 1.6 also in Lucid repo's. Can someone tell me if this will ever get fixed, or do we need to ship our own replacement package ? |
| 05:49.22 | p3nguin | Playtones(info) ? |
| 05:49.45 | ChannelZ | It takes like 5 minutes to build Asterisk from source, why torture yourself? |
| 05:49.46 | SeRi | Foxi352_work, wouldn't that be a more of a request to the ubuntu upstream chan maintainers? |
| 05:51.10 | Foxi352_work | ChannelZ: We rely on a lot of external projects and try to rely on LTS repo's as much as possible. We try to avoid building everything ourselves and maintaining own packages as much as possible .. |
| 05:52.01 | Foxi352_work | SeRi: Probably ... |
| 05:52.21 | Foxi352_work | You know who maintains the asterisk packages ? |
| 05:53.01 | SeRi | Foxi352_work, go to the #ubuntu chan and ask. I have not clue about ubuntu sorry. slackware user here :) |
| 05:53.42 | Foxi352_work | Hehe ... i thought it was the asterisk project maintaining the packages.... will try in ubuntu channel. Thanks for the hint ... |
| 05:53.52 | ChannelZ | They have their own packages |
| 05:54.01 | ChannelZ | Ubuntu/etc do their own thing |
| 05:54.17 | Foxi352_work | ok. Will try to shake them .... :-) |
| 05:54.33 | WIMPy | Foxi352_work: You are area that 1.6 os out of support? |
| 05:54.34 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 05:54.49 | ChannelZ | you might not get much action for an update in LTS |
| 05:55.07 | WIMPy | And 1.8 has addons included. |
| 06:00.07 | SeRi | p3nguin, it worked :P |
| 06:00.17 | SeRi | I got it to do what I wanted to do :) |
| 06:00.52 | SeRi | p3nguin, Thanks for the help. |
| 06:01.19 | irroot | morning peeps |
| 06:04.02 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
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| 07:08.40 | *** join/#asterisk toresbe (~toresbe@2001:1ad8:2002:5:f00d:beef:cafe:babe) |
| 07:08.45 | toresbe | hey folks |
| 07:09.41 | toresbe | I just signed up for a VoIP PSTN gateway , and one of the things I would like to do is set up SMS messaging over SIP |
| 07:10.11 | toresbe | is this possible? If I send an SMS to Asterisk, -- hang on, I'll post the log |
| 07:12.18 | toresbe | http://simula.gunkies.org/~toresbe/asterisk-sms.txt |
| 07:12.25 | toresbe | that's what happens. |
| 07:13.15 | toresbe | Does anybody understand what's going on there? :) |
| 07:17.25 | singler | I think that thing happens periodically :) asterisk uses options to "ping" peers |
| 07:17.46 | wdoekes2 | in this case it looks like the registrar is doing the same |
| 07:18.03 | wdoekes2 | s/registrar/pstn-gateway/ |
| 07:18.12 | singler | but if it really happens only when you send sms, it may be that providers asks for asterisk capabilities |
| 07:18.36 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2c74:de81:da23:593c) |
| 07:18.38 | singler | s/providers/provider |
| 07:19.37 | wdoekes2 | hm.. that sounds plausible.. I didn't see that he was sending *to* the gateway |
| 07:19.53 | wdoekes2 | in that case asterisk is replying that it doesn't to MESSAGE, so that might be a problem |
| 07:20.57 | wdoekes2 | toresbe: for starters, you'll want to use 1.8. 1.6 doesn't do message at all. |
| 07:21.05 | wdoekes2 | download the latest and compile it manually |
| 07:26.27 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:29.05 | irroot | First National Bank www.fnb.co.za here in south africa uses VOIP and uses DTMF pin auth for there call center the pin from your credit card security fail |
| 07:29.48 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:32.50 | *** join/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 07:33.02 | toresbe | wdoekes2: ok, thanks |
| 07:33.51 | toresbe | wait, what... why is there a jump from 1.8 to 10.0? |
| 07:34.28 | zealiod | Can any one please tell me why when I dial 205 nothing happens⦠what's wrong with my dial plan? http://pastebin.com/gf4fm9Fg |
| 07:36.56 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 07:37.48 | singler | toresbe: for the lols :) FF pushes numbers quickly, Chrome also, so asterisk just jumped. For serious answer read this: http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 07:38.54 | irroot | toresbe 10 = 2 :P |
| 07:39.03 | toresbe | singler: ah, thanks |
| 07:39.13 | toresbe | that does make sense, actually. |
| 07:39.19 | singler | np |
| 07:39.37 | singler | irroot: 10 = 1.10 |
| 07:40.00 | irroot | 10 = X |
| 07:40.20 | singler | if it were 2, it would be named 2, but because they probably will never reach 2, they dropped 1. part |
| 07:41.56 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 08:01.42 | zealiod | It seems to be that WaitExten is just timing out⦠not hearing my DTMF tones |
| 08:01.52 | zealiod | can anyone help me> |
| 08:02.52 | irroot | zealiod mmm you need to do a debug on the channel and make sure the core sees the DTMF |
| 08:03.03 | zealiod | irroot: how do i do that? |
| 08:03.09 | irroot | SIP ?? |
| 08:03.11 | zealiod | oh wait - i know |
| 08:03.16 | zealiod | yeah core sip debug right> |
| 08:03.25 | irroot | that is a start |
| 08:03.46 | irroot | also make sure you using the right DTMF options Inband/RFC/info |
| 08:03.56 | irroot | and that the peer setting and phone setting match |
| 08:04.05 | zealiod | hmm |
| 08:04.12 | zealiod | not sure how to do any of that :) |
| 08:04.17 | zealiod | Im debugging now |
| 08:04.31 | irroot | sip.conf and the phone setup depends on phone |
| 08:04.56 | zealiod | OK, when I press buttons nothing changes on the debug output |
| 08:05.17 | zealiod | So, my set up is, I'm calling in on a standard phone through a sip gate number |
| 08:05.26 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:05.34 | zealiod | standard dtmf on the phone |
| 08:12.26 | zealiod | irroot: OK, I've set dtmfmode = rfc2833 in my sip.conf |
| 08:12.30 | zealiod | no change... |
| 08:12.35 | zealiod | Hmm |
| 08:13.08 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:18.21 | irroot | zealiod and its set like that on phone ? |
| 08:18.36 | irroot | you need to reload sip to make the config activee ?? |
| 08:23.10 | *** join/#asterisk JohnTeddy (unstable@tor/regular/sid) |
| 08:24.01 | JohnTeddy | I'm in Shanghai... (though from NY). I need to setup a tech support phone system that is reliable for a small 3 person tech support team. Does anyone have any all in one product solutions we can buy to get this? (I'm assuming with asterisk on it). |
| 08:24.21 | JohnTeddy | Is there a software set for this kind of thing... that will record calls, and keep logs, those kinds of things |
| 08:24.44 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 08:26.49 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:27.50 | toresbe | Hurh. |
| 08:28.02 | toresbe | Asterisk simply crashes on IA64. |
| 08:28.13 | toresbe | Perhaps something some developer might want to take a look at. |
| 08:28.25 | toresbe | that is, 10beta1 |
| 08:29.05 | irroot | toresbe you can post a bt on a JIRA |
| 08:29.08 | wdoekes2 | I would, if I had an ia64.. please upload backtraces of the crash |
| 08:30.59 | toresbe | I'm not that skilled. Can I offer a user account on my machine? |
| 08:31.27 | toresbe | I keep this Itanium as my hobby machine. It's been non-IA32 my whole life; I have a nostalgic attitude towards that fact ;) |
| 08:33.44 | wdoekes2 | I have a couple of minutes to spare today, pm me with details |
| 08:35.58 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:36.00 | schmidts | good morning |
| 08:38.27 | toresbe | (can people here in general vouch for wdoekes2 being a contributor?) |
| 08:39.36 | zealiod | I can not get my extensions to goto the beginning⦠I'm using exten => i,1,Playback(pbx-invalid) |
| 08:39.37 | zealiod | exten => i,n,Goto(default, 1375387, restart) |
| 08:39.49 | zealiod | if I do this the call does the error then just ends |
| 08:39.59 | zealiod | what am i missing? |
| 08:45.06 | *** join/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 08:48.30 | kaldemar | zealiod: do you have exten => 1375387,n(restart) ? |
| 08:48.53 | zealiod | kaldemar: yes, I have "exten => 1375387,n(restart),BackGround(enter-ext-of-person)" |
| 08:49.26 | kaldemar | zealiod: in default context? |
| 08:49.31 | zealiod | yes |
| 08:49.42 | wdoekes2 | toresbe: you can find me in the ChangeLog as "Walter Doekes".. I don't have commit powers though |
| 08:50.14 | wdoekes2 | (or wdoekes, without the 2) |
| 08:51.04 | kaldemar | zealiod: remove the spaces in the Goto |
| 08:52.03 | toresbe | wdoekes2: fair enough :) |
| 08:52.34 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 08:53.23 | zealiod | kaldemar: ok, done that - but still not working |
| 08:53.53 | zealiod | my mistake - just needed to reload |
| 08:53.56 | zealiod | all good now! |
| 08:56.02 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:56.05 | jacc0 | hi all |
| 08:56.21 | *** part/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 09:01.29 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:05.19 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:19.33 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 09:28.38 | *** join/#asterisk dirkD (~dirk@84-245-20-6.dsl.cambrium.nl) |
| 09:32.53 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 09:33.18 | *** part/#asterisk rafiyr (~rafi@hex.seas.UPENN.EDU) |
| 09:36.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:43.34 | *** join/#asterisk DND (~ctcarino@94.200.7.26) |
| 09:44.00 | DND | guys i need help, our asterisk machine consumes 100% cpu usag |
| 09:44.01 | DND | e |
| 09:44.13 | DND | i check top command and it just says asterisk |
| 09:44.36 | DND | how can i know which specific inside asterisk is consuming cpu? |
| 09:45.31 | jacc0 | I had the same thing yesterday; that server hat been running without problems for months\ |
| 09:46.03 | jacc0 | @dnd: compile with compiler flags: |
| 09:46.35 | jacc0 | dont_optimize & Debug_threads |
| 09:47.21 | DND | sorry im using asterisknow |
| 09:47.24 | jacc0 | then use asterisk cli command: core show threads |
| 09:47.55 | DND | ok i have core show threads |
| 09:47.59 | jacc0 | ;) |
| 09:48.04 | jacc0 | then you are lucky |
| 09:48.12 | DND | so what do i need to check? |
| 09:48.55 | DND | i see asterisk.c, pbx.c, couple of app_mixmonitor.c, couple of chan_iax2.c |
| 09:49.16 | DND | total of 50 threads |
| 09:50.51 | DND | hello? |
| 09:51.09 | jacc0 | sorry, I'm also working; have a deadline |
| 09:51.16 | DND | oh i see sorry about that |
| 09:51.20 | jacc0 | and I'm hacking my goverment in the same time |
| 09:51.27 | jacc0 | :P |
| 09:52.30 | irroot | jacc0 that no challenge dude :P |
| 09:52.41 | jacc0 | I know |
| 09:53.04 | jacc0 | I make it in the news at least once a week |
| 09:53.07 | jacc0 | :P |
| 09:53.15 | DND | wikileaks eh? "D |
| 09:53.26 | jacc0 | but this month we promised to release a security hole eevery day |
| 09:53.30 | irroot | jacc0 just cause im nice i have sent a email to FNB re them using DTMF over SIP to callcentre to set / change pin codes |
| 09:54.33 | irroot | also when you call they ask you for last 4 digits of card and pin to validtate you totally stupid |
| 09:54.45 | jacc0 | hehehe lol |
| 10:03.29 | lkthomas | does asterisk work on virtualized environment ? |
| 10:03.35 | lkthomas | not sure if the T1 card work |
| 10:09.18 | SunTsu | lkthomas: of course asterisk does |
| 10:10.07 | lkthomas | PCI passthrough work with asterisk too ? |
| 10:11.02 | *** join/#asterisk Madkiss (~madkiss@freenode/staff-emeritus/madkiss) |
| 10:11.03 | Madkiss | hi folks |
| 10:12.20 | Madkiss | I have this in my dialplan: ChannelRedirect(${BRIDGEPEER},dynamic-nway,{CONFNO},1); |
| 10:12.35 | Madkiss | and that is what asterisk makes out of it: ChannelRedirect("SIP/gw_24_telgosiptrunk-0000006d", ",dynamic-nway,{CONFNO},1") |
| 10:12.38 | Madkiss | what am I doing wrong? |
| 10:12.39 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 10:12.50 | SunTsu | lkthomas: that's not asterisk specific, it's something your virtualization does. if it does it right then asterisk should have no problem using it |
| 10:15.09 | lkthomas | ok, thanks |
| 10:15.19 | Madkiss | i.e. why are the parameters passed to ChannelRedirect (this is ael) interpreted as one single parameter? |
| 10:15.27 | jacc0 | #Madkiss: you are missing a $ I guess : ${CONFNO} |
| 10:15.57 | Madkiss | jacc0: I noticed that right after pasting here and changed that, now I get: ChannelRedirect("SIP/gw_24_telgosiptrunk-0000006e", ",dynamic-nway,667,1") in new stack |
| 10:16.08 | Madkiss | and [Oct 4 12:13:14] WARNING[10264]: app_channelredirect.c:85 asyncgoto_exec: ChannelRedirect requires an argument (channel,[[context,]exten,]priority) |
| 10:16.31 | jacc0 | cut off the end of ${BRIDGEPEER} |
| 10:16.37 | jacc0 | it ends with : , " |
| 10:16.42 | jacc0 | that is giving the problems |
| 10:17.11 | jacc0 | because it results in an extra argument |
| 10:19.13 | jacc0 | Set(BRIDGEPEER=${CUT(BRIDGEPEER,\,,1)}) |
| 10:19.23 | jacc0 | or something like that |
| 10:19.36 | DND | guys any idea? i have restarted asterisk but im sure it will be just a couple of hours, the usage will be 100% again |
| 10:20.02 | jacc0 | yes |
| 10:20.12 | jacc0 | remove de -C from your init script |
| 10:20.33 | jacc0 | p[ast versions are known to give problems in color mode |
| 10:20.41 | jacc0 | problems like 100% cpu |
| 10:21.23 | jacc0 | running on debian/ubuntu? |
| 10:21.45 | jacc0 | https://issues.asterisk.org/view.php?id=16887 |
| 10:22.34 | DND | im using centos with asterisknow 1.5 asterisk version is 1.4.33.1 |
| 10:22.52 | jacc0 | not sure wath script you shoudl edit then |
| 10:22.59 | jacc0 | but I would try turning of color mode |
| 10:24.40 | jacc0 | "I removed the '-c' from the end /etc/init.d/asterisk line 85, and cpu utilization is now normal" |
| 10:27.12 | Madkiss | okay, this is frightening me |
| 10:27.23 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 10:28.59 | DND | i see |
| 10:29.07 | DND | ok will check on that |
| 10:29.49 | Madkiss | ouch. |
| 10:30.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 10:31.15 | DND | its kinda confusing because in other office it doesnt happen |
| 10:34.51 | DND | well i found safe_asterisk but it doesnt have the one with color |
| 10:35.40 | kaldemar | Madkiss: ",dynamic-nway,{CONFNO},1" starts with a comma. your BRIDGEPEER variable is empty. |
| 10:35.43 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 10:36.34 | DND | i only have this: http://pastebin.com/raw.php?i=rQzLY9Ep |
| 10:38.58 | jacc0 | @kaldemar: it holds |
| 10:38.58 | jacc0 | "SIP/gw_24_telgosiptrunk-0000006d", " |
| 10:39.47 | jacc0 | resulting in to much arguments |
| 10:40.05 | Madkiss | kaldemar: that'S what I am beginning to think, too, because "NoOp(${BRIDGEPEER})" results in plain nothing |
| 10:41.55 | kaldemar | jacc0: ChannelRedirect takes up to 4 arguments. "SIP/gw_24_telgosiptrunk-0000006e" is the channel that executes the command. |
| 10:42.19 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 10:43.35 | Madkiss | kaldemar: can I redirect that one by using ${CHANNEL}? |
| 10:43.42 | Madkiss | instead of $BRIDGEPEER? |
| 10:44.00 | teloniusz | hi. A simple echo application: Wait(2), Record(outfile-%d.gsm), Playback(${RECORDED_FILE}), Hangup() |
| 10:44.35 | teloniusz | but Record returns immediately, nothing specific with "core set verbose 10" |
| 10:44.45 | kaldemar | Madkiss: CHANNEL is the current channel, using redirect with yourself would not make much sense. |
| 10:44.58 | teloniusz | and nothing in error_log - how can I better debug this? |
| 10:45.10 | kaldemar | Madkiss: what do you expect BRIDGEPEER contents to be? i.e. what do you want to do? |
| 10:45.35 | jacc0 | @teloniusz: does the asterisk user have rights to write in the folder? try chmod 777 [foldername] and try again |
| 10:46.19 | kaldemar | teloniusz: have you seen app Echo? |
| 10:46.27 | teloniusz | jacc0: it seems so... |
| 10:46.37 | Madkiss | kaldemar: I want that somebody who dials the extension 667 is sent to a MeetMe room and that after that, asterisk dials another number and puts that number into the queue, too |
| 10:47.14 | toresbe | For what it's worth: I upgraded to 10-beta1 and I still got the same result: http://simula.gunkies.org/~toresbe/asterisk-sms.txt |
| 10:47.42 | kaldemar | Madkiss: what does your dialplan look like at the moment? |
| 10:47.43 | teloniusz | kaldemar: I'd like to do this on my own, I'm going to make sth more complicated later |
| 10:47.57 | kaldemar | teloniusz: ok, just checking... |
| 10:48.35 | teloniusz | jacc0: can I check which directory it is from the CLI? |
| 10:48.52 | Madkiss | kaldemar: I do not really have anything for that job. I was hoping that http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO could help me, but that looks like a dead end |
| 10:49.45 | toresbe | As far as I can tell, the problem persists? Asterisk 10 does not reply with MESSAGE in Allow:, and that's the problem, n'est sais pas? |
| 10:51.14 | wdoekes2 | toresbe: you can hack MESSAGE in and see if it does anything (it probably still won't work.. but you can see what else needs to be done) |
| 10:51.34 | toresbe | as in, write it in the source? |
| 10:51.39 | kaldemar | Madkiss: you could first do a dial to the another number and then use G() to put both parties in the conference. |
| 10:51.53 | kaldemar | Madkiss: G() being option G() of app Dial. |
| 10:53.01 | Madkiss | that sounds faboulus |
| 10:54.02 | wdoekes2 | toresbe: ~walter/*patch |
| 10:54.28 | Madkiss | kaldemar: can I do multiple Dial-Statements to dial numerous numbers sequentially (for the case that the first number doesn'T answer)? |
| 10:55.35 | kaldemar | Madkiss: sure. just set a timeout for Dial and put the next Dial in the following priority. |
| 10:57.01 | toresbe | wdoekes2: ha, that is quite ghetto :) Perhaps asterisk has always depended on the peers assuming message is supported? |
| 10:57.13 | wdoekes2 | well no.. it won't work actually |
| 10:57.24 | wdoekes2 | because asterisk will only accept message's in-dialog |
| 10:57.41 | wdoekes2 | but you should get a message from the provider, if it does what we expect |
| 10:57.41 | toresbe | no, see sip.conf option, er, um |
| 10:57.54 | kaldemar | 10 supports messages out of a dialog. |
| 10:58.01 | toresbe | accept_outofcall_message, defaults to yes |
| 10:58.10 | kaldemar | they need to be configured though. |
| 10:58.18 | wdoekes2 | oh.. that's what I get for not staying up to date |
| 10:58.45 | toresbe | As the guy with the IA64 I don't get to criticize others for not being up to date :) |
| 10:59.40 | toresbe | OK, now it's replying with Allow: Message... |
| 11:00.23 | *** join/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 11:00.27 | toresbe | ...that's weird. I guess it _was_ just the keepalive, and there is no SMS functionality. |
| 11:00.38 | zealiod | Can any one help me with the meaning of this error on the CLIâ¦.chan_sip.c:3707 sip_write: Asked to transmit frame type 8, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 11:00.48 | zealiod | It happens when a SIP call is in progress |
| 11:00.53 | zealiod | or calling.... |
| 11:01.47 | teloniusz | yeah, it works |
| 11:02.15 | teloniusz | if anyone's interested: my installation of Asterisk doesn't have any default recordings directory, it seems |
| 11:02.37 | teloniusz | I have to specify full path for it to work and there's no error message |
| 11:03.05 | teloniusz | Record(/usr/share/asterisk/sounds/recordings/outfile-%d.gsm) |
| 11:03.44 | Madkiss | kaldemar: is there any good pre-made solution to generate random numbers on the asterisk console? |
| 11:04.10 | kaldemar | Madkiss: asterisk console? |
| 11:04.20 | kaldemar | Madkiss: do you really mean CLI? |
| 11:05.37 | Madkiss | kaldemar: ahum. no. scripts, actually. |
| 11:05.50 | dym | Madkiss: theres rand() |
| 11:06.29 | kaldemar | Madkiss: if you want a random number in dialplan, you could consider func RAND. |
| 11:08.12 | Madkiss | hmkay |
| 11:08.38 | dym | Well now thats not polite :) |
| 11:11.02 | Madkiss | sorry |
| 11:11.34 | Madkiss | This is really really getting me, I have done numerous asterisk configuration steps in the last days, and every time I have to fiddle with asterisk, I remember why de facto, I don't want to have to fiddle with asterisk ;) |
| 11:12.37 | Madkiss | what's the most convenient way to put a person on hold and have music on hold played for that person? ;-) |
| 11:13.18 | kaldemar | Madkiss: press the hold button on a phone? :) |
| 11:13.46 | Madkiss | kaldemar: we don't even have phones. physical ones ;) |
| 11:13.49 | Madkiss | MusicOnHold() i guess. |
| 11:13.59 | jacc0 | park() |
| 11:14.02 | jacc0 | maybe |
| 11:24.03 | Madkiss | ouch |
| 11:24.16 | *** join/#asterisk ruied_ (~ruied@dial-b2-180-197.telepac.pt) |
| 11:27.25 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 11:31.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 11:32.09 | toresbe | kaldemar: I have a feeling it's not completely done. |
| 11:33.13 | toresbe | As far as I can tell, it actually ignores the destination address and sends it to itself. I have no idea how or why. |
| 11:37.41 | toresbe | If I do SIP:(destination number)@provider, I send it to myself. If I do SIP:(dest)@(dest)@(provider) it forms the right headers, but sends it to (null). |
| 11:43.05 | toresbe | I don't have the time to get this working right now. A shame, would have been cool if it worked. :) |
| 11:43.56 | zealiod | Is nay one successfully using extensions within mysql - there seems to be no support for n,i,t in priority or labelsâ¦. nay one else's experience> |
| 11:44.41 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 11:46.01 | defswork | anyone from sangoma here ? |
| 11:47.06 | defswork | am getting Circuit/channel congestion on a A104d - on all port - one is on a PRI, one is connected to another PBX |
| 11:47.40 | defswork | No REDs though on anything - all looks good |
| 11:49.50 | jacc0 | @defswork: /join #sangoma |
| 11:50.10 | jacc0 | they will be there in about 2,5 hour |
| 11:51.17 | defswork | cheers |
| 11:51.25 | defswork | anyone got any suggestions in the meantime ? |
| 11:51.40 | defswork | all the output suggest that the circuits there etc.. |
| 11:52.43 | singler | what does "wanrouter status" show? |
| 11:53.14 | defswork | wanpipe1 | AFT TE1 | N/A | Connected | |
| 11:53.18 | defswork | all looks ok |
| 11:53.43 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:54.55 | jacc0 | @defswork: make sure you have SSH login details for songoma support; that is the fastested way to get it fixed |
| 11:55.07 | defswork | yeah - thats not a prob |
| 11:56.24 | singler | defswork: are other ports connected too? Also do you see any other errors on asterisk cli? |
| 11:57.15 | defswork | singler, we've disconnected all but one now - to try and figure out whats happening |
| 11:57.55 | defswork | it's a 4 port card with two ports unused - one connected to E1 and one connected to another PBX |
| 11:58.02 | defswork | neither call path works now |
| 11:58.10 | defswork | was all working until 90mins ago |
| 11:58.25 | defswork | nothing apparent kicked off the problem |
| 11:58.26 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 11:58.42 | singler | did you try to restart asterisk and driver? And what about asterisk errors? |
| 11:59.16 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 11:59.19 | devil_evoxxx | hi all |
| 11:59.21 | defswork | yeah done all the usual reboot etc.. |
| 11:59.35 | defswork | * -- Executing [s@macro-dial-out-trunk:28] Dial("SIP/DVP6402-00000006", "DAHDI/g0/07766667788,,tTM(on-answer-out)") in new stack |
| 11:59.35 | defswork | [2011-10-04 12:58:53] WARNING[8892]: app_dial.c:2041 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 11:59.35 | defswork | <PROTECTED> |
| 11:59.50 | defswork | thats all in std logs |
| 12:00.12 | defswork | (soz for paste) |
| 12:00.13 | singler | did you increase verbose level? |
| 12:00.17 | defswork | yeah |
| 12:00.26 | singler | ok |
| 12:00.32 | singler | is asterisk dahdi module loaded? |
| 12:00.42 | jacc0 | did you pay the phone bill? |
| 12:00.45 | jacc0 | :P |
| 12:00.49 | singler | :) |
| 12:00.49 | defswork | yeah - shows all channels etc.. |
| 12:01.00 | defswork | In Service |
| 12:01.09 | defswork | We've called Telco just in case |
| 12:01.24 | defswork | but doesnt explain the same error on calls routed to the other PBX |
| 12:02.42 | singler | did you try to check /var/log/messages for clues? I think we could try to up the link to other PBX, this way we would eliminate telco out of equation |
| 12:03.15 | defswork | syslog/messages has nothing of note in it |
| 12:03.25 | defswork | that hw all things the circuit is connected and dany |
| 12:03.28 | defswork | dandy* |
| 12:03.54 | *** join/#asterisk Wouetr1234 (c227da0a@gateway/web/freenode/ip.194.39.218.10) |
| 12:04.03 | Wouetr1234 | hello |
| 12:04.47 | Wouetr1234 | i had a question, if im behind a password protected proxy, and i want to install asterisknow on it, can i still do that? |
| 12:05.15 | defswork | Wouetr1234, I'd expect so |
| 12:05.25 | Wouetr1234 | and do u know how? |
| 12:05.25 | defswork | specify http_proxy settings |
| 12:05.52 | Wouetr1234 | ok |
| 12:06.33 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:06.34 | singler | defswork: try to set "pri set debug <...>" on asterisk cli or something like that to enable pri debugging |
| 12:06.52 | singler | also check wanpipemon -i wXg1 -c Ta where X is number of span |
| 12:08.43 | defswork | Line Code Violation: 533 |
| 12:08.43 | defswork | Far End Block Errors: 4180 |
| 12:08.46 | defswork | is that a concern ? |
| 12:09.16 | defswork | Rx Level: > -2.5db |
| 12:09.38 | singler | errors are a concern especially because you just restarted |
| 12:10.21 | singler | are error/overrun counters in ifconfig also increase? |
| 12:11.17 | defswork | nothing high - overruns:2 overrruns:4 frame:2 |
| 12:11.19 | Madkiss | kaldemar: that works like a charm, thank you very much :) |
| 12:13.30 | defswork | ok - I'm getting D-Channel down |
| 12:13.36 | defswork | on pri debug output |
| 12:14.49 | defswork | Span: 1 Processing event: D-channel Down |
| 12:14.49 | defswork | 1 Changing from state 5 to 4 |
| 12:14.49 | defswork | 1 q931.c:7618 q931_dl_indication: link is DOWN |
| 12:17.36 | singler | hm, I guess it may be down because of errors, but then you should see this in asterisk cli too. In telco did not change settings you should try checking if cable is ok. btw I guess telco is master clock and other pbx is slave? |
| 12:18.20 | defswork | singler, this isn't my pbx so I'm not completely sure of the config - where can I check that ? |
| 12:18.42 | singler | check /etc/wanpipe/wanpipeX.conf |
| 12:18.42 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:19.03 | defswork | TE_CLOCK = NORMAL |
| 12:19.04 | defswork | TE_REF_CLOCK = 0 |
| 12:19.31 | singler | ok, so telco is master clock, what about wanpipe2? |
| 12:20.11 | defswork | wanpipe3 has MASTER |
| 12:20.19 | defswork | which is the pbx connection |
| 12:20.24 | singler | oh |
| 12:20.34 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:20.34 | singler | REF_CLOCK 0 or 1? |
| 12:20.39 | defswork | 1 |
| 12:21.56 | singler | disconnect telco line, change wanpipe3 ref_clock to be 0 and connect PBX. stop asterisk, restart wanpipe, start asterisk |
| 12:22.04 | singler | check if connection with pbx works |
| 12:22.08 | defswork | ok |
| 12:23.51 | defswork | would that require changing the pbx end to master ? |
| 12:26.12 | singler | no, card can source the clock |
| 12:26.51 | singler | ref_clock specifies where to take clock, 1 - means first port, but because we are having problems with it, we not use it now |
| 12:27.10 | singler | s/not/do not |
| 12:31.46 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:38.18 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:46.08 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:47.34 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:47.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:50.06 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 12:52.46 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 12:58.31 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 13:00.51 | ollii | what would be the best way to check if a device (SIP Peer) is BUSY without dialing him? |
| 13:00.56 | ollii | asterisk 1.4 ... |
| 13:01.57 | *** join/#asterisk cyborg-one (1000@85-238-125-30.broadband.tenet.odessa.ua) |
| 13:02.10 | ollii | i tried ChanIsAvail but it returns only "0" |
| 13:04.28 | [TK]D-Fender | ollii, show us. |
| 13:04.38 | [TK]D-Fender | ~pb |
| 13:04.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:05.22 | *** join/#asterisk serafie (~erin@nat/digium/x-aqyqecvrbetarxnt) |
| 13:05.33 | *** join/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 13:06.35 | rampage73 | anyone have experience with the calleridname.pl script? I am not sure where to put the script in asterisk |
| 13:06.42 | rampage73 | oh i have version Asterisk 1.6.2.5-0ubuntu1.4 |
| 13:06.47 | ollii | [TK]D-Fender: http://pastebin.com/wcrnEdwx |
| 13:07.10 | ollii | it is only a part of the dialplan |
| 13:07.26 | [TK]D-Fender | ollii, "core show application chanisavail" |
| 13:07.42 | [TK]D-Fender | rampage73, That is a very generic name on a script that isn't part of *. |
| 13:08.00 | ollii | core show application ChanIsAvail |
| 13:08.00 | ollii | <PROTECTED> |
| 13:08.00 | ollii | [Synopsis] |
| 13:08.42 | rampage73 | [TK]D-Fender, yes i know it is very popular according to google though! it is a script to look up caller name and put it with caller id , hang on |
| 13:09.02 | [TK]D-Fender | rampage73, My guess would be "sooner than later" |
| 13:09.19 | ollii | rampage73: maybe /var/lib/asterisk/agi-bin ... then call it via Agi(calleridname.pl,911) |
| 13:09.44 | rampage73 | it is from here http://www.teamforrest.com/blog/89/using-agi-to-get-caller-id-name-cnam/ |
| 13:10.01 | rampage73 | thanks ollii that is what i was looking for |
| 13:10.11 | ollii | [TK]D-Fender: do you mean that ChanIsAvail would not fit my requirements? |
| 13:10.37 | [TK]D-Fender | ollii, I mean read the instructions for it. There is some very evident in them for your request |
| 13:12.22 | rampage73 | ollii, [TK]D-Fender does it matter if the agi-bin directory does not exist there? |
| 13:12.46 | [TK]D-Fender | rampage73, It may be elsewhere on your system. go look at asterisk.conf |
| 13:13.30 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 13:13.56 | rampage73 | [TK]D-Fender, thanks |
| 13:14.24 | *** join/#asterisk Dale_Noll (~dale@kamino.mcts.org) |
| 13:17.13 | *** join/#asterisk cyborg-one (1000@85-238-108-99.broadband.tenet.odessa.ua) |
| 13:17.51 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 13:18.05 | *** join/#asterisk brdude (~brdude@12.200.180.147) |
| 13:27.18 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 13:31.26 | *** part/#asterisk cerienjean (~iper@95.138.77.91) |
| 13:32.46 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 13:33.13 | eduzimrs | hi i have this line im my extensions.conf * 1.8 "exten => _X.,9,MixMonitor(${TMPFILE}, /bin/mv ${TMPFILE} /var/www/integrador/gravacao/chamadas/originadas/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP:6:2}/)" where im doing wrong? |
| 13:33.48 | eduzimrs | its not moving the file to the destination folder |
| 13:35.18 | [TK]D-Fender | eduzimrs, Maybe you should show us what it is doing |
| 13:35.57 | singler | eduzimrs: is folder created? |
| 13:36.03 | eduzimrs | yeap |
| 13:36.13 | eduzimrs | sorry it was permission |
| 13:36.18 | eduzimrs | now worked |
| 13:36.21 | eduzimrs | tks |
| 13:37.29 | eduzimrs | im moving from * 1.4 to 1.8 im getting some problems |
| 13:38.50 | kaldemar | odd if it worked since it does not conform to the syntax. unless TMPFILE has a comma in it. |
| 13:42.58 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 13:45.23 | ijpalmer | Hi all, I have a really simple question, I'm trying to send a busy tone to someone if they dial a particular extension but it is just hanging up, am I doing something really stupid and not seeing it http://pastebin.com/a5ShEAw6 Thanks |
| 13:49.16 | *** join/#asterisk timahvo1 (~rogue@41.215.65.50) |
| 13:53.12 | jacc0 | who hangs up? asterisk or the client? |
| 13:54.17 | *** join/#asterisk Skeeter- (cdec4d02@gateway/web/freenode/ip.205.236.77.2) |
| 13:54.23 | Skeeter- | Good Morning |
| 13:54.47 | Skeeter- | What do you guys think about the Linksys PAP2 for a simple home use? |
| 13:54.50 | Madkiss | i am exporting two variables in an agi script with "echo SET VARIABLE FOO BAR" and "echo SET VARIABLE PING PONG". Later on, asterisk only knows about ${FOO} but ${PING} is empty |
| 13:54.52 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:54.53 | Madkiss | what might be the problem? |
| 13:55.39 | kaldemar | Madkiss: define "later on". |
| 13:56.47 | p3nguin | skeeter-: It's good. |
| 13:57.28 | Madkiss | kaldemar: right after I call the script with AG() |
| 13:57.29 | Madkiss | err |
| 13:57.31 | Madkiss | AGI() |
| 13:57.41 | [TK]D-Fender | Skeeter-, Works fine |
| 13:57.48 | Madkiss | <PROTECTED> |
| 13:57.48 | Madkiss | <PROTECTED> |
| 13:57.48 | Madkiss | <PROTECTED> |
| 13:57.50 | Madkiss | that's the code |
| 13:57.57 | Madkiss | PRINUM has a value, SECNUM does not |
| 13:58.34 | [TK]D-Fender | Madkiss, pastebin your actual code & dialplan |
| 13:58.42 | p3nguin | madkiss: Maybe you should pastebin your attendance.agi |
| 13:58.50 | Madkiss | sure, just a sec |
| 13:59.16 | Skeeter- | [TK]D-Fender: it there any alternative to that? |
| 13:59.56 | [TK]D-Fender | Skeeter-, SPA series (slightly higher end), Mediatrix 2100 series for top-end special needs |
| 14:00.10 | p3nguin | If you've chosen an ATA that is good and works fine, why do you need an alternative? |
| 14:00.12 | [TK]D-Fender | Skeeter-, Do you have some inherent issue with the one you were asking about? |
| 14:00.17 | Skeeter- | [TK]D-Fender: i tried the mediatrix and i didnt like them |
| 14:00.42 | [TK]D-Fender | Skeeter-, "I didn't like them" doesn't help us find out what charateristics to avoid or suggest based on. |
| 14:00.52 | Skeeter- | i just want to tap my home pstn line to asterisk so i can call my phones at home froma anywhere |
| 14:00.53 | Madkiss | well |
| 14:01.03 | Madkiss | [TK]D-Fender, kaldemar: While pasting, I just saw the typo |
| 14:01.03 | [TK]D-Fender | Skeeter-, PAP2 or SPA are fine |
| 14:01.05 | Madkiss | thank you anyway :) |
| 14:01.30 | Skeeter- | [TK]D-Fender: i saw some hong kong copies, are they reliable? |
| 14:01.54 | coppice | well, here in Hong Kong we use the real PAP2 |
| 14:03.58 | [TK]D-Fender | Skeeter-, could you be a little more generic please? |
| 14:04.03 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:04.31 | Skeeter- | [TK]D-Fender: this doesnt really seem legit : http://www.ebay.ca/itm/UNLOCKED-LINKSYS-PAP2T-NA-SIP-VOIP-2-Port-Phone-Adapter-/160652524007?pt=LH_DefaultDomain_2&hash=item2567a2f9e7 |
| 14:04.33 | [TK]D-Fender | Skeeter-, you aren't stating your requirement, or grievances with any others you've tried, and are asking about nameless "sopies" we can only guess at |
| 14:05.16 | [TK]D-Fender | Skeeter-, On a karmic scale you are looking like a cheap-ass who is trying his level best to get screwed. |
| 14:05.55 | [TK]D-Fender | Skeeter-, I don't see anything there that implies it is a "copy" |
| 14:05.59 | coppice | most of them aren't copies. they are originals which were once locked and now aren't |
| 14:06.48 | *** join/#asterisk master_of_master (~master_of@p57B54E27.dip.t-dialin.net) |
| 14:06.53 | Skeeter- | [TK]D-Fender: its ebay. |
| 14:07.00 | [TK]D-Fender | Skeeter-, PAP2T sells for $45 USD NEW. |
| 14:07.14 | [TK]D-Fender | Skeeter-, If you're worried about ebay, then why are you there? |
| 14:07.55 | *** join/#asterisk d1j0n (~Adium@146-115-82-21.c3-0.sbo-ubr1.sbo.ma.cable.rcn.com) |
| 14:07.58 | Skeeter- | [TK]D-Fender: here is what i want. pstn => pap2 ¨=> asterisk, pap2(2nd phone jack) => home phones |
| 14:08.30 | [TK]D-Fender | Skeeter-, We've already confirmed a whole series of models to you. Why are you not accepting what we've told you repeatedly? |
| 14:08.31 | coppice | a PAP2 won't get you very far with that requirement |
| 14:08.55 | [TK]D-Fender | Skeeter-, And you don't plug a PAP2 into the PST, it is an FXS device |
| 14:08.59 | [TK]D-Fender | PSTN* |
| 14:09.19 | [TK]D-Fender | Skeeter-, For basic PSTN (single line), get an SPA-3102 |
| 14:09.22 | Skeeter- | so its the spa |
| 14:09.27 | [TK]D-Fender | Skeeter-, It will also give you 1 FXS |
| 14:09.31 | Skeeter- | nice |
| 14:09.37 | Skeeter- | spa 3102 it is |
| 14:10.49 | Skeeter- | this 1 is more liky 60$ |
| 14:11.04 | [TK]D-Fender | Skeeter-, if you basically want to shove * between your home phone ilne & you home phones, then a single SPA-3102 should do it |
| 14:11.54 | Skeeter- | [TK]D-Fender: yes |
| 14:12.26 | Skeeter- | i hate to live in canada, electronics is to expensive |
| 14:12.44 | atheos | at least your healthcare is affordable |
| 14:13.00 | Skeeter- | affordable, its kinda free... |
| 14:13.26 | atheos | that's my point. I'd be able to buy more electronics if I didn't pay 13.5k a year in health insurance. |
| 14:13.28 | Skeeter- | service offer to the people, tho we pay it throw the taxes etc |
| 14:13.39 | Skeeter- | yeah, |
| 14:13.41 | Skeeter- | gun law sucks too |
| 14:13.48 | atheos | is jealous if single payer |
| 14:13.56 | atheos | if/of |
| 14:14.02 | coppice | we have expensive electronics and nearly free health care. maybe there is a pattern |
| 14:14.20 | Skeeter- | yep |
| 14:14.38 | *** join/#asterisk sourcode (~code@ppp-58-11-71-53.revip2.asianet.co.th) |
| 14:14.51 | Skeeter- | i buy my stuff from US then cross it at the border and pay the 5% fee. |
| 14:14.53 | jacc0 | I don't see the pattern; we have expensive electronics and expensive healt care |
| 14:14.57 | Skeeter- | most of the time |
| 14:15.07 | Skeeter- | jacc0: mexico? |
| 14:15.14 | jacc0 | the netherlands |
| 14:15.23 | Skeeter- | right |
| 14:15.29 | Skeeter- | mexico doesnt have a health care |
| 14:15.56 | Skeeter- | have you seen whats going on there actually |
| 14:16.13 | Skeeter- | cartel dropping 100 of dead bodies in the streets and stuff |
| 14:16.19 | jacc0 | we pay like 1632 euro a year and even then you have to pay for every visit to a hospital or dentist |
| 14:17.12 | Skeeter- | dentist cost 100$+ everytime you step in here |
| 14:17.25 | jacc0 | same here |
| 14:17.50 | Skeeter- | if you have an insurance, depending on it, you can get some % back |
| 14:18.00 | Skeeter- | my gf has like 80% back |
| 14:18.21 | Skeeter- | cuz her father is a contractor |
| 14:18.56 | *** join/#asterisk topro (3ef58e32@gateway/web/freenode/ip.62.245.142.50) |
| 14:19.46 | eduzimrs | anyone can help with queue problem? " -- Executing [901@internacional:1] Queue("SIP/400-00000005", "suporte|t|||60") in new stack |
| 14:19.46 | eduzimrs | <PROTECTED> |
| 14:20.00 | eduzimrs | anyone can help with queue problem? " -- Executing [901@internacional:1] Queue("SIP/400-00000005", "suporte|t|||60") in new stack -- Auto fallthrough, channel 'SIP/400-00000005' status is 'UNKNOWN'" |
| 14:20.04 | [TK]D-Fender | eduzimrs, pastebint he complete calls & configs |
| 14:20.09 | eduzimrs | ok |
| 14:22.32 | Skeeter- | aight |
| 14:22.45 | Skeeter- | th [TK]D-Fender p3nguin , have a good 1 everyone |
| 14:22.56 | eduzimrs | [TK]D-Fender: http://pastebin.com/HFwchYdF |
| 14:24.29 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:24.32 | [TK]D-Fender | eduzimrs, exten => 901,1,Queue(suporte|t) <-- | has not been a valid delimeter since it was deprecated in 1.4 |
| 14:24.58 | eduzimrs | [TK]D-Fender: hum, instead i use "," ? |
| 14:25.03 | [TK]D-Fender | eduzimrs, go read all of the upgrade docs from every branch since then. |
| 14:25.30 | [TK]D-Fender | ediYes, and don't assume that the app syntax is the same either. Go re-red your app instructions while you're at it to be sure. |
| 14:26.09 | *** join/#asterisk irroot (~irroot@196.211.38.98) |
| 14:26.15 | p3nguin | Even when using , in the dial plan, it is often printed at | in the CLI. |
| 14:26.33 | [TK]D-Fender | p3nguin, Except I saw it int he code he PB'd |
| 14:26.39 | [TK]D-Fender | p3nguin, Hook, line, and sinker |
| 14:29.33 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:29.33 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 14:30.17 | p3nguin | About that Polycom that's hosed... is it going to be possible to get into the menu to point it to the server to load the files, or will it only accept the option via dhcp? |
| 14:30.57 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:30.57 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 14:33.37 | *** join/#asterisk timahvo1 (~rogue@41.215.65.50) |
| 14:35.09 | topro | hi there, I'm trying to debug call drops of an asterisk-wifi-voip setup. Is there a good ressource as starting point for trouble-shooting? |
| 14:35.41 | [TK]D-Fender | p3nguin, Bootrom gives you full IP control |
| 14:36.14 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 14:36.50 | [TK]D-Fender | p3nguin, specify the server IP type, usr, pass and you're gold |
| 14:39.43 | *** join/#asterisk bchia (~Adium@nat/digium/x-rufzcvzrsqxptgah) |
| 14:43.13 | *** join/#asterisk celord (~celord@201.191.135.209) |
| 14:44.30 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 14:46.28 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:46.59 | p3nguin | So if it says "Error loading MAC.cfg" then I can get to the menu? |
| 14:52.21 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 14:57.46 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:59.22 | Naikrovek | before it says that, you can. |
| 14:59.38 | Naikrovek | there's a 5 second wait where you can push a 'setup' soft key and get in there that way. |
| 15:00.05 | Naikrovek | more than 5 seconds if you unplug it from the network. |
| 15:02.00 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:02.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:02.23 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:04.02 | *** join/#asterisk lmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:04.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:04.41 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 15:08.31 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:16.55 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:30.33 | devil_evoxxx | irroot: i'm going out from tree day's of test..with tcpdump i can dump rtp traffic ,but if i set rtp set debug ip in asterisk, on the no rtp stream shown |
| 15:30.48 | jaytee | just got an email to download an update for Asterisk:TDG |
| 15:31.28 | irroot | devil_evoxxx there is something not working |
| 15:32.39 | irroot | you use "rtp set debug ip 192.168.150.167:50840" with the port |
| 15:32.59 | p3nguin | Asterisk: The Dead Generation? |
| 15:33.18 | *** join/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 15:33.30 | devil_evoxxx | if i use "rtp set debug ip 87.x.x.x" no rtp stream shown in asterisk cli |
| 15:33.38 | devil_evoxxx | if i specify the port..i can see the rtp stream |
| 15:33.52 | devil_evoxxx | i think is near the specify of ":0" |
| 15:34.15 | devil_evoxxx | when you put only the ip |
| 15:34.23 | irroot | yeah |
| 15:34.35 | lmadsen | jaytee: heck ya you did :) |
| 15:34.48 | irroot | should be able to see all ports |
| 15:35.07 | irroot | i get what you saying now 100% |
| 15:35.18 | irroot | lmadsen we see you hi there dude |
| 15:35.20 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 15:35.42 | lmadsen | ohai :) |
| 15:36.05 | devil_evoxxx | :) |
| 15:36.43 | devil_evoxxx | irroot: you too have noticed this problem ? |
| 15:37.06 | irroot | now i do running it without the port |
| 15:39.07 | titter | So whats the proper way to report a non-serious but annoying bug? |
| 15:39.18 | Qwell | titter: issues.asterisk.org/jira/ |
| 15:39.24 | titter | Thank you |
| 15:39.43 | irroot | titter call qwell at work and tell him personally |
| 15:40.14 | Qwell | Yes, you can do that too. My extension is 610... |
| 15:40.29 | Qwell | cackles at lmadsen |
| 15:41.34 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:41.57 | titter | Basically you have a SIP entry of ABCd. You change it back to ABCD and reload the sip.conf. sip show peers still shows ABCd. I didn't try a restart of Asterisk. |
| 15:42.11 | lmadsen | Qwell: that's not my extension :) |
| 15:42.23 | lmadsen | Qwell: that, and it doesn't ring my phone :D |
| 15:42.25 | Qwell | lmadsen: foiled |
| 15:42.33 | Qwell | notes "cell number" listed on this wiki page |
| 15:42.40 | Qwell | 416-... |
| 15:42.45 | titter | lol |
| 15:42.48 | lmadsen | foie gras? |
| 15:42.54 | lmadsen | Qwell: oh snap |
| 15:43.01 | lmadsen | 416 is not my cell :) |
| 15:43.06 | lmadsen | 647- would be |
| 15:43.06 | Qwell | 647-... |
| 15:43.13 | lmadsen | bingo! |
| 15:43.27 | lmadsen | B I N G O! Bingo was his name-oh! |
| 15:43.30 | Qwell | lmadsen: the wiki has your extension wrong |
| 15:43.35 | lmadsen | Qwell: excellent |
| 15:43.39 | lmadsen | what does it list? |
| 15:43.55 | Qwell | 610x |
| 15:44.00 | lmadsen | heh nice |
| 15:44.28 | Qwell | lmadsen: still says russellb is your boss too! |
| 15:44.54 | lmadsen | Qwell: he is |
| 15:46.35 | p3nguin | In the Polycom 500, changing the boot server type of tftp, but it still asks for user/password? There's no user/password for tftp, so what goes in there? |
| 15:47.38 | [TK]D-Fender | p3nguin, nothing. Though for the fix I was describing to you I suggested using FTP to a closed provisioning folder with SIP app and stock only configs |
| 15:47.58 | irroot | p3nguin dont over stress re polycom odd options |
| 15:48.04 | [TK]D-Fender | p3nguin, haev 2 erleases handy so you can be sure you can force up/downgrade it |
| 15:48.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053208.ics.uci.edu) |
| 15:48.14 | *** join/#asterisk dirkD1 (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 15:48.33 | p3nguin | The same version of SIP that was on it won't work? |
| 15:49.54 | irroot | you can call me on gregory at distrotech dot co dot za |
| 15:50.14 | [TK]D-Fender | p3nguin, we want to force-refresh the app as well in case that became corrupted |
| 15:50.28 | [TK]D-Fender | p3nguin, Complete clean-up |
| 15:50.37 | titter | I handle the 500s via FTP ... just follow their matrix for upgrades. I bring them to a common bootrom, then have device.set to 1 via a custom boot.cfg that gives them the next set of FTP info to upgrade too. |
| 15:51.03 | p3nguin | the 500 is not on the matrix. |
| 15:51.07 | titter | Hmm |
| 15:51.36 | [TK]D-Fender | p3nguin, 2.2 is the last it supports IIRC |
| 15:52.15 | titter | I have 2.1.3 for my 500s at the end |
| 15:52.18 | titter | I forget where I start |
| 15:52.21 | p3nguin | It had 2.1.2 on it, so I put 2.1.2 in the tftp for it. |
| 15:53.15 | titter | Just to check tftp on the same subnet? |
| 15:54.00 | p3nguin | yes |
| 15:54.09 | mocker | Looking for success stories on using a Cisco AS5400XM to convert PRI to SIP, has anyone used these in production? |
| 15:54.29 | p3nguin | If I can't get it to work with tftp, I guess I'll switch to ftp. |
| 15:54.47 | titter | I had issues with tftp when I did this a few years ago. I settled with FTP. |
| 15:54.50 | p3nguin | I figured tftp would be easier. |
| 15:59.06 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 16:00.33 | ollii | mocker: try this one: http://www.beronet.com/ |
| 16:01.47 | mocker | ollii: Thanks, but I'll eventually be going to a DS3 so I like the Cisco because it can grow. |
| 16:02.09 | ollii | mocker: beronet box is up to 4 pri + additional boxes |
| 16:02.11 | titter | p3nguin: http://downloads.polycom.com/voice/voip/relnotes/BootROM_4_1_2RevB_release_notes.pdf ... page 7 of the PDF |
| 16:02.23 | ollii | additional boxes can be synchronized (timing) |
| 16:02.24 | *** join/#asterisk tzanger (tzanger@174.142.65.89) |
| 16:02.28 | titter | Thats the up/down paths for bootroms |
| 16:02.37 | mocker | ollii: Right, that would be a lot of boxes. :) |
| 16:02.37 | p3nguin | I'm certainly not going to upgrade to BootROM 4.1.2. |
| 16:04.00 | titter | I had some 500s that were on like 2.2 or 2.1. I went from 2.2 to 2.6.x at step one, setup the 12 0.cfg to have a custom boot.cfg which sets the device.set 1 and sets the next level of ftp info which is just a different user/pass for the next upgrades it needs to do. So from 2.6 I went to 3.2.3, then 4.1.2 |
| 16:04.31 | titter | It does this on its own, and also upgrades the sip.ld as it goes through |
| 16:05.08 | p3nguin | Looks like that one is back online. |
| 16:05.51 | titter | My 500s run on 4.1.2. without issue :) |
| 16:06.05 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:06.07 | [TK]D-Fender | p3nguin, SIP only, no need to touch the BR |
| 16:06.15 | p3nguin | Yeah. It's online again. |
| 16:06.26 | p3nguin | I never saw it pull the sip app from the tftpd, though. |
| 16:06.31 | p3nguin | So I don't know how it went online again. |
| 16:06.47 | titter | The only reason I up the BR is so I can use the variables in the config fles that weren't supported until the newer BR's |
| 16:07.12 | titter | Actually it was the sip app that supports it but required the newer BR |
| 16:07.14 | titter | Been awhile |
| 16:12.56 | *** join/#asterisk tzanger (tzanger@174.142.65.89) |
| 16:14.20 | devil_evoxxx | irroot: when you have done your test, let me say, you can find me at dalfra at irpinianetcom dot it |
| 16:15.10 | irroot | im outta town for few days |
| 16:15.36 | devil_evoxxx | no problem irroot :) when you are ready,let me say |
| 16:19.56 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 16:23.04 | *** part/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 16:23.21 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:33.14 | *** join/#asterisk bis0n (~56416@css35-2-78-238-86-105.fbx.proxad.net) |
| 16:33.21 | bis0n | hello |
| 16:33.31 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 16:34.34 | *** join/#asterisk rdegges (sngyIhKDSD@69.164.197.143) |
| 16:34.50 | rdegges | Hey all--anyone want to do some dialplan consulting work in the near future? PM me! :) |
| 16:35.28 | bis0n | anyone can help me? Have an dhadi 2.5.0.1... dahdi_hardware -vvv say 0000:03:0a.0 wcfxo- e159:0001 Wildcard X101P but I can't load wcfxo module :/ |
| 16:35.55 | bis0n | FATAL: Module wcfxo not found. |
| 16:37.23 | bis0n | have make an dahdi_genconf, good, and have only wcfxo on /etc/dahdi/modules |
| 16:38.17 | bis0n | when start dahdi I have FATAL: Module dahdi not found. |
| 16:40.01 | lmadsen | isnt' there a dahdi_genconf script that should do most of that for you? |
| 16:40.35 | bis0n | dahdi_genconf: Empty configuration -- no spans |
| 16:40.52 | lmadsen | huh |
| 16:41.02 | lmadsen | is the wcfxo driver being built? |
| 16:41.12 | bis0n | so yes script normally make this |
| 16:41.20 | lmadsen | (not going to be a ton of help as I don't use DAHDI for anything but kernel timing) |
| 16:41.36 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:42.00 | bis0n | How can I know that? |
| 16:43.35 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:44.00 | bis0n | when make config I have an line : INSTALL /usr/src/dahdi-linux-complete-2.5.0.1+2.5.0.1/linux/drivers/dahdi/wcfxo.ko |
| 16:44.13 | *** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net) |
| 16:44.26 | TheCompWiz | anyone have any tips for fixing dahdi timing issues? |
| 16:44.39 | TheCompWiz | I'm at my wits end. |
| 16:44.43 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 16:48.10 | TheCompWiz | (dahdi-timing minus the actual physical dahdi hardware) |
| 16:48.47 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:51.00 | mocker | TheCompWiz: What's the problem you are having? |
| 16:51.15 | TheCompWiz | dahdi_test with Best: 9.973 -- Worst: 8.772 -- Average: 9.770703, Difference: 9.770703 |
| 16:51.44 | TheCompWiz | but only when dynamic spans are configured. when they're not configured it's right at 99.9+% |
| 16:55.12 | TheCompWiz | (with dynamic span disabled: Best: 100.000 -- Worst: 99.996 -- Average: 99.998312, Difference: 99.998313) |
| 16:55.38 | *** join/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 16:55.54 | Rroet | good evening ;) |
| 17:00.04 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 17:00.25 | bis0n | grrr I really don't understand... |
| 17:00.53 | bis0n | need make special thing with firmware of dahdi? |
| 17:03.44 | Rroet | dahdi, that's only for hardware right? |
| 17:03.50 | Rroet | ~dahdi |
| 17:03.51 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 17:05.04 | TheCompWiz | you'd think... but it's also one of the primary timing sources for asterisk.... (up until reciently it was MANDITORY to use it) |
| 17:09.18 | Rroet | TheCompWiz: so on 1.6.xx it's still needed to run the dahdi service for timing? |
| 17:09.41 | TheCompWiz | I don't think so. |
| 17:09.49 | TheCompWiz | only for meetme... if you use it. |
| 17:10.20 | WIMPy | And there are alternative external timing sources as well. |
| 17:10.39 | TheCompWiz | sure... but I wouldn't rely on them. |
| 17:10.41 | WIMPy | Or have been. |
| 17:11.08 | TheCompWiz | we're not talking about a simple RTC... it needs to be much more precise than any network-based clock type thing. |
| 17:11.27 | *** join/#asterisk lmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:11.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:11.35 | Rroet | TheCompWiz: you're focussing on this being run as a virtual machine then? |
| 17:11.41 | TheCompWiz | Nope. |
| 17:11.47 | Rroet | I run asterisk on a normal pc with ntp setup. |
| 17:11.51 | TheCompWiz | All on dedicated hardware for me. |
| 17:11.54 | Rroet | I barelly have timedrifts. |
| 17:12.13 | WIMPy | The RTC is not related to Asterisk timing. |
| 17:12.29 | Rroet | ~meetme |
| 17:12.29 | infobot | methinks meetme is a conference bridge. More info at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 17:12.43 | TheCompWiz | Rroet: we're not talking about a "time" clock.... but rather a highly-accurate clock that can count the time between cpu-cycles |
| 17:12.49 | Rroet | RTC == realtime clock? |
| 17:13.03 | WIMPy | exactely |
| 17:13.20 | TheCompWiz | RTC not implying the device used to tell time/date |
| 17:13.49 | Rroet | yeah, those timedrifts you'd get normally occur when using a virtualised environment in which timedrifts happen cause the cpu cycle rate varies based on need |
| 17:14.08 | TheCompWiz | Rroet: do you just like spattering random dialog that isn't related in here? |
| 17:14.48 | TheCompWiz | time drift is related to the hardware-clock... that stores the date & time. it is FAAAAARRRRR to inaccurate to work for things that require processing of audio data. |
| 17:15.27 | TheCompWiz | s/to inaccurate/too inaccurate/ |
| 17:16.30 | Rroet | that much I gathered from your typing, but what I've seemed to have missed from your lengthy typing is that what makes this RTC tick better then the normal rtc available in a pc when having a single machine with a single cpu and a steady clockrate (cpu cycle rate). |
| 17:17.26 | coppice | what CPU has a steady cycle rate these days? |
| 17:17.29 | TheCompWiz | the RTC tick is much more frequent & precise than the normal clock that tell the day/time |
| 17:20.01 | TheCompWiz | the system clock is accurate within ~1 second (with the help of NTP) ... a good RTC source is accurate within 1/1000000 of 1 second. |
| 17:20.21 | *** join/#asterisk gxdssoft (~gxdssoft@200.106.102.231) |
| 17:20.24 | zamba | hi there, guys.. i have an asterisk installation that has one pstn number and several sip clients that all share this one number when calling out.. i'm now trying to set up a mechanism so that when one user dials out to a number that doesn't pick up, and this remote end then call back to the shared pstn number, they should be routed directly to the inside peer that initiated the call.. any idea how this can be done? agi scripts? |
| 17:21.42 | Rroet | TheCompWiz: thanks. I was under the impression though that with the current cpu's it wouldn't be much of an issue |
| 17:22.05 | TheCompWiz | Rroet: I wish that was the case for me... /sigh |
| 17:22.29 | zamba | anyone done anything like this? |
| 17:22.31 | TheCompWiz | my machine actually has 2 quad-core xeon processors (overkill I know) ... and is still having timing issues. |
| 17:22.55 | zamba | and of course also with a timeout of some sort, so that this direct routing doesn't exist forever, but maybe only for some hours |
| 17:23.00 | Rroet | I was unaware I had them, or never had issues with them on a single core2duo t9300 |
| 17:23.14 | *** join/#asterisk troyt (~troyt@c-67-166-68-75.hsd1.ut.comcast.net) |
| 17:23.17 | TheCompWiz | zamba: several I'm sure... but what you're asking is very vague... and sounds more like "hey... I want someone to do all the work for me... and then kick 'em to the curb" |
| 17:23.58 | TheCompWiz | Rroet: honestly... my older phone-system has 0 issues and it's got an older single dual-core xeon in it. |
| 17:23.59 | zamba | TheCompWiz: huh? |
| 17:24.26 | Rroet | zamba: (not that I'm able to help y ou) but have you made any tries to code an extension for this already. Perhaps somebody is willing to look at your coding attempt and help you correct it. |
| 17:24.27 | TheCompWiz | zamba: did I stutter? or was my response to blunt for you? |
| 17:24.43 | Rroet | TheCompWiz: truth: the latter ;) |
| 17:24.49 | zamba | TheCompWiz: i thought it was quite concrete what i want to accomplish.. and i don't really see how my question was the "do this for me!" kind.. |
| 17:24.51 | TheCompWiz | Rroet: probably. |
| 17:24.53 | zamba | god damn trolls |
| 17:25.25 | zamba | you can't ask a single question without getting abuse |
| 17:25.31 | zamba | it's insane |
| 17:25.38 | lmadsen | zamba: welcome to #asterisk.... |
| 17:26.02 | Rroet | zamba: well, they are tbh. but have you tried to do it yourself and do you have some exten lines in which you tried it. |
| 17:26.10 | Rroet | they don't bite if you show them you've actually tried. |
| 17:26.23 | TheCompWiz | zamba: You outlined a very vague situation of a "ring-back" type procedure... and basically are wanting someone to do it for you... because your question didn't appear anything like... "Hey.... I've been working on this issue... and here's a link to the code... but I'm running into a problem on line ___ " ... |
| 17:26.25 | zamba | Rroet: no, i haven't tried.. but i just wanted to get some thoughts about how this *could* be implemented |
| 17:26.47 | zamba | i have every intention of implementing this myself |
| 17:27.14 | zamba | but i just want to know that i'm heading in the right direction.. that's also why i included the part about the agi scripts, because i have a feeling myself that this is the way of doing it |
| 17:27.17 | TheCompWiz | zamba: I would suggest by starting to outline the process... then break it down into basic steps... then implement them in the dialplan with as little "agi/php" as possible. |
| 17:27.32 | _Corey_ | zamba: I think I understand your question... |
| 17:27.43 | Rroet | I already know what zamba is trying to do. |
| 17:28.29 | Rroet | it's a sort of " call back" functionality. multiple sip phones behind a single pstn. one calls out to a number that isn't answered. The number calls back and needs to be re-routed back to the originating sip phone. |
| 17:28.29 | p3nguin | It should be a basic caller/callee relationship database entry. |
| 17:28.33 | _Corey_ | zamba: If it were me, I would create an AGI to examine the call as it was coming in to see if there were any calls placed to that number within (define your window) timeframe and look at who dialed |
| 17:28.48 | zamba | Rroet: exactly! |
| 17:29.03 | _Corey_ | I would probably do it based on CDR as opposed to an external structure though |
| 17:29.03 | zamba | Rroet: was my original question vague? :) |
| 17:29.09 | Rroet | yes. |
| 17:29.19 | p3nguin | As I call out, I'd record the entry. If a call comes in, I'd check the database for the phone number, and I'd dial the appropriate extension. |
| 17:29.20 | Rroet | But I'm vague too, that makes it easier to understand ;) |
| 17:29.24 | TheCompWiz | You could implement the whole process without 1 line of AGI imho. |
| 17:29.33 | TheCompWiz | is very anti-AGI... as it breaks too often. |
| 17:29.37 | p3nguin | I'd do it in dial plan and the astDB. |
| 17:29.51 | TheCompWiz | gives +1 to p3nguin. |
| 17:29.59 | p3nguin | I'm not a programmer, so I do everything in dial plan that I possibly can. |
| 17:30.02 | Rroet | p3nguin: thanks for yesterday. Although we had it rough, you've taught a lot. |
| 17:30.17 | zamba | aight.. this was the kind of debate i was looking for |
| 17:30.25 | *** join/#asterisk irroot (~irroot@197.171.233.65) |
| 17:30.32 | zamba | i'm not very comfortable with the dialplan, so it sounds like something i should be implementing in agi :) |
| 17:30.37 | p3nguin | It would be easy to set an entry on the outbound call. |
| 17:30.45 | p3nguin | Then check that entry on the inbound. |
| 17:30.48 | TheCompWiz | indeed. |
| 17:31.04 | zamba | "easy" is quite relative :) |
| 17:31.19 | p3nguin | Yes it it... relatively easy. |
| 17:31.26 | zamba | p3nguin: but you store this in astDB? |
| 17:31.30 | p3nguin | yes |
| 17:32.03 | p3nguin | Set(DB(something/${EXTEN})=${CALLERID(num)) |
| 17:32.09 | zamba | what about timeouts? i don't want the relationship lasting potentially forever |
| 17:32.11 | TheCompWiz | points to the Set/Get methods |
| 17:32.20 | p3nguin | Something like that is how I'd write it on the outbound call. |
| 17:32.28 | [TK]D-Fender | zamba, Then also push the date into it |
| 17:32.43 | p3nguin | bugfix: Set(DB(something/${EXTEN})=${CALLERID(num)}) |
| 17:33.25 | zamba | what is "something" in this context? |
| 17:33.35 | zamba | one sec, i'll read up on the astdb |
| 17:33.40 | p3nguin | an arbitrary value |
| 17:33.46 | _Corey_ | zamba: I like the AGI approach because normally customers ask stuff like "well, ... under these circumstances we DON'T want this to happen, etc." which is difficult in the dialplan... But there are many good ways to do this |
| 17:34.16 | [TK]D-Fender | _Corey_, Hardly |
| 17:34.19 | p3nguin | Then on the inbound, Dial(SIP/${DB(something/${CALLERID(num)})}) |
| 17:34.32 | zamba | p3nguin: just to make it unique in astDB, basically? |
| 17:34.48 | _Corey_ | [TK]D-Fender: Hardly what? |
| 17:34.50 | TheCompWiz | how about ... "call-back-number" ... and you can also include a 2nd entry for an "expiration" time if you want. that way you first check if there's a call placed... by that number... then check if the call-back has been expired. if it's been expired... you delete both DB entries.... and move on... if not... you pass to the related extension. |
| 17:34.56 | [TK]D-Fender | _Corey_, Diffictul |
| 17:34.57 | TheCompWiz | yawns |
| 17:34.59 | p3nguin | The family 'something' would be used for every entry from this mechanism. |
| 17:35.00 | Rroet | p3nguin: for argument sake I'd like to differ again on the view of the "user being the phone" thing we had yesterday. |
| 17:35.11 | p3nguin | Users still aren't phones. |
| 17:35.19 | TheCompWiz | agi would be needlessly overly-complicated and inefficient. |
| 17:35.22 | p3nguin | You can argue it as much as you like, but it won't make a person into a device. |
| 17:35.43 | Rroet | p3nguin: I'll explain in another way why "hardware" should be in the sip.conf... at least in my opinion ;) |
| 17:36.13 | p3nguin | Hardware (devices) are supposed to be in sip.conf. This was never in question. |
| 17:36.22 | p3nguin | But people are not hardware. |
| 17:36.50 | [TK]D-Fender | Corect... people are firmware :) |
| 17:36.59 | p3nguin | meat bags |
| 17:37.07 | Rroet | phonenumbers should be in the sip.conf imho because it should matter from which hardware device I'm calling from. it's all based on the phonenumber I login with. hardware is irrelevant to the system. |
| 17:37.08 | _Corey_ | [TK]D-Fender: Difficult all depends on the request... many will find AGI quicker under some circumstances |
| 17:37.24 | TheCompWiz | [TK]D-Fender: I think I'm becoming more software than firmware.... too much Mt. Dew. |
| 17:37.30 | [TK]D-Fender | _Corey_, Sholdn't be for this |
| 17:37.33 | Rroet | s/it should/it shouldn't/ |
| 17:37.43 | *** join/#asterisk m_tadeu (~quassel@89.180.143.91) |
| 17:37.51 | p3nguin | You don't log in with phone numbers. |
| 17:38.03 | p3nguin | If you do, you're doing something wrong. |
| 17:38.18 | [TK]D-Fender | p3nguin, he's differentiating between a physical phone and the reg details now. |
| 17:38.25 | Rroet | same like accessing gmail for example. Gmail doesn't give a crap about the hardware I'm using. As long as I use a browser and my e-mailaddress and password. |
| 17:38.32 | Rroet | [TK]D-Fender: that's what I ment indeed. |
| 17:38.45 | p3nguin | Asterisk doesn't care what kind of phone you use. |
| 17:38.45 | _Corey_ | [TK]D-Fender: Not on its face, but I'm envisioning the inevitable "that's great but can it do ___" response. I just find the astdb/dialplan approach a little too rigid to work with for many things |
| 17:39.01 | p3nguin | As long as you use a phone, and provide the right credentials, it'll allow calls. |
| 17:39.09 | [TK]D-Fender | Rroet, * doesn't care what the hardware is. It talks SIP to *. Those credentials "are" the "phone" for all intents and purposes. |
| 17:39.15 | _Corey_ | [TK]D-Fender: Just a preference here... |
| 17:39.25 | Rroet | so rather then using unique identifiers like mac addresses or unique names.. |
| 17:39.40 | Rroet | should I just use the extention number as my unique identifier. |
| 17:40.05 | p3nguin | You do whatever you want, but I'm not going to promote it. |
| 17:40.07 | [TK]D-Fender | _Corey_, Depends on the specific need. For anything time-based like this I wouldn't use AstDB at all, but rather a SQL DB with simple lookup. Allows for a "sweeper" to clean off dead entries. |
| 17:40.13 | TheCompWiz | just wishes asterisk had some-sort of "catch-all" module that would allow snatching some sip packets and processing them outside of the core process. |
| 17:40.43 | Rroet | p3nguin: I'm trying to learn here... why shouldn't I use it? |
| 17:40.55 | p3nguin | For one thing, it's a security risk. |
| 17:41.09 | p3nguin | If I know your extension number, now I know your login name too. |
| 17:41.20 | p3nguin | that's just one example. |
| 17:41.28 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:41.56 | Rroet | that's a valid point. |
| 17:41.57 | TheCompWiz | meh... sip is transmitted in clear-text 99% of the time anyway. not exactly hard to get the username. |
| 17:42.06 | _Corey_ | [TK]D-Fender: That's kind-of what I was envisioning as the most likely twist, "can it just be the last 24-hrs, etc". IMHO, easier with AGI/SQL... |
| 17:42.10 | p3nguin | You have to be ON the system to get it. |
| 17:42.16 | TheCompWiz | or in the middle. |
| 17:42.23 | p3nguin | If you can get ON the system, you have other problems already. |
| 17:42.32 | Rroet | man in the middle *eek* |
| 17:42.39 | [TK]D-Fender | _Corey_, Can do the SQL direct in dialplan through and not require an extra process. |
| 17:43.04 | p3nguin | ~devicenames |
| 17:43.04 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 17:43.05 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:43.20 | _Corey_ | [TK]D-Fender: Yeah, I'm not a fan of SQL in the dialplan... just a preference |
| 17:43.37 | p3nguin | I use dialplan and astdb for everything. |
| 17:43.43 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 17:44.05 | p3nguin | If you name your keys appropriately, you can wipe all of them from one "group" out at once. |
| 17:44.36 | Rroet | p3nguin: makes the use of sip.conf even more confusing. it's people defined in the sip.conf that log on. not the device. I don't like numbers to be attached to a phone. |
| 17:44.55 | p3nguin | No, DEVICES are listed in sip.conf. |
| 17:44.56 | p3nguin | DEVICES. |
| 17:45.03 | p3nguin | Not people. Not extensions. |
| 17:45.10 | p3nguin | Just devices. |
| 17:45.44 | Rroet | extention has a "dialplan" that links a phonenumber/extension to a device. .. where's the people aspect then? now I'm lost. |
| 17:46.00 | Rroet | I'll get it in a sec though, I'm not far off ;) |
| 17:46.05 | p3nguin | Like I explained before, you abstract an extension and give it to a person. |
| 17:46.14 | p3nguin | My extension is 762. |
| 17:46.22 | [TK]D-Fender | extension is a number you dial. That it might case a specific device to ring is secondary |
| 17:46.39 | [TK]D-Fender | So 762 might lead to dialing SIP/Fred. |
| 17:46.42 | p3nguin | It means nothing really. It's just a number used to reach me once I am given a device. |
| 17:46.58 | Rroet | [TK]D-Fender: fred is a user, not a device ? |
| 17:47.05 | p3nguin | Fred is a device. |
| 17:47.21 | p3nguin | You can't call people, you can only call devices. |
| 17:47.23 | lmadsen | Rroet: http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceConfig.html#DeviceConfig_id283201 |
| 17:47.27 | p3nguin | If a person uses the device, that's something else. |
| 17:47.36 | Rroet | my phone isn't called fred.. it's called iPhone as stated on the back. I as a person use a softphone on the iPhone to logon as user fred to Asterisk |
| 17:47.36 | [TK]D-Fender | Rroet, Forget any special term for the physical phone. Just turf it. Phone uses a SIP account. That is the "device" or "user". not "extension" |
| 17:47.54 | Rroet | [TK]D-Fender: that's exactly the point I'm trying to make |
| 17:48.08 | Rroet | SIP account is user bound, not physical device bound.. |
| 17:48.13 | p3nguin | False. |
| 17:48.18 | [TK]D-Fender | Rroet, * doesn't know "hardware". Got it? It does not know that you are using an iPhone. It is of no importance to * |
| 17:48.19 | Rroet | in my case... |
| 17:48.24 | lmadsen | Extension numbers execute dialplan, and dialplan can either request a call to a device (or not) |
| 17:48.49 | p3nguin | The problem here is that rroet thinks PEOPLE are entered into sip.conf and PEOPLE connect to Asterisk as "users." |
| 17:48.54 | lmadsen | People is an abstracted concept that Asterisk doesn't handle |
| 17:49.03 | p3nguin | But in reality, devices connect to asterisk and devices make and receive calls. |
| 17:49.12 | lmadsen | type=user is not a "user / person" |
| 17:49.19 | lmadsen | it's just a type of device connection |
| 17:49.21 | p3nguin | It's a type of peer. |
| 17:49.42 | [TK]D-Fender | Rroet, now the fact you have a SIP app on it.. well kudos. that is a UA (user agent).v it will use SIP credentials to auto & call *. that user is the "device" (as in section of sip.conf) |
| 17:49.54 | *** join/#asterisk cerberus_za (~coert@8ta-151-108-33.telkomadsl.co.za) |
| 17:49.58 | lmadsen | Rroet: read that link I posted as it should help |
| 17:50.15 | p3nguin | To recap, devices connect to asterisk using the device entries made in sip.conf. |
| 17:50.21 | Rroet | lmadsen: I downloaded the pdf last night of oreilly's. Haven't read it yet. |
| 17:50.31 | p3nguin | People are not involved. |
| 17:50.42 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:50.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:50.44 | Rroet | p3nguin: people use sip credentials to logon to asterisk using either hardware or software devices ? |
| 17:50.45 | lmadsen | malcolmd: there you are |
| 17:50.49 | p3nguin | But if you want to name your device the same as your mother named you, go for it. |
| 17:50.52 | lmadsen | Rroet: no\ |
| 17:50.52 | [TK]D-Fender | Rroet, yes. |
| 17:51.01 | *** part/#asterisk bis0n (~56416@css35-2-78-238-86-105.fbx.proxad.net) |
| 17:51.03 | lmadsen | people don't do anything but operate a device :) |
| 17:51.07 | lmadsen | a device logs into asterisk |
| 17:51.13 | p3nguin | That's what I was saying. |
| 17:51.14 | lmadsen | a person operates that device |
| 17:51.21 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:51.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:51.27 | p3nguin | People are just operators of devices that log on. |
| 17:51.29 | Rroet | lmadsen: read it again.. people "use" sip credentials :) |
| 17:51.32 | [TK]D-Fender | Rroet, X-Lite is a "soft-phone". Stright up software running on a PC. My Polycom IP 225 on my desk is a dedicated SIP phone. |
| 17:51.40 | lmadsen | Rroet: no, the device uses the sip credential |
| 17:51.43 | p3nguin | People might use the devices. |
| 17:51.48 | p3nguin | But the devices use the credentials. |
| 17:51.49 | Naikrovek | 335 |
| 17:51.55 | [TK]D-Fender | yeah, 335 |
| 17:51.55 | lmadsen | I do not talk to asterisk (as a human) |
| 17:52.23 | Naikrovek | i see those 32X and 33X phones all the time now |
| 17:52.31 | Naikrovek | in pictures on the internets. |
| 17:52.38 | [TK]D-Fender | Naikrovek, Both of them? |
| 17:53.00 | Naikrovek | well hard to tell, so I wanted to put the whole class together. |
| 17:53.17 | Naikrovek | that style of phone, that overall polycom line, I see it all the time now |
| 17:53.17 | [TK]D-Fender | Naikrovek, I was talking about both of the "internets" :p |
| 17:53.24 | Naikrovek | oh there are many interwebs |
| 17:53.34 | Naikrovek | :) |
| 18:02.25 | *** join/#asterisk tekzilla (~jon@hmbg-4d069dc4.pool.mediaWays.net) |
| 18:02.32 | Rroet | p3nguin: if I shouldn't use my extention number as my SIP device/credential.. I'm stuck again with a nice puzzle on how to have 1 central voicemail number and have the unique SIP id's routed into the proper extensions mailbox? |
| 18:03.07 | p3nguin | SIP device names are not related to voicemail. |
| 18:03.45 | p3nguin | Are you saying you want to have one single mailbox for every person to share? |
| 18:03.52 | Rroet | no. |
| 18:04.03 | Rroet | 1 central voicemail numbe to call and each have it's own mailbox |
| 18:04.07 | p3nguin | its |
| 18:04.40 | lmadsen | Rroet: use a database lookup, or use something like setvar |
| 18:04.56 | p3nguin | I just fail to see a problem. |
| 18:05.02 | Rroet | exten = 999,1,VoiceMailMain(${EXTEN}@default) |
| 18:05.05 | lmadsen | for example, I use a mac address for the devices, then associate extension numbers to the devices, and associate a person with the extension number |
| 18:05.17 | lmadsen | Rroet: you wouldn't use ${EXTEN} there |
| 18:05.23 | p3nguin | You probably don't want to use EXTEN there. |
| 18:05.31 | p3nguin | That would be mailbox 999@default. |
| 18:05.42 | Rroet | no, 999 isn't the mailbox |
| 18:05.46 | lmadsen | exactly |
| 18:05.46 | Rroet | it's the dialin number |
| 18:05.48 | p3nguin | it is there. |
| 18:05.56 | lmadsen | that's why you woudln't use ${EXTEN} there |
| 18:05.58 | p3nguin | You wrote it as such. |
| 18:05.59 | lmadsen | because then it would always be 999 |
| 18:06.06 | p3nguin | Maybe you'd use callerid. |
| 18:06.24 | lmadsen | funny enough, all of this is explained and documented... |
| 18:06.35 | p3nguin | But that requires the admin to configure proper callerid numbers matching mailbox numbers. |
| 18:06.42 | lmadsen | I spent 9 months working on a book that documents all of what you're asking |
| 18:07.19 | Rroet | not an easy matter it is. |
| 18:07.46 | wdoekes2 | (about books, if you had anything to say about the sudo cat, I lost my irssi session last night) |
| 18:08.08 | p3nguin | It really quite simple to configure all these things to coincide. Most of us do it. |
| 18:08.24 | jaytee | I had my sudo cat de-clawed cuz he was tearin up the furniture |
| 18:08.44 | [TK]D-Fender | Rroet, "SetVar=vmbox=100" <- add to your peers with the appropriate box # and you can reference it as ${vmbox} |
| 18:09.06 | p3nguin | Also a great way to handle it. |
| 18:09.24 | lmadsen | wdoekes2: ah yes I did -- i was saying we explained how we used it earlier in the book, and that it was assumed you'd be using a base/minimal install (and thus using bash) |
| 18:09.47 | wdoekes2 | but bash will fail doing that |
| 18:10.00 | p3nguin | bash will fail doing what? |
| 18:10.05 | Rroet | lmadsen: I previously (when I used a gui) used exten = 999,1,VoiceMailMain(${CALLERID(num)}@default) and my devices had the extensionnumbers as unique identifiers. |
| 18:10.06 | lmadsen | wdoekes2: doesn't for me.... |
| 18:10.15 | wdoekes2 | it should.. |
| 18:10.18 | lmadsen | Rroet: right, and that is totally different than using ${EXTEN} |
| 18:10.24 | Rroet | lmadsen: true |
| 18:10.35 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 18:10.37 | wdoekes2 | user-bash will open the >-file, sudo-bash will run cat |
| 18:10.38 | p3nguin | And VoiceMailMain(${CALLERID(num)}@default) does not involve the SIP device identifier AT ALL. |
| 18:10.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 18:10.57 | wdoekes2 | s/sudo-bash/root/ |
| 18:11.14 | p3nguin | It only involves the caller id number configured for a device. |
| 18:11.17 | Rroet | p3nguin: then what does it involve? when I call 999 when my sip device identifier is [PHONE] it'll receive phone as callerID |
| 18:11.31 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 18:11.38 | p3nguin | CallerID number is most often going to be the extension number you apply to the device so it can be called. |
| 18:11.48 | lmadsen | wdoekes2: I just tested it and the file was created :) |
| 18:12.02 | Katty | hello my asterisk does not work at all how to fix plz?????!!!!11 |
| 18:12.08 | p3nguin | If it receives "phone" as the callerid number, you've broken it. |
| 18:12.10 | wdoekes2 | what? no.. |
| 18:12.10 | lmadsen | Katty: hammer! |
| 18:12.15 | wdoekes2 | $ sudo cat > /root/fail.txt |
| 18:12.15 | wdoekes2 | bash: /root/fail.txt: Permission denied |
| 18:12.23 | lmadsen | wdoekes2: didnt' fail for me |
| 18:12.24 | Katty | HAMMER TIME |
| 18:12.28 | lmadsen | wdoekes2: sudo cat > testing.txt |
| 18:12.29 | Rroet | problem wit hthe 3cx softphone then.. it requires CallerID, exten and ID to be set to PHONE and a password to get it to work |
| 18:12.43 | [TK]D-Fender | Rroet, CALLERID works as well.. if those numbers match |
| 18:12.44 | wdoekes2 | yes.. when testing is in your own homedir.. but it'll be owned by you as well |
| 18:12.50 | lmadsen | oic |
| 18:12.55 | p3nguin | What about sudo $(cat > /root/success.txt) ? |
| 18:13.06 | [TK]D-Fender | Rroet, then use the previous method I gave you |
| 18:13.16 | lmadsen | the whole sudo thing is probably unnecessary actually |
| 18:13.30 | lmadsen | as really you should be editing the files as the same user I guess |
| 18:13.40 | lmadsen | wdoekes2: sounds like errata |
| 18:13.47 | Rroet | [TK]D-Fender: somehow 3cx does'nt allow me to do that or else it can't authenticate to asterisk |
| 18:13.54 | [TK]D-Fender | Rroet, then use the previous method I gave you |
| 18:14.02 | [TK]D-Fender | <[TK]D-Fender> Rroet, "SetVar=vmbox=100" <- add to your peers with the appropriate box # and you can reference it as ${vmbox} |
| 18:14.03 | [TK]D-Fender | ^^ |
| 18:14.12 | p3nguin | Or just set the damned callerid value. |
| 18:14.17 | p3nguin | It's really not that difficult. |
| 18:14.34 | p3nguin | callerid=Soft Phone <1234> |
| 18:14.42 | wdoekes2 | is that an invitation for me to file an errata report? ;) |
| 18:14.58 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 18:14.58 | [TK]D-Fender | Rroet, And * ignores the callerid from your 3CX phone. that has no impact on the callerid from your SIP peer entry |
| 18:14.59 | lmadsen | wdoekes2: yes :) |
| 18:15.10 | wdoekes2 | drat |
| 18:15.25 | Rroet | p3nguin: the < > allow for an extension to be attached to a phone? |
| 18:15.39 | p3nguin | So now you have two ways to deal with it. setvar=vmbox=something and callerid=Your Name <your number> |
| 18:15.40 | [TK]D-Fender | Rroet, no "attached" |
| 18:15.50 | p3nguin | Not attached, simply used. |
| 18:15.54 | p3nguin | given |
| 18:16.58 | p3nguin | It's not uncommon for phones to give weird callerid info, so hard-code it in the peer entry to make sure it is always what you want it to be. |
| 18:17.21 | wdoekes2 | ~book |
| 18:17.21 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 18:19.16 | Katty | hi fender bender. |
| 18:19.21 | [TK]D-Fender | Katty, Mew. |
| 18:19.24 | Rroet | for example if I have a sipphone like: http://pastebin.com/xiKR9NdZ it requires the softphone to use the CallerID, Extension and ID of Phone1 to be able to connect. |
| 18:19.46 | Rroet | it might be a bad functionality in 3cx? |
| 18:19.50 | Katty | fender bender, what should i do for my bday this weekend |
| 18:19.53 | [TK]D-Fender | Rroet, When you set the callerid in your sip peer it overrides whatever your actual device sends. Is this now clear? |
| 18:20.25 | Rroet | [TK]D-Fender: thanks for stating this. Point taken. |
| 18:20.28 | [TK]D-Fender | Rroet, Forget 3cx's design-challenged quirks. They are irrelevant. We are showing you how to configure * here. |
| 18:20.35 | *** join/#asterisk bis0n (~56416@css35-2-78-238-86-105.fbx.proxad.net) |
| 18:20.43 | bis0n | o/ |
| 18:20.51 | Katty | hi bison |
| 18:21.27 | [TK]D-Fender | Katty, Do whatever you feel like. I might prioritize things you won't soon have the opportunity to do due to weather, attendance, etc. |
| 18:22.25 | bis0n | I have an asterisk + dahdi with an fxo interface... ok this is what i want, when an user call asterisk sip server, I want asterisk relay the call by fxo card |
| 18:22.45 | [TK]D-Fender | bis0n, So go Dial out the card |
| 18:23.35 | bis0n | ok, with an user on context1 for exemple, what i need on extension.conf? |
| 18:23.57 | bis0n | any other thing of extension and sip I need change? |
| 18:23.58 | Katty | [TK]D-Fender: okay well then you can take me to get my drinking on |
| 18:24.05 | [TK]D-Fender | bis0n, Dial(DAHDI/grouporchannelhere/thenumbertodial) |
| 18:24.18 | [TK]D-Fender | ~book |
| 18:24.18 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 18:24.34 | bis0n | [TK]D-Fender, thank you!!!! |
| 18:29.06 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:29.39 | michael-i | To apply changes made in gtalk.conf, do I just need to run "jabber reload"? |
| 18:32.08 | Katty | hi michael |
| 18:32.16 | Katty | i knew a guy named michael once |
| 18:32.20 | Katty | he was vera-nomnom |
| 18:32.29 | Katty | actually, i don't think i know ANY michaels that aren't nomnom |
| 18:33.43 | michael-i | Fun game in my office: yell "MIKE!" |
| 18:34.01 | Katty | is it like yelling mom in the middle of the grocery store? |
| 18:34.05 | Katty | 6 people turn around and look |
| 18:34.16 | *** join/#asterisk jits (b7521235@gateway/web/freenode/ip.183.82.18.53) |
| 18:34.21 | michael-i | Pretty much |
| 18:34.26 | Katty | sweeeet. |
| 18:34.50 | michael-i | mike-ro-management |
| 18:34.54 | michael-i | BA-ZING!!!! |
| 18:34.58 | jits | hi guys.. is there a light wt version of asterisk that we can embed in our application, we are looking for p2p voice chat on lan without need for external server |
| 18:35.01 | michael-i | boooooo |
| 18:35.48 | Qwell | jits: make menuselect, disable everything you don't need. |
| 18:36.00 | Katty | make menuselect sounds so wrong |
| 18:36.12 | Katty | or maybe that's just my head that's messed up...hmm |
| 18:36.24 | michael-i | jits: take a look at icanblink.com - they have a sip phone client for win/lin/osx which has bonjour support (auto-discovery of all peers on your network). It's also open source so you can tweak away |
| 18:36.59 | jits | thanks michael-i .. sounds good.. let me try |
| 18:38.43 | zamba | how can i log to the console in my dialplan? |
| 18:38.59 | zamba | i basically just want to see what's in a variable |
| 18:39.00 | jits | michael-i: it does not seem to be doing video :-s |
| 18:39.06 | saxa | is there a way I can use an IP address in iax.conf register => statement ? |
| 18:39.30 | michael-i | jits: true, but you also never said anything about video in your question ;) |
| 18:39.40 | saxa | [Oct 4 19:56:15] ERROR[11334]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("217.223.95.245/braserv", "(null)", ...): Name or service not known |
| 18:39.43 | saxa | [Oct 4 19:56:15] WARNING[11334]: acl.c:579 resolve_first: Unable to lookup '217.223.95.245/braserv' |
| 18:39.46 | saxa | [Oct 4 19:56:15] NOTICE[11334]: iax2-provision.c:551 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. |
| 18:40.02 | jits | michael-i: yeah sorry. I assumed most modern clients would do audio and video .. |
| 18:40.09 | saxa | or should I use some kind of variable ? |
| 18:40.33 | saxa | or am I forced to do an freedns account ? |
| 18:41.19 | [TK]D-Fender | jits, * isn't P2P. P2P largely implies there isnt' a server at all. And * isn't a softphone. |
| 18:42.14 | jits | [TK]D-Fender: I know, was wondering if we can have a really small * that can be embedded in an app. |
| 18:43.07 | [TK]D-Fender | jits, * is a server. You said "don't want server" and P2P. Big clash |
| 18:43.19 | [TK]D-Fender | zamba, If you want to output something to the console to display the contents of a varable, call "NoOp" |
| 18:43.51 | [TK]D-Fender | jits, I am not able to figure what role * possibly plays in what you have in mind. |
| 18:44.06 | mocker | Woo, get to have fun uninstalling wanpipe and getting dahdi setup tonight. |
| 18:45.00 | [TK]D-Fender | mocker, How were you using wanpipe that DAHDI wasn't already there? And what are you uninstalling Wanpipe for? |
| 18:45.08 | zamba | [TK]D-Fender: yup, got it :) |
| 18:45.12 | zamba | [TK]D-Fender: thanks |
| 18:45.18 | jits | [TK]D-Fender: I am looking at doing a training on lan. It needs to show trainer video on multiple desktops and do an audio conf. Trainees see only trainer, while the trainer will see all trainees together |
| 18:45.19 | mocker | [TK]D-Fender: Nah, dahdi is already there. |
| 18:45.28 | mocker | Just switching from sangoma cards to digium. |
| 18:45.52 | mocker | sangoma cards are giving pci errors. |
| 18:46.11 | [TK]D-Fender | jits, And now you want video conferencing as well? Don't think * is at all what you're looking for then... |
| 18:46.17 | mocker | so replacing riser card and sangoma cards. |
| 18:48.10 | jits | [TK]D-Fender: I know * doesn't do video, thats why i thought if i can have multiple one-to-one exchange, and an audio conf I am done. * does one-to-one video and audio conf, doing them together with minimal footprint is my target |
| 18:49.09 | jits | * doesn't do video conf* || one-to-one video* exchange |
| 18:49.10 | [TK]D-Fender | jits, one-to-one != conf. it = a single call. |
| 18:49.48 | *** join/#asterisk MDesade (~chatzilla@ip70-162-84-98.ph.ph.cox.net) |
| 18:49.50 | [TK]D-Fender | jits, Not sure how you'd set up * for an ad-hoc layout for this even for the audio,. |
| 18:49.51 | Qwell | actually, as of 10, confbridge can do basic video conf stuff |
| 18:49.56 | Qwell | follow-talker |
| 18:50.15 | [TK]D-Fender | Qwell, RTP stream-swapping? |
| 18:50.27 | mocker | don't cross the streams. |
| 18:50.34 | michael-i | Is the only way to apply changes to gtalk.conf through completely restarting asterisk? I can't come up with the right magic. |
| 18:50.36 | Qwell | the current talkers video gets sent out. everything else gets dropped |
| 18:50.50 | jits | yes I don't want google plus hangout. I need more like yahoo where my video is visible to multiple friends though they don't see each other .. + an audio conference .. |
| 18:50.58 | [TK]D-Fender | Qwell, Pretty smart & easy way... |
| 18:51.21 | [TK]D-Fender | mocker, But Egon I though you said crossing the streams was baaaaaddd.... |
| 18:52.32 | *** join/#asterisk rene- (~rene@189.221.128.237.cable.dyn.cableonline.com.mx) |
| 18:53.24 | rene- | helli guys, is there a way i can disable DTMF emulation for tones whose length is under a certain threshold? |
| 18:53.29 | rene- | s/helli/hello |
| 18:54.47 | jits | [TK]D-Fender: *repost* yes I don't want google plus hangout. I need more like yahoo where my video is visible to multiple friends though they don't see each other .. + an audio conference .. |
| 18:57.02 | zamba | how can i remove an astdb entry? |
| 18:57.08 | zamba | can i debug the astdb from the asterisk console? |
| 18:57.41 | Katty | Qwell: obligatory pestering. |
| 18:57.44 | Katty | Qwell: pesterpester |
| 18:57.47 | Katty | pesters Qwell |
| 18:57.58 | [TK]D-Fender | zamba, "help database" |
| 18:59.24 | zamba | [TK]D-Fender: aight, thanks |
| 18:59.40 | MDesade | hello, i got everything installed (asterisk, freepbx, FOP) and i need some help with my TDM400 cards and configuring DAHDI |
| 18:59.44 | MDesade | http://pastebin.com/sUmW73DY |
| 19:00.10 | MDesade | any help would be appreciated, (i remember there was a trick to this before?) |
| 19:01.22 | zamba | ok, i've implemented the call back feature now.. but i have a couple of more cases to figure out.. first, i need a timeout on the value in the database and purge the entry if the timeout has expired.. and i also need a default case if the database search yields nothing.. |
| 19:01.26 | [TK]D-Fender | MDesade, Would help if you showed us what you have configured so far and told us what modules you have on it... |
| 19:01.42 | zamba | so i'm thinking a goto clause if the database entry is empty |
| 19:01.58 | zamba | or.. rather.. a goto clause if the database entry is empty or the database entry has expired |
| 19:04.11 | MDesade | <[TK]D-Fender> which file would you like me to paste then? |
| 19:05.24 | MDesade | i installed using freedoh, on a Kubuntu 11.04 64-bit desktop box, core2quad, 4gig ram... i have 2x TDM400's and 3x IVC-200G (not relevant) |
| 19:06.20 | [TK]D-Fender | all of the DAHDI configs |
| 19:06.27 | MDesade | asterisk is running, dahdi is running, freepbx is running... however, DAHDI is not configured yet |
| 19:07.02 | MDesade | ok, im not sure where they are, since digium redid it all |
| 19:07.39 | MDesade | im not stupid, just don't know where the config file is... |
| 19:09.57 | MDesade | which files from /etc/dahdi would you like me to pastebin? |
| 19:11.27 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-129-148.chyn.qwest.net) |
| 19:15.26 | eduzimrs | hi, um using * 1.8 with sip realtime, im registered at the mysql server but my peers got the message (X-Lite) "Registration erro:408 - Forbidden (Bad Auth)" but i already checked the tables and its all ok, any idea? |
| 19:19.30 | zamba | ok, got the part about the default extension working.. so the last thing i have to figure out is the timeout |
| 19:20.11 | [TK]D-Fender | eduzimrs, Idea : show us everything. |
| 19:20.17 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:21.03 | eduzimrs | yeap |
| 19:28.04 | eduzimrs | [TK]D-Fender: http://pastebin.com/URDhzi0H |
| 19:30.19 | [TK]D-Fender | eduzimrs, I don't see SIP peers, register/call attempts, etc |
| 19:31.21 | eduzimrs | [TK]D-Fender; it doesnt appears at cli, thats weird |
| 19:31.28 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 19:33.17 | eduzimrs | [TK]D-Fender, it seems that * is not connected to mysql and registrations fail |
| 19:33.23 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
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| 19:35.05 | eduzimrs | [TK]D-Fender i`ll paste a peer |
| 19:36.06 | eduzimrs | [TK]D-Fender ;http://pastebin.com/207SHbJ5 |
| 19:37.15 | MDesade | nevermind, i fixed it.... uh? thanks? |
| 19:38.03 | [TK]D-Fender | eduzimrs, If MySQL isn't connected then you're pretty much dead in the water |
| 19:38.38 | *** part/#asterisk MDesade (~chatzilla@ip70-162-84-98.ph.ph.cox.net) |
| 19:41.06 | eduzimrs | [TK]D-Fender, did u see the paste? it seems to be connected |
| 19:42.07 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053208.ics.uci.edu) |
| 19:42.40 | eduzimrs | [TK]D-Fender, i guess its some kind of missing in the table. |
| 19:45.15 | [TK]D-Fender | eduzimrs, I'm not guessing anything.... I'm not seeing the actual failure yet. |
| 19:45.37 | [TK]D-Fender | eduzimrs, Go pastebin a failed reg/call |
| 19:46.49 | michael-i | Answered my own question by peeking at the source of chan_gtalk.c: " * \todo Support config reloading." |
| 19:53.22 | Katty | i just sneezed. 3 times. |
| 19:53.27 | Katty | oww. |
| 19:53.56 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 19:54.24 | carrar | PICS!! |
| 19:55.00 | SuperNull | pics or gtfo ? .. did i miss something good |
| 19:55.22 | n3hxs | Bless Katty |
| 19:55.37 | n3hxs | Asterisk is nothing to sneeze at! |
| 19:58.12 | Katty | carrar always wants pics. that's nothing new. |
| 19:58.20 | Katty | carrar: go be a photographer. |
| 19:58.28 | WIMPy | mee too |
| 19:59.19 | eduzimrs | [TK]D-Fender, this is the log from the register attempt... http://pastebin.com/JMXbEW0r |
| 20:08.21 | saxa | god, I got it working :) |
| 20:08.59 | saxa | seems that when connecting 2 * boxes, I need to have the same password in the user:pass@server part. |
| 20:09.17 | saxa | i had 2 different passwords on the boxes and it was not working |
| 20:09.35 | saxa | by putting the same password for both boxes, started to work |
| 20:09.37 | saxa | huh |
| 20:15.27 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 20:18.44 | zamba | back to my "call back" problem.. i have added everything except the timeout now.. can someone help me implement that last piece of the puzzle? |
| 20:18.59 | [TK]D-Fender | eduzimrs, fix the permit/deny for that entry. Don't use odd masks. |
| 20:19.05 | *** join/#asterisk xpot-mobile (~james@ebc-2-30.hotspot.utah.edu) |
| 20:19.13 | zamba | basically what i've done now is added a simple Set() when initiating outgoing calls.. and then a gotoif in the corresponding incoming context.. |
| 20:20.33 | zamba | http://pastebin.com/15PKz5V1 |
| 20:20.38 | zamba | that's what i've done in the incoming |
| 20:23.35 | eduzimrs | [TK]D-Fender, what? |
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| 20:29.01 | eduzimrs | [TK]D-Fender, i removed the fields (permit and deny) from the database |
| 20:29.29 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 20:29.46 | [TK]D-Fender | Checkout time, BBL |
| 20:30.05 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 20:32.54 | p3nguin | saxa: I think that's what we commonly refer to as "misconfiguration." |
| 20:33.17 | p3nguin | You certainly do not require the same password to be on each side. |
| 20:35.05 | zamba | what kind of string operations are the asterisk dialplan able to do? |
| 20:35.41 | zamba | i'm thinking about storing the extension and an epoch timestamp at an astdb key entry |
| 20:36.11 | irroot | mmm seems digium is offline |
| 20:36.39 | p3nguin | zamba: core show function STRFTIME |
| 20:36.47 | irroot | im going offline myself be in the sub tropics tommorow sunshine monkies and cocktails |
| 20:37.34 | zamba | p3nguin: instead of epoch? easier to compare seconds since that's just an int, isn't it? |
| 20:37.53 | zamba | p3nguin: but right now that's not the problem, right now i'm trying to figure out the data structure.. how i'm going to store it on outgoing calls |
| 20:37.55 | p3nguin | Whatever you want to do, I guess. |
| 20:38.44 | zamba | i was thinking something like Set(DB(something/${EXTEN})="${CALLERID(num)}:${EPOCH}") |
| 20:38.51 | zamba | and then split on ':' |
| 20:39.00 | p3nguin | I wouldn't quote it. |
| 20:39.19 | zamba | oh, ok |
| 20:39.24 | zamba | but you'd do something like this? |
| 20:39.40 | p3nguin | Something similar, probably. |
| 20:40.07 | zamba | wow, hey, why am i seeing this in the console: [2011-10-04 22:37:49] NOTICE[11666]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '+900441296857165' rejected because extension not found. |
| 20:40.32 | p3nguin | Ah, you allow anonymous calls. |
| 20:40.54 | p3nguin | Someone is trying to call 00441296857165 on your dime. |
| 20:41.10 | p3nguin | You should consider allowguest=no in sip.conf general. |
| 20:41.34 | zamba | is there any reason why it should be on? |
| 20:41.59 | p3nguin | If you allow people to call you via SIP URI and you don't always know who they are, you'd probably allowguest. |
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| 20:42.17 | zamba | nah, noone uses sip uris to make calls here |
| 20:42.21 | p3nguin | Anyone who you know, you can add a peer entry for them. |
| 20:42.23 | zamba | so i'll just turn it off |
| 20:42.50 | p3nguin | The fact that it's on tells me that you are not using the current asterisk version. |
| 20:43.00 | zamba | anyway.. back to the initial challenge.. i've now added the value like: ${CALLERID(num)}:${EPOCH} |
| 20:43.28 | zamba | so now i'm wondering how to split that |
| 20:43.50 | p3nguin | CUT()? |
| 20:44.59 | zamba | aight, sweet |
| 20:46.11 | saxa | p3nguin: yeah, but I'm sure I got the passwords written correctly on each box, but since I'm human, I'm probably the wrong one :) |
| 20:47.24 | p3nguin | The problem seems to most often be that user/passwd is on the reverse systems. |
| 20:48.05 | p3nguin | But if you have a good passwd, using the same one for both sides is probably okay anyway. |
| 20:49.34 | *** join/#asterisk f2knight (~ben@c-76-115-43-21.hsd1.or.comcast.net) |
| 20:52.44 | zamba | p3nguin: what am i doing wrong here: http://pastebin.com/ZJRqrEmB |
| 20:53.05 | zamba | oh! |
| 20:53.06 | zamba | nevermind |
| 20:53.06 | zamba | i see it |
| 20:53.50 | zamba | oh yeah, that worked |
| 20:53.51 | p3nguin | typo? |
| 20:54.02 | zamba | yeah, saw it |
| 20:56.06 | zamba | just one more question.. when setting the global timeout value.. where and how is the best way of setting that? |
| 20:56.21 | p3nguin | timeout of what? |
| 20:56.24 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 20:56.56 | zamba | well, basically i want to compare the epoch and if that's x hours off i will discard the DB entry and don't dial back |
| 20:57.08 | zamba | and it's the x i want to set somewhere and potentially also manipulate |
| 20:57.41 | p3nguin | Set it in dial plan. |
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| 21:02.56 | zamba | i should have the building blocks to be able to implement this now.. thanks! :) |
| 21:04.00 | zamba | i'll run it by you tomorrow before putting it into production.. have to know i haven't messed something up :) |
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| 21:15.37 | p3nguin | For features.conf parking lots, is it still required in 1.8 to include => parkedcalls? |
| 21:19.02 | paulc | p3nguin: I think so -I was playing with it the other week and couldn't figure out why it wasn't working.. inc'd the context and it was all good :) |
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| 21:20.04 | p3nguin | That's kind of what I figured, but when I've included it, I still see the warning that I'm including that's empty or non-existent or something like that. |
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| 21:37.51 | x86 | I got a new iPad and bought / installed the Bria (X-Lite) softphone for it... I can get the asterisk CLI to have it say that it registered, and I can make outbound calls just fine. |
| 21:38.24 | x86 | SIP presence works only when I make an outbound call, it shows offline otherwise... and sip show peers says UNKNOWN for the host |
| 21:39.17 | x86 | so, it's obviously not registering properly... I've tried UDP as well as TCP for transport in sip.conf, but it seems to do the same either way |
| 21:39.21 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:40.03 | p3nguin | UNKNOWN is an indicator from the qualify setting. If there is an IP address in the field next to the device's name, it's registered. |
| 21:40.34 | x86 | yeah I put qualify=yes on all of my peers |
| 21:40.47 | x86 | let me see if there's an IP on the sip show peers... |
| 21:42.39 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
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| 21:44.07 | p3nguin | It's the "Host" column. |
| 21:44.40 | p3nguin | Most of the time, qualify is used as a keepalive for NATs. |
| 21:44.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:46.52 | x86 | yep, there is an IP in the Host column, and now UNREACHABLE in the Status column |
| 21:46.58 | x86 | but it can make calls right now just fine |
| 21:47.36 | p3nguin | The problem is what? |
| 21:47.53 | x86 | it wont register properly / can't receive calls |
| 21:48.33 | p3nguin | If there's an IP address in the Host field, it registered. |
| 21:48.44 | p3nguin | Is the device having connectivity issues? |
| 21:48.52 | p3nguin | UNLESS... |
| 21:49.02 | p3nguin | You didn't set an IP address statically for it, did you? |
| 21:49.08 | x86 | no, it's DHCP |
| 21:49.11 | *** join/#asterisk wonderworld (~ww@port-92-201-125-208.dynamic.qsc.de) |
| 21:49.15 | p3nguin | That's not what I'm asking. |
| 21:49.26 | p3nguin | You've used host=dynamic in its sip.conf entry? |
| 21:49.28 | x86 | oh, for the "domain" in the Bria config? |
| 21:49.30 | p3nguin | no |
| 21:49.31 | x86 | yes |
| 21:49.37 | x86 | host=dynamic for all of my hosts |
| 21:49.40 | p3nguin | okay |
| 21:50.15 | x86 | should I try using the IP for the "domain" in the Bria config, instead of the hostname? |
| 21:52.10 | x86 | it's been a while since I've had to debug a SIP peer, so I guess I'm not sure what steps to take |
| 21:52.22 | wonderworld | is there documentation on the CPU usage of MeetMe conferences? i would like to know aprox. how many conferences i could host on a server. |
| 21:52.24 | x86 | Bria is setup to use ICE and STUN, would that be possibly causing an issue? |
| 21:52.35 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:53.06 | p3nguin | It would be easy enough to disable STUN. Give it a try. |
| 21:53.51 | x86 | also, there are a couple options "Support RPort" is set to yes, "Support PRACK" is set to no |
| 21:55.24 | *** join/#asterisk serafie (~erin@nat/digium/x-hqpgbttoksehabbd) |
| 21:56.20 | x86 | ok, STUN and ICE are both off, same results |
| 21:57.59 | x86 | heh, it's interesting |
| 21:58.56 | x86 | when I click Register in the app, on the Asterisk CLI it sends the presense updates to the devices that are subscribed to that extension as Available, and then immediately sends another presence update of Unavailable |
| 21:59.29 | x86 | I've also tried changing the domain to the IP of the Asterisk box, no change in results |
| 21:59.54 | saxa | p3nguin: yeah, i think thats the minimum problem. |
| 22:00.17 | saxa | p3nguin: no password is so good, but I can live with it :) |
| 22:00.56 | p3nguin | My wife has Bria on her iPod, but she's not where I can take a look at her settings. |
| 22:01.25 | x86 | well I ran a SIP debug, and never see a REGISTER |
| 22:01.28 | p3nguin | It's kind of not that great of an app. I think I had her using media5 fone or something like that. |
| 22:01.32 | p3nguin | or vaxphone |
| 22:01.34 | p3nguin | I can't remember. |
| 22:02.05 | p3nguin | But Bria just does not work very well on her device. |
| 22:02.54 | saxa | p3nguin: i need your help :) I know you can help me. |
| 22:03.07 | wonderworld | whats your fav linux softphone? i like twinkle but i am wondering if theres something better. |
| 22:03.09 | x86 | ugh, I just spent $15 on this app |
| 22:03.16 | p3nguin | I prefer twinkle. |
| 22:03.39 | p3nguin | I wish you would have asked me first! You could have given me $10 to tell you not to spend $15 on it! |
| 22:03.50 | saxa | i have a problem with one of my sip phones, i was using it without problem, but from the last ip change , it does not connect to asterisk server anymore |
| 22:04.05 | p3nguin | Whose IP address changed? |
| 22:04.14 | saxa | the routers |
| 22:04.22 | saxa | where the server is |
| 22:04.26 | p3nguin | So asterisk is behind NAT. |
| 22:04.40 | saxa | yeah, we set it up last time with you, if you remember |
| 22:04.40 | p3nguin | Are you connecting via SIP? |
| 22:04.44 | saxa | yes |
| 22:04.59 | saxa | but the server is on a DMZ |
| 22:05.14 | p3nguin | I talk to so many people, I honestly don't remember your case specifically. |
| 22:05.20 | p3nguin | Why is it on DMZ? |
| 22:05.33 | saxa | now, it was working perfectly, I do not remember if maybe I did some mess in the adsl modem |
| 22:06.01 | saxa | cause i have a few things on my server where asterisk is |
| 22:06.04 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-lynszzveqzrwcfdg) |
| 22:06.13 | saxa | so it was easier to put it on dmz |
| 22:06.15 | p3nguin | I still don't get it. |
| 22:06.16 | saxa | :) |
| 22:06.23 | p3nguin | You go ahead and fix that, then get back to me. |
| 22:06.30 | saxa | anyway, i cant connect anymore with that phone to my server |
| 22:06.55 | saxa | i do not know how can i start debugging the sip thingy from outside |
| 22:07.19 | saxa | last time I had access to all those things locally, since i was in Brasil, but now I'm in Italy |
| 22:07.34 | saxa | so I would like to figure out where to start checking |
| 22:07.50 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:08.09 | saxa | the box I have in Italy it connects to the Brasil box via IAX and works |
| 22:08.17 | saxa | thats what I set up before |
| 22:08.42 | wonderworld | wow, whats your ping from italy to brasil? |
| 22:08.53 | saxa | but that specific phone I'm talking about is a phone at my home in Brasil, and I connect via sip to that asterisk server |
| 22:09.03 | saxa | wonderworld: about 320ms |
| 22:09.16 | saxa | but the phone is also in Brasil |
| 22:09.30 | saxa | so the ping from the phone to the server was about 86ms |
| 22:09.34 | x86 | p3nguin: when would you have access to wifey's iPad? |
| 22:09.36 | wonderworld | hmm... walkie-talkie feeling in calls? or is it still acceptable? |
| 22:09.44 | saxa | no |
| 22:09.53 | saxa | it was very clear and clean |
| 22:10.02 | x86 | 86ms isn't bad at all |
| 22:10.10 | saxa | today I tried it and it worked like a charm, I was impressed |
| 22:10.15 | p3nguin | I'll see her tonight, but I'm not sure if I'll remember it by that time. |
| 22:10.31 | x86 | p3nguin: can you please try? :P |
| 22:10.33 | p3nguin | Remind me. |
| 22:10.36 | saxa | ok let me explain, 86ms is from Brasil phone to Brasil asterisk box |
| 22:10.41 | x86 | p3nguin: how long? |
| 22:10.55 | saxa | 320ms is from Italy asterisk box to Brasil asterisk box |
| 22:11.00 | p3nguin | maybe in five hours or so. |
| 22:11.04 | x86 | ugh |
| 22:11.10 | x86 | :p |
| 22:11.21 | x86 | so this Media5 Fone is free? |
| 22:11.39 | p3nguin | Yes, they have a free version in the app store. |
| 22:11.56 | saxa | so , anybody has any idea where should I start looking why that Brasil phone doesnt connect to the Brasil asterisk server ? |
| 22:12.35 | x86 | p3nguin: the "Pro" version is only $7, do you know what the difference is? |
| 22:12.54 | p3nguin | I can't remember right this moment, but if you install the free one, you'll probably find out. |
| 22:12.58 | wonderworld | saxa: you could try to reach the asterisk server with a console utility like netcat, to check if you can get thru your firewall/DMZ in general |
| 22:12.59 | p3nguin | Might be ad removal. |
| 22:13.07 | x86 | ah ok |
| 22:13.22 | p3nguin | I typically install the free stuff to see if it's worth a crap. |
| 22:13.32 | p3nguin | Then if it is, see what the paid version will give me. |
| 22:13.40 | saxa | http://pastebin.com/MfwBPu7X <- these are the UDP ports |
| 22:14.00 | x86 | but media5 fone is iphone-only, so it'll be all tiny on the iPad :( |
| 22:14.08 | saxa | wonderworld: i can reach the console without problem |
| 22:14.11 | x86 | whereas Bria is built for the iPad :P |
| 22:14.33 | p3nguin | I don't know anything about that. I just know I put iPhone apps on the iPod touch. |
| 22:14.47 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
| 22:14.53 | wonderworld | saxa: no i meant if you can connect to the SIP port (5060) from outside |
| 22:15.07 | saxa | so I thought the problem is on th ephone side, but then I tried to connect with my sip phone from here from Italy, and it wont connect either |
| 22:15.18 | saxa | wonderworld: oh I c |
| 22:15.22 | saxa | let me try |
| 22:15.28 | wonderworld | saxa: check this document -> http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
| 22:16.10 | wonderworld | you need to map ports for SIP, the media stream and maybe IAX |
| 22:16.23 | x86 | p3nguin: differences are: ad removal, multiple SIP accounts, and wideband (G.722) and HD (iSAC) codecs... |
| 22:16.45 | *** join/#asterisk Meaw (~duty@188.161.141.106) |
| 22:17.00 | saxa | sasa@quadser:~$ nc braserv.chickenkiller.com 5060 |
| 22:17.00 | saxa | braserv.chickenkiller.com [187.127.184.112] 5060 (sip) : Connection refused |
| 22:17.15 | x86 | I need multiple SIP accounts, because I want to use my iPad as a phone, so I need to switch to the external DNS name... |
| 22:17.18 | x86 | hmm |
| 22:17.39 | saxa | wonderworld: It was working few days |
| 22:17.43 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:17.49 | p3nguin | That's where you use STUN or multiple DNS views. |
| 22:17.58 | saxa | anyway, let me check that document you suggested |
| 22:18.02 | x86 | p3nguin: but media5 definitely registered and presence is consistently showing it as available... |
| 22:18.09 | p3nguin | nc braserv.chickenkiller.com 5060 <--- this is TCP |
| 22:18.18 | p3nguin | SIP uses UDP |
| 22:18.42 | saxa | thx |
| 22:18.48 | saxa | I missed an -u |
| 22:18.49 | saxa | :) |
| 22:19.03 | *** part/#asterisk Meaw (~duty@188.161.141.106) |
| 22:19.24 | saxa | ok, now it sits there and nothing happens |
| 22:19.36 | saxa | does this means it connected ? |
| 22:19.37 | p3nguin | That's probably a good sign. |
| 22:19.42 | wonderworld | ya |
| 22:19.45 | p3nguin | There is no connection... it's UDP. |
| 22:20.02 | saxa | ok got it |
| 22:20.16 | saxa | so it transmts to UDP 5060 nothing |
| 22:20.16 | p3nguin | The socket is open. We'll put it that way. |
| 22:20.27 | saxa | yep |
| 22:20.32 | p3nguin | You'd see it on sip debug. |
| 22:20.33 | saxa | also nmap shows it open |
| 22:20.48 | saxa | I can enable sip debug on the console |
| 22:21.01 | *** join/#asterisk Meaw (~duty@188.161.141.106) |
| 22:21.45 | wonderworld | saxa: retry with your softphone now while monitoring sip debug on the console. it might give you a clue whats going wrong |
| 22:24.00 | saxa | yep, let me try |
| 22:24.32 | wonderworld | i hate asterisk in combination with NAT and firewalls. gave me too many nightmares.. :) |
| 22:27.53 | *** join/#asterisk Meaw (~duty@188.161.141.106) |
| 22:30.04 | *** part/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
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| 22:32.04 | x86 | p3nguin: so, Media5 Fone seems to work just fine for inbound and outbound calls... problem is, it doesn't have a ring tone on inbound calls heh |
| 22:32.18 | x86 | doh, I had the iPad mute switch on... oops :P |
| 22:32.22 | p3nguin | haha |
| 22:32.44 | p3nguin | I see you created your own version of DND(silent). |
| 22:33.11 | x86 | wax-on-wax-off-dnd :P |
| 22:35.56 | x86 | weak, if the iPad is locked when I get an inbound call, I have to unlock it first and then go into the app to answer the call |
| 22:39.28 | wonderworld | on android, you can even replace the standard phone-app with a sip-phone |
| 22:39.45 | wonderworld | works great but empties the bat in no time |
| 22:45.23 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 22:51.10 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:51.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:00.09 | *** mode/#asterisk [-q dijib!*@*] by leifmadsen |
| 23:00.22 | p3nguin | Oh. |
| 23:00.26 | dijib | thank you |
| 23:00.27 | p3nguin | That explains it. |
| 23:00.32 | dijib | yeh |
| 23:00.45 | leifmadsen | play nice children |
| 23:00.48 | dijib | im having an issue while compiling asterisk-flite-0.5 |
| 23:00.57 | dijib | http://pastebin.com/Wkkrj3bF |
| 23:01.03 | dijib | why does it fail so bad? |
| 23:01.07 | p3nguin | I saw you here earlier, but you didn't say anything. Now I know why. |
| 23:01.30 | dijib | from the cupcake in my dialplan day. |
| 23:01.53 | p3nguin | Better than an Easter egg! |
| 23:02.34 | dijib | so why is that make dying so hard? |
| 23:02.51 | dijib | or anybody else using Flite? |
| 23:03.04 | dijib | or would there be a better speach sythisiser |
| 23:03.24 | dijib | working from this guide http://www.voip-info.org/wiki/view/Asterisk+Flite |
| 23:05.05 | wonderworld | dijib: espeak is simple to setup, but the synthesis is rather robotic |
| 23:05.28 | dijib | wonderworld, have you heard how Flite sounds? any better? |
| 23:05.42 | wonderworld | nope |
| 23:05.56 | wonderworld | but you can test espak very easily |
| 23:06.06 | wonderworld | there is a package for most distros |
| 23:06.24 | wonderworld | after installing just do "espeak blah" |
| 23:06.34 | leifmadsen | Cepstral sounds quite decent |
| 23:06.37 | dijib | k ill give her a spin. |
| 23:06.50 | leifmadsen | I think a license is like... $20 or something totally reasonable |
| 23:06.52 | dijib | how is the install? |
| 23:07.01 | leifmadsen | pretty straight forward last time I used it |
| 23:07.02 | dijib | yeah i'd rather not spend a penny |
| 23:07.08 | leifmadsen | I'd rather you diaf |
| 23:07.09 | leifmadsen | :) |
| 23:07.23 | dijib | diaf? |
| 23:07.50 | dijib | die in a fire. :'( oh |
| 23:08.16 | leifmadsen | didn't initiate artistic freedom |
| 23:08.33 | p3nguin | haha |
| 23:08.34 | wonderworld | leifmadsen: wow, cepstral really sounds cool for the price. i remember this old AT&T thing for thousands of dollars. |
| 23:08.56 | leifmadsen | wonderworld: yep I've used it back in the day; I can only imagine it sounds better now |
| 23:09.11 | leifmadsen | there is even an Allison synthesized voice |
| 23:09.19 | wonderworld | i will buy that ASAP |
| 23:09.23 | wonderworld | cool thing, tnx |
| 23:09.31 | *** join/#asterisk dms (~dms@nat/digium/x-enasfphdxclurmgo) |
| 23:10.15 | leifmadsen | wonderworld: np, I should be a reseller :D |
| 23:12.01 | wonderworld | doh, the voices are cheap. but you have to buy "voice ports" as well |
| 23:12.45 | SeRi | damn. elastix has the worst community support ever. |
| 23:14.42 | wonderworld | A Concurrency License allows all Cepstral voices on your system, licensed or not, to speak simultaneously for applications which may require it (such as telephony systems). There are several port licensing options to choose from, sold on a per-server basis. |
| 23:14.54 | wonderworld | 24 ports -> 1200$ |
| 23:15.02 | wonderworld | not so cool |
| 23:15.08 | p3nguin | This is strange. I have my timeout value in followme set to 45, but the attempt to reach me is dying at 29 seconds. What could be the problem? There is no timeout in the outbound Dial(). |
| 23:16.07 | p3nguin | seri: Yes it does; but I heard you can get decent free support for asterisk in the #asterisk channel. |
| 23:16.25 | SeRi | rofl! hahahaha thats damn straight |
| 23:16.51 | leifmadsen | wonderworld: that's for concurrent voices if you need to create voices on the fly -- if you want to just generate prompts and re-use them, then you can save them to a file and use them over and over again |
| 23:16.51 | p3nguin | That thing still did not show up. |
| 23:17.41 | leifmadsen | wonderworld: at least that's what I've used it for in the past -- just generated a whole bunch of prompts as I needed them, then essentially played the cached files |
| 23:17.44 | SeRi | o fuck. time to send in another one. Its not the firstime it happens. I have send sims in the past and they have been "flaged" |
| 23:17.58 | p3nguin | Flagged as what? |
| 23:18.02 | Micc | p3nguin, I've never used follow me, whats the advantage over just writing some stuff in the dialplan to find you? |
| 23:18.06 | leifmadsen | is totally confused by SeRi |
| 23:18.16 | leifmadsen | Micc: that's how I do it (dialplan) |
| 23:18.24 | leifmadsen | never used the FollowMe() application |
| 23:18.35 | leifmadsen | advantage is that it's supposedly quicker/easier to setup |
| 23:18.56 | wonderworld | leifmadsen: i see. not as nice as ondemand-voices but for the price absolutely fine |
| 23:18.57 | SeRi | well wrong word. I guess what I mean is that some how they ether get there or don't. |
| 23:19.05 | Micc | leifmadsen, I already spent the time to write mine just the way I like them, so I guess it doesn't apply to me. |
| 23:19.14 | leifmadsen | SeRi: what gets where? |
| 23:19.20 | leifmadsen | Micc: right :) |
| 23:19.49 | SeRi | leifmadsen, I sent out a sim card. as of lately I get people emailing me that there sim didnt make it. |
| 23:20.03 | leifmadsen | I'm not sure what that has to do with #asterisk |
| 23:20.12 | SeRi | so I have to send a second one out. This is the 3rd I lost this month |
| 23:20.48 | SeRi | leifmadsen, well it has nothing to do with you :) I was talking to p3nguin since I sent it to him. |
| 23:21.04 | SeRi | You ask so I explained :) |
| 23:22.06 | p3nguin | Interesting. I think FollowMe() must have a timeout on it when the line has not gone Up. If I answer the call quickly, it'll tell me my options to accept or reject the call for the length of my timeout setting in followme.conf. I think that's different from how it was in 1.4. |
| 23:22.55 | SeRi | p3nguin, brb I am going to set another one up. |
| 23:23.04 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 23:23.06 | SeRi | sending you tracking number this time |
| 23:23.47 | wonderworld | i am planing to host meetme conferences and was wondering if there is documentation on how much CPU they would consume |
| 23:24.23 | p3nguin | micc: I guess it's just easier. Once followme takes the call, it can record a brief message from the caller, have them hold while it tries to reach you, then calls you, plays the message and asks if you want to accept it or not. If you accept it, it connects the call to you. If you reject it or ignore it, it tells the caller it was not able to reach you and returns to the dialplan where it left off. |
| 23:25.00 | justdave | are there any plans at all to support RHEL/CentOS 6 with asterisk? |
| 23:25.22 | justdave | there still appears to be no packages built for it in the official digium/asterisk repos, and it's been out for quite a while now |
| 23:25.50 | SeRi | justdave, you can create an rpm your self :) |
| 23:26.01 | p3nguin | micc: So if my DIALSTATUS was NOANSWER, and I have followme enabled, FollowMe() will try to reach me on the pre-configured phone numbers before returning to dial plan and going to voicemail. If I have it disabled, it skips the FollowMe and just goes to voicemail. |
| 23:26.06 | leifmadsen | wonderworld: basically depends on how many people in each conference, amount of transcoding, if you're recording the conference, etc.... |
| 23:26.31 | justdave | yes, I can, but if Digium's already doing it there's not much point in my doing my own |
| 23:27.38 | justdave | they keep the RPMs pretty up-to-date for rhel5 |
| 23:28.33 | wonderworld | justdave: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29 |
| 23:28.55 | p3nguin | micc: http://pastebin.com/Qsk2QU31 |
| 23:28.58 | justdave | wonderworld: yes, that's what I'm asking about |
| 23:29.04 | wonderworld | leifmadsen: no transcoding, max 10 people per conference. MANY conferences in case my plan works out :) |
| 23:29.05 | justdave | there's no RPMs for RHEL6/CentOS6 there |
| 23:29.09 | justdave | only for 4 and 5 |
| 23:29.12 | wonderworld | justdave: ohh sorry |
| 23:30.22 | leifmadsen | wonderworld: you should experiment to see how much it uses in your situation... it's not really something where you can give much overview about how much CPU. However I would certainly look at using ConfBridge() in Asterisk 10 as it has gone over a significant overhaul and appears to handle significantly more calls than MeetMe() |
| 23:30.37 | p3nguin | leifmadsen: Is there an opposite of DB_EXISTS, or is it okay if I go ahead and use a ! with it to make it a negative? |
| 23:31.02 | wonderworld | leifmadsen: ok, i will read into it. is Asterisk 10 stable for production use? i haven't tried it yet |
| 23:31.04 | SeRi | p3nguin, your pm's are working? |
| 23:31.09 | leifmadsen | p3nguin: You could wrap it in ${ISNULL(${DB_EXISTS(...)})} too |
| 23:31.25 | p3nguin | ! is a less typing. |
| 23:31.28 | leifmadsen | :) |
| 23:31.30 | p3nguin | s/a // |
| 23:31.53 | p3nguin | was going to say a lot less typing, but then decided it's not THAT much less. |
| 23:32.00 | p3nguin | seri: Try once. |
| 23:33.52 | SeRi | p3nguin, I sent you a msg. |
| 23:34.52 | p3nguin | I didn't get it. Let me work on something real quick. |
| 23:35.08 | SeRi | ok. just ping my nick when you are ready |
| 23:36.00 | p3nguin | Try once more. |
| 23:37.04 | Micc | man I really need to update some of my old dialplans. I'm still using some stuff from 1.2 |
| 23:37.13 | SeRi | p3nguin, done. did you get it? |
| 23:37.31 | saxa | i cant get it truogh , ufff, thx to all, see you tomorrow |
| 23:37.43 | Micc | I mean I haven't updated it since then. Like the original extension example. |
| 23:38.28 | Micc | I still have goto(s-${DIALSTATUS}) |
| 23:39.22 | Micc | p3nguin, your dialplan example there gave me some ideas. |
| 23:41.07 | p3nguin | seri: Not yet, try again. |
| 23:42.03 | SeRi | p3nguin, done. |
| 23:42.28 | SeRi | p3nguin, can you msg me? I just need your info again |
| 23:44.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:47.17 | p3nguin | micc: Reload the pastebin page to see the entire thing. |
| 23:49.38 | p3nguin | seri: Did you respond to what I said in PM? |
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| 23:51.17 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 23:53.41 | SeRi | p3nguin, I just did |
| 23:57.57 | Micc | p3nguin, is that a multi-tenant system? |
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