| 00:02.03 | ChannelZ | that's (not) useful |
| 00:02.14 | ChannelZ | But again what is the significance of 929 |
| 00:02.32 | SVLD | i need dial 929 through gateway (my peer), depending on which account call provider |
| 00:03.30 | SVLD | all accounts have different username/pass, but one provider (ip) |
| 00:04.11 | ChannelZ | so 929 is the account |
| 00:04.52 | SVLD | I need use of astersik like mobile gateway, which register at provider and provider can decide which port use for call |
| 00:05.37 | ChannelZ | You're talking about *outgoing* calls? |
| 00:05.55 | ChannelZ | I don't know what the hell you're talking about. Maybe someone else can help. |
| 00:06.14 | SVLD | i route calls from provider to mobile network |
| 00:10.31 | SVLD | provider dial sip/1234/8979879, for example, 1234 - my first account in provider and I dial sip/gw1/8979879, if provider dial sip/1235/9879834, I have to dial sip/gw2/9879834 |
| 00:29.32 | *** part/#asterisk SVLD (4e1efcee@gateway/web/freenode/ip.78.30.252.238) |
| 00:44.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:44.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:50.40 | *** join/#asterisk nightwalk (~null@daimon.vixel.org) |
| 01:05.18 | *** join/#asterisk kaushal (~kaushal@115.118.155.19) |
| 01:05.20 | kaushal | Hi |
| 01:05.23 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 01:06.40 | kaushal | is there a way to blast Outbound calls to 250 known numbers and play the sound file which is basically a campaign ? |
| 01:06.53 | kaushal | from Asterisk CLI ? |
| 01:07.35 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 01:07.37 | kaushal | Basically i need to use the entire 8 PRI Lines using 8 PRI Port Sangoma Card |
| 01:07.59 | kaushal | the issue is that when i use 4 port it works perfectly fine |
| 01:08.51 | kaushal | when i start using PRI lines more than 5 nos , the call gets hung and i get error 101 |
| 01:09.16 | kaushal | I have approached the telco and the Sangoma Card techsupport team. |
| 01:09.26 | kaushal | Any clue please ? |
| 01:09.33 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:10.39 | kaushal | Please let me know if anyone needs configs or logs |
| 01:12.04 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 01:19.29 | pabelanger | kaushal: PRI or BRI? |
| 01:19.40 | pabelanger | either way, sound like a problem with you dial groups |
| 01:19.41 | kaushal | pabelanger: PRI |
| 01:19.55 | djuhl30 | Anyone live in South Korea? |
| 01:20.18 | pabelanger | my bad, 8 ports of PRI. |
| 01:20.52 | kaushal | pabelanger: shall i pastebin the configs ? |
| 01:20.53 | pabelanger | ya, so if the first 4 ports work, which I assume is card 1, then I would look at the configuration of the next 4 ports, the next card |
| 01:21.12 | kaushal | pabelanger: nope |
| 01:22.16 | pabelanger | well, pb your configs / dialplan and somebody will help. |
| 01:22.23 | pabelanger | can't right now, Pho awaits |
| 01:22.27 | pabelanger | & |
| 01:22.36 | djuhl30 | Yum Pho |
| 01:23.21 | kaushal | pabelanger: http://sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html |
| 01:23.43 | kaushal | so its a 4 port with dual mode |
| 01:25.50 | kaushal | djuhl30: do you need my configs ? |
| 01:26.19 | djuhl30 | I love free stuff. |
| 01:26.49 | djuhl30 | But for right now I am trying to buy a voip locally. Seems like South Koreans don't know what one is |
| 01:28.16 | kaushal | pabelanger: Pho awaits ? |
| 01:28.47 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 01:41.58 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 01:45.30 | WIMPy | kaushal: Use the latest libpri and dahdi. |
| 01:58.55 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:23.00 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:26.19 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 02:27.36 | *** join/#asterisk Defraz (~Defraz@65.39.108.7) |
| 02:35.33 | kaushal | WIMPy: ok |
| 02:36.23 | kaushal | WIMPy: is there a way to know the version of dahdi ? |
| 02:36.56 | WIMPy | kaushal: Actually thinking about it, I'm not sure if it wwas libpri, dahdi, or even chan_dahdi. So it could be the Asterisk Version as well. |
| 02:37.07 | kaushal | ok |
| 02:37.52 | kaushal | WIMPy: http://pastebin.ubuntu.com/665399/ |
| 02:38.57 | WIMPy | I'm pretty sure, Asterisk 1.8.5.0 should be ok. |
| 02:39.45 | kaushal | ok |
| 02:41.47 | kaushal | WIMPy: Any further clue ? |
| 02:42.28 | *** join/#asterisk Chinorro (~chino@62.82.228.34.static.user.ono.com) |
| 02:42.32 | brdude | I'm having trouble setting up an asterisk box, I can ping the box and I can ssh the box. I also have the clients set up under sip.conf, but no luck registering them. I'm folowing Asterisk: The deinitive guide 3rd edition. Any ideas where I can begin to track what the problem is? This is my first asterisk box so I'm completely lost. |
| 02:42.51 | WIMPy | I remember the last guy with that issue fixed it by upgrading. |
| 02:43.15 | kaushal | WIMPy: are you referring to me ? |
| 02:43.20 | WIMPy | kaushal: yes |
| 02:43.36 | kaushal | WIMPy: i dont see yum list updates | grep libpri |
| 02:43.42 | brdude | Sorry didn't mean to but in guys. |
| 02:43.55 | kaushal | libpri-1.4.11.5-1_centos5 |
| 02:44.17 | kaushal | WIMPy: do i need to upgrade dahdi too ? |
| 02:44.25 | WIMPy | kaushal: It is no the recent version. But libpri goes in to Asterisk, so maybe you should DIY. |
| 02:44.48 | kaushal | DIY ? |
| 02:44.51 | WIMPy | kaushal: I honestly can't remember, but I thik it was libpri or dahdi. |
| 02:45.05 | WIMPy | Compile form source. |
| 02:45.18 | kaushal | DIY Full form ? |
| 02:45.32 | kaushal | do it yourself ? |
| 02:45.41 | WIMPy | yes |
| 02:45.44 | kaushal | ok |
| 02:46.09 | kaushal | so there is no reason to upgrade dahdi ? |
| 02:46.27 | WIMPy | brdude: Turn up verbose and debug on the *CLI and sii it it tells you something when you try to register or use the phones. |
| 02:47.27 | WIMPy | kaushal: Sorry, I can't remember what exactely fixed it. But it must be mentioned somewhere. We had a few occasions of that one lately. |
| 02:47.36 | kaushal | WIMPy: np |
| 02:51.01 | brdude | WIMPy what is sii? |
| 02:51.08 | kaushal | WIMPy: I would update you |
| 02:51.41 | WIMPy | brdude: Sorry. "sip" |
| 02:52.04 | brdude | Ok thanks, |
| 02:52.48 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:55.26 | WIMPy | brdude: Sorry. Doing too many things for my age, apparently. I meant: See if it has anything to tell you. |
| 02:56.03 | brdude | It's all good. |
| 02:57.01 | brdude | WIMPy, i set verbose to 5 and set debug on for sip but no information came trough. |
| 02:58.05 | WIMPy | brdude: then Asterisk is not receiving anything. Maybe you need to habe a taly to a firewall? |
| 02:58.13 | WIMPy | damn |
| 02:58.21 | WIMPy | brdude: then Asterisk is not receiving anything. Maybe you need to have a talk to a firewall? |
| 02:59.16 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:59.25 | brdude | WIMPy no firewall in between, I'm in a local lan. I have also turned off selinux. |
| 02:59.38 | kaushal | WIMPy: when i run the command channel originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 03:00.06 | kaushal | [Aug 14 08:28:43] NOTICE[2673]: app_mp3.c:127 timed_read: Poll timed out/errored out with 0 |
| 03:00.53 | WIMPy | brdude: Did you configure the correct registrar/proxy on the phones? |
| 03:01.23 | WIMPy | kaushal: Haven't seen that before. I usually use SayUnixTime for testing. |
| 03:01.36 | kaushal | http://pastebin.ubuntu.com/665407/ |
| 03:02.16 | WIMPy | Ok, so now it calls out, but mp3 is b0rked? |
| 03:02.35 | brdude | WIMPy, no proxy setup and for the phone i just set the domain to the IP of the asterisk box. |
| 03:04.07 | WIMPy | brdude: That might be the wrong place. Usually the field(s) for the server are called registrar and proxy. But the naming and functionality differs quite a lot. Check the phones manual. |
| 03:04.51 | brdude | WIMPy, I'm using X-lite 4 for the mac. |
| 03:05.16 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:05.26 | brdude | I also tried media5 on the iPhone. |
| 03:05.28 | WIMPy | I never understood the configuration of X-Lite. |
| 03:07.32 | brdude | I used it on with a test setup I did of asteriskNow box with freePBX and it worked fine with just putting the IP in the domain field. |
| 03:12.26 | djuhl30 | polucom good? |
| 03:12.31 | djuhl30 | polycom? |
| 03:12.38 | djuhl30 | Anyone use that phone? |
| 03:13.07 | brdude | djuhl30 any specifig model or just the brand> |
| 03:13.08 | WIMPy | There seem to be a lot of Polycom fans in here. |
| 03:13.09 | brdude | ? |
| 03:13.51 | djuhl30 | http://www.alibaba.com/product-gs/419046143/VOIP_IP_phone_with_5_SIP.html |
| 03:15.01 | *** join/#asterisk zonyl (~jason@99-181-43-99.lightspeed.rcsntx.sbcglobal.net) |
| 03:16.26 | zonyl | Hi. I am trying to register fring client to my asterisk box, and am getting an "unauthorized" response back to my client. The user/password are seemingly correct, so I am thinking that this is a domain issue (all of my existing clients use IP address as server), fring needs an external dns name. |
| 03:16.41 | djuhl30 | 5 SIP lines on a VOIP phone means you can register each line with asterisks right? |
| 03:17.02 | djuhl30 | provided you configure asterisk |
| 03:17.12 | zonyl | I tried putting a "domain=" in the client sip.conf but it would appear to not be working |
| 03:17.21 | WIMPy | djuhl30: It can mean anything. But hopefully it means 5 accounts. |
| 03:17.54 | WIMPy | But I've also seen Phones supporting multiple accounts, but only on one server. |
| 03:17.58 | *** join/#asterisk DrDigi (~mmurphy@c-76-127-126-212.hsd1.ca.comcast.net) |
| 03:18.45 | zonyl | WIMPy: I have gigaset that supports multiple lines (each has its own account) |
| 03:19.47 | zonyl | I have always been a bit mystified over how asterisk decides what domain it is using for itself. |
| 03:20.56 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 03:23.48 | WIMPy | 1.5K of dialplan optimized away. Time for a flower watering break. |
| 03:25.06 | zonyl | hrm.. magically it started working |
| 03:25.46 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:25.51 | zonyl | I had been doing a "reload" all the while, which apparently didnt kick in the sip.conf change (until I did a "sip reload"?) |
| 03:34.35 | WIMPy | Ok, and now make it bigger again, adding the (hopefully) last feature. |
| 03:47.56 | *** join/#asterisk CryptixOverdrive (cryptix@pool-173-54-207-138.nwrknj.fios.verizon.net) |
| 04:01.43 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 04:35.43 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 04:44.25 | brdude | WIMPy, so iptables turned out to be the problem. |
| 04:44.47 | brdude | I turned of selinux but didn't know iptables was installed by default on centos |
| 04:45.52 | p3nguin | Firewall strikes again. |
| 04:48.07 | *** join/#asterisk Howlader (Howlader@soft.bdcom.com) |
| 05:05.16 | WIMPy | Yes. Firewalls are evil. They keep things from working. |
| 05:12.58 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 05:33.54 | DrDigital | ha! they trying to let 911 calls be done via txting |
| 05:34.57 | p3nguin | That's an interesting idea. Do you think it would work well? |
| 05:36.45 | brdude | Not sure if it would work well, but it would be convenient for people who can't for waterver reason make any noise. |
| 05:37.05 | p3nguin | I hadn't thought of that angle. |
| 05:45.59 | brdude | Would one of you guys mind looking at my sip.conf and extensions.conf and telling me why I'm only getting a busy signal when I try to dial. |
| 05:46.28 | p3nguin | Pastebin it all. Hide ONLY your passwords. |
| 05:46.30 | brdude | Actually it's not even a busy signal, the call just fails to connect |
| 05:46.33 | brdude | http://pastebin.com/KzM3TKr6 |
| 05:47.14 | brdude | There we go, I had that ready just forgot to post it. |
| 05:48.22 | p3nguin | You have no peers. |
| 05:49.04 | WIMPy | I think he has. |
| 05:49.08 | p3nguin | Oh, maybe you do but in a really weird way. |
| 05:49.18 | WIMPy | But shared passwords are innovative. |
| 05:49.25 | p3nguin | Templated items only, I guess. |
| 05:49.42 | p3nguin | I've never seen anyone do it that way before. |
| 05:50.16 | brdude | Yeah, I've been folowing the asterisk book 3rd edition and that's how it told me to set it up. |
| 05:51.12 | brdude | if I do a "sip show peers" the two phones show up and are registed. |
| 05:51.15 | p3nguin | With a password in the template, and no parameters set in the actual peer definition? WEIRD! |
| 05:51.59 | p3nguin | What does "dialplan show" show you? |
| 05:52.00 | WIMPy | That's the work of a real haxx0r :-)) |
| 05:53.22 | brdude | "dialplan show" gives me this |
| 05:53.23 | brdude | http://pastebin.com/LqGjqyiG |
| 05:53.25 | p3nguin | I'll be blown away if THAT is actually in the book. |
| 05:53.46 | p3nguin | Okay, that looks good. |
| 05:53.58 | p3nguin | If you pick up MacRod and dial 100, what happens? |
| 05:53.58 | WIMPy | Probably not that minimalistic. |
| 05:54.35 | p3nguin | or if you pick up either phone and dial 200, what happens? |
| 05:54.50 | brdude | busy signal, with message "Call failed to connect" |
| 05:54.52 | p3nguin | Oh wait! |
| 05:54.55 | p3nguin | I see the problem. |
| 05:55.04 | p3nguin | line 13 of first paste. |
| 05:55.05 | brdude | same thing for 200 |
| 05:55.09 | p3nguin | typo |
| 05:55.23 | p3nguin | conext |
| 05:55.23 | brdude | damn |
| 05:55.41 | p3nguin | Change, save, run sip reload, try again. |
| 05:55.59 | p3nguin | maybe not THE problem, but A problem nevertheless. |
| 05:57.15 | brdude | just tried 200 now that the typo is fixed and it worked. |
| 05:58.01 | brdude | everything works like a charm |
| 05:58.09 | brdude | p3nguin you the man |
| 05:58.24 | p3nguin | Then I would remove the secret from the template and add a secret in each peer, which is different. |
| 05:58.56 | WIMPy | Doing 'whaever reload' can give you warnings on invalid config options you usually miss when loading everything. |
| 05:59.22 | WIMPy | Unfortunatly you need to have an idea where there might be one hidden. |
| 06:00.35 | brdude | WIMPy what verbosity level does it have to be at? I did a dialplan reload earlier when the typo was there and nothing poped up. |
| 06:01.05 | WIMPy | it should have been 'sip reload'. |
| 06:01.05 | p3nguin | I would expect serious errors to pop up at core verbose 0 and up. |
| 06:01.20 | WIMPy | AOL |
| 06:01.28 | p3nguin | A typo in dialplan isn't necessarily a serious error. |
| 06:01.49 | WIMPy | No, but a syntax check would be nice. |
| 06:02.01 | p3nguin | Yes, yes it would. |
| 06:03.13 | p3nguin | Perhaps we can get that in Asterisk 9000. |
| 06:03.41 | WIMPy | But 9000 isn't binary. |
| 06:03.50 | p3nguin | Why would it need to be? |
| 06:03.57 | brdude | WIMPy I also did a sip reload and no errors either |
| 06:04.36 | brdude | let me put the typo back in and retry that just to be sure. |
| 06:05.04 | WIMPy | Hmm. I would have expected it at any debug/verbose level, but maybe it doesn't really car in sip.conf? |
| 06:05.16 | WIMPy | care |
| 06:06.47 | brdude | yep verbose 1 and no errors |
| 06:07.04 | WIMPy | And debug 1? |
| 06:07.44 | brdude | let me try |
| 06:09.55 | brdude | Verbose 50 and debug 50 and still nothing |
| 06:10.10 | WIMPy | Unfreindly. |
| 06:10.22 | WIMPy | I'd say no cookies for Asterisk today. |
| 06:10.24 | brdude | only thing i see is == Parsing '/etc/asterisk/sip.conf': == Found |
| 06:10.24 | brdude | <PROTECTED> |
| 06:11.12 | brdude | Yep, asterisk no get cookie from me. |
| 06:12.02 | WIMPy | renice it to -15 for an hour |
| 06:12.33 | brdude | renice? |
| 06:12.50 | WIMPy | man renice |
| 06:13.20 | p3nguin | That's an awful penalty. |
| 06:14.38 | WIMPy | What do you call a loop in the dialplan then? |
| 06:15.23 | p3nguin | Say what? |
| 06:15.45 | p3nguin | A loop in the dial plan would be called... a ... loop ... I guess. |
| 06:16.14 | DrDigital | thats what they said |
| 06:16.28 | DrDigital | if your hiding in the closet as someone is breaking into your home |
| 06:16.33 | DrDigital | you could txt police |
| 06:16.46 | DrDigital | or if your playing dead... |
| 06:16.50 | p3nguin | It's not a bad idea when looking at it at face value. |
| 06:17.04 | p3nguin | I don't know if there are other implications or not. |
| 06:17.25 | DrDigital | you can also take pictures and txt them or videos |
| 06:18.07 | WIMPy | If you attach a video the whole system will crash. |
| 06:18.08 | p3nguin | "Just text 'HELP' to POLICE" |
| 06:21.42 | brdude | It would also be helpfull if they integrated in gps. |
| 06:22.19 | WIMPy | Many phones have it. |
| 06:22.35 | p3nguin | All modern phones have it. |
| 06:22.41 | p3nguin | cellular, that is. |
| 06:22.45 | WIMPy | And there are various other ways to find the position of a phone, like wifi scans. |
| 06:23.28 | p3nguin | That's one reason the wireless company doesn't like that I still use a StarTAC. There is no GPS in something that old. |
| 06:23.28 | WIMPy | Some if the non GPS one can do GPS with some external help. |
| 06:25.21 | WIMPy | But that's only for the exact position anyway. You already get a pretty small area as a by product just from using the network. |
| 06:26.28 | p3nguin | yeah |
| 06:27.01 | p3nguin | They could find me if they had enough time and I didn't shut off my phone or move around. |
| 06:27.16 | p3nguin | But they won't get exact coordinates. |
| 06:27.43 | WIMPy | Depends on the effort they want to put in to it. |
| 06:42.42 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
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| 07:17.38 | trumee | anybody know where should i create the user 'admin' |
| 07:17.42 | trumee | authenticate: 127.0.0.1 failed to authenticate as 'admin' |
| 07:17.51 | trumee | i am getting the above error |
| 07:18.10 | trumee | manager.c:2259 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' |
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| 07:43.45 | blue | any hints on how to compile AppKonference for asterisk 1.6.2 on debian squeeze? |
| 07:44.08 | blue | makefile should point to the correct sources and includes, but make fails |
| 07:44.36 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 07:49.30 | *** join/#asterisk af_ (~getsmart@78.134.21.118) |
| 07:50.01 | nix8n82-phone | truemee manger.conf? |
| 07:51.20 | p3nguin | Just don't try to copy and paste it; use tab completion. |
| 07:51.32 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:05.28 | nix8n82-phone | I don't know how to tab on my droid x |
| 08:06.21 | p3nguin | The tab complete suggestion was for trumee, anyway. |
| 08:07.58 | trumee | nix8n82-phone: had to restart *. that problem is solved |
| 08:10.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:10.17 | nix8n82-phone | Cool glad it all worked out |
| 08:16.18 | nix8n82-phone | Anyone use an android device to make sip calls and is there one that works with bluetooth? |
| 08:17.46 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 08:26.21 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:27.39 | x1user | Hi, i have the following problem Call from 'tammari' to extension 'my_number' rejected because extension not found in context 'mycontext'. |
| 08:27.40 | x1user | Call from 'tammari' to extension '0883374478' rejected because extension not found in context 'cryptotel.net'. |
| 08:28.01 | x1user | My conf file http://pastebin.com/wEEdrPmQ |
| 08:29.39 | ChannelZ | You have no extension 0883374478 in a context named 'cryptotel.net'. It's pretty much telling you exactly what is wrong |
| 08:29.55 | ChannelZ | though I don't know where it's even getting that context based on what you pasted. |
| 08:29.57 | *** join/#asterisk xpot (~xpot@66.219.225.48.provo.dynamic.broadweavenetworks.net) |
| 08:32.34 | x1user | isnot the context [mycontext] in extensions.conf |
| 08:33.21 | ChannelZ | That is a context yes. "cryptotel.net" is not the same as "mycontext" |
| 08:33.31 | ChannelZ | Where is this call coming from? |
| 08:34.30 | x1user | From a sip client. It is a test set up but i cant set correct dialplan. |
| 08:35.05 | p3nguin | Create a peer for that client. Set the appropriate context for the peer. Put dial plan into that context. Make calls. |
| 08:35.32 | ChannelZ | ok well you didn't show us your sip setup but the problem lies there |
| 08:36.28 | p3nguin | It is beyond me how people cannot grasp the simple flow of a call. |
| 08:36.52 | x1user | http://pastebin.com/PFmsnnNT here is sip.conf |
| 08:37.19 | p3nguin | Call comes from a peer. If the peer is known, the call goes to the context as configured in the peer definition; otherwise, the call goes to the context defined in the general section. |
| 08:38.17 | p3nguin | insecure=very should be insecure=port,invite IF you even need insecure at all. |
| 08:38.51 | ChannelZ | Either you've changed something since pasting your error or something very bizarre is happening |
| 08:39.10 | p3nguin | failure to reload confs between tests? |
| 08:39.12 | x1user | I think it is because of dialplan, super user seems okay to me |
| 08:39.29 | ChannelZ | huh? |
| 08:39.48 | x1user | sip user i mean is ok to me, it should be because of dialplan |
| 08:40.30 | p3nguin | Your dial plan is usable. |
| 08:40.31 | ChannelZ | The errors you've pasted don't match the configs you pasted so I don't know what the error really is. |
| 08:41.25 | p3nguin | It could be a case of secret sip.conf and extensions.conf again. |
| 08:41.41 | p3nguin | You know... where we can't be trusted to see the entire thing. |
| 08:41.57 | ChannelZ | That too |
| 08:44.07 | p3nguin | The super secret codes could leak out on the internets. |
| 08:45.31 | ChannelZ | goes to bed |
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| 10:29.29 | usrbinfoobar | hi guys |
| 10:29.52 | usrbinfoobar | i have a question about g729 and the ipp librarys |
| 10:30.06 | usrbinfoobar | i notice v7 supports g729, a,b and some others |
| 10:30.22 | usrbinfoobar | is it possible to compile these for asterisk? |
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| 10:38.59 | usrbinfoobar | noone knows? |
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| 12:56.18 | Gugge | usrbinfoobar: http://www.google.com/search?q=asterisk+g729 first hit |
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| 13:04.34 | catphish_ | https://issues.asterisk.org/jira/browse/ASTERISK-18271 |
| 13:04.38 | catphish_ | any thoughts appreciated |
| 13:11.29 | Gugge | its slow :) |
| 13:12.47 | Gugge | but i agree, i would expect it to use _800. |
| 13:13.14 | catphish_ | the docs i've read say it loads the whole context then uses the normal pattern matching algorithm |
| 13:13.25 | catphish_ | but that doesn't appear to be the case |
| 13:13.40 | catphish_ | it loads the whole context, but then seems to select the first matching pattern |
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| 13:33.33 | x1user | If i have few mobile phone using chan_mobile, how does asterisk knows which one to answer on specific call? I think it is the dialplan but i cant uderstand it. |
| 13:37.52 | catphish_ | you want asterisk to answer the mobiles? |
| 13:43.15 | x1user | I want when i make call from the sip client connected to asterisk, the phone that is connected to asterisk via chan_mobile to call the number i dial from the sip client. |
| 13:54.15 | leifmadsen | x1user: well you call out using something like Dial(Mobile/<identifier>/${EXTEN}) |
| 13:54.28 | leifmadsen | (although I forget what the chan_mobile channel type is called in Dial()) |
| 13:57.41 | usrbinfoobar | any idea where asterisk's ring file is? |
| 13:57.49 | usrbinfoobar | i want to use the .wav in my channel driver |
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| 15:10.46 | trumee | anybody on gentoo? |
| 15:11.06 | trumee | i cant seem to get AMI switch on, even though manager.conf has it enabled |
| 15:11.24 | trumee | 'manager show settings' shows it is off |
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| 15:47.22 | *** join/#asterisk GreenWolf (Guest21436@cpe-74-77-221-5.buffalo.res.rr.com) |
| 15:47.35 | GreenWolf | hello and good morning |
| 15:47.46 | GreenWolf | is there anyone available i am having call problems in asterisk |
| 15:57.17 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 16:01.30 | WIMPy | ~ask |
| 16:01.30 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:02.07 | GreenWolf | ok im only experiencing this on incoming calls |
| 16:02.27 | GreenWolf | but when i receive an incoming call into my asterisk system it will drop the call after 30 seconds |
| 16:02.39 | GreenWolf | i seem to not be able to hold the call without it dropping |
| 16:02.59 | GreenWolf | i have made that server DMZ thru the router and have allow anyomous sip requests |
| 16:05.13 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 16:06.35 | GreenWolf | anyone have any ideas? |
| 16:11.43 | pabelanger | GreenWolf: sounds like a NAT issue, check session-timers |
| 16:12.45 | GreenWolf | but im confused i put the asterisk server at DMZ on router |
| 16:12.52 | GreenWolf | shouldn't that allow all traffic? |
| 16:13.41 | pabelanger | GreenWolf: who knows, routers tend to do some stupid things with SIP. Enable a SIP debug log and see what is happening |
| 16:13.44 | pabelanger | ~collectdebug |
| 16:13.45 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:13.52 | pabelanger | GreenWolf: ^ pb the results |
| 16:15.42 | WIMPy | Session-timers can't be <300s. That smells like rtptimeout. |
| 16:16.18 | WIMPy | GreenWolf: Can you talk in both directions? |
| 16:16.27 | GreenWolf | yes |
| 16:16.35 | GreenWolf | wait i dont know |
| 16:16.42 | GreenWolf | because i send it to an IVR prompt |
| 16:16.57 | GreenWolf | during the prompt roughly 30 seconds into the call it drops |
| 16:17.33 | pabelanger | checkout rtpkeepalive too |
| 16:17.42 | GreenWolf | but here is there weird thing... when i use flowroute for my DID service |
| 16:17.49 | GreenWolf | the call doesnt drop |
| 16:17.57 | GreenWolf | only difference is that it registers |
| 16:18.02 | GreenWolf | and my ipkall doesnt register |
| 16:18.56 | GreenWolf | do you think its something in my sip.conf file? |
| 16:19.47 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 16:20.13 | pabelanger | You will know more if you look at the debug log, and see what Asterisk is doing |
| 16:20.37 | GreenWolf | ok i will send the log is that the /var/lib/asterisk/logs? |
| 16:20.59 | pabelanger | read the wiki page above |
| 16:21.13 | pabelanger | it will explain everything |
| 16:21.47 | GreenWolf | ok i will also post what my sip file looks like maybe you can see any errors? |
| 16:21.47 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 16:25.34 | GreenWolf | externip=72.45.212.166 |
| 16:25.35 | GreenWolf | ;localnet=192.168.1.18/255.255.255.255 |
| 16:25.35 | GreenWolf | pedantic=no |
| 16:25.35 | GreenWolf | trustrpid=yes |
| 16:25.35 | GreenWolf | generaterpid=yes |
| 16:25.35 | GreenWolf | sendrpid=yes |
| 16:25.35 | GreenWolf | promiscredir=yes |
| 16:25.36 | GreenWolf | rtptimeout=120 |
| 16:25.36 | GreenWolf | videosupport=yes |
| 16:25.37 | GreenWolf | srvlookup=yes |
| 16:25.37 | GreenWolf | progressinband=yes |
| 16:25.38 | GreenWolf | bindport = 5079; Port to bind to (SIP is 5060) |
| 16:25.38 | GreenWolf | bindaddr = 192.168.1.3; Address to bind to (all addresses on machine)(use server external IP) |
| 16:25.39 | GreenWolf | qualify=yes |
| 16:25.48 | WIMPy | ~pb |
| 16:25.48 | 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. |
| 16:28.26 | GreenWolf | ok its in pastebin |
| 16:28.26 | WIMPy | The combination of externip but no localnet is certainly not good. |
| 16:29.39 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 16:30.43 | GreenWolf | please check at http://pastebin.com/raw.php?i=XermcSGv |
| 16:30.49 | GreenWolf | thats my exact copy of my sip.conf file |
| 16:40.19 | GreenWolf | fixed it |
| 16:40.22 | GreenWolf | thanks guys |
| 16:50.18 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 16:51.39 | p3nguin | Networking Rule #1: You don't know what DMZ is or how it works, so don't use it. Just forward the necessary ports and stop screwing with it. |
| 16:53.22 | p3nguin | Networking Rule #2: Use standard ports for services. User Agents don't know that you think you know what you're doing when you change listening ports. |
| 16:56.02 | GreenWolf | thanks p3nguin |
| 16:56.08 | GreenWolf | i will keep that in mind |
| 16:56.20 | p3nguin | I need to make a list of these and put somewhere. |
| 16:56.30 | GreenWolf | yes plz if you do i want the web address |
| 16:56.34 | GreenWolf | for futher reference |
| 16:56.49 | GreenWolf | i have another issue im coming across everytime i setup a system |
| 16:56.51 | WIMPy | Or tell infobot aout DMZ |
| 16:57.05 | GreenWolf | i am using trixbox and when i install phpmyadmin i get this msg |
| 16:57.14 | GreenWolf | phpMyAdmin - Error |
| 16:57.14 | GreenWolf | Cannot start session without errors, please check errors given in your PHP and/or webserver log file and configure your PHP installation properly. |
| 16:57.19 | GreenWolf | any ideas? |
| 16:57.35 | p3nguin | I think I'll buy a domain just for it. It'll be something like "the silliness people do when they don't know what they're doing but think they can make it work correctly anyway .com" |
| 17:00.16 | p3nguin | wimpy: I think it already knows, but I forget to use it. |
| 17:00.18 | p3nguin | ~dmz |
| 17:00.18 | infobot | [~dmz] De-Militarized Zone, or usually a separate physical or logical network that has limited access to your internal systems and is accessible in limited ways from untrusted networks such as the Internet. Putting Asterisk in the DMZ is not an acceptable alternative to properly forwarding the appropriate ports, so don't do it. |
| 17:27.37 | dmz | not knowing how to do something isn't acceptable; learn how & do it right; otherwise stop throwing around dmz and making my xchat keep beeping at me :) |
| 17:31.08 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
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| 17:38.25 | p3nguin | hahahaha |
| 17:38.29 | p3nguin | That's great! |
| 17:43.50 | catphish_ | having dmz as a nick does seem unwise where people discuss networks :) |
| 17:46.01 | catphish_ | speaking on d-mz, can anyone point me to some info on configuring sensible firewall rules and port-related config-option for sip |
| 17:46.26 | WIMPy | The manufacturer of your firewall. |
| 17:47.23 | catphish_ | ... |
| 17:48.29 | dmz | catphish i've been dmz since the day i was born, never give it up and it helps i do networking/security/... |
| 17:48.53 | catphish_ | http://www.voip-info.org/wiki/view/Asterisk+firewall+rules :) |
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| 17:51.46 | catphish_ | i assume each sip call requires a unique rtp port |
| 17:52.09 | catphish_ | (by default 10000-20000) |
| 17:52.21 | WIMPy | At least one. |
| 17:57.02 | pabelanger | I believe 3 ports are required for each call |
| 17:57.50 | WIMPy | An odd number seems odd. How do you get to 3? |
| 17:58.09 | catphish_ | why more than one? |
| 17:58.48 | WIMPy | RTP need not be symetric. So you might end up with one per direction. |
| 17:58.57 | pabelanger | 1 rtp port for each leg of the call (inbound and outbound) = 2 |
| 17:59.19 | WIMPy | And if I got it right things like vide open additional streams. |
| 17:59.22 | catphish_ | i see, i assumed it was symmetric |
| 17:59.49 | pabelanger | cannot remember what the 3rd was for. I think, asterisk just holds it in preparation for using it or something. Would have to ask kpfleming again |
| 17:59.54 | p3nguin | catphish_: You don't need to go anywhere to look at the info; forward or open UDP 5060 and the UDP range defined in rtp.conf, which is usually 10000-20000. |
| 18:00.14 | catphish_ | p3nguin: i already got that, i posted a link :) |
| 18:00.16 | catphish_ | but thanks |
| 18:00.43 | WIMPy | Unless you're using Linux. There you should use the sip conntrack module. |
| 18:00.46 | p3nguin | I saw that you were asking for a place with info, so I was just clearing it up here. |
| 18:01.42 | catphish_ | and i found a place with the info, though all i needed was what you said :) |
| 18:02.14 | catphish_ | using the conntrack module seems wise, but potentially a lot of stress on the conntrack table |
| 18:03.07 | catphish_ | 1,000 calls (assuming each is actually 2 sip channels bridged) will require 6,000 open ports :| |
| 18:03.30 | p3nguin | 6000? |
| 18:03.33 | WIMPy | Hmm. Wouldn't traffic on a forwarded port generate an entry anyway? So the might be no difference. |
| 18:03.48 | WIMPy | theRE |
| 18:03.51 | p3nguin | I would have thought 4000. |
| 18:04.10 | catphish_ | someone said each channel opens 3 ports, but wasn't sure why |
| 18:04.28 | p3nguin | I should test that soon. |
| 18:04.29 | catphish_ | and no, a simple ACL wouldn't need to use the conntrack table at all |
| 18:05.03 | bbryant | catphish_: it might be that each call uses 3 ports: the one that you connect on to initiate the call, and the two rtp ports for media |
| 18:05.30 | catphish_ | well you connect on 5060 so that wouldn't be an issue |
| 18:05.47 | p3nguin | But in that case 5060 is common between them all. |
| 18:05.52 | catphish_ | indeed |
| 18:05.58 | p3nguin | So 4001 ports. |
| 18:06.23 | catphish_ | i'd just allow inbound udp 10000-20000 and disable conntrack, not sure if that introduces any potential security issues |
| 18:06.32 | WIMPy | sticks with "at least 1001". |
| 18:07.21 | WIMPy | It can, if you have nat enables and didn't specify strictrtp. |
| 18:07.33 | *** join/#asterisk irroot (~irroot@197.171.177.154) |
| 18:08.21 | catphish_ | i see no reason why 1000 calls would need more than 2001 ports |
| 18:08.42 | catphish_ | but it may use a separate port for send vs receive for some odd reason |
| 18:08.48 | p3nguin | 1000 calls bridged, 2000 RTP ports |
| 18:08.57 | catphish_ | "Every call with two call legs consumes at least 4 RTP ports (RTP and RTCP in two directions)." |
| 18:09.00 | catphish_ | there's the answer |
| 18:09.09 | p3nguin | err... 2000 RTP ports per call, 4000 total |
| 18:09.16 | p3nguin | damned incompleteness |
| 18:09.36 | p3nguin | Let me start that over. |
| 18:09.43 | WIMPy | 2000 per call. Wow! |
| 18:09.51 | catphish_ | lol :) |
| 18:10.14 | p3nguin | 1000 calls bridged, 2000 RTP ports per side (2 per leg), 4000 ports total. |
| 18:10.21 | WIMPy | Ok, so it's 2, if symetric, otherwise 4. With video 3 or 6. |
| 18:10.23 | p3nguin | I THINK that's what I meant the first time. |
| 18:10.38 | WIMPy | Per cahnnel. |
| 18:10.55 | p3nguin | I'm full of fail right now. |
| 18:11.04 | catphish_ | so the default 10,000 ports should be sufficient for the 2,000 call target i'm looking for |
| 18:11.12 | WIMPy | But then we have directmedia which might save us. |
| 18:11.14 | p3nguin | 10001 default |
| 18:11.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:11.58 | catphish_ | are there any security issues with not using connection tracking? |
| 18:12.17 | WIMPy | >> It can, if you have nat enabled and didn't specify strictrtp. |
| 18:12.22 | catphish_ | i guess it depends whether asterisk validates the source ip of incoming rtp traffic |
| 18:12.31 | WIMPy | Otherwise I don;t see any harm. |
| 18:12.43 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 18:13.07 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 18:13.09 | WIMPy | By default, it doesn't. |
| 18:13.29 | catphish_ | what does strictrtp do? |
| 18:14.00 | WIMPy | That should validate the source of RTP packets. |
| 18:15.34 | WIMPy | That is the important security setting, everybody seems to ignore. |
| 18:15.55 | WIMPy | Even tho exploits have been demonstrated last year. |
| 18:16.11 | catphish_ | is there much you can exploit? |
| 18:16.22 | catphish_ | i assume the best you can do it inject data into the call |
| 18:16.59 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:17.06 | catphish_ | ah i found a pdf about it |
| 18:17.36 | WIMPy | No. Asterisk will send to the last received address. |
| 18:18.14 | WIMPy | So you can redirect the RTP stream, or rather part of it, depending on your timing. |
| 18:18.36 | catphish_ | that's a good point |
| 18:18.50 | catphish_ | i guess that's how nat=yes works |
| 18:19.00 | WIMPy | But the interesing bit is if you have "features" enabled. In that case you can inject some DTMF info to transfer te call to your 0-900 number. |
| 18:19.02 | catphish_ | i assumed it used the source ip of the sip request |
| 18:19.34 | catphish_ | but i guess that would never know the port number to send to |
| 18:19.40 | WIMPy | That wouldn't neccessarily work. There might be a proxy in between. |
| 18:19.55 | WIMPy | It's easy to scan. |
| 18:19.57 | catphish_ | yes i see |
| 18:20.24 | catphish_ | what does strictrtp do then? prevent the changing of the source part-way through a call? |
| 18:20.26 | WIMPy | Just send RTP packets to all ports and see where you get one back. |
| 18:20.30 | catphish_ | i can't find any info on it |
| 18:21.01 | catphish_ | "Every received packet on a RTP port overwrites the return IP / port combination, so the next packet will be transmitted to this peer. |
| 18:21.01 | catphish_ | " |
| 18:21.08 | catphish_ | nice |
| 18:21.12 | WIMPy | The sample config says: |
| 18:21.21 | WIMPy | Enable strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is disabled by default. |
| 18:21.38 | catphish_ | how is the source determined? |
| 18:21.43 | catphish_ | the first packet received? |
| 18:21.50 | WIMPy | yes |
| 18:22.02 | catphish_ | well that makes a lot of sense |
| 18:23.17 | catphish_ | despite what was said earlier, send and receive traffic must use the same ports |
| 18:23.26 | catphish_ | otherwise return data would never pass through a nat |
| 18:23.49 | WIMPy | Did anyone say that always has to work? |
| 18:24.24 | catphish_ | someone suggested that the traffic wasn't symmetric |
| 18:24.39 | catphish_ | anyway i think i get it |
| 18:24.43 | WIMPy | No, I said need not be. |
| 18:24.57 | catphish_ | but if it wasn't, nat traversal wouldn't work |
| 18:25.03 | WIMPy | It usually is, but it doesn't have to. |
| 18:25.07 | catphish_ | so in most cases it will be |
| 18:25.12 | catphish_ | makes sense :) |
| 18:25.17 | catphish_ | strictrtp seems very sane anyway |
| 18:25.21 | WIMPy | Not unless the other end is using conntrack. |
| 18:25.26 | WIMPy | Definitely. |
| 18:25.39 | catphish_ | of course, conntrack would be even better |
| 18:25.47 | catphish_ | but seems expensive on the firewall |
| 18:26.01 | catphish_ | my SRX240 couldn't handle more than about 500 calls |
| 18:26.08 | catphish_ | not sure about iptables |
| 18:26.16 | WIMPy | strictrtp should be good enough. |
| 18:26.56 | catphish_ | i'll try it |
| 18:27.14 | catphish_ | right now i'm being annoyed by my realtime extensions problem |
| 18:31.15 | p3nguin | Is there ever a case where strictrtp would be bad to have enabled? |
| 18:31.28 | catphish_ | i wouldn't think so |
| 18:31.33 | WIMPy | A multihomed peer. |
| 18:34.05 | catphish_ | would an individual call ever hop between hosts? |
| 18:34.14 | catphish_ | that seems like an odd situation but certainly possible |
| 18:34.20 | WIMPy | Highly unlikely, but possible. |
| 18:34.37 | catphish_ | maybe if a customer had failover DSL connections |
| 18:35.02 | WIMPy | Yes, a failover situation would kill your call. |
| 18:35.35 | WIMPy | But _if_ you have session-timers enabled, that will happen anyway, just a little later. |
| 18:35.46 | p3nguin | In which version (or branch) was strictrtp first available? |
| 18:35.56 | WIMPy | NFI |
| 18:36.27 | p3nguin | It's not available in my version. |
| 18:36.54 | p3nguin | Probably something put into one of the 1.6 branched. |
| 18:36.55 | WIMPy | So that version should not be exposed to the internet? |
| 18:37.01 | p3nguin | apparently |
| 18:37.39 | WIMPy | I honestly wonder why that topic is always missing when it comes to Asterisk security. |
| 18:38.06 | catphish_ | it seems reasonable obvious to me, hence why i asked |
| 18:38.31 | catphish_ | but it seems odd that such a feature is disabled by default |
| 18:38.51 | WIMPy | All security features are disabled by default. |
| 18:39.03 | p3nguin | That was my reason for asking if there was a case where it would be bad. |
| 18:39.07 | WIMPy | That's to make it easier to get something going. |
| 18:39.23 | p3nguin | There was recently one that was turned on by default... |
| 18:39.30 | p3nguin | alwaysauthreject |
| 18:39.37 | WIMPy | Which means that you must never use the sample configs with an internet connection. |
| 18:39.40 | p3nguin | (I think that's how it's written) |
| 18:40.13 | WIMPy | Yes, but there has also been discussion about allowguests. |
| 18:40.43 | p3nguin | That one is still yes by default? |
| 18:40.48 | catphish_ | allowguest is interesting |
| 18:40.53 | catphish_ | it's enabled by default |
| 18:40.55 | WIMPy | yes |
| 18:41.01 | p3nguin | pewpy |
| 18:41.08 | p3nguin | I figured it would have been changed by now. |
| 18:41.13 | WIMPy | The request to change that was rejected. |
| 18:41.25 | catphish_ | i have a problem at the moment where providers send calls from multiple IPs so it's been necessary to allow guests |
| 18:41.28 | catphish_ | but it's not idea |
| 18:41.30 | catphish_ | *ideal |
| 18:42.03 | WIMPy | No, you should configure on epeer per IP. |
| 18:42.13 | p3nguin | Providers have a limited number of addresses. |
| 18:42.30 | catphish_ | yes, sadly peers don't let allow multiple IPs |
| 18:42.38 | catphish_ | so the configs can get a little large |
| 18:42.39 | WIMPy | And wonder why the whole thing doesn;t work sometimes if they add another IP. |
| 18:42.44 | p3nguin | I can't think of any ITSPs that use more than a dozen IPs for calls. |
| 18:42.54 | WIMPy | Use templates. |
| 18:43.22 | WIMPy | United Internet have 16. |
| 18:43.37 | catphish_ | hmm probably is a good idea to configure them all |
| 18:43.37 | p3nguin | Okay, I can't think of any that use more than 16. :) |
| 18:45.15 | WIMPy | But somehow you were right. They only use 12. The other 4 seem to be a hot standby. |
| 18:50.09 | KavanS | any suggested links for making polycoms use a different ringtone for internal extensions? - I don't think I have my google search terms right for this :\ |
| 18:50.52 | WIMPy | SipAddHeader with something like Alert-Info? |
| 18:51.03 | KavanS | k, googling |
| 18:51.51 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 18:53.21 | x86 | so for some reason now after I rebooted my * box, when I try to make outbound calls via Google Voice / XMPP, * segfaults... was working just fine before I rebooted, and inbound calls still work fine |
| 18:54.12 | x86 | I'm not really able to get much information about why it's happening, just SIGSEGV's heh... logs are pretty non-helpful |
| 18:54.40 | x86 | has anyone seen this kind of behavior with it before? |
| 18:54.53 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 18:55.50 | ChannelZ | No. Did some package updates get applied or something in between? Something important like libc or similar |
| 18:57.20 | catphish_ | did you build your own asterisk? what version is it? |
| 19:00.25 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 19:05.49 | x86 | ChannelZ: it's certainly possible |
| 19:07.31 | x86 | catphish_: 1.8.5.0 |
| 19:07.40 | x86 | catphish_: yes I built it myself |
| 19:08.15 | catphish_ | did you try recompiling it since the problem started? |
| 19:08.22 | catphish_ | in case the libraries have changed |
| 19:08.38 | x86 | hmm, I still have the source tree, I wonder if I do a make clean, make, make install again, if that will re-build against any potentially new libraries and solve all the world's problems ;) |
| 19:08.55 | x86 | catphish_: nope, but that's a great idea :P |
| 19:08.56 | catphish_ | make clean and rerun configure too |
| 19:09.16 | catphish_ | just to ensure there isn't an incompatibility in libc or similar |
| 19:09.26 | x86 | ugh, I don't want to have to go through menuconfig again, can't I keep my existing .config's? |
| 19:09.38 | catphish_ | i don't think configure clears the menuconfig |
| 19:09.43 | catphish_ | though i may be wrong |
| 19:09.52 | WIMPy | make clean shouldn't be neccessary, but configure might be. |
| 19:10.13 | catphish_ | i'd run both to be sure |
| 19:10.13 | WIMPy | It shouldn't. |
| 19:13.17 | catphish_ | will make recompile already build binaries just because the libraries have changed? |
| 19:13.20 | catphish_ | without clean? |
| 19:14.25 | WIMPy | If the makefiels are complete, yes. |
| 19:14.35 | WIMPy | But you should run configure. |
| 19:16.18 | x86 | k |
| 19:16.21 | x86 | I'll try that |
| 19:20.40 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 19:21.30 | *** join/#asterisk cerberus_za (~coert@196-215-103-15.dynamic.isadsl.co.za) |
| 19:28.34 | trumee | is it possible to have encrypted channel between two * boxes (running * 1.8)? |
| 19:28.55 | catphish_ | trumee: can you really not be bothered to type the word asterisk? |
| 19:28.56 | x86 | WIMPy: yeah it does appear that just doing a configure and then a make, it's re-building all of it without the need for a make clean :) |
| 19:29.24 | trumee | this is what i want to do, sip ATA <> asterisk<>Internet<>asterisk<>sip ATA |
| 19:29.39 | WIMPy | trumee: Use IAX or some VPN. |
| 19:29.43 | catphish_ | i'd use ipsec |
| 19:29.57 | x86 | openvpn is much simpler |
| 19:30.07 | catphish_ | either way |
| 19:30.12 | catphish_ | with iax2 on top |
| 19:31.00 | trumee | i usually ftp things between the two asterisk servers. Can i use openvpn just for iax? |
| 19:31.29 | catphish_ | why not set up the vpn then transfer everything over it |
| 19:31.36 | catphish_ | why would you want the ftp unenctypted? |
| 19:31.43 | trumee | catphish_: because that will slow things down |
| 19:31.54 | x86 | trumee: sure you can |
| 19:32.09 | trumee | x86: right that will be nice then |
| 19:32.16 | catphish_ | it wouldn't be noticably slower |
| 19:32.17 | WIMPy | You can do whatever you want. |
| 19:32.42 | x86 | trumee: setup a dedicated subnet for the VPN between the two boxes, tell Asterisk to use those IPs for the peers on each side, then use FTP like normal |
| 19:32.45 | trumee | x86: so i can make one machine openvpn server and the other machine a client |
| 19:32.54 | x86 | trumee: yep |
| 19:32.55 | WIMPy | Depends. But it's worth to note that any vpn will break TC. |
| 19:33.15 | trumee | WIMPy: TC? |
| 19:33.18 | x86 | trumee: make sure you do openvpn over TCP |
| 19:33.26 | WIMPy | Traffic Control. |
| 19:33.36 | WIMPy | OR QoS or whatever you name it. |
| 19:33.46 | trumee | WIMPy: ah right. i dont use it at the moment |
| 19:34.19 | WIMPy | If you're using VOIP, you better should. |
| 19:34.32 | catphish_ | only if you run other traffic too :) |
| 19:34.39 | trumee | x86: The sip ATA is linksys spa3102. Can i configure asterisk so that the ATA doesnt have to register for oout going calls |
| 19:34.46 | WIMPy | right |
| 19:34.58 | WIMPy | It never has to. |
| 19:35.09 | catphish_ | you only have to register if you have a changing IP |
| 19:35.24 | WIMPy | Registering is to tell Asterisk where to send calls if you have an dynamic IP. |
| 19:35.33 | trumee | catphish_: no i dont have a changing ip. It is fixed ip ATA on the lan |
| 19:35.41 | WIMPy | And only for thst. |
| 19:35.42 | catphish_ | thn you don't need to register |
| 19:35.50 | catphish_ | just specify the ip in the peer configuration |
| 19:36.17 | trumee | nice, that means i can use the ATA to call out using multiple gateways |
| 19:36.50 | catphish_ | you can? |
| 19:37.14 | trumee | i already have a openvpn running on a openwrt router. |
| 19:37.32 | trumee | catphish_: with spa3102 i can use multiple gateways |
| 19:37.44 | x86 | ok, asterisk is rebuilt, I'm restarting it now |
| 19:38.09 | catphish_ | ah ok |
| 19:38.32 | *** join/#asterisk cusco (~tralala@a83-132-166-87.cpe.netcabo.pt) |
| 19:38.41 | p3nguin | Asterisk never requires you to register before sending calls; you'll authenticate each and every call you make. |
| 19:38.55 | p3nguin | And that ATA allows sending calls without registering first. |
| 19:39.56 | p3nguin | And there's no need to specify the host address in the peer entry unless you need to receive calls on the ATA. To get calls, either register or define the IP address. |
| 19:40.22 | x86 | catphish_: works great now :) |
| 19:40.29 | catphish_ | x86: great :) |
| 19:40.41 | x86 | heh, can't believe I didn't think about updated system libs :p |
| 19:41.05 | WIMPy | wonders what kind of update could cause that. |
| 19:41.10 | catphish_ | i tend to use the IP to authenticate calls |
| 19:41.35 | p3nguin | contemplates changing asterisk 1.4.40something to 1.8.5.0 today. |
| 19:41.38 | x86 | WIMPy: yeah me too... maybe libxmpp or libjingle or something |
| 19:41.43 | x86 | p3nguin: do it! |
| 19:41.45 | catphish_ | though sending the auth with every invite seems wise really |
| 19:41.59 | x86 | p3nguin: I made the jump about a month ago, love 1.8.5.0 :) |
| 19:42.08 | p3nguin | There's a development branch of chan_sccp-b for it now. |
| 19:42.14 | catphish_ | 1.8.5.0 works great for me, i moved last week from 1.4 |
| 19:42.17 | p3nguin | So if the channel driver works, I have no reason to stay on 1.4 that I know of. |
| 19:42.29 | p3nguin | That has been my only holdback. |
| 19:43.22 | WIMPy | p3nguin: You don't like hijacked RTP? |
| 19:43.34 | p3nguin | It hasn't even been a problem for me. |
| 19:43.49 | p3nguin | I built 1.8.5.0 a couple days ago, but didn't switch over. |
| 19:43.58 | trumee | is there any ATA which can do TLS/SRTP/ZRTP calls? |
| 19:44.02 | p3nguin | built a package, that is. |
| 19:44.49 | p3nguin | I don't like changes during working hours, so today is the day I'll change it if I change it. |
| 19:44.52 | trumee | moved from 1.6/Ubuntu to 1.8/Gentoo |
| 19:45.48 | *** join/#asterisk adnc (~akif@unaffiliated/adnc) |
| 19:46.08 | p3nguin | I guess I should do it. |
| 19:46.39 | adnc | hello, my incoming calls are interrupted exact after 15 minutes, has anyone got any ideas where I could search for that problem with asterisk 1.6.2? |
| 19:46.59 | p3nguin | Do you have an absolute timeout value defined? |
| 19:47.11 | adnc | p3nguin, I wouldn't know how |
| 19:47.16 | WIMPy | adnc: session-timers |
| 19:47.18 | adnc | is there something like this? |
| 19:47.20 | adnc | let me see |
| 19:47.52 | adnc | it is set to default in sip.conf |
| 19:48.41 | adnc | I remember I tried this setting after i read about a bug where someone was describing something similar |
| 19:51.54 | *** join/#asterisk imox1234 (~imox1234@p4FC5C35B.dip0.t-ipconnect.de) |
| 19:52.43 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 19:52.51 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:56.33 | adnc | anything I could look after? |
| 19:57.09 | WIMPy | adnc: session-timers (still) |
| 19:57.26 | adnc | WIMPy, what ‎could I do? |
| 19:57.57 | WIMPy | Switch them off, for example. |
| 19:58.01 | catphish_ | is there a way to set the moh class for both channels in a bridged call? |
| 19:58.12 | WIMPy | Or find out why they're not working. |
| 19:58.14 | catphish_ | ie set the "sent" musiconhold |
| 19:58.47 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 20:01.33 | p3nguin | Change the moh class of a call already in progress? |
| 20:01.57 | adnc | I just see that I had instead of session-timers session-timer set to default |
| 20:02.13 | adnc | maybe it helps if I set it to refuse |
| 20:02.30 | catphish_ | p3nguin: when setting up a call, it's possible to set the moh class |
| 20:02.34 | adnc | now i would have to make an inbound call and wait 15 minutes, is there an other way testing this |
| 20:02.41 | catphish_ | but i can only see how to set the moh that the caller hears |
| 20:02.49 | adnc | can I see this value from a cli session? |
| 20:02.53 | catphish_ | not the moh that the receiving party hears |
| 20:03.00 | WIMPy | adnc: Set the time to 300s and wait for 5min. |
| 20:03.38 | adnc | but the session-timers only accepts 'accept, default, refuse' |
| 20:03.51 | p3nguin | If I set moh class in an extension that is making a call to someone else, and then I put the call on hold, won't the called party hear the moh class that I just set? |
| 20:04.11 | GreenWolf | should |
| 20:05.09 | catphish_ | p3nguin: no |
| 20:05.34 | p3nguin | You're going to make me test it, aren't you? |
| 20:06.35 | catphish_ | lol |
| 20:07.22 | catphish_ | SetMusicOnHold - This sets what music the perticular channel will hear |
| 20:07.30 | catphish_ | meaning the channel that initiated the call |
| 20:07.41 | catphish_ | http://www.voip-info.org/wiki/view/Asterisk+cmd+SetMusicOnHold |
| 20:08.09 | WIMPy | That should be CHANNEL(moh) or something now. |
| 20:08.37 | catphish_ | yeah it is |
| 20:08.40 | p3nguin | The channels are bridged, so if my channel hears it, your channel would also hear it. This is my interpretation. |
| 20:09.01 | catphish_ | i don't think so |
| 20:10.54 | p3nguin | If the last command I ran in vim needs to be ran again, but I don't want to retype the entire command, what is the shortcut to just do the last command again? a single key to repeat it, not :, up, enter. |
| 20:11.16 | WIMPy | . |
| 20:12.24 | p3nguin | OH NO! That deleted the line! |
| 20:12.38 | p3nguin | I need to be more careful what my last action was. |
| 20:13.02 | WIMPy | famous last words? |
| 20:13.13 | p3nguin | . seems to repeat my last action rather than the last command I ran. |
| 20:13.39 | p3nguin | Such as dd was the last action, but the last command might have been :s/canreinvite/directmedia/ |
| 20:13.59 | WIMPy | Not sure if there's something more specific. |
| 20:14.27 | p3nguin | I guess ": up arrow enter" will have to do. |
| 20:14.27 | WIMPy | Well, there surely is, but I don't know. |
| 20:14.37 | WIMPy | is still a !Zap user. |
| 20:15.04 | cusco | woa |
| 20:16.28 | GreenWolf | slaps _Raptor_ around a bit with a large trout |
| 20:16.38 | wdoekes2 | p3nguin: are you trying to replace something over a limited set of lines? |
| 20:16.42 | _Raptor_ | GreenWolf: thx |
| 20:16.59 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1176001702.dsl.bell.ca) |
| 20:17.03 | GreenWolf | _raptor_: lol |
| 20:17.21 | _Raptor_ | GreenWolf: should i know you? |
| 20:17.27 | p3nguin | wdoekes2: I'm just trying to find out if there is a single keypress to repeat the last command that I ran. |
| 20:17.45 | wdoekes2 | but that's not your real goal, is it? |
| 20:17.55 | p3nguin | It's not a goal. |
| 20:18.00 | p3nguin | I'm just trying to find it out. |
| 20:18.09 | GreenWolf | _Raptor_: we talked along time ago. Just waking u up |
| 20:18.26 | p3nguin | An example is if I used dd to delete a line, then used :sh to go to a shell, then exited the shell to get back to vim... |
| 20:18.39 | nobodyshome | would anybody be able to convert this to 1.4 dialplan? apparently this is pre v1.2 |
| 20:18.40 | nobodyshome | http://pastebin.com/dYsSLHnU |
| 20:18.41 | p3nguin | if I wanted to get to the shell again, I could again type in :sh <enter> |
| 20:18.48 | _Raptor_ | GreenWolf: we did? |
| 20:18.49 | p3nguin | or I could use : <up> <enter> |
| 20:18.56 | _Raptor_ | GreenWolf: well, can't remember |
| 20:19.06 | p3nguin | I'm just looking for a single keypress that would rerun the last command. |
| 20:19.13 | p3nguin | If there isn't one, so be it. |
| 20:19.24 | GreenWolf | _Raptor_: i was using the nick IIHorrorII |
| 20:19.40 | WIMPy | p3nguin: If there isn;t one, it can certainly be defined. |
| 20:19.52 | p3nguin | wimpy's suggestion of . just did dd for me again rather than the :sh that was ran last. |
| 20:20.10 | _Raptor_ | GreenWolf: so? |
| 20:20.50 | GreenWolf | _Raptor_: haven't been around the scene in awhile figuring I'd jump back into swing of things. Say whats up to some ppl who have helped me in the past. How are things? |
| 20:21.17 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:22.11 | wdoekes2 | p3nguin: :map , ^V<ESC>:^V<UP>^V<ENTER> |
| 20:22.37 | p3nguin | neat. I may do that if I get tired of : up enter |
| 20:23.21 | wdoekes2 | (I didn't know either.. but google turned up the :map docs) |
| 20:23.49 | _Raptor_ | GreenWolf: things are pretty fine |
| 20:28.50 | p3nguin | If I specify a host in a sip template, and then create another peer referring to that template, and specify a host in the new one as well, there won't be any host address conflict, right? |
| 20:29.14 | p3nguin | I tested it, and the new peer using a previous as a template does show its own IP address. |
| 20:29.20 | p3nguin | But I want to be sure. |
| 20:29.43 | p3nguin | s/./ there will never be a problem doing it this way./ |
| 20:41.16 | *** join/#asterisk [netman] (netman@138.236.76.188.dynamic.jazztel.es) |
| 20:46.33 | *** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net) |
| 20:46.54 | Karen_m | i just setup my sip.conf, and sip reloaded, how do I know if it successfully connected? how can I get the 'sip status' or .. 'sip show' ? |
| 20:48.04 | *** part/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 20:48.43 | Karen_m | my goal is to answer the phone, record the conversation and call my cell. Anyone ever set this up before and able to give me hints? |
| 20:49.35 | ChannelZ | sip show peers or sip show registry depending on what you've done |
| 20:51.11 | Karen_m | beautiful :) |
| 20:51.29 | Karen_m | ChannelZ, do you know anything about how to answer the call, record it, and call my cell ? |
| 20:52.04 | WIMPy | Don't answer, Use MixMonitor and Dial. |
| 20:52.32 | dijib | guys what do i use to Read input numbers? |
| 20:52.44 | dijib | i want to do something like Read(VARNAME) |
| 20:52.44 | ChannelZ | Read(), ironically |
| 20:52.55 | leifmadsen | :) |
| 20:53.00 | dijib | then how do i add it to asterisk, its not listed in core show applications |
| 20:53.04 | leifmadsen | Read(varname,filename) |
| 20:53.12 | leifmadsen | then you didn't compile it in |
| 20:53.12 | ChannelZ | uhm |
| 20:53.28 | dijib | is there a seperate module for it? |
| 20:53.30 | leifmadsen | listed as app_read in Applications within menuselect |
| 20:53.40 | leifmadsen | it should be enabled by default though |
| 20:53.56 | dijib | im using an openwrt compiled build of asterisk |
| 20:54.05 | leifmadsen | then check with the build maintainer |
| 20:54.08 | dijib | openwrt 10.03 asterisk 1.6 |
| 20:56.07 | Karen_m | can i setup asterisk to do a voip->fax->email setup? where if someone sends me a fax, it will send it as an attachment in email? |
| 20:58.08 | catphish_ | yes you can |
| 20:58.24 | catphish_ | i recommend res_fax_spandsp |
| 20:59.41 | Karen_m | ok thank you catphish_ , that gives me something to google |
| 21:00.31 | leifmadsen | Karen_m: or just use hylafax |
| 21:00.54 | leifmadsen | although with that you can't use voip really -- for the voip aspect the other end point has to call you using T.38 |
| 21:01.28 | Karen_m | does hylafax use voip tho? |
| 21:01.41 | Karen_m | i'm not sure what t.38 is, voip? |
| 21:01.58 | leifmadsen | that's what google is for :) |
| 21:02.18 | Karen_m | i see this mixmonitor, but how do I know if my asterisk has it already included? |
| 21:02.27 | leifmadsen | 'core show application mixmonitor" |
| 21:03.13 | WIMPy | mismatched quotes |
| 21:03.32 | Karen_m | great, i do have it :) |
| 21:03.47 | leifmadsen | WIMPy: yes missed shift key |
| 21:04.07 | Karen_m | when a call is incoming to asterisk, is there a way to see what extension is being called? voip.ms is not triggering for me, so I'm not sure what they use |
| 21:05.00 | leifmadsen | it's probably based on how you register |
| 21:05.05 | leifmadsen | and you can see using a 'sip debug' |
| 21:05.21 | WIMPy | Ah, US and UK have " and @ exchanges, haven't they? |
| 21:05.22 | leifmadsen | look at the sip trace, then determine what the other end point is requesting |
| 21:05.32 | p3nguin | Oh, that reminds me... |
| 21:05.49 | leifmadsen | goes back to writing SQL join statements for a dialplan |
| 21:06.04 | p3nguin | I'm using fax for asterisk on 1.4 now. If I go to 1.8.5.0 as planned, will res_fax that I built in be enough for faxing? |
| 21:06.07 | Karen_m | sip debug, no such command ' sip debug' |
| 21:06.23 | p3nguin | sip set debug on |
| 21:08.55 | Karen_m | oh, sip:200 |
| 21:09.35 | Karen_m | that sip set debug on is awesome! |
| 21:10.30 | WIMPy | Is someone into S&M here? |
| 21:11.12 | Karen_m | rhianna here? |
| 21:12.33 | Karen_m | so i've got 2 numbers setup in my voip.ms. They both call on extension 200, how do you control which one you want? |
| 21:14.25 | p3nguin | huh? |
| 21:14.47 | p3nguin | Calls from voipms should go to the extension that is the phone number which has been called. |
| 21:15.00 | p3nguin | If I call your number 3145551212, you get a call to extension 3145551212. |
| 21:15.10 | p3nguin | At least normally, that's what happens. |
| 21:17.28 | *** join/#asterisk [netman] (netman@138.236.76.188.dynamic.jazztel.es) |
| 21:17.39 | Karen_m | i'm going back to their config and going to see what the debug prints.. maybe it goes to the lowest extension found by default |
| 21:18.15 | Karen_m | why do the docs say 'extensions reload' but it gives me command not found? |
| 21:19.34 | p3nguin | There's no such thing as a "lowest extension found." |
| 21:19.38 | p3nguin | dialplan reload |
| 21:19.48 | *** join/#asterisk [netman] (netman@138.236.76.188.dynamic.jazztel.es) |
| 21:21.16 | Karen_m | it always does: Looking for 200 in mycontext (... |
| 21:22.26 | p3nguin | If so, you configured it that way. |
| 21:22.47 | p3nguin | Have you pasted your sip.conf and extensions.conf yet? |
| 21:24.13 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 21:28.40 | Karen_m | register => xxxxxx:xxxxxxxxxxxxxxx@chicago.voip.ms:5060/200 |
| 21:28.46 | Karen_m | does that /200 mean extension by default? |
| 21:29.06 | ChannelZ | yes |
| 21:29.14 | Karen_m | yes it does, why did it get in there?!?! lol |
| 21:29.37 | ChannelZ | via your keyboard |
| 21:29.56 | Karen_m | lol |
| 21:30.08 | p3nguin | Remove /200 from that line. Don't do that. |
| 21:30.45 | p3nguin | Calls should go to the number of the DID. |
| 21:31.17 | Karen_m | now with /200 removed, it wants to call to: <sip:s@123.123.123.123> |
| 21:31.35 | p3nguin | wth... you're now the second person to tell me that. |
| 21:32.07 | p3nguin | My calls are sent to my phone numbers. |
| 21:32.11 | p3nguin | never s. |
| 21:32.38 | p3nguin | You have more than one DID routing to that host? |
| 21:32.49 | Karen_m | http://pastebin.com/Pwpt9bwB |
| 21:33.07 | Karen_m | i have 2 dids routed to main |
| 21:33.24 | Karen_m | [SIP] main account |
| 21:33.27 | Karen_m | both of them |
| 21:36.35 | ChannelZ | Seems like there's got to be something in their control panel to control how that behaves |
| 21:36.50 | ChannelZ | Or new customers are being setup with different settings |
| 21:37.22 | Karen_m | so in my [general], can i have multiple accounts listed there ? |
| 21:37.32 | Karen_m | i'm going to redirect the other number to a subaccount |
| 21:37.40 | p3nguin | Here is my peer entry: http://pastebin.com/rC6eGsuH |
| 21:37.53 | ChannelZ | 'general' isn't an 'account' |
| 21:38.06 | p3nguin | Oh, I should have included the register statement. One moment. |
| 21:38.53 | p3nguin | fixed. |
| 21:39.46 | p3nguin | I have several DIDs with VoIP.ms, and every call to any of the DIDs goes to an extension matching the DID that was called. |
| 21:40.40 | p3nguin | My only thought is that they are doing something weird with new sign-ups. |
| 21:41.07 | p3nguin | I looked around in accounts and DID management and I don't see anything related to sending calls to any defined extensions. |
| 21:41.33 | Karen_m | is that regsiter line under [general] ? |
| 21:42.07 | p3nguin | Register statements are required to be under the general section, before authentication section and before any peer definitions. |
| 21:42.11 | p3nguin | So yes, it is. |
| 21:42.29 | Karen_m | the only thing i really noticed so far is ... type=peer where i had.. type=friend |
| 21:42.40 | p3nguin | And you also use fromuser. |
| 21:44.24 | p3nguin | I think it was dijib who was having the same issue with calls going to s. He was supposed to contact support and report back to me, but I haven't heard anymore about it from him. |
| 21:44.57 | dijib | yeh i did, they found nothing wrong. told me to checkout my CDR i did, calls there are showing the correct CID |
| 21:45.03 | dijib | i said eff it |
| 21:45.14 | ChannelZ | it's not the caller ID that is the problem |
| 21:45.17 | p3nguin | What does CID have to do with it? |
| 21:45.21 | p3nguin | what he said. |
| 21:45.41 | p3nguin | It's the TO extension where lies the problem. |
| 21:46.02 | p3nguin | karen_m: What version of asterisk are you using? |
| 21:46.07 | p3nguin | or at least the branch |
| 21:46.11 | Karen_m | even with your sip.conf matching, ... still going to sip:s |
| 21:46.30 | Karen_m | Asterisk 1.6.2.9-2+squeeze3 built by pbuilder @ boomtime on a x86_64 running Linux on 2011-07-07 08:54:36 UTC |
| 21:46.35 | p3nguin | I'm using 1.4. That's the only thing different that I can tell. |
| 21:47.05 | p3nguin | I'm working to upgrade to 1.8.5.0 right this very moment. |
| 21:47.18 | p3nguin | Once I get that changed over, I'll eliminate that as being part of the cause. |
| 21:47.28 | ChannelZ | well I assume the s (or lack of exten) is coming along in the SIP packet and shouldn't make any difference what * version |
| 21:48.20 | p3nguin | To be fair, I feel like I need to eliminate it. |
| 21:48.25 | p3nguin | I do agree with you, though. |
| 21:48.43 | ChannelZ | I mean if the DID isn't to be found in the INVITE packet... it's not there to be had |
| 21:48.58 | ChannelZ | *unless* Asterisk is registering with 's' behind your back |
| 21:49.20 | dijib | p3nguin, i dont know... im a nUUber |
| 21:49.22 | ChannelZ | that would be the thing to look at, unregister and then re-register with debug on and see what it's actually saying to voip.ms |
| 21:49.32 | dijib | althought im creating one heck of a dialplan. |
| 21:51.33 | Karen_m | in one of the extension.conf sections.. i do see /s |
| 21:51.39 | Karen_m | exten => 500,1,Playback(demo-abouttotry); Let them know what's going on |
| 21:51.39 | Karen_m | exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo |
| 21:51.47 | Karen_m | darn, meant to paste 1 line sorry |
| 21:52.09 | Karen_m | so [default] being set to s, maybe it's hardcoded that default = s |
| 21:52.12 | ChannelZ | HAH check this out |
| 21:52.18 | ChannelZ | Contact: <sip:s@173.160.35.173:5060> |
| 21:52.39 | Karen_m | default has include demo! |
| 21:52.44 | leifmadsen | please do not use [default] |
| 21:52.53 | p3nguin | The problem with ANY extension in the register statement is that calls go to that extension only. When you have multiple DIDs on a single account, you'd have to way to distinguish one from another. |
| 21:52.55 | ChannelZ | That's with a normal register line to my ITSP (Vitelity) not specifying an exten. (I don't normally register with them, they know me by static IP) |
| 21:53.01 | leifmadsen | [default] should only ever be used for untrusted connections |
| 21:53.06 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-136-139.chyn.qwest.net) |
| 21:53.09 | Karen_m | [default] does have an ... include => demo, i'm removing EVERYTHING out of the extensions.conf |
| 21:53.10 | ChannelZ | Soooo maybe this is a change? |
| 21:53.19 | leifmadsen | Karen_m: obviously you didn't :) |
| 21:53.32 | leifmadsen | read i'm as i've -- opps |
| 21:53.34 | p3nguin | You shouldn't have anything in extensions.conf to begin with. |
| 21:53.34 | leifmadsen | oops too |
| 21:53.49 | p3nguin | The sample file shouldn't be USED. |
| 21:53.57 | leifmadsen | I copy extensions.conf.sample into /etc/asterisk, then remove everything below [globals] |
| 21:54.00 | p3nguin | Start with a blank file and add what you need. |
| 21:54.33 | leifmadsen | ^^^ also good advice |
| 21:54.46 | Karen_m | *crossing fingers* as I dial the number :) |
| 21:55.01 | Karen_m | no! |
| 21:55.13 | Karen_m | it still is trying ... <sip:s@...> |
| 21:55.16 | ChannelZ | p3nguin: for fun turn on sip debug and make your * re-register. What does the Contact: header say |
| 21:55.37 | p3nguin | Give me a few minutes... I use IAX2, so I have to switch over to another sub account that's routing to SIP. |
| 21:56.21 | p3nguin | I set up one for SIP the other day when dijib was having the problem; I had to ensure my SIP calls still went to the right extensions. And they did. |
| 21:56.31 | Karen_m | FOUND IT! |
| 21:56.37 | p3nguin | do tell |
| 21:56.42 | Karen_m | the REGISTER block says... Contact: <sip:s@....> |
| 21:56.45 | Karen_m | WHY IS THAT |
| 21:56.53 | ChannelZ | that's what I'm saying |
| 21:56.54 | leifmadsen | PLEASE STOP YELLING AS I CAN HEAR YOU FINE |
| 21:56.57 | p3nguin | bug in chan_sip's register? |
| 21:57.09 | leifmadsen | 1.6.2.9 *is* pretty old |
| 21:57.33 | Karen_m | is there a newer one on backports or something? let me go see :) |
| 21:57.49 | leifmadsen | shrugs |
| 21:57.53 | leifmadsen | I just compile Asterisk |
| 21:57.55 | p3nguin | Okay, you want me to make asterisk sip register and check the contact in sip debug? |
| 21:57.59 | ChannelZ | Yes |
| 21:58.01 | leifmadsen | 1.6.2.20 is the latest 1.6.2 |
| 21:58.07 | ChannelZ | You said you're on 1.4.x right? |
| 21:58.15 | p3nguin | yes |
| 21:59.10 | p3nguin | Contact: <sip:s@... |
| 21:59.21 | p3nguin | Now let me reroute the DID and make a call. |
| 22:00.58 | p3nguin | Call says To: <sip:the-did-I-called@myIPaddress> |
| 22:01.43 | p3nguin | Looking for the-did-I-called in voipms-inbound |
| 22:01.52 | Karen_m | p3nguin, when i call it comes thru as to: <sip:s@myip> |
| 22:02.01 | p3nguin | I don't get it. |
| 22:02.01 | Karen_m | which version are you using p3nguin ? |
| 22:02.20 | p3nguin | Asterisk 1.4.39.2 built by root @ cpe-448f on an i686 running Linux on 2011-04-12 07:54:20 UTC |
| 22:02.56 | Karen_m | i'm going to try this uupdate thing with debian :) |
| 22:03.36 | p3nguin | In my register packet, even though the contact says s, the call goes to my phone number. |
| 22:04.33 | p3nguin | In my register packet, I see both From and To say <myUserID@@chicago.voip.ms>. What does your From/To say in your register packets? |
| 22:05.31 | Karen_m | from: <sip:123456@chicago.> |
| 22:05.38 | Karen_m | to: <sip:123456@chicago> |
| 22:05.53 | p3nguin | so that's the same, too. |
| 22:05.58 | p3nguin | This is beyond me. |
| 22:06.08 | p3nguin | I can't be that damn lucky. |
| 22:06.12 | ChannelZ | so maybe it just is something on their end which is forcing it |
| 22:06.33 | p3nguin | It has to be something with new accounts, like you suggested. |
| 22:10.51 | *** join/#asterisk usrbinfoobar (~none@89.201.163.12) |
| 22:11.03 | usrbinfoobar | hey guys |
| 22:11.14 | usrbinfoobar | do you know if there's any way to use g729B with asterisk? |
| 22:11.19 | usrbinfoobar | the intel IPP should support it |
| 22:17.35 | leifmadsen | p3nguin: what is your real name? (just msg me so I can add you into a circle :)) |
| 22:21.10 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:21.21 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 22:22.06 | Karen_m | even with the 1.8.4 recompiled from debian, it still wants to contact <sip:s@..> |
| 22:25.58 | p3nguin | Asterisk has been eliminated. I've put in 1.8.5.0, and I'm still sending an s in the Contact, but calls are still coming to the DID number that I dial. I've tried two numbers, just to be sure. |
| 22:27.09 | p3nguin | chan_skinny sucks, by the way. I can't even make a call out. |
| 22:31.08 | p3nguin | And... I won't be using 1.8.5.0 today. |
| 22:31.38 | p3nguin | Well, maybe I can. There seems to be something wrong with the chan_sccp-b svn that I pulled in. |
| 22:31.49 | p3nguin | It says This version of chan-sccp-b only has support for Asterisk 1.6.x and below. |
| 22:33.01 | Karen_m | when I run mixmonitor, is there a way to make it save as a *.mp3 or something? it writes out as a ulaw |
| 22:33.38 | p3nguin | Not to mp3, but to other non-patented formats. |
| 22:33.46 | p3nguin | I use WAV. |
| 22:33.57 | Karen_m | how can I make it save as *.wav ? |
| 22:34.57 | p3nguin | Use wav as the file extension. |
| 22:35.26 | p3nguin | MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.WAV) |
| 22:36.10 | p3nguin | oops, that's what I use for WAV. If you want wav instead, change it to wav. |
| 22:36.19 | p3nguin | Yes, they are different. |
| 22:36.32 | Karen_m | wav and WAV are different? hrmm |
| 22:36.38 | Karen_m | which one will play on most computers? |
| 22:37.49 | *** join/#asterisk spck (~tom@h75-100-71-130.mdsnwi.dedicated.static.tds.net) |
| 22:38.15 | p3nguin | WAV |
| 22:38.19 | Karen_m | great! I have it recording the call, now how do I get the dialplan to call my cell and record now? |
| 22:38.19 | spck | hey guys i got a production server and a test server and i'm trying to create a sip trunk between them |
| 22:38.46 | p3nguin | Dial(SIP/voipms/yourcellnumber) |
| 22:38.54 | p3nguin | after the mixmonitor line. |
| 22:39.02 | spck | when i call from the test server the production server tries to connect to <sip:820@sip> |
| 22:39.03 | p3nguin | Do not use Answer() in the dial plan. |
| 22:39.30 | spck | what does the @sip part mean? |
| 22:40.26 | Karen_m | p3nguin, should there be a wait or a hangup in there after? I'm thinking of using just... 200,1,MixMonitor(), 200,n,Dial(SIP,voipms/mycellnumber) |
| 22:40.31 | Karen_m | is that all I should have? |
| 22:40.50 | p3nguin | Dial(SIP/voipms/mycellnumber) |
| 22:41.01 | p3nguin | That's enough. MixMonitor() first, then Dial() second. |
| 22:41.13 | p3nguin | I always end my extension with Hangup(), too. |
| 22:41.31 | p3nguin | You don't have any SIP phone that needs the call first? |
| 22:41.45 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:42.06 | Karen_m | what I am doing is, running a friends business for 3 weeks while she is gone. I need to record the phone calls for any cash deals, so there is transparency for me. I am honest but i want to provide these logs as well |
| 22:43.07 | p3nguin | That'll do it. Not sure why you're using extension 200, though. |
| 22:44.05 | Karen_m | i'm forcing 200 instead of s |
| 22:44.08 | p3nguin | Those two lines, plus an optional hangup after, should do what you're wanting to do. MixMonitor will run, but will not start recording until there is an answer on the phone. |
| 22:44.25 | Karen_m | the /200 was added on my part |
| 22:44.27 | p3nguin | As soon as the call is bridged, it'll begin recording. |
| 22:44.41 | Karen_m | the weird thing is, my skype phone declines any invite? |
| 22:44.44 | Karen_m | it won't chain the call for me |
| 22:44.48 | spck | make sure you are a one-party state |
| 22:45.05 | spck | in* |
| 22:46.47 | Karen_m | when someone calls, it just hangs up? |
| 22:46.58 | Karen_m | how do you poll the extension to wait for the cell line to hangup? |
| 22:48.00 | Karen_m | http://pastebin.com/WWF9bt2s |
| 22:48.03 | Karen_m | can someone look at that and see ? |
| 22:48.42 | p3nguin | It looks fine. Is something wrong? |
| 22:49.02 | Karen_m | i'm getting SIP/2.0 403 forbidden |
| 22:49.42 | p3nguin | humm |
| 22:49.51 | p3nguin | You're sending the right number format. |
| 22:49.56 | p3nguin | You've registered, right? |
| 22:50.03 | Karen_m | yes, it registers |
| 22:50.09 | p3nguin | That's how it sends calls to /200. |
| 22:50.10 | Karen_m | sip show peers, shows registered |
| 22:50.13 | p3nguin | okay... |
| 22:50.27 | Karen_m | can I somehow test outbound dialing from asterisk -r ? |
| 22:50.32 | Karen_m | maybe it's my outbound config messed up |
| 22:50.42 | p3nguin | Well, no, sip show peers shows that you know where the peer is. sip show registry shows if it is registered or not. |
| 22:51.00 | Karen_m | yes, both are registered (i have 2) |
| 22:51.07 | Karen_m | xxxxxx and xxxxxx_1 |
| 22:51.22 | p3nguin | originate SIP/voipms/14033331212 application playback tt-weasels |
| 22:51.28 | p3nguin | from the Asterisk CLI |
| 22:51.47 | Karen_m | maybe it's this: permit=64.120.22.242/255.255.255.255 |
| 22:51.52 | Karen_m | i don't know if that's their ip, or my ip? |
| 22:52.00 | p3nguin | It's supposed to be theirs. |
| 22:52.06 | p3nguin | It needs to match the host address. |
| 22:52.19 | Karen_m | yes it does, sec |
| 22:52.30 | p3nguin | How many peer entries do you have for voipms? |
| 22:53.19 | Karen_m | that originate line does nothing |
| 22:53.23 | Karen_m | doesn't even show any debugging |
| 22:54.39 | Karen_m | oh .. i think i may need .. .chicago.voip.s |
| 22:54.53 | p3nguin | What you need to do is use the conf I pasted for you. |
| 22:55.00 | p3nguin | using your username and secret. |
| 22:55.08 | Karen_m | i did |
| 22:55.38 | Karen_m | canreinvite=no ? |
| 22:55.40 | Karen_m | does that block it? |
| 22:55.51 | p3nguin | Then your peer name is voipms, and you call numbers through it with Dial(SIP/voipms/SOMEnumber). |
| 22:56.09 | p3nguin | That's for reinvites. If you need them, change it to yes. |
| 22:57.44 | Karen_m | that is what was blocking it, seems like |
| 22:57.51 | Karen_m | i called it, it went to my voice mail now lol |
| 22:57.52 | Karen_m | this is fun |
| 22:58.42 | Karen_m | i love you p3nguin |
| 22:58.46 | Karen_m | amazing :) |
| 22:58.54 | Karen_m | amazing grace, how sweet the sounds |
| 22:59.03 | p3nguin | canreinvite doesn't block anything, so that's not what made it work. |
| 22:59.03 | Karen_m | that ... something something something or other .. the somethingggggg |
| 22:59.19 | Karen_m | that canreinvite=no was making the outbound calls to be forbidden |
| 22:59.28 | Karen_m | it would be .. INVITE ... forbidden.. 403 |
| 22:59.36 | Karen_m | changed it to =yes, and now no more INVITE 403's |
| 22:59.43 | Karen_m | not sure why it want's to invite vs just call out or something |
| 22:59.53 | p3nguin | I don't see how that's possible. It just allows or disallows reinvites between the phone and the provider to occur. |
| 23:00.27 | p3nguin | If it is set to no, it will keep asterisk in the media path of the call. |
| 23:01.03 | p3nguin | If set to yes, and nothing else is forcing it to stay in the path, it'll get out of the way and the media stream will go directly between the two end points. |
| 23:02.07 | Karen_m | ok, that was not it, i changed it back to canreinvite=no and it's working now |
| 23:02.10 | Karen_m | not sure what was wrong |
| 23:02.22 | Karen_m | i think it was because i had multiple [general] register's |
| 23:02.27 | Karen_m | i don't know how to associate them or something |
| 23:02.42 | Karen_m | like, if you have 2 register lines under [general], how does it pickup [voipms] .. |
| 23:02.45 | p3nguin | You can have several register statements. |
| 23:03.01 | p3nguin | Go on down the file. Look for [voipms]. |
| 23:03.08 | ChannelZ | ugh. Spent the last 45 mins doing parental tech support. Did you make any determinations on the differences with the whole s-vs-DID exten? |
| 23:03.15 | p3nguin | Register statements tell THEM how to reach your device. |
| 23:03.51 | Karen_m | p3nguin, but i mean.. what if i called it .. [voipms] and [heyimslow] |
| 23:04.05 | Karen_m | do all those blocks get available to each register? |
| 23:04.19 | p3nguin | Then you'd have to go change the dial plan to Dial(SIP/heyimslow/numberhere) |
| 23:04.38 | Karen_m | how does it know which account to call out on tho? |
| 23:04.46 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:04.46 | p3nguin | see above |
| 23:04.55 | p3nguin | I switched over to 1.8.5.0, my Contact was still s@... and the To/From was still myUserID@chicago. Calls still go to my phone numbers as the extension. |
| 23:05.01 | Karen_m | heyimslow will use which of the register lines i meant |
| 23:05.10 | ChannelZ | So it's something on their end then. |
| 23:05.11 | p3nguin | zero |
| 23:05.26 | ChannelZ | Karen_m: Registering has almost nothing to do with calling |
| 23:05.38 | p3nguin | There is no association between register statements and the peer definition. |
| 23:05.44 | Karen_m | oh the username is the associate |
| 23:05.46 | Karen_m | association |
| 23:05.47 | Karen_m | i see |
| 23:05.48 | ChannelZ | Registeringj ust tells the other end who you are, what IP you're at, and what extension you want them to send you things. |
| 23:05.53 | p3nguin | There is no association between register statements and the peer definition. |
| 23:05.55 | p3nguin | none |
| 23:06.00 | p3nguin | zero, zilch |
| 23:06.10 | p3nguin | VoIP.ms does require you to be registered before you can make calls, though. |
| 23:06.29 | p3nguin | That is apparently a feature of SER. |
| 23:06.30 | dijib | ok, have an gotoif or if-else or if-then-else |
| 23:07.00 | dijib | issue, how do i do an if VAR = 1, then, goto. if-else Dial(VAR) |
| 23:07.06 | dijib | does that make any sense? |
| 23:07.59 | spck | dijib you have to stack them up |
| 23:08.14 | dijib | got any guides on this? |
| 23:08.27 | Karen_m | this is awesome! |
| 23:08.34 | spck | like GotoIf($[${VAR}=1]?first:second) |
| 23:08.36 | Karen_m | so, the *.WAV file sounds terrible, is there a better sounding codec? |
| 23:08.51 | p3nguin | GotoIf($["${VAR}" = 1]?:labeliffalse) |
| 23:09.02 | p3nguin | Stuff() |
| 23:09.05 | p3nguin | Hangup() |
| 23:09.24 | dijib | i think i need a gotoif guide |
| 23:09.27 | p3nguin | (labeliffalse),OtherStuff() |
| 23:09.33 | p3nguin | I'll write it for you. |
| 23:10.44 | p3nguin | What do you want it to do if true? |
| 23:11.10 | p3nguin | Goto() some other place in the dial plan? |
| 23:11.25 | dijib | if VAR=1 then dial(100) ifelse dial VAR |
| 23:13.14 | p3nguin | http://pastebin.com/LpuVTVru |
| 23:13.48 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:14.20 | p3nguin | or http://pastebin.com/jj1eXeCv |
| 23:15.03 | p3nguin | take your pick, the logic is the same. |
| 23:15.18 | spck | i can't get this sip trunk to register between my two boxes and it's driving me batty |
| 23:16.43 | spck | it honestly looks like one of the boxes is trying to register to itself, but i'm not sure how |
| 23:17.19 | Karen_m | does anyone know the config you would setup for res_fax_spandsp ... what would the exten look like? |
| 23:17.29 | Karen_m | pls and thanks |
| 23:17.30 | Karen_m | :) |
| 23:17.37 | p3nguin | karen_m: core show applications like fax |
| 23:18.20 | Karen_m | receiveFAX and SendFax |
| 23:18.33 | p3nguin | core show application SendFax |
| 23:18.38 | p3nguin | core show application ReceiveFAX |
| 23:18.46 | *** join/#asterisk jetlag (jetlag@pool-71-188-2-156.cmdnnj.east.verizon.net) |
| 23:19.15 | Karen_m | that is brilliant! |
| 23:19.16 | Karen_m | thank you |
| 23:19.22 | Karen_m | to me, it looks like 1 line |
| 23:19.24 | Karen_m | ReceiveFAX |
| 23:19.25 | Karen_m | is that it? |
| 23:19.31 | spck | any ideas what i'm doing wrong with this sip trunk? |
| 23:19.41 | ChannelZ | One simple line, yet a world of pain making it work |
| 23:19.51 | Karen_m | exten => 200,1, ReceiveFAX(/home/karen/faxes/blah) |
| 23:20.09 | Karen_m | ChannelZ, the pain for who? the people that setup the ReceiveFAX line? lol |
| 23:20.13 | ChannelZ | spck: let's see some config |
| 23:20.32 | spck | channelz: what would you like to see? |
| 23:20.36 | ChannelZ | Karen_m: depends. If you're doing ulaw/alaw and have a good stable connection between you and your ITSP, it might be fine. |
| 23:20.48 | p3nguin | I use fax for asterisk, but I'll give you my dialplan for it. |
| 23:20.51 | ChannelZ | Karen_m: if you're trying to do t.38 that's where things get nice and complicated |
| 23:20.54 | dijib | p3nguin, or anyone. how did i mess this up? |
| 23:20.55 | dijib | http://pastebin.com/7xnCSmFS |
| 23:21.15 | ChannelZ | spck: sip.conf and some console output or better explanation other than 'it doesn't work' |
| 23:21.18 | p3nguin | numbered priorities = first mistake |
| 23:21.42 | ChannelZ | you have priority 7 like 5 times |
| 23:21.51 | spck | dijib: i would say this line: exten => 600,11(labeliffalse),Goto(${OUTNUM},60) |
| 23:22.04 | spck | Goto(${OUTNUM},60) |
| 23:22.39 | p3nguin | See comment about numbered priorities. |
| 23:25.08 | p3nguin | http://pastebin.com/Pt9UMsSK |
| 23:26.11 | p3nguin | And then you HAVE TO HAVE an extension in that same context that matches the value of ${OUTNUM}. |
| 23:26.19 | p3nguin | Otherwise, the Goto() has nothing to do. |
| 23:27.46 | p3nguin | I think I like the other form of the GotoIf() now that I've written both and looked at them side by side. |
| 23:28.11 | p3nguin | But this version makes more sense to me. |
| 23:28.31 | p3nguin | If true, continue to the next line. If false, jump somewhere else. |
| 23:28.48 | p3nguin | I'M SO CONFUSED |
| 23:28.57 | *** join/#asterisk luckman212 (~do-not-re@pool-72-76-39-193.nwrknj.fios.verizon.net) |
| 23:30.30 | spck | hmm, working on a pastebin, but something changed |
| 23:30.49 | luckman212 | dear lord can someone for the love of god please tell me how to FACTORY erase a polycom IP335? i had one that i wasn't using, it was provisioned on my internal asterisk server that had a tftp server... so that was all great, but now I'm over at his house trying to set it up for him and the thing just _wants_ to still provision on my old server which is of course not accessible. been working on this phone for 2+ hours now |
| 23:30.55 | dijib | im confused |
| 23:31.03 | ChannelZ | luckman212: hammer |
| 23:31.10 | luckman212 | i know there is a "format file system" but that will leave me with a phone that won't even boot |
| 23:31.17 | luckman212 | hehe ChannelZ yeah I am at that point now |
| 23:31.19 | spck | luckman212: hold 1357 i think |
| 23:31.44 | catphish_ | http://www.google.com/search?client=ubuntu&channel=fs&q=polycom+IP335+factory+reset&ie=utf-8&oe=utf-8 |
| 23:31.50 | luckman212 | spck: christ I think that might be working |
| 23:32.05 | spck | i've had to do it a few times |
| 23:32.10 | p3nguin | dijib: about what? |
| 23:32.40 | catphish_ | yes, hold 1357 |
| 23:32.43 | dijib | just what im trying to do here. when i put in '1' as OUTNUM. it tris to dail 1 and not 100 |
| 23:32.46 | catphish_ | then it'll ask for a password |
| 23:32.46 | dijib | SIP/100 |
| 23:32.58 | p3nguin | yeah? |
| 23:33.03 | catphish_ | By default, this password is 456 |
| 23:33.11 | dijib | hey my 's' extension fixed itself somehow. |
| 23:33.16 | luckman212 | yep I do know about 45t6 |
| 23:33.21 | luckman212 | er 456 |
| 23:33.38 | p3nguin | dijib: Now calls are going to ${DID}? |
| 23:33.42 | luckman212 | it's stuck / locked up now on that "enter password" screen... i guess it's wiping itself (hopefully) |
| 23:33.48 | dijib | yes |
| 23:33.52 | p3nguin | Did you email them? |
| 23:33.53 | catphish_ | luckman212: is your ability to enter "polycom IP335 factory reset" into google failing? :) |
| 23:33.53 | dijib | weird eh. i didnt do a thing |
| 23:34.03 | dijib | yeh but they left it with me. let me check the ticket |
| 23:34.16 | luckman212 | catphish_: no but my patience for this f------g phone is |
| 23:34.21 | catphish_ | lol |
| 23:34.54 | spck | also you can use the mac address of the phone as the password |
| 23:35.06 | luckman212 | what was i thinking offering to help my friend set up an asterisk box in his house... been here since sat afternoon working on this crap |
| 23:35.13 | p3nguin | I use the MAC as the peer name and generate a password. |
| 23:35.20 | p3nguin | apg = win |
| 23:35.37 | p3nguin | alias apg='apg -a1 -m13 -n33 -p6' |
| 23:36.00 | spck | what is apg? |
| 23:36.08 | catphish_ | i assume asterisk password generator? |
| 23:36.09 | p3nguin | a passwd generator |
| 23:36.10 | spck | nm |
| 23:36.12 | ChannelZ | Apple Pig Groomer |
| 23:36.12 | catphish_ | ah |
| 23:36.33 | catphish_ | not sure why a password generator would be asterisk specific :) |
| 23:36.38 | p3nguin | It's not. |
| 23:36.56 | p3nguin | It makes awesome passwords usable in asterisk, though. |
| 23:37.18 | catphish_ | http://xkcd.com/936/ |
| 23:37.34 | p3nguin | Oh, here's the one I had to use for my Cisco phones when I used SIP with them: apg -a1 -n1 -m13 -x26 -MSNCL -E^[]{}\"? |
| 23:37.59 | p3nguin | I was not able to enter those special characters on the phone display via keypad. |
| 23:38.12 | p3nguin | So I had to exclude them. |
| 23:38.53 | spck | i was getting a 403 before, but now i'm getting: SIP/2.0 401 Unauthorized |
| 23:39.06 | ChannelZ | yay! |
| 23:39.08 | luckman212 | man this f--------g phone is just sitting there saying "Uploading log file" |
| 23:39.35 | spck | did you set the server in setup before the phone booted? |
| 23:40.00 | luckman212 | yes I did, used the LAN ip of the * server |
| 23:40.05 | luckman212 | 192.168.x.x |
| 23:40.07 | catphish_ | luckman212: please mind your language a little |
| 23:40.21 | spck | do you have an ftp server running on it? |
| 23:40.25 | p3nguin | Looks censored to me. |
| 23:40.28 | spck | check the transfer log |
| 23:40.29 | ChannelZ | Yes I find ----- highly offensive |
| 23:40.33 | catphish_ | think of the (very technically advanced) children |
| 23:40.38 | luckman212 | catphish_: i used '----'s thats not enough? sorry i was frustrated |
| 23:40.45 | spck | fuck that |
| 23:40.50 | catphish_ | lol |
| 23:40.50 | ChannelZ | YEAH! |
| 23:40.51 | p3nguin | hahahaha |
| 23:40.51 | p3nguin | bad |
| 23:41.06 | catphish_ | censoring spear words is pointless, they still read as swearwords |
| 23:41.08 | spck | hunter2 |
| 23:41.11 | luckman212 | anyway these kids today know more curses than I do... YAY internet |
| 23:41.22 | ChannelZ | catphish_: so? they obviously already know the swear words then |
| 23:41.52 | catphish_ | i was joking about the children, it was just getting on my nerves a little, not to worry though, there's no actual rules against it |
| 23:42.00 | catphish_ | it just makes me less inclined to help |
| 23:42.21 | p3nguin | That is your prerogative. |
| 23:42.51 | *** join/#asterisk jetlag (jetlag@pool-71-188-2-156.cmdnnj.east.verizon.net) |
| 23:43.09 | catphish_ | maybe i'm just getting grumpy in my old age |
| 23:43.29 | ChannelZ | F--K PU-S- S-X P-N-S BOO-S --NDOM |
| 23:45.15 | spck | Channelz: here's what i got: http://pastebin.com/xMFeYncQ |
| 23:46.11 | spck | http://pastebin.com/UyYeRQ0f |
| 23:46.14 | spck | that one actually |
| 23:46.34 | catphish_ | is anyone really knowledgeable about realtime around? |
| 23:46.36 | ChannelZ | why /sip on the end of your register line? |
| 23:46.58 | ChannelZ | and you are referring to dialing SIP/sip but your peer is really called 'asterisk-test' |
| 23:47.04 | p3nguin | He apparently wants calls going to SIP/sip. |
| 23:47.16 | p3nguin | Wait, no. |
| 23:47.23 | p3nguin | He apparently wants calls going to exten => sip |
| 23:47.31 | p3nguin | you screwed me up. |
| 23:47.33 | catphish_ | i worry that the majority of asterisk users only configure by copy/paste from the web |
| 23:47.39 | catphish_ | i know i did for a while |
| 23:47.46 | ChannelZ | has to get back to writing his primer |
| 23:47.51 | p3nguin | they do |
| 23:48.25 | p3nguin | A huge percentage of the people who come here asking for help never bothered to read the asterisk books. |
| 23:48.30 | catphish_ | maybe asterisk should include some plagiarism detection algorithms |
| 23:48.35 | p3nguin | If they had, they wouldn't be asking for the same help. |
| 23:48.43 | catphish_ | i just read the wiki |
| 23:48.54 | p3nguin | the old, outdated one? |
| 23:48.59 | catphish_ | it's mostly inaccurate and ancient |
| 23:49.10 | catphish_ | but the deprecation messages help with that |
| 23:49.18 | ChannelZ | It still gets updated, it's just a mess. |
| 23:49.25 | catphish_ | it is a mess |
| 23:49.29 | p3nguin | People who rely on voip-info without having read any of the books need to be beaten within inches of their lives. |
| 23:49.43 | catphish_ | lol, i do as far as its right |
| 23:50.01 | p3nguin | Without reading something good, how will you know if it is right or wrong? |
| 23:50.23 | catphish_ | because it works as it describes or it doesn't |
| 23:50.37 | spck | i was confused on what that should be myself |
| 23:50.43 | catphish_ | in many cases it doesn't |
| 23:50.46 | spck | sip is the name of the production server |
| 23:50.54 | ChannelZ | Dial(SIP/peername/number) |
| 23:51.11 | spck | is peername the server i'm trying to connect to? |
| 23:51.15 | ChannelZ | "asterisk-test" is the name of your peer as I see it, and you're trying to dial exten 810 on it. |
| 23:51.22 | p3nguin | It's the peer name as you have configured it in sip.conf. |
| 23:51.24 | ChannelZ | (well you're not, you're trying to dial the peer 'sip' which doesn't exist) |
| 23:51.38 | p3nguin | The Book would have told you ALL about this. |
| 23:51.43 | p3nguin | points |
| 23:51.51 | spck | i read the book and set it up that way |
| 23:51.56 | spck | apparently not i guess |
| 23:52.02 | p3nguin | They have a peer called "sip" in sip.conf? |
| 23:52.14 | spck | production or test? |
| 23:52.36 | spck | i'll try switching it |
| 23:53.19 | catphish_ | it'd be nice if someone read the 10.0 source code and replaced every page of the voip-info wiki with the correct information |
| 23:53.32 | catphish_ | but i somehow doubt anyone has time |
| 23:53.51 | ChannelZ | there's wrong things in sip.conf.sample |
| 23:53.52 | p3nguin | That sounds like a huge an impractical undertaking. |
| 23:53.55 | p3nguin | s/an/and/ |
| 23:54.28 | catphish_ | that bot is cool |
| 23:54.49 | catphish_ | s/cool/unnecessary |
| 23:54.54 | catphish_ | s/cool/unnecessary/ |
| 23:54.58 | catphish_ | damn |
| 23:55.04 | ChannelZ | and literal |
| 23:55.12 | catphish_ | i tried to be funny |
| 23:55.14 | catphish_ | it backfired |
| 23:55.50 | spck | so my test box registers to the production box via: register => asterisk-test:password@sip.XXXXX.com/sip |
| 23:56.09 | spck | should that be: asterisk-test:password@sip.XXXXX.com/asterisk-test ? |
| 23:56.12 | ChannelZ | no |
| 23:56.18 | p3nguin | So the other side can send calls to extension "sip" |
| 23:56.23 | ChannelZ | remove /sip off the first one |
| 23:56.38 | spck | off the register? |
| 23:56.40 | ChannelZ | yes |
| 23:56.44 | spck | k |
| 23:57.01 | catphish_ | otherwise you're forcing it to send a specific number with each call |
| 23:57.01 | p3nguin | The /sip tells the OTHER side to send all calls for you to extension 'sip' |
| 23:57.19 | catphish_ | rather than the number that was actually dialed |
| 23:57.31 | *** join/#asterisk jetlag (jetlag@pool-71-188-2-156.cmdnnj.east.verizon.net) |
| 23:57.51 | ChannelZ | Now on your test box your exten 8001 is doing a Dial(SIP/sip/810) which is also wrong |
| 23:57.52 | p3nguin | And unless you have exten => sip,1,Stuff(), you won't be getting any calls to extension 'sip' |
| 23:58.15 | spck | ok, what should it be? |
| 23:58.20 | ChannelZ | Make it Dial(SIP/asterisk-test/810) -- assuming 810 even exists |
| 23:58.30 | spck | it does on the production server |
| 23:58.34 | ChannelZ | or rather.. we haven't seen the sip.conf on your test box |
| 23:58.43 | p3nguin | That's 810 on the server called asterisk-test. |
| 23:59.04 | ChannelZ | your paste is confusing because part of it was from one server and part of it from another. |
| 23:59.14 | spck | i'm trying to call an extension on the other server tho |
| 23:59.37 | ChannelZ | Without having an 'outgoing' peer setup from your test box, you could try Dial(SIP/sip.unioncab.com/810) instead |