| 00:30.09 | p3nguin | I've experienced my first negative incident with VoIP.ms today. |
| 00:31.41 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 00:34.17 | carrar | air it out! |
| 00:35.42 | p3nguin | One of my clients uses VoIP.ms... I managed the PBX, but I never had portal access -- I just configured the voipms peer with the information I was given. Today I get a call asking if the PBX had been hacked. |
| 00:35.59 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 00:36.28 | p3nguin | "No, I haven't seen anything unusual going on. What's up?" |
| 00:36.48 | p3nguin | Turns out, they've had over $140 in international calls made today. |
| 00:37.47 | p3nguin | The person who has always dealt with the billing/payments had a chat with voipms, in my absence, to try to figure out what happened. |
| 00:38.02 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 00:38.46 | p3nguin | She is told that an unusual pattern was detected; all the calls had been made from two IP addresses, once which is a Romania IP address and the other a China address. |
| 00:39.44 | pdtpatrick1 | QUestion ... queue show <queuename> .. keeps should an agent is invalid.. i've tried running agent logoff Agent/number ... but this particular agent just won't logoff |
| 00:39.47 | p3nguin | They accuse us of having an insecure PBX, which might be a likely problem for someone who doesn't know anything, but I feel like I pay careful attention to things like that. |
| 00:39.49 | pdtpatrick1 | how can i force them to log off ? |
| 00:40.25 | [TK]D-Fender | pdtpatrick1: Show us the queue and your attempts |
| 00:40.27 | [TK]D-Fender | ~pb |
| 00:40.27 | 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. |
| 00:40.29 | [TK]D-Fender | ^^^^^ |
| 00:41.32 | pdtpatrick1 | here |
| 00:41.33 | pdtpatrick1 | http://pastebin.com/RXggFc1D |
| 00:41.34 | p3nguin | So within the same chat where she was told it's the fault of the PBX admin, she's told the calls were made from Romania and China. Our PBX is in the USA! If our PBX was "hacked," the calls would have been from our USA IP address.... right?! |
| 00:41.51 | carrar | no |
| 00:42.17 | p3nguin | Just wait... you'll see where I'm going with this. |
| 00:42.48 | p3nguin | Typically you'd think the calls would go through the unsecured PBX, so the call would show up as coming from the PBX. |
| 00:42.54 | [TK]D-Fender | p3nguin: Yes |
| 00:43.04 | p3nguin | So we introduce this fact to the voipms person. |
| 00:43.26 | pdtpatrick1 | any ideas regarding the Agent not being logged off or Invalid ? |
| 00:43.47 | [TK]D-Fender | p3nguin: "hacked" in the sense that yes the call may not have originated from your equipment... however they may have gotten into the box to steal the credentials |
| 00:43.50 | p3nguin | Well, now our system must have been compromised and the attacker simply took the account credentials from the conf and used it directly. |
| 00:44.02 | p3nguin | Yeah, right, that's what he suggested. |
| 00:44.32 | p3nguin | So I'm poking around trying to figure out just what the hell happened. The account used to make the calls on voipms isn't even the account we use on the PBX!! |
| 00:44.34 | [TK]D-Fender | pdtpatrick1: Logging off an agent doesn't stop them from being a member of a queue |
| 00:44.45 | [TK]D-Fender | pdtpatrick1: Think on that a little... |
| 00:44.48 | pdtpatrick1 | that Agent does not even exist |
| 00:45.05 | pdtpatrick1 | there's nothing that references 2000 or 2001 in agents.conf |
| 00:45.12 | [TK]D-Fender | p3nguin: Then that's a total "WTF". Not your account = how is it your problem? |
| 00:45.14 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 00:45.35 | [TK]D-Fender | pdtpatrick1: (dynamic) <--- |
| 00:45.47 | p3nguin | It's still there "account," but I'm talking about account numbers (aka username/defaultuser). |
| 00:45.57 | p3nguin | Think sub-accounts. |
| 00:46.43 | p3nguin | The sub account used on the PBX is not the account used to make the calls. So if someone broke in and stole the credentials, how are they making calls on an unrelated account? |
| 00:47.08 | p3nguin | So apparently someone either gave out credentials to another account or voipms underwent a bruteforce and didn't know it. |
| 00:47.34 | p3nguin | Either way, they blame us for having an unsecured PBX, when there is no evidence of this being factual. |
| 00:48.21 | pdtpatrick1 | [TK]D-Fender, what would u suggest |
| 00:48.33 | [TK]D-Fender | p3nguin: I believe the term is "slander". If they put up too much of a fight, ask your lawyer about that term :) |
| 00:48.59 | p3nguin | They offered to credit back their profit on the calls made. I thought that was polite at least. |
| 00:49.06 | [TK]D-Fender | pdtpatrick1: I suggest you think a little bit about where you dynamically added a queue member |
| 00:49.26 | SeRi | p3nguin, hacked account maybe? |
| 00:49.47 | p3nguin | I think I would have done a little more checking to see what was going on before I accused a customer of something that seems very apparent to not have happened. |
| 00:49.48 | [TK]D-Fender | p3nguin: That's not bad. See about having them restrict the ip reg range, etc and do see if you have any hardening to do |
| 00:50.01 | carrar | If you have a idle account with a bad password thats your fault actually |
| 00:50.05 | [TK]D-Fender | (That's what SHE said..._ |
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| 00:51.02 | carrar | any account that is not actively used should be disabled! |
| 00:51.11 | p3nguin | I wish they would have given me access to the portal months ago and I could have reduced (not necessary prevented) the chance of this happening. I didn't get portal access until today AFTER this happened. |
| 00:51.31 | p3nguin | So when I get in there, there's a huge list of sub accounts. |
| 00:51.36 | p3nguin | We use ONE. |
| 00:51.39 | carrar | lessoned learned, move on! :) |
| 00:52.04 | p3nguin | Now I see why they hired me to take over... that other guy was a tool. |
| 00:52.09 | carrar | heh |
| 00:52.26 | p3nguin | He's the same one who demanded I give him root access to "his" server. |
| 00:52.31 | carrar | should have requested portal access at the start? |
| 00:52.35 | p3nguin | I asked. |
| 00:52.53 | p3nguin | They were confident that it wasn't needed. |
| 00:53.09 | carrar | I would have said, obviosuly |
| 00:53.24 | carrar | thats why you fired your last guy |
| 00:54.04 | p3nguin | Oh, and since the idiot put in a callerID override on every sub-account in the portal, every one of those international calls appears to have come from his phone number. |
| 00:54.47 | p3nguin | If that wouldn't have happened, there's a small chance that I could have captured at least one real caller id of a caller. |
| 00:55.02 | carrar | maybe the previous guy did it |
| 00:55.10 | p3nguin | I actually thought about that. |
| 00:55.20 | p3nguin | But when I saw the override, I figured he didn't. |
| 00:56.08 | p3nguin | He just fouled up all kinds of settings, but I doubt he called Bulgaria and Estonia. |
| 00:59.49 | p3nguin | I wondered if there was any chance he sold or gave away credentials for an unused account. |
| 01:00.25 | p3nguin | I can't imagine he'd do that, though. I think he still works there, just doesn't get to touch systems anymore. |
| 01:00.33 | p3nguin | He was hired as a sales person anyway. |
| 01:00.53 | p3nguin | (which explains why he's such a dumbass when it comes to this stuff) |
| 01:02.55 | p3nguin | I guess I need to see if voipms has any logs of any bruteforce attempts before those calls started. |
| 01:03.56 | p3nguin | If the calls just suddenly started, the caller obviously knew the credentials ahead of time. That would be a good bit of info to know. |
| 01:04.29 | p3nguin | I almost wish it would have been through the PBX so I could have tracked the process. :/ |
| 01:06.09 | p3nguin | At 46 cents/minute, it doesn't take long to run up a phone bill. Maybe I should call the number they called the most and try to get some info out of that person. |
| 01:07.42 | p3nguin | The average call durations ranged from 15 to 30 minutes per call. |
| 01:09.24 | p3nguin | Hmm. Someone explain this to me... how are all these calls overlapping in duration? |
| 01:10.06 | p3nguin | One call starts at 10:38 and lasts 18 minutes. The next call starts at 10:40 and lasts 29 minutes. The next call starts at 10:57. |
| 01:10.26 | p3nguin | This makes me think the number being called is not the real destination. |
| 01:10.50 | p3nguin | It must be forwarded somewhere else and is being controlled during the calling. |
| 01:11.38 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 01:11.38 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:11.42 | p3nguin | It's a "Bulgaria Mobile" number. The phone can't accept a dozen calls all at the same time. |
| 01:14.08 | *** join/#asterisk dhorner_mb (~dhorner_m@184.18.32.136) |
| 01:18.42 | p3nguin | Hmm, interesting. I'm looking over old support tickets now, and I found one from Nov 8 2010 that indicates something similar happened back then. |
| 01:19.04 | p3nguin | We have noticed an unusual pattern of calls coming from your sub account xxxxxxx ... |
| 01:19.27 | kb3ien | can asterisk generate a 484 error? |
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| 01:34.49 | SeRi | p3nguin, everything will fix in a large flat rate box. thats the cheap route |
| 01:34.59 | SeRi | fit* |
| 01:37.18 | wonderworld | confbridge 10 absolutely rocks. thanks guys. |
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| 01:43.56 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 01:44.31 | nny | Getting: chan_dahdi.c:5114 dahdi_confmute: DAHDI confmute(0) failed on channel 1: Invalid argument after doing an update (1.6 to 1.8 latest, dahdi, libpri and wanpipe drivers). Is this a sangoma issue or an asterisk issue (or both)? Not sure where to start |
| 01:45.13 | nny | note this is when doing a simple Dial, nothing fancy |
| 01:52.00 | p3nguin | seri: $14.95 plus $0.70 for delivery confirmation? |
| 01:54.00 | nny | looks like a bug between newer version of dahdi and sangoma driver, asking in that channel |
| 01:55.14 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 01:59.21 | SeRi | p3nguin, Yes |
| 01:59.31 | SeRi | flat rate comes with tracking all ready |
| 01:59.43 | p3nguin | Actually it doesn't. |
| 02:00.06 | SeRi | all priority mail comes with tracking |
| 02:00.12 | p3nguin | Nope. |
| 02:00.21 | SeRi | hu? really? |
| 02:00.32 | SeRi | I all ways get tracking with my priority. |
| 02:00.43 | p3nguin | What tracking do you pay for? |
| 02:00.45 | SeRi | odd maybe I am getting charged and not know it... |
| 02:01.01 | p3nguin | The light green delivery confirmation is $0.70 extra. |
| 02:01.14 | SeRi | I was not aware I was paying for it. The recipt all ways had a tracking number. |
| 02:01.19 | p3nguin | That's the cheapest one. |
| 02:01.22 | SeRi | ooo ye thats for signature confirmation |
| 02:01.33 | p3nguin | Signature is like 1.20 or something. |
| 02:02.51 | SeRi | mhhh ok I guess I got it allc onfused. |
| 02:02.58 | SeRi | but yes thats the price. |
| 02:03.00 | p3nguin | Signature confirmation is $2.45 |
| 02:03.44 | p3nguin | It's the certificate of mailing that $1.15 |
| 02:03.49 | *** join/#asterisk gxdssoft (~gxdssoft@190.234.164.35) |
| 02:04.06 | SeRi | ok. |
| 02:04.35 | SeRi | cool. well you still want the stuff? |
| 02:04.43 | p3nguin | Yes. |
| 02:05.27 | SeRi | ok. than ill go by and drop it off sometime this week before sat. thats including the sim. |
| 02:05.51 | p3nguin | Okay, great. |
| 02:06.05 | p3nguin | It'll only take 2-3 days to get priority mail. |
| 02:06.10 | SeRi | once I have a receipt than you can pay pal me. |
| 02:06.17 | SeRi | Yes Sr. |
| 02:07.08 | SeRi | right now I am on the hunt for a sip phone |
| 02:07.26 | SeRi | I was told gigaset are good for home/office |
| 02:08.05 | p3nguin | Are you looking for a wireless phone or a desk phone? |
| 02:08.18 | SeRi | ether or. |
| 02:08.25 | SeRi | I like desk phones best |
| 02:08.34 | p3nguin | I use Cisco, but Aastra makes good phones. |
| 02:08.45 | SeRi | yes I was told ether Astar or cisco. |
| 02:09.01 | SeRi | but cisco scares me because I have to buy extra licenses etc,.... or so i was told |
| 02:09.07 | SeRi | astra* |
| 02:09.12 | carrar | PHEAR |
| 02:09.14 | p3nguin | To use them "legally" |
| 02:09.35 | SeRi | ah I see. |
| 02:09.41 | carrar | Polycom are awesome |
| 02:09.43 | p3nguin | Nothing stops the phones from working if you don't buy the licenses. |
| 02:10.15 | carrar | Cisco phones Look pretty and feel nice |
| 02:10.17 | SeRi | carrar, I have a 501 that just wont do crap. :( actually several of them |
| 02:10.30 | carrar | 501 is old |
| 02:10.39 | carrar | dump it |
| 02:10.46 | p3nguin | If you can fit more in that box, feel free to include them. |
| 02:10.48 | carrar | or learn how to configure it via ftp |
| 02:11.06 | SeRi | p3nguin, LOL realx! LOL |
| 02:11.08 | p3nguin | I don't know too much about aastra phones other than they are nice phones. |
| 02:11.17 | carrar | Aastra are nice also |
| 02:11.31 | carrar | though I would probably pick Polycom over Aastra |
| 02:11.38 | p3nguin | I've never had an occasion to configure an Aastra. |
| 02:11.52 | carrar | I have |
| 02:11.58 | p3nguin | Is it a bother? |
| 02:12.07 | carrar | i don't think so |
| 02:12.17 | p3nguin | Pretty standard for an IP phone, then? |
| 02:12.17 | carrar | jsut different |
| 02:12.44 | carrar | Their deskphone with cordless phone is a nice feature however |
| 02:12.59 | SeRi | I am looking for something with a webui... I want to stay away from cfg files and tftp stuff.... |
| 02:13.10 | carrar | haha |
| 02:13.17 | carrar | no phone as a "great" web ui |
| 02:13.19 | *** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net) |
| 02:13.28 | SeRi | I know :( lol |
| 02:13.42 | SeRi | I guess I have no choice but to setup a tftp server |
| 02:13.45 | p3nguin | That's what I need. I've been wanting to install an IP phone near my couch, but I don't want to run new cabling. A good quality wireless phone would be perfect. |
| 02:13.50 | carrar | If you really want to do it right, take the time to learn config files |
| 02:14.11 | carrar | Aastra makes some great DECT SIP phones |
| 02:14.22 | carrar | as does polycom |
| 02:14.37 | carrar | stay away from wifi phones |
| 02:14.49 | carrar | unless you live miles from anyone |
| 02:15.02 | SeRi | carrar, Thanks for the advice. I almost bought a gigaset |
| 02:15.10 | carrar | never heard of them |
| 02:15.23 | carrar | probably crap! |
| 02:15.26 | carrar | heh |
| 02:15.27 | SeRi | lol |
| 02:15.32 | WIMPy | Make sure it's CAT-ip, not DECT |
| 02:15.37 | WIMPy | s/ip/iq/ |
| 02:15.52 | carrar | you want DECT |
| 02:15.58 | carrar | for cordless phones |
| 02:16.01 | carrar | SIP DECT |
| 02:16.27 | WIMPy | CAT-iq = DECT v2 |
| 02:16.33 | dijib | p3nguin, why IP? |
| 02:16.39 | carrar | anyways |
| 02:16.45 | SeRi | I have panasonic 6ghz DECT phones... they work ok.... not the same though :) |
| 02:16.47 | p3nguin | Why IP what? |
| 02:16.52 | dijib | phone |
| 02:16.59 | p3nguin | Because I use VoIP. |
| 02:17.00 | dijib | and i cant get cdr working |
| 02:17.05 | p3nguin | You can't have VoIP without IP. |
| 02:17.09 | dijib | why not AtA -> wireless |
| 02:17.10 | SeRi | carrar, can you link me to a desk phone with dect? |
| 02:17.27 | WIMPy | CAT-iq will offer at least G.722 in addition to G.726. Many will also do G.729. |
| 02:17.32 | carrar | Aastra |
| 02:17.40 | carrar | C7iT or something |
| 02:17.47 | carrar | have to look it up |
| 02:17.54 | carrar | we have a few deployed |
| 02:18.27 | p3nguin | I'd just rather have a SIP base as opposed to an analog base and an ATA. |
| 02:19.05 | p3nguin | I could spend $100 on a good cordless phone, then I still have to get a good ATA; or I could spend like $130 for a decent SIP cordless phone. |
| 02:19.17 | SeRi | dijib, digital to digital will be best. analog to digital is overhead. |
| 02:19.28 | carrar | Aastra 6757i CT |
| 02:19.35 | carrar | newer model out |
| 02:19.37 | dijib | yeah but not for cheap smoes ike me\ |
| 02:19.37 | WIMPy | Err, where fo you buy? |
| 02:19.47 | carrar | actually thats the same one |
| 02:19.52 | carrar | 57iCT |
| 02:19.54 | carrar | is what they call it |
| 02:20.16 | SeRi | carrar, Thanks! |
| 02:20.16 | carrar | You can trasnfer from desk to cordless |
| 02:20.20 | SeRi | looking in to it now. |
| 02:20.23 | carrar | works real nice |
| 02:20.27 | SeRi | sweet |
| 02:20.46 | carrar | worked 3 houses away from my hosue when I tested it |
| 02:20.50 | carrar | house |
| 02:21.12 | SeRi | nice |
| 02:21.47 | carrar | House Base Distance (HBD) stats |
| 02:21.49 | carrar | heh |
| 02:22.21 | carrar | The DECT stuff is ncie for those places that are fooded with wifi |
| 02:22.49 | WIMPy | You can even get a DECT PCI card and use cahn_dect. But I don't know how stable that is. |
| 02:22.59 | carrar | heh |
| 02:23.11 | carrar | I'd go with a Aastra DECT solution |
| 02:23.22 | SeRi | is the 6x 0r 5x newer carrar? |
| 02:23.44 | WIMPy | The AVM routers have very good DECT integration. |
| 02:23.49 | carrar | I would guess 6 is > 5 |
| 02:24.19 | carrar | it's a about a year and ahalf since I tested it and put it into production |
| 02:24.28 | carrar | maybe 2 |
| 02:24.42 | carrar | zero complaints |
| 02:24.52 | dijib | i printed out the Asterisk Variable list but i cant find STRFTIME or similar. |
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| 02:25.01 | dijib | any direction? |
| 02:25.22 | carrar | I also have to say that the Aastra people are much friendly and easier to deal with the Polycom |
| 02:25.30 | carrar | then |
| 02:25.47 | carrar | well they all are friend on the phone |
| 02:25.49 | WIMPy | dijib: 'core show function strftime' |
| 02:25.58 | carrar | Aastra seem to go out of it's way more for us |
| 02:26.44 | SeRi | I think any company that spends a sious ammount of money on a product directly with them sort of tend to make the compnay bend a bit for there customers... |
| 02:26.54 | dijib | trunk*CLI> core show function strftime |
| 02:26.54 | dijib | No function by that name registered. |
| 02:26.54 | dijib | Command 'core show function strftime' failed. |
| 02:26.55 | dijib | huh |
| 02:27.11 | p3nguin | STRFTIME is a function not a variable. |
| 02:27.21 | dijib | ah . |
| 02:27.33 | p3nguin | core show function STRFTIME |
| 02:27.49 | p3nguin | Capitalization is important. |
| 02:28.16 | WIMPy | yes. Sorry. |
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| 02:36.06 | WIMPy | Oh, I have to make an important correction. If a CAT-iq handset is advertised as being capable og G.722 it actually won't do G.722, but G.722.1 which makes it a lot less usefull. |
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| 02:36.35 | nafg_ | What would be a good way to design an IVR menu that has a variable number of choices, possible more than 10 (maybe 30 sometimes) |
| 02:36.39 | nafg_ | It's choosing an option out of choices in a database, which aren't categorized. |
| 02:39.28 | WIMPy | DECT is only 32kbps |
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| 03:50.35 | samuelsapps | I having trouble configuring asterisk with gxw4108, when dialing the pstn number it success but when I dial the extention number it's seem the gxw try to dial to pstn |
| 03:51.46 | samuelsapps | how do I differentiate the PSTN and extention |
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| 04:23.55 | coppice | WIMPy: CAT-iq uses G.722, not G.722.1 |
| 04:26.55 | WIMPy | G.722 has min 48kbps, DECT only transmits 32kbps. |
| 04:27.19 | WIMPy | It's just that the manuals miss the .1 |
| 04:27.59 | WIMPy | I mean there is the possibility to use multiple channels, but I'm not aware that this is done on any handsets. |
| 04:28.07 | coppice | CAT-iq can send at 64k. It uses G.722. G.722 actually needs a full 64k to be of any use. at 48k its narrowband |
| 04:29.21 | WIMPy | When I dug further in to the specification of some system (can't remember which it was) it explicitly mentioned 722.1 somewhere. |
| 04:29.47 | WIMPy | If it's real G.722 with 64kpbs that sounds nice. Where did you find that? |
| 04:31.23 | coppice | what do you mean by "real" G.722? G.722.1 is a totally unrelated codec |
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| 04:32.04 | WIMPy | real=no ammex |
| 04:32.21 | WIMPy | G.722.2 is yet aother totally different codec. |
| 04:32.29 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 04:37.28 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 04:37.44 | coppice | G.722.2 is AMR-WB by another name |
| 04:37.46 | coppice | G.722.1 is mostly used by conferencing phones |
| 04:37.48 | coppice | G.722 is what the vast majority of "HD voice" is using today |
| 04:37.50 | coppice | The three are totally unrelated to each other. Only G.722 is unencumered |
| 04:38.16 | WIMPy | I know. |
| 04:38.40 | WIMPy | But where did you find that CAT-iq uses or should use G.722? |
| 04:39.27 | coppice | a) the documents |
| 04:39.29 | coppice | b) the product descriptions of CAT-iq compatible phones |
| 04:39.30 | WIMPy | I used to believe that as well until I fund G.722.1 mentioned burried in some spec. |
| 04:39.31 | coppice | c) the SDP and RTP coming out of one |
| 04:40.04 | WIMPy | What brand are you using? |
| 04:40.16 | coppice | for various reasons the industry has a huge reluctance to use G.722.1 |
| 04:41.02 | coppice | I don't use any, but I've helped other people integrate CATiq phones into their system |
| 04:41.21 | WIMPy | Seeing that .1 and .2 require licenses that makes sense. |
| 04:42.04 | WIMPy | OTOH most hardware phones include G.729. |
| 04:42.07 | coppice | it makes sense much more from the point of view of compatibility. hardly any VoIP kit can do .1 or .2 |
| 04:43.33 | WIMPy | I'm still searching for any device that does G.722 over ISDN so that it would be possible to look into getting that done for Asterisk. |
| 04:45.01 | coppice | there is very little kit that does proper ISDN G.722. Sangoma now support G.722 over ISDN (at least when used with FreeSwitch), but they don't seem to flag the codec as G.722. They flag it as a clear 64k bearer channel |
| 04:45.03 | *** join/#asterisk mintos (~mvaliyav@114.143.163.34) |
| 04:45.52 | WIMPy | I know it's only broadcasting equipment and that comes with quite a prive tag. |
| 04:45.59 | WIMPy | price |
| 04:46.28 | WIMPy | But withour interoperability that's worth a lot less. |
| 04:47.38 | coppice | they usually use it over private 64k circuits, so ISDN control doesn't really enter into it |
| 04:49.28 | WIMPy | But it's not that unlikely today that two people both use G.722 capable IP phones but are connected via the PSTN. And there is no real reason that it shouldn't work in that scenario. |
| 04:50.51 | coppice | I guess it depends on whether a high percentage of PSTN switch will actually negotiate the bearer channel |
| 04:51.33 | WIMPy | The real one should do. |
| 04:51.52 | WIMPy | But I'm pretty sure the modern SIP based lines won't. |
| 04:52.04 | *** join/#asterisk mintos (~mvaliyav@114.143.163.34) |
| 04:52.05 | WIMPy | Interestingly enough, my provider even seems to do transcoding. |
| 04:53.04 | WIMPy | But it may be neccessary to make transfers work. |
| 04:53.06 | coppice | transcoding of voice, and transcoding of bit to modem tones was supposed to be a cornerstone of ISDN. it rarely worked out, in practice |
| 04:53.47 | WIMPy | I heard about that idea for the first time in here. |
| 04:53.55 | WIMPy | But it has been in use for GSM. |
| 04:53.58 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 04:55.50 | WIMPy | Our national ISDN supported BC change during a connection. AFAIK that's not possible any more. |
| 04:56.11 | WIMPy | Might be a reason why noone tries to support G.722. |
| 05:00.36 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 05:00.36 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 05:09.12 | *** join/#asterisk shido6 (~shido6@c-98-234-181-35.hsd1.ca.comcast.net) |
| 05:12.36 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 15:49.52 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:49.52 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:50.13 | anonymouz666 | this usually happens once a week |
| 15:50.58 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 15:51.02 | anonymouz666 | I hate to do it, but it seems a restart now every day using crontab make things more stable |
| 15:53.35 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:53.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:53.59 | anonymouz666 | p3nguin: I use that heavily on distributed setups |
| 15:55.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 16:07.51 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 16:13.00 | r0m|u | p3nguin, you got a sec? |
| 16:15.17 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 16:15.59 | *** join/#asterisk infinityplus1 (~root@66.241.104.121) |
| 16:16.02 | infinityplus1 | Can you tell me if I should continue to use the 64.2.142.93 for |
| 16:16.02 | infinityplus1 | outbound |
| 16:16.04 | infinityplus1 | Also can I use 66.241.99.144 instead of inbound44.vitelity.net for |
| 16:16.07 | infinityplus1 | the inbound |
| 16:16.24 | infinityplus1 | sorry about that |
| 16:20.37 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:22.18 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:22.24 | r0m|u | [TK]D-Fender, you got a second? |
| 16:22.31 | [TK]D-Fender | possibly |
| 16:22.38 | r0m|u | ill make it quick |
| 16:22.49 | r0m|u | I am gathering the sip debug now |
| 16:22.50 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 16:23.12 | r0m|u | I am unable to make calls using sipbri.... |
| 16:23.13 | _abc_ | Has anyone succeeded to tunnel a iax2 connection over a ssh port forwarding channel? |
| 16:23.20 | r0m|u | Oct 26 11:17:51 pbx local0.warn asterisk[2519]: WARNING[2552]: chan_sip.c:8740 in process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 101 |
| 16:23.21 | r0m|u | Oct 26 11:17:51 pbx local0.warn asterisk[2519]: WARNING[2552]: chan_sip.c:8827 in process_sdp: Failing due to no acceptable offer found |
| 16:23.21 | r0m|u | Oct 26 11:18:01 pbx local0.warn asterisk[2519]: WARNING[2552]: chan_sip.c:8740 in process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 101 |
| 16:23.21 | r0m|u | Oct 26 11:18:01 pbx local0.warn asterisk[2519]: WARNING[2552]: chan_sip.c:8827 in process_sdp: Failing due to no acceptable offer found |
| 16:23.27 | r0m|u | ops didnt meant that |
| 16:23.34 | r0m|u | tring to pb |
| 16:23.39 | _abc_ | Has anyone succeeded to tunnel a iax2 connection over a ssh port forwarding channel? |
| 16:23.47 | _abc_ | oh r0m|u it's your fault :/ |
| 16:24.01 | r0m|u | lol |
| 16:24.15 | [TK]D-Fender | r0m|u, Make it a complete call. |
| 16:25.43 | r0m|u | [TK]D-Fender, http://pastebin.com/47CwsCRp |
| 16:26.15 | r0m|u | when I make the call it rings just fine. as soon as the other party answers my call drops with the above error in the pb |
| 16:27.12 | [TK]D-Fender | Something is missing... |
| 16:27.42 | r0m|u | [TK]D-Fender, I have only change what you told me to change two nights ago which was nat= |
| 16:27.56 | r0m|u | Thats it... nothing has changed since than |
| 16:28.02 | r0m|u | voipms works wonderfully |
| 16:28.23 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 16:28.23 | r0m|u | I am starting to suspect that sipbri is at fault |
| 16:28.23 | [TK]D-Fender | I'm not seeing standard verbose and I'm sensing that the debug is restricted somewhere... |
| 16:28.24 | hardwire | surprisingly well. |
| 16:28.30 | hardwire | agrees with r0m|u |
| 16:28.36 | hardwire | they have hawaii dids that don't suck |
| 16:28.50 | hardwire | going to move all my broadvoice traffic over to them. |
| 16:29.02 | r0m|u | [TK]D-Fender, want me to do another debug? |
| 16:29.17 | r0m|u | hardwire, sipbri whent down for two days and came back up yesterday |
| 16:29.52 | r0m|u | and I just decided to test the calls today to see if there was any issues with there google connection... and I found this cluster fuck... |
| 16:30.18 | [TK]D-Fender | r0m|u, what is "sipbri"? |
| 16:30.30 | r0m|u | [TK]D-Fender, sipbri.com |
| 16:30.41 | r0m|u | sipbri offers free sip to google links |
| 16:30.49 | hardwire | was hoping [TK]D-Fender would be at *con |
| 16:31.23 | r0m|u | [TK]D-Fender, I am unable to use jabber/gtalk in my system so I use sipbri to link my GV with sip |
| 16:31.34 | [TK]D-Fender | hardwire, Don't have a passport, and the costs around such a trip are more than I could ever justify |
| 16:33.17 | r0m|u | [TK]D-Fender, you want me to do another debug? |
| 16:33.30 | [TK]D-Fender | r0m|u, yes |
| 16:33.36 | hardwire | it's probably not 100% worth it if it costs you over a few grand to get here. |
| 16:33.38 | r0m|u | one sec |
| 16:33.56 | hardwire | [TK]D-Fender: maybe do a talk and see "what happens" if you haven't tried already :) |
| 16:34.07 | hardwire | Talk about "The problems with IRC communication when troubleshooting Open Source Software" |
| 16:34.12 | hardwire | hehe |
| 16:34.19 | [TK]D-Fender | lol... that couldn't possibly end well :) |
| 16:34.36 | hardwire | or "How to be snarky but helpful" |
| 16:34.43 | hardwire | which most people would enjoy. |
| 16:35.30 | [TK]D-Fender | hardwire, for nostalgia's sake perhaps, but I've backpedeled accordingly since I came back. |
| 16:35.42 | r0m|u | [TK]D-Fender, http://pastebin.com/xzZ083Nh |
| 16:35.54 | hardwire | [TK]D-Fender: came back eh? i've been out of the loop it seems. |
| 16:36.07 | snax | moin. |
| 16:36.19 | hardwire | I think because freenode keeps sacking my connection. |
| 16:36.29 | hardwire | or nick |
| 16:36.37 | hardwire | I get booted from #asterisk a lot :| |
| 16:36.37 | r0m|u | Haraken, "How to be snarky but helpful" lol that seems to be a good topic around here :) |
| 16:36.46 | snax | why bother doing a dialplan reload at the CLI> instead of just restarted the asterisk daemon? It's not like it has a syntax checker of extensions.conf. |
| 16:39.54 | [TK]D-Fender | hardwire, I was gone for 3/4 year |
| 16:40.08 | r0m|u | [TK]D-Fender, I talk to some other sipbri users and they dont have this issue. so looks like it just me..... I also remember changing directmedia=no |
| 16:40.23 | SunTsu | snax: because that might interrupt other stuff in progress? |
| 16:40.29 | r0m|u | but I know those are not the issues |
| 16:42.02 | *** join/#asterisk jrad (~jrad@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 16:45.40 | p3nguin | snax: dialplan reload only reloads the extensions and does not interrupt anything; it only takes 0m0.090s for my system. Restarting asterisk requires the daemon to shut down and start again, stopping phone calls for at least a full minute. |
| 16:46.19 | p3nguin | And besides that, why restart something that doesn't need restarted? This isn't Windows. |
| 16:46.55 | r0m|u | p3nguin, lol |
| 16:47.15 | r0m|u | p3nguin, maybe you can help as well.... I am having some issues with sipbri |
| 16:47.17 | r0m|u | http://pastebin.com/DGBmTwVR |
| 16:47.23 | r0m|u | thats the debug ^^ |
| 16:47.46 | r0m|u | process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 101 |
| 16:49.00 | r0m|u | ha! |
| 16:49.07 | r0m|u | Its a carrier issue |
| 16:49.09 | [TK]D-Fender | r0m|u, Still no verbose |
| 16:49.20 | [TK]D-Fender | and I'm sensing a single direction of debug not global |
| 16:49.31 | r0m|u | [TK]D-Fender, I just got a report that other users are having the same issue |
| 16:49.41 | jrad | Looking for any help on Asterisk Outbound DTMF signals. Using Asterisk 1.8.5 - Issue symptoms: When calling outbound to IVR's like ATT, Spring, Verison, and banks- Digits pressed quickly or swiftly are not read correclu resulting in IVR menu repeat. However, when DTMF buttons are pressed in for a longer period they are read correctly. I've tried many different settings on two different cisco phone models. We use SIP and AT |
| 16:49.41 | jrad | V/RFC2833. More info available on request. |
| 16:50.17 | anonymouz666 | anyone know why calling from sip to dahdi channel you pickup the fxs extension and after you hear the dtmfs dialed ? |
| 16:50.26 | r0m|u | [TK]D-Fender, ? I am using sip set debug on |
| 16:51.25 | p3nguin | core set verbose 3 |
| 16:51.29 | p3nguin | sip set debug on |
| 16:51.32 | p3nguin | Make a call. |
| 16:51.36 | p3nguin | Paste the results. |
| 16:52.25 | p3nguin | When you have verbose AND sip debug on the same page, it makes it a lot nicer to figure out what is causing each of the actions in the debug. |
| 16:56.12 | r0m|u | working on it |
| 16:56.34 | *** join/#asterisk irroot (~irroot@197.170.100.128) |
| 16:57.03 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-208-16.red.bezeqint.net) |
| 17:01.36 | r0m|u | p3nguin and [TK]D-Fender http://pastebin.com/a5cmZUB2 |
| 17:01.40 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 17:04.28 | p3nguin | It sure didn't add much stuff. |
| 17:04.55 | [TK]D-Fender | All I see is a ton f "we don't like ULAW". Show me where they list acceptable codecs. |
| 17:05.08 | p3nguin | I guess I need to set up the sipbri thing and make it work. |
| 17:06.08 | r0m|u | [TK]D-Fender, they dont have one. I assume they accept ulaw... |
| 17:06.15 | [TK]D-Fender | ....... |
| 17:06.40 | r0m|u | there site suck and the tech support is never online |
| 17:06.57 | [TK]D-Fender | r0m|u, http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 17:07.16 | r0m|u | LOL |
| 17:08.14 | p3nguin | Their site is broken. |
| 17:08.39 | r0m|u | there face is broken.... God do they suck! |
| 17:08.52 | r0m|u | I am wasting no more of you guys time |
| 17:09.03 | r0m|u | I am not* |
| 17:09.24 | r0m|u | screw it. there service just plain suck... its free anyways. |
| 17:10.37 | [TK]D-Fender | Go get your money back. |
| 17:10.58 | r0m|u | lol |
| 17:11.00 | r0m|u | no shit |
| 17:13.42 | r0m|u | [TK]D-Fender, others are having similar issues. |
| 17:13.50 | r0m|u | so to hell with them |
| 17:13.54 | r0m|u | I am out to lunch |
| 17:13.57 | r0m|u | Thanks guys. |
| 17:18.58 | p3nguin | One issue with sipbri: I'm not giving them my google voice account email address and password. |
| 17:19.17 | p3nguin | And I don't really feel like creating another new account just to test this. |
| 17:21.20 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:22.15 | r0m|u | Thats what I did a test account |
| 17:22.42 | r0m|u | They suck |
| 17:24.09 | p3nguin | If you want to share your test accout with me, I'll set it up in sipbri and see what I can do. |
| 17:24.22 | p3nguin | If not, I'll skip sipbri altogether. |
| 17:25.36 | r0m|u | skip it because I just dumped them. @dslr people are doing the same. the issue is at the carrier. |
| 17:25.49 | p3nguin | unregisters sipbri |
| 17:25.53 | r0m|u | everybody that had working accounts just borked |
| 17:26.13 | r0m|u | like I did so everybody is jumping ship |
| 17:26.24 | p3nguin | Actually I can't unregister... that's for things registered to me. |
| 17:26.29 | p3nguin | deletes parts of sip.conf |
| 17:26.55 | r0m|u | p3nguin, thanks for the help though |
| 17:27.18 | p3nguin | I was ready to go. |
| 17:27.30 | r0m|u | p3nguin, sorry :( |
| 17:27.33 | p3nguin | I was registered to their system and everything. |
| 17:27.42 | p3nguin | Just needed to link my GV account. |
| 17:28.11 | r0m|u | p3nguin, msg me. |
| 17:28.14 | p3nguin | And now I forgot what else I was working on. |
| 17:29.05 | r0m|u | p3nguin, msg me. |
| 17:30.19 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-208-16.red.bezeqint.net) |
| 17:32.42 | tzanger | man app_dial is huge |
| 17:34.04 | anonymouz666 | bizarre "deferred digit string" with the extension dialed upon fxs pickup |
| 17:34.25 | anonymouz666 | don't know the meaning of that, i just hear the dtmf tones |
| 17:34.33 | anonymouz666 | the callerid is sent when ringing |
| 17:34.39 | anonymouz666 | not after pickup |
| 17:34.50 | anonymouz666 | i don't have a clue why this happens |
| 17:36.06 | [TK]D-Fender | FXS CID is normally FSK not DTMF |
| 17:36.29 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:36.38 | anonymouz666 | it's an internal call from sip to fxs. |
| 17:41.13 | *** join/#asterisk mjordan (mjordan@conference/astricon/x-igqctzpccfqzbscc) |
| 17:41.36 | *** join/#asterisk gandhijee_ (~akp@50.12.169.99) |
| 17:42.56 | gandhijee_ | hey, my asterisk started complaing about a PRI error - we think we're the CPE, but they think they're the CPE too, i can change to network mode, then i get We think we're the network, but they think they're the network, too msg |
| 17:42.58 | gandhijee_ | any ideas? |
| 17:43.19 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 17:43.46 | WIMPy | You have a loop |
| 17:43.56 | WIMPy | Or your telco. |
| 17:44.10 | WIMPy | Tell them to pull the plug. |
| 17:44.56 | gandhijee_ | you mean power cycle the box? |
| 17:45.24 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 17:47.17 | anonymouz666 | damn fixed. |
| 17:47.43 | anonymouz666 | easy |
| 17:47.44 | WIMPy | No your line is looped back to you. |
| 17:50.12 | jrad | im having trouble sending DTMF digits properly, any suggestions on where to start? more details available. |
| 17:57.20 | gandhijee_ | i already had them pull the line, didn't work, so i just had them power cycle the CISCO |
| 17:57.26 | gandhijee_ | thanks WIMPy |
| 17:57.46 | WIMPy | Not pull the line, pull the loop plug. |
| 17:58.20 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-208-16.red.bezeqint.net) |
| 18:00.01 | *** join/#asterisk ew0x (~ewox@mail.brain4sale.org) |
| 18:02.04 | WIMPy | Err, and BTW: What Cisco? |
| 18:02.29 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 18:02.32 | WIMPy | AFAIK Asterisk doesn't run on Ciscos. |
| 18:03.08 | eppigy | you can run sccp or sip on cisco phones |
| 18:03.09 | p3nguin | :/ |
| 18:03.12 | eppigy | lol |
| 18:03.25 | p3nguin | But you can't run Asterisk on a Cisco device. |
| 18:03.38 | eppigy | well...yeah |
| 18:03.44 | WIMPy | And the phones don't have PRIs. |
| 18:04.13 | eppigy | i got ur pri |
| 18:04.49 | p3nguin | So there's a Cisco PBX attached to a PRI? |
| 18:04.58 | ew0x | Hi folks. Looking for some help: I'm trying to get asterisk to automatically pick-up the first non-busy trunk when placing a call via a .call file |
| 18:05.15 | [TK]D-Fender | Who's on first? |
| 18:05.16 | p3nguin | Or maybe a Cisco device converting the PRI to SIP? |
| 18:05.25 | WIMPy | My PRI? Give it back! |
| 18:05.33 | *** part/#asterisk Consolas (~Consolas@a81-84-246-186.static.cpe.netcabo.pt) |
| 18:05.45 | p3nguin | ew0x: Do you mean channel when you say trunk? |
| 18:05.49 | WIMPy | But that's a dahdi or libpri message. |
| 18:06.08 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-208-16.red.bezeqint.net) |
| 18:06.11 | ew0x | p3nguin: Yes, that's what I meant |
| 18:06.28 | p3nguin | If you are using groups of channels, that's how it should already work. |
| 18:07.10 | gandhijee_ | the the cisco 2600 in my case provides me my T1 hand off from XO since they won't let me run a straight up asterisk with SIP |
| 18:07.15 | ew0x | p3nguin: I'm not, obviously. Thanks for the tip, I'll google through the rest. |
| 18:07.22 | p3nguin | E.g., g0 would use the next open channel going upward; G0 would use the next available channel going downward. |
| 18:08.09 | p3nguin | Dial(DAHDI/g0/${EXTEN}) |
| 18:08.13 | WIMPy | gandhijee_: Then the interface is looped on the cisco. |
| 18:08.45 | ew0x | p3nguin: Okay, got the concept, I'll just setup channel groups then. |
| 18:09.13 | p3nguin | Let us know if you need more help making it work. |
| 18:09.26 | p3nguin | wimpy is our resident analog expert. |
| 18:09.46 | WIMPy | Ha |
| 18:09.49 | p3nguin | :) |
| 18:09.56 | p3nguin | I know how much you dislike VoIP. |
| 18:10.17 | WIMPy | And analog. |
| 18:10.22 | p3nguin | Oh? |
| 18:10.25 | WIMPy | I'm not THAT old. |
| 18:10.31 | p3nguin | You're opposed to telephony as a whole? |
| 18:10.34 | WIMPy | I like things that simply work. |
| 18:10.46 | WIMPy | No. It just has to work. |
| 18:10.56 | WIMPy | Which is ok of analog. |
| 18:11.03 | p3nguin | You can't get anything that just works anymore. |
| 18:11.10 | WIMPy | But I also like features. That needs digital. |
| 18:11.12 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 18:11.30 | WIMPy | You can, but it's getting harder. |
| 18:11.32 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-208-16.red.bezeqint.net) |
| 18:11.56 | WIMPy | Which reminds me that I guess I should tell my telco that they broke their service even more. |
| 18:12.49 | p3nguin | Used to, you'd get what you pay for. Now, you can't even BUY anything that's worth a shit. |
| 18:13.25 | WIMPy | The old stuff continues to work. |
| 18:13.30 | p3nguin | That's true. |
| 18:13.53 | WIMPy | And in fact it does so very well. |
| 18:14.02 | p3nguin | I think that's why you can't buy things that are worth a shit now. Ensure repeat customers by making junk products that don't last. |
| 18:14.03 | WIMPy | And with relatively small effort. |
| 18:14.35 | serafie | in dialplan world, can you access variables of unknown name, like ${${NAME}}? |
| 18:14.38 | p3nguin | A lot of old stuff I have that still works, the companies are no longer around. |
| 18:14.56 | WIMPy | serafie: Yes |
| 18:15.13 | serafie | Ooh, yay. That will make this macro prettier. |
| 18:15.23 | p3nguin | How can you get a value from a variable without knowing what variable you're checking? |
| 18:15.56 | WIMPy | By knowing later. |
| 18:16.02 | serafie | ARG1, ARG2, ARG3, ARG4.... |
| 18:16.17 | p3nguin | hmm |
| 18:16.47 | p3nguin | So you'd literally use ${${ARG1}} to refer to what ARG1 might be later? |
| 18:17.21 | p3nguin | Peculiar. I'm interested to see a snippet of dial plan where this is used... and works. |
| 18:17.30 | WIMPy | No it's about passing variables by reference. |
| 18:17.43 | snax | Do trunks use the SIP protocol? |
| 18:17.59 | WIMPy | ~siptrunk |
| 18:17.59 | infobot | somebody said siptrunk was something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 18:18.10 | WIMPy | But to be more precise, the question doesn't make much sense. |
| 18:18.12 | snax | so trunks use AIX |
| 18:18.48 | p3nguin | no |
| 18:18.49 | WIMPy | No, trunk is a bad word. Better use something more meaningful. |
| 18:19.01 | p3nguin | ~trunk |
| 18:19.01 | infobot | [trunk] a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 18:19.17 | wdoekes2 | snax: the difference between a "trunk" and a regular account in SIP is usually whether you send/get a DID or not |
| 18:19.43 | [TK]D-Fender | .. |
| 18:19.48 | p3nguin | That's not right. There is no trunk in SIP. |
| 18:19.58 | p3nguin | And DIDs cannot be sent. |
| 18:20.00 | p3nguin | ~did |
| 18:20.01 | infobot | did is, like, Direct Inward Dialing, or just a phone number |
| 18:20.05 | p3nguin | INWARD |
| 18:20.39 | WIMPy | And a SIP account doesn't have to hade a directory number, even if used as a trunk. |
| 18:20.59 | r0m|u | p3nguin, msg. |
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| 18:53.49 | jrad | <PROTECTED> |
| 18:53.49 | jrad | ATV/RFC2833. More info available on request. |
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| 19:03.49 | SuperNull | anyone play with opensips/openser.. ? |
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| 19:11.48 | bipul | CAN ANY ONE HELP ME WHEN i configure my asterisk for voip i got this message at then end is it a error http://pastebin.com/4tJwivPP |
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| 19:14.40 | kaldemar | bipul: it says it to be an error and even says what the error is. yoou're missing the context for the configurations in indications.conf. |
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| 19:18.51 | bipul | kaldemar, for example |
| 19:21.01 | kaldemar | [general]. take a look at the sample config. |
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| 19:45.20 | bipul | kaldemar, http://pastebin.com/XKrAFZdG |
| 19:45.23 | bipul | check this |
| 19:45.43 | bipul | do i need to add [general] at the bottom |
| 19:57.01 | kaldemar | the top |
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| 20:11.20 | *** join/#asterisk sogi (sogi@triton.intrak.tuke.sk) |
| 20:12.00 | sogi | hey guys |
| 20:12.04 | sogi | udptl.c: No UDPTL ports remaining |
| 20:12.14 | sogi | chan_sip.c: UDPTL creation failed |
| 20:12.17 | sogi | ast 1.8.X |
| 20:12.24 | sogi | any idea? :) |
| 20:12.40 | sogi | a bunch of opened UDP ports of course |
| 20:12.44 | sogi | in netstat -a -p -n |
| 20:15.13 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 20:16.10 | zyphlar | have you ever seen a MeetMe session go on forever thinking it has one caller even though everyone's hung up? |
| 20:16.28 | zyphlar | and/or how would i manually close a MeetMe |
| 20:18.58 | p3nguin | I would use channel request hangup ... |
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| 20:20.01 | zyphlar | hmm i'm thinking of setting a max conference time of like 8 hours |
| 20:20.09 | zyphlar | cuz this happens almost daily |
| 20:21.10 | zyphlar | still, you'd think it'd be able to better detect when nobody's in the conference anymore |
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| 20:40.11 | devil_evoxxx | hi all |
| 20:40.51 | devil_evoxxx | someone of you have connected asterisk with a nortel cs200_NGss 8.0? |
| 20:50.40 | devil_evoxxx | there is someone? |
| 20:50.44 | devil_evoxxx | *are |
| 20:52.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:52.59 | devil_evoxxx | my problem is similar to https://issues.asterisk.org/jira/browse/ASTERISK-11075 |
| 20:53.49 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 20:56.04 | bipul | http://pastebin.com/UjpNQsT4 stil having issue |
| 21:03.43 | devil_evoxxx | now, i can say that is the same problem, my prov send OPTIONS with empty username, and ip matching ( context) does not work |
| 21:03.55 | devil_evoxxx | there is some work-around? |
| 21:04.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:06.23 | devil_evoxxx | [TK]D-Fender: good morning / evening :) |
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| 21:23.57 | boch | hi all |
| 21:24.31 | amaache | hi |
| 21:25.26 | eppigy | hello |
| 21:25.28 | eppigy | i am dave |
| 21:25.48 | boch | i have a problem, im bridging two channels using Bridge() and i want to unbridge them pressing *, for that i manage to Dial() a Local channel with H option, but now, the Bridged channel does not continues with dialplan execution, without dialing local channel it does... |
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| 21:34.09 | bipul | http://pastebin.com/UjpNQsT4 some one please help me |
| 21:44.24 | puzzled | bipul: fix what is causing your warning/error |
| 21:44.55 | [TK]D-Fender | bipul: go look at all of the app_voicemail modules in usr/src and then do "noload =app_voicemail____.so" for all the ones you don't need |
| 21:44.59 | bipul | there is no mail box |
| 21:45.05 | [TK]D-Fender | .. |
| 21:45.13 | [TK]D-Fender | Disable the conflicting modules |
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| 21:47.03 | bipul | http://pastebin.com/eUMj2JaH |
| 21:47.29 | bipul | this is mine voice mail.conf |
| 21:48.00 | puzzled | bipul: you may want to remove your passwords next time... |
| 21:48.14 | [TK]D-Fender | bipul: Forget voicemail.conf |
| 21:48.20 | [TK]D-Fender | disable to modules from loading |
| 21:48.23 | [TK]D-Fender | modules.conf <---- |
| 21:48.55 | bipul | ok |
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| 21:51.21 | sogi | haha awesome passwords on pastebin |
| 21:51.23 | sogi | :) |
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| 21:52.57 | eppigy | hacked by chinese |
| 21:53.32 | [TK]D-Fender | With security holes like PB-ing configs with PW's exposed.. it's more like hacked by cheese |
| 21:53.36 | [TK]D-Fender | Swiss |
| 21:53.45 | [TK]D-Fender | Just like his security |
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| 22:49.47 | ruied | Hi, I have one computer per classroom in a Windows AD network, each professor has it's own user/pass and profile. For each computer I would like to have a unique sip account. I.E. Is there a way to have a sip extension per computer and not per user? every time a user logs in it normally has to configure a sip account, and that's a big problem.... |
| 22:51.56 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 22:52.14 | ruied | I would like to have one sip account per classroom so a professor can make internal call asking for something, and that call being identified like "Classroom 23" |
| 22:52.58 | ruied | The natural way seems to have one phone per classroom but that's a budget problem... |
| 22:54.21 | navaismo | configure the same peer in each windows session, im guessing there are only one logged session per computer |
| 22:58.15 | ruied | navaismo, each classroom has a rotation of about 15 or more professors along the five working days of the week, and a professor can log in the next hour at another classroom. There are around 90 classrooms... |
| 22:59.19 | navaismo | the same peer cant be used at the same time in different locations |
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| 23:01.32 | p3nguin | Many soft phones are free to use. No budget required, since almost every classroom has at least one computer. |
| 23:02.34 | ruied | navaismo, not at the same time, but if he goes to another classroom, the profile roams with him to the other classroom and the sip account goes with him... I would like to have a classroom extension and not a user extension... |
| 23:02.54 | p3nguin | Extensions have nothing to do with it. |
| 23:03.06 | p3nguin | You're dealing with the SIP peer at this point. |
| 23:03.35 | p3nguin | ~devicenames |
| 23:03.35 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 23:04.30 | p3nguin | If the PC in classroom 23 has a MAC address of 00001111ffff, that's going to be the SIP peer name of the soft phone you install on that PC. |
| 23:05.00 | p3nguin | To call that phone, you'd create extension 23, and extension 23 dials SIP/00001111ffff. |
| 23:10.14 | ruied | p3nguin, my problem is to configure the softphone, since the computer 23 is being used by several professors with several profiles... and one professor can go to classroom24 at the next hour roaming the sip configuration with him |
| 23:10.39 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 23:11.00 | p3nguin | Don't install the soft phone for the person -- install it for the computer. |
| 23:11.04 | blitzrage | that sounds like hot-desking |
| 23:11.34 | p3nguin | The extensions do not need to be tied to the people. |
| 23:11.37 | blitzrage | that's why you use the mac address to indentify the device -- you then use a logical layer to abstract the person and extension from the device so that extension/person can move freely |
| 23:11.44 | p3nguin | They need to be tied to the phone configured in each classroom. |
| 23:11.48 | p3nguin | No hot-desk about it. |
| 23:12.16 | blitzrage | oh nevermind I read back -- not quite working the way I was thinking |
| 23:12.24 | p3nguin | You could additionally configure an extension per person, and hot desk that. |
| 23:12.47 | p3nguin | But the phone will still stay tied to the classroom and the PC in it, not to the person. |
| 23:14.14 | p3nguin | Hmm, there's no asterisk-extra-sounds-es? |
| 23:14.32 | p3nguin | I see fr and en only. |
| 23:15.23 | ruied | p3nguin, I would like to configure per computer, but it seems when a user logs in the software is expecting the sip configuration for that user... Is there a software that can be configured per computer, and all users can us it without messing with the confs? |
| 23:15.51 | p3nguin | I was going to go multi-lingual, but with only core es and no extra es, that may be difficult to do. |
| 23:16.53 | navaismo | that remember me one post on the asterisk forums for someone complaining about racism |
| 23:16.54 | p3nguin | Try using zoiper classic soft phone and drop the configuration into a profile that is default on the computer rather than in a user profile which is a roaming profile. |
| 23:18.00 | navaismo | we always generate all 'es' files |
| 23:18.29 | ruied | p3nguin, tomorrow morning I'll go trying it... |
| 23:21.07 | p3nguin | ruied: You can install the phone and configure it on your development system, then copy the configs from your profile to put into a default profile on all computers. You'll probably want a script to provision each one after you install on each computer. I'd make my script parse the MAC address of the primary Ethernet interface and rewrite the SIP user ID in the config file. |
| 23:21.48 | p3nguin | I'd probably have a single script to unpack the re-packaged softphone and change the user ID all in one go. |
| 23:22.44 | justdave | how do I purge stale entries from a registration context? |
| 23:23.00 | p3nguin | I don't even know what a registration context is. |
| 23:23.36 | justdave | it's a config option for sip.conf on each device, that makes it create a phony extension in the named context when the device registers |
| 23:24.17 | justdave | primary use is for dundi lookups so you can have devices configured in multiple locations and it'll automatically figure out which location it's in without having to change the server config every time you move it |
| 23:24.21 | p3nguin | Are they written to a file or just stored in RAM? |
| 23:24.31 | justdave | just stored in RAM as far as I know |
| 23:24.55 | p3nguin | Can you use "dialplan remove ..." for that? |
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| 23:25.39 | justdave | that was my first thought, too |
| 23:25.41 | justdave | Command 'dialplan remove extension 283@registered-extensions' failed. |
| 23:26.11 | ruied | p3nguin, it seems logic... going to try it |
| 23:26.13 | ruied | thanks |
| 23:26.49 | justdave | what I appear to be seeing is if the phone specifically logs out, it gets removed automatically. |
| 23:27.05 | justdave | if the user just unplugs it, it's stuck there. forever (or until asterisk restarts) |
| 23:27.24 | justdave | you'd think it would go away when the registration timed out or something at least |
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| 23:28.44 | p3nguin | ruied: There used to be a zoiper distribution in zip format that works beautifully for this. Using the installer type would be silly. If it's still available in zip, that's the way to go. |
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| 23:30.15 | ruied | p3nguin, ok :) |
| 23:32.27 | p3nguin | navaismo: Do you have a "Para Español, oprime uno. For English, press two." type of prompt for callers? |
| 23:32.31 | p3nguin | aww |
| 23:36.01 | p3nguin | Now I'm confused. I got the French sounds, and it has continue-in-english, but no continue-in-french! |
| 23:37.04 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 23:37.24 | dandate2 | what is the difference between Skip Busy Agents: Yes + ringinuse=no and QueueCallsOnly +ringinuse=no |
| 23:37.38 | dandate2 | the only difference i can understand is one detects if the agent makes an outbound call and the other doesnt |
| 23:39.20 | dandate2 | does QueueMemberStatus need to be enabled? |
| 23:43.24 | ruied | going to sleep. Thanks p3nguin! |
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