| 00:08.08 | infinity1 | pankid: we've seen this to. i know one confirmed case... someone had their polycom set as the DMZ IP in their router. |
| 00:08.52 | pankid | infinity1: yeah, that would definitly do it. I am told these routers were not touched though. |
| 00:09.02 | infinity1 | pankid: are they polycom? |
| 00:09.16 | pankid | yeah, ip335 I believe |
| 00:10.41 | infinity1 | try these links. the guy that fixed it is offline. |
| 00:10.43 | infinity1 | http://community.polycom.com/t5/VoIP/Ghost-calls-polycom-phone-frustrated/td-p/66545 |
| 00:10.48 | infinity1 | http://community.polycom.com/t5/VoIP/FAQ-How-can-I-prevent-tools-like-sipvicious-or-nuisance-calls/td-p/66679 |
| 00:10.52 | infinity1 | so i'm not sure. |
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| 00:34.42 | pankid | infinity1: so maybe I dont understand nat correctly, but the phone reaches out to the server, which will then allow the server to send packets through 5060 on the firewall to the phone. |
| 00:35.13 | pankid | but only traffic originating from the server will be able to make it through the firewall on the sip port |
| 00:37.32 | infinity1 | pankid: i agree. I don't understand how our guy fixed the ghosts |
| 00:38.21 | infinity1 | pankid: i found the old ticket... hmmm |
| 00:38.57 | pankid | Incoming Signaling Validation might do it |
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| 00:39.20 | infinity1 | these are his notes: |
| 00:39.20 | infinity1 | -exported config file |
| 00:39.20 | infinity1 | -created requestValidation_source.cfg with |
| 00:39.20 | infinity1 | : <voIpProt.SIP.requestValidation voIpProt.SIP.requestValidation.1.method="source" voIpProt.SIP.requestValidation.1.request="INVITE" /> |
| 00:39.24 | infinity1 | as per -> http://community.polycom.com/t5/VoIP/FAQ-How-can-I-prevent-tools-like-sipvicious-or-nuisance-calls/td-p/66679 |
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| 00:39.47 | infinity1 | -imported config file |
| 00:40.04 | infinity1 | since he doesn't have access to the pbx, i'm guessing he logged into the web interface to import it. |
| 00:40.21 | pankid | hm |
| 00:40.29 | pankid | thanks |
| 00:40.45 | infinity1 | makes no sense why that would fix it |
| 00:40.52 | infinity1 | but he claims it did |
| 00:41.20 | infinity1 | which leads me to think i don't understand something. heh |
| 00:41.27 | pankid | yeah, seems like it would need to specify the ip address or hostname of the pbx server |
| 00:41.49 | infinity1 | try it. and educate me when you figure it out :) |
| 00:41.54 | pankid | will do |
| 00:42.22 | pankid | oh I see |
| 00:42.24 | pankid | An incoming call that does not match the reg.1.server.1.address="" Value will receive a 400 Bad Request SIP Message from the phone. |
| 00:42.24 | pankid | <PROTECTED> |
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| 01:18.04 | infinity1 | pankid: the part i don't understand is how the hell is the phone getting an incoming call from another IP? |
| 01:18.19 | infinity1 | the phone creates the connection over nat to the pbx. |
| 01:18.36 | pankid | yeah, sounds like some nonsense polycom did in their phone firmware |
| 01:18.46 | infinity1 | bizzare. |
| 01:19.25 | infinity1 | i should probably add that xml code to the polycom deployment server. |
| 01:20.17 | infinity1 | just for increased security. |
| 01:41.19 | Katty | looks in |
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| 03:52.13 | moe` | hey guys has anyone used asterisk TAPI to get a landline into the asterisk world via a modem? |
| 03:52.38 | moe` | is this even possible? |
| 03:53.07 | moe` | looking at a Hayes Optmia 28.8 modem (external)... could that even work? |
| 03:53.46 | WIMPy | No, I don't think that has ever been tried. |
| 03:54.09 | moe` | in theory could it possibly work? |
| 03:54.47 | WIMPy | AFAIK there's no possibility to transfer voice via TAPI, or is there? |
| 03:55.07 | moe` | when I asked the google machine I found some info on it |
| 03:55.21 | moe` | http://www.voip-info.org/wiki/view/Asterisk+TAPI |
| 03:55.36 | moe` | refers to both inbound and outbound calls |
| 03:56.02 | moe` | or is that just useless cruft? |
| 03:56.17 | WIMPy | But not via TAPI. That describes how to get a TAPI interface to Asterisk. |
| 03:56.27 | moe` | ah, yeh |
| 03:56.48 | moe` | I'm looking for a way to get an analog landline into asterisk |
| 03:56.53 | moe` | of course there are ways |
| 03:57.01 | moe` | but they're like $500 line cards |
| 03:57.05 | WIMPy | And it looks like even that hasn't been tried for years. |
| 03:57.12 | moe` | yeah |
| 03:57.15 | moe` | so likely no soup |
| 03:57.19 | WIMPy | Yes, analog is expensive. |
| 03:57.46 | moe` | any suggestions? |
| 03:57.49 | WIMPy | Move to a place where you can get a BRI. The cards cost next to nothing. |
| 03:58.15 | moe` | want to have a satellite asterisk trunked to the main instance, where the satellite accepts analog landlines |
| 03:58.41 | WIMPy | No other interfaces available? |
| 03:58.53 | moe` | we're stuck with a analog line |
| 03:59.24 | WIMPy | Craptastic. |
| 03:59.32 | moe` | its a case of wanting to integrate an existing analog line into the asterisk world |
| 04:00.30 | moe` | I know a PCI or PCIe card would work, but that's $500+ ... so was looking for a stupid cheap solution |
| 04:00.46 | WIMPy | Well, you've got the ultimate combination of might not work very well and is really expensive. |
| 04:00.58 | moe` | heh |
| 04:01.03 | WIMPy | Doesn't exist. |
| 04:01.03 | moe` | figures. |
| 04:01.17 | moe` | card like that do exist, no? |
| 04:01.17 | WIMPy | You could use some ATA that has an FXO. |
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| 04:01.29 | moe` | that are PCI or PCIe interface to analog landlines |
| 04:01.34 | WIMPy | Not sure how reliable that works. |
| 04:01.36 | moe` | don't PBX's use that type of stuff |
| 04:01.38 | moe` | ? |
| 04:02.17 | WIMPy | Well, in a certain part of the world perhaps. For the rest that hasn't been seen since the 80s or 90s. |
| 04:02.37 | moe` | http://www.asterisk.org/products/telephony-interface-cards |
| 04:02.38 | moe` | no? |
| 04:02.53 | moe` | Digium, of course |
| 04:03.05 | WIMPy | Sure. You already know them. |
| 04:06.01 | moe` | yeah just being optimistic and wondering if a stupid modem could do it |
| 04:06.51 | WIMPy | There used to be chan_modem up to Asterisk 1.2. |
| 04:10.28 | paraxor | hey |
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| 04:23.23 | moe` | hey do you know if on digium cards if hardware echo cancellation is important or is the software good enough these days? |
| 04:27.40 | WIMPy | With software you've got the coice which one to use. |
| 04:28.39 | moe` | indeed |
| 04:29.00 | moe` | looking at phone.conf for example I see off/low/medium/high, as a basic setting |
| 04:29.19 | WIMPy | o.O |
| 04:29.25 | moe` | ? |
| 04:29.51 | WIMPy | What are you up to now? |
| 04:30.04 | moe` | looking at options for a PCI or PCIe to analog card |
| 04:30.34 | moe` | http://www.ebay.ca/sch/i.html?_from=R40&_trksid=p2050601.m570.l1313.TR0.TRC0.H0.Xdigium+.TRS0&_nkw=digium+1A4A00F+&_sacat=0 |
| 04:30.37 | moe` | that, for example |
| 04:31.15 | WIMPy | For analog I wouldn't try to cut costs onthe hardware. |
| 04:31.43 | moe` | well it would be a test just to see if its possible, and then get a PCIe hybrid analog/digital |
| 04:31.54 | moe` | cuz those cards are $500-1500 |
| 04:31.59 | moe` | depends what you get |
| 04:32.39 | moe` | so the cheaper analog card would be a proof-of-concept I guess |
| 04:32.41 | WIMPy | hybrid? What are you up to there? |
| 04:32.56 | moe` | I assume hybrid is for both analog and digital lines |
| 04:33.04 | moe` | dunno, still just early research here :) |
| 04:33.12 | WIMPy | You got digital lines? |
| 04:35.16 | moe` | not yet |
| 04:35.41 | moe` | but if going to spend a bunch of cash on a card (after getting the cheap analog for testing)... might as well allow for the future |
| 04:36.07 | WIMPy | What's your location? |
| 04:36.28 | moe` | where the box is going is Canada |
| 04:36.33 | moe` | I'm in Curacao (Caribbean) |
| 04:36.48 | WIMPy | Then you should check, what's available there. |
| 04:37.11 | moe` | if digital lines are not available today, they will be in 5 years from now, that's the thinking |
| 04:37.14 | WIMPy | If they are anything like the US you can be sure you'll never need it. |
| 04:37.24 | WIMPy | Unlikely. |
| 04:37.39 | WIMPy | Phone lines are becoming extinct. |
| 04:37.41 | moe` | well, I was assuming that where I am, Curacao, both digital and analog lines are available |
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| 04:37.58 | moe` | I can agree with that, they are dying out |
| 04:38.36 | WIMPy | So if they aren't available now, they probably never will be. |
| 04:49.53 | moe` | point taken |
| 04:49.57 | moe` | true enough |
| 04:50.13 | moe` | will migrate from analog landlines to part of an ISP package |
| 04:50.24 | WIMPy | The good times are over. |
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| 05:05.02 | Garibaldo | anyone uses chan_dongle ? |
| 05:05.48 | WIMPy | Unless they have given up there have been users. |
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| 06:12.21 | moe` | wonder if I should inflict more pain on myself and try to get the SIP twilio to register |
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| 07:01.26 | moe` | what what what |
| 07:01.38 | moe` | twilio says register is not necessary |
| 07:04.35 | moe` | just the [block] in sip.conf |
| 07:04.37 | moe` | uhhh |
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| 07:36.10 | moe` | oddly, now I have inbound from twilio without a register statement in sip.conf - as their docs say |
| 07:36.14 | moe` | just can't get outbound |
| 07:36.16 | moe` | hmmm |
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| 07:44.43 | moe` | well I'll be damned |
| 07:44.46 | moe` | just got it working |
| 07:50.06 | moe` | well now that's amusing, anyone heard of a SIP trunk not needing a register statement in sip.conf to work? |
| 07:52.26 | WIMPy | Sure |
| 07:53.10 | WIMPy | Usually usable at home. |
| 07:54.18 | moe` | awwww via twilio SIP I cannot call myself (Curacao, 5999 area) |
| 07:54.49 | moe` | well after all my bitching and moaning about twilio, I *finally* followed the docs almost to the letter, and guess what |
| 07:54.54 | moe` | <---- stubborn |
| 07:55.43 | moe` | well I understood (misunderstood?) that a register statement was required in inbound calls |
| 07:55.46 | WIMPy | Argh. |
| 07:55.49 | moe` | outbound, not so much |
| 07:55.55 | WIMPy | Why did I use "-" in peer names? |
| 07:56.13 | moe` | I use that a lot actually |
| 07:56.17 | moe` | problem? |
| 07:57.24 | WIMPy | Well, channel names sometimes contain the peer name up to a "-". |
| 07:57.45 | moe` | oh |
| 07:57.56 | moe` | yeah, that's a bitch to parse out then if you're into scripts or whatever |
| 07:58.26 | WIMPy | I hope the other part doesn't contain "-". |
| 07:58.53 | WIMPy | Parsing names is bad anyway. |
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| 08:00.55 | WIMPy | Unfortunately I don't see a good way available. |
| 08:01.28 | moe` | meh, I use grep/awk/sed to parse the output of sip show peers and made a webpage out of it |
| 08:01.48 | WIMPy | I do it via AMI. |
| 08:02.21 | WIMPy | And I just threw out the extension status parsing so I can display a list of calls instead of just a single state. |
| 08:07.40 | moe` | that sounds like fun... what's the CLI for show current calls? (suppose I could search for it) |
| 08:08.13 | WIMPy | I build the list in the daemon. |
| 08:11.15 | moe` | ahahahaha I have asterisk setup to call all my softphones and cells at the same time, and I just did a bunch of quick dial and hangups... man, everything was ringing and would not stop |
| 08:11.43 | moe` | anyways, so you're doing coding for stuff in the daemon itself then, eh? |
| 08:12.18 | WIMPy | It IS a daemon listening to what Asterisk has to tell. |
| 08:12.46 | WIMPy | And then there's th web thign that displays what that daemon found out about. |
| 08:18.09 | moe` | ah, ok, so a socket your daemon has that then is queried by a web-thingy |
| 08:18.51 | WIMPy | Might be nice, but it just writes some files. |
| 08:20.17 | moe` | that's what 'nix is all about... its a text processing OS from the shell/script point of view |
| 08:20.27 | moe` | so yeah, I hear that |
| 08:21.10 | WIMPy | I guess with the cached data I could just write out the html from the daemon. |
| 08:21.27 | WIMPy | Maybet I'll try that next. |
| 08:21.59 | moe` | more flexible if the daemon writes the raw data and its then parsed into HTML outside of the daemon process |
| 08:22.03 | moe` | IMHO |
| 08:22.18 | WIMPy | true, but also slower. |
| 08:22.22 | moe` | indeed |
| 08:22.32 | moe` | but then the raw data can be sucked into any number of things |
| 08:22.40 | moe` | and they won't then have to parse out HTML |
| 08:22.49 | WIMPy | Yes |
| 08:28.41 | moe` | I'd leave the daemon putting out raw data, have another process HTMLize it |
| 08:29.51 | WIMPy | Yes, but at 10-40ms ist's not really efficient. |
| 08:33.39 | moe` | unless you have a lot of load on that, I'd call it a non-issue |
| 08:34.02 | moe` | that's little more than a good ping time |
| 08:34.25 | WIMPy | No, it's fine, just not nice. |
| 08:34.34 | moe` | right now, for example, from my asterisk instance, I'm 45ms ping to google |
| 08:39.23 | moe` | well, now I'm happy as a pig in shit... got toll free numbers via flowroute, got twilio hooked up |
| 08:39.44 | moe` | my skype SIP is busted, support cannot figure out why I cannot allocate channels or credit to my SIP profile |
| 08:39.45 | moe` | heh |
| 08:39.48 | moe` | stupid MS |
| 08:39.56 | moe` | so I'm gonna migrate away from that |
| 08:41.11 | moe` | already mapped what would normally be skype SIP outbound over to flowroute |
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| 11:19.34 | Phil` | when an extension is called, I want to dial 2 other phone numbers and bridge the 3 calls together |
| 11:19.40 | Phil` | what's the recommended logic for that? |
| 11:20.02 | Phil` | if I use Dial() after Dial(), it'll only do the second if the first didn't answer |
| 11:20.13 | Phil` | and if I do Dial() with 2 numbers, it'll stop ringing one when the other answers |
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| 11:33.26 | moe` | confbridge? |
| 11:33.42 | moe` | dunno |
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| 11:42.55 | moe` | Phil` - I know what you mean with Dial() after Dial() or Dial() two at the same time. don't have an answer. wait for someone with half a brain to answer your question. :) |
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| 12:27.31 | wdoekes | Phil`: moe` is right. you'll dial the invididual numbers and put the other leg into a confbridge |
| 12:27.43 | wdoekes | Phil`: see for instance 'core show help channel originate' |
| 12:28.51 | wdoekes | you'll originate one end with Tech/Destination (e.g. SIP/+12345) and the other with application ConfBridge and ID (or a custom piece of dialplan) |
| 12:36.36 | moe` | thanks wdoekes for the backup, this was beyond my experience |
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| 12:40.57 | moe` | == grasshopper on asterisk |
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| 13:11.46 | moe` | <PROTECTED> |
| 13:12.10 | moe` | that'll show whats going on with that SIP trunk |
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| 13:12.50 | moe` | the <peer/name> is what's in the sip.conf [block] |
| 13:14.02 | moe` | so that's a means to isolate details for a specific SIP connection |
| 13:14.39 | moe` | that's how I isolated the twilio issues I've had for the past week |
| 13:15.43 | moe` | maybe I'm just shitting out my ass but I think these items have a small degree of value |
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| 13:17.30 | moe` | file malcolmd gerhard7 does what I say make no sense? |
| 13:17.38 | moe` | I'm a grasshopper |
| 13:17.56 | moe` | so I invoke the greybeards |
| 13:18.00 | moe` | tell me |
| 13:19.36 | moe` | awaits an answer |
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| 13:22.33 | moe` | TK is not here |
| 13:22.44 | moe` | he's the yoda AFAIK |
| 13:27.10 | moe` | file |
| 13:27.13 | moe` | malcolmd |
| 13:27.18 | moe` | gerhard7 |
| 13:27.30 | moe` | you there? |
| 13:28.03 | moe` | TK is not here |
| 13:29.43 | wdoekes | who are you talking to? |
| 13:30.00 | moe` | wdoekes I did not mean to bypass your skills |
| 13:30.04 | wdoekes | haha |
| 13:30.08 | wdoekes | didn't mean that |
| 13:30.42 | wdoekes | your monologue looks directed at someone |
| 13:30.53 | wdoekes | but I don't see anyone else talking |
| 13:31.04 | moe` | was not directed at any particular victim |
| 13:31.15 | wdoekes | what's the question then? |
| 13:31.26 | moe` | for my part, no question |
| 13:31.29 | moe` | I'm ok |
| 13:31.58 | moe` | I was trying to answer another user's query |
| 13:32.49 | moe` | it was related to confbrige I believe |
| 13:33.20 | moe` | perhaps I should just shut the hell up on this channel and not be so verbose. |
| 13:34.10 | moe` | without this #asterisk help I would not have what I have now, so I was stupidy trying to "help" |
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| 13:35.16 | moe` | and recognition of the greybeards |
| 13:36.00 | moe` | wdoekes file malcolmd [TK-Defender] and more |
| 13:36.18 | moe` | dudes that have helped me |
| 13:37.27 | moe` | when I first bombed in here, I was useless |
| 13:37.49 | moe` | now I have three SIP trunks and am actually useful. |
| 13:37.55 | moe` | my own config sucks |
| 13:38.06 | moe` | but its a learning process |
| 13:38.14 | moe` | and people share man |
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| 13:40.04 | Hypfer | hi, how do i steal an active call from another extension? |
| 13:40.26 | moe` | yeah, why would you want to do that? |
| 13:40.43 | moe` | not my business |
| 13:40.57 | Hypfer | because I have that one sip trunk that rings constantly and I want my desk phone not to ring every time but just let the blf light light up |
| 13:41.12 | Hypfer | and when I press the blf light the caller should be transfered from waiting music to me |
| 13:41.26 | Hypfer | I have a lot of retarded people calling that number ._. |
| 13:42.19 | Hypfer | s/retarded people/children |
| 13:42.22 | moe` | not it's business. can't help you. |
| 13:42.35 | Hypfer | not possible with asteriskà |
| 13:42.47 | moe` | oh I'm sure its possible |
| 13:42.57 | Hypfer | but? |
| 13:43.08 | moe` | I'm no guru |
| 13:43.13 | Hypfer | ah I see |
| 13:43.17 | moe` | I cannot tell you how |
| 13:44.12 | moe` | do not take me as representing asterisk |
| 13:44.30 | moe` | I'm just after a few dozen beer |
| 13:44.35 | moe` | that's it |
| 13:46.45 | moe` | well on the dialplan the _[0-9] makes some sense |
| 13:47.18 | moe` | I have 9, 7, or 5 prefix for dial out means something |
| 13:47.26 | moe` | each selects a SIP trunk |
| 13:47.45 | moe` | so what is the nature of your question? |
| 13:48.41 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 13:51.39 | *** join/#asterisk stateless (~sin@saturn.2f30.org) |
| 13:53.11 | stateless | i am using a callback service, i would like to know if there are any feature phones that allow me to select a contact and pass through the number to the active call (currently i need to remember the number and type it in manually) |
| 13:53.26 | stateless | i understand this is probably not the right channel but some of you might know the term for this so i can look it up |
| 14:00.12 | *** join/#asterisk italorossi (~Adium@177.193.104.232) |
| 14:14.49 | WIMPy | Hypfer: See application Bridge(). |
| 14:18.13 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 14:18.16 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
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| 14:20.16 | moe` | WIMPy ++ |
| 14:27.27 | *** join/#asterisk rubio (~rubio@190.18.2.110) |
| 14:29.06 | rubio | Hi there, i have an agi script that will make a "transfer", in order to do this, i execute DIAL with G option, then i execute 2 new AGI on each channel, then i need to connect calling and called, but i can't. anyone can help me with this? |
| 14:31.21 | WIMPy | Same answer: See application Bridge(). |
| 14:35.53 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 14:42.38 | Hypfer | WIMPy: also, I can't manage to get the blf to work with my openstage 40 |
| 14:43.37 | Hypfer | the hint is working |
| 14:43.38 | Hypfer | <PROTECTED> |
| 14:43.38 | Hypfer | <PROTECTED> |
| 14:44.05 | WIMPy | Watchers 0 <- so noone's interested. |
| 14:44.29 | Hypfer | so its the phones fault |
| 14:48.37 | Hypfer | i've set the "program key" to "server feature" and "LED Control" to 42@asterisk_server_ip |
| 14:48.40 | Hypfer | http://wiki.unify.com/wiki/Asterisk_Feature_Busy_Lamp_Field_(BLF) |
| 14:48.44 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 14:48.44 | Hypfer | still, it doesn't connect |
| 14:48.49 | rubio | hi WIMPy, i was (before asking here) cheking on bridge app, but my problem is that i can't find the calling channel to pass to bridge when called answer :S i have test with Set(__Original_channel=${channel}) on calling 1st, but no result :( |
| 14:49.45 | rubio | any suggestion? |
| 15:02.05 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-wiyzbrlfuzqdetbm) |
| 15:12.56 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 15:19.12 | Hypfer | just can't get the phone to watch a hint |
| 15:22.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:23.25 | Hypfer | so if someone here uses openstage phones with asterisk I would be glad if they could help me out :-) |
| 15:24.40 | WIMPy | has only seen the T version so far. |
| 15:25.21 | WIMPy | Have you tried to subsrcibe to only the user part or including proxy? |
| 15:25.43 | *** join/#asterisk areski (~areski@80.174.128.122.dyn.user.ono.com) |
| 15:26.26 | rubio | i have solve the bridge issue, my G param on dial goes not to extension = channel and bridge param is exten :D |
| 15:42.51 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 17:12.11 | moe` | well that worked, the new SIP trunk |
| 17:12.12 | moe` | cool |
| 17:13.06 | moe` | twilio |
| 17:13.44 | moe` | better sound/call quality than skype SIP for sure, and apparently better than flowroute even |
| 17:13.49 | moe` | strange, but true |
| 17:14.14 | moe` | Canadian target it was, though |
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| 18:16.32 | *** mode/#asterisk [+o file] by ChanServ |
| 18:31.07 | veleiro | is there a way to easily stop brute force attempts for trying to register with asterisk? Registration from '"0121" <sip:0121@(ip):5060>' failed for '89.163.140.45:5070' - Wrong password .. Call from '' (88.150.240.96:5071) to extension '001442032903414' rejected because extension not found in context 'default' |
| 18:31.35 | veleiro | i suppose its because its reachable from public so i have to deal with that now |
| 18:32.21 | veleiro | but like sshguard for example will block hammer attempts after so many tries |
| 18:34.31 | veleiro | i guess i just need to know generally more about sysadmining |
| 18:35.54 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:cc60:4bf0:83ed:a976) |
| 18:49.48 | [TK]D-Fender | You are allowing un-authed calls in the first place which is BAD |
| 18:49.54 | [TK]D-Fender | and it's hitting your dialplan |
| 18:50.00 | [TK]D-Fender | "allowguest=no" <------- |
| 18:50.22 | [TK]D-Fender | Next, go setup something to firewall out attackers |
| 18:50.38 | [TK]D-Fender | either rate-limiting attmpts, fail2ban, etc |
| 18:51.19 | veleiro | thx |
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| 19:29.25 | igcewieling1 | veleiro: I allow the registrations and force them into a deadend context. |
| 19:35.30 | veleiro | igcewieling1: is that better than banning? |
| 19:49.58 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:50.23 | moe` | hey [TK]D-Fender I finally fixed my twilio issue |
| 19:50.37 | moe` | there was no register statement required |
| 19:50.45 | moe` | that's very strange to me |
| 19:50.55 | moe` | but inbound and outbound works |
| 19:51.43 | moe` | but it mostly came to RTFM |
| 19:51.58 | moe` | which is what you guy always suggest |
| 19:53.29 | moe` | so thanks for your tolerance on my bullshit for the past 5 days or so on this. |
| 19:56.08 | moe` | thanks guys. |
| 19:57.03 | *** join/#asterisk scinawa (~scinawa@2-230-246-32.ip204.fastwebnet.it) |
| 19:57.07 | scinawa | hi guys |
| 19:57.23 | scinawa | i ran into this bug |
| 19:57.25 | scinawa | https://issues.asterisk.org/jira/browse/ASTERISK-17722 |
| 19:57.29 | scinawa | anyone knows a solution? |
| 19:57.47 | scinawa | let /etc/hosts point to an it does not work |
| 19:57.53 | scinawa | since it's a SRV section |
| 19:58.23 | scinawa | running a dns server looks like a little overkill |
| 20:00.55 | moe` | "Just a moment, we're improving your Skype experience" |
| 20:01.03 | moe` | yeah |
| 20:01.56 | moe` | somehow that presents itself as an oxymoron |
| 20:14.21 | *** join/#asterisk last1 (~last1@109.103.111.89) |
| 20:15.26 | last1 | how can I debug an 'Unauthorized' message |
| 20:15.31 | last1 | see exactly why it says unauthorized |
| 20:21.57 | [TK]D-Fender | Because the auth is wrong. user and/or pass |
| 20:22.31 | last1 | but... it's not. I checked 100 times |
| 20:22.42 | last1 | if I switch to channel sip, it authenticates. same data in pjsip says unauthorized |
| 20:22.45 | [TK]D-Fender | Show us the actual comm |
| 20:22.57 | [TK]D-Fender | And actual config |
| 20:23.43 | last1 | http://pastebin.com/3YN6DE8a |
| 20:23.58 | last1 | Trying to connect a Cisco 7970g |
| 20:24.51 | [TK]D-Fender | [2015-11-07 15:20:58] DEBUG[21522]: res_pjsip_endpoint_identifier_ip.c:128 ip_identify: No identify sections to match against |
| 20:25.09 | last1 | yeah, but then it identifies by user no ? |
| 20:25.22 | last1 | res_pjsip_endpoint_identifier_user.c:104 username_identify: Retrieved endpoint 612 |
| 20:25.27 | moe` | omg how complex is this, now even twilio wants bullshit so I can call international numbers |
| 20:25.52 | [TK]D-Fender | Where did you setup how it is supposed to identify and auth them? |
| 20:28.39 | moe` | hey [TK]D-Fender, a query.... what about setting up failover outbound. right now I have 9 prefix skype, 7 prefix flowroute, 5 prefix twilio.... suggestions on dialplan for that? |
| 20:28.58 | last1 | in pjsip.conf |
| 20:29.28 | [TK]D-Fender | ... |
| 20:29.35 | [TK]D-Fender | doesn't tell us things are set up right |
| 20:29.35 | last1 | coming now |
| 20:29.52 | moe` | or is dialplan the wrong place to address that? |
| 20:30.05 | last1 | is there more debug output available to see exactly how it tries to authenticate ? |
| 20:31.04 | moe` | sip set debug peer "whoever" |
| 20:31.25 | moe` | where "whoever" is a [block] in your sip.conf |
| 20:32.02 | moe` | to turn it off, "sip set debug off" |
| 20:32.08 | last1 | i'm on pjsip... and that will just show me sip packets anyways |
| 20:32.16 | moe` | that assumes you have an asterisk -rv type CLI |
| 20:32.22 | last1 | I want to know what asterisk is doing with the information |
| 20:32.29 | last1 | http://pastebin.com/dtd6VmrV |
| 20:32.52 | moe` | [TK]D-Fender this is all you man, as you know I'm a grasshopper |
| 20:34.42 | last1 | could this be because I'm forwarding the requests to the second nic of my server ? |
| 20:34.57 | last1 | I have other devices which register via the public IP and they appear to be fine |
| 20:35.06 | last1 | even though they are behind nat |
| 20:35.22 | last1 | but these cisco phones on the local network give this Unauthorized message |
| 20:35.37 | moe` | what are they trying to auth to? |
| 20:35.50 | [TK]D-Fender | Looks like an initial 401 to the register which should be a challenge for the phone to come back with auth |
| 20:36.03 | moe` | I'm shitting out my arse here, just trying to be somewhat useful |
| 20:36.12 | [TK]D-Fender | Which since you aren't showing any more... I'm suspecting the phone isn't doing |
| 20:37.10 | last1 | let me go to tcpdump |
| 20:37.16 | moe` | hey [TK]D-Fender did I mention already, have you heard of SIP inbound working without a register statement? |
| 20:37.26 | moe` | twilio does that |
| 20:38.18 | moe` | strange to me, but I'm a small fish in a big pond. |
| 20:39.59 | last1 | no, I only have that. the Register and then 401 Unauthorized |
| 20:40.29 | last1 | why would it throw an Unauthorized for the Register command ? |
| 20:40.41 | moe` | yeah, well, exactly |
| 20:40.52 | moe` | I saw that with twiliio SIP |
| 20:41.07 | moe` | it does not accept register requests |
| 20:41.22 | moe` | so what is the nature of the SIP you're trying to register to? |
| 20:41.54 | [TK]D-Fender | <[TK]D-Fender> Looks like an initial 401 to the register which should be a challenge for the phone to come back with auth |
| 20:42.25 | last1 | this part you mean: WWW-Authenticate: Digest realm="asterisk",nonce="1446928715/7e0a802656646024c2acef1efb122386",opaque="5dbc90b71fe97e48",algorithm=md5,qop="auth" |
| 20:42.33 | last1 | maybe the 7970 doesn't know md5 ? |
| 20:44.29 | [TK]D-Fender | not that... |
| 20:44.54 | [TK]D-Fender | as I've seen in chan_sip they don't like "nat=yes" regardless and refuses to answer. |
| 20:45.12 | [TK]D-Fender | not sure what pjsip is doing in its answer for an auth challenge |
| 20:45.29 | last1 | does pjsip even support nat=no ? |
| 20:45.38 | last1 | I thought that was only on sip |
| 20:45.50 | last1 | because my phones are not behind a nat...they are local |
| 20:49.52 | moe` | each phone has a real IP? |
| 20:49.59 | [TK]D-Fender | add : force_rport=no |
| 20:50.05 | [TK]D-Fender | under the endpoint |
| 20:50.25 | moe` | [TK]D-Fender tell me to get lost if my comments are useless. |
| 20:51.03 | last1 | by the way, this is what a valid registration looks like under the old asterisk |
| 20:51.09 | last1 | I switched back so I could record the convo |
| 20:55.06 | last1 | http://pastebin.com/2pCetGpj |
| 20:56.26 | last1 | I see this missing from the new asterisk response in the unauthorized message: uri="sip:10.1.0.1" |
| 20:56.55 | last1 | works: Authorization: Digest username="612",realm="asterisk",uri="sip:10.1.0.1",response="c08c784a8c48667bb63f17729b660a00",nonce="4b2941c9",algorithm=MD5 |
| 20:57.20 | last1 | ah wait, I confused packets |
| 20:58.13 | last1 | this doesn't work: WWW-Authenticate: Digest realm="asterisk",nonce="1446927658/66310989aab0003f6998d9f06370a696",opaque="11299e555f7c96db",algorithm=md5,qop="auth" |
| 20:58.37 | last1 | this works: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b2941c9" |
| 21:01.13 | moe` | ok to step back a moment could I ask what changed? |
| 21:01.46 | last1 | asterisk versions... from 1.4 to 13.5 :) |
| 21:01.56 | moe` | 1.4 to 13.5 |
| 21:01.58 | moe` | wow |
| 21:02.00 | moe` | ok |
| 21:02.04 | moe` | I'm useless there |
| 21:02.11 | moe` | that's a drastic change |
| 21:02.35 | moe` | I'll stop making stupid comments |
| 21:03.09 | last1 | yeah, here's why |
| 21:03.11 | last1 | from a forum |
| 21:03.12 | moe` | (my experience is asterisk 13.x) |
| 21:03.16 | last1 | I've no idea why the last poster has a problem as they haven't provided any information, but, from the trace, it looks to me as though the Cisco isn't prepared to ignore new parameters (opaque) on the WWW-Authenticate. I don't use PJSIP, so I don't know if you can eliminate that parameter without modifying the source code and recompiling, or interposing a suitably programmed proxy. |
| 21:04.09 | moe` | is that the proper auth type? |
| 21:04.23 | moe` | I have no idea on this matter I'm just shooting thoughts |
| 21:04.31 | last1 | the recommend solution is what F-Fender said: force_rport=no, but I tried that and still nothing |
| 21:05.07 | moe` | see if we can rattle file, malcolmd, drmessano, [TK]D-Fender |
| 21:05.19 | moe` | WIMPy |
| 21:05.49 | moe` | these guys will likely know the answer |
| 21:06.10 | moe` | I can't help you, I'm an asterisk grasshopper |
| 21:08.17 | moe` | yeah gerhard7 too |
| 21:08.25 | moe` | too bad he just left |
| 21:13.23 | last1 | D-Fender, you're a genius |
| 21:13.27 | last1 | I wasn't using force_rport properly |
| 21:13.35 | last1 | enabled that stuff and it registered |
| 21:13.59 | moe` | [TK]D-Fender is an asterisk yoda |
| 21:14.49 | moe` | he's mostly guided me from point zero to where I am now with my own asterisk setup |
| 21:15.28 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 21:17.55 | *** join/#asterisk LostBoy (sid110950@gateway/web/irccloud.com/x-dpokuymbylnghpph) |
| 21:26.34 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-gjzgbmmqwgjhekps) |
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| 21:34.13 | WIMPy | Hmm. I haven't really paid attention to that detail before, but is it normal that session-timers produce very noticable audio gaps? |
| 21:34.29 | *** join/#asterisk wyoung (~wyoung@wesleyy.com) |
| 21:37.24 | moe` | ok, I'm gonna towel out, and stop offering stupid comments. |
| 21:38.13 | WIMPy | And why do thy tear the bridge each time at all? |
| 21:40.25 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 22:02.24 | scinawa | how can I prevent a Dial() to execute the same context again after an answer? (It seems that I cannot use g option) |
| 22:02.26 | last1 | D-Fender: so now with force_rport I can make calls to other local phones, I can even dial out through a trunk |
| 22:02.29 | *** join/#asterisk pjm (~pjm@uhfsatcom.plus.com) |
| 22:02.43 | last1 | but I can't call the phone itself from other phones |
| 22:02.48 | scinawa | exten => 3000,n,Dial(SIP/msg1/${tocall},,M(authcaller)g) is what I write |
| 22:02.58 | scinawa | and the macro gets executed perfectly |
| 22:03.26 | last1 | pjsip show endpoints shows this: http://pastebin.com/DWWe9MuE |
| 22:03.45 | last1 | and: Endpoint 612 is now Unreachable |
| 22:12.44 | WIMPy | scinawa@ Not sure what your questionis. If you don't want the dialplan to continue, DON'T use 'g'. |
| 22:13.39 | scinawa | i want the dialplan to continue, but it happend that after the macro call, it restart from the first exten instead on continuing.. |
| 22:13.46 | scinawa | (so I get 2 dial..) |
| 22:14.13 | WIMPy | That must be due to somethin else in your dialplan. |
| 22:14.28 | scinawa | ouch |
| 22:15.57 | scinawa | Channel: Local/3000@chiamante |
| 22:15.57 | scinawa | MaxRetries: 0 |
| 22:15.57 | scinawa | RetryTime: 30 |
| 22:15.57 | scinawa | WaitTime: 30 |
| 22:15.57 | scinawa | Context: chiamante |
| 22:15.58 | scinawa | Extension: 3000 |
| 22:16.09 | scinawa | maybe because I write 2 time the extension and the context in my callfile? |
| 22:16.20 | WIMPy | ~pb |
| 22:16.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:16.28 | scinawa | :v |
| 22:16.40 | WIMPy | And that is not your dialplan. That is a call file. |
| 22:17.07 | scinawa | yes |
| 22:17.24 | scinawa | it's the callfile that starts the context at extension 3000 |
| 22:18.35 | scinawa | http://pastebin.com/Cvz6DmhS |
| 22:18.41 | scinawa | i suspect is this |
| 22:18.55 | scinawa | I deeply suspect this. |
| 22:21.13 | WIMPy | Well, first of all you're calling the same extension twice. That's probably not what you want. |
| 22:21.42 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 22:21.44 | WIMPy | And secondly I don't see the "3001" you want to Goto. |
| 22:21.46 | scinawa | definitely |
| 22:22.44 | scinawa | but without Context: chiamante and Extension: 3000 the callfile is invalid |
| 22:23.20 | scinawa | so maybe i just had some wrong idea of how callfiles works right? :\ |
| 22:23.20 | WIMPy | You're telling it to call 3000@chiamante and connect it to 3000@chiamante. |
| 22:23.28 | scinawa | yep, fml. |
| 22:23.30 | WIMPy | What do you want to do? |
| 22:24.50 | scinawa | call 3000@chiamante, execute the dial, authenticate the user and keep listening |
| 22:25.17 | scinawa | so instead of using a macro i just have to create a new context and point my callfile there |
| 22:25.49 | WIMPy | To two different locations, yes. |
| 22:26.17 | scinawa | WIMPy: TNX. |
| 22:26.33 | scinawa | my savior :D |
| 22:26.40 | WIMPy | But if you want to call some phone an then do something, specify that phone in the Channel filed. No need to use a local channel then. |
| 22:27.07 | scinawa | yes, because i need to add an header |
| 22:27.16 | scinawa | in order to overwrite the callerid |
| 22:27.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:27.30 | WIMPy | Ok, yes. |
| 22:27.34 | scinawa | :) |
| 22:31.09 | last1 | WIMPy: do you have any experience with Cisco 79XX phones ? |
| 22:31.44 | WIMPy | Not really. Tried to use one once, but it won. |
| 22:34.13 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:34.13 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:36.52 | last1 | Have you seen another device not respond to an OPTIONS request ? http://pastebin.com/c2nYUxmK |
| 22:37.17 | last1 | the Cisco phone registers but then I think asterisk sends multiple OPTIONS packets, the phone doesn't answer and becomes unavailable |
| 22:37.25 | last1 | I can still call out from it, but not make calls to it |
| 22:38.25 | WIMPy | You can always disable qualify. |
| 22:40.03 | last1 | I bet you that's what that is |
| 22:40.16 | last1 | but qualify was set to yes/no in sip |
| 22:40.22 | last1 | in pjsip it's 60 seconds |
| 22:40.26 | last1 | can it be disabled ? |
| 22:44.05 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:48.04 | WIMPy | Maybe 0? |
| 22:48.07 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-62-18.om.om.cox.net) |
| 22:48.33 | WIMPy | But the time has always been configurable. |
| 22:52.11 | last1 | yeah, in the manual they say: Interval between attempts to qualify the contact for reachability. If 0 never qualify. Time in seconds. |
| 22:52.22 | last1 | but even at 0, it never rings |
| 22:52.40 | last1 | I mean, I see the SIP packets arriving to the phone via tcpdump...but the fscker won't ring |
| 22:52.50 | WIMPy | Then it really isn't reachable I guess. |
| 22:53.11 | WIMPy | Or it doesn't like what it receives. |
| 22:53.38 | last1 | let me compare with what asterisk 1.4 was sending |
| 22:56.54 | WIMPy | Wow. That's a long time ago. |
| 22:58.20 | last1 | yeah...that's what I'm replacing |
| 22:58.31 | last1 | still works like a charm, but it's an old centos box..security, issues, etc |
| 22:58.34 | last1 | it's gotta be done |
| 23:02.20 | last1 | I think I know what the issue is |
| 23:03.25 | last1 | http://pastebin.com/bUdRMtjE |
| 23:03.30 | last1 | first packet doesn't work, second works |
| 23:04.04 | last1 | I noticed this line here: o=- 314644781 314644781 IN IP4 69.90.179.5 <-- doesn't work, it uses the server's public IP. the phone is local |
| 23:04.21 | last1 | the one that works has this: o=root 1465 1465 IN IP4 10.1.0.1 |
| 23:04.45 | last1 | where is that IP specified ? |
| 23:05.57 | WIMPy | Don't know what the keywords are with pjsip. |
| 23:06.15 | last1 | what is it with sip ? |
| 23:09.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 23:11.54 | WIMPy | externhost/externaddr, localnet and nat. |
| 23:12.56 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:14.38 | last1 | thx, let me try that |
| 23:21.56 | last1 | no luck |
| 23:21.59 | last1 | it still rings with: o=- 1606565090 1606565090 IN IP4 69.90.179.5 |
| 23:22.41 | last1 | despite trying to force it to use the local ip |
| 23:22.42 | last1 | http://pastebin.com/LbPf1Abb |
| 23:25.16 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 23:28.59 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 23:30.36 | veleiro | so the asterisk management interface is just the CLI? |
| 23:32.04 | WIMPy | Well, the "Asterisk Management Interface" or AMI for short is text based TCP protocol for remote control. |
| 23:32.13 | WIMPy | But the administrative interface is te *CLI, yes. |
| 23:32.51 | veleiro | it looks similar somehow |
| 23:33.13 | WIMPy | No |
| 23:34.26 | veleiro | k, im going to go ahead and say its not what im looking for and leave it disabled |
| 23:34.53 | WIMPy | You need AMI for 3rd party applications. |
| 23:36.17 | veleiro | im looking to run a script that forwards a message from my sip trunkator to one of my internal asterisk dialplans |
| 23:36.28 | veleiro | so i think either the AMI or agi |
| 23:37.01 | WIMPy | Not sure what that means. |
| 23:38.19 | veleiro | twilio has a request URL you can provide for when someone sends you a message to the phone number they have |
| 23:38.32 | last1 | I officially don't have my external IP listed anywhere yet it's still in that packet |
| 23:38.40 | last1 | where could it be coming from ? |
| 23:39.11 | veleiro | im trying to learn more so i can just forward any incoming message to my asterisk user account |
| 23:40.28 | WIMPy | last1: A STUN server. |
| 23:40.52 | WIMPy | veleiro: What kind of URL? |
| 23:41.01 | veleiro | http post or get |
| 23:41.04 | WIMPy | http? sip? |
| 23:41.57 | veleiro | i can do the scripting part, but i suppose i need somehow to send commands to my asterisk |
| 23:42.23 | WIMPy | Depends on whay you want to do with the message. |
| 23:42.33 | WIMPy | what |
| 23:43.01 | veleiro | yes simple enough just send any message to a exten in the dialplan |
| 23:43.45 | WIMPy | To get to the dialplan you need a call. So you need to originate a call. |
| 23:43.54 | [TK]D-Fender | "send commands" is VERY vague |
| 23:44.11 | last1 | WIMPy: but I don't have any stun servers configured |
| 23:44.16 | last1 | that's just my public IP |
| 23:44.43 | veleiro | well, so, i figured out how to do messaging in between accounts on my asterisk server |
| 23:45.06 | veleiro | and thats now in the dialplan, for example exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) |
| 23:45.40 | [TK]D-Fender | veleiro, If you want * to GENERATE a call then you would use an ORIGIANTE or Call-File |
| 23:45.47 | [TK]D-Fender | And have that dial up a Local Channel |
| 23:47.26 | veleiro | its not a call though..? sorry total asterisk newb |
| 23:47.49 | WIMPy | Yes, but dialplan is only about calls. |
| 23:48.06 | WIMPy | So you need a call to do anything there. |
| 23:48.51 | [TK]D-Fender | veleiro, Everything hitting the dialplan is a CHANNEL |
| 23:49.00 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:49.00 | [TK]D-Fender | Some start when something CALL * |
| 23:49.12 | [TK]D-Fender | You can make * call OUT using Originate |
| 23:49.37 | [TK]D-Fender | And you can have that callout use the dialplan to execute things instead of dialing a normal comm type device (SIP, etc) |
| 23:53.04 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:1413:8219:34ff:fecf:17f0) |
| 23:53.54 | gusto | hey, when a peer comes over TLS, does it kill the connection after every failed login attempt? or does it keep up the TCP port open so that he can potentially set thousends of attempts over it |
| 23:54.07 | gusto | hm? |
| 23:55.54 | veleiro | ok, well definitely need to understand the syntax better in the dialplan. im looking for using the originate or call-file correctly through scripting i believe. thank [TK]D-Fender and WIMPy |