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| 14:45.38 | blitzrage | morning all |
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| 16:31.11 | *** join/#asterisk-doc oej (~oej@40.186.204.213.sol.worldonline.se) |
| 16:31.17 | oej | I am here |
| 16:32.22 | blitzrage | oej: hello.. I need to ask you a question which I know you've answered before... but for some reason I keep forgetting, or not getting it written to permanent memory |
| 16:32.36 | blitzrage | When do we use username= in sip.conf since we authenticate the [username] portion? |
| 16:33.05 | blitzrage | I know it has to do with authenticating a call before authentication... or something to that effect....... |
| 16:34.46 | oej | Wow. My favourite question. I need to save this in a .txt file called "Blitzrage standard questions" |
| 16:35.02 | oej | Son, now you listen to me, right? |
| 16:35.06 | oej | :-) |
| 16:35.42 | oej | If you start with a blank Asterisk (nothing in the astdb) we don't know anything about peers that are dynamic until they register |
| 16:35.42 | blitzrage | I'm going to make sure this gets written down this time :D |
| 16:35.55 | blitzrage | I have a chapter here with the perfect section to answer this question |
| 16:36.07 | oej | So in order to call between startup and theyre first registration, we need to have something |
| 16:36.08 | blitzrage | I have a note that says, "Ask Olle one more time and get it written down this time" |
| 16:36.39 | oej | So basically we call with sip:<username>@<defaultip> - it's a chance we take that may work with some phones, but not all |
| 16:36.42 | blitzrage | oh right! |
| 16:36.42 | blitzrage | stupid me |
| 16:36.42 | blitzrage | lol |
| 16:36.49 | oej | Out of security, some hosts will deny that call |
| 16:36.57 | oej | Asterisk will soon too. Ha ha ha. |
| 16:37.06 | oej | If I can convince Mark about it |
| 16:37.11 | oej | Our sneak it in somehow |
| 16:37.20 | blitzrage | hrmmmm... but I guess that can't work if host=dynamic |
| 16:38.00 | blitzrage | lol... put it in with another patch :) |
| 16:38.57 | oej | What can't work with host=dynamic - all of this is for host=dynamic |
| 16:39.26 | oej | We should actually give an error message if host=<ip> and someone tries to set a defaultip=<ip> since they don't know what they're doing then |
| 16:39.52 | oej | Asterisk *REALLY* needs to learn about SIP uri's in a better way. |
| 16:41.21 | blitzrage | agreed. |
| 16:41.35 | blitzrage | I guess what I meant is that you can't just have host=dynamic withough a defaultip= |
| 16:41.49 | blitzrage | if you do.. then you can't try to send a call to the user before they register to us |
| 16:42.55 | oej | Right |
| 16:43.22 | blitzrage | ok cool, thanks |
| 16:43.23 | oej | Which is only a problem the first time you connect the phone to a new asterisk account. In other cases, we read the old registration from astdb and use that data. |
| 16:43.52 | oej | So in most cases, this is used a *very short* time and you can't really rely on it. |
| 16:44.19 | oej | The trick most phones do in order to avoid voip spamming is that they add a parameter to the contact header at registration, a "secret code" |
| 16:44.40 | oej | When we call them we use their contact header (one of my earlier patches) and send back the code |
| 16:44.56 | oej | But with defaultIP/username, we can't do that and the phone will just ignore the call |
| 16:46.27 | blitzrage | oh wow... that was a lot more info... ok, digesting and writing a paragraph :) |
| 16:46.57 | blitzrage | this chapter is going to be pretty big.... 31 pages so far.. and I haven't even gotten to IAX yet |
| 16:47.11 | blitzrage | just SIP stuff |
| 16:47.11 | blitzrage | so many nuances |
| 16:47.11 | blitzrage | IAX *should* be easier |
| 16:47.13 | oej | Look at the number of pages of SIP vs IAX2 in the training binder... |
| 16:47.18 | oej | And I've added pages to SIP :-) |
| 16:48.40 | blitzrage | lol... so true |
| 16:48.40 | blitzrage | so what's better? :D |
| 16:51.41 | oej | What do you mean? |
| 16:53.30 | blitzrage | I was just poking fun about what protocol is better; IAX or SIP |
| 16:55.16 | oej | Oh.. Tough one. ANyone else listening? |
| 16:55.34 | oej | IAX2 is good in some cases, a short term quick solution to a lot of problems |
| 16:55.40 | oej | SIP is better in the long run |
| 16:55.48 | blitzrage | nah... :) |
| 16:55.53 | oej | IAX2 is based on PSTN quite a lot, bringing PSTN over IP |
| 16:55.59 | oej | SIP is something very different |
| 16:56.16 | blitzrage | yah. I personally like IAX for trunking between boxes and SIP for phones on the LAN (or as much LAN as possible) |
| 16:56.22 | oej | So it's hard to compare them, since they both support audio over IP, but do that in different ways |
| 16:56.43 | oej | Yes, IAX2 is very good at building ASterisk clusters. I just wish I could forward SIP uri:s over it. |
| 16:56.51 | oej | We are working on fixing that. .. |
| 16:56.54 | blitzrage | I'm not sure IAX is flexible enough for phones like SIP is |
| 16:57.32 | oej | No, but there is a lot you can do. Look at our IAXphone pro that we are going to release soon :-) |
| 16:57.55 | oej | We have over 20 registrations for "meet asterisk" already. |
| 16:58.03 | blitzrage | :D |
| 16:58.15 | blitzrage | awesome! |
| 16:59.31 | oej | We have a room for 150 so we still have capacity! |
| 17:00.13 | blitzrage | ahhhh yes.... lots of room :) |
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| 20:17.40 | blitzrage | oej: do you have a minute to explain to me what a "native bridge" is? |
| 20:30.33 | *** part/#asterisk-doc jayden (~ircatjerr@65.170.43.34) |