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| 16:41.51 | *** join/#asterisk-bugs C4colo (n=C4colo@serenity.livewirenet.com) |
| 16:44.18 | C4colo | I'm willing to test t.38 in beta9, can someone point me in the right direction? I have 1.6-beta9 installed with a basic config, have three atas capable of t.38, two fax machines on analog lines ready to go |
| 16:44.44 | C4colo | I'd also be willing to test some other areas of 1.6 if needed as well |
| 16:46.39 | C4colo | is there a doc somewhere with even a rough description of what configuration needs to be enabled? |
| 16:47.17 | symlink | there is one option in sip.conf needed, t38pt_udptl=yes |
| 16:47.22 | symlink | rest is configuration of your ATAs... |
| 16:49.13 | C4colo | ok, I have that set to yes, and the first ata is registered and I have t.38 set to auto detect (other option is ulaw passthrough) |
| 16:52.56 | C4colo | thanks, I'll try some fax calls... I was expecting some hacking of source code or complex and tedious configuration of .conf files =) |
| 16:54.47 | *** join/#asterisk-bugs UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
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| 16:58.37 | UnixDog | jsmith: hey |
| 16:58.50 | jsmith | Hello UnixDog |
| 16:59.34 | UnixDog | yeah they are testing in the lab working on callcenter interface and wont let me touch the machines |
| 16:59.42 | UnixDog | I know the fastest way to test it |
| 17:00.48 | UnixDog | btw it also happens with polycom directory sip uri dials |
| 17:00.58 | UnixDog | lots to go into for fixiing this |
| 17:01.26 | UnixDog | ok phse one |
| 17:01.30 | jsmith | UnixDog: Again, let's start with the simplest case and go from there |
| 17:01.35 | UnixDog | ok |
| 17:01.39 | UnixDog | simple do this |
| 17:01.50 | UnixDog | 2 asterisk 1.4.20 boxes |
| 17:02.02 | UnixDog | and have 10 exten 1000 -1010 |
| 17:02.05 | jsmith | UnixDog: I've tried it |
| 17:02.07 | UnixDog | on both boxes |
| 17:02.13 | jsmith | UnixDog: And I've had two other people to try it |
| 17:02.28 | jsmith | UnixDog: Which is why I've asked for a SIP trace and CLI output from *your* boxes |
| 17:02.36 | Qwell | and config |
| 17:02.46 | jsmith | Yes |
| 17:03.06 | UnixDog | the only box I have right now is a askozia very basic dialplan |
| 17:03.31 | UnixDog | and I call my fiend box with a exten 1000 and it fails |
| 17:03.44 | UnixDog | even when calling his ring group of 600 |
| 17:03.54 | UnixDog | and it does not show up on his cli |
| 17:04.04 | UnixDog | adn very little output on mine |
| 17:05.32 | UnixDog | I will paste its output |
| 17:05.46 | jsmith | UnixDog: Make sure the verbosity is at least 3 on both sides |
| 17:05.58 | jsmith | UnixDog: And make sure you've got SIP debugging turned on |
| 17:07.58 | UnixDog | http://pastebin.ca/1034209 |
| 17:08.10 | UnixDog | nothing show up on his cli |
| 17:08.16 | UnixDog | only on min |
| 17:08.25 | UnixDog | mine |
| 17:08.30 | UnixDog | there is the output |
| 17:11.06 | UnixDog | and he has a exten 1000 on his system |
| 17:11.46 | symlink | do you mean he has a user, peer, or friend in sip.conf with a username of 1000? |
| 17:12.11 | jsmith | UnixDog: Based on that post, it appears that you've got an authentication problem, not a bug |
| 17:12.35 | UnixDog | no because if he deletes the exten on his box all calls go threw fine |
| 17:13.13 | jsmith | Ok, stick with me here, while I walk through what happened on that trace |
| 17:13.32 | UnixDog | I understand the trace |
| 17:13.41 | UnixDog | but I am telling you the facts |
| 17:13.43 | jsmith | Well, let's just walk through it for sanity's sake |
| 17:13.46 | UnixDog | ok |
| 17:13.52 | jsmith | Just to make sure we're all on the same page |
| 17:14.07 | jsmith | Line 11: Your Polycom 550 dials 22222 |
| 17:14.30 | jsmith | Line 44: Asterisk says 'Hey Polycom, you need to authenticate' |
| 17:15.03 | russellb | jsmith: box2 sees 1000 in the From header, so it asks it to authenticate as user 1000, since one exists |
| 17:15.22 | russellb | jsmith: if on box1, there was a peer entry for box2 ... |
| 17:15.25 | jsmith | russellb: I know that, but I want to step through this so that everyone understands |
| 17:15.29 | russellb | and that was used for dialing box2, it could fix it. |
| 17:15.33 | russellb | by allowing you to chante the username |
| 17:15.34 | russellb | ah. |
| 17:15.35 | russellb | k! |
| 17:15.44 | russellb | :-X |
| 17:15.53 | UnixDog | ? |
| 17:16.17 | jsmith | OK... let's continue |
| 17:16.56 | jsmith | Line 61: The Polycom ACKs the 487 response |
| 17:18.07 | UnixDog | ok |
| 17:19.47 | jsmith | Line 77: The Polycom tries again |
| 17:21.13 | jsmith | Line 110: Asterisk finds a matching SIP user named [1000] |
| 17:23.12 | jsmith | Line 128 Asterisk replies back to the Polycom saying "OK, I'm trying..." |
| 17:23.28 | jsmith | Line 143: The call finally his the dialplan |
| 17:23.31 | jsmith | UnixDog: With me so far? |
| 17:23.39 | UnixDog | yes |
| 17:23.40 | jsmith | So there's the first leg of the call |
| 17:25.06 | UnixDog | ok |
| 17:27.31 | jsmith | Line 144: Asterisk tries to call Local/22222@APPLICATION-MAPPING-226591954483c56528e30f/n |
| 17:27.47 | jsmith | (Sorry for the slow responses... trying to talk on a phone call, cook lunch, and type at the same time) |
| 17:28.01 | UnixDog | ok |
| 17:29.00 | jsmith | Line 155: Asterisk sends an INVITE to sip:600@ka3uww.homeip.net |
| 17:30.05 | UnixDog | correct |
| 17:30.06 | jsmith | Line 192: Asterisk receives a "Proxy Auth Required" |
| 17:30.36 | jsmith | Line 206: Asterisk ACKs the "Proxy Auth" message |
| 17:30.38 | UnixDog | thats because there is a exten on his system on 1000 |
| 17:31.27 | jsmith | UnixDog: That shouldn't matter |
| 17:31.38 | UnixDog | it seems to |
| 17:31.45 | jsmith | UnixDog: SIP authentication happens *before* the call ever gets to the dialplan |
| 17:32.17 | UnixDog | thats the local auth |
| 17:32.17 | jsmith | UnixDog: That's one of the fundamentals of Asterisk... calls always come through a channel driver (chan_sip in this case) before hitting the dialplan |
| 17:32.24 | UnixDog | from the phone to box a |
| 17:32.32 | jsmith | UnixDog: No |
| 17:32.42 | UnixDog | yes |
| 17:32.45 | UnixDog | look at it |
| 17:32.50 | jsmith | UnixDog: I'm talking about Box2 requiring Box1 to authenticate |
| 17:33.02 | UnixDog | when it ask the polycom to auth thats a locl auth |
| 17:33.08 | UnixDog | ok |
| 17:33.16 | jsmith | Yes, but I'm not talking about that |
| 17:33.19 | UnixDog | ok |
| 17:33.24 | Qwell | line 192 is what he's referring to |
| 17:33.26 | jsmith | I'm talking about lines 155, 192, and 206 |
| 17:33.53 | jsmith | In this case, Box1 sends a INVITE to Box2, and get's a 487 back, and then ACKs it |
| 17:34.13 | C4colo | I've had that same issue with extension 100 being on two boxes |
| 17:34.26 | jsmith | In other words, Box1 has no idea how to authenticate to Box2 |
| 17:34.42 | UnixDog | hey c4colo can you give me access to your 2 boxes to get them output |
| 17:34.48 | UnixDog | correct |
| 17:34.56 | C4colo | box one could not call box two because it was claiming to be 100 and there was a 100 in the default context |
| 17:34.58 | UnixDog | but it should not have to auth |
| 17:35.06 | jsmith | UnixDog: But that's not a bug, that's a configuration issue |
| 17:35.11 | UnixDog | how |
| 17:35.14 | C4colo | one is Ryppn's and the other was a temporary xen box I threw together to test SLA |
| 17:35.39 | UnixDog | jsmith is there a way to call you and go over this |
| 17:35.44 | jsmith | UnixDog: Box2 is requiring authentication from Box1, but Box1 doesn't have a username/password to use to authenticate |
| 17:35.51 | UnixDog | I dont think your understanding the full point |
| 17:35.57 | jsmith | UnixDog: Sure... I'll pm you with my phone number |
| 17:36.02 | UnixDog | ok |
| 17:36.03 | jsmith | UnixDog: I'd say the same about you ;-) |
| 17:36.09 | C4colo | mostly it is caused by the sip users being in the same context that anon sip connections enter |
| 17:36.20 | Qwell | jsmith: grandcentral? :p |
| 17:36.28 | symlink | C4colo: not... exactly |
| 17:36.31 | jsmith | Qwell: No, I'll give him my real number |
| 17:36.48 | Qwell | actually, random and off-topic - is grandcentral like...free? |
| 17:36.51 | C4colo | well, that's what caused it on mine, because it was a test box I put everything in [default] |
| 17:37.25 | C4colo | sip calls came in there and all users were there so it matched user 100 and asked for auth |
| 17:38.04 | C4colo | also I think setting a domain on each box would help, if both are using the astersk default it will claim to be in the same domain |
| 17:38.27 | C4colo | 100@asterisk is what I think it was using |
| 17:39.33 | C4colo | and both boxes used asterisk as the domain, so it thought they were on the same "network" |
| 17:40.21 | C4colo | I never really worked too hard to clear it up as the box was littterally a throw-away install |
| 17:40.23 | symlink | C4colo: you are fundamentally correct. |
| 17:41.10 | C4colo | fundamentally? am I missing something? |
| 17:41.40 | symlink | well, your usage of terms is a little off... and understanding of sip.conf, so some of the stuff you said made it seem as though you were talking about other things... |
| 17:42.18 | symlink | but the reason you have in mind is correct |
| 17:43.33 | C4colo | hmm, reading back what I said I don't see anything that seems to be incorrect terminology, help me understand what I said incorrectly if you don't mind |
| 17:43.46 | C4colo | I'm a stickler for correct and unambiguous communication |
| 17:43.55 | symlink | you create separate entries in sip.conf, they don't all exist in [default] |
| 17:44.08 | C4colo | context=default |
| 17:44.25 | C4colo | I put that on each of the users |
| 17:44.30 | symlink | really? I thought you put in general instead... so maybe you are off :) |
| 17:46.59 | C4colo | [joe01] : username=joe01 : authname=joe01; callerid=132142134 : secret=hackmepls : context=default : etc |
| 17:47.40 | symlink | it's rather simple really... chan_sip finds users by looking at the user portion of the From header - which is also used as callerid sometimes... so a call comes in and is matched by that as 6000, it then enters the dialplan unaltered... another system is then dialed... 6000 is placed in the From header as callerid... if that machines has a user entry itself for 6000 it will match and request authentication |
| 17:48.46 | symlink | so it's best to use RPID to transport callerid/presentation information and either actually do username/pass auth or set it to something bogus that'll probably never match |
| 17:56.38 | C4colo | so to be clear, how would one say that the entire dialplan and all users were in the default context? |
| 17:57.03 | C4colo | I put the dialplan under [default] and all users were defined with context=default ? |
| 17:57.11 | symlink | that is fine when you say dialplan :) |
| 17:57.24 | symlink | I was thinking you were talking about sip.conf |
| 17:59.22 | C4colo | well, technicall I was, I set all of the users to use the default context, the way I think about it is that everything is running under default |
| 18:06.09 | mvanbaak | ls -lah symlink |
| 18:29.09 | UnixDog | jsmith: hang up |
| 18:29.17 | UnixDog | I kicked the phone |
| 18:29.22 | UnixDog | and got disconnected |
| 18:29.30 | jsmith | UnixDog: No worries |
| 18:33.28 | Qwell | UnixDog: Why'd you kick the poor phone? No need to be so violent. :) |
| 18:44.18 | UnixDog | ok done |
| 18:44.30 | UnixDog | and now to tear asterisk apart |
| 18:44.32 | UnixDog | lol |
| 18:45.12 | symlink | raises eyebrow |
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| 19:06.23 | UnixDog | ok I have fixed a macro for asterisk that works with sip uri dialing that seems to work right now thanks to jsmith |
| 19:06.32 | UnixDog | it is just a work around |
| 19:07.20 | UnixDog | bbiab and I will post it |
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| 20:08.38 | russellb | wonders why a macro was needed |
| 20:08.45 | russellb | it's just a normal sip.conf entry ... |
| 20:16.06 | seanbright | russellb: http://pastebin.ca/1032255 |
| 20:16.14 | seanbright | russellb: that was a few minutes of spit-balling |
| 20:16.33 | seanbright | err |
| 20:16.40 | seanbright | not sure why i sent that in here, but whatever. |
| 20:17.07 | russellb | heh, no problem |
| 20:17.16 | russellb | but yeah, we're on the same page |
| 20:17.23 | russellb | i'm not really picky about the format |
| 20:17.58 | russellb | just that it is expressive enough for all of the information that we want to be able to get (yet to be determined and put into a spec), and in a nice format that various tools can be written easily against |
| 20:18.00 | seanbright | will write a script o automatically update voip-info :) |
| 20:18.03 | russellb | heh |
| 20:18.23 | seanbright | russellb: well we can transform xml into anything |
| 20:18.31 | russellb | exactly |
| 20:18.43 | russellb | we could even write res_documentation that serves up documentation as HTTP via the builtin http server |
| 20:18.44 | russellb | heh |
| 20:18.44 | seanbright | russellb: we can even transform it into... xml! |
| 20:18.52 | seanbright | russellb: ohhhh |
| 20:19.07 | russellb | actually ... asterisk 1.6 will by default install the html documentation generated from the tex docs |
| 20:19.16 | russellb | but it doesn't cover apps/functions/etc |
| 20:19.19 | seanbright | right |
| 20:19.21 | russellb | because there is no way to get the info |
| 20:19.33 | russellb | yet! |
| 20:19.36 | seanbright | -yet- |
| 20:19.37 | seanbright | :) |
| 20:19.40 | russellb | w00t |
| 20:19.42 | russellb | i'm excited. |
| 20:19.49 | russellb | go go go go |
| 20:19.52 | seanbright | heh |
| 20:20.04 | seanbright | <arg name="rule" since="1.6.0"> |
| 20:20.13 | seanbright | *that* is what excites me |
| 20:20.16 | seanbright | i need to get out more |
| 20:20.17 | russellb | _yes_ |
| 20:20.21 | russellb | lol |
| 20:20.48 | russellb | so when are you moving to AL again? |
| 20:20.54 | russellb | :-p |
| 20:20.54 | putnopvut | lol |
| 20:21.42 | russellb | kicks chan_iax2 |
| 20:22.11 | seanbright | heh |
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