IRC log for #asterisk-doc on 20070411

03:30.13*** join/#asterisk-doc anders__ (n=anders@201.158.188.233)
12:58.10*** join/#asterisk-doc jsmith (n=jsmith@h46054227.area3.spcsdns.net)
12:58.10*** mode/#asterisk-doc [+o jsmith] by ChanServ
13:22.30*** join/#asterisk-doc blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:22.30*** mode/#asterisk-doc [+o blitzrage] by ChanServ
13:35.53jsmithMorning blitzrage
13:36.30blitzragejsmith: morning sir!!
13:36.49blitzrageI told you I got reinvites working with directrtpsetup=yes right?
13:37.01jsmithYes.
13:37.34blitzrageI have another thing for you :)
13:40.07blitzrageI'm trying to get presence to work...
13:40.16blitzrageso I've added a hit to the [default] context like this:
13:40.40blitzrageexten => cturner0000#mixouttst,hint,SIP/cturner0000#mixouttst@switch1-mia.mixnetworks.net
13:40.43blitzrage<PROTECTED>
13:40.49blitzragefunny enough -- this makes MWI work
13:40.49blitzrage<PROTECTED>
13:40.56blitzragehere is the SUBSCRIBE I get from the phone
13:40.56blitzrage<PROTECTED>
13:40.59blitzrage<--- SIP read from 64.251.19.190:5060 --->
13:40.59blitzrageSUBSCRIBE sip:swafford00#mixouttst@switch1-mia.mixnetworks.net:5060 SIP/2.0
13:40.59blitzrageRecord-Route: <sip:64.251.19.190;lr;ftag=B96A4AD4-692CF7C3>
13:40.59blitzrageVia: SIP/2.0/UDP 64.251.19.190;branch=z9hG4bK2e26.5694934.0
13:40.59blitzrageVia: SIP/2.0/UDP 192.168.1.249;rport=1034;received=74.100.134.250;branch=z9hG4bKf0211c9A8EA9EB2
13:41.01blitzrageFrom: "lmadsen0000#mixouttst" <sip:lmadsen0000#mixouttst@switch1-mia.mixnetworks.net>;tag=B96A4AD4-692CF7C3
13:41.04blitzrageTo: <sip:swafford00#mixouttst@switch1-mia.mixnetworks.net>
13:41.08blitzrageCSeq: 2 SUBSCRIBE
13:41.10blitzrageCall-ID: b3882e08-f29e8bc6-419ea64d@192.168.1.249
13:41.12blitzrageContact: <sip:lmadsen0000#mixouttst@74.100.134.250:1034>
13:41.14blitzrageAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
13:41.16blitzrageEvent: presence
13:41.18blitzrageUser-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
13:41.20blitzrageMax-Forwards: 69
13:41.22blitzrageExpires: 3600
13:41.24blitzrageContent-Length: 0
13:41.26blitzrage<PROTECTED>
13:41.28blitzrage<PROTECTED>
13:41.30blitzrageand the response from Asterisk
13:41.32blitzrage<PROTECTED>
13:41.47blitzrage--- (15 headers 0 lines) ---
13:41.47blitzrageCreating new subscription
13:41.47blitzrageSending to 64.251.19.190 : 5060 (no NAT)
13:41.48blitzrageFound no matching peer or user for '64.251.19.190:5060'
13:41.50blitzrageLooking for swafford00#mixouttst in default (domain switch1-mia.mixnetworks.net)
13:41.52blitzrage<--- Transmitting (no NAT) to 64.251.19.190:5060 --->
13:41.54blitzrageSIP/2.0 404 Not Found
13:41.56blitzrageVia: SIP/2.0/UDP 64.251.19.190;branch=z9hG4bK03d1.11d69525.0;received=64.251.19.190
13:41.58blitzrageVia: SIP/2.0/UDP 192.168.1.249;rport=1034;received=74.100.134.250;branch=z9hG4bKe077a4f064F11D5
13:42.00blitzrageFrom: "lmadsen0000#mixouttst" <sip:lmadsen0000#mixouttst@switch1-mia.mixnetworks.net>;tag=30F946AF-69448342
13:42.03blitzrageTo: <sip:swafford00#mixouttst@switch1-mia.mixnetworks.net>;tag=as3ab81a87
13:42.05blitzrageCall-ID: a933c593-498b5f19-34de81e4@192.168.1.249
13:42.09blitzrageCSeq: 2 SUBSCRIBE
13:42.11blitzrageUser-Agent: MixNetworks SWITCH-MIA08
13:42.13blitzrageAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
13:42.15blitzrageSupported: replaces
13:42.17blitzrageContent-Length: 0
13:42.19blitzrage<PROTECTED>
13:42.31blitzrage<PROTECTED>
13:42.31blitzrageof course that is searching for sswafford00#mixouttst, and there would be a corresponding hint in the same format as above
13:42.46blitzrage<PROTECTED>
13:42.53blitzrageany pointers?
13:45.54fileyarrr default context
13:46.49blitzrageyah I know -- its bad... but subscribecontext doesn't work anymore!
13:47.28blitzragebut the hints reside in the [default] context
13:47.43fileshow dialplan agrees with you?
13:48.06jsmithblitzrage: Uh, just one minute buddy...
13:48.15blitzragejsmith: np!
13:48.23jsmithblitzrage: Shouldn't the hint point at a *local* SIP device, such as SIP/blah
13:48.26blitzragefile: yep -- just verified
13:48.45jsmithblitzrage: I'm not sure you can point hints at external resources, such as SIP/cturner0000#mixouttst@switch1-mia.mixnetworks.net
13:48.53blitzragejsmith: yah -- I tried all sorts of formats, and the one I used at least made MWI work
13:49.04fileyou *sort of* can
13:49.12jsmithThat shouldn't have anything to do with making MWI work
13:49.20blitzrageshouldn't -- but it did
13:49.22fileunless chan_sip overrides it...
13:49.45blitzrageFYI this is a setup where the end points are registered with SER, and Asterisk is using realtime
13:49.53blitzrage(not in the dialplan though)
13:50.15filewhy do I suspect this is not going to work as you want?
13:50.34blitzragebecause everything I do with this setup doesn't work the way I want
13:51.00blitzragekinda makes sense since they aren't registered with asterisk
13:51.32fileit should still find the hint
13:51.38blitzragekinda what I figured
13:51.49blitzrageand the fact that MWI works now..... is kinda strange
13:52.00fileyeah MWI has nothing to do with hints ...
13:52.16blitzragewhy did it start working as soon as I added these hints though?
13:52.42blitzrageStatus: Idle for all hints FYI
13:54.14filewith your setup, I'm not even going to think about it
14:01.30blitzrageahhhh, ok, at least the MWI I figured out what I changed to make it work
14:11.01blitzragejsmith: when you get a chance, can you make some calls into your test PBX/trunk so I can look at it's CDRs?
14:11.19jsmithSure... gimme a few minutes
14:14.26blitzragenp! no rush
14:15.12jsmithare you ready?
14:15.34blitzrage<PROTECTED>
14:16.53jsmithOK, there was one call.
14:17.09blitzragethx!
14:17.13jsmithI'll let you look at that before I make more calls
14:17.41jsmith(calling in doesn't do much right now, as they've gone and moved their WSDL file and not told me where it is, so all the SOAP stuff is b0rk3d
14:17.50blitzrageoh fun
14:18.04jsmithNo
14:18.20jsmithBut, I've got half the day off today, and all day tomorrow, so I really don't have to worry about it until Friday
14:18.49blitzragegreat!
14:19.16jsmithOf course, I'm driving ~5 hours today and ~5 hours tomorrow... but hey...
14:19.28blitzrageoh of course as soon as I turn SIP DEBUG on the call works!
14:19.38jsmithOh, and did I mention that the concrete crew messed up the floor in my basement?
14:19.41blitzragethe previous 2 calls from my desk phone to your DID number gave me busy
14:19.45jsmithOh?
14:20.27blitzrageyah....
14:20.36blitzragethat time the call came from miami9 and it worked....
14:20.47blitzragelooks to be the same Dial() line though
14:20.47jsmithOh, wait.
14:20.52jsmithI've only got one peer defined
14:20.53blitzrageoh?
14:20.56filewhat did they mess up?
14:21.01jsmithYou said the calls would always come from that IP
14:21.10blitzragebut it should be coming from switch1-mia.mixnetworks.net (64.251.19.190)
14:21.30blitzrageyah -- it should
14:21.30blitzragewhich is why I'm confused :)
14:21.30blitzragesince it goes via SER
14:21.38jsmith[Apr 11 20:30:32] WARNING[12805]: chan_sip.c:2739 create_addr: No such host: switch1-mia.mixnetworks.net
14:22.07blitzragehrmmm... did it again
14:22.28blitzragethat's what you're getting on the rejection eh?
14:23.05jsmithTry it one more time, please
14:23.14blitzragesure thing
14:23.38blitzragegot the IVR
14:24.00jsmithStrange... DNS was all gummed up in Asterisk, so I had to restart Asterisk
14:24.07blitzrageahhh
14:24.08blitzragegotcha
14:24.11blitzrageok -- seems to be ok now
14:55.06blitzragejsmith: ping?
14:55.34jsmithblitzrage: Yes?
14:55.35blitzrageis there a way I can change something in the XML of the book so that a section is still in the code, but "marked" as not being built for inclusion in the PDF?
14:55.56jsmithYou can put it in a comment, or there's another way... hold on.
14:56.29jsmithGimme a second.
14:56.36blitzragesure thing
14:56.53jsmithhttp://www.sagehill.net/docbookxsl/Profiling.html
14:56.56brac[Chapter&nbsp;25.&nbsp;Profiling (conditional text)]
14:56.58jsmithblitzrage: Knock yourself out
14:57.07blitzragethx!
14:57.15jsmithblitzrage: Now, depending on the gyrations that O'Reilly goes through, that *should* work, but no guarantees
14:57.40blitzragehrmmmmm ok
15:04.15fileblitzrage: damn, your CLI command audit... I'm sorry...
15:07.57filejsmith: can I bounce a thought off you?
15:08.06jsmithfile: Any time!
15:08.19jsmithfile: Especially if it involves group counts and transfers
15:08.24filejsmith: it does
15:08.37jsmithw00t!
15:08.41fileI have two current thoughts in my head...
15:08.47jsmithSpit them out!
15:09.24fileactually even then...
15:12.05filewhenever I go through it in my head, I always come back to unsetting OUTBOUND_GROUP in Dial.. and forcing the user to use more logic in the dialplan to remember that, and set it again if need be
15:12.36jsmithfile: Right... but the dialplan needs to know the difference between a "new" dial and a transfer "dial"
15:13.31filesee, you can't know that in all instances...
15:14.04jsmithWhich is my big hang-up (no pun intended)
15:14.18jsmithI can *already* do everything I need to in the dialplan, except for the case of transfers
15:14.20fileso why do YOU need to know it?
15:15.24jsmithBecause I want to unset the OUTBOUND_GROUP variable if it's a transfer
15:15.46fileokay, what if Dial unset it upon use?
15:15.57*** join/#asterisk-doc ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)
15:16.09jsmithfile: That wouldn't solve the problem
15:16.17jsmithHere's how it currently works.
15:16.31ctooleyAnyone know what the max length of a SIP destination is?
15:16.42jsmithWhen app_queue distributes calls to agents, it's distributing calls to Local/4001@agents/n
15:17.07blitzragectooley: 255?
15:17.10ctooleyIE 1NXXNXXNXXX@253.212.141.235 is 27, is the max 255?
15:17.14ctooleyblitzrage, cool
15:17.20jsmithThat extension sets the OUTBOUND_GROUP channel variable, so that if the agent answers, he won't continue to get beeped for additional calls
15:17.21blitzragectooley: that's my guess
15:17.37jsmithThen it uses Dial() to dial the phone where the agent is logged in.
15:18.04jsmithThe problem is that if the agent transfers the call, they still won't get any new calls until the transferred call was hung up.
15:18.21filewhat kind of transfer?
15:18.37jsmithI'm not sure if they're blind or attended
15:18.55jsmithAnd in this case, it's a SIP transfer, not a features transfer, I think.
15:19.38fileyou're having the same issue as angler I do believe
15:19.42fileI bet it's a blind transfer
15:19.48jsmithfile: Exactly the same issue!
15:19.52fileSIP client says to the Local channel... hey go here
15:20.10fileit does another Dial, OUTBOUND_GROUP is still set so it gets set on the channel again... and the agent can't receive calls
15:20.17jsmithExactly
15:20.29fileso if OUTBOUND_GROUP was unset in the first Dial, it wouldn't be set again...
15:20.42jsmithBut *when* would it be unset?
15:21.07jsmithYou can't unset it right after the call is answered, or the agent will continue to get rung for additional calls while he's still on the first call
15:21.19fileno.
15:21.25jsmithYou'd have to unset it at the time the agent's leg of the call dropped off
15:21.39filelet me explain how it works
15:21.43jsmithOK, please do
15:21.59fileOUTBOUND_GROUP is a regular variable that app_dial checks to see if exists, if it does... it grabs the variable and passes that off to the group counting API
15:22.11fileonce that was done I would unset OUTBOUND_GROUP
15:22.26jsmithOh, I see...
15:22.43filejust because OUTBOUND_GROUP isn't set doesn't mean the channel isn't part of the group
15:22.47jsmithSo the count would still be set, but it wouldn't *continue* to be set on spawned calls... I like that
15:22.56filecorrect
15:22.58jsmithThat works for me
15:23.17filethe only scenario that is questionable is when people do Dial after Dial and want it to continue to be in the same OUTBOUND_GROUP
15:23.28fileyou would have to set that group information in a separate variable, and know when to use it
15:23.40jsmithThe other way would be to have a channel variable set on *all* transferred calls, so the dialplan can unset OUTBOUND_GROUP as appropriate
15:23.54jsmithfile: What, you suggesting we add another option to Dial()?
15:24.05fileI didn't say 'dat
15:24.07jsmithfile: Or maybe set OUTBOUND_GROUP_ONCE
15:24.11jsmithif you only want it once
15:24.18fileooh...
15:24.19jsmithor OUTBOUND_GROUP, if you want it inherited
15:24.20filethat's not a bad idea
15:24.24jsmithI like it.
15:24.33jsmithIt's simple to code.
15:24.44jsmithDoesn't add another option
15:24.55jsmithAnd it won't break existing dialplans
15:24.58filequite
15:25.11jsmithAnd as long as it's never documented, only power users will even know about it ;-)
15:25.58blitzrageit should be documented so power users can learn of it :)
15:26.52jsmithblitzrage: Of course it should... I was being sarcastic, since this is #asterisk-doc
15:27.04blitzrage:)
15:27.26jsmithGood...
15:30.19filejsmith: I do have a fix for you while I bounce this off of others
15:31.11fileSet(OUTBOUND_GROUP=) before Dials that should not have OUTBOUND_GROUP :D
15:34.27blitzragejsmith: do you have a good, complete 79[46]0 tftp configuration set? I have a really simple one... but it still seems to require changes to the phone manually....
15:34.38blitzragewhich is fine for me... but a pain to document how to set one of these phones up in the book :)
15:35.25jsmithblitzrage: Yes, I do.
15:35.31blitzragejsmith: w00t!
15:36.00jsmithfile: That's the problem... I can't tell from the dialplan whether it's a new call or a transfer, so I don't know whether or not to unset it!
15:36.36fileyou're not using the same extension for both regular dialing and agent dialing... are you?
15:36.54jsmithI'm not sure I follow
15:37.32filenevermind...
15:40.10blitzragejsmith: mind emailing that to me?
15:40.29filejsmith: I have a patch if you want to try it for me...
15:40.56jsmithI will when I get back on Friday
15:51.15*** topic/#asterisk-doc by blitzrage -> Welcome to the Asterisk Documentation Project / Idle Chat! --=-- Will work for money --==-- ISBN13: 9780596514051 --==--
15:54.49jsmithblitzrage: I'll try to send it to you tonight
15:54.52jsmithblitzrage: I've gotta run
15:54.56blitzragelates
16:34.53*** join/#asterisk-doc Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
16:35.17ZeeekI can never remember id it's docs, -docs, -doc or doc
16:36.01Zeeekwe need someone from here for the next conference on Friday about features.conf. Any volunteers?
16:36.05Zeeekhttp://x2z.eu/
16:36.07brac[About the Asterisk Users Conference/Podcast]
16:36.23Zeeekbots can eventually get on my nerves, but hey
16:36.46blitzrageheh
16:36.56ZeeekWanna hear my best debug story?
16:36.58blitzragebrac isn't really favoured by many people in here...
16:37.00blitzragesure!
16:37.00Zeeekhetre goes
16:37.06blitzragefeatures.conf eh?
16:37.29blitzragenot too much to really talk about there :)
16:37.37ZeeekSo I was having one way audio and other NAT -related probs a couple weeks ago having changed the router and updated asterisk at the same time
16:37.53ZeeekI'm going nuts, trying to find the problem
16:37.59blitzragenever change 2 things at once :)
16:38.02Zeeekeventually
16:38.12Zeeekyes, very bad to change 2 things at once
16:38.31filethe underlying code for features.conf is evil
16:38.32filethat is all.
16:38.37Zeeekeventually, after changing about 1,000 settings on the phone which was on the same side of NAT as asterisk, I'm looking at sip.conf
16:38.53Zeeekfile maybe that's why no one talks about it :)
16:39.09ZeeekBut I digress from my hilarous debug story
16:39.26Zeeekso I'm toggling every setting in sip.conf - depperate
16:39.35Zeeekdesperate beyond any logic
16:40.05Zeeekand then I notice a line: localnet=192.168.0. something
16:40.32Zeeeklong story short, that was wrong and had been for three years, but the phone worked
16:40.50Zeeekso there must have been a "bug" in asterisk's interpretation of the setting that is now fixed :)
16:41.14ZeeekI changed the localnet to the right address and the phone worked again
16:41.23ZeeekI thought it was amusing
16:41.27Zeeekoh well...
16:42.37Zeeekany suggestions for additional topics of discussion on a conf are welcome
16:42.41Zeeekwe have two days to attract a few people. Otherwise I'll be there babbling as usual
16:48.08Zeeekwho's willing to show up for 15 minutes and talk about anything? Come on guys
16:48.26Zeeekhows's the book coming?
16:50.26blitzrageZeeek: Jim and I are working on it right now actually :_)
16:50.30blitzragehence my distraction
16:50.42ZeeekI'm sure you are and I'm sure you are
16:51.23ZeeekI actually found an artificial use for applicationmap
16:51.44Zeeekbut I think it warrants some discussion - not my art(icifce, the feature itself
17:24.47*** part/#asterisk-doc Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
18:29.18*** join/#asterisk-doc Alric (n=nbowyer@avantacom.com)
19:52.14blitzrageand you just don't get it, ya keep it copacetic, and you learn to accept it, ya know, it's so pathetic!
19:53.00fileyes you are
19:53.05blitzrageoh burn
20:32.58ctooleygeeks-that-sing, pathetic, synonyms
21:23.59Alric^^
22:15.42blitzrageanyone here good with iptables?

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