irclog2html for #asterisk on 20061021

`00:00.05`	`shy_guy`	`question is does type=user, accept incoming calls?`
`00:00.23`	`shy_guy`	`host= <-- this is must set,offcourse.`
`00:00.30`	`shy_guy`	`it doesnt`
`00:00.33`	`ManxPower`	`shy_guy: no, host= is only for OUTOGING calls.`
`00:00.44`	`shy_guy`	`:) no`
`00:00.52`	`ManxPower`	`shy_guy: type=user means calls will go from device -> asterisk`
`00:01.04`	`shy_guy`	`host=dynamic if you want your outgoing peer to register with you in order for you to call him`
`00:01.21`	`ManxPower`	`shy_guy: or host=ipaddress if the device has a static ip address.`
`00:01.32`	`shy_guy`	`yup`
`00:01.40`	`shy_guy`	`only incoming calls`
`00:01.45`	`ManxPower`	`if you want to LIMIT where calls can come fro then you want permit/deny`
`00:02.02`	`ManxPower`	`host= is not checked for incoming calls.`
`00:02.06`	`benjk`	`yeah ,friends are ok for phones`
`00:02.42`	`ManxPower`	`shy_guy: what are you trying to ACCOMPLISH?`
`00:02.54`	`shy_guy`	`errata: read outgoing as incoming in above where appropriate => type=user is for incoming calls and type=peer for outgoing calls,host may be set as necessary`
`00:03.18`	`ManxPower`	`shy_guy: where is that from?`
`00:03.32`	`shy_guy`	`but what is this insecure=invite ? :P for authenticating calls coming from peer`
`00:03.33`	`ManxPower`	`for all these terms it is from the point of view of Asterisk`
`00:04.32`	`ManxPower`	`shy_guy: I don't know. I've never needed it.`
`00:04.50`	`shy_guy`	`why have they written in the book that insecure=invite is used for dealing with calls coming in from the peer whie we know peer is for only making outgoing calls to (as told in its chp4)`
`00:05.21`	`ManxPower`	`I don't know.`
`00:05.40`	`shy_guy`	`Probably no one does`
`00:05.43`	`shy_guy`	`:)`
`00:05.45`	`shy_guy`	`cheers!`
`00:05.48`	`ManxPower`	`other than to remember that as of 1.2 there is not much difference between user/peer/friend`
`00:06.06`	`shy_guy`	`it surely seems that way`
`00:06.25`	`shy_guy`	`if something registers with other proxy, it can even receive calls from it`
`00:06.43`	`shy_guy`	`all it needs is an account with type=user over there`
`00:06.44`	`De_Mon`	`when I dial #1 for a blind xfer I have like 500ms to dial the extension I want to forward to`
`00:07.06`	`shy_guy`	`has anyone tired sofia-sip with asterisk?`
`00:07.16`	`shy_guy`	`i heard there are few seconds lags`
`00:08.28`	`ManxPower`	`[root@pbx-1 root]# grep timeout /home/software/asterisk/asterisk-1.2/configs/features.conf.sample`
`00:08.28`	`ManxPower`	`;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call`
`00:08.29`	`ManxPower`	`;featuredigittimeout = 500 ; Max time (ms) between digits for`
`00:08.49`	`[TK]D-Fender`	`De_Mon : Why on earth are you using DMF transfers anyways?`
`00:08.49`	`val2`	`chown`
`00:09.21`	`ManxPower`	`[TK]D-Fender: I've stopped even TRYING to understand why people use that totally ugly hack.`
`00:09.45`	`De_Mon`	`[TK]D-Fender call is going over PSTN to a call router (outsourced) who needs to transfer the call to the correct dept`
`00:10.29`	`De_Mon`	`those are the default settings right? I definately don't get 3 seconds.. I suppose I could try increasing the transferdigittimout`
`00:10.34`	`[TK]D-Fender`	`De_Mon : ok, that works...`
`00:11.19`	`[TK]D-Fender`	`ManxPower : That answer was about the only one I could really accept :)`
`00:12.22`	`*** join/#asterisk nitrico (n=aaaa@200.81.9.176)`
`00:13.41`	`nitrico`	`hi everyone, just a question...maybe someone can help me`
`00:13.49`	`*** join/#asterisk junodixon (n=Junodixo@216-188-237-188.dyn.grandenetworks.net)`
`00:14.13`	`shy_guy`	`ask nitrico, there aint a need for theme setting`
`00:14.13`	`junodixon`	`can yall help me with a caller id problem or am i in the wrong place`
`00:14.15`	`nitrico`	`I nee to run an agi when the calls are answered, Ive tried using the G option for Dial and the context is taken, the agi is called......but the call is hangedup after that....any solution?`
`00:14.39`	`shy_guy`	`junodixon: yankees just type, others ...:p`
`00:15.03`	`junodixon`	`lol`
`00:15.34`	`shy_guy`	`nitrico:call a macro and call the agi in that macro.`
`00:15.50`	`shy_guy`	`nitram: call that macro from the Dial not individually.`
`00:16.19`	`junodixon`	`but anyway i can change the # on my caller id but i cant change the name whatever i do it just says unknow name`
`00:16.42`	`shy_guy`	`junodixon:CALLERID(name) or...`
`00:17.02`	`shy_guy`	`junodixon:depends on your version. its simple`
`00:17.15`	`junodixon`	`im useing trixbox`
`00:17.33`	`junodixon`	`i think its 1.2.2`
`00:18.16`	`shy_guy`	`see whatever it supports CALLERIDNAME or SetCallerIDName( A Juno Yankee wasting time over a simple thingie)..or CALLERID(name)="Hehe thats me"`
`00:18.37`	`nitrico`	`shy_guy....this is the log....`
`00:18.41`	`nitrico`	`at OnAnswer,s,2 failed so falling back to exten 's'`
`00:18.49`	`nitrico`	`at OnAnswer,s,2 still failed so falling back to context 'default'`
`00:18.54`	`nitrico`	`and ten hangup...`
`00:18.59`	`nitrico`	`the agi is called....this is the good news :)`
`00:19.07`	`nitrico`	`ten should be then, excuse...`
`00:19.23`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`00:19.26`	`shy_guy`	`nitrico::)`
`00:20.09`	`shy_guy`	`nitrico:you should be able to solve it now`
`00:20.29`	`shy_guy`	`um feeling quite sleepy`
`00:22.10`	`nitrico`	`thanks shy_guy, I will try :) I cannot get them bridged....I will see`
`00:23.02`	`shy_guy`	`nitram:there is a very nice paragraph in AFOT-book, ctrl-f DIALSTATUS, see Dial, see its macro option.`
`00:23.08`	`shy_guy`	`agi debug is your friend`
`00:23.15`	`shy_guy`	`good night every one, happy *ing`
`00:26.40`	`*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`00:27.46`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:27.46`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:29.22`	`[TK]D-Fender`	`nitrico : "s" is an exten, not a priority`
`00:31.29`	`nitrico`	`yes, so?`
`00:32.04`	`nitrico`	`i send the call to context OnAnswer exten s, priority 1 right?`
`00:33.29`	`*** join/#asterisk damanivu (n=damanivu@ip68-4-207-173.oc.oc.cox.net)`
`00:33.32`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`00:33.49`	`De_Mon`	`damnit, I should be able to test the blindxfr feature from a SIP phone shouldn't I? (eyebeam)`
`00:34.10`	`[TK]D-Fender`	`nitrico : TGIF :) I'm clearly too tired to think.....`
`00:34.31`	`[TK]D-Fender`	`De_Mon : From any phone I would think.`
`00:34.44`	`[TK]D-Fender`	`De_Mon : pasebin up tht call attempt`
`00:34.51`	`Marshall16`	`where can i get a free voip IAX2 account at?`
`00:34.55`	`tlp`	`Is there any particular thing that might cause the 'Echo()' application to not function?`
`00:35.13`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`00:35.16`	`De_Mon`	`DTMF send in-band rpt, sent event RTP(RFC2833) codecID 101 DTMF tone length samples 120...`
`00:35.25`	`De_Mon`	`leme debug`
`00:36.50`	`[TK]D-Fender`	`Marshall16 : Set up an * server and connect to it. There you go :)`
`00:36.54`	`*** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153)`
`00:41.25`	`De_Mon`	`odd.. shouldn't I see any DTMF tones with sip debug peer?`
`00:41.26`	`De_Mon`	`http://pastebin.ca/212698`
`00:43.27`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`00:43.36`	`junodixon`	`i have tryed all of thows things but nun of them work`
`00:46.05`	`junodixon`	`anyone know of anything that will fix my caller id`
`00:47.48`	`De_Mon`	`junodixon after you set the callerID does asterisk present the correct one if you NoOp(${CALLERID}) ?`
`00:50.42`	`junodixon`	`maby im puting it in the wrong place where do i put it`
`00:53.11`	`De_Mon`	`junodixon does that mean asterisk doesn't return the correct info?`
`00:53.48`	`junodixon`	`yeah all i get is unknown name but im geting the right #`
`00:54.13`	`De_Mon`	`at least you know where the source of the problem is :)`
`00:54.26`	`junodixon`	`lol yeah`
`00:54.41`	`De_Mon`	`junodixon paste the 1 line you're using to set the name`
`00:55.08`	`junodixon`	`SetCallerID("Name" <Number>[\|a])`
`00:56.02`	`*** join/#asterisk mpls-eric (n=joshgold@user-12l39ej.cable.mindspring.com)`
`00:56.20`	`junodixon`	`im useing trixbox 1.2.2 and i tryed setting it with the website but it just does the same`
`00:56.53`	`*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)`
`00:56.58`	`De_Mon`	`the [\|a] means that \|a is optional, the []'s should never be there`
`00:57.07`	`De_Mon`	`not that I know what \|a even does...`
`00:58.03`	`junodixon`	`i put this in extensions_custom.conf right cuz thats where i have been puting it`
`00:58.16`	`De_Mon`	`shrug depends`
`00:58.28`	`De_Mon`	`youre using trixbox, which isn't exactly supported`
`00:59.52`	`junodixon`	`well is there a trixbox room`
`01:00.10`	`junodixon`	`brb`
`01:02.49`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`01:13.03`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`01:19.17`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`01:20.01`	`*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)`
`01:25.31`	`*** part/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net)`
`01:27.06`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`01:27.06`	`*** mode/#asterisk [+o mog] by ChanServ`
`01:34.32`	`*** join/#asterisk saftsack (n=saftsack@p54A7E8A1.dip.t-dialin.net)`
`01:35.37`	`*** join/#asterisk [shodan] (n=shodan@ip164.96-113-216.pppoe1.joliette.intermonde.net)`
`01:35.57`	`[shodan]`	`any one using a gsm gateway ? any good one to recommend ?`
`01:36.37`	`SuPrSluG`	`I have an DID 800 number. Caller never hears ringing after the select an extension. any ideas?`
`01:37.02`	`SuPrSluG`	`iax is the protocol`
`01:44.40`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`01:57.02`	`*** join/#asterisk AbsTradELic (n=vldmr@201.79.157.25)`
`01:57.43`	`AbsTradELic`	`hi all !`
`02:00.51`	`AbsTradELic`	`pls, exist any site about asterisk under slackware ?`
`02:01.13`	`*** join/#asterisk Parvaresh (i=bartali@213.207.218.66)`
`02:01.18`	`AbsTradELic`	`i'm a beguinner about voip`
`02:02.02`	`SuPrSluG`	`AbsTradELic:distro doesn't matter much.`
`02:02.28`	`[shodan]`	`anyone knows where to get some MV-370 gsm gateways on the cheap ? (ebay is 350$usd , should be around 150$usd)`
`02:02.31`	`AbsTradELic`	`SuPrSluG: huMruM`
`02:02.44`	`Parvaresh`	`which forum is most active on asterisk?`
`02:03.10`	`SuPrSluG`	`AbsTradELic:?`
`02:03.49`	`AbsTradELic`	`SuPrSluG: the new release packages to slackware ?`
`02:04.34`	`SuPrSluG`	`AbsTradELic:build from source.`
`02:05.04`	`SuPrSluG`	`AbsTradELic:just get the tarballs from digium`
`02:05.13`	`DrkShdw`	`AbsTradELic: asterisk is cross platform. grab the source and comile it on whatever OS you like. then follow any available guide. Flavor it to your distros taste`
`02:05.35`	`AbsTradELic`	`huM... ok`
`02:05.42`	`SuPrSluG`	`AbsTradELic:build zaptel and libpri 1st then asterisk.`
`02:05.43`	`AbsTradELic`	`from digium`
`02:05.59`	`AbsTradELic`	`ok`
`02:06.03`	`AbsTradELic`	`I'll do`
`02:06.57`	`SuPrSluG`	`AbsTradELic:here http://www.asterisk.org/download`
`02:08.29`	`AbsTradELic`	`ok`
`02:08.50`	`SuPrSluG`	`AbsTradELic: ~wiki`
`02:09.05`	`SuPrSluG`	`AbsTradELic:has some good tutorials`
`02:09.17`	`SuPrSluG`	`~wiki`
`02:09.36`	`Parvaresh`	`any forum !?`
`02:09.53`	`Parvaresh`	`specially looking for user's comments on hardwares...`
`02:09.56`	`*** join/#asterisk JunK-Y (n=junky@modemcable198.14-83-70.mc.videotron.ca)`
`02:11.17`	`AbsTradELic`	`the zaptel 1.2 or beta ?`
`02:12.25`	`DrkShdw`	`AbsTradELic: never use beta software, when you are first learning something. you want to make sure the problems you have are actually your problems, not some bug in the beta software suite.`
`02:12.44`	`AbsTradELic`	`ok`
`02:12.54`	`AbsTradELic`	`1.2`
`02:13.01`	`SuPrSluG`	`AbsTradELic:stick w/ the 1.2 stuff. betas not documented yet. and there some purdy big changes`
`02:14.42`	`AbsTradELic`	`1.2.10`
`02:15.08`	`AbsTradELic`	`.tar.gz`
`02:15.09`	`AbsTradELic`	`ok ?`
`02:15.27`	`SuPrSluG`	`just tar xvzf *.tar.gz`
`02:16.46`	`AbsTradELic`	`SuPrSluG: yeah`
`02:17.53`	`SuPrSluG`	`iax2 DID caller never hears ringing. any ideas? it does it out of band. is that a problem?`
`02:17.55`	`AbsTradELic`	`I need to know about sources versions to 2.6 slackware extra kernel ... X;)`
`02:18.36`	`SuPrSluG`	`AbsTradELic:make linux26`
`02:19.14`	`AbsTradELic`	`yeah, i'm using it`
`02:19.29`	`SuPrSluG`	`AbsTradELic:is it compiling?`
`02:19.52`	`AbsTradELic`	`not... default kernel`
`02:20.24`	`SuPrSluG`	`AbsTradELic:install headers and u should be fine`
`02:20.41`	`AbsTradELic`	`2.6.17.13`
`02:20.46`	`AbsTradELic`	`slackware 11`
`02:20.55`	`AbsTradELic`	`full install`
`02:21.21`	`SuPrSluG`	`AbsTradELic:do you have the source or kernel headers installed`
`02:22.04`	`AbsTradELic`	`yeah`
`02:22.17`	`AbsTradELic`	`libpri 1.2.4`
`02:22.22`	`AbsTradELic`	`ok ?`
`02:22.30`	`SuPrSluG`	`yeah`
`02:23.20`	`AbsTradELic`	`ok`
`02:23.37`	`AbsTradELic`	`i have 4 other files on asterisk.org/download`
`02:24.48`	`AbsTradELic`	`i'll get only the asterisk source package`
`02:25.24`	`SuPrSluG`	`get libpri zaptel and asterisk`
`02:25.34`	`AbsTradELic`	`asterisk 1.2.13`
`02:25.39`	`SuPrSluG`	`yeah`
`02:26.08`	`AbsTradELic`	`its the voip begin`
`02:26.22`	`SuPrSluG`	`you have to compile libpri and zaptel first`
`02:26.35`	`AbsTradELic`	`ok`
`02:26.41`	`AbsTradELic`	`I'll do it now`
`02:26.44`	`AbsTradELic`	`X;)`
`02:26.56`	`AbsTradELic`	`checkinstall -y -S`
`02:29.28`	`SuPrSluG`	`AbsTradELic:check here for a good install guide (and other good stuff) http://www.asteriskguru.com/tutorials/general_asterisk_installation_compilation.html`
`02:30.38`	`SuPrSluG`	`http://www.asteriskguru.com/tutorials/basic_installation_information_asterisk_from_source.html`
`02:31.34`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`02:31.37`	`AbsTradELic`	`ok`
`02:32.25`	`SuPrSluG`	`better yet they have a slackware specific install guide http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_slackware.html`
`02:38.01`	`AbsTradELic`	`SuPrSluG: zaptel didn't created slackware packages with checkinstall`
`02:39.09`	`AbsTradELic`	`then I used the usual to done: make install`
`02:39.19`	`SuPrSluG`	`build from scratch. did you check the slack guide`
`02:39.38`	`SuPrSluG`	`any errors?`
`02:40.04`	`AbsTradELic`	`nops`
`02:40.09`	`AbsTradELic`	`only a warning`
`02:40.17`	`AbsTradELic`	`at the end`
`02:40.38`	`AbsTradELic`	`ok ?`
`02:40.53`	`SuPrSluG`	`warnings are ok`
`02:41.04`	`SuPrSluG`	`errors bad`
`02:41.21`	`AbsTradELic`	`ok... now I go to asterisk`
`02:41.38`	`SuPrSluG`	`issue a ztcfg -vv command`
`02:42.22`	`SuPrSluG`	`wait 1st do a modprobe zaptel`
`02:43.19`	`AbsTradELic`	`huM`
`02:43.22`	`AbsTradELic`	`ok`
`02:44.10`	`AbsTradELic`	`Module Size Used by`
`02:44.11`	`AbsTradELic`	`zaptel 185220 0`
`02:44.11`	`AbsTradELic`	`crc_ccitt 1920 1 zaptel`
`02:44.23`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`02:44.36`	`AbsTradELic`	`ok modules loaded`
`02:45.17`	`AbsTradELic`	`now ill compile the asterisk`
`02:45.49`	`AbsTradELic`	`make mpg123`
`02:46.10`	`AbsTradELic`	`ok ?`
`02:47.54`	`AbsTradELic`	`aff`
`02:47.56`	`AbsTradELic`	`error`
`02:48.53`	`AbsTradELic`	`only the make now`
`02:49.00`	`AbsTradELic`	`???`
`02:49.16`	`AbsTradELic`	`why it ?`
`02:50.28`	`AbsTradELic`	`SuPrSluG: ?`
`02:50.38`	`SuPrSluG`	`yo`
`02:50.49`	`AbsTradELic`	`make mpg123 error`
`02:51.08`	`AbsTradELic`	`then only make`
`02:51.20`	`AbsTradELic`	`compilling fine`
`02:51.49`	`SuPrSluG`	`do you have mpg123 installed (not mpg321)`
`02:52.07`	`AbsTradELic`	`I think no`
`02:52.12`	`SuPrSluG`	`it's only used for music on hold`
`02:52.21`	`AbsTradELic`	`its not important`
`02:52.28`	`AbsTradELic`	`?`
`02:53.06`	`AbsTradELic`	`ok`
`02:53.29`	`SuPrSluG`	`no you can use wave files or convert to native gsm.`
`02:54.51`	`AbsTradELic`	`ok... asterisk done`
`02:54.55`	`AbsTradELic`	`X:)`
`02:55.12`	`AbsTradELic`	`to test it`
`02:55.15`	`AbsTradELic`	`now`
`02:55.18`	`AbsTradELic`	`X:)`
`02:55.36`	`AbsTradELic`	`it work with webcam ?`
`02:55.47`	`AbsTradELic`	`(smiles)`
`02:57.23`	`AbsTradELic`	`ok`
`03:00.40`	`SuPrSluG`	`i'm back`
`03:00.49`	`SuPrSluG`	`~jerjer`
`03:00.58`	`jbot`	`jerjer is probably the guy who runs nufone`
`03:03.06`	`justinu\|laptop`	`~nufone`
`03:03.07`	`jbot`	`i heard nufone is Visit http://www.nufone.net for an excellent, native IAX termination service.`
`03:03.24`	`[TK]D-Fender`	`~recursion`
`03:03.25`	`jbot`	`it has been said that recursion is tell ubotu about recursion`
`03:03.28`	`[TK]D-Fender`	`:O`
`03:03.49`	`justinu\|laptop`	`~ubotu`
`03:09.13`	`AbsTradELic`	`make sample ?`
`03:09.21`	`r0d3nt\|m`	`jbot, who's the buttslut on nufone.net`
`03:09.25`	`r0d3nt\|m`	`haahha`
`03:09.43`	`AbsTradELic`	`'make samples' ???`
`03:09.55`	`AbsTradELic`	`I need it?`
`03:12.33`	`AbsTradELic`	`SuPrSluG: ?`
`03:12.44`	`SuPrSluG`	`yes you're starting with nothing`
`03:13.30`	`AbsTradELic`	`after make install command ?`
`03:13.43`	`AbsTradELic`	`or before ?`
`03:13.52`	`SuPrSluG`	`after`
`03:14.37`	`AbsTradELic`	`ok... done. only it ?`
`03:16.01`	`AbsTradELic`	`can I run '/etc/rc.d/rc.asterisk start' now ?`
`03:16.27`	`AbsTradELic`	`X:)`
`03:16.51`	`AbsTradELic`	`the samples files go to where ?`
`03:17.22`	`SuPrSluG`	`they go in /etc/asterisk`
`03:18.04`	`SuPrSluG`	`to start asterisk run -> asterisk -vvvvg &`
`03:18.16`	`SuPrSluG`	`see if you get errors`
`03:18.52`	`AbsTradELic`	`asterisk ready`
`03:19.25`	`SuPrSluG`	`now issue asterisk -r and you'll be at the CLI`
`03:19.45`	`AbsTradELic`	`like root ?`
`03:20.04`	`SuPrSluG`	`then type help and it'll give you all availble commands`
`03:20.16`	`AbsTradELic`	`huM!`
`03:20.17`	`SuPrSluG`	`yeah`
`03:20.17`	`AbsTradELic`	`ok`
`03:20.28`	`AbsTradELic`	`not like client ?`
`03:20.40`	`AbsTradELic`	`not like user ?`
`03:20.48`	`SuPrSluG`	`you can run it non root`
`03:21.24`	`AbsTradELic`	`ok... understand now`
`03:21.31`	`SuPrSluG`	`there are tutorials for that too. for now root is ok`
`03:21.33`	`AbsTradELic`	`its a client`
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`03:22.51`	`SuPrSluG`	`yeah you can connect to the command line interface. that where all yer troubleshooting is done`
`03:24.02`	`AbsTradELic`	`huM!`
`03:24.05`	`AbsTradELic`	`ok`
`03:24.25`	`AbsTradELic`	`and a qt based gui ?`
`03:24.33`	`SuPrSluG`	`the main config files are extensions.conf ( your dialplan) and iax or sip.conf for connecting to phones and providers`
`03:25.09`	`SuPrSluG`	`vi isn't purdy enuf?`
`03:25.57`	`AbsTradELic`	`SuPrSluG: sorry, I have difficulties to understand now. english is not my native language`
`03:26.02`	`SuPrSluG`	`trixbox has that stuff. I don't like it`
`03:26.36`	`AbsTradELic`	`I love command lines`
`03:26.57`	`AbsTradELic`	`but I really need to begin with gui`
`03:27.17`	`SuPrSluG`	`I think asterisk is best that way`
`03:27.34`	`AbsTradELic`	`humrum!`
`03:27.55`	`AbsTradELic`	`exist a gui to asterisk ?`
`03:28.08`	`SuPrSluG`	`no the only files you need to edit to get started are the ones mentioned above`
`03:28.22`	`AbsTradELic`	`huM!`
`03:28.31`	`AbsTradELic`	`waitt a few`
`03:29.04`	`SuPrSluG`	`so go to /etc/asterisk`
`03:30.17`	`AbsTradELic`	`ok`
`03:30.18`	`SuPrSluG`	`do you have any phones yet?`
`03:30.28`	`AbsTradELic`	`yes`
`03:30.53`	`SuPrSluG`	`what is it hard phone or softphone?`
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`03:31.17`	`AbsTradELic`	`I have a conventional phone`
`03:31.24`	`AbsTradELic`	`hardphone`
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`03:31.46`	`AbsTradELic`	`I dont have voip accessories`
`03:31.49`	`SuPrSluG`	`do you have an FXS card?`
`03:32.05`	`AbsTradELic`	`nops`
`03:32.24`	`AbsTradELic`	`I dont have nothing... only the computer`
`03:32.32`	`SuPrSluG`	`if not download kphone or xlite softphone to play with`
`03:32.54`	`SuPrSluG`	`headset with ear and mic?`
`03:33.10`	`AbsTradELic`	`yeah`
`03:33.12`	`AbsTradELic`	`X:)`
`03:33.29`	`SuPrSluG`	`then you'll edit sip.conf and add and entry for that`
`03:33.48`	`AbsTradELic`	`ok... going to /etc/asterisk`
`03:34.08`	`AbsTradELic`	`jed sip.conf`
`03:34.26`	`AbsTradELic`	`ok`
`03:35.32`	`AbsTradELic`	`the entry ?`
`03:36.21`	`AbsTradELic`	`its a big file`
`03:38.46`	`SuPrSluG`	`that's because they put EVERYTHING in ti .`
`03:38.48`	`SuPrSluG`	`here's a simple entry for a phone`
`03:38.49`	`SuPrSluG`	`[311] ; phone number`
`03:38.51`	`SuPrSluG`	`type=friend ; This device takes and makes calls`
`03:38.53`	`SuPrSluG`	`username=311 ; Username on device`
`03:38.54`	`SuPrSluG`	`secret=xxxxx ; a password -> match this on your phone`
`03:38.56`	`SuPrSluG`	`context=default`
`03:38.57`	`SuPrSluG`	`host=dynamic ; This host is not on the same IP addr every time`
`03:38.59`	`SuPrSluG`	`mailbox=311 ; Activate the message waiting light if this`
`03:39.00`	`SuPrSluG`	`nat=yes ; voicemailbox has messages in it`
`03:39.02`	`SuPrSluG`	`canreinite=no`
`03:40.13`	`SuPrSluG`	`go to asteriskguru.org for info on configuring phones and extensions. these files can be real small too!!`
`03:41.37`	`AbsTradELic`	`ok`
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`03:41.57`	`ManxPower`	`~pastebin`
`03:41.59`	`jbot`	`well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`03:42.19`	`AbsTradELic`	`but I'll insert this ok`
`03:42.23`	`AbsTradELic`	`ok`
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`03:45.56`	`AbsTradELic`	`SuPrSluG: when I insert this entries, I rerun the asterisk -r ?`
`03:46.29`	`AbsTradELic`	`restart asterisk -r ?`
`03:46.56`	`SuPrSluG`	`yes. to check if it is seen at CLI>sip show peers and you should see the device`
`03:47.06`	`SuPrSluG`	`yes restart`
`03:47.15`	`SuPrSluG`	`first`
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`04:27.23`	`junodixon`	`can anyone help me with a caller id problem`
`04:28.48`	`junodixon`	`i can change the # but i cant change the name im useing trixbox 1.2.2`
`04:29.11`	`ManxPower`	`junodixon: where is the name not appearing correctly?`
`04:29.31`	`junodixon`	`on the caller id of the phone im calling`
`04:29.41`	`ManxPower`	`and that phone is what/where?`
`04:29.52`	`ManxPower`	`is it connected to the asterisk box`
`04:29.55`	`junodixon`	`it says unknown name`
`04:29.57`	`ManxPower`	`or another asterisk box?`
`04:30.01`	`junodixon`	`on its a pots line`
`04:30.13`	`junodixon`	`no`
`04:30.13`	`ManxPower`	`You cant set the name when talking to the PSTN`
`04:30.52`	`ManxPower`	`Well, you CAN, but the terminating telco will ignore the name and do a number/name lookup and put whatever that ends up being in the name field.`
`04:33.15`	`kilobit2001`	`how can i mount a remote linux system directory locally?`
`04:33.29`	`sung`	`yeha`
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`04:34.18`	`ManxPower`	`kilobit2001: Perhaps you are looking for #linuxhelp`
`04:34.22`	`CunningPike`	`kilobit2001: Locally to what?`
`04:42.04`	`AbsTradELic`	`ok`
`04:42.14`	`AbsTradELic`	`X-Lite or KPhone ?`
`04:42.24`	`AbsTradELic`	`the best ?`
`04:43.04`	`junodixon`	`<AbsTradELic> are you talking to me`
`04:43.19`	`AbsTradELic`	`can be`
`04:43.22`	`AbsTradELic`	`X:)`
`04:43.26`	`junodixon`	`lol`
`04:44.10`	`teknoprep`	`hmmm... why not IDEFISK ?`
`04:44.15`	`teknoprep`	`IDEFISK is the best`
`04:44.39`	`AbsTradELic`	`IDEFISK ?`
`04:44.54`	`teknoprep`	`also the best is an opinion... in my opinion IDEFISK is the best softphone`
`04:44.55`	`teknoprep`	`and bro`
`04:45.04`	`teknoprep`	`ifyou don't know wtf i am talking about... www.google.com`
`04:45.37`	`teknoprep`	`http://www.google.com/search?q=idefisk&ie=utf-8&oe=utf-8&rls=org.mozilla:en-US:official&client=firefox-a`
`04:45.59`	`AbsTradELic`	`ok`
`04:46.08`	`AbsTradELic`	`it work with webcam ?`
`04:47.34`	`AbsTradELic`	`teknoprep:`
`04:48.11`	`AbsTradELic`	`X:P`
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`04:54.30`	`AbsTradELic`	`exist any free voip service ?`
`04:54.41`	`AbsTradELic`	`service ?`
`04:54.45`	`AbsTradELic`	`ops`
`04:54.51`	`AbsTradELic`	`provider ?`
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`04:59.12`	`Cephallus`	`I know this isn't the freepbx channel...buuuuut I'm getting desperate, and there's a (modest amount) of real money involved as a bounty:`
`04:59.13`	`Cephallus`	`http://www.freepbx.org/forums/viewtopic.php?p=1849`
`05:06.56`	`stephane_`	`jour`
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`06:24.19`	`leopardus1`	`any experts on Trixbox?`
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`08:43.54`	`*** mode/#asterisk [+o Corydon-w] by ChanServ`
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`09:23.19`	`blueneon`	`ok im tired of this crap now, i installed the asterisk @ home cd (which installed asterisk 1.0.9 and CentOs). But my TDM400P seems to stop taking calls out of the blue and when i try to make a call the line is dead except for a loud hissing (white noise). I restart the zaptel service and all is well again. Anyway I think I am gonna wipe the box and install fedora core 3 with the latest asterisk all manually, what else will i need>`
`09:23.20`	`blueneon`	`?`
`09:23.51`	`*** join/#asterisk ramtha (n=tk@p5088AEA2.dip0.t-ipconnect.de)`
`09:26.48`	`shellshark`	`we dont support asterisk@home here`
`09:27.09`	`shellshark`	`try #freepbx`
`09:27.35`	`shellshark`	`1.0.9 is ancient anyway`
`09:28.13`	`blueneon`	`i know`
`09:28.28`	`blueneon`	`thats why i said i want to remove it and install my own distro of linux and asterisk`
`09:28.42`	`blueneon`	`like i said fedora core 3 and the lastest asterisk`
`09:28.50`	`blueneon`	`well latest stable version`
`09:28.52`	`ramtha`	`then go on and have fun ;)`
`09:28.54`	`blueneon`	`what else will i need`
`09:29.01`	`blueneon`	`like zap drivers? etc`
`09:29.09`	`blueneon`	`<-- noob`
`09:29.14`	`ramtha`	`do you have pri interfaces?`
`09:29.18`	`blueneon`	`no`
`09:29.20`	`ramtha`	`the you need zaptel`
`09:29.22`	`blueneon`	`just a TDM400P`
`09:29.32`	`ramtha`	`the you need zaptel`
`09:29.44`	`ramtha`	`asterisk and zaptel`
`09:29.49`	`blueneon`	`ok so asterisk + zaptel is all i would need to compile and install`
`09:29.51`	`blueneon`	`sweet`
`09:30.00`	`ramtha`	`if you want realtime mysql support, you need additional asterisk-addons`
`09:30.01`	`blueneon`	`thanks mate, this should be fun`
`09:30.02`	`blueneon`	`lol`
`09:30.21`	`blueneon`	`um`
`09:30.30`	`blueneon`	`what do u mean real time support for mysql?`
`09:30.52`	`blueneon`	`you mean to edit the db? (i could use phpmyadmin surely?)`
`09:30.58`	`ramtha`	`you can use mysql for astersik configuration instead of using conf files in /etc`
`09:31.15`	`blueneon`	`hmm, i see most tutorial talk about the conf files`
`09:31.24`	`blueneon`	`should i not rather stick to using those?`
`09:31.45`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`09:31.48`	`ramtha`	`for the first time, conf files should be ok ;)`
`09:32.00`	`ramtha`	`if you have working this, try to switch to mysql`
`09:32.09`	`blueneon`	`its a really basic setup i have, one FXO and one FXS`
`09:32.23`	`blueneon`	`all calls come in and are routed to the fxs`
`09:32.27`	`blueneon`	`i have no voip etc`
`09:32.29`	`blueneon`	`(not yet)`
`09:32.42`	`blueneon`	`ok`
`09:32.44`	`blueneon`	`i'll do that`
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`10:10.03`	`skrusty`	`anyone know what this means?`
`10:10.17`	`skrusty`	`Oct 21 11:22:07 WARNING[5014]: chan_sip.c:3442 process_sdp: Unknown SDP media type in offer: image 50 udptl`
`10:10.47`	`oej`	`Asterisk does not support T.38 faxing in your version`
`10:12.00`	`skrusty`	`hmm, not trying to use T.38`
`10:12.04`	`skrusty`	`that's what's confusing me :)`
`10:12.54`	`skrusty`	`thanks though`
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`11:13.47`	`mrg82`	`My telecoms provider currently forwards calls to my sip account. I wish the calls to made to my newly build asterisk server via IAX. What details would i need to provide? just my server name? iax.domainname.com`
`11:20.07`	`tzafrir_laptop`	`Basically yes`
`11:20.35`	`tzafrir_laptop`	`But does the provider support IAX? What codecs?`
`11:20.55`	`mrg82`	`yes they do`
`11:21.02`	`mrg82`	`gsm i think`
`11:26.52`	`mrg82`	`would a username and password make it anymore secure?`
`11:27.06`	`mrg82`	`I can't see the benefit really`
`11:27.33`	`tzafrir_laptop`	`You have to authenticate calls on both directions`
`11:27.59`	`tzafrir_laptop`	`BTW: IAX2 can also use RSA keys instead of username/passwords`
`11:28.23`	`mrg82`	`i see`
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`12:36.00`	`ukh`	`I'm really at loss here. I'm a total asterisk newbie, trying to follow the book, but I cannot make a simple IAX2 FWD peer work, all I get is "Unable to negotiate codec", despite sprinkling the place liberally with "disallow=all" and "allow=ulaw"`
`12:37.43`	`ukh`	`using "iax2 debug", the outgoing call is clearly "ulaw", so I wonder if it possibly hasn't anything to do with the codec at all.`
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`13:00.51`	`mrg82`	`When attempting to use MusicOnHold i get the error: NOTICE[1133]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!`
`13:01.19`	`mrg82`	`I've installed asterisk addons that comes with format_mp3`
`13:01.49`	`xheliox`	`http://www.asteriskguru.com/tutorials/request_schedule_past.html`
`13:02.18`	`mrg82`	`yes, i found that before`
`13:02.24`	`mrg82`	`but the sound isn't playing at all`
`13:03.16`	`mrg82`	`im using format_mp3 and don't have mpg123 installed`
`13:30.47`	`*** join/#asterisk toxap (i=toxap@194.187.128.88)`
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`13:40.21`	`Druken`	`morning voip world`
`13:42.12`	`*** join/#asterisk tparcina_ (n=tomo@83-131-134-226.adsl.net.t-com.hr)`
`13:42.52`	`Rhizome`	`;)`
`13:43.20`	`DarKnesS_WolF`	`morning Druken`
`13:43.35`	`Rhizome`	`Imagine when theres only one cable going into everyones home, the power cable! phone, internet, tv and power on one cable.. yay ;)`
`13:43.37`	`*** join/#asterisk mboehn (i=mathias@65.23.156.248)`
`13:43.52`	`DarKnesS_WolF`	`Rhizome: hehe AMIN !`
`14:01.23`	`Damin`	`BEER!`
`14:01.25`	`Damin`	`Yes..`
`14:01.26`	`Damin`	`There..`
`14:01.28`	`Damin`	`I said it!`
`14:01.30`	`Damin`	`BEER!`
`14:02.00`	`xheliox`	`We're open source enthusiasts, I think only hard liquor is appropriate.`
`14:03.49`	`*** join/#asterisk lintechnokrats (n=chikki@61.17.68.129)`
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`14:05.51`	`SuPrSluG`	`hello`
`14:05.56`	`DarKnesS_WolF`	`hi`
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`14:20.00`	`SuPrSluG`	`quiet today`
`14:20.20`	`*** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se)`
`14:20.52`	`shy_guy`	`Nov 16 18:41:19 WARNING[32432] chan_sip.c: Trying to destroy "2725AC94-8189-4F6A-A3C6-DA4ED3C4F9C9@192.168.0.25", not found in dialog list?!?!`
`14:21.16`	`shy_guy`	`not found in dialog list..??`
`14:23.03`	`SuPrSluG`	`never heard that one`
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`14:39.07`	`saftsack`	`has * support for an authenthification realm user name?`
`14:40.12`	`blueneon`	`ok i just installed a fresh copy of fedora core 3 and asterisk 1.2 and zap drivers etc etc, all is up and running as it should be, except when incoming calls come through i have the following: exten=>s,1,Dial(ZAP/2,60,m) .. it dials the extention but the caller doesnt get the music on hold, just the ringing :/ any ideas?`
`14:52.34`	`*** join/#asterisk ariel_ (i=Ariel@11.sub-75-200-173.myvzw.com)`
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`15:19.04`	`tparcina_`	`blueneon: why do you use FC3? Why not FC5?`
`15:22.40`	`coppice`	`by using something no longer supported he doesn't get troubled by any of those annoying security updates`
`15:26.24`	blueneon`	`cause its the only cd's i have and dont feel in the mood to down load the latest`
`15:26.25`	blueneon`	`hehe`
`15:26.36`	blueneon`	`this is just for testing purposes not comercial use`
`15:26.42`	blueneon`	`anyway i solved that issue`
`15:26.46`	blueneon`	`installed mpg123`
`15:26.53`	blueneon`	`but now i have a new issue`
`15:27.08`	blueneon`	`im trying to write a autoattendant`
`15:27.31`	blueneon`	`but for some reason when the user dials the # the attendant isnt detecting it`
`15:29.04`	`tparcina_`	`coppice :))`
`15:29.50`	`tparcina_`	`blueneon: don't use mpg123, use nativ sounds`
`15:30.04`	`*** join/#asterisk abatista (n=Ariel@adsl-146-73-57.mia.bellsouth.net)`
`15:30.18`	`tparcina_`	`blueneon: it doen't make sense to test on something that you won't use in production anyway`
`15:30.51`	`tparcina_`	`blueneon: it just seams like time wasting`
`15:32.35`	blueneon`	`http://www.nexusisp.co.za/extensions.txt`
`15:32.38`	blueneon`	`thats what im using`
`15:32.51`	blueneon`	`but when the caller dials 1 or 2 nothing happens`
`15:34.03`	`saftsack`	`hi, is a patton smartnode owner here?`
`15:37.15`	`ManxPower`	blueneon`:you must be using a SIP phone.
`15:38.09`	blueneon`	`no u dont`
`15:38.13`	`ManxPower`	blueneon`: Also exten => s,6,Wait(10) should be WaitExten if you want asterisk to listen to DTMF while waiting
`15:38.15`	blueneon`	`and i fixed it`
`15:38.27`	blueneon`	`just had to move the 1/2 options under the s lines`
`15:38.31`	blueneon`	`instead of on tope`
`15:38.33`	blueneon`	`top*`
`15:38.54`	blueneon`	`ok i'll change it to WaitExten`
`15:52.24`	`*** join/#asterisk asdx (n=diego@200.61.236.33)`
`16:02.43`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`16:04.19`	blueneon`	`how do i add a local sip client to asterisk?`
`16:04.34`	blueneon`	`i have installed x-lite on a workstation on the same network as asterisk`
`16:05.04`	`*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)`
`16:05.45`	`DarKnesS_WolF`	blueneon`: u add a extention to that X-lite in sip.conf
`16:05.53`	`DarKnesS_WolF`	`then u write the dialplan in extension.conf`
`16:08.03`	`ManxPower`	`DarKnesS_WolF: amazing how many people ask basic questions that would be answered by reading The Book.`
`16:08.10`	`*** join/#asterisk oQPa (i=Ftv@234.Red-83-44-35.dynamicIP.rima-tde.net)`
`16:08.59`	`DarKnesS_WolF`	`ManxPower: :-)`
`16:09.00`	`DarKnesS_WolF`	`i know`
`16:09.06`	`DarKnesS_WolF`	`but i think about somehting`
`16:09.10`	`DarKnesS_WolF`	`u can give them headlines`
`16:09.15`	`DarKnesS_WolF`	`then u push them to read ;-)`
`16:10.44`	blueneon`	`DarKnesS_WolF i did that`
`16:10.50`	`ManxPower`	`people don't read!`
`16:11.08`	blueneon`	`but i dont think i did it correctly because when the xlite trys to register with asterisk it says...`
`16:11.09`	`ManxPower`	blueneon`: put your sip.conf entry on pastebin, as well as the extensions.conf entry.
`16:11.19`	blueneon`	`Registration from '"PC"<sip:3@192.168.1.1>' failed for '192.168.1.250' - Not a local SIP domain`
`16:11.20`	`ManxPower`	`~pastebin`
`16:11.25`	`jbot`	`methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`16:12.05`	`DarKnesS_WolF`	`ManxPower: hmm ur right :-) i take my answer back ;-)`
`16:12.29`	blueneon`	`http://channels.debian.net/paste/4151`
`16:14.19`	`ManxPower`	blueneon`: set X-lite to use the username of "3", not PC then fix your sip.conf entry as shown here. http://pastebin.ca/213728
`16:14.47`	`*** part/#asterisk oQPa (i=Ftv@234.Red-83-44-35.dynamicIP.rima-tde.net)`
`16:14.59`	blueneon`	`xlite is using "#"`
`16:15.00`	blueneon`	`err`
`16:15.01`	blueneon`	`3`
`16:15.03`	`ManxPower`	`In your extensions.conf dial line make it like Dial(SIP/3@3)`
`16:15.32`	`ManxPower`	blueneon`: no, if X-lite was using 3 then your error message would be Registration from '"3"<sip:3@192.168.1.1>' failed for '192.168.1.250' - Not a local SIP domain
`16:17.22`	`ManxPower`	`X-Lite's setup interface is more complicated than Rube Goldberg's wildest dream.`
`16:17.33`	blueneon`	`trust me its set to 3`
`16:17.41`	blueneon`	`"Display Name" is set to PC`
`16:17.46`	blueneon`	`username is 3`
`16:17.48`	`ManxPower`	blueneon`: then change the damn display name and see.
`16:18.04`	blueneon`	`same`
`16:18.10`	blueneon`	`Oct 21 18:18:02 NOTICE[9649]: chan_sip.c:11084 handle_request_register: Registration from '"3"<sip:3@192.168.1.1>' failed for '192.168.1.250' - Not a local SIP domain`
`16:18.12`	`ManxPower`	`I am telling you that it is passing PC into the registration request. Displayname is frequently what is passed.`
`16:18.17`	blueneon`	`^`
`16:18.30`	`ManxPower`	blueneon`: no that is different. Now did you do the sip.conf changes I gave you and then do a reload?
`16:18.39`	blueneon`	`yes`
`16:18.44`	blueneon`	`verbatium`
`16:19.17`	`ManxPower`	`is there a sip domain or realm in the X-lite config. If so, clear it out.`
`16:19.40`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`16:20.17`	blueneon`	`woot`
`16:20.19`	blueneon`	`working now`
`16:20.22`	blueneon`	`that was the issue`
`16:20.39`	blueneon`	`xlite had the proxy as 192.168.1.1 i removed it and it registered`
`16:20.40`	blueneon`	`thanks`
`16:20.50`	`*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)`
`16:20.57`	`ManxPower`	`the problem is that the username= should be the same as in the [whatever]`
`16:21.16`	`ManxPower`	`also X-lite was sending a different registration name than the username`
`16:21.33`	`ManxPower`	`http://www.voip-info.org/wiki/view/pbxnsip+X-Lite should have been helpful.`
`16:21.37`	blueneon`	`hmm i dial 7777 from xlite but dont get the simulated external call it just says the person u are trying to call is unavailable`
`16:21.54`	`DarKnesS_WolF`	`i have a question now i registered a ipkall number to my asterisk to extenton 119 .. do i need any special configurations in asterisk ?`
`16:22.36`	blueneon`	`infact anything i dial from it doesnt work`
`16:22.44`	blueneon`	`even trying to dial the zap/2 extention`
`16:22.46`	blueneon`	`:(`
`16:24.52`	`DarKnesS_WolF`	`i mean when someone will dial the number ipkall will call exten@asterisk.server right ? asterisk will make the call ? or it will need special configurations ?`
`16:25.39`	`ManxPower`	blueneon`: something should show up on the console.
`16:27.02`	blueneon`	`nope`
`16:27.09`	blueneon`	`nothing shows up on CLI`
`16:27.28`	blueneon`	`in xlite there is a message "address incomplete"`
`16:27.30`	`ManxPower`	`If it says the person is unavailabe then there should be CLI output.`
`16:27.54`	blueneon`	`i think that voice prompt is a xlite one not coming from asterisk`
`16:28.16`	`ManxPower`	`I assume you have an exten => 7777,1,Dial.... in the [internal] section of extensions.conf?`
`16:28.39`	`DarKnesS_WolF`	`got it i should have context for ipkall in sip.conf`
`16:28.40`	`DarKnesS_WolF`	`makes sense`
`16:29.28`	`ManxPower`	`you always need an extensions.conf [whatever] to match the sip.conf context=whatever`
`16:30.16`	blueneon`	`no i dont`
`16:30.29`	blueneon`	`but isnt that a build in extention in asterisk?`
`16:30.37`	blueneon`	`to simulate external calls?`
`16:30.42`	`ManxPower`	blueneon`: asterisk has no built in extensions
`16:31.21`	`ManxPower`	`I'm sure some of those sissy guis have that feature, but you are not using one of those or you would have said something earlier.`
`16:32.48`	blueneon`	`ok so how would i add an extention to asterisk that would simulate an external call`
`16:32.59`	`*** join/#asterisk VibroMax (n=phisto@83.229.70.154)`
`16:33.03`	`ManxPower`	blueneon`: I have no idea what you mean.
`16:34.13`	blueneon`	`when i installed asterisk@home, i could from any internal extention dial 7777 and that would simulate an external call on asterisk, ie, asterisk would treat me as an external calling calling the trunk`
`16:35.11`	`ManxPower`	`That is something specific to Asterisk@Home. I don't know how they do it, because either the call works or it does not. Asterisk does not even really have the concept of internal .vs. extenal calls. They are all just calls.`
`16:35.33`	`ManxPower`	`exten => 7777,1,Playback(vm-thankyou)`
`16:35.43`	`ManxPower`	`exten => 7777,n,Hangup()`
`16:35.56`	`ManxPower`	`But that is NOT a simulated call. That is a regular call to a regular extension.`
`16:36.15`	`ManxPower`	`the extension does not to much, but that does not matter.`
`16:37.04`	blueneon`	`u dont understand what i mean`
`16:37.11`	blueneon`	`so never mind ::)`
`16:37.19`	`ManxPower`	blueneon`: read The Book, then come back
`16:37.32`	blueneon`	`if i had a book to read that would be great :)`
`16:37.41`	`ManxPower`	`~book`
`16:37.47`	`jbot`	`methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`16:37.47`	`ManxPower`	`it's available online`
`16:37.59`	blueneon`	`ta`
`16:46.42`	`*** join/#asterisk ariel_ (n=Ariel@adsl-146-73-57.mia.bellsouth.net)`
`16:50.39`	`*** join/#asterisk santiago (n=santiago@debian/developer/santiago)`
`16:51.20`	`*** join/#asterisk ksh (i=ksh@indigo.nsl.it)`
`16:57.35`	`*** join/#asterisk ScurvyDawg (n=scurvyda@S0106000d883f28a0.gv.shawcable.net)`
`16:58.01`	`Nugget`	`heh`
`17:15.03`	`*** join/#asterisk Avochelm (n=damien@CPE-60-226-238-40.sa.bigpond.net.au)`
`17:15.07`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`17:20.25`	`*** join/#asterisk saftsack (n=saftsack@p54A7E8A1.dip.t-dialin.net)`
`17:20.36`	`saftsack`	`hi`
`17:25.47`	`*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`17:38.03`	`saftsack`	`i tested disa the last week and the sound quality was very bad at all :( are there some suggestions or is disas sound quality bad at all?`
`17:40.34`	`SuPrSluG`	`I have an IAX 800 number . when calling the ivr picks up and when an option is selected the caller hears no ringing. the callee does? any ideas?`
`17:45.13`	`*** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net)`
`17:46.29`	`SuPrSluG`	`when i try using sip dtmf is broken and i can't select a menu option. oh and when i hangup the call stays connected. weirdness!`
`17:47.07`	`ManxPower`	`SuPrSluG: what SIP detmfmode are you using?`
`17:47.38`	`ManxPower`	`SuPrSluG: you need a /etc/asterisk/indications.conf`
`17:47.47`	`SuPrSluG`	`nufone doesn't give it in their configs.`
`17:47.51`	`ManxPower`	`saftsack: disa does nothing with call qualifty.`
`17:48.04`	`ManxPower`	`SuPrSluG: that is because IAX2 only supports 1 DTMF mode.`
`17:48.25`	`SuPrSluG`	`ManxPower what should i have in indications?`
`17:48.36`	`ManxPower`	`SuPrSluG: whatever the default is.`
`17:49.03`	`ManxPower`	`SuPrSluG: If you are having DTMF problems with an IAX2 call then the issue is wherever the call is converted from PSTN to VoIP`
`17:49.08`	`Rhizome`	`yea im trying to get DTMF working aswell. read(DTMF) doesnt seem to work`
`17:49.55`	`ManxPower`	`DTMF issues can be caused by many, many totally different things.`
`17:50.08`	`SuPrSluG`	`dtmf work for iax, but the caller hears no ringing. when i switch to sip dtmf doesn't work`
`17:50.54`	`ManxPower`	`SuPrSluG: the caller does not hear any ringing because asterisk does not know how to indicate ringing to the calling party after the call has been answered by the IVR because you do not have /etc/asterisk/indications.conf.`
`17:51.27`	`SuPrSluG`	`indications is set to us which is correct. can i delete the other entries or do i need them when calling internationally`
`17:51.42`	`ManxPower`	`SuPrSluG: I have never ever modified inidcations.conf`
`17:51.50`	`ManxPower`	`just use the default and leave it at that.`
`17:52.18`	`SuPrSluG`	`i do have indications.conf. should i pastebin it?`
`17:52.28`	`ManxPower`	`SuPrSluG: is it the default one?`
`17:52.36`	`SuPrSluG`	`yes`
`17:52.44`	`ManxPower`	`then it should work. was it there before?`
`17:56.58`	`SuPrSluG`	`never had a problem before`
`17:57.37`	`*** join/#asterisk DarKnesS_WolF (n=wolf@217.54.209.1)`
`17:57.40`	`SuPrSluG`	`ManxPower:for sip what dtmf do you recommend?`
`18:01.22`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`18:02.16`	blueneon`	`for some reason i cant seem to get asterisk to evaluate the correct day/time of a call to send to the correct context.. here is my extentions.conf http://www.nexusisp.co.za/extentions.txt`
`18:02.24`	blueneon`	`err`
`18:02.33`	blueneon`	`http://www.nexusisp.co.za/extensions.txt`
`18:02.56`	blueneon`	`[trunk] is the context for incoming calls`
`18:03.37`	`*** join/#asterisk linlin (i=linlin@c-67-184-231-64.hsd1.il.comcast.net)`
`18:03.46`	blueneon`	`it keeps going directly to the [after-hours] context no matter what times are set`
`18:04.04`	`saftsack`	`ManxPower, i had a bad voice quality when i tried to contact to disa`
`18:04.49`	`*** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se)`
`18:10.07`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`18:11.43`	`SuPrSluG`	`blueneon:you should go to freepbx irc or nerdvittles for help with trixbox. they can help you more than here where it's pretty much straight up asterisk`
`18:13.38`	`Rhizome`	`So did you guys find the correct sip setting for DTMF? I have a PSTN nr to asterisk with IAX to another asterisk SIP account, and no DTMF ;)`
`18:14.29`	`*** join/#asterisk effenberg (n=jone@pD9E9D5F0.dip.t-dialin.net)`
`18:14.34`	`effenberg`	`moin`
`18:14.45`	`ManxPower`	`the correct sip.conf setting is almost always rfc2833`
`18:15.29`	`Rhizome`	`um, actually I was mistaken, I forgot it's a callback, so it's actually IAX to IAX to PSTN.`
`18:15.33`	`ManxPower`	`and NEVER inband if you are using any codec other than ulaw or alaw`
`18:15.43`	`jmls`	`hi guys. Is there anyway of generating a universal / global unique id from the dialplan (A uuid or guid). I want to have several asterisk servers sharing a cdr database, and want a unique reference for each call. Obviously, ${UNIQUEID} doesn't work across several * systems.`
`18:16.38`	`skopii`	`hey ManxPower remember when I was saying that I couldn't get my MWI to turn off on my polycom phone yesterday`
`18:16.49`	`ManxPower`	`skopii: yes`
`18:16.50`	`skopii`	`I finally figured out why`
`18:16.52`	`skopii`	`http://pastebin.ca/213900`
`18:16.56`	`ManxPower`	`skopii: why?`
`18:17.14`	`skopii`	`the first packet is leaving the * box`
`18:17.19`	`skopii`	`the second packet is coming into the phone`
`18:17.23`	blueneon`	`SuPrSluG im not running Trixbox`
`18:17.31`	blueneon`	`what gave u that impression?`
`18:17.44`	`ManxPower`	`skopii: so what is corrupting the packet?`
`18:17.47`	blueneon`	`im running fedora core 3 with asterisk 1.2`
`18:17.55`	`SuPrSluG`	`blueneon:Macros everywhere`
`18:18.01`	`skopii`	`so for some reason that is unknown to me the cisco 17xx is truncating hte packet`
`18:18.01`	blueneon`	`so?`
`18:18.11`	blueneon`	`whats wrong with me using macros?`
`18:18.15`	`*** join/#asterisk prttp (i=Ftv@159.Red-83-50-32.dynamicIP.rima-tde.net)`
`18:18.17`	`skopii`	`we are upgrading the firmware now...hopefully iy will fix`
`18:18.18`	blueneon`	`the feature is there for a reason no?`
`18:18.37`	`SuPrSluG`	`tougher to troubleshoot imo`
`18:18.47`	`ManxPower`	blueneon`: did you write the macros? Do you even understand what the macros do?
`18:19.00`	blueneon`	`and there is only 1 macro, how that constitutes as "everywhere" i dont know`
`18:19.04`	`ManxPower`	`skopii: I thought you were using polycoms on the same lan`
`18:19.09`	blueneon`	`yes i wrote them`
`18:19.14`	blueneon`	`and yes i understand them`
`18:19.23`	`ManxPower`	blueneon`: remove them when you need to test stuff.
`18:19.35`	blueneon`	`it`
`18:19.38`	blueneon`	`there is only 1`
`18:19.39`	blueneon`	`heh`
`18:19.43`	`skopii`	`=]`
`18:20.07`	blueneon`	`but i hear what ure saying`
`18:20.14`	blueneon`	`i guess i could just use inline coding`
`18:20.22`	`ManxPower`	`skopii: if the packets are going thru a Cisco nat router then "no service sip fixup 5060" or something like that`
`18:20.59`	`skopii`	`ahhh I figured it was something like that..I guess the cisco is actually rewriting the SIP headers or something?`
`18:21.21`	`ManxPower`	`skopii: it does that by default for NAT and SIP in some IOS versions.`
`18:21.35`	`*** join/#asterisk Anitalove (n=AnitaLov@dsl-207-112-41-18.tor.primus.ca)`
`18:21.40`	`Anitalove`	`hello everybody :)`
`18:21.45`	`ManxPower`	`with asterisk it generally causes more harm then good.`
`18:21.56`	blueneon`	`hmm actually i think i'll keep it, it makes life a little easier in that i dont have to retype the same code several times plus when / if i need to change it, i only have to do so in 1 place rather than several`
`18:22.17`	`ManxPower`	blueneon`: Macros are great --- but not while trying to diagnose problems.
`18:23.22`	blueneon`	`nod`
`18:23.28`	`Anitalove`	`I have 1 phone line at home, can I get asterisks to call me (on my cell), and then use the same line to 3-way call somebody else?`
`18:23.34`	`*** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it)`
`18:23.34`	blueneon`	`anyway i solved my issue`
`18:23.37`	blueneon`	`:D`
`18:24.14`	blueneon`	`well i kicked myself in the head really... the context was set for mon-fri (today is saturday !) lol`
`18:24.19`	`Rhizome`	`Anitalove: sure`
`18:25.40`	`Anitalove`	`Rhizome: how would I go about doing that? I know RTFM, but which manual, and where ;)`
`18:25.50`	`Rhizome`	`Anitalove: http://www.voip-info.org/wiki-Asterisk+auto-dial+out`
`18:26.02`	`Anitalove`	`tnx :)`
`18:26.03`	`Rhizome`	`Anitalove: Its pretty neat ;)`
`18:26.48`	`*** join/#asterisk segal_wor (i=segal_wo@cuscon4641.tstt.net.tt)`
`18:26.48`	`Anitalove`	`what would be cool is if when I call from my cellphone, it doesn't pickup and calls me back :)`
`18:27.00`	`Anitalove`	`(I have free incoming calls on my cell)`
`18:27.48`	`Rhizome`	`yea you can do that, what I do is make it run an agi script that makes the call file explained on that site.`
`18:28.21`	`ManxPower`	`I wish I had free incoming on my cell`
`18:29.13`	`Anitalove`	`Rhizome: nice, now, given that I have now expiernce with Asterisk and that I'm a CS Major and programmer, how much of a learning curve would it take to get this working?`
`18:29.18`	`Anitalove`	`now=no`
`18:29.55`	`ManxPower`	`Anitalove: Do you have experience with Linux, telephony, or networking?`
`18:30.20`	`Rhizome`	`there I found the site http://www.geocities.com/callbackagi/ that has a sample .agi script that does it`
`18:30.33`	`Anitalove`	`I have some experience with linux and networking... not with telephony... but I noticed that there's a Windows version of Asterisk available as well`
`18:30.46`	`Adam12`	`Yikes`
`18:31.07`	`Adam12`	`I didnt' think you could put windows and asterisk in the same sentence :\|`
`18:31.31`	`Anitalove`	`you just did :)`
`18:31.40`	`Adam12`	`;)`
`18:31.43`	`ManxPower`	`Adam12: you can't. The people that do get excommunicated and then burned at the stake`
`18:31.49`	`Anitalove`	`lol`
`18:32.18`	`Anitalove`	`I guess they have a big turnaround at http://www.asteriskwin32.com/ then`
`18:32.27`	`*** join/#asterisk effenberg (n=jone@pD9E9D5F0.dip.t-dialin.net)`
`18:32.34`	`effenberg`	`re`
`18:32.40`	`Rhizome`	`hehe, its so colorful!`
`18:32.44`	`Rhizome`	`;)`
`18:33.01`	`ManxPower`	`That is not a port of Asterisk to Windows. It is a port of Asterisk to cygwin.`
`18:33.16`	`Adam12`	`Hah. 'All circuits are busy. Please wait while we reboot Windows'`
`18:33.30`	`ManxPower`	`so you have all the complexity of linux plus all the complexity of cygwin, plus all the complexity of Windows.`
`18:33.39`	`Anitalove`	`lol`
`18:33.54`	`ManxPower`	`you can't use any PCI telephony cards either.`
`18:34.12`	`Anitalove`	`what kind of modem do I need to use for this kind of setup?`
`18:34.21`	`ManxPower`	`Anitalove: you don't.`
`18:34.32`	`Anitalove`	`huh?`
`18:34.40`	`ManxPower`	`you don't use modems with Asterisk`
`18:34.58`	`Anitalove`	`so, what do I use to interface Asterisk with a phone line?`
`18:35.19`	`tzafrir_laptop`	`Just get a spare partition and install linux on it`
`18:35.22`	`ManxPower`	`A card from Digium, Sangoma, or one of the knock-off no-name generic cards.`
`18:35.30`	`tzafrir_laptop`	`It will be simpler`
`18:35.35`	`ManxPower`	`the cards won't work under windows either.`
`18:35.36`	`tzafrir_laptop`	`(e.g: a separate HD)`
`18:35.58`	`*** join/#asterisk Dude34 (n=Aces1UP@ip68-96-234-176.lv.lv.cox.net)`
`18:36.21`	`Anitalove`	`how much do these cards go for?`
`18:36.34`	`ManxPower`	`Anitalove: $150 - $2,000 usually.`
`18:36.55`	`ManxPower`	`the one port, PSTN only generic card, not expandable can be under $30, but there is NO support for it here.`
`18:36.56`	`Anitalove`	`I was hoping to use my spare thinkpad laptop, it has a modem and I can put linux on it`
`18:37.15`	`ManxPower`	`Anitalove: you won't be able to interface to a pstn line if you do that.`
`18:37.43`	`Anitalove`	`hmm, intereting`
`18:37.50`	`ManxPower`	`You can see a list of compatable hardware on Digium's web sire.`
`18:37.53`	`ManxPower`	`site too.`
`18:38.13`	`ManxPower`	`http://www.asterisk.org/hardware`
`18:38.17`	`Rhizome`	`What kind of land line do you have?`
`18:38.19`	`Anitalove`	`O`
`18:38.37`	`Anitalove`	`normal phone line with ADSL internet access`
`18:39.10`	`bsdfreak`	`sipuras are good, too`
`18:39.22`	`bsdfreak`	`(which are external, network-based devices)`
`18:39.27`	`Rhizome`	`so thats an analog phone line?`
`18:39.40`	`Anitalove`	`yes`
`18:40.14`	`Rhizome`	`So you have a FXS PCI card.`
`18:40.19`	`Rhizome`	`need`
`18:40.36`	`ManxPower`	`Rhizome: I doubt his laptop will have such a thing.`
`18:40.44`	`ManxPower`	`and no, he needs an FXO card`
`18:40.48`	`ManxPower`	`~fxofxs`
`18:40.49`	`jbot`	`it has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.`
`18:41.19`	`Rhizome`	`right ;)`
`18:41.36`	`Anitalove`	`hmm`
`18:41.50`	`Rhizome`	`Here in Norway everyone has ISDN`
`18:41.55`	`Rhizome`	`Except my grandma ;)`
`18:42.29`	`Adam12`	`lol`
`18:42.39`	blueneon`	`is there a way to play the onhold music in the background while playing a voice over?`
`18:42.50`	`Anitalove`	`Digium's website says that it'll cost between $380 and $421 for analog Wildcard`
`18:43.21`	`ManxPower`	`Anitalove: Yes. Very cheap. Comparable cards for commercial PBXs are $2,000 - $4,000`
`18:43.52`	`Anitalove`	`yes, I was planning to do this for less than $50 lol`
`18:44.43`	`*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)`
`18:45.19`	`SuPrSluG`	blueneon`:audacity
`18:45.41`	`*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)`
`18:46.25`	`SuPrSluG`	blueneon`:then convert to gsm file
`18:46.48`	`Anitalove`	`in the hardware list, Generic X100P is listed, that's a WinModem`
`18:46.51`	blueneon`	`:/`
`18:47.22`	`Anitalove`	`can the Generic X100P be used to do what I discussed above?`
`18:47.26`	`Dude34`	`anyone here have experience with calling cards and asterisk?`
`18:47.41`	`*** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it)`
`18:48.12`	blueneon`	`in my musiconhold.conf i have : default=>quietmp3:/var/lib/asterisk/mohmp3 ... if i add -z to the end of that line (which i believe is meant to make it choose random mp3s in that dir) i get no music at all`
`18:48.15`	blueneon`	`any idea?`
`18:48.18`	blueneon`	`s*`
`18:49.22`	`ManxPower`	`try ,-z`
`18:49.43`	`ManxPower`	`assuming you are uing mpg123 0.59r of course`
`18:51.05`	`*** join/#asterisk JT (n=jon@unaffiliated/jt)`
`18:51.07`	blueneon`	`i am`
`18:52.14`	blueneon`	`i think im gonna try native`
`18:53.23`	blueneon`	`hmm`
`18:53.46`	blueneon`	`now i get a msg on the console saying "Music class default requested but no musiconhold loaded"`
`18:56.55`	`Anitalove`	`I live in Canada and use an analog phone line, what are the disadvantages to using an Generic X100P with Asterisk?`
`18:57.18`	`*** join/#asterisk BitBandit (n=polx@209.33.220.139)`
`18:57.34`	`*** join/#asterisk networkjedi (n=networkj@f3c30.gpcom.net)`
`18:59.40`	`Anitalove`	`funny, as soon as I mention the X100P, it goes really quiet in here... is it taboo?`
`19:00.18`	`*** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net)`
`19:00.55`	`*** join/#asterisk [1]BigBadHoss (n=BigBadHo@user-24-214-210-231.knology.net)`
`19:03.02`	`*** join/#asterisk Dude345 (n=Aces1UP@ip68-96-234-176.lv.lv.cox.net)`
`19:11.00`	`*** join/#asterisk Assid (n=assid@59.183.0.89)`
`19:12.04`	`*** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-eddd60983f7aec48)`
`19:19.26`	`[1]BigBadHoss`	`anybody know if polycom ever added nat keepalives`
`19:19.35`	`[1]BigBadHoss`	`i remember seeing a setting somewhere`
`19:20.55`	`*** join/#asterisk slayer192 (n=slayer19@adsl-70-137-4-246.dsl.okcyok.swbell.net)`
`19:22.41`	`teknoprep`	`i can't find the asterisk handbook.. the only one i am finding is that draft that is only 71 pages long`
`19:23.16`	`slayer192`	`Who's going to Astricon next week?`
`19:24.12`	`anthonyl`	`i am`
`19:24.34`	`anthonyl`	`slayer192, are you going to attend?`
`19:24.48`	`slayer192`	`anthonh: yeah`
`19:25.02`	`anthonyl`	`cool, we should get some drinks in that case ;)`
`19:25.22`	`slayer192`	`anthony: have to!`
`19:25.43`	`anthonyl`	`are you planningon hanging out in the code zone much?, i figure that is where i will be the majority of the time`
`19:25.51`	`anthonyl`	`brb a few mins tho, smoke`
`19:25.56`	`slayer192`	`k`
`19:30.39`	`anthonyl`	`okie`
`19:30.40`	`*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`19:31.16`	`slayer192`	`heh, how'd you know ;-)`
`19:33.07`	`slayer192`	`Not sure where I'll be most of the time. I know two of the speakers, want to catch their talks`
`19:36.49`	`slayer192`	`anthony: you from the DFW area?`
`19:37.04`	`*** join/#asterisk Qwell_ (n=north@unaffiliated/qwell)`
`19:37.04`	`*** mode/#asterisk [+o Qwell_] by ChanServ`
`19:39.50`	`anthonyl`	`slayer192, i'm from the huntsville area`
`19:40.09`	`anthonyl`	`i'm flying down monday afternoon to the DFW area tho`
`19:41.13`	`slayer192`	`not too bad of a trip`
`19:41.36`	`anthonyl`	`i hope not, i've never been to dallas before so i'm hopen it will be sort of cool`
`19:41.39`	`slayer192`	`I've got about an hour drive each day :-(`
`19:42.11`	`slayer192`	`Dallas has all sorts of things to do...name your poison`
`19:42.44`	`jtexter3`	`anybody got a good handle on how many people will be at Astricon this year?`
`19:42.59`	`shy_guy`	`i cant imagine a travelling time of more than 30mins when it comes to getting to work`
`19:43.09`	`slayer192`	`their site say "over 500 registered"`
`19:43.10`	`shy_guy`	`its tiring`
`19:43.27`	`slayer192`	`heh, I'll drive an hour for Astricon`
`19:43.47`	`shy_guy`	`slayer192:how can you live with that on daily basis`
`19:44.02`	`slayer192`	`I normally have a 25 minute drive to work now`
`19:44.11`	`shy_guy`	`slayer192:prolly its you are used to of it by now like many others`
`19:44.19`	`shy_guy`	`s/its/_`
`19:44.48`	`jtexter3`	`Personally, I miss my hour drive to work. Gave me time to just think`
`19:45.15`	`slayer192`	`shy_guy: you get used to it, time to wind down before I have to deal with ppl at home :-)`
`19:46.58`	`slayer192`	`anthonyl:what toys are you looking to play with in the code zone?`
`19:51.45`	`anthonyl`	`my laptop mostly ;)`
`19:52.46`	`anthonyl`	`atm i have access to all the voip hardware i really need too, i think the most intresting thing there will be seeing what other people are working on and going from there`
`19:53.34`	`slayer192`	`its always cool to see what other people are doing`
`19:54.31`	`slayer192`	`i recently came across an article where someone had intergrated nagios with asterisk and festival to alert you even if your net connection was down`
`19:55.40`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`19:55.55`	`[1]BigBadHoss`	`what days is astricon`
`19:56.40`	`slayer192`	`Oct 24-27`
`19:58.08`	`slayer192`	`anthonyl: have you used any othe the wifi phones?`
`20:01.09`	`linagee`	`how large of ping time is acceptable for a voip server? i have nagios set to tell me when voicepulse goes over 100ms, it's tripped lots of times. :-/`
`20:01.59`	`Juggie`	`400ms is the acceptal MOS delay`
`20:02.04`	`Juggie`	`*acceptable`
`20:02.08`	`linagee`	`for PCM`
`20:02.36`	`Juggie`	`but in that 400ms you have to count the time it takes to go from speaking to hearing`
`20:02.42`	`linagee`	`hrm`
`20:02.44`	`Juggie`	`so that includes network insertion time on any relays etc.`
`20:02.50`	`Juggie`	`time to process in *`
`20:02.52`	`Juggie`	`and so on`
`20:02.56`	`anthonyl`	`slayer192, i have`
`20:03.04`	`linagee`	`telco lag, cell phone lag, asterisk lag, voip provider lag, packet lag....`
`20:03.13`	`linagee`	`sip phone lag...`
`20:03.19`	`anthonyl`	`brb a few mins tho, i need to work on a few patches before i can irc some more`
`20:03.26`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`20:10.33`	`*** join/#asterisk AJaymn (i=TJBoi@24-159-236-181.dhcp.mdsn.wi.charter.com)`
`20:10.42`	`AJaymn`	`how can i turn the volume up on my Music on HOLD?`
`20:11.52`	`*** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net)`
`20:22.21`	`*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)`
`20:27.38`	`*** join/#asterisk linlin (n=linlin@c-71-194-70-13.hsd1.il.comcast.net)`
`20:29.53`	`*** join/#asterisk napkin (n=izaak@bas10-montrealak-1096753512.dsl.bell.ca)`
`20:32.02`	`*** join/#asterisk wl0 (n=ge@195-50-201-145-dsl.krw.estpak.ee)`
`20:37.13`	`*** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com)`
`20:39.13`	`effenberg`	`blubb`
`20:44.02`	`SuPrSluG`	`anyone come across this error for a tdm board zt_handle_event: Ring/Off-hook in strange state 6 on channel 5`
`20:44.51`	`SuPrSluG`	`and in /var/log/messages kernel: zt_rbs: Tried to set RBS hook state 0 on channel WCTDM/0/3 while span WCTDM/0 lacks rbsbits or hooksig function`
`20:46.39`	`*** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com)`
`20:49.48`	`*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)`
`20:51.41`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`20:51.41`	`*** mode/#asterisk [+o mog] by ChanServ`
`20:55.57`	`tzafrir_laptop`	`SuPrSluG, huh? What version of the driver? What board, exactly?`
`20:56.57`	`SuPrSluG`	`it's a tdm04b iv'e tried 1.2.10 and 1.2.9.1`
`20:57.54`	`SuPrSluG`	`it tries to answers and tries going to the menu when i get that error`
`20:59.15`	`*** join/#asterisk asdx (n=diego@200.61.236.33)`
`21:00.05`	`wl0`	`hi kind people, can I ask you to generate a trunk's configuration for asterisk based on data I got from provied, they gived to me an instruction for X-Lite phone configuration, but I'd like to use a Astrisk PBX with this provider but have no success.`
`21:01.28`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`21:06.47`	`CoffeeIV_`	`I have a recent asterisk that is not running any perl AGI scripts. They just exit with 0 immediately. I checked execute permissions and ownership. I installed AGI.pm the same way I did on other asterisk boxes, from source. Anyone have any ideas what I could check next ?`
`21:19.50`	`tzafrir_laptop`	`CoffeeIV_, is that a perl script?`
`21:19.54`	`napkin`	`so the deal is that cheapo digium x100 single fxo cards don't perform as well as tdm400 cards right? more interrupts or something?`
`21:21.03`	`tzafrir_laptop`	`x100p? Those are not Digium's. (Digium sold them for 100$, even though they weren't worth that much, so there weren't really cheapo digium x100p)`
`21:21.41`	`tzafrir_laptop`	`Interrupts is not the issue. One major issue is impedance`
`21:21.43`	`*** join/#asterisk nitrico (n=aaaa@200.81.9.182)`
`21:22.07`	`tzafrir_laptop`	`that is: that card will hopehully work in the US, but probably not elsewhere`
`21:22.18`	`tzafrir_laptop`	`work well, that is`
`21:23.05`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`21:27.35`	`*** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`21:32.41`	`*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)`
`21:32.48`	`Dovid`	`evening all`
`21:38.51`	`teknoprep`	`yoyo`
`21:39.07`	`teknoprep`	`Batman Begins was the bet batman movie out of all of the ones they made`
`21:39.12`	`teknoprep`	`it was actually pretty damn good`
`21:40.21`	`Dovid`	`ans this has to do with asterisk y ?`
`21:45.42`	`*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`21:45.46`	`rene1`	`hey`
`21:45.59`	`Dovid`	`hello rene1:`
`21:46.29`	`rene1`	`what is the preferred way of limiting incoming calls in 1.4.x is call-limit or group-counting?`
`21:46.54`	`rene1`	`my problem is that queues sometimes may send more than one call to a peer, which is, non cool`
`21:47.56`	`*** join/#asterisk vvard (i=1000@200.72.63.251)`
`21:48.09`	`Adam12`	`teknoprep: I guess the new one is 'Dark Night' and is supposed to come out 2008. Too bad :\| I'd love to see it this year :)`
`21:48.12`	`rene1`	`i was figuring out that if group count was already 1 i might want to send the caller back to the queue to catch those odd multiple calls`
`21:49.04`	`vvard`	`hi, im looking for compatibility list for TDM400P , any1 knows where can i find some ?`
`21:50.02`	`Dovid`	`vvard: motherboard wise ? my answer is to just go with sangoma. they will work with u instead of saying try a diffrent motherboard`
`21:50.13`	`*** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com)`
`21:50.16`	`Dovid`	`rene1: havent used ques a whole lot. cant help u there`
`21:50.34`	`Dovid`	`vvad: there should be a list on digium's site about which MB's are compatible`
`21:51.34`	`teknoprep`	`is there a site that helps with asterisk running smoothly on vmware ?`
`21:51.35`	`vvard`	`ok, thx ill check it out`
`21:53.02`	`*** part/#asterisk murdmath (n=vircuser@mail.kimballequipment.com)`
`21:53.19`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`21:53.34`	`Dovid`	`teknoprep: i dont know of a particular sites, however there are many sites out there that will have info on it`
`21:53.43`	`Dovid`	`what problems are u running in to ?`
`21:54.07`	`teknoprep`	`just a small bit of stuttering with playing recorded vm's or audio for IVR's`
`21:54.21`	`Dovid`	`ok`
`21:54.34`	`Dovid`	`u prob dont have ztdummy running properly`
`21:54.34`	`teknoprep`	`i think its bloatedness of CentOS`
`21:54.42`	`teknoprep`	`hmmm?`
`21:54.49`	`teknoprep`	`what about ztdummy ?`
`21:55.02`	`Dovid`	`teknoprep: u have to install all the kernel sources on the main machines OS`
`21:55.10`	`Dovid`	`not good enough just on the virtual machine`
`21:55.32`	`teknoprep`	`Dovid ?`
`21:55.42`	`teknoprep`	`Dovid why would my host os have anything to do with my guest os?`
`21:55.42`	`Dovid`	`yes`
`21:56.30`	`Dovid`	`cause it seems that when it looks for the kern sources it goes to the host os, unsure as to y but i ran in to issues when i was trying to run astrisk on a vm and thats what i was told`
`21:56.46`	`Adam12`	`I think that's related to Xen only`
`21:56.51`	`file`	`it depends on the underlying technology`
`21:56.55`	`teknoprep`	`yeah i would agree with Adam12`
`21:57.13`	`file`	`VMware completely virtualizes stuff, it's essentially a different computer... it would have no notion it's running on a host OS`
`21:57.14`	`*** join/#asterisk xnon (i=xnon@200.8.30.161)`
`21:57.15`	`teknoprep`	`or Vserver`
`21:58.18`	`Dovid`	`i stand to be corrected. :)`
`21:58.20`	`xnon`	`for what is the app_striplsd.so?`
`21:58.35`	`xnon`	`my asterisk cannot run for this app`
`21:58.36`	`vvard`	`another question, will asterisk run ok in a 2.x Ghz processor and 512 ram server ?`
`21:58.38`	`teknoprep`	`look like its a strip of LSD`
`21:58.50`	`xnon`	`it is important for asterisk ¿`
`21:59.03`	`xnon`	`what it is your funcction¿`
`21:59.10`	`teknoprep`	`your asterisk needs LSD ?`
`22:00.38`	`xnon`	`i dont know what is LSD`
`22:00.51`	`xnon`	`is posible delete this app`
`22:00.51`	`xnon`	`¿?`
`22:01.09`	`rene1`	`least significant digit i think it is`
`22:01.24`	`xnon`	`:S`
`22:01.45`	`xnon`	`i i only what to run my asterisk`
`22:02.01`	`rene1`	`it is a function to remove digits from variables`
`22:02.09`	`xnon`	`but i cant for this app_striplsd.so`
`22:02.21`	`xnon`	`what can i do`
`22:02.22`	`xnon`	`?`
`22:02.25`	`CoffeeIV_`	`At some point, did asterisk dialplan macros change in how they passed arguments ?`
`22:02.35`	`xnon`	`can i bloq this app?`
`22:02.43`	`xnon`	`sorry my english is not so good`
`22:03.14`	`xnon`	`Oct 21 23:56:26 WARNING[7492]: loader.c:554 load_modules: Loading module app_str iplsd.so failed!`
`22:03.15`	`rene1`	`i dont think removing such a tiny app would give you a major perfomance boost but if you must you can`
`22:03.21`	`xnon`	`this is the error in my console`
`22:03.27`	`rene1`	`i see`
`22:04.02`	`rene1`	`check modules.conf and see comment out any load app_striplsd.so modules`
`22:04.15`	`rene1`	`statements`
`22:04.16`	`rene1`	`i meant`
`22:04.28`	`xnon`	`ok hold on`
`22:04.31`	`*** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net)`
`22:05.13`	`rene1`	`is there a GSM player that will play faster than normal?`
`22:05.18`	`xnon`	`emmm no! i cant see it in modules.conf`
`22:05.43`	`rene1`	`then add a line for it that reads no load ....`
`22:05.46`	`rene1`	`use the examples`
`22:06.18`	`rene1`	`i need a gsm player that will allow me to listen to a gsm file in half time than its original duration`
`22:06.30`	`xnon`	`ok`
`22:06.44`	`xnon`	`noload => app_striplsd.so`
`22:06.45`	`xnon`	`?`
`22:06.53`	`Dovid`	`rene1: i use winamp with a plugin - i am sure u can find plugins for other players too`
`22:07.07`	`Dovid`	`i dont think winamp has the ability to play a file fast`
`22:07.35`	`CoffeeIV_`	`when you call a macro, the arguments are supposed to be in $ARG1, $ARG2, etc, right ? When I am reading those variables from an agi script in the macro, they are empty`
`22:08.09`	`teknoprep`	`wow`
`22:08.19`	`teknoprep`	`the O'Riely book for asterisk is very well written`
`22:08.43`	`Dovid`	`CoffeeIV_: ${ARG1}`
`22:09.16`	`Dovid`	`teknoprep: it was writeen real well, as it is in the opening statements, by the time u read it, it will be out dated`
`22:09.24`	`Dovid`	`which its getting too`
`22:09.53`	`teknoprep`	`anything not outdated?`
`22:09.57`	`teknoprep`	`or would that be impossble ?`
`22:10.06`	`Dovid`	`CoffeeIV_: are you using $ARG1 or ${ARG1} ?`
`22:11.00`	`xnon`	`friends`
`22:11.03`	`Dovid`	`teknoprep: i was exadurating a bit, most of it is current however there are a lot of new things that arent in the book and somethings have been out dated so when u use thier examples etc.`
`22:11.03`	`xnon`	`thanx so much`
`22:11.13`	`Dovid`	`see if they work and double check em on the web etc.`
`22:11.20`	`xnon`	`i do it and i can run my asterisk`
`22:11.21`	`xnon`	`;)`
`22:11.25`	`Dovid`	`for instance SetVar is no longer used, now its Set`
`22:13.56`	`ManxPower`	`SetVar still works in 1.2`
`22:16.01`	`Dovid`	`ManxPower: yes however when it is called uou will notice in the CLI that it tells u that its out dated and AFAIK in 1.4 it dosent work`
`22:18.28`	`asdx`	`is there a software-client for asterisk?`
`22:19.13`	`teknoprep`	`IDEFISK ?`
`22:20.09`	`qdk`	`asdx: loads...`
`22:20.13`	`Dovid`	`asdx: what do u mean by software client ? gui ?`
`22:20.58`	`asdx`	`yeah, like a jabber client so you can talk to each other using a client <-> pbx/asterisk <-> client, if is possible...`
`22:21.15`	`teknoprep`	`IDEFISK`
`22:21.25`	`teknoprep`	`asdx, idefisk`
`22:21.37`	`asdx`	`i will check it out, thanks`
`22:21.51`	`teknoprep`	`asdx, you are going to have to first know how to set up asterisk before any of it will work`
`22:22.14`	`asdx`	`teknoprep: i will do that first :)`
`22:22.34`	`Dovid`	`asdx: read the book`
`22:22.35`	`Dovid`	`~book`
`22:22.39`	`jbot`	`hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`22:23.01`	`ManxPower`	`asdx: Any SIP device/software should work`
`22:23.20`	`rene1`	`a question regarding recording.. wav49 uses gsm frames right? would i take much of a hit in performance by choosing wav49 over plain gsm?`
`22:23.45`	`teknoprep`	`dovid is that book up to date ?`
`22:23.47`	`ManxPower`	`rene1: I doubt there will be much performance difference.`
`22:24.13`	`Dovid`	`it was realeased some time ago, but it still a great refrence`
`22:24.27`	`asdx`	`ok :) thanks people`
`22:24.40`	`asdx`	`i'll read it now`
`22:24.42`	`Dovid`	`the core of what u need to know is in the book. once u get used to the gen. sytnax, google and voip-info.org u will be good to go`
`22:24.56`	`asdx`	`alright`
`22:25.01`	`ManxPower`	`Just remember that 1/2 the info on voip-info is wrong`
`22:25.33`	`asdx`	`ok, one more thing, is it possible to call to asterisk from a normal telephone, non voip telephone...`
`22:25.34`	`Dovid`	`manxpower: i learnt the hard way, its y i have a dedicated server up that i use just for testing`
`22:25.42`	`Dovid`	`asdx: yes`
`22:26.04`	`asdx`	`Dovid: wow, nice`
`22:26.06`	`Dovid`	`asdx: can handle multiple lines, VOIP, POTS, ISDN, T1/E1 etc.`
`22:26.35`	`asdx`	`Dovid: what hardware will i need for doing that?`
`22:26.40`	`Dovid`	`asdx: u will be suprised at what asterisk can do. start reading "the book" .............`
`22:27.12`	`asdx`	`ok ok`
`22:27.34`	`asdx`	`i'm going for coke, then i'll start to read the book :D`
`22:27.34`	`[1]BigBadHoss`	`Asterisk: The futtur of telephony`
`22:27.42`	`rene1`	`get me a line`
`22:27.49`	`[1]BigBadHoss`	`haha`
`22:27.58`	`[1]BigBadHoss`	`ill take a key`
`22:28.16`	`rene1`	`he will fly tru it`
`22:28.38`	`Dovid`	`asdx: i would reccomnd ur fav. bottle of whisky as well, u will bang ur head against the wall in the begining`
`22:29.00`	`[1]BigBadHoss`	`its a little slow i agree`
`22:29.30`	`[1]BigBadHoss`	`thats badass that its CC though`
`22:29.36`	`[1]BigBadHoss`	`i have it on safari`
`22:29.43`	`[1]BigBadHoss`	`and a few other books`
`22:29.50`	`[1]BigBadHoss`	`anybody else use safari?`
`22:30.03`	`Druken`	`is there any good TTS besides cepstral that can be used with asterisk ?`
`22:30.14`	`Dovid`	`safari as in the broswer ?`
`22:30.21`	`[1]BigBadHoss`	`no, Oreilly`
`22:30.34`	`[1]BigBadHoss`	`runs a service that lets you search like a million books`
`22:30.46`	`[1]BigBadHoss`	`its a sysadmin, programmers, etc dream!`
`22:31.04`	`[1]BigBadHoss`	`http://safari.oreilly.com/`
`22:31.10`	`Dovid`	`wow`
`22:31.13`	`Dovid`	`thanks for that`
`22:31.32`	`[1]BigBadHoss`	`its worth the money if you read alot of tech books`
`22:31.49`	`asdx`	`Dovid: looooool`
`22:31.51`	`[1]BigBadHoss`	`plus you can add them to your "bookshelf" and get the whole book to read`
`22:32.50`	`[1]BigBadHoss`	`someone needs to write a CC configuring polycom soundpoint ips book`
`22:33.02`	`Dovid`	`no`
`22:33.09`	`[1]BigBadHoss`	`:)`
`22:33.13`	`[1]BigBadHoss`	`anyways`
`22:33.16`	`Qwell`	`polycom manual not enough? heh`
`22:33.22`	`Dovid`	`i think polycom should put all the damn options into a gui as oposed to us having to edit the damn files every time`
`22:33.24`	`[1]BigBadHoss`	`i did find out they finally added a nat keepalive`
`22:33.32`	`[1]BigBadHoss`	`i like the files`
`22:33.43`	`[1]BigBadHoss`	`someone else should make an editor thogh`
`22:33.50`	`Dovid`	`it gives u more control but its a pain to work with`
`22:33.56`	`Qwell`	`[1]BigBadHoss: like vi?`
`22:34.04`	`[1]BigBadHoss`	`no`
`22:34.08`	`[1]BigBadHoss`	`i mean like cisco does`
`22:34.16`	`[1]BigBadHoss`	`a config file generator`
`22:34.21`	`Qwell`	`why?`
`22:34.39`	`[1]BigBadHoss`	`i personally like the files, i think its quicker to set them up`
`22:34.42`	`Qwell`	`Don't the polycoms let you download their current config or something?`
`22:34.54`	`[1]BigBadHoss`	`but it scares some people`
`22:35.02`	`Dovid`	`took me 2 days to figure out how to edit the polycom so i could get auto answer, the snom's on the other hand, tell u waht kind of sip alert to send, and that info is in the manual as opposed to having to edit the files. Arghhhhhhhhhhhhhh !!!`
`22:35.40`	`Dovid`	`Qwell: only from a reseller and even then if u want paging, u goto edit it, if u want answer on second ring u goto edit the files as well.`
`22:36.03`	`[1]BigBadHoss`	`anybody know what release they put nat keepalives in?`
`22:37.26`	`rene1`	`polycom just need to make their phones boot faster than 5 minutes`
`22:37.28`	`[1]BigBadHoss`	`<nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/> is all i have in 1.6.2 sip`
`22:37.32`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`22:37.52`	`Dovid`	`lol`
`22:37.55`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`22:38.17`	`[1]BigBadHoss`	`anybody have the latest polycom firmware?`
`22:38.37`	`[1]BigBadHoss`	`or at least one that supports nat keepalives`
`22:38.41`	`Dovid`	`nope: but some one recently posted a link on the users list`
`22:38.49`	`Dovid`	`look thru thru archive's of the last month`
`22:38.57`	`[1]BigBadHoss`	`where @?`
`22:39.05`	`[1]BigBadHoss`	`which list`
`22:39.07`	`*** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`22:39.17`	`*** join/#asterisk backblue (n=moo@87-196-33-82.net.novis.pt)`
`22:39.45`	`Dovid`	`http://lists/digium.com/asterisk-users`
`22:39.47`	`Dovid`	`have a look here`
`22:39.49`	`Dovid`	`http://www.freedomphones.net/polycom/files/`
`22:40.57`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`22:41.34`	`[1]BigBadHoss`	`does ssip work on polycom 500s?`
`22:42.35`	`[1]BigBadHoss`	`another problem i have`
`22:42.44`	`[1]BigBadHoss`	`i changed a phone from mgcp to sip`
`22:42.54`	`[1]BigBadHoss`	`and then reset the config`
`22:43.15`	`[1]BigBadHoss`	`now when i put the ip address of my tftp server in`
`22:43.25`	`[1]BigBadHoss`	`it says it couldnt contact the boot server`
`22:43.43`	`[1]BigBadHoss`	`but i ran ethereal, and it dosen't ever even look at the server`
`22:44.36`	`[1]BigBadHoss`	`can this be fixed?`
`22:47.17`	`*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)`
`22:47.43`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.161)`
`22:48.00`	`ManxPower`	`[1]BigBadHoss: I've never had that problem`
`22:48.21`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`22:48.24`	`ManxPower`	`also define "ssip"`
`22:48.54`	`[1]BigBadHoss`	`the ssip file on the freedomphones site`
`22:49.04`	`[1]BigBadHoss`	`http://www.freedomphones.net/polycom/files/spip_ssip_sip_2_0_1.zip`
`22:49.40`	`ManxPower`	`Ah. That is just standard SIP`
`22:50.25`	`ManxPower`	`Many people stick to the 1.2.x firmware since there have been so many issues with 2.0.x`
`22:50.44`	`ManxPower`	`sorry, 1.6.x`
`22:51.26`	`ManxPower`	`[1]BigBadHoss: also can YOU download the file from the server using a tftp client?`
`22:52.04`	`*** join/#asterisk slayer192 (n=slayer19@adsl-70-137-4-246.dsl.okcyok.swbell.net)`
`22:53.03`	`ManxPower`	`for 1.6.x: http://www.fnords.org/~eric/polycom-config-examples/`
`22:55.10`	`ManxPower`	`I manage about 70 polycoms, but we do it via DHCP and FTP`
`22:56.28`	`[1]BigBadHoss`	`i was sticking to it`
`22:56.38`	`[1]BigBadHoss`	`but i need nat keepalive`
`22:56.46`	`[1]BigBadHoss`	`which is only in 2.0.1`
`22:56.57`	`ManxPower`	`[1]BigBadHoss: 1) Why? 2) 1.6.x should support that.`
`22:57.05`	`[1]BigBadHoss`	`nat keepalives?`
`22:57.34`	`[1]BigBadHoss`	`looking at my current config files, theres nothing in there`
`22:58.15`	`ManxPower`	`Well you need to PUT something in there.`
`22:58.30`	`ManxPower`	`The phone doesn't just magically know the correct settings for your network.`
`22:58.36`	`[1]BigBadHoss`	`i know`
`22:58.55`	`[1]BigBadHoss`	`i have a network of 10 running behind a firewall`
`22:59.05`	`[1]BigBadHoss`	`but i need some at another office just down the road`
`22:59.09`	`[1]BigBadHoss`	`to be able to connect`
`22:59.12`	`ManxPower`	`OK.`
`22:59.24`	`[1]BigBadHoss`	`and i think ipsec will add too much latency/packet overhead`
`22:59.29`	`[1]BigBadHoss`	`or am i wrong`
`22:59.32`	`ManxPower`	`Seems pretty simple to me. Follow the wiki instructions for running Asterisk behind NAT`
`23:00.04`	`ManxPower`	`do the port forwarding, set up rtp.conf, set externip and localnet in sip.conf`
`23:00.15`	`[1]BigBadHoss`	`well, polycoms before 2.0.1 didnt send a keepalive, so the sip registrations wouldnt work`
`23:00.20`	`*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)`
`23:00.24`	`[1]BigBadHoss`	`the nat holes would time out`
`23:00.29`	`ManxPower`	`You are making this WAY more complicated then it is.`
`23:00.41`	`[1]BigBadHoss`	`ok help me out then`
`23:00.47`	`Strom_C`	`coughqualify=yescough`
`23:00.50`	`ManxPower`	`set your registration refresh time to 60 seconds or use qualify=50000`
`23:01.08`	`[1]BigBadHoss`	`in phone1.cfg?`
`23:01.08`	`ManxPower`	`Strom_C: I thinking qualify is a bad idea.`
`23:01.15`	`ManxPower`	`no in sip.conf`
`23:01.18`	`[1]BigBadHoss`	`ok`
`23:01.18`	`ManxPower`	`for the qualify`
`23:01.31`	`Strom_C`	`ManxPower: how so?`
`23:01.32`	`ManxPower`	`for the reg interval it would be in sip.cfg or phone1.cfg`
`23:01.47`	`ManxPower`	`Strom_C: things go lagged or unreachable when they should not be.`
`23:02.23`	`[1]BigBadHoss`	`is this it:`
`23:02.23`	`[1]BigBadHoss`	`reg.1.server.1.register=""`
`23:02.29`	`ManxPower`	`[1]BigBadHoss: also you need to have port forwarding as well.`
`23:02.42`	`ManxPower`	`[1]BigBadHoss: did you look at the sample config files I posted?`
`23:02.57`	`[1]BigBadHoss`	`where`
`23:03.11`	`ManxPower`	`(17:52:31) ManxPower: for 1.6.x: http://www.fnords.org/~eric/polycom-config-examples/`
`23:05.04`	`[1]BigBadHoss`	`so all i need to do is change the server and auth information in there?`
`23:07.49`	`[1]BigBadHoss`	`what versions are you using?`
`23:09.19`	`SuPrSluG`	`anyone come across this error for a tdm board zt_handle_event: Ring/Off-hook in strange state 6 on channel 5`
`23:09.48`	`*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)`
`23:10.20`	`hads`	`Yes`
`23:10.27`	`*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)`
`23:10.31`	`tzafrir_laptop`	`SuPrSluG, what is the actual problem you have?`
`23:11.45`	`SuPrSluG`	`when i place a call to the number it picks up show then menu is running. no sound. then that error message`
`23:12.24`	`tzafrir_laptop`	`set debug 10`
`23:12.30`	`SuPrSluG`	`i have an fxo and tdm04b`
`23:12.35`	`tzafrir_laptop`	`Enable the full log`
`23:12.40`	`SuPrSluG`	`at the CLI`
`23:12.48`	`tzafrir_laptop`	`SEe if there's anything interesting at the time of the ring`
`23:13.11`	`tzafrir_laptop`	`(yet another way to say that I have no idea)`
`23:15.04`	`*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)`
`23:16.43`	`SuPrSluG`	`here from the full log`
`23:16.50`	`SuPrSluG`	`ct 21 19:15:07 VERBOSE[16385]: -- Goto (eea,s,6)`
`23:16.52`	`SuPrSluG`	`Oct 21 19:15:07 VERBOSE[16385]: -- Executing BackGround("Zap/5-1", "night-greet-eea") in new stack`
`23:16.53`	`SuPrSluG`	`Oct 21 19:15:07 DEBUG[16385]: Scheduling timer at 160 sample intervals`
`23:16.55`	`SuPrSluG`	`Oct 21 19:15:07 VERBOSE[16385]: -- Playing 'night-greet-eea' (language 'en')`
`23:16.56`	`SuPrSluG`	`Oct 21 19:15:10 DEBUG[16385]: Exception on 25, channel 5`
`23:16.58`	`SuPrSluG`	`Oct 21 19:15:10 DEBUG[16385]: Got event Polarity Reversal(17) on channel 5 (index 0)`
`23:16.59`	`SuPrSluG`	`Oct 21 19:15:10 DEBUG[16385]: Ignore Reverse Polarity on channel 5, state 6`
`23:18.23`	`teknoprep`	`anyone know of any good doc's or websites on running asterisk inside of vmware ?`
`23:21.15`	`hads`	`SuPrSluG: Use pastebin`
`23:22.21`	`SuPrSluG`	`sorry`
`23:27.29`	`hads`	`Pastebin the entire log and someone may be able to help.`
`23:27.53`	`[1]BigBadHoss`	`looks like a bad card`
`23:27.58`	`[1]BigBadHoss`	`or bad wiring`
`23:28.09`	`[1]BigBadHoss`	`even:polarity reversal`
`23:28.26`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`23:29.04`	`[1]BigBadHoss`	`if you have another card you can try that`
`23:29.45`	`hads`	`A polarity reversal could be quite normal.`
`23:30.12`	`[1]BigBadHoss`	`but when combined with an exception`
`23:31.56`	`*** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com)`
`23:32.35`	`SuPrSluG`	`there's 2 cards one id xfo and the other tdm04b i have turned off the xfo card (no line to it). when i zap show channel 1 whether it is the xfo card or tdm04b it shows InAlarm 1`
`23:33.20`	`[1]BigBadHoss`	`too bad it dosent tell us whats causing the exception`
`23:33.32`	`SuPrSluG`	`it's been working fine for a year and a half`
`23:34.11`	`hads`	`It's on 4783 of chan_zap.c but I don't know the code.`
`23:34.51`	`SuPrSluG`	`i've been doing this remotely. so no one has come near the box`
`23:38.03`	`[1]BigBadHoss`	`have you changed anything lately`
`23:38.10`	`SuPrSluG`	`proc looks good`
`23:38.52`	`SuPrSluG`	`i tried to upgrade to 1.213`
`23:39.00`	`SuPrSluG`	`i tried to upgrade to 1.2.13`
`23:39.24`	`qdk`	`If i have a frontend SIP registration server and a backend Call processing server, how do i make sure that all the SIP phones doesnt transmit the voice stream through the registration server, but directly to the call processing unit?`
`23:43.51`	`[1]BigBadHoss`	`and that broke it?`
`23:44.13`	`hads`	`What happens if you roll back?`
`23:52.44`	`SuPrSluG`	`i have rolled back no change.`
`23:55.00`	`*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)`
`23:55.17`	`*** join/#asterisk goodbot (n=game@dsl-220-253-76-64.NSW.netspace.net.au)`
`23:55.27`	`goodbot`	`Howdy folks.`
`23:56.55`	`goodbot`	`Asterisk has a DNS/SIP bug.`
`23:57.08`	`*** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net)`
`23:57.10`	`goodbot`	`When a providers DNS server goes down or they use CNAMEs`
`23:57.23`	`goodbot`	`Asterisk behaves erratically.`
`23:57.32`	`goodbot`	`For example, when you do a sip show peers`
`23:57.36`	`goodbot`	`during this issue`
`23:57.44`	`goodbot`	`Providers copy all the same stat`
`23:57.46`	`goodbot`	`UNREACHABLE`
`23:57.54`	`goodbot`	`or when a provider uses CNAMEs`
`23:58.07`	`goodbot`	`the list is sorted alphabetically, and this copies over to the next provider.`
`23:58.19`	`goodbot`	`I have 2 providers listed, one Voxalot, the other WDP`
`23:58.33`	`goodbot`	`Voxalot has OK with 10ms, and that is copied over to WDP.`
`23:58.38`	`hads`	`Asterisk doesn't like it when DNS fails`
`23:58.51`	`goodbot`	`WDP has a CNAME record, and not an A record.`
`23:58.56`	`goodbot`	`Can this be error handled?`
`23:59.25`	`goodbot`	`use PHP gethostbyname() or something? :)`

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