IRC log for #asterisk on 20070521

`00:00.04`	`[hC]`	`[TK]D-Fender: hey while ive got you here, could i trouble you for a copy of your polycom microbrowser contact directory scripts?`
`00:00.22`	`Swat2`	`[TK]D-Fender: exten => _.,1,Set(CALLERID(name)=${CALLERID(num)}) <-- is that correct?`
`00:00.28`	`lee_is_me`	`kudzu reconginized it, but couldn't get it to work`
`00:00.34`	`lee_is_me`	`no lights, no nothing`
`00:00.40`	`Swat2`	`at the top of [from-zaptel]`
`00:00.52`	`[hC]`	`I want to try to build a nice web gui around managing a microbrowser based contact directory, and give it back to everyone to use. (if you didnt mind that I did that, as well, of course!)`
`00:00.57`	`[hC]`	`lee_is_me: was there another nic in the box?`
`00:01.00`	`*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)`
`00:01.01`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:01.02`	`lee_is_me`	`[hC]:ended up sticking a realtek in there`
`00:01.12`	`JT`	`realteks are evil`
`00:01.14`	`[hC]`	`lee_is_me: I have the same issue with that, with another card in the box, it screwed up the onboard nic. wouldnt load.`
`00:01.32`	`lee_is_me`	`[hC]: no, not until I disabled the onboard and put another in`
`00:01.39`	`JT`	`the only 100Mbit card you ever need is an intel eepro100 based card`
`00:02.01`	`[hC]`	`I have a couple linksys tulip cards installed, but i always prefer intel eepro100's too..`
`00:02.21`	`JT`	`eepros push a hell of a lot more data`
`00:02.23`	`lee_is_me`	`[hC]: Same thing with another entirely different pc so it might be something with my distro (Centos 4.4) not having the correct drivers...`
`00:02.30`	`JT`	`intel chipset stuff is some of the best under linux`
`00:02.48`	`Swat2`	`holy crap im late for work`
`00:02.51`	`[TK]D-Fender`	`Swat2, the line I gave you does exactly what you asked. Where you put it is up to you.`
`00:02.56`	`lee_is_me`	`JT: that is what i will use from now on and disable onboard nics`
`00:03.00`	`Swat2`	`l8r`
`00:03.11`	`[TK]D-Fender`	`Swat2, And your context name you provided tells us nothing of any value.`
`00:03.30`	`Swat2`	`[TK]D-Fender:, do i not have to put the exten => bit in?`
`00:04.05`	`[TK]D-Fender`	`Swat2, Of course it has to be an exten line, but you shouldn't have to ask us WHICH extens to add that to to get what you want.`
`00:04.08`	`JT`	`lee_is_me: good idea :)`
`00:04.13`	`[TK]D-Fender`	`Swat2, You should know that already.`
`00:04.15`	`*** join/#asterisk JerJer[mobile] (n=jj@45.129.210.2)`
`00:04.31`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net)`
`00:04.37`	`lee_is_me`	`JT: I have to break it down to the simplest I can get...life is too short.`
`00:04.43`	`Swat2`	`[TK]D-Fender: i was asking where abouts in my context i should place it.. first line or a few in?`
`00:05.20`	`[TK]D-Fender`	`Swat2, Asking that tells me you have know idea how the dialplan works or what yours does....`
`00:05.37`	`Swat2`	`[TK]D-Fender: we are all still learning.`
`00:05.45`	`lee_is_me`	`[TK]D-Fender: Do you remember when you were going to help me the other night with a TDM card? Turns out is was the card that was the problem.`
`00:05.58`	`[TK]D-Fender`	`Swat2, the "easy" answer is before any "dial" you wish to manipulate in this manner.`
`00:06.14`	`[TK]D-Fender`	`lee_is_me, Ah. Never did get that full DMESG from you.`
`00:06.22`	`[TK]D-Fender`	`lee_is_me, Everything up & running now?`
`00:06.22`	`[hC]`	`[TK]D-Fender: i take it you dont want to share? :P`
`00:06.30`	`JerJer[mobile]`	`has anyone ever pondered a ~good~ way to connect two SIP based conferencing servers? I figured do a if first participant in a specific room, fire a sip connection to the other machine`
`00:06.32`	`[TK]D-Fender`	`[hC], Share what?`
`00:06.35`	`Swat2`	`[TK]D-Fender: cheers, i gotta jet to work, thanks for your help`
`00:06.51`	`[hC]`	`[TK]D-Fender: oh you must have missed my question... here, i'll re send it`
`00:06.53`	`lee_is_me`	`[TK]D-Fender: Yeah, sorry. I got tied up and then you were gone when I got back. Ended up putting another card in the box and it worked fine.`
`00:06.55`	`[hC]`	`I want to try to build a nice web gui around managing a microbrowser based contact directory, and give it back to everyone to use. (if you didnt mind that I did that, as well, of course!)`
`00:06.58`	`[hC]`	`oops`
`00:07.00`	`[hC]`	`wrong one.`
`00:07.04`	`[hC]`	`[TK]D-Fender: hey while ive got you here, could i trouble you for a copy of your polycom microbrowser contact directory scripts?`
`00:07.11`	`lee_is_me`	`yeah, thanks`
`00:07.51`	`[TK]D-Fender`	`[hC], I never did a real PHP one or anything. Just a flat XHTML file with links to dial.`
`00:08.12`	`[TK]D-Fender`	`[hC], use "tel://1234567890" as an HREF to dial.`
`00:08.35`	`[hC]`	`[TK]D-Fender: Ah, well thats a good enough start, so that I could wrap php around it, im looking to try and save some time meddling with making sure everything is formatted correctly on the display`
`00:08.41`	`[TK]D-Fender`	`[hC], the rest I'm sure you could hack up easy depending on what kind of storage method you would prefer to use.`
`00:08.43`	`ohadz`	`hi, i'm trying to get calls in and out working. i've checked with my voip profider and my machine is registering with their iax. i am trying a simple setup calling the did number and have my sip phone ring. exten => s,1,Answer(SIP/888) --- when calling the number i get "the number you have reached is not in service" -- ideas? in the CLI i can't see the call coming in(obviously) .. Thanks.`
`00:09.26`	`[hC]`	`[TK]D-Fender: oh, i didnt realize it was easy as that HREF. I think i have an example here locally for some pre formatted vcard style stuff on a polycom, i'll work with that.`
`00:09.30`	`[hC]`	`Thanks for the info!`
`00:09.32`	`[TK]D-Fender`	`ohadz, turn up IAX2 debug and see what you can see coming in.`
`00:13.49`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`00:14.45`	`*** join/#asterisk Strom_M (n=strom@71-83-175-252.dhcp.mtpk.ca.charter.com)`
`00:20.05`	`kusznir`	`~fxs`
`00:20.07`	`jbot`	`from memory, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx`
`00:20.15`	`ohadz`	`[TK]D-Fender, iax2 debug doesn't give me any errors when i call my did number`
`00:21.27`	`poohbah431111`	`is there better documents then the AsteriskTFOT.pdf?`
`00:23.05`	`JT`	`better in what respect?`
`00:23.26`	`[TK]D-Fender`	`ohadz, pastebin the output then. It should say SOMETHING about the inbound call.`
`00:24.03`	`ohadz`	`[TK]D-Fender, which output -- the one i get when i initiate *?`
`00:24.03`	`[TK]D-Fender`	`poohbah431111, Once you've learned enought from TFOT te get off the ground, you basically pick little bits & pieces off the WIKI.`
`00:24.24`	`[TK]D-Fender`	`ohadz, the CLI output you get while IAX2 debug is enabled and a call tries to come in.`
`00:25.17`	`ohadz`	`[TK]D-Fender, i'm watching the CLI output and there's nothing. nothing changes. i just get the msg on my phone that the number is not in service`
`00:25.45`	`[TK]D-Fender`	`ohadz, Then you have a networking problem.`
`00:26.27`	`[TK]D-Fender`	`ohadz, make sure you've forwarded IAX2 if its not on a public IP, and check your firewall`
`00:27.13`	`ohadz`	`[TK]D-Fender, it's not the firewall. i was able to get to my box previously when using the sample ext.conf and sip.conf`
`00:27.50`	`[TK]D-Fender`	`ohadz, if IAX2 debug sees NOTHING then you have no packets reaching *. End of story.`
`00:29.31`	`ohadz`	`[TK]D-Fender, understood. could there be a different source for packets reaching *? messed up ext.conf file or sip file? when i try to sip show peers i can see that the voip provider is connected and that the number is nat'ing`
`00:29.42`	`ohadz`	`or registered rather then connected`
`00:29.57`	`[TK]D-Fender`	`ohadz, sip.conf & extensions.conf have nothing to do with this.`
`00:30.07`	`Hymie`	`ohadz: use tcpdump to validate where the packets go missing`
`00:30.27`	`Hymie`	`ohadz: make sure they're even coming to your firewall`
`00:30.44`	`[TK]D-Fender`	`Hymie, how you you look at something you're not GETTING? :)`
`00:30.57`	`[TK]D-Fender`	`Hymie, Nothing comes IN to the box.`
`00:31.15`	`Hymie`	`[TK]D-Fender: it's behind a separate firewall, from the sounds of it`
`00:31.32`	`Hymie`	`[TK]D-Fender: so, check to see that they're getting to the firwall, as I said`
`00:31.59`	`Hymie`	`[TK]D-Fender: regardless, trusting asterisk is like trusting a politician. always validate via tcpdump`
`00:32.00`	`ohadz`	`[TK]D-Fender, Hymie - i'm afraid that [TK]D-Fender is right...`
`00:32.03`	`[TK]D-Fender`	`Hymie, seems like you're assuming that this 2nd firewall/gateway is *NUX based and he can run a dump on it...`
`00:32.17`	`Hymie`	`[TK]D-Fender: so you know it isn't?`
`00:32.23`	`Hymie`	`[TK]D-Fender: it upsets you, that I suggested it?`
`00:33.01`	`Hymie`	`[TK]D-Fender: you can easily translate tcpdump into "see what packets you are receiving on your firewall"`
`00:33.08`	`[TK]D-Fender`	`Hymie, How many people do you know running setups like that? Smallest percentage of the population in my experience`
`00:33.14`	`Hymie`	`[TK]D-Fender: and, if he can't, that doesn't make the suggestion a bad one, it only makes his firewall a bad one`
`00:33.31`	`Hymie`	`[TK]D-Fender: so? will his head explode at the suggestion?`
`00:33.33`	`[TK]D-Fender`	`Hymie, I'm referring to a firewall/router seperate from the * box.`
`00:33.52`	`[TK]D-Fender`	`Hymie, How do you run a TCPdump on a D-Link el-cheapo router?`
`00:34.02`	`Hymie`	`[TK]D-Fender: so what? even if it is part of the box, you can tcpdump to see if the packets are being received, but dropped`
`00:34.05`	`rue_mohr`	`darn, I have to fix the gains`
`00:34.11`	`Hymie`	`[TK]D-Fender: what's your point?`
`00:34.27`	`Hymie`	`[TK]D-Fender: are you suggesting that the idea is never mentioned?`
`00:34.27`	`JT`	`easy, you connect your managed switch between the dlink and the rest ofd the network and set a monitor port for you to tcpdump on another machine ;)`
`00:34.43`	`[TK]D-Fender`	`ohadz, What kind of firewall are you running on your * box (figure we'll start asking at the source)`
`00:34.51`	`Hymie`	`[TK]D-Fender: let's all keep quiet, just in case that's not his setup? don't spread ideas, because we might be wrong?`
`00:35.04`	`[TK]D-Fender`	`Hymie, Already there with you :)`
`00:35.05`	`Hymie`	`[TK]D-Fender: quiet all! let's make sure we're never wrong!`
`00:35.35`	`[TK]D-Fender`	`Hymie, I work rather hard not to be :)`
`00:35.59`	`Sean	`[TK]D-Fender hey man can you please help me once again, i need to know of some good ATAs that have worked with Fax`
`00:36.12`	`ohadz`	`[TK]D-Fender, :) there's no firewall on my linux box. the firewall is on my router`
`00:36.34`	`[TK]D-Fender`	`Sean, Fax should be kept as far away from * as possible on a seperate analog line.
`00:36.46`	`Hymie`	`[TK]D-Fender: no, you argue very hard to not be... that doesn't mean you aren't wrong`
`00:36.49`	`[TK]D-Fender`	`ohadz, And are you forwarding IAX2 back to it?`
`00:37.12`	`Hymie`	`[TK]D-Fender: if he can't run tcpdump, SO WHAT. If he can, but hasn't thught of it, it is a valid solution`
`00:39.08`	`Sean	`[TK]D-Fender i plan to but the thing im asking is if you know of a ATA that works well for fax i dont wanna bring asterisk into this i wanna keep it to the ata all by itself`
`00:39.08`	`ohadz`	`[TK]D-Fender, forwarding IAX2 back to my firewall?`
`00:39.08`	`[TK]D-Fender`	`Hymie, True. We each had some small assumptions in there. Mine was based on differnt odds than your scenario, but both were indedd guesses as to what he's running. Which is why we're both stopping guessing and ASKING him :)`
`00:39.09`	`Hymie`	`[TK]D-Fender: anyhow, you shouldn't get so picky, it's not conducive to a solution`
`00:39.09`	`ohadz`	`[TK]D-Fender, Hymie i appreciate the tlc guys. :)`
`00:39.09`	`[TK]D-Fender`	`Hymie, We've BOTH been picky here... so truce, ok?`
`00:39.09`	`Hymie`	`[TK]D-Fender: well, he can tell us ;) TELL US OHADZ BEFORE THE END OF THE WORLD COMES`
`00:39.09`	`ohadz`	`ok. can we please fix my problem?:)`
`00:39.09`	`Hymie`	`[TK]D-Fender: no! no truce!!! the vorlons are attacking an`
`00:39.09`	`[TK]D-Fender`	`ohadz, Maybe!`
`00:39.10`	`Hymie`	`wait`
`00:39.10`	`Hymie`	`sorry, b5 was on`
`00:39.15`	`ohadz`	`guys, stop with this pissing match. lets try to help each other. that's all`
`00:39.28`	`[TK]D-Fender`	`Hymie, "Trust Ivanova, trust yourself... everyone else SHOOT 'EM"`
`00:39.37`	`Hymie`	`heh`
`00:39.59`	`[TK]D-Fender`	`ohadz, so again, have you forwarded IAX2 from your router to your * box?`
`00:40.43`	`ohadz`	`[TK]D-Fender, i believe so. how do i verify ?`
`00:41.12`	`ohadz`	`[TK]D-Fender, it "worked" before. I was able to call and reach my * machine when i had the example conf files in place`
`00:41.22`	`ohadz`	`[TK]D-Fender, i haven't touched the firewall`
`00:41.34`	`[TK]D-Fender`	`ohadz, depends on your router. 4569 UDP -> * .... thats pretty much it.`
`00:41.44`	`ohadz`	`[TK]D-Fender, i don't think that any of the iptable rules have changed`
`00:41.55`	`[TK]D-Fender`	`ohadz, I know, but you were talking SIP before.. this is IAX2 now, running on differnt ports.`
`00:42.27`	`[TK]D-Fender`	`ohadz, perhaps you should be a little pro-active and pastebin your IPTABLES setup etc....`
`00:42.56`	`ohadz`	`[TK]D-Fender, let me just verify on my firewall that all is good`
`00:43.26`	`[TK]D-Fender`	`ohadz, and do share with us so we can double-check for you.`
`00:45.56`	`*** join/#asterisk boch (n=fran@190.48.192.184)`
`00:46.33`	`boch`	`do you know a working java applet sip or iax softphone ?`
`00:47.09`	`ohadz`	`[TK]D-Fender, ok. i can't paste my iptables in pastebin`
`00:47.12`	`ohadz`	`right now`
`00:47.28`	`[TK]D-Fender`	`ohadz, How.... helpful :)`
`00:48.03`	`JT`	`got a 72 port 100Mbit/s managed switch for AUD$100`
`00:48.07`	`ohadz`	`[TK]D-Fender,sorry :/ a number of us are using it. you'll have to trust me that UDP on port 4569 is open to the *`
`00:48.13`	`*** join/#asterisk killown (n=killown@unaffiliated/killown)`
`00:48.16`	`JT`	`it also had 4 fibre ports`
`00:48.54`	`ohadz`	`[TK]D-Fender, i believe that connectivity from and too the gateway/firewall is good. what could be other reasons for the msg "the number you've reached is not in service?"`
`00:50.23`	`[TK]D-Fender`	`ohadz, if you can say that IPTABLES is not in the way and that IAX debug is enabled and you see nothing, then either the server isn't sending calls to your IP (registration failure), or there is a networking issue.`
`00:50.56`	`ohadz`	`[TK]D-Fender, ok. let me restart networking services.. brb`
`00:50.57`	`*** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net)`
`00:51.08`	`Hymie`	`as well, the asterisk box (separate from the firewall) could be dropping packets due to a firewall rule on it`
`00:51.14`	`ariel_`	`hello everyone`
`00:51.54`	`Hymie`	`hmm`
`00:52.14`	`Hymie`	`wasn't he using sip only before? couldn't iax.conf have it set to only listen on a specific IP?`
`00:52.24`	`Hymie`	`not that it would do so by default, but who knows what state that file is in`
`00:52.45`	`Hymie`	`not a very helpful lad though`
`00:52.59`	`*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)`
`00:53.08`	`[TK]D-Fender`	`Hymie, Nope. NOTHING to show us :) We've seen many of his kind in these parts, heh`
`01:00.10`	`*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)`
`01:06.11`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-69-151-213-3.dsl.hstntx.swbell.net)`
`01:06.22`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)`
`01:10.16`	`*** join/#asterisk [BAF64] (i=ferrisr@allegro/botmaster/baf)`
`01:20.51`	`*** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com)`
`01:21.14`	`ohadz`	`[TK]D-Fender, well.. same same..`
`01:21.43`	`*** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net)`
`01:21.46`	`[TK]D-Fender`	`ohadz, Well so far you haven't shown us anything so we've reached the limit of out ability to help you blindly.`
`01:24.11`	`ohadz`	`[TK]D-Fender, i'm fed up for tonight. sorry. thanks for your patience. i'll do this some other time. have a great night. I need to speak with my voip provider and rewrite the sip.conf and extensions.conf files`
`01:24.16`	`ohadz`	`thanks again`
`01:24.28`	`ohadz`	`and sorry for not being more informative`
`01:24.33`	`[TK]D-Fender`	`ohadz, hold`
`01:24.47`	`[TK]D-Fender`	`ohadz, please not that sip.conf and extensions.conf have nothing to do with your IAX2 problems.`
`01:25.13`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`01:25.13`	`[TK]D-Fender`	`ohadz, You could be completely missing a dialplan and at least see the call ATTEMPT to arrive.`
`01:25.35`	`[TK]D-Fender`	`ohadz, So either your iax2.conf is royally b0rked, or your networking is.`
`01:26.10`	`ohadz`	`[TK]D-Fender, understood. i think it might be my voip provider that is not feeding that did to me or something eventhough it registers on my * machine`
`01:26.37`	`[TK]D-Fender`	`ohadz, Perhaps you could pastebin THAT at the very least.... show us the registration is OK.`
`01:26.59`	`ohadz`	`[TK]D-Fender, hmm.. i don't have a iax2.conf only iax.conf under /etc/asterisk :/`
`01:27.13`	`[TK]D-Fender`	`ohadz, yeah that... I don't work with IAX much :)`
`01:27.48`	`ohadz`	`[TK]D-Fender, so what do you want me to show you?`
`01:28.10`	`ohadz`	`[TK]D-Fender, sip show peers? see that my voip provider is connected?`
`01:28.11`	`[TK]D-Fender`	`ohadz, pastebin the register (minus only the password, and * CLI showing that it was usccessful`
`01:28.16`	`ohadz`	`<PROTECTED>`
`01:28.29`	`[TK]D-Fender`	`ohadz, You said your ITSP is using IAX, so why are you still talking about SIP?`
`01:28.47`	`*** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net)`
`01:28.52`	`[TK]D-Fender`	`ohadz, "iax2 show registry"`
`01:28.59`	`ohadz`	`[TK]D-Fender, k`
`01:29.40`	`[TK]D-Fender`	`ohadz, You are really confusing things... like talking about tires when the mechanic is asking you about a radiator problem that just killed your car :)`
`01:29.41`	`ohadz`	`[TK]D-Fender, eek! nothing..`
`01:29.49`	`[TK]D-Fender`	`ohadz, Stay on topic! :)`
`01:30.05`	`DrukenHME`	`[TK]D-Fender: don't you ever take a break man? god damn... think you get paid for the support you hand out here or something`
`01:30.13`	`ohadz`	`<PROTECTED>`
`01:30.23`	`[TK]D-Fender`	`DrukenHME, I do on occasion :)`
`01:30.38`	`[TK]D-Fender`	`DrukenHME, But I do the little stuff for free usually`
`01:31.11`	`ohadz`	`[TK]D-Fender, ok. so what's next?`
`01:31.13`	`[TK]D-Fender`	`ohadz, the fact we don't even see a FAILED attempt tells me your register isn't done right at all (or at all)`
`01:31.19`	`ohadz`	`[TK]D-Fender, did we kill the car yet?`
`01:31.28`	`[TK]D-Fender`	`ohadz, perhaps you should pastebin your iax.conf masing only PW's`
`01:31.38`	`ohadz`	`[TK]D-Fender, ok let me do that`
`01:31.43`	`[TK]D-Fender`	`ohadz, Its already dead, its jsut not getting fixed any faster...`
`01:35.06`	`ohadz`	`[TK]D-Fender, hmm.. http://pastebin.ca/499414`
`01:35.27`	`*** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net)`
`01:35.30`	`ohadz`	`[TK]D-Fender, seems like my itsp is not defined there.. :/`
`01:35.56`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.10)`
`01:36.04`	`ohadz`	`[TK]D-Fender, that is strange... or am i missing something?`
`01:36.10`	`[TK]D-Fender`	`ohadz, Ok, what HAVE you done anyways?!?! :)`
`01:36.30`	`[TK]D-Fender`	`ohadz, And get me the name & number of your supplier!`
`01:36.53`	`[TK]D-Fender`	`ohadz, because whatever you're on... man thats good stuff...`
`01:37.02`	`ohadz`	`[TK]D-Fender, well.. i have installed the * and setup my sip phone.`
`01:37.02`	`[TK]D-Fender`	`ohadz, complete reality disconnect!`
`01:37.07`	`ohadz`	`[TK]D-Fender, :)`
`01:37.40`	`voiper1`	`Hi, Im using asterisk and snom 320/360 phones with all call's going out through ISDN. The issue im having is that on some IVR's the button presses work on some and others it doesn't`
`01:37.42`	`[TK]D-Fender`	`ohadz, and this has... WHAT to do with this mythical IAX ITSP you claim to have set up yet have no trace or clue of? :)`
`01:37.55`	`ohadz`	`[TK]D-Fender, my itsp told me that they setup my reg part on my box.`
`01:38.18`	`[TK]D-Fender`	`ohadz, Way to double-check their work!`
`01:38.23`	`ohadz`	`[TK]D-Fender, maybe my isp is not using IAX?`
`01:38.44`	`[TK]D-Fender`	`ohadz, If you have no clue what was done for you why or how, this is not a good start...`
`01:38.59`	`*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)`
`01:39.20`	`ohadz`	`[TK]D-Fender, i do know that they added a register line to sip.conf and added a context to my extensions.conf`
`01:40.18`	`voiper1`	`I the phones I have set DTMF via SIP INFO to off and the DTMF=rfc2833 on hte server.`
`01:40.28`	`ohadz`	`[TK]D-Fender, the DID was defined in extensions.conf with nat'ng enabled..`
`01:40.29`	`[TK]D-Fender`	`ohadz, ok, so far this has nothing to do with IAX.....`
`01:40.48`	`[TK]D-Fender`	`ohadz, and we've spent a LONG time marking up the wrong tree....`
`01:40.50`	`ohadz`	`[TK]D-Fender, ok. so maybe my provider does not use IAX`
`01:40.58`	`[TK]D-Fender`	`s/marking/barking/`
`01:41.06`	`ohadz`	`[TK]D-Fender, sorry :/`
`01:41.54`	`[TK]D-Fender`	`ohadz, if they are the ones who set this up, perhaps you might want to ask them what they've done since you can't seem to get yourself together about it.`
`01:42.58`	`ohadz`	`[TK]D-Fender, ..i told you what they've done. those are the only changed they caused. and they told me that all i will need to do is setup my dialplan`
`01:43.10`	`*** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal)`
`01:44.33`	`[TK]D-Fender`	`ohadz, Ok, sorry, but it sounds like you're asking to be completely hand-held through setting pretty much everything from scratch which is not something I'm really prepared to just sit here and do right now. Iwas kinda presuming you had a clue about what was going on. Perhaps you should start with the BOOK and run with it for a bit.`
`01:44.35`	`[TK]D-Fender`	`~book`
`01:44.50`	`jbot`	`rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`01:45.31`	`ohadz`	`[TK]D-Fender, yes. i'm readying the book. dialplan chapter.. page 98`
`01:46.09`	`ohadz`	`[TK]D-Fender, wanted to setup the [incoming] context and test it`
`01:46.18`	`ohadz`	`[TK]D-Fender, but alas.. no success..`
`01:46.59`	`voiper1`	`Hi, Im using asterisk and snom 320/360 phones with all call's going out through ISDN. The issue im having is that on some IVR's the button presses work on some and others it doesn't`
`01:47.00`	`ohadz`	`[TK]D-Fender, just that msg "the number you have reached is not in service"`
`01:47.21`	`ohadz`	`[TK]D-Fender, thanks for your help.`
`01:49.01`	`*** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net)`
`01:54.11`	`[TK]D-Fender`	`ohadz, Sorry we've kinda reached my free support limit for tonight`
`01:56.54`	`ohadz`	`[TK]D-Fender, nothing to be sorry about. you've helped me understand that i need to catch up on my readings`
`01:56.58`	`*** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net)`
`01:57.01`	`ohadz`	`[TK]D-Fender, thanks again`
`01:58.02`	`ariel_`	`free support limte....hummm we all have that.`
`02:02.13`	`*** join/#asterisk [shodan] (n=shodan@ip062.96-113-216.pppoe1.joliette.intermonde.net)`
`02:06.01`	`rue_mohr`	`for a channelbank, is it generally a good start to start with 0/0 gain?`
`02:06.18`	`[TK]D-Fender`	`rue_mohr, with any zaptel channel, yes`
`02:06.32`	`rue_mohr`	`this is in the actual channelbank though`
`02:06.43`	`[TK]D-Fender`	`rue_mohr, my statement stands`
`02:07.07`	`rue_mohr`	`some have rx(-3) tx(-2) and some have rx(-7) and tx(0)`
`02:07.28`	`rue_mohr`	`I'm getting about <----------------------------------> that much distortion`
`02:09.31`	`rue_mohr`	`"Invalid value. Valid range is -10.0 to -3.0 in 1 dB steps " :/`
`02:12.42`	`rue_mohr`	`I dont know what to pin down to start adjusting gains against`
`02:14.32`	`JT`	`why adjust them? default`
`02:14.41`	`rue_mohr`	`cause their all over the map right now`
`02:14.48`	`rue_mohr`	`it came that way :)`
`02:15.08`	`JT`	`i see`
`02:15.09`	`rue_mohr`	`heh, how loud should a 1mw be?`
`02:15.29`	`JT`	`voiper1: different dtmf modes/`
`02:18.58`	`J4k3`	`milliwatt is noisy. its fun to call forward your line to it ;)`
`02:19.11`	`J4k3`	`the satisfaction of telemarketers getting their ears hurt.`
`02:19.24`	`rue_mohr`	`I like the tt-busy`
`02:19.26`	`rue_mohr`	`:)`
`02:21.18`	`*** join/#asterisk Strom_M (n=strom@71-83-175-252.dhcp.mtpk.ca.charter.com)`
`02:22.24`	`*** join/#asterisk killown (n=killown@unaffiliated/killown)`
`02:24.57`	`b1shop`	`ok. i have AsteriskNow set up with an extension and a SIP phone. vm to emails not getting delivered.. any idea where i start troubleshooting`
`02:25.27`	`*** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid)`
`02:26.55`	`[TK]D-Fender`	`b1shop, try asking in #asterisknow . This is not an * issue, its a GUI / OS issue`
`02:27.15`	`b1shop`	`ok. didnt know there was a specific can for that`
`02:28.08`	`[TK]D-Fender`	`b1shop, They hid it .... in the CHANNEL TOPIC :)`
`02:28.09`	`b1shop`	`[TK]D-Fender: thanks for the advice earlier... making small progress! ;-)`
`02:28.52`	`b1shop`	`chan dead of course... lol`
`02:29.07`	`[TK]D-Fender`	`b1shop, regrettably you're heading down a troubled road where you'll find that the * GUI does not do everything for you and the bits that it does do are not necessarily supportable here much`
`02:29.56`	`J4k3`	`thirty seven different projects and none work ;)`
`02:30.02`	`b1shop`	`i also set up trixbox... i just want to get a feel for some of this`
`02:30.05`	`J4k3`	`(its the Linux Way of doing things)`
`02:30.13`	`JT`	`~trixbox`
`02:30.29`	`jbot`	`Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org`
`02:30.29`	`b1shop`	`i suppose i should just do centos and start from scratch`
`02:30.40`	`b1shop`	`too sleepy tonioght though`
`02:31.06`	`JT`	`centos, if you like it i guess..`
`02:31.08`	`J4k3`	`b1shop: "properly administrating" asterisk is about as much fun as administrating sendmail.`
`02:31.12`	`[TK]D-Fender`	`JT : Dante here is paving a road ;)`
`02:31.25`	`J4k3`	`personally I find sendmail horribly unacceptable... asterisk just doesn't have competition yet ;)`
`02:31.29`	`JT`	`dante?`
`02:31.34`	`b1shop`	`J4k3: COOL! i actually know sendmail! ;-)`
`02:31.46`	`JT`	`sendmail is crap, no reason to use it`
`02:31.51`	`[TK]D-Fender`	`JT : Dante Alighieri <-`
`02:31.58`	`JT`	`[TK]D-Fender: shrug`
`02:32.21`	`JT`	`never heard of the person`
`02:32.35`	`[TK]D-Fender`	`JT, literarily challenged :)`
`02:32.46`	`[TK]D-Fender`	`JT : "The Divine Comedy"`
`02:32.52`	`JT`	`so this is relevant why?`
`02:33.12`	`[TK]D-Fender`	`JT : Comic releif. But the topic has flown right over your head :)`
`02:33.33`	`JT`	`does that make you feel better?`
`02:34.08`	`[TK]D-Fender`	`JT : Dante made a trip to that firey place down south (WAY south) upon whose gates were writting "Abandon all hope ye who enter". I'm sure you can figure the rest out :)`
`02:34.35`	`JT`	`i see`
`02:34.44`	`[TK]D-Fender`	`JT : basically b1shop here is paving a road to hell one GUI at a time :)`
`02:35.03`	`[TK]D-Fender`	`(or should I consider them simultaneously? :D)`
`02:35.09`	`b1shop`	`heh`
`02:35.15`	`JT`	`hehe`
`02:35.38`	`J4k3`	`thats what we're using until I do something less sucky.`
`02:35.38`	`*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)`
`02:36.06`	`J4k3`	`the people I really wonder about`
`02:36.13`	`J4k3`	`is the ones who run * under vmware....`
`02:36.25`	`J4k3`	`I tried it on a machine that was plenty fast, the experience sucked profusely.`
`02:36.45`	`JT`	`may be useful for testing`
`02:37.05`	`J4k3`	`hiccups like hell for me. bad hiccups`
`02:37.14`	`J4k3`	`of course, I've yet to experience hiccup-free *`
`02:37.29`	`J4k3`	`I think our ethernet may have taken one too many lightning strikes.`
`02:37.42`	`J4k3`	`time to do a mass switch replacement`
`02:40.06`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`02:42.57`	`*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)`
`02:48.18`	`JT`	`J4k3: time to get fibre`
`02:50.06`	`*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)`
`02:53.04`	`Hymie`	`you're a switch replacement, J4k3`
`02:53.10`	`Hymie`	`YOU"RE A SWITCHG REPLACEMENT!!!!!!!!!`
`02:54.47`	`*** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70)`
`02:55.09`	`*** part/#asterisk kclaussen (n=kclausse@204.13.224.242)`
`02:56.17`	`*** join/#asterisk AMUG (n=junky@modemcable105.205-56-74.mc.videotron.ca)`
`03:01.42`	`J4k3`	`JT: yeah, I'm purchasing gig-e fiber cards off ebay tonight. Gotta find a termination method that I can afford.`
`03:01.48`	`J4k3`	`that doesn't require a phd to operate.`
`03:02.23`	`J4k3`	`I've got 1700' of duplex multimode... just gotta spread it out and hook it up`
`03:03.20`	`JT`	`J4k3: well buying pre-terminated patch leads if easiest`
`03:03.51`	`J4k3`	`JT: my first project is ~55 meters long, 3/4" electrical pvc installed already... underground.`
`03:03.52`	`*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`03:04.01`	`J4k3`	`for short jumpers pre-built is the best way`
`03:04.15`	`JT`	`right`
`03:04.21`	`J4k3`	`too many gophers here to consider direct burial without actual filled DB cable.`
`03:04.31`	`JT`	`either you'll need to buy the equipment, or hire someone`
`03:04.35`	`J4k3`	`yeah`
`03:04.51`	`J4k3`	`I'm considering just doing that with turbo-a @ 5.15`
`03:04.53`	`JT`	`you need a fibre stripper, connector oven, and fibre cleaver`
`03:05.13`	`JT`	`i have all of those items except for fibre cleaver`
`03:05.21`	`JT`	`they're expensive diamond tipped things`
`03:05.22`	`*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)`
`03:05.27`	`J4k3`	`connector oven... uv flood?`
`03:05.39`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.10)`
`03:05.42`	`tengulre`	`hi,all`
`03:05.46`	`JT`	`you might be able to do a hack job with an amary board or wet and dry paper to polish the end`
`03:06.03`	`tengulre`	`how to cancel the echo audio?`
`03:06.04`	`JT`	`the oven heats up to set fibre into connectors with epoxy i think`
`03:06.06`	`J4k3`	`yeah`
`03:06.07`	`J4k3`	`ahh`
`03:06.12`	`*** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net)`
`03:06.39`	`JT`	`s/amary/emery/`
`03:09.55`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`03:11.59`	`*** join/#asterisk brussel_ (n=brussel@cpe-72-130-172-213.san.res.rr.com)`
`03:27.17`	`yidiyuehan`	`hello everyone. i have received this error with sip trunk registration.`
`03:27.21`	`yidiyuehan`	`<PROTECTED>`
`03:27.44`	`yidiyuehan`	`any one knows the possible reason as i have googled it and found nothing refer to it`
`03:28.40`	`yidiyuehan`	`this is the debug info: http://www.pastebin.ca/499563`
`03:28.44`	`JT`	`sounds like the other end is not seeing some of the sip packets`
`03:29.59`	`yidiyuehan`	`oh, as i did not make any calls although it's registered. so can i say that it's not an error, and it will not affect the call at all?`
`03:39.28`	`killfill_`	`hey.. how is one suposted to dial using ekiga?.. whatever i type in, asterisk recieves a SIP call request... i.e. sip:xxxxxx`
`03:42.28`	`[TK]D-Fender`	`killfill_, You just enter the #. Thats all. If you've set up your auth properly in Ekiga and have a SIP entry to match in sip.conf, you're ready to go.`
`03:49.55`	`killfill_`	`[TK]D-Fender: oh sure.. SIP->SIP works.. but im trying to call to the pstn...`
`03:50.28`	`*** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net)`
`03:50.34`	`[TK]D-Fender`	`killfill_, No difference. All thats left is dialplan if * is getting the other calls fine.`
`03:50.49`	`[TK]D-Fender`	`killfill_, Which has nothing to do with Ekiga at that point.`
`03:53.53`	`kavit`	`has anyone run asterisk off a VMWare ESX cluster? how was the performance?`
`03:58.29`	`*** join/#asterisk doug437 (n=dougster@c-24-5-184-175.hsd1.ca.comcast.net)`
`03:59.10`	`doug437`	`Here's a dumb question if anyone is awake... how can I pass parameters to AGI scripts?`
`04:00.49`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`04:01.30`	`*** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net)`
`04:01.30`	`doug437`	`hmmm`
`04:01.47`	`doug437`	`oh poo`
`04:01.52`	`Qwell`	`doug437: same as any other app`
`04:03.16`	`doug437`	`Qwell: how's that?`
`04:04.16`	`doug437`	`Qwell: exten => s,n,AGI(script.py/parm1=val1/parm2=val2/) ... like that?`
`04:06.19`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`04:06.32`	`doug437`	`doesn't look like many here are alive and kicking`
`04:09.54`	`rue_mohr`	`exten => _203,1,Dial,Zap/2`
`04:09.54`	`rue_mohr`	`exten => _203,n,Voicemail(u203) there is a shorter way to do that right?`
`04:09.56`	`rue_mohr`	`heh pending I remember the \|30`
`04:10.37`	`*** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp)`
`04:10.37`	`doug437`	`Here's a dumb question if anyone is awake... how can I pass parameters to AGI scripts?`
`04:11.48`	`fujin`	`right, I accidentally registered sip user '023' from my phone. now asterisk is caching it and will not release it`
`04:11.54`	`fujin`	`how can I make it forget that I registerd 023 from my phone`
`04:13.27`	`fujin`	`should registering the extension from another phone be enough?`
`04:13.29`	`fujin`	`it doesn't seem to be working`
`04:14.17`	`doug437`	`i think I can assume you CANT pass args to AGI scripts`
`04:15.44`	`[TK]D-Fender`	`rue_mohr, Shorter than 2 lines... no.`
`04:16.04`	`rue_mohr`	`huh`
`04:16.21`	`rue_mohr`	`oh thats right, kb1's setup was sent to email by the norstar`
`04:16.36`	`rue_mohr`	`thats why it didn't have that`
`04:16.36`	`fujin`	`anyone? can I make asterisk forget a sip peer?`
`04:16.53`	`rue_mohr`	`I dont know fujin`
`04:17.13`	`[TK]D-Fender`	`fujin, comment it out of sip.conf and reload.`
`04:17.25`	`fujin`	`k`
`04:18.26`	`rue_mohr`	`wonder what I should do if nobody picks up the kitchen in 30`
`04:18.32`	`fujin`	`and then if I uncomment it and restart it will it be fine?`
`04:18.35`	`rue_mohr`	`pretty silly to have a kitchen voicemail`
`04:18.40`	`rue_mohr`	`hmm`
`04:19.03`	`[TK]D-Fender`	`fujin, Clearly.`
`04:19.39`	`doug437`	`grrrr`
`04:24.34`	`fujin`	`oh god`
`04:24.39`	`fujin`	`I uncommented it and my phone mysteriously registered it again. this is ridiculous, it must be caching it somewhere`
`04:26.00`	`JT`	`did you do a sip reload?`
`04:26.03`	`fujin`	`yup`
`04:26.17`	`fujin`	`023@blabla somehow got to my 703 phone, which is only registered 703`
`04:26.33`	`doug437`	`how can I pass parameters to AGI scripts?`
`04:27.01`	`fujin`	`i'm going to unplug it entirely and then comment 023, restart *, uncomment it`
`04:27.05`	`fujin`	`then restart it again and plug in my phone`
`04:27.06`	`fujin`	`bloody thing`
`04:29.33`	`fujin`	`heh`
`04:29.39`	`fujin`	`after commenting and restarting`
`04:29.48`	`*** join/#asterisk Nugget (i=nugget@dazed.notslacker.com)`
`04:29.52`	`fujin`	`sip show peer still shows '023' as having the ip address of my phone`
`04:30.47`	`[TK]D-Fender`	`fujin, if you've commented out your sip peer you couldn't do that...`
`04:32.58`	`rue_mohr`	`yay incomming calls work`
`04:33.04`	`rue_mohr`	`and outgoing calls work`
`04:33.28`	`*** join/#asterisk Nugget (i=nugget@dazed.notslacker.com)`
`04:37.30`	`fujin`	`what the hell. with my 192.168.108.22 phone unplugged, asterisk is still seeing the ip address as the registered location of '023'`
`04:37.36`	`fujin`	`there's some bogus stuff going on here.`
`04:37.39`	`fujin`	`is there anyway to flush it at all?`
`04:38.18`	`fujin`	`023/023 192.168.108.22 D 5060 Unmonitored`
`04:41.30`	`fujin`	`oh god`
`04:41.34`	`fujin`	`stupid damn phones, sorry guys`
`04:45.58`	`*** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net)`
`04:48.02`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`04:54.53`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`04:54.53`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`05:07.57`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`05:07.57`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`05:07.59`	`JT`	`lots of deployment...`
`05:08.05`	`drcode`	`I have some qustions about openh323 and Gatekeeper`
`05:08.09`	`drcode`	`any one up?`
`05:08.37`	`kusznir`	`My thoughts would be that we'd have one or more asterisk servers serving up native IP service to IP phones and a few ATAs (where analog is required). DIDs would be provided by channelized T1 or equiv.`
`05:08.51`	`kusznir`	`There would probably be 700-1000 stations deployed.`
`05:10.39`	`JT`	`kusznir: PRI T1 would be best`
`05:10.39`	`kusznir`	`So, first: As this would be a primary phone system in a semi-commertial enviornment, what are the emergency dialing requirements these days? Do we need to provide battery backup for all the phones (either through POE or local backup)?`
`05:10.39`	`drcode`	`is there good open source h323 mcu?`
`05:10.40`	`JT`	`i have no idea what the regulatory framework would be, but you should definitely do power backup`
`05:12.06`	`kusznir`	`Ok. My next question is similar: what are the current e911 strategies (I assume we'll be required to be e911 compliant). Do we have to use a 3rd party service for that, or is it possible to implement internally? If we do it internally, what do we need for a trunk?`
`05:12.23`	`JT`	`no idea`
`05:12.27`	`JT`	`not in the US`
`05:12.29`	`kusznir`	`With respect to power backup, is the predominant method these days PoE?`
`05:12.45`	`JT`	`that would be the smartest idea`
`05:12.59`	`kusznir`	`ok.`
`05:13.04`	`JT`	`also, a big campus would want to have a few analogue lines around the place as backup`
`05:13.36`	`kusznir`	`This would actually be a department within the campus, so we'd be confined more or less to parts of 3 buildings.`
`05:14.01`	`*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)`
`05:14.03`	`kusznir`	`Right now, we are forced to use the campus-run PBX which is over 10 years old, really expensive, and really limiting.`
`05:14.17`	`JT`	`1000 stations is a lot, so you'd still have some analogue backups`
`05:14.28`	`kusznir`	`(and from what I understand, also partially fails in the event of power failures....)`
`05:15.16`	`JT`	`hmm`
`05:15.39`	`kusznir`	`Ok. So speaking of backups, I'd think we'd want more than one asterisk server for redundancy purposes. Does Asterisk and mainstream VoIP lines handle this these days? For example, is it possible to create a cluster of 2 or 3 asterisk servers (ideally located at different places in the network) that would transparently fail over?`
`05:15.53`	`kusznir`	`(the idea being one could go down and the users wouldn't know)`
`05:16.32`	`JT`	`it is possible, as long as you don't want to maintain active calls when a server fails`
`05:16.37`	`JT`	`that's really expensive`
`05:16.49`	`JT`	`and needs carrier gear`
`05:17.19`	`kusznir`	`Yea, if a server fails, the calls it was servicing could fail as well. Just as long as a party could redial and go through using a different server (but without having to do something different to go through the other server)`
`05:17.29`	`JT`	`sure`
`05:17.42`	`JT`	`openser is good for sip clustering`
`05:18.36`	`kusznir`	`cool.`
`05:20.00`	`kusznir`	`Ok. I've been noticing that a lot of sip phones have decent-looking LCD displays (spa-921, for example). Can these be used for server-side address book yet?`
`05:20.02`	`JT`	`on some phones`
`05:20.05`	`JT`	`like polycom and cisco`
`05:21.31`	`kusznir`	`(I suspect this would be hardware-dependant, of course)`
`05:21.31`	`kusznir`	`Does polycom and cisco require you to use their software to do it?`
`05:21.31`	`mosty`	`kusznir, snom support ldap addressbooks, for example`
`05:21.31`	`killown`	`I want learn to configure asterisk which requeriments for do it?`
`05:21.31`	`kusznir`	`cool, that sounds like what I was looking for!`
`05:21.31`	`JT`	`kusznir: polycom has a microbrowser`
`05:21.31`	`JT`	`ciscos can do an xml addressbook iirc`
`05:21.44`	`mosty`	`~book`
`05:21.47`	`jbot`	`from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`05:21.58`	`mosty`	`killown, read that`
`05:22.46`	`killown`	`thanks`
`05:24.51`	`kusznir`	`Is it recommended / worth the effort to set up a dedicated vlan for voip and put all the phones on it, or can they be dropped in the same vlan as the users' computers as long as there's sufficient bandwidth available?`
`05:26.09`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`05:29.31`	`*** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net)`
`05:30.54`	`*** join/#asterisk poohbah431111 (n=rmillis@S010600183950760b.cg.shawcable.net)`
`05:31.18`	`JT`	`kusznir: i'd put them on their own vlan or network`
`05:31.45`	`poohbah431111`	`Can somone help me try and get asterisk working accross the net for a friend to be able to login?`
`05:32.14`	`poohbah431111`	`Im on alocal netowrk with my redhat AS Red Hat Enterprise Linux AS release 4 (Nahant Update 5) VM`
`05:33.18`	`poohbah431111`	`kernel is 2.6.9-55.EL, asterisk-1.2.18, on VMware Server 1.0.3 build-44356`
`05:33.18`	`*** join/#asterisk Trodnevni (n=Pjotrgra@d198-53-196-200.abhsia.telus.net)`
`05:34.57`	`poohbah431111`	`I have the asterisk machine in my DMZ`
`05:35.18`	`poohbah431111`	`we can register his machine but we have no sound when we connect`
`05:36.27`	`poohbah431111`	`The asterisk box is in a DMZ`
`05:36.39`	`poohbah431111`	`Any ideas?`
`05:36.42`	`kusznir`	`poohbah431111: My initial impression is a firewall issue. If you're using SIP, it opens up additional connections for the audio (RTP), and it needs to be able to go bidirectionally.`
`05:36.43`	`mosty`	`are you using sip or iax?`
`05:36.48`	`poohbah431111`	`sip`
`05:37.01`	`mosty`	`did you set externhost in sip.conf?`
`05:37.06`	`poohbah431111`	`he is Trodnevni and loged in here`
`05:37.07`	`mosty`	`or externip ?`
`05:37.25`	`poohbah431111`	`externhost no, externip yes`
`05:37.40`	`JT`	`localnet?`
`05:38.04`	`poohbah431111`	`I am on a localnet. Me and my girlfriend can connect and talk locally`
`05:38.08`	`mosty`	`poohbah431111, run tethereal/tshark and check that the packets have the correct source and destination ip's`
`05:38.16`	`JT`	`localnet=, that is`
`05:38.29`	`ken___`	`exit`
`05:38.32`	`ken___`	`bah`
`05:38.33`	`ken___`	`oops`
`05:38.38`	`poohbah431111`	`localnewt=my local subnet`
`05:39.39`	`poohbah431111`	`so 192.168.0.0?`
`05:39.54`	`poohbah431111`	`tehn reload from the CLI?`
`05:40.20`	`JT`	`that's an ip not a subnet`
`05:40.25`	`JT`	`192.168.0.0/24`
`05:40.56`	`poohbah431111`	`yes that did it`
`05:41.03`	`poohbah431111`	`AWESOME`
`05:41.08`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com)`
`05:41.10`	`poohbah431111`	`we screwedwith it all dam day!!!`
`05:41.25`	`JT`	`nice :)`
`05:42.02`	`*** join/#asterisk ken___ (n=ken@125.212.103.40)`
`05:42.19`	`ken___`	`yo -- i'm having problems with rxfax w/ asterisk 1.4`
`05:42.38`	`JT`	`i didn't think it was maintained for 1.4`
`05:42.47`	`ken___`	`can anyone recommend the "most" stable version of rxfax with 1.4 ?`
`05:43.57`	`ken___`	`JT: hmm`
`05:44.02`	`ken___`	`JT: that's a problem`
`05:44.07`	`JT`	`yes`
`05:44.18`	`JT`	`only works well with certain 1.2 versions`
`05:44.19`	`ken___`	`JT: the stuff i read said it worked with 1.4.2`
`05:44.25`	`JT`	`it's maintained for callweaver`
`05:44.37`	`ken___`	`JT: it compiled fine, even works fine but it core dumps asterisk`
`05:45.02`	`*** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-90-116.w86-206.abo.wanadoo.fr)`
`05:47.07`	`ken___`	`JT: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/`
`05:47.22`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`05:48.19`	`JT`	`it does say test apps`
`05:48.56`	`ken___`	`JT: this is very true -- however i was hoping it'd be "good enough" to just receive faxes ...`
`05:49.00`	`mosty`	`do all isdn services provide callerid on incoming calls?`
`05:49.23`	`JT`	`mosty: in my experience, but it probably depends on provider i guess`
`05:50.38`	`poohbah431111`	`sip and rtp is UDP?`
`05:50.44`	`poohbah431111`	`Or tcp?`
`05:51.00`	`carrar`	`What did google tell you?`
`05:51.16`	`JT`	`udp`
`05:51.23`	`poohbah431111`	`Thank you`
`05:51.28`	`JT`	`mosty: having cid issues?`
`05:51.53`	`mosty`	`JT: yes, not getting callerid on a b104p using chan_misdn`
`05:52.07`	`JT`	`what telco/`
`05:52.26`	`mosty`	`telstra`
`05:52.48`	`JT`	`paying the $5.50/months for Calling Line Identification Presentation?`
`05:52.54`	`JT`	`s/months/month/`
`05:55.02`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`05:55.02`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`05:58.42`	`mosty`	`ahh, punks`
`05:58.59`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`06:00.14`	`JT`	`mosty: they're evil :/`
`06:09.39`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
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`06:53.57`	`BugKhaM`	`anyone knows if I can upgrade to 1.4.4 from 1.2.18 straight away or there should be something I need to learn first?`
`06:56.40`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`06:56.40`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`06:57.02`	`*** join/#asterisk ptiggerdine_ (n=ptiggerd@210.18.216.66)`
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`08:19.58`	`BugKhaM`	`anyone using 1.4.4 by upgrading from 1.2.x here?`
`08:22.30`	`mosty`	`is there a changelog in the source?`
`08:30.04`	`*** join/#asterisk rikstah (n=rick@rhamnett.plus.com)`
`08:37.41`	`*** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com)`
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`08:40.03`	`mkv-`	`Argh, I cannot figure this out.. Why doesnt this work: exten => s,103,Set(name=${system(/bin/curl "http://telefon/nr.php?nr=${CALLERID(number)")})`
`08:40.08`	`mkv-`	`In my extensions.conf`
`08:40.44`	`Strom_M`	`you're missing a closing brace`
`08:40.52`	`Strom_M`	`also, n+101 syntax is highly deprecated`
`08:40.59`	`mkv-`	`Strom_M: Hmm..`
`08:41.16`	`*** join/#asterisk dreamind (n=dreamind@p54A783F8.dip0.t-ipconnect.de)`
`08:41.18`	`dreamind`	`Hi`
`08:41.24`	`file`	`Strom_M: !!!`
`08:41.27`	`Strom_M`	`also, there's a CURL() function AFAIK`
`08:41.28`	`mkv-`	`Strom_M: I see right amount of brace's.`
`08:41.28`	`BugKhaM`	`Strom_M: which version of * are you using?`
`08:41.31`	`Strom_M`	`file!`
`08:41.34`	`Strom_M`	`me?`
`08:41.40`	`Strom_M`	`why does this make a difference?`
`08:42.02`	`dreamind`	`can anybody tell me how long the timeout on Answer() is? - my guess is (based on testing) that it has a 30 second timeout. - Can this value be changed?`
`08:42.13`	`BugKhaM`	`Strom_M: I just wanna upgrade mine to 1.4.4 from 1.2.18`
`08:42.31`	`BugKhaM`	`but unsure if it will fully work`
`08:42.59`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`08:43.06`	`Strom_M`	`well, it will work, depending on how much depreated junk you have in your configurations`
`08:43.08`	`mosty`	`BugKhaM, do you need to upgrade for a specific feature?`
`08:43.26`	`mkv-`	`Strom_M: Oh, now I found the typo. Still it doesnt work. I think the variables gets messed up somewhat.`
`08:43.58`	`dreamind`	`The Answer Timeout is important to me, because I'm trying to implement some sort of predictive dialer, which uses the manager interface and originate to initiate a call.`
`08:43.58`	`dreamind`	`The only point is, the callee is the number I want to call and the destination of that originate is a local queue (so somehow "inverted" to normal usage).`
`08:44.00`	`BugKhaM`	`mosty: yeah, like PlayDTMF in manager api`
`08:44.45`	`mkv-`	`Strom_M: The curl-line works fine. But I'm not sure how to call variables etc in extensions.conf yet.`
`08:45.10`	`Strom_M`	`maybe because the function call is SYSTEM`
`08:45.11`	`Strom_M`	`not system`
`08:45.41`	`mkv-`	`Strom_M: the other system()-calls works fine.`
`08:46.01`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`08:46.08`	`_Raptor_`	`hello, how can i deal in extensions.conf when my sip peer tells me "Busy here" (f.i. say something instead of simply hanging up)?`
`08:46.31`	`mkv-`	`Strom_M: I got system()-calls to get the callers name from internet webpage (thru curl) and putting it into a mysql db.`
`08:46.32`	`Strom_M`	`_Raptor_: gotoif() and the ${DIALSTATUS} variable`
`08:46.43`	`BugKhaM`	`Strom_M: I guess I have some (depreated junks) in my config as I started using it since 1.0.x`
`08:47.42`	`BugKhaM`	`Strom_M: is there any docs I can have a look?`
`08:47.53`	`Strom_M`	`BugKhaM: the UPGRADE files in 1.2 and 1.4 source tree`
`08:48.17`	`_Raptor_`	`Strom_M: thank you for the hint, i will check ${DIALSTATUS}`
`08:49.44`	`*** join/#asterisk hest (n=tonnyjrg@fw.jay.net)`
`08:49.51`	`redax`	`hi,`
`08:50.19`	`*** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)`
`08:50.21`	`redax`	`is DEVSTATE(Custom:somestate) useable on asterisk 1.2.x ?`
`08:50.34`	`hest`	`has anybody experience with spa942 config ? i am trying to do automatic provision of configuraion using dhcp, and so far it works fine, but now i need it to acces vlan 10 and enable vlan in the config, but i cant find the syntax for doing so`
`08:50.44`	`rhombus`	`Using 1.2.18, is there a special way I should be formatting mp3s for music on hold?`
`08:51.20`	`redax`	`hest: get the config from the spa942 like http://your.spa942/admin/spacfg.xml`
`08:51.26`	`redax`	`and look for you vlan entries..`
`08:51.36`	`redax`	`s/you/your/g`
`08:51.50`	`rhombus`	`When I use MusicOnHold(), the files only play for a short time, then it stops and hangs up`
`08:53.21`	`mosty`	`what does the console show, with verbose set to 3?`
`08:54.12`	`rhombus`	`hang on, I'll get it`
`08:54.46`	`rhombus`	`here, let me pastebin is`
`08:54.53`	`rhombus`	`s/is/it/g`
`08:55.18`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`08:55.49`	`rhombus`	`http://pastebin.ca/499875`
`08:56.13`	`dreamind`	`hm, nobody ever tried to do an originate the way I do?`
`08:58.27`	`mosty`	`rhombus, try converting to 8khz mono alaw or ulaw and use that instead`
`08:58.39`	`rhombus`	`mosty: okay`
`08:59.29`	`rhombus`	`hmn. odd. it consistently plays exactly 60 seconds.`
`08:59.34`	`rhombus`	`I've confirmed that now.`
`08:59.37`	`rhombus`	`I'll try a different encoding.`
`08:59.57`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.10)`
`08:59.59`	`tengulre`	`hi,all`
`09:00.47`	`tengulre`	`<PROTECTED>`
`09:02.18`	`tengulre`	`anybody alive?`
`09:02.49`	`rhombus`	`mosty: what's the best way to convert it ulaw?`
`09:02.59`	`rhombus`	`mosty: lame doesn't seem to offer an option`
`09:03.26`	`mosty`	`lame encodes mp3, it doesn't decode`
`09:03.43`	`rhombus`	`I was using lame for transcoding earlier`
`09:03.55`	`*** join/#asterisk jacq (n=jal@203.187.143.130)`
`09:04.25`	`mosty`	`maybe it's a new feature :/`
`09:04.37`	`rhombus`	`can I use mplayer?`
`09:04.42`	`rhombus`	`how would you do it?`
`09:05.39`	`dreamind`	`sox is imho the best tool to convert mp3`
`09:06.05`	`tengulre`	`nobody help me?`
`09:06.37`	`dreamind`	`tengulre: sorry no idea about your problem :(`
`09:06.55`	`dreamind`	`tengulre: I'm only using iax2 between asterisk servers, and sip to connect softphones.`
`09:07.32`	`tengulre`	`dreamind: which softphone are u using?`
`09:07.49`	`tengulre`	`x-lite or self?`
`09:07.50`	`dreamind`	`xmeeting and sjphone.`
`09:08.08`	`tengulre`	`Oh!`
`09:08.14`	`dreamind`	`(xmeeting is osx only)`
`09:08.39`	`tengulre`	`dreamind: but sjphone not have sourcecode?`
`09:08.52`	`dreamind`	`yup its free as free beer.`
`09:09.19`	`dreamind`	`I love to see an alternative, but I don't know one which works flawlessly on Mac OS X, Linux and Windows.`
`09:09.33`	`hest`	`redax: thank you very much :-)`
`09:09.55`	`dreamind`	`tengulre: btw. x-lite is free as free beer too.`
`09:10.13`	`tengulre`	`dreamind: yes! I known.`
`09:13.01`	`hest`	`redax: can i replace group="System/VLAN_Settings" with ua="na" as with the other parameters in my provision config ?`
`09:13.15`	`dreamind`	`hm, I still don't find any way how to change the Answer() Timeout :(`
`09:13.33`	`dreamind`	`or is there any other timeout which is important for the Call made through Originate...?`
`09:13.47`	`mosty`	`dreamind, Answer should return immediately, what are you trying to do?`
`09:14.21`	`dreamind`	`mosty: I'm trying to implement some sort of predictive dialer, or overdialer...`
`09:14.42`	`dreamind`	`mosty: and I thought I could do calls through originate (via the manager interface)`
`09:14.53`	`kieranmullen2`	`http://astguiclient.sourceforge.net/vicidial.html ?`
`09:14.54`	`tengulre`	`what's mean 'i' in Dial synatx?`
`09:15.21`	`dreamind`	`kieranmullen2: no alternative, I need to implement that into our own adress management solution.`
`09:16.21`	`dreamind`	`mosty: and I thought to reverse the order of caller and destination, so the caller is the number being called and the destination is a local queue`
`09:17.03`	`dreamind`	`mosty: this way I could dispatch outgoing calls through the local queue to free agents.`
`09:17.20`	`*** join/#asterisk Fl1p (n=david@port-83-236-235-197.static.qsc.de)`
`09:19.03`	`mosty`	`dreamind: are you trying to delay before answering locally?`
`09:19.12`	`dreamind`	`no`
`09:19.26`	`dreamind`	`the local context uses Answer() as its first application`
`09:20.12`	`redax`	`hest: sure.`
`09:20.25`	`Fl1p`	`hello! Can someone tell me whats the easiest way to call someone with asterisk, and when the call is answered a Sound file will be played which prompts for DTMF then reads the DTMF e.g. 9 and connects to someone`
`09:20.55`	`mosty`	`Fl1p, call files?`
`09:21.01`	`redax`	hest: and you can manual load the xml file like `lynx "http://your.spa942/admin/resync?tftp://your.server/spacfg.xml"
`09:21.15`	`Fl1p`	`mosty : what do you mean ?`
`09:21.45`	`mosty`	`Fl1p, look up asterisk call files on google`
`09:22.08`	`Fl1p`	`mosty : The "voice" says something like : if you like to speak to the person waiting for you push the 9 otherwise hangup`
`09:23.03`	`mosty`	`well yeah, look it up`
`09:23.42`	`Fl1p`	`mosty : ok i'll have a look`
`09:24.00`	`Fl1p`	`mosty: thanks for now! Maybe i ll be back :)`
`09:24.14`	`dreamind`	`just measured using a stopwatch, the answer() timeout (if the called number doesn't pick up) seems to be somewhere around 25 Seconds.`
`09:24.36`	`*** join/#asterisk friedrich\| (n=friedric@e177254097.adsl.alicedsl.de)`
`09:26.05`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`09:31.34`	`hest`	`redax it says it will reload the profile when it is not in use, and then nothing happens ?`
`09:32.03`	`hest`	`redax is there any way to force the reload of the profile ?`
`09:32.55`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.10)`
`09:32.59`	`tengulre`	`hi,all`
`09:33.26`	`tengulre`	`how to cancel the voice echo in iax2 protocol?`
`09:34.11`	`mosty`	`tengulre, you can't. you should do echo cancellation at the analogue-digital boundary`
`09:34.25`	`mosty`	`eg in the zaptel config`
`09:34.28`	`redax`	`hest: it loaded your config from the resync?<URL>`
`09:34.55`	`redax`	`hest: configure the admin/advanced tab: provisoring or what's name`
`09:35.51`	`dreamind`	`mosty: hm, any idea? am I doing something fundamentally wrong?`
`09:36.08`	`rhombus`	`okay, none of these transcodes are working`
`09:36.59`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com)`
`09:37.11`	`rhombus`	`I cleaned the mp3 up so that I don't get that error message anymore`
`09:37.25`	`rhombus`	`and it's just too strange that the musiconhold stops after exactly 60 seconds every time`
`09:37.31`	`rhombus`	`that's got to be something else`
`09:39.09`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`09:40.15`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`09:50.16`	`redax`	`is it possible to change the ALERT_INFO within a ringing ?`
`09:50.36`	`redax`	`either way with a custom Dial() app ...`
`10:05.01`	`*** join/#asterisk Paul_UK (n=foo@email.seatwave.com)`
`10:05.02`	`*** join/#asterisk tuxinator_linux (n=tuxinato@128.187.169.195)`
`10:05.07`	`Paul_UK`	`hey guys, is there a list that specs out what kind of system you'll need per concurrent usage. say for 100 users and 50 concurrent calls, you'll need a P4 2.8 Ghz with 2Gb DDR Ram for example.`
`10:15.30`	`*** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)`
`10:17.00`	`DarKnesS_WolF`	`anyone have problems with snom and callerID ? asterisk 1.2 ?`
`10:17.05`	`DarKnesS_WolF`	`zap callerID i mean`
`10:17.53`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`10:17.58`	`puzzled`	`Paul_UK: it depends if you are going to do transcoding, what other apps run on the box etc. The minumum I deploy for T1/E1 and up is a dual xeon with 2gb and a pair of disks in a raid1 setup`
`10:18.45`	`dreamind`	`mosty: just found out where the problem lies.`
`10:18.59`	`puzzled`	`DarKnesS_WolF: setting it or receiving? Have you confirmed that you are actually receiving callerid from the telco?`
`10:19.02`	`dreamind`	`mosty: the timeout (of 30 seconds) is the one from Originate.`
`10:20.52`	`*** join/#asterisk matsk (n=mk@194.68.102.172)`
`10:32.56`	`Paul_UK`	`puzzled: what is transcoding?`
`10:36.09`	`BrokenNoze`	`Hi can anyone here help me with developing for the asterisk project? I have some additional features I've added to my own source which i'd like to include for the community`
`10:36.27`	`BrokenNoze`	`ls`
`10:40.00`	`creativx`	`ls -a`
`10:44.11`	`BrokenNoze`	`anyone help? i think I have some good additions to functionality would like someone to give me a hand`
`10:44.47`	`creativx`	`BrokenNoze, read on the wiki/asterisk.org how to commit`
`10:45.56`	`*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)`
`10:45.59`	`skirmisha`	`guys`
`10:46.09`	`skirmisha`	`is it possible with API to see current calls`
`10:46.19`	`skirmisha`	`thru api i mean`
`10:46.48`	`BrokenNoze`	`think you can see channels active somehow yeah`
`10:47.15`	`skirmisha`	`what is the command`
`10:47.28`	`BrokenNoze`	`hang on`
`10:48.04`	`*** join/#asterisk sashion (n=djbdsf@196.33.37.1)`
`10:48.32`	`sashion`	`how does one fix this issue: CAS signalling on span 3 conflicts with HDLC with FCS check on channel 38.`
`10:49.00`	`sashion`	`i have a Xorcom 16 port FXS unit and a dual PRI Digium card`
`10:49.28`	`DarKnesS_WolF`	`puzzled: receiving it .. it works with giptel but not snom.`
`10:49.45`	`puzzled`	`Paul_UK: http://en.wikipedia.org/wiki/Transcode`
`10:50.13`	`BrokenNoze`	`think you can use the COMMAND function to get it`
`10:50.19`	`puzzled`	`DarKnesS_WolF: hmm no idea`
`10:51.54`	`BrokenNoze`	`Skirmish : core show channels?`
`10:54.45`	`skirmisha`	`let me see`
`10:55.01`	`BrokenNoze`	`creativx : sorry, where is that? I can't find it.`
`10:55.22`	`creativx`	`BrokenNoze: http://www.voip-info.org/wiki/index.php?page=Asterisk`
`10:58.28`	`BrokenNoze`	`creativx: nothing to do with commiting i can find. i've got the code but apparently I need a certificate etc according to asterisk.org to get involved`
`10:59.20`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`10:59.28`	`creativx`	`well hell, then I dont know`
`10:59.33`	`creativx`	`I just consume the code`
`10:59.43`	`BrokenNoze`	`skirmish: let me know if that works. could use it myslef`
`10:59.54`	`BrokenNoze`	`creativx : ok, thanks.`
`11:04.29`	`sashion`	`never mind`
`11:04.36`	`sashion`	`ztcfg -f fixed the issue`
`11:05.43`	`*** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)`
`11:05.57`	`*** join/#asterisk ccesario (n=ccesario@201-0-124-6.dsl.telesp.net.br)`
`11:06.22`	`*** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it)`
`11:09.17`	`*** join/#asterisk defswork (n=andy@83.105.96.154)`
`11:10.00`	`defswork`	`is there any simple line status display solution for running on PCs ?`
`11:17.20`	`DarKnesS_WolF`	`hmm really i can't get teh callerID to work with snom !`
`11:17.26`	`DarKnesS_WolF`	`receving and i have no idea why !`
`11:17.40`	`DarKnesS_WolF`	`everythng works .. and it works with other phones like GIPtel200`
`11:17.58`	`creativx`	`So.. If I want queue announcements but no music on hold (just ring)... how would I go about and do that!`
`11:19.28`	`sashion`	`creativx: first, setup periodic-announce under queues.conf`
`11:19.41`	`sashion`	`creativx: also setup periodic-announce-frequency`
`11:19.56`	`sashion`	`make sure music is not set`
`11:20.03`	`creativx`	`i have sashion.. but its the music not set part im wondering aobut`
`11:20.10`	`creativx`	`i commented out musiconhold`
`11:20.16`	`creativx`	`but it still tried to do moh`
`11:20.40`	`sashion`	`and then in your dialpla add 'r' to the queue application: Queue(queuename\|rt)`
`11:20.50`	`creativx`	`yeah with 'r' it does not play announcements`
`11:21.08`	`creativx`	`that was the problem :-) heh`
`11:25.14`	`creativx`	`so what do I do... record a ringing sound and use that as MOH??`
`11:30.49`	`BrokenNoze`	`creativx: can't you just just call Ringing()`
`11:31.04`	`BrokenNoze`	`before you enter the queue, then leave MOH off inside the queue?`
`11:31.40`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:32.32`	`mosty`	`is it possible to mangle callerid during an attended transfer?`
`11:34.00`	`*** join/#asterisk octoban (n=paranoid@219.83.106.161)`
`11:35.51`	`creativx`	`BrokenNoze: I am ringing() before queue()`
`11:35.57`	`creativx`	`but how do i leave moh off inside the queue??`
`11:36.04`	`creativx`	`other than commenting out musiconhold`
`11:37.17`	`creativx`	`aha`
`11:37.58`	`sashion`	`music=`
`11:38.05`	`sashion`	`leave it blank`
`11:40.21`	`creativx`	`oh`
`11:41.59`	`creativx`	`let me attempt that`
`11:42.08`	`sashion`	`cool`
`11:42.56`	`creativx`	`let see.. musiconhold= is blank`
`11:43.09`	`creativx`	`but it still does this.. -- Started music on hold, class 'default',`
`11:43.39`	`mosty`	`creativx, noload the moh .so file in modules.conf if you want to disable moh completely`
`11:43.46`	`creativx`	`mkay`
`11:45.53`	`creativx`	`heh no wonder chan_zap didnt work as expected.. it didnt want to load the module`
`11:46.17`	`*** join/#asterisk jhiver (i=jhiver@165-242.206-83.static-ip.oleane.fr)`
`11:46.26`	`sashion`	`creativx: I'm missing here... what version of asterisk you running ?`
`11:46.33`	`sashion`	`and are you using agents ?`
`11:46.36`	`sashion`	`chan_agent`
`11:46.37`	`creativx`	`Connected to Asterisk 1.2.10`
`11:46.54`	`creativx`	`now it worked`
`11:47.01`	`creativx`	`<PROTECTED>`
`11:47.05`	`creativx`	`:)`
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`11:47.18`	`sashion`	`lol a reload app_queue.so would help`
`11:47.20`	`creativx`	`not using chan_agent`
`11:47.26`	`creativx`	`i stopped asterisk`
`11:51.04`	`*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net)`
`12:00.52`	`*** join/#asterisk rgomes1997 (n=chatzill@212.49.223.210)`
`12:02.27`	`rgomes1997`	`Hi guys.. just a quick memory refreshment.... Where do you usually post copied/pasted text and screenshots?`
`12:04.39`	`creativx`	`pastebin`
`12:05.03`	`rgomes1997`	`oh! yes!.. thanks a lot!`
`12:08.17`	`creativx`	`np`
`12:09.23`	`*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net)`
`12:12.11`	`*** join/#asterisk punani (n=m@87.127.7.210)`
`12:12.27`	`*** join/#asterisk djmarin (n=marin@64.9.33.130)`
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`12:14.52`	`creativx`	`sashion.. after the first announce it doesnt resume ringing (as expected..)`
`12:15.04`	`creativx`	`I need some good ideas.. I don't want to torture callers with music`
`12:15.40`	`*** join/#asterisk Gringo_ (n=N3TW4LK3@62.50-245-81.adsl-dyn.isp.belgacom.be)`
`12:15.54`	`Gringo_`	`hey`
`12:15.59`	`Gringo_`	`can someone explain to me:`
`12:16.01`	`Gringo_`	`park:701@parkedcalls`
`12:16.09`	`Gringo_`	`making that hint`
`12:16.26`	`Gringo_`	`for example exten => 90,hint,park:701@parkedcalls`
`12:16.56`	`Gringo_`	`i found that somewhere as being the solution for a BLI-parking lot`
`12:17.40`	`Gringo_`	`but i don't really understand what this line does`
`12:17.58`	`*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)`
`12:18.18`	`Gringo_`	`exten => 50,hint,SIP/22`
`12:18.42`	`Gringo_`	`that i understand: it hints SIP peer 22 when extension 50 is called, sure`
`12:19.07`	`Gringo_`	`but park:701??`
`12:20.30`	`Gringo_`	`ok sorry :) got it :)`
`12:20.32`	`Gringo_`	`;If you want to subscribe to the status of a parking space, this is`
`12:20.32`	`Gringo_`	`;how you do it. Subscribe to extension 6600 in sip, and you will see`
`12:20.32`	`Gringo_`	`;the status of the first parking lot with this extensions' help`
`12:20.33`	`Gringo_`	`;exten => 6600,hint,park:701@parkedcalls`
`12:20.33`	`Gringo_`	`;exten => 6600,1,noop`
`12:20.59`	`Gringo_`	`it's even in the official documentation :$`
`12:22.05`	`*** join/#asterisk penguinFunk (n=penguin@87.224.86.46)`
`12:29.00`	`LeddyHM`	`~paste`
`12:29.24`	`jbot`	`paste is, like, http://rafb.net/paste/`
`12:29.53`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
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`12:33.02`	`*** join/#asterisk pif (n=ldm@zenon.apartia.fr)`
`12:33.22`	`pif`	`hi, does asterisk 1.4.x support multiple sip registrations?`
`12:33.34`	`pif`	`(on the same sip account)`
`12:33.39`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
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`12:42.56`	`*** join/#asterisk berktr (n=canberk@teknopet.com)`
`12:43.00`	`berktr`	`hello everybody!`
`12:45.00`	`iCEBrkr`	`pif: There's better ways to do that`
`12:45.18`	`berktr`	`i am new do asterisk and wondering if somebody can send me his/her extensions.conf`
`12:45.25`	`berktr`	`so that I can have a look at`
`12:45.32`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell)`
`12:45.32`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`12:45.40`	`iCEBrkr`	`berktr: didn't you do a 'make examples' ?`
`12:46.57`	`berktr`	`i have the examples`
`12:47.08`	`berktr`	`however , let me tell you what I need in our company`
`12:47.57`	`berktr`	`a ivr will answer the phone and ask the user which extension he wants to call, then it will forward the call to that extension, and after 45seconds of ringing, if nobody answers it, it will redirect it to voicemail`
`12:48.33`	`berktr`	`and I need my users to be able to hold the call and transfer it to another extension`
`12:51.55`	`*** join/#asterisk shinao1 (n=shinao1@196.3.63.252)`
`12:55.16`	`*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)`
`12:55.36`	`*** join/#asterisk cazze (n=pc@unaffiliated/kammicazze)`
`12:56.24`	`SuPrSluG`	`berktr:most decent phone are able to hold and transfer. so, just write an ivr. plenty of examples out there.`
`12:56.39`	`SuPrSluG`	`~book`
`12:56.51`	`jbot`	`book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`12:57.37`	`*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)`
`13:01.01`	`sashion`	`berktr: check features.conf.. you looking for attended transfer`
`13:01.16`	`sashion`	`that automatically puts the call on hold`
`13:01.55`	`berktr`	`thank you!`
`13:04.51`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`13:08.26`	`berktr`	`sashion, how is this parking works`
`13:08.30`	`berktr`	`let's say i am talking to a customer`
`13:08.36`	`berktr`	`how can i park the conversation?`
`13:08.50`	`berktr`	`shall i just dial i.e 700 to hold the call?`
`13:09.02`	`berktr`	`during conversation?`
`13:10.16`	`creativx`	`yes`
`13:10.26`	`creativx`	`and then 700 tells you "i parked your callee on extension 705"`
`13:10.42`	`creativx`	`then you can run wherever you want to and dial 705 and be connected with your callee`
`13:10.46`	`berktr`	`for example`
`13:11.11`	`berktr`	`during a call, can i set asterisk to park the call when my user pushes to # button of the phone`
`13:11.32`	`berktr`	`and then give the user a dial tone, then make the user call the local extension, then transfer the call`
`13:12.03`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:12.40`	`*** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net)`
`13:13.21`	`coolbeans`	`Hi all. I'm emailing voicemail via asterisk through postfix (ubuntu/debian). Postfix rewrites the outbound email address regardless of how I specify it in my voicemail.conf. Any suggestions?`
`13:14.03`	`creativx`	`berktr: that sounds like blind transfer`
`13:14.10`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`13:14.11`	`mosty`	`coolbeans, rewriting how?`
`13:14.45`	`coolbeans`	`To the fqdn.`
`13:14.57`	`mosty`	`give me an example`
`13:15.08`	`*** join/#asterisk neuwald (n=neuwald@200.199.198.61)`
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`13:17.22`	`coolbeans`	`In my voicemail.conf, fromstring=something@somewhere.foo, but postfix is sending email as the fqdn, i.e., host@some.fqdn.foo`
`13:20.42`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
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`13:24.35`	`coolbeans`	`No clue?`
`13:25.38`	`creativx`	`what does sendmail do coolbeans`
`13:25.58`	`*** join/#asterisk ta^3 (n=tacvbo@189.136.34.99)`
`13:26.20`	`mosty`	`coolbeans, actually, i am not sure what purpose the from string serves- do you need it to be a valid reply address?`
`13:26.40`	`mosty`	`i use exim on my debian asterisk boxes, and we use noreply@fqdn for the from address`
`13:28.34`	`*** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-026b0572937a57cd)`
`13:28.45`	`sashion`	`berktr: you are making something complicated there`
`13:28.53`	`sashion`	`doing an attended transfer is a lot easier`
`13:29.17`	`sashion`	`attended will simply put the caller on hold, and allow you to dial any other extension`
`13:29.33`	`sashion`	`you can then introduce the caller on hold to the person on the other extension`
`13:29.49`	`sashion`	`and when you hangup, the caller on hold will be connected to the new extension`
`13:34.23`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`13:34.23`	`*** mode/#asterisk [+o anthm] by ChanServ`
`13:38.53`	`penguinFunk`	`coolbeans: check the entry in /etc/hosts for your localhost, it may be taking the fqdn from that file there`
`13:38.57`	`*** join/#asterisk saftsack (n=oliver@p54A7FD25.dip.t-dialin.net)`
`13:39.11`	`berktr`	`yes sashion this is what i want to do`
`13:40.05`	`*** join/#asterisk kleiner_hacker (n=Miranda@X0dbc.x.pppool.de)`
`13:40.13`	`*** part/#asterisk kleiner_hacker (n=Miranda@X0dbc.x.pppool.de)`
`13:41.33`	`sashion`	`berktr: Then just enable attended transfer in features.conf`
`13:41.48`	`berktr`	`is it that easy?`
`13:41.50`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`13:41.56`	`*** join/#asterisk oej (n=olle@65-182-39-213.cre.bil.biltmorecommunications.net)`
`13:41.56`	`sashion`	`and restart asterisk... and viola.. providing your phones support DTMF, you won't have a problem`
`13:42.04`	`berktr`	`sounds cool`
`13:42.26`	`sashion`	`you might need to adjust this: featuredigittimeout in features.conf`
`13:42.31`	`berktr`	`and how can i play the music on hold files when they are waiting during attended transfer`
`13:42.37`	`Paul_UK`	`does anyone here have asterisk running in a vmware environment, how is it running? how many clients do you have?`
`13:42.42`	`berktr`	`i mean how can I choose which mp3 to play`
`13:42.44`	`sashion`	`as the default is 500ms and might be to fast for people with slow finger s:P`
`13:42.58`	`mosty`	`i have a minor hassle with atxfer: having to set __TRANSFER_CONTEXT`
`13:43.12`	`berktr`	`:)`
`13:43.28`	`*** join/#asterisk axisys (n=axisys@155.70.141.45)`
`13:43.28`	`mosty`	`Paul_UK, why would you want to run asterisk inside vmware?`
`13:44.12`	`sashion`	`mosty: one only needs that to set __TRANSFER_CONTEXT if you are transfering outside of the current context you are in`
`13:45.04`	`mosty`	`hmm`
`13:45.16`	`berktr`	`do i have to do anything to activate the fax support of asterisk`
`13:45.43`	`mosty`	`berktr, fax support in asterisk is not worth using`
`13:45.52`	`berktr`	`y?`
`13:46.21`	`mosty`	`berktr, because fax over voip doesn't work well, and asterisk doesn't support fax over ip much yet`
`13:46.41`	`berktr`	`fax over ip works well here for long time, but with a private voip company`
`13:46.43`	`sashion`	`T.38 will solve all our delimas :P`
`13:46.55`	`JT`	`delimas?`
`13:46.55`	`mosty`	`sashion, one day, we hope`
`13:46.57`	`berktr`	`they are most probably not using asterisk though`
`13:47.00`	`berktr`	`dilemma?`
`13:47.12`	`sashion`	`yeah ok so I spelt that wrong`
`13:47.14`	`JT`	`berktr: you need to use t.38 for fax`
`13:47.18`	`*** join/#asterisk fourcheeze (n=rich@82.153.23.79)`
`13:47.22`	`berktr`	`means how?`
`13:47.53`	`*** join/#asterisk imesper (n=ian@201-92-86-82.dsl.telesp.net.br)`
`13:47.55`	`mosty`	`berktr, t.38 is a fax over ip protocol which asterisk only supports in pass-through mode currently`
`13:48.14`	`berktr`	`what does pass-through means`
`13:48.32`	`coppice`	`think eating and shitting :-)`
`13:48.38`	`berktr`	`what i want to do is connect my fax machine to my voip network and send faxes to pstn through a termination company`
`13:48.43`	`JT`	`think google`
`13:49.31`	`imesper`	`I am having a problem with incoming call in my IVR, when the caller send dtmf too fast it's lock the channel, no one hear a thing, and the channel keeps locked until I kill asterisk and start again, Using asteriskSVN-branch-1.4-r65123M`
`13:50.07`	`berktr`	`imesper, maybe you should consider switching to 1.2 version, which is much more stable`
`13:51.10`	`imesper`	`It's kinda a hard to go back, since I have some modifications on meetme and a web interface for conference based on 1.4`
`13:51.42`	`berktr`	`then try to switch back to latest stable 1.4, not a svn`
`13:53.27`	`imesper`	`I'll try`
`13:53.40`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`13:54.55`	`*** join/#asterisk deeperror (n=deeperro@mail.banctel.com)`
`13:55.33`	`LeddyHM`	`bigger isn't always better when referring to product version numbers :\|`
`13:59.10`	`sashion`	`imesper: how are you doing IVR? Standard Background() or Read() ?`
`13:59.18`	`*** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru)`
`13:59.20`	`drrt`	`hi`
`13:59.57`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)`
`14:00.46`	`drrt`	`f`
`14:00.47`	`imesper`	`Background()`
`14:00.49`	`drrt`	`Playback(hello)`
`14:00.55`	`sashion`	`imesper: are you setting TIMEOUT?`
`14:01.35`	`imesper`	`Yes,`
`14:01.54`	`sashion`	`can you paste bin CLI output of when this lockup happens ?`
`14:02.06`	`sashion`	`perferrably from beginning of call`
`14:02.07`	`imesper`	`Actually TIMEOUT(digit)=5 and absolute =0`
`14:02.25`	`imesper`	`I can, but I doesn't show anything`
`14:02.31`	`imesper`	`One minute`
`14:03.05`	`sashion`	`with versose = 6 and debug = 5?`
`14:03.16`	`sashion`	`hmmm try without setting absolute`
`14:04.00`	`*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)`
`14:05.32`	`*** join/#asterisk saftsack (n=oliver@p54a7d0ab.dip.t-dialin.net)`
`14:06.01`	`drrt`	`did anybody face with garbage CDRs in the log? they have "NO-ANSWER" in disposition and "s" in dst fields`
`14:06.30`	`mosty`	`drrt, was the call answered?`
`14:06.56`	`imesper`	`I found out that when I call with a inband dtmf it doesn't lock, only with info`
`14:07.11`	`imesper`	`I took off absolute, the same thing`
`14:07.33`	`imesper`	`and zap some times with zap channels`
`14:08.03`	`drrt`	`mosty, no. there are no lastapp at all`
`14:08.27`	`drrt`	`mosty, i cant find out where they come from`
`14:08.43`	`mosty`	`drrt, looks like it came from an unanswered call`
`14:10.02`	`drrt`	`i put such strings to the extension they are: [misc]`
`14:10.03`	`drrt`	`exten => t,1,NoCDR()`
`14:10.03`	`drrt`	`exten => t,n,PlayBack(invalid)`
`14:10.03`	`drrt`	`exten => t,n,Hangup()`
`14:10.03`	`drrt`	`exten => i,1,NoCDR()`
`14:10.06`	`drrt`	`exten => i,n,PlayBack(invalid)`
`14:10.07`	`*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)`
`14:10.08`	`drrt`	`exten => i,n,Hangup()`
`14:10.10`	`drrt`	`exten => s,1,NoCDR()`
`14:10.12`	`drrt`	`exten => s,n,Hangup()`
`14:10.25`	`sashion`	`drrt: ~paste`
`14:10.29`	`sashion`	`~paste`
`14:10.41`	`jbot`	`it has been said that paste is http://rafb.net/paste/`
`14:10.56`	`sashion`	`imesper: I'd suggest upgrading to 1.4.4`
`14:10.57`	`drrt`	`exuse`
`14:11.07`	`*** join/#asterisk Fieldy (i=9kovjdPm@gentoo/contributor/Fieldy)`
`14:11.11`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`14:11.14`	`sashion`	`there have been a couple fixed on dtmf before 1.4.4`
`14:13.44`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`14:13.47`	`*** join/#asterisk Plantseeker (n=Plantsek@83.167.161.28)`
`14:14.15`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`14:15.13`	`drrt`	`for the extension i put to prevent such garbage cdrs. http://rafb.net/p/UNCfds28.html`
`14:15.42`	`coolbeans`	`Is there an easy way to replace "in files" from the linux bash shell?`
`14:15.51`	`coolbeans`	`I need to replace settings.cfg with central.cfg in a bunch of files.`
`14:16.15`	`sashion`	`drrt: have considered using ${DIALSTATUS} ?`
`14:16.19`	`drrt`	`use vi`
`14:16.20`	`deeperror`	`sed/awk`
`14:16.41`	`coolbeans`	`deeperror: Can you give me an example?`
`14:16.56`	`coolbeans`	`Nevermind.. google`
`14:17.05`	`coolbeans`	`Ahh.. easy`
`14:17.17`	`drrt`	`sashion, i ve tried that, but no way. then i move to put NoCDR at "s" extension`
`14:17.34`	`deeperror`	`sed 's/this/that/g' FILE > TMPFILE && mv TMPFILE FILE`
`14:18.00`	`sashion`	`drrt: ok what are you trying to do exactly ?`
`14:18.30`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`14:18.38`	`drrt`	`:%:s/pattern_to_change/target`
`14:18.42`	`drrt`	`using vi`
`14:18.59`	`*** join/#asterisk saftsack (n=oliver@p54a7d8be.dip.t-dialin.net)`
`14:19.18`	`coolbeans`	`Would this get everything in a dir? sed 's/settings.cfg/central.cfg/g' *`
`14:19.41`	`jm\|laptop`	`how do you have two files in the same directory with the same name?`
`14:19.46`	`jm\|laptop`	`oic`
`14:19.46`	`Mercestes`	`drrt: I'm pretty sur ethat's wrong. You have an extra :`
`14:19.58`	`jm\|laptop`	`:% s/match/switch/`
`14:20.26`	`Mercestes`	`s/%:s\/pattern_co_change\/target/%s\/pattern_toOchange\/target/`
`14:20.29`	`jm\|laptop`	`coolbeans: are you trying a mass rename?`
`14:20.38`	`jm\|laptop`	`Mercestes: ARGH`
`14:20.41`	`jm\|laptop`	`my eyes`
`14:20.44`	`coolbeans`	`No. settings.cfg is text WITHIN the file.`
`14:20.56`	`jm\|laptop`	`oh`
`14:21.03`	`jm\|laptop`	`if you're sedding a file to the same file you have to use a switch`
`14:21.13`	`jm\|laptop`	`I've just forgotten which: otherwise you might find you end up with a blank file`
`14:21.32`	`jm\|laptop`	`use vi :)`
`14:21.59`	`Mercestes`	`hehe. Correct a sed statement using sed. so classic`
`14:22.06`	`jm\|laptop`	`:)`
`14:22.34`	`drrt`	`sashion, let me show u this cdrs. http://rafb.net/p/RWAwO695.html`
`14:22.37`	`jm\|laptop`	`it might be -i`
`14:23.32`	`jm\|laptop`	`jamie@jamie:~$ echo foo bar meh > test`
`14:23.32`	`jm\|laptop`	`jamie@jamie:~$ cat test`
`14:23.32`	`jm\|laptop`	`foo bar meh`
`14:23.32`	`jm\|laptop`	`jamie@jamie:~$ sed -i s/bar/bleh/ test`
`14:23.32`	`jm\|laptop`	`jamie@jamie:~$ cat test`
`14:23.35`	`jm\|laptop`	`foo bleh meh`
`14:23.43`	`jm\|laptop`	`<tada>`
`14:23.48`	`drrt`	`Mercestes, i m using the string to change file contents using vim 7.1`
`14:24.15`	`*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)`
`14:24.20`	`coolbeans`	`Will this do it: sed 's/settings.cfg/central.cfg/g' * ??`
`14:24.22`	`Zeeek`	`hey`
`14:24.27`	`sashion`	`drrt: hmmm lots of stuff missing there...`
`14:24.52`	`jm\|laptop`	`coolbeans: I'd still use -i`
`14:25.00`	`jm\|laptop`	`in bash you could go with for`
`14:25.10`	`sashion`	`drrt: can you paste your outcome context ?`
`14:25.28`	`coolbeans`	`Ok. So: sudo sed -i 's/settings.cfg/central.cfg/g' * Should do it?`
`14:25.28`	`drrt`	`sashion, sure`
`14:25.29`	`jm\|laptop`	`for i in ./* do sed -i s/settings\.cfg/central\.cfg/g; done`
`14:25.44`	`jm\|laptop`	`oops`
`14:25.53`	`jm\|laptop`	`for i in ./* do sed -i s/settings\.cfg/central\.cfg/g "$i"; done`
`14:26.07`	`*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)`
`14:26.24`	`sashion`	`jm\|laptop: big oops :)`
`14:26.26`	`coolbeans`	`perl -pi -w -e 's/settings.cfg/central.cfg/g;' *.cfg`
`14:26.31`	`jm\|laptop`	`sashion: shushhh! ;)`
`14:26.41`	`sashion`	`jm\|laptop: hahaha`
`14:27.49`	`drrt`	`sashion, http://rafb.net/p/wgPycJ92.html`
`14:28.25`	`coolbeans`	`Ok, done. Thanks guys/gals.`
`14:28.36`	`jm\|laptop`	`ole!`
`14:29.21`	`drrt`	`sashion, http://rafb.net/p/3Gb07l54.html`
`14:29.32`	`sashion`	`drrt: how does the caller get to SIP/phone86 ?`
`14:30.03`	`drrt`	`sashion, macro-simpledial`
`14:30.53`	`drrt`	`sashion, i pasted it too`
`14:30.56`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`14:32.21`	`sashion`	`drrt: try this`
`14:32.24`	`Paul_UK`	`mosty, running asterix on a vm, is to consolidate server apps, i have a server which really isnt doing much and want to put it on there`
`14:32.32`	`sashion`	`under macro-simpledial, the line exten => s,n,Dial(${ARG1},${ARG2},${ARG3})`
`14:32.48`	`sashion`	`I assume ${ARG3} is your options...`
`14:32.59`	`sashion`	`pass the g option along with ${ARG3}`
`14:33.12`	`drrt`	`let me try`
`14:33.21`	`sashion`	`alternatively, add a n+101,Goto(s-${DIALSTATUS},1)`
`14:33.41`	`*** join/#asterisk rwsr-xr-x (i=maikeru@p5231-ipad12motosinmat.mie.ocn.ne.jp)`
`14:35.24`	`drrt`	`sashion, do u think it skips the strings with actual dialstatus and NoCDR ?`
`14:35.35`	`sashion`	`its a possibility`
`14:36.24`	`sashion`	`cause Dial fails, might try to jump to n+101, then exit cause n+101 doesn't exist`
`14:37.27`	`drrt`	`sashion, perhaps. trying`
`14:39.25`	`*** part/#asterisk Fl1p (n=david@port-83-236-235-197.static.qsc.de)`
`14:39.47`	`*** join/#asterisk \lart (i=foobar@arwen.jasons.org)`
`14:40.37`	`*** join/#asterisk saftsack (n=oliver@p54a7f8f2.dip.t-dialin.net)`
`14:40.37`	`\lart`	`greetings.. anyone have ability to share latest polycom firmware/bootroms? The reseller I purchased from has been utterly useless.. Looking for sip 2.1.1c and bootroom 3.2.3b.`
`14:40.44`	`*** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91)`
`14:40.56`	`axisys`	`anyone here know of a voip provider who is linux freindly.. plus offer BYOD plan ?`
`14:41.30`	`axisys`	`i think there was a guy here works for some company something like shark.. cant remember`
`14:41.35`	`mosty`	`axisys, most voip providers won't care what OS you use`
`14:41.36`	`[[blah]asfd`	`getting an authentication error when i try to dial using iax2. Could anyone look at my stuff and tell me where I am going wrong? kinda new to iax... not used to how it authenticates I guess... Here is the info: http://pastebin.ca/500238`
`14:42.10`	`[[blah]asfd`	`axisys: I use binfone and have been happy with them`
`14:42.18`	`axisys`	`mosty: agreed.. i am looking for this compnay something like shark.. he said they allow BYOD plan`
`14:42.28`	`[[blah]asfd`	`have tried nufone and was not happy`
`14:42.48`	`mosty`	`axisys, just about every provider lets you BYOD`
`14:42.49`	`SwK`	`axisys: are you looking for DIDs or TF/LD`
`14:42.51`	`*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)`
`14:42.52`	`axisys`	`i have been using vonage.. but want to get out of it.. i want to try sipura 3000 w/ a voip provider`
`14:43.02`	`[[blah]asfd`	`binfone will support BYOD as well as they are pay as you go`
`14:43.19`	`axisys`	`mosty: vonage and sunrocket does not.. it has to be their approved device`
`14:43.33`	`SwK`	`axisys: check out the wiki theres a list a mile long there`
`14:43.38`	`SwK`	`wiki?`
`14:43.47`	`[[blah]asfd`	`axisys: are you planning on porting your number from vonage to the new carrier? If so good luck. have been trying to get mine ported since December and Vonage keeps failing the LNP`
`14:43.52`	`axisys`	`SwK: i did.. but that sark company name wasnt there..`
`14:44.12`	`SwK`	`axisys most of the companies there allow byod`
`14:44.19`	`axisys`	`[[blah]asfd: hmm thats not a good news for ,e`
`14:44.23`	`axisys`	`me*`
`14:44.32`	`*** join/#asterisk phpboy (n=shane@196.212.30.186)`
`14:44.54`	`axisys`	`binfone front page looks good..`
`14:45.01`	`[[blah]asfd`	`I have really liked them.`
`14:45.09`	`\lart`	`I suspect if you were to clue the FCC &/or your local board of utilities that vonage won't let you port your number out, that just may light a fire under them..`
`14:45.11`	`SwK`	`if your vonage number is not in a major market they can have a few excuses why its not portable`
`14:45.16`	`phpboy`	`hey all, I'm having trouble recording my click2call calls. it doesn't seem like asterisk is making it a 'true' call in the cdr, why would this be?`
`14:45.22`	`axisys`	`[[blah]asfd: so how long have u been trying to move your vonage to binfone?`
`14:45.25`	`[[blah]asfd`	`just know up front that they do not answer the phone... use their email address and you will get a reply withing 24 hours or less.`
`14:45.39`	`[[blah]asfd`	`axisys: since December 5th 2006`
`14:45.49`	`axisys`	`[[blah]asfd: hmm`
`14:46.01`	`[[blah]asfd`	`The problem is that they are a reseller of other carriers.`
`14:46.03`	`axisys`	`SwK: what u mean by major market?`
`14:46.12`	`[[blah]asfd`	`my number is owned by XO`
`14:46.18`	`[[blah]asfd`	`Vonage is the client of XO`
`14:46.30`	`[[blah]asfd`	`so the CSR does not have any of my personal information on it.`
`14:46.56`	`[[blah]asfd`	`and there is no one at Vonage who is responsible for handling this LNP issue.`
`14:47.02`	`SwK`	`[[blah]asfd, just send them a copy fo the bill`
`14:47.13`	`[[blah]asfd`	`that was the first attempt.`
`14:47.18`	`axisys`	`LNP?`
`14:47.21`	`[[blah]asfd`	`but the name on the CSR record is not mine.`
`14:47.22`	`SwK`	`whatever carrier you are trying to port to is probably screwing up the LNP request`
`14:47.27`	`[[blah]asfd`	`LNP= local number port`
`14:47.29`	`\lart`	`[[blah]asfd: call your local board of utilities. file a complaint. call the fcc too.`
`14:47.44`	`[[blah]asfd`	`yeah... done that.`
`14:47.46`	`SwK`	`the name on the CSR when it comes to VoIP providers is rarely the name of the customer`
`14:48.04`	`[[blah]asfd`	`have tried to port to 3 different carriers thinking that it could possibly be the new carriers fault.`
`14:48.08`	`[[blah]asfd`	`does not seem to be the case.`
`14:48.36`	*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
`14:48.45`	`axisys`	`[[blah]asfd: so i should use sunrocket since they promise to transfer my number`
`14:49.09`	`[[blah]asfd`	`They all promise to transfer the number... The issue lies on the side of Vonage releasing it.`
`14:49.13`	`axisys`	`and then from sunrocket go to binfone.. may be sunrocket will be more helpful?!`
`14:49.34`	`axisys`	`[[blah]asfd: that is really bad!!`
`14:49.46`	`[[blah]asfd`	`Binfone is actually working very hard for me. they have made many phone calls and have done most of the complaints to the regulatory agencies for me.`
`14:49.51`	`axisys`	`that means I am stuck w/ vonage`
`14:50.16`	`[[blah]asfd`	`You may have a different experience. (cross your fingers) that is just what I am going through.`
`14:51.23`	`*** join/#asterisk fbffff (n=fbffff@c-67-167-83-228.hsd1.il.comcast.net)`
`14:51.24`	`[[blah]asfd`	`If you would like, You can talk to the support team about it before you pull the trigger on anything. it is support@binfone.com`
`14:51.29`	`axisys`	`[[blah]asfd: so i dont see the price anywhere`
`14:51.31`	`drrt`	`sashion, cdrs still here but u show me the way i should check. thanks alot`
`14:51.45`	`[[blah]asfd`	`axisys: Were are you at?`
`14:51.53`	`[[blah]asfd`	`where are you located, I should say`
`14:51.56`	`axisys`	`http://www.binfone.com/`
`14:52.09`	`axisys`	`falls church, va`
`14:52.37`	`sashion`	`drrt: did it help ?`
`14:52.38`	`[[blah]asfd`	`most everywhere in the continental US is about 2 cents per min. pretty standard rate for a pay as you go provider`
`14:53.10`	`*** join/#asterisk vopp (i=vopp@70.169.155.226)`
`14:53.33`	`sashion`	`drrt: only a pleasure man... hope you come right.`
`14:53.44`	`axisys`	`[[blah]asfd: hmmm.. they dont have pkg plan.. unlimited US`
`14:53.46`	`axisys`	`?`
`14:53.52`	`[[blah]asfd`	`no`
`14:54.04`	`[[blah]asfd`	`havent seen anything like that with any of the ones I have found.`
`14:54.33`	`[[blah]asfd`	`would anyone have a second to take a peek at my issue? http://pastebin.ca/500238`
`14:54.37`	`vopp`	`anyone willing to help with an now newbie question? the now channel has no responses...`
`14:55.01`	`drrt`	`sashion, no. it doesn't but i ll try to read order of context executing once again )`
`14:55.37`	`*** part/#asterisk joeTSUNAMI (n=joeTSUNA@unaffiliated/joetsunami)`
`14:56.00`	`[[blah]asfd`	`vopp: what is your question.`
`14:57.01`	`vopp`	`:) I am experienced in , but now I'm trying to install now on a CF card to use it in an industrial type appliance... When I download the install image and boot a system with it, it only sees the system's attached disks (SATA disks) and does not see the USB storage device.`
`14:57.20`	`vopp`	`I've tried on several different systems and used several different USB CF readers`
`14:57.57`	`vopp`	`so I can't install to CF`
`14:57.57`	`vopp`	`:(`
`14:57.57`	`Strom_M`	`well, you can use CF, so long as you don't use USB, it seems`
`14:58.34`	`vopp`	`so... does anyone have a suggestion for how to get it to install onto a CF?`
`14:58.52`	`Strom_M`	`use a CF -> IDE adaptor? :)`
`14:59.24`	`vopp`	`so... for sure, then, there is a limitation of the *now that it will only install to IDE devices? Then why does it say it's loading the usb-storage kernel module when it boots? :)`
`14:59.36`	`Strom_M`	`beats me`
`14:59.42`	`Strom_M`	`i dont use asterisknow`
`15:01.43`	`Strom_M`	`im just proposing the most obvious solution to your immediate problem`
`15:01.43`	`vopp`	`think they have an IDE->CF at radio shack?`
`15:01.43`	`Strom_M`	`i doubt it. Fry's Electronics, however...`
`15:01.43`	`vopp`	`:)`
`15:02.51`	`*** part/#asterisk Gringo_ (n=N3TW4LK3@62.50-245-81.adsl-dyn.isp.belgacom.be)`
`15:03.15`	`mosty`	`vopp: is your usb chip supported in your kernel?`
`15:03.38`	`vopp`	`I mean. Honestly, I have no idea, that's why I tried 3 different systems, ... I figured one would have to be...`
`15:03.46`	`vopp`	`:)`
`15:04.03`	`vopp`	`what is the equivalent to see if it sees the device?`
`15:04.06`	`vopp`	`usbstat in bsd`
`15:04.10`	`vopp`	`what is it in linux?`
`15:04.10`	`mosty`	`lsusb`
`15:04.13`	`vopp`	`thx`
`15:04.19`	`vopp`	`checking`
`15:05.57`	`*** join/#asterisk x86_ (n=x86@p3m/member/x86)`
`15:06.19`	`x86_`	`hey guys, what kernel module does a Rhino R4T1 use?`
`15:06.50`	`*** join/#asterisk alrs (n=lars@170.206.224.58)`
`15:06.55`	`vopp`	`jesus, *now doesn't run a virtual/rescue terminal`
`15:11.26`	`*** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk)`
`15:14.33`	`*** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net)`
`15:14.35`	`*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)`
`15:14.39`	`vAd0r`	`what is the default password?`
`15:14.42`	`vAd0r`	`i forgot`
`15:14.46`	`Strom_M`	`cocks`
`15:17.16`	`Strom_M`	`well, you didnt specify what you wanted the default password to`
`15:17.26`	`vAd0r`	`of course it would be root`
`15:17.35`	`vAd0r`	`i downloaded the vm ware image but need to get the ip`
`15:17.55`	`Strom_M`	`the VMware image of /what/?`
`15:18.03`	`vAd0r`	`trixbox`
`15:18.09`	`Strom_M`	`um`
`15:18.12`	`Strom_M`	`this isnt #trixbox`
`15:18.16`	`Strom_M`	`go to #trixbox`
`15:18.21`	`vAd0r`	`i know but it uses asterisk`
`15:18.24`	`Strom_M`	`plzreadtopickthx`
`15:18.31`	`vAd0r`	`so i figured the default password would be the same`
`15:18.52`	`Strom_M`	`asterisk is not linux and has no default password`
`15:19.06`	`Strom_M`	`from topic: -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`15:19.16`	`drrt`	`<PROTECTED>`
`15:25.16`	`*** join/#asterisk saftsack (n=oliver@p54a7d368.dip.t-dialin.net)`
`15:30.24`	`rue_mohr`	`will the line to my phone company be loopstart or something fancy, say dpt ?`
`15:31.29`	`Strom_M`	`most likely loopstart`
`15:31.33`	`rue_mohr`	`cause I know they hang up with a line reversal`
`15:31.40`	`Strom_M`	`oh`
`15:31.47`	`Strom_M`	`hm`
`15:31.54`	`Strom_M`	`do they answer with a polarity reversal?`
`15:32.16`	`rue_mohr`	`phones cant reverse polarity`
`15:32.21`	`rue_mohr`	`hmmm`
`15:32.28`	`Strom_M`	`what do you mean?`
`15:32.43`	`Strom_M`	`you can use a polarity reversal to indicate when the call answers`
`15:32.45`	`Strom_M`	`it's quite common`
`15:32.56`	`rue_mohr`	`... hmm`
`15:33.00`	`rue_mohr`	`I'm not that sure`
`15:33.04`	`Strom_M`	`well, find out.`
`15:33.29`	`rue_mohr`	`yousuggesting I try to get technical information from my phone company?`
`15:33.41`	`rue_mohr`	`though I might know a retired tech`
`15:33.51`	`Strom_M`	`no, i'm suggesting you clip a voltmeter to the line and measure it for yourself`
`15:34.27`	`rue_mohr`	`so your saying that there may be a polarity reversal between the ring tones and the other end answering`
`15:34.52`	`Strom_M`	`no. what i'm saying is that there may be a polarity reversal at the exact moment the other end answers`
`15:34.57`	`rue_mohr`	`or do you mean when the set is picked up, before the dialtone`
`15:35.09`	`Strom_M`	`now what use would the latter option be?`
`15:35.20`	`rue_mohr`	`not sure`
`15:35.28`	`Strom_M`	`come on - brain time`
`15:35.41`	`rue_mohr`	`:) how long are the reversals usually for?`
`15:35.46`	`rue_mohr`	`their pretty short aren't they?`
`15:35.47`	`Strom_M`	`the duration of the call`
`15:35.51`	`rue_mohr`	`OH`
`15:35.55`	`Strom_M`	`cocks.`
`15:36.07`	`rue_mohr`	`hmm`
`15:37.00`	`rue_mohr`	`wheres my volt meter...`
`15:37.04`	`rue_mohr`	`wheres my bix tap...`
`15:37.27`	`*** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net)`
`15:37.34`	`Strom_M`	`anyway, you should hear the reversal`
`15:38.12`	`rue_mohr`	`I wan tot be sure`
`15:38.22`	`rue_mohr`	`where's my bix punch...`
`15:38.22`	`*** join/#asterisk irule (n=irule@189.164.43.19)`
`15:38.48`	`rue_mohr`	`boots...`
`15:39.01`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`15:39.22`	`irule`	`upon * compilation I also executed make progdocs, it supposedly generated some documentation, so, where is it? how can I look at it? thanks`
`15:39.55`	`*** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu)`
`15:40.39`	`b11d`	`.`
`15:43.53`	`*** join/#asterisk jsolares (n=jsolares@216.106.164.144)`
`15:43.53`	`*** part/#asterisk vopp (i=vopp@70.169.155.226)`
`15:45.59`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)`
`15:46.57`	`[TK]D-Fender`	`rue_mohr, You'd be looking to use Kewlstart (fxs_ks)`
`15:48.16`	`*** join/#asterisk saftsack (n=oliver@p54A7DC6D.dip.t-dialin.net)`
`15:49.17`	`Strom_M`	`[TK]D-Fender: not if the answer is indicated by a polarity reversal`
`15:49.37`	`rue_mohr`	`I'm working on that`
`15:50.04`	`rue_mohr`	`slight brain fart as the link between two ports onteh patch panel isn't working, breakfast time`
`15:50.05`	`[TK]D-Fender`	`Strom_M, I'm answering his questions, not yours :)`
`15:50.30`	`*** join/#asterisk tsurko (n=tsurko@77.70.24.142)`
`15:50.43`	`[TK]D-Fender`	`Strom_M, If you decide to start talking about grilled-cheese sandwiches don't go thinking you're the center of attention either ;)`
`15:50.54`	`[TK]D-Fender`	`....mmmmm FOOD`
`15:51.09`	`*** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91)`
`15:51.28`	`rue_mohr`	`:)`
`15:51.36`	`Strom_M`	`shrug whatever.`
`15:52.44`	`Strom_M`	`yum`
`15:52.53`	`Strom_M`	`you're still wrong about the line type though`
`15:53.12`	`rue_mohr`	`port 2 to 7, this isn't hard, white/orange -> red/orange, to the bix strip...odd`
`15:53.35`	`boch`	`do you know what enconding uses iax instead ascii?`
`15:54.00`	`*** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net)`
`15:55.08`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com)`
`15:56.33`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:56.59`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:57.24`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:57.51`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:58.12`	`errr`	`Im having echo issues when calling any number in this 1 city. I spoke to my PRI provider and they told me our PBX needed to have the DB levels changed. Where do you configure this in asterisk?`
`15:58.16`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:58.58`	`Strom_M`	`errr: uh, you shouldnt have to adjust rxgain and txgain on digital entrance facilities...`
`15:59.12`	`*** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net)`
`15:59.22`	`Strom_M`	`but, if you feel the need to screw with those settings, they're in zapata.conf`
`16:00.05`	`rue_mohr`	`hmm bad set of hte bix block, thats explains the noise onteh second set`
`16:00.07`	`errr`	`Strom_M: ok, well I think its our provider because we have asterisk, and Intertel for PBX systems here and both systems have the same echo issue`
`16:01.12`	`Strom_M`	`well it wouldnt be the PRI provider, it would be on the terminating end`
`16:01.25`	`Strom_M`	`if it were the PRI's problem, you'd have echo problems all the time`
`16:01.31`	`errr`	`ah ok`
`16:01.59`	`Strom_M`	`who is your PRI from>?`
`16:02.03`	`Strom_M`	`s/>//`
`16:02.05`	`errr`	`time warner`
`16:02.10`	`Strom_M`	`oh christ`
`16:02.16`	`errr`	`=)`
`16:02.22`	`Strom_M`	`time warner doesn't know ISDN from a hole in the ground`
`16:02.25`	`mosty`	`the company that produces walker texas ranger?`
`16:03.41`	`errr`	`Strom_M: what should the rx and tx gain be in my zapata.conf mine are both 0.0`
`16:03.47`	`*** join/#asterisk Skarmeth (n=Skarmeth@201009092213.user.veloxzone.com.br)`
`16:03.57`	`Strom_M`	`on ISDN? 0.0 :)`
`16:04.53`	`*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)`
`16:05.02`	`errr`	`hmm, I wonder what the best way to resolve this echo issue will be`
`16:05.33`	`Strom_M`	`where are you, and where does the echo seem localized to?`
`16:06.00`	`errr`	`Im in san antonio texas, the echo only happens (that we have found to this point) when calling gonzales texas`
`16:06.14`	`errr`	`and its any number in that city`
`16:06.22`	`Skarmeth`	`hi all, I'm running Asterisk 1.2.18 and my pickup exten stop's working... I have a exten _585XX,1,Pickup(SIP/${EXTEN:1}) extension on my dialplan. I just upgraded from 1.2.12 to 1.2.18 last saturday. I know abour features,conf *8 extension, but it never worked with Polycom phones to me...`
`16:06.23`	`Strom_M`	`are all the numbers localized to the same switch?`
`16:06.30`	`*** join/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com)`
`16:06.36`	`Skarmeth`	`Any idea about the pickup app not working?`
`16:06.41`	`errr`	`Strom_M: I have no idea, how would I find that out?`
`16:06.53`	`Strom_M`	`look in LERG`
`16:06.56`	`errr`	`that city is very small..`
`16:06.59`	`Strom_M`	`or some equivalent`
`16:07.10`	`Strom_M`	`well is it all wireline numbers in that city, or is it mobile phones there too?`
`16:07.11`	`errr`	`sorry but what is lerg?`
`16:07.23`	`Strom_M`	`the telcordia local exchange routing guide`
`16:07.24`	`*** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net)`
`16:07.49`	`rue_mohr`	`the reversal is onyl for half a second, and its only there when the other end hangs up`
`16:07.59`	`Strom_M`	`rue_mohr: odd`
`16:08.13`	`Strom_M`	`see if kewlstart will work, but I doubt it`
`16:08.25`	`errr`	`hmm, I dont know anyone with a cell there. I will call someone to see if I can get a cell number to someone from there.`
`16:08.42`	`Strom_M`	`errr: what kind of t1 interface card are you using?`
`16:08.57`	`errr`	`Strom_M: snagoma a101`
`16:09.06`	`errr`	`sangoma*`
`16:09.29`	`Strom_M`	`have a hardware EC module for it?`
`16:09.43`	`psi0n`	`ok here's my problem: when i call in to my pbx and activate MoH, it's all choppy - except if i make some noise on the cell phone end. so something is causing the MoH to stutter whenever there is no traffic coming from the cell phone. anyone know what's causing this and how to fix it?`
`16:09.58`	`errr`	`Strom_M: I dont think it does. I think it uses software`
`16:10.03`	`Strom_M`	`psi0n: what codec are you using for the trunk into the PBX?`
`16:10.12`	`Strom_M`	`errr: hmm. might be worth trying hardware EC`
`16:10.16`	`PioneerVM2`	`Anyone know if its possible to do the following: Someone calls in to Asterisk to connect to an extension -- put them on some kind of "hold", dial an extension and have festival (or some automated announcement) tell the person a message and allow them to refuse or accept the call -- if they accept, connect the two calls together`
`16:10.30`	`Strom_M`	`PioneerVM2: yes`
`16:10.32`	`psi0n`	`Strom_M: alaw i think, but i'll doublecheck`
`16:10.36`	`psi0n`	`brb`
`16:10.40`	`Strom_M`	`type "show application Dial" at the CLI`
`16:10.41`	`Hmmhesays`	`what up folks`
`16:11.25`	`vAd0r`	`I am trying to get my trixbox to register a trunk w/ my cisco call manager and under asterisk info it says unreachable. I can ping the call manager though`
`16:11.49`	`Strom_M`	`~trixbox`
`16:11.50`	`jbot`	`Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org`
`16:12.09`	`vAd0r`	`crap i meant asterisk not trixbox`
`16:12.15`	`vAd0r`	`different one`
`16:12.23`	`Strom_M`	`you're still using trixbox though`
`16:12.27`	`vAd0r`	`i edited the sip.conf and the extension.conf`
`16:12.31`	`vAd0r`	`no i was earlier`
`16:12.40`	`Strom_M`	`groan`
`16:12.54`	`vAd0r`	`i was testing the differences to see which was better`
`16:12.56`	`psi0n`	`Strom_M: hmm it seems to be converting between alaw and ulaw for some reason.. i'll try to fix that first`
`16:13.16`	`PioneerVM2`	`strom: reading now thx`
`16:14.13`	`frigidzephyr`	`does ChanIsAvail work correctly with sip peers, with ${availstatus} i only get two status numbers return 0 (unknown) whether the peer is on a call, busy, available, and then only time it changes is i get a 3 (busy) when the peer is unregistered`
`16:14.20`	`frigidzephyr`	`oops forgot the ?`
`16:15.42`	`psi0n`	`Strom_M: ok, now it's all alaw, but of course the problem's still there`
`16:18.33`	`frigidzephyr`	`ive tried this on multiple brands of sip phones, also asterisk 1.2.x and 1.4.x`
`16:18.36`	`psi0n`	`how do i check if internal_timing is really enabled in asterisk?`
`16:18.53`	`*** join/#asterisk \BAF64\ (i=ferrisr@allegro/botmaster/baf)`
`16:19.52`	`PioneerVM2`	`strom: do you know which option i should be looking at here`
`16:20.00`	`Strom_M`	`PioneerVM2: the one for call screening`
`16:20.32`	`PioneerVM2`	`hmm it doesnt really say much`
`16:20.37`	`PioneerVM2`	`i figured that was it but it is like 1 line`
`16:21.04`	`Strom_M`	`well, try it`
`16:21.16`	`PioneerVM2`	`lol i dont know how to use it :)`
`16:21.22`	`PioneerVM2`	`i gues i can just put in ok hold`
`16:21.32`	`*** join/#asterisk saftsack (n=oliver@p54a7c917.dip.t-dialin.net)`
`16:24.31`	`PioneerVM2`	`Well P was not exactly what I was looking for lol`
`16:24.34`	`PioneerVM2`	`"Torture" :)`
`16:25.27`	`PioneerVM2`	`strom: I waslooking to potentially not even "pickup" the caller`
`16:25.46`	`PioneerVM2`	`strom: this works but it picks up the call and asks them to say their name -- i want to use my own announcement to the receiving party`
`16:26.09`	`PioneerVM2`	`for example "You are receiving a business call from 123-4567"`
`16:26.20`	`PioneerVM2`	`and then connect the person`
`16:26.20`	`Strom_M`	`um, thats what caller ID is for`
`16:26.23`	`PioneerVM2`	`sort of`
`16:26.30`	`PioneerVM2`	`strom: that wont work for my scenario`
`16:26.40`	`PioneerVM2`	`strom: im looking for a way for end users to share biz and personal phone`
`16:26.48`	`PioneerVM2`	`strom: and they might not recognize the incoming #`
`16:26.51`	`Strom_M`	`distinctive ringing?`
`16:26.56`	`PioneerVM2`	`strom: cell phone`
`16:26.57`	`Strom_M`	`multiple line appearances?`
`16:27.09`	`PioneerVM2`	`not sure what u mean by MLA`
`16:27.20`	`Strom_M`	`hmm`
`16:27.22`	`PioneerVM2`	`im giong to offer a biz zservice to people`
`16:27.33`	`PioneerVM2`	`and im trying to find a way to give them warning its a biz call`
`16:27.41`	`PioneerVM2`	`the only thing i can think of is altering caller ID #`
`16:27.44`	`Strom_M`	`I suppose you'd have to write some AGI script or do some dialplan trickery...`
`16:27.47`	`PioneerVM2`	`like changing last 2 digits to "99" or something`
`16:27.56`	`PioneerVM2`	`yea thats what i was wondering but couldnt find anything on that`
`16:28.05`	`PioneerVM2`	`technically it should be possible but not sure if asterisk can support it`
`16:28.21`	`PioneerVM2`	`i need to spawn off another call so that the incoming and outgoing calls are separate in a way then connect them together`
`16:28.48`	`PioneerVM2`	`maybe this would help -- is there a way to have a program dial a call without having an incoming call first?`
`16:28.57`	`PioneerVM2`	`meaning is there a way to launch an AGI without having to use a phone`
`16:29.17`	`PioneerVM2`	`like going to a shell and typing "call_someone.agi"`
`16:29.27`	`PioneerVM2`	`and it will dial a phone an dplay a recorded msg for example`
`16:29.42`	`PioneerVM2`	`(think wanting to trigger a call based upon an alarm system or something)`
`16:30.09`	`*** join/#asterisk keulin (n=cray@AMontpellier-152-1-25-137.w81-251.abo.wanadoo.fr)`
`16:31.40`	`psi0n`	`could the provider perhaps be using silence suppression on the line or something? and could that be the reason why MoH stutters when theres no voice traffic?`
`16:31.42`	`[TK]D-Fender`	`PioneerVM2, lookup ".call files" and "AMI originate" on the WIKI`
`16:32.03`	`Hmmhesays`	`I like call files better`
`16:32.22`	`PioneerVM2`	`cool thx`
`16:32.26`	`PioneerVM2`	`yea this may solve my problem`
`16:32.36`	`PioneerVM2`	`i could have the AGI spawn a call (fork)`
`16:32.45`	`PioneerVM2`	`and then if i can figure out how to connect or conference them together`
`16:33.04`	`Strom_M`	`psi0n: possibly`
`16:33.16`	`*** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-53-5.pskn.east.verizon.net)`
`16:33.29`	`*** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net)`
`16:34.56`	`PioneerVM2`	`DF -- I see there is a "FollowMe" option in 1.4 to "fork" (create multiple calls) may be what i want`
`16:35.55`	`*** join/#asterisk punani (n=m@87.127.7.210)`
`16:37.19`	`psi0n`	`is there a reason why the internal_timing option isn't enabled by default in asterisk.conf? and is there a way to check if it really is enabled?`
`16:38.05`	`*** join/#asterisk ManOfMilk (n=CpnPlnet@71-210-14-218.eugn.qwest.net)`
`16:42.52`	`*** join/#asterisk jmacz (n=jmacz@201.244.170.3)`
`16:45.21`	`psi0n`	`hmm is this the case of noone knowing the answers, or me asking dumb-ass questions that i should've managed to find the answers to myself? :S`
`16:46.36`	`*** join/#asterisk tecnico (n=tecnico@24.96.146.69)`
`16:49.13`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`16:49.29`	`*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)`
`16:49.39`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`16:52.08`	`[TK]D-Fender`	`psi0n, what "internal_timing" feature?`
`16:53.19`	`psi0n`	`it's supposedly an option you can enable in asterisk.conf`
`16:53.23`	`*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) [NETSPLIT VICTIM]`
`16:53.29`	`*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM]`
`16:53.50`	`[TK]D-Fender`	`psi0n, can you be more precies please.....`
`16:54.11`	`psi0n`	`hmm hang on, i'll provide some urls`
`16:55.07`	`*** join/#asterisk fbffff (n=fbffff@c-67-167-83-228.hsd1.il.comcast.net)`
`16:55.07`	`psi0n`	`http://bugs.digium.com/view.php?id=9357 - a bugreport, referring to the internal_timing option at the end`
`16:56.23`	`psi0n`	`links to http://forums.digium.com/viewtopic.php?p=50753&sid=521b797880d02619d6091477cd90dd76`
`16:57.01`	`[TK]D-Fender`	`psi0n, Ok, never heard of`
`16:57.30`	`psi0n`	`it's pretty new then perhaps?`
`17:00.24`	`lee_is_me`	`has anyone used "p" in Dial command successfully?`
`17:01.11`	`lee_is_me`	`(call screening)`
`17:01.17`	`psi0n`	`gah! how could i have been so blind! that feature was introduced in v1.4, and i'm still using 1.2.16!`
`17:04.53`	`*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net)`
`17:05.49`	`vAd0r`	`can someone help me with asterisk / cisco call manager integration`
`17:14.22`	`*** join/#asterisk JerJer[mobile] (n=jj@45.129.210.2)`
`17:16.02`	`JerJer[mobile]`	`does one need a type=peer entry to make an SRV lookup happen? doing SIP/foo@bob.com doesn't seem to look up SRV for bob.com`
`17:16.32`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com)`
`17:17.00`	`JerJer[mobile]`	`even with srvlookup=yes in general`
`17:20.34`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`17:21.47`	`*** join/#asterisk deeperror (n=deeperro@mail.banctel.com)`
`17:23.56`	`psi0n`	`asterisk's inability to generate outgoing RTP packets when incoming packets stop arriving (due to silence supression) seems to be fixed in versions 1.4+, but i found a patch ( http://bugs.digium.com/view.php?id=5374 ) which supposedly takes care of it. i'm afraid of other things breaking if i update asterisk, but i'm totally new to patching. does anyone have any tips in this matter?`
`17:24.42`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`17:25.27`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-69-151-213-3.dsl.hstntx.swbell.net)`
`17:38.56`	`anonymouz666`	`what did you do guys for a machine who has dynamic external IP and it is behind a NAT? what solution to externip= entry in sip.conf?`
`17:39.17`	`*** join/#asterisk file2 (n=IrcNet@asterisk/developer-and-muffin-lover/file)`
`17:39.17`	`*** mode/#asterisk [+o file2] by ChanServ`
`17:44.43`	`*** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it)`
`17:50.01`	`psi0n`	`i chose the easy fix - get a static ip`
`17:50.33`	`deeperror`	`could write a script ha`
`17:51.25`	`*** join/#asterisk nitrus^ (n=Sarah@cpe-76-166-246-175.socal.res.rr.com)`
`17:51.39`	`*** join/#asterisk killown (n=killown@unaffiliated/killown)`
`17:51.54`	`irule`	`is there some website that compares audio codecs?`
`17:52.14`	`nitrus^`	`is there a major difference between the cisco 7940 and 7960? I'm trying to find a lower cost solution than the 7960 but i want to see what features i'd be losing. they can both integrate with sugarcrm right?`
`17:52.29`	`nitrus^`	`irule yes`
`17:52.32`	`nitrus^`	`let me get you the link`
`17:52.38`	`irule`	`thanks`
`17:52.59`	`nitrus^`	`http://www.voip-info.org/wiki/view/Bandwidth+consumption`
`17:53.04`	`lee_is_me`	`anyone have experience with Aastra's? I'm trying to figure out how to get MWI working on a 9112`
`17:53.13`	`nitrus^`	`that talks about bandwidth consumption and there is another document on relative quality linked in there`
`17:53.37`	`*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)`
`17:53.40`	`irule`	`nitrus^ openning, great, thanks again`
`17:53.50`	`nitrus^`	`np`
`17:54.01`	`nitrus^`	`click the link on the trunking comparison it talks about some quality stuff`
`17:55.14`	`nitrus^`	`i use ilbc and trunking as the other document explains and it requires hardly any bandwidth and the quality is fine`
`17:55.24`	`nitrus^`	`i'd use g729 but i dont want to pay the money`
`17:55.39`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`18:01.17`	`*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)`
`18:02.38`	`*** join/#asterisk mrdigital (n=mrdigita@207-172-228-21.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com)`
`18:04.20`	`*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net)`
`18:04.53`	`mocker`	`Can anyone point me to documentation on MeetMe administration? If I login as a conference admin to a bridge, how do I do things like kick users?`
`18:05.06`	`mocker`	`(w/o having to drop to the CLI or a web GUI)`
`18:05.19`	`*** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com)`
`18:05.45`	`anonymouz666`	`psi0n: the internal IP it can get static but not the external IP it's dynamic`
`18:06.05`	`*** join/#asterisk etfonhomey (n=etfonhom@74-140-209-191.dhcp.insightbb.com)`
`18:06.10`	`anonymouz666`	`unless I use externhost=`
`18:09.01`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@192.117.42.170.static.012.net.il)`
`18:09.03`	`anonymouz666`	`which I don't think it is good to use an hostname in c= parameter SDP`
`18:09.38`	`etfonhomey`	`Anyone use Nuvox for phone service?`
`18:09.57`	`psi0n`	`anonymouz666: i meant getting a static ip from your ISP`
`18:10.26`	`anonymouz666`	`psi0n: I can't change the way that ISP do for million customers`
`18:10.33`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@192.117.42.170.static.012.net.il)`
`18:11.21`	`psi0n`	`anonymouz666: most ISPs give you the option to purchase a static IP-address, afaik anyway`
`18:11.37`	`deeperror`	`how about dyndns?`
`18:11.42`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`18:12.11`	`*** join/#asterisk btsteve (n=travis@204.10.20.30)`
`18:12.40`	`btsteve`	`Hello`
`18:12.55`	`psi0n`	`hi`
`18:13.25`	`btsteve`	`I need some help with asterisk with realtim configuration connecting to odbc database`
`18:15.30`	`*** join/#asterisk tbic (n=tbic@207.148.218.162)`
`18:18.26`	`*** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com)`
`18:18.27`	`tzafrir_laptop`	`btsteve, so pastebin some relevant details about your setup and/or ask some specific questions`
`18:20.33`	`*** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net)`
`18:23.22`	`btsteve`	`I am trying to set up a Asterisk Box to be a voicemail server and it need to have a database backend for dynamicaly adding users to it. I have been trying to setuo unixodbc with no luck. I have a clean centos 5 box with unixodbc installed, when i do ./configure --with-odbc it tells me the odbc is broken`
`18:24.41`	`*** join/#asterisk MindTheGap_ (n=iote@BHE201062133118.res-com.wayinternet.com.br)`
`18:29.37`	`btsteve`	`can anyone help me with a odbc setup on centos 5 for asterisk`
`18:30.55`	`*** join/#asterisk mmartinn (n=martins@128.227.123.22)`
`18:31.26`	`mrdigital`	`anyone familar with sugarcrm`
`18:31.40`	`mmartinn`	`Anyone familiar with features.conf? I'm trying to put in an entry to [applicationmap] like the following, but it won't substitute variables! Here's the line: recordcall => *1,caller,MixMonitor,/mnt/ramdisk/auto-${EPOCH}-${EXTEN}-recording.wav\|b`
`18:31.41`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`18:33.40`	`btsteve`	`<PROTECTED>`
`18:34.00`	`cpm`	`I've looked at sugercrm a bit, hard to get past all the hype and salesdrivel`
`18:34.27`	`mrdigital`	`what hype?`
`18:36.58`	`cpm`	`depending on what you are actually trying to do, egroupware has a functional crm-ish thing, call infolog. works pretty well.`
`18:37.00`	`*** join/#asterisk dasuberdavid (i=david@nat/digium/x-d49f6e06d284052d)`
`18:37.42`	`mrdigital`	`cpm: pm?`
`18:38.00`	`cpm`	`sure`
`18:40.27`	`*** join/#asterisk mattchis (n=mattchis@216.54.143.246)`
`18:42.23`	`*** join/#asterisk jm\|home (n=jm\|home@dilbert.jamiem.com)`
`18:42.58`	`*** part/#asterisk btsteve (n=travis@204.10.20.30)`
`18:43.35`	`drako`	`anyone using a b410p here?`
`18:43.42`	`vAd0r`	`can someone help me with this error. i am calling from my cisco phone system to asterisk. http://pastebin.ca/500770`
`18:44.39`	`*** join/#asterisk berktr (n=canberk@212.156.213.131)`
`18:45.11`	`*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)`
`18:45.25`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-69-151-213-3.dsl.hstntx.swbell.net)`
`18:46.18`	`CoffeeIV`	`I have an asterisk server behind a NAT and on the gateway I am trying to put in iptables rules so an IAX softphone from the internet can connect to asterisk. Do I need to handle anything besides UDP port 4569 ?`
`18:46.45`	`dasuberdavid`	`CoffeeIV: no , that should be it`
`18:47.09`	`berktr`	`port 5060`
`18:47.28`	`dasuberdavid`	`berktr: thats for sip messaging`
`18:47.28`	`CoffeeIV`	`dasuberdavid: thanks`
`18:47.28`	`berktr`	`aha you said iax right`
`18:47.31`	`berktr`	`you need the rtp ports i think`
`18:48.08`	`dasuberdavid`	`for sip you need to forward 5060 and wahtever ports you have set in rtp.conf for your rtp stream`
`18:48.27`	`dasuberdavid`	`for iax2 you only need to forward 4569 because signalling and media are on the same port`
`18:48.46`	`mattchis`	`Has anyone successfully setup T38?`
`18:50.05`	`*** join/#asterisk phillipk (n=pkey@216.248.143.87)`
`18:50.26`	`ManxPower`	`berktr: IAX2 does not use RTP`
`18:50.40`	`ManxPower`	`The only port IAX2 uses is UDP 4569. It uses no other ports.`
`18:50.52`	`*** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com)`
`18:51.39`	`*** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-45-2.pskn.east.verizon.net)`
`18:52.53`	`*** part/#asterisk nitrus^ (n=Sarah@cpe-76-166-246-175.socal.res.rr.com)`
`18:54.31`	`*** part/#asterisk mmartinn (n=martins@128.227.123.22)`
`18:57.02`	`vAd0r`	`what does this error mean? Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)`
`19:02.29`	`PioneerVM2`	`anyone know if the asterisk mailing list is moderated?`
`19:02.47`	`killfill_`	`what does this mean=`
`19:02.49`	`killfill_`	`Detected alarm on channel 1: Yellow Alarm`
`19:02.55`	`killfill_`	`its a te110p`
`19:03.39`	`Strom_M`	`killfill_: it means the other end of the connection stopped receiving data from you`
`19:03.45`	`rikstah`	`tzafrir, ping`
`19:04.44`	`rikstah`	`tzafrir, just responded to your post about the firefox greasemonkey script. You make a good point. If this is being used in an untrusted source, add a password check to the .php script and then pass the password as a param from the .user.js GM script.`
`19:04.47`	`killfill_`	`Strom_M, is this a normal situation?.. i get that at 6am.. noone is using the thing in that hour..`
`19:04.53`	`rikstah`	`tzafrir, unless you have some better suggestion? :)`
`19:18.55`	`*** join/#asterisk lorinc (n=ang@pool-5266.adsl.interware.hu)`
`19:21.31`	`ManxPower`	`vAd0r: That is NOT an error`
`19:23.17`	`*** join/#asterisk santiago (n=santiago@debian/developer/santiago)`
`19:24.47`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`19:27.53`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`19:30.31`	`*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)`
`19:33.24`	`*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net)`
`19:33.49`	`*** join/#asterisk Tamnun (i=Tamnun@80.178.108.198.adsl.012.net.il)`
`19:34.12`	`Tamnun`	`anyone may help with zaptel compiling issue ?`
`19:34.23`	`Tamnun`	`autoconf: no input file`
`19:34.30`	`Tamnun`	`make[3]: *** [config.h] Error 1`
`19:34.36`	`*** part/#asterisk mattchis (n=mattchis@216.54.143.246)`
`19:35.14`	`Tamnun`	`couldn't find any info regarding this on the net.. would appreciate any help with this one`
`19:37.13`	`*** join/#asterisk bill4242 (n=bill@66.60.191.200)`
`19:37.24`	`*** join/#asterisk emiquelito (n=evandro@c934eab7.virtua.com.br)`
`19:38.28`	`bill4242`	`ALL: I need help with outbound calling... I can't seem to get the syntax correct... Trying to set it up so from any analog extension if you dial 9 it will give you access to an outside line..`
`19:40.21`	`_VoiceMeUp_COM`	`hmm can we iax prune reload on IAX ?`
`19:41.25`	`bill4242`	`umm.. I'm not using IAX..`
`19:47.44`	`*** part/#asterisk emiquelito (n=evandro@c934eab7.virtua.com.br)`
`19:51.02`	`*** part/#asterisk deeperror (n=deeperro@mail.banctel.com)`
`19:52.52`	`*** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net)`
`19:52.58`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`20:06.47`	`*** join/#asterisk NirS (i=Nir@87.68.0.156.cable.012.net.il)`
`20:06.52`	`NirS`	`hey all`
`20:06.54`	`NirS`	`anybody home ?`
`20:08.51`	`*** join/#asterisk GuruJee (n=tilde@12.186.161.117)`
`20:08.55`	`anonymouz666`	`this channel is dead today`
`20:09.02`	`GuruJee`	`I need help with DUNDi dialplan`
`20:09.03`	`anonymouz666`	`is holiday in USA?`
`20:09.08`	`GuruJee`	`hello people :)`
`20:09.20`	`GuruJee`	`no its not a holiday`
`20:09.21`	`GuruJee`	`its monday`
`20:09.21`	`*** join/#asterisk techie (n=gus@voip.routedsystems.com)`
`20:09.29`	`GuruJee`	`anony666`
`20:09.46`	`GuruJee`	`so people`
`20:09.49`	`GuruJee`	`geniuses!`
`20:10.02`	`GuruJee`	`I need help in routing DIDs over DUndi`
`20:10.10`	`GuruJee`	`between 2 server`
`20:10.53`	`*** join/#asterisk elriah (n=e@adsl-074-164-217-189.sip.bhm.bellsouth.net)`
`20:10.57`	`GuruJee`	`ok`
`20:11.07`	`GuruJee`	`I have DUNDi over IAX2 trunk`
`20:11.11`	`GuruJee`	`internal calls work great`
`20:11.14`	`GuruJee`	`ivr is working`
`20:11.16`	`elriah`	`Hi all. Has anyone ever seen a Polycom phone (new out of the box) loose it's registration icon by the line label? It's just not there on about 15 of our phones.`
`20:11.36`	`GuruJee`	`elriah -> you need to change its firmware`
`20:11.50`	`GuruJee`	`make sure the sip.ld and sip.cfg are the same versions`
`20:11.57`	`GuruJee`	`for all your phones`
`20:12.09`	`elriah`	`If it's a newer revision, how do I get it to "downgrade"?`
`20:12.11`	`elriah`	`(and thanks)`
`20:12.15`	`GuruJee`	`well`
`20:12.37`	`NirS`	`hey all`
`20:12.50`	`NirS`	`anyone has experience with coupling OpenSER and Asterisk together ?`
`20:12.58`	`GuruJee`	`change in the <mac>.cfg file (which would be something like 0004f2something.cfg) to the appropriate sip.ld`
`20:13.06`	`NirS`	`not for registration purposes, but for load balancing purposes`
`20:13.06`	`GuruJee`	`and sip.cfg`
`20:13.18`	`elriah`	`Thanks. Might just be FTP permissions.`
`20:14.07`	`GuruJee`	`well u can use WinSCP to connect to your asterisk box using the username PlcmSpIp and password PlcmSpIp`
`20:14.08`	`GuruJee`	`for polycom`
`20:14.08`	`GuruJee`	`thats default`
`20:14.08`	`GuruJee`	`i dont know if u have changed it or not`
`20:15.11`	`elriah`	`Everything is right. Nothing has changed. hrmm....`
`20:15.20`	`GuruJee`	`need help with DIDs over IAX2 DUNDi network`
`20:15.26`	`GuruJee`	`helllllllloooooooooo`
`20:15.28`	`GuruJee`	`anyone here`
`20:15.29`	`GuruJee`	`damnit`
`20:15.57`	`*** join/#asterisk clive- (n=pirch@dsl-242-132-117.telkomadsl.co.za)`
`20:17.45`	`clive-`	`hi, anyone got any advice for me wrt itfs toll free access numbers`
`20:19.12`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)`
`20:19.36`	`*** join/#asterisk kfuller (n=kfuller@74.95.33.162)`
`20:20.13`	`kfuller`	`Off the wall question that I'm hoping someone can answer for me: Does anyone know of a SIP-T channel driver for asterisk?`
`20:20.38`	`*** join/#asterisk johann8384 (n=johann83@206.80.75.240)`
`20:22.14`	`elriah`	`Hi all. Has anyone ever seen a Polycom phone (new out of the box) loose it's registration icon by the line label? It's just not there on about 15 of our phones.`
`20:23.26`	`*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)`
`20:23.29`	`Katty`	`allo.`
`20:23.47`	`vAd0r`	`what do people use to plug in their own phone line and use it for outbound local calls?`
`20:23.52`	`vAd0r`	`modem?`
`20:24.37`	`kfuller`	`There are several ways to do that, including FXO cards and some ATA's have FXO ports as well.`
`20:24.51`	`kfuller`	`Unfortunately a modem cannot.`
`20:25.28`	`vAd0r`	`i have a vg200`
`20:25.47`	`vAd0r`	`cisco that i used to use as the same function for cisco call manager`
`20:25.55`	`kfuller`	`Unfortunately, I'm not fammiliar with that one.`
`20:25.59`	`vAd0r`	`is there a way to set that up`
`20:26.10`	`kfuller`	`If its call manager, are you sure it speaks SIP?`
`20:26.45`	`vAd0r`	`im not sure if that device uses sip`
`20:26.48`	`Katty`	`can someone explain to me why a pri is better than a t1 for calls?`
`20:26.59`	`vAd0r`	`because it has a d channel that is always up`
`20:27.12`	`vAd0r`	`quicker call setup and they can pass more information across it`
`20:27.32`	`vAd0r`	`when you say t1 calls you mean did trunks as compared to a pri because a pri is over a t1 as it is just the pathway`
`20:27.48`	`Katty`	`okay slow down`
`20:27.52`	`vAd0r`	`sry`
`20:27.52`	`Katty`	`right now we use analog lines`
`20:28.00`	`Katty`	`i know nothing about t1s and pris (=`
`20:28.06`	`vAd0r`	`are you speaking in general or about asterisk`
`20:28.13`	`Katty`	`except that t1 gives 24 channels, and pri gives 23 channels`
`20:28.21`	`GuruJee`	`hi people`
`20:28.26`	`Katty`	`why should i choose a 23 over a 24?`
`20:28.28`	`vAd0r`	`here is the differnce katty`
`20:28.31`	`GuruJee`	`I need some help regarding DID routing over a DUNDI network`
`20:28.39`	`vAd0r`	`can i pm you so i dont flood the channel`
`20:28.49`	`Katty`	`k`
`20:28.54`	`GuruJee`	`I need some help regarding DID routing over a DUNDI network`
`20:28.55`	`GuruJee`	`I need some help regarding DID routing over a DUNDI network`
`20:29.05`	`GuruJee`	`anyone?`
`20:31.57`	`PioneerVM2`	`Anyone know why caller ID wont work thru my SIP trunk? If i change to IAX it works fine`
`20:32.00`	`PioneerVM2`	`thru SIP it doesnt get sent`
`20:32.10`	`PioneerVM2`	`Using Voicepulse`
`20:32.54`	`GuruJee`	`pioneer`
`20:32.57`	`kfuller`	`Does anyone know of a SIP-T channel driver for asterisk?`
`20:32.59`	`GuruJee`	`sip has some CLI issues`
`20:33.01`	`*** join/#asterisk Fieldy (i=FWWg4ik2@gentoo/contributor/Fieldy)`
`20:33.02`	`GuruJee`	`its the header of the SIP`
`20:33.08`	`GuruJee`	`packet`
`20:33.51`	`GuruJee`	`how can I create a custom-trunk for IAX2 and DUNDi`
`20:33.55`	`GuruJee`	`whats the syntax?`
`20:39.05`	`bill4242`	`i got an easy one.. how the hell do you dial out on an analog interface.. i can't get the dial rules right... i either get a fast busy or it falls back to the default context and gets forwarded to the IVR...`
`20:41.40`	`bill4242`	`... hmm, well i thought it'd be an easy one.. so no one can dial out on their PBX... neat..`
`20:42.12`	`anonymouz666`	`Katty is in da house`
`20:42.13`	`elriah`	`Hi all. Has anyone ever seen a Polycom phone (new out of the box) loose it's registration icon by the line label? It's just not there on about 15 of our phones.`
`20:42.14`	`sbingner`	`bill4242, /me points at Dial(Zap/INTERFACE/NUMBER)`
`20:42.33`	`elriah`	`And furthermore, those same polycome phones don't light up when dialed, but the calls go through...`
`20:43.02`	`*** join/#asterisk rene- (n=rene@200.34.66.137)`
`20:43.06`	`rene-`	`heye`
`20:43.28`	`rene-`	`guys is anyone familiar with cisco catalyt express switches? those that are web based`
`20:43.37`	`rene-`	`and not console based?`
`20:43.37`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`20:43.49`	`sbingner`	`catalyst, and as far as I know they all have a console you can use...`
`20:44.00`	`sbingner`	`so if I'm not right... no ;)`
`20:44.02`	`rene-`	`i have a catalyst express 500`
`20:44.06`	`bill4242`	`OK, but how does asterisk know the number? like how do i capture what is punched in the keypad? is that done through the pattern mattching ?? or..`
`20:44.20`	`sbingner`	`bill4242, read extensions.conf.sample`
`20:44.20`	`rene-`	`it is like cisco with small wheels on the side`
`20:44.25`	`GuruJee`	`hey people`
`20:44.29`	`GuruJee`	`i need some help`
`20:44.44`	`GuruJee`	`I am trying to route DIDs over a DUNDi network over IAX2`
`20:45.16`	`elriah`	`rene: Yep, just configured 3.`
`20:46.29`	`clive-`	`gurujee are any toll free did's?`
`20:47.27`	`rene-`	`elriah: in the initial setup stage, i cannot get to the cisco configuration page`
`20:47.33`	`elriah`	`Using vista?`
`20:47.37`	`rene-`	`a macintosh`
`20:47.43`	`rene-`	`do i have to use windows?`
`20:48.06`	`sbingner`	`elriah, you can't just configure it using a console port if you don't want a gay web interface? that sucks`
`20:48.14`	`GuruJee`	`clive`
`20:48.17`	`elriah`	`That's why. Has to be Win 2000 or XP. Something to do with the way it's DHCP server is giving out addresses.`
`20:48.17`	`GuruJee`	`I have two offices`
`20:48.29`	`GuruJee`	`one office has PRI lines, and all calls and DIDs terminate there`
`20:48.44`	`elriah`	`sbingner: Yep. It does. It works great if you follow the instructions. I just spend 2 hours trying to get one configured using Vista. Did it with XP (per the instructions) in about 30 seconds.`
`20:48.45`	`GuruJee`	`the other server is in a different city and I just need to route my DIDs to that server in the different city`
`20:49.27`	`elriah`	`Anyone ever seen where a polycom phone will receive calls but the light doesn't blink? And also the registration icon next to the line label is just not there. weird.`
`20:49.38`	`GuruJee`	`yeah, elriah`
`20:49.42`	`elriah`	`sbingner: But the switched is very powerful when used in voice applications.`
`20:49.44`	`ManxPower`	`eltech: Yes.`
`20:49.47`	`GuruJee`	`i have a whole fleet of 400 polycoms`
`20:49.54`	`sbingner`	`elriah, I bet a Cat 3550 would do as well`
`20:49.58`	`GuruJee`	`its a bad sip.cfg file`
`20:50.06`	`elriah`	`sbingner: Sure would, at $5k more. lol`
`20:50.17`	`sbingner`	`worth it if that crap makes you have a web interface`
`20:50.18`	`bill4242`	`OK I think I got that the rule right... exten => _9XXXXXXX,1,Dial(ZAP/5/${ARG1})`
`20:50.21`	`clive-`	`gurujee, ok:)\`
`20:50.23`	`rene-`	`cisco 500 is like ~2500`
`20:50.24`	`elriah`	`GuruJee: It's the same one that works with about 100 of our other phones, it's just these 15 phones at a remote site...`
`20:50.29`	`ManxPower`	`elriah: you are using the newer polycom models with .cfg files that don't have the info for those phones in them`
`20:50.33`	`bill4242`	`but, now i get a zt_set_hook error.`
`20:50.36`	`elriah`	`A PoE 500 is 1900, a non-poe is only $500.`
`20:50.37`	`GuruJee`	`hey bill424 is that for the DID`
`20:50.40`	`GuruJee`	`under what context?`
`20:51.02`	`GuruJee`	`hey clive`
`20:51.08`	`GuruJee`	`so any idea how that will be done?`
`20:51.09`	`elriah`	`GuruJee: They all use the same bootrom.ld and sip.ld ...`
`20:51.11`	`bill4242`	`under [local] which is included by [outboud] which is set ast the default context for Zap1,2,3,4`
`20:51.16`	`aptura`	`ManxPower I had a issue wiring up a t1 extention to a off the shelf rj45 jack. What jacks do you use? amp?`
`20:51.33`	`GuruJee`	`elriah, what polycoms u have ? 301 , 501 or 601?`
`20:51.37`	`ManxPower`	`aptura: I have no idea what you just said.`
`20:51.39`	`elriah`	`501's.`
`20:51.44`	`GuruJee`	`hmm`
`20:51.56`	`ManxPower`	`elriah: at the remote site too?`
`20:51.57`	`elriah`	`I've deployed a ton of them.. not my first rodeo so to speak but I've never seen this...`
`20:52.07`	`elriah`	`ManxPower: At a bunch of remote sites...`
`20:52.15`	`GuruJee`	`so basically ur problem is that for some phones u see the extension for the other phones u just see something as "IP ..."`
`20:52.15`	`GuruJee`	`?`
`20:52.25`	`mcab`	`1st rule of Polycoms - Thou shalt always use the right config file for the app you're using`
`20:52.26`	`elriah`	`GuruJee: Nope, see the extention (label) no problem.`
`20:52.26`	`*** join/#asterisk fujin (n=aj@unaffiliated/fujin)`
`20:52.34`	`elriah`	`mcab: I am.`
`20:52.36`	`ManxPower`	`elriah: I've only seen the issue with 430 phones config files that do not have the 430 settings in the,`
`20:52.49`	`elriah`	`hrm...`
`20:52.58`	`JerJer[mobile]`	`does one need a type=peer entry to make an SRV lookup happen? doing SIP/foo@bob.com doesn't seem to look up SRV for bob.com - even with srvlookup==yes in general`
`20:52.58`	`aptura`	`Manx basiscly I am not to happy with these rj45 jacks used to end the T1 extention. The telco guy did not end the demark in the wiring closet so I had to extend the demark into the closet but the jacks design was not what I expected.`
`20:53.01`	`ManxPower`	`I assume any of the non x00, x01 phones would have the same problem`
`20:53.10`	`mcab`	`elriah: well, it really sounds like you're not :-)`
`20:53.26`	`mcab`	`but I'll actually read some scrollback and see what your problem is :-D`
`20:53.26`	`ManxPower`	`JerJer[mobile]: I think you only need one =`
`20:53.47`	`fujin`	`how stable is the 1.4 branch?`
`20:53.51`	`elriah`	`Bootrom 3.2.2.0019`
`20:53.59`	`JerJer[mobile]`	`typo`
`20:54.01`	`JerJer[mobile]`	`into irc`
`20:54.08`	`*** join/#asterisk brussel (n=brussel@cpe-72-130-172-229.san.res.rr.com)`
`20:54.52`	`ManxPower`	`JerJer[mobile]: are you sure bob.com has a valid srv record?`
`20:54.53`	`JerJer[mobile]`	`i am running 1.2`
`20:54.57`	`JerJer[mobile]`	`yes`
`20:55.07`	`aptura`	`ManxPower what t1 extentions jacks do you use ?`
`20:55.13`	`ManxPower`	`aptura: I use the exact same plugs and outlets I use for ethernet`
`20:55.18`	`aptura`	`mmm`
`20:55.31`	`mcab`	`elriah: what SIP version are you running?`
`20:55.49`	`elriah`	`mcab: 2.10.2708`
`20:56.41`	`mcab`	`elriah: what are you using for a bootserver?`
`20:56.44`	`JerJer[mobile]`	`in fact we get: chan_sip.c:1989 create_addr: No such host`
`20:57.05`	`JerJer[mobile]`	`yet i can lookup that dns name manually on the same machine`
`20:57.13`	`JerJer[mobile]`	`with or without srv`
`20:57.14`	`elriah`	`mcab: vsftpd`
`20:58.10`	`mcab`	`elriah: and the phones are downloading the sip.cfg file OK?`
`20:58.13`	`fujin`	`this bloody contractor has used 1.4`
`20:58.40`	`mcab`	`(sorry, not trying to belabour a point, but the symptoms you're describing scream missing/incorrect sip.cfg)`
`20:58.53`	`elriah`	`mcab: Yep.`
`21:00.06`	`mcab`	`elriah: is it only on a few of the phones you're deploying, or is it consistent?`
`21:02.16`	`elriah`	`Just the 15 new ones at a specific site but it is all of them and they are right out of the box.`
`21:02.19`	`elriah`	`(new)`
`21:02.38`	`JerJer[mobile]`	`it does appear that one needs a type=peer entry to do SRV lookups`
`21:02.42`	`JerJer[mobile]`	`which totally sucks ass`
`21:03.18`	`ManxPower`	`JerJer[mobile]: Seems logical`
`21:04.08`	`JerJer[mobile]`	`why? i feed a dns name and if srvlookup does equal yes, it should attempt to lookup srv`
`21:04.10`	`ManxPower`	`elriah: Did you physically handle those phones or even seen the new phones?`
`21:04.26`	`*** join/#asterisk r3n4n (n=renan@200.210.51.130)`
`21:04.39`	`ManxPower`	`JerJer[mobile]: because calls are sent to peers and calls are received from users. Seems silly to support SRV for INCOMING calls.`
`21:04.46`	`elriah`	`Yep.`
`21:04.50`	`GuruJee`	`Need help with DUNi !! anyone help!!`
`21:04.52`	`elriah`	`It's a weird issue.`
`21:04.58`	`elriah`	`Firewall maybe stripping off something?`
`21:05.06`	`elriah`	`Like I said, they place and receive calls normally.`
`21:05.10`	`ManxPower`	`You know all this. Or maybe you actually believe Olle when he says there is no longer a difference between user and peer?`
`21:05.18`	`elriah`	`But why the line label 'registration' icon is gone is beyond me.`
`21:05.42`	`*** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net)`
`21:05.53`	`ManxPower`	`I like Olle. He is a smart guy, but this whole peer and user are the same thing is crap.`
`21:06.56`	`*** join/#asterisk [hC] (n=hardcore@66.119.167.162)`
`21:07.15`	`JerJer[mobile]`	`how am i doing an incoming call?`
`21:07.34`	`ManxPower`	`JerJer[mobile]: If your type= is not peer then what is it?`
`21:07.47`	`elriah`	`Hey [TK]D-Fender, you in channel?`
`21:07.51`	`JerJer[mobile]`	`Dial => Something,1,Dial,SIP/$EXTEN}@SOMEDOMAINANME.COM`
`21:07.59`	`JerJer[mobile]`	`+ {`
`21:08.33`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`21:08.35`	`diclophis-work`	`hello all`
`21:08.46`	`diclophis-work`	`whats the context for calls coming in from a s SIP connection?`
`21:08.47`	`ManxPower`	`JerJer[mobile]: Sorry, I never do that.`
`21:09.02`	`diclophis-work`	`i got box a registered to box b, i am calling from box b, but i get a all circuits busy error`
`21:09.04`	`ManxPower`	`But I can see how dialing by IP would make asterisk ignore sip.conf`
`21:09.09`	`diclophis-work`	`the registry shows up fine`
`21:09.13`	`JerJer[mobile]`	`try running a VoIP provider where ppl don't want to register with your proxy`
`21:09.22`	`ManxPower`	`diclophis-work: it is whatever context you specified in sip.conf!`
`21:09.30`	`diclophis-work`	`in the default of sip.conf?`
`21:09.38`	`*** join/#asterisk anthm (n=anthm@adsl-76-228-194-178.dsl.milwwi.sbcglobal.net)`
`21:09.38`	`*** mode/#asterisk [+o anthm] by ChanServ`
`21:09.44`	`ManxPower`	`JerJer[mobile]: Those are called "former customers" in my world.`
`21:09.51`	`diclophis-work`	`or in the register statement?`
`21:09.56`	`ManxPower`	`diclophis-work: the default sip.conf is not a working file.`
`21:10.08`	`ManxPower`	`it is an example of all the config options, not a file you can drop in and use.`
`21:10.16`	`diclophis-work`	`right`
`21:10.17`	`diclophis-work`	`i know that`
`21:10.25`	`diclophis-work`	`i have the registry part working`
`21:10.36`	`ManxPower`	`diclophis-work: So what is the context= line set to?`
`21:10.41`	`diclophis-work`	`its just when i call from boxb->boxa boxb says curcuits are busy`
`21:10.47`	`ManxPower`	`that is the context the calls will call into.`
`21:10.48`	`diclophis-work`	`i dont have a context= line`
`21:10.59`	`ManxPower`	`diclophis-work: thew it won't work`
`21:11.08`	`ManxPower`	`I am, of course referring to the sip.conf on boxa`
`21:11.23`	`diclophis-work`	`ok, so on box a i have a register->boxb, and a context=inbound`
`21:11.26`	`diclophis-work`	`let me try that`
`21:11.47`	`ManxPower`	`diclophis-work: I really don't care about your register line. You said it works, I believe you.`
`21:11.58`	`diclophis-work`	`what should i make my exten?>`
`21:11.59`	`ManxPower`	`so on boxa you have a line that says context=inbound?`
`21:12.02`	`diclophis-work`	`i was using exten => s,1,`
`21:12.10`	`diclophis-work`	`yes i just put that into place`
`21:12.12`	`ManxPower`	`diclophis-work: that will never work.`
`21:12.19`	`diclophis-work`	`...`
`21:12.28`	`ManxPower`	`diclophis-work: Tell me what the "s" extension is for.`
`21:12.34`	`diclophis-work`	`start?`
`21:12.47`	`ManxPower`	`diclophis-work: no.`
`21:12.51`	`diclophis-work`	`i thought i was like the wildcard for any context?`
`21:12.59`	`ManxPower`	`that is also wrong.`
`21:13.15`	`ManxPower`	`exten => s will match any call with NO DESTINATION DIGITS`
`21:13.34`	`ManxPower`	`Typically this only happens on FXO ports that do not have DIDs`
`21:13.56`	`ManxPower`	`diclophis-work: what is the Dial line on boxa`
`21:14.18`	`diclophis-work`	`exten => _8XXXXXXXXX,1,Dial(SIP/${EXTEN}@prometheus,60)`
`21:14.20`	`ManxPower`	`diclophis-work: it sounds like you have been using the Wiki as a source for Asterisk docs?`
`21:14.29`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`21:14.32`	`diclophis-work`	`ues`
`21:14.33`	`diclophis-work`	`er yes`
`21:14.39`	`ManxPower`	`Then you need a matching exten line on the destination`
`21:14.55`	`ManxPower`	`diclophis-work: do not ever use the Wiki for Asterisk docs. half the docs on the Wiki are wrong.`
`21:15.05`	`ManxPower`	`use the docs that come with asterisk.`
`21:15.25`	`diclophis-work`	`bingo`
`21:15.30`	`diclophis-work`	`thanbks`
`21:15.36`	`diclophis-work`	`i had a frekin extra X on boxa`
`21:15.49`	`diclophis-work`	`man, i have been working with asterisk for 2 years now`
`21:15.52`	`diclophis-work`	`you think i would know this stuff`
`21:15.58`	`ManxPower`	`<PROTECTED>`
`21:16.07`	`ManxPower`	`also use "show applications" for application docs`
`21:16.12`	`mvanbaak`	`those docs are in latex now`
`21:16.15`	`mvanbaak`	`annoying`
`21:16.40`	`ManxPower`	`mvanbaak: You are referring to 1.4, I assume.`
`21:16.40`	`mvanbaak`	`-trunk`
`21:17.04`	`ManxPower`	`mvanbaak: We generally talk about released versions here.`
`21:17.15`	`*** join/#asterisk ardor (n=Miranda@las-static-66.18.135.148.mpowercom.net)`
`21:17.18`	`ManxPower`	`so your comment does not refer to any released asterisk version,. correct?`
`21:17.22`	`mvanbaak`	`soon it will be released`
`21:17.33`	`mvanbaak`	`ManxPower: I dont know about 1.4`
`21:17.39`	`mvanbaak`	`I went from 1.2 to -trunk`
`21:17.41`	`ManxPower`	`I think people will raise holy hell if they change the docs format in 1.4`
`21:17.54`	`mvanbaak`	`nothing to stop it`
`21:17.59`	`ardor`	`Can anyone point me to app_confrence HowTo play Music for an empty room`
`21:18.01`	`mvanbaak`	`everything is converted to .tex`
`21:18.39`	`ManxPower`	`mvanbaak: 1.4 gets only bug fixes. As a documentation format change would not be to fix a "bug" it should not happen in 1.4`
`21:19.28`	`ManxPower`	`Since trunk will become 1.6 and there is not even a release schedule for 1.6, I think we can safely not worry about the docs format change for a while.`
`21:20.15`	`[TK]D-Fender`	`elriah, just got in`
`21:21.01`	`ManxPower`	`It just annoys me when someone spouts off something about a version not many people here run and not even bother to mention the verison this applies to.`
`21:21.10`	`*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)`
`21:21.27`	`mvanbaak`	`lemme check`
`21:21.45`	`elriah`	`[TK]D-Fender: Ever seen a problem where a Polycom phone (501) running latest bootrom.ld/sip.ld that works just fine placing and receiving calls but the MWI light doesn't blink when receiving a call and the little phone icon next to the line label is simply not there.`
`21:21.57`	`elriah`	`Behind nat, asterisk 1.2.18`
`21:22.06`	`[TK]D-Fender`	`elriah, Nope`
`21:22.09`	`mvanbaak`	`1.4 still has .txt files`
`21:22.26`	`mvanbaak`	`ManxPower: I'm deeply sorry I said something to annoy you`
`21:22.51`	`ManxPower`	`[TK]D-Fender: mvanbaak says that all the docs in -trunk are in TeX format. Have you heard anything about that?`
`21:23.10`	`elriah`	`[TK]D-Fender: Thanks. Batting 100 here, must be a firewall issue of some kind. hrm...`
`21:23.27`	`ManxPower`	`elriah: I cannot imagine how it could be a firewall problem`
`21:23.48`	`mvanbaak`	`ManxPower: http://svn.digium.com/view/asterisk/trunk/doc/`
`21:23.55`	`ManxPower`	`unless your firewall is "sip aware NAT" in which case, as always, all bets are off`
`21:24.01`	`elriah`	`ManxPower: I'm all out of options, everything is right in the configs... And we have a tone o fthese phones... These are the only ones with the issue.`
`21:24.08`	`ManxPower`	`mvanbaak: um, most of us will never ever use -trunk.`
`21:24.09`	`elriah`	`ManxPower: It's a Fortigate.`
`21:24.23`	`ManxPower`	`elriah: never heard of it.`
`21:24.23`	`elriah`	`Does the same thing even when I remove the SIP proxy rule.`
`21:24.36`	`elriah`	`Fortigate? Military grade stuff...`
`21:24.42`	`mvanbaak`	`ManxPower: I know. like I said: I'm sorry`
`21:24.51`	`[TK]D-Fender`	`ManxPower, I don't use trunk, never have. Sorry`
`21:24.54`	`*** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91)`
`21:25.15`	`fujin`	`anyone know about sylantro bridged line appearance (BLA) or Broadsoft Shared Call appearance (SCA)?`
`21:25.20`	`ManxPower`	`[TK]D-Fender: Just wondering. I can't imagine the developers allowing let another dependency creep into Asterisk`
`21:25.40`	`[TK]D-Fender`	`ManxPower, "This website is best viewed with... YOUR EYES"`
`21:25.45`	`mvanbaak`	`ManxPower, [TK]D-Fender: look at that svn.digium.org url I gave you`
`21:26.33`	`[TK]D-Fender`	`ManxPower, mvanbaak : I'm sure the two of you can work this out, or concede to give a hoot without me becoming a mediator :)`
`21:26.50`	`mvanbaak`	`[TK]D-Fender ;)`
`21:26.58`	`[TK]D-Fender`	`fujin, Doesn't exist in *.`
`21:28.29`	`GuruJee`	`Need help with DUNi !! anyone help!!`
`21:28.36`	`ManxPower`	`[TK]D-Fender: nothing to really work out. If changing the docs format is a policy decision then the user community needs to do something about it.`
`21:29.05`	`GuruJee`	`need help with a dial-plan, can anyone help me please?`
`21:29.07`	`[TK]D-Fender`	`ManxPower, right.. you grab the pitchforks & torches, and I'll gather the villagers!`
`21:29.26`	`[TK]D-Fender`	`GuruJee, Ask a specific question and you might get a specific answer`
`21:29.32`	`GuruJee`	`thanks tk`
`21:29.54`	`ManxPower`	`[TK]D-Fender: My main problem is that you need to install this MASSIVE suite of things just to read the damn docs.`
`21:30.11`	`GuruJee`	`well, i connected two asterisk boxes. One of them has a PRI. I want the one without PRI to be able to dial out on the trunk that connects it to the system with a pri`
`21:30.14`	`[TK]D-Fender`	`ManxPower, ALLEGEDLY <-`
`21:30.32`	`GuruJee`	`basically, i want to dial out to the pstn`
`21:31.00`	`GuruJee`	`tk..`
`21:31.01`	`ManxPower`	`[TK]D-Fender: blame russellb, he's the one that converted everything`
`21:31.01`	`[TK]D-Fender`	`GuruJee, http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers`
`21:31.13`	`GuruJee`	`i have seen that`
`21:31.15`	`ManxPower`	`[TK]D-Fender: Hence, my wanting to know if this was a policy or not.`
`21:31.19`	`[TK]D-Fender`	`ManxPower, Blame? Feh!`
`21:31.28`	`GuruJee`	`the problem is that I am running DUNDi`
`21:31.43`	`mvanbaak`	`it was a policy`
`21:31.51`	`mvanbaak`	`seesh, why dont you believe me ?`
`21:31.51`	`[TK]D-Fender`	`ManxPower, Never touched those docs before and the claim is unsubstantiated and I believe still falls into the "Do I actually care?" category :)`
`21:32.04`	`ManxPower`	`mvanbaak: do you work for Digium as a developer?`
`21:32.17`	`mvanbaak`	`no, but I was there when they discussed it`
`21:32.28`	`ManxPower`	`[TK]D-Fender: the docs are the most current and accurate docs for Asterisk.`
`21:33.01`	`ManxPower`	`The Wiki is a cesspool of outdated information that makes people waste time.`
`21:33.38`	`[TK]D-Fender`	`ManxPower, Well as I always run fully loaded systems I'm sure I will continute to not care either way for the time being. If you wish to mount a crusade against this action, more power to you....`
`21:34.13`	`mvanbaak`	`ah well, never mind`
`21:34.17`	`ManxPower`	`[TK]D-Fender: you never have to look anything up in README.variables?`
`21:34.40`	`*** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net)`
`21:34.43`	`[TK]D-Fender`	`ManxPower, Nope. My intuition and the WIKI have gotten me far.`
`21:34.51`	`mvanbaak`	`it's been decided, done, and committed`
`21:35.07`	`mvanbaak`	`if you dont want to believe it, that's fine`
`21:35.15`	`[[blah]asfd`	`quick question... is there a way to show the current status of an SIP phone to a web interface? for example is there an AGI program which is capable of parsing this status to php? idle, ringing, on call... whatever?`
`21:35.16`	`mvanbaak`	`you'll notice when 1.6 is there`
`21:36.04`	`ManxPower`	`mvanbaak: I fully and totally and completely believe that the docs have been converted to TeX`
`21:36.05`	`jsolares`	`[[blah]asfd: There's the Flash Operator Panel now with DHTML`
`21:36.15`	`[TK]D-Fender`	`mvanbaak, "When its done", and "when I'm ready for it", sure, whatever :)`
`21:36.18`	`ManxPower`	`This is a fact.`
`21:36.30`	`[TK]D-Fender`	`[[blah]asfd, Yeah, go write a quick script`
`21:36.42`	`ManxPower`	`What you cannot help me with is understanding why and help me form an arguement as to why this is a very bad idea.`
`21:36.45`	`*** join/#asterisk zm23 (n=chatzill@zombie.cc.columbia.edu)`
`21:36.59`	`mvanbaak`	`ManxPower: I agree with you it's a bad idea`
`21:37.01`	`mvanbaak`	`totally`
`21:37.02`	`[[blah]asfd`	`wanting to do the script, just not sure which command will give me the info I need. that is where I am looking for help`
`21:37.21`	`mvanbaak`	`I hate reading .tex source files`
`21:37.38`	`*** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de)`
`21:38.14`	`[[blah]asfd`	`i am looking to add this phone status to a custom app without having to reverse engineer the flash operator panel.`
`21:39.00`	`ManxPower`	`there we go. discussion can be continued on asterisk-dev`
`21:39.34`	`[TK]D-Fender`	`[[blah]asfd, How "live" does this have to bbe?`
`21:39.39`	`ManxPower`	`[[blah]asfd: /path/src/asterisk/docs/manager.txt or "help manager" in the asterisk CLI, IIRC`
`21:39.43`	`mvanbaak`	`ManxPower: I'm even thinking of putting some script online that does: svn up && convert doc/*.tex to txt && upload to webspace`
`21:39.49`	`*** part/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`21:40.02`	`vAd0r`	`how many calls can i make over a sip trunk?`
`21:40.13`	`fujin`	`AHH`
`21:40.18`	`[[blah]asfd`	`[TK]D-Fender: doesnt have to be 100% realtime. it is for managers to see if their people are really working.`
`21:40.21`	`[TK]D-Fender`	`vAd0r, As many as each end and your bandwidth allows`
`21:40.21`	`mvanbaak`	`vAd0r: depends on your upload bandwidth and provider`
`21:40.25`	`[[blah]asfd`	`just a status indicator.`
`21:40.33`	`[[blah]asfd`	`so maybe within a few seconds of realtime`
`21:40.34`	`ManxPower`	`vAd0r: There is no such thing as a SIP "trunk". Perhaps you mean a "sip connection".`
`21:40.41`	`vAd0r`	`yes`
`21:40.41`	`fujin`	`^5 ManxPower`
`21:40.46`	`ManxPower`	`The answer is unlimited, subject to your bandwidth.`
`21:40.50`	`vAd0r`	`i have a sip trunk to my office pbx cisco phone system`
`21:40.55`	`[TK]D-Fender`	`[[blah]asfd, use the AMI to call a CLI "show hints" and parse it out`
`21:40.57`	`fujin`	`you have a sip CONNECTIOn`
`21:41.01`	`fujin`	`blargggh`
`21:41.06`	`ManxPower`	`vAd0r: You do not have a SIP trunk.`
`21:41.10`	`vAd0r`	`i can make and receive calls but when one call is up the other does not work`
`21:41.11`	`vAd0r`	`ok`
`21:41.25`	`vAd0r`	`im calling it that because that is what cisco called it upon creation`
`21:41.28`	`[[blah]asfd`	`[TK]D-Fender: ok.. I will check that out.`
`21:41.32`	`*** join/#asterisk don_martino (n=martin@pd95b5be9.dip0.t-ipconnect.de)`
`21:41.35`	`[[blah]asfd`	`thank you for the advice.`
`21:41.40`	`[TK]D-Fender`	`vAd0r, then one side is limiting how many connections it feels like allowing.`
`21:41.49`	`ManxPower`	`There are two things that can be called "trunk" in Asterisk, and a SIP peer/user/friend is not one of them.`
`21:42.02`	`vAd0r`	`do i have to put some command in that will allow more than one ?`
`21:42.07`	`[TK]D-Fender`	`[[blah]asfd, I have one for my Polycom MicroBrowser for global presence`
`21:42.14`	`ManxPower`	`vAd0r: Since Asterisk has no built in limit, then it must be the Cisco`
`21:42.22`	`vAd0r`	`ok`
`21:42.27`	`vAd0r`	`ill check the trunk to see`
`21:42.28`	`vAd0r`	`thx`
`21:42.29`	`ManxPower`	`vAd0r: Asterisk will allow you to have 10,000 or more calls between two asterisk servers`
`21:42.32`	`ManxPower`	`by default`
`21:42.34`	`ManxPower`	`automatically`
`21:42.41`	`[TK]D-Fender`	`ManxPower, Of course its the Cisco... now why would they want to pass up the ability to charge per session? :)`
`21:43.01`	`[TK]D-Fender`	`ManxPower, jsut like NorHell`
`21:43.07`	`fujin`	`lol`
`21:43.09`	`*** join/#asterisk Greek-Boy (n=g@196.45.144.42)`
`21:43.16`	`fujin`	`thank god I just got rid of the norhell here`
`21:43.20`	`fujin`	`terrible bloody thing`
`21:43.21`	`don_martino`	`hello world. i'm fairly new to asterisk. i want to connect to very simple asterisk boxes via iax2. i try the example i read in a book. asterisk does not complain while reading the config files, i testet port forwarding, it all works. the problem is that neither box is trying to send a packet to the other box. the line just remains silent. so i gues si need to tell asterisk to establish the connection how do i do that?`
`21:43.29`	`mvanbaak`	`Nortell Schmortell`
`21:43.30`	`*** join/#asterisk Dovid (n=Dovid@bzq-88-155-44-193.red.bezeqint.net)`
`21:43.34`	`Dovid`	`hi guys`
`21:44.11`	`bill4242`	`I'm lost..`
`21:44.31`	`[TK]D-Fender`	`bill4242, You're in #asterisk .... now you know where you are!`
`21:44.39`	`Dovid`	`how do i do trunk grouping with a TDM400p`
`21:44.49`	`Dovid`	`so i can use Dial(ZAP/g1/.......`
`21:44.49`	`bill4242`	`haha... yeah`
`21:44.51`	`rene-`	`elriah: are u still there`
`21:45.07`	`rene-`	`elriah: i ve got a windows xp pc`
`21:45.13`	`[TK]D-Fender`	`ManxPower, Doesn't that word just make you want to shoot the person uttering it? :)`
`21:45.18`	`mvanbaak`	`rene-: my condoleances`
`21:45.19`	`bill4242`	`I can't seem to make any outbound calls.. i know my dialing rules aren't right I just can't seem where i'm going wrong..`
`21:45.39`	`Dovid`	`TK: did u see my question ? (sorry first time I am dealing with hardware.....)`
`21:46.00`	`rene-`	`mvanbaak: hehe`
`21:46.07`	`fujin`	`Dovid: put group=1 above your channel definitions`
`21:46.09`	`ManxPower`	`config files and CLI output on pastebin.ca is a good place to start.`
`21:46.10`	`rene-`	`~seen: elriah`
`21:46.21`	`mvanbaak`	`~pb`
`21:46.34`	`jbot`	`hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`21:46.35`	`[TK]D-Fender`	`Dovid, "group=[0-63]" for your channels`
`21:46.42`	`Dovid`	`fujin: i have that. I will test. thanks`
`21:47.02`	`rene-`	`well, i have mitel iphones and windows PCs`
`21:47.16`	`mvanbaak`	`I'm off to get some sleep`
`21:47.17`	`mvanbaak`	`latero`
`21:47.18`	`fujin`	`wee, mitel phones`
`21:47.21`	`fujin`	`i have 5224s here`
`21:47.27`	`fujin`	`wish they had have gone with ciscos or polycom`
`21:47.41`	`fujin`	`Dovid: http://rafb.net/p/6kP05V76.html`
`21:47.44`	`rene-`	`and a POE switch, the PC's should connecto to the phones ethernet port`
`21:47.56`	`rene-`	`but they should live on different networks`
`21:48.03`	`rene-`	`so i am trying to get this vlan thing going on`
`21:48.09`	`fujin`	`learn2network`
`21:48.12`	`rene-`	`but well it is kind of hard`
`21:48.26`	`fujin`	`basically you need to configure the port on the switch as a trunkport, and then configure it with two vlans`
`21:48.29`	`fujin`	`I use 2 and 3`
`21:48.29`	`rene-`	`yeah i wish i had polycom and not mitel`
`21:48.43`	`rene-`	`and do you asign the vlan to the mitel phone?`
`21:48.47`	`fujin`	`correct`
`21:49.20`	`fujin`	`and then the switchport in the back of the mitels expansion dock will automatically grab the other one`
`21:49.29`	`rene-`	`oh`
`21:49.49`	`vAd0r`	`what do people use to connect their own phoneline to use as an outbound line`
`21:49.53`	`rene-`	`and to talk to the asterisk server`
`21:50.05`	`rene-`	`<PROTECTED>`
`21:50.16`	`fujin`	`rene-: well no, the asterisk server should be ont he same vlan as the phones`
`21:50.22`	`rene-`	`oh`
`21:50.31`	`rene-`	`so should i set up linux vlan there too?`
`21:50.37`	`[TK]D-Fender`	`vAd0r, www.digium.com <- hav fun`
`21:50.40`	`ManxPower`	`vAd0r: I use a T-1 card, but many people use analog cards`
`21:50.48`	`fujin`	`you won't need to setup a vlan on the linux box, just terminate a vlan onto the port that it's on as an access port`
`21:50.51`	`irule`	`is it a good idea to create a macro for zal trunk? I want to set something more interesting than a busy signal quen I get the s-CHANUNAVAIL`
`21:50.55`	`vAd0r`	`lol i odnt have a T1 at the house`
`21:51.12`	`[[blah]asfd`	`[TK]D-Fender: so i did the hint... but show hints just shows the phone as idle no matter what state it is in. Do I need to add something to sip.conf?`
`21:51.12`	`fujin`	`vAd0r: a digium td400p or whatever. thats' what I had at my old place`
`21:51.18`	`fujin`	`or a 200p`
`21:51.19`	`vAd0r`	`how much you think`
`21:51.25`	`fujin`	`few hundred, check ebay`
`21:51.33`	`fujin`	`the tiger cards are cheap and reasonable`
`21:51.47`	`vAd0r`	`what sux is i have a vg200 cisco and can't get it to work`
`21:51.54`	`hansin321`	`vAd0r: if you just have one line, maybe you could use the 'wildcard X100P' or 'X101P' card.`
`21:51.58`	`[TK]D-Fender`	`irule, you tell us. Macros only do what we tel them to. You need to imagin more than the odd sample or two you've seen already...`
`21:52.23`	`hansin321`	`vAd0r: that would allow you to connect a single POTS line to your PC.`
`21:52.30`	`[[blah]asfd`	`[TK]D-Fender: here is what i did in extensions.conf`
`21:52.30`	`[[blah]asfd`	`exten => 100,hint,SIP/100`
`21:52.30`	`[[blah]asfd`	`exten => 100,1,Dial(SIP/${EXTEN}\|15\|)`
`21:52.34`	`vAd0r`	`yeah that is what i want`
`21:53.17`	`*** join/#asterisk saftsack (n=saftsack@pd9e07656.dip.t-dialin.net)`
`21:53.24`	`rene-`	`fujin: i didnt got your last statement...`
`21:53.37`	`fujin`	`sorry?`
`21:53.46`	`[TK]D-Fender`	`[[blah]asfd, with 1.4.x+ you should have "type=peer" and "call-limit=[big # like 99 or something]"`
`21:53.52`	`rene-`	`fujin: are you able to perform a factory reset on the 5224 by means of the web interface`
`21:53.55`	`fujin`	`you don't need to put a vlan onto the box, you just need to configure a port to be an access port (Not a trunk port)`
`21:54.05`	`fujin`	`rene-: uhh, no.. don' think so - i use the phones interface`
`21:54.08`	`[TK]D-Fender`	`[[blah]asfd, SIP spresence got kicked in the nads in 1.4`
`21:54.09`	`rene-`	`fujin i got you`
`21:54.14`	`rene-`	`now`
`21:54.18`	`bill4242`	`OK, in detail, as requested.. I can't make outbound calls they fallback to the default context.. my zapata.conf files have group1 defined as ZAP=5-7 with default context and group2 defined as ZAP=1-4 with context=outbound... Here's my extensions.conf file: [general]`
`21:54.18`	`bill4242`	`static=yes`
`21:54.18`	`bill4242`	`writeprotect=no`
`21:54.18`	`bill4242`	`[globals]`
`21:54.18`	`bill4242`	`BILL=ZAP/1`
`21:54.20`	`bill4242`	`MILA=ZAP/2`
`21:54.22`	`bill4242`	`WES=ZAP/3`
`21:54.24`	`fujin`	`~pb`
`21:54.29`	`jbot`	`extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`21:54.29`	`bill4242`	`JOHN=ZAP/4`
`21:54.29`	`bill4242`	`KARL=1234567890`
`21:54.29`	`bill4242`	`TRUNK=Zap/g1`
`21:54.30`	`bill4242`	`BILLCELLPHONE=0987654321`
`21:54.31`	`Fieldy`	`stop`
`21:54.32`	`bill4242`	`TECHSUP=${BILL}&${WES}&${JOHN}`
`21:54.33`	`[TK]D-Fender`	`SOMEBODY KICK HIM`
`21:54.34`	`zm23`	`Hello all, need help transferring calls "to" asterisk. are their any config settings to allow attended call transfers?`
`21:54.34`	`bill4242`	`BILLING=${WES}`
`21:54.36`	`Fieldy`	`can someone please +q him`
`21:54.36`	`bill4242`	`SALES=${MILA}&${WES}&${KARL}`
`21:54.38`	`bill4242`	`TRUNKMSD=1`
`21:54.40`	`bill4242`	`[default]`
`21:54.42`	`bill4242`	`include => mainmenu`
`21:54.44`	`bill4242`	`[outbound]`
`21:54.46`	`bill4242`	`include => local`
`21:54.48`	`bill4242`	`[mainmenu]`
`21:54.50`	`bill4242`	`exten => s,1,Wait,1`
`21:54.52`	`Fieldy`	`angler / anthm / Corydon76-home / file / Qwell[]: ping: flooder`
`21:54.52`	`bill4242`	`exten => s,2,Wait,1`
`21:54.54`	`bill4242`	`exten => s,3,Answer`
`21:54.56`	`bill4242`	`exten => s,4,Dial(${BILL})`
`21:54.58`	`bill4242`	`[local]`
`21:54.59`	`[[blah]asfd`	`[TK]D-Fender: k, will try that, thanks`
`21:55.00`	`bill4242`	`exten => _9NXXXXXX,1,Dial(ZAP/5/${ARG1})`
`21:55.01`	`rene-`	`fujin: i have some nice ruby scripts that perform some nice stuff like restarting the phones, upgrading firmware but so far i cannnot tell how to factory reset them`
`21:55.02`	`bill4242`	`....`
`21:55.04`	`bill4242`	`that didnt paste as expected`
`21:55.12`	`Fieldy`	`try http://rafb.net/paste`
`21:55.18`	`Fieldy`	`angler / anthm / Corydon76-home / file / Qwell[]: unping.`
`21:55.24`	`zm23`	`Hello all, need help transferring calls "to" asterisk. are their any config settings to allow attended call transfers?`
`21:55.44`	`[TK]D-Fender`	`zm23, nope`
`21:56.27`	`zm23`	`not using asterisk to manage calls.. only transferring to asterisk. aah.. TK-D-fender.. hi... i've been here before with same issue..`
`21:56.32`	`zm23`	`:(`
`21:56.38`	`*** join/#asterisk WindBack (n=jorge@host44.200-117-61.telecom.net.ar)`
`21:56.53`	`vAd0r`	`will they sell the x100p at best buy or any place like that or is it an ebay item`
`21:57.31`	`[TK]D-Fender`	`vAd0r, its an out of production crap option. Buy a TDM400.`
`21:57.40`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`21:57.43`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`21:57.54`	`[[blah]asfd`	`[TK]D-Fender:that did it, thank you`
`21:58.16`	`vAd0r`	`i can't find it`
`21:58.19`	`WindBack`	`I'm trying to programing AGI with python. Are there any framework who make the things easy???`
`21:58.39`	`frigidzephyr`	`vAd0r: www.digium.com`
`21:59.27`	`ManxPower`	`vAd0r: the X100P and X101P have not been made in several years. remove it from your list of devices you could buy`
`21:59.47`	`*** join/#asterisk init6 (n=chad@d198-53-142-118.abhsia.telus.net)`
`22:00.08`	`vAd0r`	`k`
`22:00.16`	`vAd0r`	`that one is 30 bux as opposed to 300`
`22:00.18`	`WindBack`	`I saw a lot, but i don't know who is the best. And I didn't find any documentation`
`22:00.21`	`pipwerk`	`there are quite a few refurbished available`
`22:00.24`	`vAd0r`	`is there any others that will do just one line`
`22:00.46`	`pipwerk`	`sangoma (sp?)`
`22:01.17`	`*** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-026b0572937a57cd)`
`22:02.15`	`init6`	`we upgraded asterisk and now incoming fax doesn't work. the rxfax() starts and the tiff is created but the system() to email the tiff never gets run. any idea why?`
`22:02.16`	`[TK]D-Fender`	`vAd0r, TDM400P is about $130 with 1 FXO module, not $300`
`22:02.23`	`vAd0r`	`on that site`
`22:02.35`	`vAd0r`	`http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=17`
`22:02.50`	`[TK]D-Fender`	`vAd0r, Slightly more affordable is the Linksys SPA-3102. An ATA/gateway that ill give you 1 FXO, and 1 FXO`
`22:02.53`	`*** join/#asterisk jm\|laptop (n=jm\|home@zen.jamiem.com)`
`22:03.11`	`ManxPower`	`the SPA is a bitch to get PSTN calls sent to asterisk`
`22:03.26`	`[TK]D-Fender`	`init6, this has nothing to do with *. check your script out`
`22:03.53`	`[TK]D-Fender`	`ManxPower, hardly... so many quick & easy guides out there that anyone struggling jsut doesn't ahve a cliue.`
`22:03.53`	`*** join/#asterisk duckz (n=duckz@141.85.3.18)`
`22:03.58`	`vAd0r`	`i think i have something like that`
`22:03.58`	`PioneerVM2`	`Hey, anyone know of a bug in Set(CALLERID(num)=) vs. SetCallerID() with SIP?`
`22:04.04`	`vAd0r`	`i signed up for vonage`
`22:04.06`	`vAd0r`	`and caneled`
`22:04.14`	`vAd0r`	`they sent me some sore of linksys w/ 2 ports on it`
`22:04.23`	`vAd0r`	`but i think they are prob fsx though`
`22:04.28`	`vAd0r`	`fxs`
`22:04.40`	`init6`	`[TK]D-Fender, the script worked before the upgrade. the console shows RxFax() being called but not the System() command.`
`22:04.45`	`*** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ma.comcast.net)`
`22:04.55`	`WindBack`	`I'm trying to programing AGI with python. Are there any framework who make the things easy???`
`22:04.56`	`aptura`	`[TK]D-Fender has digium on occation had a module burn out on the tdm400p?`
`22:04.58`	`PioneerVM2`	`I was using Set(CALLERID(num)) with SIP and it wouldnt work, then switched to IAX and it worked fine -- but once I used SetCallerID it worked with both.`
`22:05.00`	`progmonkey`	`Can I ask a question about AEL?`
`22:05.31`	`progmonkey`	`I'm about to change all my .conf files into .ael`
`22:05.35`	`[TK]D-Fender`	`PioneerVM2, setcalledid is deprecated, and the function works.`
`22:05.39`	`init6`	`The debug log and the sending FAX machine both indicate a successfull transmission.`
`22:05.45`	`PioneerVM2`	`Yes I know its depricated`
`22:05.50`	`PioneerVM2`	`but there is something wrong with SET(CALLERID)`
`22:05.59`	`PioneerVM2`	`they apparently don't work the same way`
`22:06.00`	`[TK]D-Fender`	`progmonkey, its a changing spec I wouldn't bother with.....`
`22:06.31`	`WindBack`	`I saw a lot, but i don't know who is the best. And I didn't find any documentation`
`22:06.36`	`PioneerVM2`	`I was using SET(CALLERID and it would not work on a SIP trunk, but SetCallerID did -- however either worked on IAX`
`22:06.37`	`[TK]D-Fender`	`PioneerVM2, Sure it does. Go ahead and show us what you're trying and we'll tell you where you went wrong.`
`22:06.41`	`ManxPower`	`PioneerVM2: perhaps you need Set(CALLERID(num)=) instead of Set(CALLERID(num))`
`22:06.44`	`progmonkey`	`[TK]D-Fender: that was my question... is it a good idea?`
`22:06.48`	`PioneerVM2`	`no i was doing that Manx`
`22:06.59`	`PioneerVM2`	`i troubleshooted with VoicePulse`
`22:07.01`	`WindBack`	`who know something about AGI???`
`22:07.04`	`[TK]D-Fender`	`PioneerVM2, Show us or move on....`
`22:07.04`	`PioneerVM2`	`they told me to go back and try SetCallerID and it worked`
`22:07.16`	`progmonkey`	`Does AEL have a future?`
`22:07.17`	`PioneerVM2`	`what do u want to see`
`22:07.21`	`[TK]D-Fender`	`progmonkey, I wouldn't waste my time with it personally...`
`22:07.36`	`ManxPower`	`PioneerVM2: then you are an excellent example of why you should use copy/paste rather than retyping to the channel`
`22:07.57`	`*** join/#asterisk santiago (n=santiago@debian/developer/santiago)`
`22:08.06`	`PioneerVM2`	`Manx: i was trying to find out in general if there was a known issue before i got into details`
`22:08.09`	`WindBack`	`please, somebody answer to me`
`22:08.19`	`[TK]D-Fender`	`PioneerVM2, we clearly want to see how you're calling it in your dialplan, and how you set up the peers you're dialing out of.`
`22:08.20`	`WindBack`	`I'm new here`
`22:08.23`	`progmonkey`	`WindBack: hello WindBack`
`22:08.25`	`ManxPower`	`PioneerVM2: I doubt many people try to set the callerid to empty.`
`22:08.35`	`[TK]D-Fender`	`WindBack, Ask a SPECIFIC question and you might get a specific answer`
`22:08.38`	`PioneerVM2`	`exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=2031234567)`
`22:08.57`	`PioneerVM2`	`exten => _1NXXNXXXXXX,1,SetCallerID(2031234567)`
`22:09.06`	`WindBack`	`[TK]D-Fender, I did an specific answer`
`22:09.07`	`[TK]D-Fender`	`PioneerVM2, pastebin the whole mess please, not in HERE.`
`22:09.09`	`WindBack`	`I'm trying to programing AGI with python. Are there any framework who make the things easy???`
`22:09.10`	`ManxPower`	`PioneerVM2: I fail to see how that sets it to empty`
`22:09.11`	`PioneerVM2`	`thats all`
`22:09.20`	`[TK]D-Fender`	`WindBack, Go look on the WIKI.`
`22:09.31`	`ManxPower`	`WindBack: have you tried the python bindings for AGI?`
`22:09.39`	`*** join/#asterisk saftsack (n=saftsack@pd9e07656.dip.t-dialin.net)`
`22:09.47`	`Fieldy`	`indeed`
`22:09.54`	`PioneerVM2`	`it's just those two lines -- when I used version 1, it would work on IAX lines but not on SIP -- when iused version two it fixed the SIP issue.`
`22:10.09`	`*** part/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net)`
`22:10.13`	`PioneerVM2`	`The only difference between IAX and SIP configs was doing a search/replace in extensions.conf`
`22:10.15`	`ManxPower`	`PioneerVM2: I thought you were trying to set it to empty.`
`22:10.18`	`PioneerVM2`	`no`
`22:10.24`	`PioneerVM2`	`I was trying to set it to something`
`22:10.43`	`ManxPower`	`You were too vague.`
`22:10.52`	`WindBack`	`ManxPower, I was loking in it on www.vo-ip.org. But there are any documentation`
`22:10.54`	`PioneerVM2`	`apparently so`
`22:11.03`	`ManxPower`	`I use Set(CALLERID(num)=12345) ALL THE TIME and it works just fine.,`
`22:11.07`	`PioneerVM2`	`sorry for the confusion -- i was trying to find out if there was a known difference in the two formats`
`22:11.17`	`PioneerVM2`	`yes, well it doesnt work here on 1.4.4 with VoicePulse connect`
`22:11.39`	`WindBack`	`Sorry with my English`
`22:11.56`	`ManxPower`	`PioneerVM2: your SetCallerID line sets the NAME, not the NUMBNER`
`22:12.10`	`PioneerVM2`	`I am well aware`
`22:12.17`	`ManxPower`	`SetCallerID(Persons Name <thetelephonenumber>`
`22:12.35`	`ManxPower`	`so to make a VALID compairson yo would have to do xten => _1NXXNXXXXXX,1,Set(CALLERID(name)=2031234567)`
`22:12.46`	`PioneerVM2`	`well it came through with the number`
`22:12.47`	`*** join/#asterisk Fieldy (i=12Wjuuwn@gentoo/contributor/Fieldy)`
`22:12.51`	`PioneerVM2`	`ok, here is my point`
`22:12.57`	`*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)`
`22:13.04`	`PioneerVM2`	`Set(CALLERID(num)= worked with SIP but not with IAX`
`22:13.09`	`ManxPower`	`PioneerVM2: My pioint is that you are not comparing the same things`
`22:13.17`	`PioneerVM2`	`no changes other than changing SIP to IAX with search/replace`
`22:13.49`	`ManxPower`	`SIP devices prolly cares there the callerid info is fucked up where iax devices do not.`
`22:14.20`	`ManxPower`	`and there is no arguement that SetCallerID(2031234567) sets a messed up callerid`
`22:14.24`	`PioneerVM2`	`what does that have to do with Set(CALLERID=`
`22:14.27`	`PioneerVM2`	`nono`
`22:14.29`	`PioneerVM2`	`SetCallerID works!`
`22:14.34`	`PioneerVM2`	`Set(CALLERID= does not!`
`22:15.04`	`ManxPower`	`SetCallerID(2031234567) is the equiv of Set(CALLERID(name)=2031234567)`
`22:15.10`	`PioneerVM2`	`Ugh.`
`22:15.10`	`ManxPower`	`try it. I bet it will work`
`22:15.18`	`*** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net)`
`22:15.23`	`PioneerVM2`	`Ok, forget about SetCallerID`
`22:15.38`	`PioneerVM2`	`can you please explain why Set(CALLERID(num)=2031234567) does not work on SIP`
`22:15.42`	`PioneerVM2`	`but it does on IAX`
`22:15.43`	`ManxPower`	`try what I suggested. It will work they way you want.`
`22:16.18`	`PioneerVM2`	`that does not explain why it is not working properly`
`22:16.25`	`PioneerVM2`	`I don't want a hack, i want to know why it is not working properly`
`22:16.43`	`ManxPower`	`PioneerVM2: I cannot. I can only tell you what is correct and what is not correct usage of setting the callerid`
`22:16.45`	`[TK]D-Fender`	`PioneerVM2, Show us it not working.`
`22:17.06`	`*** join/#asterisk oej (n=olle@65-182-39-213.cre.bil.biltmorecommunications.net)`
`22:17.12`	`vAd0r`	`an someone help me w/ upgrading my phone`
`22:17.27`	`ManxPower`	`vAd0r: I'm sorry, but Asterisk does not make phones.`
`22:17.27`	`PioneerVM2`	`Is Set(CALLERID(num)=2031234567) correct?`
`22:17.33`	`vAd0r`	`lol`
`22:17.39`	`vAd0r`	`put sip firmware on 7940`
`22:17.42`	`init6`	`[TK]D-Fender, the rxfax problem is that rxfax does not return anymore. it triggers a hangup. I had to use the "h" extension.`
`22:17.43`	`ManxPower`	`PioneerVM2: it is for setting the callerid name`
`22:17.47`	`WindBack`	`[TK]D-Fender, Are there any repository for Deabian. I want to download and install FreePBX??`
`22:17.48`	`vAd0r`	`sry for vagueness`
`22:17.51`	`ManxPower`	`vAd0r: try a cisco channel`
`22:17.57`	`PioneerVM2`	`CALLERID(num) is??`
`22:18.03`	`ManxPower`	`..er...`
`22:18.11`	`PioneerVM2`	`Isn't that CALLERID(name)?`
`22:18.16`	`ManxPower`	`sorry, it is correct for setting thecallerid number.`
`22:18.20`	`[TK]D-Fender`	`WindBack, go ask in #freepbx . it is NOT supported here.`
`22:18.22`	`PioneerVM2`	`tahnk you.`
`22:18.36`	`WindBack`	`[TK]D-Fender, ok`
`22:18.44`	`ManxPower`	`I was spending too much energy stopping myself from bitchslapping vAd0r`
`22:19.55`	`ManxPower`	`PioneerVM2: You have some OTHER problem, as setting the callerid number on SIP calls happens hundreds of times per day on my PBX without even 1 problem`
`22:20.23`	`ManxPower`	`So you know it is not a general problem with that function`
`22:20.31`	`WindBack`	`[TK]D-Fender, don't you like freepbx??`
`22:20.35`	`WindBack`	`why??`
`22:20.39`	`ManxPower`	`~freepbx`
`22:20.57`	`jbot`	`it has been said that freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there`
`22:20.57`	`ManxPower`	`~amp`
`22:21.01`	`jbot`	`i guess amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP)`
`22:21.02`	`ManxPower`	`WindBack: FreePBX is not really Asterisk.`
`22:21.04`	`[TK]D-Fender`	`WindBack, No, it is a falaming pile of shit to be used only by those who are too dumb to deserve to be administering a PBX`
`22:21.30`	`PioneerVM2`	`I may be kinda preoccupied for next 20 mins as I M awatching my son so cant fully trouble shoot this`
`22:21.39`	`WindBack`	`<PROTECTED>`
`22:21.45`	`[TK]D-Fender`	`WindBack, Asterisk gives you control over how you process calls and the first thing you are asking how to do is GIVE THAT UP.`
`22:21.50`	`*** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net)`
`22:21.59`	`*** join/#asterisk mcab (n=mb@66.195.193.151)`
`22:22.22`	`ManxPower`	`PioneerVM2: then come back when you an devote your full attention to debugging this problem`
`22:22.35`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`22:22.37`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`22:23.48`	`ManxPower`	`[TK]D-Fender: today has been a weird day.`
`22:23.54`	`WindBack`	`[TK]D-Fender, I was studying well haw to create a dialplan whit the .conf files. But I thougth that free PBX was good. I good for me to hear your opinion. Thank You`
`22:24.14`	`WindBack`	`Sorry. Is good for mi`
`22:24.16`	`WindBack`	`Sorry. Is good for me`
`22:25.02`	`fakhir`	`has anyone had problems with jitter on a WRT54G running dd-wrt`
`22:25.03`	`fakhir`	`http://img61.imageshack.us/img61/1427/jittermx1.png`
`22:26.20`	`hansin321`	`Sounds like you can't get the X101P cards anymore, but if one already has one (and isn't doing much but learning at the moment anyway), is there any argument as to why you wouldn't use this card to connect you PC to a POTS line? Thanks.`
`22:28.03`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell)`
`22:28.03`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`22:28.15`	`[TK]D-Fender`	`ManxPower, Lazy day for me. Stat holiday so I got to go out for my first bikeride of the season since getting it back from the shop. Rode to work jsut to see how long it'll take (since I plan on doing this regularly).`
`22:28.45`	`punani`	`<[TK]D-Fender> WindBack, No, it is a falaming pile of shit to be used only by those who are too dumb to deserve to be administering a PBX`
`22:28.48`	`[TK]D-Fender`	`ManxPower, dropped by the grandparents place in so-doing, and am feeling pretty relaxed.`
`22:28.49`	`punani`	`what gui do you use?`
`22:29.00`	`ManxPower`	`punani: we don't use guis`
`22:29.06`	`punani`	`lol`
`22:29.06`	`ManxPower`	`since they are so limiting`
`22:29.08`	`*** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`22:29.12`	`[TK]D-Fender`	`ManxPower, I've spent the rest of this weekend specing out my new projector setup for my livingroom.`
`22:29.12`	`Qwell`	`#astridevcon FTW`
`22:30.01`	`[TK]D-Fender`	`#AEIOUandsometimesY !`
`22:30.46`	`[TK]D-Fender`	`ManxPower, http://www.tigerdirect.ca/applications/searchtools/item-details.asp?EdpNo=2887690&Tab=5`
`22:30.48`	`[TK]D-Fender`	`ManxPower, http://www.insight.ca/apps/productpresentation/index.php?alert=categoryresults&product_id=IF4052284`
`22:31.46`	`ManxPower`	`[TK]D-Fender: I'm starting to think that doing any video/audio streaming is a task thought up by Dante for a new, even more horrible, level of hell.`
`22:32.18`	`[TK]D-Fender`	`ManxPower, Level 8! a whole new world of pain!`
`22:32.46`	`[TK]D-Fender`	`ManxPower, what kind of streaming were you referring to?`
`22:32.58`	`ManxPower`	`On linux everything requires ffmpeg or one of its parts. ON Windows, everything uses WME as the backend`
`22:33.16`	`[TK]D-Fender`	`ManxPower, I sue VLC on both and it runs very nicely...`
`22:33.27`	`ManxPower`	`[TK]D-Fender: Video on Demand streamed from a Linux server to Windows clients, the files are in WMV/ASF format`
`22:33.46`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`22:33.55`	`[TK]D-Fender`	`ManxPower, well I'm not working with that level of video... jsut basic Xvid decode + DVD`
`22:34.20`	`ManxPower`	`I found out that MS Media Player (which is what I want to support) will only play streamed ASF/WMV format.`
`22:34.24`	`*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)`
`22:34.35`	`irule`	`ok, how can I give the user the same continous tone that you hear when you pick up the sip phone?`
`22:34.53`	`[TK]D-Fender`	`ManxPower, WMP = flaming pile of @%$# too :) VLC > ALL and is multi-platform`
`22:34.56`	`ManxPower`	`irule: "show application playtones"`
`22:35.01`	`irule`	`thanks`
`22:35.15`	`ManxPower`	`[TK]D-Fender: Yes, but if I make them install software then all future computing problems will be my fault.`
`22:35.17`	`ManxPower`	`I'm not going there.`
`22:35.42`	`[TK]D-Fender`	`ManxPower, so who pissed in your Cornflakes today? :)`
`22:35.55`	`ManxPower`	`[TK]D-Fender: VLC, actually.`
`22:35.59`	`[TK]D-Fender`	`LO`
`22:36.00`	`[TK]D-Fender`	`:O`
`22:36.28`	`ManxPower`	`I hope to get to compile eventually.`
`22:36.49`	`[[blah]asfd`	`i got an error which crashed my asterisk. I am not familiar enough with linux to understand the error. if this was caused by the kernel, or if asterisk caused the kernel to crash it. could anyone help me to understand what this is and why i got it?`
`22:36.51`	`[[blah]asfd`	`May 21 16:00:27 slk-apbx-01 kernel: asterisk[4927]: segfault at 0000003105b86630 rip 0000002a96aafc4e rsp 0000000040246e10 error 6`
`22:37.06`	`fujin`	`nasty`
`22:37.28`	`ManxPower`	`[[blah]asfd: it should have created a core file in /tmp`
`22:37.36`	`ManxPower`	`[[blah]asfd: what version of Asterisk?`
`22:38.33`	`*** join/#asterisk jm\|laptop (n=jm\|home@zen.jamiem.com)`
`22:38.49`	`[[blah]asfd`	`ManxPower: Asterisk 1.2.14`
`22:40.03`	`WindBack`	`[TK]D-Fender, If I want to make a Log of all calls (Duration, time of beginig, callerId).. Are any soft who do that, or I have to do thath with AGI and for example MySQL???`
`22:40.25`	`ManxPower`	`[[blah]asfd: they won't take bug reports for anything but the latest 1.2.x`
`22:40.33`	`[[blah]asfd`	`ManxPower: looking at the dump file it is mostly odd characters. Should i be able to read all of this?`
`22:40.36`	`*** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net)`
`22:40.49`	`ManxPower`	`WindBack: why not just use the info in /var/log/asterisk/messages`
`22:41.05`	`ManxPower`	`[[blah]asfd: you will be able to after reading README.backtrace`
`22:41.40`	`[[blah]asfd`	`;-) gotcha`
`22:41.53`	`WindBack`	`ManxPower, ahh, ok, but this info isn't so organizated`
`22:42.35`	`ManxPower`	`WindBack: better to write something to organize it than try to do silly hacks in the dialplan`
`22:42.55`	`WindBack`	`ok`
`22:44.45`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`22:46.04`	`pigpen`	`if I do a simultainious ring to a cell phone, can asterisk detect if the call falls into the cell phone voicemail?`
`22:46.13`	`ManxPower`	`pigpen: no`
`22:46.23`	`pigpen`	`ok..simple timers?`
`22:46.49`	`ManxPower`	`pigpen: your only option unless your telco provides a "signal when call goes to cell voicemail" option`
`22:46.50`	`pigpen`	`I have a customer telling me that Intertel and Cisco support this....`
`22:46.53`	`pigpen`	`I think it is crap.`
`22:47.21`	`ManxPower`	`pigpen: if they "support" it then it is some hack of listening to the audio, since the telco does not provide this infomration`
`22:47.22`	`pigpen`	`k...great answer.`
`22:47.40`	`pigpen`	`k..that is what I thought...just checking....`
`22:47.43`	`pigpen`	`thanks again.`
`22:47.49`	`ManxPower`	`And if you want an ugly and unreliable hack for detecting answering machines, well there's supposed to be a link to such an app on the Wiki`
`22:48.21`	`pigpen`	`thanks.`
`22:53.52`	`PioneerVM2`	`Manx: http://www.pastebin.ca/501219`
`22:53.54`	`*** part/#asterisk Fieldy (i=12Wjuuwn@gentoo/contributor/Fieldy)`
`22:54.04`	`PioneerVM2`	`i am now child free and can devote full attention`
`22:54.17`	`PioneerVM2`	`I wrote up a short, succint, and complete test of both SIP/IAX`
`22:57.01`	`PioneerVM2`	`manx: u still here?`
`22:57.38`	`[TK]D-Fender`	`WindBack, lookup CDR on the WIKI`
`22:58.23`	`*** join/#asterisk Joss^ (n=dotcommu@84.229.80.247)`
`22:58.49`	`PioneerVM2`	`and d-f if u dont mind looking as well`
`23:06.33`	`ManxPower`	`PioneerVM2: have you tried this with any OTHER provider.`
`23:06.44`	`ManxPower`	`I suspect VoicePulse is just not letting you set the number`
`23:07.13`	`ManxPower`	`Also, exactly where are you seeing the callerid? on an analog phone connected to the PSTN outside of Asterisk?`
`23:07.19`	`*** join/#asterisk davidcsi (n=davidcsi@53.red-82-158-35.user.auna.net)`
`23:09.35`	`davidcsi`	`anyone knows whether idefisk supports symmetric RTP`
`23:11.59`	`*** join/#asterisk duckz (n=duckz@141.85.3.18)`
`23:13.06`	`*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net)`
`23:14.38`	`ManxPower`	`for devoting his attention to the problem, PioneerVM2 sure is taking a long time to respond`
`23:15.09`	`ManxPower`	`I give up`
`23:15.12`	`*** part/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)`
`23:15.50`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`23:17.34`	`lee_is_me`	`has anyone had problems with Aastra phones and MWI? Most say that it works out of the box, but I have yet to get it working on some 9112's.`
`23:25.32`	`pigpen`	`so has anyone done/know anything with the usb port on the polycom 650?`
`23:26.40`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com)`
`23:27.50`	`MindTheGap_`	`anyone running asterisk-gui (the one from asteriskNow) on a 1.4 asterisk server using realtime (ldap) sip.conf? would it work?`
`23:35.22`	`*** join/#asterisk Swat2 (n=bler@218-215-192-135.people.net.au)`
`23:38.16`	`[TK]D-Fender`	`pigpen, No real point yet`
`23:40.28`	`pigpen`	`yeah...just thinking maybe a usb headset, ringtone access....maybe a special port to take over 3rd world countries...you know...the American Dream....`
`23:42.33`	`*** join/#asterisk MindTheGap_ (n=iote@BHE201062133118.res-com.wayinternet.com.br)`
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`23:44.02`	`*** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM]`
`23:44.02`	`*** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal)`
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`23:44.02`	`*** join/#asterisk Vec (n=Vec@dsl-243-119-36.telkomadsl.co.za) [NETSPLIT VICTIM]`
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`23:44.02`	`*** join/#asterisk JT_ (n=jon@unaffiliated/jt)`
`23:44.03`	`*** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net)`
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`23:44.03`	`*** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) [NETSPLIT VICTIM]`
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`23:44.03`	`*** mode/#asterisk [+o Corydon76-home] by irc.freenode.net`
`23:44.29`	`MGSsancho`	`hello`
`23:44.32`	`*** join/#asterisk fnordus (n=dnall@24.85.128.203)`
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`23:48.04`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`23:52.59`	`rikstah`	`tzafrir, ping`
`23:57.36`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com)`

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