IRC log for #asterisk on 20080103

`00:00.18`	`riddlebox`	`do you guys suggest having your data on a separate network than your voice? I will be talking to a guy tonight about putting in data and voice and want to see if I can get him to bite on an asterisk system`
`00:00.52`	`NovceGuru`	`define seperate network`
`00:00.52`	`fujin`	`I do.`
`00:00.52`	`fujin`	`seperate being vlan (802.1q)`
`00:01.05`	`fujin`	`redundant switch pathing, and n+1 routers`
`00:01.08`	`riddlebox`	`I was thinking one switch for data and a different switch for voip`
`00:01.34`	`NovceGuru`	`riddlebox: how big of a setup`
`00:01.45`	`riddlebox`	`I think about 20 phones`
`00:01.50`	`fujin`	`most phones support 802.1q trunking, as such I went the route of cisco 3560's, 24 port PoE's, they carry both VLAN's down asingle wire, the phoen breaks out the connection to the PC`
`00:01.55`	`fujin`	`so 1 wire, 1 phone, 1 pc`
`00:02.23`	`NovceGuru`	`thats nice`
`00:02.37`	`fujin`	`well, most 'good' phones`
`00:02.42`	`fujin`	`(we went with linksys spa942's)`
`00:02.50`	`fujin`	`but I would go with polycom ip330's now,`
`00:02.59`	`NovceGuru`	`I would say any smaller I would put them on the same network and just use vlan (or not, depends) but deffintly QoS`
`00:03.23`	`fujin`	`yeah, expedited forwarding sip/rdp`
`00:04.13`	`*** join/#asterisk coppice (n=chatzill@137.192.17.210.dyn.pacific.net.hk)`
`00:05.30`	`riddlebox`	`so any smaller than 20 you think it would be ok to put them on one network?`
`00:05.43`	`fujin`	`with qos, sure`
`00:05.58`	`fujin`	`and qos capable devices`
`00:08.13`	`CrashSys`	`Sometimes I realllly wish I knew C`
`00:09.41`	`riddlebox`	`I guess the other question would be, how they use their network too, right, if they pull down big files and stuff like that?`
`00:10.41`	`outtolunc`	`multicast multiple dvd's <G>`
`00:10.53`	`[TK]D-Fender`	`riddlebox, Save on the Cisco router and passthrough concerns and just get a Linksys PoE Switch and IP 320's instead. It'll pay for the difference in equipment savings and offer greater flexibility down the road`
`00:11.18`	`jblack`	`Is it possible for extensions.conf to load another config file? I want to make a telemarketer hell dialplan, but I don't want to load my real dialplan with 1800 lines of garbage`
`00:11.43`	`[TK]D-Fender`	`jblack, #include "otherconfig.cfg"`
`00:11.47`	`jblack`	`Awesome`
`00:12.11`	`jblack`	`My daughter and I are going to make it a project.`
`00:12.20`	`riddlebox`	`[TK]D-Fender, cool I will do that`
`00:12.39`	`[TK]D-Fender`	`jblack, I just feed mine to the weasels`
`00:12.51`	`De_Mon`	`[TK]D-Fender what? #include?`
`00:12.56`	`riddlebox`	`[TK]D-Fender, I feed mine to screaming monkeys`
`00:12.57`	`NovceGuru`	`So do you guys think once my PSTN number is transfered to a VOIP provider, I could then transfer to a provider that cant port my number directly from the PSTN provider?`
`00:13.07`	`[TK]D-Fender`	`De_Mon, telemarketers silly!`
`00:13.10`	`coppice`	`[TK]D-Fender: you feed weasels to weasels?`
`00:13.21`	`De_Mon`	`[TK]D-Fender I use include => somefile.conf myself...`
`00:13.23`	`[TK]D-Fender`	`coppice, Cannibalism at its best :)`
`00:13.25`	`riddlebox`	`NovceGuru, broadvoice can take up to a month to port your number`
`00:13.28`	`[TK]D-Fender`	`coppice, I like the irony`
`00:13.44`	`NovceGuru`	`riddlebox: they seem to be the only ones that can port my number :(`
`00:13.45`	`[TK]D-Fender`	`De_Mon, that include format is to include a CONTEXT.`
`00:13.51`	`jblack`	`Yeah, I'll put weasels in there... after they go through 50 prompts of "If you are calling from Alabama, please press 1. If you are calling from Alaska, please press 2"`
`00:13.52`	`[TK]D-Fender`	`De_Mon, he wants to include a FILE <-`
`00:14.00`	`NovceGuru`	`I wondered if once with them, I could transfer to someone say, voicepulse`
`00:14.47`	`riddlebox`	`NovceGuru, I like broadvoice they always did just fine for me, I think I am going to port my cell number to them when my plan goes up`
`00:14.50`	`De_Mon`	`oh, humm I think I've misplaced my brain... have you seen it by chance?`
`00:14.57`	`_ShrikE`	`NovceGuru: The port process is not trivial and will take a few weeks for most carriers.`
`00:15.46`	`NovceGuru`	`riddlebox: I have one client that uses them, they seem to have been ok. I was just gonna tranfer to voicepulse once I got it ported to broadvoice since I already have a voicepulse account`
`00:16.20`	`NovceGuru`	`and 4 lines to play with on there, and there's like a $1/month charge for a number`
`00:16.55`	`riddlebox`	`I need to look at voicepulse, I want to port my cell number and have asterisk use it so that when someone calls it, it rings my house phones and my work cell`
`00:17.33`	`NovceGuru`	`thats exactly what I'm doing atm`
`00:17.58`	`NovceGuru`	`with a different number, was just gonna port my parents number, but voicepulse cant port it`
`00:18.05`	`De_Mon`	`yikes! that could be bad(tm)`
`00:18.25`	`NovceGuru`	`then have it ring both their cellphones and get naked dsl`
`00:18.39`	`riddlebox`	`De_Mon, what could be bad?`
`00:18.39`	`[TK]D-Fender`	`riddlebox, do a separate metered test. If they answer the call before ringing your cell, you're DOA.. also if you cell is off it's VM will steal calls from your home IMMEDIATELY.`
`00:19.16`	`mvanbaak`	`zzzzzzzzzzzzzzzzz time`
`00:19.18`	`mvanbaak`	`latero all`
`00:19.38`	`NovceGuru`	`could have it ring the house, then ring your cell`
`00:19.42`	`NovceGuru`	`or whatever order you want it in`
`00:20.08`	`riddlebox`	`[TK]D-Fender, yeah, I really just want to keep the number, since I have had it for 7 years, but work is paying for a cellphone with text messaging and unlimited data`
`00:20.24`	`dacs`	`~context`
`00:20.24`	`jbot`	`it has been said that context is like LaTeX but less messy and more oriented to DTP instead of academics.`
`00:20.37`	`NovceGuru`	`im paying $11/month for 4 "channels" and 1 number`
`00:20.53`	`NovceGuru`	`a call in then out to your home phone and cell phone takes 3 of those channels`
`00:21.14`	`NovceGuru`	`unless your home phone is a sip/iax client of the * system`
`00:21.23`	`[TK]D-Fender`	`dacs, ......`
`00:21.30`	`[TK]D-Fender`	`dacs, .... Chapter 5 ;)`
`00:21.42`	`craigk`	`quick zaptel question ... I am using an analog phone connected to a zaptel card to place a call out a PSTN line connected to the same card. The problem is where I am calling wants me to enter data separated by the # key - but the # key does not seem to be sent ... any ideas ?`
`00:22.29`	`dacs`	`[TK]D-Fender: am still in 4`
`00:22.31`	`dacs`	`:)`
`00:22.35`	`dacs`	`got busy at work`
`00:22.44`	`[TK]D-Fender`	`craigk, watch out for the "tT" parameters in Dial stealing your DTMF for transfers.`
`00:23.07`	`NovceGuru`	`_ShrikE: I know it takes a while, but since voicepulse cant directly port my number, I wondered if they could _after_ it gets ported to broadvoice`
`00:23.16`	`craigk`	`[TK]D-Fender: I do have thsoe set ... but have redefined them to not use # key`
`00:23.18`	`_ShrikE`	`probably not`
`00:23.21`	`[TK]D-Fender`	`dacs, You rally shouldn't try to kid yourself about doing this right. Just go read the chapter already`
`00:23.36`	`[TK]D-Fender`	`dacs, its right there.`
`00:23.54`	`NovceGuru`	`_ShrikE: the more I repeated that the more wrong it sounded`
`00:23.59`	`[TK]D-Fender`	`craigk, Oh? I don't recall the ability to set the char for basic transfers.`
`00:24.32`	`*** join/#asterisk MrFollies (n=Miranda@60-242-243-193.static.tpgi.com.au)`
`00:24.47`	`riddlebox`	`[TK]D-Fender, did you see that I got my call-waiting to work, but I havent figured out how to transfer on my cordless phone for some reason`
`00:25.02`	`craigk`	`[TK]D-Fender, in features.conf i set various values in the featuremap section. The new values work so I assuemd that the default # values are not being used`
`00:25.56`	`_ShrikE`	`NovceGuru: Did voicepulse tell you why the couldnt port that number in?`
`00:26.14`	`[TK]D-Fender`	`craigk, Where in there did you ever see transfer defined anyways?`
`00:26.45`	`[TK]D-Fender`	`craigk, I don't see "transfer" in there....`
`00:27.12`	`[TK]D-Fender`	`craigk, straight transfer is hard-coded in app_dial last I checked. "show application dial"`
`00:27.13`	`craigk`	`[TK]D-Fender: I set disconnect, atxfer, blindxfer, and parkcall`
`00:27.44`	`[TK]D-Fender`	`craigk, looking at it now...`
`00:28.05`	`[TK]D-Fender`	`craigk, hrm.`
`00:28.16`	`craigk`	`[TK]D-Fender: show application dial gives me for tT "Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf"`
`00:28.33`	`craigk`	`[TK]D-Fender: so i redefined them to free up # key :)`
`00:28.44`	`[TK]D-Fender`	`craigk, yup, seems to be...`
`00:29.05`	`[TK]D-Fender`	`craigk, pastebin your features.conf`
`00:30.49`	`dacs`	`got to go , will talk to you later`
`00:30.52`	`dacs`	`bye all`
`00:30.52`	`craigk`	`[TK]D-Fender: pastebin'd http://pastebin.com/m349f127a`
`00:30.58`	`craigk`	`later dacs :)`
`00:32.11`	`craigk`	`[TK]D-Fender I am 'assuming' that the # key is not being sent, because if i call using a phone connected direct to the PSTN line then everything works, but when i go via asterisk/zaptel then the menu i am interacting with just times out instead of accepting my data`
`00:33.07`	`[TK]D-Fender`	`craigk, does "show features" compare properly with your config?`
`00:34.41`	`craigk`	`[TK]D-Fender show features tells me <sorry about this>:`
`00:34.44`	`craigk`	`Builtin Feature Default Current`
`00:34.56`	`craigk`	`--------------- ------- -------`
`00:35.00`	`craigk`	`Pickup 8 8`
`00:35.02`	`*** join/#asterisk nny_1 (n=Scott@64.20.138.159.dyn-e-pool11.pool.hargray.net)`
`00:35.04`	`craigk`	`Blind Transfer # 1`
`00:35.08`	`craigk`	`Attended Transfer *1`
`00:35.12`	`craigk`	`One Touch Monitor`
`00:35.16`	`craigk`	`Disconnect Call * *0`
`00:35.20`	`*** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.mn.comcast.net)`
`00:35.20`	`craigk`	`Park Call *7`
`00:35.24`	`craigk`	`Dynamic Feature Default Current`
`00:35.27`	`*** join/#asterisk vrtk (n=vyrotiko@201.9.57.7)`
`00:35.28`	`craigk`	`--------------- ------- -------`
`00:35.32`	`craigk`	`(none)`
`00:35.36`	`craigk`	`Call parking`
`00:35.39`	`nny_1`	`~pastebin`
`00:35.39`	`jbot`	`[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`00:35.40`	`craigk`	`------------`
`00:35.44`	`craigk`	`Parking extension : 70`
`00:35.44`	`craigk`	`Parking context : parkedcalls`
`00:35.44`	`craigk`	`Parked call extensions: 71-79`
`00:35.44`	`craigk`	`<apologies all for that paste>`
`00:35.50`	`nny_1`	`craigk: stop`
`00:35.55`	`nny_1`	`craigk: ~pastebin`
`00:36.01`	`CrashSys`	`Ohh the humanity of it all`
`00:36.30`	`nny_1`	`any other aastra geeks here?`
`00:37.34`	`riddlebox`	`weird I cannot transfer ** from my analog phone`
`00:37.57`	`craigk`	`[TK]D-Fender so show features seem to indicate that the default value of blind transfer is #, but that i have redefined it to be 1 ... again, i assumed this means that # is no longer trapped`
`00:37.59`	`Sauron	`wtf is aastra?`
`00:38.24`	`[TK]D-Fender`	`craigk, yeah its all starting to look "kosher".`
`00:38.40`	`[TK]D-Fender`	`Sauron, a very popular maker of SIP phones
`00:39.54`	`nny_1`	`yeah trying to discern if i can tell a 480i ct to stop trying to hit a config server.. every boot yields different results.. it hangs.. (reseting to factory defaults). I seem to remember the web config stuff can conflict with the stuff loaded by provisioning.. (long story I would provision if all the factors were right)`
`00:40.13`	`vrtk`	`is this the ultimate resource for using SRTP on Asterisk: http://www.e164.org/wiki/AsteriskSRTP ?`
`00:42.03`	`craigk`	`[TK]D-Fender: does that mean i have it configured correctly, but for some reason it is not working ? maybe there is some way i can see what chan_zap is doing with my DTMF ... guess i am off to read the code :)`
`00:42.05`	`nhuisman_work`	`how can I figure out what zaptel version I have installed?`
`00:42.19`	`nhuisman_work`	`is there some -V flag somewhere?`
`00:42.35`	`nhuisman_work`	`this is asterisk business edition and it masks it with the business edition version`
`00:42.43`	`[TK]D-Fender`	`craigk, Yeah I'm a bit confused on why this wouldn't work. For a sanity check, remove all dial options and test.`
`00:42.47`	`nhuisman_work`	`when I use the package manager to look`
`00:42.54`	`[TK]D-Fender`	`nhuisman_work, "ztcfg -v`
`00:43.01`	`craigk`	`[TK]D-Fender great suggestion, thanks .. .trying now`
`00:43.03`	`Sauron	`oh`
`00:43.07`	`nhuisman_work`	`nope`
`00:43.36`	`nhuisman_work`	`[TK]D-Fender, didn't display a version`
`00:43.40`	`[TK]D-Fender`	`nhuisman_work, well the version it froze at should be documented somewhere.`
`00:43.51`	`[TK]D-Fender`	`nhuisman_work, then again very few of us here use ABE`
`00:43.54`	`nhuisman_work`	`yeah I know`
`00:44.07`	`nhuisman_work`	`the same goes for asterisk`
`00:44.13`	`nhuisman_work`	`is there a way to display that version?`
`00:44.25`	`nhuisman_work`	`nm the overwrote it`
`00:44.33`	`nhuisman_work`	`asterisk -V just displays the business edition version`
`00:45.26`	`craigk`	`[TK]D-Fender that did it, without the rTWK flags it works fine`
`00:45.39`	`tzafrir_home`	`modinfo zaptel \| grep ^version`
`00:46.00`	`tzafrir_home`	`or: cat /sys/module/zaptel/version`
`00:46.29`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b78c56f782b5f88d)`
`00:46.30`	`*** join/#asterisk Maliuta_ (n=nikolai@203.201.152.211)`
`00:46.47`	`[TK]D-Fender`	`craigk, well sounds like tis time to start layering your tests`
`00:46.49`	`nhuisman_work`	`boy they are tricky, they even removed the version in the modinfo`
`00:47.11`	`tzafrir_home`	`nhuisman_work, switchvox?`
`00:47.13`	`nhuisman_work`	`digium`
`00:47.16`	`[TK]D-Fender`	`nhuisman_work, So... you you have sold a smaller pieve of your immortal soul? :)`
`00:47.29`	`nhuisman_work`	`well it seems like it has tdmoe support so I guess it's fine`
`00:47.48`	`tzafrir_home`	`nice`
`00:48.22`	`tzafrir_home`	`nhuisman_work, modinfo xpp`
`00:48.38`	`nhuisman_work`	`no such module`
`00:48.45`	`tzafrir_home`	`even better`
`00:51.16`	`nny_1`	`anyone using res_snmp successfully in here yet?`
`00:51.29`	`craigk`	`[TK]D-Fender so if I ahve any of the T, W or K flags it does not work :(. I was hoping that it was going to be one flag/feature in particular but it does not look like it.`
`00:52.22`	`[TK]D-Fender`	`craigk, now start commenting in/out options in features.conf`
`00:54.04`	`craigk`	`[TK]D-Fender but if i comment out blind transfer, that will revert to the default of # :(`
`00:54.17`	`[TK]D-Fender`	`craigk, try, then start messing with it.`
`00:56.24`	*** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com)
`00:57.35`	`nny_1`	`so i am trying to change the local dial plan to allow (essentially) 1917145551212 and 197145551212 and 195551212 in this aastra phone... i add (because god forbid they make a wildcard) 19xxxxxxxxxxx 19xxxxxxxxxx and 19xxxxxxx and the damn phone won't register...`
`00:57.46`	`nny_1`	`lol yeah smores!`
`00:58.45`	`nny_1`	`i mean regardless of what is or isn't right with it.. why would a local dialplan entry cause the phone not to boot.. it's not like that kind of state is something you could diagnose..`
`00:58.52`	`nny_1`	`er not boot, register`
`01:03.28`	`*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)`
`01:03.57`	`*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)`
`01:04.24`	`craigk`	`[TK]D-Fender commenting out the features one at a time did not help. I left blindxfer to the end, and when I pressed the # key it started a transfer instead of sending the DTMF. So it seems there is no way to turn the features off completely. I will look into the code and see what I can see.`
`01:06.18`	`[TK]D-Fender`	`craigk, So * is getting "grabby"`
`01:18.59`	`De_Mon`	`so um... what would you expect this to return? exten => s,n,NoOp($["test string" : "(.*)s"])`
`01:21.44`	`*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)`
`01:24.44`	`craigk`	`[TK]D-Fender i think so .... but so far there is nothign obvious in the code (not that i am an expert, but i _think_ i have found where the dtmf is being handled at least ... nearly there :) )`
`01:27.51`	`nny_1`	`so someone poke a cell for me here.. i have added 1[2-9]xxxxxxxxxx as well as 19xxxxxxxxxx and the phone keeps cutting the user off at the last digit.. any other sip client can dial out that way, so i don't suspect a dialplan issue, as they are all in the same contexts...`
`01:27.56`	`nny_1`	`i get == Spawn extension (from-internal, 19184338472, 2) exited non-zero on 'SIP/141-b760d980'`
`01:28.08`	`nny_1`	`er`
`01:28.18`	`nny_1`	`(yay for spaz clients testing)`
`01:28.30`	`nny_1`	`i have an example of the actual number i told him to dial one sec`
`01:28.38`	`*** join/#asterisk clusco (i=clusco@203.114.50.232)`
`01:28.51`	`nny_1`	`<PROTECTED>`
`01:28.51`	`nny_1`	`<PROTECTED>`
`01:28.51`	`craigk`	`[TK]D-Fender hmmm, i am running asterisk 1.4.14, res_feature.c Revision: 89248. On line 1076 I can see where it is decided to sotre the feature digits instead of passing them through (in my case, this means store # instead of passing it on) ... still looking but i think this might be the error`
`01:28.57`	`clusco`	`g00d morning!`
`01:29.01`	`nny_1`	`so it is cutting off the last 2`
`01:29.08`	`nny_1`	`this must* be int he aastra local dial plan`
`01:29.34`	`nny_1`	`any thoughts appreciated.. gonna have a smoke and marinate on it`
`01:30.10`	`[TK]D-Fender`	`nny_1, no, thats not the same channel at all.`
`01:30.26`	`rob0`	`s/mar/ur/`
`01:30.33`	`[TK]D-Fender`	`nny_1, SIP/141-b7609268 != SIP/141-b760d980`
`01:37.40`	`craigk`	`[TK]D-Fender forget what i said before ... this is only an issue when i use zaptel ... so there is nothing 'wrong' with res_feature.c ... changing to look at chan_zap.c now :)`
`01:38.11`	`[TK]D-Fender`	`craigk, Actually I have concerns with app_dial directly....`
`01:38.25`	`[TK]D-Fender`	`craigk, As that's what uses it.`
`01:39.01`	`craigk`	`[TK]D-Fender oh ... i guess i can not definitivly say that it works with SIP as my SIP phone uses # internally and does not send it :/`
`01:39.22`	`nny_1`	`[TK]D-Fender: hmm the 141-bXXXXXXX seems to change on each call.. i think i may have figured it out.. the aastra may be defaulting to the 19xxxxxxxxx entry before mine.. trying a different angle`
`01:39.58`	`*** join/#asterisk BajaEd (n=ednagy@72.168.135.209)`
`01:40.34`	`[TK]D-Fender`	`nny_1, it IS a different call, that *'d channel name`
`01:40.38`	`[TK]D-Fender`	`*'s`
`01:40.53`	`nny_1`	`gotcha`
`01:41.38`	`nny_1`	`[TK]D-Fender: ahh got it`
`01:42.50`	`nny_1`	`[TK]D-Fender: in the aastra dialplan "2xx#\|[2-9]11\|9911\|1[01]xx\|[2-9]xxxxxx\|1[2-9]xxxxxxxxx\|1[2-9]xxxxxxxxxx\|011x+#\|xx\|xx+#\|xxxxxxxxxxxxxxxxx#"`
`01:42.50`	`nny_1`	`the second 1[2-9]x etc is ignored if you dialed 198435551212 and it defualts to the first one`
`01:43.02`	`nny_1`	`this i did not know`
`01:45.46`	`*** join/#asterisk apardo (n=apardo@107.65.220.87.dynamic.jazztel.es)`
`01:46.42`	`*** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk)`
`01:50.26`	`*** join/#asterisk tengulre (n=tengulre@124.42.50.54)`
`01:52.23`	`De_Mon`	`nny_1 dial patterns always use the FIRST matching pattern`
`01:53.32`	`*** join/#asterisk ariel_ (n=ariel_@c-66-176-41-202.hsd1.fl.comcast.net)`
`01:54.12`	`*** join/#asterisk m160858 (n=m160858@200.106.120.204)`
`01:54.15`	`ariel_`	`hello everyone`
`01:54.21`	`m160858`	`hello`
`01:54.59`	`m160858`	`i need a little support about queues with asterisk 1.4`
`01:55.11`	`ariel_`	`ask the question`
`01:55.14`	`nny_1`	`De_Mon: yeah i am now going on the notion this excersie has had some pebkac in it... I have dealt with minor changes to polycoms etc. in the past, but never had to add one in this matter`
`01:55.44`	`m160858`	`i've a queue with 60 agents`
`01:56.58`	`ariel_`	`a queue with 60 agents that is nice. And?`
`01:57.13`	`m160858`	`with AgenLogin command, the call are quick ... one by one`
`01:58.01`	`m160858`	`but with AgentCallBackLogin ... the calls are slowly`
`01:58.09`	`m160858`	`sorry by my english`
`01:58.38`	`[TK]D-Fender`	`m160858, elaborate on "slowly" please, and provide CLI output in a pastebin for us to see.`
`01:58.39`	`[TK]D-Fender`	`~pb`
`01:58.40`	`jbot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`01:58.59`	`m160858`	`ok`
`01:59.29`	`ariel_`	`~ktmb`
`01:59.49`	`ariel_`	`~weather ktmb`
`02:00.02`	`nny_1`	`De_Mon: i am still learning (and will always be) i have been working with asterisk since august of last year, and until i cough up the dough for boot camp, some lessons come a bit harder`
`02:00.25`	`ariel_`	`boot camp`
`02:00.26`	`ariel_`	`wow`
`02:00.42`	`nny_1`	`alternatively (i have the old book, getting the new one soon) if anyone has any ways to sharpen skills outside of working on systems, i am all ears`
`02:01.14`	`ariel_`	`work, work, on test box, work try things read the wiki`
`02:01.18`	`ariel_`	`help here,`
`02:01.27`	`ariel_`	`google is your friend`
`02:01.41`	`nny_1`	`ariel_: yeah thats what have been doing... even been getting into some of the crazier stuff (snmp etc)`
`02:01.54`	`ariel_`	`nice`
`02:02.01`	`ariel_`	`but I would suggest basic first`
`02:02.12`	`ariel_`	`as it's what will keep you up and running in the long term`
`02:02.15`	`nny_1`	`ariel_: i will say it's by far the most interesting software i have had the pleasure of working with`
`02:02.23`	`ariel_`	`really`
`02:02.34`	`nny_1`	`ariel_: yeah i have a box here we have setup scratch dialplans on multiple times`
`02:02.49`	`ariel_`	`I do have to say it's improved allot since I first started with it. Back a few years ago at version .5`
`02:02.57`	`nny_1`	`ariel_: also reinstalled different os (centos, ubuntu, debian) and installed asteriska nd zap siz ways of sundays`
`02:03.14`	`nny_1`	`yeah my hat goes off to the more experienced developers who dealth with asterisk in it's infancy`
`02:03.17`	`*** join/#asterisk RoyK (n=roy@ip-131-23-149-91.dialup.ice.no)`
`02:03.32`	`nny_1`	`sounds like that includes you ;)`
`02:03.40`	`ariel_`	`me no never`
`02:04.27`	`ariel_`	`just remember there are over 100 different ways to do the same thing in asterisk`
`02:04.28`	`nny_1`	`ariel_: will say i have had the fortune of having a more experienced asterisk guy working with us on projects as a consultant.. i try to not pretend I know enough outside to do basic basic things, break stuff, and be dangerous, at least right now`
`02:04.38`	`ariel_`	`so what might work for someone one way might not work for you.`
`02:04.42`	`nny_1`	`ariel_: :) that's what makes it great`
`02:05.33`	`ariel_`	`yes, there are allot of things that can be done with the systems`
`02:05.38`	`nny_1`	`ariel_: oh i can imagine, we have a client who uses asterisk in some pretty unique ways`
`02:05.49`	`ariel_`	`that is good`
`02:06.51`	`nny_1`	`ariel_: the reality is from my past experience with (nortel, avaya, etc.) this thing runs circles around the customization stuff.. although simplicity does have it's benefits, it's not without costs (literally, after seeing the price tags)`
`02:07.35`	`ariel_`	`Nortel's and Avaya have there place.`
`02:07.40`	`nny_1`	`indeed`
`02:07.44`	`ariel_`	`Asterisk still has allot of growth ahead of it.`
`02:08.21`	`ariel_`	`It's a good product, but there are still many things need to be fixed, upgraded, and worked on.`
`02:08.32`	`nny_1`	`indeed`
`02:08.52`	`nny_1`	`hopefully someday soon i'll be able to do more than just work with it and be able to work on it`
`02:09.08`	`ariel_`	`it's a start`
`02:09.23`	`nny_1`	`hehehe learning nuances now :)`
`02:09.24`	`ariel_`	`where are you located? And seems your working for someone that puts systems up.`
`02:09.38`	`m160858`	`ok ... hello again`
`02:09.44`	`m160858`	`i've this configuration`
`02:09.45`	`m160858`	`http://rafb.net/p/6PlyFv77.html`
`02:09.53`	`nny_1`	`ariel_: south carolina, and i have a small firm that does IT here`
`02:10.40`	`nny_1`	`ariel_: I am an owner, so as much as I focus on asterisk, I also deal with MS sys admin (bleah), linux fileservers, gateways, ddwrt stuff, spyware, pebkac, and just about everything else under the sun`
`02:11.00`	`m160858`	`i need to know why the command AgentLogin() don't work same to the command AgentCallBackLogin()`
`02:11.37`	`*** join/#asterisk bintut (n=bintut@203.125.63.150)`
`02:11.43`	`nny_1`	`I like it though.. I expect to be able to pass some of the lesser duties off here soon to some new recruits, but I am going to focus on * for a bit`
`02:11.49`	`nny_1`	`anyways`
`02:11.51`	`nny_1`	`<PROTECTED>`
`02:11.56`	`nny_1`	`but yeah it's awesome`
`02:12.11`	`ariel_`	`it's hard finding good asterisk people`
`02:12.32`	`nny_1`	`ariel_: indeed.. I am working on a nation support site that hires out to consultants`
`02:12.34`	`ariel_`	`seems all the good ones have to be either home grown or worked with via a consultant.`
`02:12.56`	`nny_1`	`ariel_: it's very very premature but http://ipbxsupport.com is the base`
`02:13.39`	`nny_1`	`ariel_: I would rather pay someone a consultant's fee up front who really knows the system than elsewise. i can always charge accordingly, and still be reasonable or even low priced (gotta be careful with low price, you attract the wrong people)`
`02:13.44`	`ariel_`	`nice,`
`02:14.07`	`ariel_`	`some times it does`
`02:14.41`	`nny_1`	`yeah I have to read those cases individually`
`02:15.26`	`ariel_`	`I used to work for myself. But wife got to me and I have been working for a large call center for almost 2 years now.`
`02:15.26`	`nny_1`	`but nah it's good stuff, and our closest competition locally is a strict MS and cisco shop that has gone elitist.. the rest are all mom and pop stores that do the basics and sell PCs`
`02:15.44`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`02:15.44`	`*** mode/#asterisk [+o russellb] by ChanServ`
`02:15.45`	`ariel_`	`But sometimes I think of going back to working for my self.`
`02:15.50`	`nny_1`	`ariel_: nice what kind of call work?`
`02:16.06`	`ariel_`	`www.elephantgroup.com`
`02:16.17`	`ariel_`	`it's kinda very hard to discrible all we do`
`02:16.49`	`nny_1`	`reading`
`02:16.51`	`nny_1`	`but i understand`
`02:17.09`	`drmessano`	`Do you get free Pureeez water?`
`02:17.17`	`nny_1`	`my old bix partner was a marketing major, learned a lot from him`
`02:17.23`	`m160858`	`hello? somebody help me?`
`02:17.34`	`nny_1`	`m160858: i could try :)`
`02:17.42`	`m160858`	`thanks`
`02:17.52`	`m160858`	`http://rafb.net/p/6PlyFv77.html`
`02:18.15`	`m160858`	`i'm to using 2 command for login`
`02:18.30`	`m160858`	`AgentLogin and AgentCallBackLogin`
`02:18.39`	`m160858`	`AgentLogin works great`
`02:18.42`	`ariel_`	`We just got PureEZ up, in our office we have them setup. At home not yet. Can't keep enought of them in stock yet`
`02:18.59`	`m160858`	`but AgentCallBackLogin ... works slowly`
`02:19.08`	`m160858`	`i don't understand why`
`02:19.17`	`nny_1`	`m160858: i did a core show application AgentLogin and core show application AgentCallbackLogin and it seems there are some differences`
`02:20.14`	`m160858`	`yes, but both are using for login`
`02:20.20`	`nny_1`	`m160858: any output from console when you dial 554 you could post?`
`02:20.40`	`m160858`	`no,i don't have`
`02:20.51`	`m160858`	`the problem is about calls`
`02:20.58`	`nny_1`	`speed you said?`
`02:21.04`	`nny_1`	`er whothefuckami yoda?`
`02:21.06`	`nny_1`	`:)`
`02:21.24`	`nny_1`	`you said callback is slow right?`
`02:21.30`	`m160858`	`i don't speak english .. but i try`
`02:21.35`	`nny_1`	`ahh no problem :)`
`02:21.40`	`nny_1`	`i apparently don't eithetr`
`02:21.46`	`nny_1`	`-_-`
`02:21.49`	`m160858`	`yes`
`02:23.02`	`ariel_`	`m160858, need more info as all your users on same network? You seem to have a basic call setup not local settings for call routes?`
`02:23.15`	`nny_1`	`but yeah could you open an asterisk console (asterisk -r), turn up the verbosity of asterisk so it gives you a lot of detail (try core set verbose 5) and ( debug level 5) and c/p the output of what happens when you try AgentCallbackLogin`
`02:23.16`	`m160858`	`the callback is slow ... why? .. i don't understand the reason`
`02:24.06`	`m160858`	`mmmmm .. ok`
`02:24.42`	`m160858`	`ariel: yes, all the user are on the same network`
`02:25.10`	`ariel_`	`in agent call back do they stay logged in?`
`02:25.34`	`ariel_`	`what is the status of each agents when you check there available status`
`02:25.53`	`m160858`	`all available`
`02:26.02`	`[TK]D-Fender`	`m160858, SHOW US`
`02:26.13`	`nny_1`	`:) hope I am not getting in the way`
`02:26.24`	`m160858`	`i can't 4 today`
`02:26.45`	`[TK]D-Fender`	`m160858, Come back when you're in a position to do something about your problem then.`
`02:26.53`	`[TK]D-Fender`	`m160858, if we can't see it, we can't help it.`
`02:27.26`	`m160858`	`ok, not problem`
`02:27.42`	`m160858`	`thanks`
`02:28.09`	`*** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx)`
`02:29.44`	`nny_1`	`m160858: yeah, getting to the console and really getting it to churn out output is usually the best start to digging down on an issue`
`02:31.46`	`*** join/#asterisk RoyK (n=roy@ip-131-23-149-91.dialup.ice.no)`
`02:32.23`	`m160858`	`ok`
`02:33.49`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bcf524d45ecab4c1)`
`02:33.54`	`m160858`	`by otherhand ... you know any console like isymphony but not as expensive?`
`02:34.36`	`*** join/#asterisk suvir (n=suvir@ppp-124.120.239.93.revip2.asianet.co.th)`
`02:35.18`	`*** join/#asterisk Maliuta (n=nikolai@119.11.96.104)`
`02:36.10`	`*** join/#asterisk UserReg_CL (n=COB@pc-117-24-120-200.cm.vtr.net)`
`02:36.33`	`UserReg_CL`	`mmm hi`
`02:37.03`	`nny_1`	`ariel_: http://www.i9technologies.com/isymphony/`
`02:37.05`	`ariel_`	`oh it's just another gui type of setup.`
`02:37.14`	`*** join/#asterisk DaveCanoe (n=Dave@H49.C20.B96.tor.eicat.ca)`
`02:37.29`	`nny_1`	`hmm windows api no less`
`02:38.20`	`nny_1`	`ariel_: to be honest with you, no matter what console you dealt with, nothing would show you (unless this isymphony really has verbose console logging) the information you would need to diagnose an issue of this nature`
`02:38.24`	`nny_1`	`oops mt`
`02:38.32`	`nny_1`	`m160858: that was for you ^`
`02:38.45`	`ariel_`	`m160858, your better off starting with asterisknow`
`02:38.59`	`m160858`	`it's for my client`
`02:39.02`	`[TK]D-Fender`	`nny_1, did YOU see him say he was USING it?`
`02:39.21`	`[TK]D-Fender`	`nny_1, He wouldn't be complaining about its cost if he did <-`
`02:39.36`	`[TK]D-Fender`	`nny_1, now you need to focus on the small print :)`
`02:39.37`	`m160858`	`they want other console`
`02:39.40`	`nny_1`	`[TK]D-Fender: nah didn't mean to make it sound like i did.. :)`
`02:40.04`	`m160858`	`Dislike FOP`
`02:40.15`	`nny_1`	`my bad, thought he was trying to address his issue with it`
`02:40.31`	`[TK]D-Fender`	`m160858, perhaps you should be clearer about what it is you are looking for exactly. the term "console" is not enough to imply what you are looking for.`
`02:41.13`	`m160858`	`oh ... sorry`
`02:41.33`	`m160858`	`software for managing phone calls via the Open Source Asterisk platform`
`02:41.47`	`m160858`	`on windows`
`02:41.47`	`[TK]D-Fender`	`m160858, now manage in what way?`
`02:42.40`	`ariel_`	`m160858, depends on how many users, queues, reports. AsteriskNow is a good start. Also switchvox, Or depending on how you see things even freepbx is a good way to start without much cost.`
`02:42.41`	`m160858`	`any`
`02:43.03`	`[TK]D-Fender`	`m160858, Again we need SPECIFICS. define "manage".`
`02:43.14`	`[TK]D-Fender`	`m160858, These mean different things to different people.`
`02:43.14`	`ariel_`	`yike`
`02:43.33`	`nny_1`	`only a suggestion, damned if i have been able to get it working`
`02:43.39`	`m160858`	`no, i need more ... that hudlite`
`02:43.51`	`m160858`	`i have 200 users`
`02:44.08`	`[TK]D-Fender`	`m160858, how about telling us? You seem to keeping yours needs awefully secret and badgering it out of you is getting tiring...`
`02:44.49`	`m160858`	`4 queues ... i need to see the calls waiting, agents logged, channels availables`
`02:44.55`	`ariel_`	`200 users it's not a system your going to be able to see the users like a FOP or Hublite.`
`02:45.24`	`m160858`	`i know`
`02:45.58`	`[TK]D-Fender`	`m160858, indeed Custom may be worth it.`
`02:46.17`	`*** join/#asterisk UnixDog (n=unixdog@adsl-69-234-208-201.dsl.irvnca.pacbell.net)`
`02:46.24`	`*** part/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net)`
`02:46.38`	`UnixDog`	`when is asterik going to move from 8 khz to 16khz ?`
`02:46.43`	`ariel_`	`users... hummm we have well over 300 plus so there is no off the shelve system that will give us what we need.`
`02:46.51`	`nny_1`	`yeah i have a bid for close to 2000 phones in the works, and we have already decided we are gonna have to hire some help to work on the backend interface`
`02:47.10`	`UnixDog`	`?`
`02:47.15`	`UnixDog`	`interface for what`
`02:47.19`	`[TK]D-Fender`	`UnixDog, Since the pstn uses 8khz we don't really care.`
`02:47.19`	`UnixDog`	`pbx`
`02:47.28`	`ariel_`	`2000 phones... well having phones does not equal queues and agents.`
`02:47.38`	`UnixDog`	`but pstn is not the say all in voip`
`02:47.54`	`UnixDog`	`and native audio is 16khz`
`02:47.57`	`nny_1`	`yeah not the same as a call center, more of a apartment complex/ resort / office front end`
`02:48.06`	`[TK]D-Fender`	`UnixDog, Native to WHO?`
`02:48.20`	`[TK]D-Fender`	`UnixDog, thats just another rate like another other.`
`02:48.25`	`ariel_`	`nny_1, you don't need much of a gui or setup for that`
`02:48.28`	`nny_1`	`just a pretty way for the managers to parse the cdr data and print out pdfs..`
`02:48.34`	`ariel_`	`it's fairly simple what your looking at`
`02:48.40`	`nny_1`	`ariel_: yeah the asteriskcdr stuff works well so far`
`02:48.46`	`russellb`	`"native audio is 16 kHz" ... where did you get that from?`
`02:48.57`	`[TK]D-Fender`	`russellb, I roasted him first!`
`02:49.07`	`russellb`	`oh, darn`
`02:49.08`	`UnixDog`	`most audio files nowdays I pull are 16khz and endup having to be resampled for only asterisk to 8khz`
`02:49.22`	`russellb`	`oh well, besides, asterisk 1.4 has G.722 passthrough support`
`02:49.24`	`UnixDog`	`there is no roasting`
`02:49.31`	`russellb`	`asterisk trunk can transcode g.722`
`02:49.35`	`m160858`	`yes, but i need any visual environment 4 supervisors`
`02:49.43`	`russellb`	`i just merged a new channel driver, chan_console, that operates in 16 kHz natively ...`
`02:49.48`	`[TK]D-Fender`	`UnixDog, Oh, YOU'RE the standard! Its all so much clearer to me now! :)`
`02:49.53`	`russellb`	`and wrote a new module, codec_resample, which resamples between 8 kHz and 16 kHz`
`02:49.58`	`russellb`	`so, it already does.`
`02:50.04`	`ariel_`	`m160858, I have over 70 different queues and about 30 managers`
`02:50.30`	`UnixDog`	`no I am not the real world is applications like sphinx and recordit and most others are all native 16khz`
`02:50.32`	`ariel_`	`an off the shelve product is not going to work out. You need a ture setup configured to your needs and call center views.`
`02:50.57`	`[TK]D-Fender`	`UnixDog, And when you think about it are your 16khz samples.... STEREO? Because even G.722 isn't.`
`02:51.03`	`ariel_`	`UnixDog, mono`
`02:51.22`	`russellb`	`hey, i answered the question, asterisk already supports it`
`02:51.27`	`m160858`	`ariel: i dont understand ur comment`
`02:51.30`	`russellb`	`no reason to continue arguing about it :)`
`02:52.06`	`russellb`	`~flog [TK]D-Fender`
`02:52.06`	`jbot`	`ACTION whips [TK]D-Fender with a cat-5 o' nine tails`
`02:52.11`	`UnixDog`	`ok when will it be merged into a release version`
`02:52.22`	`ariel_`	`m160858, with 200 users for a call center and 4 plus managers that will be changing adding agents around you will not be able to get that with a predone off the selve product`
`02:52.24`	`russellb`	`when 1.6 is released, which will be the first quarter of this year`
`02:52.25`	`[TK]D-Fender`	`russellb, sticks & stones, whips and chains, whee!`
`02:52.40`	`[TK]D-Fender`	`russellb, z0mg LEAK!!!!`
`02:52.46`	`russellb`	`:-X`
`02:53.28`	`russellb`	`i intended to start the process a couple months ago`
`02:53.31`	`russellb`	`but have had to hold off .....`
`02:53.42`	`UnixDog`	`astgui aka vikidial`
`02:54.01`	`UnixDog`	`is good for callcenters`
`02:54.01`	`ariel_`	`astgui/vicidial has a very large short coming`
`02:54.08`	`UnixDog`	`yes`
`02:54.11`	`ariel_`	`it's uses meetme for calls yuke`
`02:54.17`	`UnixDog`	`I know I tried setting it up once`
`02:54.21`	`UnixDog`	`no`
`02:54.24`	`ariel_`	`yes`
`02:54.27`	`UnixDog`	`it uses app confrence`
`02:54.35`	`ariel_`	`yes same shit`
`02:54.46`	`*** join/#asterisk Maliuta_ (n=nikolai@203.201.152.211)`
`02:55.04`	`[TK]D-Fender`	`app_meetme & app_conference all become nearly defunct with 1.6 :)`
`02:55.11`	`UnixDog`	`ok`
`02:55.19`	`m160858`	`but it consumes many HW`
`02:55.24`	`UnixDog`	`cant wait to see what they use for confrencing in 1.6`
`02:55.47`	`ariel_`	`UnixDog, we have 5 dialers working on asterisk with allot of trafic. AstGui could not even come close`
`02:55.59`	`UnixDog`	`what dialer`
`02:56.23`	`UnixDog`	`m160858, what phones are you planning to use`
`02:56.48`	`m160858`	`just softphones`
`02:57.00`	`UnixDog`	`if I where you I would look at a asterisk server + fxs channel banks`
`02:57.01`	`UnixDog`	`ok`
`02:57.06`	`ariel_`	`softphones..... ahh there windows phone I bet.`
`02:57.30`	`*** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com)`
`02:57.32`	`UnixDog`	`you can do a server + 200 fxs for about 10 grand`
`02:57.56`	`UnixDog`	`and then add a few hunderd on for manager sip phones`
`02:58.19`	`ariel_`	`humm putting two 4 port 412 boards.... into one box..... hummm`
`02:58.27`	`nny_1`	`anyone know of a sip client for macs?`
`02:58.31`	`UnixDog`	`xorcom channel banks`
`02:58.37`	`russellb`	`nny_1: xlite`
`02:58.42`	`nny_1`	`russellb: thanks`
`02:58.42`	`russellb`	`or whatever they call it these days ...`
`02:58.46`	`russellb`	`nny_1: or asterisk ;)`
`02:59.15`	`nny_1`	`russellb: hehe`
`02:59.21`	`ariel_`	`My frist setup with with 7 adtrans 750's just for that type of setup.`
`03:00.06`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a46a422edadd3aff)`
`03:00.12`	`UnixDog`	`I just did a 75 exten setup for a client and I hated doing it`
`03:00.20`	`UnixDog`	`they where so cheap about it`
`03:00.44`	`ariel_`	`nny_1, I have a customer that I setup in CA with 4 TE410e in a box running many channel banks for an old office building. It works.`
`03:01.06`	`craigk`	`[TK]D-Fender so, I found my problem with zaptel and the missing # .... <hangs head in shame> I was pressing the keys to dman fast, it was dropping keys :)`
`03:01.10`	`UnixDog`	`they used gxp1200 for 70 desk ad a gxp2020 for the front desk and the rest got gxp2010`
`03:01.31`	`[TK]D-Fender`	`craigk, that'll learn' ya! :p`
`03:01.51`	`UnixDog`	`but thanks gawd they did not use trashbox`
`03:01.58`	`UnixDog`	`I would have choked`
`03:02.03`	`nny_1`	`wow http://kerneltrap.org/Linux/Dusting_Off_the_0.01_Kernel`
`03:02.31`	`craigk`	`[TK]D-Fender :). I eventually worked out how to get debugging output showing what dtmf keys were being handled ... i could see some missing if i was too fast. Time for me to change to decaf and slow down a bit me thinks`
`03:02.52`	`nny_1`	`ariel_: yeah i am waiting for the day i get an install that requires integration with older phones.. looked at the xircom channel banks..`
`03:03.54`	`[TK]D-Fender`	`nny_1, non recyclable non redundant. I wouldn't touch'em`
`03:03.57`	`ariel_`	`nny_1, xircom are ok for small setups. But for larger buildings your better off with a real good channel banks like Adtran`
`03:04.17`	`UnixDog`	`I am getting ready to do a install with 5 32 channel fxs xorcom units and 1 32 channel xorcom server and 12 polycoms 1 601+sidecars`
`03:04.36`	`nny_1`	`[TK]D-Fender: i tend to agree.. and hope it never ever happens,`
`03:04.36`	`UnixDog`	`wich adtran`
`03:04.55`	`nny_1`	`[TK]D-Fender: i think the general feeling right now is, either go all the way or let em go lesewhere`
`03:04.59`	`ariel_`	`most of the ones I have used are the 750 and 850`
`03:05.53`	`m160858`	`bye`
`03:05.57`	`m160858`	`and thanks`
`03:06.38`	`ariel_`	`my heats are going down in flames.... It's been a bad year for south florida teams....`
`03:08.50`	`[TK]D-Fender`	`nny_1, Zaptel can flake out. Thats why I advise SIP gateways. Far more flexible and take all the load off of *`
`03:09.48`	`hmmhesays`	`absolutely`
`03:10.35`	`[TK]D-Fender`	`no more dial options to fiddle with, single user interface, multi-homes servers, etc.`
`03:11.39`	`ariel_`	`[TK]D-Fender, and what large sip gateways do you recommend?`
`03:12.14`	`[TK]D-Fender`	`ariel_, Mediatrix 1124 or AudioCodes MP-124. For smaller installs Linksys SPA-8000`
`03:12.32`	`hmmhesays`	`1124's pretty much rock`
`03:12.48`	`hmmhesays`	`i've used them heavily in the past`
`03:12.52`	`[TK]D-Fender`	`hmmhesays, they are dead easy...`
`03:13.05`	`ariel_`	`I need something in the DS3 range`
`03:13.22`	`UnixDog`	`TDM0E`
`03:13.40`	`ariel_`	`ha`
`03:13.57`	`UnixDog`	`just not usb channel banks please`
`03:14.08`	`UnixDog`	`small sites maybe`
`03:14.15`	`UnixDog`	`day no more the 6 lines`
`03:14.58`	`ariel_`	`I have 2 asterisk (custom) setup for gateways that can handle 644 channels of sip from Qwest. but need to back them up with some old fashion DS3 setups.`
`03:15.57`	`hmmhesays`	`you could always go with a cms`
`03:16.12`	`hmmhesays`	`by quintum, you have to break your ds3 off into t1's`
`03:16.14`	`ariel_`	`I was looking at an Excel switch`
`03:16.32`	`ariel_`	`lucent but it's 250k to start with`
`03:17.35`	`ariel_`	`To one of our locations which is withing 100 feet of a major L3 pop I can get a great DS3 setup from them. But need to be able to terminate it correctly into our 22 asterisk setups accross the contry`
`03:19.23`	`ariel_`	`it's so hard to scale up our systems as it is. Need to figure out how to setup 2 more call centers into our mix.`
`03:24.18`	`*** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net)`
`03:24.18`	`*** mode/#asterisk [+o mog] by ChanServ`
`03:26.19`	`ariel_`	`mog, long time no see. Hope all is well`
`03:36.02`	`*** join/#asterisk Lucky7 (n=abaird@207.200.28.175)`
`03:36.27`	`Lucky7`	`Hey - whats the command that will show you where your Digum cards are plugged into? (like what slot and stuff)`
`03:36.54`	`De_Mon`	`lspci`
`03:37.09`	`Lucky7`	`sorry - i ment span`
`03:37.36`	`Lucky7`	`I've got a WE12XP T1 card that randomly alarms yellow after a restart`
`03:38.07`	`De_Mon`	`zttool? not sure about that one`
`03:38.30`	`Lucky7`	`nah- its not tool`
`03:39.27`	`ariel_`	`zap show status`
`03:40.47`	`*** join/#asterisk L\|NUX (i=L_NUX@unaffiliated/lnux/x-10290)`
`03:41.34`	`Lucky7`	`i'm getting a ZT_SPANCONFIG failed on span 0`
`03:41.43`	`Lucky7`	`its almost like it doesn't know that span 0 exists`
`03:41.54`	`Lucky7`	`i remember seeing a tool that told me what is on each span`
`03:42.50`	`ariel_`	`zttool will tell you allot of info`
`03:43.03`	`Lucky7`	`oo - thats neat`
`03:43.08`	`Lucky7`	`now it doesn't see my TDM card at all`
`03:43.11`	`Lucky7`	`er, TE card`
`03:43.32`	`De_Mon`	`zomg what did you doo!`
`03:43.41`	`Lucky7`	`probably something to do with genzaptel`
`03:43.45`	`ariel_`	`http://www.voip-info.org/wiki/view/Asterisk+zttool`
`03:44.07`	`Lucky7`	`ahh!`
`03:44.07`	`Lucky7`	`thats it`
`03:44.13`	`Lucky7`	`cat /proc/zaptedl*`
`03:46.06`	`ariel_`	`you can get that as well by doing ztcfg -vv`
`03:46.26`	`Lucky7`	`Not if it isn't loaded`
`03:46.34`	`Lucky7`	`it'll just be really pissed off and give errors`
`03:47.38`	`ariel_`	`Ok it's sleep time. Need to be up by 5:30 am. It's sure cold outside here. Please send the cold weather away don't need it.`
`03:50.32`	`Lucky7`	`hm`
`03:53.56`	`Lucky7`	`the system doesn't recogize its a TE12XP`
`03:54.12`	`Lucky7`	`03:02.0 Ethernet controller: Digium, Inc. Unknown device 0120 (rev 11)`
`03:57.40`	`Lucky7`	`How do i tell what is causing a YEL alarm?`
`04:01.20`	`*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)`
`04:01.49`	`Lucky7`	`http://rafb.net/p/cOcdKu53.html`
`04:01.49`	`Lucky7`	`thats with lsmod`
`04:01.49`	`Lucky7`	`whatelse am I supposed to have there?`
`04:04.34`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f4c9604990e6d20e)`
`04:07.46`	`*** part/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net)`
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`04:11.41`	`[TK]D-Fender`	`Lucky7, modprobe the driver`
`04:20.28`	`Lucky7`	`Yea`
`04:20.29`	`Lucky7`	`i did`
`04:20.47`	`Lucky7`	`what actually landed up doing it was doing an lsmod`
`04:20.58`	`Lucky7`	`and somehow the it loaded like 6-7 drivers on zaptel`
`04:21.03`	`Lucky7`	`I modprobe -r'd everything`
`04:21.15`	`Lucky7`	`and the second i removed tor2 - it changed to OK`
`04:21.49`	`[TK]D-Fender`	`Lucky7, So all good now?`
`04:21.55`	`Lucky7`	`Yup yup`
`04:23.12`	`*** part/#asterisk Lucky7 (n=abaird@207.200.28.175)`
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`04:35.02`	`egecko`	`has anyone tried to implement some kind of automatic gain control on the audio from voicemails? would something like "sox" do this?`
`04:35.31`	`jlewis`	`if anyone has access or knows someone...the link for asterisk 1.4.17 at http://www.asterisk.org/downloads actually points to 1.4.1. Looks like someone typo'd/dropped a 7.`
`04:35.35`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`04:39.34`	`[TK]D-Fender`	`egecko, AGC adjusts the overall interface once to match the telco. Not message to messge. So if they are poor relative to one another, its the endpoints fault`
`04:40.15`	`*** join/#asterisk angom (n=Angel@201.170.49.106)`
`04:41.39`	`egecko`	`I was thinking of doing some post-processing to voicemails after they have been saved.`
`04:42.02`	`egecko`	`before emailing it to the user as an mp3 .. running it through some "effects" filters first`
`04:46.34`	`*** join/#asterisk jochieng (n=jochieng@217.194.147.193)`
`04:53.45`	`[TK]D-Fender`	`egecko, there are clear places to trigger such scripts in voicemail.conf`
`04:59.17`	`*** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl)`
`05:04.24`	`*** join/#asterisk hfb (n=hfb@75.80.37.175)`
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`05:20.16`	`Mavvie`	`http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO <- anybody got a copy of the conf.agi mentioned there?`
`05:24.56`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
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`05:41.54`	`*** join/#asterisk emike79 (n=temp@216.254.186.68)`
`05:42.08`	`emike79`	`All, I got a quick question regarding codecs`
`05:42.18`	`emike79`	`I know asterisk does not support g729B`
`05:42.45`	`emike79`	`I also know that it announces codec as g729 - does not specify g729a or g729b`
`05:43.09`	`emike79`	`what happens is the remote end assumes g729b and uses that causing poor voice quality`
`05:43.23`	`emike79`	`and many messages to be printed on the console`
`05:43.30`	`*** part/#asterisk nny_1 (n=Scott@64.20.138.159.dyn-e-pool11.pool.hargray.net)`
`05:43.39`	`*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)`
`05:43.45`	`emike79`	`how do I force it to announce g729a such that the remote end does not use g729b`
`05:43.46`	`emike79`	`?`
`05:44.30`	`emike79`	`any ideas?`
`05:48.02`	`emike79`	`hmm.. maybe this is already done -- I see that in the SDP, asterisk does announce "annexb=no"`
`05:48.19`	`emike79`	`anyone know how to get NexTone to honor this?`
`05:56.57`	`[TK]D-Fender`	`emike79, If they don't want to then thats the end of it. You don't MAKE them do anything. They either cooperate, or they don't`
`05:57.10`	`*** join/#asterisk tengulre (n=tengulre@124.42.50.9)`
`06:02.41`	`zeeesh`	`hi getting one way trafic ... wil u pls guide how to troubleshoot ...`
`06:02.58`	`[TK]D-Fender`	`zeeesh, describe the scenario`
`06:05.39`	`zeeesh`	`i have server in usa where i registered abt 10 sip peers .. in my office abt 6 peers registered from 6 different comuters or cell phones are communicating each other fine .. but if a user register from outside the country .. i m unable to hear his voice .. he can hear me ..`
`06:06.05`	`[TK]D-Fender`	`zeeesh, well if either side has NAT involved, then here :`
`06:06.07`	`[TK]D-Fender`	`~sipnat`
`06:06.08`	`jbot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`06:06.09`	`[TK]D-Fender`	`^^^^^^^^^^`
`06:06.25`	`zeeesh`	`i hv mentioned nat=yes with all the peers`
`06:06.47`	`[TK]D-Fender`	`zeeesh, there is more than just that...`
`06:07.25`	`zeeesh`	`as well by using sip .. when i involove any route getting same problem .. just getting one way trafic .. anyway let me check ..`
`06:09.23`	`tengulre`	`RTP?`
`06:13.16`	`*** join/#asterisk denon (n=denon@tooth.decay.org)`
`06:13.16`	`*** mode/#asterisk [+o denon] by ChanServ`
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`08:07.22`	`R1ck`	`would you guys recommend configuring asterisk by hand or using a gui like asterisknow or trixbox?`
`08:08.51`	`drmessano`	`To answer a question much like religious preference`
`08:09.02`	`drmessano`	`I would use Asterisk + FreePBX, but not Trixbox`
`08:10.20`	`R1ck`	`why freepbx? is it a lot of work to set up a trunk and extensions?`
`08:10.41`	`drmessano`	`Depends on how you look at it`
`08:10.44`	`R1ck`	`cause right now I have a configuration I understand very little of, because freepbx made it very complicating`
`08:11.12`	`R1ck`	`and i'm thinking of reinstalling the box and doing things by hand`
`08:11.23`	`Alexandre_fr`	`If you have time, the best way to understand asterisk it's doing it by hand`
`08:11.26`	`drmessano`	`If you use a WYSWIYG web editor, you create "working" HTML that probably makes little sense and is hard to edit manually.. but works`
`08:11.40`	`drmessano`	`If you want real control, you hand edit`
`08:12.03`	`drmessano`	`If you want a quickie interface that does the job for casual admins, FreePBX is nice`
`08:12.09`	`drmessano`	`Depends on the admin`
`08:12.28`	`drmessano`	`as Alexandre_fr, if you want to REALLY learn Asterisk, hand edit is the way to go`
`08:12.48`	`R1ck`	`i would need to configure ring groups, queues, voicemail.. at least`
`08:12.58`	`R1ck`	`about 20 extensions`
`08:13.07`	`*** join/#asterisk ralfep (n=ralfe@dsl-240-41-38.telkomadsl.co.za)`
`08:13.19`	`drmessano`	`Are you going to be the admin?`
`08:13.23`	`drmessano`	`Day to day?`
`08:13.33`	`R1ck`	`yeah`
`08:13.58`	`R1ck`	`right now i have a fairly good understanding of how it works`
`08:14.05`	`R1ck`	`i just dont like the complicated configs :P`
`08:14.07`	`ralfep`	`Hi all. Could someone either briefly explain to me how conference calls work in asterisk, or poit me to a good tutorial please?`
`08:14.10`	`drmessano`	`Expect lots of changes?`
`08:14.21`	`R1ck`	`no not really`
`08:14.57`	`drmessano`	`I'll be honest.. I like digging to config files, but FreePBX makes it easier.. and if you know something about Asterisk, you can troubleshoot most FreePBX problems`
`08:15.04`	`drmessano`	`But again, its religion.. lol`
`08:15.21`	`R1ck`	`well, the problem i'm having, is that asterisk doesnt set the correct MSN when dialing out, which is really difficult to debug like this`
`08:15.57`	`drmessano`	`I think FreePBX does what it does very well.. but.. it only does as much as it does`
`08:16.06`	`drmessano`	`Theres a lot asterisk will do that FreePBX doesnt touch`
`08:16.21`	`drmessano`	`For most people it does enough`
`08:16.25`	`drmessano`	`err`
`08:16.30`	`drmessano`	`For a lot of people it does enough`
`08:16.34`	`drmessano`	`Id say try it`
`08:17.18`	`ronr`	`is there a way to measure echo and the effects of echo cancellation other than just making a call?`
`08:17.23`	`drmessano`	`But if you want a GUI, dont get a box like Trixbox, it will mess with your head`
`08:17.35`	`drmessano`	`Build something up and put just FreePBX on it`
`08:18.01`	`R1ck`	`hmm`
`08:18.58`	`drmessano`	`I'm sure if you ask the right person, they'll tell you "ZOMG UR A N00B MORON DONUT UZE A GEWI", but go with what works for ya`
`08:20.26`	`tzafrir_laptop`	`ronr, in ztmonitor you can record either before echo cancellation or after it`
`08:20.40`	`tzafrir_laptop`	`Though not in the same run`
`08:21.02`	`tzafrir_laptop`	`But also look at OSLEC's scary zaptap patch`
`08:22.16`	`R1ck`	`is it better to run asterisk 1.4 instead of 1.2?`
`08:22.25`	`R1ck`	`if you have a zaptel device`
`08:22.45`	`tzafrir_laptop`	`zaptel 1.4 is improved, slightlly`
`08:22.48`	`R1ck`	`cause the junghanns drivers seems to be experimental`
`08:22.53`	`tzafrir_laptop`	`chan_zap is improved`
`08:23.53`	`drmessano`	`1.4 has been out long enough that I think it's silly not to be using it, unless you've got some very specific dependancies`
`08:24.59`	`R1ck`	`yeah im just googling for user experiences with bristuff 1.4`
`08:26.26`	`tzafrir_laptop`	`R1ck, what card do you have?`
`08:26.42`	`R1ck`	`01:02.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01)`
`08:26.52`	`R1ck`	`Junghanns.net quadBRI`
`08:27.29`	`tzafrir_laptop`	`From what I see the most up-to-date drivers are in his latest zaptel 1.4 release`
`08:27.37`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`08:27.48`	`tzafrir_laptop`	`Though IIRC the difference in qozap is not as big as the one in cwain`
`08:30.20`	`ronr`	`tzafrir_laptop: ok, I'll try that (in some calls I have really annoying echos and in others none at all)`
`08:37.21`	`*** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk)`
`08:40.37`	`*** join/#asterisk clusco (n=clus@203.114.50.228)`
`08:43.20`	`*** join/#asterisk af_ (n=getsmart@88-149-240-55.dynamic.ngi.it)`
`08:45.04`	`clusco`	`hi......`
`08:45.19`	`clusco`	`how does to connect between 2 asterisk ????`
`08:49.42`	`*** join/#asterisk bakarat (n=arnath@d54C1C929.access.telenet.be)`
`08:50.18`	`*** join/#asterisk Ccomp5950 (n=Owner@dpc6935139171.direcpc.com)`
`08:50.36`	`bakarat`	`supposing i have a telefone line, can i set asterisk up as a sip gateway for myself? (so basically i can access it as a sip account from another internet-connected device)`
`08:50.58`	`*** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl)`
`08:58.46`	`*** join/#asterisk Maliuta (i=nikolai@119.11.103.56)`
`09:08.27`	`fujin`	`bakarat: yes, of course`
`09:08.52`	`fujin`	`get an fxo/fxs card, from digium or sangoma`
`09:09.00`	`fujin`	`configure a sip account, and a basic dialplan`
`09:09.03`	`fujin`	`and you're done ;)`
`09:09.10`	`bakarat`	`fujin: sweet, thanks :D`
`09:09.34`	`fujin`	`if you're not really interested in the nitty gritty, other projects like asterisknow/freepbx/trixbox etc will take care of the magic for you`
`09:11.50`	`bakarat`	`fujin: ah, good to know, might be a better option for a beginner like me :)`
`09:20.57`	`*** join/#asterisk Itiliti (n=Itiliti@mail.itility.us)`
`09:24.45`	`*** join/#asterisk arcanine (n=saxon_m2@203.82.44.181)`
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`09:38.55`	`*** join/#asterisk Defraz (n=tim@24-116-152-177.cpe.cableone.net)`
`09:40.35`	`*** join/#asterisk ToTo (i=Administ@207.176.6.149)`
`09:41.45`	`ToTo`	`hi all`
`09:54.07`	`*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)`
`10:09.22`	`*** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com)`
`10:10.03`	`*** join/#asterisk KnowWhat (n=sajid@203.99.178.41)`
`10:10.06`	`KnowWhat`	`Hi all`
`10:10.17`	`KnowWhat`	`i need some tutorial on configuring x100p card please`
`10:10.26`	`KnowWhat`	`or its clone`
`10:10.36`	`KnowWhat`	`md3200 chipset may be`
`10:24.35`	`*** join/#asterisk qdk (n=qdk@195.242.194.41)`
`10:26.47`	`*** join/#asterisk Zap-W (i=xaszx@87.69.35.201.cable.012.net.il)`
`10:26.48`	`Zap-W`	`hi`
`10:26.53`	`Zap-W`	`does asterisk support pam`
`10:27.00`	`Zap-W`	`or kerberos or gSSAPI`
`10:28.38`	`*** join/#asterisk Porks (n=Porks@201.62.79.12)`
`10:29.22`	`tzafrir_laptop`	`Zap-W, no`
`10:29.49`	`tzafrir_laptop`	`For what kind of application?`
`10:30.00`	`*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)`
`10:30.27`	`tzafrir_laptop`	`authenticate phones to asterisk? agents to asterisk?`
`10:30.50`	`Zap-W`	`<PROTECTED>`
`10:31.00`	`Zap-W`	`sip clients`
`10:33.10`	`Zap-W`	`?`
`10:39.41`	`KnowWhat`	`tzafrir: got any clue of md3200 configuration`
`10:41.27`	`tzafrir_laptop`	`KnowWhat, how does it show up on lspci?`
`10:43.17`	`R1ck`	`anyone here running hylafax with iaxmodem and asterisk?`
`10:44.08`	`R1ck`	`iaxmodem is succesfully registered to asterisk, it creates the device, hylafax opens the modem but when calling the extension, I get no fax beeps`
`10:44.40`	`tzafrir_laptop`	`KnowWhat, It should be picked up by 'modprobe wcfxo'`
`10:45.22`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-03ebd2317f5821a8)`
`10:46.15`	`tzafrir_laptop`	`But I recently fixed wcfxo.c in zaptel svn to recognize the md3200-based clone I have at home`
`11:00.38`	`*** join/#asterisk Winkie (n=urmom@149.254.192.192)`
`11:04.05`	`*** join/#asterisk PepOSX (n=pepOSX--@190.78.221.19)`
`11:04.30`	`*** join/#asterisk l0verb0y (i=daemon@210.1.137.43)`
`11:04.35`	`*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)`
`11:04.35`	`l0verb0y`	`hey hows it going`
`11:06.48`	`KnowWhat`	`yeah`
`11:06.53`	`KnowWhat`	`it is picked up`
`11:07.02`	`KnowWhat`	`but i am having an error here`
`11:07.54`	`KnowWhat`	`Changing signalling on channel 1 from Unused to FXO Kewlstart ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling?`
`11:09.05`	`l0verb0y`	`hey does anyone have any advice on faxing with asterisk`
`11:09.08`	`*** join/#asterisk wakku (n=eurulo@unaffiliated/wakku)`
`11:09.10`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-eb612aae2af4b0fa)`
`11:16.04`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`11:18.50`	`tzafrir_laptop`	`KnowWhat, you need fxsks=1`
`11:18.56`	`tzafrir_laptop`	`not kxoks=1`
`11:19.19`	`tzafrir_laptop`	`that is: not fxoks=1`
`11:19.38`	`tzafrir_laptop`	`try running: xpp/utils/zapconf`
`11:28.30`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)`
`11:28.39`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`11:36.47`	`lilalinux`	`how do I transfer a call from a zap channel to a sip channel and vice versa?`
`11:37.55`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0926cfd72ce42fd5)`
`11:38.57`	`KnowWhat`	`ok thanks tzafrir_home`
`11:41.14`	`*** join/#asterisk [gnubie] (n=[gnubie]@cm205.gamma183.maxonline.com.sg)`
`11:41.45`	`*** join/#asterisk glen2 (n=glen@212.54.184.93)`
`11:45.36`	`*** join/#asterisk vrtk (n=bb@189.21.178.20)`
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`11:59.11`	`lilalinux`	`I have atxfer => 2 in my features.conf, but asterisk ignores when I dial 2. blindxfer with # works, but I need attended call transfer`
`12:00.15`	`waKKu`	`dont you have any other feature using * as start ?`
`12:00.59`	`waKKu`	`do a "feature show" on cli`
`12:01.48`	`lilalinux`	`No such command 'feature show'`
`12:02.26`	`lilalinux`	`do I have to activate features.conf somehow?`
`12:03.05`	`waKKu`	`are you using 1.4 or 1.2 ?`
`12:03.47`	`lilalinux`	`1.2`
`12:04.33`	`waKKu`	`maybe you need to set it as DYNAMIC_FEATURES global in extensions.conf... but i'm not sure`
`12:04.55`	`lilalinux`	`I have DYNAMIC_FEATURES=automon`
`12:05.10`	`waKKu`	`DYNAMIC_FEATURES=automon#atxfer`
`12:05.12`	`waKKu`	`try it`
`12:06.31`	`JunK-Y`	`in 1.2, it's show features.`
`12:06.46`	`waKKu`	`read it too: http://www.voip-info.org/wiki-Asterisk+config+features.conf`
`12:07.36`	`lilalinux`	`thx, I'll try`
`12:09.05`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`12:09.10`	`*** join/#asterisk glen2 (n=glen@212.54.184.93)`
`12:13.24`	`*** join/#asterisk RoyK (n=roy@ip-131-23-149-91.dialup.ice.no)`
`12:24.36`	`lilalinux`	`waKKu: thx, I configured atxfer now as #. *2 didn't work`
`12:24.47`	`*** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60)`
`12:26.35`	`*** join/#asterisk glen2 (n=glen@212.54.184.93)`
`12:27.43`	`*** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br)`
`12:29.11`	`*** join/#asterisk rdis (n=Kenoby@195-23-23-14.net.novis.pt)`
`12:31.19`	`lilalinux`	`Is there something like a blind-attended xfer?`
`12:31.54`	`lilalinux`	`I want attended xfer, but sometimes want to be able to hangup before the other picks up`
`12:31.55`	`*** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru)`
`12:32.29`	`lilalinux`	`with attended xfer I can't hangup before the otherone picks up`
`12:32.59`	`lilalinux`	`and with blind xfer I can't announce the call`
`12:34.14`	`rdis`	`Hi, using chan_misdn, I get "extensions can never match" . Using latestet fedora kernel (x86_64) with mISN 1.1.7. Any ideas?`
`12:36.58`	`*** join/#asterisk stuarta (n=stuarta@unaffiliated/stuarta)`
`12:37.04`	`*** join/#asterisk ltd (n=z@pat.transact.net.au)`
`12:38.38`	`puga`	`is there a link that specifies every varible created by asterisk during a call?`
`12:38.48`	`puga`	`variable*`
`12:40.10`	`puga`	`I found`
`12:40.11`	`puga`	`xD`
`12:40.29`	`*** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177611293.dsl.bell.ca)`
`12:48.21`	`javb`	`people, have an issue, with asterisk 1.2 everything was ok, now, with 1.4, a softphone and a polycom 501, can transfer well, but my 8 polycom 330 cant transfer or use callwaiting, asterisk cli says "cant authenticate the user" ... any ideas?`
`12:48.27`	`*** join/#asterisk rdis (n=Kenoby@2002:c317:170e:1:217:f2ff:fec6:5f6d)`
`12:48.40`	`jwh`	`javb: using insecure=very?`
`12:48.47`	`javb`	`jwh, no.`
`12:49.41`	`rdis`	`Hi! Any chan_misdn users/experts around here/`
`12:49.43`	`rdis`	`?`
`12:53.32`	`javb`	`i placed it in sip.conf, did sip reload, and still getting the same... any other idea?`
`12:55.34`	`*** join/#asterisk coppice (n=chatzill@137.192.17.210.dyn.pacific.net.hk)`
`12:55.42`	`puga`	`* Install ncurses to use the menu interface! *`
`12:55.42`	`puga`	`**************************************************`
`12:55.42`	`puga`	`menuselect changes NOT saved!`
`12:55.52`	`puga`	`ncurses is already installed`
`12:55.53`	`puga`	`Oo`
`12:56.45`	`tzafrir_laptop`	`you need development curses`
`12:56.47`	`*** join/#asterisk rdis (n=Kenoby@2002:c317:170e:1:217:f2ff:fec6:5f6d)`
`12:57.06`	`tzafrir_laptop`	`ncurses-dev(el)`
`12:57.08`	`*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)`
`12:57.13`	`puga`	`I've already installed ncurses-devel`
`12:57.35`	`puga`	`Installed Packages`
`12:57.35`	`puga`	`ncurses.i386 5.5-24.20060715 installed`
`12:57.35`	`puga`	`ncurses-devel.i386 5.5-24.20060715 installed`
`12:57.36`	`coppice`	`I thought all development was cursed`
`12:58.01`	`javb`	`jwh, it works with the polycom 501, and softphones, isnt it weird ?`
`12:58.06`	`*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)`
`12:58.19`	`jwh`	`javb: very`
`12:58.44`	`jwh`	`not sure what to suggest other than double check the ones that aren't working, and making sure the settings are the same`
`13:00.29`	`zeeesh`	`i m unable to hear ivr like ""tt-weasels"" or any other asterisk ivr ... by using sip client .. getting the same problem when peer calling .. i m getting one way trafic ... wil u pls guide?`
`13:00.46`	`jwh`	`nat/firewall?`
`13:01.07`	`FlatFoot`	`zeeesh: duplex mismatch ?`
`13:01.35`	`zeeesh`	`<jwh>: nat=yes ....fiewall disabled .`
`13:01.59`	`jwh`	`zeeesh: may need stun if you're behind nat`
`13:01.59`	`*** join/#asterisk anonymouz666 (n=anonymou@201.19.83.173)`
`13:02.10`	`puga`	`I've already installed asterisk 1.4 in CentOS 5, but I dont remember what package I had to install to stop this alert`
`13:03.58`	`zeeesh`	`<FlatFoot>: duplex mismatch ? sorry .. what u mean ?`
`13:06.00`	`FlatFoot`	`duplex , network connection try google . Duplex mismatch can cause one way traffic only`
`13:07.49`	`zeeesh`	`ok`
`13:11.43`	`*** join/#asterisk lirakis (n=lirakis@65.200.189.231)`
`13:14.54`	`puga`	`damn`
`13:15.05`	`puga`	`it insists that I dont have ncurses installed`
`13:15.36`	`stuarta`	`missing the -devel package?`
`13:16.43`	`puga`	`no`
`13:17.34`	`puga`	`Installed Packages`
`13:17.34`	`puga`	`ncurses.i386 5.5-24.20060715 installed`
`13:17.34`	`puga`	`ncurses-devel.i386 5.5-24.20060715 installed`
`13:21.53`	`slima`	`~book`
`13:21.53`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`13:22.48`	`puga`	`cant believe`
`13:22.50`	`puga`	`=\`
`13:23.09`	`*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)`
`13:23.49`	`lilalinux`	`May I have a [globals] section inside an #included file?`
`13:24.41`	`*** join/#asterisk logyati (n=logyati@201.29.102.153)`
`13:24.46`	`logyati`	`hello guys`
`13:25.00`	`logyati`	`can i use * 1.4 with zaptel drivers 1.2?`
`13:25.52`	`*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`13:31.38`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`13:34.30`	`*** join/#asterisk Skarmeth (n=Skarmeth@201.9.74.32)`
`13:35.05`	`*** join/#asterisk grEvenX (n=even@1elt2pn.ip.hipercom.no)`
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`13:44.45`	`*** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2)`
`13:45.37`	`fiXXXerMet`	`I am calling my softphone (1002) from my ip phone (1000) just fine, but I am not able to call the ip phone from the softphone. The softphone rings, but the ip phone never does - it just sits there.`
`13:47.22`	`fiXXXerMet`	`Now they can't call each other.`
`13:47.32`	`fiXXXerMet`	`Is QoS needed, or something special?`
`13:49.54`	`fiXXXerMet`	`It is weird because if I unplug the phone, or disable and reenable the account on the soft phone, things start working again.`
`13:51.21`	`iCEBrkr`	`<PROTECTED>`
`13:51.21`	`iCEBrkr`	`<PROTECTED>`
`13:51.24`	`iCEBrkr`	`Brrrrrrr`
`13:51.27`	`iCEBrkr`	`<PROTECTED>`
`13:51.29`	`iCEBrkr`	`Umm.`
`13:51.30`	`lilalinux`	`Erm`
`13:51.38`	`fiXXXerMet`	`I wish it were 31F here. It's like 11F!`
`13:52.15`	`*** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar)`
`13:52.36`	`coppice`	`its 20C here, and I think its damned cold`
`13:53.01`	`FlatFoot`	`20C the average for summer`
`13:53.04`	`FlatFoot`	`in the uk`
`13:54.18`	`fiXXXerMet`	`Can I use a name, instead of a number, for an extension?`
`13:55.40`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`13:57.21`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`13:57.59`	`RoyK`	`coppice: shut your mouth`
`13:58.34`	`RoyK`	`fiXXXerMet: a little colder than here....`
`13:58.42`	`anonymouz666`	`20C for RoyK it's summer`
`13:58.50`	`RoyK`	`bingo`
`13:58.51`	`coppice`	`hey, we're suffering. I'm actually sleeping under a duvet at the moment`
`13:59.39`	`RoyK`	`coppice: I do that something like 350 days a year`
`13:59.48`	`lilalinux`	`Is there some magic I have to take care off, when using #include "foobar"? It doesn't work, and it doesn't even complain when the file doesn't exist`
`14:00.21`	`coppice`	`RoyK: does that mean you travel south for 15 days a year?`
`14:00.57`	`RoyK`	`hehe`
`14:01.02`	`fiXXXerMet`	`OK, I have my phone registered as 'kyle'. Now... how do I dial that extension? I tried 5953 but that didn't work)`
`14:01.28`	`lilalinux`	`fiXXXerMet: SIP/kyle`
`14:01.56`	`lilalinux`	`fiXXXerMet: from where do you want to dial?`
`14:01.57`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`14:01.59`	`fiXXXerMet`	`How do i do that from an IP phone with a normal number pad`
`14:02.08`	`fiXXXerMet`	`Linksys SPA941`
`14:02.29`	`*** join/#asterisk k-man (n=jason@unaffiliated/k-man)`
`14:02.37`	`k-man`	`hello`
`14:02.57`	`fiXXXerMet`	`hi`
`14:03.23`	`k-man`	`i have created an extension called 1000, but when i dial it, i get the error "call to 1000 rejected because extension not found"`
`14:03.28`	`lilalinux`	`fiXXXerMet: don't know that, but you could always create another extension that is reachable via numbers and do the Dial(SIP/kyle)`
`14:04.34`	`lilalinux`	`exten => 5953, 1, Dial(SIP/kyle)`
`14:04.45`	`[TK]D-Fender`	`coppice: Yeah its 20c here alright....`
`14:04.52`	`[TK]D-Fender`	`coppice: NEGATIVE <-`
`14:04.52`	`fiXXXerMet`	`Ahh, thanks lilalinux`
`14:05.17`	`lilalinux`	`fiXXXerMet: maybe there is something more generic, but for the moment ...`
`14:05.40`	`[TK]D-Fender`	`fiXXXerMet: Quick lesson : you do not dial a "phone", you dial an "extension".`
`14:06.34`	`coppice`	`[TK]D-Fender: holidaying in Siberia?`
`14:06.55`	`[TK]D-Fender`	`fiXXXerMet: lilalinux gave you some dialplan code that upon dialing "5953" will then call your SIP device. Never mix those 2 ideas up. You can have 1,000,000 extensions that never lead to making a phone ring.`
`14:07.16`	`[TK]D-Fender`	`coppice: Nope... Welcome to Quebec winter. We also just broke our record for snow in December.`
`14:07.29`	`[TK]D-Fender`	`coppice: we got another 20cm on Ney Years eve`
`14:07.30`	`fugitivo`	`is agentcallbacklogin deprecated in 1.4.x?`
`14:07.33`	`[TK]D-Fender`	`New*`
`14:07.42`	`fiXXXerMet`	`[TK]D-Fender: I'm still trying to get my head around that, but thanks`
`14:07.46`	`[TK]D-Fender`	`fugitivo: Yes, you really should read upgrade.txt`
`14:08.00`	`javb`	`[TK]D-Fender: i have a an issue with asterisk 1.4, and polycom 330 sets... in 1.2 those phone were able to transfer and use call waiting, now, the cant, asterisk says "[Jan 3 10:07:09] NOTICE[2831]: chan_sip.c:13794 handle_request_invite: Failed to authenticate user" ... had google a lot.. and had studied sip debug... can u help ?`
`14:08.06`	`[TK]D-Fender`	`fiXXXerMet: well hopefully this is helping get you you on the right path`
`14:08.20`	`coppice`	`[TK]D-Fender: I think another 20cm is what they promised me in those e-mails I keep getting`
`14:08.38`	`[TK]D-Fender`	`coppice: ;)`
`14:08.40`	`fiXXXerMet`	`Yes it is [TK]D-Fender`
`14:08.45`	`javb`	`a 501 and a softphone (Ekiga), works great. The polycoms 330 worked GREAT in 1.2`
`14:09.19`	`[TK]D-Fender`	`javb: pastebin your sip.conf entry for them. and then pastebin a Good call followed by a bad call.`
`14:09.26`	`[TK]D-Fender`	`javb: With sip debug of course`
`14:09.28`	`*** join/#asterisk wulfy814 (n=askme@static-acs-24-154-122-15.zoominternet.net)`
`14:09.55`	`k-man`	`so when i load my extensions.conf file, it says extension 1000 (and 500) created, but when i dial those extensions, it says extension not found`
`14:10.18`	`*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)`
`14:10.47`	`k-man`	`any ideas how i can work out why i can't dial the extensions?`
`14:11.39`	`iCEBrkr`	`You know what I wish?`
`14:11.43`	`iCEBrkr`	`I wish VoicePulse was usable.`
`14:11.53`	`wulfy814`	`so I'm being stupid -- I'm trying to set outbound callerID based on the extension that dialed out http://pastebin.com/d66b03f1c`
`14:12.01`	`iCEBrkr`	`[Jan 3 09:09:42] NOTICE[31272]: chan_iax2.c:8101 __iax2_poke_noanswer: Peer 'vpconnect-t01' is now UNREACHABLE! Time: 54`
`14:12.05`	`iCEBrkr`	`All morning long.`
`14:12.10`	`fiXXXerMet`	`iCEBrkr: What's wrong with em?`
`14:12.28`	`iCEBrkr`	`It keeps bouncing`
`14:12.30`	`wulfy814`	`but, it doesn't seem to passing the correct info`
`14:12.35`	`[TK]D-Fender`	`k-man: probably because they aren't in the CONTEXT that the device is using <-`
`14:12.41`	`iCEBrkr`	`[Jan 3 09:12:17] NOTICE[31267]: chan_iax2.c:8101 __iax2_poke_noanswer: Peer 'vpconnect-t01' is now UNREACHABLE! Time: 1057`
`14:12.45`	`iCEBrkr`	`[Jan 3 09:12:28] NOTICE[31271]: chan_iax2.c:7295 socket_process: Peer 'vpconnect-t01' is now REACHABLE! Time: 1049`
`14:12.47`	`[TK]D-Fender`	`k-man: Or you might have just done them wrong.`
`14:12.54`	`iCEBrkr`	`Seems like every 10 seconds or so.`
`14:13.14`	`*** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net)`
`14:13.21`	`fugitivo`	`honestly the new way to add agents to queues in 1.4 is not good`
`14:13.24`	`iCEBrkr`	`I had to setup a second IAX channel with another provider`
`14:14.16`	`javb`	`[TK]D-Fender ---> good call with 501: http://pastebin.com/m65107217 \| bad call with 330: http://pastebin.com/m168e3328 \| good call with softphone: http://pastebin.com/m78f970b2 \| sip.conf: http://pastebin.com/m51593a29`
`14:14.30`	`k-man`	`[TK]D-Fender, when i create a device section, called say [1000], how do i set it up so that when i dial 1000, it goes to that device?`
`14:14.34`	`lilalinux`	`What trunk do I use, when I have ZAP and SIP and CAPI? (for example)`
`14:14.54`	`lilalinux`	`in [globals]`
`14:15.13`	`k-man`	`is that what this line does? exten => 1000,n,Dial(SIP/1000,30)`
`14:16.36`	`fugitivo`	`k-man: yes but you should have a line before with order number 1 or replace that n by 1`
`14:16.49`	`k-man`	`fugitivo, yeah, i do`
`14:16.57`	`k-man`	`i was just checking i unserstood what that meant`
`14:17.31`	`k-man`	`so if i have a device called [xyz] i could change SIP/1000 to SIP/xyz?`
`14:17.42`	`fugitivo`	`if it's in sip.conf yes`
`14:17.52`	`k-man`	`fugitivo, righto, thanks`
`14:17.55`	`fugitivo`	`if it's in iax.conf it should be IAX2/xyz`
`14:18.00`	`k-man`	`ok`
`14:18.06`	`javb`	`[TK]D-Fender -> Now, for SOME reason, all the 330 in this office has gone like this "sip show peers" http://pastebin.com/m47cae7a4 ... <--- those peers were not like that.`
`14:18.18`	`k-man`	`ok`
`14:18.53`	`k-man`	`so how do i tell asterisk that my sip device [1000] can dial my test numbers?`
`14:18.57`	`k-man`	`i think im missing something`
`14:19.11`	`fugitivo`	`what test numbers?`
`14:21.01`	`k-man`	`500 say`
`14:21.19`	`k-man`	`i mean, just a local dialplan`
`14:21.33`	`k-man`	`can i paste 3 lines here?`
`14:21.38`	`fugitivo`	`pastebin`
`14:21.47`	`javb`	`[TK]D-Fender : Those "unmmonitore" peers are able to call, but not to be called.`
`14:21.48`	`k-man`	`ok, 2 secs`
`14:22.31`	`jblack`	`k-man: The context you set for an account in sip.conf is the context that they start in, in extensions.conf`
`14:22.33`	`wulfy814`	`I want to set outbound callerID based on the extension I'm calling out from`
`14:23.04`	`k-man`	`jblack, so if its in [1000].... ?`
`14:23.13`	`jblack`	`No.`
`14:23.15`	`k-man`	`in sip.conf`
`14:23.18`	`wulfy814`	`can't I do: exten => _1NXXNXXXXXX,1,Macro(pri-out\|${CALLERIDNUM}\|${EXTEN})`
`14:23.29`	`jblack`	`The context= line in sip.conf`
`14:23.34`	`k-man`	`jblack, ah...`
`14:23.35`	`fugitivo`	`k-man: when you setup a device, you set the context, be sure that 500 is in the same context`
`14:23.40`	`k-man`	`that might be what im missing`
`14:23.45`	`k-man`	`ok`
`14:24.10`	`k-man`	`ok its in default`
`14:24.55`	`k-man`	`how do i let it do [internal] calls?`
`14:25.48`	`jblack`	`lets say you have 3 internal phones, authenticating with sip.`
`14:25.59`	`jblack`	`Default them all to the internal context`
`14:26.24`	`jblack`	`ANd let's say you have a sip/iax account with SomeCompany. Put them in the incoming context.`
`14:26.40`	`jblack`	`of course, these names are arbitrary.`
`14:27.11`	`k-man`	`ok`
`14:28.41`	`k-man`	`ah, that fixed it`
`14:28.44`	`k-man`	`thanks jblack`
`14:28.54`	`*** join/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com)`
`14:34.15`	`k-man`	`ok, so when i dial my Echo() test dial plan, it doesn't seem to echo anything`
`14:34.48`	`jblack`	`Make sure your microphone is up, not muted, etc.`
`14:34.55`	`jblack`	`That's a conventional adminitration problem`
`14:35.23`	`k-man`	`its a handset`
`14:35.35`	`k-man`	`spa942`
`14:35.44`	`k-man`	`the mute button is not lit`
`14:36.15`	`jblack`	`try reading through the sip debug then (asterisk -rvvvvvvvvvv and type "sip debug"`
`14:36.23`	`k-man`	`ok`
`14:36.52`	`jblack`	`also, if you turn sip debug off, you can watch the call traverse the dialplan.`
`14:37.02`	`jblack`	`Watch it take the steps, and make sure it's getting to echo.`
`14:37.28`	`jblack`	`If you like, you can put your sip.conf and extensions.conf up on pastebin, and I'll take a look at them`
`14:37.49`	`k-man`	`ok`
`14:37.51`	`k-man`	`thanks`
`14:38.06`	`jblack`	`I know that some of the hardware phones need special settings in sip.conf to make them work just right.`
`14:38.25`	`jblack`	`I don't know which phones need what, but google should be able to tell you.`
`14:39.12`	`[TK]D-Fender`	`k-man: No, defining a SIP.CONF entry has nothing to do with having an EXTENSION lead to it.`
`14:39.21`	`[TK]D-Fender`	`k-man: see my above comment to fiXXXerMet`
`14:39.43`	`jblack`	`[tk]d-fender Including setting context= ?`
`14:39.47`	`*** join/#asterisk ming_zym (n=ming_zym@124.14.234.119)`
`14:40.16`	`jblack`	`I may have given you bad advice, k-man.`
`14:40.20`	`[TK]D-Fender`	`jblack: in order to set the actual context, yes. And you should never have a "default" context that does anything important. un-authed calls fall there.`
`14:40.32`	`[TK]D-Fender`	`jblack: Every device definition should set their context.`
`14:40.54`	`jblack`	`I agree with those points. =)`
`14:41.14`	`jblack`	`Having a good day so far?`
`14:42.06`	`[TK]D-Fender`	`jblack: Not too bad.`
`14:42.16`	`[TK]D-Fender`	`jblack: 6.5h to go :)`
`14:42.26`	`javb`	`[TK]D-Fender : Did you see the pastebin ?`
`14:42.29`	`javb`	`:/`
`14:42.50`	`jblack`	`Some day, you may come to miss work. A little.`
`14:43.08`	`[TK]D-Fender`	`javb: thats a remote phone isn't it?`
`14:43.21`	`javb`	`no.. all of those are local.`
`14:43.36`	`k-man`	`jblack, http://pastebin.com/m780ed238`
`14:43.36`	`mtryfoss`	`are there any good explanation on how a sip 486 (busy) message becomes a 603 (declined) when passing through an asterisk server ?`
`14:43.48`	`jblack`	`k-man;: Mom`
`14:44.07`	`*** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu)`
`14:44.11`	`[TK]D-Fender`	`javb: I want to see 1 good call with the 330, 1 bad.`
`14:44.23`	`[TK]D-Fender`	`javb: and just it's config`
`14:44.26`	`jblack`	`Ok, so your 1000 phone will go to the phones context. that's good`
`14:44.37`	`jblack`	`Aha.`
`14:44.45`	`jblack`	`k-man: Ok, look at your phones context.`
`14:44.49`	`k-man`	`jblack, yeah, i worked that out after what you said about contexts`
`14:44.55`	`jedaustin`	`Anyone here familliar with provisioning Polycom phones? How do I set the transfer and other buttons to dial asterisk # and * codes?`
`14:45.06`	`jblack`	`You're missing your start extension. Pull out your book, and take it to page....`
`14:45.30`	`jblack`	`127`
`14:45.30`	`[TK]D-Fender`	`jedaustin: You don't want that...`
`14:45.32`	`k-man`	`ok`
`14:45.32`	`javb`	`[TK]D-Fender: mmm, if the 330 calls a 501, and then trys to transfer the call to the softphone, PERFECT. ... if the phone calls a 501, and trys to tranfer to a 330, "failed to authenticate"`
`14:46.04`	`k-man`	`jblack, which book are you refering to?`
`14:46.05`	`jblack`	`Wait. How many phones do you have? Just one?`
`14:46.10`	`jblack`	`The asterisk book.`
`14:46.22`	`[TK]D-Fender`	`jedaustin: Leave Polycom's transfers the way they are. DTMF transfers are BS`
`14:46.29`	`k-man`	`jblack, yeah... so far just one, but i do intend to connect a second phone`
`14:46.42`	`jblack`	`Ok. You need two phones.`
`14:46.45`	`[TK]D-Fender`	`jedaustin: Go read the user's guide to learn how to handle calls on them. They are currently the best in the industry.`
`14:46.58`	`jblack`	`so you can call yourself. Go get another one, either hard, or soft`
`14:47.04`	`k-man`	`jblack, the o'riely one? or osmething else?`
`14:47.06`	`jblack`	`Is there a ~ for soft phone suggestions?`
`14:47.06`	`[TK]D-Fender`	`jblack: Umm... whats this about him needing a "start extension"?`
`14:47.14`	`[TK]D-Fender`	`~sofphone`
`14:47.17`	`[TK]D-Fender`	`~softphone`
`14:47.18`	`jbot`	`[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam`
`14:47.23`	`k-man`	`jblack, ok`
`14:47.24`	`[TK]D-Fender`	`yup`
`14:47.31`	`jblack`	`k-man: Get one of those, so that you can call between yoruself.`
`14:47.48`	`javb`	`wow, this is driving me crazy, need coffee`
`14:47.51`	`k-man`	`any recommendations as to which one?`
`14:47.58`	`k-man`	`for windows`
`14:48.11`	`jblack`	`[tk]: I figure there's two ways to go about it.. Set him up with the nodephone entry, so he can call his extension, which leads to worrying qabout firewalling and such.`
`14:48.43`	`jblack`	`k-man: Look at the softphone line above. [tk]d-fender (who I think a supergenius) prefers zoiper. I found x-lite easier to deal with.`
`14:48.55`	`k-man`	`ok`
`14:48.56`	`k-man`	`thanks`
`14:48.58`	`lilalinux`	`I'm looking for some free music on hold mp3`
`14:49.16`	`[TK]D-Fender`	`X-lite IS very friendly, but lacks a native "Transfer". Zoiper is also multi-protocol which is nice and handles more calls.`
`14:49.25`	`jblack`	`tk: Or making him do a second local extension, so he can make * works internally.`
`14:49.31`	`[TK]D-Fender`	`lilalinux: Google it up.`
`14:49.52`	`lilalinux`	`Of course I did that, but I found only non-free ;-)`
`14:49.55`	`jblack`	`If he's dialing extensions directly in his phones context, he wouldn't need an s right off the bat, I figure.`
`14:50.00`	`mocker`	`I hate Zoiper.`
`14:50.08`	`[TK]D-Fender`	`lilalinux: your Google-Fu is weak. Try again`
`14:50.22`	`[TK]D-Fender`	`jblack: "s" has no place in SIP calls :)`
`14:50.26`	`jblack`	`If I get him able to dial 1000 to 1001, then I can say "Oh, and s is the answering extension"`
`14:50.38`	`[TK]D-Fender`	`jblack: only for IVRs & MACROs`
`14:50.52`	`[TK]D-Fender`	`jblack: Nope, try again....`
`14:51.09`	`tzafrir_laptop`	`try again`
`14:51.21`	`jblack`	`Well, isn't it used for when you just goto(context) ?`
`14:51.55`	`k-man`	`jblack, well, im going to try setting up zoiper, and creating a new extension for it`
`14:52.04`	`javb`	`those anyone here has the default musiconhold.conf file created by asterisk 1.4 on make samples?`
`14:52.05`	`jblack`	`Please do.`
`14:52.08`	`[TK]D-Fender`	`jblack: Only if YOU tell it to go to "s"`
`14:52.17`	`javb`	`does anyone here has the default musiconhold.conf file created by asterisk 1.4 on make samples?`
`14:52.23`	`[TK]D-Fender`	`javb: its in your source folder....`
`14:52.28`	`jblack`	`Heh, and I do.`
`14:53.06`	`jblack`	`k-man: You're catching on what's going on here, right? I'm pretty new at things, and I don't have a completely accurate understanding yet.`
`14:53.07`	`[TK]D-Fender`	`jblack: and you can't just Goto to a context. check your parameter precedence.`
`14:53.12`	`*** join/#asterisk ToTo (n=ToTo@207.176.6.58)`
`14:53.21`	`*** join/#asterisk CapRicORN^80 (n=you@209.8.41.156)`
`14:53.25`	`k-man`	`yeah`
`14:53.25`	`ToTo`	`hiall`
`14:53.26`	`jblack`	`I'm looking at them now. I'm using s in a pointless way.`
`14:53.28`	`k-man`	`getting the hang of it`
`14:53.36`	`k-man`	`i played with asterisk about 6 months ago`
`14:53.39`	`jblack`	`It works, but it's irrelevant.`
`14:53.47`	`[TK]D-Fender`	`jblack: Well I can't say that your use of it is "pointless"... go show me :)`
`14:53.51`	`k-man`	`but then didn't do anything and promplty forgot all i had learned`
`14:53.56`	`k-man`	`it's coming back to me though`
`14:54.29`	`[TK]D-Fender`	`CapRicORN^80: So.... figured out what you need to fix in your dialplan (or what you were dialing) in order to call out to FWD yet?`
`14:54.58`	`jblack`	`Well, for example, I have an inbound context, that IPKall and teliax go to. When those extensions pick up, I send them to public,s,1`
`14:55.03`	`[TK]D-Fender`	`CapRicORN^80: (beyond that 1 fixed test # you got up the other day)`
`14:55.19`	`[TK]D-Fender`	`jblack: Well that IS a perfectly good thing to do.`
`14:55.33`	`jblack`	`Yeah, it's valid. If it wasn't, it wouldn't work. :)`
`14:55.34`	`[TK]D-Fender`	`jblack: Running an IVR in that context off the "s" exten?`
`14:55.46`	`jblack`	`i don't know what IVR means`
`14:55.46`	`*** part/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)`
`14:55.46`	`*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)`
`14:55.51`	`[TK]D-Fender`	`jblack: By valid I don't just mean syntax, but your approach as a whole.`
`14:55.53`	`jblack`	`Internet Value Resaller?`
`14:56.11`	`CapRicORN^80`	`[TK]D-Fender: hi`
`14:56.12`	`[TK]D-Fender`	`jblack: IVR = menu where they can punch in DTMF to go through your extensions.`
`14:56.16`	`[TK]D-Fender`	`~ivr`
`14:56.17`	`jbot`	`from memory, ivr is Interactive Voice Response`
`14:56.19`	`CapRicORN^80`	`thanks for your help`
`14:56.29`	`CapRicORN^80`	`as i said i can dail that no`
`14:56.31`	`jblack`	`Ahh.`
`14:56.39`	`CapRicORN^80`	`it was mistake in context name`
`14:57.05`	`CapRicORN^80`	`well fWD . not working yet`
`14:57.11`	`[TK]D-Fender`	`CapRicORN^80: So... you had some number like 600 for echo or something working right? But as I recall you didn't figure out how to dial any other # yet. Is that still the case?`
`14:57.15`	`jblack`	`I think it's ok. Perhaps a little confusing if I give it to other newbs.`
`14:57.34`	`javb`	`i have in musiconhold.conf, mode=files, and the directory, i have the files in the directory in gsm format...`
`14:57.39`	`[TK]D-Fender`	`CapRicORN^80: enable SIP debug, try another call and pastebin it alogn with your dialplan. You did something SILLY last time.`
`14:57.45`	`CapRicORN^80`	`listen my sip users can call each other`
`14:57.46`	`javb`	`but when i press hold, it says started and stopped, an nothing else`
`14:57.49`	`javb`	`any ideas?`
`14:57.54`	`CapRicORN^80`	`but the problem is with FWd`
`14:58.04`	`[TK]D-Fender`	`CapRicORN^80: I left you a while to see if it would come to you but the answer WAS remarkably quick and easy, but I wanted you to try to come to it yourself.`
`14:58.25`	`[TK]D-Fender`	`javb: pastebin EVERYTHING <-`
`14:58.35`	`CapRicORN^80`	`right . but sorry i didnt`
`14:58.37`	`k-man`	`jblack, cool, got that working`
`14:58.41`	`k-man`	`i can now call myself`
`14:58.44`	`jblack`	`Great.`
`14:58.46`	`[TK]D-Fender`	`CapRicORN^80: so show me a new call and I'll tell you straight up where you went wrong.`
`14:58.51`	`jblack`	`So now you know you've made progress. :)`
`14:58.57`	`k-man`	`yeah`
`14:59.02`	`jblack`	`Let me look over the extensions you set up`
`14:59.07`	`[TK]D-Fender`	`k-man: Next so psychotherapy.... stop talking to yourself dammit!`
`14:59.23`	`jblack`	`Ok, with your soft phone, dial 501`
`14:59.23`	`CapRicORN^80`	`hmm . ok i am working on it`
`14:59.37`	`CapRicORN^80`	`actually i am little confused with things mention on fwd website`
`15:00.00`	`javb`	`[TK]D-Fender ---> http://pastebin.com/m6f08224b (musiconhold)`
`15:00.06`	`[TK]D-Fender`	`CapRicORN^80: Just give me the 2 bits. You had completely failed to see what you were doing.`
`15:00.24`	`[TK]D-Fender`	`javb: keep going...`
`15:00.30`	`k-man`	`jblack, seems to work too`
`15:00.44`	`CapRicORN^80`	`;exten => _393.,2,Dial(IAX2/${myid number}:${mypassword}@iax2.fwdnet.net/${EXTEN:3},60,r)`
`15:00.47`	`javb`	`[TK]D-Fender: http://pastebin.com/m10bee0ed (asterisk output)`
`15:00.55`	`CapRicORN^80`	`exten => _393.,2,Dial(IAX2/${myid number}:${mypassword}@iax2.fwdnet.net/${EXTEN:3},60,r)`
`15:01.17`	`[TK]D-Fender`	`javb: keep going...`
`15:01.20`	`jblack`	`k-man: So your sound problem is local to the hard phone?`
`15:01.26`	`yang`	`Which are some well supported ISDN cards to use with Asterisk ?`
`15:01.33`	`javb`	`[TK]D-Fender here is everything, even the "ls" http://pastebin.com/m6f3cf21a`
`15:01.45`	`[TK]D-Fender`	`CapRicORN^80: PASTEBIN the complete call at verbose 10 and SIP debug enabled as I requested, along with your complete dilaplan`
`15:01.51`	`k-man`	`jblack, i think so... but i can hear myself from the spa942 on the softphone`
`15:02.02`	`jblack`	`Try calling into 500 again on the spa`
`15:02.03`	`*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)`
`15:02.05`	`k-man`	`jblack, so im not sure what it is`
`15:02.30`	`CapRicORN^80`	`ok i will`
`15:02.33`	`k-man`	`jblack, anyway`
`15:02.39`	`k-man`	`its time for bed..`
`15:02.42`	`k-man`	`thanks for your help guys`
`15:02.43`	`yang`	`Is ISDN controller Cologne chip design HFC-43 any good ?`
`15:02.46`	`CapRicORN^80`	`but the above line is correct ?`
`15:02.46`	`jblack`	`ok`
`15:03.00`	`k-man`	`ill probably be back tomorrow :)`
`15:03.01`	`k-man`	`bye`
`15:03.02`	`Alexandre_fr`	`Do somebody know a sip call generator ?`
`15:03.11`	`coppice`	`sipp`
`15:03.14`	`*** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl)`
`15:03.26`	`Alexandre_fr`	`thx`
`15:03.40`	`[TK]D-Fender`	`javb: ls -l plese`
`15:03.42`	`[TK]D-Fender`	`please`
`15:04.50`	`[TK]D-Fender`	`CapRicORN^80: MAYBE. Now please provide that pastebin`
`15:05.03`	`CapRicORN^80`	`[TK]D-Fender: sure . actually i am afraid of you`
`15:05.13`	`CapRicORN^80`	`thats why i am thinking to paste it or not :)`
`15:05.14`	`[TK]D-Fender`	`lol`
`15:05.23`	`defswork`	`anyone recommend a handset model suitable for a hotel's rooms ?`
`15:05.40`	`[TK]D-Fender`	`CapRicORN^80: If you don't let your mechanic look under the hood don't expect him to be able to fix anything <-`
`15:05.48`	`[TK]D-Fender`	`defswork: Analog phone.`
`15:06.00`	`defswork`	`[TK]D-Fender: how so ?`
`15:06.04`	`javb`	`[TK]D-Fender, saw it?`
`15:06.05`	`mocker`	`[TK]D-Fender: You should charge what mechanics charge.`
`15:06.06`	`[TK]D-Fender`	`defswork: Something you can afford to have raped like drunk guests do to them`
`15:06.23`	`[TK]D-Fender`	`mocker: I am a consultant...... I jsut help a lot free here too`
`15:06.30`	`mocker`	`[TK]D-Fender: I know. :)`
`15:06.39`	`[TK]D-Fender`	`javb: "ls -l", not "ls"`
`15:06.53`	`[TK]D-Fender`	`javb: I doubt your pirv's`
`15:06.54`	`defswork`	`[TK]D-Fender: that's not a real problem for me - I'm not buying em :) But I'd really like IP phone`
`15:07.11`	`[TK]D-Fender`	`defswork: Polycom IP 320 on a PoE Switch`
`15:07.20`	`[TK]D-Fender`	`defswork: but no, you DON'T actually.`
`15:07.25`	`CapRicORN^80`	`well i did and you really slap me :(`
`15:07.33`	`[TK]D-Fender`	`defswork: that will allow them ot forward calls and do other evil shit`
`15:08.06`	`[TK]D-Fender`	`CapRicORN^80: You really needed it if I did. So get over it and we'll get this solved.`
`15:08.23`	`defswork`	`[TK]D-Fender: customer will have to pay for calls on checkout - I can restrict some stuff too`
`15:08.28`	`CapRicORN^80`	`ok`
`15:08.38`	`CapRicORN^80`	`let me figure out my files`
`15:08.40`	`CapRicORN^80`	`then i paste it`
`15:08.57`	`[TK]D-Fender`	`defswork: Hey, if you insist.... but I really wouldn't... costs more, adds nothing and frankly might confuse a guest.`
`15:09.53`	`defswork`	`[TK]D-Fender: hmm - I'd still rather not do analog if I can avoid it`
`15:10.01`	`mocker`	`defswork: Why?`
`15:10.10`	`defswork`	`mocker: cabling`
`15:10.18`	`[TK]D-Fender`	`defswork: Your call, but I've already advised the best low-end high-quality SIP phone for you.`
`15:10.32`	`defswork`	`have single CAT5 in rooms - no voice`
`15:10.37`	`[TK]D-Fender`	`defswork: You can run RJ11 of Cat5 you know...`
`15:10.50`	`defswork`	`[TK]D-Fender: lose IP in room then though`
`15:10.57`	`mocker`	`wifi`
`15:10.58`	`[TK]D-Fender`	`defswork: Get a splitter then`
`15:11.03`	`[TK]D-Fender`	`defswork: Or wifi`
`15:11.13`	`[TK]D-Fender`	`defswork: Splitter would be cool.`
`15:11.23`	`javb`	`[TK]D-Fender http://pastebin.com/m4d838d0d`
`15:11.47`	`defswork`	`what are the real issues with an IP phone in the room though ? apart from cost ?`
`15:11.48`	`[TK]D-Fender`	`defswork: 10/100 runs on 1,2,3,6. POTS would sue 5,6.`
`15:11.58`	`Qwell`	`theft`
`15:12.00`	`[TK]D-Fender`	`defswork: security.`
`15:12.03`	`mocker`	`support`
`15:12.13`	`mocker`	`Why does my phone have all these buttons?`
`15:12.16`	`javb`	`[TK]D-Fender what pirv ?`
`15:12.23`	`[TK]D-Fender`	`plenty of good reasons not to. Also CONFUSING to your guests <---`
`15:12.27`	`[gnubie]`	`gtg now..`
`15:12.29`	`FlatFoot`	`mocker: to hit when angry`
`15:12.29`	`*** part/#asterisk [gnubie] (n=[gnubie]@cm205.gamma183.maxonline.com.sg)`
`15:12.36`	`[TK]D-Fender`	`javb: Priveledges <-`
`15:12.47`	`mocker`	`Ohh yeah, and launching ethereal to listen to other guests.`
`15:12.52`	`defswork`	`mocker: I've not stayed in a hotel in the uk that had a phone without feature buttons`
`15:12.57`	`Qwell`	`mocker: that'd be fun!`
`15:13.03`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`15:13.03`	`*** mode/#asterisk [+o anthm] by ChanServ`
`15:13.06`	`[TK]D-Fender`	`Did I forget to mention "security"? ;)`
`15:13.18`	`mocker`	`defswork: People in the UK aren't as easily baffled.`
`15:13.23`	`javb`	`ajajaj`
`15:13.25`	`defswork`	`mocker: how'd they listen on a switched network ?`
`15:13.30`	`javb`	`did u see the ls -la`
`15:13.32`	`[TK]D-Fender`	`defswork: stop ^@#@ing with hotel room phone you h4x0r!`
`15:13.41`	`Qwell`	`defswork: surely you don't have a switch for each room`
`15:13.54`	`Qwell`	`erm, nm`
`15:13.58`	`Qwell`	`</tired>`
`15:14.16`	`[TK]D-Fender`	`javb: please chown those and fix the rights`
`15:14.35`	`[TK]D-Fender`	`...`
`15:14.41`	`[TK]D-Fender`	`omg, they are SUPPOSED to look like that?`
`15:14.48`	`yang`	`Which are some prefered ISDN cards to use with Asterisk ? any special ones from Digium ?`
`15:15.00`	`Qwell`	`yang: isdn as in bri?`
`15:15.11`	`yang`	`Qwell: yes bridge you mean?ž`
`15:15.16`	`Qwell`	`bridge?`
`15:15.24`	`[TK]D-Fender`	`javb: Are you running * as root?`
`15:15.27`	`Qwell`	`I said nothing of the sort`
`15:15.34`	`javb`	`[TK]D-Fender, YES`
`15:15.46`	`[TK]D-Fender`	`javb: Ok, now things are getting wierd...`
`15:15.52`	`yang`	`Qwell: ISDN as an asterisk to the pstn ISDN gateway`
`15:16.00`	`CapRicORN^80`	`[TK]D-Fender: http://pastebin.com/m33c3d27f`
`15:16.00`	`Qwell`	`yang: bri or pri?`
`15:16.11`	`javb`	`[TK]D-Fender, funny question: "EASY WAY TO GO BACK TO ASTERISK 1.2 ? "`
`15:16.29`	`yang`	`Qwell: i dont know the difference`
`15:16.38`	`Qwell`	`you'll need to find out`
`15:16.45`	`[TK]D-Fender`	`CapRicORN^80: I said your entire diaplan and CLI output with SIP DEBUG enabled <-`
`15:17.00`	`defswork`	`yang: BRI is 2 channel PRI is 30`
`15:17.09`	`[TK]D-Fender`	`javb: Just flush your modules folder, and recompile * 1.2`
`15:17.25`	`[TK]D-Fender`	`javb: if yous till have your source folders, even a make install alone will do.`
`15:18.08`	`javb`	[TK]D-Fender, its a shame that i `m not able to use 1.4 because of two INCREDIBLE weird bugs...
`15:20.05`	`yang`	`Qwell: BRI then , a small PCI card`
`15:20.13`	`Qwell`	`Digium b410p`
`15:20.23`	`yang`	`Qwell: thanks`
`15:20.58`	`yang`	`Qwell: much better than the ISDN Controller cologne Chip HFC-4S if you know it?`
`15:21.49`	`javb`	`can use asterisk 1.2 with zaptel 1.4 ?`
`15:22.08`	`[TK]D-Fender`	`javb: nope`
`15:22.36`	`*** join/#asterisk powerkill (n=powerkil@office.annatel.net)`
`15:22.38`	`dmz`	`morning y'all`
`15:22.47`	`dmz`	`anyone here use *1.4 w/app_conference?`
`15:22.55`	`CapRicORN^80`	`[TK]D-Fender: http://pastebin.com/m20961ce9`
`15:23.02`	`javb`	`so, i dont understadn the process to clean everything, downloading zaptel 1.2 and doing make clean and make and make install will overwrite zaptel 1.4 ?`
`15:23.58`	`javb`	`?`
`15:24.12`	`dmz`	`you should try to remove 1.2 files & modules to be "safe" :)`
`15:24.20`	`dmz`	`i don't think the make file has an uninstall option`
`15:24.34`	`dmz`	`you can try just overwriting w/1.2 but who knows what wil be different`
`15:25.38`	`yang`	`Qwell: I googled around and that ones costs around 900 USD...Its quite a lot ?`
`15:26.25`	`javb`	`guys, any help on the correct way to remove everything asterisk and zaptel ?`
`15:27.53`	`CapRicORN^80`	`[TK]D-Fender: check my sip enabled here : http://pastebin.com/m5d82b6f2`
`15:28.29`	`*** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net)`
`15:28.39`	`dmz`	`yang, what are you trying to buy?`
`15:30.27`	`*** join/#asterisk anonymouz666 (n=anonymou@201.19.83.173)`
`15:33.01`	`jblack`	`hmmm`
`15:33.35`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`15:34.36`	`jblack`	`Aha. I know where I got the misunderstanding about s. Macros.`
`15:35.58`	`*** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60)`
`15:36.54`	`lilalinux`	`can I have a [globals] section in an #included file?`
`15:37.53`	`lilalinux`	`I want to separate the distribution supplied conf from my own, and would like to keep the original one mostly untouched.`
`15:38.19`	`yang`	`dmz: something for around 100-200 eur?`
`15:38.43`	`tzafrir_laptop`	`lilalinux, yes`
`15:39.15`	`*** join/#asterisk mog (n=mog@nat/digium/x-85992d6a0cea6f7c)`
`15:39.15`	`*** mode/#asterisk [+o mog] by ChanServ`
`15:39.35`	`dmz`	`yang missed the start of your questions, what are you tryng to buy :)`
`15:39.59`	`tzafrir_laptop`	`javb, mostly. zaptel 1.2 and 1.4 differ by locations of the userspace .h files they install`
`15:40.05`	`yang`	`dmz: BRI, a small PCI card`
`15:40.30`	`*** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net)`
`15:40.34`	`javb`	`tzafrir_laptop, i see. how do i uninstall zaptel, librpri, and asterisk 1.4 ?`
`15:40.49`	`logyati`	`is anyone here familiar to asterisk-gui?`
`15:40.55`	`lilalinux`	`tzafrir: thx, do entries in the included [globals] override the original ones?`
`15:41.04`	`AlexTO`	`Hi, there is some familiar setting up CDRs on MYSQL?`
`15:41.16`	`[TK]D-Fender`	`CapRicORN^80: looks like your phone is trying to reinvite.`
`15:41.24`	`De_Mon`	`AlexTO yeah it was easy, just followed the directions...`
`15:41.39`	`[TK]D-Fender`	`CapRicORN^80: set "canreinvite=no" under [general] and every other section of sip.conf`
`15:41.51`	`CapRicORN^80`	`ok`
`15:42.08`	`dmz`	`yang ah, sorry i dno't know any good sources for bri cards`
`15:42.16`	`[TK]D-Fender`	`logyati: Yes, but this isn't the support channel for it`
`15:42.20`	`AlexTO`	`de_Mon, I already install the addon and the module has been loaded fine but i dont have samples to follow, the help for that is not to mucho`
`15:42.51`	`*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)`
`15:43.09`	`logyati`	`[TK]D-Fender, i know but there is only ghosts there, they dont answer, and dont even chat`
`15:43.20`	`yang`	`dmz: ok`
`15:43.30`	`logyati`	`[TK]D-Fender, asterisk-gui didnt detect my analog ports :(`
`15:43.36`	`[TK]D-Fender`	`logyati: Here we will actively berate you :)`
`15:44.10`	`[TK]D-Fender`	`logyati: Sorry but that stuff is definitely not supported here.`
`15:44.32`	`logyati`	`[TK]D-Fender, anyway, ty :( i understand`
`15:44.49`	`logyati`	`if anyone here can help me, please pm me then`
`15:45.06`	`CapRicORN^80`	`[TK]D-Fender: ok i did that`
`15:45.22`	`AlexTO`	`De_Mon, u there?`
`15:45.25`	`[TK]D-Fender`	`CapRicORN^80: Sorry I misread something in there.`
`15:45.29`	`CapRicORN^80`	`every thing else is fine ? i mean can i move to fwd`
`15:45.38`	`[TK]D-Fender`	`CapRicORN^80: You have 1 leg as SIP the other is IAX, you CAN'T be re-inviting...`
`15:45.41`	`*** join/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net)`
`15:45.45`	`nny_1`	`~book`
`15:45.46`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`15:46.23`	`nny_1`	`hmm not on amazon`
`15:46.27`	`nny_1`	`luckily i hat eamazon`
`15:46.30`	`CapRicORN^80`	`ok i made changes in sip.conf users and set canreinvite=no`
`15:46.30`	`nny_1`	`hate too`
`15:46.35`	`[TK]D-Fender`	`CapRicORN^80: FWD isn't even responding`
`15:46.45`	`[TK]D-Fender`	`CapRicORN^80: Let me read their guide & instructions.`
`15:47.01`	`CapRicORN^80`	`ok`
`15:48.38`	`*** join/#asterisk Toerkeium (i=oo@201.216.206.221)`
`15:48.44`	`logyati`	`[TK]D-Fender, ok the, so, about asterisk... the command zap show channels will show any analog port is i didnt configure zapata.conf?`
`15:48.52`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`15:50.54`	`nny_1`	`any other asterisk books recommended around here? I know the cookbook is in utero`
`15:52.17`	`*** join/#asterisk teletouch (n=teletouc@213.8.90.92)`
`15:53.16`	`*** join/#asterisk Havokmon (n=None@mail.valeoinc.com)`
`15:54.03`	`Havokmon`	`Morning all. Where is the best 'place' for a sip proxy? It doesn't help if it's outside the registrer's network, but the phone is still behind a nat, does it?`
`15:54.55`	`*** join/#asterisk af_ (n=getsmart@88-149-240-55.dynamic.ngi.it)`
`15:55.08`	`logyati`	`[TK]D-Fender, WARNING[30149]: app_system.c:107 system_exec_helper: Unable to execute '/sbin/zapscan.bin' this zapscan comes with asterisk?`
`15:56.50`	`*** join/#asterisk kshaw\|work (n=kshaw@nat/ibm/x-be9b4db508a432b6)`
`16:01.01`	`*** part/#asterisk lirakis (n=lirakis@65.200.189.231)`
`16:04.21`	`[TK]D-Fender`	`logyati: Sorry, if there's an issue with it, thats a binary build on a custom OS.`
`16:04.32`	`*** join/#asterisk binary-zero (n=binary--@unaffiliated/binary-zero)`
`16:04.32`	`[TK]D-Fender`	`logyati: All the reasons we don't want to deal with them.`
`16:04.39`	`*** join/#asterisk dswillia (n=me2@199.3.247.34)`
`16:04.55`	`binary-zero`	`guys i am getting a stupid error: file.c:517 ast_openstream_full: File /var/lib/asterisk/mohmp3/fpm-calm-river does not exist in any format`
`16:04.58`	`dswillia`	`is there a list of default commands that can be issued within a meetme conference?`
`16:04.59`	`binary-zero`	`can any one help on this`
`16:05.03`	`[TK]D-Fender`	`Havokmon: it should on the inside LAN and the sole thing forwarded out from there`
`16:05.25`	`[TK]D-Fender`	`binary-zero: Go show us the file is there.`
`16:05.35`	`binary-zero`	`[TK]D-Fender: i can assue it is`
`16:05.43`	`binary-zero`	`in mp3 format & permissions are also good`
`16:05.44`	`[TK]D-Fender`	`binary-zero: And show us the line that generated the error, not just the error itself`
`16:05.55`	`[TK]D-Fender`	`binary-zero: * doesn't support MP3 by default`
`16:06.01`	`binary-zero`	`yup i am using mpg123`
`16:06.07`	`[TK]D-Fender`	`binary-zero: You need to install format_mp3 which is part of asterisk-addons`
`16:06.15`	`[TK]D-Fender`	`binary-zero: that is NOT using mpg123.`
`16:06.19`	`binary-zero`	`um let me check that`
`16:06.26`	`[TK]D-Fender`	`binary-zero: that error is saying * is trying to play it, not mpg123`
`16:06.37`	`[TK]D-Fender`	`binary-zero: You're probably on mode-files`
`16:06.50`	`[TK]D-Fender`	`binary-zero: and frankly you should be using mpg123 anymore anyways`
`16:06.52`	`*** part/#asterisk harpal (n=Harpal@124.125.79.212)`
`16:07.05`	`binary-zero`	`correct , should be or shouldn't be ? [TK]D-Fender`
`16:07.06`	`[TK]D-Fender`	`dswillia: "show application meetme"`
`16:07.13`	`logyati`	`[TK]D-Fender, but app_zapsan isnt an asterisk module?`
`16:07.23`	`[TK]D-Fender`	`binary-zero: shouldtn't <- sorry`
`16:07.29`	`logyati`	`zapscan`
`16:08.12`	`[TK]D-Fender`	`logyati: perhaps, but your version is FIXED. If you can download and install the latest zaptel on top we might be able to support that. But I'm not sure thats even part of * the core zaptel`
`16:09.30`	`*** join/#asterisk puga (n=none@189.5.213.166)`
`16:09.41`	`FlatFoot`	`daft question , building a * that is only NETWORK no cards ( BRI/PRI ) do i need zaptel ??`
`16:09.45`	`binary-zero`	`[TK]D-Fender: thanks, format_mp3 solved the issue`
`16:10.08`	`dswillia`	`[TK]D-Fender: thanks, but are there default codes a user can enter to say record a call on demand, or mute parties, etc`
`16:10.09`	`CapRicORN^80`	`[TK]D-Fender: you said you have configure fwd with your asterisk`
`16:10.15`	`[TK]D-Fender`	`CapRicORN^80: Found the problem`
`16:10.27`	`[TK]D-Fender`	`CapRicORN^80: You setup iax.conf for FWD.`
`16:10.35`	`[TK]D-Fender`	`CapRicORN^80: but look at your DIAL : -- Executing [613@internal:1] Dial("SIP/saji-08cff460", "SIP/123456@iax2.fwdnet.net") in new stack`
`16:10.47`	`[TK]D-Fender`	`CapRicORN^80: You are trying to use SIP to call them there`
`16:10.52`	`tzafrir_laptop`	`FlatFoot, maybe for timing and for mixing (Meetme)`
`16:10.59`	`logyati`	`[TK]D-Fender, im using zaptel SVN--r downloaded from http://svn.digium.com/svn/zaptel/branches/1.4/ isnt it supported here?`
`16:11.04`	`CapRicORN^80`	`yes`
`16:11.12`	`logyati`	`[TK]D-Fender, i downloaded it today`
`16:11.18`	`*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-60925a6cea7cec28)`
`16:11.26`	`CapRicORN^80`	`i am calling from my sip user to call to other sip user`
`16:11.40`	`CapRicORN^80`	`which is not in my asterisk . thats what fwd do`
`16:11.42`	`FlatFoot`	`tzafrir_laptop: in that case do i need to fool zaptel into thinking it has a card ? is it ztdummy ?`
`16:11.59`	`*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`16:11.59`	`*** mode/#asterisk [+o lmadsen] by ChanServ`
`16:13.22`	`BCS-Satori`	`Has anyone configured Cisco 7960 (SIP) on asterisk but has skipped line buttons on the phone, say only register line button 1 and line button 6. I can get 1 to register but not 6, but if i move 6 to line button 2 it works. (yes i know sounds confusing)`
`16:14.54`	`*** join/#asterisk lirakis (n=lirakis@65.200.191.253)`
`16:14.58`	`*** join/#asterisk lupino3 (n=lupino3@217-133-45-108.b2b.tiscali.it)`
`16:15.05`	`lupino3`	`hello everybody`
`16:15.29`	`*** join/#asterisk Neil_L (n=NLiningt@81.171.129.186)`
`16:15.36`	`lupino3`	`is there any way to group together all CDR entries related to a single queue call?`
`16:15.41`	`*** join/#asterisk rantsh (n=rantsh@190.36.185.139)`
`16:15.41`	`lupino3`	`I see different entries`
`16:15.46`	`rantsh`	`hi people`
`16:15.55`	`rantsh`	`Happy new year to all`
`16:16.05`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`16:16.09`	`lupino3`	`and I'd like to associate one ID in the CDR userfield`
`16:16.15`	`lupino3`	`in order to do some further processing`
`16:16.22`	`rantsh`	`I'm having a hard time setting ilbc on 20 ms mode on asterisk 1.4.2`
`16:16.22`	`*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`16:16.27`	`lupino3`	`can anybody help me, please?`
`16:16.28`	`Steven_elvisda_`	`any tutorial how to make two people talk each other by useing asterisk?`
`16:16.38`	`[TK]D-Fender`	`CapRicORN^80: No, you are trying to CALL FWD using the SIP protocol, but you set them up for IAX`
`16:16.45`	`[TK]D-Fender`	`CapRicORN^80: pastebin your dialplan again`
`16:16.49`	`lupino3`	`(forgot to mention: asterisk 1.2.2x)`
`16:16.50`	`Steven_elvisda_`	`i know i need to config in extern.conf and sip.conf`
`16:16.53`	`Steven_elvisda_`	`but i don know how`
`16:16.54`	`rantsh`	`I keep setting it as "ilbc:20" in sip.conf but when I do sip show settings it shows "ilbc:30"`
`16:16.56`	`[TK]D-Fender`	`Steven_elvisda_: ....`
`16:16.58`	`[TK]D-Fender`	`~book`
`16:16.58`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`16:17.01`	`[TK]D-Fender`	`~jerjerguide`
`16:17.02`	`jbot`	`[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/`
`16:17.03`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^`
`16:17.45`	`[TK]D-Fender`	`lupino3: Yes, the UNQUEID field is what ties them together.`
`16:17.50`	`Steven_elvisda_`	`thanks [TK]D-Fender`
`16:18.09`	`rantsh`	`~ilbc`
`16:18.09`	`jbot`	`it has been said that ilbc is at http://www.ilbcfreeware.org`
`16:18.13`	`lupino3`	`[TK]D-Fender, I tried to write CDR(UNIQUEID) in the userfield`
`16:18.19`	`lupino3`	`but I get different values :(`
`16:18.29`	`[TK]D-Fender`	`lupino3: it IS a field aready!`
`16:18.39`	`lupino3`	`yes but it doesn't get written`
`16:18.45`	`[TK]D-Fender`	`lupino3: sure it does...`
`16:18.52`	`lupino3`	`maybe I need to enable it in source code`
`16:18.56`	`[TK]D-Fender`	`lupino3: check your field listing and show us a dump`
`16:19.02`	`lupino3`	`yes`
`16:19.18`	`lupino3`	`I can confirm that uniqueID is empty :(`
`16:19.32`	`*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)`
`16:19.36`	`[TK]D-Fender`	`lupino3: Something is definitely screwed up then`
`16:20.03`	`lupino3`	`uhm...`
`16:20.12`	`lupino3`	`but`
`16:20.16`	`lupino3`	`even if it doesn't get written`
`16:20.27`	`lupino3`	`shouldn't CDR(uniqueid) give me the same value?`
`16:20.50`	`rantsh`	`In any case, I've made a paste bin of the relevant parts`
`16:20.58`	`rantsh`	`you may see it here: http://pastebin.com/d78b8c005`
`16:22.35`	`CapRicORN^80`	`[TK]D-Fender: http://pastebin.com/m2f98fc47`
`16:24.30`	`FlatFoot`	`anyone in from amsterdam ?`
`16:25.39`	`[TK]D-Fender`	`CapRicORN^80: this is the line you showed me from your CLI output : -- Executing [613@internal:1] Dial("SIP/saji-08cff460", "SIP/123456@iax2.fwdnet.net") in new stack`
`16:25.54`	`[TK]D-Fender`	`CapRicORN^80: and THIS is the line that appears in your dialplan : exten => 613,Dial(IAX2/iaxfwd/613)`
`16:26.02`	`[TK]D-Fender`	`CapRicORN^80: you didn't apply your changes! <---`
`16:26.23`	`*** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru)`
`16:26.58`	`CapRicORN^80`	`what changes i should make ?`
`16:27.01`	`CapRicORN^80`	`i mean i am not getting you`
`16:27.58`	`[TK]D-Fender`	`CapRicORN^80: Look at the dialplan thats being executed. taht isn't what you just pasted for me. You changed your extensions.conf but didn't put your changes into EFFECT <-`
`16:28.11`	`[TK]D-Fender`	`CapRicORN^80: * has NOT reloaded your ned configs and`
`16:28.18`	`[TK]D-Fender`	`CapRicORN^80: new*`
`16:28.26`	`[TK]D-Fender`	`CapRicORN^80: do "dialplan reload"`
`16:28.28`	`rantsh`	`no one knows what to do with my ilbc issue?`
`16:28.54`	`*** part/#asterisk binary-zero (n=binary--@unaffiliated/binary-zero)`
`16:29.40`	`CapRicORN^80`	`i didnt`
`16:30.24`	`[TK]D-Fender`	`CapRicORN^80: Go reload your dialplan`
`16:30.40`	`[TK]D-Fender`	`CapRicORN^80: because you showed me 2 totally different things and thats why its not working.`
`16:30.50`	`CapRicORN^80`	`i did`
`16:31.12`	`[TK]D-Fender`	`CapRicORN^80: look at those 2 lines I pasted. You are either wrong, or you are lying to me.`
`16:31.12`	`CapRicORN^80`	`i didnt . seriouly i did reload asterisk`
`16:31.32`	`rantsh`	`~ilbc:20`
`16:32.12`	`[TK]D-Fender`	`CapRicORN^80: LOOK AT THEM. they are clearly not the same line. You are either showing me the wrong file or are showing me thing from different points in time that do not apply.`
`16:32.28`	`[TK]D-Fender`	`[11:25]<[TK]D-Fender>CapRicORN^80: this is the line you showed me from your CLI output : -- Executing [613@internal:1] Dial("SIP/saji-08cff460", "SIP/123456@iax2.fwdnet.net") in new stack`
`16:32.29`	`[TK]D-Fender`	`[11:25]<[TK]D-Fender>CapRicORN^80: and THIS is the line that appears in your dialplan : exten => 613,Dial(IAX2/iaxfwd/613)`
`16:32.35`	`[TK]D-Fender`	`^^^ NOT the same`
`16:32.56`	`[TK]D-Fender`	`CapRicORN^80: So Go pastebin another call attempt along with your "current" dialplan.`
`16:33.07`	`CapRicORN^80`	`ok`
`16:33.18`	`*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)`
`16:33.21`	`[TK]D-Fender`	`rantsh: I can't see any reference to the syntax you are using to set the bitrate as being valid.`
`16:33.30`	`*** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com)`
`16:34.00`	`VJFROMGT`	`anyone know about nxtvox analog cards ?`
`16:34.01`	`VJFROMGT`	`https://store.nxtvox.com/product_info.php?products_id=29&osCsid=ce71c25a538744bebd1873ac52553eef`
`16:34.17`	`rantsh`	`[TK]D-Fender, I remember reading it in a book or something, but... it was a long time ago and I don't have the book anymore`
`16:34.58`	`[TK]D-Fender`	`rantsh: well so far nothing I can find validates what you're doing so with that in mind my first guess is that it will not work and cannot be done`
`16:35.06`	`rantsh`	`[TK]D-Fender, but if you know of any way I could get ilbc working on 20 ms mode I'd very much appreciate it if you could lighten my path`
`16:36.05`	`[TK]D-Fender`	`rantsh: Sorry, not offhand, thats for sure. Get Googling`
`16:37.00`	`RoyK`	`how's the t.38 support status in asterisk atm? still only passthrough?`
`16:37.02`	`rantsh`	`[TK]D-Fender, well that's ok... thank you very much`
`16:37.23`	`[TK]D-Fender`	`RoyK: Correct`
`16:37.45`	`lupino3`	`[TK]D-Fender, I use MySQL backend, and UNIQUEID was not getting written. I had to modify the addons' Makefile in order to make it write the UNIQUEID field (as indicated in http://www.voip-info.org/wiki-Asterisk+cdr+mysql)`
`16:37.49`	`RoyK`	`any idea if there will be any more progress with t.38 endpoint/gateway?`
`16:37.54`	`*** join/#asterisk PepOSX (n=pepOSX--@190.78.221.19)`
`16:38.08`	`lupino3`	`[TK]D-Fender, but still.. I get different uniqueid's for calls related to a queue`
`16:38.19`	`[TK]D-Fender`	lupino3: `clid` varchar(80) NOT NULL default '', <-------
`16:38.49`	`lupino3`	`[TK]D-Fender, that is the caller id, right?`
`16:38.50`	`[TK]D-Fender`	`lupino3: hrm : A: You need to define MYSQL_LOGUNIQUEID at compile time for it to use that field.`
`16:38.55`	`*** join/#asterisk m160858 (n=m160858@200.48.6.67)`
`16:39.08`	`lupino3`	`[TK]D-Fender, isn't the field 'uniqueID'?`
`16:39.17`	`coppice`	`royK: "more" progress? there hasn't been any so far :-)`
`16:39.25`	`VJFROMGT`	`$50 for 8 port FXO card https://store.nxtvox.com/product_info.php?products_id=29&osCsid=ce71c25a538744bebd1873ac52553eef`
`16:40.11`	`[TK]D-Fender`	`lupino3: You know.. I think I'll elave this alone right now.... Its insane that it wouldn't include the full details you get in the CSV output.... but as I haven't really worked with it directly much I think you may be best off continuing your research based on the book.`
`16:40.11`	`RoyK`	`coppice: bingo`
`16:40.53`	`RoyK`	`coppice: only someone told me "it's in progress" some weeks back`
`16:41.00`	`lupino3`	`[TK]D-Fender, thanks however :)`
`16:41.03`	`[TK]D-Fender`	`coppice: technically getting T.38 passthrough is progress from not having any T.38 support at all so his term is valid :)`
`16:41.37`	`coppice`	`he said gateway and termination. he didn't mention passthrough`
`16:42.57`	`coppice`	`people will tell you all sorts of things are in progress, like they'll tell you code for X exists when its just a few lines thrown onto a web page somewhere`
`16:43.09`	`[TK]D-Fender`	`coppice: RoyK>how's the t.38 support status in asterisk atm? still only passthrough? <-- sorry, my opinion stands :)`
`16:43.13`	`RoyK`	`imho 'having' support for t38 passthrough is state, not progress :P`
`16:43.38`	`[TK]D-Fender`	`RoyK: "Off" is a state too...`
`16:43.42`	`coppice`	`OK. I only saw "any idea if there will be any more progress with t.38 endpoint/gateway?"`
`16:43.55`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`16:43.55`	`*** mode/#asterisk [+o russellb] by ChanServ`
`16:48.25`	`logyati`	`russellb, are u there?`
`16:48.40`	`logyati`	`russellb, ops, wrong channel`
`16:49.22`	`RoyK`	`I think there was some talk about doing full t38 for asterisk some two years back or so :)`
`16:49.24`	`russellb`	`i am here`
`16:49.36`	`*** join/#asterisk Greek-Boy (n=grb@41.221.58.2)`
`16:50.17`	`coppice`	`RoyK: I think that was me talking, but then the digium people tried their hardest to piss me off.`
`16:52.02`	`Greek-Boy`	`Is the Digium Wildcard TE412P the top-of-the-range card?`
`16:52.14`	`RoyK`	`coppice: I remember`
`16:52.19`	`[TK]D-Fender`	`coppice: Does it take that much effort? You are 99% "cynically snide" then it seems :)`
`16:52.48`	`coppice`	`the digium people have driven away most of the major contributors`
`16:52.51`	`[TK]D-Fender`	`coppice: Endearing in its own way of course ; you are "original".`
`16:53.17`	`[TK]D-Fender`	`coppice: Yeah, thats the optome of commercialization... you go for money and control.`
`16:53.22`	`[TK]D-Fender`	`outcome*`
`16:53.28`	`fugitivo`	`taking away Agentcallbacklogin was the worst decision`
`16:54.01`	`RoyK`	`Greek-Boy: well, unless you count sangoma in, it might be. if you _do_ count sangoma in, it's not`
`16:54.11`	`[TK]D-Fender`	`fugitivo: no need with the new login metheods.`
`16:54.59`	`fugitivo`	`[TK]D-Fender: new login methods = that thing called queues-with-callback-members ?`
`16:55.05`	`Greek-Boy`	`RoyK: and it's a PCI-X card, right?`
`16:55.26`	`[TK]D-Fender`	`fugitivo: "AddQueueMember" <-`
`16:55.56`	`[TK]D-Fender`	`fugitivo: Since you can add any channel as a memeber live in the dialplan, who needs a dedicated app for Dialplan based?`
`16:56.00`	`RoyK`	`Greek-Boy: Sangoma have PCI and PCI express cards. no need for PCI-X for a card with a maximum throughput of 8Mbps`
`16:56.18`	`[TK]D-Fender`	`fugitivo: I don't think you've realized quite how to use that app yet...`
`16:56.36`	`RoyK`	`Greek-Boy: digium's stuff is PCI as well, or have they started selling pci express?`
`16:56.37`	`fugitivo`	`[TK]D-Fender: well, that breaks every application for call centers developed for 1.2.x`
`16:56.55`	`[TK]D-Fender`	`fugitivo: Welcome to the wornderful world of GROWTH.`
`16:56.57`	`Qwell`	`fugitivo: 2 major versions isn't long enough?`
`16:57.16`	`Qwell`	`nobody is being forced to change their agents until 1.6`
`16:57.17`	`fugitivo`	`i'm not the only one saying that`
`16:57.25`	`[TK]D-Fender`	`fugitivo: "But Why won't my old Apple ][ programs work on my new Inte C2D Mac Pro>?!?!?!"`
`16:57.26`	`fugitivo`	`there's a lot of people complaining about that`
`16:57.46`	`Greek-Boy`	`RoyK: It's just PCI, PCI-X`
`16:57.47`	`[TK]D-Fender`	`fugitivo: Yea yeah... keep whining...`
`16:58.07`	`fugitivo`	`[TK]D-Fender: ? are you a kid?`
`16:58.08`	`[TK]D-Fender`	`fugitivo: Seriously... new versions, things change, time for them to get off their asses and adapt. Or simply die off`
`16:58.30`	`RoyK`	`Greek-Boy: PCI, then. I doubt they have 64bit cards for that. no point`
`16:58.36`	`Greek-Boy`	`RoyK: And PCI-X is 1064 MB/s, not 8Mbps.`
`16:58.40`	`fugitivo`	`Qwell: well, that's good news, thanks`
`16:58.43`	`[TK]D-Fender`	`fugitivo: Things change. Its bound to happen. You don't maintain backwards compatibility forever without becoming unstable & bloated`
`16:59.14`	`[TK]D-Fender`	`fugitivo: And given its flaws some things are best left FAR behind`
`16:59.17`	`RoyK`	`Greek-Boy: sure, but the actual load on the card is max 8Mbps since there are four 2Mbps ports on it, given you're running E1s`
`16:59.37`	`fugitivo`	`[TK]D-Fender: I know things change man, MY bussiness is software development, but when you have to change a BIG feature, you think it twice`
`17:00.31`	`coppice`	`If I thought twice about a big change, I'd never make it. I have to blunder in quick before good sense takes hold :-)`
`17:00.34`	`RoyK`	`Greek-Boy: I think those cards are PCI 2.3 compliant, meaning 32bit 66MHz, theoretically 200MBps, quite sufficient for the load`
`17:00.37`	`[TK]D-Fender`	`fugitivo: Well this is also OSS, not a stagnant fixed-purpose commercial platform. You have chosen the platform for your development, so you shouldn't be too quick to judge when the ground moves out from under you.`
`17:00.37`	`Greek-Boy`	`yes I'm running E1's`
`17:01.08`	`[TK]D-Fender`	`fugitivo: * will not stop for any 1 higher-level project that depends on it.`
`17:01.25`	`[TK]D-Fender`	`fugitivo: And if it does, well... watch people flock to the "next big thing"`
`17:01.29`	`RoyK`	`Greek-Boy: they might even be PCI 2.1, meaning max 33MHz, 32bit, theoretical speed 120MBps or so, which is also quite sufficient`
`17:01.32`	`Greek-Boy`	`RoyK: I have no choice but to go for a digium card since I'm running a SS7 channel`
`17:01.41`	`Havokmon`	`Umm Actually, I can run my Apple ][ programs on my new Mac Pro ;)`
`17:01.42`	`RoyK`	`Greek-Boy: why?`
`17:01.51`	`Havokmon`	`:P`
`17:01.53`	`[TK]D-Fender`	`Greek-Boy: Oh? Where does it say that Sangoma cards won't work?`
`17:01.56`	`coppice`	`Greek-Boy why would SS7 require a Digium card?`
`17:01.58`	`RoyK`	`Greek-Boy: sangomas use zaptel as well. asterisk doesn't see the difference`
`17:02.08`	`fugitivo`	`[TK]D-Fender: whatever...`
`17:02.19`	`[TK]D-Fender`	`Havokmon: s'ok.. I'll come up with another nasty attempt, the point is the same :)`
`17:02.33`	`Havokmon`	`lol I hear ya.`
`17:02.36`	`RoyK`	`Greek-Boy: what sort of ss7 solution is this? sangoma has their own which is quite robust......`
`17:02.52`	`[TK]D-Fender`	`fugitivo: Sorry to be the messenger. But let me say that the 1.4 way IS a lot better....`
`17:03.18`	`RoyK`	`coppice: what's the name of the L2 standard for E1 and so on again? Q.932?`
`17:03.34`	`Greek-Boy`	`RoyK: I'm playing around with chan_ss7 and want to try out libss7 too`
`17:03.49`	`fugitivo`	`[TK]D-Fender: sure it is`
`17:04.18`	`Greek-Boy`	`the chan_ss7 implies that it only works with Digium E1 cards.`
`17:04.24`	`Greek-Boy`	`maybe a mistake on the developers part`
`17:04.25`	`RoyK`	`Greek-Boy: short advice: drop it. Get an ss7box and a sangoma card and go with their solution. wasim, sometimes in here, has a few setups with that scattered across pakistan`
`17:04.42`	`RoyK`	`Greek-Boy: sounds like bull to me`
`17:04.56`	`Greek-Boy`	`hmmm`
`17:04.58`	`*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)`
`17:04.59`	`Greek-Boy`	`thanks for warning me`
`17:05.17`	`[TK]D-Fender`	`Greek-Boy: where does it imply that?`
`17:05.19`	`Greek-Boy`	`RoyK: what do u mean by ss7box?`
`17:05.49`	`RoyK`	`Greek-Boy: it probably works with zaptel and zaptel only runs _natively_ over digium cards and compatible ones, but sangoma wanpipe abstrahises that so zaptel runs with sangoma.`
`17:05.59`	`RoyK`	`Greek-Boy: sec`
`17:06.14`	`RoyK`	`Greek-Boy: http://wiki.sangoma.com/wanpipe-linux-asterisk-ss7-install`
`17:06.55`	`RoyK`	`Greek-Boy: that's a good solution...`
`17:07.35`	`Greek-Boy`	`[TK]D-Fender: http://www.sifira.dk/chan-ss7/0.9/readme.txt`
`17:07.42`	`tzafrir_laptop`	`RoyK, q921 (layer2) and Q931 (layer3)`
`17:09.02`	`Greek-Boy`	`RoyK: Looks robust. Thanks a lot.`
`17:09.47`	`RoyK`	`tzafrir_laptop: thanks`
`17:10.12`	`[TK]D-Fender`	`- Supports Digium E1 (T1 and other zap-compatible cards should be easy to add).`
`17:10.47`	`[TK]D-Fender`	`Greek-Boy: I wouldn't take it as "Digium only", but I can see a point of poetential doubt.`
`17:10.50`	`*** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org)`
`17:11.08`	`[TK]D-Fender`	`Greek-Boy: keeping in mind that a Zaptel interface at the channel-driver level all uses the same base.`
`17:14.10`	`RoyK`	`I'd guess if Greek-Boy is going to use ss7 in production, sangoma's solution might be better. It only costs $4k or so for the first ss7 link and less for the remaining ones. It doesn't license per traffic link either, and an ss7 link and hold the traffic for a medium-sized city`
`17:15.36`	`RoyK`	`s/hold/be enought for/`
`17:19.32`	`Greek-Boy`	`dont u think I should try out chan_ss7`
`17:19.37`	`Greek-Boy`	`might work`
`17:19.50`	`Greek-Boy`	`if i was a telco then I would go with the sangoma solution`
`17:19.57`	`Greek-Boy`	`but I just want to terminate a few calls`
`17:20.03`	`Greek-Boy`	`and originate them`
`17:21.46`	`RoyK`	`Greek-Boy: then why on earth would you want to use ss7?`
`17:22.21`	`RoyK`	`if it's just a few calls, do it over SIP`
`17:22.39`	`RoyK`	`if it's a hundred concurrent calls, use PRI termination`
`17:22.43`	`RoyK`	`if it's big, use ss7`
`17:22.48`	`Greek-Boy`	`because I've got a deal with a mobile CDMA company. They only use SS7 for interconnect`
`17:23.07`	`RoyK`	`you don't use exterior BGP on your LAN, do you?`
`17:23.20`	`RoyK`	`oh`
`17:23.21`	`RoyK`	`ic`
`17:23.32`	`hi365`	`how can i set a database value as blank?`
`17:24.07`	`RoyK`	`anyway - if you can afford the sangoma solution, use it. it's supported and it works. the other ss7 stacks around are non-supported and lack a lot of stuff last I checked`
`17:24.26`	`RoyK`	`hi365: I'd guess like a variable Set(ASDF=)`
`17:25.08`	`Greek-Boy`	`RoyK as long as I can get it to make and receive calls then I'm happy. And ofcourse DTMF should work...`
`17:25.28`	`Greek-Boy`	`as for the sangoma solution, I will be sure to use it in my next big project.`
`17:26.03`	`Greek-Boy`	`is the $4k just for software or complete solution?`
`17:26.39`	`*** join/#asterisk shadebob (n=chatzill@84.16.28.38)`
`17:26.42`	`shadebob`	`hi,`
`17:26.43`	`RoyK`	`Greek-Boy: that's for the ss7box. in addition you will need an asterisk server and a Sangoma card for PSTN connectivity`
`17:26.46`	`*** join/#asterisk harryr (n=harryr@cpc3-lamb3-0-0-cust913.bmly.cable.ntl.com)`
`17:27.22`	`RoyK`	`Greek-Boy: I'm not 100% sure of the price, though - contact sangoma`
`17:27.54`	`Greek-Boy`	`thanks RoyK`
`17:29.28`	`RoyK`	`Greek-Boy: np :)`
`17:31.33`	`[TK]D-Fender`	`hi365: .... What database?`
`17:31.42`	`hi365`	`astdb`
`17:31.42`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`17:31.49`	`[TK]D-Fender`	`hi365: Just delete the key.`
`17:32.04`	`hi365`	`and if i dont wan to?`
`17:32.09`	`*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca)`
`17:38.19`	`russellb`	`fwiw, asterisk trunk has some ss7 support in chan_zap as well, via libss7`
`17:38.28`	`russellb`	`which matt f. from Digium has developed`
`17:40.19`	`RoyK`	`methinks a well-proven solution from sangoma might perhaps be slightly better`
`17:40.41`	`russellb`	`you sound like a marketing droid`
`17:40.48`	`*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)`
`17:41.03`	`RoyK`	`russellb: nope`
`17:41.09`	`russellb`	`you probably didn't even know that existed, but you're pretty quick to dismiss it`
`17:41.26`	`russellb`	`so you have no idea what it supports, how many people are using it, ...`
`17:41.32`	`RoyK`	`russellb: or you might say so, but somehow I prefer solutions from sangoma. Those I've tried work for me`
`17:41.46`	`RoyK`	`well`
`17:41.48`	`russellb`	`exactly my point, you dismissed a _free_ option without even knowing a thing about it`
`17:41.54`	`russellb`	`sounds like someone i really want advice from`
`17:41.54`	`RoyK`	`[18:38] <russellb> fwiw, asterisk trunk has some ss7 support in chan_zap as well, via libss7`
`17:42.13`	`RoyK`	`sounds like you're advising people to use pre-alpha software in production`
`17:42.30`	`russellb`	`it's not pre-alpha`
`17:42.35`	`RoyK`	`now that's a good advice....`
`17:42.37`	`BCS-Satori`	`Does anyone have a Cisco 79xx with multiple lines registed none sequentialy. I am able to registed line 1 and 2 on the cisco phone, but if I move line 2 to say line 6, the phone never attempts to register it.`
`17:42.38`	`Nugget`	`they're already using Linux, how much worse can it be? :)`
`17:42.50`	`*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)`
`17:42.54`	`Corydon76-vcch`	`Uh, the Sangoma alterations to Zaptel are, at best, pre-alpha quality`
`17:43.09`	`RoyK`	`"You get what you're paying for and it's free" :)`
`17:43.16`	`Corydon76-vcch`	`Why do you think their changes never got accepted into the codebase?`
`17:43.25`	`RoyK`	`Corydon76-dig: have you seen the actual zaptel patch?`
`17:43.26`	`russellb`	`RoyK: you're such a troll`
`17:43.42`	`Corydon76-vcch`	`RoyK: yes, I have, and it breaks every time we fix a bug in Zaptel`
`17:43.44`	`russellb`	`RoyK: yes, we have all seen their changes`
`17:43.58`	`RoyK`	`russellb: it's six lines or so with defines`
`17:44.05`	`Corydon76-vcch`	`Their patches are essentially band aids to make their stuff work...`
`17:44.19`	`russellb`	`i know what they are`
`17:45.55`	`RoyK`	`well`
`17:45.55`	`RoyK`	`http://karlsbakk.net/zaptel.patch`
`17:46.18`	`RoyK`	`if zaptel is so crippled it can't stand a patch like this, I have a hard time thinking of zaptel as good software`
`17:46.39`	`Corydon76-vcch`	`Then why hasn't Sangoma written their own drivers?`
`17:47.37`	`*** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net)`
`17:49.20`	`RoyK`	`Corydon76-dig: sangoma has their own drivers - it's called wanpipe. only it's easier to interface with zaptel than to write something from scratch. Also, that patch is only necessary if you want firmware HDLC and not HDLC in zaptel`
`17:50.07`	`RoyK`	`Corydon76-dig: now, what about that patch would break anything else in zaptel?`
`17:50.37`	`Corydon76-vcch`	`RoyK: sorry, I really can't go into this right now. I have accounting work to take care of`
`17:51.00`	`JunK-Y`	`Corydon76-vcch: cant or want? :)`
`17:51.18`	`Corydon76-vcch`	`JunK-Y: I have real work to do`
`17:51.33`	`RoyK`	`well - nothing can. it's a patch with three #define statements and an if clause needing one dedicated flag to be set. it can't fail`
`17:52.03`	`Corydon76-vcch`	`You know, the stuff I get paid for?`
`17:52.04`	`Greek-Boy`	`russellb: I have actually being thinking about trying out libss7. But it currently only runs with asterisk trunk`
`17:52.34`	`ZaVoid`	`morning guys`
`17:53.05`	`RoyK`	`evening, ZaVoid`
`17:53.27`	`Absorto`	`Hello! ls`
`17:53.32`	`*** join/#asterisk cesar_CR (n=cesar@201.192.86.6)`
`17:53.48`	`RoyK`	`russellb: ping`
`17:54.08`	`russellb`	`i also have real work to do`
`17:54.33`	`*** join/#asterisk jlar (n=chatzill@santana.office-ww.wideideas.net)`
`17:54.45`	`RoyK`	`russellb: so I guess neither you can explain why those lines of code are so bad?`
`17:55.37`	`JunK-Y`	`RoyK: i really dont understand why each time you come on IRC, it's only to start fights.`
`17:55.50`	`russellb`	`it's a dirty hack for something which can be done another way in zaptel ... there have been list discussions on the topic`
`17:55.59`	`RoyK`	`JunK-Y: I didn't start one.....`
`17:56.07`	`russellb`	`and quite frankly, i don't care to continue discussing this right now`
`17:56.18`	`JunK-Y`	`u keep trying to start one...`
`17:57.17`	`RoyK`	`Honestly, I didn't start it. I was merely talking about Sangoma's ss7 solution and someone started to insult me for not looking at other solutions in the asterisk trunk, which is, by default, unstable`
`17:57.48`	`JunK-Y`	`give a try and report bugs so we can make it stable.`
`17:58.27`	`russellb`	`no, i mentioned another solution, and you immediately dismissed it without knowing a thing about it, and then i called you out on it`
`17:58.38`	`russellb`	`and then it went downhill from there`
`17:58.39`	`*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)`
`17:58.43`	`russellb`	`anyway, back to real work`
`17:59.53`	`*** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com)`
`18:03.10`	`*** join/#asterisk dacs (n=haiger@unaffiliated/dacs)`
`18:03.20`	`dacs`	`Good morning all`
`18:04.36`	`outtolunc`	`if you say so <G> jk'n`
`18:09.16`	`*** join/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk)`
`18:10.37`	`Unst4ble`	`Hey all, I have a SIP account with a provider which i can use from both a SPA3102 and softphone by entering the SIP details. How would i set this up on asterisk? No matter what settings i use i cant get it to make outgoing calls through the trunk.`
`18:11.01`	`*** join/#asterisk jcims (n=chatzill@cpe-71-72-93-210.columbus.res.rr.com)`
`18:11.49`	`jcims`	`good viop provider for inbound 800 access? i want to stand up a conference calling capability for my employees`
`18:12.23`	`jcims`	`lol, s/viop/voip`
`18:14.34`	`jblack`	`~providers`
`18:14.40`	`jblack`	`It was worth a try`
`18:14.43`	`Qwell`	`~itsplist-us`
`18:14.44`	`jbot`	`[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com`
`18:15.03`	`jblack`	`Um, There's a variety of providers. Most of them offer 800, usually for around 4-5 US cents a minute.`
`18:16.07`	`jblack`	`That'll be per caller, and there's usually a 20-25 monthly fee for the number. So, a five person conference will end up costing you about 15 bucks an hour. Probably worth it. :)`
`18:17.07`	`Qwell`	`4-5c? nowhere near`
`18:17.14`	`Qwell`	`usually 2c at worst`
`18:18.32`	`jblack`	`For 800 numbers?`
`18:18.37`	`Qwell`	`yes`
`18:18.39`	`puga`	`anyone knows where can I find CentOS 5 repositories with asterisk pre-compiled?`
`18:18.46`	`*** join/#asterisk tripps (n=ss@72.20.150.196)`
`18:18.52`	`russellb`	`jbot: itsplist-us is also http://www.jnctn.com`
`18:18.53`	`jbot`	`okay, russellb`
`18:18.58`	`russellb`	`another good one that i like ...`
`18:19.04`	`tripps`	`happy new year * people`
`18:19.24`	`mvanbaak`	`how about bandwidth.com ?`
`18:19.29`	`russellb`	`also good`
`18:19.58`	`NovceGuru`	`hey guys, settings up a very very simple IVR here, after the inital call is answered, and exten => s,5,Background(sai-welcome) plays, the call hangs up,I have done a Set(TIMEOUT(response)=10) but I don't believe this is enough`
`18:19.59`	`russellb`	`jbot: itsplist-us is also http://www.bandwidth.com`
`18:20.00`	`jbot`	`russellb: okay`
`18:20.56`	`tripps`	`before I rewrite app_nv_faxdetect, I wanted to ask if anyone had seen anything that ALWAYS assumes inbound calls are faxes, i.e., to set up a dedicated fax extension that plays a proper fax tone and would work with old faxes as well as new ones`
`18:21.12`	`jblack`	`http://connect.voicepulse.com/Rates.aspx says 4.9c I wasn't able to quickly find the toll free rates at the other providers`
`18:21.40`	`russellb`	`i seem to remember 3.9 cents for junction networks ..`
`18:22.37`	`Qwell`	`yikes`
`18:24.07`	`jblack`	`So, I don't quite understand the correction you're providing me. I'd love an amplification so that I don't give that wrong info out again`
`18:25.48`	`NovceGuru`	`after playing my menu I see Auto fallthrough, channel 'SIP/REMOVED' status is 'UNKNOWN'`
`18:26.22`	`jblack`	`novce: have you seen the i and t extensions?`
`18:26.54`	`NovceGuru`	`I guess not :(`
`18:26.55`	`*** join/#asterisk mtryfoss (n=mtryfoss@6.81-166-192.customer.lyse.net)`
`18:27.11`	`jblack`	`Check 'em out. One is for errors, and one is a... kind of default if you fall through`
`18:27.25`	`NovceGuru`	`I see, thanks :)`
`18:27.47`	`NovceGuru`	`actually, just alt-tabbed back to firefox and scrolled down to the I extenion`
`18:28.15`	`jblack`	`I figured something like that.`
`18:29.34`	`*** part/#asterisk ddunavant (n=David@66.170.97.28)`
`18:29.41`	`NovceGuru`	`I need it to just "wait" 10 seconds or something`
`18:29.46`	`*** join/#asterisk ddunavant (n=David@66.170.97.28)`
`18:30.14`	`Havokmon`	`I'm trying to get a handle on SIP and NAT. If I have an asterisk box on an internal corp network, and want to deploy sip phones that could be behind NAT - what's the best network layout for that?`
`18:30.35`	`Havokmon`	`uhh I should say, the deployed phones would be at user's homes, across the 'net`
`18:30.40`	`*** part/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk)`
`18:30.45`	`jcims`	`vpn between the two`
`18:30.48`	`mvanbaak`	`~sipnat`
`18:30.49`	`jbot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`18:30.57`	`*** part/#asterisk jcims (n=chatzill@cpe-71-72-93-210.columbus.res.rr.com)`
`18:31.04`	`Havokmon`	`vpn would be good, but we're cheap`
`18:31.09`	`NovceGuru`	`PPTP then :P`
`18:31.15`	`Havokmon`	`lol`
`18:31.15`	`NovceGuru`	`stun server and port forwarding`
`18:31.23`	`NovceGuru`	`or just port fowarding`
`18:31.44`	`NovceGuru`	`or put the * server in a DMZ`
`18:31.54`	`NovceGuru`	`or give it it's own WAN ip..`
`18:32.02`	`Havokmon`	`port forwarding? I don't want to configure end-user firewalls - or am I overcomplicating the incoming call info?`
`18:32.09`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:32.12`	`NovceGuru`	`no, port fowarding where the * server is`
`18:32.18`	`[TK]D-Fender`	`NovceGuru: You need to set "autofallthrough=no" under [general} <---`
`18:32.23`	`mvanbaak`	`Havokmon: read the guide I posted`
`18:32.35`	`Havokmon`	`yeah I can get to the * - it's receiving calls on the phones I'm worried about..`
`18:32.43`	`Havokmon`	`mvanbaak: I'm bringing it up now - thanks :)`
`18:33.04`	`[TK]D-Fender`	`Havokmon: Read that 1st guide. It covers what you need.`
`18:33.09`	`NovceGuru`	`mvanbaak: i'm behind a nat and recieve calls fine`
`18:33.14`	`Havokmon`	`oh that's it? Damn`
`18:33.18`	`Havokmon`	`Thanks :)`
`18:33.19`	`NovceGuru`	`fender, thanks <3`
`18:33.26`	`*** join/#asterisk Porks (n=Porks@201.62.79.12)`
`18:33.32`	`mvanbaak`	`NovceGuru: same here`
`18:33.49`	`NovceGuru`	`nat=yes`
`18:34.02`	`NovceGuru`	`and it seems to have magically handled about ANY hotel/corp network i've been in`
`18:34.06`	`[TK]D-Fender`	`NovceGuru: more than just that...`
`18:34.19`	`Havokmon`	`I've 2 problems.. 1st, my kphone doesn't seem to work with alsa right, 2nd, the inter-tel phone can't call out because sonicwall causes problems shakes fist`
`18:34.22`	`NovceGuru`	`canreinvite=no,`
`18:34.27`	`NovceGuru`	`host=dynamic`
`18:34.28`	`NovceGuru`	`:P`
`18:34.42`	`[TK]D-Fender`	`Havokmon: Tell your SonicWALL to NOT do any SIP transform.`
`18:34.49`	`Havokmon`	`ok.. so I was just making it more complex than it really is.`
`18:34.52`	`[TK]D-Fender`	`Havokmon: And then jsut forward the ports on it and you'll be jsut fine`
`18:34.58`	`NovceGuru`	`I'm just a n00b, but the defaults for the xlite phone in the example seem to work great`
`18:35.06`	`Havokmon`	`fender: hmmm k`
`18:35.55`	`[TK]D-Fender`	`Havokmon: I'm running jsut fine behind a SonicWALL TZ170 at the office myself...`
`18:36.19`	`Havokmon`	`nope.. something else is going on. Only the Intel-Tel phone doesn't work.. kphone and Grandtech work fine (I don't have the grandtech with me now though)`
`18:36.57`	`mvanbaak`	`we use OpenBSD for our firewalls`
`18:37.10`	`Havokmon`	`I think it has something to do with the application filtering, it WAS catching the Inter-Tel phone as a vulnerability, and dropping the packet.. but I disabled that check - still no go.`
`18:37.28`	*** join/#asterisk `paul (n=aldee@125.252.68.68)
`18:39.46`	`Havokmon`	`Hrm.. this is a pro 3060.. Guy from corp came in and spent a week trying to combine my old Linux fw, and separate OpenSwan box into this beast...`
`18:40.32`	`mvanbaak`	`get OpenBSD, spent an hour and be done with it`
`18:40.50`	`tripps`	`is it possible to skip nvfaxdetect entirely and just call rxfax application? if so, would this work on older faxes or would I have to generate the fax answer tone?`
`18:40.53`	`daven`	`hello, do any of you know where to get help for doing IAX connections to Free World Dialup?`
`18:41.15`	`puga`	`anyone knows where can I find CentOS 5 repositories with asterisk pre-compiled?`
`18:41.23`	`mvanbaak`	`daven: the fwd pages ?`
`18:41.27`	`daven`	`It is trying to register, but appears to be unable to do so, giving errors such as`
`18:41.30`	`daven`	`Jan 3 18:40:28 NOTICE[23716]: chan_iax2.c:7536 socket_read: Registration of '885454' rejected: 'Registration Refused' from: '192.246.69.186`
`18:41.46`	`mvanbaak`	`daven: check your username and password`
`18:41.59`	`daven`	`yep, they seem ok`
`18:43.05`	`[TK]D-Fender`	`puga: just compile from source like the rest of us`
`18:43.25`	`puga`	`[TK]D-Fender okay xD`
`18:44.46`	`outtolunc`	`daven, remember using 'user/peer' is directional`
`18:45.26`	`*** join/#asterisk Assid (n=assid@unaffiliated/assid)`
`18:47.01`	`*** join/#asterisk RoyKa (n=roy@ip-238-3-149-91.dialup.ice.no)`
`18:48.57`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`18:50.26`	`daven`	`outtolunc: is there a place to discuss this that's more relevant to freeworlddialup`
`18:51.36`	`outtolunc`	`no idea, i used FWD like twice... umm like 4-5 years ago`
`18:52.07`	`outtolunc`	`check your user/pass make sure you remove teh ""'s and check your firewall for port 4569`
`18:52.38`	`outtolunc`	`you can also enable iax2 debug`
`18:52.52`	`outtolunc`	`and see 'why' its being rejected`
`18:53.20`	`daven`	`mvanbaak: my username/password all seem to be correct, I've tried the username as both the FWD number and the actual UserName that I've signed up with. I've set up IAX to work with other servers fine before so that should all be fine; I'm guessing it could perhaps be fwd is slow to update or something.`
`18:53.31`	`outtolunc`	`actually you that might not help <G>`
`18:54.14`	`errr`	`is there a device where I can take my pots line from home and plug it in and then have the device connect to my server over the internet which is in a datacenter running asterisk?`
`18:54.43`	`rob0`	`~fxo`
`18:54.44`	`jbot`	`foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo`
`18:54.46`	`mvanbaak`	`errr: yes`
`18:54.58`	`errr`	`sweet`
`18:54.59`	`errr`	`thanks`
`18:55.36`	`errr`	`(that link is bad btw)`
`18:57.28`	`*** join/#asterisk beighto (n=chatzill@12.176.156.130)`
`18:59.59`	`*** join/#asterisk juniper (n=juniper@151.77.143.138)`
`19:01.48`	`juniper`	`i have to do a newbie question`
`19:04.49`	`Havokmon`	`Mint - The nat settings seem to work, I even called from behind the same nat I received to - thanks guys :)`
`19:06.53`	`beek`	`errr: spa3102 Sipura`
`19:07.13`	`*** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com)`
`19:08.43`	`[TK]D-Fender`	`Havokmon: You're welcome`
`19:08.54`	`errr`	`beek: sweet, I was just reading about that wondering if that was what i needed or not. Thanks`
`19:09.11`	`[TK]D-Fender`	`errr: Yes, very cost-effective and flexible little box.`
`19:10.35`	`puga`	`SetVar() was removed from * 1.4 ?`
`19:10.50`	`[TK]D-Fender`	`puga: Yes, and replaced by Set since 1.2`
`19:11.04`	`[TK]D-Fender`	`puga: Yuo had a few years warning on that...`
`19:13.31`	`*** join/#asterisk P4C0 (n=Dark@200.124.22.34)`
`19:14.26`	`P4C0`	`hello guys, can someone explain me why the registry is important? i mean to registry with a proxy server, is that for incoming or outgoing calls??`
`19:15.06`	`[TK]D-Fender`	`~book`
`19:15.08`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`19:15.09`	`[TK]D-Fender`	`~jerjerguide`
`19:15.10`	`jbot`	`[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/`
`19:15.20`	`[TK]D-Fender`	`P4C0:...`
`19:15.22`	`[TK]D-Fender`	`~sipregister`
`19:15.23`	`jbot`	`[~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW)`
`19:16.21`	`P4C0`	`[TK]D-Fender, thank you :) just having problems placing calls, my sip provider doesn't want to give any support so i have to figured it out by myself, thanks for your time`
`19:16.45`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`19:16.47`	`*** join/#asterisk [hC] (n=hardcore@ip67-90-234-94.z234-90-67.customer.algx.net)`
`19:17.07`	`[TK]D-Fender`	`P4C0: perhaps if you pastebinned your failed call with SIP DEBUG enabled for us to look at we might be able to advise you...`
`19:18.06`	`P4C0`	`[TK]D-Fender, thanks, i will do it later, just let me try a couple of things first, i can get calls, but when i place a call the invite doesn't have any username or password, and the provider replies with forbidden`
`19:18.24`	`[TK]D-Fender`	`P4C0: And why DON'T you have a user & pass?`
`19:18.44`	`syzygyBSD`	`can anyone recommend a soft phone for osx?`
`19:18.45`	`[TK]D-Fender`	`P4C0: "registering" has nothing to do with authing calls you place or receive.`
`19:18.46`	`puga`	`using Set() I can change some channel variable like CALLERID(num) ?`
`19:19.13`	`[TK]D-Fender`	`puga: Yes, Set is an "in-place" replacement for SetVar, and CALLERID is a function`
`19:19.20`	`P4C0`	`[TK]D-Fender, that's what i'm checking, i think i have them specified int he sip.conf`
`19:22.31`	`*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)`
`19:24.23`	`*** join/#asterisk Toerkeium (i=oo@201.216.206.221)`
`19:26.40`	`*** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net)`
`19:26.48`	`Havokmon`	`ah ha-.. FYI. Sonicwall Pro has an Intrusion Prevention setting under VoIP "sipXtapi Remove Buffer Overflow" blocks Inter-Tel 8622 from dialing if enabled`
`19:28.16`	`*** join/#asterisk CVirus (n=GoD@196.205.193.171)`
`19:34.56`	`P4C0`	`[TK]D-Fender, i don't know why it's not placing the user/password in the sip... maybe the "insecure=very" have something to do?`
`19:35.21`	`[TK]D-Fender`	`P4C0: perhaps you can SHOW US what you're doing so we don't have to guess....`
`19:35.30`	`P4C0`	`[TK]D-Fender, yes, moment`
`19:42.04`	`*** join/#asterisk patrickteng0615 (n=patrickt@dsl081-050-020.sfo1.dsl.speakeasy.net)`
`19:43.49`	`patrickteng0615`	`hi, i know this probably isn't the right channel for this question, but I was wondering if anyone can point me in the right direction on how to configure a cisco catalyst switch with linksys voip phones`
`19:43.52`	`patrickteng0615`	`?`
`19:44.57`	`fiXXXerMet`	`patrickteng0615: Go figure that I have a 50-page booklet on just that which one of our consultants set up`
`19:45.25`	`fiXXXerMet`	`I've never done it though, so`
`19:45.51`	`patrickteng0615`	`fiXXXerMet: thanks though...that booklet wouldn't be in pdf format would it?`
`19:45.51`	`patrickteng0615`	`:-D`
`19:46.43`	`fiXXXerMet`	`No :(`
`19:47.02`	`*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)`
`19:47.05`	`P4C0`	`[TK]D-Fender, here it is: http://pastebin.com/m20259dc7`
`19:47.19`	`patrickteng0615`	`fiXXXerMet: it's cool, no worries`
`19:48.01`	`rantsh`	`hello people`
`19:48.13`	`rantsh`	`~queue`
`19:48.14`	`jbot`	`Innovative load-balancing/batch-processing system and rsh replacement. URL: http://bioinfo.mbb.yale.edu/~wkrebs/queue.html`
`19:49.07`	`rantsh`	`hey [TK]D-Fender, jbot is your bot isn't it?`
`19:49.42`	`[TK]D-Fender`	`P4C0: SIP/2.0 403 Forbidden (Not Proxy/Gateway) <--- I have a suspicion that they may not allow * as a UA`
`19:49.46`	`*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)`
`19:49.48`	`[TK]D-Fender`	`~jbot`
`19:49.49`	`jbot`	`extra, extra, read all about it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch`
`19:49.55`	`rantsh`	`[TK]D-Fender, the http://bioinfo.mbb.yale.edu/~wkrebs/queue.html link it has on ~queue is broken`
`19:49.56`	`[TK]D-Fender`	`rantsh: Not "officially" ;)`
`19:50.18`	`P4C0`	`[TK]D-Fender, humm, let me check, thanks`
`19:50.31`	`rantsh`	`[TK]D-Fender, oh! so he's just your b*tch ... I get it XD`
`19:50.40`	`[TK]D-Fender`	`rantsh: The Bot belongs to FreeNode and is a member of MANY channels`
`19:50.53`	`[TK]D-Fender`	`rantsh: that is not an * related answer`
`19:51.17`	`rantsh`	`[TK]D-Fender, I noticed...`
`19:51.30`	`[TK]D-Fender`	`rantsh: I can MAKE one for us if there is something relevant to say.`
`19:51.43`	`P4C0`	`[TK]D-Fender, but i'm sending User-Agent: Asterisk PBX in the invite`
`19:51.44`	`[TK]D-Fender`	`rantsh: I maintain the majority of the jbot trainings.`
`19:51.55`	`rantsh`	`[TK]D-Fender, nice`
`19:52.06`	`[TK]D-Fender`	`P4C0: I know.... and I also know of some providers who REFUSE you based on it`
`19:52.31`	`[TK]D-Fender`	`P4C0: "We don't want to hear about your *!"`
`19:52.58`	`[TK]D-Fender`	`P4C0: Some providers are jerks that way. Not saying thats whats happening here, but the error looks a bit like it...`
`19:53.15`	`P4C0`	`[TK]D-Fender, humm but that UA is hard coded? right?`
`19:53.20`	`rantsh`	`[TK]D-Fender, since we're already chatting allow me to ask you something... if I'm using queue(some_queue\|t) the agent is supposed to be able to transfer the call to another extension... I just forgot the most basic little thing`
`19:53.22`	`[TK]D-Fender`	`P4C0: you can override it.`
`19:53.35`	`P4C0`	`[TK]D-Fender, in the config? without recompiling?`
`19:53.39`	`[TK]D-Fender`	`rantsh: What sort of phone are they using?`
`19:53.48`	`rantsh`	`[TK]D-Fender, what is the sequence of buttons? #exten# ? ?`
`19:53.50`	`[TK]D-Fender`	`P4C0: yes, in sip.conf. Go read the sample.`
`19:53.52`	`outtolunc`	`queue(somequeue\|\|t)`
`19:53.56`	`P4C0`	`[TK]D-Fender, thanks`
`19:53.57`	`rantsh`	`sip`
`19:54.08`	`[TK]D-Fender`	`rantsh: "show application queue"`
`19:54.15`	`[TK]D-Fender`	`rantsh: that is NOT a proper answer.`
`19:54.24`	`[TK]D-Fender`	`rantsh: what MODEL exactly?`
`19:54.37`	`[TK]D-Fender`	`rantsh: And what kind of agents?`
`19:55.04`	`rantsh`	`[TK]D-Fender, :O sorry... dinamyc, and some use softphones`
`19:55.15`	`*** join/#asterisk ob_graldo (n=graldo@206.71.78.172)`
`19:55.34`	`ob_graldo`	`i am having an issue with dtmf tones.`
`19:55.42`	`ob_graldo`	`i can dial an internal extension and get it to work,`
`19:55.51`	`ob_graldo`	`but if i dial out to an outside number it does not.`
`19:55.55`	`rantsh`	`outtolunc, according to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue it should be only one \|`
`19:56.00`	`[TK]D-Fender`	`rantsh: .... MODELS!!!!!!!!!!!!`
`19:56.13`	`[TK]D-Fender`	`rantsh: "show application queue" <--`
`19:56.19`	`ob_graldo`	`my set up is i have one asterisk server vs 1.2 connecting to an asterisk gateway vs 1.4 at which point it goes out to a PRI`
`19:56.40`	`[TK]D-Fender`	`ob_graldo: pastebin the failed call at verbose 10, and SIP DEBUG enabled.`
`19:56.42`	`[TK]D-Fender`	`~pb`
`19:56.43`	`jbot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`19:56.44`	`[TK]D-Fender`	`^^^^^^^^^^`
`19:56.57`	`ob_graldo`	`k... sec`
`19:57.01`	`rantsh`	`[TK]D-Fender, they are proprietary softphones, I don't know what else to tell you, they're not grandstream (hard) necessarily, and their certainly not xlite`
`19:57.20`	`[TK]D-Fender`	`rantsh: Do they have a transfer feature of their own built-in?`
`19:57.36`	`rantsh`	`nope`
`19:57.40`	`[TK]D-Fender`	`rantsh: Much better answer BTW.`
`19:57.59`	`rantsh`	`[TK]D-Fender, hehe`
`19:57.59`	`[TK]D-Fender`	`rantsh: Ok, well read the Queue instructions. It'll tell you what it uses.`
`19:59.03`	`rantsh`	`[TK]D-Fender, I did... all it says on transfer is --> 't' -- allow the called user transfer the calling user`
`19:59.09`	`rantsh`	`[TK]D-Fender, it doesn't say how :s`
`19:59.18`	`[TK]D-Fender`	`rantsh: it if doesn't say, start with the assumption that it uses the same means as features.conf`
`19:59.42`	`rantsh`	`[TK]D-Fender, I'll try that... thanks`
`19:59.47`	`P4C0`	`how is called the devices that act like a sipphone but with rj11 jack for normal phones? ATA?`
`20:00.00`	`paul	`can i connect a local number/line to my asterisk server? what are the hardware requirements?`
`20:00.00`	`[TK]D-Fender`	`P4C0: Yes.`
`20:00.07`	`[TK]D-Fender`	`~ata`
`20:00.08`	`jbot`	`rumour has it, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA`
`20:00.12`	`De_Mon`	`I think we might need to improve the documentation of that so it matches the Dial command`
`20:00.45`	`[TK]D-Fender`	`paul: what kind of "line"?
`20:01.16`	`paul	`hmmm... analog line.... i mean a regular phone with a regular number`
`20:01.19`	`paul	`argggg`
`20:01.40`	`ob_graldo`	`[TK]D-Fender: http://pastebin.com/d33dded59`
`20:01.57`	`ob_graldo`	`there is a few other calls in there... too much to take out... but the number dialed was 9832001`
`20:02.32`	`*** part/#asterisk juniper (n=juniper@151.77.143.138)`
`20:03.31`	`paul	`?`
`20:05.44`	`*** part/#asterisk patrickteng0615 (n=patrickt@dsl081-050-020.sfo1.dsl.speakeasy.net)`
`20:06.43`	`*** join/#asterisk lizor (n=liz@office-nat.popcap.com)`
`20:06.57`	`P4C0`	`hum is there a way to get a list of user agents by manufacturer/model?`
`20:08.53`	`P4C0`	`thanks [TK]D-Fender`
`20:09.15`	`ob_graldo`	`if i change my dtmfmode from rfc2833 to dtmfmode=info then it works outboud, but then it stops working for inbound.`
`20:10.16`	`jblack`	`I decided to setup fwd on my server. Does it take some time for them to setup the iax connections?`
`20:13.01`	`*** join/#asterisk Telamon (n=telamon@bridge.isn.net)`
`20:14.25`	`Telamon`	`Is there some way to log something to the asterisk message file from the dialplan? Like a Noop alternative that gets put in /var/log/asterisk/messages?`
`20:16.00`	`jblack`	`Oh well.`
`20:16.12`	`jblack`	`telamon: Hold`
`20:16.36`	`jblack`	`Look at the Verbose() option`
`20:16.47`	`jblack`	`Verbose([level,]message)`
`20:17.32`	`Telamon`	`Excellent, that's exactly what I was looking for. Thanks jblack. :)`
`20:17.38`	`jblack`	`Welcome.`
`20:17.52`	`jblack`	`Have you head anything about iax2 on fwd not working?`
`20:18.11`	`[TK]D-Fender`	`ob_graldo: SIP/2.0 401 Unauthorized <-- YOUR SIP PHONE'S AUTH IS BAD`
`20:18.17`	`Telamon`	`Sorry, no, I just use SIP.`
`20:19.20`	`[TK]D-Fender`	`P4C0: setup X-Lite on your . Then call from X-Lite to and steal its UA string :)`
`20:21.01`	`beighto`	`This may not have anything to do with Asterisk, but after installing an Asterisk system with 8 Polycom phones the network pretty much died. Only one or two computers will work on the internet at a time. The VPN works and one can ping through the VPN tunnels, but not out to the internet. After things got screwy the hub was replaced with a nice beefy switch. The only other networking...`
`20:21.02`	`beighto`	`...component that hasn't been replaced is a Cisco PIX. This is probably just a networking issue, but I was wondering if anybody has encountered this before after installing a new phone system.`
`20:21.04`	`P4C0`	`[TK]D-Fender, they use some occtel ata`
`20:21.42`	`[TK]D-Fender`	`paul: to take in ana analog line you'd use one of any of these for example : Digium TDM400P, Sangoma A200 , or Linksys SPA-3102. For small-time use I'd suggest the Linksys
`20:21.55`	`[TK]D-Fender`	`P4C0: Well just pick a good one to fake out`
`20:22.07`	`*** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk)`
`20:22.22`	`P4C0`	`[TK]D-Fender, but why in the invite there's no digest information?? do * waits for reply to re-send the invite with the digest information?`
`20:22.45`	`[TK]D-Fender`	`P4C0: Could be you need to set the "realm"`
`20:22.53`	`[TK]D-Fender`	`P4C0: that could be it all by itself`
`20:23.08`	`P4C0`	`[TK]D-Fender, what do you mean by real?`
`20:23.10`	`P4C0`	`~realm`
`20:23.37`	`[TK]D-Fender`	`P4C0: Go read up on SIP. I don't understand it well enough to explain properly.`
`20:24.03`	`P4C0`	`[TK]D-Fender, ok, thank you`
`20:28.08`	`NovceGuru`	`dialplan is kinda fun once you barely grasp it`
`20:28.09`	`NovceGuru`	`:P`
`20:29.50`	`jblack`	`I found realm is a great thing to fiddle with to make things not work. ;)`
`20:30.28`	`jblack`	`I think, that to get it to work on linphone for the day I used it, that I set realm to "asterisk". Either that, or the hostname of the server. I can't remember`
`20:32.13`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`20:34.26`	`tripps`	`does anyone have an ancient (G2) fax machine around here we can run a quick test with?`
`20:34.44`	`tripps`	`i.e., one that doesn't generate tones automatically`
`20:35.09`	`tripps`	`i think i have the solution i was looking for to receive faxes from them . .`
`20:35.16`	`jblack`	`heh. From what I'm reading, now is a good time to abandon trying fwd`
`20:36.06`	`NovceGuru`	`funny, vlc wont play the default wav format asterisk sends voicemails in`
`20:36.43`	`P4C0`	`need to get the user agent of octtel SP4220... anyone? :)`
`20:37.23`	`*** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com)`
`20:37.30`	`jstew`	`Greeetings`
`20:38.18`	`jstew`	`I have a bit of a strange request. Anyone know what a milliwatt test number is in the 616 area code? Do they follow a certain scheme?`
`20:40.03`	`*** part/#asterisk Assid (n=assid@unaffiliated/assid)`
`20:40.39`	`Telamon`	`jstew: Try 616-<exchange>-9994. IE 616-958-9994`
`20:41.52`	`jstew`	`Muchos gracias`
`20:43.36`	`*** part/#asterisk nny_1 (n=Scott_My@64.203.239.83.static-pool-4.pool.hargray.net)`
`20:44.32`	`*** join/#asterisk drmessano-LT (n=nonya@207.230.140.240)`
`20:46.27`	`[TK]D-Fender`	`NovceGuru: works for me...`
`20:46.47`	`NovceGuru`	`it plays for about 1 second then stops, wonder whats up with that`
`20:47.01`	`*** join/#asterisk h4lt (n=Gustavo@201-14-145-69.fnsce701.dsl.brasiltelecom.net.br)`
`20:48.18`	`h4lt`	`wow... how many people here! :D`
`20:49.26`	`russellb`	`h4lt: only 2 ... the rest are bots`
`20:49.34`	`russellb`	`it's just me and you`
`20:49.55`	`jblack`	`Input error -3`
`20:51.18`	`[TK]D-Fender`	`h4lt: Should have taken the RED pill :p`
`20:52.11`	`jstew`	`Has callerid reception been a bitch for anyone else with the TDM400P?`
`20:54.39`	`h4lt`	`really? :(`
`20:55.11`	`h4lt`	`I wasn't in irc for a long time (maybe five years). now I returned, to join in the #asterisk-br channel`
`20:55.44`	`[TK]D-Fender`	`jstew: pastebin your zaptel & zapata ana tell us where you're located`
`20:55.46`	`[TK]D-Fender`	`~pb`
`20:55.47`	`jbot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`20:55.48`	`[TK]D-Fender`	`^^^^^^^^^^`
`20:57.24`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`20:58.15`	`*** join/#asterisk ManxPower (n=manxpowe@20.sub-70-218-95.myvzw.com)`
`20:58.22`	`jstew`	`US 616 area 454 exchange and Here you go: http://pastebin.com/d721d14a1`
`20:59.12`	`SwK`	`wanted: iraq DID...`
`20:59.40`	`NovceGuru`	`hmmm doing Dial(extensionhere&extensionthere) is pretty unreliable`
`21:00.01`	`jwh`	`alexcf: did you find out what was causing the zombie channels?`
`21:00.03`	`[TK]D-Fender`	`jstew: immediate=no <- move that ABOVE your "channel" line`
`21:00.23`	`[TK]D-Fender`	`jstew: the restart * and test. Does * wait for the 2nd ring before picking up after that>`
`21:00.30`	`jstew`	`arrggh... that must be it.`
`21:00.30`	`NovceGuru`	`wonder if I can set the first extension to ring X number of times before going to voicemail, even if that client isnt connected`
`21:00.55`	`[TK]D-Fender`	`SwK: We've found the Telco's of Mass DID!`
`21:01.01`	`jstew`	`should it be under [general]?`
`21:01.11`	`[TK]D-Fender`	`jstew: just 2 lines up`
`21:01.33`	`[TK]D-Fender`	`jstew: under the [channels] heading, but above the "channel=" directive`
`21:01.39`	`jstew`	`I see`
`21:02.05`	`[TK]D-Fender`	`jstew: so update, restart *, test, and report.`
`21:06.43`	`jstew`	`yes, it answers after the 2nd ring now. Still no cid info though`
`21:07.14`	`jstew`	`I can see it being sent with ztmonitor`
`21:07.30`	`[TK]D-Fender`	`jstew: Does it work with a regular phone? Also do NoOp(CallerID is "${CALLERID(all)}") as your first line and paste the output`
`21:08.24`	`jstew`	`I'll have to get my hands on a regular phone and report back.`
`21:10.59`	`[TK]D-Fender`	`jstew: would be nice to know you actually HAVE CID :)`
`21:11.34`	`jstew`	`haha, yes it would.... it was working when we pulled the pots lines off of our old phone system.`
`21:12.28`	`syzygyBSD`	`can asterisk act as a softphone?`
`21:14.06`	`[TK]D-Fender`	`syzygyBSD: I suppose in a sense with Chan_oss`
`21:14.18`	`[TK]D-Fender`	`syzygyBSD: Thogh why would you do that?`
`21:14.47`	`syzygyBSD`	`because I can't find any softphones for OSX that don't crash every call`
`21:15.30`	`NovceGuru`	`x-lite?`
`21:15.49`	`syzygyBSD`	`the mac image isn't valid`
`21:16.13`	`[TK]D-Fender`	`syzygyBSD: Ekiga? Zoiper?`
`21:16.20`	`syzygyBSD`	`haven't tried those`
`21:16.21`	`beighto`	`<PROTECTED>`
`21:17.08`	`jstew`	`syzygyBSD: I'm using eyebeam right now as we type.`
`21:17.25`	`syzygyBSD`	`how is eyebeam?`
`21:17.31`	`jstew`	`Zoiper is poop on OSX. So is eyebeam, but it's less poopy.`
`21:18.02`	`syzygyBSD`	`it could just be my computer... I need to reinstall everything. Been testing way too much out on my first mac`
`21:18.03`	`jstew`	`Well it works and has most of the features I need, but the interface is ugly IMHO`
`21:18.48`	`jstew`	`syzygyBSD: Probably not your computer. There are many broken softphones out there for Mac OS. Leopard did something that borked most of them`
`21:19.00`	`*** join/#asterisk merkurie (n=merkurie@c-68-60-85-88.hsd1.mi.comcast.net)`
`21:20.54`	`daven`	`aaaargh, this is IAX free world dialup issue is really beginning to get on my nerves`
`21:21.33`	`hmmhesays`	`why are you using iax with fwd?`
`21:21.48`	`jstew`	`daven: I never was able to register. SIP works fine though`
`21:22.09`	`daven`	`hm`
`21:22.13`	`jstew`	`I think they lie about IAX support lol`
`21:22.20`	`daven`	`well`
`21:22.21`	`syzygyBSD`	`nm, for some reason the x-lite download only did 4 megs`
`21:22.29`	`daven`	`no, there is an option to activate IAX support`
`21:22.42`	`daven`	`but I guess I'll stick to SIP`
`21:23.23`	`mocker`	`If you check their forums, they acknowledge that IAX support blows.`
`21:23.49`	`jstew`	`syzygyBSD: I haven't checked in a month or so but x-lite does not support leopard. Only the pay version (eyebeam) did.`
`21:24.29`	`jstew`	`I used the IAX activate option, waited a week, then 2 weeks... still rejected my registrations, so I just said screw it.`
`21:25.04`	`fiXXXerMet`	`Trying to access my voicemail. I've recorded one, and setup my box in voicemail.conf. Following the debug on command line, I see that I've entered my mailbox id and password correctly, but it tells me incorrect.`
`21:26.07`	`mmlj4`	`how can I tell how many voicemails a particular box has? aside from inside the vm enviromnent, i mean`
`21:26.19`	`merkurie`	`i have dialplan set to playback a sound file on an fxo channel, incoming call, but its like the first second of the sound file gets cut off, i setup an extension to play the sound file and when i call it internally, it plays fine... something with the phone co?`
`21:26.40`	`fiXXXerMet`	`I see that context = default but I set the context to tvicorp... Or so I think that I did.`
`21:26.55`	`fiXXXerMet`	`:q`
`21:28.58`	`*** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep)`
`21:28.59`	`teknoprep`	`ls`
`21:29.02`	`teknoprep`	`hey all`
`21:29.22`	`fiXXXerMet`	`yo teknoprep`
`21:29.24`	`teknoprep`	`hey is it possible to have a hosted PBX ... and then behind a NAT have say 20 phones`
`21:29.26`	`teknoprep`	`hey fiXXXerMet`
`21:29.37`	`teknoprep`	`the NAT'd phones do not have a SIP proxy`
`21:29.48`	`teknoprep`	`they just connect to the Hosted PBX`
`21:29.57`	`teknoprep`	`will this scenario work fine ?`
`21:30.08`	`teknoprep`	`or do i need a SIP Proxy at the location with 20 phones >?`
`21:30.20`	`*** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net)`
`21:30.39`	`hmmhesays`	`server.1.register.1.expires is the registration timeout on polycom right?`
`21:30.40`	`paul	`can i have a custom hold music for each sip users??`
`21:30.48`	`rantsh`	`~transfer`
`21:30.54`	`hmmhesays`	`'paul yes`
`21:31.09`	`rantsh`	`~book`
`21:31.10`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com`
`21:31.36`	`paul	`hmmm: how? i mean where...?`
`21:31.52`	`paul	`sip.conf?`
`21:31.56`	`teknoprep`	`paul are you using FreePBX ?
`21:31.59`	`fiXXXerMet`	`Wondering if anyone can help with my voicemail issue. I am following the example in the book. Debug @ http://pastebin.com/m14c95fdd and .conf files @ http://pastebin.com/m572246d6`
`21:32.03`	`teknoprep`	`if so try joining #freepbx`
`21:32.29`	`teknoprep`	`so anyone on my problem ?`
`21:33.01`	`hmmhesays`	`~setmusiconhold`
`21:33.11`	`paul	`using asterisk`
`21:33.11`	`hmmhesays`	`bah`
`21:33.35`	`hmmhesays`	`show application SetMusicOnHold`
`21:34.14`	`paul	`i mean if a user pressd the hold button the other person will hear a custom music based on the user(who pressd hold)`
`21:35.32`	`hmmhesays`	`set a channel variable that sets the music on hold class based on who called in`
`21:35.37`	`*** join/#asterisk brad[] (i=brad@TMA-1.brad-x.com)`
`21:36.07`	`brad[]`	`Hi folks, can someone enlighten me on what DIALSTATUS=CANCEL means?`
`21:36.20`	`hmmhesays`	`it means the calling party cancelled the call`
`21:36.39`	`hmmhesays`	`before the call was bridged`
`21:39.36`	`brad[]`	`hmmhesays: Okay, so if I'm seeing that in the log of a failed SIP call, that would indicate the call was successfully placed from the SIP client through asterisk and on to the ITSP before failing?`
`21:39.54`	`brad[]`	`hmmhesays: Excluding issues that could have disconnected it`
`21:40.38`	`hmmhesays`	`brad[]: that would indicate that either asterisk or your client are cancelling the call before the itsp terminates it successfully`
`21:40.46`	`hmmhesays`	`you would get a different dialstatus if the itsp was failing the call`
`21:40.53`	`fiXXXerMet`	`Wondering if anyone can help with my voicemail issue. I've left myself a voicemail, but am unable to log in to access it. I am following the example in the book. Debug @ http://pastebin.com/m14c95fdd and .conf files @ http://pastebin.com/m7639e7da`
`21:41.16`	`brad[]`	`hmmhesays: Hm, okay. Thanks`
`21:41.19`	`hmmhesays`	`example: you place a call with your phone to asterisk, you hang up while it is ringing, your phone is going to send a sip cancel message, resulting in dialstatus cancel`
`21:41.45`	`*** join/#asterisk denon (n=denon@tooth.decay.org)`
`21:41.45`	`*** mode/#asterisk [+o denon] by ChanServ`
`21:41.56`	`hmmhesays`	`its actually one of those sip messages that accurately describes the situation`
`21:42.04`	`brad[]`	`hahah.`
`21:43.19`	`hmmhesays`	`fiXXXerMet: you dialing 400 to try and access the voicemail?`
`21:43.42`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`21:44.07`	`lirakis`	`later all`
`21:44.11`	`fiXXXerMet`	`hmmhesays: yes. I get the prompt, enter my informatoin, and get denied.`
`21:44.15`	`*** part/#asterisk lirakis (n=lirakis@65.200.191.253)`
`21:44.54`	`Mavvie`	`both our PABXs which I upgraded to SVN-branch-1.4-r96102M have just paniced.`
`21:45.11`	`*** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.mn.comcast.net)`
`21:45.30`	`syzygyBSD`	`calm them down with some rum`
`21:45.41`	`Qwell`	`Mavvie: use the latest zaptel release, rather than svn right now`
`21:45.42`	`fiXXXerMet`	`lol`
`21:46.32`	`*** join/#asterisk corporeal (n=corporea@24.143.85.194)`
`21:46.53`	`corporeal`	`how easy is it to get a sip server going using asterisk`
`21:46.58`	`Mavvie`	`Qwell: nice :-/`
`21:47.29`	`Mavvie`	`one day the boss says "We need to upgrade immediately", and the next day he says "We have a serious problem" :-P`
`21:47.37`	`paul	`how do you set a custom hold music during transfer (# by default)?`
`21:48.10`	`hmmhesays`	`fiXXXerMet: what is the message that is played back?`
`21:48.10`	`fiXXXerMet`	`I am also getting WARNING[3412]: channel.c:3281 ast_request: No channel type registered for 'IAX2' and I don't know why..... I have an iax provider setup in iax.conf`
`21:48.25`	`hmmhesays`	`and lets see your voicemail.conf`
`21:48.26`	`fiXXXerMet`	`"Login Incorrect"`
`21:48.37`	`fiXXXerMet`	`hmmhesays: http://pastebin.com/m7639e7da`
`21:48.50`	`*** join/#asterisk Porks (n=Porks@201.62.79.12)`
`21:49.55`	`hmmhesays`	`VoicemailMain(@tvicorp)`
`21:50.19`	`hmmhesays`	`I take donations via paypal`
`21:50.43`	`fiXXXerMet`	`Why does it require that, if it's already under the tvicorp context?`
`21:51.03`	`hmmhesays`	`If a mailbox is not provided, the`
`21:51.03`	`hmmhesays`	`calling party will be prompted to enter one. If a context is not specified,`
`21:51.03`	`hmmhesays`	`the 'default' context will be used.`
`21:51.12`	`hmmhesays`	`voicemail context not dialplan context`
`21:52.58`	`fiXXXerMet`	`ah, thank you sir.`
`21:53.10`	`hmmhesays`	`work now?`
`21:53.14`	`fiXXXerMet`	`Yes it does. :)`
`21:53.15`	`Mavvie`	`heh... hangs faster than I can login to it...`
`21:53.48`	`hmmhesays`	`looks like you'll have to start it manually`
`21:54.01`	`*** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com)`
`21:54.04`	`rantsh`	`anyone knows how I can get a chart like this? -> http://lists.digium.com/pipermail/asterisk-dev/2007-January/025690.html`
`21:54.19`	`*** join/#asterisk zuchmir (n=zuchmir@ool-18ba7e18.dyn.optonline.net)`
`21:55.00`	`zuchmir`	`I get "configure: *** The ISDN PRI installation on this system appears to be broken." any ideas?`
`21:55.02`	`Mavvie`	`[~] edwin@k7>ftp -a downloads.digium.com`
`21:55.02`	`Mavvie`	`ftp: connect: Connection refused`
`21:55.03`	`Mavvie`	`brilliant.`
`21:55.10`	`Mavvie`	`oh, it's http`
`21:55.34`	`_ShrikE`	`was there a revision to the tc400b driver that now allows it to handle g.729 calls?`
`21:55.37`	`syzygyBSD`	`Mavvie: just use wget`
`21:55.41`	`_ShrikE`	`i mean 120 calls`
`21:55.50`	`Qwell`	`_ShrikE: it's always been able to do g729`
`21:56.07`	`_ShrikE`	`sorry. i meant 120 concurrent g729 calls.`
`21:56.11`	`murdmath`	`rantsh: show translation in the * consol`
`21:56.13`	`*** join/#asterisk asagage (i=asagage@12.192.197.15)`
`21:56.35`	`*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)`
`21:56.38`	`murdmath`	`rantsh: Ooops wrong table.`
`21:56.59`	`rantsh`	`murdmath, that isn't exactly translation... it's some sort of supported codecs mode`
`21:57.23`	`rantsh`	`murdmath, sorry, hitted enter before noticing you noticed`
`21:57.28`	`asagage`	`I seem to be having a problem between my asterisk server and Cisco call manager. The SIP trunk has no audio on every other call through the trunk. The first call through the trunk works fine. The second concurrent call is silent on both ends. The RTP stream on second call goes from phone to asterisk, but not from asterisk to call manager or vice versa. A third concurrent call will work fine and the forth will fail in the same way.`
`21:57.47`	`Mavvie`	`asagage: scary.`
`21:58.14`	`asagage`	`very`
`21:58.47`	`asagage`	`i see no erros anywhere, just missing rtp`
`21:59.01`	`zuchmir`	`is there a special configuration to use librpi with 64bit linux?`
`21:59.19`	`Qwell`	`zuchmir: no, it just works`
`22:00.06`	`zuchmir`	`quell: i get this when i run configure --with-pri=/usr/lib "configure: *** The ISDN PRI installation on this system appears to be broken."`
`22:00.27`	`jwh`	`21:38:50 < Sharkz> can't remember my darn password..LMAO..`
`22:00.29`	`Qwell`	`did you install libpri?`
`22:00.31`	`jwh`	`oops`
`22:01.14`	`jblack`	`asagage: Neat.`
`22:01.14`	`zuchmir`	`quell: i did "make install" in the libpri-1.4.x folder`
`22:01.28`	`zuchmir`	`then i did the configure`
`22:01.28`	`jblack`	`So, if you have four calls at the same time, the first and third work, the second and fourth don't ?`
`22:01.56`	`Qwell`	`jblack: makes it sound like a feature`
`22:02.02`	`asagage`	`thats right`
`22:02.10`	`jblack`	`Yeah....`
`22:02.33`	`jblack`	`Scientific studies say 1/2 of calls are worse than average, and 1/2 are better than average.`
`22:02.48`	`Qwell`	`we just doubled average`
`22:02.49`	`jblack`	`You're being spared all the bad calls.`
`22:02.55`	`Qwell`	`..or halved it`
`22:03.27`	`Qwell`	`asagage: you would have to look at a SIP debug to see what's different about the calls`
`22:03.41`	`murdmath`	`Qwell: Or made it so all the calls are bad because your dropping all the good ones.`
`22:03.50`	`asagage`	`i dont see anything different`
`22:03.58`	`Mavvie`	`brilliant... now the chan_zap doesn't get loaded.`
`22:04.12`	`asagage`	`would anyone like to see?`
`22:04.17`	`Mavvie`	`[Jan 4 09:04:06] WARNING[19456]: chan_zap.c:904 zt_open: Unable to specify channel 1: Device or resource busy`
`22:04.38`	`*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582264.dsl.bell.ca)`
`22:04.38`	`Mavvie`	`PRI span 1 is down at this moment...`
`22:04.42`	`Mavvie`	`could that be the issue?`
`22:14.40`	`Mavvie`	`that elqRedir script doesn't really work....`
`22:15.09`	`Mavvie`	`mental note: don't upgrade two asterisk boxes at the same time, even if $boss is annoying.`
`22:17.55`	`*** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net)`
`22:20.00`	`murdmath`	`I swear, all telco's are in a race for last place.`
`22:20.14`	`zuchmir`	`when i do make testprilib (in the libpri-1.4.x folder), i get "testprilib.c:46:26: error: linux/zaptel.h: No such file or directory`
`22:20.18`	`zuchmir`	`"`
`22:20.49`	`zuchmir`	`is that normal?`
`22:21.07`	`fiXXXerMet`	`Is ${ARG1} something special that is used in macros?`
`22:21.35`	`fiXXXerMet`	`Like, it'll use the first variable that was used in the context that called the macro?`
`22:23.36`	`Mavvie`	`asagage: do you have the released version of asterisk or the SVN version?`
`22:24.09`	`fiXXXerMet`	`http://pastebin.com/m354bdef7 is my dialplan and I can't figure out why it isn't working. After the 10 second delay, I get a fast busy signal.`
`22:24.10`	`obnauticus`	`If anyone here's good with Chan_Mobile then i need some help`
`22:24.10`	`obnauticus`	`<PROTECTED>`
`22:24.38`	`Qwell`	`Mavvie: your zaptel problem is fixed`
`22:25.05`	`Mavvie`	`Qwell: you're a chamption :-)`
`22:25.06`	`Mavvie`	`chamption`
`22:25.08`	`Mavvie`	`champion`
`22:25.09`	`Qwell`	`I didn't fix it`
`22:25.18`	`Mavvie`	`but still :-)`
`22:25.29`	`Qwell`	`well, feel free to send lots of money then`
`22:25.41`	`Mavvie`	`we already did.`
`22:25.48`	`Mavvie`	`we got free quad E1s for them!`
`22:25.52`	`hmmhesays`	`fiXXXerMet: dialing 260?`
`22:25.53`	`Qwell`	`no, I mean, to me personally :p`
`22:25.59`	`fiXXXerMet`	`hmmhesays: Yes`
`22:26.22`	`fiXXXerMet`	`hmmhesays: Fixed it. The book said MCARO_EXTEN and I put that, but it meant MACRO_EXTEN :)`
`22:26.35`	`fiXXXerMet`	`I've seen quite a few mistakes in the boox`
`22:26.37`	`fiXXXerMet`	`book*`
`22:27.09`	`hmmhesays`	`${ARG1} is equal to voicemail in that case`
`22:28.07`	`hmmhesays`	`i'm guessing you want Dial(${ARG2})`
`22:30.31`	`Mavvie`	`Qwell: I'm kind of confused about the mailman overview, can you tell me which commit message it resolved?`
`22:30.32`	`zuchmir`	`any ideas about this: "testprilib.c:51:17: error: zap.h: No such file or directory"`
`22:31.02`	`Qwell`	`Mavvie: no idea`
`22:31.39`	`Qwell`	`zuchmir: did you...install zaptel?`
`22:33.59`	`zuchmir`	`quell i did (make install; make install-include) in zaptel-1.4.x`
`22:34.52`	`zuchmir`	`i also tried find . -name zap.h in zaptel-1.4.x and zothing showed up`
`22:35.27`	`syzygyBSD`	`zuchmir: try automake`
`22:35.55`	`zuchmir`	`in which folder ? (zaptel / libpri - or both)?`
`22:36.05`	`syzygyBSD`	`um.. both`
`22:36.15`	`Qwell`	`no idea where zap.h is supposed to come from`
`22:37.18`	`zuchmir`	`http://pastebin.com/d3dfbd230`
`22:37.51`	`Mavvie`	`oops... now I have asterisk running in console mode on my laptop.`
`22:37.56`	`nhuisman_work`	`so i'm wondering what I need to put in zapata.conf`
`22:38.10`	`nhuisman_work`	`i currently have a loopback connector on my card`
`22:38.20`	`nhuisman_work`	`but i will be using a t1 pri eventually`
`22:39.11`	`AlexTO`	`Someone can tell me how disable the default cdr and enable the cdr_mysql, i already set the addon and create the DB in MySQL? Thanks..!`
`22:39.55`	`asagage`	`Mavvie: Asterisk 1.4.16-1 RPM by vc-rpms@voipconsulting.nl`
`22:40.04`	`zuchmir`	`after commenting out that "include <zap.h>" i get: http://pastebin.com/d7cd0ef27`
`22:40.10`	`Qwell`	`asagage: trixbox?`
`22:40.14`	`asagage`	`yes`
`22:40.18`	`Qwell`	`~trixbox`
`22:40.18`	`jbot`	`from memory, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org`
`22:40.18`	`nhuisman_work`	`can someone point me in the right direction`
`22:40.24`	`Qwell`	`there's your problem`
`22:40.25`	`ManxPower`	`asagage: except that 1.4.16. has a major security vuln.`
`22:40.58`	`ManxPower`	`~zeeek`
`22:40.59`	`jbot`	`i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."`
`22:41.39`	`jblack`	`ohh, nice fortune cookie. "They will be grateful that you cared enough to make it"`
`22:41.59`	`asagage`	`i am seeing the same issue with another server running Asterisk 1.2.14 svn rev 48468`
`22:42.17`	`nhuisman_work`	`does anyone know how I would find out what signalling and switchtype my t1 pri use?`
`22:42.32`	`syzygyBSD`	`nhuisman_work: where are you?`
`22:42.41`	`syzygyBSD`	`besides work...`
`22:42.50`	`Mavvie`	`nhuisman_work: ask your telco is the first step.`
`22:42.55`	`tzafrir_laptop`	`zap.h is from the obsolete zaptel library`
`22:43.19`	`tzafrir_laptop`	`libzap in Debian until Etch, I think`
`22:43.23`	`nhuisman_work`	`hmm`
`22:43.27`	`nhuisman_work`	`syzygyBSD, hawaii`
`22:43.42`	`syzygyBSD`	`switchtype=national,signalling=pri_cpe`
`22:43.43`	`nhuisman_work`	`I have a cisco vg200 already using the pri`
`22:43.45`	`syzygyBSD`	`try that`
`22:44.07`	`nhuisman_work`	`if i'm testing a loopback will that still work?`
`22:44.12`	`nhuisman_work`	`using a loopback`
`22:44.31`	`zuchmir`	`i'm using libpri-1.2.7, and the testpri.c has that in there`
`22:44.42`	`nhuisman_work`	`isdn switch-type primary-dms100`
`22:44.47`	`nhuisman_work`	`that's in my cisco config`
`22:44.48`	`syzygyBSD`	`uhh, someone else will be able to answer that, I have never tested anything, I just put it all right into production, deal with the screams then`
`22:44.50`	`nhuisman_work`	`i guess that means I need to change it`
`22:44.52`	`Mavvie`	`good, all up and running now.`
`22:44.58`	`nhuisman_work`	`to dms100`
`22:45.02`	`Mavvie`	`mental note: don't upgrade a redundant system in one step.`
`22:45.25`	`nhuisman_work`	`Mavvie, you mean you upgraded both at once instead of one at a time?`
`22:45.42`	`Mavvie`	`yes.`
`22:45.45`	`Mavvie`	`you can shoot me.`
`22:45.47`	`nhuisman_work`	`hehe`
`22:46.03`	`obnauticus`	`Err`
`22:46.08`	`obnauticus`	`on a digium bug report`
`22:46.12`	`Mavvie`	`nhuisman_work: silly $boss was so afraid of the security alert send out yesterday.....`
`22:46.17`	`obnauticus`	`how do I freaking upload a patch and a source that came with my package.`
`22:46.29`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`22:46.43`	`nhuisman_work`	`does anyone know how to test asterisk with my hardware and use a loopback on my t1 port?`
`22:46.49`	`obnauticus`	`nevermind`
`22:46.50`	`obnauticus`	`lol`
`22:47.01`	`mocker`	`Mavvie: Don't let $boss read those.`
`22:47.19`	`Mavvie`	`nhuisman_work: but a common solution is to just try some standard settings and see if they work.`
`22:47.44`	`nhuisman_work`	`it's mostly just wanting to check the gateway hardware`
`22:47.47`	`nhuisman_work`	`it's not a digium card`
`22:48.03`	`Mavvie`	`if you have it working on a Cisco voice router, you should know them.`
`22:48.18`	`nhuisman_work`	`no, its another card`
`22:48.26`	`Mavvie`	`except that that line you mentioned didn't really give me any clues netiher.`
`22:48.28`	`nhuisman_work`	`i can't unplug our current system right now to test it.`
`22:48.44`	`nhuisman_work`	`test plugging in the t1 pri to my redfone gateway.`
`22:48.53`	`nhuisman_work`	`so I made a loopback cable`
`22:48.59`	`zuchmir`	`libpri-1.4.3 also has the include zap.h`
`22:49.03`	`nhuisman_work`	`i just wanted to make sure asterisk saw it.`
`22:49.20`	*** join/#asterisk ez` (n=ez@c75.152.78-116.clta.globetrotter.net)
`22:51.01`	`asagage`	`anyone have any ideas about the sip trunk problem?`
`22:51.57`	`obnauticus`	`~sip trunk`
`22:52.01`	`obnauticus`	`nuts.`
`22:52.03`	`obnauticus`	`nevermind.`
`22:52.30`	`lesouvage`	`nhuisman_work: that is why it wise to always buy a 2 ports card. Then you can use a cross cable to connect both ports and do our testing with one port in TE mode and the other in NT mode.`
`22:52.39`	`nhuisman_work`	`i do have a 2 port`
`22:52.53`	`*** join/#asterisk dkatz334 (n=guest@66.238.199.82.ptr.us.xo.net)`
`22:52.53`	`lesouvage`	`nhuisman_work: great!`
`22:53.01`	`nhuisman_work`	`i'm just trying to find docs to read to tell me how to test that`
`22:53.17`	`lesouvage`	`nhuisman_work: is it a sangoma card`
`22:53.26`	`nhuisman_work`	`no, it's a fonebridge2`
`22:54.19`	`lesouvage`	`nhuisman_work: the idea is to configure one port in TE mode (terminal equipment, normally your asterisk box) and the other one in NT mode (playing telco like kpn)`
`22:54.20`	`nhuisman_work`	`i guess i'd need to set one side to pri_cpe and one side to pri_net`
`22:55.15`	`lesouvage`	`nhuisman_work: make yourself a little isdn crosscable and connect the two ports.`
`22:55.19`	`nhuisman_work`	`yeah I have that now`
`22:55.23`	`nhuisman_work`	`where do I set the types?`
`22:55.28`	`obnauticus`	`K i made a bug report for da stuff: http://bugs.digium.com/view.php?id=11673`
`22:56.01`	`nhuisman_work`	`zttool shows both as "ok"`
`22:56.16`	`lesouvage`	`nhuisman_work: but what are the modes?`
`22:56.33`	`nhuisman_work`	`where do I set the modes? zaptel.conf?`
`22:56.36`	`*** join/#asterisk Maliuta_ (i=nikolai@119.11.102.46)`
`22:56.54`	`nhuisman_work`	`hmm guess not`
`22:57.09`	`lesouvage`	`nhuisman_work: wait a moment, I check for you. Takes a moment.`
`22:57.13`	`nhuisman_work`	`sure`
`22:58.18`	`lesouvage`	`nhuisman_work: /etc/asterisk/zapata.conf`
`23:00.58`	`nhuisman_work`	`yeah i'm trying to figure out what to change in that file`
`23:02.06`	`lesouvage`	`nhuisman_work: I can use pastbin to show you a working example of a 2 port sangoma card`
`23:02.12`	`nhuisman_work`	`sure`
`23:02.24`	`lesouvage`	`nhuisman_work: wait a minute`
`23:02.31`	`*** join/#asterisk fujin (n=aj@unaffiliated/fujin)`
`23:03.16`	`*** part/#asterisk corporeal (n=corporea@24.143.85.194)`
`23:05.12`	`dkatz334`	`Can anybody help me with a really annoying problem related to BLFs and "Early Dial?"`
`23:07.25`	`lesouvage`	`nhuisman_work: check http://www.pastebin.be/7952`
`23:09.18`	`nhuisman_work`	`has anyone here ever had ztcfg crash your server?`
`23:10.09`	`dkatz334`	`yes... with nethdlc enabled.`
`23:10.30`	`nhuisman_work`	`hmm trying to remember what the error message on mine was`
`23:10.38`	`nhuisman_work`	`kernel panic me twice so far though`
`23:10.41`	`nhuisman_work`	`and now its fine`
`23:10.47`	`dkatz334`	`Are you trying to use nethdlc?`
`23:10.56`	`nhuisman_work`	`how can I check`
`23:11.09`	`nhuisman_work`	`my device is tdmoe`
`23:12.47`	`*** join/#asterisk EvilMetal (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net)`
`23:13.00`	`dkatz334`	`what's in your /etc/zaptel.conf?`
`23:13.00`	`nhuisman_work`	`why would asterisk run on span 2 and 3 instead of 1 and 2?`
`23:13.27`	`dkatz334`	`nhuisman: post your zaptel.conf`
`23:13.33`	`nhuisman_work`	`dynamic=ethmf,eth1/00:50:C2:65:D2:52/0,24,0`
`23:13.33`	`nhuisman_work`	`dynamic=ethmf,eth1/00:50:C2:65:D2:52/1,24,0`
`23:13.33`	`nhuisman_work`	`bchan=1-23`
`23:13.33`	`nhuisman_work`	`dchan=24`
`23:13.33`	`nhuisman_work`	`bchan=25-47`
`23:13.34`	`nhuisman_work`	`dchan=48`
`23:13.36`	`nhuisman_work`	`shit`
`23:13.38`	`nhuisman_work`	`i was trying to past the pastbin address`
`23:13.44`	`nhuisman_work`	`http://pastebin.com/m5ac7f6fb`
`23:14.35`	`nhuisman_work`	`also, do you know what these errors mean ? http://pastebin.com/m5bb982ff`
`23:16.41`	`nhuisman_work`	`PRI got event: HDLC Bad FCS (8) on Primary D-channel of span`
`23:16.46`	`nhuisman_work`	`anyone know what those men?`
`23:16.47`	`nhuisman_work`	`mean`
`23:17.04`	`dkatz334`	`never seent he ethmf`
`23:17.05`	`lesouvage`	`nhuisman_work: check http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE`
`23:17.56`	`nhuisman_work`	`ethmf is something special for this gateway I think`
`23:20.59`	`lesouvage`	`nhuisman_work: google is your best friend. check http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.html`
`23:21.11`	`nhuisman_work`	`i am searching google now`
`23:21.19`	`nhuisman_work`	`guess I just didn't see that one`
`23:21.45`	`nhuisman_work`	`ugg that looks bad`
`23:22.44`	`nhuisman_work`	`i wonder if this is happening because it's really just a loopback`
`23:22.49`	`*** join/#asterisk EvilMetal (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net)`
`23:27.47`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`23:34.11`	`*** join/#asterisk HybridStorm (n=HybridSt@adsl-066-156-078-028.sip.asm.bellsouth.net)`
`23:34.48`	`HybridStorm`	`Does anyone know the best way to have a sip asterisk box failover to another location should the internet connection drop out?`
`23:36.30`	`*** join/#asterisk trippss (n=ss@72.20.150.196)`
`23:37.09`	`hmmhesays`	`you can qualify the peer you are calling`
`23:37.24`	`hmmhesays`	`I've found that it works intermittently though`
`23:37.51`	`lesouvage`	`nhuisman_work: Have you checked http://www.mapleleaf-technologies.com/webstore/redfone_ethernetbridges.php -> install_guide link.`
`23:39.52`	`*** join/#asterisk cli4me (n=shizm@cpe-071-070-229-009.nc.res.rr.com)`
`23:40.03`	`hmmhesays`	`low sip registration timeouts help too`
`23:40.05`	`cli4me`	`anyone familiar with polycom dialplan?`
`23:40.38`	`cli4me`	`ip650`
`23:40.50`	`hmmhesays`	`its a pretty basic sudo regex isn't it?`
`23:41.07`	`HybridStorm`	`hmmhesays: so you are saying basically have two running at once and keep the timeout low so it will move over when the first one fails?`
`23:41.35`	`hmmhesays`	`or modify chan_sip to fail right away instead of sending out 6 invites when it doesn't receive a response`
`23:41.36`	`HybridStorm`	`I was thinking more along the lines of having the second asterisk start when something detects the failure of the first`
`23:41.41`	`hmmhesays`	`thats actually what I do`
`23:42.00`	`nhuisman_work`	`lesouvage, yes`
`23:43.17`	`hmmhesays`	`write a small perl script to ping one server and start asterisk if it fails`
`23:45.00`	*** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com)
`23:45.32`	`hmmhesays`	`simple enough`
`23:51.11`	`hmmhesays`	`you could then use the manager interface to modify the extension and point it at your second server`
`23:51.17`	`hmmhesays`	`until the first server comes back online`
`23:51.56`	`hmmhesays`	`hell you could do it based on anything, if your ping times become erratic`
`23:55.05`	`hmmhesays`	`can you set a global variable with the manager?`
`23:57.01`	ez`	`where could i read about redundant server asterisk and incomming zap line example ?`
`23:57.58`	`hmmhesays`	`that gets difficult`
`23:58.48`	`hmmhesays`	`I should write that script for changing the server IP`

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