IRC log for #asterisk on 20100407

`00:00.53`	`jpcansa`	`how can i recognize a 3G phone connecting to my *, so i can open iptables for the connection?`
`00:01.32`	`*** join/#asterisk KavanS (~KavanS@173-12-184-89-oregon.hfc.comcastbusiness.net)`
`00:02.40`	`jsjc`	`Thanks WIMPy, I will start mocking around with it!`
`00:08.48`	`hardwire`	`mucking`
`00:08.50`	`hardwire`	`:P`
`00:09.01`	`hardwire`	`mocking would be counterproductive.`
`00:11.31`	`carrar`	`jpcansa, just open up iptables for the carrier of your phone`
`00:12.25`	`carrar`	`and what are you gonna do when you are on some random wifi`
`00:13.16`	`*** join/#asterisk talntid (~t@173.160.174.126)`
`00:13.53`	`talntid`	`[TK]D-Fender, can I PM you? business opportunity.`
`00:14.02`	`*** part/#asterisk knarfly (~vlad@c-98-242-237-166.hsd1.fl.comcast.net)`
`00:14.24`	`carrar`	`in nigeria?`
`00:14.29`	`talntid`	`no :)`
`00:20.03`	`*** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net)`
`00:20.27`	`bmoraca_work`	`just renumbered my internal network here at work...that was fun. not.`
`00:20.44`	`talntid`	`yeah, that sucks`
`00:20.48`	`talntid`	`did that before`
`00:22.55`	`*** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net)`
`00:24.05`	`*** join/#asterisk manxpower (~ewieling@216.186.151.147)`
`00:25.19`	`talntid`	`any competent PHP coders here, with Asterisk experience, want some work?`
`00:26.10`	`hardwire`	`I can make python whistle dixie.`
`00:26.14`	`hardwire`	`and I avoid PHP`
`00:26.56`	`hardwire`	`which is great because for the most part I don't have limits on what I can use if I'm also using asterisk.. it sort of defines the need for dedicated hardware vs colocated accounts.`
`00:27.29`	`rdircio`	`what needs be done?`
`00:28.05`	`talntid`	`Auto Dialing`
`00:28.11`	`talntid`	`not predictive, but auto dialing`
`00:28.18`	`rdircio`	`spam ?`
`00:28.19`	`rdircio`	`:)`
`00:28.26`	`talntid`	`my auto dialing?`
`00:28.27`	`hardwire`	`yeh.. that's usually my first question :0`
`00:28.39`	`talntid`	`negative. business to business calls.`
`00:28.50`	`hardwire`	`insurance?`
`00:28.54`	`talntid`	`sell the ads and coupons on the back of the reciept tape`
`00:28.59`	`hardwire`	`gotcha`
`00:29.15`	`talntid`	`call the businesses surrounding grocery stores, to set up appointments (not sell over phone)`
`00:29.39`	`talntid`	`have a fully functional call center, calendar system (php) already working..`
`00:29.55`	`*** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com)`
`00:29.59`	`talntid`	`so, auto dialing the leads is the next step in progression :)`
`00:31.58`	`bmoraca_work`	`it's been my experience that projects done for people over the internet suffer from feature creep, and ambiguous requirements`
`00:32.50`	`hardwire`	`and lack of beer`
`00:33.51`	`talntid`	`feature creep?`
`00:34.00`	`talntid`	`as in, "can you add this?" later?`
`00:35.03`	`*** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus)`
`00:35.48`	`*** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net)`
`00:39.45`	`*** join/#asterisk marienz (marienz@freenode/staff/marienz)`
`00:40.08`	`*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)`
`00:43.03`	`*** join/#asterisk coppice (~chatzilla@93.176.64.202.dyn.pacific.net.hk)`
`00:46.35`	`*** join/#asterisk chendy (~chatzilla@204.152.211.137)`
`00:48.16`	`*** join/#asterisk jhirley (~jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net)`
`01:07.26`	`*** join/#asterisk overfien (~overfien@c-98-247-121-77.hsd1.wa.comcast.net)`
`01:12.11`	`*** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net)`
`01:12.11`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`01:17.43`	`*** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68)`
`01:31.01`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`01:48.16`	`*** join/#asterisk inova (~inova@mail.copierts.ru)`
`01:50.35`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`01:53.58`	`*** join/#asterisk OrNix (~ornix@host89-251-107-43.hnet.ru)`
`01:54.29`	`talntid`	`jaytee hey...`
`01:54.33`	`talntid`	`are you a DALneter?`
`01:54.45`	`jaytee`	`not really`
`01:55.00`	`jaytee`	`think I've only been in one channel on that net`
`01:55.10`	`talntid`	`#helpdesk?`
`01:55.25`	`talntid`	`i recognize the nick. not sure if its same person :)`
`02:00.43`	`*** join/#asterisk brendansterne (~brendanst@cpe-70-124-61-17.austin.res.rr.com)`
`02:02.07`	`brendansterne`	`Greetings. I'm new to Asterisk, but comfortable with VoIP / SIP. I'm having an issue with identifying the peer on an inbound Axvoice call.`
`02:02.31`	`brendansterne`	`I was wondering how Asterisk tries to match the incoming invite to a peer definition.`
`02:02.35`	`p3nguin`	`And by what method are you intending to identify said call?`
`02:02.53`	`brendansterne`	`What do you mean? I'm seeing the invite come in.`
`02:03.52`	`p3nguin`	`Calls inbound to Asterisk usually go into a context where an extension match is attempted.`
`02:04.19`	`jsjc`	`what is the easiest fax2email email2fax solution with asterisk 1.6?`
`02:04.25`	`brendansterne`	`How do they know what context to use?`
`02:04.32`	`jsjc`	`using T.38`
`02:04.54`	`brendansterne`	`I'm registering fine with Axvoice. And they're sending the invite.`
`02:05.08`	`p3nguin`	`Contexts are determined by matching peer definitions in your sip.conf, or, if no match is found, the default context is used.`
`02:05.22`	`brendansterne`	`How does peer matching work?`
`02:05.51`	`p3nguin`	`It can be done by IP address and port or by username.`
`02:05.55`	`brendansterne`	`It appeared like it was trying to match the sip From: field`
`02:06.10`	`brendansterne`	`How do you control if it uses IP, port or username?`
`02:06.57`	`p3nguin`	`The type= option determines that. You'll most likely want to use type=peer, which will match on IP/port.`
`02:07.22`	`p3nguin`	`The host= option sets the peer's IP address.`
`02:07.45`	`p3nguin`	`The port= option, which only needs to be used if it is a non-standard SIP port, controls the port.`
`02:08.37`	`brendansterne`	`What if the SIP provider could be sending incoming requests from a variety of IPs?`
`02:09.11`	`*** join/#asterisk jerlique (~traveler@staffmail.esc.net.au)`
`02:09.22`	`p3nguin`	`You can specify multiple peers for the provider, each having its own IP address.`
`02:09.33`	`p3nguin`	`I'm sure there is also a way using ACLs.`
`02:10.01`	`brendansterne`	`I'm getting a strange error message:`
`02:10.22`	`jerlique`	`Hi all - we have a legacy asterisk box running 1.2. It is giving an unauthorised msg when receiving sip INFO dtmf messages from a vega (channel bank). Calls to and from the vega work no problems, so what could cause this?`
`02:10.37`	`brendansterne`	`No user '5129928589' in SIP users list`
`02:10.37`	`brendansterne`	`No matching peer for '5129928589' from '69.90.174.98:5060'`
`02:10.37`	`brendansterne`	`No user '5129928589' in SIP users list`
`02:10.37`	`brendansterne`	`No matching peer for '5129928589' from '69.90.174.98:5060'`
`02:10.37`	`brendansterne`	`No user '5129928589' in SIP users list`
`02:10.38`	`brendansterne`	`No matching peer for '5129928589' from '69.90.174.98:5060'`
`02:10.40`	`brendansterne`	`No user '5129928589' in SIP users list`
`02:10.42`	`p3nguin`	`~pb`
`02:10.43`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`02:11.10`	`brendansterne`	`Ah ok thanks..`
`02:12.03`	`brendansterne`	`http://pastebin.com/88G37umr`
`02:12.07`	`p3nguin`	`I would start with the sip.conf being in the pastebin, then go from there.`
`02:12.25`	`brendansterne`	`You can see my sip.conf`
`02:13.11`	`brendansterne`	`You can also see the error. It seems like it's trying to match the SIP From: field`
`02:13.22`	`p3nguin`	`axvoice is sending calls to you via the 5129928589 username?`
`02:15.36`	`brendansterne`	`I'll add the INVITE to the pastebin`
`02:15.38`	`p3nguin`	`I don't know which version invalidates insecure=very, but just for the heck of it, change that to insecure=port,invite .`
`02:16.28`	`brendansterne`	`ok, let me try that`
`02:18.20`	`jerlique`	`Also (regarding my dtmf problem) I notice that on some traces, my vega transmits "event DTMF 1 8" whilst others show "Signal=8 (new line) Duration=250"`
`02:18.36`	`jerlique`	`and by others - i mean other examples on the internet.`
`02:23.12`	`brendansterne`	`p3nguin: I think that helped. I set the host= parameter.`
`02:23.39`	`brendansterne`	`Now it is matching. Now I'm seeing something else: Looking for s in default`
`02:24.04`	`brendansterne`	`Probably because when Asterisk is registering, it's Contact is s@myip`
`02:30.31`	`p3nguin`	`brendansterne: If the call is trying to match the 's' extension, then they are not sending your phone number appropriately.`
`02:31.07`	`manxpower`	`you could always add the requested extension in your registration statement`
`02:31.31`	`manxpower`	`That may or may not work, depending on the provider`
`02:31.33`	`p3nguin`	`Based on your register statement, it should be sending to the 650352xxxx number.`
`02:31.58`	`brendansterne`	`I can't figure out how to set my register string so that my contact is set properly`
`02:32.09`	`brendansterne`	`Check my pastebin, and you can see the format I'm using`
`02:32.15`	`p3nguin`	`Do you have two phone numbers from that provider?`
`02:32.24`	`brendansterne`	`register=> callvine1:PASSWORD@magnum.axvoice.com:9060/6503520300`
`02:32.28`	`brendansterne`	`No, just one phone number`
`02:32.36`	`manxpower`	`you already have /6503520300 so it should be working.`
`02:32.37`	`p3nguin`	`650352.... ?`
`02:32.42`	`manxpower`	`Do you know what I would do at this point?`
`02:32.50`	`brendansterne`	`Asterisk is REGISTERing with a contact "s@ip"`
`02:32.51`	`manxpower`	`Find a provider that doesn't suck.`
`02:32.57`	`brendansterne`	`Agreed`
`02:33.12`	`brendansterne`	`But Asterisk is setting the REGISTER Contact field to s@ip`
`02:33.20`	`manxpower`	`Oh! I've never heard of that happening before.`
`02:33.51`	`p3nguin`	`Maybe a full sip debug would be useful?`
`02:33.52`	`manxpower`	`brendansterne: I hope that is not your entire sip.conf`
`02:34.08`	`brendansterne`	`nope`
`02:34.11`	`manxpower`	`Um, how do we know asterisk is sending s@ if sip debug has not been run?`
`02:34.45`	`p3nguin`	`manxpower: You know the drill... we don't get the important details unless we demand them half a dozen times.`
`02:35.03`	`brendansterne`	`I'll paste the REGISTER....`
`02:35.09`	`manxpower`	`p3nguin: but where the hell did this come from? 09:32:50 PM) brendansterne: Asterisk is REGISTERing with a contact "s@ip"`
`02:35.19`	`brendansterne`	`By the way, thanks for your assistance with this`
`02:35.32`	`p3nguin`	`shrug`
`02:35.39`	`manxpower`	`sounds to me someone is making unfounded assumptions.`
`02:35.44`	`p3nguin`	`I had to take his word for it, since he wasn't willing to share.`
`02:35.51`	`manxpower`	`ah. sucks to be him.`
`02:36.18`	`manxpower`	`when he provides the sip debug of a REGISTER then I guess someone can help him proceed.`
`02:36.18`	`brendansterne`	`I pasted the REGISTER message. You can see the Contact line.`
`02:36.21`	`brendansterne`	`Contact: <sip:s@204.236.221.166:9060>`
`02:36.31`	`brendansterne`	`Into the pastebin`
`02:36.37`	`brendansterne`	`http://pastebin.com/VtGkzCuB`
`02:36.59`	`manxpower`	`brendansterne: Ah, you smushed everything into one big pastebin.`
`02:37.09`	`brendansterne`	`Sorry... I'm new to this`
`02:37.43`	`manxpower`	`that looks to me like an invite, not a register`
`02:38.06`	`brendansterne`	`Above the invite I pasted a register`
`02:38.18`	`brendansterne`	`Should I be using a new pastebin for each update?`
`02:38.43`	`manxpower`	`brendansterne: every time you do a sip reload.`
`02:39.13`	`manxpower`	`but it does look like asterisk is actually sending that. Does your password or username have an @ in it?`
`02:39.15`	`brendansterne`	`I think I should be able to specify my Contact name with the register => line somehow`
`02:39.30`	`brendansterne`	`No, my username and password are alphanumeric`
`02:39.31`	`manxpower`	`brendansterne: you can and you are. the /whatever is that`
`02:40.02`	`brendansterne`	`It doesn't seem to be taking effect`
`02:40.04`	`p3nguin`	`Maybe sip reload was skipped after that was added?`
`02:40.18`	`brendansterne`	`Let me try it again....`
`02:40.36`	`manxpower`	`brendansterne: perhaps you need to spend some time reading The Book.`
`02:40.39`	`manxpower`	`~answers`
`02:40.40`	`infobot`	`somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt`
`02:41.27`	`p3nguin`	`And I'm still seeing insecure=very, even though I totally recommend changing it to insecure=port,invite since you run * 1.6.0.5.`
`02:42.14`	`brendansterne`	`Sorry... I updated the sip.conf, but not that part of the pastebin`
`02:42.37`	`brendansterne`	`I reloaded. Asterisk is still using Contact: s@...`
`02:42.45`	`jerlique`	`How can I debug why asterisk is responding to an Unauthorised sip message when receiving SIP INFO dtmf? (The sip trace shows the unauthorised, but doesnt give a clue as to why its rejected)?`
`02:43.15`	`p3nguin`	`Contact: <sip:5129928589@69.90.174.98>`
`02:43.24`	`p3nguin`	`Where does this 512 phone number come from?`
`02:43.38`	`p3nguin`	`the calling number?`
`02:43.39`	`brendansterne`	`That's the number I'm calling FROM`
`02:43.42`	`brendansterne`	`yep`
`02:44.37`	`brendansterne`	`If I have a /1234 at the end of my register, does my peer section need to be named [1234] ?`
`02:44.43`	`p3nguin`	`no`
`02:45.06`	`[TK]D-Fender`	`brendansterne: http://pastebin.com/VacyPJst`
`02:45.07`	`p3nguin`	`I prefer to make the peer name something sensible, like the name of the provider.`
`02:46.22`	`brendansterne`	`D-Fender, I'll try that`
`02:46.56`	`p3nguin`	`How about this: 1) make the changes to sip.conf, 2) save, 3) sip reload, 4) show your entire sip debug output`
`02:48.07`	`p3nguin`	`All the piecing things together makes my brain melt.`
`02:48.52`	`p3nguin`	`Or was that the PCP?`
`02:49.00`	`p3nguin`	`Ah well, I can't remember, now.`
`02:49.57`	`brendansterne`	`ok`
`02:51.46`	`brendansterne`	`http://pastebin.com/MVUePidg`
`02:52.15`	`brendansterne`	`As far as I can tell, the problem I'm having is getting Asterisk to set the REGISTER Contact properly`
`02:55.52`	`[TK]D-Fender`	`brendansterne: the REGISTER 200OK's just fine`
`02:56.04`	`[TK]D-Fender`	`brendansterne: Also proven by the fact they send you actual calls following it`
`02:56.21`	`brendansterne`	`Yes. But I'm trying to set the Contact field in the REGISTER`
`02:56.30`	`brendansterne`	`According to the docs /1234 should set the Contact.`
`02:56.39`	`brendansterne`	`\/1234 is the Asterisk contact extension. 1234 is put into the contact header in the SIP Register message. The contact extension is used by remote SIP server when it needs to send a call to Asterisk. See the example below. The default context extension is "s".`
`02:56.57`	`brendansterne`	`Sorry.. that came from: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf`
`02:57.31`	`brendansterne`	`My register = user:pw@server:port/contact doesn't seem to be setting the contact`
`02:58.58`	`p3nguin`	`Yeah, it works for almost everyone else, so not sure how to proceed.`
`03:00.13`	`[TK]D-Fender`	`brendansterne: Contact has no impact on this at all`
`03:00.37`	`p3nguin`	`how can it not?`
`03:00.50`	`p3nguin`	`Contact: <sip:s@204.236.221.166:9060>`
`03:01.00`	`[TK]D-Fender`	`brendansterne: you aren't matching a peer and the contact only changes the INVITE header, which relates to the Extension dialed, not the peer to match`
`03:01.02`	`p3nguin`	`The ITSP doesn't know what contact info to include.`
`03:01.05`	`funtoo_nbu`	`im converting stereo wav files to mono for asterisk moh`
`03:01.10`	`funtoo_nbu`	`anyone know the preferred format?`
`03:01.21`	`[TK]D-Fender`	`[23:00]<p3nguin>Contact: <sip:s@204.236.221.166:9060> <- exten, not account`
`03:01.24`	`funtoo_nbu`	`http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk is this outdated?`
`03:01.29`	`[TK]D-Fender`	`p3nguin: CRAY PERSON :p`
`03:01.31`	`brendansterne`	`D-Fender.... the peer is matching based on the host=`
`03:01.38`	`[TK]D-Fender`	`brendansterne: correct`
`03:01.58`	`brendansterne`	`But the problem I'm now having is that I cannot identify the incoming call`
`03:02.00`	`[TK]D-Fender`	`brendansterne: feel free to drop to username by doing "type=user" instead`
`03:02.12`	`[TK]D-Fender`	`brendansterne: that is the next step`
`03:02.29`	`brendansterne`	`Because the INVITE is addressed to s@ip`
`03:02.43`	`brendansterne`	`Ok, let me try type=user and see what happens`
`03:04.36`	`brendansterne`	`D-Fender... the problem I'm having is identifying the DID of the incoming call.`
`03:04.44`	`brendansterne`	`I'm getting a peer match`
`03:04.52`	`brendansterne`	`But the EXTEN is "s"`
`03:04.58`	`[TK]D-Fender`	`brendansterne: Again the INVITE line is the basis of the exten to match once the SOURCE is matched.`
`03:05.12`	`[TK]D-Fender`	`BrOf so it IS matching the peer now?`
`03:05.16`	`brendansterne`	`My INVITE line is "s@ip"`
`03:05.22`	`brendansterne`	`Yes it IS matching`
`03:05.23`	`[TK]D-Fender`	`brendansterne: Oh, so it IS matching the peer now?`
`03:05.26`	`brendansterne`	`Yes`
`03:05.29`	`[TK]D-Fender`	`braGood.. about time..`
`03:05.34`	`[TK]D-Fender`	`brendansterne: Then go fix your REGISTER`
`03:05.35`	`brendansterne`	`But the INVITE line is "s@ip"`
`03:05.41`	`brendansterne`	`I'm trying to fix my register! :)`
`03:05.47`	`brendansterne`	`How?`
`03:06.19`	`brendansterne`	`I have /6503520300 but the REGISTER doesn't use Contact 6503520300@ip`
`03:06.26`	`[TK]D-Fender`	`brendansterne: if this is your register : register=> callvine1:PASSWORD@magnum.axvoice.com:9060/6503520300 <-- as it is NOW and having restarted , then you need to force to forget the old one...`
`03:06.34`	`[TK]D-Fender`	`brendansterne: Is this "as current"?`
`03:07.07`	`brendansterne`	`I'm not sure what you mean. I have done a sip reload. Asterisk is sending out a REGISTER with Contact: s@ip`
`03:07.19`	`brendansterne`	`Every time I reload it sends out a new registration`
`03:07.20`	`[TK]D-Fender`	`brendansterne: pastebin your actual sip.conf`
`03:07.23`	`p3nguin`	`I think the question was something like, "How do I force it to forget the old one?"`
`03:07.36`	`[TK]D-Fender`	`p3nguin: that's the next step`
`03:07.41`	`*** join/#asterisk paulc (~Paul@unaffiliated/paulc)`
`03:07.52`	`[TK]D-Fender`	`brendansterne: PB "database show sip"`
`03:08.14`	`brendansterne`	`0 results found`
`03:09.00`	`p3nguin`	`Well there's yer problem. :P`
`03:10.21`	`brendansterne`	`I don't understand. When I SIP reload, Asterisk re-registers. When it does so, it is still using s@ip`
`03:10.31`	`brendansterne`	`Isn't that the problem?`
`03:10.37`	`p3nguin`	`Something is amiss, for sure.`
`03:10.59`	`brendansterne`	`when I sip show registry it shows me the registration`
`03:11.09`	`[TK]D-Fender`	`brendansterne: PB "database show"`
`03:11.16`	`brendansterne`	`Host Username Refresh State Reg.Time`
`03:11.16`	`brendansterne`	`magnum.axvoice.com:9060 callvine1 105 Registered Tue, 06 Apr 2010 23:10:13`
`03:11.16`	`brendansterne`	`1 SIP registrations.`
`03:11.25`	`p3nguin`	`But show registry doesn't show the contact info.`
`03:12.16`	`brendansterne`	`database show doesn't show anything except /dundi/secret and /dundi/secretexpiry`
`03:12.31`	`brendansterne`	`Do I need to have something in users.conf ?`
`03:12.44`	`[TK]D-Fender`	`brendansterne: Ok, pastebiin your sip. in its entirety.`
`03:12.49`	`[TK]D-Fender`	`sip.conf`
`03:12.54`	`brendansterne`	`ok ...`
`03:13.20`	`[TK]D-Fender`	`brendansterne: And pastebin a current failed call attempt`
`03:15.13`	`brendansterne`	`http://pastebin.com/CH62PvbZ`
`03:17.30`	`[TK]D-Fender`	`failed call now`
`03:17.32`	`brendansterne`	`http://pastebin.com/7yXWT7BD`
`03:17.40`	`brendansterne`	`But it's not a failed call.`
`03:17.49`	`brendansterne`	`Just a failure to identify the called number`
`03:18.09`	`[TK]D-Fender`	`No matching peer for '5129928589' from '69.90.174.98:9060' <- NOT matching`
`03:18.22`	`[TK]D-Fender`	`brendansterne: Please do not tell me it matches and then show me this`
`03:18.31`	`p3nguin`	`:/`
`03:18.42`	`[TK]D-Fender`	`brendansterne: Ditch the insecure line now and retry`
`03:18.56`	`p3nguin`	`This is a result of type=user.`
`03:19.11`	`brendansterne`	`ok... let me switch back to type=peer`
`03:19.33`	`p3nguin`	`user matches on username, peer matches on IP/port.`
`03:20.33`	`brendansterne`	`switching back to type=peer and I get the match: http://pastebin.com/BJC6exN0`
`03:20.43`	`p3nguin`	`On a different note, I really dislike that you not only allowguest, but your peer's context is also default.`
`03:21.07`	`brendansterne`	`Here's what I'm using asterisk for..`
`03:21.21`	`brendansterne`	`I'm working on an outbound conference calling service`
`03:21.30`	`brendansterne`	`And I need some "robot" people.`
`03:21.48`	`brendansterne`	`Basically phone numbers that, when called, will answer, wait a little bit, then press 1`
`03:21.54`	`brendansterne`	`Then wait some more and play some music.`
`03:22.04`	`brendansterne`	`I'm going to hook asterisk up to several trunk providers`
`03:22.13`	`brendansterne`	`such as Broadvoice, Axvoice, Broadvox`
`03:22.22`	`brendansterne`	`I want 30 + numbers`
`03:22.42`	`p3nguin`	`incoming numbers? You can get them all from a single provider.`
`03:23.11`	`brendansterne`	`Sure... but I'm not sure about reliability right now`
`03:23.19`	`brendansterne`	`I've got Broadvoice to work no problem`
`03:23.27`	`brendansterne`	`I've got Broadvox coming up next`
`03:23.37`	`brendansterne`	`But Axvoice I cannot seem to identify the incoming call`
`03:23.49`	`brendansterne`	`I've used another IP PBX with Axvoice no problem (SoftSPOT)`
`03:24.04`	`p3nguin`	`Maybe they are just crap.`
`03:24.05`	`brendansterne`	`But with Asterisk I cannot seem to figure out how to configure it`
`03:24.19`	`brendansterne`	`But why can't I set the REGISTER Contact field?`
`03:24.26`	`brendansterne`	`The documentation says I can`
`03:24.30`	`brendansterne`	`with my register => line`
`03:24.33`	`p3nguin`	`because of the aforementioned crap.`
`03:25.10`	`p3nguin`	`I use register statements to change my Contact field successfully... just not on axvoice.`
`03:25.33`	`p3nguin`	`Without /number they send to 's' and with it, they send to the number specified.`
`03:25.50`	`brendansterne`	`But it's Asterisk that's sending the Contact`
`03:25.58`	`brendansterne`	`I don't see any problem with the provider`
`03:26.00`	`p3nguin`	`and it's axvoice ignoring it.`
`03:26.11`	`brendansterne`	`Axvoice is not ignoring it`
`03:26.15`	`[TK]D-Fender`	`brendansterne: Lets look at one final potential oddity: do "sendrpid=yes", and "trustrpid=yes" and retest. make sure you have "insecure=no".`
`03:26.29`	`brendansterne`	`ok`
`03:26.44`	`*** join/#asterisk spanning_tree (~ring@189.102.114.22)`
`03:27.13`	`brendansterne`	`are sendrpid=yes and trustrpid=yes global options or part of the peer definition?`
`03:28.24`	`*** join/#asterisk pawz (~pawz@ppp118-208-2-99.lns20.bne1.internode.on.net)`
`03:28.52`	`[TK]D-Fender`	`brendansterne: peer`
`03:31.36`	`Kobaz`	`mmm`
`03:31.46`	`Kobaz`	`okay so, i'm trying to get this adtran 904 going...`
`03:32.08`	`brendansterne`	`D-Fender... still problematic`
`03:32.20`	`Kobaz`	`call comes in on t1, it redircts it to sip... now i'm having problems accepting calls where the from: matches a local sip peer`
`03:32.25`	`brendansterne`	`Maybe I'll just have to give up on Axvoice`
`03:32.39`	`[TK]D-Fender`	`brendansterne: something is awry here...`
`03:32.41`	`brendansterne`	`Even though it appears that it's behaving fine`
`03:32.55`	`Kobaz`	`as seen here: http://pastebin.com/HTGjTMkj`
`03:32.56`	`brendansterne`	`I just can't get Asterisk to set the Contact`
`03:33.01`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`03:33.26`	`brendansterne`	`p3nguin and D-Fender, thanks for your assistance.`
`03:34.03`	`[TK]D-Fender`	`brendansterne: that is OK... if its passed in another header.`
`03:34.13`	`Kobaz`	`actually this is the original invite request that that adtran sends: http://pastebin.com/mQFd8Eh7`
`03:34.15`	`brendansterne`	`It's not passed in another header`
`03:34.29`	`brendansterne`	`The INVITE and TO are both addressed to the Contact`
`03:34.43`	`brendansterne`	`Maybe I can adjust this on my Axvoice settings.`
`03:34.45`	`[TK]D-Fender`	`brendansterne: To = S?`
`03:34.54`	`brendansterne`	`Yep To=s`
`03:34.55`	`[TK]D-Fender`	`brendansterne: pastebin the final matching call`
`03:35.09`	`brendansterne`	`Sure...`
`03:35.17`	`Kobaz`	`how would i make asterisk not match the callerid (4139) against currently defined peers`
`03:36.15`	`brendansterne`	`D-Fender: http://pastebin.com/fbWKbgnd`
`03:37.01`	`p3nguin`	`And we're back to no matching user or peer.`
`03:37.44`	`*** join/#asterisk nosbig (~nosbig@cpe-75-179-4-171.neo.res.rr.com)`
`03:37.48`	`Kobaz`	`[TK]D-Fender: i know your the sip guru :P`
`03:37.59`	`brendansterne`	`Bloody hell yes. But that's not the problem`
`03:38.22`	`brendansterne`	`The problem is: How do I get Asterisk to send a REGISTER with a Contact other than "s"`
`03:38.36`	`p3nguin`	`Yes, I know it's not _the_ problem, but it will eventually be _a_ problem.`
`03:38.49`	`brendansterne`	`Fair enough`
`03:38.58`	`*** part/#asterisk spanning_tree (~ring@189.102.114.22)`
`03:39.11`	`brendansterne`	`But any idea why Asterisk wont send anything other than REGISTER Contact=s@ip`
`03:39.43`	`brendansterne`	`If I want to add /number to my register, does this number have to match up somewhere?`
`03:39.48`	`brendansterne`	`Can I set /number to anything?`
`03:39.50`	`p3nguin`	`I still say that asterisk is sending it correctly and they are ignoring it.`
`03:40.04`	`brendansterne`	`I can show you the SIP trace from Asterisk...`
`03:40.07`	`[TK]D-Fender`	`brendansterne: comment out your register statement. Complete restart *. verify that sip show registry is blank. then re-apply with the /suffix part there.`
`03:40.08`	`*** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net)`
`03:40.18`	`Kobaz`	`do de do`
`03:41.00`	`brendansterne`	`http://pastebin.com/tCkQzXYR`
`03:41.11`	`brendansterne`	`You can see Asterisk is not sending Contact properly`
`03:41.15`	`brendansterne`	`or at least as expected`
`03:42.06`	`brendansterne`	`Here's something else odd, my REGISTER Contact is s@ip:9060`
`03:42.10`	`brendansterne`	`But why 9060?`
`03:42.17`	`brendansterne`	`I don't want to be contacted on 9060`
`03:42.33`	`brendansterne`	`That's the port my provider uses`
`03:43.16`	`Kobaz`	`anyone wanna take a stab before i fall asleep`
`03:43.16`	`*** join/#asterisk Wildy (~simba@91.205.147.94)`
`03:43.23`	`p3nguin`	`Just because they use a silly port (and you use it in your register statement), they should send back to you on a standard port, right?`
`03:43.27`	`nosbig`	`I am trying to resolve an issue I think falls within SIP. I have an IAX provider and SIP phones registered locally. I can make outbound calls, but some inbound calls will ring across my IAX trunk and leave the following SIP trace: http://pastebin.com/QAsd9U0C`
`03:43.40`	`brendansterne`	`Yep.`
`03:43.53`	`brendansterne`	`They're supposed to reach me at my REGISTER Contact`
`03:44.26`	`brendansterne`	`D-Fender... I'll try that.`
`03:44.45`	`nosbig`	`Calls between local SIP extensions work just fine, and outbound calls seem to be working well... Any idea what direction I might be looking for?`
`03:45.15`	`[TK]D-Fender`	`nosbig: how long does the ringing last?`
`03:45.44`	`nosbig`	`My phone rings for a few seconds, but I see that the SIP connection terminates immediately...`
`03:46.31`	`nosbig`	`In the Asterisk console.`
`03:46.45`	`Kobaz`	`i can do a callerid override, and then any source numbers will work, but then i lose callerid`
`03:46.57`	`Kobaz`	`this is so frusterating`
`03:47.15`	`[TK]D-Fender`	`nosbig: * aborts the dial... how long exactly?`
`03:47.28`	`brendansterne`	`ok... I found something interesting`
`03:47.48`	`brendansterne`	`if I remove the :9060 from the register line, asterisk starts sending the /number as my contact`
`03:48.18`	`[TK]D-Fender`	`brendansterne: \o/`
`03:48.23`	`p3nguin`	`But they don't answer you on the standard port, do they?`
`03:48.43`	`brendansterne`	`It appears that they do`
`03:48.51`	`brendansterne`	`\o/ indeed`
`03:48.55`	`brendansterne`	`WTF?`
`03:49.13`	`p3nguin`	`Why would they want you to use an usual port which breaks things?`
`03:49.23`	`p3nguin`	`if the normal port works, that is`
`03:49.34`	`brendansterne`	`Why would sending to an usual port cause * to mess up the contact?`
`03:49.53`	`[TK]D-Fender`	`brendansterne: I've seen some similar odd bugs....`
`03:50.02`	`Kobaz`	`okay so... it seems like the only way to fix my problem is to force callerid to be a fixed number`
`03:50.06`	`[TK]D-Fender`	`brendansterne: I'm thinking "screwball parsing error"`
`03:50.07`	`Kobaz`	`this is retarded`
`03:50.13`	`[TK]D-Fender`	`brendansterne: your format looked just fine`
`03:50.34`	`brendansterne`	`I too am thinking screwball parsing error. Should I investigate + report this?`
`03:50.35`	`p3nguin`	`bug needs to be filed?`
`03:50.46`	`[TK]D-Fender`	`brendansterne: recommended`
`03:50.50`	`[TK]D-Fender`	`Kobaz: do "sendrpid=yes", and "trustrpid=yes"`
`03:50.55`	`Kobaz`	`either that, or figure out a way to have asterisk not try and authenticate against a peer, when a new call comes in from the gateway`
`03:50.59`	`p3nguin`	`1.6.0.5 is pretty old, though.`
`03:51.12`	`Kobaz`	`[TK]D-Fender: the problem is... the gateway doesn't send remote party id`
`03:51.20`	`brendansterne`	`I'm using it because it was pre-built for Amazon EC2`
`03:51.21`	`Kobaz`	`http://pastebin.com/mQFd8Eh7`
`03:51.22`	`[TK]D-Fender`	`Kobaz: ew`
`03:51.25`	`Kobaz`	`[TK]D-Fender: that's what i get`
`03:52.02`	`[TK]D-Fender`	`Kobaz: complete cal please and details to match`
`03:52.09`	`Kobaz`	`[TK]D-Fender: yeah i use the *rpid settings for all my sip links so i can pass callerid nicely`
`03:52.17`	`nosbig`	`The call is aborted as soon as I see the call having been accepted by Asterisk... I would say less than 1 second after the call shows up.`
`03:52.20`	`Kobaz`	`k`
`03:52.28`	`Kobaz`	`this is an incoming call`
`03:52.33`	`Kobaz`	`so there's no much more than that`
`03:52.35`	`Kobaz`	`but here's the ret`
`03:52.37`	`Kobaz`	`rest`
`03:53.01`	`brendansterne`	`Gents... thanks for your assistance... I didn't understand the peer matching etc before.`
`03:53.18`	`brendansterne`	`If I get a chance, I'll see if that bug exists on the latest version of *`
`03:53.47`	`Kobaz`	`[TK]D-Fender: allrightey... http://pastebin.com/PuBegtQm that's the whole thing`
`03:54.19`	`Kobaz`	`Foo"<sip:4139`
`03:54.23`	`Kobaz`	`that's the callerid information`
`03:54.37`	`Kobaz`	`but asterisk is treating it as part of the authentication for the call`
`03:55.12`	`Kobaz`	`it's matching a peer against it`
`03:56.06`	`jerlique`	`guys - any ideas on this one? http://pastebin.com/BNixjExk`
`03:56.43`	`Kobaz`	`jerlique: looks like dtmf`
`03:57.01`	`jerlique`	`Kobaz: yes, but why is it unauthorised?`
`03:58.24`	`ChannelZ`	`(OT) anyone use RapidSSL?`
`03:58.28`	`Kobaz`	`that would depend on your authentication settings`
`03:59.06`	`Kobaz`	`[TK]D-Fender: so far... the only way i can fix it... is if i override callerid to be something that's not a peer`
`03:59.23`	`Kobaz`	`[TK]D-Fender: or just hope that the callerid never comes in as the same as a peer`
`03:59.36`	`jerlique`	`Kobaz: we have tried insecure=very and insecure=port,invite to no avail.`
`03:59.59`	`Kobaz`	`jerlique: it would be username/secret related probably`
`04:01.37`	`Kobaz`	`[TK]D-Fender: everything works great except for that`
`04:02.47`	`[TK]D-Fender`	`Kobaz: type=user, no insecure`
`04:03.00`	`Kobaz`	`mm`
`04:03.01`	`Kobaz`	`k`
`04:03.40`	`*** join/#asterisk diegomad (mad@190.158.78.204)`
`04:03.52`	`Kobaz`	`same`
`04:04.23`	`Kobaz`	`it's matching agsinst the peer 4139 (which is the callerid)`
`04:04.27`	`jerlique`	`Kobaz: We dont require the vega to register to asterisk. I have tried having no username/secret specified too. Whats strange is that the vega has already successfully passed the call to asterisk, without auth, but to send dtmf as part of the call it fails with the unauth msg`
`04:04.28`	`[TK]D-Fender`	`Kobaz: :/`
`04:06.07`	`Kobaz`	`isn't there a way i can like... match anything from that ip`
`04:06.07`	`Kobaz`	`From: "Foo"<sip:4139@192.168.5.90:`
`04:06.08`	`Kobaz`	`192.168.5.90 is the gateway`
`04:06.08`	`Kobaz`	`anything from that ip, is the gateway, and only the gateway`
`04:06.08`	`Kobaz`	`those other peers don't even have ips`
`04:06.23`	`Kobaz`	`http://pastebin.com/dFkGxbAN`
`04:06.26`	`Kobaz`	`it's matching based on username`
`04:07.29`	`Slugs_`	`~book`
`04:07.30`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`04:07.57`	`Kobaz`	`do i have to rename all my peers so that they don't match?`
`04:08.17`	`jerlique`	`Kobaz: http://pastebin.com/LrKgHgaz`
`04:08.35`	`*** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-ifnqfdglktpludfq)`
`04:09.10`	`Kobaz`	`jerlique: that doesnt help`
`04:09.18`	`Kobaz`	`jerlique: you need to make sure everything matches on both sides etc`
`04:09.19`	`Kobaz`	`anyways`
`04:09.25`	`Kobaz`	`i need to take a shower and pass out`
`04:09.45`	`Kobaz`	`[TK]D-Fender: there's gotta be a way to fix this`
`04:11.14`	`jerlique`	`Is there a way to may asterisk not require the username and seceret?`
`04:11.23`	`[TK]D-Fender`	`Kobaz: restrict your peers. permit/deny, etc`
`04:11.26`	`jerlique`	`(like it does for the INVITE)`
`04:11.26`	`*** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205)`
`04:12.09`	`aceio`	`i am trying too understand why my extention are assigned the same ip address`
`04:12.44`	`aceio`	`this is highlighted in sip show peers`
`04:14.13`	`*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uvtmymokjqiqsais)`
`04:14.18`	`aceio`	`this is my sip show peers result http://pastebin.com/YGHandew`
`04:20.29`	`aceio`	`can anyone help`
`04:24.01`	`ChannelZ`	`you have two devices registering from the same IP?`
`04:24.13`	`TJNII`	`First off, those arn't extensions. They are sip peers. Secondly, are 1260 and 1261 on the dame device>`
`04:24.44`	`ChannelZ`	`I'm guessing you changed the phone's config and re-regged but * hasn't forgotten about the other yet`
`04:28.32`	`*** join/#asterisk tengulre (~tengulre@125.71.208.16)`
`04:28.33`	`TJNII`	`Uggggh.... So pastebin is hosting a poll about disallowing anonymous posting, and googling the name of one rabid poster I found this gem: http://www.paynelessdesigns.com/about.shtml`
`04:28.37`	`Kobaz`	`[TK]D-Fender: k, thanks, that sounds like it'll work... night night`
`04:28.47`	`TJNII`	`Turn off javascript to see some awesome use of the blink tag....`
`04:28.56`	`aceio`	`ChannelZ: correct`
`04:31.27`	`*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)`
`04:32.57`	`*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)`
`04:34.26`	`aceio`	`ChannelZ: are you saying i should register the phones`
`04:35.17`	`ChannelZ`	`I'm saying that you should unregister the bogus one`
`04:35.27`	`ChannelZ`	`sip unregister 1260 (or whichever one is wrong)`
`04:36.35`	`aceio`	`ChannelZ: i understand mow`
`04:38.45`	`*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)`
`04:44.05`	`aceio`	`ChannelZ: i don't have this command sip unregister on my cli`
`04:44.47`	`*** join/#asterisk williamtaft (~seangrego@c-71-57-7-193.hsd1.il.comcast.net)`
`04:45.25`	`williamtaft`	`can anybody help me understand an asterisk configuration?`
`04:45.33`	`aceio`	`i am version Asterisk 1.4.30`
`04:47.57`	`funtoo_nbu`	`if i want an inbound call to go directly on hold i can just go exten => number,1,MusicOnHold() on my inbound trunk ye?`
`04:48.34`	`aceio`	`ChannelZ: what is the command to re-register a sip phone`
`04:49.03`	`ChannelZ`	`That's up to the phoen`
`04:49.06`	`ChannelZ`	`phone`
`04:50.52`	`aceio`	`ChannelZ: okay i am useing a cisco 7940 phone`
`04:51.15`	`ChannelZ`	`reboot it`
`04:52.09`	`aceio`	`ChannelZ: yeah i completed more than once already`
`04:52.50`	`aceio`	`ChannelZ: and still i am getting dup ip address`
`04:53.09`	`ChannelZ`	`well there's any number of explanations then; it's not configured to register correctly, or it already has and you haven't realized it, or the network is not functioning, or....`
`04:54.16`	`ChannelZ`	`Is this perhaps one setup with 2 lines as separate peers?`
`04:55.12`	`aceio`	`ChannelZ: let me show you my sip.conf file`
`04:55.41`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`04:55.45`	`aceio`	`and SIP<mac-addr.cnf>`
`04:56.01`	`*** part/#asterisk williamtaft (~seangrego@c-71-57-7-193.hsd1.il.comcast.net)`
`04:56.09`	`ChannelZ`	`I don't think I need to see it`
`04:57.00`	`funtoo_nbu`	`no one quickly can tell me how to auto put all incoming calls on hold? im trying to test my moh changes remotely :/`
`04:57.25`	`aceio`	`or maybe P003-08-11-00.bin has a bug`
`04:57.26`	`ChannelZ`	`funtoo_nbu: put all currently active calls..?`
`04:57.39`	`funtoo_nbu`	`nope if i just call in on a certian number`
`04:57.52`	`funtoo_nbu`	`can just go exten => number,1,MusicOnHold() on my inbound trunk ye?`
`04:58.08`	`ChannelZ`	`Yes`
`04:58.10`	`ChannelZ`	`Answer first`
`04:58.25`	`*** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net)`
`04:58.39`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`04:58.46`	`*** join/#asterisk pinoyskull (~pinoyskul@122.55.80.194)`
`04:58.47`	`funtoo_nbu`	`answer 1st, okay`
`04:58.58`	`ChannelZ`	`I think MusicOnHold is one of those apps that doesn't do it for you but it won't kill anything either way`
`04:59.00`	`jerlique`	`Why would INFO be missing from an asterisk response..... "Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"`
`05:00.21`	`ChannelZ`	`jerlique: I think INFO is only applicable to an established call, but I could be wrong.`
`05:01.00`	`aceio`	`exit`
`05:01.22`	`Defraz`	`I have a few random calls that the audio to the caller is delayed at the first of the call. For example someone calls into a DID of ours and the person on the SIP phone that answers has to say hello a couple times before the caller can hear.`
`05:01.39`	`Defraz`	`Being that it is random would that be the SIP carrier we are getting our trunks from?`
`05:01.54`	`funtoo_nbu`	`ChannelZ: okay my asterisk buddy final question`
`05:01.55`	`Defraz`	`I can't seem to find anything on those calls tha tlooks any differenet in the logs.`
`05:02.08`	`funtoo_nbu`	`the volume is too high on my music on hold, which is mono wav`
`05:02.19`	`manxpower`	`turn off reinvites`
`05:02.30`	`funtoo_nbu`	`can this be altered via asterisk or should i mess with the wav files themselves`
`05:03.11`	`ChannelZ`	`funtoo_nbu: just lower the files themselves. If you're using mp3 and an external encoder you could do it there but you're just adding extra processing`
`05:03.28`	`funtoo_nbu`	`yes i have opted not to do that`
`05:03.32`	`funtoo_nbu`	`okay thx a milli`
`05:04.56`	`ChannelZ`	`I normalize to around -13db for a a good 'background music' level that's not obnoxious`
`05:05.14`	`funtoo_nbu`	`how to determine what they are currently at?`
`05:05.22`	`jerlique`	`ChannelZ: Ta, so then here is the full trace... http://pastebin.com/hT0ZYdYi`
`05:05.28`	`funtoo_nbu`	`looks like sox just axes a percetange`
`05:05.39`	`ChannelZ`	`funtoo_nbu: hmm`
`05:05.44`	`ChannelZ`	`not sure with sox`
`05:05.50`	`funtoo_nbu`	`ill just keep a backup`
`05:05.57`	`funtoo_nbu`	`i wish we could juse use the awesome asterisk moh`
`05:06.02`	`funtoo_nbu`	`use*`
`05:06.52`	`*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)`
`05:07.27`	`funtoo_nbu`	`ive caught myself dancing to macroform-robot_dity.wav on several occasions`
`05:11.38`	`ChannelZ`	`jerlique: is this a fax?`
`05:11.50`	`*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)`
`05:12.19`	`jerlique`	`ChannelZ: no incoming voice call.`
`05:13.57`	`ChannelZ`	`hmm well it's sending dtmf digits as SIP INFO messages`
`05:15.19`	`ChannelZ`	`but they don't look quite right from what I've seen, but generally you use rfc2833. Is this coming from an ITSP?`
`05:18.06`	`jerlique`	`ChannelZ the DTMF is coming from the VEGA a BRI to sip gateway`
`05:18.34`	`jerlique`	`ChannelZ: Yeah the weird thing is that asterisk is rejecting the DTMF`
`05:19.17`	`ChannelZ`	`they don't look formatted right which is probably why`
`05:20.00`	`ChannelZ`	`Can you configure the way it's doing it (preferably to do rfc2833)?`
`05:29.07`	`funtoo_nbu`	`Anyone know if I can have a certain file played 1st on moh and then a random one afterwards?`
`05:30.40`	`*** join/#asterisk styelz (~yoohoo@m0o0.mooo.com)`
`05:34.48`	`ChannelZ`	`the only way I could think is to call MusicOnHold(first,30) where 'first' is a MOH context with just your one file, and 30 is the length of said file`
`05:35.12`	`ChannelZ`	`Then call MusicOnHold(random) where 'random' is an MOH context with your random files`
`05:35.41`	`*** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485)`
`05:36.05`	`ChannelZ`	`So for instance if your first track is 48 seconds, pad it with a couple of seconds of silence at the end and call MusicOnHold(first,49) to be safe`
`05:37.41`	`ChannelZ`	`runs off to shower`
`05:51.05`	`*** join/#asterisk ktwilight[m] (~ktwilight@91.179.89.252)`
`06:02.59`	`*** join/#asterisk e-jones (~jkastner@nat/redhat/x-jojznaetvzmtkqnt)`
`06:14.59`	`*** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire)`
`06:15.08`	`*** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire)`
`06:18.42`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`06:32.11`	`*** join/#asterisk mykhyggz (~col@evolone.org)`
`06:36.20`	`*** join/#asterisk mykhyggz (~col@evolone.org)`
`06:56.11`	`*** join/#asterisk mpe (~mpe@gate.ipvision.dk)`
`06:57.21`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`06:59.49`	`*** join/#asterisk tamiel (~tamiel@213.30.183.226)`
`06:59.59`	`*** join/#asterisk soman (~somnath@stargate.starnet.fi)`
`07:00.55`	`jsjc`	`anyplace where I can find some info in how to setup fax for asterisk res_fax res_fax_digium to send faxes to email address and to transmit from email?`
`07:01.21`	`*** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net)`
`07:11.35`	`Polysics`	`wow, pjsua is truly great`
`07:11.41`	`Polysics`	`anyone using that for testing?`
`07:11.53`	`Polysics`	`i can't find out what wav format it wants from me :-)`
`07:13.25`	`*** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)`
`07:16.24`	`*** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net)`
`07:18.24`	`*** join/#asterisk c0rnoTa (~c0rnoTa@178.176.244.174)`
`07:27.05`	`*** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net)`
`07:27.05`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`07:30.34`	`*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)`
`07:37.23`	`*** join/#asterisk Z_God (~julius@wlan232215.mobiel.utwente.nl)`
`07:39.30`	`ChannelZ`	`Polysics: What is that some sort of pajama service?`
`07:39.36`	`Polysics`	`:-)`
`07:39.43`	`Polysics`	`pjsua is a command line SIP client`
`07:39.49`	`ChannelZ`	`oh. hmm`
`07:40.07`	`Polysics`	`which has some very useful features, such as multiple accounts and auto-answer with a sound file`
`07:40.25`	`Polysics`	`so far i have tried a lot of formats, and the only one that works is the bundled test file`
`07:40.36`	`Polysics`	`sox reports that and my files to be idnetical, but hey :-)`
`07:40.45`	`Polysics`	`it is sort of OT`
`07:40.56`	`Polysics`	`but i was hoping someone had the same problem`
`07:41.42`	`ChannelZ`	`"File MUST be single channel/mono, 16bit signed PCM, with any sampling rate."`
`07:41.49`	`ChannelZ`	`I assume this means raw, not wav or any wrapper`
`07:44.15`	`*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)`
`07:44.32`	`*** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net)`
`07:45.24`	`Polysics`	`i will try that too`
`07:48.46`	`*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)`
`07:52.15`	`*** join/#asterisk smooth_penguin (~smoove@59.95.38.210)`
`07:52.58`	`Polysics`	`ChannelZ, nah, says not a valid WAVE file xplicitly`
`07:57.30`	`*** join/#asterisk Tim_Toady (~moi@77.49.52.245.dsl.dyn.forthnet.gr)`
`07:58.01`	`*** join/#asterisk Madkiss (~madkiss@freenode/staff-emeritus/madkiss)`
`07:58.02`	`Madkiss`	`hi folks`
`07:58.23`	`Madkiss`	`is anybody aware of an OCF ressource agent for asterisk?`
`08:00.43`	`*** join/#asterisk smooth_penguin (~smoove@59.95.38.210)`
`08:01.20`	`*** join/#asterisk adnc (~numer@unaffiliated/adnc)`
`08:01.46`	`*** join/#asterisk Tim_Toady (~moi@77.49.52.245.dsl.dyn.forthnet.gr)`
`08:13.08`	`*** join/#asterisk Faustov (user@gentoo/user/faustov)`
`08:23.14`	`*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)`
`08:23.18`	`*** join/#asterisk c0rnoTa (~c0rnoTa@178.176.244.174)`
`08:28.15`	`*** join/#asterisk sulex (~sulex@93-35-4-103.ip52.fastwebnet.it)`
`08:30.28`	`*** join/#asterisk smooth_penguin (~smoove@59.95.38.210)`
`08:33.15`	`jsjc`	`I have found that my SIP provider has fax passtrough so where could I find some info to start mocking around with that?`
`08:35.56`	`*** join/#asterisk krion (~seb@unaffiliated/krion)`
`08:49.48`	`*** join/#asterisk Buggaboo (~Buggab00@5353059E.cable.casema.nl)`
`08:49.50`	`Buggaboo`	`hi`
`08:52.21`	`*** join/#asterisk adnc (~numer@unaffiliated/adnc)`
`08:52.24`	`*** join/#asterisk chendy_ (~chatzilla@204.152.211.137)`
`08:56.52`	`*** join/#asterisk smooth_penguin (~smoove@59.95.3.157)`
`09:01.12`	`*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)`
`09:09.08`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`09:10.59`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`09:12.38`	`*** join/#asterisk pentanol (~pentanol@77.35.49.164)`
`09:12.43`	`*** join/#asterisk adnc (~numer@unaffiliated/adnc)`
`09:15.48`	`*** join/#asterisk Researcher (~user@unaffiliated/unafilliate)`
`09:16.27`	`*** join/#asterisk chendy_ (~chatzilla@116.24.38.161)`
`09:16.45`	`*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)`
`09:25.28`	`*** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35)`
`09:46.22`	`*** join/#asterisk WimD (~Wim@83.101.79.13)`
`09:54.50`	`*** join/#asterisk sulex (~sulex@93-35-4-103.ip52.fastwebnet.it)`
`10:03.49`	`*** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110)`
`10:12.39`	`*** join/#asterisk ttwhy (~tekkno@p4FECFEEC.dip.t-dialin.net)`
`10:13.00`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`10:17.34`	`*** join/#asterisk ChrisWi (~admin@mx2.wwserver.net)`
`10:19.59`	`*** join/#asterisk mikkel (~mikkel@130.226.36.170)`
`10:26.50`	`krion`	`anyone with a software name who "analyse" asterisk logs`
`10:26.59`	`krion`	`sort of awstats for asterisk ?`
`10:37.32`	`*** join/#asterisk pawz (~pawz@210.56.87.168)`
`10:37.54`	`tzafrir_laptop`	`krion, what for?`
`10:38.09`	`tzafrir_laptop`	`if you're after stas, maybe use CDR?`
`10:49.29`	`*** join/#asterisk aceio (~90fe6658@gateway/web/freenode/x-dyatgppbkoydaogc)`
`10:51.47`	`*** join/#asterisk ttwhy (~tekkno@p4FECF938.dip.t-dialin.net)`
`11:09.14`	`*** join/#asterisk pawz (~pawz@210.56.87.168)`
`11:13.38`	*** join/#asterisk Intel`` (~fdjgjd@213.132.40.2)
`11:14.16`	Intel``	`hi guys. how come my x-lite suddeny disconnects and gives me 503 service unavailable?`
`11:14.27`	Intel``	`does this has to do with re-registering?`
`11:16.40`	`*** join/#asterisk the_weard (~mitch@196.212.100.148)`
`11:17.19`	`the_weard`	`hey`
`11:17.29`	`vlt`	`Hello. I compiled t38modem from source on a fresh Debian install. When I try to run it I get "error loading avcodec - avcodec: cannot open shared object file: No such file or directory". Any idea what pkg is missing here?`
`11:17.46`	`*** join/#asterisk Tulga (~chatzilla@203.91.113.10)`
`11:18.02`	`Tulga`	`I have queue and 10 agents. I need monitor them. how to do it?`
`11:18.19`	Intel``	`what kind of monitor?`
`11:18.43`	Intel``	`call listen?`
`11:18.44`	`Tulga`	`listen agent voice`
`11:18.49`	`Tulga`	`yes`
`11:18.55`	`the_weard`	`how do i install asterisk-sounds-core-en-g729 on asterisk only using g729 codec for all phones`
`11:19.11`	`*** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it)`
`11:19.45`	`*** part/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it)`
`11:19.58`	Intel``	`Tulga, what asterisk are you using?`
`11:20.10`	`*** join/#asterisk sourcode (~code@ppp-58-8-131-16.revip2.asianet.co.th)`
`11:20.12`	Intel``	`did you compile it yourself or used a bundles one?`
`11:20.16`	`Tulga`	`1.6.2.6`
`11:20.19`	Intel``	`like asterisnow`
`11:20.20`	`the_weard`	`so when my phone is put on hold or aney thing else i get your call cannot be completed pleas check the number and try again later but everything works fine`
`11:20.23`	`Tulga`	`yes compile myself`
`11:20.28`	`the_weard`	`transver and hold works`
`11:20.44`	`Tulga`	`how about queuemetrics?`
`11:23.13`	Intel``	`i could not find my backedup dial pla for that chanspy`
`11:23.35`	`Tulga`	`please find. I'll wait you :)`
`11:23.46`	Intel``	`please check this. http://www.freepbx.org/forum/chanspy-customization-in-freepbx`
`11:24.16`	Intel``	`it might be freepbxbut most of it can be used for asterisk installed from source`
`11:24.46`	`Tulga`	`ok`
`11:25.18`	Intel``	`basically you ned to create a configuration for channel spy(chanspy)`
`11:25.46`	Intel``	`then restart asterisk then dial the number you assigned, input password then input extension`
`11:26.53`	`*** join/#asterisk Faustov (user@gentoo/user/faustov)`
`11:27.25`	`pentanol`	`hello, may I use list ip's via ODBC so as to connect there over trunks... i.e to avoid definition much trunks in sip.conf and use it over ODBC from database...?`
`11:27.35`	`ttwhy`	`hmm does someone have any good softphone which is (whenever possible) crossplatform ready?If tried ekiga, but i doesn't seem to be reliable`
`11:27.36`	`Tulga`	Intel``: I changed extensions.conf
`11:27.41`	`Tulga`	`how to reload without restart?`
`11:28.50`	Intel``	`im using asterisknow so it might be different`
`11:29.13`	Intel``	`what i do is "amportal restart"`
`11:29.21`	`pentanol`	`I've clue to define in database just trunk names and then checkout it available or used`
`11:31.44`	`*** join/#asterisk kotp (~vgoff@96.2.187.66)`
`11:35.49`	`*** join/#asterisk smooth_penguin (~smoove@59.96.228.109)`
`11:38.30`	`Tulga`	Intel``: dialplan reload works
`11:38.36`	`Tulga`	`now chanspy works best`
`11:38.38`	`Tulga`	`I love it!`
`11:48.28`	`*** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55)`
`11:50.55`	`*** join/#asterisk hluesea (~hluesea@88.247.127.66)`
`11:53.33`	`the_weard`	`how the hell do i use make menuselect on asterisk`
`11:53.45`	`the_weard`	`it does nothinf`
`11:53.47`	`the_weard`	`it does nothing`
`11:55.13`	`carrar`	`You have gcc installed?`
`11:59.12`	`tzafrir_laptop`	`the_weard, nothing? no error?`
`12:06.16`	`aceio`	`i am still trying to resolve a problem were my sip peers have dup ip addresses`
`12:06.24`	`*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)`
`12:06.44`	`aceio`	`rebooting the phone dosen't help`
`12:08.25`	`aceio`	`i have also tryed to fixed the mac address to ip-addr at dhcp level`
`12:10.49`	`aceio`	`have anyone seen this problem before`
`12:12.48`	`*** join/#asterisk sourcode (~code@ppp-58-8-131-141.revip2.asianet.co.th)`
`12:13.28`	`carrar`	`ddres, just define a pool and let dhcp hand out IP's`
`12:13.43`	`carrar`	`err <acei`
`12:13.45`	`*** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca)`
`12:13.46`	`carrar`	`aceio`
`12:13.55`	`aceio`	`carrar: hi`
`12:15.11`	`aceio`	`carrar: i have allowed dhcp to hand out the ip-addr but i am still getting the same results`
`12:15.46`	`carrar`	`How big is your IP poool of addresses and lease time?`
`12:15.58`	`aceio`	`i am not sure if this a a sip firmware problem`
`12:16.40`	`carrar`	`Are you sure you phones are polling your dhcp server?`
`12:17.06`	`aceio`	`sure they are i see from the dhcp log file`
`12:17.24`	`carrar`	`How big is your IP poool of addresses and lease time?`
`12:18.09`	`aceio`	`default-lease-time 3600`
`12:18.56`	`aceio`	`pool is 10.192.30.70 - 80`
`12:19.04`	`carrar`	`so 10 IP's`
`12:19.09`	`carrar`	`how many sip phones do you have`
`12:19.15`	`aceio`	`5`
`12:19.19`	`carrar`	`and why do you think you have phones with duplicate IP's?`
`12:19.50`	`aceio`	`when i do a sip show peers`
`12:20.07`	`aceio`	`i am see two dial peers with the same ip addr`
`12:20.28`	`carrar`	`it's possible that you can have that`
`12:20.33`	`carrar`	`and perfectly ok`
`12:20.51`	`carrar`	`if you have a phone with more then 1 registration`
`12:22.11`	`aceio`	`for exmple ext 1260 and 1261 have the same ip address`
`12:22.27`	`carrar`	`So look at your config on the phone`
`12:22.37`	`aceio`	`and is i dial from 1260 to call 1261`
`12:22.37`	`carrar`	`see why it has 2 registrations if it was not suppose to have two`
`12:22.59`	`*** join/#asterisk pif (~ldm@zenon.apartia.fr)`
`12:23.14`	`aceio`	`if so have so i clear it`
`12:23.18`	`carrar`	`if you didn't intentionally do that you probably accidently put two registrations on the same phone`
`12:23.57`	`aceio`	`logically i think you are correct`
`12:23.59`	`[TK]D-Fender`	`aceio: pastebin "database show", your sip.conf and place a call from one to the other with SIP DEBUG enabled`
`12:24.24`	`[TK]D-Fender`	`aceio: Does the phone itself say that it has the same IP as another?`
`12:24.35`	`aceio`	`dam! i am not at home at this point in time to get you the info`
`12:25.43`	`[TK]D-Fender`	`aceio: Are you telling me you only looked at "sip show peers" like I saw you do last night and thats it?`
`12:25.46`	`aceio`	`i can get you the info later about 11pm uk time`
`12:26.03`	`carrar`	`I'm sure there will be someone here`
`12:26.22`	`aceio`	`cool`
`12:26.32`	`*** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com)`
`12:26.33`	`[TK]D-Fender`	`aceio: get me the rest of what I asked then. and go nmap your subnet`
`12:26.46`	`aceio`	`[TK]D-Fender: correct`
`12:26.54`	`beek`	`mornin' [TK]D-Fender`
`12:27.07`	`[TK]D-Fender`	`beek: 'lo`
`12:28.24`	`aceio`	`thank guys`
`12:28.47`	`*** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com)`
`12:29.26`	`*** join/#asterisk coppice (~chatzilla@202.64.176.93)`
`12:32.51`	`*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)`
`12:33.39`	`*** join/#asterisk Researcher (~user@unaffiliated/unafilliate)`
`12:40.18`	`[TK]D-Fender`	`aceio: ...not going to get anything to actually debug with, are we?`
`12:40.31`	`carrar`	`<aceio> i can get you the info later about 11pm uk time`
`12:40.37`	`*** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)`
`12:42.34`	`aceio`	`[TK]D-Fender: sorry i am not at home and i don't have my server switch on to acces it remotely`
`12:43.15`	`[TK]D-Fender`	`aceio: Then ask when you are in a position to do something about it.`
`12:43.56`	`aceio`	`[TK]D-Fender: np`
`12:44.09`	`[TK]D-Fender`	`carrar: I hoped that was only for the "place a call" test not being loacl to the phones in question. Wouldn't preclude getting me the rest`
`12:44.28`	`*** join/#asterisk Footman (~Footman@gwdev.creape.unilim.fr)`
`12:44.38`	`Footman`	`hello`
`12:45.12`	`carrar`	`Hi France`
`12:50.28`	`Footman`	`I have a random problem with Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c since few months and I don't find an answer : during a conversation, RTP paquets doesn't go to the phone anymore. We can see paquets on both sides for the side Asterisk/Provider but only one side for the side Asterisk/Phone (with tcpdump). Have you already seen this problem and/or do you have an solution ? I use Thomson ST2030. Could this model make this problem ? Thanks.`
`12:54.22`	`ManxPower-work`	`random problems are the hardest to solve.`
`12:55.11`	`carrar`	`They sure are!!`
`12:56.08`	`*** join/#asterisk brad_mssw (~brad@shop.monetra.com)`
`13:01.59`	`*** join/#asterisk foxfire (~foxfire@2001:690:2200:9a51:1aa9:5ff:fe2d:eae4)`
`13:04.22`	Intel``	`so my problem also random? :D`
`13:04.37`	Intel``	`about my softphone doesnt re-register`
`13:05.58`	`*** join/#asterisk zenpan (~kevin@c-174-59-96-138.hsd1.pa.comcast.net)`
`13:06.05`	`Mark22`	`Hello, I am looking for ways to let the attended transfer work. Currently I have the following situation http://yourpaste.net/5088/ I am calling with a phone within sip-internal to number 1012 (also in sip-internal) and I want to do a attended transfer to 1011`
`13:06.56`	`manxpower`	`Mark22: your phone doesn't have a transfer button?`
`13:07.09`	`Mark22`	`manxpower: no`
`13:07.14`	`manxpower`	`Also, we need to see the output of a failed call`
`13:07.34`	`manxpower`	`Mark22: Are you using a GUI like FreePBX or Trixbox?`
`13:07.59`	`foxfire`	`hello i am migrating from 1.4 to 1.6.1.18, i have changed almost all code to be compatible, but i have encountered a weird behavior with SIPAddHeader, i use it to autoanswer some click and dial calls, but now when i do a click and dial of an internal number it not only answers the first leg of the call but also on the destination. I noticed that there is a function in 1.6.2.x SIPRemoveHeader but not in 1.6.1.x is there any retro`
`13:07.59`	`foxfire`	`patch out there for 1.6.1.x ?`
`13:08.04`	`Mark22`	`nothing in the logs when it is not working, I don't use a gui (I just update changes in the configs)`
`13:09.04`	`manxpower`	`Mark22: Ah, OK. Can't help then. Parhaps someone else can.`
`13:09.28`	`manxpower`	`Without any CLI output it is much harder to diagnose.`
`13:09.49`	`manxpower`	`foxfire: yes, there is a patch, no I don't know where.`
`13:10.37`	`waa`	`what is the better way to PauseQueueMember with a user defined reason? I'm following building_queues.txt and I think if define a extension that accept *01<queue_number><text_reason> will work fine, suggestions?`
`13:11.13`	`foxfire`	`i found one for 1.4.x but not for 1.6.1.x , also took a look to backport , but 1.6.1.x looks a lot more diferent that 1.6.2.x`
`13:11.50`	`manxpower`	`Ah, OK. I assumed if it was available for 1.4, it was available for 1.6`
`13:13.08`	`foxfire`	`nope , seems to have to port it then arrggg or upgrade to 1.6.2.x`
`13:13.10`	`foxfire`	`thanx`
`13:13.35`	`Mark22`	`I am looking to see if I can produce some logs`
`13:13.58`	`zenpan`	`Good morning all`
`13:17.23`	`the_weard`	`ok im baffeled everytime the operator puts me on hold it doesent play the music but tells me your call cannot be completed as dialed pleas try again later`
`13:17.43`	`the_weard`	`how do i fix it im using g729 codec with my sip trunk`
`13:17.51`	`the_weard`	`and my phones`
`13:18.07`	`the_weard`	`everytinh works but no hold music`
`13:18.24`	`the_weard`	`also no transver music`
`13:19.05`	`the_weard`	`so how would i go about fixing it googled al day`
`13:20.28`	`manxpower`	`the_weard: how many G729 licenses do you have?`
`13:20.44`	`*** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com)`
`13:21.32`	`the_weard`	`1`
`13:21.32`	`the_weard`	`licence`
`13:21.32`	`manxpower`	`you should put a copy of the CLI output of a failed call on pastebin.com. Maybe someone can help you.`
`13:21.32`	`the_weard`	`but the call comes in and then gets put on hold`
`13:21.48`	`manxpower`	`the_weard: and then it fails. pastebin please`
`13:21.54`	`the_weard`	`and get the stupid lady telling me that`
`13:22.16`	`the_weard`	`ok where do i get the cli output?`
`13:22.24`	`manxpower`	`the_weard: in the Asterisk CLI.`
`13:22.27`	`the_weard`	`newb at my bussiness`
`13:22.31`	`manxpower`	`Maybe you should read The Book`
`13:22.34`	`manxpower`	`~book`
`13:22.35`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`13:22.39`	`manxpower`	`We are not here to teach you Asterisk.`
`13:22.50`	`carrar`	`heh`
`13:23.34`	`the_weard`	`ok not that a newb i alredy configuerd everything text base but the damm codecs`
`13:23.41`	`the_weard`	`stuffing me around`
`13:24.28`	`manxpower`	`the_weard: You must be using a GUI for Asterisk`
`13:24.40`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`13:25.20`	`the_weard`	`jip for the other pepole that doesnot know what is going on`
`13:25.35`	`manxpower`	`jip?`
`13:26.33`	`the_weard`	`means jes`
`13:26.47`	`the_weard`	`sorry im afrikaans a bit difficalt`
`13:27.01`	`manxpower`	`Wish you the best of luck, but GUIs are not supported here.`
`13:27.10`	`manxpower`	`~freePBX`
`13:27.11`	`infobot`	`[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx`
`13:27.26`	`manxpower`	`anyway, I wish you the best of luck.`
`13:27.58`	`the_weard`	`http://pastebin.com/vqaRKaz1`
`13:28.03`	`zenpan`	`Which is the proper channel to ask questions about hardware and Asterisk?`
`13:28.16`	`manxpower`	`zenpan: here is a good place for that.`
`13:28.44`	`manxpower`	`zenpan: No, you cannot use a "voice modem" with Asterisk.`
`13:29.02`	`hurdman`	`:)`
`13:33.19`	`*** join/#asterisk bdfoster (~5710e77c@gateway/web/freenode/x-kazabzczieqcdwwb)`
`13:33.21`	`bdfoster`	`heyoh`
`13:33.25`	`bdfoster`	`anyone around?`
`13:35.51`	`elred_`	`just ask..`
`13:37.32`	`*** join/#asterisk jasonwoot (~some@69.73.89.233)`
`13:37.50`	`*** join/#asterisk muiro (~muiro@unaffiliated/muiro)`
`13:38.29`	`[TK]D-Fender`	`bdfoster: No.`
`13:38.30`	`bdfoster`	`is it better to leave NAT on or off on the router if I want to have a few users in the company connect to the asterisk server?`
`13:38.49`	`bdfoster`	`?`
`13:39.09`	`ManxPower-work`	`bdfoster, your question makes no sense.`
`13:39.36`	`ManxPower-work`	`Usually you either need NAT or you don't. Doesn't have anything to do with Asterisk itself.`
`13:40.02`	`carrar`	`Leave NAT off!`
`13:40.16`	`Chainsaw`	`network goes dark world ends`
`13:40.30`	`carrar`	`unless you are on IPv6 VCC`
`13:40.31`	`bdfoster`	`yea, kinda lost on that subject, just saw a way to turn nat off on my router, didnt know if that would help anything as far as SIP`
`13:40.44`	`bdfoster`	`only on IPv4`
`13:40.48`	`ManxPower-work`	`bdfoster, do you understand what NAT is?`
`13:41.09`	`bdfoster`	`router rewrites packets basically`
`13:41.29`	`ManxPower-work`	`bdfoster, Do you need that?`
`13:41.37`	`bdfoster`	`hell if i know`
`13:41.51`	`carrar`	`Then you shouldn't be touching the router`
`13:41.58`	`ManxPower-work`	`bdfoster, That makes no sense. You either need it or you don't. This has nothing to do with Asterisk`
`13:42.17`	`carrar`	`Please remove your access to the router`
`13:42.22`	`[TK]D-Fender`	`[09:41]<carrar>Then you shouldn't be touching the router <- or the IntarWebs`
`13:42.35`	`carrar`	`heh`
`13:42.41`	`bdfoster`	`carrar: my apologies, i asked another dumbass question`
`13:43.03`	`carrar`	`it's ok`
`13:43.07`	`ManxPower-work`	`try to make it less dumbass this time. 8-)`
`13:43.41`	`bdfoster`	`well, just trying to broaden my understanding of networking in general`
`13:44.07`	`bdfoster`	`im a hardware guy`
`13:44.08`	`ManxPower-work`	`You might want to try asking on a networking channel.`
`13:45.03`	`ManxPower-work`	`without a good understanding of Networking (UDP, NAT, routing, etc), Linux, VoIP and SIP, and Asterisk.`
`13:45.13`	`ManxPower-work`	`you will have major issues trying to do anything with Asterisk.`
`13:47.31`	`*** join/#asterisk UQlev (~yuriy@212.50.99.8)`
`13:49.46`	`*** join/#asterisk bdfoster (~5710e77c@gateway/web/freenode/x-xbfjermfjvcmnuoi)`
`13:49.55`	`bdfoster`	`sry, firefox crashed`
`13:49.58`	`carrar`	`bdfoster, pick up a copy of TCP/IP Illustrated, Volume 1`
`13:50.01`	`*** join/#asterisk geneticx_wrk (~jorge@host-208-88-126-198.biznesshosting.net)`
`13:50.20`	`bdfoster`	`yea, i think im gonna need to hang that book around my neck`
`13:50.33`	`carrar`	`it works better if you read it`
`13:50.59`	`bdfoster`	`carrar: so i can reference it anytime before i ask another braindead question`
`13:51.39`	`carrar`	`it's on googlebooks also`
`13:52.06`	`bdfoster`	`ill have to print it out`
`13:53.31`	`carrar`	`it actually doesn't cover NAT, but you can read about that elsewhere`
`13:53.44`	`bdfoster`	`ping`
`13:54.00`	`bdfoster`	`nvm it kicked back on`
`13:54.44`	`*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)`
`13:54.53`	`*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)`
`14:01.31`	`*** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca)`
`14:05.53`	`*** join/#asterisk moy (~moy@74.12.128.42)`
`14:06.55`	`*** join/#asterisk Fs0L (~Fs0L@136.223.19.35)`
`14:06.56`	`*** part/#asterisk manxpower (~ewieling@216.186.151.147)`
`14:09.24`	`*** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net)`
`14:13.16`	`*** join/#asterisk wcselby (~wcselby@216.110.88.194)`
`14:13.19`	`wcselby`	`o/`
`14:14.56`	`*** join/#asterisk rgsteele (~rgsteele@207.106.239.81)`
`14:17.20`	`*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)`
`14:20.35`	`*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)`
`14:24.16`	`*** join/#asterisk boch (~fran@200.61.191.9)`
`14:24.24`	`ariel_`	`Morning everyone`
`14:24.53`	`boch`	`Hi all, how can i know if a sip hardphone supports MESSAGE method? is this method part of RFC3261?`
`14:31.35`	`*** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net)`
`14:36.20`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`14:39.03`	`zenpan`	`I understand that I can use an ATA instead of an FXS analog card in my Asterisk (trixbox) system. I've looked around and found one that looks interesting: Grandstream Handytone 286/486. Anyone use these? Do they work well with Asterisk?`
`14:39.35`	`*** join/#asterisk [1]sassyn (~sassyn@bzq-82-80-242-222.cablep.bezeqint.net)`
`14:39.39`	`[1]sassyn`	`hi all`
`14:39.48`	`[1]sassyn`	`can someone please help with an issue i'm having`
`14:39.49`	`WIMPy`	`~gs`
`14:39.50`	`infobot`	`GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.`
`14:40.01`	`Chainsaw`	`zenpan: For FXS I use a Patton SmartNode 4118.`
`14:40.07`	`ariel_`	`they work with Asterisk yes, but just so you know Trixbox is not asterisk it's a gui that changes allot on the system and it does not react normally`
`14:40.31`	`[TK]D-Fender`	`zenpan: ...`
`14:40.33`	`[TK]D-Fender`	`~gs`
`14:40.34`	`infobot`	`GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.`
`14:40.42`	`[1]sassyn`	`it go like this: i have install asterisk 1.4.29 in my local network with freepbx - and everything seems to work fine. i even build the debian pkg so if some need them please send me an email`
`14:41.04`	`[1]sassyn`	`now, the server is in the local network - 172.17.0.20 for example`
`14:41.16`	`[1]sassyn`	`and i have configure using a sip trunk`
`14:41.43`	`rdircio`	`i got "File all-circuits-busy-now does not exist in any format"`
`14:41.52`	`rdircio`	`now i dlded all sounds`
`14:41.56`	`rdircio`	`where do i place them?`
`14:42.03`	`[1]sassyn`	`i configure my firewall to do nat - so when system is getting to the internet it's ip address on the remote host is the external one`
`14:42.19`	`[1]sassyn`	`i can call to external network`
`14:42.25`	`[1]sassyn`	`i can even get internat calls`
`14:42.31`	`[1]sassyn`	`but i can't hear a thing`
`14:42.38`	`[1]sassyn`	`it is working internal`
`14:42.42`	`[1]sassyn`	`but not external`
`14:42.45`	`[1]sassyn`	`why?`
`14:43.12`	`Fs0L`	`I have an HT-503 that I use`
`14:43.14`	`[1]sassyn`	`the machine in the firewall can access outside to the internet in all ports`
`14:43.19`	`Fs0L`	`and yeah, in general they are a pain`
`14:43.24`	`Fs0L`	`but I got mine to work`
`14:43.49`	`[TK]D-Fender`	`[1]sassyn: What have you forwarded to your server?`
`14:44.21`	`[1]sassyn`	`from the external i did a virtual ip - of one of my public ip address that goes to the dmz`
`14:44.46`	`[1]sassyn`	`and i have made a rule like this 212.212.212.x --- > 172.17.0.20 - all ports are open`
`14:44.59`	`[1]sassyn`	`[TK]D-Fender, thank for the help ;-)`
`14:45.15`	`[TK]D-Fender`	`[1]sassyn: pastebin a COMPLETE call attempt with SIP DEBUG enabeld.`
`14:45.17`	`[TK]D-Fender`	`~pb`
`14:45.18`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`14:45.25`	`[1]sassyn`	`a sec`
`14:45.36`	`*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)`
`14:45.45`	`*** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek)`
`14:47.40`	`Fs0L`	`xm list`
`14:47.43`	`Fs0L`	`oops...sry`
`14:47.48`	`Fs0L`	`wrong window`
`14:47.49`	`*** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com)`
`14:48.01`	`wcselby`	`[1]sassyn - what type of firewall?`
`14:48.38`	`hluesea`	`how can i insert the realtime coming calls to mysql ?`
`14:49.04`	`[1]sassyn`	`[TK]D-Fender, http://pastebin.com/TWshUbGg`
`14:49.06`	`[TK]D-Fender`	`hluesea: Wording is broken. Try again.`
`14:49.58`	`*** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca)`
`14:50.05`	`*** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it)`
`14:50.23`	`[1]sassyn`	`wcselby, Fortigate`
`14:50.34`	`[TK]D-Fender`	`[1]sassyn: -- Executing [s@macro-dial:7] Dial("SIP/012-SIP-Out-00000034", "SIP/6510\|15\|tr") in new stack <-- didn't even really try. do "sip show peer 6510"`
`14:50.48`	`wcselby`	`[1]sassyn - ahhh`
`14:51.00`	`wcselby`	`it's doing SIP mangling`
`14:51.14`	`wcselby`	`you have to disable it in the cli of the firewall`
`14:51.23`	`[1]sassyn`	`[TK]D-Fender, it is ture the peer 6510 is down at the moment`
`14:51.29`	`[1]sassyn`	`but even when it is up`
`14:51.33`	`[1]sassyn`	`i can't hear a thing`
`14:51.35`	`wcselby`	`i had a client with a firewall from them, it was the source of all kinds of trouble until we disabled the SIP crap`
`14:51.38`	`rdircio`	`make sounds install..........`
`14:51.41`	`[TK]D-Fender`	`[1]sassyn: Show me a REAL CALL.`
`14:51.48`	`[TK]D-Fender`	`[1]sassyn: What failed in this?`
`14:52.08`	`[1]sassyn`	`[1]sassyn, Even in this call I shoudl hear the VoiceMail`
`14:52.15`	`[1]sassyn`	`and nothing`
`14:52.55`	`[1]sassyn`	`wcselby, so u saying it is a FW problem - i starting to think it is the problem. since On a pure network with just simple FW i manage to do so with a problem`
`14:53.09`	`[TK]D-Fender`	`[1]sassyn: <--- Transmitting (NAT) to 212.199.157.154:5060 ---> <--- They are not behind NAT. your entry for them should say "nat=no"`
`14:53.10`	`[1]sassyn`	`[TK]D-Fender, a real call? what do u mean?`
`14:53.29`	`[1]sassyn`	`hmm`
`14:53.38`	`[1]sassyn`	`but my server is behind nat`
`14:54.11`	`[TK]D-Fender`	`[1]sassyn: ....`
`14:54.23`	`[TK]D-Fender`	`[1]sassyn: *THEY* are not behind NAT`
`14:54.55`	`[1]sassyn`	`[TK]D-Fender, let me understand`
`14:54.59`	`hluesea`	`[TK]D-Fender : how can i insert realtime calls to mysql ? (actually i want to insert "sip show channels" results to mysql or like monitoring things for realtime calls )`
`14:55.00`	`wcselby`	`[1]sassyn - just be sure to disable the sip session helper and the sip alg on the firewall, that's a major issue with the fortigate's I've dealt with`
`14:55.42`	`[1]sassyn`	`u saying that nat=no should be if my SIP provide is not behiend NAT?`
`14:55.44`	`[TK]D-Fender`	`hluesea: Then go write a program to poll for that and do your SQL inserts`
`14:56.04`	`[TK]D-Fender`	`[1]sassyn: For their entry, not your [general]`
`14:56.53`	`[1]sassyn`	`[TK]D-Fender, can u be more specific`
`14:56.55`	`[1]sassyn`	`I lost u`
`14:57.05`	`[1]sassyn`	`wcselby, where to disbale this?`
`14:57.37`	`[TK]D-Fender`	`[1]sassyn: what part of "they are not behind NAT" are you ahving trouble with?`
`14:57.39`	`wcselby`	`[1]sassyn - this is the method my last client used, found it on a trixbox forum, it seemed to work. --> http://pastebin.com/5HJ23Tb4`
`14:57.59`	`wcselby`	`[1]sassyn -also, you should read up on sip / nat stuff related to asterisk`
`14:58.02`	`wcselby`	`~sipnat`
`14:58.02`	`infobot`	`[~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`
`14:58.43`	`[1]sassyn`	`wow cool info`
`14:59.33`	`[1]sassyn`	`[TK]D-Fender, as i said my server is in my local network - 172.16.0.17 - this is rhe asterisk server. in the firewall it access to the internal with all port open and with nat address of 212.212.212.X`
`14:59.59`	`[1]sassyn`	`coming network traffic is router to 212.212.212.X -- > 172.16.0.17 - all port are open`
`15:00.09`	`[TK]D-Fender`	`[1]sassyn: That's nice... but completely not what I just told you.`
`15:00.11`	`[1]sassyn`	`so i don't get what do u mean in saying nat=no ?`
`15:00.45`	`wcselby`	`[1]sassyn - so you're asterisk server is behind NAT, since it uses a private IP address, but the world sees a public. the other entity you are trying to call is using a public IP address, they have no nat, so you need to configure them in your sip.conf file with "nat=no"`
`15:01.00`	`*** part/#asterisk Fs0L (~Fs0L@136.223.19.35)`
`15:01.01`	`[1]sassyn`	`OK`
`15:01.05`	`[1]sassyn`	`now I got it`
`15:01.16`	`[1]sassyn`	`checking - thank thank thank`
`15:01.17`	`[TK]D-Fender`	`...`
`15:01.23`	`wcselby`	`read the first link (www.aocomputing.net/?p=3) for a better understanding`
`15:01.25`	`[1]sassyn`	`wcselby, u too my friend :-)`
`15:01.35`	`[TK]D-Fender`	`[1]sassyn: Good... just saved a heavy dose of ClueBat (tm)`
`15:03.50`	`hardwire`	`want thank.`
`15:05.24`	`[1]sassyn`	`http://pastebin.com/YbKcQbwN`
`15:08.38`	`[TK]D-Fender`	`[1]sassyn: make sure your router is not performing any SIP transform as suggested, and ensure that your [general] has "canreinvite=no"`
`15:09.37`	`DarthPointer`	`How does asterisk specify the queue order when using strategy=roundrobin? A show queue XXXX shows a different order than what is specified in queues.conf`
`15:09.46`	`ariel_`	`I have a very strange issue. 2 boxes they setup with iax2 peers, Box A sends the call box B. But box B does nothing with it. It displays nothing does not answer or even reject it normally just sits there. If I put debug on for iax2 here all I get: http://pastebin.ca/1857832`
`15:10.03`	`[TK]D-Fender`	`[1]sassyn: Retransmitting #1 (NAT) to 172.17.0.52:5060: <--- and you seem to think your local devices ARE behind NAT. You don't seem to have grasped this very well. Go read the guides again`
`15:10.35`	`[1]sassyn`	`OK`
`15:12.42`	`[TK]D-Fender`	`DarthPointer: Show us`
`15:12.50`	`hardwire`	`ariel_: I'm checking my mail before heading in to my office :)`
`15:12.53`	`hardwire`	`err`
`15:15.35`	`DarthPointer`	`http://pastebin.com/EFwfK0Na`
`15:15.41`	`DarthPointer`	`Asterisk 1.4.30`
`15:15.48`	`*** join/#asterisk tunk (~tunk@unaffiliated/tunk)`
`15:17.13`	`DarthPointer`	`Previously I was using RingGroup w/ a RingStrategy set hunt, however we came across a deadlock that was crashing the system (Leif found that); so I am now putting members directly in the queue`
`15:17.28`	`DarthPointer`	`where the RingGroup was the only member of the queue`
`15:18.21`	`hardwire`	`DarthPointer: I'm not sure how it sorts the list, try using a member name (described in the member => format)`
`15:18.46`	`hardwire`	`You want the first member to be called each time the queue is called, right?`
`15:19.19`	`DarthPointer`	`yes that is correct`
`15:19.41`	`hardwire`	`well you might want to try something`
`15:20.10`	`hardwire`	`change the member priority, add a member name, and add a device state extension vs using Local`
`15:20.36`	`hardwire`	`well.. either don't use local or define both Local and a TECH/Exten to monitor.`
`15:21.49`	`hardwire`	`looks like roundrobin was depreciated in 1.4 in favor of roundrobin+memory`
`15:21.55`	`hardwire`	`so you may not get what you want anyways.`
`15:22.22`	`rdircio`	`now i restarted and dahdis how channels shows no channel :(`
`15:22.47`	`wcselby`	`roundrobin was deprecated in 1.6 I thought`
`15:22.58`	`rdircio`	`when i try make a call i get "f type 'DAHDI' (cause 66 - Channel not implemented)"`
`15:23.00`	`wcselby`	`no, i'm getting my memory mixed up`
`15:23.00`	`rdircio`	`:(`
`15:23.03`	`DarthPointer`	`the member priority; do you mean the penalty?`
`15:23.04`	`*** join/#asterisk Fubard (~brawr@173.10.235.205)`
`15:23.10`	`rdircio`	`:(`
`15:23.36`	`wcselby`	`rdircio - did your dahdi startup?`
`15:23.39`	`DarthPointer`	`my reading of it means that it won't automatically ring the next caller http://www.voip-info.org/wiki/view/Asterisk+call+queues`
`15:24.05`	`[TK]D-Fender`	`DarthPointer: What does your examination of an actual call have to say?`
`15:24.10`	`[TK]D-Fender`	`wcselby: clearly not`
`15:24.13`	`rdircio`	`/etc/init.d/dahdi start was ok`
`15:24.22`	`rdircio`	`not sure what else to check`
`15:24.25`	`*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.79)`
`15:24.46`	`[TK]D-Fender`	`rdircio: Didi you compile * AFTER building and isntalling DAHDI?`
`15:25.12`	`rdircio`	`yes and it used to work jsut before i restarted the server :(`
`15:25.30`	`rdircio`	`modules wcfxo wctc4xxp dahdi, dahdi_transcode are loaded`
`15:25.51`	`[TK]D-Fender`	`rdircio: stop . do "dahdi_cfg -vvvv" restart . Test`
`15:25.53`	`*** join/#asterisk [1]sassyn (~sassyn@bzq-82-80-242-222.cablep.bezeqint.net)`
`15:26.00`	`Naikrovek`	`grumbles.`
`15:26.06`	`[1]sassyn`	`[TK]D-Fender, sorry i lost connection`
`15:26.24`	`[1]sassyn`	`wcselby, i have disbale the setting in the fortigate client`
`15:26.31`	`[1]sassyn`	`it still not workin`
`15:26.45`	`zenpan`	`OK, if the Grandstreams aren't a good choice, does anyone have a recommendation?`
`15:26.57`	`[TK]D-Fender`	`zenpan: Linksys PAP2T-NA`
`15:27.22`	`DarthPointer`	`It seems to be working with as a round robin w/ memory`
`15:27.29`	`wcselby`	`[1]sassyn - did you read the nat settings page and follow those instructions as well?`
`15:28.28`	`[1]sassyn`	`yes i did`
`15:28.37`	`[1]sassyn`	`i just don't get it why?`
`15:28.38`	`[TK]D-Fender`	`[1]sassyn: Not from the last PB you gave us`
`15:29.01`	`[TK]D-Fender`	`[1]sassyn: perhaps you should PB your entire sip.conf and what's linked to it masking only passwords.`
`15:29.07`	`*** part/#asterisk tunk (~tunk@unaffiliated/tunk)`
`15:29.34`	`[1]sassyn`	`ok`
`15:29.40`	`[1]sassyn`	`thanks`
`15:31.13`	`[1]sassyn`	`http://pastebin.com/Az32KY1u`
`15:32.26`	`[TK]D-Fender`	`....`
`15:32.38`	`[TK]D-Fender`	`senses that reading is not this one's strong suit`
`15:33.36`	`DarthPointer`	`zenpan I have ~100 Aastra 57i's and am very happy with them`
`15:34.08`	`[1]sassyn`	`again the only prblem i have is that i can't hear nothing when calling from external to internal or internal to exteral`
`15:34.17`	`*** join/#asterisk slashtom (~tom@k-rad.co.uk)`
`15:34.17`	`[1]sassyn`	`interal to interal works fine!`
`15:34.40`	`[TK]D-Fender`	`goes off to do something productive.`
`15:35.28`	`[1]sassyn`	`wcselby, any more ideas?`
`15:39.55`	`*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)`
`15:42.01`	`Skeeter-`	`[TK]D-Fender, my polycoms is kinda bricked. i can boot it but on the Main page, i cant press any button`
`15:42.25`	`Skeeter-`	`i tried the *468 during the boot to flash to factory the phone , but it doesnt work, do you have any work around?`
`15:43.28`	`Chainsaw`	`Skeeter-: Use TFTP instead of any other protocol.`
`15:43.39`	`Chainsaw`	`Skeeter-: That has the highest chance of succeeding.`
`15:43.48`	`Skeeter-`	`Chainsaw, whats the point?`
`15:43.50`	`Chainsaw`	`Skeeter-: (Feel free to revert to FTP/HTTP later, but not now)`
`15:44.05`	`Chainsaw`	`Skeeter-: The point is that TFTP is easiest to handle for the bootrom. It's the least amount of work.`
`15:44.11`	`*** join/#asterisk theBruno (~theBruno@128-177-27-78.ip.openhosting.com)`
`15:44.13`	`p3nguin`	`Most Polycom guys freak out at the mentin of tftp.`
`15:44.27`	`Naikrovek`	`it's not ideal, tftp. ftp > tftp`
`15:44.35`	`p3nguin`	`see?`
`15:44.42`	`Skeeter-`	`Chainsaw, i takes its configs, BUT someone probably went into the web gui and changed sth`
`15:45.00`	`Skeeter-`	`it takes*`
`15:45.05`	`Chainsaw`	`Naikrovek: With a fully functional bootrom and application suite, yes.`
`15:45.26`	`Naikrovek`	`Skeeter-: change the admin password on the phones`
`15:45.27`	`Skeeter-`	`Chainsaw, its not even related, its a protocol`
`15:45.46`	`Skeeter-`	`Naikrovek, what do i do with the bricked phone?`
`15:45.48`	`Naikrovek`	`yes, but tftp isn't called trivial ftp for nothing. it is easier to implement`
`15:45.56`	`Naikrovek`	`and easier on devices`
`15:46.02`	`Naikrovek`	`is the phone REALLY bricked?`
`15:46.04`	`[TK]D-Fender`	`Skeeter-: Re-provision it from a new account. change the SIP app version in so doing. Then rebuild configs for the desired version and revert.`
`15:46.06`	`Chainsaw`	`Skeeter-: You know what. Call Polycom support and go second-guess them.`
`15:46.16`	`[TK]D-Fender`	`It isn't bricked if SIP can boot.`
`15:46.34`	`p3nguin`	`You aren't likely to even talk to Polycom support about it.`
`15:46.36`	`[TK]D-Fender`	`That means it passes the BootROM and can be reflashed`
`15:46.51`	`Skeeter-`	`new sip app = reflash?`
`15:47.09`	`[TK]D-Fender`	`Skeeter-: Yes`
`15:47.26`	`Naikrovek`	`if it can boot into the sip (fail or not) then the phone isn't bricked. yes, upgrading sip = reflash`
`15:48.12`	`Skeeter-`	`Naikrovek, [TK]D-Fender downloading and trying it right away`
`15:48.44`	`*** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru)`
`15:49.26`	`*** join/#asterisk Knightfal (~JK@mailer.1callres.com)`
`15:50.44`	`Knightfal`	`Morning guys!`
`15:51.04`	`Knightfal`	`Had a couple of questions.`
`15:51.38`	`Naikrovek`	`ask them`
`15:51.49`	`Naikrovek`	`(we're not much for pleasantries; no offense)`
`15:51.50`	`wcselby`	`he HAD questions`
`15:51.52`	`wcselby`	`past tense`
`15:51.58`	`Naikrovek`	`is corrected.`
`15:52.03`	`wcselby`	`lol :)`
`15:52.04`	`Naikrovek`	`glad you got that worked out, Knightfal :)`
`15:53.02`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`15:53.06`	`p3nguin`	`hahaha`
`15:53.17`	`p3nguin`	`Another happy customer?`
`15:53.23`	`Knightfal`	`Im working with Attended xfers in 1.4.30 and was wondering about performing an action after the original call. So A calls B, B then Warm xfers to C , B Then hangs up. Currently the dial plan will not finish executing untill A and C are finished with the call`
`15:53.35`	`zenpan`	`thank you D-Fender and DarthPointer`
`15:54.39`	`Naikrovek`	`Knightfal: i see this question pretty often, and i've never noticed an answer to it. I BELIEVE (i'm often wrong) that this is just how it works.`
`15:54.50`	`*** join/#asterisk Faithful (~Faithful@202.6.145.116)`
`15:55.29`	`Knightfal`	`Have you seen Set(GOTO_ON_BLINDXFER=todo-xfer^s^1)`
`15:55.47`	`*** join/#asterisk jmacz (~jmacz@190.144.75.22)`
`15:55.58`	`Knightfal`	`on a blind xfer the original callee is sent to the todo-xfer`
`15:56.00`	`*** join/#asterisk AsteriskNoob (~AsteriskN@host86-128-145-94.range86-128.btcentralplus.com)`
`15:56.22`	`AsteriskNoob`	`Hello all. Any Cisco 7940 or 7970 experts in here today?`
`15:56.28`	`Knightfal`	`looking for the same type of functionality for attended xfer`
`15:56.43`	`p3nguin`	`asterisknoob: Just ask the question.`
`15:57.00`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`15:57.27`	`AsteriskNoob`	`I have a 7971 which I tried to configure today and all I can get is inbound audio, no outgoing audio. No codec selected and I can't find where to fix it.`
`15:58.06`	`AsteriskNoob`	`But now my 7940s have developed the same problem, even though I have not upgraded/changed their loads.bin files.`
`15:58.14`	`Naikrovek`	`NAT?`
`15:58.23`	`p3nguin`	`Which firmware are you running on the 7940s?`
`15:58.47`	`p3nguin`	`Cisco phones are not NAT-friendly, from what I hear.`
`15:59.24`	`AsteriskNoob`	`8.12.00`
`15:59.46`	`AsteriskNoob`	`All was fine until I tried to add the 7971.`
`16:00.41`	`p3nguin`	`There's no chance that one of your files for the 7971 is being read by the 7940s?`
`16:01.10`	`AsteriskNoob`	`I thought so, but I have now removed the 7971 files and the problem persists..`
`16:01.34`	`AsteriskNoob`	`I'm ready to throw every Cisco phone in the river!`
`16:01.43`	`p3nguin`	`I'll pay shipping to my place.`
`16:01.47`	`AsteriskNoob`	`hehehe`
`16:03.31`	`rdircio`	`bleh`
`16:03.42`	`rdircio`	`all dahdi commands in the CLI fail`
`16:03.43`	`rdircio`	`:(`
`16:03.46`	`rdircio`	`dahdi restart`
`16:03.50`	`rdircio`	`or dahdi show channels`
`16:03.52`	`rdircio`	`:(`
`16:07.55`	`rdircio`	`why?`
`16:08.54`	`*** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org)`
`16:08.57`	`*** join/#asterisk rEv9 (~rEv9@koppenaal.nl)`
`16:10.03`	`rEv9`	`is it possible to find out what meetme conference number I'm in from the AGI script that is running in the background of the meetme conference ? is it stored in a variable or something? I cant find out that conference the agi script is running in..`
`16:12.36`	`*** join/#asterisk rubbs (~rubbs@cpe-71-72-56-140.neo.res.rr.com)`
`16:12.46`	`Footman`	`bye bye`
`16:13.21`	`tzafrir_laptop`	`rdircio, "fail" == ?`
`16:13.22`	`[1]sassyn`	`[TK]D-Fender, it is working`
`16:13.30`	`tzafrir_laptop`	`"no such command"?`
`16:13.39`	`[1]sassyn`	`wcselby, it is working thnaks`
`16:13.45`	`*** join/#asterisk AndyGraybeal (~AndyGray@128-177-27-78.ip.openhosting.com)`
`16:14.09`	`AndyGraybeal`	`is ther a channel to talk about voip hardware ? .. or is this it even though it might be a little offtopic?`
`16:14.19`	`rdircio`	`yea`
`16:14.39`	`rdircio`	`kraftek*CLI> dahdi show channels`
`16:14.39`	`rdircio`	`No such command 'dahdi show channels' (type 'core show help dahdi show' for other possible commands)`
`16:14.47`	`rdircio`	`sorry for paste, only 2 lines`
`16:14.57`	`Chainsaw`	`rdircio: So the dahdi channel driver is not loaded.`
`16:15.03`	`rdircio`	`how do i?`
`16:15.34`	`Chainsaw`	`rdircio: Normally modules auto-load as required, so it's possible that you built Asterisk without DAHDI support.`
`16:15.44`	`rdircio`	`that is odd`
`16:15.57`	`rdircio`	`lemme try rebuild it :(`
`16:16.07`	`AsteriskNoob`	`Miracle of miracles... Changing NAT options in sip.conf and the problem is fixed...`
`16:16.13`	`AsteriskNoob`	`Why??? I give up..`
`16:16.18`	`Chainsaw`	`rdircio: If you see anything about a DAHDI channel driver... you want that.`
`16:16.31`	`rdircio`	`is there a flag in "configure"?`
`16:16.44`	`rdircio`	`cos i do have dahdi started and kernel modules loaded`
`16:16.51`	`Chainsaw`	`rdircio: Worth a go with ./configure --help`
`16:16.57`	`Chainsaw`	`rdircio: If it does, it'll be listed there.`
`16:17.06`	`rdircio`	`dahdi_cfg -vvv shows my device`
`16:18.19`	`rdircio`	`only --with-dahdi=PATH`
`16:18.21`	`rdircio`	`hmm`
`16:19.53`	`rdircio`	`did a make uninstall-all jsut to make sure :)`
`16:22.31`	`*** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net)`
`16:22.56`	`rdircio`	`configure passes thru "checking for enhanced dahdi vmwi support... yes"`
`16:23.00`	`rdircio`	`so it does find dahdi`
`16:23.19`	`Chainsaw`	`Alright. Go through make & make install again, just to be sure.`
`16:23.31`	`Chainsaw`	`If you still can't use dahdi commands after that, see if you have to load the module by hand.`
`16:23.33`	`*** join/#asterisk pigo (~7c7cf78c@gateway/web/freenode/x-sljytdgejagnpwlc)`
`16:23.35`	`Chainsaw`	`(chan_dahdi.so)`
`16:23.44`	`pigo`	`Hi everybody`
`16:24.09`	`pigo`	`Anybody have any idea about this error ---------->channel.c:2976 __ast_read: Dropping incompatible voice frame on SIP/1000-00000002 of format ulaw since our native format has changed to 0x2 (gsm)`
`16:24.16`	`rdircio`	`<PROTECTED>`
`16:24.16`	`rdircio`	`<PROTECTED>`
`16:24.21`	`rdircio`	`it's building it....`
`16:24.48`	`Chainsaw`	`pigo: That suggests you're allowing more codecs than you actually support.`
`16:24.57`	`p3nguin`	`pigo: What's the question about it? It says what the problem is.`
`16:25.02`	`Chainsaw`	`pigo: So carefully review your allow= entries for that peer. And make sure you have a disallow=all in there.`
`16:25.32`	`pigo`	`thanks chainsaw, i try that one`
`16:28.42`	`*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)`
`16:32.03`	`*** join/#asterisk bdfoster (~5710ed83@gateway/web/freenode/x-nrosmsrspyznxnjz)`
`16:32.33`	`bdfoster`	`im trying to figure out the best way to do a few things with the asterisk server`
`16:32.44`	`bdfoster`	`one of them is vpn`
`16:33.01`	`bdfoster`	`is it ok to install openvpn on an asterisknow install?`
`16:33.25`	`Qwell`	`bdfoster: sure, it's just Linux`
`16:33.33`	`[TK]D-Fender`	`bdfoster: No. You'll break the space-time continuum`
`16:33.45`	`bdfoster`	`no complications with httpd or anything?`
`16:34.01`	`[TK]D-Fender`	`bdfoster: And they have what to do with each otehr?`
`16:34.21`	`bdfoster`	`point taken. what about installing webmin?`
`16:34.33`	`[TK]D-Fender`	`bdfoster: Can I have an FTP server on my web server as well? I mean... Firefox can access HTTP & FTP... won't it get confused?`
`16:35.01`	`bdfoster`	`problem is that i already have a webserver on it, freepbx`
`16:35.03`	`[TK]D-Fender`	`bdfoster: And what does webmin have to do with anything....?`
`16:35.17`	`bdfoster`	`connects on the same port as freepbx`
`16:35.29`	`bdfoster`	`i know openvpn doesnt, but webmin does`
`16:35.33`	`[TK]D-Fender`	`bdfoster: Webmin has its own embedded server, doesn't use Apache OR sit on the standard HTTP port 80`
`16:35.50`	`bdfoster`	`what port does it sit on?`
`16:35.52`	`[TK]D-Fender`	`bdfoster: Webmin defaults to TCP 10000`
`16:36.03`	`[TK]D-Fender`	`facepalms`
`16:36.04`	`bdfoster`	`ah, ok`
`16:36.20`	`bdfoster`	`i was reverting to m prior experience with piaf`
`16:36.32`	`bdfoster`	`which was on port 80`
`16:36.44`	`[TK]D-Fender`	`bdfoster: What was on 80?`
`16:36.48`	`bdfoster`	`webmin`
`16:36.51`	`bdfoster`	`im pretty sure it was`
`16:36.57`	`*** join/#asterisk NMR_1122 (~rahl@adsl-068-209-105-089.sip.mia.bellsouth.net)`
`16:37.02`	`[TK]D-Fender`	`bdfoster: No....`
`16:37.07`	`bdfoster`	`on piaf?`
`16:37.27`	`[TK]D-Fender`	`Should never be`
`16:37.47`	`rdircio`	`good`
`16:38.00`	`rdircio`	`dahdi show channels now shows pseudo`
`16:38.08`	`*** join/#asterisk the_weard (~arthur@196-210-36-183.dynamic.isadsl.co.za)`
`16:38.09`	`rdircio`	`can't remember how to generate channel 1`
`16:38.13`	`rdircio`	`will find out now`
`16:38.31`	`*** join/#asterisk paulc (~paulc@unaffiliated/paulc)`
`16:41.11`	`NMR_1122`	`Hi, does anybody know a way to set a variable to a predetermined value at a certain time of day? Not using gotoIfTime(), but more like a cron job that (re)sets the variable once each day ?`
`16:41.44`	`[TK]D-Fender`	`NMR_1122: Variables are generally local only to a given channel and gone when it ends. use AstDB`
`16:42.27`	`bdfoster`	`best way to do ddns? does ddclient work with asterisknow?`
`16:42.47`	`p3nguin`	`What do you want to use ddns for?`
`16:43.31`	`bdfoster`	`track the ip address of the asterisk server`
`16:43.41`	`bdfoster`	`so external users can connect`
`16:43.42`	`p3nguin`	`For what purpose?`
`16:43.44`	`p3nguin`	`oh`
`16:43.46`	`p3nguin`	`okay, then.`
`16:44.32`	`[TK]D-Fender`	`bdfoster: AsteriskNOW = CentOS = RHEL = Linux.`
`16:44.35`	`p3nguin`	`There are multiple ways to handle dynamic DNS, but it usually depends on what service you use as to which one you should implement.`
`16:44.53`	`[TK]D-Fender`	`bdfoster: Any LINUX client will probably work.`
`16:45.06`	`bdfoster`	`[TK]D-Fender: had a hard time with ddclient`
`16:45.11`	`bdfoster`	`on piaf`
`16:45.27`	`p3nguin`	`PiaF can DiaF.`
`16:45.30`	`[TK]D-Fender`	`bdfoster: Unqualified`
`16:45.58`	`bdfoster`	`hence the reason im attempting to use asterisknow`
`16:46.05`	`NMR_1122`	`And the values in AstDB can be modified external to the dialplan, like through crontab?`
`16:46.15`	`bdfoster`	`and, i know, im unqualified [TK]D-Fender`
`16:46.27`	`p3nguin`	`Which ddns service will you use?`
`16:46.41`	`bdfoster`	`i guess dyndns, i already have a free acct with them`
`16:46.42`	`[TK]D-Fender`	`bdfoster: I was referring to your recounting of your previous attempt`
`16:47.04`	`bdfoster`	`as far as trying to get it to work with piaf?`
`16:47.05`	`DarthPointer`	`bdfoster: piaf runs webmin on port 9001`
`16:47.05`	`p3nguin`	`ddclient should work well for that one. At least it did that last time I needed it.`
`16:47.13`	`[TK]D-Fender`	`bdfoster: Yuo seem to think EVERYTHING will fail.`
`16:47.35`	`Fubard`	`i deserve a cake. finally got this phone to register properly. apparently i'm horrible with autoprovisioning.`
`16:47.50`	`bdfoster`	`[TK]D-Fender: probably because it has at one point or another, and i would like to build a stable system`
`16:48.09`	`p3nguin`	`AsteriskNOW is stable.`
`16:48.10`	`[TK]D-Fender`	`bdfoster: "DDclient is a small but full featured client requiring only Perl and no additional modules. It runs under most UNIX OSes and has been tested under GNU/Linux and FreeBSD"`
`16:48.26`	`bdfoster`	`[TK]D-Fender: yes, ive read the sites`
`16:48.35`	`[TK]D-Fender`	`bdfoster: that is THEIR site`
`16:48.48`	`[TK]D-Fender`	`bdfoster: Its a Perl script... my PHONE could probably run this`
`16:48.51`	`bdfoster`	`yup, which is where i downloaded it the first time`
`16:49.04`	`bdfoster`	`and never worked`
`16:49.15`	`p3nguin`	`It might even be available via yum.`
`16:49.26`	`[TK]D-Fender`	`bdfoster: well unfortunately we can't go back to figure out where it all went wrong`
`16:49.33`	`*** join/#asterisk DennisG (~DennisG@84.30.136.208)`
`16:49.41`	`bdfoster`	`[TK]D-Fender: no need to, just gotta do it right this time`
`16:50.13`	`p3nguin`	`Yep, there it is.`
`16:50.14`	`[TK]D-Fender`	`bdfoster: That would help`
`16:50.38`	`p3nguin`	`in rpmforge.`
`16:52.16`	`bdfoster`	`p3nguin: adding it now`
`16:52.37`	`*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)`
`16:55.20`	`bdfoster`	`still getting used to this, im used to debian..`
`17:01.07`	`*** join/#asterisk DMeloUK (~DominicMe@64.129.93.210)`
`17:02.11`	`*** join/#asterisk lanning (~lanning@208.87.235.224)`
`17:02.38`	`Fubard`	`on polycom's, its normal to autoprovision auth.userId/password?`
`17:03.05`	`Fubard`	`it feels like it just doesn't want to take it`
`17:04.15`	`*** part/#asterisk DMeloUK (~DominicMe@64.129.93.210)`
`17:07.48`	`[TK]D-Fender`	`Fubard: phone local overrides take precedence`
`17:08.32`	`Katty`	`hai`
`17:09.13`	`[TK]D-Fender`	`Katty: Mew.`
`17:09.56`	`hardwire`	`indeed.`
`17:09.58`	`hardwire`	`mew.`
`17:16.18`	`*** join/#asterisk TJ^ (~tjq@host86-142-46-156.range86-142.btcentralplus.com)`
`17:16.28`	`TJ^`	`hi`
`17:16.43`	`TJ^`	`was hoping someone could help me out with a x100p`
`17:16.50`	`*** join/#asterisk Ad-Hoc (~nimbus@194.219.186.16.dsl.dyn.forthnet.gr)`
`17:16.58`	`TJ^`	`as far as i know its been installed correctly but i keep getting:`
`17:17.08`	`TJ^`	`-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/301-09768ff0", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack`
`17:21.58`	`*** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net)`
`17:22.18`	`QbY`	`anyone know of a softphone for both windows and linux that supports video and works with asterisk?`
`17:23.10`	`[TK]D-Fender`	`QbY: ...`
`17:23.12`	`[TK]D-Fender`	`~ekiga`
`17:23.13`	`infobot`	`[~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org`
`17:23.45`	`[TK]D-Fender`	`TJ^: that line says nothing of value. We do not see your configs or your actual attempt to call out.`
`17:23.46`	`*** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl)`
`17:23.56`	`[TK]D-Fender`	`TJ^: PASTEBIN is your friend`
`17:23.58`	`[TK]D-Fender`	`~pb`
`17:23.59`	`infobot`	`[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.`
`17:24.00`	`[TK]D-Fender`	`^^^^^^^6`
`17:24.02`	`TJ^`	`ok here it comes`
`17:24.19`	`QbY`	`i seen that one, but the windows download doesn't work.. we wanted to see if its possible to have a video conference with asterisk`
`17:24.20`	`*** join/#asterisk Madoc (~Madoc@74.12.181.50)`
`17:24.32`	`TJ^`	`QbY have u tried the linux version of xlite?`
`17:24.52`	`Madoc`	`Is there a way from an app to send a userevent? Anyone remember the function so I can look it up in the documentation?`
`17:25.02`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`17:25.52`	`QbY`	`TJ.. no... I was just looking for suggestions.. we are going to play..`
`17:26.24`	`[TK]D-Fender`	`Madoc: http://www.voip-info.org/wiki/view/asterisk+manager+events#UserEventEvent`
`17:27.22`	`TJ^`	`[TK]D-Fender http://pastebin.org/140244`
`17:28.20`	`[TK]D-Fender`	`TJ^: complete DAHDI configs please`
`17:28.27`	`Madoc`	`Yea I saw that, and instructions for how to use it in dialplans, but I'm doing a compiled c app`
`17:28.33`	`Madoc`	`Guess I can go look at how app_userevent does it`
`17:29.01`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
`17:29.11`	`TJ^`	`[TK]D-Fender /etc/dahdi/system.conf ?`
`17:29.31`	`[TK]D-Fender`	`TJ^: no, chan_adhdi.conf and everything linked to it`
`17:29.31`	`Madoc`	`wow actually app_userevent has exactally what I need, thanks for the link, pointed me at what I needed :)`
`17:31.01`	`*** part/#asterisk Madoc (~Madoc@74.12.181.50)`
`17:31.46`	`[TK]D-Fender`	`TJ^: /etc/asterisk`
`17:33.24`	`TJ^`	`http://pastebin.org/140253`
`17:33.39`	`*** join/#asterisk farkus_ (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com)`
`17:35.09`	`[TK]D-Fender`	`TJ^: #include dahdi-channels.conf <-----------`
`17:35.24`	`[TK]D-Fender`	`TJ^: #include chan_dahdi_additional.conf <------`
`17:35.35`	`*** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net)`
`17:35.38`	`[TK]D-Fender`	`[13:29]<[TK]D-Fender>TJ^: no, chan_adhdi.conf and everything linked to it <---`
`17:36.52`	`TJ^`	`http://pastebin.org/140255`
`17:37.10`	`*** join/#asterisk Brian_H (~Brian_H@static-173-50-141-22.ptldor.fios.verizon.net)`
`17:37.40`	`*** part/#asterisk Brian_H (~Brian_H@static-173-50-141-22.ptldor.fios.verizon.net)`
`17:37.40`	`TJ^`	`http://pastebin.org/140257`
`17:37.48`	`TJ^`	`sorry bout the [TK]D-Fender`
`17:39.42`	`*** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net)`
`17:40.26`	`Skeeter-`	`The Polycoms' Messages button can be configure via the web GUI to dial the asterisk voicemail directly, can it be coded within the configuration files?`
`17:42.50`	`[TK]D-Fender`	`Skeeter-: Forget the web config. NEVER touch it again. Ther is nothing that it can do that provisioning can't.. and can't do better.`
`17:43.27`	`Skeeter-`	`[TK]D-Fender, im aware of that, but i coudnt find it`
`17:44.21`	`WIMPy`	`Shouldn't that work automgically if the VM-extension is set correctely in sip.conf?`
`17:44.39`	`[TK]D-Fender`	`Skeeter-: Yes and we all know your aversion the reading the VERY FINE MANUALS`
`17:44.48`	`jdoe`	`Skeeter-: the way I did it was to have the voicemail button dial its own extension and match that in extensions.conf.`
`17:46.54`	`TJ^`	`[TK]D-Fender do my conf files look alright?`
`17:46.58`	`[TK]D-Fender`	`TJ^: do "core show channels concise"`
`17:49.55`	`TJ^`	`in cli?`
`17:49.58`	`TJ^`	`nothing happens`
`17:50.20`	`[TK]D-Fender`	`TJ^: core set verbose 10`
`17:50.24`	`Skeeter-`	`still cant find the way to convert a wav into polycomc readable format`
`17:50.27`	`[TK]D-Fender`	`TJ^: then repeat the other`
`17:50.38`	`Skeeter-`	`i found it 1 day but didnt write it down...`
`17:50.51`	`TJ^`	`changed from 3 to 10`
`17:50.56`	`TJ^`	`no change on 2nd command`
`17:52.09`	`[TK]D-Fender`	`TJ^: ... PASTEBIN`
`17:52.22`	`[TK]D-Fender`	`TJ^: should see something for "core show channels"`
`17:52.45`	`TJ^`	`i do on core show channe;s`
`17:52.51`	`TJ^`	`but nor channels concise`
`17:52.54`	`TJ^`	`*not`
`17:53.08`	`[TK]D-Fender`	`TJ^: fine, show that`
`17:53.25`	`TJ^`	`http://pastebin.org/140277`
`17:55.37`	`[TK]D-Fender`	`TJ^: try another call. If that fails, restart * completely.`
`17:55.55`	`*** join/#asterisk h1kz0r (~MJz@c-68-44-81-246.hsd1.nj.comcast.net)`
`17:56.10`	`TJ^`	`all circuits are busy ><`
`17:56.18`	`h1kz0r`	`hello everyone`
`17:56.26`	`[TK]D-Fender`	`TJ^: PASTEBIN. That message is meaningless`
`17:56.57`	`[TK]D-Fender`	`h1kz0r: ...`
`17:56.59`	`[TK]D-Fender`	`~book`
`17:57.00`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`17:57.08`	`[TK]D-Fender`	`h1kz0r: :)`
`17:57.18`	`h1kz0r`	`yea i read that yesterday ;)`
`17:57.32`	`[TK]D-Fender`	`h1kz0r: Excellent`
`17:57.34`	`h1kz0r`	`btw it's MJz here, i had to change and register a nick to come here`
`17:57.44`	`[TK]D-Fender`	`h1kz0r: Well aware... I read hostnames you know...`
`17:58.06`	`h1kz0r`	`hehehe harcore irc.. been like 10 years since i joined here`
`17:58.11`	`[TK]D-Fender`	`[13:55]-->\|h1kz0r (~MJz@c-68-44-81-246.hsd1.nj.comcast.net) has joined #asterisk`
`17:58.13`	`[TK]D-Fender`	`[13:47]-->\|MJz (~MJz@c-68-44-81-246.hsd1.nj.comcast.net) has joined #freepbx`
`17:58.46`	`h1kz0r`	`hopefully i can get some help here :)`
`17:59.00`	`TJ^`	`[TK]D-Fender http://pastebin.org/140286`
`17:59.25`	`[TK]D-Fender`	`TJ^: that is post restart?`
`17:59.46`	`TJ^`	`sorry m8 no`
`18:01.40`	`h1kz0r`	`any asterisk gurus in the house willing to give a hand? ;)`
`18:01.59`	`p3nguin`	`h1kz0r: Did you ever ask your question?`
`18:02.04`	`p3nguin`	`~ask`
`18:02.05`	`infobot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`18:02.21`	`*** join/#asterisk Wildy (~simba@83.149.9.59)`
`18:02.32`	`rdircio`	`s/gurus/noobs/`
`18:03.27`	`TJ^`	`[TK]D-Fender rebooting`
`18:03.39`	`h1kz0r`	`ok i'm trying to setup 2 sip providers, one for international outgoing calls only and another providing a DID number, i want to be able to call that number from my cell phone for example and based on my ANI make international calls using the other sip provider`
`18:04.33`	`h1kz0r`	`i did this on a SPA but with a land line, here i want to do the same with a DID from some provider, all voip`
`18:04.50`	`p3nguin`	`Easy enough.`
`18:05.28`	`paulc`	`wonders who's here against their will`
`18:06.43`	`p3nguin`	`h1kz0r: Just remember that people can spoof caller information, so make sure you implement another form of authentication, such as a PIN.`
`18:07.51`	`*** join/#asterisk bdfoster (~5712ea8c@gateway/web/freenode/x-xkaqkxtmippgrzkp)`
`18:08.02`	`bdfoster`	`[TK]D-Fender: #freepbx`
`18:08.05`	`*** part/#asterisk grishnav (~grishnav@e2180-8777.securedservers.com)`
`18:08.17`	`h1kz0r`	`yes i want to do both (ANI and/or PIN), as soon as I know how to do any :)`
`18:08.28`	`*** join/#asterisk Circlefusion (~circlefus@74-142-193-26.dhcp.insightbb.com)`
`18:09.09`	`h1kz0r`	`does switchvox does this or I'm better using the asterisk directly?`
`18:10.42`	`[TK]D-Fender`	`bdfoster: I'm sorry... you don't seem to know what you're doing or any idea of anything to look at on your own.`
`18:10.50`	`[TK]D-Fender`	`h1kz0r: * is just fine for this.`
`18:11.28`	`[TK]D-Fender`	`h1Take call in. Look at funny. Determin what do do with it. Send it out A/B/C/ refuse, etc.`
`18:11.34`	`[TK]D-Fender`	`h1kz0r: All pretty easy stuff.`
`18:11.40`	`h1kz0r`	`so p3nguin, how easy is it to do this`
`18:11.56`	`bdfoster`	`[TK]D-Fender: all i did was yum install openvpn`
`18:12.02`	`[TK]D-Fender`	`h1kz0r: 1st step. Set up your fist provider. Get calls to come in.`
`18:12.02`	`p3nguin`	`h1kz0r: You should be able to have it done in under an hour.`
`18:12.04`	`bdfoster`	`reboot`
`18:12.09`	`[TK]D-Fender`	`bdfoster: Yeah, what about configuring?`
`18:12.16`	`paulc`	`h1kz0r: very easy - look at the Authenticate command if you want to verify with PIN`
`18:12.30`	`bdfoster`	`well, it would be nice if i could get to that page as well, but i cant`
`18:13.08`	`[TK]D-Fender`	`bdfoster: Page? What page?`
`18:13.16`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`18:13.23`	`p3nguin`	`h1kz0r: Based on the calling phone number, allow access to a DISA, but use Authenticate to provide PIN access.`
`18:13.29`	`bdfoster`	`192.168.X.X:943`
`18:13.39`	`bdfoster`	`https or http`
`18:13.40`	`[TK]D-Fender`	`bdfoster: And what is supposed to be there?`
`18:13.47`	`bdfoster`	`openvpn page`
`18:14.02`	`p3nguin`	`h1kz0r: I've tried using DISA with a password, but it seems to work better using Authenticate() first, then DISA() without a password.`
`18:14.14`	`[TK]D-Fender`	`bbdWho said there was a web page for it?`
`18:14.44`	`bdfoster`	`shit, that was the client`
`18:14.52`	`h1kz0r`	`now we're talking, i don't have a DISA as a destination, let me install the module and see what i can play with`
`18:15.28`	`[TK]D-Fender`	`h1kz0r: You want to prompt for a number to dial out?`
`18:15.37`	`h1kz0r`	`yes precisely`
`18:15.50`	`h1kz0r`	`disa or a tone back should be fine`
`18:15.55`	`[TK]D-Fender`	`h1kz0r: Ok, DISA maye be a decent option for this...`
`18:16.14`	`[TK]D-Fender`	`b1app_disa is basically a dumb dialtone with access to the context you point it to.`
`18:16.16`	`p3nguin`	`You could even use Read() for taking a number in, but I don't like that the dial tone doesn't cancel after the first key press.`
`18:16.41`	`[TK]D-Fender`	`p3nguin: Or make a basic IVR using a recording of dialtone background-ed :)`
`18:16.57`	`p3nguin`	`interesting`
`18:17.22`	`[TK]D-Fender`	`p3nguin: DISA offers no i/t control and no timeout delay functions`
`18:17.28`	`[TK]D-Fender`	`p3nguin: and creates extra CDR's`
`18:17.37`	`[TK]D-Fender`	`p3nguin: Most of the time I consider it a waste`
`18:18.02`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`18:18.17`	`*** join/#asterisk diegomad (mad@186.81.136.65)`
`18:19.05`	`h1kz0r`	`ok i installed the DISA module, i have it now`
`18:19.14`	`*** part/#asterisk rdircio (~admin@201.137.57.68)`
`18:19.23`	`h1kz0r`	`i just need something simple now, we'll complicate it later`
`18:19.40`	`TJ^`	`[TK]D-Fender post reboot if fails too, i will pastebin the log for u`
`18:19.42`	`bdfoster`	`[TK]D-Fender: http://192.168.X.X/admin/config.php does not show`
`18:19.48`	`[TK]D-Fender`	`h1kz0r: Then start with that`
`18:19.55`	`h1kz0r`	`dial in, detect my ANI, give me tone, i dial the number, * connects us and then hang up`
`18:20.04`	`TJ^`	`[TK]D-Fender am i correct in assuming these cards need a specific type of RJ11 cable?`
`18:20.38`	`*** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-170-134.clienti.tiscali.it)`
`18:21.09`	`[TK]D-Fender`	`bdfoster: I have no idea how you screwed this up. You're showing jack shit. You don't seem to have configured anything or show even awareness of WHERE the configs are for any of this. You aren't showing any networking status dumps or anything else at all. How is anyone supposed to help you?`
`18:21.31`	`[TK]D-Fender`	`bdfoster: "It doesn't work" <- offers nothing.`
`18:21.38`	`bdfoster`	`[TK]D-Fender: tell me what you want and where and ill give it to you`
`18:22.10`	`[TK]D-Fender`	`bdfoster: You want to run OpenVPN and have no clue how it works. Go ask in #openvpn`
`18:22.29`	`bdfoster`	`[TK]D-Fender: right now, this is freepx, wanna meet me over there?`
`18:22.52`	`[TK]D-Fender`	`bdfoster: No, this is #ASTERISK . You aren't even aware of what room you are in NOW.`
`18:23.06`	`[TK]D-Fender`	`eraches for his ClueBat (tm)`
`18:23.10`	`[TK]D-Fender`	`reaches*`
`18:23.23`	`h1kz0r`	`lmao`
`18:23.26`	`Kobaz`	`damnit, this is horrible`
`18:23.37`	`Kobaz`	`[TK]D-Fender: that allow/deny thing worked perfectly`
`18:23.41`	`bdfoster`	`i know what room this is [TK]D-Fender but im dealing with a FREEPBX issue so do you want to meet me over in #freepbx`
`18:23.48`	`[TK]D-Fender`	`Kobaz: Loev to say "I told you so" :p`
`18:23.55`	`Kobaz`	`this t1 card/drivers/etc keeps dieing`
`18:24.07`	`Kobaz`	`i'm so happy we get to replace it with a gateway tomorrow`
`18:24.16`	`[TK]D-Fender`	`bdfoster: This isn't a freePBX issue. This is a "I fucked up all of the networking on my server" issue.`
`18:24.25`	`bdfoster`	`fuck it`
`18:24.36`	`Kobaz`	`i don't even want to really bother debugging it any more`
`18:24.36`	`Kobaz`	`but`
`18:24.55`	`[TK]D-Fender`	`Kobaz: You don't need to... it's working!`
`18:24.57`	`Kobaz`	`anyone know why the d channel would be up, no alarms, no errors, no nothing... but the pri just stops working`
`18:25.15`	`Kobaz`	`[TK]D-Fender: heh.. my gateway works.. but now we need to replace the card`
`18:25.23`	`[TK]D-Fender`	`Which & why?`
`18:25.29`	`Kobaz`	`sangoma a101d`
`18:25.46`	`Kobaz`	`i think it's getting worse`
`18:25.51`	`bdfoster`	`[TK]D-Fender: by the way, not a networking issue either, as I'm SSH'ed into it, call centric says it's registered, and my extensions are all registered`
`18:25.56`	`bdfoster`	`looks like a PHP issue`
`18:25.56`	`Kobaz`	`it used to happen once a week, then it was twice a week`
`18:26.00`	`Kobaz`	`now it's two times in one day`
`18:26.07`	`[TK]D-Fender`	`bdfoster: Which Open VPN has nothing to do with`
`18:26.21`	`Kobaz`	`it's almost like i need to do an asterisk restart after every completed call`
`18:26.26`	`p3nguin`	`h1kz0r: http://pastebin.com/zc7XZQrg`
`18:26.27`	`[TK]D-Fender`	`bdfoster: What leads you to believe its a PHP issue?`
`18:27.04`	`bdfoster`	`[TK]D-Fender: by the way, not a networking issue either, as I'm SSH'ed into it, call centric says it's registered, and my extensions are all registered`
`18:27.15`	`[TK]D-Fender`	`bdfoster: You're skipping...`
`18:27.24`	`Kobaz`	`[TK]D-Fender: i think sangoma makes great hardware, but the hardware<->asterisk layer is just too flakey`
`18:27.29`	`bdfoster`	`im repeating what i said`
`18:27.48`	`DarthPointer`	`kobaz are you running freepbx?`
`18:27.48`	`[TK]D-Fender`	`Kobaz: Dunno... not a single client of mine has ever had any issue with them`
`18:27.55`	`[TK]D-Fender`	`bdfoster: Yes... and not adding anything.`
`18:27.58`	`Kobaz`	`DarthPointer: of course not`
`18:28.14`	`[TK]D-Fender`	`bdfoster: You poitn the finger at PHP. Why? How do I know what daemons are listening? On what ports?`
`18:28.18`	`Kobaz`	`[TK]D-Fender: i've had several problems over the last two years, but nothing this big before`
`18:28.24`	`[TK]D-Fender`	`bdfoster: Being accessed on what machine? Over what interfaces?`
`18:28.35`	`[TK]D-Fender`	`bdfoster: You offer absolutely no backup`
`18:28.36`	`DarthPointer`	`kobaz: what happens w/ debug and verbose on in the los?`
`18:28.53`	`Kobaz`	`DarthPointer: nothing... and there is no los according to the driver`
`18:28.55`	`p3nguin`	`h1kz0r: Alternatively, you may want an IVR which allows you to not go straight into the DISA. That's how I do it, because I may or may not want to dial out when I call in.`
`18:29.03`	`bdfoster`	`[TK]D-Fender: i just love this community`
`18:29.26`	`[TK]D-Fender`	`bdfoster: You ask for help and show nothing? What do you think we are, psychic?`
`18:29.33`	`[TK]D-Fender`	`bdfoster: You are doing nothing to help yourself`
`18:29.35`	`DarthPointer`	`kobaz: so your solution now is to restart *?`
`18:29.44`	`bdfoster`	`[TK]D-Fender: i dont know where to start`
`18:29.48`	`Kobaz`	`DarthPointer: either restart asterisk or module unload/load chan_dahdi... both work`
`18:29.57`	`[TK]D-Fender`	`bdfoster: Yet you already ahev a culprit?`
`18:29.58`	`bdfoster`	`[TK]D-Fender: ive said this way to many times`
`18:30.05`	`bdfoster`	`[TK]D-Fender: i took a guess`
`18:30.15`	`[TK]D-Fender`	`bdfoster: You don't know how to see what interfaces are running? You can't even tell me HOW you're looking at it?`
`18:30.16`	`Kobaz`	`DarthPointer: d channel reports as being up... but there's no data passing around`
`18:30.43`	`bdfoster`	`[TK]D-Fender: eth0, i am ssh'ed into it on port 22`
`18:30.54`	`DarthPointer`	`kobz: what version of *? Dahdi? Zaptel?`
`18:31.14`	`Kobaz`	`DarthPointer: 1.6.0.26, dahdi is umm 2.2.1`
`18:31.18`	`wcselby`	`well, that was a good lunch`
`18:31.22`	`Kobaz`	`i should probably downgrade to 2.2.0`
`18:31.24`	`wcselby`	`how was ya'lls?`
`18:31.29`	`TJ^`	`anyone here have a xp100?`
`18:31.40`	`bdfoster`	`TJ^: yea, piece of shit`
`18:31.45`	`TJ^`	`haha y?`
`18:31.49`	`TJ^`	`xp100 se?`
`18:31.58`	`bdfoster`	`idk lemme get it`
`18:31.58`	`[TK]D-Fender`	`bdfoster: What interfaces are active? What ports is it listening on? What daemons are running?`
`18:32.02`	`beek`	`piece of shite, special edition.`
`18:32.30`	`[TK]D-Fender`	`Shittier Edition`
`18:33.04`	`bdfoster`	`TJ^: i guess just the regular one`
`18:33.13`	`Kobaz`	`[TK]D-Fender: that's the general problem that i see with dahdi related issues... is that 99% of the population doesn't have any problems, and in thsoe situations where libpri crashes, or there's mysterious data loss... noone knows what's going on`
`18:33.44`	`TJ^`	`well i am new to this and didnt want to spend much money on an fxo card`
`18:33.54`	`TJ^`	`anyone know what kind of rj11 these things need?`
`18:33.54`	`Kobaz`	`~cheap`
`18:33.55`	`infobot`	`rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.`
`18:34.14`	`wcselby`	`eclipse is open source, right?`
`18:34.20`	`Kobaz`	`an rj11 is an rj11, is an rj11... there is no 'types of rj11'`
`18:34.20`	`wcselby`	`<PROTECTED>`
`18:34.25`	`wcselby`	`is eclipse open source?`
`18:34.26`	`Corydon76-dig`	`wcselby: Yes, written in Java`
`18:34.34`	`wcselby`	`Corydon76-dig - thanks`
`18:34.44`	`bdfoster`	`TJ^: get a voip line`
`18:34.56`	`bdfoster`	`cheaper, works`
`18:35.53`	`TJ^`	`lol`
`18:37.00`	`wcselby`	`TJ^ - those xp100's are bad knock off clones of digium boards that are really old. you will have more trouble trying to get one to work than would be worth the investment in a slightly more expensive, yet insanely better, card`
`18:37.36`	`TJ^`	`wcselby i read all about those clones and made sure i got one that wasnt cloned`
`18:37.42`	`h1kz0r`	`thanks p3nguin, and that extension number is the ANI right?`
`18:38.06`	`TJ^`	`wcselby im a poor student man the next best thing was £60`
`18:38.07`	`Kobaz`	`TJ^: they are all clones`
`18:38.08`	`[TK]D-Fender`	`h1kz0r: No, its the DID`
`18:38.25`	`[TK]D-Fender`	`TJ^: PB : "dahdi show channels`
`18:38.46`	`h1kz0r`	`oh ye ic`
`18:38.47`	`Kobaz`	`TJ^: pwnie up and get an audiocodes gateway`
`18:38.53`	`*** join/#asterisk murraytm (~murraytm@wsip-68-224-219-238.br.no.cox.net)`
`18:39.07`	`TJ^`	`!google audiocodes gateway`
`18:39.50`	`Kobaz`	`the linksys ones aren't that bad, grandstream is usable for the single port thingers`
`18:40.04`	`wcselby`	`TJ^ - VOIP lines should be < 5 quid a month`
`18:40.05`	`Kobaz`	`dont buy grandstream phones though`
`18:40.18`	`wcselby`	`fuck if I know what a quid is`
`18:40.22`	`TJ^`	`[TK]D-Fender http://pastebin.org/140343`
`18:40.40`	`TJ^`	`wcselby its not so much about making calls but receiving them`
`18:40.50`	`TJ^`	`weve had the number for a long time etc etc`
`18:40.52`	`murraytm`	`how does the associative array function (HASH) work with channel variable inheritance in 1.6.2? do you put the underscore before the function or before the hashname?`
`18:41.03`	`[TK]D-Fender`	`TJ^: dahdi show status`
`18:41.21`	`TJ^`	`wcselby dyu know any good voip providers in the uk apart from gradwel and voip talk?`
`18:42.00`	`TJ^`	`[TK]D-Fender http://pastebin.org/140355`
`18:42.06`	`wcselby`	`TJ^ - sorry I'm in the US`
`18:42.13`	`wcselby`	`i haven't shopped UK voip providers`
`18:42.23`	`[TK]D-Fender`	`TJ^: Wildcard X100P Board 1 RED 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)`
`18:42.39`	`[TK]D-Fender`	`TJ^: RED. IIRC in the UK you need to watch out for what pair you use for analog <-`
`18:42.52`	`[TK]D-Fender`	`TJ^: double check this. Looks like it isn't connected on the right pair`
`18:43.20`	`TJ^`	`i knew it would be something like that`
`18:43.31`	`TJ^`	`thats why i was asking bout rj11 cables`
`18:43.36`	`*** join/#asterisk meesterarend (~frans@vc-41-30-43-56.umts.vodacom.co.za)`
`18:43.41`	`TJ^`	`sorry`
`18:43.47`	`TJ^`	`looks like jumped the gun`
`18:43.49`	`TJ^`	`whats the pair?`
`18:44.30`	`h1kz0r`	`hey p3nguin/tk do you mind breaking this out a bit for me, the flow of the call`
`18:45.52`	`h1kz0r`	`i'm using freepbx also, for instance Authenticate(${DB(disa/pin)}) is this the pin i put in the disa config?`
`18:46.47`	`wcselby`	`TJ^ - don't know if this is valid, but from a quick google i found this --> http://www.velocityreviews.com/forums/t594683-cable-for-fxo-connections-spa-3102-a.html`
`18:47.51`	`TJ^`	`The cable used to connect the X100P SE to the PSTN telephone socket must be an RJ11`
`18:47.51`	`TJ^`	`modem cable. A modem cable uses the centre pair of wires (3 and 4) whereas a telephone`
`18:47.52`	`TJ^`	`cable may use a different pair. For example, in the UK a telephone cable will often (not`
`18:47.52`	`TJ^`	`always) use wires 1 and 5. If the modem cable included with the X100P SE is being used for`
`18:47.52`	`TJ^`	`this connection, then a country specific adapter may be required for the telephone socket end.`
`18:47.52`	`TJ^`	`For example, in the UK a RJ11 female socket port to BT/UK male telephone socket plug`
`18:47.53`	`TJ^`	`adapter will be required.`
`18:47.54`	`TJ^`	`shit`
`18:47.58`	`TJ^`	`(sorry)`
`18:48.13`	`[TK]D-Fender`	`TJ^: yeah, please don't do that again....`
`18:48.37`	`TJ^`	`so looks like i got the wrong cable`
`18:48.39`	`[TK]D-Fender`	`TJ^: also make sure you're on the right jack at all on the back of that card`
`18:48.51`	`TJ^`	`made sure of that :)`
`18:48.53`	`[TK]D-Fender`	`TJ^: could require only a minor hack`
`18:49.20`	`[TK]D-Fender`	`h1kz0r: For now just replace ${DB(disa/pin)} with a number of your choice`
`18:49.35`	`[TK]D-Fender`	`h1kz0r: He offered to help abstract it so you could also dial in to change the pass, etc`
`18:49.48`	`[TK]D-Fender`	`h1kz0r: Was generous of him, but requires a little bit more work to complete`
`18:50.27`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`18:50.53`	`h1kz0r`	`yes i can see that, i'm trying to put it to work though and this is a bit overwhelming`
`18:51.15`	`[TK]D-Fender`	`h1kz0r: swap as I sugegsted, and we'll look at this a bit at a time`
`18:51.15`	`p3nguin`	`Oops, I forgot I store my PIN in astDB, but he doesn't (yet).`
`18:51.22`	`h1kz0r`	`the freepbx thing confused me i guess, nothing from what i'm trying to do can be done through the interface it seems`
`18:51.59`	`[TK]D-Fender`	`h1kz0r: as I said, FreePBX offers you nothign for this`
`18:52.24`	`[TK]D-Fender`	`h1kz0r: You'll have to code it all anyways, and then learn all that GUI crap just to get off the ground`
`18:52.32`	`h1kz0r`	`can you break down line 1 for me please?`
`18:52.41`	`[TK]D-Fender`	`h1kz0r: GUI is good if you are running a more common PBX implementation.`
`18:52.56`	`[TK]D-Fender`	`h1kz0r: pastebin it`
`18:52.56`	`*** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org)`
`18:53.03`	`h1kz0r`	`ye well now i know "where" to work anyway :)`
`18:53.24`	`h1kz0r`	`http://pastebin.com/zc7XZQrg`
`18:53.41`	`*** join/#asterisk meesterarend (~frans@vc-41-30-43-56.umts.vodacom.co.za)`
`18:54.03`	`[TK]D-Fender`	`h1kz0r: 1st line compares the CallerID and if it matches jumps down to authenticate you`
`18:54.05`	`p3nguin`	`exten 3149691077 is your phone number that people (you) call to reach the * system.`
`18:54.05`	`*** join/#asterisk Chibat (~Chibat@67.214.225.82)`
`18:55.01`	`h1kz0r`	`i do a fair amount of programming under linux and at my job we never install the GUI, jeos only so i definitely know what you mean`
`18:55.24`	`*** join/#asterisk meesterarend (~frans@vc-41-30-43-56.umts.vodacom.co.za)`
`18:55.27`	`Chibat`	`Voicemail: no sound over sip, sound files are still valid, these were working about an hour ago just fine any suggestions to check i'm watching console right now. thanks`
`18:56.00`	`h1kz0r`	`and 3148665101 is "my" cell phone for example?`
`18:56.38`	`p3nguin`	`yes, the caller ID number of the callING phone.`
`18:57.02`	`Chibat`	`Playing 'vm-message' (language 'en')`
`18:57.11`	`Chibat`	`but i do not hear anything`
`18:57.12`	`h1kz0r`	`does this mean if the callerid matches jump to line 5?`
`18:58.00`	`[TK]D-Fender`	`h1kz0r: yes`
`18:58.05`	`p3nguin`	`h1kz0r: Yes, the ''?disa'' part of that GotoIf() command says if the evaluation was true, go to the label "disa", which is on line 5.`
`18:58.41`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`18:58.59`	`h1kz0r`	`ok and line 9 means go to the outcontext which will be defined as my other sip provider right?`
`19:00.16`	`p3nguin`	`The "outcontext" will be used when you dial a phone number from inside the DISA.`
`19:01.23`	`p3nguin`	`If you dial 005 in the DISA, * is going to try to find exten => 005,1,command in [outcontext].`
`19:02.14`	`h1kz0r`	`what if i just want it to send whatever i type as is to the other provider?`
`19:02.39`	`[TK]D-Fender`	`h1kz0r: then use a more generic pattern`
`19:03.07`	`[TK]D-Fender`	`h1kz0r: exten => _X.,1,Dial(SIP/provider/${EXTEN})`
`19:03.10`	`[TK]D-Fender`	`h1kz0r: for example`
`19:03.35`	`p3nguin`	`Pattern matching is a wonderful thing.`
`19:03.44`	`bmoraca_work`	`holy hell i hate telephones, telephony, and everything that goes with them`
`19:04.24`	`h1kz0r`	`Assuming i'm an expert on linux regex, is this the same concept?`
`19:04.50`	`wcselby`	`bmoraca_work - i feel that way about faxing mixed with voip`
`19:05.04`	`bmoraca_work`	`h1kz0r: only in the sense that assembler is like BASIC in that it can print messages on the screen`
`19:07.19`	`h1kz0r`	`[TK]D-Fender: ur example makes sense but this isn't X. like the wildcard and will not allow me to put additional extentions or how does it work`
`19:08.33`	`p3nguin`	`h1kz0r: regex is not the same as extension pattern matching at all.`
`19:10.15`	`Chibat`	`Does: anyone know why possibly after years of my voicemail sound files playing over the phone now suddenly might not? Nothing has changed files are still valid.`
`19:10.16`	`h1kz0r`	`so exten => _X.,1,Dial(SIP/provider/${EXTEN}) should work in this scenario without affecting other stuff?`
`19:11.13`	`h1kz0r`	`what does _X. means here? extentions that are not matched anywhere else?`
`19:11.56`	`p3nguin`	`~dialplan patterns`
`19:11.57`	`infobot`	`dialplan patterns are explained here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns`
`19:13.51`	`*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)`
`19:15.20`	`*** join/#asterisk capitan (~captain@76.91.206.32)`
`19:16.23`	`p3nguin`	`h1kz0r: _ indicates we are going to start a pattern match.`
`19:16.27`	`p3nguin`	`h1kz0r: X is any digit from 0 to 9.`
`19:16.45`	`p3nguin`	`h1kz0r: . is any one or more character.`
`19:16.49`	`h1kz0r`	`thatnks p3nguin i bookmarked this page`
`19:16.57`	`hardwire`	`hmmm.`
`19:17.10`	`h1kz0r`	`ye i kinda knew what it means technically but in terms of call behavior`
`19:17.33`	`h1kz0r`	`it's like the .* in regex or * in linux`
`19:17.36`	`p3nguin`	`h1kz0r: _X. matches any two or more, so a single number would not match.`
`19:18.41`	`h1kz0r`	`ok so using this, i can make different routing through different providers based on country codes for example right?`
`19:19.18`	`p3nguin`	`And we don't (typically) use _. because that matches everything, including the standard (special?) extensions such as i, h, s, t, et cetera.`
`19:19.29`	`Chibat`	`voicemail isn't being created now as well`
`19:19.38`	`[TK]D-Fender`	`h1kz0r: Yes you can make les generic ones where you can pick provider A who is cheaper in country 1, and B for countery 2, etc based on the # dialed`
`19:19.54`	`h1kz0r`	`makes sense`
`19:20.04`	`p3nguin`	`_00480XXXXXX,1,Dial()`
`19:20.39`	`h1kz0r`	`and I should put all this in the same context right?`
`19:21.10`	`p3nguin`	`I probably would. [outgoing] or something.`
`19:21.36`	`h1kz0r`	`yes, this is the outcontext which will be called from the disa one right?`
`19:21.45`	`p3nguin`	`It's your dialplan, so you can do anything you want.`
`19:22.32`	`p3nguin`	`You can even have a few layers of contexts if you desire.`
`19:22.59`	`h1kz0r`	`do I have to do anything special if I have to register with the outgoing sip provider? maybe register in the sip.conf first?`
`19:23.01`	`p3nguin`	`[outcontext] would then include => outgoing_international for example.`
`19:23.17`	`p3nguin`	`For outgoing, you don't need to register at all.`
`19:23.35`	`h1kz0r`	`this provider requires me to register for outgoing as well`
`19:23.40`	`p3nguin`	`I doubt that.`
`19:23.55`	`p3nguin`	`They might require you to authenticate, but that has nothing to do with registration.`
`19:23.58`	`h1kz0r`	`this is the reason i can't do it in the SPA, i can't register with 2`
`19:24.32`	`p3nguin`	`Registration tells the remote system how to reach you. If you are not receiving a call, you don't need to be reached.`
`19:24.39`	`h1kz0r`	`i tried but didn't work and they told me i have to register`
`19:25.07`	`h1kz0r`	`technically i should be able to recieve there as well, but those guys don't have DIDs, just local extentions which i'm not interested in`
`19:26.03`	`[TK]D-Fender`	`[15:23]<h1kz0r>this provider requires me to register for outgoing as well <- rare, but possible`
`19:26.12`	`p3nguin`	`Without a DID, you won't be getting many calls.`
`19:26.43`	`h1kz0r`	`i know but this is what it is now and i do wanna keep them for the destinations i call the most`
`19:27.42`	`h1kz0r`	`will the DIAL() register or it's enough if i register separately in the sip.conf?`
`19:27.58`	`p3nguin`	`Dial() only dials.`
`19:29.28`	`h1kz0r`	`but in this scenario i should be able to register and the dial is "supposed to" work?`
`19:30.05`	`p3nguin`	`I still maintain that you do not REGISTER to dial outbound.`
`19:30.19`	`h1kz0r`	`k i'll test and get back to you`
`19:30.27`	`h1kz0r`	`thanks a lot p3nguin [TK]D-Fender`
`19:30.44`	`h1kz0r`	`i'll be back in few hours :)`
`19:33.21`	`[TK]D-Fender`	`p3nguin: tehre are a few providers that require this as they auth by IP not during each call.`
`19:33.57`	`*** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org)`
`19:34.15`	`p3nguin`	`I suppose if he has one of those, he'll be asking for help to form his register statement pretty soon.`
`19:34.46`	`*** join/#asterisk atis_work (~atis_work@193.238.212.171)`
`19:34.48`	`*** join/#asterisk endigo-ceo (~5712d112@gateway/web/freenode/x-guxsmskubszrsivb)`
`19:34.52`	`*** join/#asterisk uqlev (~yuriy@91.184.221.31)`
`19:35.09`	`*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)`
`19:35.27`	`endigo-ceo`	`is logging all chat dialog in this channel`
`19:36.29`	`p3nguin`	`Why?`
`19:37.08`	`Chibat`	`hellO?`
`19:37.54`	`wcselby`	`Chibat - do you have any disk space left on the drive where your voicemails are stored?`
`19:39.40`	`[TK]D-Fender`	`Chiis your filesystem mounted RW?`
`19:39.56`	`*** join/#asterisk Alagar (~Administr@122.164.39.167)`
`19:41.01`	`bmoraca_work`	`needs a new job`
`19:41.34`	*** join/#asterisk byteme` (~blah@69.73.167.236)
`19:42.00`	`Naikrovek`	`bmoraca_work: why`
`19:42.12`	`*** join/#asterisk meesterarend (~frans@vc-41-30-43-56.umts.vodacom.co.za)`
`19:42.19`	`*** join/#asterisk lynxsys (~lynxsys@213-152-36-251.dsl.eclipse.net.uk)`
`19:42.22`	`Chibat`	`oh i will check`
`19:42.49`	`wcselby`	`Naikrovek - prolly has to do with this statement ---> <bmoraca_work> holy hell i hate telephones, telephony, and everything that goes with them`
`19:43.04`	`Chibat`	`wcselby: yes i do`
`19:43.25`	`bmoraca_work`	`Naikrovek: because people here don't listen to me when i tell them that shit isn't going to work and make me implement it anyway. then they expect me to have all the answers for fixing it. i usually do, but that's besides the point.`
`19:43.33`	`Chibat`	`yes mounted RW`
`19:43.36`	`*** join/#asterisk mcateer (~jmcateer@75.76.254.162)`
`19:43.48`	`Qwell`	`blinks at mcateer`
`19:43.54`	`*** join/#asterisk spenguin[work] (~penguin@122.182.0.38)`
`19:43.57`	`Chibat`	`iit was working all morning, every once in a while ftp server crashes and I have to restart the whole box`
`19:44.09`	`Chibat`	`but then it works just fine again, hold music doesn't work either`
`19:44.35`	`Naikrovek`	`bmoraca_work: do it your way anyway`
`19:44.42`	`bmoraca_work`	`Naikrovek: also, they refuse to let me train someone else to help with all the bullshit. i've explained a hundred times to one office that *66 is the code to toggle nightmode, but they think that my time is best spent telling people how to "make their phones stop dialing letters" and fixing their isymphony displays`
`19:44.48`	`Naikrovek`	`bmoraca_work: FAAARRRR easier to apologize than to ask for permission`
`19:45.04`	`Fubard`	`Chibat: has moh ever worked or just stopped working recently?`
`19:45.11`	`bmoraca_work`	`Naikrovek: my way would have cost more and worked...it would have needed more equipment which wasn't authorized`
`19:45.15`	`*** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58)`
`19:45.18`	`Naikrovek`	`bmoraca_work: ah`
`19:45.59`	`Naikrovek`	`bmoraca_work: i agree. you need a new job. or to get some respect`
`19:46.05`	`Naikrovek`	`from your employer i mean`
`19:46.06`	`Naikrovek`	`i respect you`
`19:46.09`	`Naikrovek`	`but i don't count`
`19:46.14`	`mcateer`	`Qwell: I never realized you could blink _at_ someone. lol`
`19:46.20`	`bmoraca_work`	`lol`
`19:46.28`	`Fubard`	`from what i hear, a little respect is all you need. R-E-S-P-E-C-T.`
`19:46.36`	`Qwell`	`mcateer: it's the Internet. You can do lots of things at someones.`
`19:46.38`	`Fubard`	`gonna google: what does respect mean to you?`
`19:46.45`	`Qwell`	`for example:`
`19:46.50`	`Qwell`	`throws a fish taco at file`
`19:47.06`	`mcateer`	`nice`
`19:47.09`	`bmoraca_work`	`Naikrovek: i basically keep this place running. when someone has a problem internally, they call me first. hell, i even get the bookkeeper asking me accounting questions.`
`19:47.33`	`Naikrovek`	`bmoraca_work: same here. i'm a bit forceful when people ask me goofy questions first`
`19:47.51`	`Naikrovek`	`i mean, i understand if computing isn't your skillset... but to ask me how to look up a word definition online? i got that today`
`19:47.56`	`Chibat`	`Fubard: yes it did work, also using cat on the hold music mp3's`
`19:48.35`	`bmoraca_work`	`Naikrovek: i've already gotten chastised for being frustrated when someone asked me to email him the i386 folder from Win 2k3 while he was on a job site that he KNEW he would need it at. fucker didn't bring it with him and expected me to drop everything i was doing and get it to him immediately`
`19:48.45`	`Naikrovek`	`what i don't understand, is how computing ISN'T in your skillset, when your resume says it is, and that you have 5 years experience writing computer programs, and I have to tell you how to log on to your computer.`
`19:48.57`	`spenguin[work]`	`lol email the i386 folder`
`19:49.00`	`Naikrovek`	`bmoraca_work: hah`
`19:49.05`	`spenguin[work]`	`haha`
`19:49.25`	`Naikrovek`	`bmoraca_work: i don't have tolerance for that. mgmt. anger or not, someone does something stupid, I tell them`
`19:49.34`	`bmoraca_work`	`yep`
`19:49.35`	`Naikrovek`	`your lack of intelligence is not my emergency`
`19:49.48`	`Naikrovek`	`not yours, theirs points`
`19:49.59`	`Naikrovek`	`your lack of planning is not my emergency`
`19:50.01`	`bmoraca_work`	`he also asked me to download goodsync and email it to him. not able to download it himself?`
`19:50.13`	`Naikrovek`	`your lack of foresight is not my emergency`
`19:50.30`	`Chibat`	`restarted asterisk no go, now i'm going to restart the whole box`
`19:50.34`	`Naikrovek`	`your lack of planning\|execution\|listening is not my emergency`
`19:50.45`	`bmoraca_work`	`Naikrovek: EVERYTHING is my emergency`
`19:50.47`	`bmoraca_work`	`lol`
`19:50.48`	`p3nguin`	`He couldn't download it, but he was going to be able to gasp download it from his email account?`
`19:51.13`	`Naikrovek`	`i email my supervisor all the time about things that are going to happen if we don't do X, Y, or Z. X, Y, or Z happen all the time, and I told them it would happen, and it's always my emergency to fix`
`19:51.14`	`spenguin[work]`	`in anycase which email provider supports that large an attachment`
`19:51.28`	`*** join/#asterisk docelmo (~docelmo@147.177.249.119)`
`19:51.40`	`Naikrovek`	`"Naikrovek. call the phone company. ask them if they can get us a couple new T1s this week. They'll do it if you ask nice." R O F L`
`19:51.42`	`Chibat`	`haha its called 1TB.com`
`19:51.43`	`Naikrovek`	`no they will not`
`19:51.47`	`bmoraca_work`	`Naikrovek: i'm expected to be on call 24/7, too. yet when i try to call my boss 20 minutes after the shop closes about a service affecting issue, he never answers`
`19:51.48`	`docelmo`	`Say can anyone tell me what Version of Dadhi is compatible with Asterisk Biz edition?`
`19:51.56`	`Naikrovek`	`of course`
`19:52.31`	`Naikrovek`	`docelmo: the version that comes with asterisk biz edition i would guess. digium would be who i'd contact about that`
`19:52.49`	`Naikrovek`	`bmoraca_work: my mangement are kings of double standards`
`19:52.51`	`docelmo`	`eh.. figured I would ask..`
`19:52.58`	`docelmo`	`thanks..`
`19:53.02`	`Naikrovek`	`docelmo: excellent inst... nevermind`
`19:54.05`	`Naikrovek`	`i hate when people do that`
`19:54.06`	`wcselby`	`lol`
`19:54.16`	`*** join/#asterisk xuser (~xuser@unaffiliated/xuser)`
`19:54.18`	`Naikrovek`	`unhappy with answer? I'm out!`
`19:54.23`	`Naikrovek`	`wtf kind of response is that`
`19:54.32`	`Naikrovek`	`stick around... maybe i have more to say...`
`19:54.43`	`p3nguin`	`IRC's version of storming away and slamming a door.`
`19:54.52`	`Naikrovek`	`yeah i know but come on`
`19:55.29`	`p3nguin`	`I know today isn't your first time encountering an ingrate.`
`19:55.34`	`Naikrovek`	`no.`
`19:55.43`	`Naikrovek`	`but it still itches where i can't scratch when people do that`
`19:55.57`	`Naikrovek`	`bmoraca_work: hear my tale of woe and enjoy your employer a bit more, perhaps.`
`19:56.28`	`Naikrovek`	`bmoraca_work: our core network consists of several unmanaged linksys switches`
`19:56.39`	`Naikrovek`	`bmoraca_work: no backup solution, 2tb or more of customer data`
`19:56.44`	`p3nguin`	`yikes`
`19:56.45`	`Naikrovek`	`bmoraca_work: saturated single t1`
`19:57.01`	`Naikrovek`	`bmoraca_work: no vlans (see unmanaged switches)`
`19:57.28`	`Naikrovek`	`bmoraca_work: disks are at about 95% utilization, with no plans (other than my own) to upgrade`
`19:57.45`	`wcselby`	`Naikrovek - hope you have your resume up to date.... :)`
`19:57.52`	`Naikrovek`	`hehe`
`19:57.53`	`Naikrovek`	`i do`
`19:58.10`	`Naikrovek`	`but i asked my bro-in-law, who is a master of obtaining IT funds from his employer, how to do it`
`19:58.10`	byteme`	`Naikrovek: small company?`
`19:58.19`	`Naikrovek`	byteme`: small, but lots of money flowing through here
`19:58.31`	byteme`	`Naikrovek: I'm in the same boat pretty much`
`19:58.38`	byteme`	`but I enjoy working for startups`
`19:58.52`	byteme`	`I guess I like pain or something`
`19:58.53`	`Naikrovek`	byteme`: the $40k i need to upgrade core infrastructure is chump change when compared to a single project budget forecast. and we have 6 going at once
`19:59.14`	`Naikrovek`	`downstairs (engine testing facility) they spend $100k on upgrades per month.`
`19:59.18`	byteme`	`we have a single Dell 2950 running over 150 Concurrent postgres databases`
`19:59.28`	`Naikrovek`	`i spent $5k last year and got a good talking to because of it`
`19:59.35`	`Naikrovek`	byteme`: whoa
`19:59.39`	byteme`	`the I/O is starved the SQL queries perform like crap`
`19:59.46`	byteme`	`+and`
`19:59.49`	`Naikrovek`	`yeah i have several IO problems as well`
`20:00.00`	byteme`	`if the server ever crashes, we are so ... @#$%^@#$^ screwed`
`20:00.21`	byteme`	`the DBAs don't even have log shipping configured to a backup server.`
`20:00.29`	`Naikrovek`	`i've been talking about tacking IT upgrades onto engine test cell improvement budget.. they wouldn't even notice, I don't think`
`20:00.31`	`Naikrovek`	`ew`
`20:00.33`	`Naikrovek`	`ewewewewew`
`20:00.46`	`Naikrovek`	byteme`: they're going to get bytten in the arse soon
`20:01.00`	`bmoraca_work`	`Naikrovek: i just had to deploy a second PBX at one site with overlapping extension ranges because of management's lack of listening to me...not pretty`
`20:01.06`	byteme`	`but you know what? I don't care because as long as the checks cash every two weeks I don't give a rip. All I can show them the capacity planning documents I've created`
`20:01.40`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`20:01.44`	`Naikrovek`	`i need to see some good upgrade proposal documents so i can scare them good`
`20:01.54`	`Naikrovek`	`bmoraca_work: overlapping extensions? wtf?`
`20:01.56`	byteme`	`the thing that doesn't make sense either is I get paid pretty well there, but they can't seem to find 5k to buy some extra disk/servers`
`20:02.15`	`Naikrovek`	byteme`: they can, they just don't see the point. upon failure, they will see the light.
`20:02.22`	`*** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68)`
`20:02.43`	`wcselby`	`i've worked at places like that`
`20:02.45`	byteme`	`Naikrovek: DBAs were sending emails last night saying the entire database servers has 350MB of space left on a raid 10 3 TB array.`
`20:02.50`	`Naikrovek`	`everyone sees the light when something fails`
`20:02.58`	`wcselby`	`the owner once told me "i don't see the point of warranties, I thnk they're just designed to take your money"`
`20:03.10`	`Naikrovek`	`then holy F*** THIS IS PRIORITY NUMBER ONE (just like you told them)`
`20:03.12`	`wcselby`	`and would never let me order a warranty on any piece of hardware I ever bought`
`20:03.23`	`Naikrovek`	byteme`: whoa
`20:03.47`	`bmoraca_work`	`Naikrovek: the PBX is more than sufficient. the WAN links between the sites are not. two alternatives: add a second PBX in location2 ($5k or so expense) or move the single PBX to a central location...of course, they don't want to be down and they want to test it, so i had to deploy the PBX in the central location and give them the ability to 4-digit dial between the two PBXes. but, they don't want to test all phones, so I've got`
`20:03.47`	`bmoraca_work`	`phones in the middle of their extension range on the second PBX.`
`20:04.12`	`Naikrovek`	`bmoraca_work: ugh.`
`20:04.16`	`*** join/#asterisk tessier_ (~treed@kernel-panic/copilotco)`
`20:04.25`	`bmoraca_work`	`Naikrovek: double the work, double the fun!`
`20:04.56`	`Naikrovek`	`if some work is good, more work must be great, right?`
`20:04.59`	`Naikrovek`	`wait...`
`20:05.30`	byteme`	`Naikrovek: I've stopped caring for the most part after I let management know the situation. When they get pissed, I just refer to a printed physical copy of the email I sent them months ago alerting them to a potential disaster`
`20:05.42`	byteme`	`that shuts them up pretty fast`
`20:05.44`	`Naikrovek`	`i am seriously considering tacking IT needs onto projects.`
`20:05.59`	`Naikrovek`	`yeah that would shut them up fast.`
`20:06.02`	`Naikrovek`	`i need to do that`
`20:06.37`	`Qwell`	`looks at jonmasters and boggles at IRC drama`
`20:06.40`	`*** join/#asterisk voipnet-tech (~russ@rrcs-24-39-187-146.nys.biz.rr.com)`
`20:07.08`	byteme`	`heh :)`
`20:07.44`	`jonmasters`	`Qwell: huh?`
`20:07.51`	`Qwell`	`jonmasters: jbj..`
`20:08.12`	`jonmasters`	`Qwell: I dunno, I've nothing against him personally`
`20:08.14`	`*** join/#asterisk meesterarend (~frans@vc-41-26-81-242.umts.vodacom.co.za)`
`20:09.02`	`jonmasters`	`Qwell: but I don't want to bother him if he prefers I hang out with the RH rpm folks :)`
`20:09.05`	`Qwell`	`jonmasters: because you work for the evil corporation, apparently :p silly`
`20:09.19`	`jonmasters`	`Qwell: etc. Anyway, I don't really care :)`
`20:10.44`	`jonmasters`	`hopes his second 7940G turns up today`
`20:11.11`	`p3nguin`	`I can understand that. I prefer talking on two phones at once.`
`20:11.43`	`jonmasters`	`:)`
`20:12.27`	`Qwell`	`jonmasters: good to know you're here here. I may try to bug you some time for kmod stuff :p`
`20:12.39`	`jonmasters`	`Qwell: sure thing`
`20:12.43`	`Naikrovek`	`doesn't know who jonmasters is`
`20:12.46`	`Qwell`	`s/here./though./`
`20:13.03`	`jonmasters`	`Jon Masters (jcm@redhat.com`
`20:13.10`	`Naikrovek`	`ah redhat guy`
`20:13.14`	`voipnet-tech`	`hello everyone- i just noticed my Asterisk 1.6.2.0 box that my Full log file is getting spammed frequently with the same message: [Apr 7 16:05:31] WARNING[703] udptl.c: No UDPTL ports remaining`
`20:13.16`	`Naikrovek`	`i turned down a redhat job once :)`
`20:13.24`	`jonmasters`	`I'm also a huge Asterisk fan`
`20:13.32`	`Naikrovek`	`as evidenced by your presence`
`20:13.34`	`Naikrovek`	`nice`
`20:13.49`	`jonmasters`	`All my personal telephony is Asterisk - DIDs in the US/UK with lots of ATAs and IP phones ;)`
`20:14.13`	byteme`	`jonmasters: what ATAs do you have?`
`20:14.27`	`jonmasters`	`when I moved to the US it let me do nice things like keep my UK cellphone number and have calls to that re-route to my US cellphone using two DIDs and some fancy rules`
`20:14.45`	`Defraz`	`SO I have my Asterisk server and a PRI and it is working just dandy for me, I have another guy upstairs that wants to tie his allworx box into mine for outbound calling and inbound.`
`20:14.57`	`jonmasters`	byteme`: mostly PAP2, I just got a locked one after a bad eBay listing. I had to "fix" it...PAPv2 is quite painful.
`20:15.14`	`Defraz`	`I ahve the inbound working but he can't seem to make calls, it keeps making the called DID the Caller ID for some reason.`
`20:15.18`	byteme`	`jonmasters: I've got a PAP2 that I can't unlock :(`
`20:15.29`	byteme`	`PAPv2`
`20:15.40`	`*** join/#asterisk Circlefusion (~circlefus@adsl-80-74-57.sdf.bellsouth.net)`
`20:16.05`	`jonmasters`	byteme`: it's tricky. And even if you do it it's unreliable with the provisioning "fix".
`20:16.08`	`*** join/#asterisk mhaddog (~mhaddog@173-122-96-45.pools.spcsdns.net)`
`20:16.12`	`jonmasters`	byteme`: better use it as a doorstop
`20:16.28`	`Chibat`	`Restarting did it or clearing out the my db i have set up with it.`
`20:17.01`	`jonmasters`	byteme`: I recommend a real PAP2-NA or the T model, or a real IP Phone. I love the Cisco phones.
`20:17.13`	byteme`	`jonmasters: yeah`
`20:17.28`	`jonmasters`	`took out a SMARTnet contract to aid with that. Worth the hassle.`
`20:17.35`	byteme`	`:(`
`20:18.07`	`jonmasters`	`Anyway, bbl. Nice to talk. Qwell, I'm on about 160 channels here and elsewhere, so pretty contactable ;)`
`20:18.57`	byteme`	`160?!?!?`
`20:19.29`	`[TK]D-Fender`	`Polycom > All`
`20:19.36`	`Naikrovek`	`agreed`
`20:19.41`	`Skeeter-`	`+1`
`20:20.05`	`jonmasters`	byteme`: I have some custom plugins I wrote that do IRC->SMS text gateway, email, twitter, etc. I mostly respond to pings.
`20:20.22`	`jonmasters`	`-> back later`
`20:21.27`	byteme`	`jonmasters: that is really cool :) take care`
`20:21.41`	`wcselby`	`lol you just pinged him back into the channel lol`
`20:21.46`	`wcselby`	`ah well`
`20:21.49`	`wcselby`	`think it's that time for today`
`20:21.59`	`wcselby`	`o/ all`
`20:23.13`	`*** join/#asterisk jmacz (~jmacz@190.144.102.122)`
`20:28.51`	`[TK]D-Fender`	`checkout time, heading home. Later all`
`20:30.45`	`ariel_`	`hardwire: the adding Ww to the dial string did not help at all.`
`20:31.38`	`hardwire`	`media path wasn't the problem then :(`
`20:31.41`	`*** part/#asterisk NMR_1122 (~rahl@adsl-068-209-105-089.sip.mia.bellsouth.net)`
`20:31.44`	`p3nguin`	`Doesn't that allow both the caller and the callee to start automon?`
`20:31.53`	`ariel_`	`[Apr 7 12:28:05] WARNING[17538]: chan_iax2.c:3181 __attempt_transmit: Max retries exceeded to host 172.16.4.234 on IAX2/shiptrunk-16922 (type = 6, subclass = 1, ts=9, seqno=0)`
`20:32.00`	`ariel_`	`but at least I have something different now`
`20:32.29`	`ariel_`	`p3nguin: yes`
`20:32.52`	`p3nguin`	`I missed the initial complaint.`
`20:32.59`	`p3nguin`	`Are you unable to record?`
`20:33.11`	`ariel_`	`I am trying to send calls from one asterisk box to another over iax2 but box B shows nothing on the cli even with debug on`
`20:33.14`	`hardwire`	`I was having him test media involvement`
`20:33.19`	`hardwire`	`locking the media to asterisk`
`20:33.57`	`p3nguin`	`What are you using to "send calls from one asterisk box to another"?`
`20:34.02`	`ariel_`	`I switched it to sip with same almost settings and it works via sip`
`20:34.56`	`ariel_`	`p3nguin: it's 2 asterisk boxes being used as gateways, Box A is connected to a pbx, and box B is connected to 4 E1's`
`20:35.23`	`p3nguin`	`irrelevant`
`20:35.38`	`ariel_`	`well just trying to set them up to use iax2 trunking`
`20:35.42`	`ariel_`	`so save b/w`
`20:35.55`	`p3nguin`	`I was expecting an answer such as, "I use a basic Dial() command to send calls from one to the other."`
`20:36.30`	`hardwire`	`He was use a basic Dial() command to send calls from one to the other.`
`20:36.31`	`hardwire`	`:P`
`20:37.01`	`ariel_`	`exten => _1NXXNXXXXX,n,Dial(IAX2/${EXTEN}@shiptrunk,60,t)`
`20:37.02`	`p3nguin`	`And it fails when using IAX2, but works with SIP.`
`20:37.06`	`ariel_`	`that is the actual command`
`20:37.14`	`ariel_`	`yes`
`20:37.24`	`*** join/#asterisk CraigW76 (~techcaw@addr33.mimc.com)`
`20:37.25`	`p3nguin`	`Dial(IAX2/shiptrunk/${EXTEN})`
`20:37.47`	`ariel_`	`done it that way as well,`
`20:38.06`	`p3nguin`	`And there's no sign of the call reaching B?`
`20:38.28`	`p3nguin`	`I would be interested in the iax2 peer definition.`
`20:38.33`	`ariel_`	`issue is not sending the call out of box A, it's that Box B does not even give a rejected, or any display that it does not like the call. Nothing even with core set debug 99 and iax2 debug on`
`20:39.42`	`hardwire`	`ariel_: I forgot to ask if this is diffrent for any other call using the same trunk.`
`20:40.02`	`ariel_`	`pastebin.ca seems down going to use pastebin.com`
`20:40.14`	`p3nguin`	`.ca always gives me trouble.`
`20:40.27`	`ariel_`	`hardwire: it's different and it's not. Different gateways and trunks`
`20:40.38`	`hardwire`	`I'm a huge nut for pastey.net`
`20:41.01`	`hardwire`	`mostly because of the handle.pastey.net ability`
`20:41.19`	`ariel_`	`http://pastebin.com/sZEnTHqw`
`20:41.59`	`hardwire`	`ariel_: did you try transfer=no as a diagnostic?`
`20:42.01`	`ariel_`	`I will use that next time`
`20:42.19`	`ariel_`	`yes but will try it again`
`20:43.23`	`hardwire`	`btw`
`20:43.49`	`hardwire`	`http://pastebin.com/tFzMjcu9`
`20:44.41`	`hardwire`	`http://pastebin.com/diff.php?i=tFzMjcu9`
`20:44.42`	`*** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net)`
`20:44.42`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`20:44.44`	`hardwire`	`neat.. you can diff`
`20:45.57`	`ariel_`	`yes I also had tried type=friend`
`20:46.02`	`ariel_`	`same diff`
`20:46.18`	`hardwire`	`ok`
`20:46.23`	`hardwire`	`wraps up lunch`
`20:46.31`	`hardwire`	`half a pizza..`
`20:46.34`	`hardwire`	`sigh..`
`20:46.38`	`hardwire`	`burp`
`20:46.43`	`ariel_`	`oh I had pizza today as well`
`20:47.10`	`hardwire`	`hmm.. make that 5/8ths of a pizza`
`20:47.17`	`ariel_`	`but that was almost 5 hours ago`
`20:47.29`	`ariel_`	`just 2 slices here`
`20:47.32`	`hardwire`	`cause you're on the wrong side of the lpanet`
`20:47.34`	`hardwire`	`planet`
`20:47.46`	`ariel_`	`not wrong side`
`20:48.05`	`ariel_`	`in any case I will be around your area from May 10th to 15th`
`20:48.34`	`ariel_`	`Seattle, Juenu, Kachica or how ever you spell them`
`20:48.43`	`hardwire`	`kakikahchu`
`20:48.54`	`hardwire`	`are you going to Homer, AK or Seward, AK at all?`
`20:49.04`	`ariel_`	`no`
`20:49.08`	`hardwire`	`:(`
`20:49.10`	`ariel_`	`going through the path`
`20:49.25`	`ariel_`	`one day at Victoria`
`20:49.35`	`hardwire`	`That would be nice`
`20:49.57`	`ariel_`	`but only have to work on 2 ships on this trip`
`20:50.11`	`hardwire`	`specialty sushi chef.. right?`
`20:50.17`	`ariel_`	`lol`
`20:52.13`	`*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)`
`20:53.11`	`ariel_`	`wow just finally got some info on the box B but it really does not tell much: http://pastebin.com/aDajCDbU`
`20:53.34`	`*** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110)`
`20:53.45`	`hardwire`	`what does /var/log/asterisk/debug say?`
`20:56.30`	`ariel_`	`no debug log`
`20:56.39`	`ariel_`	`ls`
`20:58.31`	`*** join/#asterisk rdircio (~admin@201.137.57.68)`
`20:58.35`	`rdircio`	`chan_sip.c: Call from '1001' to extension '958458908' rejected because extension not found.`
`20:58.37`	`rdircio`	`grrrrr`
`20:58.59`	`rdircio`	`it is not an extension and should go thru a dahdi line :(`
`20:59.17`	`ariel_`	`you have a dial rule for that?`
`20:59.27`	`rdircio`	`9\|.`
`20:59.54`	`rdircio`	`a "route" in freepbx`
`20:59.58`	`ariel_`	`9\|. = evil`
`21:00.11`	`rdircio`	`9\|X. ?`
`21:00.22`	`ariel_`	`where do you have that rule`
`21:00.41`	`ariel_`	`in any case that number is missing digits`
`21:00.51`	`rdircio`	`?`
`21:00.58`	`rdircio`	`it's a valid number here`
`21:01.06`	`ariel_`	`9\|1NXXNXXXXX`
`21:01.21`	`rdircio`	`why should i want a 1?`
`21:01.24`	`ariel_`	`9 digits`
`21:01.30`	`rdircio`	`nope`
`21:01.35`	`rdircio`	`here in MX`
`21:01.49`	`Naikrovek`	`anyone in here know anything about ibm bladecenter network modules`
`21:01.55`	`rdircio`	`we can use 8 digits`
`21:02.02`	`p3nguin`	`<rdircio> it is not an extension and should go thru a dahdi line :( <-- Not an extension? Why are you dialing it if it isn't an extension?`
`21:02.16`	`*** join/#asterisk zomenox (~brock@254120.firewall.winthrop.edu)`
`21:02.22`	`rdircio`	`from a sip client`
`21:02.29`	`*** part/#asterisk zomenox (~brock@254120.firewall.winthrop.edu)`
`21:02.37`	`rdircio`	`i want all numbers that start with 9 to go to the dahdi channel 1`
`21:02.55`	`p3nguin`	`thinks someone forgot to read The Book and understand that numbers dialed from phones need matching extensions.`
`21:03.01`	`ariel_`	`you need to setup the correct routes in the outbound routes then in your turnk setting`
`21:03.11`	`ariel_`	`you should see the folks on channel #freepbx`
`21:03.31`	`p3nguin`	`First, you need an extension. Then the extension needs to "route" based on the peer in the Dial() command.`
`21:03.31`	`rdircio`	`oks, you mean the sip trunk or the zap trunk?`
`21:03.55`	`ariel_`	`how do you want to send the call out?`
`21:03.59`	`rdircio`	`i got a zap trunk without dial rules`
`21:04.13`	`ariel_`	`trunks are not direct outbound until you set that up in the freebpx gui`
`21:04.34`	`ariel_`	`what is your outbound route setup as?`
`21:05.28`	`rdircio`	`just route name "rdircio" with dial pattern "9\|." trunk sequence 0 goes to "ZAP/1"`
`21:05.33`	`p3nguin`	`WTF are you asking about FreePBX here, anyway?`
`21:05.35`	`rdircio`	`which is in dahdi compat mode`
`21:05.50`	`rdircio`	`well i thought it was a generic error`
`21:05.56`	`rdircio`	`chan_sip.c: Call from '1001' to extension '958458908' rejected because extension not found.`
`21:05.59`	`p3nguin`	`It probably is.`
`21:06.05`	`p3nguin`	`It's clear.`
`21:06.23`	`p3nguin`	`It says there is no matching extension for '958458908' within the context where your device is dialing.`
`21:06.26`	`p3nguin`	`That's pure asterisk.`
`21:06.29`	`rdircio`	`somehow the conversation came to freepbx, but i want to try fixing it by hand`
`21:06.31`	`rdircio`	`yea`
`21:06.36`	`*** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net)`
`21:06.39`	`rdircio`	`pure asterisk...`
`21:06.42`	`p3nguin`	`So add the friggin' extension.`
`21:06.48`	`rdircio`	`nah, why?`
`21:06.52`	`ariel_`	`he needs to set that up in the gui`
`21:06.52`	`rdircio`	`it's an external #`
`21:06.53`	`[TK]D-Fender`	`rdYou do not fix dialplan errors like that by hand. Your outbound routes are done wrong`
`21:07.02`	`p3nguin`	`Are you stupid?`
`21:07.03`	`rdircio`	`it's not an extension!`
`21:07.15`	`p3nguin`	`There we have it. You are stupid.`
`21:07.35`	`rdircio`	`that will not solve much`
`21:07.37`	`rdircio`	`:)`
`21:07.38`	`ariel_`	`no he does not edit the dial rules directly he needs to set them up via the gui`
`21:08.06`	`ariel_`	`rdircio: step one, go to #freepbx, step 2 ask for help with setting your outbound rules`
`21:08.13`	`DerkKo`	`Im using mixmonitor to record calls. But i want to record the calls and the recording be on 2 different audio channels channels... (Ie i want the caller to be on the stereo left and the callee on the rigth) How can i accomplish this?`
`21:08.16`	`rdircio`	`will do, thx`
`21:08.23`	`p3nguin`	`and 3, learn asterisk.`
`21:08.33`	`rdircio`	`that's what i'm trying to do here :)`
`21:08.55`	`ariel_`	`rdircio: yes but the gui is blocking your from direct learning rules`
`21:08.55`	`p3nguin`	`You think the number dialed doesn't need to match an extension. You've got a long way to go.`
`21:09.15`	`rdircio`	`somehow i made it worked yesterday`
`21:09.22`	`rdircio`	`without defining an extension`
`21:09.23`	`p3nguin`	`luck`
`21:09.27`	`rdircio`	`correct`
`21:09.31`	`ariel_`	`you do not need an extension`
`21:09.36`	`p3nguin`	`You clearly don't even kno what an extension is.`
`21:09.36`	`ariel_`	`you need to set your outbound rule`
`21:09.53`	`p3nguin`	`You probably thing phones are extensions or something equally as ignorant.`
`21:09.54`	`rdircio`	`and i could dial anything starting with 9 and went to the pots line`
`21:10.18`	`ariel_`	`hint, go to #freepbx`
`21:10.23`	`rdircio`	`will do, thx`
`21:10.43`	`ariel_`	`there more able to help then want you to learn actually asterisk dial plan and context`
`21:11.59`	`p3nguin`	`Another hint: if there is no extension that matches the number dialed, nothing can happen. Extensions are rules that determine how calls are handled.`
`21:13.06`	`ariel_`	`extensions are not rules`
`21:13.12`	`p3nguin`	`yes they are.`
`21:13.18`	`ariel_`	`there a dial plan`
`21:13.28`	`p3nguin`	`Anyone that knows anything about Asterisk would know this.`
`21:13.32`	`*** join/#asterisk githogori (githogori@57.sub-75-229-111.myvzw.com)`
`21:13.33`	`ariel_`	`hehe`
`21:14.08`	`rdircio`	`oh well`
`21:14.18`	`rdircio`	`thx guys`
`21:15.09`	`p3nguin`	`An extension says what happens to a call when the dialed characters match. This is the textbook definition of "rule."`
`21:15.21`	`ariel_`	`exten =`
`21:16.08`	`p3nguin`	`Was there more to that?`
`21:16.41`	`ariel_`	`yes just got interrupted by a person here...`
`21:16.58`	`p3nguin`	`Tell him/her to take a number. :)`
`21:17.05`	`ariel_`	`lol`
`21:17.11`	`*** join/#asterisk ibercom (~96d6cd0f@gateway/web/freenode/x-lnbkgiamujyvfkkb)`
`21:17.30`	`p3nguin`	`Put up one of those red number tag machines like at the customer service desk.`
`21:17.48`	`ariel_`	`I understand your term, but I only use that term for final end point setting as you can have allot of things working with the exten=> command`
`21:18.11`	`p3nguin`	`All exten => lines in extensions.conf are rules.`
`21:18.18`	`ariel_`	`I call a did number a did, not and extension`
`21:18.37`	`p3nguin`	`You call the number, asterisk matches it to an extension. That's how it works.`
`21:18.48`	`ariel_`	`yes there rules, but they don't need to be in the extensions.conf file,`
`21:18.51`	`p3nguin`	`Without the extension match, nothing happens.`
`21:19.04`	`ariel_`	`ok what ever you want to call it`
`21:19.14`	`p3nguin`	`You could put them in other files, as long as you include those files appropriately.`
`21:19.25`	`ariel_`	`yes`
`21:19.34`	`p3nguin`	`The parent file is still extensions.conf, though.`
`21:20.09`	`ariel_`	`wonders....`
`21:20.13`	`DerkKo`	`can i execute the monitor application from the cli ? ie cool-ast# asterisk -rx "monitor start SIP/10.20.2.198-00000000 test"`
`21:20.14`	`p3nguin`	`Or you could rewrite pbx_config and rename the file where the extensions are kept.`
`21:25.04`	`p3nguin`	`That gives me an idea.`
`21:27.28`	`p3nguin`	`There aren't that many spots in pbx/pbx_config.c that would need to be changed.`
`21:30.19`	`ibercom`	`Hello, i haven't astribank hardware, but when i run "/etc/init.d/dahdi start" i get the message "Missing astribank_is_starting". Is this a bug ? I use dahdi-tools-2.2.1.`
`21:31.02`	`p3nguin`	`What kind of hardware do you have that you are going to use dahdi?`
`21:31.05`	`*** join/#asterisk endigo-ceo (~5712d112@gateway/web/freenode/x-nsvgnwaxjdxsbxme)`
`21:31.05`	`Qwell`	`ibercom: neither of those are errors`
`21:31.59`	`*** join/#asterisk githogori (githogori@75.229.7.255)`
`21:32.28`	`ibercom`	`I have digium card, but with dahdi-tools-2.2.0 i start OK, no error messages.`
`21:32.40`	`p3nguin`	`Then use 2.2.0`
`21:32.44`	`p3nguin`	`Problem solved.`
`21:32.48`	`p3nguin`	`Next!`
`21:33.30`	`ibercom`	`Yes, That's the easy solution.`
`21:34.51`	`*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)`
`21:36.11`	`ibercom`	`<Qwell>This can be a cosmetic bug.`
`21:36.23`	`p3nguin`	`Hmm. I don't remember him saying that.`
`21:38.54`	`ibercom`	`He says 'no error'.`
`21:39.21`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`21:40.41`	`*** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-fkjjkqunjtrxzrdv)`
`21:41.26`	`*** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380)`
`21:44.14`	`*** join/#asterisk jtodd (vlv96qfsmj@asterisk/community-director-and-tie-dye-shirt-lover/jtodd)`
`21:44.14`	`*** mode/#asterisk [+o jtodd] by ChanServ`
`21:46.35`	`aceio`	`hi`
`21:46.52`	`*** join/#asterisk xuser (~xuser@unaffiliated/xuser)`
`21:47.46`	`paulc`	`Anyone know of a prepaid cell provider that will work in Canada (primarily for texting) (without getting shafted on the rates)?`
`21:49.03`	`paulc`	`sorry - a US based prepaid cell service provider*`
`21:50.27`	`*** join/#asterisk Circlefusion (~circlefus@adsl-80-74-57.sdf.bellsouth.net)`
`21:51.29`	`*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)`
`21:52.56`	`*** join/#asterisk yg (~yg@200.152.178.153)`
`21:53.00`	`yg`	`hi`
`21:53.06`	`*** join/#asterisk megalomano (~klonstein@38.124.169.126)`
`21:53.43`	`megalomano`	`hi people`
`21:55.04`	`megalomano`	`one question , how can I put the codecs ( specialy ) g729 , to amd architecture`
`21:56.20`	`megalomano`	`any sugestion`
`21:56.23`	`megalomano`	`?`
`21:57.31`	`aceio`	`i am still seening duplicate ip at sip show peers`
`21:59.11`	`*** part/#asterisk ibercom (~96d6cd0f@gateway/web/freenode/x-lnbkgiamujyvfkkb)`
`21:59.15`	`aceio`	`http://pastebin.com/WE0ceHif`
`22:00.01`	`*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)`
`22:00.27`	`p3nguin`	`megalomano: There is a g.729 for AMD.`
`22:00.43`	`ariel_`	`megalomano: 2 ways, one you buy the board from digium or you get the correct license from them for your system.`
`22:00.49`	`aceio`	`[TK]D-Fender: hi`
`22:01.12`	`ariel_`	`p3nguin: yes it's a board which works really well.`
`22:01.28`	`p3nguin`	`?`
`22:01.34`	`p3nguin`	`WTF are you talking about?`
`22:01.45`	`ariel_`	`megalomano: ask about g729`
`22:01.48`	`*** part/#asterisk rdircio (~admin@201.137.57.68)`
`22:02.16`	`*** join/#asterisk _penfold1972_ (~penfold19@66-194-25-11.static.twtelecom.net)`
`22:02.17`	`p3nguin`	`megalomano asked if there was g.729 for AMD, and I answered appropriately.`
`22:02.31`	`*** join/#asterisk ido- (~wtf@109.160.210.212)`
`22:02.59`	`ariel_`	`so did I`
`22:03.06`	`p3nguin`	`Not really.`
`22:03.07`	`_penfold1972_`	`Hi all! Can someone help me understand what this debug message means:`
`22:03.07`	`_penfold1972_`	`[Apr 7 17:23:20] DEBUG[6157] chan_sip.c: Skipping packet - Bad request protocol Queue-5804"@192.168.231.52 SIP/2.0`
`22:03.43`	`ido-`	`Hrm, can any please help me out, I've added a trunk to my asterisk, and this is what I'm getting in my log: http://pastebin.org/140028`
`22:03.45`	`p3nguin`	`The codec is available for Athlons, Opterons, and who knows what other AMDs.`
`22:04.05`	`*** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net)`
`22:04.05`	`ChannelZ`	`_penfold1972_: looks like you have a stray " character somewhere perhaps`
`22:04.06`	`p3nguin`	`It has nothing to do with a "board" as you put it.`
`22:05.00`	`aceio`	`i am seening duplicate ip at sip show peers`
`22:05.04`	`_penfold1972_`	`ChannelZ, that is what I was wondering... Just wanted to make sure I was going in the right direction`
`22:05.11`	`_penfold1972_`	`thanks`
`22:05.47`	`aceio`	`can anyone help understand why`
`22:07.17`	`*** join/#asterisk h1kz0r (~h1kz0r@c-68-44-81-246.hsd1.nj.comcast.net)`
`22:08.27`	`ChannelZ`	`Your SIP username or something, not sure where it's coming from`
`22:08.43`	`*** join/#asterisk Fubard (~brawr@173.10.235.205)`
`22:09.03`	`ChannelZ`	`aceio: as per last night, you somehow have two devices as the same IP or more likely a single device which registers multiple times, for separate lines for instance`
`22:10.53`	`ido-`	`anyone looked at my pastebin ?`
`22:11.54`	`aceio`	`ChannelZ: have a looking at this http://pastebin.com/WE0ceHif`
`22:12.20`	`aceio`	`ChannelZ: please it's doing my head in`
`22:12.43`	`[TK]D-Fender`	`aceio: pastebin "sip show peer 1261" <- same for 1262`
`22:13.30`	`megalomano`	`ariel_: I'm require a codec g729 to my asterisk box`
`22:13.56`	`ChannelZ`	`aceio: You tell me - look in whatever device is at that IP and see where 1261 and 1260 are coming from`
`22:14.07`	`ChannelZ`	`megalomano: http://digium.com`
`22:15.58`	`aceio`	`[TK]D-Fender: 1 sec`
`22:17.31`	`megalomano`	`ChannelZ: its necesary buy a digium card?`
`22:17.58`	`*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)`
`22:18.30`	`megalomano`	`ChannelZ: I'm refer by the codec`
`22:19.37`	`aceio`	`[TK]D-Fender: http://pastebin.com/rhr3JcPg`
`22:19.48`	`ChannelZ`	`megalomano: You can buy software codecs for $10/ea channel`
`22:20.11`	`aceio`	`[TK]D-Fender: i ran sip show peers`
`22:22.39`	`aceio`	`[TK]D-Fender: and now i am getting dup ip http://pastebin.com/tKdGfJ5L`
`22:23.24`	`*** join/#asterisk ido- (~wtf@94.159.203.234)`
`22:23.28`	`aceio`	`it seems too have shifted onto anther sip peer`
`22:23.32`	`ido-`	`hrm`
`22:23.43`	`ChannelZ`	`aceio: is NAT involved between * and the devices?`
`22:24.06`	`ariel_`	`megalomano: there is 2 ways to get this, one is via software for each channel which ChannelZ posted the link, And there is also a DSP board sold by Digium with license for g729 on them. Sometimes it less expensive via the board if you need the full 120 channels`
`22:24.14`	`aceio`	`ChannelZ: no natting at all`
`22:25.14`	`ChannelZ`	`Are the devices getting their IP via DHCP?`
`22:25.43`	`aceio`	`ChannelZ: DHCP works fine`
`22:26.08`	`ChannelZ`	`Is the lease expire ridiculously low or something?`
`22:26.20`	`ariel_`	`exit`
`22:26.38`	`ChannelZ`	`Either your devices are batshit crazy or something else is causing them to switch IPs all the time..`
`22:26.49`	`ChannelZ`	`from the looks of it anyway`
`22:27.52`	`talntid`	`mmhmmm`
`22:28.40`	`aceio`	`[TK]D-Fender: http://pastebin.com/xytzqgX8`
`22:30.21`	`aceio`	`ChannelZ: the default lease time is default-lease-time 3600;`
`22:31.00`	`aceio`	`i just can't see why`
`22:32.31`	`thehar`	`anyone familar with the adtran 904?`
`22:32.58`	`thehar`	`curious if you can take say dsl into the ethernet for connectivity and output dsx`
`22:34.03`	`ChannelZ`	`eh?`
`22:34.28`	`h1kz0r`	`heyyo`
`22:34.58`	`ChannelZ`	`aceio: well I don't know why your phones are switching IPs then and I couldn't tell from here`
`22:35.01`	`*** join/#asterisk hluesea (~hluesea@88.247.127.66)`
`22:35.11`	`thehar`	`guess not :)`
`22:35.19`	`ChannelZ`	`aceio: you have 5 physical phones?`
`22:35.34`	`aceio`	`ChannelZ: correct`
`22:35.45`	`ChannelZ`	`thehar: well I'm not quite sure I know what you're asking..`
`22:36.17`	`thehar`	`with an adtran 904.. there is one pri out and pri in.. and ethernet`
`22:36.30`	`ChannelZ`	`aceio: and if you ask the devices themselves, are their IPs show up the same way Asterisk is showing? Do two really in fact have the same IP at any given moment? Or are they staying consistent over the course of 5 minutes say?`
`22:36.32`	`thehar`	`curious if you can take say.. a dsl connection into it.. and SIP trunks on it. and output a 24 chan pri`
`22:36.58`	`h1kz0r`	`is it normal i have no sip.conf, sip_additional.conf or any sip commands by defaut? not even sample files... (v1.4)`
`22:37.17`	`h1kz0r`	`they're referenced in the book and i'm gonna be using them so just wanted to make sure i'm not missing anything`
`22:38.10`	`ChannelZ`	`h1kz0r: did you "install config" ? The sample confs are put in by default with just 'make install'`
`22:38.43`	`h1kz0r`	`i installed asterisknow, didn't compile from scratch`
`22:38.58`	`ChannelZ`	`SORRY 'make samples' not config`
`22:38.59`	`h1kz0r`	`i have plenty of config files in there and samples, everything except the sip files`
`22:39.03`	`ChannelZ`	`config installs the init script`
`22:39.22`	`aceio`	`sip 1261 has 10.192.30.81 and sip 1263 has dot 80`
`22:39.25`	`ChannelZ`	`Oh. AsteriskNOW puts that shit in a MySQL database IIRC`
`22:39.47`	`ChannelZ`	`so yes I guess that's normal`
`22:40.12`	`ChannelZ`	`If you're following the Asterisk book, you shouldn't use AsteriskNOW which is all hotwired by FreePBX`
`22:40.38`	`h1kz0r`	`hmmm so i can't configure it using the files and just ignore the web interface?`
`22:41.37`	`*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)`
`22:41.37`	`aceio`	`ChannelZ: i just rebooted the 1261`
`22:42.13`	`aceio`	`sip show peers is not show dup ip`
`22:42.16`	`ChannelZ`	`h1kz0r: if you undo all the config done by FreePBX so it's not jacking it up, yeah`
`22:42.34`	`ChannelZ`	`aceio: wait around a couple of minutes`
`22:43.16`	`aceio`	`ChannelZ: yeah iam also watching the dhcp logs`
`22:43.29`	`ChannelZ`	`h1kz0r: do you have a "Extconfig.conf"`
`22:44.06`	`h1kz0r`	`nope, only extensions_custom.conf.sample which i renamed to extensions.conf and created an empty sip.conf`
`22:45.14`	`h1kz0r`	`i don't wanna waste too much time and then it doesn't work for a silly reason, I have a centos server here somewhere you think i should just install it from the yum repo? at least i get the latest version`
`22:45.45`	`ido-`	`Hey guys. I'm getting a 'Not acceptable here' error, can anyone please take a look: http://pastebin.com/Te2YaMxn`
`22:46.41`	`*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)`
`22:48.57`	`h1kz0r`	`any ideas?`
`22:49.00`	`aceio`	`ChannelZ: i think it's something in my Dialplan.xml`
`22:49.31`	`carrar`	`h1kz0r, use centos, install from source`
`22:49.37`	`Fubard`	`having trouble finding info on remapping polycom hardkeys, anyone gotta link that might assist?`
`22:49.54`	`carrar`	`ido, purchase some g729 codec licenses`
`22:50.32`	`ido-`	`are g729 trunk specific ?`
`22:50.52`	`ido-`	`I'm pretty sure I already have a g729 license. how can I check ?`
`22:50.56`	`ChannelZ`	`h1kz0r: if you don't want the GUI then yeah.. either just remove all the packages in the install you have now and put on just Asterisk, or build your own`
`22:51.10`	`*** join/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net)`
`22:51.18`	`*** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net)`
`22:51.30`	`ChannelZ`	`I'm not sure where FreePBX sets up realtime config for sip peers. If you haven't added any in the GUI perhaps it doesn't create the configs, no idea.`
`22:51.42`	`ido-`	`carrar ?`
`22:51.44`	`ChannelZ`	`ido-: g729 show licenses`
`22:52.21`	`ChannelZ`	`they're not "trunk specific" but you need as many licenses as you have simultaneous active channels needing it`
`22:52.28`	`ido-`	`there's only show license`
`22:52.35`	`ido-`	`and its asterisks`
`22:52.49`	`ChannelZ`	`then you don't even have the module loaded`
`22:52.55`	`ChannelZ`	`so chances are you don't have any licenses either`
`22:53.16`	`ido-`	`whats the name of the module ?`
`22:53.16`	`Mark22`	`config information: http://yourpaste.net/5088/ cli information http://yourpaste.net/5102/ I want to do an attended transfer, I can use #2 for that and the person I want to transfer the call to can hear me (and not the caller), but I can't hear him and at my side I can hear beep beep beep even after he did answer my call. unattended transfer works`
`22:53.17`	`ChannelZ`	`/usr/lib/asterisk/modules/codec_g729a.so`
`22:54.01`	`ido-`	`i dont see it`
`22:54.28`	`ChannelZ`	`and format_g729`
`22:54.31`	`ChannelZ`	`then you don't have it`
`22:54.37`	`ido-`	`while in call, how can i see which codec is being used ?`
`22:54.38`	`ChannelZ`	`It's separate`
`22:54.59`	`carrar`	`sip show channels`
`22:55.10`	`megalomano`	`hi again , I would to know , in the line of extensions.conf ... i have the line exten => _521XXXXXXXXXX,1,Dial(SIP/${EXTEN}@Mega,,L(180000) , question , what is the meaning of :SIP/${EXTEN}@Mega`
`22:55.42`	`ChannelZ`	`${EXTEN} is replaced with the extension dialed`
`22:56.08`	`ido-`	`hrm.`
`22:56.13`	`ido-`	`ok.`
`22:56.16`	`ChannelZ`	`@Mega is a SIP device name, probably from a register line in sip.conf that tells * where to send the call`
`22:56.27`	`ido-`	`can i disable a codec from within asterisk shell ?`
`22:56.35`	`ido-`	`without editing the conf files`
`22:57.09`	`carrar`	`ido, it's your OpenRG device`
`22:57.15`	`Corydon76-dig`	`ido-: you can disable the codec, but you cannot avoid the codec being negotiated`
`22:57.29`	`Corydon76-dig`	`"module unload codec_whatever.so"`
`22:57.29`	`ido-`	`yes carrar`
`22:57.42`	`carrar`	`ido-: OpenRG is saying it only talks g729`
`22:57.50`	`carrar`	`so fix openrg`
`22:57.53`	`ido-`	`thats weird.`
`22:58.04`	`ido-`	`its talking alaw too`
`22:58.10`	`ido-`	`i just checked using another trunk`
`22:58.13`	`carrar`	`it's not sayign it is`
`22:58.58`	`ido-`	`are codecs trunk specific ?`
`22:59.09`	`ido-`	`that is, where are they set to be trunk specific`
`22:59.13`	`Corydon76-dig`	`They can be`
`22:59.13`	`ido-`	`sip.conf ?`
`22:59.21`	`carrar`	`but it might be whatever is passing the call to openrg`
`22:59.27`	`carrar`	`aqnd openrg is passing through`
`22:59.30`	`ido-`	`disallow=all`
`22:59.30`	`ido-`	`allow=alaw:20`
`22:59.30`	`ido-`	`allow=g729:20`
`22:59.35`	`ido-`	`in [general]`
`22:59.51`	`carrar`	`remove g729 since you don't have licenses for that`
`23:00.44`	`carrar`	`and remove g729 from openrg`
`23:01.24`	`ido-`	`ok`
`23:02.22`	`ido-`	`codecs set only to alaw now`
`23:02.35`	`carrar`	`restart?`
`23:02.59`	`*** join/#asterisk jks (jks@193.189.93.254)`
`23:03.25`	`ido-`	`Apr 8 02:02:27 WARNING[154]: channel.c:2764 ast_channel_make_compatible: No path to translate from SIP/ido3-xxx-dcf3 () to SIP/xxx-bf19 (alaw)`
`23:03.46`	`carrar`	`because your openrg wants g729`
`23:04.12`	`ido-`	`erm.`
`23:04.31`	`ido-`	`sip show codecs has Codecs: (alaw)`
`23:04.58`	`megalomano`	`ChannelZ:thanks`
`23:05.42`	`ido-`	`so how can that be ?`
`23:05.52`	`*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)`
`23:08.27`	`tvaughn`	`what can I use to open the .ul sound files bundled with asterisk?`
`23:08.28`	`ido-`	`carrar, http://pastebin.com/sMYXGUCK`
`23:08.34`	`ido-`	`this is the full log now`
`23:10.17`	`carrar`	`not seeing any codecs being offered by the openrg`
`23:10.25`	`carrar`	`enable 711 on it`
`23:10.32`	`ido-`	`it is`
`23:10.41`	`carrar`	`call their support`
`23:11.04`	`carrar`	`or purchase some g729 licenses form digium`
`23:11.09`	`carrar`	`from`
`23:11.16`	`ido-`	`I'd prefer setting it with alaw first`
`23:11.24`	`ido-`	`then I'll get myself g729 too`
`23:12.54`	`ido-`	`carrar, i have access to asterisk conf files on openrg. i can change them myself.`
`23:19.08`	`jdoe`	`do dynamic hints work in 1.6.2? The example shown at the foot of http://www.voip-info.org/wiki/view/Asterisk+standard+extensions doesn't seem to.`
`23:24.23`	`ido-`	`carrar, found a bug over there, thanks!`
`23:24.31`	`ido-`	`solved it too.`
`23:24.42`	`ido-`	`will try to create patch if i have time`
`23:37.11`	`*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)`
`23:37.27`	`*** join/#asterisk mchou (~quassel@unaffiliated/mchou)`
`23:39.28`	`*** join/#asterisk coppice (~chatzilla@202.64.176.93)`
`23:46.55`	`*** join/#asterisk Teevo_ (~steve@154.94.96.58.static.exetel.com.au)`
`23:47.18`	`Teevo_`	`Hello, I need some help.`
`23:47.37`	`carrar`	`call 911`
`23:47.55`	`Teevo`	`lol they won't help me. :p`
`23:48.09`	`carrar`	`heh`
`23:48.30`	`Teevo`	`I'm trying to use Asterisk Now as a proxy between my SIP account and my SIP hardware.`
`23:48.48`	`Teevo`	`And I must be doing something wrong because it's not working.`
`23:49.00`	`Chainsaw`	`Teevo: Please use the #asterisknow channel.`
`23:49.09`	`Teevo`	`Okies. Thanks.`
`23:57.20`	`*** part/#asterisk ido- (~wtf@94.159.203.234)`
`23:59.47`	`*** join/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com)`

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.