| 00:01.00 | boodu | WIMPy, Signalling Type: ISDN BRI Point to MultiPoint |
| 00:03.04 | boodu | err witchtype ??? |
| 00:03.42 | WIMPy | What flavour of ISDN. You set that using switchtype= |
| 00:05.18 | ujjain | Can somebody help me find out why I can connect with X-Lite, Iphone to Asterisk after placing a firewall, but not from PAP2T? |
| 00:06.22 | boodu | I don't where i can found that i know i use coding ami and framing css |
| 00:06.38 | boodu | *I don't know where ... |
| 00:07.33 | boodu | and switchtype = euroisdn |
| 00:08.38 | WIMPy | Ok. And you are sure, you're using dss1 (=euroisdn)? |
| 00:08.51 | WIMPy | or rather should use |
| 00:11.48 | boodu | I'm not sure |
| 00:11.56 | WIMPy | I have heard about lines where you'd have to configure different SPIDs to place multiple calls |
| 00:12.25 | WIMPy | The you should check what your telco delivers/expects. |
| 00:12.53 | boodu | ok i'm going to search about that |
| 00:13.57 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:14.03 | WIMPy | Things can be so easy. You just have to live in the right place... |
| 00:14.13 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 00:46.04 | pabelanger | anyway to access astvarlibdir, astdbdir via a dialplan variable? |
| 00:47.02 | pabelanger | guess so |
| 00:53.16 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 00:54.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 01:01.59 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:03.53 | *** join/#asterisk w0rk3r (~Senny@85.15.90.36) |
| 01:08.09 | *** join/#asterisk Pio (~sean@h77.149.4.12.cable.frzr.cablerocket.net) |
| 01:26.11 | atan | Does asterisk support SCCP? |
| 01:26.44 | pabelanger | chan_skinny |
| 01:27.25 | atan | Is Skinny a secure protocol or do I need to hide the traffic somehow? |
| 01:28.12 | pabelanger | nothing is secure |
| 01:28.36 | pabelanger | however, depends on what you consider secure |
| 01:28.42 | atan | I figured as much. |
| 01:28.56 | atan | Well, currently I use SIP over to connect over the internet |
| 01:29.10 | atan | I guess I am just asking if I could expect to do the same with Skinny |
| 01:29.24 | atan | Of if it would be as silly as connecting a networked printer directly to a cable modem these days |
| 01:30.03 | pabelanger | both traverse across IP, so you would get the same result |
| 01:30.04 | WIMPy | That's just a modern Fax :-) |
| 01:30.17 | atan | WIMPy, =) hahaha |
| 01:31.01 | pabelanger | you can include TLS / SRTP into your network, however each proxy you hop thru would require the keys to decrypt / route / encrypt the protocol |
| 01:31.30 | pabelanger | best bet to start is create a VPN between your 2 networks |
| 01:34.19 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
| 01:37.00 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 01:38.41 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 01:40.08 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 01:48.27 | candrews | I have followed the directions at https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google but cannot get outgoing calls to work |
| 01:48.47 | candrews | When I run "gtalk show channels" i don't get any results |
| 01:49.05 | candrews | However, jabber is connected |
| 01:49.21 | candrews | when I attempt to place a call, I get this output in the log: |
| 01:49.21 | candrews | [Nov 3 21:39:17] WARNING[9326] chan_gtalk.c: Could not find recipient. |
| 01:49.22 | candrews | [Nov 3 21:39:17] WARNING[9326] app_dial.c: Unable to create channel of type 'Gtalk' (cause 0 - Unknown) |
| 01:49.28 | candrews | What can I do next? |
| 01:54.18 | ManxPower | candrews, did you check the doc/ directory of your asterisk source |
| 01:54.32 | ManxPower | that should have the gtalk info for YOUR version of Asterisk |
| 01:55.00 | candrews | I'm using 1.8, btw. |
| 02:08.07 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 02:29.41 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 02:33.46 | *** join/#asterisk PMantis (~sswitzer@cpe-74-74-216-216.rochester.res.rr.com) |
| 02:37.42 | atan | Is there a command I can run to see who's connected via SIP? |
| 02:37.42 | *** join/#asterisk PMantis (~sswitzer@cpe-74-74-216-216.rochester.res.rr.com) |
| 02:37.58 | WIMPy | sip show channels |
| 02:38.37 | atan | Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r ? |
| 02:38.49 | atan | I'm connected up using shell, if that means anything =\ |
| 02:39.00 | atan | Where is this 'sip' thinger I am trying to invoke? |
| 02:39.11 | pabelanger | yes, it tells you what you need to do... 'asterisk -r' |
| 02:39.22 | atan | err, yes. My bad. |
| 02:39.28 | WIMPy | It means what it says. Asterisk is already running. |
| 02:40.40 | atan | Oh rock on. Okay so my phone actually connected up without any issue =D too awesome |
| 02:40.55 | atan | Are there any default remote logins enabled by, uh, default? |
| 02:41.05 | PMantis | Hi, having a problem with several Aastra 6731i phones cutting off after on a call for 3600 seconds. Asterisk sends an SDP invite, Aastra sends a "481 Call Leg/Transaction Does Not Exist" and disconnects, but Asterisk keeps the channel open indefinitely. And ideas here? |
| 02:41.23 | WIMPy | atan: Are you sure, you know what you mean by "connected"? |
| 02:41.53 | atan | WIMPy, nope! I setup my sip phone to connect to the asterisk server... and the server shows it under 'sip show channels' |
| 02:42.06 | pabelanger | PMantis: sounds like a NAT issue |
| 02:42.12 | atan | Shows the remote IP of the phone correctly, user (none) and some random-looking call ID |
| 02:42.30 | PMantis | pabelanger, On the same subnet as the server, no proxy. |
| 02:42.45 | atan | Now I'm just trying to configure it up so my outbound calls function for me =) and then to figure out how to set the Caller Name, and Caller Number for the caller ID |
| 02:42.52 | WIMPy | atan: I think you're actually looking for sip show peers. That shows registrated peers. |
| 02:43.24 | atan | show peers shows me this: 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 02:43.35 | atan | One being my remote server I'm trying to setup with, in my case voip.ms |
| 02:43.43 | WIMPy | PMantis: Sounds like you have session-timers enables and your phones don't like them. |
| 02:44.19 | PMantis | pabelanger, I have several Polycom phones that behave with the same server. I've actually had them in a conference all night long playing MOH without any issues. |
| 02:44.35 | PMantis | WIMPy, session timers enabled in * ? |
| 02:44.55 | WIMPy | sip.conf |
| 02:46.14 | PMantis | WIMPy, I'll bite. |
| 02:46.21 | WIMPy | Me? |
| 02:46.52 | PMantis | WIMPy, Heh, meaning "Sounds sane enough for me to investigate" |
| 02:46.54 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 02:47.09 | WIMPy | gooood |
| 02:48.21 | PMantis | WIMPy, I don't have anything in the sip.conf that looks like a timeout. I'm looking at 'sip show settings' now.. |
| 02:49.02 | WIMPy | session-* |
| 02:49.45 | WIMPy | are you using a sip.conf from an older version, than you're using? |
| 02:50.00 | PMantis | WIMPy, in 'sip show settings', these look like they could apply: "Qualify Freq : 60000 ms" "Qualify Freq : 60000 ms" "Session Expires: 1800 secs" |
| 02:50.26 | PMantis | WIMPy, It's possible. I actually am using a perl script to auto-generate the sip.conf from XML. |
| 02:50.35 | WIMPy | Qualify is something else. |
| 02:50.55 | WIMPy | But Session expires might be it. |
| 02:50.58 | PMantis | "Reg. max duration: 3600 secs" |
| 02:51.50 | WIMPy | Yes. Session expires is session-expires. |
| 02:52.07 | WIMPy | try session-timers=refuse |
| 02:52.18 | PMantis | in global? |
| 02:52.20 | atan | Is there a way I can view phone call attempts from the SIP devices? |
| 02:52.23 | WIMPy | yes |
| 02:52.41 | atan | Right now I dial, and it goes blank... but seems to be connected to both my provider + my sip phone |
| 02:52.47 | PMantis | WIMPy, If it helps I have a tcpdump capture of the call beginning to end. |
| 02:52.55 | WIMPy | to try it out you can lower the time. I think session-expires=600 is the shortest possible. |
| 02:54.17 | PMantis | WIMPy, How can the Polycom phones I use stay in a conference playing MOH all night long, though? It points to Aastra as the issue, although it could easily be an incompatibility... Just like to make sense of things. |
| 02:54.19 | WIMPy | I think I remember seing something on the issue tracker about session timers being activated even when the peer didn't indicate support for it. |
| 02:54.48 | PMantis | Ahh, your last statement clears it up. |
| 02:55.07 | pabelanger | WIMPy: they default to originate |
| 02:55.29 | pabelanger | sorry, accept |
| 02:56.31 | PMantis | WIMPy, I'll try that on the Aastra peer itself. Isn't that reasonable? |
| 02:57.02 | WIMPy | Not sure if you can set it per peer. |
| 02:57.40 | pabelanger | you can |
| 02:58.23 | PMantis | That approach makes sense to me, then... since the Polycom's work anyway. |
| 02:58.50 | atan | What do I need to insert into my config files (I assume sip.conf for maybe extensions.conf) to route all outbound calls though the SIP trunk I have setup inside sip.conf? =S the tutorials I am looking at all show examples of local extensions but not actually connecting to the outside world |
| 02:59.28 | WIMPy | atan: Asterisk makes no diffrence. |
| 02:59.36 | PMantis | reads RFC 4028 |
| 02:59.45 | WIMPy | A call comes in and the rest is dialplan. |
| 03:00.16 | WIMPy | There is no internal and external or trunk or whatever. Just calls. |
| 03:00.41 | atan | Here is what I have currently for it http://pastie.org/1271331 |
| 03:01.01 | pabelanger | ~book |
| 03:01.01 | 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, or http://ofps.oreilly.com/ |
| 03:01.01 | PMantis | WIMPy, pabelanger, Thank you for the help. I'll dig in a little and hope this resolves the issue. |
| 03:01.04 | pabelanger | atan: ^ |
| 03:01.07 | atan | I assume this should push calls over to the @voipms connection |
| 03:03.08 | pabelanger | atan: no. Dial(SIP/voipms/${EXTEN} |
| 03:03.15 | pabelanger | atan: no. Dial(SIP/voipms/${EXTEN}) |
| 03:03.55 | atan | reloads |
| 03:06.08 | atan | <PROTECTED> |
| 03:06.24 | atan | 123 should get forwarded over to the voipms thinger |
| 03:07.07 | boodu | what is framing (i have css) and coding (i have ami) with DAHDI ? |
| 03:07.19 | WIMPy | Read the part about contexts. |
| 03:07.23 | atan | Heck, a full number doesn't get sent over to the voipms connection =\ |
| 03:07.26 | WIMPy | atan ^^ |
| 03:07.38 | atan | WIMPy, on which link? |
| 03:07.55 | WIMPy | the book |
| 03:08.09 | WIMPy | Or any book or wiki or whatever. |
| 03:09.50 | atan | When I make changes to extensions.conf can I just reload or do I need to restart? |
| 03:10.14 | WIMPy | dialplan relaod will do |
| 03:11.20 | atan | So then I could say exten => 123,1,Dial(SIP/voipms/100) ? |
| 03:11.39 | atan | Then if one dials 123, it forwards to the voipms connection requesitng #100 ? |
| 03:11.47 | pabelanger | boodu: Framing should be D4 or ESF |
| 03:11.49 | [TK]D-Fender | atan: Since when is voipms going to consider "100" a valid number on their side? |
| 03:12.22 | atan | [TK]D-Fender, sorry, perhaps I could include the actual extension but inside their accounts you can list an extension |
| 03:13.01 | atan | [TK]D-Fender, I was assuming there would be a new error message of some kind to indicate it did try to connect it up |
| 03:13.03 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:14.02 | [TK]D-Fender | atan: they are an ITSP. you use them to call real world numbers. 100 isn't a real world number |
| 03:14.46 | atan | [TK]D-Fender, when I connect directly to their servers using a sub account I can dial my other connected sip devices using their extension |
| 03:15.11 | atan | If I tell one sub account that it is extension 101 my other phones can call it by pressing 101 |
| 03:15.12 | [TK]D-Fender | "sub account"? |
| 03:15.42 | atan | I suppose sub-account in their terms is just another sip login |
| 03:16.29 | [TK]D-Fender | atan: I'd still like to know how it considers those numbers legit... |
| 03:16.46 | atan | Okay, well, should I substitute it for my cellphone number for now then? |
| 03:17.07 | atan | I'm just logging in here now, will grab a screenshot to give you more detail if you would like =) |
| 03:17.18 | atan | For whatever reason their site is acting slow as a slug for me right now. |
| 03:20.25 | atan | Wouldn't exten => _1NXXNXXXXXX,1,Dial(SIP/voipms/${EXTEN}) appear to accept a 1905XXXXXXX input, and forward it along to the voipms connection? |
| 03:20.44 | leifmadsen | yes |
| 03:21.31 | atan | So then I am confused why my logs show handle_request_invite: Call from '7940' to extension '19024404444' rejected because extension not found. |
| 03:21.50 | pabelanger | because it is in the wrong context |
| 03:22.12 | atan | pabelanger, you mean where I have it positioned within my extensions.conf file? |
| 03:22.23 | pabelanger | yes |
| 03:22.25 | atan | Right now it's under [mycontext] |
| 03:22.35 | atan | Does this mean my sip.conf should point to that same one? |
| 03:22.38 | pabelanger | does you phone have access to that context? |
| 03:22.50 | pabelanger | yes or included |
| 03:22.53 | atan | Ahhhh. Okay! Now we're cooking with gas. One sec while I fix that =) |
| 03:23.01 | atan | I had it set to context=ciscotest |
| 03:23.16 | atan | When I make a change to my sip.conf, should I reload, restart, or ? |
| 03:23.22 | leifmadsen | pabelanger: I build my AGI earlier to strip off all that XML junk :) |
| 03:23.27 | leifmadsen | dialplan reload |
| 03:23.31 | leifmadsen | errr |
| 03:23.32 | leifmadsen | sip reload |
| 03:23.42 | ChannelZ | life reboot |
| 03:23.48 | pabelanger | high-fives leifmadsen |
| 03:24.01 | leifmadsen | pabelanger: only 95 lines of PHP :) |
| 03:24.15 | pabelanger | >.< |
| 03:24.25 | atan | Oh my goodness. I love you guys. We're up and running here now!! |
| 03:24.45 | atan | pabelanger, thank you for the tip on using the context properly |
| 03:24.47 | leifmadsen | nice :) |
| 03:24.59 | atan | same to you, WIMPy |
| 03:25.02 | leifmadsen | 166 lines of dialplan so far tonight |
| 03:25.11 | pabelanger | atan: like I said, read the book. Lots of information |
| 03:25.14 | leifmadsen | would be further along if it wasn't for that PHP script :) |
| 03:25.20 | leifmadsen | the book is junk! |
| 03:25.22 | leifmadsen | :D |
| 03:25.44 | atan | So does this mean that 1.8 has been changed, and as such "Dial(SIP/1${EXTEN}@voipms)" is now void in favor of Dial(SIP/voipms/${EXTEN})? |
| 03:26.00 | pabelanger | leifmadsen: same, up to 60 now. Playing with CELGenUserEvent too |
| 03:26.14 | leifmadsen | nice! |
| 03:26.22 | leifmadsen | I have to add that to this system in the 2nd iteration |
| 03:26.44 | leifmadsen | atan: I prefer that latter since it uses the [peer] information in sip.conf |
| 03:27.05 | leifmadsen | perhaps the former does as well, but i've never seen it like that really |
| 03:27.20 | *** join/#asterisk ups (~ups@c-67-172-114-234.hsd1.ca.comcast.net) |
| 03:27.24 | atan | Now can I pre-configure the user caller ID + caller num within my sip.conf? *googles for possible sip.conf configurations* |
| 03:27.36 | leifmadsen | callerid= |
| 03:27.58 | atan | callerid represents the name, or number? or both in some funky format like DIsplayName\15551212 |
| 03:28.10 | ups | Hello guys. Is there a way to include a context in musiconhold so that a caller can jump out for example into a voicemail ? |
| 03:29.14 | p3nguin_ | atan: There are examples of every setting in the sample file included with your source distribution. |
| 03:29.48 | atan | ty! |
| 03:29.56 | atan | If the "We're sorry due to technical difficulties we are unable to route your call" recording something in Asterisk or elsewhere? |
| 03:30.04 | atan | s/if/is |
| 03:30.14 | p3nguin_ | So close! |
| 03:30.36 | pabelanger | do you see it in your CLI? |
| 03:30.39 | p3nguin_ | s/So close/You missed the trailing slash/ |
| 03:32.36 | leifmadsen | how do I verify with a regex whether something is a number or not? |
| 03:33.18 | atan | s/s\/if\/is/s\/if\/is\// |
| 03:34.28 | leifmadsen | hmm, this might work: ^[0-9]+$ |
| 03:36.17 | pabelanger | <PROTECTED> |
| 03:36.23 | pabelanger | minus the ? |
| 03:37.23 | leifmadsen | apparently \d could match outside of 0-9 since unicode defined more than just 10 number |
| 03:37.29 | leifmadsen | http://stackoverflow.com/questions/273141/regex-for-numbers-only |
| 03:38.20 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 03:39.14 | *** join/#asterisk w0rk3r (~Senny@85.15.87.159) |
| 03:39.31 | WIMPy | [\d0-9]+ would only match an even number of digit, I guess. |
| 03:40.09 | WIMPy | Where the even ones are arabic and the odd ones anything digit. |
| 03:40.21 | *** join/#asterisk BeeBuu (~chatzilla@58.252.73.164) |
| 03:40.31 | WIMPy | bullshit |
| 03:40.37 | BeeBuu | is there new sip client in linux? |
| 03:40.47 | WIMPy | It just lists 0-9 twice. |
| 03:41.40 | atan | Is there some funky free service somewhere that reads out your caller ID name? For some reason mine isn't cooperating with me right now. Cellphones also don't display the names for me. Hmm. |
| 03:42.45 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 03:47.53 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ozrghhkingkgjvlm) |
| 03:53.41 | PMantis | WIMPy, This time the Aastra dropped the call at about 45 minutes. Nothing in the tcpdump output that indicated WHY. :( |
| 03:55.04 | WIMPy | No information on the *cli either? |
| 03:55.57 | PMantis | WIMPy, The first time, I walked out of the room for about 10 minutes. When I came back, the call was gone... so not sure exactly when it happened. Might not related to the 481 error I saw at the end before. |
| 03:56.20 | PMantis | WIMPy, No. and 'core show channels' still shows the call active: "SIP/device18_1-cc05c 555@user-502:15 Up MeetMe(555,wqMs,)" |
| 03:57.21 | WIMPy | Maybe you should activate rtptimeout to at least get rid of the zombe calls. |
| 03:57.42 | PMantis | I have to run a manual channel request hangup |
| 03:57.54 | WIMPy | But unfortunaletly no idea springs to mind what your Aastras do. |
| 03:58.03 | PMantis | WIMPy, Ahhh. me looks for that. |
| 04:00.46 | PMantis | Ok, I set the rtptimeout to 60, performed a sip reload, and we'll see if that channel disappears. |
| 04:01.13 | PMantis | WIMPy, Buut, I'd *sure* like to know what's causing this annoying call drop. :( |
| 04:02.40 | PMantis | WIMPy, Either way, THANK YOU for your time and help on this - I'm at least closer to determining why. |
| 04:02.54 | WIMPy | Are you? |
| 04:03.49 | PMantis | WIMPy, Yeah, there's a timeout. I didn't know about session timers before I talked to you... |
| 04:04.11 | WIMPy | But they weren't the cause. |
| 04:04.59 | PMantis | After I read, I set session-timers=accept this time. Perhaps I should try your original suggestion. |
| 04:05.33 | WIMPy | Did you set the time as well? |
| 04:05.36 | atan | So I've told this thing to dial me for 20 seconds here, with Dial(SIP/7940,20), how do I tell it I want to hang up if the number is unresponsive? |
| 04:05.40 | atan | ie. no pickup |
| 04:05.49 | atan | Or perhaps forward to another extension/voicemail |
| 04:06.08 | PMantis | WIMPy, Oh .... no. |
| 04:06.28 | WIMPy | atan: You continue (in) your dialplan. |
| 04:06.43 | [TK]D-Fender | atan: NEXT priority |
| 04:06.51 | PMantis | WIMPy, session-expires=600 ? |
| 04:07.00 | [TK]D-Fender | atan: no/negative response = continue processing |
| 04:07.06 | WIMPy | PMantis: yes |
| 04:07.15 | atan | So say my first line reads (the end of it) 1,Dial(SIP/7940,20) my next rule could be 2,Dial(SIP/7941,20) ? |
| 04:07.42 | WIMPy | atan: For example |
| 04:09.31 | PMantis | WIMPy, Hmmm, that channel still didn't drop... perhaps because it's in MOH so * is always sending a stream. |
| 04:09.55 | PMantis | kills a channel |
| 04:12.57 | PMantis | WIMPy, Hmmmmm about 1/2 way dow this page: http://www.voip-info.org/wiki/view/Aastra I see: "You should avoid using "sip session timer" with any value other than 0" |
| 04:14.12 | PMantis | WIMPy, But now... I've changed session-timers=refuse *and* changed an Aastra option. |
| 04:14.42 | PMantis | WIMPy, I'll try it and see... but I'm going to bed - check the phone in the AM. :) |
| 04:14.45 | atan | WIMPy, you popular fella, what were you saying? |
| 04:15.22 | PMantis | atan, he meant: You are correct, for an example. |
| 04:15.48 | atan | can I use 'n' in place of a number? To represent 'after the thinger above me' ? |
| 04:15.58 | PMantis | atan, yes |
| 04:16.01 | [TK]D-Fender | atan: yes |
| 04:16.45 | atan | So I'm sending a hangup() after that if there's no response but the cellphone calling does not disconnect... it plays a 'doon, doon, doon, doon' noise like forever |
| 04:16.48 | PMantis | You can also use: n+1, n+100, etc. (useful for apps that 'jump') |
| 04:16.57 | atan | Normally if I hang up the cellphone drops the call |
| 04:17.28 | atan | PMantis, sweet. Thanks for that tidbit. |
| 04:17.38 | atan | I assume this means there are listen functions in there too, eh? |
| 04:18.03 | atan | Is there a nice list of these functions I can graze over? Or print for that matter =D |
| 04:18.38 | PMantis | atan, type: core show applications |
| 04:18.56 | PMantis | Then search for one that interests you on voip-info.org |
| 04:19.03 | atan | wow. |
| 04:19.25 | PMantis | atan, Yup. |
| 04:19.48 | PMantis | atan, I prefer NOT to think of asterisk as a phone system... but rather a programming language that can be used to build one. |
| 04:20.22 | atan | Does SayAlpha actually read English? |
| 04:20.38 | atan | ie. I could collect an ID, and read a sentence? Or grab an account balance using a URL? |
| 04:20.39 | atan | O_O |
| 04:21.14 | PMantis | atan, I think if you do SayAlpha(a), it'll look for an a.ulaw file to play (letter A), etc. |
| 04:21.21 | atan | Woah woah woah. Woah. Woah. Let's talk ab out this thinger... " SendText:" |
| 04:21.29 | atan | Does that send a text message to like, a cell phone? |
| 04:21.37 | atan | I never had any idea how those thingers worked to begin with |
| 04:22.24 | WIMPy | PMantis: rtptimeout is meant to cath in when no rtp is _received_. |
| 04:22.38 | PMantis | atan, haha. I don't know of any providers (SIP/IAX/TDM) that allow you to send text messages. |
| 04:23.01 | PMantis | WIMPy, Hmmm, didn't work for me. Maybe a full restart is needed, not just a 'sip reload'? |
| 04:23.12 | atan | Well darn. I was going to say, like, AWESOME!! =) Make a gmail for text messages. Man would that be awesome! Threaded conversations... spam filtering... |
| 04:23.40 | PMantis | atan, you can send messages to supporting SIP phones. |
| 04:23.52 | WIMPy | PMantis: Shouldn't, but I've had occasions where a kill and restart did cure things, variations of reload didn't. |
| 04:24.01 | PMantis | atan, I've just never done that. |
| 04:24.17 | PMantis | I'll try 'module reload' |
| 04:24.19 | atan | I suppose I should move along to the second most important thing, now that inbound/outbound calls are sorted out. That is, voicemail. |
| 04:24.42 | atan | I assume I just add a rule inside extensions.conf to transfer to voicemail as needed |
| 04:25.30 | PMantis | atan, If you're building your dialplan yourself, BE SURE to keep contexts in mind, and CORRECT. Otherwise, you might be a SIP gateway for strangers to make phone calls on your dime. |
| 04:26.00 | PMantis | Anyway... time for sleep before I get grumpy. :) |
| 04:26.03 | atan | PMantis, uht-oh. |
| 04:26.09 | atan | Oh yes, now you leave :P lol |
| 04:26.15 | PMantis | lol |
| 04:26.18 | atan | Right after that "make his heart beat like mad" moment there |
| 04:26.30 | PMantis | heh |
| 04:26.44 | atan | Before you go, can you throw me a bone as to what you mean? Right now I just have sip.conf which has three sections |
| 04:26.49 | atan | general, voipms, and 7940 |
| 04:27.45 | atan | PMantis, do you mean I should seperate my inbound & outboind contexts? |
| 04:27.46 | WIMPy | First take a look at allowguests and th contex under general. |
| 04:28.06 | atan | Under general I ONLY have register => myserverinfos |
| 04:28.14 | WIMPy | That's usually a very good idea. |
| 04:28.24 | PMantis | atan, Look at sip.conf, and be sure that "allowguest=no" is there and uncommented. The "context=default" in general also states that if there is a request coming in that doesn't match another entry below, throw them into "default" and look for a phone # match. |
| 04:28.56 | PMantis | atan, Add allowguest=no... just to be SURE it's no. |
| 04:29.03 | atan | So allowguest=no under [general] ? |
| 04:29.16 | PMantis | yes |
| 04:29.30 | atan | Got'r. |
| 04:29.33 | atan | She's in there now |
| 04:29.48 | PMantis | atan, On my system (Ubuntu) I have "/usr/share/asterisk/conf/sip.conf.sample". Find something similar on your system and read. |
| 04:30.11 | atan | I have /etc/asterisk/sip.orig.conf I'll look over |
| 04:30.24 | atan | But this guest thing for now, covers it mostly? =S |
| 04:30.26 | WIMPy | Yes. The sample configs are very informative. |
| 04:30.33 | atan | I assume I just need to be careful if I am accepting data from the user. |
| 04:30.34 | PMantis | atan, One last point before I go... |
| 04:30.46 | ChannelZ | shake your booty |
| 04:30.51 | PMantis | Pay attention to: deny=0.0.0.0/0.0.0.0 and permit=192.168.0.60/255.255.255.0 |
| 04:30.54 | atan | shake shake |
| 04:31.01 | PMantis | Uhmmmmmmmmm |
| 04:31.07 | PMantis | thinks not |
| 04:31.12 | atan | Sorry. |
| 04:31.16 | PMantis | LOL |
| 04:31.26 | atan | Where might I see deny / permit stuff like this? |
| 04:31.31 | ChannelZ | OK Jebadiah |
| 04:31.40 | PMantis | atan, sip.conf examples. |
| 04:31.59 | atan | What does it do? Deny access to account from such IP/allow it? |
| 04:32.07 | PMantis | atan, I assume that 7940 is an internal phone? |
| 04:32.33 | atan | Not sure what you mean? It's a desk phone. It connects to my asterisk which is miles, mile, miles away =) |
| 04:32.41 | atan | Server is not local |
| 04:32.47 | PMantis | atan, ahhhh |
| 04:32.49 | atan | But I do plan to attempt to use it for some other local customers |
| 04:33.07 | atan | So I don't want to limit the IP range too much. I mean, the local isps ip ranges would be okay but not too much. |
| 04:33.23 | atan | I also use SIP on my cellphone to skip cell charges. woot. |
| 04:33.29 | atan | My phone company hates me, I swear. |
| 04:33.35 | PMantis | atan, But yes, deny all, then only open what you have to to allow each entry to work (that's ideal). |
| 04:33.52 | atan | PMantis, to aid in preventing brute forcing? |
| 04:33.59 | PMantis | If you can't use a large deny scope, then be sure your password is set. |
| 04:34.01 | PMantis | atan, Exactly. |
| 04:34.07 | atan | PMantis, makes sense then |
| 04:34.25 | atan | What does disallow=all mean? |
| 04:34.28 | atan | I have that on my drunk |
| 04:34.38 | atan | does that mean it saves it for outbound stuff, no inbounders allowed? |
| 04:34.39 | PMantis | atan, refers to codecs |
| 04:34.47 | atan | Oh. |
| 04:34.51 | PMantis | then you allow=(whatever you want) |
| 04:35.07 | trollasaurus | Anyone have a suggestion for a cheap, but good hardware SIP client ? |
| 04:35.11 | atan | I want to pastebin you one config, only one sec |
| 04:35.54 | atan | http://pastie.org/1271461 |
| 04:35.55 | PMantis | atan, If you DO have a guest account, limit them to dialing only extensions that don't cost you money for them to dial. Contexts are your friend - and enemy. Get to understand them! :) |
| 04:36.18 | atan | That link is my config for my trunk, I believe |
| 04:36.19 | WIMPy | sleep mode |
| 04:36.25 | PMantis | atan, First thing... do you have a context of "mycontext" in extensions.conf? |
| 04:36.33 | atan | Is there anything I should have in there to prevent any malicious fools like myself from using it? |
| 04:36.41 | atan | I do, and am using [mycontext] for everything right now |
| 04:36.49 | atan | local dialing rules, inbound calls, and so on |
| 04:36.51 | PMantis | atan, BAD idea. |
| 04:36.59 | atan | bangs head on wall |
| 04:37.11 | PMantis | Easy to fix. |
| 04:37.14 | atan | Separate it somehow? |
| 04:37.33 | PMantis | Make a context for JUST that provider... like [voipms-inbound] |
| 04:37.59 | atan | I have that now. I assume I place my inbound extens in there? |
| 04:38.13 | PMantis | Then add dial rules in the extensions.conf under that context that match DID's (numbers), and call dial rules accordingly. |
| 04:38.13 | PMantis | Yes |
| 04:38.38 | PMantis | atan, Then your sip.conf should have: context=voipms-inbound |
| 04:38.53 | atan | PMantis, under which section? [general] ? |
| 04:38.56 | PMantis | But only for the snippet you pasted me. |
| 04:39.20 | atan | Let me pastebin my entire sip.conf file to prevent me from messing this up too hard, one sec =D |
| 04:39.42 | PMantis | Remember what I said above... that context line in there tells * to look in extensions.conf for the context to see if there's a match for the inbound #. |
| 04:40.03 | atan | http://pastie.org/1271465 is sip.conf |
| 04:40.10 | PMantis | You must know what you're matching up and your desired results. |
| 04:40.40 | atan | So then, in [general] I place context=voipms-inbound because [general] handlee my inbound calls since they are not being routed elsewhere yet? |
| 04:40.52 | PMantis | no |
| 04:41.20 | atan | Under [voipms] in sip.conf, since it gets inbound calls from the voipms trunk? |
| 04:41.33 | PMantis | under general, place a context=something_that_is_empty |
| 04:41.59 | PMantis | Then, under [voipms], set the context to "voipms-inbound" so it jumps there to process the call. |
| 04:42.38 | atan | Got it. |
| 04:42.53 | PMantis | Maybe even use my exact "context=something_that_is_empty" in general, so you look at it later and remember that it was on purpose. |
| 04:42.54 | PMantis | :0 |
| 04:42.54 | atan | and my [7940] phone SIP extension I tell to use my outbound context, with the rules and such? |
| 04:43.11 | atan | I set it to context=empty, and made a context called [empty] in extensions |
| 04:43.18 | atan | ..of course with nothing in it. |
| 04:43.26 | PMantis | LOL |
| 04:43.40 | atan | Just to get one thing strait. If I have two contexts in a file, like [1] and [2], #1 doesn't load data under #2 does it? |
| 04:43.46 | atan | It stops when it hits another [, right? |
| 04:43.57 | [TK]D-Fender | atan: yes |
| 04:44.07 | atan | Cause that would be hell'a mess. |
| 04:44.08 | PMantis | Then the 7940 would need a context that can dial other extensions, and if you want... external numbers. |
| 04:44.54 | PMantis | atan, heh, yes it would... but as you gain experience, you'll want to include (confusing sometimes), there is an "include => context" option. |
| 04:44.57 | atan | For the sake of this I'll call them inbound + outbound |
| 04:45.15 | atan | PMantis, that would avoid repeating my own code is all, no? |
| 04:45.17 | PMantis | atan, Sure. |
| 04:45.57 | atan | This is freakin awesome. You guys totally rock =) |
| 04:46.00 | PMantis | atan, You might want to have a phone that can dial internally only, and another that can dial internal & local, internal & local & LD... etc.. |
| 04:46.07 | atan | Now I best figure out this voicemail.conf file :D |
| 04:46.31 | atan | PMantis, so like, I could lock people down... ie. free calls only, no iraq, and so on? |
| 04:46.51 | atan | Yes yes, okay that makes sense. |
| 04:47.26 | PMantis | atan, Exactly. |
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| 04:48.13 | atan | A courtesy phone for example, could maybe only call local... or require a code to get outbound =) |
| 04:48.15 | PMantis | atan, Example: exten => _NXXNXXX,1,Dial.... that only matches 7 digit US numbers (usually local). |
| 04:49.01 | atan | but your Dial would likely need to inject the area code then, no? |
| 04:49.05 | atan | As such, only one area? |
| 04:49.06 | PMantis | atan, Lots and lots you can do to manipulate the numbers, too.. |
| 04:49.08 | PMantis | atan, Yes |
| 04:49.17 | atan | =D |
| 04:49.38 | PMantis | atan, You can also do: exten => 411,1,Dial(SIP/provider/18001234567) |
| 04:49.39 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |
| 04:49.53 | PMantis | So, force 411 calls to use a free directory service. :) |
| 04:49.55 | atan | Too bad Goog411 is gone now! |
| 04:50.43 | atan | One could write their own system to look it up on yellowpages.com or something then, no? |
| 04:50.57 | drmessano | Force 411 to the request line at a local radio station |
| 04:50.58 | atan | just grab the name via numpad or something, and query it ^_^ |
| 04:51.38 | PMantis | drmessano, LOL |
| 04:52.18 | PMantis | atan, With perl and AGI, you may be able to do that... just wondering about typing in a name to search on. :) |
| 04:52.43 | PMantis | Ok, almost an hour later... I *really* need to get to bed. |
| 04:52.47 | PMantis | TTYL guys! |
| 04:53.15 | atan | Ciao baby, ciao! Thank you kindly for all your lines of text. |
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| 04:55.33 | PMantis | atan, You're welcome |
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| 05:35.27 | atan | Err, chan-sccp talks about working with 1.4 & 1.6... but the latest is 1.8. Anyone fiddle with it? =S |
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| 05:42.05 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 06:40.05 | atan | Well crap. Does SCCP not support a password of any kind? =S |
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| 06:42.26 | carrar | craps south park style |
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| 07:05.29 | atan | Can anyone here recommend an awesome multi-line SIP phone? =) |
| 07:06.14 | atan | I am about to chuck these Cisco 7940/7960/7914 things across the room and then light the house on fire |
| 07:08.00 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 07:08.15 | Tim_Toady | go with polycoms |
| 07:09.20 | atan | Tim_Toady, why do you say that? Fireproof? |
| 07:09.39 | Tim_Toady | i guess so |
| 07:10.14 | atan | Tim_Toady, is there a particular model you find particularly awesome? |
| 07:10.36 | atan | Ideally I am looking for something that supports multi lines |
| 07:10.41 | atan | At least 3 are requires |
| 07:10.58 | atan | Speed dial is also a nice addition but I can work around it if need be |
| 07:19.41 | boodu | WIMPy, I can't resolve my problem with dahdi |
| 07:19.52 | boodu | and my moh problem |
| 07:19.56 | boodu | :( |
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| 07:23.31 | ChannelZ | which are what exactly |
| 07:38.04 | boodu | ChannelZ, you 're talking to me ? |
| 07:38.51 | boodu | my dahdi proble it's i can use 2 channel of a port ISDN with outgoing mode |
| 07:38.55 | boodu | I use dahdi |
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| 07:58.29 | schmidts | good morning |
| 08:02.50 | atan | Where does Asterisk get the modules from? |
| 08:03.01 | atan | I would like to remove chan_skinny.so but don't see a config file with it |
| 08:03.06 | atan | I see a module unload option but, err |
| 08:03.18 | atan | I'd much prefer to see module.conf with the modules listed where I could remove & reload |
| 08:04.00 | Tim_Toady | /etc/asterisk/modules.conf |
| 08:04.13 | Tim_Toady | noload => chan_skinny.so |
| 08:05.07 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 08:05.30 | verywiseman | how can i call sip phone through cli? |
| 08:11.21 | kaldemar | verywiseman: console dial ... |
| 08:13.06 | verywiseman | kaldemar, i can't find dial cli, i am working on * 1.4 |
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| 08:15.17 | kaldemar | you need either chan_alsa.so or chan_oss.so loaded for console dial to work. "module show like chan" will show if you have one of those. |
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| 08:21.58 | verywiseman | i have this message "Got SIP response 302 "Moved Temporarily"" , what is that meaning? |
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| 08:26.01 | kaldemar | verywiseman: the phone is redirecting the call somewhere else, most likely. |
| 08:26.56 | kaldemar | is there a call transfer activated in the phone? the Contact header should point where the call is redirected. |
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| 08:28.14 | verywiseman | kaldemar, ok |
| 08:28.24 | verywiseman | kaldemar, what is that meaning |
| 08:28.25 | verywiseman | Now forwarding SIP/2424-096e4b58 to 'Local/ 1234@DLPN_SeniorManagers' (thanks to SIP/3123-098687c8) |
| 08:28.33 | verywiseman | where my ext is 2424 |
| 08:28.40 | verywiseman | and i call 3123 |
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| 08:31.15 | kaldemar | SIP/3123 transfers the call to 1234. |
| 08:32.18 | verywiseman | kaldemar, thank you :) |
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| 08:50.23 | atan | Err, can I send a user to their own voicemail listed within their config somehow? Right now exten => *98,1,VoiceMailMain wants a mailbox number |
| 08:50.53 | atan | But seeing the user already has a mailbox= in their own config there must be a way? |
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| 08:51.57 | kaldemar | atan: you need to specify it to the application. |
| 08:52.16 | atan | kaldemar, for each user's outbound? |
| 08:52.37 | kaldemar | atan: come again? |
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| 08:53.10 | atan | Inside extensions I have my users (sips), should it be setup in there so it calls their preset voicemail? |
| 08:53.13 | kaldemar | if there is no mailbox parameter for VoiceMailMain, it will prompt for one. |
| 08:53.43 | atan | Okay so I have exten => *98,1,VoiceMailMain as a rule |
| 08:53.53 | atan | So like *98 brings it up for them |
| 08:53.56 | kaldemar | the extension needs to be *98,1,VoiceMailMain(<mailbox>) |
| 08:54.03 | b_d | anyone set an ivr/message for the callee from a call initiated from asterisk? almost like a collect call message? I have been working with phpagi, but cant seem to get it right |
| 08:54.24 | atan | kaldemar, is there a way to make exten => *98,1,VoiceMailMain pull that data from the sip.conf for the person using it? |
| 08:54.44 | kaldemar | atan: yes. |
| 08:54.56 | atan | Err, <mailbox> ? |
| 08:55.48 | atan | exten => *98,1,VoiceMailMain(<mailbox>) still requests the user enter a mailbox number |
| 08:55.49 | kaldemar | atan: VoiceMailMail(${SIPPEER(mailbox)}) |
| 08:55.51 | atan | nice |
| 08:55.59 | atan | <3 you long time |
| 08:56.10 | kaldemar | <mailbox> is not a literal parameter, just a sample. |
| 08:56.32 | atan | No wait that's not working as it should |
| 08:56.35 | kaldemar | i had a small typo in there... |
| 08:56.45 | kaldemar | how should it work? |
| 08:56.54 | atan | Sorry for the troubles |
| 08:57.02 | atan | So inside sip.conf I have my [user] with mailbox=1000@voicemail |
| 08:57.14 | atan | Inside extensions I have exten => *98,1,VoiceMailMain(${SIPPEER(mailbox)}) |
| 08:57.23 | atan | ...and voicemail I have 1000 => 1234, atan, |
| 08:57.33 | atan | I want the [user] to get to their own box byu *98 |
| 08:58.16 | atan | I assume we are on the right track with exten => *98,1,VoiceMailMain(${SIPPEER(mailbox)}) |
| 08:58.26 | atan | Except that for whatever reason it still wants the mailbox # |
| 08:58.35 | kaldemar | what do you see in the CLI? |
| 08:59.07 | atan | [Nov 4 08:59:10] WARNING[7932]: app_voicemail.c:6369 vm_authenticate: Couldn't read username |
| 08:59.17 | kaldemar | what else? |
| 08:59.32 | kaldemar | show the verbose output |
| 08:59.32 | atan | When I dial *98 that is all I get |
| 08:59.38 | atan | How do I do that? -v? |
| 08:59.46 | kaldemar | "core set verbose 10" in CLI |
| 09:00.33 | atan | http://pastie.org/1271743 |
| 09:00.51 | atan | I did supply an invalid password & username though |
| 09:00.58 | atan | I could try back with the real one |
| 09:01.05 | ups | thank you |
| 09:01.18 | ups | but i have no idea what are you talking about :) |
| 09:01.48 | atan | Err. It does not let me in even if I supply the mailbox number, 1000, and the password |
| 09:02.22 | kaldemar | "Executing [*98@outbound:1] VoiceMailMain("SIP/7940-00f54220", "")" suggests that VoiceMailMain gets no arguments. |
| 09:02.59 | atan | So then it is indeed this thinger? exten => *98,1,VoiceMailMain(${SIPPEER(mailbox)}) |
| 09:03.09 | ups | what voicemail context user is in ? |
| 09:03.18 | atan | [7940] |
| 09:03.29 | ups | in voicemail.conf ? |
| 09:03.37 | atan | [7940] ... mailbox=1000@voicemail is in sip.conf |
| 09:03.51 | atan | the voicemail thinger says [voicemail] |
| 09:03.51 | atan | 1000 => 1234, atan, user@addr |
| 09:03.53 | kaldemar | double check that you see a mailbox defined in sip show peer 7940 |
| 09:04.17 | atan | Oh? This [voicemail] thing actually means something within voicemail.conf? |
| 09:04.19 | atan | I did wonder about that |
| 09:04.23 | ups | then use @voicemail |
| 09:04.37 | atan | will try, one sec |
| 09:05.29 | atan | moving it to exten => *98,1,VoiceMailMain(${SIPPEER(mailbox)@voicemail}) didn't change it |
| 09:05.43 | kaldemar | atan: that's plain wrong |
| 09:05.53 | atan | =S |
| 09:06.36 | atan | show peer 7940 shows Mailbox : 1000@voicemail |
| 09:07.44 | atan | I switched the exten back to exten => *98,1,VoiceMailMain(${SIPPEER(mailbox)}) |
| 09:08.18 | ups | exten => 6245,1,Set(SIPID=${SIPCHANINFO(peername)}) exten => 6245,2,Set(SIPMAIL=${SIPPEER(${SIPID}:mailbox)}) exten => 6245,3,VoiceMailMain(${SIPMAIL}) exten => 6245,4,Hangup() |
| 09:09.24 | atan | Sweet. |
| 09:09.27 | kaldemar | it was missing the peer name. |
| 09:09.28 | atan | That works beautifully! |
| 09:09.43 | atan | Why is ups more advanced than mine? |
| 09:09.58 | kaldemar | that can be done with only one line too, skipping the Set's. |
| 09:10.09 | atan | Awesomeness. |
| 09:10.16 | atan | Can you tell a rule to look for or? Like |
| 09:10.22 | atan | *98|*97|*99 |
| 09:11.52 | kaldemar | ?? |
| 09:12.24 | atan | So it would match wither *97, or *98.. and so on |
| 09:12.32 | atan | instead of me putting the rule 3 times for 3 different numbers |
| 09:13.36 | kaldemar | exten => _*9[7-9],1,VoiceMailMain(${SIPPEER(${SIPCHANINFO(peername)},mailbox)}) |
| 09:13.48 | atan | A regex... creative |
| 09:13.50 | atan | nice =D |
| 09:14.38 | kaldemar | actually not a regexp, but a pattern. |
| 09:15.33 | atan | Well that's just too cool. =) thank you kindly for your expertise |
| 09:15.51 | atan | same to you there ups |
| 09:16.07 | ups | np |
| 09:16.13 | atan | I suppose there is one other question I have about it |
| 09:16.19 | atan | Why do voicemails need a name, and a number? |
| 09:16.32 | atan | like in voicemail I have [voicemail] |
| 09:16.41 | atan | and [other] [general] and so on |
| 09:16.44 | atan | What's the big deal? |
| 09:16.48 | atan | Why not just have numbers? |
| 09:17.05 | atan | Is that so if you were moving users over to a new system you could change the name(folder) and retain the mailbox numbers? |
| 09:17.11 | ups | so u could use voicemail box numbe 1 numerous of times but in different contexts :) |
| 09:17.46 | atan | I see. Like, for example, daytime voicemail, after hours voicemail, and so on? |
| 09:17.53 | atan | Keep different messages on them and rotate it around as needed? |
| 09:18.15 | ups | this can be done in one context |
| 09:18.33 | ups | i see it as a way to differentiate between companies |
| 09:22.00 | atan | Can the user change the number of rings before it goes to voicemail? Does it touch that file at all? |
| 09:22.11 | atan | For some reason I didn't think it did. So I assume it's unchanged? |
| 09:22.14 | ups | nope |
| 09:22.20 | atan | k |
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| 09:28.53 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:36.27 | atan | Woah doggy! Using VoiceMail(1@1000) doesn't play the greeting, hmm |
| 09:36.50 | atan | Is there a PlayBack(vm-greeting) or something? +S |
| 09:36.51 | atan | =S |
| 09:39.28 | *** join/#asterisk Phlog (~Phlogisto@mjoelnir.net.in.tum.de) |
| 09:40.10 | Phlog | Hi, I wonder whether ZRTP passthrough has made it into asterisk 1.8 ? |
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| 09:53.10 | atan | Well I'm miffed. Voicemail won't play the greeting. |
| 09:53.19 | krion | anyone with a link in order to stress test an asterisk ? |
| 09:53.20 | atan | It lets you record one, but it won't play it back to the caller |
| 09:53.23 | krion | with linux client |
| 09:53.30 | krion | i want to reproduce deadlock |
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| 10:00.58 | *** join/#asterisk gled (~gled@212.99.103.3) |
| 10:15.43 | atan | chan_ooh323.c:3161: error: expected ')' before string constant |
| 10:15.52 | atan | I can't make asterisk-addons |
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| 10:45.35 | Uzzi | I want to use askterisk with analogic line, is it possible? |
| 10:49.59 | fauxalliance | Uzzi, of course... you need an ATA or an interface card.... with either FXO or FXS ports respectively. |
| 10:50.10 | fauxalliance | ~fxo |
| 10:50.10 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 10:53.09 | Uzzi | TDM410 with 2fxo 1voice number and 1 fax number it's possibile? hylafaxserver works with this card? |
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| 10:56.10 | kjs | sigh doing voice prompt recording is painful |
| 10:59.28 | tzafrir_laptop | atan, what version of asterisk? asterisk-addons? |
| 10:59.44 | atan | tzafrir_laptop, I ended up getting it all fixed up & compiled |
| 10:59.49 | atan | Just working out how to setup my config |
| 10:59.55 | atan | I want to log everything to the database |
| 10:59.59 | atan | inbound, outbound, and so on |
| 11:00.03 | tzafrir_laptop | atan, what versions? What did you need to fix? |
| 11:00.20 | atan | 1.8, and I just installed the prereqs |
| 11:00.23 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 11:00.29 | atan | Suppose nothing was broken ^_^ |
| 11:05.16 | tzafrir_laptop | atan, in 1.8 the addons are included in the the main tarball |
| 11:05.28 | tzafrir_laptop | Is that what you use? |
| 11:05.33 | atan | tzafrir_laptop, I was just apt-getting it all I didn't grab the tarball |
| 11:05.43 | atan | Foolish me, yes, I know =) |
| 11:05.45 | atan | <3 the apt-get |
| 11:06.28 | tzafrir_laptop | atan, asterisk provides module compatibility only within the same major version |
| 11:06.41 | atan | Now I just have issues with connecting |
| 11:06.54 | atan | <PROTECTED> |
| 11:06.58 | atan | but that is my database server |
| 11:07.22 | atan | server cannot see it though, which is messed up |
| 11:07.23 | atan | hmm |
| 11:07.33 | atan | will get to that in a bit |
| 11:10.07 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 11:10.10 | zplinux | hi all |
| 11:10.20 | atan | Hey rockstar |
| 11:11.10 | zplinux | when I route a call from one pbx to another pbx, should I see the channel the call got on the remote pbx? |
| 11:11.22 | zplinux | both pbx are connected via sip |
| 11:11.41 | zplinux | in pbx I use a trunk rule to route the cal the the other pbx |
| 11:12.17 | zplinux | If the other pbx replays with the ID the call got over there - it is to trace the call |
| 11:12.26 | zplinux | assuming both [bx are my |
| 11:12.39 | zplinux | s/[bx/pbx |
| 11:12.45 | zplinux | s/[bx/pbx/ |
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| 11:18.39 | atan | Is there a simple way to log to mysql the account login used to make call, and the inbound number? |
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| 11:25.54 | Phlog | hi all, does anybody know whether ZRTP pass throug has made it into asterisk 1.8? |
| 11:26.02 | gnuday | What NAT traversal technology does asterisk support? i.e. STUN, ICE, TURN etc. Thanks. |
| 11:36.29 | fauxalliance | gnuday, astertisk has more of a symmetric RTP really... the UA tends to support the fancy stuff. (nat traversal) |
| 11:37.05 | *** join/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 11:37.24 | fauxalliance | Asterisk can be used as a gateway in order to avoid SIP NAT Traversal! |
| 11:38.01 | fauxalliance | strokes the NAT friendly pet IAX/2 |
| 11:38.32 | gnuday | IAX/2 still requires a port forwrding though ? |
| 11:39.01 | fauxalliance | IAX2 uses just one UDP port, 4569 |
| 11:39.44 | gnuday | I need a zeroconf solution for home users connecting behind an ADSL router/modem |
| 11:40.33 | fauxalliance | good luck with that. |
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| 11:43.00 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 11:43.26 | gnuday | If I use a client such as xlite which supports ICE, TURN, STUN etc, ( and this feature is switched on). will it do NAT traversal out of the box if they are asymmetrically NATEDed. (with asterisk 1.6) |
| 11:43.26 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 11:43.43 | gnuday | i.e. just the sip client behind a router... |
| 11:44.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:44.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:45.07 | jpmcallister | gnuday: if just the sip client is behind nat you should not have any problem |
| 11:46.17 | jpmcallister | gnuday: I have a setup like that and have no problem with connecting sip clients to my asterisk box |
| 11:47.30 | gnuday | Can I ask what sip soft clients did you use? What method of NAT traversal? What version of asterisk? Sorry for all the questions! |
| 11:48.09 | jpmcallister | gnuday: Sip clients: zoiper, sjphone, xlite Asterisk 1.6 |
| 11:48.33 | jpmcallister | my asterisk server is nated behind a firewall. I Just define the sip users as dynamics and nat=yes |
| 11:50.03 | jpmcallister | of course I dnat ports 5060, 10000-20000, and 4569 at the firewall |
| 11:50.40 | gnuday | Were your users also behind a router (/ADSL modem) Did you manage to not have to set any port forwarding up at all at the user's end? Thanks |
| 11:51.17 | jpmcallister | yes, not port forward at the user end. Just define nat=yes at the sip user definition |
| 11:51.52 | jpmcallister | and all my users are behind routers. |
| 11:52.22 | jpmcallister | Nobody is directly connected to the internet |
| 11:52.37 | gnuday | Thanks. What NAT detection and traversal method was set in the soft clients? e.g. ICE? |
| 11:52.48 | jpmcallister | and that setup worked even with asterisk 1.4 |
| 11:53.06 | jpmcallister | gnuday: none |
| 11:53.21 | jpmcallister | gnuday: I just configure zoiper without worring about that |
| 11:53.31 | jpmcallister | I guess * takes care of that for me |
| 11:53.53 | gnuday | Cool. Thanks, that's great much obliged :-) |
| 11:54.14 | jpmcallister | gnuday: it just works. nat was never an issue |
| 11:54.39 | gnuday | Great, very grateful for all your help |
| 11:54.42 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 11:56.06 | kjs | hmm, I am using a snom 360, when a call comes in and it's still ringing I can hit transfer and then type the extension I want to transfer it to and it works fine. If i pick up the call and then try and transfer it, * drops the call with the following error " == Spawn extension (phones, 123, 1) exited non-zero on 'SIP/194.145.191.131-000000a4'" |
| 11:56.23 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:03.57 | ukine_work | anyone know a quick-start guide or tut for *now / freepbx? |
| 12:06.49 | atan | I want some way to flag calls using the mysql logger so I know who is responsible for what charges. Is there such a way? Like perhaps I can add a line to the inbound extension + add a line to outbound for that account?? |
| 12:08.27 | ukine_work | also, can asterisk dialout on google voice? |
| 12:08.43 | russellb | using 1.8, yes |
| 12:09.11 | kjs | any ideas why my calls wont transfer? |
| 12:09.13 | russellb | ~googlevoice |
| 12:09.26 | fauxalliance | ~gv |
| 12:09.26 | infobot | . URL: http://wino.physik.uni-mainz.de/~plass/gv/ |
| 12:09.32 | russellb | infobot: googlevoice is <reply> For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 12:09.32 | infobot | okay, russellb |
| 12:10.23 | leifmadsen | russellb: you so smooth |
| 12:10.42 | russellb | that's how i roll |
| 12:11.40 | ukine_work | ty russellb |
| 12:11.45 | russellb | you're welcome. |
| 12:13.35 | atan | Ahuh! accountcode win. <3 |
| 12:14.36 | atan | Can I specify an accountcode on an inbound extension so it will count ring time/voicemail/transfers & such? |
| 12:15.45 | atan | SetAccount([account]) ^_^ rockin! |
| 12:18.06 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 12:19.05 | atan | Oh wait it's not working as I expected... I was thinking exten => 8662102157,n,SetAccount(9999) would cause that 9999 account code to get logged but it's not |
| 12:19.07 | atan | hmm |
| 12:19.30 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 12:20.51 | atan | perhaps the wrong order |
| 12:21.55 | *** join/#asterisk _omer (~omer@119.158.52.249) |
| 12:22.12 | _omer | hello |
| 12:22.28 | _omer | any suggestion to make it possible ? http://www.pastebin.ca/1980856 |
| 12:22.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:23.54 | russellb | atan: that looks like ancient syntax ... try Set(CHANNEL(accountcode)=whatever) |
| 12:24.20 | atan | russellb, you got it |
| 12:24.31 | atan | I used Set(CDR(accountcode)=9999) |
| 12:24.40 | russellb | ah, i was close |
| 12:25.27 | atan | Now I just need to figure out why it generates two log entries |
| 12:25.57 | atan | The inbound call to the DID gets set properly as far as billing is concerned |
| 12:26.23 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:26.34 | atan | But there is a secone entry fgor the same durations, same disposition, except no dstchannel & dst=s |
| 12:27.30 | atan | I suppose that's the log entry for the outbound entry to the SIP device itself |
| 12:29.46 | ManxPower | atan, a single call can, and usually does, create more than 1 CDR. Depending on your dialplan it could create 3, 4 or even 5 CDRs |
| 12:30.07 | atan | ManxPower, well I suppose that would make sense since it gets bounces around so much |
| 12:30.35 | atan | I need to figure out an effective way to capture the $$ packets here so I know who to bill for what =D |
| 12:30.48 | atan | No huge amounts, so I'm not really at a great loss if I fail. Maybe $5 a year. hah. |
| 12:31.30 | atan | If I catch them on the inbound DID, and catch them on the outbound calls... that should just about cover me |
| 12:32.09 | atan | The in between action can go unnoticed I guess |
| 12:32.45 | ukine_work | russellb, how do i update to 1.8 from the 1.6 in *now? |
| 12:33.15 | ukine_work | replace files in /etc/asterisk with those in asterisk-1.8.0.tar.gz ? |
| 12:33.20 | ManxPower | ~asterisknow |
| 12:33.20 | infobot | somebody said asterisknow was based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 12:34.22 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 12:34.31 | *** join/#asterisk PMantis (~sswitzer@cpe-74-74-216-216.rochester.res.rr.com) |
| 12:34.44 | PMantis | WIMPy, ping |
| 12:35.17 | kjs | hmm calls are transfering ok from linksys spa942's but not snom 360's... |
| 12:36.15 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |
| 12:36.26 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 12:40.00 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
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| 12:42.30 | schmidts | kjs no suprise at all :D |
| 12:42.54 | kjs | schmidts: really? |
| 12:43.27 | kjs | are snoms the suck then? |
| 12:43.39 | schmidts | kjs IMHO yes |
| 12:44.34 | kjs | i admit the buttons on this thing do rage me, double pressing numbers. It also randomly doesn't dial out first time as well, yet if you hang up and redial it then works... |
| 12:45.10 | kjs | maybe ill just swap it out. :) |
| 12:48.56 | atan | What is this users.conf file? does it auto-generate stuff that is uneeded? |
| 12:49.27 | ManxPower | ~usersconf |
| 12:49.33 | ManxPower | ~users.conf |
| 12:49.33 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 12:49.49 | russellb | infobot: forget users.conf |
| 12:49.49 | infobot | russellb: i forgot users.conf |
| 12:50.00 | russellb | that's not helpful in the least bit |
| 12:50.23 | ManxPower | atan, Asterisk does not "auto-generate" config files. |
| 12:50.27 | [TK]D-Fender | russellb: It doesn't suggest a direct course of action.... |
| 12:50.52 | russellb | atan: it's alternative configuration mechanism that is suitable for a small and simple PBX installation. The Asterisk GUI uses it. |
| 12:51.14 | atan | Righto then. Can one delete users.conf without any ill effects? |
| 12:51.24 | atan | I'd prefer to setup an extension/sip and go that way |
| 12:51.46 | russellb | atan: yes. |
| 12:52.02 | ManxPower | ~toolkit |
| 12:52.02 | infobot | Remember, Asterisk isn't really a PBX: Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 12:52.09 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 12:53.19 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 12:55.10 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:59.19 | PMantis | atan, You're back already? :) |
| 12:59.28 | atan | PMantis, I never left |
| 12:59.45 | atan | I've been at it all night. Love this Asterisk thing. |
| 12:59.52 | PMantis | Haha |
| 12:59.54 | atan | Just setting up a second debian box to fool around on. |
| 12:59.59 | PMantis | True geek |
| 13:00.24 | atan | I'm out about 2l of coke & 1/4 a monster energy drink right now but... oh well. I need to be up until UPS shows up. |
| 13:00.32 | atan | I went to school (classes started at 8) and all my teachers are not there |
| 13:00.42 | atan | They decided they would just write it on the board for us to see. |
| 13:00.44 | atan | And not tell us in advance. |
| 13:00.45 | atan | Buggers. |
| 13:01.03 | atan | All is well though =) very nice treat |
| 13:01.05 | PMantis | 's Aastra phones are finally NOT disconnecting after 45 minute calls. |
| 13:01.27 | atan | Over night I switched my cisco phone to SCCP and hated every second of it |
| 13:01.37 | PMantis | lol |
| 13:01.48 | atan | That was totally a wasted 4 hours messing with that |
| 13:01.56 | atan | 4 hours Cisco stole from me, that I will never get back |
| 13:02.07 | atan | Needless to say it was back to the SIP image pretty quickly |
| 13:02.16 | atan | Too bad it doesn't support the expansion module I just ordered =( |
| 13:02.17 | petern_ | SCCP works nicely for me |
| 13:02.48 | PMantis | Cisco is a wonder-ful company... you wonder where they come up with such assenine protocols. |
| 13:02.59 | atan | petern_, I suppose if I were to dedicate the time to learning how to use it perfectly... I'd be okay. It does seem like an interesting concept but I can't wrap my head around how it's supposed to authenticate over the internet semi-securely |
| 13:03.09 | [TK]D-Fender | PMantis: On your Aastra issue, did you actually find the cause? |
| 13:03.13 | petern_ | it's not |
| 13:03.51 | PMantis | [TK]D-Fender, I might have to turn off some of my settings and try again to be sure, because the last test I changed 2 things, then tested.... but I have it narrowed down. |
| 13:03.52 | petern_ | it's meant to be on a segregated LAN |
| 13:04.30 | PMantis | [TK]D-Fender, I have a hunch that it was an Aastra config option. |
| 13:04.47 | petern_ | SIP is indeed better for going across the 'net |
| 13:05.05 | [TK]D-Fender | PMantis: StopPissingMeOff=yes? ;) |
| 13:05.16 | PMantis | [TK]D-Fender, Yup! hah |
| 13:05.51 | PMantis | [TK]D-Fender, This should perhaps be expanded, but I found my clue here: http://www.voip-info.org/wiki/view/Aastra |
| 13:06.06 | PMantis | [TK]D-Fender, 1/2 way down: "You should avoid using "sip session timer" with any value other than 0" |
| 13:06.59 | PMantis | [TK]D-Fender, That wasn't defined in my configs. I also followed some of WIMPy's suggestions, which also revolved around session timers. |
| 13:07.32 | [TK]D-Fender | PMantis: I've heard of several issues with *'s lack of proper support for this. |
| 13:08.43 | PMantis | [TK]D-Fender, What's the theory here? devices supposed to send "pings" in SIP or RDP to ensure that the endpoints are up? |
| 13:09.20 | [TK]D-Fender | PMantis: I don't actually know the technical details for this unfortunately... I jsut remember the ranting some have had over it.... |
| 13:09.29 | PMantis | k |
| 13:09.34 | [TK]D-Fender | PMantis: few isolated cases, but the frustration factor was pretty massive |
| 13:10.03 | PMantis | [TK]D-Fender, Yeah! Could barely find anything on it. |
| 13:10.18 | PMantis | I'm writing an article on everything I did for my website. |
| 13:11.18 | [TK]D-Fender | PMantis: I'd recommend adding to voip-info & the new * wiki |
| 13:11.41 | PMantis | [TK]D-Fender, I'll likely add a blurb there, too. |
| 13:12.54 | *** join/#asterisk Delido1983 (~Delido198@web.its-medienservice.de) |
| 13:14.39 | Delido1983 | Hello, i have an problem with the callerid(name) and spezial chars like äöüÃ. The Display dont show the name if there is one of this chars.. |
| 13:15.21 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 13:15.34 | fauxalliance | Delido1983, does the display support the umlauts? |
| 13:16.39 | fauxalliance | s/umlauts/umlauts and es-tsets |
| 13:16.49 | Delido1983 | fauxalliance: I think so, all settings are in UTF-8 and in the local addressbook i can save names with spezial chars, but realy i dont know if the phone support utf-8 URI-Strings |
| 13:17.06 | fauxalliance | ^^check it out |
| 13:17.30 | Delido1983 | mhm okay i search again... :D |
| 13:19.05 | Katty | good morning! |
| 13:19.24 | fauxalliance | hands Katty a steaming cup of sunshine. |
| 13:19.25 | Katty | if anyone is interested in participating in the Asterisk Christmas Card Exchange, please /query me |
| 13:19.36 | Katty | fauxalliance: :>> |
| 13:19.37 | atan | Guys, are there any super-cool Asterisk add-ons I'm missing? Goodness knows what exactly. You know, like the CFL bulb is to the incandescent =) |
| 13:19.38 | Katty | hugs fauxalliance |
| 13:19.49 | atan | Katty, this sounds interesting! |
| 13:20.55 | atan | Where does asterisk stash sound files by default? |
| 13:21.02 | Katty | var lib asterisk sounds en |
| 13:21.10 | tzanger | heh |
| 13:21.19 | tzanger | I thought you were making fun of canadians, I read that "var lib asterisk sounds, eh?" |
| 13:21.35 | Katty | :P |
| 13:21.54 | atan | ^_^ abandon-all-hope.wav |
| 13:22.44 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 13:23.03 | McBoingbo | does context = from-<bla> have special meaning? |
| 13:23.07 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 13:24.49 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 13:25.39 | ManxPower | McBoingbo, only in GUI addons which are not supported here. |
| 13:26.19 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 13:26.32 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 13:27.08 | fauxalliance | [TK]D-Fender, I can't send you illicit drugs, ugh, a christmas card if you don't sign up. |
| 13:27.10 | McBoingbo | oh the article I was reading about asterisk dual servers was talking about editing sip.conf which in GUI is typically not done |
| 13:27.42 | [TK]D-Fender | McBoingbo: A name is a name. Might as well be ^fredç for all * cares |
| 13:27.52 | ukine_work | anyone update *now to 1.8 ? what's the process? (asking again since there's only a handful of people in #asterisknow) |
| 13:28.25 | [TK]D-Fender | ukine_work: rad their channel TOPIC '- |
| 13:28.29 | [TK]D-Fender | ??? |
| 13:28.31 | [TK]D-Fender | read* |
| 13:28.42 | Katty | [TK]D-Fender: you going to participate in the christmas card thing? |
| 13:29.00 | ukine_work | [TK]D-Fender, ty, man i feel dumb now :} |
| 13:29.08 | fauxalliance | ukine_work, that's his job |
| 13:29.29 | ukine_work | hah |
| 13:30.39 | [TK]D-Fender | [08:21]=-=Topic for #asterisknow is âAsteriskNOW 1.7.1 has been released (2010/09/14): http://www.asterisknow.org/downloads and http://blogs.digium.com/2010/09/14/asterisknow-171-add-on-module/ -=- Switching to Asterisk 1.8: http://forums.asterisk.org/viewtopic.php?t=75822â |
| 13:30.47 | [TK]D-Fender | [09:27]<ukine_work>anyone update *now to 1.8 ? what's the process? |
| 13:31.08 | [TK]D-Fender | Something we hide it in the BIG PRINT :p |
| 13:31.18 | [TK]D-Fender | sometimes* |
| 13:31.19 | [TK]D-Fender | gah |
| 13:31.48 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:31.58 | *** join/#asterisk lep (~lep@static-217-133-61-144.clienti.tiscali.it) |
| 13:32.30 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:38.20 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 13:38.52 | Weazel | hey guys, i got an annoying problem where my snom phones drop register and other ip phones won't |
| 13:39.01 | Weazel | can someone help with that area ? |
| 13:39.48 | aberrios_ | any output from the * CLI? |
| 13:40.00 | atan | When I make an outbound call to a regular landline I can hang up on the SIP phone while it's ringing but the other device keeps ringing |
| 13:40.09 | Weazel | well from the cli I just see it as unreachable |
| 13:40.09 | atan | Like if I call my cellphone, let it ring once, then hang up, the phone keeps ringing |
| 13:40.11 | ManxPower | Hey, I need help with Postfix! I was on the postfix channel but nobody was helping me so I decided to come here! |
| 13:40.42 | Weazel | i think its something with my network but i don't know how to find it since i've opned any/any on my firewall |
| 13:40.59 | ManxPower | Weazel, set qualify=yes for the problem sip peer |
| 13:41.28 | Weazel | ManxPower: can u set it from freepbx ? |
| 13:41.29 | fauxalliance | ManxPower, sup with the MDA? |
| 13:41.43 | Delido1983 | fauxalliance: in the datasheet from the soundpoint ip 320 is written "Support Unicode UTF-8" |
| 13:42.04 | Weazel | ManxPower: its already set to Yes in the Freepbx |
| 13:42.38 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 13:42.50 | atan | exten => NXXXXXX,1,Dial(SIP/voipms/1902${EXTEN}) isn't playing nice with 7 digit numbers =\ |
| 13:43.15 | Katty | russellb: poke |
| 13:44.16 | ManxPower | Weazel, FreePBX is not supported here. |
| 13:44.22 | ManxPower | you can set it, but FreePBX will overwrite it |
| 13:44.38 | ManxPower | Faustov, MDA? |
| 13:44.44 | Weazel | well it is already set to Qualify = yes so it can't be it right? |
| 13:44.54 | ManxPower | atan, try making it a pattern with _ |
| 13:45.05 | ManxPower | Weazel, I don't know. I don't use FreePBX |
| 13:45.36 | ManxPower | I assume Qualify = yes in the PoS GUI you are using will set qualify on the phone, but it might not. who knows? |
| 13:45.37 | McBoingbo | for this article to setup 2 asterisk servers to call each other, the SIP method mentions registration is not needed for static IP's, but then does not go into details about how to setup the trunk, it only talks about IAX so do I simply copy IAX method, I dont think so |
| 13:45.46 | atan | ManxPower, majik! |
| 13:46.03 | Weazel | is there a good network management channel around Freenode ? i have a feeling its something to do with my network setup |
| 13:46.16 | atan | Now just to figure out why the voip phone hanging up doesn't discontinue the call in progress |
| 13:47.14 | atan | Weazel, I always used ##networking |
| 13:48.09 | [TK]D-Fender | McBoingbo: You donèt need to register if they know what HOST to contact you at '------- CLUE |
| 13:48.18 | Weazel | atan: thanks\ |
| 13:49.02 | McBoingbo | [TK]D-Fender: which would indicate I still need to setup peer entries in sip.conf, but this article does not explain this |
| 13:49.44 | [TK]D-Fender | McBoingbo: "this article". Care to actually show us what you're talking about? |
| 13:50.18 | McBoingbo | [TK]D-Fender: oops I thought I pastededed it http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, mai bad |
| 13:51.30 | [TK]D-Fender | McBoingbo: "When the ip of the peer is unknown, a user has no way to place a call (e.g. when an office/user calls a teleworker/peer at home, where the teleworker has only a dynamic ip or is behind NAT.) To compensate for this, the teleworker/peer actively registers with the office/user by providing its identity and ip location. " |
| 13:51.51 | [TK]D-Fender | McBoingbo: It spends the whole article SHOWING you peers, and telling you when you NEED Register. |
| 13:52.00 | McBoingbo | yes I stated I am aware no registration is needed for my setup |
| 13:52.13 | [TK]D-Fender | McBoingbo: McBoingbo Peer = always required to auth calls. |
| 13:52.18 | McBoingbo | so the answer is yes, I use the IAX examples to setup the sip.conf peers |
| 13:52.37 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:52.45 | [TK]D-Fender | McBoingbo: Nobody said you CEASED to need it just because you register. And when you register... you are registering to THE OTHER SIDE'S PEER |
| 13:53.10 | [TK]D-Fender | Which clearly has to be " |
| 13:53.14 | [TK]D-Fender | host=dynamic" |
| 13:53.23 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-oehsdayxkgmqhbbn) |
| 13:53.58 | McBoingbo | Im confused now lol |
| 13:54.19 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 13:54.20 | ManxPower | McBoingbo, have you read The Book? |
| 13:54.24 | ManxPower | ~answers |
| 13:54.24 | infobot | [~answers] 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 |
| 13:54.34 | asteriskmonkey | has anyone got dahdi to work in a freebsd jail? |
| 13:54.41 | McBoingbo | guess I am gonna have to |
| 13:54.50 | ManxPower | it should have been the first thing you did |
| 13:55.15 | [TK]D-Fender | asteriskmonkey: How can you jail something that requires kernel & hardware access |
| 13:55.28 | ManxPower | You will need to read the UPGRADE*.txt files that come with Asterisk so you can translate the book (based on asterisk 1.4) to whatever version of Asterisk you are using. |
| 13:55.33 | *** join/#asterisk BONO (~Bruno@187.59.136.96) |
| 13:55.36 | *** part/#asterisk BONO (~Bruno@187.59.136.96) |
| 13:56.14 | ManxPower | ~toolkit |
| 13:56.14 | infobot | Remember, Asterisk isn't really a PBX: Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 13:57.11 | [TK]D-Fender | McBoingbo: When you want to contact a peer you have to know WHERE to send the call. If you KNOW the IP/host you can specify it in host=WHERETOFINDTHEM. |
| 13:57.50 | [TK]D-Fender | McBoingbo: If they are a MOVING TARGET (dynamic IP), they can REGISTER to your server (YOU obvious have to be a FIXED TARGET). This informs * WHERE to send the calls to. |
| 13:57.58 | Katty | Nugget: hippo birdie two ewe. |
| 13:58.02 | Katty | Nugget: hippo birdie two ewe. |
| 13:58.08 | [TK]D-Fender | It still has NOTHING to do with authing the actual calls. |
| 13:58.08 | Katty | Nugget: hippo birdie deer ewe. |
| 13:58.13 | Katty | Nugget: hippo birdie two ewe. |
| 13:58.19 | ManxPower | .kick Katty |
| 13:58.45 | Katty | ManxPower: you should wish Nugget a happy birthday instead. |
| 13:59.04 | [TK]D-Fender | Nugget: Happy telnet to you! |
| 13:59.05 | Nugget | telnet is eeeeeeevil! |
| 13:59.07 | [TK]D-Fender | :D |
| 13:59.10 | [TK]D-Fender | PWNED |
| 13:59.21 | ManxPower | Katty, or you could stop spamming the channel with nonsense |
| 13:59.41 | [TK]D-Fender | ManxPower: hukt on fonix faled 4 u! |
| 14:00.44 | Katty | ManxPower: i will stop spamming 4 lines (= |
| 14:00.49 | Katty | ManxPower: but i will not stop being nice to people |
| 14:00.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:00.55 | asteriskmonkey | [TK]D-Fender: you can jail asteirsk nicley, http://www.voip-info.org/wiki/view/Installing+Asterisk+In+A+FreeBSD+Jail , jailing is not para or full virtualization its kernel segmentation/security locking |
| 14:02.48 | [TK]D-Fender | asteriskmonkey: Asterisk != DAHDI |
| 14:03.28 | [TK]D-Fender | asteriskmonkey: Then again if the guide shows you how to make it work through the HV, go for it |
| 14:03.32 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 14:04.06 | ManxPower | sends Nugget a sympathy card |
| 14:04.38 | kjs | does ENUM cause problems in general? |
| 14:05.14 | [TK]D-Fender | kjs: Problems with what? |
| 14:05.57 | *** join/#asterisk chigambamukoko (~smokesmok@fl-76-5-60-186.dhcp.embarqhsd.net) |
| 14:06.11 | kjs | I am wondering if it's going to cause problems with calls, I have not used it before. Just checking for responses like "Don't use it, it's a pile of crap" |
| 14:06.55 | ManxPower | kjs, use it if you need it, don't if you don't. It is far more work than other methods |
| 14:07.03 | [TK]D-Fender | kjs: Stop fishing for problems and jsut deal with the ones you HAVE :p |
| 14:07.05 | chigambamukoko | anyone looking to make a few extra $ by helping me with Fax on asterisk, |
| 14:07.24 | kjs | [TK]D-Fender: ;) |
| 14:07.35 | ManxPower | chigambamukoko, Free advice: don't run fax over voice over ip over internet |
| 14:08.08 | chigambamukoko | many have gotten it to work, |
| 14:08.40 | chigambamukoko | I'm sure I am missing a few steps, since I'm not that savvy |
| 14:08.49 | Katty | chigambamukoko: it's advice. |
| 14:08.55 | Katty | chigambamukoko: it's something you take, or leave. |
| 14:08.59 | chigambamukoko | thanks friend |
| 14:09.32 | ukine_work | Asterisk communicates with Google using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant on the iksemel library files as well as the OpenSSL development libraries presence on your system. |
| 14:09.38 | chigambamukoko | when using g711, I can only get about 70% success, I need to set up for t.38 |
| 14:09.55 | chigambamukoko | my upstream do support t.38 |
| 14:10.11 | ukine_work | where do i find chan_gtalk, res_jabber, and iksemel/openssl ? |
| 14:11.58 | pabelanger | ukine_work: the first two are part of asterisk, the other 2 are libraries |
| 14:12.53 | kjs | anyone got an example of musiconhold.conf for asterisk 1.8 the wiki version is out of date |
| 14:13.16 | [TK]D-Fender | kjs: Look in your SAMPLES folder |
| 14:13.24 | ukine_work | package openssl already installed and latest version |
| 14:13.42 | pabelanger | ukine_work: openssl-dev? |
| 14:13.46 | ukine_work | is there a yum install foo for iksemel i wonder? |
| 14:14.01 | pabelanger | ukine_work: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google or http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ |
| 14:15.54 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 14:17.09 | Katty | hugs coppice |
| 14:20.01 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:21.10 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:23.30 | atan | When I set a voicemail greeting using the voicemail program by calling it it does record it, but does not play it to future callers |
| 14:23.36 | atan | Any ideas where I went wrong? |
| 14:24.51 | pabelanger | atan: *CLI> core show application Voicemail |
| 14:25.48 | *** part/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 14:26.06 | atan | pabelanger, when I run that command it just gives me the doc on how to use it |
| 14:26.21 | pabelanger | correct |
| 14:26.29 | p3nguin_ | haha |
| 14:26.37 | p3nguin_ | Imagine that! |
| 14:26.40 | SuPrSluG | atan: what does your dialplan tell it to play? |
| 14:26.51 | pabelanger | you need to pass the busy or unavailable flags |
| 14:26.52 | leifmadsen | SuPrSluG: hey I met you! :) |
| 14:27.19 | SuPrSluG | liefmadsen: u goin to the bears game lol |
| 14:27.31 | leifmadsen | heh nah :) |
| 14:27.41 | leifmadsen | I'm flying to Punta Cana tomorrow :) |
| 14:27.45 | atan | And right now it doesn't do any of that. Just, VoiceMailMain(${SIPMAIL}) |
| 14:28.18 | atan | So I assume there is a voicemail greeting then, eh? VoiceMailGreeting() or something |
| 14:28.51 | atan | Sorry, VoiceMail(1@1000) even |
| 14:28.51 | ManxPower | atan, read the Asterisk book. |
| 14:29.40 | ManxPower | <atan> So I assume there is a voicemail greeting then, eh? VoiceMailGreeting() or something <--- indicates you have not read the book or the UPGRADE*.txt files |
| 14:29.51 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:30.29 | atan | No =\ I have not =/ |
| 14:31.02 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:34.42 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 14:36.00 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 14:36.42 | *** join/#asterisk Baylink (~jra@65.34.94.26) |
| 14:37.41 | Baylink | QUERY: I'm trying to build libiax against speex-devel-1.1.999_1.2rc1-5.1.i586 (and the related speex and libogg, packaged for SuSE 11.3), and I'm getting this error: |
| 14:37.44 | Baylink | audio_encode.c: In function âiaxc_input_postprocessâ: |
| 14:37.44 | Baylink | audio_encode.c:146:5: error: dereferencing pointer to incomplete type |
| 14:38.14 | pabelanger | atan: If you read the output of *CLI> core show application Voicemail it will explain how to use it properly |
| 14:38.17 | Baylink | while building audio_encode.c. Google says that's because I need a beta1 or newer speex... which theoretically, rc1 should be. |
| 14:38.28 | Baylink | Any other suggestions? |
| 14:39.19 | jblack | baylink: It could be the headers you have in /usr/include are from the older ones, or /usr/local/include |
| 14:39.34 | [TK]D-Fender | atan: "core show application voicemail" |
| 14:39.48 | Baylink | jblack: Good point; I'll check. |
| 14:39.58 | atan | Yes, I see the options list =) |
| 14:40.10 | atan | Was just pecking around on another doc about faxes |
| 14:40.50 | Baylink | Irritatingly enough, jblack; the headers don't have *version numbers* in them. I did get the -devel, though, and previously had *no* speex, or so the build told me, so... |
| 14:40.59 | Baylink | Wait, /u/l/i |
| 14:41.14 | Baylink | Nope, nothing there, either. |
| 14:41.47 | jblack | you're including the codec from -somewhere-. =) |
| 14:41.54 | Katty | jblack: <3 |
| 14:42.00 | jblack | Hey Katty. =) |
| 14:42.06 | Katty | jblack: how're you feeling dear |
| 14:42.26 | jblack | I'm sad about ssdi, but made up for it by ordering a 3d monitor. |
| 14:42.38 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:42.54 | jblack | How about you? |
| 14:43.05 | Katty | content and happy (= |
| 14:43.08 | jblack | baylink: How about in-tree? |
| 14:43.09 | Baylink | jblack: Oh yes: i've installed it *now*. |
| 14:43.21 | jblack | Oh, so you're good. |
| 14:43.29 | jblack | Katty: How is your rodent feeder doing? |
| 14:43.46 | Baylink | I'm working with an SVN release-tag build tree from Xelatec; their XIPPR iaxRpt console. They build with it, so I assume the tree is clean. |
| 14:43.51 | Katty | jblack: rodent free, sadly, but the birds visit regularly |
| 14:44.12 | jblack | I'd rather bird flu than rabies. =) |
| 14:44.22 | Katty | :P |
| 14:45.54 | *** part/#asterisk Delido1983 (~Delido198@web.its-medienservice.de) |
| 14:46.00 | Katty | jblack: ready for thanksgiving? |
| 14:46.37 | jblack | I don't have any plans. I'm having a big party the week before |
| 14:47.11 | jblack | I have this irrational fear that peple will confuse me for the turkey, and I'll be stuffed, cooked, and served. |
| 14:47.26 | Katty | LOL |
| 14:49.22 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:49.52 | jblack | How about you? Didja get stuck with the cooking? |
| 14:54.14 | *** join/#asterisk sbszulu (~dundubala@41.16.235.43) |
| 14:54.40 | Katty | jblack: attending a potluck this year. |
| 14:54.45 | Katty | jblack: so no one has to cook a lot at all |
| 14:54.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:54.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:55.34 | kjs | jblack: yeah rabies will kill you. |
| 14:56.12 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 14:56.18 | Katty | rabies won't always kill |
| 14:56.25 | *** part/#asterisk Baylink (~jra@65.34.94.26) |
| 14:56.26 | Katty | if it's caught early, it can be handled |
| 14:56.30 | Katty | but there will still probably be some brain damage |
| 14:56.43 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:00.57 | jblack | It's still rabies. |
| 15:01.25 | jblack | Can you imagine telling a (boy/girl)friend "btw hon, I should let you know that I have rabies. literally. Yes. The mouth foaming thing" |
| 15:01.41 | coppice | yeah, but one way your dead, and the other way you can still have a career in politics |
| 15:01.52 | coppice | s/your/you're/ |
| 15:01.54 | ManxPower | rabies *ALWAYS* kills by the time you have symptoms. |
| 15:01.55 | jblack | either way, you're not getting laid |
| 15:02.10 | jblack | ManxPower: I don't think that's true any more |
| 15:02.46 | ManxPower | jblack, other than a single patient that was put into a coma, they all die once you get symptoms |
| 15:02.58 | kjs | If I place: [default] |
| 15:02.58 | kjs | mode=files |
| 15:02.58 | kjs | directory=moh |
| 15:03.19 | jblack | I thought we had made progress on it |
| 15:03.26 | ManxPower | if you catch it before symptoms happen, most people survive just fine. I suppose I could cite a source but I bet it would be too much work to find it. |
| 15:03.31 | kjs | in musiconhold.conf that should play mp3's placed in /var/lib/asterisk/moh/ correct |
| 15:03.32 | coppice | ManxPower: I expect pretty much everyone wishes they were dead once they get the symptoms |
| 15:03.34 | kjs | ? |
| 15:03.55 | Katty | kjs: i think that would be /moh |
| 15:03.56 | jblack | Ok, post symptoms, the survival rate is 8% |
| 15:03.58 | ManxPower | jblack, rabies is a virus, your only hope is to catch it before it crosses the blood/brain barrier (which is a while after infection) |
| 15:04.07 | Katty | kjs: directory=/var/lib/asterisk/moh/ |
| 15:04.13 | p3nguin | kjs: There are very clear and precise samples of mp3 moh in the sample musiconhold.conf file. |
| 15:04.25 | Katty | kjs: if you would like a sample, i can pastebin mine |
| 15:04.49 | ManxPower | jblack, much higher than I thought |
| 15:06.24 | kjs | p3nguin: well I have read that mpg123 is no longer required for mp3 playback, yet the sample conf still contains examples of this... Which is a little confusing |
| 15:06.35 | jblack | There's something called the "maxwell protocol". Go wikipedia |
| 15:06.52 | Katty | September 28th is World Rabies Day |
| 15:06.59 | Katty | so get yourself vaccinated |
| 15:07.20 | jblack | I'd say you're more right than I am, manx |
| 15:07.23 | jblack | A lot more right |
| 15:08.08 | p3nguin | kjs: [mp3] |
| 15:08.08 | p3nguin | mode=quietmp3 |
| 15:08.08 | p3nguin | directory=/var/lib/asterisk/mohmp3 |
| 15:08.13 | p3nguin | kjs: How is that using mpg123? |
| 15:09.01 | jblack | kjs: There's more than one way to do it. =) |
| 15:10.26 | kjs | "does not exist in any format" is what im getting on the CLI |
| 15:10.44 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 15:10.50 | [TK]D-Fender | [11:08]<p3nguin>mode=quietmp3 <-- THIS is how |
| 15:10.54 | ManxPower | kjs, again, you are reading old and outdated docs |
| 15:11.05 | [TK]D-Fender | p3nguin: that = mpg123 |
| 15:11.12 | p3nguin | Oh yeah? |
| 15:11.23 | [TK]D-Fender | YA RLY |
| 15:11.30 | kjs | haha |
| 15:13.04 | p3nguin | Interesting. |
| 15:14.21 | p3nguin | application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s <-- this is what I thought of when I think "using mpg123" |
| 15:14.50 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 15:14.58 | p3nguin | I guess app_mp3.so uses mpg123. |
| 15:16.57 | kjs | http://pastebin.ca/1981805 this should work right? |
| 15:17.36 | p3nguin | It should play wav files just fine. |
| 15:18.03 | [TK]D-Fender | or any file that * has a FORMAT for |
| 15:19.00 | kjs | right, ill just convert it to wav if its going to be easier. |
| 15:19.53 | p3nguin | Easier? EASIER? How hard could it be to install mpg123 and use app_mp3 via mode=quietmp3? |
| 15:20.10 | [TK]D-Fender | kjs: what ver of * are you using? |
| 15:20.18 | kjs | 1.8 |
| 15:20.36 | [TK]D-Fender | kjs: There may be a simple extra step to add MP3 support from within the tarball |
| 15:20.46 | p3nguin | (1608.04) <@Qwell> 1.8 isn't a version |
| 15:20.47 | [TK]D-Fender | kjs: It IS included, but I'm not sure if it's built by default |
| 15:21.11 | kjs | p3nguin: well mpg123 is not in the centos repos so I would have to add an additional repo. |
| 15:21.29 | kjs | plus mpg123 is no longer in dev and has known security holes. |
| 15:22.04 | p3nguin | mpg123.i386 1.12.5-1.el5.rf rpmforge |
| 15:22.22 | kjs | yeah id have to add rpm forge ;) |
| 15:22.23 | ManxPower | p3nguin, mpg123 was deprecated in 1.4, dude. |
| 15:22.30 | Katty | what should i do for lunch today |
| 15:22.39 | kjs | Katty: just thinking the same |
| 15:22.55 | kjs | rabies on toast? |
| 15:23.05 | Katty | that's not funny |
| 15:23.22 | kjs | and not possible either? |
| 15:23.30 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 15:23.53 | p3nguin | Lunch? I barely had breakfast. |
| 15:24.17 | Katty | that's not my fault ;) |
| 15:24.23 | p3nguin | Oh, I know. |
| 15:24.39 | p3nguin | I'm just saying it's quite early for lunch. |
| 15:25.28 | *** join/#asterisk FreezeS (~kvirc@89.238.223.70) |
| 15:26.14 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 15:28.50 | *** join/#asterisk brettnem (~brett@user-0vvd88f.cable.mindspring.com) |
| 15:28.59 | brettnem | Hello all! |
| 15:29.14 | brettnem | long time no chat. :) |
| 15:30.06 | brettnem | has anyone seen DIALTIME and ANSWEREDTIME be completely blank in answered calls? this just popped up and nothing's changed form what I know.. running 1.6.2.12 |
| 15:32.03 | brettnem | pokes group |
| 15:35.37 | beardy | Katty: Scrambled eggs? |
| 15:36.36 | Katty | beardy: decided on chicken sammich. |
| 15:37.15 | beardy | Katty: Also nice. |
| 15:40.12 | *** join/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com) |
| 15:40.22 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:41.21 | *** join/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com) |
| 15:41.57 | *** part/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com) |
| 15:46.09 | tzafrir_laptop | hi, a question about nortel phones (chan_unistim): |
| 15:46.17 | tzafrir_laptop | Where is the time zone set there? |
| 15:46.38 | tzafrir_laptop | Is it something in unistim.conf? (can't find it there) |
| 15:51.05 | brettnem | anyone seen a problem like this? Where DIALEDTIME and ANSWEREDTIME end up empty? :/ |
| 15:53.54 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn210.78-98-252.t-com.sk) |
| 15:57.45 | tzafrir_laptop | brettnem, what's the minimal setting required to reproduce this? |
| 15:57.55 | tzafrir_laptop | Does it always happen, or occasionally? |
| 16:05.03 | Katty | mm, chicken sammich with swiss |
| 16:05.28 | p3nguin | Swiss cheese is nice. |
| 16:05.55 | p3nguin | I like French Imported Swiss from the deli. |
| 16:05.56 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:07.12 | p3nguin | Lorraine cheese is also good. |
| 16:10.06 | *** join/#asterisk goomail (~Adium@r186-48-193-11.dialup.adsl.anteldata.net.uy) |
| 16:14.20 | goomail | Hi everyone! I'm wondering if anyone here might be able to lend a hand with a SIP registration problem I'm experiencing... |
| 16:14.39 | WIMPy | ~ask |
| 16:14.39 | 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. |
| 16:14.42 | p3nguin | Only if you're specific about the problem and describe what you've done to troubleshoot it. |
| 16:17.24 | goomail | Every couple of weeks, my main SIP registration goes awry and I am unable to place outgoing calls through my provider. (Incoming works OK.) The only significant thing I've noticed is that the remote IP under "sip show" has disappeared when this happens. Doing a "sip reload" gets everything working again instantly. I'm running * 1.6.13 on an embedded ar71xx. If anyone cares to click a link, all of the gory details are here: http |
| 16:17.39 | p3nguin | You truncated at http. |
| 16:17.55 | goomail | s/1.6.13/1.6.2.13/ and http://lists.digium.com/pipermail/asterisk-users/2010-October/255458.html |
| 16:19.10 | p3nguin | If your registration is messed up, you probably won't be getting incoming calls anymore. |
| 16:20.16 | goomail | incoming actually works fine from that provider (i've tested it multiple times). when i wrote "registration goes awry", perhaps what i meant to write was "my sip settings for my remote peer goes awry". |
| 16:21.24 | p3nguin | You said you run "sip show", but that shouldn't actually show anything but an error message. What did you actually show? sip show peers? sip show registry? |
| 16:21.43 | thehar | 4/win 14 |
| 16:21.45 | thehar | fail |
| 16:22.06 | p3nguin | I prefer Alt+R for that. |
| 16:22.08 | goomail | sorry. sip show peer <peername> |
| 16:23.09 | *** join/#asterisk DodgeThis (~DodgeThis@bl17-56-108.dsl.telepac.pt) |
| 16:27.55 | kjs | asterisk will only play 8k sample rate wavs ? |
| 16:30.59 | *** join/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 16:31.36 | jpmcallister | tried to install asterisk18 in a 64 bit centos and i'm gettin that message: chan_dahdi.so: undefined symbol: pri_retrieve_ack |
| 16:31.46 | russellb | jpmcallister: update libpri |
| 16:35.09 | jpmcallister | russellb: tank you very much |
| 16:35.13 | russellb | np |
| 16:35.41 | russellb | jpmcallister: which version of libpri did you have installed already? do you know? |
| 16:35.49 | jpmcallister | 1.4.10 |
| 16:35.56 | russellb | thanks |
| 16:36.32 | *** join/#asterisk [T]ank (~chwall@206.71.78.158) |
| 16:38.42 | [T]ank | i have set my localnet, externhost and externrefresh in sip.conf but in sip debug it still shows me coming from my local IP address... is there somewhere else I am supposed to set my ip address so it shows my public IP? |
| 16:40.28 | [T]ank | here is my general section of sip.conf and an example of what it is showing i am coming from. |
| 16:40.29 | [T]ank | http://pastebin.com/6AAci2wR |
| 16:42.01 | [TK]D-Fender | [T]ank: externhost=206.71.78.158 <- should be externip. Also you don't have nat=yes under[general] AND we don't see your actual peers, and complete debug |
| 16:42.06 | *** join/#asterisk imcdona (~imcdona@173.160.189.68) |
| 16:42.36 | [T]ank | k, let me try the change in syntax... thanks |
| 16:43.06 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 16:51.32 | [T]ank | still no luck... here is the debug. |
| 16:51.33 | [T]ank | http://pastebin.com/HasLqGcH |
| 16:51.47 | [T]ank | peer is included. |
| 16:53.11 | *** join/#asterisk razu_ (~razu@razu.data.ee) |
| 16:54.08 | *** join/#asterisk simplydrew (~simplydre@66.181.225.250) |
| 16:56.22 | [TK]D-Fender | [Trestart * |
| 16:58.33 | [TK]D-Fender | [T]ank: localnet=10.40.0.8/0 <-- and try setting a VALID netmask |
| 16:59.37 | fauxalliance | <haiku> TCP/IP, learn how it fits together, there is no escape</haiku> |
| 17:02.08 | [TK]D-Fender | I should learn haikus. They're moderately easy. Yeah I think I will..... |
| 17:02.36 | Kobaz | i haven't played much with natted asteriskeses |
| 17:02.57 | Kobaz | would using externip be enough in sip.conf for forcing the source ip for sip stuff |
| 17:03.46 | [TK]D-Fender | Kobaz: No. |
| 17:03.47 | Kobaz | asterisk is behind a nat router, with ports 5060 and 10000-20000 forwarded.... and i have the usual one way audio issue... it looks like asterisk is sending out 192.168.1.160 as the server ip |
| 17:03.50 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:04.14 | [TK]D-Fender | wishes he put aside the time to get his server subnet back on-line |
| 17:04.32 | carrar | don't do it |
| 17:04.37 | carrar | china will attack you |
| 17:05.04 | Kobaz | i wish the world was ipv6 already |
| 17:05.05 | carrar | China will steal your IP's!!!! |
| 17:05.06 | Kobaz | come on, hurry up world |
| 17:05.56 | *** join/#asterisk simplydrew (~simplydre@66.181.225.250) |
| 17:06.28 | carrar | Kobaz |
| 17:06.34 | carrar | your host doesn't even have IPv6 |
| 17:06.36 | ANurmi | what is the patternmatching scheme for international calling? |
| 17:06.55 | Kobaz | nope |
| 17:07.00 | carrar | HURRY!! |
| 17:07.04 | carrar | Before it's TOOOOO late |
| 17:07.35 | carrar | The world is gonna crash! |
| 17:08.03 | [TK]D-Fender | ANurmi: Depends what your carrier says |
| 17:08.21 | telnettech | ANurmi: depends on what country you are in |
| 17:08.30 | ANurmi | US to Europe. |
| 17:08.47 | ANurmi | Germany specifically* |
| 17:09.21 | [TK]D-Fender | ANurmi: depends on your CARRIER |
| 17:10.29 | telnettech | 011 is the universal code for US carriers but TK is right, check with your Carrier to verify what they expect |
| 17:14.30 | jpmcallister | Anyone familiar with that error: ERROR[6137]: chan_dahdi.c:12405 dahdi_pri_error: !! Don't know how to pre-handle message type Unknown Message Type (100) |
| 17:15.06 | [T]ank | [TK]D-Fender: haha... woops. Thanks. /0 = /24 |
| 17:15.09 | [T]ank | that fixed it. |
| 17:15.28 | jpmcallister | It happens when I try to setup "follow me" from an analog extension at my old pbx to an extension at * |
| 17:15.35 | *** join/#asterisk ccesario (~ccesario@189-18-198-93.dsl.telesp.net.br) |
| 17:20.35 | *** join/#asterisk ccesario (~ccesario@189-18-198-93.dsl.telesp.net.br) |
| 17:20.56 | *** join/#asterisk Tim_Toady (~moi@77.49.109.71.dsl.dyn.forthnet.gr) |
| 17:28.47 | *** part/#asterisk goomail (~Adium@r186-48-193-11.dialup.adsl.anteldata.net.uy) |
| 17:29.47 | *** join/#asterisk jkroon (~jkroon@dsl-242-7-115.telkomadsl.co.za) |
| 17:30.04 | *** join/#asterisk ccesario (~ccesario@189-18-198-93.dsl.telesp.net.br) |
| 17:33.33 | *** join/#asterisk ccesario (~ccesario@189-18-198-93.dsl.telesp.net.br) |
| 17:37.24 | citywok | jpmcallister: get rid of the old pbx :P |
| 17:37.40 | jpmcallister | citywok: I wish :) |
| 17:40.31 | citywok | wish harder! |
| 17:40.53 | jpmcallister | If I wish harder, my brain will explode |
| 17:41.53 | [TK]D-Fender | <PROTECTED> |
| 17:50.37 | tzafrir_laptop | jpmcallister, what version of asterisk? libpri? |
| 17:50.51 | tzafrir_laptop | Can you reproduce that? |
| 17:51.05 | tzafrir_laptop | If no progress, try providing a pri-level trace |
| 17:51.26 | tzafrir_laptop | asks again about Nortel phones |
| 17:51.54 | tzafrir_laptop | Where is the time zone set for them? Anywhere explicitly? |
| 17:52.20 | tzafrir_laptop | Alternatively: what needs to be reset to apply the change of timezone settings on the asterisk host? |
| 17:59.40 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 18:00.48 | ANurmi | So I figured out to dial through vonage it is 011 + country code + city code + number, I am dialing from the US, would exten => _011XXXXXXXXXXXX,1,Dial(DAHDA/5/${EXTEN}) work for pattern matching? |
| 18:02.21 | p3nguin | anurmi: Consider using _011.,1,Dial() because not every number you call will be the same length. |
| 18:03.08 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 18:03.08 | ANurmi | ok, I had looked at that example, but it was saying that would only anticipate one more digit. |
| 18:03.46 | p3nguin | _011. means 011 + at least one more |
| 18:05.01 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:05.29 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 18:09.49 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 18:11.41 | ANurmi | OK, good to know. |
| 18:14.52 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:15.14 | p3nguin | ~dialing patterns |
| 18:15.25 | p3nguin | ~pattern matching |
| 18:15.25 | infobot | it has been said that pattern matching is explained here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 18:19.18 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 18:19.21 | wcselby | o/ |
| 18:25.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:25.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:27.47 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 18:28.37 | fullstop | I'm writing an AGI script.. and no matter what my exit code is in the script, asterisk says "AGI Script script.agi completed, returning 0" |
| 18:29.01 | fullstop | I'm sort-of familiar with python, but I believe sys.exit(1) should exit with a return code of 1. |
| 18:29.04 | wcselby | your script isn't executing properly |
| 18:29.11 | wcselby | oh |
| 18:29.14 | wcselby | i don't know about that |
| 18:29.40 | fullstop | Oh, the script runs fine.. but I'm trying to make sure that I can tell if * can see if it terminates unexpectedly, etc.. |
| 18:31.02 | fullstop | I've checked in the shell -- echo $? returns 1 when the agi runs. |
| 18:31.37 | wcselby | i don't know if the "returning 0" is the same "0" that you're trying to return |
| 18:32.13 | wcselby | i'll look in a little bit, kind of busy at the moment |
| 18:33.20 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 18:42.05 | *** part/#asterisk josephnexus (~josephnex@71-209-40-81.bois.qwest.net) |
| 18:43.53 | fullstop | wcselby: I looked in the asterisk res_agi.c source... it checks to see if the return code is set. If it is, it sets it to -1. |
| 18:48.00 | *** join/#asterisk Alric (~nbowyer@64.6.54.218) |
| 18:48.59 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 18:49.39 | fullstop | I misread the code. you can only determine if the channel was hung up or not. The return code of the agi process is not checked at all. |
| 18:52.37 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 18:55.56 | wcselby | fullstop - yeah, that's what I thought as well |
| 18:56.36 | fullstop | I suppose that this is understandable, considering that an agi process can be yanked at any time. |
| 18:58.51 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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| 18:59.29 | *** part/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 18:59.38 | kjs | anyone recommend a decent sip soft phone for the iPhone ? |
| 19:03.05 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 19:05.27 | *** join/#asterisk bsaxon (~bsaxon@rrcs-67-78-132-142.se.biz.rr.com) |
| 19:14.10 | wcselby | kjs - i use iSip, i've been happy with it when I need to use it. it has wifi and 3g support |
| 19:21.39 | Katty | if anyone is interested in participating in the Asterisk Christmas Card Exchange, please /query me |
| 19:22.43 | Kobaz | hehe |
| 19:23.56 | *** part/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 19:25.57 | *** join/#asterisk simplydrew (~simplydre@66.181.225.250) |
| 19:27.15 | jdoe | is anyone here doing asterisk ha with only open source tools? (ie drbd/linux-ha etc.) |
| 19:29.59 | leifmadsen | yes |
| 19:30.28 | leifmadsen | I use linux-ha, dundi, replicated mysql, func_odbc, subversion, rsync, unison, etc. |
| 19:31.31 | jdoe | need to provision a remote site that's a PITA to get to, so looking to make it fairly resilient... just not entirely sure what my options are, or what gotchas there might be out there. The wiki seems to be mostly proprietary options. |
| 19:32.02 | jdoe | I guess I'm mostly concerned with keeping voicemail, configs, and a couple custom sound files in sync. The phone configs would be nice too, but that's probably fine with rsync alone. |
| 19:32.23 | leifmadsen | you could keep the voicemail in IMAP too |
| 19:33.14 | jdoe | that's not really an option... we're spread across three separate email systems, two of which don't have HA storage, and the other is gmail :/ |
| 19:33.31 | leifmadsen | use unison to sync the voicemails then I guess |
| 19:33.44 | jdoe | yeah, it was going to be either that or trying to figure out drbd + gfs or something. |
| 19:33.59 | fullstop | I use drbd here -- not for asterisk, but for other systems. |
| 19:34.35 | fullstop | The problem with asterisk is that it usually involves system-specific hardware or licenses. |
| 19:34.58 | jdoe | in this particular setup it's all software, and (currently) with no g729 licenses etc. |
| 19:35.05 | fullstop | that is, you would have to have a 2nd TDM card ready and waiting.. as well as 729 licenses if you use them |
| 19:35.06 | fullstop | ahh |
| 19:35.44 | jdoe | call volume is low enough that I wouldn't care especially about dupe licenses, I just need it to work most of the time. |
| 19:35.46 | fullstop | jdoe: what sort of bandwidth do these systems have? |
| 19:36.02 | jdoe | fullstop: between them, or to the world? |
| 19:36.07 | fullstop | between them |
| 19:36.13 | jdoe | gigE |
| 19:36.43 | fullstop | I have a central server here, and they share a network filesystem between the two for a central voicemail location. |
| 19:37.12 | fullstop | This could be a drbd volume if you wish for it to be replicated. |
| 19:37.50 | jdoe | I'm thinking about it, but I'm trying to avoid a central server. |
| 19:38.01 | jdoe | ah, yeah. |
| 19:38.26 | fullstop | it doesn't work for everyone; it really depends on your setup. |
| 19:39.05 | fullstop | For us, the central server is the one with the dahdi hardware.. and remote locations place calls over an iax2 trunk to get to the outside world. |
| 19:41.08 | jdoe | leifmadsen: how well does unison handle voicemail dirs? Presumably you'd get a lot of conflicting filenames since they're incrementally named. |
| 19:41.25 | leifmadsen | I've never used it for that, so I have no idea |
| 19:41.29 | leifmadsen | I've used it to sync recordings |
| 19:42.30 | jdoe | oh. You do imap storage for voicemail or something different? |
| 19:42.49 | jdoe | hmm... wait... I could do odbc storage... |
| 19:42.52 | jdoe | that works. |
| 19:43.03 | jdoe | thanks guys :) |
| 19:43.49 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 20:22.55 | wcselby | heh |
| 20:23.32 | wcselby | client is bitching about a $40 license fee for FOP2, but didn't blink an eye at a 17.5k bill for brand new imacs |
| 20:27.07 | leifmadsen | heh |
| 20:27.10 | leifmadsen | priorities :) |
| 20:41.40 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 20:42.39 | fullstop | how is fop2, btw? |
| 20:45.38 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 20:46.14 | Nugget | I like it |
| 20:47.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:56.55 | *** join/#asterisk sbszulu (~dundubala@41.14.103.215) |
| 20:59.21 | wcselby | it does it's job |
| 20:59.25 | wcselby | fop2, that is |
| 21:00.50 | p3nguin | Yes, it does it is job. |
| 21:04.20 | wcselby | hmmm |
| 21:04.38 | wcselby | i suppose i deserved that |
| 21:11.19 | *** join/#asterisk MiserySoft (~Adium@94-116-26-44.dynamic.thecloud.net) |
| 21:18.54 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 21:19.45 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 21:21.12 | joesuffceren | I have a quad-core Xeon platform running 1.4.27. Up to now this box has been SIP only, but I am being forced to add 3 analog trunks to it. I am planning to buy an analog interface card from digium. I'm wondering with the processing power I have at my disposal if software echo cancellation is a good path, or if you would recommend that I spend the extra coin for the hardware echo cancellation |
| 21:24.04 | bneff | wow Asterisk 1.8 is amazing |
| 21:25.31 | joesuffceren | bneff: care to elaborate? |
| 21:26.03 | bneff | joesuffceren: I'd spend 2 days getting an older(er) pre-built ec2 asterisk setup working |
| 21:26.15 | pabelanger | joesuffceren: you should be fine with software echo, but if you can spare the extra $$ grab the module |
| 21:26.37 | bneff | joesuffceren: I was having weird issues so I decided to just go to 1.8...and everything is working with minimal configuration - I'm using SIP only |
| 21:26.49 | joesuffceren | pabelanger: I'll have to look for some better pricing. digium store it goes from 200 to 600 when you add the echo module :-/ |
| 21:27.15 | p3nguin | $400 for an echo canceler is a lot of money. |
| 21:27.33 | pabelanger | joesuffceren: you save some $$$ through a reseller |
| 21:27.35 | joesuffceren | bneff: cool deal. I'm getting rid of a 1.2 installation tomorrow. Excited about that |
| 21:27.42 | bneff | voice,video,nat, presence...working beautiful |
| 21:28.02 | bneff | I just started with Asterisk this week... to all the devs...awesome work |
| 21:28.48 | joesuffceren | bneff: interesting. I'll have to look into that for the video and presence! what endpoints? |
| 21:28.52 | *** join/#asterisk NickNick (~nicholas@93-97-188-195.zone5.bethere.co.uk) |
| 21:29.09 | bneff | using x-lite ( Free version ) and media-fone on iphones |
| 21:29.28 | joesuffceren | x-lite does video? |
| 21:29.48 | bneff | yeah, I just tried it with a co-worker |
| 21:30.03 | bneff | windows 7 to snow leopard |
| 21:30.26 | joesuffceren | bneff: nice! |
| 21:30.33 | *** join/#asterisk guilhermebr (~Guilherme@189.63.47.66) |
| 21:30.44 | joesuffceren | will have to play with that. *tries to decide which old laptop wants to be a 1.8 testbed... :-) |
| 21:31.21 | bneff | amazon aws is free for a year...don't even need hardware to test it on =) |
| 21:31.40 | joesuffceren | and your video is working fine over that? would think latency would kill you... |
| 21:32.09 | *** part/#asterisk BMJ (~bjohns@cpe-098-026-116-043.nc.res.rr.com) |
| 21:32.10 | bneff | nope...quality is not the greatest, but its good enough and works |
| 21:32.26 | bneff | and the video is being routed through ec2, haven't figured out how to to p2p yet |
| 21:32.59 | bneff | also, I'm west coast, and the ec2 instance is on the east coast..and still really usable |
| 21:42.19 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 21:42.20 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:42.48 | mercutioviz | hey, does anybody else notice that voip-info.org's main page is missing all the news items? |
| 21:43.00 | citywok | voip-info has a main page? |
| 21:43.07 | mercutioviz | hehe |
| 21:43.11 | mercutioviz | more or less |
| 21:43.14 | WIMPy | voip-info has news? |
| 21:43.16 | citywok | i thought it's main page was google |
| 21:43.24 | p3nguin | its |
| 21:43.32 | citywok | sorry, i just did that like 5 minutes ago too |
| 21:43.38 | citywok | it doesn't own anything, nor is it plural |
| 21:43.46 | mercutioviz | thhehe |
| 21:43.48 | mercutioviz | oops |
| 21:43.49 | mercutioviz | hehe |
| 21:43.51 | citywok | and it isn't a contraction :P |
| 21:43.55 | mercutioviz | yeah, its vs. it's |
| 21:44.08 | citywok | stupid grammar |
| 21:44.51 | p3nguin | Being able to communicate is horrible. |
| 21:47.13 | *** join/#asterisk sbszulu (~dundubala@41.14.103.215) |
| 21:47.14 | p3nguin | hukd on foniks |
| 21:48.45 | Nugget | S O C K S |
| 21:48.53 | drmessano | shur werkd fer me |
| 21:49.19 | drmessano | I love a good smore with grammer crackers |
| 21:49.29 | p3nguin | chortles |
| 21:52.01 | citywok | so i just discovered my philippines office MPLS network didn't have the output QoS map running. Amazing that for 8 months we didn't have any issues or notice. |
| 21:52.14 | p3nguin | I guess your QoS policy was overrated. |
| 21:52.53 | citywok | heh, fortuantely the packets were already tagged, so the MPLS network itself could prioritize, but the bandwidth must not have ever spiked high enough for anybody to care/notice. lol |
| 21:54.01 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 21:55.56 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 22:05.10 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:09.13 | *** join/#asterisk telnettech_lap (~telnettec@74-132-208-121.dhcp.insightbb.com) |
| 22:18.37 | *** part/#asterisk sbszulu (~dundubala@41.14.103.215) |
| 22:21.26 | *** join/#asterisk AAronCI (~Jimota@CPE0018397aac73-CM000f212fb09b.cpe.net.cable.rogers.com) |
| 22:21.53 | AAronCI | hello world |
| 22:22.05 | AAronCI | oh good, working |
| 22:24.18 | AAronCI | hello? |
| 22:24.48 | p3nguin | We still don't know how to answer that one. |
| 22:24.52 | p3nguin | ~ask |
| 22:24.52 | 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. |
| 22:24.56 | p3nguin | ~answers |
| 22:24.56 | infobot | [~answers] 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 |
| 22:26.13 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 22:26.54 | AAronCI | I am having trouble making a core dump. Specifically, I don't know where to find the resulting file |
| 22:27.10 | p3nguin | Why do you want to dump core? |
| 22:27.25 | p3nguin | That's typically the result of something bad. |
| 22:27.50 | AAronCI | yes, asterisk crashes when I call playback() |
| 22:27.51 | citywok | AAronCI: it's in the folder that asterisk started from |
| 22:28.11 | citywok | so if you are in /etc/asterisk, and type asterisk -gvvvvdddddddd the core dump will land in /etc/asterisk |
| 22:28.41 | AAronCI | hmm, ok. let my try that now |
| 22:28.48 | tzafrir_laptop | in /etc/asterisk ? |
| 22:29.09 | citywok | it goes whereever you are... so if you are in /root it goes there, or /i_heart_puppes, it will go there |
| 22:29.23 | citywok | or even in /i_heart_puppies lol |
| 22:29.59 | p3nguin | /home/aaron/.pr0n/ |
| 22:30.19 | citywok | nice |
| 22:30.37 | AAronCI | :) |
| 22:30.46 | AAronCI | I have long since moved to .pr0n2 |
| 22:32.12 | AAronCI | ok, so no core dump |
| 22:32.26 | AAronCI | ach, now it's even worse |
| 22:32.29 | AAronCI | damn |
| 22:32.33 | citywok | did you start asterisk with -g? |
| 22:32.37 | p3nguin | How about using strace? |
| 22:32.39 | citywok | when it crashes it will say (core dumped) |
| 22:32.42 | AAronCI | I used asterisk -gvvvvdddddddd |
| 22:32.59 | citywok | did it say (core dumped) ? |
| 22:33.25 | AAronCI | it froze here: == Parsing '/opt/etc/asterisk/cli.conf': == Found |
| 22:33.36 | AAronCI | I quit the process |
| 22:33.40 | AAronCI | it said segmentation fault |
| 22:33.48 | AAronCI | let me start again without all that verboseness |
| 22:33.51 | citywok | croe dump only happens on a segfault, i'm not sure it will do it if you kill it |
| 22:33.58 | tzafrir_laptop | AAronCI, or maybe it actually crashed after that |
| 22:34.19 | citywok | and you may want to check your cli.conf and make sure nothing is wrong with it |
| 22:34.38 | tzafrir_laptop | if you have a core file, use: gdb -c path/to/core path/to/sbin/asterisk |
| 22:34.48 | tzafrir_laptop | and get a backtrace: bt |
| 22:34.49 | AAronCI | ok, it's running. Now I'll replicate my crashing behaviour... |
| 22:35.10 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:36.33 | AAronCI | still no file. maybe it's not crashing. Is there a difference between a crash and 'Disconnected from Asterisk server'? |
| 22:36.40 | p3nguin | quite |
| 22:37.03 | p3nguin | Run it normally (using the init script), then connect to the CLI with asterisk -r. |
| 22:37.08 | AAronCI | I can't connect to the asterisk prompt though. |
| 22:37.23 | p3nguin | Did you run it using the init script? |
| 22:37.47 | AAronCI | I ran it by typing 'asterisk -g'. where would this init script be |
| 22:37.59 | p3nguin | /etc/init.d/asterisk |
| 22:38.39 | tzafrir_laptop | AAronCI, do you normally run Asterisk as root? |
| 22:38.42 | AAronCI | hmm, nothing there |
| 22:38.51 | AAronCI | yes, I'm running as root here. |
| 22:38.53 | p3nguin | How did you install asterisk and on what distro? |
| 22:39.16 | AAronCI | It's running on a Asus RT-N16. Installed via optware from NSLU2 linux |
| 22:39.26 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 22:39.27 | AAronCI | quite the oddball system, I know |
| 22:39.29 | bmoraca_work | join #cisco |
| 22:39.32 | bmoraca_work | er |
| 22:40.43 | AAronCI | I haven't touched it other than add a couple peers and extensions |
| 22:41.00 | AAronCI | so I'm puzzled as to why it's not working |
| 22:41.20 | AAronCI | it's 1.8.0.1 |
| 22:41.26 | AAronCI | which may make a difference |
| 22:45.46 | AAronCI | ahh, I see I scared everyone off. :( |
| 22:45.52 | AAronCI | I'm just going to restart here |
| 22:48.04 | citywok | that router may not have all the features built in? |
| 22:48.24 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 22:48.27 | citywok | i have no idea what that thing is even capable of doing, it may not have core-dump support (is that even possible to do? any digium guys around?) |
| 22:48.39 | AAronCI | I kind of doubt that since others are successfully running systems on it. |
| 22:49.15 | AAronCI | I think I'm going to install asterisk 1.4 to see if that makes a difference |
| 22:49.42 | citywok | i'm sure it supports *, but whether or not the * that runs on it has the ability to produce a core dump is what i'm curious about |
| 22:51.27 | AAronCI | ahh, I see |
| 22:56.02 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:56.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:58.54 | *** join/#asterisk [canniballllera] (~cannibale@201-14-149-18.fnsce703.dsl.brasiltelecom.net.br) |
| 22:59.24 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 23:02.45 | fireman_biff | I just had to make a copy of an asterisk system (1.4.2.2 in trixbox 2.6.2.3) by copying the config files and the databases. The new system is running but the extensions, follow me's etc won't work until I re-submit the form for each one in FreePBX and apply the changes. Is there a better way to do this than go through each extension, follow me etc? (I already tried asterisk -rx reload) |
| 23:03.56 | [TK]D-Fender | fireman_biff: Wrong channel... 3rd from the left please... |
| 23:03.59 | [TK]D-Fender | ~trixbox |
| 23:03.59 | infobot | it has been said that trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 23:04.09 | [TK]D-Fender | #trixbox <- |
| 23:04.20 | citywok | lol |
| 23:04.49 | fireman_biff | alright, I'll try there |
| 23:05.05 | [TK]D-Fender | fireman_biff: And I'm betting you didn't copy all the pertinent databases |
| 23:05.12 | WIMPy | Has anyine here ever taken a look at Gemeinschaft, BTW? |
| 23:05.13 | fireman_biff | I copied all |
| 23:05.18 | citywok | i imagine trixbox has a backup button, and a restore button? |
| 23:05.22 | [TK]D-Fender | fireman_biff: Define "all" |
| 23:05.36 | fireman_biff | every database except mysql and test |
| 23:05.46 | [TK]D-Fender | fireman_biff: from where? |
| 23:06.24 | fireman_biff | using mysqlhotcopy on the original pbx and then dropped into /var/lib/mysql on the new one |
| 23:07.02 | [TK]D-Fender | fireman_biff: So far it sounds like you left of AstDB <------ |
| 23:07.09 | [TK]D-Fender | off |
| 23:07.57 | fireman_biff | thank you |
| 23:08.01 | fireman_biff | indeed I did |
| 23:08.59 | fireman_biff | can I just do a regular copy of /var/lib/asterisk/astdb ? |
| 23:09.43 | fireman_biff | nevermind, I found something saying that I can |
| 23:15.03 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 23:20.23 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 23:27.42 | ManxPower | a trixbox user that knows something? How the hell did *that* happen? |
| 23:29.50 | fireman_biff | given that people here don't seem to like trixbox much, what do you guys recommend? |
| 23:33.38 | p3nguin | fireman_biff: How about ... gasp ... Asterisk? |
| 23:34.41 | fireman_biff | but is there any real difference between installing Asterisk on Cent (for example) and installing trixbox? Do you get anything you don't get with trixbox? |
| 23:35.58 | p3nguin | You don't get a forked version of Asterisk nor a forked version of FreePBX. |
| 23:36.52 | p3nguin | If you need something that you can at least get support for, go with AsteriskNOW. |
| 23:37.34 | fireman_biff | I'm not trying to be difficult or anything, but what would a non-forked version of asterisk/freepbx give me that the version in trixbox don't? |
| 23:37.44 | fireman_biff | I'm just trying to figure out how much sense the switch would make for me |
| 23:38.29 | WIMPy | You can get support here. |
| 23:38.41 | fireman_biff | hmm... true :) |
| 23:39.48 | WIMPy | And you can learn about ALL features Asterisk provides. |
| 23:40.30 | fireman_biff | so the regular asterisk has features that the trixbox version doesn't have? |
| 23:41.17 | WIMPy | The trixbox version might have them as well, but that dosn't mean you can use them easily. |
| 23:41.17 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:42.00 | WIMPy | Is there a way to see if a call is using srtp? |
| 23:44.05 | fireman_biff | I don't know what that is, much less how to check for it |
| 23:44.34 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:45.11 | fireman_biff | is that something I would need internally? |
| 23:45.46 | WIMPy | Probably not. |
| 23:46.19 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 23:46.20 | fireman_biff | good, at least that's one bit of research I don't need to do tonight :) |
| 23:52.05 | AAronCI | well, asterisk 1.4 works perfectly |
| 23:55.41 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:56.17 | riddlebox | man these freaking VoIP spammers all use the same scripts and dictionaries |
| 23:57.12 | WIMPy | They probably paid lots of money for the most common username/password combinations. |
| 23:57.20 | WIMPy | LOL |
| 23:58.30 | p3nguin | I can't imagine that v2cYR/#dxu@.dLVW;FefkQK00 is a common password found in a list. |
| 23:59.07 | WIMPy | That doesn't matter if you can convice a "customer" that it's a likely one. |
| 23:59.31 | p3nguin | Good point. Marketing always seems to win. |
| 23:59.36 | WIMPy | I mean the most e-mail addresses being sold have never existed, either. |