| 00:00.31 | asdx | nm |
| 00:01.35 | hsv-al | man, these cepstral voices cool w/ asterisk |
| 00:01.44 | hsv-al | i got that british sounding lady reading back dynamic weather, when users input zipcode |
| 00:02.28 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 00:09.46 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:10.48 | *** join/#asterisk DrkShadow_ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:12.21 | DrkShadow_ | so, IAX is set up; Machine A is behind NAT and connects to machine B. All works correctly, calls go to the right context and work. However, when a call goes from B--> A, I get the error, "Rejected connect attempt from <machine B>, request '101@default' does not exist".. what's up? |
| 00:12.34 | DrkShadow_ | machine A registers on B, but not vice-versa (NAT issue) |
| 00:12.45 | Strom_C | is there an extension called "101" in the context "default"? |
| 00:12.56 | DrkShadow_ | the context "default" isn't referenced anywhere. |
| 00:13.02 | DrkShadow_ | except this error. |
| 00:13.26 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:13.29 | Strom_C | did you specify a context at all on the receiving end? |
| 00:13.50 | DrkShadow_ | yes, which makes me think it's not coming in on the correct IAX... "profile" |
| 00:14.05 | DrkShadow_ | and rather taking the guest one (which is context=default) |
| 00:14.36 | Strom_C | pastebin your iax.confs |
| 00:14.39 | DrkShadow_ | but, without registering, I just don't see how to make it use the correct IAX context.. |
| 00:14.43 | DrkShadow_ | ehm.. hold. |
| 00:15.47 | *** join/#asterisk moy (n=moyhu@189.169.83.78) |
| 00:16.53 | DrkShadow_ | http://pastebin.com/de6d7997 |
| 00:17.27 | Strom_C | the whole thing, please |
| 00:17.40 | *** join/#asterisk RoyK (n=roy@ip-168-52-149-91.dialup.ice.no) |
| 00:17.56 | *** part/#asterisk RoyK (n=roy@ip-168-52-149-91.dialup.ice.no) |
| 00:18.04 | Strom_C | also, NAT has nothing to do with your problem |
| 00:18.15 | DrkShadow_ | that is so difficult... |
| 00:18.16 | Strom_C | the fact that machine A is responding proves that it isn't the NAT |
| 00:22.36 | Mercestes | pasting the whole thing is difficult? lol. Would, say, paying a consultancy fee be easier for you? |
| 00:22.48 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:26.15 | *** join/#asterisk DrkShadow__ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:26.22 | DrkShadow__ | egh.. network sucks. |
| 00:26.35 | DrkShadow__ | strom: http://pastebin.com/d7c923111 |
| 00:27.47 | Strom_C | do you need guest, iaxtel, and demo? |
| 00:27.50 | Strom_C | if not, remove them |
| 00:28.41 | DrkShadow__ | they shouldn't be hurting anything, and perhaps you should read the note about the guest context on voip-info. |
| 00:28.51 | DrkShadow__ | For the sake of security, it says, you need it. |
| 00:29.53 | *** join/#asterisk glaz (n=strke@mofu.ca) |
| 00:31.36 | DrkShadow__ | any other input? |
| 00:31.51 | Mercestes | can you add the appropriate extensions.conf lines? |
| 00:32.15 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3446d2cd555c0e77) |
| 00:32.34 | DrkShadow__ | exten => _***.,1,Dial(IAX2/trent-phonebox/${EXTEN:3}) |
| 00:32.45 | Mercestes | Oh, you aer using type=friend. Try seperating them out into type=user and type=peer |
| 00:32.51 | Mercestes | I am pretty sure I mentioned I never made "friend" work. |
| 00:33.01 | DrkShadow__ | mercestes: you said that with sip.. |
| 00:33.17 | Mercestes | I said that "in general" :P In reference to sip. |
| 00:33.18 | Strom_C | DrkShadow__: I don't consider voip-info to be even remotely reputable |
| 00:33.31 | DrkShadow__ | strom: well, you're the only one. |
| 00:33.39 | Strom_C | i've found so much stuff on that stupid thing that's so blatantly wrong... |
| 00:33.40 | Mercestes | no, he's really not. |
| 00:34.14 | Mercestes | unreputable doesn't mean you should not read it. It just means it's not *the source* of information. |
| 00:34.20 | Mercestes | People like me write stuff on there all the time. |
| 00:34.55 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:35.13 | Mercestes | but anyways, back to the issue at hand, try seperate user and peer entries in iax.conf just for laughs. That's the only way I could get it to work properly |
| 00:35.51 | Mercestes | if it does magically work....you might want to file a bug report with copies of all this. Could be you (and me) or it could be something with type=friend. |
| 00:36.02 | Mercestes | I'm just too lazy to dig deep enough into it to find out. |
| 00:36.10 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582050.dsl.bell.ca) |
| 00:36.48 | *** join/#asterisk skyggen (n=edward@CPE-65-31-137-130.wi.res.rr.com) |
| 00:36.56 | skyggen | hello? |
| 00:37.10 | Strom_C | I still think you should try removing or commenting out the guest and iaxtel stuff just for testing |
| 00:37.14 | Strom_C | hi skyggen |
| 00:37.24 | skyggen | Anyone ever had the D-channel drop randomly? |
| 00:37.32 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 00:37.32 | Strom_C | ISDN PRI? |
| 00:37.46 | skyggen | Yar |
| 00:37.46 | DrkShadow__ | strom: I did. It works with guest removed, which suggests it's an authentication error, and emphasizing that I _do_not_ want to remove the guest context. |
| 00:38.01 | Strom_C | DrkShadow__: try putting the guest context at the /end/ of your iax.conf file |
| 00:38.09 | Strom_C | and don't be such a stick in the mud |
| 00:38.52 | DrkShadow__ | strom: wtf? I suppose you're just being your usual self. |
| 00:39.46 | Mercestes | Damn you! Damn you strom, for helping me for free! A curse upon you! |
| 00:39.54 | Strom_C | yes, dear god |
| 00:39.57 | Strom_C | I'm such a horrible person |
| 00:40.03 | Strom_C | I should go off myself right now |
| 00:40.18 | DrkShadow__ | help? I haven't received that here yet.. only people telling me to do things, completely unrelated to the problem I'm having. Dunno where this "Help" is, but it's sure as hell not here. |
| 00:40.21 | *** part/#asterisk DrkShadow__ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:40.27 | Strom_C | CLOMP CLOMP CLOMP *CLICK* *BANG!!!!* THUD |
| 00:40.28 | unpaidbill | strom, before you do it scream 'IT'S NOT CALI YOU RETARDS' |
| 00:40.43 | skyggen | So anyone got any ideas why the D-channel on my T1 span drop? |
| 00:40.57 | Strom_C | skyggen: probably a transmission error |
| 00:41.31 | Strom_C | how far is the asterisk box from the smartjack? |
| 00:41.31 | skyggen | 408 + feet |
| 00:41.31 | skyggen | the conduit is 408 plus prolly another 50 |
| 00:41.31 | Strom_C | do you have the gain set properly in zaptel.conf? |
| 00:41.44 | skyggen | gain is at 0 |
| 00:41.51 | Strom_C | zzzt, wrong :) |
| 00:41.59 | *** join/#asterisk routerweasel (n=4stfed@66.208.251.241) |
| 00:42.20 | skyggen | ok, whats a good number? |
| 00:42.30 | Strom_C | zaptel.conf, not zapata.conf |
| 00:42.37 | Strom_C | the "build-out" section of the T1 span definition |
| 00:42.43 | skyggen | yeah, thats what I was thinking you were talking about |
| 00:42.43 | Strom_C | that's set to 0? |
| 00:42.45 | Mercestes | ...this means he got type=friend to work, btw...I'll have to look into that. =/ |
| 00:42.59 | skyggen | txgain=0.0 rxgain=0.0 |
| 00:43.02 | Strom_C | no |
| 00:43.04 | Strom_C | ZAPTEL.CONF |
| 00:43.07 | Strom_C | not zapata.conf |
| 00:43.34 | Strom_C | look at your span= line |
| 00:43.46 | skyggen | span=1,1,4,esf,b8zs |
| 00:44.03 | Strom_C | ok, so LBO gain is set properly |
| 00:44.18 | Strom_C | what's your zttest score? |
| 00:44.37 | skyggen | 99.987793% |
| 00:45.19 | Strom_C | hm |
| 00:45.32 | skyggen | and its not always dropping |
| 00:45.44 | Strom_C | is there a consistent pattern to when it drops? |
| 00:45.44 | skyggen | it drops randomly |
| 00:46.32 | Strom_C | which telco? |
| 00:46.38 | skyggen | One Communications. |
| 00:46.44 | Strom_C | never heard of them |
| 00:46.48 | *** join/#asterisk hads (n=hads@120.138.17.30) |
| 00:46.51 | Strom_C | CLEC, I'm guessing? |
| 00:47.02 | skyggen | One Comunications -> Us Exchange -> AT&T |
| 00:47.22 | Strom_C | whose switch is at the other end of your T1? |
| 00:47.31 | skyggen | One Comunications |
| 00:47.39 | Strom_C | ok, so they own their own switch? |
| 00:47.45 | skyggen | we had TDS and everythign worked fine |
| 00:48.04 | skyggen | yes |
| 00:48.26 | Strom_C | I'm willing to suspect somethign on their end |
| 00:48.42 | Strom_C | s/gn/ng/ |
| 00:49.13 | skyggen | if it was on my end what would it be? |
| 00:49.44 | Strom_C | some sort of weird software issue, I think...but your zttest score is good, and your LBO is correct |
| 00:49.58 | Strom_C | what kind of card are you using? |
| 00:50.01 | skyggen | yeah, only running asterisk and postgresql |
| 00:50.16 | skyggen | Digium, Inc. Wildcard TE205P |
| 00:50.51 | skyggen | The only other thing I could think to be on my end would be Humidity. |
| 00:50.55 | Strom_C | you may want to give Digium a ring and have them make sure things look good to them before opening a ticket with your carrier |
| 00:51.05 | Strom_C | why, are you underwater and/or in Tallahassee? |
| 00:51.43 | skyggen | midwest flooding |
| 00:52.01 | skyggen | its that or mice |
| 00:52.13 | Strom_C | did your equipment get wet? |
| 00:52.16 | skyggen | no |
| 00:52.42 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-246-91-9.nycmny.east.verizon.net) |
| 00:52.51 | Strom_C | call your carrier; the most likely thing I can think of is a problem in their linecard or somewhere in the outside plant |
| 00:53.22 | skyggen | yeah, I'm still waiting for a technician |
| 00:53.41 | VJFROMGT | my trunk gives up too fast when a call rings out, where can i set timeout value for outgoing calls? |
| 00:53.44 | VJFROMGT | yes |
| 00:54.48 | Strom_C | "trunk"? "rings out"? |
| 00:55.08 | VJFROMGT | yes |
| 00:55.31 | VJFROMGT | if i make a call to someone, their phone rings and rings and my asterisk gives up |
| 00:55.46 | VJFROMGT | but if it can wait a few more rings i will get voicemail |
| 00:55.57 | Strom_C | what does your Dial() line look like? |
| 00:56.29 | VJFROMGT | let me check |
| 00:57.53 | *** part/#asterisk VJFROMGT (n=vjfromgt@pool-96-246-91-9.nycmny.east.verizon.net) |
| 00:58.38 | Strom_C | that's some Dial() line |
| 00:58.59 | routerweasel | hehe...my turn! |
| 00:59.02 | routerweasel | I'm having a strange issue with incoming calls being rejected. I have everything registered and functioning correctly with internal extensions and my SIP provider (nexvortex). When i attempt to dial into the asterisk box from pstn the call is droped and i see "chan_sip.c: Call from 'sip_username' to extension 'DID' rejected because extension not found. I do have an incoming wildcard to forward all callers to the welcome menu. Im obviously missing so |
| 00:59.18 | Strom_C | no. |
| 00:59.21 | Strom_C | that's not what s is for |
| 00:59.59 | routerweasel | would $50 in your paypal account change your mind? |
| 01:00.10 | Strom_C | does the error message actually contain 'DID' or are you replacing a number with 'DID'? |
| 01:00.24 | routerweasel | it contains actual DID |
| 01:00.51 | Strom_C | did your question end with "obviousl missing s" or is there more past the "s" that was truncated on my end? |
| 01:00.57 | Strom_C | s/sl/sly/ |
| 01:01.26 | routerweasel | aah...on my end i see the full message... |
| 01:01.59 | *** join/#asterisk phenobitch (i=juma@gateway/gpg-tor/key-0xB9002659) |
| 01:02.05 | routerweasel | Im obviously missing something and im sure it simple...Outbound calls work prefectly...Can anyone offer some suggestions? |
| 01:02.24 | *** part/#asterisk hads (n=hads@120.138.17.30) |
| 01:02.33 | *** part/#asterisk phenobitch (i=juma@gateway/gpg-tor/key-0xB9002659) |
| 01:02.37 | Strom_C | pastebin the inbound section of your dialplan |
| 01:02.55 | routerweasel | you'll have to forgive me... |
| 01:02.59 | routerweasel | new to Asterisk |
| 01:03.20 | Mercestes | http://pastebin.ca |
| 01:03.26 | Strom_C | ~pb |
| 01:03.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:03.26 | Mercestes | ~pastebin |
| 01:03.27 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:03.32 | Mercestes | lmao |
| 01:03.35 | Strom_C | cocks |
| 01:04.01 | Mercestes | ~mercestes |
| 01:04.02 | jbot | mercestes is definitely a total nub |
| 01:04.08 | Mercestes | I shall always be remembered. |
| 01:04.09 | Strom_C | routerweasel: pastebin your full extensions.conf and your full sip.conf (but mask any passwords) |
| 01:04.12 | Strom_C | ~strom_c |
| 01:04.12 | jbot | extra, extra, read all about it, strom_c is just some nub |
| 01:04.24 | routerweasel | will do... |
| 01:06.00 | *** join/#asterisk gbr_ (n=gbr@201-24-166-176.gnace703.dsl.brasiltelecom.net.br) |
| 01:06.31 | *** join/#asterisk erojasv (n=erojasv@190.40.80.88) |
| 01:08.30 | CrazyTux | ~crazytux |
| 01:08.37 | CrazyTux | :( |
| 01:12.09 | *** join/#asterisk routerweasel (n=4stfed@66.208.251.241) |
| 01:12.14 | routerweasel | bah! |
| 01:12.58 | routerweasel | i obviously messed that up. here is the url to extensions.conf: http://pastebin.com/d5cee3058 |
| 01:13.59 | routerweasel | and sip.conf: http://pastebin.com/d47cf5a93 |
| 01:14.03 | routerweasel | sorry about that |
| 01:15.07 | Strom_C | ok...wow, this is a mess |
| 01:15.25 | Mercestes | I was just thinking that. |
| 01:15.28 | Strom_C | however, the most glaring problem is that the context you specify in sip.conf doesn't exist in extensions.conf |
| 01:16.11 | Mercestes | ....damnit Strom...you keep beating me to it |
| 01:16.19 | Mercestes | lol |
| 01:16.34 | routerweasel | ok. question: this is a fresh install...I havent mucked with anything. |
| 01:16.41 | Strom_C | routerweasel: a fresh install of what? |
| 01:16.52 | Strom_C | asterisk? or asterisknow? |
| 01:16.59 | routerweasel | asterisknow |
| 01:17.03 | routerweasel | because im a n00b |
| 01:17.06 | Strom_C | points at #asterisknow |
| 01:18.29 | plik | routerweasel: do aspire to be better then n00b ever? |
| 01:18.39 | routerweasel | bah! didnt realize there was a seperate channel. |
| 01:18.42 | Mercestes | Short answer, change DID_trunk_1 to trunk_1 in extensions.conf so SIP will know where to go, and add a exten => DID,1,Goto(voicemen-custom-1|s|1) to fix your strange problem. |
| 01:18.44 | Strom_C | plik: that's not even English |
| 01:18.56 | routerweasel | plink: yes... |
| 01:19.08 | plik | Strom_C: close enough ;) |
| 01:19.09 | Mercestes | Also, google asterisk wiki extensions.conf and asterisk wiki sip.conf and asterisk wiki to educate yourself. |
| 01:19.19 | Strom_C | routerweasel: also, read the book |
| 01:19.21 | Strom_C | ~book |
| 01:19.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:20.04 | plik | routerweasel: then I strongly encourage you to ditch asteriskNOW when you've had a little play, and install from source of your favourite distro ... perhaps not now, but before too long |
| 01:20.40 | Mercestes | and stay away from GUIs in general. |
| 01:20.47 | Mercestes | GUIs are the path to the dark side. |
| 01:21.13 | Mercestes | GUIs lead to anger, anger leads to noobness, noobness leads to suffering (for us). |
| 01:21.14 | plik | Mercestes: calling it the dark side makes it sound fun |
| 01:21.51 | Mercestes | oh.... |
| 01:21.58 | Mercestes | GUIs are the path to perpetual virginity. |
| 01:22.04 | routerweasel | i certainly will..my beef with Asterisk is that i've simply been too busy to read up on everything and learn something new. |
| 01:22.05 | plik | heh :) |
| 01:22.05 | Mercestes | nods sagely. |
| 01:22.22 | routerweasel | which is my fault but regardless i appricieate all of your help! |
| 01:22.24 | Mercestes | technically....that would not constitute a "beef with asterisk" btw. |
| 01:22.28 | routerweasel | making changes |
| 01:22.35 | Strom_C | depeche mode |
| 01:22.37 | Strom_C | boners |
| 01:22.40 | Strom_C | some other word that's funny |
| 01:22.43 | Mercestes | No problem. Find out why what's his face is sending you "DID" as a phone number |
| 01:23.01 | Strom_C | Mercestes: DID isn't being sent; routerweasel is just masking his real DID |
| 01:23.09 | Mercestes | oh... |
| 01:23.24 | Mercestes | then ignore that whole exten => DID,1,Goto thing I said. |
| 01:23.31 | Mercestes | just fix your extensions.conf |
| 01:23.40 | Mercestes | and say no to Guis. |
| 01:23.43 | Mercestes | friends don't let friends click. |
| 01:24.00 | Strom_C | click the dick and win $20 !!!!!!!!!!! |
| 01:24.26 | Mercestes | clicks on Strom_C and selects "whois" |
| 01:24.35 | Strom_C | hahahah |
| 01:24.58 | routerweasel | Mercestes: I did, changed context DID_trunk_1 to trunk_1, restart asterisk, dial in from outside = same result |
| 01:25.15 | Mercestes | :) |
| 01:25.28 | Mercestes | you restarted *? |
| 01:26.00 | Mercestes | You did this instead of an "extensions reload" because....? |
| 01:26.42 | plik | perhaps he couldn't find the "extensions reload" button |
| 01:26.46 | plik | to click on |
| 01:27.45 | routerweasel | lol...none the less, i still get the same result. |
| 01:29.56 | Mercestes | Random question, is your incoming call coming from SIP? |
| 01:30.02 | routerweasel | yes... |
| 01:30.11 | routerweasel | sip provider=nexvortex |
| 01:31.23 | Mercestes | I hate to do it but can you repastebin your extensions.conf and sip.conf with what you ahve now then? And, if possible output of extensions reload? |
| 01:31.35 | Mercestes | if there is a parse error in extensions.conf it won't actually update your changes. |
| 01:33.59 | routerweasel | http://pastebin.com/d7146fd6e |
| 01:34.04 | routerweasel | http://pastebin.com/d4394d8ff |
| 01:34.15 | routerweasel | after changes above... |
| 01:34.24 | routerweasel | http://pastebin.com/d5cee3058 |
| 01:34.28 | routerweasel | http://pastebin.com/d47cf5a93 |
| 01:34.30 | routerweasel | orig |
| 01:34.40 | Mercestes | dont need orig. |
| 01:34.44 | Mercestes | I need "now" |
| 01:34.50 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 01:34.56 | routerweasel | now is the first two urls |
| 01:35.21 | Mercestes | where is the sip definition for your sip provider? |
| 01:35.56 | *** join/#asterisk techie (n=techie@m0f0e36d0.tmodns.net) |
| 01:35.57 | routerweasel | that is all my sip provider gave me |
| 01:36.13 | Mercestes | yea but they aren't actually defined here. |
| 01:36.25 | Mercestes | you just set defauts for all other sip definitions beneath that. |
| 01:37.14 | Mercestes | You need a [nexvortex] with a username=nexvortex secret=nexvortex host=nextvortex_ip etc. |
| 01:38.07 | [TK]D-Fender | routerweasel: For the error you provided WAY back there, you have a DIALPLAN error. |
| 01:39.05 | Mercestes | ah crap, he's back |
| 01:39.06 | [TK]D-Fender | routerweasel: "chan_sip.c: Call from 'sip_username' to extension 'DID' rejected because extension not found. " <-- DIALPLAN. Now forget everything else, and provide a PASTEBIN of a failed call attempt at verbose 10, SIP DEBUG enabled. |
| 01:39.11 | [TK]D-Fender | ~mercestes |
| 01:39.12 | jbot | mercestes is definitely a total nub |
| 01:39.20 | Mercestes | :P |
| 01:39.39 | [TK]D-Fender | Mercestes: So.... escapen MN yet? :) |
| 01:39.44 | [TK]D-Fender | escaped* |
| 01:39.55 | Mercestes | MN as in Minnesota? |
| 01:39.58 | [TK]D-Fender | yes |
| 01:40.00 | routerweasel | i assumed it was dialplan related...but there is a wildcard to forward to the welcome voice menu |
| 01:40.03 | Mercestes | Yea, I'm back in Tx for now. |
| 01:40.10 | [TK]D-Fender | routerweasel: clearly not in the PLACE |
| 01:40.14 | Mercestes | Hey, where are you at? If I'm ever in your state we can grab a beer. |
| 01:40.30 | Mercestes | I was getting to that... |
| 01:40.34 | Mercestes | actually... |
| 01:40.37 | [TK]D-Fender | routerweasel: a baseball bat left in my car won't help help me win the pennent |
| 01:40.53 | Mercestes | because if there is a context= in his sip.conf for his provider that goes somewhere nonexistent (again) then ...etc. |
| 01:40.56 | [TK]D-Fender | Mercestes: I'm in "Disarray" ;) |
| 01:41.04 | Mercestes | I'm due there next week, actually |
| 01:41.19 | [TK]D-Fender | Mercestes: Montreal, QC actually.... you should remember... |
| 01:41.43 | routerweasel | ill get the verbose debug...in the meantime...this is what nexvortex gave me to setup asterisk! |
| 01:41.44 | routerweasel | http://www.spokanecomputing.com/nexvortexAsteriskNOW.pdf |
| 01:41.51 | [TK]D-Fender | Mercestes: Escape while you still can! |
| 01:41.58 | Mercestes | I do cover Montreal...and I should remember...but I consistently try to purge your memory. |
| 01:42.08 | [TK]D-Fender | routerweasel: Stop. NOW. Provide what I have requested. |
| 01:42.34 | [TK]D-Fender | Mercestes: You can't afford that kind of therapy :p |
| 01:42.39 | Mercestes | lol |
| 01:46.04 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 01:49.03 | routerweasel | http://pastebin.com/d262bf0fd |
| 01:49.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:50.48 | [TK]D-Fender | routerweasel: This is the part where you go back and read that I said WITH SIP DEBUG, and try it again... |
| 01:51.21 | Mercestes | add debug to logger.conf under console=> |
| 01:51.29 | [TK]D-Fender | NO |
| 01:51.35 | Mercestes | No don't do that! |
| 01:51.45 | [TK]D-Fender | just do "sip debug" at CLI and be done with it. |
| 01:51.57 | Mercestes | right.. |
| 01:52.05 | [TK]D-Fender | then "sip no debug" after you've captured the end of the attempt. |
| 01:53.43 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:54.28 | routerweasel | i did just that...CLI: sip debug. make the call, sip no debug" review log = same as what i pasted. |
| 01:55.14 | routerweasel | seriously...im at the point where i will pay someone to resolve this. VNC into my machine for all i care. |
| 01:55.27 | [TK]D-Fender | routerweasel: you've done it wrong then. ddon't type "cli", just do "sip debug" |
| 01:55.32 | routerweasel | name your *reasonable* price |
| 01:55.56 | [TK]D-Fender | routerweasel: just try it again already.. |
| 01:56.15 | [TK]D-Fender | "sip debug" <- 2 little words to pump in at * CLI... it ain't Raw-Cat Science |
| 01:56.50 | Mercestes | Did someone say pay? |
| 01:57.04 | [TK]D-Fender | routerweasel: and I asked for CLI OUTPUT WITH SIP DEBUG. You should not be looking for "logs"! |
| 01:57.30 | [TK]D-Fender | routerweasel: Log into * CLI and do what I told you. |
| 01:57.47 | Mercestes | ...but.....:( he said "pay"... |
| 01:57.54 | Mercestes | routerweasel, go into your linux shell and type asterisk -r |
| 01:57.58 | *** part/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 01:58.20 | Mercestes | routerweasel, then 'sip debug' |
| 01:58.34 | Mercestes | routerweasel, then set verbose 98 |
| 01:58.44 | Mercestes | then try to make the call...then copy paste that output to pastebin for Fender |
| 01:59.42 | routerweasel | sorry...i've been using the web gui...logging in with putty |
| 01:59.54 | Mercestes | No....you haven't..... |
| 01:59.59 | Mercestes | you are mistaken... |
| 02:00.14 | Mercestes | doesn't want to see fender assplode. |
| 02:06.22 | *** join/#asterisk rcy (n=rcy@66.183.58.28) |
| 02:06.23 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 02:09.23 | LiNeTuX|Home | Hey folks.. anyone here have any exp/tips when working with SIP and an MS ISA Server? |
| 02:09.58 | *** part/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 02:10.18 | [TK]D-Fender | LiNeTuX|Home: IIRC MS only uses TCP for SIP so they tend to put SipX in front of *, etc. |
| 02:10.51 | [TK]D-Fender | LiNeTuX|Home: mind you 1.6 is supposed to support TCP, but I wouldn't touch for a while even after it leaves "beta" |
| 02:11.05 | LiNeTuX|Home | [TK]D-Fender: Well, this is going the other way (an ext. behind an ISA box, trying to get to my * box) |
| 02:11.13 | LiNeTuX|Home | using 1.4 |
| 02:11.29 | LiNeTuX|Home | and I'm not cow-towing my server for this other person's problems |
| 02:11.48 | [TK]D-Fender | LiNeTuX|Home: well... what kind of "problems" do you have? |
| 02:12.09 | LiNeTuX|Home | [TK]D-Fender: The phone just won't register... (Aastra 53i) |
| 02:12.29 | [TK]D-Fender | LiNeTuX|Home: SIP debug please.. you know the deal... |
| 02:12.34 | LiNeTuX|Home | [TK]D-Fender: he's looking for a place to put credentials in to have the phone reg. with the ISA box |
| 02:13.06 | [TK]D-Fender | LiNeTuX|Home: and tell him to make sure ISA doesn't fudge up SIP transform or anything stupid. I've heard MS ISA doing strange crap before. |
| 02:13.09 | LiNeTuX|Home | [TK]D-Fender: I don't have any - there's nothing hitting my side and this isn't my network... I'm about >this< close to telling him to get a Linksys or something |
| 02:13.39 | [TK]D-Fender | LiNeTuX|Home: if its not reaching you it could eb the phone, or it could be ISA itself. |
| 02:13.40 | LiNeTuX|Home | [TK]D-Fender: I've been reading, and it seems like there are some 3rd party addons... but I'm not real impressed with this guys' network-fu. |
| 02:14.17 | LiNeTuX|Home | [TK]D-Fender: I'm sending another phone JIC, but I think it's the ISA. Haven't had an issue with 20 other deploys on the same system. |
| 02:14.31 | [TK]D-Fender | LiNeTuX|Home: I've heard it before. |
| 02:14.51 | LiNeTuX|Home | [TK]D-Fender: Heard what? It not working or it does work? |
| 02:15.24 | LiNeTuX|Home | It IS A MS product after all... I mean, what should we really expect ... standards support?!? |
| 02:16.15 | LiNeTuX|Home | thinks he also mucked with the phone config and screwed that up too. |
| 02:19.15 | NovceGuru | LiNeTuX|Home: make sure ISA doesn't do any source port re-writing |
| 02:19.53 | LiNeTuX|Home | NovceGuru: Thanks for the 411, but I don't think me saying that to this guy is going to help anything. I think it'll be wasted breath. |
| 02:20.00 | *** join/#asterisk techie (n=techie@m6e0e36d0.tmodns.net) |
| 02:20.04 | NovceGuru | ahhh phone support yayyy |
| 02:20.30 | LiNeTuX|Home | sometimes the worst kind of help is when the other end is a know-it-all |
| 02:20.43 | NovceGuru | esp a M$ know-it-all |
| 02:20.50 | LiNeTuX|Home | uh ... yeah |
| 02:21.01 | LiNeTuX|Home | because the MS way is the only way. |
| 02:21.14 | NovceGuru | duh |
| 02:22.36 | LiNeTuX|Home | How's that saying go... "There's 'A' way, and there is no 'B' way." |
| 02:23.31 | NovceGuru | it's called |
| 02:23.33 | LiNeTuX|Home | rummages around for a good ISA exploit to take it offline... |
| 02:23.46 | NovceGuru | "Sir I'll come look at it for $199/hr" |
| 02:24.13 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:24.18 | LiNeTuX|Home | I wish I could do that. But this is a 'sister company' for lack of a better way of explaining it. |
| 02:24.28 | NovceGuru | ahh |
| 02:24.56 | LiNeTuX|Home | So they do it their way, and they think everyone else is an idiot who doesn't know what they're doing. |
| 02:26.31 | NovceGuru | Can sadly say I can relate |
| 02:26.46 | LiNeTuX|Home | I think we've all been there to some degree. |
| 02:26.57 | NovceGuru | yeah |
| 02:27.33 | LiNeTuX|Home | Anyone ever put an extension behind a Juniper appliance/firewall? |
| 02:28.26 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:28.31 | NovceGuru | Can't say I have |
| 02:29.00 | LiNeTuX|Home | We kept getting timeouts on any ext. put behind this one box... they tried changing all kinds of settings and eventually just put the phone right on the 'net. |
| 02:29.15 | LiNeTuX|Home | can't believe it's really that hard to manage. |
| 02:29.35 | NovceGuru | LiNeTuX|Home: what is the udp timeout on the appliance vs the qualify time on the asterisk box |
| 02:29.59 | NovceGuru | all of my bsd boxes have a default of 60 sec, which is the default qualify time in * |
| 02:29.59 | LiNeTuX|Home | NovceGuru: Phone was 120, they tried putting the fw at 122... then the phone wouldn't even register at all! |
| 02:30.19 | LiNeTuX|Home | actually, maybe it was 60 on my side. I can't remember now. |
| 02:30.34 | NovceGuru | should up it on the juniper to like 1.5 the timeout |
| 02:30.49 | LiNeTuX|Home | they gave up i think. |
| 02:31.05 | NovceGuru | I had to fight hard to get it working on pfsense |
| 02:31.27 | NovceGuru | udp timeouts, tftp doesn't work behind it, couldn't have more then 1 device behind a single wan ip |
| 02:31.30 | LiNeTuX|Home | I thought pfsense was fairly easy to work with in that regard? |
| 02:31.56 | NovceGuru | it just SUCKs in it's default settings for sip |
| 02:32.02 | LiNeTuX|Home | uses Cisco or linux-based fw's |
| 02:32.40 | NovceGuru | http://www.pfsense.com/index.php?option=com_content&task=view&id=40&Itemid=43 |
| 02:32.43 | NovceGuru | ctrl F sip |
| 02:33.06 | NovceGuru | 1.3 will have siproxd and a tftp proxy, so until then, its hax hax hax |
| 02:33.25 | LiNeTuX|Home | ahhhh |
| 02:33.57 | LiNeTuX|Home | SmoothWall has a nice SIP proxy setup if you need it. Well, at least the commercial version does. |
| 02:34.18 | LiNeTuX|Home | But it seems to 'just work' |
| 02:35.18 | NovceGuru | yeah |
| 02:35.35 | NovceGuru | from what I investigated everything that actually "works" is breaking rfc |
| 02:35.49 | NovceGuru | in terms of tftp/nat |
| 02:35.58 | mchou | LiNeTuX|Home: so you get any traces on Juniper fw? |
| 02:36.24 | mchou | LiNeTuX|Home: be interested in what you find out |
| 02:36.44 | LiNeTuX|Home | mchou: I could, but I don't have any. Again, not my fw... and I'm not sure how much time the other side wants to spend messing around with it. |
| 02:37.02 | mchou | LiNeTuX|Home: ok, fair enough |
| 02:37.21 | mchou | LiNeTuX|Home: I thought it was your fw :) |
| 02:37.37 | LiNeTuX|Home | I'll talk with 'em again... I'm going to change some setups on the phone to allow for more dynamic changing of the registration settings as well |
| 02:37.59 | LiNeTuX|Home | mchou: No, other side... I have the * server, they've got the extension. |
| 02:41.01 | mchou | lol |
| 02:41.12 | mchou | grandcentral just went down hard |
| 02:43.46 | NovceGuru | the service or just the website? |
| 02:44.04 | NovceGuru | wtf is that phone hooked to a reel to reel player |
| 02:44.10 | NovceGuru | and a analog tv :( |
| 02:44.59 | mchou | NovceGuru: I think both |
| 02:45.14 | mchou | NovceGuru: Web is definitely down |
| 02:45.20 | NovceGuru | yeah |
| 02:45.28 | NovceGuru | I dont have/care to have an account so I donno :| |
| 02:45.45 | mchou | NovceGuru: that's otherwise known as a Tivo :) |
| 02:46.07 | NovceGuru | ahhh |
| 02:46.15 | NovceGuru | still doesn't fit! |
| 02:46.25 | NovceGuru | they need a phone that just got ran over by a truck or something |
| 02:46.46 | mchou | Tivo actually isnt connected to a phone |
| 02:46.54 | mchou | just a phone line |
| 02:46.57 | NovceGuru | or just http://www.tomandang.com/blog/images/smashedComputer.jpg |
| 02:47.10 | NovceGuru | that would be _AWESOME_ to set a high traffic site to |
| 02:47.40 | NovceGuru | along with some choice words |
| 02:47.46 | mchou | NovceGuru: which site you referring to? Tivo or the pic? |
| 02:47.54 | *** join/#asterisk dmz (n=dmz@m250f36d0.tmodns.net) |
| 02:48.03 | NovceGuru | mchou: the pic on grandcentral.com |
| 02:54.46 | LiNeTuX|Home | loves this one... I found a vendor that has an addon for ISA server to make it work with SIP... they say they "Innovating ISA Server" ... I say they're making it work... |
| 02:55.59 | NovceGuru | haha |
| 02:56.32 | NovceGuru | can you setup a vpn? |
| 02:56.51 | LiNeTuX|Home | hey... now there's a thought... |
| 02:59.28 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 02:59.34 | NovceGuru | throw openvpn on both ends |
| 02:59.38 | NovceGuru | on the winders boxes |
| 03:00.06 | NovceGuru | the l2tp junk built into the servers might even work, as long as they aren't ipsec based |
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| 03:12.41 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
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| 03:38.40 | smach | guys I'm wondering, do sip compliant phones accept a "305 use proxy" response to a "REGISTER" |
| 03:46.48 | *** join/#asterisk pcrack (n=pcack@202.164.176.106) |
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| 04:04.33 | [TK]D-Fender | Wow, just found a guy selling Linksys SPA-1001's for 15$ea.... |
| 04:05.39 | *** join/#asterisk hfb (n=hfb@cpe-76-87-173-149.socal.res.rr.com) |
| 04:07.44 | *** join/#asterisk Gwayne (n=Gwayne@bb219-74-53-242.singnet.com.sg) |
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| 04:23.44 | unpaidbill | that reminds me, i need to buy a sip hardphone |
| 04:31.43 | Mercestes | what's wrong with them? |
| 04:34.19 | Mercestes | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 04:34.28 | Mercestes | http://www.demotivateus.com/ebay-its-cheaper-for-a-reason-demotivational-poster/ |
| 04:34.29 | Mercestes | rather |
| 04:37.27 | [TK]D-Fender | http://despair.com/med24x30prin.html |
| 04:38.35 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 04:48.16 | Mercestes | lol |
| 05:00.17 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
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| 05:07.39 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:08.39 | trnzmeta | guys: the file that contains the registry string? |
| 05:09.33 | *** join/#asterisk steliosk (n=Stelios@85.75.202.96) |
| 05:10.39 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 05:11.00 | Strom_C | trnzmeta: that's not even a complete sentence, let alone a question that makes any sense |
| 05:11.19 | trnzmeta | hahaha nvm got it now :) |
| 05:13.27 | Strom_C | what were you looking for, anyway |
| 05:15.27 | *** join/#asterisk MrNaz (n=naz@ppp121-44-247-162.lns4.mel4.internode.on.net) |
| 05:15.33 | [TK]D-Fender | Strom_C: AstDB |
| 05:15.51 | [TK]D-Fender | Strom_C: If he;s referring to getting info on his phones. |
| 05:16.01 | [TK]D-Fender | Strom_C: Channel driver file otherwise |
| 05:16.07 | Strom_C | [TK]D-Fender: yeah, that's the thing; i can think of at least three different ways to interpret that |
| 05:17.31 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 05:17.54 | Robba | hi guys. |
| 05:18.20 | Robba | is anyone using Polycom Soundpoint 501 SIP handsets? |
| 05:18.40 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.228) |
| 05:18.54 | Strom_C | no -- no one in the history of history itself has ever used any sort of polycom phone for anything ever ever ever |
| 05:19.20 | mcab | Strom_C: I sense a hint of sarcasm... just a hunch though |
| 05:19.21 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.106.228) |
| 05:19.29 | Strom_C | sarcasm?! me?! |
| 05:19.33 | Robba | i was hoping so |
| 05:19.37 | Strom_C | hides the polycom phone under the rug |
| 05:19.40 | mcab | Strom_C: neavar |
| 05:20.07 | mcab | Robba: what's up with the Polycoms? |
| 05:20.26 | Robba | i can't seem to figure out how to get a polycom handset to allow ## as a blind transfer. any suggestions? |
| 05:20.31 | Strom_C | Robba: also, the handset is only the part you hold in your hand. |
| 05:20.41 | Robba | sip device? |
| 05:20.45 | Strom_C | Robba: you don't do blind transfer with ## on a polycom phone |
| 05:20.53 | Strom_C | you use the "transfer" softkey |
| 05:21.04 | Robba | curses |
| 05:21.09 | Strom_C | ## transfer is a nasty hack for phones that don't have any sort of other transfer capability |
| 05:21.19 | Robba | i see |
| 05:21.36 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 05:21.43 | Robba | hmmm... my collegues are now used to ## as a bxfer |
| 05:21.59 | [TK]D-Fender | Robba: poor schmucks |
| 05:21.59 | Robba | so is there any way to make it work? |
| 05:22.16 | [TK]D-Fender | Robba: Holy crap you don't WANT stupid DTMF transfers |
| 05:22.24 | Strom_C | Robba: tell them to get used to that button that says "XFER" |
| 05:22.28 | Robba | lol |
| 05:22.30 | [TK]D-Fender | Robba: go read the USERS guide (not even the ADMIN guide) for the phone. |
| 05:23.31 | Robba | i already undertand that you can use the soft button and even how to reprogram it. I just thought there maybe a way. |
| 05:23.44 | [TK]D-Fender | Robba: features.conf |
| 05:23.54 | [TK]D-Fender | Robba: but thats jsut wrong. |
| 05:26.25 | Robba | its already configured in there. just when in call pressing ## does nothing. you can hear the ## tones the "handset" lol but no prompt for transfer |
| 05:27.48 | [TK]D-Fender | Robba: pastebin the call attempt at verbose 10 |
| 05:28.04 | Robba | ok brb |
| 05:31.05 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 05:31.24 | Robba | ~pastebin |
| 05:31.25 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:32.56 | *** join/#asterisk af_ (n=getsmart@88-149-240-27.dynamic.ngi.it) |
| 05:33.00 | Robba | http://rafb.net/p/ZuEbJi53.html |
| 05:33.08 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 05:33.31 | Robba | i think i may have just clicked to what it could be? |
| 05:33.36 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 05:33.41 | Robba | Tt |
| 05:33.44 | [TK]D-Fender | Robba: After updating features.conf, did you restart *? |
| 05:33.46 | Robba | shouldn't it be tT |
| 05:33.54 | [TK]D-Fender | Robba: order doesn't matter |
| 05:33.56 | Strom_C | Robba: doesn't matter |
| 05:34.21 | Robba | ## as blind xfer is working on our linksys handsets |
| 05:34.33 | [TK]D-Fender | Robba: Then again, you should be giving the PSTN access to transfer themselves, but that isn't the problem you're working on. |
| 05:34.36 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 05:34.47 | [TK]D-Fender | Robba: perhaps you didn't set the DTMFmode properly |
| 05:34.56 | Robba | its set to auto |
| 05:35.02 | [TK]D-Fender | Robba: make sure its rfc2833 |
| 05:35.25 | *** join/#asterisk robeph (i=1022@grumpy.asteriasgi.com) |
| 05:35.27 | [TK]D-Fender | Robba: What Linksys models? |
| 05:35.36 | robeph | off hand is there any way to make the asterisk console display wider than 80 columns? |
| 05:35.42 | robeph | cos I'm getting my channel list chopped =\ |
| 05:35.49 | Robba | yeah... i'd like to but, it seems our carrier doesn't like rfc2833 |
| 05:35.56 | Robba | it seems to like INFO |
| 05:36.02 | [TK]D-Fender | Robba: this has NOTHING to do with your carrier |
| 05:36.16 | Robba | sorry side tracked thats another problem |
| 05:36.34 | [TK]D-Fender | Robba: * talks to your phone with one set of characteristics. It uses a completely separate set for your carrier |
| 05:36.47 | [TK]D-Fender | Robba: and * sits int he middle translating |
| 05:36.58 | Robba | sorry i will question that in a minute |
| 05:37.14 | [TK]D-Fender | robeph: what option are you using? |
| 05:37.38 | Robba | but yeah pressing ## doesn't seem to initiate the bxfer |
| 05:37.51 | [TK]D-Fender | Robba: because the mode is probably wrong. |
| 05:38.16 | Robba | ok now here is the kicker |
| 05:38.33 | Robba | we are a channel partner to optus australia |
| 05:38.47 | Robba | so we call them support numbers quite a bit |
| 05:38.48 | [TK]D-Fender | Robba: time is short, don't waste it. |
| 05:39.03 | robeph | [TK]D-Fender: just -r |
| 05:39.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e7dc791a5fc7e5f1) |
| 05:39.20 | [TK]D-Fender | robeph: No, what option is being truncated? |
| 05:39.25 | robeph | oh show channels |
| 05:39.26 | Robba | there IVR doesn't work properly when its set to anything other than auto |
| 05:39.35 | robeph | it's like bridged(SIP/2322 ~~~~~~ |
| 05:39.42 | [TK]D-Fender | robeph: IIRC that will ALWAYS truncate. use "show channels conciser" |
| 05:39.45 | robeph | ah |
| 05:39.51 | robeph | it chops the end off? |
| 05:39.52 | [TK]D-Fender | robeph: IIRC that will ALWAYS truncate. use "show channels concise" |
| 05:40.08 | [TK]D-Fender | robeph: it "columnizes" it |
| 05:40.17 | [TK]D-Fender | robeph: and the defaults suck |
| 05:40.24 | [TK]D-Fender | robeph: no matter. |
| 05:40.42 | [TK]D-Fender | Robba: your itsp setting have NOTHING TO DO WITH THE PHONE! |
| 05:40.59 | robeph | ah |
| 05:41.03 | robeph | how annoying ;p |
| 05:41.43 | Robba | itsp? |
| 05:42.24 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 05:42.38 | [TK]D-Fender | Robba: .... nevermind. your linksys use "info" by default. just sset the damned mode to rfc2833 for your polycom. |
| 05:43.58 | Robba | ok thats done |
| 05:44.27 | [TK]D-Fender | Robba: ...and? |
| 05:44.40 | Robba | the phone seems to take a while to restart |
| 05:44.52 | [TK]D-Fender | Robba: SDSDDJHSDGJKSDJGSDKSDGUIOWEUIOWE |
| 05:45.00 | [TK]D-Fender | Robba: in the stupid asterisk sip.conf peer! |
| 05:45.28 | [TK]D-Fender | Robba: there goes several minutes of reboot for nothing |
| 05:49.36 | Robba | ok well the 501 is in use now but it seems to work on the 430 |
| 05:51.00 | [TK]D-Fender | Robba: fine, should work on both, but don't go rebooting them for nothing. |
| 05:51.22 | [TK]D-Fender | Robba: and FFS you have REAL PHONES. You should not be using DTMF garbag for this. ESPECIALLY with Polycoms. |
| 05:51.31 | robeph | I've noticed a very odd corrolation with stuck/locked channels and a strange cascade of events |
| 05:51.36 | [TK]D-Fender | Robba: You are throwing away their superior call handling. |
| 05:52.04 | Robba | i understand this |
| 05:52.16 | Robba | but the collegues don't |
| 05:52.17 | robeph | if we notice a bunch of locked channels...and then you keep doing "show channels" eventually the client goes bonkers... eg. hangs real bad, you can ctrl-c and then rejoin it, but you can't so much as "restart now" it just sits there and keeps giving you the prompt heh |
| 05:52.57 | robeph | if you know you have stuck channels from other symptoms (eg a phone showing as the line in use, but no way to kick it or otherwise) but you don't do show channels, you can easily just restart asterisk |
| 05:53.36 | Strom_C | Robba: just tell them "It won't work with these phones -- use the XFER button" |
| 05:53.40 | Strom_C | problem solved |
| 05:53.46 | Strom_C | let them whine all they want |
| 05:53.51 | Robba | lol |
| 05:54.35 | Robba | then i have the boss buggin me |
| 05:54.41 | [TK]D-Fender | Robba: this is far beyond stupid. Polycoms can split megre and transfer calls better than any other model and you are treating them no better than some shitty 10$ phone on a zaptel channel. |
| 05:54.50 | [TK]D-Fender | merge* |
| 05:55.21 | [TK]D-Fender | Robba: Do your groceries in a Ferrari as well? |
| 05:55.38 | Robba | hey i don't have many choices here |
| 05:55.43 | Robba | i just do what i get told |
| 05:55.48 | [TK]D-Fender | INFORM THEM |
| 05:55.49 | Robba | i'm not the boss |
| 05:55.54 | *** join/#asterisk dssman (n=no@CPE00146c48b625-CM0011aec559de.cpe.net.cable.rogers.com) |
| 05:55.57 | [TK]D-Fender | try actual training. |
| 05:55.58 | dssman | yoo all |
| 05:56.09 | [TK]D-Fender | Anyways, I'm off. Nite all. |
| 05:56.10 | Robba | hahahahaha i wish |
| 05:56.14 | Robba | thanks |
| 05:56.44 | dssman | anyone awake that has a good north american dial plan for cisco? |
| 05:56.56 | Strom_C | dssman: NANP is easy as hell |
| 05:57.00 | Strom_C | 1NXXNXXXXXX |
| 05:57.09 | dssman | well then :P |
| 05:57.16 | dssman | I guess I can use that for now |
| 05:57.17 | Robba | i wish AU was the same as that |
| 05:57.34 | Strom_C | dssman: check www.nanpa.com for details |
| 05:58.17 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 05:58.33 | dssman | fanx |
| 05:59.01 | dssman | :@ I cant get a dial tone on my cisco ata 186 :@:@ |
| 05:59.26 | robeph | sup strom :p |
| 06:00.22 | Strom_C | hey rob |
| 06:00.24 | Strom_C | not a lot |
| 06:00.28 | Strom_C | i made a new 80s mix :) |
| 06:00.39 | robeph | heheh |
| 06:00.47 | robeph | still in the 80s |
| 06:00.49 | Strom_C | want a copy? |
| 06:00.50 | robeph | any abba in this one? |
| 06:00.52 | robeph | sure ting |
| 06:00.54 | Strom_C | heh, no abba |
| 06:00.57 | robeph | :'( |
| 06:01.06 | Strom_C | i'm looking to branch out and do other genres, but i havent found one i like enough yet |
| 06:01.08 | robeph | ok I don't...*backs away* really like abba... |
| 06:01.14 | robeph | dnb ;) |
| 06:01.36 | Strom_C | http://www.stromcarlson.com/dj/Strom%20Carlson%20-%20Time%20and%20Salt.mp3 |
| 06:02.50 | robeph | if ff3 would stop trying to play it with mplayerplugin....meh |
| 06:03.01 | Strom_C | go up a level, right click, save as :) |
| 06:03.11 | robeph | I just save page |
| 06:03.17 | Strom_C | that works too |
| 06:03.28 | robeph | then close the window, my gf was lookin at me funny lol |
| 06:03.45 | Strom_C | hehehe |
| 06:03.57 | robeph | think I should keep my volume down a bit =\ .... |
| 06:04.17 | Strom_C | http://www.stromcarlson.com/dj/Strom%20Carlson%20-%20Time%20and%20Salt.txt <-- there's a tracklist and md5sum in case you were curious |
| 06:05.07 | robeph | anyone got an idea how to say, if you got a call made outbound, that conects to an ivr, then the called presses 1, transfers to another number (say 2345678901) is there an easy way to see this transfer in the channel list? eg how would it show up (in conjunction with the original number that was called and transfered to the secondary number post ivr?) |
| 06:05.37 | Strom_C | Dial() plus the LOCAL channel? |
| 06:06.04 | robeph | ah yeh, its gonna Dial 234..... |
| 06:06.30 | Strom_C | adding a call leg onto an existing call really isn't a "transfer" |
| 06:06.34 | robeph | yeah |
| 06:06.39 | robeph | misnomer |
| 06:07.03 | robeph | thats how its referred to to the end point clients, so just guess I'm confusing myself |
| 06:08.26 | robeph | does 1.4 fix the overflow on a -rx? |
| 06:08.47 | Strom_C | overflow? |
| 06:08.48 | robeph | well overflow is bad definition. eg -rx 'show channels' and I get all the dial strings and such |
| 06:08.52 | robeph | that isn |
| 06:09.00 | robeph | t' related to the show channels |
| 06:09.33 | *** join/#asterisk [hC] (n=hardcore@12.46.54.90) |
| 06:09.35 | robeph | like I get way more outta the console return than just 'show channels' should produce, I get 'spillage' |
| 06:09.36 | Strom_C | well, the Dial() is there to show you what's being executed, IIRC |
| 06:09.52 | robeph | no no I mean I see ACTUAL dials... |
| 06:10.01 | Robba | ok so can someone tell me what Tt and tT does? |
| 06:10.04 | Strom_C | as in what people are dialing? |
| 06:10.14 | Strom_C | Robba: "core show application Dial" at the CLI |
| 06:10.31 | robeph | <PROTECTED> |
| 06:10.31 | robeph | <PROTECTED> |
| 06:10.42 | hardwire | wowsa |
| 06:10.45 | robeph | like all that crap comes out, in conjunction with -rx 'show channels' |
| 06:10.52 | Strom_C | ah, as in what's happening live on the CLI |
| 06:10.55 | robeph | yeah |
| 06:10.58 | robeph | spillage ;p |
| 06:11.15 | robeph | its overflowing into the buffer before it closes it after 'show channels' is executesd |
| 06:11.37 | Robba | so if Tt is there it really doesn't make a difference which order it appears? |
| 06:11.44 | Strom_C | yeah, i don't know really, robeph |
| 06:11.58 | robeph | I could hack that out maybe... need to look how asterisk handles the -rx, maybe make a differnt tmp handle to output just the executed commands stdout |
| 06:12.07 | robeph | not sure how asterisk hands its output to stdout.. |
| 06:12.19 | Strom_C | Robba: none of the option flags have any order preference |
| 06:12.26 | Robba | ok |
| 06:12.27 | Robba | thanks |
| 06:13.08 | robeph | Strom_C: ultra annoying when you have massive amounts flowing by and need just a channel list, I have to make somewhere around 500+ scroll back to catch the channel list heh with all that spills out, |
| 06:14.07 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 06:14.12 | Strom_C | i'll ask on your behalf if you don't get an answer |
| 06:14.28 | robeph | I'm sure its a problem in the way its output is handled |
| 06:14.49 | robeph | seems it has a timewait perhaps on the output from a command line execute.. |
| 06:14.57 | bijit | hi everyone, new to asterisk... any recomendations for me to start? |
| 06:15.03 | Strom_C | bijit: read the book |
| 06:15.05 | Strom_C | ~book |
| 06:15.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 06:15.07 | robeph | bijit: that's a somewhat...generalized question |
| 06:15.41 | bijit | sorry. I just want to know where I can start. |
| 06:15.59 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 06:16.13 | robeph | bijit: here, try not thinking of it in those turms |
| 06:16.15 | robeph | terms... |
| 06:16.19 | robeph | or spell like me... don't do that |
| 06:16.20 | *** join/#asterisk ManxPower (n=manxpowe@228.sub-70-222-71.myvzw.com) |
| 06:16.28 | bijit | jbot, thank you I am downloading. |
| 06:16.28 | jbot | bijit: de nada |
| 06:16.41 | robeph | try thinking more about "what you want to do" and where to start will come naturally to that |
| 06:17.43 | bijit | tu hablas español, cool |
| 06:18.08 | bijit | thanks for the tip robeph |
| 06:18.52 | robeph | bijit: it's like coding, you want to learn a language... but why? you can't start if you have no concept of what you'll want to do with it. =s |
| 06:19.14 | robeph | are there any good books in spanish for asterisk? that i dunno |
| 06:20.00 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:20.28 | bijit | I don't know, I prefer english books, when they are translated they tend to be totally different |
| 06:20.40 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 06:20.52 | Strom_M | dear Internet: |
| 06:20.54 | Strom_M | what the fuck |
| 06:20.55 | robeph | hahahaha |
| 06:20.56 | Strom_M | love, |
| 06:20.58 | Strom_M | Strom |
| 06:21.12 | J4zen | Did anyone ever import hardware from the USA to Europe? |
| 06:21.23 | J4zen | with the profitable exchange rate in mind |
| 06:21.41 | robeph | I frown at your euros |
| 06:21.47 | robeph | now I can't buy anything from europe =\ |
| 06:21.49 | robeph | used to all the time |
| 06:22.01 | robeph | now, 50$ is like 100 after import and all meh |
| 06:22.14 | J4zen | yep, good business |
| 06:22.20 | J4zen | blame bush |
| 06:22.30 | robeph | I'd fire him if I was his boss |
| 06:22.33 | robeph | wait...aren't? |
| 06:22.34 | robeph | I |
| 06:22.43 | J4zen | lol |
| 06:22.45 | robeph | I should send him a letter telling him he's fired... |
| 06:22.53 | robeph | I wonder what kinda whitehouse canned response I'd get, |
| 06:23.12 | robeph | I always get canned responses from my congress people... |
| 06:23.29 | J4zen | newsbroadcast today stated that he's spent 800 billion on the war in iraq up until now, go figure where your dollars went ;) |
| 06:23.34 | robeph | about current issues etc...sorta canned like...oddly worded... i think they have an adlib email -> snail mail reply system |
| 06:23.49 | robeph | i dont care where the dollars went, actually, its that they keep printing more... |
| 06:24.15 | J4zen | That be true, yarr |
| 06:24.23 | robeph | instead of printing more they should recoup the losses by fining people who gum up the works by forcing us to spend / overcharging / whatever the hell they do =) |
| 06:24.43 | robeph | eg. halliburton... why print money when you can just take it back! |
| 06:24.58 | rift0r | any of you guys use astguiclient |
| 06:25.00 | rift0r | or vicidial |
| 06:25.04 | J4zen | nope |
| 06:25.24 | robeph | J4zen: I got told to not tip the ladies at a club with euros that I traded some dollars in for at my bank =s |
| 06:25.59 | robeph | of course i did it for the humor of it all, and the reaction was less ...bad for me than when I was using those coin dollars... they didn't like those much either |
| 06:26.15 | robeph | rift0r: not i |
| 06:26.25 | J4zen | lol, funny how an import question can lead to tipping ladies in a club with dollar coins |
| 06:27.07 | robeph | well it was euros latter, before I'd gon in with a tube full of susan b anthony dollar coins =\ |
| 06:27.28 | robeph | i'm not partial to strip joints, but it's always fun to go in seeing the reaction to akward currencies |
| 06:27.37 | J4zen | lol |
| 06:27.49 | J4zen | 1 dolla love u long time |
| 06:28.00 | robeph | hahah |
| 06:28.17 | robeph | ok bed time its 1.30.... |
| 06:28.26 | J4zen | aye, gnight |
| 06:28.29 | robeph | latez |
| 06:30.04 | J4zen | Ok. So back to my original question: Any european who imported hardware from USA? |
| 06:30.07 | J4zen | and doesn't feel like sharing the story of his life with me ;-) |
| 06:33.29 | Strom_M | when I was just a young boy... |
| 06:33.36 | Strom_M | [thirty-eight screens of text later] |
| 06:34.19 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-8099985d6957fab4) |
| 06:35.39 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 06:42.19 | Robba | does anyone know about DTMF regeneration? |
| 06:51.32 | *** part/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 06:53.46 | creativx | J4zen: just for fun.. i import some goods from the us.. but not telephony :) |
| 06:56.51 | *** join/#asterisk c0ldk1ll3r (n=matias@67-207-133-226.slicehost.net) |
| 06:57.23 | c0ldk1ll3r | hi.. why does the guys in #freeswitch talks too much about asterisk sucking? |
| 06:58.27 | creativx | who knows, they might enjoy sucking |
| 06:58.35 | creativx | do you think asterisk scks? |
| 06:58.44 | c0ldk1ll3r | no |
| 06:59.11 | c0ldk1ll3r | it's just funny |
| 06:59.18 | c0ldk1ll3r | they mention asterisk every 5 minutes there |
| 06:59.34 | creativx | does it really matter? |
| 06:59.56 | J4zen | creativx: How do you handle taxes? do you report them to your local tax-office, ever imported "grey"? |
| 07:01.50 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-222-63.telkomadsl.co.za) |
| 07:02.15 | xacatecas | hi guys, if I get "Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from 192.168.1.100" in my asterisk logs, what could possibly be causing this? |
| 07:02.19 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 07:02.31 | xacatecas | I've put that string into google with no results. |
| 07:03.07 | xacatecas | ok, never mind (for now), removing the IP gets better results :) |
| 07:15.59 | creativx | J4zen: the package value is reported by the company transporting it.. that be ups or usps for instance.. ups handles tax claims themselves, based upon the value.. usps uses the national postal service which then does the tax claims |
| 07:17.07 | creativx | J4zen: but usually when buying from exporting firms they will sell the goods without taxes |
| 07:17.26 | J4zen | So it is in fact worthwhile to order in the US right now? |
| 07:17.29 | J4zen | i have a couple of polycom |
| 07:17.53 | J4zen | and the prices of those phones are 300Euros, while only 250 $ (don't see how that happened in the first place) |
| 07:18.22 | J4zen | or would you say the total price after taxes + import would be over 300E ? |
| 07:19.38 | *** join/#asterisk smach (n=smach@bas15-ottawa23-1279406959.dsl.bell.ca) |
| 07:28.50 | *** join/#asterisk eyecue (n=eyecue@miranda/admin/eyecue) |
| 07:29.10 | eyecue | Morning, can one serve the asterisk-gui via lighttpd/apache/someotherwebserver ? |
| 07:30.01 | eyecue | And further, can one modify the path to the gui docroot, I tend not to remember '/asterisk/static/config/cfgbasic.html' |
| 07:34.33 | *** join/#asterisk steliosk (n=Stelios@athedsl-168642.home.otenet.gr) |
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| 07:39.02 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 07:39.11 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:40.07 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 07:40.54 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:41.17 | creativx | J4zen: indeed very worthwile.. i order heaps of car parts for my project from over the pond :) |
| 07:41.32 | creativx | J4zen: but it ofcourse depends on your country |
| 07:41.48 | creativx | or the eu i guess |
| 07:42.01 | *** part/#asterisk bijit (n=benji@200.122.188.156) |
| 07:42.23 | J4zen | creativx: alright, thanks for the info :) |
| 07:42.44 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 07:44.06 | eXistenZ | ganz komisch |
| 07:44.38 | J4zen | nein, leider nicht |
| 07:44.43 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:45.29 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 07:46.10 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 07:46.59 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:47.23 | eXistenZ | J4zen, hast du schon mal spa3102 benutzt? |
| 07:47.39 | flenders | anyone has any bad things to say about the TDM410? I was gonna go for the A200 with HW echo, but now the TDM410 with HW EC is cheaper... any reason why I should still go with sangoma? |
| 07:49.22 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:57.27 | bijit | hi |
| 07:59.21 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:00.38 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 08:04.29 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 08:06.51 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 08:08.55 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 08:14.07 | bijit | bijit: |
| 08:14.58 | bijit | bijit: |
| 08:15.16 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 08:16.23 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 08:16.28 | bijit | bijit: |
| 08:16.38 | bijit | bijit |
| 08:17.43 | *** join/#asterisk und3r (i=root@76.76.103.22) |
| 08:17.44 | und3r | hello |
| 08:18.31 | bijit | hi |
| 08:18.32 | und3r | is there anyone that can send me an example configuration about a Patton SmartNode 4960 for work with Asterisk? |
| 08:19.08 | *** join/#asterisk clive- (n=pirch@dsl-244-222-197.telkomadsl.co.za) |
| 08:19.17 | clive- | is anyone here in germany ? |
| 08:19.18 | und3r | i've googled it but i've not find it :) |
| 08:20.49 | *** join/#asterisk salzh (n=salzh@116.233.145.18) |
| 08:21.15 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 08:25.55 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 08:30.18 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-12-53.revip2.asianet.co.th) |
| 08:30.56 | HaMYaI | is there a default hangup key in asterisk? |
| 08:31.21 | HaMYaI | like a '*' or something |
| 08:31.39 | clive- | hamyal. * but you need a h in your dial parameter |
| 08:32.34 | HaMYaI | clive-: right, I have the h ext, but wanna disable the '*' |
| 08:32.40 | HaMYaI | is it possible? |
| 08:32.58 | clive- | yes, look in features.conf |
| 08:33.41 | clive- | or leave outthe "h", not sure, play around and see |
| 08:34.36 | HaMYaI | yeah, in features.conf is where I was trying to find |
| 08:35.27 | *** join/#asterisk XnOSX (n=XnOSX@212.145.170.219) |
| 08:36.23 | HaMYaI | okay found that, thanks |
| 08:52.24 | fiddur | I can't find anything about RealTime agents... isn't there any support for that in asterisk? ...or am I supposed to use realtime queues to get a similar result...= |
| 08:52.26 | fiddur | ? |
| 09:01.18 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
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| 09:30.21 | J4zen | Does anyone have any expierence programming an interface with MS Exchange server? |
| 09:30.34 | J4zen | Trying to integrate * into Exchange |
| 09:33.25 | gr0mit | erm, J4zen - i think we need more info! |
| 09:37.22 | J4zen | Actually no, why would you? Im just curious if anyone ever coded any soft of value-added application for Exchange |
| 09:37.38 | J4zen | And how they did that, MAPI, SDK, 3rd party launchers, activeX? |
| 09:37.39 | J4zen | etc |
| 09:38.13 | gr0mit | aah ok |
| 09:38.22 | gr0mit | no - we all use a proper mail server ;-) |
| 09:39.09 | J4zen | depends on the implementation |
| 09:39.40 | J4zen | we're operating with large networks in Terminal sessions using shared inboxes and a lot of office-integrated applications |
| 09:39.57 | J4zen | it is by far the easiest and most solid solution to go for exchange in such an implementation |
| 09:39.59 | gr0mit | hmmm. YAOYO then |
| 09:40.30 | J4zen | lol YAOYO? |
| 09:40.44 | gr0mit | You Are On Your Own ;-) |
| 09:40.59 | J4zen | i was hoping i wouldn't be, but so be it ;) |
| 10:00.16 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 10:00.19 | HaMYaI | anyone knows how to ignore SIGHUP signal in bash agi script |
| 10:03.30 | *** join/#asterisk marc7 (n=marc@S0106001c10b1e865.vc.shawcable.net) |
| 10:04.23 | marc7 | is there any way to encode a higher quality audio clip for use in asterisk to be played back as a prompt/greeting than by using the following command -- sox src.mp3 -r 8000 -c 1 -s -w dest.wav |
| 10:07.00 | *** join/#asterisk LuisTorres (n=luistorr@bl6-204-40.dsl.telepac.pt) |
| 10:07.18 | LuisTorres | Howdy |
| 10:10.30 | jblack | marc7: You want to use a different format? |
| 10:11.03 | jblack | You could add the mp3 app and use mp3s. |
| 10:11.08 | marc7 | jblack: i'm just trying to figure out if there's any way to bump up the quality. it takes a significant hit... but I understand there's no 128kbps+ audio quality on a 64kbps medium |
| 10:11.19 | marc7 | the MP3's sound moderately better, but they're very quiet |
| 10:11.46 | jblack | so, can you use sox to remix it for a higher volume? |
| 10:12.27 | *** join/#asterisk colulu (n=jgreen@58.251.75.157) |
| 10:13.43 | *** join/#asterisk xenonex (n=xenonex@82.200.211.194) |
| 10:17.04 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:17.42 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 10:20.58 | marc7 | jblack: default sox on my system can't write mp3 files... anyway, i'm not entirely certain I can tell if the quality is much better with MP3's than with these desampled WAV's |
| 10:23.44 | Strom_M | marc7: telephone network is only 8khz 8-bit logarithmically companded audio anyway |
| 10:24.11 | Strom_M | feed asterisk an 8khz 16-bit audio file and you're at the best quality you'll ever hear over the PSTN |
| 10:24.21 | Strom_M | also, don't confuse 64kbps PCM with 64kbps MP3 |
| 10:24.44 | Strom_M | each byte represents a COMPLETELY different encoding method |
| 10:25.06 | marc7 | yeah, I suppose it doesn't matter if I'm trying to have the clips sound as good as possible for the people who are using VoIP phones on our internal network... ultimately that won't reflect what the majority of people will be hearing |
| 10:25.28 | Strom_M | what codec are you using internally? |
| 10:25.29 | marc7 | i should just sox them to sln's and be done with it |
| 10:25.43 | Strom_M | wav == slin |
| 10:26.21 | marc7 | right, they're in mp3 format right now and play quietly, converting them to sln alone seems to raise the volume of the playback considerably |
| 10:26.34 | Strom_M | MP3 is a waste of time for telephones :) |
| 10:26.39 | Strom_M | what codec are your phones using internally? |
| 10:26.45 | marc7 | just the format we were provided the files in |
| 10:26.56 | marc7 | I believe gsm |
| 10:27.06 | Strom_M | ugh, that'll sound like total shit regardless |
| 10:27.23 | jblack | heh. |
| 10:27.23 | marc7 | either that or 711u |
| 10:27.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:27.31 | Strom_M | switch them over to ulaw (or alaw if you're outside of north america) and you'll hear a HUGE difference |
| 10:27.32 | marc7 | i'll have to check |
| 10:27.59 | Strom_M | but anyway, G711 is the codec that the PSTN uses |
| 10:28.03 | marc7 | yep |
| 10:28.24 | Strom_M | so you're not going to have "better quality" internally unless your PSTN link is using some shit codec like GSM |
| 10:28.28 | marc7 | 'g726, ulaw, gsm' is the order of the codecs we use in sip.conf |
| 10:28.30 | marc7 | *nod* |
| 10:28.44 | marc7 | thank you Strom |
| 10:28.52 | marc7 | i just needed to stop obsessing and just be done with it |
| 10:28.56 | Strom_M | :) |
| 10:29.02 | marc7 | it's 3AM, I shouldn't be nitpicking |
| 10:29.05 | Strom_M | in short -- what you're doing now is the best option |
| 10:30.07 | mvanbaak | or go wideband ;) |
| 10:30.23 | Strom_M | mvanbaak: an almost total waste since the PSTN isn't wideband at all yet |
| 10:30.31 | jblack | better would be boxes with talking parrots that communicate with semaphors.... well, perhaps not better, but at least more fun |
| 10:30.42 | Strom_M | voice over IP over carrier pigeon? |
| 10:30.51 | jblack | nah. parrots. |
| 10:30.53 | Strom_M | VoIPoCP |
| 10:31.08 | Strom_M | (just try to pronounce that one) |
| 10:31.13 | jblack | carrier pigeons are little more than "bum chicken". Parrots are cool, as some of them can talk |
| 10:31.40 | marc7 | oh man, that reminds me of packet over sheep |
| 10:31.49 | marc7 | http://www.cluebat.ca/rfc3203/ |
| 10:31.59 | Strom_M | "Sir, the order is in from Engineering." "Oh?" "Yes, Apparently we need 34 monitors and 47 Braaaawk servers." |
| 10:32.06 | jblack | icmp eh-eh-eh-eh-eh-eh-eh-cho ? |
| 10:32.49 | jblack | 34 pah-ah-ah-ah-kets sent 34 pah-ah-ah-ah-kets received. |
| 10:40.35 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 10:48.03 | defswork | I'm had an issue (twice now) where incoming calls to a ISDN30 site get engaged tones - the site can dial out ok etc.. If I restart asterisk incoming calls work again. Can't diagnose it now because priority was to get it working but anyone have any thooughs (using sangoma E1 btw) |
| 10:48.17 | defswork | thoughts* |
| 10:48.51 | Strom | defswork: you sure it's an engaged tone and not a congestion tone? |
| 10:49.01 | Strom | (I don't know how similar the two are outside of north america9 |
| 10:49.04 | Strom | s/9/)/ |
| 10:49.04 | defswork | Strom - fairly certain |
| 10:49.34 | defswork | strom - I thought more likely to get the pip peep tone it circuit was down |
| 10:49.56 | Strom | if and when it happens again, see if the telco switch is talking to your PBX when you attempt a call |
| 10:50.11 | Strom | if so, see how the PBX is responding (pri debug will help) |
| 10:50.14 | defswork | theres nothing in the asterisk logs at all on the incoming call (when engaged tone received) |
| 10:50.31 | Strom | yeah, you have to turn pri debug |
| 10:50.35 | defswork | which makes me think it's not got to the * box at all |
| 10:50.38 | Strom | s/turn/turn on/ |
| 10:51.20 | Strom | if pri debug shows conclusively that the telco isn't talking to you, call them up and see why |
| 10:51.27 | Strom | is asterisk restarting your B channels regularly? |
| 10:51.38 | defswork | aye - can't do anything now until it stops again |
| 10:51.59 | Strom | yes, which is why i'm giving you some checklist things i'd do :) |
| 10:52.07 | defswork | strom - I don't know - how do I check that ? |
| 10:52.28 | Strom | you'll see periodic messages on the CLI (usually once per hour) that asterisk is restarting idle B channels |
| 10:52.31 | defswork | this box has been in and fine for 8 months btw |
| 10:52.36 | Strom | hm |
| 10:52.43 | defswork | with no config changes |
| 10:52.56 | defswork | last week they had calls terminating also |
| 10:53.04 | defswork | I think telco or line issues :) |
| 10:53.07 | Strom | how long was asterisk running when this occurred? |
| 10:53.07 | defswork | (hope) |
| 10:53.22 | defswork | last week - probably months |
| 10:53.28 | Strom | hm |
| 10:53.40 | defswork | it's been rebooed a coupld of times since |
| 10:53.51 | defswork | yesterday when incoming failed |
| 10:53.58 | Strom | i really don't know; i'd have to see how exactly it behaves when broken to figure out where the problem is |
| 10:54.01 | defswork | (I couldn't sort it so told them to restart the box) |
| 10:54.13 | Strom | you could have just restarted asterisk, probably :) |
| 10:54.21 | defswork | yes - thats what I did today |
| 10:54.31 | Strom | it broke twice? |
| 10:54.34 | defswork | yesterday I was in a meeting so couldn't get on |
| 10:54.39 | defswork | yes |
| 10:54.45 | Strom | ok |
| 10:55.10 | Strom | next time, see if just physically unplugging and reconnecting the E1 fixes things |
| 10:55.41 | Strom | the telco's equipment may be going bonkers |
| 10:57.29 | defswork | will do |
| 10:57.32 | defswork | thanks ;) |
| 10:58.59 | defswork | it's curious that outgoing calls work |
| 10:59.00 | *** join/#asterisk RoyK (n=roy@ip-168-52-149-91.dialup.ice.no) |
| 10:59.04 | defswork | and nothing logged on incoming |
| 10:59.16 | defswork | makes me think the call doesn't touch the * box |
| 11:00.02 | defswork | should telco be issuing engaged tones though on an E1 ? |
| 11:00.45 | defswork | I thought that was down to the terminating equipment |
| 11:01.53 | Strom | the equipment just sends an engaged message |
| 11:02.01 | Strom | it's up to the originating switch to actually send the tone |
| 11:02.19 | defswork | Line Code Violation: 270 |
| 11:02.19 | defswork | Far End Block Errors: 112 |
| 11:02.26 | defswork | are those anything to worry about ? |
| 11:02.32 | Strom | perhaps |
| 11:04.23 | defswork | heh - I just look at same info on another site - that has 26777 line code violations and 448 far end block errors :) |
| 11:06.45 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:07.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:10.14 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:13.43 | *** join/#asterisk juanjoc (n=juanjoc@host223.190-225-200.telecom.net.ar) |
| 11:13.57 | *** join/#asterisk aethelrick (n=richard@212.248.154.2) |
| 11:16.18 | aethelrick | can someone give me a little help loading zaptel and ztdummy... I'm running openSuSE 10.3 I have the latest version of Asterisk (1.4.21) and Zaptel from the SuSE asterisk community repository. When I run modprobe zaptel I get no feedback and the module does not load |
| 11:17.21 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 11:17.36 | Strom | you should receive no feedback |
| 11:17.50 | Strom | are you sure the module doesn't load? does it not show up when you run lsmod? |
| 11:18.06 | aethelrick | I run lsmod | grep zaptel and it is not listed |
| 11:18.19 | Strom | how about just lsmod | grep zap |
| 11:18.31 | aethelrick | still nothing |
| 11:18.41 | Strom | is it the zaptel built for your specific kernel? |
| 11:18.47 | aethelrick | I've also run lsmod and eyeballed the list :) |
| 11:19.07 | *** join/#asterisk kami- (n=user@p4FD3AC1B.dip0.t-ipconnect.de) |
| 11:19.11 | Strom | what happens if you modprobe ztdummy and then do lsmod | grep zap |
| 11:19.14 | aethelrick | I'm running a stock kernel from he suse repository and a stock zaptel from the suse repository |
| 11:19.27 | aethelrick | I'm assuming (possible incorrectly I suppose) that they match |
| 11:19.31 | kami- | hello |
| 11:20.04 | aethelrick | ztdummy does not load either |
| 11:20.24 | Strom | maybe someone fucked up with the packages |
| 11:20.30 | aethelrick | could be |
| 11:20.35 | Strom | download the source, build against your actual kernel, and try again |
| 11:20.45 | aethelrick | okey dokey, I'll give that a go |
| 11:20.48 | aethelrick | thanks |
| 11:20.58 | Strom | welcome |
| 11:23.20 | mvanbaak | gheh |
| 11:23.33 | mvanbaak | wish all problems could be solved that easy |
| 11:23.42 | mvanbaak | 'download the source, compile, and be done with it' |
| 11:23.59 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 11:24.00 | Strom | mvanbaak: it's not a solution; it's a troubleshooting step |
| 11:24.28 | kami- | I'm trying to connect an E1 line to a fonebridge2 and to asterisk |
| 11:24.49 | kami- | the span is up and active |
| 11:25.00 | kami- | but asterisk doesn't answer the call |
| 11:25.18 | kami- | Ext: 1 Cause: Unallocated (unassigned) number (1) |
| 11:25.36 | Strom | how does the fonebridge connect to asterisk |
| 11:25.47 | kami- | ztd-ethmf |
| 11:26.02 | Strom | TDMoE? |
| 11:26.05 | kami- | yes |
| 11:26.10 | aethelrick | hi, Strom... any configure options you reccommend for zaptel with just the ztdummy driver (I only want it for the meetme() app) |
| 11:26.35 | Strom | kami-: pastebin your zapata.conf and your extensions.conf |
| 11:26.39 | Strom | ~pb |
| 11:26.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 11:26.49 | kami- | Strom: will do |
| 11:26.56 | Strom | aethelrick: run make menuselect and turn off all the modules you don't need :) |
| 11:28.57 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 11:30.53 | Vec | Hi, I found this perl script that deletes and renames old msgs, http://www.voip-info.org/wiki/view/Script+to+Age%252FDelete+old+Voicemails%7Cvmspool_manager, can't get it anywhere ? |
| 11:31.24 | Vec | I would write a script to just delete files in /var/spool/asterisk older then a specified date, but AFAIK I need to rename them as well ? |
| 11:31.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:32.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:33.26 | Strom | Vec: that would be relatively simple for even a moderately competent coder to throw together |
| 11:34.29 | Vec | Strom : yeh I guess I'm below moderately competent. |
| 11:35.00 | flenders | anyone has any bad things to say about the TDM410? I was gonna go for the A200 with HW echo, but now the TDM410 with HW EC is cheaper... any reason why I should still go with sangoma? |
| 11:35.33 | Strom | flenders: ive had a TDM800 (essentially the same card) for a year now with no problems |
| 11:36.42 | flenders | Strom: I had a couple of TDM400Ps running for a couple of years too, and the only problem with them was getting rid of echo |
| 11:37.02 | flenders | but now with the TDM410P, it would be a lot easier :o) |
| 11:37.12 | Strom | yeah -- it's a great card |
| 11:37.18 | *** join/#asterisk zepmantra (n=dea@124.107.177.41) |
| 11:37.39 | Strom | plus, unlike the sangoma, you don't have to have those funky RJ9 connectors on your cable :) |
| 11:37.51 | flenders | yeah, true |
| 11:38.01 | flenders | I might go with the TDM410 then |
| 11:38.16 | flenders | it's $200 cheaper |
| 11:38.19 | flenders | :D |
| 11:38.59 | Strom | go for it |
| 11:39.05 | zepmantra | hello there we will be setting up 2 sites which are connected via a leased line, 25 users per site, 8FXO per site ... is there still a need for the g729A codec? btw we will be using linksys papt2-na |
| 11:39.35 | Strom | zepmantra: that depends entirely on the bandwidth available on the leased line |
| 11:39.47 | Strom | and the terminal equipment has NOTHING to do with it |
| 11:40.01 | kami- | Strom: I pasted my zapata.conf and extensions.conf at http://paste.lisp.org/display/62891 and the log is available at http://www.m-creations.com/downloads/log-after-connect.txt.gz |
| 11:41.02 | zepmantra | ic the client says they have 1mbps connection |
| 11:41.50 | Strom | kami-: way way way too much data in the log -- just pastebin the CLI output from the failed call at verbose 10 with debug turned off |
| 11:42.07 | Strom | zepmantra: is that solely for voice? |
| 11:42.20 | zepmantra | yes ... |
| 11:42.46 | Strom | what percentage of the users do you anticipate calling over that leased line simultaneously? |
| 11:42.53 | kami- | Strom: sorry. But don't have access to the machine right now. I will cut the middle part and pastebin it. |
| 11:43.38 | Strom | kami-: how the hell do you expect to fix the problem if you have no access to it? |
| 11:44.26 | zepmantra | Storm around 20, 8 of those calls would use the fxo of the other site to avoid long distance charge |
| 11:44.49 | Strom | zepmantra: my name isn't storm :) |
| 11:45.16 | zepmantra | sorry for the typo .. |
| 11:45.26 | Strom | tab-complete is your friend |
| 11:45.26 | kaldemar | talubator completion. |
| 11:47.00 | Strom | zepmantra: you could probably get away with using G.726 on that circuit (and it would sound a hell of a lot better than G.729) |
| 11:47.06 | kami- | Strom: I just wanted to ask someone whether it _seems_ fixable. I can send someone there (takes 1,5 hours). |
| 11:47.28 | Strom | kami-: it's an extremely fixable proble |
| 11:47.36 | Strom | why don't you have ssh access? |
| 11:47.39 | kami- | Strom: :-) |
| 11:48.03 | kami- | Strom: I have ssh. But the line is currently connected to a proprietary pbx. |
| 11:48.29 | Strom | well, that's different than not having access to the machine |
| 11:48.32 | kami- | I would have to send someone to connect the line (intermittently) |
| 11:48.53 | kami- | sorry for the wrong wording. I'm not a native speaker. |
| 11:48.53 | Strom | anyway, the error is extremely simple |
| 11:49.18 | zepmantra | Strom, ic thanks for the very great recommendation .... :) |
| 11:49.28 | kami- | A problem with the dialplan? |
| 11:50.19 | Strom | the PRI context is set to "default" and you don't seem to have any sort of method for handling inbound calls at all in your dialplan |
| 11:50.47 | kami- | What about the 's' extension? I thought that would answer any call. |
| 11:50.50 | Strom | no |
| 11:50.52 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 11:50.54 | Strom | that's not what s is for |
| 11:51.02 | Strom | s is only for calls that come in with NO dialed number information at all |
| 11:51.08 | Strom | i.e. an analog line ringing |
| 11:51.19 | Strom | in all other cases, you must pattern match |
| 11:51.35 | jack_sparo | hi, in freepbx the follow me are in which find if i want to look for it from the cli command guys? |
| 11:51.35 | kami- | OK, I'll re-read the dial plan chapter in the book. |
| 11:51.53 | Strom | jack_sparo: this is not #freepbx -- please go ask there |
| 11:52.06 | jack_sparo | sorry :) |
| 11:52.49 | kami- | Strom: now I understand why the same dial plan worked with a TDM2xx card. The analog line was picked up through 's'. Right? |
| 11:53.11 | Strom | yes |
| 11:53.25 | Strom | there's no such thing as TDM2xx though :) |
| 11:53.34 | kami- | Sorry TDM4xx |
| 11:56.18 | kami- | Strom: I think we also called <number of the pri line>600. Shouldn't that answer the call? |
| 11:56.38 | kami- | No, it doesn't have Answer in it! |
| 11:58.06 | Strom | kami-: you need to figure out what the telco is actually sending your PBX for DNIS |
| 11:58.15 | Strom | arbitrary guessing isn't going to help |
| 12:03.57 | kami- | Strom: the DNIS is the number I have to dial to connect to the E1 line? |
| 12:04.13 | Strom | DNIS is Dialed Number Identification Service |
| 12:04.23 | Strom | i.e. the number the telco sends YOU when someone calls you |
| 12:04.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 12:05.23 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:05.45 | kami- | The pri debug log says: Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '82144600' |
| 12:06.39 | kami- | 82144 is the 'base number' and 600 was the extension we were trying to reach (to hear the demo). |
| 12:06.50 | Strom | ok, so you need to pattern match against that |
| 12:07.20 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:07.46 | kami- | Strom: so, I will put an entry for 82144 into extensions.conf. |
| 12:07.58 | Strom | just "82144" isnt going to match |
| 12:09.33 | kami- | Strom: would 82144600 do? |
| 12:10.00 | eXistenZ | guten tach |
| 12:10.06 | eXistenZ | allerseits :] |
| 12:10.20 | Strom | kami-: yes. it helps if you think logically about this stuff :) |
| 12:10.30 | *** join/#asterisk andresmujica (n=andresmu@190.25.192.24) |
| 12:10.41 | kami- | Strom: thank you very much for helping such a moron as I am! |
| 12:11.31 | *** join/#asterisk berspolis (n=bartoloc@190.25.228.235) |
| 12:11.45 | berspolis | hello |
| 12:11.45 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 12:11.48 | *** join/#asterisk chendy (n=chatzill@58.60.115.219) |
| 12:11.50 | Mike8861 | hello all! |
| 12:11.58 | andresmujica | hello all! |
| 12:12.00 | kami- | Strom: I promise to re-read the book. ;-) |
| 12:12.15 | berspolis | i have a question |
| 12:12.23 | kami- | bye |
| 12:12.26 | *** part/#asterisk kami- (n=user@p4FD3AC1B.dip0.t-ipconnect.de) |
| 12:12.31 | andresmujica | hey i`m subscribed to asterisk bootcamp and dcap cert. Any recomendations, study guides for success? |
| 12:13.03 | Strom | berspolis: ask away |
| 12:13.05 | berspolis | i have a channel that is member of 3 queues |
| 12:13.15 | Strom | berspolis: ask away but preferably on one line :) |
| 12:13.27 | berspolis | ok i'm sorry |
| 12:13.37 | Strom | andresmujica: study everything. practice setting up a PBX within 90 minutes |
| 12:13.53 | Strom | know Asterisk inside and out, know telephony theory |
| 12:14.02 | Strom | know your protocols |
| 12:14.24 | andresmujica | hmm 90 minutes... and pure scratch config? |
| 12:14.30 | Strom | yup |
| 12:14.43 | Strom | unless they've changed it since I last administered the test |
| 12:14.57 | Strom | but I suspect the format is still the same |
| 12:15.01 | andresmujica | i`m planning to follow FOT with a test install |
| 12:15.02 | aethelrick | Hi Strom... no change on the zaptel front, I compiled the zaptel module and installed it and still does not load |
| 12:15.03 | andresmujica | nice. |
| 12:15.55 | Strom | aethelrick: hmm, do any errors appear in syslog? |
| 12:15.56 | berspolis | i have 4 channels that are members of 3 queues, but sometimes, in the last defined queue the call doesn´t ring all four members, even if they are available, what can be wrong? |
| 12:16.03 | andresmujica | the actual FOT version is v2 right? |
| 12:16.17 | Strom | andresmujica: the test doesnt follow the book per se |
| 12:16.30 | Strom | andresmujica: but the book is a good reference to read fully anyway |
| 12:16.50 | Strom | andresmujica: really, the only thing which will truly prepare you for the test is plenty of experience |
| 12:17.23 | Strom | but I've seen people take the course and pass the test even without having ever touched asterisk |
| 12:17.41 | andresmujica | ohh.. |
| 12:18.04 | [TK]D-Fender | berspolis: pastebin your queue config, and an agent dump at CLI. |
| 12:18.06 | [TK]D-Fender | ~pb |
| 12:18.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:18.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 12:18.20 | Strom | of course, i may be off; again, it's been a while since I administered the dCAP test |
| 12:18.29 | aethelrick | Strom - sadly nothing hits the logs... it's all a bit silent |
| 12:18.51 | andresmujica | ok. no prob. anyway i'll try to do my best. i've got about a month for the course and test itself |
| 12:18.54 | andresmujica | thanks!!! |
| 12:18.54 | Strom | aethelrick: well, I'm lost then. |
| 12:19.00 | berspolis | ok |
| 12:19.01 | aethelrick | you and me both |
| 12:21.28 | Strom | wasn't that a Yazoo album? |
| 12:22.06 | plik | it was |
| 12:22.11 | *** part/#asterisk andresmujica (n=andresmu@190.25.192.24) |
| 12:22.31 | berspolis | http://pastebin.com/d4320e01a |
| 12:22.31 | plik | recently remastered as part of a box set |
| 12:22.48 | berspolis | ok, i have pastebin my queue.conf |
| 12:23.05 | Strom | yeah -- I have a copy of Upstairs at Eric's, and I have one or two singles from You and Me Both, but no copy of You and Me Both per se :) |
| 12:23.44 | aethelrick | Hi Strom, I just ran the init script and got an error... |
| 12:23.52 | aethelrick | line 0: Unable to open master device '/dev/zap/ctl' |
| 12:24.01 | Strom | aethelrick: are you root? :) |
| 12:24.02 | aethelrick | I'm assuming this is because the module is not loaded? |
| 12:24.06 | aethelrick | yes |
| 12:24.31 | Strom | aethelrick: if the module isn't loading, a script isn't going to help |
| 12:24.48 | [TK]D-Fender | berspolis: Now CLI output of a call attempt where they are not called as expected, along with the aget AND channel dumps |
| 12:25.26 | [TK]D-Fender | aethelrick: modprobe your cards driver and rerun ztcfg |
| 12:25.57 | aethelrick | I'm only after zaptel for the ztdummy timer... I don't have any actual hardware to drive with it |
| 12:26.00 | *** join/#asterisk GreeneMile (n=alan@87.192.131.123) |
| 12:26.08 | Strom | [TK]D-Fender: he is modprobing it; it's not showing up when he runs lsmod |
| 12:26.21 | [TK]D-Fender | aethelrick: ok, modprobe them. Also what OS & Kernel are you using? |
| 12:26.25 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:26.28 | *** part/#asterisk GreeneMile (n=alan@87.192.131.123) |
| 12:26.52 | aethelrick | openSuSE 10.3 kernel 2.6.22.17-0.1-default |
| 12:27.20 | aethelrick | I have tried the stock zaptel rpm from the SuSE repo and I've built a copy from source |
| 12:28.14 | aethelrick | i type "modrobe zaptel" and/or "modprobe ztdummy" from the console logged in as root and lsmod shows the modules are not loaded. I get no errors, just silent failure. I am of course logged in as root |
| 12:28.23 | aethelrick | *modprobe |
| 12:28.50 | [TK]D-Fender | aethelrick: If you've changed your kernel you'll need to recompile zaptel. |
| 12:29.04 | [TK]D-Fender | aethelrick: I'd suggest redoing that process in its entirety now. |
| 12:29.26 | aethelrick | I recompiled zaptel about five minutes ago :( |
| 12:29.43 | aethelrick | the kernel has been the same one for a while now |
| 12:30.00 | Strom | [TK]D-Fender: I walked him through everything I could think of |
| 12:30.27 | Strom | [disclaimer: I've been up way too long and was drunk not too long ago) |
| 12:30.45 | [TK]D-Fender | aethelrick: I've heard all sorts of trouble regarding Zaptel & SUSE.... |
| 12:30.49 | aethelrick | is there any way I can increase the logging level on modprobe a simple "-v" just returns install /bin/true (or similar, I'm working from memory on that one) and there must be something erroring, I just can't see what |
| 12:31.28 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 12:32.02 | [TK]D-Fender | aethelrick: http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+SuSE |
| 12:32.08 | [TK]D-Fender | aethelrick: See if it does you any good. |
| 12:32.14 | aethelrick | thanks... I'll have a read |
| 12:33.15 | eXistenZ | [TK]D-Fender, What might be the problem if the CID cannot be resolved? |
| 12:34.12 | [TK]D-Fender | eXistenZ: maybe there isn't any. |
| 12:34.29 | eXistenZ | [TK]D-Fender, how come? |
| 12:34.59 | [TK]D-Fender | eXistenZ: blocked, or coming from telco not supporting, or YOUR line doesn't support it... yaddayadda.. |
| 12:35.16 | eXistenZ | [TK]D-Fender, well, I checked it with another phone, which works |
| 12:35.24 | eXistenZ | dect |
| 12:35.27 | [TK]D-Fender | eXistenZ: please provide BACKUP VIA PASTEBIN instead of these unvalidated questions. |
| 12:35.54 | [TK]D-Fender | eXistenZ: Maybe you're screwing something up, but since we can't see, we'll never know |
| 12:42.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:42.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:44.52 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:47.22 | *** join/#asterisk smach (n=smach@bas15-ottawa23-1279406959.dsl.bell.ca) |
| 12:50.09 | *** part/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 12:58.52 | *** join/#asterisk ManxPower (n=manxpowe@28.sub-75-251-129.myvzw.com) |
| 12:59.41 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 13:01.57 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 13:03.35 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 13:07.07 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 13:07.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:07.57 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-20d62fb69e697510) |
| 13:10.54 | ManxPower | Nothing quite like waking up to a phone call telling you "the internet is down" |
| 13:13.18 | plik | I had one of those on Tuesday :/ |
| 13:13.47 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.41 | [TK]D-Fender | "What are you doing Dave?" |
| 13:15.38 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-21-203.revip2.asianet.co.th) |
| 13:23.11 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:24.02 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 13:24.26 | ManxPower | once I figured out what was wrong, I switched to their backup internet connection |
| 13:29.23 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 13:36.38 | J4zen | How do i send out a dial command via the manager interface? |
| 13:36.43 | J4zen | I can get it to register and logon as a manager |
| 13:36.53 | J4zen | but it should be calling, ill pastebin the datastream if needed |
| 13:37.01 | *** join/#asterisk hsv-al (n=hsv-al@user-24-214-126-81.knology.net) |
| 13:37.03 | ManxPower | J4zen: have you read manager.txt? |
| 13:37.10 | hsv-al | is there a way to limit how often i see this, it occurs in console to frequently |
| 13:37.14 | hsv-al | [Jun 27 07:34:20] NOTICE[7706]: chan_sip.c:15092 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 |
| 13:37.51 | ManxPower | hsv-al: turn off subscribe on the phone or add a mailbox= line for the peer [1000] |
| 13:37.52 | *** join/#asterisk smach (n=smach@bas15-ottawa23-1279406777.dsl.bell.ca) |
| 13:38.09 | J4zen | lol is this normal : Message: Thanks for all the fish. |
| 13:38.16 | J4zen | when logging of as a manager |
| 13:38.26 | *** part/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 13:38.52 | [TK]D-Fender | J4zen: Care to show us the actual problem? |
| 13:39.25 | oej | hsv-al: That should not be a NOTICE... Hmmm. |
| 13:39.50 | J4zen | I'm testing a small application that will log on as a manager and perform a call between two extensions. I ported a php Click-To-Call script for this. |
| 13:39.57 | ManxPower | [TK]D-Fender: he still has not answered my original question. |
| 13:40.18 | [TK]D-Fender | ManxPower: True, but his solution doesn't hinge on it. |
| 13:40.26 | J4zen | When sniffing the packets my application sends to the PBX it shows that the logon was accepted, however no call is performed |
| 13:40.29 | J4zen | one moment for the pastebin |
| 13:40.35 | ManxPower | [TK]D-Fender: it would give an idea of if he's read the docs or not. |
| 13:40.55 | ManxPower | (8:37:02 AM) ManxPower: J4zen: have you read manager.txt? |
| 13:40.58 | [TK]D-Fender | ManxPower: That doc anyways... |
| 13:41.09 | ManxPower | [TK]D-Fender: it's the best place to start |
| 13:41.25 | [TK]D-Fender | ManxPower: Take your word for it.. |
| 13:41.40 | eXistenZ | [TK]D-Fender, here is my configuration as mht file: http://existenz.googlepages.com/spa.rar |
| 13:41.42 | ManxPower | [TK]D-Fender: anyway, have fun with this one. |
| 13:41.59 | [TK]D-Fender | eXistenZ: No. |
| 13:42.04 | ManxPower | and that one that pasts a .rar file. |
| 13:42.06 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 13:42.51 | J4zen | [TK]D-Fender: http://pastebin.com/md02bf7f , Thats the servers respons packets and a SIP debug when my application tries to call |
| 13:42.55 | J4zen | response* |
| 13:42.57 | [TK]D-Fender | eXistenZ: and WTF is this supposed to tell me? |
| 13:43.24 | eXistenZ | [TK]D-Fender, You asked me for the configuration |
| 13:43.26 | [TK]D-Fender | eXistenZ: Last Called Number: fallback@gw0 / Last Called VoIP Number: 127.0.0.1:5060 |
| 13:43.29 | J4zen | Sorry ManxPower: i didn't see your message |
| 13:43.58 | [TK]D-Fender | eXistenZ: Your ata is talking to itself, and I want to see the call coming in <--- |
| 13:44.06 | *** join/#asterisk JoAkKiNeN (n=jalvarez@cm13109.red91-117.mundo-r.com) |
| 13:44.08 | ManxPower | J4zen: that's ok. Your opportunity to receive my help has passed anyway. |
| 13:44.13 | J4zen | ManxPower: i havn't read the mhat specific file, i have read the click-to-call application code to see how its done. |
| 13:44.25 | ManxPower | J4zen: that file is the official manager docs |
| 13:44.32 | eXistenZ | [TK]D-Fender, not last called number |
| 13:44.39 | [TK]D-Fender | eXistenZ: AND that doesn't show me you configured it to take CID and pass it on either. |
| 13:44.45 | J4zen | ManxPower: I'll look into it, thank you thusfar. |
| 13:44.56 | eXistenZ | [TK]D-Fender, "Last PSTN Caller:" |
| 13:45.01 | [TK]D-Fender | eXistenZ: Seeing what you dialed there makes me wonder if you had any clue when setting this up. |
| 13:45.44 | eXistenZ | [TK]D-Fender, It is supposed to take CID automatically |
| 13:46.02 | [TK]D-Fender | eXistenZ: O RLY? I beg to differ. |
| 13:46.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:46.19 | [TK]D-Fender | eXistenZ: And I've owned an SPA-3000 & SPA-3102 |
| 13:46.30 | eXistenZ | [TK]D-Fender, I haven't used the spa yet to dial |
| 13:46.51 | eXistenZ | [TK]D-Fender, I am talking now only about the pstn. |
| 13:46.54 | eXistenZ | just displaying the caller id |
| 13:47.20 | [TK]D-Fender | eXistenZ: that was BOTH ports is pasted there. Either way you have made blanket stements I know are incorrect. |
| 13:48.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:49.25 | *** join/#asterisk sack (n=sack@42.Red-83-44-211.dynamicIP.rima-tde.net) |
| 13:50.32 | [TK]D-Fender | eXistenZ: \o/ |
| 13:50.45 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 13:50.58 | eXistenZ | [TK]D-Fender, Sorry, I was disconnected. |
| 13:54.06 | *** join/#asterisk orionr (n=arosen@c-76-26-221-76.hsd1.sc.comcast.net) |
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| 13:59.30 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
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| 14:02.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.30 | FinboySlick | Hmmmm... Running out of iax2 threads in 1.4.21. I'm already rolling back to see if that fixes things but all the bug reports I've found were for 1.4.11 or so. |
| 14:13.53 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 14:18.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:20.25 | ^shark_ | hi friends,i installed c-client and * with no problem but now i am seeing this error >> http://pastebin.com/ma3bc8c5. |
| 14:21.27 | FinboySlick | ^shark_: It doesn't seem to like your self-signed imap certificate. |
| 14:21.57 | [TK]D-Fender | ^shark_: init_mailstream: Can't connect to imap server {localhost:143/imap/authuser=vmail/user=vmail}Voicemail <- go prove imap is running. |
| 14:22.36 | ^shark_ | [TK]D-Fender: i have telneted to the imap server & imap is running. |
| 14:22.36 | Nugget | telnet is eeeeeeevil! |
| 14:22.51 | FinboySlick | [TK]D-Fender: If it can see the ssl certificate, there must be an imap server running ;) |
| 14:23.02 | ^shark_ | is thinking how to change the imap certificate |
| 14:23.46 | FinboySlick | ^shark_: Since you're connecting to localhost, you could also just not use ssl for that specific connection. |
| 14:24.20 | ^shark_ | FinboySlick: how do i change my settings, i am abit of a newbie |
| 14:24.52 | ^shark_ | FinboySlick: i like the idea of using ssl, since this is a VM and i am just doing practice. |
| 14:25.26 | FinboySlick | ^shark_: Then you'll have to convince the voicemail app to accept self-signed certs. I wouldn't know, I'm a bit of a newbie too. |
| 14:26.24 | *** join/#asterisk zepmantra (n=hehe@210.23.188.100) |
| 14:28.22 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 14:28.26 | FinboySlick | ^shark_: http://noizyland.net/node/8 might be a good starting point. |
| 14:28.46 | FinboySlick | ^shark_: Especially the last line. |
| 14:29.01 | ^shark_ | k |
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| 14:35.10 | *** join/#asterisk KorcaN (n=KOrcaN@65.167.121.90) |
| 14:37.24 | KorcaN | hey all.. |
| 14:38.17 | KorcaN | I am thinking about installing asterisk for my company, I have written an app in PHP and want to be able to do screen pop based on the incoming call and dial from the browser... |
| 14:38.20 | KorcaN | any ideas? |
| 14:38.47 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-6d8197a4d982f1cf) |
| 14:39.08 | Inez | easy breezy |
| 14:39.30 | *** join/#asterisk dijungal (n=kdaniel@205.244.151.188) |
| 14:39.37 | KorcaN | Inez, you got some tutorials? |
| 14:39.54 | [TK]D-Fender | KorcaN: No such thing. |
| 14:39.57 | dijungal | hi guys... how dto i set the accountcode variable in the CDR? |
| 14:40.21 | KorcaN | [TK]D-Fender, what do you mean? can this not be done? |
| 14:40.28 | KorcaN | how do call centers do it? |
| 14:40.28 | [TK]D-Fender | KorcaN: As you call out or answer you can trigger a script to do whatever you'd like it to do. |
| 14:40.49 | [TK]D-Fender | KorcaN: No, it can certainly be done. I'm saying there's no such "tutorial". |
| 14:40.52 | KorcaN | [TK]D-Fender, script where? on the asterisk box? |
| 14:41.01 | KorcaN | ahh.. |
| 14:41.45 | [TK]D-Fender | KorcaN: Go set up * and use a soft-phone to test with. |
| 14:41.53 | KorcaN | first things first.. screen pop customers info to CSR browser... how would you tackle that? |
| 14:42.16 | [TK]D-Fender | KorcaN: how would YOU do it? |
| 14:42.51 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 14:42.59 | KorcaN | when csr picks up phone run script on * box to somehow pop the record... |
| 14:43.10 | *** join/#asterisk mgdm (n=michael@serenity.mgdm.net) |
| 14:43.19 | [TK]D-Fender | KorcaN: keep thinking on the "somehow". |
| 14:43.30 | FinboySlick | I keep seeing this: http://pastebin.com/m71c5575 despite there being no new iax calls... I'm guessing it's some sort of keepalive but wouldn't the fact that the call number changes every time be a probable reasons for iax running out of threads? |
| 14:43.57 | KorcaN | I don't know the somehow pop the record part... can you point me in the right direction? |
| 14:44.19 | [TK]D-Fender | KorcaN: If you can't imagine this then you'd better hire a programmer. |
| 14:44.21 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 14:44.33 | KorcaN | maybe throw it in a DB and have the csr click a button in mthe browser to pull the record... |
| 14:44.45 | [TK]D-Fender | KorcaN: that could work. |
| 14:45.05 | *** part/#asterisk ^shark_ (n=^shark_@41.222.2.65) |
| 14:45.43 | *** join/#asterisk nuonguy (n=john@c-24-6-187-202.hsd1.ca.comcast.net) |
| 14:45.44 | RoyK | two ducks in a cup http://img.dailymail.co.uk/i/pix/2007/07_03/mallardMS2807_468x397.jpg |
| 14:45.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
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| 14:46.17 | KorcaN | [TK]D-Fender, how would you do it? |
| 14:46.31 | naggers | how did you guys like that microsoft pbx software vid i think that was the most disgusting propaganda ive ever seen |
| 14:46.45 | naggers | they even had those twins from vh1 80s series |
| 14:47.40 | [TK]D-Fender | KorcaN: have a client run on the agents PC and send a TCP message of what to pop to it, and have that launch a browser targeting a lookup URL |
| 14:48.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:49.12 | KorcaN | [TK]D-Fender, nice :) is there an example of that anywhere? |
| 14:49.14 | codefreeze-lap | dijungal: try "core show function CDR" |
| 14:49.24 | Katty | bleh |
| 14:49.26 | [TK]D-Fender | KorcaN: this is not a "programming" channel. |
| 14:49.37 | Katty | i'll program you in a minute. |
| 14:49.42 | [TK]D-Fender | KorcaN: JFGI |
| 14:49.51 | Katty | someone needs to hire me away from my stupid company. |
| 14:50.05 | KorcaN | Katty, where you live? |
| 14:50.11 | Katty | in hell. |
| 14:50.15 | KorcaN | heh |
| 14:50.16 | Katty | also misery. |
| 14:50.30 | Katty | [TK]D-Fender: you're not going to believe what they just did. |
| 14:50.31 | [TK]D-Fender | Katty: c'mon... its not Milwaukee ;) |
| 14:50.37 | Katty | [TK]D-Fender: one of the sales reps is supposed to get 53 cents a mile. |
| 14:50.47 | dijungal | codefreeze-lap: got it |
| 14:50.47 | Katty | [TK]D-Fender: but, they didn't tell him yet, so he was still getting 47. |
| 14:50.47 | dijungal | thanks |
| 14:51.00 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 14:51.11 | codefreeze-lap | dijungal: no prob |
| 14:51.13 | Katty | [TK]D-Fender: but then they didn't want to actually pay him 47 cents a mile, so management skewed the numbers so he was getting paid 20 cents a mile. |
| 14:51.29 | [TK]D-Fender | Katty: fuckers. |
| 14:51.32 | Katty | [TK]D-Fender: i know. |
| 14:51.38 | Katty | [TK]D-Fender: it's so stupid here. nothing is by the book. |
| 14:51.46 | eXistenZ | :o |
| 14:52.05 | [TK]D-Fender | Katty: Sure it is... they've just made it a "(we) Choose your own adventure" |
| 14:52.38 | Katty | i'm just tired of all of it. |
| 14:52.49 | *** join/#asterisk iobug (n=gustav@212.124.71.46) |
| 14:52.51 | Katty | it's a lot of stress. |
| 14:52.53 | iobug | Aloha! |
| 14:52.57 | *** join/#asterisk spokra (n=spokra@host093-179-142.sea0.speakeasy.net) |
| 14:53.01 | *** join/#asterisk kami- (n=user@p4FD3AC1B.dip0.t-ipconnect.de) |
| 14:53.24 | kami- | hello |
| 14:53.33 | iobug | how to set in my asterisk sip account from another provider and use in my local PBX?! |
| 14:53.59 | iobug | can anybody help me with link |
| 14:54.10 | [TK]D-Fender | iobug: .... |
| 14:54.14 | [TK]D-Fender | ~book |
| 14:54.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:54.16 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 14:54.31 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 14:54.39 | iobug | ok, thank you |
| 14:55.21 | kami- | Strom: still around? |
| 14:55.36 | FinboySlick | iobug: I'm pretty sure it's what got me started too. |
| 14:57.07 | *** join/#asterisk shtoom (n=shtoom@59.93.124.6) |
| 14:58.30 | shtoom | Hi, when I stream audio file using AGI, is there a way to interrupt playback from agi program it self? not by user interruption by pressing digits. |
| 14:59.06 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:59.27 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 14:59.51 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 14:59.54 | ThoMe | hm |
| 14:59.54 | ThoMe | exten => _XXXXX.,n,GotoIf(${CALLERID(num)}" = "8"]?setze-thomas-absender:setze-default-absender) |
| 15:00.00 | ThoMe | he jump only to "setze-default-absender" |
| 15:00.16 | ThoMe | i would like if 8 then goto setze-thomas-absender |
| 15:00.30 | RoyK | ThoMe: perhaps GotoIf($[${CALLERID(num)}" = "8"]?setze-thomas-absender:setze-default-absender) |
| 15:00.57 | RoyK | eh |
| 15:01.08 | RoyK | GotoIf($["${CALLERID(num)}" = "8"]?setze-thomas-absender:setze-default-absender) |
| 15:01.18 | *** join/#asterisk mrmeek (n=mike@207-5.97-97.tampabay.res.rr.com) |
| 15:01.21 | mrmeek | wavs |
| 15:01.24 | mrmeek | waves* too |
| 15:01.35 | RoyK | .wavs |
| 15:01.44 | mrmeek | .gsms |
| 15:01.53 | mrmeek | (-; |
| 15:02.56 | *** join/#asterisk rickbradley (n=rickbrad@www.eastcore.com) |
| 15:04.12 | rickbradley | I'm on ubuntu 8.04, asterisk 1.4.9 from source, everything works fine, except, if I tell voicemail.conf to do volgain (and I do have sox installed), I see error messages like these when I record a voicemail: |
| 15:04.13 | rickbradley | Failed to open log file: /var/spool/asterisk/voicemail/default/1002/tmp/8gH04V.WAV: No such file or directory |
| 15:04.22 | rickbradley | without volgain voicemail works fine |
| 15:04.34 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 15:04.35 | [TK]D-Fender | rickbradley: upgrade |
| 15:04.46 | rickbradley | [TK]D-Fender: upgrade asterisk? |
| 15:04.55 | [TK]D-Fender | rickbradley: yes |
| 15:05.00 | rickbradley | okey dokey -- thanks |
| 15:06.47 | FinboySlick | People probably ask this once every couple days, but is there a way to have voicemail store as mp3. I know you can run a script to convert the files, but that means you have another set of files to manage separately. |
| 15:07.30 | Maliuta | FinboySlick: read the book |
| 15:08.11 | mrmeek | FinboySlick: Agree w/ Maliuta |
| 15:08.13 | ThoMe | RoyK: mh. |
| 15:08.15 | rickbradley | realizes what he did now -- scrolled down through the 1.4.* series downloads and picked the last one (1.4.9) which was not the most *recent*, just sorted last |
| 15:08.42 | ThoMe | RoyK: |
| 15:08.42 | ThoMe | exten => _XXXXX.,1,NoOp(${CALLERID(num)}) |
| 15:08.43 | ThoMe | exten => _XXXXX.,n,GotoIf($["${CALLERID(num)}" = "8"]?setze-thomas-absender:setze-default-absender) |
| 15:09.03 | russellb | rickbradley: that's why we have only the latest release in the main asterisk directory on the downloads site :) |
| 15:09.09 | russellb | we even have a asterisk-1.4-current link :) |
| 15:09.15 | ThoMe | RoyK: and if i dial: |
| 15:09.15 | ThoMe | <PROTECTED> |
| 15:09.18 | ThoMe | <PROTECTED> |
| 15:09.20 | rickbradley | russellb: now I know :-) |
| 15:09.26 | ThoMe | its "8" but jump to setze-default-absender" |
| 15:09.36 | russellb | ~pb |
| 15:09.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:09.51 | RoyK | try NoOp(GotoIf($["${CALLERID(num)}" = "8"]?setze-thomas-absender:setze-default-absender)) |
| 15:09.52 | RoyK | and |
| 15:09.55 | ThoMe | RoyK: excusa |
| 15:09.57 | RoyK | try NoOp(GotoIf($["${CALLERID(num)}"="8"]?setze-thomas-absender:setze-default-absender)) |
| 15:10.36 | shtoom | Hi, when I stream audio file using AGI, is there a way to interrupt playback from agi program it self? not by user interruption by pressing digits. |
| 15:10.41 | ThoMe | RoyK: you mean this: http://paste.keks.be/53 ? |
| 15:10.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:11.04 | FinboySlick | Maliuta: So long as it's not just to be told that 'you can't because of licensing issues with mp3', I don't mind reading. |
| 15:11.17 | RoyK | ThoMe: try without spaces around the = |
| 15:11.25 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:11.49 | ThoMe | RoyK: eem http://paste.keks.be/54 |
| 15:11.51 | ThoMe | yes |
| 15:11.51 | ThoMe | mom |
| 15:12.24 | ThoMe | RoyK: result http://paste.keks.be/55 |
| 15:13.50 | RoyK | if it jumps to default absender with that, asterisk must be pretty fscked |
| 15:14.08 | ThoMe | "fscked" ? |
| 15:14.15 | [TK]D-Fender | LOL |
| 15:14.17 | [TK]D-Fender | chumps. |
| 15:14.22 | [TK]D-Fender | Read that pastebin again! |
| 15:14.35 | ThoMe | [TK]D-Fender: öh. hi. hmm? |
| 15:14.40 | ThoMe | *read* |
| 15:14.43 | [TK]D-Fender | rolls his eyes a full 360 |
| 15:14.52 | ThoMe | ° |
| 15:14.53 | Zeeek | tsk |
| 15:15.21 | ThoMe | [TK]D-Fender: set this two caller id hm? |
| 15:15.32 | [TK]D-Fender | ThoMe: Nope, not the problem... |
| 15:15.37 | ThoMe | not? hm |
| 15:15.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:15.41 | [TK]D-Fender | This is jsut kinda sad... |
| 15:15.57 | [TK]D-Fender | c'mon people! |
| 15:16.05 | RoyK | ThoMe: fsck is a geek's replacement of fuck |
| 15:16.16 | ThoMe | [TK]D-Fender: one tip? :-( |
| 15:16.26 | [TK]D-Fender | ThoMe: READ YOUR PASTEBIN. |
| 15:16.30 | Zeeek | RoyK spoiled the magic of geek talk |
| 15:16.34 | ThoMe | [TK]D-Fender: yes yes.. you mean ";" ? |
| 15:17.09 | [TK]D-Fender | ThoMe: No, the problem is so glaringly huge its still funny... |
| 15:17.13 | ThoMe | [TK]D-Fender: have a big screen, you dont need capitals :-) |
| 15:17.19 | ThoMe | hmmm |
| 15:17.28 | ThoMe | *loooking again* |
| 15:17.31 | [TK]D-Fender | ThoMe: look at what its *executing* |
| 15:17.56 | ThoMe | one "CALLERID(num)=089143346182" and "CALLERID(num)=08914334610" ? |
| 15:18.09 | ThoMe | ah ok |
| 15:18.09 | *** join/#asterisk afaict (n=afaict@gw2.integral.com.br) |
| 15:18.12 | afaict | hii Folks! |
| 15:18.20 | [TK]D-Fender | ThoMe: No, not that. |
| 15:18.24 | ThoMe | mhhh! |
| 15:18.43 | ManxPower | normally you don't have a 0 in the callerid when sending to the telco |
| 15:18.49 | ManxPower | as 0 is not part of the phone number |
| 15:18.53 | ThoMe | then.. i dónt know. i give up... |
| 15:19.17 | afaict | I have two PSTN lines and I would like to use those to do outgoing calls from my office. Can I use a TDM440E Digium card to accomplish that ? |
| 15:20.06 | ManxPower | afaict: I would have to look it up, but a TDM card with two FXO modules will do it. |
| 15:20.06 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 15:20.08 | [TK]D-Fender | ThoMe: Open your eyes and look at the APPLICATION that are getting called. |
| 15:20.25 | [TK]D-Fender | RoyK: Please dear God tell me you see it and are just holding back... |
| 15:20.27 | RoyK | ManxPower: he's callerid is '8' and that's what he's testing for. the test returns 1 and GotoIf jumps to the alternate position |
| 15:20.34 | outtolunc | you are noop'n not exec gotoif |
| 15:20.39 | [TK]D-Fender | RoyK: You don't, do you? |
| 15:20.57 | [TK]D-Fender | hands outtolunc the prize |
| 15:20.59 | RoyK | outtolunc: the noop's where what I told him to add to see the output of things |
| 15:21.16 | [TK]D-Fender | RoyK: and he wants to know why its not JUMPING. |
| 15:21.30 | afaict | ManxPower yes it has. TDM440E has. |
| 15:21.31 | RoyK | what he said, was why the gotoif jumped to the wrong place |
| 15:21.31 | [TK]D-Fender | well DUH of course ti won't jump... hes not CALLING gotoif! |
| 15:21.41 | afaict | ManxPower thanks |
| 15:21.46 | [TK]D-Fender | RoyK: its not jumping period! |
| 15:21.46 | ManxPower | afaict: I thought that card had FOUR not TWO FXO modules |
| 15:21.51 | afaict | ManxPower it has |
| 15:21.51 | ajohnson | rofl |
| 15:21.56 | [TK]D-Fender | roy he never undid the "noop! |
| 15:22.04 | afaict | ManxPower and I have four PSTN lines. I used two as example |
| 15:22.09 | afaict | ManxPower :) |
| 15:22.23 | ManxPower | afaict: next time don't use wrong examples |
| 15:22.27 | afaict | ManxPower ok |
| 15:22.33 | ThoMe | [TK]D-Fender: have try this: http://paste.keks.be/56 |
| 15:23.03 | ajohnson | ThoMe: That's not going to work |
| 15:23.05 | outtolunc | royk, the point was that *second noop (eindehend:3) should have not been a noop, only the first |
| 15:23.09 | [TK]D-Fender | ThoMe: And the output? |
| 15:23.10 | ManxPower | ThoMe: EXTEN:0 should do nothing |
| 15:23.32 | [TK]D-Fender | ManxPower: We're not there yet :) |
| 15:23.36 | ajohnson | If it hits thomas-absender it's going to follow the next priority |
| 15:23.45 | ajohnson | ThoMe: Which is default-absender |
| 15:23.51 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:24.04 | afaict | ManxPower Does it exists? A d-link router with four RJ11 to connect my four phones and setup this router to connect to asterik ? |
| 15:24.06 | [TK]D-Fender | ajohnson: We'll deal with his complete lack of logic once we get the GotoIf right ;) |
| 15:24.07 | ajohnson | So no matter what the clid will be 0 |
| 15:24.08 | RoyK | ThoMe: if it jumps to thmas-absender, it'll continue and set the callerid again |
| 15:24.14 | ThoMe | "setze-default-absender" > exten => _XXXXX.,n(setze-default-absender),Set(CALLERID(num)=08914334610) |
| 15:24.15 | afaict | s/asterik/asterisk/ |
| 15:24.16 | ajohnson | urgh copy/paste |
| 15:24.21 | ThoMe | setze-thomas-absender > exten => _XXXXX.,n(setze-thomas-absender),Set(CALLERID(num)=089143346182) |
| 15:24.25 | RoyK | s/asterisk/asstrix/ :P |
| 15:24.35 | ThoMe | ah ok |
| 15:24.36 | ThoMe | RoyK: lol |
| 15:24.37 | ThoMe | ok :) |
| 15:24.48 | ThoMe | RoyK: and a good solution is? ;) |
| 15:24.52 | RoyK | ThoMe: sec |
| 15:24.55 | ThoMe | i would jump to DIAL |
| 15:24.55 | ManxPower | afaict: I don't know. I use digium equipment |
| 15:24.59 | ThoMe | and not to the nother SET |
| 15:25.35 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:25.39 | afaict | ManxPower so i would have to buy a digium card to connect my four phones to asterisk. |
| 15:25.58 | afaict | ManxPower i thought d-link has a cheaper solution for that |
| 15:26.00 | RoyK | ThoMe: http://pastebin.com/d2faa3d64 |
| 15:26.07 | ManxPower | afaict: that is one option and the only option I can comment on. |
| 15:26.22 | afaict | ManxPower uhm. ok! thanks! |
| 15:26.28 | ThoMe | RoyK: ah.. easy ;) |
| 15:26.28 | ManxPower | RoyK: handing out fish again? |
| 15:26.30 | RoyK | ManxPower: does digium have ATAs? |
| 15:26.43 | ManxPower | RoyK: only that horrid little IAXy |
| 15:27.11 | russellb | heh |
| 15:27.14 | ManxPower | ThoMe: I call that "stone skipping" dialplan |
| 15:27.29 | ThoMe | ManxPower: and? :P |
| 15:27.37 | *** part/#asterisk iobug (n=gustav@212.124.71.46) |
| 15:27.41 | ManxPower | ThoMe: it's a VERY common thing in dialplans |
| 15:27.47 | shtoom | is there a non blocking file playback application in asterisk ? |
| 15:27.59 | ManxPower | shtoom: define "non-blocking" |
| 15:28.06 | RoyK | ManxPower: how is that pronounced? Yaiksy! ? |
| 15:28.19 | ManxPower | RoyK: p i e c e o f s h i t |
| 15:28.26 | ManxPower | that's how it's pronounced |
| 15:28.29 | RoyK | ah |
| 15:28.40 | ThoMe | ManxPower: ok |
| 15:29.08 | shtoom | ManxPower:return immediately and continue execution of dialplan while playing the file |
| 15:29.10 | ManxPower | shtoom: thread blocking, dtmf blocking, interruptable, socket blocking.... |
| 15:29.18 | ManxPower | shtoom: no. |
| 15:29.30 | ManxPower | shtoom: but you can get close with the Background application |
| 15:29.35 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:29.45 | Zeeek | In about 30 minutes, we'll be talking to JR Richardson about asterisk scaling, clustering and fun stuff. http://x2z.eu to join, IRC #voip-users-conference |
| 15:29.50 | ManxPower | background can be interrupted by DTMF, as can READ while playing a file. |
| 15:30.07 | shtoom | ManxPower:my problem is i need to inerrupt file play and play some other file on receiving some data over network |
| 15:30.31 | ManxPower | shtoom: that is a very complex thing |
| 15:31.05 | shtoom | ManxPower: ExternalIVR 1.6 version seems to be right choice but that is still in beta I am looking for some thing similar in 1.4 |
| 15:31.21 | ManxPower | there is externalivr in 1.4 |
| 15:31.28 | ManxPower | heck, I think it is even in 1.2 |
| 15:32.13 | shtoom | ManxPower: but it won't continue execution of dialplan in 1.4 it simply hangup when its done thats rude. in 1.6 there is 'E' option to continue dialplan |
| 15:32.29 | Vec | The Digium B410P does it use Zaptel, mISDN, both or either ? |
| 15:32.42 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 15:33.25 | Mike8861 | hello, can any help me, I think my PBX got some memoery issue, how to determine physcal memoery usuage ? |
| 15:33.31 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:33.31 | shtoom | ManxPower: I've tried to compile 1.6 version with 1.4 but there seems to be lots of things changed from 1.4 to 1.6 so its simply failing to compile |
| 15:33.47 | [TK]D-Fender | Mike8861: Boot something like an Ubuntu CD and run memtest |
| 15:34.08 | [TK]D-Fender | Mike8861: and for usage, "top" |
| 15:34.17 | shtoom | or else a backport of latest ExternalIVR to 1.4 will do |
| 15:34.17 | [TK]D-Fender | (aside from testing) |
| 15:34.42 | Mike8861 | op - 23:34:31 up 36 days, 36 min, 1 user, load average: 0.08, 0.02, 0.01 |
| 15:34.42 | Mike8861 | Tasks: 83 total, 1 running, 82 sleeping, 0 stopped, 0 zombie |
| 15:34.42 | Mike8861 | Cpu(s): 0.4%us, 0.2%sy, 0.0%ni, 99.4%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st |
| 15:34.42 | Mike8861 | Mem: 450508k total, 444896k used, 5612k free, 34528k buffers |
| 15:34.42 | Mike8861 | Swap: 779144k total, 128k used, 779016k free, 68168k cached |
| 15:34.54 | Mike8861 | does this sounds any proble m? |
| 15:35.02 | shtoom | apart from ExternalIVR is there a way to interrupt file playback ? |
| 15:35.24 | [TK]D-Fender | Mike8861: Looks like a lot of memory used, but that doesn't break it down for us. |
| 15:35.32 | [TK]D-Fender | Mike8861: and never spam in here again. |
| 15:35.43 | [TK]D-Fender | Mike8861: thats what pastebin is for |
| 15:36.04 | mrmeek | lol irc elitism |
| 15:36.22 | ajohnson | bah |
| 15:36.27 | Mike8861 | [TK]D-Fender: what should i do by now, memoery issue occurs after i settle IM problem |
| 15:36.33 | ajohnson | which codec is the high-bandwidth codec supported by polycoms? |
| 15:36.41 | Mike8861 | IM wass solve by installing openfire on the same machine = =.... |
| 15:36.56 | ThoMe | RoyK: last question: http://paste.keks.be/57 |
| 15:36.58 | Mike8861 | seems openfire use up the resources |
| 15:36.59 | [TK]D-Fender | Mike8861: I think you should be looking at whats USING your memory. That summary doesn't tell us WHAT is using up so much. |
| 15:37.16 | ThoMe | RoyK: if i jump to the ausgehend-isdn (outgoing-isdn) can i not use ${EXTEN:0} ? |
| 15:37.29 | rickbradley | [TK]D-Fender: hm. So, upgraded to 1.4.21, restarted, etc., (verified 1.4.21 is running by version inside asterisk -r) and I still get the same behavior: http://pastie.org/private/f7bideu1p8g3xwg3vpbsa |
| 15:37.30 | ajohnson | g722 |
| 15:37.58 | RoyK | ThoMe: EXTEN will be 's' in there |
| 15:38.07 | Mike8861 | [TK]D-Fender: maybe i got virus ??? |
| 15:38.09 | RoyK | ThoMe: make a macro and add the extension as an argument |
| 15:38.19 | Mike8861 | i saw a user running process, call halo_daemon |
| 15:38.42 | RoyK | ThoMe: sec |
| 15:38.56 | ManxPower | Mike8861: This channel is for Asterisk, not operating system or virus problems |
| 15:39.40 | [TK]D-Fender | rickbradley: Looks like a full-on, current bug. Go check Mantis for it, and post it if there isn't an entry already. |
| 15:39.46 | Mike8861 | slaps himself |
| 15:39.47 | rickbradley | [TK]D-Fender: ok, will do |
| 15:40.01 | RoyK | ThoMe: http://pastebin.com/d1fb42bd2 |
| 15:41.30 | Rem| | does RDNIS work for chan_sip? |
| 15:42.39 | RoyK | Rem|: presumably |
| 15:43.05 | ThoMe | RoyK: ah. very good :) |
| 15:43.47 | ThoMe | RoyK: thank you very much. That's very good of you. |
| 15:44.02 | Rem| | RoyK: have you seen it work between two sip servers? |
| 15:44.43 | RoyK | Rem|: I don't think I've tried. the code's in there, but I don't know if it works |
| 15:44.51 | ManxPower | Rem|: if it does not, you can transport it using SIPAddHeader, etc |
| 15:44.56 | mmlj4 | does anyone maintain a current debian repository? I can't abide * version 1.1.2 |
| 15:45.12 | ManxPower | mmlj4: compile from source |
| 15:45.17 | [TK]D-Fender | mmlj4: Compile from source like the rest of us |
| 15:46.15 | mmlj4 | sure, did that, but I'd prefer to have the distro's maintainers do that for me |
| 15:46.35 | [TK]D-Fender | mmlj4: Well they don't. Deal with it. |
| 15:46.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:46.56 | ManxPower | mmlj4: if you want to use a package, then you'll have to contact the package maintainer for support. |
| 15:47.18 | ManxPower | They tend to move files around and compile with weird options |
| 15:47.27 | mmlj4 | this I know...debian especially |
| 15:48.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:49.31 | RoyK | terrorizes his neighbors with playing Rammstein a little too loud :) |
| 15:49.59 | *** join/#asterisk smth (n=chatzill@CPE000d48380389-CM00195eda2522.cpe.net.cable.rogers.com) |
| 15:52.51 | smth | hi, does asterisk surpport early media so far ? |
| 15:52.54 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 15:52.54 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 15:54.58 | *** part/#asterisk shtoom (n=shtoom@59.93.124.6) |
| 15:55.17 | RoyK | smth: yeah, it does |
| 15:55.22 | RoyK | smth: use Progress() |
| 15:55.59 | *** join/#asterisk Trey-- (i=bacchus@ppp-70-129-183-188.dsl.rcsntx.swbell.net) |
| 15:56.26 | smth | <PROTECTED> |
| 16:06.10 | Zeeek | Listen to this call with JR on asterisk scaling and clusters: http://x2z.eu bye for now :) |
| 16:06.22 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 16:08.36 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 16:08.59 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 16:09.20 | DSpair | [TK]D-Fender, Sorry I didn't get back to you yesterday, I was swamped with Help Desk calls. |
| 16:09.37 | DSpair | I finally have some decent sip history on when those channels hang. |
| 16:10.40 | DSpair | I have narrowed the indications down to this: |
| 16:11.01 | DSpair | 1. Call in to voice mail and change the "unavailable message" |
| 16:11.06 | DSpair | 2. Approve the message |
| 16:11.10 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:11.21 | DSpair | 3. Hang up without using the menus to exit the voice mail application. |
| 16:11.34 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:12.30 | DSpair | 4. Sip channel will hang almost invariably. |
| 16:12.42 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 16:12.45 | [TK]D-Fender | DSpair: pastebin everything... |
| 16:13.02 | DSpair | http://pastebin.com/d36c82c90 |
| 16:13.32 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 16:13.39 | *** join/#asterisk J4zen (n=Jeroen@213-84-194-178.adsl.xs4all.nl) |
| 16:13.41 | [TK]D-Fender | DSpair: try again, this time with * CLI channel dumps, etc |
| 16:13.44 | krondorl | Hi, is there a channel for the FOP? |
| 16:14.05 | [TK]D-Fender | krondorl: nope. |
| 16:14.13 | DSpair | FOP? |
| 16:14.26 | krondorl | dang.. I can't seem to get something I want to do working right.. |
| 16:14.42 | krondorl | Front Operator Panel (interface to asterisk) |
| 16:14.44 | DSpair | ~FOP |
| 16:14.45 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 16:15.07 | krondorl | Opps Flash, not front.. LOL |
| 16:15.08 | DSpair | Ahhh . . . IC |
| 16:16.04 | krondorl | [TK]D-Fender: Can I still ask my question here then? |
| 16:16.14 | [TK]D-Fender | krondorl: So far you're fine.. |
| 16:17.29 | DSpair | I think that's jbot's link is wrong. That for an XSL formatter, not the Flash Operator Panel.... |
| 16:17.46 | DSpair | Duh, I just can't read... |
| 16:17.57 | DSpair | Nevermind, ignore the moron in the room. |
| 16:18.51 | krondorl | Cool, what I want to do is have some of the buttons show the status of 2 channels instead of just one as some of the phones actually have 2 different extensions. the buttons are getting way to small with the number of channels it has to monitor. |
| 16:19.52 | krondorl | And I still need to add more.. |
| 16:20.13 | krondorl | DSpair: Don't worry, we all go through that once in our lifetime.. |
| 16:20.38 | [TK]D-Fender | DSpair: That's right....you've hit your quota so don't do that again! |
| 16:20.48 | krondorl | lol |
| 16:21.34 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 16:21.43 | DSpair | S'ok... |
| 16:21.45 | krondorl | I was looking at the regexp buttons but I can't figure out how that works. |
| 16:22.32 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:23.09 | De_Mon | is there a release schedule for the 1.6beta's? I'm looking everywhere and can't really find a place that tells me proactively when to expect the next release |
| 16:23.18 | Qwell | De_Mon: when it's ready |
| 16:23.22 | Qwell | and not a moment later |
| 16:23.48 | De_Mon | no roadmap to speak of, and I should just keep an eye on the announcements mailing list, eh? |
| 16:24.11 | russellb | basically, once the dahdi transition is done |
| 16:24.13 | russellb | we're going RC |
| 16:24.23 | creativx | dahdi da |
| 16:24.33 | De_Mon | googles |
| 16:25.28 | *** join/#asterisk jreign (n=reign@router.asteriasgi.com) |
| 16:25.29 | jreign | <PROTECTED> |
| 16:25.30 | jreign | [Jun 27 13:17:29] NOTICE[25291]: app_directed_pickup.c:159 pickup_exec: No target channel found for 1556. |
| 16:26.12 | De_Mon | I was looking on mantis and there arn't a whole lot of open bugs, I was very encouraged :) |
| 16:26.33 | Qwell | De_Mon: soon though, no doubt |
| 16:27.58 | [TK]D-Fender | Qwell: It'll be SEVERAL moments later I'm sure ;) |
| 16:28.07 | Qwell | [TK]D-Fender: it's all scripted |
| 16:28.20 | *** join/#asterisk blackhole (n=Mishu@unaffiliated/blackhole) |
| 16:28.21 | Qwell | russellb has had his finger on the magic button for weeks now - just waiting to lift it |
| 16:28.25 | [TK]D-Fender | Qwell: And scripts take time to execute ;) |
| 16:28.31 | DSpair | Will 1.6 address some of the issues with AMI conflicts? |
| 16:28.44 | Qwell | DSpair: you're gonna have to be a little more specific... |
| 16:28.55 | blackhole | Is there some channel for asterisk development also? |
| 16:29.01 | Qwell | blackhole: #asterisk-dev |
| 16:29.17 | DSpair | Qwell, When more than one client connects to AMI there are sometimes "collisions" that cause crashes, at least as I understand it. |
| 16:29.27 | Qwell | DSpair: that hasn't been an issue in a very long time |
| 16:29.43 | DSpair | gets more and more annoyed at his vendor. |
| 16:30.34 | russellb | who is your vendor? |
| 16:30.59 | russellb | as they are probably running a 3-year old version |
| 16:31.08 | DSpair | I wrote a status monitor app in Java to show who in our department is on the phone, etc... My vendor said that it would conflict with his iSymphony application because they both use AMI. |
| 16:31.19 | russellb | that makes absolutely no sense |
| 16:31.34 | russellb | you can tell him the asterisk project lead said that :-p |
| 16:31.54 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 16:32.03 | DSpair | Hey, I'm new to using Asterisk, so I've been placing a lot of trust in our vendor, and they are quickly losing that trust and soon after our business. |
| 16:32.03 | Qwell | besides, even when it *was* a problem, there were very well known workarounds |
| 16:32.17 | Qwell | ie; astmanproxy |
| 16:32.29 | creativx | astmanproxy ftw! |
| 16:32.47 | DSpair | fumes. |
| 16:34.00 | DSpair | I am so tempted to call over there are chew them out right now.... Gah!!! |
| 16:34.51 | Qwell | DSpair: record it |
| 16:34.57 | russellb | heh |
| 16:37.32 | DSpair | I've been copying down my IRC sessions to show to their "dCap certified" guy. You guys over the last few days have refuted about 20 things that he told me couldn't or shouldn't be done. |
| 16:37.36 | *** join/#asterisk Firass-VC22 (n=Firass-z@method.vikcomm.wwu.edu) |
| 16:38.12 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:38.26 | Qwell | DSpair: Tell them that if they're running into such a critical problem as that, which makes it so only their program can run... |
| 16:38.38 | Qwell | that they need to report a bug. Don't just sit on it and tell your customers. |
| 16:38.41 | Qwell | that's stupid |
| 16:39.05 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 16:39.31 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:42.02 | jreign | anyone know why I'm getting no target channel found when I try to execute an Originate using the pickup application |
| 16:42.22 | DSpair | Qwell, The straw that broke the came; |
| 16:43.24 | [TK]D-Fender | DSpair: Anybody can pass a test.... |
| 16:43.24 | [TK]D-Fender | DSpair: Its reality that will bit back hard. |
| 16:43.24 | [TK]D-Fender | bite* |
| 16:43.24 | jjshoe | DSpair who is your vendor? |
| 16:43.24 | DSpair | Qwell, The straw that broke the camel's back was they told me that they were blaming my fax server (connected via IAX) was the cause of my hung SIP channels and that there was no way to trace a SIP session to troubleshoot. |
| 16:43.24 | DSpair | jjshoe, Bluegrass.net |
| 16:43.25 | Qwell | oh dear |
| 16:44.32 | DSpair | Qwell, I've been working with Linux and Open Source fore over 12 years now, and I have yet to run across a server app that you couldn't do some kind of debugging on. |
| 16:46.13 | DSpair | Heck, I still remember the first time that Asterisk was demo'd at IBM's booth for LinuxWorld New York. |
| 16:46.52 | jjshoe | I remember when regan was shot :) |
| 16:46.56 | DSpair | I don't think that Digium even existed yet. |
| 16:47.11 | DSpair | jjshoe, ALMOST shot... |
| 16:47.27 | DSpair | What'shisface hot hit in the head instead. |
| 16:47.33 | jjshoe | DSpair right |
| 16:49.13 | jjshoe | The sixth and final bullet ricocheted off the side of the limousine and hit the president under his left arm, grazing a rib and lodging in his lung, near his heart.[9] |
| 16:49.16 | jjshoe | <3 wikipedia |
| 16:51.01 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 16:51.03 | *** part/#asterisk rickbradley (n=rickbrad@www.eastcore.com) |
| 16:51.07 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 16:51.39 | routerweasel | anyone know how much the g729 codec license is? currently using GSM without any issues but just curious if there would be any quality advantages |
| 16:51.53 | ManxPower | routerweasel: $10/channel |
| 16:51.57 | DSpair | stands corrected. |
| 16:52.05 | spokra | anyone use the free g729 |
| 16:52.14 | Qwell | spokra: There is no such thing. |
| 16:52.19 | ManxPower | spokra: there is no such thing. That is an unlicensed codec and not legal anywhere. |
| 16:52.33 | ManxPower | and no it's not a software patent |
| 16:52.42 | spokra | it's compiled from a IBM lib |
| 16:52.46 | routerweasel | thats not bad...where does one buy the license? digium? |
| 16:52.52 | Qwell | routerweasel: yes. |
| 16:53.00 | DSpair | Would anyone be willing to take a look at this SIP history to see if anything jumps out as the cause of my hung SIP channels? |
| 16:53.04 | routerweasel | thanks... |
| 16:53.05 | ManxPower | spokra: Intel's actually. Intel can license the code for free, but that DOES NOT give you a license for the actual codec and the Intel docs SAY THIS. |
| 16:53.26 | Qwell | nor does Intel even give you rights to use the source |
| 16:53.38 | spokra | well it is avalible |
| 16:53.40 | ManxPower | nope, just the binary lib |
| 16:53.46 | ManxPower | spokra: so is MS office |
| 16:54.10 | spokra | so i guess the answer is NO ONE uses it!! |
| 16:54.13 | ManxPower | spokra: if digium allowed that pirated codec they could lose their license to use that codec. |
| 16:54.22 | Qwell | spokra: Correct. |
| 16:54.30 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 16:54.43 | ManxPower | spokra: no legitimate Asterisk uses it, it's not supported here or anywhere else. |
| 16:54.48 | spokra | or no one admits to using it |
| 16:55.42 | beek | the licensed g729 is compiled on intel only, right? |
| 16:55.42 | ManxPower | The only time I have ever seen Digium censor the mailing list and the mailinglist archives is when that pirated codec is discussed. |
| 16:55.53 | ManxPower | beek: you would have to look. |
| 16:55.59 | Qwell | beek: nope |
| 16:56.00 | [TK]D-Fender | beek: No. Go look at Digiums download site for a full list. |
| 16:56.10 | Qwell | beek: i386 and up, and also I believe sparc on Linux |
| 16:56.17 | Qwell | (though sparc is unsupported) |
| 16:56.19 | beek | heads to digium |
| 16:56.55 | beek | I'm running my * on an Asus GL-500P but want another small, low-power box that can run g729. |
| 16:58.39 | [TK]D-Fender | beek: going to be pretty tough. G729 is CPU intensive. |
| 16:59.35 | beek | That's what I was just reading.... it's not a necessity but I'm looking for a small box with PCI interface so that I can put in a TDM400 card. |
| 16:59.43 | DSpair | But WikiPedia claims that G.729a is more CPU friendly. |
| 17:00.01 | Qwell | DSpair: it's still intensive |
| 17:00.16 | [TK]D-Fender | DSpair: Shit looks very good when compared to CRAP |
| 17:00.26 | DSpair | [TK]D-Fender, Point well taken. |
| 17:01.11 | De_Mon | all my shit and crap look about the same :( |
| 17:01.40 | DSpair | OK, perhaps a better analogy would be "Piss looks pretty good when compared to shit."? |
| 17:02.14 | DSpair | thinks this has gone too far already... |
| 17:02.21 | De_Mon | yay more mental images |
| 17:02.28 | DSpair | [TK]D-Fender, Did you see my pastebin link? |
| 17:02.51 | [TK]D-Fender | DSpair: no |
| 17:03.03 | DSpair | http://pastebin.com/d36c82c90 |
| 17:03.11 | DSpair | That's the SIP history for the hung channel. |
| 17:03.24 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:03.30 | Keltus | s |
| 17:03.33 | DSpair | Do you see anything glaringly obvious? If not, how should I go about contacting Digium for help? |
| 17:04.17 | russellb | to get help ... 1) use latest version 2) bugs.digium.com |
| 17:04.24 | Qwell | DSpair: how'd you get that debug? O.o |
| 17:04.48 | Qwell | oh, sip history. use sip debug |
| 17:04.57 | DSpair | Qwell, Enabled "sip history" and once I had a hung channel, I typed "sip show history <chan info>" |
| 17:05.13 | *** join/#asterisk ianweller (n=ianwelle@fedora/ianweller) |
| 17:05.13 | DSpair | Qwell, I have not recompiled with debugging yet. |
| 17:05.19 | Qwell | don't need to |
| 17:05.22 | DSpair | Oh? |
| 17:06.09 | DSpair | Yikes!! How do I turn that back off? |
| 17:07.05 | De_Mon | sip no debug |
| 17:07.37 | DSpair | Thanks... |
| 17:07.49 | DSpair | My fax server generates too much traffic for that to be useful. |
| 17:08.03 | De_Mon | do sip debug <tab> |
| 17:08.16 | De_Mon | or help sip debug |
| 17:08.30 | DSpair | K, I gotta go get some curry. BBL, thanks for the help |
| 17:09.20 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:09.22 | artz22 | hi all |
| 17:09.48 | artz22 | i was wondering, is it possible to do software echo cancellation on asterisk? I mean, is there already a module or such that does it? |
| 17:10.07 | ManxPower | artz22: you have not read the book. |
| 17:10.28 | artz22 | ManxPower: you're right.. I only read the sip part I guess |
| 17:10.33 | ManxPower | There are at least 10 different echo cancelers that can be used with Asterisk, 4 or 5 of them are included in the source. |
| 17:10.43 | ManxPower | artz22: since EC doesn't happen with SIP..... |
| 17:10.49 | artz22 | thanks... i will look it up now |
| 17:12.11 | artz22 | yes, i'm trying to use it with a custom interface, since there is no hw echo cancellation |
| 17:12.37 | *** part/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 17:12.55 | ManxPower | artz22: it's impossible to do EC when the latency is as high as a SIP call |
| 17:13.23 | ManxPower | that's why EC needs to happen at the PSTN interface |
| 17:13.29 | Qwell | ManxPower: I wouldn't say impossible |
| 17:13.37 | Qwell | I mean, polycoms do EC |
| 17:13.55 | Qwell | it's just not feasible for Asterisk do try to EC VoIP channels |
| 17:14.01 | Qwell | to try* |
| 17:14.13 | ManxPower | Qwell: they still do it at the analog/digital conversion |
| 17:14.25 | Qwell | sure |
| 17:15.09 | ManxPower | and unless it's changed, it only happened on speakerphone calls |
| 17:15.44 | artz22 | there is a lot of realimentation on hybrid lines |
| 17:15.54 | artz22 | (feedback) |
| 17:16.35 | *** join/#asterisk gbr_ (n=gbr@201-41-126-43.gnace703.dsl.brasiltelecom.net.br) |
| 17:17.20 | [TK]D-Fender | ManxPower: Polycoms do AEL on any audio interface on them. |
| 17:17.26 | [TK]D-Fender | AEC* |
| 17:18.54 | artz22 | i'm with the book now, echo cancellation is mostly for zaptel telephones |
| 17:19.19 | artz22 | i think the best would be to do hw echo cancellation, although now it gets complicated to do so |
| 17:20.21 | FinboySlick | likes echo, it makes people not want to call him ;) |
| 17:24.31 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 17:25.08 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 17:25.31 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 17:27.18 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 17:27.28 | *** join/#asterisk thx2000 (n=bob@netblock-208-127-150-56.dslextreme.com) |
| 17:28.24 | thx2000 | I'm a complete newb to PRI. I've got a customer who has one on it's way, and to avoid looking like a complete newb, I was wondering how I'm supposed to get the info for the span statement in zaptel.conf |
| 17:28.35 | *** part/#asterisk ianweller (n=ianwelle@fedora/ianweller) |
| 17:31.45 | *** join/#asterisk neurosys0 (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 17:31.53 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-169-185.dsl.tul2ok.sbcglobal.net) |
| 17:33.05 | routerweasel | i've got a few grandstream gpx-2000 handsets and am tinkering with the multi purpose keys. I'd like to configure them as BLF red/green so that i can see when an extension is being used. I can get them to function as a speed dial and show a green LED when the phone is registered but the LEDS dont change colors when the extension is in use. Any thoughts? |
| 17:33.06 | [TK]D-Fender | thx2000: ~book ~wiki & google |
| 17:33.28 | [TK]D-Fender | routerweasel: Go set up your hints for them |
| 17:33.39 | [TK]D-Fender | routerweasel: lookup "presence" on the WIKI |
| 17:34.20 | routerweasel | doing so now...also, i think you were helping me last night on my inbound calling issues. it was a simple context issue which i have since resolved. Thanks again for your help |
| 17:34.49 | rwaite | * reminds me of sendmail. |
| 17:36.30 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:36.46 | thx2000 | I've read the wiki and the pdf's. I'm just wondering how I'm supposed to gather the vendor/manuf. specific info |
| 17:36.51 | thx2000 | Is this something they typically just hand you? |
| 17:37.02 | thx2000 | Is it something I'm just supposed to scour the web for? |
| 17:37.11 | thx2000 | Will I need to speak w/ their tech staff? |
| 17:38.12 | *** join/#asterisk MihiNomenEst (n=argh@65-85-36-130.client.dsl.net) |
| 17:38.43 | *** join/#asterisk implicit (n=bayan@ip68-105-92-248.sd.sd.cox.net) |
| 17:38.56 | jsmith | thx2000: If it's PRI, it's typically pretty easy |
| 17:39.30 | thx2000 | I'm sure it is...just a fear of the unknown I guess :P |
| 17:39.33 | jsmith | thx2000: Make sure you take timing from the telco, make sure your Framing/LineCoding is ESF and B8ZS, and your LBO is typically set to 0, unless your Asterisk box is more than 133 ft from the demarc |
| 17:39.55 | jsmith | Something like span => 1,1,0,esf,b8zs |
| 17:40.12 | jsmith | thx2000: (I'm assuming this is a PRI over a T1 in North America) |
| 17:40.21 | thx2000 | yup |
| 17:41.11 | thx2000 | It's going to have 8 voice channels...is that something that needs to be defined? |
| 17:41.42 | [TK]D-Fender | thx2000: in zapata.conf |
| 17:42.24 | thx2000 | fxsks=1-8 ? |
| 17:42.38 | *** join/#asterisk asdx (n=diego@adsl-133-186.click.com.py) |
| 17:42.44 | asdx | hi, does 1.6 already supports tls? |
| 17:42.46 | asdx | in sip |
| 17:42.54 | [TK]D-Fender | thx2000: not on PRI |
| 17:43.04 | [TK]D-Fender | thejsut go read the WIKI |
| 17:43.15 | russellb | asdx: yes |
| 17:43.35 | asdx | russellb: how can i configure it? |
| 17:43.38 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 17:46.49 | russellb | asdx: by reading sip.conf.sample and searching for tls |
| 17:46.55 | russellb | or doc/siptls.txt |
| 17:46.55 | asdx | russellb: ok thx |
| 17:48.56 | asdx | russellb: is there something like mod_event_sockets in asterisk, any plans for doing something like this? |
| 17:49.26 | asdx | or curl for retriving information |
| 17:49.36 | russellb | the manager interface has existed for years |
| 17:49.44 | russellb | which is a socket interface into asterisk events |
| 17:49.49 | asdx | but is that the same thing? |
| 17:49.55 | asdx | ok |
| 17:49.56 | russellb | it can also be accessed over http, and encoded in xml |
| 17:53.04 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:53.37 | *** join/#asterisk jjg (n=jjg@12.40.200.74) |
| 17:53.49 | jjg | hi, any of you guys familiar with h.281? |
| 17:54.24 | Qwell | jjg: for camera control? |
| 17:54.33 | jjg | Qwell, yeah |
| 17:54.40 | Qwell | only very slightly |
| 17:54.47 | Qwell | (basically just that it's for camera control :P) |
| 17:54.50 | jjg | Qwell, do you have any traces for it? |
| 17:54.56 | Qwell | no |
| 17:55.08 | Qwell | it's a spec though.. shouldn't be too hard to figure out |
| 17:55.14 | jjg | yah, i don't have the spec |
| 17:55.25 | jjg | need to look through itu website |
| 17:55.32 | jjg | but wireshark won't decode it |
| 17:55.48 | jjg | do you know if there is a sip analog? |
| 17:57.03 | Qwell | http://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.281-199411-I!!PDF-E&type=items |
| 17:57.39 | jjg | Qwell, thank you sir |
| 17:57.50 | jjg | itu docs are free, right? |
| 17:57.57 | Trey-- | hola Qwell |
| 17:58.16 | Qwell | Trey--: Who let you out of your cage? |
| 17:58.44 | Qwell | jjg: some are |
| 17:59.13 | jjg | Qwell, did you google or use itu site search engine to find the doc so quickly? |
| 17:59.22 | Qwell | both |
| 18:00.16 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 18:00.44 | bsdwarrior | I changed the server in iax.conf, how do I get to to register again. iax2 show peers shows the old one |
| 18:01.03 | bsdwarrior | already did iax2 reload |
| 18:01.08 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 18:01.11 | [TK]D-Fender | bsdwarrior: kill off its registry entry in astdb and reload |
| 18:01.36 | bsdwarrior | astdb ? |
| 18:01.43 | asdx | how do i generate the .pem files? |
| 18:01.46 | [TK]D-Fender | bsdwarrior: .... |
| 18:01.47 | asdx | with openssl? |
| 18:02.08 | [TK]D-Fender | asdx: http://www.google.ca/search?hl=en&q=how+do+i+generate+the+.pem+files%3F&btnG=Google+Search&meta= |
| 18:02.16 | bsdwarrior | how do you kill of its registry entry |
| 18:02.21 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 18:02.31 | Qwell | pem files..for..what? |
| 18:02.37 | [TK]D-Fender | bsdwarrior: go read the CLI command list. |
| 18:02.37 | Qwell | oh, sip tls |
| 18:02.57 | asdx | [TK]D-Fender: thx, you are the man |
| 18:06.29 | bsdwarrior | iax2 provision ? |
| 18:07.12 | [TK]D-Fender | bsdwarrior: "help database" <- load chan_clue.so |
| 18:08.08 | bsdwarrior | tkd-fender I asked the questions because I dont know, you dont need act like your 12 |
| 18:09.17 | jsmith | bsdwarrior: "database show" will show you the AstDB database, and you can use "database del" and "database deltree" to remove items from it |
| 18:09.38 | [TK]D-Fender | bsdwarrior: http://www.google.ca/search?hl=en&q=astdb+cli+registry&btnG=Search&meta= |
| 18:10.04 | [TK]D-Fender | bsdwarrior: a 2 second Google would have garnered you a page full of examples on this with no more than these 3 key words. |
| 18:11.21 | bsdwarrior | it worked on its on, teliax just sucks sometimes, thanks |
| 18:12.04 | asdx | do i need two servers for sip tls? |
| 18:12.18 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 18:13.20 | [TK]D-Fender | bsdwarrior: Can be said of jsut about every ITSP. |
| 18:13.41 | asdx | http://pastie.org/223537 |
| 18:14.39 | bsdwarrior | at least its not me. |
| 18:14.58 | bsdwarrior | later |
| 18:16.32 | *** join/#asterisk jicksta (n=jicksta@adsl-69-109-235-72.dsl.pltn13.pacbell.net) |
| 18:17.42 | shido6 | why, when i try to recompile asterisk it seems to start with the file i editted? whats telling it to do that? |
| 18:18.29 | russellb | magic |
| 18:18.59 | russellb | it doesn't recompile stuff that hasn't changed ... |
| 18:19.03 | FinboySlick | shido6: Unless you 'make clean', I think it will only recompile what is affected by your changes. |
| 18:19.05 | russellb | pretty standard build system stuff |
| 18:19.15 | shido6 | i did a make clean |
| 18:19.23 | shido6 | then tried again |
| 18:19.27 | Strom | russellb: really? all this time I thought you had miniaturized Depeche Mode and put them into "make" |
| 18:19.31 | shido6 | same thing |
| 18:19.51 | FinboySlick | shido6: What about make distclean? |
| 18:20.08 | shido6 | http://pastebin.com/m16eef714 |
| 18:20.12 | shido6 | did a make distclean, too |
| 18:20.26 | [TK]D-Fender | shido6: "./configure" <- |
| 18:20.36 | shido6 | did a configure before i did a make |
| 18:20.47 | [TK]D-Fender | shido6: Trash the whoel folder! |
| 18:21.11 | FinboySlick | [TK]D-Fender: But he might destroy a revolution in compiling efficiency! |
| 18:21.39 | shido6 | http://pastebin.com/m16eef714 is what i get as an error |
| 18:25.11 | shido6 | hrmm |
| 18:27.10 | *** join/#asterisk hardwire (n=hardwire@rdbk-9064.mtaonline.net) |
| 18:32.10 | asdx | russellb: when i do a sip reload i get an error... i have server.pem, not asterisk.pem... http://pastie.org/223555 |
| 18:32.20 | asdx | russellb: is this "asterisk.pem" name hardcoded or something? |
| 18:36.11 | *** join/#asterisk angryuser (n=angryuse@d04m-89-83-111-62.d4.club-internet.fr) |
| 18:36.38 | DSpair | OK, I have a complete SIP debug of a session which ended in a hung SIP channel. Would someone please take a look? http://pastebin.com/d70152d9a |
| 18:36.43 | seanbright | mv server.pem asterisk.pem |
| 18:36.59 | seanbright | that'll be $125 |
| 18:38.53 | DSpair | Ay caramba!! 3 hung sip channels and a server load of 5.33... the suck! |
| 18:40.11 | asdx | lol |
| 18:42.25 | *** join/#asterisk proute (n=anthony@ARouen-153-1-49-112.w90-17.abo.wanadoo.fr) |
| 18:42.36 | proute | Hello, |
| 18:42.39 | markid | <PROTECTED> |
| 18:44.01 | proute | I use asterisk 1.4.21. In my extensions.conf I put my did: exten 123456,1,Dial(SIP/15) It's ok. Now i use a new VoIP provider (Trunk SIP) this provider don't send me the DID. So I must put this line in my extensons.cond: |
| 18:44.17 | proute | exten => s,1,Dial(SIP/15). |
| 18:44.35 | proute | But How can i distinct thie DID number with "s" |
| 18:45.02 | proute | How can I route DID to extn15 and other DID to extn 16? |
| 18:45.09 | proute | thanks for your help... |
| 18:45.16 | Qwell | proute: Tell your provider you want them to send the DID |
| 18:45.16 | [TK]D-Fender | proute: Only reason it lands on "s" is because you didn't not tell your ITSP what exten to dial on your register statement and * told them "x" |
| 18:45.20 | Qwell | (it's trivial) |
| 18:45.23 | [TK]D-Fender | "s" |
| 18:45.31 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-131-249.32-151.iol.it) |
| 18:46.03 | proute | Qwellit's sewan a frencj provider |
| 18:46.28 | Qwell | ..what? |
| 18:46.36 | proute | oups |
| 18:46.41 | proute | french provider |
| 18:46.57 | proute | so i have for example 2 DUID number: 123456 and 456789 |
| 18:47.18 | proute | So If i put exten => 123456.... and exten => 456789... |
| 18:47.30 | proute | In my debug SIP i have this error: |
| 18:47.44 | [TK]D-Fender | proute: fix your REGISTER statements in sip.conf ,- |
| 18:47.50 | [TK]D-Fender | ^^ |
| 18:49.31 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 18:49.49 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:50.15 | proute | [TK]D-Fender: My register.? |
| 18:50.45 | [TK]D-Fender | proute: register => user:pass@itsphost.tld/thisistheextentheywilldial <-- |
| 18:52.03 | proute | [TK]D-Fender: If I understand correctly, if I have 10 DID via my count SIP, I must enter 10 times this line it's true? |
| 18:52.43 | asdx | Maitei*CLI> sip reload |
| 18:52.43 | asdx | SSL cert error <asterisk.pem> |
| 18:52.47 | asdx | i have asterisk.pem |
| 18:52.48 | asdx | there |
| 18:52.52 | asdx | wtf |
| 18:53.53 | nn | Anyone know much about unlocking RT31P2-NA's |
| 18:53.59 | nn | So i can use with *? |
| 18:56.03 | asdx | any softphones that support tls? |
| 18:56.52 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 19:00.14 | [TK]D-Fender | proute: depends on your ITSP. |
| 19:03.26 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 19:04.45 | ThoMe | hm. |
| 19:04.49 | ThoMe | i would like set dcontext |
| 19:04.50 | ThoMe | but [Jun 27 21:04:08] ERROR[26728]: cdr.c:304 ast_cdr_setvar: Attempt to set the 'dcontext' read-only variable!. |
| 19:05.01 | ThoMe | a solution is not posible or? :) |
| 19:05.08 | ThoMe | try "exten => s,1,Set(CDR(dcontext)="ausgehend-isdn") |
| 19:05.09 | ThoMe | " |
| 19:05.30 | flush | yo |
| 19:05.42 | flush | is this adaptable to the pstn http://cgi.ebay.ca/Antique-Telephone-w-Bell-Box-202-Oval-102-Works_W0QQitemZ330246597520QQihZ014QQcategoryZ38037QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 19:05.43 | flush | ? |
| 19:05.58 | Strom | flush: common-battery telephone technology hasn't changed in 100 years |
| 19:06.01 | flush | you plug the phone in the box? and the box.. to the wall or its just too old |
| 19:06.20 | Strom | it's essentially the same circuit as any analog phone you buy today |
| 19:06.26 | flush | so i could use this and plug it to my asterisk fxs port ? |
| 19:06.30 | Strom | yes |
| 19:06.39 | flush | theres a phone jack? |
| 19:06.45 | flush | cause id like to get an old phone.. i like them |
| 19:06.54 | DSpair | Woohoo!!! I've narrowed down my problem even more... Pressing "#" in the voice menus is what hangs my SIP channels. |
| 19:07.05 | DSpair | s/voice/voice mail/ |
| 19:07.11 | flush | but just dont want to figure out i cant use it you know.. so theres a phone jack going out of the box you see that i can plug in my tdm400p fxs module ? |
| 19:07.14 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:07.14 | nn | The only issue i suspect might be the pulse vs tone dialing thing' |
| 19:07.21 | jjg | anybody doing any HD videoconferencing with sip? |
| 19:07.42 | Strom | flush: no, you'll have to wire tip and ring in yourself |
| 19:07.56 | flush | unnh |
| 19:07.59 | Strom | flush: the phone jack as such was invented in 1972 |
| 19:08.45 | flush | ok cool i didnt even know, could this phone box be adapted to a phone jack as we know today without much complications ? |
| 19:09.03 | Strom | flush: there is no "adaptation" |
| 19:09.09 | Strom | it's the same damned electrical circuit |
| 19:09.28 | flush | haha ok cool i get this, but for connecting it to the fxs port i need to pass via a phone jack |
| 19:09.44 | jjshoe | die lol |
| 19:09.48 | jjshoe | oops, wrong iwndow |
| 19:09.53 | riddlebox | lol |
| 19:10.01 | Strom | yeah, so you put a modular plug on one end of a wire and then connect tip and ring to the network in the ringer box |
| 19:10.08 | Strom | simple simple simple |
| 19:10.54 | riddlebox | hey Strom, whats the biggest Asterisk install you have done? |
| 19:11.05 | flush | sounds quite cool |
| 19:11.33 | Strom | riddlebox: about fifty users -- but I've helped plan larger installs |
| 19:12.06 | russellb | i know of > 10k phone installs ... |
| 19:12.07 | riddlebox | hrmm so far mine is 8 users, I am hoping that our salesmen try to sell it more |
| 19:12.20 | flush | haha mine is 2 users |
| 19:12.23 | flush | i guess you gotta start somwhere |
| 19:12.54 | russellb | that's what's cool. it has worked for anywhere from 1 to 10's of thousands :) |
| 19:12.55 | flush | but i have set up recording for in and out calls automatically.. i really love asterisk thanks to the guy who started the project |
| 19:13.18 | russellb | no thanks to the people working on it today? :) |
| 19:13.25 | jjshoe | I really liked gaim in the beginning too... |
| 19:13.25 | riddlebox | ;) |
| 19:13.26 | flush | of courrrssee |
| 19:14.02 | asdx | can i use any file names for the .pem files in asterisk, or it has to be "asterisk.pem" ? |
| 19:14.08 | asdx | coz i still get an error |
| 19:14.11 | russellb | any filename |
| 19:14.21 | asdx | SSL cert error <asterisk.pem> |
| 19:14.23 | russellb | you may have not created it properly ... |
| 19:14.25 | asdx | http://pastie.org/223555 |
| 19:14.41 | russellb | where did you put the key? |
| 19:14.53 | asdx | sip.conf |
| 19:15.02 | russellb | literally, on the system |
| 19:15.04 | russellb | where is the file located |
| 19:15.16 | flush | Strom, a modular plug is a phone jack as i know it? sorry im quite n00b i just enjoy learning telephony |
| 19:15.28 | asdx | russellb: well the file is in /etc/asterisk/ssl/server.pem |
| 19:15.36 | Strom | flush: the jack is the hole in the wall; the plug is the plastic bit on the end of the wire |
| 19:15.45 | Strom | flush: read this |
| 19:15.47 | Strom | ~101 |
| 19:15.48 | jbot | well, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 19:15.55 | flush | thanks a lot |
| 19:15.58 | asdx | tlsenable=yes |
| 19:15.58 | asdx | tlscertfgile=/etc/asterisk/ssl/server.pem |
| 19:16.01 | asdx | russellb: ^^ |
| 19:16.20 | russellb | certfgile? |
| 19:17.10 | asdx | russellb: thats what the documentation says |
| 19:17.22 | asdx | On server_b: |
| 19:17.22 | asdx | [general] |
| 19:17.22 | asdx | tlsenable=yes |
| 19:17.23 | asdx | tlscertfgile=/etc/asterisk/asterisk.pem |
| 19:17.28 | russellb | tlscertfile |
| 19:18.02 | russellb | that was a typo in that doc |
| 19:18.29 | asdx | ok |
| 19:18.35 | asdx | i just copy/pasted it |
| 19:19.05 | russellb | nods |
| 19:19.45 | asdx | Maitei*CLI> sip reload |
| 19:19.46 | asdx | SSL cert error </etc/asterisk/ssl/server.pem> |
| 19:20.00 | asdx | hmmmmmmmmm |
| 19:20.17 | russellb | how did you create server.pem |
| 19:20.26 | ThoMe | is it not posoble? exten => s,n,Set(CDR("out-over")="ausgehend-isdn") |
| 19:20.33 | ThoMe | have the field created in the database |
| 19:20.39 | russellb | remove quotes |
| 19:21.29 | asdx | russellb: openssl req -new -days 3650 -out certreq.req -keyout server.pem |
| 19:21.50 | ThoMe | russellb: i have now. : -- Executing [s@macro-ausgehend-isdn:2] Set("SIP/8-b5977850", "CDR(out-over)=ausgehend-isdn") in new stack |
| 19:21.56 | ThoMe | but no update |
| 19:22.02 | *** join/#asterisk Superbartt (n=bart@82-168-44-79.ip.telfort.nl) |
| 19:22.16 | russellb | you have to use cdr_adaptive_odbc for it to get created IIRC |
| 19:22.23 | russellb | i have to go .. |
| 19:23.18 | DSpair | If a SIP channel will not respond to a "soft hangup" does anyone know of an alternative way to force it to hangup? |
| 19:23.51 | DSpair | Obviously excluding a restart of asterisk. |
| 19:23.59 | asdx | how do i know when i have TLS working? |
| 19:25.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:29.08 | *** join/#asterisk techie (n=techie@m120e36d0.tmodns.net) |
| 19:38.37 | [TK]D-Fender | asdx: Easy guess : place a call, dumpt the channel |
| 19:38.40 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 19:40.08 | rwaite | can i put allow=g729,ulaw in sip.conf or do i need a seperate line |
| 19:40.20 | outtolunc | yes you can |
| 19:40.36 | outtolunc | you should do disallow=all above that tho |
| 19:40.40 | rwaite | yeah |
| 19:40.49 | rwaite | thanx |
| 19:40.52 | outtolunc | np |
| 19:42.02 | [TK]D-Fender | rwaite: Separate lines |
| 19:42.11 | rwaite | ? |
| 19:42.22 | rwaite | the reason i ask is that in iax.conf there is a comma |
| 19:42.31 | rwaite | but not in sip |
| 19:50.37 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:52.41 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 19:57.04 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 20:00.16 | ddunavant | Can someone look at this and answer the question enclosed at this pastebin or at least attempt to help me resolve it. http://pastebin.com/d43559c5f |
| 20:00.19 | ddunavant | thanks |
| 20:00.32 | *** join/#asterisk makkksimal (n=makkksim@e177212241.adsl.alicedsl.de) |
| 20:01.59 | rwaite | i think its supposed to happen |
| 20:02.29 | ddunavant | Well crud, that wasn't the answer I was hoping for... |
| 20:02.31 | [TK]D-Fender | ddunavant: In your macro you have the ability to count the call as not answered, so I agree |
| 20:03.01 | ddunavant | is there anyway that I could set it to stop ringing the other lines at that point? |
| 20:03.15 | rwaite | i think callgroups are the answer> |
| 20:03.16 | rwaite | ? |
| 20:04.00 | rwaite | nevermind, no, i dont think they are |
| 20:04.07 | rwaite | voip-info says no :) |
| 20:04.16 | ddunavant | hmmm... |
| 20:04.37 | rwaite | i'm lookin at sum stuff |
| 20:04.45 | ddunavant | thanks |
| 20:04.47 | rwaite | what is the purpose of rining these phones in tandem> |
| 20:04.57 | ddunavant | a group of techs getting a call |
| 20:04.59 | rwaite | s/rining/ringing/ |
| 20:05.18 | rwaite | okay |
| 20:05.26 | ddunavant | one needs to pick up and the menu provides them with how the call got there and gives them the option to pick up or send to voicemail... |
| 20:05.29 | rwaite | i think a queue is what ur looking for |
| 20:05.45 | ddunavant | hmmm... |
| 20:05.56 | *** join/#asterisk J4zen (n=Jeroen@213-84-194-178.adsl.xs4all.nl) |
| 20:05.56 | rwaite | place all the extensions in a queue and set the ring settings to ring all at once |
| 20:06.28 | rwaite | http://www.voip-info.org/wiki/index.php?page=Asterisk+call+queues |
| 20:06.41 | rwaite | ringall will behave the way you want |
| 20:07.12 | ddunavant | thank you good sir, I will look this over and hopeful discern my answer :) |
| 20:07.24 | rwaite | i am setting up a system similar to that now |
| 20:07.57 | rwaite | we've 3 helpdesk techs and they all have to ring at once and share a common support vmbox. not easy for a beginner lemme tell you |
| 20:09.30 | [TK]D-Fender | rwaite: exten =>123,1,Dial(SIP/tech1&SIP/tech21&SIP/tech3,20) exten =>123,2,Voicemail(123@default) |
| 20:09.41 | [TK]D-Fender | rwaite: </the> |
| 20:10.04 | rwaite | heh. you are correct sir. but ... mwi on all phones was the hardest part :) |
| 20:10.14 | rwaite | which is a polycom thing |
| 20:10.25 | rwaite | but i got it working finally and now its smooth |
| 20:11.09 | [TK]D-Fender | rwaite: MWI? Complete non-issue |
| 20:11.24 | [TK]D-Fender | rwaite: there is NOTHING to configure on the Polycom to get MWI |
| 20:11.45 | [TK]D-Fender | rwaite: mailbox=[box#]@[vmcontext] is all |
| 20:11.53 | rwaite | well the prob is the incoming has an option to leave a msg for support |
| 20:12.03 | rwaite | well yeah, and im not saying this isnt cause i am dumb, |
| 20:12.33 | rwaite | but its just i had to figure out how to add a second line to each phone that would indicate when there was a msg in the support mailbox |
| 20:12.51 | rwaite | i eventually did it with 123a, 123b, and 123c, each sharing the 123@default mailbox |
| 20:12.52 | *** join/#asterisk asdx (n=diego@adsl-133-186.click.com.py) |
| 20:13.09 | rwaite | then each polycom phone has the a/b/c line as a second line |
| 20:13.53 | rwaite | if i had them share *one* 123, only one of the phones (the last one to start/subscribe) would get mwi sip messages |
| 20:13.57 | [TK]D-Fender | rwaite: Well you can work off a single reg & have MWI for both, but you'll have to check both to know which |
| 20:13.59 | asdx | will TLS encrypt the SIP traffic? |
| 20:14.11 | rwaite | yeah. i'm no expert in this stuff thats for sure |
| 20:14.12 | asdx | do i still need a VPN for bypass my telco? |
| 20:14.19 | asdx | if i use TLS |
| 20:14.34 | [TK]D-Fender | asdx: They will know its SIP. |
| 20:14.59 | rwaite | bypass telco? like, as in telco doesnt allow sip traffic? |
| 20:15.06 | asdx | rwaite: yeah |
| 20:15.12 | rwaite | ouch |
| 20:16.34 | rwaite | [TK]D-Fender: do you know about do not disturb and polycom phones |
| 20:17.23 | *** join/#asterisk vladtser1 (n=Vladimir@adsl-065-015-017-142.sip.mia.bellsouth.net) |
| 20:18.59 | rwaite | nevermind i found it out by testing it |
| 20:27.57 | rwaite | i have a question, if i have a regular line thru a std telco... if i set up that number to forward to my voip provider's did number ... will that tie up the std telco line, or will it release so if someone calls again while the original forwarded call is still in process |
| 20:28.07 | rwaite | they can still get thru |
| 20:28.50 | hardwire | heh.. so does anybody else here add every single zaptel module to /etc/modprobe.d/blacklist.zaptel ? |
| 20:28.56 | hardwire | I love it, fixes so many problems |
| 20:29.19 | hardwire | tells /etc/modules what to load.. I WILL BE THE MASTER OF THIS COMPUTER! |
| 20:29.30 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 20:30.27 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:31.20 | asdx | [netman]: and what is what gets encrypted then? the identifier, protocol? |
| 20:31.35 | asdx | err |
| 20:31.40 | asdx | [TK]D-Fender i mean |
| 20:34.01 | *** part/#asterisk dijungal (n=kdaniel@205.244.151.188) |
| 20:47.07 | asdx | Maitei*CLI> sip reload |
| 20:47.08 | asdx | SSL cert error </etc/asterisk/ssl/server.pem> |
| 20:50.21 | hardwire | pulls the pin and throws |
| 20:50.28 | hardwire | "MySQL or PostgreSQL?" |
| 20:51.11 | asdx | hardwire: the one it works for you better |
| 20:51.23 | hardwire | sigh |
| 20:51.42 | asdx | i'm using mysql lately and works quite well for me |
| 20:51.49 | hardwire | I just don't know how each asterisk db interface works... it seems like they were developed completely out of sync with eachother |
| 20:51.49 | asdx | even though i still like postgresql more |
| 20:51.54 | hardwire | odbc being the only real option |
| 20:52.03 | hardwire | I like postgresql cause I'm a postgis nut |
| 20:52.29 | hardwire | most docs for asterisk are all mysql based.. I also kind of pull my hair out trying to decide on the best cdr in sql schema |
| 20:52.39 | asdx | postgresql always was more serious... mysql lacked a lot of features in 4.x |
| 20:52.53 | hardwire | yar |
| 20:53.06 | hardwire | the mysql "Carrier Grade Extensions" are spiffy however. |
| 20:53.13 | hardwire | but maybe overkill for what I need. |
| 20:53.27 | hardwire | I should probably stick with MySQL since most asterisk toys only speak mysql |
| 20:53.29 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:55.31 | asdx | [TK]D-Fender: how do they will know it's SIP? what will the TLS encrypt? the protocol? |
| 20:55.39 | asdx | [TK]D-Fender: what about if i run on a non standard port? |
| 20:55.47 | asdx | with TLS |
| 20:57.10 | [TK]D-Fender | asdx: just because your call is encryped, you think they can't tell its a call? |
| 20:57.32 | [TK]D-Fender | asdx: the protocol headers are probably very easily detected. |
| 20:57.49 | [TK]D-Fender | asdx: Is the risk of what you're doing really worth it? |
| 20:59.41 | asdx | uhm |
| 21:00.21 | rwaite | hardwire mysql is garbage |
| 21:00.34 | rwaite | use postgres |
| 21:00.35 | hardwire | :) |
| 21:00.51 | asdx | [TK]D-Fender: just want to make this secure |
| 21:01.21 | *** join/#asterisk RoyK (n=roy@ip-168-52-149-91.dialup.ice.no) |
| 21:03.14 | [TK]D-Fender | asdx: Well AFAIK, SIP TLS would only encrpt the SIP messaging. This doesn't imply anything about RTP. |
| 21:03.26 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 21:03.35 | [TK]D-Fender | asdx: and it depends on your idea of "secure". |
| 21:03.51 | mcab | correct, SIP & TLS only handles signalling, you need SRTP to encrypt the RTP as well |
| 21:04.10 | asdx | i agree that security is a broad term, it can mean many things |
| 21:04.19 | [TK]D-Fender | mcab: And thats something I haven't seen a definitive doc confirmed for 1.6 yet |
| 21:04.24 | asdx | for different people |
| 21:04.45 | mcab | [TK]D-Fender: yeah, is TLS even supported in 1.6 yet? |
| 21:04.50 | russellb | TLS, yes |
| 21:04.54 | mcab | cool |
| 21:04.54 | russellb | SRTP, no |
| 21:04.55 | [TK]D-Fender | mcab: yup |
| 21:05.07 | [TK]D-Fender | russellb: <- the man has spoken |
| 21:05.11 | asdx | [TK]D-Fender: i just want to encrypt everything, protocol, media and change the ports |
| 21:05.18 | russellb | then use openvpn |
| 21:05.21 | [TK]D-Fender | asdx: And you've just been shot down. |
| 21:05.36 | mcab | TLS is a good start, at least you can do Point to Point sRTP with that... |
| 21:05.40 | asdx | russellb: is sRTP coming? |
| 21:05.40 | russellb | openvpn is eaaaasy |
| 21:05.40 | [TK]D-Fender | asdx: Better stok up on K-Y |
| 21:05.46 | russellb | yes, it is coming |
| 21:05.50 | asdx | [TK]D-Fender: K-Y? |
| 21:05.55 | russellb | lube. |
| 21:06.30 | asdx | russellb: so with TLS+SRTP vpn wont be necessary anymore to encrypt everything? |
| 21:07.35 | asdx | i will give openvpn a try |
| 21:07.46 | implicit | asdx, one prob |
| 21:07.52 | asdx | i will test 1.6 features first ho |
| 21:07.53 | asdx | tho |
| 21:07.53 | implicit | your VPN will not give point-point encryption |
| 21:07.59 | implicit | best way to do encryption |
| 21:08.12 | implicit | is with TLS for signaling and DTLS-SRTP for media |
| 21:08.23 | asdx | k |
| 21:08.44 | implicit | you can do DTLS-SRTP in multiple modes |
| 21:08.48 | implicit | including short-auth strings |
| 21:09.02 | bkruse | tzafrir_home: ping |
| 21:09.11 | implicit | and then you can send hashes through TLS as extra header fields on sip messages to assure that you have no media stream MITM |
| 21:09.12 | asdx | thx |
| 21:09.19 | implicit | in addition to verbal confirmation w/ short-auth strings |
| 21:09.31 | implicit | so even voice-impersonators couldn't break into the ad-hoc setup |
| 21:09.37 | implicit | w/ MITM |
| 21:09.56 | asdx | cool |
| 21:10.13 | implicit | let me know if you have more detailed questions, i've done a lot of work in voice encryption |
| 21:10.18 | implicit | w/ SIP |
| 21:10.30 | asdx | ok |
| 21:10.41 | [TK]D-Fender | asdx: And again, none of this to my awareness will stop them from knowing you're placing SIP calls |
| 21:11.46 | mcab | [TK]D-Fender: I guess it's based on the premise that to the provider it just looks like a TLS connection |
| 21:11.46 | asdx | hm |
| 21:12.00 | implicit | if you use TLS |
| 21:12.04 | implicit | it is indistinguishable |
| 21:12.10 | implicit | just use TLS on the port of another service |
| 21:12.12 | implicit | like 443 |
| 21:12.14 | mcab | although, the RTP streams still look like RTP streams |
| 21:12.16 | implicit | and run your sip server there |
| 21:12.35 | rabelais | can I use echo cancellation like oslec on audio coming in from a sipura 3000? |
| 21:12.36 | implicit | mcab, not necessarily, take a look at dtls-srtp standard |
| 21:12.41 | implicit | just looks like udp trash |
| 21:12.46 | [TK]D-Fender | rabelais: No. |
| 21:12.55 | asdx | http://en.wikipedia.org/wiki/Image:Ssl_handshake_with_two_way_authentication_with_certificates.svg |
| 21:13.18 | rabelais | *sniff* that sucks...the sipura's echo cancellation blows |
| 21:13.25 | implicit | asdx, I recommend you talk to Dan Wing if you are interested in implementing a DTLS-SRTP client |
| 21:13.26 | mcab | implicit: *nod* interesting; I've only dealt with plain TLS/SRTP and it shows up a RTP in wireshark :-) |
| 21:13.58 | asdx | implicit: who is him |
| 21:14.13 | implicit | he is working on some of the internet drafts for key-exchange verification on dtls-srtp |
| 21:14.24 | implicit | he's a good guy, send him and email and he'll probably do a conf w/ you |
| 21:14.33 | asdx | ok |
| 21:14.41 | implicit | i've talked to him before about this stuff a lot, he's active in the IETF in DTLS-SRTP right now |
| 21:14.47 | implicit | mikey is dead |
| 21:14.51 | implicit | and sdescriptions too |
| 21:15.54 | implicit | mikey is slow and has some major issues especially w/ short auth strings, which i'm interested in |
| 21:16.02 | implicit | zrtp you have to license |
| 21:16.12 | implicit | and it is not as flexible as dtls-srtp |
| 21:16.18 | asdx | what about sip over ssh, is that possible |
| 21:16.19 | asdx | ? |
| 21:16.28 | implicit | sure but then you don't have point-point encryption |
| 21:16.35 | implicit | which is a problem |
| 21:16.38 | implicit | at least for my applications |
| 21:17.04 | implicit | ideally I would use DTLS-SIP and DTLS-SRTP |
| 21:17.07 | implicit | do everything over UDP |
| 21:17.12 | [TK]D-Fender | asdx: unrecommended. |
| 21:17.21 | implicit | [TK]D-Fender: yeah |
| 21:17.31 | implicit | if you just want signaling encryption |
| 21:17.32 | [TK]D-Fender | asdx: voice is not carried over TCP |
| 21:17.36 | implicit | do TLS-SIP or DTLS-SIP |
| 21:17.49 | implicit | for media you have a few options |
| 21:18.03 | implicit | sdes, mikey, dtls-srtp, and zrtp |
| 21:18.09 | [TK]D-Fender | implicit: What he wants is to run on nonstandard ports and for that traffic, even if encrypted to not be detectable as a VoIP call. |
| 21:18.21 | implicit | yep |
| 21:18.24 | implicit | i've done that sort of thing before |
| 21:18.25 | [TK]D-Fender | asdx: And your packet rate, etc will still give you away |
| 21:18.26 | implicit | i understand |
| 21:18.42 | implicit | i know a way to get around that eaily |
| 21:18.44 | implicit | packet rate issues |
| 21:18.48 | implicit | you can make dynamic frame size |
| 21:19.01 | implicit | and even rebuild it using something like the new rtpproxy |
| 21:19.05 | implicit | and normalize it |
| 21:19.08 | [TK]D-Fender | implicit: Sounds like some supoer funky intermediary gateway trickery to be added... |
| 21:19.14 | *** join/#asterisk proute (n=anthony@ARouen-153-1-49-112.w90-17.abo.wanadoo.fr) |
| 21:19.31 | implicit | signaling and media could take different paths even |
| 21:19.38 | implicit | and also no b2buas or anything |
| 21:19.46 | proute | Sorry, I have a bug with my pc :( |
| 21:19.46 | asdx | [TK]D-Fender: i see |
| 21:19.52 | *** part/#asterisk gbr_ (n=gbr@201-41-126-43.gnace703.dsl.brasiltelecom.net.br) |
| 21:20.20 | implicit | i use OpenSER for all of my core routing and many of the sip application proxies i run |
| 21:20.37 | proute | [TK]D-Fender, your answer about my problem is ok. But I have again a litle problem about this line: |
| 21:20.41 | implicit | and if i need to do things that sip proxies can't do i use signaling-only b2bua's on the periphery |
| 21:20.45 | proute | <[TK]D-Fender> proute: register => user:pass@itsphost.tld/thisistheextentheywilldial <-- |
| 21:20.53 | implicit | and use asterisk etc purely as media-servers |
| 21:21.22 | proute | When I call the DID 12345 it's ok, sometimes when i call DID 12345 it's forwarded to 456789 |
| 21:25.51 | [TK]D-Fender | proute: PASTEBIN is your friend. |
| 21:25.53 | [TK]D-Fender | ~pb |
| 21:25.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:25.54 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:25.55 | rabelais | can I use echo cancellation like oslec on an X100P card? |
| 21:26.00 | [TK]D-Fender | rabelais: Yes |
| 21:26.16 | rabelais | wheee, guess I found my next purchase then, hehe |
| 21:26.48 | [TK]D-Fender | proute: pastebin your sip.conf masking only passwords, your dialplan, and SIP debug of calls coming in. |
| 21:27.06 | [TK]D-Fender | rabelais: Scraping the bottom of the barrel. Good luck with that. |
| 21:27.32 | asdx | does asterisk uses a socket for doing "asterisk -r" |
| 21:28.04 | rabelais | [TK]D-Fender, ya, well...I'm a student, I'm used to scraping the bottom of the barrel |
| 21:28.40 | asdx | can i create a ruby script that does the same thing as "asterisk -r" over TCP, etc? is this what AMI is for? |
| 21:29.11 | [TK]D-Fender | asdx: Go read the book and get a clue. The purposes of things like that are well defined |
| 21:29.44 | asdx | ... ok |
| 21:29.47 | [TK]D-Fender | asdx: And that is a very open-ended case of scope for "is AMI just like CLI?" |
| 21:30.02 | [TK]D-Fender | asdx: if all you care about is 1 function, then maybe, but probably not. |
| 21:30.10 | asdx | [TK]D-Fender: i'm asking for a interface to write scripts |
| 21:30.51 | [TK]D-Fender | asdx: Another wonderfully vague goal. |
| 21:31.00 | asdx | is not vague |
| 21:31.01 | asdx | it's fun |
| 21:31.13 | [TK]D-Fender | asdx: Details will get you ^!%@ing killed a lot of the time. |
| 21:31.27 | asdx | why? |
| 21:31.28 | asdx | :( |
| 21:31.38 | *** join/#asterisk cestchiant (n=anthony@ARouen-153-1-49-112.w90-17.abo.wanadoo.fr) |
| 21:31.54 | cestchiant | oups sorry again problem proute is my ghost |
| 21:32.08 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:32.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:32.14 | cestchiant | I re-post my question: |
| 21:32.27 | cestchiant | [TK]D-Fender, your answer about my problem is ok. But I have again a litle problem about this line: |
| 21:32.29 | [TK]D-Fender | cestchiant: We all heard you already |
| 21:33.06 | [TK]D-Fender | cestchiant: And I asked you for BACKUP. |
| 21:33.27 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 21:34.12 | *** join/#asterisk Superbartt (n=bart@82-168-44-79.ip.telfort.nl) |
| 21:35.02 | [TK]D-Fender | asdx: "scripting" can mean anything. AMI can do CERTAIN things, CLI can do CERTAIN things. They are not all the SAME things. So therefor if you want to know if AMI will do the job, you'd better describe the job in detail. |
| 21:35.20 | creativx | AMI != CLI |
| 21:35.40 | asdx | [TK]D-Fender: ok |
| 21:35.47 | asdx | i see it's not possible then |
| 21:35.58 | [TK]D-Fender | asdx: Like if you were to ask me if glass was poisonous, I'd say no. Then I'd grind some up, put it in your food, and it'd kill you. So watch out for the details. |
| 21:36.30 | [TK]D-Fender | asdx: "Its not possible"? You didn't even say what "it" was you were trying to do! |
| 21:38.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 21:38.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21 (2008/06/12) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 21:39.00 | *** join/#asterisk Veggen (n=vegard@213.187.177.230) |
| 21:39.46 | [TK]D-Fender | OMFG Firefox 3 beyond rocks.... its zoom even works on embedded video. |
| 21:39.49 | asdx | computer illiterate? lol |
| 21:39.59 | asdx | i'm just curious |
| 21:40.11 | asdx | [TK]D-Fender: yeah for flash |
| 21:40.16 | [TK]D-Fender | asdx: Remember the cat..... |
| 21:40.26 | asdx | the cat killed the curious? |
| 21:40.39 | asdx | was it that? |
| 21:40.57 | creativx | [TK]D-Fender: you should be glad you get paid :-) |
| 21:41.17 | Veggen | I'm considering coding a messaging-service for my private asterisk. It's gonna work like this: I dial in to home, dial in a number that will receive that message when they call, and then read in the message... |
| 21:41.18 | [TK]D-Fender | creativx: Not my norm... it was such a slow day though... |
| 21:41.23 | Veggen | has anyone done similar? |
| 21:41.23 | lmadsen | [TK]D-Fender: now you can zoom into your porn! :) |
| 21:41.45 | [TK]D-Fender | Veggen: that sounds like a private VM system. |
| 21:41.57 | [TK]D-Fender | Veggen: very easy to make. |
| 21:42.10 | Veggen | Fender: shouldn't be that hard, I think. |
| 21:42.16 | [TK]D-Fender | Veggen: not at all. |
| 21:42.28 | [TK]D-Fender | lmadsen: mmm pr0n |
| 21:42.44 | creativx | [TK]D-Fender: hehe.. wait.. not your norm?? do nothing all the time? does not compute! |
| 21:42.50 | lmadsen | [TK]D-Fender: I wish I liked prawns :) |
| 21:42.55 | Veggen | Fender: would it be useful with ragi or similar, or overkill, I wonder? |
| 21:43.02 | [TK]D-Fender | lmadsen: wimp! |
| 21:43.06 | lmadsen | :D |
| 21:43.24 | [TK]D-Fender | lmadsen: I've eaten whole shrimp (heads & all). |
| 21:43.25 | creativx | lmadsen i might have asked you this before, but how come your name is so damn scandinavian |
| 21:43.26 | Veggen | <---sort of a beginner in asterisk, this is more a learning project :) |
| 21:43.45 | lmadsen | creativx: because my late grandfather was born in Denmark :) |
| 21:43.45 | [TK]D-Fender | creativx: He's blond too! How dare he! |
| 21:43.51 | lmadsen | and blue eyes! |
| 21:44.01 | creativx | lmadsen: ahh nice |
| 21:44.03 | creativx | hehe |
| 21:44.03 | [TK]D-Fender | lmadsen: Now you've gone TOO far!!! |
| 21:44.04 | creativx | damn blondines |
| 21:44.05 | lmadsen | good viking blood :) |
| 21:44.11 | lmadsen | actually I'm not blond |
| 21:44.15 | lmadsen | blonde* |
| 21:44.15 | creativx | im not blond either |
| 21:44.18 | creativx | +e |
| 21:44.21 | lmadsen | lol |
| 21:44.26 | creativx | although i live in vikingland |
| 21:44.36 | lmadsen | I was blonde when I was like 2-3 |
| 21:44.40 | lmadsen | creativx: Denmark? |
| 21:44.41 | creativx | we all were |
| 21:44.45 | creativx | no norwegien |
| 21:44.47 | lmadsen | ah gotcha |
| 21:44.54 | creativx | right up the street |
| 21:44.59 | cestchiant | it's stange, When I do a debug SIP in filed "TO" and "contact" I have not the same data (To: <sip:012345642@xx.xx.xx.xx> Contact: <sip:012345640@xx.xx.xx.xx>) Why? |
| 21:45.27 | lmadsen | creativx: haha... I'm Toronto, Canada :) |
| 21:45.42 | creativx | lmadsen: hehe cool |
| 21:45.43 | cestchiant | I try to call 012345642 and I speak with 012345640 .... |
| 21:45.46 | routerweasel | is it possible to set up a 'dummmy' extension is configured to forward calls outside the system regardless if a phone is registered? |
| 21:45.52 | creativx | so you know the trailerpark boys then lmadsen |
| 21:46.09 | [TK]D-Fender | cestchiant: pastebin the ENTIRE message. |
| 21:46.15 | lmadsen | creativx: haha very much so :) |
| 21:46.26 | cestchiant | [TK]D-Fender can you give me the url please? |
| 21:46.28 | creativx | lmadsen: heheh, great series. i love those guys |
| 21:46.29 | [TK]D-Fender | routerweasel: Extensions dow hatever you tell them to. |
| 21:46.36 | lmadsen | creativx: I come from a town of 4500 people... the trailer park boys was just life for about 18 years :) |
| 21:46.46 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:46.53 | [TK]D-Fender | lmadsen: And now live in a "town" of about 3/4 million :p |
| 21:47.01 | lmadsen | pretty much :) |
| 21:47.04 | routerweasel | TKD: Do you know the syntax for this? I assume it would go in extensions.conf? |
| 21:47.09 | lmadsen | I got my new condo yesterday, w00t! |
| 21:47.26 | [TK]D-Fender | routerweasel: that's where extensions go you know... |
| 21:47.36 | [TK]D-Fender | lmadsen: Rent or buy? |
| 21:47.45 | lmadsen | [TK]D-Fender: buy this time! |
| 21:47.53 | [TK]D-Fender | lmadsen: Wow. What part of town? |
| 21:47.54 | creativx | lmadsen: hehe nice. 4500 ppl should be enough to either know everybody or have slept with everyones daughter. |
| 21:47.59 | lmadsen | [TK]D-Fender: Yonge and Dundas |
| 21:48.06 | lmadsen | creativx: or cousin |
| 21:48.07 | [TK]D-Fender | lmadsen: Wait.. you moved to central TO didn't you? |
| 21:48.19 | lmadsen | [TK]D-Fender: about as central as you can get, ya |
| 21:48.37 | [TK]D-Fender | lmadsen: Increast that 3/4 Mil then ;) And OUCH, morgage must be huge |
| 21:48.40 | creativx | hehe |
| 21:48.54 | lmadsen | [TK]D-Fender: not toooo bad... monthly "rent" will be cheaper than what I'm paying now |
| 21:49.01 | lmadsen | [TK]D-Fender: I stayed under $300k |
| 21:49.20 | [TK]D-Fender | creativx: He WAS in Mississauga (just outside of Toronto proper). That alone is HUGE. Toronto itself is a true metropolis. |
| 21:49.28 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:49.57 | creativx | [TK]D-Fender: oh well my knowledge of canadian geography limits itself to what ive learned from trailerpark boys |
| 21:50.11 | [TK]D-Fender | lmadsen: unless I was assured of real gains, any kind of property would indeed be a liability for me, and not an investment. |
| 21:50.18 | lmadsen | creativx: don't worry... Mississauga isn't really worth knowing :) |
| 21:50.25 | [TK]D-Fender | lmadsen: now now... |
| 21:50.49 | creativx | [TK]D-Fender: you talk like an investor |
| 21:50.55 | cestchiant | [TK]D-Fender: http://pastebin.com/d59136c58 |
| 21:50.55 | lmadsen | Petrolia -> Oakville -> Mississauga -> Toronto downtown (rent) -> Toronto downtown (buy) |
| 21:50.55 | creativx | a homeowner wouldnt worry about gains |
| 21:51.10 | [TK]D-Fender | creativx: just a person who won't board an already sinking ship :) |
| 21:51.28 | creativx | [TK]D-Fender: theres nothing wrong about being patient :) im in the same ship |
| 21:51.28 | [TK]D-Fender | creativx: Gains is a way of offsetting "losses" :) |
| 21:51.57 | [TK]D-Fender | lmadsen: get to enjoy the nightlife more now? |
| 21:52.12 | lmadsen | [TK]D-Fender: yeppers, way more. Mississauga might as well be in Kitchener |
| 21:52.18 | *** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 21:52.32 | lmadsen | Mississauga is totally not accessible for those of us without cards |
| 21:52.34 | lmadsen | cars* |
| 21:52.42 | cestchiant | [TK]D-Fender: Sometime, fields To and contact are the same, sometimes no :( |
| 21:52.54 | lmadsen | and I figure a condo downtown should be somewhat shielded from a housing market collapse |
| 21:52.55 | [TK]D-Fender | lmadsen: Comparatively, yeah... its super quite. But then again... there's only a half dozen cities in Canada I'd put in the same grouping as TO |
| 21:53.15 | [TK]D-Fender | cestchiant: And we can't explain any of it to you till we see the full packet. |
| 21:53.29 | cestchiant | [TK]D-Fender: http://pastebin.com/d59136c58 |
| 21:53.30 | creativx | hehe |
| 21:53.39 | [TK]D-Fender | quiet* |
| 21:54.17 | creativx | so lmadsen how many m2 you get for 300 k canadians? |
| 21:54.27 | Alowishus | Any idea how to troubleshoot "asterisk.c:2982 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection."?? zaptel modules load fine and ztcfg runs fine and sees the hardware |
| 21:54.42 | [TK]D-Fender | cestchiant: And you are hiding the headers you are referring to. Stop wasting our time. |
| 21:54.54 | lmadsen | creativx: about 180 m^2 |
| 21:54.57 | [TK]D-Fender | lol |
| 21:55.05 | cestchiant | The header?? |
| 21:55.10 | creativx | lmadsen: woah! |
| 21:55.11 | [TK]D-Fender | cestchiant: "To: |
| 21:55.15 | lmadsen | creativx: ya :) |
| 21:55.28 | lmadsen | creativx: although that is about 1/3 of the price of Manhatten :) |
| 21:55.43 | [TK]D-Fender | cestchiant: To: <sip:+012345642@sysun> <- this is who you are CALLING. |
| 21:55.44 | lmadsen | creativx: one bedroom condo, 34th floor |
| 21:55.48 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:55.48 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:55.57 | [TK]D-Fender | cestchiant: Contact: <sip:012345640@xx.xx.xx.xx> <- this is who you ARE |
| 21:56.02 | creativx | lmadsen: yeah my comparison would be bergen, norway |
| 21:56.03 | cestchiant | yes I try to call To: <sip:+012345642@sysun> |
| 21:56.14 | creativx | lmadsen: which isnt even funny to think about |
| 21:56.14 | creativx | heh |
| 21:56.21 | cestchiant | and it's 012345642 who ring.... |
| 21:56.24 | cestchiant | oups |
| 21:56.25 | lmadsen | creativx: oh ya? how much is that place? :) |
| 21:56.30 | cestchiant | 012345640 who ring... |
| 21:57.03 | creativx | lmadsen: well.. for ~100 m2 central place here, 2 bedroom.. that would be around 700k canadian dollars plus minus.. |
| 21:57.20 | lmadsen | creativx: jeebuz |
| 21:57.32 | creativx | lmadsen: i'm glad we have the oil a couple of more years |
| 21:57.36 | lmadsen | creativx: how the heck do you get a two bedroom with 100m2? |
| 21:57.38 | creativx | after that we better figure out some smart shit! |
| 21:57.55 | lmadsen | creativx: google said 600 sqft == 182 m2 |
| 21:57.59 | [TK]D-Fender | creativx: Bergen Population (2008) 247,746. Toronto, Population (2006) 2,503,281 <- SLIGHT difference in scale ;) |
| 21:58.02 | creativx | lmadsen: you rarely see >100m2 condos here |
| 21:58.12 | creativx | [TK]D-Fender: yep :) slight.. |
| 21:58.12 | lmadsen | creativx: really?! that is tiny... |
| 21:58.39 | creativx | lmadsen: wel someone has to clean all those 600 sqf! |
| 21:58.40 | creativx | hehe |
| 21:58.40 | *** join/#asterisk AsteriskRo (n=rosadesa@190.36.184.217) |
| 21:58.45 | lmadsen | haha |
| 21:59.08 | creativx | and unless you live in chile, where the maid follows the condo, its gonna be you! |
| 21:59.18 | [TK]D-Fender | lmadsen: And for reference : Mississauga Population (2008) 704,246 (6th largest in Canada) |
| 21:59.18 | creativx | :-) |
| 21:59.30 | [TK]D-Fender | lmadsen: for that "dead" town you left :p |
| 21:59.33 | lmadsen | [TK]D-Fender: and spread out like LA :) |
| 21:59.34 | creativx | [TK]D-Fender: norway never had many people.. just a lot of oil and money :) |
| 21:59.47 | creativx | and booze.. and hot women. |
| 21:59.48 | lmadsen | [TK]D-Fender: Mississauga has a LOT of people -- nothing to do in it though, and you HAVE to drive everywhere |
| 21:59.55 | [TK]D-Fender | creativx: And hot blond chicks. Export dammit, EXPORT! |
| 21:59.57 | lmadsen | creativx: count me in! |
| 22:00.05 | creativx | heheh |
| 22:00.21 | creativx | and to rub it in ofcourse.. we are the closest neighbour to sweden.. where there are MORE blonde chicks and even cheaper alcohol.. |
| 22:00.22 | creativx | :D |
| 22:00.25 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 22:00.47 | lmadsen | ya... but at least we have sunlight in the winter :) |
| 22:00.58 | creativx | hehe yeah |
| 22:01.03 | creativx | them timezones |
| 22:01.22 | lmadsen | well... not timezones... just lower latitude :) |
| 22:02.19 | creativx | hehe |
| 22:02.20 | creativx | yeah |
| 22:02.20 | [TK]D-Fender | All about the 'tude y0! |
| 22:02.37 | cestchiant | [TK]D-Fender: I don't understand in my fiel "TO" I have +12345642@... So I try to call via my phone 12345642... And in my contact field I have 12345640 ... And it's this phone that is contacted... Why it's not the "good" phone? :( |
| 22:02.39 | AsteriskRo | hi all, any help on freepbx topic? nobody on freepbx channel :( |
| 22:03.19 | creativx | AsteriskRo: nietski |
| 22:03.28 | [TK]D-Fender | cestchiant: the "contact" is YOUR NUMBER AT YOUR ITSP |
| 22:04.01 | cestchiant | yes I'm ok but i have somes number 12345640 -> 12345650 |
| 22:04.06 | [TK]D-Fender | AsteriskRo: No, and not our problem. |
| 22:04.22 | cestchiant | so when i call 12345642 sometime it's forwarded to other DID... |
| 22:04.26 | [TK]D-Fender | cestchiant: Then you should try to speicify which one you want to appear as. |
| 22:04.37 | Veggen | creativx: you live in Bergen? |
| 22:04.43 | creativx | Veggen: last time i checked, yes. |
| 22:04.48 | AsteriskRo | thanks, TKD-Fender...you are always so nice |
| 22:04.51 | Veggen | so do I, in fact. |
| 22:05.08 | creativx | not bad |
| 22:05.19 | creativx | feel free to join me in my whoring of our motherland |
| 22:05.20 | creativx | hehe |
| 22:05.25 | [TK]D-Fender | AsteriskRo: And you learn your lessons so well... |
| 22:05.48 | AsteriskRo | what do you do here???? |
| 22:06.01 | AsteriskRo | always with bad answers.... |
| 22:06.07 | [TK]D-Fender | AsteriskRo: Help people who are using Asterisk. |
| 22:06.15 | AsteriskRo | so? |
| 22:06.20 | AsteriskRo | I'm using asterisk too |
| 22:06.27 | [TK]D-Fender | AsteriskRo: No, you're not. |
| 22:06.30 | AsteriskRo | you never are willing to help me |
| 22:06.34 | creativx | you are using the bastard child |
| 22:06.40 | [TK]D-Fender | creativx: not even. |
| 22:06.51 | AsteriskRo | i've make asterisk questions and no answers from you |
| 22:06.54 | [TK]D-Fender | creativx: Need to look at it from the appropriate point of scale. |
| 22:07.05 | [TK]D-Fender | AsteriskRo: Perhaps I didn't have the answers. |
| 22:07.07 | creativx | in lack of a worse definition, [TK]D-Fender. |
| 22:07.07 | creativx | hehe |
| 22:07.18 | [TK]D-Fender | creativx: Lol, good comeback :) |
| 22:07.22 | lmadsen | AsteriskRo: we can't answer trixbox questions here because it does funny things to the configuration files |
| 22:07.25 | [TK]D-Fender | creativx++ |
| 22:07.52 | [TK]D-Fender | AsteriskRo: For reference, FreePBX is using Asterisk. You are using FreePBX. |
| 22:08.29 | AsteriskRo | i'm not talking about my question today, i'm talking about other times... |
| 22:08.31 | [TK]D-Fender | AsteriskRo: Since it controls everything, it isn't "you" that needs help, and we are not hear to dissect how it works. |
| 22:08.53 | [TK]D-Fender | AsteriskRo: and as I said, if I didn't answer, perhaps I didn't have an answer to you question at the time. |
| 22:09.17 | AsteriskRo | baaaaa |
| 22:09.45 | creativx | would you like some cheese with that whine, AsteriskRo? |
| 22:10.17 | Veggen | AsteriskRo: If you come and ask for free help on free channels, you *should* be prepared to do some work yourself. |
| 22:10.25 | Veggen | A lot of work yourself, actually. |
| 22:11.14 | AsteriskRo | i know, but I "is not our problem" is not the answer i expected |
| 22:11.41 | creativx | feel free to deploy asterisk |
| 22:11.46 | [TK]D-Fender | AsteriskRo: Funny, thats the stock answer to "can I ask a question about FreePBX, nobody is answering there" |
| 22:11.52 | creativx | and encounter lots of questions, problems, whatever.. |
| 22:11.56 | hardwire | anybody here use users.conf religiously? I'm hoping to set my voicemail context from users.conf for a specific template a set of users are using |
| 22:12.23 | [TK]D-Fender | AsteriskRo: As ManxPower most recently stated "#asterisk is not 2nd level support for #freepbx". Very apt. |
| 22:12.43 | [TK]D-Fender | hardwire: Yes, pray for it... you'll need all the help you can get :) |
| 22:12.52 | hardwire | [TK]D-Fender: damnit |
| 22:12.56 | [TK]D-Fender | hardwire: users.conf is a flaming pile of shit :) |
| 22:13.01 | hardwire | isn't there some sort of ASTDB love going on there? |
| 22:13.15 | hardwire | [TK]D-Fender: I didn't want to hear that. |
| 22:13.15 | [TK]D-Fender | hardwire: Shouldn't be |
| 22:13.20 | hardwire | holds fingers up to ears. |
| 22:13.30 | [TK]D-Fender | hardwire: LA LA LA CAN'T HEAR YOu!! |
| 22:13.37 | hardwire | damnit |
| 22:13.45 | hardwire | goes back to manually entering in tons of stuff |
| 22:13.47 | *** join/#asterisk remibemol (n=remibemo@m44.net81-66-213.noos.fr) |
| 22:13.58 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 22:14.04 | lmadsen | hardwire: there are templates you can build in all configuration files |
| 22:14.12 | eXistenZ | [TK]D-Fender, What might be the problem if "Last Called PSTN Number" is always empty? |
| 22:14.13 | hardwire | I know, thats fine. |
| 22:14.21 | lmadsen | hardwire: what is the question I guess.... ? |
| 22:14.39 | [TK]D-Fender | eXistenZ: perhaps you are calling your PSTN channel and not GIVING it a number. |
| 22:15.05 | hardwire | lmadsen: instead of setting userscontext globally in voicemail.conf I'd like to set it IN users.conf as "voicemailcontext" or something |
| 22:15.19 | hardwire | it's no big deal |
| 22:15.20 | [TK]D-Fender | hardwire: thats "vmcontext" IIRC |
| 22:15.25 | hardwire | lies |
| 22:15.56 | AsteriskRo | i use asterisk itself for many purposes now, i have servers with conference, ivr, b2bua, etc... all setup by myself, but now i want to setup a pbx without having to deal with interface for user names, dialplans, etc... that's why i'm asking about it |
| 22:16.32 | hardwire | [TK]D-Fender: nodice |
| 22:16.41 | [TK]D-Fender | hardwire: guess not... |
| 22:16.52 | hardwire | [TK]D-Fender: even YOU thought it should be there. |
| 22:17.05 | lmadsen | AsteriskRo: asterisk-gui is easier to work around than trixbox |
| 22:17.08 | lmadsen | imho |
| 22:17.37 | [TK]D-Fender | hardwire: I ttok a small jab thinking they might have done SOMETHING right in there... I guess my "flaming pile of shit" comment stands in all its glory :p |
| 22:17.57 | AsteriskRo | but there are a lot of things i had to configure outside the interface, and after that the asterisk-gui doesn't parse the files i changed |
| 22:18.08 | [TK]D-Fender | AsteriskRo: And its not getting any more supported in here the more you ask. |
| 22:18.20 | AsteriskRo | i know... |
| 22:18.41 | AsteriskRo | only the reason of my question... |
| 22:20.25 | eXistenZ | [TK]D-Fender, calling it and not giving it a number? |
| 22:20.37 | [TK]D-Fender | eXistenZ: Yes. |
| 22:20.45 | eXistenZ | [TK]D-Fender, How can I give it a number? |
| 22:20.59 | [TK]D-Fender | eXistenZ: go look at how you are placing calls to it |
| 22:22.47 | eXistenZ | [TK]D-Fender, it says: "Connected to PSTN" and "Line 1 Fallback To PSTN Gateway", but doesn't display the number dialed |
| 22:23.39 | *** join/#asterisk kippi (n=kippi@cpc3-ware2-0-0-cust165.lutn.cable.ntl.com) |
| 22:23.42 | kippi | hey |
| 22:23.49 | [TK]D-Fender | eXistenZ: go look at how you are placing calls to it <-- |
| 22:24.14 | eXistenZ | [TK]D-Fender, How exactly? |
| 22:24.23 | [TK]D-Fender | eXistenZ: in your DIALPLAN. |
| 22:24.32 | eXistenZ | [TK]D-Fender, aha |
| 22:24.33 | Veggen | Are people using ruby-agi or ragi? |
| 22:24.53 | kippi | I am runing version 1.4.20 verything is going well apart from the voicemail, the quailty is very poor, how can I track down the problem, I have read people have had this problem when not using any hardware, I have a diguim IDSN card |
| 22:26.00 | [TK]D-Fender | kippi: descrip a complete scenario including protocols & hardware incolved for a "low quality call". |
| 22:26.54 | [TK]D-Fender | Veggen: probably only a handful of people using Ruby for AGI |
| 22:27.01 | kippi | is there any dug I can run that will help?? |
| 22:27.15 | [TK]D-Fender | Veggen: most usu PHP / Perl / Python |
| 22:27.45 | *** part/#asterisk jjg (n=jjg@12.40.200.74) |
| 22:27.48 | [TK]D-Fender | kippi: that LOOKED like english... but I can't understand what the hell you're saying. |
| 22:27.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:27.56 | Veggen | Fender: ok. I'm sort of a bit into ruby lately. But if it's not considered stable and/or have less features, I'll reconsider. |
| 22:29.10 | [TK]D-Fender | Veggen: Oh, I don't know that either won't work fine. Just staing the the likelyhood of finding people who use it here is really low. |
| 22:29.14 | *** join/#asterisk kippi (n=kippi@cpc3-ware2-0-0-cust165.lutn.cable.ntl.com) |
| 22:29.16 | [TK]D-Fender | stating* |
| 22:30.05 | Veggen | Fender: ok, I'll have a go at ragi, then :) I was just wondering *which* ruby for asterisk tool that was considered "best" nowadays? |
| 22:30.25 | Veggen | there's both ruby-agi and ragi, and also RAI (ruby asterisk integration) I see. |
| 22:30.30 | [TK]D-Fender | Veggen: So few use ANY of it, now you're looking for someone who has used BOTH? |
| 22:30.39 | Veggen | :P |
| 22:31.07 | Veggen | I take it that if you're using one, you're bound to have looked at and/or noticed the others. oh well :) |
| 22:35.21 | [TK]D-Fender | AsteriskRo: ... is? |
| 22:37.44 | kippi | from the asterisk cli is there away to see what codec it will be using for internal calls? |
| 22:39.50 | Veggen | kippi: "sip show user XX" ? |
| 22:40.31 | [TK]D-Fender | kippi: WILL be using? * isn't psychic, so unless you only allow ONE codec, who knows. |
| 22:41.31 | [TK]D-Fender | kippi: and "it" is a very poor choice of terms. Individual peers set their own permissible codecs, and that doesn't say which amongst multiple options will actually get used. |
| 22:41.40 | asdx | how do you connect to the manager interface in asterisk? |
| 22:41.54 | asdx | i already created a user, etc |
| 22:42.05 | jjshoe | asterisk -r |
| 22:42.09 | asdx | done |
| 22:43.03 | [TK]D-Fender | asdx: Go read Chapter 10 |
| 22:43.14 | [TK]D-Fender | jjshoe: No. |
| 22:43.25 | asdx | asterisk -r is the cli |
| 22:44.05 | jjshoe | [TK]D-Fender correct me if I'm wrong, but doesn't it just make a socket connection back to itself? |
| 22:44.16 | [TK]D-Fender | jjshoe: AMI not CLI |
| 22:44.20 | jjshoe | [TK]D-Fender correct me if I'm wrong, but doesn't it just make a socket connection back to itself? |
| 22:44.26 | asdx | lol |
| 22:44.35 | [TK]D-Fender | jjshoe: Apples & orangutans. |
| 22:45.02 | jjshoe | I wasn't arguing they where different, just asking a question. but, nevermind. |
| 22:45.15 | jjshoe | beer time anyways & |
| 22:45.26 | [TK]D-Fender | jjshoe: "it" just make a socket connection back to "itself"? Huh? |
| 22:45.38 | asdx | lol |
| 22:45.46 | asdx | i'm going to buy coke |
| 22:46.02 | [TK]D-Fender | asdx: You don't need drugs... or at least THAT kind... |
| 22:46.10 | asdx | [TK]D-Fender: coca cola |
| 22:46.15 | asdx | the drink ;) |
| 22:46.40 | asdx | coca cola & dark chocolate & pringles ftw |
| 22:47.56 | asdx | hm not sure |
| 22:48.00 | asdx | i'd better stay more and learn ami |
| 22:49.34 | asdx | ldap and openvpn would be nice to learn too |
| 22:50.06 | Veggen | openvpn is nice. Not so difficult either. |
| 22:51.37 | Veggen | I run sip over openvpn to my Asterisk, from my Linux tablet ;) (when I'm out in the world on a wlan somewhere) |
| 22:52.09 | asdx | cool |
| 22:53.52 | *** join/#asterisk afg_ch (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 22:57.14 | kippi | Failed to write frame |
| 22:57.16 | kippi | opps |
| 22:57.24 | eXistenZ | [TK]D-Fender, now it works :) |
| 22:57.33 | afg_ch | Anyone out there who compiled Zaptel 1.4.11 successfully on CentOS 5.2 (Kernel 2.6.18-92.1.6.el5) ? |
| 22:58.20 | [TK]D-Fender | ~centos52bug |
| 22:58.20 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 22:58.21 | eXistenZ | [TK]D-Fender, I changed the PSTN Answer Delay to 10, and now it receives the caller id |
| 22:58.22 | [TK]D-Fender | ^^^ |
| 22:58.49 | kippi | ok, looking more into my voicemail problem, and I cam getting warnings saying "failed to write frame" just getting more info on the codec etc, but does this error point me in the right dirrection, it sounds like the voicemail lady is holding her nose whist talking |
| 22:58.50 | [TK]D-Fender | eXistenZ: Like I said... "CID by default" is NOT the case. |
| 22:58.56 | eXistenZ | [TK]D-Fender, the weird thing is now, that when I enable "Block ANC Setting" under User 1, all calls are blocked. |
| 22:59.14 | ManxPower | kippi: failed to write frame is usually "caller hung up" |
| 22:59.16 | eXistenZ | although it identifies the caller id |
| 22:59.36 | [TK]D-Fender | eXistenZ: Stop blockingon the ATA level and just do it in * |
| 22:59.41 | ManxPower | kippi: are you using SIPura ATAs? |
| 23:00.46 | kippi | ManxPower: one of the handsets is, but tired it on other handsets to, softphone, grandstream, cisco and all get the same, out going calls are clear as could be |
| 23:03.09 | eXistenZ | [TK]D-Fender, how should the Vertical Service Activation Codes of spa3102 be dialed? in the first dial tone? |
| 23:03.39 | kippi | card i am using is a TE110p t1/e1 |
| 23:04.10 | [TK]D-Fender | eXistenZ: Yes. |
| 23:04.52 | eXistenZ | [TK]D-Fender, am I supposed to hear a kind of confirmation tone? when I dial *77 for example, it doesn't activate the corresponding functionality |
| 23:05.27 | [TK]D-Fender | eXistenZ: Go check the forms for it, and I don't recall seeing any valuable backup |
| 23:06.08 | [TK]D-Fender | forums* |
| 23:08.49 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 23:10.28 | kippi | can using ulaw codec |
| 23:10.32 | kippi | and |
| 23:10.36 | kippi | and |
| 23:16.37 | afg_ch | jbot: in which path would I find the MENUCONFIG file you refer to? I couldn't locate it. |
| 23:16.58 | kippi | i have read on some forums etc it could be to do with the timmers? but as I have hardware this should all be ok? |
| 23:17.30 | [TK]D-Fender | afg_ch: Don't talk to the bot... its too late for that kind of hunour |
| 23:20.23 | afg_ch | [TK]D-Fender: oh god! I didn't figure some users are bots |
| 23:20.44 | [TK]D-Fender | afg_ch: --> jbot |
| 23:20.45 | afg_ch | [TK]D-Fender: Would you hang in instead? |
| 23:21.01 | [TK]D-Fender | afg_ch: what "menuconfig" file are you looking for? |
| 23:21.17 | [TK]D-Fender | afg_ch: Oh, thats the message *I* triggered |
| 23:21.29 | [TK]D-Fender | afg_ch: do "make menuconfig" and unselec XPP <-- |
| 23:23.56 | kippi | anyone got anyideas on my voicemail problem, really not sure where to look, |
| 23:26.25 | eXistenZ | [TK]D-Fender, if the caller id is generated only after the call is accepted, what might be the problem? |
| 23:27.39 | [TK]D-Fender | eXistenZ: Statement doesn't make sense. |
| 23:28.17 | eXistenZ | [TK]D-Fender, I have "PSTN Peer Number:" (empty) and "PSTN State: Ringing" |
| 23:28.32 | afg_ch | [TK]D-Fender: Just feeling lucky I am not hassling with a production system with pressure to have a success soon. Guess what "make menuconfig" is telling me? It's telling me: *** Install ncurses to use the menu interface! *** but wait... |
| 23:28.35 | eXistenZ | [TK]D-Fender, I see the number only after the conversation is hooked up |
| 23:28.45 | [TK]D-Fender | eXistenZ: CID is typically received betweent he 1st and 2nd rings |
| 23:29.04 | eXistenZ | weird |
| 23:29.12 | [TK]D-Fender | eXistenZ: and maybe thats jsut the way that page displays things. |
| 23:29.18 | eXistenZ | maybe |
| 23:29.35 | afg_ch | [TK]D-Fender: ... guess what yum is telling me! It's telling me: "Package ncurses - 5.5-24.20060715.x86_64 is already installed" |
| 23:29.49 | [TK]D-Fender | afg_ch: then it is. |
| 23:29.55 | eXistenZ | [TK]D-Fender, when I activate the "Block ANC Setting", all calls are blocked =/ |
| 23:30.15 | [TK]D-Fender | eXistenZ: Go read your admin guide. |
| 23:32.03 | afg_ch | [TK]D-Fender: I think I'll have a go with the diff patch and don't update the Kernel to 2.6.18 on any machine for some good time. |
| 23:34.34 | [TK]D-Fender | afg_ch: just change 1 little option in menuconfig. This isn't Raw-Cat Science.... |
| 23:34.43 | *** part/#asterisk Alowishus (n=jpenix@adsl-69-109-156-138.dsl.sndg02.pacbell.net) |
| 23:37.11 | *** join/#asterisk asdx (n=diego@adsl-133-186.click.com.py) |
| 23:40.04 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582050.dsl.bell.ca) |
| 23:43.06 | kippi | will * be using ztdummy or my card for the voicemail part? |
| 23:43.09 | afg_ch | [TK]D-Fender: maybe I was indirect. "make menuconfig" failed telling me to install ncurses. |
| 23:44.03 | *** join/#asterisk Superbartt (n=bart@82-168-44-79.ip.telfort.nl) |
| 23:44.07 | [TK]D-Fender | afg_ch: And you've clearly not read *'s requirements. |
| 23:44.32 | [TK]D-Fender | kippi: No |
| 23:46.12 | afg_ch | [TK]D-Fender: Not sure, until now I got along like this. Do you have an idea why I'm suggested to install ncurses when it IS INSTALLED already. Now you have been indirect. |
| 23:46.46 | [TK]D-Fender | afg "ncurses-devel" <----- |
| 23:47.01 | kippi | [TK]D-Fender the recorded call sounds fine, it is just the menu system that sounds like she is holding her nose |
| 23:47.03 | [TK]D-Fender | afg_ch: you need the devel library |
| 23:47.24 | [TK]D-Fender | kippi: describe the complete call. I've asked you this already. |
| 23:48.57 | afg_ch | [TK]D-Fender: Be assured I have checked that and yum was telling me: Package ncurses-devel - 5.5-24.20060715.x86_64 is already installed |
| 23:49.27 | [TK]D-Fender | afg_ch: then trash your extracted folder, reectract it, and redo "./configure" |
| 23:49.29 | kippi | [TK]D-Fender how do you mean, the codec is alaw |
| 23:49.52 | [TK]D-Fender | kippi: I said describe the entire call. EVERY interface codec, networking, etc involved. |
| 23:50.42 | afg_ch | [TK]D-Fender: However, full credits go to YOU since your hint to the bugtracker entry has now made me apply the patch with a breeze and the compile went its way with no more hiccups. COOOL! |
| 23:51.02 | eXistenZ | [TK]D-Fender, so you say that it is possible to block anonymous callers with asterisk? |
| 23:51.17 | [TK]D-Fender | eXistenZ: Yeah. Its called DON'T ANSWER |
| 23:51.38 | eXistenZ | [TK]D-Fender, I see you are always on high torques :) |
| 23:53.40 | kippi | ok, the handset is set to use alaw, networking is a 100Mb to the handset and 1GB to the switch, asterisk hardware is te110p setup for UK, dialing 1050 which goes to voicemailman |
| 23:54.09 | eXistenZ | [TK]D-Fender, Anyways, I will download AsteriskNOW and try it out, it might help. Thanks for your help :) |
| 23:54.10 | [TK]D-Fender | kippi: Describe the call that left the VM. |
| 23:54.19 | eXistenZ | I should hit the sack |
| 23:54.31 | afg_ch | [TK]D-Fender: what do you understand by "trash the extracted folder"? Do you mean "make clean", rm -r -f "zaptel-1.4.11", or "format c:" ;-) |
| 23:54.56 | [TK]D-Fender | afg_ch: rm -r -f "zaptel-1.4.11" |
| 23:55.14 | kippi | call on voicemail is from the IDSN which plays back fine, its the menu where she sounds like she is under water or holding her noise |
| 23:55.55 | ManxPower | kippi: sounds to me like an RTP packetization mismatch, but I won't know until you answer my question |
| 23:56.17 | ManxPower | ah, I see you did. |
| 23:56.35 | *** join/#asterisk juantelez (i=be12619e@gateway/web/ajax/mibbit.com/x-16a1d81e2637b2f5) |
| 23:56.45 | ManxPower | kippi: I suspect the RECORDING is bad. Was it recorded from a SIPura, if so you need to change the RTP packet size on the SIPura to .20, from .30 like it is now. |
| 23:57.34 | afg_ch | [TK]D-Fender: well, I could do that just for the sake of testing, but I think I push the imaginary button "close case, works for me now". Any objections to that? Where to send the check for the solution or does a THANK YOU! satisfy your needs? |
| 23:57.34 | kippi | its the asterisk lady who says press 1 for this, 2 for that, its hr that sound under water etc |
| 23:58.09 | [TK]D-Fender | afg_ch: Not a need. But you're welcome. |
| 23:58.49 | *** part/#asterisk juantelez (i=be12619e@gateway/web/ajax/mibbit.com/x-16a1d81e2637b2f5) |
| 23:59.00 | kippi | its sounds the same from 3 differnet types of handsets |
| 23:59.01 | [TK]D-Fender | kippi: the default sounds are in GSM and if you compiled * with certain late GCC's (4.2, maybe 4.1) its had some real issues that would make GSM (specifically) sound like shit. |
| 23:59.15 | [TK]D-Fender | kippi: So far that looks like a viable culprit |
