| 00:06.52 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 00:11.56 | *** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net) |
| 00:12.08 | tzafrir_home | aninoSAdilim, asterisk can send any tone you'd like, if you put the right sound file... |
| 00:23.59 | tzafrir_home | coppice, why do you use ZT_IOMUX_NOWAIT in sliptest ? |
| 00:24.23 | tzafrir_home | I removed it, and suddenly I wasn't consuming 100% CPU |
| 00:28.48 | tzafrir_home | It's also very sensitive to noises. So it seems that I must use such a resistor instead of a real phone |
| 00:39.21 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-decb98b12d5b6ed2) |
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| 01:00.10 | De_Mon | tzafrir you around? |
| 01:00.11 | aninoSAdilim | tzafrir_home, i know but thanks anyway. rxfax is my real solution |
| 01:00.33 | tzafrir_home | De_Mon, yes |
| 01:00.46 | De_Mon | was adding --with-imap to the asterisk build args intentional on dfsg-3? |
| 01:01.06 | De_Mon | err, chaning the voicemail storage to imap |
| 01:01.14 | tzafrir_home | De_Mon, you have problems building it on Etch? |
| 01:01.42 | De_Mon | no, I just installed the package and it broke the previously working install |
| 01:01.44 | tzafrir_home | It means changing? Not just adding an option? |
| 01:02.00 | De_Mon | yep, its filesystem OR odbc OR imap |
| 01:02.14 | ManxPower | well don't use packages |
| 01:02.15 | De_Mon | thats what I learned today ;) |
| 01:02.48 | De_Mon | ManxPower I just got around to recompiling, but as it wasn't mentioned in the debian changelog... |
| 01:03.17 | De_Mon | that, and I haven't bothered to see if theres a bugreport about it |
| 01:05.28 | tzafrir_home | Can you please reportbug on this? |
| 01:06.19 | tzafrir_home | De_Mon, to save you time on package rebuilding, just extract the package and run debian/backports/etch.xorcom |
| 01:06.37 | tzafrir_home | It shows I never really liked that feature ... |
| 01:07.52 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:08.24 | De_Mon | ooo? |
| 01:08.32 | *** join/#asterisk Cresl1n (n=matt@wsip-70-184-124-51.ph.ph.cox.net) |
| 01:08.32 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 01:11.58 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 01:17.37 | De_Mon | oh that reminds me, I need to try getting snmp working again in * |
| 01:32.48 | *** part/#asterisk Cresl1n (n=matt@wsip-70-184-124-51.ph.ph.cox.net) |
| 01:46.12 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 01:46.22 | hrmphh | how can i get incoming calls to go to the 's' extension |
| 01:47.32 | hrmphh | for some reason those coming over the PRI are trying to send to the full 10 digit |
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| 01:54.34 | tzafrir_home | hrmphh, what's wrong with that? |
| 01:54.56 | tzafrir_home | hrmphh, anyway, if you won't have that number in that context it will fall through to s |
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| 01:56.12 | kiscokid | Anybody else seen wierdness in AgentCallbackLogin()? |
| 01:59.05 | CCFL_Man2 | i'm ircing from my shell through a serial port on my lantronix box which telnets into my freebsd server |
| 01:59.25 | hrmphh | no it didnt fall through |
| 01:59.27 | hrmphh | that was the problem |
| 02:00.51 | hrmphh | <PROTECTED> |
| 02:01.04 | hrmphh | where XXXXXXXXXX is our main# |
| 02:01.11 | hrmphh | and YYYYYYYYYY is the number i called from |
| 02:02.00 | *** join/#asterisk gankhuu (n=luken@d104.digis.net) |
| 02:03.34 | De_Mon | hrmphh exten => _.,1,Goto(s) |
| 02:04.41 | De_Mon | even better would be _NXXNXXXXXX,1,Goto(s,1) (You need exten and prio sorry) |
| 02:07.23 | hrmphh | ael is fun |
| 02:10.59 | hrmphh | why the two n's in there? |
| 02:11.54 | gankhuu | because those two numbers do not start with 1 in US |
| 02:12.07 | De_Mon | because thats the pattern for numbers in ... what he said |
| 02:12.31 | De_Mon | international numbers arnt 10 digits afaik just us ones |
| 02:13.41 | *** join/#asterisk Rants (i=Rants@ool-182f4d05.dyn.optonline.net) |
| 02:15.39 | hrmphh | coo |
| 02:15.46 | hrmphh | how can i goto the last 4 digits from that? |
| 02:16.37 | Rants | Looking for some help. I'm im connecting to the manager and geting it to dial a number, but I want it to use the same rules as if it was dial from an extension (pick a trunk) and all. What is the command? |
| 02:16.58 | De_Mon | voip-info.org |
| 02:18.13 | Rants | any particular area de_mon? |
| 02:18.26 | De_Mon | variables |
| 02:18.34 | De_Mon | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 02:18.58 | *** join/#asterisk deb_user (n=debian_l@70-59-107-53.albq.qwest.net) |
| 02:19.22 | Rants | thanks |
| 02:19.49 | De_Mon | if you still cant find the answer leme know I'll look at the page |
| 02:20.00 | deb_user | what does this mean: |
| 02:20.00 | deb_user | Auto fallthrough, channel 'IAX2/xxx' status is 'UNKNOWN' |
| 02:20.31 | deb_user | I changed my dialplan to have a catchall...like so: exten => _X.,1,Answer() |
| 02:20.58 | deb_user | now when it cycles through the catchall on an IVR, and you actually have to enter the extension number, it gives that weird error message and hangs up |
| 02:21.19 | De_Mon | deb_user you need a WaitExten() in there |
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| 02:22.28 | deb_user | De_Mon: is that a new feature? I just upgraded from 1.4.0 to 1.4.11 a few minutes ago... |
| 02:22.48 | De_Mon | no.. it's been around as I've known |
| 02:22.56 | deb_user | strange... |
| 02:22.58 | De_Mon | as long as I've known |
| 02:23.10 | deb_user | maybe with the upgrade my old faulty dialplan became obsolete |
| 02:23.13 | deb_user | it used to work before |
| 02:23.18 | deb_user | I just added it to extensions.conf |
| 02:23.23 | deb_user | let me test it; thanks |
| 02:23.39 | De_Mon | the autofallthrough behavior may have changed, but yeah not having a wait is a bad practice |
| 02:23.42 | riddlebox | can someone tell me how I can get asterisk to see when a calling party hangs up and not keep ringing my phones? I checked my line coming in it detects it? |
| 02:23.44 | rozelli | guys, Im trying to start asterisk but I am having this error: Sep 28 23:19:27 WARNING[10229]: manager.c:1750 init_manager: Unable to bind socket: Address already in use |
| 02:24.02 | *** join/#asterisk heartones (n=root@196.218.146.98) |
| 02:24.04 | De_Mon | deb_user search voip-info.org for autofallthrough in extensions.conf |
| 02:24.07 | riddlebox | rozelli, try sudo asterisk -r |
| 02:24.33 | rozelli | riddlebox unable to connect to remote asterisk |
| 02:24.40 | CCFL_Man2 | shit |
| 02:24.47 | deb_user | De-Mon: it works now |
| 02:24.52 | De_Mon | CCFL_Man2 I'll wait till I'm in the bathroom |
| 02:24.57 | deb_user | thanks, i think the autfallthrough behavior has changed |
| 02:24.59 | CCFL_Man2 | i put a bid in for 51A lamps, i need 51B lamps |
| 02:25.10 | De_Mon | deb_user Yup, http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+waitexten |
| 02:25.23 | De_Mon | its the very first NOTE |
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| 02:25.37 | deb_user | i suppose the upgrades force us to use best practices, which is a good thing, and which is why i'm upgrading on a friday night instead of a weekday |
| 02:25.53 | deb_user | there's always a few wrinkles to iron out after an upgrade |
| 02:26.49 | rozelli | riddlebox any idea? |
| 02:27.13 | De_Mon | deb_user there should be an upgrade doc of some sort that tells you about things like that |
| 02:27.49 | De_Mon | how you set callerid changed too |
| 02:28.39 | heartones | any one has installed the sangoma wanpipe with asterik here some I am getting 4 channels configured but ZT_CHANCONFIG failed on channel 1 I think it's related to my wanpipe configurationn but I couldn |
| 02:28.46 | heartones | t thing of any thing |
| 02:28.59 | heartones | can any one troubleshoot this |
| 02:29.52 | tzafrir_home | heartones, sounds like an error from ztcfg . So your /etc/zaptel.conf does not match the reality (e.g: /proc/zaptel/* ) |
| 02:30.17 | tzafrir_home | pastebin /etc/zaptel.conf and the output of cat /proc/zaptel/* |
| 02:30.33 | heartones | but when I do wanrouter start it fails |
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| 02:30.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:30.56 | heartones | complaining about some modules I guess |
| 02:31.38 | heartones | ok let me check that |
| 02:31.49 | hrmphh | ok q |
| 02:32.02 | hrmphh | lets say i have _XXXX => { } |
| 02:32.07 | hrmphh | how can i make it dial SIP/XXXX |
| 02:32.08 | hrmphh | ? |
| 02:32.39 | deb_user | DeMon: i don't think CID changed from 1.4.0 to 1.4.11 |
| 02:33.06 | deb_user | but, yes, I did have to deal with that when I configured Hoodahek on my 1.4 system, the documentation didn't account for the changes in callerid syntax |
| 02:33.33 | heartones | I am not getting any thing out of cat /proc/zaptel/* |
| 02:33.59 | heartones | I guess it's empty |
| 02:34.08 | heartones | what suppose to be there |
| 02:35.01 | De_Mon | deb_user uh, the autofallthrough didn't change from 1.4.0 to 1.4.11 either I don't believe |
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| 02:35.55 | De_Mon | tzafrir_home since we can't compile app_voicemail with both and, maybe separate packages for each ;) |
| 02:36.29 | tzafrir_home | Long ago I have engeneered separate voicemail.conf with pgsql support |
| 02:37.08 | De_Mon | odbc didn't cut it? |
| 02:37.12 | tzafrir_home | The makefiles have changed since, though |
| 02:38.33 | deb_user | De_Mon: that's strange though...because the system was working just fine before the upgrade, and I'm SURE I didn't have a WaitExten() in the dialplan, I'm positive I didn't |
| 02:38.42 | De_Mon | I'm talking about a package that contains app_voicemail.so compiled for filestorage, one compiled for odbc, and one compiled for imap that replace eachother |
| 02:39.01 | deb_user | De_Mon: maybe that was supposed to be in 1.4.0 but it got fixed in subsequent versions |
| 02:39.28 | De_Mon | deb_user in 1.2 the default autofallthrough setting was 'no' in 1.4 that setting defaults to 'yes' |
| 02:39.32 | *** join/#asterisk jmacz (n=jmacz@190.24.103.32) |
| 02:39.56 | De_Mon | it has been thant way for all 1.4 releases, maybe your dialplan setting for that got changed during the upgrade |
| 02:41.55 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 02:43.28 | ThoMe | is it posibile when i use "ael" if, elseif , and, or ? |
| 02:44.58 | ThoMe | ah "else if" |
| 02:45.09 | De_Mon | http://www.voip-info.org/wiki/view/AEL+Example+Snippets |
| 02:46.20 | *** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home) |
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| 03:23.27 | ThoMe | De_Mon: u know queues? |
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| 03:49.31 | ThoMe | hello? |
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| 04:09.57 | CCFL_Man2 | i love fxs ports] |
| 04:11.21 | WilliamK | CCFL, I love faxing over Sipura ATA's |
| 04:11.54 | WilliamK | I finally got something that worked allowing a modem to dial out to refill the postage on a pitney bowes postage meter today |
| 04:12.55 | ThoMe | servetux*CLI> show channels |
| 04:12.55 | ThoMe | Channel Location State Application(Data) |
| 04:12.55 | ThoMe | 0 active channels |
| 04:12.55 | ThoMe | 0 active calls |
| 04:13.05 | ThoMe | how i can get the variable for 0 active calls ?? |
| 04:13.12 | ThoMe | for 0 ? is is posible? |
| 04:16.42 | CCFL_Man2 | WilliamK: they are shit |
| 04:16.49 | CCFL_Man2 | well, not shit |
| 04:16.57 | CCFL_Man2 | but a channel bank is better |
| 04:17.10 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 04:17.24 | destructure | anybody here worked with ani2, or the sip equivalent? |
| 04:26.23 | *** join/#asterisk techie (n=techie@adsl-68-127-127-133.dsl.frsn02.pacbell.net) |
| 04:26.45 | ThoMe | hello? |
| 04:26.49 | ThoMe | CCFL_Man2: can u help me? |
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| 04:32.55 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 04:38.38 | Qwell_ | ThoMe: Haven't I asked you multiple times not to ctcp ping the channel? |
| 04:38.48 | ThoMe | Qwell_: sorry. error :-( |
| 04:39.29 | ThoMe | Qwell_: in gotoif, is this: => allowed? |
| 04:39.46 | CCFL_Man2 | i'm getting tired |
| 04:39.47 | ThoMe | Qwell_: s,n,GotoIf($["${active_calls}" => = "2"]?3) |
| 04:39.48 | ThoMe | this? |
| 04:39.53 | ThoMe | is this allowed? |
| 04:40.17 | CCFL_Man2 | but my radio still bleeds audio from my other inputs |
| 04:40.28 | *** join/#asterisk |[_Caf_]| (i=___Caf__@229.85.252.216.dsl1.colba.net) |
| 04:40.31 | CCFL_Man2 | and i still never got a 684A subset |
| 04:41.54 | |[_Caf_]| | Can someone help me with this |
| 04:41.57 | *** topic/#asterisk by Qwell_ -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. -=- AstriCon is over :( |
| 04:42.39 | |[_Caf_]| | i run asterisk 1.2.24, with realtime voicemail on mysql databases, and cannot get my queues from queues.conf to load |
| 04:43.27 | |[_Caf_]| | basically i get this queue_exec: Unable to join queue 'colbanet-ts-en' |
| 04:43.36 | *** topic/#asterisk by Qwell_ -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php -=- AstriCon is over :( |
| 04:45.23 | |[_Caf_]| | i'm lost and try the bug report section on the digium websites but no lock |
| 04:46.03 | |[_Caf_]| | the closest article i found was 0005300 |
| 04:46.11 | ThoMe | exten => s,n,GotoIF($[${active_calls} > 1]?hauptnummer_ausserhalb_der_geschaeftszeiten,s,1) |
| 04:46.16 | ThoMe | is it wrong?? |
| 04:47.05 | *** join/#asterisk dishem (n=dishem@ip68-109-166-1.ph.ph.cox.net) |
| 04:47.11 | dishem | hi guys |
| 04:47.48 | dishem | can anyone tell me if it's possible to have the calling party hear a dtmf tone when the called party picks up the phone? |
| 04:49.26 | CCFL_Man2 | i need an audio switcher and audio distribution amp |
| 04:49.35 | CCFL_Man2 | this'll be tough |
| 04:49.36 | Corydon76-dig | dishem: not the way it's written currently, no |
| 04:49.56 | *** join/#asterisk Dimitripietro (n=dimitrip@65-122-15-162.dia.static.qwest.net) |
| 04:50.01 | Dimitripietro | Bump !!! |
| 04:50.11 | ThoMe | Corydon76-dig: can u help me please? what is here wrong? :-( >> exten => s,n,GotoIF($[${active_calls} > 1]?hauptnummer_ausserhalb_der_geschaeftszeiten,s,1) |
| 04:50.14 | dishem | corydon76-dig: i was thinking something like transfering both parties into a bridge and playing the tones in the bridge, tho i don't know if it's even possible to play tones to a bridge |
| 04:50.44 | Corydon76-dig | dishem: generally not |
| 04:50.59 | russellb | |[_Caf_]|: 1.2 is no longer supported. |
| 04:51.08 | Corydon76-dig | ThoMe: what is setting active_calls ? |
| 04:51.23 | |[_Caf_]| | i'll compile 1.4 tomorrow then |
| 04:51.27 | dishem | it sure would be an awesome feature in addition to dial()'s D option |
| 04:51.40 | |[_Caf_]| | thanks for tonight. |
| 04:51.52 | ThoMe | Corydon76-dig: a variable from me. noop has said in the variable are "1" |
| 04:52.25 | Corydon76-dig | ThoMe: you realize, of course that 1 is NOT greater than itself, right? |
| 04:52.43 | ThoMe | jepp |
| 04:52.47 | ThoMe | oh ok |
| 04:53.18 | *** join/#asterisk brc_ (n=brc__@pdpc/supporter/basic/brc) |
| 04:53.25 | Corydon76-dig | If you meant ">=", that's what you should write |
| 04:55.35 | ThoMe | sorry and thank you :-) |
| 04:55.47 | ThoMe | in germany is it now 6:56 .. i go to bed :-) |
| 04:55.50 | ThoMe | sleep well! |
| 04:56.06 | Corydon76-dig | ThoMe: no big deal, I've done silly things before, too |
| 04:56.34 | brc_ | oi |
| 04:59.01 | dishem | corydon76-dig: what about with some kind of answering dealy? |
| 04:59.27 | dishem | corydon76-dig: answer, play tones, then ring extension |
| 05:01.02 | Corydon76-dig | That doesn't play tones when someone answers. That plays tones regardless |
| 05:02.18 | dishem | right |
| 05:02.36 | dishem | that would be an acceptable alternative, i think. |
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| 08:05.33 | *** join/#asterisk batphone (n=batphone@cpe-70-116-10-87.austin.res.rr.com) |
| 08:05.41 | batphone | greetings |
| 08:08.44 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 08:09.39 | bartpbx | hello from germany |
| 08:10.56 | bartpbx | since a few day i see strange resulst on some of our servers. If i do a show channels i get 5 active channels but 24 active calls. I see the 5 active calls an than 24 messages "avoided deadlock". |
| 08:11.10 | bartpbx | this is on 1.2.24 |
| 08:11.44 | bartpbx | anyone seen this before? |
| 08:15.11 | batphone | those calls are probably not really going on |
| 08:15.33 | bartpbx | yes, but where are they from an how can i trace this problem? |
| 08:16.11 | batphone | its been a while since ive worked on one but ill try |
| 08:16.18 | batphone | put it up on pastebin or something |
| 08:17.18 | bartpbx | the problem is, i can not recreate this on my test box, it only happens on production an i dont want to enabele debug threads on them |
| 08:17.40 | bartpbx | but wait |
| 08:17.43 | bartpbx | ill put it |
| 08:20.05 | bartpbx | ok, its 3694 on pastbin |
| 08:21.30 | batphone | link please |
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| 08:21.37 | bartpbx | http://www.pastebin.org/3694 |
| 08:22.14 | batphone | lemme see show channels |
| 08:23.21 | batphone | what kind of phones are you using? |
| 08:23.26 | batphone | describe your set up a little |
| 08:23.40 | bartpbx | our customers use many different phones |
| 08:23.48 | batphone | ATA? |
| 08:23.52 | bartpbx | most of them have their own asterisks |
| 08:23.57 | batphone | i see |
| 08:24.10 | batphone | and this machine, is, a hub? |
| 08:24.14 | bartpbx | jes |
| 08:24.34 | batphone | what distro? |
| 08:24.41 | bartpbx | ist concentrates call and routes them from the customers to our gate or from the gate to the customer |
| 08:25.02 | bartpbx | debian 4.0 |
| 08:25.26 | batphone | hmm |
| 08:25.39 | batphone | im glad you didn't say "gentoo" |
| 08:25.41 | batphone | or something |
| 08:25.45 | batphone | "asslinux" |
| 08:25.47 | batphone | heh |
| 08:25.55 | bartpbx | :-) |
| 08:27.24 | bartpbx | http://www.pastebin.org/3696 |
| 08:27.25 | batphone | 32 channels.. thats an E1 right? |
| 08:27.34 | bartpbx | only ip traffic |
| 08:27.36 | bartpbx | no zaptel |
| 08:28.02 | bartpbx | the * reroutes the traffic to a SER That does a carrier selection and terminates the call |
| 08:28.12 | bartpbx | into the pstn |
| 08:28.28 | batphone | and your customers connect to you via sip trunk |
| 08:28.33 | bartpbx | yes |
| 08:28.40 | batphone | i see iax too |
| 08:28.59 | bartpbx | jey, some customers use IAX to ceonnect to us |
| 08:29.26 | bartpbx | and some inbound calls get routet from carries via iax to this server |
| 08:29.42 | bartpbx | we are using realtime |
| 08:29.45 | batphone | nice |
| 08:29.48 | bartpbx | an some fastagi stuff |
| 08:30.02 | batphone | what? no AEL? hehe |
| 08:30.13 | bartpbx | no not jet |
| 08:30.22 | batphone | its been a couple years since i saw that problem last |
| 08:30.44 | batphone | if i remember right it was at some funky setup we were trying to work between a call manager in mexico and some system in the states |
| 08:31.03 | batphone | 2 digit dialing to mexico city haha |
| 08:31.17 | batphone | an upgrade fixed it though |
| 08:32.12 | batphone | this does not look good |
| 08:32.14 | batphone | http://osdir.com/ml/telephony.ooh323.c/2005-11/msg00023.html |
| 08:32.23 | batphone | it may have something to do with a deep configuration |
| 08:32.33 | bartpbx | Ive theen this but be are not using h323 |
| 08:32.49 | bartpbx | but ill try to seperate a testcustomer tomorrow onto one test server |
| 08:32.55 | bartpbx | there i'll enable debug theads |
| 08:33.02 | bartpbx | and than we will see |
| 08:33.13 | bartpbx | I'll than open an issue in the bugtracker |
| 08:33.22 | batphone | wish i knew more |
| 08:33.31 | batphone | deadlocks are a pain |
| 08:33.32 | bartpbx | thanks a lot for your help |
| 08:33.41 | bartpbx | thats right |
| 08:34.04 | batphone | the more you pile into a system like that the better your chances of seeing wierd stuff like that are |
| 08:34.28 | bartpbx | hehe |
| 08:34.51 | batphone | several asterisk boxes routing through yours, over some different protocols, through RTA, then to SER, coming back in via different protocols |
| 08:35.02 | batphone | dozens of contexts... |
| 08:35.05 | batphone | hah! |
| 08:35.12 | batphone | thats why you make the big bucks man ;) |
| 08:35.34 | batphone | we ran it on gentoo for years |
| 08:35.43 | batphone | boy did i learn alot about linux hehe |
| 08:36.03 | bartpbx | we whre using suse before but switched to debian middle of last year |
| 08:36.04 | batphone | about the kernel especially |
| 08:36.07 | batphone | why? |
| 08:36.12 | batphone | cost? |
| 08:36.27 | bartpbx | the ugly support of suse and the "strange" licensing |
| 08:36.35 | batphone | yeah... |
| 08:36.54 | batphone | im just a network/unix admin now |
| 08:37.04 | batphone | we only have 1 * box at my friends desk |
| 08:37.25 | bartpbx | oh, my wife calls.. breakfast |
| 08:37.28 | batphone | i quit working with * about a year ago but i still use it from time to time |
| 08:37.30 | bartpbx | where are your from? |
| 08:37.31 | batphone | have fun ;) |
| 08:37.32 | bartpbx | germany? |
| 08:37.40 | batphone | Austin Texas |
| 08:37.46 | bartpbx | we are still searching for linux / asterisk guys |
| 08:37.47 | bartpbx | hm |
| 08:38.00 | bartpbx | okay.. thats to far to treavel every moring :-) |
| 08:38.01 | bartpbx | cu |
| 08:38.10 | batphone | well, ive built hundreds of them |
| 08:38.10 | dishem | tell your company to move me to germany :D |
| 08:38.18 | dishem | i was born there |
| 08:38.19 | dishem | hehe |
| 08:38.30 | batphone | hey i saw him first lol |
| 08:38.33 | dishem | :) |
| 08:38.41 | batphone | get me out of the US |
| 08:38.48 | batphone | and I'll work for, well, less |
| 08:39.07 | dishem | yeah like that's possible livin on euros :) |
| 08:39.28 | batphone | i like strudel |
| 08:39.36 | batphone | and bier |
| 08:39.54 | batphone | haha... i dont think id be interested in support phones though |
| 08:39.58 | batphone | carrier stuff, maybe |
| 08:40.21 | batphone | "waaah.. how do i change the ring tone on my polycom.... i want my mp3.." |
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| 08:49.13 | sysreq | i'm using a polycom ip 501 and i'm having this weird problem: when i put a call on hold and that call is disconnected by the other end somehow, i can still see it on hold and i can't resume/end it. |
| 08:50.06 | sysreq | anyone ever had that problem? |
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| 08:59.00 | sysreq | i can't even soft reset it at that stage.. it says 'reboot initiated', waits, then goes back to the main screen instead of actually restarting.. |
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| 09:05.21 | zumbush | when upgrading from asterisk 1.2 to 1.4 on my centos box do i need to remove the old 1.2 first? |
| 09:05.37 | zumbush | or can i just compile in the now 1.4 |
| 09:05.47 | zumbush | *new |
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| 09:06.55 | sysreq | zumbush: did you install it (1.2) with yum, or from sources? |
| 09:08.07 | zumbush | source |
| 09:09.09 | sysreq | the 1.4 files should override the old ones. |
| 09:09.16 | zumbush | thx |
| 09:10.02 | zumbush | from what i read i just need to backup my sounds cuz 1.4 have a different file structure |
| 09:10.24 | sysreq | did you want to keep your configuration files as well? |
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| 09:11.01 | zumbush | yea ive backup up them to |
| 09:12.18 | sysreq | if you don't use 'make samples', they should be fine but it's still good to back them up. |
| 09:12.47 | zumbush | thx |
| 09:12.47 | sysreq | but that's pretty much it. |
| 09:12.51 | sysreq | no problem |
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| 09:38.01 | zumbush | i get this when i try to compile zaptel |
| 09:38.19 | zumbush | zaptel-1.4.5/ztdummy.c:89: error: storage size of `ztd_tlet' isn't known |
| 09:38.28 | zumbush | *** [/usr/src/zaptel-1.4.5/ztdummy.o] Fel 1 |
| 09:38.46 | zumbush | any ideas whats wrong anyone? |
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| 09:41.33 | zumbush | forget it my bad |
| 09:41.45 | zumbush | was using old info on installing |
| 09:53.31 | zumbush | gettin "No rule to make target `install-inlcude', needed by `install-programs'. Stop." |
| 09:53.51 | zumbush | is there a typo in zaptel sourcefiles? |
| 09:54.03 | zumbush | thats doin this |
| 09:57.18 | zumbush | yep .when i edited the Makefile to correct type its all well |
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| 10:12.59 | sysreq | zumbush: * Makefile, /: * Fix Makefile typo in 'install-progrms:'.. that's in zaptel 1.4.5.1's ChangeLog. |
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| 10:27.58 | Skty | Evening all. I've had a couple drinks so please excuse my Saturday night'ish questions but just a quick one that has resulted in a bit of Googl'ing. Is there still no way to bridge two calls from the Dialplan without having to put them insomething like a meetme room? |
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| 10:44.39 | tzafrir_home | zumbush, what kernel version? |
| 10:47.35 | zumbush | <PROTECTED> |
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| 10:54.56 | tzafrir_home | zumbush, hmmm... tasklets are only used in the RTC code, I believe |
| 10:55.03 | tzafrir_home | let's see |
| 10:58.10 | tzafrir_home | USE_RTC only gets set for kernel >= 2.6.13 |
| 10:58.54 | tzafrir_home | That said, you should get 1.4.5.1 or current zaptel svn |
| 10:59.18 | tzafrir_home | due to the 'make install' issue mentioned earler here |
| 11:00.42 | tzafrir_home | zumbush, output of 'uname -r' is 2.6.9-55.0.9.ELsmp and you don't set KSRC/KVERS explicitly in the make command-line? |
| 11:00.53 | zumbush | thx tzafrir but when i manually edited the Makefile it all worked out well |
| 11:01.04 | zumbush | zaptel seems working now |
| 11:01.04 | mvanbaak | Skty: not as far as I know |
| 11:01.21 | zumbush | is there any more issues than the typo that i need to be aware about |
| 11:02.48 | tzafrir_home | There were also some minor fixes in 'make config' later on |
| 11:03.10 | tzafrir_home | zumbush, does it mean you don't have that error with ztdummy? |
| 11:03.40 | zumbush | i can load my te110p card and it seems to work when running zttool |
| 11:04.02 | mvanbaak | tzafrir_home: can you remind me how to use ztdummy inside xen ? |
| 11:04.17 | mvanbaak | I remember it's possible by editing some file |
| 11:04.23 | mvanbaak | but I cant remember what it was |
| 11:04.58 | tzafrir_home | mvanbaak, there seems to be a patch for that. Never tried it |
| 11:07.09 | zumbush | tzafrir -> yea no problem with ztdummy now when compiling |
| 11:16.03 | Skty | Hrmm righto mvanbaak - I guess what I will have to do is create a Meetme room for each instance and transfer the two calls into it. |
| 11:16.07 | mvanbaak | tzafrir_home: I thought you only needed to comment out something bout RTC |
| 11:17.00 | tzafrir_home | mvanbaak, hmm... can't find the ztdummy xen bug on the Mantis |
| 11:22.39 | mvanbaak | meh, nice mail signature: GDB has a 'break' feature; why doesn't it have 'fix' too? |
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| 13:10.47 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php -=- AstriCon is over :( |
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| 13:40.03 | dongs | who was i haveing a conversation about analog to ISDN conversion few days ago :D |
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| 13:45.03 | *** join/#asterisk |[_Caf_]| (i=___Caf__@229.85.252.216.dsl1.colba.net) |
| 13:45.12 | |[_Caf_]| | Hello |
| 13:46.17 | |[_Caf_]| | I have a problem comiling asterisk 1.4.11, it gives me an error in menuselect.makeopts error type 255, any idea how to correct this. |
| 13:47.50 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:49.34 | mvanbaak | make clean; ./configure; make |
| 13:50.00 | |[_Caf_]| | thats what i do but gives me the error 255 |
| 13:50.28 | mvanbaak | can you pastebin the output ? |
| 13:50.50 | |[_Caf_]| | make[2]: Leaving directory `/usr/src/asterisk-1.4.11/menuselect' |
| 13:50.50 | |[_Caf_]| | make[1]: Leaving directory `/usr/src/asterisk-1.4.11/menuselect' |
| 13:50.50 | |[_Caf_]| | menuselect/menuselect --check-deps menuselect.makeopts |
| 13:50.50 | |[_Caf_]| | mxml: Character entity "cur" not terminated under parent <menu>! |
| 13:50.50 | |[_Caf_]| | Error parsing 'menuselect-tree'! |
| 13:50.51 | |[_Caf_]| | make: *** [menuselect.makeopts] Error 255 |
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| 13:55.58 | riddlebox | how can I get asterisk to actually stop ringing my phones when someone hangs up? |
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| 13:58.52 | dongs | what |
| 13:58.54 | dongs | it does that. |
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| 14:02.43 | riddlebox | dongs, if I call from my cellphone, wait two rings, then hang up, my asterisk system will keep ringing until voicemail answers, then there will a message of nothing |
| 14:05.09 | _ShrikE | riddlebox: Pots? Sounds like disconnect supervision issue. |
| 14:05.52 | riddlebox | _ShrikE, I have connected an analog phone to my pots line, and it does stop ringing when I hang up |
| 14:07.44 | _ShrikE | kewl start? |
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| 14:11.43 | riddlebox | yeah I have it fxs_ks |
| 14:14.27 | _ShrikE | If you have a analog phone with lit keypad that is span powered you should see the light flash when the far end hangs up if you telco is providing disconnect supervision. |
| 14:15.26 | riddlebox | _ShrikE, so there is nothing in the dialplan that could cause it? |
| 14:16.54 | _ShrikE | What you are describing sounds like a disconnect supervision to me. When the remote party hangs up nothing is telling * that it has happened. It is usually done by droppig the loop current for a few milliseconds or reversing the polarity. Depending on what part of the world you are in. |
| 14:17.14 | riddlebox | one other question, I used to have exten => 522,1,Dial(SIP/522,20,tr) is the tr correct? |
| 14:17.27 | _ShrikE | r is almost never a good idea. |
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| 14:18.11 | riddlebox | see I used to be able to transfer a call without pressing the # key, and now I do, I have taken the tr off of every extension |
| 14:18.20 | keulin | in fact and after a very long thinking, a phone with an rj45 connector seems to be so more natural |
| 14:20.09 | |[_Caf_]| | nobody has any idea about my problem ? or i'm just not seeing what everybody knows ? |
| 14:20.29 | riddlebox | I talked to another phone vendor(a competitor) yesterday and he said they do not trust VoIP and walk away from any sales opportunity that the potential customer asks about VoIP |
| 14:21.13 | _ShrikE | That ridiculous |
| 14:21.20 | _ShrikE | er.. thats |
| 14:21.29 | riddlebox | _ShrikE, so even if an analog phone connected to the line stops ringing I still could not have disconnect supervision on the line? |
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| 14:21.46 | _ShrikE | The analog phone is not your fxo card |
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| 14:22.05 | riddlebox | _ShrikE, they sell Avaya Partner and Avaya Magix ->which is discontinued |
| 14:25.49 | [TK]D-Fender | analog phones ringing & "call disconnect supervision" have nothing to do with each other. |
| 14:27.01 | riddlebox | ohh ok |
| 14:27.29 | [TK]D-Fender | Its the weekend, and the crazies have come to #asterisk ..... |
| 14:27.47 | riddlebox | well in that case I will call Charter and tell them to come out and replace my modem, I talked to someone last night and they said they think my modem doesnt respond to it |
| 14:27.49 | _ShrikE | ha |
| 14:28.19 | [TK]D-Fender | riddlebox, So you would-be customer fears VoIp do they? Was that your plan for termination? |
| 14:29.31 | riddlebox | [TK]D-Fender, not my company, I was talking to a competitor yesterday and he told me that about his company |
| 14:34.06 | [TK]D-Fender | riddlebox, Answering that I ususlay terminate via PRI or POTS. VoIP for me in * installations is between the phone & PBX only. |
| 14:35.18 | riddlebox | yeah I know, I just walked away from the guy, thinking they only do small business' but he offered me a job, they want to "eventually do VoIP" |
| 14:42.40 | riddlebox | [TK]D-Fender, you know who I really have a problem with in the St. Louis area, all the cisco vendors that are selling the cisco phone system. None of them know anything about wiring or telecom |
| 14:44.05 | [TK]D-Fender | riddlebox, And how is that a problem? |
| 14:46.13 | riddlebox | its a problem because alot of corporations around here which used to be our customers are going Cisco, so these vendors go out, cant get things to work like a T1, so the customer gets mad because these guys dont know what they are doing, we get called out to go "help" the cisco guys, but we still dont get the customer |
| 14:46.49 | [TK]D-Fender | riddlebox, no profit for you? |
| 14:47.06 | riddlebox | we get that one last service call from them |
| 14:47.30 | riddlebox | I told my boss we should start working on Cisco then we could kill these guys |
| 14:48.11 | riddlebox | I argued with one of these guys for an hour, because he said you couldnt put a fax line on a cat5e cable, he said it just doesnt work |
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| 15:01.32 | jozu | hi |
| 15:02.40 | jozu | i need some help |
| 15:03.07 | *** join/#asterisk shido6 (n=shido6@74-130-59-184.dhcp.insightbb.com) |
| 15:04.09 | asdx | hi |
| 15:04.10 | jozu | in the a2billing my clients can see the calls in a web page? total, subotatal, taxes.. etc |
| 15:04.41 | asdx | im getting a vps, will i be able to use asterisk with pstn there? |
| 15:04.57 | asdx | ~book |
| 15:04.58 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 15:05.59 | tzafrir_home | asdx, how exactly do you intend to connect PSTN there? |
| 15:06.14 | tzafrir_home | Not through local hardware, I gather |
| 15:06.43 | asdx | tzafrir_home: yeah, i thought so |
| 15:07.21 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583653.dsl.bell.ca) |
| 15:07.23 | asdx | tzafrir_home: but i should be able to use voip phones (hardware phones and soft phones) right? |
| 15:07.47 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 15:08.58 | tzafrir_home | asdx, how good a control do you have on the network settings to your VPS? |
| 15:09.21 | asdx | tzafrir_home: i don't have access to the computer (hardware) |
| 15:09.27 | tzafrir_home | Can you guarantee enough bandwidth for your VoIP traffic? |
| 15:09.32 | asdx | i think i will have root access thought |
| 15:09.39 | asdx | tzafrir_home: yeah |
| 15:10.00 | tzafrir_home | On that system, yes. But can you guarantee good traffic to there? |
| 15:10.34 | asdx | tzafrir_home: yes, i think |
| 15:12.22 | asdx | well, i will try |
| 15:14.24 | jozu | tzafrir_home, i need a homepage whith login/pass for my potential clients, in this page they can see the cost of calls.. a2billing is capable to do this? |
| 15:15.05 | tzafrir_home | jozu, sorry, I don't know a2billing |
| 15:15.20 | jozu | oks, thanks |
| 15:16.23 | shido6 | yes it does |
| 15:28.09 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 15:28.19 | VJFROMGT | when a trunk is busy i get "Dial failed due to trunk reporting BUSY |
| 15:28.34 | VJFROMGT | is tehre a way for asterisk to try the other trunk in the route before giving up |
| 15:29.32 | shido6 | yes |
| 15:29.40 | VJFROMGT | how shido? |
| 15:32.45 | VJFROMGT | still there? |
| 15:32.53 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 15:39.29 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.52) |
| 15:40.41 | tzafrir_home | VJFROMGT, yes. This is why you have a dialplan for |
| 15:40.45 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.227) |
| 15:41.06 | VJFROMGT | dialplan works great, providing asterisk know in advance what is busy and what is available |
| 15:41.34 | tzafrir_home | Check ${DIALSTATUS} after the dial |
| 15:41.36 | VJFROMGT | if asterisk thinks that the line is available and then tries it to find out that it is not then it just gives up |
| 15:41.54 | tzafrir_home | e.g: goto(s-${DIALSTATUS},1) |
| 15:42.05 | VJFROMGT | what does dialstatus 1 mean? |
| 15:42.21 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:43.58 | VJFROMGT | this is what my dial plan looks like http://pastebin.ca/719725 |
| 15:50.43 | riddlebox | does anyone know how to program those telephony modems that cable companies put in? |
| 15:54.14 | *** join/#asterisk Yourname` (i=Miranda@unaffiliated/yourname/x-837320) |
| 15:58.09 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:07.13 | darkskiez | if a phone is convinced its got a different ip after doing stun and is setting up the rtp session to talk to its public address rather than its natted address for communication within a lan, is there an asterisk param to override this and send the rtp of the source of the sip query? |
| 16:10.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:12.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:13.23 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:16.55 | |[_Caf_]| | is there a way to force asterisk to read the file queues.conf ? |
| 16:17.05 | |[_Caf_]| | from extconfg.conf |
| 16:19.48 | Corydon76-dig | for static or dynamic realtime? |
| 16:19.57 | |[_Caf_]| | for static |
| 16:20.12 | Corydon76-dig | "module reload app_queue.so" |
| 16:20.20 | |[_Caf_]| | i tough i had a problem with realtime versus static |
| 16:20.34 | |[_Caf_]| | because i had voicemail on mysql and queue in queues.conf |
| 16:21.09 | |[_Caf_]| | but even if a reset voicemail to not use mysql my queues don't come up |
| 16:21.29 | |[_Caf_]| | even after a reload app_queue.so it doesn't work |
| 16:22.14 | Corydon76-dig | Do you have queues enabled in extconfig.conf? |
| 16:22.24 | Corydon76-dig | Because if you do, that's why it's not reading the file |
| 16:22.27 | |[_Caf_]| | my ivr in extensions.conf is read but i get this queue_exec: Unable to join queue 'colbanet-ts-en' |
| 16:22.45 | |[_Caf_]| | not it's commented with ; |
| 16:22.50 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 16:22.56 | Corydon76-dig | Pastebin your configs |
| 16:23.21 | |[_Caf_]| | to you or the channel ? |
| 16:24.00 | Corydon76-dig | ~pb |
| 16:24.00 | jbot | A Pastebin is a place to paste your stuff 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 |
| 16:24.19 | Daejeo1 | my phone polycom-501 was registered, but it is not registering anymore |
| 16:24.51 | Daejeo1 | where can i see error log files? |
| 16:25.09 | Daejeo1 | nothing on cli |
| 16:25.37 | Corydon76-dig | Daejeo1: try rebooting your phone |
| 16:26.54 | Daejeo1 | Corydon76-dig : how can i reboot manually? |
| 16:27.20 | Corydon76-dig | Hold down -, +, Messages, and Hold |
| 16:27.21 | De_Mon | Daejeo1 disconnecting it from power generally works |
| 16:28.15 | Daejeo1 | i am 1000 miles away from the phone |
| 16:28.32 | |[_Caf_]| | Corydon76-dig http://pastebin.ca/719760 |
| 16:28.57 | |[_Caf_]| | i did it quick |
| 16:29.12 | |[_Caf_]| | i know that the queue name will not match |
| 16:29.12 | Corydon76-dig | Daejeo1: "sip notify polycom-check-cfg <peername>" |
| 16:29.16 | De_Mon | whats the recommended way to create zaptel.conf? I see a genzaptelconfig ztcfg AND zapconf |
| 16:29.36 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:29.53 | |[_Caf_]| | i just realize that but i have the queue colbanet-ts-en also configured in my queues.conf |
| 16:30.17 | |[_Caf_]| | and i posted colbanet-bs-en |
| 16:30.23 | |[_Caf_]| | my mistake |
| 16:30.34 | riddlebox | has anyone ever configured the phone modem that a cable company puts in? |
| 16:30.47 | De_Mon | riddlebox huh? |
| 16:30.55 | Corydon76-dig | |[_Caf_]|: typos are frequently the cause here |
| 16:31.16 | Corydon76-dig | riddlebox: it's not a modem |
| 16:32.08 | riddlebox | Corydon76-dig, hrmm thats what I thought I read, it is a Arris TM502A, I just want to look at it and see if I can turn on disconnect supervision on the device |
| 16:32.22 | Corydon76-dig | Daejeo1: if you're 1000 miles away from the phone, how can you verify the phone is even plugged in? |
| 16:32.27 | Daejeo1 | Corydon76-dig: how can i check the firmware version? |
| 16:33.01 | Daejeo1 | Corydon76-dig: i am accessing it thri web |
| 16:33.13 | Daejeo1 | thats wahy i know it is plugged in |
| 16:33.34 | Corydon76-dig | I really don't recommend using its web interface |
| 16:33.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:34.23 | Daejeo1 | how can I edit the sip and lines parameters remotely ? |
| 16:34.23 | De_Mon | Daejeo1 the web interface doesnt have a reboot button? |
| 16:34.56 | Corydon76-dig | Daejeo1: most people use a central provisioning server |
| 16:35.03 | |[_Caf_]| | Corydon76-dig i will redo my post because i get to my queue cs correctly but not my ts |
| 16:36.44 | Daejeo1 | well when I change certain parameters . it says " please wait your phone will restart" |
| 16:37.37 | Daejeo1 | Corydon76-dig: how can i check the firmware version? |
| 16:37.43 | |[_Caf_]| | Corydon76-dig here's the new post http://pastebin.ca/719771 |
| 16:37.47 | Corydon76-dig | Daejeo1: have no idea |
| 16:38.01 | riddlebox | Corydon76-dig, do you have any clue how I could connect to it and program it, there is an Ethernet cable port, but I tried a straight thru and a crossover, neither allowed me to connect |
| 16:38.30 | Corydon76-dig | riddlebox: no |
| 16:38.40 | Daejeo1 | Corydon76-dig: did u ever update the firmware version on polycom? |
| 16:38.56 | Corydon76-dig | Daejeo1: yes, but I do it when I'm right next to the phone |
| 16:39.30 | Corydon76-dig | |[_Caf_]|: you're still referencing the same queue that is in queues.conf |
| 16:39.51 | Daejeo1 | Corydon76-dig: it was registered, and working fine |
| 16:39.55 | Corydon76-dig | |[_Caf_]|: I don't see any difference whatsoever |
| 16:40.11 | Corydon76-dig | Daejeo1: well, somebody obviously changed something |
| 16:40.14 | Daejeo1 | now don't know what ahppened |
| 16:40.52 | Corydon76-dig | Maybe someone tightened firewall rules? |
| 16:41.03 | Daejeo1 | yeah it could be |
| 16:41.31 | Daejeo1 | i doubt about router |
| 16:42.03 | Daejeo1 | i asked my friend to connect the phone directly to isp modem |
| 16:42.17 | |[_Caf_]| | Corydon76-dig no, no the other post was with colbanet-bs-en, this is colbanet-ts-en, i also have a colbanet-ac |
| 16:43.22 | Daejeo1 | Corydon76-dig: do you think that ISP is blocking ports? |
| 16:44.18 | Daejeo1 | my friend was using nokia 80 ie. it was registered |
| 16:44.42 | Daejeo1 | now both the phones are regsitering |
| 16:44.46 | Daejeo1 | sorry |
| 16:45.05 | Daejeo1 | now both the phones are not registering |
| 16:47.30 | Daejeo1 | trendnet- wireless router |
| 16:48.38 | file | yay free wifi |
| 16:50.44 | |[_Caf_]| | thank you for your help i found the errors of my way |
| 16:51.15 | |[_Caf_]| | i've got no registered sip for my members so the queue went straight to my voice mail. |
| 16:57.03 | |[_Caf_]| | i have another problem though, when i compile zaptel it doesn't start wct4xxp, i have to start manually with modprobe wct4xxp t1e1override=0 any one knows how to solve this ? |
| 16:57.11 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:07.32 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:15.19 | Daejeo1 | Corydon76-dig: are you around? |
| 17:18.36 | De_Mon | |[_Caf_]| that would be a question for your distribution, or startup scripts |
| 17:18.45 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 17:19.14 | |[_Caf_]| | ok thnak you |
| 17:20.25 | *** join/#asterisk ManxPower (n=manxpowe@ip68-11-78-112.no.no.cox.net) |
| 17:22.01 | Daejeo1 | Corydon76-dig: router trouble |
| 17:22.12 | Daejeo1 | phone is registered now |
| 17:22.19 | Daejeo1 | :) |
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| 17:35.28 | karleeto | i'm having trouble finding some directions to compile a custom kernel on trixbox.. i just need to have the nvnet driver for my onboard nic |
| 17:35.41 | karleeto | is there a way to just get that module, or do i need to compile a new kernel? |
| 17:35.52 | karleeto | or forcedeth |
| 17:36.41 | Nugget | This is not a trixbox support channel. |
| 17:36.43 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.153.79) |
| 17:37.17 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-59-195.pskn.east.verizon.net) |
| 17:37.20 | CCFL_Man2 | this tone dialer i got is from british telecom |
| 17:37.23 | CCFL_Man2 | wtf |
| 17:38.01 | CCFL_Man2 | a collector of western electric and i have a british telecom tone dialer |
| 17:38.14 | karleeto | Nugget: yes i know that, but not everyone is crappy about helping someone |
| 17:39.08 | riddlebox | karleeto, there is a #trixbox support channel, trixbox changes things from regular asterisk, that is why it is best to go to the #trixbox channel |
| 17:42.49 | karleeto | riddlebox: yes, i understand.. i'm in the channel, its just famous for being idle |
| 17:44.21 | ManxPower | unfortunatly trixbos changes so much stuff it really isn't "asterisk" |
| 17:44.40 | karleeto | yeah |
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| 17:49.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:49.49 | riddlebox | I really wish I could get into this modem for my telephone |
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| 18:07.25 | tzafrir_home | karleeto, instructions for that is a bit tricky. People on #centos don't like answering that either :-) |
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| 19:49.32 | hmmhesays | [TK]D-Fender: where do you go for your polycoms? |
| 19:50.02 | [TK]D-Fender | hmmhesays, Usually buy from CCP here. Telephonydepot / voipsupply in the US |
| 19:50.18 | [TK]D-Fender | hmmhesays, VoOP Supply's canadian store is pretty close to par |
| 19:52.12 | hmmhesays | k |
| 19:52.19 | hmmhesays | atacomm closed down |
| 19:53.37 | [TK]D-Fender | yup |
| 19:53.53 | hmmhesays | bastards, they were in the same state as me so I got next day delivery on stuff |
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| 20:00.05 | *** join/#asterisk Grandfrere (n=dustin_n@c-67-166-113-138.hsd1.ut.comcast.net) |
| 20:01.04 | Grandfrere | I'm looking for somebody that can help me modify an Asterisk app. I've got most of the work done. I'm just missing one small piece, but I'm a sysadmin, not a programmer. :) |
| 20:01.49 | *** join/#asterisk FTexcom (n=hola@169.Red-88-2-81.staticIP.rima-tde.net) |
| 20:01.54 | FTexcom | Hello, good night |
| 20:02.09 | Grandfrere | Hey FTexcom. Are you a developer? |
| 20:02.13 | FTexcom | nop |
| 20:02.18 | FTexcom | I'm here for asking help |
| 20:02.22 | So3kris | hello |
| 20:02.26 | Grandfrere | Me too. Nobody's answering. |
| 20:02.26 | hmmhesays | Grandfrere: #asterisk-dev |
| 20:02.29 | hmmhesays | probably have better luck |
| 20:02.35 | Grandfrere | Ah, okay. Thanks. |
| 20:02.39 | So3kris | have someone a blackbarry with asterisk working |
| 20:02.50 | hmmhesays | i might be able to help you, but probably not |
| 20:03.01 | Grandfrere | Do you know C? |
| 20:03.19 | hmmhesays | enough to be dangerous |
| 20:03.32 | Grandfrere | That's more than I know. I'll give you $50 if you can help me get this working. |
| 20:03.33 | FTexcom | whenever I issue the reload command it never ends, wheter I do it on freepbx or on the asterisk CLI |
| 20:03.55 | hmmhesays | what are you trying to do and where are you stuck? |
| 20:04.14 | Grandfrere | Okay, I'm working with the dictation application, app_dictate.c... |
| 20:04.24 | Fluor_ | lol hmmhesays |
| 20:04.47 | hmmhesays | hmm? |
| 20:04.53 | Grandfrere | I've modified it from its original functionality so that when somebody calls in, they get a message saying to enter a code... |
| 20:05.01 | Fluor_ | 'enough to be dangerous' ;) |
| 20:05.09 | Grandfrere | It's essentially an IVR, the code is an extension. |
| 20:05.19 | Grandfrere | When they dial the extension, the dictate app is called. |
| 20:05.29 | Grandfrere | I've assigned the filename to "foo" |
| 20:05.36 | hmmhesays | ok |
| 20:05.42 | Grandfrere | Then they can hit the # key and it asks them to enter a new filename. |
| 20:05.56 | Grandfrere | They can do that, but I WANT them to be able to hit the # key and have the app assign a random filename. |
| 20:06.14 | *** join/#asterisk ta^3 (n=tacvbo@65-122-15-162.dia.static.qwest.net) |
| 20:06.35 | hmmhesays | random with some structure? |
| 20:06.40 | hmmhesays | or completely random |
| 20:06.51 | Grandfrere | Well, let's say a random four-digit number. |
| 20:07.15 | hmmhesays | is hitting the # key part of the app currently? |
| 20:07.50 | Grandfrere | Yes. If they're recording, they hit the pound key and it plays a prompt asking them to enter a filename, followed by the pound key. |
| 20:08.23 | Grandfrere | Once they do that, they get another prompt that tells them to begin recording. |
| 20:08.36 | hmmhesays | so what is the behavior you want? |
| 20:09.00 | Grandfrere | I want it to work like this: they are recording, they hit the pound key, and instead of asking for a filename, it assigns a random one, then plays the "you can now begin recording" prompt. |
| 20:09.08 | hmmhesays | hit the pound key, random file name assigned, message to begin recording |
| 20:09.13 | Grandfrere | Yes. |
| 20:09.47 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com) |
| 20:10.20 | Grandfrere | I just don't know enough about C to even begin messing with types and pointers. |
| 20:10.57 | Grandfrere | I could email you the section of code where the existing behavior is displayed, if that would help. |
| 20:11.16 | hmmhesays | yeah I'm looking at it now |
| 20:11.28 | Grandfrere | Well, I've modified it from the original. |
| 20:11.41 | Grandfrere | The version I'm using isn't the one in the main source tree. |
| 20:11.46 | hmmhesays | hmm |
| 20:11.55 | hmmhesays | you should probably ask in asterisk-dev |
| 20:12.06 | Grandfrere | Okay. I'll try there. Thanks. |
| 20:12.20 | hmmhesays | I could probably help you but it sure wouldn't be in a timely fashion |
| 20:12.47 | Grandfrere | Hehe. I understand. The problem is I've put it off and put it off and now the customer wants it delivered by Monday. :) |
| 20:13.27 | hmmhesays | yeah, there is no flag for app dictate to set the filename? |
| 20:14.03 | Grandfrere | There is a way to pass it the first time. I'm doing that. My problem is doing it the second and third times. |
| 20:14.32 | hmmhesays | set a channel variable with a random name and pass it each time |
| 20:15.16 | Grandfrere | I don't know how to do that. |
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| 20:18.44 | hmmhesays | easily |
| 20:21.25 | hmmhesays | Grandfrere: one sec I'll show you |
| 20:21.31 | Grandfrere | Okay. |
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| 20:24.36 | hmmhesays | you don't have to recode the app to do what you want |
| 20:26.06 | hmmhesays | Grandfrere: give me a couple of minutes I'm looking for the right function |
| 20:27.39 | *** join/#asterisk blq (n=Bl@dslb-088-064-138-243.pools.arcor-ip.net) |
| 20:28.48 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 20:29.12 | hmmhesays | http://www.pastebin.ca/719960 |
| 20:29.27 | hmmhesays | theres an example of setting your destination file name the same as the channel name |
| 20:29.35 | hmmhesays | which should be unique |
| 20:34.37 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 20:43.09 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 20:53.28 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 21:01.02 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:05.29 | *** join/#asterisk DM_Lazybone (n=lazybone@dslb-084-061-235-205.pools.arcor-ip.net) |
| 21:05.35 | DM_Lazybone | Hello |
| 21:06.05 | DM_Lazybone | i need german help |
| 21:08.01 | DM_Lazybone | speak anybody german? |
| 21:08.07 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 21:10.17 | DM_Lazybone | hi famicon |
| 21:10.29 | tzafrir_laptop | DM_Lazybone, try English... |
| 21:10.50 | DM_Lazybone | my english is bad |
| 21:11.05 | tzafrir_laptop | My German is even worse ;-) |
| 21:11.16 | DM_Lazybone | ;-) |
| 21:11.46 | DM_Lazybone | gives a german asterisk chan ? |
| 21:12.18 | tzafrir_laptop | If there was, it would probably be called #asterisk-de . I don't see anybody there |
| 21:12.23 | *** part/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 21:12.41 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 21:13.28 | DM_Lazybone | hmm |
| 21:17.04 | hmmhesays | where would I go for a did from japan? |
| 21:17.10 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 21:20.11 | sysreq | hmmhesays: didww.com seems to have japanese (tokyo) dids.. |
| 21:20.35 | hmmhesays | I guess i'm looking for somewhere somebody has experience with |
| 21:21.14 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 21:21.22 | WilliamK | does anyone else have probs with being logged out of a queue(s) if you type reload on the console within asterisk? |
| 21:21.57 | FTexcom | whenever I issue the reload command it never ends, wheter I do it on freepbx or on the asterisk CLI |
| 21:25.11 | sysreq | hmmhesays: i have a couple canadian dids with them.. they support sip/iax/h323, use ulaw, and they're pretty cheap considering it's a flate rate plan.. |
| 21:25.49 | sysreq | i've never really had problems with them. |
| 21:26.43 | hmmhesays | no compressed codecs? |
| 21:29.04 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:31.03 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:31.22 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:31.38 | DM_Lazybone | keiner da der deutsch kann? |
| 21:31.57 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 21:32.50 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:32.52 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:33.15 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:33.46 | rbd | hi guys, I'm trying to do some capacity planning around asterisk. I'm wondering if the usual load of a AGI (IVR-type) call on the box is about as much as a caller in a meetme conference.... e.g. should a box that can do 100 IVR sessions normally be able to do 100 meetme sessions, or 50 IVR sessions and 50 meetme sessions? |
| 21:34.02 | sysreq | hmmhesays: at least gsm, because i just tested it.. but i'm not sure about others. |
| 21:34.02 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:34.08 | rbd | or is the load nature of these two things different enough to where it's like apples and oranges? |
| 21:34.19 | hmmhesays | g729 is what i would be interested in |
| 21:34.46 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:36.29 | hmmhesays | also would be interested in backup routing to pstn |
| 21:36.32 | hmmhesays | like vitelity has |
| 21:37.23 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:37.41 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:37.50 | *** part/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:38.24 | *** join/#asterisk atomicd (n=atomicd@adsl-68-123-236-193.dsl.irvnca.pacbell.net) |
| 21:40.47 | hmmhesays | sysreq: are you happy with their service? |
| 21:43.12 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:48.19 | sysreq | hmmhesays: yes, but i've used my dids with them mainly for low traffic (residential) purposes.. i have no idea how they perform under a heavy load. |
| 21:48.25 | sysreq | i'm not of much help, am i? ;p |
| 21:49.54 | *** join/#asterisk r- (i=rob@rob.tc) |
| 21:50.15 | r- | hy |
| 21:52.07 | jozu | who is the autofalltrougt channel sip/200 status is 'unknown' problem? |
| 21:52.53 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 21:53.20 | jozu | someone can helpme? |
| 21:53.40 | jozu | auto fallthrough, channel sip status is 'unknown' |
| 21:53.41 | jozu | ??? |
| 21:53.56 | jozu | what this? |
| 21:55.14 | r- | huh |
| 21:58.08 | *** join/#asterisk ManxPower (n=manxpowe@11.sub-70-197-51.myvzw.com) |
| 22:00.57 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 22:01.01 | _pepo_ | hi friends |
| 22:01.48 | _pepo_ | where do I can look up information to use asterisk console ? |
| 22:02.10 | r- | asterisk.org |
| 22:03.06 | _pepo_ | I see |
| 22:06.41 | *** join/#asterisk blq (n=Bl@dslb-088-065-171-218.pools.arcor-ip.net) |
| 22:18.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:21.49 | ectospasm | _pepo_: you can also type help at the CLI |
| 22:24.14 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 22:28.55 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 22:51.13 | rbd | is it possible to run a CLI command through the dial plan? (I don't see the function listed....I probably missed it) |
| 22:51.46 | *** join/#asterisk Boones (n=bytewalk@port-212-202-170-97.dynamic.qsc.de) |
| 22:54.54 | rbd | looks like I'd have to use system to run asterisk -r -x <mycommand> |
| 22:55.05 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:57.07 | ManxPower | "show applications like system" |
| 23:02.04 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
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| 23:06.19 | *** part/#asterisk Grandfrere (n=dustin_n@c-67-166-113-138.hsd1.ut.comcast.net) |
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| 23:14.25 | *** join/#asterisk silvertip257 (n=d009c2c9@unaffiliated/silvertip257) |
| 23:17.00 | silvertip257 | I'm in need of a few SCTP packet captures for a project and would like to know if I can/how I can use Asterisk to create them |
| 23:19.36 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:19.46 | hmmhesays | sysreq: all I need is low traffic at the moment |
| 23:25.06 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:29.40 | jozu | where is the exten => to giveme a second tone to dial an external number? |
| 23:30.40 | sysreq | jozu: check out the DISA application, http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 23:30.51 | jozu | thanks |
| 23:31.23 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 23:39.52 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 23:42.43 | *** join/#asterisk SpaceBassLaptop (n=ls@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 23:42.47 | SpaceBassLaptop | hey folks |
| 23:42.53 | SpaceBassLaptop | anyone have a reccomendation for a wifi phone? |
| 23:43.03 | SpaceBassLaptop | I'm replacing a Hitachi IP-5000 |
| 23:43.16 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:45.27 | [TK]D-Fender | ~wifisip |
| 23:45.28 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 23:45.44 | [TK]D-Fender | SpaceBassLaptop, And that one is one of the ever so slightly better ones... |
| 23:46.10 | SpaceBassLaptop | I loved the IP-5000 but it died on me |
| 23:46.11 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 23:46.18 | SpaceBassLaptop | Frankly, it didn't suck too badly |
| 23:46.39 | silvertip257 | I'm in need of a few SCTP packet captures for a project and how can I create them? |
| 23:47.00 | SpaceBassLaptop | so what I'm hearing is that I should probably get another IP-5000 rather than a cisco 7920 or something else |
| 23:49.13 | SpaceBassLaptop | [TK]D-Fender, btw its been a long long time....how have you been? |
| 23:49.40 | ManxPower | silvertip257: I don't believe asterisk support STCP. You are out of luck. |
| 23:52.37 | ectospasm | I've got a minor IAX registration/authentication issue that I wish I could find a cleaner fix for... I've got an IAX trunk and an IAXy, and the only way I can receive incoming calls through the IAX trunk when the IAXy is configured is by having the plaintext passwords of the IAX trunk and IAXy be the same... |
| 23:53.16 | ManxPower | ectospasm: The IAXy does not support IAX2 trunking. |
| 23:53.24 | silvertip257 | ManxPower: what can I do to get SCTP then? |
| 23:53.31 | ectospasm | ManxPower: That's not what I'm trying to do |
| 23:53.40 | ManxPower | Also, I suspect the IAXy does not support anything except plaintext passwords. |
| 23:53.56 | ectospasm | I've got an IAX provider, and calling in on that IAX provider tries to authenticate as the IAXy user, and I can't figure out why. |
| 23:53.58 | ManxPower | silvertip257: heck if I know. Since Asterisk does not do STCP, nobody here has experience with it. |
| 23:54.08 | ectospasm | ManxPower: I know that |
| 23:54.29 | silvertip257 | oh ok sorry ManxPower - I really need to figure out how I can get a packet capture of it.... |
| 23:54.35 | ManxPower | ectospasm: The provider only authenticates as who you tell it to authenticate. |
| 23:54.53 | ectospasm | so... |
| 23:54.59 | ManxPower | ectospasm: you have done an iax2 debug to verify? |
| 23:55.21 | ectospasm | ManxPower: not in several weeks... lemme try it again... |
| 23:55.50 | silvertip257 | ManxPower: I messed up - it's SCTP |
| 23:56.06 | silvertip257 | stream control transport protocol |
| 23:56.09 | ManxPower | silvertip257: Asterisk does not support any kind of encrypted SIP or RTP. |
| 23:56.18 | silvertip257 | ok :( |
| 23:56.28 | ManxPower | if you want an encrypted connection use a VPN |
| 23:56.39 | silvertip257 | nah SCTP isnt' encrypted |
| 23:56.54 | ManxPower | silvertip257: It is a media thing, not a VoIP thing |
| 23:57.10 | *** join/#asterisk nighty^ (n=nighty@72.0.72.123) |
| 23:57.19 | silvertip257 | no SCTP is a VoIP thing - used in VoIP installs/protocols |
| 23:57.34 | ManxPower | You are doing the equiv of asking about Windows Media Files on a Samba channel. |
| 23:57.51 | ManxPower | Whaever |
| 23:57.58 | silvertip257 | but yes I can understand if Asterisk in its infantcy will not support all VoIP protocols |
| 23:58.15 | silvertip257 | no it's not the equiv of that, but what the hell |
| 23:58.23 | ManxPower | Asterisk will never support all VoIP protocols. |
| 23:58.38 | silvertip257 | fine I give up |
| 23:58.43 | ectospasm | ManxPower: It could, if someone who wanted it bad enough would write a driver for it |
| 23:59.02 | silvertip257 | instead of finding what I need, we keep picking each others sentences apart .... |
| 23:59.17 | ManxPower | ectospasm: It would be pretty hard to support things like Skype, NetMeeting, CUSeeMe, VAT, etc |
| 23:59.28 | *** part/#asterisk silvertip257 (n=d009c2c9@unaffiliated/silvertip257) |
| 23:59.35 | ectospasm | Skype I agree, since it's a closed, proprietary protocol |
| 23:59.37 | ManxPower | silvertip257: what you need is to find a channel where people are familiar with what you are looking form. |
| 23:59.40 | [TK]D-Fender | Jerry! Jerry! Jerry! Jerry! |
| 23:59.42 | ectospasm | heh, there is chan_skype |