| 00:06.42 | bpun | angryuser, cool! will look into it! |
| 00:07.04 | dark_one | rpm: how are you provisioning your phones? |
| 00:08.46 | rpm | dark_one: ftp based |
| 00:09.23 | dark_one | humm, just updated a 330 to 4.0.0 sip 3.0.1 and got an new error 0x4000 |
| 00:09.55 | *** join/#asterisk moy (n=moy@189.169.62.96) |
| 00:10.46 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:10.56 | rpm | Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 <- bootrom.ld with sip load 3.0.0 |
| 00:12.03 | rpm | you more likely have a config error. |
| 00:13.07 | *** join/#asterisk angryuser (n=angryuse@88.140.144.209) |
| 00:14.14 | dark_one | the only config the phone downloads looks like this: http://pastebin.com/d253b28d7 |
| 00:14.35 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 00:14.37 | dark_one | the phones are not requesting any of the other files..... |
| 00:14.50 | rpm | i think thats wrong. |
| 00:15.56 | rpm | http://pastebin.com/m6102f060 |
| 00:16.48 | dark_one | rpm: its what polycom wrote in their white paper on configuration management |
| 00:20.17 | rpm | oh |
| 00:21.02 | dark_one | its also the configration that sipX's prvisioning tool generates |
| 00:21.11 | rpm | ah.. |
| 00:21.21 | *** join/#asterisk Corydon76-dig (i=beige@pdpc/supporter/bronze/Corydon76-home) |
| 00:21.22 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 00:21.39 | dark_one | however reading the admin manual seems to indicate that the sip application is having trouble downloading the config files |
| 00:25.49 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:25.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:33.30 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:37.05 | [TK]D-Fender | DarkoBetter to build yours off the samples provided with your firmware |
| 00:38.16 | dark_one | [TK]D-Fender: just looking at them, however the phone never trys to contact the provisioning server after it loads the sip app, it does the dhcp request then quits.... |
| 00:38.41 | [TK]D-Fender | dark_one: Go make sure those exact files exist with the right permissions. |
| 00:39.53 | dark_one | [TK]D-Fender: all files are 644 |
| 00:40.27 | dark_one | and all the dirs in the tree are 755 |
| 00:41.52 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 00:41.53 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 00:43.23 | [TK]D-Fender | dark_one: pastebin your log files and an "ls -la of your folder |
| 00:44.30 | dark_one | [TK]D-Fender: http://pastebin.com/d74cb163a |
| 00:47.50 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:49.23 | [TK]D-Fender | dark_one: seriously, go fix the user on all of those. |
| 00:50.01 | dark_one | [TK]D-Fender: done |
| 00:50.35 | [TK]D-Fender | dark_one: retry |
| 00:52.57 | javb | does zaptel 1.4 needs to get access to internet, ? |
| 00:53.05 | javb | can i install it without access to internet? |
| 00:53.20 | Qwell | you'll have to put the firmware in the source directory manually |
| 00:53.28 | Qwell | it downloads that at install |
| 00:53.36 | javb | Qwell, how can i do that? |
| 00:53.42 | javb | And what firmware? |
| 00:53.49 | dark_one | [TK]D-Fender: still no joy the sip application is not contacting the boot server after doing the dhcp request |
| 00:55.10 | javb | Qwell ? |
| 00:55.29 | Qwell | javb: what hardware do you have? |
| 00:56.27 | javb | Digium Wildcard TE212P |
| 00:56.42 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 00:56.50 | javb | Qwell: Digium Wildcard TE212P |
| 00:56.58 | Qwell | with or without echo can? |
| 00:56.59 | [TK]D-Fender | DarkOH... its not even TRYING it. |
| 00:57.08 | javb | Qwell, WITH echo can |
| 00:57.13 | Qwell | damn |
| 00:57.14 | [TK]D-Fender | dark_one: What did you fill in in the BootROM while its flashing on start? |
| 00:57.25 | Qwell | that's one of the things you need firmware for. |
| 00:57.49 | javb | Qwell ... :/ .... is it that mmm "hard" ? to install the firmwares? |
| 00:57.56 | dark_one | [TK]D-Fender: ? did not set any settings in the boot rom |
| 00:58.00 | Qwell | not when you have net access :) |
| 00:58.22 | Qwell | I'm not really sure how to download it or where to put it, right this second |
| 00:58.24 | [TK]D-Fender | dark_one: pastebin your dhcpd.conf |
| 00:59.19 | dark_one | [TK]D-Fender: http://pastebin.com/m68ce435b <-- at the end |
| 01:01.24 | [TK]D-Fender | dark_one: ok, I missed something there... yeah, you're trying to do this off TFTP. By default Polycom's are set to pick up via FTP. You'll have to reboot, go into the BR and set the mode for TFTP |
| 01:02.00 | dark_one | [TK]D-Fender: nope ftp... I have both ftp and tftp server set to the same root |
| 01:02.36 | [TK]D-Fender | dark_one: FTP will try to hit a "home" unix user... |
| 01:02.46 | [TK]D-Fender | DarkWhcih is by default the PlcmSpIp user |
| 01:03.09 | dark_one | [TK]D-Fender: yes, thats working for the boot rom stage |
| 01:03.42 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 01:04.18 | dark_one | i'm not seeing any traffic from the phone after sip boots and does the dhcp request |
| 01:05.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0e2ef77a830d3497) |
| 01:06.06 | dark_one | [TK]D-Fender: the manual says to set option 66 for the boot server option 66 is tftp-server-name in isc dhcpd |
| 01:07.17 | [TK]D-Fender | dark_one: yeah, your DHCP looks fine. |
| 01:07.46 | [TK]D-Fender | dark_one: double-checked to make sure your server daemons are running? |
| 01:09.17 | dark_one | [TK]D-Fender: the daemons are running (vsftpd, xinetd, dhcpd) but still the phone does nothing after the sip starts |
| 01:09.36 | [TK]D-Fender | dark_one: do you get the "welcome" splash? |
| 01:09.56 | dark_one | [TK]D-Fender: yes, but no network traffic |
| 01:10.17 | [TK]D-Fender | dark_one: Does it proceed to the "idle" screen from there? |
| 01:10.46 | dark_one | [TK]D-Fender: no error 0x4020 on a 650 and 0x4000 on a 330 and then they both just reboot |
| 01:11.34 | dark_one | actually the 650 just rebooted with 0x4000 |
| 01:11.53 | [TK]D-Fender | dark_one: ok, if SIP loads and then errors out I'd be betting that you're running generated configs that are not compatible with 3.X |
| 01:12.05 | [TK]D-Fender | dark_one: this is something yuo have to be very careful of with Polycom |
| 01:12.36 | dark_one | [TK]D-Fender: thing is the 650 is using the 00000000.cfg and sip.cfg and phone1.cfg from the distribution |
| 01:12.53 | [TK]D-Fender | dark_one: most minor revisions are compatible, but any 2nd place decimal version (1.5.X vs 1.6.X etc) is likely to break things |
| 01:13.16 | [TK]D-Fender | dark_one: 650's come stock with 2.1.2 IIRC |
| 01:13.27 | [TK]D-Fender | dark_one: 3.0 SHOULD break them. |
| 01:13.34 | dark_one | ? |
| 01:13.41 | [TK]D-Fender | dark_one: I strongly advise you attempt a rebuild from scratch |
| 01:13.58 | dark_one | [TK]D-Fender: this is a rebuild from scratch :( |
| 01:14.13 | [TK]D-Fender | dark_one: 2.X configs under 3.X firmware are very likely incompatible and likely to crash out with an error like that |
| 01:14.35 | [TK]D-Fender | dark_one: using SipX it seems. I can't vouch for its wuality. |
| 01:14.35 | dark_one | the configs are the defaults from the 3.0.1 distribution |
| 01:14.40 | [TK]D-Fender | quality* |
| 01:15.58 | dark_one | I just tried a new 650 with no mac.cfg, it dl's 00000000.cfg from the ftp server then errors with 0x4000 and reboots..... |
| 01:16.58 | dark_one | same with the 330 which has a copy of 00000000.cfg as mac.cfg |
| 01:18.56 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 01:22.11 | *** join/#asterisk settntrenz (n=joe@37.204.175.24.cfl.res.rr.com) |
| 01:23.30 | [TK]D-Fender | dark_one: You need to make sure the sip.cfg and so on is there and good though |
| 01:24.02 | *** join/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 01:25.22 | dark_one | [TK]D-Fender: sip.cfg is never transfered from the ftp or tftp server :( |
| 01:27.16 | dark_one | i cant see how it can be the problem if it never gets copied to the phone.... |
| 01:27.43 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:29.31 | [TK]D-Fender | dark_one: ok, I still advise fluch out the folder, re-extracting from scratch, not mucking with the fine names and starting minimalistic. |
| 01:32.22 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 01:32.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:36.02 | dark_one | [TK]D-Fender: the strange thing is that the same set of default configs used and booted from a FreeBSD server would casue the phone to go to the idle screen.... |
| 01:36.21 | [TK]D-Fender | dark_one: not sure what to say at this point... |
| 01:38.23 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 01:38.41 | silvertip257 | [TK]D-Fender: it's been a long day - I'll analyze the fw rules and see what's goin on - these embedded devices are rather unique/quirky |
| 01:39.21 | [TK]D-Fender | silvertip257: You're tring to turn a toaster into a convetion oven. Don't be surprised if you get burnt ;) |
| 01:40.19 | silvertip257 | [TK]D-Fender: har har har ... it's the darn firewall/vlans/bridging -- I'm almost sure of that. Heck I had to go back a few releases to get a version that would leave enough space for ASTERISK!!! ;-) I got burnt a lil on that one too! |
| 01:40.44 | dark_one | m***her ****ing pile of steaming **** it was the netmask........ |
| 01:40.56 | dark_one | for the 330 anyway..... |
| 01:41.25 | [TK]D-Fender | dark_one: that wasn't a clean class-C? |
| 01:41.35 | dark_one | [TK]D-Fender: no it was a /23 |
| 01:41.40 | [TK]D-Fender | dark_one: Don't get creative with CIDR.... |
| 01:41.53 | [TK]D-Fender | dark_one: I've seen picky devices before... |
| 01:42.41 | silvertip257 | hheh |
| 01:42.41 | dark_one | right time to try the real config for this phone... |
| 01:43.10 | silvertip257 | good luck fellas |
| 01:43.15 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 01:43.27 | dark_one | wooooohooooo |
| 01:48.25 | florz | any ideas as to how to replace all *s in a string with +es in dialplan code? |
| 01:48.47 | florz | (one that doesn't involve looping in the dialplan? =:-) |
| 01:54.58 | *** join/#asterisk Segnale007 (n=Segnale0@host141-4-dynamic.18-79-r.retail.telecomitalia.it) |
| 01:56.03 | [TK]D-Fender | florz: Loop it. Its what you've got. |
| 01:58.50 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
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| 02:07.58 | *** join/#asterisk bsaxon (n=bsaxon@179.sub-75-248-9.myvzw.com) |
| 02:09.04 | *** join/#asterisk javb (n=javb@190.6.142.16) |
| 02:09.24 | javb | hello, when trying to start asterisk 1.4, it keeps on restarting, is there i way i can see the reason ? |
| 02:09.55 | javb | maybe an output or something ? i had 1.2, and uninstalled it using : http://astrecipes.net/index.php?q=AstRecipes/Removing%20Asterisk ... and installed zap 1.4 and ast 1.4 |
| 02:10.00 | Gwayne | javb, it can be rights |
| 02:10.17 | javb | Gwayne... ? |
| 02:10.39 | javb | I did the same process im used to do with 1.2 ... |
| 02:10.50 | javb | ./configura ; make ; make install |
| 02:10.55 | *** part/#asterisk settntrenz (n=joe@37.204.175.24.cfl.res.rr.com) |
| 02:12.01 | Gwayne | javb, with what user do you start it ? |
| 02:12.15 | javb | root |
| 02:12.17 | javb | all the time. |
| 02:12.30 | javb | service asterisk start |
| 02:12.43 | *** join/#asterisk [intra]lanman (n=lanman@75-105-17-160.cust.wildblue.net) |
| 02:14.28 | bbryant | javb: start it with "asterisk -vvvvvvvvvvvgc" |
| 02:14.32 | bbryant | and pastebin the output |
| 02:16.32 | javb | bbryant, THANKS A LOT. . . a problem with zaptel! |
| 02:16.46 | bbryant | welcome |
| 02:24.40 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:25.42 | jaytee | I've setup * 1.4 on a test box at home and I've got a Polycom phone working for inbound and outbound calls and voicemail working but even though I've setup the voicemail.conf options to send me email with the voicemail attachment sendmail never sends it. |
| 02:26.23 | *** join/#asterisk ta^3 (n=tacvbo@189.146.188.205) |
| 02:30.03 | javb | jaytsee, maybe it is sending it, the problem is that your email is rejecting it due to spam policies around internet ? |
| 02:30.27 | jaytee | javb, I was thinking that might be the case |
| 02:30.46 | javb | i ve had that case. |
| 02:32.52 | javb | i spen like 15 minutes with my T1 disconnected... i think that my service provider may disable it ... is there a way i can notice that? even if zttool says NO ALARMS ? |
| 02:33.55 | jaytee | do you have smartjacks or does the T1 go through a CSU? |
| 02:43.51 | *** join/#asterisk bsaxon (n=bsaxon@72.sub-70-221-151.myvzw.com) |
| 02:44.29 | *** join/#asterisk autoditac (n=autodita@p579E1CBA.dip.t-dialin.net) |
| 02:44.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:45.18 | *** join/#asterisk admin0 (n=admin@bb121-6-1-64.singnet.com.sg) |
| 02:45.34 | *** join/#asterisk kash (n=J4CK@89.248.172.3) |
| 02:45.50 | kash | is there a way to have two servers with conferences bridge each other |
| 02:47.01 | troubled | is there a way to check the serial of an iaxy connect to the system? |
| 02:47.35 | kash | http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe#Mergingconferences |
| 02:48.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:50.13 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
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| 02:51.28 | [TK]D-Fender | kash: their "Originate" method is advisable. Equally functional are "call files". |
| 02:52.10 | [TK]D-Fender | kash: Only issue is that the bridge will have to be taken down manually or the channel will be kept ope on both sides indefinitely |
| 02:52.35 | kash | ok |
| 02:52.39 | kash | :) |
| 02:53.21 | javb | [TK]D-Fender, i installed Asterisk 1.4; sip phones are registrared, but when i make a call, i mean, dial an exten, asterisk DOESTN SAY anything in CLI, verbose and debug is on... it just says freeze, any idea? |
| 02:53.58 | [TK]D-Fender | javb: is SAYS something? please elaborate... |
| 02:54.19 | troubled | [TK]D-Fender: are you aware of any way to fetch the serial number of an iaxy from the console? |
| 02:54.36 | [TK]D-Fender | troubled: Nope, never used, never advise. |
| 02:54.41 | troubled | np, thanks |
| 02:56.45 | javb | [TK]D-Fender, i mean, i dont get anything in the "cli" |
| 02:56.51 | javb | not a single error, or debug |
| 02:56.54 | autoditac | : hi. we are experiencing severe problems with the combination of freepbx 2.4.1 and asterisk 1.4.21 (and a few minor versions before that). on ca. 20% of the calls, either inbound, outbound or local sip2sip calls, we get a connection but no or only one audio channel. no error messages in the asterisk logs. we don't use NAT. any hints? |
| 02:57.04 | [TK]D-Fender | javb: Enable SIP DEBUG and pastebin a failed attempt |
| 02:57.54 | [TK]D-Fender | autoditac: pastebin the CLI & SIP DEBUG output of a COMPLETE call attempt from beginning to end. |
| 02:57.57 | [TK]D-Fender | ~pb |
| 02:57.57 | 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 |
| 02:57.59 | kash | [TK]D-Fender: i made the call file, but it's not making the outgoing call |
| 02:58.17 | [TK]D-Fender | kash: And you haven't shown me anything. |
| 02:59.04 | jaytee | I'm trying to setup my voicemail to email the voicemail message. I don't get any console errors but in my maillog I see where it tries to send the mail and it has DSN: Service unavailable. |
| 02:59.08 | autoditac | [TK]D-Fender: wait a minute ... |
| 02:59.21 | kash | http://pastebin.ca/1075003 |
| 03:00.42 | [TK]D-Fender | kash: Now try to be THOOROUGH in what you show me. One tiny little excerpt isn't going to tell me anything. I see no CLI output, no debug, no file listings, no prrof of your implementing the call-file properly (not just the contexts) |
| 03:01.04 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 03:01.04 | kash | [TK]D-Fender: nothing happened. |
| 03:01.06 | kash | in cli |
| 03:01.46 | [TK]D-Fender | kash: You'd better eloborate a whole lot more on what you did. If youwere at verbose 10 and saw nothing I'd think you didn't deal with your call file properly |
| 03:02.07 | De_Mon | jaytee try sending a message normally, not using asterisk |
| 03:02.07 | kash | i put it into /var/spool/asterisk/outgoing |
| 03:02.19 | jaytee | De_Mon, trying that now |
| 03:03.29 | De_Mon | jaytee sounds like you're using sendmail, google has solutions to that all over the place. |
| 03:03.47 | [TK]D-Fender | kash: how? |
| 03:03.53 | jaytee | yeah, I'm reading a few of them now |
| 03:03.54 | kash | [TK]D-Fender: mv |
| 03:04.03 | [TK]D-Fender | kash: And your dialplan? |
| 03:04.08 | kash | [TK]D-Fender: what about it? |
| 03:04.09 | De_Mon | my favorite is "install postfix or exim" |
| 03:04.18 | [TK]D-Fender | kash: I can't validate a call files without validating the dialplan... |
| 03:04.45 | kash | well, the extension 10000 is a MeetMe |
| 03:04.52 | De_Mon | kash be nice to [TK]D-Fender he lost his mind reading abilities a few weeks ago |
| 03:04.53 | kash | extension 1111 is an IAX link |
| 03:05.27 | De_Mon | is reminded of houses favorite line, "people lie" |
| 03:05.40 | [TK]D-Fender | kash: I trust pastebin (mostly), not loose descriptions. |
| 03:06.18 | [TK]D-Fender | De_Mon: entirely true. Do doctors no less. How incredibly stupid. |
| 03:06.25 | [TK]D-Fender | To* |
| 03:07.21 | De_Mon | where does the to go? |
| 03:07.42 | De_Mon | ack burn notice came on tonite? |
| 03:07.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.03 | [TK]D-Fender | De_Mon: entirely true. To doctors no less. How incredibly stupid. |
| 03:08.14 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 03:08.18 | javb | [TK]D-Fender, ok solved it, but still with the same bug in 1.2, it was not detecting dtmf from cell phone, bu it was from other normal phone... added the "relaxdtmf" ... and it is working better, i mean detect more, but sometimes doesnt detect what i dialed... any idea? |
| 03:09.34 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 03:09.48 | [TK]D-Fender | javb: You are saying NOTHING. Random incorherent thoughts are coming out with no sense of context. First you say calls aren't coming in, not something about DTMF. PLease get a grip and come up with something complete. |
| 03:10.22 | [TK]D-Fender | javb: and "dialed" can mean several things. |
| 03:10.44 | De_Mon | ooh doctors lie to doctors, it didn't make sense till I read it |
| 03:11.15 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 03:12.56 | [TK]D-Fender | De_Mon: actually I meant patients lying to doctors. Its their health and they are lying to the people who are there to help them. |
| 03:13.20 | implicit | not all doctors want to help you |
| 03:13.26 | implicit | some want to screw you over |
| 03:13.35 | implicit | one of my friends was a doctor like that |
| 03:13.53 | javb | [TK]D-Fender, Fine, forget about everything before... see: I have an asterisk box, with a T1. I have an IVR working on it, if you call it from a regular(whatever but not a cellphone) phone (out side the *pbx) and you press any key, Asterisk respond inmidately, in the other hand, if i dial from my cell phone, it will NOT LISTEN the dtmf im sending.. after googling and testing, i added the option (zapata.conf) relaxdtmf = yes, and after that, it will listen, b |
| 03:13.53 | javb | ut not very accurate.. if you dial an exten with 3 digits, it may listen to the first two, or maybe 3, repeating the first digit pressed . . . thats the issue im having now . . . |
| 03:15.01 | [TK]D-Fender | javb: all calls coming in over T1? |
| 03:15.13 | javb | yes |
| 03:15.23 | [TK]D-Fender | javb: Because your loose description could mean that cells come in differently |
| 03:15.32 | *** join/#asterisk bsaxon (n=bsaxon@82.sub-70-220-57.myvzw.com) |
| 03:15.43 | javb | just T1 |
| 03:16.02 | [TK]D-Fender | javb: 2 key factors in dtmf : origin signal quality & gain. you can play around with your rxgain on your zapata channels and that can help clear things up |
| 03:16.28 | [TK]D-Fender | javb: if DTMF is inband from the cell itself expect things to suck on occasion. |
| 03:17.15 | javb | Mmmm, you mean that i may be an specific cell problem ? |
| 03:17.25 | javb | it* |
| 03:19.13 | [TK]D-Fender | javb: entirely possible. |
| 03:19.28 | [TK]D-Fender | javb: first 2 things to test is relaxdtmf, and rxgain. |
| 03:19.37 | [TK]D-Fender | javb: up the gain a nothch see if it helps |
| 03:21.38 | javb | rxgain = 10 ; relaxdtmf = yes .... same, sometime detect it... but what is worse, if you press or dial an exten bigger than 1 digit, the IVR will confuse them, and mess the final exten, it also just select the las digit pressed. |
| 03:23.22 | javb | If i increse the rxgain more, it will get an "s" exten when a call is comming |
| 03:24.45 | [TK]D-Fender | javb: What card, and what signalling? |
| 03:25.55 | javb | Digium TE212P ; 2 T1 ; With ECHO CAN MODULE ; ami,d4 ' em_w |
| 03:26.53 | [TK]D-Fender | javb: ugly signalling... |
| 03:26.55 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 03:27.03 | *** join/#asterisk tengulre (n=tengulre@125.70.73.201) |
| 03:27.12 | [TK]D-Fender | javb: analog T1, DTMF DIDs... bleh |
| 03:27.33 | javb | [TK]D-Fender, ... ? |
| 03:27.52 | javb | [TK]D-Fender, it doesnt depend on me ... i think .. |
| 03:27.53 | [TK]D-Fender | javb: Couldn't get real PRI signalling? |
| 03:28.23 | [TK]D-Fender | javb: And if your gain is already at 10 you might be glaring it. have you tried lowering it as well? |
| 03:28.34 | moy | korihor: I just reviewed your patch and I am about to commit it, thanks again! |
| 03:28.48 | javb | [TK]D-Fender, mmm, no, this guys had this T1 over a year on Nortel Norstar, with an IVR working |
| 03:29.28 | [TK]D-Fender | javb: what */zaptel ar you on exactly? |
| 03:30.14 | javb | u mean version ? |
| 03:30.41 | [TK]D-Fender | clrealy |
| 03:30.45 | [TK]D-Fender | clearly. |
| 03:30.49 | javb | 1.4 |
| 03:32.49 | javb | If i dial 312, it get 332, so, the first twice and the last ... and this is JUST happing when the cellphone originates the call ! |
| 03:34.47 | *** join/#asterisk chendy (n=chatzill@58.251.111.151) |
| 03:35.25 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 03:36.06 | [TK]D-Fender | javb: maybe the cell is shit |
| 03:36.17 | [TK]D-Fender | javb: did you try to reduce the gain? |
| 03:37.16 | javb | |:/ ... yes, had put the gain to 0,1,2,5,9,10 ... 15, 20... the problem is that the T1 is working great in a Nortel NorStar which is right next to the server . . . |
| 03:37.35 | javb | Kind of hard to make people understand that maybe is the Cell ! |
| 03:49.59 | chendy | did asterisk 1.4.x 's queue realy support call-limit as defined in sip.conf ? |
| 03:52.32 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:53.14 | chendy | any work around or patch? |
| 03:56.59 | bbryant | chendy: what do you mean? |
| 03:57.18 | *** join/#asterisk adorah (n=Michael@87.69.57.141.cable.012.net.il) |
| 03:59.40 | *** join/#asterisk FoneHome (n=ben@c-69-180-119-205.hsd1.fl.comcast.net) |
| 04:00.46 | FoneHome | [TK]D-Fender: Thank you for your help earlier TK, with the outbound blf notifications. unfortunately it did not work. |
| 04:01.13 | chendy | i have setup limitonpeer=yes and call-limit for each sip friend in sip.conf, but the app_queue.c still complaint reject call attempt to sip interface and infor me to read UPGRADE.txt |
| 04:02.00 | FoneHome | REQUESTING BLF HELP! Outbound calls to not register BLF notifications but inbound calls do.. any advice or help is greatly appreciated. |
| 04:03.55 | mosty | fonehome: limitonpeers=yes in general section of sip.conf |
| 04:05.14 | FoneHome | mosty: yes i have limitonpees=yes as well as call-limit=50 |
| 04:05.42 | FoneHome | I can provide copies of all configs or sections |
| 04:06.03 | chendy | why 50? there are sip phones |
| 04:06.45 | FoneHome | yes they are all sip phones. Grandstream GXP-2010 phones |
| 04:07.14 | FoneHome | there is 10 phones in the office. and 10 (VERY PISSED) people because then can not tell when someone is on the phone |
| 04:08.06 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 04:10.13 | mosty | what version of asterisk? |
| 04:11.44 | FoneHome | Asterisk 1.4.20.1 |
| 04:13.16 | mosty | it works on 1.4.21.1 here |
| 04:14.27 | FoneHome | would it be possible for you to share your configs? |
| 04:14.43 | FoneHome | maybe i can find whats missing. if anything |
| 04:16.17 | korihor | moy: ok :) , you are welcome |
| 04:17.24 | moy | korihor: do you work a lot with Asterisk? do you work for some telco? |
| 04:19.06 | korihor | moy: yes, here in Venezuela i make a company for Premium Service on telefony, sms, mms, voice, etc :) |
| 04:19.17 | *** join/#asterisk hads (n=hads@120.138.17.30) |
| 04:19.40 | moy | nice |
| 04:20.09 | korihor | moy: I make that with other 2 university freinds :p |
| 04:20.51 | FoneHome | korihor: do you happen to have working experience with BLF setups? |
| 04:21.45 | moy | korihor: cool!, hey, are you going to Astricon? |
| 04:22.50 | korihor | moy: no :( , I haven't VISA for USA |
| 04:23.15 | korihor | moy: and you? |
| 04:23.52 | moy | too bad, I'd have liken to meet you there, yeah, I will be giving a small talk about openr2 |
| 04:24.50 | FoneHome | what is openr2? |
| 04:24.54 | korihor | moy: nice :) , I will try go |
| 04:25.38 | moy | FoneHome: library that implements MFC/R2 signaling ... pretty much like libmfcr2, but does not depend on Unicall framework |
| 04:25.50 | korihor | moy: a question. why don't port r2 variant found in Unicall to libopenr2? |
| 04:26.10 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 04:26.43 | moy | I have not checked if libmfcr2 is already LGPL, is it? when I started the project coppice had not released his stuff for LGPL, so that was a non technical limitation |
| 04:27.05 | korihor | FoneHome: I worked with few a Snom and Grandstream BLF |
| 04:28.10 | moy | now coppice at least released SpanDSP as LGPL, that helps a lot, but have not checked for their other software stacks, but, since I believe he is working towards implement unicall in FreeSwitch, probably he has already done that |
| 04:29.55 | korihor | moy: coppice tell me something about that, few weeks ago. Unicall for Freeswitch :) nice idea |
| 04:30.23 | moy | cool indeed, I hope to have some time the next weeks to put openr2 in FreeSwitch as well, in their openzap stuff |
| 04:30.53 | moy | korihor: are you using some other modules of Unicall? aside of the r2 module? |
| 04:31.04 | mosty | FoneHome, i had the exact issue you had, and when i made the change i mentioned before, it fixed it |
| 04:31.14 | FoneHome | korihor: I am using GXP 2000/2010 phones but i only get BLF notifications one way .. that is on incoming calls. Any out bound calls do not register. I am having the damndest time finding help on getting out bound blf to work. |
| 04:31.27 | korihor | moy: exist other? :p |
| 04:31.40 | FoneHome | mosty: updating from 1.4.20 to 21? |
| 04:32.24 | moy | korihor: lol ... unfortunately I don't know why coppice did not release their other modules (he told me he has FXO, SS7, PRI and probably others), had he released those, probably Unicall would have more users |
| 04:32.27 | mosty | FoneHome, no i had the undesired behaviour on 1.4.21.1 and fixed it with the setting in sip.conf |
| 04:33.01 | FoneHome | mosty: sadly that has not worked for me. :( |
| 04:33.04 | korihor | moy: I can help you with openr2 for freeswitch and openzap? |
| 04:33.30 | moy | korihor: sure, be my guest if you have the time :) |
| 04:33.39 | FoneHome | I notice on the CLI that the hint is never even passed i am wondering if its a dialplan issue. |
| 04:35.02 | korihor | moy: yes, I have :) I guess :p |
| 04:35.30 | moy | korihor: sounds great, really, I'd appreciate it, I have been trying to make me some time to do it, but my "real" job had me tied |
| 04:36.12 | korihor | moy: are you working for IBM rigth? |
| 04:37.05 | moy | yeah, for IBM (Immense Boring Meetings) |
| 04:37.21 | korihor | moy: jajajajaja |
| 04:37.52 | pputman | IBM is a horrible company to work with |
| 04:38.08 | pputman | never let their technicians in your server room unaccompanied |
| 04:38.19 | moy | korihor: I wish were that funny :) ... I hope I can get another job soon ... pputman: kind of, it depends on the country and area though |
| 04:38.45 | korihor | moy: sangoma people talk with you about openr2? |
| 04:38.52 | moy | pputman: but yes, in general, there is too much protocol and burocracy |
| 04:39.09 | pputman | moy, well at least in my experience with storage devices. I'm sure they have some intelligent people, but I've just had too many IBM techs go out to remote sites and start ripping out hard drives in live systems without telling anyone, etc... |
| 04:39.26 | moy | korihor: yeah, they were the first ones in supporting this development |
| 04:39.45 | korihor | moy: nice. congrats |
| 04:40.35 | moy | korihor: thanks ... btw, have you ever worked with PIKA cards? I just talked with one of their devs some days ago, he told me now PIKA cards also support R2 |
| 04:40.40 | korihor | in Venezuela, IBM techs guys are sell guys :) |
| 04:41.54 | korihor | korihor: I haven't one on my hands for now :(. I will buy one for testint it |
| 04:42.06 | moy | korihor: In Mexico they are too, it's just that they are called engineers and fix bugs ... the thing I don't like is that most IBM projects here in Mexico are just maintenance of crappy code they sent us from the states |
| 04:42.08 | korihor | moy: PIKA are canadian rigth? |
| 04:42.22 | moy | yeah, they are in Ottawa, Sangoma is in Toronto |
| 04:43.52 | korihor | ok |
| 04:44.38 | mosty | moy, in the states it's probably just crappy code they were sent from india ;) |
| 04:44.50 | korihor | moy: I make a company because don't like me the boss :p |
| 04:45.27 | korihor | moy: authority problems , jajajajaja :) |
| 04:45.32 | moy | mosty: hahaha, yeah, likely :) |
| 04:46.23 | moy | korihor: haha, yeah, sometimes sucks, but working at the right company can be cool too, as long as your boss is a technical guy who understand programmers |
| 04:46.35 | moy | korihor: are you still studying? |
| 04:46.50 | moy | university? high school? how old are you? |
| 04:47.00 | korihor | moy: no, I'm graduate |
| 04:47.24 | korihor | moy: from here http://www.usb.ve/ |
| 04:47.56 | korihor | moy: 25 yaer old :) |
| 04:48.02 | korihor | moy: and you? |
| 04:48.09 | moy | hey, me too |
| 04:48.26 | moy | 25, I will be 26 in 2 months more |
| 04:48.32 | korihor | moy: nice |
| 04:49.42 | korihor | moy: me be in the future we can make a few business :p |
| 04:49.57 | moy | you bet |
| 04:50.48 | korihor | moy: my english suck :( but I'm learning |
| 04:52.16 | moy | korihor: I think you are doing just fine, I assume you have never been in the states (since you told me you don't have a visa)? |
| 04:52.48 | korihor | moy: thats rigth :( |
| 04:53.25 | korihor | moy: libmfcr2 is GPL |
| 04:54.14 | moy | korihor: as I expected, until steve has ready his unicall module for FreeSwitch I don't think he is going to LGPL it, so I have to wait before taking the tone definitions from there |
| 04:55.10 | korihor | moy: ok |
| 04:55.41 | korihor | moy: astricon is in Arizona rigth? |
| 04:55.48 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:58.06 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 04:58.20 | moy | korihor: yup, near to Phoenix ... Glendale Arizona ... don't ask me why they did it there :P ... I suppose they could have chosen a better place |
| 04:58.47 | moy | s/did/will do it |
| 04:59.20 | korihor | moy: I have a frinde in Arizona :) |
| 04:59.44 | korihor | moy: http://www.patriots.com/team/index.cfm?ac=playerbio&bio=31812 :p |
| 05:00.33 | moy | korihor: yay ... where did you meet him? |
| 05:00.44 | korihor | moy: he is Todd Mortensen and your wife Lori :) http://www.missamerica.org/competition-info/national-contestants.aspx?state=Arizona&year=1999 |
| 05:01.15 | moy | korihor: ur kidding me? |
| 05:01.22 | korihor | moy: here in Caracas, Venezuela. He live here for 2 year. good guy |
| 05:01.30 | moy | cool |
| 05:01.31 | korihor | moy: don't is true |
| 05:02.09 | korihor | moy: yeah, he call me sometime |
| 05:02.58 | moy | cool indeed ... I have been wishing visit Venezuela for a while, are they near to the galapagos? |
| 05:03.53 | moy | korihor: I meant, is Venezuela near of the galapagos |
| 05:03.55 | moy | ? |
| 05:04.29 | korihor | moy: no, Ecuador |
| 05:05.13 | moy | korihor: ah, yeah, my Geography notion sucks as you can see |
| 05:05.33 | korihor | moy: no problem |
| 05:06.51 | korihor | moy: are you working with ss7? |
| 05:07.39 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:10.26 | moy | korihor: not really, I have a book about it here though, it's in my TODO list of things to learn |
| 05:10.50 | korihor | moy: me too |
| 05:10.54 | moy | korihor: do you have a specific need or just curious about it? |
| 05:12.06 | korihor | moy: I waiting for access a Ericsson central in CANTV :p |
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| 05:12.39 | korihor | moy: i have freinds in side this telco :) |
| 05:13.29 | moy | korihor: I see, cool, so have you already played with libss7? |
| 05:13.38 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 05:13.38 | korihor | moy: I believe Ericsson have 2 books of ss7. blue book and white book |
| 05:14.03 | *** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
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| 05:14.14 | korihor | moy: no, i will try |
| 05:15.04 | korihor | moy: first I need learn a few theory about that |
| 05:15.48 | moy | korihor: indeed, me too ... I gotta go dude, it's late (0:15AM here) and my wife is yelling at me :) |
| 05:16.12 | moy | korihor: it has been nice to chat with you, see you later! |
| 05:16.25 | korihor | moy: ok bro, i see you |
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| 05:29.32 | *** join/#asterisk PakiPenguin (n=pkbill@linuxpakistan/admin/pakipenguin) |
| 05:29.36 | PakiPenguin | hello everyone |
| 05:30.02 | *** part/#asterisk _nn (n=nn@unaffiliated/nn) |
| 05:31.37 | PakiPenguin | can anyone help me with a registration issue please |
| 05:32.05 | FoneHome | i can try to help you out PakiPenguin |
| 05:32.47 | PakiPenguin | okay i have a sip server to which xpro is registering fine , but asterisk doesnt , and gives me wrong password , here's the successful log from xpro http://pastebin.ca/1075141 |
| 05:36.41 | FoneHome | PakiPenguin: do you have a register line in your register line correct |
| 05:36.44 | FoneHome | Format: |
| 05:36.44 | FoneHome | <PROTECTED> |
| 05:36.45 | FoneHome | <PROTECTED> |
| 05:36.45 | FoneHome | <PROTECTED> |
| 05:37.38 | FoneHome | that would be in the sip.conf file |
| 05:41.14 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 05:42.26 | *** join/#asterisk pakipenguin (n=pkbill@linuxpakistan/admin/pakipenguin) |
| 05:42.29 | pakipenguin | hello |
| 05:42.32 | pakipenguin | FoneHome, sorry |
| 05:42.39 | pakipenguin | lost connectivity |
| 05:43.44 | bbryant | pakipenguin: what asterisk version? |
| 05:45.15 | pakipenguin | 1.4.18 |
| 05:45.32 | pakipenguin | bbryant, did you see my xpro's succesful logs? |
| 05:46.54 | mosty | pakipenguin, look at asterisk's logs? |
| 05:47.05 | pakipenguin | yeah , holdon , pasting the debug |
| 05:48.20 | pakipenguin | mosty, bbryant http://pastebin.ca/1075165 <-- thats what asterisk has to say |
| 05:49.17 | FoneHome | PakiPenguin: do you have a register line in your register line correct |
| 05:49.44 | FoneHome | PakiPenguin: do you have a register line in your sip.conf correct |
| 05:49.54 | pakipenguin | yeah |
| 05:49.55 | mosty | either your username or password are wrong in your register line |
| 05:49.57 | bbryant | pakipenguin: looks like the password is missing from the register line |
| 05:50.15 | pakipenguin | register => 0218000342@wateen.net:PWD098@58.27.240.22:9060 |
| 05:52.26 | pakipenguin | this is my register line |
| 05:53.42 | FoneHome | i tried pinging your 58.27.240.22 server with no response |
| 05:53.52 | bbryant | pakipenguin: the sip server is rejecting the call because the domain is not allowed |
| 05:54.04 | bbryant | s/call/registration/ |
| 05:54.45 | pakipenguin | bbryant, but the same configuration works perfectly alright in xpro , so i bet its the registration string , thats messing it up or something |
| 05:55.38 | FoneHome | <good night guys |
| 05:56.35 | bbryant | pakipenguin: are you running xpro on the same computer that asterisk is trying to registering from? |
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| 06:11.09 | PakiPenguin_ | hmm |
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| 06:16.13 | PakiPenguin_ | hmm |
| 06:16.29 | PakiPenguin_ | anyone who can help me out ? |
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| 06:18.05 | gramulhaozin | anyone ever seen FXO PCI MASTER ABORT ? |
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| 06:45.19 | PakiPenguin_ | hmm |
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| 07:11.25 | CoffeeIV | is there any free way to get a POTS number to go an asterisk VoIP server ? (legal of course) Even just for testing ? |
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| 07:30.06 | PakiPenguin_ | CoffeeIV, ipkall.com |
| 07:30.22 | CoffeeIV | thanks, I'll check it out |
| 07:33.18 | PakiPenguin_ | why am i stuck with the worst sip servers in the world :( |
| 07:33.22 | PakiPenguin_ | *sigh* |
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| 08:30.42 | Vec | Would it be a bad idea to run an IAX trunk with no hardware timing source with Asterisk 1.4 ? |
| 08:33.02 | mvanbaak | it's impossible |
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| 08:37.21 | mog | i think he just means iax2 peer not iax2 trunking which is impossible without timing source on 1.4 |
| 08:37.27 | mog | gnite |
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| 08:41.43 | gormux | hello |
| 08:42.11 | gormux | I have a strange thing : I have 15 phones, connected to an * server |
| 08:42.36 | gormux | and some connects via UDP without a problem, and others can only connect via TCP |
| 08:42.53 | gormux | any idea of a reason ? |
| 08:47.01 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 08:48.59 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:49.09 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
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| 08:53.08 | *** join/#asterisk hadronzoo (n=user@adsl-70-250-184-20.dsl.rcsntx.swbell.net) |
| 08:53.47 | jack_sparo | who knows a good solution for asterisk billing solution? |
| 08:53.56 | hadronzoo | does anyone have any recommendations about the best 4-line voip phone to use with asterisk? |
| 08:54.26 | mvanbaak | jack_sparo: a2billing |
| 08:54.41 | mvanbaak | ~phones |
| 08:54.41 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 08:55.40 | many | such as what? |
| 08:55.44 | hadronzoo | So, no Grandstream, then |
| 08:56.18 | many | 's surprised that cisco and linksys come before snom. and what is aastra |
| 08:57.37 | *** join/#asterisk mrnick (n=basement@kulnet-nat-2.kulnet.kuleuven.be) |
| 08:58.31 | mrnick | hi |
| 08:58.40 | hadronzoo | Thanks mvanbaak |
| 08:59.04 | Gary | we use the Linksys SPA-942's here and they are great imho |
| 09:00.44 | many | well, wouldnt consider snom great but good enough |
| 09:01.10 | hadronzoo | Gary: How many lines does it have? The referring link states "2(4)" |
| 09:01.13 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-5432689e9e6fb9bd) |
| 09:02.46 | Gary | hadronzoo: it has four, each with a button (and light) |
| 09:02.59 | Gary | we only use one number per handset though |
| 09:03.07 | mrnick | is there a way to redirect dialed channels to an ivr on hangup, sorry i'm not very good at this yet... |
| 09:03.27 | hadronzoo | Gary: Thanks. I guess the 2 referred to the two ethernet ports. |
| 09:05.07 | Gary | yeah, you have a mini network switch, 10/100 ont he bottom, so the pc can use the other port, saves network ports |
| 09:06.02 | many | normally you cant use both ethernet ports for outbound sip, let alone configure them individually as sip line |
| 09:06.15 | hadronzoo | That makes sense, so it acts as a switch. |
| 09:06.28 | Gary | yeah, one is marked WAN and the other PC |
| 09:06.45 | many | it usually does, so you can put your phone between your network and your pc and keep your cabling clean |
| 09:06.54 | Gary | the handset is cool though, we get ours to auto upload the company logo on to them |
| 09:06.59 | many | usually = thats what most phone vendors do |
| 09:07.08 | hadronzoo | Easy enough |
| 09:07.40 | many | kinda like some keyboards have usb ports to directly attach your usb mouse |
| 09:08.28 | hadronzoo | Are there hold and transfer features present? |
| 09:09.01 | hadronzoo | I think I just answered my own question by reading the linksys site |
| 09:09.16 | *** join/#asterisk mmisiak (n=mmisiak@212.191.72.227) |
| 09:09.22 | mmisiak | Hi |
| 09:09.33 | mmisiak | I have problem with CDR for Call Transfer |
| 09:10.11 | mmisiak | so A calls B and then B transfers call to the C |
| 09:10.38 | mmisiak | as a result I recive only 1 CDR A to C |
| 09:11.03 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-176b91f73e0a8f4b) |
| 09:11.04 | mmisiak | and I want to have a A->B and A->C CDR |
| 09:11.28 | mmisiak | do you know what is the problem ? |
| 09:11.34 | mmisiak | thx for help |
| 09:17.57 | *** join/#asterisk joobie (n=joobie@joobie.org) |
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| 09:55.13 | Vec | @mog thanks |
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| 10:06.48 | gramulhaozin | hey |
| 10:06.50 | gramulhaozin | guys |
| 10:07.07 | mvanbaak | dont forget the gals |
| 10:07.07 | gramulhaozin | ever had a problem with NO AUDIO on the IVR but I get audio working for PHONE 2 PHONE |
| 10:07.16 | gramulhaozin | ? |
| 10:07.23 | gramulhaozin | anyone ever had a problem like that ? |
| 10:07.34 | gramulhaozin | I'm using Cisco 7940 phones |
| 10:07.40 | mvanbaak | gramulhaozin: do you have an Answer() call before you start the IVR ? |
| 10:07.53 | gramulhaozin | I'm using FreePBX |
| 10:07.58 | gramulhaozin | it works out of the box already |
| 10:08.00 | mvanbaak | ~freepbx |
| 10:08.01 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:08.02 | gramulhaozin | but not in that phone |
| 10:08.18 | gramulhaozin | mvanbaak: it's not a freepbx issue, it's an issue in the phone configuration |
| 10:09.06 | gramulhaozin | It's a sip configuration issue |
| 10:10.45 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 10:12.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:12.27 | gormux | re |
| 10:12.47 | gormux | is there a way to make * use only one interface ? |
| 10:13.12 | gormux | err only one IP addr |
| 10:13.33 | gormux | because i have an alias that i'd like to use rather than the "main" ip address |
| 10:14.02 | gormux | and my phones connects trough that IP, but * responds through the other ip |
| 10:18.06 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 10:19.02 | joobie | hey boys.. anyone got a good article for setting up asterisk? i have 2 polycom 320's i want to setup.. and have 1 sip account with a provider.. wanted to get asterisk to conect to the sip provider and get the phones to connect to asterisk |
| 10:22.52 | mrnick | is there a way to redirect dialed channels to an ivr on hangup? |
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| 10:48.18 | gnorbert | Hi, does somebody know, how can I play a wav file for a meetme conference? |
| 10:48.22 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 10:49.18 | gnorbert | That everybody should hear it (And of course it shouldn't be done from a .conf file) |
| 10:50.57 | viraptor | hi - can I access the original INVITE's 'From:' domain in an AGI script somehow? I've seen SIPDOMAIN variable, but that's for destination only... |
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| 11:04.49 | *** join/#asterisk taggy (i=75c16295@gateway/web/ajax/mibbit.com/x-8193a55311e11848) |
| 11:05.40 | taggy | hi guys ,i have a few questions abt asterisk . Can asterisk be used to create a conference call .dial in to a number andlet everyone joinin conversation . what kind of setup does it require. it wu dbe great if anyone can respond |
| 11:09.29 | gnorbert | taggy: It's a meetme conference, you have to edit sip.conf, meetme.conf, extensions.conf. |
| 11:09.39 | gnorbert | http://www.voip-info.org/ Is good to start, I think... |
| 11:09.50 | gnorbert | But I am beginner too. :) |
| 11:11.44 | taggy | <PROTECTED> |
| 11:13.32 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:14.46 | gnorbert | taggy: At least I tried, that is a nice thing too. :))) |
| 11:16.59 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 11:17.07 | pputman | asterisk 1.4 does not support srtp does it? |
| 11:18.13 | taggy | wow gnorbert great ! that lets u create simultaneous many conferences ? |
| 11:18.24 | *** join/#asterisk kippi (n=kippi@untrust-gct.equinoxit.net) |
| 11:18.26 | kippi | hey |
| 11:18.39 | kippi | should this stop asterisk gracefully? http://pastebin.com/m3992f847 |
| 11:18.45 | pputman | taggy, yeah you can. |
| 11:19.56 | pputman | gnorbert, I would take voip-info with a grain of salt. There's lots of good information on there, but also some that's outdated |
| 11:22.16 | gnorbert | pputman: I found it helpfull, I'm sure, there is better site, however I could understand Asterisk (As much as I understand it) from there most easily. |
| 11:23.26 | *** join/#asterisk implicit (n=bayan@ip68-4-97-211.oc.oc.cox.net) |
| 11:23.28 | pputman | there really isn't a better site, I use it a lot, I'm just saying that don't take everything there at face value. |
| 11:23.48 | gnorbert | Does somebody know, how can I play a sound file started from CLI? |
| 11:23.56 | pputman | the book is very good too |
| 11:23.59 | pputman | ~book |
| 11:23.59 | 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 |
| 11:24.00 | gnorbert | In a meetme conference. |
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| 11:25.02 | gnorbert | pputman: The book was good also, but I think it's a bit long to read it, if you are looking for a concrete thing. |
| 11:25.18 | pputman | yeah |
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| 12:00.29 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 12:00.30 | raz | hi guys |
| 12:01.12 | raz | i got the "simple ivr example" from www.the-asterisk-book.com to work |
| 12:01.14 | raz | now i want more :D |
| 12:01.50 | raz | can anyone point me to a tutorial that teaches me how make asterisk connect to a SIP account and answer incoming calls from there? :) |
| 12:02.34 | *** part/#asterisk gormux (n=julien@ks361262.kimsufi.com) |
| 12:03.02 | *** join/#asterisk Hyphenex (n=User@60-241-154-242.tpgi.com.au) |
| 12:03.54 | Hyphenex | G'Day, I'm looking for getting a rough guide about how much it'd cost to build a server to support around 12-13 people with a gateway to POTS for outbound calls |
| 12:04.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.37 | gr0mit | Hyphenex, where POTS = ISDN2, ISDN30 or analogue? |
| 12:06.10 | Hyphenex | I'm not too sure |
| 12:06.23 | Hyphenex | I just thought a card would do all of them? |
| 12:06.24 | gr0mit | ah!! |
| 12:06.26 | gr0mit | nope |
| 12:06.29 | Hyphenex | I'm kind of new to this |
| 12:06.32 | gr0mit | different cards |
| 12:06.49 | Hyphenex | Do they differ in price much, depending on what system we have? |
| 12:07.13 | Hyphenex | and does it help if I tell you the system is fairly new? |
| 12:07.24 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 12:07.24 | gr0mit | yup - a card for ISDN2 can set you back from £20 to £500 |
| 12:07.49 | gr0mit | important thing is how you connect to the PSTN |
| 12:08.01 | gr0mit | age is not really an indication |
| 12:09.35 | gnorbert | Does somebody know, how can I play a sound file in a meetme conference with asterisk from CLI? |
| 12:11.07 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:11.28 | Hyphenex | gr0mit, Ahh |
| 12:11.49 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:11.51 | Hyphenex | gr0mit, What if the company is willing to re-work how the phone systems work? |
| 12:12.02 | Hyphenex | wait, Can it be ISDN if we get ADSL through it? |
| 12:12.03 | *** join/#asterisk heison (n=heison@209.195.83.65) |
| 12:12.03 | gr0mit | then definitely go for ISDN |
| 12:12.31 | gr0mit | some telcos will deliver ADSL over isdn. BT will not. |
| 12:12.52 | gr0mit | and if you go for an E1 (ISDN-30) you will never get adsl over it. |
| 12:13.09 | gr0mit | what is the function of the office? |
| 12:13.11 | Hyphenex | so I've probably got ISDN |
| 12:13.16 | gr0mit | is it a call centre? |
| 12:13.33 | gr0mit | a clue: do you have direct dial in to extensions? |
| 12:13.54 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:13.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:14.04 | gr0mit | if 'yes' you will certainly have ISDN |
| 12:14.27 | Hyphenex | Yeah, I know the office I'm at has a direct number I can get into it |
| 12:14.44 | Hyphenex | but I still have to push '0' to get a dial tone before I can dial out |
| 12:14.57 | gr0mit | but does each extension on the pbx have their own direct dial number? |
| 12:15.05 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 12:15.27 | Hyphenex | gr0mit, well, I think so (I know for a fact mine does, and I'm not that important, so I can't picture me being any different from other people) |
| 12:16.05 | gr0mit | ok, so you either have one or more ISDN2 or an ISDN30 |
| 12:16.35 | gr0mit | we have an ISDN30 in our Melbourne office with 10 channels |
| 12:17.13 | Hyphenex | That's cool. Probably what we have then :) |
| 12:17.41 | Hyphenex | so 500 pounds for a good card to interface with the ISDN line hey? |
| 12:18.08 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 12:18.21 | gr0mit | approx, yes |
| 12:18.53 | Hyphenex | we could do away with it all together, but we're going to be hosting terminal services, so I'm a bit worried about the extra bandwith going through a VoIP provider and the QoS that would provide (on both the Terminal services and VoIP sides) |
| 12:19.13 | gr0mit | well get two adsl |
| 12:19.24 | gr0mit | one for voip, the other for the rest of the traffic |
| 12:20.48 | gr0mit | you can run 6-8 voip calls through a UK adsl line with G726 codec |
| 12:20.56 | gr0mit | uplink is 448 |
| 12:21.01 | gr0mit | downlink varies |
| 12:21.11 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-74-16.vif.net) |
| 12:21.26 | mrnick | is there a way to redirect dialed calls to an ivr on hangup or transfer them in some way? |
| 12:21.54 | Hyphenex | Hmm, that might be even more expensive then the ISDN card though, no? |
| 12:22.07 | gr0mit | no idea what Telstra charge you |
| 12:22.21 | Hyphenex | Heh :P |
| 12:22.31 | Hyphenex | well, I'll look into that later then :P |
| 12:23.26 | many | Hyphenex: how man voice channels do you have on how many physical ports? |
| 12:24.32 | many | mrnick: if the call is inbound, just continue the dialplan |
| 12:24.42 | gnorbert | Hope dies last...:) Does somebody know, how can I play sound in a meetme conference with a command at CLI? |
| 12:24.48 | many | when your internal phone hangs up, itll continue in the dialplan |
| 12:25.13 | mrnick | many: thx for the respons, but unfortunately it's outbound, i've been searching for weeks (as i'm not an expert...) |
| 12:25.18 | Hyphenex | many, Don't know, but if we've got a dial in number for each person, would that not mean we've got plans for each person now? |
| 12:25.28 | many | mrnick: then maybe the 'h' extension helps you |
| 12:25.34 | many | iam not too sure wether its the case |
| 12:26.04 | Hyphenex | We're paying off our current phone system, but I think it works out to be about $20,000 which is WAAAY to expensive, so I think we were looking at replacing it all toegether if VoIP could work out cheaper (considering we'll be setting up the same system at 4 other sites to also reduce the cost of calling each site) |
| 12:26.07 | many | Hyphenex: usually you get routed a block in that case, which doesnt mean much about the physical |
| 12:26.16 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 12:26.21 | Hyphenex | Ahhk |
| 12:26.25 | Hyphenex | I'm not sure then sorry |
| 12:27.18 | many | Hyphenex: in isdn terms, one or more DIDs (who do not need to be one block) are PTMP, one DID with the opition of adding digits after the DID is PTP |
| 12:27.44 | many | both of them can be 2 channels (BRI) or more (usually 30, PRI) over one 4wire wire. |
| 12:28.01 | many | we for example have 3 4wire wires with 2 channels each (BRI, but PTP) |
| 12:28.04 | Hyphenex | Ahh, interesting |
| 12:28.21 | Hyphenex | so I could get a single card in an asterisk box, and have it provide 30 ISDN channels over one 4 wire? |
| 12:28.42 | many | if your connection is a PRI (or ISDN30 as gromit called it), yes. |
| 12:28.56 | many | basically its an T1/E1 with 64k voice channels. |
| 12:29.16 | many | thats why these isdn cards are usually also referred to as E1 voice cards |
| 12:29.43 | Hyphenex | and why they can be about 500 pounds? |
| 12:30.09 | gr0mit | i would recommend the Sangoma cards |
| 12:30.15 | mrnick | many: i'm think of "smooth" solutions here, but can't really think of one now |
| 12:30.18 | gr0mit | their tech support is top nitch |
| 12:30.28 | gr0mit | notch even |
| 12:30.30 | mrnick | maybe i can just dail back on the same number |
| 12:30.50 | mrnick | and start the ivr |
| 12:30.53 | many | Hyphenex: without advocating the vendor, there are several ones: http://www.digium.com/en/products/digital/te122.php (ISDN30), http://www.digium.com/en/products/digital/b410p.php (ISDN2) |
| 12:30.54 | Hyphenex | gr0mit, but 500 pounds is about 1000 AUD, so it might even be better to do away with ISDN all together and just get the dedicated 2nd phone line put in... |
| 12:31.30 | many | single ISDN2 cards are about 20 euro, quad isdn2 cards are 500 euro, E1 cards are 500 euro, too. i believe |
| 12:31.35 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 12:31.51 | gr0mit | well, Hyphenex we have answered your questions, but we are still not quite sure what you are trying to achieve! can you give us the 'big picture' ? |
| 12:32.15 | many | Hyphenex: and dont get confused by the number of channels of T1/E1, E1 is 32 channels with 2 data-channels included (which are needed to signal isdn), for T1 its 24/2, i believe. |
| 12:32.45 | many | Hyphenex: so the actual number of channels delivered depends on your carrier, for australia i'd expect them to deliver T1 |
| 12:32.57 | gr0mit | Aus is E1 |
| 12:33.26 | gr0mit | everywhere except USA, Canada Hongkong and Japan use E1 |
| 12:33.37 | gr0mit | IIRC |
| 12:33.56 | many | ah, beyond my expectactions. :) |
| 12:34.00 | Hyphenex | hehe, okies. We're a glass company. We have a few offices in the building, not many. Most people work out on the factory floor and don't have need for them, but the guy who owns our company, also owns a few others around Australia, and such, there are a few phone calls being made between the companies (VoIP could save cost in this area.) Migrating to a VoIP method of providing cheaper phone calls could also be a big bonus |
| 12:34.00 | gr0mit | I know oz is E1 cos I manage an asterisk box in Melbourne. |
| 12:34.41 | many | Hyphenex: what you need depends on what you get from your carrier and/or wether you are willing to switch. |
| 12:35.05 | many | the usually scenario includes one E1 for the main office and several BRI for the branch offices. |
| 12:35.14 | gr0mit | ok Hyphenex first thing to do is to establish what you currently have |
| 12:35.38 | gr0mit | coz then we can better advise on moving forward |
| 12:36.10 | many | actually it also depends on what your carrier is ABLE to deliver you and how many simultaneous channels you really need. |
| 12:36.21 | Hyphenex | Ahhk, I'll talk to you about that next week then I guess... but can we assume we're replacing everything then and talk about options from there? |
| 12:36.51 | gr0mit | sure |
| 12:36.54 | many | if you made your homework by then, i s |
| 12:36.57 | many | 'pose so |
| 12:37.05 | Hyphenex | because would it not be feasable to have 2 ADSL2+ lines (even be them naked) and just delieve pure VoIP on one, and terminal services on the other? |
| 12:37.16 | gr0mit | yup |
| 12:37.22 | gr0mit | however ...... |
| 12:37.35 | many | Hyphenex: it is, but then you need someone to terminate POTS for you and deliver voip to you |
| 12:37.44 | gr0mit | certainly in my experience, using just voip is a bit of a risk |
| 12:37.51 | many | it certainly is |
| 12:38.01 | gr0mit | as adsl never seems to be as reliable as a phone line. |
| 12:38.10 | Hyphenex | no, that's very true... |
| 12:38.23 | gr0mit | so what i recommend is for example, to get 1 or 2 ISDN2 lines |
| 12:38.24 | *** join/#asterisk sgtpepper (n=ncorrare@190.12.99.66) |
| 12:38.41 | gr0mit | to handle incoming calls, emergency calls etc |
| 12:38.43 | Hyphenex | or 1 ISDN2 line and one ADSL line? |
| 12:38.55 | gr0mit | and use voip to make your outgoing calls as first choice |
| 12:39.06 | gr0mit | yup - that would also work |
| 12:39.07 | many | you can ofcourse mix'em, but you usually can not use the same DIDs for both |
| 12:39.29 | many | also, the size of the isdn and adsl lines depend on the number of simultaneously channels you need |
| 12:39.30 | sgtpepper | Anyone with experience with Unicall? I'm getting this weird message in asterisk chan_unicall.c:998 unicall_call: Make call failed - Blocked |
| 12:39.40 | many | as i said, thats part of your homework to do |
| 12:39.46 | gr0mit | well sgtpepper are the channels blocked? |
| 12:39.52 | sgtpepper | gr0mit, no |
| 12:39.54 | sgtpepper | just Idle |
| 12:39.58 | gr0mit | if you look at zttool you should see 1001 |
| 12:40.17 | Hyphenex | hmm, why wouldn't I mix and match then and use one for POTS phone system and VOIP, and the ADSL for Terminal Services then? |
| 12:40.27 | sgtpepper | I'm seeing 1001 in tx and 1101 in rx |
| 12:40.34 | gr0mit | have aah well that is why |
| 12:40.47 | gr0mit | your telco is blocking the channels |
| 12:40.53 | sgtpepper | Actually |
| 12:40.57 | many | i mean you cant use the same DIDs (pots numbers) on an incoming voip line terminted by someone else and on incoming pots |
| 12:41.08 | sgtpepper | I'm Using a Panasonic PABX against asterisk |
| 12:41.21 | sgtpepper | gr0mit, you mean the panasonic is blocking the channels? |
| 12:41.22 | many | what you certainly can do is terminte pots in the main office and then route dids on to branch offices via voip |
| 12:41.25 | gr0mit | yup |
| 12:41.31 | sgtpepper | Ohh |
| 12:41.36 | gr0mit | what flavour of R2 are you running? |
| 12:41.36 | sgtpepper | thank you very much then |
| 12:41.40 | sgtpepper | MFCR2 |
| 12:41.47 | gr0mit | which country vartient? |
| 12:41.50 | sgtpepper | ar |
| 12:42.00 | gr0mit | ok 1 sec |
| 12:42.09 | gr0mit | let me look at my box there |
| 12:42.45 | sgtpepper | I'm a master clock, since I'm connected to a private branch... I'm using asterisk as an ATA, but instead of usings fxs ports, I'm using an E1 |
| 12:42.53 | Hyphenex | many, Correct me if I'm wrong, but I could have my POTS service provider give me the numbers for incomming calls (like we do now) but also provide an extension in the dial plan that goes through the VoIP lines (if possible) to the asterisk boxes at another site before it tries the POTS line? |
| 12:43.04 | Hyphenex | so I'd only need a single DID number then, right? |
| 12:43.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:44.05 | many | the branches would need to have DIDs too, they dont need to be published. then you could possible route via voip or deflect via pots. |
| 12:44.29 | many | ifyou'd route via pots, you might use two channels on the main office for one call because asterisk continues to bridge the call |
| 12:44.48 | gr0mit | sgtpepper, my box is showing 1001 in both directions on my system in BA. |
| 12:44.51 | many | incoming main -> asterisk -> outgoing main -> incoming branch |
| 12:45.00 | *** join/#asterisk rjizzle (n=rjizzle@216.207.245.1) |
| 12:45.01 | sgtpepper | BA=Buenos Aires gr0mit ? |
| 12:45.04 | gr0mit | yup |
| 12:45.33 | Hyphenex | I'd have asterisk boxes at each site though? Every "branch" is it's own company and would want their own physical Dial in numbers on the POTS, it'd just be nice to have the Asterisk boxes between them able to save the cost of a phone call there |
| 12:45.53 | many | Hyphenex: i would take another path then. |
| 12:46.13 | Hyphenex | oh? |
| 12:46.52 | sgtpepper | gr0mit, I'll check with the panasonic guys... |
| 12:46.53 | many | Hyphenex: every branch and main gets its own DIDs. inbound is to each branch and main seperately. between them you can choose to try voip first and then pots. thats okay, unless you want to forward calls between branches, which would need some extra brainwork |
| 12:46.56 | gr0mit | sgtpepper, you need to see why your panasonic is blocking the channels |
| 12:46.59 | gr0mit | i hate R2 |
| 12:47.00 | many | (but possibly this is what you meant anyway) |
| 12:47.12 | gr0mit | it causes nothing but problems!!!! |
| 12:47.43 | sgtpepper | gr0mit, never had an Issue with PRI |
| 12:48.01 | sgtpepper | I'll ping you if I see something else |
| 12:48.16 | gr0mit | yup, PRI much easier to manage! |
| 12:48.42 | gr0mit | could not get it from our telco there |
| 12:49.47 | gr0mit | Hyphenex, starting with a clean sheet, I would put an asterisk box in each site |
| 12:50.00 | gr0mit | a BRI in each site |
| 12:50.05 | gr0mit | with some DDI numbers |
| 12:50.18 | gr0mit | and have intersite calls running over voip |
| 12:50.35 | gnorbert | Does somebody know, how can I play sound in a meetme conference started from CLI? |
| 12:50.35 | gr0mit | outbound calls running out via a VIOIP provider |
| 12:51.48 | Hyphenex | Yep, so I'd need the ISDN2 at each site to get that to work (and possibly an extra ADSL2 line on the main one to reduce the load on the Terminal Services server) |
| 12:51.55 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 12:52.13 | many | yup |
| 12:52.17 | gr0mit | how you get your internet there is left as an excercise for the reader! |
| 12:52.33 | gr0mit | i.e. it depends on your telco |
| 12:52.49 | Zeeek | excercise is for the non lazy |
| 12:52.55 | Hyphenex | Coolies :) |
| 12:53.22 | gr0mit | if you want a voip provider in UK let me know, but otherwise you will need to find one down under |
| 12:54.43 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:54.58 | [TK]D-Fender | gnorbert: create a "call file" that will dial into the conference upon connect play its sound, and then hangup |
| 12:56.33 | gnorbert | [TK]D-Fender: Thanks, it was a good idea again. :) |
| 12:57.06 | Hyphenex | gr0mit, for home use, Mynetfone have been pretty good in AUS |
| 12:57.07 | gr0mit | yeah, [TK]D-Fender - that is a neat trick! |
| 12:57.34 | gr0mit | Hyphenex, for biz, you will want one which lets you send a range of caller id |
| 12:57.48 | gr0mit | and which lets you have multiple calls at the same time. |
| 12:57.58 | [TK]D-Fender | gr0mit: Easy enough o do. I can envision an entire "live feeder" mechanism through this as well. |
| 12:58.48 | Hyphenex | gr0mit, I didn't even think of caller ID... if I've got caller ID on incomming POTS service, I can't then set that caller ID for an outputbound call through a different VoIP company, can I? |
| 12:59.11 | gr0mit | Hyphenex, that depends on your voip provider. |
| 12:59.19 | Hyphenex | gr0mit, wouldn't that be illegal? |
| 12:59.30 | Hyphenex | I mean, what's to stop me from setting my caller ID number as you? |
| 12:59.42 | gr0mit | I allow my customers do to that but only if they provide proof for each number that they own it |
| 12:59.50 | gr0mit | Hyphenex, nothing. caller id is not secure |
| 13:00.12 | Hyphenex | Heh, that's interesting |
| 13:00.17 | gr0mit | never to be trusted for any authentication. If I can set any caller id I want, then the crooks can too. |
| 13:00.17 | Hyphenex | I'll have to give MyNetFone a call then |
| 13:00.41 | gr0mit | Hyphenex, can't comment on ozziliegaliies |
| 13:00.46 | *** join/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 13:01.01 | Hyphenex | that's cool, all part of my homework :) |
| 13:01.18 | Hyphenex | but what sort of computer will I be looking at then to host asterisk for my, say 12 users max? |
| 13:01.19 | *** part/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 13:01.33 | gr0mit | Hyphenex, anything from a P2 up will be fine |
| 13:01.43 | gr0mit | but i recommend a P3-500 at least |
| 13:01.49 | Hyphenex | Really? Even if they're all talking at the same time? |
| 13:01.51 | gr0mit | i.e. any old junk you have. |
| 13:02.02 | Hyphenex | heh, fair enough :) |
| 13:02.12 | gr0mit | however |
| 13:02.24 | gr0mit | you are well advised to have raid hard drives |
| 13:02.44 | gr0mit | raid1 is mirrored iirc |
| 13:03.01 | Hyphenex | why do I need RAID hard drives? |
| 13:03.22 | [TK]D-Fender | Hyphenex: not special drives, just redundent |
| 13:03.23 | gr0mit | coz hard drives will fail. |
| 13:03.26 | Hyphenex | can't I just have a copy of it installed on our database server our VoIP server goes down? |
| 13:03.38 | [TK]D-Fender | Hyphenex: Because you don't want some spinning piece of steel to lock up and take you down. |
| 13:03.39 | many | running a critical application such as a telephony server one one harddisk only is suicide. |
| 13:03.43 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:03.44 | gr0mit | yup, but how long to rebuild the system? |
| 13:03.49 | Hyphenex | Good point |
| 13:03.57 | Hyphenex | I could always skip the hard disks all toegether? |
| 13:04.09 | Hyphenex | and Netboot it from a fileserver that just happens to be raided anyway? |
| 13:04.12 | [TK]D-Fender | Hyphenex: Flash dies too... limited writes, etc |
| 13:04.26 | many | Hyphenex: sure. fibre channel will love you |
| 13:04.26 | [TK]D-Fender | Hyphenex: Then you are placing network load etc on it. |
| 13:04.34 | [TK]D-Fender | Hyphenex: And I'm sure would require a larger box |
| 13:04.40 | gr0mit | i don't want to be the person trying to make a 999 call when the phones are down |
| 13:04.52 | [TK]D-Fender | Hyphenex: 2 HD < 100$ total. Don't be a schmuck :) |
| 13:04.54 | many | i dont want to be the person who meets such a person |
| 13:05.06 | Hyphenex | Hehe, ok, ok |
| 13:05.37 | gr0mit | for once the whole of #asterisk is in violent agreement on a point! |
| 13:06.15 | many | actually, we're not. a good nas will cover your ass pretty well |
| 13:06.25 | many | but anyway :> |
| 13:06.30 | gr0mit | hehe!!!!! |
| 13:06.54 | Hyphenex | what does nas stand for? |
| 13:07.16 | Hyphenex | oahh yes, the file server thing |
| 13:07.18 | Hyphenex | Got ya |
| 13:07.20 | jbeez | network attached storage |
| 13:07.30 | many | okay, he doesnt own a good nas. probably just a linux server with JBOD |
| 13:07.36 | jbeez | usually slower than a san in my experience |
| 13:08.03 | many | jbeez: not in mine, the environment just needs to be planned more carefully. |
| 13:08.28 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 13:08.36 | many | loves his fc/iscsi/nfs/cifs allineonebox |
| 13:08.41 | Kyoshi | gmornin |
| 13:09.02 | *** join/#asterisk phpcodemonkey (n=jeremy@82-43-235-140.cable.ubr02.pres.blueyonder.co.uk) |
| 13:09.10 | jbeez | it could just be that every "nas" I've worked with has been like a linksys disk share or a snap server, and with sans I've worked with like enterprise grade equipment |
| 13:09.38 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 13:09.45 | many | hehe |
| 13:09.52 | Hyphenex | I'm afraid it'd be windows |
| 13:09.59 | Hyphenex | we're on the process of upgrading and everything is just about windows |
| 13:10.03 | many | err, stick to a 3ware |
| 13:10.12 | many | or something alike |
| 13:11.21 | phpcodemonkey | hi - anyone know why upgrading from asterisk 1.2.19/zaptel 1.2.18 to 1.2.29/1.2.16 would stop previously working callerid (UK) from working? |
| 13:11.56 | Hyphenex | ahh well, I'm off to bed |
| 13:12.06 | Hyphenex | Thanks for your help :) |
| 13:12.33 | gr0mit | g'night Hyphenex |
| 13:12.52 | gr0mit | phpcodemonkey, analogue or ISDN? |
| 13:13.01 | phpcodemonkey | PSTN analogue |
| 13:13.15 | gr0mit | which card? |
| 13:13.25 | phpcodemonkey | TDM400P Rev I |
| 13:13.36 | gr0mit | hmmm no idea |
| 13:13.39 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 13:14.16 | phpcodemonkey | gonna try going back to older zaptel |
| 13:14.20 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 13:14.27 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:14.40 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:16.45 | Kyoshi | If I have asterisk configured as a call switch, most pbx functions disabled, SIP only, 8GB or 16GB ram, dual Xeon 3Ghz proc, all configs via ARA, what kind of concurrent call volume can I expect? are there any benchmarks i can read? |
| 13:17.06 | [TK]D-Fender | Kyoshi: What do you NEED to support? |
| 13:17.21 | [TK]D-Fender | Kyoshi: And many other factors can have severe impact |
| 13:18.16 | Kyoshi | "need" to support as in? codecs? |
| 13:18.42 | [TK]D-Fender | Kyoshi: Describe you needs and we minght know if that is sufficient |
| 13:18.50 | Kyoshi | a call switch |
| 13:18.55 | Kyoshi | thats all |
| 13:18.57 | [TK]D-Fender | Kyoshi: MEANINGLESS |
| 13:18.58 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:19.02 | Kyoshi | ulaw, gsm |
| 13:19.25 | [TK]D-Fender | Kyoshi: How many calls are you passing through? Any transcoding? Call recording? AGI? Live synth? |
| 13:19.29 | *** join/#asterisk bkw___ (n=brian@adsl-70-234-164-166.dsl.tul2ok.sbcglobal.net) |
| 13:19.31 | Kyoshi | call switch/ soft switch |
| 13:19.59 | [TK]D-Fender | Kyoshi: Those are not miracle terms you can just through around |
| 13:20.06 | Kyoshi | <PROTECTED> |
| 13:20.12 | Kyoshi | i said MOST functions disabled |
| 13:20.22 | Kyoshi | its ONLY a call switch nothing more |
| 13:20.33 | [TK]D-Fender | Kyoshi: You're better off with SER then. |
| 13:20.37 | Kyoshi | its doing sip auth to ensure only authorized users get thru |
| 13:20.40 | [TK]D-Fender | Kyoshi: * is not a "switch" |
| 13:20.47 | Kyoshi | mine is now |
| 13:20.50 | Kyoshi | :p |
| 13:21.13 | [TK]D-Fender | Kyoshi: No, you're just CALLING it a "switch". You are quite adept at abusing terminology. |
| 13:22.02 | Kyoshi | or you're quite adept and saying what you like to make others look bad? |
| 13:22.03 | Kyoshi | pick one |
| 13:22.08 | Kyoshi | this can go on and on |
| 13:22.16 | Kyoshi | i stripped the code down completely |
| 13:22.17 | Zeeek | get a room |
| 13:22.25 | Kyoshi | took only the modules i wanted to compile |
| 13:22.30 | Kyoshi | changed the makefile |
| 13:22.37 | Kyoshi | compiled |
| 13:22.50 | [TK]D-Fender | Kyoshi: You're right. It can. the book says * is a B2BUA, not a switch. The asterisk primary docs say the same. Keep arguing, but the creators of * beg to differ. |
| 13:22.51 | Kyoshi | im serious when i said it serves no pbx functions |
| 13:22.56 | Kyoshi | its JUST a switch |
| 13:22.57 | Corydon76-dig | Kyoshi: then you're the only one who can produce that benchmark |
| 13:22.59 | *** join/#asterisk moy (n=moy@nat/ibm/x-15ce96ab4cffcb23) |
| 13:23.19 | [TK]D-Fender | Kyoshi: And how many calls are you looking to push through it? |
| 13:23.24 | Kyoshi | cor: currently, unfortunately |
| 13:23.33 | Kyoshi | about 1000 concurrent hopefully |
| 13:23.39 | *** join/#asterisk gaetronik (n=gaetan@200.111.138.170) |
| 13:25.06 | [TK]D-Fender | Kyoshi: I'd really advise SER at this point. |
| 13:25.32 | Kyoshi | i dont wanna think that i threw away a few months of work to hear that, i'd rather see what i can do with this first |
| 13:26.24 | *** join/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 13:26.42 | EricL | Is there a way to use MeetMe without a Zap card? |
| 13:27.01 | gaetronik | EricL, ztdummy |
| 13:27.27 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 13:27.33 | EricL | gaetronik: Is that a kernel module? |
| 13:28.40 | [TK]D-Fender | Kyoshi: You're perfectly entitled to try whatever you feel like. |
| 13:29.13 | [TK]D-Fender | Kyoshi: But if its the right too for the right job, * isn't it as the primary front-end for most larger implementations. |
| 13:29.19 | [TK]D-Fender | tool* |
| 13:29.29 | [TK]D-Fender | EricL: Yes, part of Zaptel |
| 13:29.35 | *** join/#asterisk eliel (n=eliel@151-202-114-200.fibertel.com.ar) |
| 13:29.38 | *** join/#asterisk spike008t (n=spikie@ven69-2-82-228-116-153.fbx.proxad.net) |
| 13:29.44 | spike008t | Hi everybody |
| 13:30.27 | spike008t | Does anyone know how it's work the hint with the protocol IAX? |
| 13:30.52 | EricL | [TK]D-Fender: But I don't need a zaptel card to build zaptel drivers right? |
| 13:31.15 | [TK]D-Fender | EricL: Correct |
| 13:31.29 | [TK]D-Fender | spike008t: Same as any other device |
| 13:31.37 | Kyoshi | <PROTECTED> |
| 13:31.41 | EricL | [TK]D-Fender: Excellent. Then its a Gentoo problem I am having and nothing to do with Asterisk or hardware. |
| 13:31.53 | Kyoshi | you'd be surprised how many larger implementations chose * over ser |
| 13:32.07 | gaetronik | EricL, |
| 13:32.11 | jbeez | ser? |
| 13:32.14 | EricL | gaetronik: Yes? |
| 13:32.15 | Kyoshi | i was pretty shocked to hear it |
| 13:32.21 | gaetronik | it is |
| 13:32.24 | [TK]D-Fender | Kyoshi: But when you remove the thing that * is good for, you are picking it for the aprts its worse at. |
| 13:32.27 | spike008t | [TK]D-Fender: thank's and can I catch it with the iaxclient lib? other wise I'll make it myself... |
| 13:32.27 | gaetronik | you have to compile zaptel |
| 13:32.31 | Kyoshi | jb, openser, sip express router |
| 13:32.34 | gaetronik | with the ztdummy module |
| 13:32.38 | jbeez | ah, sorry :D |
| 13:32.47 | [TK]D-Fender | spike008t: Sorry, I don't follow you |
| 13:33.00 | EricL | Yep...I have a few times, I just keep getting an invalid module format error no matter what I do. |
| 13:33.10 | EricL | I just wanted to be sure that it had nothing to do with the hardware. |
| 13:33.14 | [TK]D-Fender | spike008t: "hints" (presence" don't ahve anything to do with iaxclient |
| 13:33.32 | Kyoshi | tkd-fender, you cant say 'it's worst at' when you would basically be making a blanket statement that ALL components are horrible, but combined they just simply work, bad approach |
| 13:33.56 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 13:34.14 | [TK]D-Fender | Kyoshi: No... I never said that together it'll all "simply work" :) |
| 13:34.23 | [TK]D-Fender | Kyoshi: Don't go putting words in my mouth! |
| 13:34.52 | Kyoshi | i didnt, but you said by tearing it apart, im chosing what * is worst at |
| 13:34.58 | [TK]D-Fender | Kyoshi: You just boiled your intentions for * down to a single function : auth calls and terminate. Thats one of *'s strong suits. |
| 13:34.58 | Kyoshi | what are you implying then |
| 13:35.08 | Kyoshi | fine |
| 13:35.10 | [TK]D-Fender | Kyoshi: Not tearing it apart. |
| 13:35.24 | [TK]D-Fender | Kyoshi: oops, no *'s strong suit, but rather SER |
| 13:35.26 | spike008t | [TK]D-Fender: ok I see thank's, in fact i'm writing an iaxphone, and i didn'y know how to know the presence of the other phone |
| 13:35.39 | Kyoshi | <[TK]D-Fender> Kyoshi: But when you remove the thing that * is good for, you are picking it for the aprts its worse at. |
| 13:36.08 | [TK]D-Fender | spike008t: Ah... sorry I'm not familiar with what interface that lib offers for it. I suspect it'll be the same as jsut about every other device's capabilities however |
| 13:36.26 | Kyoshi | again, you need to consider that statement is pretty blanketing and says something dangerous |
| 13:36.31 | Kyoshi | very dangerous |
| 13:36.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:37.00 | spike008t | [TK]D-Fender: yes u're right. Thank you men for the answers :) |
| 13:37.06 | Kyoshi | the SIP stack i'd think is one of the stronger components of * and such a statement implies that the stack is in fact not as such |
| 13:37.22 | [TK]D-Fender | Kyoshi: Lets try this again. * isn't so great at just passing off calls, it sits in the middle, has load issues with tons of calls. * is great a processing calls and doing interesting stuff. * is good as an application server (which is what a lot of people using SER in front use * for) |
| 13:37.27 | Kyoshi | but in any case |
| 13:37.35 | Kyoshi | i appreciate the time thought on this |
| 13:37.49 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:37.54 | [TK]D-Fender | Kyoshi: If you think *'s SIP stack is strong I'm beginning to wonder who your dealer is :) theres a reason there are 3 complete rewrites of it in progress. |
| 13:38.23 | [TK]D-Fender | Kyoshi: it has been the cause of massive ravings. |
| 13:38.28 | EricL | Thanks all. |
| 13:38.30 | *** part/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 13:40.19 | gaetronik | when a caller to a queue hangup, it takes seconds before asterisk detect the hangup |
| 13:40.44 | gaetronik | is there anyway to improve the hangup detection |
| 13:41.03 | gaetronik | i use a digium card |
| 13:42.46 | Corydon76-dig | gaetronik: analog or digital? |
| 13:42.50 | gaetronik | digital |
| 13:42.56 | gaetronik | i was looking for the exact model |
| 13:43.26 | gaetronik | te420B |
| 13:43.41 | *** join/#asterisk |||Mad||| (n=mad@69.95.51.232) |
| 13:43.52 | Corydon76-dig | Are you using the card in CAS mode or as a PRI? |
| 13:44.23 | Corydon76-dig | PRI will give you the best signalling. |
| 13:44.36 | |||Mad||| | Good morning! Can someone tell me where in Asterisk it sends the MWI codes? |
| 13:44.47 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 13:44.53 | gaetronik | as a pri |
| 13:44.57 | *** join/#asterisk ManxPower (n=manxpowe@134.sub-75-203-231.myvzw.com) |
| 13:45.16 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 13:46.25 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 13:46.29 | Corydon76-dig | gaetronik: then you should already be getting the most prompt signalling |
| 13:47.02 | Kyoshi | <[TK]D-Fender> Kyoshi: If you think *'s SIP stack is strong I'm beginning to wonder who your dealer is :) |
| 13:47.12 | Kyoshi | i need a new dealer |
| 13:47.12 | gaetronik | the agents hear the tut tut tut |
| 13:47.14 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:47.14 | Kyoshi | :( |
| 13:47.25 | russellb | define ... strong ... |
| 13:47.41 | [TK]D-Fender | Kyoshi: Sorry, that'd be in #pharmaceuticals |
| 13:47.44 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 13:47.51 | [TK]D-Fender | russellb: Odour isn't everything you know! |
| 13:48.08 | gaetronik | the strangest is that the configuration i use was the same on an other server with a sangoma card |
| 13:48.37 | Corydon76-dig | gaetronik: why don't you call tech support? |
| 13:48.56 | gaetronik | Corydon76-dig, here is the first step |
| 13:49.58 | gaetronik | since talking in english in irc is a way more easy |
| 13:50.07 | gaetronik | than making a phone call |
| 13:50.35 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:50.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:51.12 | Zeeek | the next step: Digium International phone support |
| 13:51.14 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 13:51.19 | Zeeek | might be worth looking int |
| 13:51.29 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 13:51.37 | Zeeek | o |
| 13:51.41 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.1 (2008/06/30), 1.2.29 (2008/06/03), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), 1.2.9 (2008/06/04), Zaptel 1.4.11 (2008/05/28), 1.2.26 (2008-05-28), Libpri 1.4.5 (2008/07/11), 1.2.7 (2008/03/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Swit |
| 13:51.48 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 13:51.54 | [TK]D-Fender | hrm. too many versions :) |
| 13:51.59 | [TK]D-Fender | 1 sec |
| 13:52.00 | Zeeek | FLOOD |
| 13:52.01 | russellb | heh, pwnt! |
| 13:52.08 | russellb | take off 1.2 |
| 13:52.09 | [TK]D-Fender | russellb: indeed! |
| 13:52.09 | *** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 13:52.14 | russellb | or i will :-p |
| 13:52.19 | [TK]D-Fender | russellb: I have solutions, don't you fear |
| 13:52.26 | russellb | seriously, take off 1.2 |
| 13:52.26 | [TK]D-Fender | russellb: ... mean! |
| 13:52.28 | russellb | it's not supported |
| 13:52.33 | [TK]D-Fender | russellb: I will... in a manner. |
| 13:52.36 | russellb | k |
| 13:52.40 | Zeeek | but is it defecated? |
| 13:52.51 | russellb | it's deprecated, at least |
| 13:52.52 | [TK]D-Fender | Zeeek: No, CONSTIPATED ;) |
| 13:52.58 | M1s3ry | lol |
| 13:53.03 | [TK]D-Fender | zing! |
| 13:53.08 | gaetronik | Zeeek, yes |
| 13:53.16 | [TK]D-Fender | Wow... thats pretty good for a Friday... |
| 13:53.24 | gaetronik | first step find a phone with international right |
| 13:53.37 | Zeeek | nows humbly for having been a good straight man |
| 13:53.48 | Zeeek | s/nows/bows |
| 13:54.15 | Zeeek | or should that be good, straight, man on this channel? |
| 13:54.45 | *** part/#asterisk phpcodemonkey (n=jeremy@82-43-235-140.cable.ubr02.pres.blueyonder.co.uk) |
| 13:54.48 | russellb | blinks |
| 13:55.06 | Zeeek | thinks |
| 13:55.13 | Zeeek | then drinks |
| 13:55.31 | M1s3ry | follow suit with Zeeek's last comment |
| 13:55.43 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.1 (2008/06/30), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.5 (2008/07/11) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX |
| 13:55.57 | Zeeek | passes round a bottle of fruity 5.9% lemonade |
| 13:56.18 | Zeeek | (94.1% alcohol) |
| 13:56.34 | [TK]D-Fender | ~asterisk1.2 |
| 13:56.35 | jbot | [~asterisk1.2] Asterisk 1.2.29 (2008/06/03), Addons 1.2.9 (2008/06/04), Zaptel 1.2.26 (2008-05-28), Libpri 1.2.7 (2008/03/13) |
| 13:56.41 | [TK]D-Fender | russellb: decent now? |
| 13:56.53 | Zeeek | ~alcohol |
| 13:56.54 | jbot | it has been said that alcohol is the answer to most of lifes problems.., or a good excuse to alias rm='rm -i', or a cause of "rm -rf . / "'s, or the cause of, and solution to, all of lifes problems, or ask me about ambrosia |
| 13:57.09 | ManxPower | [TK]D-Fender: You forgot to add "Height of Asterisk stability" |
| 13:57.11 | M1s3ry | [TK]D-Fender, "Now Defecated" should be added to that 1.2 statement for jbot |
| 13:57.11 | ManxPower | 8-) |
| 13:57.11 | Qwell | ~ambrosia |
| 13:57.12 | jbot | [ambrosia] ask me about alcohol |
| 13:57.15 | Zeeek | ~lemonade |
| 13:57.16 | jbot | lemonade is, like, probably a liquidy form of chocolate |
| 13:57.32 | russellb | [TK]D-Fender: very nice, thanks :) |
| 13:57.52 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 13:57.52 | [TK]D-Fender | russellb: Minor tweaking to come for this, but almost there. |
| 13:57.53 | russellb | ManxPower: trollllllllllllllll |
| 13:58.44 | ManxPower | russellb: 8-) |
| 13:58.56 | Zeeek | as Mark Twain once noted, "rumours of the defecation of 1.2 has been greatly exagerated" |
| 13:59.35 | ManxPower | Zeeek: It's the User's .vs. Digium Asterisk Smackdown! |
| 13:59.41 | russellb | ~manxpower |
| 13:59.42 | jbot | somebody said manxpower was Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Contact eric@fnords.org, or a trollllllllllllll |
| 13:59.52 | russellb | :-p |
| 14:00.04 | *** join/#asterisk Xaviertoor (n=Xavierto@200-146-243-009.xf-static.ctbcnetsuper.com.br) |
| 14:00.05 | Zeeek | did you guys see the NYC job offer on the ML? |
| 14:00.07 | M1s3ry | nice |
| 14:00.15 | Zeeek | speaking of consultants |
| 14:00.42 | M1s3ry | ~russellb |
| 14:00.43 | jbot | hmm... russellb is Russell Bryant <russell@digium.com>, or not a fan of jbot, or http://www.russellbryant.net/ |
| 14:01.13 | M1s3ry | russellb, how could you not be a fan of jbot? :( |
| 14:01.16 | russellb | aw, i love you jbeez |
| 14:01.17 | russellb | lol |
| 14:01.20 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 14:01.29 | russellb | jbeez: not you, jbot .... wow, owned by tab completion |
| 14:01.36 | Qwell | uh huh |
| 14:01.36 | M1s3ry | fail |
| 14:01.39 | Qwell | I buy that |
| 14:01.39 | russellb | mega fail |
| 14:01.55 | Qwell | scurries off |
| 14:02.22 | russellb | jbot: you rock |
| 14:02.22 | jbot | russellb: aw, gee |
| 14:05.04 | [TK]D-Fender | ~asteriskversions |
| 14:05.05 | jbot | [~asteriskversions] Always check the channel topic for updates, otherwise for the latest bundled listing of Asterisk & supporting packages by major release type ~asterisk1.2 / ~asterisk1.4 / ~asterisk1.6 |
| 14:05.09 | [TK]D-Fender | russellb: All done. |
| 14:05.14 | [TK]D-Fender | ~asterisk1.2 |
| 14:05.14 | jbot | [~asterisk1.2] Asterisk 1.2.29 (2008/06/03), Addons 1.2.9 (2008/06/04), Zaptel 1.2.26 (2008-05-28), Libpri 1.2.7 (2008/03/13) |
| 14:05.15 | russellb | yay. |
| 14:05.16 | [TK]D-Fender | ~asterisk1.4 |
| 14:05.16 | jbot | [~asterisk1.4] Asterisk 1.4.21.1 (2008/06/30), Addons 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.5 (2008/07/11) |
| 14:05.17 | [TK]D-Fender | ~asterisk1.5 |
| 14:05.21 | russellb | pwnt |
| 14:05.30 | [TK]D-Fender | ~asterisk1.6 |
| 14:05.30 | jbot | [~asterisk1.6] Asterisk 1.6.0-beta9 (2008/05/14), Addons 1.6.0-beta4 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.5 (2008/07/11) |
| 14:05.32 | [TK]D-Fender | oops :) |
| 14:05.33 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:06.04 | *** join/#asterisk freakynl (n=freaky@unaffiliated/freakynl) |
| 14:06.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:06.11 | sgtpepper | ~asterisk1.4 |
| 14:06.12 | jbot | [~asterisk1.4] Asterisk 1.4.21.1 (2008/06/30), Addons 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.5 (2008/07/11) |
| 14:06.16 | sgtpepper | me likey |
| 14:06.21 | russellb | ~asterisk1.5 |
| 14:06.21 | jbot | WATCH OUT, SHARKS!!!! |
| 14:06.22 | freakynl | hiya, our siemens phone central just died and i have some hardware q's if that's ok. |
| 14:06.41 | russellb | freakynl: ask away. |
| 14:06.43 | gr0mit | ask away! |
| 14:06.50 | Zeeek | hardware is deprecated |
| 14:06.54 | sgtpepper | gr0mit, finally it was on the panasonic side |
| 14:07.01 | sgtpepper | the interface is'nt configured |
| 14:07.02 | freakynl | first of, i have experience with e-phone. e-phone requires expensive eicon diva cards. can asterisk deal with telephony with cheap cards? Like avm's for like 30 euro's? |
| 14:07.03 | gr0mit | aah, sgtpepper, working now? |
| 14:07.10 | sgtpepper | not yet |
| 14:07.16 | sgtpepper | they're going to configure that on monday |
| 14:07.27 | freakynl | second can we use the siemens phones? their neither isdn nor analog it's some proprietary digital thing |
| 14:07.44 | gr0mit | freakynl, what type of interface - isdn? |
| 14:07.48 | gnorbert | Maybe it's a noob question, but shouldn't Playback work from the CLI? |
| 14:07.56 | freakynl | the central is isdn, well was it died :D |
| 14:08.13 | freakynl | the bottom of the phone says optiset e standard |
| 14:09.02 | freakynl | it has a normal rj-12 (or was it 11?) connection, the smaller version of rj-45. pretty familiar with computers and networks, but telephony is quite new to me. redirecting me to docs is fine, but what i've seen so far doesn't really mention much on telephony hardware |
| 14:09.19 | russellb | freakynl: you can't use the phones with asterisk directly, but there are some companies that make devices that let you use them, and then their device speaks SIP to Asterisk. For example, check out citel.com |
| 14:10.06 | [TK]D-Fender | gnorbert: No, you can't just call DIALPLAN apps from the CLI like that |
| 14:10.13 | Corydon76-dig | gnorbert: why would it? |
| 14:10.49 | [TK]D-Fender | freakynl: if its a smaller version of RJ.45 then its a proprietary digital set taht is useless with * |
| 14:10.51 | gnorbert | Sorry, missread... I thought, it's a CLI command. |
| 14:11.05 | ManxPower | freakynl: The problem with those converter devices is that they are as expensive per port as a decent IP phone |
| 14:11.06 | Corydon76-dig | gnorbert: think about it. Asterisk is designed to service multiple channels at the same time. |
| 14:11.54 | gnorbert | Corydon76-dig: Well, it sounds logic.:) |
| 14:11.54 | Corydon76-dig | The CLI is simply ill-suited to manage a single call, let alone multiple |
| 14:12.17 | ManxPower | gnorbert: Where would sound be played back if you ran it on the CLI? |
| 14:12.34 | ManxPower | Most Asterisk systems don't even have a sound card |
| 14:14.28 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 14:15.41 | [TK]D-Fender | ManxPower: Statistcally speaking I doubt that. But its true that extremely few have any need of one. |
| 14:16.16 | kamanashisroy | hi .. I am spending hard time doing ami in perl .. the authentication technique is not working like manager.txt .. it seems it has MD5 authentication technique .. any idea ? |
| 14:17.57 | russellb | MD5 auth is optional |
| 14:18.33 | Zeeek | I now have the world's record for ATA overkill |
| 14:18.45 | Zeeek | an old asterisk box with 3 FXS |
| 14:18.54 | Zeeek | but it works |
| 14:19.50 | raz | hrm... is there a "simple" way to receive calls over a bog-standard remote sip account? |
| 14:19.55 | bboschman | hi |
| 14:20.15 | freakynl | ManxPower: yea i was afraid of that |
| 14:20.26 | bboschman | can I delete some of the default asterisk .conf files? |
| 14:20.29 | kamanashisroy | russellb: ah .. it is working .. |
| 14:20.36 | bboschman | e.g. manager.conf |
| 14:20.42 | freakynl | gr0mit: the voip dsl connection outputs what they call here ISDN-2. they also call it s0 bus, albeit I think that's less specific |
| 14:21.08 | freakynl | it has 2 channels. they also sell ISDN-30 here, with 30 lines on one cable. don't believe there is anything in between |
| 14:21.59 | freakynl | there's also something called vdsl, which i think is phone, not internet related, but would have to dig into that. currently we have adsl-2 (which is fake as the modem outputs that but basically it's voip towards the modem, then isdn from there to the central) |
| 14:22.01 | gr0mit | ok isdn2 is well known here too! |
| 14:22.38 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:22.50 | freakynl | can i just use cheap isdn cards for that? forgot the chipset, there's a quite famous one i used to build linux internet routers with. cards were like 40 euro back then |
| 14:23.10 | freakynl | billion incorporated them, should be able to find it in linux kernel |
| 14:24.13 | [TK]D-Fender | raz: the same kind of deal as setting up any other ITSP with * |
| 14:24.16 | freakynl | it's called a passive card anyways. |
| 14:24.25 | *** part/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 14:24.30 | *** join/#asterisk tomoconnor (n=toconnor@rabbit.dbplc.com) |
| 14:24.40 | freakynl | the active ones are pretty expensive (well not compared to other phone centrals, but they are very expensive compared to passive isdn cards) |
| 14:24.57 | freakynl | hisax |
| 14:25.48 | sax | hifreakynl |
| 14:26.01 | raz | [TK]D-Fender, ok, i'm a total asterisk newbie. i followed the "simple answering machine" tutorial and that worked. now i'd like to receive inbound call via my sip-provider to that extension (ext 30). i put this line in sip.conf under [general]: register => user:pass@provider.com/30 .. but asterisk doesn't seem to connect there. what's missing? :) |
| 14:27.12 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 14:27.35 | [TK]D-Fender | raz: You made no mention of creating a peer entry to auth the incoming calls against, no dialplan to match the call and steps to process it with, or any voicemail setup, etc. |
| 14:27.43 | [TK]D-Fender | raz: In short, you are missing EVERYTHING. |
| 14:27.51 | raz | i know |
| 14:28.23 | raz | i think i have the dialplan (in extensins.conf) but what is peer entry? |
| 14:28.24 | [TK]D-Fender | raz: * has a somewhat steep learning curve at the start and there isn't going to be a "Here's exactly how to do funky thing XYZ". |
| 14:28.46 | raz | well, as said. the voicemail thingy already works. i can dial 30 on my softphone and babble to it. |
| 14:28.49 | [TK]D-Fender | raz: Go look up how your ITSP suggests setting it up, install a soft-phone, go read THE BOOK, and get started. |
| 14:28.51 | [TK]D-Fender | ~book |
| 14:28.51 | 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 |
| 14:29.10 | raz | k, thx |
| 14:29.16 | [TK]D-Fender | raz: Feel free to look at how other ITSP's have you set up their inbound peer entry |
| 14:29.27 | tomoconnor | hey folks... i'm trying to set up asterisk with a Digium Wildcard E100P card, and i keep getting these errors when i start asterisk, i've been through all the zaptel config, and it all seems to be valid..[Jul 18 15:11:42] ERROR[9261]: asterisk.c:2982 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. |
| 14:29.29 | tomoconnor | any ideas? |
| 14:29.45 | tomoconnor | hey folks... i'm trying to set up asterisk with a Digium Wildcard E100P card, and i keep getting these errors when i start asterisk, i've been through all the zaptel config, and it all seems to be valid..[ |
| 14:29.51 | raz | [TK]D-Fender, ITSP? ;) |
| 14:30.24 | M1s3ry | tomoconnor, the E100P a very old card... what version of zaptel are you running? |
| 14:30.24 | freakynl | hi sax, nice to meat you, but was referring to hisax chips :) |
| 14:30.24 | russellb | tomoconnor: it means you don't have zaptel configured and initialized properly |
| 14:30.48 | raz | what's really annoying about the book (for a newbie) is that the examples often don't tell which config file they belong in. |
| 14:30.50 | M1s3ry | tomoconnor, ^ ditto russellb's comment |
| 14:30.57 | raz | http://tfot.leifmadsen.com/ch04s08.html <-- in what file does that snippet go? |
| 14:30.58 | tomoconnor | uh, 1.4.11 |
| 14:31.21 | tomoconnor | and zttool says the card's working ok... |
| 14:31.34 | russellb | raz: since you're doing SIP configuration, it goes in sip.conf :) |
| 14:32.15 | tomoconnor | i've got a book on asterisk, but it doesn't make much sense for ISDN cards, only FXO ones |
| 14:32.43 | raz | russellb, you say that as if it was obvious. ;) i have only touched 3 out of the 62 config files so far, so i wasn't sure. |
| 14:32.54 | M1s3ry | tomoconnor, ^ ditto russellb's comment... pastebin your zaptel.conf, zapata.conf, and "lsmod | grep zaptel" |
| 14:32.57 | raz | thanks though |
| 14:33.08 | [TK]D-Fender | ~itsp |
| 14:33.08 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 14:33.10 | [TK]D-Fender | raz ^^^^ |
| 14:33.26 | raz | yup i figured that one out |
| 14:34.22 | freakynl | so can i use passive isdn cards? |
| 14:34.51 | raz | will asterisk -c log something to the console when and if it connects to the sip provider? |
| 14:35.04 | mags2 | when trying to dial iax2/ to another machine, we started seeing channel.c: No channel type registered for '' |
| 14:35.04 | mags2 | seems like it works fine for a while but then starts to do this. sometimes reloading iax fixes it sometimes not. |
| 14:35.11 | Kobaz | i've set up some fxs channels, I plug in a handset, I get dialtone, but I hit some digits and asterisk doesn't see any, and it doesn't matter what phone i use... what could be the problem? |
| 14:37.15 | Kobaz | oh |
| 14:37.21 | Kobaz | hmm, i turned on relaxdtmf and that worked |
| 14:42.38 | *** join/#asterisk Drognan (n=Drognan@rrcs-24-129-157-34.se.biz.rr.com) |
| 14:42.43 | gaetronik | the issue of not detecting the end of a call can be due to a bad country in zaptel |
| 14:42.45 | gaetronik | ? |
| 14:43.00 | tomoconnor | M1s3ry, http://pastebin.com/m3171615f |
| 14:43.03 | Drognan | What program can I use for IAX traffic analysis? |
| 14:43.11 | gaetronik | wireshark |
| 14:43.45 | Drognan | Ok, is there a plugin for looking at the call details? |
| 14:44.51 | gaetronik | Drognan, we use the cdr odbc module and the pentaho |
| 14:44.55 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:45.34 | gaetronik | but you can use whatever tools with a db |
| 14:45.34 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:46.48 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 14:46.51 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 14:47.28 | raz | anyone know what could be wrong when asterisks doesn't even seem to try to connect, desipte a "register=>" line in sip.conf? |
| 14:49.42 | xiando | raz: Just to start with basics, does normal SIP software like Twinkle and Ekiga connect to the service? |
| 14:49.52 | raz | xiando, yes |
| 14:50.07 | raz | it's sipgate.de, nothing exotic |
| 14:50.19 | raz | i set it up following this guide: http://www.geisterstunde.org/drupal/?q=asterisk_sipgate |
| 14:50.34 | raz | i wonder because "asterisk -c" doesnt show any "attempting to connect" messages during startup |
| 14:51.02 | xiando | does the debug info (sip set debug) give cluez? |
| 14:51.12 | browser | anyone who can tell why * 1.2 sends a fake call progress tone as 183 with SDP early-audio to a remote peer even if actual audio from the ISDN trunk is available. I have caught * to send at the beginning "correct" RTP and then steering over the fake progress tone. However, Asterisk does send the correct early-audio as RTP to a friend registered to the server. |
| 14:51.29 | raz | xiando, ahh |
| 14:51.35 | raz | it shows an empty read |
| 14:51.48 | raz | does that mean anything? |
| 14:51.58 | raz | in what config file can i put that line to have it enabled on startup? |
| 14:52.04 | freakynl | well thx a bunch gotta run |
| 14:52.12 | raz | oh sec, now there was a request |
| 14:52.23 | xiando | "sipdebug = no" or yes in sip.conf |
| 14:52.45 | raz | cool thx, i think i have something to chew on now |
| 14:52.49 | raz | sipgate gave me a 401 |
| 14:53.12 | gaetronik | Is there anyway to learn zaptel that the country changed without stopping asterisk? |
| 14:54.56 | coppice | wow, that's some major politcal change you're trying to cope with there :-) |
| 14:55.28 | M1s3ry | tomoconnor, try recompiling from scratch to make sure there are no issues there. I didn't see anything wrong with your configuration. Note though that that card is very old and is unsupported. If it comes down to it you may need to purchase a new E1 card |
| 14:55.39 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:55.41 | Kobaz | wtf |
| 14:55.41 | Zeeek | indeed every asterisk in the USA would need to reboot after 2000 and 2004 |
| 14:55.50 | Kobaz | relaxdtmf isn't helping dtmf detection anymore |
| 14:56.01 | tomoconnor | M1s3ry, this is from source |
| 14:56.02 | Kobaz | asterisk completely isn't getting any dtmf at all now |
| 14:56.11 | xiando | Hinweis: Ersetzen sie jeweils SIPID durch Ihre SIP-ID und PASSWD durch Ihr SIP-Passwort. |
| 14:56.15 | xiando | http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257 |
| 14:56.23 | tomoconnor | M1s3ry, i tried the ubuntu binaries.. and they sucked, so i got the source and built that by hand |
| 14:56.34 | Zeeek | ich habbe kein Camera |
| 14:56.40 | tomoconnor | M1s3ry, and everything else worked, like zaptel config tools and stuff |
| 14:56.45 | tomoconnor | except asteisk won't run |
| 14:57.19 | raz | xiando, hmm strange. outbound registration seems to succeed (i get a NOTICE for that) but inbound registration apparently gives a 401. using the identical credentials.. hrm |
| 14:57.19 | xiando | maby SIPID is different for password somehow, I don't know, I'm not German. Would be odd, though, since you get normal SIP clients working just fine |
| 14:57.30 | M1s3ry | how do you mean asterisk won't run... as in asterisk -r won't bring you to CLI, /etc/init.d/asterisk start won't start asterisk? |
| 14:58.10 | [TK]D-Fender | tomoconnor: pastebin your attempt to do "asterisk -gvvvvvc" |
| 14:58.29 | tomoconnor | ok |
| 14:58.31 | *** join/#asterisk ajricoveri (n=chatzill@201.248.93.18) |
| 14:59.06 | tomoconnor | [TK]D-Fender, http://pastebin.com/m68c28189 |
| 14:59.52 | [TK]D-Fender | tomoconnor: and now "ztcfg -vvvv" |
| 15:00.52 | tomoconnor | [TK]D-Fender, http://pastebin.com/m220f1947 |
| 15:01.04 | gaetronik | ANyone can confirm the loadzone and defaultzone paramters of zaptel can be correlated with end call detection? |
| 15:01.43 | [TK]D-Fender | tomoconnor: If you restart as I jsut had you test and it fails, pastebin "cat /proc/interrupts" |
| 15:02.02 | [TK]D-Fender | gaetronik: On what kind of circuit? |
| 15:02.10 | gaetronik | pri |
| 15:02.13 | gaetronik | e1 |
| 15:02.35 | tomoconnor | [TK]D-Fender, restart the entire box? |
| 15:03.01 | gaetronik | ztcfg will do the trick to load new conf |
| 15:03.12 | gaetronik | without killing everything |
| 15:03.40 | [TK]D-Fender | tomoconnor: no, "asterisk -gvvvvc" |
| 15:04.02 | tomoconnor | [TK]D-Fender, still fails as earlier pastebin |
| 15:04.03 | [TK]D-Fender | gaetronik: then no. Tones have nothing to do with disconnect. Thats digital to the telco |
| 15:04.13 | [TK]D-Fender | tomoconnor: then provide the next PB |
| 15:04.16 | tomoconnor | ok |
| 15:04.38 | gaetronik | [TK]D-Fender, so my problem is bigger |
| 15:04.57 | tomoconnor | [TK]D-Fender, http://pastebin.com/m1eb896c9 |
| 15:06.11 | [TK]D-Fender | tomoconnor: I'm not 100% sure but "t1xxp" seems suspicious as the driver name for that old card. I believe you should have an "e100p" module to load instead... |
| 15:08.30 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 15:08.43 | tomoconnor | [TK]D-Fender, the output from make config of zaptel drivers is: |
| 15:08.45 | tomoconnor | I think that the zaptel hardware you have on your system is: |
| 15:08.45 | tomoconnor | pci:0000:81:01.0 wct1xxp+ e159:0001 Digium Wildcard T100P T1/PRI or E100P E1/PRA Board |
| 15:09.00 | *** join/#asterisk moy (n=moy@nat/ibm/x-0c3ef15c718bc69f) |
| 15:09.15 | tomoconnor | i dunno if it's t100 or e100 |
| 15:09.25 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 15:09.41 | tomoconnor | [TK]D-Fender, it's uk isdn30 if that's important |
| 15:09.42 | [TK]D-Fender | tomoconnor: I'll take your word for it... not sure what to suggest ATM |
| 15:09.56 | CunningPike | ~centos52bug |
| 15:09.56 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 15:10.24 | tomoconnor | ok [TK]D-Fender, your help has been most helpful :) |
| 15:12.42 | ajricoveri | hi all, i'm trying to get my sip phone registered to my asterisk pbx over the internet, i have setup an extra interface on my pbx with a public address, when the pbx gets the REGISTER method from my sip phone it replies with a SIP/401 Unauthorized packet, what could this be?? http://pastebin.com/m3e47ce64 |
| 15:13.00 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 15:14.43 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:15.39 | [TK]D-Fender | ajricoveri: auth doesn't match your sip.conf |
| 15:15.43 | *** part/#asterisk tomoconnor (n=toconnor@rabbit.dbplc.com) |
| 15:16.01 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:16.31 | gaetronik | i've any lines like that in my log [Jul 18 10:59:32] NOTICE[25916] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 15:16.57 | ajricoveri | [TK]D-Fender: can you explain that in detail, please?? =) thanks |
| 15:17.12 | Kobaz | gaetronik: bad cable? |
| 15:17.28 | Kobaz | gaetronik: is it a rhino card? |
| 15:17.29 | Kobaz | heh |
| 15:17.30 | [TK]D-Fender | ajricoveri: The auth doesn't match. Either their IP/host is restricted, bad password, etc. |
| 15:17.56 | [TK]D-Fender | gaetronik: What card and what Zaptel version? |
| 15:17.58 | *** join/#asterisk skirmisha (n=5c425811@67.159.55.26) |
| 15:18.16 | ajricoveri | [TK]D-Fender: it is weird actually because sip phone does get get registered easily over the internal ip network, but not over the internet ?? |
| 15:18.38 | [TK]D-Fender | ajricoveri: Go look at his peer entry |
| 15:20.25 | ajricoveri | [TK]D-Fender, http://pastebin.com/m4a8ac572 |
| 15:20.29 | gaetronik | Te420B zaptel-1.4.11 |
| 15:20.49 | *** join/#asterisk s0lid (n=s0lid@122.53.100.86) |
| 15:20.54 | [TK]D-Fender | gaetronik: pastebin "cat /proc/interrupts" |
| 15:21.20 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 15:21.26 | [TK]D-Fender | ajricoveri: You've probably entered t wrong on the phone. |
| 15:21.33 | gaetronik | ~pb |
| 15:21.34 | 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 |
| 15:22.56 | ajricoveri | [TK]D-Fender, there had never been issues about asterisk and public addresses, doesn't the sip phone or sip.conf need extra config for those scenarios ?? |
| 15:23.03 | gaetronik | [TK]D-Fender, http://pastebin.com/m7b7ae257 |
| 15:24.08 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:25.15 | [TK]D-Fender | gaetronik: 18: 475425114 475409642 475324319 475178170 IO-APIC-fasteoi uhci_hcd:usb4, wct4xxp <- sharing an IRQ = bad |
| 15:25.21 | [TK]D-Fender | gaetronik: try to get it on its own. |
| 15:25.48 | [TK]D-Fender | ajricoveri: Yes, it does, but you have a straight auth problem. It's looking like the password si wrong. |
| 15:25.48 | gaetronik | how can i make it |
| 15:25.59 | [TK]D-Fender | gaetronik: in your BIOD |
| 15:26.01 | [TK]D-Fender | BIOS* |
| 15:26.34 | gaetronik | fuck it implies shutdown the server |
| 15:26.45 | gaetronik | and i don't use usb? |
| 15:26.52 | gaetronik | so does it really matter |
| 15:28.11 | *** join/#asterisk huey23 (n=huey23@65.111.241.212) |
| 15:29.23 | gaetronik | [TK]D-Fender, |
| 15:29.52 | huey23 | could someone take a look and see why we are dropping calls please? : http://pastebin.com/m5fd984eb |
| 15:29.54 | [TK]D-Fender | gaetronik: your error is indicative of lost interrupts. |
| 15:30.38 | [TK]D-Fender | huey23: span=0,1,0,esf,b8zs <-span *0*? WTF |
| 15:31.02 | [TK]D-Fender | huey23: pastebin ztcfg -vvvv" and "ca /proc/interrupts" |
| 15:31.26 | gaetronik | [TK]D-Fender, ok |
| 15:31.26 | *** join/#asterisk xarmiex (n=xarmiex@208.58.18.210) |
| 15:31.52 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:31.59 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 15:32.04 | raz | how do i play *.gsm files in linux? |
| 15:32.18 | Qwell | raz: play |
| 15:32.33 | raz | lol okay |
| 15:32.35 | raz | that was easy ;) |
| 15:33.36 | gaetronik | is there any guide to make it |
| 15:34.27 | huey23 | [TK]D-Fender: is there another command that i can use besides ca? |
| 15:34.40 | gaetronik | cat |
| 15:34.48 | gaetronik | huey23, cat |
| 15:34.49 | huey23 | i got it |
| 15:34.56 | huey23 | i figured he would have |
| 15:35.29 | *** join/#asterisk angryuser (n=sldf@88.140.144.209) |
| 15:35.40 | huey23 | [TK]D-Fender: http://pastebin.com/m1d6bdfa7 |
| 15:35.56 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 15:36.21 | rwaite | grr. this headset is echoing badly |
| 15:36.46 | [TK]D-Fender | huey23: 11: 1913393864 XT-PIC Mylex eXtremeRAID 2000, t1xxp, eth0 <--- shared with your RAID ADAPTER. Yuo should be dragged out and shot :) |
| 15:37.02 | [TK]D-Fender | huey23: AND *eth0* |
| 15:37.08 | [TK]D-Fender | loads up another clip |
| 15:37.14 | huey23 | i didn't set this bad boy up :P |
| 15:37.24 | huey23 | but i will die with honor |
| 15:37.48 | xarmiex | hm i got a weird issue here, went from 1.2 to 1.4 and now my agi's work at first and then eventually they will all stop working, i actually have to reboot the machine for them to start working again, the rest of the pbx will function fine when this is happening, has anyone ever seen that before ? |
| 15:37.57 | youseenothing | is g.711 good to use when you could possibly have 25 active calls on a t1 connect? |
| 15:38.17 | youseenothing | with a potential active call spike to 50? |
| 15:38.18 | [TK]D-Fender | youseenothing: Yes |
| 15:38.36 | youseenothing | thanks [TK]D-Fender |
| 15:38.38 | [TK]D-Fender | Yourname``: G.711 will incur almost no load on *. You're more than fine. |
| 15:38.51 | youseenothing | outstanding...thanks |
| 15:39.02 | xiando | $ echo 128*25|bc |
| 15:39.02 | xiando | 3200 |
| 15:39.10 | huey23 | [TK]D-Fender: do you have anything in that noodle that might have a fix for the HDLC and FCS errors from what you've seen? |
| 15:39.20 | xiando | 25 calls = 400 KB/s. What's "T1"? |
| 15:39.28 | huey23 | 1.5 |
| 15:39.46 | *** join/#asterisk angryuser (n=sldf@88.140.144.209) |
| 15:39.49 | xiando | 1.5 what? apples? pies? megabit? byte? |
| 15:40.31 | Corydon76-dig | 1.544Mbps |
| 15:40.32 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:40.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:41.00 | [TK]D-Fender | huey23: that combo FUBAR's you big time. |
| 15:42.03 | [TK]D-Fender | youseenothing: OH.. you mean T1 for DATA over the internet |
| 15:42.11 | [TK]D-Fender | youseenothing: No, that really won't work. |
| 15:43.46 | [TK]D-Fender | youseenothing: You'd need to use a much lighter codec to support 25 calls, and that's assuming you are locking it out for VoIP only. G.726 / GSM / G.729 would fit. |
| 15:43.56 | xiando | http://www.thrallingpenguin.com/articles/voip-dimensioning-and-bandwidth.htm story is that g711 uses 64Kbps, 87.2Kbps including overhead. That indicates that you can make 17 calls on the "T1" if it's 1544Mbps |
| 15:44.12 | youseenothing | yeah, locking it out for strictly voip |
| 15:44.30 | [TK]D-Fender | youseenothing: Whats your circuit cost? |
| 15:44.30 | huey23 | [TK]D-Fender: OK, we haven't seen dropped calls until today, something tripped them off, anything come to mind? i have checked the digium site and it is a big circle, dropped calls are caused by HLDC (8 and 6) errors and HLDC (8 and 6) errors cause dropped calls |
| 15:44.54 | [TK]D-Fender | huey23: Sounds like frame slips. A classic timing issue |
| 15:44.59 | Corydon76-dig | xiando: double that (approximately) with G.726 |
| 15:45.10 | youseenothing | well, it is only a temporary solution that i was handed...so i don't know the specifics...just trying to figure out best codec for use |
| 15:45.29 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 15:45.43 | [TK]D-Fender | youseenothing: G.726 is the best quality you can push through it. |
| 15:45.47 | youseenothing | ok |
| 15:45.50 | youseenothing | great...thanks |
| 15:46.43 | xiando | various sites indicate that GSM uses 13.2 Kbps, that's about 110 calls |
| 15:46.49 | youseenothing | but if my active calls is way below the 17 then i would be fine? |
| 15:47.01 | youseenothing | with g.711 |
| 15:47.23 | xiando | youseenothing: it does appear that way, yes. |
| 15:47.27 | huey23 | [TK]D-Fender: will changing the timing (which looks like someone did according to zaptel.conf) while in production cause issues or will the new config take place after new channels are opened? |
| 15:47.34 | youseenothing | outstanding...thanks guys |
| 15:47.47 | xiando | I don't actually know if alaw uses bandwidth when there is silence, does it? |
| 15:48.29 | gaetronik | why only 4 choices for interrupt |
| 15:48.48 | gaetronik | is this a limitation of x86 arch? |
| 15:49.21 | [TK]D-Fender | huey23: they cannot go into effect without reloading chan_zap |
| 15:51.13 | gaetronik | [TK]D-Fender, is there a good link to understand IRQ APIC and this bunch of things |
| 15:51.26 | gaetronik | french english or spanish |
| 15:52.13 | [TK]D-Fender | gaetronik: no idae |
| 15:52.39 | gaetronik | google is my friend |
| 15:53.08 | gaetronik | http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 15:55.11 | browser | anyone who can tell why * 1.2 sends a fake call progress tone as 183 with SDP early-audio to a remote peer even if actual audio from the ISDN trunk is available. I have caught * to send at the beginning "correct" RTP and then steering over the fake progress tone. However, Asterisk does send the correct early-audio as RTP to a friend registered to the server. |
| 15:56.08 | angryuser | .. |
| 15:56.11 | angryuser | ,, |
| 15:56.26 | |||Mad||| | Hi, all! Can someone tell me where the MWI logic is in Asterisk? I would like to have it send a DTMF to tell our analog phones to turn the MWI on/off |
| 15:56.34 | *** join/#asterisk angryuser (n=sldf@88.140.144.209) |
| 15:56.39 | angryuser | test |
| 15:56.55 | angryuser | test |
| 15:57.27 | [TK]D-Fender | |||Mad|||: How are thsoe phones connected to *? |
| 15:58.01 | |||Mad||| | They will be coming in on an analog port |
| 15:58.33 | [TK]D-Fender | |||Mad|||: What kind exactly? |
| 15:58.44 | angryuser | jj |
| 15:58.50 | gaetronik | [TK]D-Fender, this confirm interrupt problem |
| 15:58.51 | gaetronik | --- Results after 44 passes --- |
| 15:58.51 | gaetronik | Best: 100.000 -- Worst: 98.838 -- Average: 99.787640, Difference: 100.205781 |
| 15:59.04 | |||Mad||| | I've discovered that if the PBX receives at DTMF 62xxx it turns the light on, 63xxx turns it off for extension xxx |
| 15:59.06 | [TK]D-Fender | angryuser: Please stop spamming |
| 15:59.13 | angryuser | test |
| 15:59.46 | |||Mad||| | I will be installing a Sangoma A200 card with FXO ports |
| 16:00.25 | [TK]D-Fender | |||Mad|||: Go look in the source for where MWI events are triggered. "grep"-able. |
| 16:01.32 | Zeeek | *sorry about that |
| 16:01.41 | |||Mad||| | OK, can you point me in the right direction of where to start looking, please? |
| 16:01.44 | *** join/#asterisk kombi_ (n=kombi@port-92-198-15-96.static.qsc.de) |
| 16:01.54 | *** join/#asterisk berspolis (n=berspoli@190.25.228.235) |
| 16:02.34 | kombi_ | what do I do to not accept a second call for an extension? In other words, when boss speaks, signal busy to other callers? |
| 16:02.36 | berspolis | hello |
| 16:02.38 | Zeeek | VoIP Users Conference SIP 123@ts.x2z.eu DTMF 11847# 1# |
| 16:04.09 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 16:04.28 | [TK]D-Fender | kombi_: "core show chanisavail" |
| 16:04.38 | [TK]D-Fender | kombi_: "core show application chanisavail" |
| 16:05.23 | [TK]D-Fender | |||Mad|||: you prepared to write a large patch and maintain it through new releases? |
| 16:05.51 | *** part/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 16:07.03 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 16:07.13 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 16:07.58 | berspolis | I have a channel bank connected to asterisk server in the FXS Port of the card; the 24 ports of the channel bank have been configured as FXS channels...i need to use one of this ports as FXO channel to connect a GSM line |
| 16:08.07 | berspolis | is that possible? |
| 16:09.19 | [TK]D-Fender | berspolis: No. On standard PCI cards out there the port module is fixed as either FXO or FXS. It is physical circuitry and you can't tell it to try to be the other kind of module. |
| 16:10.49 | berspolis | so i should connect the channel bank to FXO port of the card |
| 16:11.36 | [TK]D-Fender | berspolis: You need to connect ports that are a proper match. If one side is FXO, the other must be FXS. |
| 16:11.51 | |||Mad||| | [TK]D-Fender: Nope, I guess I was on the mistaken assumption that it would be something simple :) |
| 16:12.08 | [TK]D-Fender | |||Mad|||: very much not. |
| 16:12.23 | |||Mad||| | Damn |
| 16:14.22 | rwaite | does ajam modify config files or is it kind of a read only thing |
| 16:14.33 | rwaite | i -dont- want it to edit the conf files, as i do those by hand |
| 16:14.36 | |||Mad||| | I noticed that it already talks to softphones and leaves a MWI, I assume it does the same with VOIP phones... I figured it was a command or two that shot a code to the device, and that could be replaced with the command to send a DMTF |
| 16:14.41 | |||Mad||| | DTMF |
| 16:15.30 | neurosys | hmm LookupCIDName is deprecated. ugh |
| 16:17.09 | [TK]D-Fender | rwaite: AJAM is for AMI, has nothing to do with your "config files" |
| 16:17.30 | [TK]D-Fender | |||Mad|||: No. |
| 16:18.00 | [TK]D-Fender | |||Mad|||: there is no quick plug-in to indicate this stuff. It is channel specific and requires C code directly. |
| 16:18.07 | [TK]D-Fender | |||Mad|||: A very serious undertaking. |
| 16:18.53 | [TK]D-Fender | |||Mad|||: You might be able to hack an ugly fake out however. Write a script that will poll the VM on your box, if the # changes, issue a "call file" or "AMI originate" to dial out your 'cancel code' |
| 16:19.43 | rwaite | [TK]D-Fender: so its like a framework for making web apps? |
| 16:19.53 | rwaite | (that communicate thru the ami) |
| 16:19.58 | [TK]D-Fender | rwaite: Its just another way to get to AMI. |
| 16:20.07 | rwaite | oh okay, i read incorrectly. good. |
| 16:20.29 | [TK]D-Fender | rwaite: Not entirely appriate to excessively tie the word "web" to it. |
| 16:20.37 | [TK]D-Fender | appropriate* |
| 16:21.49 | gaetronik | i will be glad to announce to the client that qe need to shutdown the prod server to change a bios setting |
| 16:21.57 | rwaite | [TK]D-Fender: well, considering it uses an http server built into asterisk, and is accessible thru a browser... |
| 16:22.15 | gaetronik | first step find an ather guy to say it |
| 16:22.25 | [TK]D-Fender | rwaite: Draw a line and I'll drop a bucket of paint on it... |
| 16:22.33 | [TK]D-Fender | gaetronik: SMRT |
| 16:23.11 | rwaite | _____________ |
| 16:24.35 | javb | hi, the pickup exten will not work when the call is comming via a zap chan . . . any idea? |
| 16:24.51 | javb | Pickup app, but when using *8 it will work |
| 16:24.53 | |||Mad||| | D-F: Makes sense, perhaps call that script every time the VM extension is accessed or something |
| 16:25.03 | huey23 | [TK]D-Fender: did you pull the trigger on me yet? i was just wondering if it was ok to die now |
| 16:25.24 | |||Mad||| | Thanks for the tip, I will look into scripting some more. |
| 16:26.06 | [TK]D-Fender | huey23: I said shot. I never said anything about being so merciful as to allow you to die ;) |
| 16:26.35 | huey23 | [TK]D-Fender: falls right in line with how this day's going...let me suffer |
| 16:26.35 | phpboy | Hey all I need to send through a dial command, but I've got a trunk-fallover Macro, how do I send through the fall over Macro or is it as simple as sending TRUNK1/${EXTEN},TRUNK2/${EXTEN} |
| 16:26.59 | [TK]D-Fender | phpboy: Depends on what that macro DOES. |
| 16:27.06 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 16:27.10 | [TK]D-Fender | phpboy: and GUI's are not supported here. |
| 16:28.32 | phpboy | [TK]D-Fender: this is not for a GUI, I'm writing an API, the macro just checks which of the two trucks in order of what args is sent through |
| 16:28.53 | [TK]D-Fender | phpboy: well that macro and what it does is your responsibility. |
| 16:29.07 | [TK]D-Fender | phpboy: what kind of "API" are you creating? |
| 16:29.44 | phpboy | [TK]D-Fender: I know, I got the Macro working now i need to send that directly to asterisk through an AGI script |
| 16:30.16 | gaetronik | javb, context |
| 16:30.24 | phpboy | I just need to figure out how to tell asterisk to use the macro instead of try to dial directly to the truck |
| 16:30.28 | phpboy | *trunk |
| 16:31.45 | javb | gaetronik, could you elaborate? |
| 16:32.06 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:32.45 | gaetronik | javb, pickup take a context |
| 16:32.49 | gaetronik | as an argument |
| 16:33.20 | gaetronik | and to pickup the call the context may different between sip call and zap call |
| 16:33.24 | javb | gaetronik, so i shouuld specifie the context to where the zap is attached? |
| 16:33.32 | gaetronik | javb, maybe |
| 16:33.41 | gaetronik | i'm not a specialist of this feature |
| 16:37.22 | Qwell | Zeeek: no call today? |
| 16:37.50 | phpboy | :/ |
| 16:38.41 | Zeeek | we had a problem |
| 16:38.45 | Zeeek | VoIP Users Conference SIP 123@ts.x2z.eu DTMF 11847# 1# |
| 16:38.54 | Qwell | use Asterisk. |
| 16:38.59 | Zeeek | but we're going to reschedule to next week |
| 16:39.05 | Zeeek | naw |
| 16:39.21 | Zeeek | wasn't a voip problem |
| 16:41.35 | Qwell | so then? |
| 16:43.34 | *** join/#asterisk CVirus (n=Satan@82.201.178.112) |
| 16:44.44 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:44.46 | phpboy | ah, I figured out what I'm going to do |
| 16:44.47 | phpboy | :D |
| 16:45.25 | mikealeonetti | should I put the IP phones on the network lines I already have for the office, or should I run separate lines? |
| 16:45.46 | gaetronik | mikealeonetti, depend on your network use |
| 16:45.51 | javb | gaetronik, now it works, but wont work for any call coming from another context... example, another T1 |
| 16:46.12 | javb | How would i make my pickup cmd to work for no matter which context the call is comming, works |
| 16:46.15 | *** join/#asterisk geek_cl (n=geek@190.54.42.62) |
| 16:46.20 | geek_cl | hi all |
| 16:46.33 | gaetronik | javb, if you find feel free to tell me |
| 16:51.29 | mikealeonetti | gaetronik: well, it has at least 20 workstations and at the most we might have 40 computers on the network at once. |
| 16:52.11 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 16:53.04 | [TK]D-Fender | mikealeonetti: Its technically advisable to run a completely separate subnet and wiring for your phone network. |
| 16:53.58 | De_Mon | pish, thats what .... shoot I just went blank |
| 16:54.09 | De_Mon | traffic shaping is fore |
| 16:54.31 | De_Mon | ducks (because he said fore) |
| 16:55.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:56.10 | mikealeonetti | [TK]D-Fender: okay. Tha'ts what I'll do. Thanks. |
| 16:57.20 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:59.31 | dark_one | anyone got some experiance with polycom phones, I am having trouble picking up calls, the line idicates that its ringing, but the remote party get sent to voice mail and the answer buttons do not work.... |
| 17:01.07 | rpm | dark_one: still having problems? if you can't answer calls it sounds like you have a nat problem and the ack/sip 200 ok is never being recieved by your proxy. |
| 17:01.46 | dark_one | rpm: well, i made some progress the phones boot now (btw it was having a /23 net that caused the phones to hang) |
| 17:02.15 | rpm | dark_one: ah, i remember hearing about that 255.255.254.0 mask bug. |
| 17:03.10 | dark_one | yeah, its kinda anoying as we have ~260 phones, but oh well just have to use /16.... |
| 17:04.29 | rpm | dark_one: just hopefully you don't have a power-outage and every phone try to reboot at the same time :P |
| 17:05.31 | *** join/#asterisk DexTerDDIT (n=xxx@ppp2481078633.ambra.ro) |
| 17:06.07 | *** join/#asterisk Hydrant (n=aj@74.210.126.107) |
| 17:06.11 | DexTerDDIT | i have a question , can i use a motorola pci modem for asterisk , or do i need a special pc card ? |
| 17:06.35 | Hydrant | hey all... playing with my openmoko... is there any chance of doing voip with 40kbits/s up 50kbits/s down ? |
| 17:07.03 | Hydrant | DexTerDDIT: you need a special card |
| 17:07.09 | Hydrant | DexTerDDIT: or SIP phones |
| 17:07.27 | Hydrant | DexTerDDIT: you can get a converter to make a normal phone a network device too |
| 17:08.55 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 17:09.15 | *** join/#asterisk pbxcocr (n=pbxcocr@201.195.71.60) |
| 17:09.43 | DexTerDDIT | hmm... so i can`t use the motorola modem to connect my asterisk pbx to my analog line? |
| 17:10.46 | pbxcocr | mmm intresting question, and i want to know if i can connect a regular modem and use it as a line of the asterisk. for test pourposes only. |
| 17:11.14 | *** join/#asterisk pta200 (n=paolo@63.162.54.226) |
| 17:11.28 | DexTerDDIT | pbxcocr we have the sama pourposes :D... |
| 17:11.48 | Hydrant | my understanding is that it won't work |
| 17:12.06 | Hydrant | you can check a wiki or something... there was some particular reason why, I forget it though |
| 17:12.06 | pta200 | anybody know what this error is? rtp.c: Unknown RTP codec 126 received from ... |
| 17:12.39 | DexTerDDIT | aha Hydrant 10x ... for the info :) |
| 17:12.58 | Hydrant | DexTerDDIT: 10x ? |
| 17:14.08 | pta200 | can't find that number listed anywhere |
| 17:16.48 | DexTerDDIT | Hydrant 10x = thank you .... sry old irc habit \ |
| 17:16.56 | Hydrant | DexTerDDIT: np |
| 17:17.05 | [TK]D-Fender | pbxcocr: No. |
| 17:19.31 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 17:19.42 | doolph | hello |
| 17:19.49 | doolph | how can I send commands via manager |
| 17:20.16 | *** join/#asterisk huey23 (n=huey23@65.111.241.212) |
| 17:21.01 | [TK]D-Fender | doolph: telnet <- |
| 17:21.02 | Nugget | telnet is eeeeeeevil! |
| 17:21.09 | doolph | really |
| 17:21.09 | [TK]D-Fender | puts the nugget-bot |
| 17:21.12 | *** part/#asterisk pta200 (n=paolo@63.162.54.226) |
| 17:21.35 | Hydrant | Nugget: right... that's why I use SSH when I'm communicating via two laptops in a car connected via a switch... |
| 17:21.40 | [TK]D-Fender | doolph: There is a nice chapter for this in THE BOOK. Go read it. |
| 17:21.49 | [TK]D-Fender | Hydrant: Its a triggered statement... |
| 17:22.07 | Wayhigh | sup all |
| 17:22.41 | Hydrant | [TK]D-Fender: did you see my question on whether voip will work with my low bandwidth ? |
| 17:22.53 | *** join/#asterisk Segnale007 (n=Segnale0@host141-4-dynamic.18-79-r.retail.telecomitalia.it) |
| 17:23.18 | [TK]D-Fender | Hydrant: More than enough. |
| 17:23.18 | doolph | i just get Asterisk Call Manager/1.0 |
| 17:23.38 | Hydrant | [TK]D-Fender: think so? I've gotten some conflicted reports |
| 17:23.58 | Hydrant | [TK]D-Fender: I'm thinking that I'll just get a data plan, and use my Freerunner with voip |
| 17:24.44 | [TK]D-Fender | Hydrant: I suggest GSM or G.729. I doubt its got the CPU for the latter, so work on GSM |
| 17:25.27 | Hydrant | [TK]D-Fender: thx... any interest yourself in getting a FR? |
| 17:25.44 | [TK]D-Fender | Hydrant: ? |
| 17:25.55 | Hydrant | I think it could make an awesome wireless voip phone |
| 17:25.58 | [TK]D-Fender | Hydrant: Oh... no, I'm stuck on CDMA |
| 17:25.58 | Hydrant | [TK]D-Fender: freerunner |
| 17:26.04 | Hydrant | [TK]D-Fender: CDMA ?> |
| 17:26.09 | *** part/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 17:26.13 | [TK]D-Fender | Hydrant: Well technically YES, but economically, no :) |
| 17:26.31 | Qwell | well, the battery life would be about the same as other voip phones |
| 17:26.40 | [TK]D-Fender | Hydrant: FR is a GSM phone. I'm with Bell Canada which uses CDMA for cell tech |
| 17:26.48 | Hydrant | [TK]D-Fender: ah... |
| 17:26.55 | Hydrant | [TK]D-Fender: I had to get eeevil rogers in canada |
| 17:26.56 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:27.11 | Hydrant | Qwell: what battery life is that ? |
| 17:27.18 | Qwell | garbage, basically |
| 17:27.26 | Hydrant | Qwell: I'm thinking you could get a couple days out of it once suspend / resume is in |
| 17:27.31 | Hydrant | Qwell: if it's always on |
| 17:27.36 | coppice | the battery life of most new phones is poor |
| 17:27.44 | [TK]D-Fender | Hydrant: My cell plan is pretty good : 37$+tax for 250dt, unlim eve/wk @ 6pm, VM, CID, unlimited internet on my HTC Touch |
| 17:27.51 | Qwell | coppice: sure - they've all got 500mhz processors in them now |
| 17:28.04 | Qwell | it's become pretty necessary to satisfy people these days |
| 17:28.23 | coppice | they are all 1mm too thin to have enough battery space |
| 17:28.32 | Qwell | eh? |
| 17:29.39 | jameswf-home | charges the blackberry every few days |
| 17:30.23 | Qwell | jameswf-home: my wife has to charge her sidekick every night |
| 17:30.32 | Qwell | it won't last 2 days |
| 17:30.53 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 17:31.59 | Hydrant | how flexible is asterisk? Like I want to run a python script when a call comes in... basically pass the call to the python script in a sense |
| 17:32.15 | Qwell | sure, AGI |
| 17:32.31 | Qwell | Asterisk can do pretty much anything you can think of. "It's just software." |
| 17:32.36 | Kobaz | where can i get a pinout diagram for an amphanol connector |
| 17:32.42 | Qwell | Kobaz: google. ;) |
| 17:32.54 | Kobaz | yeah i'm googleing i'm googleing |
| 17:33.17 | Qwell | first hit for 'amphenol connector diagram'... |
| 17:33.25 | Kobaz | ah |
| 17:33.31 | Kobaz | i clicked on images |
| 17:33.34 | phpboy | Hydrant: what info do you want passed to the python script? |
| 17:33.35 | Kobaz | and got a few |
| 17:33.44 | Qwell | 3rd hit on images |
| 17:34.01 | Kobaz | Qwell: heh, i was searching for amphanol wireing diagram and got a bunch of pdfs with no diagrams |
| 17:34.01 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 17:34.06 | Kobaz | i found some on images though |
| 17:34.33 | jaytee | If you were ever able to get ahold of Felix the Cat, The wonderful, wonderful cat's bag of tricks and looked inside, Asterisk is what you'd find. |
| 17:34.50 | phpboy | LOL :( |
| 17:35.03 | phpboy | ok, I need to get a grip |
| 17:35.20 | phpboy | it's 19h35 on a Friday night and I'm still at the office :( |
| 17:35.34 | Hydrant | well I can tell you what I want to do... have someone call my DID... then when I get a call I want to see if my freerunner phone is on... if not send them to voicemail, otherwise I want to then send a command to my phone to get it ready to receive a call via voip... then have the script route the call through to that SIP device once it's on... think push voice |
| 17:35.35 | Qwell | phpboy: go home |
| 17:35.43 | phpboy | I best do that |
| 17:35.49 | Qwell | tell your boss Qwell said you could. |
| 17:35.58 | phpboy | I will do that :D |
| 17:35.59 | jaytee | says "Whoa!" in his best impression of Keanu Reeves. Dude, go home! |
| 17:36.04 | jaytee | or go drinkin! |
| 17:36.15 | phpboy | I best go drinking |
| 17:36.17 | phpboy | BUT |
| 17:36.17 | Qwell | jaytee: emotion? |
| 17:36.27 | phpboy | i have a meeting tomorrow morning at around 9am |
| 17:36.33 | Qwell | from a Keanu impression? |
| 17:36.34 | phpboy | although, I don't think it's on anymore :T |
| 17:36.37 | jaytee | Qwell: context? |
| 17:36.46 | Qwell | context: he's the worst "actor" ever. :p |
| 17:36.51 | phpboy | ok, bye |
| 17:36.53 | phpboy | *gone* |
| 17:36.53 | Qwell | (though, he's awesome) |
| 17:37.46 | jaytee | Qwell, he always plays the same 'character' just with different names, kinda like Steven Seagall only with less angry whispering and martial arts. |
| 17:39.08 | [TK]D-Fender | jaytee: http://geekadelphia.com/wp-content/uploads/2008/04/steven_segal_emotion_chart.jpg |
| 17:39.11 | jaytee | I liked Keanu in Matrix 1, the other two sucked |
| 17:39.21 | cpm | agrees |
| 17:39.40 | jaytee | and A Walk In The Clouds made me want to slash my ankles while standing in the shower (neatness freak) |
| 17:40.20 | jaytee | but I liked Sweet November although that was because of Charlize Theron for whom I'd commit genocide and sell my soul to the devil for one night with. |
| 17:40.57 | jaytee | but the trailers for the remake of The Day The Earth Stood Still look pretty damn good. |
| 17:41.32 | jaytee | [TK]D-Fender, that's both frikken hilarious and absolutely true!!! |
| 17:49.59 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 17:55.53 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:58.47 | Wayhigh | what's the name of the bot these days so I can ask it for some info? |
| 17:59.31 | M1s3ry | jbot |
| 17:59.44 | Wayhigh | jbot: recording? |
| 17:59.44 | jbot | i heard recording is a little bit of a weakness as you can only use the internal microphone which is small by necessity. At least I think that is the case. |
| 18:00.13 | M1s3ry | ~[TK]D-Fender |
| 18:00.14 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:00.30 | Wayhigh | ~Wayhigh |
| 18:00.31 | jbot | Asterisk mouse WAZ in his 1U, eatinz his thermo ribbons.. HE R MOUSEKILLA |
| 18:00.46 | Wayhigh | hahaha still has that.. awesome |
| 18:00.51 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 18:00.59 | M1s3ry | hhhm... lets see |
| 18:01.08 | M1s3ry | ~M1s3ry |
| 18:01.25 | M1s3ry | cries |
| 18:04.30 | Wayhigh | is there an easier way to record all incoming calls than setting up Monitor or MixMonitor per extension? |
| 18:04.47 | Wayhigh | I've got outgoing figured out via the dialout-trunk but I haven't found a global recording for incoming |
| 18:05.05 | [TK]D-Fender | Wayhigh: No. |
| 18:05.16 | [TK]D-Fender | Wayhigh: Your dialplan does what you tell it to, and you have to tell it everything |
| 18:05.56 | [TK]D-Fender | Wayhigh: You don't have to call those apps IMMEDIATELY before your "dial" however. |
| 18:05.56 | Wayhigh | fender: thanks.. I'll have to create a script to do it for me then |
| 18:06.21 | [TK]D-Fender | Wayhigh: So if you have a place higher up in your heirarchy then that'll do. |
| 18:06.52 | Wayhigh | fender: I appreciate the information. Thanks a bunch |
| 18:11.48 | *** join/#asterisk Itiliti (n=Itiliti@75.150.198.1) |
| 18:12.06 | *** join/#asterisk atis_work (n=atis_wor@c158.csc.lv) |
| 18:18.04 | Wayhigh | fender: so, on trixbox, if I was to setup mixmonitor to record from the [ext-local] section of extensions_additional.conf, it would affect all the extensions listed under [ext-local], right? |
| 18:18.36 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 18:19.08 | Qwell | ~trixbox |
| 18:19.09 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 18:22.23 | jameswf-home | heh http://youtube.com/watch?v=7ZuT5mCvki0 |
| 18:23.27 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 18:24.28 | Wayhigh | qwell: nice FO.. |
| 18:24.45 | Qwell | Wayhigh: the answer is "we don't know" |
| 18:25.03 | jameswf-home | FO? |
| 18:25.06 | Wayhigh | guess I'll just try it out and see what happens |
| 18:25.21 | Wayhigh | FO.. Fuck Off message |
| 18:25.55 | jameswf-home | oh thats about 80% of this room :)) |
| 18:26.24 | Wayhigh | yep.. |
| 18:26.36 | Wayhigh | ~seen stotaro |
| 18:26.42 | jbot | i haven't seen 'stotaro', Wayhigh |
| 18:26.55 | jameswf-home | asterisk users are elitest a-holes.. not as bad as debian or bsd folks but yeah pretty bad... find someone who uses asterisk on debian or bsd and its all over |
| 18:27.08 | Wayhigh | <-- bsd elitist.. |
| 18:27.21 | Wayhigh | definitely wont use asterisk on bsd though |
| 18:27.33 | Wayhigh | there's something to be said for ease of use ya know |
| 18:27.56 | jameswf-home | Qwell: can be nice but you have to bribe him with scooby snacks |
| 18:28.25 | Qwell | when it comes to GUIs, we cannot help, because we don't use them |
| 18:28.39 | Qwell | so expecting help for something like that is perhaps a bit silly |
| 18:28.59 | jameswf-home | pitures a funny accent "Gui? we need no stinkin gui" |
| 18:29.08 | Qwell | they each have their own support channels, with people who know/use them. |
| 18:29.53 | Qwell | jameswf-home: have they fixed that stupid vuln yet? |
| 18:30.08 | jameswf-home | learned when it takes 20 min to figure out how to do it in a gui or 2 min to type its faster to do it by hand |
| 18:30.44 | jameswf-home | Qwell: dunno even if they have folks are to afraid to yum update so ... |
| 18:30.53 | Qwell | heh |
| 18:30.56 | Qwell | they should be afraid |
| 18:31.15 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 18:31.39 | jameswf-home | I was quite impressed 8 seconds to break a box with zero work, wtf why wouldnt you start attacking boxes |
| 18:31.56 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 18:32.52 | jameswf-home | if i didnt have 30,000 projects I would probably go capture a hundred boxes and have them all vote for reality TV |
| 18:33.17 | ibnolqaiyem | if i have wildcard tdm 400p and analog telephone , am i needing channel bank for analog telephone? |
| 18:33.44 | jameswf-home | ~tdm400p |
| 18:33.44 | jbot | tdm400p is, like, http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 18:34.31 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 18:36.16 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 18:36.22 | neurosys | how do i get into the asterisk CLI to watch realtime debuging? |
| 18:36.37 | putnopvut | neurosys: asterisk -r |
| 18:36.50 | putnopvut | or when you start asterisk, use the -c flag. |
| 18:37.00 | neurosys | cool. thx |
| 18:37.43 | [TK]D-Fender | ibnolqaiyem: No. The phone plugs directly into the card |
| 18:38.24 | ibnolqaiyem | [TK]D-Fender, thank you |
| 18:38.27 | neurosys | "Failed to authenticate on invite" for an outbound call. i have canreinvite=yes in my sip.conf for that phone. |
| 18:39.35 | [TK]D-Fender | Wayhigh> fender: so, on trixbox, if I was to setup mixmonitor to record from the [ext-local] section of extensions_additional.conf, it would affect all the extensions listed under [ext-local], right? <- This shows you have no comprehension of how the dialplan works at all. |
| 18:43.21 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:43.46 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 18:47.16 | [TK]D-Fender | neurosys: means the creds are bad. "canreinvite" doesn't come into play here. |
| 18:48.53 | neurosys | [TK]D-Fender: Creds are bad to the asterisk box or the provider? |
| 18:49.14 | [TK]D-Fender | neurosys: The provider is never wrong :) |
| 18:49.25 | neurosys | [TK]D-Fender: lol too right |
| 18:50.30 | neurosys | hmm the username and pw are correct :( |
| 18:51.17 | pputman- | http://www.sonnyradio.com/oldphones.htm haha that's great |
| 18:51.29 | *** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 18:53.13 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 18:56.50 | rpm | dark_one: 40632: Phones hang at the welcome screen when DHCP server specifies a |
| 18:56.53 | rpm | subnet mask of 255.255.254.0 |
| 18:56.54 | rpm | there it is. |
| 18:57.42 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 18:57.51 | [TK]D-Fender | bad mask again. |
| 18:58.04 | [TK]D-Fender | CIDR breaks tons of consumer devices |
| 18:58.11 | [TK]D-Fender | (non class based) |
| 18:58.15 | DSpair | Hye gang... Can someone point me to information on debugging iaxmodem connections? My iaxmodems are not even trying to dia out through asterisk. |
| 18:58.20 | DSpair | s/dia/dial/ |
| 18:59.06 | DSpair | When I use minicom, I attempt "ATDT XXXXXXX" and I get "ERROR" as the response with no activity within asterisk. |
| 18:59.06 | [TK]D-Fender | DSpair: Do you see IAXmodem registering? |
| 18:59.11 | DSpair | [TK]D-Fender, Yes |
| 18:59.41 | DSpair | Is there a way to increase logging output from iaxmodem? |
| 18:59.47 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 18:59.50 | [TK]D-Fender | DSpair: no idea. |
| 19:00.02 | DSpair | The documentation on iaxmodem is rather sparse. |
| 19:00.11 | neurosys | [TK]D-Fender: What else besides the username and pw would cause the auth failure? |
| 19:00.41 | [TK]D-Fender | neurosys: SIP domain, PW encoding, all sorts. Go check with your ITSP as to what your peer should look like. |
| 19:05.06 | neurosys | [TK]D-Fender: but it doesnt appear to be leaving my asterisk box at all |
| 19:05.19 | [TK]D-Fender | neurosys: PASTEBIN is your friend. |
| 19:05.26 | neurosys | :) |
| 19:06.53 | DSpair | [TK]D-Fender, Stangely enough, inbound calls to the IAX modems pick up!!! |
| 19:06.56 | MatBoy | has someone ever seen a script that can point someone to an IVR when a caller has entered some digits and these digits are used to quiry a DB ? |
| 19:07.02 | MatBoy | *query |
| 19:07.25 | [TK]D-Fender | MatBoy: Fo read the section of the book on func_odbc |
| 19:07.28 | [TK]D-Fender | Go* |
| 19:07.46 | MatBoy | [TK]D-Fender: that is nice advice, thanks |
| 19:08.18 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.11.249) |
| 19:08.28 | hardwire | he's a pretty nice guy |
| 19:08.31 | hardwire | send him cookies |
| 19:09.44 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:09.49 | FinboySlick | Under 1.4.20.1, I get: "sSMTP[21072]: 501 5.1.7 Bad sender address syntax" ever in syslog every time asterisk tries to send voicemail notification. Is there a way for me to see *how* asterisk tries to send that mail? |
| 19:09.51 | outtolunc | hash brownies ! |
| 19:10.18 | [TK]D-Fender | FinboySlick: TCPDUMP port 25 |
| 19:10.26 | *** join/#asterisk gaetronik (n=gaetan@190.22.6.108) |
| 19:10.43 | MatBoy | [TK]D-Fender: always nice to have you aroind |
| 19:10.45 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
| 19:10.46 | MatBoy | *around |
| 19:11.14 | gaetronik | hi |
| 19:11.25 | neurosys | [TK]D-Fender: ok up on pastebin |
| 19:11.42 | outtolunc | guess which one <G> |
| 19:11.56 | *** join/#asterisk ecrist (n=ecrist@t111-gw.c7200-1.bdr6.mpls.iphouse.net) |
| 19:15.05 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 19:17.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:17.07 | dark_one | rpm: yeah, switched to a /24 however now I cant make the answer button work ... |
| 19:18.00 | neurosys | [TK]D-Fender: http://pastebin.com/d2a924450 |
| 19:18.05 | *** join/#asterisk DevilSoulBlacK (n=devilsou@srv.ec-gye.internet.geainternacional.com) |
| 19:18.33 | DevilSoulBlacK | hi |
| 19:19.08 | DevilSoulBlacK | any one know search engine for look up the audio record from asterisk ? |
| 19:19.43 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-150.nys.biz.rr.com) |
| 19:20.50 | [TK]D-Fender | neurosys: Doesn't look like comprehensive SIP debug to me.. |
| 19:21.03 | [TK]D-Fender | neurosys: And that IS an answer. |
| 19:21.14 | [TK]D-Fender | neurosys: your creds are still wrong. |
| 19:22.08 | [TK]D-Fender | DevilSoulBlacK: ...huh? |
| 19:22.35 | FinboySlick | [TK]D-Fender: Thanks, that got me a little further. Apparently " postfix/smtpd[20283]: warning: Illegal address syntax from unknown[w.x.y.z] in MAIL command: <asterisk@>" Interstingly, that is not what 'servermail=' is set to in voicemail.conf |
| 19:22.41 | Whitor | Hi. I've having What I believe to be codec issues connecting a "analog -> voip" gateway device to my asterisk server... Codecs available are G711a, G711u, G7231, G729, GSM and ILBC.... Any suggestions on which ones work best ? |
| 19:23.11 | neurosys | [TK]D-Fender: heh ok :) Ill keep digging. sip set debug? |
| 19:23.20 | [TK]D-Fender | FinboySlick: missing a domain there. |
| 19:23.54 | [TK]D-Fender | neurosys: For the next time, yes, but nothing to dig in there now. You HAVE a response and its saying "your setup is bad". |
| 19:24.41 | [TK]D-Fender | Whitor : way to stick around for an answer :p |
| 19:24.50 | jaytee | "Hi! I've got my head lodged in my ass and I'm sure it's a codec issue. Can someone give me an immediate solution to my problem before I quit?" |
| 19:25.19 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-150.nys.biz.rr.com) |
| 19:25.33 | [TK]D-Fender | Whitor: Oh, you DO want an answer :p |
| 19:25.36 | Whitor | sorry abut that ... I accidentally killed my session |
| 19:25.56 | [TK]D-Fender | Whitor: Any of thiose will work save G723 |
| 19:25.56 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 19:26.06 | Whitor | heh [TK]D-Fender That would be nioce ... or at least a suggestion :) |
| 19:26.06 | [TK]D-Fender | Whitor: G.711 recommended if you can spare the BW |
| 19:26.46 | Whitor | its on G711a riht now... and I'm getting some x-talk dropouts |
| 19:27.11 | [TK]D-Fender | Whitor: Whats the networking between it and *? |
| 19:27.20 | Whitor | hehe... there is the crux |
| 19:27.28 | Whitor | Good Q... its a wireless bridge |
| 19:27.40 | Whitor | but its the only voip traffic on it |
| 19:27.42 | jaytee | gaaaaahhhh!!!!!!! |
| 19:27.56 | jaytee | makes sign of the cross |
| 19:27.58 | Whitor | and there should be ample bandwidth |
| 19:28.00 | Whitor | Thanks Jaytee |
| 19:28.07 | FinboySlick | [TK]D-Fender: Well, not in serveremail= in my voicemail.conf. It's set to a full and valid address. Maybe 1.4.x gets the value for its outgoing mail elseqhere? It worked in 1.2.x |
| 19:28.20 | Whitor | I know... I know... tell my super .... He just like ... make it better - |
| 19:29.08 | [TK]D-Fender | Whitor: Dropouts is an easy guess of packet loss & jitter which WiFi could never EVER be susceptable to. </sarcasm> |
| 19:34.47 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 19:35.02 | MatBoy | mh, not much info about examples for func_odbc and query an external DB and move to an IVR |
| 19:35.23 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 19:36.01 | [TK]D-Fender | MatBoy: IVR is jsut IVR. Dozens of samples. Func_odbc is in the book, and man pages, docs in your source folder, etc. |
| 19:36.28 | neurosys | [TK]D-Fender: you know what . It appears by the verbose debug that the provider IS rejecting it. |
| 19:37.40 | [TK]D-Fender | neurosys: I've told you that repeatedly and even that 1-liner you've pasted twice confirm it instantly. |
| 19:37.50 | MatBoy | [TK]D-Fender: yes true, but most of the time tehre are some live examples out there... I will investigate further |
| 19:38.50 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 19:40.54 | neurosys | [TK]D-Fender: :) But thanks for your patience anyways ;) |
| 19:41.26 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 19:41.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:48.38 | rpm | in the polycom 3.0.3revb firmware, i just noticed that bitmap.IP_[MODEL].61.name is not always at IdleDefault image.. anyone know the alternate XML attribute for IdleImage? |
| 19:48.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:49.23 | [TK]D-Fender | rpm: They are not the same #'s for each model. |
| 19:49.31 | [TK]D-Fender | rpm: not a trait I like about them... |
| 19:49.50 | rpm | yeah, it is bad design. |
| 19:51.55 | FinboySlick | serveremail in voicemail.conf is apparently completely ignored in my setup. I tried to force it on a per-user basis too and it still doesn't work :P It insists on using asterisk@ instead of what I specified. |
| 19:52.31 | [TK]D-Fender | FinboySlick: How did you apply the change? pastebin your config as well |
| 19:53.31 | FinboySlick | [TK]D-Fender: I tried reload from the console, and I also flat-out restarted asterisk. I'll remove passwords from the config and pastebin it, one moment |
| 19:54.00 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 19:54.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:58.27 | FinboySlick | [TK]D-Fender: http://www.pastebin.ca/1075869 |
| 20:00.26 | MatBoy | [TK]D-Fender: did you already used it btw ? I mean, not that you have to give me a full solution but the info is really not that people are really using it |
| 20:02.03 | [TK]D-Fender | FinboySlick: You don't have a "mailcmd" set in there. You might want to trap exactly how your binary is getting called |
| 20:02.22 | [TK]D-Fender | MatBoy: No, but there is plety of docs out there on it. |
| 20:02.50 | [TK]D-Fender | MatBoy: Go read, go try and when you have gotten somewhere and need that little something extra, come back and we might be able to help. |
| 20:04.05 | MatBoy | [TK]D-Fender: yes, I'm doing that :) |
| 20:06.16 | FinboySlick | [TK]D-Fender: That was my initial question... Though I guess I can toss together a shell script that will dump what's being called. |
| 20:07.01 | [TK]D-Fender | FinboySlick: Good idea to strip out the comments from there for readability as well. check the domin set in your hosts file,etc |
| 20:07.31 | MatBoy | [TK]D-Fender: Asterisk cmd MYSQL may be nicer |
| 20:07.44 | [TK]D-Fender | MatBoy: If thats available, sure |
| 20:07.49 | MatBoy | [TK]D-Fender: :) |
| 20:08.00 | [TK]D-Fender | MatBoy: Then again, AGI is much nicer still. |
| 20:08.08 | MatBoy | [TK]D-Fender: my life is about php and mysql :) |
| 20:08.10 | MatBoy | I love it |
| 20:10.00 | FinboySlick | [TK]D-Fender: Yeah, sorry about the comments... I assume I might get it working through default system settings but I still wonder why asterisk would ignore its own config in this respect. |
| 20:10.35 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 20:11.06 | rpm | i cannot find anywhere documented on polycoms extranet resource site the listing of phone configurations with which element is the IdleDisplay.. err |
| 20:12.43 | DSpair | Woohoo... Figured out how to enable debugging in iaxmodem. |
| 20:13.02 | DSpair | There's config file options for dspdebug and iax2debug |
| 20:20.51 | [hC] | rpm: what are you trying to find out? |
| 20:22.58 | rpm | [hC]: sip load 3.0.3revb which xml element in the <bitmaps></bitmaps> section for each phone i put for the name of the "IdleImage" in version 2.1.2 it was always bitmap.IP_[model].61.name="".. now it has changed. |
| 20:23.40 | [hC] | ohh. I see. I was going to lead you to the bitmap.IP_[model] |
| 20:23.44 | [hC] | Didnt know it changed in 3.0.3 |
| 20:24.24 | rpm | in 3.0.3 for the Polycom 600/601 the default for .61 is bitmap.IP_600.61.name="DiagnosticFrame6" |
| 20:24.58 | *** part/#asterisk DevilSoulBlacK (n=devilsou@srv.ec-gye.internet.geainternacional.com) |
| 20:24.58 | *** join/#asterisk DevilSoulBlacK (n=devilsou@srv.ec-gye.internet.geainternacional.com) |
| 20:25.46 | [TK]D-Fender | ok, heading home, bbiab |
| 20:27.12 | rpm | wow, lightning storm.. making all my power flicker. good thing for power conditioners. |
| 20:35.40 | jbeez | rpm: thunder just rolling in here |
| 20:37.26 | rpm | jbeez: you in alberta too? |
| 20:38.00 | rpm | http://www.theweathernetwork.com/index.php?product=alerts&placecode=caab0049®ion=wwcaab0006 <- severe thunderstorm watch.. seems we've been getting these almost every day for the last 3 weeks |
| 20:38.45 | rpm | gotta love global warming |
| 20:39.00 | jbeez | rpm: philly area |
| 20:48.17 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 20:48.22 | Idle | Qwell: poke |
| 20:48.29 | Qwell | ? |
| 20:48.41 | Idle | Qwell: libpri 1.4.5, what changed in it? |
| 20:48.52 | Qwell | dunno, read the ChangeLog |
| 20:48.58 | huey23 | version numbers :0 |
| 20:49.05 | Idle | Qwell: :P |
| 20:49.11 | Qwell | every commit is in there |
| 20:49.17 | Idle | we've done that, but aparently one was a branch merge |
| 20:49.32 | Idle | ie: not everything :( |
| 20:53.56 | Idle | of course, that wasn't your commit.. |
| 21:01.09 | *** join/#asterisk gaetronik (n=gaetan@190.22.17.146) |
| 21:02.56 | *** part/#asterisk fogo (n=fogo@rs-69-169-132-121-0003.broadweave.net) |
| 21:03.21 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:06.53 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-121-0003.broadweave.net) |
| 21:08.33 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 21:09.05 | arekm | hello, anyone using bristuffed zaptel/asterisk? I have a problem with tons of "ztgsm: TX buffer overflow on span 0" and not working setup |
| 21:12.12 | *** join/#asterisk moy (n=moy@nat/ibm/x-4acde9cad8df01fe) |
| 21:27.47 | *** join/#asterisk angryuser (n=sldf@88.140.144.209) |
| 21:31.19 | Idle | jesus... looking at these PRI debugs, there is something seriously wrong with asterisk.... |
| 21:31.21 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 21:31.48 | WhiteWolf | how so |
| 21:32.01 | Idle | heres the phone number being sent as a called party: |
| 21:32.06 | Idle | 20 8f 02 82 01 40 48 51 8c 81 d4 ....@HQ... |
| 21:32.31 | Idle | sometimes it sends channel number of 52 (0x34) |
| 21:32.41 | Idle | and it never sends the called party correctly when it fails |
| 21:32.48 | WhiteWolf | fun |
| 21:33.00 | Idle | yea, Digium tech support is all 'uhhh, what the hell' |
| 21:33.28 | Idle | asterisk, via its own PRI debug, says its OK |
| 21:36.13 | Idle | like, it only sees the 1, correct message |
| 21:36.23 | Deeewayne | Idle: would you mind pastebin'ing more of that trace ? |
| 21:36.34 | Idle | sure |
| 21:36.37 | Idle | I |
| 21:36.47 | Idle | er, I am going to obfuscate the phone numbers |
| 21:37.18 | Deeewayne | ok |
| 21:38.00 | jbeez | its, 1a8a0a0a5a5a5a1a2a1a2a there, no one will ever figure it out :D |
| 21:38.08 | Idle | http://uuoc.com/2072 |
| 21:38.38 | Idle | the numbers listed as 000000000 are actually correct calling and called party |
| 21:38.46 | Idle | the first message is raw, no modifications |
| 21:39.24 | Idle | it sends both of these message directly after each other. the first one _sometimes_ sends the wrong channel id, its 17 there, but sometimes it sends 0x34, etc |
| 21:40.12 | Idle | the second of the two, I am told, is correct, but because it already recieved that reference number, it dumps the call |
| 21:43.01 | Deeewayne | what did you use to get that trace ? |
| 21:43.47 | Qwell | Idle: the guy who runs uuoc.com is a newb |
| 21:43.51 | Qwell | Idle: oh, and flyback thinks you're dead. |
| 21:44.04 | Idle | Qwell: seriously, tell him I am dead |
| 21:44.11 | Idle | Deeewayne: the Telco has some analyzer |
| 21:44.19 | Idle | Deeewayne: MOST calls are successful |
| 21:45.11 | Idle | Qwell: and yea, hes a total newb, I hate that guy |
| 21:47.11 | Deeewayne | hrm. I was hoping to see pri intense debug output so I can see what might be happening. |
| 21:47.28 | Idle | Deeewayne: thats just it, pri intense debug shows no issues |
| 21:47.42 | Idle | Deeewayne: talk to John in tech support, hes been scritinizing the whole deal |
| 21:47.49 | Deeewayne | ok |
| 21:47.51 | Idle | hes the T1 guy, aparently |
| 21:53.22 | Idle | I wonder if the T1 card is sending the first setup message before asterisk has filled its buffer |
| 21:53.55 | Idle | cause, its always at aprox the channel number... and then everything after is all blown to hell |
| 21:54.10 | Idle | so it just sends an uniitialized buffer.... hmmm |
| 21:54.22 | Idle | maybe I'm on to sometihng... or maybe I'm an idiot... |
| 21:54.37 | Idle | I can't spell, thats a vote for #2 |
| 21:57.07 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 21:58.18 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 22:00.14 | *** join/#asterisk _khan (n=shariq@124.29.194.207) |
| 22:05.55 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 22:07.27 | ZX81 | hi all, I have a pri that is not hanging up when it gets a hangup message from pstn |
| 22:07.34 | ZX81 | http://pastebin.ca/1075985 |
| 22:07.38 | ZX81 | Cause code is 16 |
| 22:07.47 | ZX81 | Message type: DISCONNECT (69) |
| 22:07.54 | ZX81 | Asterisk receives the message |
| 22:08.00 | ZX81 | (pri intense debug span 1) |
| 22:08.09 | ZX81 | Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) |
| 22:08.13 | ZX81 | but then does nothing |
| 22:08.29 | ZX81 | the call hangs up about 15 seconds later on a timer recovery |
| 22:08.37 | MikeJ | ZX81: is the channel actually up at the time? |
| 22:08.46 | ZX81 | nah |
| 22:08.47 | MikeJ | or does asterisk think it is |
| 22:08.53 | ZX81 | well before the hangup it is |
| 22:08.54 | ZX81 | :) |
| 22:08.58 | MikeJ | heh |
| 22:09.00 | ZX81 | :) |
| 22:09.07 | MikeJ | should go --> DISCO |
| 22:09.07 | ZX81 | Asterisk takes the call |
| 22:09.13 | MikeJ | <--- RELEASE |
| 22:09.16 | ZX81 | I press hangup on my cell |
| 22:09.19 | MikeJ | ---> RELEASE COMPLETE |
| 22:09.27 | ZX81 | the message comes in |
| 22:09.40 | ZX81 | disconnect is from far end |
| 22:09.44 | ZX81 | so reverse the picture? |
| 22:09.51 | outtolunc | 15 second disconnect supervision on a cell phone can be fairly normal |
| 22:09.56 | ZX81 | < Message type: DISCONNECT (69) |
| 22:10.06 | MikeJ | outtolunc: he gets the disconnect right away |
| 22:10.10 | ZX81 | nothing gets sent out |
| 22:10.19 | outtolunc | which libpri is he using? |
| 22:10.31 | ZX81 | latest |
| 22:10.34 | ZX81 | 1.4svn |
| 22:10.46 | MikeJ | and your comment about dico supervision 15 secs on cell is nonsense |
| 22:10.51 | ZX81 | q931.c:3779 q931_receive: call 3 on channel 1 enters state 12 (Disconnect Indication) |
| 22:10.58 | ZX81 | Sending Receiver Ready (114) |
| 22:11.09 | ZX81 | > Supervisory frame: |
| 22:11.15 | ZX81 | > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] |
| 22:11.22 | ZX81 | but no release |
| 22:11.29 | ZX81 | and hence no release complete |
| 22:11.44 | MikeJ | ZX81: crank 931 debug all the way up.. and turn 921 debug off.. 921 is fine |
| 22:11.54 | ZX81 | how? |
| 22:11.56 | ZX81 | :) |
| 22:12.08 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 22:12.25 | ZX81 | how do I just log 931? |
| 22:12.32 | MikeJ | hmmm |
| 22:12.37 | MikeJ | been a while.. I don't recall |
| 22:12.48 | ZX81 | can't see a way |
| 22:12.49 | MikeJ | libpri has a bitmask for the debug... |
| 22:12.55 | ZX81 | oh in src? |
| 22:13.03 | MikeJ | so I know you can from the api.. it must be exposed somehow |
| 22:13.11 | MikeJ | i can't recall |
| 22:13.55 | ZX81 | hmmm, only pri commands are set, show, intense etc |
| 22:13.59 | ZX81 | gimme a sec |
| 22:14.16 | ZX81 | ah |
| 22:14.20 | ZX81 | drop intense |
| 22:14.54 | ZX81 | pri debug span 1 |
| 22:15.24 | ZX81 | yeah |
| 22:15.28 | ZX81 | so it doesn't send release |
| 22:15.35 | ZX81 | sec I'll pastebin it |
| 22:16.15 | ZX81 | http://pastebin.ca/1075995 |
| 22:16.27 | ZX81 | so it goes into an echo test |
| 22:16.34 | ZX81 | I hang up during message |
| 22:16.39 | ZX81 | pri gets the message |
| 22:16.43 | ZX81 | asterisk continues |
| 22:16.49 | ZX81 | one interesting thing |
| 22:16.57 | ZX81 | Progress Description: Inband information or appropriate pattern now available. |
| 22:17.06 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 22:17.23 | ZX81 | but again, < Message type: DISCONNECT (69) followed by Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) |
| 22:18.42 | ZX81 | I put line breaks to show time spaces :) |
| 22:19.20 | ZX81 | I thought I saw something somewhere recently |
| 22:19.28 | ZX81 | regarding pri not hanging up |
| 22:19.35 | ZX81 | but can't find it in mlist |
| 22:19.40 | ZX81 | checking insects :) |
| 22:20.14 | ZX81 | :) |
| 22:20.14 | ZX81 | Q.931 not release properly |
| 22:20.21 | cesar_CR | hello guys dummy question ? |
| 22:20.21 | ZX81 | http://bugs.digium.com/view.php?id=12587 |
| 22:20.22 | outtolunc | zx81 just lookup inbandrelease |
| 22:20.28 | ZX81 | checking it now |
| 22:20.40 | ZX81 | kk ty |
| 22:20.41 | cesar_CR | how can I have colors on CLI ?? |
| 22:20.46 | ZX81 | -c |
| 22:21.12 | cesar_CR | or colors are only for 1.6 ? |
| 22:21.14 | x86 | _ShrikE: you around? |
| 22:21.32 | ZX81 | cesar_CR: -v |
| 22:21.33 | ZX81 | *-c |
| 22:21.37 | ZX81 | hmm |
| 22:21.37 | x86 | how do you set an Adit 600 to use internal timing instead of receiving timing from a T1 |
| 22:21.49 | ZX81 | x86: see zaptel.conf |
| 22:22.06 | cesar_CR | ZX81, "asterisk -v" ?? |
| 22:22.11 | ZX81 | Your search - inbandrelease - did not match any documents. |
| 22:22.18 | ZX81 | cesar_CR nah -c |
| 22:22.20 | ZX81 | but |
| 22:22.24 | ZX81 | its a weird option |
| 22:22.27 | ZX81 | it seems to be console |
| 22:22.30 | ZX81 | if without -r |
| 22:22.33 | ZX81 | best bet |
| 22:22.40 | x86 | ZX81: um, note how i said "adit 600" not "asterisk", but that's cool I figured it out heh |
| 22:22.40 | ZX81 | is to edit asterisk.conf |
| 22:22.49 | ZX81 | x86 oh :) |
| 22:22.54 | x86 | -c is console |
| 22:22.54 | cesar_CR | ok thanks |
| 22:22.55 | x86 | err |
| 22:22.56 | x86 | color |
| 22:22.56 | ZX81 | dip switch |
| 22:23.05 | ZX81 | :) |
| 22:23.06 | x86 | ZX81: no, telnet ;) |
| 22:23.09 | ZX81 | ah |
| 22:23.12 | ZX81 | :) |
| 22:23.25 | x86 | -c == color |
| 22:23.41 | ZX81 | yeah -c is a bit weird |
| 22:23.54 | ZX81 | asterisk -h reports it as console I think |
| 22:24.04 | ZX81 | <PROTECTED> |
| 22:24.51 | ZX81 | outtolunc: maybe typo? |
| 22:28.21 | gaetronik | chao |
| 22:30.27 | ZX81 | :D |
| 22:30.28 | ZX81 | sweeet |
| 22:30.33 | ZX81 | upgrade of libpri fixed it |
| 22:30.34 | ZX81 | :D |
| 22:30.41 | Idle | fixed what? |
| 22:30.49 | Idle | should pay attention :( |
| 22:32.07 | ZX81 | :) not releasing |
| 22:32.11 | ZX81 | the channel |
| 22:32.13 | Idle | ah, fun |
| 22:32.15 | ZX81 | :D |
| 22:32.16 | ZX81 | yeah |
| 22:32.19 | MikeJ | good stuff |
| 22:32.25 | Idle | at least yours makes the call ;) |
| 22:32.28 | ZX81 | :D |
| 22:32.30 | ZX81 | Of course |
| 22:32.32 | ZX81 | except |
| 22:32.40 | ZX81 | every call plays a message from the telco |
| 22:32.45 | cesar_CR | ZX81, did not work -c |
| 22:32.48 | ZX81 | because the telco screwed up the install |
| 22:32.49 | ZX81 | :) |
| 22:33.02 | ZX81 | cesar_CR: asterisk -rc or asterisk -c? |
| 22:33.05 | Idle | pft, you would expect any better from a telco? |
| 22:33.10 | cesar_CR | when asterisk is running I mean |
| 22:33.10 | ZX81 | lol true |
| 22:33.14 | ZX81 | yeah |
| 22:33.18 | cesar_CR | already running |
| 22:33.24 | ZX81 | if you do a reload |
| 22:33.27 | ZX81 | you get no colour? |
| 22:33.45 | ZX81 | like |
| 22:33.46 | ZX81 | [Jul 19 10:33:39] NOTICE[9973]: chan_iax2.c:8862 __iax2_poke_noanswer: Peer '8666' is now UNREACHABLE! Time: 68 |
| 22:33.52 | ZX81 | with notice in yellow |
| 22:34.00 | cesar_CR | nop, could it be the start-up script ? |
| 22:34.00 | Idle | ~centos52bug |
| 22:34.01 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 22:34.15 | ZX81 | :) |
| 22:34.31 | ZX81 | cesar_CR: try stop now |
| 22:34.33 | ZX81 | then |
| 22:34.35 | ZX81 | safe_asterisk |
| 22:34.37 | ZX81 | then |
| 22:34.40 | ZX81 | asterisk -r |
| 22:34.42 | ZX81 | then |
| 22:34.43 | ZX81 | reload |
| 22:34.44 | ZX81 | and see |
| 22:34.48 | cesar_CR | ok |
| 22:36.00 | ZX81 | grrr my fop says "1 channel in use" on chan 1, and correct info on all the others, and yet the conf is the same for all :) |
| 22:37.49 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 22:38.22 | cesar_CR | ZX81, I only have color when doing "asterisk -c", but when asterisk is started by /etc/init.d/asterisk start I have no colors |
| 22:38.50 | ZX81 | instead of init.d |
| 22:38.53 | ZX81 | try safe_asterisk |
| 22:39.04 | ZX81 | or |
| 22:39.09 | ZX81 | edit /etc/init.d/asterisk |
| 22:39.14 | ZX81 | so it does -c |
| 22:39.37 | cesar_CR | ok I'll try the second one |
| 22:45.05 | xiando | ~centos52bug |
| 22:45.06 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 22:47.22 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:15.08 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 23:18.07 | Idle | YAY! it works |
| 23:18.11 | Idle | <3 digium support |
| 23:18.58 | unpaidbill | hoorah |
| 23:19.21 | Idle | gonna update the firmware tho |
| 23:19.33 | Idle | and nuke some devices on the PCI express bus |
| 23:23.46 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:25.17 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 23:26.09 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 23:27.10 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 23:27.24 | joobie | hey boys.. are there asterisk rpm's for redhat? |
| 23:27.53 | joobie | http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 23:27.54 | joobie | found it :P |
| 23:28.15 | joobie | ahh |
| 23:28.17 | joobie | what about centos5? |
| 23:28.29 | jaytee | what about it? you can download it |
| 23:28.45 | angryuser | ? |
| 23:29.04 | angryuser | it was a question ? |
| 23:31.12 | joobie | hmm |
| 23:31.15 | joobie | what is freepbx? |
| 23:31.21 | joobie | i see reference to freepbx and asterisk... |
| 23:31.25 | unpaidbill | cracked out asterisk |
| 23:31.36 | jaytee | it's pain, lots of pain and sometimes blood |
| 23:31.41 | unpaidbill | as in someone smoked a bunch of crack and made it |
| 23:31.41 | joobie | hehe |
| 23:31.51 | joobie | so it's like asterisk on steroids?:P |
| 23:31.54 | MikeJ | blah.. |
| 23:32.01 | MikeJ | its a gui frontend to asterisk |
| 23:32.06 | joobie | ahh |
| 23:32.08 | joobie | thanks Mike |
| 23:32.12 | MikeJ | ignore everyone elses fud |
| 23:32.19 | joobie | cheers :P |
| 23:32.24 | jaytee | imagine Asterisk being a full pardon from a life sentence, freepbx is just being out on furlough for a day and then right back in the slammer. |
| 23:32.25 | unpaidbill | i thought it was that offshoot of asterisk |
| 23:32.42 | MikeJ | you were wrong :D |
| 23:32.43 | unpaidbill | with all the xml configurations and whatnot |
| 23:32.47 | unpaidbill | wtf am i thinking of |
| 23:33.00 | unpaidbill | oh, freeswitch |
| 23:33.03 | jaytee | nope, it's Asterisk as a base with a mysql database and all kinds of custom scripts glommed on top of * |
| 23:33.12 | joobie | what is zaptel? |
| 23:33.20 | jaytee | ~book |
| 23:33.21 | 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 |
| 23:33.27 | MikeJ | zaptel are drivers for the digium (and some other) tdm cards |
| 23:33.34 | joobie | i see i need, asterisk-1.4.4.tar.gz , zaptel-1.4.2.1.tar.gz , libpri-1.4.0.tar.gz , asterisk-addons-1.4.1.tar.gz |
| 23:33.36 | angryuser | ~zaptel |
| 23:33.37 | jbot | methinks zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company |
| 23:33.53 | joobie | ahh |
| 23:33.59 | joobie | not needed in a sip install i guess |
| 23:34.01 | joobie | thanks |
| 23:34.06 | unpaidbill | dahdi now! |
| 23:34.28 | jaytee | not needed for SIP only but you will need it in 1.4 for timing if you use MeetMe conferencing. |
| 23:34.45 | MikeJ | so unpaidbill the guys who wrote freeswitch are crackheads? |
| 23:34.49 | angryuser | joobie install it anyway |
| 23:34.51 | joobie | thanks jay |
| 23:34.51 | jaytee | it has a "dummy" driver for use as a timer |
| 23:35.09 | jaytee | yw |
| 23:35.17 | unpaidbill | mikej, yes |
| 23:35.18 | joobie | is there an article that is specific to setup where i have my asterisk setup connected to the sip provider.. and the voip phones connected to my asterisk box? |
| 23:35.30 | joobie | i dont have any of these hardware cards, it's just rj45 poe int he lan and sip to the asterisk box |
| 23:35.35 | MikeJ | oh.. do you know any of them? |
| 23:35.37 | joobie | so many howtos i find are more complex.. |
| 23:36.04 | unpaidbill | i know them all personally |
| 23:36.07 | unpaidbill | i used to sell crack |
| 23:36.19 | jaytee | joobie, the voip-wiki entry for * and CentOS will give you a solid install |
| 23:36.25 | MikeJ | or anything about the software? |
| 23:36.33 | joobie | jay, http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos this one? |
| 23:36.54 | jaytee | that and the book you can download will answer 89% of your questions at least. |
| 23:37.03 | jaytee | yes that one |
| 23:37.06 | joobie | okie |
| 23:37.08 | joobie | cheeres |
| 23:37.12 | jaytee | and the downloadable pdf of the book |
| 23:37.17 | jaytee | ~book |
| 23:37.18 | 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 |
| 23:37.56 | joobie | k thanks |
| 23:38.00 | joobie | i have that bookmarked |
| 23:38.03 | joobie | one more Q .. :P |
| 23:38.09 | joobie | v 1.4 or 1.6? |
| 23:38.14 | joobie | the ftp looks like 1.6 is beta |
| 23:38.16 | jaytee | joobie, I'd advise reading as much of the book as possible BEFORE you install and make sure you thoroughly read Chapter 5 and 6. |
| 23:38.19 | joobie | asterisk-1.6-current.tar.gz |
| 23:38.21 | joobie | but then i see that |
| 23:38.23 | jaytee | 1.6 is beta |
| 23:38.29 | joobie | kk |
| 23:38.32 | joobie | cheers |
| 23:38.36 | joobie | im up to like ch3 in the book |
| 23:38.42 | joobie | but i need to get this install done this weekend |
| 23:38.46 | joobie | so pressed for time |
| 23:39.02 | jaytee | if you rush you'll overlook things and end up with a mangled dialplan |
| 23:39.13 | joobie | can it be cleaned up once it's mangled? |
| 23:39.31 | jaytee | sure, but it might end up costing you more time than it's worth. |
| 23:39.40 | joobie | ahh k |
| 23:39.49 | joobie | if i read ch5 and 6.. think that would be enough ? |
| 23:39.52 | joobie | .. i mean, as a minimum |
| 23:40.48 | joobie | thanks jay |
| 23:40.52 | joobie | ill skim through the book whilst installing |
| 23:40.57 | joobie | cheers for the help |
| 23:41.12 | angryuser | my oracle book is depressing 1818 pages ...... |
| 23:42.17 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 23:42.30 | angryuser | i like the begginning of chapter 3 "You must find the force to do not skip this chapter, it is important" |
| 23:42.32 | jaytee | my O'Reilly MySql in A Nutshell is depressing but it's only 545 pages |
| 23:42.34 | joobie | what cha trien to do in oracle angry? |
| 23:42.44 | MikeJ | use the force? |
| 23:42.51 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 23:43.06 | angryuser | yes the dark side |
| 23:43.26 | jaytee | I thought the dark side was MSSQL 2005? |
| 23:44.08 | angryuser | or should i say "courage", my translation is far from perfect |
| 23:44.35 | jaytee | although it's widely rumored that Larry Ellison is a giant dickwad and puffs the pickle in public restrooms. |
| 23:45.00 | *** part/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 23:46.31 | angryuser | never used mssql 2005, hopefully newer will |
| 23:46.57 | jaytee | once you do, the rectal itch will never go away :-( |
| 23:47.54 | angryuser | that's a chance to change the job |
| 23:48.43 | angryuser | jaytee bad experience with that one ? |
| 23:49.02 | jets | rumor has it larry ellison and larry craig are friends |
| 23:49.22 | bkw_ | omg its jets |
| 23:49.40 | bkw_ | jets: ltns |
| 23:50.06 | jets | hey whats up! |
| 23:50.10 | jaytee | let's just say that from one version to the next the migration wasn't all that smooth |
| 23:50.18 | bkw_ | jets: just workin on goodies |
| 23:50.30 | jets | ya i know how that is |
| 23:50.38 | _khan | how can i dial number from a specific zap channel??? |
| 23:50.40 | jaytee | anyway, I gotta run. Watching a friend's autistic son tonight and playing guitar with him. 13 and a prodigy at music. |
| 23:50.50 | bkw_ | _khan: to a specific channel? |
| 23:50.53 | jaytee | later all |
| 23:50.58 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:51.19 | _khan | yes, if i have 30 channels, i want to dial a number from channel 25 |
| 23:51.22 | VIPCarrier | I updated freePBX and I start getting this massage in CDR page "YOU MUST ACCESS THE CDR THROUGH THE ASTERISK MANAGEMENT PORTAL!" |
| 23:51.33 | angryuser | _khan Dial(Zap/N) where N is number declared in zapata.conf |
| 23:51.37 | VIPCarrier | any one have any ideas? |
| 23:51.59 | angryuser | ~freepbx |
| 23:51.59 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:52.38 | joobie | freepbx sounds like a cop out to learning * |
| 23:53.19 | jblack | In all fairness, I don't want to learn how to rebuild an engine in order to drive a car. |
| 23:53.55 | bkw_ | joobie: its really not. Not everyone wants or needs to learn every detail of Asterisk. |
| 23:54.08 | MikeJ | joobie: no.. it's just a different approach |
| 23:54.26 | MikeJ | lots of people have no need to "learn asterisk" ... they just want something to work.. |
| 23:54.26 | jblack | An in more fairness, one shouldn't go to the auto-shop bar to ask how to get jiffie lube to do an oil change. |
| 23:54.46 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 23:54.46 | MikeJ | meh.. |
| 23:54.51 | bkw_ | haha |
| 23:55.02 | _khan | is it possible to listen dial tone before dialing a number from Zap channel (analogue line) |
| 23:55.12 | bkw_ | _khan: just dial Zap/25 |
| 23:55.15 | bkw_ | no number |
| 23:55.19 | bkw_ | it should give you dialtone |
| 23:55.28 | _khan | ok |
| 23:56.11 | angryuser | i think if you want to learn asterisk, start by understanding config files, when thing go wrong with 'gui' you have no way to find error fast |
| 23:57.38 | _khan | bkw_: how can i dial zap/25 from sip phone? or i need to dial on console?? |
| 23:58.42 | angryuser | _khan have you read a book about writing dialplans ? |
| 23:58.48 | angryuser | ~book |
| 23:58.49 | 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 |
| 23:58.54 | MatBoy | mhh that IVR MySQL thing seems to be more difficult than it's displayed |
| 23:59.13 | MikeJ | angryuser: don't be mad.. it's all going to be ok |
| 23:59.23 | Qwell | MatBoy: look at func_odbc |
| 23:59.24 | MikeJ | :D |
| 23:59.34 | MikeJ | ok.. off .. ttfn |
