| 00:00.15 | beek | primesoft: Run dahdi_genconf once, then never again. The only way I could get the parameters I needed was to hand-edit the file. |
| 00:01.43 | primesoft | beek: Ahhhh!.... now we are getting somewhere...so it doesn't run as part of the boot process and if it does it should be disabled? |
| 00:02.30 | beek | primesoft: yes. I was put off by that message for quite some time. It appears to me to be a red herring. |
| 00:02.52 | citywok | yay i've been segfault free using chanspy+monitor in asterisk 1.6, even though it says it shouldn't have been happening in 1.4.21 |
| 00:03.51 | Micc | why do I keep seeing this? sched.c: Scheduled event in 0 ms? |
| 00:05.43 | *** join/#asterisk coppice (n=chatzill@201.162.17.210.dyn.pacific.net.hk) |
| 00:05.48 | primesoft | beek: Any suggestion of where this little work around should be posted to ensure our fellow head scratchers speed up their troubleshooting process? |
| 00:06.39 | *** part/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 00:07.05 | beek | primesoft: Not really... I just happened to look up, saw your message and had a sense of deja vu |
| 00:08.16 | beek | primesoft: Perhaps there is an official way that this is supposed to work and my suggestion short circuits it. If that's the case I'd love to know the real story. From what I can tell though, dahdi_conf is useful to discover your cards and create a basic configuration file for them. |
| 00:08.53 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 00:12.47 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 00:12.52 | primesoft | beek: Thank goodness for that. Yeh more than likely I'll make a dick of myself if I post it somewhere with someone countering it with the correct way to do it. It could genuinely be another example of because it works in the US its not counted as a bug. :-) |
| 00:13.48 | bombaclat667 | I configured a trunk via asterisk-gui(my x100p is detected), but I get an unable to create channel of type 'Zap' (Cause 0 -Unknowm). I also noticed I do not have a zaptel.conf file in /etc/asterisk (but I do have a etc/zaptel.conf file) |
| 00:14.06 | bombaclat667 | is the zaptel.conf deprected or should I create it? |
| 00:14.31 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 00:14.52 | ManxPower | bombaclat667: /etc/zaptel.conf = configure hardware /etc/asterisk/zapata.conf = set up asterisk to use the hardware |
| 00:16.32 | bombaclat667 | ok..zapata and not zaptel |
| 00:18.31 | *** join/#asterisk javb (n=javb@tdev213-76.codetel.net.do) |
| 00:18.42 | javb | hi, im getting this error http://threebit.net/mail-archive/asterisk-users/msg29145.html |
| 00:18.56 | javb | Calls just last 10 seconds while using the sip trunking |
| 00:18.58 | javb | any idea? |
| 00:20.20 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 00:22.46 | ManxPower | javb: that means "hey! the remote side is not responding!" |
| 00:23.24 | ManxPower | maybe you have asterisk behind nat and don't have it configured for nat or you have the client behind nat or you don't have canreinvite=no (required in NAT setups) |
| 00:24.33 | ManxPower | So this company calls up, wanting our consulting services. They fired the consultant that built the MASSIVE conferencing solution for then, but they had not docs. |
| 00:24.55 | ManxPower | Turns out there is a patent filed for the service and that has very useful docs. |
| 00:25.06 | ManxPower | (MASSIVE = 2000+ people in one conference) |
| 00:25.08 | javb | Problem solved, qualify! |
| 00:25.17 | ManxPower | javb: you only think that solved it. |
| 00:25.38 | javb | ManxPower, what do you mean? |
| 00:25.57 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 00:26.07 | ManxPower | javb: qualify has it's own issues |
| 00:26.50 | ManxPower | IMNSHO, setting a low registration time on the client is better than qualify=, 60 seconds is a good one |
| 00:28.21 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 00:30.04 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 00:33.02 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 00:38.19 | *** join/#asterisk eric2 (n=nobody@69.60.247.142) |
| 00:43.14 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 00:43.50 | bombaclat667 | is zapata.conf still neeed when running on 1.4.23.1 with trunks? |
| 00:47.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:49.25 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 00:50.05 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 00:50.06 | NovceGuru | ponders a battery backup, "wireless gaming adapter" and a polycom soundstation |
| 00:50.22 | NovceGuru | stupid concrete floors and lack of pre-thought out conduit |
| 00:51.48 | jaytee | I once installed a channel bank in a bathroom at a retail office |
| 00:52.52 | NovceGuru | wooh! |
| 00:53.23 | eppigy | did it dispense novelty condoms |
| 00:56.02 | jaytee | no, but I think the blinking light factor was soothing to people using the facilities :-) |
| 00:56.58 | *** join/#asterisk dalbaech (n=dalbaech@knight.darkvoip.net) |
| 00:58.22 | jaytee | it was a T1 for voice and data at a retail store in a small town in Maine and I had to have it up in less than 4 hours. The smartjack was installed in a "closet" 6" deep and I had no extra cable to extend from the smartjack to any other location than the wall behind the closet which was in the bathroom. |
| 00:59.20 | jaytee | I also spent half a day searching for a T1 smartjack in a Vermont town only to find that it had been installed in the laundromat next door. |
| 00:59.23 | beek | evening jaytee |
| 00:59.29 | jaytee | evening beek |
| 01:00.30 | beek | jaytee: Finally fully implemented the Asterisk system between the PSTN and our legacy PBX. It's running really well. |
| 01:00.50 | beek | New PRIs are in and will be turned up next Tuesday evening. |
| 01:00.58 | jaytee | cool! |
| 01:01.22 | beek | I drove by the building which houses our new provider. I can now choke the living shit out of a physical person if I need service and they're not providing it. |
| 01:01.52 | jaytee | I'll be migrating the last of the Nortel lines over to Asterisk by the end of the month. Pulling the plug on the Option 11C at the end of the first week in March. |
| 01:02.10 | beek | jaytee: So you'll be strictly Asterisk then? |
| 01:02.27 | jaytee | being able to choke the living shit out of a provider would be a great reassurance I think |
| 01:02.45 | jaytee | yeah, Asterisk, sipX and Exchange 2007 Unified Messaging. |
| 01:02.59 | beek | jaytee: Except for the latter of the three that sounds great. |
| 01:03.09 | mchou | jaytee: sipX? |
| 01:03.10 | jaytee | Exchange UM is pretty nice |
| 01:03.21 | mchou | jaytee: is sipX any good? |
| 01:03.23 | jaytee | mchou, yes, sipX as a proxy |
| 01:03.30 | beek | jaytee: Perhaps, but I'll avoid putting anything Microsoft in my shop, thank you. |
| 01:03.49 | jaytee | we only use it for udp to tcp proxy to Exchange since Exchange only speaks SIP TCP |
| 01:04.16 | mchou | jaytee: cause I've been eying that but there are only 5 or so ppl on the sipX irc channel |
| 01:04.20 | jaytee | beek, you're lucky you have that option. I don't have choices, I just have directives. |
| 01:04.31 | beek | jaytee: Any idea of the savings you'll be seeing due to this new setup? |
| 01:05.52 | jaytee | mchou, it's not heavily supported for the open source version. Nortel bought Pingtel that makes the commercial version and they even contributed code back to the open source version to improve it. It has some nice features but it only talks SIP so if you need to interface with a PRI you need to use a SIP/PRI gateway device like Audiocodes, etc. |
| 01:06.21 | mchou | jaytee: I happen to have a pingtel hard phone.... |
| 01:06.43 | mchou | jaytee: so I am aware of theier (checkered?) history :) |
| 01:06.47 | jaytee | beek, well in terms of migration costs we've spent less than a third of what we paid for the Nortel pbx plus phones and licensing and we'll also save 18K a year in support contract costs. |
| 01:06.58 | beek | jaytee: Now that is impressive. |
| 01:07.58 | mchou | jaytee: what upsets me is that sipfoundry just randomly "disappears" code..... |
| 01:08.27 | jaytee | and that gives us two servers, one as a "hot" spare. Eventually I'd like to sell them on buying a T1 failover box so I can cable both of them and use linux HA with shell scripts to change IP interfaces and bring the "hot spare" online while taking the active server offline if there's a problem. |
| 01:08.36 | *** join/#asterisk TommyTheKid (n=tommy@97-118-14-90.hlrn.qwest.net) |
| 01:08.36 | mchou | jaytee: that's what I'm very leery of |
| 01:08.46 | beek | jaytee: I've been mapping out our exit strategy for our legacy PBX... I'd like to see it go away within two years, saving us $8K/year in support costs. |
| 01:09.08 | TommyTheKid | what does it mean if a DAHDI line proceeds with progress status = 31 |
| 01:09.58 | TommyTheKid | ... its a PRI if that matters |
| 01:10.08 | *** join/#asterisk jupeterson (n=John@c-24-126-160-141.hsd1.ga.comcast.net) |
| 01:10.16 | jaytee | beek, I expected this migration to take longer and wanted it to in order to do it right but the micromanaging pinheads got very agressive with the schedule. |
| 01:11.04 | jaytee | I told them, Good, Fast and Cheap, pick any two. Now they want all three and I've flat out told them, Nope! You get Fast and Cheap since you insist on keeping to this schedule. No guarantees, mofo. |
| 01:11.45 | carrar | MAKE IT HAPPEN |
| 01:11.51 | beek | jaytee: Is this the thing you were struggling to get spanish TTS for? |
| 01:11.57 | *** join/#asterisk highlevelcode (n=highleve@adsl-75-51-89-110.dsl.lsan03.sbcglobal.net) |
| 01:12.36 | jaytee | no, I meant the entire migration project including the speech enable IVR. |
| 01:13.00 | jaytee | I absolutely hate coding in a rushed fashion. |
| 01:13.14 | jupeterson | does anyone know the default mailbox size in asterisk for all extensions.. size of entire mailbox system, and can it be configured? |
| 01:13.17 | beek | jaytee: That's where the mistakes come from. Take this example... |
| 01:13.21 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 01:13.51 | beek | jaytee: I coded a dialplan to replace our existing voice mail server. |
| 01:14.19 | beek | jaytee: I had a failsafe in case the phone system sent DTMF that I couldn't identify and during testing |
| 01:14.47 | beek | I needed a relatively long prompt, so I used the included "spam" message. I was rushed to get it done so that I could |
| 01:15.10 | beek | move to another project. Yesterday, a customer called and the phone system sent an unexpected code. |
| 01:15.13 | beek | Need I say more? |
| 01:15.32 | jaytee | sounds about par for the course |
| 01:15.44 | beek | Fortunately she had a good sense of humor. |
| 01:15.55 | beek | You famliar with that prompt? |
| 01:17.58 | beek | Oh well... time to head home. Have a good night jaytee. |
| 01:18.10 | jaytee | nite beek |
| 01:24.30 | jaytee | TommyTheKid, here's a link that may be of help: http://lists.digium.com/pipermail/asterisk-users/2007-March/181488.html |
| 01:25.16 | jaytee | your code 31 is Normal, Unspecified. It's common and means the system doesn't know what to do with the call. |
| 01:27.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-dc3224652628e8c9) |
| 01:29.42 | *** join/#asterisk ingenius (n=alektro@host27.190-224-187.telecom.net.ar) |
| 01:35.39 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:35.39 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:37.23 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 01:38.19 | kb3ien | my polycom phones will only try to register at boot-time, is there a setting i can use to avoid powercycling them every time they loose connectivity to the ast 1.4 box? |
| 01:42.29 | wonderworld | hey. i have got problems with ChanSpy() it always exits after the end of the second spied on call. i attach it to a specific SIP channel..eg. ChanSpy(SIP/100). |
| 01:43.01 | wonderworld | asterisk just hangs up on me instead of waiting on the channel for the next call |
| 01:43.56 | dalbaech | kb3ien: Change the SIP Expiration rate to "60" seconds |
| 01:44.01 | dalbaech | http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+7#722Polycom |
| 01:44.11 | dalbaech | kb3ien: search / find that string I gave you it might help |
| 01:44.18 | dalbaech | I don't use polycom, meh. |
| 01:44.42 | dalbaech | wonderworld: don't add the channel |
| 01:45.19 | dalbaech | wonderworld: when the channel isn't there, there's nothing to spy on. |
| 01:47.40 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 01:47.48 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 01:49.21 | wonderworld | dalbaech: ChanSpy waits for a call on the channel. when i start it and there is no call on the spied on channel, the app just waits and connects the moment, a call is started... |
| 01:50.04 | wonderworld | it works fine, but only twice.. |
| 01:50.21 | dalbaech | is the prefix of the channels "SIP/100***" by chance? |
| 01:52.35 | wonderworld | dalbaech: sorry, not sure what you mean... |
| 01:52.58 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:56.40 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:56.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:59.31 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 02:01.29 | *** join/#asterisk gones (n=gones@116.24.223.108) |
| 02:03.46 | eric2 | murdock_ut got blf working :) |
| 02:04.28 | eric2 | and now I got more game than coleco |
| 02:06.26 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:19.19 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:19.41 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.133) |
| 02:23.57 | thehar | ohhhh queue stats you are difficult. can't ever please anyone. |
| 02:31.28 | [T]ank | i am on asterisk 1.4.23.1 and I am having trouble dialing out: http://pastebin.ca/1333439 |
| 02:31.32 | [T]ank | can anyone help? |
| 02:33.17 | eric2 | try Dial(Zap/g0/${EXTEN}) |
| 02:33.45 | *** join/#asterisk tobias (n=tobias@cpe-071-070-218-098.nc.res.rr.com) |
| 02:34.26 | [T]ank | same result |
| 02:34.55 | wonderworld | the problem with /dev/dsp means, that your local sound hardware can't be accessed |
| 02:35.00 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:36.14 | [T]ank | Im asking about the chanunavailable issue. |
| 02:37.01 | [T]ank | The t1s work... I have them working in a server running asterisk 1.4.11 |
| 02:37.24 | wonderworld | what does "zap show channels" say? |
| 02:38.03 | [T]ank | I have built the replacement for that server... I copied the configs over directly and moved the t1s. Then restarted asterisk. |
| 02:38.35 | [T]ank | pastebin.ca/1333453 |
| 02:40.08 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:40.25 | [T]ank | im guessing that I should see all 96 channels there. |
| 02:40.30 | wonderworld | yes right |
| 02:40.36 | [T]ank | so, what have i screwed up? |
| 02:41.07 | wonderworld | what hardware dou you have? |
| 02:41.12 | wonderworld | a digium 4-span card? |
| 02:41.43 | [T]ank | yep |
| 02:41.49 | [T]ank | 420p |
| 02:42.36 | wonderworld | ok, you should find out first, if zap is working at all |
| 02:42.48 | wonderworld | there was a tool to check the status of the hardware card |
| 02:43.03 | wonderworld | are the LED's green on the card for your lines? |
| 02:43.07 | [T]ank | yep |
| 02:43.15 | wonderworld | ok |
| 02:43.16 | [T]ank | zap show status shows all ports as "ok" |
| 02:43.28 | wonderworld | i am usind dahdi, so i forgot the name of the zap-tool |
| 02:43.30 | wonderworld | just a second |
| 02:43.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 02:43.53 | [T]ank | i only changed to zap because I couldnt figure out the differences in dahi |
| 02:43.58 | [T]ank | dahdi |
| 02:44.04 | *** join/#asterisk dalbaech (n=dalbaech@knight.darkvoip.net) |
| 02:44.12 | dalbaech | hey guys |
| 02:44.15 | dalbaech | [TK]D-Fender: you around? |
| 02:44.42 | wonderworld | hey |
| 02:44.45 | dalbaech | heya wonderworld |
| 02:44.48 | dalbaech | sorry about earlier |
| 02:44.57 | dalbaech | my box crashed when I was compiling |
| 02:44.57 | dalbaech | heh |
| 02:45.00 | wonderworld | Tank: it *should* look like that: http://pastebin.ca/1333463 |
| 02:45.04 | wonderworld | dalbaech: np |
| 02:45.56 | [T]ank | wonderworld: yeah, the server I copied and moved the t1s from is showing that correctly |
| 02:46.42 | *** join/#asterisk joako (n=joako@99-153-162-33.lightspeed.miamfl.sbcglobal.net) |
| 02:46.52 | wonderworld | try the linux console |
| 02:47.00 | wonderworld | type zap and tab tab |
| 02:47.06 | [T]ank | nadda |
| 02:47.06 | dalbaech | I upgraded to asterisk1.6, now meetme doesn't work... I have the dahdi_dummy module loaded and it's new |
| 02:47.34 | dalbaech | anything you think I'm missing? |
| 02:47.37 | dalbaech | http://xipv6.net/34 |
| 02:47.44 | wonderworld | i think it was zttool |
| 02:47.46 | wonderworld | try that |
| 02:48.03 | wonderworld | should give you the status of the card and the ability to see the available channels |
| 02:48.11 | [T]ank | wonderworld: how about dahdi_tool |
| 02:48.36 | wonderworld | maybe try dahdi show status in the asterisk cli too |
| 02:50.05 | wonderworld | yes, i think dahdi_tool replaced zttool |
| 02:50.12 | wonderworld | mine says: Span 1: 31 total channels, 31 configured |
| 02:50.23 | [T]ank | pastebin.ca/1333466 |
| 02:50.40 | [T]ank | mine just took me to the next blank line. no output |
| 02:51.07 | wonderworld | hmm |
| 02:51.12 | wonderworld | dahdi show channels ? |
| 02:51.48 | [T]ank | dame |
| 02:51.49 | *** join/#asterisk Brijn (n=Freax@S01060016b6848f75.vn.shawcable.net) |
| 02:51.51 | [T]ank | im missing something... |
| 02:52.03 | Brijn | Hello all |
| 02:52.12 | wonderworld | hi Brijn |
| 02:52.17 | [T]ank | it used to be if you ran ztcfg it would configure your spans.... then you restarted asterisk |
| 02:53.25 | [T]ank | wonderworld: pastebin.ca/133471 |
| 02:53.42 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 02:53.44 | Brijn | Somebody used my * box to make several thousands calls.. I never paid much attention to security, not expecting it to be an interesting/likely target.. I was clearly wrong.. Want to make sure I learn how to best protect my * box (at the application level, I'll also put proper fw rules in place). Somebody willing to provide me with a few pointers |
| 02:54.38 | [T]ank | wonderworld: dahdi_cfg does nothing |
| 02:54.50 | [T]ank | i would have expected that to configure and load each of my spans |
| 02:55.08 | Brijn | I'm guessing that somebody brute forced my SIP phone user and pwd (they where not very complicated). Is that a common "hack"? |
| 02:55.09 | wonderworld | you kast paste in pastebin was a mirc-script |
| 02:55.24 | [TK]D-Fender | Brijn: make sure all calls have to be auth'd, NEVER EVER use contexts named "default", make sure your contexts don't included extens they shouldn't be able to dial, etc |
| 02:56.16 | [TK]D-Fender | Brijn: It could simply be because you allowed un-auth'd calls and set your contexts poorly |
| 02:56.18 | [T]ank | wonderworld: so that was not helpful? |
| 02:57.08 | Brijn | [TK]D-Fender: In the cdr they all showed with Src 300 (300 being my SIP deskphone), does that match un-auth'ed calls. What keywords to google for in regards to unauth'ed calls? |
| 02:57.44 | [TK]D-Fender | Brijn: paste 2-3 lines of CDR for this |
| 02:58.05 | [TK]D-Fender | [T]ank: show us |
| 02:58.15 | dalbaech | heya [TK]D-Fender |
| 02:58.35 | wonderworld | tank: did you compile asterisk yourself on the server? |
| 02:58.47 | [T]ank | wonderworld: yes |
| 02:58.54 | Brijn | [TK]D-Fender: Would the output of this: select src,dst,start,answer,end,duration,billsec from cdr; be usefull. A full line makes a bit of a mess |
| 02:59.00 | [T]ank | [TK]D-Fender: let me grab the past pastebin links really quick |
| 02:59.51 | [TK]D-Fender | Brijn: I asked for 2-3 full lines |
| 03:00.08 | dalbaech | ~pastebin |
| 03:00.09 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:00.24 | [T]ank | pastebin.ca/1333439 pastebin.ca/1333453 pastebin.ca/1333466 pastebin.ca/133471 |
| 03:00.27 | dalbaech | wowza |
| 03:00.46 | Brijn | 10483|"RIJNIERSCE" <300>|300|15757466351|lspeed-out|SIP/300-0841e7c0|SIP/lightspeed-out-08459f60|Dial|SIP/15757466351@lightspeed-out|30|rwW|2009-02-10 09:21:48|2009-02-10 09:21:48|2009-02-10 09:21:59|11|11|8|3| |
| 03:00.48 | Brijn | 10484|"RIJNIERSCE" <300>|300|15757466381|lspeed-out|SIP/300-0841e7c0|SIP/lightspeed-out-0840fa00|Dial|SIP/15757466381@lightspeed-out|30|rwW|2009-02-10 09:22:22|2009-02-10 09:22:22|2009-02-10 09:22:33|11|11|8|3| |
| 03:01.04 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 03:01.34 | dalbaech | [TK]D-Fender: after the queue is over (haha, made a funny) can you look at mine too? hehe http://xipv6.net/34 |
| 03:02.00 | wonderworld | Tank: did you compile libpri and zaptel before asterisk? |
| 03:02.19 | [TK]D-Fender | Brijn: You did have a secret set for your peer, right? |
| 03:02.20 | [T]ank | well, i did the dahi |
| 03:02.24 | [T]ank | dahdi |
| 03:02.38 | [T]ank | let me redo it all and make sure its all in the right order. |
| 03:02.49 | [TK]D-Fender | [T]ank: FFS include the http:// next time so the links are clickable :p |
| 03:02.56 | [T]ank | sorry |
| 03:03.03 | Brijn | [TK]D-Fender: Yes, but it wasn't very hard. But somebody would have needed to guess user and secret |
| 03:03.21 | wonderworld | Tank: you should check on the right order, before doing it again |
| 03:03.29 | wonderworld | my order musn't be correct, but order matters |
| 03:04.12 | thehar | anyone heard of queue-ninja? |
| 03:04.26 | [TK]D-Fender | [T]ank: What are on your ports? |
| 03:05.03 | [TK]D-Fender | [T]ank: you running NFAS over 2 PRI? |
| 03:05.21 | [T]ank | nfas on 4 |
| 03:05.44 | [TK]D-Fender | [T]ank: Ah yes... I missed registering it was 96 and not 48 |
| 03:06.02 | [TK]D-Fender | [T]ank: [Feb 10 19:27:05] WARNING[4150]: chan_oss.c:687 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory <--- do a VALID test from a friggen phone that is an IMMEDIATE failure |
| 03:06.10 | [TK]D-Fender | :) |
| 03:06.31 | [T]ank | I have always been able to do it from console before. |
| 03:06.40 | [T]ank | is that changed in latest versions? |
| 03:07.04 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:07.11 | [TK]D-Fender | [T]ank: No, maybe * simply can't seize exclusive access. |
| 03:07.24 | [TK]D-Fender | [T]ank: test with a softphone or something... console is a SHIT test I never trust |
| 03:07.42 | [T]ank | k |
| 03:11.17 | Brijn | [TK]D-Fender: Is adding a deny/accept clause in the peer a valid way to prevent outside host from registering? |
| 03:11.36 | [TK]D-Fender | Brijn: Yup |
| 03:11.47 | dalbaech | [TK]D-Fender: got it... |
| 03:11.50 | dalbaech | :-D |
| 03:11.54 | dalbaech | , instead of | now |
| 03:11.55 | dalbaech | hehe |
| 03:12.06 | Brijn | [TK]D-Fender: Any other best practice things when it comes to security |
| 03:12.20 | Brijn | [TK]D-Fender: I'll rename my default contact as well |
| 03:13.29 | [TK]D-Fender | Brijn: use crptic peer names not [300]. So instead [agdjkgasdkjahsgd] |
| 03:13.41 | *** join/#asterisk etfonhomey (n=chatzill@74-131-85-108.dhcp.insightbb.com) |
| 03:14.28 | wonderworld | Brijn: are other services running on your server as well? |
| 03:14.54 | Brijn | wonderworld: No, it's a tiny via ITX running of flash |
| 03:15.06 | wonderworld | thats good :) |
| 03:15.14 | Brijn | wonderworld: With a firewall in front of it with only SIP being forwarded |
| 03:15.33 | wonderworld | are you the only person using this server? |
| 03:15.46 | Brijn | I'm really suprised somebody apparently wrote a scanner for SIP, and then bruteforcing it |
| 03:15.49 | Brijn | wonderworld: Yep |
| 03:16.22 | wonderworld | you could add an extra password prompt in front of your trunk |
| 03:16.36 | Brijn | Good thing I have unlimted NA calls.. But I hate to be the person that deliver SPAM calls (I guess that what it was) to other ppls phone |
| 03:16.41 | wonderworld | send an email if someone enters a wrong code and log the ip |
| 03:16.47 | Brijn | wonderworld: How do you mean? |
| 03:16.54 | Brijn | wonderworld: Where would I do that? |
| 03:17.22 | Brijn | I would expect the "attacker" to come back, and would like to log some info |
| 03:17.42 | wonderworld | http://www.voip-info.org/wiki-Asterisk+cmd+authenticate |
| 03:19.44 | Brijn | wonderworld: Ahh, I was thinking about something completely else.. Could do that, but rather not |
| 03:23.40 | [T]ank | wonderworld: same result after reinstall |
| 03:23.41 | [TK]D-Fender | wonderworld: overkill |
| 03:24.20 | [TK]D-Fender | [T]ank: Whats all this talk of reinstall? I have yet to see a valid test. |
| 03:25.20 | Brijn | [TK]D-Fender: Any suggestions on what logging to enable to get an IP on the remote end. Files in /var/log/asterisk don't seem to have that |
| 03:25.48 | [T]ank | [TK]D-Fender: was already into the reinstall when you suggested the softphone... working on that now. |
| 03:26.04 | [TK]D-Fender | Brijn: look at fail2ban |
| 03:26.46 | Brijn | [TK]D-Fender: Cool! TX |
| 03:28.33 | Brijn | [TK]D-Fender: Ahhh, yep, it was a sip brute force attach alright. Have the IP now, see what I can find |
| 03:29.12 | [TK]D-Fender | Brijn: allow/deny are good if you want to restrict to your local LAN |
| 03:29.21 | [TK]D-Fender | Brijn: VPn for remote use if plenty valid. |
| 03:29.22 | kb3ien | i set the timer to 60 seconds, all that did was enable to phone to notice the lack of connectivity in 60 seconds (or less) it didn't allow the phone to reregister. |
| 03:30.27 | Brijn | [TK]D-Fender: All SIP phones are local to the PBX. At work I have phone in other office, but inside VPN, so also known IP range |
| 03:31.25 | Brijn | [TK]D-Fender: For my understanding. I see a lot of registration failed, but then 300 is missing there. Can you register without the correct password. So asterisk basically "leaks" valid peer names, correct. Auth will only fail if they actually make a call? |
| 03:31.29 | kb3ien | any means to allow the polycom to re-register sans reboot? |
| 03:35.00 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 03:35.21 | [TK]D-Fender | Brijn: briThewy don't need to register to try a call direct |
| 03:35.35 | [T]ank | [TK]D-Fender: http://pastebin.ca/1333524 |
| 03:35.59 | [TK]D-Fender | kb3ien: I believe a call-attempt on it triggers a re-reg attempt if it does not believe its registered |
| 03:36.08 | Brijn | [TK]D-Fender: TX, learned some new things. Very useful!!! |
| 03:36.29 | Brijn | is firing of a few mails to abuse addresses |
| 03:36.43 | [TK]D-Fender | [T]ank: All the backup now please. 1 big pastebin |
| 03:37.11 | [T]ank | backup? you mean all the configs? |
| 03:38.20 | [TK]D-Fender | [T]ank: EVERYTHING related to zaptel setup |
| 03:38.33 | [T]ank | k - sec |
| 03:38.34 | [TK]D-Fender | [T]ank: CLI verifications (* and LINUX) |
| 03:38.35 | [TK]D-Fender | etc |
| 03:42.08 | kb3ien | it does, sortof, it never reregsters well enought to rececive calls. |
| 03:42.48 | [T]ank | [TK]D-Fender: http://pastebin.ca/1333529 |
| 03:43.04 | [T]ank | not sure what all to send. If there is something i should add let me know. |
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| 03:46.27 | *** mode/#asterisk [+o Mog] by ChanServ |
| 03:47.03 | [TK]D-Fender | [T]ank: Why the hell are you MIXING Zaptel & DAHDI? Its one or the other dammit! |
| 03:47.33 | [T]ank | ... what do you mean? |
| 03:47.36 | [T]ank | what have I done wrong? |
| 03:47.41 | [T]ank | I am so lost with this change. |
| 03:47.43 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 03:47.52 | [TK]D-Fender | [T]ank: Stop using ^%#$ Mecedes deisel engine parts in your friggen unleaded Ford Fiesta |
| 03:48.20 | [TK]D-Fender | [T]ank: Ditch Zap. It no longer exists. Period. No Zaptel. ANYWHERE |
| 03:48.35 | [T]ank | ok...... |
| 03:50.30 | *** join/#asterisk Kyosh (i=48595dad@gateway/web/ajax/mibbit.com/x-a6046ad4abba90ce) |
| 03:52.39 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 03:56.14 | *** join/#asterisk mosty (n=mosty@202.164.195.71) |
| 04:02.06 | [TK]D-Fender | [T]ank: BRB |
| 04:03.09 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 04:13.06 | kb3ien | Anyone using xlite? |
| 04:13.22 | [T]ank | yep |
| 04:13.37 | prg3 | Does anyone know what CID signalling Telus in Canada uses? |
| 04:14.28 | kb3ien | everytime i fireup xlite i get failed outbound calls, until i do a 'module reload'. THe failure is odd in that the calling numbers seems blank: Call from '' to extension '18005551212' |
| 04:15.10 | kb3ien | any good sample configs? |
| 04:16.15 | [TK]D-Fender | prg3: Standard bellcore |
| 04:17.47 | kb3ien | everything i have read about such failures always list the 'username' value from sip.conf in the "Call from" field. |
| 04:18.09 | Brijn | Hmmm, I changed the basic phone name to something hard: But now I get: username mismatch, have <telefoonboven>, digest has <T`> |
| 04:18.21 | Brijn | While the sip.conf has exactly trhe same string? |
| 04:18.25 | kb3ien | i need to do a module reload after starting xlite to get the calls out to work. |
| 04:19.28 | prg3 | [TK]D-Fender: ring or polarity? |
| 04:21.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:21.10 | [TK]D-Fender | prg3: between 1st & 2nd rings |
| 04:22.31 | prg3 | [TK]D-Fender: know offhand the zapata.conf flag to make it behave? |
| 04:22.58 | kb3ien | gtg bed. |
| 04:23.10 | [TK]D-Fender | prg3: shouldn't need any with the basic zone data |
| 04:23.53 | prg3 | Strange.. this machine of mine just won't pull CID off the lines, I've checked the lines, and all have CID according to an analog phone.. |
| 04:24.25 | prg3 | fxs_ks wouldn't be interferring? |
| 04:25.01 | [TK]D-Fender | prg3: pastebin your actual configs... |
| 04:25.01 | prg3 | http://pastebin.com/m763fe25e |
| 04:25.06 | Brijn | I see quite a few posts on the username mismatch, but no solutions.. Any tips? |
| 04:25.11 | prg3 | zapata.conf, anything else you want? |
| 04:25.22 | [TK]D-Fender | WOW. Fast turnaround. |
| 04:25.30 | [TK]D-Fender | GOD, 1 million dollars please!! |
| 04:25.32 | [TK]D-Fender | waits |
| 04:26.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.201) |
| 04:26.56 | [TK]D-Fender | prg3: http://pastebin.com/m65d7e778 |
| 04:27.19 | [TK]D-Fender | prg3: you missed the "calledid=" and I removed the other junk. Just drop this in |
| 04:27.42 | [TK]D-Fender | prg3: OH, and your gains are F-ing psycho... |
| 04:27.42 | *** join/#asterisk mick_laptop (n=mweiss@clamwin/admin/mickhome) |
| 04:28.49 | mick_laptop | if i want people to dial each other via a 3 digit extension (via iax) -- what do i need to add in the extensions.conf? i have the users setup in iax.conf |
| 04:28.54 | prg3 | [TK]D-Fender: one suggestion I found was playing with gains.. I shoulda set that down. |
| 04:28.59 | mick_laptop | i figure that i need something under [local] |
| 04:29.10 | [TK]D-Fender | mick_laptop: EXTENSIONS. Go read chaper 5 of THE BOOK... |
| 04:29.11 | [TK]D-Fender | ~book |
| 04:29.12 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 04:29.12 | prg3 | [TK]D-Fender: No change.. zap show channel 7 should give the CID data? |
| 04:29.30 | [TK]D-Fender | prg3: high gains can kill CID |
| 04:29.39 | [TK]D-Fender | prg3: did you completely restart *? |
| 04:30.10 | LemensTS | which agi command acts like read? i need to capture the keypresses after the sound is played...i see get data, get option, stream file, etc |
| 04:30.17 | [TK]D-Fender | prg3: "zap show channel 7 should give the CID data?" - No |
| 04:30.27 | [TK]D-Fender | LemensTS: GET DATA |
| 04:30.38 | [TK]D-Fender | LemensTS: Ther are 2 whoe chapters on this in THE BOOK. |
| 04:30.42 | [TK]D-Fender | whole* |
| 04:33.33 | LemensTS | TK: ahh i thought that was opening a config file and getting data out of it or something from the name of it. |
| 04:33.39 | LemensTS | thx |
| 04:34.11 | prg3 | with /etc/init.d/asterisk restart , and zaptel restart as well.. |
| 04:34.11 | prg3 | [TK]D-Fender: Then my test is faulty.. where's the "best" place to see CID data? |
| 04:34.11 | prg3 | for debugging.. |
| 04:34.55 | mick_laptop | thanks - i'm reading now |
| 04:35.49 | [TK]D-Fender | prg3: NoOp(Name is "${CALLERID(name)}" number is "${CALLERID(num)}") |
| 04:36.40 | prg3 | [TK]D-Fender: That's in the extension? |
| 04:36.55 | Brijn | Hmm, changed username back to 300 and all is OK, so something funny between Ploycom 510 and my version of * |
| 04:37.19 | [TK]D-Fender | prg3: Something like the first priority |
| 04:37.42 | [TK]D-Fender | Brijn: Nope. |
| 04:37.43 | prg3 | [TK]D-Fender: Thank you very much! it appears to be working |
| 04:38.01 | [TK]D-Fender | prg3: You're welcome |
| 04:38.32 | prg3 | Why does zap show channel have "Caller ID:" if it doesn't fill it? |
| 04:38.57 | [TK]D-Fender | prg3: because its meant for FXS |
| 04:40.28 | prg3 | Ah, so it's CID that * is sending, not CID that * is recieving.. nice |
| 04:41.48 | prg3 | Next, I get to sort out insane dialplan mess to get a queue behaving with remote dial in agents.. any insights into that one, or just search, read and some back with something specific and a pastebin url? |
| 04:42.58 | [TK]D-Fender | prg3: Yes... please to to clerly IDENTIFY your problem and then maybe we have a chance to fix it |
| 04:44.59 | prg3 | [TK]D-Fender: Current problem is mostly lack of experience, but, I'll try to explain what I want to do. I am trying to get a inbound caller queue system working, where the agents will register as either internal SIP clients or external coming in throug Zap. |
| 04:45.35 | prg3 | The agents will phone into an extension, and an interactive menu will get their agent id and their password, and then add them to the agents for the queue |
| 04:47.21 | SkramX | ooh |
| 04:47.24 | SkramX | sounds like what I just did |
| 04:47.28 | SkramX | prg3 have fun! |
| 04:47.32 | stabler | lol |
| 04:47.33 | [TK]D-Fender | prg3: Messy. |
| 04:48.09 | prg3 | [TK]D-Fender: Yeah, that's about where I am.. I have some of it behaving. |
| 04:48.14 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 04:48.17 | prg3 | SkramX: Got an extensions.conf you can share? :) |
| 04:48.24 | SkramX | I wrote it all in an AGI |
| 04:48.39 | SkramX | and.. I love open source... but I have to make money and I did do part of this on company time |
| 04:48.56 | SkramX | so my extensions.conf is only a couple lines ;) |
| 04:49.43 | prg3 | :) |
| 04:53.14 | SkramX | sorry |
| 04:53.33 | prg3 | Not a problem.. |
| 04:53.34 | SkramX | but there is plenty of free documentation/tutorials as well as people that will code this up for you for a fee |
| 04:53.52 | prg3 | Got a number/url of someone good? |
| 04:53.59 | SkramX | haha |
| 04:54.23 | SkramX | my way uses dynamic agents and they authenticate with a database independent of asterisk |
| 04:56.04 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:56.40 | stabler | im having some issues with a super simple IVR... when i select 2 which should dial an extension |
| 04:56.53 | stabler | i get this on the console and a busy signal |
| 04:56.55 | stabler | Called 1002/30 |
| 04:56.55 | stabler | <PROTECTED> |
| 04:56.55 | stabler | <PROTECTED> |
| 04:56.55 | stabler | <PROTECTED> |
| 04:56.55 | stabler | <PROTECTED> |
| 04:57.09 | stabler | this is my conf |
| 04:57.10 | stabler | http://pastebin.com/m2818c3d5 |
| 04:57.43 | stabler | if i just dial it from the extension is works fine |
| 04:59.04 | [TK]D-Fender | stabler: SIP/1002 is calling... **ITSELF** and you're surprised its reporting back BUSY? |
| 04:59.23 | [TK]D-Fender | stabler: And what does "dial it from the extension" mean? |
| 04:59.30 | stabler | this happens from any extension |
| 04:59.44 | [TK]D-Fender | stabler: Show another call. |
| 04:59.59 | [TK]D-Fender | stabler: a COMPLETE CALL. |
| 05:00.41 | stabler | Called 1002/30 |
| 05:00.42 | stabler | <PROTECTED> |
| 05:00.42 | stabler | <PROTECTED> |
| 05:00.42 | stabler | <PROTECTED> |
| 05:00.42 | stabler | <PROTECTED> |
| 05:01.35 | stabler | Called 1002 |
| 05:01.35 | stabler | <PROTECTED> |
| 05:01.35 | stabler | <PROTECTED> |
| 05:01.35 | stabler | <PROTECTED> |
| 05:01.36 | [TK]D-Fender | stabler: PASTEBIN |
| 05:01.43 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 05:02.15 | stabler | sorry |
| 05:02.52 | [TK]D-Fender | stabler: PB "sip show peers" and PASTEBIn the ENTIRE call. |
| 05:05.15 | stabler | ok |
| 05:09.11 | stabler | http://pastebin.com/m262b1e85 |
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| 05:09.36 | stabler | didnt mean to post an amendment posting |
| 05:09.36 | stabler | but it works |
| 05:09.41 | [TK]D-Fender | stabler: I asked to for a COMPLETE call attempt. You keep showing only part of the call. Keep this up and I will stop wasting my time on this. |
| 05:10.13 | [TK]D-Fender | stabler: And in this last one everything seemed to work. |
| 05:10.41 | LemensTS | TK: $keypress = $agi->get_data("vm-Work", 10000, 1); it doesnt play, just shows in CLI AGI Rx << GET DATA vm-Work 10000 1 |
| 05:11.16 | [TK]D-Fender | LemensTS: I also don't debug single lines fo code I don't get to see executed along with eveyrhting around it |
| 05:12.51 | stabler | http://pastebin.com/m1dfb5462 |
| 05:13.16 | stabler | [TK]D-Fender, sorry about that |
| 05:14.03 | LemensTS | TK: http://pastebin.com/m15b48675 |
| 05:14.35 | [TK]D-Fender | LemensTS: ENTIRE CALL |
| 05:15.13 | TommyTheKid | Hi, is anyone here familiar with using PRI's and Dial "progress" codes? |
| 05:16.06 | [TK]D-Fender | TommyTheKid: loike? |
| 05:16.18 | stabler | http://pastebin.com/m1dfb5462 |
| 05:16.37 | stabler | the only time i have the issue is when is use the ivr... the ext to ext call is fine |
| 05:16.40 | TommyTheKid | Usually when a number is "busy" I get code 17, and dial fails immediately and ${DIALSTATUS} is "BUSY" .. as expected.. but sometimes I am getting code 31, and it *sounds* like a busy tone or sometimes "fast busy" .. but its not reporting to Dial that its busy, Dial thinks the call succeeded ( I think) |
| 05:16.41 | stabler | doesnt make sense |
| 05:17.15 | [TK]D-Fender | stabler: Where's the failure in there? |
| 05:17.34 | stabler | i posted that earlier |
| 05:17.41 | [TK]D-Fender | TommyTheKid: Less "think" more "show" |
| 05:18.00 | [TK]D-Fender | stabler: Last time : Show me a COMPLETE &^#$%ING call with your FAILURE IN IT |
| 05:19.31 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 05:19.56 | clyrrad | quick question if any of you know the answer. RealTime is teliing me it failed to connect to database server asterisk on localhost, its telling me to check "debug" for more information, but there is no "debug" file in /var/log/asterisk. Anyone know where this "debug" info is? |
| 05:20.54 | [TK]D-Fender | clyrrad: could be its implying debug of your DB server |
| 05:20.58 | stabler | http://pastebin.com/m578db768 |
| 05:21.02 | stabler | i misunderstood |
| 05:21.11 | stabler | i thought you meant a complete call ext to ext |
| 05:21.22 | stabler | sorry |
| 05:23.00 | TommyTheKid | chatzilla? |
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| 05:23.40 | [TK]D-Fender | stabler: exten => 2,1,Dial(SIP/1002/30) <-- you put a "/" instead of a "," |
| 05:23.47 | clyrrad | [TK]D-Fender: Yup, that seems to be the case thanks |
| 05:23.55 | [TK]D-Fender | stabler: Next time read your syntax closer |
| 05:24.12 | TommyTheKid | [TK]D-Fender: Sorry, this happened earlier today, and it has apparently left my scrollback.. is there a way to get old console history to re-play sortof? :) I couldn't find anything in the logs (they aren't verbose=3+) |
| 05:24.53 | [TK]D-Fender | TommyTheKid: Sorry res_fluxcapacitor.so is only going to get released 3 weeks ago. Check back then. |
| 05:25.14 | [TK]D-Fender | TommyTheKid: And logs = BS. Live CLI or bust |
| 05:25.28 | TommyTheKid | understood, I will be sure to capture it tomorrow when it happens again |
| 05:25.51 | *** join/#asterisk andrewn (n=andrew@76-191-212-109.dsl.dynamic.sonic.net) |
| 05:25.54 | TommyTheKid | is there anywhere you suggest looking for reference on the PRI status codes that DIAL reports (17 and 31) to see what 31 might be? |
| 05:26.14 | stabler | [TK]D-Fender, thanks.. i dont know the syntax well yet |
| 05:27.53 | [TK]D-Fender | stabler: You don't have to know it well, you only have to look enough at what you've done to see that you aren't consistent and ask. If you were consistently wrong I might simply think you're misinformed. As it is I'd figure you really just missed this due to a twitchy "/" finger |
| 05:28.13 | [TK]D-Fender | TommyTheKid: Quick guess "PRI debug" |
| 05:30.26 | TommyTheKid | http://asterisk.pastebin.ca/1333647 |
| 05:31.35 | TommyTheKid | one of my users reported the number... basically if we get a busy, they have an option to have the system redial it for them till it rings through, but its like its not being detected as busy |
| 05:31.41 | [TK]D-Fender | TommyTheKid: no PRI debug or SIP debug in there |
| 05:31.48 | TommyTheKid | no ;) |
| 05:32.11 | TommyTheKid | dont think its a sip issue, the outbound call is PRI (it just so happens I was calling from SIP) |
| 05:32.37 | [TK]D-Fender | TommyTheKid: Think? Try looking first, its supposed to be easier apparently ;) |
| 05:33.39 | TommyTheKid | I know you get pummeled with stupid ass questions all day long, sorry I did not understand you were looking for debug info.. |
| 05:34.00 | carrar | TK likes the abuse! |
| 05:34.06 | TommyTheKid | I have turned on debugging for all four spans and now my console has filled with "Sending Set Asynchronous Balanced Mode Extended" |
| 05:36.38 | TommyTheKid | http://asterisk.pastebin.ca/1333648 |
| 05:38.56 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:39.20 | TommyTheKid | Note that the call is *ringing* right now, but I am seeing the same message as before "PROGRESS with cause code 31 received" |
| 05:39.55 | TommyTheKid | again, I will have to try to reproduce the non-detected-busy tomorrow |
| 05:40.31 | [TK]D-Fender | TommyTheKid: Whats on the other end of your TDM card? |
| 05:40.35 | TommyTheKid | Do you think it could possibly be something like they have a PBX at the other end and they are effectively doing a "Playtones(busy)" but not sending back the Busy ? |
| 05:40.51 | TommyTheKid | [TK]D-Fender: I wish I knew, its a freaking radio station |
| 05:41.04 | [TK]D-Fender | TommyTheKid: No... what is the CARD plugged INTO ? |
| 05:41.27 | [TK]D-Fender | TommyTheKid: Please tell me the "I don't know" wasn't intended to answer that... |
| 05:41.31 | TommyTheKid | heh, oh sorry, tis plugged into an Avaya PBX (our corporate phone system) |
| 05:41.51 | [TK]D-Fender | TommyTheKid: You absolutely sure the 9 in front is legal? |
| 05:41.57 | TommyTheKid | I thought you meant what was at the *far* end of the call |
| 05:42.29 | *** join/#asterisk woonix (i=woot@c-24-20-125-121.hsd1.or.comcast.net) |
| 05:42.41 | [TK]D-Fender | TommyTheKid: < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Access information discarded (43), class = Network Congestion (resource unavailable) (2) ] |
| 05:42.43 | TommyTheKid | absolutely certain, we are just getting our PRI service from our corporate PBX, which requires a "9" to get an outside line, otherwise 5 digit dial |
| 05:43.24 | carrar | Did you try without |
| 05:43.40 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 05:43.58 | TommyTheKid | without the 9 would be difficult, my cisco has a "dialplan" which would only accept 5 digits if it didn't start with 9 :) |
| 05:44.11 | [TK]D-Fender | TommyTheKid: cheat with * <- |
| 05:44.24 | carrar | :1 |
| 05:44.30 | TommyTheKid | i could make an extension that forced the dial, but all other numbers work with the 9, its just that one |
| 05:45.00 | [TK]D-Fender | TommyTheKid: Fair enough |
| 05:45.07 | TommyTheKid | and its not every time either thats what bothers the hell out of me. I saw my "busy-auto-redial" macro hit 170 calls to that number, then sometimes it goes to progress 31 after one or two |
| 05:45.32 | [TK]D-Fender | TommyTheKid: Ok, noop the HANGUPCAUSE, etc after the call attempt. |
| 05:45.44 | TommyTheKid | I am strongly suspecting that its something on *their* side, i was just curious as thats the only time I have really ever seen progress cause code 31 ;) |
| 05:46.33 | TommyTheKid | the auto-dialer does, it actually does GotoIf based on it too, but Dial still has control, while hte user can "hear" a busy signal |
| 05:47.46 | TommyTheKid | Normal, Unspecified. |
| 05:47.46 | TommyTheKid | <PROTECTED> |
| 05:47.46 | TommyTheKid | This is a very common cause code and happens when the network is not able to determine what to do |
| 05:47.46 | TommyTheKid | with the call being made. |
| 05:47.57 | TommyTheKid | uh, sorry, I thought that would go on one line |
| 05:48.05 | [TK]D-Fender | TommyTheKid: Sounds like inband progress could be its passed through a carrier that doesn't pass it back normally |
| 05:48.45 | TommyTheKid | I have no idea where to even start looking in the code, but it almost looks like cause code 31 is not being handled, maybe? |
| 05:49.37 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:49.58 | [TK]D-Fender | TommyTheKid: go NoOp all the usualy reason code vars and look. |
| 05:50.24 | TommyTheKid | http://www.cisco.com/en/US/docs/ios/11_0/debug/command/reference/disdn.html#wp1532 |
| 05:51.50 | TommyTheKid | sorry that was not in response to your last statement, it was just a reference with "cause code values" :) |
| 05:54.13 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 05:56.04 | rob0 | yo, Wally had several Vonage DLink VTA-VR's on clearance for $30. Master Google suggests that they're unlockable! |
| 05:57.59 | [TK]D-Fender | In other related news strap-on's are %30 off while quantities last! |
| 05:59.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 06:02.08 | TommyTheKid | I guess I need to delve into the code of Dial to start? Dial still has control, it appears to see the "cause code 31" ... although maybe that message is being emitted by DAHDI as it looks like a sub-element of "Processing IE 30 (cs0, Progress Indicator)" .. Line 60 of my pastebin might be the killer, "further call progress information may be available inband" |
| 06:03.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:04.19 | [TK]D-Fender | TommyTheKid: I think you need ot NoOp all those damn variables after your dial like I already told you to do |
| 06:04.39 | [TK]D-Fender | grabs his ClueBat (tm) |
| 06:05.23 | TommyTheKid | 1. what variables? ${DIALSTATUS} and ${HANGUPCAUSE} I assume? (any others?) |
| 06:06.13 | TommyTheKid | 2. will they not be tainted by the fact that I had to "hang up" on my end to end the call instead of Dial exiting and the dial plan going to the next priority cause it detected busy |
| 06:06.55 | [TK]D-Fender | TommyTheKid: IMMEDIATELY after your DIAL |
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| 06:29.57 | [TK]D-Fender | checkout time. Later all |
| 06:33.26 | harry_v | exit |
| 06:34.15 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 06:34.59 | TommyTheKid | guh! |
| 06:35.08 | TommyTheKid | I was just getting done with my debug paste :p |
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| 06:45.52 | TommyTheKid | haha nice name |
| 06:55.27 | TommyTheKid | if anyone else has been idly listening and has any idea how to help me, http://asterisk.pastebin.ca/1333716 |
| 06:58.49 | nix8n82 | is 303 denver? |
| 06:59.12 | TommyTheKid | yes |
| 07:04.03 | nix8n82 | cool just checking I"m wyoming..you a broncos fan? |
| 07:05.05 | TommyTheKid | sometimes, as much as a computer geek can be, I suppose |
| 07:05.05 | nix8n82 | I came in late..what was your problem? |
| 07:06.03 | TommyTheKid | When I "Dial()" a certain number, and its busy, its supposed to exit dial with DIALSTATUS=BUSY ... but sometimes its getting stuck (not exiting when it gets PRI Cause code 31) |
| 07:06.10 | TommyTheKid | .. if its supposed to exit there |
| 07:06.38 | TommyTheKid | either way, my user *hears* an inband busy signal tone, but Asterisk is not aware that the line is busy (making it so that my auto-busy-on-redial macro gets stuck |
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| 07:16.16 | nix8n82 | I hate team sports especially a fancy version of smeer the queer..anyway I don't know what to tell ya about the pri thing..can I see the macro code? |
| 07:18.54 | *** join/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 07:19.29 | ultrav1olet | If my users are defined in users.conf how can I Dial() them? I don't know whether they are connected using IAX2 or SIP |
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| 07:22.43 | TommyTheKid | macro code has been massively simplified at the top (in the dialplan show testing) |
| 07:23.31 | TommyTheKid | basically its all about being caller 106 when they give the cue to call for a metal shop v key ;p |
| 07:24.45 | ultrav1olet | anyone? |
| 07:24.46 | TommyTheKid | ultrav1olet: in our configuration we have an extension for the username |
| 07:25.06 | TommyTheKid | that extension calls all their numbers, sip's iax's trunk (cell) |
| 07:25.18 | ultrav1olet | can you give the example? |
| 07:25.20 | TommyTheKid | but that was decidedly an asterisk 1.2 thing, not sure how the new fancy users.conf works :) |
| 07:26.23 | ultrav1olet | TommyTheKid: probably you have a macro, right? |
| 07:26.30 | ultrav1olet | how does it look like? |
| 07:27.05 | TommyTheKid | i am looking, its been forever, I am only dabling with the pbx cause I want to win a radio contest ;p |
| 07:27.48 | ultrav1olet | ;) |
| 07:28.02 | kaldemar | ultrav1olet: from sample users.conf: "Creating entries in users.conf is a "shorthand" for creating individual entries in each configuration file." |
| 07:29.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 07:29.25 | TommyTheKid | ultrav1olet: my extensions.conf has a [users] section that has exten => 12345,2,Macro(stdexten,SIP/user&SIP/user-cisco&SIP/user-e71&IAX2/user&${TRUNK}/913031234567) |
| 07:29.47 | frogonwheels | ~users.conf |
| 07:29.48 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 07:29.56 | frogonwheels | ultrav1olet: ^^^^ |
| 07:29.59 | TommyTheKid | stdexten is pretty simple, it just basically dials ${ARG1} then on busy/noanswer it goes to voicemail |
| 07:30.08 | TommyTheKid | .. it voicemail fails, it says f'off |
| 07:30.14 | TommyTheKid | (er .. goodbye) |
| 07:30.35 | TommyTheKid | haha |
| 07:31.17 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:31.19 | TommyTheKid | ultrav1olet: you could just as easily replace the "Maro(stdexten," with "Dial(" |
| 07:31.40 | TommyTheKid | but yea, see jbot's response, I have never used users.conf (at least the official one) |
| 07:32.26 | TommyTheKid | I can see how it might be considered desirable, but it sounds like it may have been poorly implemented ? |
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| 07:33.41 | TommyTheKid | somehow it just feels like that should be a database to me, but that probably makes things worse... even if the database dumped its config out to flat asterisk files |
| 07:34.02 | kaldemar | as long as you can't configure clients for different protocols identically, using users.conf is like giving oneself a shot in the leg. |
| 07:34.03 | TommyTheKid | my configuration is *veeery* simple, I have about 40 users, 10 of which might actually use it :) |
| 07:34.18 | *** join/#asterisk joobie (n=joobie@210-84-27-211.dyn.iinet.net.au) |
| 07:35.34 | ultrav1olet | frogonwheels: OK, I've already moved all users to iax |
| 07:35.59 | ultrav1olet | kaldemar: I just like unification and simplification |
| 07:36.11 | TommyTheKid | so you have to have a section for each user in iax.conf, voicemail.conf, sip.conf, extensions.conf, (am I missing any.conf?) |
| 07:36.32 | frogonwheels | ultrav1olet: which is fine - .. but 'build a system even a fool can use, and only a fool will want to use it' springs to mind as well :) |
| 07:36.38 | TommyTheKid | in my case, some users have several sip entries due to sip not being able to handle multiple registrations |
| 07:36.56 | TommyTheKid | some users have their own home asterisk box, so they have an IAX2 entry too |
| 07:37.02 | TommyTheKid | its really quite chaotic :) |
| 07:38.00 | frogonwheels | TommyTheKid: you could possibly aggregate entries to a single user using agents.conf - but don't know all caveats |
| 07:38.15 | TommyTheKid | interesting :) |
| 07:39.26 | TommyTheKid | The problem I have is that I have to be able to support a softphone, hard-ip phone, and potentially a mobile-sip client (mostly E71's at this point). I would prefer they all registered with the same sip account and it just sent the "invite" to all the registered peers, but I guess it doesn't work that way |
| 07:39.58 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:41.19 | TommyTheKid | instead I manage 3 sip channels, an iax (or two), maybe a few dial-out numbers for each user,..... then I and (&) them all together in their "users" entry (extensions.conf) |
| 07:41.51 | TommyTheKid | thats why I was thinking maybe a database might be the way to go given the requirements and limitations |
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| 07:43.28 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 07:46.31 | ultrav1olet | the default asterisk configuration includes stdexten macro - is it safe to use it? is it well tested? I don't understand some actions it performs - so I'm asking |
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| 07:48.08 | ultrav1olet | oops, in fact it's an extension |
| 07:48.44 | TommyTheKid | ultrav1olet: I think I am probably using it, considering its the same name :) |
| 07:50.56 | TommyTheKid | yep, I am using that, I did add something to the bottom to pick up an "unconfigured" voicemail box |
| 07:52.43 | TommyTheKid | sortof a "sorry this user doesn't have voicemail, goodbye" |
| 07:56.37 | TommyTheKid | needs sleep |
| 08:01.27 | ttk-sleep | ultrav1olet: http://www.voip-info.org/wiki/view/Stdexten+macro take a look at the v 0.2 .. it has a check for voicemail for example |
| 08:01.55 | ttk-sleep | .. wait no thats broken, disregard |
| 08:03.33 | ttk-sleep | anyhow, stdexten works fine, use it *if* you want voicemail, otherwise, just use Dial |
| 08:03.34 | ttk-sleep | nite! |
| 08:04.42 | *** join/#asterisk af_ (n=getsmart@88-149-230-21.dynamic.ngi.it) |
| 08:05.32 | gones | Have anyone heard about sip 185 ? |
| 08:09.10 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 08:09.26 | Zeeek | so, is 2.0 out yet? |
| 08:10.10 | denon | 1,6 |
| 08:10.12 | denon | 1.6 |
| 08:10.49 | denon | of course, version number has very little bearing on functionality |
| 08:10.58 | denon | look at OSX for example :) |
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| 08:22.24 | imcdona | does anyone have an example amd.conf I could look at? |
| 08:26.19 | imcdona | found it http://tinyurl.com/akgb3o |
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| 08:38.43 | Zeeek | what about OS X? do you use it? |
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| 08:46.49 | Zeeek | denon, your example doesn't work, OS X is worlds away from OS 9 |
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| 09:05.40 | jermey_g | how can i put some initial values in the asterisk db |
| 09:06.13 | jermey_g | e.g. i imagine something in [globals] DBPut(f1/k1)=value1; |
| 09:06.15 | Zeeek | using the CLI |
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| 09:06.55 | jermey_g | i have read that the values are never lost even if you reboot. where are they stored? |
| 09:07.59 | Zeeek | in a file |
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| 09:08.59 | Zeeek | jermey_g: take a look at this: http://www.voip-info.org/wiki-Asterisk+Database |
| 09:11.52 | jermey_g | Zeeek: Thanks! |
| 09:12.15 | Zeeek | it's actually one of the better pages on voip-info! |
| 09:12.21 | imcdona | what purpose does the predefined extension 'n' provide in extensions.conf? |
| 09:12.29 | Zeeek | everything you need is there, where it is, how to change add delete via CLI, etc |
| 09:12.43 | Zeeek | imcdona: n allows you to renumber |
| 09:12.54 | Zeeek | rather than 1,2,3 you use 1,n,n,n,n |
| 09:13.15 | Zeeek | that way you can insert and move around without having to renumber |
| 09:13.32 | Zeeek | actually it is a priority, not an extension |
| 09:14.41 | imcdona | zeek in this example :http://pastebin.com/m9951e68 after it reaches status-machine for exacmple, can I just start putting in piroties after that? |
| 09:15.39 | Zeeek | yes but the goto lines will obviously make execution go to the destinations |
| 09:16.06 | imcdona | Thanks zeeek, for some reason I was thinking it was an extensino. I am on the right track now ty |
| 09:16.22 | Zeeek | no problemo |
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| 09:24.52 | *** join/#asterisk kamh (n=qmpelkam@xdsl-1814.wroclaw.dialog.net.pl) |
| 09:24.58 | kamh | hi guys |
| 09:25.38 | kamh | do You know if UA should send # character instead of code 11 to SIP server? |
| 09:26.02 | kamh | when I send 9# by DTMF I see 9\r\n and then 11\r\n |
| 09:26.11 | kamh | is it proper behaviour? |
| 09:26.18 | kamh | or maybe it should be 9\r\n |
| 09:26.24 | kamh | #\r\n |
| 09:26.25 | kamh | ? |
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| 09:53.49 | jermey_g | how are extensions matched? is the extenion order important in dial plan? e.g. if i put _X. before an exten 2848 then what happens? |
| 09:54.30 | jermey_g | if a call come for 2848, would it take it to _X.? no, right? |
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| 09:56.14 | kim0 | Hi gurus ... I have a TDM410 card ... It needs a PC power cord, but my box doesn't have any ... Will I really need the power cord ? |
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| 10:06.20 | jermey_g | is this possible Set(DB(f1/${DB(ff1/k1)}=3}) |
| 10:06.51 | jermey_g | where f1/k1 contains a valid entry |
| 10:11.55 | Gido-E | kim0 my experience yes. |
| 10:12.17 | kim0 | Gido-E: I think I only need it for FXS |
| 10:12.21 | kim0 | which I dont have |
| 10:12.22 | Gido-E | yep |
| 10:12.27 | kim0 | great |
| 10:18.09 | dwery | 'morning... mmm my grandstream bt102 doesn't store called/callers list.. any clue? |
| 10:28.58 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 10:29.12 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 10:35.10 | ultrav1olet | where can I read about the syntax of Voicemail command in Asterisk 1.6? |
| 10:37.18 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:38.09 | *** part/#asterisk kim0 (n=kimoz@unaffiliated/kim0) |
| 10:39.49 | mort_gib | Hi Mikkel |
| 10:40.08 | mikkel | Hi |
| 10:40.16 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 10:40.17 | mort_gib | How are things?? |
| 10:42.10 | Gido-E | ultrav1olet core show application voicemail<tab> ? |
| 10:42.14 | mikkel | Things are good, have gotten my * to run stable now. How are you, did you get that order you talked about ? |
| 10:42.53 | ultrav1olet | Gido-E: thanks! |
| 10:43.45 | ultrav1olet | When I call from outside of my SIP provider back to my office softphone I get zillion of errors: [Feb 11 15:42:41] WARNING[23131]: chan_sip.c:5041 sip_write: Asked to transmit frame type 2, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 10:43.57 | ultrav1olet | and I hear no sound - what can be done about that? |
| 10:44.03 | *** join/#asterisk jape44 (n=jape44@85.233.236.59) |
| 10:53.01 | xrmx__ | is there any tool available to parse sip debug output? |
| 10:55.10 | imcdona | e164.org has a no number harvesting policy. But they give you the option to opt in for teleresearch calls. What is the point of having the option to opt in or out? |
| 10:56.45 | imcdona | xrmx wireshark does an awesome job of parsing VoIP calls. It even captures the RTP. If you have an old hub laying arond or a mirrored switch port I would hightly recommend wireshark |
| 10:59.23 | xrmx__ | imcdona, thanks, the ideal would be a tool to parse asterisk cli debug output and let me follow calls |
| 11:03.06 | *** join/#asterisk fdz (n=francois@c2cpc3.camptocamp.com) |
| 11:05.22 | mort_gib | mikkel: Well I'm still talking to them... |
| 11:06.49 | mort_gib | mikkel: So many other projects :-( |
| 11:07.22 | mikkel | mort_gib, I got my hands full aswell. |
| 11:08.54 | mort_gib | mikkel: I have some 120 users on Asterisk now... |
| 11:09.07 | mort_gib | in groups of 10-15 |
| 11:09.36 | mort_gib | A lot of strange dial plans.... |
| 11:10.52 | mikkel | mort_gib, I only go 2 phones and no avnaced dialplans. But have finally found out the the problems I have, was the B&O phones fault. Interferens from the wireless box. |
| 11:16.35 | mort_gib | mikkel: Well... Vendor independent... Uhm |
| 11:17.22 | *** join/#asterisk arejay (i=arejay@dawnshosting.com) |
| 11:18.29 | mort_gib | mikkel: I had a spree of Firmware issues, after a certain Firmware update none of my Snom phones could *8 |
| 11:20.15 | mikkel | mort_gib, Yes, was a wird problem, only sound one way. So always assumed it was *, but that is always a mistake to assume something :) |
| 11:21.15 | mort_gib | Yeah, I had a serial cable costing me 2 hours... Faulty but I blamed the BIOS in my Soekris box |
| 11:22.03 | mort_gib | mikkel: never trust hardware |
| 11:22.22 | ultrav1olet | How can I set the language of voice messages spoken out from extensions.conf? |
| 11:22.33 | ultrav1olet | I run asterisk 1.6 |
| 11:22.44 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:25.45 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 11:29.37 | vlt | Hello. I moved from asterisk 1.2.13 (Debian Etch) to 1.4.21 (Debian lenny). Now my SIP subscriptions don't work anymore. `sip show subscriptions` shows me a valid list but the "last state" field is "idle" everywhere even when a channel is actually active. Was there a syntax change how to define the hints? |
| 11:36.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:39.58 | ultrav1olet | vlt: enable sip debug and look how it all goes on |
| 11:44.37 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.79.73) |
| 11:49.48 | *** join/#asterisk badcfe (i=cso@78.156.5.233) |
| 11:50.02 | badcfe | is it possible to Playback a video-stream? |
| 11:52.08 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:52.39 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 11:53.17 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.24 | *** join/#asterisk coppice (n=chatzill@201.162.17.210.dyn.pacific.net.hk) |
| 12:09.28 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 12:09.49 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 12:12.12 | bombaclat667 | I am a bit confused here....using asterisk 1.2.23.1 . Via asterisk gui, my trunk_1 is set to Zap/g1 . Whenever I use that it fails (unable to create channel of type 'Zap'). |
| 12:12.26 | bombaclat667 | WHen I use Zap/1 , it works, but does not show the trunk line as busy in the gui |
| 12:12.56 | bombaclat667 | (yet the line zap/1 is being used to make a call and works) |
| 12:13.52 | *** join/#asterisk imcdona (i=imcdona@c-24-17-45-79.hsd1.wa.comcast.net) |
| 12:15.20 | beek | bombaclat667: What group are those Zap channels configured for? |
| 12:18.32 | bombaclat667 | what do you mean by grp? context? and its 1.4.23.1, not 1.2 ..my bad |
| 12:20.37 | *** part/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 12:23.13 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
| 12:28.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 12:29.57 | *** join/#asterisk Kage` (n=Kage@c-71-60-67-135.hsd1.wv.comcast.net) |
| 12:34.47 | kaldemar | bombaclat667: there are no contexts in zapata.conf. groups are defined above channel-lines with "group => X", where X is the number of the group, 1 in your case. |
| 12:35.01 | bombaclat667 | exit |
| 12:35.15 | bombaclat667 | lol...wrong window...was gonna check if I even set that var up |
| 12:35.45 | bombaclat667 | yeah..it wasnt defined |
| 12:43.10 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
| 12:44.11 | pithen | I really hate that the US and Canada share the same numberspace... is there any programatic way to tell a US number from a Canadian number, other than importing the entire list of area codes? |
| 12:49.51 | bombaclat667 | bleh..I edited the configs and now asterisk doesn't see the channel anymore..so I cannot tell if adding teh group worked :( |
| 12:51.21 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-219-150.lyn.east.verizon.net) |
| 12:51.42 | bombaclat667 | it shows the line in the trunk, but 'show channels' in the CLI shows none |
| 12:52.44 | kaldemar | show channels shows active channels |
| 12:53.18 | kaldemar | i.e. when there are ongoing calls |
| 12:53.18 | bombaclat667 | 0 active channels |
| 12:53.41 | kaldemar | you can tell by making a call that uses the group. |
| 12:54.46 | kaldemar | btw, editing configs by hand and using a GUI at the same time is an extremely bad idea. |
| 12:55.28 | *** join/#asterisk qdk_ (n=qdk@81.7.168.130) |
| 12:55.47 | bombaclat667 | I agree..lol, but I wasn't able to get the line to work with the gui |
| 12:56.05 | bombaclat667 | I had to manually create the zapata file for the lines |
| 12:56.25 | kaldemar | the gui will probably screw the configurations then at some point. |
| 12:56.25 | bombaclat667 | exit |
| 12:56.30 | bombaclat667 | grr..wonrg window.. |
| 12:57.14 | bombaclat667 | are trunks from teh GUI dahdi only? |
| 12:57.31 | bombaclat667 | now I think of it...maybe im actually fucking shit up by using both |
| 12:57.46 | kaldemar | yes you are |
| 12:57.48 | bombaclat667 | and maybe thats why its sporadic |
| 12:57.58 | bombaclat667 | ok |
| 12:58.47 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 12:59.28 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 12:59.36 | bombaclat667 | all I need to do is remove the zapata,conf file and it showld only be dahdi, or is there another one? |
| 13:00.07 | kaldemar | but you're using zaptel, right? |
| 13:00.29 | bombaclat667 | yes I did install teh zaptel drivers |
| 13:00.34 | stevetotaro | don't forget mahmi |
| 13:00.58 | kaldemar | to use DAHDI you'd have to compile DAHDI and asterisk again against DAHDI. |
| 13:01.30 | vlt | Hello. I moved from asterisk 1.2.13 (Debian Etch) to 1.4.21 (Debian lenny). Now my SIP subscriptions don't work anymore. `sip show subscriptions` shows me a valid list but the "last state" field is "idle" everywhere even when a channel is actually active. Was there a syntax change how to define the hints? |
| 13:01.40 | *** join/#asterisk adnc (n=adnc@unaffiliated/adnc) |
| 13:01.54 | vlt | I enabled sip debug and there's no status message sent out from Asterisk. |
| 13:01.56 | kaldemar | after, mahmi, there'll be sahni. |
| 13:02.31 | bombaclat667 | ok, but is the gui revolving around both or dahdi only? basicly all I want is to be able to monitor the line from it |
| 13:03.23 | bombaclat667 | I agree that the name was poorly chosen |
| 13:04.14 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.21.163.adsl.gvt.net.br) |
| 13:04.20 | vncsnvs | hello |
| 13:04.29 | vncsnvs | anyone works w freepbx? |
| 13:04.45 | kaldemar | vncsnvs: #freepbx |
| 13:04.50 | vncsnvs | thx |
| 13:06.42 | adnc | is there a dial pattern rule that can distinguish a @ sign, so that all calls to a address with an at-sign go to a particular provider? |
| 13:06.48 | *** join/#asterisk path_ (n=path@92-106-21-190.adsl.terra.cl) |
| 13:06.50 | kaldemar | bombaclat667: ask in #asterisk-gui also |
| 13:06.59 | vncsnvs | hey |
| 13:07.07 | vncsnvs | how can i log cdr @ mysql? |
| 13:07.49 | bombaclat667 | ok..will go there. Im going to restart from scratch though....that way it wont be mixed |
| 13:08.26 | kaldemar | vncsnvs: http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc |
| 13:08.49 | vncsnvs | kaldemar, thx! :) |
| 13:09.02 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 13:09.41 | kaldemar | adnc: asterisk's dialplan doesn't handle calls by addresses, but mainly by numbers. how are you planning to dial? |
| 13:10.07 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 13:10.44 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:10.45 | adnc | kaldemar: from a softphone like sip communicator. i've one provider that forwards landline calls for me and iptel. since my landline provider does not forward calls to other networks i would like to have calls like 1000@iptel.org forwarded to iptel.org |
| 13:11.29 | adnc | i can not distinguish between numbers because iptel also has numbered services |
| 13:11.38 | vncsnvs | noone answer @ freepbx, anyone could help me? |
| 13:12.02 | adnc | how do you normaly handle this, many people should have this problem |
| 13:12.27 | *** join/#asterisk waKKu (n=vaKKu@unaffiliated/wakku) |
| 13:12.29 | waKKu | !books |
| 13:12.41 | waKKu | !tfot |
| 13:12.42 | kaldemar | adnc: configure iptel to the client itself |
| 13:12.50 | waKKu | anybody, please ? link for tfot |
| 13:12.58 | kaldemar | ~book |
| 13:13.00 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 13:13.00 | adnc | kaldemar: no other way? |
| 13:13.14 | waKKu | tks |
| 13:13.15 | *** part/#asterisk waKKu (n=vaKKu@unaffiliated/wakku) |
| 13:14.47 | kaldemar | adnc: check what the client actually sends to asterisk. if it manages to send a possibly retarded sip uri of form 1000@iptel.org@asterisk and asterisk parses that the way you want it to, maybe you could make a dialplan pattern to match it. go try. |
| 13:15.21 | adnc | to match what? |
| 13:15.29 | adnc | thats the problem |
| 13:15.48 | kaldemar | but you'll skip a few headaches if you configure your client so that it recognizes entered uri's right. |
| 13:16.03 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:16.40 | kaldemar | exten _. <--- but that is dangerous and should not be used as it matches to special extensions too. you could pad the . with X's for numbers it you know how long the numbers are. |
| 13:16.46 | adnc | my idea was to hook up my client, asterisk should manage the forwarding of the call, so doesnt it make sence to manage it at asterisk side |
| 13:17.53 | kaldemar | then test if the last 10 chars of EXTEN are @iptel.org and proceed accordingly. but still, i won't recommmend that. |
| 13:18.18 | kaldemar | no, it makes sense to configure clients and gateways/proxies the way threy're supposed to. |
| 13:19.43 | adnc | kaldemar: you are suggesting not to make such rules, instead configure the clients to connect directly to iptel.org? |
| 13:20.33 | kaldemar | yes |
| 13:20.49 | *** join/#asterisk troubled_ (n=troubled@unaffiliated/troubled) |
| 13:21.30 | [TK]D-Fender | kaldemar: Lot of supposition thre and we've yet to see any call debug... I'd wait on it.. |
| 13:21.51 | kaldemar | adnc: or direct calls based on numbers |
| 13:22.18 | yondaime | hey what's hardware I need to learn asterisk? |
| 13:22.19 | kaldemar | [TK]D-Fender: a huge amount of it. but as i told him earlier, "go try". |
| 13:22.50 | [TK]D-Fender | yondaime: a PC running an OS that you can sanely install * on. Thats it |
| 13:22.53 | adnc | kaldemar: but what if someone wants to reach me via user@domain.tld or i want to make such a call |
| 13:23.15 | yondaime | [TK]D-Fender, freebsd here |
| 13:23.28 | [TK]D-Fender | adnc: "allowguest=yes" , "context=miscsipcalls" in [general] |
| 13:23.33 | [TK]D-Fender | yondaime: That'll do |
| 13:24.35 | adnc | [TK]D-Fender: yes, thats no problem, but how does the dialing pattern need to look in order to forward such a request to the correct provider like iptel.org |
| 13:25.01 | [TK]D-Fender | adnc: Depends if they need calls to them being auth'd or not |
| 13:25.19 | adnc | mhh |
| 13:25.46 | adnc | are you reachabel via an address like tk@yourdomain.tld |
| 13:25.46 | [TK]D-Fender | adnc: if they allow un-authed randome calls you could just do Dial(SIP/1234@iptel.org) |
| 13:25.54 | [TK]D-Fender | adnc: Yes I am |
| 13:26.26 | adnc | and you probably also do have a provider that forwards to landlines, and for this you need to have a dialpattern that can distinguish between them |
| 13:27.03 | [TK]D-Fender | adnc You have now mixed how I RECEIVE calls and how I PLACE calls up |
| 13:27.19 | adnc | yes, right |
| 13:27.30 | [TK]D-Fender | adnc: And stop using the word "forward" <- it does not apply |
| 13:27.34 | *** join/#asterisk path_ (n=path@92-106-21-190.adsl.terra.cl) |
| 13:27.46 | adnc | [TK]D-Fender: which word does apply correctly? |
| 13:28.36 | [TK]D-Fender | adnc: My * gets a call of type X (Zaptel, SIP ITSP, unauthed misc SIP call, etc), I then do whatever I want with it. I might DIAL out to some other resource, maybe a SIP phone on my desk, maybe to the PSTN via some resource I have access to, but it is not "forwarding" |
| 13:29.16 | vncsnvs | done |
| 13:29.18 | [TK]D-Fender | adnc: The correct term is "just dial" and then a noun... preferably one that implies what kind of call you're placing :) |
| 13:29.34 | vncsnvs | there is a way to configure SNMP + * ? |
| 13:29.55 | [TK]D-Fender | vncsnvs: http://www.google.ca/search?hl=en&q=asterisk+snmp&btnG=Google+Search&meta= |
| 13:30.06 | vncsnvs | [TK]D-Fender, thx! :) |
| 13:30.12 | vncsnvs | aheuiaehaehiauehieaa |
| 13:30.13 | vncsnvs | lol |
| 13:34.47 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:35.47 | [TK]D-Fender | jaytee: Sobered up you you lush? :p |
| 13:36.28 | jaytee | [TK]D-Fender, morning! yeah, I'd hope in 48 hours I'd be sober. My liver isn't that weak. |
| 13:36.52 | [TK]D-Fender | jaytee: We'll I simply avoided the assumption that it wasn't a chronic cycle ;) |
| 13:37.06 | jaytee | I woke up feeling fine yesterday in fact, other than my back still hurting from slipping on ice on the weekend. |
| 13:37.39 | jaytee | no, I rarely drank until I started using Asterisk :-) |
| 13:37.50 | *** join/#asterisk bahjons (n=robert@140.99.23.26) |
| 13:37.54 | [TK]D-Fender | jaytee: So... I have wiped my IBM server that ran Novell, and upped to CentOS 5.2 on it and ported my Samba config over and am DONE. w00t. |
| 13:41.19 | jaytee | [TK]D-Fender, way to go! |
| 13:51.31 | [TK]D-Fender | jaytee: Old backup speed = 680mb/s, new 1800mb/s, almost TRIPLE on the same hardware |
| 13:53.01 | adnc | [TK]D-Fender: in order for you to be reachable via a mail adress do you use dns srv records or are there more elegant ways? |
| 13:53.21 | *** join/#asterisk DarkRift (n=dark@65.92.169.237) |
| 13:53.22 | jaytee | [TK]D-Fender, that's great! I've heard people in the past complaining about throughput speeds on samba shares but I've never run into it myself and I just use a basic config. I suspect in the cases I've heard people whine it's some kind of flaky network card or crappy disk controller hardware. |
| 13:53.23 | [TK]D-Fender | adnc: it is NOT an "e-mail" address it is a SIP URI |
| 13:53.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:53.54 | [TK]D-Fender | adnc: extension@host. Nothing more. |
| 13:54.05 | [TK]D-Fender | adnc: It ain't Raw-Cat Science. |
| 13:54.13 | mphill_ | ouch |
| 13:54.34 | adnc | [TK]D-Fender: i was talking about username@domain.tld like his mail adress |
| 13:54.51 | [TK]D-Fender | adnc: "looks" like. Isn't. |
| 13:55.01 | adnc | i suppose it is then a simple alias |
| 13:55.19 | [TK]D-Fender | adnc: exten => jondoe,1,NoOp(OMFG) |
| 13:55.39 | [TK]D-Fender | adnc: Noone said it had to be a NUMBER. |
| 13:55.55 | adnc | [TK]D-Fender: ohh, beautiful |
| 13:58.13 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279393977.dsl.bell.ca) |
| 14:01.21 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 14:01.29 | *** part/#asterisk bahjons (n=robert@140.99.23.26) |
| 14:03.38 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 14:03.53 | *** join/#asterisk pruonckk (n=mike@firewall-spo-1.intranet.braslink.com) |
| 14:04.07 | *** join/#asterisk keebler (n=keebler@h149.148.20.98.dynamic.ip.windstream.net) |
| 14:04.22 | *** part/#asterisk pruonckk (n=mike@firewall-spo-1.intranet.braslink.com) |
| 14:04.30 | keebler | Has anyone been able to use a WBP54G with their ATAs? |
| 14:08.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:09.40 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:10.01 | fskrotzki | RW1gk1112 |
| 14:13.56 | defswork | anyone has a nokia n810 working with asterisk ? Mine used to but since I upgraded the firmware I can't get it working again |
| 14:20.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.57 | Zeeek | Hey TK: Sony's latest gadget is bound to interest you: http://www.youtube.com/watch?v=8AyVh1_vWYQ |
| 14:25.10 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 14:25.55 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:26.44 | *** join/#asterisk ingenius (n=alektro@host174.201-253-221.telecom.net.ar) |
| 14:29.51 | *** join/#asterisk emiller (n=ed@static-98-117-7-38.hrbgpa.fios.verizon.net) |
| 14:30.46 | emiller | I had a user record their own 'unavailable' voicemail recording, but they have been let go; is there a way to reset their voicemail greetings back to the default asterisk voice? |
| 14:32.16 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:250:daff:fe7c:a503) |
| 14:33.59 | *** join/#asterisk waKKu (n=vaKKu@unaffiliated/wakku) |
| 14:34.03 | waKKu | hi folks.. afternoon.. |
| 14:34.56 | waKKu | could someone help me with this problem: http://rafb.net/p/i1d8pF22.html - I'm not able to receive calls on a SIP trunk... and i was able at morning, and still able if registering sip using a softphone |
| 14:35.57 | waKKu | sip.conf + extensions.conf : http://rafb.net/p/IpW59v72.html |
| 14:36.19 | waKKu | PS: i'm able to call using sip trunk.. i cant only receive calls |
| 14:36.24 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:36.39 | emiller | wakku: i had this problem last week |
| 14:36.56 | waKKu | emiller any hint ? |
| 14:37.01 | emiller | are you still registered with your provider? |
| 14:37.07 | emiller | sip show registry |
| 14:37.08 | waKKu | emiller yes.. |
| 14:37.18 | emiller | our problem was Level 3 |
| 14:37.26 | waKKu | vono.net.br:5060 angraconstr 115 Registered |
| 14:37.29 | emiller | apparently they were having issues on their end. |
| 14:37.37 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 14:37.43 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 14:38.32 | waKKu | emiller hm.. but I can receive calls if registering into a softphone |
| 14:38.32 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:38.52 | emiller | hmm, you can receive calls on your softphone, but not on physical phones? |
| 14:38.52 | *** join/#asterisk eric2 (n=ejc@dsl-66-185-205-254.vianet.ca) |
| 14:38.52 | lbt_ | I just ported the UK CLI patch to 1.6.1-rc1 and dahdi-2.1.0.4 .... and it seems to work OK |
| 14:39.50 | waKKu | emiller not on asterisk |
| 14:40.12 | waKKu | if I configure sip trunk on to softphone, it works perfectly |
| 14:40.24 | zeeesh | error- after disconnecting the call both end disconnected and after 1 or 2 second i m receiving again ring at dialed number ,,, getting error at console "Remote host can't match request CANCEL to call '503a773b4728b4300689a311457b4c3c@203.215.179.38'. Giving up."? |
| 14:40.41 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 14:40.42 | telnettech | when making a change in the zapata file, what is the proper procedure to make the change take affect? |
| 14:41.31 | jaytee | module reload chan_zap.so |
| 14:41.46 | telnettech | ok thats what i thought thanks jaytee |
| 14:41.54 | jaytee | that's only for zapata |
| 14:42.00 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-376ae6f8297f3f96) |
| 14:42.11 | jaytee | if you change zaptel you need to restart the zaptel service |
| 14:42.27 | telnettech | right i am making a change on the zapata.conf file only |
| 14:42.55 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 14:43.05 | dlynes | ~seen cunningpike |
| 14:43.16 | jbot | cunningpike <n=arodgers@vpn.dnv.org> was last seen on IRC in channel #asterisk, 28d 14h 54m 35s ago, saying: 'Is voip-info kaput?'. |
| 14:43.16 | *** join/#asterisk Khratos (n=khratos@190.166.103.112) |
| 14:43.17 | dlynes | !seen cunningpike |
| 14:43.29 | dlynes | wow....guess he went on vacation :) |
| 14:43.35 | Khratos | good $TIMEOFDAY |
| 14:44.01 | jaytee | dlynes, I'm sure I saw him come in here yesterday or the day before. |
| 14:44.36 | dlynes | jaytee: under a different nick? |
| 14:44.42 | talirk81 | I have "include LevelCall/*.conf " in my extensions_custom.conf file. with a file Handler.conf setting up a context, but when asterisk try to GoTo() that context it just dies. Do astersik not support * in includes? |
| 14:44.58 | dlynes | jaytee: Just looking for him because I might have some consulting work for him |
| 14:45.29 | jaytee | dlynes, pretty sure it was the same nick although it might have had an underscore at the end. I remember seeing his nick joining within the last couple days. |
| 14:45.46 | dlynes | !seen cunningpike_ |
| 14:45.55 | dlynes | ~seen cunningpike_ |
| 14:45.57 | jbot | cunningpike_ <n=arodgers@S010600095b33697f.vc.shawcable.net> was last seen on IRC in channel #asterisk, 93d 10h 48m 1s ago, saying: 'How are you?'. |
| 14:45.59 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 14:46.08 | dlynes | Guess he's got another nick yet, :) |
| 14:46.18 | jaytee | maybe |
| 14:46.25 | dlynes | ~seen cunningpike__ |
| 14:46.26 | jbot | i haven't seen 'cunningpike__', dlynes |
| 14:46.32 | jaytee | ~seen jaytee |
| 14:46.32 | jbot | jaytee is currently on #asterisk (1h 11m 45s). Has said a total of 12 messages. Is idling for 0s, last said: '~seen jaytee'. |
| 14:46.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:46.39 | dlynes | ~seen _cunningpike |
| 14:46.39 | jbot | i haven't seen '_cunningpike', dlynes |
| 14:46.42 | dlynes | hrm |
| 14:46.49 | dlynes | ~seen cunning_pike |
| 14:46.49 | jbot | dlynes: i haven't seen 'cunning_pike' |
| 14:46.56 | dlynes | shurgs. |
| 14:47.15 | jaytee | thought maybe jbot's record keeping wasn't up to date since it was offline for a couple days a week or so ago |
| 14:47.48 | dlynes | jaytee: ah |
| 14:47.55 | dlynes | I haven't been on here regularly for a while |
| 14:48.08 | dlynes | Been too busy with other stuff |
| 14:48.22 | dlynes | Haven't been doing asterisk 24 hours a day any more either |
| 14:51.00 | beek | morning guys. |
| 14:51.10 | yondaime | I have a problem compiling zaptel /kernel/ztdummy.c:202: error: ‘struct hrtimer’ has no member named ‘expires’ do anyone know if it require any dep to fix it? |
| 14:53.17 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:55.20 | *** join/#asterisk [gnubie] (n=[gnubie]@119.56.59.6) |
| 14:55.53 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 14:56.05 | [gnubie] | i can't seem to start the asterisk v1.4.21.2~dfsg-1 here |
| 14:57.42 | [gnubie] | this used to work |
| 14:58.11 | [gnubie] | executing "/etc/init.d/asterisk start" does not the service |
| 14:58.26 | [gnubie] | but "asterisk -cvvv" works |
| 14:59.39 | murdock_ut | eric2: So what was the problem? |
| 15:00.05 | [gnubie] | how will i know what causes the problem? |
| 15:00.10 | *** join/#asterisk prxtien (n=pro@ppp121-45-82-113.lns10.adl6.internode.on.net) |
| 15:00.23 | prxtien | im using 1.6.0.5, how can i make asterisk confirm if a file in the file system exists |
| 15:00.39 | prxtien | i want calls to be routed to voicemail if my house is in 'movie' mode |
| 15:00.54 | prxtien | i want it to act one way if the file exists, another if it doesnt |
| 15:01.11 | *** part/#asterisk waKKu (n=vaKKu@unaffiliated/wakku) |
| 15:01.37 | *** part/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 15:03.35 | dlynes | prxtien: is the file you're looking for a pid file created by a movie player? |
| 15:04.19 | beek | prxtien: Use the "SYSTEM" application to call a quick script to look for the file and return success/failure. |
| 15:04.22 | dlynes | [gnubie]: check your asterisk log (/var/log/asterisk/full) |
| 15:04.55 | [gnubie] | dlynes: i don't have that log |
| 15:05.31 | dlynes | [gnubie]: check your /etc/asterisk/logger.conf file...you probably have the full log commented out (it's usually commented out, by default) |
| 15:06.05 | dlynes | [gnubie]: in the absence of that file, /var/log/asterisk/messages might have enough information to determine what went wrong |
| 15:06.51 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:07.24 | [gnubie] | [Feb 11 23:04:57] ERROR[30243] logger.c: Unable to create event log: Permission denied |
| 15:07.26 | Gido-E | howto see the current amount of calls in Asterisk 1.4? |
| 15:07.38 | eppigy | hello |
| 15:07.40 | eppigy | i am dave |
| 15:07.44 | Whitor | Hi. What config file is responsible for the options available when an IVR is handling a call? |
| 15:08.08 | dlynes | [gnubie]: that's not a huge issue, but then you've gotta wonder why it wasn't able to create the log, too |
| 15:08.10 | eppigy | both of you need to read some basic stuff |
| 15:08.16 | eppigy | basicaly |
| 15:08.33 | eppigy | Gido-E: type help at the CLI |
| 15:08.35 | dlynes | [gnubie]: like maybe you don't have correct permissions on the /var/log/asterisk directory, or wrong ownership, or you're out of disk space, or ... |
| 15:08.36 | eppigy | and figure it out |
| 15:08.39 | prxtien | beek, yeah but how do you audit that |
| 15:08.42 | Gido-E | eppigy not the channels, the amount of calls! |
| 15:08.50 | prxtien | dlynes, nah it isnt in this case mate |
| 15:08.52 | eppigy | !!!!!! |
| 15:09.02 | [gnubie] | dlynes: drwxr-xr-x 4 asterisk asterisk 4096 2009-02-11 23:05 asterisk |
| 15:09.03 | eppigy | it says how many calls |
| 15:09.11 | dlynes | prxtien: ok, so the movie in this case is running on your entertainment center? |
| 15:09.23 | prxtien | beek, i can call the system script no worries but, how do i know if it returns 0 or 1 |
| 15:09.26 | dlynes | prxtien: i.e. an external source that's not connected to the asterisk box |
| 15:09.31 | prxtien | dlynes: yeah thats right |
| 15:09.42 | dlynes | prxtien: why not just use a database variable, then? |
| 15:09.47 | prxtien | so i am touching a file, to indicate the state of the system |
| 15:09.52 | Gido-E | eppigy which command? |
| 15:09.55 | beek | prxtien: man bash ; look at the -x test; use that as an exit code. |
| 15:10.03 | dlynes | prxtien: makes everything easier, and you don't have to subject the asterisk box to the load of a system call |
| 15:10.36 | prxtien | mm its a possibility i guess, would rather not involve a database if i dont need |
| 15:10.51 | beek | prxtien: The database is already in Asterisk... |
| 15:10.54 | dlynes | prxtien: you don't understand...I meant the asterisk database, not mysql, ... |
| 15:11.08 | prxtien | beek, i understand that part but, when asterisk gets its return code.. how does it do one thing for 0 and another for 1 |
| 15:11.12 | Gido-E | howto see the current amount of calls in Asterisk 1.4? |
| 15:11.23 | dlynes | Gido-E: show channels |
| 15:11.24 | *** join/#asterisk medjr (n=medjr@41.226.129.49) |
| 15:11.27 | prxtien | dlynes, but asterisk isnt starting the action, so how do i get the variable into th asterisk db in the first place? |
| 15:11.38 | beek | prxtien: show application system on your asterisk console. That should give you the hint. |
| 15:11.38 | Gido-E | that is not the ammount of calls. |
| 15:11.43 | dlynes | prxtien: asterisk -rx "database set varname value" |
| 15:11.49 | prxtien | oh |
| 15:11.54 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:11.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:12.13 | prxtien | well thats a totally different story then |
| 15:12.22 | prxtien | ! |
| 15:12.32 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 15:12.47 | dlynes | prxtien: I think that's what it was, anyways...I've only ever used the asterisk database a couple of times |
| 15:12.50 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-8e76de0be7be790d) |
| 15:12.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:12.53 | medjr | hi all |
| 15:13.12 | medjr | i'm very new to asterisk and i need some help |
| 15:13.38 | dlynes | prxtien: nope... asterisk -rx "database put <family> <key> <value>" |
| 15:13.53 | *** join/#asterisk Wangster (n=Wangster@host-253.epicnet.ca) |
| 15:14.09 | medjr | does anybody here speak french btw ? |
| 15:14.37 | dlynes | prxtien: so.... asterisk -rx "database put /MOVIE/mode on 1" |
| 15:14.56 | dlynes | prxtien: and then change the 1 to a 0 to turn it off |
| 15:15.21 | prxtien | yeah thats perfect mate just looking at it now |
| 15:15.41 | prxtien | thats even better, then asterisk always knows my house state |
| 15:15.46 | prxtien | and can route calls accordingly |
| 15:16.12 | dlynes | prxtien: actually....don't need the '/' in front of MOVIE |
| 15:16.24 | Wangster | I have a strange problem, my asterisk box won't play back any internal sounds, not even things like milliwatt. Mainly this is effecting voicemail. It doesn't show any errors on the console. Any idea what this could be? I tried and then backed out of an upgrade from 1.4.21.2 to 1.4.23.1 last night and thats when it stopped working. |
| 15:16.49 | dlynes | prxtien: i think it's a much more elegant, and flexible method than using system calls, anyways |
| 15:17.03 | prxtien | yeah i like that |
| 15:17.15 | beek | prxtien: I'd agree. |
| 15:17.17 | prxtien | /usr/sbin/asterisk -rx "database put x10 statecurr `cat /etc/heyu/state/state.curr`" |
| 15:17.22 | prxtien | will suite my needs perfectly |
| 15:17.28 | *** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 15:18.02 | dlynes | prxtien: there ya go |
| 15:18.06 | prxtien | thanks guys |
| 15:18.18 | dlynes | prxtien: and you get the added benefit of being able to script it using perl, python, bash, whatever your heart desires |
| 15:18.35 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-a5547c16a692cd3d) |
| 15:18.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:19.02 | dlynes | prxtien: or even microsoft scripting language, if you're stuck on windows :) |
| 15:19.16 | prxtien | dlynes, take that back! |
| 15:19.32 | beek | holds cross out at dlynes and screams "Out, Satan!" |
| 15:19.41 | *** join/#asterisk JerJer (n=jmcnamar@c-71-207-221-127.hsd1.al.comcast.net) |
| 15:22.19 | *** join/#asterisk Juggie (n=kavando@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:24.12 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 15:29.17 | *** join/#asterisk laggo (n=user@80.175.142.209) |
| 15:29.26 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 15:29.26 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:30.59 | *** join/#asterisk savage1 (n=savageon@static-66-212-194-134.cpe.metrocast.net) |
| 15:31.09 | savage1 | is anyone using snmp to monitor asterisk? |
| 15:31.26 | savage1 | I see this info: Under debian, get snmpd snmp libsnmp9-dev libsnmp9 for adding it so I can monitor asterisk via hyperic |
| 15:31.30 | savage1 | but, I'm running centos |
| 15:31.37 | savage1 | anyone have any experience with this? |
| 15:32.53 | laggo | i'm trying to install asterisk-addons for the mysql cdr addon, but make doesnt seem to be doing anything to do with mysql, just format_mp3 and chan_ooh323. any suggestions? |
| 15:33.34 | codefreeze-lap | laggo: check make menuselect, and make sure the cdr_mysql is turned on; |
| 15:33.36 | lowtek | laggo: You have to have the mysql-client-dev tools installed. |
| 15:33.46 | lowtek | laggo: what distro? |
| 15:33.52 | laggo | centos |
| 15:34.01 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:34.22 | codefreeze-lap | laggo: and as lowtek says, make sure the depended-on libs are installed |
| 15:34.24 | *** join/#asterisk cesau (n=cesau@66.94.94.66) |
| 15:34.31 | cesau | morning #asterisk! |
| 15:34.33 | laggo | ah ok i did make menuselect and im obviously missing some dependency for mysql |
| 15:34.35 | laggo | thanks for that |
| 15:34.45 | madgeek | morning cesau |
| 15:34.51 | lowtek | laggo: can't help you there, on debian you would install libmysqlclient15-dev which would install the libs |
| 15:35.44 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 15:35.45 | jaytee | i wrote and advanced Powershell script that would do MS SQL replication and exports for backups on a scheduled basis. It ended up becoming "self-aware" and RDP'd into my home computer and ate my homework for a college course I was taking nights. |
| 15:36.06 | cesau | if a sip qualify yields "unreachable", does that indicate traffic out isnt working or does it indicate either traffic out _or_ traffic in isnt working? |
| 15:36.33 | lowtek | jaytee: /usr/local/sbin/skynet.sh |
| 15:36.37 | Gido-E | laggo did you already do: yum install -y gcc ncurses-devel libtermcap-devel kernel-devel kernel-smp-devel gcc-c++ openssldevel |
| 15:36.45 | Zeeek | jaytee that was before it ran away with your significant other? |
| 15:37.06 | cesau | jaytee: hah |
| 15:37.16 | laggo | Gido-E: im RTFM'ing |
| 15:38.01 | lowtek | jaytee: What TIME did it become self aware, please, tell us the time, this is our last ch @$(@# NO CARRIER |
| 15:38.08 | eric2 | murdock_ut: can't use context's when configuring the phone, only the extension number |
| 15:39.33 | murdock_ut | Correct. |
| 15:40.39 | *** join/#asterisk l2trace99 (n=asd@p1-bh-mco-1.prismone.net) |
| 15:42.04 | telnettech | got a problem with my dialplan: http://pastebin.com/da31bdf9 |
| 15:42.45 | telnettech | If I put the WaitExten statement right after the background, then anyone putting digits into the system while the sound file is plying errors out and doesnt work |
| 15:43.17 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 15:43.26 | telnettech | the error that is on the pastebin, I get if there is no digit entered while it is timing out and transferring to where i need it to go to |
| 15:44.01 | telnettech | any ideas |
| 15:44.11 | Gido-E | swap, s,2 and s,3 |
| 15:44.15 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.66.155) |
| 15:44.54 | laggo | hooray dependency crisis resolved |
| 15:45.18 | *** join/#asterisk JerJer (n=jmcnamar@router.asteriasgi.com) |
| 15:45.24 | telnettech | Gido: If i do that, then anyone that enters digits while the sound file is playing cant reach the extension they dial |
| 15:45.47 | telnettech | and the timeout is then invoked and they are transferred to front desk operators |
| 15:46.01 | telnettech | anyways, which is what we are trying to avoid |
| 15:46.30 | jpcansa | anybody knows what this means: "-- SIP/1200-b6402020 requested special control 16, passing it to Zap/1-1" ?? thx in advance |
| 15:47.34 | *** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime) |
| 15:48.57 | laggo | is there any point in logging CDR to both CSV and MySQL? |
| 15:49.42 | codefreeze-lap | laggo: that's up to you |
| 15:51.15 | laggo | just trying to think of reasons why mysql would fail |
| 15:52.12 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 15:52.18 | telnettech | anybody else have any ideas about my dialplan problem? |
| 15:53.47 | telnettech | I mean it is working like we want but we want to be able to reduced the numberof any error or warning messages we get in our logs. Our support structure is being moved offshore to Singapore to people that have no idea how our product works |
| 15:55.09 | *** join/#asterisk seanmh (i=seanmh@216.31.101.27) |
| 15:56.36 | Whitor | How do I enable off site calling from an IVR ? I keep getting a recording "I'm sorry, That is not a valid extension. Please try again" Any ideas ? |
| 15:57.20 | lowtek | Whitor: What do you mean by "off site calling", more details please ... |
| 15:57.31 | jaytee | telnettech, just get rid of the WaitExten and use Set(TIMEOUT(digits)="large enough interval for slow witted fools in seconds") as your first priority. |
| 15:57.58 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:58.05 | telnettech | jaytee: before the background priority? |
| 15:58.37 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.112.83) |
| 15:58.45 | Whitor | I have 2 systems connected via an IAX2 trunk System A and System B. System A has a ZAP trunk too.... when a person calls in on the zap trunk, they get the IVR. I want to dial an extension on System B... but the IVR is only accepting extensions for the local system |
| 15:59.09 | Whitor | System A |
| 15:59.10 | jaytee | telnettech, yes, before the background app runs, FIRST PRIORITY |
| 15:59.20 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:59.28 | telnettech | k thanks |
| 16:01.22 | Whitor | System B is on a remote network... phones on system A and B can call each other directly |
| 16:01.42 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 16:01.54 | jaytee | telnettech, try not to make the value too high. I wouldn't go over 10 seconds. If they can't dial at least 4 or 7 digits in that period of time they need to be culled from the gene pool |
| 16:02.31 | telnettech | i have it set as 5 which i feel is enough for a 4 digit number |
| 16:04.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:05.10 | telnettech | anyway to see CDR on the CLI? |
| 16:05.28 | laggo | when i call my asterisk context, i always have to pad the beginning of the call with Wait() for a few seconds. if i just Answer() and Playback() right away, the sound gets cut off - does everybody have this problem? |
| 16:06.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:08.05 | carrar | laggo, need to give everything time to get into the connect state |
| 16:08.15 | carrar | thats typical with telcos |
| 16:08.22 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:08.40 | laggo | carrar: i thought so, just wondering why all tutorials have just Answer() and Playback() :) |
| 16:08.57 | carrar | telnettech, like "core show channels verbose"? |
| 16:09.00 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 16:09.17 | carrar | sip to dialplan might be fast enough |
| 16:09.30 | carrar | depending on how it's connected and the phone type |
| 16:10.00 | telnettech | carrar: sorry should have included that I have 1.2.28 version |
| 16:10.08 | carrar | for calls coming in from providers I have to wait typically |
| 16:10.24 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:10.29 | carrar | telnettech, there are no "view cdr" commands |
| 16:10.36 | carrar | other then viewing the active calls |
| 16:10.44 | telnettech | ok thanks |
| 16:12.29 | carrar | Whitor, just point the extension range to the other box |
| 16:12.46 | carrar | or use dundi |
| 16:13.20 | carrar | sips on his coffee |
| 16:14.44 | Whitor | carrar, in what config file do I do the extension range pointing ? |
| 16:14.49 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:14.54 | Whitor | goes to get coffee |
| 16:15.12 | carrar | in the conext that needs to dial the extension |
| 16:15.31 | carrar | in the ivr context I would guess |
| 16:16.34 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:17.23 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 16:17.49 | carrar | something like exten => _4XXX,1,Dial(IAX2/IAX-link-name/${EXTEN},100) |
| 16:18.21 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:19.17 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 16:24.34 | mphill_ | is the HD voice function of polycom phones something that is strictly phone related or is it a function of something asterisk has to support? |
| 16:24.55 | carrar | phone related |
| 16:25.03 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.04 | carrar | however asterisk can pass that codec through |
| 16:25.28 | mphill_ | carrar: is hd voice worth the extra money? |
| 16:25.31 | carrar | I don't think it can terminate it yet |
| 16:25.45 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 16:25.45 | carrar | it is if you do a lot of interoffice calls |
| 16:25.51 | carrar | and all the phones are 72 |
| 16:25.53 | carrar | 722 |
| 16:26.11 | carrar | but if most of your calls are outbound then no |
| 16:26.33 | carrar | or unless they just want a nice fancy phone |
| 16:26.39 | carrar | which most execs do |
| 16:26.44 | *** join/#asterisk aut (n=aut@modemcable207.231-201-24.mc.videotron.ca) |
| 16:27.10 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:27.14 | aut | hi, how accurate is the latency shown in "sip show peers"? im finding that my softphone users show about 100ms more latency there than my hardware atas |
| 16:27.25 | aut | and that corresponds to about 100ms more than if i were to ping them |
| 16:27.32 | carrar | not very accurate |
| 16:27.52 | mphill_ | carrar: off the top of your head the polycom 501 and 601 can not do hd voice, correct? |
| 16:28.01 | aut | whereas the hardware ata latency pretty much matches what i get when i ping it |
| 16:28.03 | carrar | correct, they can not |
| 16:28.26 | carrar | 601 is however a very nice phone |
| 16:28.44 | aut | carrar: do you know if softphones are noticably slower (because of the soundcard or whatever)? |
| 16:29.09 | carrar | I don't, but I would think they might be due to the computer part |
| 16:29.35 | carrar | I wouldn't worry about the reading too much |
| 16:29.44 | carrar | just as long as it's not like 500 |
| 16:29.45 | carrar | heh |
| 16:30.22 | madgeek | the 601 just has the expansion modules... |
| 16:30.26 | madgeek | has a 501 on his desk |
| 16:30.42 | madgeek | gets the job done for the day to day |
| 16:30.46 | carrar | yeah |
| 16:30.51 | carrar | I used 3 on that phone before |
| 16:30.55 | carrar | pretty nice |
| 16:32.30 | Zeeek | as far as I've seen, softphones are not slower in any way than hardphones |
| 16:32.38 | carrar | pign wise |
| 16:32.40 | carrar | ping |
| 16:32.45 | Zeeek | The ip450 does HD Voice |
| 16:32.52 | madgeek | i've seen probalems with them on shitty switches |
| 16:32.53 | carrar | I haven't done any side by side tests |
| 16:33.04 | lowtek | Soft phones suck. |
| 16:33.08 | carrar | yeah |
| 16:33.14 | Zeeek | my anecdotal comment is that I've used both for years and don't see any diff |
| 16:33.15 | carrar | but if thats all ya got |
| 16:33.21 | carrar | or can afford |
| 16:33.28 | Zeeek | and they're both useful |
| 16:33.45 | Zeeek | if you are already sitting at a computer and need features, a softphone is fine |
| 16:34.01 | Zeeek | anywhere else, it's a PITA to have to always have a computer running |
| 16:34.41 | Zeeek | I'venever had a USB phone whioch combines the DISadvantages of both hard and soft :) |
| 16:35.19 | lowtek | Zoiper is OK, but I would love to find a soft phone + mic that would reproduce the quality of even a cheap polycom. |
| 16:35.52 | carrar | I like the xlite/logitech USB headset combo |
| 16:36.05 | carrar | seems to work flawlessly |
| 16:36.17 | sub | i'm using zoiper on my macbook, i couldn't get nat traversal to work with xlite |
| 16:36.24 | lowtek | Which logitech USB? I have their high end one and a cheap one, the high end one works great but it has those earphones that cover your ears and isn't really usable long term in an office. |
| 16:36.30 | carrar | I use xlite on all my OSX boxes |
| 16:36.52 | Zeeek | love Xlite |
| 16:37.01 | lowtek | sub: Are you sure your problem wasn't that X-Lite was trying to connect via SRV record instead of directly to asterisk? |
| 16:37.16 | lowtek | carrar: Which headset are you using? |
| 16:37.20 | Zeeek | and we have Jason from Counterpath live next Friday 20th feb on the VUC |
| 16:37.23 | carrar | lawtek, no model number on it, but it's a stereo headset with a boom mic |
| 16:37.25 | sub | it runs fine, i like the UI a bit more than zoiper's, but my i don't actually know the password to my router so i can't forward any ports and was having issues registering (all would timeout) |
| 16:37.26 | *** join/#asterisk sheri_rao (n=sheri_ra@115.186.130.188) |
| 16:37.49 | sub | lowtek: not sure, i was using my asterisk's IP address only |
| 16:37.53 | lowtek | sub: right, I bet it's really trying to find the host via SRV record ... |
| 16:38.05 | sub | hmm |
| 16:38.25 | sub | i could actually see the registrations on asterisk console |
| 16:38.34 | lowtek | sub: I had this issue and was able to fix it by *I think* putting my asterisk server(s) hostname in the proxy setting, forget the actual dialog verbage |
| 16:39.14 | sub | hmm, i'll have to give that a shot |
| 16:39.29 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 16:40.13 | lowtek | sub: It's "send outbound via" "proxy" |
| 16:40.17 | *** part/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 16:40.34 | Zeeek | carrar: I agree Xlite+USB (logitec or plantronics) will give great sound unless the sound card is crap |
| 16:40.35 | carrar | lawtek, Looks like it's the Logitech Stereo USB Headset 200 |
| 16:41.04 | lowtek | plantronics makes a usb headset? cool, will look, thanks |
| 16:41.08 | sub | ahhh i had tried that |
| 16:41.18 | Zeeek | Mine is a DSP400 |
| 16:41.21 | sheri_rao | can anyone test my voip termination system |
| 16:41.27 | carrar | I use a plantronics in my Cisco phone headset jack |
| 16:41.40 | carrar | I love that setup |
| 16:41.52 | Zeeek | I have a plantronics for my Polycom but it doesn't work so well (unamplified) |
| 16:42.31 | sheri_rao | can anyone check my voip termination system |
| 16:42.58 | carrar | whats wrong with it? |
| 16:43.17 | sheri_rao | i want someone forward terrific on my asterisk server |
| 16:43.28 | carrar | sounds scary |
| 16:44.05 | sheri_rao | it is not scary |
| 16:44.33 | carrar | How do you want to send traffic to it? |
| 16:45.03 | sheri_rao | i have IP address and i need to know your IP address so i can add yours in my white list |
| 16:45.37 | carrar | I think Digium has a echo back test you can use |
| 16:45.42 | carrar | or someone does |
| 16:46.02 | sheri_rao | tel me the address of echo test |
| 16:47.31 | mvanbaak | it's in the default configs |
| 16:47.53 | mvanbaak | in your sourcetree, have a look at configs/extensions.conf.sample |
| 16:48.22 | mphill_ | carrar: one more polycom question, the phones support TLS. Is that supported with asterisk 1.6 it looks like. Have you used this at all? |
| 16:48.33 | sheri_rao | carrar, thanks |
| 16:48.44 | carrar | I haen't use TLS with polycoms |
| 16:48.59 | Dovid | anyone have issues only with the Cisco 7940 ? |
| 16:49.06 | carrar | I only us polycom in controlled enviroments |
| 16:49.10 | carrar | since they use ftp |
| 16:49.14 | Dovid | all other phones work with NAT excpet the Cisco 7960 |
| 16:49.25 | carrar | 7940 work awesome |
| 16:49.33 | lowtek | Dovid: Use SIP 8.2 and your NAT problems will go away with the 7940/7960 |
| 16:49.36 | Dovid | I get one way audio with it on inbond calls. |
| 16:49.38 | carrar | 7940/60 are the same |
| 16:49.41 | carrar | configwise |
| 16:49.49 | Dovid | lowtek: thanks I will try that |
| 16:49.56 | Dovid | just 8.2 ? |
| 16:50.00 | lowtek | Yep. |
| 16:50.03 | Dovid | wuts special about that version ? |
| 16:50.06 | carrar | I don't use 8.2 |
| 16:50.20 | lowtek | I'm sure others work to some extent, but 8.2 is by far the most stable with asterisk IMHO. |
| 16:50.25 | carrar | nat works fine with other versions also |
| 16:50.31 | carrar | least with 7.4 |
| 16:50.36 | Dovid | carrar: What version r u using ? |
| 16:50.49 | eric2 | any aastra users here? recommended? |
| 16:51.00 | lowtek | eric2: They are fine phones, but most prefer Polycoms. |
| 16:51.03 | carrar | P0S3-07-4-00 |
| 16:51.12 | lowtek | 8.2 is better than 7.4 |
| 16:51.18 | lowtek | And it's free. |
| 16:51.19 | carrar | how so |
| 16:51.21 | Dovid | eric2: I have a client that is in love with Aastra. let me give him a call |
| 16:51.25 | eric2 | I don't like the look of the polycom's for some reason |
| 16:51.27 | carrar | What am I missing? |
| 16:51.30 | *** join/#asterisk sack (n=sack@208.Red-81-33-111.dynamicIP.rima-tde.net) |
| 16:51.37 | eric2 | I'm using snom for everything so far |
| 16:51.47 | carrar | eric2, I think the cisco are the best looking and feeling |
| 16:51.53 | eric2 | over rated! |
| 16:51.54 | carrar | but the polycom has better features |
| 16:51.57 | eric2 | imho |
| 16:52.10 | lowtek | carrar: The Cisco will only do a single g729 channel, this is a deal killer for us. |
| 16:52.16 | eric2 | which polycom model for office use do you suggest? |
| 16:52.22 | carrar | I only use g.711 |
| 16:52.24 | Dovid | lowtek: Is that a cisco issue ? |
| 16:52.33 | lowtek | Polycom 330 is a great entry level phone, $100 |
| 16:52.48 | Dovid | eric2: 53 55 or 57 was his response |
| 16:52.57 | lowtek | Dovid: yep, those phones were designed in 1999, they don't have the processing power to do two g729 streams. |
| 16:53.04 | carrar | Polycom 430 is nice single user phone also |
| 16:53.12 | carrar | we use 430 for "cheap polycoms" |
| 16:53.21 | sheri_rao | can anyone dial my asterisk server i want to test |
| 16:53.29 | Dovid | lowtek: including the 7940/7960 ? |
| 16:53.42 | Dovid | sheri_rao: Sure. where are you located ? |
| 16:53.46 | carrar | Dovid, you use 729 you need to pay for g729 codec licenses |
| 16:53.54 | eric2 | tx Dovid |
| 16:53.59 | lowtek | Dovid: Yep, specifically them. |
| 16:54.00 | Dovid | carrar: I pay for them. want to know if the cisco has an issue with it |
| 16:54.01 | aut | i find the cisco sounds better to the ear |
| 16:54.03 | aut | not sure why that is |
| 16:54.07 | carrar | ah |
| 16:54.13 | lowtek | Carrar: Doesn't matter what you pay for, the phone simply won't do more than one g729 stream. |
| 16:54.29 | carrar | I don't ever remember having that problem |
| 16:54.40 | Dovid | lowtek: will it roll over for the second call to G711u ? or will it crap out trying to do another G711U ? |
| 16:54.43 | sheri_rao | Dovid, i can give u IP address where u can forward terrific, then u have to tell me yours so it can be allowed from urs |
| 16:54.48 | lowtek | Dovid: precisely |
| 16:54.56 | Dovid | geez |
| 16:55.07 | lowtek | Dovid: Any subsequent channels will be g711. |
| 16:55.50 | Dovid | ah ok. not the end of the workd |
| 16:55.54 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:55.54 | lowtek | Dovid: but I will say, they are *solid* phones running 8.2 and the audio quality is nice. |
| 16:56.13 | lowtek | Dovid: oh, and they look cool |
| 16:56.15 | Dovid | lowtek: thanks. I am going to try that |
| 16:56.18 | sheri_rao | Dovid, please help me |
| 16:56.37 | *** join/#asterisk propellerhead (n=yogurt2u@host135.190-138-101.telecom.net.ar) |
| 16:56.47 | aut | lowtek: do you find the audio quality higher than the polycoms too? |
| 16:57.18 | lowtek | aut: No, I think the polycoms have a much better dynamic range, but the Cisco's seem to be louder and more crisp, but I think this is just personal preference really. |
| 16:57.19 | Qwell | aut: Ciscos? |
| 16:57.20 | Dovid | sheri_rao: what is your IP ? |
| 16:57.21 | aut | been a while since i had my polycom hooked up, but i remember it sounding muted or something like that vs the cisco |
| 16:57.32 | aut | qwell: yea |
| 16:57.34 | Qwell | aut: Cisco licensed their audio stuff *from* Polycom. |
| 16:57.40 | Qwell | So, no, absolutely not better. :) |
| 16:57.43 | aut | qwell: i know, but there's still a difference ! |
| 16:57.51 | *** join/#asterisk esperegu (n=esperegu@145.116.15.244) |
| 16:57.52 | aut | mayb eit's preference, but it's certainly different |
| 16:57.54 | Qwell | Polycoms are amazing. |
| 16:58.01 | aut | maybe it's the speaker (not sure if that's made by polycom) |
| 16:58.08 | lowtek | Qwell: agreed++ |
| 16:58.20 | lowtek | aut: yep |
| 16:58.29 | lowtek | aut: likely just different speakers |
| 16:58.48 | esperegu | hi. I have core debug on lvl 5 but when I get an incomming sip call I see nothing on the console. Sip debug gives lots of sip info though any idea what might be wrong? |
| 16:59.00 | aut | i think the speakerphone on the cisco is better too, but again, been a whiel since i compared |
| 16:59.13 | lowtek | aut: I'd agree with that |
| 16:59.16 | aut | besides the sound, the polycom is 1032142x better than cisco |
| 16:59.26 | aut | simply because of the configuration issue :) |
| 16:59.36 | lowtek | aut: in that the mic seems to be better than my Polycom 550's |
| 16:59.37 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:59.40 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.21.163.adsl.gvt.net.br) |
| 16:59.44 | eric2 | polycom 330 looks like sin |
| 16:59.54 | eric2 | yuck |
| 17:00.02 | lowtek | eric2: I think that's the best bang for the buck in IP phones right now. |
| 17:00.07 | carrar | Cisco wins on look and feel |
| 17:00.30 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 17:00.35 | aut | but consider that it takes probably 20x longer to figure out how to configure the ciscos for SIP |
| 17:00.47 | carrar | naw |
| 17:00.50 | carrar | they are hella easy |
| 17:00.51 | aut | unless the newest firmwares are better |
| 17:00.53 | lowtek | aut: Well, firmware upgrade issues asside, the configs are *really* simple |
| 17:00.53 | carrar | not many options |
| 17:01.17 | ttk-sleep | aut: they are easy, but upgrading the firmware can be a pain... following advice on voip-info.org usually works |
| 17:01.19 | aut | when i was last messing with this (maybe a year or two ago), there was no documentation, etc |
| 17:01.20 | lowtek | aut: our cisco configs are maybe 12-15 lines. |
| 17:01.27 | aut | had to get my info from voip-info.org, and that's a fucking mess |
| 17:01.38 | ttk-sleep | hah |
| 17:02.00 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:02.22 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:02.30 | TommyTheKid | is anyone real good with PRI cause codes and what dial should be doing with them? I have a case where my users are hearing busy tones, and Dial is not exiting with cause 17, its still connected, thinking the call is in progress |
| 17:03.31 | carrar | TommyTheKid, your pri provider can help and do traces |
| 17:03.36 | TommyTheKid | debug info: http://asterisk.pastebin.ca/1333716 ... first call is getting cause code 31, then generally goes busy (last night it was ringing) |
| 17:03.46 | TommyTheKid | regular calls (the second and third calls) are fine |
| 17:04.09 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 17:04.18 | TommyTheKid | I honestly think its the other end, they are "accepting" the call into their local phone system of some sort (who knows what) and playing busy tones, but not sending back the busy status on the line |
| 17:04.35 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 17:08.27 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:11.30 | [TK]D-Fender | TommyTheKid: Your PB shows [ Context 'testing' created by 'pbx_config' ] but your call shows -- Executing [502@itcto:1] |
| 17:11.44 | [TK]D-Fender | TommyTheKid: Not the same context and my trust factor shrinks rapidly |
| 17:12.08 | [TK]D-Fender | TommyTheKid: and it seems to show that it didn't answer but that the call terminates immediately and we don't get that NoOp |
| 17:12.15 | [TK]D-Fender | TommyTheKid: Yet we SHOULD. |
| 17:12.59 | *** join/#asterisk seanmh (n=johndoe@74.95.19.70) |
| 17:16.37 | *** join/#asterisk ingenius (n=alektro@host73.190-136-126.telecom.net.ar) |
| 17:20.26 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:23.26 | timeshell_atwork | Is agentcallbacklogin broken? |
| 17:24.25 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 17:27.40 | yondaime | does anyone here have this problem when compile zaptel error: ‘struct hrtimer’ has no member named ‘expires’ |
| 17:28.10 | yondaime | I can't see anything wrong here overrun = hrtimer_forward(&zaptimer, htmr->expires, |
| 17:28.49 | [TK]D-Fender | timeshell_atwork: Care to be a lot less generic? |
| 17:30.07 | timeshell_atwork | [TK]D-Fender : When I'm trying to log into the agentlogincallback, it's sayign that the the application isn't there. |
| 17:30.27 | [TK]D-Fender | timeshell_atwork: VERSION PLEASE |
| 17:30.34 | timeshell_atwork | 1.6.0-svn |
| 17:30.42 | [TK]D-Fender | timeshell_atwork: Correct, does not exist. |
| 17:30.50 | timeshell_atwork | WHY? |
| 17:31.11 | [TK]D-Fender | timeshell_atwork: Replaced by AddQueuemember, etc |
| 17:31.16 | [TK]D-Fender | timeshell_atwork: Go read the docs. |
| 17:31.19 | timeshell_atwork | lol |
| 17:31.24 | timeshell_atwork | I hate reading docs |
| 17:32.41 | [TK]D-Fender | timeshell_atwork: We know. |
| 17:32.46 | timeshell_atwork | :D |
| 17:33.48 | manxpower | timeshell_atwork: read the UPGRADE.* files in 1.6. All of them. Then read them again. |
| 17:34.06 | timeshell_atwork | manxpower, you know, that's a great idea. |
| 17:34.08 | timeshell_atwork | :p |
| 17:34.44 | [TK]D-Fender | manxpower: if only he'd do it |
| 17:38.19 | talirk81 | in a channel i am setting some variables but when I am in AGI , and print_r on an array i created from the AGI STDIN i dont see them. Do I have to run GET VARIBABLE for each custom varible , i was under the impression asterisk is supposed to put the channel varibles into the STDIN with the default varibles like dnid and callerid and context. |
| 17:39.40 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 17:41.40 | manxpower | talirk81: asterisk only puts the pre-defined variables in STDOUT/STDIN. Any vars you set would be accessed with GET VARIABLE. |
| 17:42.46 | talirk81 | ok i can do that, to bad it can put all defined varibles in stdin, that would make things easier. |
| 17:42.51 | talirk81 | cant i mean |
| 17:43.11 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 17:43.48 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 17:45.29 | esperegu | hi. I have core debug on lvl 5 but when I get an incomming sip call I see nothing on the console. Sip debug gives lots of sip info though any idea what might be wrong? |
| 17:46.18 | *** join/#asterisk ingenius (n=alektro@host85.190-136-99.telecom.net.ar) |
| 17:49.30 | l2trace99 | is there anything I have to set in order to get the CallerID(name) on a DAHDI channel ? |
| 17:50.16 | tzafrir_laptop | l2trace99, what type of channel? analog or digital? |
| 17:50.55 | l2trace99 | analog |
| 18:02.53 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:03.44 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 18:04.50 | *** join/#asterisk path_ (n=path@92-106-21-190.adsl.terra.cl) |
| 18:05.08 | tzafrir_laptop | l2trace99, have you set: callerid = asrecieved ;(sp?) |
| 18:06.28 | *** join/#asterisk sheri_rao (n=sheri_ra@115.186.130.188) |
| 18:07.08 | l2trace99 | in chan_dahdi.conf yes |
| 18:07.29 | *** join/#asterisk madler_navaera (n=madler@c-68-38-82-98.hsd1.nj.comcast.net) |
| 18:07.37 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 18:07.43 | [TK]D-Fender | l2trace99: pastebin your configs |
| 18:08.03 | watchy | sometimes when a call comes in and 2 people pick it up, one person ends up with it showing answered but nothing on the line. anyway to fix that? |
| 18:08.35 | madler_navaera | hello. I seem to be having an issue configuring my outbound route. For some reason the requests aren't being forwarded on to my trunk. The CLI output is here: http://www.pastebin.ca/1334188. Please let me know any advice/guidance. |
| 18:09.47 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 18:10.06 | sheri_rao | IAX works but SIP sucks , any comment? |
| 18:10.16 | ruben23 | hi |
| 18:10.19 | [TK]D-Fender | madFreePBX is NOT supported here |
| 18:10.26 | [TK]D-Fender | madler_navaera: FreePBX is NOT supported here |
| 18:10.41 | [TK]D-Fender | sheri_rao: IAX may work, and SIP is fine. |
| 18:12.13 | watchy | tk you ever have that issue? |
| 18:12.58 | [TK]D-Fender | watchy: No, both think they got it till one looks at there phone and realizes its sitting on IDLE |
| 18:13.29 | l2trace99 | http://pastebin.com/m207a43b9 |
| 18:13.49 | watchy | no way to fix it? |
| 18:15.00 | watchy | its just annoying at our office if we both pickup at the same time sometimes, my phone will show it answered and is online with someone but it actually didn't get it |
| 18:15.18 | [TK]D-Fender | l2trace99: permanently remove all commented out crap and repastebin |
| 18:15.29 | [TK]D-Fender | watchy: that I have not seen |
| 18:15.38 | watchy | hmm maybe its a firmware issue |
| 18:16.10 | watchy | i'm only runing 2.2.2 here |
| 18:16.13 | watchy | ill upgrade |
| 18:18.04 | watchy | yea but its really strange, i'll still see caller id and the phone will think its connected. |
| 18:18.15 | watchy | and stay like that forever until i hang up. |
| 18:18.17 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 18:22.25 | watchy | either way i blame tk for the issues. |
| 18:23.28 | [TK]D-Fender | "He who laughs in the face of adversity.... has a good scapegoat" |
| 18:24.15 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 18:25.39 | l2trace99 | http://pastebin.com/m50bc0902 |
| 18:26.37 | [TK]D-Fender | l2trace99: looks OK |
| 18:28.26 | watchy | tk: you think followme's are useful? |
| 18:30.16 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 18:31.38 | riddlebox | what is the best way to group sip trunks? to do something like sip/g1? |
| 18:31.50 | l2trace99 | [TK]D-Fender: thanks i was wondering if I was missing something |
| 18:32.08 | [TK]D-Fender | riddlebox: Not possible |
| 18:32.09 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:32.09 | sheri_rao | is it possible that SIP remote server can dial out my number but RTP does not get through |
| 18:32.12 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 18:32.19 | [TK]D-Fender | sheri_rao: Entirely |
| 18:32.29 | [TK]D-Fender | loads res_8ball.so |
| 18:33.22 | sheri_rao | then could it be firewall problem |
| 18:33.37 | [TK]D-Fender | sheri_rao: Yup. Improper NAT setup is the #1 common cause |
| 18:33.47 | riddlebox | [TK]D-Fender, so you would basically create a macro to go through the channels if there was congestion until it finds an open one? |
| 18:34.03 | [TK]D-Fender | riddlebox: Yes, its all dialplan. |
| 18:34.31 | riddlebox | or i guess mot voip providers would group channels together on one register |
| 18:34.39 | riddlebox | mot+most |
| 18:35.17 | *** part/#asterisk madler_navaera (n=madler@c-68-38-82-98.hsd1.nj.comcast.net) |
| 18:35.19 | [TK]D-Fender | riddlebox: Most providers don't run * as a back-end and don't trunk to OTHER providers (multiple) |
| 18:40.20 | carrar | How can you say that TK!! |
| 18:40.26 | carrar | EVERYONE runs Asterisk!! |
| 18:40.58 | carrar | Next you are gonna say there is no Santa clause! |
| 18:42.14 | stevetotaro | asterisk is small potatos |
| 18:42.20 | stevetotaro | just a blip on the radar |
| 18:42.26 | carrar | NO!!! |
| 18:42.53 | stevetotaro | carriers with half a brain would investigate freeswitch |
| 18:43.07 | carrar | hahah |
| 18:43.09 | [TK]D-Fender | punches carrar's inner child in the face |
| 18:43.22 | carrar | Ouch! |
| 18:43.59 | stevetotaro | i know a few carriers running teles switches |
| 18:44.14 | doug | once asterisk gets support for dwdm, maybe it'll be worth carriers' time to look at |
| 18:44.41 | stevetotaro | it just cannot compete with FS as far as switching goes |
| 18:50.42 | doug | hm, my sip provider doesn't like periods in the number. |
| 18:50.48 | doug | there an easy way to strip those out? |
| 18:51.22 | doug | ideally, something akin to $EXTENSION =~ s/[^0-9]+//g in perl... |
| 18:51.48 | *** join/#asterisk docelm0 (n=vircuser@pool-141-152-198-243.lyn.east.verizon.net) |
| 18:52.50 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 18:52.55 | carrar | How are you getting periods in your extensions? |
| 18:52.57 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 18:53.03 | rene- | hello |
| 18:53.06 | Gido-E | doug, check the cut application |
| 18:53.09 | doug | pasting them from web pages and/or emails |
| 18:53.41 | doug | gido-e: cut function? |
| 18:53.53 | Gido-E | core show application cut |
| 18:53.57 | rene- | looking for Vijay Shan |
| 18:54.03 | rene- | are you here? |
| 18:54.32 | doug | Your application(s) is (are) not registered |
| 18:54.51 | doug | ah, "Asterisk Slimming" |
| 18:54.54 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:55.17 | doug | hm, "The application Cut is deprecated." |
| 18:56.33 | doug | asteriskguru has a description: " Cut - this application allows you to split the content of a variable by using a specified delimiter." |
| 18:56.39 | doug | not sure that's what i want |
| 18:57.02 | carrar | could run it through a perl agi |
| 18:57.15 | doug | that seems a little excessive. |
| 18:57.30 | carrar | yeah |
| 18:57.37 | doug | i might just, though. |
| 18:57.47 | Gido-E | doug, yep it is an function |
| 18:57.53 | Gido-E | show function CUT |
| 18:59.18 | doug | gido-e: so that looks like it'll let me cut a variable into fields, but i'm unsure how i'd use it to eliminate unwanted characters... |
| 18:59.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:59.33 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:02.30 | Gido-E | doug, something like this: SET(VARbla = ${CUT($VARNAME,$DELIMITER,1)}) |
| 19:06.28 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:10.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:11.12 | cesau | ok so - i figured out how to fix our nat issue in asterisk, but our network admin wont forward ports bound to our firewall onto our asterisk box, so... anyone have a recommendation for proxy software? |
| 19:11.50 | [TK]D-Fender | cesau: Tell the idiot to wake up and do what's necessary |
| 19:12.22 | Whitor | Hi. I'm using asterisk and I have two systems connected via an iax2 trunk. (they both call each other fine) When the IVR on system A answers the call, I want to type in an extension of System B and have the call routed appropriately. Problem is... System A's IVR does not let me enter an extension for System B. What file defines the options available in an IVR ? |
| 19:12.28 | cesau | he says its a "security risk", though he fails to give me any specifics =/ |
| 19:12.48 | [TK]D-Fender | cesau: He's a moron and you're screwed |
| 19:13.11 | [TK]D-Fender | Whitor: EXTENSIONS.CONF |
| 19:13.27 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 19:13.38 | sub | +1 [TK]D-Fender |
| 19:13.50 | Whitor | thx [TK]D-Fender |
| 19:13.57 | *** join/#asterisk trapa (n=trapa@S010600a0cc25f8b0.vs.shawcable.net) |
| 19:14.22 | cesau | the downside to this situation is that someone else had this working (at least able to dial 1 call) without port forwarding, and i have no idea how as we wiped the box and are no longer in contact with that person |
| 19:15.24 | sub | well |
| 19:16.04 | sub | your network admin's (and everyone's job) is to support the business, not carry on about bullshit scurity risks |
| 19:16.09 | [TK]D-Fender | cesau: You can do OUTBOUND calls without a full NAT setup, but largely forget inbound, etc |
| 19:16.15 | sub | especially since security is really just cost/risk analysis anyways |
| 19:16.20 | [TK]D-Fender | cesau: And your network admin is a fuck-off |
| 19:16.34 | trapa | Hello, I'm having issues with the command Hangup not actually hanging up a sip channel. It comes back from voicemail, i send playback(goodbye) (that works) and hangup .. then next thing i know it's back at s,1 |
| 19:16.54 | rob0 | Being on a network is a security risk |
| 19:17.06 | sub | "I unplugged our internet. It was a huge security risk" |
| 19:17.10 | trapa | I'm using a new IRC client, did that whole message get through, or did it start at "not actually" |
| 19:17.29 | Whitor | trapa, whole thing got through |
| 19:17.29 | sub | trapa: it went through. Paste your extensions.conf to a pastebin |
| 19:17.50 | trapa | works on pasting. |
| 19:19.07 | *** join/#asterisk sipy (n=bob@modemcable160.54-37-24.mc.videotron.ca) |
| 19:20.37 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 19:21.41 | kannan | hello, when i try to use text2wav in phpagi based agi script, it says /text2_wav<some-random0name> not found in any format. Where to start debug? |
| 19:22.33 | trapa | sub: http://pastebin.com/d63c78d70 |
| 19:23.42 | kannan | in phpagi, where is the /tmp dir , do we need to define it seprately in Section [festival]? |
| 19:23.44 | *** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com) |
| 19:23.50 | [TK]D-Fender | trapa: pastebin your failed calls CLI output at verbose 10 |
| 19:23.50 | cesau | where was that udp port tester tool... voipuser.org? |
| 19:23.59 | trapa | Sub: So it answers, great, plays the message, pauses, then it goes to the macro-voicemail ... that rings my cellphone, and sends them to voicemail successfully, then it exits and returns to the stack .. |
| 19:24.10 | cesau | (found it) |
| 19:24.13 | trapa | fender: Okay will do. |
| 19:24.18 | citywok | [TK]D-Fender: 500 calls and an hour of using chanspy on 1.6.0.5 didnt segfault where 1.4.21 would tank out within 100 calls and 15 minutes |
| 19:24.55 | citywok | now i am going to have to upgrade my primary server from 1.4.21 to 1.6.0.5 -- anything major i should worry about? Are all the AMI commands the same for telnetting in, etc? |
| 19:25.10 | *** join/#asterisk ThatKidKel (n=Kelvin@c-24-99-104-97.hsd1.ga.comcast.net) |
| 19:26.42 | [TK]D-Fender | citywok: No. Go read the upgrade DOCS |
| 19:27.46 | trapa | fender: http://pastebin.com/d4f376ae1 |
| 19:28.04 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:28.34 | trapa | fender, sub: At line 57, it seems to execute the hangup .. or try or somthing, but then it goes right back to the top of the s, stack |
| 19:29.20 | trapa | I tried playing with variables in order to maybe make it let it through the first time and hangup the second, but now i realize it's processing the hangup command without actually hanging up. |
| 19:29.36 | sub | line 57 is part of your 1-900 dialing though? |
| 19:30.03 | trapa | sub: Sorry i mean line 57 on the second patebin http://pastebin.com/d4f376ae1 the messgae output |
| 19:30.15 | telnettech | ok guys need a little more help with an error that Im seeing from time to time: http://pastebin.com/d18bc630d |
| 19:30.57 | telnettech | im not worried about the 1 set of warnings as this is something to do with my dialplan setup. It is the 2nd warning that Im getting that I need help with |
| 19:31.17 | ThatKidKel | I'm looking for a device or devices that will support 150 analog lines. I want it to be able to program each "port" on the unit so that i can just do a Dial(SIP/1@UNITA) (where 1 is the port).... anyone know of such a device? |
| 19:32.04 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 19:34.35 | trapa | sub: So yeah on the first pastebin (the dialplan) they get to line 29 and retun from line 29 successfully, but then line 31 doesn't hangup on them. Tho in the verbose output, i can see it trying to execute that. Then it starts again at the top (s,1) |
| 19:37.04 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 19:39.11 | *** part/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 19:39.46 | sub | trapa: i'm scratching my head on this one |
| 19:40.33 | trapa | Yeah me too .. On lines 18,19,20 and 22 .. i was trying to do a variable so that you set it the first time, and then skip to hangup when it comes around the second time |
| 19:40.41 | trapa | but every time it comes around the variable is empty again |
| 19:42.14 | *** join/#asterisk kimo_sabe (n=nick@zappa.azrackspace.net) |
| 19:42.56 | trapa | I'm kinda wondering if it's possible that Voipgo is re-sending the call instead of hanging up. Which would explain why the channel variables are reset. BUt then when you look at the output log the call identifier is the same |
| 19:42.59 | kimo_sabe | is faxing out a PRI usually problematic? |
| 19:47.16 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 19:49.10 | sub | yeah i am lost man |
| 19:49.15 | sub | i am very new to * though ; |
| 19:49.18 | sub | ;) |
| 19:49.31 | sub | gotta head to work, bye all |
| 19:50.40 | trapa | Have fun, thanks for the attempt at help tho. |
| 19:50.54 | trapa | Will have to see if anyone else has a idea |
| 19:51.44 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 19:53.51 | trapa | So, Nugget, Grantm, you guys good for a question about SIP calls not paying attention to the Hangup command? |
| 19:56.38 | Nugget | no clue, sorry |
| 19:58.07 | trapa | Thanks anyway :) ... I'm pretty stumped on this one. Trying to call my voip provider and see if they have any guesses, trouble is they don't support asterisk. |
| 19:58.23 | trapa | so it'll just be up to the tech to find out if he's ever heard of sucha problem |
| 19:58.44 | Nugget | nutty. that's like an isp that doesn't support tcp/ip. "Sorry, we only sell NetBEUI PPPoE connections" :) |
| 19:59.19 | trapa | Well i think they don't want to support it because they don't want to "officially" support it .. and have to take calls from people who don't know how to configure it |
| 20:00.19 | Nugget | well I wouldn't want to support asterisk either, but then again, I also don't expect to earn a living off selling voip services. if I did I guess I'd sort of figure I'd have to in order to be successful. |
| 20:00.52 | trapa | Hehe .. apparently not .., I mean look at vonage they don't support asterisk either. |
| 20:02.14 | Nugget | yeah, i guess you have a point. I sometimes forget about the huge market for consumer phone service like vonage. |
| 20:05.36 | telnettech | ok guys need a little more help with an error that Im seeing from time to time: http://pastebin.com/d18bc630d |
| 20:05.41 | telnettech | im not worried about the 1 set of warnings as this is something to do with my dialplan setup. It is the 2nd warning that Im getting that I need help with |
| 20:05.55 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 20:06.30 | kimo_sabe | analog fax->FXS card->PRI, reliable? |
| 20:09.10 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 20:09.12 | kimo_sabe | I've been getting horrible reliability with this setup, is that normal? Should I not have bothered trying? |
| 20:12.42 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 20:13.32 | trapa | kimo: Can't say never even tried getting FXS, FXO cards working. (Just so you know we're not ignoring you) |
| 20:13.52 | kimo_sabe | trapa: thanks |
| 20:16.08 | HeMan | anyone using Snoms? |
| 20:17.03 | HeMan | I'm trying to get multiple identitys to work |
| 20:22.26 | talirk81 | can someone tell me what im doing wrong? http://pastebin.com/d414db85d |
| 20:22.36 | *** join/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 20:22.52 | *** join/#asterisk becks` (n=sdfgsfdg@169-244.104-92.cust.bluewin.ch) |
| 20:24.12 | talirk81 | nvm i figured it out, i used AGIecho not AGIexecute |
| 20:24.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:25.04 | telnettech | what causes dropped calls when the phone is an ip phone |
| 20:27.49 | trapa | telnettech: I'd suspect dropped packets. But not just like 1 or 2 but a series |
| 20:27.49 | Nugget | telnet is eeeeeeevil! |
| 20:28.04 | *** part/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 20:28.16 | telnettech | anything else? |
| 20:28.52 | trapa | telnettech: Oh probably i'm no expert, but that'd be my first thoughts ... |
| 20:29.08 | kimo_sabe | telnettech: does it work for a little bit then drop, or is it a drop on answer or transfer that you're getting? |
| 20:29.44 | telnettech | it is sporadic but it is a drop on answer |
| 20:29.57 | telnettech | within seconds of the answering of the call |
| 20:30.18 | kimo_sabe | telnettech: but you do get audio for those seconds? So it's not a codec negotiation thing... |
| 20:31.55 | telnettech | no you hear someone say he......... |
| 20:35.58 | telnettech | any other ideas |
| 20:36.25 | telnettech | im not even seeing the dropped calls on the CLI and I have the verbosity up to 99 |
| 20:38.13 | manxpower | telnettech: do you have callprogress=yes or busydetect=yes in zapata.conf? |
| 20:39.30 | telnettech | i dont have a callprogress param but the busydetect=yes |
| 20:40.10 | manxpower | telnettech: remove it and your calls should stop dropping |
| 20:40.20 | telnettech | ok |
| 20:41.01 | telnettech | manxpower: what about busycount? |
| 20:42.21 | manxpower | telnettech: busycount doesn't count is busydetect is not set |
| 20:42.31 | telnettech | k |
| 20:48.11 | ruben23 | hi |
| 20:50.20 | ruben23 | how much the amount of bandwidth with a g729 codec consume... |
| 21:01.50 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 21:09.01 | seanmh | anyone using 7941/7961s with SIP successfully? |
| 21:09.14 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:10.13 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 21:12.40 | WHYS | I am relooking at email-to-fax applicatoins. I was looking at RxFax, but it only handles a few formats, and I am sure people here will want a variety of different options. |
| 21:12.40 | WHYS | Does anyone have other suggestions for fax-out solutions? |
| 21:13.48 | HeMan | WHYS: have you looked at iaxmodem and hylafax? |
| 21:14.18 | WHYS | Yes, I setup incomming fax that way, but Havn't tried it for fax out yet. You? |
| 21:15.04 | HeMan | No, we don't need to send faxes |
| 21:15.32 | *** join/#asterisk freddyk (n=freddyk@host61-6-dynamic.42-79-r.retail.telecomitalia.it) |
| 21:15.34 | savage1 | http://www.theonion.com/content/video/sony_releases_new_stupid_piece_of |
| 21:16.11 | WHYS | I'll have a look at that again. I'll try to find a recipe. :) Thanks |
| 21:17.48 | TommyTheKid | oh man, someone said my name, (probably [TK]) but I was ... away, and my scroll back doesnt go back far enough |
| 21:18.01 | TommyTheKid | I thought I saw something then it was gone |
| 21:19.21 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 21:20.02 | Kobaz | is there an easy way to make the audio on calls a bit louder |
| 21:20.10 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 21:20.18 | Kobaz | like the speakerphone on polycom 320's isn't very loud |
| 21:20.32 | Micc | Anyone know why all my customers caller id only shows the name but not the number? |
| 21:20.44 | Micc | Are they linked into one on some phones? |
| 21:21.09 | Micc | setting the name makes the name only show up. but not setting the name makes the number only show. |
| 21:21.44 | *** join/#asterisk timte (n=tim@c-0d7ce255.04-86-6b6c6d10.cust.bredbandsbolaget.se) |
| 21:26.21 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 21:26.38 | timte | I'm about to try asterisk/pbx out for the first time. I recommend that asterisk comes without configuration. Do you recommend to start with AsteriskGUI or FreePBX or just learning the config files and edit them manually? |
| 21:26.59 | timte | s/recommend/read/ |
| 21:28.57 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:29.26 | jplank | timte, should always start with learning how to config manually. FreePBX is great (I'm a big fan personally) but since it takes complete control over the config files, its good for someone who doesn't know how to use asterisk, and doesn't care to learn, but not so great for someone who wants to learn how to use it |
| 21:30.18 | jplank | I've only used asterisk GUI on a warp box, and I can't say I'm a big fan |
| 21:31.23 | Khratos | brb, going home... |
| 21:31.43 | timte | I read that FreePBX comes with sane defaults so you get something running directly. I guess that could be nice. |
| 21:31.54 | jplank | yes, that is very true |
| 21:32.20 | jplank | its a great GUI, like I said, I'm a big fan, I use it on almost every customer prem install |
| 21:34.08 | timte | If this should be a long successful project, perhaps I should start easy. |
| 21:34.13 | jplank | but if you really want to learn how asterisk works, FreePBX is not really the way. You don't learn the asterisk way of doing things, you learn the freepbx way of doing it |
| 21:34.27 | jplank | whats your pplan? setting up * to play with, or use? |
| 21:35.18 | jplank | your best first move is to get the asterisk book to be honest |
| 21:35.22 | jplank | before you even install it |
| 21:35.44 | timte | for private use, mostly so I can use any computer as a phone |
| 21:36.12 | TommyTheKid | timte: skype? :p |
| 21:36.29 | timte | TommyTheKid: I have skype, but nah |
| 21:36.58 | madgeek | having your own PBX vs sharing your bandwidth so other ppl can make their calls |
| 21:37.00 | madgeek | that's a big diff |
| 21:37.17 | timte | this has more uses, like being able to call home to control everything connected to your home network :) |
| 21:38.59 | timte | jplank: oh, a book, it's that advanced? I'm not gonna use it professionally |
| 21:39.00 | *** join/#asterisk ivan__zxc (n=ivan@204-56-235-201.fibertel.com.ar) |
| 21:39.31 | ivan__zxc | hi all |
| 21:39.40 | lanning | ~book |
| 21:39.41 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:40.07 | madgeek | any o'reilly book is a good investment |
| 21:40.14 | ivan__zxc | can someone help me with libopenr2? |
| 21:41.03 | *** join/#asterisk Grufft3ch (n=Adam@140.239.216.61) |
| 21:41.27 | timte | madgeek: hmm, did you suggest that you can make two calls from the same phone number if you have pbx? |
| 21:43.21 | TommyTheKid | timte: if you have more than one phone line, then probably yes, we have 4 PRI's (T1) with about 100 different phone numbers assigned to them, I can make up to about 92 phone calls from the same number if I want |
| 21:44.06 | TommyTheKid | lots of times a business will use their main phone number for all outgoing calls |
| 21:45.00 | TommyTheKid | if you have regular analog (POTS) lines, then, no |
| 21:45.42 | jplank | if you use SIP trunking, then most likely yes |
| 21:45.45 | n3hxs | You might be able to get the Telco to give all your POTS the same CID. |
| 21:45.59 | TommyTheKid | thats true |
| 21:46.03 | TommyTheKid | (both of them) |
| 21:46.15 | jplank | n3hxs: thats a big 911 no no |
| 21:46.19 | timte | brain overheated, too many new words for me :) |
| 21:47.19 | *** join/#asterisk dmz (n=dmz@216.16.217.9.dyn-cm-pool45.pool.hargray.net) |
| 21:47.22 | n3hxs | Doesn't mean it can't happen. Especially if it is in the same building. |
| 21:47.30 | TommyTheKid | timte: if you have TWO phone lines, you can make TWO phone calls, with Asterisk, you can always get Internet based phone lines (SIP/IAX2) as well to increase your capacity |
| 21:47.45 | n3hxs | We have done things with Telco that wern't supposed to happen. |
| 21:48.08 | timte | TommyTheKid: you mean physical copper wires? |
| 21:48.14 | TommyTheKid | has it easy cause my "telco" is my companies corporate PBX |
| 21:48.41 | TommyTheKid | timte: the physical wires have nothing to do with how many lines you have... |
| 21:49.10 | TommyTheKid | (except in analog) |
| 21:49.45 | TommyTheKid | if you have regular phone lines from the phone company (like a normal home phone line), then yes, each pair of copper wires is a new phone line |
| 21:49.45 | jplank | n3hxs: not saying it can't happen. We used to get focal to do it all the time way back when, but today, I don't see many companies doing it, unless its really some type of voip trunking handed off as analog. I know we sure as hell wouldn't do it on regular POTS, I don't want the fine |
| 21:51.24 | TommyTheKid | haha, reminds me of the time my stupid IP phone was getting double digits and I dialed 9911 in an attempt to get 91800....... |
| 21:51.52 | TommyTheKid | I was like... um, ok I know my number is in California, but I am in Colorado and there is no emergency, just a phone system malfunction .. blah blah blah |
| 21:51.56 | jplank | heh, just tell them your from the phone company testing 911, then ask for the address your calling from |
| 21:52.18 | TommyTheKid | lukily I know the actual street address the numbers are assigned to :) |
| 21:52.35 | jplank | screw it, if your not there, who cares? :) |
| 21:52.44 | n3hxs | Way back when, the phone number was the house number! |
| 21:53.10 | TommyTheKid | that doesnt seem like it would scale very well |
| 21:53.24 | jplank | "Yes operator, I'd like to be connected to 'Rose 5134' please" |
| 21:53.41 | TommyTheKid | if they would just put 10,000 houses on a street, maybe |
| 21:54.03 | jplank | wasn't it the street name and a individual number? |
| 21:54.18 | jplank | this is way before my time, I just remember it from old TV shows like The Honeymooners |
| 21:55.02 | n3hxs | I lived in Turkey for a while and we had a 3 digit phone number. |
| 21:55.17 | n3hxs | But they also had a cord board for switching. |
| 21:55.42 | jplank | heh, my grandma used to tell me about how it was when she worked at the switchboard |
| 21:55.53 | TommyTheKid | even as few as 15 years ago, in small exchanges, they could just dial the last 4 digits (much like an extension in a business I guess) |
| 21:56.11 | TommyTheKid | everyone in that community only would share the last 4 digits when someone asked their number |
| 21:56.19 | jplank | give it another 2 or 3 years and all 50 states will be 10 digit dialing |
| 21:56.28 | TommyTheKid | I live in Denver, and we have two area codes that are local, so I always assume someone wants all 10 digits :) |
| 21:56.49 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 21:56.53 | TommyTheKid | 2 area codes and 10 digit dial |
| 21:57.06 | jplank | a lot of smaller carriers (like the one I work for) already converted to only 10 digit dialing, its easy for routing and setting up translations |
| 21:57.11 | TommyTheKid | they are talking about eliminating our Internal extensions at work, and moving to 10 digit dial |
| 21:57.13 | n3hxs | Run out of numbers we will have to go to 11 digit dialing. |
| 21:57.26 | Micc | Anyone know why one of my spa8000's won't reset to factory defaults? |
| 21:57.28 | jplank | TommyTheKid: what?!? |
| 21:57.31 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 21:57.31 | jplank | thats crazy |
| 21:57.48 | TommyTheKid | jplank: I agree |
| 21:58.02 | TommyTheKid | they want it to be the same experience if you are "inside" or "outside" |
| 21:58.03 | jplank | revolt! |
| 21:58.27 | TommyTheKid | also we have 5 and 7 digit, have to dial 70, then a 5 digit to get overseas |
| 21:58.34 | jplank | I have customers who get mad because by default we start with a 4 digit extensions |
| 21:58.35 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:58.44 | jplank | that makes sense |
| 21:59.01 | TommyTheKid | jplank: I suppose if the company had less than 100 people, 4 digits does seem silly |
| 21:59.50 | jplank | we always ask the customer if they want to pick their extension range, they say no, our scripts default to 2xxx and then all of a sudden, they want to pick it, AFTER everything is configured |
| 21:59.51 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 21:59.56 | TommyTheKid | but then it also depends on whether you are just shortening a 10 digit number (taking the last four digits) or if you have one number and then dial ext 1003 to reach me sort of setup |
| 22:00.00 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:00.33 | TommyTheKid | I am sooooo glad I don't have to support voice services for companies.. its bad enough running their email server :p |
| 22:00.49 | jplank | 10 years ago, or even 5, it was easy to get DID ranges that match your extension ranges |
| 22:00.54 | cesau | is there a place other than extconfig.conf that needs to be configured to enable realtime extensions? |
| 22:01.01 | jplank | now your lucky if you can get blocks longer then 20 |
| 22:01.13 | cesau | (sippeers realtime is working) |
| 22:01.43 | jplank | TommyTheKid: its funny, PBX support isn't as bad as we thought it was going to be when we first started supporting PBXs |
| 22:01.45 | TommyTheKid | yea, our latest wasn't even contiguous (nor working right) .. only 3 of the 4 PRIs are working and they wouldn't give us the whole 100 numbers |
| 22:01.55 | cesau | but im not even seeing requests for extensions hit the db server |
| 22:02.31 | jplank | its hard to break out full blocks of 100 numbers |
| 22:02.49 | jplank | most numbers you get today are recycled numbers |
| 22:02.55 | TommyTheKid | in our CA building, I have ext. 688xx (650-xxx.88xx), here in CO, we have maybe 30-40 dispersed across ~3000 numbers |
| 22:02.56 | jplank | unless you get a new area code, or exchange |
| 22:03.01 | jplank | and no one wants to give those out |
| 22:03.13 | TommyTheKid | heh |
| 22:03.21 | TommyTheKid | recycled numbers are SOO much fun |
| 22:03.50 | jplank | heh |
| 22:04.09 | jplank | 212 867 5309 was lost to VZ's pool a couple weeks ago |
| 22:04.12 | TommyTheKid | we also had the pleasure of getting assigned the extensions 186xx-188xx ... so you can imagine how many calls we get at the top of each hour from people dialing 1-888..., 1-866.... etc |
| 22:04.31 | TommyTheKid | 877 too I guess |
| 22:04.32 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:04.52 | jplank | thats crazy, I stay so far away from extension ranges like that |
| 22:05.00 | TommyTheKid | I literally had to macro 10 extensions out of each range with a "this is not the number... dial 9 .. monkeys .. etc " :) |
| 22:05.10 | jplank | lol |
| 22:05.46 | TommyTheKid | like I said, we get a range of 5 digit extensions assigned from our corporate PBX, controlled by a third party |
| 22:05.58 | jplank | we had a customer who wanted their extension range to be 91xx |
| 22:06.03 | TommyTheKid | the 5 digits map 1:1 with a 10 digit, but we got some really shitty ones |
| 22:06.07 | TommyTheKid | eep |
| 22:06.11 | jplank | there was so many things wrong with that, I didn't know where to start |
| 22:06.24 | TommyTheKid | say.. uhm... no |
| 22:07.07 | cesau | hrm, switch statement |
| 22:08.17 | TommyTheKid | not saying it couldn't be done, but I wouldn't |
| 22:08.20 | cesau | is there any way to have realtime extensions without a switch statement in the dialplan? |
| 22:08.32 | TommyTheKid | it would make the dialplan on the phone rather difficult as well |
| 22:08.33 | jplank | it could be done, but its too prone to mess ups |
| 22:08.37 | jplank | yea |
| 22:08.42 | TommyTheKid | 9 usually re-dialtones |
| 22:08.47 | jplank | I could see lots of calls to 911 every day |
| 22:09.04 | jplank | 9 only gets second dialtone if you tell it to |
| 22:09.11 | jplank | but most people use 9 to dial outside anyway |
| 22:09.22 | TommyTheKid | right, thats the "typical" number for that |
| 22:09.46 | TommyTheKid | note - we use ancient cisco 7960's |
| 22:09.51 | jplank | I'm not even a big fan of starting extensions with a 1 unless I HAVE to, let alone 91 |
| 22:10.07 | jplank | I got a 7970g on my desk I use as a house phone |
| 22:10.09 | jplank | I love this phone |
| 22:10.17 | jplank | too bad it doesn't work great |
| 22:10.27 | jplank | unless you have cisco call manager |
| 22:10.29 | jplank | looks great though |
| 22:11.23 | TommyTheKid | 1 has the obvious defect of long distance (and forgetting to dial 9) .. but honestly stupid people are everywhere, and unless you live somewhere like Denver (that has manditory 10 digit dialing), someone is going to try to dial a random local number without 9 and get someone they didn't expect |
| 22:11.55 | TommyTheKid | here of course, I would just avoid 7 and 3 (720 / 303) |
| 22:12.12 | talirk81 | anyone know why I got the exited with a non-zero here? http://pastebin.com/m7645844a |
| 22:13.24 | TommyTheKid | i get non-zero every time I hang up :) |
| 22:13.48 | jplank | wait, my clients are in NYC. I have a hosted PBX client that HAD to have 3 digit extensions that started with 1 AND refused to dial 9 to get an outside line. Since its NYC everything is 11 digit dialing, and since their extensions started with 1 it became a challenge. Especially since they didn't want to have to wait for the call to complete or dial # after the number |
| 22:14.06 | talirk81 | further more why didnt the hangup phase intiate :( |
| 22:14.47 | jplank | I kept begging my boss to let them out of their contract and bother some other company |
| 22:15.33 | TommyTheKid | well as long as they have less than 10 employees, 10x could work (no areacodes start with 0?) |
| 22:16.05 | jplank | less then 10 employees, but it had to be the same extensions as their old key system |
| 22:16.07 | jplank | 110 |
| 22:16.07 | jplank | 11 |
| 22:16.09 | jplank | 111 |
| 22:16.10 | jplank | 112 |
| 22:16.13 | jplank | 113 |
| 22:16.14 | jplank | etc |
| 22:16.42 | jplank | it had to work just like their key system did. |
| 22:16.57 | *** join/#asterisk mrsci (n=mrsci@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 22:17.06 | jplank | which is like kryptonite when it comes to asterisk and most IP PBXs |
| 22:18.20 | jplank | luckily it was only like 9 employees so it wasn't too pain in the ass, I just to program the dialplan specifically for every extension and 1xxxxxxxxxx |
| 22:18.46 | jplank | then after all was said and done I realised I could of used NXX and been fine |
| 22:18.57 | TommyTheKid | I was just wondering,, I am sure no area codes start with 0, do any start with 1 or is it just 2 and on? |
| 22:19.04 | TommyTheKid | 2-9 that is |
| 22:19.18 | jplank | NXX - first digit 2-9 2nd and 3rd 0-9 |
| 22:19.38 | jplank | it didn't dawn on me while setting up the dialplan of the phone (polycom) |
| 22:19.41 | TommyTheKid | ok so it wouldn't be too hard to go from ext 100 - 119 |
| 22:19.55 | TommyTheKid | right, then you just gotta figure out how the phone will deal with it |
| 22:20.00 | jplank | no, not at all, once you realize you can do NXX |
| 22:20.15 | ruben23 | hi i tried to dial my local phone on the asterisk CLI by dial 1000, want is the hangup command for this.. |
| 22:20.22 | jplank | I was also trying to figure out who I could do SLA |
| 22:20.33 | jplank | how I could* |
| 22:20.51 | jplank | because again, it had to work just like their old key system |
| 22:22.01 | jplank | I wound up having to hack together call parking, hinting and BLF to get it to almost appear like SLA, but it was far from elegant |
| 22:24.26 | jplank | would probably work better on a aastra though. I need to try that one day |
| 22:25.01 | TommyTheKid | honestly, I have a 1.2.x box that I am not really looking forward to upgrading, how sad is that.. it works, I dont want to screw with it :) |
| 22:25.18 | TommyTheKid | I am sure its scary vulnerable, but its at least inside the network :) |
| 22:30.16 | *** join/#asterisk Chex (n=Stefan@bas1-montreal48-1176171542.dsl.bell.ca) |
| 22:35.09 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 22:37.06 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:37.21 | jplank | the only issue I really had with 1.4 (which urged us to upgrade our softswitches based on asterisk) is that terrible DTMF bug |
| 22:39.05 | jplank | I meant issue with 1.2 |
| 22:40.07 | mrsci | how stable is 1.6.0.5? |
| 22:41.24 | russellb | jplank: the rfc2833 stuff? |
| 22:42.11 | *** part/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 22:42.38 | *** join/#asterisk adnc (n=adnc@unaffiliated/adnc) |
| 22:43.21 | jplank | russellb: yea |
| 22:43.24 | jplank | it was killing us |
| 22:43.53 | TommyTheKid | [TK]D-Fender: I know you said something to me, but before I got a chance to read it, it went out of my scroll back (damn meetings) .. do you have it in your history or could you re-look? |
| 22:43.59 | russellb | jplank: using cisco gateways? |
| 22:44.08 | *** join/#asterisk laggo (n=user@80.175.142.209) |
| 22:44.12 | jplank | actually, it was a nortel gateway |
| 22:44.18 | russellb | ah |
| 22:44.19 | jplank | using their new cards |
| 22:44.25 | russellb | gotcha. |
| 22:44.31 | russellb | it was the cisco ones that we normally heard about |
| 22:44.52 | eppigy | ALLO |
| 22:44.56 | jplank | their old cards worked fine, but once VZ started upgrading the cards in their c2k's we started having the problem |
| 22:45.16 | jplank | it was actually pretty funny, ratecenter by ratecenter DTMF stopped working |
| 22:45.59 | [TK]D-Fender | TommyTheKid: Nope, no history here... |
| 22:46.26 | TommyTheKid | ok it was like 5 hours go |
| 22:46.34 | TommyTheKid | it was the debug output with the code 31 stuff |
| 22:46.41 | [TK]D-Fender | TommyTheKid: many miles and netowrks ago as well |
| 22:46.53 | [TK]D-Fender | TommyTheKid: Oh I do remember something now |
| 22:47.00 | [TK]D-Fender | TommyTheKid: Your NoOp NEVER EXECUTED |
| 22:47.02 | TommyTheKid | I showed the dialplan entries at the top, itcto is the inbound context for my phone, which imports "testing" |
| 22:47.14 | [TK]D-Fender | TommyTheKid: And your contexts of dialplan didn't matcht he CLI you showed |
| 22:47.24 | [TK]D-Fender | TommyTheKid: trast factor is near 0 |
| 22:47.28 | [TK]D-Fender | trust* |
| 22:47.33 | TommyTheKid | trust? |
| 22:47.42 | TommyTheKid | did you miss the "Include" ? |
| 22:48.26 | [TK]D-Fender | TommyTheKid: You did... it wasn't in the PB... so show me something complete |
| 22:48.44 | TommyTheKid | oh maybe we are talkign about different PB's |
| 22:48.57 | TommyTheKid | http://asterisk.pastebin.ca/1333716 |
| 22:49.52 | *** part/#asterisk ThatKidKel (n=Kelvin@c-24-99-104-97.hsd1.ga.comcast.net) |
| 22:49.53 | trapa | So anyone up for trying to help diagnose why a sip channel is ignoring the hangup() command? |
| 22:50.12 | jplank | what makes you think its ignoring the hangup command? |
| 22:50.30 | trapa | Extensions.conf: http://pastebin.com/d63c78d70 , Verbose Debug: http://pastebin.com/d4f376ae1 |
| 22:50.34 | jplank | I went weeks thinking the same thing, until I realized I was just miss understanding that the hangup command does |
| 22:51.09 | trapa | Line 58 says executing hangup, line 59 says executing s,1,answer (of the verbose debug) |
| 22:51.29 | jplank | I'm pretty sure either fendor or drmessano kept abusing me until I figured it out |
| 22:51.38 | jplank | trapa, what do you expect the hangup command to do? |
| 22:51.55 | TommyTheKid | [TK]D-Fender: the [itcto] context is inbound for my SIP device(s), its basically a permission statement.. it "Include => "'s all the different context's |
| 22:52.32 | trapa | Send the voip provider information that the call has been disconnected on my end |
| 22:52.37 | [TK]D-Fender | TommyTheKid: I don't see anything that would qualify as an 'Answer' in your PRI debug, and can't see why your NoOpe would not be called otherwise... |
| 22:52.42 | trapa | question is, is there a channel_destroy or some better function to use? |
| 22:53.11 | jplank | nay, thats not what it does, it pushes the call to the h priority if there is one, if not, it goes on to the next priority (in your case being s) |
| 22:53.53 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 22:53.56 | trapa | Really .. Interesting! |
| 22:54.10 | TommyTheKid | [TK]D-Fender: my guess -- when I hung up the SIP channel (cause the other side didn't answer) it halted execution |
| 22:54.11 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.20.90) |
| 22:54.25 | trapa | So there's a H priority (h,1,blah) that i can use to execute more commends like a AGI? |
| 22:54.32 | [TK]D-Fender | TommyTheKid: Oh, so YOU hung up? |
| 22:54.43 | TommyTheKid | yes, I thought I put that in the pb |
| 22:54.45 | [TK]D-Fender | TommyTheKid: What happens if you don't? |
| 22:54.59 | jplank | trapa, add something like exten => h,1,NoOp("DONE!") or something like that to the end of the dialplan |
| 22:54.59 | TommyTheKid | it plays "busy" tones |
| 22:55.22 | TommyTheKid | (or in the case of last night, when no one was there, it plays ringing) |
| 22:55.40 | TommyTheKid | sometime before 5 I will be able to reproduce the busy tones |
| 22:55.56 | jplank | this is only for SIP trunks though, I believe over a PRI or any ZAP channel, it does actually go on hook before moving the the H priority |
| 22:56.34 | *** join/#asterisk coppice (n=chatzill@96.196.17.210.dyn.pacific.net.hk) |
| 22:57.01 | *** join/#asterisk mortsmel (n=andrew@corp-nat.kaslnet.net) |
| 22:57.08 | [TK]D-Fender | TommyTheKid: Let it sit rhough and put a timeout on that dial as well |
| 22:57.19 | TommyTheKid | [TK]D-Fender: Note like 85, i think thats the problem. I think they are taking the call into their system and giving me status inband |
| 22:57.28 | TommyTheKid | s/like/line/ |
| 22:58.03 | TommyTheKid | thanks jbot :p |
| 22:58.07 | [TK]D-Fender | TommyTheKid: That would be.... unfortunate... |
| 22:58.24 | TommyTheKid | yea, inband status sucks ass |
| 22:58.28 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 22:58.46 | *** join/#asterisk trapa (n=trapa@S010600a0cc25f8b0.vs.shawcable.net) |
| 22:59.14 | trapa | jplank Sorry about that .. Somehow rebooted. You're definatly onto somthing. |
| 22:59.31 | jplank | trapa: you get my last message about that only working with SIP, and on PRI and most all ZAP channels, it actually puts the call "onhook" |
| 22:59.40 | trapa | jplank .. It did go to the h priority .. So now .. how do i destroy the sip channel and actually end the call? |
| 23:00.07 | trapa | jplank: I'll remember that |
| 23:00.19 | TommyTheKid | does "Dial" stop Playtones ? |
| 23:00.21 | jplank | if it hits the h priority, it will do whatever you set in the h priority, and assuming you don't use any goto |
| 23:00.29 | jplank | 's or anything like that, end the call |
| 23:00.41 | trapa | jplank: It did hit the h priority,t hen it went to the s priority again. |
| 23:00.46 | TommyTheKid | I noted the example in the Playtones has "&s@no-op" |
| 23:00.52 | TommyTheKid | (which just hangs up) |
| 23:00.55 | jplank | is the h priority AFTER the s priority? |
| 23:01.29 | jplank | trapa in that PB you sent me, what context are we talking about? |
| 23:03.05 | *** join/#asterisk henkoegema (n=henkoege@78-21-68-204.access.telenet.be) |
| 23:03.12 | trapa | Uhm Incoming |
| 23:03.18 | trapa | it goes to the voicemail macro then back |
| 23:03.57 | TommyTheKid | [TK]D-Fender: search down to ~ line 276 for my "control" experiment (to a line I had control over)... the Noop works when Dial exits :) ... meanwhile, I will add a timeout to the Dial (and maybe option "H" ?) |
| 23:04.04 | trapa | i added the h priority as "exten => h,1,NoOp("Done!");" immidately before [outbound] at the end of the [inbound] context |
| 23:04.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:05.48 | jplank | are you saying it hits s,1 on the incoming context, runs through the priorities, then hits hangup, goes to the h priority, then goes back up top to the s,1 priority? |
| 23:06.00 | trapa | jplank exactally |
| 23:06.20 | [TK]D-Fender | TommyTheKid: Start with the timeout.... though if you're inband I think you're DOA from there. |
| 23:07.03 | TommyTheKid | I was afraid of that |
| 23:07.54 | [TK]D-Fender | trapa: I don't see an "h" in your CLI output... it looks like it jumps back immediately.... |
| 23:08.00 | TommyTheKid | fuck they answered :p |
| 23:08.05 | TommyTheKid | I was like, uh... |
| 23:08.14 | citywok | I'm playing with transmitting callerid info into my pbx, it says *ANI*DNIS* -- i tried dialing that way but it doesn't work, i'm using E&M trunks, any ideas? |
| 23:08.18 | jplank | fender, he added it after I told him to |
| 23:08.23 | jplank | after he took these PBs |
| 23:08.34 | jplank | I was just going to ask him for two new PBs |
| 23:09.14 | trapa | Just prepping them |
| 23:09.19 | *** join/#asterisk maruz (n=maumar@host86-19-dynamic.0-79-r.retail.telecomitalia.it) |
| 23:10.14 | [TK]D-Fender | jplank: Just in the one he showed H never got hit and there is NO reason for "s" to be afterwards. The SIP session appears to eb the same as it it ignored it |
| 23:11.51 | jplank | I dont get it either to be honest. Before he was using hangup() without a h prioirty, but now he said he is using h, but its still going back up to s,1 after h. I was expecting to see a goto or something, but based on what he's shown so far, there isn't |
| 23:12.06 | *** join/#asterisk Khratos (n=Khratos@190.80.231.209) |
| 23:12.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:13.38 | trapa | Extensions.conf - http://pastebin.com/d26c900e Debug: http://pastebin.com/m13394f02 |
| 23:14.33 | ruben23 | hi anyone have idea on this error: http://pastebin.com/m1cb63493 trying to setup a local SIP phone on my asterisk.. |
| 23:15.22 | jplank | fender, I'm not telling him to use the h priority wrong, am I? |
| 23:16.03 | trapa | jplank, Fender: Is there a session_destroy() for asterisk, that will just like kill the sip channel? |
| 23:16.26 | jplank | thats really what hangup is supposed to do |
| 23:16.28 | cesau | with the alternate structure of originate "application <applicationdata>", and seeing "Dial" as a loaded application, would that suggest that you could: originate Local/1234 application Dial(SIP/provider/1231231234,10) ? |
| 23:16.47 | harry_v | [TK]D-Fender, I have been working on setting up paging configuration for my ast box. Do you know if call files can include more then one channel or make multiple call files for each channel? I have tried appending other SIP channels in the call file. Idealy, would like the setup to call all files as much as the CPU can handle and broadcast the message. |
| 23:16.47 | jplank | move the call past the rest of the priorities so it hits the end of the context |
| 23:16.51 | jplank | hmmmm |
| 23:17.06 | jplank | whats the CLI command to check the order of your priorities in your context? |
| 23:17.20 | *** join/#asterisk mphill (n=mphill@CPE-24-167-202-52.wi.res.rr.com) |
| 23:17.24 | jplank | could * be reading the h priority before the s priority? |
| 23:17.32 | [TK]D-Fender | harry_v: Channel: is just like what you would put in a Dial. |
| 23:18.12 | harry_v | yea, for some reason did not want to except that method. |
| 23:18.20 | harry_v | Ill take a look at it again. |
| 23:18.31 | [TK]D-Fender | harry_v: then pick another channel type |
| 23:18.50 | jplank | trapa, can you try something for me. put the h priority inside its own context, and just include it at the bottom of your context, that will make 100% sure its reading the h priority last |
| 23:19.03 | jplank | I don't know if that will change anything, but it will at least rule that out |
| 23:19.46 | trapa | jplank: Inside of it's own context? What do i call the context? |
| 23:19.56 | jplank | anything you want |
| 23:20.19 | jplank | just do |
| 23:20.26 | jplank | [endcall] |
| 23:20.26 | jplank | exten => h,1,NoOp("Done!"); |
| 23:20.34 | *** join/#asterisk prxtien (n=pro@ppp121-45-82-113.lns10.adl6.internode.on.net) |
| 23:20.36 | jplank | then at the bottom of your context put |
| 23:20.41 | jplank | include => endcall |
| 23:20.52 | harry_v | SIP/201&SIP/202&SIP/203 |
| 23:20.52 | harry_v | That is what I used last night |
| 23:20.54 | jpcansa | HI, i got a problem, telephone A calls B, then A transfers B to C, after that transfer, B still listens to MOH while C can listen to B. A and C are sip extensions in the same * while B is and outside Zap Channel. Any idea? |
| 23:21.09 | prxtien | how do i make asterisk perform a task when a call is hung up, asterisk puts my house into phoen mode when a call comes in, but i want it to go back into its previous mode after the call is hung up |
| 23:21.24 | harry_v | JP, put your dial plan on pastebin.ca |
| 23:21.34 | *** join/#asterisk AndyT (n=a@81-86-57-147.dsl.pipex.com) |
| 23:21.44 | trapa | jplank: OKay. done, and testing. |
| 23:22.07 | AndyT | anyone know how to suppress - Parsing '/etc/asterisk/manager.conf': Found - from the console? |
| 23:22.10 | laggo | my asterisk freezes during calls occasionally. the whole terminal becomes unresponsive (cant even ctrl-c out). it's probably one of the simplest configuration and dialplan possible, but i have no idea how to diagnose what's causing this. any suggestions would be helpful |
| 23:22.39 | jpcansa | harry_v: ok |
| 23:23.09 | harry_v | laggo, put your entire dial plan on pastebin including errors from debbiging then paste it here |
| 23:23.15 | trapa | jplank: Okay still executes h and shows "Done!" .. but then still goes to s,1 |
| 23:23.25 | laggo | harry_v, errors from debugging? |
| 23:23.27 | sipy | hey Harry |
| 23:23.33 | harry_v | ohh ho |
| 23:23.36 | harry_v | hi |
| 23:23.37 | harry_v | :) |
| 23:23.47 | jplank | prxtien: use hangup() and the h priority, hopefully you wont have the same issue as trapa |
| 23:23.51 | harry_v | your back at your office? |
| 23:23.53 | sipy | just got back in town |
| 23:23.56 | harry_v | nice |
| 23:24.08 | prxtien | jplank, i just noticed that, trying it now |
| 23:24.28 | sipy | laggo, doesn't sound like a dial plan issue. Sounds more like a hardware issue. what cards do you have in it x100p? |
| 23:24.34 | laggo | no cards, all IP |
| 23:24.36 | jplank | trapa, what about making a new inbound context and just have it match for a specific number, answer the call, play monkeys and then use the hangup command |
| 23:24.48 | harry_v | tzdummy installed laggo? |
| 23:24.56 | trapa | jplank can/will do. |
| 23:25.10 | laggo | harry_v: no |
| 23:25.15 | laggo | i cant recompile my kernel |
| 23:25.48 | jplank | you could try matching for _. , but DEF don't leave it like that |
| 23:25.55 | jplank | brb going to smoke a stog, let me know what happens |
| 23:26.19 | harry_v | I have never installed tzdummy but if kernel compile is a requirment may not have a choice |
| 23:26.40 | laggo | harry_v: but the issue isnt reproducible 100%, it happens sometimes and at different points in the call |
| 23:26.55 | laggo | harry_v: wouldnt tzdummy be a case of all or nothing |
| 23:27.27 | trapa | jplank: Didn't do any matching because this system is just my private system, so no biggie .. just made a new inbound context and renamed the old one ... Made it answer() playback(tt-weasels) and hangup |
| 23:27.45 | trapa | Now whats interesting is that calling it (try it yourself if you like) it plays tt-weasels twice, and then stops |
| 23:28.10 | *** join/#asterisk qdk_ (n=qdk@94.191.232.187.bredband.3.dk) |
| 23:28.22 | sipy | its a timing thing. if it freezes under certain instances timiong is involved that could be it. is sb accessable? |
| 23:28.35 | sipy | what do the logs say? |
| 23:28.41 | harry_v | think its irq issues? |
| 23:29.15 | sipy | with what? ethernet card? |
| 23:29.33 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 23:29.51 | sipy | Laggo, do a cat /proc/interrupts and see if the ethernet card is sharing irq with something else. I doubt it would be that though |
| 23:30.42 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:30.43 | laggo | doesnt look like it |
| 23:30.57 | AndyT | anyone know how do stop Parsing '/etc/asterisk/manager.conf': Found from spamming the console? |
| 23:31.07 | laggo | sipy, what is 'sb' |
| 23:31.11 | sipy | eth0 is on a line by itself? |
| 23:31.30 | laggo | yeah, it's under xen - is that an issue? |
| 23:31.33 | harry_v | sound blaster |
| 23:31.40 | hardwire | laggo: check out procinfo if you have it. procinfo -n1 -d |
| 23:31.47 | hardwire | that's my favorite way of finding conflicts |
| 23:33.51 | sipy | man procinfo |
| 23:33.58 | prxtien | im having a problem when i run a bash script from asterisk... if u su asterisk and run it from command line it works, when i run it from asterisk, it executes, but it doesnt turn my sound off like it should |
| 23:34.41 | harry_v | brb |
| 23:34.47 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 23:34.54 | Qwell | prxtien: turn your sound off? |
| 23:35.14 | prxtien | qwell, im automating my house and intergrating asterisk into that, so if i get an incoming phone call, it mutes the house speakers |
| 23:35.34 | Qwell | prxtien: have you looked into muted in the asterisk source? |
| 23:35.51 | laggo | well doesnt look like any irq's are shared |
| 23:36.11 | prxtien | Qwell, no i havnt mate |
| 23:36.38 | sipy | laggo, check the logs and try running debug and capture output when it freezes if possible |
| 23:39.12 | jplank | trapa what does the debug show? |
| 23:39.19 | jplank | does it show it hit the h priority then end? |
| 23:39.48 | trapa | I didn't include the h priority lemme include it and try again |
| 23:40.51 | trapa | Shows the h prority both times. But the second time it actually stops at it |
| 23:41.11 | jplank | can I see? |
| 23:41.42 | jplank | the actual context and the debug |
| 23:41.44 | trapa | jplank : Yup gimme a sec |
| 23:43.15 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 23:43.25 | trapa | extensions on top, debug on bottom http://pastebin.com/d4a2b1dfb |
| 23:44.36 | jplank | do it without the s |
| 23:44.59 | trapa | how? Don't i need the s for it to answer? |
| 23:45.12 | jplank | thats why I said to match to a number for testing |
| 23:45.39 | trapa | Oooh! ... hrm ... I'm not sure i know how to do that just exten => 7782161820,1,Answer; |
| 23:46.05 | jplank | do _. it will match for everything |
| 23:46.24 | jplank | just don't leave it like that |
| 23:47.05 | trapa | jplank: Same result as before. |
| 23:47.26 | harry_v | sipy, you have worked with jail before? |
| 23:48.42 | jplank | give me one sec, I'm going to try it on my box |
| 23:49.56 | *** join/#asterisk mnicholson_ (n=matthew@72.146.52.161) |
| 23:52.24 | *** join/#asterisk etherealite_ (n=evan@adsl-75-18-203-219.dsl.pltn13.sbcglobal.net) |
| 23:54.04 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:55.49 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 23:55.51 | Dovid | hi |
| 23:56.09 | Dovid | is the wait waitexten command supposed to work with the h extension ? |
| 23:56.33 | jplank | trapa |
| 23:56.39 | trapa | Yup? |
| 23:56.40 | jplank | it seems to work for me... |
| 23:56.49 | Dovid | not working for me. just fails there |
| 23:56.57 | jplank | let me set it with _. just to test |
| 23:57.01 | trapa | damn .. that sucks .. I am wondering if it's my voip provider causing the issue |
| 23:57.04 | Dovid | oh. thought u were talking to me ;) |
| 23:57.13 | harry_v | Dovid, can i see some of your dial plan with the WaitExten in it? |
| 23:57.26 | jplank | actually, I'm going to try it with s |
| 23:58.22 | jplank | ohhhhh |
| 23:58.25 | jplank | I'm a fool |
| 23:58.33 | jplank | it will never work with _. or s |
| 23:58.40 | laggo | blah im going to keep messing with this to reproduce the bug |
| 23:58.40 | jplank | well |
| 23:59.14 | jplank | it won't work with _. because that matches ANYTHING, including h,2 (which is what it would go to after h,1) |
| 23:59.23 | jplank | and I suspect the same with s |