| 00:08.05 | ManxPower | scooby2: plug in a loopback plug |
| 00:08.18 | ManxPower | you can do a software loopback via zttool |
| 00:08.34 | scooby2 | obviously but I would like to go home and not have to sit here waiting for the telco:) |
| 00:08.35 | ManxPower | but that is mainly useful only for the telco testing the lines. |
| 00:08.48 | ManxPower | a loopback plug will test the card, port, and config |
| 00:09.42 | scooby2 | correct |
| 00:10.01 | scooby2 | but sometimes they like it looped back the other way to see if they can see it. If its a line issue or not |
| 00:14.13 | ManxPower | scooby2: they can loop to the smartjack. |
| 00:14.17 | ManxPower | that is enough for line testing |
| 00:14.41 | *** join/#asterisk SuxieDapper (n=SuxieDap@190.245.108.2) |
| 00:14.44 | SuxieDapper | Hello |
| 00:15.23 | SuxieDapper | Does anybody know if I can send SMS with my Vonage account? |
| 00:15.37 | drmessano | WUT |
| 00:15.50 | ManxPower | SuxieDapper: no you cannot do that |
| 00:16.00 | ManxPower | Asterisk SMS only support the European SMS standards |
| 00:16.56 | SuxieDapper | Somebody told that I could use my Vonage account to send SMS with it. But I wasn't sure if it was using Asterisk, some Vonage site or something like that. |
| 00:17.04 | drmessano | Yeah, Asterisk doesn't support 568A SMS |
| 00:17.09 | drmessano | only 568B |
| 00:17.15 | drmessano | Sorry |
| 00:17.37 | C4colo | ok, g729 question |
| 00:17.54 | C4colo | let's assume I"m not in latvia and I need to pay for the licenses |
| 00:17.56 | scooby2 | and speak of the devil. GBLX circuit is now up/up. |
| 00:18.14 | SuxieDapper | Thank you guys |
| 00:18.24 | C4colo | I have a sip gateway, a call router, and a registration/feature server |
| 00:18.28 | C4colo | the call hits all three systems |
| 00:18.46 | C4colo | (I have multiples of each, but for the sake of simplicity let's assume I have only these three systems) |
| 00:19.16 | drmessano | Like in series? |
| 00:19.27 | C4colo | do I need to license g.729 on EVERY system even though the call will be routed through these systems |
| 00:19.29 | C4colo | yea in series |
| 00:19.38 | drmessano | Are any of them transcoding? |
| 00:19.46 | C4colo | first and last might |
| 00:19.47 | drmessano | Remember it's a "transcoding license" |
| 00:19.57 | drmessano | So anywhere you're transcoding |
| 00:20.04 | C4colo | hmm |
| 00:20.28 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:20.36 | C4colo | so sip phone (729) -> asterisk -> asterisk -> asterisk -> PSTN |
| 00:20.56 | C4colo | the last one would need a license to transcode to the PSTN right? |
| 00:21.17 | drmessano | I believe so, since Asterisk is doing media conversion |
| 00:21.34 | C4colo | how does one configure the other servers to pass g.729 through without needing a license? |
| 00:21.48 | C4colo | just install the codec? |
| 00:22.42 | drmessano | I know "out of the box" * supports passthru.. |
| 00:22.49 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:22.52 | C4colo | oh? |
| 00:22.55 | C4colo | hmm |
| 00:22.57 | C4colo | never tried it |
| 00:23.41 | drmessano | I dont know how to explain it.. But it will pass G729 but not transcode.. I guess it comes with a Passthru codec (like the early G722 support) |
| 00:24.02 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 00:24.35 | drmessano | I'm gonna assume the licensed codec is smart enough to count properly |
| 00:24.54 | C4colo | yea |
| 00:25.07 | C4colo | I'll have to do some testing |
| 00:25.29 | drmessano | A default install will lock you into passthru only.. which actually may be better for testing |
| 00:25.46 | drmessano | Easier for it not to work than to keep watching the CLI, i guess |
| 00:28.25 | *** join/#asterisk pixlated (n=not@ool-44c0746a.dyn.optonline.net) |
| 00:33.18 | *** part/#asterisk jan1607 (n=niewerth@dslb-084-061-121-080.pools.arcor-ip.net) |
| 00:41.40 | *** join/#asterisk hi365_m (n=hi365@213.151.44.101) |
| 00:57.07 | *** join/#asterisk Itiliti (n=Itiliti@216.159.238.110) |
| 01:02.57 | Itiliti | I am trying to get faxing working through my PRI. I am running spanDSP spandsp-0.0.4-9.pre15.i386, and app_nv_faxdetect-1.0.6_1.4 and background detect of the same version. But when it detects the fax and kicks it over to the rx_fax app, it crashes asterisk? Any ideas? |
| 01:03.27 | Itiliti | this is only happening when calls coming in on PRI.. |
| 01:04.35 | *** join/#asterisk marc7 (n=marc@S0106001ff33f8523.vc.shawcable.net) |
| 01:04.53 | Itiliti | .join asterisk-dev |
| 01:05.32 | *** join/#asterisk cnielsen (n=Cole@209.181.98.68) |
| 01:06.39 | cnielsen | Can anyone tell me how to resolve an issue with Asterisk rejecting calls due to codecs on FC7 ? |
| 01:06.54 | *** join/#asterisk Compy (n=compy@h123.192.18.98.dynamic.ip.windstream.net) |
| 01:06.57 | cnielsen | I'm using the FC7 packages |
| 01:07.34 | tmckay | I'm using ekiga.net for sip and trying to configure IPKall with it. What do I enter for my SIP proxy? |
| 01:07.45 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 01:07.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:08.05 | v4mp | hey guys any idea to this after u get the audio menu options it hangs up rather than w8ing for u to pick an option |
| 01:08.06 | v4mp | http://pastebin.com/d9d2f949 |
| 01:08.34 | *** part/#asterisk cnielsen (n=Cole@209.181.98.68) |
| 01:12.23 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 01:12.30 | v4mp | this is my extensions.conf http://pastebin.com/d5bfe77b7 |
| 01:13.25 | l2trace99 | ~softphones |
| 01:13.27 | l2trace99 | crap |
| 01:13.29 | l2trace99 | there's not jbot response for softphone recommendations ? |
| 01:14.43 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:15.40 | *** join/#asterisk moy (n=moy@189.169.68.109) |
| 01:17.37 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 01:17.37 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 01:17.44 | v4mp | ~softphones |
| 01:17.56 | v4mp | l2trace99, u tyed x-lite ? |
| 01:18.25 | l2trace99 | yes |
| 01:18.38 | subdolus | is there a way to send a DTMF tone down the line once a call has been established with asterisk? |
| 01:18.47 | jeff_smoker | If you receive a gsm based call on a COM port, can you enter AT commands to forward that call once its been picked up? |
| 01:19.02 | l2trace99 | i was looking other alternatives that support multiple accounts |
| 01:19.36 | l2trace99 | and that I can remotely configure |
| 01:20.20 | v4mp | hmm i used to know of 1 but i cant remember name of it no more |
| 01:24.58 | l2trace99 | mmmmm Ekiga for windows |
| 01:26.48 | *** join/#asterisk mackes (n=root@cpe-76-180-145-138.buffalo.res.rr.com) |
| 01:27.10 | *** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 01:34.39 | jeff_smoker | subdolus: what are you trying to accomplish? |
| 01:35.51 | subdolus | i want to idle in a teleconference, but the conference kicks people if they dont press a button (send a DTMF tone) within a certain period of time |
| 01:36.17 | subdolus | so i want it to, say, every 2 minutes press a button / send DTMF |
| 01:36.26 | subdolus | possible? |
| 01:37.22 | jeff_smoker | subdolus: I think it is, but I haven't worked with asterisk enough to know. i think you might be able to write a looping php script to do this. |
| 01:37.53 | [TK]D-Fender | subdolus: Tricky to set up, but quite doable |
| 01:38.51 | [TK]D-Fender | ~softphone |
| 01:38.51 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 01:45.39 | jeff_smoker | does anyone know of a voip service provider that lets you make a direct socket connection to their voip server? |
| 01:46.28 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 01:46.59 | subdolus | kicks router |
| 01:47.15 | subdolus | joker, you still around matey? |
| 01:48.20 | Compy | ~xlite |
| 01:48.21 | jbot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 01:49.28 | [TK]D-Fender | jeff_smoker: ....HUH? |
| 01:49.47 | Carlos_PHX | Um, yeah, why would you do that? |
| 01:50.51 | [TK]D-Fender | Forget "why"... more like "what". As in "what is that supposed to mean?" |
| 01:52.00 | LiNeTuX | You know, a plug-in for the socket. So you can port your XML over SIP for real-time Web 2.0. |
| 01:52.16 | [TK]D-Fender | With *, SIP & RTP is all UDP. Thats right, CONNECTIONLESS. So did he meam to imply some sort of VPN-like connection over which he would alter pass that same traffic? We may never know.... |
| 01:53.17 | Carlos_PHX | Huh, interesting idea actually. Like tunneling UDP in TCP without the overhead of GRE? |
| 01:53.20 | [TK]D-Fender | Just hearing someone saying "but I'm connected to the provider" makes me look at them funny... |
| 01:53.32 | Carlos_PHX | Heh |
| 01:53.39 | [TK]D-Fender | Carlos_PHX: Possible... not great on reliability, jitter, etc..... |
| 01:53.52 | LiNeTuX | why would you want TCP overhead? <shrug> |
| 01:54.26 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:55.02 | Carlos_PHX | You get back information on the state of your connectivity. I've only lightly experimented with SIP over TCP but there are some benefits. I'm not saying it's great, just that there may be reasons. |
| 01:55.05 | *** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net) |
| 01:55.10 | jeff_smoker | TDK: similar to how you can make a direct socket connection to a POP server, can you make a direct socket connection to a VOIP server and then just enter commands? |
| 01:55.42 | [TK]D-Fender | With *, SIP & RTP is all UDP. Thats right, CONNECTIONLESS. So did he meam to imply some sort of VPN-like connection over which he would alter pass that same traffic? We may never know.... |
| 01:55.46 | [TK]D-Fender | jeff_smoker: ^^^ |
| 01:55.58 | Compy | ick, voice over tcp... *quivers* |
| 01:56.00 | [TK]D-Fender | jeff_smoker: there aren't jsut "commands". |
| 01:56.23 | [TK]D-Fender | jeff_smoker: Can you try to describe a specific scenario of what it is your looking to do... |
| 01:56.24 | LiNeTuX | You might imply the commands are in the UDP packets. |
| 01:57.08 | [TK]D-Fender | jeff_smoker: SIP is not a scripting language, its a call setup and teardown protocol. |
| 01:57.32 | jeff_smoker | So if you want to make VOIP calls without asterisk, you still need to pay some company that has a voip server connection to a pstn trunk, right? |
| 01:58.02 | [TK]D-Fender | jeff_smoker: Asterisk has nothing to do with the PSTN in and of itself. |
| 01:58.31 | [TK]D-Fender | jeff_smoker: ITSP's use their protocols and accounts to allow you access to the PSTN. |
| 01:58.40 | [TK]D-Fender | jeff_smoker: * is not needed for this. |
| 01:58.49 | jeff_smoker | ok, right |
| 01:58.55 | [TK]D-Fender | jeff_smoker: You could cset up a softphone direct just the same |
| 01:59.03 | jeff_smoker | exactly, ok |
| 01:59.20 | [TK]D-Fender | jeff_smoker: Again, these are CALLS, not just some kind of arbitrary "command" |
| 01:59.40 | jeff_smoker | So what to know is if you write your own softphone application, you have to write a connection script to the ISTP yes or no? |
| 01:59.51 | jeff_smoker | correction: what I want to knw |
| 01:59.53 | jeff_smoker | konw |
| 01:59.55 | jeff_smoker | know |
| 02:00.20 | [TK]D-Fender | jeff_smoker: and "making voip calls" doesn't require anything other than 2 pieces of software communicating with each other. You can point 1 softphone DIRECTLY at another an bam, thats a "voip call" |
| 02:01.02 | LiNeTuX | jeff: SIP is a protocol. If you write an application to use SIP, you'd be able to connect to any provider that uses SIP. |
| 02:01.07 | [TK]D-Fender | jeff_smoker: 1 : there is no CONNECTION. No such thing. Period. |
| 02:01.18 | Carlos_PHX | SIP is connectionless. |
| 02:01.23 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 02:01.40 | jeff_smoker | but if you have to get access to the pstn, you have to make a connection to something right? |
| 02:01.43 | Carlos_PHX | I say "INVITE" and you reply, and I reply, and then we both start shooting voice packets. |
| 02:01.48 | Carlos_PHX | That's all. |
| 02:01.56 | Carlos_PHX | jeff_smoker: NO |
| 02:02.04 | jeff_smoker | ?? |
| 02:02.04 | [TK]D-Fender | jeff_smoker: You place your call. They ask for auth. You give it. They say OK. You both set up RTP. You Talk. You say "I'm done", and thats the end of it |
| 02:02.18 | Carlos_PHX | You are shooting packets at a PSTN gateway, but you don't literally "connect" in the networking sense. |
| 02:02.23 | [TK]D-Fender | jeff_smoker: You need to go read a VoIP protocol primer guide. |
| 02:02.28 | Carlos_PHX | Yup |
| 02:02.36 | jeff_smoker | so why use a voip service provider? |
| 02:02.41 | jeff_smoker | what role do they play? |
| 02:02.48 | Carlos_PHX | You send your packets to them to send to the PSTN. |
| 02:02.51 | [TK]D-Fender | jeff_smoker: because THEY will bridge your call to the PSTN. |
| 02:03.13 | Carlos_PHX | jeff_smoker: Let me try to say it another way. |
| 02:03.18 | jeff_smoker | Ok |
| 02:03.23 | Carlos_PHX | "Connection" in the context of networking means something specific. |
| 02:03.29 | [TK]D-Fender | jeff_smoker: You can feel free to pass all the VoIP traffic around the world that you want but it won't magically get onto the PSTN unless someone is providing you termination <- |
| 02:03.31 | Carlos_PHX | SIP is connectionless. |
| 02:03.58 | Carlos_PHX | "Connection" in layman's parlance could be what you mean, IE you are negotiating with a PSTN gateway to connect you to the PSTN. |
| 02:04.11 | Carlos_PHX | But at the networking level there is no connection. |
| 02:04.31 | jeff_smoker | Carlos, are you in Phoenix? |
| 02:04.37 | Carlos_PHX | Yes |
| 02:04.44 | jeff_smoker | I'm in Tempe |
| 02:04.49 | Carlos_PHX | Small world |
| 02:04.57 | jeff_smoker | Know of any club meetings that have to do with VOIP? |
| 02:05.13 | Carlos_PHX | I've been talking with cohorts about setting something up, but nothing so far. |
| 02:05.31 | Carlos_PHX | However a few of us are open to beer and wings to talk VoIP here and there. |
| 02:05.46 | jeff_smoker | Well, I'd be very interested. |
| 02:05.52 | jeff_smoker | I'm at ASU |
| 02:06.14 | jeff_smoker | I just sold my party bus, otherwise I'd throw some girls in and host |
| 02:06.29 | Carlos_PHX | Shoot me an e-mail at carlos at televolve.com and I'll drop you a line some time. |
| 02:06.33 | Carlos_PHX | Heh |
| 02:08.55 | LiNeTuX | Carlos_PHX: I'm jealous of you guys... you have The Yard House... |
| 02:09.03 | jeff_smoker | Carlos_PHX: Alright, thank...I just sent you an email. |
| 02:09.08 | LiNeTuX | doesn't have good beer on tap where he is. :( |
| 02:09.13 | jeff_smoker | thank(s) |
| 02:10.57 | jeff_smoker | Alright, so if you have a softphone you can connect to basically anyone else who has a softphone as long as you point to them. But as far as PSTN call termination goes, that's when you have to pay someone $.019 / minute or whatever. And you guys are saying what protocol does that? |
| 02:11.57 | [TK]D-Fender | jeff_smoker: the protocol is the carrier. There are MANY VoIP protocols. SIP, IAX2, H.323, MCGP, Jabber, Skype, and so on... |
| 02:12.40 | jeff_smoker | So why choose one over the others? |
| 02:12.41 | [TK]D-Fender | jeff_smoker: it does not matter WHICH one this "provider" uses. Naturally though by choosing to acquire their service one would think that they would tell you how to actually USE it. |
| 02:13.20 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 02:13.30 | [TK]D-Fender | jeff_smoker: 1 provider over another? Maybe one provider has cheaper rates to a country you want to call more often. Maybe one provider is cheap but only lets you have 1 channel at a time when you need to support multiple. |
| 02:13.33 | UnixDawg | so is the new asterisk-now iso out ? |
| 02:13.41 | jeff_smoker | Other than voice quality and reliability, what are other considerations when selecting a provider? |
| 02:13.48 | jeff_smoker | and price of course |
| 02:13.56 | UnixDawg | the one the had downloadable at astricon ? |
| 02:14.01 | [TK]D-Fender | jeff_smoker: Maybe some providers utilization rates such and they run out of lines to terminate your calls so you can't even GET to the PSTN |
| 02:14.24 | [TK]D-Fender | jeff_smoker: Or their servers are in a bad location and jitter, latency etc screw with your calls |
| 02:14.27 | jeff_smoker | What providers do you use? |
| 02:14.38 | jeff_smoker | Could you become your own provider? |
| 02:14.46 | LiNeTuX | jeff_smoker: latency is a big key in choosing a provider that works for you. |
| 02:14.46 | [TK]D-Fender | jeff_smoker: Why are there so many cellphone carriers? because some suck more than others <- |
| 02:15.28 | [TK]D-Fender | jeff_smoker: Sure. You have an analog line? get a piece of hardware that lets you connect your line and something that will accept VoIP calls and use it <- |
| 02:15.38 | [TK]D-Fender | jeff_smoker: BTW... that would be ASTERISK |
| 02:15.43 | jeff_smoker | Is there any kind of definitive resource on the net that tracks the average latency of providers? Or is this just something you have to test on your own. |
| 02:15.55 | [TK]D-Fender | jeff_smoker: Have to test for yourself. |
| 02:15.59 | Compy | Are there any other decentralized carriers out there besides skype? |
| 02:16.02 | LiNeTuX | jeff_smoker: YMMV. |
| 02:16.08 | Compy | has wondered that |
| 02:16.28 | [TK]D-Fender | jeff_smoker: How is every testing company going to know how bad YOUR connection will suck? Do they have computers on evvery network? In every segment? |
| 02:16.30 | LiNeTuX | jeff_smoker: Latency is a factor of what is between you and them. |
| 02:17.03 | jeff_smoker | LiNeTux: I see. |
| 02:17.06 | LiNeTuX | Compy: of course. VoicePulse (one example) has at least 4 locations you can 'pick'. |
| 02:17.11 | [TK]D-Fender | jeff_smoker: common sense answer just about all of these questions. |
| 02:18.34 | LiNeTuX | jeff_smoker: ex: I can get to provider "A" from my house in 80ms. I can get to the same provider from my colo server in 20ms. That box is physically only 10 miles from me, but with another (much better) provider that has better perring arrangements than my home ISP. |
| 02:18.37 | jeff_smoker | [TK]D-Fender: So you're saying that providers have literally thousands of PSTN termination numbers? |
| 02:18.42 | Compy | LiNeTuX: hmm, that doesn't exactly strike me as decentralized/mesh... |
| 02:19.10 | mosty | is there a way to do GotoIfTime for a specific timezone that's different to the system timezone? |
| 02:19.11 | [TK]D-Fender | jeff_smoker: yes |
| 02:19.33 | LiNeTuX | Comy: depends on how you do it. You can use a single DNS entry and do much trickery based on location to route/reroute/failover. you can also use SIP gateways to help load balance and other such thigns. |
| 02:19.43 | [TK]D-Fender | jeff_smoker: they are each a small telephone company so THOUSANDS of calls at a time |
| 02:20.14 | [TK]D-Fender | mosty: AGI. Have fun :) |
| 02:20.39 | mosty | [TK]D-Fender, yeah i figured |
| 02:21.25 | Compy | LiNeTuX: yeah, i know you can do that, just wondering. Skypes is for the most part fully peer based, just wondering if any others had picked up on that kind of architecture. |
| 02:21.26 | jeff_smoker | [TK]D-Fender: How do they operate the lines so cheap? |
| 02:22.18 | LiNeTuX | Compy: trust me, Skype isn't the first to do this - and remember, they went completely down not too long ago due to some "windows updates" <cough>. So even the best plans can fail. |
| 02:22.21 | jeff_smoker | [TK]D-Fender: I mean, when it costs you $15 to get a basic phone number...they can have literally thousands of phone numbers for far cheaper, right? |
| 02:22.34 | [TK]D-Fender | jeff_smoker: Are you really expecting an answer to "Wow, can you explain exactly how phone companies work in detail?" |
| 02:22.39 | LiNeTuX | jeff_smoker: a phone number doesn't mean it's a phone line. |
| 02:22.49 | LiNeTuX | heh |
| 02:23.25 | [TK]D-Fender | jeff_smoker: Lets say I have 5 lines. I then take on TWENTY customers on the assumption that no more that 25% of them will actually even be on the phone at a given time <- |
| 02:23.36 | Compy | LiNeTuX: oh ofcourse, I assumed they werent the first, hence I asked if you knew of any others. But yeah, I remember the massive outage when they lost all their supernodes, lol. |
| 02:23.48 | tzanger | [TK]D-Fender: sounds like erlang equations |
| 02:23.58 | [TK]D-Fender | jeff_smoker: why do you think phone systems lock up in major emergencies? Because the telcos themselves can't survive huge concurrency |
| 02:24.04 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 02:24.33 | LiNeTuX | Compy: Skype does do it differently than most... their model is very akin to how Hamachi works if you're familiar with them. |
| 02:24.39 | [TK]D-Fender | jeff_smoker: its all about percentages. When a company offers an "unlimited" plan, do you think they do this blindly? |
| 02:24.58 | [TK]D-Fender | jeff_smoker: this is big math. |
| 02:25.37 | jeff_smoker | [TK]D-Fender: So YOU CONNECT to the PROVIDER via VOIP...the PROVIDER routes your VOIP connection to the PSTN as a normal phone call and still can only put one call through each line at a time, is that right? |
| 02:25.44 | Compy | LiNeTuX: yeah, they use a variant of the fasttrack protocol, they literally route everything over the mesh, even logins. |
| 02:26.28 | [TK]D-Fender | jeff_smoker: they can accept as manny calls from you at a time as they feel like offering you based on what you're paying for. |
| 02:26.32 | mosty | jeff_smoker, no. digital telephone lines that telephone companies use aren't the same as a single line/phone number you can get at home |
| 02:26.48 | jeff_smoker | mosty: how so? |
| 02:27.04 | [TK]D-Fender | jeff_smoker: now is the point to take the word "line" and throw it right out the window. We are in a world of CHANNELS |
| 02:27.23 | mosty | jeff_smoker, look up ISDN on wikipedia |
| 02:27.40 | LiNeTuX | Compy: SIP has other such devices - like SIP gateways (can't recall the name of them at the moment)... plus you can give SIP devices 'backup' registration info as well... so it's very flexable. |
| 02:27.58 | jeff_smoker | Ok, so how channels can I use concurrently on my GSM mobile phone, 2? What about if I start sending AT commands to a voice modem...how many channels can I use? |
| 02:28.09 | jeff_smoker | (how many channels) |
| 02:28.15 | [TK]D-Fender | jeff_smoker: a "line" is a piece of wire. In the case of an analog phone line in your home it has a ver fixed signalling on it. It can only support 1 call at a time. the telco switch can conference in on its end, but coming in to you is 1 audio stream period. |
| 02:28.36 | [TK]D-Fender | jeff_smoker: leave MODEMS out of this. |
| 02:28.53 | [TK]D-Fender | jeff_smoker: if you are talking DATA, then that is where voip comes in, and that is based on bandwidth |
| 02:29.13 | [TK]D-Fender | jeff_smoker: and through your concept of "AT commands out along with. |
| 02:29.57 | Compy | yeah, LiNeTuX. I have and still am considering those options. I guess I will just have to design a protocol to offload this traffic onto commodity hardware-based peers. |
| 02:30.08 | jeff_smoker | [TK]D-Fender: Ok. I'm just wondering why you couldn't put make multiple calls on your cellphone by utilizing a bunch of channels at once? What's the limitation? |
| 02:30.32 | LiNeTuX | Compy: there were some good talks at Astricon around this subject. |
| 02:30.49 | [TK]D-Fender | jeff_smoker: if you're talking about GSM as a data transmission protocol for raw data, only the lack of hardware designed to talk that way. |
| 02:31.05 | Compy | bummer that I missed it |
| 02:31.23 | LiNeTuX | "Why can't I get HDTV on my old TV"? |
| 02:31.42 | pcrane | jeff_smoker: are you looking at initiating a call on asterisk, and having it go out via mobile phone? |
| 02:31.43 | jeff_smoker | [TK]D-Fender: But if you're talking about GSM for voice transmission on multiple channels in the same sense that the voip providers use ISDN like mosty says |
| 02:32.15 | [TK]D-Fender | jeff_smoker: What GSM device do you have that supports multiple simultaneous channels? |
| 02:32.41 | LiNeTuX | Compy: Leif Madsen, who's in here from time to time, knows a lot about that subject. Well, he helped write "the book"...so he should :) |
| 02:33.31 | [TK]D-Fender | Leif is Leif! Na na na na na! |
| 02:34.42 | Compy | LiNeTuX: excellent :) I will also scour for some info concerning this topic |
| 02:35.45 | LiNeTuX | had lunch with Leif at Astricon and didn't even know who he was :) |
| 02:36.05 | LiNeTuX | (at least by sight!) |
| 02:36.49 | *** join/#asterisk dmz (n=dmz@12.25.86.34) |
| 02:36.52 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 02:37.29 | jeff_smoker | pcrane:Good question. I was working with AT commands on my Samsung SGH-t639 mobile phone which the AT commands indicate does not support voice transmission over the modem. I wanted to circumvent asterisk because I don't know linux, and put together my own pbx using php and apache...and receiving voice directly over the COM ports on a server. But now I'm moving away from this and towards VOIP |
| 02:38.06 | *** join/#asterisk cryptnix (n=andrew@firewall.kasltechnology.net) |
| 02:38.32 | [TK]D-Fender | jeff_smoker: Voice is not a web page. your concepts of "com" don't apply here. |
| 02:38.38 | jeff_smoker | pcrane: Because I can't seem to find any evidence that shows a gsm voice modem can receive and forward more than one voice call at a time |
| 02:38.53 | pcrane | I'd have thought that it would depend on the phone |
| 02:38.56 | tzanger | jeff_smoker: I'm not aware of any that can |
| 02:39.12 | jeff_smoker | [TK]D-Fender: What do you mean? |
| 02:39.22 | [TK]D-Fender | jeff_smoker: And AT commands are jsut a way of telling a modem to dial. from there it is jsut a data link. Not even a packet interface by definition. |
| 02:39.28 | pcrane | e.g. I've got a motorolla e770 which has a different feature set (via AT commands) than a nokia 6121 classic |
| 02:40.07 | [TK]D-Fender | tzanger: Plenty of crack floating around here... |
| 02:40.07 | jeff_smoker | [TK]D-Fender: I became curious about this when somebody mentioned chan_bluetooth to me |
| 02:40.08 | Compy | LiNeTuX: thanks a bunch. I'll definitely document my progress on this. |
| 02:40.17 | tzanger | [TK]D-Fender: heh :-) |
| 02:40.56 | [TK]D-Fender | jeff_smoker: Yes, well Bluetooth IS a packet protocol and gee, I dunno.. there are HEADSETS out there for it for some strange reason, which seems to have the word "voice" hanging over it for some other equally mysterious reason. |
| 02:41.13 | tzanger | you also don't use AT commands to transfer voice data to bluetooth headsets |
| 02:41.38 | [TK]D-Fender | Somebody is definitely stuck in 1990 in the BBS years. |
| 02:41.48 | jeff_smoker | tzanger: No, but it appears you can use them to answer, dial, and join calls. |
| 02:41.52 | [TK]D-Fender | Get back into your Delorean. |
| 02:41.54 | subdolus | jeff_smoker: my router cracked the shits earlier... do you know of a command to send DTMF down a channel? |
| 02:41.54 | tzanger | they were fun times, I can't balme him :-) |
| 02:42.13 | [TK]D-Fender | subdolus: "core show application senddtmf" <- |
| 02:42.24 | tzanger | jeff_smoker: again, I am almost positive that the calls are joined at the telco switch and not the cell phone |
| 02:42.48 | [TK]D-Fender | tzanger: Yes, I was a sysop back to the 80's, ran my own, wrote my own, and when the internet came to the common people I LET GO. |
| 02:43.11 | jeff_smoker | tzanger: I see. |
| 02:43.24 | tzanger | [TK]D-Fender: agreed, I did too, but there was something very cool about it being local instead of global |
| 02:43.36 | [TK]D-Fender | jeff_smoker: And for chan_bluetooth, its a single voice channel, just like your headset to your cellphone. There is no magic there. |
| 02:44.12 | [TK]D-Fender | tzanger: I know..... I occasionally hit onne of the TW2002 telnet sites from time to time just for nostalgia ;) |
| 02:44.13 | Nugget | telnet is eeeeeeevil! |
| 02:44.20 | [TK]D-Fender | pets the nugget-bot |
| 02:44.29 | tzanger | :-) |
| 02:44.35 | jeff_smoker | So in other words, you guys are saying that you CANNOT just hook up a bunch of mobile phones to your laptop and transfer and dial calls back and forth between them? |
| 02:45.14 | Qwell | I don't know about transfer |
| 02:45.23 | [TK]D-Fender | jeff_smoker: Well if you have a pile of BT receivers on your server the concept is quite possible. |
| 02:45.27 | Nugget | eyes [TK]D-Fender |
| 02:45.29 | tzanger | jeff_smoker: sure, with bluetooth if the phone has the appropriate profiles, or with a bunch of sound cards and usb cables. |
| 02:45.37 | [TK]D-Fender | jeff_smoker: And you can get GSM direct cards as well |
| 02:45.45 | [TK]D-Fender | hands Nugget a spoon. |
| 02:46.20 | jeff_smoker | Ok, but you guys are saying you can't split these phones into multiple channels so that they can handle multiple calls at once? |
| 02:46.38 | Qwell | jeff_smoker: no.. |
| 02:46.42 | [TK]D-Fender | jeff_smoker: So aside from all of this field&stream of consciousness are you actually fishing for something specific, or just kinda ramblimg? |
| 02:46.46 | Qwell | I've never seen a cell phone that can do multiple calls at once |
| 02:47.15 | [TK]D-Fender | jeff_smoker: Again, what kind of phone have you ever seen that lets you talk to 3 people INDEPENDANTLY? |
| 02:47.17 | jaytee | Field and Stream is good reading for the crapper |
| 02:47.23 | tzanger | heh |
| 02:47.33 | [TK]D-Fender | jaytee: nice to see you appreciated the mixed metaphor ;) |
| 02:47.42 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 02:47.43 | tzanger | [TK]D-Fender: it's one of the new ones; they're made of unobtanium |
| 02:48.02 | tzanger | I've got an iphone and it won't even do that :-p |
| 02:48.03 | jaytee | although I also am fond of It Pays To Enrich Your Wordpower in Reader's Digest. |
| 02:48.27 | [TK]D-Fender | jaytee: Now imagine that my normal state of consciousness operations on 2+ tiers more often than not :) God it sucks trying to communicate with mortals sometimes ;) |
| 02:48.40 | [TK]D-Fender | tzanger: lol! |
| 02:49.06 | [TK]D-Fender | tzanger: on the "period"ic table, its the one that plays "hard to get", isn't it? ;) |
| 02:50.49 | jaytee | [TK]D-Fender, I don't have a background in psychology but I've detected what I believe may be a streak of masochism and martrydom in your psyche. You continue to "suffer fools gladly nite after nite. |
| 02:51.32 | tzanger | [TK]D-Fender: absolutely |
| 02:51.36 | [TK]D-Fender | jaytee: I won't let it kill me, so martyrdon is out. |
| 02:52.04 | [TK]D-Fender | jaytee: "Why do we always come here? I guess we'll never know. It's like a kind of torture... TO HAVE TO WATCH THIS SHOW!" |
| 02:52.07 | jaytee | And I've actually witnessed you working on a few noobs and literally beating the sense into their thick skulls. |
| 02:52.23 | [TK]D-Fender | "ANIMAL'S" OUT |
| 02:52.42 | tzanger | [TK]D-Fender has amazing patience |
| 02:52.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:53.17 | [TK]D-Fender | tzanger / jaytee : Its been a real strain lately and you've seen it run out FAST in some cases. |
| 02:53.50 | [TK]D-Fender | tzanger: It hurt my previous rep and was a wake-up call as it wasn't restricted to jsut in here. |
| 02:54.24 | tzanger | I bet, you are one of a handful of regulars in here who don't seem to give up and lurk after a while... like me, for example. |
| 02:54.59 | tzanger | you, manxpower (who's not here?!), qwell... those are probably the top three active non-paid digium people here |
| 02:55.05 | [TK]D-Fender | tzanger: No, I don't give up, and thats the problem. Burn out. |
| 02:55.08 | Qwell | tzanger: ... |
| 02:55.13 | Qwell | I AM paid-Digium |
| 02:55.17 | tzanger | oh |
| 02:55.24 | tzanger | I did not know that :-) |
| 02:55.25 | Qwell | have been for like 2 years :p |
| 02:55.33 | jaytee | Manxpower isn't paid Digium |
| 02:55.33 | tzanger | and I am an idiot |
| 02:55.36 | tzanger | I know he isn't |
| 02:55.41 | tzanger | I didn't think qwell was though |
| 02:55.56 | Carlos_PHX | Well, I'm a noob to the channel, not to Asterisk. But I reserve the right to ask stupid questions anyway. I'm gonna go have a couple beers and see what I can think up. |
| 02:56.05 | tzanger | so does digium have all their new customer service people hang out in here for 3 months as a training period? :-) |
| 02:56.16 | Qwell | tzanger: good idea :p |
| 02:56.28 | LiNeTuX | is already on his way to the 2+ beer limit. |
| 02:58.18 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 02:59.08 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
| 03:01.37 | drmessano | Oh lord |
| 03:01.49 | drmessano | No more Digium new employees... please |
| 03:01.49 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 03:02.06 | Qwell | drmessano: go test |
| 03:02.16 | drmessano | lol |
| 03:02.18 | Qwell | I put up new versions of stuff today |
| 03:02.26 | drmessano | o.o |
| 03:02.27 | Qwell | new ISOs too |
| 03:02.30 | drmessano | O.O |
| 03:02.38 | Qwell | just updates stuff |
| 03:03.00 | drmessano | In the words of Donald duck |
| 03:03.07 | drmessano | ZOMGGGG TORRENT PLZ |
| 03:03.11 | drmessano | hang on |
| 03:03.15 | tzanger | drmessano: "rabbit season" ? |
| 03:03.16 | Qwell | btw, if anybody wants a super-exclusive pre-beta of AsteriskNOW, lemme know :p |
| 03:03.38 | LiNeTuX | raises his hand |
| 03:03.56 | jaytee | is it still using that custom distro? |
| 03:03.57 | drmessano | http://farm2.static.flickr.com/1174/1477939087_a56d8bdbfa.jpg?v=0 <--- PWNS HARD |
| 03:04.03 | Qwell | jaytee: CentOS |
| 03:04.13 | Qwell | drmessano: oh, it's got a hawt splash screen too |
| 03:04.15 | jaytee | no way! when did that change? |
| 03:04.17 | Qwell | you should download the new one just for that |
| 03:04.50 | Qwell | jaytee: technically it was announced last week |
| 03:04.54 | Qwell | "officially" |
| 03:05.02 | LiNeTuX | yeah, where is it? lol! |
| 03:05.03 | Qwell | but we told TMC like...over a month ago |
| 03:05.26 | drmessano | ZOMG |
| 03:05.27 | jaytee | I'd like to test it |
| 03:05.33 | voxter | Qwell: asterisknow++ |
| 03:05.36 | voxter | Qwell: link me! |
| 03:05.43 | drmessano | TORRENT PLZ |
| 03:05.44 | drmessano | ha |
| 03:05.51 | drmessano | !!!!ONES!!!11!!!! |
| 03:05.55 | voxter | haha |
| 03:07.15 | drmessano | File size is smaller |
| 03:07.27 | drmessano | ++ |
| 03:07.40 | Qwell | drmessano: it actually should be 1 ISO... |
| 03:07.51 | drmessano | Oh, people still burn CDs? |
| 03:08.01 | Qwell | I screwed up when I created these ones.. the size was a little larger on my test build of it, then I cleaned it up and rebuilt |
| 03:08.01 | drmessano | downloads the DVD |
| 03:08.07 | drmessano | Ah |
| 03:08.38 | drmessano | so it was 700 + extra, and you rebuilt the 700 |
| 03:08.49 | drmessano | But extra needs to be moved over |
| 03:14.29 | Carlos_PHX | So you guys at Digium know who (if anyone) is working on the T.38 stuff? |
| 03:15.12 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 03:15.13 | Qwell | drmessano: no, there were duplicate packages on disc2 that I cleaned out |
| 03:15.23 | Qwell | those were an extra 100mb or so, pushing it over the 650mb |
| 03:15.24 | drmessano | ah |
| 03:15.33 | drmessano | So only need CD1? |
| 03:15.39 | Qwell | get the dvd |
| 03:15.49 | drmessano | But... but |
| 03:15.52 | Qwell | you need both CDs if you do it that way |
| 03:16.01 | Qwell | CD2 has @development-tools |
| 03:16.04 | drmessano | Im not using the CDs anyway |
| 03:16.12 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:16.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:16.16 | drmessano | Youre just making me confused and stuff |
| 03:16.17 | Qwell | right russellb ? |
| 03:16.27 | Qwell | drmessano: yeah, just get the damn dvd :p |
| 03:16.42 | drmessano | Where is CD3 with Skype on it? |
| 03:16.51 | Qwell | drmessano: refresh |
| 03:17.08 | LiNeTuX | heh |
| 03:17.14 | drmessano | Ah |
| 03:17.15 | LiNeTuX | that's another 1G file |
| 03:17.15 | Carlos_PHX | russellb: Since you happen to be here...do you know who, if anyone, is working on T.38? |
| 03:17.56 | russellb | Carlos_PHX: not currently, no |
| 03:18.00 | russellb | Qwell: huh? |
| 03:18.04 | Qwell | russellb: exactly |
| 03:18.11 | voxter | soooo why does festival make my asterisk eat 100% cpu and not hang up the call |
| 03:18.18 | voxter | Is there a free license/trial of cepstral out there? |
| 03:18.31 | russellb | oic |
| 03:18.41 | Qwell | voxter: I implemented festival stuff in dialplan once |
| 03:18.44 | Qwell | it was hot |
| 03:18.59 | voxter | Ive made it work on another box, but this one is being bitchy. |
| 03:19.08 | Carlos_PHX | Bah, I guess I should give up on T.38, but its *so* close to working. |
| 03:19.13 | Qwell | most of it was System() and Playback().. had md5 caching of the text and everything |
| 03:19.21 | Carlos_PHX | Hey, I can reliably crash Asterisk with a T.38 call, that's something. |
| 03:19.35 | voxter | Qwell: oh you mean THAT way. haha. was this before the festival module, or were you just trying to torture yourself? :) |
| 03:19.35 | Qwell | Carlos_PHX: is there a bug open for it? |
| 03:19.45 | Qwell | voxter: it wasn't too hard, really |
| 03:19.52 | Qwell | worked really well too |
| 03:19.58 | Carlos_PHX | Yes, and the fix is to get a version of Spandsp that supports a specific call. |
| 03:20.06 | Carlos_PHX | But...that version doesn't seem to exist. |
| 03:20.15 | voxter | Qwell: im sure it was easy! probably easier than using Festival() in my case! |
| 03:21.00 | Carlos_PHX | But Kevin Fleming says he's seen it working, so someone must have it. |
| 03:21.52 | coppice | Carlos_PHX: use spandsp-0.0.5pre4 |
| 03:22.18 | Carlos_PHX | coppice: That's what I'm using. |
| 03:22.31 | Carlos_PHX | Reinstalled twice, let another Asterisk guy have a go, no dice. |
| 03:22.58 | Qwell | Carlos_PHX: what version does it say to use? coppice is the man who'd know |
| 03:23.11 | coppice | then the problem is with the * code, or you are doing something else wrong. That version is in heavy use |
| 03:23.20 | *** join/#asterisk SanityIO (n=SanityIO@77.242.103.109) |
| 03:23.50 | Carlos_PHX | Interesting to hear. I've found no docs, so I'm winging it, could be something I've done wrong. |
| 03:24.09 | Carlos_PHX | The symptoms match those in bug 13473. |
| 03:24.24 | Carlos_PHX | Someone else is reporting the same thing also with 0.0.5pre4. |
| 03:24.53 | Carlos_PHX | coppice: Are you saying that it's in use with T.38 and *? |
| 03:25.03 | coppice | I have no idea |
| 03:25.05 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 03:26.29 | coppice | if the system can't find t38_terminal_init, you are not using 0.0.5pre4 |
| 03:26.53 | Carlos_PHX | I can't be sure that's the case, it just crashes *. |
| 03:27.21 | coppice | no. * crashes. |
| 03:27.31 | Carlos_PHX | Well, yeah, sorry. |
| 03:27.38 | Carlos_PHX | I can be sure I have this though: spandsp-0.0.5pre4.tgz |
| 03:27.48 | Carlos_PHX | Clean system, no other version possible. |
| 03:28.11 | *** join/#asterisk MrNaz (n=naz@ppp59-167-150-186.lns4.mel6.internode.on.net) |
| 03:28.11 | coppice | maybe, but do you have other older versions installed? from the distro itself, maybe? |
| 03:28.38 | *** join/#asterisk Steve_J-obs (n=Chris123@ip70-173-64-212.lv.lv.cox.net) |
| 03:28.41 | Carlos_PHX | Are you saying Asterisk has a version of Spandsp? I will have to look then. |
| 03:29.21 | coppice | many distros install a spandsp RPM or whatever your local package manager might be |
| 03:29.26 | Steve_J-obs | hi, anybody familiar with the Mysql command on the dialplan? |
| 03:30.07 | Carlos_PHX | Oh, it's apt, I will double check. Fairly sure not, since menuselect was not showing app_fax as an option until I installed spandsp manually. |
| 03:31.04 | Carlos_PHX | I did use apt to install libspandsp, which was not included with spandsp. |
| 03:32.26 | coppice | that last line made no sense |
| 03:33.11 | coppice | the core thing which spandsp does is install libspandsp |
| 03:33.57 | Carlos_PHX | Ok, then I had one problem first,and another now. I installed spandsp from source, and Asterisk said it could not find libspandsp. |
| 03:34.18 | Carlos_PHX | So then I installed libspandsp via apt, and Asterisk could find it, but crashes. |
| 03:34.50 | coppice | then you must get rid of the other libspandsp you installed, and tell your system how to find the correct one. |
| 03:35.24 | Carlos_PHX | I was unable to locate libspandsp in the expected locations. |
| 03:35.32 | Qwell | coppice: does it install to somewhere like /usr/local/ by default? |
| 03:35.32 | coppice | This is fully covered in the spandsp FAQ http://www.soft-switch.org/spandsp_faq/ar01s12.htm |
| 03:35.36 | voxter | lame. app_swift is telling me failed to set voice now. TTS is just not meant to be on this box... |
| 03:35.45 | Carlos_PHX | Ok, I'll take a look through there. |
| 03:35.49 | Qwell | coppice: 404? |
| 03:36.09 | Carlos_PHX | coppice: Thank you very much for your time and attention. |
| 03:36.25 | coppice | ./configure type packages normall install in /usr/local by default, and spandsp follows the rules :-) |
| 03:37.02 | Carlos_PHX | Yeah, but this fails to find it, so... find /usr/lib/ -name libspandsp.so\* |
| 03:37.43 | Carlos_PHX | FYI, the URL above is 404, but I can open the main FAQ page. |
| 03:38.49 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:38.59 | coppice | whoops. a letter got chopped off - http://www.soft-switch.org/spandsp_faq/ar01s12.html |
| 03:41.54 | Carlos_PHX | Heh, yeah, point taken on that. Now I just have to figure out how to tell * where the right version is. |
| 03:41.55 | *** join/#asterisk fredonIRC (n=fredonIR@c-98-243-174-131.hsd1.mi.comcast.net) |
| 03:41.56 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 03:42.39 | ReDNeQ | sup all |
| 03:45.19 | coppice | If you want spandsp to install in /usr/lib use "./configure --prefix=/usr/lib" |
| 03:46.49 | Carlos_PHX | coppice: Thanks. I am going to assume I should uninstall/reinstall spandsp followed by *. |
| 03:47.52 | coppice | The dumb thing with most distros is they install by default to scan /usr/local/include for headers, but not /usr/local/lib for the associated libraries. this causes endless support hassles |
| 03:48.19 | phix | coppice: yeah |
| 03:48.35 | phix | coppice: I say just install under /usr not /usr/local |
| 03:48.40 | *** join/#asterisk roatgider (i=psyoptix@vmh3.adv.duffnoc.net) |
| 03:48.55 | phix | coppice: apt-get source asterisk if you want to compile it your self |
| 03:49.12 | *** join/#asterisk fredonIRC (n=fredonIR@c-98-243-174-131.hsd1.mi.comcast.net) |
| 03:49.31 | Carlos_PHX | Great info. At the end of this I'll put the documentation on voip-info so there's at least a starting point for others. |
| 03:49.34 | phix | hey fredonIRC |
| 03:50.46 | voxter | is there seriously no function to streplace in asterisk? |
| 03:51.00 | *** part/#asterisk fredonIRC (n=fredonIR@c-98-243-174-131.hsd1.mi.comcast.net) |
| 03:51.06 | Carlos_PHX | coppice: Are you Steve Underwood? |
| 03:51.25 | coppice | only if you have money to pay me |
| 03:51.29 | Carlos_PHX | Heh |
| 03:51.48 | Carlos_PHX | Thank you for your contributions, Spandsp has helped me tremendously over the years. |
| 03:52.08 | roatgider | Hey all, I just install *NOW in a Virtual Server to test it out, everything is working great with the exception of the default IVR |
| 03:52.15 | roatgider | when I dial 7000 from any sip phone, and watch on the cli |
| 03:52.20 | roatgider | the call is answered, the 1 second wait period passes, and then the call is dropped |
| 03:52.27 | roatgider | it's weird, because it was working fine the first few times |
| 03:52.31 | drmessano | ----> #asterisknow |
| 03:52.42 | roatgider | no one is responding in there :\ |
| 03:52.51 | drmessano | You're cross spamming |
| 03:53.11 | roatgider | I'm restating my problem in a different channel that has more active users |
| 03:53.18 | roatgider | didnt mean to spam |
| 03:53.36 | drmessano | I took offense to it |
| 03:53.42 | roatgider | apologies |
| 03:54.21 | drmessano | Really, really, really hurt here |
| 03:54.27 | drmessano | I feel so... used |
| 03:54.32 | roatgider | <.< |
| 03:54.36 | drmessano | Ok, more than normal |
| 03:54.38 | Carlos_PHX | And you like it |
| 03:54.44 | jaytee | hehehe |
| 03:55.38 | *** join/#asterisk TeamFrrst (n=TeamFrrs@c-98-243-174-131.hsd1.mi.comcast.net) |
| 03:55.48 | [TK]D-Fender | drmessano: how long... do the memories last?! |
| 03:56.18 | drmessano | I still hear the cries of the clowns.. it's so.... tragic.. |
| 03:59.17 | voxter | Is there no dialplan function to cut all occurrences of, say, @ from a variable? |
| 03:59.36 | jaytee | um CUT? |
| 04:00.30 | *** part/#asterisk LemensTS (n=matthew@adsl-70-238-160-24.dsl.stlsmo.sbcglobal.net) |
| 04:03.17 | voxter | jaytee: that doesnt exactly remove them so much as give you specific fields around every @ |
| 04:04.49 | [TK]D-Fender | voxter: aND YOU GLUE THOSE BACK TOGETHER AND THERE YOU HAVE IT |
| 04:04.57 | [TK]D-Fender | (along with my capslock) |
| 04:05.10 | voxter | So then the answer is no :) |
| 04:05.50 | voxter | Hmm, there also doesnt seem to be a way without using LEN() first to chop just the LAST character/integer off the end of a variable using :#:# |
| 04:06.01 | voxter | i guess ill ahve to make this code a bit uglier than i'd hoped |
| 04:16.02 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 04:27.42 | [TK]D-Fender | voxter: Well you could do it in 1 line with grep/sed and app_backticks (3rd party app) |
| 04:29.03 | Qwell | func_shell |
| 04:29.12 | voxter | quivers |
| 04:29.24 | voxter | I did it nested inside of 3 CUT's |
| 04:29.26 | voxter | ugly, but it'll do |
| 04:29.49 | voxter | someone asked me for a way to have it read back their SIP URI, so i set up sip:getsipuri@voxter.ca to read it back to them |
| 04:29.57 | voxter | of course app_swift is bitchy about the characters it will read. |
| 04:36.45 | [TK]D-Fender | voxter: I'm sure it'd have been an easy job to invent a function or app to do what you needed. |
| 04:46.32 | jaytee | time to snooze |
| 04:46.34 | jaytee | nite all |
| 04:46.42 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:51.19 | voxter | [TK]D-Fender: yeah. It would really be nice to see a STR_REPLACE function that understood regex. it's come up more than once. |
| 04:56.14 | voxter | found a chanisavail bug |
| 04:56.15 | voxter | time to report |
| 04:56.43 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 04:59.09 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 05:02.48 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 05:12.45 | [TK]D-Fender | checkout time. Later all |
| 05:18.44 | *** join/#asterisk Cutlass (n=chatzill@c-67-176-208-15.hsd1.il.comcast.net) |
| 05:20.21 | Cutlass | I just installed asterisk 1.6 and I can't find the config files...has anyone seen this before? I used "/usr/local" as the installation PREFIX... |
| 05:20.43 | Cutlass | ...there is nothing in the whole filesystem.... |
| 05:25.55 | mosty | they're in the configs directory in the asterisk source aren't they? |
| 05:27.00 | Cutlass | ah!!...so they're in the location that I unpack the tar file??? I see them now...doesn't it install these somewhere else? |
| 05:33.15 | mosty | there is a make target that will install the sample configs |
| 05:33.43 | jameswf-home | make broken |
| 05:34.11 | jameswf-home | make configs_incompatible |
| 05:34.14 | jameswf-home | heh |
| 05:34.30 | mosty | make install-useless-default-configs |
| 05:35.05 | jameswf-home | that is just make configs.. it was much to long before |
| 05:36.19 | Cutlass | thanks!...that help :) |
| 05:36.25 | Cutlass | *helps |
| 05:45.38 | *** part/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 05:51.05 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:03.05 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 06:03.20 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:16.51 | *** join/#asterisk [netman] (n=netman@199.Red-83-38-222.dynamicIP.rima-tde.net) |
| 06:20.56 | jameswf-home | heh http://linuxslut.net/album/linuxsluts/ |
| 06:29.05 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 06:42.15 | *** join/#asterisk Corydon76-dig (i=gray@pdpc/supporter/bronze/Corydon76-home) |
| 06:42.15 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 06:43.51 | *** join/#asterisk shine (n=stroll@lamantin.achamo.net) |
| 06:47.51 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 06:54.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d2e4a34824b49ce4) |
| 06:59.39 | andrewgodwin | this one has been puzzling me for a while - if my laptop's phone port appears as a second soundcard (which it does |
| 06:59.49 | andrewgodwin | is it possible to get asterisk to use it? |
| 07:10.30 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 07:14.50 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 07:17.51 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:28.24 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:32.37 | *** join/#asterisk hi365_m (n=hi365@213.151.44.101) |
| 07:39.21 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:55.42 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
| 08:09.16 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:14.54 | *** join/#asterisk Steve_Ollis (n=Steve_Ol@cms.hillsong.com) |
| 08:16.58 | *** join/#asterisk nicox (n=nicox@212-183-32-224.adsl.highway.telekom.at) |
| 08:27.43 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 08:32.56 | nicox | hi, anyone there? |
| 08:34.22 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:35.41 | nicox | hi |
| 08:36.48 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:37.05 | Steve_Ollis | /me is a n00b .. just watching and learning |
| 08:40.17 | *** join/#asterisk oilinki (n=oil@ppp-124-120-1-200.revip2.asianet.co.th) |
| 08:45.37 | *** join/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 08:45.52 | anebi | hi |
| 08:46.35 | anebi | i have a problem with one of the trunks and i can't understand how to solve this. i have a sip account from telsome and i have set it to asterisk on our server. |
| 08:47.00 | anebi | it works well some times, but during the day we get a message wrong password and then it disconect and stop to work |
| 08:47.25 | anebi | also we get timed out messages for this trunk, what can it be the reason and how can i solve this problem? |
| 08:47.49 | anebi | we have installed asterisk with freepbx |
| 08:49.08 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 08:49.19 | Kernel_Core | hi all |
| 08:51.39 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:51.40 | anebi | the password is correct, i can't understand why this happen |
| 08:52.04 | anebi | after asterisk restart it works again, register to sip provider |
| 08:52.35 | *** join/#asterisk op3r (n=op3r@121.96.100.177) |
| 08:54.25 | defswork | anebi: sounds like something to ask tour provider first I would guess |
| 08:56.17 | *** join/#asterisk niros (n=nir@89-139-53-119.bb.netvision.net.il) |
| 08:56.21 | anebi | defswork: as my boss told me, the provider told him that our server register every 60 secs |
| 08:56.29 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:57.47 | defswork | anebi: right - but someone is telling you your credentials are bad |
| 08:58.05 | defswork | anebi: that's pretty explicit - not a timeout etc.. |
| 08:58.25 | Kernel_Core | hi all |
| 08:58.46 | nicox | <PROTECTED> |
| 08:58.52 | Kernel_Core | after installing OSLEC with asterisk , it isnot possible to increase txgain or rxgain in zapta.conf |
| 08:59.14 | Kernel_Core | I want to increase my txgain |
| 08:59.18 | Kernel_Core | how do I do it ?! |
| 08:59.37 | anebi | defswork: yes, you are right. |
| 08:59.56 | defswork | anebi: are they tetchy about registering so often ? |
| 09:00.42 | anebi | defswork: not at all |
| 09:01.02 | nicox | any idea why a call Transfer is not working with IAX? |
| 09:02.11 | defswork | did I read somewhere that aastra are making blf's tri-state so that a ringing call can be collected ? |
| 09:05.21 | defswork | aah - directed call pickup is already there |
| 09:06.54 | *** part/#asterisk Steve_Ollis (n=Steve_Ol@cms.hillsong.com) |
| 09:10.27 | *** join/#asterisk dwagner (n=dwagner@195.202.166.182) |
| 09:10.36 | dwagner | hello, i would like to talk to murf |
| 09:25.36 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 09:25.47 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 09:25.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:26.21 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:27.14 | Andre101 | hello.. Is it possible to get a dialplan to match on characters? Trying to match dialing a skype username |
| 09:27.51 | kaldemar | yes |
| 09:28.47 | Andre101 | kaldemar: How do I go about doing it? |
| 09:31.20 | kaldemar | exactly the same as with numbers. |
| 09:31.33 | Pagautas | exten => _hello,1,dial(SIP/hello@skype) |
| 09:31.43 | kaldemar | _ isn't needed in that. |
| 09:33.32 | Pagautas | i've got a question about queues |
| 09:33.44 | Pagautas | i had a dialplan like this |
| 09:33.45 | Pagautas | http://pastebin.ca/1215633 |
| 09:33.54 | Pagautas | everything worked fine |
| 09:34.12 | Pagautas | until my boss asked to put an announcement |
| 09:34.15 | *** part/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 09:34.26 | Pagautas | to calling side before |
| 09:34.34 | Pagautas | agent phone begins to ring |
| 09:34.41 | Pagautas | so i've done like this |
| 09:34.46 | Pagautas | http://pastebin.ca/1215632 |
| 09:35.02 | Pagautas | but then asterisk stopped to record calls |
| 09:35.06 | Pagautas | why? |
| 09:38.21 | nicox | hi, anyone an idea why Call transfer on IAX is not working after upgrading to asterisk 1.6.0-rc6? |
| 09:43.20 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 09:51.30 | *** join/#asterisk steliosk (n=Stelios@91.140.121.44) |
| 10:14.06 | *** join/#asterisk Ng_ (n=cmsj@mairukipa.tenshu.net) |
| 10:14.31 | Ng_ | if one has a PRI (E1) hooked up via a zaptel device, is there a way to see which channels are actually being used? |
| 10:14.46 | Ng_ | the nearest i can find is "sip show channels", but that just shows active SIP calls, which may not be the same thing |
| 10:15.36 | nicox | zap show channels |
| 10:15.47 | nicox | or simple show channels verbose |
| 10:15.58 | nicox | or core show channels verbose |
| 10:16.21 | nicox | there you see everything, which channel is connected with which channel and how long.... |
| 10:16.36 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 10:16.38 | Ng_ | nicox: oh interesting, I must have only run "zap show channels" when nobody was on the phone, I can actually see which extension is using a channel \o/ |
| 10:16.48 | Ng_ | I don't appear to have "simple" or "core" though |
| 10:17.14 | Ng_ | (I get abused for running an old asterisk each time I pop in here with a question ;) |
| 10:17.49 | Ng_ | nicox: thanks :) |
| 10:18.05 | nicox | i'm using 1.2.17 to 1.6.0-rc6 so i have to use everything.. |
| 10:18.08 | nicox | your welcome |
| 10:19.38 | Ng_ | 1.2.12 here. I will upgrade it happily when a need arises, but it's been in operation for 18 months with no problems (i had to reset the sangoma driver twice in that time, but that's not asterisk's fault) |
| 10:20.41 | nicox | on my site sangoma driver runs better then asterisk *g* also ss7box running great |
| 10:21.50 | Ng_ | on the whole it's been fine, but every so often the provider drops the line over a weekend, and sometimes the sangoma driver doesn't bring it back properly. calls work, but they sound really distorted. A quick poke of the driver restores normality so I've not really traced it any deeper |
| 10:22.52 | nicox | hm, try a firmware-upgrade and upgrade of the wanpipe-driver. this should help |
| 10:23.48 | Ng_ | yeah, I've had that queued up in my todo list for some months now ;) |
| 10:35.36 | *** join/#asterisk _Roman (n=roman@87.254.78.150) |
| 10:43.51 | _Roman | Hello, I am using asterisk 1.4 (under trixbox), with an X100P based pstn card. The problem I am having is that when a call comes in, I send it to a queue, I answer one of the internal SIP phones (a grandstream 200) and then hang the internal phone up. The problem is that the external line does not hangup. Does anyone have any suggestions? |
| 10:44.47 | *** part/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 10:52.01 | *** join/#asterisk Squeeb (i=squeeb@eggwee.co.uk) |
| 10:52.41 | Squeeb | I keep getting an error, and also my Queue has stopped reporting the position status to customers.. This is the error: |
| 10:52.44 | Squeeb | [Oct 1 11:46:07] WARNING[66413]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 1 |
| 10:52.47 | Squeeb | Any ideas where it's coming from? |
| 10:56.25 | *** join/#asterisk padski (n=paddy@5ac311a1.bb.sky.com) |
| 10:56.29 | Squeeb | Nobody here? |
| 10:57.11 | kaldemar | when did you start getting the error? when do you get the error? |
| 10:57.43 | Squeeb | When someone is in the queue, just before it switches extension, it's in rrmemory stratigy.. I'm not sure when I started getting this error |
| 10:59.16 | Squeeb | hmmm .. actually it appears to be whenever the queue tries to call my 6002 extension |
| 11:01.48 | kaldemar | show the extension |
| 11:02.30 | Squeeb | http://pastebin.ca/1215690 |
| 11:03.08 | Squeeb | hmm |
| 11:03.15 | Squeeb | it happens if I try and call that extension directly |
| 11:06.27 | kaldemar | that is not an extension, that is a users.conf entry. an extension is in your dialplan, extensions.conf. |
| 11:06.28 | Squeeb | I wish it told me the line number the fault occures |
| 11:08.59 | Squeeb | http://pastebin.ca/1215694 |
| 11:09.41 | Squeeb | not sure what o,1, does |
| 11:11.46 | kaldemar | looks like you're using some gui. |
| 11:12.10 | Squeeb | yea, the digium web panel |
| 11:13.04 | kaldemar | exten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3) might be a problem if ARG1 has spaces in it for some reason. |
| 11:13.35 | Squeeb | ah |
| 11:13.56 | kaldemar | variables should be quoted in comparisons like that. |
| 11:14.01 | Squeeb | what does ARG1 contain? |
| 11:15.04 | kaldemar | no way of knowing without seeing the whole dialplan. you better ask digium about that. that is not direct asterisk stuff. |
| 11:15.23 | kaldemar | but rather built on asterisk. |
| 11:18.28 | _Roman | Hello, I am using asterisk 1.4 (under trixbox), with an X100P based pstn card. The problem I am having is that when a call comes in, I send it to a queue, I answer one of the internal SIP phones (a grandstream 200) and then hang the internal phone up. The problem is that the external line does not hangup. Does anyone have any suggestions? |
| 11:21.33 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582612.dsl.bell.ca) |
| 11:22.10 | nicox | hi, anyone expierence with Call transffer on IAX calls, and why they are not working? |
| 11:22.51 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 11:25.52 | gr0mit | _Roman, you mentioned trixbox, which is not really supported here |
| 11:26.12 | gr0mit | however, clearing of analogue channels is somewhat complex |
| 11:26.45 | gr0mit | can you explain your config in a bit more details pls |
| 11:29.44 | *** join/#asterisk InsolentDreams (n=Insolent@ip68-5-17-7.oc.oc.cox.net) |
| 11:31.00 | InsolentDreams | Hey is there a trick to unregister a SIP device and re-register it? I'm having a issue where someone's phone rings, but is unable to pick it up. Nothing happens when they pick it up, I think it's a registration/cache issue. |
| 11:32.00 | kaldemar | why would you think it's a registration issue? |
| 11:33.31 | kaldemar | registering just tells asterisk where the phone is. if the phone rings, the phone must either be registered or defined with a static ip. either way, location or registering shouldn't be the problem. |
| 11:38.54 | *** join/#asterisk mesfet (n=iw3grx@89-96-227-146.ip14.fastwebnet.it) |
| 11:40.14 | mesfet | Hi. Just a question: what about a Playback(message|m) to play a message with music on hold background? |
| 11:41.15 | kaldemar | rephrase your question, please. |
| 11:41.52 | Squeeb | What's the astdb ? |
| 11:42.11 | mesfet | I believe that in interesting feature (not available, now) is the possibility to play a message (a sound file) with music in the background. |
| 11:42.35 | mesfet | This can be done mixing the sound file with the music-on-hold. |
| 11:42.38 | kaldemar | Squeeb: asterisk's internal database |
| 11:42.44 | Squeeb | aah cool |
| 11:43.29 | mesfet | Playing "welcome" message and "ivr-instructions" with music in the background should be nice for the caller. |
| 11:43.53 | mesfet | I don't know if this feature was evaluated or not. |
| 11:44.01 | *** join/#asterisk LiNeTuX (n=LiNeTuX@64.132.248.206) |
| 11:44.06 | InsolentDreams | kaldemar: Thanks mate, The phone appeared to be in a messed up state, cheapo polycom's seemed to return a bunch of gibberish when I was sip debugging it |
| 11:44.54 | kaldemar | np |
| 11:44.59 | InsolentDreams | I reloaded from firmware defaults and re-set the settings and it's fine ;) I asked a bit prematurely before digging in properly. :) |
| 11:45.48 | *** join/#asterisk nicox (n=nicox@212-183-32-224.adsl.highway.telekom.at) |
| 11:46.45 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 11:50.20 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 11:51.00 | *** join/#asterisk martijn_ (n=martijn@office-nl.dbcorp.nl) |
| 11:51.13 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 12:01.24 | *** join/#asterisk traxxos (n=trax@clubnix100.esiee.fr) |
| 12:01.30 | traxxos | hello all |
| 12:07.24 | *** join/#asterisk steliosk (n=Stelios@rouss--1.static.otenet.gr) |
| 12:07.46 | *** join/#asterisk kielstrand (n=yes@p5B3EB3A3.dip0.t-ipconnect.de) |
| 12:08.18 | kielstrand | hi there, anyone here can help me with sccp and a cisco 7960 |
| 12:09.25 | kielstrand | message after booting: File Auth Fail: CTLFile.tlv |
| 12:16.32 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 12:16.56 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 12:20.58 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 12:22.48 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:22.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:24.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:25.23 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:30.28 | mort_gib | HI |
| 12:30.36 | Squeeb | HELLO |
| 12:31.35 | mort_gib | I'm having problems getting SIPPEER(100:curcalls) to work in a macro -Are there any problems with SIPPEER ?? |
| 12:33.48 | kaldemar | replace : with , or | |
| 12:35.13 | *** join/#asterisk Corydon76-dig (i=yellow@pdpc/supporter/bronze/Corydon76-home) |
| 12:35.28 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 12:35.33 | *** join/#asterisk mav3rick (n=t0m@trankil.biz) |
| 12:39.55 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:43.12 | *** join/#asterisk steliosk (n=Stelios@91.140.121.44) |
| 12:45.28 | *** join/#asterisk stony (n=oloch@serverbu.de) |
| 12:45.31 | stony | hi |
| 12:46.07 | stony | i have a strange problem - out of nowhere the asterisk binary hang and now it produces 100% cpu load on start and isn't booting completly |
| 12:46.16 | stony | any idea where this could come from ? |
| 12:47.40 | *** join/#asterisk write_erase (n=Olivier@goodgw.m2m-fleet.com) |
| 12:47.43 | stony | i haven't changed anything |
| 12:48.57 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 12:51.09 | *** join/#asterisk v4mp (n=Gary@82.118.111.250) |
| 12:51.28 | mort_gib | Kaldemar: Yeah? |
| 12:52.58 | padski | I'm fighting with rfc3389 comfort noise packets again today. I'm being told that they are coming from BT and that BT can't be persuaded to switch them off, but our upstream provider is acting as some kind of proxy between us and it is unclear whether that may impact this problem. anyone got experience with asterisk talking to BT IPXes in the UK ? |
| 12:54.19 | Squeeb | I get the same thing with Sipgate |
| 12:54.26 | Squeeb | Oct 1 12:58:48] NOTICE[66413]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.10.77.24 |
| 12:54.29 | Squeeb | et... |
| 12:55.21 | eric2 | Squeeb - I get that same problem :( |
| 12:55.24 | kaldemar | mort_gib: what yeah? |
| 12:55.53 | mort_gib | kaldemar: Was that for me (replace : with , or |) |
| 12:55.58 | kaldemar | yes, it was. |
| 12:56.54 | mort_gib | Okay thanks, but in voip-info.org it states SIPPEER(<peername>[:item]) |
| 12:56.56 | padski | Squeeb, have you spoken to them about it ? |
| 12:57.58 | Squeeb | no |
| 12:58.27 | kaldemar | mort_gib: well, voip-info has been known to have errors. core show function SIPPEER uses pipes. |
| 12:58.34 | [TK]D-Fender | mort_gib: SIPPEER(<peername>[|item]) <- |
| 12:58.43 | padski | Squeeb, one of the fun symptoms I am seeing is music-no-hold that only plays when you talk to it. have you got that too ? |
| 12:58.49 | [TK]D-Fender | mort_gib: "core show function SIPPEER" |
| 12:58.53 | kaldemar | mort_gib: try it and fix it to the wiki if it's wrong there. |
| 12:59.01 | [TK]D-Fender | mort_gib: You shouldn't be using the WIKI for instructions. |
| 12:59.24 | v4mp | :o |
| 13:00.07 | kaldemar | i'd use wiki for instructions, but test every single thing myself before believing it works. |
| 13:00.29 | mort_gib | As did I, but alas it didn't work, which is why I'm here :-) |
| 13:00.45 | v4mp | hey guys i have a problem where as when u call in u get the welcome then the options read to you then when you try pick an option it doesn't wrong right 1 way i had the line it spammed loads of line in the cli i will paste my config and cli output |
| 13:01.15 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:01.28 | kaldemar | mort_gib: use cli helps, docs in the source package and the book as primary docs. |
| 13:01.42 | mort_gib | Ok |
| 13:01.43 | km- | padski: This may not be even remotely relevant to that music-on-hold thing, but I have heard that asterisk will not start sending rtp until it receives rtp first. |
| 13:01.48 | mort_gib | Thanks |
| 13:02.13 | km- | padski: if the lines have VAD, maybe the handset isn't sending rtp until there's energy on the handset? |
| 13:02.43 | mort_gib | Is incominglimit of ANY use in 1.4 -> ?? |
| 13:02.45 | v4mp | this is the config http://pastebin.com/d5bfe77b7 not too sure about after the welcome part what the line should look like to read what to do... 1 line i did have kind of worked but when u tryed to go to option 1 it said it didn't exist |
| 13:02.50 | mort_gib | in sip.conf |
| 13:02.59 | [TK]D-Fender | CLI is the #1 way to get instruction on app syntax. |
| 13:03.36 | stony | even if i start asterisk without any hardware (no modules in the kernel loaded) it still hangs |
| 13:03.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:03.49 | mort_gib | I'm beginning to realize that. I have always used .Wiki, Google or similar |
| 13:04.09 | mort_gib | Examples works better for me.... |
| 13:04.31 | padski | km-, there is already rtp in progress. the silence at the asterisk end corresponds to rfc3389 CN packets incoming in the tcpdump. the first CN packet arrives, if anythin, earlier than I would expect the next rtp packet. |
| 13:04.47 | v4mp | [TK]D-Fender, yes if you know what to look for :p |
| 13:05.01 | Squeeb | padski: I've not tried that |
| 13:05.03 | Squeeb | I will check |
| 13:05.10 | mort_gib | v4mp: Yes! |
| 13:05.14 | Squeeb | I don't think it's a problem though |
| 13:05.21 | Squeeb | It's only a notice .. not a warning |
| 13:05.24 | *** join/#asterisk TeamFrrst (n=TeamFrrs@c-98-243-174-131.hsd1.mi.comcast.net) |
| 13:05.41 | [TK]D-Fender | v4mp: "core show functions" , "core show function [functionname]" , "core show applications" , "core show application [applicationname]" |
| 13:05.51 | [TK]D-Fender | v4mp: There is EVERYTHING. |
| 13:06.43 | padski | km-, but I suppose that if the CN packets don't count as rtp, and every packet had to be a reply then this would account for it, but I can't help thinking that would be a nonsense, since then we would drop packets in response to dropped incoming packets ?? The explanation I have seen is that is is timing related. |
| 13:06.51 | *** join/#asterisk TeamFrrst (n=TeamFrrs@c-98-243-174-131.hsd1.mi.comcast.net) |
| 13:07.08 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 13:07.55 | km- | padski: ah. I don't really have much knowledge of cng to comment on it, sorry. |
| 13:07.59 | phpboy | hey all, I have 2 PRI cards, one for land lines (incoming and outgoing) and one for mobile calls (outbound only) what would be the best way to configure this in zapata.conf? |
| 13:08.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.00 | [TK]D-Fender | phpboy: Just about the only thing those might have in common is just sharing a group # for applicable channels. |
| 13:09.49 | padski | km-, it doesn't seem to me that the actual generation of comfort noise is all that important. silence would do fine. I'm seeing issues that go way beyond CNG. but thanks for your thought, maybe it will turn out to be part of the puzzle :-) |
| 13:09.52 | phpboy | I thought so, that's all I'll have to worry about? |
| 13:10.00 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 13:11.16 | phpboy | [Oct 1 15:10:48] WARNING[23735]: channel.c:3025 ast_request: No channel type registered for 'ZAP' |
| 13:11.16 | phpboy | [Oct 1 15:10:48] WARNING[23735]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 13:11.26 | Hertzy3 | Does anyone know where I can find instructions on how to install * on a Xen DomU running CentOS? I am having problems installing zaptel |
| 13:11.32 | phpboy | it's giving me those errors now on the outbound mobile pri :( |
| 13:11.52 | *** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 13:11.57 | phpboy | Hertzy3: what problems are you experiencing? |
| 13:13.46 | Hertzy3 | upon running 'make', I get this error: You do not appear to have the sources for the 2.6.18-92.el5xen kernel installed. |
| 13:13.52 | phpboy | [TK]D-Fender: which paste bin can I use to show you my config? |
| 13:14.07 | Hertzy3 | i tried doing yum install xen kernel-xen kernel-xen-devel |
| 13:14.11 | phpboy | Hertzy3: easy to fix |
| 13:14.13 | Hertzy3 | but its a host |
| 13:14.18 | phpboy | did it install? |
| 13:14.19 | Hertzy3 | err, vm |
| 13:14.30 | Hertzy3 | they all installed just fine |
| 13:14.33 | phpboy | ok |
| 13:14.35 | phpboy | then run |
| 13:14.40 | phpboy | yum update |
| 13:14.42 | phpboy | then reboot |
| 13:14.46 | phpboy | and it should work |
| 13:14.52 | Hertzy3 | thank you I will give that a try |
| 13:15.02 | phpboy | I'm pretty certain that will sort you out |
| 13:15.08 | phpboy | would you like the reasoning behind this? |
| 13:15.22 | [TK]D-Fender | phpboy: Doesn't matter which |
| 13:15.23 | stony | is there a way to strace the asterisk process ? it always detaches and then the strace isn't working |
| 13:15.35 | stony | i added the -f switch, but that doesn't stop asterisk from detaching |
| 13:15.38 | [TK]D-Fender | phpboy: And clearly chan_zap.so is not even loaded |
| 13:15.49 | Hertzy3 | yes |
| 13:15.52 | [TK]D-Fender | phpboy: Sounds like you compiled zaptel after * |
| 13:16.00 | *** join/#asterisk awannabe (n=brad@ip24-251-147-29.ph.ph.cox.net) |
| 13:16.07 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 13:16.08 | phpboy | [TK]D-Fender: no no, you'll see now what the problem is, hang ten... |
| 13:16.41 | awannabe | hey guys, ever seen where a polycom registerrs can take in/out calls, but if you try to dial *99 to access voicemail (without the button) asterisk says unable to authenticate?! |
| 13:16.57 | phpboy | Hertzy3: Zaptel requires the src of the kernel you are running at the point of compile, which in this case you don't have |
| 13:17.06 | phpboy | so after a new kernel and yum upgrade |
| 13:17.07 | c4t3l | awannabe: check your polycom configs |
| 13:17.09 | phpboy | and a reboot |
| 13:17.12 | phpboy | you should be fine |
| 13:17.28 | Hertzy3 | that does make sense, thank you, i am currently rebooting, hopefully it works |
| 13:17.33 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:18.24 | phpboy | [TK]D-Fender: http://pastebin.com/m7f752733 - if I uncomment the commented lines, zap does not work... if it's commented then zap works |
| 13:18.43 | awannabe | c4t3l, i have over and over, it only does it for *99 its really weird |
| 13:19.06 | *** join/#asterisk andrewgodwin (n=andrew@southill.torchboxapps.com) |
| 13:19.19 | c4t3l | awannabe: what about your dialplan? This works for the mapped button on the phone?? |
| 13:19.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:19.58 | awannabe | on like the 550s the button works fine, but no workie if i just dial *99 |
| 13:20.30 | [TK]D-Fender | phpboy: phpboy>[Oct 1 15:10:48] WARNING[23735]: channel.c:3025 ast_request: No channel type registered for 'ZAP' <--- no chan_zap loaded. En of story |
| 13:21.09 | *** part/#asterisk InsolentDreams (n=Insolent@ip68-5-17-7.oc.oc.cox.net) |
| 13:21.11 | c4t3l | awannabe: and your msg.mwi.1.callBack whatever is mapped tp *99?? |
| 13:21.33 | mort_gib | I need some advice.... I have ONE out of a bunch of clients that complains about dropped calls. I can turn on sip debug, but it only works while I'm logged in. Any way of making it permanent?? |
| 13:21.46 | phpboy | [TK]D-Fender: but it works with the bottom lines commented out |
| 13:22.00 | phpboy | So I think it's safe to assume zap is working |
| 13:22.03 | mort_gib | They have this issue, maybe once a week or so, some weeks 3-4 times |
| 13:22.11 | phpboy | it's just not happy with those last commented lines :( |
| 13:22.21 | awannabe | c4t3l, correct, is that the prob you think somehow? |
| 13:22.36 | phpboy | It works perfectly without the last comment lines, but I need the last lines for outbound mobile :( |
| 13:22.38 | c4t3l | mort_gib: change your logger.conf settings to debug and grep away at the logs :) |
| 13:22.48 | mort_gib | SIp debug too?? |
| 13:22.52 | awannabe | its like the phone is trying to not use the registration when you dial *99, the phones have one registration on them, and it spans to all linekeys |
| 13:24.29 | phpboy | [TK]D-Fender: Any ideas? |
| 13:24.33 | c4t3l | awannabe: sounds like some parameter is not being passed to you dialplan |
| 13:24.38 | mort_gib | How do I get sip debug into the log file?? |
| 13:24.39 | c4t3l | your** |
| 13:25.17 | awannabe | c4t3l, yeah, thats what i thought to, i can dial any other extension number, but *99 its like star codes are screwed up almost |
| 13:26.07 | fred-tmft | awannabe: do any other star codes work directly? |
| 13:26.10 | phpboy | Can anybody help please, http://pastebin.com/m7f752733 <-- zapata.conf .... the commented out lines at the bottom seem to be breaking zap :( |
| 13:26.13 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:26.15 | kaldemar | phpboy: does your zaptel have such channels? |
| 13:26.28 | phpboy | kaldemar: it does |
| 13:26.35 | kaldemar | where have you come up with the channel numbers? |
| 13:26.37 | Hertzy3 | phpboy: Now the VM won't boot back up. Which I will let you know that if something got messed up it is not a big deal, the whole point of this server is for me to learn how Xen works and this is the only VM on there. But heres the error I receive now. http://pastebin.com/d97e237f |
| 13:27.15 | phpboy | kaldemar: /etc/zaptel.conf |
| 13:27.22 | phpboy | it comes out of there |
| 13:27.23 | awannabe | oh wtf |
| 13:27.29 | awannabe | *90 works lol |
| 13:27.46 | awannabe | oh wait, nevermind no it doesnt |
| 13:28.03 | awannabe | if you hit speakerphone and dial it does, if you hit a line key then it it fails, for both * codes |
| 13:28.20 | phpboy | Hertzy3: did yum upgrade run properly? |
| 13:28.38 | [TK]D-Fender | phpboy: You have not proved that it is loaded, or which channels * should recognize. You have not shown FULL CLI output for your failed attempt either |
| 13:28.52 | Hertzy3 | phpboy: I had planned to run that after the reboot |
| 13:29.00 | fred-tmft | awannabe: this will be within your polycom dialstring... make sure you have *XX allowed |
| 13:29.09 | phpboy | [TK]D-Fender: ZAP is working for group=1 |
| 13:29.16 | kaldemar | phpboy: so you have 2 cards with four ports? |
| 13:29.27 | phpboy | kaldemar: correct |
| 13:29.27 | awannabe | uri="sip:*@ |
| 13:29.30 | awannabe | yeah, bastards |
| 13:29.37 | [TK]D-Fender | phpboy: you aren't showing me anything of value. I cannot help you |
| 13:29.39 | phpboy | and group=2 is the first port on the second card |
| 13:29.45 | phpboy | [TK]D-Fender: no problem |
| 13:29.56 | phpboy | thanks anyway |
| 13:30.24 | awannabe | i bet thats it, rebooting now |
| 13:30.38 | kaldemar | phpboy: show your zaptel.conf, /proc/zaptel/1 and /proc/zaptel/5 |
| 13:30.47 | phpboy | kaldemar: anyhoo, so, giving my zapata.conf file, if I comment out group=2 and it's channels... it works without a problem... only for the first cards channels of course... problem is, when I wanna bring the second card into the picture... it breaks |
| 13:30.49 | phpboy | kaldemar: ok |
| 13:31.38 | *** join/#asterisk Kobaz (n=kobaz@64.27.7.21) |
| 13:31.57 | v4mp | on my config the extensions 1 4 and 5 are setup correct ? |
| 13:32.39 | *** join/#asterisk moy (n=moy@nat/ibm/x-6d01573306b2f990) |
| 13:32.52 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-151.washdc.east.verizon.net) |
| 13:33.09 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 13:33.38 | phpboy | kaldemar: http://pastebin.com/m11d92c65 |
| 13:33.46 | asteriskmonkey | has any one had this error before? chan_zap.c:1045 zt_digit_begin: Couldn't dial digit 5 |
| 13:33.53 | phpboy | I see one problem, now that I look at that, I left a channel out on span 5 |
| 13:34.07 | phpboy | Although I don't think that'll damage anything? |
| 13:34.26 | asteriskmonkey | it seems dtmf isnt passing out or in after a call is established but it is showing and stating it cant dial |
| 13:34.41 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 13:35.36 | Katty | morning |
| 13:35.42 | phpboy | Katty :D |
| 13:35.50 | LiNeTuX | Katty: yes, it is! |
| 13:36.03 | Katty | puts on Warning, I am Grumpy sticker |
| 13:36.20 | tzanger | hah |
| 13:36.20 | phpboy | My business day is almost finished |
| 13:36.27 | phpboy | kaldemar: what are your thoughts? |
| 13:36.43 | tzanger | you don't know grumpy until you've met my wife when she's reconciling the books |
| 13:36.57 | v4mp | i get this with WaitExten == Auto fallthrough, channel 'SIP/84455707-08837138' status is 'UNKNOWN' |
| 13:37.42 | kaldemar | phpboy: set your dchan for span5 as 140 and bchan=125-139,141-155 and try again. |
| 13:38.17 | awannabe | fred-tmft, thanks dude, that was it, added *xxT |
| 13:38.43 | fred-tmft | my pleasure. |
| 13:39.22 | phpboy | kaldemar: otherwise everything looks fine? |
| 13:39.42 | write_erase | Hi, someone know French FranceTelecom RNIS connectivity ? I'm not sure how to disconnect the existing PABX and connect my Asterisk server. thx |
| 13:39.54 | kaldemar | phpboy: with a quick look. |
| 13:40.10 | Squeeb | Hmm |
| 13:40.14 | kaldemar | phpboy: then you'd want to fix the channels in your zapata.conf too. |
| 13:40.24 | Squeeb | how can I pass caller ID to an AGI script? |
| 13:40.41 | Squeeb | it doesn't seem to turn up if i just do AGI(test.agi) |
| 13:40.53 | Squeeb | although it appears in Master.csv |
| 13:42.31 | phpboy | kaldemar: you are too kind, it worked. thanks :D |
| 13:43.02 | kaldemar | phpboy: did you figure out what was wrong? |
| 13:43.20 | Squeeb | or is there a better way of using MySQL with asterisk? |
| 13:43.26 | Squeeb | to record calls in and out? |
| 13:44.45 | *** join/#asterisk mog (n=mog@nat/digium/x-662e59cbe2f1b0f4) |
| 13:44.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:46.25 | phpboy | kaldemar: I believe it's because I was missing one channel, why it would do this, i don't know |
| 13:46.26 | phpboy | but ja |
| 13:46.28 | phpboy | it's working now |
| 13:46.35 | phpboy | thanks a million :D |
| 13:46.46 | c4t3l | Katty: Good morning! |
| 13:46.53 | *** join/#asterisk angryuser (n=Miranda@43.252.146.195.dynamic.adsl.abo.nordnet.fr) |
| 13:46.59 | Katty | c4t3l: hi |
| 13:47.25 | kaldemar | it wasn't because of the missing channel, but your d-channel being set as the 17th channel on the span, asterisk doesn't like any other than 16th. |
| 13:49.02 | padski | What does asterisk do if it misses rtp packets from the other end due to packet drops ? |
| 13:49.45 | Katty | c4t3l: i'm a bit grumpy today due to being up with riddick half the night )= |
| 13:50.10 | c4t3l | Katty: How's the pup doin? |
| 13:51.41 | phpboy | kaldemar: this I did not know |
| 13:52.07 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 13:52.47 | phpboy | Thanks again, I really do appreciate your help |
| 13:55.59 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:58.44 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 14:03.09 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-151.washdc.east.verizon.net) |
| 14:10.23 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:10.49 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-176.usadatanet.com) |
| 14:11.28 | ctaloi | hey all - anyone have an example of a rollover type dial plan? example, call comes in to DIDA, if DIDA busy try DIDB etc... |
| 14:11.29 | ctaloi | ? |
| 14:11.57 | [TK]D-Fender | ctaloi: Inbound rollover is ont *'s job, its the telco's <- |
| 14:12.17 | ctaloi | [TK]D-Fender - in this case, I am the telco |
| 14:12.24 | ctaloi | providing DID trunking |
| 14:12.27 | [TK]D-Fender | s/is ont/isn't/ |
| 14:12.46 | phpboy | kaldemar: what does it mean if asterisk says that a zap channels status is yel? |
| 14:12.53 | phpboy | yellow I'm guessing |
| 14:13.11 | UnixDawg | that it has turned chicken |
| 14:13.12 | [TK]D-Fender | ctaloi: if the call comes in on another DID then you do whatever you want with it. |
| 14:13.19 | coppice | or the gain is too high |
| 14:14.05 | [TK]D-Fender | ctaloi: Sound like you need to keep a rollover DB for the order to attempt channels in. |
| 14:14.07 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:14.26 | [TK]D-Fender | ctaloi: start loop, try1, increment, try 2, etc. |
| 14:15.08 | ctaloi | D-Fender - that sounds about right...... do you know of any examples of that in a dial plan I can refrence? |
| 14:16.01 | [TK]D-Fender | ctaloi: No such thing. How you do it is up to you and your DB method, etc. And the mere fact you're asking this and claiming to be a telco scares the hell out of me. |
| 14:16.13 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a1191a1782757a23) |
| 14:16.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:16.46 | ctaloi | D-Fender - No reason to be scared, the call flow isn't what you think it is... |
| 14:16.54 | [TK]D-Fender | ctaloi: Up to the method you handle your peers with. What tech... only about a few dozen different factors.. |
| 14:17.19 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:17.52 | ctaloi | [TK]D-Fender - Agreed, was just wondering if anyone had any examples, no biggie |
| 14:18.11 | [TK]D-Fender | ctaloi: Too many factors for there to be a usable sample |
| 14:19.05 | ctaloi | Okay... Thanks |
| 14:19.58 | [TK]D-Fender | ctaloi: If you're looking at a hard-coded method , in order you would check the status of 1st. If in-use jump to next, otherwise dial it. |
| 14:20.35 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 14:21.09 | phpboy | woah, this second span is freaking out |
| 14:21.28 | ctaloi | yah, that's what I'm thinking... I'm not routing based on the DID, I'm routing a 1k block to the Asterisk, and then matching and routing to a username on an Adtran at the customer CPE, the user equals a physical DS0 on the CSR's gear |
| 14:21.40 | shriven | hello. I am curious about asterisk 1.6.1, I am curious if anyone knows why it was split fro 1.6.0? Especially as 1.6.0 isn't even released? |
| 14:21.40 | phpboy | getting the following for all the channels on the second span on the second card:- [Oct 1 16:19:38] WARNING[9608]: chan_zap.c:1465 zt_disable_ec: Unable to disable echo cancellation on channel 185 |
| 14:22.19 | [TK]D-Fender | shriven: Because 1.6.0 RC is feature locked <- |
| 14:22.31 | shriven | ahhh ok thanks |
| 14:23.05 | adr3nalin3 | phpboy: I have been getting that too. ' |
| 14:23.31 | shriven | okay, another question. Is the ldap integration built into 1.6.0 usable? |
| 14:23.40 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:24.01 | phpboy | adr3nalin3: it's got something to do with the dchan :/ |
| 14:24.18 | phpboy | kaldemar: you in for a little bit of final advice (hopefully) |
| 14:24.33 | adr3nalin3 | mine did too. I kept "losing" my d-channel |
| 14:24.40 | phpboy | bchan=156-170,172-186 \n dchan=171 <--- that looks correct? |
| 14:24.51 | adr3nalin3 | Is it a te122p? |
| 14:25.10 | beek | morning |
| 14:25.23 | phpboy | hi |
| 14:25.29 | phpboy | no |
| 14:25.45 | [TK]D-Fender | shriven: Go try |
| 14:25.50 | phpboy | adr3nalin3: 2 x Digium Quads |
| 14:26.02 | phpboy | I forgot the model number :( |
| 14:26.15 | shriven | well I am hoping to, but if someone here already knows that it's jacked, no point in wasting my time. ;) |
| 14:26.38 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000022.dsl.bell.ca) |
| 14:27.12 | adr3nalin3 | phpboy: did your log look similar to this? http://pastebin.com/m45615c9a |
| 14:27.26 | beek | [TK]D-Fender: I just had an interesting conversation with my proprietary system vendor. I want to put an * box in front of it and have my PSTN PRI connect to the * box, then from * to the Iwatsu. They asked if * can replicate NI2 (e.g. act like the CO). I assume yes, but can it? |
| 14:27.40 | adr3nalin3 | yes |
| 14:28.01 | [TK]D-Fender | beek: Yes |
| 14:28.48 | beek | [TK]D-Fender: That's excellent news. Thanks very much. |
| 14:28.52 | phpboy | adr3nalin3: mine looks identical |
| 14:28.57 | phpboy | I can help you fix yours though |
| 14:29.08 | [TK]D-Fender | beek: "signalling=pri_cpe" <- to telco. "signalling=pri_net" <- to PBX |
| 14:29.08 | phpboy | in /etc/zaptel.conf |
| 14:29.33 | [TK]D-Fender | beek: You would take timing from your telco span, and pass it to your PBX span |
| 14:29.40 | phpboy | adr3nalin3: you using T1 or E1? |
| 14:29.43 | adr3nalin3 | t1 |
| 14:30.03 | phpboy | adr3nalin3: I don't think your dchan on zaptel.conf should be 24 :( |
| 14:30.25 | *** join/#asterisk murraytm (n=murraytm@wsip-68-224-219-238.br.no.cox.net) |
| 14:30.31 | phpboy | my country uses E1, so I'm not 100% sure about T1 but I'm pretty sure your dchan shouldn't be 24 |
| 14:30.44 | beek | [TK]D-Fender: I really appreciate your help. Thanks. |
| 14:30.52 | adr3nalin3 | I think it should, bchan 1-23 dchan 24 |
| 14:31.23 | *** join/#asterisk cguerrero (n=cuauhtem@200.79.231.94.static.cableonline.com.mx) |
| 14:31.26 | cguerrero | hello |
| 14:32.06 | cguerrero | can any one help me, I like to know if it is posible to dial a cel phone and get the voicemail mesage |
| 14:32.18 | adr3nalin3 | Yes on my other box it is that way working well, but when I checked my /etc/zaptel.conf it has bchan 1-22, which I have fixed already but the damn gui overwrote again |
| 14:32.20 | [TK]D-Fender | adr3nalin3: It should.... |
| 14:32.21 | cguerrero | because right now when I dial a cel i get a busy tone |
| 14:32.24 | phpboy | adr3nalin3: Try to confirm that, because I think I read otherwise about T1's and asterisk |
| 14:32.45 | cguerrero | after 30 seconde |
| 14:32.57 | phpboy | I wonder why my config isn't happy with the dchan :( |
| 14:33.07 | cguerrero | and i f I dial from a landline I get the recording for the voicemail |
| 14:33.20 | phpboy | [TK]D-Fender: Does my config for E1 seem right for the 6th span? bchan=156-170,172-186 \n dchan=171 |
| 14:35.34 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 14:35.44 | murraytm | i'm having a problem where when asterisk is started from /etc/init.d/asterisk, my AGI scripts can't load some libraries. if asterisk is started from /usr/sbin/asterisk, everything works fine. what's the difference between those two environments? |
| 14:35.51 | [TK]D-Fender | phpboy: Depends on what cards, what order, etc. I'm not going to answer based on 1 single line like that. |
| 14:36.25 | c4t3l | <PROTECTED> |
| 14:36.32 | murraytm | centos 5 |
| 14:36.32 | [TK]D-Fender | murraytm: Because your init script might be running * as non-root. |
| 14:36.44 | c4t3l | good point! |
| 14:36.58 | [TK]D-Fender | murraytm: And you shoud already have been fully aware of that. |
| 14:37.07 | [TK]D-Fender | should* |
| 14:37.37 | murraytm | i figured as much, but can't tell what user it's running as. i didn't configure it to run as non-root |
| 14:38.15 | angryuser | i need to save all sip debug output of defined peer to a file, any help ? |
| 14:38.49 | murraytm | so then i need to get with some centos / redhat people then? |
| 14:39.05 | [TK]D-Fender | murraytm: Easy to see. |
| 14:39.12 | c4t3l | murraytm: did you install via rpm? |
| 14:39.14 | lowtek | angryuser: asterisk -rx "sip show peer XXX" > outputfile |
| 14:39.21 | [TK]D-Fender | murraytm: How did you install * in the first place? |
| 14:39.37 | lowtek | oh, debug output, sorry |
| 14:39.38 | murraytm | installed from source |
| 14:39.52 | tzafrir_laptop | phpboy, I just use zapconf / genzaptelconf to generate zaptel.conf without worrying about the exact numbers |
| 14:39.58 | [TK]D-Fender | murraytm: go look at the running daemon and see what user its running as. |
| 14:41.41 | murraytm | looks like it's running as root |
| 14:44.03 | [TK]D-Fender | murraytm: go run System with an open shell script that will confirm things. |
| 14:44.55 | *** join/#asterisk leif[astricon] (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 14:45.20 | murraytm | [TK]D-Fender: "System"? |
| 14:45.32 | [TK]D-Fender | murraytm: "core show application system" |
| 14:45.55 | phpboy | tzafrir_laptop: thanks a lot, it generated it the way i did, which is good news, anyway... asterisk is still giving me issues about the dchan getting yellow alarm(4), any ideas? |
| 14:46.39 | tzafrir_laptop | phpboy, do you use crc4 on that line? |
| 14:46.46 | tzafrir_laptop | is that line ok? |
| 14:49.17 | *** join/#asterisk seanmh (i=seanmh@216.31.101.24) |
| 14:49.34 | phpboy | tzafrir_laptop: yeah, seems fine |
| 14:49.35 | phpboy | crc4, yes |
| 14:49.50 | phpboy | oh, you mean the physical line? |
| 14:50.21 | tzafrir_laptop | Yes. Someone along the line has a red alarm |
| 14:50.31 | tzafrir_laptop | At least in one direction |
| 14:51.09 | phpboy | hmmmm, I think it may be on my telco's side |
| 14:51.25 | phpboy | this is possible, yes? |
| 14:57.59 | *** part/#asterisk leif[astricon] (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 14:58.55 | *** join/#asterisk propellerhead (n=yogurt2u@200.41.65.114) |
| 15:01.08 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 15:02.16 | Katty | [TK]D-Fender: my boss just discovered sip trunking :< |
| 15:03.07 | cguerrero | how can I gate ringing tones from asterisk before early media |
| 15:03.46 | cguerrero | get |
| 15:04.47 | *** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 15:05.00 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:05.33 | Katty | hi james. |
| 15:05.48 | Katty | seanmh: ping? |
| 15:06.42 | seanbright | ~siptrunk |
| 15:06.42 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 15:06.54 | seanbright | ~seanbright |
| 15:06.55 | jbot | from memory, seanbright is a girl with standards |
| 15:06.59 | seanbright | correct! |
| 15:07.03 | jameswf | ~me |
| 15:07.03 | jbot | no u |
| 15:07.08 | Katty | ~Katty |
| 15:07.08 | jbot | you are probably the only girl in the channel, so be nice to her |
| 15:07.21 | Katty | jbot: Katty? |
| 15:07.22 | jbot | you are probably the only girl in the channel, so be nice to her |
| 15:07.26 | *** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 15:07.26 | jameswf | ~jameswf |
| 15:07.27 | jbot | jameswf loves unsolicited technical support, or http://jameswf.info |
| 15:07.29 | Katty | we need mutiple replies )= |
| 15:07.34 | [TK]D-Fender | Katty: ... |
| 15:07.36 | [TK]D-Fender | ~siptrunk |
| 15:07.37 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 15:07.40 | jameswf | ~porn |
| 15:07.41 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
| 15:07.46 | jameswf | ~botabuse |
| 15:07.46 | jbot | [botabuse] fun |
| 15:07.52 | Katty | [TK]D-Fender: i'm scared )= |
| 15:07.55 | tzanger | ha |
| 15:08.08 | tzanger | it's a good thing the porn link and the siptrunk link aren't related |
| 15:08.09 | [TK]D-Fender | Katty: that's "skeered" ;) |
| 15:08.11 | Katty | [TK]D-Fender: i hear bad things about sip trunks )= |
| 15:08.37 | jameswf | got junk in the sip trunk hayyyyyy |
| 15:08.44 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 15:08.46 | creativx | i see dead sip trunks |
| 15:08.54 | Katty | cries. |
| 15:09.04 | Katty | i just wanna go home and play with my puppy |
| 15:09.07 | Katty | the world is too skeery today |
| 15:09.09 | tzanger | the best take on that was "I see dumb people.. they're everywhere.. I work with them." |
| 15:09.16 | jameswf | creativx: you know some dont even know they are dead |
| 15:09.32 | gr0mit | gaaaa - my asterisk box just died. |
| 15:09.33 | [TK]D-Fender | tzanger: "If I don't work for a circus.... then why am I surrounded by clowns?" |
| 15:09.37 | *** join/#asterisk diverscuba (n=diverscu@d-72-9-4-157.cpe.metrocast.net) |
| 15:09.39 | tzanger | [TK]D-Fender: that's another good one |
| 15:09.51 | jameswf | Everytime you boot trixbox god kills a puppy.... thats what i hear |
| 15:09.53 | tzanger | this vhdl is killing me |
| 15:10.22 | Katty | holds riddick :< |
| 15:10.23 | creativx | jameswf: indeed |
| 15:10.28 | Katty | NOT MY PUPPEH! |
| 15:10.42 | gr0mit | sneezes |
| 15:10.44 | jameswf | Katty: put down the lime green box |
| 15:10.47 | Katty | i haven't had a chance to nom on his ears yet )= |
| 15:10.58 | murraytm | [TK]D-Fender: i must not understand what an open shell script is or how i should be using system. i put it in the dialplan and ran it, but came up with nothing useful. |
| 15:11.05 | Katty | jameswf: i stay farrrr away from trixbox |
| 15:11.40 | jameswf | Lime should be reserved for alchoholic beverages |
| 15:11.54 | jblack | And concrete |
| 15:11.59 | *** join/#asterisk diverscuba (n=diverscu@d-72-9-4-157.cpe.metrocast.net) |
| 15:12.19 | gr0mit | and cheesecake |
| 15:12.23 | *** join/#asterisk logicwrath (n=no@c-68-42-253-39.hsd1.mi.comcast.net) |
| 15:12.24 | jameswf | I thought that was lye... also good for disposing of bodies |
| 15:12.27 | Katty | i like cheesecake. |
| 15:12.57 | *** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 15:13.32 | jblack | jameswf: Absolutely true. Just a few days ago, I saw a pile of dead puppies, and figured "someone must have rebooted their trixbox server a couple dozen times". |
| 15:13.36 | jameswf | I am here today to talk about the effects of ADD/ADHD in the business world....... oh crap is that the new ipod |
| 15:14.07 | murraytm | did i join 4chan or something when i wasn't looking? :) |
| 15:14.10 | creativx | lye? lutefisk? si si |
| 15:17.36 | [TK]D-Fender | murraytm: #!/bin/bash whoami > someplace&file |
| 15:18.03 | [TK]D-Fender | murraytm: No, this is clearly SomethingAwful |
| 15:18.21 | jameswf | Awful is relitive |
| 15:20.08 | murraytm | thanks, it's definitely running as root |
| 15:20.26 | jameswf | heh http://houghi.org/shots/vim001.gif |
| 15:20.51 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 15:21.03 | *** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 15:21.21 | tzanger | jameswf: haha |
| 15:21.24 | tzanger | old but fun |
| 15:21.38 | murraytm | jameswf: awesome |
| 15:21.59 | jameswf | I use to have one... it appears you are trying to write a suicide note.... |
| 15:22.05 | tzanger | yeah |
| 15:24.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:27.22 | [TK]D-Fender | jameswf: I have that one at home. |
| 15:29.37 | *** join/#asterisk jsolis (n=jimmy@190.232.168.56) |
| 15:29.37 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 15:36.29 | *** join/#asterisk Lorax (n=lorax@loki.aceldama.com) |
| 15:38.31 | *** join/#asterisk adr3nalin3_ (n=afink@asa.redglaze.com) |
| 15:45.38 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 15:46.56 | *** join/#asterisk afink (n=afink@asa.redglaze.com) |
| 15:46.59 | murraytm | [TK]D-Fender: any other suggestions? if it's running as root, what else could be different? |
| 15:47.33 | [TK]D-Fender | murraytm: You never actually showed your problem or any of your backup so I have nothing to comment on. |
| 15:49.24 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 15:49.34 | murraytm | sorry, i thought i had. the problem is that my AGI scripts can't load some libraries when it was started by /etc/init.d/asterisk (as opposed to just /sbin/asterisk). what else do you need? |
| 15:49.46 | [TK]D-Fender | ~pb |
| 15:49.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:49.49 | [TK]D-Fender | ^^^^ |
| 15:50.24 | [TK]D-Fender | murraytm: I would need to see the problem, not your description of it. |
| 15:52.24 | murraytm | i'll see if i can put together a simple reproduction case and come back later |
| 15:52.27 | murraytm | thanks for your help |
| 15:52.28 | *** part/#asterisk murraytm (n=murraytm@wsip-68-224-219-238.br.no.cox.net) |
| 15:54.33 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:55.11 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:56.29 | [TK]D-Fender | Amazing... ask to actually see a problem and *poof*, out they go. |
| 15:56.29 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 16:00.10 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-151.washdc.east.verizon.net) |
| 16:01.48 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-151.washdc.east.verizon.net) |
| 16:03.41 | *** join/#asterisk DJ_HaMsTa (i=DJ_HaMsT@c-76-124-82-218.hsd1.nj.comcast.net) |
| 16:06.26 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-178-166.phil.east.verizon.net) |
| 16:12.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:16.55 | *** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net) |
| 16:22.11 | *** join/#asterisk hardwire (n=hardwire@rdbk-996.mtaonline.net) |
| 16:22.16 | hardwire | hai' |
| 16:25.49 | hardwire | I mean it |
| 16:25.59 | *** join/#asterisk ElSonico (n=tav@213-157-93-129.localloop.fi) |
| 16:26.15 | hardwire | blah |
| 16:27.21 | jameswf | has the sky fallen yet? |
| 16:30.21 | bkw_ | jameswf: yep.. I'm sweeping it up as I type |
| 16:30.29 | bkw_ | jameswf: do you want that in a bag or a box? |
| 16:30.58 | jameswf | ummmm is the bag recycled? |
| 16:31.24 | thehar | paper or plastic? |
| 16:31.30 | thehar | or did you bring your own bag with you this am? |
| 16:31.46 | c4t3l | good lord what is with ppl today!!!??? |
| 16:31.57 | c4t3l | not you guys of course |
| 16:32.03 | c4t3l | you guys are cool |
| 16:32.06 | jameswf | i feel like a crazy peson bringing my own bags |
| 16:32.11 | jameswf | I need to go hug a tree |
| 16:32.35 | jaytee | if it's a giant redwood, better bring alot of friends |
| 16:33.01 | bkw_ | haha |
| 16:33.03 | jameswf | If its a gialnt redwood i am going to bring a chain saw psh |
| 16:33.07 | c4t3l | Katty: You still grumpy? |
| 16:33.09 | thehar | the checkers at whole foods look down on you when you don't bring a bag and ask for plastic |
| 16:34.22 | jameswf | I look at em and say F___ off i use linux i am a better hippy than you will ever be |
| 16:35.04 | tzanger | haha |
| 16:35.13 | jameswf | every year billions of trees are killed so microsoft and apple can compete..... tree killers |
| 16:35.33 | hardwire | those bastards |
| 16:35.40 | thehar | the signature in my company's announcements is "save a tree use email!" |
| 16:35.42 | thehar | tender, i know. |
| 16:35.45 | hardwire | jameswf: I put my hippy foot out the other day |
| 16:35.59 | hardwire | Sun was proclaiming being eco friendly by using x86 in their proprietary chassis |
| 16:36.09 | hardwire | "Can I replace the mainboard with any ATX mainboard?" |
| 16:36.10 | hardwire | no.. |
| 16:36.18 | jameswf | Stop global warming kill a cow |
| 16:36.22 | hardwire | "Can I do anything to it that would use off the shelf parts?" |
| 16:36.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:36.25 | hardwire | not really.. no |
| 16:36.35 | *** part/#asterisk mesfet (n=iw3grx@89-96-227-146.ip14.fastwebnet.it) |
| 16:36.37 | thehar | we use 100% reuseable energy in our full colocation and our offices. http://www.xmission.com/community/green/ |
| 16:36.40 | hardwire | "So you guys made yet another chassis we have to melt down and recycle?" |
| 16:36.44 | hardwire | yes.. yes we did. |
| 16:37.02 | hardwire | thehar: you work for xmisson eh? |
| 16:37.04 | thehar | yes |
| 16:37.07 | hardwire | GODO 4 U |
| 16:37.10 | thehar | oh? |
| 16:37.12 | hardwire | err |
| 16:37.12 | hardwire | good |
| 16:37.13 | hardwire | oh |
| 16:37.20 | hardwire | is emailing all the people he met @ astricon today |
| 16:37.20 | *** part/#asterisk jsolis (n=jimmy@190.232.168.56) |
| 16:37.24 | hardwire | a bit late.. better late than never. |
| 16:38.02 | thehar | how do you know about xm? |
| 16:38.06 | thehar | if you say maddox i'll laff |
| 16:38.06 | hardwire | dude |
| 16:38.09 | hardwire | I get around |
| 16:38.11 | hardwire | and yes. |
| 16:38.21 | thehar | best page in the universe |
| 16:38.21 | hardwire | not directly |
| 16:38.38 | hardwire | I'm the occasional onlooker |
| 16:38.54 | thehar | hehe |
| 16:39.00 | thehar | yeah XM rocks. |
| 16:39.06 | hardwire | jobplz? |
| 16:39.17 | hardwire | kthx. |
| 16:39.32 | thehar | you want to live in the big slc, eh? |
| 16:39.43 | hardwire | I'm not holy enough for SLC |
| 16:39.56 | jjshoe | slc? |
| 16:40.03 | thehar | if you live in the metro area you will be just fine.. outside of downtown you're a big sinner |
| 16:40.08 | thehar | salt lake CITY, utard |
| 16:40.22 | hardwire | way to wave the politically correct flag :) |
| 16:40.32 | hardwire | I love pointing my finger at PC violators |
| 16:40.35 | hardwire | it gives me a rush |
| 16:40.40 | thehar | :D |
| 16:40.44 | hardwire | except I think I said retarded at least 15 times yesterday |
| 16:40.51 | hardwire | and I inspired somebody else to say it |
| 16:41.03 | thehar | no that's the word for utahns. it's really the perfect definition. |
| 16:41.10 | jameswf | Retarded is PC |
| 16:41.13 | hardwire | I said "not like this is all that PC.. but.. this cement is retarded.. sorry, that's just how I feel" |
| 16:41.22 | hardwire | which made the person I was talking to somehow say it 6 more times. |
| 16:41.42 | hardwire | jameswf: anything derogatory isn't PC. :) |
| 16:41.48 | jameswf | the support groups are all caller "arc" XXXX associatonn for retarded citizins |
| 16:42.52 | hardwire | like.. you never here "and lets give it up for all the retards here at the special olympics.. bang up job guys." |
| 16:43.00 | hardwire | hear |
| 16:43.01 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:43.01 | hardwire | blah |
| 16:43.04 | hardwire | I'm a bit out of whack today. |
| 16:43.10 | bkw_ | TODAY only? |
| 16:43.14 | bkw_ | I think ALWAYS |
| 16:43.18 | hardwire | do I know you? |
| 16:43.38 | bkw_ | no clue... but apparently you haven't a sense of humor :P |
| 16:43.50 | Cutlass | dumb question...I just upgraded to 1.6 from some pre-1.2 version....isn't there a CLI command to show all the SIP users? |
| 16:43.51 | jameswf | YOu are all differently abled |
| 16:43.58 | hardwire | bkw_: I never said you were wrong. |
| 16:44.02 | thehar | wat is a sense of humor, precious? |
| 16:44.05 | hardwire | I just never know who's looking over my shoulder lately. |
| 16:44.06 | bkw_ | hardwire: hehe |
| 16:44.14 | bkw_ | hardwire: I know I'm totally freakin nutz |
| 16:45.08 | hardwire | bkw_: that's a matter or perspective. |
| 16:45.13 | bkw_ | could be |
| 16:46.13 | hardwire | hugs his repaired laptop |
| 16:46.14 | thehar | submits order for Asterisk: The Future of Telephony, 2nd Edition |
| 16:46.17 | hardwire | I missed it soooo |
| 16:46.22 | hardwire | slowest laptop evar |
| 16:46.23 | hardwire | but I love it |
| 16:46.56 | jameswf | thanks for the reminder i need to order a couple books for give aways |
| 16:48.36 | thehar | i have the lovely pdf but a physical book makes life so much easier |
| 16:49.05 | bkw_ | thehar: I use mine as a monitor stand! |
| 16:49.58 | thehar | i have lovely three 22" LG lcds.. i don't need monitor stands :) |
| 16:50.36 | bkw_ | I have a single 24 inch iMac |
| 16:50.36 | jameswf | I find consoles look funny when you put it on big screens |
| 16:50.47 | bkw_ | jameswf: I love it.. my terminal is 160x61 |
| 16:50.52 | bkw_ | at 14 pt |
| 16:50.52 | jameswf | ooh washington DC calling |
| 16:51.09 | thehar | mine is 93 x 28 |
| 16:51.17 | thehar | at like 8 pt |
| 16:51.24 | bkw_ | you must not love your eyes |
| 16:51.25 | bkw_ | :P |
| 16:51.54 | thehar | i have a lot of terminals to use and windows! i need space |
| 16:52.07 | [TK]D-Fender | Mine is 80x25 @ 1.2 IPC ;) |
| 16:52.44 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 16:55.29 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 16:55.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:58.09 | thehar | how easily asterisk-users takes over my mailbox |
| 17:02.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:03.22 | *** join/#asterisk nny_2 (n=scott@64.203.244.146) |
| 17:04.04 | *** join/#asterisk J0n5555 (n=caldwell@cpe-71-67-178-48.neo.res.rr.com) |
| 17:04.33 | logicwrath | im using all SIP, with softphones and a 1 or 2 desk phones. I have no digium hardware in my server. What kind of EC solutions can I use? I have soft echo but I think it required a digium card. I would buy a cheap card if that would work but all the channels will be using SIP and skipping teh card. I was not sure if that would work. Can someone advise? |
| 17:04.47 | *** join/#asterisk hardwire` (n=hardwire@rdbk-11447.mtaonline.net) |
| 17:04.49 | logicwrath | *I have seen soft echo.. |
| 17:05.09 | angryuser | can someone add a tag to ~siemens like this 'C675ip 2 phones OK: C650ip 2 phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00, : C450ip 2 phones OK' |
| 17:05.18 | Greek-Boy | any call center admins here? |
| 17:06.44 | [TK]D-Fender | logicwrath: If you've got echo with an ITSP, its their fault & problem. |
| 17:07.03 | [TK]D-Fender | logicwrath: There is no EC over VoIP, its done at the PSTN level. |
| 17:08.08 | logicwrath | is ITSP internet telephone service provider? |
| 17:08.15 | *** join/#asterisk docelmo (n=docelmo@206.248.239.194) |
| 17:08.37 | J0n5555 | i have, I hope, a simple question... I'm wanting to setup a voip pbx for 4 of my family members and me... we call each other alot, but still need to call out as well.... If i get to voip trunks for 5 houses, i know I can save upwards of $200-300 a month (done the math already)... however, is there any way to report different addresses for 911 service? three of us are disabled and are concerned |
| 17:09.08 | docelmo | e911 is attached to your DID |
| 17:09.14 | [TK]D-Fender | ~seimens |
| 17:09.27 | docelmo | so if every house has a seperate DID then the e911 info will be registered to this house. |
| 17:09.47 | [TK]D-Fender | ~siemens |
| 17:09.48 | jbot | [~seimens] Seimens C675IP, C650IP, C450IP phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00 |
| 17:09.57 | docelmo | Your outbound routing will need to be setup as such the correct callerid is presented to the outbound provider correctly for the caller |
| 17:10.02 | J0n5555 | docelmo: hmm... I was hopeing to save money by just having 1 did and using extensions... but if that's the way it needs to be I can go read up on it |
| 17:10.04 | angryuser | thank you |
| 17:10.34 | docelmo | j0n5555 DID's are fairly inexpensive $2-10 per DID in most cases |
| 17:11.10 | docelmo | then just purchase pay as you go outbound this way your MRC is only the DID's it will end up being maybe ~60 a month for 5 house holds at worst case scenario |
| 17:11.15 | J0n5555 | docelmo: great, yea, i remember seeing i on the list... I'm looking at bandwidth.com, but haven't shopped around much yet... just getting my feet warm |
| 17:11.36 | docelmo | j0n check www.aretta.net they have hosted pbx's also |
| 17:12.08 | J0n5555 | docelmo: ty! |
| 17:12.13 | docelmo | all good |
| 17:12.27 | [TK]D-Fender | logicwrath: Yes |
| 17:12.57 | docelmo | say anyone done any load testing with meetme on asterisk to find the max number of conferences that can be brought up under ztdummy or dahdidummy? |
| 17:12.59 | [TK]D-Fender | ~siemens |
| 17:12.59 | jbot | [~siemens] Siemens C675IP, C650IP, C450IP phones OK: C470ip -> bad firmware for * V02123: 021230000000 / 041.00 |
| 17:13.06 | [TK]D-Fender | There, fixed the spelling |
| 17:14.25 | angryuser | [TK]D-Fender : you need to specify the number of phones, siemens is able to support 6 for some bases, and the rumor says the instability is more important with more phones, but i cant confirm or deny that, i have tested with 2 only |
| 17:15.03 | docelmo | [tk]d-fender any knowledge of meetme? |
| 17:15.28 | [TK]D-Fender | docelmo: Not to answer your previous question with. |
| 17:15.43 | docelmo | wonder if there is anything on google.. |
| 17:16.06 | docelmo | I guess we shall see or I could use SIPp and just bombard a box with ulaw and see where it explodes |
| 17:16.32 | [TK]D-Fender | ~siemens |
| 17:16.32 | jbot | [~siemens] Siemens C675IP, C650IP, C450IP phones OK using 2 phones. C470ip -> bad firmware for * V02123: 021230000000 / 041.00 |
| 17:19.01 | angryuser | nice phones, on the paper ;) |
| 17:19.03 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:19.58 | [TK]D-Fender | angryuser: Same can be said of Grandstream |
| 17:20.49 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 17:20.58 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:22.00 | UnixDawg | anyone know when the new asterisk-now iso is going to be released ? |
| 17:22.24 | [TK]D-Fender | UnixDawg: "when its done" |
| 17:22.58 | UnixDawg | well I thought it was as they had a dlsite atastricon some one said. |
| 17:23.04 | UnixDawg | was it only a beta |
| 17:23.23 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 17:23.50 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.207.97) |
| 17:23.57 | scooby2 | offtopic kinda: Anyone know if you can make an 800# failover? We have two pri's from different providers. If one provider goes down, is there any way to make the 800's flip to the other? |
| 17:24.10 | scooby2 | or any other way to make it more robust |
| 17:24.51 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 17:24.53 | *** join/#asterisk zacwolfe (n=zxwolfe@c-67-171-236-13.hsd1.or.comcast.net) |
| 17:25.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:25.38 | [TK]D-Fender | UnixDawg: I'll ammend that with "When its no longer beta" :) |
| 17:26.44 | [TK]D-Fender | scooby2: Yes, thats the telco's job. |
| 17:28.33 | scooby2 | I was hoping someone here had experience with that. We have two different providers so we have a little redundancy but it would be nice if the numbers could roll over some how |
| 17:28.36 | *** join/#asterisk hardwire` (n=hardwire@srv001.gandi.brutetech.com) |
| 17:28.44 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 17:29.35 | [TK]D-Fender | scooby2: Nothing for us to have experience with. Its the TELCO's job |
| 17:30.20 | gsiener | Recommendations for US iax trunk provider? |
| 17:31.01 | [TK]D-Fender | gsiener: IAX is generally unrecommended unless necessary. Otherwise, Teliax |
| 17:31.17 | *** join/#asterisk chigital (n=chigital@tmo-122-1.customers.d1-online.com) |
| 17:31.21 | gsiener | I guess I meant voip trunk provider |
| 17:31.22 | jksM | why is it that IAX is not recommended? |
| 17:31.33 | [TK]D-Fender | jksM: Stability & quality |
| 17:31.58 | jksM | hmm, I'm using IAX2 for trunking right now... should I consider switching to SIP? - what kind of stability and quality problems can I run into? |
| 17:32.02 | jksM | are we talking sound quality or? |
| 17:32.14 | jameswf | Where do i get a voip trunk google says i cant have one |
| 17:32.17 | hardwire | sigh |
| 17:32.33 | jameswf | :) |
| 17:32.53 | gsiener | <PROTECTED> |
| 17:33.11 | [TK]D-Fender | jksM: If you are running fine, then leave it. |
| 17:33.49 | [TK]D-Fender | gsiener: what is "looping"? What is "logging in" supposed to mean? |
| 17:34.09 | jksM | well, I have a weird problem with audio not functioning properly when the machine is under high load (problem = parties cannot hear each other, even though both sound streams are available when doing a network sniff at the asterisk server) |
| 17:34.38 | gsiener | [TK]D-Fender: when I log into asterisk-gui, it enters a loop of parsing the config files repeatedly, instead of logging in |
| 17:35.13 | [TK]D-Fender | gsiener: Definitely a problem for their channel then. This is not 2nd level GUI support |
| 17:35.29 | [TK]D-Fender | jksM: any transcoding? |
| 17:35.34 | jksM | no transcoding |
| 17:35.42 | gsiener | [TK]D-Fender: fair enough, just wondering if anyone had seen it before |
| 17:36.28 | [TK]D-Fender | jksM: Try using SIP and see if it improves. |
| 17:37.28 | *** join/#asterisk mltlnx (n=mltlnx@207-237-36-133.c3-0.nyw-ubr3.nyr-nyw.ny.static.cable.rcn.com) |
| 17:37.50 | jksM | hmm, I would like to be more sure that it is IAX that is the culprit before switching... it's hardcoded a lot of places |
| 17:37.56 | mltlnx | Hello, I have the need to block inbound calls from a certain area code....What is the best way to do this? |
| 17:38.03 | jksM | and it's very hard to debug as it seems to happen only once a week or something like that |
| 17:38.39 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 17:39.04 | Katty | is there a way to do something /after/ a caller hangs up |
| 17:39.23 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 17:39.23 | Katty | exten => foo,1,Dial - exten => foo,2,if user hangs up do this |
| 17:39.35 | nny_2 | Is anyone here well versed with Tellabs ECs? I am in a bind here. |
| 17:41.19 | [TK]D-Fender | Katty: read up on your Asterisk Standard Extensions.... |
| 17:41.52 | *** join/#asterisk Talirk (i=434e2716@gateway/web/ajax/mibbit.com/x-241dfef0ddd48b2f) |
| 17:42.02 | Katty | i need something more specific than that. |
| 17:42.09 | Katty | not all the time |
| 17:42.13 | angryuser | Katty: : g option in dial or use 'n' in priority |
| 17:42.15 | Katty | just when certain numbers are dialed. |
| 17:43.10 | Katty | ooh, g will be useful |
| 17:43.10 | Talirk | Does anyone here use FreePBX, I have a question on apply a custom dialrule , but I want to do it thru the gui verus directly editing the conf files. Is it possible |
| 17:43.26 | jaytee | mltlnx, you could use a GotoIf where if the area code of the callerid matches the unwanted area code it would jump to a priority that hangs up the call or plays a message and hangs up and if it doesn't match just continues in the dialplan. |
| 17:43.39 | [TK]D-Fender | Katty: "h" <- |
| 17:43.54 | [TK]D-Fender | Talirk: Ask in #freepbx . It is not supported here |
| 17:43.54 | mltlnx | thanx |
| 17:43.59 | Katty | h? |
| 17:44.01 | Katty | looks at h |
| 17:44.13 | Katty | [TK]D-Fender: that's not what i want. |
| 17:44.22 | Katty | angryuser: g is what i want. |
| 17:44.24 | Katty | angryuser: thanks |
| 17:46.14 | zacwolfe | ANNOUNCEMENT: Visual call-flow editor and stand-alone server now in Beta and available for free download from http://www.safisystems.com |
| 17:46.58 | Katty | angryuser: i'll pastebin what i'm doing if it works |
| 17:47.12 | jameswf | zacwolfe: for linux? |
| 17:47.41 | jjshoe | zacwolfe nice, based on eclipse rcp? |
| 17:47.54 | *** join/#asterisk tkbeat (n=tk@p54B9474A.dip.t-dialin.net) |
| 17:47.59 | zacwolfe | yeah its an RCP-based app |
| 17:48.04 | Katty | does anyone know where rm is? |
| 17:48.16 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 17:48.19 | zacwolfe | the server runs on Linux but designer is only avail for Windows at this time |
| 17:48.21 | jameswf | which rm |
| 17:48.28 | Katty | System,1,(rm /this/file) |
| 17:48.48 | jameswf | james@james-laptop:~$ which rm |
| 17:48.48 | jameswf | /bin/rm |
| 17:49.03 | Katty | cheers. |
| 17:49.39 | jameswf | doesnt own a windoze box :( |
| 17:50.13 | Katty | yay it works |
| 17:51.19 | Katty | angryuser: http://pastebin.ca/1215999 |
| 17:51.40 | angryuser | h is good too, but not like option to dial(), at hangup dial searches for h in context where dial is and executes it's code. I am not sure if you need to add another option to dial() to make it happen |
| 17:51.53 | Katty | needs a few changes |
| 17:51.54 | angryuser | executed* |
| 17:51.56 | Katty | but basically accomplishes what i want |
| 17:53.15 | zacwolfe | jameswf: yeah there were some minor graphical glitches in the Linux version of the designer but that was around last Christmas and I know there were some fixes that came out for the graphical lib we're using. I'll try making a linux build again and see how it looks. |
| 17:53.15 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 17:54.10 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 17:54.28 | nny_2 | can someone explain the difference between a -48 vdc power supply and a 48vdc power supply? |
| 17:55.22 | nny_2 | I am in a greta bind here. Had someone sell me a cage and EC and now that I have them all I realize this setup is absolutely not mean for 1 card, and the person I was talking to is not available |
| 17:55.35 | angryuser | nny_2 : alternative or continiuos power supply (not sure if it sound like this in english ;) |
| 17:55.42 | nny_2 | AC |
| 17:57.02 | Katty | mmm, variable power signals! |
| 17:57.02 | nny_2 | Says INPUT AC 100-240v 2.0A OUTPUT 48v 1.3A The card is reported to need "-48vdc power" |
| 17:57.25 | nny_2 | Nearly all telephone central offices, where these cards were meant to live, are powered from -48. The shelf, ground, and the positive terminal of the power supply are all connected together. The "hot" side of the supply is negative with respect to ground. Why? Because any leakage current on a telephone line will cause metal to build up on something that is negatively charged with respect to its surroundings (earth) and will cause the metal to dissolve into |
| 17:57.28 | nny_2 | from the wiki |
| 17:57.50 | [TK]D-Fender | Katty: "h" was what you asked for. You said when the caller hangs up. If you want the caller to continue when the CALLEE hangs up, then thats "g". And if you don't do "g", BOTH will end up in "h" anyways |
| 17:57.59 | nny_2 | I was told to order this power supply, but not clear if it is the proper one or not, as it says NOTHING about -48vdc |
| 17:58.20 | Katty | [TK]D-Fender: just be happy it works (= |
| 17:58.26 | Katty | [TK]D-Fender: stop making a fuss over the little things in life. |
| 17:59.14 | Katty | nny_2: it's AC |
| 17:59.53 | nny_2 | well crap this power supply I just blew cash on says output DC |
| 17:59.58 | Katty | nny_2: you'd need something to convert the dual AC voltage into DC negative and positive |
| 18:00.09 | nny_2 | I swear screw echo |
| 18:00.12 | nny_2 | this is such bullshit |
| 18:00.15 | Katty | like a diode bridge or something |
| 18:00.19 | hardwire | nny_2: :( |
| 18:00.28 | scooby2 | Wierd issue with 11.4.21.2. Occasionally an Agent will login, get one call, then Asterisk kicks the agent out. No mention of anything in warnings or verbose. |
| 18:00.28 | nny_2 | fuck that, I just want my god damn 1000 dollar echo canceler to work |
| 18:00.30 | Katty | which would make sure the ouput would remain the same directions, regardless of the input polarity |
| 18:00.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:00.44 | nny_2 | <-- pissed, brb |
| 18:00.47 | Katty | i'm not sure your average power adapter is that smart ^_- |
| 18:00.55 | Katty | but what do i know (= |
| 18:00.56 | *** part/#asterisk killfill (n=killfill@200.63.96.244) |
| 18:01.02 | Katty | i just like puppehs |
| 18:05.35 | *** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com) |
| 18:06.42 | bipolar | Does anyone know of a way to track all CDR entries for a call from ring to hangup in mysql? |
| 18:07.08 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 18:07.17 | bipolar | wonders if that question makes any sense |
| 18:07.57 | nny_2 | bipolar: http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 18:08.42 | v4mp | does this look right exten => 84455707,WaitExten() as i dont see any viable option i would need there from the cli |
| 18:08.48 | bipolar | nny_2: yes, it's installed and running. Afaik that page doesn't cover what I'm asking about. |
| 18:09.10 | bipolar | maybe there's something in the cdr config I'm missing |
| 18:10.05 | nny_2 | bipolar: it seems to track all of them in our implementation, not sure what you are missing |
| 18:10.44 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:10.52 | bipolar | nny_2: the issue is that if a call comes in and gets passed around between extentions each transfer makes a new entry in the cdr table. |
| 18:10.54 | nny_2 | Katty: thanks for trying to help, i dunno. I feel like I am building a bridge to get over a puddle right now |
| 18:11.03 | nny_2 | bipolar: ahh |
| 18:11.09 | nny_2 | bipolar: yeah mine does that too |
| 18:11.20 | bipolar | k. I'm not insane then :P |
| 18:11.25 | *** join/#asterisk XnOSX (n=XnOSX@212.145.55.118) |
| 18:11.41 | Katty | nny_2: i know that feeling all too well |
| 18:11.48 | Katty | nny_2: sometimes i feel like a rag that just cleans up messes |
| 18:12.06 | jameswf | Neat: http://ocw.mit.edu/OcwWeb/Electrical-Engineering-and-Computer-Science |
| 18:12.34 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:12.40 | nny_2 | jameswf: i saw that, looks cool. Just need to find the time |
| 18:13.30 | coppice | jameswf: that stuff has been there several years |
| 18:13.36 | nny_2 | Katty: yup. Installed a system. Dialplan and underlying is solid, but the T1 echo has been a nightmare |
| 18:14.13 | jameswf | is slow... |
| 18:14.56 | jameswf | always ammused by the frantic subject " I got hacked" |
| 18:15.02 | coppice | there's some really good stuff there for comms, DSP and other fun topics |
| 18:15.03 | jameswf | :)) dont do that |
| 18:16.50 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 18:17.20 | FruitBasket | uhm... help... when I perform a transfer, ${CDR(accountcode)} ends up blank. How can I fix that? I'd prefer the account code of the incoming call be preserved through a transfer... |
| 18:18.01 | Qwell | JerJer: ping |
| 18:18.04 | bipolar | the CDR webscripts are able to track the CID as it gets passed around, but I don't know to do it in mysql. time to dig into the CDR webscript code. |
| 18:18.55 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 18:21.30 | *** join/#asterisk citywok (n=andrewp@65.249.42.130) |
| 18:21.43 | jjshoe | Qwell: No Route to Host |
| 18:21.53 | Qwell | jjshoe: see link on efnet |
| 18:21.58 | Qwell | they responded |
| 18:22.11 | citywok | i have a question about sip bandwidth. i'm using ulaw (g711), and each call is generating 160kbps of bandwidth. 10 calls / 1.5mbit. why is it so high? i thought g711 was supposed to be a bit less than half of that? |
| 18:22.24 | Qwell | citywok: each direction |
| 18:23.05 | citywok | oh really. if the provider (voicepulse) doesnt do g729, is that my bst option then? |
| 18:23.22 | [TK]D-Fender | citywok: 80kbps per DIRECTION |
| 18:23.38 | citywok | yea, i understood that when qwell said it |
| 18:23.58 | [TK]D-Fender | citywok: Yeah, I was typing and then distracted as he answered... |
| 18:24.01 | jjshoe | Qwell everything but the last response was good too, shocking. |
| 18:24.13 | citywok | oh haha, no worries then :-) |
| 18:24.13 | Qwell | good? it's terrible. |
| 18:24.49 | nny_2 | citywok: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 18:24.58 | Qwell | jjshoe: This means that hundreds (thousands?) of Asterisk users are in violation of Ciscos licensing terms |
| 18:25.10 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:25.17 | Qwell | and finding that out, was exactly the point of this whole ordeal |
| 18:25.51 | citywok | thank you nny_2 i was trying to compare bandwidth usage looking up codecs on voip-info |
| 18:27.24 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 18:27.24 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 18:27.39 | d33p5n | that's interesting, i ought to be able to pump 8 concurrent calls over a 1.5mbps pipe with room to spare |
| 18:27.53 | citywok | 10 was hitting 100% or so for me d33p5n |
| 18:28.09 | *** join/#asterisk nicox (n=nicox@213-33-20-72.adsl.highway.telekom.at) |
| 18:28.14 | FruitBasket | Does anyone know how I can make Asterisk keep the accountcode when I do a transfer?... |
| 18:28.20 | d33p5n | ADSL for me means do your max calculations based on 70-80% of your peak |
| 18:28.25 | d33p5n | to be safe |
| 18:29.02 | d33p5n | FruitBasket can you assign it to a var and then pick up that var after the transfer? |
| 18:29.04 | citywok | i want to know per 1.5mbit, testing with a solitary T1, soon to be a full ds3 |
| 18:29.34 | [TK]D-Fender | d33p5n: Really? Using what codec? What's your upstream? |
| 18:29.36 | jjshoe | Qwell what's your legal think? |
| 18:29.42 | FruitBasket | deep: dunno, but that was the next step.. even if I can, though, it's still problematic as I can't guarantee where they will be going. |
| 18:30.01 | Qwell | jjshoe: we don't use Cisco phones. It wouldn't affect us :p |
| 18:30.27 | [TK]D-Fender | citywok: IAX2 trunking can save you 16-20 kbps / call from the 2nd onwards. |
| 18:30.27 | citywok | so i set the codec to be GSM, i see it set channel to write format gsm, and if i do sip show channel it says GSM for a few seocnds, then switches to 0x4 ulaw, any idea why? |
| 18:30.33 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:30.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:30.49 | jjshoe | Qwell ah |
| 18:32.28 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:32.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:32.36 | lmadsen | ~ciscolicense |
| 18:33.07 | Qwell | jbot: ciscolicense is unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. |
| 18:33.08 | jbot | okay, Qwell |
| 18:33.09 | coppice | Qwell: don't cisco ensure that all their licencing is so arcane every user is in breach of it? |
| 18:33.16 | Qwell | coppice: http://www.ntbox.com/cisco-openletter.html |
| 18:33.20 | Qwell | coppice: I got official answers. |
| 18:33.38 | Qwell | jbot: ciscolicense is also see http://www.ntbox.com/cisco-openletter.html |
| 18:33.39 | jbot | Qwell: okay |
| 18:33.46 | lmadsen | beat me too it :) |
| 18:33.58 | lmadsen | [TK]D-Fender: ^^^ remember that for Cisco license questions :) |
| 18:34.03 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 18:34.08 | lmadsen | ~ciscolicense |
| 18:34.09 | jbot | ciscolicense is probably unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html |
| 18:34.18 | *** join/#asterisk hardhatpat (n=hardhatp@c-67-189-23-114.hsd1.or.comcast.net) |
| 18:34.24 | Qwell | Jeremy wanted to blog about it last week.. I said to hold off. I emailed him to open the flood gates. |
| 18:34.33 | lmadsen | go JerJer! |
| 18:34.44 | [TK]D-Fender | lmadsen: I would, but thats a pipe-dream appended to a cheap shot :) |
| 18:35.00 | hardhatpat | i am trying to use the web-based voicemail client and i am getting this error: Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 152. |
| 18:35.20 | Qwell | hardhatpat: Do you have a (readable) /etc/asterisk/voicemail.conf? |
| 18:35.27 | hardhatpat | Qwell: yes |
| 18:35.37 | jjshoe | hardhatpat readable by your apache user? |
| 18:35.39 | hardhatpat | i even tried 777'ing it to see if that would fix the problem |
| 18:35.50 | hardhatpat | i owned it to the apache group |
| 18:36.03 | jjshoe | hardhatpat put this vmail.cgi on pastebin.com or something |
| 18:36.13 | jjshoe | so we can see wtf line 152 is |
| 18:36.25 | coppice | Qwell: that someone needs to ask these questions would in a sane world make cisco's licencing invalid |
| 18:36.37 | Qwell | coppice: *shrug* |
| 18:37.07 | Qwell | sadly, I've been trying to figure that out for like 3 years |
| 18:37.12 | jjshoe | Qwell invalidated all opporutnities for any cisco usage for even asking questions probably :P |
| 18:37.31 | hardhatpat | http://pastebin.ca/1216048 |
| 18:37.32 | Qwell | You can see why it took them nearly 2 weeks to respond to me |
| 18:37.50 | FruitBasket | ... no, I can't set a channel variable to store the account code. Chan vars don't appear to persist through a transfer. |
| 18:37.53 | [TK]D-Fender | jjshoe: I'm sure it was duly filed in the "Ask me if I give a shit" drawer ;) |
| 18:38.05 | coppice | Qwell: I would think that in many countries they are claiming control which is illegal |
| 18:38.16 | [TK]D-Fender | Qwell: And do we get to see the response? |
| 18:38.19 | FruitBasket | so I'm getting desparate. How can I make an account code persist through a transfer. |
| 18:38.19 | jjshoe | hardhatpat change open(VMAIL, "<$filename") || die("Bleh, no $filename"); to open(VMAIL, "<$filename") || die("Bleh, no $filename: " $! . "|" . $@); |
| 18:38.22 | Qwell | ~ciscolicensing |
| 18:38.25 | jjshoe | hardhatpat and give us the error again |
| 18:38.33 | Qwell | ~ciscolicense |
| 18:38.33 | jbot | somebody said ciscolicense was unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html |
| 18:38.38 | [TK]D-Fender | Qwell: And a crying shame you weren't able to get my advice prior... |
| 18:38.38 | Qwell | I can't believe I already forgot the entry |
| 18:38.50 | jjshoe | Qwell shows how much you care ;) |
| 18:38.51 | Qwell | [TK]D-Fender: bottom of the link |
| 18:39.37 | hardhatpat | jjshoe: syntax error at /var/www/cgi-bin/vmail.cgi line 152, near ""Bleh, no $filename: " $! " |
| 18:39.37 | hardhatpat | Execution of /var/www/cgi-bin/vmail.cgi aborted due to compilation errors. |
| 18:39.57 | jjshoe | hardhatpat add, missed a . |
| 18:40.09 | jjshoe | hardhatpat open(VMAIL, "<$filename") || die("Bleh, no $filename: " . $! . "|" . $@); |
| 18:40.23 | hardhatpat | jjshoe: yeah i just caught it |
| 18:40.27 | coppice | even with the current consumer hostile attitude of most governments, this kind of "we sell you nothing. everything is licenced at our whim" attitude is surely heading for trouble |
| 18:40.34 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 18:40.50 | hardhatpat | it says permission denied ... but /etc/asterisk/voicemail.conf is 777 |
| 18:41.22 | [TK]D-Fender | Qwell: Asterisk : CAN HAZ FUR-WARE & SUPER-T? Cisco : NO KTHXBIBI |
| 18:41.22 | jjshoe | hardhatpat :) |
| 18:41.33 | coppice | why exactly would anyone *want* to use a cisco phone these days? |
| 18:41.35 | hardhatpat | jjshoe: what? |
| 18:41.39 | jjshoe | hardhatpat I presume it's a parent directory permission issue |
| 18:42.22 | jjshoe | really stupid cgi, since it doesn't check for such a common issue and give hints on how to correct, or even look for where the issue is |
| 18:42.46 | jaytee | "Dear customer (he who we sodomize at our whim), we, the executives of Cisco (insert list 'o dickwads here) have determined that you have only the rights we deem you worthy of and only after excessive price gouging has commenced." |
| 18:43.04 | jjshoe | jaytee and to that, we drink. |
| 18:43.09 | hardhatpat | drwxrwx--T ... waht does that T mean? |
| 18:44.32 | hardhatpat | ok im an idiot who cant type |
| 18:44.38 | jjshoe | hardhatpat ? |
| 18:45.01 | hardhatpat | hold on ... i changed the permissions of /etc/apache instead of /etc/asterisk |
| 18:45.06 | jjshoe | roffle |
| 18:45.14 | jjshoe | 777? swift :P |
| 18:45.19 | jjshoe | what's the url to your website? :) |
| 18:45.31 | hardhatpat | jjshoe: just chown, not chmod :) |
| 18:45.32 | *** join/#asterisk Specialist1 (n=me@119.160.105.172) |
| 18:45.38 | Specialist1 | hi everyone |
| 18:45.46 | jaytee | hi |
| 18:45.48 | FruitBasket | anyone... is there a way to persist channel variables or account code through a transfer? I don't really want to write the data to a database for each ID for each call, that seems slow and.. bad.. especially since they never go away. |
| 18:46.32 | jaytee | FruitBasket, read channelvariables.txt in the docs folder of your asterisk source |
| 18:46.53 | hardhatpat | none of my messages are showing up, however |
| 18:46.55 | Specialist1 | anyone using MERA here? |
| 18:47.12 | *** join/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 18:48.30 | FruitBasket | jaytee: thanks, will read. |
| 18:48.38 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:49.05 | jaytee | FruitBasket, adding a single or double underscore to the beginning of the variable name allows inheritance to another channel |
| 18:50.40 | FruitBasket | oooohhhhh... does that apply to the account code, by chance? |
| 18:51.01 | FruitBasket | actually, if someone calls in and gets transferred somewhere else.. does the billing period end at the transfer, or does it continue? |
| 18:51.14 | FruitBasket | and if it continues... why the hell would the account code ever go away?!? |
| 18:54.57 | citywok | whatever protocol i set sip to use (gsm, g726), it uses for 10 seconds in the call, and then falls back to ulaw. is it dtmf or something causing it to fall back? why would it do this? |
| 18:56.42 | CrazyTux | [TK]D-Fender: quick question, say I'm using Asterisk Manager, and Action: Originate, to send calls, about 30/per second, I'll stop sending the asterisk manager calls, but it'll still have calls to send for a while, is there a way to view/clear this asterisk manager queue, and see which calls have been "ringing" / "called", and which calls are still "queued" ? |
| 19:02.20 | smth | Something weird about inband dtmf. it only works with 'backgroud' . 'waitexten' or 'read ' did recognized the inband digits on the sip inconming call. I use 1.4.20.1 . and set ulaw codec in sip channel. Can anyone help figure out? |
| 19:02.20 | jaytee | holy shit! netsplit! |
| 19:02.20 | jaytee | "Cap'n!!!! The engines!!!!!" |
| 19:02.20 | smth | sorry did not recognize |
| 19:02.38 | *** join/#asterisk walhala (n=walhala@resonix.fr) |
| 19:02.38 | [TK]D-Fender | *b00m* |
| 19:03.34 | keith4 | wow... my other nick got split off. that hasn't happened before |
| 19:04.15 | v4mp | can anyone see what i' |
| 19:04.24 | keith4 | no |
| 19:05.49 | v4mp | ve done wrong here its giving SIP Response message for INCOMING dialog BYE arrived on waitexten this is my conf |
| 19:05.49 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk nicox (n=nicox@213-33-20-72.adsl.highway.telekom.at) |
| 19:05.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk lanning (n=lanning@66.151.128.195) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk chigital (n=chigital@tmo-122-1.customers.d1-online.com) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk grantm (n=grant@68.142.138.4) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 19:05.49 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-178-166.phil.east.verizon.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk DJ_HaMsTa (i=DJ_HaMsT@c-76-124-82-218.hsd1.nj.comcast.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk diverscuba (n=diverscu@d-72-9-4-157.cpe.metrocast.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk seanmh (i=seanmh@216.31.101.24) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk c4t3l (n=root@74.95.210.124) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk awannabe (n=brad@ip24-251-147-29.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk phpboy (n=shane@196.211.1.45) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk stony (n=oloch@serverbu.de) [NETSPLIT VICTIM] |
| 19:05.50 | *** join/#asterisk steliosk (n=Stelios@91.140.121.44) |
| 19:05.50 | *** join/#asterisk mav3rick (n=t0m@trankil.biz) |
| 19:05.51 | *** join/#asterisk [netman] (n=netman@199.Red-83-38-222.dynamicIP.rima-tde.net) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk MrNaz (n=naz@ppp59-167-150-186.lns4.mel6.internode.on.net) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.ifa.hawaii.edu) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk JT_ (n=j@unaffiliated/jt) |
| 19:05.51 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:05.51 | *** join/#asterisk linagee (n=jalton@about/linux/staff/linagee) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 19:05.51 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk beek (n=klinebl@65.211.106.242) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk keith4_ (n=kbe2@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) [NETSPLIT VICTIM] |
| 19:05.51 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 19:05.51 | *** join/#asterisk polerin (n=erin@c-71-228-216-37.hsd1.tn.comcast.net) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk infinity1 (i=brendon@saleen.netcal.com) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk gongoputch (n=kseel@74.95.184.161) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk brodiem (n=brodiem@67.18.114.226) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk badcfe (i=christia@morra.di.er.kjip.no) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk km- (n=pgrace@66.92.234.252) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 19:05.52 | *** join/#asterisk savaticus (n=savaticu@sta-206-168-231-41.rockynet.com) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk krdian (i=krdian@killer.radom.net) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk jblack (n=jblack@pool-71-181-244-67.sctnpa.east.verizon.net) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM] |
| 19:05.52 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) [NETSPLIT VICTIM] |
| 19:05.53 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 19:05.53 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM] |
| 19:05.53 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) [NETSPLIT VICTIM] |
| 19:05.53 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) [NETSPLIT VICTIM] |
| 19:05.53 | v4mp | http://pastebin.com/d60b95b34 |
| 19:06.19 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 19:06.19 | [TK]D-Fender | CrazyTux: Go look at your active channels |
| 19:07.09 | smth | any idea about inband digits working? |
| 19:08.20 | *** join/#asterisk Shotygun (n=thorn@82.166.246.66) |
| 19:09.06 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 19:09.54 | v4mp | [TK]D-Fender, can u see whats wrong with my config? |
| 19:10.06 | CrazyTux | [TK]D-Fender: they are not all active |
| 19:10.13 | CrazyTux | [TK]D-Fender: is that what you wanted to point out? |
| 19:10.41 | [TK]D-Fender | CrazyTux: If you want to know if you are clear to feed more, then go look at how many are in progress |
| 19:10.58 | [TK]D-Fender | v4mp: Who cares wha your config looks like when I can't see the CALL. |
| 19:11.06 | *** part/#asterisk spaetzle (n=spaetzle@gw02.easterngraphics.com) |
| 19:11.42 | Alan_Hicks | Howdy folks. I'm troubleshooting a terribly odd echo problem that is only affecting a subset of my phones. Basically, when a caller dials in, asterisk rings the receptionists. When one of them answers, there is a terrible echo on the line until the caller answers. |
| 19:11.44 | v4mp | [TK]D-Fender, i did say further up that it gives this |
| 19:11.44 | v4mp | SIP Response message for INCOMING dialog BYE arrived on waitexten |
| 19:12.01 | v4mp | everything above that works thats when it hangs up |
| 19:12.06 | v4mp | that with debug on |
| 19:12.20 | jjshoe | hardhatpat get your shit working? |
| 19:12.27 | Alan_Hicks | These users are on Polycom Soundpoint IP 650 phones. If instead they use a Polycom 320 phone, there is no echo. Is there some setting in the polycom configuration that might cause this? |
| 19:12.31 | [TK]D-Fender | v4mp: And do i feel like guessing where I should be looking in the call? Never waste time with anything less than complete CLI output. |
| 19:12.37 | hardhatpat | jjshoe: halfway |
| 19:12.42 | Alan_Hicks | I've ruled out cabling as a culprit entirely. |
| 19:12.42 | hardhatpat | no errors ... but no voicemails show up |
| 19:13.07 | [TK]D-Fender | Alan_Hicks: Sidetone on the 650. there are base gains you can adjust and IIRC the stock firmware values weren't great and caused this. |
| 19:13.08 | jjshoe | hardhatpat hot |
| 19:13.13 | v4mp | [TK]D-Fender, i dont see how it would make a difference as i told u where it gets to on config :/ |
| 19:13.16 | v4mp | but ok 1 sec |
| 19:13.19 | hardhatpat | jjshoe: any ideas? |
| 19:13.21 | *** join/#asterisk Ironhand (i=x@xyx.nl) |
| 19:13.26 | jjshoe | hardhatpat link to the pastebin again? |
| 19:13.36 | hardhatpat | http://pastebin.ca/1216048 |
| 19:13.39 | Alan_Hicks | [TK]D-Fender: Thank you. Do you have a link to a discussion of this issue or can tell me what to google/grep for please? |
| 19:14.07 | [TK]D-Fender | Alan_Hicks: sip.cfg |
| 19:14.09 | Alan_Hicks | Or should I try to contact Polycom? |
| 19:14.21 | [TK]D-Fender | Alan_Hicks: Just fix your provisioning |
| 19:14.47 | Alan_Hicks | [TK]D-Fender: Thank you. Do you know what values might need to be changed? |
| 19:15.02 | Alan_Hicks | I'll grep the polycom manual in a moment for sidetone. |
| 19:15.10 | jjshoe | hardhatpat did you set the context? |
| 19:15.15 | hardhatpat | jjshoe: i can log in with my mailbox and password, but NO vmails show up whatsoever |
| 19:15.24 | hardhatpat | jjshoe: what context? |
| 19:15.31 | [TK]D-Fender | Alan_Hicks: Go look a bit. I'm sure you'll gind it. |
| 19:15.38 | Alan_Hicks | Thanks. |
| 19:15.40 | jjshoe | hardhatpat perhaps you need to read the cgi? |
| 19:16.02 | *** part/#asterisk Ironhand (i=x@xyx.nl) |
| 19:16.27 | jjshoe | hardhatpat perhaps at least to line $24 |
| 19:16.32 | hardhatpat | jjshoe: im reading it |
| 19:16.35 | hardhatpat | what do i set it as? |
| 19:16.46 | jjshoe | hardhatpat what's the context for your voicemail? |
| 19:16.50 | v4mp | [TK]D-Fender, this is full debug of call http://pastebin.com/d3d0ab5f5 |
| 19:17.12 | jjshoe | hardhatpat ls /var/spool/asterisk/voicemail/ |
| 19:17.36 | hardhatpat | oh |
| 19:17.43 | hardhatpat | default |
| 19:18.00 | jjshoe | so add that to the cgi |
| 19:18.10 | [TK]D-Fender | v4mp: Go look at your dialplan. The error is quite obvious. |
| 19:18.29 | v4mp | needs a goto ? |
| 19:18.44 | [TK]D-Fender | v4mp: No. |
| 19:18.44 | hardhatpat | do i add just 'default'? or the entire path? |
| 19:18.52 | smth | [TK]D-Fender, something weird about inband dtmf. it only works with 'backgroud' . 'waitexten' or 'read ' did not recognize the inband digits on the sip incoming call. I use 1.4.20.1 . and set ulaw codec in sip channel. any idea? |
| 19:19.20 | v4mp | then i have no clue because the lines below it are ; out |
| 19:19.28 | [TK]D-Fender | v4mp>ve done wrong here its giving SIP Response message for INCOMING dialog BYE arrived on waitexten this is my conf <--- Wrong |
| 19:19.43 | v4mp | hmm |
| 19:19.47 | [TK]D-Fender | v4mp: waitexten isn't even CALLED. Pay attention to your OUTPUT |
| 19:19.54 | jjshoe | hardhatpat you add the context. |
| 19:20.11 | [TK]D-Fender | v4mp: So go stare at your dialplan untill you see the rather clear mistake. |
| 19:20.53 | [TK]D-Fender | smth: Aside from "read" "waitexten", etc, where should * care about dtmf? |
| 19:21.11 | [TK]D-Fender | smth: Do feel free to show me an actual problem. |
| 19:21.30 | hardhatpat | jjshoe: yeah i made it 'default' |
| 19:21.52 | Katty | anyone work with bandwidth.com? |
| 19:21.55 | v4mp | would i need AGI or is that just an extra ? |
| 19:22.07 | [TK]D-Fender | v4mp: No. |
| 19:22.15 | v4mp | ok |
| 19:22.39 | [TK]D-Fender | v4mp: There is an extremely clear reason why Waitexten isn't being called.... keep staring at that line and the one that preceeds it. |
| 19:23.00 | v4mp | i noticed [TK]D-Fender which is why im on next part :) |
| 19:25.35 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:26.28 | *** join/#asterisk l337ingDisorder (n=chatzill@66.183.159.54) |
| 19:27.24 | bipolar | Does anyone know any way to put a value in the userfield column of the CDR table to track a call on a zap channel from begining to end? Is there another way to do that? |
| 19:28.12 | l337ingDisorder | hey folks, just installed Asterisk and gastman on a fresh Ubuntu 8.10 install. When I run gastman I can connect to localhost but it asks for a login and password... I wasn't prompted to enter an admin login/password to use during the install process and I can't seem to find the default asterisk login info anywhere.. can anyone tell me how to log in? |
| 19:28.16 | codefreeze-lap | bipolar: what do you mean, "track a call on a zap channel from begining to end" ? |
| 19:28.35 | v4mp | ok does anyone see a problem with this because i dont but cli says its meant to have '=' |
| 19:29.18 | v4mp | exten #,1,Playback(sai-thanks) |
| 19:29.23 | bipolar | codefreeze-lap: if someone gets calls in, the CDR system tracks the call in multiple segments. If the call gets transfered I get a new record in the CDR but no way to tie that recored to the previous segment. |
| 19:29.36 | bipolar | at least I havn't found a way... |
| 19:29.44 | codefreeze-lap | l337ingDisorder: check you manager.conf file and set yourself up an account there. |
| 19:29.51 | scooby2 | v4mp: your missing an important part |
| 19:30.11 | scooby2 | exten => invalid-format,1,Playback(prompts/dial-first) |
| 19:31.06 | v4mp | i dont get it o_O |
| 19:32.57 | bipolar | I need a way to track an incoming or outgoing call and see every part of the converstation, no matter how many times the call was transfered. doesn't seem to be a way with the existing cdr data. :\ |
| 19:33.03 | codefreeze-lap | bipolar: Not yet. If you look at bmd's work on using CEL, you'll see that we keep a channel field for an id, that is viral when two channels interact. All the legs of call xfers can be tied together this way. |
| 19:33.16 | scooby2 | bipolar: track using the zap channel |
| 19:33.22 | smth | [TK]D-Fender, I have a 'background' called before waitexten in dialplan, the inband digit could be known during 'background' a sound file. but if sending inband digits within the period 'waitexten(5)' when the sound file finished playing ,they were not being recognized until time out. if same case used on rfc2833 or info . both work fine. |
| 19:33.31 | Kobaz | how would i make a custom feature code in [applicationmap] that is active both for caller and calle on the same feature code |
| 19:34.04 | bipolar | scooby2: If I can insert the zap channel into the userfield of the cdr table that would work, but i don't know how to do that. |
| 19:34.18 | bipolar | at least it would be a start. |
| 19:35.26 | [TK]D-Fender | smth: Show me a failed call with full debug... |
| 19:35.48 | bipolar | codefreeze-lap: hmmm..... |
| 19:36.31 | smth | [TK]D-Fender, where I paste them to show you the details |
| 19:36.41 | [TK]D-Fender | ~pb |
| 19:36.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:36.42 | bipolar | codefreeze-lap: So it looks like the dstchannel becomes the channel field of the recieving extention. is that correct? |
| 19:37.03 | smth | got it |
| 19:37.15 | [TK]D-Fender | v4mp: exten => 84455707,6,Background(sai-choose) ..... exten #,1,Playback(sai-thanks) |
| 19:37.22 | [TK]D-Fender | v4mp: Notice anything MISSING? |
| 19:38.06 | v4mp | i hate help sites that dont tell u much info at all about things :/ |
| 19:38.19 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 19:38.31 | [TK]D-Fender | v4mp: you meant he part where the "=>" after the word EXTENS is NECESSARY? |
| 19:38.48 | [TK]D-Fender | v4mp: You must be completely blind to how you even formatted all your other lines. |
| 19:38.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:38.54 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:39.08 | v4mp | good point :| |
| 19:39.27 | v4mp | was doing that at 5am this morning ¬_¬ lol |
| 19:39.40 | [TK]D-Fender | v4mp: What's your excuse for right now? |
| 19:40.15 | v4mp | i haven't slept much for last week or sumthin |
| 19:40.55 | [TK]D-Fender | goes off to do something productive |
| 19:42.25 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 19:46.15 | angryuser | i have 3 outgoing peers from different providers, they serve for the same destinations, i would like to develep a script 1: Change them each time the destination is called(astdb&groupcount) i do know how to do that, and 2: Verify the registry status BEFORE call so if the provider 1 is dead, callout with another one + round up them, i hope it was clear ;) |
| 19:47.29 | Kobaz | hmmmm |
| 19:47.58 | Kobaz | i can't seem to get [applicationmap] feature codes to be available to the callee |
| 19:49.47 | *** join/#asterisk gewuerzwiesel (n=gewuerzw@unaffiliated/gewuerzwiesel) |
| 19:49.49 | *** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net) |
| 19:49.51 | gewuerzwiesel | hello |
| 19:50.54 | [TK]D-Fender | angryuser: Register status is just another value in AstDB. |
| 19:50.57 | jeff_smoker | Does anyone know if a complete mid-call handoff is possible with SIP REFER? |
| 19:52.16 | [TK]D-Fender | jeff_smoker: Sounds like a basic transfer to me... |
| 19:52.21 | gewuerzwiesel | I just want to ask you some little questions to start :) what version should I use? install from ubuntu reps or better compile, and then 1.4 or 1.6? |
| 19:52.41 | nny_2 | what does chan_dahdi.c:1442 dahdi_enable_ec: Echo cancellation already on in console mean? If you had to guess at what the situation is that would cause that, what would you say? |
| 19:52.49 | [TK]D-Fender | gewuerzwiesel: 1.6 is not yet a full elease and should e notably less stable, documented, and able to be supported |
| 19:52.53 | angryuser | [TK]D-Fender : database show give me the register status of my local peers and agents not the outgoing ones ;( , maybe i need to enable some option ? |
| 19:52.57 | [TK]D-Fender | gewuerzwiesel: Your call. |
| 19:53.00 | nny_2 | I mean it's obvious, but is this the software EC reporting that hardware EC is active? |
| 19:53.07 | [TK]D-Fender | angryuser: Yes, it DOES. |
| 19:53.48 | [TK]D-Fender | angryuser: Last i checked anyways |
| 19:53.49 | gewuerzwiesel | [TK]D-Fender: ok, so if I compile it by myself, what do I need to get my fritzcard pci working? do I need zaptel? |
| 19:54.01 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:54.08 | jeff_smoker | [TK]D-Fender: I mean to say being there is a difference between a conference call and call forwarding. When you forward a call using SIP refer and then terminate the call...will the referred parties remain connected? |
| 19:54.10 | [TK]D-Fender | angryuser: For everything else there's SIPPEER |
| 19:54.23 | [TK]D-Fender | gewuerzwiesel: Can't answer BRI questiosn, sorry. |
| 19:54.26 | voxter | has anyone put up what came of astridevcon? |
| 19:54.35 | gewuerzwiesel | tkbeat: ok, no prob. thx |
| 19:54.40 | [TK]D-Fender | jeff_smoker: thats the point |
| 19:54.42 | angryuser | [TK]D-Fender : http://www.pastebin.ca/1216119 ..... |
| 19:54.45 | implicit | voxter, we are too tiredd :) |
| 19:55.02 | implicit | voxter, i'm sure some stuff will be going up soon. we had 4 groups |
| 19:55.19 | [TK]D-Fender | angryuser: I see, please refer to the function I jsut mentioned |
| 19:55.32 | implicit | media negotiation, cli stuff, new architecture for dev & web stuff, and other stuff |
| 19:56.09 | angryuser | [TK]D-Fender : nice |
| 19:56.15 | Kobaz | so, would anyone here happen to be a ninja jedi master of [applicationmap] in features.conf? |
| 19:56.46 | jeff_smoker | So is a mid-call handoff possible with SIP, in the sense that PARTY1 forwards inbound call PARTY2 to an outside line PARTY3, then terminates the call and is no longer billed by the provider...but PARTY2 and PARTY3 are now independently connected? |
| 19:56.55 | [TK]D-Fender | Kobaz: Funny... I don't see a pastebin from you anywhere... |
| 19:57.04 | voxter | implicit: man i wish i could have stayed for it. Even though my main focus is not dev, i follow it all closely |
| 19:57.21 | Kobaz | [TK]D-Fender: well it's really just two lines |
| 19:57.33 | [TK]D-Fender | Kobaz: No, it should be a lot more than that. |
| 19:57.54 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:58.23 | lowtek | Hi all. How does one set the default Polycom ring tone via the XML file? I thought it was with one of the rt.x lines but apparently not. |
| 19:58.33 | implicit | jeff_smoker, it is possible but in asterisk it is a b2bua so it can't leave the signaling stream |
| 19:58.44 | implicit | jeff_smoker: but in sip, proxies can leave both the media and signaling stream |
| 19:59.38 | Katty | so. if i wanted to Set(Email=foo), and then have h do h,1,Gotoif ( email isn't empty) |
| 19:59.41 | Katty | how would i word that? |
| 20:00.04 | Katty | the gotoif part |
| 20:00.47 | Kobaz | [TK]D-Fender: okay well, it's more than two |
| 20:00.49 | Kobaz | [TK]D-Fender: http://pastebin.ca/1216128 |
| 20:00.54 | Kobaz | :) |
| 20:01.19 | [netman] | $["${Email}" != ""] I think |
| 20:01.59 | [TK]D-Fender | Kobaz: Go prove that devices DTMF even works. |
| 20:02.13 | jeff_smoker | implicit: So if I [1] receive an inbound SIP stream on a softphone and then [2] I turn around and refer that stream back to my provider and then [3] the inbound call I received is now connected to the outbound line that I just connected it to and [4] I terminate the call, what would I need to do to make sure the connected parties remain connected ? |
| 20:02.16 | Katty | [netman]: okay, now how do i tell it where to go :/ |
| 20:02.36 | [TK]D-Fender | Kobaz: and I don't see the call being ANSWERED, nor do I see any debug level that indicated DTMF being detected |
| 20:02.42 | lowtek | Hey [TK]D-Fender: Do you know which XML setting sets the default ring-type for Polycom phones? I tried the <DEFAULT rt.1 ...> but that's apparently only used for alertInfo ... |
| 20:02.54 | Katty | GotoIf($["${Email}" != ""],s,1)? |
| 20:02.54 | implicit | jeff_smoker: REFER is more difficult |
| 20:03.02 | implicit | cause it has to be processed by a UA |
| 20:03.02 | [TK]D-Fender | lowGo override it, and look at the overrides. |
| 20:03.05 | implicit | like an endpoint |
| 20:03.07 | implicit | or a B2BUA |
| 20:03.09 | lowtek | Thanks! |
| 20:03.10 | seanbright | Katty: core show application GotoIf |
| 20:03.25 | jeff_smoker | implicit: what method would you use then, other than refer? |
| 20:03.27 | implicit | so it is still two dialogs |
| 20:03.29 | implicit | in SIP |
| 20:03.33 | implicit | even if the audio is connected |
| 20:04.01 | implicit | jeff_smoker: no method other than refer can be used after 200OK is received for initial INVITE |
| 20:04.13 | Katty | seanbright: i already did that. it was not helpful. |
| 20:04.16 | implicit | you can only send 3xx replies before dialog is established |
| 20:04.22 | Katty | seanbright: the wiki page was more helpful, but doesn't show an extension jump anywhere |
| 20:04.24 | seanbright | Katty: how was it not helpful? |
| 20:04.36 | seanbright | Katty: it's helpful if you read it. |
| 20:04.42 | seanbright | Katty: consume words and such. |
| 20:04.48 | implicit | you can do after 180/183, but not after 200 |
| 20:04.51 | Katty | seanbright: i got 90% of what i needed, yeah |
| 20:04.57 | implicit | after 200 REFER needs to go to the endpoint |
| 20:05.00 | implicit | asterisk is an endpoint though |
| 20:05.02 | implicit | since it's a b2bua |
| 20:05.08 | seanbright | Katty: GotoIf(condition?[labeliftrue]:[labeliffalse]) |
| 20:05.17 | angryuser | [TK]D-Fender : i see if i do SIPPEER(peer|status) where '- status Status (if qualify=yes)' but it's the qualify status ? not the registration status? (i got qualify set to no) |
| 20:05.46 | jeff_smoker | implicit: Would you argue that you can change endpoints or that you cannot change endpoints? |
| 20:05.49 | [TK]D-Fender | angryuser: look at "sip show peer [peer]" and see if you're inspired to look at something else. |
| 20:05.58 | implicit | ok |
| 20:06.03 | implicit | lets use more accurate terminology |
| 20:06.03 | angryuser | or maybe i need to check EXISTS(SIPPEER(peer|regexten) ? |
| 20:06.05 | [TK]D-Fender | jeff_smoker: "core show application transfer" |
| 20:06.07 | seanbright | Katty: if e-mail is blank, where do you want to go? |
| 20:06.11 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-0f27561830b960e8) |
| 20:06.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:06.18 | [TK]D-Fender | angryuser: Nope. |
| 20:06.19 | implicit | jeff_smoker UAC and UAS cannot be changed after dialog is established (200OK on initial INVITE) |
| 20:06.28 | implicit | unless REFER is sent |
| 20:06.30 | implicit | by one of them |
| 20:06.32 | implicit | to the other |
| 20:06.36 | [TK]D-Fender | angryuser: that implies your provider is registering to YOU |
| 20:06.40 | implicit | and it ends up being a new dialog |
| 20:06.48 | Katty | seanbright: GotoIf($["${Email}" != ""]?foo,s,1) i think |
| 20:06.53 | implicit | before 200OK |
| 20:07.00 | implicit | you can have parallel/serial forking |
| 20:07.03 | implicit | and tons of stuff w/ early media |
| 20:07.04 | Katty | seanbright: if false, do nothing |
| 20:07.07 | implicit | or provisional replies |
| 20:07.11 | seanbright | Katty: priority 1 of extension s in the foo context? |
| 20:07.17 | implicit | and be dealing with many UAs |
| 20:07.18 | Katty | seanbright: yes. |
| 20:07.22 | seanbright | Katty: looks right to me |
| 20:07.28 | implicit | jeff_smoker: i gotta head out but send me a message |
| 20:07.37 | jeff_smoker | implicit: ok thanks |
| 20:07.39 | Katty | seanbright: i just gotta see things. |
| 20:07.52 | Katty | seanbright: the examples are good and all, but it doesn't always sink it like it hsould )= |
| 20:08.21 | seanbright | Katty: i forgive you |
| 20:09.12 | angryuser | [TK]D-Fender : i can set the qualify and check it, but is there any other way ? |
| 20:09.28 | [TK]D-Fender | jeff_smoker: the first refer goes from your endpoint to *. from there you call TRANSER. |
| 20:09.49 | [TK]D-Fender | angryuser: I've already said what else to look at |
| 20:10.07 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 20:10.28 | angryuser | cal me blind Subscriptions: Yes ? |
| 20:10.42 | Katty | seanbright: katty learn by do, not read |
| 20:10.58 | seanbright | Katty: well do more doing and less asking :) |
| 20:10.58 | [TK]D-Fender | angLook at the entire dump. I'm sure something sueful will stand out. Just go try for a while. |
| 20:11.04 | Katty | seanbright: psh. |
| 20:11.20 | seanbright | i know... it's crazy talk. |
| 20:11.22 | [TK]D-Fender | seanbright: Yeah... that'll work... |
| 20:11.25 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 20:11.30 | Katty | hugs [TK]D-Fender |
| 20:11.36 | Katty | [TK]D-Fender: i know you love me. |
| 20:12.29 | angryuser | expire ......... |
| 20:12.34 | seanbright | Katty gets special treatment cuz she's a gurl. |
| 20:12.37 | jeff_smoker | [TK]D-Fender: I'm not sure if I'm being clear about what I aim to do. I'd like to receive an INBOUND VOIP call and then use a SIP method to send that call out to an EXTERNAL PARTY, and then TERMINATE the connection that I have with my VOIP PROVIDER. I think I understand what you're saying is that asterisk can transfer or forward the call, but asterisk in this case will continue to host the... |
| 20:12.39 | jeff_smoker | ...call. I do not want to continue to host the call. |
| 20:12.40 | seanbright | s'not fair |
| 20:12.52 | Qwell | snot fair? |
| 20:12.57 | Katty | are you sure it's not because i ask politely? |
| 20:13.12 | *** join/#asterisk boch (n=fran@customer191-9.iplannetworks.net) |
| 20:13.15 | Katty | Qwell: what do you think? |
| 20:13.16 | seanbright | Katty: most definitely |
| 20:13.18 | [TK]D-Fender | jeff_smoker: I understand perfectly and have handed you the answer. |
| 20:13.23 | seanbright | Katty: it's cuz you're a chick |
| 20:13.25 | Qwell | Katty: I wasn't paying attention |
| 20:13.34 | Katty | Qwell: you have a habbit of that. |
| 20:13.40 | Qwell | of what? |
| 20:13.43 | seanbright | ohhh |
| 20:13.44 | seanbright | touche! |
| 20:13.46 | Qwell | sorry, dozed off a second there |
| 20:13.50 | jeff_smoker | [TK]D-Fender: Excellent. i love answers. i will review it exhaustively. Thank you. |
| 20:14.12 | Katty | Qwell: seanbright here thinks that people help me and tolerate my questions because i'm a female. |
| 20:14.12 | boch | could you recommendme a good iax client for linux/gtk ? |
| 20:14.16 | Katty | Qwell: what's your viewpoint on this? |
| 20:14.25 | Qwell | Katty: seanbright is a wise person |
| 20:14.30 | Alan_Hicks | [TK]D-Fender: Ok. I'm stuck. I'm not sure that I'm even modifying the right setting. I've raised voice.handset.s |
| 20:14.34 | Alan_Hicks | idetone.adjust.IP_650 |
| 20:14.35 | Qwell | but, no, you've been around forever :P |
| 20:14.44 | seanbright | Katty: tolerated is a strong word |
| 20:14.45 | Alan_Hicks | Oops... |
| 20:15.03 | Katty | forever is a strong word, if you ask me. |
| 20:15.06 | seanbright | Katty: anyone else asking how GotoIf worked would get the following response from you-know-who... |
| 20:15.08 | Katty | let's go with Years. |
| 20:15.11 | seanbright | Katty: go READ the DOCUMENTATION. |
| 20:15.32 | Alan_Hicks | [TK]D-Fender: Ok. I'm stuck. I'm not sure that I'm even modifying the right setting. I've raised voice.handset.sidetone.adjust.IP_650 from "-3" to as high as "9" with no perceivable change. Should I be looking elsewhere to solve this echo problem? |
| 20:15.33 | Katty | Qwell: i think seanbright might have a point :< |
| 20:15.35 | seanbright | am i right or am i right? or am i right? |
| 20:15.47 | *** join/#asterisk shaw22dog (n=shaw@pacman.oaklandcorp.com) |
| 20:15.48 | seanbright | Katty <3 |
| 20:15.59 | Katty | my nagging persistance pays off |
| 20:16.04 | shaw22dog | Hello. |
| 20:16.07 | gewuerzwiesel | do I need libpri and zaptel? there are tutorials compiling just asterisk and asterisk-sounds? |
| 20:16.07 | [TK]D-Fender | Alan_Hicks: base gains as well. |
| 20:16.11 | Katty | answer me, or i shall bug you every 5 minutes for the next YEAR! |
| 20:16.41 | Katty | seanbright: it also has a downside. |
| 20:16.49 | Alan_Hicks | [TK]D-Fender: voice.gain.tx.analog.chassis.IP_650 correct? |
| 20:16.52 | *** join/#asterisk mltlnx (n=mltlnx@32.136.47.207) |
| 20:16.54 | Katty | seanbright: when you mentioned i was a female, i instantly got 5 /queries from random people :/ |
| 20:17.13 | [TK]D-Fender | ASL? |
| 20:17.15 | gewuerzwiesel | Katty: lol :) |
| 20:17.20 | Katty | [TK]D-Fender: close. |
| 20:17.23 | shaw22dog | I'm running Asterisk 1.2, with a Sangoma card. When I park calls, and then pick them back up, I get an echo on my side for approx. 20. Although there was no echo on the initial connection, any ideas on what is causing this? |
| 20:17.30 | seanbright | Katty: my bad. |
| 20:17.37 | Katty | seanbright: FOR SHAME! |
| 20:17.39 | shaw22dog | *approx 20 sec. |
| 20:17.51 | Alan_Hicks | ASL? I'm afraid I don't know what that means. |
| 20:17.54 | seanbright | Katty: /nick BigManTypePerson |
| 20:17.57 | Katty | seanbright: you may be right, i don't know. never thought about it before. |
| 20:18.04 | *** part/#asterisk nny_2 (n=scott@64.203.244.146) |
| 20:18.12 | seanbright | Katty: i'm just giving you a hard time |
| 20:18.14 | Alan_Hicks | Oh wait.... Age, Sex, location. I'm a dumb-ass. |
| 20:18.22 | Katty | seanbright: i've just always tried to ask my questions politely and wait patiently for an answer. and let the person know i appreicate their help. |
| 20:18.30 | Katty | seanbright: i think it's worked okay for the last 5 years or so (= |
| 20:18.48 | Katty | seanbright: SO BUZZ OFF ;> |
| 20:18.48 | seanbright | 5 years? |
| 20:18.54 | seanbright | and you don't know how GotoIf works? |
| 20:18.57 | seanbright | FOR SHAME! |
| 20:19.03 | Katty | of course i know how it works, kinda. |
| 20:19.15 | Katty | just nothing overly complicated ;) |
| 20:19.18 | [TK]D-Fender | seanbright: the term is "reciprocal lamprey" |
| 20:19.25 | *** join/#asterisk gabegundy (n=gabe@nat.parentlink.net) |
| 20:19.27 | Corydon76-dig | Alan_Hicks: and the answer to "ASL" for you was? |
| 20:19.34 | Katty | it'd be different if i used all these fancy features! |
| 20:19.54 | [TK]D-Fender | Katty: Next challenge... app_dial! World beware! |
| 20:20.05 | Katty | [TK]D-Fender: hush dear ;) |
| 20:20.12 | seanbright | interesting... |
| 20:20.37 | *** join/#asterisk sheri_rao (n=rao@203.99.179.244) |
| 20:20.43 | Katty | seanbright: most of the stuff i set up is pretty simple. (= |
| 20:20.50 | [TK]D-Fender | seanbright: It's all about obfuscative selection ;) |
| 20:21.17 | [TK]D-Fender | Katty needs no IFs, ANDs, or BUTs ! |
| 20:21.24 | seanbright | Katty: well from now on when someone has a GotoIf question, i'll expect you to be the goto person (ba dum dum) |
| 20:21.37 | Katty | seanbright: oh look at you mister funnypants. |
| 20:21.38 | Alan_Hicks | Corydon76-dig: 0x1C, XY, Lizella. |
| 20:22.14 | Corydon76-dig | Alan_Hicks: hawt, though still older than my BF |
| 20:22.36 | Alan_Hicks | Corydon76-dig: 1C is much younger that BF. |
| 20:22.42 | seanbright | 11100? |
| 20:22.45 | seanbright | that's old |
| 20:22.47 | Corydon76-dig | Alan_Hicks: the BF is 0x18 |
| 20:23.17 | seanbright | i am 11101 though |
| 20:23.21 | seanbright | so that makes me ancient. |
| 20:23.33 | Corydon76-dig | seanbright: mega |
| 20:23.46 | Corydon76-dig | hands seanbright his cane |
| 20:24.03 | seanbright | ewww... that is *not* a cane! |
| 20:24.03 | seanbright | heh |
| 20:24.12 | thehar | eh? |
| 20:24.22 | seanbright | wha? |
| 20:24.35 | Corydon76-dig | With your eyesight, I'm surprised you noticed :-P |
| 20:24.46 | seanbright | well played. |
| 20:25.20 | angryuser | [TK]D-Fender : well i have compared the dump from registered and not registered (same) outgoing peer, i can use only status with SIPPEER, i dont see more, see by yourself http://www.pastebin.ca/1216151 |
| 20:27.22 | [TK]D-Fender | angryuser: I suppose you could also script something to look at "sip show registry" easily enough. Messy but more than doable |
| 20:27.54 | [TK]D-Fender | ok, heading home. Later all |
| 20:30.30 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:30.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:33.00 | angryuser | well i will enable qualify for 2 days, wait&see what happens, if my provider dont cut the qualify packets to reduce traffic i will use that, lazy solution but easyer of coding all stuff with bash |
| 20:37.57 | codefreeze-lap | tomas-- you around? |
| 20:38.41 | Katty | seanbright: http://pastebin.ca/1216172 (= |
| 20:39.00 | Katty | seanbright: i will now have more spam than i can realistically handle! |
| 20:40.30 | Qwell | spams Katty |
| 20:41.48 | *** part/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:44.01 | Katty | Qwell: it all start with a company who had analog lines, who needed to know the person who was transfered to their cellphones. |
| 20:44.13 | Katty | Qwell: and then i decided to put an attachment on the email |
| 20:44.41 | Katty | Qwell: and then i got the bright idea of always recording certain extensions in our call center, to a quality control email address...that could have a few audio files listened to, and the rest dumped. |
| 20:46.20 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:47.56 | gewuerzwiesel | do I need zaptel or is it only for digium hardware? |
| 20:49.23 | *** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:49.27 | Kobaz | awww |
| 20:49.27 | seanbright | Katty: pretty sure you can leave out the ,s,1 part for both of those |
| 20:49.34 | Kobaz | the fender went byebye |
| 20:49.34 | seanbright | Katty: but not 100% |
| 20:49.50 | Katty | seanbright: it works. i'm happy with the working. |
| 20:50.03 | seanbright | Katty: working schmirking. |
| 20:50.16 | Kobaz | okay so |
| 20:50.21 | Kobaz | maybe someone else can shed some light |
| 20:50.31 | Kobaz | http://pastebin.ca/1216128 |
| 20:50.40 | *** part/#asterisk Talirk (i=434e2716@gateway/web/ajax/mibbit.com/x-241dfef0ddd48b2f) |
| 20:50.41 | Kobaz | having an issue with features.conf and custom feature codes |
| 20:51.48 | smth | seanbright,http://pastebin.com/m79b7d310 I use this test inband dtmf . it does not work on the incoming call. (ulaw/1.4.20.1) any idea? |
| 20:52.00 | seanbright | smth: no |
| 20:52.30 | seanbright | smth: i don't really help here. i just put people down, usually. |
| 20:52.30 | v4mp | how do you create the queues ? |
| 20:52.54 | Kobaz | seanbright: damn right |
| 20:52.57 | seanbright | v4mp: queues.conf |
| 20:53.00 | smth | who takes care dtmf stuff . any help? ;) |
| 20:53.31 | angryuser | Kobaz : add W option to your dial, see what happens |
| 20:53.41 | gewuerzwiesel | do I need zaptel or is it only for digium hardware? :) |
| 20:54.04 | angryuser | Kobaz : ah no |
| 20:55.13 | bijit | I am having problems registering my sip trunk. |
| 20:55.35 | *** join/#asterisk tkbeat (n=tk@p54B9474A.dip.t-dialin.net) |
| 20:55.35 | bijit | I tried alot of stuff and I can't get it to work. |
| 20:55.45 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 20:55.54 | angryuser | Kobaz : what kind of client is 2605 and 2608 ? same hardware ? |
| 20:55.58 | bijit | anyone have any idea what could it be? |
| 20:56.07 | *** join/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
| 20:56.18 | Maliuta | bijit: firewall, your settings, the visp's settings .... |
| 20:56.23 | angryuser | bijit : pastebin cli messages |
| 20:56.34 | *** join/#asterisk s0lid (n=s0lid@122.53.107.186) |
| 20:56.35 | Maliuta | bijit: the list is longer than my schwong |
| 20:56.44 | Maliuta | and that long |
| 20:57.22 | hardwire | anything better than a2b in the free software world? |
| 20:57.54 | seanbright | hardwire: apache? open office? |
| 20:58.02 | hardwire | asterisk2billing |
| 20:58.05 | seanbright | right |
| 20:58.11 | hardwire | right |
| 20:58.12 | seanbright | apache and open office are both better |
| 20:58.19 | hardwire | I refuse |
| 20:58.25 | *** part/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:58.30 | Maliuta | OpenBSD has to make that list |
| 20:58.36 | angryuser | no spectrum zx webserver is the best ;) |
| 20:58.37 | hardwire | don't forget teh hurd |
| 20:58.44 | seanbright | GNU/Hurd, thank you |
| 20:58.58 | seanbright | stallman would kill you for not using "GNU" in front of it |
| 20:59.06 | Maliuta | gcc? |
| 20:59.13 | hardwire | stallmans sword is weak |
| 20:59.13 | seanbright | indeed |
| 20:59.20 | seanbright | i think the point is |
| 20:59.23 | Maliuta | where would we be without the g |
| 20:59.36 | seanbright | when it comes to free software, many things are better than a2b |
| 20:59.38 | hardwire | Maliuta: in a world with less grubby g keys |
| 20:59.44 | seanbright | glad we got that cleared up. |
| 20:59.58 | Maliuta | hardwire: you could live in such a world? |
| 21:00.22 | *** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 21:00.25 | Maliuta | accuses hardwire of being typograph-ist |
| 21:00.53 | Qwell | Maliuta: he already lives in Alaska! can't get much worse. |
| 21:01.17 | hardwire | Qwell: you said you wouldn't tell |
| 21:01.19 | Maliuta | alsaka eh? |
| 21:01.20 | bijit | here is pastebin http://pastebin.ca/1216195 |
| 21:01.24 | Qwell | hardwire: Where's my million dollars? |
| 21:01.32 | hardwire | Qwell: where's my fuzzy peach? |
| 21:01.48 | Qwell | we threw the core away |
| 21:01.51 | Qwell | peaches have cores, right? |
| 21:01.52 | hardwire | aww |
| 21:02.00 | hardwire | they have pits |
| 21:02.00 | seanbright | Qwell: pits |
| 21:02.04 | Qwell | that then |
| 21:02.09 | Qwell | Deeewayne ate it :( |
| 21:02.15 | hardwire | Deeewayne: ! |
| 21:02.19 | Maliuta | I have a million monkeys at a million typewriters ... one bunch has some shakespeare thing, another has a functioning kernel |
| 21:02.38 | Qwell | a million typewriters per monkey, or a million typewriters total? |
| 21:02.45 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:02.47 | hardwire | how many fingers does each monkey have? |
| 21:02.53 | hardwire | do toes count? |
| 21:02.55 | Qwell | hardwire: I'm guessing 10 million |
| 21:03.14 | Kobaz | mr fender! |
| 21:03.21 | Kobaz | :) |
| 21:03.32 | bijit | [Oct 1 15:02:46] WARNING[18589]: chan_sip.c:1950 retrans_pkt: Cancelling retransmit of OPTIONs (call id 03a7769506945f6359343dd760ad358f@190.241.15.48) |
| 21:03.35 | bijit | is that bad? |
| 21:03.37 | Alan_Hicks | [TK]D-Fender: Welcome back. |
| 21:03.39 | Deeewayne | hardwire: I'm de-fuzzing the peach |
| 21:03.45 | Maliuta | Qwell: well the number of typewriters is constant, the per monkey thing changes as they split into groups to do "writers stuff" or jerk off to pr0n |
| 21:03.56 | bijit | sup Alan_Hicks |
| 21:04.05 | hardwire | Deeewayne: like how you decorn corn? |
| 21:04.07 | Alan_Hicks | [TK]D-Fender: When you told me I must change the default gains as well, did you mean this? voice.gain.tx.analog.chassis.IP_650 |
| 21:04.11 | Kobaz | [TK]D-Fender: so anyways... dtmf works on both phones.... if i dial from 2605->2608, 2605 can use the feature codes... if i dial from 2608->2605, 2608 can use the feature codes |
| 21:04.25 | Kobaz | [TK]D-Fender: i'll make you some cookies |
| 21:04.26 | Alan_Hicks | bijit: Troubleshooting a damn irritating echo problem on a subset of my phones. |
| 21:04.32 | Maliuta | bijit: there is a firewall between you and the sip server? |
| 21:05.26 | bijit | Maliuta: nope. It was registering very well till yesterday all of a sudden it stopped. If I connect a ATA works fine. |
| 21:05.45 | bijit | Alan_Hicks: Me struggling with sip trunk register. :-) |
| 21:06.03 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 21:06.09 | *** join/#asterisk pluggo (n=bburns@mookie.synset.net) |
| 21:06.57 | v4mp | which option should i be using to name queues as i couldn't see that part in queues.conf |
| 21:06.58 | franck | Hi all |
| 21:07.02 | pluggo | alright so... i am trying to set up an asterisk server with a te122p and dahdi... everything seems to be right, the module is loaded and everything, but i don't see the channel show up when i type dahdi show channels |
| 21:07.13 | franck | I'm looking for info on how to interface asterisk with cisco pbx |
| 21:07.51 | Kobaz | crisco |
| 21:07.59 | pluggo | i see the pseudo channel but no numbered ones |
| 21:08.06 | pluggo | and it shows up ok in dahdi_scan |
| 21:08.42 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:09.13 | pluggo | any ideas anyone? |
| 21:09.27 | pluggo | (apparently dadhi = the new name for zaptel) |
| 21:10.09 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:18.52 | *** join/#asterisk TedC (n=cabeen@milli.chem.ucsb.edu) |
| 21:19.34 | *** part/#asterisk FengShui (n=cabeen@milli.chem.ucsb.edu) |
| 21:23.14 | pluggo | *poke poke* |
| 21:23.39 | Cutlass | dahdi = zaptel as of 1.6 ....legal issues |
| 21:23.57 | pluggo | yeah... i figured that out |
| 21:24.16 | pluggo | i actually got it where its showing up as a channel in dahdi show channels |
| 21:24.24 | Maliuta | "legal issues" very much depends on which part of the world you live in :) |
| 21:24.47 | Cutlass | true... |
| 21:24.49 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:24.51 | pluggo | now, if i select pri_net signalling it says it thinks it's the network and we think we're the network... ditto for cpe |
| 21:27.25 | pluggo | according to our globalcom, we need a straight through cable to connect the te122p to the adtran... they didnt seem very sure tho |
| 21:27.56 | pluggo | is it possible this thing is looped up in software somehow? |
| 21:29.06 | pluggo | btw... dahdi is also zaptel as of 1.4.22 |
| 21:29.32 | Cutlass | ok |
| 21:30.15 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 21:30.47 | pluggo | alright |
| 21:30.52 | pluggo | y'all have been right helpful |
| 21:32.17 | Cutlass | question: I installed 1.6 with prefix=/usr/local/src ...now I can't seem to find the config files. There are some under the configs directory in the root of where I built asterisk, but I don't find any in the /usr/local/etc/asterisk directory (isn't that where it should be??)....The ones that I did find don't seem to affect the operation of asterisk |
| 21:32.43 | Cutlass | ...sorry pluggo, I don't know the answer to your quesiton...I'm waiting for a response so I can learn too :) |
| 21:33.26 | Kobaz | [TK]D-Fender: yaaaaaaaaaaay |
| 21:33.41 | Kobaz | angryuser: and a yaaaaaaay |
| 21:34.00 | Kobaz | angryuser: wW doesn't work if i dial from an iax extension to an iax trunk |
| 21:34.10 | angryuser | Cutlass : and if you start asterisk -vvvvvv does it say where he cant find conf files ? |
| 21:34.29 | Kobaz | angryuser: i used M(macro) in my Dial() to turn on the DYNAMIC_FEATURES for the called channel as well |
| 21:34.42 | angryuser | Kobaz : answer my last question please |
| 21:34.58 | Kobaz | angryuser: 2605 is iax, 2608 is sip |
| 21:35.19 | Kobaz | angryuser: i'll show you the stuff that works |
| 21:35.34 | angryuser | Kobaz : and you have dtmfmode set ? |
| 21:35.44 | Kobaz | yeah it's nothing to do with dtmf |
| 21:35.57 | Kobaz | it's based on DYNAMIC_FEATURES not being set for the called channel |
| 21:36.10 | Kobaz | exten => _91NXXXXXXXXX,n,Dial(SIP/2608,600,rM(foo)) |
| 21:36.23 | Kobaz | [macro-foo] |
| 21:36.24 | Kobaz | exten => s,1,Set(DYNAMIC_FEATURES=toggleRecorda#toggleRecordb) |
| 21:36.29 | Kobaz | well |
| 21:36.39 | Kobaz | that works for setting the features for the called phone |
| 21:36.49 | Kobaz | put it before the Dial, and then you get it on both ends |
| 21:36.56 | Cutlass | it says that modules.conf and features.conf could not be found...does not specify where it is looking |
| 21:37.48 | angryuser | Cutlass : it's a new install ? |
| 21:37.54 | Cutlass | yes |
| 21:38.12 | angryuser | Cutlass : so do make samples and find them |
| 21:38.23 | Cutlass | I didn't do make samples |
| 21:38.29 | Cutlass | just make, make install |
| 21:38.55 | Cutlass | is that the suggested method?....make samples? |
| 21:39.46 | angryuser | Cutlass : it will generate you the sample config files, you can wipe them if you want after |
| 21:39.54 | Cutlass | actually...I think I may have done that...I have all the sample files in the configs directory |
| 21:40.16 | [TK]D-Fender | Cutlass: If you are working off a 100% fresh install then do "make samples" |
| 21:40.26 | Cutlass | ok |
| 21:40.42 | Cutlass | and then rename all the files to remove the ".samples" extension? |
| 21:41.07 | angryuser | Cutlass : so "find / | grep extensions.conf" and here you are |
| 21:41.17 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:41.17 | [TK]D-Fender | Cutlass: it will do everything |
| 21:42.47 | Kobaz | [TK]D-Fender: i figured it out.... the Dial(M(macro)) feature is amazing |
| 21:43.07 | Kobaz | this will solve like, a bazillion problems |
| 21:43.19 | *** part/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
| 21:43.33 | *** join/#asterisk chigital (n=chigital@tmo-099-1.customers.d1-online.com) |
| 21:47.10 | Cutlass | thanks guys...I think it may have worked...I now see config files under /usr/local/etc/asterisk |
| 21:48.03 | v4mp | how would i setup a queue with a name as i didn't see how to in queues.conf |
| 21:48.33 | hardwire | [queueoftheawesome] |
| 21:48.37 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-174.phlapa.east.verizon.net) |
| 21:48.47 | hardwire | queue(queueoftheawesome,...) |
| 21:50.39 | *** join/#asterisk ManxPower (n=manxpowe@c-76-105-105-153.hsd1.ga.comcast.net) |
| 21:54.57 | v4mp | that go in queues.conf yea ? |
| 21:58.02 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 21:59.25 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 22:00.42 | *** join/#asterisk af_ (n=getsmart@88-149-240-19.dynamic.ngi.it) |
| 22:04.20 | *** join/#asterisk AndyML (n=alauppe@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 22:04.39 | AndyML | asterisk 1.4.19.1 - how/where would I enable callwaiting? |
| 22:05.16 | AndyML | actually - I think this is a freepbx question - i bet its this dialparties.agi causing the problem. nevermind! |
| 22:05.35 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 22:06.31 | Maliuta | AndyMillar: look at call parking |
| 22:11.54 | jmacz | Hi everyone, I'm trying 1.6 with TLS and I'm getting a "SSL cert error" when starting *, but no aditional information (asterisk logs shows nothing). What may be causing this? |
| 22:19.48 | angryuser | can someone help me with GROUP_MATCH_COUNT i need to count the total amount of channels of groups (group1,group2,group3) i dont see how to use the pattern, syntax is here http://www.pastebin.ca/1216252 , maybe something like this Gotoif($[${GROUP_MATCH_COUNT(group?)} >= 5]?true:false) |
| 22:21.00 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:21.27 | nicox | i'm sorrx what do you mean with tls, iax/tls? |
| 22:26.19 | angryuser | Gotoif($[${GROUP_MATCH_COUNT(^group[1-3])} >= 5]?true:false) that must be it.. testing ;) |
| 22:26.45 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 22:29.13 | *** join/#asterisk sircco (n=sircco@dh207-102-138.xnet.hr) |
| 22:29.32 | sircco | what's the difference in dialing queue with Dial() and with Queue() ? |
| 22:30.12 | [TK]D-Fender | sircco: "Dial" is not "Queue" |
| 22:30.20 | sircco | i know.. |
| 22:30.41 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:30.46 | sircco | but lets say i have queue 1000 I can go to it with Dial(1000@context) or with Queue(1000) |
| 22:31.01 | [TK]D-Fender | sircco: Dial just calls the devices you list for a single timeout. Queue will dial multiple devices with weighted priorities over MULTIPLE attempts,e tcx |
| 22:31.28 | [TK]D-Fender | sircco: Dial(1000@context) <- invalid syntax |
| 22:31.34 | sircco | so going to queue with Dial() is a bad idea? |
| 22:31.59 | [TK]D-Fender | sircco: So far your "plan B" is invalid from the start. |
| 22:32.17 | sircco | well that's why i came here to ask :) |
| 22:32.53 | [TK]D-Fender | sircco: Dial(1000@context) <- simply not valid. What more is there to say? You want the call to fall into a Queue then go ahead an call Queue. |
| 22:33.27 | sircco | here is exact one Dial(Local/${NLABQUEUE}@from-internal) |
| 22:33.47 | [TK]D-Fender | sircco: No point to doing a dial there that I can see. |
| 22:34.17 | [TK]D-Fender | sircco: Nothing you can't do just by doing a straight Goto, or jsut calling the same apps as that exten would lead to. |
| 22:34.51 | sircco | this is some trixbox box im trying to extend with queues from mysql |
| 22:34.57 | [TK]D-Fender | sircco: And of course this new sample doesn't prove to me what it will execute even upon attempt. |
| 22:35.16 | [TK]D-Fender | sircco: Well Trixbox & FreePBX are not supported here. They have their own channels |
| 22:35.51 | sircco | yes i know but im making my own dialplan, nothing to do with clicking and web :) |
| 22:49.46 | citywok | [TK]D-Fender, i learned today that having 4 codecs in allow statements in sip.conf means that the other end just might pick any of those codecs it likes, not in preference order. |
| 22:49.58 | citywok | contrary to all of the documentation online |
| 22:50.52 | [TK]D-Fender | citywok: Its a negotiation of preference. |
| 22:51.07 | [TK]D-Fender | citywok: Doesn't mean * will win the fight in its order |
| 22:51.25 | citywok | hah, yea, thats what i learned from the tech at voicepulse. well, actually, we learned this together. |
| 22:52.08 | citywok | gsm is a very, very cpu light codec. 15 concurrent calls and a p3 1.13 isn't even breaking a sweat |
| 22:53.21 | [TK]D-Fender | citywok: Unless * is transcoding there is nothing to load your system |
| 22:53.36 | [TK]D-Fender | citywok: And you should be picking ONE codec per peer |
| 22:53.50 | citywok | it's doing the encoding to gsm, from a zap channel (ulaw right?) |
| 22:54.25 | citywok | and yea, that was there fuckup for putting all supported codecs in their pre-configged sip.conf file, and my fucup for using it. |
| 22:54.38 | *** join/#asterisk Great_Anta_Baka (i=c4219f6d@gateway/web/ajax/mibbit.com/x-46a64bb5baa583c8) |
| 22:59.08 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 23:02.49 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 23:04.03 | *** join/#asterisk Ei5GTB (n=Paul@78.16.216.222) |
| 23:06.24 | *** join/#asterisk ctooley (n=ctooley@209.33.108.195) |
| 23:07.46 | *** join/#asterisk Entranced (n=entrance@191.23.119.70.cfl.res.rr.com) |
| 23:11.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:11.36 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 23:12.07 | brian | a bit offtopic but does anyone know how those free teleconferencing / telephone partyline services make money? |
| 23:12.36 | citywok | they generally provide free local numbers, but charge you if you want to provide a TFN |
| 23:13.14 | brian | That's not what I meant |
| 23:14.18 | citywok | sorry, i wasnt clear. my thought on the matter is they dont make money at all, except for when people pay per minute to be able to provide a TFN to their clients/customers/etc. -- they then make money by charging 8cents/minute or something retarded for that |
| 23:14.56 | brian | How do people get numbers in Minnesota or Iowa and get kickbacks? |
| 23:15.14 | citywok | i'm not sure what you mean |
| 23:15.49 | brian | Like, whenever a long distance call is made to the number, the phone company gets paid for it, and they give you a fraction of what they get. |
| 23:16.55 | _ShrikE | brian: I think you are talking about cabs |
| 23:17.00 | brian | cabs? |
| 23:17.22 | brian | I don't know what the proper word for it is |
| 23:19.16 | brian | I think you're messing with me though...I'm being serious... |
| 23:21.09 | angryuser | why when i do Noop(The quantity of calls is "${GROUP_MATCH_COUNT(group[1-3])}") i got "The quantity of calls is "0"") in new stack why "0"" with 2 "" at the end ? |
| 23:22.21 | angryuser | oh found it nevermind |
| 23:27.44 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:27.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:29.21 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-0f27561830b960e8) |
| 23:34.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:37.20 | v4mp | how would i setup a queue with a name as i didn't see how to in queues.conf |
| 23:37.43 | angryuser | my script is over, accepting any critics ;) http://www.pastebin.ca/1216317 ;) |
| 23:37.55 | angryuser | v4mp : what ? |
| 23:38.12 | v4mp | to load in new voicemail settings from voicemail.conf do i need to restart * |
| 23:38.15 | hardwire | angryuser: hah hah.. line 20 amuses me... |
| 23:38.18 | hardwire | and 21 |
| 23:38.21 | hardwire | and 32 |
| 23:38.22 | v4mp | angryuser, want to setup a queue for callers |
| 23:38.36 | hardwire | actually. I didn't look at it |
| 23:38.37 | Carlos_PHX | v4mp: reload app_voicemail |
| 23:39.01 | v4mp | ty Carlos_PHX |
| 23:39.39 | v4mp | say a queue for 'sales' queue for 'support' etc |
| 23:39.42 | Carlos_PHX | v4mp: forgot .so at the end. I just press tab after app_v |
| 23:39.46 | angryuser | v4mp : so create a queue with a name you want [thenameofqueue) |
| 23:40.07 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 23:40.17 | angryuser | v4mp : no just do reload in CLI |
| 23:40.26 | thehar | or reload app_voicemail |
| 23:40.34 | *** join/#asterisk RB2 (n=RB2@pool-71-125-95-246.nwrknj.east.verizon.net) |
| 23:40.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:40.51 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:44.29 | v4mp | angryuser, what conf does that go in and how do i lay it out or know of somewhere to help ? |
| 23:46.12 | angryuser | v4mp : hm, i am no sure if i understood the question, so you want to setu a queue right ? |
| 23:46.33 | angryuser | v4mp : how would you like to name it ? |
| 23:47.02 | v4mp | yes want to setup queues for certain extens when theres noone available to take the call |
| 23:47.06 | v4mp | yes would like to name it |
| 23:50.48 | *** join/#asterisk ManxPower (n=manxpowe@c-76-105-105-153.hsd1.ga.comcast.net) |
| 23:55.08 | v4mp | if i want to allow wav should it be allow=wav ? |
| 23:55.39 | [TK]D-Fender | v4mp: wav is not a call codec |
| 23:55.40 | angryuser | v4mp : wait a sec i need to do something |
| 23:55.59 | v4mp | ok |
| 23:56.34 | v4mp | [TK]D-Fender, wasn't sure because it uses allow=gsm and uses gsm for some of the recordings so weren't sure :) |
| 23:56.53 | seanbright | there are gsm encoded wavs |
| 23:57.22 | ManxPower | v4mp: The codecs installed on your system are the ones listed in "core show translations" (I think that's the command. |
| 23:58.48 | v4mp | nope |
| 23:58.59 | v4mp | without the s works |
| 23:59.08 | ManxPower | that would be it then. |
| 23:59.35 | ManxPower | do a help on the command. adding the documented recalc option to that command can make the numbers more accurate. |
