00:03.54 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
00:04.19 | *** join/#asterisk PASCA (n=peter@abkz236.neoplus.adsl.tpnet.pl) |
00:04.40 | PASCA | Hello all. |
00:05.15 | PASCA | Quick question... Is it possible to assign two public static IPs to one asterisk box? |
00:06.47 | PASCA | Hello? Is there anyone here? |
00:06.49 | ManxPower | PASCA: There is nothing special about two public IPs. Packets will be routed according to the linux kernel routing table. |
00:07.33 | ManxPower | of course, it could very well be a living hell to make the box route packets the way you want when using two public IPs, but that is a linux/networking issue, not an Asterisk issue. |
00:07.57 | ManxPower | (look up "source routing") |
00:08.41 | PASCA | ManxPower: Ok thanks. The issue I am experiencing with Asterisk 1.4 is the following. I have two public IPs on one ethernet card. eth0 is 51, eth1 is 48. |
00:08.50 | [TK]D-Fender | PASCA: You cannot tell * to bind to 2 IP, its either one, or all. |
00:08.57 | PASCA | Clients are able to register on 51, but not on 48. |
00:09.20 | ManxPower | PASCA: put in a route to the clients to go out via 48 |
00:09.20 | [TK]D-Fender | PASCA: And multi-homed * boxes cause real problems. |
00:09.50 | ManxPower | PASCA: on the asterisk linux box |
00:10.02 | ManxPower | then they should be able to register to 51 |
00:10.06 | ManxPower | sorry, to 48 |
00:10.25 | PASCA | ManxPower: So this is a linux kernel routing table issue correct? |
00:10.59 | ManxPower | PASCA: correct. The "multi-homed * boxes cause real problems." [TK]D-Fender is talking about is, I strongly suspect, all linux routing issues. |
00:11.23 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
00:11.28 | ManxPower | but I do agree you should not use bindip= option |
00:11.40 | [hC] | <PROTECTED> |
00:11.55 | PASCA | I have bindip set to 0.0.0.0. |
00:12.09 | ManxPower | PASCA: just comment it out. |
00:12.54 | PASCA | ManxPower: Ok thanks I will comment it out and add a route in the routing table. This should allow registrations on 48 correct? |
00:13.24 | [TK]D-Fender | ManxPower: Picture that UDP is stateless, a call comes in on A but * doesn't have a "state" (UDP!!) to know which interface to send it out. Hilarity ensues |
00:13.51 | [TK]D-Fender | ManxPower: Maybe with some conntrack zaneyness it might work... but boy, what a lot of trouble. |
00:14.12 | ManxPower | [TK]D-Fender: Asterisk DOESN'T SEND IT OUT AN INTERFACE. Asterisk sends it out, the kernel handles the source ip because asterisk did not bind to a specific interface. |
00:14.16 | [TK]D-Fender | PASCA: Whats the point of running multiple IP's both public on the same box for this? |
00:14.24 | lanning | actually the issue is that when bound to the global socket (0.0.0.0), the source IP is the interface it routes out of, if you have two IP addresses on one card, it picks the one assigned to the physical interface (as opposed to the eth0:1 interface) |
00:14.42 | [TK]D-Fender | ManxPower: Sorry, correct, not "interface", just that it'll sounce by one arbitrary interface or another |
00:14.59 | [TK]D-Fender | IP -> to go and "interface" |
00:15.05 | [TK]D-Fender | bleh, you know what I mean |
00:15.15 | [TK]D-Fender | Anyway you go about it... trouble. |
00:15.40 | [TK]D-Fender | PASCA: If you want to do something like this I suggest you run a proxy in front |
00:16.03 | PASCA | [TK]D-Fender: I have some extensions registered on 51 and have about fifty on 48. Instead of reprogramming the ATAs with the 51 IP, I want to group the all on one server. |
00:16.05 | ManxPower | lanning: what if you put a route in the routing table to send packets destined for 209.61.27.185 goes out interface/IP 206.87.43.12 |
00:16.15 | ManxPower | (just random example addresses) |
00:16.19 | *** join/#asterisk __jeff_O (n=silas@ool-18bab2c5.dyn.optonline.net) |
00:16.41 | ManxPower | PASCA: it is better to change them all to use a hostname or to the same IP. |
00:17.16 | __jeff_O | if my asterisk server is behind a firewall, what are the necessary ports to forward so that computers outside the firewall can make sip calls? |
00:17.22 | ManxPower | otherwise sometime in the future when you totally forgot about the route line you put in /etc/rc.d/rc.local and bad things happen |
00:17.42 | __jeff_O | Ports 5060 to 5070 and 10001 to 20000 ? |
00:17.43 | ManxPower | __jeff_O: Read the sipnat document *CAREFULLY* |
00:17.46 | ManxPower | ~sipnat |
00:17.46 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
00:17.58 | ManxPower | the wiki one is confusing, use the first lin, |
00:18.00 | ManxPower | link |
00:18.19 | PASCA | ManxPower: That's the thing. It may be difficult to change all devices. So I would rather have 2 ips on 1 * box. |
00:18.53 | PASCA | Do you really think this will cause many more problems than it is worth? |
00:19.05 | ManxPower | PASCA: yes. |
00:19.32 | ManxPower | I would do it as a temporary measure to make the clients registering to the old address still work -- as a transition thing. |
00:19.51 | ManxPower | PASCA: are the clients PCs or phones? |
00:20.16 | __jeff_O | ManxPower: ok, so are the 10000-20000 ports for audio? That seems to be what I've read elsewhere and the doc you gave me said they should be open. |
00:20.25 | PASCA | ManxPower: Phones... Combination of ATAs and softphones. |
00:21.16 | ManxPower | PASCA: I would try this. Have one of the softphone pc's ping the 48 address. See what the response is. I suspect it will be from the 51 address. |
00:21.31 | ManxPower | if so, then you can test the route changes on the linux box by just running a ping from one of the clients |
00:22.22 | ManxPower | __jeff_O: configurable in /etc/asterisk/rtp.conf |
00:22.51 | PASCA | ManxPower: Thanks a lot for the info. You have been a great help! :) |
00:22.56 | PASCA | [TK]D-Fender: Thanks for your help as well. |
00:23.03 | ManxPower | PASCA: let me know how it goes. |
00:23.22 | ManxPower | routing is bitch to get right, even on dedicated routers. |
00:24.13 | [TK]D-Fender | PASCA: You're welcome, and ManxPower easily outclasses me on the networking side of this. Profit from whatever he can offer you. |
00:24.20 | __jeff_O | ManxPower: cool, thanks. Hopefully opening those ports will fix my probs |
00:24.36 | PASCA | ManxPower: I won't be able to give it a shot until tomorrow, but I will come back and let you know for sure. |
00:24.40 | [TK]D-Fender | PASCA: Mine was more warning than guidance in that respect |
00:25.27 | PASCA | [TK]D-Fender: It was some useful information indeed. Your warning and his guidance. I will give it a go, hopefully as a temp fix. |
00:25.42 | PASCA | Will let both of you know how it goes. |
00:25.52 | [TK]D-Fender | PASCA: You've walked away ahead of the game then. Best of luck to you. |
00:26.59 | PASCA | Thanks! Have a good one guys. |
00:30.00 | *** join/#asterisk phl4kx (i=phl4kx@190.40.104.189) |
00:30.02 | phl4kx | hi for all |
00:30.18 | phl4kx | what is the IAX or SIP phone for call to DIGIUM SUPPORT???? |
00:32.57 | phl4kx | what is the IAX or SIP phone for call to DIGIUM SUPPORT???? |
00:35.10 | phl4kx | exten => 500,1,Playback(demo-abouttotry); Let them know what's going on |
00:35.10 | phl4kx | exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo |
00:35.10 | phl4kx | exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site |
00:35.10 | phl4kx | exten => 500,n,Goto(s,6) ; Return to the start over message. |
00:35.11 | phl4kx | :D |
00:36.30 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d325993de75d3c1d) |
00:38.14 | rue_mohr | look its asterisk related |
00:39.34 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
00:40.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
00:48.47 | *** part/#asterisk PASCA (n=peter@abkz236.neoplus.adsl.tpnet.pl) |
00:58.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
01:00.10 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
01:05.45 | *** part/#asterisk jsmith-teaching (n=njsmith@72.21.36.138) |
01:32.36 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
01:36.38 | *** join/#asterisk synchris (n=synchris@athedsl-218704.home.otenet.gr) |
01:39.58 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
01:41.21 | Damin | Please DIGG and pass on: http://digg.com/linux_unix/Ohio_LinuxFest_2008 |
01:42.13 | jaytee | hmmm, Ohio |
01:42.35 | jaytee | wonder if any of the Ubuntu Loco team from here in Indiana are going |
01:42.59 | Damin | Yep.. they party like rockstars.. |
01:43.06 | Damin | We have Jono Bacon keynoting.. |
01:43.14 | Damin | I suspect there will be a large Ubuntu presence.. |
01:44.02 | jaytee | Don't know if I'll be able to go the way my schedule is |
01:45.30 | jaytee | they do party hard. When they first got team approval they partied so hard they almost got kicked out of the Claddagh Irish Pub |
01:49.23 | *** join/#asterisk moy (n=moy@189.169.91.147) |
01:50.31 | *** join/#asterisk dwayne (i=dwayne@76.29.245.9) |
02:11.06 | *** part/#asterisk Steve_J-obs (n=Chris123@209.58.251.50) |
02:11.18 | lmadsen | well hello there! |
02:11.25 | Qwell | ohai |
02:11.30 | lmadsen | omghi2u |
02:11.46 | *** join/#asterisk sapere (n=sapere@c-71-206-107-228.hsd1.mi.comcast.net) |
02:12.30 | sapere | evening all -- question about the asterisk service provider, ironvoice ... maybe i don't understand how the process works, but where do i get to pick a phone number? |
02:14.32 | lmadsen | anyone have suggestions for Asterisk Cookbook recipes they would like to see in a book? |
02:18.52 | *** join/#asterisk oilinki (n=oil@ppp-124-120-4-102.revip2.asianet.co.th) |
02:19.33 | Qwell | Mr. Madsen |
02:26.51 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
02:28.19 | lmadsen | Mr. North |
02:28.24 | Qwell | looks around |
02:28.28 | Qwell | he's dead |
02:28.31 | lmadsen | oh really |
02:28.34 | Qwell | mmhmm |
02:28.37 | Qwell | shh |
02:28.41 | lmadsen | oic |
02:28.49 | lmadsen | Mr. Parker. |
02:29.03 | Qwell | meh, I don't like that guy |
02:29.14 | Qwell | he's a jerk |
02:29.21 | lmadsen | Mexico just scored on Canada in the World Cup qualifying, so I will now find something else to watch since that game is over |
02:29.26 | lmadsen | Qwell: I agree |
02:29.33 | lmadsen | Qwell: give me recipe ideas! :) |
02:29.35 | mchou | Is there a way in Asterisk to send DTMF after caller and callee have been bridged? Kind of like 'Dial(blah,timeout,D:12345)' except after both sides have been bridged |
02:29.37 | lmadsen | I have 91 so far |
02:29.47 | Qwell | I gave you one! |
02:29.54 | lmadsen | that's why I have 91 :) |
02:29.55 | lmadsen | and not 90 |
02:29.56 | Qwell | :D |
02:30.04 | file | lmadsen: new TV channels! |
02:30.05 | Qwell | how about... |
02:30.11 | Qwell | a youtube "viewer" |
02:30.12 | lmadsen | file: oh? |
02:30.24 | Qwell | can you convert flv to ulaw? :D |
02:30.40 | file | lmadsen: Joy! |
02:30.44 | *** join/#asterisk EI5GTB (n=paul@78.16.158.28) |
02:31.14 | lmadsen | has no idea what file is talking about |
02:31.27 | lmadsen | hrmmm... someone just told me he thinks dialplan is what is holding asterisk back |
02:31.31 | lmadsen | scoffs |
02:32.11 | mchou | I think so too. |
02:32.32 | lmadsen | funny, because I've created some pretty good dialplan |
02:32.41 | mchou | most keywords dont make any sense..... |
02:33.06 | lmadsen | define: keywords |
02:33.20 | mchou | like what's the diff betw. 'username' and 'fromuser' |
02:33.27 | mchou | lol |
02:33.31 | Qwell | blame sip, not Asterisk |
02:33.34 | file | that's not dialplan |
02:33.37 | lmadsen | exactly |
02:33.45 | mchou | and the classic: 'user vs. peer' |
02:33.49 | mchou | lol |
02:33.51 | Qwell | blame sip, not Asterisk |
02:34.01 | lmadsen | also not dialplan |
02:34.04 | mchou | user vs. peer is sip? |
02:34.21 | Qwell | it was easier to hit up+enter |
02:34.23 | lmadsen | user vs. peer is not really THAT hard |
02:34.39 | Qwell | but, yeah, basically. SIP has multiple types of servers |
02:34.47 | mchou | no, but nobody is clear on the concept |
02:35.07 | lmadsen | incoming calls try to match on users from bottom to top first on the [username], and if that doesn't match, it'll search peers for IP address from bottom to top |
02:35.24 | mchou | where ppl at digium basically says 'just use peer!' |
02:35.26 | mchou | lol |
02:35.29 | mchou | lame |
02:35.37 | Qwell | user = gone |
02:35.37 | lmadsen | that's a bit simplified, but is pretty much the jist of it |
02:35.41 | lmadsen | also true |
02:36.39 | lmadsen | I wish I had chocolate |
02:36.56 | mchou | not to mention 'include' and extension matching priority madness |
02:37.04 | mchou | more lameness |
02:37.17 | lmadsen | I believe that has more to do with your lack of understanding |
02:37.25 | mchou | no it doesn't |
02:38.04 | lmadsen | I've never had a problem understanding matching order in my dialplans |
02:38.05 | mchou | an include stanza gets matched last |
02:38.10 | lmadsen | right |
02:38.13 | lmadsen | that makes sense |
02:38.17 | lmadsen | match local first, then follow includes |
02:38.23 | mchou | make no sense whatsoever |
02:38.29 | lmadsen | I disagree |
02:38.49 | lmadsen | you think it should search the includes before the local context? |
02:38.54 | mchou | cause there are plenty of conflicting 'namepaces', if you will |
02:38.56 | lmadsen | that makes no sense to me |
02:39.53 | mchou | no, a rule like 'match specific prior to general (regardless of includes)' makes MOST sense |
02:40.13 | lmadsen | uhhhh.... it does do that |
02:40.23 | lmadsen | _123XXX is matched before _XXXXXX |
02:40.38 | mchou | meaning include is more like a c preprocessor #include |
02:41.03 | mchou | lmadsen: it doesnt if includes ALWAYS gets matched last |
02:41.24 | lmadsen | right, it should match locally first |
02:41.40 | mchou | cause an include can clearly have more 'specific' matches than the local |
02:41.51 | Qwell | mchou: There's this book that some totally cool guys (who owe me money) wrote, that you should check out |
02:41.53 | Qwell | ~book |
02:41.54 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
02:42.07 | mchou | Qwell: stop lecturing me on the book |
02:42.26 | mchou | nothing I said contradicts anythingf mentioned in the book |
02:42.35 | Qwell | lecturing? I brought it up once |
02:42.48 | lmadsen | this conversation is going no where, so I'm going to find something to eat |
02:43.42 | mchou | I just think the way asterisk parses includes and pattern matching is lame when includes are ALWAYS matched last. |
02:44.15 | Qwell | you can use it to your advantage fairly easily |
02:44.23 | mchou | Asterist could be a bit smarter about pattern matching rather than relegating includes last |
02:46.44 | lmadsen | I still disagree on your position. I've been writing dialplans for 5 years, and never found it to be a problem; quite the opposite |
02:46.53 | mchou | lol |
02:47.18 | mchou | it's not a problem if you don't use includes, just like 'user' |
02:47.28 | lmadsen | I use plenty of includes |
02:47.33 | Qwell | it isn't a problem if you know how to use them |
02:48.46 | mchou | yeah, now I dont know how to write a dial plan since I point out the sortcoming with include processing in asterisk |
02:48.56 | mchou | shortcomings* |
02:49.43 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
02:49.45 | mchou | Does Digium or asterisk have anything that you guys find 'irregular?' |
02:50.03 | mchou | I mean they can do no wrong, right? |
02:50.15 | lmadsen | I have found lots of things |
02:50.24 | lmadsen | I just disagree on your position about includes |
02:50.50 | lmadsen | and you can disagree with me |
02:50.51 | lmadsen | NEXT!!! |
02:51.12 | mchou | lmadsen: fine. I can respect that. But saying "I've never had a problem' is hardly a rebuttal |
02:51.36 | mchou | when I said an include can have more specific patterns than local. |
02:51.49 | lmadsen | I don't understand what you want me to say. I've written plenty of 1000+ line dialplans that utilize includes heavily, and never found a problem |
02:52.11 | lmadsen | perhaps I've not run into the same problems as you |
02:52.16 | lmadsen | but the rules are quite clear |
02:52.41 | mchou | maybe they are a 1000+ lines since asterisk didnt do smarter pattern matching :) |
02:52.51 | lmadsen | sure thing |
02:52.56 | lmadsen | must be it |
02:53.08 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
02:54.41 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
02:55.26 | dwayne | I'm computering... |
02:56.51 | [TK]D-Fender | mchou: If you want to talk about "irregular" I'll let you know when the next bowel movement is on its way... |
02:57.18 | dwayne | lol |
02:57.30 | mchou | [TK]D-Fender: You clearly need more prunes :) |
02:57.52 | mchou | [TK]D-Fender: Constipation is a killer! |
02:58.06 | [TK]D-Fender | mchou: "constant" isn't an enviable measurement of "success" |
02:58.36 | dwayne | especially when it is combined with high frequency |
02:58.45 | [TK]D-Fender | mchou: As for dialplan issues, there is little I can see to complain about. Takes very little effort to acheive 99% of peoples needs. |
02:59.16 | [TK]D-Fender | mchou: Do you have an actual issue you'd like advise with? |
02:59.25 | [TK]D-Fender | mchou: Or are you just trying to stir the pot? |
02:59.47 | mchou | [TK]D-Fender: [19:29] <mchou> Is there a way in Asterisk to send DTMF after caller and callee have been bridged? Kind of like 'Dial(blah,timeout,D:12345)' except after both sides have been bridged |
03:00.02 | mchou | [TK]D-Fender: and no, I'm not trolling |
03:00.43 | [TK]D-Fender | mchou: After they've been bridged huh? That'd be tough indeed. Whats the purpose? |
03:00.48 | mchou | [TK]D-Fender: |
03:00.48 | mchou | [19:31] <lmadsen> hrmmm... someone just told me he thinks dialplan is what is holding asterisk back |
03:00.48 | mchou | [19:31] * lmadsen scoffs |
03:01.11 | mchou | [TK]D-Fender: that was all I was responding to |
03:01.51 | mchou | [TK]D-Fender: Lame ass grandcentral is the purpose |
03:02.08 | Qwell | once the channels have been bridged, you are no longer in the dialplan |
03:02.10 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
03:02.11 | [TK]D-Fender | mchou: what was intended by use of "dialplan"? Is it a pattern matching aspect? Priority logic? Lack of great SIP signalling control via it? That is an open ended wide-angle attack. |
03:02.12 | Qwell | hence the "bridge" |
03:02.30 | mchou | [TK]D-Fender: I dunno what's going on w/ Grand Central as far ast DTMF |
03:02.32 | [TK]D-Fender | mchou: Why do both sides need the dtmf? |
03:02.34 | mchou | as* |
03:02.56 | [TK]D-Fender | mchou: When you dial with "Dial" why do YOU need DTMF? |
03:03.28 | mchou | [TK]D-Fender: I dont need DTMF. GC does. |
03:03.31 | [TK]D-Fender | mchou: I can see why you'd want to pass THEM digits. D() does already do that. What are you trying to have * ask itself? |
03:03.52 | [TK]D-Fender | mchou: but you said AFTER bridge. Whats wrong with sending it to them BEFORE the bridge? |
03:04.13 | [TK]D-Fender | mchou: To them the call is set up, so whats the difference? |
03:04.16 | mchou | [TK]D-Fender: the programmtic way doesnt work successfully with GC. On manual works |
03:04.28 | mchou | s/On/Only |
03:04.42 | [TK]D-Fender | mchou: What kind of failure? Perhaps * sends the digits too fast? |
03:04.47 | *** join/#asterisk nr4q (n=dave@wsip-70-183-143-230.hr.hr.cox.net) |
03:04.55 | mchou | nope. plenty of wwwwww |
03:04.59 | [TK]D-Fender | mchou: as in they need you to wait a sec before litening for your answer... |
03:05.06 | mchou | w=0.5sec delay |
03:05.14 | nr4q | man... asterisk gives me a chub |
03:05.20 | [TK]D-Fender | mchou: I'm not certain "w" works inside of D() has this been confirmed? |
03:05.29 | Qwell | I think w only works on zap |
03:05.37 | [TK]D-Fender | mchou: AFAIK, "w" only works in the tech part of a zap call |
03:05.45 | mchou | w works. At lest it ways before sending DTMF to me |
03:05.49 | [TK]D-Fender | mchou: Do you see it documented somewhere? |
03:06.08 | mchou | core shore application Dial in * 1.4 |
03:06.17 | mchou | shoe* |
03:06.22 | mchou | show** |
03:06.24 | mchou | lol |
03:06.27 | file | oh it'll work any time Asterisk sends DTMF to a channel... |
03:06.40 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
03:07.55 | [TK]D-Fender | mchou: I don't see it in the insrtuctions. |
03:08.33 | file | mchou: are you sending a string of DTMF digits? |
03:09.24 | mchou | file: just :wwwww1 |
03:09.47 | mchou | various amounts of w's of course :) |
03:10.08 | Qwell | :? |
03:10.25 | file | the internal API call would just discard that as an invalid digit |
03:10.27 | Qwell | oh, right |
03:10.49 | mchou | file: huh?? what's an invalid digit? |
03:10.51 | file | 0-9*#aAbBcCdDfF are valid |
03:11.08 | file | : is invalid |
03:11.15 | file | but not your problem. |
03:11.37 | [TK]D-Fender | Sounding like 'w" is not an option for D() |
03:11.45 | mchou | file: ':' is not a part of the DTMF string :) |
03:11.47 | Qwell | wait, :? |
03:11.55 | file | it's to separate calling/called |
03:11.56 | Qwell | you're sending the DTMF to the calling party? |
03:12.03 | [TK]D-Fender | mchou: here's a though, call it yourself manually and the very instant you see it answer, try dialing. |
03:12.03 | mchou | D([called][:calling]) - Send the specified DTMF strings *after* the called |
03:12.03 | mchou | <PROTECTED> |
03:12.03 | mchou | <PROTECTED> |
03:12.03 | mchou | <PROTECTED> |
03:12.03 | mchou | <PROTECTED> |
03:12.19 | [TK]D-Fender | mchou: Nowhere does that say "w" is valid |
03:12.26 | file | [TK]D-Fender: It's valid. |
03:12.31 | mchou | Qwell: yup. |
03:12.38 | Qwell | did you answer? |
03:12.42 | mchou | [TK]D-Fender: use the source luke :) |
03:12.43 | [TK]D-Fender | file: I'm still looking through the code for it |
03:12.51 | file | [TK]D-Fender: ast_dtmf_stream in main/app.c |
03:13.06 | [TK]D-Fender | mchou: I am looking in the code, just haven't gotten there yet. If you care to direct me I'll happily look there./ |
03:13.19 | mchou | Qwell: If by 'answer' you mean did I pick up the phone, then yes :) |
03:13.26 | Qwell | The calling chanenl |
03:13.30 | Qwell | channel too |
03:14.01 | Qwell | If grand central doesn't allow DTMF before the call is answered... |
03:14.01 | mchou | Qwell: I'm not sure the channel is explicitly answered...... |
03:14.18 | file | Qwell: hrm... I wonder if Dial() doesn't do that... |
03:14.24 | Qwell | file: pre-bridge? |
03:14.27 | [TK]D-Fender | file: I'm seeing it. If that is what's referenced for all realted apps, then ok/fine/sure. |
03:14.32 | Qwell | it might |
03:14.35 | file | Qwell: this would certainly be executed pre-bridge... |
03:15.02 | Qwell | worth a try though. put an answer/wait before the dial |
03:15.02 | [TK]D-Fender | No, the line would have to be answered to pass DTMF no? |
03:15.13 | file | [TK]D-Fender: maybe. |
03:15.14 | Qwell | [TK]D-Fender: the question is whether it's answered at that point yet or not, implicitly |
03:15.14 | mchou | Qwell: oh, Answer would work. but it would have side effects |
03:15.21 | Qwell | mchou: such as? |
03:15.30 | Qwell | nevermind |
03:15.32 | [TK]D-Fender | mchou: a thought, try it in M() <- |
03:15.52 | mchou | if you 'Answer programmatically' GC regards that called as picked up. |
03:16.20 | mchou | I dont want to that to happen b4 I pick up...... |
03:16.33 | mchou | physically pick up, that is |
03:17.02 | file | [TK]D-Fender: that is a solid idea |
03:17.15 | mchou | [TK]D-Fender: sorry? you mean macro or ? |
03:17.19 | [TK]D-Fender | file: I prefer to think of it as "all thats left" :) |
03:17.24 | [TK]D-Fender | mchou: Yes. |
03:17.40 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
03:17.41 | file | is too tired to look through app_dial |
03:17.47 | [TK]D-Fender | mchou: in yoru macro do "answer", "wait(2)", SendDTMF", etc |
03:18.54 | mchou | I dont see how that avoid the side effect of having the call answered b4 I physically pick up? |
03:19.02 | mchou | avoids* |
03:19.36 | file | the Macro only executes once a called channel is answered |
03:19.59 | file | (if done via the M dial option) |
03:20.03 | [TK]D-Fender | mchou: Your side isn't the issue is it? you are calling OUT to them, no? |
03:20.18 | mchou | [TK]D-Fender: nono. GC calls me |
03:20.43 | [TK]D-Fender | mchou: tehn ANSWER their call, and jsut do SendDTMF before calling inside anywhere |
03:21.05 | file | drifts off to bed |
03:21.12 | [TK]D-Fender | mchou: and D() should send back to them anyways. |
03:21.15 | mchou | [TK]D-Fender: back up. lemme descibe how GC works |
03:21.28 | [TK]D-Fender | mchou: yeah I know you have to ACK the call |
03:21.44 | [TK]D-Fender | mchou: I ran into others trying to automate the "annoy the human" to answer |
03:22.24 | mchou | [TK]D-Fender: the stated purpose is NOT to annoy the human |
03:22.26 | mchou | :) |
03:22.54 | [TK]D-Fender | mchou: yeah, thats YOUR goal, not theirs. |
03:23.03 | mchou | it's for callee me to decibe whether I want it to go to VM and the like :) |
03:23.13 | mchou | decide* |
03:23.27 | [TK]D-Fender | mchou: Have you tried just Answer first, then SendDTMF direct in the dialplan and THEN passing off to your inside phone, etc? |
03:24.12 | mchou | [TK]D-Fender: no, not yet, but others have and they say it seems to work (according to the intarweb) |
03:25.05 | mchou | [TK]D-Fender: I have no reason to doubt that might work |
03:25.10 | [TK]D-Fender | mchou: well m() is definately no use, so if you can't get D() to work (DO Answer the call before the dial at the bare minimum, and play a sound (even silence) BEFORE You Dial (might help to set up the audio path). |
03:25.23 | [TK]D-Fender | mchou: Aside from that you have my only other suggestion |
03:25.48 | [TK]D-Fender | mchou: and others have gotten it to work. I HIGHLY recommend the "Answer" + "Playback" right from the start |
03:26.07 | [TK]D-Fender | mchou: Just saved someone elses sanity with that earlier today |
03:26.27 | mchou | [TK]D-Fender: yeah. I just dont like the side-effect of programmatic answer |
03:26.56 | mchou | that's the whole point, isnt it? Automating this crap :) |
03:28.01 | [TK]D-Fender | mchou: GC is generally considered a toy for cheap-ass kiddies who haven't gotten around to trying to bypass CID blcok with that other kiddie schmuck provider they saw a youtube video for. |
03:28.18 | mchou | haha |
03:28.37 | mchou | I resemble that remark :) |
03:29.07 | [TK]D-Fender | A slow painful progression towards the "I give up on VoIP, so I can use my 5$winmodem as FXS/FXO right?@!?! plaeze!!!" |
03:29.14 | mchou | but I got it from David Pogue of NYT :) |
03:29.37 | [TK]D-Fender | mchou: Pogue reminds me of "Hate By Numbers" videos. |
03:29.48 | Nugget | http://macnugget.org/stuff/asterisk-irc.txt |
03:30.01 | mchou | [TK]D-Fender: Never seen those videos |
03:30.23 | [TK]D-Fender | mchou: Same kinda sarcastic tone in their teardowns of things. |
03:30.33 | Nugget | hate by numbers is a cracked.com thing |
03:31.55 | [TK]D-Fender | That guy is kinda funny... |
03:32.35 | mchou | well, at least david pogue is not completely clueless |
03:32.46 | Qwell | yeah, neither is Markoff |
03:32.52 | mchou | unlike 2bigtelephony guys |
03:33.17 | mchou | I heard their crap the other night any nearly died laughing |
03:33.30 | Qwell | link? |
03:33.41 | mchou | CLEC==Common Local Exchange Carrier. |
03:33.43 | mchou | lol |
03:33.56 | mchou | and other faux pas |
03:34.32 | Qwell | mchou: who? |
03:34.57 | mchou | Qwell: some jerks doing audio blog |
03:35.10 | Qwell | got a link? |
03:35.20 | [TK]D-Fender | mchou: well I think I'll more permanently move you off of the "troll" list and into the "rabble rousing / trouble seeking / agitant" list :) |
03:35.25 | mchou | http://2bigtelecomguys.com/ |
03:35.26 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
03:35.35 | mchou | it's a real hoot |
03:35.55 | mosty | is there a manager command to take two channels and bridge them? |
03:35.59 | Qwell | this weeks? |
03:36.36 | mchou | Qwell: nah |
03:36.55 | mchou | Qwell: Aug 19 |
03:40.33 | Qwell | their voices are annoying |
03:40.40 | Qwell | record in > 8khz, thanks |
03:41.00 | mchou | Qwell: Try not to focus on their voice but content |
03:41.06 | Qwell | trying |
03:41.35 | mchou | I mean the voices are annoying but that's not what get's my goat |
03:41.41 | mchou | gets* |
03:44.53 | *** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
03:45.28 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
03:45.48 | prg3 | I'm having a problem with asterisk-gui/http, it's giving me 403 forbidden for files that are there.. and I can't seem to get any sort of debugging output from it. Ideas? |
03:46.29 | mchou | prg3: did you authenticate? |
03:46.54 | prg3 | mchou: in the browser? I'm not given any options to do so.. is there a .htaccess that I'm missing or something? |
03:47.02 | mchou | sigh |
03:47.33 | *** join/#asterisk Levonk (n=lk@75.62.129.25) |
03:47.51 | mchou | prg3: you think asterisk will let _anyone_ mess with it over the web? |
03:48.25 | prg3 | mchou: I'd expect it to prompt me for a username and password, yes. I was expecting to have the first page I see being the authentication page. |
03:49.09 | mchou | prg3: afaik it doesnt prompt. you need to go to the approriate page to gog on |
03:49.15 | mchou | log* |
03:50.31 | prg3 | according to this page: http://www.asteriskguru.com/tutorials/asterisk_gui.html I've done all of that, yet it still will not prompt for authentication info.. |
03:50.39 | prg3 | is there a debugging flag for http somewhere? |
03:51.09 | Qwell | what URL are you trying to hit? |
03:51.38 | prg3 | asterisk/static/cfgbasic.html although all of them do the same thing, and cfgbasic.html is just a meta refresh to index.html |
03:51.42 | *** part/#asterisk dwayne (i=dwayne@76.29.245.9) |
03:53.14 | Qwell | T3=45mbit? |
03:53.37 | Qwell | oh, I guess it is |
03:55.38 | prg3 | Is there something better then asterisk-gui? I'm mostly looking for a reporting, and listening in on recorded (or live) conversations.. call center managers want to keep an eye on the call center type thing |
03:56.27 | mchou | prg3: most of us here dont support GUIs |
03:56.59 | mchou | prg3: and better is a rather subjective term |
03:57.41 | [TK]D-Fender | prg3: Asterisk-GUI was not made for any of that |
03:58.29 | prg3 | mchou: Yeah, I know :) |
03:58.54 | prg3 | I'd prefer it.. but the managers won't be able to do anything CLI based.. |
04:00.25 | prg3 | Oh, maybe better question, is there a decent (pick your own definition of decent) web frontend for CDR data? (mysql) |
04:00.26 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
04:00.31 | [TK]D-Fender | prg3: If you're at all competent at programming it'd be easy enough to make a web interface for them for this. |
04:00.53 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
04:01.08 | [TK]D-Fender | prg3: For reporting there are already 3rd party web-based solutions available. Spying would be a small custom job |
04:01.17 | heedly | having people write programs for you is pretty smart though. |
04:01.18 | prg3 | I'm not.. but that's where I might need to go. |
04:01.27 | mosty | is there an easy way to bridge two existing channels in asterisk 1.4? |
04:01.32 | prg3 | Spying we can leave out, or they can drive that via a phone |
04:01.55 | [TK]D-Fender | prg3: then go hit the WIKI for a list of these other reporting tools. |
04:01.59 | prg3 | [TK]D-Fender: Any suggestions for the first one I should look at? |
04:02.11 | [TK]D-Fender | prg3: Asterisk-stat, etc. |
04:02.20 | [TK]D-Fender | Areski... |
04:03.43 | prg3 | [TK]D-Fender: Thanks |
04:04.09 | *** join/#asterisk Ardi (n=NiMiTz_C@cpe-72-227-190-157.nyc.res.rr.com) |
04:04.48 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
04:07.07 | *** join/#asterisk hi365_m (n=hi365@213.151.58.24) |
04:08.51 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
04:17.30 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
04:20.34 | *** join/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
04:20.48 | a1fa | are there any tools that can stress test a number based on SIP? |
04:23.15 | voxter | has anyone implemented multicast paging in asterisk yet? i saw a couple posts on the list, but nothing definitive |
04:23.50 | jameswf-home | a1fa: sipp |
04:24.32 | a1fa | sipp can dial a did number? |
04:24.48 | Qwell | brownie points for anybody who can tell me the longitude of the north pole |
04:25.35 | a1fa | i need to join a conference bridge and basically flood the shit out of it with number of concurent calls from my * box |
04:25.56 | Qwell | (tip, latitude is 90) |
04:26.43 | jameswf-home | 0? |
04:27.05 | Qwell | you lose |
04:27.41 | jameswf-home | eh well |
04:30.02 | Ardi | Qwell: which one geographic or magnetic north pole |
04:30.25 | Qwell | true north |
04:30.45 | Qwell | (geographic) |
04:30.46 | jameswf-home | Longitude is meaningless at the north pole (and the south pole too). It has all longitudes at the same time. |
04:30.55 | jameswf-home | google foo hah |
04:31.02 | Qwell | :D |
04:31.26 | jameswf-home | sadly found on kde.org wtf ever that has to do with anything |
04:31.27 | *** join/#asterisk techman97 (n=myweiner@97-91-103-181.dhcp.roch.mn.charter.com) |
04:32.20 | jameswf-home | Jesse ventura looks like crap |
04:32.25 | techman97 | evening all - my mind is blanking, hopefully you can help me. I'm setting up a Sangoma card (US PRI) and the line I put in the zapata.conf file that allows me to send my station ANI is escaping me...help? |
04:32.37 | techman97 | jameswf-home: yes he does. I'm from MN and he never looked *good* |
04:32.38 | techman97 | hahahaha |
04:33.00 | Qwell | techman97: ...and yet you elected him |
04:33.01 | Qwell | :p |
04:33.37 | techman97 | yeah...well....It's a lot like South Park's episode of choosing between a douche and a turd sandwich.. |
04:33.52 | techman97 | the three people that were running - Ventura was the lesser of the three evils. |
04:34.04 | techman97 | and he didn't do crap - good or bad - while in office, so it was a net wash. |
04:34.19 | jeev | Qwell |
04:34.26 | Qwell | jeev |
04:34.39 | jameswf-home | http://boozhy.typepad.com/my_weblog/images/jesse_ventura.jpg <<< i would put change in his coffee |
04:34.43 | techman97 | I think I found the command - "usecallingpres=yes" |
04:34.52 | jeev | hi. |
04:35.43 | jameswf-home | So is obama the douche and mccain the turd sandwitch |
04:35.43 | techman97 | he made an appearance at the Ron Paul event during the RNC....if he would just shave and style his hair, he wouldn't look like a cracked out street pimp |
04:35.56 | techman97 | jameswf-home: it's kinda turning out that way |
04:36.07 | techman97 | you tell me the last election where that WASN'T the choice...LOL |
04:36.07 | jeev | slaps jameswf-home, obama aint no douche! he's gonna save us f00| |
04:36.15 | *** part/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
04:36.16 | jameswf-home | I am voting for the pig in lipstick |
04:36.19 | techman97 | hahaahahah |
04:36.21 | jeev | wow |
04:36.22 | jeev | really ? |
04:36.40 | jameswf-home | VPILF bayyyybeeee |
04:36.41 | jeev | jameswf-home must be one of those who installed asterisk from a package! |
04:36.55 | Qwell | jeev: I hear he uses trixbox |
04:37.15 | jeev | wow |
04:37.19 | jameswf-home | uses ubuntu-server (shuddup) |
04:37.25 | jeev | ubuntu |
04:37.27 | jeev | EVEN worse |
04:37.32 | jeev | real men use slackware |
04:37.44 | jeev | republicans use ubuntu |
04:38.07 | jameswf-home | http://dontcallmyboss.blogspot.com/2008/09/my-pet-project.html << what i use :) |
04:38.33 | jeev | http://www.screamingpoints.com/archives/idiot-genius.jpg |
04:38.34 | jeev | hahahahahah |
04:39.18 | jeev | your pet project blows |
04:39.38 | techman97 | ouch jeev, don't hurt the boy's feelings or nuttin' |
04:39.44 | techman97 | hehehe |
04:39.50 | jameswf-home | I am not a fush ban i mean bush fan but im not sure obama is the cure no matter how well he parts the waters and turns water to wine |
04:40.03 | techman97 | but I love wine man... |
04:40.32 | jeev | http://tbn0.google.com/images?q=tbn:PHVlzf-bH_e41M:http://willful-ignorance.com/uploaded_images/pro-war-735076.jpg |
04:40.32 | jameswf-home | I am more of a 151 and coke guy |
04:40.32 | jeev | hahahah |
04:40.36 | techman97 | the biggest problem is that they're both the same ol' politics....big talk to get in, do nothing but status quo when in office. |
04:40.55 | jeev | jameswf-home, doesn't matter, obama talks change, mcbush talks the same, perm tax cuts, war in iraq.. this and that, then picks that hot ass milf and then says it's time for change. |
04:41.06 | jameswf-home | Moran is minisotan for moron also used in wiscansan |
04:41.22 | jeev | hence why i dont consider minisotan's or whatever people. |
04:41.22 | jeev | hahah |
04:41.32 | techman97 | oh SNAP jeev |
04:41.53 | techman97 | 'dem fighin' werds ya know....don't make me get out my waders and show ya' who's the best on the lake |
04:42.01 | techman97 | yea fer s00re |
04:42.02 | jameswf-home | I want a Ron Paul Ross Perot ticket |
04:42.09 | jeev | ok ok o |
04:42.14 | jeev | my 2 other colo'd servers are online |
04:42.17 | jeev | what the hell should i use them for |
04:42.22 | techman97 | pr0n |
04:42.31 | techman97 | what else is the intertubes good for? |
04:42.31 | jameswf-home | ipv6 porn |
04:42.56 | techman97 | setup a pr0n site and then advertise on CraigsList for developers to help you out for free access |
04:43.20 | jeev | lol |
04:43.43 | jameswf-home | was contacted to set up a 900 line |
04:43.48 | jeev | lol |
04:44.00 | jeev | who pimps out 900 lines anyway |
04:44.12 | jameswf-home | umm 800 lines |
04:44.33 | jameswf-home | this phone seks powered by Asterisk |
04:45.07 | jameswf-home | oh crap memory leak crash |
04:45.28 | techman97 | that's what you get for hosting fone seks |
04:45.33 | techman97 | dirty memory |
04:45.38 | techman97 | and leaky intertubes |
04:46.15 | jeev | jameswf-home, your e-penis is too small in this channel |
04:46.32 | jameswf-home | tigerdirect has 1TB drives for 149... I hink i shal set up a 1 TB raid |
04:46.56 | jameswf-home | I read somewhere that someone had voipsex with kerry's wife |
04:47.30 | jeev | lol |
04:48.35 | jameswf-home | 167 for "voipsex". (0.25 seconds |
04:48.59 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
04:49.34 | *** join/#asterisk SpaethCo (n=SpaethCo@cache.spaethco.com) |
04:50.33 | drmessano | SIPSEX.com |
04:51.20 | drmessano | I w_n_ t_ _??__ _ou al_ _ve_ oh _?_ |
04:51.44 | trnzmeta | can I buy an e? |
04:51.53 | trnzmeta | oops already there |
04:51.59 | techman97 | I won't take your clothes off while you're drunk? |
04:52.22 | drmessano | That's why phone sex ops don't use SIP |
04:52.45 | drmessano | ooohhh |
04:52.55 | drmessano | I wonder if IAXME.com would take off |
04:52.56 | techman97 | ooooohhhh |
04:53.15 | a1fa | lol |
04:54.20 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
04:58.16 | techman97 | nite all |
04:58.36 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
05:01.25 | a1fa | ah midnight |
05:01.31 | jameswf-home | The price of gold canyon candles are going up |
05:02.18 | jeev | wtf is that |
05:03.06 | jameswf-home | candles sold in house parties... the local sheriff Raided their warehouse and arrested 65 illegals |
05:03.26 | jameswf-home | need a job? |
05:03.27 | jameswf-home | :)) |
05:03.56 | trnzmeta | weeet weeet |
05:06.01 | jeev | huh |
05:06.03 | jeev | i'm lost |
05:06.10 | jeev | we need to arrest illegals and mcbush |
05:11.16 | jameswf-home | look you can put lipstick on a pig http://www.mondaymorningmemo.com/mmm_images/June6_2005MMM.jpg oh wait its still a pig |
05:14.17 | jeev | hahah |
05:15.39 | *** join/#asterisk Bananaskin (n=Banana@user-5af01b01.wfd96.dsl.pol.co.uk) |
05:20.24 | jameswf-home | HOLY CRAP http://www.bunnyranch.com/virgin/ |
05:35.17 | *** join/#asterisk costal79 (n=ivan@203.84.225.2) |
05:35.33 | costal79 | hi people |
05:35.41 | costal79 | I have this group |
05:35.57 | costal79 | GROUPSUPPORT=SIP/620&SIP/624&SIP/621&SIP/622&SIP/623 |
05:36.26 | costal79 | and also I'm doing something like this |
05:36.29 | costal79 | exten => o,1,Dial(${GROUPSUPPORT},20,tTwWr) |
05:36.34 | costal79 | so my question here is |
05:36.50 | costal79 | the order of the group define which extension will get the call first ? |
05:40.31 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-893bd2a09f2fc996) |
05:41.49 | rue_mohr | no |
05:41.54 | rue_mohr | they all ring togethor |
05:42.25 | rue_mohr | costal79, |
05:43.19 | rue_mohr | the only time you might see a difference is if they are all set to autoanswer |
05:43.42 | rue_mohr | but network topology would effect is just as much |
05:43.49 | rue_mohr | I should think |
05:45.31 | rue_mohr | I also think your doing it wrong for a set of support phones, but I dont know how to do any of the rotation type stuff I think you should be doing |
05:46.28 | rue_mohr | I think it involves some lists with for loops and if busys |
05:46.43 | costal79 | mmm |
05:46.59 | costal79 | this is an old asterisk system and I'm trying to do some changes |
05:47.10 | costal79 | they have only this group definition |
05:47.18 | costal79 | and it works like a queue |
05:47.24 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
05:48.18 | *** join/#asterisk jsin (n=jsin@gentoo/developer/jsin) |
05:48.59 | jsin | Can someone please make a soft phone recommendation? |
05:49.23 | mchou | jsin: like don't use one |
05:49.31 | jsin | :) |
05:49.41 | mchou | jsin: not joking, really |
05:49.56 | jsin | how come? |
05:50.22 | mchou | poor audio & inconvienence |
05:50.50 | jsin | I just want to play while I wait on my hard phone to come in the mail |
05:51.05 | rue_mohr | jsin, what os? |
05:51.07 | mchou | that's what they all say |
05:51.17 | jsin | windows |
05:51.21 | rue_mohr | ok |
05:51.26 | mchou | haha. |
05:51.26 | rue_mohr | whast was it called... |
05:51.32 | mchou | MS Messenger :) |
05:51.33 | jsin | I have a GXP2000 on the way |
05:51.38 | rue_mohr | justa sec |
05:51.44 | mchou | xlite? |
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05:51.56 | rue_mohr | no |
05:53.43 | rue_mohr | expresstalk |
05:54.00 | rue_mohr | http://www.nch.com.au/components/talksetup.exe |
05:54.20 | rue_mohr | to configure it.. |
05:54.20 | rue_mohr | click options... |
05:54.20 | rue_mohr | you need to get to "sip setup" |
05:54.20 | rue_mohr | http://i80.photobucket.com/albums/j192/tpad-com/ExpressTalkSIPProvider.jpg |
05:54.20 | rue_mohr | click the "I already have an account" (then next) |
05:54.32 | rue_mohr | the rest is specific to your thing |
05:55.12 | rue_mohr | my experiance is that its 'descent' |
05:55.18 | mchou | lol |
05:55.27 | mchou | descent into hell is right! |
05:55.47 | rue_mohr | 2 hours past my bedtime |
05:55.48 | rue_mohr | gnight |
05:56.09 | rue_mohr | twinkle under linux |
05:59.24 | jsin | thanks rue_mohr, good night |
05:59.40 | jsin | this better than x-line in your experience? |
06:04.20 | drmessano | Gotta love any software that offers you to download a whole effing PBX when registering |
06:04.43 | drmessano | X-Lite is 10x better than ExpressTalk.. that thing gave me daymares |
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06:10.37 | jsin | expresstalk installed and setup quickly... I can make outbound calls, now to figure out how to get incoming calls :) |
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06:18.50 | trnzmeta | can you have skins to x-lite? |
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07:01.20 | r0bis | The funniest thing I discovered about my problem when inbound calls do not get hung up properly (i.e. remote side remains with open line after asterisk issues hangup()). It appears telco problem, since it is exactly reproducible when just an analog phone and nothing else is connected to the line. lol and I think so far the condition has not been detected because its so natural for the other side to hangup after speaking |
07:01.20 | r0bis | <PROTECTED> |
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07:04.37 | johnakabean | hey room i need some help with 1.4 (latest stable) |
07:09.59 | kaldemar | ~ask |
07:10.00 | jbot | [ask] Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
07:11.18 | kaldemar | either jbot scared the living poo out of him or he didn't really need help. |
07:11.23 | MooingLemur | uh huh |
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07:31.06 | [gnubie] | waves to all.. |
07:33.41 | tzafrir_laptop | hi [gnubie] |
07:35.30 | [gnubie] | i am running 1.4.21.2 here.. my ip phone is connected remotely.. if i call another extension phone located inside the lan where the asterisk is located, the callee hears me.. but if i call a telephone connected to a local pstn (pots) via my fxo, the callee cannot hear me at all.. no problem with all incoming calls regardless if it comes from one of the extension phones or from the pstn (pots) |
07:35.40 | [gnubie] | hello tzafrir_laptop |
07:37.12 | tzafrir_laptop | if you call voicemail of playback? (not direct sip<->sip)? |
07:37.23 | [gnubie] | in short, inbound calls to my remote phone is ok but outbound calls from my remote phone, the callee cannot hear a sound |
07:38.11 | [gnubie] | tzafrir_laptop: no problem if the remote phone calls to another extension number or vice versa |
07:39.02 | [gnubie] | calling the echo test is good |
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07:39.55 | [gnubie] | remote_phone => asterisk => pstn(pots) => telephone |
07:39.58 | [gnubie] | that's the problem |
07:40.11 | [gnubie] | but if the other way around, there's no problem at all.. |
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07:43.00 | [gnubie] | i'm using sip here by the way |
07:43.53 | [gnubie] | outbound call from a remote phone produces a one way audio where the callee cannot hear the caller |
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07:44.49 | [gnubie] | any idea? |
07:45.20 | mchou | yeah. it's your firewall |
07:49.00 | [gnubie] | mchou: firewall.. where? on the asterisk box or on where the remote phone is connected? |
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07:51.43 | mchou | whereever there is a firewall needs to be checked |
07:51.51 | [gnubie] | mchou: my firewall rules => http://paste.debian.net/16965/ |
07:51.54 | costal79 | exit |
07:51.58 | costal79 | \q |
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07:52.47 | mchou | I'm not going to debug your issue for you, sorry |
07:53.24 | mchou | one way audio is almost always a firewall traversal issue |
07:54.12 | [gnubie] | i didn't changed anything except that i upgraded my asterisk installation from 1.4.17 to 1.4.21.2 |
07:54.30 | [gnubie] | the configurations are all the same |
07:54.51 | [gnubie] | now, i have this issue already.. |
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08:01.21 | drmessano | ~sipnat |
08:01.22 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
08:01.28 | drmessano | It's not a firewall problem |
08:01.45 | drmessano | Sounds like you need to fix your localnet externhost/externip settings |
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08:10.34 | [gnubie] | drmessano: you mean on my sip.conf ? |
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08:13.47 | [gnubie] | drmessano: as far as i understand, it's good.. and it worked before with the same configuration as i have for asterisk-1.4.17.. |
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08:15.11 | dandre | Hello, |
08:15.24 | dominic1 | if I program a forward in my telephone I get a 302 move temporarily from the device. Is there any possibility to read that statement in the dialplan? |
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08:15.32 | dandre | How can I put a call on hold using the manager interface? |
08:15.53 | dandre | I haven't found an action: hold |
08:17.51 | [gnubie] | besides, my asterisk box is facing the internet already.. eth0 is using a public ip and eth1 is the one that faces to my lan.. my remote phone is located somewhere the internet outside the lan |
08:18.09 | [gnubie] | so when i try to place a call, the route is: |
08:18.43 | [gnubie] | remote_phone => internet => asterisk => pstn(pots) => telephone/callee |
08:19.14 | [gnubie] | sorry, it's not like that.. here is the right one: |
08:20.01 | [gnubie] | remote_phone => lan => modem/router => internet => asterisk => pstn(pots) => telephone/callee |
08:20.34 | [gnubie] | if it's the other way around, it's ok.. like this: |
08:21.01 | [gnubie] | remote_phone <= lan <= modem/router <= internet <= asterisk <= pstn(pots) <= telephone/callee |
08:21.24 | [gnubie] | s/callee/caller |
08:23.13 | dominic1 | hold ist not possible |
08:23.19 | dominic1 | you can park and unpark somebody |
08:23.24 | dominic1 | but there is no command to hold sb |
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08:39.35 | dominic1 | sorry, back again |
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08:59.00 | Rico29 | hi |
08:59.15 | Rico29 | i'm trying to configure zaptel through /etc/zaptel.conf |
08:59.46 | Rico29 | i get this error msg : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
09:00.18 | Rico29 | http://debian.pastebin.com/m7869a72b |
09:01.14 | tzafrir_laptop | Rico29, what is the output of: zaptel_hardware |
09:01.27 | tzafrir_laptop | what device(s) do you actually have? |
09:01.42 | Rico29 | oxygen:/etc# zaptel_hardware |
09:01.45 | Rico29 | nothing |
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09:01.48 | Rico29 | tdm411B |
09:02.20 | Rico29 | 1FXS / 1FXO |
09:03.09 | tzafrir_laptop | do you see that card on lspci ? |
09:03.22 | tzafrir_laptop | that card? or a TigerJet? |
09:03.41 | tzafrir_laptop | What kernel? What linux distribution is it? |
09:03.46 | Rico29 | mmmh |
09:04.02 | Rico29 | my boss told me that pluging the card to the computer's alim was not needed... |
09:04.09 | Rico29 | i think he was wrong |
09:04.13 | Rico29 | debian 4 |
09:04.14 | JT | alim? |
09:04.20 | Rico29 | 2.6.18-6-686 |
09:04.25 | Rico29 | power |
09:04.35 | JT | alim == power, huh? |
09:04.55 | Rico29 | yes there is a slot to plug a power cable on the card |
09:05.00 | JT | you do need to hook it up if you want the fxs port to work |
09:05.06 | Rico29 | like these on HD |
09:05.10 | JT | since when has that been called "alim"? |
09:05.16 | Rico29 | sorry for my bad english... |
09:05.26 | Rico29 | french word sorry |
09:05.28 | Rico29 | :) |
09:05.30 | JT | ok |
09:05.51 | JT | it's a molex connector |
09:06.20 | Rico29 | yes |
09:06.23 | Rico29 | that's it |
09:07.55 | tzafrir_laptop | Rico29, as a hint: dmesg | tail |
09:08.13 | tzafrir_laptop | I think you should see there an error message telling you just that :-) |
09:08.51 | tzafrir_laptop | Zaptel is kernel, so looking at dmesg (or in Debian: better: /var/log/kern.log) can be useful |
09:08.59 | Rico29 | ok |
09:09.00 | Rico29 | thanks |
09:09.18 | Rico29 | i'll see |
09:09.21 | Rico29 | i'm currently rebooting |
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09:31.11 | Rienzilla | hey all |
09:32.06 | Rienzilla | I was wondering whether a linksys voip-phone adapter should be able to carry fax and modem signals? |
09:32.25 | mgdm | faxes and especially modems don't work very well across VoIP |
09:32.41 | Rienzilla | yeah I suspected that |
09:33.16 | Rienzilla | won't it work at all? or just with low bitrates/unreliably? |
09:33.42 | mgdm | Modems I suspect won't work at all, faxes can be made to work, I believe, with a lot fo tweaking |
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09:41.51 | gr0mit | Rienzilla, it is all rather flakey |
09:41.55 | Rienzilla | ok. Thanks. |
09:42.24 | Rienzilla | yeah the point is we switched to voip, but we forgot to take into account some old modem connection to the building's heating system :) |
09:43.02 | Rienzilla | so ill probably have to keep some pots line into the building to operate that modem |
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09:43.58 | tzafrir_laptop | Rienzilla, will it be OK for the modems to work on lower baud rates? |
09:44.09 | Rienzilla | 9600 8n1 is probably what they're talking |
09:44.29 | Rienzilla | it's connected to the serial port of some management system |
09:44.35 | tzafrir_laptop | so this is basically like a fax |
09:44.38 | Rienzilla | I bet it's 9600 baud |
09:44.41 | Rienzilla | yeah |
09:44.48 | Rienzilla | I don't expect 56k6 to work across voip ;) |
09:45.18 | tzafrir_laptop | that's marginal for an ATA. You need a good network |
09:45.52 | Rienzilla | I'm pretty well connected (100mbit fiber to the datacenter where the signal goes onto isdn) |
09:46.14 | WimpMan | What about Serial-Ethernet thingies? |
09:46.22 | tzafrir_laptop | There are some ATAs that support T.38 (fax over IP). But T.140(?) (modem over IP) is hardly supported |
09:46.37 | Rienzilla | hmmkay |
09:46.54 | Rienzilla | Ill keep the analog line for the next few years then, until this is better supported |
09:46.59 | Rienzilla | thanks for the advice |
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11:44.35 | bugotaf | Hi |
11:46.27 | bugotaf | I'm using an digium AA50 and i have some trouble with voicemail, i' have define some user with voice mail with password equal to phone number, when i call voicemail and type phone number then pincode the autorisation is always refused |
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11:46.44 | Specialist1 | hello everyone |
11:47.41 | bugotaf | i have another problem with voicemail when i call internaly my voicemail sound is in french but when outdoor call fall in voicemail voice are english |
11:48.27 | Specialist1 | <PROTECTED> |
11:50.17 | gr0mit | hands bugotaf a dictionary for his external callers ;-) |
11:50.49 | gr0mit | bugotaf, i think you need to set the language for the SIP or IAX peer that handles your incoming calls |
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11:51.49 | heedly | Specialist1: it always doesn't ring? |
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11:52.01 | heedly | sounds like a provisioning thing to me... |
11:52.19 | Specialist1 | hi heedly, actually it rings at times but not always |
11:52.51 | Specialist1 | first I saw "auto congesting " message on the console.. |
11:53.28 | Specialist1 | then I changed 'autoqualify' to 'no' ! to overcome auto congesting. |
11:54.35 | Specialist1 | Before I set 'autoqualify' to 'no' It showed two times ringing tone and after that it said "the number is not avaialble" |
11:55.12 | Specialist1 | Now after I change it to 'yes' It shows ringing on the calling side but acutally it does not ring on the called side!! |
11:57.09 | heedly | maybe a nat problem? |
11:58.36 | Specialist1 | hmm . Do you think so. Is there any way that I can communicate from the server with the client . Like ping or something? |
11:59.12 | Specialist1 | But ping can only tell if the client machien is responding! |
12:04.57 | heedly | there are a number of things you can do to test. |
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12:08.10 | Phob[0S] | hi all |
12:08.13 | Rienzilla | hmm |
12:08.27 | Rienzilla | is there a straightforward solution for cancelling echo in phonecalls? |
12:09.14 | gr0mit | yes - use isdn |
12:09.27 | gr0mit | or anything with 4-wire connections the whole way |
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12:09.30 | Rienzilla | well |
12:09.30 | *** mode/#asterisk [+o russellb] by ChanServ |
12:09.38 | Rienzilla | I don't use any analogue lines |
12:09.41 | gr0mit | ok |
12:09.50 | Rienzilla | and my voip provider does use isdn bundles afaik |
12:09.54 | gr0mit | ok |
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12:10.16 | gr0mit | so if you are using a voip provider then it is them who needs to cancel the echo |
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12:11.08 | Rienzilla | oh |
12:11.22 | Phob[0S] | hi can i ask support for a strange issue? |
12:11.29 | Specialist1 | heedly sorry for the delayed reply... How can i test if its due to nat |
12:11.31 | gr0mit | the echo points will be any 2-4 wire conversion point |
12:11.54 | gr0mit | or a speakerphone or handset with accoustic path from Rx to Tx |
12:12.04 | WimpMan | Phob[0S]: We are unaware of your abilities to do so. |
12:12.20 | Phob[0S] | ??? |
12:12.39 | WimpMan | ~ask |
12:12.40 | jbot | rumour has it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
12:12.40 | Phob[0S] | i have a strange issue with rtp stream |
12:12.46 | Phob[0S] | can we talk in pvt? |
12:13.19 | Great_Anta_Baka | what does silence suppression do? |
12:13.41 | gr0mit | messes up Asterisk, Great_Anta_Baka |
12:13.44 | [TK]D-Fender | Great_Anta_Baka: Stops sending RTP at all when the endpoint decides its too weak to consider conversation |
12:13.58 | Great_Anta_Baka | i see |
12:14.06 | Great_Anta_Baka | so i should disable it? |
12:14.16 | gr0mit | always disables it |
12:14.25 | [TK]D-Fender | Great_Anta_Baka: * doesn't support it, so yes. |
12:14.25 | Great_Anta_Baka | mmm kk |
12:14.50 | kalel008 | how do i stop this from playing http://pastebin.com/m4009fb94 |
12:15.20 | [TK]D-Fender | kalel008: "core show application voicemail" <- |
12:16.14 | Odd_Bloke | Does this stack trace look familiar to anyone: http://paste.pocoo.org/show/85007/ ? |
12:16.28 | Phob[0S] | Ok, i've an asterisk implementation with a Cisco voicegateway. Cisco Voice Gateway had an E1 PRI Isdn. RTP streams for internal calls are setup P2P between phones, but external calls (From Internal Phone to public Network through Cisco) RTP Stream it setup through Asterisk Server |
12:16.47 | Odd_Bloke | I'm about to start debugging an issue with a channel we're writing, but want to check we're not hitting a known problem in Asterisk itself. |
12:17.02 | Phob[0S] | anyone can help me on how to resolve this issue? |
12:18.12 | Phob[0S] | no one? |
12:19.14 | Rienzilla | gr0mit: thanks so far |
12:19.35 | gr0mit | so Rienzilla under what conditions do you get echo? |
12:19.42 | Rienzilla | Im not sure |
12:19.49 | Rienzilla | some calls experience echo, some don't |
12:20.08 | gr0mit | have you raised this with your carrier? |
12:20.17 | Rienzilla | I just have :) |
12:20.25 | Phob[0S] | how can i avoid rtp proxying? |
12:20.27 | gr0mit | my carrier had prbs on calls to French landlines |
12:20.27 | Rienzilla | I was just wondering whether this could be my own ault |
12:20.43 | gr0mit | so i juest send calls via a different carrier and they are fine |
12:21.23 | gr0mit | are these calls within .nl or different countries? |
12:21.50 | Rienzilla | almost exclusively .nl |
12:22.01 | Rienzilla | it's customers of a dental practice, so hardly any international calls |
12:22.10 | gr0mit | ah ok |
12:22.18 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
12:22.34 | gr0mit | can you try another carrier? |
12:22.55 | Rienzilla | not for incoming calls |
12:23.00 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
12:23.10 | Rienzilla | I could set something up for outgoing calls, maybe |
12:23.18 | gr0mit | yup |
12:23.36 | Rienzilla | but I'll first wait to see what th ecarrier says |
12:23.47 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
12:23.53 | Rienzilla | I know they just received new incoming lines from another phone company |
12:24.03 | Rienzilla | so maybe it's a new problem which they dont know yet |
12:24.06 | gr0mit | yup |
12:24.12 | gr0mit | which carrier is this? |
12:24.16 | Rienzilla | speakup.nl |
12:24.41 | gr0mit | just looking |
12:27.29 | gr0mit | not the cheapest, Rienzilla |
12:27.55 | gr0mit | so not a bargain basement company! |
12:28.12 | Rienzilla | I know the company owners personally |
12:28.36 | Rienzilla | but they have a quite ok service |
12:28.38 | gr0mit | ok, well what I did was to run a tcpdump on my asterisk box |
12:28.45 | gr0mit | log the traffic |
12:28.52 | gr0mit | and send it to my wholesaler |
12:28.59 | Great_Anta_Baka | is there a tool i can use to analyze the route to another pbx to check if placing a sip call over it is viable? |
12:29.13 | gr0mit | when they heard the echo they said ' oh yes, that is bad' |
12:29.40 | Rienzilla | Well I'll have my users write down the phone numbers they have problems with |
12:29.51 | gr0mit | yup - date time and destination |
12:30.01 | gr0mit | then they can look through the logs |
12:30.04 | Rienzilla | yep |
12:30.07 | Rienzilla | k |
12:30.17 | gr0mit | I have had issues with calling DECT phones |
12:30.34 | gr0mit | coz they introduce latency and echo |
12:30.40 | jblack | They work fine for me. |
12:30.57 | jblack | It's probably the particular model of phone you're using. |
12:30.59 | gr0mit | just some dect phones on some lines |
12:31.12 | gr0mit | most as you say are perfect |
12:31.16 | Rienzilla | I have voip phones exclusively |
12:31.31 | Rienzilla | (and all the same models, too) |
12:31.40 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
12:31.53 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
12:32.12 | darkskiez | how can i find out what Pickup() is expecting.. |
12:32.15 | Rienzilla | and the weird thing is, I hav used the same voip provider at home, with the same model phone, and not experienced any problem |
12:32.25 | Rienzilla | so it might just be their new outgoing lines |
12:32.40 | darkskiez | i'm trying with the text displayed under location col in show channels |
12:32.46 | gr0mit | Rienzilla, could be anything really |
12:33.11 | jblack | It could be anything. Perhaps a bad ground in one phone. Maybe a little space alien has set up camp in one phone. |
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12:33.42 | gr0mit | Rienzilla, best get them to look into it |
12:34.43 | gr0mit | Rienzilla, if you really don;t get any joy let me know - might be able to help. |
12:37.51 | Rienzilla | ok, thanks so far |
12:39.10 | gr0mit | graag gedaan. |
12:40.52 | [TK]D-Fender | [08:28]<Great_Anta_Baka>is there a tool i can use to analyze the route to another pbx to check if placing a sip call over it is viable? <- Dial |
12:42.12 | Phob[0S] | anyone can help my on my rtp trouble? |
12:42.28 | Phob[0S] | i need to know if i can solve my problem |
12:43.06 | Phob[0S] | ?? |
12:43.56 | Phob[0S] | c'mon guys i only would to avoid rtp stream pass through my asterisk PBX |
12:44.21 | Phob[0S] | this happens only in particular cases |
12:44.52 | Rico29 | with canreinvite=yes ? |
12:45.07 | Phob[0S] | hi rico |
12:45.10 | Phob[0S] | thnx for your reply |
12:45.25 | Rico29 | you're welcome |
12:45.29 | Phob[0S] | i've already set it to yes |
12:45.39 | Phob[0S] | i can esply to you from beginning |
12:45.52 | Rico29 | sorry i've a lot of work |
12:45.56 | Phob[0S] | ok |
12:45.59 | Phob[0S] | tnx |
12:46.00 | Rico29 | i'm at the office |
12:46.02 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
12:46.06 | Phob[0S] | np |
12:46.51 | lmadsen | I can't seem to remember where this is enabled, but there is an option I thought that would display 'hint' information to the manager interface? |
12:46.58 | lmadsen | I can't seem to find it in sip.conf though |
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12:47.28 | WimpMan | lmadsen: That's 1.6 only, isn't it? |
12:47.43 | lmadsen | WimpMan: that still doesn't answer my question :) |
12:47.49 | lmadsen | uses all versions of asterisk |
12:47.49 | Phob[0S] | i've resolved my issue |
12:48.12 | Phob[0S] | this happens cause i passed 'tr' option in Dial application in my dialout macros |
12:48.19 | Phob[0S] | tnx to all for the supports |
12:48.25 | Phob[0S] | Google it's your friend |
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12:58.10 | Great_Anta_Baka | how can i calculate how big my jitterbuffer should be? |
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13:01.46 | [TK]D-Fender | lmadsen: "core show hints" |
13:01.56 | lmadsen | thanks |
13:01.58 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
13:02.10 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
13:02.23 | WimpMan | Didn't I read manager? |
13:02.46 | lmadsen | WimpMan: I talked to a developer, and what I just requested does not exist in asterisk, but I'm gonna attempt to fix that... |
13:02.52 | [TK]D-Fender | WimpMan: Yes, but you didn't use your imagination :) |
13:02.54 | dominic1 | can anybody tell me, why I am not able to set the callerid of a number from my isdn pstn and not always show the network number? |
13:03.05 | [TK]D-Fender | lmadsen: *I* do it this way |
13:03.25 | lmadsen | I'm not gonna parse through the output of the CLI if I can help it |
13:03.28 | lmadsen | hacks |
13:03.33 | lmadsen | coughs |
13:03.41 | [TK]D-Fender | lmadsen: And have for years. my call center Polycom Idle app mointors 2 queues, 2 VM boxes, and 4 presence entries |
13:03.53 | [TK]D-Fender | lmadsen: Parse it! |
13:04.22 | [TK]D-Fender | lmadsen: If you don't like brute force, you clearly haven't found a big enough stick! |
13:04.24 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
13:04.41 | [TK]D-Fender | has an even BIGGER stick, and I use ti too! *WHAM* |
13:05.42 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-149.dsl.lsan03.sbcglobal.net) |
13:05.53 | Alton2 | Postel, a famous name in computing. |
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13:12.44 | dominic1 | I want to transfer the ddi of my isdn pri. The main msn is 15450. I set the callerid of my outgoing call to 1545123 but the called party only sees 15450. Is there any option, which I will have to adjust? |
13:13.54 | *** join/#asterisk Xaviertoor (n=xavierto@189-015-91-171.xd-dynamic.ctbcnetsuper.com.br) |
13:14.36 | *** join/#asterisk nacer (n=God@l.alcolo.a.mpl.pastIX.net) |
13:14.40 | nacer | hi |
13:14.50 | nacer | i need a information about digium card :) |
13:14.55 | WimpMan | dominic1: Check the type of number. |
13:15.04 | lmadsen | dominic1: sounds like no, as the provider is restricting your number |
13:15.08 | nacer | i need to know the exact size of the eax 2400 card |
13:15.17 | lmadsen | 42! |
13:15.24 | nacer | :) |
13:15.27 | Great_Anta_Baka | check out the specifications page |
13:15.30 | Great_Anta_Baka | on the digium site |
13:15.31 | lmadsen | nacer: your best bet is to just call digium sales -- they'll tell you |
13:15.33 | WimpMan | dominic1: Might be you have to set the area code as well, or set the correct TON. |
13:15.34 | lmadsen | or that :) |
13:15.52 | nacer | thereis no informtion about this in the documentation |
13:16.12 | Great_Anta_Baka | normally the specifications pag eincludes all of that crap |
13:16.20 | nacer | yeahh normally |
13:16.22 | Great_Anta_Baka | otherwise call digium |
13:16.33 | nacer | dam my english speaking is bad |
13:16.56 | lmadsen | your english seemed fine -- just the spelling, but even native english speakers seem to have that problem :) |
13:17.48 | lmadsen | and the only word I actually see wrong is s/dam/damn :) |
13:18.18 | dominic1 | WimpMan: TON? |
13:22.59 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
13:22.59 | WimpMan | type of number |
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13:29.50 | WimpMan | dominic1: -> msg |
13:30.14 | dominic1 | ah okay, I understand |
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13:32.11 | *** mode/#asterisk [+o mog] by ChanServ |
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13:39.27 | sapere | Hey guys, anyone familiar with the Asterisk service provider, IronVoice.net? |
13:39.48 | Rienzilla | hmm is it possible with an asterisk queue to put "important" clients in the front of the queue? |
13:40.04 | sapere | make that IronVoice.com |
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13:42.31 | lmadsen | Rienzilla: that is a very good question. I had someone ask me that recently, let me see if I can find any configuration option or variable that allows that -- but I have a feeling it may not be possible |
13:42.42 | lmadsen | unless there is a weighting system I can find that I don't know about |
13:43.07 | Rienzilla | I cant find it in the Queue command documentation |
13:43.30 | lmadsen | ya, I think it'd have to be a channel variable or something since you have to apply it to the channel before it enters the queue |
13:44.56 | *** join/#asterisk moy (n=moy@nat/ibm/x-94b8c591a6d565b4) |
13:45.46 | lmadsen | Rienzilla: ${QUEUE_PRIO} Queue priority |
13:45.56 | lmadsen | Rienzilla: looking for more docs on it -- but that might be the ticket |
13:45.59 | [TK]D-Fender | lol.... pre 1.0 stuff only discovered now! |
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13:46.26 | lmadsen | oh my god.... JTODD! |
13:47.16 | nacer | lmadsen: the hotline give the information |
13:47.17 | Rienzilla | ahh |
13:47.42 | lmadsen | nacer: the rooster flies at midnight |
13:48.01 | nacer | lmadsen: i dont understand :) |
13:48.09 | lmadsen | nacer: me either :) |
13:48.15 | [TK]D-Fender | gives lmadsen the secret handshake |
13:48.30 | lmadsen | I don't want to know your name! |
13:48.31 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
13:48.39 | Rienzilla | http://lists.digium.com/pipermail/asterisk-users/2006-May/150398.html |
13:48.39 | [TK]D-Fender | lmadsen: I jsut want... |
13:48.41 | Rienzilla | looks like it |
13:48.42 | lmadsen | ! ! ! |
13:48.46 | [TK]D-Fender | lmadsen: zomg! |
13:48.54 | lmadsen | lawlz |
13:49.39 | lmadsen | Rienzilla: I will have to test. It may be possible that variable doesn't work and needs to be fixed. If that is the case, then I will file a bug -- unless you get to it before me :) |
13:50.09 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
13:50.17 | Rienzilla | i'll try now |
13:51.47 | Rienzilla | hmm |
13:51.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:52.01 | Rienzilla | my queue is not large enough to see where they are queued at the moment :) |
13:52.47 | jeev | aaaaaaaaaa |
13:52.56 | Rienzilla | how do you enable debug output for queue? |
13:56.20 | jeev | leef |
13:58.10 | l2trace99 | leaf | life | loof whats the difference |
13:58.11 | l2trace99 | ? |
13:58.42 | jeev | dunno |
13:58.43 | jeev | i'm tired |
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14:00.11 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-118-239.pools.arcor-ip.net) |
14:01.05 | Blackvel | hi all. found GotoIf cmd. I am in the need to manually change to callerid for outgoing call depending on various callerid strings. Is there an simple If/else statement too? where could I look at? |
14:02.50 | Kobaz | Blackvel: nope, you're stuck with BASIC-style syntax unless you go with ael |
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14:03.20 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
14:03.49 | l2trace99 | or use agi |
14:04.28 | Kobaz | i like the freeswitch method better for language integration |
14:05.04 | [TK]D-Fender | Blackvel: How many different values? What are the calls coming in from? |
14:05.41 | Blackvel | probably 4 (so far I can think for) |
14:05.42 | Blackvel | http://radio-downtown.de/voip-info/wiki/view/Asterisk+func+if.html |
14:06.04 | Blackvel | hmm no if/else :) |
14:06.50 | [TK]D-Fender | Blackvel: go read IF's **instructions** |
14:07.00 | [TK]D-Fender | Blackvel: You've clearly missed the big print |
14:09.32 | Blackvel | [TK]D-Fender: What are the calls coming in from? how do you mean that? the calls come in as sip from pstn. but the the callerid can have at least 4 different numbers (when I set clip no screening on my mobile phone I have to completely change the incoming callerid otherwise clip is not working with the p |
14:09.40 | Blackvel | with the sip provider... |
14:10.49 | [TK]D-Fender | Blackvel: Ok, so in from a non "local phone", and then right out another channel. Then 4 IF's are in order |
14:12.54 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
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14:15.22 | Blackvel | too bad func_logic.so is in 1.4, not 1.2 :) and asterisk-1.4.21.2 segfaults with patton bri gateway. |
14:15.43 | heedly | just copy iy over! |
14:15.47 | heedly | and hope it works.. |
14:16.47 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
14:17.00 | Blackvel | i dont need func_logic.so in 1.2? |
14:17.13 | [TK]D-Fender | total waste. 4 little "IF"s |
14:17.13 | Blackvel | i mean necessarily need func_logic.so? |
14:17.20 | Blackvel | I see |
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14:17.23 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
14:18.09 | Blackvel | I feel lost so many times. I always miss the entry point to find the stuff with * I need (like if) |
14:19.12 | [TK]D-Fender | Blackvel: "core show applications" , "core show application [appwithoutbraces]" , "core show functions" , "core show function [functionwithoutbraces]" <- all you need. |
14:19.50 | [TK]D-Fender | Blackvel: So before you ask "how do I XXXXX", go look at the list first. |
14:20.22 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
14:21.57 | Blackvel | very good tip with show functions. looks like I was missing that :( |
14:25.34 | Rienzilla | hmm, anyone ever used a linksys pap2 ata thingy? My analogue device insists the line is off hook if I plug it into the ata |
14:26.09 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
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14:31.39 | gr0mit | Rienzilla, yup. |
14:31.49 | gr0mit | i have instaled one at a customer site |
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14:35.22 | [TK]D-Fender | Rienzilla: It'll do that if it isn't registered |
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14:36.53 | gabegundy | I have a client that wants CentOS/Asterisk/FreePBX and wants RPMs for everything. I'd love to build it for him, but from what I understand, AsteriskNOW will provide all of that. Does anyone know if that is true? |
14:37.10 | Rienzilla | [TK]D-Fender: the registration stat eis online though |
14:37.16 | Rienzilla | according to the web interface |
14:37.29 | gabegundy | The next version of AsteriskNOW that is. |
14:38.47 | [TK]D-Fender | gabegundy: I've heard as much myself |
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14:39.26 | gr0mit | Rienzilla, you have checked the pins are correct? |
14:39.51 | gr0mit | if you plug a phone in, do you get dialtone? |
14:40.18 | gabegundy | What are the chances someone in this channel can confirm that and save this guys some money (even if it means I get less billable hours)? |
14:40.56 | Rienzilla | yes, a phone will give a dialtone |
14:41.00 | Rienzilla | and the wiring must be correct |
14:41.14 | Rienzilla | it's just standard rj11 and rj45 cabling |
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14:44.49 | Rienzilla | (if I plug a phone to the fax wall outlet which is connected to the ata I also get a dialtone) |
14:45.22 | gr0mit | ok |
14:45.46 | gr0mit | I wonder if you can increase the line voltage in the pap2 |
14:46.18 | Rienzilla | I can increase the ring voltage |
14:46.30 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
14:47.10 | Rienzilla | and I read something similar (same problem on a different device) where a solution was to increase a CPC value, but I don't know what that is :) |
14:47.20 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
14:48.07 | rwaite | hi all. i am scanning my server and port 1720 is open ... i looked for a module i could disable to close this port but i dont know what could be using it? |
14:48.21 | Rienzilla | rwaite: are you using a linux system? |
14:48.53 | lmadsen | Blackvel: you could use the IF() function too |
14:49.08 | lmadsen | may be way out of date now though as he didn't read the entire scrollback |
14:49.23 | Rienzilla | rwaite: (if so, run netstat -tapn | grep LISTEN | grep 1720 to find the process which is listening on port 1720 |
14:49.24 | gr0mit | CPC is calliing party clear. should not affect line voltage detection |
14:49.41 | Rienzilla | rwaite: or lsof -i -n | grep 1720 |
14:50.13 | rwaite | ahah |
14:50.15 | rwaite | thanks |
14:50.15 | [TK]D-Fender | lmadsen: ClueBat action has been missed... |
14:50.22 | Rienzilla | so probabnly my device is thinking the line is open because the voltage on the wires is different than it would expect in an on-hook situation? |
14:50.30 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
14:51.00 | jameswf-home | so asterisknow is debian? |
14:51.26 | rwaite | Rienzilla: hmm. i get nothing for that, but nmap still lists it as open. maybe i'm missing something |
14:51.49 | rwaite | it says "filtered" |
14:51.52 | Rienzilla | ah :) |
14:52.00 | Rienzilla | filtered means a firewall is blocking that port |
14:52.21 | Rienzilla | (so a connect to the port does not yield "connection refused", but it yields nothing) |
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14:53.25 | rwaite | what would cause this? |
14:53.34 | rwaite | (no firewall on this machine, yet) |
14:54.21 | [TK]D-Fender | jameswf-home: rPath currently |
14:54.40 | jameswf-home | rpath uses apt? |
14:54.40 | Rienzilla | rwaite: some firewall between the machine you scanned from, and the machine you scanned |
14:54.51 | [TK]D-Fender | jameswf-home: conary.. |
14:54.52 | rwaite | ok |
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14:54.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:55.46 | jameswf-home | ah must be old instructions |
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14:59.05 | weinerk | Hello, please help - why do I have a line repeating over and over in the log |
14:59.18 | weinerk | [Sep 11 02:12:47] VERBOSE[26681] logger.c: -- <SIP/ZDt92ruS96-b7800468> Playing 'vm-extension' (language 'en') |
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15:01.37 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
15:02.00 | pjz | anyone have recommendations on firmware version to use with Polycom IP330s ? |
15:02.51 | pjz | 3.0.3B (since it's the latest) ? |
15:02.56 | Rienzilla | gr0mit: I don't think the pap2t can change the line voltage |
15:03.17 | gr0mit | hmmm oh dear |
15:03.35 | *** join/#asterisk rdgr (n=rich@82.32.1.139) |
15:04.11 | x86 | anyone have any thoughts on the Linksys SPA921 phone? |
15:05.54 | [TK]D-Fender | x86: FFS WHY!?! :p |
15:06.56 | [TK]D-Fender | x86: Polycom IP 320 kicks its ass for the same price. |
15:08.27 | gr0mit | [TK]D-Fender, so should i get hold of one of these Polyjuice phones? |
15:09.00 | [TK]D-Fender | gr0mit: If you are looking for a good SIP phone, Polycom is it. |
15:09.52 | gr0mit | they look better than the snom 300 |
15:11.05 | rwaite | the the asterisk .X releases fixes for severe bugs that couldnt wait until the next .XX release to fix? |
15:11.11 | [TK]D-Fender | gr0mit: Very little point to Snom in my books. To pricey for normal/high models, too lacking in features on lower, too unstable. Linksys is my #2 choice. |
15:11.26 | [TK]D-Fender | rwaite: 4th position, yes |
15:11.36 | ManxPower | The Polycom 300s are pretty ugly, but the other models (320, etc) are quite nice. |
15:11.43 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
15:11.54 | gr0mit | well i like the busy lamps on the snom phones. |
15:11.58 | rwaite | that's what i thought. thanks. |
15:12.00 | [TK]D-Fender | ManxPower: 30x = never buy > $50 |
15:12.04 | gr0mit | have not seen this on an others |
15:12.20 | ManxPower | [TK]D-Fender: my customer has many many 301s |
15:12.23 | gr0mit | do polycom do any thing like that? |
15:12.25 | ManxPower | ..er.. 300s, that is |
15:12.39 | [TK]D-Fender | gr0mit: Thats 1 thing Snosm is good for on the middle models... but its a very specific segment that profits from the # it has based on model |
15:12.42 | *** join/#asterisk thansen|laptop (n=thansen@c-67-172-249-106.hsd1.ut.comcast.net) |
15:12.56 | [TK]D-Fender | ManxPower: I just got my first 2 IP320's here. They look nice |
15:13.00 | ManxPower | gr0mit: Polycom allows you to set any line appearance as a BLF. Polycom 60x has a side car available, 14 line appearances per sidecar, up to 3 side cars |
15:13.20 | ManxPower | [TK]D-Fender: my customer moved to the 320s for new phones |
15:13.30 | [TK]D-Fender | ManxPower: cool |
15:14.43 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
15:16.58 | *** join/#asterisk wacky__ (n=root@nat/digium/x-ae82c3d51cf8a4f2) |
15:17.42 | wacky__ | with Asterisk 1.6, when it run "core show translation", I get some 1-digit values, and some 5-digits values.. |
15:18.01 | wacky__ | is there a cache somewhere to clear ? anything to update to get all 4 to 5-digits micro-seconds numbers ? |
15:18.59 | ManxPower | wacky__: show translation has some options to help |
15:19.16 | ManxPower | I believe it's a "recalc" option |
15:20.16 | wacky__ | oh wow.. |
15:20.24 | wacky__ | that's neat! thanks |
15:21.01 | ManxPower | BTW, thanks to the person that sent me a paypal donation! |
15:21.13 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
15:23.07 | x86 | [TK]D-Fender: I've had nothing but problems with the IP330's |
15:23.20 | x86 | [TK]D-Fender: 330's and 430's both have serious flaws |
15:23.23 | ManxPower | x86: then you set them up wrong. 8-) |
15:23.29 | x86 | [TK]D-Fender: where the IP301 and IP501 work fine |
15:23.37 | x86 | ManxPower: not according to Polycom's engineer |
15:24.01 | ManxPower | We had massive problems with the 320s. Until we updated the sip.cfg and phone1.cfg to add the required options for those phones. |
15:24.19 | ManxPower | volume problems, the icons next to the line appearance buttons not working. |
15:25.00 | x86 | the problem we had with the 330's and 430 is that they would start to boot, and then just lock up |
15:25.16 | x86 | some times they would sit there locked up and never do anything, sometimes they would go into an infinite reboot |
15:25.30 | x86 | Polycom couldn't figure the issue out |
15:25.41 | ManxPower | x86: we never had those problems and we have a fair number of 320s and 330s |
15:25.43 | x86 | not had a single problem with 301's, 501's, nor 601's |
15:26.01 | x86 | ManxPower: *shrugs* |
15:26.19 | x86 | ManxPower: the network cabling here is cat4 (*ugh*), so that might have something to do with it |
15:26.30 | x86 | ManxPower: but the 301, 501, and 601 have no problems with that at all |
15:29.28 | ManxPower | x86: are you *sure* your sip.cfg and phone1.cfg files support the new options for the 320/330? |
15:30.41 | [TK]D-Fender | x86: Neither myself nor any of my clients ever have problems with any of them. |
15:31.42 | pjz | anyone have recommendations on firmware version to use with Polycom IP330s ? 3.0.3revB ? |
15:32.45 | pjz | x86: I'm having the infinite reboot problem with my phones right now, but I thought it was due to upgrading my asterisk server |
15:33.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:35.40 | ManxPower | I'll look at what version we use. |
15:35.52 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
15:38.04 | [TK]D-Fender | pjz: I would switch to the latest 3.X release and if that causes issues, the latest 2.2.X release |
15:38.18 | Rienzilla | gr0mit: would changing settings for FXS Port impedance, FXS input gain and FXS output gain help anything? |
15:38.22 | Rienzilla | could* |
15:38.23 | ManxPower | Polycom SIP firmware 2.1.1.0052 |
15:38.34 | ManxPower | (according to the phone menu) |
15:38.38 | pjz | ManxPower: on a 330? |
15:38.50 | ManxPower | pjz: we only have one firmware version. |
15:39.04 | ManxPower | IIRC 2.1 was what version supported 330/320/430 |
15:40.18 | ManxPower | You would have to check the release notes to be *sure*. |
15:40.53 | ManxPower | We tend not to change things unless they are broken |
15:40.56 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
15:43.26 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
15:46.39 | Rienzilla | aargh this is driving me nuts :) |
15:47.58 | Specialist1 | I want to send all my outgoing traffic to a H323 server . do i need to create an H323 trunk for that on my server ? |
15:48.57 | gr0mit | Rienzilla, nope |
15:50.09 | *** join/#asterisk weinerk (n=irc@82.80.129.212) |
15:50.53 | ManxPower | Specialist1: not many people here use H323. Which H323 channel are you using in Asterisk? |
15:51.53 | *** join/#asterisk ManxPower (n=manxpowe@40.sub-70-221-156.myvzw.com) |
15:52.11 | ManxPower | stupid verizon |
15:52.49 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
15:52.53 | Specialist1 | actually i am using a sip server but i want to forward my traffic to the h323 sever |
15:53.35 | *** join/#asterisk scurb (i=scurb@94.191.130.55.bredband.tre.se) |
15:57.51 | ManxPower | Specialist1: Best of luck. As I said, virtually nobody here uses H323 |
16:00.00 | Blackvel | oh my dear, I wish I had choosen external Java AGI for this ivr thing. how can I avoid that 4x IF...SET gets executed line after line (even on 1st match)? is there any way to combine the IF with GOTOIF? like "if .. else if .. else if .. else if ..." |
16:00.39 | aliver | I'm using an ITSP (bandwidth.com). In my sip.conf they want me to have "fromuser=+1234567890" where the number there is one of my DIDs they handle for me. If I comment out the line, I think the "callerid=Some Young Guy <1234567890>" is sent to them instead and they reject the call because they only want the number in it's 11 digit form. |
16:01.00 | aliver | Does that sound unlikely. Mainly I want to know, does fromuser == callerid more or less? |
16:01.15 | aliver | ie.. is callerid sent as "fromuser" when there is no "fromuser" ? |
16:02.45 | [TK]D-Fender | Blackvel: FFS its jsut 4 lines of dialplan, stop complicating it |
16:03.16 | lmadsen | Blackvel: sounds like you want to use a case statement in AEL (which is just a set of GotoIf()'s |
16:04.55 | lmadsen | GotoIf($["${foo}" = "${bar}"]?true:${IF($["${jim}" = "${beam}"]?true:${IF($["${jack}" = "${daniels}"]?true:false)})})}) |
16:05.04 | lmadsen | if you really wanna make it hard and put it all on one line |
16:06.04 | lmadsen | so yes, there IS a way -- but even *I* don't do that, and I love nesting functions |
16:06.36 | Blackvel | so you guys dont care in normal extensions.conf when the four ifs get executed one by one |
16:06.42 | lmadsen | and actually I have one too many sets of closing I believe |
16:06.59 | lmadsen | Blackvel: explain again simply what you're trying to solve |
16:07.10 | lmadsen | you're trying to execute 4 GotoIf()'s at the same time? That's what it sounds like... |
16:07.20 | lmadsen | because obviously we aren't getting it |
16:07.22 | Knightfal | can I limit the amount of calls I recieve to an inbound route on a per day basis |
16:07.27 | [TK]D-Fender | lmadsen: Even that is more complex than it deserves |
16:07.34 | Knightfal | I have a client that has 100 calls per day in thier contract and they are sending me 300+ |
16:07.35 | lmadsen | Knightfal: yes you can |
16:07.52 | Knightfal | Can you point me in the right direction |
16:07.57 | lmadsen | Knightfal: combination of GROUP(), GROUP_COUNT() and STRFTIME() I would think |
16:08.14 | lmadsen | or STRFTIME and DB() perhaps |
16:08.33 | Blackvel | lmadsen: sorry I dont get it to describe in English :) |
16:08.45 | x86 | lmadsen: group and group count only count the number of calls currently in use in that group, right? they don't do cumulative totals over a period of time, I thought |
16:09.02 | lmadsen | x86: yes, that's why I realized using a DB() call is more appropriate :) |
16:09.09 | x86 | :) |
16:09.18 | lmadsen | thx though for making that point obvious |
16:09.20 | sapere | Hey guys -- Asterisk/VOIP newb question here: Asterisk-based appliances provide what benefits that a simple analog VoIP adapter doesn't? |
16:09.29 | lmadsen | Blackvel: then good luck! :) |
16:09.29 | x86 | or MYSQL() that ties right in with your billing system ;) |
16:09.45 | lmadsen | yep, something like that :) |
16:09.48 | Blackvel | 4 lines with SET cmd combined with prior IF. when I have 4 lines, the if gets called 4 times (which you normally try to avoid in programming languages) |
16:10.28 | lmadsen | if you normally try to avoid it in programming languages, why would you do it in asterisk? |
16:10.44 | Blackvel | lmadsen: [TK]D-Fender tries to tell me that I should program those 4 if/set lines and shut up :) |
16:11.00 | lmadsen | I still don't get what you're trying to do |
16:11.05 | Blackvel | exactly |
16:11.35 | Blackvel | forget it. maybe I can explain it in 10 years when my English progresses :) |
16:11.53 | lmadsen | I don't think its an english problem -- I think it is a logic problem |
16:12.22 | sapere | Do I even need an analog adapter if I receive service over ethernet? |
16:12.48 | sapere | ah, wait ... nvm |
16:12.51 | lmadsen | IF($[expression]?true:else) |
16:12.54 | sapere | analog phones, not analog incoming |
16:14.18 | lmadsen | [10:00] <Blackvel> hi all. found GotoIf cmd. I am in the need to manually change to callerid for outgoing call depending on various callerid strings. Is there an simple If/else statement too? where could I look at? |
16:14.35 | lmadsen | Blackvel: do you mean based on the incoming callerID string, you need to change the callerID string going outbound? |
16:15.26 | lmadsen | exten => _NXXNXXXXXX/4165551212,1,Set(CALLERID(num)=8002223334) |
16:15.46 | lmadsen | exten => _NXXNXXXXXX/5195915119,1,Set(CALLERID(num)=8665551212) |
16:16.11 | ManxPower | Blackvel: See "ex-girlfriend option" in the Asterisk docs |
16:16.14 | lmadsen | exten => _NXXNXXXXXX/5195915119,n,Goto(standard_stuff,1) |
16:16.49 | lmadsen | exten => standard_stuff,1,NoOp(This is where all major logic happens, regardless of who called in -- you've already set the CID for the outgoing call) |
16:16.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
16:18.10 | *** join/#asterisk enno (n=enno@lightsource.verbrennung.org) |
16:19.40 | enno | "Extension can never match" <-- i tried to find out where in the externsion.conf to place an i or _X rule for testing first isdn dialin. Any hints? Google didnt help |
16:20.21 | ManxPower | enno: "i" is only for IVRs |
16:20.32 | ManxPower | you would place it in the context your ISDN dialing ports are located |
16:20.44 | ManxPower | where "it" is _X. |
16:20.57 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
16:22.39 | enno | ManxPower: thanks |
16:25.32 | *** join/#asterisk jsin_ (n=jsin@gentoo/developer/jsin) |
16:26.58 | *** join/#asterisk StormD (n=stormd@64.3.54.171.ptr.us.xo.net) |
16:27.29 | StormD | how do I determine what version I'm running? |
16:27.55 | [TK]D-Fender | StormD: "show version" |
16:29.00 | lmadsen | "core show version" |
16:29.01 | StormD | so I think I did something stupid, running version 1.2.23 |
16:29.15 | lmadsen | heh |
16:29.18 | [TK]D-Fender | StormD: Thats a self-fulfilling statement :) |
16:30.18 | StormD | Thank you for the vote of confidence in my self-appraisal. |
16:31.31 | StormD | anyway, what happened is, a couple months ago, one of my clients demanded I enable voice mail on their system which had apparently been set up and configured without any voice mail. |
16:32.31 | StormD | Figuring that was a completely retarded way to set up a voip system, I went ahead, read a few docs, and set up basic voice mail, which worked pretty well until two weeks ago, when it became apparent why voice mail was never set up in the first place, when the 10gig hard drive that the system was running on completely overflowed. |
16:34.15 | StormD | So I went through /var/spool/asterisk/voicemail/<group>/<boxes>/INBOX/ and deleted a bunch of messages to clear up a little space, and I also created some room by deleting /var/log/asterisk/messages.0 |
16:34.23 | StormD | I think that last part was retarded of me, correct? |
16:35.06 | [TK]D-Fender | StormD: no, thats fine |
16:35.12 | l2trace99 | it should be a log file by the path of it |
16:35.39 | StormD | now, several users can't log into their mailboxes...they dial into their mailbox, enter their password, and then the system just disconnects them |
16:36.40 | l2trace99 | do the users show up in 'show voicemail users' ? |
16:37.17 | StormD | yes. |
16:37.23 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
16:38.06 | StormD | oh hmmm... |
16:38.56 | StormD | Being connected to the console when logging into the maibox is a valuble troubleshooting tool, innit? |
16:39.21 | l2trace99 | does the same user that owns the asterisk process owns the /var/spool/asterisk dir |
16:39.22 | l2trace99 | yes |
16:39.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
16:40.09 | StormD | app.c:1232 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/<group>/<user>/Old': File exists |
16:40.54 | StormD | well, root owns that directory |
16:41.04 | [TK]D-Fender | StormD: Did you delete the txt files AND all of the recordings to match? |
16:41.05 | l2trace99 | and asterisk is running as ? |
16:41.17 | StormD | root |
16:41.47 | StormD | Fender: yeah I pretty much scoured their boxes clean |
16:41.51 | *** join/#asterisk arpu (n=arpu@chello080109017114.12.14.vie.surfer.at) |
16:43.31 | Kobaz | wow, our main asterisk hub just crashed |
16:43.44 | [TK]D-Fender | StormD: copy their recordings, delete the entire vm folder, leave a new VM to reinit the box, then copy the recordings back |
16:44.19 | StormD | but when I was cleaning the "Old" directory, I think I just ran ls, and didn't 'ls -al' and missed that .lock file which I now see in there and just cleared out. |
16:44.25 | StormD | testing again |
16:44.29 | Kobaz | [Sep 11 12:37:59] WARNING[20140] res_features.c: Bridge failed on channels IAX2/AlbanyIn-5 and Zap/1-1, res = -1 |
16:44.42 | Kobaz | that's the last message in the log before asterisk crashed |
16:44.43 | Kobaz | weird |
16:45.04 | StormD | oh that totally fixed it. |
16:45.17 | *** join/#asterisk bmg505 (n=leon@196-209-8-66-ndn-esr-2.dynamic.isadsl.co.za) |
16:46.17 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
16:46.24 | *** join/#asterisk newvn (n=newvn@70.252.57.249) |
16:47.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
16:47.23 | StormD | Now, the main thing I need to do is install a bigger hard drive on this damn computer to mount on /var so that this doesn't happen again in another month. |
16:52.13 | *** join/#asterisk jtodd (i=fn8t2q0n@ns2.loligo.com) |
16:52.34 | StormD | so /var/log/asterisk/messages, messages.0, messages.1 just cascading records of voice-mails gone by....not necessary for functioning of system? |
16:53.28 | [TK]D-Fender | StormD: Feel free to trash them |
16:53.29 | StormD | I notice those files get huge, being voice-recordings. |
16:53.45 | StormD | and I only have a 10G drive for the whole system here. |
16:58.35 | CrashSys | Use mp3 |
17:02.22 | [TK]D-Fender | ... |
17:02.30 | [TK]D-Fender | ~cluebat CrashSys |
17:02.31 | jbot | ACTION pulls out a ClueBat (tm) and thwaps CrashSys. |
17:02.39 | CrashSys | Thank you sir may I have another |
17:02.41 | CrashSys | err, sup? |
17:03.34 | StormD | I'll look into that, but the thing is, installing a hard drive and mounting a partition is something I know how to do in less than an hour and only costs my client $100 or so. Tracking down documentation on converting voice-mails to utilize some compression I'm not sure how to use in a system I'm only barely familiar with, and teaching myself how to implement that is... ??? on my schedule and billing sheet. |
17:04.10 | CrashSys | lol :) |
17:04.21 | mort_gib | StormD: you are better off doing both anyway, if you have enough CPU power |
17:04.33 | mort_gib | And it's not really difficult |
17:05.00 | [TK]D-Fender | ~cluebat CrashSys |
17:05.00 | jbot | ACTION pulls out a ClueBat (tm) and thwaps CrashSys. |
17:05.02 | CrashSys | Hard-drive is the easier option |
17:05.09 | [TK]D-Fender | StormD: the answer you're looking for is "more" :) |
17:05.12 | CrashSys | plus 10-gig's is pretty miniscule |
17:05.15 | *** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk) |
17:05.23 | CrashSys | by today's standards anywyas |
17:05.35 | bkw_ | 10 gig of voicemail? |
17:05.39 | mort_gib | Yes, how did you manage to find a 10GB drive?? |
17:05.43 | [TK]D-Fender | My cell phone has 4, and thats because I got it free :) |
17:05.50 | bkw_ | aha |
17:06.14 | [TK]D-Fender | \o/ WinMo6.1 upgrade |
17:06.20 | [TK]D-Fender | (pending) |
17:06.45 | StormD | I have no idea...this system came with the client. One of their IT-guys who no longer works for them since before they hired me set it up. |
17:07.29 | StormD | The weird thing is, it's a Dell that looks like it's 2 years old at the max....I have no idea how it shipped with such a tiny drive. |
17:07.42 | StormD | I suspect some hanky-panky might have gone on. |
17:08.27 | StormD | in fact, I think I'm going to look up the Service Tag on Dell's site and see what size drive it actually shipped with. |
17:09.04 | Kobaz | maybe the drive failed and a crappy old spare was quickly found |
17:09.17 | Kobaz | or it came with a 500gig drive and someone jacked it |
17:09.18 | StormD | bkw: 10 gig of total hard disk...that's OS AND voice mail. |
17:09.46 | StormD | all mounted on a single partition, naturaly |
17:09.52 | Kobaz | StormD: 10 gig isn't too shabby for a small linux system, i used to run linux on like 100 meg drives |
17:11.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
17:12.07 | *** join/#asterisk Levonk (n=lk@adsl-75-62-140-156.dsl.lsan03.sbcglobal.net) |
17:12.24 | StormD | Kobaz: oh, I know...I used to run Red Hat on a 800MB drive when I was in college. But I wasn't trying to record and serve voice-mail for an office, either. |
17:13.00 | Kobaz | heh |
17:13.19 | *** join/#asterisk sapere (n=sapere@209-249-12-72.ip.openhosting.com) |
17:13.27 | coppice | I remember doing voicemail on a 10M drive |
17:13.53 | StormD | you get 1 30 second message |
17:14.05 | mvanbaak | it all depends on the ammount of users and the count of messages |
17:14.26 | mvanbaak | oh, and the codec ;) |
17:14.31 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
17:15.00 | StormD | anyway, thanks for the help, I gotta jet. |
17:16.18 | sapere | afternoon all, quick question about minimum hardware required for an IP phone small business asterisk setup: I'd need a computer to run my asterisk server, a switch, and an IP phone. That's it correct? (other than the ITSP service) |
17:16.46 | sapere | well, and cat5 cables between the PC & switch, and switch & phone(s) |
17:17.21 | [TK]D-Fender | sapere: Yup, thats about it. |
17:17.46 | mort_gib | sapere: and some time on your hands |
17:17.47 | sapere | thanks, [TK]D-Fender |
17:17.55 | [TK]D-Fender | sapere: You could technically do without the switch by adding more NIC's to your server, but who wants to do all that? :) |
17:17.59 | [TK]D-Fender | (except me) |
17:18.39 | [TK]D-Fender | loves his everything-and-the-kitchen-sink-home-server |
17:18.42 | sapere | If I wanted to run something like Elastix or Trixbox, instead of Asterisk directly, am I really gaining anything? |
17:18.52 | [TK]D-Fender | sapere: Yes, our wrath :) |
17:19.01 | mort_gib | sapere: flaming from TK |
17:19.03 | [TK]D-Fender | sapere: Free with every install! |
17:19.45 | [TK]D-Fender | sapere: GUI's don't do much of anything you can't do yourself and force you to do things based on their method of thinking. |
17:19.53 | errr | Im having some issues with an ivr.. http://paste2.org/p/72794 I have pasted the error Im getting here |
17:19.57 | sapere | hah -- i can't find any real value they provide, other than claims that make me skeptical (trixbox claims to "lockdown" security in asterisk) |
17:20.06 | mvanbaak | [TK]D-Fender: you forget that vim is also a gui |
17:20.18 | mort_gib | sapere: If you do have issues, doing it ground up will prepare you better... |
17:20.42 | [TK]D-Fender | mvanbaak: Real programmers use SOLDER |
17:20.51 | mvanbaak | or ed |
17:20.53 | sapere | [TK]D-Fender : are they strictly a GUI? If they're only a GUI, then FreePBX is also in that group |
17:21.03 | mort_gib | Or rather, WHEN you have issues, HAVING done it ground up prepares you better |
17:21.20 | mvanbaak | sapere: freepbx is not a GUI, it's a virus |
17:22.04 | sapere | and I only need FXS/FXO to connect to analog phones or analog phone lines, correct |
17:22.09 | [TK]D-Fender | sapere: Those distros you mentioned all USE FreePBX and is the core problem. |
17:22.17 | [TK]D-Fender | sapere: Correct. |
17:22.31 | [TK]D-Fender | sapere: for that I'd suggest a Linksys SPA-3102 if for home use |
17:22.54 | sapere | The cheapest ITSP i've found is Broadvoice's BYOD service at $6/mo for 100 minutes, any opinions on them? |
17:23.06 | mvanbaak | those SPA thingies are that good ? |
17:23.50 | jaytee | I just got a 3102 in today. haven't set it up yet |
17:24.10 | *** join/#asterisk carpenike (n=ircap@82.ecb7d1.client.atlantech.net) |
17:24.16 | sapere | [TK]D-Fender : but I wouldn't need a Linksys SPA-3102 if I'm doing IP phones and no FXO/FXS interface |
17:24.29 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
17:24.29 | [TK]D-Fender | sapere: thats $.06/min with SUCKS |
17:24.39 | [TK]D-Fender | sapere: Yes, but you ASKED |
17:24.39 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
17:24.56 | [TK]D-Fender | mvanbaak: Not awesom for FXO, but good enough for home. |
17:24.57 | sapere | oh, sorry I misunderstood your response |
17:25.09 | carpenike | Hey anybody know much about faxing inside 1.6? |
17:25.13 | mvanbaak | [TK]D-Fender: ah. good to know |
17:25.15 | [TK]D-Fender | mvanbaak: and the FXS is Linksys regular. Excellent value overall |
17:25.28 | CrashSys | I pay $0.0089/min |
17:25.43 | CrashSys | only have to spend about $2000/mo to get it too :) |
17:25.44 | sapere | [TK]D-Fender : that price is outgoing, I should mention. Free incoming |
17:25.45 | [TK]D-Fender | I pay $0.000/min:) |
17:25.53 | CrashSys | DAMN |
17:25.57 | [TK]D-Fender | sapere: bleh |
17:26.08 | mvanbaak | I just want a device that will reroute calls going from $outside to local analog line to sip/iax/skinny |
17:26.19 | [TK]D-Fender | sapere: could still do better probably. You need to reall calculate your usage when shopping around. |
17:26.25 | mvanbaak | so ppl can keep their loved/hated analog line and still experiment with voip |
17:26.31 | [TK]D-Fender | mvanbaak: It'd do the job OK |
17:26.46 | [TK]D-Fender | mvanbaak: its ESPECAILLY good because of the power failover |
17:27.07 | mvanbaak | yeah, I read that |
17:27.17 | carpenike | Hey anybody know much about faxing inside 1.6? |
17:27.59 | *** join/#asterisk moy (n=moy@nat/ibm/x-cbcb7d536591e364) |
17:28.46 | carpenike | Specifically, I'm tring to figure out if it's possible to do a Email to Fax gateway using the new termination/origination built into Asterisk 1.6 all locally installed... And if so, what I actually need installed to do so... |
17:29.00 | carpenike | I'd like to get a T.38 provider and pass the faxing SIP/IAX2. |
17:32.48 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
17:33.08 | sapere | Do you guys generally recommend AsteriskNOW over Asterisk as a separate server? In my case I have no need to reuse my server for any other purpose |
17:33.58 | heedly | sapere: most people in here will recommend compiling asterisk from source. |
17:34.13 | heedly | and using a distro you are comfortable with. |
17:34.21 | sapere | fair enough, thanks |
17:34.50 | carpenike | Gentoo. :-D |
17:35.28 | *** join/#asterisk makkksimal (n=makkksim@f054120109.adsl.alicedsl.de) |
17:35.48 | errr | isnt asterisknow going to start to come with freepbx? |
17:36.07 | heedly | most modern linux distributions include ways to build from source. |
17:36.16 | heedly | gentoo is the only one that requires it ;) |
17:36.33 | heedly | slap an "R" on it goes fffffaaaast |
17:36.56 | carpenike | I just got Gentoo + OpenVZ + Asterisk 1.6 Source + Freepbx to all play together. :) |
17:37.08 | carpenike | Inside of VMWare. |
17:39.22 | heedly | a VM inside a VM huh |
17:39.32 | heedly | my reality is shriking man |
17:39.54 | carpenike | Heh... Goal is to put it into ESX. |
17:40.42 | carpenike | And use the same VM to run additional isolated services like mail. |
17:43.54 | *** join/#asterisk ohayden (i=ohayden@adhd.irule.net) |
17:44.40 | CrashSys | carpenike: How did you get a reliable hardware timer through openVZ? |
17:45.39 | carpenike | installed Zaptel inside of the HN and gave the client access via vzctl set --devnodes |
17:45.56 | CrashSys | what's zttest report? |
17:47.16 | Blackvel | how can I call functions on a dialplan? within {} or something? |
17:47.32 | Blackvel | like IF(expr? Set(....)) |
17:47.34 | *** join/#asterisk makkksimal (n=makkksim@92.224.50.208) |
17:47.45 | Blackvel | it interpretes my if as cmd |
17:47.47 | carpenike | openvz ~ # zttest |
17:47.48 | carpenike | Opened pseudo zap interface, measuring accuracy... |
17:47.48 | carpenike | 99.963379% 99.926758% 99.987793% 99.926758% 99.987793% 99.902344% 99.926758% |
17:47.48 | carpenike | 99.987793% 99.938965% 99.987793% 99.902344% 99.963379% 99.914551% 99.914551% 99.975586% |
17:48.07 | CrashSys | impressive... at idle load? |
17:48.27 | carpenike | yeah. |
17:48.47 | CrashSys | So you are running VMWare inside OpenVZ on top of Gentoo? |
17:49.06 | [TK]D-Fender | Blackvel: Go read the basics on functions and variables on the WIKI.... |
17:49.15 | carpenike | Gentoo Host --> VMWare Server --> Gentoo VM --> OpenVZ --> Asterisk |
17:49.16 | jeev | as fender always says: reading is fundamental |
17:49.23 | carpenike | Err |
17:49.26 | carpenike | Gentoo VM is OpenVZ. |
17:49.46 | CrashSys | is the Gentoo/VMWare just for testing or actual production set-up? |
17:50.07 | CrashSys | In other words, is Gentoo/VZ the actual platform as what you would do in production? |
17:50.16 | carpenike | For my house yeah. |
17:50.21 | carpenike | At work... |
17:50.36 | carpenike | I'd do ESX -> GentooOpenVZ --> Asterisk |
17:50.52 | CrashSys | Hmmm, ok... |
17:51.13 | carpenike | And use the same Gentoo OpenVZ VM to run a few other VPSes. |
17:52.04 | carpenike | --- Results after 313 passes --- |
17:52.04 | carpenike | Best: 100.000000 -- Worst: 92.480469 -- Average: 99.764670 |
17:52.25 | CrashSys | ehhh... 92 = hell for IAX/MeetMe |
17:52.43 | carpenike | Yeah. |
17:52.58 | CrashSys | I have IAX/MeetMe dependencies... |
17:53.52 | carpenike | Hmm.. What's the main draw in hosting your own conference room and not going with a hosted provider IE: freeconferencecall.com? Aside from branding and I can see it if everybody is internal to the office or dialing in through VPNs... |
17:54.01 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
17:54.39 | x86 | anyone know of a good classic rock icecast stream that I could stream via an Asterisk extension? |
17:54.57 | Blackvel | have a good evening |
17:57.04 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
17:58.23 | *** join/#asterisk ManxPower (n=manxpowe@231.sub-70-222-214.myvzw.com) |
17:58.47 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
17:58.47 | *** mode/#asterisk [+o russellb] by ChanServ |
17:59.13 | Kobaz | x86: you could stream shoutcast/icecast via mplayer, and pipe the output to music on hold |
17:59.58 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
18:01.50 | *** join/#asterisk soulfreshner (n=root@dsl-241-171-94.telkomadsl.co.za) |
18:02.31 | soulfreshner | I'm struggling to get my tdm400 card working |
18:02.40 | soulfreshner | I compiled the drivers |
18:03.42 | soulfreshner | but when I run ztcfg -vv I get "failed on channel 1: No such device or address (6) |
18:03.56 | Qwell | did you load the drivers? |
18:04.25 | soulfreshner | yep - modprobe zaptel; modprobe wctdm |
18:05.16 | *** join/#asterisk skr1p7 (n=toor@line111-36.adsl.kirov.ru) |
18:05.55 | Yourname` | Hi, is there a way to 1) Login an agent via AMI 2) Transfer a call via AMI? |
18:05.57 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
18:06.12 | [TK]D-Fender | Yourname`: Yes, Yes, and BOOK <- |
18:06.46 | skr1p7 | hi all. how to fix that: "system('asterisk -rx "dialplan reload"');" return "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" ? |
18:06.54 | skr1p7 | srwx---r-x 1 asterisk asterisk 0 Sep 7 00:43 /var/run/asterisk/asterisk.ctl |
18:07.01 | skr1p7 | ^ file is exist |
18:07.31 | ManxPower | skr1p7: remove the ' |
18:07.35 | ManxPower | both of the ' actually |
18:07.43 | skr1p7 | where? |
18:07.48 | ManxPower | I doubt that's the problem, but it's a good idea. |
18:07.50 | Yourname` | [TK]D-Fender: Which book? |
18:08.01 | ManxPower | system('asterisk -rx "dialplan reload"') <-- remove the 's |
18:08.02 | soulfreshner | I get another error at startup as well: wcopenpci:[00] Firmware version 0 not supported by this driver contact Voicetronix to have it updated |
18:08.03 | [TK]D-Fender | ~book |
18:08.04 | jbot | methinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
18:08.29 | skr1p7 | run sudo -u www asterisk -rx "dialplan reload" return same error in shell |
18:08.41 | ManxPower | soulfreshner: edit /etc/sysconfig/zaptel remove the drivers you don't need |
18:08.51 | ManxPower | skr1p7: but that was not your question |
18:09.12 | ManxPower | as you can see the file is owned by ASTERISK but you are trying to access it as WWW |
18:09.44 | *** join/#asterisk UtopiahGHML (n=rrrec@lns-bzn-33-82-252-24-232.adsl.proxad.net) |
18:09.50 | skr1p7 | yeah. need own by www? i think if chmod o+rx is good :) |
18:10.02 | soulfreshner | ManxPower, I don't have /etc/sysconfig/zaptel (using ubuntu) |
18:10.27 | *** join/#asterisk makkksimal (n=makkksim@92.224.50.208) |
18:11.19 | skr1p7 | ManxPower: thx. chown to www fix that problem |
18:11.56 | ManxPower | skr1p7: not going to help in the long run. |
18:12.25 | ManxPower | That file is re-created every time Asterisk starts. |
18:12.32 | *** join/#asterisk Bananaskin (n=Banana@user-5af01b01.wfd96.dsl.pol.co.uk) |
18:12.44 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
18:13.23 | skr1p7 | ok. i will add chown command to rc.d script :) |
18:14.24 | ManxPower | skr1p7: that is the wrong way to do it. |
18:14.30 | ManxPower | Why don't you fix the actual problem? |
18:14.59 | ManxPower | The problem being you want to run Asterisk as user "www", but Asterisk is configured to run as user "asterisk" |
18:15.25 | skr1p7 | run asterisk as www? that not good fix too |
18:15.40 | ManxPower | skr1p7: no, that is the CORRECT fix. |
18:18.30 | Yourname` | [TK]D-Fender: The QueueAdd, does that matter if it's AgentLogin()? |
18:18.36 | Yourname` | ..in the dialplan I mean |
18:19.50 | [TK]D-Fender | Yourname`: And I've answered this many times, agentlogin has NOTHING to do with dynamic members |
18:20.23 | Yourname` | Ok, cool. |
18:20.26 | Yourname` | God bless you. |
18:20.45 | Kobaz | we don't want blessings, we just want food and beer |
18:20.45 | ManxPower | god has nothing to do with it |
18:21.38 | Yourname` | Well, [TK]D-Fender needs 'em, ok? |
18:21.56 | ManxPower | I doubt that. |
18:22.02 | Yourname` | With the blessings of your employer, your requisite for food and beer will be fulfilled. |
18:22.28 | Katty | herroes. |
18:22.32 | ManxPower | most tech people don't seem to believe in those superstitions. |
18:23.08 | Yourname` | Of an employer fulfilling your food/beer needs? |
18:23.44 | ManxPower | no. of some super powerful being that watches over people. |
18:23.59 | ManxPower | kind of like Santa Claus or the Easter Bunny |
18:25.10 | *** join/#asterisk imcdona (i=imcdona@2001:5c0:8fff:fffe:0:0:0:b917) |
18:25.10 | Yourname` | Ah, I don't know tech world you live in Mr. Power, but a couple engineers here who happen to be the geekiest of techs have little Hindu shrines on their desks. |
18:25.11 | scooby2 | damn this issue happened again today. 5 calls waiting to be answered, 3 people logged in but idle, asterisk thinks everyone is busy. |
18:25.26 | Yourname` | scooby2: autofill=yes? |
18:25.28 | ManxPower | scooby2: tell your agents to stop tranfering people |
18:25.54 | ManxPower | IIRC an agent that transfers a call is still considered in use. |
18:26.18 | scooby2 | Yourname`: no autofill |
18:26.41 | scooby2 | not sure about transfering but I doubt all three agents transferred someone |
18:27.45 | scooby2 | just moved to 1.4.21.2 looks like something to try |
18:28.53 | scooby2 | Yourname`: thanks, I will try it now |
18:28.56 | carpenike | Hey does anybody know when the HP Proliant dl380 G4s were introduced? |
18:29.04 | ManxPower | scooby2: moved to 1.4.12.2 FROM what version |
18:30.05 | scooby2 | 1.2.15 |
18:30.27 | ManxPower | scooby2: Did you read upgrade.txt and upgrade-1.2.txt? |
18:30.52 | scooby2 | yeah |
18:30.58 | MindTheGap_ | hello all, i get this: |
18:31.02 | MindTheGap_ | <PROTECTED> |
18:31.03 | MindTheGap_ | <PROTECTED> |
18:31.03 | MindTheGap_ | Segmentation fault |
18:31.20 | carpenike | Could try moving the modules directory to a backup and re-installing |
18:31.44 | MindTheGap_ | when registering a peer w md5secret |
18:31.51 | MindTheGap_ | asterisk 1.6.beta9 |
18:32.23 | russellb | update to a 1.6.0-rc |
18:32.31 | russellb | downloads.digium.com/pub/telephony/asterisk |
18:32.35 | russellb | up to rc6 now |
18:33.01 | russellb | and also make sure you empty out /usr/lib/asterisk/modules when upgrading |
18:34.21 | Kobaz | i always build with a prefix |
18:34.37 | Kobaz | ./configure --prefix=/opt/asterisk-1.6.0rc6 |
18:34.48 | Kobaz | and then you just make symlinks |
18:35.30 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
18:35.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
18:37.06 | Kobaz | so if i wanna switch asterisk versions, i just change one symlink |
18:39.21 | *** join/#asterisk makkksimal (n=makkksim@92.224.50.208) |
18:39.29 | thedonvaughn | Kobaz: doh, so simple i feel dumb for not doing that myself. I run vicidialer with a few asterisk servers here. I've gotten in trouble with vici with an upgrade. Changing with symlinks woulda been the smooth. |
18:41.04 | Kobaz | heh |
18:41.28 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
18:46.27 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
18:46.41 | x86 | Kobaz: or you can use an icecast stream directly as MoH ;) |
18:46.49 | x86 | Kobaz: I was looking for suggestions on a stream |
18:47.28 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
18:48.54 | Kobaz | x86: oh hmm, dunno |
18:48.58 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
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19:02.04 | jameswf-home | my MOH is leekspin |
19:12.53 | *** join/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
19:16.26 | MindTheGap_ | russelb, thanks... just upgraded to rc6 and it wont crash anymore but the password wont match. Realmed password should match peer@realm:password (from memory may not be it) yes? but i need to match it to just a password, no realm no peer, just password. i should use MD5Secret? im using MD5Secret but it fails authentication. |
19:17.27 | MindTheGap_ | btw, i'm authenticating against an ldap server |
19:17.39 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
19:18.01 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
19:18.12 | MindTheGap_ | s/russelb/russellb/ |
19:18.32 | russellb | i do not recall |
19:19.25 | *** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk) |
19:20.05 | rgsteele||work | So, I'm constructing a dialplan in which 011 gets stripped from the number, but only if that 011 is present. My original attempt stripped it all the time, so I worked up this to circumvent that problem, but was hoping to have some eyes more experienced with AEL give the yay or nay before I test it early tomorrow morning before anyone gets to the office: exten => s,n,Set(${NUMTODIAL}=${IF($["${MAC |
19:20.07 | rgsteele||work | RO_EXTEN}" : "(....)"] = "011")?${MACRO_EXTEN:3}:${MACRO_EXTEN}) |
19:20.24 | rgsteele||work | Bah, formatting. Let me try that again: exten => s,n,Set(${NUMTODIAL}=${IF($["${MACRO_EXTEN}" : "(....)"] = "8011")?${MACRO_EXTEN:4}:${MACRO_EXTEN:1}) |
19:20.28 | MindTheGap_ | russellb, i asked about asterisk crashing using MD5Secret, you told me to upgrade do rc6. i did upgrade and it wont crash anymore but the password wont match. Realmed password should match peer@realm:password (from memory may not be it) yes? but i need to match it to just a password, no realm no peer, just password. i should use MD5Secret? im using MD5Secret but it fails authentication. |
19:21.06 | heedly | rgsteele||work: why not just write two seperate extens that match the different formats? |
19:21.23 | heedly | that just looks confusing... |
19:22.22 | rgsteele||work | heedly: I suppose I could, I just thought about doing it in one go initially. |
19:22.49 | [TK]D-Fender | rgsteele||work: Close, but you are not setting a varibale properly there... |
19:23.09 | rgsteele||work | [TK]D-Fender: That's the part I was worried about :) |
19:23.33 | rgsteele||work | I was pretty sure the ternary statment was okay (though certainly I could be wrong about that too!) |
19:24.00 | [TK]D-Fender | exten => s,n,Set(${NUMTODIAL}= <-- already bad |
19:24.35 | rgsteele||work | Ergh, yeah, that should be ...Set(NUMTODIAL=....) |
19:25.14 | rgsteele||work | Which leaves me with: exten => s,n,Set(NUMTODIAL=${IF($["${MACRO_EXTEN}" : "(....)"] = "8011")?${MACRO_EXTEN:4}:${MACRO_EXTEN:1}) |
19:26.55 | *** join/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
19:26.57 | rgsteele||work | [TK]D-Fender: Thanks for the once-over - I hate chasing down silly things like that really early before those first two cups of coffee have kicked in :) |
19:27.35 | *** join/#asterisk nr4q (n=dave@24-183-225-98.dhcp.kgpt.tn.charter.com) |
19:28.51 | nr4q | scratching my head over this. I think it's replated to the dialplan on the phone. with a polycom phone, handset on phone type a 3 digit extension (103), press end and it dials. handset off hook dial 103 and it sends 10 to asterisk. default dialplan on the phone and in sip.cfg |
19:29.50 | nr4q | only thing i can think of is that the phone sees the 0 in "103" and things i'm trying to call the operator "0" and automatically sends "10" |
19:29.51 | [hC] | So I got an email from polycom about a polycom/digium-created wifi voip phone. is this anything new, or is this the same phone weve known about for ages that polycom had out? |
19:30.07 | nr4q | hc: oh that sounds cool |
19:30.44 | *** join/#asterisk Levonk (n=lk@adsl-76-238-250-64.dsl.lsan03.sbcglobal.net) |
19:30.52 | nr4q | hc: do you have a link? |
19:31.13 | [hC] | not yet |
19:31.37 | [TK]D-Fender | [hC]: Ages ago Polycom bought out Spectralink |
19:32.04 | nr4q | are there any decent voip 802.11g phones these days? i know a few years ago i heard that wasn't the case |
19:32.12 | rgsteele||work | Haven't the VoIP over WiFi phones been out for a bit, though? |
19:32.20 | rgsteele||work | (Quality aside) |
19:33.41 | [hC] | [TK]D-Fender: yeah. I know, I just got this email from a dude at polycom, so it seemed like maybe something new happened. turned out its just a spectralink 8002 |
19:33.56 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
19:33.57 | [hC] | http://www.polycom.com/usa/en/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html |
19:34.45 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
19:35.58 | jameswf-home | [hC]: that doesnt mention digium as a partner just that it is ABE Certified |
19:38.14 | [hC] | jameswf-home: my email said "Polycom and Digium join forces to bring an enterprise grade VoWifi wireless solution to the SMB market segment!!!!!!!!!" |
19:38.41 | MindTheGap_ | nr4q, we got some nokias e65 and e51. quality is decent but they cant handle re-registration (when phone looses wifi coverage) properly sometimes. |
19:39.34 | nr4q | i have a FXS card for a tdm400. i thought the best solution may be to just use an analog phone on that FXS port |
19:39.38 | sapere | Asterisk The Future of Telephony speaks highly of the Linksys SPA942, has anyone had any negative experiences with one? |
19:39.58 | jameswf-home | [hC]: i see nothing newer than 07 in the googlesphere so maybe hasnt hit the blogs |
19:40.50 | [hC] | jameswf-home: its probably just announcing ABE certification.. |
19:40.59 | jameswf-home | nothing on russellb's blog either |
19:41.05 | [TK]D-Fender | sapere: Forget Linksys, go Polycom. www.telephonydepot.com |
19:41.22 | nr4q | i'm impressed with my polycom phones |
19:42.08 | nr4q | I just wish I weren't clueless |
19:42.15 | russellb | what what what? |
19:42.30 | WimpMan | sapere: They try to use the in the company I used to work for but ar very dissatisfied with speech quality. I don;t knlow however what else might be wrong there nowadays. |
19:42.52 | russellb | [hC]: I got that email, too. I lol'd at its sillyness level. |
19:43.02 | WimpMan | I just disliked the UI, but then I haven't seen a voip phone with a UI I liked, so far. |
19:43.06 | jameswf-home | OpenR2 in Asterisk - MFC/R2 Free of Headaches or Your Money Back << funny :) |
19:43.17 | [hC] | russellb: I actually had to check wether it was from a "real person" |
19:43.20 | russellb | VoWifi wireless solution from the department of redundancy department!!!!!111111111oneone |
19:43.34 | voxter | damnit i need to change my xchat config. |
19:43.41 | russellb | voxter: was it? |
19:43.51 | bkruse | voxter: I remember those phones coming into Digium |
19:43.54 | voxter | russellb: @polycom.com address. |
19:44.03 | russellb | voxter: didn't look very ... um ... well, it didn't look like something that polycom would do |
19:44.22 | voxter | russellb: no kidding. i bet that guy got in shit at lunch. :) |
19:44.27 | jameswf-home | i must only be on the asia spammer list :( |
19:44.28 | russellb | i hope he did |
19:44.30 | voxter | "Bill Newman" |
19:44.40 | russellb | headers say it came from polycom in atlanta ... |
19:44.48 | [TK]D-Fender | WimpMan: For user interface I haven't a complaint about Polycom.... |
19:44.49 | voxter | never heard of him, but i dont exactly pay attention to polycom's employees either. |
19:44.59 | russellb | polycom is a big company .. |
19:45.54 | jameswf-home | Bill Newman - Manager, Dect Pre-Sales, Americas |
19:47.01 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
19:47.25 | voxter | has multicast/rtp paging made its way into asterisk yet? |
19:47.32 | voxter | i would love to stop using unicast Page()... |
19:49.26 | Katty | anyone use exim4 and mutt a lot? i'm having some email problems i can't seem to figure out. |
19:50.36 | tzafrir_laptop | mutt and postfix here... |
19:50.48 | voxter | crazy. people still use mutt. |
19:50.55 | Yourname` | I do. |
19:51.02 | *** join/#asterisk soulfreshner (n=D@dsl-241-171-94.telkomadsl.co.za) |
19:51.08 | Katty | it's easy to test with. |
19:51.29 | Katty | i'm just not sure why emails aren't sending, and i'm not sure which logs to check |
19:52.16 | soulfreshner | at last I got the tdm400 card working - I had to uncomment #define TDM_REVH_MATCHALL in zconfig.h ... |
19:52.17 | voxter | no idea about your system, but if you're sending from it, id start with /var/log/mail.log, as a suggestion |
19:52.17 | voxter | that may not exist on your system. |
19:52.22 | *** join/#asterisk wacky__ (n=root@nat/digium/x-4763810caeefbe6b) |
19:52.42 | soulfreshner | something to do with the PCI subvendor ID being out of date |
19:53.09 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
19:53.26 | Katty | well that's a good place to check i guess. |
19:53.26 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
19:53.30 | soulfreshner | now what - how do I set up hunting lines for outgoing calls? |
19:54.10 | [TK]D-Fender | soulfreshner: "group=" in zapata.conf |
19:54.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
19:54.45 | [TK]D-Fender | soulfreshner: Dial(Zap/g1/1234567890) , etc for example |
19:54.59 | soulfreshner | thanks, [TK]D-Fender |
19:55.04 | Katty | voxter: mailling to remote domains is not support, claims eximlog |
19:55.14 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
19:55.25 | *** join/#asterisk phl4kx (n=supervis@200.48.200.66) |
19:55.31 | phl4kx | hi for all |
19:55.35 | phl4kx | I call digium asterisk support |
19:55.45 | phl4kx | but the system say I dont have registered products |
19:55.50 | voxter | Katty: so your exim is misconfigured. there's a howto on the google. |
19:55.57 | phl4kx | Im register the product in digium |
19:56.26 | Katty | voxter: mmhmm, yeah. i'm rerunning dpkg reconfigure |
19:58.08 | russellb | phl4kx: hmm? what product do you have? |
19:58.17 | phl4kx | AEX 800P |
19:58.39 | russellb | where did you get it from? |
19:59.07 | sapere | If I have only one phone line and I want to add fax functionality, I'm kinda screwed right? I mean, since fax machines don't know how to enter an extension... |
19:59.08 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
19:59.12 | *** join/#asterisk CamargoBP (n=BP@jive.fttp.xmission.com) |
19:59.47 | jaytee | [TK]D-Fender, do you think it might be possible with Asterisk and an SPA3102 to have the line for the FXS port always off-hook and in an on-hold condition with MOH playing? |
19:59.57 | CamargoBP | Anyone know a good tool to monitor Jitter? |
20:00.05 | CamargoBP | Hopefully that is opensource. |
20:01.04 | [TK]D-Fender | jaytee: Umm.... why? |
20:01.15 | voxter | CamargoBP: using just icmp, you can use mtr. |
20:01.23 | Katty | oh |
20:01.27 | CamargoBP | That's what we do right now |
20:01.34 | [TK]D-Fender | Katty: Mew. |
20:01.37 | Katty | something just came to me about our stupid exchange reverse dns lookup server |
20:01.44 | jaytee | I'm trying to jury rig something with the paging PA system I told you about using Valcom's equipment. it has a separate audio source connection for music. |
20:01.45 | Katty | it's floating about my head, like dust particles |
20:01.47 | CamargoBP | voxter: I'm trying to make a nice web interface or maybe a plugin for Zenos |
20:01.48 | CamargoBP | s |
20:01.50 | Katty | [TK]D-Fender: ohai |
20:01.53 | Katty | [TK]D-Fender: howrechu |
20:02.25 | jameswf-home | http://www.hanlongtek.com/pro_2_4.html <<looks strangely like a GS Handytone |
20:03.24 | [TK]D-Fender | Katty: gesundheit :) |
20:03.43 | Katty | voxter: ahahahahahahaha |
20:03.44 | voxter | CamargoBP: well jitter is really just large deviation between round trip measurements... so however youd want to capture that. but its something that you can only determine by monitoring over time |
20:03.48 | Katty | voxter: my exhcnage server was dumping it |
20:03.53 | Katty | voxter: cause reverse DNS FAILED |
20:04.18 | jaytee | jameswf-home, best way to tell if it's a Handytone on the inside with a different case is just make a call with it and say hello and see if you hear yourself saying hello back. :-) |
20:04.29 | CamargoBP | voxter: When you talk time does that mean secons, minutes, hours or days? |
20:04.32 | Katty | [TK]D-Fender: well come on then |
20:04.34 | Katty | [TK]D-Fender: how're you? |
20:04.38 | Katty | [TK]D-Fender: mister gesundheit |
20:04.42 | CamargoBP | I've seen some solutions where you can see real time jitter |
20:04.58 | CamargoBP | Maybe they aren't really real time |
20:05.08 | jaytee | jitter should be in milliseconds |
20:05.23 | phl4kx | russellb: |
20:05.25 | phl4kx | Im from peru |
20:05.46 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
20:06.19 | n3hxs | jitter is sometimes caused by too much coffee. |
20:06.51 | CamargoBP | n3hxs: lol |
20:07.12 | CamargoBP | I'll get all my customers to stop drinking coffee |
20:07.34 | n3hxs | Everclear will help clear things up. |
20:07.45 | n3hxs | At least they won't care as much. |
20:08.07 | [TK]D-Fender | Katty: Getting by, a piece at a time. |
20:08.22 | CamargoBP | You bet send them all a 40 and hope they don't complain |
20:08.40 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
20:10.11 | CamargoBP | Would you say encapsulating all the VOIP traffic in TCP rather than UDP will help with jitter? |
20:10.25 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-230-174.balt.east.verizon.net) |
20:11.45 | SteveTotaro | hey guys, i want to do a dial on the h exten using asterisk 1.2 has anybody done this, with a regular dial, it tries and "Cancels" I believe because there is only one channel at that point |
20:12.20 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
20:12.20 | SteveTotaro | I am trying to use local channels but I get caught in a loop and I lose the variables that were set in a macro |
20:14.29 | [TK]D-Fender | SteveTotaro: No, there is no channel at all. The call is DEAD |
20:15.08 | *** join/#asterisk xacatecas (n=jkroon@196.46.173.152) |
20:15.46 | xacatecas | hi all, hope all is well. i'm new to the isdn arena and was hoping someone can point me at some good resources. |
20:15.59 | *** join/#asterisk bbryant (n=Brett_Br@adsl-153-53-11.chs.bellsouth.net) |
20:16.08 | soulfreshner | is there a simple sample dialplan out there? |
20:16.24 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
20:16.33 | soulfreshner | the one included with the asterisk installation is a bit over the top for me |
20:16.53 | [TK]D-Fender | soulfreshner:... |
20:16.56 | [TK]D-Fender | ~jerjerguide |
20:16.56 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
20:16.57 | xacatecas | i'd like to learn a little bit about the technology, specifically, i've got a potential client that claims he's got 10 BRI's along with 5 BRI based premis plugged into 3 x 4-port Digium BRI cards ... but 15 > 12 ... so I can't figure how that works. |
20:16.59 | [TK]D-Fender | ^^^ |
20:17.19 | afink | Hey guys is there any specific linux distro that seems to work best with asterisk? I was using opensuse but I wasn't able to use digium HPEC software b/c suse doesn't use incremental init scripts, then I tried Debian etch which gave me kernel panics and now I am using debian etch n half that just stops working almost every 24 hrs. Can someone point me in the right direction please? |
20:17.44 | soulfreshner | thanks [TK]D-Fender :) |
20:17.45 | [TK]D-Fender | soulfreshner: the sample one includes tons of crap that is not necessarily sanely assembled. This above guide has a very minimalistic structure for you to get an idea from |
20:18.13 | [TK]D-Fender | afink: Debian is normally OK if you know what you're doing, as is CentOS (my recommendation) |
20:18.30 | afink | Thanks [TK]D-Fender |
20:19.43 | SteveTotaro | the call is not dead |
20:20.02 | SteveTotaro | i can do a dial from the h exten using a local chan |
20:22.46 | gr0mit | afink, i use Debian on all my boxes |
20:22.56 | gr0mit | it Mostly Just Works |
20:23.12 | afink | gr0mit: What version and are you using any HPEC software? |
20:23.29 | gr0mit | am using etch on two production asterisk boxes |
20:23.36 | *** join/#asterisk jtodd (i=bqs9e3jl@ns2.loligo.com) |
20:23.55 | gr0mit | HPEC?? |
20:24.13 | afink | Digium High Performance Echo Cancellation for TDM cards |
20:24.34 | gr0mit | ah nope |
20:24.49 | sapere | If I have only one phone line and I want to add fax functionality, I'm kinda screwed right? I mean, since fax machines don't know how to enter an extension... right? |
20:24.56 | gr0mit | no reason to believe they would not work though! |
20:24.56 | boch | where can i read main diff between * 1.4 and 1.6 ? |
20:24.57 | afink | that seems to be the source of all my debian problems. It worked fine before I installed HPEC |
20:25.44 | [TK]D-Fender | sapere: "go read up on |
20:25.54 | [TK]D-Fender | sapere: "go read up on "Asterisk Standard Extensions" on the WIKI |
20:25.56 | [TK]D-Fender | ~wikis |
20:25.56 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
20:26.05 | [TK]D-Fender | boch: changelog.txt |
20:26.11 | [TK]D-Fender | boch: upgrade.txt |
20:26.14 | sapere | [TK]D-Fender : thanks |
20:27.47 | sapere | [TK]D-Fender : another question for ya: if my ITSP gives me 4 channels of service, and each channel allows me a separate incoming or outgoing call, what does this mean for bandwidth? In other words, what does asterisk generally use per channel? |
20:28.09 | sapere | i've read some conflicting information, including some saying as little as 16k per channel |
20:28.55 | [TK]D-Fender | sapere: go read up on "bandwidth consumption on the WIKI. this is protocol, and codec-sensitive |
20:29.01 | [TK]D-Fender | out for now, back later |
20:29.20 | *** join/#asterisk amiral_ (n=amiral@gob75-2-81-56-65-10.fbx.proxad.net) |
20:29.46 | amiral_ | hello ;) how can i have email => fax and fax => email with asteris |
20:30.12 | *** part/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
20:34.23 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
20:34.30 | x86 | amiral_: take out asterisk and insert hylafax |
20:34.56 | kraptv | Does anyone know if there's any ztdummy workaround for some of the stranger operating systems like OSX (Darwin) ? |
20:36.40 | kraptv | Or, another way of putting it, is there an alternative to MeetMe that doesn't rely on zaptel timings? |
20:37.33 | Katty | i had some zaptel problems on boot. it said to check dmesg. |
20:37.40 | Katty | but dmesg doesn't show the zaptel errors. |
20:38.36 | Katty | chan_zap doesn't appear to be loaded. it was complaining about lib modules |
20:38.49 | Katty | any idea where to start on this? |
20:41.09 | amiral_ | x86 hylafax can send fax over sip ? |
20:41.23 | tzafrir_laptop | Katty, what libs? |
20:41.28 | Katty | http://pastebin.ca/1200408 |
20:41.50 | Katty | tzafrir_laptop: still looking for the lib thing i saw. |
20:42.54 | tzafrir_laptop | Katty, is that pastebin related? |
20:42.56 | Katty | tzafrir_laptop: does that pastebin mean anything to you? |
20:42.57 | Katty | aye |
20:43.03 | *** part/#asterisk wacky__ (n=root@nat/digium/x-4763810caeefbe6b) |
20:43.09 | Katty | i'm not sure what i'm looking at really |
20:43.27 | tzafrir_laptop | it means that the kernel module "wanpipe" failed to load |
20:44.20 | tzafrir_laptop | because of missing exports such as wanpipe_ec_register |
20:44.25 | Katty | hmm. |
20:44.29 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
20:44.51 | Katty | do you think i should try to recompile wanpipe? |
20:44.51 | jameswf-home | I dont know why pastebin.ca hates me. |
20:44.54 | x86 | amiral_: not over sip, but using iaxmodem you can send faxes via IAX2 |
20:45.54 | sapere | Would using IAX over SIP be preferable with an ITSP, if it were available as an option? |
20:46.00 | tzafrir_laptop | Katty, no idea |
20:46.18 | Katty | k |
20:47.45 | tzafrir_laptop | grep in the sources for EXPORT_SYMBOL(wanpipe_ec_register) |
20:48.02 | x86 | sapere: it's less bandwidth if you're running multiple concurrent channels, but in my experiences, sip seems more reliable |
20:48.03 | *** join/#asterisk zydoon (n=zydoon@41.225.153.182) |
20:48.22 | x86 | amiral_: I wouldn't recommend fax over IP at all... it's never really reliable |
20:48.25 | *** part/#asterisk zydoon (n=zydoon@41.225.153.182) |
20:48.44 | x86 | amiral_: just setup a fax server with some regular old external faxmodems, and hylafax |
20:49.48 | amiral_ | x86: problem, my company don't have pstn |
20:50.05 | amiral_ | x86: big problem... |
20:51.55 | x86 | very big problem if they want reliable faxing |
20:52.07 | x86 | perhaps you can solve that problem by installing some POTS lines? |
20:53.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
20:53.23 | Katty | tzafrir_laptop: i'm just recompiling wanpipe |
20:53.33 | Katty | tzafrir_laptop: i have no idea what's wrong |
20:53.40 | Katty | tzafrir_laptop: but i recall i updated grub. |
20:53.48 | MindTheGap_ | hello all, how do i take advantage os an md5 password already in a database to authenticate users? our md5 password contain only the pasword as there is samba, ldap, etc authenticating against it, but asteris expects a hashed "peer:realm:secret" in the db not a hashed "secret". |
20:53.49 | Katty | tzafrir_laptop: maybe that did something funky. |
20:54.07 | MindTheGap_ | s/ldap/imap/ |
20:54.26 | tzafrir_laptop | Katty, strange stuff in /proc/cmdline ? yeah, a very longshot |
20:54.51 | jameswf-home | MindTheGap_: AGI |
20:56.51 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
20:57.17 | MindTheGap_ | jameswf-home, sorry but i dont get it. how will an agi help me? |
20:57.17 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
20:57.40 | tvirus | If someone calls in on a land line (zap module) is it possible to transfer that over to SIP to free up the land line? |
20:57.59 | Katty | tzafrir_laptop: i really don't know :/ |
20:58.12 | Katty | tzafrir_laptop: i was tinkering with the grub splash image and it suggested to apt-get upgrade it... |
20:58.18 | Katty | tzafrir_laptop: so i updated grub. |
20:58.26 | Katty | tzafrir_laptop: reboot, and wanpipe spewed. |
20:58.55 | Katty | tzafrir_laptop: only thing i can think of. |
20:59.08 | Katty | tzafrir_laptop: i'm fscking the disk right now, just to make sure |
20:59.58 | *** join/#asterisk Levonk (n=lk@adsl-76-230-108-247.dsl.lsan03.sbcglobal.net) |
21:02.24 | Katty | tzafrir_laptop: well the recompile worked. |
21:02.29 | Katty | tzafrir_laptop: i dunno know really happened |
21:02.37 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:02.42 | tzafrir_laptop | different kernel? |
21:03.01 | jaytee | quittin time, be back later |
21:11.06 | ManxPower | tvirus: generally No |
21:13.54 | legis | How do I send a call via SIP without a username? |
21:14.20 | lmadsen | ... |
21:14.29 | lmadsen | eh? |
21:14.45 | legis | heh |
21:16.01 | legis | Is that i'm dialing this VSP and the call goes through fine through console, which I suppose it doesn't send a username? |
21:17.09 | mchou | which VSP is this? |
21:17.11 | legis | If i call from the softphone it gets rejected |
21:17.43 | legis | dialing through this VSP I mean |
21:17.59 | legis | mchou: one that you probaly haven't heard of |
21:18.09 | mchou | legis: try me anyways |
21:18.21 | WimpMan | I think the console sends "asterisk" just as an originate will. |
21:18.42 | mchou | yup, agree with WimpMan |
21:19.00 | legis | mchou: UNE, in Colombia |
21:19.53 | legis | WimpMan: nice, the softphone is using a number instead of letters, so maybe thats it, let me try |
21:20.37 | Katty | tzafrir_laptop: hrmm, no. i didn't do a kernel update. |
21:20.53 | Katty | not that i can remember, anyway ^_- |
21:20.57 | WimpMan | Could it be the choice of codec? |
21:22.11 | legis | no, cause if I use a DID of the them as the user, the call works too. |
21:23.57 | *** join/#asterisk PakiPenguin_ (n=junaid@linuxpakistan/admin/pakipenguin) |
21:23.59 | PakiPenguin_ | http://www.youtube.com/watch?v=NLRS1b5-Kg8 |
21:25.31 | legis | WimpMan: thanks!, looks like they only accept letters or they own DID |
21:26.39 | *** join/#asterisk hfb (n=hfb@96.247.52.72) |
21:26.54 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583722.dsl.bell.ca) |
21:28.34 | sapere | exit |
21:28.44 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
21:28.57 | legis | now I have to test up fax with them :/ |
21:30.32 | *** join/#asterisk great_Anta_Baka (i=c636caf6@gateway/web/ajax/mibbit.com/x-e0ded14cd1bb3220) |
21:30.37 | *** join/#asterisk hmodes (i=hmodes@B1-66ER.matrix.gs) |
21:30.52 | legis | is there an app to send faxes? |
21:30.53 | great_Anta_Baka | how do i get rid of this how do i get rid of this linux/compiler.h: No such file or directory? |
21:31.03 | great_Anta_Baka | trying to install h323 support for asterisk |
21:31.53 | legis | great_Anta_Baka: looks like you are missing dependacies |
21:32.18 | great_Anta_Baka | that file doesnt exist in the kernel any more |
21:32.35 | legis | great_Anta_Baka: what distro? |
21:32.46 | great_Anta_Baka | this is a fedora box |
21:33.13 | legis | does it have the kernel headers installed? |
21:33.13 | great_Anta_Baka | its when i am compiling open h323 |
21:33.33 | great_Anta_Baka | i can install them again but linux/compiler.h doesnt exist in the headers |
21:33.42 | great_Anta_Baka | it was removed in an earlier kernel version |
21:34.13 | legis | install the kernel headers |
21:34.18 | great_Anta_Baka | i have done this before just cant remember what i did :( |
21:34.22 | ManxPower | great_Anta_Baka: What kernel version are you running? What version of Asterisk? What version of Asterisk-Addons? |
21:34.40 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
21:34.41 | great_Anta_Baka | latest asterisk on the digium page |
21:34.59 | ManxPower | great_Anta_Baka: I am not going to go to Digium's page to look that up. |
21:35.07 | ManxPower | Anyway, best of luck. |
21:35.18 | great_Anta_Baka | 1.4.21.2 or something like that |
21:35.23 | ManxPower | I was not aware that Digium had linux kernels for download. |
21:35.40 | great_Anta_Baka | 2.6.18-53.1.4.el5 |
21:35.44 | great_Anta_Baka | kernel version |
21:36.02 | great_Anta_Baka | i mean asterisk version |
21:36.22 | great_Anta_Baka | but this hasnt got to do with asterisk |
21:36.29 | great_Anta_Baka | asterisk isnt compiled yet |
21:36.37 | great_Anta_Baka | need to install openh323 first |
21:36.48 | ManxPower | well nevermind then |
21:37.04 | ManxPower | Sorry, I thought you were having Asterisk problems. |
21:37.16 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
21:37.23 | great_Anta_Baka | legis kernel headers installed |
21:37.35 | ManxPower | great_Anta_Baka: which of the 3 or 4 H323 drivers do you want to install? |
21:38.00 | great_Anta_Baka | ManxPower: installing from this guide http://astrecipes.net/index.php?from=40&q=AstRecipes/Compiling%20Asterisk%201.4%20with%20TDM400%20and%20H323 |
21:38.06 | scooby2 | Another strange one. Agent logs out. Directly as logging out, a call goes to him. It cancels the logout thus leaving him logged in. |
21:39.09 | ManxPower | great_Anta_Baka: I suggest you use the one in Asterisk-addons |
21:39.47 | great_Anta_Baka | i see... do i compile asterisk then the addons and then asterisk again? |
21:39.58 | ManxPower | great_Anta_Baka: you follow the instructions for it. |
21:40.26 | great_Anta_Baka | kk |
21:40.27 | ManxPower | the one in asterisk-addons is the one DIGIUM PAID FOR. |
21:41.33 | great_Anta_Baka | according to to voip-wiki asterisk addons on has 4 packages and none of them has h323 in it o_0 |
21:41.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
21:41.56 | ManxPower | great_Anta_Baka: perhaps you should look at the asterisk-addons package rather then the old and outdated information on the Wiki. |
21:43.21 | great_Anta_Baka | mmm looking |
21:43.25 | ManxPower | You should report it as a bug if the H323 channel driver is not in asterisk-addons |
21:45.12 | scooby2 | would hitting dnd before trying to logout help? |
21:47.55 | *** join/#asterisk nr4q (n=dave@12.160.78.184) |
21:52.35 | great_Anta_Baka | ManxPower: after installing the asterisk-addons package, do i have to ./configure and make asterisk again? |
21:52.54 | ManxPower | great_Anta_Baka: I have no idea what you have to do. I assumed there would be instructions. |
21:53.00 | great_Anta_Baka | cos in the make menuselect of asterisk chan_h323 still has [XXX] next to it |
21:53.07 | great_Anta_Baka | no instructions |
21:53.44 | ManxPower | Then I guess you need to report a bug |
21:53.55 | ManxPower | and it is NOT chan_h323 |
21:53.59 | great_Anta_Baka | well restarted asterisk and did module show like h323 |
21:54.03 | ManxPower | as you can see in the source code in the asterisk-addons. |
21:54.04 | great_Anta_Baka | and it seems like its loaded |
21:54.07 | ManxPower | it is chan_ooh323 |
21:54.08 | great_Anta_Baka | so thanks :) |
21:54.20 | great_Anta_Baka | yeah |
21:55.13 | great_Anta_Baka | oh and how do i stop my iax trunk from reregistering every 10-20 seconds? |
21:56.05 | [TK]D-Fender | great_Anta_Baka: "noload => chan_iax2.so" in modules.conf |
21:56.25 | *** join/#asterisk Paige (n=Paige@208.89.241.31) |
21:56.40 | [TK]D-Fender | ;) |
21:56.44 | Paige | hi, i need some help getting mysql realtime to work |
21:56.46 | great_Anta_Baka | lol |
21:56.56 | *** join/#asterisk Levonk (n=lk@adsl-76-238-248-235.dsl.lsan03.sbcglobal.net) |
21:57.15 | [TK]D-Fender | great_Anta_Baka: Could be your provider is insisting on that timeout. Otherwise hit up the WIKI page for iax.conf and I'm sure you'll find it... |
21:58.28 | Paige | my error is: ERROR[8206]: res_config_mysql.c:845 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on /var/lib/mysql/mysql.sock (err 1045). |
21:58.39 | great_Anta_Baka | mmm well i set the minexpire time to 60 on the side where the trunk is registering to |
21:58.44 | great_Anta_Baka | will see if that changes anything |
21:58.46 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
22:00.09 | Paige | can someone help me trouble shoot this error? |
22:01.59 | scooby2 | does /var/lib/mysql/mysql.sock exist? |
22:02.06 | Paige | yes |
22:02.20 | great_Anta_Baka | ok so now that chan_ooh323 is installed... in my sip.conf file do i just put allow=h323 for the user that? |
22:02.32 | scooby2 | username/password exists on the mysql db you are trying to use? |
22:02.45 | great_Anta_Baka | http://www.google.co.za/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fforums.mysql.com%2Fread.php%3F11%2C19531%2C19531&ei=dpXJSI27B6TIsgKrpvXUBg&usg=AFQjCNEhCbVzgWUDqpEmpjYFBSD0t1jPEg&sig2=cLvq0NGEsB_EtLR97OolVg |
22:02.57 | Paige | matches in res_mysql.conf and works by command line |
22:03.06 | great_Anta_Baka | access denied |
22:03.07 | great_Anta_Baka | error |
22:03.37 | great_Anta_Baka | check your privileges Paige |
22:03.49 | Paige | i did |
22:04.23 | scooby2 | set the logs to include verbose and debug? What does messages say? |
22:04.32 | *** part/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
22:05.20 | great_Anta_Baka | did you set up a custom user to connect the mysql database or you using the root user? |
22:05.40 | Paige | custom user |
22:06.02 | great_Anta_Baka | and what grant command did you use? |
22:06.19 | Paige | everything is yes |
22:07.17 | jameswf-home | As mom said in mysql dont forget to flush |
22:07.24 | great_Anta_Baka | lol |
22:07.28 | great_Anta_Baka | my thoughts exactly |
22:07.44 | great_Anta_Baka | also just see if it works connecting with the root user to the database |
22:07.46 | scooby2 | Paige said that he/she could login manually |
22:07.50 | Paige | i lushed |
22:07.57 | Paige | flushed |
22:08.09 | jameswf-home | did you jiggle the handle? |
22:09.03 | great_Anta_Baka | are you connecting with the -h option? |
22:09.32 | Paige | ? |
22:09.47 | great_Anta_Baka | mysql -h 127.0.0.1 -u root -p |
22:09.57 | great_Anta_Baka | or mysql -u root -p |
22:09.59 | Paige | yes |
22:10.06 | great_Anta_Baka | yes? |
22:10.13 | great_Anta_Baka | <PROTECTED> |
22:10.35 | Paige | mysql -u astuser -p |
22:11.10 | great_Anta_Baka | any white spaces in your cofig file? not that i think that makes a difference |
22:11.22 | Paige | no whitespace |
22:11.22 | great_Anta_Baka | just check if you can log in with your root mysql user |
22:11.28 | great_Anta_Baka | in the asterisk conf file |
22:11.48 | great_Anta_Baka | i mean the mysql |
22:11.51 | Paige | the database config is in res_mysql.conf |
22:11.57 | great_Anta_Baka | yes |
22:11.59 | great_Anta_Baka | that one |
22:12.12 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
22:14.56 | Paige | i found it. typo |
22:15.34 | scooby2 | lol |
22:15.43 | scooby2 | always something simple |
22:17.29 | great_Anta_Baka | anyone done any video through asterisk.. i dont know where to start other than having installed chan_ooh323 |
22:20.32 | Paige | thanks for the help guys! |
22:20.39 | *** part/#asterisk Paige (n=Paige@208.89.241.31) |
22:26.29 | j0 | does anyone have a nice pdf of the voicemail options? |
22:26.42 | j0 | i guess every system is different, but at least one for the generic comedian mail |
22:30.34 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
22:36.29 | hohum | hey guys |
22:36.43 | hohum | what happened to the "odbc show" command int he latest version of asterisk 1.4 |
22:37.45 | hohum | help please |
22:40.35 | hohum | never mind I don't have res_odbc even compiled |
22:46.12 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
22:49.33 | jaytee | anyone know of a cheap analog line audio coupler that can plug into a line jack, seize the line in an off-hook condition and has an audio output like a handset without a mike? |
22:50.26 | *** join/#asterisk sapere (n=sapere@c-71-206-107-228.hsd1.mi.comcast.net) |
22:50.33 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
22:51.27 | pjz | like a cheap phone? |
22:51.31 | WimpMan | jaytee: Any old modem? |
22:52.04 | *** part/#asterisk pjz (n=pj@zachs.place.org) |
22:52.14 | jaytee | yeah, but I only need to recieve audio, don't need mic output on the analog line, don't need a dtmf keypad either. |
22:55.14 | jaytee | this would be used in place of a phone so I can use it with an SPA3102 FXS port setup as a hotline to an extension on Asterisk that answers and immediately plays musiconhold so I can take the audio output and hook it into the music channel of a PA system. The PA side works with an FXO port but I can't use the MOH on that channel because the Page application doesn't support MOH. |
23:00.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
23:04.01 | *** join/#asterisk imcdona (i=imcdona@2001:5c0:8fff:fffe:0:0:0:b917) |
23:05.01 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-192-41.try.wideopenwest.com) |
23:05.18 | tmccrary | Can anyone recommend a somewhat decent sip termination service provider? |
23:06.35 | bminish | tmccrary, where are you ? |
23:06.44 | tmccrary | oh sorry, midwest usa |
23:07.31 | bminish | ok then blueface.ie will not be that great for you (they also do IAX trunking if you ask them.) |
23:07.40 | mchou | tmccrary: what's you definition of decent? |
23:07.58 | mchou | tmccrary: and what's your call volume min/mo? |
23:08.04 | bminish | !crap I would guess ;-) |
23:08.04 | mchou | your* |
23:08.19 | mchou | bminish: lol |
23:09.25 | mchou | bminish: would that include vonage? :) |
23:10.03 | tmccrary | decent, something that is fairly reliable |
23:10.10 | bminish | dunno, never used them, happy with blueface.ie though |
23:10.14 | tmccrary | call volume very low, like maybe a few calls a weak |
23:10.18 | tmccrary | *week |
23:10.39 | mchou | tmccrary: if you call volume is low go with these guys: diamondcard.us |
23:10.59 | mchou | tmccrary: no affliation except a reasonably happy customer |
23:11.11 | tmccrary | thanks I'll check it out |
23:11.43 | mchou | DIDs in all US and termination a-z |
23:12.10 | mchou | tmccrary: dont be discourage by their mickey mouse kindergarten web site |
23:12.18 | mchou | discouraged* |
23:12.59 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
23:13.10 | phix | hey |
23:13.40 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
23:16.55 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
23:16.55 | *** mode/#asterisk [+o mog] by ChanServ |
23:17.46 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
23:17.46 | *** mode/#asterisk [+o russellb] by ChanServ |
23:18.26 | jaytee | my boss just got approval for me to go to Advanced Asterisk training, YAY!!!!! |
23:18.34 | russellb | jaytee: nice!! |
23:18.44 | *** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.158.28) |
23:18.45 | russellb | jaytee: it sounds like a pretty cool course |
23:18.50 | jaytee | it does |
23:19.06 | russellb | wants to take it |
23:19.06 | russellb | :) |
23:19.21 | jaytee | russellb, are you in Huntsville? |
23:19.25 | russellb | yes, i am |
23:19.36 | jaytee | cool, maybe we can do lunch! |
23:19.40 | russellb | sure! |
23:19.49 | file | it's a trap! |
23:19.52 | russellb | just let me know when you're here, and I can come eat lunch with the whole group |
23:20.38 | jameswf-home | jaytee: has a big club... |
23:20.44 | russellb | o.O |
23:20.46 | mchou | There's no such thing as a free lunch :) |
23:20.58 | jaytee | I actually kind of had the choice between Huntsville and Vegas but Vegas just isn't anything that thrills me. not a gambling man and I don't care for hookers. |
23:21.20 | russellb | Huntsville has ... um ... cotton fields |
23:21.22 | jaytee | but I figured I'd stay an extra day and see if I can get into a tour of the Marshall Space Flight Center |
23:21.26 | russellb | and a space center |
23:21.27 | mchou | jaytee: what attractions does Huntsville have? |
23:21.37 | mchou | ok, besides the space ctr |
23:21.41 | jaytee | I'm too old for Space Camp :-) |
23:21.47 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-230-174.balt.east.verizon.net) |
23:22.01 | *** join/#asterisk Levonk (n=lk@adsl-76-227-116-162.dsl.lsan03.sbcglobal.net) |
23:22.05 | jameswf-home | I wanna go to the flea market in MG |
23:22.09 | russellb | yes! |
23:22.51 | jameswf-home | i hear its just like a minimall |
23:23.36 | jaytee | MG? |
23:23.41 | russellb | Montgomery |
23:23.45 | russellb | ~nowwhat |
23:23.45 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
23:23.49 | russellb | see the video |
23:23.59 | jameswf-home | cant spell Montgomery |
23:24.22 | russellb | huh ... what's with the music on that one |
23:24.23 | russellb | lame |
23:24.38 | jaytee | ok, that's almost scary! |
23:24.46 | russellb | anyway, search for "flea market montgomery" |
23:24.48 | jameswf-home | has leekspin as his MOH it is awesome |
23:26.50 | mchou | I dont even get the concept |
23:27.08 | mchou | the whole flea market is owned by one guy?? |
23:27.39 | mchou | he must have lots of crap if he can fill 73000 sq ft :) |
23:27.54 | russellb | it's just like a mini-mall, obviously |
23:28.23 | mchou | russellb: Is it mostly new or used stuff? |
23:28.29 | russellb | i have never been there |
23:29.16 | mchou | ahh, too bad. I was hoping you'd enlighten me |
23:31.02 | jaytee | ok, I'm looking for something similar to this little gizmo but I need it to act like a phone that's off-hook, not a passive tap into an extension line. http://www.omnicronelectronics.com/analog/lic-390.htm |
23:31.17 | jaytee | anyone know of anything "cheap" |
23:32.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:33.16 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
23:34.22 | mchou | anyone here ever installed HW for a telemarketing call center? |
23:35.48 | russellb | telemarketing is eeeeeevil |
23:36.13 | mchou | russellb: I know it's evil. that's why I'm seeking info |
23:36.29 | russellb | edits the Asterisk license the exclude telemarketer's from being allowed to use it |
23:36.39 | mchou | lol |
23:36.44 | Rienzilla | hehe |
23:37.00 | Rienzilla | maybe asterisk could be patched to include an equivalent of the evil bit |
23:37.11 | Rienzilla | the m extension, for telemarketeers |
23:37.23 | mchou | russellb: I'm sure the telemarketers will find an escape clause somewhere. |
23:37.50 | russellb | well, we can't retroactively change the license of every version previously released |
23:38.24 | jameswf-home | you can if you hire chuck norris |
23:38.32 | mchou | haha |
23:39.48 | mchou | This is what REALLY irritates me: http://800notes.com/articles/News.aspx/0PX8Dr_Y_AAcJgjKvtDs0g |
23:40.25 | mchou | you break the law, get a wrist slap |
23:40.40 | *** join/#asterisk onescomplement (n=dave@mail.davefuller.net) |
23:41.16 | mchou | "Based on the companies' ability to pay, the amounts were reduced to $20,000 and $75,000 respectively." |
23:41.20 | mchou | wtf?? |
23:42.49 | mchou | And I dont understand why the govt didmt go after dish network instead of their contractors |
23:42.57 | mchou | didnt* |
23:45.38 | jameswf-home | http://dontcallmyboss.blogspot.com/2008/09/epic-battle.html << hitler beat Chuck norris 5 to 1 |
23:48.36 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-192-41.try.wideopenwest.com) |
23:50.24 | mchou | so here is a question. suppose at telemarketer calls. I use asterisk to defeat AMD and get a live person on the line. Then I play this: "The number you have called is on the Federal "Do not call" registry. This call may be recorded. You will risk legal action if you call again." |
23:50.58 | mchou | Does this have teeth or am I whistling in the wind? |
23:53.29 | jaytee | mchou, whistling into the wind |
23:53.40 | mchou | jaytee: why do you say that?? |
23:54.13 | mchou | jaytee: even if I get a live telemarketer? |
23:54.31 | Rienzilla | what is AMD? |
23:54.52 | mchou | Answering Machine Detection (on their end) |
23:55.09 | Rienzilla | ah |
23:55.25 | jaytee | Do Not Call has not real teeth. Legal action? yeah, so you sue, the lawyers settle, your lawyer keeps their cash and bills you an additional amount. feelin better yet? |
23:55.49 | mchou | sure I fell better. They wont bother me again |
23:55.54 | mchou | feel* |
23:56.06 | jaytee | mchou, wanna know my solution? |
23:56.14 | mchou | that's the whole point |
23:56.21 | mchou | jaytee: sure |
23:56.57 | mchou | jaytee: plus there's no lawyer involved. It's all small claims |
23:57.18 | mchou | cause each fine I think is <$5K |
23:57.47 | jaytee | I filter by callerid, if it's blocked they get a message that says, "I don't know who you are so go away". If it's any number with an 800 NPA I route it into a queue with music on hold and announcement every 30 seconds that says, "Your call is unimportant to us, please continue to hold and someone will be with you when they are bored senseless" |
23:58.10 | mchou | jaytee: I do something similar already |
23:58.33 | jaytee | you could also do Zapateller and then dump them in a queue |
23:58.42 | mchou | but that's not enough. I dont want them to call "on my dime," as it were |
23:59.26 | mchou | jaytee: I do Zapateller already. You know that doesnt work cause telemarketers get PRI signalling |