| 00:03.31 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 00:04.21 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 00:04.30 | Dr-Linux|home | [TK]D-Fender: since my internet is slow, i just downloaded your music :P |
| 00:04.53 | bkruse_home | geez.....thats slow |
| 00:06.00 | *** join/#asterisk dendrite (n=ph@dsl092-150-203.wdc2.dsl.speakeasy.net) |
| 00:07.11 | *** join/#asterisk AJaymn (n=root@216-55-162-53.dedicated.abac.net) |
| 00:08.24 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:10.40 | AJaymn | ? |
| 00:16.30 | ManxPower | Are there ant RF Geeks around? /msg me, I have a couple of questions WRT 1.9Ghz antennas |
| 00:17.43 | *** join/#asterisk SLiNK (n=KNOCK@c-68-63-34-189.hsd1.al.comcast.net) |
| 00:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:18.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:21.05 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:23.28 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 00:23.34 | Asterman | Manx : Doh, I'm in big trouble :)) I just recompiled * after I had compiled zap first, I spotted that it compiled chan_zap, so I thought I was in good shape, but when I try to start * now it complains that /etc/zaptel.conf isn't there :( |
| 00:23.41 | Asterman | Manx: and * won't start at all :( |
| 00:24.13 | [TK]D-Fender | Asterman : "touch /etc/zaptel.conf |
| 00:25.35 | Dr-Linux|home | i'm having an issue, when an agent logged in to the queue and closed his client, this way if the call comes, it goes to his/her vicemail via macro. Do i need to change my macro or where i'm wrong? |
| 00:25.35 | Asterman | TK : ok, now it's complaining that /etc/zapata.conf isn't there now, I was told earler by someone to remove that as I'm using ztdummy with no card....is that right? |
| 00:26.04 | [TK]D-Fender | Asterman : "touch /etc/asterisk/zaptel.conf |
| 00:26.10 | [TK]D-Fender | Asterman : NEXT!!!@!@ |
| 00:26.16 | [TK]D-Fender | (c) BKW |
| 00:26.27 | Dr-Linux|home | as i'm using round robin, so it rings expected user and after that goes to his voicemail, but it should not |
| 00:26.54 | [TK]D-Fender | Dr-Linux : the you shouldn't be running extens that lead to vm. |
| 00:28.27 | Dr-Linux|home | [TK]D-Fender: hhm.. but agent extension also leads to VM. |
| 00:28.56 | [TK]D-Fender | Dr-Linux : So stop doing things that are clearly not bright.... |
| 00:28.56 | Dr-Linux|home | [TK]D-Fender: if i use simple members in queues.conf, then all fine, but if i use agent callbacklogin, then this issue |
| 00:29.14 | alamantia | humm, have any of you seen this? http://www.digg.com/linux_unix/Mark_Spencer_Presents_AsteriskNOW |
| 00:29.32 | [TK]D-Fender | Dr-Linux : All that does is dil an exten in the context you tell it to. its YOUR fault for letting it call macros that fallback to VM. |
| 00:29.35 | Asterman | TK : ok, touched that too.... how do I go about configuring channels in zaptel.conf if I don't have any physical cards in the box and I'm just using ztdummy at the moment I'm getting an error that says the device is busy |
| 00:29.50 | [TK]D-Fender | Asterman : Should have to. |
| 00:30.09 | [TK]D-Fender | Asterman : modprobe ztdummy |
| 00:30.16 | [TK]D-Fender | Asterman : modprobe zaptel |
| 00:30.20 | [TK]D-Fender | Asterman : thenr etry |
| 00:30.34 | Asterman | TK : rmmod then first? (as they're already loaded) |
| 00:30.55 | [TK]D-Fender | Asterman : Dunno... jsut try |
| 00:31.00 | Dr-Linux|home | hhm.. |
| 00:31.19 | Dr-Linux|home | [TK]D-Fender: our clients always report new issue :P |
| 00:31.59 | Asterman | TK : Ok, did that I get multiple errors when doing the modprobe ztdummy (regarding Cannot get number of tones for channel 1) |
| 00:33.32 | Asterman | TK : sorted it out ;) |
| 00:33.38 | bkruse_home | ManxPower: i need to become an rf geek |
| 00:33.38 | bkruse_home | need/want |
| 00:33.54 | *** join/#asterisk awannabe (n=brad@207-114-155-214.static.twtelecom.net) |
| 00:33.57 | bkruse_home | woah |
| 00:34.04 | bkruse_home | check it out http://www.digg.com/linux_unix/Mark_Spencer_Presents_AsteriskNOW |
| 00:34.09 | awannabe | anyone used the metermaid patch? |
| 00:34.38 | Asterman | TK : ztdummy loaded without any problems, but asterisk still won't start up, complaining about unable to register channel 1 |
| 00:35.04 | Mad|Cow | Can someone tell me why when I use type=user in my sip.conf file my SIP clients fail to register, but if I change them to type=friend, they work? |
| 00:35.38 | Asterman | TK : arggggggggggghhhhhhhh .... ok, so I fixed that last one too |
| 00:36.07 | Asterman | TK : but now I'm getting a new error on startup of asterisk : ast_register_application: Already have an application 'Pickup' |
| 00:36.27 | Asterman | looks like somewhere I'm trying to load the same thing twice??? |
| 00:36.30 | SLiNK | Im trying to log in Asterisk API with MD5 authentication by PHP generation challenge+password - I trim the string before hashing -Authentication Fails, all is good clear-text wise. The only think I can think references Ive found said use option "md5 -s" on command line. Though I have found no digest utilty that will take this option for confirmation of my php hash. Anyone know anything I should check into? |
| 00:37.52 | [TK]D-Fender | Asterman : go into /usr/lib/asterisk/modules and clear it out, and redo "make install" for zapte, then * |
| 00:42.57 | Asterman | TK : joy joy happy happy joy joy :))) asterisk fires up once again *phew*.... ok, so now that I when I dial meetme, I'm not geting any of the warning about not being able to load up the zap channel, but it's still giving me the invalid pin message even though I know the pin is correct |
| 00:44.33 | Asterman | TK : oh hang on, I thnk I might know what might behind this....brb |
| 00:44.40 | *** join/#asterisk X-Rob (n=rob-x@CPE-61-9-217-229.static.qld.bigpond.net.au) |
| 00:45.42 | Asterman | TK : nope, I thought for a sec it might have been the dtmf setup, but it's not that |
| 00:46.05 | awannabe | what access do you have to have to download patches off the digium site?/ |
| 00:46.59 | Asterman | awannabe : me? |
| 00:47.06 | [TK]D-Fender | Asterman : verify between meetme.conf and how yoou call it in extensions.conf. |
| 00:47.13 | X-Rob | You need to know how to use SVN or a Web browser, apart from that, nothing special. |
| 00:47.46 | awannabe | X-Rob, it says access denied! |
| 00:47.50 | De_Mon | some tard set my asterisk startup script with the args -gvvvcr. no wonder they stopped working. |
| 00:48.14 | awannabe | http://bugs.digium.com/file_download.php?file_id=10350&type=bug%22 -< no workie :( |
| 00:48.44 | Asterman | TK : my meetme.conf is blank in the [general] section and in the [rooms] section I have just one room that says conf => 0001,0001,0001 |
| 00:49.09 | X-Rob | I doubt there should be a %22 on the end |
| 00:49.29 | awannabe | ahh, SOB!! lol |
| 00:49.36 | perd | duhhh |
| 00:50.02 | Asterman | TK : and then for my dialplan I have two lines for the conf call, one to answer and the other to invoke meetme (exten => 9000,2,Meetme(0001) ) |
| 00:50.36 | Asterman | TK : sorry, typo, the conf line in meetme says conf => 0001,0001,0002 |
| 00:54.52 | [TK]D-Fender | Asterman : You are supposed to pass a pin, try working that way first |
| 00:56.04 | *** part/#asterisk mog (n=mog@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:56.55 | Asterman | TK : you mean in the dial plan it should be set up to read exten => 9000,2,Meetme(0001,0001) (this is different from the docs I've seen, maybe I've been reading docs from an earlier version) |
| 00:57.02 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
| 01:00.21 | awannabe | anyone messed with the metermaid patch, to have lights on the Snom and other phones for parked calls? |
| 01:01.58 | Asterman | TK : ok, I've got the dialplan setup as exten => 9000,2,Meetme(0001,i,0001) but still no joy |
| 01:02.59 | [TK]D-Fender | Asterman : "show application meetme". |
| 01:03.09 | [TK]D-Fender | Asterman : aim SUPER basic to start,a nd work your way up |
| 01:05.11 | flenders | hey, I spoke to our phone carrier yesterday, asking them if we could change the outgoing caller id on all our lines so they would look the same as our main number. they said they couldn't do it, but asked me to speak to the phone system guy (me), as it was possible to do it on the phone system. can I do that on asterisk? I mean, can I spoof the caller id on my outgoing calls? |
| 01:05.38 | Strom_M | flenders, are you using an ISDN PRI? |
| 01:05.49 | flenders | nope, a bunch of PSTN lines |
| 01:06.01 | Strom_M | well then the telco has to change the caller ID |
| 01:06.07 | Strom_M | or you need to upgrade to ISDN |
| 01:06.24 | JT | flenders: the answer i gave yesterday is still as valid today |
| 01:06.27 | [TK]D-Fender | flenders : if your telco sys they can't change the callerid to match your primary they are FULL OF SHIT. |
| 01:06.32 | perd | owned |
| 01:06.53 | Asterman | TK : I thought you couldn't get much more basic than that :) I would have thought the most basic would be not to have a pin at all :( |
| 01:07.09 | flenders | JT: sorry, my box went down yesterday, and i only got nugget's reply |
| 01:07.13 | [TK]D-Fender | Asterman : NOW your getting a idea ;) |
| 01:07.49 | Asterman | TK : what's irritating is there's no warning or error messages |
| 01:07.49 | Asterman | TK : even with the verbose turned right up |
| 01:07.49 | JT | flenders: it's not pysically possible to send outgoing callerid from your end if you do not have digital lines |
| 01:07.58 | JT | telcos can set them statically to whatever they like |
| 01:08.07 | flenders | JT: got it |
| 01:08.12 | flenders | I'll ring them again |
| 01:08.22 | perd | your telco are lazy jerks |
| 01:08.50 | flenders | perd: I know they are... they used to be a government company. |
| 01:09.20 | flenders | telstra in AU. if there are any aussies in here, they would know what I'm talking about |
| 01:09.26 | [TK]D-Fender | flenders : then aim where it hurts. their POCKETS. threaten to leave unless they give you a level 2 tech who'll actually get something DONE. |
| 01:09.53 | perd | probably the only carrier out in the middle of ozzieland |
| 01:09.55 | flenders | [TK]D-Fender: I thought I should ask here first |
| 01:09.56 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 01:09.58 | flenders | :D |
| 01:10.15 | [TK]D-Fender | flenders : "I wan all of my POTS lines (list them) to match my pilot number (1st number in your hunt group), OR ELSE". |
| 01:11.12 | flenders | will ring them now |
| 01:13.50 | perd | get a sip client, connect to my asterisk box and we can call them in a #asterisk conference |
| 01:13.54 | perd | and verbally abuse them |
| 01:14.03 | perd | and their silly australian accents |
| 01:14.03 | SLiNK | Ill be damned I got the FreeBSD md5 utility with -s extension and Im sending a proper hash. but authentication fails even entered manually in telnet. There must be something I need to switch on in manager.conf or asterisk although I got a challenge code |
| 01:14.21 | JT | flenders: yes, i'm in aust, and with telstra for some stuff |
| 01:14.29 | JT | perd: what are you taking about? |
| 01:14.50 | perd | his phone carrier |
| 01:14.57 | [hC] | Ive got a polycom phone here that claims inuse, even though nobody's on it (in sip hints) any idea why a hint might get stuck inuse? |
| 01:15.47 | [hC] | ah. found it. asterisk thinks hes got a channel stuck in app voicemailmain |
| 01:15.51 | [hC] | I wonder why that would happen. |
| 01:15.53 | Asterman | TK : when I have it set without a PIN it works fine, but the moment I try to put a PIN in there it always rejects it |
| 01:15.56 | [hC] | never had that happen before. |
| 01:16.01 | JT | perd: "silly australian accents" "only carrier out in the middle of ozzieland" how about you get some facts, first? |
| 01:16.13 | JT | and not be an arsehole |
| 01:16.34 | SLiNK | well hmm i had to have AuthType: MD5 twice |
| 01:17.52 | [TK]D-Fender | Asterman : prove that your dtmf works. |
| 01:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:18.12 | [TK]D-Fender | Asterman : make an exten that answers, does a read into a variable thenr eads it back. |
| 01:21.11 | [hC] | If i have a call stuck in VoiceMailMain, and a soft hangup doesnt drop it, where do i start looking to figure out why that would have happened? |
| 01:21.17 | perd | jt, you're a little sensitive |
| 01:21.56 | JT | perd: you're a little arrogant |
| 01:22.05 | perd | yeah i suppose i am |
| 01:22.07 | perd | so what? |
| 01:22.08 | flenders | spoke to them again, they're still saying that it is done on the phone system. |
| 01:22.31 | perd | <PROTECTED> |
| 01:22.33 | JT | flenders: ask to speak to someone with a clue, make sure you make it cystal clear that they are analogue lines and not ISDN |
| 01:22.44 | JT | perd: /die |
| 01:23.01 | flenders | JT: man, I tried... they said they're a sales office, and she would not give me a tech support number |
| 01:23.07 | flenders | it's just the business sales |
| 01:23.10 | JT | flenders: say you want them statically set |
| 01:23.12 | JT | do not ring sales |
| 01:23.16 | JT | ring tech support |
| 01:23.19 | JT | business tech |
| 01:23.38 | dendrite | Odd, around here, sales will tell you *anything* is possible... |
| 01:24.07 | flenders | JT: I'm trying to find their number |
| 01:24.53 | JT | 132253 |
| 01:25.42 | flenders | that's the one I got |
| 01:25.49 | flenders | and the one I rang before |
| 01:25.54 | flenders | options 2 and 3 |
| 01:26.17 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 01:27.00 | Asterman | TK : damn it.... I took a look at the dtmf settings, they had been changed since the last time I'd looked.... everything is now working perfectly.... thank you for all the help and patience :) |
| 01:27.07 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 01:28.25 | JT | flenders: ask for line inquiries or something like that |
| 01:28.36 | JT | say you have a technical question about the configuration of the line |
| 01:30.03 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:33.07 | flenders | just spoke to one of my mates who used to sell alcatel systems, and he said people used to ask him for that, and telstra would never set static caller ids |
| 01:33.17 | [TK]D-Fender | flenders : BS... try someone else. |
| 01:33.33 | [TK]D-Fender | Asterman : Quite welcome |
| 01:35.27 | JT | flenders: ah, there you go then |
| 01:35.48 | dendrite | flenders: If all else fails, maybe you can play an orange-box code at the start of the call... http://artofhacking.com/files/OB-FAQ.HTM |
| 01:36.25 | JT | orange box code, are you serious? |
| 01:36.25 | flenders | JT: so that's a no no I think |
| 01:36.34 | JT | this is a digital phone network |
| 01:36.53 | dendrite | JT: I thought he said it was analog. |
| 01:37.28 | JT | the network is digital, his line is analogue |
| 01:37.44 | JT | most phone exchanges in the first world have been digital for at least 2 decades now |
| 01:37.56 | dendrite | Ah. So OB (after call established) wouldn't redraw the CID? |
| 01:38.12 | JT | i doubt it |
| 01:38.29 | JT | most coloured "boxes" don't work anymore |
| 01:38.45 | JT | because exchanges use Out Of Band, or Common Channel Signalling |
| 01:38.57 | *** join/#asterisk demigod2k (n=joey@cpe-65-29-113-212.twmi.res.rr.com) |
| 01:38.58 | demigod2k | hi |
| 01:39.04 | dendrite | I thought the CID was quite audible. |
| 01:39.18 | JT | inbound sure |
| 01:39.31 | JT | outbound, none is set, it's set by a database in the exchange |
| 01:39.40 | coppice | JT: 2 decades is an exagerration. the first place to achieve an all digital network only did so in the early 90s |
| 01:39.52 | dendrite | JT: Yep, and possibly reset, by caller squawk. |
| 01:40.12 | JT | coppice: all digital vs. mostly digital |
| 01:40.36 | JT | dendrite: what? outbound callerid doesn't even come into play over analogue line customers' lines |
| 01:41.05 | JT | it is send as ANI and CLI over SS7 links to the destination terminating exchange |
| 01:41.07 | dendrite | JT: That's what the orange box does. It plays the signal after the answer. |
| 01:41.35 | JT | dendrite: when it's too late? the exchange would have already sent the callerid |
| 01:41.42 | JT | during the alerting phase |
| 01:41.44 | coppice | in 1986 many countries were still deciding about their all digital exchanges. most got pretty aggressive about their rollouts once they decided, though |
| 01:41.44 | dendrite | JT: Yes, but the CID may redraw. |
| 01:42.04 | dendrite | JT: YOu won't be able to affect the pre-answer CID, sure... |
| 01:42.16 | JT | dendrite: sounds pretty dodgy, and it won't affect HEAPS of terminating lines |
| 01:42.29 | JT | as a lot terminate to digital ISDN, digital mobile telephony, etc |
| 01:42.36 | dendrite | JT: flenders seems to be hitting a wall, and my (probably useless) suggestion was just a possible last gasp idea. |
| 01:42.39 | Mad|Cow | I have a Cisco 7940 at a friends house (behind his firewall) which is configured to use my Asterisk server as its proxy. When I try and call the phone, I get a "Unable to create channel of type 'SIP' (cause 3 - No route to destination)". However when I do a sip show peers, I see the phone registered. Anyone have any ideas? |
| 01:42.52 | dendrite | JT: "Pretty dodgy" hardly covers it. :-) |
| 01:42.58 | JT | most businesses terminate with isdn |
| 01:43.05 | JT | and a lot of calls go to mobiles these days :) |
| 01:43.17 | dendrite | JT: Point, and point. |
| 01:43.34 | demigod2k | is isdn affordable anywhere these days? the rates in michigan are awful |
| 01:43.51 | sudhir492 | anyone using Flash Operator panel here, specifically the Panel_Context feature? |
| 01:43.54 | flenders | dendrite: thanks anyway mate |
| 01:43.57 | JT | flenders: how important is it to change outbound callerid? |
| 01:44.13 | dendrite | flenders: Hopefully you can get it sorted. |
| 01:44.19 | flenders | JT: well, it would be pretty good if we could do that |
| 01:44.38 | JT | flenders: what is the main advantage, from your viewpoint? |
| 01:44.39 | flenders | if telstra won't do it, then we'll have to live with that |
| 01:44.53 | dendrite | flenders: Maybe ask counsel if the telco is publishing false information? |
| 01:45.10 | flenders | JT: call back, calling clients, calling staff |
| 01:45.16 | JT | ah ok |
| 01:45.34 | JT | flenders: my only suggestion that will definately work is going digital |
| 01:46.19 | flenders | JT: yeah, if ISDN cards weren't so expensive, we probably would |
| 01:47.33 | JT | depends, single bri cards are pretty cheap |
| 01:47.59 | JT | multibri cards can get expensive, i know :) |
| 01:48.39 | JT | but they're still at least equivalent in price to a TDM400P on a per line/B channel comparison basis |
| 01:49.00 | demigod2k | and still way way way way way cheaper than a traditional PBX |
| 01:50.06 | [hC] | anyone using hylafax for email-to-fax? |
| 01:50.08 | demigod2k | my music on hold sounds like garbage when you call from a cellphone. any suggestions? |
| 01:50.14 | demigod2k | hc, not me sorry |
| 01:50.32 | [hC] | trying to figure out how to get it to get confirmation on successfully sent faxes.. i get reports on errors only right now |
| 01:51.10 | ManxPower | demigod2k: That is pretty common. Music that is more musical and less lyrical tends to sound pretty bad over GSM codec. |
| 01:51.17 | ManxPower | or most any compressed codec |
| 01:51.28 | *** join/#asterisk flenders_ (n=fserto@unaffiliated/flenders) |
| 01:51.45 | demigod2k | ya its sprint PCS in particular in this case. I figured there was almost no way around it. their silence detection is really aggressive with music |
| 01:52.05 | *** join/#asterisk perd (i=[U2FsdGV@207.44.158.6) |
| 01:52.43 | ManxPower | demigod2k: try different music. It DOES make a difference. |
| 01:52.56 | dendrite | demigod2k: Yeah, try some punk! =) |
| 01:52.58 | ManxPower | I Want Rock-n-Roll by Joan Jett sounded pretty good. |
| 01:52.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:53.09 | ManxPower | mostly lyrics, not a lot of instruments |
| 01:53.10 | demigod2k | had a santana CD previously, total garbage. switched to blues brothers and it works marginally better |
| 01:53.11 | JT | demigod2k: does it just sound bad? or is it cutting in and out? |
| 01:53.14 | *** join/#asterisk perd (i=[U2FsdGV@207.44.158.6) |
| 01:54.02 | demigod2k | JT: landline sounds fine, from a sprint cellular cutting in-an-out and terrible sound |
| 01:54.16 | blitzrage | hey all!.... I'd like to conduct a poll: The 'h' extension is a reliable method of cleaning up channels: a) all the time, b) in most cases, c) in few cases, d) in no cases, e) it's file's fault |
| 01:54.46 | blitzrage | s/cleaning up channels/cleaning up after channels (such as database cleanup, etc...) |
| 01:55.03 | infinity1 | i have a few polycom phones in the office and the pbx is outside the office so we're using NAT. Obviously we're having problems getting the polycoms to work 100%. Is there a solution? |
| 01:55.20 | JT | demigod2k: cutting in and out can probably be fixed with different music or volume levels, sounding terrible, not much you can do |
| 01:55.28 | blitzrage | infinity1: you're having problems? What does your sip.conf look? |
| 01:55.40 | JT | demigod2k: low bandwidth voice codecs are designed for voice signalls, not music |
| 01:55.43 | demigod2k | JT: ya. its already loud. I'm going to try turning it down a few dB tomorrow just in case. |
| 01:55.51 | infinity1 | blitzrage: the phone works fine if there is ONE polycom. as soon as we get more, things go crazy. |
| 01:56.08 | JT | a lot of music appears as background noise to a vocoder |
| 01:56.13 | [TK]D-Fender | infinity1 : Set each onto its ownSIP signalling port and unique RTP range. |
| 01:56.21 | blitzrage | infinity1: what [TK]D-Fender said |
| 01:56.24 | demigod2k | ya I'm not totally surprised by that. sprint's silence detection seemed to make no sense to me :) |
| 01:56.42 | JT | it's looking for voice, not music |
| 01:56.50 | JT | they're very different |
| 01:56.58 | infinity1 | [TK]D-Fender: i thought that might be the answer. do we to setup the NAT /firewal to forward ports? |
| 01:57.15 | demigod2k | JT: fortunately, other than that, I've been really quite happy with it. we replaced a panasonic system with 2 weeks notice during an office move |
| 01:57.22 | [hC] | god damn hylafax is a botched piece of crap. |
| 01:57.31 | [hC] | its too bad it works well, or id ditch it |
| 01:57.54 | flenders_ | demigod2k: we're doing the same, but it's a NEC system |
| 01:58.03 | flenders_ | we're moving in 2 weeks. |
| 01:58.12 | demigod2k | flenders_: good luck with it. I ended up buying an off-the-shelf system from thevoipconnection |
| 01:58.35 | demigod2k | the move was pretty miserable, but we were next door to a daycare which was probably even worse to begin with |
| 01:59.24 | perd | haha hc |
| 01:59.40 | perd | hylafax is awesome, what botched problems are you having |
| 01:59.45 | [TK]D-Fender | infinity1 : should need to forward. |
| 01:59.56 | [TK]D-Fender | shouldn't* |
| 02:00.41 | demigod2k | flenders_: my only advice after all that is to try and keep a homogenous system. buying all the same model of polycom (301) helped things along |
| 02:00.44 | blitzrage | I have 6 phones behind a NAT with no forwarding |
| 02:00.52 | blitzrage | nat=yes |
| 02:01.17 | [hC] | perd: shit like trying to tell faxemail to always send out one fax machine, instead of randomly picking an available fax machine, as each has a different baner w/ diff company name... or, getting confirmation emails sent to the SENDER that their email-to-fax job went thru |
| 02:01.24 | [hC] | just stupid shit like this.. is such a pain. |
| 02:01.43 | flenders_ | demigod2k: I installed it myself, and the basic PBX funcitions are all working |
| 02:01.50 | perd | oh, i write custom scripts to do all that |
| 02:02.01 | infinity1 | [TK]D-Fender: k. will give it a shot. |
| 02:02.10 | infinity1 | tnx for confirming what i didn't want to hear :) |
| 02:02.14 | flenders_ | demigod2k: we'll have, at least, the same as we have now |
| 02:02.20 | [hC] | perd: so do i. and it sucks. |
| 02:02.26 | perd | works well for me |
| 02:02.45 | [hC] | working is not the issue, i just said it works fine |
| 02:02.52 | perd | i think im going to try out app_txfax and app_rxfax though |
| 02:02.58 | perd | i had to downgrade to 1.2 for friggen chan_sccp |
| 02:03.05 | [hC] | i said its a botched pain in the ass, the crap you have to go thru to get to a final stage. |
| 02:03.26 | perd | heh, i didnt find it all that difficult to set up |
| 02:03.31 | [hC] | i tried rxfax and txfax... i wasnt super happy with those either.. they had some reliability issues |
| 02:03.43 | perd | do you use hylafax with iaxmodem? |
| 02:03.48 | [hC] | yes. |
| 02:03.54 | perd | yeah, that seems to work well |
| 02:04.08 | [hC] | Its not difficult to set up, its just a pain. the way they laid it out is a mess. |
| 02:04.27 | perd | i have used hylafax for several years so i guess i'm used to it |
| 02:04.35 | perd | it's no worse than any other unix based faxing software |
| 02:05.23 | flenders_ | JT: what sort of card would I need if I would go with ISDN? |
| 02:05.27 | [hC] | yeah the problem is they're all designed to work with legacy old crap |
| 02:05.33 | [hC] | hylafax is what, 10 years old? |
| 02:05.39 | perd | something like that |
| 02:06.05 | [hC] | do you have any idea how to instruct hylafax to send out a specific faxmodem? |
| 02:06.10 | [hC] | I still have yet to figure that out |
| 02:06.14 | perd | yeah |
| 02:06.19 | JT | flenders_: either a multiport cologne card like a Junghanns quadBRI or OctoBRI, or a few HFC Cologne single port cards (which are cheap as, available from lots of manufacturers) |
| 02:06.21 | perd | that's how i do my print to fax |
| 02:06.30 | perd | one sec i'll paste the line |
| 02:06.50 | [hC] | perd: have a suggestion on where to look? I need to be able to filter email so that certain customers emailing in from email-to-fax get queued into THEIR fax modem |
| 02:06.52 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:07.19 | flenders | JT: any resellers you'd know of? |
| 02:07.39 | JT | not off hand |
| 02:07.47 | JT | none of the ones for sale in .au are cheap |
| 02:07.59 | rpm | has anyone here done much reviewing of conferencing apps.. meetme, sems, sipx boston-bridge? |
| 02:08.01 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 02:08.07 | flenders | JT: I was afraid you were gonna say that |
| 02:08.18 | JT | just buy them from overseas |
| 02:08.19 | JT | problem solved :) |
| 02:08.20 | perd | sendfax -h ttyIAX01@localhost |
| 02:08.23 | perd | sendfax -h ttyIAX02@localhost |
| 02:08.30 | perd | sendfax -h ttyIAXjenny@localhost |
| 02:08.43 | perd | uses the modem name @ the host |
| 02:08.50 | perd | then you pass the other arguments as normal |
| 02:09.04 | perd | sendfax -h ttyIAX02@localhost -n -d 911 saveus.pdf |
| 02:09.07 | [hC] | aha |
| 02:09.16 | [hC] | fantastic |
| 02:09.32 | [hC] | thanks. |
| 02:09.35 | perd | np |
| 02:09.48 | [hC] | do you use faxmail for email->fax gateway? |
| 02:09.59 | perd | nah, im not fond of email to fax |
| 02:10.04 | perd | i force my people to print to fax |
| 02:10.11 | [hC] | wish i could :) im not doing one client |
| 02:10.17 | [hC] | im doing 50+ |
| 02:10.42 | perd | yeah i dunno, i would probably use fetchmail or procmail for that |
| 02:11.11 | perd | just set up a mailbox that is checked every minute |
| 02:11.24 | [hC] | i might just ditch hylafax's included faxmail script |
| 02:11.36 | [hC] | the output is horrid, and its kinda kludgy |
| 02:11.42 | [hC] | not very much flexibility |
| 02:12.09 | perd | yeah, the ones i use i replace, like faxrcvd |
| 02:15.35 | [hC] | ya i worked on the notification script so it outputs a ton differently |
| 02:17.53 | perd | i have mine set up to email the faxes based on the tty they come in on |
| 02:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:19.20 | [hC] | yup, thats how i deliver received faxes |
| 02:20.30 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:23.14 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 02:29.26 | *** join/#asterisk `Sean (i=Un1x@CPE000c148d127c-CM00140458831c.cpe.net.cable.rogers.com) |
| 02:33.35 | *** join/#asterisk mog (n=mog@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:33.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:41.48 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 02:42.17 | *** part/#asterisk mog (n=mog@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:46.05 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 02:57.42 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 03:03.12 | rpm | jesus i wish i took comp. sci, i cannot put the pieces together of mixing multiple rtp streams |
| 03:06.13 | *** join/#asterisk welby (n=welby@ivonova.whmcr.com) |
| 03:07.30 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 03:14.43 | *** join/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com) |
| 03:14.56 | connecta | how do i pipe sip debug messages to a log file |
| 03:15.19 | Nugget | in another window do asterisk -rvvv > logfile.log |
| 03:15.29 | Nugget | it's not perfect but it works. |
| 03:17.36 | [hC] | that is a really bad idea. |
| 03:17.37 | [hC] | :) |
| 03:17.51 | [hC] | /var/log/asterisk/full (on a lot of distros) contains a full debug log |
| 03:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:18.27 | Nugget | that's also a really bad idea, but in different ways. :) |
| 03:18.39 | [hC] | yep :) |
| 03:18.58 | connecta | why are they bad ideas |
| 03:21.53 | *** join/#asterisk doolph (n=doo@200.124.21.169) |
| 03:23.17 | jtexter3 | anyone online have experience with the Audiocodes Mediant gateways? |
| 03:23.57 | blitzrage | I don't like them |
| 03:23.59 | blitzrage | pain in the butt |
| 03:24.08 | nick125_lappy | connecta: Well, one reason might be because it would fill your disk with a ton of logs |
| 03:24.54 | jtexter3 | I have it working 99%. But I'm having trouble with ringback when doing an attended transfer of a call that goes through the gateway |
| 03:25.21 | jtexter3 | blind transfer works perfect, attended transfer results in no audio until the 3rd party answers, or goes to voicemail |
| 03:25.36 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 03:25.57 | doolph | are you using r? |
| 03:26.20 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 03:27.18 | jtexter3 | doolph: I tried r, but no luck |
| 03:27.28 | blitzrage | 'r' just simulates the ringing |
| 03:27.35 | blitzrage | you should really be getting the ringing indication |
| 03:27.54 | jtexter3 | Looking at the debug output on the gateway, it looks like it's because Asterisk sends a BYE to the gateway when you complete the attended transfer |
| 03:28.09 | jtexter3 | Originally, the gateway was hanging up the PSTN call |
| 03:29.11 | danp | are there any major asterisk conferences besides astricon? |
| 03:29.46 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:29.54 | [TK]D-Fender | danp : Cluecom |
| 03:29.56 | [TK]D-Fender | danp : Cluecon* |
| 03:30.33 | jtexter3 | I really just don't know where to start on the troubleshooting with this gateway. This darn attended transfer is the only thinging keeping me from having the gateway in production |
| 03:31.06 | ManxPower | jtexter3: So, no hold music? |
| 03:31.24 | jtexter3 | ManxPower: hold music works. |
| 03:31.30 | ManxPower | In attended transfers the person that is being trasfered would normally hear hold music |
| 03:32.07 | jtexter3 | ManxPower: yep, that part works. When I complete the attended transfer( using polycom 501's, so pressing transfer the second time), hold music stops, and I just get dead air |
| 03:32.37 | *** part/#asterisk orlok (i=[IHSJAyU@202.44.174.4) |
| 03:35.00 | ManxPower | jtexter3: have you done mailing list searches? |
| 03:35.55 | jtexter3 | ManxPower: I've been trying google, but either no one has hit this, or i haven't come up with the right search phrase |
| 03:36.10 | jtexter3 | I tried sending an email the asterisk-users, but it appears there is no mailing list traffic today.... |
| 03:38.01 | jtexter3 | s/email the/email to/g |
| 03:39.54 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 03:42.37 | EyeCue | hmm gstreamer codecs |
| 03:43.52 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-160-147.dsl.irvnca.pacbell.net) |
| 03:54.32 | connecta | will someone please look at a debug and let me know why my incoming calls go to oblivion... |
| 03:54.46 | connecta | oh my god |
| 03:55.13 | [TK]D-Fender | connecta : ;;; |
| 03:55.14 | [TK]D-Fender | ~pb |
| 03:55.26 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:59.22 | connecta | nvm, i got it, im a jackass |
| 04:00.33 | *** part/#asterisk connecta (n=Administ@175.6.188.72.cfl.res.rr.com) |
| 04:09.28 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 04:09.57 | rbd | hey guys, with meetme is there a way to get confirmation on the room number entered (if not specifying a room number and causing meetme to prompt you for one)? |
| 04:10.22 | De_Mon | what do you mean? |
| 04:10.53 | De_Mon | 1) enter number 2) enter number again 3) if numbers dont match have meetme prompt you for a number? |
| 04:11.02 | rbd | you get the "please enter your conference ID followed by # sign" menu...enter the conf ID, then it normally just throws you into the room |
| 04:11.33 | rbd | no, instead it will say "you entered conference ID xxxx, press 1 if this is correct, press 2 to re-enter your conf ID" |
| 04:11.35 | rbd | something like that |
| 04:11.49 | [TK]D-Fender | rbd : jsut make you're own little IVR in front of it. no big deal. |
| 04:12.03 | De_Mon | you can do that in the dialplan and then throw them into MeetMe(${ROOM}|args) |
| 04:12.11 | [TK]D-Fender | ^^^^^ |
| 04:12.34 | De_Mon | you can do anything with a dialplan and enough creativity |
| 04:13.01 | rbd | okay, sounds good |
| 04:13.24 | De_Mon | just remember there are any number of ways to impliment the idea and most of them are right :) |
| 04:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 04:19.21 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 04:20.57 | *** join/#asterisk AtomicStack (n=matt@malibu.ecomtel.com.au) |
| 04:27.20 | arcanine | ` |
| 04:29.45 | Corydon76-home | ~seen shadowhntr |
| 04:30.16 | jbot | shadowhntr <i=sentinel@wikipedia/Shadowhntr> was last seen on IRC in channel #asterisk, 4d 11h 53s ago, saying: 'sweeper: haven't tried it lately? :P'. |
| 04:33.12 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 04:33.42 | perd | anyone know why when i transfer someone to park, then i pick them up i cant use feature #1 (transfer) again? |
| 04:35.19 | *** join/#asterisk fastfinge-deskto (n=samuel@interfree.ca) |
| 04:36.15 | *** part/#asterisk fastfinge-deskto (n=samuel@interfree.ca) |
| 04:36.53 | [TK]D-Fender | perd : What kind of phone? |
| 04:36.53 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 04:47.18 | *** join/#asterisk CrashHD (n=crashhd@67.182.167.222) |
| 04:48.37 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 04:50.46 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 04:51.37 | perd | [TK]D-Fender: sip phone, cisco 79XX, whatever |
| 04:51.51 | perd | damn nick completion |
| 04:52.00 | *** join/#asterisk paavum (n=dorphals@pcsp163-73.supercabletv.net.co) |
| 04:52.02 | paavum | Hello |
| 04:52.42 | dlynes_laptop | ceshia galee |
| 04:52.58 | paavum | I was wondering if I can get an asterisk server to do video streaming ... like connect to a h264 streaming server and then rebroadcast it |
| 04:53.02 | CrashHD | ceshkabob what? |
| 04:53.33 | dlynes_laptop | shishkabob |
| 04:53.43 | CrashHD | :) |
| 04:53.50 | paavum | wtf u speaking about? :P |
| 04:53.57 | CrashHD | rambling |
| 04:54.00 | rbd | is there a command to play a prompt and gather any DTMF input into a variable that can be used in the asterisk diaplan (I see Background but that will transfer the caller to the extension they entered) |
| 04:54.01 | dlynes_laptop | paavum, guess you're not punjabi :) |
| 04:54.11 | CrashHD | rbd: Read() |
| 04:54.18 | rbd | thanks |
| 04:54.25 | paavum | Nope, but I can speak spanish if ya want |
| 04:54.39 | CrashHD | no habla espanol mi amigo |
| 04:54.46 | dlynes_laptop | nah...just figured you might be indian because of your nick |
| 04:55.03 | dlynes_laptop | a double a is often indicative of an indian name |
| 04:55.34 | paavum | hehehe nope, thats just "turkey" in my own variation of latin |
| 04:55.48 | dlynes_laptop | ah |
| 04:56.29 | paavum | but now I have to think what a turkey could be doing in india |
| 04:56.29 | dlynes_laptop | anyways...i was just more or less saying welcome in punjabi :0 |
| 04:56.44 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:56.46 | joelsolanki | Hi all |
| 04:56.48 | dlynes_laptop | paavum, getting eaten :) |
| 04:56.48 | CrashHD | hah |
| 04:56.53 | dlynes_laptop | speaking of indians |
| 04:56.54 | paavum | lol |
| 04:56.58 | CrashHD | internationally friendly |
| 04:57.21 | joelsolanki | Hi daniel :) |
| 04:57.25 | joelsolanki | Jan 16 10:17:57 NOTICE[27624]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 04:57.31 | joelsolanki | i get this on asterisk console. |
| 04:57.36 | dlynes_laptop | Yeah...no big deal |
| 04:57.38 | dlynes_laptop | Just ignore it |
| 04:57.42 | dlynes_laptop | I get it all the time, too |
| 04:57.43 | joelsolanki | hmm ok. gr8 |
| 04:57.46 | paavum | as a rule of thumb |
| 04:57.47 | bkruse_home | joelsolanki: i think they call it a notice for a reason :P |
| 04:57.49 | paavum | you can ignore notices |
| 04:58.01 | dlynes_laptop | It's because asterisk doesn't support voice activity detection |
| 04:58.07 | paavum | as a rule of thumb you acn ignore everything thats not a warning |
| 04:58.14 | bkruse_home | notice just says "hey, check this out, probably useful for debugging......i dont even know why i exist!" |
| 04:58.17 | joelsolanki | yes agree. but it can suddenly so thought to just what it is . |
| 04:58.20 | joelsolanki | :) |
| 04:58.26 | dlynes_laptop | and you're using the g729 annex that's trying to use it from another device that supports it |
| 04:58.26 | joelsolanki | how are u doing dlynes ? |
| 04:58.32 | JT | paavum: ignore everything that's not a warning... like an "error"? |
| 04:58.33 | dlynes_laptop | good |
| 04:58.34 | joelsolanki | hmm ok. |
| 04:58.41 | dlynes_laptop | JT, Good idea! |
| 04:58.42 | bkruse_home | JT: agreed. |
| 04:58.51 | paavum | JT --> Yeah |
| 04:58.56 | JT | hah |
| 04:59.07 | paavum | u sewe |
| 04:59.17 | paavum | warnings are what you can eventually fix with hard work |
| 04:59.25 | dlynes_laptop | even then |
| 04:59.26 | paavum | errors just say "man u're f*cked" |
| 04:59.35 | dlynes_laptop | some of the warnings in asterisk are mislabelled notices |
| 04:59.45 | JT | warnings mean "doesn't need fixing" errors mean "doesn't work" |
| 05:00.20 | *** join/#asterisk SwK_ (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 05:00.26 | dlynes_laptop | paavum, warnings are usually because you haven't met a requirement, and it's asterisk's way of telling you to wise up |
| 05:00.29 | bkruse_home | yep, but pay attention to the ones that say "YOU SHOULD NEVER SEE ME!!" |
| 05:00.43 | dlynes_laptop | bkruse_home, NO!!!! |
| 05:00.45 | paavum | bkruse_home --< NO! |
| 05:01.13 | paavum | uhhhh |
| 05:01.34 | paavum | You take note that you shouldn't be seeing them and act accordingly |
| 05:01.56 | [TK]D-Fender | perd : If you're using a Cisco, you shouldn't be using DTMF for transfers.... |
| 05:01.57 | bkruse_home | if(HACKER == "true") |
| 05:01.57 | bkruse_home | { |
| 05:01.57 | bkruse_home | ast_log(LOGWARNING, "YOU SHOULD NEVER SEE ME!!!" |
| 05:01.57 | bkruse_home | } |
| 05:02.11 | bkruse_home | :X |
| 05:03.16 | bkruse_home | ~seen bkruse |
| 05:03.36 | jbot | bkruse <i=bkruse@nat/digium/x-c4bcfeee8bef66f6> was last seen on IRC in channel #asterisk, 4d 9h 6m 28s ago, saying: 'mplayer?'. |
| 05:03.36 | [TK]D-Fender | bkruse_home : Very mal-formed :) "hacker" should be a boolean, which you couldn't compare to a string. At which point you wouldn't even NEED a comparison. so - if(HACKER == "true") { ast_log(LOGWARNING, "YOU SHOULD NEVER SEE ME!!!"); } |
| 05:03.38 | [TK]D-Fender | bkruse_home : (also corrected your missing bracket) |
| 05:03.40 | [TK]D-Fender | :O |
| 05:03.40 | bkruse_home | [TK]D-Fender: :P |
| 05:03.57 | [TK]D-Fender | bkruse_home : Very mal-formed :) "hacker" should be a boolean, which you couldn't compare to a string. At which point you wouldn't even NEED a comparison. so - if(HACKER) { ast_log(LOGWARNING, "YOU SHOULD NEVER SEE ME!!!"); } |
| 05:04.05 | [TK]D-Fender | (missed the DELETE for that bad part :) |
| 05:04.08 | [TK]D-Fender | aak;sdkja;skldj;as;dasd |
| 05:04.11 | [TK]D-Fender | gah |
| 05:04.18 | bkruse_home | ha, close enough |
| 05:04.20 | bkruse_home | :P |
| 05:04.32 | [TK]D-Fender | bkruse_home : Yeah, I'll leave it at that :) |
| 05:04.35 | dlynes_laptop | [TK]D-Fender, he's used to perl :) |
| 05:04.39 | bkruse_home | :P |
| 05:04.52 | bkruse_home | I should have started with C instead of a scripting language |
| 05:04.54 | [TK]D-Fender | bkruse_home : I was king of "perverse (ab)use of boolean logic"/ |
| 05:05.01 | bkruse_home | :P |
| 05:05.13 | bkruse_home | haha, thats quite a role to fill |
| 05:05.23 | bkruse_home | but im learning :] |
| 05:05.24 | [TK]D-Fender | bkruse_home : s'ok. I wrote a language in Turbo Pascal back in the day :) The worlds first mid-level language. |
| 05:05.36 | bkruse_home | woot! |
| 05:05.59 | file | Strom_C: eep |
| 05:06.04 | [TK]D-Fender | bkruse_home : yeah, it was craptastic. Mind you its starting to look a LOT like extensions.conf, based on the GotoIf section... |
| 05:06.06 | file | I'm telling on you |
| 05:06.13 | bkruse_home | LOL |
| 05:06.14 | Strom_C | fine! |
| 05:06.19 | bkruse_home | [TK]D-Fender: i know how that is, trust me |
| 05:06.20 | [TK]D-Fender | bkruse_home : actually... it never really did more than 1 operation at a time... |
| 05:06.31 | bkruse_home | [TK]D-Fender: i shouldnt have started with scripting in bash though.....its spoiled me |
| 05:07.06 | [TK]D-Fender | file : ..... your continued ambiguity may arouse bkw_ .... take heed ;) |
| 05:07.33 | file | [TK]D-Fender: SNOW |
| 05:07.38 | [TK]D-Fender | bkruse_home : I don't know bash at all really. only the most very basic. SUB basic even.... |
| 05:07.46 | [TK]D-Fender | file : JSDLSLhkl;hs;dhf;hg;fdkg'hfdkj |
| 05:07.53 | bkruse_home | [TK]D-Fender: thats what i grew up in unfortunatly. |
| 05:07.56 | [TK]D-Fender | file : TONS of the friggen white shit! |
| 05:08.02 | bkruse_home | its good for dirty things, but not at all for any kind of real use |
| 05:08.35 | bkruse_home | unforunately |
| 05:08.36 | [TK]D-Fender | bkruse_home : Perl <- all that is good AND bad in the *nix world. Great to lrean from. |
| 05:08.46 | bkruse_home | [TK]D-Fender: i agree. |
| 05:08.56 | bkruse_home | i dabbled in perl as my first book to read, but never really used it |
| 05:08.58 | [TK]D-Fender | (or so I hear. I don't do Perl eiher) |
| 05:09.12 | bkruse_home | i always found perl as another dirty scripting language, far more powerful than bash though |
| 05:09.22 | file | farrrrrr |
| 05:09.28 | bkruse_home | file: agreed |
| 05:09.47 | dendrite | Perl... The Swiss Army Chainsaw of Scripting. |
| 05:10.01 | bkruse_home | but i figure, php for my scripting/web stuff and C for everything else will suite me fine |
| 05:10.05 | Strom_C | <camel> There's More Than One Way To Do Me |
| 05:10.25 | file | trying to adjust your glasses when you aren't wearing them doesn't work too well |
| 05:10.39 | bkruse_home | file: lies! |
| 05:12.06 | bkruse_home | [TK]D-Fender: i just thought that it would probably be better to just do if (hacker) to see if hacker exists no? |
| 05:13.42 | bkruse_home | ~seen blitzrage |
| 05:13.44 | jbot | blitzrage is currently on #asterisk-doc (3h 20m 45s). Has said a total of 29 messages. Is idling for 1h 46m 9s, last said: 'you should really be getting the ringing indication'. |
| 05:14.00 | bkruse_home | hmm |
| 05:14.09 | *** join/#asterisk zeeesh (i=aadilism@9-237-154-202.wol.net.pk) |
| 05:14.28 | zeeesh | hi |
| 05:15.20 | [TK]D-Fender | dendrite : http://www.potse.nl/tex.7.png |
| 05:15.53 | dendrite | [TK]D-Fender: :-) |
| 05:16.31 | [TK]D-Fender | file : I had a phantom-glasses moment today where if felt like they needed to be pushed back up the bridge.... |
| 05:16.55 | [TK]D-Fender | file : And then it like... hot-damn I don't have glasses! |
| 05:17.37 | file | lol |
| 05:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:18.48 | bkruse_home | well im off to bed |
| 05:18.49 | bkruse_home | night all |
| 05:19.40 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:20.33 | file | there goes the snow plow... |
| 05:21.05 | [TK]D-Fender | file "Mr. Plow, thats my name. That name again is Mr. Plow!" |
| 05:22.11 | [TK]D-Fender | wunderkin : Get the ^%#$ off my telepathic frequency or I'll have the FCC give you an invasive "scan". This time WITHOUT the K-Y :D |
| 05:22.28 | wunderkin | [TK]D-Fender <3 |
| 05:22.58 | wunderkin | look for a pot, they work better |
| 05:23.22 | Strom_C | I'm a would-be |
| 05:23.25 | Strom_C | w o o d |
| 05:23.32 | Strom_C | I'm a would-be would-be |
| 05:23.33 | Strom_C | b e e z |
| 05:23.35 | Strom_C | etc |
| 05:23.46 | [TK]D-Fender | Strom_C : "You're not a has-been, you're a never-was" |
| 05:23.57 | Strom_C | that's not how the song goes! |
| 05:24.19 | [TK]D-Fender | Strom_C : Its called "improvisation". or "stylized" even ;) |
| 05:24.42 | [TK]D-Fender | Strom_C : perhaps a "medley"? |
| 05:24.51 | Strom_C | catsex |
| 05:26.17 | [TK]D-Fender | -> C Texas? |
| 05:26.28 | [TK]D-Fender | I do :) |
| 05:26.37 | [TK]D-Fender | Dyslexics of the world... untie! |
| 05:27.07 | dendrite | Dylsexics? |
| 05:27.20 | Strom_C | Dyslexus |
| 05:28.05 | CrashHD | hey there strom |
| 05:28.05 | [TK]D-Fender | http://dictionary.reference.com/browse/Dyslexics |
| 05:28.25 | Strom_C | hi |
| 05:28.30 | CrashHD | how goes it |
| 05:28.39 | Strom_C | peachy |
| 05:28.46 | CrashHD | not apply? |
| 05:28.49 | CrashHD | appley? |
| 05:28.59 | CrashHD | lol |
| 05:29.09 | [TK]D-Fender | This channel has gone Fruit Loops..... |
| 05:29.14 | Strom_C | it was going more pomegranatey, actually |
| 05:29.18 | CrashHD | hah |
| 05:29.26 | CrashHD | where are my damn lucky charms |
| 05:30.11 | CrashHD | I wasn't able to return to the comp last night |
| 05:30.14 | [TK]D-Fender | "If ever I reach the end of the rainbow, as good fortune did intend, Murphy would be there to tell me the pot's at the OTHER end." |
| 05:30.23 | CrashHD | hah |
| 05:30.28 | CrashHD | murphy and his damn laws |
| 05:30.44 | Strom_C | he should stick to furniture products |
| 05:30.49 | codefreeze | Tell me about it! (steve murphy here!) |
| 05:31.05 | CrashHD | a murphy with the alias of codefreeze |
| 05:31.10 | CrashHD | irony...? |
| 05:31.11 | CrashHD | I think so |
| 05:32.23 | [TK]D-Fender | CrashHD : Who would by a book labeled "Murphy's Suggestions"? No, people need to be TOLD what they think! |
| 05:32.43 | CrashHD | hah I agree, very true. |
| 05:32.59 | codefreeze | CrashHD: Ah, be careful of lightly treating Murphy's Laws. You will someday understand their importance in life and their role in its purpose. |
| 05:33.40 | CrashHD | they bite me in the ass all the time...I of all people do understand |
| 05:33.54 | file | having codefreeze as a coworker just makes Murphy's Laws kick in even MORE! |
| 05:34.00 | CrashHD | HAHA |
| 05:34.33 | codefreeze | Yes, time and space do seem at times to bend in my presence. I will not bore you with my tales of woe! |
| 05:34.41 | CrashHD | but what about crashes paradox....the one that poses the question if murphy would have been shot before him and his damn laws....would the laws still apply! |
| 05:35.03 | CrashHD | I choose to think not |
| 05:35.09 | CrashHD | *laugh* |
| 05:35.32 | CrashHD | :) |
| 05:35.35 | codefreeze | No, of course they would still be! They'd just be labeled as "McGillicutty's Laws" or whatever! |
| 05:35.47 | file | codefreeze: if you're going to San Diego be sure to wear some flowers in your hair |
| 05:36.00 | CrashHD | hah |
| 05:36.02 | CrashHD | ohhh flowers |
| 05:36.13 | CrashHD | file is feeling frisky codefreeze...better watch yourself |
| 05:36.31 | [TK]D-Fender | codefreeze .... and thats just your gravitational field! |
| 05:36.31 | codefreeze | file: San Diego's not my kind of town; you know the rest! |
| 05:36.57 | CrashHD | hey fella's if you were building a dialplan macro for a multiple destination (simul. dial) follow me find me how would you do it? |
| 05:37.03 | [TK]D-Fender | :D |
| 05:37.21 | *** join/#asterisk conver2 (n=marc3234@206-248-153-49.dsl.teksavvy.com) |
| 05:37.22 | Strom_C | "The Complete Idiot's Guide For Dummies" |
| 05:37.24 | CrashHD | tk is that volume two?...you really should get your updated copy. |
| 05:37.34 | *** join/#asterisk bkw_ (n=brian@88-110-74-241.dynamic.dsl.as9105.com) |
| 05:37.34 | [TK]D-Fender | CrashHD : Simul dail is tricky when involving the PSTN. |
| 05:37.45 | [TK]D-Fender | CrashHD : Using a PRI at least I hope. |
| 05:37.49 | CrashHD | ya of course |
| 05:37.58 | file | [TK]D-Fender: analog partyline! |
| 05:38.05 | CrashHD | I built it using the dial() with M() |
| 05:38.09 | CrashHD | but I found out later |
| 05:38.12 | Strom_C | analog partyline ringdown circuit |
| 05:38.20 | CrashHD | that dial() considers a line picked up when it is picked up |
| 05:38.24 | CrashHD | not after a macro is completed |
| 05:38.28 | [TK]D-Fender | Strom_C : the problem with idiot-proofing everything is morons keep screwing it all up. |
| 05:38.31 | CrashHD | so it stops the other outbound channels |
| 05:38.35 | [TK]D-Fender | file : Whee! |
| 05:39.06 | Strom_C | CrashHD: didnt I already give you a solution to this problem? |
| 05:39.07 | [TK]D-Fender | CrashHD : Analog? You're SCREWED. |
| 05:39.14 | Strom_C | .call files, a meetme, and some macros |
| 05:39.25 | CrashHD | Strom_C, I wasn't able to make it back to the comp last night |
| 05:39.37 | CrashHD | no analog |
| 05:39.53 | CrashHD | ya |
| 05:39.59 | CrashHD | I guess meetme is the only way |
| 05:40.04 | [TK]D-Fender | Strom_C : trying the out-calls so know to halt the other attempts must be a real bitch, and worse if timing allows both to be answered when a mutual-kill event gets triggered and EVERYONE gets hung up on... |
| 05:40.27 | CrashHD | I'm probably going to agi this |
| 05:40.35 | codefreeze | Here's another philosophical question: how many of those "XXX for Dummies" books are there? How many are possible? |
| 05:40.36 | CrashHD | give me a bit more logic |
| 05:40.38 | [TK]D-Fender | s/trying/tying |
| 05:40.54 | Strom_C | codefreeze: stupid for dummies |
| 05:41.00 | Strom_C | drooling for dummies |
| 05:41.06 | CrashHD | dumies guide to reading dummies books |
| 05:41.07 | Strom_C | advanced neurosurgery for dummies |
| 05:41.08 | [TK]D-Fender | codefreeze : Ther are SOOOO many stupid people out there, I doubt we'll max out. |
| 05:41.27 | CrashHD | stupid people o' plenty |
| 05:41.53 | CrashHD | stupid people whom know they are stupid, few, and a sad fact at that |
| 05:41.59 | [TK]D-Fender | "two things are infinite: the universe and human stupidity" - Einstein |
| 05:42.47 | CrashHD | so strom you basically suggest opening up a meetme conference and dumping the first callee into it after they accept the call |
| 05:44.16 | [TK]D-Fender | CrashHD : its a Catch-22 psycholgical defense mechanism. If stupid people were truely aware of how stupid and useless they were, man-kind would not have a sufficient gene-pool (due to self-termination) to continue creating the 1 / 1,000,000 people whose IQ is befitting more than drinking from the toilet. |
| 05:44.34 | CrashHD | hah |
| 05:44.42 | CrashHD | sure would solve the overpopulation problem though |
| 05:44.50 | [TK]D-Fender | Holy crap I think too much...... |
| 05:44.58 | codefreeze | Good to see you guys have such an upbeat attitude!! |
| 05:45.08 | CrashHD | you should give that hamster a break tk |
| 05:45.18 | CrashHD | your hamster wheel needs some wd40 |
| 05:45.46 | [TK]D-Fender | CrashHD : The lights are on, the wheel is spinning but the hampster is just ^&%#ing DEAD, ok? :) |
| 05:45.52 | CrashHD | hah |
| 05:45.58 | CrashHD | lizards run faster anyway right? |
| 05:46.23 | CrashHD | ok |
| 05:46.31 | CrashHD | have my quote of retarded comments for the night |
| 05:46.36 | [TK]D-Fender | CrashHD : WD40 + duct tape - the only 2 tools in a real man's toolbox. If it moves and shouldn't : duct-tape. If it doesn't, but should : WD40. |
| 05:46.38 | CrashHD | quota |
| 05:46.40 | x86 | anyone know anything about DVI cables? |
| 05:46.56 | CrashHD | if you can't duct it...fek it |
| 05:47.12 | [TK]D-Fender | x86 : gota more specific question? they are used by MONITORS. next! |
| 05:47.34 | CrashHD | and are a replacement for dsub calbes |
| 05:47.43 | dlynes_laptop | [TK]D-Fender, you scared CP away |
| 05:47.45 | CrashHD | and hook up to cool tv's |
| 05:47.50 | [TK]D-Fender | Oddly enough one thing any plumber will tell you is that duct tape is useful for all sorts of stuff EXCEPT sealing ducts.... |
| 05:47.58 | CrashHD | hah |
| 05:48.00 | x86 | [TK]D-Fender: well, i bought a DVI cable at wall-mart, and it's missing a set of 9 pins (by my count) right in the middle of the cable, and on both ends |
| 05:48.02 | CrashHD | soooo true |
| 05:48.09 | CrashHD | to seal ducts you use beutal tape |
| 05:48.22 | dlynes_laptop | x86, don't buy it at walmart, then :) |
| 05:48.23 | CrashHD | that is what you get for buying cables at walmart |
| 05:48.41 | [TK]D-Fender | x86 : Could be an exclusively DVI-D (digital only). DVI can simultaneously support an analog signal IIRC. |
| 05:48.54 | CrashHD | pshhhhhh |
| 05:49.03 | CrashHD | no real answers here.....that's the rule |
| 05:49.10 | CrashHD | hah |
| 05:49.12 | [TK]D-Fender | :O |
| 05:49.31 | [TK]D-Fender | Ask a stupid ascii, get a stupid ansi ;) |
| 05:49.33 | CrashHD | only condesending rederric |
| 05:49.52 | CrashHD | damn I have to format this damn comp tonight |
| 05:49.56 | x86 | [TK]D-Fender: yeah, DVI-I combines both analog and digital... |
| 05:49.56 | CrashHD | sooooo not looking forward to it |
| 05:49.59 | dlynes_laptop | asterisk seems to be getting a lot of press lately |
| 05:50.09 | dlynes_laptop | php architect even had an article on it this month |
| 05:50.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:50.32 | CrashHD | I want to see an opensource class 5 softswitch |
| 05:50.35 | [TK]D-Fender | CrashHD : "rhetoric" |
| 05:50.45 | CrashHD | you'll have to excuse my spelling |
| 05:50.59 | dlynes_laptop | [TK]D-Fender, spelling skills are not normally associated with Americans |
| 05:51.04 | CrashHD | it's not a lack of intelligence but more or less a lack of caring |
| 05:51.16 | CrashHD | ya what dlynes said |
| 05:51.22 | CrashHD | hah |
| 05:51.37 | CrashHD | pulls out the give a shit nuke and laughs as the mushroom cloud goes up |
| 05:51.42 | CrashHD | :) |
| 05:51.58 | [TK]D-Fender | Today's magic word is "Ebonics" |
| 05:52.18 | CrashHD | how many languages do you speek tk? |
| 05:52.22 | CrashHD | speak |
| 05:52.37 | dlynes_laptop | [TK]D-Fender, I guess he didn't understand the joke :) |
| 05:52.38 | [TK]D-Fender | CrashHD : 32 empty missile silos, a mushroom could on the horizon ; NOW its Miller Time! |
| 05:52.50 | CrashHD | hah |
| 05:53.06 | CrashHD | now I feel retarded...going to have to go download a dictionary applet |
| 05:53.07 | [TK]D-Fender | CrashHD : Several, most practical is my fluency in Gibberish :) |
| 05:53.27 | *** join/#asterisk mog_home (n=rah@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:53.27 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 05:53.40 | dlynes_laptop | CrashHD, ebonics is black america's version of English |
| 05:53.49 | CrashHD | hahah |
| 05:53.52 | CrashHD | yes, this I knew |
| 05:53.55 | CrashHD | I meant in general |
| 05:54.15 | CrashHD | I'm not going to be able to have a conversation in here without making sure my spelling is correct |
| 05:54.27 | CrashHD | ebbbowhat |
| 05:54.30 | CrashHD | lol |
| 05:54.38 | [TK]D-Fender | dlynes_laptop : And thanks to the like of Eminim, Snow, Vanilla Ice, K-Fed, JT, and crew, soon to hit critical mass amongst white-trash America! |
| 05:54.45 | CrashHD | hah |
| 05:54.47 | dlynes_laptop | hehehee |
| 05:54.48 | CrashHD | ohh so true |
| 05:54.59 | dlynes_laptop | JT, you gonna take that? |
| 05:55.12 | CrashHD | alright off I go to start this damn format |
| 05:55.18 | CrashHD | catch you fellas later |
| 05:55.22 | dlynes_laptop | latas |
| 05:55.35 | CrashHD | p.s. no comments on my class 5 softswitch idea? |
| 05:55.38 | CrashHD | come on |
| 05:55.59 | dlynes_laptop | no idea what the diff is between a class 5 and a class 1 softswitch, personally |
| 05:56.13 | CrashHD | class 5 sounds cool? |
| 05:56.14 | JT | dlynes_laptop: arg :( |
| 05:56.25 | JT | [TK]D-Fender: s/JT/Justin Timberlake/ damnit :] |
| 05:56.55 | dlynes_laptop | Is that what your nick is an abbreviation for? :) |
| 05:57.02 | [TK]D-Fender | JT : He's famous..... get your OWN INITIALS! |
| 05:57.44 | JT | dlynes_laptop: no, it's an abbreviation for my name |
| 05:57.49 | dlynes_laptop | ah |
| 05:58.04 | [TK]D-Fender | At least now Cameron Diaz is available again :) |
| 05:58.07 | CrashHD | later |
| 05:58.16 | dlynes_laptop | [TK]D-Fender, you're thinking about asking her out? |
| 05:58.50 | [TK]D-Fender | P&P... dear God I haven't seen that since my l33t g4m3r dayz... |
| 05:58.51 | [TK]D-Fender | dlynes_laptop : No need to be so polite! |
| 05:59.00 | [TK]D-Fender | *yum* |
| 05:59.09 | dlynes_laptop | frankly |
| 05:59.19 | dlynes_laptop | I'd rather do that other chick that's in Girl Next Door |
| 05:59.36 | [TK]D-Fender | Mind you I don't know what goes on "upstairs" in there.... brains are a turn-on (or off).... |
| 05:59.36 | dlynes_laptop | elisha cuthbert |
| 05:59.55 | [TK]D-Fender | EW. Dumberer Bauer is her moniker in my circles thanks to "24". |
| 05:59.57 | dlynes_laptop | plus, she's canadian, eh? :) |
| 06:00.09 | [TK]D-Fender | She is USELESS. |
| 06:00.12 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 06:00.23 | dlynes_laptop | i dunno...i thought she was pretty hot looking |
| 06:00.34 | *** join/#asterisk inv_Arp (n=junya@c-75-74-183-191.hsd1.fl.comcast.net) |
| 06:00.48 | dlynes_laptop | but i've never watched 24, either |
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| 06:01.53 | joelsolanki | <PROTECTED> |
| 06:02.12 | dlynes_laptop | joelsolanki, different codecs |
| 06:02.15 | joelsolanki | yesterday my linux hdd crashed. so i have built up the new server on hdd |
| 06:02.45 | joelsolanki | i had taken backup of licenses and i have restored too in new asterisk |
| 06:03.02 | joelsolanki | show g729 shows me 30 channels |
| 06:03.08 | dlynes_laptop | joelsolanki, different codecs |
| 06:03.20 | joelsolanki | dlynes_laptop: means ? |
| 06:03.36 | dlynes_laptop | joelsolanki, one leg is codec 1, the other is codec 256 |
| 06:03.41 | dlynes_laptop | joelsolanki, show codecs |
| 06:04.12 | x86 | heh heh... is the third leg codec 69? |
| 06:04.40 | dlynes_laptop | guess ya had to be there, huh? |
| 06:04.46 | joelsolanki | 1 is g723 and 256 is g729 |
| 06:04.57 | dlynes_laptop | there ya go |
| 06:04.59 | JT | of course asterisk cannot transcode g.723 |
| 06:05.08 | dlynes_laptop | and i'm guessing you don't have anything to transcode g723 |
| 06:05.10 | joelsolanki | oh not possible ? |
| 06:05.19 | joelsolanki | are u sure JT? |
| 06:05.21 | JT | not possible |
| 06:05.22 | JT | yes |
| 06:05.23 | dlynes_laptop | nopenopenope...not legally anyways |
| 06:05.26 | dlynes_laptop | however |
| 06:05.32 | joelsolanki | hmm ok. NP. |
| 06:05.35 | dlynes_laptop | the patent for g723 expires in June(?) |
| 06:10.18 | *** join/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 06:10.55 | *** join/#asterisk juice (n=juice@mo-76-0-43-77.dhcp.embarqhsd.net) |
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| 06:16.31 | Qwell | s/the/a/ |
| 06:17.14 | Corydon76-home | The final patent for G.723.1 doesn't expire until 2014 |
| 06:17.46 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 06:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 06:31.07 | dlynes_laptop | Corydon76-home, so what part of it is expiring this summer? |
| 06:31.21 | Qwell | the g |
| 06:32.14 | dlynes_laptop | so g.723 expires in June or July, but not g.723.1? |
| 06:32.41 | Qwell | g.723.1 is what people are talking about when they say g723 |
| 06:33.04 | conver2 | in queue application, how can I enable/disable moh? |
| 06:33.14 | dlynes_laptop | ok so what's the 'g' you referred to that expires this summer, then? |
| 06:33.26 | Qwell | a failed attempt at humor |
| 06:33.35 | dlynes_laptop | sheesh |
| 06:34.08 | *** join/#asterisk azidenth (n=aby_azid@60.50.220.139) |
| 06:34.11 | dlynes_laptop | so any idea what exactly expires this summer then? |
| 06:34.43 | JT | g.723 with no .1 it sounds like |
| 06:34.46 | azidenth | hello..so sorry to interrupt...but is there anyone here can help me with Asterisk Realtime? |
| 06:34.52 | dlynes_laptop | someone was telling me it was kind of pointless to license g723, because the patent expires this summer |
| 06:34.55 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 06:35.11 | dlynes_laptop | and the amount of money you would need to invest to license it wouldn't be worth it for 6 months |
| 06:35.24 | Corydon76-home | The original G.723 is already free and clear of patents |
| 06:35.53 | dlynes_laptop | But it's not compatible with g.723.1 right? |
| 06:37.47 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 06:39.19 | azidenth | i need to configure asterisk with mysql.. |
| 06:39.28 | azidenth | anyone can help? please |
| 06:40.41 | dendrite | azidenth: I'm not a regular here, but in general, on IRC, if you ask specific questions you tend to get better results... |
| 06:41.00 | Grnd-Wire | dlynes_laptop: Are you referring to g.729 ? |
| 06:41.05 | azidenth | ok.. |
| 06:41.17 | dlynes_laptop | Grnd-Wire, no |
| 06:41.19 | Grnd-Wire | or am I the one that's confusing the two.. ? |
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| 06:41.51 | dlynes_laptop | Grnd-Wire, i'm curious about g.723 because the patent on it is much older than g.729 |
| 06:42.07 | dlynes_laptop | Grnd-Wire, and because you cannot buy codec licenses for g.723 from digium |
| 06:42.25 | Grnd-Wire | dlynes_laptop: ahh, ok.. |
| 06:42.35 | x86 | dlynes_laptop: where do you get g.723 licenses from? |
| 06:42.42 | dlynes_laptop | x86, you don't |
| 06:42.47 | dlynes_laptop | x86, at least not legally |
| 06:43.03 | dlynes_laptop | x86, so iow, you don't |
| 06:43.04 | x86 | i was gonna say, on my dev machine with the IPP g.729 codec, g.723 just works ;) |
| 06:43.16 | dlynes_laptop | x86, cause if it was legally, they wouldn't be called 'licensed' |
| 06:43.29 | x86 | hehe |
| 06:44.20 | azidenth | 9 sip peers [Monitored: 2 online, 4 offline Unmonitored: 1 online, 2 offline] |
| 06:44.21 | azidenth | [Jan 16 14:53:39] WARNING[22203]: config.c:1231 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 06:44.21 | azidenth | localhost*CLI> |
| 06:44.32 | azidenth | 9 sip peers [Monitored: 2 online, 4 offline Unmonitored: 1 online, 2 offline] |
| 06:44.33 | azidenth | [Jan 16 14:53:39] WARNING[22203]: config.c:1231 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 06:44.33 | azidenth | localhost*CLI> |
| 06:44.42 | dlynes_laptop | azidenth, you don't have the mysql realtime engine loaded |
| 06:44.54 | azidenth | can u help me ? |
| 06:45.04 | dlynes_laptop | azidenth, nope...I don't use realtime, or mysql |
| 06:45.09 | x86 | azidenth: do you have asterisk-addons installed? |
| 06:45.09 | dlynes_laptop | Maybe someone else does, though |
| 06:45.14 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 06:45.17 | *** join/#asterisk shinux__ (n=shinux@196.220.29.36) |
| 06:45.19 | dlynes_laptop | but yeah...you need asterisk-addons for it |
| 06:45.25 | *** join/#asterisk groogs[h] (n=chatzill@66.102.80.229) |
| 06:45.27 | azidenth | yup |
| 06:45.31 | azidenth | already intalled |
| 06:45.36 | azidenth | from the svn |
| 06:45.37 | groogs[h] | is there any practical limit on the number of includes that can be in a context? |
| 06:46.02 | dlynes_laptop | groogs[h], limited by your memory, and how long you feel like waiting for asterisk to boot up |
| 06:46.44 | azidenth | change the configuration in extconfig.conf and res_mysql.conf |
| 06:47.18 | groogs[h] | dlynes_laptop: how memory intensive is it? if i had say, 500 includes in one context.. |
| 06:47.28 | dlynes_laptop | i have no idea |
| 06:47.41 | dlynes_laptop | but if you tell the makefile to build memory debugging into asterisk |
| 06:47.43 | dlynes_laptop | you can find out |
| 06:47.53 | groogs[h] | well, are we talking using like, 2 more MB of memory, or 200? :) |
| 06:48.09 | dlynes_laptop | depends on how big the contexts are, i would imagine |
| 06:48.18 | dlynes_laptop | and how entry lines each one has |
| 06:48.28 | groogs[h] | one extension, maybe 5 priorities |
| 06:48.31 | Grnd-Wire | So is the READ application still supported in 1.2.14 ? It's like it's not even executing.. |
| 06:48.43 | dendrite | azidenth: Did http://voip-info.org/wiki/view/Asterisk+RealTime help? |
| 06:48.52 | JT | why do you need so many includes, groogs[h] ? |
| 06:49.10 | azidenth | done that dendrite... |
| 06:49.42 | groogs[h] | JT: having different contexts for various users.. so eg, one user can call 101, 102, 103, 104, another can only call 101.. |
| 06:50.02 | JT | i'm sure there are far smarter ways to omplement that |
| 06:50.05 | JT | implement |
| 06:50.17 | groogs[h] | rather than duplicating the dial lines for each extension for each context that has that extension |
| 06:50.51 | naftali5 | groogs, http://www.asterisk.org/node/112 |
| 06:51.23 | azidenth | anyone here had similar problem like me? |
| 06:52.10 | JT | groogs[h]: what sort of includes are you talking about? |
| 06:53.32 | groogs[h] | uh .. include=>context |
| 06:53.44 | azidenth | im trying to add sip user/peers dynamically using mysql.. |
| 06:54.05 | azidenth | any other solution besides asterisk realtime? |
| 06:54.08 | JT | groogs[h]: you can include files too, that's why i asked |
| 06:54.23 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:54.30 | groogs[h] | JT: oh, right, sorry. |
| 06:57.42 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 06:58.36 | JT | groogs[h]: how fine grained does it need to be? are you including a context for every extension, or are there groups of extensions in these contexts? |
| 06:59.26 | *** join/#asterisk X-Rob (n=rob-x@203.37.172.162) |
| 07:00.03 | groogs[h] | JT: a context for every extension.. |
| 07:01.58 | niZon | anyone ever used an audicodes channel bank with astersk? |
| 07:02.03 | niZon | +i |
| 07:02.19 | JT | groogs[h]: that sounds highly unoptimal |
| 07:03.24 | groogs[h] | JT: yeah, i think it might make more sense just to duplicate the exten=> lines (theres at most 5, and at least 2 .. so its not a HUGE deal..) |
| 07:03.45 | JT | umm |
| 07:03.50 | JT | what about groups |
| 07:04.03 | JT | surely you don't need such ridiculously fine grained access control |
| 07:04.12 | groogs[h] | well, thats an option too, but adds an extra step to creating everything |
| 07:04.15 | JT | you could have context for groups like sales, management, etc |
| 07:04.31 | JT | sounds way easier to manager than a context for every extension |
| 07:04.38 | JT | and programmatically more efficient |
| 07:05.05 | groogs[h] | yeah, it might work better like that |
| 07:05.14 | dendrite | groogs[h]: Forgive my curiosity, but why are the complex restrictions needed in your installation? |
| 07:05.44 | JT | and if you need such a fine grain, you could use macros to dial, and set variables per extension to define what it has access to |
| 07:05.48 | JT | or use realtime |
| 07:05.48 | groogs[h] | probably the case where you want to create an extension that can only call one other extension is more rare than when you want an extension to only be able to call 'sales' group |
| 07:06.09 | groogs[h] | dendrite: doing work on freepbx |
| 07:06.17 | groogs[h] | (oops, said the bad word..) |
| 07:06.40 | zeeesh | hi |
| 07:06.52 | JT | not sure how that relates to your business problem, groogs[h] |
| 07:07.25 | CunningPike | niZon: We have a SIP FXS gateway - is that what you mean? |
| 07:08.34 | *** join/#asterisk alamantia (n=Anthony@65.4.1.65) |
| 07:09.09 | *** join/#asterisk Corydon76-home (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 07:09.09 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:09.47 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:10.28 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 07:10.31 | zeeesh | i have been installed asterisk at redhat9 .. now want to test at my own cell number as access number ... if my cell number is 92321XXX then what shud be the extensions phase.... like "exten => 92321XXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) "??? |
| 07:12.28 | Grnd-Wire | Does anyone know of a replacement macro for SayDigits that would allow me to use my own voice, and hopefully speak the numbers alot quicker (so it's not so choppy?) |
| 07:12.28 | Strom_C | Grnd-Wire: write your own :) |
| 07:12.53 | Grnd-Wire | Strom_C: Well, I might end up doing that (what a learning experience that will be.. :) - I was just thinking someone else may have done it already? Me-thinks I'll go google for "saydigits replacement".. Be right back. ;) |
| 07:12.54 | JT | does SayDigits play prompt files? |
| 07:12.57 | JT | if so, change them |
| 07:13.11 | Grnd-Wire | JT: hmm.. but what about the timing, is it really that simple? |
| 07:13.23 | JT | unless it adds delays internally, yes |
| 07:13.36 | JT | find out by making a test extension that plays the digits with Playback |
| 07:13.48 | JT | if the gaps are still there, they're probably in the file or unavoidable anyway |
| 07:14.06 | Grnd-Wire | cool! I'll mess with that when I get to that part, thanks for the advice. |
| 07:15.09 | JT | i think you can also have multiple copies of the prompts, for different regions, so you may be able to keep your prompt files, and just get certain dialplan stuff to use different "region" prompts |
| 07:15.36 | Grnd-Wire | ooh, that would make me even happier.. |
| 07:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:19.07 | JT | it may just be a matter of modifying the existing prompts |
| 07:19.15 | JT | if you're lucky you won't have to re-record |
| 07:22.22 | *** join/#asterisk nahirean (i=nahirean@unaffiliated/nahirean) |
| 07:23.56 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:29.04 | Grnd-Wire | This is really cool.. I'm building an IVR so people can RSVP to my wedding just by calling in.. :) |
| 07:31.53 | jeremy_g | Grnd-Wire: you rule |
| 07:32.57 | nahirean | congrats on the wedding |
| 07:33.26 | nahirean | i'll be taking your balls now, you've no more use for them. |
| 07:33.31 | Grnd-Wire | jeremy_g: ya.. My woman is sitting behind me designing the invitation, while I setup the RSVP system.. (website and phone IVR) haha..We're such geeks (art and technology) |
| 07:33.56 | jeremy_g | LOLz |
| 07:33.57 | Grnd-Wire | oh no, she lets me still use them - In fact, if you'll excuse me, I need to go use them right now.. Night guys! |
| 07:34.03 | nahirean | Ugh.. |
| 07:34.05 | nahirean | good night. |
| 07:34.07 | Grnd-Wire | :S |
| 07:38.43 | zeeesh | <PROTECTED> |
| 07:39.51 | Strom_C | zeeesh: I suggest you start with something simpler |
| 07:40.35 | zeeesh | like?? |
| 07:41.32 | nahirean | also depends on how the ITSP sends the call to you |
| 07:41.44 | nahirean | Do they send 11 digits, or 10? |
| 07:41.56 | zeeesh | <Strom_C> : what kind of simpler ?? |
| 07:42.02 | nahirean | (assuming you're using an itsp) |
| 07:42.16 | *** join/#asterisk mog_home (n=rah@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:42.16 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 07:50.49 | tzafrir | zeeesh, redhat9? Don't you have something that is a bit more in touch with reality? |
| 07:51.48 | tzafrir | (for RH fans: a later Fedora/CentOS/RHEL . Or other distros) |
| 07:53.31 | Nugget | heh |
| 07:54.09 | Nugget | http://lnk.nu/macnugget.org/d44 <-- this is for you, tzafrir |
| 07:54.13 | zeeesh | <tzafrir>:so i would like to face .. reality .. .so thats y using ... without .. entering into water .. how can u swim ... catch u later .. now going for lunch... |
| 07:55.13 | tzafrir | Nugget, wow! |
| 07:55.29 | *** join/#asterisk alamantia (n=Anthony@65.4.1.65) |
| 07:55.29 | *** join/#asterisk susinths (n=susinths@hesthoi.ifi.uio.no) |
| 07:56.14 | Nugget | I was going to install it a while ago, just on a lark, but I was thwarted because it's so old that the CD isn't bootable. |
| 07:56.38 | susinths | hello everybody |
| 07:56.41 | tzafrir | Nugget, can't you get something from ftp.redhat.com? |
| 07:56.56 | Nugget | that's too much work. |
| 07:57.09 | susinths | is this the place where most asterisk knowledge is exchanged? |
| 07:57.27 | Nugget | it's about 50/50 knowledge and insults. |
| 07:57.36 | Nugget | you're lucky though because today is a knowledge day. |
| 07:57.36 | tzafrir | zeeesh, while RH isn't my favorite distro, I have nothing against using it. |
| 07:57.58 | susinths | Nugget: OK |
| 07:58.18 | tzafrir | However working today with a Linux system from 2001 is just going to cause you extra grieff. Not to mention tons of unpatched security holes |
| 07:59.53 | tzafrir | zeeesh, if that system is a new installation, install a newer linux system over it, and be done. You'll e.g., have a newer kernel which is more responsive |
| 08:00.19 | susinths | Can asterisk be used form a VOIP company offering IP telephony? |
| 08:00.23 | *** join/#asterisk apardo (n=apardo@87.217.145.94) |
| 08:00.34 | Strom_C | susinths: let me guess, you want to be the next vonage |
| 08:00.52 | susinths | :) |
| 08:01.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:01.24 | susinths | for example vonage |
| 08:02.15 | susinths | Strom_C: can * be used for such ? |
| 08:03.40 | Strom_C | susinths: yes, although unless you have the technical skill and the telephony experience, i wouldnt recommend you attempt it |
| 08:04.36 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 08:04.43 | susinths | Strom_C: OK, i do have tech skills, but NO tel experience at all. What about u? |
| 08:04.45 | phpboy | Jan 16 10:03:38 WARNING[2695]: translate.c:88 powerof: Powerof 0: No power?? |
| 08:04.46 | phpboy | Jan 16 10:03:38 WARNING[2695]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown |
| 08:04.48 | phpboy | what does this mean? |
| 08:05.05 | Strom_C | susinths: six years of telephony experience and I still wouldn't go for it |
| 08:05.17 | susinths | Strom_C: Oh, i see |
| 08:05.38 | phpboy | it happens when ever a call goes through pstn |
| 08:05.40 | phpboy | :/ |
| 08:05.43 | phpboy | in or out |
| 08:05.47 | susinths | Strom_C: What can be the biggest challenge? |
| 08:06.15 | Strom_C | susinths: pay me $150 per hour for my telcommunications consulting services and I'll be happy to tell you everything |
| 08:07.50 | susinths | Strom_C: I see. I'll think about it. I've been testing * , it seems for work very good. Thats why was wondering to offer something with asterisk |
| 08:09.04 | susinths | Strom_C: Although i haven't tested with SIP trunks yet, but read a lot about it. |
| 08:09.35 | susinths | Strom_C: I'm thinking of offering such in South Asia |
| 08:10.37 | phpboy | hey all |
| 08:10.44 | phpboy | I'd really appreciate some help :/ |
| 08:11.54 | niZon | well i think i just bricked this MP-124 |
| 08:11.54 | niZon | audiocodes and their broken firmware |
| 08:11.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:12.35 | susinths | anyone with comments to my plan to start a VOIP comapany with asterisk in south asia? |
| 08:12.54 | *** part/#asterisk irasnyd (n=irasnyd@cpe-75-85-119-184.socal.res.rr.com) |
| 08:15.09 | phpboy | please guys |
| 08:15.12 | phpboy | I love you!! |
| 08:15.14 | phpboy | all of you!! |
| 08:15.17 | phpboy | please help me :T |
| 08:16.36 | *** join/#asterisk azidenth (n=aby_azid@60.50.220.139) |
| 08:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:20.31 | azidenth | anyone here have ever used asterisk realtime?.. |
| 08:23.17 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:24.22 | *** join/#asterisk apardo (n=apardo@87.217.145.176) |
| 08:28.52 | Kigh | i did |
| 08:30.15 | azidenth | alrite.. |
| 08:30.36 | azidenth | i have a problem here mate.. |
| 08:31.05 | azidenth | i already installed the asterisk-addons..create dbase and table.. |
| 08:31.26 | azidenth | and configured the extconfig.conf and res_mysql.conf.. |
| 08:31.46 | Kigh | whats the problem |
| 08:32.13 | phpboy | Why do you guys hate me so much? :T |
| 08:32.28 | Kigh | what does "realtime mysql status" in manager console say? |
| 08:32.45 | azidenth | but when i reload my asterisk server..and try to view the sip peers and users..it says there mysql engine is not available |
| 08:33.15 | azidenth | "realtime mysql status" <-- wats the command? |
| 08:33.25 | phpboy | azidenth: check your mysql logs to see if asterisk is even trying to make the connection... |
| 08:33.49 | Kigh | phpboy: "No translator path from gsm to unknown" means the asterisk cant _transcode_ the voice from gsm codec to "unknown" |
| 08:34.05 | dlynes_laptop | susinths, as Strom_C suggested, I would hire someone to help you get started |
| 08:34.09 | Kigh | i assume you have a typo somewhere in a codec definition or something |
| 08:34.13 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
| 08:34.13 | dlynes_laptop | susinths, however, before you even get to that stage |
| 08:34.20 | phpboy | Kigh: where can I check this? |
| 08:34.27 | dlynes_laptop | susinths, I would suggest you actually sit down with asterisk, and try it out more in depth |
| 08:34.38 | dlynes_laptop | susinths, find out its strengths, and its pitfalls |
| 08:34.41 | azidenth | kigh: yeah wer? |
| 08:34.41 | Kigh | azidenth: thats the status command that shows wether res_mysql is loaded and connected to DB or not |
| 08:34.52 | dlynes_laptop | susinths, so that you're not going around a blind corner |
| 08:35.16 | Kigh | phpboy: you need to read codecs.conf, sip.conf and misdn.conf/zapata.conf (depending on what driver youre using) |
| 08:35.31 | Kigh | and the telephone settings of course. every part of configuration regarding the transcode |
| 08:36.05 | Kigh | azidenth: and "sip peers" wont show a realtime peer until it is conntected. if i remember correctly |
| 08:36.08 | phpboy | Kigh: my zapata.conf does not have anything in it regarding codecs |
| 08:36.38 | azidenth | kigh --> == Parsing '/etc/asterisk/asterisk.conf': Found |
| 08:36.39 | azidenth | <PROTECTED> |
| 08:36.39 | azidenth | <PROTECTED> |
| 08:36.39 | azidenth | <PROTECTED> |
| 08:36.42 | Kigh | azidenth: the asterisk fetches the settings in "realtime" from the mysql table. use "realtime mysql status" to check if the module is loaded |
| 08:36.52 | azidenth | kihg:is that wat u mean? |
| 08:36.58 | Kigh | no. |
| 08:37.04 | susinths | dlynes_laptop: Thanx a lot |
| 08:37.09 | phpboy | Kigh: my codecs.conf is the default codecs.conf |
| 08:37.12 | Kigh | azidenth: type "realtime mysql status" |
| 08:37.16 | susinths | dlynes_laptop: I thought the same. |
| 08:37.20 | Kigh | phpboy: then you dont have to check that part |
| 08:37.47 | dlynes_laptop | susinths, yeah...anyone that has the telecommunications experience has the upper hand over you |
| 08:37.58 | dlynes_laptop | susinths, their skills are more valuable for such a business than the technical skills |
| 08:38.20 | susinths | dlynes_laptop: I see |
| 08:38.35 | Kigh | azidenth: here's an example what the result should look alike: |
| 08:38.35 | Kigh | asterisk*CLI> realtime mysql status |
| 08:38.36 | Kigh | Connected to asterisk@localhost, port 3306 with username root for 20 days, 15 hours, 36 minutes, 46 seconds. |
| 08:38.36 | phpboy | Kigh: where else do you think I should look? |
| 08:38.38 | azidenth | kigh: where? in the asterisk CLI? |
| 08:38.44 | susinths | dlynes_laptop: sounds reasonable |
| 08:38.45 | Kigh | azidenth: damn yes =) |
| 08:39.10 | azidenth | it says there no such command.. |
| 08:39.56 | azidenth | hmm.. |
| 08:40.05 | Kigh | phpboy: for greater debugging possibilities type "set verbosity 4" on the asterisk CLI and do whatever you did again. you will get more output about whats going on then |
| 08:40.14 | Kigh | azidenth: then the module isnt loaded |
| 08:40.22 | azidenth | kigh: i followed the instruction in www.voip-info.org.. |
| 08:40.35 | *** join/#asterisk tengulre (n=tengulre@61.185.220.249) |
| 08:42.05 | *** join/#asterisk s0lid (n=jlq@210.213.240.100) |
| 08:42.46 | Kigh | azidenth: do "!find /usr/lib/asterisk/modules/ -name 'res_config_mysql.so'" (note the exclamation mark) in the asterisk CLI to see if the module exists ÃÃphysically on your HDD |
| 08:42.46 | susinths | dlynes_laptop: I've the computer skills, but that alone is not sufficient i think |
| 08:42.56 | dlynes_laptop | susinths, not even close |
| 08:43.15 | Kigh | azidenth: do you use asterisk 1.2 or 1.0? |
| 08:43.21 | azidenth | is asterisk 1.4 |
| 08:43.25 | Kigh | k |
| 08:43.30 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 08:43.34 | Kigh | azidenth: so did you find the module? |
| 08:43.47 | azidenth | wait |
| 08:44.43 | phpboy | Kigh: set it to 6 |
| 08:44.46 | phpboy | nothing more |
| 08:44.47 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:45.11 | susinths | dlynes_laptop: I understand |
| 08:45.26 | susinths | dlynes_laptop: U got exp with tele? |
| 08:45.44 | dlynes_laptop | I've got experience with both, yes |
| 08:46.08 | phpboy | exact same info |
| 08:46.55 | susinths | dlynes_laptop: But is Asterisk good enough to handle real life VOIP connections to the mass? |
| 08:47.21 | phpboy | pomply :T |
| 08:47.26 | tzafrir | azidenth, it is built from asterisk_addons |
| 08:47.32 | dlynes_laptop | susinths, yes, there's a few companies that have mass deployments that are using asterisk |
| 08:47.57 | azidenth | tzafir: i know |
| 08:48.05 | susinths | dlynes_laptop: OK, good to hear. |
| 08:48.08 | phpboy | pomple :T |
| 08:48.13 | susinths | dlynes_laptop: Which country r u from? |
| 08:48.16 | phpboy | this is sakkie de kok :T |
| 08:48.20 | susinths | dlynes_laptop: I'm from norway. |
| 08:48.21 | dlynes_laptop | susinths, Canada |
| 08:48.30 | susinths | dlynes_laptop: Oh i see |
| 08:48.36 | phpboy | Kigh: what else do you think I should have a look at? |
| 08:48.52 | susinths | dlynes_laptop: I live here, i'm immigrant |
| 08:49.10 | dlynes_laptop | susinths, the people with experience in mass deployments you'll find all over the world...they're not really concentrated in any one country |
| 08:49.51 | susinths | dlynes_laptop: that's true |
| 08:50.14 | dlynes_laptop | but, i suspect in coming years |
| 08:50.39 | dlynes_laptop | you'll probably find Indians to have the experience with the largest deployments |
| 08:50.46 | dlynes_laptop | because of the size of the call centers there |
| 08:50.57 | susinths | dlynes_laptop: i see |
| 08:51.04 | azidenth | kigh:the module exist..wats next |
| 08:51.32 | susinths | dlynes_laptop: U mean because of leasing over there? |
| 08:51.46 | susinths | i'm from country next to India |
| 08:52.00 | dlynes_laptop | susinths, because american companies (and I'm assuming others) offshore their call centers to India |
| 08:52.10 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 08:52.14 | dlynes_laptop | susinths, if you hang out in this channel long enough |
| 08:52.23 | dlynes_laptop | susinths, you're sure to run across some Indians and Pakistanis |
| 08:52.53 | susinths | dlynes_laptop: really |
| 08:52.56 | susinths | dlynes_laptop: i see |
| 08:53.29 | dlynes_laptop | Which country are you in? |
| 08:53.52 | susinths | norway |
| 08:53.56 | susinths | but from srilanka |
| 08:54.14 | dlynes_laptop | susinths, well, you could try talking to some of the chaps from India |
| 08:54.23 | dlynes_laptop | susinths, perhaps they'd be willing to work for you |
| 08:54.40 | dlynes_laptop | susinths, but somehow I kinda doubt it |
| 08:54.47 | dlynes_laptop | susinths, most of them are business owners already |
| 08:54.58 | susinths | dlynes_laptop: i see |
| 08:55.25 | Kigh | azidenth: type "show modules like mysql" in CLI |
| 09:00.08 | azidenth | kigh: nothing..no modules... |
| 09:04.34 | azidenth | kigh: im lost here..when i typed show modules like mysql it returned 0 |
| 09:05.58 | *** join/#asterisk bkw_ (n=brian@88-110-74-241.dynamic.dsl.as9105.com) |
| 09:06.45 | jeremy_g | i know a lot of indians looking for work in stockhol |
| 09:06.47 | jeremy_g | m |
| 09:06.54 | jeremy_g | sip guys |
| 09:10.46 | *** join/#asterisk apardo (n=apardo@87.217.144.198) |
| 09:14.46 | susinths | jeremy_g: where is stockhol? uk? |
| 09:15.56 | *** join/#asterisk yansolo90 (n=yansolo9@par69-3-82-224-162-203.fbx.proxad.net) |
| 09:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:18.11 | dlynes_laptop | susinths, sweden |
| 09:18.20 | dlynes_laptop | susinths, he meant stockholm |
| 09:18.39 | azidenth | kigh: anyway kigh i managed to load res_config_mysql.so |
| 09:19.06 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:21.42 | susinths | dlynes_laptop: i see |
| 09:24.44 | azidenth | kigh: when i type realtime mysql status...fail to connect database |
| 09:24.59 | azidenth | kigh: which config files should i look into? |
| 09:25.10 | azidenth | anyone? |
| 09:28.52 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:30.52 | *** join/#asterisk drega (n=dforeman@mail.amcat.co.uk) |
| 09:31.53 | yansolo90 | check /etc/asterisk/res_mysql.conf |
| 09:32.22 | azidenth | yansolo90: im checking it |
| 09:32.37 | jeremy_g | susinths:stockholm,SE |
| 09:32.54 | susinths | jeremy_g: i se |
| 09:32.58 | susinths | jeremy_g: see |
| 09:33.42 | jeremy_g | susinths:what do you do |
| 09:38.15 | susinths | jeremy_g: study |
| 09:38.47 | susinths | jeremy_g: has been working as a service techican earlier |
| 09:40.43 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
| 09:40.54 | *** join/#asterisk Gled|work (n=sguser@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
| 09:42.17 | *** join/#asterisk Dibbler_XP (n=Dibbs@212.248.196.179) |
| 09:44.55 | susinths | jeremy_g: u? |
| 09:44.59 | *** join/#asterisk Dibbler_XP (n=Dibbs@212.248.196.179) |
| 09:46.51 | *** join/#asterisk e-ddie (n=oal@62.61.133.90.generic-hostname.arrownet.dk) |
| 09:47.44 | azidenth | yansolo90: i checked the res_mysql.conf |
| 09:47.50 | azidenth | but still fail to connect |
| 09:47.53 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:47.59 | azidenth | any ideas |
| 09:51.15 | yansolo90 | you can try to connect your mysql DB with an other client |
| 09:53.40 | Chris-NB | hi |
| 09:53.46 | Chris-NB | anybody using astlinux? |
| 09:55.59 | azidenth | kigh: if u still around..can i asked u again same time tomorrow.. |
| 09:57.23 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 10:02.40 | potsboy | Chris-NB, used it a while back whats the prob? |
| 10:02.58 | Chris-NB | potsboy, I try to upgrade kernel and asterisk |
| 10:03.13 | Chris-NB | potsboy, kernel and modules where successfull |
| 10:03.13 | potsboy | and?? |
| 10:03.30 | Chris-NB | potsboy, but I've problems with asterisk (and other binaries I installed) |
| 10:03.39 | potsboy | example? |
| 10:03.51 | Chris-NB | when I try to start asterisk i get this err: -sh: /usr/sbin/asterisk: No such file or directory |
| 10:04.12 | Chris-NB | I tried to install wanrouter (sangoma) mdecrypt and asterisk |
| 10:04.16 | potsboy | have you run a "which asterisk" ? may be that simple |
| 10:04.34 | Chris-NB | what do you mean with that? |
| 10:04.38 | potsboy | also check your path |
| 10:04.55 | Chris-NB | path is deffinitly right |
| 10:05.02 | Chris-NB | also the file is read and execute able |
| 10:05.21 | potsboy | so you can run it lik # /usr/sbin/asterisk |
| 10:05.24 | tzafrir | ldd /usr/sbin/asterisk |
| 10:05.28 | tzafrir | ldd -s /usr/sbin/asterisk |
| 10:05.47 | Chris-NB | aahhhh ... that could be the prob : D |
| 10:05.50 | Chris-NB | fine, I'll try |
| 10:05.50 | tzafrir | hmm... wrong message |
| 10:05.54 | Chris-NB | just a mom. |
| 10:06.33 | tzafrir | Chris-NB, and there is no chroot magic involved here, right? |
| 10:06.53 | Chris-NB | tzafrir, ähm .... don't think so. |
| 10:07.52 | tzafrir | if you have no better idea, the next step may be to find which script exactly runs this, and run it in traced mode (#!/bin/sh -x |
| 10:07.57 | tzafrir | ) |
| 10:08.35 | Chris-NB | I've manually tried to start asterisk |
| 10:08.50 | yansolo90 | anybody knows ipvsadm and ldirectord ? |
| 10:09.30 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:16.38 | Chris-NB | tzafrir, jep. dynam libs where missing. better, they ARE missing : / |
| 10:17.12 | Chris-NB | tzafrir, but now I know there to look at. thanks mate |
| 10:17.32 | *** join/#asterisk shinux_ (n=shinux@196.220.30.208) |
| 10:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:18.43 | andrebarbosa | hi guys, i'm making a call from a fxs extension (zap) to a sip phone |
| 10:19.11 | andrebarbosa | on my SIP header i got the from as: "asterisk@ip" |
| 10:19.20 | andrebarbosa | can i change this fromuser? |
| 10:19.48 | andrebarbosa | i know i can change the fromuser from a sip peer with fromuser option, but in a zap extension i don't know how to do it |
| 10:21.11 | *** join/#asterisk santibiotico (n=santi@160.Red-83-58-126.dynamicIP.rima-tde.net) |
| 10:21.12 | santibiotico | hi |
| 10:21.20 | santibiotico | can anyone help me with iax?? |
| 10:21.43 | santibiotico | i have 2 asterisk servers in the same network and i want to get them connected |
| 10:22.02 | santibiotico | in order to call users in asterisk#1 from asterisk#2 |
| 10:22.34 | *** join/#asterisk bkw__ (n=brian@82.152.100.241) |
| 10:29.10 | tzafrir | santibiotico, there should be a relevant page in http://voip-info.org , too lazy to search now |
| 10:29.21 | tzafrir | Ask here specific questions... |
| 10:29.55 | tzafrir | andrebarbosa, is the callerid set in zapata.conf? |
| 10:31.04 | santibiotico | tzafrir of course i've read all i've found in voip-info.org |
| 10:31.28 | tzafrir | santibiotico, so what's your specific problem? |
| 10:31.46 | andrebarbosa | yep |
| 10:31.51 | andrebarbosa | caller ID is fine |
| 10:31.56 | santibiotico | i've defined in iax.conf the 2 "friends" |
| 10:32.20 | andrebarbosa | but i wonder how we change the fromuser |
| 10:32.22 | santibiotico | but i always get a registration error |
| 10:32.40 | santibiotico | i am using static ip addresses, so i do not need to register |
| 10:33.00 | *** join/#asterisk mkl1525 (n=qwertz@pD9532738.dip0.t-ipconnect.de) |
| 10:33.36 | *** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it) |
| 10:33.58 | andrebarbosa | i got something like: "my callerid" from: asterisk@ip |
| 10:34.01 | andrebarbosa | in my SIP header |
| 10:34.11 | andrebarbosa | i want to change "asterisk" to my extension name |
| 10:36.18 | *** join/#asterisk susin (n=susinths@trollhoin.ifi.uio.no) |
| 10:37.56 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 10:40.14 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 10:42.23 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 10:44.00 | tzafrir | andrebarbosa, what do you have on the callerid field in zapata.conf? |
| 10:44.26 | tzafrir | Full Name <number> |
| 10:44.29 | tzafrir | or just: |
| 10:44.33 | tzafrir | Full Name |
| 10:50.37 | andrebarbosa | full name |
| 10:50.41 | andrebarbosa | only |
| 10:50.43 | andrebarbosa | let me try |
| 10:50.48 | andrebarbosa | with callerid <number> |
| 10:51.13 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:51.48 | andrebarbosa | hum |
| 10:51.49 | andrebarbosa | :) |
| 10:52.10 | andrebarbosa | Thanks, tzafrir |
| 10:52.14 | andrebarbosa | now I have |
| 10:52.28 | andrebarbosa | "full name" from: number@ip |
| 10:53.52 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:54.26 | andrebarbosa | can you give me a hand to find that in the source? |
| 10:54.39 | dlynes_laptop | is something like the following: exten => s,n(mylabel),dothis() |
| 10:55.01 | dlynes_laptop | A label for that dialplan priority, so that you can do goto(mylabel)? |
| 10:55.13 | dlynes_laptop | instead of goto(10) or something similar? |
| 10:55.47 | dlynes_laptop | So I could also define it as exten => s,10(mylabel),dothis() |
| 10:56.05 | dlynes_laptop | and then i could goto(10), or goto(mylabel), however I chose to do it? |
| 10:56.34 | *** join/#asterisk viperdude (n=viperdud@195.74.96.120) |
| 10:57.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:57.58 | santibiotico | while trying to configure iax...whenever i try to make a call i get the following cli output error: |
| 10:57.59 | santibiotico | auto_congest: Auto-congesting call due to slow response |
| 10:58.03 | santibiotico | any idea?? |
| 10:58.18 | santibiotico | (of course, the call is not working...) |
| 10:58.34 | dlynes_laptop | santibiotico, try decreasing your qualify= value in your sip.conf file |
| 10:58.48 | dlynes_laptop | santibiotico, the default if I remember correctly is 3000 |
| 10:59.00 | dlynes_laptop | santibiotico, the default is applied if you do qualify=yes |
| 10:59.33 | dlynes_laptop | santibiotico, it's probably some kind of issue with your firewall |
| 10:59.56 | santibiotico | i have no firewall ;) |
| 11:00.00 | dlynes_laptop | santibiotico, i.e. it might be closing the port sooner than asterisk expects it to |
| 11:00.07 | dlynes_laptop | santibiotico, you're not behind a router? |
| 11:00.10 | santibiotico | i'm changing the qualify value right now |
| 11:00.35 | dlynes_laptop | santibiotico, on either end, that is? |
| 11:00.45 | dlynes_laptop | santibiotico, or is this a purely lan setup for the sip? |
| 11:01.16 | santibiotico | i've changed the qualify value to 300 |
| 11:01.19 | santibiotico | but still the same |
| 11:01.32 | dlynes_laptop | what's your topology? |
| 11:01.33 | santibiotico | dlynes_laptop: i have both servers in the same lan |
| 11:01.46 | santibiotico | dlynes_laptop: no firewall, no router between them... |
| 11:01.46 | dlynes_laptop | Ok, and is this for iax2, or sip? |
| 11:01.53 | santibiotico | dlynes_laptop: iax2 |
| 11:02.26 | dlynes_laptop | Have you tried changing to different ports on your switch? |
| 11:03.51 | dlynes_laptop | Also, try forcing duplex and rate on your nics to match that of the switch |
| 11:04.14 | dlynes_laptop | But I would suggest switching ports first |
| 11:04.24 | santibiotico | yes, i've tried |
| 11:04.41 | santibiotico | obviusly i can ping each peer |
| 11:04.52 | dlynes_laptop | Have you tested your cabling to make sure it complies with a minimum of cat 5? |
| 11:05.30 | santibiotico | sure |
| 11:05.30 | dlynes_laptop | i.e. set up testers at both ends of it |
| 11:05.30 | dlynes_laptop | to make sure there's no significant signal loss? |
| 11:05.30 | santibiotico | i've tried different cables |
| 11:05.36 | santibiotico | connection is working between the servers |
| 11:05.41 | santibiotico | it's not a connection issue |
| 11:05.49 | santibiotico | it might be a misconfiguration ossue |
| 11:05.51 | santibiotico | issue |
| 11:05.58 | dlynes_laptop | yeah...signal loss won't cause loss of connection |
| 11:06.08 | dlynes_laptop | it'll just cause degradation in the quality of the signal |
| 11:06.27 | dlynes_laptop | which can incur lost packets, decreases in speed, network latency, ... |
| 11:07.19 | dlynes_laptop | There are also a number of fine tuning parameters for iax2 as well |
| 11:07.30 | dlynes_laptop | I don't know if any of them will help in your situation, or not |
| 11:07.45 | dlynes_laptop | But you can look at the sample iax2 config file that came with asterisk to find out what they are |
| 11:08.58 | santibiotico | thx, i'll be checking it all |
| 11:13.41 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 11:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 11:20.07 | queuetue | Does anyone make a 6 or 8-port ata? |
| 11:21.10 | queuetue | I suppose I should be looking at PCI cards then, huh? |
| 11:24.16 | tzafrir | queuetue, there are a number of those. But for 8 ports they are sometimes called "channel banks" |
| 11:25.08 | tzafrir | queuetue, audiocodes and such have ones. There should be a rather cheap one by grandstream |
| 11:25.57 | tzafrir | and there is our USB Zaptel device (Xorcom Astribank) |
| 11:27.45 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:28.44 | backblue | anyone have used siemens sl75? |
| 11:28.48 | *** join/#asterisk lorinc (n=ang@caracas-2107.adsl.interware.hu) |
| 11:30.04 | JT | queuetue: there are also real channel banks that connect to a PRI |
| 11:30.12 | JT | queuetue: it really depends what you need to use it for |
| 11:31.06 | queuetue | JT: Just providing 8 local cordless phones with connectivity to an Asterisk server. |
| 11:31.20 | queuetue | tzafrir: I don't see prices on your website - who do you resell through? |
| 11:32.17 | queuetue | Ah, VOIP connection has them. |
| 11:33.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:35.49 | queuetue | Looks like I'll stick with 4 spa's for now - I was hoping price would go down as ports went up. |
| 11:36.12 | queuetue | Price per port, that is. |
| 11:40.49 | tzafrir | queuetue, there are prices there |
| 11:41.04 | tzafrir | look at the bottom of the pages |
| 11:44.51 | *** join/#asterisk AvoidingDeadlock (n=brian@82.152.100.241) |
| 11:47.16 | *** join/#asterisk Newbie___ (n=me@espeed24-92.brunet.bn) |
| 11:47.46 | Newbie___ | hi, can anyone please help me on ADtran 750 |
| 11:47.58 | Newbie___ | cant seems to get the FXS light lit |
| 11:48.36 | Newbie___ | and i am getting a T1 alarm on Adtran, but TE110P says is ok |
| 11:51.27 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 11:58.58 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 12:04.37 | JT | queuetue: there's also a card like a TDM2400P |
| 12:05.06 | *** part/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 12:05.28 | badcfe | ./configure in asterisk-1.4.0 ends with "*** termcap support not found" on my debian. and i _have_ installed termcap-compat. |
| 12:06.24 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 12:08.30 | tzafrir | badcfe, apt-get build-dep asterisk |
| 12:08.44 | tzafrir | specifically: apt-get install libncurses-dev |
| 12:08.52 | badcfe | tzafrir: is this safe when i will afterwards compile 1.4.0 from source? |
| 12:08.55 | tzafrir | or something similar |
| 12:09.15 | badcfe | tzafrir: i mean -- etch dont contain asterisk 1.4.0 yet right? |
| 12:09.25 | tzafrir | badcfe, this installs build dependencies. Feel free to remove some of them (e.g.: libpri) |
| 12:09.34 | tzafrir | no, Etch has 1.2 |
| 12:10.15 | badcfe | tzafrir: ill do that. and then ill remove libpri. so zaptel does not get installed as dependecy? |
| 12:10.33 | puzzled | hi |
| 12:10.39 | tzafrir | libtonezone-dev is |
| 12:11.54 | tzafrir | and also zaptel-source |
| 12:14.53 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:15.55 | *** join/#asterisk reber (i=reber@gateway/tor/x-2af9f67ef5b4eddb) |
| 12:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:23.33 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 12:48.49 | *** join/#asterisk lupino3 (n=andrea@217-133-98-121.b2b.tiscali.it) |
| 12:48.57 | lupino3 | hello * |
| 12:49.17 | lupino3 | I've just posted a message to *-users, and I thought that I might find here an answer |
| 12:49.29 | lupino3 | seems like IAX channels don't respect language setting |
| 12:49.42 | lupino3 | I have "language=it" setting in iax.conf |
| 12:49.50 | lupino3 | and still meetme plays english messages |
| 12:49.58 | lupino3 | how can I get rid of this behaviour? |
| 12:50.00 | lupino3 | TIA |
| 12:54.01 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 13:00.39 | DrukenLPY | good morning asterisk world :) |
| 13:01.19 | *** join/#asterisk orsobob (n=orsobob@static-pro-212-101-21-185.adsl.solnet.ch) |
| 13:03.13 | *** join/#asterisk Makenshi (n=chaz@kenshi.fox.furry.be) |
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| 13:29.18 | orsobob | hello, somebody could help me with LumenVox Speech Engine? |
| 13:30.02 | orsobob | we have bought LumenVox Power Kit for tests... but have some questions |
| 13:30.26 | orsobob | about German, Franche and Italian lamguages |
| 13:30.58 | *** part/#asterisk orsobob (n=orsobob@static-pro-212-101-21-185.adsl.solnet.ch) |
| 13:31.59 | *** join/#asterisk orsobob (n=orsobob@static-pro-212-101-21-185.adsl.solnet.ch) |
| 13:32.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:41.14 | Dr-Linux | which SIP firmware is stable and would be fine for cisco 7960? |
| 13:41.25 | Dr-Linux | currenlty i'm using 7.4 |
| 13:41.49 | Dr-Linux | as i got 10 new phones, so i need suggestion |
| 13:42.13 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:43.02 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 13:49.24 | jeremy_g | what a hairpin call? |
| 13:49.37 | Ahrimanes | nat hairpin? |
| 13:51.04 | Ahrimanes | jeremy_g, http://searchvoip.techtarget.com/sDefinition/0,,sid66_gci1037278,00.html |
| 13:52.53 | jeremy_g | Ahrimanes:thanks |
| 13:52.54 | jeremy_g | :) |
| 13:54.26 | Ahrimanes | jeremy_g, :) |
| 14:01.54 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:05.55 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:09.17 | *** join/#asterisk drako (n=laptop@unaffiliated/luisjose) |
| 14:09.22 | drako | Morning |
| 14:13.24 | *** join/#asterisk susinthsss (n=susinths@marmolada.ifi.uio.no) |
| 14:15.42 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:24.27 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 14:24.44 | jtexter3 | Any Audiocodes users online? |
| 14:25.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:25.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:25.49 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 14:26.52 | qdk | jtexter3: doing a survey? |
| 14:27.03 | jtexter3 | qdk: No, trying to get it to work :D |
| 14:27.19 | *** join/#asterisk treat (n=fhe@tvalk.campus.luth.se) |
| 14:27.27 | jtexter3 | Just having an issue with audio when doing attended transfers |
| 14:29.29 | jtexter3 | When doing an attended transfer, if you complete the transfer, Asterisk appears to send a BYE to the Gatway, so the gateway thinks the connection is broken |
| 14:29.39 | qdk | jtexter3: ok, I would use such a box for a thing like that, so no input from me. |
| 14:30.12 | jtexter3 | It appears to be a nice little gateway. Makes for an easy way to get a lot of channels in a server |
| 14:30.46 | qdk | "get a lot of channels in a server" <- say what? |
| 14:31.43 | *** join/#asterisk Mad|Cow (n=thirt@c-69-242-72-104.hsd1.de.comcast.net) |
| 14:31.46 | jtexter3 | I haven't tested this myself, but from everything I've read, you can't really put more than 2 Digium cards in a server |
| 14:31.55 | jtexter3 | That limits you to 8 T1's/E1's |
| 14:32.22 | jtexter3 | The Audiocodes gateway I have right now is capable of handling a DS-3 (28 T1's |
| 14:32.36 | treat | i can only register sip clients to asterisk when the password is blank otherwise i get "Wrong password" |
| 14:32.45 | treat | what am i doing wrong? |
| 14:33.05 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.148.215) |
| 14:34.03 | qdk | jtexter3: well is a lot more hardware in the world than just digium (and audiocodes). |
| 14:34.51 | qdk | there is* |
| 14:35.26 | *** join/#asterisk zapata (n=herbert@norway.ath.cx) |
| 14:35.27 | jtexter3 | qdk: Agreed. I'll be using Aculab soon, which supports 8 ports in a single blade. What I haven't gotten an answer on is whether the 2 card limitation is a fundamental limitation in the channel architecture, or simply a limit with the Digium hardware |
| 14:36.28 | [TK]D-Fender | jtexter3: Well You could get 2 Sangoma's in there nice and comfy @ 8 ports ea :) |
| 14:36.48 | qdk | jtexter3: the limits is probably i both digium and asterisk. Mostly asterisk. |
| 14:37.06 | jtexter3 | [TK]D-Fender: I've been wanting to try a Sangoma card, just haven't gotten around to purchasing one yet |
| 14:37.55 | qdk | jtexter3: I have only heard good things about the hardware... not so much on their own drivers, but zaptel drivers actually make good use of the sangoma cards. |
| 14:39.13 | [TK]D-Fender | qdk: All of my experiences have been great. 0 echo, no PCI/IRQ issues, nada |
| 14:42.17 | *** join/#asterisk awannabe (n=brad@216.161.165.212) |
| 14:43.20 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 14:43.38 | qdk | [TK]D-Fender: with what and to what? |
| 14:43.38 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 14:44.04 | *** join/#asterisk bkw__ (n=brian@82.152.100.241) |
| 14:44.18 | qdk | [TK]D-Fender: inspired is reselling Sangoma cards, perhaps he can help you with a new shinny one. ;-) |
| 14:44.53 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:44.54 | inspired | shitty? |
| 14:45.17 | qdk | inspired: ? |
| 14:45.25 | inspired | :-) |
| 14:45.40 | inspired | if he's in canada, I can't help him. there should be plenty of other resellers there though :-) |
| 14:46.44 | qdk | ok |
| 14:51.49 | [TK]D-Fender | qdk: I'd worked with the A102/4d, and A200d |
| 14:52.07 | [TK]D-Fender | And yes, I'm in Canada :) |
| 14:53.10 | *** join/#asterisk atnonis (n=Java6936@cityairnetworks.ath.forthnet.gr) |
| 14:53.15 | atnonis | o/ |
| 14:53.30 | atnonis | anyone have experience with asterisk and OpenBSD(3.9) ? |
| 14:54.14 | atnonis | im missing iax.conf in OpenBSD, is that a port problem or i have to create it my self? |
| 14:54.23 | mercestes | atnonis: YEs! Just not together |
| 14:54.31 | atnonis | :P |
| 14:54.47 | atnonis | i mean together |
| 14:55.15 | qdk | [TK]D-Fender: ok, a bit wage on the details, but at least I know its Sangoma and not digium you are talking about. |
| 14:55.22 | *** join/#asterisk dongc (i=gr8Heale@bb219-74-171-89.singnet.com.sg) |
| 14:55.35 | [TK]D-Fender | qdk: ? |
| 14:56.36 | qdk | atnonis: are you sure IAX is compiled? coz I had to patch OpenPBX (and now fixed in release candidate) to make IAX work on OpenBSD. |
| 14:57.44 | De_Mon | asterisk 1.2.14 Ive got agents in bridged calls, or in meetme rooms and suddenly get |
| 14:57.48 | De_Mon | update_header: Unable to find our position errors |
| 14:57.51 | De_Mon | err warnings |
| 14:58.14 | De_Mon | The call is being recorded by mixmonitor |
| 14:58.37 | De_Mon | SHIT nevermind hd is full |
| 15:01.06 | atnonis | qdk: i have download asterisk-1.2.4p0.tgz (i386) from ports i dont know if iax is compiled in |
| 15:01.53 | De_Mon | atnonis 1.2.14? |
| 15:01.57 | atnonis | qdk: how can i see that? |
| 15:02.01 | qdk | atnonis: ok, I dont know either. |
| 15:02.09 | atnonis | De_Mon: yes 1.2.14 |
| 15:02.50 | qdk | atnonis: see if the chan_iax2.so is in libs i guess. |
| 15:03.37 | Mad|Cow | For my sip clients, in sip.conf, should I be using type=friend or type=user? It doesnt seem to be clear when I should be using one v.s. the other |
| 15:05.12 | mercestes | Mad|Cow: I like having friends. |
| 15:06.18 | atnonis | qdk: it seems that no modules are installed :( |
| 15:06.45 | Mad|Cow | mercestes: Any reason? |
| 15:06.46 | qdk | atnonis: that doesnt seem right. |
| 15:06.58 | mercestes | Mad|Cow: Because I hate being lonely. |
| 15:07.12 | Mad|Cow | mercestes: That was a cheap setup ;-) |
| 15:07.21 | qdk | atnonis: but for anything other than linux i would recommend using OpenPBX... and probably on Linux as well. |
| 15:07.35 | mercestes | Mad|Cow: hehe. Should be friends tho so they can make and recieve calls. |
| 15:08.18 | Mad|Cow | mercestes: What happens if it type=user? They cant make calls? |
| 15:08.45 | hwt | anyone have problems with thomson st2030 and calls that fail to hang up? |
| 15:09.03 | hwt | even though i have rtptimeout = 60, it still stays open. |
| 15:09.13 | mercestes | Mad|Cow: It breaks ChanIsAvail(). They should probably be type=friend. Type=user or type=peer is more for * to * sip connects. |
| 15:09.20 | hwt | and the phone has status = unreachable |
| 15:09.23 | hwt | asterisk 1.2.10. |
| 15:09.45 | Mad|Cow | mercestes: got it. thanks |
| 15:10.20 | hwt | the weird thing is that it's only st2030's. |
| 15:10.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:10.49 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:12.15 | *** join/#asterisk mavior (n=Miranda@88-149-160-22.f5.ngi.it) |
| 15:12.25 | mavior | hello everybody |
| 15:13.13 | hwt | oh, and they're all nat-ed. |
| 15:13.18 | hwt | so that could also be the reason. |
| 15:13.21 | mkl1525 | Hi, atm I'm trying to get hinting with my snoms to work. basically it works but on the display monitoring extension 201 I see "From: 201 To: 201" - does anybody know what I've got to do so that it shows "From: 4566656 To:201"? |
| 15:14.29 | mavior | people anyone experienced some problems with incoming calls using some sip provider (i mean by registering to them with asterisk) ? |
| 15:14.40 | mavior | i have problem with skypho an italian provider |
| 15:15.13 | mavior | seems that some minutes after started asterisk the incoming calls stop to work!!! |
| 15:15.50 | mavior | foe the first 5-8 miinutes it works...sometimes more, and then it stop |
| 15:16.31 | *** join/#asterisk elriah (n=johnny@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:16.44 | mitcheloc | hey guys, front page of digg -- http://www.digg.com/linux_unix/Mark_Spencer_Presents_AsteriskNOW |
| 15:16.46 | elriah | Hi all. Where does one download the national 911 database of psaps? |
| 15:16.46 | mitcheloc | digg it! |
| 15:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.13 | mavior | i had pastebinned my debug , can someone help me? seems that is something related to the registration expiry time http://pastebin.ca/318415 !!!! |
| 15:18.41 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:20.32 | [TK]D-Fender | mavior: SIP/2.0 401 Unauthorized |
| 15:20.50 | [TK]D-Fender | mavior: Looking for me in mycontext (domain 192.168.1.102) |
| 15:21.00 | [TK]D-Fender | mavior: SIP/2.0 404 Not Found |
| 15:21.19 | [TK]D-Fender | mavior: Bad passwords, bad target extens. Your setup is a mess |
| 15:21.44 | mavior | mhhh.but it works for the first minutes |
| 15:21.59 | mavior | and i' mlogged to my asterisk server with x-lite |
| 15:22.10 | mavior | "me" is my username on asterisk |
| 15:23.30 | [TK]D-Fender | mavior: Just because you can register doesn't mean std auth on calls will work or surpass timeouts. |
| 15:23.44 | [TK]D-Fender | mavior: Fic your password issues |
| 15:23.45 | [TK]D-Fender | fix* |
| 15:24.11 | mavior | for example, now it is working and the log is here http://pastebin.ca/318473 |
| 15:24.13 | Marty-OTT | PRetty neat.. |
| 15:24.16 | Marty-OTT | asterisknow |
| 15:25.43 | mavior | [TK]D-Fender: i don't understand what is my pass problem... |
| 15:26.45 | [TK]D-Fender | mavior: That looks like an OUTGOING call... |
| 15:26.48 | Marty-OTT | Is AsteriskNow only BETA version? |
| 15:27.46 | *** join/#asterisk hassler (n=hassler@r-corp.hcst.com) |
| 15:28.12 | hassler | Mornin folks! I'm curious about any folks working with Asterisk Business Edition -- I'm not impressed with it so far. |
| 15:28.47 | mavior | [TK]D-Fender: the second log? |
| 15:28.58 | [TK]D-Fender | mavior: yes |
| 15:29.04 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 15:29.23 | [TK]D-Fender | hassler: Wow... very qualified remark. Care to actually state your problem? :) |
| 15:29.33 | hassler | :-) |
| 15:29.44 | mavior | [TK]D-Fender: wait i'll post my little sip.conf and ext conf |
| 15:30.58 | hassler | so far we've backed off the rPath distribution, and now having difficulties getting freePBX installed with it (doesn't look like that's going to work either). |
| 15:32.08 | hassler | We know 1.2 + freePBX works well, which is basically what my client wants, but wanted the "supported" version |
| 15:32.30 | [TK]D-Fender | hassler: If I'm not mistaken, FreePBX tries to keep with more mainstream releases of *, and as ABE is sometimes noticably behind (Don't think they're on 1.4), it may not be compatible. |
| 15:33.27 | [TK]D-Fender | hassler: And who said you had to use rPath? While we're at it, what are you expecting out of ABE? Does Digim support FreePBX for you just because you want to run it on top of ABE? |
| 15:33.30 | hassler | you are correct, it is not compatible. Now, the only real problem we're having (with freePBX -- that we've noticed so far) is that the extra config files (sip_additional.conf for instance) are not generated. |
| 15:34.26 | hassler | fender - no one, which is why we backed off of it. However, only RHEL and FCx are "supported" -- we backed off to CentOS for expediency. |
| 15:35.09 | Dr-Linux | why is this happening >> Jan 16 07:31:02 WARNING[25129]: chan_iax2.c:7532 socket_read: Received mini frame before first full voice frame |
| 15:35.11 | mavior | [TK]D-Fender: this is my sip.conf configuration http://pastebin.ca/318489 and this one is my only two extensions http://pastebin.ca/318492, and i register to voip.eutelia.it trough register command and i put my server online passing asterisk the externhostname option |
| 15:35.41 | [TK]D-Fender | hassler: Well so far your problem appear to be FreePBX, not * in any way..... |
| 15:35.51 | mavior | http://pastebin.ca/318492i read one teoric manual and made this conf by myself |
| 15:37.09 | [TK]D-Fender | mavior: You don't have a [general] section? |
| 15:37.18 | mavior | [TK]D-Fender: yes i have |
| 15:37.35 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:37.48 | mavior | [TK]D-Fenderi only posted the most important sections |
| 15:38.03 | mavior | [TK]D-Fender |
| 15:38.19 | [TK]D-Fender | mavior: Ok, well I don't know what they require, but I suggest you show them your config to see if they can see anything obvious that I'm missing... |
| 15:39.11 | mavior | [TK]D-Fender:who the other people here? |
| 15:39.26 | [TK]D-Fender | mavior: Your ITSP |
| 15:40.07 | mavior | [TK]D-Fender: mhhh they require this http://www.skypho.net/download/asterisk.html |
| 15:40.25 | hassler | problems, correct. but what's the real motivation for Asterisk Business Edition? |
| 15:41.45 | [TK]D-Fender | hassler: Ask yourself "why did I pay for it?". What were YOU expecting? |
| 15:42.13 | *** join/#asterisk Marty-OTT (n=marty@209.50.87.3) |
| 15:42.23 | mavior | [TK]D-Fender: even though some options seems to be not necessary.....anyway my config is shit ? do you think that it could be made better? i mean there are some more "right ways" to do this simple thing: have an asterisk server, connect to it with xlite(so i made my user "me" ), use voipcheap for out calls and voip.eutelia.it for incoming calls ? |
| 15:42.26 | Marty-OTT | I'm going to try an install of AsteriskNOW |
| 15:43.51 | [TK]D-Fender | mavior: Funny.. your config doesn't look anything like that.... |
| 15:43.59 | hassler | *I* didn't -- client did. *THEY* are expecting SUPPORT, and were struggling with installation and expecially GUI interface (they first tried ThirdLane, which was also a failure). Now I'm trying to help them out. Rather experienced with Asterisk (using 1.2 on several installations) |
| 15:44.08 | Marty-OTT | Does anyone have that link on you tube again for AsteriskNow Mark Spencer? |
| 15:44.19 | file | http://www.digg.com/linux_unix/Mark_Spencer_Presents_AsteriskNOW |
| 15:44.43 | hassler | Which brings up another point, can AsteriskNOW fit in the picture with AsteriskBE? Sure looks good! |
| 15:44.43 | mavior | [TK]D-Fender: ehmm..why? after all it does the job right now, less this small imcoming problems |
| 15:44.55 | mavior | [TK]D-Fender can you suggest one way to do it, please? |
| 15:45.25 | [TK]D-Fender | hassler: Well I guess they should have done their homework. If all they wanted was FreePBX they might as well have gone with Trixbox. |
| 15:45.44 | Marty-OTT | thx |
| 15:45.53 | x86 | [TK]D-Fender: i actually prefer FreePBX to a full-blown trixbox |
| 15:45.59 | [TK]D-Fender | mavior: Check your [general] section to see if it looks right |
| 15:46.00 | queuetue | What is the linux distro sitting under asterisknow? |
| 15:46.04 | x86 | [TK]D-Fender: trixbox's interface is crap, IMHO |
| 15:46.04 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 15:46.09 | Marty-OTT | Mark Spencer is young - holy crap |
| 15:46.18 | Marty-OTT | Very impressive |
| 15:46.20 | [TK]D-Fender | x86: Which interface would that be? |
| 15:46.51 | queuetue | Is it debian-based? |
| 15:48.38 | tzafrir | queuetue, it is rPath |
| 15:49.00 | x86 | [TK]D-Fender: the trixbox web interface |
| 15:49.13 | x86 | [TK]D-Fender: the purple thing that puts freepbx in a little frame ;) |
| 15:49.24 | [TK]D-Fender | x86: What aspects of that are seperate from FreePBX itself? |
| 15:49.38 | [TK]D-Fender | x86: Thats IT> The whole of your complaint? |
| 15:50.14 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:51.43 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:51.46 | mavior | [TK]D-Fender: can you please have a look? this is the entire sip.conf http://pastebin.ca/318518 |
| 15:51.46 | x86 | it was never really a complaint :) |
| 15:51.54 | x86 | [TK]D-Fender: personal preference |
| 15:52.04 | x86 | [TK]D-Fender: i dont appreciate the extra cruft that trixbox adds |
| 15:52.13 | [TK]D-Fender | x86: I'll lump that into the "whoopee shit" category, kplzthxbibi |
| 15:52.16 | x86 | [TK]D-Fender: freepbx itself is enough bloat for me ;) |
| 15:52.26 | x86 | [TK]D-Fender: lol, exactly :) |
| 15:52.34 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 15:52.40 | x86 | i'm gonna check out this asterisknow thingy though |
| 15:53.13 | [TK]D-Fender | mavior: Your [general] section doesn't have any of the parameters that I saw in their sample.... and remove all that commented out crap |
| 15:54.04 | queuetue | So, an *Now box basically can't be used for anything but asterisk... |
| 15:54.07 | Marty-OTT | hmmm... can I use Nero to burn my AsteriskNOW ISO image? |
| 15:54.25 | queuetue | Marty-OTT: If it burns ISOs, then yes. |
| 15:54.27 | mavior | [TK]D-Fender: mhh but i have added them to my [voip.eutelia.it] section, is it not enough? |
| 15:54.50 | [TK]D-Fender | mavior: How about you FOLLOW THE INSTRUCTIONS, before wondering why it doesn't work... |
| 15:55.39 | *** join/#asterisk tpetrosky (n=tpetrosk@asfs.agencysacks.com) |
| 15:56.20 | mavior | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+settings+for+skypho, i followed this one |
| 15:57.29 | mavior | and all seems to work |
| 15:57.35 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:58.19 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
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| 16:06.44 | *** part/#asterisk lupino3 (n=andrea@217-133-98-121.b2b.tiscali.it) |
| 16:07.08 | *** part/#asterisk dongc (i=gr8Heale@bb219-74-171-89.singnet.com.sg) |
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| 16:10.00 | *** join/#asterisk ManxPower (n=manxpowe@217.sub-75-200-162.myvzw.com) |
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| 16:12.42 | *** join/#asterisk shinux__ (n=shinux@196.220.28.157) |
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| 16:16.59 | vader-- | hey |
| 16:17.18 | vader-- | anyone ever see where a cisco 7940G phone would have the Message waiting indicator on and no message is waiting |
| 16:17.22 | *** join/#asterisk RichL (n=RichL@68.143.17.4.nw.nuvox.net) |
| 16:17.27 | vader-- | and the user never had one? |
| 16:17.43 | mercestes | vader--: Yes. If I set teh "mailbox" incorrectly in sip.conf. |
| 16:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 16:22.44 | vader-- | mercestes good call |
| 16:22.48 | vader-- | thanks |
| 16:24.25 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:24.28 | mercestes | NP..:) |
| 16:24.58 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
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| 16:28.32 | *** join/#asterisk bipolar (n=bflong@146.145.26.91) |
| 16:29.08 | bipolar | hey everyone. I'm having a lunch meeting with the boss in about an hour, and one of the things we're going to discuss is a new phone system. |
| 16:29.44 | bipolar | I'm going to push for an asterisk solution, but I'm wondering if anyone has used the Grandstream GXP-2000 phones and can share their experiances with it. |
| 16:30.01 | bipolar | it seems, well, a bit on the cheap side. |
| 16:30.38 | bipolar | and I'm wondering if it really is cheap, in the negitive sense.... or just amazingly inexpensive ;) |
| 16:31.03 | [TK]D-Fender | bipolar: GrandSuck is a terrible idea |
| 16:31.18 | [TK]D-Fender | bipolar: Avoid with extreme prejudice |
| 16:31.22 | bipolar | [TK]D-Fender: ok... thats what I was looking for. :) |
| 16:31.30 | bipolar | [TK]D-Fender: any recomendations? |
| 16:31.32 | [TK]D-Fender | bipolar: Where are you located? |
| 16:31.53 | [TK]D-Fender | bipolar: Planning on PoE? Need speakerphone? |
| 16:32.10 | [TK]D-Fender | bipolar: Backlight a heavy selling point? |
| 16:32.26 | bipolar | [TK]D-Fender: No PoE atm.... speakerphone yes, backlight no, in northeast USA |
| 16:33.03 | [TK]D-Fender | bipolar: Polycom IP 501's for general users then, IP 601 for receptionist. www.telephonydepot.com |
| 16:33.11 | [TK]D-Fender | bipolar: $170 USD |
| 16:33.14 | bipolar | [TK]D-Fender: cool. thank you :) |
| 16:33.20 | [TK]D-Fender | bipolar: np, and gl |
| 16:33.25 | bipolar | thanks again. |
| 16:33.38 | vader-- | bipolar defender is the man |
| 16:33.40 | vader-- | listen to him |
| 16:33.42 | hassler | Very happy with my Grandstreams (using GXP-2000), the BT-102's are hard to read with the flat displayes |
| 16:34.16 | vader-- | i went with cisco 7940G |
| 16:34.20 | hassler | Agree that Polycoms are even better |
| 16:34.25 | vader-- | wish i would went with newer cisco |
| 16:34.31 | bipolar | hassler: cool. yes, we need quality |
| 16:34.36 | vader-- | but the cisco 7940G work great |
| 16:34.58 | *** join/#asterisk amdtech (i=adaniel@nat/digium/x-fa84570583ded71b) |
| 16:35.07 | vader-- | like a cisco 7941g or somthing |
| 16:35.31 | [TK]D-Fender | bipolar: In order of suggestability : (Polycom (any), Aastra 480i, Cisco (7960+), Linksys. |
| 16:36.00 | bipolar | [TK]D-Fender: again... thanks! Lots of info :) |
| 16:36.13 | [TK]D-Fender | vader--: 7941 adds 802.3af IIRC and a double-res screen. Pics looked nice. Then again it wouldn't be "marketing" if they weren't ;) |
| 16:38.45 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:39.47 | vader-- | defender ya |
| 16:39.58 | vader-- | i ran into the issue with the 7940g where they used cisco POE |
| 16:40.04 | vader-- | kinda put a burden on things |
| 16:40.09 | vader-- | but i worked around it |
| 16:40.31 | Mad|Cow | If I have RTP configured to listen on a certain port range (in my rtp.conf file) should I not see that range listening if I do a netstat -na on the system running asterisk? |
| 16:40.52 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:40.54 | Qwell[] | Mad|Cow: nope, you shouldn't unless a call is initiated |
| 16:41.17 | mercestes | Try doing an nmap --insane on it. |
| 16:41.21 | mercestes | no wait, don't do that. |
| 16:41.26 | Mad|Cow | Gotcha... so the asterisk server decies which port to use |
| 16:41.33 | Qwell[] | Mad|Cow: yes, and when |
| 16:41.51 | mercestes | I mean I guess you could, if you didn't plan on passing any calls on it for a while. |
| 16:45.31 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:47.33 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
| 16:48.09 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 16:49.12 | Mad|Cow | If I have a Asterisk server that is public accessiable on the I-Net (not NAT or firewall in front), if I am using xlite at a remote site (behind a firewall) can I connect to Asterisk? |
| 16:50.09 | [TK]D-Fender | Mad|Cow: typically, sure |
| 16:50.15 | HarryR | Mad|Cow, depending on the firewall, you should be able to when using STUN or something like xTunnels |
| 16:50.33 | [TK]D-Fender | Mad|Cow: I run double-nat scenarios all the time without trouble. |
| 16:50.54 | [TK]D-Fender | HarryR: * doesn't support STUN, nor is it needed in the client most of the time. |
| 16:51.09 | CrashHD | no need for stun with client behind firewall |
| 16:51.12 | CrashHD | just nat=yes |
| 16:51.15 | CrashHD | qualify=yes |
| 16:51.20 | CrashHD | works pretty well |
| 16:51.28 | CrashHD | for most setups anyway |
| 16:51.42 | *** part/#asterisk BarnacleBob (n=karl@38.99.18.98) |
| 16:51.54 | [TK]D-Fender | CrashHD: *cough* canreinvite=no *cough* |
| 16:52.00 | CrashHD | heh ya and that |
| 16:52.13 | CrashHD | but I run that as a default |
| 16:52.29 | Mad|Cow | qualify=yes? If they are behind a firewall though, it wont be able to ping... will that still work? |
| 16:52.29 | [TK]D-Fender | CrashHD: Assume nothing :) |
| 16:52.45 | CrashHD | it can still qualify |
| 16:52.56 | [TK]D-Fender | Mad|Cow: it pings through a SIP packet which keeps the UDP port "open" on the remote NAT router. |
| 16:53.10 | CrashHD | assume everything, explore nothing, help only when required |
| 16:53.13 | [TK]D-Fender | Mad|Cow: Thats how it succeeds |
| 16:53.21 | CrashHD | :) |
| 16:53.29 | Mad|Cow | ahhh... that explains a lot |
| 16:53.32 | [TK]D-Fender | CrashHD: If you have to give 1 setting, you probably have to revisit them ALL :) |
| 16:53.42 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:54.00 | Mad|Cow | If I dont have qualify=yes, then it might try to jump around on different ports? |
| 16:54.21 | [TK]D-Fender | Mad|Cow: No, the port will close and your client won't get calls. |
| 16:55.10 | CrashHD | Mad|Cow: basically your phone opens a port when it goes outbound to register with the system...you have to keep that hole open |
| 16:55.30 | CrashHD | the router will only open holes when going from the lan to wan...and not wan to lan (security) |
| 16:55.57 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 16:56.13 | Mad|Cow | Makes sence |
| 16:59.09 | naitram | Ive got a lot of repition in this http://pastebin.ca/318595 . Need to know how to run setVar in a context regardless of which extension is dialed in the context |
| 16:59.40 | variable_office | dtmf is not working for me in any way, inband, rfc2833, info; none work; what are some common problems that could make this happen? |
| 17:00.27 | *** join/#asterisk andyshack (n=andyshac@203-59-134-11.perm.iinet.net.au) |
| 17:00.33 | andyshack | evening. |
| 17:00.49 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 17:00.50 | *** join/#asterisk watchy (n=wgag@work.gwhsi.com) |
| 17:01.01 | watchy | anyone seen a 601 not bootup its sidecars? |
| 17:01.04 | SuPrSluG | hi all |
| 17:01.10 | andyshack | this is asterisk now gui related question. hope you dont mind.. |
| 17:01.11 | andyshack | quick question, sorry i havent the time to go through the faqs : i was going to deploy this on a nice server with lots of scsi raid. i was going to use to raid 5 drives. using the expert gui config will it be simple enough to ask asterisk to save all recordings on raid drive #2 whereas the os etc. is on raid drive #1 ? |
| 17:01.13 | hoobastooba | which line in the iax.conf is it that I use to cause the server to register periodically in case a connection get dropped. |
| 17:01.20 | watchy | i just put in a new 601 in a place and its not loading its side cars for some reason |
| 17:01.48 | [TK]D-Fender | exten => 101,7,SetVar(PLCUNITexten => 101,1,SetVar(MEMAREA=7400) |
| 17:01.48 | robl^ | watchy: what do you mean not loading? |
| 17:01.55 | hoobastooba | i periodically needed to log in and restart asterisk so that it will register with my iax provider again. |
| 17:01.58 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 17:01.58 | [TK]D-Fender | naitram: Looks kinda broken |
| 17:02.04 | [TK]D-Fender | naitram: and very deprecated |
| 17:02.18 | watchy | robi: you know when most 601s boot finish then the side cars start to init on up and light up |
| 17:02.20 | [TK]D-Fender | naitram: Make a Macro out of that. |
| 17:02.21 | watchy | it never happens |
| 17:02.43 | *** join/#asterisk reber (i=reber@gateway/tor/x-27b8d83d80c028a8) |
| 17:02.47 | RichL | having a problem with call routing (2 connections to voicepulse) working fine, (3 pots lines connected via TDM400P) inbound is working, but outbound is only allowing the first line to used everyone else get a scretching sound. |
| 17:03.03 | robl^ | watchy: no polycom logo.. nothing? |
| 17:03.05 | naitram | [TK]D-Fender: yeah, this silly gedit broke it , sometimes the paste doesnt put it where i think it is. It does work (not as shown) |
| 17:03.11 | watchy | i read it should be auto if they have more then 6 or 8 or something contacts. the phone has like 30 |
| 17:03.18 | watchy | robl: nothin man |
| 17:03.45 | fall0ut | watchy: sounds like the power supply is messed up |
| 17:04.06 | watchy | wierd, ill take another 601 out there and test it |
| 17:04.08 | fall0ut | watchy: we had a bunch of them that the cube was messed up and caused that |
| 17:04.29 | Mad|Cow | hmm... it's strange though... I just cant get xlite to register to Asterisk. tcpdump shows my responce to the port xlite is at as failing (port unreachable). Its like my xlite client isnt keeping the port open on the firewall. Any ideas? |
| 17:04.39 | [TK]D-Fender | bbiab |
| 17:04.50 | robl^ | watchy: make sure you assign the contacts a speed dial number |
| 17:05.16 | robl^ | if they don't have a speed dial number, then they won't show on sidecars |
| 17:05.39 | watchy | robl: yea well i copied over a 50 user directory to the phones conf on the server and it loaded it |
| 17:05.50 | StephenL | Any idea when I try to dial an extension it always goes to their voicemail? It does it for all extensions. We just have a default asterisknow install with a couple of extensions defined. |
| 17:05.51 | watchy | and it still didnt load up the sidecars |
| 17:05.56 | *** part/#asterisk andyshack (n=andyshac@203-59-134-11.perm.iinet.net.au) |
| 17:06.28 | naitram | [TK]D-Fender: as far as depricated, ill upgrade when i get time. its working now and am approaching a deadline. Do you think that ver 1.0.7-BRI is substantially depricated? |
| 17:06.47 | robl^ | watchy: they have <sd>{some number]</sd> tags for each entry in your directory file? |
| 17:06.54 | watchy | yea |
| 17:07.01 | watchy | they should ill check |
| 17:07.07 | watchy | they work on the other 601s |
| 17:07.28 | watchy | i have 5 of them deployed, i just added a new one and its not working |
| 17:07.35 | variable_office | is there any way to check the workings of dtmf to see if asterisk is even sending it? |
| 17:07.40 | robl^ | if it works on other 60s, but not that one.. then I'd say it may be a hardware issue |
| 17:08.00 | watchy | ill trade out the 601 next time im out there then i guesss |
| 17:09.11 | watchy | i got a issue with a 501 dialing fine passing it to asterisk showing it dialing the right # but im getting the operater giving me an error 90% of the time |
| 17:09.56 | CrashHD | variable_office: sip debug or iax debug usually shows dtmf activity |
| 17:10.35 | variable_office | CrashHD what does it mean if it shows nothing? |
| 17:10.51 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 17:10.52 | CrashHD | I would verify you are getting dtmf from the phone first |
| 17:11.16 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 17:11.30 | CrashHD | if you are not seeing sip debug messages about dtmf my guess would be inband is being used |
| 17:11.31 | variable_office | how can i verify that? |
| 17:11.39 | CrashHD | same way you would see outbound |
| 17:11.46 | CrashHD | should see dtmf even inbound as well |
| 17:11.59 | variable_office | it is currently set to rfc2833 |
| 17:11.59 | CrashHD | make sure to use rfc if possible |
| 17:12.37 | CrashHD | sip show channel $#$@$@PTJ@JTP@JT@PTJ should show you your negotiated dtmf mode |
| 17:12.44 | variable_office | what line should i be looking for in the prinout of sip debug? |
| 17:12.45 | CrashHD | I would verify the devices have negotiated rfc |
| 17:12.53 | Dr-Linux | again forgot |
| 17:13.09 | Dr-Linux | how can i unblock cisco 7960? |
| 17:13.09 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:13.29 | CrashHD | you should see a dtmf event |
| 17:13.33 | CrashHD | when you press a key |
| 17:13.42 | Mad|Cow | I think its **#* |
| 17:13.47 | Mad|Cow | or **# |
| 17:14.06 | variable_office | if i type sip show channels i get 0 active channels (possibly because of realtime)? |
| 17:14.18 | Mad|Cow | CrashHD: but it depends on your firmware version |
| 17:14.35 | ManxPower | Dr-Linux: you go to cisco's web site and find the docs for it. |
| 17:14.37 | Mad|Cow | CrashHD: 8.x requires you to do it within the menu |
| 17:14.38 | CrashHD | you have to have an active phone call for it to show a channel |
| 17:14.57 | variable_office | ah, gotcha, my bad |
| 17:15.01 | variable_office | trying now |
| 17:15.15 | CrashHD | and you want show channel <channel id> |
| 17:15.23 | CrashHD | will give you more details |
| 17:16.40 | variable_office | <PROTECTED> |
| 17:16.44 | variable_office | is what it says |
| 17:16.44 | CrashHD | ok |
| 17:16.49 | CrashHD | for which channel |
| 17:16.51 | CrashHD | your phone |
| 17:16.55 | CrashHD | or your outbound connection? |
| 17:17.02 | variable_office | phone, ill check outbound now |
| 17:17.06 | CrashHD | I forgot to ask how you were sending outbound |
| 17:17.10 | CrashHD | but I assumed it was sip |
| 17:17.29 | variable_office | sip, yes |
| 17:17.33 | CrashHD | brb |
| 17:17.37 | CrashHD | breakfast |
| 17:17.43 | variable_office | outbound also: |
| 17:17.43 | variable_office | DTMF Mode: rfc2833 |
| 17:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:21.07 | *** join/#asterisk meshugga (i=philip@loeblich.linuxteam.at) |
| 17:21.11 | meshugga | hey chaps |
| 17:21.24 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 17:21.26 | *** part/#asterisk FireW0lf (n=me@velocitycs1.demon.co.uk) |
| 17:21.27 | meshugga | i wonder if i could somehow get ringing events with the asterisk manager api? |
| 17:21.33 | meshugga | i havent yet found a way to do that |
| 17:21.44 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:22.06 | cpm | hrmm |
| 17:22.35 | cpm | how can I make my asterisk box 'answer' with a fast busy? aside from recording a fast busy and using that as welcome? |
| 17:22.51 | hoobastooba | if my iax registration drops for whatever reason, is it qualifyfreqok that makes it reregister? |
| 17:23.07 | meshugga | to be more specific: i do "Originate", and asterisk won't give me any status about the ringing channel unless the called party takes the call |
| 17:23.07 | ManxPower | cpm: Congestion() |
| 17:23.07 | CrashHD | cpm: playtones(congestion) |
| 17:23.07 | fetcher | cpm: 1,Answer, 2,Congestion ? |
| 17:23.27 | CrashHD | cpm: after you setup....indications.conf I believe |
| 17:23.30 | cpm | ManxPower, Doh! |
| 17:23.31 | cpm | thanks |
| 17:23.39 | variable_office | CrashHD did you see that outbound was rfc2833 as well? |
| 17:23.43 | cpm | CrashHD, fetcher et al, thanks |
| 17:23.45 | CrashHD | scrolling |
| 17:24.04 | StephenL | Any idea when I try to dial an extension it always goes to their voicemail? It does it for all extensions. We just have a default asterisknow install with a couple of extensions defined. |
| 17:24.10 | ManxPower | If you Answer first, then the caller will be billed, as the call will have been connected. If you simply run Congestion() then the call will not be connected, but the calling device will be signaled as to the congestion condition with whatever sound their telco uses for that indication |
| 17:24.19 | CrashHD | variable_office: so now make a call through your box and hit a key while sip debug is on |
| 17:24.27 | *** join/#asterisk SomethingISODD (n=dan@HS196-230-91.nt.net) |
| 17:24.32 | CrashHD | actually |
| 17:24.36 | SomethingISODD | hello what Stun server is recommended to use with asterisk? |
| 17:24.38 | CrashHD | terminate at your box somehow |
| 17:24.44 | ManxPower | StephenL: the phone you are trying to call is not registered. pastebin.ca the CLI output if you are NOT running FreePBX/whatever |
| 17:24.45 | CrashHD | to make sure phone to asterisk dtmf is working |
| 17:24.49 | ManxPower | SomethingISODD: no |
| 17:24.50 | CrashHD | then call through |
| 17:24.56 | SomethingISODD | ManxPower? |
| 17:25.13 | CrashHD | is it certain dtmf not getting through? or all? |
| 17:25.19 | ManxPower | SomethingISODD: STUN is not recommend to be used with Asterisk. Asterisk's nat=yes setting does the same thing. |
| 17:25.33 | adde | I just purchased a USB Bluetooth Dongle... Anyone who could point me in the direction to get it set up? using centos... |
| 17:25.43 | SomethingISODD | hrm ok i am having a problem with one of my ATA`s they will not connect to asterisk |
| 17:25.47 | ManxPower | adde: try #CentOS |
| 17:25.50 | adde | is there a general driver or do i kneed something specific for my device? |
| 17:25.50 | CrashHD | adde: search the web, best bet |
| 17:25.53 | SomethingISODD | i don`t even see the attempt |
| 17:26.00 | ManxPower | SomethingISODD: STUN will not solve that problem. |
| 17:26.04 | variable_office | CrashHD while i pressed numbers in the call, nothing was printed out |
| 17:26.09 | SomethingISODD | ManxPower oh ok. |
| 17:26.11 | SomethingISODD | thanks. |
| 17:26.18 | CrashHD | nothing at all....? |
| 17:26.21 | CrashHD | what version of asterisk? |
| 17:26.29 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 17:26.41 | variable_office | 1. |
| 17:26.43 | ez` | i bought a polycom ip500 75$ US ebay ; they are good as ip501 ??? i always use ip501 |
| 17:26.46 | variable_office | 1.2.9.1 |
| 17:26.51 | CrashHD | upgrade |
| 17:27.04 | variable_office | i tried on 1.4, and the same thing |
| 17:27.06 | cpm | ManxPower, the idea is to allow folks who know me to connect through to my extension(s) but everyone else, go away |
| 17:27.13 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:27.21 | CrashHD | well I remember around that version there was a bug having to do with feature codes and dtmf |
| 17:27.31 | CrashHD | try disabling all your features |
| 17:27.33 | CrashHD | in features.conf |
| 17:27.37 | CrashHD | just to see |
| 17:27.48 | CrashHD | but I highly recommend atleast upgrading to newest 1.2 |
| 17:27.54 | CrashHD | if not 1.5 |
| 17:27.56 | CrashHD | 1.4 |
| 17:28.25 | CrashHD | secondly try terminating straight to your asterisk box |
| 17:28.33 | CrashHD | see if the phone is actually getting the dtmf to your * box |
| 17:28.42 | CrashHD | then try terminating the phone directly to your sip provider |
| 17:28.43 | variable_office | i ran 1.4 on brand new machine, and no dtmf, although i did not watch sip debug on 1.4 |
| 17:28.51 | CrashHD | see if dtmf is getting through your sip provider |
| 17:29.13 | ManxPower | I have not had DTMF issues in the last 3 years. |
| 17:29.21 | CrashHD | just narrow down the actual problem, where it is, what criteria must be met for he problem to occur |
| 17:29.22 | ManxPower | It is almost always user or design or ITSP errors. |
| 17:29.31 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:29.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:29.39 | variable_office | CrashHD what do you mean terminate straight to asterisk box? |
| 17:29.55 | CrashHD | ManxPower, I had some dtmf issues with feature digit buffering |
| 17:30.11 | variable_office | disabling features did nothing |
| 17:30.15 | Mad|Cow | hmm... it's strange though... I just cant get xlite (behind NAT) to register to Asterisk. tcpdump shows my * responce to xlite as failing (port unreachable). Its like my xlite client isnt keeping the port open on the firewall. Any ideas? |
| 17:30.47 | CrashHD | variable_office, answer a call on your asterisk and do a read() or background() to see if digits are atleast getting to your asterisk box from your phone |
| 17:30.56 | ManxPower | CrashHD: I use the native transfer feature of my phones. |
| 17:31.11 | CrashHD | earlier when you made a call with sip debug, you should had seen two dtmf events |
| 17:31.17 | variable_office | CrashHD when i use voicemailmain from the phone it works fine |
| 17:31.22 | CrashHD | ok |
| 17:31.24 | variable_office | doesnt that use dtmf? |
| 17:31.27 | ManxPower | Mad|Cow: disable ALL NAT features of X-Lite. They will cause a problem if you are using nat=yes in Asterisk |
| 17:31.28 | CrashHD | ya |
| 17:31.45 | CrashHD | variable_office: I would open up a ticket with your itsp and see if they are getting your dtmf |
| 17:31.58 | CrashHD | they hand off their signal to another sip provider usually |
| 17:32.05 | CrashHD | and sometimes that handoff messes up dtmf |
| 17:32.19 | CrashHD | (where they don't negotiate the protocol correctly) |
| 17:32.29 | Mad|Cow | ManxPower: I'm using the most recent version of xlite; there doesnt seem to be any NAT settings. Am I missing something? |
| 17:32.36 | CrashHD | *but you should be seeing dtmf events, I'm guessing you are just missing htem) |
| 17:32.51 | variable_office | i am going to print it out and ctrl f the output |
| 17:32.54 | ManxPower | Most ITSPs that use Level 3 as their carrier have problems unless you use INBAND DTMF and the ULAW or ALAW codec. It sucks. |
| 17:33.01 | ManxPower | Mad|Cow: Yes. |
| 17:33.10 | CrashHD | ya itsp with level 3...big problem |
| 17:33.16 | CrashHD | which is sad lol |
| 17:33.17 | variable_office | i tried inband, and i am using ulaw, no luck |
| 17:33.35 | ManxPower | Of course, I never send calls across the internet, so I've never had that problem. |
| 17:33.37 | CrashHD | variable are you doing canreinvite? |
| 17:33.43 | variable_office | nope |
| 17:33.53 | CrashHD | for both connections? |
| 17:34.16 | CrashHD | keep your provider in the media path |
| 17:34.18 | variable_office | i have canreinvite = no for ALL connections |
| 17:34.22 | CrashHD | hmmmmmmmmm |
| 17:34.34 | CrashHD | well upgrade to the newest 1.2 |
| 17:34.44 | CrashHD | as a desperation measure |
| 17:34.57 | Mad|Cow | ManxPower: Do you know what xlite calls NAT? |
| 17:35.01 | CrashHD | look at your sip debug messages more closely |
| 17:35.03 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:35.11 | CrashHD | nat is nat |
| 17:35.25 | CrashHD | Mad|Cow, let me look |
| 17:35.51 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 17:35.56 | ManxPower | http://www.google.com/search?hl=en&q=x-lite+nat+configure&btnG=Google+Search |
| 17:35.56 | variable_office | i put all the output into notepad did ctrl f > dtmf and it finds nothing |
| 17:36.07 | Mad|Cow | Thanks Crash - its killing me |
| 17:36.08 | ManxPower | If you want me to READ the pages for you then I expect dinner and drinks before holding your hand. |
| 17:36.24 | CrashHD | Mad|Cow: when you setup your sip account under topology there are firewall traversal settings |
| 17:36.37 | variable_office | CrashHD i could pastebin the output if you want? |
| 17:36.50 | CrashHD | I was thinking sexual favors were more appropriate for this kind of effort, ManxPower |
| 17:36.52 | CrashHD | lol |
| 17:36.52 | TripleFFFF | i get Looking for 5141231234 in default (domain 70.15.55.12) |
| 17:36.56 | TripleFFFF | is that normal ? |
| 17:37.11 | TripleFFFF | i get 404's with trixbox.. all is configed ok |
| 17:37.13 | CrashHD | variable_office, sure I can take a quick look. I need to format this comp though so have to hurry |
| 17:37.21 | CrashHD | trix be damned |
| 17:37.40 | ManxPower | Drixbox wraps the Dial in an AGI so it is impossible to debug |
| 17:38.02 | variable_office | CrashHD -> http://pastebin.ca/318661 |
| 17:38.08 | wunderkin | trix is for kids |
| 17:38.21 | TripleFFFF | yes well a client is using it |
| 17:38.27 | TripleFFFF | so i got to work with that i got lol |
| 17:38.38 | TripleFFFF | i passed the message ;) |
| 17:38.42 | ManxPower | Asking for help with Trixbox on #asterisk is like doing a total customization of your car, right down to the electrical system and then bringing it to the dealer for repairs. |
| 17:39.04 | TripleFFFF | well its yeah i get it.. but its a general asterisk issue at core.. |
| 17:39.06 | TripleFFFF | <PROTECTED> |
| 17:39.19 | TripleFFFF | meaning is it looking for that EXTEN 5141231234 or USER ? |
| 17:39.34 | ManxPower | TripleFFFF: so what is the PROBLEM with that. It is looking for the extension 5141231234 in the [default] context in extensions.conf |
| 17:39.45 | TripleFFFF | hmm |
| 17:39.53 | variable_office | CrashHD in the sip debug printout while dialing into voicemailmain and entering my passwd i had no dtmf printout |
| 17:40.05 | CrashHD | hmm |
| 17:40.05 | TripleFFFF | duh ok |
| 17:40.11 | CrashHD | maybe I should verify it is even there |
| 17:40.13 | CrashHD | brb |
| 17:40.54 | *** join/#asterisk cygar (n=cygar@201.216.200.33) |
| 17:41.01 | cygar | hello |
| 17:41.13 | variable_office | it doesnt EVER seem to print out a dtmf event? |
| 17:41.31 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 17:41.58 | ellisdee | in my auto attendant that i have setup. when a end user press 0 for the directory. i have a slight pause before the enduser goes into the directory. |
| 17:41.58 | CrashHD | hmmmmmmm |
| 17:42.01 | CrashHD | doesn't on mine either |
| 17:42.06 | CrashHD | I could have sworn it did |
| 17:42.10 | ellisdee | exten => 0,1,Goto(directory,4636,1) |
| 17:42.15 | CrashHD | my apologies then |
| 17:42.26 | ellisdee | [directory] |
| 17:42.26 | ellisdee | <PROTECTED> |
| 17:42.29 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:42.33 | TripleFFFF | wanna have fun ? |
| 17:42.52 | variable_office | CrashHD no problem, we have essentially proven that asterisk is getting dtmf though correct? |
| 17:42.56 | cygar | I have configured an E1 and it looks everything alright... I receive incoming calls without any problems but got No Circuit/channel unavailable when I try to make an outgoing call. Can anyone give me a hint what to check for ? |
| 17:43.22 | ManxPower | ellisdee: you have another extension in the same context as exten => 0 that matches 0 as the first digit. |
| 17:43.24 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 17:43.34 | skirmisha | hello everybody |
| 17:43.41 | ManxPower | cygar: paste JUST the Dial line you are using. |
| 17:43.50 | ManxPower | ~trixbox |
| 17:43.57 | jbot | hmm... trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 17:43.57 | ManxPower | ~freebpx |
| 17:43.59 | skirmisha | i've got a problem with asterisk |
| 17:44.26 | skirmisha | is there a solution of doing call divert/redirect but not forward or using follow me |
| 17:44.42 | cygar | ManxPower: -- Executing Dial("SIP/2339-08af1020", "Zap/g1/0212188684247|30|T") in new stack |
| 17:44.52 | *** join/#asterisk Asterman (n=newkinet@shell4.sea5.speakeasy.net) |
| 17:45.12 | ManxPower | cygar: In the line after the Dial do a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) |
| 17:45.50 | Asterman | Could someone suggest a simple to setup yet reliable RTP proxy? I've taken a look at a couple but so far they've seemed kinda flakey |
| 17:46.15 | meshugga | anyone an idea how i could get a "ringing" state when originating a call? |
| 17:46.25 | meshugga | in the manager api |
| 17:46.56 | TripleFFFF | add agent 1000,pass 1234; put exten => 777 ,1,AgentCallbackLogin(1000|1000@something) ; yes i know im missing the options seperator.. next dial 777 , put a wrogn pass..123... then asks for agent put 1000, pass 1234, then extensions put 1000.. INFINITE LOOP CREATED |
| 17:47.15 | cygar | ManxPower: then I get "requested transfer capability: 0x00 - speech' ... called g1/..... and then 'channel 0/1 , span 1 got hangup' . It's not up now, but it will help to give me the cause... |
| 17:47.15 | CrashHD | ya if your asterisk is reacting to dtmf it is getting it |
| 17:47.25 | CrashHD | contact your itsp |
| 17:47.26 | CrashHD | best bet |
| 17:47.51 | PupenoR | What frequency should wavs be for Asterisk ? |
| 17:47.53 | cygar | ManxPower: I just wanted to get sure cause every failed attempt they charge you a lot ! and tomorrow will make the second try. |
| 17:48.25 | variable_office | CrashHD is there another sip itsp that i could use to test this on for a buck or something? |
| 17:48.39 | ManxPower | HANGUPCAUSE will tell you the EXACT reason for the failed call. Assuming you are using PRI, of course. |
| 17:48.42 | CrashHD | not that I know of, I'm sure someone here can suggest |
| 17:49.09 | ManxPower | PupenoR: 8,000Hz, 16-bit, mono |
| 17:49.11 | CrashHD | I think sip uses hangupcause as well right ManxPower? |
| 17:49.21 | ManxPower | CrashHD: yes, but he's not using SIP |
| 17:49.37 | CrashHD | *smiles* |
| 17:49.45 | PupenoR | Thankse |
| 17:50.01 | CrashHD | just didn't scroll |
| 17:50.58 | ellisdee | ManxPower, nope. no other entries in that context that match 0 |
| 17:51.20 | ellisdee | ManxPower, just a 2sec pause before the directory wav's are played after executing 0 over the phone. |
| 17:51.21 | skirmisha | any ideas? |
| 17:51.53 | ManxPower | ellisdee: how about include =>'d contexts |
| 17:51.59 | Dr-Linux | difficult to me to assist my US client that how she can change the IP address on the cisco 7960 phone |
| 17:52.04 | Dr-Linux | she says: i only have a choice of select and cancel. It doesn't work when I type on keypad |
| 17:52.12 | ManxPower | Aslo watch the CLI to see if something happens as soon as your press 0 or if there is a delay before anything happens on the CLI |
| 17:52.20 | CrashHD | variable_office: you still here? |
| 17:52.55 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 17:53.35 | CrashHD | that should get you some dtmf events in console |
| 17:53.40 | CrashHD | err |
| 17:53.43 | CrashHD | that was meant for a notice |
| 17:53.46 | CrashHD | ignore me |
| 17:54.20 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
| 17:54.43 | skirmisha | hello |
| 17:55.52 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:56.42 | *** join/#asterisk demigod2k (n=joey@adsl-75-58-206-177.dsl.sfldmi.sbcglobal.net) |
| 17:57.00 | cygar | ManxPower: thanks. Yes, I am using PRI. Will try to see what the noop() hangup cause gives me. |
| 17:57.20 | demigod2k | on a Polycom -- is it possible to change how often you get the message waiting trill? I've seen how to get rid of it entirely. |
| 17:57.46 | demigod2k | by default it seems to be every 30 seconds which is really annoying in a cubicle farm |
| 17:57.56 | variable_office | ah crap, i missed him. what will get me some dtmf events in the console? |
| 17:58.47 | JunK-Y | ztdummy compiled fine, but when doing a modprobe ztdummy, im getting WARNING: Error inserting zaptel (/lib/modules/2.6.17-10-server/misc/zaptel.ko): Invalid module format |
| 18:01.32 | shepimport | demigod2k: I could not see a setting for it in its config file... |
| 18:01.42 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 18:02.06 | demigod2k | shepimport, ya I didnt find one in the 2.0.x firmware admin guide either. Might not exist.... |
| 18:02.27 | demigod2k | I'm surprised based on how annoying the sound is. :) |
| 18:02.45 | *** join/#asterisk flashnet (i=flashnet@166.15.185.213.dk-hvi.res.dyn.perspektivbredband.net) |
| 18:03.14 | skirmisha | anybody home |
| 18:03.21 | shepimport | You can change it by setting the tone value to a different sound |
| 18:04.12 | demigod2k | ya. I may have to change the sound to "silence" and rely on the LED only. there was a snow/ice storm today so everybody's phone is trilling every 30 seconds with nobody around to answer |
| 18:04.16 | shepimport | demigod2k: you can do it with all the tones on the polycom, just not the interval value |
| 18:06.34 | skirmisha | huh |
| 18:06.43 | skirmisha | is there anyone familar with asterisk |
| 18:06.51 | demigod2k | skirmisha, somewhat. just ask your question |
| 18:07.11 | variable_office | when i do dtmf sometimes it prints out something along the lines of comfort noise generation not supported? |
| 18:08.18 | *** join/#asterisk AlfaScorpii (n=alfascor@host218.201-252-175.telecom.net.ar) |
| 18:08.28 | AlfaScorpii | Hello all |
| 18:08.45 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 18:09.57 | *** join/#asterisk hardwire (n=hardwire@rdbck-2624.wasilla.mtaonline.net) |
| 18:10.04 | hardwire | doobiedoobiedooooo |
| 18:10.08 | hardwire | hey there little fellas |
| 18:10.40 | skirmisha | demigod2k i asked already |
| 18:10.46 | skirmisha | is there a solution of doing call divert/redirect but not forward or using follow me |
| 18:11.17 | skirmisha | means asterisk will not dial but will return new number to user |
| 18:12.32 | skirmisha | the problem is that i bill customers on caller id, and if they set follow me to their mobile's number, no one can reach them |
| 18:12.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:12.59 | skirmisha | as user caller id is diff |
| 18:13.51 | *** join/#asterisk reber (i=reber@gateway/tor/x-2116c1965f54ce94) |
| 18:13.58 | *** join/#asterisk brain- (n=brain@2001:618:400:0:0:0:c8b5:aadb) |
| 18:15.07 | adde | Ive been searching alkl about ... but im not getting further.... I need to install my USB Dongle for Bluetooth... Please can someone help me out.... |
| 18:15.38 | CrashHD | adde, this isn't a hardware support or linux support channel...highly recommend searching google |
| 18:15.52 | skirmisha | so any ideas |
| 18:15.58 | *** part/#asterisk AlfaScorpii (n=alfascor@host218.201-252-175.telecom.net.ar) |
| 18:16.32 | Dr-Linux | my cisco 7960 is requesting still for SEP<mac>.cnf , it's not requestiong for SIP<mac> and relevant files, what could be the issue? :S |
| 18:16.38 | *** join/#asterisk acehunky (n=chat_jok@61.17.18.28) |
| 18:17.08 | *** part/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 18:17.08 | acehunky | does anyone know what is the right room to talk on ss7 with asterisk ? |
| 18:17.11 | Rhizome | what a strange question to be asked here :p |
| 18:17.27 | adde | Obviosly i have already tried google. ANd this is an Asterisk channel and most likely a few users are using bluetooth as a trunk and therefor most likely are sitting with knowledge that might help me... And beeing a channel sharing information that is relevant. If you do not know an answer just sit quite. You dont always have to me a smartass... |
| 18:17.52 | [TK]D-Fender | skirmisha: If you're using a PRI you can set your outbound callerid when calling their cell so it looks like its coming direct from the customer... |
| 18:17.58 | acehunky | i have installed chan_ss7 and i keep on seeing this message on CLI " mtp2_read_su: MTP2 CRC error " |
| 18:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:18.02 | acehunky | can anyone help me with that ? |
| 18:18.11 | ManxPower | acehunky: I don't think there is any, since asterisk does not come with SS7 support |
| 18:18.21 | Rhizome | adde: lol |
| 18:18.35 | skirmisha | [TK]D-Fender i am using sip for outbound |
| 18:19.03 | Rhizome | adde: Still doesn't change the fact that your asking in the wrong channel. |
| 18:19.17 | ManxPower | Dr-Linux: SEP, I believe means the phone is runnning SCCP, not SIP. |
| 18:19.17 | [TK]D-Fender | skirmisha: Then its up to your ITSP as to whether they support setting CID or not. |
| 18:19.27 | adde | How do get Bluetooth set up in/with Asterisk.. |
| 18:19.33 | adde | Thats wrong chan? |
| 18:19.38 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:19.56 | Rhizome | adde: Ask a more specific question then. |
| 18:20.00 | Dr-Linux | ManxPower: yes, you are correct, but how can i install SIPfirmware on it |
| 18:20.11 | BSDTech | ok I have a issue |
| 18:20.18 | ManxPower | adde: Asterisk does not come with Bluetooth support, perhaps you can ask the person that wrote the Bluetooth stack you are using. |
| 18:20.21 | Dr-Linux | ManxPower: Cisco phone often comes with SCCP, but i always install SIP firmware on it |
| 18:20.27 | ManxPower | Dr-Linux: you have to purchase the SIP firmware |
| 18:20.38 | EmleyMoor | If I pass a call to VoiceMail with two mailboxes specified, which unavailable/busy etc. messages does it use? |
| 18:20.41 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 18:20.42 | acehunky | ManxPower yeah thats right .. but i see that there are a few trunks for ss7 and mailing list too |
| 18:20.57 | BSDTech | I have 1+NXXNXXXXXX on out bound trunks for matching but it all works fine for 5 calls on the 6th it fails and this is on zap channels |
| 18:21.06 | Dr-Linux | ManxPower: :) i've perchased a login, and i can download everything |
| 18:21.08 | ManxPower | acehunky: I strongly doubt anyone that comes here is running SS7 |
| 18:21.10 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-5c20388581bcb57d) |
| 18:21.17 | skirmisha | [TK]D-Fender currently asterisk is setting correct caller id for outbound calls |
| 18:21.37 | skirmisha | but the call is not auth at remote side and it fails |
| 18:21.40 | ManxPower | EmleyMoor: I think it's the first maibox specified |
| 18:21.59 | [TK]D-Fender | skirmisha: Then your provider is most likely not allowing you to set it. |
| 18:22.07 | Dr-Linux | ManxPower: it's not my first phone, i'm already using a couple of cisco 7960's |
| 18:22.10 | ManxPower | EmleyMoor: I use that feature, but play my own custom greeting and turn off the mailbox greeting |
| 18:22.25 | skirmisha | [TK]D-Fender let me give u example |
| 18:22.33 | *** join/#asterisk flashnet (i=flashnet@166.15.185.213.dk-hvi.res.dyn.perspektivbredband.net) |
| 18:22.58 | skirmisha | lets say mobile number(123456) is calling user within asterisk (2222) |
| 18:23.15 | skirmisha | user 2222 has set follow me to his mobile number 654321 |
| 18:23.21 | *** join/#asterisk jake[work] (n=Administ@74.92.120.54) |
| 18:23.47 | skirmisha | what asterisk do is to follow this call to sip trunk out with caller id 123456 and dialed number 654321 |
| 18:24.00 | ManxPower | skirmisha: STOP! Which of the 5 or so followme methods are you using? |
| 18:24.01 | skirmisha | but my provider bills calls based on called id |
| 18:24.07 | skirmisha | and user id set is 2222 |
| 18:24.12 | skirmisha | and thus above call fails |
| 18:24.27 | ManxPower | skirmisha: I don't see why it would fail. |
| 18:24.52 | skirmisha | because asterisk set caller id to be the caller id of calling user (123456) |
| 18:24.59 | ManxPower | Since the provider bills on called id, and the called id is 654321 |
| 18:25.02 | skirmisha | and my provider does not auth that call |
| 18:25.17 | acehunky | ManxPower Do you know of any room where ppl talk bout it ? |
| 18:25.20 | ManxPower | skirmisha: set the callerid in the sip.conf section for that provider |
| 18:25.41 | ellisdee | or set it in macro-dialout context in your extensions.conf |
| 18:25.42 | ManxPower | acehunky: no. So few people use SS7 with Asterisk there is practically no community supporting it. |
| 18:26.06 | skirmisha | i want to do it per user basis |
| 18:26.23 | jake[work] | i have several phones behind 1 router (NAT) that work fine and show different port numbers for each phone. another group of phones are behind a different router and they all show as port 5060 when doing a sip show peers. is this a router setting? what might be the setting to look for? |
| 18:26.28 | ellisdee | then go with the ManxPower method, and set the appropriate caller id nfo for each peer |
| 18:26.41 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:26.42 | ManxPower | skirmisha: Then you have to do it within the dialplan using a complex set of lines and logic. |
| 18:26.55 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:26.56 | Katty | mew. |
| 18:26.57 | ManxPower | why not just use the fromuser= and similar options in sip.conf |
| 18:27.05 | wunderkin | jake[work], if it is doing nat, crappy router, update the firmware |
| 18:27.10 | skirmisha | ManxPower provider auth calls on caller id |
| 18:27.28 | ManxPower | ellisdee: he could also set up TWO entries for the the provider, 1 with override and one without. |
| 18:27.46 | Katty | it's snowing out today. |
| 18:27.53 | Katty | and i'm craving ice cream... what next )= |
| 18:28.08 | BSDTech | BSDTech> it seems zaptel is haveing issues 5 calls work fine but on a 6th call it fails to pass the dialing codes |
| 18:28.08 | BSDTech | <BSDTech> and only happeneds afeter updating from 1.2.12 to 1.2.14 |
| 18:28.08 | BSDTech | <BSDTech> and to the current branch of 1.4 zaptel and libpri |
| 18:28.10 | *** join/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
| 18:28.17 | ManxPower | skirmisha: I doubt the provider auths on callerid, as it would be trivial for people to spoof that. I suspect it actually auths on from id (which defaults to being the same as callerid unles you override it with fromuser=_ |
| 18:28.20 | skirmisha | if i set a caller id to all outgoing calls, user will make free calls |
| 18:28.25 | BSDTech | any input on this |
| 18:28.38 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-c4a23dcdbcccdc65) |
| 18:28.41 | jake[work] | wunderkin: tnx - i'll try that |
| 18:28.46 | ManxPower | BSDTech: I think the users is taking some very good drugs |
| 18:28.58 | Katty | mmm, rocky road. |
| 18:29.08 | BSDTech | no I have tested it |
| 18:29.11 | ManxPower | BSDTech: does it work with the latest 1.2.x zaptel? |
| 18:29.23 | Qwell[] | ManxPower: Did that fix work? |
| 18:29.31 | ManxPower | Qwell[]: which fix? |
| 18:29.34 | Qwell[] | 8822? |
| 18:29.40 | Qwell[] | getgroupcount or whatever |
| 18:29.47 | ManxPower | Oh! No by the time all users went home, I was drunk. |
| 18:29.50 | Qwell[] | heh |
| 18:29.51 | Dr-Linux | anybody is using Cisco 7960 phone? |
| 18:29.54 | Qwell[] | fair enough |
| 18:29.55 | jake[work] | yes |
| 18:30.12 | jake[work] | Dr-Linux: i'm using it |
| 18:30.14 | ManxPower | Qwell[]: I don't really have any interest in fixing the problem and using a custom patch set. I'll wait for 1.2.15 to come out. |
| 18:30.14 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3a7dc65eefff1071) |
| 18:30.14 | BSDTech | I hooked up a polycom to thier system and it dials fine for 5 calls on the 6th you get your call could notbe completed . you let the systems set for a few min and it dials out fine again |
| 18:30.31 | *** join/#asterisk bradoaks (n=bradoaks@dog.lost-habit.com) |
| 18:30.34 | ManxPower | BSDTech: PRI? |
| 18:30.41 | BSDTech | to me this says that the zaptel is not reseting properly |
| 18:30.47 | BSDTech | tdm |
| 18:31.01 | ManxPower | maybe you just need a "w" in the dial plan |
| 18:31.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-64-105.red.bezeqint.net) |
| 18:31.07 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:31.07 | ManxPower | as asterisk is starting to dial too fast. |
| 18:31.10 | ManxPower | Classic problem |
| 18:31.11 | *** join/#asterisk foRza (n=forza@firewall.hikt.no) |
| 18:31.14 | BSDTech | ok |
| 18:31.18 | BSDTech | hmmmm |
| 18:31.27 | Qwell[] | Dial(Zap/1/w5551212) |
| 18:31.47 | BSDTech | ok I will test that |
| 18:31.54 | skirmisha | ManxPower your solution won't work |
| 18:32.22 | ManxPower | skirmisha: it worked for every provider I've used. |
| 18:32.37 | Qwell[] | it's probably only an issue on analog lines |
| 18:32.57 | skirmisha | as follow me just dial the number listed with caller id -> caller id of user dialing that number |
| 18:34.14 | ManxPower | skirmisha: what is fromuser= set to in sip.conf? |
| 18:34.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-64-105.red.bezeqint.net) |
| 18:35.01 | skirmisha | ManxPower i have many users in that asterisk and can not set in sip.conf fromuser because it will overwrite all others callers id's |
| 18:35.14 | ManxPower | skirmisha: NO IT WILL NOT |
| 18:35.32 | ManxPower | Set fromuser= in the [section] for your provider |
| 18:35.46 | ManxPower | skirmisha: do you REALLY think you are the only one in the world that has had this problem? |
| 18:36.19 | ManxPower | fromuser overides the SIP user id, not the SIP callerid |
| 18:36.45 | ManxPower | now, your provider may be stupid and actually use the SIP callerid info, but that would be a massive security hole. |
| 18:37.02 | ManxPower | In which case there is no solution to your problem. |
| 18:37.09 | demigod2k | BSDTech, I had the same problem you described with polycoms "could not be completed" on analog lines, picking up and dialing too quickly |
| 18:37.27 | BSDTech | hmmm |
| 18:37.56 | BSDTech | well Imake 5 calls fine |
| 18:37.56 | demigod2k | BSDTech, I dont know if you're on pots or pri. if you've got a pots interface try changing the delay in your zaptel |
| 18:37.56 | BSDTech | then the 6th 7th fail |
| 18:37.56 | skirmisha | ManxPower let me see |
| 18:37.56 | demigod2k | BSDTech, if you're on PRI I dont think the delay applies or makes a difference |
| 18:37.56 | BSDTech | and then I wait 3 min and dial fine |
| 18:38.06 | BSDTech | this is a tdm card |
| 18:38.14 | demigod2k | BSDTech, tdm with plain analog lines right? |
| 18:38.38 | wunderkin | are the phone lines being released |
| 18:39.04 | ManxPower | BSDTech: Classhc symptom of a slightly overloaded telco switch. |
| 18:39.05 | foRza | I'm having trouble with an IAX trunk. Status is "Unreachable" on one server, but is works fine on the other one though... |
| 18:39.14 | BSDTech | well I get your call cannot be completed from the phone comany |
| 18:39.17 | bkruse | is it getting to hot? |
| 18:39.23 | *** part/#asterisk hassler (n=hassler@r-corp.hcst.com) |
| 18:39.28 | ManxPower | BSDTech: Correct. the telco is missing the first digit. |
| 18:39.28 | BSDTech | so Ithink it tries to dial with out dial tone |
| 18:39.41 | ManxPower | BSDTech: Asterisk does NOT check for dialtone before dialing. |
| 18:39.48 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:39.55 | BSDTech | that sucks |
| 18:39.57 | BSDTech | lol |
| 18:40.15 | demigod2k | ya. in my situation adding a 0.5s delay solved everything |
| 18:40.37 | wunderkin | BSDTech, are the phone lines being released |
| 18:40.47 | [TK]D-Fender | ManxPower: like a wise man once said ""Listen up maggots. You are not special. You are not a beautiful or unique snowflake. You are the same decaying organic matter as everything else."" |
| 18:41.13 | ManxPower | [TK]D-Fender: he was a wise man |
| 18:41.26 | *** join/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 18:41.53 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 18:44.00 | mercestes | That makes me want to cry. |
| 18:44.42 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 18:45.12 | ManxPower | Granted he was prolly a wise son of satan, but wise none the less. |
| 18:45.26 | [TK]D-Fender | ManxPower: Tyler Durden :D |
| 18:45.47 | [TK]D-Fender | ManxPower: The all-singing, all-dancing crap of the world :) |
| 18:46.28 | [TK]D-Fender | Chuck Palahniuk rocks.... |
| 18:46.47 | [TK]D-Fender | mercestes: You haven't seen it?!?!?! z0mg, no wonder your world view is so easily shattered! |
| 18:47.35 | *** part/#asterisk foRza (n=forza@firewall.hikt.no) |
| 18:48.35 | mercestes | ?? |
| 18:48.51 | mercestes | my world view is so not easily shattered..:P |
| 18:48.55 | [TK]D-Fender | mercestes: "Fight Club" |
| 18:49.01 | mercestes | I saw. |
| 18:49.13 | [TK]D-Fender | mercestes: Watch it again! |
| 18:49.20 | mercestes | ...yessir. |
| 18:50.14 | [TK]D-Fender | knot* |
| 18:50.16 | [TK]D-Fender | :O |
| 18:50.29 | [TK]D-Fender | Remins me of Kurt Kobain.... |
| 18:50.47 | [TK]D-Fender | Do know what colour his eyes were when he died |
| 18:50.49 | [TK]D-Fender | ? |
| 18:51.05 | mercestes | [TK]D-Fender: You would be deadly if you could type. and red. |
| 18:51.26 | [TK]D-Fender | ? |
| 18:51.36 | mercestes | * [TK]D-Fender tightens the know and prepares the trap-door |
| 18:52.06 | [TK]D-Fender | my office is too damned cold. muscles just not responding right :| |
| 18:52.21 | perd | my quarter of a cubicle is cold |
| 18:52.25 | perd | so cold |
| 18:55.08 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:57.30 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 18:58.55 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 19:01.18 | RichL | Newbie..., Can someone help me solve a call routing issue? |
| 19:02.11 | RichL | i have voicepulse and tdm400p to route for |
| 19:02.28 | Dr-Linux | please tell me someone **# locks the entire phones button, or only configuration? |
| 19:02.49 | RichL | the voicepulse it working fine, but I have issues with the tdm400p not rolling over to the next available line |
| 19:03.15 | EmleyMoor | It even works with a rotary phone again now |
| 19:03.37 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.43) |
| 19:04.49 | ManxPower | RichL: paste JUST the Dial(... line to the channel. If you are using FreePBX you must ask on #freepbx |
| 19:05.51 | RichL | oh ok |
| 19:07.25 | fetcher | does Argentina use T1 circuits or E1? Anyone happen to know? |
| 19:07.45 | *** join/#asterisk xxxAGENTExxx (n=x@201.65.48.130) |
| 19:07.49 | ManxPower | fetcher: it should |
| 19:07.55 | ManxPower | E1 that is |
| 19:08.03 | xxxAGENTExxx | whats message problem?? Jan 16 13:19:30 NOTICE[840]: rtp.c:577 ast_rtp_read: Unknown RTP codec 96 received ??? |
| 19:08.13 | xxxAGENTExxx | whats problem this message |
| 19:08.26 | ManxPower | xxxAGENTExxx: It means that the device tried to use a codec Asterisk does not know about |
| 19:08.43 | xxxAGENTExxx | mmmmm |
| 19:09.01 | terrapen | has anybody ever tried a distributed call queue? |
| 19:09.10 | terrapen | like, two servers handling two pools of operators |
| 19:09.22 | terrapen | and sending callers to the secondary server when the first queue is full? |
| 19:10.11 | ManxPower | xxxAGENTExxx: specifically the device is sending DTMF using codec number 96, rather than what Asterisk is expecting. |
| 19:10.11 | terrapen | i'm going to be needing to run a queue for hundreds of callers and I'm afraid that one single machine cannot handle the load |
| 19:10.11 | xxxAGENTExxx | ManxPower how solucion? |
| 19:10.11 | ManxPower | But you knew that already from your google searches, right? |
| 19:10.11 | xxxAGENTExxx | solution this problem |
| 19:10.24 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 19:10.34 | ManxPower | The solution is to fix the device or the config on the device. There is nothing you can do in your |
| 19:10.39 | ManxPower | Asterisk config to fix the problem |
| 19:10.55 | xxxAGENTExxx | ok |
| 19:10.57 | xxxAGENTExxx | thanks |
| 19:11.02 | ManxPower | unless you want to patch asterisk and recompile |
| 19:11.12 | ManxPower | but the links in your google search will show you how/where to do that |
| 19:12.32 | terrapen | I'm thinking that I could use the timeout= parameter of Queue() to send people over to the secondary server |
| 19:12.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.44 | terrapen | what I really want is a way to balance people between the two |
| 19:12.48 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 19:16.03 | *** join/#asterisk jimmy_deanPB (n=jhodapp@209.131.196.174) |
| 19:17.11 | [TK]D-Fender | terrapen: All an AGI that will check the queue size of both and transfer to the best choice. |
| 19:17.18 | *** join/#asterisk Mad|Cow (n=thirt@c-69-242-72-104.hsd1.de.comcast.net) |
| 19:17.30 | [TK]D-Fender | s/All/Call/ |
| 19:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:23.56 | *** join/#asterisk lsl23_ (n=chatzill@141.214.234.28) |
| 19:24.07 | lsl23_ | What is the URL for the att site that lets you choose an 800 number |
| 19:25.11 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:25.25 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 19:25.36 | mtaht4 | join #ardour |
| 19:25.45 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:26.38 | wunderkin | DDoS mthah4 & #ardour :P |
| 19:27.42 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 19:33.42 | *** join/#asterisk Strom_M (n=strom@dsl081-035-115.lax1.dsl.speakeasy.net) |
| 19:34.18 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 19:35.35 | ellisdee | anyone care to recommend a iax softphone for windows? |
| 19:37.01 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:40.29 | [TK]D-Fender | ellisdee: Idefisk |
| 19:42.55 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 19:44.48 | Mad|Cow | Is there anyone that could do a little handholding with me? ;-) |
| 19:45.57 | [TK]D-Fender | Mad|Cow: pastebin your sip.conf so we can see if you did what was already advised to you. |
| 19:45.58 | [TK]D-Fender | ~pb |
| 19:46.06 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:47.52 | ellisdee | thanks [TK]D-Fender ..great app |
| 19:49.02 | Mad|Cow | TK: pb doesnt look like it supports this channel? |
| 19:49.03 | [TK]D-Fender | ellisdee: Gets the job done. Icky interface, but better than some. Suports native transfer so its not "crap". |
| 19:49.15 | [TK]D-Fender | mad? |
| 19:49.40 | [TK]D-Fender | Mad|Cow: Pastebin.ca is working jsut fine for me.... |
| 19:50.09 | Mad|Cow | TK: Ahh... didnt see the other address... one minute |
| 19:50.33 | [TK]D-Fender | Mad|Cow: BOTH seem to work just fine... |
| 19:51.49 | mercestes | Mad|Cow: Umm... If you can't use pastebin I don't think we can help you. Have you considered hiring a consultant maybe? |
| 19:53.00 | CunningPike | For those of you in the Pacific Northwest, our office is hosting a Digium Asterisk Bootcamp next month: http://www.digium.com/en/training/locator/enroll/46 |
| 19:53.17 | *** part/#asterisk RichL (n=RichL@68.143.17.4.nw.nuvox.net) |
| 19:54.25 | *** join/#asterisk jjhall (n=jjhall@mail.dbsupply.com) |
| 19:57.06 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 19:57.17 | Mad|Cow | mercestes: ouch... :-) OK... I got it. My number is 5107 |
| 19:57.18 | jjhall | I'm having some issues connecting to Broadvoice. It has worked in the past, then I temporarily suspended my account. After re-enabling it, it will not register. They are telling me my system is trying to register multiple times per second and is being IP blocked. Nothing has changed on my end, are they full of it or is there something I can check from my end? |
| 19:57.45 | Mad|Cow | TK: Sorry... I thought it was supposed to go to the channel... I didnt realize it would just post on a site |
| 19:58.05 | mercestes | 5107 of which PB service? |
| 19:58.21 | Mad|Cow | http://channels.debian.net/paste/5107 |
| 19:58.29 | mercestes | :) Much better. |
| 19:58.35 | Mad|Cow | sorry.. new to this ;-) |
| 19:59.06 | Mad|Cow | I have tried several differnt configs.... none seem to work |
| 19:59.30 | mercestes | you have externip and localnet commented out? |
| 19:59.32 | Mad|Cow | If I put the phone in front of the firewall, it works. It only fails behind the firewall (NAT) |
| 20:00.03 | Mad|Cow | I tried both ways (with commented out and uncommented out) |
| 20:00.09 | Mad|Cow | neither seem to work |
| 20:01.17 | Mad|Cow | 69.242.72.104 <-- Is the external IP address of the firewall that my PC is behind (running the soft phone) |
| 20:01.46 | Mad|Cow | 74.92.109.0/255.255.255.0 <-- Is the network my asterisk server is on |
| 20:01.50 | Mad|Cow | Am I missing something? |
| 20:02.18 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 20:02.32 | mercestes | don't think so |
| 20:05.06 | *** join/#asterisk mkl1525 (n=qwertz@82.193.233.238) |
| 20:05.53 | *** join/#asterisk UlbabraB (n=salama@81.72.43.241) |
| 20:06.53 | *** join/#asterisk ping2921 (n=marc3234@206-248-153-49.dsl.teksavvy.com) |
| 20:08.51 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:10.32 | terrapen | d-fender, that's an awesome idea |
| 20:10.38 | terrapen | the AGI that checks the size of the queue |
| 20:11.18 | Mad|Cow | My output from tcpdump: 15:10:43.937632 IP voip.foxtailsolutions.com > c-69-242-72-104.hsd1.de.comcast.net: ICMP voip.foxtailsolutions.com udp port 3478 unreachable |
| 20:12.26 | Mad|Cow | TK/mercestes: Any ideas? |
| 20:13.37 | terrapen | the trick would be to be able to move people to the other queue if the queue they are currently in does not empty as fast as the queue on the other server |
| 20:16.07 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 20:16.08 | *** join/#asterisk mrg82 (n=na@dsl82-163-126-23.as15444.net) |
| 20:16.28 | ellisdee | Mad|Cow, which protocol are you using |
| 20:16.38 | Mad|Cow | ellisdee: SIP |
| 20:17.18 | ellisdee | you have port 5060 open? |
| 20:17.22 | ellisdee | this is a softphone? |
| 20:17.34 | Mad|Cow | 5060 is open, and this is a softphone |
| 20:17.45 | ellisdee | tcp/udp |
| 20:17.47 | Mad|Cow | FYI - If i put the softphone in front of my firewall... it works fine |
| 20:17.58 | Mad|Cow | tcp and udp are both open |
| 20:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.01 | ellisdee | what port does the softphone initiate communications with the pbx? |
| 20:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.18 | ellisdee | when you have the soft phone outside of the firewall/nat |
| 20:18.45 | ellisdee | is the port in the range of 10-20k? |
| 20:18.46 | *** join/#asterisk fjean5 (n=fjean5@207.107.208.137) |
| 20:18.51 | fjean5 | hi guys |
| 20:18.51 | mrg82 | I'm editing my extensions.conf and i want the wait() time to be at random between 10 and 20 seconds |
| 20:18.59 | Mad|Cow | it appears to come from 3478 |
| 20:20.02 | mrg82 | can i use math functions in the extensions.conf? |
| 20:20.02 | [TK]D-Fender | terrapen: What's an excellent idea? |
| 20:20.02 | Mad|Cow | ellisdee: no, it looks like 3478 (from my tcpdump) |
| 20:20.03 | [TK]D-Fender | Mad|Cow: You did NOT set qualify=yes as you were instructed to... |
| 20:20.03 | ellisdee | Mad|Cow, listen to [TK]D-Fender |
| 20:20.04 | Mad|Cow | TK: I have tried that... no luck |
| 20:20.09 | ellisdee | host=dynamic |
| 20:20.10 | ellisdee | nat=yes |
| 20:20.13 | ellisdee | qualify=yes |
| 20:20.34 | ellisdee | altho qualify=yes with multiple remote phones can be a bit of traffic on the wan interface =P |
| 20:20.41 | ellisdee | i learned the hard way. |
| 20:20.44 | ellisdee | 100+ remote phones =P |
| 20:20.53 | [TK]D-Fender | ellisdee: TFB |
| 20:21.25 | fjean5 | guys, how can i redirect all CLI messages to a file ? |
| 20:21.56 | ellisdee | thats done for you by default |
| 20:22.03 | ellisdee | /var/log/asterisk/messages |
| 20:22.04 | ellisdee | i believe |
| 20:22.26 | ellisdee | /var/log/asterisk |
| 20:23.27 | fjean5 | not really, I have var/log/asterisk but no messages file |
| 20:24.23 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 20:24.54 | ellisdee | check /etc/asterisk/logger.conf |
| 20:24.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:25.01 | fjean5 | ok, right |
| 20:25.03 | fjean5 | thanks |
| 20:27.29 | ellisdee | trixbox is nice |
| 20:27.38 | ellisdee | never used it before. just installed it. |
| 20:27.47 | ellisdee | that op panel sure is nice =P |
| 20:27.54 | CunningPike | ~lart ellisdee |
| 20:27.57 | CunningPike | ;) |
| 20:27.59 | ellisdee | lol |
| 20:28.14 | ellisdee | dial plan is really messy tho |
| 20:28.14 | ellisdee | man |
| 20:28.25 | CunningPike | ellisdee: Hence, /topic |
| 20:28.25 | tzanger | woo |
| 20:28.31 | tzanger | get to set up a branch office in Calgary |
| 20:28.37 | dlynes_laptop | ellisdee, CunningPike just likes spanking people :) |
| 20:28.52 | ellisdee | lol |
| 20:29.11 | dlynes_laptop | tzanger, ooh...don't you live in Calgary? |
| 20:29.12 | tzanger | thinking small form factor PC with Sangoma S518 for DSL and Polycom 501s |
| 20:29.23 | tzanger | port a # to Unlimitel and rock |
| 20:29.26 | CunningPike | tzanger: Bring a sweater - -20C today |
| 20:29.27 | tzanger | dlynes_laptop: no, Ontario |
| 20:29.32 | dlynes_laptop | ah |
| 20:29.38 | ellisdee | tzanger, sounds like a piece of cake =) |
| 20:29.40 | ellisdee | easy money |
| 20:29.44 | tzanger | yeah it's -9 here, -16 with windchill |
| 20:29.49 | tzanger | ellisdee: unfortunately I don't get paid :-) |
| 20:29.51 | dlynes_laptop | In order to get the S518 to work though, don't you have to get the settings from Telus? |
| 20:29.52 | Strom_M | here's a really stupid question - how do you do an iax2 debug in 1.4? |
| 20:29.56 | ellisdee | tzanger, doh! |
| 20:30.10 | tzanger | dlynes_laptop: yes, but that's a piece of cake, I have a dozen S518s working with Linux and Bell/Ikano/etc |
| 20:30.15 | tzanger | Telus PPPoE's no different |
| 20:30.21 | ellisdee | iax2 debug? |
| 20:30.35 | tzanger | iax2 set debug? |
| 20:30.44 | dlynes_laptop | tzanger, ah...any idea where you get the info to set them up with? Or do you just copy the settings off an already configured telus modem? |
| 20:30.46 | ellisdee | you debugging a iax channel? |
| 20:30.50 | `Sean | ANNOUNCEMENT: |
| 20:30.57 | CunningPike | What's the concensus on hardware for an "install and forget" asterisk installation? By that I mean something that gets installed and sits unnoticed in the corner for years |
| 20:30.57 | `Sean | i got my Cisco 7970 today |
| 20:30.57 | `Sean | :D:D:D: |
| 20:31.01 | CunningPike | Bluefin? |
| 20:31.04 | tzanger | dlynes_laptop: you ask them. Bell writes it down in the little book they give you |
| 20:31.05 | CunningPike | Or a PC? |
| 20:31.12 | `Sean | CunningPike CISCO 7970 |
| 20:31.14 | `Sean | its a phone |
| 20:31.14 | `Sean | :D |
| 20:31.15 | tzanger | CunningPike: anything without fans or hard drives |
| 20:31.16 | `Sean | ip phone from cisco |
| 20:31.20 | dlynes_laptop | tzanger, I've asked telus until I'm blue in the face |
| 20:31.26 | dlynes_laptop | tzanger, they don't know shit |
| 20:31.26 | tzanger | dlynes_laptop: really |
| 20:31.30 | CunningPike | tzanger: OK - that's what I was thinking |
| 20:31.33 | ellisdee | CunningPike, i use only intel hardware. |
| 20:31.44 | CunningPike | Anyone used any of the Xorcom boxes? |
| 20:31.45 | tzanger | dlynes_laptop: you can't just say "what is my DSL login information?" |
| 20:31.51 | ellisdee | CunningPike, disable everything except for video, keyboard, and mouse ports. |
| 20:31.57 | dlynes_laptop | tzanger, I'm not talking about the login info |
| 20:32.03 | ellisdee | CunningPike, supermicro motherboards. digium tdm cards. |
| 20:32.05 | dlynes_laptop | tzanger, I'm talking about the DSL setup information |
| 20:32.06 | ellisdee | CunningPike, crucial memory |
| 20:32.23 | dlynes_laptop | tzanger, Or does the S518 autodetect all that? |
| 20:32.24 | ellisdee | CunningPike, motherboard *has* to have two eth interfaces for my setups |
| 20:32.46 | tzanger | dlynes_laptop: huh? S518 autodetects VPI/VCI, just as any modern DSL modem does |
| 20:33.02 | tzanger | all you need is user@realm and password |
| 20:33.07 | dlynes_laptop | tzanger, hrm...ok...the modem I tried a while back didn't autodetect anything |
| 20:33.10 | CunningPike | ellisdee: Agreed - ours do. One for admin and one for the VOIP VLAN |
| 20:33.19 | ellisdee | tzanger, why dont you just bridge that pos into a linksys wrt ? =P |
| 20:33.28 | tzanger | feed that info into /etc/ppp/pap-secrets and run adsl-start after setting up pppoe.conf to point to w1ad |
| 20:33.38 | tzanger | ellisdee: because I don't like external hardware for this stuff |
| 20:33.42 | tzanger | I want plug in and leave it alone forever |
| 20:33.53 | ellisdee | will allow for remote phones as well in the event that the customer wants to expand. |
| 20:34.03 | ellisdee | vpn on a wrt |
| 20:34.14 | tzanger | besides, if the * box is also handling the firewalling/NAT then your * box is not behind NAT, and also you can tune it out for QoS |
| 20:34.21 | ellisdee | or some see it as another point of failure =\ |
| 20:34.29 | tzanger | ellisdee: how does a linux box doing the same thing not achieve that? |
| 20:34.32 | Hmmhesays | how hard would it to make app_directory return the users name in a variable? |
| 20:34.34 | tzanger | without another piece of hardware |
| 20:34.42 | dlynes_laptop | tzanger, btw...they also tie their dhcp to specific mac addresses |
| 20:34.42 | CunningPike | OK - next question - what are the 'good' solid state platforms? Stealth? |
| 20:34.58 | tzanger | dlynes_laptop: uh, PPPoE != DHCP |
| 20:35.21 | dlynes_laptop | tzanger, ic |
| 20:35.30 | ellisdee | tzanger, this is a pci dsl ? |
| 20:35.33 | tzanger | ellisdee: yes |
| 20:35.41 | dlynes_laptop | thought pppoe still used dhcp under the hood |
| 20:35.47 | tzanger | I wish it had a Zaptel-friendly FXO port on it too but you can't get everything |
| 20:35.48 | dlynes_laptop | much like ppp did on dialup |
| 20:35.48 | ellisdee | tzanger, straight sip? |
| 20:35.56 | tzanger | ellisdee: ? |
| 20:36.05 | tzanger | asterisk would be using SIP yes, as would the ip501s |
| 20:36.23 | ellisdee | tzanger, no digium card, etc? |
| 20:36.36 | ellisdee | tzanger, i have had problems with irq's, bus |
| 20:36.37 | ellisdee | s |
| 20:36.52 | ellisdee | interference from dsl modems with tdm cards. |
| 20:36.57 | ellisdee | too many headaches. |
| 20:36.58 | dlynes_laptop | ellisdee, that's the reason I switched to sangoma almost a year ago |
| 20:37.04 | dlynes_laptop | ellisdee, haven't had any problems since |
| 20:37.35 | ellisdee | pbx should only be a pbx =P |
| 20:37.41 | [TK]D-Fender | tzanger: 5$ for an RJ11 splitter & filter. |
| 20:38.14 | [TK]D-Fender | ellisdee: Its a Sangoma card. It doesn't care about what you install it in (within reason) |
| 20:38.23 | dlynes_laptop | [TK]D-Fender, yeah...those $5 splitters aren't worth the plastic they're made out of |
| 20:38.43 | tzanger | ellisdee: I've had no issues |
| 20:38.44 | [TK]D-Fender | dlynes_laptop: I know.... thats where the 2$ prfit comes from :) |
| 20:38.53 | dlynes_laptop | [TK]D-Fender, the wilcom ones are the only ones that are useful |
| 20:38.55 | tzanger | [TK]D-Fender: don't need the filter, the S518 has it inbuilt |
| 20:39.10 | tzanger | [TK]D-Fender: but what I'm saying is that id' have been nice to have the S518 with an FXO port on it internal like |
| 20:39.19 | `Sean | wow |
| 20:39.19 | tzanger | so you have ONE card that gets you both internet and PSTN connectivity |
| 20:39.20 | [TK]D-Fender | tzanger: I meanin terms of splitting off so you can go into Zaptel card. |
| 20:39.23 | *** join/#asterisk brain- (n=brain@2001:618:400:0:0:0:c8b5:aadb) |
| 20:39.29 | tzanger | [TK]D-Fender: yeah bu tnow I need two PCI slots :-) |
| 20:39.46 | [TK]D-Fender | tzanger: Oh for USE as an interface, not jsut a dumb filtered port... thats ANOTHER matter ;) |
| 20:40.16 | tzanger | yes, mostly for 911 |
| 20:40.33 | dlynes_laptop | [TK]D-Fender, well, why pay $60 for a wilcom, plus another $100 for an a200? |
| 20:40.50 | [TK]D-Fender | dlynes_laptop: "wilcom" is what? |
| 20:41.06 | dlynes_laptop | [TK]D-Fender, a highgrade dsl filter...the same ones that telus, bell, et al all use |
| 20:41.17 | [TK]D-Fender | dlynes_laptop: You mean MODEM, no? |
| 20:41.39 | dlynes_laptop | [TK]D-Fender, no...the black or white filters that you put line in, phone out, modem out all onto |
| 20:42.06 | dlynes_laptop | [TK]D-Fender, you usually glue them or screw them onto the wall in the phone closet |
| 20:42.08 | [TK]D-Fender | dlynes_laptop: That would be a power bar then with line conditioning? |
| 20:42.17 | dlynes_laptop | [TK]D-Fender, no...it's only for DSL, not for power |
| 20:42.31 | [TK]D-Fender | dlynes_laptop: Absurb.... never seen heard of or needed... |
| 20:42.43 | dlynes_laptop | [TK]D-Fender, so that you can have analog conversations and DSL data communications, and the two don't interfere with one another |
| 20:43.06 | [TK]D-Fender | dlynes_laptop: Sounds like you've been scammed for what should be a 5$ filter (I'm rounding UP) |
| 20:43.11 | dlynes_laptop | [TK]D-Fender, nah |
| 20:43.23 | dlynes_laptop | [TK]D-Fender, we've tried the $5 and even the $20 filters |
| 20:43.30 | dlynes_laptop | [TK]D-Fender, they don't work in all cases |
| 20:43.37 | [TK]D-Fender | dlynes_laptop: then something is royally screwed up where you are. |
| 20:43.59 | dlynes_laptop | [TK]D-Fender, well, not all of our customers are in pristine downtown core offices |
| 20:43.59 | [TK]D-Fender | dlynes_laptop: I've done installs across ON, and QC without need for more. |
| 20:44.09 | *** join/#asterisk X-Rob (n=rob-x@203.37.172.162) |
| 20:44.22 | dlynes_laptop | [TK]D-Fender, some of them are in the likes of surrey, or really old buildings with really old wiring |
| 20:44.29 | [TK]D-Fender | dlynes_laptop: I'm talking 50 year old crappy residential wiring too... |
| 20:44.34 | tzanger | damn |
| 20:44.41 | tzanger | I just had an echoey call out to Vancouver through Unlimitel |
| 20:44.48 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 20:44.52 | dlynes_laptop | imagine that |
| 20:45.01 | tzanger | [TK]D-Fender: that's why I put this stuff in right at the demarc |
| 20:45.04 | dlynes_laptop | an imperfect call on a $10/mo unlimited minute plan |
| 20:45.15 | tzanger | dlynes_laptop: eh? |
| 20:45.26 | [TK]D-Fender | dlynes_laptop: unlimited for $10? really... |
| 20:45.32 | dlynes_laptop | Unlimitel is like $10/mo for unlimited calls out here |
| 20:45.41 | tzanger | dlynes_laptop: tell me where I can get $10/mo for unlimited minutes for a business |
| 20:45.47 | dlynes_laptop | Oh..not for business |
| 20:45.47 | robl^ | yay! gotta love crappy ancient wiring. pre-modular 4 conductor, wired for "party-lines" |
| 20:45.50 | dlynes_laptop | that's for residential |
| 20:46.04 | Strom_M | robl^, ? |
| 20:46.14 | tzanger | dlynes_laptop: regardless, I terminate directly to their NAS equipment (not to their servers), and call quality is usually exceptional |
| 20:46.25 | Strom_M | robl^, party-line wiring is more dependent on the telephone set than the 4-wire quad cable |
| 20:46.34 | dlynes_laptop | [TK]D-Fender, so you don't work from demarc? |
| 20:47.10 | robl^ | Strom_M: right. but the wiring was still going to a party line switch at the CO. they have to move it. |
| 20:47.31 | Strom_M | oh okj |
| 20:47.36 | [TK]D-Fender | dlynes_laptop: usually |
| 20:47.44 | Strom_M | so what was this "party line switch"? |
| 20:47.50 | Strom_M | a step-by-step exchange? :) |
| 20:48.06 | aydiosmio | I bet it has a crank |
| 20:48.22 | tzanger | hmm so I need to find a small form factor fanless PC with ethernet and 2 PCI |
| 20:48.23 | dlynes_laptop | party lines were cool |
| 20:48.34 | dlynes_laptop | You could catch up on all the latest gossip about your neighbors :) |
| 20:48.41 | *** join/#asterisk endikos (n=endikos@gtlgateway.nmsu.edu) |
| 20:48.45 | robl^ | Strom_M: not sure. it's a VERY old swtich at Cicninnati bell. only for like all 6 party line customers. People that haven't changed service in the past half century |
| 20:49.00 | Strom_M | robl^, define VERY old |
| 20:49.14 | Mad|Cow | [TK]D-Fender: I currently have: type=friend; regexten=3069; username=3069s secret=mypassword; host=dynamic; nat=yes; externip=69.242.72.104 ; localnet = 74.92.109.0/255.255.255.0; qualify = yes ; canreinvite=no. |
| 20:49.14 | robl^ | Strom_M: older than me |
| 20:49.21 | Strom_M | robl^, and how old are you? |
| 20:49.38 | robl^ | Strom_M: lets change the subject. ;-) |
| 20:49.57 | Strom_M | because you can do party-line stuff in 1ESS, 5ESS, and DMS |
| 20:50.04 | [TK]D-Fender | Mad|Cow: 1/2 of that doesn't belong in your phone setup, but rather up in [general] |
| 20:50.13 | Strom_M | so it's a line class code thing, not necessarily an equipment change |
| 20:50.19 | tzanger | *coughs* voipsupply has a Pentium-D Shuttle PC with 1G RAM, 100G SATA HDD, a Linksys SPA-841 and a TDM11B for USD$1500 |
| 20:50.36 | CunningPike | None of you commercial guys are using solid state servers for Asterisk? |
| 20:50.42 | tzanger | CunningPike: I will be |
| 20:50.50 | [TK]D-Fender | tzanger: *cough* couldn't find anything WORSE? *cough* |
| 20:50.57 | CunningPike | tzanger: Which one are you looking at? |
| 20:51.00 | dlynes_laptop | CunningPike, we're looking at it...trying to find more free time |
| 20:51.06 | tzanger | [TK]D-Fender: oh Im' sure there's worse, that was just typing "asterisk smal form factor" into google |
| 20:51.40 | Mad|Cow | [TK]D-Fender: Even if some of this isnt in the general, would it still cuase this not to work? |
| 20:51.45 | robl^ | Strom_M: Dunno all the details. This is just something that was relayed to me by a relative 1200 miles away |
| 20:51.55 | Mad|Cow | [TK]D-Fender: The phone works if I'm not behind a firewall |
| 20:52.19 | [TK]D-Fender | Mad|Cow: Do you think this stuff doesn't matter? yes its important where it occurs! Pastebin the whole thing so we can clean up the mess. |
| 20:52.26 | endikos | Hey folks, when Using a SIP extension to Dial out to a POTS number via my TDM400, the other party can hear me just fine, but they're a bit soft to me. rxgain doesnt seem to be having any effect... what can I do to increase that volume? |
| 20:52.57 | dlynes_laptop | CunningPike, what exactly are you looking for? Something you can mount to the wall, or something you can throw into a small rack that's mounted on the wall? |
| 20:52.57 | Strom_M | robl^, you should know by now that ordinary folks dont know phones :) |
| 20:52.58 | robl^ | endikos: press the volume up button on the sip phone? |
| 20:53.11 | DocHolliday | loll |
| 20:53.40 | CunningPike | dlynes_laptop: Something that can be left in an office that has no server room, no IT staff and can handle being unplugged by the cleaning lady |
| 20:53.50 | endikos | robl^: never woulda thought of that one. shucks and darn yer shoor smart! |
| 20:53.59 | dlynes_laptop | CunningPike, Do you want to be able to put any tdm cards in it? |
| 20:54.03 | CunningPike | dlynes_laptop: Not in terms of call preservation, of course - just HDD preservation :) |
| 20:54.06 | dlynes_laptop | CunningPike, or is it pure SIP? |
| 20:54.14 | CunningPike | dlynes_laptop: We're hoping to put a PRI card in it |
| 20:54.27 | CunningPike | dlynes_laptop: Or an external gateway would also work |
| 20:54.54 | dlynes_laptop | CunningPike, if external gateway is fine, try looking at something like WRAP |
| 20:54.59 | CunningPike | Has anyone had any success with the Xorcom devices? |
| 20:55.10 | CunningPike | dlynes_laptop: Thanks - I'll take a look at that one |
| 20:55.19 | dlynes_laptop | CunningPike, they have miniPCI, but sangoma doesn't have any miniPCI cards yet |
| 20:55.22 | *** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161) |
| 20:55.32 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
| 20:55.35 | CunningPike | dlynes_laptop: WRAP? |
| 20:55.37 | dlynes_laptop | CunningPike, otherwise, take a look at the dell sc430(?), or the Soekris boards |
| 20:55.46 | dlynes_laptop | CunningPike, yeah |
| 20:56.14 | CunningPike | dlynes_laptop: I was looking at Soekris - but they're just boards, right? I'd need a Tupperware container or something for it, right? :) |
| 20:56.26 | dlynes_laptop | Nah....Soekris you can get fully enclosed |
| 20:56.37 | *** join/#asterisk FutZ247 (n=rob@209.248.134.245.nw.nuvox.net) |
| 20:56.39 | CunningPike | dlynes_laptop: Oh, OK - thanks - good info to go on |
| 20:56.45 | dlynes_laptop | CunningPike, for WRAP, you buy a board, a case, a flash card, ... |
| 20:57.07 | dlynes_laptop | CunningPike, you might be able to get something fully assembled for WRAP, but generally, it's a do it yourselfer |
| 20:57.08 | CunningPike | I'd sure be interested in real world Xorcom experience, though - their products look really neat |
| 20:57.30 | dlynes_laptop | CunningPike, if you order WRAP systems in quantities of 10 or more, you usually get discounts, too |
| 20:57.31 | CunningPike | dlynes_laptop: OK - probably rules it out then - has to be commercial-grade reliable |
| 20:57.37 | CunningPike | dlynes_laptop: OK - thanks |
| 20:57.44 | aydiosmio | 10-4 |
| 20:57.45 | dlynes_laptop | CunningPike, it is commercial grade reliable |
| 20:57.47 | CunningPike | Anyway, meeting time - ttyl |
| 20:57.59 | aydiosmio | B - YE |
| 20:58.00 | dlynes_laptop | CunningPike, I know someone that's doing a rollout of hundreds of them in Japan, atm |
| 20:58.02 | FutZ247 | could someone here possibly give me a hand with bridging PRI and SIP? |
| 20:58.08 | CunningPike | dlynes_laptop: OK - good to know |
| 20:59.34 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 21:00.30 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 21:00.44 | Mad|Cow | [TK]D-Fender: http://channels.debian.net/paste/5110 |
| 21:01.26 | mercestes | Is the Sip Firmware 2.0.1 pretty stable?? |
| 21:01.30 | Ryanw | Hello All!, i'm using a TDM400P (4xFXO) and ocasionally when i place a call the called party only hears me speak at a very low volume, any ideas? |
| 21:01.43 | Mad|Cow | [TK]D-Fender: I really apprichate this.. I just dont understand where I am going wrong |
| 21:01.59 | dlynes_laptop | Ryanw, increase your txgain |
| 21:02.44 | [TK]D-Fender | mercestes: Pretty much like the rest. Though mind you we're up to at least 2.0.3 now |
| 21:02.48 | Ryanw | dlynes, i tried increasing my txgain and it intorduced echo, it only happens ocasionally 95% of calls work fine. |
| 21:02.54 | mercestes | Ryanw: Yea, I tend to suggest txgain=38497 for a good, default starterpoint. |
| 21:03.07 | mercestes | I'm kidding btw. Don't do that. |
| 21:03.10 | [TK]D-Fender | ~lart mercestes |
| 21:03.15 | FutZ247 | lol |
| 21:03.22 | dlynes_laptop | Ryanw, sounds like you need to change your echo cancel algorithm, buy a hardware echo canceller, and/or get a faster cpu |
| 21:03.37 | [TK]D-Fender | mercestes: In a vacuum-sealed can... no one can hear you cry :D |
| 21:03.40 | mercestes | Ryanw: Or reject your T1 and yell at your telco |
| 21:03.43 | Ryanw | txgain={if problem?txgain=6:txgain=2.4} |
| 21:03.43 | Ryanw | lol |
| 21:04.07 | ellisdee | Ryanw, play with echotraining |
| 21:04.10 | ellisdee | echotraining=400 or something |
| 21:04.18 | ellisdee | echocancel=64 |
| 21:05.11 | mercestes | I'm going to do that next time I want to yell at someone. txgain=97 or something....and then yell. |
| 21:05.31 | mercestes | I'll crank up the .tx values on my polycom too |
| 21:05.32 | FutZ247 | hehe make it a feature code |
| 21:05.42 | Ryanw | ellisdee, does echocancelwhenbridged effect calls from a sip phone out a zap channel? |
| 21:06.00 | c4t3l | hello ellisdee |
| 21:06.16 | Hmmhesays | bah my func_odbc is not working right |
| 21:06.19 | c4t3l | i say hello ellisdee |
| 21:06.23 | mercestes | So I get a "Got sip 500, internal server error back from" errors on phones sometimes and i have to reboot them. any hints on what causes that? Polycom phones, firmware 2.0.1 |
| 21:06.24 | ellisdee | c4t3l, hello |
| 21:06.29 | mercestes | Do I need to upgrade? |
| 21:06.38 | c4t3l | yes! |
| 21:06.44 | ellisdee | Ryanw, not sure |
| 21:06.46 | c4t3l | upgrade of what ?? |
| 21:07.05 | [TK]D-Fender | mercestes: Messing with base gains in Polycom's is NOT a good idea... |
| 21:07.06 | mercestes | Shush, c4t3l. Your still using Sip 1.6.9 because you like the version number. |
| 21:07.15 | c4t3l | true |
| 21:07.26 | [TK]D-Fender | mercestes: And that message gets spewed out by ALL firmware releases....\ |
| 21:07.35 | mercestes | [TK]D-Fender: Actually.....I fixed some call clipping issues on speakerphone messing with those gain settings, but..I had a few engineers on the phone at the time. |
| 21:08.21 | mercestes | [TK]D-Fender: ya. I fig'd. What causes it? I seem to be getting one phone a day... |
| 21:08.29 | mercestes | rebooty clears it. |
| 21:08.42 | [TK]D-Fender | mercestes: I got an answer to that a while back, but don't recall the details. |
| 21:09.25 | mercestes | [TK]D-Fender: What was the general consensus? Was there a fix? was it just a "deal with it" situation? |
| 21:09.49 | [TK]D-Fender | mercestes: Apparently "deal with it". I should look it up again though. Check Mantis |
| 21:10.04 | mercestes | Alrighty, thanks. |
| 21:10.18 | `Sean | mercestes i got my phone today |
| 21:10.46 | *** part/#asterisk FutZ247 (n=rob@209.248.134.245.nw.nuvox.net) |
| 21:11.26 | mercestes | `Sean: yay! Are you happy with it? |
| 21:12.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:14.05 | PupenoR | [OT] Does anybody know howo to find out the provisioning names of the fields of a Linksys PAP2 ? |
| 21:15.11 | *** join/#asterisk Mad|Cow (n=madcow@c-69-242-72-104.hsd1.de.comcast.net) |
| 21:17.10 | queuetue | Does anyone maintain "turnkey parts" lists for linux hardware configurations? Like ARS Technica's old "budget box"... I just want to put together a low-level linux NAS and Asterisk server, but am just not up to speed on the hardware anymore... |
| 21:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:18.04 | `Sean | mercestes of course manim in love with it especialy the receiver lol its like perfectly made |
| 21:18.24 | Mad|Cow | [TK]D-Fender: I dont know if you saw my post to you ealier, but I have my sip.conf out at http://channels.debian.net/paste/5110. If you could give it a once over, I would apprichate it. |
| 21:18.24 | `Sean | i spent 120 on this hunk of crap phone i have now wich is anologue |
| 21:18.26 | `Sean | and i regret it |
| 21:18.42 | `Sean | Now, i gotta get a support contract with Cisco |
| 21:18.46 | `Sean | so i can get login info for Cisco.com |
| 21:18.51 | `Sean | and download product updates and so on |
| 21:19.23 | [TK]D-Fender | Mad|Cow: Does your * box have a public IP? |
| 21:19.43 | Mad|Cow | [TK]D-Fender: yes (74.92.109.201) |
| 21:20.20 | *** part/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
| 21:20.26 | perd | anyone know how to create an extension in CCM that will dial to asterisk? i have my dial-patterns set up on the router, i can get calls externally to go to my asterisk box but i cant get internal phones to be able to dial the extensions on asterisk :/ |
| 21:21.05 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:26.42 | *** part/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
| 21:26.44 | [TK]D-Fender | Mad|Cow: http://www.pastebin.ca/318928 |
| 21:26.52 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:27.15 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:29.54 | Mad|Cow | [TK]D-Fender: No go :-( |
| 21:30.09 | [TK]D-Fender | Mad|Cow: You'll hagve to reset the remote device |
| 21:30.33 | Mad|Cow | [TK]D-Fender: I restrated Asterisk and restrated my SIP phone |
| 21:31.04 | endikos | <PROTECTED> |
| 21:34.05 | Hmmhesays | one of the very best star trek episodes is on |
| 21:34.29 | endikos | Hmmhesays: Which one? |
| 21:34.38 | *** join/#asterisk Globetrotter (n=eric@205.211.214.167) |
| 21:34.41 | Hmmhesays | the tng episode where they question data's rights |
| 21:34.55 | endikos | that was a good one. |
| 21:35.16 | endikos | Data had a lot of the best episodes |
| 21:35.23 | Hmmhesays | true |
| 21:35.23 | Globetrotter | hi Guys, iwant to record all calls in and out of my PBX.. is that possible? if so how?? |
| 21:35.43 | Hmmhesays | right up to where they killed his character off on the big screen |
| 21:35.52 | endikos | yeah |
| 21:36.18 | endikos | My jaw dropped about three feet when they did that |
| 21:36.33 | Hmmhesays | yeah no doubt, that was a pretty awesome scene of him flying through space though |
| 21:37.11 | Mad|Cow | [TK]D-Fender: Is there anything else you might suggest? |
| 21:37.56 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-137.mtco.com) |
| 21:38.05 | perd | such crap, i need to set up an h.323 channel to communicate with my old ass CCM crap |
| 21:39.09 | monsted | upgrade your old ass CCM crap and you can use SIP :) |
| 21:39.21 | Qwell[] | ditch your old ass CCM crap and you don't need either |
| 21:39.43 | monsted | and you can use a new ass asterisk crap instead :) |
| 21:39.51 | Qwell[] | exactly |
| 21:39.54 | Qwell[] | wait a minute :P |
| 21:41.37 | perd | yeah im replacing CCM with asterisk |
| 21:41.42 | perd | i want to migrate users over slowly |
| 21:41.44 | perd | so i can work out kinks |
| 21:41.55 | perd | and i dont have a service contract with cisco any more |
| 21:42.01 | perd | <PROTECTED> |
| 21:42.07 | perd | unless of course i pay them a shitton of money |
| 21:42.38 | Mad|Cow | [TK]D-Fender: I have to run, I'll check back with you tonight again if your on. Thanks for all the help! |
| 21:42.45 | aptura | ccm with asterisk? |
| 21:43.05 | aptura | ahh okay |
| 21:43.08 | endikos | Damn. looks like the busydetect routines are hardcoded. |
| 21:43.32 | aptura | What is the most bug free solid working version of asterisk as of late ? |
| 21:43.54 | Qwell[] | aptura: 1.4.0 of course |
| 21:43.57 | perd | 1.4 just came out, i had to stay with 1.2 though |
| 21:44.01 | perd | thanks chan_sccp! |
| 21:44.06 | Globetrotter | <PROTECTED> |
| 21:44.18 | Qwell[] | perd: yeah...about that |
| 21:44.22 | perd | globetrotter hells yes it is! |
| 21:44.25 | aptura | globe its possible but you may have to do that on your own. |
| 21:44.25 | Strom_M | what is the option to frob in the polycom configuration file to change the admin password of the phone? |
| 21:44.29 | Qwell[] | maybe later this week I'll try to get to it :p |
| 21:45.05 | aptura | or find somone that is willing to help. So far two people on this site that said thay would pay a fee to me never have so I dont except typed promisses anymore :) |
| 21:45.07 | perd | http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
| 21:45.08 | Globetrotter | soi t is not possble to get help in here for call recording? |
| 21:45.15 | perd | check that globe |
| 21:45.23 | Globetrotter | ok thanks |
| 21:45.58 | aptura | Qwell, do you do in field consulting |
| 21:46.03 | Dr-Linux | can we send our own differen callerID's in ${CALLERID(num)} variable? |
| 21:46.11 | Qwell[] | aptura: I don't do consulting anymore |
| 21:46.25 | perd | qwell haha later this week you say? i'm not holding my breath :P |
| 21:46.30 | aptura | okay so you found another way to suplment your income then. |
| 21:46.33 | Qwell[] | aptura: If, however, I help you with something, and you feel like taking care of something on my wish list... :P |
| 21:46.47 | Qwell[] | (I don't actually know if I can accept bribes like that either, heh) |
| 21:46.50 | perd | anyone know where to get the recommended versions of pwlib and openh323 |
| 21:47.03 | Qwell[] | perd: good idea :P |
| 21:50.13 | ellisdee | a/s/l |
| 21:50.30 | perd | please no |
| 21:50.41 | ellisdee | lol |
| 21:50.51 | aptura | Qwell, still not alot of people up here know what a pbx is or does or in come cases dont know what linux is :) BTW I am looking at embeded linux boards to run asterisk. Looking at ideas. Want rock solid stability and reliability or I become a liability :) Like it to handle 8 connections at same time using the g729 codec. Perhaps g711 ulaw since its not licenced. BTW what is the cost of 729 and really does it make a difference? |
| 21:51.30 | aptura | ohh and qwell what is your wish list |
| 21:53.23 | Qwell[] | aptura: You know, it's never come up, so I've never created one :p |
| 21:53.25 | perd | dude, the wrt54g runs asterisk |
| 21:53.27 | perd | with openwrt |
| 21:53.30 | perd | it's amazingly cool |
| 21:53.50 | perd | and i ran it on a WRAP 1b board from pcengines.ch |
| 21:53.54 | aptura | yes but it needs one pci slot |
| 21:53.56 | perd | that was also nifty |
| 21:54.04 | perd | the WRAP board has minipci |
| 21:54.06 | Qwell[] | aptura: If you want a pre-made, supported embedded box, you can check out our Asterisk Appliance Developers Kit |
| 21:54.22 | Qwell[] | and g729 is $10 per channel |
| 21:54.32 | perd | i asume you want to drive an FXS though |
| 21:54.43 | Qwell[] | You don't get "rock solid stability and reliability" from a linksys router :) |
| 21:54.46 | perd | i dont think they make minipci FXS :/ but maybe you can make an adaptor |
| 21:54.56 | aptura | Qwell[] and really, other then bandwith savings which I dont know what it takes up sound quality is the same right? |
| 21:55.03 | perd | haha qwell whatever man! my little wrt54g rocks the voip :) |
| 21:55.20 | Qwell[] | aptura: sound quality is a bit less than ulaw |
| 21:55.38 | Qwell[] | but, it's like 1/8th the size, so that's to be expected |
| 21:55.39 | tzanger | no I have not seen minipci fxs or fxo |
| 21:55.41 | tzanger | unfortunately |
| 21:55.41 | aptura | These will be in small bussiness so my reputation is on the line. |
| 21:55.47 | tzanger | I would LOVE minipci adsl but can't find it either |
| 21:56.01 | Qwell[] | aptura: and if you want fxo/fxs on an embedded device, ours has modules :p |
| 21:56.03 | tzanger | yes WRAP boards do rock |
| 21:56.03 | perd | tzanger, time to get to work then, sir. make us some hardware. |
| 21:56.07 | Qwell[] | </commercial> |
| 21:56.13 | aptura | Qwell, which ones? |
| 21:56.14 | tzanger | perd: I've taken a look at it |
| 21:56.19 | perd | screw digium, pcengines for you! :P |
| 21:56.19 | tzanger | unfortunately I don't think the market's there |
| 21:56.31 | perd | i didnt know digium made an embedded device |
| 21:56.33 | perd | i gotta check this out |
| 21:56.39 | tzanger | yes they did |
| 21:56.40 | Qwell[] | http://www.digium.com/en/products/hardware/aadk.php |
| 21:56.58 | Qwell[] | It's currently a Developer Kit (ie; not for mass consumption yet), but that will change soon |
| 21:57.02 | Grnd-Wire | I'm trying to find the best way to "walk" through a variable that contains a phone number, and extra each number out individually.. Does anyone know how to make that work? |
| 21:57.15 | perd | man, all that coolness without any damn minipci fxo/fxs cards. for shame. |
| 21:57.22 | perd | i want a minipci pri also |
| 21:57.42 | Grnd-Wire | I'm guessing I'll just use the ${VAR:x:y} with an incrementing variable below it.. right?? So the question is, how do I get the TOTAL LENGTH of the variable? That's the one thing I don't have right now. |
| 21:57.49 | perd | man that's a cool looking developer kit though, is it affordable? :) |
| 21:58.03 | tzanger | not really |
| 21:58.04 | tzanger | $4k |
| 21:58.10 | perd | jesus |
| 21:58.10 | aptura | Qwell, I like to support digium but of course I worry about giving the companies name away may encourage other business to go on there own and buy the system. |
| 21:58.12 | Qwell[] | with training |
| 21:58.15 | perd | i was hoping like, 400 bucks |
| 21:58.24 | Qwell[] | aptura: it will be OEMable, afaik |
| 21:58.40 | tzanger | yes |
| 21:58.42 | Hmmhesays | that was a hell of a speech picard gave |
| 21:58.44 | tzanger | but at what quantities |
| 21:58.50 | tzanger | I'm guessing no less than 1k/a |
| 21:58.52 | aptura | What do you mean by OEMable? |
| 21:59.03 | Qwell[] | aptura: ie; rebranded |
| 21:59.49 | aptura | yea okay. Last thing I would want is a client get pissed off at me and say why are you charging above what we can buy online? |
| 22:00.12 | Grnd-Wire | aptura: Because you get support, and a custom tailored configuration.. |
| 22:00.36 | Grnd-Wire | You can go buy a server from HP, but that doesn't mean it's got the OS on it, setup for them.. That takes ALOT of extra time, and money to pay for that work@! |
| 22:00.52 | aptura | I have done enough asterisk configurations. |
| 22:01.06 | aptura | But i am also getting rusty :) |
| 22:01.35 | Grnd-Wire | aptura: I was explaining how you could charge more than Digium charges, and explain it to your customer.. |
| 22:01.55 | aptura | Grnd, Canadians are different |
| 22:02.02 | Grnd-Wire | That box doesn't come with tailored onsite support, or hookup.. hell, it doesn't even come with phones.. |
| 22:02.17 | Grnd-Wire | HAHA.. Well uhh.. Is this going to become some sort of socialist vs. capitalist discussion? :D |
| 22:02.31 | aptura | hehe |
| 22:04.20 | aptura | There was a article about Canadian buying behaviors and I dont recall if it was specifically pointing out British Columbians or Just Canada as a whole but a sale agent mentioned that you can try and sell a life saving device to a residential owner and most of the time thay would reject. Then the house would burn down and the owner would say "we need to buy a smoke alarm" |
| 22:04.45 | greendisease | is anyone who manages the asterisk.org accounts around? |
| 22:04.54 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 22:04.55 | aptura | But now the table has turned more and going into debt thay used to have really good saving accounts. |
| 22:08.02 | aptura | I hope this is the last snow of the year. |
| 22:15.34 | *** join/#asterisk zriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 22:15.55 | zriah | Hi all. Is there a way to implement e911 with just asterisk? |
| 22:16.39 | *** join/#asterisk mencken (n=shadow@63.163.216.254) |
| 22:16.51 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
| 22:16.59 | CunningPike | zriah: You - you'll need a PSAP as well |
| 22:17.09 | CunningPike | s/You -/No,/ |
| 22:17.23 | shepimport | Hey guys- has anyone messed with the routing of an RTP stream with asterisk? |
| 22:17.48 | Hmmhesays | anyone know how hard it would be to return a variable from app_directory? |
| 22:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.19 | zriah | How would one implement e911 then? Just forward calls based on a psap database lookup? |
| 22:18.38 | shepimport | I.e. asterisk box in data center... users are all offsite in "WAN" envirmont ... user A calls user B.. will asterisk route the RTP stream through itself? |
| 22:19.04 | *** part/#asterisk amdtech (i=adaniel@nat/digium/x-fa84570583ded71b) |
| 22:23.29 | mercestes | we got allthe "local" 911 numbers for our area and set a "call center" variable for each of our customers so we could direct them to the proper 911 call center, so we basically had our own database local. |
| 22:23.33 | mercestes | but...it was an isolated area. |
| 22:29.39 | *** join/#asterisk amdtech (i=adaniel@nat/digium/x-25bd454e1d16e384) |
| 22:30.46 | [TK]D-Fender | Hmmhesays: You know rather than having * dial from app_directory, it SHOULD jsut exit with an exten that YOU choose how and if to dial |
| 22:33.02 | Hmmhesays | [TK]D-Fender: it does exit with the exten you have defined in vm.conf or wherever you store your directory info, the problem is I need to return the users name for an sql query later in the dialplan |
| 22:36.38 | [TK]D-Fender | Hmmhesays: I mean it DIALS it in the target exten. It would be better to exit with a variable in the CURRENT context. |
| 22:38.25 | *** join/#asterisk Juxt (n=denis@66.151.46.42) |
| 22:39.50 | *** join/#asterisk xnon (n=xnon@190.38.157.59) |
| 22:40.19 | shepimport | Does anyone know how asterisk would handle this? An asterisk box in data center... users are all offsite in "WAN" envirmont ... site A calls site B .. will asterisk route the RTP stream through itself/the datacenter? |
| 22:40.38 | shepimport | I could not find a value to enable direct RTP streams |
| 22:40.58 | Hmmhesays | [TK]D-Fender: thats what I'm looking to make it do |
| 22:41.20 | [TK]D-Fender | Hmmhesays: I'm beeting it'd be a damn easy hack. |
| 22:41.28 | [TK]D-Fender | betting* |
| 22:42.15 | [TK]D-Fender | shepimport: "canreinvite=yes" in sip.conf. Guess you didn't look too hard. |
| 22:42.45 | [TK]D-Fender | shepimport: Thogh if you're talking about 2 * boxes, you should be trunking through IAX to save on bandwidth. |
| 22:43.44 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 22:44.41 | shepimport | Fender: no, no ... that was what i was looking for... we use a reinvite in different context so i passed right over it... thanks! |
| 22:46.08 | *** part/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 22:46.37 | Hmmhesays | [TK]D-Fender: we'll find out in a second |
| 22:46.50 | Hmmhesays | i'm compiling my change right now |
| 22:47.16 | [TK]D-Fender | Hmmhesays: :O |
| 22:56.51 | EmleyMoor | Does setting a default caller ID (for use when caller ID can be set, of course) for a Zap FXS channel work? |
| 22:56.59 | EmleyMoor | (in zapata.conf) |
| 22:57.06 | EmleyMoor | Not had much luck with it yet |
| 23:00.13 | EmleyMoor | Ah, it does work |
| 23:00.51 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 23:09.00 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 23:10.49 | mavior | hello everybody...can someone please help me out with this http://pastebin.ca/319021 ? i'm getting this strange behaviour: i'm using xlite to connect to my asteris server with user called "me", and incoming calls works at the start , then stop working and then randomly after a while re-start to work for a bit.... |
| 23:12.11 | mavior | i guess is something related to the sip/404 not found error...seems like my asterisk cannot found anymore my user , even though x-lite shows status as connected |
| 23:12.39 | mavior | i really don't know how to get out from this one... |
| 23:13.00 | perd | hmmm, asterisk is sending no audio through the h323 channel, but i can hear incoming audio (from ccm) fine.. any ideas why this is? |
| 23:13.32 | De_Mon | firewall? |
| 23:13.53 | perd | each audio channel is on a separate port? |
| 23:15.24 | *** join/#asterisk prags2626 (n=pragsup@adsl-66-126-122-10.dsl.lsan03.pacbell.net) |
| 23:15.43 | De_Mon | maybe youre blocking inbound but not outbound or somesuch, I duno |
| 23:15.49 | perd | nah |
| 23:15.51 | perd | nothing blocked |
| 23:16.08 | Dr-Linux | for some reason i can't unlock my cisco 7960's phone :S there are 3 :S |
| 23:16.23 | Dr-Linux | if i press any key it says "key is not active" |
| 23:16.26 | perd | press **# then hit the settings button |
| 23:16.37 | perd | all fast like |
| 23:16.43 | Dr-Linux | perd: tried but it doesn't help |
| 23:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.34 | perd | did you try going into setttings, have network settings highlighted, hit **# then the select softkey |
| 23:19.37 | Dr-Linux | perd: hhm.. |
| 23:19.42 | Dr-Linux | nothing seems to work |
| 23:19.53 | Dr-Linux | perd: how can reset default factory |
| 23:20.07 | mavior | any ideas with this http://pastebin.ca/319021 ? |
| 23:20.09 | perd | under the main settings menu hit 'more' and you should have an option for it |
| 23:20.50 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:20.50 | Dr-Linux | perd: aww i hve to load SIP firmware on about 50 phones :( |
| 23:21.00 | Dr-Linux | 12 are done |
| 23:21.09 | perd | you dont need to physically access the phone |
| 23:21.21 | perd | just plugging it in should be enough assuming you have all your config files set up correctly |
| 23:21.34 | mercestes | bye |
| 23:21.47 | Dr-Linux | perd: i know that, but alreast i should unlock the phone and put the TFTP server ip |
| 23:21.58 | perd | DHCP should provide that |
| 23:24.22 | Dr-Linux | DHCP server is different and TFTP server is different |
| 23:25.16 | CunningPike | Hmm - What's the 'Linksys One' like - anyone tried it? |
| 23:25.40 | Dr-Linux | perd: default password "cisco" is invalid :S |
| 23:26.16 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 23:27.06 | zapp-branigan | hi, i have compiled the asterisk but the speex codec do not compile |
| 23:27.44 | zapp-branigan | and ./configure tell me : |
| 23:27.47 | zapp-branigan | checking for speex_encode in -lspeex... yes |
| 23:27.47 | zapp-branigan | checking speex/speex.h usability... yes |
| 23:27.47 | zapp-branigan | checking speex/speex.h presence... yes |
| 23:27.47 | zapp-branigan | checking for speex/speex.h... yes |
| 23:28.01 | zapp-branigan | but the .o and .so not appear |
| 23:28.14 | zapp-branigan | in the codecs directory |
| 23:28.23 | Qwell[] | Is it enabled in menuselect? |
| 23:28.30 | zapp-branigan | where ? |
| 23:28.35 | zapp-branigan | some file ? |
| 23:28.54 | zapp-branigan | please tell me where to activate |
| 23:29.25 | perd | oh shit RTP was messing me up |
| 23:29.28 | perd | thank god for h.323 debubg |
| 23:32.59 | zapp-branigan | in the menuselect directory the reference of speex is only in one file example_menuselect-tree |
| 23:35.04 | *** join/#asterisk Skarmeth (n=Skarmeth@201009032109.user.veloxzone.com.br) |
| 23:37.02 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 23:41.02 | tzafrir_laptop | zapp-branigan, when you run menuselect? |
| 23:41.17 | zapp-branigan | run ? |
| 23:41.28 | zapp-branigan | in the asterisk directory |
| 23:41.37 | zapp-branigan | i put ./configure |
| 23:41.43 | tzafrir_laptop | make menuselect or: ./menuselect/menuselect |
| 23:41.48 | zapp-branigan | make make install |
| 23:42.21 | zapp-branigan | i have estered in menuselect |
| 23:42.26 | zapp-branigan | ./configure |
| 23:42.31 | zapp-branigan | and cd .. |
| 23:42.36 | zapp-branigan | make clean |
| 23:42.38 | zapp-branigan | make |
| 23:43.01 | zapp-branigan | and speex.o is not generated |
| 23:43.10 | zapp-branigan | codec_speex.so |
| 23:43.15 | zapp-branigan | is not done |
| 23:43.35 | tzafrir_laptop | what is the value of MENUSELECT_CODECS in menuselect.makeopts ? |
| 23:43.47 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:44.27 | tzafrir_laptop | nick125_lappy, so what problems do you run into? |
| 23:44.51 | nick125_lappy | tzafrir_laptop: the ebuild kept trying to grab the file from the wrong mirror :P |
| 23:45.08 | zapp-branigan | the file menuselect.makeopts is not generated |
| 23:45.16 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:45.28 | nick125_lappy | Wewt! My PAP2T-NA should be here tomorrow! |
| 23:45.32 | zapp-branigan | yes yes i see |
| 23:45.34 | tzafrir_laptop | zapp-branigan, have you run 'make' in the toplevel directory? |
| 23:45.53 | zapp-branigan | MENUSELECT_CODECS=codec_speex |
| 23:46.16 | tzafrir_laptop | this means that it is disabled (menuselect configuration) |
| 23:46.30 | tzafrir_laptop | just remove codec_speex from there |
| 23:46.48 | zapp-branigan | ok |
| 23:47.03 | Dr-Linux | can set my callerID variable like this? >> ${CALLERID}(num)=${myvariable} |
| 23:47.11 | blitzrage | Corydon-w: usage question here... say I have a While() call such as follows: exten => *99,n,While($[${EXISTS(${PAGER_GROUP_MEMBERS(${ZONE},${PBX},${OFFSET})})}]). Now, it seems to be that I obviously have to make the call each time through the loop, but I'd like to not have to make two calls to the same ODBC function to get the data (since I just returned it in the While()). Is there a variable set, or some other met |
| 23:47.11 | blitzrage | hod I'm not thinking of that will allow me access to the "cached" copy of that function call? |
| 23:47.45 | Corydon-w | EXISTS? |
| 23:47.55 | hardwire | TRUE! |
| 23:47.55 | blitzrage | opposite of ${ISNULL()} |
| 23:48.02 | tzafrir_laptop | ${Set(CALLERID(num)=${myvariable}} |
| 23:48.20 | tzafrir_laptop | ${Set(CALLERID(num)=${myvariable})} |
| 23:48.33 | blitzrage | functions should be in uppercase |
| 23:49.37 | blitzrage | it'd be nice to use the value that was called from the While() instead of calling it again from the Set() application. |
| 23:49.52 | Corydon-w | blitzrage: no, it is not cached anywhere |
| 23:50.02 | blitzrage | ok cool -- just wanted to check to make sure while I was thinking of it |
| 23:50.02 | Dr-Linux | tzafrir_laptop: thanks, so will it work this way? |
| 23:53.09 | *** part/#asterisk amdtech (i=adaniel@nat/digium/x-25bd454e1d16e384) |
| 23:56.06 | aptura | what does vonage use for its sip backbone SER? |
| 23:56.37 | Corydon-w | blitzrage: in some While loops, you might actually want the function to re-execute and re-evaluate |
| 23:57.46 | blitzrage | Corydon-w: true -- would be nice to have that option though :) |
| 23:58.05 | Corydon-w | blitzrage: just do the Set just before you execute the While |
| 23:58.10 | blitzrage | just seems inefficient to have to call it twice each time through the loop |
| 23:58.21 | blitzrage | Corydon-w: but I'm doing the loop because I'm calling back multiple rows with OFFSET |
| 23:58.32 | blitzrage | so calling before the loop defeats the purpose of the loop :) |
| 23:58.33 | Corydon-w | Aha |
| 23:58.41 | Corydon-w | Right ;-) |
| 23:58.44 | blitzrage | indeed aha :) |
| 23:58.45 | blitzrage | heheh |
| 23:59.04 | blitzrage | so you see my dilemma :) |
| 23:59.11 | blitzrage | well... not dilemma... but issue |
| 23:59.17 | Corydon-w | I don't see a dilemna at all |
| 23:59.29 | Corydon-w | You know exactly what you have to do |
| 23:59.39 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 23:59.49 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:59.52 | blitzrage | right, I'm just curious about scalability. It's probably not that big a deal because SELECTs are pretty low cost. |
