| 00:01.04 | [TK]D-Fender | Zodiacal : generate a fax/modem tone |
| 00:01.30 | Zodiacal | that would be kinda loud |
| 00:01.56 | Ryanw | Dex...dial the original dialed number in a context at the remote office over the iax trunk |
| 00:03.43 | dexteruk | Ryanw: what do you mean? |
| 00:03.43 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 00:04.33 | Ryanw | its been ages since i played with an iax trunk, but grab the incoming sip call and dial iax/remoteoffice/${EXTEN} then at the remote office catch it with _X. and do your matching on ${EXTEN} |
| 00:05.46 | dexteruk | Ryanw: I have tried but i dont see the original number dialed |
| 00:06.34 | dexteruk | this is how the call starts exten => +1XXXXXXXXXX,1,Goto(1XXXXXXXXXX-In,s,2) |
| 00:07.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:13.54 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:15.01 | kronic | [TK]D-Fender: it still remembers the member that answer and starts off on the next |
| 00:17.41 | Ryanw | is it possible to set a default periodic announcement in queues.conf ? |
| 00:17.46 | *** join/#asterisk zeppelin_ (n=zeppelin@201.35.144.2) |
| 00:17.49 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 00:20.02 | ma-requin | do I really have to have extra hardware other than PC and wireless card to call PSTN ?? |
| 00:20.31 | [TK]D-Fender | ma-requin : depends HOW you intend to get to the PSTN. |
| 00:21.19 | Ryanw | ma-requin, you can but a cheap & nasty $30 FXO card off ebay and use a single PSTN phone line, or you can use an interhet connection and voip provider to terminate the calls for you. |
| 00:21.38 | ma-requin | <[TK]D-Fender> I am behind a belkin N1 router which is connected to a gateway |
| 00:23.02 | [TK]D-Fender | ma-requin : and that in no way answers my question. |
| 00:25.03 | *** join/#asterisk nortex (n=barracud@adsl-66-142-12-231.dsl.amrltx.swbell.net) |
| 00:25.27 | ma-requin | we already have a voip island and the NAT router is connected to "vega150 ip voice gateway" |
| 00:27.34 | kronic | Ryanw: that doesn't help still, if I were to pickup a particular call, it remember the member and next call progress |
| 00:27.56 | [TK]D-Fender | ma-requin : Well if you have a VoIP gateway then * can use that for PSTN connectivity and needs no extra special hardware |
| 00:28.59 | fiber0pti | how do you turn off auto fallthrough |
| 00:29.07 | ma-requin | <[TK]D-Fender> can I pm you ? |
| 00:29.18 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
| 00:30.36 | [TK]D-Fender | ma-requin : just ask in here |
| 00:31.29 | ma-requin | so I hook my NAT router to the voice gateway and configure SIP and it should work ?? |
| 00:32.16 | Nugget | what could possibly go wrong? |
| 00:33.25 | [TK]D-Fender | ma-requin : Could be. What do you mean by hooking your Voice Gateway to your NAT router? What do they have to do with each other? |
| 00:33.40 | [TK]D-Fender | Nugget : Ask Murphy..... no wait, he's DEAD. |
| 00:34.23 | [TK]D-Fender | Nugget : "If ever I reacht he end of the rainbow as good fortune did intend, Murphy would be there to tell me the pot's at the OTHER end" |
| 00:34.32 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:35.09 | Nugget | heh |
| 00:35.33 | ma-requin | <[TK]D-Fender> we already have a working voip island with sip phones. I am trying 2 hook a NAT router to the voip island and make succesful calls to lan wan and pstn from a linux box using 802.11n wireless |
| 00:36.21 | JT | hmm, a strange question, but, does asterisk support DTMF ABCD digit pattern matching |
| 00:36.28 | JT | ie. it won't take it as dialing by name |
| 00:43.36 | ma-requin | <[TK]D-Fender> did you see my last msg ??? |
| 00:47.50 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 00:47.50 | *** join/#asterisk fredflintstone (i=nbdy@gateway/tor/x-c047b6e7ea214274) |
| 00:48.07 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:48.26 | [TK]D-Fender | ma-requin : What is a "voi-island" and what exact service is your NAT router supposed to be doing in relation to it? |
| 00:50.15 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 00:51.25 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:51.52 | ma-requin | <[TK]D-Fender> my computer is connected to the NAT router with a wireless card |
| 00:52.44 | ma-requin | <[TK]D-Fender> voipisland = a fully functional voip system |
| 00:56.14 | [TK]D-Fender | ma-requin : Sounds like its just "networking" so yeah no reason the independant bits shouldn't work just fine. |
| 01:01.01 | *** join/#asterisk _DAW (n=_DAW@adsl-157-54-57.msy.bellsouth.net) |
| 01:01.49 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 01:02.50 | _DAW | hello mates |
| 01:03.53 | shy_guy | hi |
| 01:04.02 | shy_guy | insecure=very |
| 01:04.37 | shy_guy | if i put that in a friend's context, does that mean it won't be authenticated when i receive a call from him |
| 01:06.18 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:06.29 | [hC] | anyone used sangoma's hardware echo debugging tools? |
| 01:07.25 | _DAW | shy_guy: doesnt that go in sip.conf? |
| 01:07.36 | [TK]D-Fender | [hC] : have you spoken with "The Korbic" yet? |
| 01:07.47 | shy_guy | yeah it is in sip.conf |
| 01:08.12 | shy_guy | by context, i refer to a sip account in sip.conf |
| 01:08.52 | _DAW | ahh |
| 01:11.29 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:11.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:12.47 | [hC] | [TK]D-Fender: not about this.. ive been speaking to yuan s or something |
| 01:13.01 | [hC] | echo is making me crazy |
| 01:13.11 | [hC] | i had a complaint today about sip->sip echo, on the same pbx. (!!) |
| 01:13.54 | [TK]D-Fender | [hC] : Call..... The Korbic..... |
| 01:14.02 | [TK]D-Fender | [hC] : What phones? |
| 01:14.06 | [hC] | [TK]D-Fender: polycom. |
| 01:14.15 | [hC] | [TK]D-Fender: 501's. |
| 01:14.41 | [hC] | [TK]D-Fender: recently turned on acoustic echo can and suppression on them, and moved all internal calling to g729 instead of ulaw, to try to solve it, but that doesnt seem to have helped. |
| 01:14.45 | [TK]D-Fender | [hC] : Ok, you've entered the realm of "High Improbability". You should have taken the other pill.... |
| 01:15.08 | [TK]D-Fender | [hC] : Something else has gone horrifically wrong. |
| 01:15.08 | [hC] | [TK]D-Fender: and their hwec tool seems to return like 1000+ ms ranges when taking a sample with audio on the line.. messed up. |
| 01:15.18 | [hC] | [TK]D-Fender: what else could it BE?! |
| 01:15.38 | [hC] | [TK]D-Fender: i mean... sip->sip echo?? can that be caused by a f'ed up LAN? jitter maybe.. but.. echo? |
| 01:15.42 | [TK]D-Fender | [hC] : For SIP/SIP I dunno.... nasty network conditions? |
| 01:15.59 | [TK]D-Fender | [hC] : the pnly phones with local echo I've ever heard of were GrandSucks |
| 01:16.07 | [hC] | [TK]D-Fender: I didnt think echo was possible w/ data |
| 01:16.12 | shy_guy | _DAW:so what would insecure imply ?? |
| 01:16.31 | _DAW | check the wiki..... insecure=very ; To allow registered hosts to call without re-authenticating |
| 01:16.58 | [TK]D-Fender | [hC] : No, there were a couple of nasty GXP-2000 firmware releases that caused some truely unfortunate echo (2nd hand info) |
| 01:17.02 | shy_guy | _DAW:i read the whole wiki, but dude u put it very clear.thxs :) |
| 01:17.09 | [hC] | Hmmmmmmmmmmmm. i just reloaded the wanrouter module and got a kernel paging error! on a dell! |
| 01:17.23 | [hC] | [TK]D-Fender: I thought that was acoustic echo that it picked up. |
| 01:18.00 | JT | [hC]: echo is possible with data |
| 01:18.05 | JT | becase humans don't speak digital |
| 01:18.12 | JT | echo can be caused in the handpiece |
| 01:21.56 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
| 01:24.59 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:25.48 | tengulre | HI,all |
| 01:28.02 | shy_guy | _DAW:insecure=invite, do not allow auth of incoming INVITES |
| 01:29.07 | shy_guy | does that mean ALL the invites received from this registered host would never be authenticated. isnt that equivalent to that registered host calling us (INVITE are calls anyway) |
| 01:30.31 | shy_guy | plz answer anyone? gotta remove this confusion |
| 01:32.06 | *** join/#asterisk lule (i=lule@host233.201-252-36.telecom.net.ar) |
| 01:32.32 | [hC] | JT: thats not digital, now is it? thats acoustic echo |
| 01:32.40 | *** part/#asterisk lule (i=lule@host233.201-252-36.telecom.net.ar) |
| 01:32.45 | JT | yes, but it can cause echo |
| 01:33.54 | [hC] | i mean like, problems in your LAN can not CAUSE it, its acoustic feedback |
| 01:34.30 | JT | they can exacerbate a pre-existing accoustic echo problem |
| 01:39.14 | [hC] | yes, sure, but as i said... again... data networks can not -CAUSE- echo. |
| 01:39.36 | tessier | My ethernet cable was too long causing my voice to echo from one end to the other. |
| 01:40.16 | JT | [hC]: i guess not, unless there is something really screwy in the network |
| 01:40.17 | Un1x | anyone have some kinda bash or some script to back up Asterisk configs like everyweek and put it into a tarball and name it Astback-date.tar.gz or .tgz wichever you like |
| 01:40.20 | *** join/#asterisk requin (n=youssef@dyn10.tcom.ou.edu) |
| 01:40.21 | JT | tessier: uhh, okay? |
| 01:40.39 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 01:40.45 | [hC] | JT: well, that is what im trying to figure out, wether or not it is "possible" - which i dont believe it is. |
| 01:40.59 | [hC] | echo, from what I have read, and as i understand, is an acoustic side-effect |
| 01:41.22 | [hC] | unless you have some sort of .... delayed packet re-injection... |
| 01:41.44 | [hC] | What I also wonder is if you are on a latent link, if sidetone to the server can cause echo, or if sidetone is 100% handeld by the telephone itself |
| 01:41.52 | [hC] | (sidetone = hearing your own voice in the speaker, after you speak into the mic) |
| 01:41.58 | [hC] | normal |
| 01:42.12 | tessier | On purely digital links echo should not be possible unless it happens in the analog components such as the handset. |
| 01:42.22 | JT | yeah, i'm aware of sidetone... well it's possible to cause echo in software |
| 01:42.28 | JT | so buggy software could cause it |
| 01:42.29 | *** join/#asterisk coolhp (n=Toshiba@modemcable239.6-80-70.mc.videotron.ca) |
| 01:42.32 | tessier | But when you go digital to analog such as through a POTS line you can have echo. |
| 01:43.00 | [hC] | absolutely |
| 01:43.44 | tengulre | how to using TAPI to communicate to asterisk server? |
| 01:44.03 | sahafeez | if i have something in sip.conf marked as nat=yes but sip show peers marks it as N .. what am i missing |
| 01:44.38 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 01:45.56 | [hC] | sahafeez: N means "NAT" |
| 01:46.01 | [hC] | not "NO" |
| 01:46.04 | sahafeez | ah, ok. |
| 01:46.05 | [hC] | misleading. |
| 01:46.10 | sahafeez | yes. thank you. |
| 01:46.14 | [hC] | np. |
| 01:47.27 | sahafeez | next question - if i had a call from a sip provider - inbound to asterisk the call goes thru but i cannot hear the caller but the caller can hear me...what should i look for? it is a nat'd connection however i move the box outside of nat and tried it before asking about this to rule out that nat as an issue |
| 01:48.35 | *** join/#asterisk andresmujica (n=AndresMu@201.245.238.140) |
| 01:50.27 | JT | it sounds exactly like a nat issue |
| 01:51.04 | sahafeez | ok, let me move the box outside of the nat again and try. bbr. |
| 01:51.15 | [hC] | have you tried setting externip and localnet? |
| 01:51.27 | JT | lol |
| 01:51.30 | JT | patient |
| 01:53.48 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 01:54.56 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 01:55.40 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:56.50 | TripleFFFF | how to replicate a dealock... |
| 01:56.50 | TripleFFFF | use realtime.. sned a call to provider 1 that you know will come back o your switch.. ecxample one of your numbers that you would route externaly.. YOU -> provider -> YOU -> sip client .... if you send to sip/12345 and there no entry for sip/12345.. tyou get " channel.c:787 channel_find_locked: Avoided initial deadlock for '0x97b1170', 10 retri" |
| 01:58.45 | sahafeez | ok, not nat'd or firewalled at all. inbound sip call to asterisk - the caller can hear me but i cannot hear them? |
| 01:59.31 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
| 02:01.04 | sahafeez | thoughts anyone? |
| 02:02.06 | [TK]D-Fender | sahafeez : Disable reinvites between all your clients |
| 02:02.13 | sahafeez | ok. one sec. |
| 02:04.25 | sahafeez | that worked. thank you. can you tell me why now? |
| 02:04.34 | andresmujica | hi there! |
| 02:04.44 | andresmujica | anyone knows if digium cards support PCI-X slots? |
| 02:05.23 | bsdfreak | bif |
| 02:05.56 | [hC] | sahafeez: are any of the sip clients themselves natted or firewalled? (aside from the server) |
| 02:05.58 | Qwell | andresmujica: The cards should work okay in a PCI-X slot. It's just an enhanced version of PCI, which is generally backward compat |
| 02:06.32 | andresmujica | ahh.. that's what i thought!!! |
| 02:06.50 | andresmujica | where can i find more specs about it? |
| 02:06.58 | [hC] | andresmujica: http://www.google.com |
| 02:07.06 | sahafeez | at this point no, i moved everything outside the firewall. since i was having so much trouble getting it working. the reivites fixed it. now i am going to try to put it back behind the firewall and see if i can get it working. |
| 02:11.11 | sahafeez | thank you everyone. i have it working thru nat thru the firewall and am very happy. |
| 02:11.11 | [hC] | sahafeez: I have turned reinvites off for ever. the idea is nice but it never seems to work right. |
| 02:11.57 | JT | you can only guarantee it will probably work if you have control of all the networks involved |
| 02:12.07 | JT | like a lan |
| 02:12.09 | JT | etc |
| 02:12.20 | [TK]D-Fender | sahafeez : the remote side has no way to initiate the RTP inbound to your inside phone and it would likely not be able to know that it needs to forge the SIP return address anyways so communications are bound to get screwed up. re-invites = evil |
| 02:13.13 | Ryanw | reinvites + nat = evil. reinvites work fine on public ip addresses. |
| 02:15.23 | Marshall16 | bah i am in need of a IAX provider so i can call from name and number any ideas? |
| 02:16.05 | sahafeez | [hC]: thanks, i never had them on until someone here told me to put them on. this was a pain because i am not a sip guy and my sip provider was a bit on the clueless side, when i moved the firewall outside of the firewall to remove any issues there it turns out that my upstream, XO, had an routing issue to GX (RTP). it took a few days to settle that and an email to XO's NOC suggesting they get a copy of TCP/IP Illustrated Vol. 1 |
| 02:16.25 | sahafeez | move the pbx outside of the firewall that is.. |
| 02:16.33 | [hC] | haha. |
| 02:16.39 | javar | Marshall16, what do you think teliax.com |
| 02:16.59 | javar | they are IAX provider |
| 02:18.48 | Un1x | javer you wont understand his problem |
| 02:18.49 | Un1x | lol |
| 02:19.35 | [TK]D-Fender | Talks does dunny Yoda, hmmmMMMMMMMM?!?!!!?! |
| 02:20.16 | Un1x | now i need a script that emails my tarballs |
| 02:20.58 | Un1x | im thinking a perl bot |
| 02:21.19 | Un1x | iono gotta think of something fast :P |
| 02:23.39 | *** part/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 02:23.54 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 02:25.53 | Marshall16 | Un1x |
| 02:26.49 | Marshall16 | do i know you? |
| 02:27.21 | Un1x | Marshall is that supposed to be a joke |
| 02:27.54 | Un1x | i got it a perl script envoked with crontab |
| 02:28.05 | Un1x | no but thats risky my configs flying over email |
| 02:28.09 | Un1x | hmm back to thinking |
| 02:28.41 | Marshall16 | no? |
| 02:28.45 | Marshall16 | do i know you from somewhere? |
| 02:29.21 | Un1x | umm you think about it and tell me |
| 02:29.41 | Un1x | anyway brb |
| 02:31.59 | [TK]D-Fender | Un1x : Its a question about where you want to send things to. If you want something a fari bit more secure, try using SCP to copy them over to another box. |
| 02:32.56 | *** join/#asterisk lord-fu (n=lord-fu@cpe-65-24-185-26.columbus.res.rr.com) |
| 02:36.52 | *** join/#asterisk lord-fu (n=lord-fu@cpe-65-24-185-26.columbus.res.rr.com) |
| 02:40.38 | *** join/#asterisk kimo_sabe (i=nick@zappa.azrackspace.net) |
| 02:50.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 02:50.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:53.29 | *** join/#asterisk lule (i=lule@host233.201-252-36.telecom.net.ar) |
| 02:59.06 | Un1x | Hmm |
| 02:59.36 | Un1x | im thinking of a cd-rom small mini ones i got a million |
| 02:59.55 | Un1x | since im doing backup once weekly i could have it jkust write that along with my other bakcups into a cd |
| 03:00.02 | Un1x | where once written cant be fondled with |
| 03:01.02 | Un1x | man i found these awesome cooling method i never knew before better then watercooling |
| 03:01.07 | Un1x | keeps ur cpu at 0c all the time |
| 03:01.30 | Un1x | well i found the technology for it but never knew about it before in other words im saying i didn't create it :p |
| 03:01.47 | tengulre | Un1x: do you know the asterisk supported the Microsoft TAPI? |
| 03:02.09 | Un1x | tengulre sorry i dont understand want to rephrase |
| 03:02.11 | JT | i assume you're talking about peltier coolers? |
| 03:02.16 | Un1x | JT yea |
| 03:02.33 | JT | they do not keep your cpu at 0degC all the time |
| 03:02.49 | Un1x | hmm well it apperently freezes the thing if left without a load covered in ice |
| 03:02.49 | JT | they have a max temperature delta and an operating range |
| 03:03.12 | JT | if the change they need to make is too great, they don't be able to keep it that cool |
| 03:03.12 | Un1x | so what are you specificly saying that if server gets too high of a temp on the cpu it will stop working? |
| 03:03.24 | JT | or ambient temp |
| 03:03.27 | tengulre | JT: hi! NICT TO MEET U!! |
| 03:03.38 | JT | it will work, just may not be able to maintain 0degC |
| 03:03.38 | Un1x | you no u |
| 03:03.49 | Un1x | well it works better then a fan and water cooling i hear |
| 03:03.55 | JT | tengulre: hi, can't say we've met in person though :P |
| 03:03.59 | JT | yes it does |
| 03:03.59 | Un1x | since they keep it at a fluctating 30ish degrees |
| 03:04.03 | JT | although it pulls power |
| 03:04.10 | tengulre | JT: haha.. |
| 03:04.11 | Un1x | Yea i got a 1kw psu :) |
| 03:04.23 | tengulre | JT: I want ask you some question. ;) |
| 03:04.42 | Un1x | JT do you happen to know any other cooling metods except the peltier? |
| 03:05.18 | tengulre | JT: which linux are you using for asterisk? |
| 03:05.29 | JT | tengulre: debian |
| 03:05.34 | JT | Un1x: refrigeration |
| 03:05.38 | JT | nitrogen |
| 03:05.43 | JT | liquid helium |
| 03:06.09 | Un1x | i was thinking liquid nitro |
| 03:06.16 | Un1x | i heard people doing it |
| 03:06.17 | tengulre | JT: I m too. but the asterisk cost many memory in my machine, do you know why? |
| 03:06.22 | [TK]D-Fender | Sodium tri-iodide |
| 03:06.26 | *** part/#asterisk lule (i=lule@host233.201-252-36.telecom.net.ar) |
| 03:06.29 | Un1x | have you tried it? |
| 03:06.50 | JT | Un1x: no, my computers work perfectly fine with their standard fans |
| 03:07.00 | JT | tengulre: dunno, mine doesn't use much |
| 03:07.11 | Un1x | no that was ment for d-fender sorry |
| 03:07.18 | Un1x | [TK]D-Fender have you tried it? |
| 03:07.20 | *** join/#asterisk SplasPood (n=jwb@ip-160-79-255-7.autorev.intellispace.net) |
| 03:08.08 | tengulre | which one is author of asterisk ,http://www.markocam.com/? |
| 03:08.31 | [TK]D-Fender | <- Iron Chef of the Anarchist's Cook-book |
| 03:08.35 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 03:10.10 | Qwell | tengulre: grey hoodie |
| 03:11.01 | directory | oh no, people in the office! |
| 03:11.09 | tengulre | Qwell: hehe..., does he in here? |
| 03:11.14 | Qwell | tengulre: no |
| 03:11.17 | Qwell | directory: I know..scary |
| 03:11.23 | directory | Qwell: why aren't YOU there |
| 03:11.31 | Qwell | directory: because I went home :p |
| 03:11.37 | directory | lame |
| 03:12.18 | tengulre | They are watching film...? |
| 03:18.22 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:20.34 | shmaltz | realy? |
| 03:22.26 | Qwell | shmaltz: yep |
| 03:22.39 | shmaltz | thats so interestin |
| 03:22.58 | shmaltz | anybody here in centeral south jersey interested in a job? |
| 03:24.59 | *** join/#asterisk [hC] (n=hardcore@66.119.176.118) |
| 03:25.08 | [hC] | Hmm. anyone installed vicidial? |
| 03:25.25 | [hC] | Im curious how long it takes to get it up and to a working point, to try to decide wether or not to go thru with a demo of it for someone tomorrow |
| 03:28.06 | *** join/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 03:28.13 | shmaltz | [hC], my understanding is that if you have never done this before, that you are too late now :P |
| 03:28.24 | [hC] | really. shit. |
| 03:28.25 | [hC] | :P |
| 03:28.39 | [hC] | i am by no means a noob to asterisk, or linux, or lamp, etc... but.. |
| 03:29.09 | shmaltz | [hC], this is quite complicated though, you still have to be able to use it as well |
| 03:29.31 | [hC] | shmaltz: exactly. I think the big thing here is tying it into some sort of customer database to have a clear usable demo |
| 03:33.10 | *** join/#asterisk LakeSolon (n=blake@64-83-227-227.dhcp.stcd.mn.charter.com) |
| 03:42.20 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 03:43.25 | *** join/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 03:43.39 | *** part/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 03:46.46 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 03:52.37 | *** join/#asterisk bmg505 (n=leon@c1-205-4.rndf.isadsl.co.za) |
| 04:08.11 | *** join/#asterisk Krooks (n=kvirc@202.184.116.212) |
| 04:08.16 | *** join/#asterisk orlock (i=[8B0sy60@202.44.174.4.static.nexnet.net.au) |
| 04:08.16 | Krooks | hi |
| 04:08.20 | orlock | Does anybody have advice on how to debug/diagnose the causes of sip/rtp call quality? |
| 04:08.22 | orlock | packets are in order, low latency, low jitter.. |
| 04:09.23 | Krooks | Do I need any extra hardware to use asterisk besides my PC ? |
| 04:09.29 | *** join/#asterisk SwK (n=Silik0nJ@c-71-199-179-121.hsd1.ga.comcast.net) |
| 04:09.37 | JT | Krooks: depends what you want to do with it |
| 04:17.11 | *** join/#asterisk benjk_ (n=benjamin@aa2005080232001.userreverse.dion.ne.jp) |
| 04:17.21 | *** join/#asterisk SwK (n=Silik0nJ@c-71-199-179-121.hsd1.ga.comcast.net) |
| 04:17.53 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 04:20.02 | Krooks | well for the time-being I just wanna play with it. What can I do with only a PC and an ethernet card ? |
| 04:20.38 | Krooks | Hmm, where do I jack in the RJ11 ? Do I need a modem ? |
| 04:25.10 | JT | Krooks: with a pc and ethernet card you can use soft phones or sip hard phones, connect to a VoIP provider over the Internet |
| 04:25.27 | JT | if you want to connect a real phone line you'll need hardware with an FXO interface |
| 04:29.12 | orlock | anybody? audio quality help? |
| 04:31.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:44.52 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 04:50.51 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:53.56 | brookshire | !! |
| 04:57.11 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 04:59.42 | *** join/#asterisk kronic (n=user@mail.stabat.com) |
| 05:04.36 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:09.03 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:09.21 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 05:14.32 | *** join/#asterisk dome_c (n=dome_c@203.156.4.37) |
| 05:15.04 | *** join/#asterisk saabo (n=saab_va@user-0c8hjld.cable.mindspring.com) |
| 05:15.17 | dome_c | hello Thailand here :) |
| 05:15.27 | saabo | hi thai |
| 05:15.31 | CunningPike | Hello Thailand there |
| 05:16.00 | saabo | hmmm |
| 05:16.06 | dome_c | Need some help about asterisk-gui |
| 05:16.14 | saabo | i was hopin i would come in and learn a few things |
| 05:16.23 | dome_c | I can't add provider |
| 05:16.30 | dome_c | Can someone help me ? |
| 05:16.30 | saabo | <---- does ... a whole lot |
| 05:16.31 | saabo | lol |
| 05:17.14 | saabo | their all sleep |
| 05:17.23 | saabo | :( |
| 05:17.27 | dome_c | i see 12.17 here :) |
| 05:17.52 | saabo | same here |
| 05:18.08 | dome_c | where you from saabo? |
| 05:18.26 | saabo | i using fedroa for the first time |
| 05:18.29 | saabo | NC |
| 05:18.42 | saabo | ant linux based OS for that matter |
| 05:18.49 | saabo | any^ |
| 05:19.05 | dome_c | i like ubuntu and debian |
| 05:19.21 | saabo | i have no clue on what i'm doin ... but i'm makin my way so far |
| 05:19.38 | saabo | fedora was the first i came across |
| 05:21.05 | saabo | i did d/l a morpix with debian on but it was a live copy |
| 05:21.16 | *** join/#asterisk aadilismail (n=adilisma@202.38.55.114) |
| 05:21.24 | aadilismail | hi |
| 05:21.36 | dome_c | hi |
| 05:21.39 | saabo | hi aad .... room is dead |
| 05:22.34 | aadilismail | let them nudje |
| 05:23.14 | saabo | go for it |
| 05:23.25 | aadilismail | ok |
| 05:23.37 | saabo | hey dome |
| 05:23.52 | dome_c | hi |
| 05:23.52 | saabo | you have asterisk workin? |
| 05:23.56 | dome_c | yes |
| 05:24.04 | saabo | i gotta a ? |
| 05:24.06 | aadilismail | nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje nudje |
| 05:24.25 | aadilismail | new to askerisk |
| 05:24.25 | saabo | do i have to install all 5 disks to get it up n goin? |
| 05:24.33 | wunderkin | nudie? |
| 05:24.54 | dome_c | what's fedora version ? |
| 05:24.58 | saabo | 6 |
| 05:25.19 | aadilismail | workin redhat9 |
| 05:25.27 | dome_c | http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation |
| 05:25.31 | dome_c | can help you |
| 05:25.40 | aadilismail | i have been installed |
| 05:26.08 | aadilismail | i need to know ... about context... how many context we mention ? |
| 05:26.32 | aadilismail | in vi sip.conf... how many context can mention? |
| 05:26.37 | saabo | thankx dome |
| 05:26.41 | dome_c | i don't know but i think limit on text file size |
| 05:27.01 | dome_c | why you need more context ? |
| 05:28.58 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 05:29.03 | hads | ~book |
| 05:29.08 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:29.17 | aadilismail | i just want to dial digit like 431234xnxn and 44208xnxnxn .. through user agent.. and want both numbers different reply from asterisk like congestion , busy ,, hold etc??? |
| 05:29.54 | topping | is there a trick to getting a tdm2400 FXO port to listen for CallerID? |
| 05:29.56 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 05:30.12 | topping | it just picks up the phone as soon as the first ring arrives, missing CID |
| 05:31.22 | CunningPike | topping: Wait(2) |
| 05:31.39 | topping | CunningPike: presumably before the Answer, yah? |
| 05:31.56 | CunningPike | topping: Aye |
| 05:32.57 | topping | hmm, worked better that time, thanks! |
| 05:35.17 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:35.19 | saabo | anyone in here use fore 6? |
| 05:35.32 | saabo | core 6 that is |
| 05:35.37 | saabo | fedora |
| 05:36.39 | CunningPike | topping: Ytw |
| 05:36.51 | CunningPike | saabo: Nope - RHEL |
| 05:37.13 | JT | topping: did it pickup CID? |
| 05:37.21 | topping | yup |
| 05:37.30 | topping | just trying to log it now |
| 05:37.42 | topping | sugoi! |
| 05:37.48 | *** join/#asterisk alib80 (i=hidden-u@tsjb01.motorite.co.za) |
| 05:37.52 | *** part/#asterisk dome_c (n=dome_c@203.156.4.37) |
| 05:38.01 | alib80 | hi all |
| 05:38.29 | alib80 | is everyone asleep still? |
| 05:38.31 | alib80 | :) |
| 05:39.29 | alib80 | so here's my amazing problem |
| 05:39.38 | alib80 | i may be talking to myself |
| 05:39.45 | alib80 | but i will give it a try |
| 05:40.11 | alib80 | how do you stop the zaptel drivers from automatically loading on boot? |
| 05:42.54 | CunningPike | alib80: Remove it from your init.d? |
| 05:43.10 | alib80 | i tried that |
| 05:43.30 | alib80 | there is soemthing that is auto loading |
| 05:43.30 | CunningPike | alib80: What distro? |
| 05:43.35 | alib80 | debian sarge |
| 05:43.44 | alib80 | intel dual xeon servers |
| 05:44.48 | alib80 | its very strange. when i try run my load scripts automatically vi cron job |
| 05:44.59 | alib80 | they load the modules in the wrong order |
| 05:45.18 | alib80 | when i run them by hand i.e. /etc/init.d/zaptel start |
| 05:45.24 | alib80 | they do them in the correct order |
| 05:45.49 | alib80 | and on boot it just does its own thing even though |
| 05:45.59 | alib80 | i disable those scripts |
| 05:46.08 | JT | alib80: /etc/modules? |
| 05:46.15 | alib80 | i checked that |
| 05:46.29 | alib80 | but didn't find anything pertaining to zaptel |
| 05:46.37 | alib80 | i may be looking for the wrong |
| 05:46.41 | alib80 | thing though |
| 05:46.47 | alib80 | any ideas?? |
| 05:46.57 | JT | <PROTECTED> |
| 05:47.20 | alib80 | let me check quick;y |
| 05:47.35 | alib80 | aah |
| 05:47.36 | alib80 | yes |
| 05:47.50 | alib80 | there is a zaptel file |
| 05:47.52 | alib80 | here |
| 05:47.58 | alib80 | thanks for the help |
| 05:48.00 | alib80 | :) |
| 05:48.01 | JT | np |
| 05:48.29 | alib80 | do i just need to remove those files |
| 05:48.49 | alib80 | i don't have to also change some entry in a file somewhere? |
| 05:51.13 | hads | udev |
| 05:52.42 | *** join/#asterisk [shodan] (n=shodan@ip089.96-113-216.pppoe1.joliette.intermonde.net) |
| 05:53.15 | alib80 | Hi hads |
| 05:53.36 | alib80 | how do i do this? |
| 05:54.23 | [shodan] | emm , how do I insert a pause in a dial string ? I tried adding a "w" but it doesn't work , I'm using a SPA-3002 as a fxo .. |
| 06:04.00 | CunningPike | [shodan]: 'w' works for me.......... - show us your Dial() statement |
| 06:06.19 | *** join/#asterisk kilobit2001 (n=locid@206-248-129-91.dsl.teksavvy.com) |
| 06:06.23 | kilobit2001 | hello |
| 06:07.25 | [shodan] | it's simply dial(sip/fxo1/4505551111wwwwwww5551111) but it just ignores the "w" |
| 06:16.41 | saabo | do i need to have an account befor i set aterisk up? |
| 06:16.53 | saabo | like at egika? |
| 06:18.34 | CunningPike | [shodan]: Bet the other end answers after the the first set of numbers is dialed....... |
| 06:19.15 | [shodan] | yes , that is why I want to insert a pause |
| 06:19.18 | JT | saabo: only if you want to use such an account? |
| 06:19.36 | CunningPike | [shodan]: Try either WaitExten() or the 'D' option to Dial() |
| 06:20.03 | saabo | well the instucktions are askin me to enter my info |
| 06:20.19 | JT | are you using asterisk? |
| 06:20.29 | JT | if so, what instructions are telling you this? |
| 06:20.33 | saabo | what other otions do i have? |
| 06:20.59 | saabo | from the link that dome gave me |
| 06:21.01 | CunningPike | [shodan]: Actually, the 'D' option for Dial() is your best bet |
| 06:21.03 | [shodan] | JT , like so ? dial(sip/fxo1/4505551111,D(wwwwwwww5551111)) ? |
| 06:21.12 | [shodan] | oops I mean CunningPike |
| 06:22.06 | JT | saabo: you don't have to follow example instructions step to step |
| 06:22.11 | CunningPike | [shodan]: Kinda - Dial(SIP/fxo1/450555111,,D(wwwwwwwww55511111)) - you need the two commas for the timeout |
| 06:22.46 | saabo | ohhh ok ... well as you can see i'm new to this so i wanted to do it right |
| 06:22.59 | saabo | so what should i do? |
| 06:23.31 | JT | you should be able to get a free sip account at some place |
| 06:23.33 | JT | like FWD |
| 06:24.17 | saabo | ekiga is not free? |
| 06:25.41 | JT | i dunno if it is, haven't dealt with it |
| 06:25.43 | JT | it might be |
| 06:30.01 | [shodan] | hmm the behavior of dial with the D flag is very different ! |
| 06:30.46 | [shodan] | without it , I get silence , then the remote line rings , but with D it rings instantaneously |
| 06:31.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:32.06 | [shodan] | but it doesn't work, I listen to it dialing on the line , the first part of the number is clear dtmf , then it waits wwwwwww, then it sends the extra dtmf but all the digits are heavily distorted |
| 06:32.20 | [shodan] | the remote box doesn't even recognize them as dtmf |
| 06:32.44 | [shodan] | it's just a loud sharp noise instead of dtmf |
| 06:32.50 | saabo | hey jt? |
| 06:33.02 | [shodan] | (I'm using ulaw fxs -> * -> fxo(SPA3k2)) |
| 06:33.18 | JT | hey? |
| 06:33.46 | saabo | read this and tell me what you take from it please ............- In iax.conf, make the changes outlined in the QUICKSTART section of the sample file, save the file and close it. |
| 06:33.46 | saabo | - In extensions.conf make the changes outlined in the QUICKSTART section of the sample file, save the file and close it. |
| 06:34.22 | saabo | i'm not sure what to change in there |
| 06:34.42 | CunningPike | [shodan]: Try using Inband DTMF for your FXO |
| 06:38.10 | [shodan] | how do I tell * to do that when using the D flags with dial ? or do you mean just to try to enter dtmf signal with my analog phone it this chain analog phone -> fxs -> * fxo -> pstn (the dtmf tones aren't clear but probably understandable for a machine) |
| 06:38.29 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 06:38.58 | shellshark | titaniumsoft.... i like that name |
| 06:39.56 | mitcheloc | thanks, |
| 06:40.18 | mitcheloc | it's a pun |
| 06:40.32 | JT | "titanium" has been a marketer's wet dream recently |
| 06:40.43 | JT | products are "titanium" coloured |
| 06:40.48 | JT | even when they contain none |
| 06:41.12 | saabo | jt: couldn't make out what that could be |
| 06:41.39 | saabo | ? |
| 06:41.41 | JT | saabo: not sure |
| 06:41.44 | saabo | ok |
| 06:47.29 | [shodan] | hmm , I set "DTMF Tx Method:" on the SPA to inband but same thing , instead of deee douuu daaa , I get *chirp* *chirp* *chirp* |
| 06:47.55 | [shodan] | I think the dtmf * is generating are just too short |
| 06:49.12 | [shodan] | hmm , well at least now the dtmf I send from my analog phone through analog -> fxs -> * -> fxo -> pstn , are a lot cleaner , I probably won't be getting more misread numbers from remote pbxs |
| 06:55.59 | CunningPike | [shodan]: I think you can set the dtmf length somewhere....... |
| 06:56.12 | CunningPike | Anyway, I'm off to bed |
| 06:58.24 | crosslimits | NEW BORN... need IVR response.. need to dial any digit like 91XNXNXN by user agent and get any IVR response like congestion, busy , ? |
| 07:04.07 | crosslimits | hi |
| 07:07.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 07:07.04 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 07:07.05 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) [NETSPLIT VICTIM] |
| 07:07.05 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) [NETSPLIT VICTIM] |
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| 07:07.05 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 07:07.13 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-70-128-250-253.dsl.ksc2mo.swbell.net) |
| 07:08.23 | *** join/#asterisk benjk (n=benjamin@aa2005080232001.userreverse.dion.ne.jp) [NETSPLIT VICTIM] |
| 07:09.49 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 07:12.05 | *** join/#asterisk [shodan] (n=shodan@ip089.96-113-216.pppoe1.joliette.intermonde.net) [NETSPLIT VICTIM] |
| 07:21.48 | crosslimits | new born ... can u tell me ... i need IVR response ...exten => 200,2,Dial_915XXNN(SIP/0.0.0.0:5060_915XXNN) where is the problem ??? |
| 07:23.14 | kaldemar | your syntax is weird. are you using asterisk? |
| 07:25.35 | JT | crosslimits: a new born is a baby that has just been born |
| 07:26.20 | crosslimits | yup |
| 07:26.33 | crosslimits | new born to asterisk |
| 07:27.48 | crosslimits | kaldemar:so can u plz correct this . syntax? |
| 07:29.01 | crosslimits | need IVR response ...using xpro |
| 07:29.09 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 07:29.11 | kaldemar | what exactly do you want to do? |
| 07:29.24 | JT | crosslimits: totally incorrect usage of the term |
| 07:29.34 | JT | i think you mean "newbie" |
| 07:29.36 | kaldemar | call some number if the user dials 200? |
| 07:31.39 | crosslimits | want to dial some digits like 44208XNXNXXX by using xpro.. and want IVR ... like user busy ... congestion... etc... ? |
| 07:33.00 | crosslimits | like 1st is .... "exten => 200,1,Answer" wat would be 2nd? |
| 07:33.45 | kaldemar | exten => 44208XNXNXXX,1,Dial(SIP/${EXTEN}@x.x.x.x) <-- that's how you would call that number with SIP |
| 07:33.58 | crosslimits | ok |
| 07:34.03 | crosslimits | thank u |
| 07:34.28 | kaldemar | but take a good look at |
| 07:34.29 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 07:34.31 | kaldemar | and |
| 07:34.37 | crosslimits | ok |
| 07:34.47 | kaldemar | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 07:36.13 | *** join/#asterisk apardo (n=apardo@87.217.147.18) |
| 07:40.07 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 07:44.03 | *** join/#asterisk saabo (n=saab_va@user-0c8hjld.cable.mindspring.com) |
| 07:48.02 | [shodan] | when using the D flag or dial to send extra dtmf tones , is there a way to control how those tones are generated (gain/lenght/interdigit delay) or the method (inband or SIP INFO msg) ? |
| 07:51.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:52.44 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:56.46 | *** join/#asterisk voodoonofx (n=CheeseHe@pool-71-104-61-26.lsanca.dsl-w.verizon.net) |
| 08:09.55 | *** part/#asterisk saabo (n=saab_va@user-0c8hjld.cable.mindspring.com) |
| 08:11.37 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 08:14.01 | *** join/#asterisk Alexphil (n=chatzill@daemon.aktel.kg) |
| 08:14.58 | Alexphil | Hi all! http://bugs.digium.com/view.php?id=8266 it's my trouble... |
| 08:16.44 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 08:18.11 | *** join/#asterisk neoalex (n=neoalex@user-12ldt1n.cable.mindspring.com) |
| 08:18.37 | neoalex | hello... is there any way to play a RealMedia stream in asterisk? |
| 08:26.29 | neoalex | actually is there anyway to get weather forecasts, without having my server do the text to speech |
| 08:26.47 | neoalex | basically any online weather stream would do |
| 08:31.38 | _Vile | hmm |
| 08:32.02 | _Vile | non tts ur no fun http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration |
| 08:37.02 | neoalex | :))... I know, but I'm not sure the machine I'm running it on would be able to handle it properly |
| 08:39.37 | _Vile | http://tools.ietf.org/html/rfc1 |
| 08:45.21 | Simplix | hello all |
| 08:45.51 | Simplix | can I store call forwarding in odbc ? |
| 08:50.25 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:51.29 | *** join/#asterisk Skymarshal (n=Skymarsc@195.141.78.162) |
| 08:51.55 | Skymarshal | Is there an easy way to start parts of the extensions.conf whenever asterisk starts? |
| 08:53.11 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 08:54.24 | NDT | What is a voip provider that truely has an unlimited outbound? The lines we would want would probably be up about 22 hours a day...any of these $25-$50 per account places have that? |
| 08:55.13 | _Vile | heh |
| 08:55.25 | _Vile | heh heh heh |
| 08:55.32 | SheriF_SpacE | NDT: check voip.shellshark.net |
| 08:55.49 | NDT | k... |
| 08:56.36 | NDT | heh...we all know that most places that say unlimited only mean about 650 min per month and make their profits off the law of averages heh...why I was curious |
| 08:57.19 | _Vile | well |
| 08:57.54 | _Vile | divide x minutes by .008, add in bandwidth costs.. |
| 08:58.02 | _Vile | *shrug* |
| 08:58.12 | _Vile | hitting bed night |
| 08:58.43 | NDT | night |
| 08:59.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:59.40 | *** join/#asterisk qdk (n=qdk@193.226.189.115) |
| 09:06.14 | dzh | Hi all, any ideas WHY loosing of packets happens on bridging between IAX and ZAP ? Asterisk 1.2. output from " iax2 show netstats |
| 09:06.14 | dzh | " |
| 09:06.21 | *** join/#asterisk svenna_ (n=svenna@p548D35CE.dip0.t-ipconnect.de) |
| 09:06.44 | dzh | Channel RTT Jit Del Lost % Drop OOO Kpkts |
| 09:06.53 | dzh | IAX2/gw054b-4 6 1 58 159 10 0 0 9 |
| 09:07.05 | *** join/#asterisk smig (n=hegemoni@sd-792.dedibox.fr) |
| 09:07.09 | smig | Hello |
| 09:08.02 | smig | Does anyone know why I get such an error: "Realtime mapping for 'sipusers' found to engine 'odbc', but the engine is not available" ? |
| 09:08.25 | smig | Any URL, advice, or help would be very appreciated! |
| 09:12.59 | Simplix | smig : your odbc connexion is misconfigured |
| 09:13.13 | smig | only this... ok |
| 09:14.09 | Simplix | smig : test "odbc show" in your CLI |
| 09:14.41 | smig | at startup, there is no error on this point, I set up the "pre-connect" variable to yes in res_odbc |
| 09:14.52 | smig | hu? ok, I will try |
| 09:15.34 | *** join/#asterisk prttp (i=Ftv@187.Red-81-44-146.dynamicIP.rima-tde.net) |
| 09:16.04 | dzh | guys - any ideas about "iax2 show netstat" ??? |
| 09:17.58 | *** join/#asterisk Skymarshal (n=Skymarsc@195.141.78.162) |
| 09:18.45 | Skymarshal | When ever my asterisk server starts I want to save some values in the asterisk database (it has to be a fresh set of values for every start). What is the easiest way for that? |
| 09:19.14 | *** join/#asterisk tsurk0 (n=tsurko@80.72.68.86) |
| 09:21.06 | *** part/#asterisk Alexphil (n=chatzill@daemon.aktel.kg) |
| 09:22.53 | X-Rob_ | ~centosbug |
| 09:22.55 | jbot | well, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 09:26.36 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:27.09 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:30.30 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:33.20 | SheriF_SpacE | any one has multitech devices as sip clients for asterisk server? |
| 15:15.13 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:15.13 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) Join #freepbx for freepbx/trixbox support. |
| 15:16.12 | *** join/#asterisk djMax (n=chatzill@66.92.91.134) |
| 15:16.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:17.08 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:17.12 | djMax | hi. I am trying to setup a failover for our fax. If Dial(Sip/number1) fails, I want to go to Dial(Sip/number2). I tried just adding the second dial next in the loop (and +101, and with j dial option) but nothing works. Pointers? |
| 15:19.20 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:19.44 | file | webmask12345: are you messaging all the ops, then working your way down? that is quite rude |
| 15:20.12 | djMax | (my simulated failure mode is unplugging the ATA) |
| 15:22.07 | webmask12345 | :) |
| 15:22.11 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:22.18 | *** join/#asterisk packetman (i=packetma@d141-9-96.home.cgocable.net) |
| 15:22.47 | packetman | Where can I report a serious asterisk bug I have discovered |
| 15:23.08 | file | bugs.digium.com is the bug tracker |
| 15:23.22 | file | although I would be curious to know what this serious asterisk bug is |
| 15:24.17 | packetman | K I will pastebin it |
| 15:25.11 | packetman | It may not be a bug, I might be doing something incorrectly, or it may be a hardware issue but maybe someone can test to see if it occurs on their hardware |
| 15:25.20 | packetman | Have a read and tell me what you think |
| 15:25.22 | packetman | http://pastebin.ca/234416 |
| 15:25.29 | djMax | so does Dial() go to the next line on failure? |
| 15:25.34 | djMax | I sure thought it did. |
| 15:26.11 | *** join/#asterisk prttp (i=Ftv@187.Red-81-44-146.dynamicIP.rima-tde.net) |
| 15:26.12 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 15:26.14 | *** join/#asterisk asdx|work (n=asdx@200.85.40.178) |
| 15:27.00 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 15:27.23 | file | if that was occuring for more people they would definitely make it known |
| 15:27.43 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:28.14 | packetman | true thats what I thought. |
| 15:28.27 | packetman | I'm I using the tT options in proper form? |
| 15:28.56 | packetman | Is there any real documentation on the tT options. I could not find much of anything |
| 15:29.18 | *** part/#asterisk webmask12345 (n=albert@125.212.19.184) |
| 15:31.45 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 15:32.45 | *** join/#asterisk Skarmeth (n=Skarmeth@201009073170.user.veloxzone.com.br) |
| 15:34.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 15:36.00 | *** join/#asterisk Nivex (n=kjotte@user-0ce2nsk.cable.mindspring.com) |
| 15:40.33 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 15:40.42 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
| 15:41.24 | GaVak | I've a question about the sip.conf file. I'm setting up a system using the Polycom 501 phones. So far, I have 5 people in the local office using the phones just fine. |
| 15:42.05 | GaVak | This morning, I was adding the settings externip= and localnet= ... after adding these settings, all of the local phones starting throwing registration errors. |
| 15:42.08 | *** join/#asterisk xezz (n=root@serial.trust-it.gr) |
| 15:42.44 | *** join/#asterisk Un1x (n=x@CPE006323108591-CM00407b87dc81.cpe.net.cable.rogers.com) |
| 15:44.05 | GaVak | I tried setting nat=never for those phones, but registration only started working after i commented out the externip setting. |
| 15:44.14 | GaVak | Is there a way to apply it to only certain phones? |
| 15:44.53 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:46.47 | *** join/#asterisk riksta (n=rick@89.240.134.21) |
| 15:50.26 | *** join/#asterisk seele_ (n=seele@208.35.117.195) |
| 15:53.07 | seele_ | how can I reduce the log debug level ??? |
| 15:53.42 | jeremy_g | seele_:set debug 3 |
| 15:54.42 | *** join/#asterisk onglipo (n=onglipo@125.22.81.2) |
| 15:55.23 | onglipo | what's up with jabber.org ? http://status.jabber.org/ says connections 0 - they've been down for some time |
| 15:58.28 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 15:58.40 | sevard | Hello friends. |
| 15:58.43 | *** join/#asterisk MooingLemur (n=troy@shells195.pinchaser.com) |
| 15:59.13 | seele_ | hello sevard |
| 15:59.44 | sevard | I'm currently using wine with the spc. Does anyone know of any reports of issues using spc with wine? As there isn't a linux version this is kind of bothersome. |
| 16:00.27 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 16:01.05 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 16:01.39 | *** join/#asterisk bmg505 (n=leon@c1-205-4.rndf.isadsl.co.za) |
| 16:02.59 | sevard | `DATE |
| 16:03.07 | sevard | grr. |
| 16:03.33 | *** join/#asterisk Brokennoze (n=Simon@82.109.134.10) |
| 16:04.22 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 16:04.25 | b11d | morning lads |
| 16:04.39 | sevard | mornin'. |
| 16:05.40 | Brokennoze | hello, badly need some help with a 12 second delay on outbound calls, anyone help |
| 16:07.45 | b11d | wow.. 12 second delay eh |
| 16:07.50 | b11d | that licks man |
| 16:08.20 | b11d | i'd help you but honestly, I dont have any experience troubleshooting that kind of issue.. |
| 16:08.41 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:10.36 | seele_ | I can update the asterisk without stopping the service ???? |
| 16:11.06 | Brokennoze | we go live in a 999 call centre |
| 16:11.14 | b11d | seele_.. you'd need to restart asterisk. |
| 16:11.20 | Brokennoze | if anyone has ANY ideas please help me out. I |
| 16:11.33 | b11d | Brokennoze.. when did this start happening? |
| 16:11.35 | Brokennoze | ve got 10 hours before go live!!!!!! |
| 16:11.46 | Brokennoze | 999 - or rather 911 |
| 16:11.46 | b11d | did you upgrade some equipment, software, change config? |
| 16:11.58 | b11d | 911.. fuck.. you're fucked man. :) |
| 16:12.04 | b11d | no one messes with 911 |
| 16:12.05 | b11d | :) |
| 16:12.14 | Brokennoze | thanks for that |
| 16:12.18 | Brokennoze | :-) |
| 16:12.22 | b11d | just trying to cheer you up |
| 16:12.24 | b11d | but seriously |
| 16:12.26 | b11d | did it work before? |
| 16:12.27 | Brokennoze | any ideas though, seriously |
| 16:12.31 | b11d | did it ever work? |
| 16:12.39 | Brokennoze | Inbound fine, outbound takes ages |
| 16:12.47 | b11d | and this is how its been since you started the project? |
| 16:12.51 | Brokennoze | no, not via the trunk |
| 16:12.54 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:13.01 | b11d | ok.. what did you do just before it started happening? |
| 16:13.09 | Brokennoze | yeah, we use a vega 400? heard of it? |
| 16:13.16 | b11d | no.. sorry.. i havent |
| 16:13.28 | Brokennoze | always happened |
| 16:13.34 | b11d | maybe you should call the people who make this "vega" or your trunk provider? |
| 16:13.50 | seele_ | b11d, ok but only asterisk not all the system |
| 16:14.01 | b11d | seele_.. yep |
| 16:14.15 | b11d | just compile the new asterisk.. stop the asterisk program.. back it up! and then install the new asterisk. |
| 16:14.24 | Brokennoze | have done. just hoping its a asterisk thing |
| 16:14.32 | b11d | what did they say about it Brokennoze? |
| 16:14.38 | b11d | that it must be asterisk, and not their stuff? |
| 16:15.30 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:15.35 | b11d | hello CunningPike.. |
| 16:15.42 | b11d | you always remind me of a stunning array of cunts |
| 16:15.44 | Brokennoze | just wondering if theres something silly in the dial plan |
| 16:15.46 | b11d | or that joke, that is |
| 16:15.54 | *** join/#asterisk saftsack (n=saftsack@pD9E065E3.dip.t-dialin.net) |
| 16:16.00 | b11d | Brokennoze.. well... i really am not sure if i can help, but lets take a look at it |
| 16:16.12 | b11d | can you post your dialplan somewhere? (be careful with this) |
| 16:16.17 | Brokennoze | its a standard _9X.,1,Dial(SIP/ etc) |
| 16:16.18 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:16.35 | b11d | well.. you wonder if its somethign funky with the dialplan, then you just say its simple one line stuff.. |
| 16:16.43 | b11d | we need to see the entire thing |
| 16:17.31 | *** join/#asterisk apardo (n=apardo@87.217.144.170) |
| 16:17.36 | Brokennoze | yeah, it is. so no config you can think of, standrad sip devices.. nowhere could have a delay switched on, differenct config for the gateway we need to change?? |
| 16:17.41 | Brokennoze | hang on |
| 16:17.58 | b11d | not that im aware of.. like I said in the beginning here.. i dont really know enough about this to really troubleshoot it effectively. |
| 16:18.08 | b11d | if I can see it, i might catch something.. but no guarantees. |
| 16:18.21 | b11d | there are others who come in here who are much more likely to be able to help you out |
| 16:18.24 | b11d | but its still early yet |
| 16:18.58 | Brokennoze | OK, thanks, I got a lead, gonna go see what I can find on the provider for now. |
| 16:19.06 | Brokennoze | Cheers guys , i'll be back |
| 16:19.18 | b11d | ok |
| 16:19.35 | b11d | best of luck |
| 16:19.45 | Brokennoze | cheers |
| 16:19.59 | seele_ | If I have asterisk 1.2.x and I want to install 1.4.x, can I use my old asterisk configuration? |
| 16:20.15 | Qwell[] | seele_: mostly. Be sure to read the CHANGES and UPGRADE.txt files |
| 16:20.26 | Qwell[] | (the latter is very important) |
| 16:20.46 | b11d | good advice |
| 16:21.24 | b11d | dammit.. i've been messing with "progress_ind" for like, 6 hours, on my vg-224 only to now learn those settings do not do anything with SIP. |
| 16:21.26 | b11d | that bothers me :) |
| 16:21.48 | b11d | i guess its time to bite it and try MGCP. |
| 16:21.53 | b11d | or SCCP. |
| 16:21.59 | Qwell[] | b11d: skinny... |
| 16:22.00 | b11d | in fact.. is there a difference between MGCP and SCCP? |
| 16:22.02 | Qwell[] | mgcp is... |
| 16:22.07 | b11d | oh ok |
| 16:22.27 | b11d | i dont know much about mgcp, and know even less about skinny. |
| 16:22.37 | Qwell[] | b11d: What phone? |
| 16:22.51 | b11d | im trying to connect some analog phones through a cisco vg-224. |
| 16:22.57 | b11d | to the * box |
| 16:23.00 | Qwell[] | ooo |
| 16:23.08 | b11d | it's about 70% working |
| 16:23.19 | Qwell[] | yeah, 10/1 says it won't work with chan_skinny or chan_sccp |
| 16:23.22 | b11d | but im suffering from one-way audio problems |
| 16:23.23 | Qwell[] | BUT |
| 16:23.44 | Qwell[] | If you were to try it with chan_skinny, and get me a little bit of debug output, I might be able to hack something up |
| 16:23.57 | sevard | Qwell[]: Do you use spc? |
| 16:24.03 | b11d | i appreciate that.. i'll let you know when i get to that point |
| 16:24.03 | Qwell[] | sevard: spc? |
| 16:24.11 | sevard | the sipura config compiler |
| 16:24.20 | Qwell[] | no |
| 16:24.28 | sevard | does anyone? |
| 16:24.37 | b11d | i thought SCCP and Skinny were the same things |
| 16:25.00 | Qwell[] | They are |
| 16:25.02 | Qwell[] | sorta |
| 16:25.06 | b11d | oh |
| 16:25.16 | *** join/#asterisk zamba (i=marius@flage.org) |
| 16:25.27 | b11d | well, i see some documentation on sccp with this vg, so im gonna try that first. |
| 16:26.13 | *** join/#asterisk Barnes (n=barnes@c193-150-214-238.bredband.comhem.se) |
| 16:26.17 | Qwell[] | SCCP is ambiguous |
| 16:26.41 | Qwell[] | SCCP can either be Skinny Client Control Protocol, or ...waiting for google |
| 16:26.47 | Qwell[] | some SS7 thing |
| 16:26.51 | b11d | yeah im there too :) |
| 16:27.05 | Qwell[] | Signaling Connection Control Part |
| 16:27.19 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:27.28 | CunningPike | Morning, b11d |
| 16:27.31 | b11d | morning |
| 16:27.36 | Qwell[] | a bug? rofl |
| 16:27.43 | b11d | quick.. we need to get airwolf in the air right now.. |
| 16:27.44 | CunningPike | Too early to think of a witty response :) |
| 16:27.46 | b11d | the goverment is coming |
| 16:27.50 | BurtyB | god its great to be on freenode again |
| 16:27.58 | b11d | yeah wtf was with it earlier eh? |
| 16:28.01 | b11d | i couldnt get on it at all |
| 16:28.11 | CunningPike | I was on the Milan node yesterday :) |
| 16:28.12 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:28.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:29.00 | BurtyB | b11d you forgot about blue thunder! |
| 16:29.02 | Qwell[] | b11d: so, yeah... When you try to register the device with asterisk, it very very likely will not work |
| 16:29.18 | b11d | well the vg224 is successfully "registering" |
| 16:29.21 | Qwell[] | but, if you turn on 'skinny debug', you can find the device number |
| 16:29.23 | b11d | sip show peers shows it listed |
| 16:29.27 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.24) |
| 16:29.29 | Qwell[] | b11d: I mean with chan_skinny |
| 16:29.33 | b11d | oh |
| 16:29.37 | b11d | :( |
| 16:29.44 | b11d | i'll keep hacking away at it.. |
| 16:29.57 | Qwell[] | if you get me the device number (it may already be defined), I can add it, and let you try it again, heh |
| 16:30.01 | b11d | i got a lower revision of an IOS for it, im running 12.4 and i got 12.3.. I might give it a shot |
| 16:30.27 | b11d | Qwell.. i think it wil be awhile.. i've never done anything with skinny before. |
| 16:30.34 | Qwell[] | b11d: yeah... |
| 16:30.50 | Qwell[] | b11d: feel free to donate one of those to Digium. I'll hack on it if I have one here. :P |
| 16:30.53 | b11d | its nice to have options & people to talk with though.. thanks |
| 16:31.05 | b11d | Qwell, i'd do that actually.. maybe in the future sometime |
| 16:31.21 | b11d | but then again, i might just as well never use these god damn satan-spawned beasts again :) |
| 16:31.23 | DarKnesS_WolF | what is VLDTMF ? i mean what dose it means and for what use ? i read in asterisk.org about that |
| 16:31.27 | Qwell[] | b11d: I know squat about cisco routers/gateways though, unfortunately |
| 16:31.29 | b11d | Very Loud DTMF |
| 16:31.38 | b11d | :) |
| 16:31.39 | b11d | im probably wrong |
| 16:31.48 | Qwell[] | DarKnesS_WolF: Variable Length DTMF. It means that you can hold the key down, and hear the tone until you let go |
| 16:32.01 | DarKnesS_WolF | Qwell[]: and what use of that ? |
| 16:32.10 | b11d | playing funky town |
| 16:32.12 | Qwell[] | DarKnesS_WolF: To follow the RTP specs. :) |
| 16:32.12 | b11d | on the phone |
| 16:32.18 | Qwell[] | b11d: nah, res_indications |
| 16:32.22 | DarKnesS_WolF | Qwell[]: cool :-) i love standers ;-) |
| 16:32.30 | Qwell[] | b11d: Which is quite funny that you bring that up... Somebody here actually did that |
| 16:32.38 | Qwell[] | and because of it, I added midi support to Playtones() :P |
| 16:32.43 | b11d | nice! |
| 16:32.57 | Qwell[] | I "composed" a couple of my own too :D |
| 16:32.59 | b11d | :) |
| 16:33.14 | DarKnesS_WolF | Qwell[]: another question .. i have some devices reigstering to asterisk but after like 1 or 2 days they become unregister and i have to restart them .. can i put options in asterisk like this devices registeration never expired ? |
| 16:33.14 | Qwell[] | tetris theme A, Mario bros underwater theme... some doors song |
| 16:33.22 | b11d | no way.. that must have taken a little time eh |
| 16:33.30 | Qwell[] | couple hours maybe |
| 16:33.38 | b11d | not bad |
| 16:33.38 | DarKnesS_WolF | Qwell[]: hehe cool this midi thing |
| 16:33.57 | Qwell[] | b11d: If you have an existing midi, you can fairly easily "port" it |
| 16:34.04 | b11d | oh really? |
| 16:34.05 | seele_ | I have this message in my logs ... ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 16:34.07 | Qwell[] | just need to know the tone "number" and the length |
| 16:34.10 | seele_ | any suggest |
| 16:34.12 | b11d | oh yeah |
| 16:34.35 | Qwell[] | midi has well defined tone numbers too |
| 16:34.35 | b11d | i'll hit that up when I get some free time :) |
| 16:34.35 | *** join/#asterisk p1p (i=j@mail.comp911.com) |
| 16:34.37 | b11d | i'll "port" the canadian national anthem |
| 16:34.42 | Qwell[] | b11d: Ask me later tonight, and I'll get you a sample |
| 16:34.45 | b11d | just to annoy my american counterparts |
| 16:34.54 | b11d | cool Qwell.. i'll do that |
| 16:34.55 | Qwell[] | b11d: one thing to note, is that you can only do 2 tones simultaneously |
| 16:34.56 | *** join/#asterisk adorah (n=admin@87.68.160.129.cable.012.net.il) |
| 16:35.16 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:36.32 | b11d | brb.. |
| 16:37.53 | b11d | there is a more fundamental issue with SIP and this vg224.. like.. while the analog phone actually rings, no ring is generated in the earpeice of the SIP phone im dialing from. |
| 16:38.13 | b11d | and the reverse is true as well.. when I call the SIP from the analog, I hear no ring on the analog, while the SIP rings. |
| 16:38.27 | DarKnesS_WolF | b11d: i think there is options to force asterisk to genrate dialtone |
| 16:38.31 | DarKnesS_WolF | check the Dial cmd |
| 16:38.34 | b11d | i dont need dialtone.. |
| 16:38.36 | b11d | i need a ring back. |
| 16:38.45 | DarKnesS_WolF | or ring back i remember i read something like this |
| 16:38.49 | b11d | i have dialtone. |
| 16:39.00 | b11d | i've already set "tone ringback alert-no-PI" in the dial-peer. |
| 16:39.11 | b11d | which is supposed to force a ringback if no Progress Indicator has been sent |
| 16:40.23 | b11d | yeah |
| 16:40.27 | b11d | you're right DarKnesS_WolF |
| 16:40.29 | b11d | the "r" flag.. |
| 16:40.30 | Qwell[] | It's a hack, but you COULD try Dial with r |
| 16:40.32 | b11d | im going to try that now |
| 16:40.38 | b11d | yeah it is a hack.. |
| 16:40.40 | Qwell[] | That does very bad things though |
| 16:40.45 | Qwell[] | like...mask busy signals and such |
| 16:40.48 | b11d | i'll try it quickly |
| 16:40.49 | b11d | oh really? |
| 16:40.50 | b11d | ugh |
| 16:40.57 | DarKnesS_WolF | b11d: sorry my connection is very slow i was trying to load voip-info |
| 16:41.35 | Qwell[] | vg == voice gateway? |
| 16:41.41 | b11d | yeah |
| 16:43.04 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 16:44.02 | DarKnesS_WolF | i'm having problems between 2 asterisk using IAX, |
| 16:44.11 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:44.22 | blitzrage | morning all -- anyone have a simple example of connecting to AMI via PHP? |
| 16:44.44 | DarKnesS_WolF | i'm testing the open source g729 codecs but the qualty is very bad :-s the voice very electraed and worried to ask my boss to buy codecs and it turn out the same bad results :D |
| 16:45.10 | b11d | you were right Qwell.. |
| 16:45.15 | Qwell[] | ? |
| 16:45.19 | b11d | it did force the ringback,but it wasnt good |
| 16:45.21 | Qwell[] | yeah |
| 16:45.25 | Qwell[] | it's a major hack |
| 16:45.27 | b11d | it is a mistake to use that |
| 16:45.28 | b11d | :) |
| 16:45.34 | blitzrage | DarKnesS_WolF: just buy one then... it's only $10 |
| 16:45.40 | DarKnesS_WolF | i tried tweaiking with tos / jittbuffer nothing seems useflul |
| 16:45.40 | b11d | why would audio only go one way though |
| 16:45.40 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 16:45.41 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 16:45.44 | b11d | from the SIP to the Analog |
| 16:45.47 | b11d | always works |
| 16:45.52 | b11d | I can NEVER get audio back from the Analog.. |
| 16:45.52 | Qwell[] | firewall? :D |
| 16:45.54 | b11d | no.. |
| 16:46.01 | Qwell[] | Where is the analog? |
| 16:46.04 | b11d | the vg and the pbx are on the same network, on the same switch. |
| 16:46.05 | DarKnesS_WolF | blitzrage: i'll soon but i think it will be same results i'll buy more than one anywya when i got to prodection |
| 16:46.14 | b11d | the vg is at 10.0.5.5 and the pbx is at 10.0.5.11 |
| 16:46.22 | b11d | no firewalls in place |
| 16:46.29 | Qwell[] | b11d: localnet=10.0.5.0/24 ? |
| 16:46.31 | blitzrage | DarKnesS_WolF: yah, but just buy one to test -- if its still bad, then its the network |
| 16:46.41 | b11d | yep |
| 16:47.02 | Qwell[] | analog is internal to the pbx? |
| 16:47.05 | Qwell[] | ie; a pci card or something? |
| 16:47.11 | DarKnesS_WolF | blitzrage: i have another device using g729 connected to same 2nd asterisl via sip and qualty is perfect " so i thin it's not that connection thing " |
| 16:47.12 | b11d | the analog is on the vg224. |
| 16:47.19 | Qwell[] | okay, so it's sip to sip |
| 16:47.24 | b11d | pretty much yep |
| 16:47.30 | Qwell[] | where is your sip phone? |
| 16:47.43 | Qwell[] | also 10.0.5.0/24? |
| 16:47.49 | seele_ | help my log file is filling of this ... chan_sip.c: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 16:48.07 | b11d | no |
| 16:48.07 | DarKnesS_WolF | seele_: sounds like codecs to me |
| 16:48.14 | Qwell[] | now we're getting somewhere ;) |
| 16:48.14 | b11d | the sip is on 10.3.237.0/24 |
| 16:48.23 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:48.25 | b11d | sip to sip works great otherwise |
| 16:48.29 | Qwell[] | and I assume asterisk doesn't have a 10.3 address? |
| 16:48.32 | DarKnesS_WolF | seele_: start with disallow=all allow=ulaw in ur sip.conf and make sure both ends using ulaw codecs |
| 16:48.33 | seele_ | DarKnesS_WolF, how can i solve this ? |
| 16:48.34 | b11d | yes, it does |
| 16:48.38 | b11d | there is NO NAT in the way |
| 16:48.43 | Qwell[] | well, there is |
| 16:48.46 | Qwell[] | between the endpoints |
| 16:49.16 | b11d | ok.. theres nothing commonly known as NAT between the devices.. |
| 16:49.17 | DarKnesS_WolF | yes |
| 16:49.28 | b11d | which is really PAT. |
| 16:49.33 | DarKnesS_WolF | 24=255.255.255.0 whcih means both are not i nteh same netmask |
| 16:49.34 | Qwell[] | Have you run an rtp debug, to see if the audio is flowing through asterisk? Is it being sent to the right addresses? |
| 16:49.39 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 16:49.43 | Qwell[] | DarKnesS_WolF: he knows that :p |
| 16:49.51 | DarKnesS_WolF | Qwell[]: b11d ok sorry ;-) |
| 16:49.55 | b11d | no.. im not doing an rtp debug on the * box.. i was doing it on the vg.. |
| 16:49.57 | b11d | one sec. |
| 16:50.02 | DarKnesS_WolF | b11d: test with nat=yes canreinvite=no |
| 16:50.05 | DarKnesS_WolF | in sip and see ;-) |
| 16:50.16 | b11d | would that be for both endpoints? |
| 16:50.20 | b11d | for both sip entries, that is? |
| 16:50.20 | Qwell[] | yes |
| 16:50.21 | b11d | ok |
| 16:50.23 | b11d | one sec. |
| 16:50.23 | DarKnesS_WolF | Qwell[]: i know he knows that too but i'm just fat ass who like to show a little bit :P? |
| 16:50.27 | Qwell[] | heh |
| 16:50.36 | DarKnesS_WolF | Qwell[]: see he didn't know :P |
| 16:50.45 | b11d | yes.. fight over me |
| 16:50.46 | b11d | :) |
| 16:50.49 | Juggie | Qwell, did you have a chance to look @ that bug for me? |
| 16:50.57 | seele_ | DarKnesS_WolF, like this http://pastebin.ca/234524 |
| 16:51.05 | Qwell[] | well, I'm guessing that the network is really more like 10.0.0.0/16 or something |
| 16:51.21 | Qwell[] | erm, 10.3...so, bigger than /16 :D |
| 16:51.27 | b11d | no dice lads |
| 16:51.28 | Qwell[] | maybe a /14 |
| 16:51.36 | Qwell[] | b11d: Do you get any output from rtp debug? |
| 16:51.36 | b11d | yes.. i restarted asterisk after making the changes |
| 16:51.41 | Qwell[] | in either direction? |
| 16:51.41 | DarKnesS_WolF | Qwell[]: acutally if i'll use 10.0.0.0 i use /8 i think thats standerd |
| 16:51.45 | b11d | going to do that now Qwell |
| 16:52.03 | DarKnesS_WolF | seele_: yes |
| 16:52.27 | p1p | Im seeing lots of "channel.c: Avoiding initial deadlock for 'SIP/211-093b0478' " in my asterisk full logs for all different extensions. What exactly does that mean? |
| 16:52.33 | DarKnesS_WolF | seele_: make sure both ends has G711U / G711A " ulaw/alaw " support |
| 16:53.09 | seele_ | DarKnesS_WolF, how can I see this |
| 16:53.18 | b11d | it just goes crazy Qwell |
| 16:53.23 | b11d | non stop "sent and received" lines |
| 16:53.24 | DarKnesS_WolF | seele_: what softphone ur using ? |
| 16:53.30 | Qwell[] | b11d: pastebin some of it...say...100 lines |
| 16:53.31 | b11d | or "got and sent" |
| 16:53.36 | b11d | ok give me a sec |
| 16:53.36 | blitzrage | Qwell[]: you have 1.4beta in the topic twice |
| 16:53.38 | Qwell[] | It should have IPs in it |
| 16:53.53 | Qwell[] | blitzrage: nope :) |
| 16:53.58 | seele_ | DarKnesS_WolF, no softphone ... linksys boxes |
| 16:53.59 | blitzrage | I see it! |
| 16:54.08 | Juggie | i see dead people. |
| 16:54.13 | Qwell[] | blitzrage: asterisk 1.4beta, zaptel 1.4beta, addons 1.4beta |
| 16:54.13 | blitzrage | I see dumb people. |
| 16:54.15 | DarKnesS_WolF | seele_: don't know anything about it ;-) RTFM or give it a try |
| 16:54.41 | blitzrage | Qwell[]: Zaptel 1.2, 1.4beta |
| 16:54.46 | blitzrage | it reads weird :) |
| 16:54.53 | Qwell[] | yeah, blame Russell :p |
| 16:54.57 | blitzrage | but I'm a topic nazi, so continue about your way |
| 16:55.05 | blitzrage | -!- Topic set by Qwell |
| 16:55.13 | Qwell[] | I added the last part :D |
| 16:55.17 | blitzrage | uh huh |
| 16:55.20 | b11d | i dont seem to know how to get the rtp debug info into syslog so i can paste it |
| 16:55.38 | Juggie | blameing russel is the safe thing to do |
| 16:55.44 | b11d | i'll try putty.. one sec |
| 16:55.46 | Juggie | hes busy w/ school and unable to defend himself |
| 16:56.33 | Qwell[] | brb,, |
| 16:58.06 | b11d | qwell.. |
| 16:58.07 | b11d | http://pastebin.ca/234532 |
| 16:58.27 | b11d | 10.3.237.50 is the sip phone im calling from |
| 16:58.33 | b11d | 10.0.5.5 is the vg224 |
| 16:58.40 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 16:58.57 | *** join/#asterisk tillo (i=tillo@85-218-17-14.dclient.lsne.ch) |
| 16:59.34 | seele_ | DarKnesS_WolF, yes the linksys has G711U / G711A ulaw/alaw support |
| 16:59.51 | b11d | where exactly is alaw used? |
| 16:59.54 | BurtyB | yay asterisk with YAC seems to be going down well :) |
| 17:00.05 | Juggie | b11d, everywhere that ulaw isnt? :) |
| 17:00.10 | b11d | :) |
| 17:00.17 | b11d | you lippy bastard :) |
| 17:00.32 | b11d | i love this channel |
| 17:00.45 | seele_ | DarKnesS_WolF, http://pastebin.ca/234533 |
| 17:01.03 | Juggie | b11d, the real answer is hard to say |
| 17:01.19 | Juggie | probally most of non-north america uses alaw |
| 17:01.29 | b11d | ahh ok |
| 17:02.04 | DarKnesS_WolF | seele_: give it a try |
| 17:02.33 | b11d | bbl everyone |
| 17:02.38 | b11d | if qwell comes back, tell him i had to go.. |
| 17:02.41 | b11d | but will return later |
| 17:03.37 | BurtyB | ://pastebin.ca/234533 |
| 17:03.53 | BurtyB | bah sorry about that i went on a click fest and pasted |
| 17:04.05 | Qwell[] | b11d: You aren't getting any RTP packets from 10.0.5.5 |
| 17:04.24 | Qwell[] | SO...you need to see what IP the vg224 is sending them to |
| 17:04.38 | Qwell[] | or, what IP asterisk is telling it to send them to |
| 17:05.46 | seele_ | brb |
| 17:06.13 | DarKnesS_WolF | any one seen the old systm eps. about asterisk ? with john todd ? |
| 17:07.28 | Qwell[] | DarKnesS_WolF: I downloaded it months ago, but never got around to watching it |
| 17:10.34 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 17:21.25 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 17:23.14 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:25.27 | *** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com) |
| 17:26.34 | blitzrage | yah, never watched it |
| 17:26.34 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 17:27.37 | DarKnesS_WolF | Qwell[]: it's nice very simple but i love it ... who is john todd ? |
| 17:27.38 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 17:27.55 | Qwell[] | DarKnesS_WolF: He's...somebody...who...does stuff? |
| 17:28.10 | DarKnesS_WolF | Qwell[]: no no i mena he comes here around ? nickname so on |
| 17:28.13 | Qwell[] | If you don't know who he is, that's a hard question to answer. :) |
| 17:28.14 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:28.15 | Qwell[] | ahh, jtodd |
| 17:28.27 | blitzrage | good old jtodd :) |
| 17:28.30 | DarKnesS_WolF | see easy :P |
| 17:28.34 | Qwell[] | blitzrage: indeed |
| 17:28.59 | DarKnesS_WolF | ok i'll go get some food and some tea ;-) |
| 17:29.07 | blitzrage | you'll see him log on irc... but rarely talk... but when he does... he's always got lots to say :) |
| 17:29.17 | file | jtodd wouldn't give me a hug :( |
| 17:29.22 | Qwell[] | file: lame! |
| 17:29.22 | file | Allison did though |
| 17:29.27 | Qwell[] | also lame! |
| 17:31.18 | blitzrage | file: jtodd doesn't seem like the hugging kind of fellow |
| 17:31.41 | file | blitzrage: I know, that's why it was fun to try |
| 17:31.45 | blitzrage | heh |
| 17:31.59 | blitzrage | bring it back, that old new york rap |
| 17:33.03 | file | blitzrage: oh no... it's you |
| 17:33.38 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:33.40 | blitzrage | sweet... no rain on Saturday for football |
| 17:34.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net) |
| 17:35.10 | hoobastooba | I am looking for a good SIP Provider in the US. I was looking at BroadVoice.com. Can anyone make recommendations. I need to make sure they do t38. |
| 17:36.18 | blitzrage | NuFone? |
| 17:36.29 | Qwell[] | Does nufone do t.38? |
| 17:36.35 | blitzrage | I won't be doing t.38 passthrough for another couple of months |
| 17:36.41 | Juggie | blitzrage, pm. |
| 17:36.43 | blitzrage | Qwell[]: no idea... I assume they do... ? |
| 17:36.48 | blitzrage | Juggie: no way :) |
| 17:39.06 | *** join/#asterisk shellshark (n=x86@69.13.8.224) |
| 17:39.27 | *** join/#asterisk saftsack (n=saftsack@pD9E0631E.dip.t-dialin.net) |
| 17:40.16 | *** join/#asterisk sexyken (n=sexyken@c-24-23-203-168.hsd1.ca.comcast.net) |
| 17:40.26 | sexyken | Hey people, anyone use the WIP 330 from Linksys? |
| 17:41.35 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 17:41.51 | *** join/#asterisk VibroMax (n=phisto@83.229.118.250) |
| 17:44.25 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 17:44.55 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
| 17:45.03 | TexasJay | Quick phpagi question: for fastpass_get_data(), what should I use for the first param ($buffer)? |
| 17:48.09 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 17:50.04 | hoobastooba | I am being told if i want to do email to fax and fax to email, i have to use openpbx. I am looking at both sides of the argument... Is there no solution for Astersik? |
| 17:50.18 | *** join/#asterisk monsted (n=monsted@rootweiler.dk) |
| 17:50.19 | hoobastooba | i installed SpanDSP and it is fairly unreliable. |
| 17:50.53 | hoobastooba | i dont want to have to get an SIP provider, and pay the rates. I would rather get the rate I have on my T1s. |
| 17:50.54 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:51.16 | hoobastooba | plus I already have local dids, which looks like an issue with t38 providers of sip |
| 17:51.29 | Corydon-w | spandsp is very reliable, if you follow the installation instructions |
| 17:51.42 | hoobastooba | i was told that it is no longer supported for asterisk |
| 17:51.56 | Corydon-w | It was never supported in Asterisk. It's a third party addon |
| 17:52.08 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:52.57 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:53.27 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:54.15 | hoobastooba | Corydon-w: I have installed it, but I can never get a multi page fax to go through successfully. I get line errors on the fax machine. |
| 17:54.35 | hoobastooba | are you successfully using this for fax? |
| 17:54.38 | Corydon-w | Yep |
| 17:55.02 | luke-jr_work | wtf |
| 17:55.15 | luke-jr_work | I can use spandsp, but not zap-to-zap bridging?? |
| 17:55.22 | Corydon-w | Follow the installation instructions to the letter... Don't think that the tiff version is 'close enough' |
| 17:55.25 | *** join/#asterisk Crescendo (n=Master@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:55.49 | luke-jr_work | If I bridge the fax line (zap) to the POTS line (also zap), the fax fails |
| 17:55.58 | hoobastooba | are you talking about this instructions included in the download? |
| 17:56.03 | luke-jr_work | but if I use rxfax to make a tiff, and then txfax that to the destination using the same lines, it works fine |
| 17:56.38 | Corydon-w | hoobastooba: yes |
| 17:56.57 | hoobastooba | ok, I will go through it again. you open to me PMing you this afternoon if I have any questions? |
| 17:57.15 | Corydon-w | hoobastooba: 99% of the problems are due to people using the wrong libtiff version |
| 17:57.32 | Corydon-w | I don't know where I'll be this afternoon |
| 17:57.38 | hoobastooba | ok |
| 17:57.38 | hoobastooba | thanks |
| 17:57.43 | hoobastooba | i appreciate the advice |
| 17:58.46 | TexasJay | Quick phpagi question: for fastpass_get_data(), what should I use for the first param ($buffer)? |
| 17:58.52 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 18:00.55 | Juggie | whatever you want. |
| 18:06.46 | luke-jr_work | is it possible to prevent native bridging of Zap channels? |
| 18:08.18 | *** join/#asterisk xaka (n=xaka2004@83.239.13.202) |
| 18:08.48 | *** join/#asterisk sexyken (n=sexyken@c-24-23-203-168.hsd1.ca.comcast.net) |
| 18:08.57 | sexyken | Hey guys, does Asterisk support Shared Lines yet? |
| 18:09.06 | xaka | what mean next message: "ERROR[18960] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info."? |
| 18:09.12 | *** join/#asterisk [shodan] (n=shodan@ip089.96-113-216.pppoe1.joliette.intermonde.net) |
| 18:11.18 | *** join/#asterisk p1p (n=p1p@mail.comp911.com) |
| 18:11.25 | luke-jr_work | sexyken, MeetMe |
| 18:11.44 | DarKnesS_WolF | xaka: means u need to check the logs for why it's not connected to mysql database |
| 18:12.04 | xaka | hm...no more logs presented :( |
| 18:12.20 | *** join/#asterisk davewise (i=icechat5@66.251.196.193) |
| 18:12.57 | DarKnesS_WolF | xaka: then edit ur /etc/asterisk/logger.conf and make sure u have full enabled and the console commented |
| 18:13.04 | DarKnesS_WolF | xaka: logs placed /var/log/asterisk/ |
| 18:13.14 | sexyken | MeetMe? what's that have to do with SharedLines? |
| 18:13.59 | davewise | Has anyone here compiled asterisk-1.2.13 from the ftp site? I keep getting compile errors from it. |
| 18:14.21 | blitzrage | what errors? |
| 18:14.30 | davewise | chan_phone.c:41:29: error: linux/compiler.h: No such file or directory |
| 18:14.37 | blitzrage | davewise: you're using fc6 aren't you? |
| 18:14.48 | blitzrage | remove that line from chan_phone.c at line 41 |
| 18:14.59 | davewise | you r correct :) |
| 18:15.05 | Qwell[] | latest 1.2 svn should have that fixed already |
| 18:15.07 | Qwell[] | if not, it's a bug |
| 18:15.13 | davewise | it's a problem w/FC6? |
| 18:15.20 | DarKnesS_WolF | davewise: yes i did compile it |
| 18:15.20 | sexyken | Is 1.2 the latest stable AsterisK? |
| 18:15.21 | blitzrage | its a problem with FC6 and newer kernels |
| 18:15.30 | blitzrage | 1.2 is the latest release |
| 18:15.31 | DarKnesS_WolF | davewise: which is ur destro ? |
| 18:15.32 | file | not newer kernels, just FC6 |
| 18:15.36 | davewise | Latest Stable is 1.2.13 |
| 18:15.38 | file | :P |
| 18:15.41 | blitzrage | file: supposed to be going into newer kernels though afaik |
| 18:15.42 | Qwell[] | well, newer kernels technically |
| 18:15.48 | davewise | just downloaded it again to make sure it was good |
| 18:15.50 | sexyken | blitz: How can you say "1.2 is the latest release" |
| 18:15.51 | blitzrage | davewise: latest release -- not stable :) |
| 18:15.52 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:15.52 | DarKnesS_WolF | ah fc6 |
| 18:15.52 | Qwell[] | it's been deprecated for a while... RedHat just decided to take it out |
| 18:15.53 | sexyken | That's false information |
| 18:15.53 | DarKnesS_WolF | yak :P |
| 18:15.57 | sexyken | The latest release is: Asterisk 1.4.0-beta3 released! |
| 18:16.00 | sexyken | :-) |
| 18:16.06 | blitzrage | notice the beta3 |
| 18:16.15 | sexyken | So? |
| 18:16.18 | blitzrage | unless you like running beta's in production |
| 18:16.21 | sexyken | The latest RELEASE is 1.4.0-beta3 |
| 18:16.21 | davewise | I am using the latest stable kernel from kernel.org |
| 18:16.33 | Qwell[] | sexyken: No, the latest BETA RELEASE is 1.4.0-beta3 |
| 18:16.39 | xaka | where i can get cdr_mysql.conf example? |
| 18:16.46 | sexyken | No |
| 18:16.46 | Qwell[] | xaka: asterisk-addons |
| 18:16.50 | sexyken | The latest stable release is 1.2.13 |
| 18:16.57 | sexyken | The latest release, is 1.4.0 |
| 18:16.58 | DarKnesS_WolF | xaka: ur problem in cdr or in res ? |
| 18:16.59 | sexyken | eta3 |
| 18:17.00 | blitzrage | there is not such thing as "stable" anymore -- its just called release |
| 18:17.02 | Qwell[] | 1.2.13 is not stable, it is a release |
| 18:17.07 | *** join/#asterisk mixi (n=mixi@the.one.and.only.iammixi.de) |
| 18:17.11 | sexyken | Well |
| 18:17.12 | sexyken | That too |
| 18:17.18 | sexyken | I think we're all wrong |
| 18:17.19 | sexyken | ;-) |
| 18:17.23 | blitzrage | I think you're more wrong |
| 18:17.27 | luke-jr_work | Qwell, it's a stable release =p |
| 18:17.34 | sexyken | I think Qwell is wrong |
| 18:17.36 | sexyken | It is a stable release |
| 18:17.37 | sexyken | slut. |
| 18:17.38 | sexyken | ;-) |
| 18:17.39 | xaka | Qwell, hm...i installed it |
| 18:17.40 | blitzrage | I'm sure Qwell[] knows |
| 18:17.41 | Qwell[] | I assure you, Qwell is not wrong. :) |
| 18:17.50 | blitzrage | Qwell[] is no nub |
| 18:17.51 | sexyken | Qwell is wrong. |
| 18:18.13 | luke-jr_work | well, Qwell bans people for helping other people... so... o.o |
| 18:18.16 | blitzrage | 1.2.13 is latest packaged release, 1.4.0-beta3 is latest beta release |
| 18:18.29 | xaka | DarKnesS_WolF, i don't configure connection options and get error (look at above) |
| 18:18.33 | Qwell[] | blitzrage: There you go |
| 18:18.47 | blitzrage | 1.2 SVN is latest SVN release :) |
| 18:19.12 | DarKnesS_WolF | xaka: mv /etc/asterisk/res_mysql.conf res_mysql.conf_ |
| 18:19.18 | hoobastooba | sexyken: you should read the release notes and such. |
| 18:19.19 | DarKnesS_WolF | xaka: or comment the file if it's not |
| 18:19.40 | davewise | So, to make asterisk-1.2.13 work with FC6, get the latest from svc correct? |
| 18:19.51 | hoobastooba | sexyken: if beta looks ok to you then use it |
| 18:20.01 | blitzrage | well, latest SVN 1.2 branch is not 1.2.13 -- its post-1.2.13 |
| 18:20.10 | sexyken | I'm running Asterisk 1.2.1 -- is that bad? |
| 18:20.18 | blitzrage | davewise: or just remove that one line from the chan_phone.c file |
| 18:20.18 | Qwell[] | davewise: easiest way, is to just comment out that #include |
| 18:20.22 | Qwell[] | OR |
| 18:20.28 | Qwell[] | go into menuselect, and disable chan_phone |
| 18:20.35 | blitzrage | Qwell[]: there is no menuselect in 1.2.... |
| 18:20.37 | Qwell[] | oh, wait, heh |
| 18:20.38 | hoobastooba | sexyken: you are missing out on improvements and bug fixes |
| 18:20.45 | blitzrage | Qwell[]: you've been in trunk too long, lol |
| 18:20.52 | sexyken | hoobastooba, where can I find out what changes have been made? |
| 18:20.53 | Qwell[] | blitzrage: yeah, trunk land has got me |
| 18:20.57 | blitzrage | and security vulnerabilities |
| 18:20.58 | davewise | which line (in the Make file or the phone.c?) |
| 18:21.01 | Qwell[] | sexyken: in the CHANGES file |
| 18:21.02 | hoobastooba | sexyken: the release notes |
| 18:21.02 | blitzrage | davewise: 41 |
| 18:21.08 | Qwell[] | davewise: In the chan_phone.c |
| 18:21.10 | blitzrage | davewise: like I mentioned earlier... :) |
| 18:21.12 | sexyken | I'm looking for the changes |
| 18:21.14 | sexyken | But I have tod ownload? |
| 18:21.18 | hoobastooba | they are on the digium site as well as i think included in the download |
| 18:21.18 | sexyken | Not an online changes file? |
| 18:21.29 | davewise | Thanks eveyone.... |
| 18:21.30 | DarKnesS_WolF | Qwell[]: what is chan_phone ? i didn't undersatnd what is writen in voip-info anyway |
| 18:21.31 | hoobastooba | sexyken: should be |
| 18:21.35 | davewise | I'll try it now! |
| 18:21.37 | davewise | :) |
| 18:21.42 | Qwell[] | DarKnesS_WolF: it's a channel driver for a specific type of device |
| 18:21.47 | Qwell[] | one that nobody actually...uses |
| 18:21.52 | Qwell[] | davewise: hell...delete the file |
| 18:21.52 | hoobastooba | sexyken: go to the download page and look for the release notes |
| 18:21.58 | hoobastooba | should be a link if i remember correctly. |
| 18:21.59 | Qwell[] | That'll take care of it :) |
| 18:21.59 | DarKnesS_WolF | Qwell[]: yes what kind of devices ? |
| 18:22.03 | Qwell[] | DarKnesS_WolF: dunno |
| 18:22.07 | DarKnesS_WolF | Qwell[]: lol |
| 18:22.11 | DarKnesS_WolF | ok i'll unload it then ;-) |
| 18:22.13 | blitzrage | Qwell[]: I think there is a dep. on chan_phone for some reason... |
| 18:22.19 | blitzrage | if I remember right |
| 18:22.20 | Qwell[] | oh |
| 18:22.34 | blitzrage | I think Dial() needs it |
| 18:22.42 | Qwell[] | silly |
| 18:22.45 | blitzrage | quite |
| 18:22.50 | DarKnesS_WolF | hm |
| 18:22.53 | Qwell[] | somebody should fix that :P |
| 18:22.55 | blitzrage | heh |
| 18:23.04 | DarKnesS_WolF | Qwell[]: ur a pure evil ! |
| 18:23.06 | blitzrage | yep... I haven't seen somebody around for a while |
| 18:23.15 | Qwell[] | ~seen somebody |
| 18:23.29 | jbot | somebody <n=somebody@200.189.112.30> was last seen on IRC in channel #debian-br, 391d 23h 18m 47s ago, saying: 'como faço para adicionar uma impressora da rede do windows no debian'. |
| 18:23.29 | DarKnesS_WolF | Qwell[]: asterisk without Dial is liek a car without doors :P |
| 18:23.38 | sexyken | Do you guys run Asterisk with any type of Graphical Interface? |
| 18:23.40 | Qwell[] | yeah, he really dropped the ball on that one |
| 18:23.48 | Qwell[] | sexyken: xterm |
| 18:24.14 | hoobastooba | I have upgraded to the latest polycom firmware version, have double checked my dial plan and hardware settings. I am still having issues where people call into a queue and when the caller is answered the queue member cannot hear them. It is intermittent. can anyone offer some suggestions as to what else I could try to correct this issue? |
| 18:24.28 | hoobastooba | phones and server on same network, no nat. |
| 18:24.39 | hoobastooba | i have added nat=no and canreinvite=no to each sip device |
| 18:24.40 | *** join/#asterisk fx0 (n=fx0@tor.blaqhat.com) |
| 18:24.50 | sexyken | Qwell, I simply meant a GUI Interface for managing asterisk. |
| 18:25.06 | sexyken | Not a console you use to manage asterisk using the CLI and directly editing config files. |
| 18:25.17 | *** part/#asterisk xaka (n=xaka2004@83.239.13.202) |
| 18:25.25 | DarKnesS_WolF | sexyken: i used to have FreePBX but i didn't like it ;-) |
| 18:25.35 | Qwell[] | blitzrage: ! |
| 18:25.39 | Qwell[] | ping:D |
| 18:25.42 | mcab | sexyken: I suspect Qwell knew exactly what you ment :-) |
| 18:25.48 | sexyken | hoobastooba: Do you have that thing turned off? |
| 18:26.02 | hoobastooba | sexyken: there are many gui interfaces out there, but most are more work to configure and administer than just doing it from the cli and configs |
| 18:26.10 | hoobastooba | <PROTECTED> |
| 18:27.00 | hoobastooba | do i have which thing turned off? |
| 18:27.04 | sexyken | hoobastooba: nat=yes, host=dynamic, type=friend |
| 18:27.11 | sexyken | dtmfmode=auto |
| 18:27.13 | sexyken | All those set properly? |
| 18:27.19 | *** join/#asterisk I-MOD (i=opticron@71.207.209.230) |
| 18:27.24 | davewise | That worked! Thanks |
| 18:27.37 | hoobastooba | if both the phone and the server are on the same network with no nat, shouldnt it be nat=no? |
| 18:27.40 | luke-jr_work | is it possible to prevent native bridging of Zap channels? |
| 18:27.45 | *** part/#asterisk fx0 (n=fx0@tor.blaqhat.com) |
| 18:27.51 | *** join/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 18:27.55 | Cherebrum | How old is Asterisk? |
| 18:28.02 | Qwell[] | Cherebrum: 7 years? |
| 18:28.05 | blitzrage | 80 yrs tomorrow |
| 18:28.06 | Qwell[] | something like that |
| 18:28.10 | sexyken | What IP is your SIP Phone connecting to? |
| 18:28.18 | davewise | I noticed that make mpg123 doesn't work w/FC6... Is this new (does anyone know) |
| 18:28.27 | blitzrage | don't use mpg123 |
| 18:28.30 | hoobastooba | sexyken: you asking me? |
| 18:28.32 | blitzrage | use native moh |
| 18:28.32 | sexyken | Yes |
| 18:28.35 | sexyken | hoba, yes. |
| 18:28.39 | hoobastooba | 10.0.1.183 |
| 18:28.45 | sexyken | What IP is your phone connecting from |
| 18:28.48 | Cherebrum | yea.. using mpg123 for music on hold is a terrible idea |
| 18:29.08 | davewise | What do you use instead of mpg123 and does it require any changes in any config files? |
| 18:29.09 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.120) |
| 18:29.10 | hoobastooba | 10.0.4.195 |
| 18:29.18 | blitzrage | native moh |
| 18:29.21 | luke-jr_work | davewise, they told you |
| 18:29.22 | hoobastooba | davewise: native |
| 18:29.23 | Qwell[] | davewise: I would convert your MoH files to wav or ulaw or something |
| 18:29.29 | hoobastooba | and put the sound files into ramdisk |
| 18:29.33 | hoobastooba | good quality sound |
| 18:29.34 | sexyken | hoobah: Turn on NAT for now |
| 18:29.35 | sexyken | Test it out |
| 18:29.41 | hoobastooba | ok. |
| 18:29.46 | Qwell[] | If you're using the standard MoH files, you can get wav/ulaw/etc versions from ftp.digium.com/ |
| 18:29.47 | hoobastooba | davewise: explain |
| 18:29.50 | davewise | ok thanks |
| 18:29.54 | hoobastooba | sorry... not davewise |
| 18:29.55 | sexyken | Also -- I'd suggest forcing your codec if you haven't already |
| 18:29.58 | *** part/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 18:30.23 | hoobastooba | sexyken: explain the nat=yes in my case. I must be misunderstanding its purpose. |
| 18:30.43 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 18:31.14 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:31.18 | I-MOD | i have an issue that i'm trying to debug, but i'm not sure what the intended behavior of callwaiting and threewaycalling are when callwaiting=yes and threewaycalling=no |
| 18:31.25 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 18:31.35 | blitzrage | Qwell[]: pong |
| 18:31.41 | sexyken | NAT=yes causes Asterisk to ignore the address information in the SIP and SDP headers, and reply to the sender's IP address and port. NAT=yes enables a form of Symmetric RTP in Asterisk. |
| 18:31.49 | I-MOD | the behavior that i'm seeing is that the channel i'm trying to transfer to is being dropped |
| 18:31.57 | Qwell[] | blitzrage: preceed that with a /j :D |
| 18:32.13 | Qwell[] | d'oh |
| 18:32.13 | hoobastooba | sexyken: so would reinvite still = no? |
| 18:32.45 | sexyken | I always have reinvite = no |
| 18:32.51 | Corydon-w | canreinvite=... there is no option called reinvite |
| 18:33.01 | sexyken | Doesn't matter |
| 18:33.03 | sexyken | He knows what it is |
| 18:33.07 | *** join/#asterisk VooDooNOFX (n=CheeseHe@pool-71-104-61-26.lsanca.dsl-w.verizon.net) |
| 18:33.12 | sexyken | Why type more when not nesescary? |
| 18:33.23 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:33.24 | Qwell[] | blitzrage: /j and /q are way too close to each other, eh? |
| 18:33.25 | Corydon-w | because people misinterpret |
| 18:33.26 | hoobastooba | sorry, typing from short term memory Corydon-w |
| 18:33.27 | hoobastooba | :-D |
| 18:33.28 | blitzrage | Qwell: oops... :) |
| 18:33.33 | blitzrage | Qwell: way too close :) |
| 18:33.46 | sexyken | Our Polycoms dont work very well with reinvite |
| 18:33.52 | Corydon-w | blitzrage: must not be qwerty, eh? |
| 18:34.01 | luke-jr_work | dvorak ftw! |
| 18:34.04 | Qwell[] | Corydon-w: jwerty - silly canadians |
| 18:34.08 | blitzrage | nope... qjerty |
| 18:34.14 | Qwell[] | heh |
| 18:34.31 | luke-jr_work | q and j are next to each other on dvorak |
| 18:34.50 | Qwell[] | dvorak rocks |
| 18:35.16 | Qwell[] | I have a dvortyboard at home... I never USE the "dvorak lock" key, but hey |
| 18:35.40 | *** join/#asterisk knobenheimer_lap (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 18:35.52 | Qwell[] | It sure is fun when people come over and try to use my computer though |
| 18:35.55 | luke-jr_work | KDE supports it |
| 18:36.13 | hoobastooba | sexyken: I have made the changes and will see how it goes. Thank you for chiming in. |
| 18:36.25 | luke-jr_work | in software |
| 18:36.55 | sexyken | Also, hoobastooba....when reinvite is enabled and successful, Asterisk is not in the middle any longer. This means that call recording, parking, etc can not be used. |
| 18:37.46 | luke-jr_work | sexyken, reinvite is not enabled if recording, etc |
| 18:38.34 | sevard | Damnit. Does anyone know a list of characters you CAN'T use in SIP/sip.conf's secret/Sipura ATA's SIP password? |
| 18:38.36 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:38.48 | sevard | One of those three things is rejecting one character from my awesome password. |
| 18:39.04 | luke-jr_work | is it possible to prevent native bridging of Zap channels? |
| 18:39.05 | sevard | I could do the experiment, one by one character, but if anyone has an idea that'd save me a few hours of bs. |
| 18:39.21 | luke-jr_work | sevard, use a less secure password |
| 18:39.24 | Qwell[] | sevard: stuff like % would probably be interpreted as an escape sequence thing |
| 18:39.30 | Qwell[] | like %20, html escapes.. |
| 18:39.32 | luke-jr_work | like 'my-pap2' |
| 18:39.34 | sevard | Luke-Jr: .... |
| 18:39.40 | luke-jr_work | lol |
| 18:39.41 | sevard | Luke-Jr: pap smear |
| 18:40.03 | luke-jr_work | jk |
| 18:40.10 | blitzrage | ok... gym time -- back later |
| 18:40.13 | luke-jr_work | but 3294jr34gerjg is pretty secure and compat |
| 18:40.14 | Qwell[] | blitzrage: :( |
| 18:40.24 | sevard | So stick to alphanumeric passwords |
| 18:40.29 | sevard | as characters are butt*#&#@ers. |
| 18:40.43 | aptura | Interesting. TISP with * has some security issues. http://forums.whirlpool.net.au/forum-replies-archive.cfm/498645.html |
| 18:42.01 | blitzrage | aptura: that has nothing to do with asterisk |
| 18:42.25 | luke-jr_work | aptura, I've seen that before |
| 18:43.13 | aptura | I see |
| 18:46.37 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 18:48.07 | sahafeez | hey, reading about getting a g.729 license from digium. question - i need only one per conversation - so if i want to support g.729 on 3 sip phones i just need 3 license. am i reading that right? |
| 18:49.38 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 18:50.48 | GaVak | Anyone ever work with a D-Link DFL-300 Router/Firwall in reference to getting SIP to work? I've allowed UDP/TCP on 5060 on the firewall for all IP's, but it keeps setting any traffic to DROP. I was wondering if there was something special I had to set to allow SIP to function. |
| 18:51.29 | luke-jr_work | sahafeez, you probably don't need any |
| 18:51.51 | luke-jr_work | sahafeez, unless Asterisk itself is decoding or encoding the call |
| 18:52.09 | luke-jr_work | but if the SIP phones are doing all your encoding/decoding, they came with licenses |
| 18:52.36 | sahafeez | i have sip provider that does g.729. right now the calls are not the best. doing ulaw/alaw |
| 18:52.49 | sahafeez | the other issue is i am putting some phones remote over a vpn |
| 18:53.04 | sahafeez | polycom 501s |
| 18:53.10 | luke-jr_work | sahafeez, do your phones support g.729? |
| 18:54.03 | sahafeez | i was told by the polycom rep yes.. |
| 18:54.11 | luke-jr_work | then you don't need any licenses |
| 18:54.20 | luke-jr_work | the SIP provider does, and the company producing your phones do |
| 18:54.24 | Qwell[] | Unless you plan on doing any transcoding |
| 18:54.28 | luke-jr_work | but your Asterisk isn't touching the stream |
| 18:54.30 | b11d | im back |
| 18:54.38 | luke-jr_work | well, not the codec part anyhow |
| 18:54.38 | Qwell[] | ie; doing a Playback of any non g729 soundfiles |
| 18:54.41 | Qwell[] | b11d: scroll up |
| 18:54.44 | b11d | ok |
| 18:54.52 | b11d | to what time? |
| 18:54.58 | Qwell[] | just after you left ;) |
| 18:55.10 | b11d | ok |
| 18:55.17 | luke-jr_work | Qwell, does Digium have the recordings available in g729? |
| 18:55.22 | Qwell[] | luke-jr_work: yep |
| 18:55.30 | luke-jr_work | so that shouldn't be a problem :) |
| 18:55.32 | Qwell[] | and even g722 now |
| 18:55.37 | mog | in 1.4 |
| 18:55.40 | Qwell[] | luke-jr_work: no, but custom ones could be |
| 18:55.41 | mog | select them in menuselct |
| 18:55.52 | luke-jr_work | Qwell, not if they use the phone to record them ;) |
| 18:55.54 | b11d | [11:06] <Qwell[]> b11d: You aren't getting any RTP packets from 10.0.5.5 |
| 18:55.54 | b11d | [11:06] <Qwell[]> SO...you need to see what IP the vg224 is sending them to |
| 18:55.54 | b11d | [11:06] <Qwell[]> or, what IP asterisk is telling it to send them to |
| 18:55.59 | b11d | this is what you meant I take it |
| 18:56.02 | Qwell[] | yep |
| 18:56.05 | b11d | interesting.. |
| 18:56.27 | b11d | the vg is sending them to 10.0.5.11 I think |
| 18:56.29 | b11d | the * box |
| 18:56.35 | Qwell[] | make sure |
| 18:56.37 | b11d | ok |
| 18:56.40 | b11d | brb |
| 18:56.48 | Qwell[] | because asterisk isn't seeing any packets from the vg224 |
| 18:57.04 | sahafeez | luke-jr_work: sorry i do not follow - you are saying asterisk will do g.729 without the license as long as the end point does it |
| 18:57.20 | monsted | has anyone here attempted to make a trunk connection between asterisk and a call manager express? |
| 18:57.22 | luke-jr_work | sahafeez, Asterisk does not need a license for mere passthrough |
| 18:57.57 | luke-jr_work | sahafeez, as long as you don't use actual g729 code, you're good |
| 18:57.58 | sahafeez | hum. ok, i can test this by changing my sip.conf for a phone to g.729 |
| 18:58.06 | luke-jr_work | sahafeez, sure |
| 18:58.51 | luke-jr_work | Qwell, how can I block Asterisk from bridging zap channels? |
| 18:59.07 | zamba | how do i deactivate the zap-module? |
| 18:59.50 | Qwell[] | luke-jr_work: dunno.. |
| 19:00.27 | luke-jr_work | :\ |
| 19:00.40 | I-MOD | luke-jr_work, what are you trying to accomplish by preventing the channels from being bridged? |
| 19:01.04 | luke-jr_work | I-MOD, higher quality |
| 19:01.12 | luke-jr_work | bridging sounds like crap |
| 19:01.39 | I-MOD | maybe you could force it through a local channel before sending it back out |
| 19:01.52 | I-MOD | but bridging channels should improve the quality |
| 19:02.01 | *** join/#asterisk duderock (n=natrix@host-216-153-249-214.mil.choiceone.net) |
| 19:02.24 | luke-jr_work | lots of noise during ringing and a loud squeal when answered such that nobody can hear the other party... |
| 19:02.26 | I-MOD | *hardware bridging |
| 19:02.57 | duderock | Anyone use ARI to display call details? Mine is displaying two entries for every call.... |
| 19:03.24 | I-MOD | luke-jr_work, do you have hardware echocan and is it enabled when zap channels are bridged? |
| 19:03.39 | luke-jr_work | I-MOD, yes and yes |
| 19:03.58 | sahafeez | like-jr_work: ok, i have a call up between to phones. how do i see what they are useing? |
| 19:04.19 | luke-jr_work | sahafeez, no clue-- maybe set verbose 100 before calling |
| 19:04.45 | TexasJay | ok, hopefully the third time is the charm |
| 19:04.46 | TexasJay | Quick phpagi question: for fastpass_get_data(), what should I use for the first param ($buffer)? |
| 19:04.48 | I-MOD | luke-jr_work, in zapata.conf, set echocancelwhenbridged=no |
| 19:04.54 | sahafeez | 192.168.119.66 4206 5b804133256 00102/00000 g729 No Tx: ACK |
| 19:04.58 | sahafeez | cool thanks! |
| 19:05.15 | luke-jr_work | I-MOD, tried that first |
| 19:06.57 | I-MOD | luke-jr_work, tdm2400? |
| 19:07.12 | luke-jr_work | I-MOD, Sangoma A102 |
| 19:07.56 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 19:08.15 | I-MOD | have you talked to sangoma's tech support about this? |
| 19:08.38 | luke-jr_work | nope |
| 19:08.58 | luke-jr_work | I suspect it's the wiring here |
| 19:09.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 19:10.15 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 19:11.02 | I-MOD | luke-jr_work, possible....try connecting phones directly to the card via patch cable to see if that fixes the problem, otherwise, i know very little about the specifics of sangoma cards |
| 19:11.47 | luke-jr_work | I-MOD, well, the phone wiring isn't the problem nearly so much as the POTS end |
| 19:11.51 | brc_ | anybody know if there is a trixbox channel? |
| 19:12.05 | luke-jr_work | it's mostly just cut up Cat5e cables twisted together |
| 19:12.10 | luke-jr_work | brc_, RTFT |
| 19:14.00 | I-MOD | luke-jr_work, any way to have the telco come out and test the line? |
| 19:14.14 | luke-jr_work | I-MOD, dunno, do they usually do that for free? |
| 19:14.19 | luke-jr_work | there's 3 lines |
| 19:14.42 | sahafeez | luke-jr_work: well you were not quite right. Unable to open /var/spool/asterisk/voicemail/default/4211/unavail (format g729): No such file or directory |
| 19:15.00 | luke-jr_work | sahafeez, you need to rerecord it, yes |
| 19:15.11 | sahafeez | ah, ok. cool. |
| 19:15.16 | luke-jr_work | I mentioned that ;p |
| 19:15.55 | sahafeez | i missed it then. sorry. |
| 19:15.55 | sahafeez | thanks |
| 19:15.55 | Qwell[] | and existing voicemails need to be changed |
| 19:15.55 | I-MOD | luke-jr_work, they *SHOULD* do it for free, but they'll probably charge for it |
| 19:15.56 | luke-jr_work | you also need to get the g729 voicemail messages from * 1.4 |
| 19:15.56 | I-MOD | call and ask them |
| 19:16.31 | sahafeez | luke-jr_work: so all of my exisiting voice will not work unless it is rerecored. also the voicemail files do not work |
| 19:16.54 | b11d | holy mother fuck Qwell |
| 19:16.56 | b11d | you nailed it |
| 19:16.58 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 19:16.59 | luke-jr_work | sahafeez, you could also transcode it manually, but that requires licensed software |
| 19:17.00 | diclophis-work | hwllo all |
| 19:17.16 | TexasJay | hello? can ANYONE answer my phpagi question?? |
| 19:17.17 | sahafeez | ok. so dl 1.4 beta and get the files from there. |
| 19:17.28 | diclophis-work | TexasJay: what was your question? |
| 19:17.29 | Qwell[] | b11d: works? |
| 19:17.32 | TexasJay | Quick phpagi question: for fastpass_get_data(), what should I use for the first param ($buffer)? |
| 19:17.36 | b11d | a log from my vg224: rtp_udp_unreachable: Entered ICMP unreachable for dest port 17801 |
| 19:17.58 | diclophis-work | TexasJay: looks like it might be a referenace variable to some resource type |
| 19:17.58 | b11d | it doesnt say what IP its trying to connect to |
| 19:18.13 | diclophis-work | TexasJay: or an integer value for the amount of buffer to use |
| 19:18.38 | TexasJay | Well according to the docs, it's supposed to be a string, but it's a referenced var. |
| 19:18.50 | diclophis-work | yea, so just pass in a blank string |
| 19:19.01 | diclophis-work | and when the function is done, it will be filled with the content of the message? |
| 19:19.02 | TexasJay | fastpass_get_data(&$buffer ... ) |
| 19:19.03 | diclophis-work | probably |
| 19:19.27 | TexasJay | That's what I thought. But passing it a blank string, an empty variable, an empty string and an integer all do nothing. |
| 19:19.56 | diclophis-work | does the function work? |
| 19:20.07 | diclophis-work | you probably need to dive deeper into the code |
| 19:20.15 | TexasJay | it's supposed to play an audio file and accept dtmf input. |
| 19:20.38 | TexasJay | It doesn't do anything. The console only shows an error when I have it play a file that does not exist. |
| 19:20.54 | b11d | should that port be open on the * box, or the SIP phone? |
| 19:21.20 | Qwell[] | b11d: probably the * box |
| 19:21.22 | TexasJay | And now it's not even showing that error. Ugh. |
| 19:21.26 | b11d | hmm |
| 19:21.30 | TexasJay | Where's docelmo when you need him? ;) |
| 19:21.51 | b11d | from what I can tell, it IS sending that data to the * Box. but gets that ICMP error.. |
| 19:21.54 | b11d | im not firewalling it.. |
| 19:22.07 | TexasJay | Welp, looks like I need to look into it further. Thanks diclophis-work |
| 19:22.11 | diclophis-work | no prob |
| 19:22.14 | b11d | i'll try to manually "permit" icmp through this vg.. |
| 19:22.16 | b11d | just to be sure |
| 19:22.41 | luke-jr_work | ICMP shouldn't be blocked anyhow |
| 19:22.56 | b11d | yeah I know |
| 19:23.07 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 19:23.10 | b11d | but I cant really explain this ICMP error, which I figure is causing my one-way audio problem |
| 19:26.38 | b11d | i see that sip to sip only dumps about 6 of those RTP messages.. |
| 19:26.53 | b11d | it looks exactly the same as sip to the analog, but those keep going forever.. |
| 19:27.09 | b11d | and sip to sip uses much lower port numbers. |
| 19:27.13 | b11d | 2248 and 2246 |
| 19:27.24 | b11d | 2262 and 2248 that is |
| 19:28.27 | b11d | well.. icmp IS wide open on this vg.. |
| 19:28.37 | b11d | and there is NO firewall preventing it from making the connection |
| 19:28.46 | b11d | hrm.. it just must be trying to connect to the wrong IP. |
| 19:30.08 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 19:30.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) [NETSPLIT VICTIM] |
| 19:30.09 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) [NETSPLIT VICTIM] |
| 19:30.21 | *** join/#asterisk lorinc (n=ang@caracas-3779.adsl.interware.hu) [NETSPLIT VICTIM] |
| 19:30.31 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) [NETSPLIT VICTIM] |
| 19:32.10 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) [NETSPLIT VICTIM] |
| 19:33.12 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 19:33.13 | *** join/#asterisk VooDooNOFX (n=CheeseHe@pool-71-104-61-26.lsanca.dsl-w.verizon.net) [NETSPLIT VICTIM] |
| 19:33.14 | *** join/#asterisk Malawar (n=Malawar@adsl-75-21-165-54.dsl.sgnwmi.sbcglobal.net) |
| 19:33.16 | Malawar | hum |
| 19:33.20 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 19:33.22 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) [NETSPLIT VICTIM] |
| 19:33.23 | b11d | sigh |
| 19:33.24 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 19:33.32 | Malawar | when I do something like a get_data |
| 19:33.38 | Malawar | does the # key always end data gathering? |
| 19:35.12 | *** join/#asterisk _Vile (i=vile@198.175.14.242) |
| 19:36.02 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 19:36.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:36.53 | *** part/#asterisk BurtyB (n=chris@alxa.camelnetwork.com) |
| 19:37.03 | *** join/#asterisk Mavvie (n=edwin@59.167.33.110) |
| 19:38.53 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) [NETSPLIT VICTIM] |
| 19:39.15 | *** join/#asterisk bigjb (n=nbigjb@195.60.10.114) |
| 19:39.28 | *** join/#asterisk evilstoy (i=HydraIRC@83-131-15-19.adsl.net.t-com.hr) |
| 19:39.52 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:42.05 | b11d | yeah this is making more sense than ever Qwell |
| 19:42.18 | b11d | i see the vg224 tries to return RTP data to the same port its sent on, except up one. |
| 19:42.37 | Qwell[] | That's normal |
| 19:42.42 | b11d | oh |
| 19:42.49 | Qwell[] | unless asterisk is expecting it elsewhere |
| 19:42.52 | *** join/#asterisk andresmujica (n=AndresMu@190.24.70.13) |
| 19:42.58 | Qwell[] | Check the rtp debug from the very start of the call |
| 19:43.07 | Qwell[] | or the sip debug, I guess it is.. see what the sdp says |
| 19:43.24 | b11d | i am doing that right now |
| 19:43.26 | andresmujica | hello! |
| 19:43.34 | diclophis-work | so i have a question regarding a sip connection between two asterisk servers... |
| 19:43.38 | andresmujica | anyone knows what should i do for speex testing? |
| 19:43.43 | andresmujica | i'm on centos |
| 19:43.46 | diclophis-work | ... why would the "ringing" status not be sent over that connection? |
| 19:43.57 | b11d | aster wants it on 2246 |
| 19:44.09 | b11d | cisco is trying to send it to 18331 but itself is sending out on 18330 |
| 19:44.24 | diclophis-work | i have notifyringing=yes in the configs on both machines |
| 19:44.46 | Qwell[] | b11d: You sure 2246 isn't what asterisk is sending on? |
| 19:44.50 | Qwell[] | s/on/from/ |
| 19:44.54 | b11d | not yet.. it could be |
| 19:45.09 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 19:45.31 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:45.37 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 19:45.47 | b11d | let me paste a sip debug & rtp debug |
| 19:45.51 | b11d | brb |
| 19:46.15 | andresmujica | hello, anyone knows how can i make speex works with asterisk??? |
| 19:47.07 | diclophis-work | i also get this error: app_dial.c:666 wait_for_answer: Unable to forward voice frame |
| 19:47.18 | diclophis-work | i don't like that line number at all |
| 19:48.37 | *** join/#asterisk zazzizza (n=gabriel@syrah.cespi.unlp.edu.ar) |
| 19:48.54 | zazzizza | hi there |
| 19:49.27 | b11d | http://pastebin.ca/234700 |
| 19:49.31 | zazzizza | does anybody know for sure whether tdm400 cards are 3.3v or 5v? |
| 19:49.39 | b11d | thats the sip calling the analog, analog answers, then hangs up, then I hang up the SIP. |
| 19:49.53 | b11d | thats with sip debug peer 5454 and sip debug peer 2637309 and rtp debug |
| 19:52.36 | b11d | zazz.. digium has a 1-800 number which can answer that shit easier eh |
| 19:52.50 | b11d | their sales dept would love to answer that for you :) |
| 19:52.59 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:53.24 | zazzizza | ha! roger. cause they seem to be pretty sh..y on posting info ;-) |
| 19:54.04 | b11d | yeah i hear you |
| 19:54.44 | b11d | i note this in the vg224 output: laddr = 10.0.5.5 lport = 19410, raddr = 10.3.237.50 rport=2260 |
| 19:54.46 | b11d | thats the problem isnt it |
| 19:54.54 | b11d | raddr needs to be 10.0.5.11 |
| 19:54.57 | b11d | i think |
| 19:55.18 | b11d | 10.3.237.50 is the sip phone, 10.0.5.11 is the * box |
| 19:57.02 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 19:57.03 | b11d | come on.. we need to get Julian out of Jail today |
| 19:57.11 | aptura | julian? |
| 19:57.20 | b11d | go to google video and watch Trailer Park Boys |
| 19:57.24 | b11d | :) |
| 19:57.26 | aptura | ohh |
| 19:57.33 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 19:58.04 | aptura | keep him there mabey the show wont be broadcast :) |
| 19:58.35 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 19:58.44 | b11d | that show rules.. |
| 19:58.48 | GaVak | Ok, I think I'm losing my mind. For a sip phone, 5060 is the default port that it registers with Asterisk. Does it use TCP or UDP for that connection? |
| 19:58.54 | b11d | UDP |
| 19:59.08 | b11d | but it can use TCP.. iirc |
| 19:59.22 | *** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 19:59.40 | GaVak | do some old firewalls drop SIP just because? |
| 19:59.48 | b11d | I cant speak to that |
| 19:59.51 | seele_ | how can I recors a call with a phone keyboard command like |
| 19:59.57 | seele_ | # or * |
| 20:00.02 | seele_ | ? |
| 20:00.05 | b11d | i dont know what recors means |
| 20:00.10 | GaVak | ok, thanks. I'm about to throw the asterisk server outside the firewall and be done with it. |
| 20:00.27 | b11d | yeah.. that sucks |
| 20:00.33 | b11d | but you gotta do what you gotta do |
| 20:00.40 | monsted | GaVak: depending on the type of firewall, it may drop protocols that don't seem to make sense |
| 20:01.06 | b11d | which is likely a smarter way to act :) |
| 20:01.07 | blitzrage | TPB! |
| 20:01.12 | b11d | there we go |
| 20:01.15 | b11d | someone knows it |
| 20:01.18 | monsted | GaVak: currently, our Cisco FWSM drops any H323 v4 packets it sees, since they don't look like the H323 v3 it knows |
| 20:01.25 | blitzrage | of course...I live in a trailer park |
| 20:01.29 | b11d | nice! |
| 20:01.35 | b11d | do you have a Lahey? |
| 20:01.40 | blitzrage | and funny enough... I like rye and ginger (julian likes rye and coke) |
| 20:01.48 | b11d | julian drinks rum and coke |
| 20:01.54 | blitzrage | rum? really? ewwww |
| 20:01.57 | b11d | yeah.. |
| 20:02.00 | monsted | that's a pity for the rum and the coke :) |
| 20:02.03 | b11d | but it doesnt matter.. great show |
| 20:02.08 | seele_ | b11d, sorry record |
| 20:02.15 | b11d | there ARE some good rum's out there |
| 20:02.15 | blitzrage | aye |
| 20:02.38 | b11d | nice |
| 20:02.45 | b11d | lets beer bong that :) |
| 20:02.53 | b11d | out of disrespect for the aging process |
| 20:02.57 | monsted | oi! |
| 20:03.04 | monsted | that's $100 worth of rum around here! |
| 20:03.13 | b11d | its probably great eh |
| 20:03.19 | monsted | it is :) |
| 20:03.23 | b11d | smoothest rum I've ever enjoyed came from the Barbados.. |
| 20:03.47 | b11d | I have no idea of its name, or age. |
| 20:03.52 | b11d | (just like those girls I was with last night) |
| 20:03.55 | b11d | :) |
| 20:03.56 | b11d | kidding |
| 20:04.27 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 20:04.40 | monsted | zacapa is from guatamala and has several "best rum in the world" awards on its resume :) |
| 20:04.56 | monsted | fabulous stuff |
| 20:04.59 | b11d | thats cool.. i'll have to hunt some down |
| 20:05.17 | b11d | I was recently introduced to the world of scotch lately.. |
| 20:05.21 | b11d | i had no idea it could be so good |
| 20:05.29 | monsted | if you find one with a black label in the older style reed-covered bottle, send it to me :) |
| 20:05.36 | b11d | umm.. i'll do that :) |
| 20:06.30 | b11d | im going to blow away my vg224 config and start this from scratch again |
| 20:06.36 | b11d | i think im beginning to peice this together |
| 20:06.40 | monsted | it's a special single-cask edition that's nowhere to be found any more :( |
| 20:06.45 | b11d | not even on ebay? |
| 20:06.51 | b11d | single-cask edition>? |
| 20:06.52 | b11d | wow |
| 20:06.54 | b11d | that must be $$ |
| 20:07.41 | Crescendo | Where can I get support for trixbox...? |
| 20:08.10 | b11d | #trixbox |
| 20:08.12 | monsted | dunno what it'd sell for now, but the regular one is $70ish now and the black label wasn't really any more expensive |
| 20:08.16 | b11d | or maybe #openpbx |
| 20:08.21 | Crescendo | b11d, empty channel. |
| 20:08.23 | Qwell[] | #freepbx.. |
| 20:08.36 | b11d | dammit |
| 20:08.41 | seele_ | Is posible start/stop call recording with DTMF commands ?? |
| 20:08.42 | b11d | i keep saying openpbx |
| 20:09.06 | monsted | try #callmanager! |
| 20:09.09 | monsted | or not ;) |
| 20:09.57 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 20:10.14 | luke-jr_work | can a dialplan initiate a new call? |
| 20:10.47 | *** join/#asterisk sitedude (n=sitedude@cpe-076-182-082-228.sc.res.rr.com) |
| 20:10.53 | Qwell[] | luke-jr_work: app_dial |
| 20:11.06 | Qwell[] | What is your definition of "call"? |
| 20:11.12 | sitedude | Is asterisk compatible with Lightyear voip? |
| 20:11.27 | luke-jr_work | Qwell, same as a .call file |
| 20:11.37 | Qwell[] | then sure |
| 20:11.40 | luke-jr_work | Qwell, I want to TxFAX after RxFAX completes |
| 20:11.51 | Qwell[] | System(cp my.call /path/to/spool/stuff/) |
| 20:11.56 | luke-jr_work | :( |
| 20:11.59 | luke-jr_work | that's hackish |
| 20:12.09 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 20:12.39 | luke-jr_work | and means I need to contruct a file |
| 20:12.45 | TexasJay | Anyone here a PHPAGI whiz? I have some code I'd like to post in a privmsg to you (so as to not clutter up the channel). |
| 20:12.54 | Qwell[] | TexasJay: pastebin |
| 20:12.56 | sitedude | Is asterisk easy to configure? |
| 20:13.01 | Qwell[] | sitedude: sure |
| 20:13.06 | TexasJay | pastebin doesn't load for me :( |
| 20:13.17 | Qwell[] | TexasJay: pastebin.com, pastebin.ca, uuoc.com |
| 20:13.19 | seele_ | TexasJay, pastebin.ca |
| 20:13.19 | luke-jr_work | TexasJay, none of them do? |
| 20:13.20 | Qwell[] | there are many out there |
| 20:13.26 | luke-jr_work | rafb.net |
| 20:13.38 | Qwell[] | rafb.net gives you ugly long urls |
| 20:13.47 | Qwell[] | free |
| 20:13.49 | luke-jr_work | rafb.net is most reliable in my experience |
| 20:13.50 | TexasJay | well, pastebin is working today. Hurrah. |
| 20:13.52 | Qwell[] | erm, stupid swap |
| 20:13.55 | sitedude | Qwell: do you know if asterisk would work with my lightyear alliance voip phone? |
| 20:14.06 | Qwell[] | sitedude: What protocol does it use? |
| 20:14.13 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 20:14.31 | seele_ | someone help me |
| 20:14.44 | luke-jr_work | seele_, too little info |
| 20:15.06 | TexasJay | ok, scratch that. pastebin is being a jerk again. |
| 20:15.18 | Qwell[] | TexasJay: try uuoc.com |
| 20:16.00 | sitedude | Qwell: i'm trying to find out which protocal it works off of |
| 20:16.15 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 20:16.25 | seele_ | luke-jr_work, i need to record a call but no automatically ... only when I want .... to start the record I need to use a key of the phone like * or # |
| 20:16.34 | TexasJay | rafb.net worked fine. |
| 20:16.40 | seele_ | is this possible ? |
| 20:17.41 | pifiu | hey everyone |
| 20:18.14 | b11d | hey |
| 20:18.21 | TexasJay | Ok. I'm trying to use fastpass_get_data() to, well, get data. My PHP is at http://rafb.net/paste/results/zCAues40.html |
| 20:18.28 | TexasJay | My extension info is at http://rafb.net/paste/results/Ge7bT921.html |
| 20:18.33 | TexasJay | What on earth am I doing wrong? |
| 20:19.16 | TexasJay | The console shows the script running, and 4 NoOp commands showing no data. |
| 20:19.56 | TexasJay | My phone sits at "Dialing..." |
| 20:19.57 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
| 20:19.58 | tessier | What's that company other than digium that makes T-1 cards for asterisk... |
| 20:20.04 | tessier | It's on the tip of my tongue... |
| 20:20.15 | tessier | I own one of their cards. Just don't have the box nearby at the moment... |
| 20:20.18 | justinu|laptop | sangoma |
| 20:20.22 | tessier | That's it. Thanks! |
| 20:21.44 | b11d | i wonder how many more years i'll have to listen to students get all hyped up over linux.. |
| 20:21.44 | b11d | linux is OLD NEWS |
| 20:21.44 | b11d | christ. |
| 20:21.44 | b11d | :) |
| 20:21.47 | blitzrage | I'm just getting excited over Win 3.1! |
| 20:21.48 | mog | lol |
| 20:21.50 | b11d | hahah |
| 20:21.56 | pifiu | hey b11d wasup |
| 20:22.03 | mog | if i havent seen it b11d its new to me |
| 20:22.06 | b11d | not much pifiu.. working on this vg224 issue still.. |
| 20:22.11 | b11d | fair enough mog :) |
| 20:22.12 | sitedude | qwell: is there an easy wat to see which protocal my voip works off of |
| 20:22.13 | pifiu | ouch |
| 20:22.21 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 20:22.21 | pifiu | well something to keep you busy! |
| 20:22.27 | b11d | im making progress.. and learning a shitload in the mean time |
| 20:22.30 | b11d | which is fantastic |
| 20:22.33 | pifiu | yeah |
| 20:22.35 | mog | my grandma still gets kicks from electricity |
| 20:22.35 | pifiu | thas always good |
| 20:22.38 | blitzrage | is it possible to adjust gain on a SIP channel? |
| 20:22.40 | pifiu | LOL! mog |
| 20:22.51 | b11d | hahaha |
| 20:23.04 | pifiu | ok brb |
| 20:23.13 | C6Vette | Tell her to stop putting her fingers in the outlets. |
| 20:23.13 | blitzrage | the silence is deafening :) |
| 20:23.15 | C6Vette | :) |
| 20:23.17 | b11d | does she still fear movable typeface? |
| 20:23.24 | sitedude | is there an easy way to see which protocal my voip works off of |
| 20:23.34 | blitzrage | sitedude: sip debug peer <foo> and look at the SDP |
| 20:23.48 | *** join/#asterisk [Outcast] (n=bill@222-154-61-84.jetstream.xtra.co.nz) |
| 20:24.20 | tzanger | So this UGLY woman walks into Wal*Mart with her two kids in tow. The greeter says "Welcome to Wal*Mart. Tell me, are your children twins?" The woman looks at her kids, and back to the greeter. "Uh, no. This one is 9 and the other is 5. Why on earth would you think they're twins?" "My apologies, ma'am," says the greeter, "I just can't believe you got laid twice." |
| 20:24.31 | b11d | HAHAHAHAH |
| 20:24.37 | b11d | thats great! |
| 20:24.40 | blitzrage | lol |
| 20:24.42 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 20:24.48 | C6Vette | lol |
| 20:25.06 | [Outcast] | does it make sense to buy a dual processor system if asterisk only uses one of them |
| 20:25.10 | b11d | yes |
| 20:25.15 | blitzrage | tzanger: that's a good one :) |
| 20:25.18 | b11d | just in case you want to run something other than asterisk |
| 20:25.28 | b11d | you dont want it bogging down the cpu dedicated to asterisk, right? |
| 20:25.31 | blitzrage | I'm totally turning that around on my room mate |
| 20:27.08 | CunningPike | [Outcast]: You can bind your zaptel card's interrupts to one processor and your NIC et al to the other |
| 20:27.54 | b11d | lets get into farming.. |
| 20:28.23 | blitzrage | when I do a top on my dual CPU, it seems that both CPUs are working |
| 20:28.31 | b11d | I have "The Teacher" by Jethro Tull stuck in my head.. |
| 20:28.38 | b11d | im going to have to overplay it.. |
| 20:28.59 | blitzrage | probably just other processes using the 2nd CPU |
| 20:29.24 | CunningPike | b11d: Reef - Place your hands |
| 20:29.48 | b11d | Oh place your hands, on my hope, |
| 20:29.48 | b11d | Run your fingers through my soul, |
| 20:29.48 | b11d | And the way that I feel right now, |
| 20:29.48 | b11d | Oh lord it may go |
| 20:30.19 | blitzrage | ok.... no more Jethro Tull please |
| 20:30.29 | b11d | ok |
| 20:30.30 | blitzrage | more Guns n' F'n Roses |
| 20:30.41 | b11d | haha you're all hard but cant write fuck |
| 20:30.42 | b11d | ;) |
| 20:30.54 | blitzrage | I used to do a little, but a little wouldn't do it, so a little got more and more |
| 20:31.02 | Qwell[] | blitzrage: ! |
| 20:31.08 | blitzrage | lol -- I could.... but I choose not to :) |
| 20:31.11 | b11d | :) |
| 20:31.13 | Qwell[] | blitzrage: /j it :p |
| 20:31.24 | blitzrage | fool me once, shame on you! |
| 20:31.38 | Qwell[] | you could /q it instead, and maybe it'll work :D |
| 20:31.44 | b11d | :) |
| 20:31.46 | tessier | fool me twice...can't be fooled agani |
| 20:31.53 | b11d | hey Bush is here |
| 20:31.53 | b11d | :) |
| 20:32.04 | blitzrage | I love 'the google' |
| 20:32.12 | b11d | all hail the google |
| 20:32.16 | blitzrage | hail! |
| 20:32.22 | b11d | I wonder how long until google's dead? |
| 20:32.25 | b11d | what'll the next thing be.. |
| 20:32.51 | b11d | infinity no doubt. |
| 20:36.17 | iCEBrkr | Altavista! |
| 20:37.38 | *** part/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 20:38.04 | b11d | haha |
| 20:38.08 | b11d | i remember when av was the bomb |
| 20:38.17 | b11d | back in like, 1997 |
| 20:38.17 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 20:38.28 | iCEBrkr | I remember when you typed in 'sex' on Yahoo, you got 55 results. |
| 20:38.36 | b11d | no way :) |
| 20:38.38 | iCEBrkr | way |
| 20:38.45 | b11d | thats impressive |
| 20:39.19 | iCEBrkr | I also remember Damin hooking me up with a SLIP account and me saying "Ok, now what?" |
| 20:39.19 | b11d | ahh.. brb all |
| 20:39.44 | b11d | well, i remember the first time i connected to a BBS at a whopping 1200 baud.. |
| 20:39.49 | iCEBrkr | haha |
| 20:39.54 | b11d | and i just about shit my pants watching that ANSI come across the terminal |
| 20:39.55 | iCEBrkr | b11d: www.cyberdyne.org |
| 20:40.03 | iCEBrkr | speaking of BBS/ANSi |
| 20:40.12 | b11d | NICE |
| 20:40.18 | b11d | that was the shit right there |
| 20:40.21 | b11d | I loved the old scene |
| 20:40.23 | iCEBrkr | hehe |
| 20:40.35 | b11d | good ANSI took skillz |
| 20:40.42 | b11d | thank god RIP never "caught on" |
| 20:40.50 | iCEBrkr | RIP was 5yrs too late |
| 20:41.07 | b11d | yeah |
| 20:41.13 | b11d | is this board operational?? I can call it? |
| 20:41.17 | iCEBrkr | lol |
| 20:41.17 | iCEBrkr | no |
| 20:41.21 | iCEBrkr | I took it down in 1996 |
| 20:41.23 | b11d | oh |
| 20:41.23 | iCEBrkr | or so |
| 20:41.36 | b11d | heh.. I ran "The Final Nightmare" on Renegade for a long time.. |
| 20:41.42 | b11d | i took mine down in 96 too |
| 20:42.01 | iCEBrkr | Basically when I discovered IRC. My phone line was busy all night |
| 20:42.03 | b11d | its when "the web" got popular amongst the BBS community where I was from |
| 20:42.04 | justinu|laptop | PCBoard was 1337 |
| 20:42.12 | b11d | I never used PCB.. I hated Wildcat.. |
| 20:42.12 | iCEBrkr | justinu|laptop: hahahahahaahahhahahah |
| 20:42.23 | b11d | I ran Renegade and for a short while OBV/2 |
| 20:42.28 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:42.31 | iCEBrkr | OH!BlowME/too! |
| 20:42.35 | b11d | :) |
| 20:42.37 | justinu|laptop | seriously... PCBoard out of the box sucked |
| 20:42.41 | justinu|laptop | but you could really bling it out |
| 20:42.41 | *** join/#asterisk evilstoy (i=HydraIRC@83-131-78-209.adsl.net.t-com.hr) |
| 20:42.47 | iCEBrkr | I knew a few people who pimped PCBoard |
| 20:42.52 | iCEBrkr | justinu|laptop: yeah |
| 20:42.55 | b11d | bust out those hex editors lads :) |
| 20:42.57 | justinu|laptop | i lauged at it at first, but i was a convert |
| 20:43.00 | iCEBrkr | It was amazing when I found out it was PCBoard |
| 20:43.32 | tzanger | haha |
| 20:43.46 | tzanger | Telegard 2.5i man |
| 20:43.49 | tzanger | then Renegade |
| 20:43.53 | iCEBrkr | ha |
| 20:44.01 | iCEBrkr | Telegard was pretty bling-able |
| 20:44.16 | iCEBrkr | I based my menu editing off Telegard |
| 20:44.18 | iCEBrkr | kinda |
| 20:44.23 | justinu|laptop | telegard had its limits |
| 20:44.29 | justinu|laptop | but was better out of the box |
| 20:44.33 | tzanger | I was a remote op on both the most hated BBS in the community (Ice-Nine, 519-888-0085) and the one voted most family-friendly, New Gold Dream (725-something) |
| 20:44.46 | tzanger | Ice-Nine rocked |
| 20:44.52 | tzanger | totally, totally rocked |
| 20:45.26 | iCEBrkr | I ran my own, because I was tired of being a 'courier' |
| 20:45.36 | iCEBrkr | Let the demos/ansi's come to me! |
| 20:46.10 | iCEBrkr | I wasn't into warez. |
| 20:48.05 | b11d | i did the warez thing, but we traded tapes in the mail |
| 20:48.14 | b11d | none of this downloading all night at 14.4 shit |
| 20:48.35 | iCEBrkr | haha |
| 20:48.37 | b11d | haha i remember having to think hard about dedicating 10MB of my 20MB drive to my board.. |
| 20:48.47 | b11d | fucking 20MB hard drives.. |
| 20:48.48 | iCEBrkr | yeah, that's why I didn't do warez. |
| 20:48.49 | b11d | i cant even belive that now |
| 20:48.54 | b11d | god im getting old |
| 20:48.55 | iCEBrkr | demo and ansi packs were smaller :P |
| 20:49.00 | iCEBrkr | s3m's and mods |
| 20:49.01 | b11d | terabytes are no big deal today.. |
| 20:49.35 | b11d | in my old town of 2000 people we had 10 boards.. |
| 20:49.40 | b11d | all running renegade :) |
| 20:49.43 | iCEBrkr | ha |
| 20:49.43 | b11d | and BRING BACK FIDONET!! |
| 20:49.59 | iCEBrkr | I was in a fidonet for a short time. |
| 20:50.04 | iCEBrkr | FrontDoor.. ew |
| 20:50.08 | b11d | frontdoor :) |
| 20:50.12 | b11d | qemm, desqview |
| 20:50.13 | b11d | etc.. |
| 20:50.15 | b11d | all great |
| 20:50.16 | iCEBrkr | haha |
| 20:50.22 | iCEBrkr | qemm rocked! |
| 20:50.25 | b11d | fuck yeah |
| 20:50.27 | b11d | it was a godsend |
| 20:50.30 | b11d | I could play doom :) |
| 20:50.35 | b11d | and run Trade Wars 2002 as a door. |
| 20:50.41 | b11d | with my 8mb of ram |
| 20:50.56 | iCEBrkr | and 1meg of video ram that qemm sucked up |
| 20:51.15 | b11d | then MS had to come along with memmaker |
| 20:51.43 | b11d | I enjoyed using TheDrawr, but Aciddraw owned it.. |
| 20:51.55 | iCEBrkr | aciddraw sucked, what are you talking about? |
| 20:51.56 | b11d | i also used TERMINAT for my terminal app |
| 20:52.00 | iCEBrkr | TheDraw kicked ass. |
| 20:52.05 | *** join/#asterisk sitedude (n=sitedude@cpe-076-182-082-228.sc.res.rr.com) |
| 20:52.05 | iCEBrkr | haha terminate |
| 20:52.08 | b11d | thedraw was limited to 25 lines.. |
| 20:52.13 | sitedude | Does anyone know anything about lightyear alliance voip? |
| 20:52.26 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/ss.png |
| 20:52.30 | b11d | I did use Qmodem for awhile |
| 20:52.32 | iCEBrkr | Dude. QmodemPRO rocked |
| 20:52.36 | b11d | yeah |
| 20:52.40 | iCEBrkr | check that screenshot |
| 20:52.57 | b11d | haha thats great! |
| 20:53.10 | iCEBrkr | I found that the other night |
| 20:53.14 | iCEBrkr | I was laughing |
| 20:53.16 | iCEBrkr | 836 calls |
| 20:53.19 | b11d | I should break out my old Colorado 350 and check my old tapes out |
| 20:53.29 | tzanger | heh |
| 20:53.34 | b11d | i wonder if they still work |
| 20:53.34 | tzanger | that looks like my MC screen |
| 20:54.39 | iCEBrkr | My drive took a shit, I still have the tapes |
| 20:54.47 | iCEBrkr | I was pissed |
| 20:56.11 | b11d | ahh the good old days.. |
| 20:56.29 | b11d | I miss playing LORDs |
| 20:56.41 | b11d | or Usurper |
| 20:56.42 | luke-jr_work | can I tell RxFAX to give an error to the remote end? :) |
| 20:56.56 | GaVak | Ever play Crossroads or Global Destruction? |
| 20:56.59 | b11d | no |
| 20:57.04 | GaVak | The used to run on Majornet boards. |
| 20:57.05 | b11d | i missed out on that one |
| 20:57.07 | GaVak | *they |
| 20:57.13 | *** join/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
| 20:57.14 | luke-jr_work | preferably by having the remote end wait for a response, processing the received doc, and then sending back any errors |
| 20:57.36 | iCEBrkr | http://www.fortunecity.com/victorian/jacobean/234/Merr1.html |
| 20:57.36 | iCEBrkr | LOL |
| 20:58.28 | b11d | nice |
| 20:58.32 | b11d | Saskatoon rocks |
| 20:58.48 | b11d | I got put under the table by some 75 year old men drinking scotch till 4am.. |
| 20:59.17 | b11d | ok |
| 20:59.21 | b11d | time to get back on this vg224 bullshit.. |
| 20:59.23 | iCEBrkr | http://www.textfiles.com/phreak/nha-001.txt |
| 20:59.23 | iCEBrkr | omg |
| 20:59.26 | b11d | i've got to square this up |
| 20:59.34 | iCEBrkr | NHA was out of 216 |
| 20:59.44 | iCEBrkr | and Inphinit was a total jackass |
| 20:59.52 | b11d | I was an 807 lad |
| 20:59.59 | b11d | nothing came out of 807 :) |
| 21:00.29 | CunningPike | :) |
| 21:00.44 | iCEBrkr | http://www.flashback.se/bibliotek/index.php?lang=se&group=166&item=1267 |
| 21:00.47 | iCEBrkr | LOL |
| 21:00.49 | iCEBrkr | this is too much |
| 21:01.30 | b11d | CunningPike.. your life was ruined at age 12 then |
| 21:01.58 | b11d | i hate tv so much :| |
| 21:02.14 | CunningPike | b11d: Ya - I don't watch a lot. I was a radio kid |
| 21:02.17 | b11d | good man |
| 21:02.30 | b11d | I still try to catch Coast To Coast AM every night.. |
| 21:02.30 | CunningPike | b11d: Still prefer to listen to the radio than watch tv |
| 21:02.35 | b11d | but its too long |
| 21:03.03 | CunningPike | I grew up with the BBC |
| 21:03.08 | b11d | i shouldnt say theres anything wrong with tv, its just that people give in to it too easilyt. |
| 21:04.49 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 21:06.00 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 21:06.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:06.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:07.30 | Juggie | russel! |
| 21:07.37 | Juggie | +l |
| 21:07.41 | russellb | hi :) |
| 21:07.48 | Juggie | whats up. |
| 21:07.55 | russellb | school .. |
| 21:08.00 | Juggie | sounds fun :) |
| 21:08.40 | russellb | it's ... not |
| 21:09.11 | *** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net) |
| 21:09.42 | CunningPike | Hey russellb |
| 21:13.19 | Qwell[] | CunningPike: Did I ever meet you at astricon? |
| 21:13.22 | Qwell[] | I don't recall :p |
| 21:13.30 | CunningPike | Qwell[]: Not sure :) |
| 21:13.38 | Qwell[] | You'd know it if you did |
| 21:13.45 | CunningPike | Qwell[]: lol |
| 21:14.14 | CunningPike | Qwell[]: I think [hC] pointed you out to me in the CZ, but I didn't get a chance to say hi |
| 21:14.27 | blitzrage | I didn't get a chance to meet like... anyone |
| 21:14.39 | b11d | what do you mean? |
| 21:14.39 | b11d | [03:15] <CunningPike> I ran into Qwell at Astricon.. what a dick.. i hate that guy! |
| 21:14.44 | b11d | :) |
| 21:14.46 | b11d | j/k |
| 21:14.49 | CunningPike | b11d: lol |
| 21:14.54 | syzygyBSD | anyone know how to mount an FTP site? or if it is possible |
| 21:15.01 | b11d | thats not possible |
| 21:15.08 | CunningPike | syzygyBSD: You'll need a ladder |
| 21:15.17 | syzygyBSD | on windows they have webdrive |
| 21:15.28 | b11d | windows is the superior OS. |
| 21:15.35 | syzygyBSD | lmao |
| 21:15.37 | b11d | its light years ahead of anything else |
| 21:15.48 | syzygyBSD | ah... i needed a good laugh |
| 21:15.51 | b11d | :) |
| 21:15.54 | syzygyBSD | but seriously |
| 21:16.14 | russellb | Qwell[]: did I meet you at Astricon? |
| 21:16.30 | syzygyBSD | I realize that there would have to be additional drivers, but I wouldn't think it would be that much harder than any other FS driver... |
| 21:16.37 | syzygyBSD | probably not the right channel for this... |
| 21:16.42 | blitzrage | b11d: makes sense.. that's why Nortel uses it |
| 21:16.56 | luke-jr_work | yay |
| 21:17.09 | luke-jr_work | you can prevent Zaptel bridging by using Monitor :) |
| 21:17.17 | syzygyBSD | nfs isn't that far off... |
| 21:17.25 | iCEBrkr | syzygyBSD: There's a ton of VirtualFS type programs for windows |
| 21:17.42 | syzygyBSD | ya, but I want it for linux |
| 21:17.46 | blitzrage | ok, off to dinner, lates |
| 21:17.48 | syzygyBSD | I already have one for windows |
| 21:18.10 | luke-jr_work | MONITOR_EXEC=/bin/rm; Monitor(); |
| 21:18.12 | luke-jr_work | :) |
| 21:18.15 | iCEBrkr | syzygyBSD: oh.. then isn't there something in Konqueror? |
| 21:18.25 | b11d | hehe |
| 21:18.29 | syzygyBSD | uh... I don't use gui's |
| 21:18.40 | luke-jr_work | syzygyBSD, FUSE |
| 21:18.42 | diclophis-work | does any one know why a SIP channel would not indicating ringing on the calling side? |
| 21:18.49 | iCEBrkr | syzygyBSD: It's um.. 2006. and you're not 15 anymore. Stop being an elitest and get with the program :P |
| 21:20.00 | syzygyBSD | uh, the only thing I use linux for is servers, and I ssh into all of them, get with the program and realize you can't do 1/4 the stuff in a gui you can in the command line |
| 21:20.02 | luke-jr_work | iCEBrkr, GUIs suck for file management |
| 21:20.16 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:20.31 | luke-jr_work | syzygyBSD, get with the program and realize Windows has no place even on the desktop :) |
| 21:20.40 | syzygyBSD | :P |
| 21:20.50 | Juggie | syzygyBSD, uses MSDOS 6.22 For life. |
| 21:20.56 | iCEBrkr | syzygyBSD: Dude, I grew up on DOS 2.11, I was a CLI elitest for years. Because yes, I can type a command line or cd into a file path faster than you can clicky-clicky. |
| 21:20.57 | syzygyBSD | lol, not quite |
| 21:21.40 | iCEBrkr | Yea, even Midnight Commander is too GUI for him. |
| 21:22.02 | syzygyBSD | oh, I use a gui on the desktop, but for server management there is no need for GUI's they waste resources for configuration you shouldn't have to change if it is setup right |
| 21:22.46 | iCEBrkr | The argument of 1998. |
| 21:23.05 | iCEBrkr | Meanwhile you have 3Ghz+++ CPUS and dual core this and 64bit that. |
| 21:23.16 | *** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-49-32.w83-114.abo.wanadoo.fr) |
| 21:23.18 | iCEBrkr | anyhow |
| 21:23.26 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 21:23.37 | iCEBrkr | There's a VFS something or other to mount FTP |
| 21:23.42 | syzygyBSD | maybe you do, my servers are still old |
| 21:23.47 | *** part/#asterisk smig (n=hegemoni@sd-792.dedibox.fr) |
| 21:23.48 | syzygyBSD | thanks |
| 21:24.09 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 21:24.38 | iCEBrkr | hrrm, shit |
| 21:24.40 | iCEBrkr | maybe not |
| 21:24.51 | iCEBrkr | It loaded my FTP up in Konqueror |
| 21:24.53 | iCEBrkr | :-/ |
| 21:24.56 | syzygyBSD | found http://gentoo-wiki.com/HOWTO_FTP_Mount after searching for fuse |
| 21:24.59 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 21:25.15 | syzygyBSD | thanks Luke-Jr |
| 21:25.32 | iCEBrkr | and besides, FTP is so 1992 |
| 21:25.35 | iCEBrkr | SCP |
| 21:25.43 | Manipura | Any Open Source PHP projects need help? I'm going to have some free time later this month |
| 21:25.43 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 21:25.59 | Qwell[] | Manipura: php itself |
| 21:26.02 | Manipura | Asterisk is my first choice for open source contributions |
| 21:26.17 | syzygyBSD | oh, between any servers I can I SCP, but I dont' administrate all the servers I use |
| 21:26.36 | Manipura | Qwell, yeah, but I wana help out asterisk |
| 21:26.37 | iCEBrkr | I've become addicted to scp |
| 21:26.46 | iCEBrkr | and I put winscp on my windows box. |
| 21:27.09 | justinu|laptop | pscp |
| 21:27.38 | b11d | well Qwell, I think that the RTP stuff is right.. |
| 21:27.44 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:28.19 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 21:28.35 | elriah | Hey guys, what's a good softphone for linux? Something not crazy, just simple and works well. |
| 21:28.47 | iCEBrkr | cornfed |
| 21:28.49 | iCEBrkr | lol |
| 21:29.15 | Nugget | There's no such thing as a good softphone. |
| 21:29.46 | iCEBrkr | Nugget: ExpressTalk for windows isn't half bad |
| 21:30.05 | iCEBrkr | Actually, it's one of the better ones. It doesn't clutter your desktop and it doesn't have some gayed-up skin |
| 21:30.23 | elriah | Thanks... |
| 21:30.38 | iCEBrkr | The buttons are NOT duplo-sized like other softphones. |
| 21:30.42 | Nugget | heh |
| 21:30.46 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:30.54 | iCEBrkr | Makes you wonder what the developers screens are. 1600x1200 or something? |
| 21:31.01 | CunningPike | elriah: Not sure iff SJPhone works on Linux, but I've found it to be rock solid |
| 21:31.04 | iCEBrkr | Every softphone I've seen it's 800x600 |
| 21:31.06 | Qwell[] | 1900x1280 |
| 21:31.13 | iCEBrkr | Qwell[]: It's all your fault then! |
| 21:31.39 | Nugget | "whenever a programmer thinks, "Hey, skins, what a cool idea", their computer's speakers should create some sort of cock-shaped soundwave and plunge it repeatedly through their skulls." ( http://jwz.livejournal.com/123070.html ) |
| 21:31.40 | iCEBrkr | A Softphone shouldn't much bigger than Calculator |
| 21:31.54 | iCEBrkr | Someone needs SERIOUS UI training |
| 21:32.04 | b11d | agreed |
| 21:32.12 | iCEBrkr | Nugget: haha |
| 21:32.53 | Nugget | xlite is the worst offender. Who had the dumb idea of mimicing the crappy UI of a mobile phone on a computer? it's like those computer games that make you put in your high score initials with the joystick when there's a keyboard RIGHT THERE IN FRONT OF YOU |
| 21:33.04 | Qwell[] | heh |
| 21:33.04 | iCEBrkr | ExpressTalk is straight forward and to the point.. |
| 21:33.22 | iCEBrkr | Nugget: haha exactly |
| 21:33.44 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 21:34.10 | iCEBrkr | And then there's cornfed, which is CLI |
| 21:34.17 | Nugget | NFS:Porche Unleashed is the only one that's worth playing. :) |
| 21:34.26 | DrkShdw | I remember :) |
| 21:34.57 | DrkShdw | I remember decibull running me into a few overpasses, back in the day |
| 21:35.05 | DrkShdw | or.. was that dbaker? I forget |
| 21:35.19 | Nugget | dbaker sucked at it. he never played because he'd always wreck |
| 21:35.50 | DrkShdw | Thats kinda why I stopped playing :P I'd wreck and watch you guys keep on tooling along |
| 21:36.09 | DrkShdw | they need a NFSGSXRE ;) |
| 21:36.14 | Nugget | heh |
| 21:36.16 | DrkShdw | (NFS GSXR edition) |
| 21:36.25 | justinu|laptop | tourist trophy |
| 21:36.46 | iCEBrkr | Need For Speed GSX-R? |
| 21:36.49 | iCEBrkr | Mmmm |
| 21:36.54 | DrkShdw | yeah, i'd buy it. |
| 21:37.08 | Nugget | I'm getting bored with my bmw and I'm starting to eye a porsche to replace it. |
| 21:37.17 | Nugget | better for autocross |
| 21:37.24 | Qwell[] | Nugget: send the bmw here, I'll pay for shipping |
| 21:37.26 | Nugget | but since I only drive 5k miles a year it's hard to justify |
| 21:37.33 | b11d | no its not |
| 21:37.35 | b11d | its materialism |
| 21:37.38 | b11d | it needs no explanation |
| 21:37.42 | b11d | just quick death :| |
| 21:37.45 | iCEBrkr | Yes! you HAVE to have it. |
| 21:37.50 | iCEBrkr | If you need it or not |
| 21:38.11 | Nugget | just doing my part to keep the wheels of industry movin, b11d. |
| 21:38.16 | Nugget | you should be more appreciative |
| 21:38.17 | b11d | yep |
| 21:38.20 | b11d | you're helping. |
| 21:38.27 | b11d | I thank thee |
| 21:38.43 | DrkShdw | Nugget is directly contributing to the import/export economy of the US. |
| 21:39.06 | b11d | yeah except we need more exports.. not imports |
| 21:40.29 | b11d | fudge.. im running out of options. |
| 21:40.43 | DrkShdw | does the US manufacture anything worth exporting? we manufacture crackheads and such, but I doubt that counts |
| 21:40.43 | b11d | downgrading my IOS to two other versions did not fix this one-way-audio problem |
| 21:40.48 | b11d | we manufacture weapons |
| 21:41.01 | DrkShdw | one way audio? you sure you have RTP forwarded correctly (assuming NAT) |
| 21:41.07 | b11d | no nat.. |
| 21:41.19 | b11d | im almost 100% sure RTP is working the way its supposed to.. but its obviously not. |
| 21:41.41 | b11d | fuck I hate calling the cisco TAC.. they are always jerks. |
| 21:41.47 | fx0 | b11d |
| 21:41.52 | fx0 | which codec are you using ? |
| 21:41.56 | b11d | g711ulaw |
| 21:42.01 | fx0 | that happened to me and it was a codec mismatch |
| 21:42.03 | b11d | it negotiates it fine (according to the logs) |
| 21:42.06 | Qwell[] | It's not a codec thing |
| 21:42.08 | fx0 | try g729 on the asterisk |
| 21:42.09 | Qwell[] | it's an rtp thing |
| 21:42.30 | b11d | i'm willing to give it a shot fx0. |
| 21:42.37 | b11d | im getting down to my last options |
| 21:42.50 | Qwell[] | The rtp wouldn't even start flowing if that were the case |
| 21:42.50 | b11d | I just dont see where RTP is failing Qwell |
| 21:42.55 | b11d | ahh |
| 21:43.17 | Qwell[] | somewhere between asterisk and the vg, the rtp is being lost |
| 21:43.27 | iCEBrkr | Give it a map |
| 21:43.30 | b11d | haha |
| 21:43.34 | b11d | well.. back to the logs. |
| 21:43.35 | Qwell[] | the sdp IS the map :p |
| 21:43.36 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 21:43.45 | Qwell[] | eeps! |
| 21:43.51 | Qwell[] | Now you've done it |
| 21:43.53 | b11d | no gui! |
| 21:43.54 | russellb | yup. |
| 21:43.57 | russellb | i've done it |
| 21:43.59 | Un1x | ok i just talked to digium |
| 21:44.00 | russellb | and now, I RUN AWAY |
| 21:44.12 | Un1x | they say any a Xeon processor 3ghz + will support 98 calls on g729 |
| 21:44.21 | Un1x | and if u have 2 of them u can push abit more |
| 21:44.24 | b11d | there you go |
| 21:44.28 | b11d | are you still on that? :) |
| 21:44.30 | docelmo | Not if you use the hardware codec board.. |
| 21:44.30 | fx0 | i just talked to digium as well, they have 898 dlls yearly from me now. haha |
| 21:44.52 | Un1x | docelmo hardware codec board? |
| 21:45.47 | hads | It's probably slightly more variable than exactly 98 calls :) |
| 21:48.05 | Un1x | well im getting a Xeon chip |
| 21:48.18 | Un1x | screw the amd lol |
| 21:48.42 | b11d | that sounds well thought out.. |
| 21:48.44 | sevard | Did you see that video of Intel VS AMD procesors? They took two processors, the newset at the time, let them run for about 5 minutes then pulled off the fans |
| 21:48.45 | b11d | go for it |
| 21:48.59 | b11d | no |
| 21:48.59 | b11d | what happened? |
| 21:49.09 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 21:49.10 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 21:49.27 | sevard | smoke started billowing out of the AMD chip in under 10 seconds |
| 21:49.32 | b11d | really |
| 21:49.35 | sevard | then intel chip locked up, the fan was replaced on it and functionality resumed like normal |
| 21:49.38 | b11d | that doesnt speak well :/ |
| 21:49.40 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:49.42 | b11d | for amd |
| 21:49.52 | cekc | is there a way from the command line that I could tell asterisk to make one of my sip extensions ring, and when the extension is answered to immediately dial another extension? |
| 21:49.56 | DrkShdw | sevard: I've seen videos facing either way.. I've seen some where the intel smoked.. it's all a bunch of hype :) use the CPU you like |
| 21:50.04 | sevard | intel has this control board on it that says "oh shit i'm really hot, halt all operation immediatly" |
| 21:50.24 | cekc | halt? or slow down |
| 21:50.30 | sevard | more like pause |
| 21:50.35 | Juggie | mobile processors slow down. |
| 21:50.43 | xai | are there any decent linux tools for getting summary network stats for voip? |
| 21:50.44 | sevard | DrkShdw: I haven't seen avideo the either way, this video was from Tom's HArdware, a pretty good resource. |
| 21:51.16 | xai | Is there something that can detect jitter, drop rate, burst that goes through multiple hops? |
| 21:51.40 | sevard | xai: a tool like that would be pretty nifty. If you find/build one tell me. |
| 21:51.48 | DrkShdw | sevard: I'm not saying it was faked. I'm saying others have done the same thing, and the drama happened the other way. bottom line with either chip is: if you run it without a fan, it can be damaged.. which is just common sense (and common knowledge) |
| 21:51.54 | fx0 | why only for linux ? |
| 21:52.01 | b11d | damn this line: |
| 21:52.01 | b11d | *Mar 1 00:08:45.791: :rtp_udp_unreachable:Entered: ICMP unreachable for dest port16655 |
| 21:52.14 | sevard | DrkShdw: Common knowledge, yes, but fans are elecro-mechanical instruments which fail. |
| 21:52.16 | b11d | Nov 2 15:48:11 443: Source IP Address (Media): 10.0.5.5 |
| 21:52.16 | b11d | Nov 2 15:48:11 444: Source IP Port (Media): 16654 |
| 21:52.16 | b11d | Nov 2 15:48:11 445: Destn IP Address (Media): 10.0.5.11 |
| 21:52.16 | b11d | Nov 2 15:48:11 446: Destn IP Port (Media): 19776 |
| 21:52.16 | b11d | Nov 2 15:48:11 447: Orig Destn IP Address:Port (Media): 0.0.0.0:0 |
| 21:52.18 | b11d | oops |
| 21:52.19 | b11d | sorry |
| 21:52.26 | sevard | ouch b11d. |
| 21:52.33 | b11d | ouch? |
| 21:52.38 | sevard | paste, ouch |
| 21:52.41 | b11d | oh yeah |
| 21:52.44 | cekc | MY EYES' |
| 21:52.57 | DrkShdw | sevard: I'm *not* disputing that. by any means. I'm saying.. AMD or intel, run what you like. if the fan fails, you are more than likely replacing either one. |
| 21:53.07 | sevard | very true. |
| 21:53.11 | b11d | if its production anyway |
| 21:53.30 | sevard | I'm under the presumetion that if it's a production server go intel, if you're a home user or gamer and need more bang for your buck, it's always amd. |
| 21:53.31 | Qwell[] | fan failing and the heatsink being removed are completely different :) |
| 21:53.37 | DrkShdw | most motherboards have a CPU thermal protection option, that is really your only hope ;) |
| 21:55.16 | Nugget | Just buy a powermac. :) |
| 21:55.25 | cekc | iBook G4 ftw |
| 21:56.26 | *** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) |
| 21:56.52 | b11d | qwell, could you please take a look at this? |
| 21:56.53 | b11d | http://pastebin.ca/234898 |
| 21:56.59 | b11d | its more rtp debug shit from the vg.. |
| 21:57.45 | qdk | CunningPike: sjphone works great in linux too. |
| 21:57.56 | b11d | looks right to me |
| 21:58.40 | b11d | hrm.. it says it negotiates ulaw, but it does add: |
| 21:58.41 | b11d | Preferred Codec : g729r8, bytes :20 |
| 21:59.18 | Qwell[] | That's fine |
| 21:59.21 | Qwell[] | The interesting part is this |
| 21:59.30 | Qwell[] | Nov 2 15:54:48 793: *Mar 1 00:15:22.603: RTP(8931): fs tx s=10.0.5.5(17010), d=10.0.5.11(10494), pt=0, ts=0, ssrc=1CC50505 |
| 21:59.49 | CunningPike | qdk: Cool |
| 21:59.50 | Qwell[] | and more interesting, is the rx. rtp is flowing between them. Go ahead and pastebin an asterisk rtp debug again |
| 22:00.00 | b11d | ok give me a sec plz. |
| 22:00.20 | qdk | sevard: so... what decade are you presently living in? |
| 22:01.42 | qdk | CunningPike: yes, its nice... instead of the crappy xlite software. |
| 22:02.00 | CunningPike | qdk: Ya - I use it all the time on Mac OSX |
| 22:02.04 | *** join/#asterisk kronic (n=user@mail.stabat.com) |
| 22:02.44 | b11d | qwell.. |
| 22:02.44 | qdk | CunningPike: ;-) |
| 22:02.45 | b11d | http://pastebin.ca/234902 |
| 22:02.54 | b11d | rtp debug, and sip debug for 5454 and 7309 |
| 22:03.40 | Qwell[] | okay, here is what is happening |
| 22:03.56 | Qwell[] | rtp is coming from the phone, asterisk receives it, and sends it on to the vg |
| 22:04.08 | Qwell[] | the vg receives that packet, then tries to send one back... |
| 22:04.13 | Qwell[] | asterisk never receives it |
| 22:04.28 | Qwell[] | both are sending the rtp to the right place |
| 22:04.46 | Qwell[] | now, it could be that the vg is sending on the wrong port |
| 22:04.55 | b11d | that seems likely |
| 22:07.14 | CunningPike | Qwell[]: wireshark? |
| 22:09.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:09.38 | b11d | hello TK |
| 22:09.41 | [TK]D-Fender | y0 |
| 22:10.23 | CunningPike | wb, [TK]D-Fender |
| 22:10.50 | [TK]D-Fender | Not sure how much longer I'll be able to log in at work for the enxt while |
| 22:10.59 | [TK]D-Fender | Getting busy and eyes are shifting |
| 22:12.36 | iCEBrkr | [TK]D-Fender: GET TO WORK! |
| 22:13.55 | justinu|laptop | [TK]D-Fender: noobs everywhere will cry out in agony |
| 22:14.49 | xai | sevard: roger wilco. |
| 22:15.00 | *** join/#asterisk saftsack (n=saftsack@pD9E058EA.dip.t-dialin.net) |
| 22:21.51 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 22:22.19 | b11d | i assume mgcp depends on rtp to function as well as SIP? |
| 22:22.33 | Qwell[] | yes |
| 22:22.38 | b11d | im screwed |
| 22:22.40 | b11d | :) |
| 22:23.07 | b11d | if the vg is sending the packets to the wrong port.. nothing I can do about that. |
| 22:23.25 | b11d | ive tried the newest version of the IOS.. and ive tried the two previous versions too.. |
| 22:23.28 | b11d | so i doubt its an IOS thing |
| 22:23.47 | *** join/#asterisk icallandy (n=andy@cpe-74-72-215-143.nyc.res.rr.com) |
| 22:23.59 | b11d | thing meaning bug :| |
| 22:24.06 | Qwell[] | b11d: and you're SURE that asterisk doesn't have a firewall? |
| 22:24.14 | b11d | i am 100% sure.. i run the box. |
| 22:24.24 | b11d | it DOES have a firewall, but the only rule is to allow packets.. |
| 22:24.32 | b11d | there is nothing being denied |
| 22:24.32 | icallandy | Anyone interested in some part-time work please PM me |
| 22:24.45 | b11d | i'll disable the firewall right now just to be absoutly sure |
| 22:25.19 | b11d | if this fixes it i'll be hanging from a cat5 noose in about ten minutes :) |
| 22:25.33 | *** join/#asterisk ryguillian (n=ryguilli@c-24-12-98-36.hsd1.il.comcast.net) |
| 22:26.37 | *** part/#asterisk ryguillian (n=ryguilli@c-24-12-98-36.hsd1.il.comcast.net) |
| 22:29.20 | b11d | umm.. weird.. now I dont get a dialtone on the vg.. |
| 22:29.21 | b11d | one sec |
| 22:31.13 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 22:33.41 | b11d | qwell |
| 22:33.48 | b11d | there is NO firewall in place |
| 22:33.49 | b11d | nothing |
| 22:33.52 | b11d | same prob.. |
| 22:33.58 | Qwell[] | fun |
| 22:34.06 | b11d | yeah its a blast :) |
| 22:34.16 | b11d | i'll call the TAC ASAP.. maybe put something out on the mailing lists.. |
| 22:35.38 | monsted | b11d: use wireshark to examine and document the issue before attempting a TAC case |
| 22:35.41 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.120) |
| 22:36.08 | b11d | what, do you work for the TAC or something? :) |
| 22:36.30 | monsted | nope, but i've opened enough TAC cases to know what their first question will be :) |
| 22:36.34 | b11d | I will do that though.. i'll capture a few sessions |
| 22:41.00 | zazzizza | <PROTECTED> |
| 22:41.21 | monsted | the Statistics->VOIP Calls menu option is a great tool to trace a call across the network |
| 22:42.49 | b11d | ? |
| 22:43.50 | b11d | good call |
| 22:44.02 | b11d | looking at that, my SIP phone is only transmitting.. not receiving ANY data. |
| 22:44.02 | zazzizza | blld. what happens to be the problem? i didnt read the whole before |
| 22:44.19 | b11d | one-way audio problem.. |
| 22:44.25 | b11d | sip to analog works.. but analog back to sip fails. |
| 22:44.34 | b11d | this is through an asterisk box, and through a cisco vg224.. |
| 22:44.42 | zazzizza | asterisk & cisco? |
| 22:44.44 | b11d | the vg224 is suspected to be the problem |
| 22:44.56 | b11d | yeah look at that.. |
| 22:45.08 | b11d | the rtp data is getting back to the * box, but * isnt passing it back to the SIP phone |
| 22:45.11 | zazzizza | is nat out of question? |
| 22:45.16 | b11d | there is no nat.. no firewall |
| 22:45.25 | zazzizza | codec negotiation? |
| 22:45.34 | b11d | g711ulaw |
| 22:45.38 | b11d | negotiates sucessfully |
| 22:45.39 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 22:46.30 | toerkeium | hello everyone |
| 22:46.32 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 22:46.33 | b11d | hi |
| 22:46.48 | toerkeium | does anyone know of a good VoIP phone for * ? |
| 22:46.54 | ManxPower | b11d: did you make any progress? |
| 22:46.54 | b11d | Polycom series |
| 22:46.59 | b11d | im making some progress |
| 22:47.13 | ManxPower | b11d: Are you running an SMP kernel? |
| 22:47.13 | b11d | on the polycom.. im seeing an tx of 10k/sec but no rx.. |
| 22:47.16 | zazzizza | do you have debugs for both sides? and some basic configs to peek to? |
| 22:47.24 | b11d | yes I am |
| 22:47.39 | b11d | yeah I do.. but really. i've been at this all day and im about to go home.. |
| 22:47.45 | ManxPower | b11d: try NOT running an SMP kernel. There have been a few reports of audio issues with SMP kernels. |
| 22:47.47 | zazzizza | ok no problem |
| 22:48.07 | b11d | this is clearly a network problem of some sort.. |
| 22:48.23 | b11d | the rtp data is making it to the vg224 from the sip phone.. but not back.. |
| 22:48.28 | ManxPower | b11d: I assume your Asterisk box has only 1 IP? |
| 22:48.31 | b11d | yep |
| 22:49.37 | b11d | my sip phone is sending rtp data to the asterisk box, and asterisk is sending it to the vg224.. the vg224 looks like its sending the data back to the sip phone but its not making it.. |
| 22:49.56 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 22:49.58 | ManxPower | b11d: the VG should be sending it to Asterisk |
| 22:50.14 | zazzizza | both endpoints can communicate with each other??? |
| 22:50.21 | h0 | hello all anyone have an ETA for 1.4? |
| 22:50.36 | b11d | the VG is sending it to asterisk.. my mistake |
| 22:50.37 | zazzizza | same net, etc? i've seen ouy posted something about icmp unreachable |
| 22:50.42 | b11d | asterisk isnt passing it back to the SIP phone. |
| 22:50.54 | zamba | what ports do asterisk use? i have an asterisk server behind NAT, so i need to forward some ports to make it work |
| 22:50.58 | b11d | zazzizza.. the analog phone on the vg224 can successfully hear whats being said on the SIP phone. |
| 22:51.07 | b11d | so.. its kind of like they can communicate. |
| 22:51.23 | b11d | zazzizza.. it turns out that unreachable thing is normal |
| 22:52.00 | *** join/#asterisk bjohnson (n=bjohnson@209.195.75.92) |
| 22:53.02 | *** join/#asterisk p1p (i=j@mail.comp911.com) |
| 22:53.58 | zazzizza | i had the same problem yesterday, but it turned out to be that the cisco and the asterisk were talking to each another, rather than between endpoints |
| 22:54.20 | b11d | wtf.. are they supposed to talk to asterisk or each other? |
| 22:54.25 | b11d | im getting conflicting info here then |
| 22:54.38 | Qwell[] | the rtp packets are being sent to the appropriate places |
| 22:54.47 | Qwell[] | b11d: tcpdump it or something |
| 22:55.00 | b11d | to what end? |
| 22:55.14 | Qwell[] | on the asterisk box |
| 22:55.19 | b11d | no i mean, why? |
| 22:55.28 | b11d | its just going to show the same data.. i suspectr.. |
| 22:55.29 | Qwell[] | to see if the packets are making it to the asterisk box |
| 22:55.33 | Qwell[] | they ARE being sent there |
| 22:55.35 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 22:55.42 | b11d | hmm.. fair enough |
| 22:55.48 | b11d | leave no stone unturned and all that |
| 22:55.48 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:55.57 | Qwell[] | from the same call, it would be really useful to see the sdp on both the vg and asterisk, as well as the rtp debug from both, and a tcpdump from the asterisk box |
| 22:56.12 | Qwell[] | if it's all from the same call, we can verify that the rtp ports are right |
| 22:56.24 | zazzizza | it's really simple to come to a conclusion |
| 22:56.41 | zazzizza | put EVERYTHING in the same network/vlan or whatever |
| 22:56.50 | Qwell[] | the vg is definitely getting rtp packets from asterisk, and the vg is definitely attempting to send them back to the right IP (but maybe not the right port...or something) |
| 22:57.07 | zazzizza | if it works, it's something to do with sip handshaking |
| 22:57.14 | b11d | ok, youre a bastard.. |
| 22:57.17 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 22:57.20 | zazzizza | ? |
| 22:57.22 | b11d | no.. it looks like its NOT getting to the * box |
| 22:57.31 | b11d | im seeing my asterisk box send UDP to the vg.. |
| 22:57.35 | b11d | but not seeing anything back from the vg.. |
| 22:57.40 | b11d | I did a few pings in there, and saw that.. |
| 22:57.43 | b11d | so i know its logging right |
| 22:57.46 | b11d | that DOES NOT ADD UP!!! |
| 22:57.49 | zazzizza | sip IS shitty about the set up |
| 22:57.55 | b11d | i should see that traffic without regard to the port being open or not.. |
| 22:58.05 | Qwell[] | b11d: hence the tcpdump |
| 22:58.20 | Qwell[] | but the rtp packets are definitely going to the right place FROM asterisk |
| 22:58.21 | b11d | why do I need to keep learning that lesson |
| 22:58.24 | *** join/#asterisk OneBinary (n=OneBinar@64.161.217.18) |
| 22:58.26 | b11d | yeah.. |
| 22:58.47 | b11d | I wonder if I can setup a span port mirroring whats going across teh port that the vg is attached to, and dump all that traffic.. |
| 22:59.56 | b11d | hrmm... |
| 23:00.04 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 23:02.31 | b11d | tcpdump sucks :/ |
| 23:02.32 | b11d | :) |
| 23:02.52 | b11d | it only captures 202 packets, yet its screaming by on the asterisk console.. and it still reports no packets lost.. |
| 23:02.53 | b11d | what a lie |
| 23:04.16 | b11d | hmmm |
| 23:04.30 | b11d | I just noticed there IS a few packets being exchanged between the vg and asterisk when the call is placed.. |
| 23:04.31 | b11d | that does work.. |
| 23:05.15 | *** join/#asterisk EFI-VJ (n=vj@ool-45756cb1.dyn.optonline.net) |
| 23:05.39 | ManxPower | b11d: look at the IP addresses in the data portion of the packets |
| 23:05.56 | EFI-VJ | I need info about vendors, am I allowed to ask those questions? |
| 23:06.09 | b11d | I am manx.. |
| 23:07.31 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:07.41 | b11d | so.. my sip phone isnt receiving ANY data from the vg.. and neither is the Asterisk box, despite the fact that the cisco logs show its sending rtp data to the Asterisk box. |
| 23:07.54 | Qwell[] | as it should be |
| 23:08.03 | Qwell[] | so, figure out where those packets are going. :) |
| 23:08.14 | Qwell[] | It's certainly not an Asterisk issue |
| 23:08.21 | b11d | well thats the thing.. cisco says its sending to the asterisk box, yet tcpdump shows NO packets coming from that vg.. |
| 23:08.37 | b11d | tcpdump -AEvvv host 10.0.5.5 and host 10.0.5.11 |
| 23:08.41 | b11d | is what im doing |
| 23:09.09 | klasstek | Is there anyone currently located in Denver, CO that would be interested in a full time * gig? |
| 23:09.10 | b11d | ohhhhh.. the polycom's network graph doesnt show data hitting the port.. it shows only its utilization |
| 23:09.20 | b11d | hrm |
| 23:09.42 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 23:09.47 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:09.47 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:09.55 | b11d | packets.. where oh where are you going.. |
| 23:11.22 | zazzizza | blld start from the ground up, test whether conn exists (ie ping) between * & phone, * & vg, vg & phone |
| 23:11.40 | mrichmanM | I have an odd issue that is arising in the last week or so. It is that asterisk will play the all circuits busy message eventhough there are lines available on both internal and external calls |
| 23:11.46 | Qwell[] | the call is up, there is obviously connectivity |
| 23:11.58 | zazzizza | then check the dialpeers on the vg, and the dialplan on * + the contexts |
| 23:12.19 | zazzizza | the call doest seem to be up, otherwise he wouldnt be here |
| 23:12.22 | zazzizza | :-S |
| 23:12.39 | Qwell[] | it *IS* up |
| 23:12.41 | zazzizza | ok ;-) |
| 23:12.42 | b11d | yeah |
| 23:12.43 | b11d | it is |
| 23:12.51 | b11d | the analog phone can hear the sip phone |
| 23:12.54 | b11d | they can both dial each other fine |
| 23:12.59 | mrichmanM | restarting the * service fixes it but it shouldn't be happening right? |
| 23:13.00 | b11d | its just that confounded audio |
| 23:15.05 | *** join/#asterisk saabo (n=saab_va@user-0c8hjld.cable.mindspring.com) |
| 23:15.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:16.04 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:16.47 | b11d | ok.. |
| 23:16.54 | b11d | im going to call it a night.. hit this up again tomorrow. |
| 23:17.52 | b11d | goodnight all.. and thanks again for the discussion |
| 23:18.32 | b11d|bbl | zazzizza.. what can you tell me about dial-peers? |
| 23:18.43 | b11d|bbl | i likely have mine all muddled up.. |
| 23:21.16 | mrichmanM | What sort of things would cause the error NoOp("SIP/721-9e8c", "Dial failed due to CONGESTION") in new stack |
| 23:21.38 | Qwell[] | mrichmanM: see the channel topic |
| 23:22.01 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 23:22.08 | *** join/#asterisk kronic (n=user@mail.stabat.com) |
| 23:22.45 | zazzizza | if you want to ride cisco, then dialpeers are a must to be bored with :S |
| 23:22.59 | b11d|bbl | yep |
| 23:23.28 | mrichmanM | Qwell: Is there something wrong in my question? |
| 23:23.59 | b11d|bbl | get a new version I think |
| 23:24.10 | Qwell[] | mrichmanM: You're using freepbx.. |
| 23:24.14 | b11d|bbl | oh :) |
| 23:24.29 | b11d|bbl | i thought he said Asterisk |
| 23:24.36 | zazzizza | usually u have 4 "legs" in each "conversation" according to cisco, but in the most general case you worry just about whether there exists a match to dial out, and another for dialin |
| 23:24.37 | Qwell[] | he did |
| 23:24.39 | SplasPood | anyone else having issues with Voicepulse connect? |
| 23:24.52 | mrichmanM | it is from a log file |
| 23:25.13 | Qwell[] | mrichmanM: from freepbx |
| 23:25.16 | b11d|bbl | heres my cisco config for all to laugh at: |
| 23:25.29 | b11d|bbl | <PROTECTED> |
| 23:25.34 | mrichmanM | freepx is not installed on the machine |
| 23:25.45 | zazzizza | that's pretty much it. when it's h323 (default) there's not much problems. sip takes an extra effort, though not that much on cisco |
| 23:25.55 | mrichmanM | amp is but is not being used |
| 23:26.48 | Qwell[] | mrichmanM: It is being used. Freepbx kills configs |
| 23:27.10 | *** join/#asterisk saftsack (n=saftsack@pD9E07423.dip.t-dialin.net) |
| 23:27.16 | saftsack | hi |
| 23:27.16 | monsted | b11d|bbl: what kind of box is that? a plain router with a PRI ISDN line? |
| 23:27.24 | b11d|bbl | its a cisco vg-224. |
| 23:27.36 | saftsack | how is it possible to run asterisk without ncurses? |
| 23:27.38 | mrichmanM | if you are saying amp==freepbx then OK |
| 23:27.54 | monsted | oh |
| 23:28.08 | b11d|bbl | do my dial-peers make any sense? |
| 23:28.12 | b11d|bbl | are they fuxxed? |
| 23:28.41 | zazzizza | they do make sense since they enable you to receive and place calls |
| 23:28.53 | zazzizza | can be fucked up? yes |
| 23:28.55 | b11d|bbl | yeah but my config in specific.. |
| 23:29.21 | CunningPike | saftsack: I'm curious as to why you need to |
| 23:29.38 | monsted | i need to find a way to interconnect a cisco call manager express with my asterisk |
| 23:30.01 | zazzizza | as i said, in a outofthebox scenario, you only care about dialpeers. with sip it's that + sip-ua config |
| 23:30.53 | b11d|bbl | did you look at the config? |
| 23:31.07 | zazzizza | nope, where? |
| 23:31.10 | b11d|bbl | <PROTECTED> |
| 23:31.37 | b11d|bbl | ^^^ there |
| 23:33.40 | zazzizza | alright |
| 23:33.44 | zazzizza | looks ok |
| 23:34.11 | b11d|bbl | damn.. i was hoping you'd notice something obviously wrong with it.. |
| 23:34.20 | zazzizza | but |
| 23:34.34 | zazzizza | there is nothing to dial out |
| 23:34.55 | zazzizza | the dialpeer voip`doesnt have a destination pattern or something like it |
| 23:34.57 | b11d|bbl | yeah I know.. i removed that when I redid the config a little while ago |
| 23:35.02 | b11d|bbl | yeah i was setting it to .... |
| 23:35.06 | b11d|bbl | or . |
| 23:35.11 | zazzizza | shouldnt be a problem if you only receive calls |
| 23:35.13 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 23:35.40 | *** join/#asterisk reza_ (i=reza@abort.boom.net) |
| 23:35.57 | b11d|bbl | well thanks for taking a look |
| 23:36.04 | reza_ | i recently upgraded to 1.4-beta2 of asterisk and i'm now getting this error : No channel type registered for 'Zap' |
| 23:36.10 | reza_ | did some configuration file change it's format? |
| 23:36.18 | b11d|bbl | read the changelog? |
| 23:36.25 | zazzizza | nop |
| 23:36.31 | reza_ | the zap module is loaded and compiled, and the card shows up in the dmesg |
| 23:36.34 | zazzizza | where? right there= |
| 23:37.04 | reza_ | i also ge this error : Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 23:37.13 | zazzizza | what's the number of the sip phone? |
| 23:37.21 | b11d|bbl | 2627309 |
| 23:37.22 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:37.31 | b11d|bbl | 7309 also works |
| 23:37.57 | zazzizza | ok but not now, is it? with that config it shouldnt work |
| 23:38.10 | b11d|bbl | no not right now.. |
| 23:38.18 | *** join/#asterisk xAD (n=xAD@85-18-136-111.fastres.net) |
| 23:38.19 | b11d|bbl | it was working for 7309 when I had destination-pattern .... |
| 23:38.42 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:38.50 | zazzizza | ok, but 5.11 is the address of what? asterisk? |
| 23:39.13 | b11d|bbl | yep |
| 23:39.23 | b11d|bbl | i just re-added that destination-pattern |
| 23:39.25 | zazzizza | and what's the address of the sip phone? |
| 23:39.26 | b11d|bbl | dialing to 7309 works |
| 23:39.29 | b11d|bbl | 10.3.237.53 |
| 23:39.35 | b11d|bbl | i can ping 10.3.237.53 from the vg-224. |
| 23:39.48 | reza_ | hmm |
| 23:40.03 | zazzizza | ok and what do you have loaded on *? |
| 23:40.11 | b11d|bbl | what do you mean? |
| 23:40.21 | zazzizza | what's the dial plan that routes the call to the vg? |
| 23:40.23 | reza_ | when i did a module load chan_zap.so, it gave me an error that ZapSendKeypadFacility was already loaded, but then several other things registered, and now it seems to work. |
| 23:40.24 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@71-10-182-240.dhcp.stls.mo.charter.com) |
| 23:40.25 | reza_ | any thoughts? |
| 23:40.42 | b11d|bbl | oh.. exten => 5454,1,macro(standard,${EXTEN},SIP/5454) |
| 23:40.49 | b11d|bbl | 5454 is in sip.conf with the ip of 10.0.5.5 |
| 23:40.58 | b11d|bbl | and "sip show peers" shows 5454 as registered from 10.0.5.5 |
| 23:41.43 | zazzizza | ok, it shouldnt matter since you have it "registered" from the vg's sip-ua "register" command |
| 23:41.44 | b11d|bbl | oh.. no sip.conf has host=dynamic for 5454, but sip show peers shows it at 10.0.5.5 |
| 23:41.49 | b11d|bbl | yeah |
| 23:42.03 | zazzizza | remove register on the vg |
| 23:42.08 | zazzizza | and issue sip server |
| 23:42.17 | b11d|bbl | for the dial-peer or the sip-ua? |
| 23:42.29 | zazzizza | sip server ipv4:10.0.5.11 |
| 23:42.31 | zazzizza | sip-ua |
| 23:42.32 | b11d|bbl | ok |
| 23:42.36 | zazzizza | sip server ipv4:10.0.5.11:5060 |
| 23:43.15 | zazzizza | i'm not sure if you can issue both |
| 23:43.23 | zazzizza | :-S |
| 23:43.28 | b11d|bbl | sip server isnt a command |
| 23:43.35 | b11d|bbl | sip ipv4:10.0.5.11:5060 worked |
| 23:43.37 | zazzizza | sorry |
| 23:43.38 | zazzizza | sip-server |
| 23:43.39 | zazzizza | :S |
| 23:43.48 | b11d|bbl | ok |
| 23:43.57 | b11d|bbl | done |
| 23:44.21 | CunningPike | reza_: It _might_ be the order in which your modules are loading...... |
| 23:44.32 | b11d|bbl | 5454 is now re-registering.. |
| 23:44.33 | b11d|bbl | oen sec |
| 23:44.38 | reza_ | what file controls that? |
| 23:44.41 | b11d|bbl | oh wait |
| 23:44.48 | b11d|bbl | now it doesnt show 5454 as a line to register.. |
| 23:45.00 | OneBinary | OT: are there any good chans/forums for discussing phone hardware (aka cisco -> sip) |
| 23:45.18 | reza_ | need to run... i'll buy you guys late |
| 23:45.19 | reza_ | r |
| 23:45.37 | zazzizza | no, ok but still you can dial out ;) |
| 23:45.48 | zazzizza | registration doesnt mean unreachable |
| 23:45.59 | b11d|bbl | hang on a sec.. somethings weird.. brb |
| 23:46.05 | zazzizza | ha |
| 23:46.22 | zazzizza | it won't get any weirder :S |
| 23:46.38 | b11d|bbl | :) |
| 23:46.47 | b11d|bbl | show sip reg stat |
| 23:46.49 | b11d|bbl | doesnt list anything now.. |
| 23:46.55 | b11d|bbl | dialing 5454 returns a busy signal |
| 23:47.03 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:47.07 | zazzizza | mmm |
| 23:48.37 | zazzizza | i dont want to start changing things as madman but why dont you just have the dial out like 'exten => 5454,1,Dial(SIP/5454) |
| 23:48.46 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:49.21 | b11d|bbl | ok.. |
| 23:49.22 | zazzizza | fuck |
| 23:49.24 | b11d|bbl | change whatever.. this is all test |
| 23:49.32 | zazzizza | wait |
| 23:49.45 | zazzizza | sip/5454 is an endpoint |
| 23:49.55 | zazzizza | but your number on the cisco is 5454 |
| 23:50.12 | zazzizza | so it should be sip/5454/5454 right? |
| 23:50.14 | zazzizza | :S |
| 23:50.16 | b11d|bbl | ? |
| 23:50.29 | b11d|bbl | one sec |
| 23:50.38 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:50.57 | zazzizza | i guess i'm pretty much fucked up as well, i need some pizza |
| 23:50.59 | zazzizza | :-S |
| 23:51.09 | b11d|bbl | sip/5454 could really be "sip/vg224" |
| 23:51.16 | b11d|bbl | i just named it 5454.. maybe I should change that.. |
| 23:51.40 | zazzizza | can be named whatever you want |
| 23:52.57 | b11d|bbl | no dice |
| 23:53.05 | zazzizza | :-( |
| 23:53.17 | zazzizza | i have a c2600 defined in sip.conf |
| 23:53.29 | diclophis-work | ManxPower: you around? |
| 23:53.30 | zazzizza | it's not registered on the * box |
| 23:53.57 | b11d|bbl | hmm |
| 23:54.07 | zazzizza | the voice ports are named 100, 101, 102, ... |
| 23:54.56 | zazzizza | host=<ip address>, type=friend |
| 23:55.07 | OneBinary | Nov 2 15:52:33 NOTICE[17983]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:55.10 | OneBinary | what does that mean? |
| 23:55.20 | b11d|bbl | hmm |
| 23:55.23 | *** join/#asterisk terinjokes (n=spader@adsl-11-150-123.mia.bellsouth.net) |
| 23:55.27 | zazzizza | when i place calls to it i do: ;exten => _90X.,1,Dial(SIP/VoIP-0.28/${EXTEN},10,T) |
| 23:55.37 | b11d|bbl | hmm |
| 23:56.15 | monsted | OneBinary: i got those when using host=dynamic instead of host=ip.of.the.phone with my cisco phones |
| 23:56.23 | terinjokes | hey!!! this is a general SIP question (sorry!) but can anyone recomend a provider that can link a phone number to a SIP |
| 23:57.02 | OneBinary | monsted: really? hmm, i'll try that, but i'd ideally like to have the phones use dhcp |
| 23:57.12 | Ryanw | OneBinary, sip show peers and see if there is an ip address next to the phone |
| 23:57.43 | OneBinary | 365/365 10.0.0.24 D 5060 UNREACHABLE |
| 23:57.56 | OneBinary | 364/364 10.0.0.26 D 5060 OK (9 ms) |
| 23:58.01 | Ryanw | does your asterisk computer have an ip route to 10.0.0.0/24 ? |
| 23:58.12 | Ryanw | never mind, i can see it does. |
| 23:58.26 | Ryanw | can your asterisk computer ping 10.0.0.24 ? |
| 23:58.30 | terinjokes | ? |
| 23:58.57 | Ryanw | OneBinary, i take it 365 is the phone you're having troubles with. |
| 23:59.08 | OneBinary | Ryanw: Yes, ping is ok to 24 |
| 23:59.17 | OneBinary | Yes, I can not call to 365, but i can make calls from it |
| 23:59.46 | OneBinary | I take that back, I can not call 364 from 365 |
| 23:59.56 | b11d|bbl | no dice zazzizza |
| 23:59.56 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 23:59.59 | Ryanw | OneBinary, looks like 365 has lost its registration to the gatekeeper, a reboot of 365 will probably make it re-register |
