00:14.40 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
00:21.52 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
00:36.02 | *** join/#asterisk kaz0358 (n=kurtzogl@asterisk.telecom.ksu.edu) |
00:36.56 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
00:39.42 | kaz0358 | when you create an entry in an enum registrar, you can specify a sip uri. is it valid to specify an extension? ie guest@hostname.com/1234 |
00:41.15 | gaupe | kaz0358: http://www.voip-info.org/wiki/view/ENUM+syntax |
00:46.43 | kaz0358 | gaupe, thanks.. but i'm not for sure that answers my question. if you have several PSTN DIDs listed in e164.arpa and they all point back to the same asterisk server, how do you direct them to the right extension? |
00:48.52 | gaupe | ENUM is just the registry, the call has to be set up with standard SIP - so you will pass the phone number there |
00:49.11 | gaupe | then it's up to you :) |
00:50.44 | kaz0358 | gaupe, so for instance.. if you want to call 1-234-567-8901 and it comes back with a sip entry of guest@asterisk.com.. the equilavent command in asterisk would be dial(SIP/guest@asterisk.com/12345678901,60)? |
00:51.38 | gaupe | yes, I think so - just started looking in to this my self |
00:53.42 | kaz0358 | gaupe, okay.. we i wasn't a 100% sure.. while registering on e164.org i did a search for anyone registered in my city and found 1 hit.. but when i did a lookup, it wouldn't go through.. it appears that he has incorrectly listed his info b/c the sri contains the pstn number... ie guest@asterisk.com/pstn-num.. |
00:54.23 | kaz0358 | s/we\ i/we/; s/sri/uri/; |
00:55.42 | gaupe | seen this? http://www.e164.org/wiki/AsteriskExamples |
00:55.49 | *** join/#asterisk inv_Arp (i=junya@adsl-10-153-159.mia.bellsouth.net) |
00:56.49 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
00:57.20 | kaz0358 | gaupe, no.. i hadn't seen that.. but i was looking at http://www.voip-info.org/wiki/view/RFC+Compliant+ENUM+Macro |
00:58.12 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-210-73.dsl.irvnca.pacbell.net) |
00:58.23 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
01:00.18 | kaz0358 | gaupe, what i'm not getting is why on both the url you gave me and the one i gave you.. you basically find.. Set(DIALSTR=SIP/${ENUM:4}) instead of Set(DIALSTR=SIP/${ENUM:4}/${EXTEN}) |
01:01.49 | kaz0358 | gaupe, that would seem to imply for instance that an enum lookup pointed to an ansterisk server and a default extension.. right? |
01:02.37 | kaz0358 | gaupe, if the sip uri in the naptr doesn't contain the destination extension.. and it isn't being supplied back in the dial script, then you end up with that situation.. right? |
01:06.19 | gaupe | yes, that seems right - SIP/${ENUM:4} should give you the phonenumber too |
01:07.46 | kaz0358 | but the uri doesn't contain the phone number if you make it "guest@asterisk.com/phonenum".. asterisk will think the hostname is "asterisk.com/phonenum" |
01:08.19 | kaz0358 | okay that wasn't clear.. |
01:09.01 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
01:23.59 | *** join/#asterisk talljon84 (n=jonathan@66-188-104-144.dhcp.mdsn.wi.charter.com) |
01:24.11 | talljon84 | Is anyone aware of a SIP client for Palm OS? |
01:24.44 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
01:24.53 | Druken | evening everyone |
01:24.59 | talljon84 | evening Druken |
01:25.01 | kaz0358 | hi druken |
01:25.32 | Druken | so uhmm, yeah... who's got a WORKING wakeup call agi ?? :) |
01:26.03 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
01:26.22 | Qwell | why AGI? |
01:27.05 | Druken | doesn't have to be, just something that works |
01:27.35 | Druken | i guess agi would only be for the phone to set wakeups |
01:27.38 | kaz0358 | druken, you could just setup a cron job to create a .call file |
01:28.12 | Druken | kaz0358: i'm very awear of that... but i'm lookin for the backend to that |
01:28.31 | Druken | i'm being a lazy fuck and don't want to make my own :) |
01:28.55 | kaz0358 | druken, ahh. :) |
01:30.02 | Druken | Qwell: you got something for me?? :) |
01:33.34 | Qwell | Druken: $$$ |
01:33.35 | Qwell | :p |
01:33.51 | Qwell | $$$ talks, rather |
01:34.58 | Druken | Qwell: my checks are so tight, they would have withstood katrina |
01:35.11 | Druken | er.. cheeks i guess |
01:35.28 | Qwell | eh? |
01:36.29 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
01:39.17 | wunderkin | you don't want to say that in here :D |
01:40.22 | Druken | why not? i give back to the community here... |
01:40.43 | wunderkin | as long as you will take it for the team |
01:41.01 | Druken | someone has to... hehe |
01:41.31 | Druken | hehe |
01:43.02 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
01:44.19 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
01:44.51 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
01:48.03 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
01:50.47 | Druken | is it just me, or shouldn't the call files to changed so we can schedule calls into the future? i can think of alot of placed that would come in handy... fax back services, wakeup calls, etc |
01:55.21 | Qwell | Druken: You already can. `touch` it to a specific date/time |
01:55.29 | Qwell | before moving it |
01:57.33 | Druken | oh... so if the creation date is in the future, asterisk will ignore it? |
01:59.22 | Qwell | That's the rumor |
01:59.36 | Druken | ahh, excelent... |
02:02.06 | *** join/#asterisk tomcontr3 (n=gcontrer@62-76-246-201.adsl.terra.cl) |
02:02.59 | tomcontr3 | hello guys, I have found a problem with asterisk, that might be a bug or maybe a bad configuration |
02:04.08 | tomcontr3 | I have 2 trunks cofigured from the same privider, and I can only use the last trunk that I added. |
02:04.21 | tomcontr3 | I thoung I might be a problem from the provider |
02:04.25 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
02:04.57 | tomcontr3 | but, the I configured an X-lite softphone, with one of the trunks |
02:05.11 | tomcontr3 | and I was able to make calls using both lines |
02:05.22 | tomcontr3 | so I must be a problen regarding asterisk, |
02:05.24 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
02:05.36 | tomcontr3 | it |
02:10.33 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
02:15.17 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
02:18.23 | rpm | i finally figured out how to get the wgt634u running in client-bridge mode, except the bridge interface keeps going through the spanning-tree recalculations |
02:28.42 | bsdfreak | heh |
02:30.45 | *** join/#asterisk RES2 (n=RES@chello213047231029.tirol.surfer.at) |
02:31.17 | RES2 | hi |
02:31.24 | ManxPower | ~docs |
02:31.25 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
02:31.26 | ManxPower | ~mailinglist |
02:31.28 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
02:32.41 | RES2 | Are there people, who have spandsp on asterisk running? |
02:33.43 | RES2 | my problem: "loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str" |
02:33.48 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
02:34.52 | wwalker | Anyone know of a reason that the manager interface limits the listen() backlog to 2? I want to set it to 40 instead... |
02:35.58 | *** join/#asterisk Junbug (i=junya@adsl-10-153-159.mia.bellsouth.net) |
02:42.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
02:42.16 | *** mode/#asterisk [+o denon] by ChanServ |
02:44.36 | ManxPower | wwalker, try it and see |
02:44.59 | ManxPower | RES2, did you search the mailing list archives for that error message? |
02:45.22 | RES2 | ManxPower: yes |
02:45.50 | ManxPower | I THINK that error message is either 1) problem with spandsp or 2) a problem with the libtiff libraries, check the spandsp readme and confirm you have a correct version. |
02:46.15 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
02:47.09 | RES2 | ManxPower: I already use the correct version of libtiff and libtiff-devel. :-( |
02:47.13 | wwalker | ManxPower: thx. I know it will fix my problem. I just wonder if anyone knows of another problem it will create that might not be obvious in testing and will bite me hard in production under load. |
02:47.53 | ManxPower | RES2, Try using the code from here if you have not already done so: http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre25/ |
02:48.37 | RES2 | ManxPower: I already use this code. |
02:48.44 | ManxPower | wwalker, The manager interface has problems with handling a large number of connections, which is why astmanproxy was created, as talked about on the mailing lists in the past few days. |
02:48.57 | wwalker | Ah, thx |
02:48.57 | ManxPower | RES2, I don't know what to suggest then. You can ask coppice if he's around. |
02:49.39 | ManxPower | wwalker, astmanproxy also does REALLY COOL stuff like provide a translation layer to talk to the manager interface using things like XML and other types of stuff. |
02:49.56 | RES2 | MaxPower: OK. Thank you. |
02:51.25 | wunderkin | ManxPower, the manager should not have a problem with a large number of connections anymore |
02:51.42 | ManxPower | rpm, That sounds like a network loop |
02:53.00 | ManxPower | wunderkin, Cool. |
02:53.24 | ManxPower | wunderkin, so why is the connection backlog set to 2? |
02:53.39 | wunderkin | i only use like 3 manager connections at a time, but i reported some manager issues awhile back, i would think that should have fixed most of that |
02:53.42 | wunderkin | i don't know what that is |
02:53.53 | ManxPower | wwalker, you can also ask on #asterisk-dev |
02:54.04 | wwalker | thx, will do |
02:56.37 | [hC] | anyone here have/used a linksys wip300? |
02:56.41 | [hC] | mine seems to be... broken? |
02:56.46 | Qwell | [hC]: the new wifi one? |
02:56.50 | [hC] | Yeah. |
02:56.56 | Qwell | I used it VERY briefly at VON |
02:57.00 | Qwell | like, 2 minutes :p |
02:57.09 | [hC] | mine doesnt seem to want to detect any wireless networks anymore |
02:57.17 | [hC] | site survey returns 'no records' and specifying manually just does... nothing. |
02:57.32 | tainted- | broken antenna |
02:57.35 | tainted- | probably |
02:57.42 | [hC] | like either the radio is asleep, or the firmware is messed up |
02:57.45 | [hC] | the antenna is inside the thing |
02:57.54 | [hC] | it did work for a while |
02:57.55 | [hC] | then it just stopped. |
02:58.14 | tainted- | expensive brick |
02:58.33 | *** join/#asterisk johngalt (n=john@eowyn.blacksun.net) |
02:59.14 | RES2 | cu |
02:59.24 | [hC] | looks that way so far |
02:59.48 | [hC] | qwell, i think i might have enough time tonight to load trunk on a box here and try your skinny driver with my 7914 FINALLY |
03:00.02 | Qwell | [hC]: well, you're in luck. I made a branch this morning |
03:00.05 | [hC] | I had to spend all last week writing a billing interface and phone mgmt interface for a client |
03:00.11 | Qwell | team/north/chan_skinny-fixup |
03:00.28 | [hC] | OOooh really. |
03:00.38 | MikeJ[Laptop] | go team north! |
03:00.46 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
03:00.53 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
03:01.09 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
03:01.21 | key2 | !seen kram |
03:01.28 | Qwell | key2: weeks |
03:01.30 | Qwell | ~seen kram |
03:01.34 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 26d 23h 54m 55s ago, saying: 'oh most certainly :)'. |
03:01.41 | MikeJ[Laptop] | key2, yeah, I've seen him.. . |
03:01.48 | key2 | lol |
03:02.06 | MikeJ[Laptop] | I heard him more recently tho |
03:02.28 | [hC] | Qwell: ive never used a team branch, i presume this patches against 1.2.7.1, is that the idea? |
03:02.40 | Qwell | [hC]: nope, it's a full pre-patched version of trunk |
03:02.50 | johngalt | our asterisk system was working fine earlier today but now I recieve incomming calls fine but get dead air when I try to dial out. the phones here are all sip based, 3 grandstream bt-101, and a sipoura adapter for a cordless pots phone. no config changes have been made and the system has been working well for months. any idea what I should be looking at? |
03:03.19 | xachen | johngalt: * in general |
03:03.21 | [hC] | Qwell: oh yeah i see that now that im browsing the tree |
03:03.24 | ManxPower | johngalt, did you reboot? |
03:04.27 | johngalt | yes, I rebooted all the phones, no go...then I rebooted asterisk, then rebooted the phones again. still same issue |
03:04.43 | ManxPower | xachen, how are your calls going out to the PSTN |
03:04.53 | xachen | I just use SIP termination |
03:05.26 | ManxPower | xachen, what does the asterisk console say when you dial out? Use pastebin.ca if it's more than 3 lines |
03:05.34 | johngalt | combination of sip and t1 interface. the strange thing is that we can recieve incomming calls fine. |
03:05.49 | ManxPower | sorry, that was for johngalt |
03:05.50 | johngalt | just can not outdial |
03:05.53 | xachen | ManxPower: I'm not having problems now |
03:06.00 | xachen | I just cracked a funny |
03:06.05 | xachen | hah hah laugh |
03:06.07 | xachen | :) |
03:07.44 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
03:07.51 | johngalt | ideas? |
03:08.12 | xachen | johngalt: Have you stopped asterisk and restarted it? |
03:08.32 | justinu|laptop | turn on sip debug |
03:11.59 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
03:12.32 | johngalt | the reboot should have done that but I have limited access to the box. I have a email in to our main admin to get root but I can't exactly call him. mainly looking for what to look at when I can look further. I realize that only so much can be done without logging in. |
03:14.46 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
03:25.34 | [TK]D-Fender | <PROTECTED> |
03:26.30 | Druken | hey tk, your village called, they want their idiot back :) hehe |
03:28.48 | [TK]D-Fender | Tell them I'm still looking, but its hard to sift though so many here for ours :) |
03:29.02 | Druken | hehe too true |
03:30.13 | Druken | probably he one in here the otherday looking for a provider that allows 1-900 calls :) |
03:31.40 | [TK]D-Fender | Dear God the people we get in here.... Every schmuck who insists their POS modem MUST work with *. And then the semi-enlightened who use all the stuff we divert to other channels for... |
03:32.52 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
03:34.01 | file | hrm, a spoon |
03:34.04 | file | how... |
03:34.05 | file | spoon like |
03:34.13 | [TK]D-Fender | how... astute |
03:34.45 | file | yay |
03:36.28 | tainted- | what's the proper syntax for setting calleridname and calleridnum in dialplan? |
03:37.17 | [TK]D-Fender | Set(CALLERID(num)=1234567890) |
03:37.26 | tainted- | what about name |
03:37.27 | [TK]D-Fender | Set(CALLERID(name)=Joe Blow) |
03:37.36 | tainted- | what variable uses holds name |
03:37.41 | [TK]D-Fender | Assuming 1.2.x of course |
03:37.43 | tainted- | ${CALLERIDNAME} ? |
03:38.00 | [TK]D-Fender | tainted- : the function in reverse ${CALLERID(name)} |
03:38.16 | [TK]D-Fender | tainted- : Go creck out "asterisk functions" on the WIKI |
03:38.43 | tainted- | i lose callername and callerid when i send a call between asterisk boxes |
03:38.47 | X-Rob | or even better 'show function CALLERID' |
03:38.47 | Qwell | spork == school-house shank |
03:39.28 | [TK]D-Fender | tainted- : How are you sending them across? |
03:39.45 | [TK]D-Fender | Qwell : Man.... you outta south-central? |
03:39.55 | Qwell | [TK]D-Fender: I am now! |
03:41.01 | tainted- | [TK]D-Fender just Dial(IAX2/secondserver/${EXTEN}) from first server |
03:41.24 | [TK]D-Fender | tainted- : Shouldn't lose it unless you have a caller-id line for the user end... |
03:43.19 | *** join/#asterisk george____ (n=hanpc14@p161ds3xi.xDSL-1mm.sentex.ca) |
03:43.21 | george____ | Hi |
03:45.43 | [TK]D-Fender | SHHH!!!! You'll wake the crickets! |
03:53.44 | *** join/#asterisk bmg505 (n=leon@dsl-146-24-53.telkomadsl.co.za) |
03:55.57 | [TK]D-Fender | Ok, I'm baked... outta here. later all. |
03:56.02 | *** join/#asterisk esculapio_ (n=ESCulapi@224stb68.codetel.net.do) |
04:02.06 | *** join/#asterisk Mark987 (n=Nikk@modemcable068.243-131-66.mc.videotron.ca) |
04:04.24 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
04:16.28 | *** join/#asterisk techie (n=gus@antibala.com) |
04:18.57 | rpm | bahah, i love this telemarketter torture.. (fwdtel: 712906) |
04:23.27 | *** join/#asterisk gursikh (n=m@adsl-68-92-63-196.dsl.hstntx.swbell.net) |
04:25.23 | *** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
04:35.12 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
04:37.52 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
04:41.31 | coppice | RES2: don't mix softeware versions |
04:43.14 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
04:44.46 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
04:45.02 | *** join/#asterisk kn0x (n=atlantic@c-71-194-235-251.hsd1.il.comcast.net) |
04:48.30 | *** part/#asterisk kn0x (n=atlantic@c-71-194-235-251.hsd1.il.comcast.net) |
04:55.31 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
04:57.57 | *** join/#asterisk litage (n=nick@203.220.55.70) |
05:02.18 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
05:08.16 | Alystair | hmmmmmmmmmmmmm |
05:10.03 | *** part/#asterisk Mark987 (n=Nikk@modemcable068.243-131-66.mc.videotron.ca) |
05:13.13 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
05:15.21 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
05:21.23 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
05:29.05 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-170-44.dsl.chcgil.sbcglobal.net) |
05:29.18 | Flauto | anyone here tried to use icall with asterisk? |
05:32.50 | tainted- | Fellatio: what is icall? |
05:40.08 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
05:46.45 | Alystair | Is there a way to do some neat integration between asterisk and our ActiveDirectory server at the office? |
05:46.52 | Alystair | Eg. when I make a new account it gives them an extension etc |
05:49.49 | Alystair | ignore the question |
05:54.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:00.35 | *** join/#asterisk blackgecko (i=blackgec@201.152.14.187) |
06:01.56 | blackgecko | anyone has had problems with the tdm2400p ?? im using it but it randomly strips some numbers and the dialed number is incorrect, any idea ? |
06:03.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
06:04.21 | Flauto | tainted, it is a free service, check out www.icall.com |
06:04.37 | websae[A] | icall |
06:04.39 | websae[A] | what is that? |
06:04.58 | *** join/#asterisk kavit (n=kavit@60-240-44-231.static.tpgi.com.au) |
06:05.07 | websae | what is icall about? |
06:05.27 | Flauto | it is a free service |
06:05.36 | Flauto | the enables calls to usa |
06:05.40 | Flauto | for free |
06:06.01 | websae | that wont last long |
06:06.02 | Flauto | but i am trying to figure out how to make it to work with asterisk |
06:06.04 | kavit | say is it possible to use asterisk as the gateway for video calls with h323 driver? |
06:06.20 | websae | h323 sucks |
06:06.25 | websae | especially on asterisk |
06:06.38 | Flauto | tried i tried it and it works only sometimes |
06:06.43 | Flauto | i mean video |
06:07.29 | Flauto | websae, i got the windows client for icall and it works |
06:07.42 | Flauto | i tired a few calls and it sounded okay |
06:07.54 | Flauto | but the thing is that i want to use it through asterisk |
06:07.59 | kavit | well i just want to use it as a gateway and pass the call to an mcu |
06:08.06 | coppice | icall claim better sound quality than skype. i wonder what they use |
06:08.06 | Flauto | i tired a few things and it did not work |
06:08.36 | Flauto | cppice, it uses sip |
06:08.47 | blackgecko | anyone has had problems with the tdm2400p ?? im using it but it randomly strips some numbers and the dialed number is incorrect, any idea ? |
06:08.51 | Flauto | but i dont' know what codec they use |
06:09.41 | coppice | you should be able to find something about the codec from the SIP messages |
06:10.29 | Flauto | when i use that client, it pops a notpad file |
06:10.37 | Flauto | with some info |
06:10.48 | Flauto | so i figured that the server is beta.icall.com |
06:10.52 | Flauto | and they are using sip |
06:11.10 | Flauto | but for authantication, i am not sure |
06:12.11 | coppice | signup by invitation only. how very exclusive :-) |
06:14.06 | Flauto | coppice, use FLAUTO |
06:14.15 | Flauto | as invitation |
06:21.05 | *** part/#asterisk blackgecko (i=blackgec@201.152.14.187) |
06:21.57 | Flauto | coppice, are you there? |
06:22.18 | coppice | yep |
06:23.03 | Flauto | did you get the invitation |
06:24.03 | Flauto | Apr 30 01:23:37 NOTICE[10595]: chan_sip.c:9548 handle_response_invite: Failed to authenticate on INVITE to '"Zhao Liu" <sip:flauto@beta.icall.com>;tag=as732b7366' |
06:24.10 | Flauto | i got this when i tried to call |
06:24.29 | coppice | I didn't try to sign up |
06:24.37 | Flauto | oh |
06:25.18 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
06:26.31 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
06:27.22 | *** join/#asterisk rdgzt (n=joakim@201.137.86.15) |
06:27.40 | Flauto | would anyone here be interested in figuring out how to make icall to work with asterisk? |
06:28.05 | rdgzt | I'm having a weird problem, despite my best efforts, I can't register via SIP, asterisk tells me the password is wrong even when I'm sure it isn't. |
06:30.41 | rdgzt | Doesn't matter what I set as secret in sip.conf or anything. |
06:32.55 | Flauto | it does matter |
06:33.22 | rdgzt | Well, yes, obviously, but I'm sure the secret in sip.conf and the password I'm using in the softphone are the same. |
06:33.30 | rdgzt | I've tested with two different softphones to be sure. |
06:33.38 | websae | Flauto: by the time you figure out how to connect to icall they will be shut down because of loss |
06:33.46 | rdgzt | I'm following the example setup in Asterisk: The Future of Telephony. |
06:34.23 | Flauto | websae, hehe, possible, but they must have some way to be able to provide free service |
06:34.37 | rdgzt | And I've used both the xten softphone and ekiga, both give the same result. |
06:34.50 | Flauto | rdgzt, show your sip.conf settings |
06:34.54 | Flauto | it might tell something |
06:35.42 | websae | sure....they incorporate it as a startup cost.............and then they a) either have to start charging the end user a nominal fee to start making up for overhead, etc b) they shut down, and even if they do a) they could become like nufone and not have the capacity to support paying customers |
06:36.12 | rdgzt | Flauto: Ok to paste here? |
06:36.21 | Flauto | not here |
06:36.25 | Flauto | pastebin.ca |
06:37.02 | rdgzt | Ok, let me see... |
06:38.28 | rdgzt | http://pastebin.ca/52345 |
06:38.36 | Flauto | is insecure=very still working in asterisk 1.2? |
06:38.46 | rdgzt | That has my sip.conf, and also, below the ---, what asterisk says when I try to register. |
06:38.54 | Flauto | or it is port,invite, that kind of things now |
06:39.11 | Flauto | okay |
06:39.47 | rdgzt | It's really weird, it's an incredibly basic config, and it just doesn't want to work. |
06:39.51 | rdgzt | Kind of has me stumped. |
06:42.42 | rdgzt | It was badly wrapped in pastebin.ca, I updated it now. |
06:43.10 | Flauto | i can not open it |
06:43.35 | rdgzt | No? Let me see. |
06:43.55 | rdgzt | http://pastebin.ca/52346 |
06:44.00 | rdgzt | That doesn't work for you? |
06:45.44 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
06:46.20 | Flauto | it does not |
06:46.33 | rdgzt | What do you get as an error message? |
06:49.04 | tainted- | Flauto what's icall all about |
06:49.25 | websae | free |
06:49.29 | tainted- | is it softphone? |
06:49.33 | websae | beta voip |
06:49.40 | tainted- | SIP? |
06:49.50 | websae | free softphone interface |
06:49.52 | Flauto | yes |
06:50.03 | websae | to make long distance PSTN calls fro free |
06:50.06 | tainted- | is it locked to icall netowrk? |
06:50.26 | websae | do you know if they even offer sip connections via your own sip client? |
06:50.41 | websae | can terminate to PSNT lines free |
06:50.46 | Flauto | tained, if you want to register, you can use FLAUTO as invitation code |
06:51.04 | Flauto | that is the part i don't know |
06:51.13 | Flauto | i just see the connection they use is sip |
06:54.11 | kavit | Does asterisk work well with ISDN <---> H323 for video conferencing? Can't seem to find a definitive answer |
06:54.54 | websae | H323 is a pian to get going on asaterisk to begin with |
06:55.00 | websae | so i would not recommend that at all |
06:55.25 | kavit | well can it be done, pain aside? |
06:55.49 | websae | do you like pain where you feel like you're at the point of death? |
06:56.27 | kavit | no but if I have to endure it to get something working |
06:59.03 | kavit | basically I want to know if I can get asterisk to relay ISDN video calls as H323 to a gatekeeper |
07:00.47 | bmg505 | isn't a video call just a straight data connection that u can handle with v4l? |
07:01.28 | bmg505 | s/v4l/i4l/ |
07:01.44 | bmg505 | :) |
07:02.21 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
07:03.10 | distortion | ok, im reduced to asking- what is the substring function in c? |
07:03.47 | distortion | i want to effectively do this: ${EXTEN:4} in an asterisk application |
07:14.40 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
07:21.18 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
07:33.49 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
07:33.55 | *** join/#asterisk gr0mit (n=Gr0mit@extrt.txrx.org.uk) |
07:40.34 | bmg505 | distortion: memove could do the job in straight ansi C |
07:40.45 | bmg505 | memmove |
07:41.29 | bmg505 | be mindful of the strcpy command as it is know to not do what u expect |
07:42.01 | distortion | memmove(temp, temp+4, 11); -- is what i was playing with, cool |
07:42.07 | distortion | thx bmg |
07:47.12 | distortion | sweet that worked!! |
07:51.56 | *** join/#asterisk lorinc (n=ang@caracas-1478.adsl.interware.hu) |
07:54.36 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
07:58.24 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
08:01.32 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
08:26.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:46.46 | *** join/#asterisk netsurfer (n=bbjunkie@dreambox.myvnc.com) |
08:59.02 | Alystair | are polycoms really good phones? |
09:00.17 | Alystair | I mean is the price justified |
09:01.37 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
09:01.46 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
09:04.38 | *** join/#asterisk Olobola (n=casper_s@netblock-68-183-67-158.dslextreme.com) |
09:05.38 | Olobola | dudes, can't I just bypass a provider and recieve phone calls in some magical way? I can't wait around for this shit! |
09:09.09 | tainted- | Olobola what do u mean? |
09:09.34 | evilbuny | Olobola: you mean having the call go directly to your SIP device? |
09:10.22 | *** join/#asterisk kavit (n=kavit@203-158-58-2.dyn.iinet.net.au) |
09:22.25 | Olobola | oh I dunno -- I used to go through a webhost, now I host my own site and consequently never have to worry about downtime. I wish I could do the same with incoming toll free service. |
09:25.35 | Alystair | outsource to india :D |
09:27.38 | evilbuny | Olobola: you technically can |
09:27.38 | evilbuny | although you are still dependent on carriers to route the call |
09:33.57 | Alystair | why isn't there a nice wiki which has reviews of all these ip phones |
09:35.02 | kristalino | Alystair, i agree |
09:35.35 | coppice | Alystair: start one :-) |
09:35.55 | Alystair | only if I get free phones ;) |
09:37.10 | coppice | a sort of tomshardwareoverip, where you get more than just free phones - you get kickbacks :-) |
09:37.53 | evilbuny | Olobola: http://voip.wikispaces.com/IPDialling |
09:38.27 | Alystair | heh |
09:47.50 | *** join/#asterisk MaddieBoi (n=MaddieBo@210-84-15-248.dyn.iinet.net.au) |
09:55.59 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
09:56.12 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
10:04.13 | *** join/#asterisk ToTo (n=ToTo@host235-158.pool875.interbusiness.it) |
10:26.49 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
10:29.01 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
10:47.56 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
11:07.42 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
11:08.12 | SheriF_WorK | i have diguim card with 2 moduls / FXS and FXO only. what additional modules should i load for zaptel to work ? |
11:09.01 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
11:13.33 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
11:16.42 | key2 | none |
11:23.11 | SheriF_WorK | key2: sure ? |
11:26.06 | *** join/#asterisk viperdudeuk (n=viperdud@84-45-168-60.no-dns-yet.enta.net) |
11:28.24 | robin_sz | Alystair, there cant be a wiki about the IP phones for legal reasons |
11:29.28 | robin_sz | because eventually, someone will review the GXP2000 and then we'll be in trouble with the ISP for bad language and obscenities |
11:30.04 | poisoner | hrhr |
11:30.21 | poisoner | Has anyone experiences with chan_sccp? |
11:31.20 | poisoner | I run into problems if I call a meetme-number from a skinny phone |
11:33.04 | viperdudeuk | poisoner: there is a chan_sccp mailing list you might be better asking on there. However I think there is a issue with chan_sccp and MeetMe |
11:33.19 | poisoner | yap |
11:33.27 | poisoner | The question is who to blame for it |
11:33.48 | robin_sz | he is |
11:33.51 | viperdudeuk | well there is a guy called Sergio that maintains the code |
11:34.23 | viperdudeuk | but as its open source and essentially born out of good will I wouldn't think complaining will get you anywhere |
11:34.43 | poisoner | the chan_sccp developers for letting chan->type be NULL or the developers of app_meetme.c for not putting a check before stcapcmp'ing chan->tpye... |
11:34.51 | robin_sz | I had lots of problems with chan_sccp and some DECt->sccp adaptors |
11:35.06 | robin_sz | the biggest problem being the docs :( |
11:35.11 | poisoner | yap |
11:35.23 | viperdudeuk | chan_sccp is not officially part of * which is the main reason for the issue |
11:36.00 | robin_sz | it was assumed you knew how it works and great chunks of "what to do to get it going" were not mentioned at all |
11:36.10 | poisoner | viperdudeuk: but not checking before comparing a string... isn't this a generic fault? |
11:36.39 | poisoner | chan_skinny perhaps crashes asterisks as soon as my 7940 want's to register |
11:36.54 | viperdudeuk | i agree but if you feel strongly about it sumbit a patch |
11:37.18 | poisoner | for app_meetme.c ? |
11:37.25 | viperdudeuk | we have stopped using chan_sccp with our 7960 due to crashes |
11:37.45 | viperdudeuk | whatever fixes it |
11:38.02 | poisoner | viperdudeuk: What are you using now? chan_skinny oder SIP-Firmware von 7960? |
11:38.06 | poisoner | s/von/on |
11:38.09 | viperdudeuk | you might want to read the chan_sccp mailing list for more details |
11:38.17 | viperdudeuk | SIP firmware |
11:38.42 | robin_sz | ... |
11:38.46 | viperdudeuk | we only tried sccp as we wanted to use 7914's, however in the end we went with FOP |
11:38.50 | robin_sz | has it happened yet? |
11:38.58 | poisoner | a coworker thinks that the SIP-firmware has missing some features... |
11:39.06 | poisoner | FOP? |
11:39.07 | viperdudeuk | such as? |
11:39.20 | viperdudeuk | Flash Operator Panel |
11:39.38 | poisoner | the "conference" thing |
11:39.46 | viperdudeuk | google it |
11:39.47 | poisoner | but in 7.5.something it worked |
11:40.07 | viperdudeuk | 3 party conf works on 7.5 on 7940's I use it at work |
11:40.37 | poisoner | yap.. This is wat I discovered. |
11:40.50 | poisoner | only 3party? or also more? |
11:41.03 | viperdudeuk | while we are talking SIP does anyone know how to save the debug to a file? |
11:41.32 | viperdudeuk | 3 party only on the phone |
11:42.02 | poisoner | at work, with our CCM an sccp firmware I think it could be more... |
11:42.03 | viperdudeuk | 7940's at least not tried on a SIP 7960 |
11:42.14 | viperdudeuk | ok |
11:42.17 | poisoner | ok... thx for the infos. |
11:51.53 | poisoner | iiiek... |
11:52.21 | poisoner | viperdudeuk FOP uses swf as the name says, right? |
12:05.26 | viperdudeuk | yes |
12:26.39 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
12:27.50 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
12:33.18 | *** join/#asterisk ToTo (n=ToTo@host235-158.pool875.interbusiness.it) |
12:36.35 | *** join/#asterisk ToTo (n=ToTo@host235-158.pool875.interbusiness.it) |
12:37.10 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
12:55.33 | *** join/#asterisk saftsack (n=saftsack@p54A7EC2D.dip.t-dialin.net) |
13:08.11 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
13:13.35 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
13:16.18 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
13:16.32 | stoffell_h | hm, is "avoiding initial deadlock for sip/..." a bad thing? |
13:17.35 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
13:19.10 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
13:20.19 | gaupe | stoffell_h: I'm seeing it when a phone does a redirect, at the same time I got error messages all over the console |
13:21.45 | stoffell_h | gaupe, ah, weird. I have this happening on an asterisk 1.2.7 box.. could be related to this maybe: http://bugs.digium.com/view.php?id=7004&nbn=6 |
13:23.16 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
13:23.45 | gaupe | stoffell_h: have seen that one with call parking and unattended transfers, but that bug is supposed to be fixed at 1.2.7 |
13:24.41 | gaupe | this is what I'm seeing - http://bugs.digium.com/view.php?id=4101 |
13:27.15 | stoffell_h | oh, that's a different one i guess, yep |
13:31.11 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
13:31.12 | *** join/#asterisk frk2 (n=kvirc@202.141.251.102) |
13:31.20 | frk2 | whatsup people |
13:31.28 | frk2 | whos working on a sunday? :) |
13:32.01 | frk2 | guess its just me :) |
13:32.57 | frk2 | i just wanted to tell everybody of my revived sudden faith in pa1688 based phones |
13:33.16 | lesouvage | I added exten => 300,1,MeetMe(,MD,) to my configuration. The idea is to be able to open a conference room with a number of choice and be prompted to also add a pincode to the conference. |
13:33.39 | lesouvage | The problem is that I'm not prompted to add a pin code to the new conference room. |
13:34.45 | lesouvage | Is this supposed to work (starting new conference room if added number does not exist and be asked to add a pin on the new conference room) |
13:38.25 | frk2 | is anybody using the atcom's and has this sudden 'silence' in the middle problem? |
13:38.33 | frk2 | the phone just stops for like a second |
13:38.55 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
13:41.23 | frk2 | it dont happen in the older firmware |
13:41.30 | frk2 | the newer firmwares seem to have this issue though |
13:42.32 | Zeeek | frk2 see the yahoo group on PA1688 |
13:42.46 | *** join/#asterisk bmonty (n=bmontgom@ubuntu/member/bmonty) |
13:43.08 | frk2 | i did zeeek |
13:43.14 | frk2 | seems like a unresolved issue |
13:43.23 | Zeeek | go back to older firmw |
13:43.27 | frk2 | i hate these yahoo groups.. such bad navigation |
13:43.31 | Zeeek | yeah |
13:43.42 | Zeeek | but that one's so quiet it don't matter so much |
13:43.44 | frk2 | zeeek i cant find the older firmware anyways |
13:43.54 | frk2 | do you know where i could get it from? |
13:43.59 | Zeeek | heh you should have been saving them all |
13:44.01 | frk2 | aredfox only has the latest one |
13:44.03 | frk2 | hahah |
13:44.06 | frk2 | which one do you use? |
13:44.11 | Zeeek | which do you need? |
13:44.22 | Zeeek | I think .49 but I don't have the phone here |
13:44.30 | frk2 | 1.49 has issues too |
13:44.37 | frk2 | the SIP version atleast |
13:44.43 | Zeeek | I don't use the phones much (I have three of them) |
13:44.53 | Zeeek | SIP? Who uses SIP on those IAX phones? |
13:45.06 | Zeeek | shit if I want SIP I'll use a real phone :) |
13:45.12 | frk2 | hahah |
13:45.14 | frk2 | hmmm |
13:45.22 | frk2 | so you saying their iax stuff is better than sip? |
13:45.25 | Zeeek | I think I tried SIp once to see if it worked |
13:45.32 | Zeeek | no it's IAX though |
13:45.44 | Zeeek | I never used SIP with those things. The main advantage is they do IAX |
13:46.01 | frk2 | true |
13:46.06 | frk2 | the phones actually arent that bad |
13:46.21 | frk2 | the voice quality is okkkk.. nothing great. but what do you expect for $40 |
13:46.28 | frk2 | they dont crash much |
13:46.43 | Zeeek | depends on the phone. But the idea of compilable firmware is cool |
13:46.44 | frk2 | and look decent |
13:46.59 | Zeeek | all PA1688 are not the same |
13:47.14 | frk2 | the atcom's have a pa1688S |
13:47.25 | frk2 | im guessing all pa1688S's are essentially the same |
13:47.27 | frk2 | no? |
13:47.32 | Zeeek | the white atcom? |
13:47.36 | frk2 | black atcom |
13:47.42 | Zeeek | of course they're not! |
13:47.50 | Zeeek | different keypads and speakers |
13:47.53 | frk2 | oh yeah |
13:47.57 | Zeeek | and plastic case |
13:47.57 | frk2 | well other than that i mean |
13:48.04 | frk2 | functionality wise |
13:48.12 | Zeeek | those are the most important features of most phones |
13:48.28 | frk2 | none of my clients mind the quality of the black atcom much |
13:48.30 | Zeeek | the atcoms I have, white ones aren't the greatest but they work |
13:48.44 | frk2 | the white ones look too chinese |
13:48.45 | frk2 | :) |
13:48.46 | Zeeek | the black which I think is a yixin |
13:48.49 | Zeeek | is cool |
13:49.05 | Zeeek | the women prefer the white ones... that's what cracked me up |
13:49.10 | Zeeek | at the office |
13:49.13 | frk2 | hahahah |
13:49.15 | frk2 | really? |
13:49.18 | frk2 | i hate the white ones |
13:49.20 | Zeeek | I like the black one |
13:49.29 | Zeeek | what is the model number of your black phone? |
13:49.35 | frk2 | if i could only solve the 'voice gap' issue on these phones.. i think they're a good bargain |
13:49.38 | frk2 | AT-320 |
13:49.42 | frk2 | i think you have the AT-323 |
13:49.48 | Zeeek | I think I have the same in white |
13:50.00 | frk2 | no no- hang on |
13:50.10 | Zeeek | actually mine arenet branded atcom but I guess there about the ame |
13:50.19 | frk2 | 'oh yeah |
13:50.24 | frk2 | did you get it for $40 |
13:50.32 | frk2 | thats the price atcom sells t |
13:50.33 | frk2 | at |
13:50.37 | frk2 | $35 in wholesale |
13:50.45 | Zeeek | I've had it for a long time. At that time they were $50 |
13:50.48 | frk2 | ive been able to get them down to $32 |
13:50.53 | Zeeek | $60 with a HUB* |
13:51.03 | frk2 | oh this is $32 with a hub |
13:51.09 | Zeeek | I don't need a bunch of cheap phones at the moment |
13:51.11 | frk2 | i ordered 150 of them |
13:51.33 | Zeeek | I wonder what 150 Sipuras would cost? |
13:51.37 | frk2 | exactly |
13:51.41 | frk2 | thats the problem |
13:51.50 | frk2 | in a third world country, the cost of phones is unbearable |
13:51.53 | Zeeek | what? Double? They're 5 times better |
13:51.55 | frk2 | i need to find a cheap ass phone |
13:52.04 | frk2 | Sipura phones? |
13:52.15 | Zeeek | I think it's more that no one gives a shit about whether the user is comfortable |
13:52.37 | Zeeek | anyway, working for $20/month, how comfortable ya need to be? |
13:52.56 | frk2 | you mean $200/month |
13:53.06 | Zeeek | not that much difference |
13:53.12 | frk2 | lol |
13:53.24 | Zeeek | unless an appartment is $25/month |
13:53.39 | frk2 | hmm |
13:53.40 | Zeeek | anyway, those are good starter phones but that's it IMO |
13:53.48 | frk2 | i know |
13:53.53 | frk2 | you using 841s? |
13:53.59 | Zeeek | lots of fun to mess with too |
13:54.03 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
13:54.18 | Zeeek | I have one 941 - it's very good speaker sound |
13:54.27 | Zeeek | boots in about 10 seconds |
13:54.39 | frk2 | see my prob is that 941 is like $130? |
13:54.46 | frk2 | for that price i'll just get a aastra |
13:54.48 | Zeeek | 150 of them might be a lot less |
13:54.57 | Zeeek | aastra is supposed to be decent |
13:55.05 | frk2 | aastra is awesome |
13:55.10 | Zeeek | Grandstreams are "deprecated" now |
13:55.24 | Zeeek | the first one I had was decent though, worked fine |
13:55.26 | frk2 | ive had too many issues with the GXPs |
13:55.39 | Zeeek | I'm talkin BT100 |
13:55.43 | frk2 | The 102 is okay |
13:55.51 | frk2 | but then its not too much better than the atcom |
13:55.57 | frk2 | similar build/voice quality |
13:56.29 | Zeeek | is anyone using IAX providers ? |
13:56.37 | robin_sz | sip phoens will be like $30 within 12 months |
13:56.37 | frk2 | i use voicepulse |
13:56.49 | frk2 | robin i hope so |
13:56.50 | Zeeek | yeah me too, works ok |
13:56.55 | frk2 | that'll jump start the industry |
13:57.07 | Zeeek | look at the price of cellphones! |
13:57.08 | frk2 | the main problem is the cost of a phone |
13:57.11 | robin_sz | frk2, you just have to watch whats happening with the Big Players ... |
13:57.27 | frk2 | zeeek cell phones are personal items |
13:57.31 | robin_sz | even BT are pushing VOIP now ... |
13:57.33 | frk2 | ip phones are usually corporate items |
13:57.55 | Zeeek | the point is, sell the service and give the phone away. Coming soon to all providers |
13:57.55 | robin_sz | nah, ip phoens are now corportae, but by 12 months will be domestic |
13:58.07 | frk2 | oh definitely |
13:58.13 | Zeeek | many people have them in the us along with voip routers |
13:58.14 | frk2 | i would wanna give a low cost phone away :) |
13:58.18 | robin_sz | every broadband user will have some |
13:58.36 | frk2 | yes |
13:58.38 | frk2 | death of the PSTN |
13:58.45 | frk2 | sad demise of the POTS network |
13:58.47 | Zeeek | the best use of those atcoms is actually to take it in your suitcase and plug it in at the hotel |
13:58.49 | frk2 | i would be so freakin happy |
13:58.54 | Alystair | hmmmm |
13:59.08 | Alystair | So what's a good brand to go with? Polycom? |
13:59.13 | Zeeek | what about junction networks (for IAX providers) ? |
13:59.14 | frk2 | how long do you think it'll be before the PSTN dies off completely |
13:59.24 | Cybertoy | zeeek, yeah ... except in most hotels you have to go through some web-page before you actually have access to the internet. |
13:59.28 | frk2 | Polycom is the safest bet |
13:59.30 | Cybertoy | how do you get around that? |
13:59.32 | Zeeek | Polycom is good but getting up in the price scale |
13:59.35 | frk2 | aastra is damn good too |
13:59.40 | Zeeek | Cybertoy |
13:59.53 | frk2 | yeah |
13:59.54 | Zeeek | Cybertoy I didn't have that problem in Madrid |
14:00.01 | frk2 | and some hotel idiots dont even NAT |
14:00.14 | Zeeek | NAT has nothing to do with IAX |
14:00.14 | frk2 | or give good bandwidth |
14:00.15 | robin_sz | gxp2000 has been a big disappointment for me .. the firmware upgrade is a disaster |
14:00.23 | frk2 | NAT needs to work man |
14:00.32 | Cybertoy | zeeek, I had that experience in many hotels in Zurich, London, Paris, Tokyo, Singapore, Orlando ... |
14:00.37 | frk2 | im sure the hotel aint providing you a public IP |
14:00.38 | Zeeek | there were 200 people talking into phones in tat Madrid hotel on the same connection! |
14:00.40 | Cybertoy | just to name a few where I stayed in the last 6 months. |
14:00.56 | *** join/#asterisk Lino` (n=Lino@i577BC51D.versanet.de) |
14:01.00 | frk2 | robin |
14:01.01 | Zeeek | your don't need a public ip |
14:01.02 | frk2 | it is |
14:01.04 | Cybertoy | in USA many hotels now stopped charging... |
14:01.11 | Cybertoy | but you still have to go through a web-page ... |
14:01.13 | frk2 | my GXP's crash all the time if the call load increases |
14:01.23 | frk2 | yes zeek, then you need NAT |
14:01.26 | Cybertoy | in Europe many want to charge... through ipass or credit-card. |
14:01.34 | frk2 | some hotels ive been to WONT nat.. they'll want you to use a proxy |
14:01.49 | Zeeek | frk2 I have used this phone with no problem with NAT ever |
14:02.22 | frk2 | zeeek yes- no issues.. but you NEED NAT |
14:02.59 | Zeeek | I see what you're saying. I've been lucky I guess |
14:03.13 | Cybertoy | also many WiFi hotspots are like that... |
14:03.26 | Zeeek | wifi is a whole n'other subject |
14:03.26 | Cybertoy | so a small phone with some small built-in browser would be nice. |
14:03.32 | frk2 | the worst thing is they wanted to charge $25/hr in Zurich airport :) |
14:04.07 | Cybertoy | frk2, yes... but if you get close enough to the business class lounge you can pick up their signal ... and there it's free.. :D |
14:04.19 | frk2 | dude i was in the business class lounge!!! |
14:04.28 | Cybertoy | hmm.. then they changed it. |
14:04.37 | Cybertoy | year ago that was the case. |
14:05.15 | frk2 | but who ever pays that much? |
14:05.39 | Cybertoy | I wouldn't... but it seems like people do. |
14:05.48 | Alystair | my wifi connected printer prints out money! |
14:06.00 | Cybertoy | someone mentioned the junxion box? |
14:06.09 | Zeeek | no junction networks |
14:06.47 | Cybertoy | ah ... I am looking at getting the junxion box for my car... |
14:06.50 | Cybertoy | and then a wifi phone. |
14:06.55 | tzafrir | Cybertoy, how much would a minimal PC cost? |
14:07.17 | Cybertoy | trafrir, for what? |
14:07.23 | Cybertoy | tzafrir that is |
14:07.24 | Zeeek | I thought a junction box was a thing with screws that put two fat wires together |
14:07.44 | Cybertoy | no ... www.junxionbox.com ... it's for wireless high speed broadband internet. |
14:08.08 | Zeeek | so cellphone in the left hand and keyboard in the right? None on the wheel? |
14:08.16 | Cybertoy | lol |
14:08.35 | Zeeek | what'you, Britney Spears? |
14:08.43 | Cybertoy | latin blood... :) |
14:08.54 | Zeeek | Cybertoy kinda says it all |
14:09.52 | frk2 | hahahahah |
14:10.41 | poisoner | hmmm |
14:10.43 | frk2 | dude... when would GPRS be low-latency enough for VOIP? |
14:10.53 | Zeeek | anyone in the USA at the moment? |
14:11.01 | Cybertoy | EV-DO or EDGE ... no gprs... |
14:11.03 | poisoner | frk2: try UMTS with HSDPA |
14:11.09 | Cybertoy | Zeeek, I am. |
14:11.12 | Zeeek | funny, I was just about to ask a GPRS question |
14:11.26 | Zeeek | Do any cell companies do GPRS now? |
14:11.36 | Cybertoy | yeah .... |
14:11.36 | frk2 | zeek we share the same chain of thoughts.. atcom, gprs :) lol |
14:11.38 | Zeeek | I just bought a quad-band phone |
14:11.39 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
14:11.43 | frk2 | in pakiland they do.. all of then |
14:11.51 | Zeeek | in the USA? |
14:11.54 | Cybertoy | yes. |
14:12.08 | Zeeek | because I've seen some web pages that say few providers do GPRS |
14:12.22 | Cybertoy | Cingular has it ... that's the one I'm on ... |
14:12.24 | Zeeek | I need to buy a SIM when we're over there in the summer |
14:12.30 | frk2 | im sure last time i was in the US cingual/tmobile had it |
14:12.31 | Zeeek | How are they? |
14:12.32 | Cybertoy | I would guess that Sprint does it as well. |
14:12.36 | Cybertoy | and T-Mobile |
14:12.44 | Cybertoy | I'm quite happy with Cingular. |
14:12.45 | Zeeek | Sprint? The used the okld crap system |
14:12.51 | frk2 | sprint uses something else |
14:12.55 | frk2 | not gprs |
14:12.57 | Zeeek | I wonder if Cingular sells prepaid SIM? |
14:13.02 | Cybertoy | they do |
14:13.03 | frk2 | its cdma 1xRTT i think |
14:13.09 | Zeeek | PCS is the old system right? |
14:13.14 | frk2 | since they aint gsm phones |
14:13.28 | Zeeek | I'll have a look at Cingular |
14:13.37 | frk2 | man.. why cant be just have a BIG HUGE 802.11b network |
14:13.41 | frk2 | that would be quite awesome |
14:13.43 | Cybertoy | zeeek, did you look at www.united-mobile.com if you travel a lot? |
14:13.51 | Zeeek | no but I will, thx |
14:13.58 | Cybertoy | no roaming cost to many countries. |
14:14.05 | Zeeek | I have seen a few sites that have a bunch of info |
14:14.24 | Zeeek | it was really co,nfusing at first with all those different bands and standards |
14:14.35 | Zeeek | way worse thazn voIP :) |
14:15.00 | Cybertoy | junxion box and wifi voip phone... |
14:15.03 | Cybertoy | way to go... :) |
14:15.13 | Zeeek | google should habe a swith "NOT ebay" |
14:15.25 | frk2 | junxion |
14:15.29 | frk2 | let me check that shit |
14:17.22 | frk2 | yo cybertoy |
14:17.34 | frk2 | what speeds can you get? |
14:17.41 | frk2 | what latency to the first router? |
14:17.49 | Cybertoy | no idea... I'm trying to find out more about it as well ... |
14:18.01 | frk2 | damn |
14:18.12 | Cybertoy | but sure sounds sexy... |
14:18.17 | frk2 | even if it gives 64/128k with less than 100ms its pretty usable for voi |
14:18.18 | frk2 | voip |
14:18.28 | Zeeek | united-mobile calls are very expensive |
14:18.30 | frk2 | however |
14:18.44 | blitzrage | morngin all |
14:18.47 | Zeeek | US to US nearly $2/minute? |
14:18.48 | blitzrage | morning even |
14:18.48 | frk2 | coolest thing is nokia is coming out with phones, ordinary phones, that would have WiFi and a SIP phone built in |
14:18.51 | Zeeek | hi blitz |
14:18.54 | Cybertoy | zeeek, frigg that... |
14:18.58 | blitzrage | Zeeek: hey ho! |
14:19.03 | Zeeek | I don't get it |
14:19.23 | blitzrage | frk2: yah --- any idea if it comes with a mini-browser too? |
14:19.25 | Zeeek | what's the point? I need to just buy a US SIM |
14:19.26 | Cybertoy | zeeek, I think that only makes sense if you receive many calls... but not make them. |
14:19.37 | Cybertoy | zeeek, and USA is not in the free roaming list. |
14:19.38 | blitzrage | wifi phones with no mini-browser are useless imho |
14:19.43 | Zeeek | in Europe recd calls are free anyway |
14:19.51 | Zeeek | Cybertoy ok I see now |
14:19.53 | Cybertoy | zeeek, that's right... but not in the usa. |
14:20.03 | frk2 | blitz.. dont matter.. if I can just use a cell phone to register to asterisk and make gsm calls too thats kick ass |
14:20.10 | frk2 | eliminates the need for separate IP phones |
14:20.18 | Zeeek | so, folks, who sells USA SIM cards as prepaid with no locked phone? |
14:20.32 | Cybertoy | ebay |
14:20.37 | Zeeek | Cingular doesn't apper to by the way |
14:20.38 | blitzrage | frk2: yah, but if you'r ein an airport or public wifi and want to save on the cell costs, a mini-browser is necessary to login |
14:20.53 | frk2 | oh true |
14:21.01 | Cybertoy | zeeek, they do .. I have friends that use cingular... but they don't sell an unlocked phone with it. |
14:21.03 | frk2 | but dude... im just looking at benefits in the office |
14:21.09 | frk2 | even those are amazing |
14:21.15 | Zeeek | I want the phone to do GPRS because it has a built in email client. That was the whole point of using it |
14:21.19 | frk2 | employees have their own cell phone + sip phone |
14:21.45 | Zeeek | Cybertoy ? you mean you buy a phone and put the SIM in your international phoen? |
14:21.47 | *** join/#asterisk Dimitripietro (i=Wut@modemcable017.237-202-24.mc.videotron.ca) |
14:22.17 | Cybertoy | Zeeek, you can't just buy a sim card?? |
14:22.34 | Zeeek | that's what I'm asking about. I don't see the possibility on their site |
14:23.08 | Dimitripietro | I'm using a TDM400P and i'M getting some background noise when using it. Anyone could gice me some trick too look at ? |
14:23.24 | Zeeek | take a good look at cables and phones first |
14:23.46 | Dimitripietro | Polycom IP phone |
14:24.05 | Dimitripietro | If i'm doing a SIP TO SIP, no problem |
14:24.32 | Zeeek | what kind of noise? |
14:24.38 | Zeeek | crackling? |
14:24.55 | Zeeek | could be IRQ conflicts, bad cable or wall connector |
14:24.56 | Dimitripietro | Not cracking, constant noise |
14:25.02 | Zeeek | hummmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm? |
14:25.19 | Zeeek | applause? |
14:25.28 | [TK]D-Fender | Dimitripietro : Could be an impedence issue or grounding. |
14:25.38 | Zeeek | yes, or card placement |
14:25.54 | Zeeek | sometimes video cards interfere with others on the bus |
14:25.57 | Dimitripietro | Is there a way in Linux to manually mange the IRQ ? |
14:26.05 | [TK]D-Fender | Yeah, general outside interference is a possibilty |
14:26.16 | Zeeek | it's done in the BIOS |
14:26.18 | [TK]D-Fender | Dimitripietro : Not really. Check your BIOS |
14:26.37 | Dimitripietro | There is no options in the Bios ... |
14:26.37 | [TK]D-Fender | Dimitripietro : Do a "cat /proc/interrupts" and pastebin it. |
14:26.38 | [TK]D-Fender | ~pb |
14:26.40 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
14:26.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
14:26.43 | coppice | constant noise sounds like my kids |
14:26.50 | coppice | or my mother in law |
14:27.22 | Cybertoy | zeeek, I can't see it on the web-page either.. but just go to one of their stores... |
14:27.30 | Zeeek | I FOUND IT YAY! |
14:27.43 | Zeeek | only in California, but that's where I'll be! YES!!! |
14:27.52 | Zeeek | http://www.cingularwireless.com/download/Prepaid%20Brochure%20(2%20MB).pdf |
14:28.01 | Cybertoy | what a scam |
14:28.11 | Dimitripietro | <PROTECTED> |
14:28.20 | Zeeek | shit, no that's prepaid wireless |
14:28.25 | Zeeek | damn |
14:28.32 | Dimitripietro | The tdm is not sharring IRQ |
14:28.47 | Zeeek | try another slot if you can |
14:28.55 | Dimitripietro | Already tried |
14:28.59 | Dimitripietro | same noise |
14:29.03 | [TK]D-Fender | Dimitripietro ; hmm, made sure your loadzone and so on are rigtht ro your area? |
14:29.05 | Zeeek | I'm out of suggestions |
14:29.19 | Dimitripietro | loadzone zapata ? |
14:29.32 | Cybertoy | nedd to get breakfast... see ya all. |
14:29.46 | Zeeek | prepaid wireless is a cellphone? What is this marketing giberish? |
14:29.49 | Zeeek | bye |
14:29.52 | [TK]D-Fender | Dimitripietro : zaptel |
14:30.07 | Cybertoy | zeeek, ah .. yeah .. they call mobile phone telephony here wireless... don't confuse it with wifi |
14:30.23 | Zeeek | well the news is good then despite the terminology |
14:30.27 | Zeeek | bon appétit |
14:30.30 | Dimitripietro | loadzone=us and i'm in canada |
14:30.32 | Cybertoy | merci |
14:30.34 | Dimitripietro | should be fine |
14:30.52 | Zeeek | loadzone=timbuktu |
14:31.10 | Zeeek | works fine but calls are very expensive! |
14:31.33 | [TK]D-Fender | Dimitripietro : You on DSL? Got it filtered before entering the card? |
14:31.48 | Zeeek | good call; that'd do it |
14:32.13 | [TK]D-Fender | No, he's on cable... |
14:32.23 | [TK]D-Fender | hmm |
14:32.30 | Zeeek | heh well, strike five for the debug team :) |
14:32.38 | [TK]D-Fender | I'm not done yet... |
14:32.52 | Dimitripietro | <[TK]D-Fender> I can send you a wav file |
14:33.02 | [TK]D-Fender | Dimitripietro : Try plugging in-line with a powerbar telephone surge suppressor |
14:33.14 | [TK]D-Fender | Dimitripietro : Its jsut a loud hum right? |
14:33.25 | Dimitripietro | yep |
14:33.52 | [TK]D-Fender | Dimitripietro : Yuo can also try changing the module's position on the card. Have seen a few that flaked out because of where it was. |
14:34.12 | [TK]D-Fender | Mine was DOA till i swapped the FXS & FXO modules for some unknown reason. |
14:34.46 | Dimitripietro | Ok, I,M goona try the power telephone filter (have one) first |
14:35.24 | Dimitripietro | <[TK]D-Fender> May I send you a recording of the problem ? |
14:36.02 | [TK]D-Fender | Dimitripietro : if you wish |
14:37.00 | Dimitripietro | Have you received the send request ? |
14:39.45 | [TK]D-Fender | well.. that failed :) |
14:39.47 | [TK]D-Fender | PM |
14:40.26 | rpm | kudos to whoever wrote that telemarketter script, now i just need to find a way to get a list of telemarketters numbers. |
14:41.13 | Dimitripietro | sent |
14:41.38 | Alystair | yay RDC to VMWare to CentOS |
14:46.00 | [TK]D-Fender | Dimitripietro : Thats just your breathing normally during a call with both sides mutually quiet? |
14:47.46 | Dimitripietro | yep |
14:48.07 | [TK]D-Fender | Dimitripietro : Whats your rxgain at currently? |
14:48.12 | [TK]D-Fender | in zapata |
14:48.24 | Dimitripietro | 8 |
14:48.27 | Dimitripietro | tx: 0 |
14:48.41 | [TK]D-Fender | lower to 0.0 across the board |
14:49.11 | Dimitripietro | When the gain is at 0.0, the sound is to low |
14:49.14 | Dimitripietro | too |
14:50.21 | saftsack | are any chan_capi users here? |
14:50.52 | [TK]D-Fender | Dimitripietro : try to scale it back a bit then and raise the default volumes on your phones |
14:51.46 | *** join/#asterisk esculapio_ (n=ESCulapi@66.98.18.236) |
14:52.11 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
14:52.34 | Dimitripietro | K, give me 5 minutes, I just changed the order of the FX0 modules |
14:59.20 | *** join/#asterisk \home\Carloz (n=gnunix@unaffiliated/packet) |
14:59.31 | *** part/#asterisk \home\Carloz (n=gnunix@unaffiliated/packet) |
14:59.48 | ManxPower | [TK]D-Fender, Be careful or you will become known as "the nice version of ManxPowr" |
15:00.12 | *** join/#asterisk isaiah (n=test@208-187-93-4.br1.hnv.mi.frontiernet.net) |
15:00.51 | [TK]D-Fender | ManxPower : Yin to your yang :) |
15:01.30 | Dimitripietro | [TK]D-Fender : rx:0 and swaped the order of modules. The sound is still there but lower. I will now try to raise the sound in the polycom configuration files |
15:01.39 | *** join/#asterisk ramo (n=ramo@59.92.141.2) |
15:02.12 | [TK]D-Fender | Dimitripietro : Better than doing it per-call. It amps better on the phone than doing it on the card. |
15:02.44 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.150.Dial1.SanJose1.Level3.net) |
15:04.46 | tekati | exten => _*7.,1,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${ARG1:2},60) |
15:05.07 | frk2 | guys |
15:05.31 | frk2 | why do i have to adjust my txgain/rxgain to ridiculously negative numbers for proper echo cancellation on PRIs? |
15:05.32 | tekati | Why when I dial *7 things appear to work right but when I dial *7*?????? as soon as I hit that second * it gives me a busy in the phone? |
15:05.54 | frk2 | tekati some phones treat * as 'send' or 'dial' |
15:06.14 | frk2 | sorry thats hash |
15:06.22 | frk2 | ignore what i just said |
15:06.23 | frk2 | :) |
15:06.54 | frk2 | i mean is a txgain of -15 and rxgain of -14 normal? |
15:07.05 | frk2 | or is my telco doing funny stuff |
15:07.10 | Zeeek | tekati what phone? |
15:07.39 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.150.Dial1.SanJose1.Level3.net) |
15:07.46 | tekati | I use a cordless phone with a LinkSys PAP2 |
15:08.01 | Zeeek | the phone it self may use service codes |
15:08.28 | tekati | Interesting. Let me look at the PAP2 config. See if there is anything in there. |
15:08.37 | Zeeek | try looking at the web page of the linksys (I assume it show a page?) |
15:08.51 | Zeeek | many SIP phones have a table of service codes |
15:09.55 | [TK]D-Fender | tekati : You are using ARG1 where you should be using EXTEN. Looks like you ripped that off some guy's macro... |
15:10.24 | tekati | Sorry I changed it using my own macro to make sense here. I do use Exten and Arg in the right places my bad. |
15:10.47 | tekati | I think I found the issue in the PAP2 config. It has a Dial Plan string: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
15:10.50 | [TK]D-Fender | tekati : Pastebin the whole thing and everything related to it (except the constant definitions |
15:10.53 | [TK]D-Fender | ~pb |
15:10.56 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
15:11.18 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
15:11.47 | [TK]D-Fender | tekati : Try the much simpler : (*.T,#.T,x.T) |
15:12.10 | CunningPike | tekati: We have a PRI with a gain of -10dB - if the numbers you are using work for you, use them :) |
15:12.39 | tekati | CunningPike: that was meant for frk2 I think. |
15:12.45 | CunningPike | Oops - sorry |
15:12.47 | [TK]D-Fender | Gain should be 0.0 across the board. Anything else is typically a sad last attempt to alleviate echo... |
15:13.12 | tekati | Fender: Is that for the dial plan for the PAP2? |
15:13.22 | [TK]D-Fender | tekati : yup |
15:13.38 | [TK]D-Fender | basically "take anything I feel like dialing and jsut do it" |
15:14.06 | frk2 | Fender thats what i thought |
15:14.14 | [TK]D-Fender | Real dialplans on en-points can be an adminstrative PITA. |
15:14.25 | frk2 | but unless txgain/rxgain are BOTH less than -14, echo happens |
15:14.26 | Dimitripietro | [TK]D-Fender : with gain 0.0 there is a huge differnce in the volume of a sip to sip call and a sip to pstn call throught TDM |
15:14.34 | [TK]D-Fender | Just let the phone do "whatever" and control it on the * level |
15:15.02 | [TK]D-Fender | Dimitripietro : hmmm yeah... sip - sip would need to be adjusted. work something out in between. |
15:15.32 | tekati | I agree. I am still having the same issue after switching both Line1 and Line2 to the (*.T,#.T,x.T) for the Dial Plan. Must be another setting somewhere. |
15:15.37 | [TK]D-Fender | frk2 : Try all the other settings before ever touching gains.... |
15:15.51 | frk2 | Fender there arent much settings |
15:16.04 | frk2 | echo is being created on the PRI box |
15:16.16 | frk2 | i need to cancel it there |
15:16.24 | frk2 | MG2 helps, but not without gain adjustment |
15:16.24 | [TK]D-Fender | frk2 : echocancel=[taps], echotraining, recompile Zaptel with several other options... |
15:16.44 | [TK]D-Fender | frk2 : Maybe its time you get a better PRI card. |
15:16.46 | frk2 | yes tried that.. what other options? I've tried MG2 |
15:16.59 | frk2 | I got a digium TE 110P |
15:17.01 | [TK]D-Fender | frk2 : MG2 is DEFAULT isn't it? |
15:17.07 | frk2 | no KB1 is default |
15:17.10 | frk2 | MG2 is the new one |
15:17.17 | frk2 | supposed to be more effective on PRI cards |
15:17.19 | ghenry | Hi, is it a Tele provider or Asterisk dialplan that allows someone to dial 01358 279644, when the internal extension is 44 and the main line is say 01358 279600 |
15:17.36 | frk2 | and it is |
15:17.41 | frk2 | but not without gain adjustment |
15:17.52 | ghenry | So each internal extension can be reached from outside the company |
15:17.57 | [TK]D-Fender | frk2 : Oh the joys of software echo cancellation.... |
15:18.13 | frk2 | heheh |
15:18.20 | frk2 | maybe my telco is amplifying |
15:18.29 | frk2 | cuz at gain = 0, the voice is REALLY loud |
15:18.35 | frk2 | to the point that MOH hurts |
15:18.44 | [TK]D-Fender | frk2 : Could be..... |
15:19.08 | frk2 | I say damn the PSTN to hell |
15:19.11 | CunningPike | frk2: We dicked around with software settings for months before finally purchasing a Ditech echo can |
15:19.15 | CunningPike | No more ecbho |
15:19.25 | frk2 | ditech echo can? |
15:19.26 | CunningPike | Except when I type :S |
15:19.41 | CunningPike | Ditech make carrier-grade echo cancellation boxes |
15:19.45 | ManxPower | CunningPike, We got Tellabs EC off eBay. Works great. |
15:19.52 | [TK]D-Fender | I did the same for about a month before switching things at my company as well. |
15:20.12 | CunningPike | ManxPower: We have 4 PRIs, so the Ditech is actually cost-effective at that scale |
15:20.14 | [TK]D-Fender | ManxPower : A bit of a PITA to set up, but works decent right? |
15:20.44 | frk2 | Pike the new quad digium hardware echo can cards are really good |
15:21.28 | [TK]D-Fender | frk2 : I've heard plenty of mixed reviews at least about their 1st gen. Not as much "in the know" about the newer VPM though. |
15:21.39 | CunningPike | Not in our experience, I'm afraid - we found them to be worse than MG2, and they introduced audio drop outs |
15:21.43 | ManxPower | [TK]D-Fender, Tellabs? Yes. About $250 - $300, for the shelf and 20 or so EC cards and a -48V power supply. The docs SUCK, but it works very well. |
15:22.13 | ghenry | Anyone? ;-) |
15:22.17 | frk2 | thats weird man |
15:22.24 | [TK]D-Fender | ManxPower : Yeah I think I read a few things on the WIKI about that... looked scary. |
15:22.24 | frk2 | we got the newer tdm400 and te411p |
15:22.28 | frk2 | they are awesome |
15:22.38 | frk2 | tdm2400 |
15:22.40 | frk2 | sorry |
15:22.48 | frk2 | the echo cancellation was pretty decent |
15:22.49 | ManxPower | [TK]D-Fender, I have the official tellabs docs in PDF format..... |
15:23.43 | frk2 | so basically there are others running very negative gains |
15:23.47 | Dimitripietro | frk2 : Are you using hardphone with your tdm2400 ? |
15:23.59 | frk2 | yup |
15:24.03 | CunningPike | ghenry: It is your provider - you need to purchase DIDs from them |
15:24.07 | frk2 | basically the tdm2400p is a failover for the PRI |
15:24.35 | *** join/#asterisk miguel3239 (n=miguel32@ns1.nashuacs.com) |
15:24.37 | Dimitripietro | did you needed to turn software echo canceller on even with the hardware echo canceller ? |
15:24.49 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
15:24.58 | frk2 | hmmm |
15:25.09 | frk2 | it is actually on now that i think of it |
15:25.18 | frk2 | will try turning it off |
15:25.18 | CunningPike | Dimitripietro: The VPM disables the software EC |
15:25.25 | frk2 | however echo training refuses to work |
15:25.49 | CunningPike | frk2: That's because the software EC is disabled |
15:25.55 | frk2 | yeah probably |
15:26.01 | frk2 | but the echo cancellation was pretty good |
15:26.43 | Dimitripietro | With my tdm2400p and hardware echo canceller, if I turn in zapata echocancel=no then I'm getting echo in my polycom phone |
15:27.05 | frk2 | hmm |
15:27.14 | Dimitripietro | I needed to turn echocancel=yes to eliminate the echo |
15:27.16 | frk2 | ill double check |
15:27.41 | [TK]D-Fender | ManxPower : An eve those suck huh? |
15:27.55 | ManxPower | [TK]D-Fender, correct. |
15:28.01 | [TK]D-Fender | Dimitripietro : Thats the NORMAL thing to do regardless. |
15:28.03 | *** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:28.31 | Dimitripietro | Ok |
15:28.46 | *** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:28.59 | *** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:29.02 | Dimitripietro | Even with an hardware echo can, we need to turn the echocancel=yes |
15:29.08 | *** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:29.52 | *** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:29.57 | *** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
15:33.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
15:34.16 | ManxPower | Dimitripietro, if you need to do echocancel=yes with external EC then the external EC is not working correctly. |
15:35.04 | [TK]D-Fender | ManxPower : I believe for the hardware he's using "echochancel=yes" just actives the HWEC, not soft |
15:35.33 | frk2 | thats what i thought |
15:35.39 | frk2 | but theres no way to check i guess |
15:36.27 | ManxPower | [TK]D-Fender, so he is using the VPM addon board for the Digium card? If so, then echocancel=yes enables the EC on the daughter card. If he's using an actual external box, then my previous statement stands. |
15:36.57 | ManxPower | I just don't trust Digium's EC. |
15:37.28 | [TK]D-Fender | ManxPower : Careful... the "Man" is watching! ;) |
15:37.37 | frk2 | hahahah |
15:39.51 | *** part/#asterisk bmonty (n=bmontgom@ubuntu/member/bmonty) |
15:46.47 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
15:50.04 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
15:50.38 | poisoner | has anyone linked succesfully asterisk mit an older Cisco CM? |
15:50.48 | DoktorGreg | arrghhh |
15:50.54 | DoktorGreg | sip phone... |
15:50.58 | DoktorGreg | no matter what i do |
15:51.03 | DoktorGreg | if i let it sit for while |
15:51.06 | DoktorGreg | or reboot it |
15:51.13 | DoktorGreg | first call |
15:51.17 | DoktorGreg | cant hear anything |
15:51.25 | DoktorGreg | but can talk |
15:51.28 | DoktorGreg | after that |
15:51.30 | DoktorGreg | works fine |
15:51.32 | Alystair | DoktorGreg: I'm doing the unthinkable and trying a VM setup ;) |
15:51.33 | Dimitripietro | [TK]D-Fender : If I use the zapbarge fuction on one of my fxo, i'm hearing the same noise even if my line isn't plugged in the fxo |
15:51.46 | DoktorGreg | cant hurt to try |
15:51.49 | Alystair | yep |
15:51.59 | DoktorGreg | I thought you found an old smp p3 system? |
15:52.00 | Dimitripietro | So it's sure the noise is coming from my box |
15:52.10 | Alystair | I did but this monster machine is going on unused :( |
15:52.42 | DoktorGreg | its not going unused |
15:52.47 | DoktorGreg | it is your domain server |
15:53.11 | Alystair | so, nothing important :) |
15:53.28 | tekati | Wow LinkSys guards the Admin Guide to PAP2 like it is their pot of GOLD!!! |
15:53.32 | Alystair | Oh, also our profile server and central backup |
15:53.34 | DoktorGreg | arnt you using roaming desktops? |
15:53.40 | tekati | Does anyone have the Admin Guide for the PAP2-NA? |
15:53.52 | Alystair | DoktorGreg: yeah this machine is basically the heart of the company's office hah |
15:54.04 | [TK]D-Fender | Dimitripietro You might jstu have a defective FXO module which does happen... call up Digium, explain everything you've tried and they may take you througha few more tests, but ultimately I believe thy my change your card. |
15:54.24 | DoktorGreg | well again, if all your are useing is its samba functions, nuke the os and put samba on it |
15:54.48 | DoktorGreg | samba works better as windows fileserver than windows does anyhow |
15:55.17 | Dimitripietro | On your side, the sound using a analog card is clear ? |
15:55.37 | Alystair | DoktorGreg: including security settings? |
15:55.42 | Alystair | and rights management? |
15:55.42 | DoktorGreg | yah |
15:55.44 | Alystair | hrm |
15:55.48 | DoktorGreg | read about samba |
15:55.55 | DoktorGreg | also |
15:56.00 | [TK]D-Fender | <PROTECTED> |
15:56.01 | DoktorGreg | no CAL's in future |
15:56.13 | DoktorGreg | you can also set samba up as domain servers |
15:56.14 | Dimitripietro | Ok |
15:56.26 | ghenry | CunningPike: Thanks. Will look into it ;-) |
15:56.46 | CunningPike | ghenry: np |
15:56.59 | ghenry | CunningPike: What does it stand for? |
15:57.20 | CunningPike | DID? Direct Inward Dial |
15:57.39 | ghenry | ah, yes. Thanks |
15:58.24 | ghenry | Isn't it DDI for Europe/UK? |
15:59.01 | CunningPike | It may well be |
15:59.07 | ghenry | thought so ;-) |
15:59.14 | ghenry | Maybe be a bad idea anyway for security |
15:59.14 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
15:59.24 | CunningPike | Bad for security? How? |
15:59.29 | ghenry | better going through IVR or Receptionist |
15:59.30 | tekati | Anyone have a Admin Guide for the PAP2? |
15:59.32 | ghenry | bbl |
16:03.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
16:03.21 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
16:04.08 | brif8 | Reviewing log files I see this SIP OPTIONS (404 not found) asterisk@my.voip.provider to s@my.ip.address:5060, this is not the register as I see that also ? |
16:04.27 | ManxPower | DID and DDI are just different terms for the same thing |
16:07.22 | brif8 | any ideas |
16:09.01 | ManxPower | brif8, that means that for some reason your provider is trying to send an options packet to extension "s" on your server. |
16:09.21 | brif8 | ManxPower: what is an options packet ? |
16:09.43 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
16:10.08 | ManxPower | brif8, it can be used for a zillion things. Asterisk uses them when you use qualify in sip.conf |
16:10.48 | brif8 | ok I have qualift = yes to my voip provider, would it occur roughly every minute |
16:10.59 | brif8 | qualift => qualify |
16:11.13 | ManxPower | brif8, I have no idea |
16:11.21 | brif8 | ok |
16:12.35 | brif8 | thanks anyways |
16:16.20 | *** join/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
16:17.18 | Seyr | I have an * server with 25 7960 phones using SIP and they want one to have a 7914 module. From what I read, the 7914 requires SCCP? |
16:17.37 | Qwell | Seyr: yes |
16:17.48 | Seyr | Any problems having 24 7960 use SIP and one use SCCP for the 7914? |
16:17.52 | Qwell | no |
16:18.07 | Seyr | any "gotchas" i need to look out for? |
16:18.54 | Seyr | should I seperate the TFTP for the SCCP from the SIP? or will the configs work side by side? |
16:19.05 | Seyr | the xml and loads and stuff? |
16:24.17 | tekati | In the PAP2 dial plan Fender sent me (*.T,#.T,x.T) What does the T do does anyone know? |
16:25.18 | ManxPower | tekati, Timeout |
16:25.20 | Dimitripietro | Timeout |
16:26.28 | viperdudeuk | Seyr: we have had SIP and SCCP configs on the same TFTP before now without a issue |
16:26.47 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
16:29.05 | DoktorGreg | http://video.google.com/videoplay?docid=109587088318033773&q |
16:29.56 | Seyr | thanks |
16:31.13 | DoktorGreg | yargghhh |
16:31.25 | DoktorGreg | ok, ATA, sipura 1001 |
16:31.31 | DoktorGreg | let it sit for while |
16:31.37 | DoktorGreg | pick up phone, dial |
16:31.42 | DoktorGreg | cant hear anything |
16:31.46 | DoktorGreg | but can talk ok |
16:31.50 | DoktorGreg | hang up phone |
16:31.55 | DoktorGreg | dial again |
16:31.58 | DoktorGreg | all works good |
16:32.04 | DoktorGreg | wait 10 minutes or so |
16:32.06 | DoktorGreg | try it again |
16:32.11 | DoktorGreg | same thing |
16:32.11 | *** join/#asterisk ToTo (n=ToTo@host235-158.pool875.interbusiness.it) |
16:32.20 | DoktorGreg | any idea where to look? |
16:32.56 | ManxPower | DoktorGreg, sounds like you have NAT involved and are not using qualify=yes |
16:33.17 | DoktorGreg | in sip.conf? |
16:33.23 | ManxPower | of course. |
16:33.41 | ManxPower | usually if you have nat=yes you want qualify=yes too |
16:33.51 | DoktorGreg | i have qualify=200 |
16:34.00 | ManxPower | that will also work |
16:34.32 | DoktorGreg | so its something else then... |
16:34.32 | ManxPower | you're not doing any port forwarding on the NAT router on the remote side, right? |
16:34.41 | DoktorGreg | lemme double check |
16:34.45 | ManxPower | I assume Asterisk is on a public IP and the SIP client is behind NAT |
16:34.50 | DoktorGreg | yup |
16:35.00 | DoktorGreg | well i take that back |
16:35.08 | DoktorGreg | asterisk is on a one to one nat |
16:35.27 | DoktorGreg | asterisk has its own internal ip address |
16:35.32 | DoktorGreg | er |
16:35.33 | ManxPower | Ah. I don't really feel like giving you a tutorial on running Asterisk behind nat. Check the Wiki |
16:35.52 | Zeeek | vas ist das problem? |
16:35.54 | DoktorGreg | every other phone works no prob though... also behind nat |
16:38.05 | *** join/#asterisk gursikh (n=guriskh1@adsl-69-151-246-132.dsl.hstntx.swbell.net) |
16:41.42 | *** part/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
16:42.38 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
16:57.54 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
16:58.03 | Assid | heya |
16:58.09 | Assid | hows everybuddy doing? |
16:58.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
16:58.40 | Zeeek | we are micromanaging the current adversity, thank you |
16:58.40 | Qwell | Assid: poorly, since gaim got popular |
16:59.31 | Assid | hehe |
16:59.44 | Assid | i was waiting for someone to notice that |
16:59.44 | *** join/#asterisk blackgecko (n=blackgec@201.152.14.187) |
16:59.47 | Qwell | in fact, it's dead |
17:00.34 | blackgecko | hi, anyone with a tdm2400p ?? |
17:01.38 | Assid | so what you been upto ? |
17:01.41 | *** join/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
17:02.21 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
17:04.47 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
17:04.54 | Zeeek | what is the parameter that holds the timeout time for a sip register? |
17:05.54 | Dimitripietro | <blackgecko> me |
17:06.15 | rdgzt | I have a weird SIP auth problem on register. With various softphones, I can't register, Asterisk says the password is wrong, although it's clearly correct. |
17:06.23 | rdgzt | Anyone have any ideas? |
17:07.07 | ManxPower | PASTE the error message. |
17:07.22 | rdgzt | Apr 30 02:06:04 NOTICE[14457]: chan_sip.c:10817 handle_request_register: Registration from '<sip:john@10.0.254.35>' failed for '10.0.2.2' - Wrong password |
17:07.28 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
17:07.40 | ManxPower | And you have a [john] section in sip.conf? |
17:07.45 | rdgzt | I'm following the examples from the Asterisk: The Future of Telephony book. |
17:07.45 | rdgzt | Yes. |
17:08.03 | ManxPower | put the [john] section on pastebin.ca |
17:08.41 | VoicePulse | Anyone know if Asterisk::LCR in widespread use? |
17:08.46 | file | do a lil' dance, get down tonight |
17:08.47 | blackgecko | dimitripetro; have you had any problem with it ?? |
17:09.33 | *** join/#asterisk rdgzt (n=joakim@201.137.86.15) |
17:09.34 | blackgecko | dimitripietro: have you had any problem with it ?? |
17:09.38 | rdgzt | Sorry, got disconnected. |
17:09.50 | rdgzt | I'll put the section on pastebin, hold on. |
17:10.39 | Dimitripietro | blackgecko : what kind of problem ? |
17:11.15 | blackgecko | striped numbers from the dialed ones, worng number dialed, etc |
17:11.37 | Dimitripietro | No |
17:12.06 | ManxPower | blackgecko, sounds like your telco does not start accepting DTMF fast enough. Try prepending w or ww to your dial string to add a .5 or 1 second wait before sending digits |
17:12.09 | blackgecko | dimitripietro: cause i get random errors from my telco "the number you have dialed is incorrect" |
17:12.14 | Dimitripietro | Is yor card sharing an irq with something else ? |
17:12.24 | ManxPower | classic problme, talked about at least once per month on the mailing lists. |
17:12.40 | rdgzt | ManxPower: http://pastebin.ca/52402 |
17:12.54 | blackgecko | dimitripietro: no irq sharing, ive already added www to the dialed number, without luck |
17:13.08 | ManxPower | blackgecko, paste the Dial line |
17:13.09 | rdgzt | ManxPower: It's my whole sip.conf, since it's very small, in case that's helpful. |
17:13.39 | *** join/#asterisk ramo (n=ramo@59.92.141.2) |
17:13.42 | blackgecko | the dial string is Zap/g0/wwwwww54234569 |
17:14.17 | blackgecko | the problem is the error isnt constant, it happens randomnly |
17:14.40 | blackgecko | dimitripietro: can you share your zapata.conf ?? |
17:14.56 | ManxPower | blackgecko, no, that is only part of your dial line. |
17:15.21 | ManxPower | I need the actual dial line from your extensions.conf. |
17:15.32 | ManxPower | rdgzt, looks good to me, other than you are not allowing any codecs. |
17:16.30 | Strom_C | no no no, the answer is "dead hookers" |
17:16.37 | blackgecko | manxpower using freepbx for the config, is it bad ? |
17:16.39 | rdgzt | ManxPower: I get the default set if I don't specify any, don't I? Besides, the error message pretty explicitly says that it's "Wrong Password" |
17:17.05 | ManxPower | blackgecko, I cannot help you with FreePBX. Did you not look at the /topic of this channel? |
17:17.10 | rdgzt | As I said, following the instructions from the Asterisk book, and I've tried this with two different softphones, both do the same thing. |
17:17.14 | rdgzt | So I'm a bit stumped here. |
17:17.30 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
17:17.42 | Zeeek | rdgzt are you cutting and pasting the password or username? |
17:17.53 | ManxPower | rdgzt, what happens if you comment out the secret= line and do a reload? |
17:18.01 | ManxPower | and not touch the config on the SIP device |
17:18.36 | rdgzt | Zeeek: The password is "welcome", and even though I'm not cutting and pasting it, I'm typing it very carefully, and have several times. |
17:18.40 | rdgzt | ManxPower: Let me see... |
17:18.40 | blackgecko | manxpower, yeah but mi problem isnt with freepbx, ive done it without a gui, and with amp, and is the same problem |
17:19.25 | ManxPower | blackgecko, Yeah, but FreePBX uses such complicated config files that nobody wants to spend the day or so working with you to understand your setup. |
17:19.32 | ManxPower | As I said, I cannot help you. |
17:19.59 | blackgecko | is this correct ?? exten => _ZXXXXXXX,1,Dial(Zap/g0/www{EXTEN}) |
17:20.53 | ManxPower | blackgecko, not even close. |
17:20.53 | Strom_C | it's ${EXTEN} silly |
17:21.13 | ManxPower | Strom_C, Obvious that he did not even paste the line. |
17:21.23 | rdgzt | ManxPower: If I remove the secret line, I don't get that error message, I get some sip debug output in asterisk, but not that exact error message, and then the softphone says "registration failed: timeout". |
17:21.33 | Strom_C | ManxPower: well yeah |
17:21.37 | ManxPower | also that exten line will only match an 8 digit number |
17:21.48 | Strom_C | there's never a line that simple in freepbx |
17:21.50 | ManxPower | rdgzt, you have me stumped. |
17:21.54 | Strom_C | it's like [code_vomit] |
17:21.55 | rdgzt | ManxPower: Will putting that SIP debug output in a pastebin help? |
17:22.17 | rdgzt | ManxPower: Yeah, I'm stumped myself, it's had me scratching my head since yesterday. |
17:22.21 | ManxPower | rdgzt, only if 1) I wanted to think hard on a sunday. Someone else may be able to help you. |
17:22.46 | Zeeek | what the heck, go for it, there's a big audience out there |
17:22.52 | blackgecko | manxpower: im willing to try it on a no gui asterisk but i think the problem isnt the gui |
17:23.13 | rdgzt | At this point I'm considering building and installing Asterisk from source, since I'm currently using the Ubuntu packages, in case there's something weird with the defaults or something in the Ubuntu packages. |
17:23.23 | rdgzt | But that's kind of grasping at straws. |
17:23.50 | ManxPower | blackgecko, The line you pasted says Match an 8-digit number, the first digit matcing 1-9, dial out Zap group 0, wait 1.5 seconds, then dial the DTMF digits { E X T E N } of which, of course there are no such DTMF digits. |
17:24.08 | ManxPower | blackgecko, if you continue to try to convince me to help you with this problem I will put you on /ignore. |
17:24.11 | Zeeek | ManxPower, I want to benefit from your long, hard earned wisdom (or anyone who can answer this) : |
17:24.30 | Strom_C | Zeeek: the answer is "dead hookers" |
17:24.34 | rdgzt | http://pastebin.ca/52408 is the SIP debug output, if anyone's interested. |
17:24.44 | blackgecko | manxpower: sorry to bother you |
17:24.53 | rdgzt | From when I removed the secret line in the config. |
17:25.28 | ManxPower | rdgzt, be sure to mention what SIP device you use, it might be important |
17:25.40 | Zeeek | I thought is was 42. I'll have change that part of sip.conf |
17:25.59 | rdgzt | ManxPower: I've tried with the Xten softphone and Ekiga, both get the same result. Currently testing with Ekiga, since the Xten interface is horrible. |
17:26.02 | stoffell_h | it seems avoiding initial deadlock is not a "good thing" ? I have (http://pastebin.ca/52409) a LOT of those errors, just before my * server stopped doing anything.. any idea on how to troubleshoot ? |
17:26.05 | Strom_C | 42 is The Answer. different than just the answer :) |
17:26.08 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
17:26.21 | Zeeek | and THe ansWer ? |
17:26.41 | Strom_C | that's "They spinnin' yo" |
17:27.12 | Zeeek | incidentally, X-Ten is great for testing SIP, but remember out there, it automatically uses STUN so it won't be the same on asterisk |
17:27.36 | Zeeek | unless you disable STUN in the X-Ten of course |
17:27.48 | *** join/#asterisk IceManRISK (n=kart@201.14.2.169) |
17:30.22 | rdgzt | The softphone and the server are on the same network with nothing inbetween, shouldn't that make STUN irrelevant? |
17:31.01 | Zeeek | indeed |
17:31.11 | *** part/#asterisk neilvince (n=Neil@nevis-systems.demon.co.uk) |
17:31.18 | Zeeek | the comment is a general avertissement |
17:31.57 | rdgzt | Zeeek: Right, thanks. I guess that's why the Asterisk book recommends it. Although I insist that the interface is horrible. |
17:32.48 | Zeeek | it works well when you get used to all those clicks |
17:33.22 | rdgzt | Software that tries to look like a physical object is generally a bad idea, but never mind, I'm sure that's not related to my problem. :) |
17:33.53 | Zeeek | aiming at the youth element |
17:33.56 | Zeeek | "skins" |
17:34.04 | rdgzt | I'm in a bind here, I have a bunch of hardphones that I'm supposed to set up with this thing, but that seems like a tall order if I can't even get a simple softphone working. :) |
17:34.11 | Zeeek | I have seen the password issue somewhere |
17:34.16 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
17:34.19 | PakiPenguin | evening |
17:34.21 | rdgzt | Zeeek: Really? |
17:34.26 | Zeeek | but where? |
17:34.30 | rdgzt | Zeeek: I tried to google, but I didn't have much luck. |
17:34.39 | Zeeek | you've obviously googled it to death |
17:34.39 | Zeeek | heh |
17:34.48 | Zeeek | and looked on the mailing lists |
17:35.46 | rdgzt | I haven't explicitly looked on the mailing lists, mostly because I assume that Google would search their archives, but... |
17:37.25 | Zeeek | I'd just look for password in the ml |
17:37.25 | Strom_C | rdgzt: what version of asterisk are you running? |
17:37.25 | rdgzt | Strom_C: My package is 1.2.1.dfsg-4ubuntu1 |
17:37.27 | Qwell | Strom_C: I uploaded the code into my chan_skinny branch |
17:37.28 | key2 | for what purpose people use asterisk with postgres ? |
17:37.30 | Strom_C | ok, time to upgrade. we're on 1.2.7.1 now, you know |
17:37.35 | Qwell | team/north/chan_skinny-fixup/ |
17:37.44 | Strom_C | Qwell: oooh |
17:37.54 | Strom_C | i will totally have to play with it when I get out of bed |
17:37.55 | rdgzt | Strom_C: Yeah, I was actually considering building asterisk from the latest source instead of using packages. |
17:38.00 | Qwell | ... |
17:38.05 | Strom_C | rdgzt: yes, that would be a good idea |
17:38.11 | rdgzt | Strom_C: In case there's a bug that's been fixed or something. Although I generally prefer using the package system. |
17:38.14 | Qwell | note to self: make sure to read things IN CONTEXT |
17:38.17 | rdgzt | I'm going to test that now, I think. |
17:38.41 | Strom_C | rdgzt: with asterisk, packages are almost always a bad idea |
17:38.53 | Qwell | s/almost // |
17:39.13 | rdgzt | Strom_C: I noticed a tendency to that, actually, why is it that distributions don't package the zaptel drivers, for instance? |
17:39.19 | *** join/#asterisk tomcontr3 (n=gcontrer@247-79-246-201.adsl.terra.cl) |
17:39.31 | Strom_C | because they blow donkeys for nickels? |
17:39.36 | tomcontr3 | hi dows ony one know about this? http://www.voip-forum.com/news.php?p=187 |
17:39.40 | rdgzt | The drivers or the distributions? |
17:40.03 | Strom_C | the distros, obviously |
17:40.23 | Strom_C | zaptel is good stuff |
17:40.56 | rdgzt | Building kernel drivers from source is generally a pretty bad idea, since they might break arbitrarily if you upgrade the kernel through the package system. Which is one of the reasons I generally prefer the package system. |
17:41.23 | Strom_C | thats why you rebuild zaptel after you upgrade the kernel. |
17:41.25 | Strom_C | duh |
17:42.32 | rdgzt | Strom_C: Yes, well, that's why packages would be good, since the distro would just install zaptel drivers built against the new kernel. |
17:42.37 | *** join/#asterisk budmang (i=budman@12.206.134.162) |
17:43.25 | rdgzt | Building stuff from source is so 1999. :) |
17:44.10 | Zeeek | only true for WIndows |
17:44.21 | Zeeek | assembler is best for that |
17:45.57 | *** join/#asterisk mutilator (i=WebChat@65.111.201.122) |
17:46.18 | mutilator | do the zaptel drivers control the hardware echo can? |
17:46.22 | Zeeek | why is there no distro called 'Kernel Klink' ? |
17:46.23 | mutilator | i have a sangoma a104d |
17:55.46 | tomcontr3 | Im having problems installing an SVN version of asterisk |
17:55.56 | tomcontr3 | gcc -g -o menuselect ../strcompat.o menuselect.o menuselect_curses.o ../mxml/libmxml.a -lncurses |
17:55.56 | tomcontr3 | gcc: ../strcompat.o: No such file or directory |
17:55.59 | tomcontr3 | any idea? |
17:56.34 | Strom_C | tomcontr3: you downloaded the stable version from SVN, right? |
17:58.50 | tomcontr3 | its a version that a guy name OEJ is building |
17:58.55 | tomcontr3 | to fix a SIP problem |
17:59.02 | tomcontr3 | http://svn.digium.com/view/asterisk/team/oej/sipregister/ |
17:59.17 | tomcontr3 | http://www.voip-forum.com/news.php?p=187 |
17:59.58 | Strom_C | you've got the ncurses stuff installed, right? |
18:00.15 | tomcontr3 | dont know what that is... lolol |
18:00.29 | *** join/#asterisk esculapio_ (n=ESCulapi@187stb68.codetel.net.do) |
18:00.31 | *** join/#asterisk tparcina (n=tparcina@83-131-143-11.adsl.net.t-com.hr) |
18:01.22 | Strom_C | tomcontr3: uh, yeah. stick to installing stable releases for now. |
18:01.46 | tomcontr3 | hmmm, but I need to fix that problem, |
18:01.56 | tomcontr3 | and that installation it seems to be the only way |
18:05.43 | *** join/#asterisk darby_t (i=darby_t@aaoy166.neoplus.adsl.tpnet.pl) |
18:09.41 | tomcontr3 | yes I have installed the ncurses-devel-5.4-17 |
18:16.37 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
18:19.36 | blackgecko | manxpower: now its working, it was tx rx gain problem, the txgain was to high. thanks |
18:23.27 | *** join/#asterisk bmg505 (n=leon@dsl-146-56-106.telkomadsl.co.za) |
18:24.25 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
18:25.18 | SwK_ | misfire |
18:26.57 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
18:29.00 | *** join/#asterisk frk2 (n=kvirc@202.141.251.102) |
18:29.09 | frk2 | dude |
18:29.23 | frk2 | have you guys ever sat and wondered how cool asterisk really is? :) |
18:29.47 | frk2 | i mean damn man. mark spencer has caused a semi-revolution at the age of 28 |
18:29.53 | frk2 | or 29 |
18:30.25 | MikeJ[Laptop] | isn't that a beatles song? |
18:30.37 | frk2 | revolution? |
18:30.39 | frk2 | hell yes |
18:31.15 | frk2 | i wonder if mark spencer had the digium idea in mind before he started asterisk |
18:31.50 | frk2 | cuz if he did not - that makes it even more noble |
18:32.17 | frk2 | i guess nobody cares :) |
18:35.10 | Nugget | noble? it's software. |
18:35.23 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
18:35.23 | Nugget | it's not like asterisk is curing cancer. |
18:35.38 | dpryo | Ever wondered about how neat Apache really is? |
18:35.40 | dpryo | ;) |
18:35.41 | file | in the end, it's only software |
18:35.51 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
18:35.57 | dpryo | or any other piece of open source software.. |
18:36.34 | dpryo | Firefox, postfix, .. gimp, gnome, kde.. etc |
18:36.46 | frk2 | yeah man |
18:36.48 | frk2 | it is neat |
18:37.01 | frk2 | specially shit like KDE |
18:37.09 | frk2 | i mean damn thats a lot of code |
18:37.20 | file | yessir |
18:37.44 | frk2 | But man asterisk is screwing a lot of larger companies over |
18:38.12 | dpryo | Yeah, and I still have to use a sucky Avaya-system. |
18:38.13 | Nugget | It's likely that none of that open source software would exist were it not for a robust and thriving commercial software industry. |
18:38.18 | *** part/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
18:38.23 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
18:38.30 | frk2 | this cisco biz dev was complaining about asterisk |
18:38.58 | frk2 | whatdya mean nugget? |
18:39.27 | file | I like being part of Asterisk... it's nice to see people use it |
18:39.35 | Nugget | well, for one, much of the open source software that exists is just re-implementing the innovations that come from the closed-source world. The open source community hasn't shown itself to be very innovative at all. |
18:39.57 | file | Nugget: slacker. |
18:40.01 | frk2 | oh come on i dont think so |
18:40.11 | frk2 | maybe stuff like KDE, etc |
18:40.18 | Nugget | and secondly, a staggering majority of the "donated" time, energy, and code is coming from people who are gainfully employed as programmers, for companies who release closed-source products. |
18:40.24 | stoffell_h | Nugget, uh, asterisk is the example that the OS communicate IS innovative ! |
18:40.34 | stoffell_h | communicate=community :) |
18:40.42 | Nugget | if it wasn't possible to earn a living coding, there would be a lot fewer volunteers with the expertise and disposable income to donate their time |
18:41.08 | Nugget | with the possible exception of dundi, asterisk isn't very innovative at all. |
18:41.10 | Nugget | it's just free. |
18:41.15 | frk2 | i dont know if ALL of these coders are employed by companies who make money selling code |
18:41.28 | Nugget | what's asterisk do that people haven't been doing for decades with commercial solutions? |
18:41.45 | frk2 | Nugget, the ability to customize |
18:41.52 | Nugget | that's not unique either |
18:41.56 | stoffell_h | software pbx.... |
18:41.59 | frk2 | thats pretty unique |
18:42.11 | file | fight fight fight! |
18:42.14 | frk2 | hahahah |
18:42.16 | Zeeek | I think you're all missing the religious connatations here |
18:42.20 | stoffell_h | no fight! :d |
18:42.22 | Nugget | open source is a great way to commoditize innovation that comes from the commercial world, but it hasn't really demonstrated that it's a good way to actually innovate directlyu |
18:42.34 | stoffell_h | gotta go catch a beer (hope it's free :p ) |
18:42.45 | Nugget | gimp isn't innovative. gimp is a free clone of photoshop (or, at least, tries to be) |
18:42.49 | frk2 | dude |
18:42.57 | frk2 | KDE did in like 3 years what took windows 10 |
18:43.09 | Nugget | but only after windows did it first. |
18:43.22 | frk2 | and i think by kde 4 it would have innovated past windows |
18:43.33 | frk2 | gotta give them time man |
18:43.34 | Nugget | and kde is still years behind the current state of the art when it comes to desktop interfaces. |
18:43.53 | dpryo | Nugget: What do you consider "state of the art desktop"? |
18:43.54 | Nugget | what is kde working on, then, that's not already present in windows or os x? |
18:43.58 | frk2 | oh come on. kde is pretty damn good - many people i know compare it to windows |
18:44.16 | Nugget | I didn't say it wasn't good. I said it wasn't innovative. |
18:44.17 | frk2 | nugget check their site out... they are gonna go pretty crazy |
18:44.29 | frk2 | with kde 4 |
18:44.36 | Nugget | why do you assume I am not familiar with kde? |
18:45.23 | dpryo | All those nice things like you can remove window borders, have different styles on different windows etc.. That's pretty cool. |
18:45.28 | frk2 | cuz if you were, you wouldn't say many things werent innovative |
18:45.37 | frk2 | yeah man - STYLES |
18:45.40 | frk2 | thats unknown to windows |
18:45.44 | dpryo | It sure has been available in enlightenment for 10 years, but I haven't seen it in windows ;) |
18:45.57 | SwK_ | nugget is just a irc troll he has never done anything important or well known |
18:46.03 | frk2 | lol |
18:46.04 | DoktorGreg | are the debian update servers only down for me? |
18:46.23 | Nugget | skins are your idea of innovation? wow. |
18:46.27 | dpryo | DoktorGreg: Those are distributed around the world, I don't think they all go down at once |
18:46.45 | frk2 | dude its a desktop!!!! yes in that scenario skins are innovative |
18:46.49 | SwK_ | nugget trolling again eh? |
18:46.49 | DoktorGreg | um, i cant get on ... any right now... |
18:47.04 | DoktorGreg | skins? |
18:47.26 | DoktorGreg | all i wana know, is can i set the default windows 2000 'skin' |
18:47.44 | Nugget | I'd rather use an environment where the clipboard works between applications and for data other than plain 7bit clean text. |
18:47.57 | Nugget | or an environment that's making good use of 3d acceleration for common desktop tasks. |
18:48.27 | Nugget | but if the epitome of an environment for you is to be able to put rounded corners on windows, then I guess I can see where you'd be impressed by kde. |
18:48.39 | frk2 | hahah |
18:48.48 | frk2 | i see your point |
18:49.24 | file | someone guess what Nugget uses for an OS... |
18:49.30 | Nugget | I use them all. |
18:49.39 | frk2 | but you must have one primary one |
18:49.41 | SwK_ | nugget uses DOS |
18:49.48 | Nugget | ok, I don't use dos. :) |
18:49.49 | dpryo | The abillity to have more users using a single computer is pretty innovative.. Connect more keyboards, displays etc and run more desktops on the very same computer. |
18:50.00 | DoktorGreg | DOS rocks! |
18:50.01 | Nugget | that's not an open source innovation. |
18:50.02 | blitzrage | DOS rox! |
18:50.08 | dpryo | Nugget: Sure it is. |
18:50.13 | Nugget | no, it plainly is not. |
18:50.28 | SwK_ | dpryo no its not |
18:50.29 | dpryo | It wouldn't work in windows, nor OSX. |
18:50.29 | Nugget | people did that with dos, with proprietary software, in the early '80s |
18:50.33 | DoktorGreg | FreeDos has expanded DOS intil a 32 bit multitasking operating system |
18:50.35 | dpryo | It works with X.org |
18:51.14 | frk2 | i guess its impossible to say whether something is an opensource innovation or a closed source one anyways. |
18:51.25 | Nugget | how so? |
18:51.36 | Nugget | you just have to look to see who first came up with the idea and implemented it. |
18:51.40 | SwK_ | doktorgreg and thats not even innovative... I ran majorBBS back in the day that was dos based 32bit multitasking and could handle 100s of users on 386s |
18:51.44 | frk2 | nugget |
18:51.57 | frk2 | how do you decide who invented the wheel for example? |
18:52.05 | frk2 | the 'idea' of the wheel |
18:52.20 | Nugget | that's hardly a fair comparison. |
18:52.32 | frk2 | no man |
18:52.32 | SwK_ | its not the inventing that you are pointing out here, its all the reinvention of old ideas |
18:53.03 | VoicePulse | Nugget is correct in saying that most of the things you guys are mentioning existed a long time ago, developed by companies like IBM, Xerox and AT&T -- long before the Internet or open-source communities as large as current ones existed. |
18:53.20 | *** join/#asterisk zyth (n=Anon5017@66.244.197.93) |
18:53.36 | VoicePulse | www.uspto.gov is where you look to see who came up with it first |
18:54.32 | SwK_ | and dont forget people like QuarterDeck that extended DOS into a multitaking environment with QuarterDesk, and allowed it to access more then 1M of ram w/ QEMM, and Pharlaps which made it even easier to access 32bit based memory |
18:54.35 | DoktorGreg | lol, thats right, nothing has been invented since 1968 |
18:55.00 | frk2 | hehe |
18:55.20 | frk2 | i dont know whats counted as innovation |
18:55.37 | VoicePulse | DoktorGreg: That's why he asked you to name one to support your argument... |
18:55.45 | SwK_ | i mean how innovative is a telephone really? yeah we can make it do new tricks, but you can still hook up A.G.Bell's original phone to a modern pots like and make a call with it |
18:55.47 | DoktorGreg | ok, voip |
18:55.51 | DoktorGreg | voip is new |
18:55.57 | SwK_ | VoIP is not new |
18:56.09 | VoicePulse | I believe the discussion is if open-source invented XYZ before closed-source. |
18:56.54 | DoktorGreg | http://www.voipreview.org/news.details.aspx?nid=51 |
18:56.54 | Nugget | open source has a poor track record for actually producing innovative software and features. |
18:56.55 | SwK_ | Packet Switched Networks and pushing realtime (or near realtime) data feeds have been around forever... see TymeNet and x.25 |
18:57.07 | DoktorGreg | there you go, voip is new |
18:57.18 | DoktorGreg | well except for voip |
18:57.39 | frk2 | :) |
18:57.49 | Nugget | to my eyes, open source excels mainly at solving the challenge of "I want that cool software, but I don't want to pay what they charge for it" |
18:58.06 | DoktorGreg | like Apache??? |
18:58.15 | DoktorGreg | php |
18:58.18 | Nugget | or, to be fair, "I want that cool software but the terms of use for it are ridiculous" |
18:58.27 | DoktorGreg | databases |
18:59.01 | frk2 | well see doktor what nugget is saying is that 'mysql' is not innovative, which i can understand, because databases have been around for many years |
18:59.03 | DoktorGreg | bittorrent |
18:59.16 | Nugget | bittorrent is an excellent example of open source innovation, yes. |
18:59.29 | DoktorGreg | so is voip |
18:59.46 | frk2 | voip isnt dude |
18:59.48 | SwK_ | VoIP != OpenSource innovation |
18:59.49 | Nugget | so cisco callmanager came after asterisk? |
18:59.53 | frk2 | voip s been around forever |
18:59.57 | DoktorGreg | since 95 |
19:00.01 | SwK_ | CCM != innovation |
19:00.08 | DoktorGreg | 95 != forever |
19:00.20 | DoktorGreg | or even a long time |
19:00.31 | SwK_ | cisco didnt innovate shit in VoIP they bought it |
19:00.35 | Nugget | what is an example of an open source voip solution that came prior to 1995, then? |
19:00.50 | Nugget | that first, innovative voip implementation that was open source. |
19:01.53 | DoktorGreg | its maybe the first product that is open soice |
19:02.06 | DoktorGreg | but the foundations of voip were opensource |
19:02.16 | Nugget | why do you say that? |
19:02.31 | Nugget | and how does that make voip, then, an open source innovation? |
19:03.10 | zyth | I have 16 rooms in a hotel I need to provide phone service to - basically, the call from telco comes to a main #, I then want people to be able to punch in the room # and talk to whomever. Is VoIP a viable solution to that? |
19:03.19 | zyth | w/o it costing much.. |
19:03.41 | SwK_ | zyth: I have asterisk in a 200 room hotel |
19:03.48 | Nugget | heh |
19:04.08 | zyth | SwK_: any appreciable cost outlay I'd need to be aware of before trying to implement such a thing? |
19:04.24 | SwK_ | channel banks T1 Cards etc... |
19:04.26 | justinu|laptop | probably around 5 grand |
19:04.28 | justinu|laptop | tops |
19:04.34 | SwK_ | its just another PBX |
19:04.51 | tomcontr3 | how can I doenload this SVN version? |
19:04.52 | tomcontr3 | http://svn.digium.com/view/asterisk/team/oej/sipregister |
19:04.59 | DoktorGreg | unless you have someone install it for you, in which case it is about 30k |
19:05.07 | zyth | is that apart from our internet link? |
19:05.28 | zyth | DoktorGreg: yeah, for that we could lay regular phones to the building. |
19:05.43 | kaz0358 | i have done a bit of searching and i cannot find what all the valid characters are for an iax2 password.. i know letters and numbers.. but what symbols are also valid? |
19:05.45 | SwK_ | zyth you dont necesarily need as IP link to the internet to use asterisk |
19:06.04 | SwK_ | it can replace a legacy Avaya, nortel or siemens hospitality solution |
19:06.18 | frk2 | definitely |
19:06.35 | frk2 | but do you HAVE to go pots? |
19:06.37 | Zeeek | zyth you'd usually be safe with RADIX50 characters |
19:06.38 | SwK_ | you can do 16 stations for 5 to 10K USD |
19:06.45 | zyth | hmm ok |
19:07.09 | Zeeek | nah, that' can't be right |
19:08.34 | tomcontr3 | can any one help me to install this SVN version? |
19:08.34 | tomcontr3 | <PROTECTED> |
19:09.01 | Qwell | tomcontr3: svn co http://svn.digium.com/view/asterisk/team/oej/sipregister asterisk-sipregister |
19:09.02 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
19:09.09 | Qwell | cd asterisk-sipregister && make install |
19:09.25 | tomcontr3 | just make install |
19:09.26 | tomcontr3 | ? |
19:09.31 | kaz0358 | radix50.. do you think & and ! are valid characters? |
19:09.35 | Qwell | yes |
19:09.37 | tomcontr3 | ok |
19:09.45 | Qwell | unless it bitches about needing to restart make...then just run it again |
19:10.17 | tomcontr3 | Generating the configure script ... |
19:10.29 | tomcontr3 | its running a config script , is that normal? |
19:11.07 | SwK_ | yes |
19:11.13 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-170-22.bflony.east.verizon.net) |
19:11.17 | hwt | i'm having some serious problems getting anything but xten to send dtmf to asterisk |
19:11.24 | hwt | or rather getting asterisk to understand dtmf |
19:11.26 | hwt | e.g. in meetme. |
19:11.28 | hwt | any tips? |
19:11.41 | hwt | i suspect it's trying to send inband, and that's not very smart when using gsm. |
19:12.13 | *** join/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net) |
19:12.25 | tomcontr3 | Qwell, it didn worked |
19:12.37 | tomcontr3 | it seems that this needs to be configured first |
19:12.48 | tomcontr3 | and now it says |
19:12.48 | tomcontr3 | gcc -g -o menuselect ../strcompat.o menuselect.o menuselect_curses.o ../mxml/libmxml.a -lncurses |
19:12.49 | tomcontr3 | gcc: ../strcompat.o: No such file or directory |
19:12.57 | hwt | tried rfc, auto and inband |
19:12.59 | hwt | no luck |
19:13.41 | *** part/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net) |
19:13.52 | SuPrSluG | i'm having a similar issue. but it's w/ broadvoice. When I dial in through voicepulse I can get to the meetme 22. when I got through broadvoice it dials 2 in my ivr and never waits for the second 2 |
19:13.53 | kaz0358 | hwt, what do you have dtmfmode set to in sip.conf? |
19:14.41 | kaz0358 | hwt, the default is rfc2833 unless you have changed it. |
19:15.21 | hwt | kaz0358: everything. auto, rfc and inband |
19:15.29 | Zeeek | with reloads each time? |
19:15.37 | hwt | yes, of course. |
19:15.48 | Zeeek | did you change the setting in X-Lite |
19:15.51 | Zeeek | ? |
19:16.02 | hwt | Zeeek: i haven't done anything for it to work in x-lite. |
19:16.10 | hwt | but i have tried calling in from my gsm |
19:16.12 | hwt | and no luck |
19:16.15 | Zeeek | there is a setting that forces inband |
19:16.23 | *** join/#asterisk trimi` (i=Whatt@62.162.243.194) |
19:16.25 | hwt | yeah, but that won't work with gsm |
19:16.31 | trimi` | hey pla any on can tell me what does this error mean Apr 30 20:33:10 NOTICE[13712]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
19:16.35 | trimi` | <PROTECTED> |
19:16.37 | Zeeek | exactly, you'd want to make sure it wasn't set |
19:17.02 | hwt | yeah, but i've tried setting the option globally. |
19:17.03 | SwK_ | trimi that means the devices is most likely not registered or asterisk doesnt know that endpoints IP |
19:17.08 | Zeeek | trimi` use pastebin to show the dial command |
19:17.30 | trimi` | its working if i use directly |
19:17.43 | trimi` | but if i register my asterisk in other user |
19:17.47 | trimi` | not dialing |
19:17.53 | Zeeek | ok |
19:18.12 | trimi` | btw how do i see which codec its used while a sip its calling |
19:18.26 | Zeeek | hwt what did yiou mean calling from your gsm? |
19:18.29 | robin_sz | sip show channels |
19:18.37 | trimi` | thnx |
19:18.37 | hwt | Zeeek: yes, calling from my gsm |
19:19.03 | Zeeek | what gsm? |
19:19.08 | PakiPenguin | cellphone |
19:19.11 | PakiPenguin | :p |
19:19.16 | Zeeek | gsm doesn't mean cellphone |
19:19.23 | robin_sz | .me nods |
19:19.25 | kaz0358 | trimi, not following too closely.. but asterisk does tell you the codec used if you are on the console and gave -vvvv |
19:19.33 | Zeeek | nor does a cellphone put gsm to asterisk |
19:19.42 | trimi` | <kaz0358> if i use iax yes |
19:19.45 | trimi` | if i use sip no |
19:20.00 | PakiPenguin | :) |
19:20.25 | Zeeek | hwt plus cellphones are so bad they often can't send tones to regular lines let alone asterisk |
19:20.31 | *** join/#asterisk gr0mit (n=Gr0mit@extrt.txrx.org.uk) |
19:20.45 | PakiPenguin | Zeeek, mine works alright :) |
19:20.56 | Zeeek | it's connection dependent |
19:21.05 | Zeeek | anyway |
19:21.11 | robin_sz | dtmf over a gsm compressed link can be dodgy |
19:21.22 | Zeeek | eggsakly |
19:21.30 | PakiPenguin | yup |
19:21.42 | Zeeek | someone has a nice little test out there |
19:23.24 | robin_sz | i'll tell you what I want, what I really, really, want, |
19:24.00 | robin_sz | I wanna, I wanna ... someone to send me thevery latest, not on the wiki anymore, GXP2000 firmware |
19:24.00 | [hC] | proper polycom BLF. |
19:24.21 | [hC] | or fixit instructions for my linksys wip300! |
19:24.31 | SuPrSluG | anyone know why asterisk dials exten 2 immediately instead of waiting for 22? only in one context. the other inbound line works fine. dtmf? |
19:24.50 | Zeeek | Banana-Rama? |
19:24.58 | Zeeek | that's dredging pretty low! |
19:25.04 | robin_sz | well, yes |
19:25.12 | robin_sz | but we are talking grandstream here |
19:25.19 | robin_sz | I had to go as low as I could |
19:25.20 | Zeeek | heh |
19:25.34 | [hC] | you got a 2 for 1 there with spice girls, too. |
19:25.37 | Zeeek | to think I *almost* bought one of those |
19:25.52 | hwt | Zeeek: well, it doesn't work from snom phones or another ata in the network |
19:25.55 | hwt | Zeeek: either. |
19:26.15 | Zeeek | DTMF? no where? |
19:26.25 | robin_sz | ahh yes, the Spice Girls. a group so .. so ... unsullied by talent |
19:26.38 | hwt | Zeeek: yes, the dtmf doesn't work from anywhere except my x-lite. |
19:26.43 | Zeeek | well the talent was only visible to managerial elements prolly |
19:27.07 | robin_sz | you mean your boss liked them? |
19:27.19 | Zeeek | no their managers prolly saw "talen" |
19:27.31 | robin_sz | ahh |
19:27.46 | robin_sz | their managers probably saw $$$ ... |
19:27.51 | Zeeek | One famous actor years ago said "talent? Yeah it's spelled s.u.c.k." |
19:28.03 | Zeeek | but we digress |
19:28.08 | robin_sz | we do ... |
19:28.16 | robin_sz | so, the latest firmware? |
19:28.21 | Zeeek | so SIP, gsm, no dtmf - a geeks' dream |
19:28.31 | Zeeek | find a GS geek forum |
19:28.56 | Zeeek | or... try the google related time machine thingie |
19:29.03 | Zeeek | or the google cache |
19:29.12 | robin_sz | ahh ... point. |
19:29.19 | robin_sz | wayback |
19:29.20 | Zeeek | sometimes the shit is there and they just changed the links :) |
19:29.41 | Zeeek | but before you get all wet, I doubt that'll work |
19:30.05 | Zeeek | it's just a pacifier I sent along so yopur crying in the middle of the night doesn't keep us awake |
19:30.46 | Zeeek | damn, well I solved a few of my little problems tonight. 2 down one to go |
19:31.21 | Zeeek | have you ever walked in a store and asked for a SIM card? |
19:31.22 | gr0mit | robin_sz, what sw do you want for the GS? |
19:31.48 | Zeeek | he's looking for foimware |
19:34.37 | gr0mit | yes which voision? |
19:35.59 | Zeeek | the "foim" part indicates a recent, but not the most recent version. Whereas "firware" are wooden trousers |
19:36.33 | Zeeek | it's either time to go to bed or time for a beer. |
19:36.44 | gr0mit | both prolly |
19:37.03 | Zeeek | naw, I don't like balancing it in bed |
19:39.50 | hwt | hmm, pbx.c:1406 ast_func_write: Function LANGUAGE not registered |
19:39.52 | hwt | what does that mean? |
19:39.58 | hwt | what module am i missing? |
19:40.08 | Qwell | func_language, would be my guess |
19:40.25 | hwt | Qwell: nah. |
19:40.42 | hwt | perhaps pbx_functions.so |
19:40.46 | Qwell | func_language.so Channel language dialplan function |
19:40.54 | Qwell | No, func_language |
19:41.37 | hwt | uhm, i don't have that module. |
19:41.51 | Qwell | Then...sounds like you've got problems |
19:42.01 | Qwell | such as using a version of asterisk that's too old |
19:42.33 | hwt | pbx_functions.so did the trick |
19:42.42 | hwt | Qwell: maybe func_language.so is deprecated? |
19:42.46 | Qwell | no |
19:43.23 | Qwell | oh, heh |
19:43.32 | Qwell | Deprecated. Use CHANNEL(language) instead. |
19:44.09 | robin_sz | dang, going through the tiki-wiki page history, it seems that the latest firmware never made it to the wiki ... bummer. |
19:45.15 | robin_sz | gr0mit, I would like any version that makes my display work :) |
19:45.40 | gr0mit | which version do you have? |
19:45.46 | robin_sz | 1.1.0.1 |
19:46.00 | gr0mit | hmmm i think i have 1.1.0.4 |
19:46.03 | gr0mit | let me check |
19:46.08 | robin_sz | 1.1.0.4 wold be lovely :) |
19:46.30 | robin_sz | or at lease, stands a better chance of getting my display working |
19:46.44 | robin_sz | it goes blank the moment after the phone boots :( |
19:47.01 | robin_sz | apparently I have an older phone. |
19:49.36 | gr0mit | need to got upstairs and check |
19:49.47 | robin_sz | just access the phone web onteface |
19:51.54 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-75-9-244-217.dsl.sfldmi.sbcglobal.net) |
19:54.48 | *** join/#asterisk SparFux (n=player@e182022054.adsl.alicedsl.de) |
19:55.27 | gr0mit | sorry but i have the same 1.1.0.1 as you |
19:55.43 | *** join/#asterisk darby_t (i=darby_t@aaph204.neoplus.adsl.tpnet.pl) |
19:55.53 | gr0mit | the phone crashed when i logged in t o the web interface. not entirely stable |
19:56.11 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
19:56.31 | robin_sz | ahh, ok |
19:56.35 | robin_sz | thanks for looking |
19:59.34 | *** part/#asterisk darby_t (i=darby_t@aaph204.neoplus.adsl.tpnet.pl) |
20:01.11 | SparFux | I would like to have an application app_streamsound(program,parameters) which would use <program> to play a sound to stdout and stream that to an asterisk extension. Like MP3Player does it with mpg123, but just an arbitrary program, not only mpg123. |
20:01.22 | *** join/#asterisk dlynes_ (i=1000@S010600c09f9a0fc4.vc.shawcable.net) |
20:03.10 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
20:03.39 | *** join/#asterisk Cardoe (n=Cardoe@gentoo/developer/Cardoe) |
20:03.57 | Cardoe | how do you guys feel about freePBX as a viable system for a home Asterisk setup? |
20:04.19 | Cardoe | anything business related is bad with *@Home |
20:04.50 | Hmmhesays | i disagree |
20:04.50 | gr0mit | Cardoe - just use raw asterisk. |
20:05.19 | gr0mit | once you get the hang of config files you can do anything from vi |
20:05.29 | SparFux | gromit: ack. |
20:07.48 | dlynes_ | Cardoe: Well, there's one advantage of AMP |
20:07.57 | dlynes_ | Cardoe: Telephone techs can understand how to use it |
20:09.49 | gr0mit | prolly fine but not very flexible |
20:10.30 | Cardoe | I'm debating if I should just toss in freePBX on top of my Asterisk box here at home. |
20:10.33 | Cardoe | or just do it myself. |
20:10.39 | dlynes_ | telephone techs only need something that can do a very finite set of things, anyways |
20:10.40 | gr0mit | just do it yourself |
20:11.00 | Cardoe | dlynes: what does it include that telephone techs want? |
20:11.14 | dlynes_ | Cardoe: ease of configuring |
20:11.15 | Cardoe | I also don't have a POTS line anymore. Gonna go through SIP purely. |
20:11.31 | dlynes_ | Cardoe: most telephone techs don't know computers well enough to log into linux, fire up vi, and edit some text files |
20:12.03 | Cardoe | what telephone techs would be messing with my system? |
20:12.14 | dlynes_ | Cardoe: so for them, it's a gui such as amp or something similar, or nothign |
20:12.27 | dlynes_ | Cardoe: none...I'm just saying for offices, not for homes |
20:12.58 | Cardoe | Well I was talking about my home machine |
20:13.07 | dlynes_ | telephone techs working for interconnects are the ones that traditionally install office pbxes, and maintain them |
20:13.13 | Cardoe | at the office we have a propritary system developed by some company. |
20:13.16 | gr0mit | just keep it as raw text. if you give them a gui you are doing yourself out of a job ;-) |
20:13.18 | Cardoe | that provides a web interface to asterisk |
20:13.36 | dlynes_ | Cardoe: Yeah...I was just saying for the office, amp is somewhat useful...for the home, it's total overkill |
20:14.01 | Cardoe | any features you guys find particularly useful? |
20:14.09 | Cardoe | I played with asterisk like 3 years ago |
20:14.09 | dlynes_ | Dial() |
20:14.16 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:14.21 | SparFux | Aha! I get a feeling that eagi is just what I am searching for. I can use fd 3 to pipe any sound from any application to asterisk, right? :-) |
20:14.22 | Nugget | Aren't you glad you used Dial()? |
20:14.27 | dlynes_ | Yep |
20:14.34 | Strom_C | Don't you wish everybody did? |
20:14.37 | dlynes_ | It's the most powerful tool in my whole arsenal :) |
20:14.52 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
20:15.07 | dlynes_ | Good morning, Dr-Linux |
20:15.16 | Cardoe | any recommendations for outgoing SIP/IAX? |
20:16.35 | Dr-Linux | dlynes_: thanks friend |
20:16.41 | Dr-Linux | it's night though heere |
20:16.53 | dlynes_ | Dr-Linux: oh...thought it'd be like 1 or 2 am |
20:17.45 | Dr-Linux | dlynes_: yah it is |
20:17.56 | *** join/#asterisk Lord_Drachenblut (n=Lord@12.210.117.62) |
20:18.07 | dlynes_ | well then, it's morning :) |
20:18.24 | dlynes_ | hanji? |
20:23.10 | *** join/#asterisk ozverenm20 (n=ozverenm@162.27.103-84.rev.gaoland.net) |
20:23.44 | ozverenm20 | someone expert in ISDN card ? |
20:27.01 | dlynes_ | Cardoe: you mean providers? |
20:27.05 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
20:27.39 | dlynes_ | Cardoe: Try www.calltermination.com....there's a whole slew of them on there |
20:28.13 | Cardoe | thx |
20:28.36 | dlynes_ | a good number of them will do voip escrow, too |
20:31.13 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
20:35.13 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
20:38.47 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
20:38.48 | *** mode/#asterisk [+o kram] by ChanServ |
20:48.06 | ozverenm20 | i need help about DACS and audio buffer hooking |
20:48.32 | ozverenm20 | i want to make asterisk between my PBX and my telco |
20:48.56 | ozverenm20 | i also need to record B channels traffic without decrease audio quality |
20:49.03 | ozverenm20 | how can i do it ? |
20:59.54 | *** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net) |
21:00.08 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
21:03.24 | tainted- | ozverenm20 PRI? |
21:04.09 | ozverenm20 | yes |
21:04.13 | ozverenm20 | yes pri |
21:04.45 | tainted- | u'd need some zaptel cards |
21:04.53 | ozverenm20 | wich ? |
21:04.56 | tainted- | i'd recommend sangoma |
21:05.00 | RoyK | ~ser |
21:05.02 | jbot | hmm... ser is Sip Express Router - see http://www.iptel.org/ser/ |
21:05.03 | ozverenm20 | sangoma ? |
21:05.20 | tainted- | how many B channels |
21:05.20 | kram | i'd recommend digium ;-) |
21:05.35 | tainted- | sangoma works better though |
21:05.36 | RoyK | sangoma has good cards, but they have a higher cpu overhead than digium's due to the wanrouter stuff |
21:06.06 | tainted- | how much higer |
21:06.08 | tainted- | higher |
21:06.10 | DoktorGreg | it is really hard to recommend anything but digium, because that is the only thing i have used |
21:06.17 | RoyK | about the double for what i can see |
21:06.28 | kram | not to mention we actually support the project |
21:06.49 | ozverenm20 | i am searching for a card that do hardware switching ( bridging ) and sent all audio buffer to me |
21:06.51 | RoyK | sangoma's stuff can be used to do other stuff, though, like ss7 with ss7box |
21:06.51 | DoktorGreg | well that too, digium actually makes asterisk |
21:07.18 | RoyK | but I beleive I'd recommend digium's E1/T1 cards for plain asterisk PRI works |
21:07.50 | tainted- | RoyK do u have screencaps of cpu usage? |
21:08.26 | kram | tainted: what makes you recommend sangoma out of curiosity |
21:08.36 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
21:08.39 | ozverenm20 | digium are reputed to be not stables , no ? |
21:08.46 | ozverenm20 | digium cards |
21:08.46 | kram | the digium card does support on-card switching |
21:08.51 | DoktorGreg | why would you want to route old slow single megabit connections anyhow? |
21:08.57 | kram | how would we be less reliable? |
21:09.08 | ozverenm20 | sangoma support on card switching ? |
21:09.11 | kram | i don't know |
21:09.34 | DoktorGreg | ive had zero stability problem with the two digium cards ive had |
21:10.01 | RoyK | kram: the old cards have lots of bad habits. dunno about the new ones, though. we still run zaptel 1.0.7 since we can't upgrade to newer zaptel since it doesn't work with them, and we can't take the cards out of production to have them upgraded :P |
21:10.04 | DoktorGreg | and i was using 8 channels at a time last week |
21:10.07 | kram | i should clarify the quad T1/E1 supports on-card switching not the single |
21:10.18 | RoyK | ozverenm20: sangomas doesn't support oncard switching either |
21:10.28 | DoktorGreg | whats on channel switching? |
21:10.34 | DoktorGreg | er oncard |
21:10.55 | tainted- | kram ever since 1.0.4 we've had various issues with digium cards |
21:11.03 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
21:11.03 | kram | means that when a call starts and ends on a card, it doesn't have to pass all the way up to the zaptel layer |
21:11.05 | RoyK | DoktorGreg: tell card to attach channel n to channel m and forget about it |
21:11.09 | kram | it shaves a millisecond or two off of the latency |
21:11.24 | kram | it also means that no matter what happens in interrupt land there is no possibility of missing a sample |
21:11.25 | RoyK | it also saves cpu load.... |
21:11.30 | tainted- | recently after discovering sangoma, we've realized that is JUST WORKS |
21:11.53 | DoktorGreg | the native bridging thing that digium cards do when a call just goes in and out of the pri ports? |
21:11.59 | Assid | is there any way of adding gsm support to polycom? |
21:12.01 | tainted- | but it could be zaptel like RoyK was saying |
21:12.13 | kram | tainted: what was the situation in which your digium cards did not work, if i may ask |
21:12.30 | RoyK | tainted-: with a single te410p in a couple of single cpu boxes, they use something like 40% cpu with full load, whereas a newer dual cpu box gives about the same load with a sangoma |
21:12.42 | RoyK | meaning about double or a little more cpu |
21:12.51 | RoyK | per channel, no transcoding |
21:13.03 | RoyK | simply more abstraction layers etc |
21:13.05 | kram | our 2nd gen firmware includes some special SMP optimizations that allow the performance to be improved significantly |
21:13.27 | RoyK | kram: but i still have to mail you the cards to get the new firmware :P |
21:13.40 | Assid | kram: you work for digium? |
21:13.45 | RoyK | lol |
21:13.47 | I-MOD | lol |
21:14.10 | RoyK | hm. i might have one a card spare. kram, how long does it take to get one upgraded and mailed up here? |
21:14.12 | Assid | whatd i say so funny? |
21:14.23 | brookshire | assid: kram is the creator of asterisk |
21:14.23 | RoyK | Assid: ever heard the name Mark Spencer? |
21:14.28 | kram | royk: i tell you what roy, i'll talk to malcolm and if you've been staying off the trolling i'll see what we can work out |
21:14.30 | Assid | oh thats him?!?!!? |
21:14.33 | Assid | damn.. i didnt know |
21:14.36 | SwK_ | lol |
21:14.38 | RoyK | kram: thanks |
21:14.55 | Assid | shit.. sorry man.. didnt know |
21:15.01 | RoyK | :) |
21:15.47 | ozverenm20 | junghanns cards does support on card switching ? |
21:15.53 | kram | tainted: can you clarify a bit your issues? |
21:16.10 | kram | tainted: if you're having real issues, i'd like to try to make sure they're things that have been resolved |
21:16.44 | kram | oz: i don't know much about his e1 cads |
21:16.46 | kram | cards |
21:16.55 | tzanger | whoa, kram is actually on IRC? |
21:17.00 | file | don't everyone attack kram |
21:17.11 | Strom_C | it's kram! |
21:17.14 | file | tzanger: oh goody you're here... I'm doing some chan_iax2 work right now... that I'll want you to test |
21:17.15 | Assid | hehe |
21:17.33 | tzanger | file: I can try but I've gotta head out very very shortly |
21:17.38 | Assid | man.. i feel like everyones doing something important except me |
21:17.47 | ozverenm20 | only eicon support on card switching ? |
21:17.47 | file | tzanger: it can wait till tomorrow or whatever |
21:17.53 | tzanger | ok |
21:17.54 | Nivex | Assid: don't worry, I just lurk here. |
21:17.59 | kram | you mean for BRI? |
21:18.32 | tainted- | kram there were audio problems, asterisk would segfault, IRQ incompatibilities |
21:18.45 | kram | you seriously think an asterisk segfault would be related to hardware? |
21:18.52 | kram | tell me about the audio problems and irq issues though |
21:19.09 | Assid | hrmm.. im segfault.. check ram..? |
21:19.35 | ozverenm20 | if segfault do 112 call ? :) |
21:19.49 | tainted- | kram well audio issues turned out to be problems dealing with MMX CPU |
21:20.00 | kram | okay |
21:20.05 | kram | so what were the issues with the cards then |
21:20.11 | Assid | Nivex: i just dont wanna be sitting around doing nothing.. lets see.. what can i do |
21:20.16 | tzanger | tainted-: did you get my messages? |
21:21.32 | Nivex | Assid: spread the word! |
21:21.59 | Assid | of? |
21:22.34 | tzanger | Assid: the word is legs! |
21:22.37 | Nivex | A |
21:22.37 | *** join/#asterisk faljse (n=martin@83-65-243-10.dynamic.xdsl-line.inode.at) |
21:22.45 | kram | work on the bug tracker :) |
21:22.50 | kram | help us fix issues in asterisk etc |
21:22.51 | tainted- | kram all i'm saying is that for me, sangoma was drop and play |
21:23.01 | faljse | is there a limit for agi parameter length(100 chars maybe?) |
21:23.04 | kram | but wait, i don't understand yet |
21:23.10 | tainted- | u don't have to |
21:23.13 | ozverenm20 | does digium cards support on chip card + audio buffer recording ? |
21:23.21 | kram | you changed general zaptel parameters that don't have anything to do with the hardware |
21:23.22 | Assid | kram: dunno C .. want php ? |
21:23.29 | tzanger | tainted-: not for me. I have an A104D that has zero audio... D channel works great but no audio :-) |
21:23.32 | kram | i'm trying to understand what actually was the hardware issue |
21:23.48 | kram | i want to be sure we're doing well for people |
21:23.55 | *** join/#asterisk cr_0 (n=y@toronto-HSE-ppp4334596.sympatico.ca) |
21:24.03 | tzanger | the only hardware issues I know of are the VPMs that trigger DTMF on (generally) female voices |
21:24.05 | kram | i mean if you just want sangoma that's fine but if you're actually having issues with digium hardware i'd like to know about it |
21:24.22 | Assid | hell.. you guys are.. i love *.. i'd love to find a way for it to be a source of income one day for me |
21:24.23 | Assid | hehe |
21:24.30 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-75-9-244-217.dsl.sfldmi.sbcglobal.net) |
21:24.36 | kram | assid: where are you located? |
21:24.36 | cr_0 | i don't need cdr_odbc.conf so i removed it, but now asterisk warns about disabling some odbc module when restarting. how do i tell asterisk not to try to load the module? |
21:24.43 | Assid | india |
21:25.00 | tzanger | speaking of VPMs... kram, is there any thought on a VPM with a heavier (i.e. Tellabs-quality, long tail, bidirectional) echo cancel? |
21:25.09 | file | India? you can be imported! just like me. |
21:25.29 | kram | :) |
21:25.38 | brookshire | file: you haven't been imported yet |
21:25.51 | file | pesky country |
21:25.51 | Assid | nah.. my lifes wayyyy tooo complicated.. would take forever for my papers to go through.. one of the reasons i dont live with my dad.. he fucked my life up |
21:25.56 | Damin | Sigh.. |
21:26.08 | ozverenm20 | no response for my problem ? |
21:26.18 | tainted- | kram i'm sure the irq issues were fixed, and the mmx thing wasn't exactly a digium card problem |
21:26.20 | Damin | First day that I get a break in two weeks and I have to put together a bike for my 4 year old.. |
21:26.24 | brookshire | oz: what's your problem? |
21:26.39 | kram | assid: are you a web coder? :) |
21:26.43 | Assid | yep |
21:26.51 | Assid | php mostly.. |
21:26.54 | kram | javascript? |
21:27.04 | brookshire | hah |
21:27.06 | Assid | not that well.. but decent enough to get my work done |
21:27.10 | brookshire | ajam! |
21:27.11 | brookshire | :D |
21:27.13 | Assid | mostly just php |
21:27.34 | Assid | wrote a call logger to integrate with asterisk's cdr |
21:27.48 | Damin | Assid: I need a fully Multitasking Kernel written in Javascript that can run inside of IE. Can you do it? ;) |
21:27.50 | Assid | pretty simple shit |
21:28.45 | brookshire | damin: i need a virtual machine that runs inside of java, that can run firefox |
21:28.49 | tainted- | kram also i received better phone support from sangoma |
21:29.58 | brookshire | i serious doubt that |
21:30.03 | brookshire | seriously |
21:30.17 | Assid | im planning to maybe create a web interface similar to AMP.. but from a business angle.. where users can make their own pbx and stuff |
21:30.27 | Damin | brookshire: Hehehe.. While it doesn't run inside Java the Vmware Player Browser Appliance.. http://www.vmware.com/vmtn/appliances/directory/browserapp.html |
21:30.35 | kram | tainted: can you explain what you didn't like from our phonoe support? |
21:30.41 | ozverenm20 | tainted: its sure sangoma support on-card switching ? |
21:31.46 | *** join/#asterisk Eggplants (i=No@dsl-201.cascadeaccess.com) |
21:32.11 | kram | i think tainted is just a fan of sangoma and either can't or doesn't want to express a real beef with digium |
21:32.26 | kram | which is unfortunate since, on the whole, it's digium's code and not sangomas that he's using ;-) |
21:32.42 | tainted- | well now that's not fair |
21:32.58 | tainted- | i've supported digium many times |
21:33.02 | Assid | kram: why whatd you have in mind? |
21:33.06 | kram | okay then express your beef in something that i can take to my management in order to try to get your issues resolved |
21:33.17 | tainted- | i've purchased g729 licenses, hardware, supported new users |
21:34.16 | kram | assid: just generally looking for people that want to explore the ajam stuff i've been working on |
21:34.16 | kram | web enabling asterisk |
21:34.16 | Assid | oh.. |
21:34.16 | Assid | realtime database ? |
21:34.16 | brookshire | manager in xml |
21:34.17 | Assid | not bad.. seems to have good potential |
21:34.29 | kram | there's an ajamdemo.html that lets you demo it a little with transfer and hangup |
21:34.42 | Assid | could also use it for interfacing dundi if its on xml |
21:34.59 | kram | tainted-: okay so please do me the favor of telling me what issues you had with digium so i can try to improve for future customers |
21:35.09 | Assid | kram: using the manager ? |
21:35.33 | kram | tainted-: i'm not trying to convince you to buy digium, it's obvious you want to stick with sangoma but i'd like to at least try to improve upon our processes if there is anything we actually did wrong to push you that way |
21:35.35 | brookshire | assid: no, it is manager |
21:35.43 | brookshire | only not manager ;) |
21:36.03 | Assid | isnt the manager just a port open... and running commands through it |
21:36.25 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-182-210.37-151.net24.it) |
21:36.29 | brookshire | yeah.. but the ajam stuff doesn't use manager |
21:36.30 | Assid | i havent done much readup on the manager.. so maybe im wrong |
21:36.39 | Assid | right. prolly just an api does the same |
21:36.44 | brookshire | it is a direct xml interface to asterisk |
21:36.50 | Assid | nice |
21:37.09 | Assid | maybe i can work on it once im done with this project i have in hand |
21:37.09 | brookshire | it uses an internal webserver |
21:37.14 | Assid | on my free time |
21:37.17 | tainted- | kram i don't think u understand |
21:37.42 | tainted- | kram i recommended sangoma based on my past experiences with digium hardware and my current experiences with sangoma |
21:37.46 | Assid | i think im losing it .. wheres it on the site? |
21:38.03 | kram | tainted-: so what about your past experience with digium is the issue |
21:38.16 | brookshire | assid: is so fresh.. there is not much to go on |
21:38.29 | brookshire | assid: but you can find some docs/readme in trunk |
21:38.34 | Assid | aiee |
21:38.41 | tainted- | there's no issue |
21:38.47 | tainted- | lol |
21:38.56 | kram | tainted-: i mean seriously, in the absense of any other differentiation, i would hope that you would support digium since we support the project and even the channel you're chatting in here, recommending our competition |
21:39.00 | Assid | for some strange reason.. if i use svn.. on debian.. it screws up.. coz of libapr.. then my apache gets messed up if im using source |
21:39.10 | brookshire | assid: http://svn.digium.com/view/asterisk/trunk/doc/ajam.txt?view=markup |
21:39.15 | tainted- | i do support digium! |
21:39.18 | kram | so presumably there must be *some* reason that you would choose not to support us but instead to recommend our competitor |
21:39.28 | tainted- | i'm in here every day helping folks with dialplans etc |
21:39.38 | kram | and i'm trying to understand what that reason is, concretely, so that i can be sure that we can improve |
21:39.50 | kram | and yet you either cannot or choose not to express any such reason whatsoever |
21:39.59 | kram | and i cannot fathom any reason why |
21:40.10 | kram | are they your employer? |
21:40.10 | tainted- | hey listen |
21:40.25 | kram | give you free hardware? |
21:40.25 | tainted- | if u don't want me recommending competitors' products, just flat out say it |
21:40.37 | tainted- | i'll keep my mouth shut |
21:40.39 | kram | i'm not saying that, i'm saying i want to understand why |
21:40.54 | kram | after all, if that's truly how you feel, it's not my place to tell you to go against your feelings |
21:41.00 | Assid | brookshire: any chance thats running php ? |
21:41.02 | kram | your opinions expressed here are obviously and should be your own |
21:41.05 | brookshire | assid: zero :) |
21:41.06 | kram | i'm simply trying to understand the background |
21:41.10 | Assid | hehe |
21:41.11 | tainted- | i told u |
21:41.12 | kram | and you're not helping me with that much |
21:41.14 | brookshire | assid: it was written in c |
21:41.27 | Assid | my c knowledge sucks.. :( |
21:41.37 | brookshire | assid: yah, but it outputs xml |
21:41.39 | tainted- | audio issues, problems with irq, a few frustrating phone calls with support |
21:41.52 | brookshire | so.. think of a webservice :) |
21:41.52 | kram | you said your audio issues were MMX related |
21:41.53 | tainted- | and i haven't touched digium cards since then |
21:41.55 | kram | which is not specific to us |
21:42.09 | kram | and you haven't specified what the irq or phone call issues were |
21:42.15 | Assid | right.. so what part of it needs development? |
21:42.24 | kram | but i'm very eager to find out if you'll explain, again so that i can try to improve in the future |
21:42.42 | Assid | tainted-: help him help you! |
21:42.48 | tainted- | i just want the voip/zaptel platform to WORK so i can focus on writing value adding apps |
21:42.59 | Assid | tainted-: explain "WORK" |
21:43.05 | Assid | whats wrong EXACTLY |
21:43.16 | tainted- | right now, there's nothing wrong |
21:43.17 | Assid | whats the symptoms |
21:43.22 | kram | okay so help me do that by sharing your experience more concretely |
21:43.24 | tainted- | i'm just explaining my preferences |
21:43.37 | kram | well at least what was the phone issue? |
21:43.44 | *** join/#asterisk jql (n=jql@ip68-6-153-27.sd.sd.cox.net) |
21:43.46 | Assid | tainted-: how do you prefer 1 product over another, if there is no pros/cons |
21:43.50 | kram | and when you say nothing is wrong now do you mean with your digium h/w? |
21:44.12 | tainted- | i don't own anymore digium hardware |
21:44.12 | Assid | tainted-: do you like apples/oranges ? |
21:44.15 | Nivex | random side question: has anyone here had luck getting Zaptel to compile and function as perscribed under Xen? |
21:44.18 | kaz0358 | is there something special you have to do if you want to be able to dial *12345 if you are punching it in with dtmf? because as soon as you hit * followed by two numbers, you automatically get a busy/congested tone.. |
21:44.40 | Assid | kaz0358: transfer ? |
21:44.42 | kram | okay so you had issues with the digium h/w |
21:44.51 | kram | do you have your support ticket numbers from when you contacted tech support? |
21:45.07 | tainted- | probably somewhere |
21:45.22 | kram | did support resolve your issues? |
21:45.29 | kaz0358 | assid, no i was wanting to use sipbroker which allows you to dial another provider with a *XXX-enduser-num |
21:46.10 | ozverenm20 | tainted: what do you think about cologne chipsets ?? |
21:46.13 | tainted- | no, i recall scouring lists and trying different mobos |
21:47.07 | kram | tained: if you can send me your ticket numbers, i'd like to research them and see what happened |
21:47.33 | tainted- | ok |
21:48.22 | tainted- | BUY DIGIUM |
21:48.34 | kram | lol |
21:48.43 | tainted- | but switch to freeswitch |
21:48.44 | Damin | No.. vote for pedro! |
21:48.48 | tainted- | lol |
21:48.49 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
21:48.52 | kram | tainted: seriously i'm asking you to help me try to make our product better |
21:49.15 | tainted- | since i got into asterisk i have been |
21:49.15 | kram | tainted: i'm not asking you to switch, just to help me understand your experience |
21:49.22 | tainted- | i am not a hard core c developer |
21:49.43 | brookshire | k |
21:49.51 | kram | maybe tainted just likes trolling :) |
21:50.04 | russellb | kram: that's what i think |
21:50.14 | Strom_C | tainted-: it's very simple. tell kram, in a reasonable amount of detail, what happened and why you were unsatisfied |
21:50.14 | kram | oh well, if you're willing to, e-mail me your ticket numbers, markster@digium.com |
21:50.23 | kram | i'd be interested in seeing it |
21:50.29 | kram | i'm gonna get back to coding :) |
21:50.38 | kram | i think i remember now why i'm so rarely on irc lol |
21:51.10 | *** join/#asterisk swytch (n=ezcall@d83-179-214-255.cust.tele2.fr) |
21:51.23 | Assid | hahaa |
21:51.42 | tainted- | lol |
21:51.46 | tainted- | he really got worked up |
21:52.07 | swytch | You dont have a true random source in your computer? Then use the h323-disconnect-cause. |
21:52.14 | russellb | worked up when you were not able to provide any reasoning, yes. |
21:52.36 | russellb | which means you're just a troll |
21:52.43 | blitzrage | wow.... kram is on irc? |
21:52.46 | blitzrage | :) |
21:52.49 | file | blitzrage: scary eh? |
21:52.52 | blitzrage | quite |
21:53.08 | ariel_ | kram, your always welcome here.... |
21:53.10 | file | soooo IAXtel *should* be operational |
21:53.12 | Strom_C | well either that or sangoma's really putting some high-quality crack into their kool-aid these days |
21:53.53 | xachen | haha |
21:53.56 | xachen | you mean it finally is? |
21:54.00 | tainted- | russellb the issues with digium hardware are long past. i made a recommendation based on my current positive experiences with sangoma |
21:54.05 | file | xachen: it is. |
21:54.31 | tainted- | but i haven't seen/heard about the cpu overhead issues royk was talking about |
21:55.23 | tainted- | i can see why my public recommendation for a competitors product would peeve so many in this channel |
21:55.46 | ManxPower | tainted-, I'll prolly go with Sangoma when I do my personal telco stuff |
21:55.53 | tainted- | after all, most of the respondants work for digium |
21:56.02 | tainted- | but the personal attacks should stop |
21:56.13 | X-Rob | sangoma makes good hardware. |
21:56.16 | tainted- | it makes digium look like a bunch of tards |
21:56.18 | X-Rob | There's no issue with that |
21:56.33 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
21:56.37 | X-Rob | and, c'mon, the TDM4xx is a pile of puss. |
21:56.55 | tainted- | X-Rob u'd better explain yourself with support tickets |
21:56.56 | X-Rob | but I like the TE110, that's a nice little card. Cheap, cost effective, and does what you want. |
21:57.07 | ManxPower | tainted-, I suspect that it's less an issue of hardware design, and more of kernel driver design. |
21:57.33 | X-Rob | tainted-, heh |
21:58.20 | Strom_C | I think it's time for a bike ride |
21:59.06 | X-Rob | However, I am pissed at digium's tech support at the moment. |
21:59.12 | ManxPower | Things like HDLC Abort problems and random TDM400P FXS modules lockups are the main reason I'm not 100% pro Digium cards. |
21:59.41 | X-Rob | russellb, you reading this? |
21:59.45 | ozverenm20 | anyone know junghanns cards here ? |
21:59.54 | kram | manx: have you tried hardware hdlc? |
22:00.02 | kram | what sort of FXS lockups? |
22:00.18 | X-Rob | I've never heard of FXS lockups. |
22:00.25 | ManxPower | X-Rob, my issues with the TDM400P cards only seem to happen on production machines, and so are a miserable horror to diagnose. |
22:00.30 | kram | and do you have any support ticket numbers? |
22:00.31 | ManxPower | kram, no dialtone. |
22:00.32 | X-Rob | but, I don't use TDM00's 8) |
22:00.50 | kram | manx: what driver version? There was a known issue with this with older zaptels |
22:01.01 | tainted- | kram u'd better go code, u'll be fielding digium hardware problems all day |
22:01.03 | X-Rob | Hey, kram. Why are digium tech support sending people who buy tdm400's to #freepbx, purely because they're using A@H or freePBX? |
22:01.19 | ManxPower | kram, I could look them up. We fixed the problem by rebooting the server every monday morning. |
22:01.30 | X-Rob | there's no magic zaptel differences that we make. |
22:01.35 | kram | as i understand it from tech support A@H is very difficult to support outside of the GUI |
22:01.40 | kram | but i have no first hand experience |
22:01.42 | ManxPower | kram, what qualifies as "older zaptel" |
22:01.47 | tainted- | it's macro madness |
22:01.47 | kram | x-rob are you involved with a@h? |
22:01.47 | robin_sz | X-Rob, *@h != * |
22:02.00 | X-Rob | kram, no. They want help setting up their TDM400's. All they want is the FXS and FXO ports set up in zapata.conf |
22:02.03 | kram | i think most of the 1.0 series, i'd have to look for sure |
22:02.10 | X-Rob | kram, I'm a freePBX developer, which is the gui that A@H uses |
22:02.28 | X-Rob | freePBX does everything else for them, except for zapata.conf |
22:02.31 | X-Rob | and zaptel.conf |
22:02.44 | kram | x-rob: any reason you guys don't do that? |
22:02.48 | X-Rob | too hard? 8) |
22:02.51 | kram | x-rob: maybe we can setup a call or something |
22:02.58 | ManxPower | kram, we gave up before 1.2 was released and reboot the server every monday. We seldom have problems now. The odd FXS module just stops working, but I assume that's just fried. We are moving to SIP phones. |
22:03.02 | kram | if you're interested just send an e-mail |
22:03.05 | tainted- | X-Rob exactly |
22:03.20 | X-Rob | kram, to which email? |
22:03.26 | kram | markster@digium.com |
22:03.31 | X-Rob | coolo |
22:03.59 | kram | if you know the a@h people we might involve them too down the road |
22:04.03 | X-Rob | nah |
22:04.09 | kram | have they considered changing their name yet btw? |
22:04.10 | X-Rob | the a@h guy pretty much keeps to himself |
22:04.24 | kram | it's really causing us a lot more confusion than i would have originally thought |
22:04.41 | X-Rob | He doesn't get on IRC, and barely responds via email. |
22:04.46 | kram | *nods* |
22:04.49 | ariel_ | kram, asterisk@home uses freepbx but it's just an ISO which puts things together as a boot ISO for install. |
22:04.54 | kram | yah |
22:05.58 | ariel_ | name change for asterisk@home is something we all have email him about. But he never replys. |
22:06.19 | X-Rob | why do you want him to change the name? |
22:06.41 | ariel_ | @home |
22:07.00 | X-Rob | we've changed from AMP to freePBX because we're also gunna be supporting softswitch/openpbx etc if they ever get to a stage where they need a seperate dialplan maker. |
22:07.10 | ozverenm20 | kram: does digium offer me a solution to on card switch ? |
22:07.37 | tainted- | X-Rob sweet! |
22:07.38 | file | 'round the world... are you travelling |
22:08.14 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
22:08.28 | tainted- | X-Rob i think it's great that u cater to the 'i just want shit to work' market |
22:08.46 | tainted- | X-Rob it's much needed |
22:08.47 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
22:09.01 | X-Rob | tainted-, it's not just that - we've also got hooks everywhere for people who want to futz around with stuff |
22:09.13 | X-Rob | admittedly, people have to actually read the .conf files to see that the hooks are there |
22:09.42 | X-Rob | and yes, the extensions.conf _is_ insanely complex, but we've cut that down a hell of a lot since the AMP days |
22:10.00 | tainted- | well it's understandable |
22:10.48 | tainted- | i prefer to duct tape asterisk to agi apps to minimize dialplan logic |
22:10.51 | kaz0358 | do you know an easy way to make _*X. work while entering the number in over dial tone? :) |
22:11.04 | kram | oz: the TE410P/TE405P and families do support on card switching yes |
22:11.57 | X-Rob | tainted-, we had a big discussion about that a year ago, and we decided to do as much as we could in dialplan logic, pretty much 'coz it was there' 8) |
22:11.59 | tainted- | pulls biz logic out of voip box so i can migrate to freeswitch or whatever whenever they get a pbx module together |
22:12.16 | kaz0358 | x-rob, it seems that asterisk is looking for *67 and such and prevents you from dialing something like *373612 |
22:12.44 | X-Rob | kaz0358, I don't know why you're telling me that. |
22:12.53 | tainted- | i've heard the argument that dialplan logic is X times faster than agi |
22:12.54 | X-Rob | I know that the zaptel channels have some built-in feature codes. |
22:13.00 | Assid | anyone know a decent place for cheap DID's |
22:13.08 | Assid | like 2-3 bucks |
22:13.09 | Assid | tops |
22:13.09 | kaz0358 | x-rob, you mentioned that extensions.conf is much easier than it use to be. heheh.. i still haven't figured out how to make that work yet |
22:13.47 | X-Rob | kaz0358, the extensions.conf that's supplied with freePBX is less complex than the extensions.conf supplied with AMP |
22:14.02 | tainted- | but to me, hardware is cheap, and time to market is much more important |
22:14.27 | tainted- | X-Rob is srvlookup = yes by default in freepbx? |
22:14.50 | X-Rob | nope |
22:15.13 | X-Rob | purely because I've got zero experience with it |
22:15.40 | kaz0358 | AMP? freePBX? i just compiled and downloaded the source.. i am happy with the way i have my extensions.conf setup. i'd just like to be able to dial *373612 for instance to use sip broker as the lookup for the uri of the provider |
22:15.41 | tainted- | some providers don't have SRV records, so it's best to set = no |
22:16.03 | tainted- | i've had to support a few freepbx users who couldn't get up and running b/c of that |
22:16.55 | X-Rob | tainted-, we're using trac as a bugtracker now - if you think it's an issue, raise a ticket. But when I looked at it last time it caused more issues than it solved. |
22:17.04 | X-Rob | www.freepbx.org/newticket I think |
22:17.06 | *** join/#asterisk avilv (i=justacas@Real.IRC.Masters.Use.Bitch-X.us) |
22:17.09 | tainted- | k |
22:17.33 | *** join/#asterisk anthm (n=anthm@000-450-899.area4.spcsdns.net) |
22:17.33 | *** mode/#asterisk [+o anthm] by ChanServ |
22:17.36 | avilv | whats up everybody? |
22:18.55 | *** join/#asterisk rigid (n=The@port-212-202-73-9.dynamic.qsc.de) |
22:19.00 | rigid | re |
22:19.24 | avilv | its.... quiet :( |
22:20.22 | rigid | i have a asterisk configured to forward 2 sip-accounts to the 2 lines of a sip/analog box... i see calls to <mynumber> (which work) and i see calls to <mynumber>0 (that don't work) when using "sip debug" |
22:20.44 | *** join/#asterisk ozverenm21 (n=ozverenm@162.27.103-84.rev.gaoland.net) |
22:20.44 | rigid | the destination number is then <mynumber>0 ... how can i configure this number in asterisk? |
22:21.08 | ManxPower | rigid, just like you set it up in extensions.conf |
22:21.13 | rigid | to be exact, i want to MP3Player() a file when it's called... |
22:21.30 | ManxPower | rigid, are you using AMP/FreePBX/Asterisk@Home |
22:21.57 | rigid | ManxPower, no... i'm using a compiled packet (gentoo ebuild) |
22:22.24 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
22:22.44 | rigid | just adding: "exten => 123450,0,MP3Player(/var/lib/asterisk/mohmp3/song.mp3)" doesn't work |
22:22.54 | rigid | ManxPower, (if my number is 12345) |
22:23.05 | ManxPower | rigid, priorities start at 1 not at 0 |
22:23.05 | avilv | Could somebody tell me why my 2 month old A104 works perfectly but the Digium Wildcard TE411P doesn't? |
22:23.12 | rigid | while calls to 12345 work (using Ringing) |
22:23.14 | avilv | which I just got earlier today |
22:23.16 | rigid | ManxPower, ahh |
22:23.30 | b4ka | sangoma > * |
22:23.33 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
22:23.42 | avilv | asterisk just segfaults when I try to make a zap/ call using the wildcard |
22:23.52 | avilv | and i recompiled the whole thing |
22:23.58 | tainted- | i heard there are asterisk advisory council trading cards |
22:24.09 | kram | avilv: have you placed a trouble ticket with tech support? |
22:24.10 | tainted- | is that true? |
22:24.23 | avilv | tainted-: this wildcard I picked up must jsut be one |
22:24.24 | kram | i haven't heard any such thing |
22:24.25 | avilv | completely useless |
22:24.43 | file | you get technical support with the card, so use it... |
22:24.51 | rigid | ManxPower, doesn't work either :( |
22:24.51 | ManxPower | avilv, what specific model of Digium card do you have? |
22:24.53 | kram | what was completely useless? |
22:24.57 | kram | do you have a ticket number? |
22:24.58 | rigid | ManxPower, with priority 1 |
22:25.12 | ManxPower | rigid, did you do a reload after changing it? |
22:25.20 | rigid | ManxPower, ;) ofcoz |
22:25.45 | rigid | ManxPower, ok... s.o. that starts priorities with 0 could also forget to reload i admit :) |
22:26.29 | avilv | i got the card on ebay |
22:26.48 | ManxPower | avilv, Then I strongly doubt that is an actual Digium card. |
22:27.26 | avilv | http://cgi.ebay.ca/Digium-Wildcard-TE411P-Quad-T1-ASterisk-VoIP-PCI-Card_W0QQitemZ9718822238QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
22:27.26 | kram | avilv: so what kind of card is it? |
22:27.28 | Strom_C | either that or the card got destroyed by an inept user who tried to pawn it off on ebay |
22:27.41 | kram | avilv: okay so did you call or e-mail tech support yet about your problem? |
22:27.57 | ManxPower | Strom_C, I saw someone recently that got a "Genuine X100P Clone", complete with hologram sticker. |
22:28.09 | Strom_C | hahahahahahahahahahahahahahahaha |
22:28.11 | Strom_C | oh god |
22:28.13 | Strom_C | that tickles |
22:28.24 | rigid | ManxPower, the incoming call is to uri="sip:123450@asterisk-server" ... don't i have to add a new context or something? |
22:28.54 | avilv | kram: no. but I just got it to work bridging to an IAX call and it drifts more more when I dial dtmf :( |
22:29.06 | Strom_C | avilv: what version of zaptel and asterisk are you running, out of curiosity? |
22:29.20 | avilv | the latest |
22:29.23 | budmang | is FREEpbx nice? |
22:29.38 | Strom_C | avilv: by "the latest" do you mean "stable" or do you mean "svn trunk"? |
22:29.39 | avilv | 1.2.7 for asterisk |
22:29.39 | mds2 | anybody have regular stability issues with WCTDM24XXP cards using 2.4.32? |
22:29.43 | avilv | 1.2.5 for zaptel |
22:29.52 | kram | avilv: if you didn't contact tech support then how are we supposed to help you? |
22:30.11 | Splas | anyone know if it's at all possible to play the voicemail messages asterisk can attach to the notification emails on a blackberry.. specially an 8700 |
22:30.37 | tainted- | where's your fucking support ticket asshole |
22:30.48 | file | tainted-: okay that was just rude |
22:30.56 | Strom_C | tainted-: chill, for fuck's sake |
22:31.04 | avilv | it should just work. when its in dual opteron with a nice supermicro mobo and the latest fedora with the latest asterisk and zaptel |
22:31.06 | avilv | it should just work! |
22:31.23 | file | avilv: but it's not, so you contact support so they can troubleshoot and try to solve the problem |
22:31.32 | Strom_C | avilv: was the card used or brand new? |
22:31.36 | ManxPower | avilv, Why do you expect a non-name fake card to work with Asterisk? |
22:31.38 | avilv | brand new |
22:31.38 | Strom_C | woohoo, I'm a training card |
22:31.42 | Strom_C | er |
22:31.43 | Strom_C | trading card |
22:31.47 | avilv | I got it off voip supply on ebay |
22:31.52 | avilv | so i hope its not a dud |
22:31.55 | Strom_C | yes, I'm thinking, really I am |
22:32.36 | kram | it should be good |
22:32.40 | ManxPower | avilv, Digium has not sold X100P/X101P cards in at least 2 years. |
22:32.42 | kram | you can contact either of us for support |
22:32.48 | kram | this is a TE411P |
22:33.04 | ManxPower | Ah! nevermind |
22:33.07 | avilv | this is fucking bullshit. when I want to benchmark and move to all digium cards and I don't even have a good testing platform (because asterisk segfaults or gives bad quality) then what am I supposed to do? |
22:33.11 | kram | :) |
22:33.28 | kram | you're supposed to contact tech support so they can figure out what's wrong with your asterisk installation |
22:33.34 | kram | support@digium.com |
22:33.43 | kram | if you haven't had a happy resolution in 3 days you can drop me an e-mail |
22:33.47 | ManxPower | avilv, You have 2 choices. 1) return the card, or 2) contact tech support. |
22:33.48 | kram | include your support ticket number |
22:33.54 | ManxPower | I recommend option 2 |
22:34.05 | Strom_C | avilv: calm down, man. there may be a minor problem or the card may have gotten damaged in shipment or something. bitching about it won't help, but talking to support will. |
22:34.07 | kram | i'd like to point out that option 1 is the dumb one since changing hardware won't change an asterisk segfault |
22:34.30 | tainted- | can i start trolling about asterisk yet? |
22:34.31 | avilv | ok i will do that |
22:34.35 | russellb | tainted-: no. |
22:34.42 | ManxPower | kram, option 1 is dumb, but this guy is screaming about problems without evening having contacted support..... |
22:34.46 | tainted- | i have some beef with scaling |
22:34.46 | avilv | kram: are you mark? |
22:34.51 | budmang | I am looking for an easy install of asterisk with a web interface. Any suggestions? |
22:35.04 | ManxPower | budmang, try #freePBX |
22:35.06 | Strom_C | budmang: switchvox ;) |
22:35.17 | avilv | while i'm hear i'm just curious |
22:35.27 | russellb | avilv: yes |
22:35.35 | avilv | why does the console say avoiding deadlok on every iax call i make? |
22:36.01 | rigid | ManxPower, can i somehow debug how asterisk decides when walking through extensions.conf? |
22:36.03 | ManxPower | avilv, your name isn't RoyK is it? |
22:36.09 | tainted- | b/c ur supporse to use SIP for calls dude |
22:36.16 | tainted- | IAX is just for NAT traversal |
22:36.23 | avilv | no my name is jason |
22:36.30 | ManxPower | rigid, ask me on a week day. |
22:36.38 | rigid | ManxPower, :) |
22:36.53 | ManxPower | rigid, I'm too tired to do complicated dialplan debugging right now. |
22:36.57 | blitzrage | ManxPower: yo |
22:37.24 | ManxPower | hiya, royk |
22:37.33 | ManxPower | blitzrage, 'sup? |
22:37.48 | rigid | ManxPower, hmm... how to make asterisk extension-stuff more verbose, you don't know by heart? |
22:37.55 | rigid | ManxPower, i can debug it then myself |
22:37.56 | avilv | no answer?! THIS IS FUCKING BULLSHIT! I'M BUYING ALL SANGOMA CARDS ON MONDAY AND AM WAITING TIL THAT FREESWITCH COMES OUT |
22:38.07 | ManxPower | avilv, go for it |
22:38.10 | russellb | avilv: what are you talking about? |
22:38.18 | ManxPower | rigid, asterisk -rvvv |
22:38.29 | blitzrage | ManxPower: not to much, just listening to tune |
22:38.31 | blitzrage | tunes* |
22:38.52 | tainted- | damn |
22:38.54 | Strom_C | note to the short-tempered: you can get wall padding professionally installed now. |
22:39.00 | rigid | ManxPower, tnx |
22:39.53 | Assid | anyone here using sipbroker? |
22:40.16 | RoyK | ManxPower: wrong guy :) |
22:40.21 | ozverenm21 | does it plained to asterisk to work as stateless sip proxy ? |
22:41.02 | RoyK | ozverenm21: asterisk != proxy |
22:41.19 | RoyK | ozverenm21: what you're looking for is something like SER |
22:41.31 | ManxPower | Maybe avilv is The Person Formerly Known as Timecop.... |
22:42.00 | tainted- | ozverenm21 u need to run SER/asterisk |
22:42.03 | ManxPower | I'm REALLY REALLY REALLY starting to hate working with DVD video. |
22:42.10 | tainted- | ser handle SIP headers, asterisk handle media |
22:42.48 | ozverenm21 | maybe but asterisk b2bua is not natural and break somes CDR functionnalities because callid change |
22:43.01 | xachen | :O |
22:43.06 | xachen | that avilv has a short temper |
22:43.48 | X-Rob | ok, that was funny. |
22:44.04 | tainted- | ozverenm21 if u need to do billing, ur in for a ride |
22:44.57 | tainted- | ozverenm21 how soon are u going to production? |
22:45.31 | tainted- | ozverenm21 or are u already on asterisk life support |
22:45.44 | ozverenm21 | i am using asterisk in production, just for doing testing on voip and isdn ... |
22:47.03 | ozverenm21 | do you plain to add alcatel UA protocol support in a* ? |
22:47.35 | tainted- | ozverenm21 try #asterisk-dev |
22:48.05 | russellb | any UA should work. |
22:48.07 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
22:48.08 | russellb | that's not a dev question. |
22:48.08 | surfdue | hi |
22:48.14 | surfdue | I cant connect to asterisk with my linksys pap2? |
22:48.17 | surfdue | can anyone help |
22:48.29 | Strom_C | surfdue: are they both on the same network segment? |
22:48.36 | surfdue | what do you mean? |
22:48.39 | xachen | surfdue: #freepbx |
22:48.44 | surfdue | huh |
22:48.48 | xachen | your using FreePBX? |
22:48.54 | surfdue | maybe no. |
22:48.59 | RoyK | tainted-: asterisk cannot handle media itself, it needs to do both |
22:49.05 | anthm | sure blame the segment it's segment's fault! |
22:49.21 | Strom_C | surfdue: are the asterisk box and the PAP2 on the same subnet? |
22:50.04 | surfdue | no |
22:50.12 | surfdue | storm_c its not local |
22:50.22 | Strom_C | surfdue: is the asterisk box behind NAT? |
22:50.25 | *** join/#asterisk talljon84 (n=jonathan@66-188-104-144.dhcp.mdsn.wi.charter.com) |
22:50.34 | Strom_C | and is the PAP2 also behind NAT? |
22:50.36 | surfdue | Strom_C, its on a dedicated server |
22:50.43 | xachen | so the server wouldn't b |
22:50.45 | xachen | I hope |
22:50.52 | RoyK | a dedicated server may well be behind nat, surfdue |
22:50.52 | surfdue | Strom_C, the pap2 may be i did setup port forwarding though |
22:50.52 | Strom_C | surfdue: dedicated or not, is there NAT involved? |
22:51.07 | talljon84 | Is anyone aware of a SIP client for Palm OS? If not, any suggestions where I could post a bounty for one? |
22:51.08 | surfdue | Strom_C, i dont think so |
22:51.19 | xachen | rather |
22:51.20 | Strom_C | surfdue: is it behind a firewall? |
22:51.24 | RoyK | talljon84: there are a few for wince |
22:51.24 | xachen | is your PAP2 behind a firewall |
22:51.37 | RoyK | firewalls are for pussies :) |
22:51.37 | talljon84 | RoyK: Yes, but I have a Treo 650 so that won't work. |
22:51.42 | *** join/#asterisk wenko (n=wenko@142.232.8.200) |
22:51.48 | wenko | hey there |
22:51.50 | surfdue | my pap2 is but the ports unblocked |
22:51.51 | surfdue | AND |
22:51.55 | wenko | anyone here good with iaxi? |
22:51.55 | surfdue | my other line works |
22:51.56 | surfdue | :| |
22:52.23 | xachen | RoyK: no, the only firewalls to use are pf and ipf |
22:52.25 | surfdue | its from another provider though |
22:52.25 | Strom_C | which "other line"? |
22:52.26 | surfdue | it has 2 lines |
22:52.26 | Strom_C | oh, on the PAP2 itself |
22:52.26 | RoyK | talljon84: http://www.google.com/search?client=opera&rls=en&q=sip+palm&sourceid=opera&ie=utf-8&oe=utf-8 |
22:52.29 | wenko | i need to do a password recouvery on my iaxi, is that possible?? |
22:52.49 | surfdue | Strom_C, is there a test i can run to see if its behind a NAT the server? |
22:52.53 | RoyK | wenko: google for 'hammer' |
22:52.59 | wenko | hammer? |
22:53.10 | Strom_C | surfdue: um, log in and then run ifconfig? :) |
22:53.25 | RoyK | wenko: a good tool by Steel or Iron |
22:53.33 | talljon84 | RoyK: That was my first idea. The first heading doesni't really exist (hoax it appears). The rest are all talking about the lack-of-a-client. |
22:53.38 | surfdue | what am i looking for Strom_C ? |
22:53.42 | wenko | ...that means im fucked right?? |
22:54.04 | RoyK | wenko: sorry. don't know. contact digium :) |
22:54.13 | Strom_C | surfdue: whether or not the IP address of the ethernet interface is the same one that faces the public network |
22:54.16 | wenko | :S |
22:54.25 | wenko | i get an error when trying to telnet |
22:54.25 | Nugget | telnet is eeeeeeevil. |
22:54.38 | wenko | it tells me that there is no password set |
22:55.02 | Dr-Linux | surfdue: what does say >> ifconfig |
22:55.07 | surfdue | Strom_C, it does |
22:55.10 | surfdue | it is |
22:55.27 | Strom_C | surfdue: is the PAP2 failing to register? |
22:55.29 | xachen | surfdue: Is this server the one I was on earlier? |
22:55.39 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
22:55.41 | surfdue | xachen, yes |
22:55.44 | xachen | oh |
22:55.49 | xachen | there is no NAT on that machine then |
22:55.55 | xachen | Layeredtech doesn't NAT servers |
22:55.58 | talljon84 | Is anyone aware of a deb package that will provide the libstdc++.so.2.8 lib that's required for some games? |
22:56.03 | RoyK | wenko: http://www.google.com/search?hs=dze&hl=en&lr=&ie=UTF-8&oe=UTF-8&client=opera&rls=en&q=iaxy+reset+password&btnG=Search |
22:56.08 | talljon84 | err. damnit. wrong room, sorry |
22:56.18 | Strom_C | surfdue: is the PAP2 failing to register? |
22:56.23 | surfdue | yes |
22:56.31 | Strom_C | is asterisk spitting out an error? |
22:56.32 | surfdue | Registration State:Can't connect to login server |
22:56.41 | surfdue | Strom_C, whereis hte log file again? |
22:56.53 | Strom_C | just connect to the console and see if there's an error there |
22:57.34 | Dr-Linux | surfdue: you are using A@home? |
22:57.42 | surfdue | whats the link |
22:57.44 | surfdue | no |
22:57.48 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
22:58.02 | Dr-Linux | surfdue: type asterisk -r |
22:58.13 | surfdue | oh |
22:58.13 | Dr-Linux | if asterisk is already running |
22:58.14 | surfdue | lol |
22:58.17 | Strom_C | surfdue: what link? log into the box and connect to the asterisk console after you're whatever user asterisk is running as |
22:58.17 | surfdue | it is |
22:58.55 | Dr-Linux | surfdue: the try debug |
22:59.04 | surfdue | im guessing you mean type debug? |
22:59.06 | Dr-Linux | then* |
22:59.07 | Strom_C | not yet |
22:59.10 | Strom_C | set verbosity to 10 |
22:59.13 | surfdue | k |
22:59.19 | Strom_C | see whether asterisk is kicking out an error |
22:59.30 | surfdue | whats the command |
22:59.30 | surfdue | lol |
22:59.35 | Strom_C | set verbose 10 |
22:59.51 | surfdue | k |
23:02.18 | Strom_C | anything yet? |
23:02.28 | surfdue | no ? |
23:02.42 | Strom_C | try rebooting the PAP2 and then see what happens |
23:04.03 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
23:04.04 | surfdue | k |
23:05.21 | orlok | Hey, has anybody here ever used the Sail asterisk manager? |
23:05.56 | surfdue | link me orlok |
23:06.01 | surfdue | Strom_C, no luck nothing errors? |
23:06.30 | Strom_C | surfdue: well, OK, then it sounds like it's a PAP2 issue |
23:06.41 | surfdue | Strom_C, heres whats odd |
23:07.00 | surfdue | i use asterlink and I have it connected directly to asterlink on line 1 |
23:07.03 | orlok | surfdue: k, it runs on top of sme server |
23:07.08 | surfdue | and on line 2 asterisk which is linked to asterlink |
23:07.13 | orlok | which is based on centos |
23:07.59 | Strom_C | surfdue: do a sanity check and make sure you have all the settings correct on the PAP2 |
23:08.03 | orlok | surfdue: http://contribs.org/modules/phpwiki/index.php/SME7Contribs has the link to the addon fo smeserver |
23:08.21 | orlok | it seems to work, just that i get errors trying to dial out |
23:08.28 | surfdue | Strom_C, sanity check hmm ? |
23:08.41 | Strom_C | yes |
23:08.45 | Strom_C | sanity checks are good things |
23:08.55 | wenko | i dont understad how to provision an iaxi, anyone able to help me out? |
23:08.57 | file | insanity check. |
23:09.01 | xachen | haha |
23:09.11 | Strom_C | i like those too ;) |
23:09.11 | xachen | 100% in my case |
23:09.42 | surfdue | Strom_C, is there a port check i can do to make sure i can talk to my pap2 through my firewall? |
23:10.18 | Strom_C | surfdue: you just said that you're able to talk SIP through your firewall direct to asterlink, so that rules out firewall issues in my book |
23:10.31 | surfdue | ya |
23:10.32 | surfdue | just making sure |
23:10.33 | Strom_C | surfdue: like I said, do a sanity check and make sure your settings are correct |
23:10.53 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
23:10.57 | file | did you do a sip debug on your personal server and then restart the PAP2? |
23:13.00 | surfdue | Strom_C, http://img244.imageshack.us/my.php?image=untitled1nm.jpg |
23:13.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net) |
23:13.15 | surfdue | file i dont know how to do sipdebug |
23:13.37 | file | you type sip debug |
23:13.40 | Strom_C | surfdue: that doesn't help me at all. I don't know the name of your server |
23:13.43 | file | and then watch SIP traffic appear |
23:14.31 | Strom_C | surfdue: for your asterlink config do you have a domain name or an IP address in the "SIP Proxy" field? |
23:14.35 | xachen | You know your a nerd when you watch SIP traffic flow through :P |
23:14.54 | file | hehe |
23:14.56 | Strom_C | oh baby oh baby oh baby SIP me harder |
23:14.57 | surfdue | strom i have the name |
23:15.14 | Strom_C | surfdue: your asterisk box is just "host41.com"? |
23:15.33 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
23:15.44 | surfdue | well its an ip |
23:15.47 | surfdue | ya |
23:15.48 | surfdue | lol.. |
23:15.54 | surfdue | host41 directs to the ip |
23:16.01 | Strom_C | and host41.com resolves to the correct IP? |
23:16.20 | surfdue | yep |
23:17.35 | Strom_C | 72.232.27.132 right? |
23:17.39 | surfdue | look at sip debug http://pastebin.com/691309 |
23:17.45 | surfdue | s aparently it works ya Strom_C :) |
23:17.57 | surfdue | the sip debug looks fine but not logged in still? |
23:18.27 | Strom_C | SIP/2.0 404 Not found |
23:18.30 | Strom_C | thats no good |
23:18.31 | surfdue | whats that mean :P |
23:18.43 | file | set nat=yes |
23:18.51 | surfdue | in the pap2? |
23:18.56 | Strom_C | show me the relevant section of sip.conf |
23:18.57 | file | no, sip.conf |
23:19.08 | surfdue | k |
23:19.17 | file | and do whatever Strom says |
23:19.27 | Strom_C | do the hokey pokey |
23:20.02 | surfdue | nat isnt in sip.conf ? |
23:20.28 | file | put it in general, nat=yes and do a sip reload |
23:20.38 | surfdue | k |
23:20.42 | surfdue | at the top? |
23:20.43 | surfdue | im guessing |
23:20.44 | file | and get the PAP2 to register again... and see what happens |
23:20.51 | file | in the general section, somewhere |
23:20.51 | surfdue | nvm |
23:20.52 | surfdue | i see it |
23:21.25 | Strom_C | hey, speaking of which: is there an IAX provider that offers local DIDs in the los angeles area with unlimited inbound and no restrictions on concurrent calls? |
23:21.35 | surfdue | reregistering |
23:21.52 | *** join/#asterisk msg43 (n=msg43@adsl-68-255-187-222.dsl.bcvloh.ameritech.net) |
23:22.03 | msg43 | surfdue, did you realize what you did |
23:22.17 | msg43 | surfdue, anyone who is from my area that wants to join the irc channel can't |
23:22.35 | surfdue | msg43, please dont spam this room about other rooms matters. |
23:22.37 | msg43 | surfdue, great job on getting rid of costomers |
23:22.54 | msg43 | Nah I'm just point out what I moran you are :) |
23:23.09 | Strom_C | msg43: take the drama elsewhere, please. |
23:23.13 | surfdue | . |
23:23.15 | *** join/#asterisk kietlak (n=kietlak@apn-99-84.gprspla.plusgsm.pl) |
23:23.38 | msg43 | Strom_C, I just like pissing of surfdud |
23:23.57 | surfdue | please stop this msg.. |
23:24.00 | surfdue | thats why your banned. |
23:24.24 | surfdue | Strom_C, sorry anyways I am still getting cant register its odd |
23:24.27 | msg43 | surfdud, we can take to a personal level in pm |
23:24.33 | surfdue | Strom_C, i dont see sip404 anymore though? |
23:24.40 | surfdue | msg43, dont pm me. |
23:24.49 | Strom_C | surfdue: show me your sip.conf |
23:25.02 | msg43 | surfdue, woops already did |
23:25.05 | russellb | surfdue: /ignore works wonders |
23:25.06 | msg43 | surfdue, what you gonna do |
23:25.16 | russellb | msg43: please take this elsewhere ... |
23:25.16 | surfdue | arg, who has ops in here? |
23:25.27 | surfdue | Strom_C, It has passwords lol let me take them out |
23:26.32 | russellb | gutes? |
23:26.38 | Strom_C | what the hell are gutes? |
23:26.51 | surfdue | Strom_C, http://pastebin.com/691325 |
23:26.55 | russellb | Strom_C: i hate your gutes so much! |
23:27.10 | Strom_C | russellb: i hate your gutes even more! |
23:27.27 | surfdue | lol.. |
23:27.33 | ihateyourgutes | surfdue, your so cute |
23:27.38 | ihateyourgutes | banning everyone :) |
23:27.39 | ihateyourgutes | ahh |
23:27.45 | Strom_C | surfdue: so, uh, where is the entry for your PAP2? |
23:28.06 | surfdue | Strom_C, is tehre suppose to be ? :P |
23:28.10 | Strom_C | duh |
23:28.25 | surfdue | what do i need to add |
23:29.24 | Strom_C | one sec |
23:30.38 | Strom_C | http://pastebin.com/691335 is a sample entry for one of my PAP2 SIP lines |
23:32.15 | Strom_C | welcome back, kram |
23:32.19 | kram | thank ye |
23:35.01 | Strom_C | i really need to bring an ethernet connection here into my bedroom so I can get my bedside phone as an extension off my pbx... |
23:35.19 | orlok | hmm.. |
23:35.23 | orlok | i'm having dialplan issues |
23:35.55 | orlok | i think - is there any way to debug why a call doesnt go through? |
23:36.03 | *** join/#asterisk esculapio_ (n=ESCulapi@145stb68.codetel.net.do) |
23:36.06 | Strom_C | orlok: what is your specific problem? |
23:36.35 | orlok | Strom_C: asterisk server is also the net gateway. phones behind asterisk ca dial each other, but get a fast busy signal trying to dial outbound |
23:36.49 | Strom_C | orlok: paste extensions.conf please |
23:36.51 | Strom_C | er |
23:36.53 | Strom_C | pastebin |
23:37.15 | Strom_C | and btw, that fast busy signal is called a reorder |
23:38.39 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
23:38.46 | orlok | http://pastebin.ca/52450 |
23:39.24 | Strom_C | well theres your problem |
23:39.40 | Strom_C | you cant wildcard match if you dont put an underscore first |
23:40.00 | ms34 | surfdue = http://z.about.com/d/politicalhumor/1/0/n/U/moran.jpg |
23:40.01 | ms34 | surfdue = http://z.about.com/d/politicalhumor/1/0/n/U/moran.jpg |
23:40.22 | Strom_C | orlok: exten => XXXXXXXXX,1,agi(selintra,OutRoute,Outgoing) is the problem line |
23:40.34 | surfdue | :| |
23:40.41 | surfdue | stop spamming channels! |
23:40.50 | *** mode/#asterisk [+b %ms34!*@*] by russellb |
23:41.28 | orlok | heh |
23:41.30 | orlok | #teenlinux |
23:41.48 | *** kick/#asterisk [ms34!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (bye!) |
23:41.57 | orlok | Strom_C: XXXXXXXXX is specifiable, what should it be? |
23:42.08 | Strom_C | what do you mean "specifiable" |
23:42.13 | Strom_C | you're doing wildcard matching, right? |
23:42.48 | orlok | Strom_C: I am setting it up using sail, which is an asterisk config manager using the e-smith templating system |
23:43.01 | Strom_C | to me, that sounds like "blah blah blah blah" |
23:43.03 | orlok | yes, i am |
23:43.23 | orlok | Strom_C: Well, what specifically about that line is the problem? |
23:43.26 | Strom_C | orlok: therefore, if you are doing wildcard matching, you must prefix the wildcard characters with an underscore |
23:43.30 | orlok | ahh |
23:43.34 | orlok | cool |
23:43.39 | Strom_C | which is what I already said |
23:43.48 | orlok | i always saw the underscore, but the docs never seemed to mention it, only XXX and whitespace |
23:43.57 | orlok | ahh, yeah |
23:44.04 | orlok | up before the flood from ms34 |
23:44.08 | Strom_C | yes |
23:44.55 | orlok | ok, now its _XXXXXXXXX, same issue |
23:45.01 | Strom_C | you did a reload, right? |
23:45.34 | orlok | yup |
23:45.46 | Strom_C | what is the console saying? |
23:46.36 | orlok | nothing when i try to dial |
23:46.42 | Strom_C | set verbose 10 |
23:46.44 | Strom_C | dial again |
23:46.55 | *** join/#asterisk msg43 (n=msg43@adsl-68-255-187-222.dsl.bcvloh.ameritech.net) |
23:47.12 | ms89 | rule 1, never ban a nickname |
23:47.16 | ms89 | :) |
23:47.19 | *** mode/#asterisk [+b *!*n=msg43@*.dsl.bcvloh.ameritech.net] by russellb |
23:47.19 | *** kick/#asterisk [ms89!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
23:47.41 | surfdue | russellb, atleast there is an op in here.. |
23:48.41 | russellb | heh, you're welcome... |
23:48.47 | russellb | i'm trying to let you guys talk in peace |
23:52.23 | orlok | Hmm |
23:52.38 | orlok | should the sip povider asterisk is registering with show up in sip show peers? |
23:52.53 | Strom_C | orlok: what. does. the. console. say? |
23:53.54 | orlok | nothing! |
23:54.04 | Strom_C | is your AGI even working? |
23:54.25 | Strom_C | (speaking of which, why in god's name are you using an AGI to dial when Dial() will work much more cleanly?) |
23:55.04 | *** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net) |
23:55.26 | orlok | Strom_C: Only time i've ever gotten asterisk working was using Asterisk@Home |
23:55.32 | Strom_C | ugh |
23:55.35 | Strom_C | I hate guis |
23:55.36 | orlok | <PROTECTED> |
23:55.40 | orlok | yeah, so do i |
23:55.48 | Strom_C | !thebook |
23:55.49 | Strom_C | er |
23:55.52 | Strom_C | ~thebook |
23:55.53 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
23:55.59 | *** join/#asterisk Isaiah (n=Isaiah@208-187-93-4.br1.hnv.mi.frontiernet.net) |
23:56.04 | orlok | got it printed out next to m |
23:56.27 | Strom_C | orlok: you should try to set up a system without any guis or prototyping tools or anything |
23:56.32 | Strom_C | just you, asterisk, and vim |
23:56.42 | orlok | yeah |
23:56.57 | orlok | cos i dont know what the gui/framework system i'm using now is or isnt doing right |
23:57.07 | orlok | and its a layer of abstraction that pisses off the people that do know |