| 00:02.44 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 00:03.24 | znoG | to get the first digit of $EXTEN, is it just ${EXTEN:0:1} ? |
| 00:04.06 | *** join/#asterisk NewSole (n=dave@d226-107-112.home.cgocable.net) |
| 00:04.25 | NewSole | Hello Spooks... |
| 00:05.46 | crshman | I am trying to set up a connection to broadvoice but i get this in the logs: "Allocating new SIP dialog for xxxxxxxxxxxxxx@127.0.0.1 - REGISTER (No RTP)" |
| 00:06.06 | crshman | the thing that is of particular interest to me is the ip address and that (No RTP) at the end....what's that about? any ideas? |
| 00:07.42 | *** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
| 00:09.26 | PMantis | crshman, register => 5852198656@sip.broadvoice.com:dsfshfkshffhuhdsfsdf:5852198656@sip.broadvoice.com |
| 00:10.01 | crshman | do i need the ">" i have the right line just no ">" |
| 00:10.03 | PMantis | I stopped using BV this past month, so you can't do anything with the above info. :) |
| 00:10.25 | PMantis | Yes, you nee\d it. |
| 00:10.29 | PMantis | need |
| 00:10.33 | PMantis | sorry, old KB |
| 00:10.50 | matthewsimpson | pmantis: it's working for me... i just called china and left the phone off the hook |
| 00:10.56 | matthewsimpson | :-o |
| 00:11.06 | PMantis | heh |
| 00:11.16 | PMantis | I change the PW anyhow, just to be safe. :) |
| 00:11.20 | matthewsimpson | too late |
| 00:11.22 | matthewsimpson | i already changed |
| 00:11.23 | matthewsimpson | buahahahaha |
| 00:11.33 | PMantis | No, I mean before I pasted. :) |
| 00:11.40 | matthewsimpson | oh, darn :( |
| 00:12.07 | PMantis | Here's my peer config, too |
| 00:12.08 | PMantis | http://pastebin.com/764112 |
| 00:12.09 | generalhan | hey guys ... when you do a # transfer what context does the transfer look to? i need to run it through a macro before the call is trnasfered and i cant get it to work |
| 00:12.47 | *** part/#asterisk matthewsimpson (i=matthews@67.58.10.44) |
| 00:12.50 | crshman | is it supposed to be @127.0.0.1? or my ip address? |
| 00:12.52 | PMantis | ok, now for my question. :) I have an og_fax contect for my fax machine. |
| 00:13.16 | *** join/#asterisk IeatPaste (i=matthews@67.58.10.44) |
| 00:13.27 | PMantis | I use an exten => _NXXXXXX(dial...) line, plus include other contects, that have my own numbers. |
| 00:13.57 | PMantis | if I dial my own 7 digit number, it still is caught by the above exten, no matter where I place the include line. |
| 00:15.29 | *** join/#asterisk ids2500 (i=matthews@67.58.10.44) |
| 00:15.45 | *** join/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 00:15.53 | *** part/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 00:15.58 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 00:16.14 | PMantis | Ok, here's my relevent exten config: http://pastebin.com/764118 |
| 00:16.47 | PMantis | my in-voicepulse context includes an eten for my 7 digit number. |
| 00:17.23 | PMantis | dialing this 7 digit number always sends the call out, which then returns to my server anyhow... silly! |
| 00:17.51 | PMantis | Unless I comment the exten line like it is in this example |
| 00:18.20 | litage | in the asterisk-users mailing list today, there's been a lot of talk about DTMF issues (the subject is "DTMF feedthru again..."). i can't figure out if the problem they're referring to pertains only to asterisk used with sipura devices, or if the problem pertains to any device connected to *. any ideas? |
| 00:20.14 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 00:21.46 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 00:26.51 | *** join/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 00:27.02 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 00:28.45 | PMantis | What in the world does this mean? |
| 00:28.45 | PMantis | Jun 6 20:28:03 WARNING[17997]: channel.c:2328 set_format: Unable to find a codec translation path from unknown to unknown |
| 00:29.07 | PMantis | Or more correctly asked, how do I fix that? |
| 00:30.21 | JoseBravo | Im looking for Billing solution, what is the best one, AstBill? |
| 00:34.43 | bkw_ | PMantis, you have a looped up context |
| 00:34.48 | bkw_ | calls come in and loop around |
| 00:35.47 | PMantis | bkw_, No. I mean that if I dial a number that's hosted on my * machine, my dialplan sends it out one provider only to return on another provider, and get answered by my * box. |
| 00:35.58 | PMantis | bkw_, It should simply stay on this box to begin with. |
| 00:36.46 | PMantis | I found that placing my includes above or below has no effect... the _NXXXXXX line matches it and is never checks the included context. |
| 00:37.26 | Dr-Linux | what's new in 1.2.9.1 ? |
| 00:37.38 | PMantis | Dr-Linux, IAX2 security fix. |
| 00:38.03 | Dr-Linux | i see, i read something on asterisk.org |
| 00:38.31 | Dr-Linux | not sure when asterisk will give some new features :S |
| 00:38.54 | PMantis | Dr-Linux, You have the code.. add some. :) |
| 00:39.14 | Dr-Linux | PMantis: i don't know languages :( |
| 00:40.28 | PMantis | Dr-Linux, I don't code * either, but I thought I'd jump on the bandwagon of the typical IRC response. :) |
| 00:41.17 | Dr-Linux | PMantis: today i installed 1.2.9.1 on my new Dual server |
| 00:41.34 | PMantis | Hmmmmmmmm |
| 00:41.47 | PMantis | I get to do that soon.. |
| 00:42.01 | dlynes_office | Anyone on that's familiar with sangoma a200d's? |
| 00:42.10 | PMantis | creating a * box for a client... will be a Dual chip, dual core machine. :) |
| 00:43.32 | Dr-Linux | PMantis: mine is dual core |
| 00:43.36 | Dr-Linux | Dell |
| 00:44.03 | PMantis | Dr-Linux, Yeah, saw that... This one will be two dual core chips. SuperMicro |
| 00:44.29 | PMantis | That'll be lots of fun! :) |
| 00:45.02 | dlynes_office | PMantis: another race issue, i'm guessing in 1.2.9.1? |
| 00:45.25 | PMantis | dlynes_home, race issue? |
| 00:45.31 | Dr-Linux | ? |
| 00:47.49 | dlynes_office | PMantis: threadlocking? |
| 00:52.25 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 00:52.29 | PMantis | dlynes_office, What are you referring to? I don't follow... |
| 00:53.36 | dlynes_office | PMantis: when an operation is taking place inside a thread, and it accesses a resource, then another operation outside that thread tries to access the same resource, both will try to lock the resource to use it (usually) |
| 00:54.01 | dlynes_office | PMantis: but one of them might forget to relinquish the lock, or might be in a tight loop waiting for the lock to be released |
| 00:54.16 | PMantis | ok... |
| 00:54.52 | dlynes_office | anyways...i'm guessing that's what 1.2.9.1 fixes, but I haven't checked the changelog yet, to be sure |
| 00:55.09 | PMantis | Ahhhhhhhh |
| 00:55.19 | PMantis | That's what this was in reference to! heh |
| 00:55.33 | dlynes_office | i remember someone talking on asterisk-dev last night about a race condition that they had found...perhaps that's what 1.2.9.1 fixes |
| 00:55.48 | russellb | no, 1.2.9.1 fixes a security issue in chan_iax2 |
| 00:55.56 | dlynes_office | ah |
| 00:56.06 | russellb | well, 1.2.9 fixes the security issue, and 1.2.9.1 fixes a bug introduced by the security fix :) |
| 00:56.08 | dlynes_office | then the race condition they were talking about last night only affects trunk? |
| 00:56.20 | russellb | yeah, that was me, just trunk |
| 00:56.42 | dlynes_office | ah...i thought it was you, but my memory's bad, and so I didn't want to mention any names in case i was wrong |
| 00:58.31 | russellb | lol, right. |
| 00:58.34 | dlynes_office | heh |
| 00:58.41 | Nivex | dlynes_office: your funeral. |
| 00:58.42 | russellb | and you'll be able to process an astounding 3 calls at a time |
| 00:58.44 | PMantis | dlynes_office calls it Javterisk |
| 00:58.46 | russellb | with no hardware support |
| 00:59.10 | dlynes_office | russellb: nah...java's not that bad...it's bad, but not as bad as visual basic |
| 00:59.10 | Sedorox | java.... sucks... |
| 00:59.15 | Sedorox | my my $0.01 |
| 00:59.19 | Sedorox | just my* |
| 00:59.33 | dlynes_office | Sedorox: just your * on the line? |
| 00:59.37 | russellb | Sedorox: agreed |
| 00:59.45 | Sedorox | dlynes_office: eh? |
| 00:59.55 | Sedorox | I meant just my $0.01 |
| 00:59.59 | dlynes_office | Sedorox: s-p-e-l-l i-t o-u-t |
| 01:00.05 | Sedorox | see above :p |
| 01:00.06 | dlynes_office | Sedorox: just my asterisk |
| 01:00.19 | dlynes_office | Sedorox: just my assterisk on the line :p |
| 01:00.28 | Sedorox | lol |
| 01:01.40 | dlynes_office | yeah...just imagine the call quality that would result every time the Java garbage collector kicked in :p |
| 01:03.02 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 01:10.39 | *** join/#asterisk cybergyp1y (n=mark@APoitiers-156-1-10-247.w86-207.abo.wanadoo.fr) |
| 01:11.22 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 01:12.12 | dlynes_office | anyone familiar with sangoma cards? |
| 01:19.51 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 01:20.27 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 01:27.45 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 01:28.59 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 01:29.43 | *** join/#asterisk holy_wood (n=benjamin@modemcable024.197-203-24.mc.videotron.ca) |
| 01:29.48 | holy_wood | vous ĂȘtes gay ? |
| 01:30.05 | *** part/#asterisk holy_wood (n=benjamin@modemcable024.197-203-24.mc.videotron.ca) |
| 01:31.39 | *** part/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 01:34.02 | ManxPower | OK everyone, this is a poor geek. If anyone that I've helped in the past can send a few dollars to eric@fnords.org via paypal it would be appreciated. |
| 01:37.08 | NewSole | hey manx |
| 01:39.59 | *** join/#asterisk iq|mobile (n=iq@71-215-55-11.omah.qwest.net) |
| 01:42.12 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 01:44.30 | *** join/#asterisk surye (i=1045@ix.c14n.org) |
| 01:45.35 | surye | Hey, when my Cisco 7940 attempts to register with asterisk using SIP 7.3, it's not registering, and the telnet debug on the phone reports this: E640 REG msg unsupported: in 404, request failure. Any idea's? |
| 01:45.35 | Nugget | telnet is eeeeeeevil! |
| 01:46.05 | PMantis | telnet is useful for debugging |
| 01:46.24 | surye | It seems to be a knee-jerk bot ;) |
| 01:46.48 | surye | But yea, it seems the phone doesn't understand the asterisk server.. |
| 01:47.24 | *** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net) |
| 01:47.54 | russellb | it doesn't understand a 404? heh |
| 01:47.56 | *** part/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 01:48.43 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 01:55.28 | generalhan | ok all have fun ... i cant take another minute of work i gotta get outta here ! |
| 01:55.32 | generalhan | talk to everyone tomorrow |
| 01:55.38 | generalhan | ~generalhan |
| 01:55.40 | jbot | you are, like, THE MAN |
| 01:55.47 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:55.48 | generalhan | yeah ... had to see that one more time ! lol |
| 01:55.52 | generalhan | hasta everyone ! |
| 01:56.31 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 01:57.18 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 01:59.29 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 02:03.46 | *** part/#asterisk AndrewKT (n=andrewkt@user-0c8h5qn.cable.mindspring.com) |
| 02:08.47 | techman97_andy | wtf is that thing? |
| 02:08.54 | techman97_andy | ~techman97_andy |
| 02:09.05 | techman97_andy | doh, I'm not as cool as generalhan |
| 02:09.07 | techman97_andy | =P |
| 02:13.16 | sevard | So |
| 02:13.33 | sevard | What does one do when the boss commands the engineer to have access to the linux servers |
| 02:14.09 | sevard | shutdown |
| 02:14.11 | sevard | reboot |
| 02:14.12 | sevard | log out |
| 02:14.25 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 02:14.34 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 02:14.38 | *** join/#asterisk litecode (n=andrewb@ip-66-235-230-20.sterlingnetwork.net) |
| 02:15.23 | litecode | for some reason, i have some fax only calls that are taking 142,000 seconds. My bill from my upstream was massive. is there a way to set a timer, than if a call is not completed, in say... 120 seconds, it's killed? |
| 02:16.45 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 02:17.24 | techman97_andy | hey all, so in the CLI / Asterisk Manager, what is DBGet and DBPut? What can they do for me? |
| 02:18.33 | sevard | techman97_andy: show application DBput |
| 02:21.53 | techman97_andy | hmmmm |
| 02:28.16 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 02:31.48 | ManxPower | Hello, NewSole |
| 02:36.51 | *** part/#asterisk P-NuT (n=P-NuT@fw.office.unitedip.net.au) |
| 02:39.01 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 02:57.16 | *** join/#asterisk b00mer (n=b00mer@ip68-100-65-89.dc.dc.cox.net) |
| 02:57.59 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 02:59.06 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 02:59.27 | Telamon | I'm having a problem with SIP registering between two servers. Basically, I have a user account on server A, and I want server B to register as being that user so I can put the dialplan on server B. But when I call the user from server A, I'm getting "congestion error", even though server B shows the user as being registered. What am I doing wrong? |
| 03:00.41 | Telamon | Err, sorry, error is circuit busy, not congestion error. |
| 03:03.39 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 03:14.10 | *** join/#asterisk crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net) |
| 03:14.19 | crshman | what can cause an SIP line timeout? |
| 03:14.22 | crshman | i can ping the host via ip and hostname (sip.broadvoice.com) what else am i missing? |
| 03:15.10 | Telamon | crshman: Did you check your firewall logs? Are you behind a nat? |
| 03:15.33 | crshman | no i'm not, it worked just fine for like 5 minutes i added an extension and reloaded the config and it started to fail |
| 03:16.09 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:17.43 | Telamon | Hmm, dunno. |
| 03:18.37 | crshman | erm today is bad day, nothing but problems with asterisk today =( |
| 03:19.27 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:20.07 | *** join/#asterisk nain (n=nain@202.59.90.182) |
| 03:20.13 | nain | Hi Every body |
| 03:20.52 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 03:21.36 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 03:21.52 | Freman | I've got a little issue with placing calls.. I'm getting double ringtones when I'm calling people (ie: normal aus ringtone + the us one) |
| 03:22.15 | littleball | hi, i am thinking how to combine SER and asterisk to setup a big media relay system... Anyone has experience ? |
| 03:22.15 | feld | Freman: whats the aus ringtone sound like? lol =/ |
| 03:22.59 | Freman | I don't know if it's happening on zap calls, but it's happening on sip > iax calls |
| 03:24.38 | nain | Can any one guide HOw to create multiple fallback routes if 1 or 2 fails...? Any Macro or help ful Dial plan ??? |
| 03:25.11 | CunningPike | nain: Easy. Have consecutive Dial() statements |
| 03:26.48 | nain | CunningPike: mean Dial(Route1&Route2&Route3) ? |
| 03:27.29 | CunningPike | nain: No. Consecutive, as in one after the other. |
| 03:27.50 | CunningPike | nain: exten => s,1,Dial(route1) |
| 03:28.01 | CunningPike | nain: exten => s,2,Dial(route2) |
| 03:28.10 | CunningPike | nain: And so forth |
| 03:28.41 | nain | CunningPike: If Route 1 is Ok then no need to dial route2 but this might call second route as well ?? |
| 03:28.53 | [TK]D-Fender | nain |
| 03:28.55 | CunningPike | nain: It won't |
| 03:28.57 | [TK]D-Fender | nain : nope |
| 03:29.11 | [TK]D-Fender | nain : normally after a successful call it will hang up |
| 03:29.15 | nain | How ? mean it won't run next priority ? |
| 03:29.29 | feld | [TK]D-Fender: what if nobody answers |
| 03:29.38 | feld | wont it go to the next line, Dial the next route? |
| 03:29.40 | nain | but if call is not successfuly or even a busy, or chanunavail or different dialstatus then ? |
| 03:30.05 | CunningPike | nain: A Dial() statement effectively ends the dialplan execution when one party hangs up |
| 03:30.15 | nain | Exactly if s-NOANSWER won't it dial next route ? |
| 03:30.17 | [TK]D-Fender | might be worth checking dialstatus |
| 03:30.17 | *** part/#asterisk downunder33 (n=robert@219.95.248.213) |
| 03:30.41 | CunningPike | nain: Well, it's what we do....... |
| 03:31.50 | nain | Actually I want to setAccount(ROUTEn) for each route, for which call successfully routed |
| 03:32.57 | nain | In consective Dial statment how i can set AccountCode for successfull Route? |
| 03:34.04 | Freman | whoot, fixed the double ringtone... |
| 03:34.14 | Freman | disable callprogress |
| 03:34.39 | Freman | now I have a problem with how long it takes to set up a call.. |
| 03:35.29 | CunningPike | nain: Interleave setAccount() statements with your Dial() statements |
| 03:35.30 | feld | :( |
| 03:37.26 | nain | CunningPike: It could be fine...., But if can any one suggest a good macro which perform call routing according to dial status and set their account code as well.... to make dial plan more neat |
| 03:38.08 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 03:43.11 | nain | Any body guide me how can i create macro that perform condition like this: if CHANUAVAIL GOTO ROUTE 1, if route 1 fail goto route 2 and so on... |
| 03:45.55 | [TK]D-Fender | nain : GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?10) |
| 03:47.45 | nain | [TK]D-Fender: Well this is for first time check, how to check that the same statement for 2nd time if dial statement at extensions 10 fail? |
| 03:48.01 | [TK]D-Fender | nain : Another GotoIf just like that... |
| 03:50.16 | nain | <[TK]D-Fender>: i will appreciate and thankful to you if you can create a macro here.... |
| 03:51.09 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 03:51.14 | [TK]D-Fender | nain : just interlace your dial's and gotoif's |
| 03:51.15 | Kerry_G | ~ centosbug |
| 03:51.17 | jbot | methinks centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit. |
| 03:51.32 | nain | exten => _NXXNXXXXXX,3,macro(routing,H323/1${EXTEN}@P-ROUTE0) |
| 03:51.38 | [TK]D-Fender | nain : nothing more to say... do your dial, if you don't like the result jump to another dial... sue your imagination... |
| 03:51.53 | nain | Sorry that is not for here. |
| 03:52.09 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 03:52.12 | nain | <[TK]D-Fender>: i got your point let me create and show you if that is right.. |
| 03:52.20 | *** join/#asterisk bmg505 (n=leon@196.209.33.145) |
| 03:53.51 | file | [TK]D-Fender: hrm, sue your imagination... marvelous idea |
| 03:54.15 | feld | file: how much u think u can get in America for suing your imagination? |
| 03:54.32 | [TK]D-Fender | file : Beter dividends :) |
| 03:54.36 | file | not a clue! :D |
| 03:54.48 | [TK]D-Fender | wasjhgdflasdf |
| 03:54.59 | [TK]D-Fender | I can't type tonight.... |
| 03:55.03 | CunningPike | My imagination doesn't have any money |
| 03:55.07 | file | [TK]D-Fender: or any other time |
| 03:55.16 | [TK]D-Fender | file ; SHUP YUO! |
| 03:55.17 | CunningPike | file: Bah - just beat me to it |
| 03:55.47 | file | I <3 e-tickets |
| 03:58.20 | Freman | so... can anyone explin the value of hint's and subscriptions? |
| 03:59.03 | [TK]D-Fender | Freman : So you can see the status of extensions on your system |
| 03:59.57 | Freman | I suppose that'll be more valuable when I have extensions with programmable leds and buttons huh |
| 04:00.38 | file | [TK]D-Fender: I haven't found an excuse yet to go back to Montreal for something :\ |
| 04:00.59 | *** join/#asterisk wigalowski (n=wigalows@c-67-161-244-209.hsd1.ut.comcast.net) |
| 04:01.32 | wigalowski | i am looking for open source call center style reporting. Something to do queue reporting with, what exists already? |
| 04:01.35 | [TK]D-Fender | Jazz festival and Jut for Laughs is soming up... Grand Prix as well... |
| 04:02.11 | file | ooh true |
| 04:02.12 | [TK]D-Fender | wigalowski : Queuemetrics. AMP also has a module for that which you should be able to exorcise. |
| 04:02.28 | wigalowski | are they anygood? |
| 04:02.32 | wigalowski | any good? |
| 04:02.43 | [TK]D-Fender | wigalowski : QueueMetrics is, not sure on the others |
| 04:02.59 | wigalowski | awesome, will give it a shot, thanks |
| 04:03.11 | [TK]D-Fender | wigalowski : Check them out on the WIKI |
| 04:03.50 | wigalowski | are they all license based? |
| 04:03.54 | wigalowski | look like QueueMetrics is |
| 04:04.37 | [TK]D-Fender | some are free, others not so... |
| 04:04.44 | [TK]D-Fender | just start looking.... |
| 04:04.52 | [TK]D-Fender | YGWYPF as well often |
| 04:05.05 | wigalowski | ok, any that we know just suck and I should stay away from? |
| 04:05.24 | file | he who expects the world for nothing may find themselves with a black hole of DOOM |
| 04:05.47 | feld | file: 1, 2, or 3? which DOOM? |
| 04:05.54 | file | 42 |
| 04:06.11 | wigalowski | yes, but i just shelled out over 300,000 on an NEC. So I am broke. Stay away from NEC. |
| 04:06.12 | feld | ahhh that's a good one too :) Doom: the answer to everything |
| 04:06.25 | nain | <[TK]D-Fender>: Would you please check the dial plan for fallback route according to your suggestion here ? http://hashphp.org/pastebin.php?pid=6940 |
| 04:06.47 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:07.13 | [TK]D-Fender | nain : Priority jumping is DEAD... make that 1.2 compliant.... |
| 04:07.51 | nain | <[TK]D-Fender>: won't it work and how to compliant it with 1.2 ? mean what to change in this dial plan |
| 04:09.04 | [TK]D-Fender | nain : The idea is pretty close though |
| 04:09.33 | nain | <[TK]D-Fender>: mean just move to next priority with GotoIF statement or something else? |
| 04:11.33 | [TK]D-Fender | nain : http://hashphp.org/pastebin.php?pid=6941 |
| 04:12.30 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:13.44 | nain | <[TK]D-Fender>: Aha thanks..... |
| 04:17.23 | nain | <[TK]D-Fender>: One more question plz... I am using NuFone H323 driver and i have set accountcode and amaflag in h323.conf but unable to find the path where CDR is being generated in box. I have checked /var/log/asterisk/cdr-csv and custom folder... |
| 04:18.12 | [TK]D-Fender | No idea... never played with CDR really. |
| 04:18.31 | nain | <[TK]D-Fender>: Ok np |
| 04:18.38 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 04:18.39 | [TK]D-Fender | ok, I'm fried..... back tomorrow *yawn* |
| 04:18.57 | nain | Any body else have idea where NuFone H323 driver is generating CDR |
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| 04:43.16 | supjigatr | Anyone here using the chan_ss7 |
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| 05:11.12 | yxa | can someone point me to a link for crimping my own E1 cable? |
| 05:11.44 | techman97_andy | http://kb.digium.com/entry/1/124/ |
| 05:14.32 | yxa | techman97_andy thanks |
| 05:14.35 | techman97_andy | np |
| 05:14.39 | techman97_andy | that do it for you? |
| 05:15.13 | yxa | techman97_andy yeah, looks easy enough |
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| 05:30.08 | Snake-Eyes | what sort of things do people use account code for in cdr? I've seen a few uses for it. |
| 05:32.46 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 05:33.10 | h0 | good evening everyone |
| 05:33.11 | dlynes_office | Snake-Eyes: easy billing |
| 05:34.00 | dlynes_office | Snake-Eyes: then you don't have to try and figure out what contexts were used for individual customers, figure out whether the call was billable or not, ... |
| 05:35.13 | dlynes_office | Snake-Eyes: if you set amaflags to billable for an outbound long distance call, and leave default at documentation, otherwise, and also for any incoming or outgoing call (or however you choose to define it), set an account code, it's quite obvious which calls belong to that customer |
| 05:35.47 | dlynes_office | Snake-Eyes: also, if you're using disa, or a common number for dialin access to asterisk, you have no other way of knowing who made that call |
| 05:36.10 | dlynes_office | Snake-Eyes: and if you use authentication codes, you can use those to adjust your account code for more billing info |
| 05:37.11 | glm2k | well said |
| 05:40.14 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 05:40.57 | Snake-Eyes | dlynes_home, wow thanks, most lilly take me while to digest it all :) |
| 05:42.18 | dlynes_office | Snake-Eyes: yeah...like for one customer i've got, they only have one main account, but each user enters their own authentication code; then I add that authentication code on to the end of the main account code to form the real account code |
| 05:42.56 | dlynes_office | Snake-Eyes: that way i can have ten different accounts for one office that uses their phone system for a number of different companies within the same office |
| 05:43.44 | Snake-Eyes | dlynes_home, ah pbx for a whole office block of companies, cool |
| 05:44.09 | dlynes_office | yeah, but they're all sharing three sipura 2000's |
| 05:44.21 | dlynes_office | and then those three sipura 2000's connect to our main asterisk softswitch |
| 05:44.36 | Snake-Eyes | nice |
| 05:44.44 | dlynes_office | they only use it for long distance |
| 05:44.59 | dlynes_office | so they have six outbound long distance lines |
| 05:45.10 | dlynes_office | and something like ten different authentication codes |
| 05:45.47 | dlynes_office | It's a showcase center for condos, so they rent space out to various condo developers |
| 05:45.56 | *** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com) |
| 05:46.04 | dlynes_office | and rent out phone extensions to go with those showcases |
| 05:46.16 | Gabriel25 | anyone know if avaya IP phone 4624 is working SIP ? |
| 05:46.21 | dlynes_office | some of the condo developers even set up call centers there, too |
| 05:46.27 | Snake-Eyes | hehe |
| 05:46.50 | dlynes_office | so there's currently two call centers there, but each project only lasts maximum three months |
| 05:47.06 | Snake-Eyes | so there are only 6 outside numbers which all these extensions use |
| 05:47.18 | dlynes_office | only six outside numbers for long distance calls |
| 05:47.27 | dlynes_office | they have a number of analog lines, too |
| 05:47.34 | dlynes_office | everything is all hooked up to an nec pbx |
| 05:48.12 | *** join/#asterisk downunder33 (n=robert@60.51.217.62) |
| 05:48.16 | dlynes_office | so it's all outbound calls only; no inbound calls |
| 05:49.04 | Snake-Eyes | I was thinking of having every number associated to account in accounting software or have numbers that belong to one company grouped |
| 05:49.05 | Snake-Eyes | ah |
| 05:49.28 | Snake-Eyes | might rethink this abit :) |
| 05:49.47 | dlynes_office | Snake-Eyes: yeah...it's easier to use accountcodes, and then you have the flexibility to decide how the customer's billing is grouped |
| 05:50.41 | dlynes_office | Snake-Eyes: so for the customers where you want to group certain phone numbers, just before you dial, do a SetAMAFlags(billing) |
| 05:50.47 | Snake-Eyes | dlynes_office, so every time a new number is created for existing customer give it the same accountcode |
| 05:50.56 | dlynes_office | Snake-Eyes: and then for each of their sip.conf files, do an accountcode=xxxxxxxxxxx |
| 05:51.22 | dlynes_office | Snake-Eyes: but set the same accountcode for every sip device associated iwth that number |
| 05:51.55 | Snake-Eyes | dlynes_office, why use this SetAMAFlags(billing)? |
| 05:52.10 | dlynes_office | Snake-Eyes: so you know what constitutes a billable call, and what doesn't |
| 05:52.22 | dlynes_office | Snake-Eyes: we don't charge for local, or long distance calls |
| 05:52.46 | dlynes_office | Snake-Eyes: we charge for long distance, 411, 1-NPA-555-1212, ... |
| 05:53.03 | Snake-Eyes | dlynes_office, ah ok, my setup differs, everything that goes through/out of asterisk will be billable |
| 05:53.11 | dlynes_office | that way when you're reading the billing records, you know what constitutes a billable call, and what doesn't |
| 05:53.33 | dlynes_office | Snake-Eyes: ah...yeah...we have a pri and a voip provider |
| 05:53.33 | Snake-Eyes | crap just remmeber emergency numbers |
| 05:54.02 | Snake-Eyes | they cant be billable, guess i have to use the flag :) |
| 05:54.10 | dlynes_office | Snake-Eyes: yeah...when they want to dial 911, if they only have one analog line, we drop whoever's talking on there, and make a 911 call on the analog line |
| 05:54.42 | dlynes_office | Snake-Eyes: otherwise, we try to grab a free analog line to do 911 on |
| 05:54.54 | dlynes_office | Snake-Eyes: if there isn't, find one that's not on a 911 call, drop the call, and make a 911 call |
| 05:55.58 | Snake-Eyes | dlynes_office, yea, I was thinking more some one on our voip network make 911/000 call by mistake on network, dont want to cut call off cause they dont have enough prepaid credits |
| 05:56.19 | *** join/#asterisk Splat (n=Splat@220-253-97-230.TAS.netspace.net.au) |
| 05:56.26 | dlynes_office | operator calls aren't free calls |
| 05:56.33 | dlynes_office | even your ilec charges for those |
| 05:57.00 | dlynes_office | erm...nvm |
| 05:57.01 | Snake-Eyes | im pretty sure 000 calls are free |
| 05:57.03 | dlynes_office | i'm not thinking |
| 05:57.11 | dlynes_office | I'm thinking of 0-NPA-NXX-xxxx |
| 05:57.13 | dlynes_office | not 0 |
| 05:57.14 | dlynes_office | :) |
| 05:57.18 | Snake-Eyes | hehe |
| 05:57.19 | *** join/#asterisk clive- (n=pirch@dsl-146-96-02.telkomadsl.co.za) |
| 05:57.28 | dec | so the SIP plc/jitterbuffer stuff... is that included in /branches/1.2 or only in /trunk/ ? |
| 05:57.40 | dlynes_office | I haven't used the operator in so long, I can't even remember why i would call the operator |
| 05:58.10 | Snake-Eyes | whats a operator :P |
| 05:58.23 | dlynes_office | Snake-Eyes: the only reason you would call the operator is to make a station to station call or a collect call, right? |
| 05:58.57 | Snake-Eyes | dlynes_office, most collect calls are partly automated now, no person |
| 05:59.17 | dlynes_office | Snake-Eyes: ok, so if the user called 0 to make a station to station call |
| 05:59.17 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 05:59.28 | dlynes_office | Snake-Eyes: how are you going to know that they're making a long distance call on your dime? |
| 05:59.35 | dlynes_office | Snake-Eyes: and at a premium rate, at that? |
| 05:59.43 | vooduhal | Is anyone else having problems getting monitor-join=yes to actually do anything from queues.conf? |
| 06:00.49 | Snake-Eyes | dlynes_office, ive heard of station to station call before but cant remmeber defination, phoning some one on the same pbx/exchange ? |
| 06:01.21 | dlynes_office | Snake-Eyes: you can do station to station, operator to operator, station to operator and operator to station from 0 |
| 06:01.28 | dlynes_office | Snake-Eyes: they all cost money |
| 06:01.34 | dlynes_office | Snake-Eyes: and no discounts, either |
| 06:01.44 | stephane_ | jour |
| 06:01.51 | dlynes_office | Snake-Eyes: station to station call is the same as 1-NPA-XXX-xxxx, but it's 0 instead |
| 06:02.19 | dlynes_office | Snake-Eyes: well, for north america...not sure what it would be in oz |
| 06:03.00 | Snake-Eyes | dlynes_home, ah ok, so from one operator center to another |
| 06:03.35 | Snake-Eyes | dlynes_home, only remmeber it from some movie, never heard of it used outside the north america |
| 06:06.19 | dlynes_office | ah |
| 06:06.53 | Snake-Eyes | dlynes_office, would something simiarly be, some makes call to operator then operator transfer the call to whom ever X, and one our records the call is only to operator not X ? |
| 06:07.29 | *** join/#asterisk Eggplant (i=No@dsl-216-155-212-168.cascadeaccess.com) |
| 06:08.25 | Snake-Eyes | * and when our cdr's show call only went to operator |
| 06:09.07 | vooduhal | Anyone here have a lot of experience with app_queue? |
| 06:10.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:11.04 | vooduhal | I'll take that as a no. |
| 06:12.41 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 06:15.54 | dlynes_office | Snake-Eyes: correct |
| 06:17.27 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:17.50 | dlynes_office | heya littlebals |
| 06:17.52 | littleball | hello, who can recommed a media relay system architecture to me? |
| 06:22.09 | littleball | dlynes_office |
| 06:22.14 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-251-57.telkomadsl.co.za) |
| 06:22.18 | X-Gen | hey freaks |
| 06:22.44 | littleball | who can recommed an archititecture of a media relay system? |
| 06:23.04 | littleball | one should be able to scale |
| 06:23.39 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:24.41 | InHisName | When using Sipura for extensions, should the "sip show registry" have an entry for each extsion ? |
| 06:26.45 | clive- | x-gen howzit |
| 06:29.13 | *** join/#asterisk v_farmer (i=rvilleri@xs6.xs4all.nl) |
| 06:32.38 | *** join/#asterisk v_farmer (i=rvilleri@xs6.xs4all.nl) |
| 06:34.05 | dlynes_office | InHisName: not usually, no |
| 06:38.17 | dlynes_office | anyways...heading home...ttfn |
| 06:39.48 | sevard | gayyy |
| 06:39.59 | sevard | this fricken router has a faster cpu than my main machine |
| 06:40.21 | Snake-Eyes | dlynes_office, night |
| 06:40.23 | sevard | come on boss, does that tell you 'buy new equpitment' |
| 06:40.33 | sevard | $30 dollar fricken router |
| 06:40.39 | Snake-Eyes | lol |
| 06:40.47 | sevard | :( |
| 06:41.12 | sevard | i'm not joking dude, it has 2.3x RAM and 2x cycles to play with |
| 06:41.29 | sevard | i'm going to be using this router as my main machine from now on :| |
| 06:41.45 | *** join/#asterisk Trimor (n=trimor@host202-147-179-183.isb.dancom.net.pk) |
| 06:42.27 | Trimor | hi! |
| 06:42.42 | Trimor | can ne body tell me the differenc ebetween fxs and fxo |
| 06:43.06 | Trimor | i've purchased a tdm 400 card with one fxs module |
| 06:43.16 | sevard | fxo == foriegn exchange office, fxs == forigen exchange service, fxo talks fxs and fxs talks fxo |
| 06:43.47 | sevard | so what you should be doing now is heading over to voip-info.org and reading about your new card and what fxo/fxs is and how it works. |
| 06:43.53 | sevard | before you eeeeeeven plug it in. |
| 06:43.56 | Trimor | ahan |
| 06:43.57 | Trimor | right |
| 06:44.06 | Trimor | thank you |
| 06:44.09 | sevard | no problem. |
| 06:44.39 | sevard | (p.s. my rule of thumb is to read about the product before i go spending a buttload on it :P ) |
| 06:45.14 | InHisName | When using Sipura for extensions, should the "sip show peers" have (unmonitored) or a ms time for each extension? |
| 06:46.55 | kaldemar | InHisName: if you have qualify=yes defined for a peer, it has a time in ms or unreachable, if qualify=no or the parameter is not defined, it has unmonitored. |
| 06:47.09 | sevard | I believe the "ms" is lag in miliseconds from the pbx to the sip client and you have to set something to make it monitored |
| 06:47.14 | sevard | yeah, listen to him. |
| 06:50.11 | dlynes_home | InHisName: if it's dynamic it should be in milliseconds; otherwise, it should be unmonitored |
| 06:50.30 | dlynes_home | InHisName: erm i mean in seconds |
| 06:50.40 | kmilitzer | Morning everyone ... I have strange CDR records since update to 1.2.8 on saturday morning ... anyone else seeing something? |
| 06:50.51 | dlynes_home | kmilitzer: type /topic |
| 06:50.56 | dlynes_home | kmilitzer: you should be running 1.2.9.1 now |
| 06:51.06 | sevard | it's fricken 2 am dude |
| 06:51.13 | sevard | who wants to update at 2 am |
| 06:51.18 | dlynes_home | me!!!!!!!!!! |
| 06:51.25 | dlynes_home | but seriously...it's midnight |
| 06:51.29 | sevard | heh |
| 06:51.30 | dlynes_home | you're on the wrong coast |
| 06:51.58 | kmilitzer | kmilitzer: I have 1.2.9 since yesterday ... the described bugs are only IAX, I just use SIP |
| 06:51.58 | kaldemar | seriously it's almost 10 am, you should check your clocks. |
| 06:52.11 | kmilitzer | Argh, too early. I meant dlynes_home |
| 06:52.14 | dlynes_home | yeah, no doubt, eh? |
| 06:52.42 | dlynes_home | kaldemar: hauschtenappa? |
| 06:53.12 | dlynes_home | kaldemar: don't know how to spell it...i only know how to say it :) |
| 06:53.15 | clive- | kmilitzer hi, what are the bugs, or where can I read about them>? |
| 06:53.55 | dlynes_home | kmilitzer: the cdr bugs are only iax? |
| 06:55.22 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:56.39 | kmilitzer | clive-: The "bug" is, that I get strange/wrong CDR records, if I got sent an SIP REFER |
| 06:57.01 | kmilitzer | dlynes_home: I am not aware of CDR bugs in 1.2.8 ... are there any? |
| 06:57.24 | dlynes_home | kmilitzer: apparently you are -> Morning everyone ... I have strange CDR records since update to 1.2.8 on saturday morning ... anyone else seeing something? |
| 06:57.39 | kaldemar | dlynes_home: uhh? |
| 06:57.51 | dlynes_home | kaldemar: oh..sorry...I thought you knew Finnish |
| 06:58.21 | kmilitzer | dlynes_home: Meant: I am not aware of any _already known" CDR bugs |
| 06:58.25 | kaldemar | dlynes_home: i do, i am finnish, but that's definitely not finnish. :) |
| 06:58.38 | dlynes_home | kaldemar: ah...maybe it's only Canadian Finnish |
| 06:58.46 | dlynes_home | kaldemar: i.e. a Finnish dialect spoken here |
| 06:58.59 | dlynes_home | kaldemar: it means "How's the belly button?" |
| 06:59.17 | dlynes_home | kaldemar: but like i said...I don't know how to spell Finnish words/phrases |
| 06:59.40 | sevard | how am I on the wrong coast |
| 06:59.44 | kaldemar | oh, so it's mixed english and finnish. "how's the napa?" napa = belly button. |
| 06:59.50 | sevard | you're wrong. |
| 06:59.54 | dlynes_home | kaldemar: ah...hahaha |
| 07:00.06 | dlynes_home | kaldemar: and then they usually say Keska la maha |
| 07:00.29 | dlynes_home | kaldemar: i.e. it's in the middle |
| 07:01.03 | kaldemar | that would be "keskellä mahaa" :) |
| 07:01.07 | dlynes_home | ah |
| 07:01.18 | InHisName | Is there a preference to whether I use qualify=x000 or not in defining my extensions ? (unmonitored or ms time) |
| 07:01.58 | dlynes_home | anyways...it's a bit of Canadian Finnish humor I picked up from living in Thunder Bay for so many years |
| 07:02.19 | dlynes_home | It's the largest Finnish population outside of Finland |
| 07:02.46 | dlynes_home | sevard: the west coast is the best coast :)) |
| 07:03.09 | dlynes_home | InHisName: it depends |
| 07:03.12 | sevard | the west coast is only good when you hit alaska |
| 07:03.17 | sevard | everything below can go. |
| 07:03.20 | dlynes_home | sevard: lol |
| 07:03.38 | dlynes_home | InHisName: well, if your sip extensions are natted, it'll make a huge difference |
| 07:05.53 | kmilitzer | So as I see, nobody else have issues with SIP REFERs? |
| 07:06.56 | InHisName | I am runing a router/qos etc on same linux box as *, does that define a choice that I need ? (qualifiy defined or not) |
| 07:07.03 | dec | any opinions on the best echo cancellation filter to use with zaptel? |
| 07:08.08 | InHisName | How about the other west coasts? Japan, England, Europe, Africa, Austraila(sp). Does Asia have a west coast somewhere ? |
| 07:08.26 | dec | Australia. |
| 07:08.27 | InHisName | South america too |
| 07:08.57 | dlynes_home | kmilitzer: well, i don't use sip refers in my cdrs, so it's not an issue for me |
| 07:08.57 | dec | I'm on the South coast. |
| 07:09.12 | dlynes_home | dec: mg2, or mark2 |
| 07:09.42 | dec | dlynes_home - which one of those is better? ;) |
| 07:09.45 | InHisName | South sounds warm, hopefully not south coast of greenland. |
| 07:09.53 | dlynes_home | dec: it depends on your echo problems |
| 07:09.57 | dec | InHisName - South coast of Australia. |
| 07:10.05 | dec | dlynes_home - hmm okay, i'll read about those two filters a little more. |
| 07:10.07 | *** join/#asterisk apardo (n=apardo@87.217.146.210) |
| 07:10.10 | dlynes_home | dec: one echo canceller may not get rid of it and the next one will...ymmv |
| 07:10.16 | dec | Thanks. |
| 07:10.27 | kmilitzer | dlynes_home: Well, I usually do not too. I just had two sitations where users with a Sipura started to send REFERs and my CDRs got mangeld ... I cannot explain why |
| 07:10.34 | InHisName | Ahhh that spelling looks better than mine, Australia. |
| 07:10.43 | dlynes_home | kmilitzer: define mangled? |
| 07:11.21 | kmilitzer | dlynes_home: I get src and destination that both are not local. That can not happen, as I do not allow such calls |
| 07:11.40 | kmilitzer | dlynes_home: And I have durations of 0 and billsec of > 0 |
| 07:11.43 | dlynes_home | ah |
| 07:12.21 | kmilitzer | ~pastebin |
| 07:12.25 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 07:12.25 | dlynes_home | yeah...i remember a while back, asterisk 1.0.something, I was getting all kinds of garbled characters in my cdrs, too |
| 07:12.25 | dlynes_home | ~pb |
| 07:12.27 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 07:12.44 | dlynes_home | ah...cool...he's got more in there now |
| 07:13.36 | dlynes_home | ~pastebin |
| 07:13.39 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 07:13.41 | dlynes_home | ~pb |
| 07:13.43 | jbot | i guess pb is aka pastebin |
| 07:13.52 | *** join/#asterisk lorinc (n=ang@caracas-1593.adsl.interware.hu) |
| 07:15.01 | kmilitzer | dlynes_home: See here what I mean: http://pastebin.com/764595 |
| 07:16.05 | dlynes_home | yeah...that's weird |
| 07:16.46 | kmilitzer | dlynes_home: In that case destination and source were the same number ... I have another one, that is quite normal, except, that destination and source are both not local :( |
| 07:17.31 | kmilitzer | dlynes_home: I now disallowed REFERs on my SER in front of my asterisk as a workaround, but that still does not explain why I got these things since saturday |
| 07:17.32 | dlynes_home | kmilitzer: and it's that behaviour in 1.2.8, but not in 1.2.7.1? |
| 07:18.46 | kmilitzer | dlynes_home: Correct. As far as I can tell it started after an update to 1.2.8 on saturday morning. I had to update there because my asterisk ran into an deadlock because of a logrotate reload ... |
| 07:19.24 | InHisName | dlynes_home: I am runing a router/qos etc on same linux box as *, does that define a choice that I need for my sipura extensions ? I think extensions are NATted ? (qualifiy defined or not) |
| 07:19.39 | dlynes_home | InHisName: no, they're not natted in that case |
| 07:19.47 | dlynes_home | InHisName: if they're on the other side of the nat from asterisk |
| 07:20.04 | dlynes_home | InHisName: but you're connecting to them on the asterisk machine's local interface, not the external interface, correct? |
| 07:20.36 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:20.46 | InHisName | internet - eth0 - linux/asterisk - eth1 - 16port switch - sipura - analog phone (NATted or not ?) |
| 07:22.17 | InHisName | we'd be dvorak then |
| 07:22.47 | JackEstorm | mitcheloc: umm, it was |
| 07:23.00 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 07:23.37 | yxa | the led is steady green on my te411p card but cat /proc/zaptel/1 tells me: Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RECOVERING ClockSource |
| 07:23.43 | mitcheloc | ah, then nevermind me, would have been funny to see dozens of keyboard layouts though ;) |
| 07:25.53 | dlynes_home | InHisName: obviously not |
| 07:26.16 | dlynes_home | mitcheloc: the keyboard patent's long since expired |
| 07:27.12 | kmilitzer | dlynes_home: I just found a bug, that may match with my CDR problems ... 6579 |
| 07:27.31 | *** join/#asterisk halorgium (n=tim@202.50.176.27) |
| 07:27.32 | dlynes_home | kmilitzer: there ya go |
| 07:27.36 | *** part/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 07:27.36 | halorgium | evening |
| 07:27.47 | dlynes_home | kmilitzer: so you can download 1.2.9.1, patch it, and then install it :) |
| 07:27.59 | halorgium | what are the recommendations for installing asterisk on debian, from source or using the binary packages? |
| 07:28.47 | kmilitzer | dlynes_home: If there was a patch for it ... the bug was closed unresolved because of uncooperative reporter :( |
| 07:29.03 | dlynes_home | kmilitzer: what a putz |
| 07:29.08 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:29.31 | kmilitzer | dlynes_home: Is there a way to reopen it? |
| 07:31.04 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:31.58 | dlynes_home | kmilitzer: talk to Russell Bryant (russellb)...he's the one that closed it, or Kevin P Fleming (kevinpfleming), he's the CTO or something of Digium |
| 07:32.13 | dlynes_home | kmilitzer: they're both quite active in #asterisk-dev |
| 07:32.30 | dlynes_home | erm kpfleming, not kevinpfleming |
| 07:32.45 | kmilitzer | dlynes_home: Thanks, I'll try it there ... |
| 07:34.04 | *** part/#asterisk phonic (i=phonic@antisocial.nu) |
| 07:35.41 | dlynes_home | kmilitzer: btw...you might want to try when someone's awake |
| 07:35.48 | dlynes_home | kmilitzer: most of them are in North America |
| 07:35.56 | dlynes_home | kmilitzer: especially russell and kevin |
| 07:36.13 | dlynes_home | kmilitzer: it's almost 4am where they are right now |
| 07:36.33 | kmilitzer | dlynes_home: I hate timezones :( |
| 07:36.44 | dlynes_home | heh |
| 07:38.03 | *** join/#asterisk motu (n=motu@192.165.166.143) |
| 07:38.23 | littleball | hello, who can recommend an architecture for media relay system? |
| 07:39.16 | dlynes_home | littleball: doesn't SER do that? |
| 07:42.19 | littleball | dlynes_home, if all sip phones behind firewall, media relay will be used to relay the voice. Then the architecture is important to scale the system |
| 07:42.56 | dlynes_home | littleball: yeah...and doesn't SER do that? |
| 07:43.44 | *** join/#asterisk Ayatolah (n=mike@pool-141-149-114-131.pghk.east.verizon.net) |
| 07:43.47 | littleball | SER, i think SER is just a SIP proxy. how can it support thousands of current sip calls (relay calls) |
| 07:43.53 | *** part/#asterisk Ayatolah (n=mike@pool-141-149-114-131.pghk.east.verizon.net) |
| 07:44.58 | dlynes_home | littleball: yeah, and it sounds to me like you're looking for a SIP media proxy...I was under the impression SER did that |
| 07:45.08 | dlynes_home | littleball: i've never used it though, so I could be wrong |
| 07:45.11 | zoa | littlebalb: what are you looking for ? |
| 07:46.44 | littleball | dlynes_home, i know asterisk have nat=yes which works as media relay. Just want to know how to make it scalable. |
| 07:47.40 | littleball | zoa, i am designing a media relay system. I prefer to using asterisk work as the media relay component. But i need to put a lot of box to make the system to be scalable. Then what should be a good architecture? |
| 07:49.19 | dlynes_home | anways |
| 07:49.23 | dlynes_home | i need sleep |
| 07:49.23 | dlynes_home | laters |
| 07:49.30 | mitcheloc | littleball: freeswitch |
| 07:50.38 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 07:50.44 | littleball | why not asterisk? |
| 07:51.28 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 07:51.30 | dlynes_home | littleball: because he wants you to play games with your business with a piece of software that's still in its infancy |
| 07:51.37 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:51.43 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:51.51 | dlynes_home | littleball: freeswitch is going to be good when it's finished |
| 07:51.58 | dlynes_home | littleball: but it needs a lot of work yet |
| 07:52.05 | sevard | blah blah gayswitch |
| 07:52.12 | dlynes_home | littleball: i wouldn't consider it for anything serious at this point |
| 07:52.28 | littleball | dlynes_home, why not asterisk? i prefer to use asterisk because it already has goood ecosystem... |
| 07:52.47 | sevard | it can support desert and jungle creatures. |
| 07:52.49 | dlynes_home | littleball: taht's what i'm saying |
| 07:53.34 | littleball | so, come back to topic how to design a media relay system based on asterisk (maybe + ser) to make the system scalable. |
| 07:54.29 | dlynes_home | littleball: you might try asking cunningpike or ManxPower in the morning, too...they've both built some pretty large systems |
| 07:54.41 | dlynes_home | littleball: and [tk]-dfender, too |
| 07:54.55 | littleball | ok. thankns. |
| 07:55.08 | dlynes_home | on that note, i'm going to get some sleep |
| 07:55.12 | dlynes_home | it's 1am here |
| 07:56.06 | sevard | 3am |
| 07:56.11 | sevard | still at work |
| 07:56.12 | sevard | since 8am |
| 07:56.15 | sevard | :'( |
| 07:58.09 | zoa | littleball i have such a system |
| 07:58.23 | zoa | look at ser + rtpproxy or media proxy |
| 07:58.28 | littleball | zoa, can u explain the architeture? |
| 07:58.38 | littleball | or give me a useful link |
| 07:58.38 | littleball | ? |
| 07:58.55 | littleball | how about PSTN termination then? |
| 08:00.09 | yxa | guys other than bchan=1-15,17-31 |
| 08:00.09 | yxa | dchan=16 |
| 08:00.27 | yxa | i should also set clear= or fxsks= lines? |
| 08:00.34 | yxa | in zaptel.conf |
| 08:01.07 | tdi | does sb know the guy who wrote chan_fax? |
| 08:07.08 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 08:07.18 | tparcina | goodmorning group |
| 08:07.51 | sevard | good morning miss tparcina |
| 08:08.17 | tparcina | sevard, why you call me miss? |
| 08:10.36 | tparcina | i have problem (it isn't the fact that sevard calls me miss). every now and then stablished call hangs up. u use cisco phones, preaty strong hardware (P4@3GHz, 512MB) and Digium E1 interface card connected to my providers Cisco router with E1 interface |
| 08:13.37 | tparcina | when two phonecalls are established, only one hangs up randomly - is it isn't that interface goes down |
| 08:14.07 | tparcina | i will tourn on full loging, and I'll try to katch packets with ethereal |
| 08:14.53 | tparcina | but, can anybody sugest something more? has anybody have the same problem before? how can i check is the problem on my or providers side? |
| 08:15.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:15.25 | tparcina | dlynes, where are you when I need you most? :)) |
| 08:15.28 | sevard | we have most strong hardware for you five for two dolla |
| 08:17.44 | puzzled | hi |
| 08:18.51 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:20.42 | tparcina | and yes, my processor is 99% idle |
| 08:21.03 | Trimor | Can a simple hard voice modem can be used with asterisk as pstn gateway ??? |
| 08:21.15 | tparcina | mem is 200MB free (of 512 MB) |
| 08:21.36 | tparcina | Trimor, no you can't |
| 08:22.13 | Trimor | ahan, why so |
| 08:22.21 | tparcina | Trimor: you shoul buy some FXO gateways - analog cards |
| 08:22.30 | Trimor | well i've got that |
| 08:22.44 | Trimor | but i was lookin for the reason y a modem can't be used |
| 08:24.06 | tparcina | Trimor: do some readings and you'll find out why. i know it can't and that is enough for me. don't wona to spend several hours reading why not. I'll spend that time on more usefull way |
| 08:28.16 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 08:29.38 | Trimor | ahan |
| 08:29.50 | Trimor | k |
| 08:29.57 | Trimor | thnx |
| 08:30.54 | *** join/#asterisk Vahram (n=VX@83.139.6.86) |
| 08:33.04 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 08:35.05 | Dico_ | hello |
| 08:35.46 | Dico_ | humm, since i've patched my asterisk to version .9.1 i get a weird frame subclass type : -1 |
| 08:36.01 | Dico_ | do you know where this subclass type come come from ? |
| 08:58.04 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 09:02.00 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.38 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 09:04.49 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 09:04.55 | *** join/#asterisk lorinc (n=ang@caracas-1593.adsl.interware.hu) |
| 09:05.41 | RoyK | morning |
| 09:07.27 | Vahram | yep |
| 09:07.31 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 09:07.44 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 09:08.52 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 09:10.24 | *** join/#asterisk mfedyk (n=mfedyk@adsl-63-194-240-129.dsl.lsan03.pacbell.net) |
| 09:13.01 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:13.22 | mfedyk | Hi, is there a way to configure asterisk to only request rtp ports in a certain range? |
| 09:13.54 | mfedyk | I have a vonage ATA behind my firewall, so I have to map ports 10000:20000 to it |
| 09:14.15 | mfedyk | and I'd rather not have to use the same range for asterisk also. |
| 09:15.12 | *** join/#asterisk apardo (n=apardo@213.27.175.185) |
| 09:17.58 | Poincare | mfedyk: sip.conf? |
| 09:18.08 | mfedyk | actually, rtp.conf |
| 09:18.28 | mfedyk | someone pointed that out to me just now, thanks |
| 09:18.39 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:22.42 | Poincare | mfedyk: depending on your setup you might want to decrease the range also. 10000 ports is a lot... at least for a small company with 10 users :-) |
| 09:23.25 | mfedyk | what are the rules for port range rollover? |
| 09:23.58 | mfedyk | they may not make 10000 calls in a day, but they can in a week or less |
| 09:24.04 | mfedyk | (call center) |
| 09:24.34 | zoa | port range: |
| 09:24.37 | zoa | 1 port per call |
| 09:24.41 | mfedyk | also, how do I reload rtp.conf without restarting? |
| 09:24.49 | zoa | random allocated in this range |
| 09:24.50 | mfedyk | sip reload didn't do it. |
| 09:25.00 | zoa | so you would have enough with a range of 100 :) |
| 09:25.07 | mfedyk | and reload <tab> didn't show anything obvious. |
| 09:27.11 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 09:27.19 | Stephnie | hi |
| 09:28.14 | *** join/#asterisk eivindtr (n=wingnut-@ti211310a080-15945.bb.online.no) |
| 09:28.47 | Stephnie | Jun 7 14:30:58 WARNING[10337]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 09:29.00 | Stephnie | I am getting this problem....its a Big loop.. |
| 09:29.33 | Stephnie | any help? |
| 09:30.17 | eivindtr | Does anyone know if there is another way than using Agents I can make Asterisk realize an account is busy when one conversation is active, basically inhibiting the invite? |
| 09:30.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:33.32 | Stephnie | knock knock |
| 09:36.50 | MGSsancho | whos there |
| 09:36.51 | Stephnie | any one???? |
| 09:36.58 | Stephnie | Jun 7 14:30:58 WARNING[10337]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 09:37.17 | MGSsancho | oh a reall asterisk question pshh i dunno sorry |
| 09:37.43 | Stephnie | dont be....no one knows ;) |
| 09:45.06 | Stephnie | NOC NOC |
| 09:49.05 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 09:53.00 | *** join/#asterisk visik7 (n=visi@unaffiliated/visik7) |
| 09:53.02 | visik7 | hi |
| 09:53.05 | visik7 | I'm a noob |
| 09:53.18 | visik7 | is there a way to use asterisk with skypeout ? |
| 09:53.55 | Stephnie | Jun 7 14:30:58 WARNING[10337]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 09:54.09 | Stephnie | no one is answering |
| 09:55.41 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 09:56.19 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 09:57.27 | queuetue | Hello. how would I send someone "straight to the beep"? IE, voicemail without hearing the Digium Voice's instructions first? |
| 09:59.56 | queuetue | Append the mailbox name with "s"? That simple? |
| 09:59.59 | tparcina | sheck app(voicemail), there is somthing you need to at at the end (a or s, i'm not sure |
| 10:00.18 | tparcina | yes, that simple it is |
| 10:04.41 | Trimor | has any body any information regarding WC_FXO |
| 10:05.32 | Stephnie | can I ask my question again? |
| 10:05.37 | Stephnie | Jun 7 14:30:58 WARNING[10337]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 10:06.25 | viperdude | Stephnie: looks like what ever is trying to communicate with asterisk using GSM is using a incompatible GSM codec |
| 10:06.46 | Stephnie | everything was fine till yesterday.. |
| 10:06.57 | viperdude | so something changed |
| 10:07.03 | Stephnie | no.... |
| 10:07.11 | Stephnie | GSM codec was working fine.... |
| 10:07.24 | viperdude | is this communicating with a third party ? |
| 10:07.54 | Stephnie | yes dialing out |
| 10:08.15 | viperdude | so maybe the change was at the outbound provider? |
| 10:08.34 | Stephnie | but Asterisk is doing codec conversion .... |
| 10:08.47 | Stephnie | that provider is at ULAW and Softphone is using GSM .. |
| 10:08.58 | Stephnie | GSM - ULAW conversion |
| 10:08.58 | viperdude | yes but there are different versions of GSM codec... if the other party changed the |
| 10:09.10 | viperdude | <PROTECTED> |
| 10:09.39 | Stephnie | ok I tried changing the route.... |
| 10:09.45 | Stephnie | but the same problem with all the routes |
| 10:09.49 | viperdude | route? |
| 10:10.00 | Stephnie | I mean...the Peer which is dialing out |
| 10:10.22 | viperdude | tried another codec? ulaw |
| 10:10.35 | Stephnie | yes...no problem with other codecs...but only with GSM |
| 10:10.57 | viperdude | ok so something is sending GSM data to asterisk that it doesn't like |
| 10:11.40 | Stephnie | I have used 2 Different Softphones... |
| 10:11.51 | Stephnie | these phones were working fine till yesterday.. |
| 10:11.57 | Stephnie | but today...both are not working .. |
| 10:12.21 | viperdude | well something changed but i can't help you other than that |
| 10:12.29 | Stephnie | 6-6-06 date was not good for my box...thats the what I think ;) |
| 10:12.35 | viperdude | lol |
| 10:12.51 | Stephnie | :) |
| 10:13.21 | Stephnie | I have recompiled asterisk... |
| 10:13.23 | Stephnie | but same problem |
| 10:13.29 | Stephnie | now going to reinstall linux... |
| 10:14.37 | tzafrir | reinstall linux? why??? |
| 10:14.58 | Stephnie | I couldnt get this problem resolved .. |
| 10:15.05 | Stephnie | thats the only way I have now |
| 10:15.22 | Stephnie | Jun 7 14:30:58 WARNING[10337]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 10:15.27 | hads|home | I don't think reinstalling your distro will help. |
| 10:16.06 | Stephnie | yeah I think so ..but I dont know the reason for this problem.. |
| 10:16.20 | Stephnie | I have searched wiki & google....but couldnt get any solution |
| 10:17.01 | Stephnie | tried to read codec_gsm.c but it looks like french to me |
| 10:17.22 | hads|home | So your softphone is the one using GSM? Or the provider? |
| 10:17.56 | Stephnie | I have cut out the provider ...now just going to do BACKGOUND with softphone.. |
| 10:18.00 | Stephnie | thats it... |
| 10:18.23 | hads|home | And still the same WARNING? Tried a different softphone? |
| 10:19.02 | Stephnie | yes tried different softphone..but stil the same warning.. |
| 10:19.09 | Stephnie | ok I got to know 1 thing.. |
| 10:19.21 | Trimor | http://www.voip-info.org/wiki/view/X100P+clone <-<tparcina> - herez some info regarding modems |
| 10:19.23 | Stephnie | from softphone to ASteisk....... |
| 10:19.50 | Stephnie | I am doing only exten => 1,1,Background(beep) |
| 10:19.57 | Stephnie | it works....no WARNING |
| 10:20.53 | Stephnie | going to check codec conversion |
| 10:21.55 | tparcina | Trimor, somebody else was asking about modems, i have just give response. it seams that there are some cards (modems) that work, but they have registred problems... |
| 10:24.21 | tparcina | i need to buy conference station - SIP phone in which i can plug several speakers and one (or more) microphones. is there anything like this on market? |
| 10:25.02 | Stephnie | hads|home : Conversion from GSM to anycodec ......thats the problem |
| 10:25.14 | Stephnie | when I do this codec conversion I start getting this WARNING |
| 10:25.54 | viperdude | Stephnie: what does "show translation" on the CLI tell you? |
| 10:27.08 | Stephnie | I have GSM and ULAW...... |
| 10:27.29 | viperdude | and the conversion time? |
| 10:27.32 | Stephnie | GSM to ULAW ...conversion.....Getting WARNING...and show translation says |
| 10:27.42 | *** join/#asterisk alucard064 (n=vircuser@ABayonne-152-1-63-126.w83-193.abo.wanadoo.fr) |
| 10:27.49 | alucard064 | re all |
| 10:28.03 | Stephnie | 2 |
| 10:28.08 | Stephnie | 2 milliseconds |
| 10:28.09 | viperdude | hmm |
| 10:28.42 | viperdude | what version of asterisk? |
| 10:29.12 | Stephnie | Asterisk 1.2.7.1 |
| 10:29.28 | viperdude | try upgrading to 1.2.9 |
| 10:29.49 | Stephnie | ok whats he upgrade command? |
| 10:29.50 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 10:30.00 | viperdude | download the source and recompile |
| 10:30.11 | iceyp | hey guys, anyone know where i can download a whole lot of cisco 7940 ringtones? |
| 10:30.24 | iceyp | There were a whole heap I got off a website once before |
| 10:30.34 | iceyp | but cant find a reference to it on voip-info anywhere |
| 10:30.56 | Stephnie | do I need to download Zaptel and libpri as well or only the asterisk? |
| 10:31.06 | viperdude | took me 5 secs to find 7940 ringtone sites with google |
| 10:31.13 | viperdude | only asterisk |
| 10:31.47 | Stephnie | okey |
| 10:32.50 | iceyp | viperdude can u suggest one please i cant find any :/ |
| 10:33.12 | viperdude | http://www.thecaretakers.net/CMS/content/section/77/204/ |
| 10:33.32 | *** join/#asterisk kiddy (n=achu@59.93.35.232) |
| 10:33.39 | kiddy | hi |
| 10:33.59 | iceyp | thanks viperdude |
| 10:34.40 | kiddy | I can't use the conference call facility with asterisk from outside(PSTN) |
| 10:35.40 | kiddy | When I call like "PhoneNumber+8+extensionNumber" it says number not exists |
| 10:35.48 | kiddy | anybody know what I want to change? |
| 10:36.02 | *** join/#asterisk bastien040 (n=bastien@ABayonne-152-1-63-126.w83-193.abo.wanadoo.fr) |
| 10:36.16 | bastien040 | re all |
| 10:38.08 | alucard064 | yop |
| 10:38.18 | alucard064 | c la fete a la grenouille |
| 10:38.45 | *** join/#asterisk RoyK (n=roy@85.166.58.24) |
| 10:39.47 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 10:40.20 | bastien040 | Hi, |
| 10:40.21 | bastien040 | Does anybody have the signification of this debug information: |
| 10:40.21 | bastien040 | '3 !! Unknown IE 36 (cs6, Unknown Information Element) ?' |
| 10:40.21 | bastien040 | Thanks for reply, |
| 10:40.21 | bastien040 | Regards. |
| 10:41.49 | kiddy | anybody know how to make outside conference call to asterisk extension? |
| 10:42.44 | motu | im trying to have the callee transfer the call, but there is no time to input the extension after pressing #, only the first digit is recognized |
| 10:43.31 | motu | what can I do to prolong the digit timeout time for the callee after pressing #? |
| 10:47.19 | Stephnie | viperdude: Asterisk 1.2.9.1 |
| 10:47.34 | Stephnie | same WARNING.....on Codec Conversion from GSM to anycodec |
| 10:47.50 | *** part/#asterisk downunder33 (n=robert@60.51.217.62) |
| 10:47.51 | Stephnie | I think I should better reinstall distro... |
| 10:49.51 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 10:50.54 | swytch | when using the action "UserEvent" with the ^ character to brak lines (as documented), i just get the ^ character instead of CRLF in the output of the Manager API. what do i do wrong? |
| 11:02.51 | *** join/#asterisk lorinc (n=ang@caracas-1593.adsl.interware.hu) |
| 11:05.13 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 11:13.57 | Stephnie | I am getting this error while "make clean" zaptel |
| 11:13.58 | Stephnie | make: *** /lib/modules/2.6.11-1.1369_FC4/build: No such file or directory. Stop. |
| 11:13.58 | Stephnie | make: *** [clean] Error 2 |
| 11:14.33 | Stephnie | its a new FC4 Distro...do I need to install kernel or something like that? |
| 11:15.00 | *** join/#asterisk ness (n=Tom@pppin-5d-b6.pop-kaltenengers.rz-online.NET) |
| 11:15.55 | ness | hi, can you please have a look at http://pastebin.com/764636? |
| 11:16.12 | ness | it's something like a "session dump" |
| 11:16.19 | *** join/#asterisk rleyba (n=root@60-241-132-21.tpgi.com.au) |
| 11:17.09 | ness | it looks like a call is coming in from willich, going through us and back to willich |
| 11:17.38 | ness | but actually someone called willich from here (through lotus notes) |
| 11:18.03 | key2 | Stephnie: no |
| 11:18.08 | rleyba | hi there...just got myself a new IP phone.....on the asterisk server, I keep getting username/auth name mismatch but I am SURE I have the name and secret set correctly. would appreciate any help |
| 11:18.18 | Stephnie | key2: then what about the error? |
| 11:18.26 | Stephnie | make: *** /lib/modules/2.6.11-1.1369_FC4/build: No such file or directory. Stop. |
| 11:18.35 | key2 | Stephnie: first, do you need zaptel ? |
| 11:18.50 | key2 | Stephnie: do you have any interface that uses zaptel ? |
| 11:18.54 | ness | I guess it is somewhat related to the "Application: Bridged Call" given by show channel IAX2/iax_dialout_willich/3 |
| 11:18.58 | Stephnie | no.... |
| 11:19.04 | ness | what does this mean? |
| 11:19.25 | ness | *afk* |
| 11:21.24 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 11:22.30 | swytch | quit "evil bit" |
| 11:22.42 | mr_horsepower | morning all |
| 11:25.26 | *** join/#asterisk drew___ (n=foo@zux221-156-100.adsl.green.ch) |
| 11:27.29 | drew___ | i am trying to get 2 Wildcard X100P's to work in the same dell box... somehow it only works with one card at a time... any ideas on what i could do about it? |
| 11:29.21 | mr_horsepower | drew___: there are a couple of problems with dell machines. |
| 11:29.42 | mr_horsepower | drew___: do you have a clone of x100p or a real x100p? |
| 11:29.49 | *** join/#asterisk kiddy (n=achu@59.93.39.43) |
| 11:29.53 | mr_horsepower | i have some problems like that with clones. |
| 11:30.13 | kiddy | how can I connect to asterisk conference from external call? |
| 11:33.11 | drew___ | some clone that identifys itself as the real one |
| 11:34.00 | drew___ | could it be a interrupt issue? |
| 11:34.34 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 11:35.07 | tzafrir | drew___, one at a time == ? do you get two spans under /proc/zaptel/ ? |
| 11:35.31 | drew___ | tzafrir - hang on.. |
| 11:36.17 | *** join/#asterisk assert_true (n=anil@59.176.23.160) |
| 11:36.27 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:36.44 | mr_horsepower | drew___: clones does not work just like the one original x100p. |
| 11:36.59 | mr_horsepower | i have exacly the same issues, they dont even appear in lspci. |
| 11:37.06 | mr_horsepower | they are not detected |
| 11:37.09 | tzafrir | mr_horsepower, there are no "originals" for quite a while. |
| 11:37.24 | tzafrir | what do call "clones" and what originals? |
| 11:37.32 | mr_horsepower | yes i know, but i dont know when you bought yours. |
| 11:37.45 | drew___ | i could only find "clones" on ebay and i dont have the money for the newer digium cards... |
| 11:38.00 | drew___ | if i have a single card in the box - it works... |
| 11:38.02 | mr_horsepower | http://www.x100p.com/ |
| 11:38.20 | drew___ | both cards work in both slots, aslong as only one card is in the box |
| 11:38.35 | mr_horsepower | i'm waiting for 2 of these, to see if they work, i have about 10 clones here, and none of them work callerid and a couple of another things. |
| 11:38.51 | mr_horsepower | they crash the machine, and some weird problems with irq and stuff like that. |
| 11:39.22 | mr_horsepower | drew___: try to assign, specific irq to each pci slot. |
| 11:39.32 | mr_horsepower | i dont know if dell bios, suport it. |
| 11:39.44 | tzafrir | drew___, again, what exactly do you call "doesn't work"? doesn't shows up in lspci? not identified when module is nnloaded (no span generated , no file under /proc/zaptel) |
| 11:40.52 | drew___ | ill have to check that... |
| 11:45.28 | ness | re |
| 11:45.34 | ness | any idea? |
| 11:45.46 | ness | (wrt http://pastebin.com/764636) |
| 11:47.49 | *** part/#asterisk assert_true (n=anil@59.176.23.160) |
| 11:50.44 | kmilitzer | ness: What exactley is you problem there? |
| 11:50.51 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 11:51.43 | ness | kmilitzer: well, I wonder what "bridged call" means |
| 11:51.54 | ness | I described it above: |
| 11:52.32 | ness | <ness> it looks like a call is coming in from willich, going through us and back to willich |
| 11:52.32 | ness | <ness> but actually someone called willich from here (through lotus notes) |
| 11:52.33 | kmilitzer | ness: I would say it is a "connected" call ... don't you have any asterisk logs? |
| 11:52.49 | ness | sure I have, but not here |
| 11:53.15 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 11:53.16 | kmilitzer | ness: I am not very with the manager interface and the logs of it ... IMHO the best way to see what happend is to take a look at an asterisk verbose log |
| 11:53.31 | ness | I'll bring them from the office this evening |
| 11:53.36 | kmilitzer | s/not very with/not very familiar with/ |
| 11:54.23 | ness | I'm just asking because it makes our status monitor display shit |
| 11:55.25 | kmilitzer | ness: What does your setup look like? Asterisk with iax clients an TExxxP as PSTN Interface? |
| 11:57.06 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 11:57.14 | ness | kmilitzer: I'm not too familiar with it, but we have sip clients and * at offices in different town communicate via iax |
| 11:57.22 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 11:57.26 | fourcheeze | is it possible to renegotiate codec after the start of a call? |
| 11:57.36 | drew___ | ok i rebuild the box with both cards - only one card shows in /proc/zaptel |
| 11:58.19 | kmilitzer | ness: And you somehow use the manager interface to get a overview of which calls are terminated where, etc? |
| 11:58.22 | coppice | fourcheeze: yes. that is how T.38 works |
| 11:58.50 | fourcheeze | coppice: well that's interesting because I'm talking faxes here |
| 11:58.57 | fourcheeze | however not t.38 |
| 11:58.59 | ness | kmilitzer: I use the manager interface to see who is calling |
| 11:59.18 | ness | I'm most interested in knowing who is locally not available |
| 11:59.32 | ness | (because phoning) |
| 11:59.35 | fourcheeze | what I want to do is when I get a call coming in on a particular extension on g729 to renegotiate to g711 |
| 11:59.35 | kmilitzer | ness: OK, I am getting to understand your problem I think. So if a call comes in via IAX localy it is displayed wrong in your interface? |
| 11:59.51 | fourcheeze | then dial out to a fax machine |
| 12:00.00 | fourcheeze | fax machine is on POTS |
| 12:00.19 | fourcheeze | I understand that g711 often works for sending faxes |
| 12:00.24 | fourcheeze | but someone tell me if I'm wrong |
| 12:01.00 | ness | kmilitzer: no, it works most of the time. It some ugly way of calling out from lotus notes that is displayed wrongly |
| 12:01.00 | fourcheeze | coppice: any idea how to do that? |
| 12:01.01 | ness | gtg |
| 12:01.22 | *** part/#asterisk ness (n=Tom@pppin-5d-b6.pop-kaltenengers.rz-online.NET) |
| 12:01.31 | coppice | its done with a reinvite, but don't expect G.711 to work for FAX, except by fluke |
| 12:02.05 | kmilitzer | Just leaves when someone's going to help him ... tss |
| 12:02.45 | fourcheeze | coppice: hmm why is that - I thought it would be high enough quality |
| 12:03.01 | fourcheeze | how does one route to a fax machine on POTS? |
| 12:03.28 | drew___ | on init of zaptel there is a error "Running ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address(6)" |
| 12:03.30 | coppice | http://www.soft-switch.org/foip.html |
| 12:07.38 | fourcheeze | coppice: so basically it's all pretty crap right now |
| 12:08.02 | drew___ | anybody have any ideas on those wildcards? |
| 12:08.19 | coppice | there are reasons why T.38 exists :-) |
| 12:09.56 | fourcheeze | I understand t.38 is fairly bleeding edge in asterisk though |
| 12:10.10 | drray | it bleeds less than faxing does |
| 12:10.17 | fourcheeze | it doesn't seem to help me talk to a phone on the end of an analogue line either |
| 12:10.27 | coppice | a few people are lucky enough to have everything working in their favour, and do actually FAX over their LAN with some success. be very skeptical of any claims to "fine" or "perfect" though. the typical person saying those things probably had one successful page one day :-) |
| 12:11.13 | coppice | the T.38 going into * right now is just passthrough. I have termination and gateway basically working, but my code is GPL, so it can't go into * |
| 12:11.35 | coppice | or rather into *'s SVN |
| 12:11.46 | fourcheeze | coppice: what do you use on the client end? |
| 12:12.00 | drray | fax to fax over PSTN with no pbx/asterisk involved is not 100% |
| 12:12.20 | coppice | what do you mean by client end |
| 12:12.42 | coppice | drray: it should be well over 99%, unless you have really shitty lines |
| 12:13.09 | drray | or crappy old fax machines at the other end |
| 12:14.25 | coppice | i can send hundreds of FAX pages across town without a single bit error in the images |
| 12:18.01 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:21.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:30.24 | *** join/#asterisk phpboy (n=shane@196.26.21.106) |
| 12:31.11 | phpboy | hey all, I have a TDM100 Card with 2 FXO modules... when I run ztcfg -vvvvv it tells me 0 channels configured... could this mean that my card is broken? |
| 12:32.54 | *** join/#asterisk ChrisDE4 (n=ChrisDE@tmo-031-222.customers.d1-online.com) |
| 12:34.16 | *** join/#asterisk littlejohn (n=little@host12-254.pool8717.interbusiness.it) |
| 12:34.53 | coppice | i've got an e-mail that says its from god. i'm a little suspicious of its genuineness |
| 12:35.43 | jsharp | No. Its real. I really did send it. |
| 12:35.57 | *** join/#asterisk Kis (i=vlad@p5080DF3F.dip.t-dialin.net) |
| 12:36.12 | coppice | seems god's grammar is a bit iffy |
| 12:38.03 | ChrisDE4 | hi. have one question: I experienced the problem that a phone registeres at asterisk and sets a timeout of 180. After some time (most likely 180) the phone tries to reregister, so it sends a "REGISTER" to asterisk. Asterisk answeres "Trying" and then "401 Unauthorized".... after resetting the phone it registeres normally. ... This only seems to happen when having stun enabled at the phone. |
| 12:38.24 | [TK]D-Fender | phpboy : It likely means you didn't configure your zaptel.conf file properly |
| 12:39.55 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 12:40.17 | InHisName | userlist |
| 12:41.04 | SheriF_WorK | i want to test something on my asterisk anyone knows any Free SIP providers ? i want to try asterisk as a client. |
| 12:41.04 | InHisName | \ |
| 12:41.26 | phpboy | [TK]D-Fender: ah, so I have to config it manually first? |
| 12:41.35 | [TK]D-Fender | phpboy : Yes. |
| 12:41.37 | phpboy | I thought that ztcfg handles that for me? |
| 12:41.42 | [TK]D-Fender | SheriF_WorK : Sign up with FWD. |
| 12:41.49 | SheriF_WorK | thx ;-) |
| 12:41.55 | [TK]D-Fender | phpboy : ztcfg TESTS zaptel.conf settings |
| 12:42.30 | phpboy | I see |
| 12:42.43 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 12:43.13 | tparcina | does anybody have cisco sereen picture for cisco 7905? |
| 12:43.26 | tparcina | or, how can I get back cisco image on phone? |
| 12:43.45 | *** part/#asterisk ChrisDE4 (n=ChrisDE@tmo-031-222.customers.d1-online.com) |
| 12:44.55 | InHisName | I am connecting spa2000 to eth1 of * router box and when one ext calls other, rings, moh, answer, neither hears other. |
| 12:45.10 | *** join/#asterisk v_farmer (i=rvilleri@xs6.xs4all.nl) |
| 12:46.21 | [TK]D-Fender | InHisName : Describe the other extension. |
| 12:47.14 | InHisName | both connected to spa2000 thusly: internet-cablemodem-eth0-asterisk/router-eth1-spa-2 extns. |
| 12:47.54 | [TK]D-Fender | ok, so 1 port on the SPA is calling the other port? |
| 12:48.06 | InHisName | yes, via * |
| 12:48.32 | [TK]D-Fender | you can dial from A > B and B < A and they both ring? |
| 12:49.08 | InHisName | both: ring, moh, answer, BUT no speach heard by either one / direction. |
| 12:49.18 | InHisName | Identical setups in sip.conf |
| 12:49.21 | key2 | [TK]D-Fender: do you know why when someone gets into the queue, after a position announcement, there we can't hear the musiconhold anymore ? |
| 12:49.24 | CoaxD | inhisname: nat=yes |
| 12:49.31 | CoaxD | inhisname: (in sip.conf) |
| 12:49.34 | InHisName | yes |
| 12:49.51 | *** join/#asterisk ToyMan (n=stuq@74-32-59-52.dsl1.mdl.ny.frontiernet.net) |
| 12:49.58 | [TK]D-Fender | CoaxD : No need... its direct plugged to the * box |
| 12:49.58 | CoaxD | inhisname: If you've got NATs on both sides of that link, it'll take a whole lot to get sip working |
| 12:50.08 | CoaxD | tk: Yes need. one side is NATted |
| 12:50.22 | [TK]D-Fender | key2 : Never heard of that before... |
| 12:50.29 | [TK]D-Fender | InHisName : Pastebin your sip.conf |
| 12:50.30 | [TK]D-Fender | ~pb |
| 12:50.38 | jbot | i guess pb is aka pastebin |
| 12:50.38 | key2 | [TK]D-Fender: how could I debug ? |
| 12:50.47 | [TK]D-Fender | CoaxD : Not by his description it isn't. read again |
| 12:50.50 | InHisName | how do i pastebin the sip.conf ? |
| 12:51.00 | CoaxD | +internet-cablemodem-eth0-asterisk/router-eth1-spa-2 extns. |
| 12:51.19 | CoaxD | if 'router' == 'nat', it'll need it |
| 12:51.32 | [TK]D-Fender | CoaxD : ETH1 does not have INTERNET on it anywhere, does it? |
| 12:51.41 | [TK]D-Fender | CoaxD : That'd do it... |
| 12:52.02 | [TK]D-Fender | InHisName : Describ the hardware hanging off ETH1 please... |
| 12:52.15 | InHisName | I have nat=yes in extn defin [1021] only. All "internal" extns do not have nat. |
| 12:52.16 | CoaxD | you know, nat=yes works even if there's no nat |
| 12:52.18 | [TK]D-Fender | InHisName : Specifically your use of the term "router" |
| 12:52.52 | CoaxD | this is why iax2 is better than sip. by far. |
| 12:52.56 | InHisName | Hardware: eth1-16port switch [4 computers, 1 spa, more to come later] |
| 12:53.29 | CoaxD | inhisname: Lets try again. describe 'router' |
| 12:53.30 | InHisName | Router as in network router |
| 12:53.58 | [TK]D-Fender | InHisName : so no "router". you're just running DHCPD for IP's and thats it? |
| 12:54.09 | InHisName | I am running linux box with router / qos functions along with asterisk on it. |
| 12:54.16 | CoaxD | inhisname: Router as in linksys? or router as in cisco? |
| 12:54.37 | [TK]D-Fender | InHisName : Or do you just mean a dumb SWITCH? |
| 12:54.47 | InHisName | router as in 350 mhz cpu with linux runninng eth0 & eth1 |
| 12:54.48 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.48.Dial1.SanJose1.Level3.net) |
| 12:55.53 | InHisName | internet - cablemodem - [eth0 + router + qos + linux + asterisk + eth1] - switch - ata |
| 12:56.52 | [TK]D-Fender | InHisName : Ok, so nat = irrelevent. Can the extensions each use the echo test independantly? |
| 12:57.18 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34) |
| 12:57.26 | tparcina | come on guys, somebody has to know how to reset logo on cisco 7905 |
| 12:57.47 | InHisName | not sure, how do I echo test on * only and not fwd ? |
| 12:57.48 | tparcina | or, if somebody has the original cisco picture... |
| 12:57.55 | ManxPower | I think you can upgrade the LOGO to Pascal |
| 12:57.55 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.48.Dial1.SanJose1.Level3.net) |
| 12:57.59 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 12:58.40 | [TK]D-Fender | ManxPower : Hey... baby-steps! BASIC first! ;) |
| 12:58.49 | tparcina | ManxPower, is that some realy old computer language that you are only familiar with? :== |
| 12:58.50 | SheriF_WorK | sorry guys what is the test number for FWD ? the page not exists on fwd website |
| 12:59.01 | InHisName | extn 613 gives fast busy, may be due to _61xxxxxx dial code. |
| 12:59.29 | [TK]D-Fender | tparcina : No, it was a great learning tool for its day and you needn't be that old to know it... |
| 12:59.34 | InHisName | FWD greeting line 55555 |
| 12:59.52 | [TK]D-Fender | InHisName : Make an exten for the echo test like in the sample file |
| 13:00.24 | [TK]D-Fender | tparcina : If you want a language only 1 person in the world actually knows I can give you a copy later :) |
| 13:00.34 | InHisName | I'll check for it. Is there an application to call like echo() or such ? |
| 13:01.20 | [TK]D-Fender | InHisName : Yes |
| 13:02.04 | tparcina | Fender, thank you but I don't think i'll need that :) |
| 13:02.15 | SheriF_WorK | InHisName: someone picked the call up :P |
| 13:02.56 | InHisName | Here is what I put into extensions.conf, trying out now. |
| 13:04.10 | InHisName | <PROTECTED> |
| 13:04.10 | InHisName | <PROTECTED> |
| 13:04.10 | InHisName | <PROTECTED> |
| 13:04.21 | InHisName | No echo occurred |
| 13:05.08 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:05.48 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 13:05.52 | coppice | let's see. is echo free good or bad today? :-) |
| 13:06.32 | Pj_ | lol |
| 13:06.48 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:06.56 | [TK]D-Fender | InHisName : You hear dial-tone at least right? |
| 13:07.07 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-142.usadatanet.com) |
| 13:07.43 | InHisName | yes, got dial tone, dialed 618, quiet line no echoing. |
| 13:08.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:08.56 | [TK]D-Fender | in, pastebin your sip.conf |
| 13:08.57 | [TK]D-Fender | ~pb |
| 13:08.59 | jbot | somebody said pb was aka pastebin |
| 13:09.04 | [TK]D-Fender | eek |
| 13:09.13 | [TK]D-Fender | What dumbass changed the bot!? |
| 13:09.22 | [TK]D-Fender | InHisName : www.pastebin.com |
| 13:09.32 | kmilitzer | ~pastebin |
| 13:09.33 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 13:09.56 | InHisName | I'll go there and see what to do. |
| 13:10.06 | feld_ | Is there any reason to have ztdummy loaded when you have an actual TDM400P? I thought that was for only if you DIDNT have one?! |
| 13:11.40 | *** join/#asterisk sandos (n=sandos@83.233.97.253) |
| 13:12.05 | key2 | [TK]D-Fender: I get that |
| 13:12.06 | key2 | <PROTECTED> |
| 13:12.19 | key2 | <PROTECTED> |
| 13:12.19 | key2 | <PROTECTED> |
| 13:12.19 | key2 | <PROTECTED> |
| 13:12.26 | key2 | but it never plays the musiconhold again |
| 13:12.27 | key2 | :( |
| 13:14.10 | *** join/#asterisk frk2 (n=faraz@202.5.145.13) |
| 13:14.20 | frk2 | guys whats going onnnnn |
| 13:14.43 | frk2 | Whats a good ATA to use? |
| 13:15.01 | frk2 | im thinking of deploying ATAs whereever low cost IP phones are required |
| 13:15.04 | ManxPower | SIPura |
| 13:15.08 | frk2 | i need something that just WORKS- does not hang, crash or reboot |
| 13:15.36 | key2 | ManxPower: do you have an idea about my queue problem ? |
| 13:15.37 | drray | <PROTECTED> |
| 13:15.46 | [TK]D-Fender | frk2 : Linksys SPA-2002 |
| 13:15.58 | InHisName | OK, I pasted sip.conf with pwd xxxx etc. |
| 13:16.09 | [TK]D-Fender | InHisName : link pease.... |
| 13:16.12 | tzafrir | ~pb |
| 13:16.13 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 13:16.33 | frk2 | i heard reports of it hanging |
| 13:16.34 | frk2 | no? |
| 13:16.42 | ManxPower | key2, I almost never use queues. |
| 13:16.46 | ManxPower | Hate them. |
| 13:17.02 | InHisName | Hmmhesays, OH link here you go: http://pastebin.com/765108 |
| 13:17.03 | drray | I've been driving payphones with IAXY's for about a year, and they are bulletproof. the only annoyance I have is the bright blue LED but that is fixable with tape |
| 13:17.26 | Ahrimanes | haha |
| 13:17.34 | Ahrimanes | the wellknown ducttape-bugfix :) |
| 13:17.37 | frk2 | yeah ive heard good things about iaxys |
| 13:17.50 | [TK]D-Fender | Ahrimanes : VCR configuration tool of choice! |
| 13:17.51 | coppice | there should be legislation to control the brightness of blue LEDs |
| 13:17.58 | frk2 | hahhah |
| 13:18.05 | frk2 | so spa 2002 or iaxy? |
| 13:18.13 | Ahrimanes | [TK]D-Fender: :P |
| 13:18.19 | drray | I have my iaxy's behind a linux gateway so I can remote config them from remote |
| 13:18.27 | frk2 | i am not gonna go chinese on this shit this time around |
| 13:18.49 | frk2 | tkd--- dude whatsss uppp |
| 13:18.51 | Ahrimanes | coppice: but blue leds are 31337... |
| 13:19.03 | coppice | you can go chinese, if you want. some of us can read it :-) |
| 13:19.33 | [TK]D-Fender | frk2 :IAXY = IAX = only *. SPA-2002 = SIP = any IP-PBX really. IAXY = 1 pot, SPA = 2 port, IAXY = no web interface, SPA = web interface, SPA is cheaper. Easy choice to me... |
| 13:19.33 | frk2 | okay |
| 13:19.33 | frk2 | spa then |
| 13:19.41 | frk2 | done |
| 13:19.52 | frk2 | am about to order 250 of these bad boys |
| 13:20.01 | [TK]D-Fender | frk2 : Reality time : EVERYTHING is made in China.... you just want BETTER Chinese crap :) |
| 13:20.08 | [TK]D-Fender | frk2 : 250?! |
| 13:20.14 | frk2 | does it have a builtr nin switch (i know i can google this) :) |
| 13:20.16 | coppice | but why are the blue LEDs always *so* much brighter than any of the other LEDs on a piece of equipment? there's a blue LED on a computer here that's so damn bright its annoying across the room |
| 13:20.30 | frk2 | yeah man. |
| 13:20.41 | frk2 | medium organization |
| 13:20.52 | frk2 | whats wrong with 250? |
| 13:20.52 | Ahrimanes | coppice: blue light travels faster than other colors? |
| 13:20.59 | [TK]D-Fender | coppice : Well on the colour scale blue is "hotter" that white when you think of stars.... |
| 13:21.12 | frk2 | not on the same network dude |
| 13:21.21 | frk2 | maybe 80 max on one network |
| 13:21.42 | [TK]D-Fender | frk2 : You do NOT use that many SPA's for a large install like that you use mass gateways like AudioCodes/Mediatrix in a rack frame... |
| 13:21.57 | coppice | why don't black bodies radiate black? :-\ |
| 13:21.59 | [TK]D-Fender | frk2 : you need 500 extensions? |
| 13:22.03 | *** join/#asterisk asda13123sd (n=mitka@62.76.244.194) |
| 13:22.10 | asda13123sd | hi |
| 13:22.20 | frk2 | well |
| 13:22.27 | frk2 | at their HO its more like 80 extensions |
| 13:22.28 | [TK]D-Fender | coppice : Master of DSP's / Failure of physics :) |
| 13:22.34 | coppice | a building full of SPAs appeals to me. its really wacky :-) |
| 13:22.34 | frk2 | putting 20 polycom phones |
| 13:22.44 | frk2 | so 60 "cheap" phones |
| 13:23.01 | frk2 | how else do i go cheap? |
| 13:23.24 | [TK]D-Fender | frk2 : get a couple of AudioCodes MP-124 FXSgateways then, not SPA's.... just about the same cost-effectiveness in a much more manageable setup |
| 13:23.26 | asda13123sd | could someone please suggest how to implement callback system using asterisk |
| 13:23.46 | [TK]D-Fender | frk2 : You already have the phones? |
| 13:23.57 | [TK]D-Fender | frk2 : (the analog ones that is) |
| 13:24.39 | [TK]D-Fender | YES! My blade is in town on delivery! |
| 13:24.53 | [TK]D-Fender | dance even! |
| 13:25.12 | frk2 | oh sure |
| 13:25.27 | frk2 | if phone lines were laid out.. I would install gateways or even rhino channel banks |
| 13:25.43 | frk2 | some smaller offices DONT have phone lines laid out |
| 13:25.50 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 13:25.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:25.57 | asda13123sd | could someone please suggest how to implement callback system using asterisk |
| 13:26.06 | frk2 | for the 60 analog phones I would just shove a channel bank or a gateway |
| 13:26.18 | frk2 | my expereince has been VERY good with Rhino's |
| 13:26.29 | [TK]D-Fender | frk2 : So you're going to put the SPA's at the users desk and wire up extra RJ45's for them, AND buy the phones? |
| 13:26.54 | asda13123sd | hi |
| 13:26.58 | frk2 | the SPA dont got a internal switch????? |
| 13:27.02 | [TK]D-Fender | frk2 : Channel bank isn't as cost effective as a 24-port SIP gateway anymore.... |
| 13:27.06 | [TK]D-Fender | frk2 : NOPE |
| 13:27.10 | frk2 | fuck me |
| 13:27.21 | frk2 | okay that just screws everything |
| 13:27.21 | [TK]D-Fender | frk2 : Not likely :/ |
| 13:27.32 | frk2 | haha |
| 13:27.35 | [TK]D-Fender | frk2 : Just get them low end Polycom phones! |
| 13:27.46 | [TK]D-Fender | frk2 : Spend a little, get a lot! |
| 13:27.48 | frk2 | lemme search these gateways |
| 13:28.01 | frk2 | dude thats TOO much spending :( |
| 13:28.03 | frk2 | they'll screw me |
| 13:28.05 | frk2 | literally |
| 13:28.11 | frk2 | i totally convinced them to go voip :) |
| 13:28.13 | [TK]D-Fender | frk2 : If they don't have the wiring, you're better off just buying them an IP phone with the switch built in. |
| 13:28.29 | [TK]D-Fender | The gateway IS Voip :) |
| 13:28.33 | stephane_ | re |
| 13:29.03 | [TK]D-Fender | asda13123sd : Lookup "call files" on the WIKI for some inspiration. |
| 13:29.16 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 13:29.25 | frk2 | I know |
| 13:29.35 | frk2 | crap |
| 13:29.49 | frk2 | I know the gateawy is voip-- what do you mean? :) |
| 13:29.51 | frk2 | okay |
| 13:30.13 | frk2 | audiocodecs MP 124 definitely rocks the casbah |
| 13:30.20 | frk2 | for $1500 thats better than a freaking channel bank |
| 13:31.07 | InHisName | [TK]D-Fender I assume you are studying the sip.conf file between conversations. Is cutting and pasteing the only way to avoid hand typing the nicks in a msg ? |
| 13:31.11 | techman97_andy | Fender recommended those to me too - he said they were great...strange to setup the first time, but easier the 2nd. |
| 13:31.42 | [TK]D-Fender | frk2 : Thats what I was saying.... |
| 13:32.06 | [TK]D-Fender | InHisName : I asked you for the LINK to your posting, but never saw you copy it here.... |
| 13:32.48 | InHisName | Try again: Hmm, OH link here you go: http://pastebin.com/765108 |
| 13:33.11 | syle | i find the rhino channel banks better, never a problem |
| 13:33.22 | [TK]D-Fender | frk2, techman97_andy : Correct. The first one is a bit of a boar, but you can export a text config file from the web interface, mod it in 2 seconds, and upload it to your next unit.... |
| 13:33.30 | *** join/#asterisk titoxx69 (n=fobada@neu69-1-82-232-162-41.fbx.proxad.net) |
| 13:34.08 | [TK]D-Fender | syle : No, the Rhino seem pretty good, and I have one myself, but with a T1 card to add, just not as cost efectivae and adds load to your * box directly. SIP passthrough is irrelevent by comparison. |
| 13:35.00 | syle | they give free card now |
| 13:35.06 | titoxx69 | hello :) I have a small problem. I would like to record an interview made by VoIP (SIP), but there's no client that has this feature. So, does Asterisk feature this, and is there any simple tutorial to achieve this ? |
| 13:35.42 | frk2 | yeah man |
| 13:35.43 | frk2 | and plus |
| 13:35.52 | frk2 | the T1 means ports in the server, kinda expensive these days |
| 13:36.02 | titoxx69 | I think Asterisk would be a proxy, but it is far more complex than just a proxy and I'm lost :( |
| 13:36.13 | [TK]D-Fender | InHisName : the phones are in different contexts, get rid of the NAT statements, and DEFAULTIP. then flush out all the commented out junk. |
| 13:36.14 | frk2 | Asterisk is simple dude |
| 13:36.21 | frk2 | try SER if you wanna bang your head against the wall |
| 13:36.43 | asda13123sd | how do i implement callback system |
| 13:36.47 | asda13123sd | in * |
| 13:36.58 | [TK]D-Fender | titoxx69 : Look at features.conf and use "show application dial" at the * CLI for how to use it. Monitor works great all by itself also |
| 13:37.15 | InHisName | [TK]D-Fender maybe I need your nick in line: Hmm, OH link here you go: http://pastebin.com/765108 |
| 13:37.22 | [TK]D-Fender | asda13123sd : I told you where to go look.... what did you discover? |
| 13:37.38 | titoxx69 | [TK]D-Fender, thanks :) |
| 13:37.52 | frk2 | TKD-Fender... its done then.. AudioCodecs gateways it is then |
| 13:38.02 | frk2 | but yeah.. Rhino is awesome |
| 13:38.05 | InHisName | [TK]D-Fender got your note, editing now |
| 13:38.07 | [TK]D-Fender | InHisName : I just told you what to fix... get cleaning, apply the changes, test then come back with a new pastebin if things don't improve. |
| 13:38.14 | frk2 | my most satisfied client is the one using Analog :( :( |
| 13:38.25 | InHisName | \ |
| 13:38.26 | [TK]D-Fender | frk2 : BUT do they have the phones already? |
| 13:38.27 | frk2 | the chinese shit for Voip phones ive recommened are ALL gay |
| 13:38.29 | asda13123sd | thats for me ok |
| 13:38.29 | InHisName | [TK]D-Fender will do |
| 13:38.31 | asda13123sd | thanks |
| 13:38.32 | asda13123sd | hhe |
| 13:38.33 | asda13123sd | sorry |
| 13:38.47 | frk2 | the analog phones? |
| 13:38.50 | frk2 | yes- they have many |
| 13:39.09 | frk2 | but come on man.. even $5-$10 analog phones are pretty descent |
| 13:39.16 | frk2 | compared to even $80 ip phones |
| 13:39.41 | [TK]D-Fender | frk2 : No, they simply WORK..... I still always suggest people to go with GOOD phones whenever possible. |
| 13:40.03 | frk2 | exactly.. DONT HANG.. thats my requirement |
| 13:40.14 | frk2 | apparently its not for chinese voip manufacturers :) |
| 13:40.16 | [TK]D-Fender | frk2 : If they aren't wired for it already you may lose a lot of the difference in costt right there.... |
| 13:40.27 | frk2 | they are 90% wired |
| 13:40.38 | frk2 | analog phone acquisition costs are not that high |
| 13:40.43 | coppice | $10 is very expensive for an analogue phone. that's caller ID, and a big LCD display pricing |
| 13:40.59 | frk2 | but yeah what im thinking is if I order 250 polycom phones.. i might get a good price discount. |
| 13:41.04 | [TK]D-Fender | frk2 : Ok, well if you need analog, mass gateways is the way to go, otherwise there's noone I can suggest over Polycom for IP phones... |
| 13:41.57 | frk2 | man being a consultant is SO much more than being a simple IT guy |
| 13:41.59 | frk2 | damnit |
| 13:42.14 | RaYmAn-Bx | you also get paid more (hopefully) =P |
| 13:42.35 | InHisName | <PROTECTED> |
| 13:42.35 | InHisName | <PROTECTED> |
| 13:42.35 | InHisName | <PROTECTED> |
| 13:42.35 | InHisName | <PROTECTED> |
| 13:42.35 | InHisName | <PROTECTED> |
| 13:42.36 | InHisName | <PROTECTED> |
| 13:42.38 | InHisName | <PROTECTED> |
| 13:42.58 | InHisName | will be inputing pastebin soon sip.conf |
| 13:43.32 | *** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
| 13:43.35 | [TK]D-Fender | InHisName : do NOT paste like that here again please... |
| 13:43.49 | frk2 | or HE will come crashing down on you |
| 13:45.26 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 13:45.41 | frk2 | coppice - you are right.. but i need to give atleast THAT for the average user to know its a VOIP system :) |
| 13:46.14 | InHisName | http://pastebin.com/765166 |
| 13:46.24 | [TK]D-Fender | frk2 : A "GE" branded home phone with call-waiting beeps and RJ11 jacks hardly feels like VoIP :) |
| 13:46.25 | coppice | if POE were sensibly priced, using low end VoIP phones would be a no-brainer |
| 13:47.21 | [TK]D-Fender | InHisName : Those are not the default ports... sure everything matches on the ATA? |
| 13:47.48 | InHisName | default ports ? |
| 13:47.50 | [TK]D-Fender | coppice : Its getting better. IP430 does it integrated cheaper than the IP 501 without it at all. |
| 13:48.10 | [TK]D-Fender | InHisName : the SIP port #'s in your phone setups |
| 13:48.38 | InHisName | 5060 & 61 = default ? |
| 13:48.48 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 13:49.06 | [TK]D-Fender | InHisName : correct... and I'd leave them that way if at all possible |
| 13:49.13 | coppice | D-Fender: the hubs still cost too much |
| 13:49.31 | techman97_andy | coppice: what is *too much*? |
| 13:49.50 | asda13123sd | a lot |
| 13:50.19 | coppice | well, if a 24 port switch could actually do 24 ports of POE and cost <$150, I guess that would be OK |
| 13:50.51 | InHisName | OK, then I'll redo the spa to reflect the dfault port nos 5060 & 5061 and switch the sip.conf to see the same. |
| 13:51.10 | techman97_andy | it's still a new technology...give it a few years to come down in price. |
| 13:51.13 | techman97_andy | =P |
| 13:51.47 | [TK]D-Fender | coppice : $400 = 24 port... not terrible... but Yeah, room for improvement still... |
| 13:52.03 | asda13123sd | whats the best gsm gateay |
| 13:52.05 | asda13123sd | gateway |
| 13:52.14 | coppice | its hard to make small switches cheap, as they must be smart switches. getting the 24 port ones down should be easier |
| 13:52.43 | [TK]D-Fender | coppice : Well the switch I bought last year at $1000 is not under HALF that price now. |
| 13:52.47 | coppice | D-Fender $400 for a 24 port with all of them doing POE? most only allow a few ports of POE |
| 13:53.02 | frk2 | TKD-Fender... thats basically what the 'cheaper' VOIP phones do anyways... AND they hang, AND they echo, AND they have voice breaks.. etc etc |
| 13:53.13 | [TK]D-Fender | coppice : All 24 ports PoE. D-Link DES-1526. I run 2 of them here. |
| 13:53.35 | [TK]D-Fender | frk2 : What is this about what cheaper phones do? |
| 13:53.47 | techman97_andy | NetGear FS726TP - $285 |
| 13:54.36 | [TK]D-Fender | techman97_andy : only 12 of 24 ports are PoE..... |
| 13:54.43 | techman97_andy | on the NetGear? |
| 13:54.44 | [TK]D-Fender | techman97_andy : Not good... |
| 13:54.51 | coppice | D-Fender: hum. 15W a port. it does a proper job. not too bad |
| 13:55.08 | [TK]D-Fender | techman97_andy : CORRECT "Choose to plug in up to 24 Ethernet or Fast Ethernet devices and mix in up to 12 802.3af IP-based devices.Power-over-Ethernet (PoE)," |
| 13:55.25 | frk2 | Dude... according to me.. a voip phone thats say.. less than $120 is shit compared to a analog setup which would end up costing you roughly $60 per port |
| 13:55.28 | [TK]D-Fender | coppice : Works great... I run my all-Polycom IP 600 setup off of them. |
| 13:55.31 | techman97_andy | hmmm - gotta love it when different vendor sites say differnet things |
| 13:55.53 | [TK]D-Fender | frk2 : Polycom IP 301 = $115 and is a great little phone.... |
| 13:56.02 | mishehu | the problem with PoE is that it's still so damn expensive. |
| 13:56.30 | frk2 | it is... but its too damn expensive |
| 13:56.35 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 13:56.47 | frk2 | and besides.. after import.. its gonna end up costing me $160 |
| 13:57.03 | [TK]D-Fender | frk2 : Where are you located? |
| 13:57.11 | frk2 | pakistan :) |
| 13:57.19 | [TK]D-Fender | frk2 : Ok, go analog :) |
| 13:57.24 | frk2 | hahah |
| 13:57.30 | asda13123sd | best is to go analog |
| 13:57.31 | mishehu | analog urdu |
| 13:57.42 | [TK]D-Fender | frk2 : Cost is a factor to be respected sometimes.... |
| 13:57.53 | asda13123sd | i am having the same problem |
| 13:57.56 | asda13123sd | cost!!! |
| 13:57.58 | frk2 | dude... always |
| 13:58.16 | frk2 | otherwise its cheaper to put in a seimens voip exchange |
| 13:58.18 | [TK]D-Fender | I bought Polycom's for my HOME! You cheap bastards ;) |
| 13:58.33 | frk2 | TKD I got Cisco's at my house too :) :) |
| 13:58.35 | frk2 | two of them |
| 13:58.36 | SplasPood | [TK]D-Fender: how many? |
| 13:58.39 | frk2 | home = luxury |
| 13:58.44 | frk2 | office = business |
| 13:58.44 | SplasPood | I've got a 7960 and an IP501 at home |
| 13:58.47 | mishehu | frk2 |
| 13:58.49 | [TK]D-Fender | SplasPood : 2 for now, 3rd on the way when the IP 430 ships |
| 13:58.50 | coppice | D-Fender: that D-Link looks like the genuine article. seems POE is finally getting there |
| 13:58.58 | mishehu | frk2: how many female sex slaves do you have at home? |
| 13:59.02 | SplasPood | [TK]D-Fender: IP430, eh? Haven't looked at those |
| 13:59.04 | asda13123sd | haha |
| 13:59.08 | frk2 | I'm thinking of putting up a small datacentre at my new house |
| 13:59.10 | mishehu | if the answer is "none", then that's not luxury |
| 13:59.17 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 13:59.20 | frk2 | opterons and shit... just for kicks :) |
| 13:59.25 | mosty | is it possible to let two sip phones register with the same sip account, so when somebody calls that extension both phones ring? |
| 13:59.31 | *** join/#asterisk lorinc (n=ang@caracas-4331.adsl.interware.hu) |
| 13:59.33 | [TK]D-Fender | coppice : It is... like I said I've been running them for a year now... dead simple and have some decent features... I didn't need the management really since it is a dedicated LAN for my phones... |
| 13:59.37 | frk2 | only TKD-Fender knows about the polycom IP 4xx |
| 13:59.39 | asda13123sd | anyone know of good gsm gateway |
| 13:59.40 | [TK]D-Fender | mosty : NO |
| 13:59.42 | SplasPood | mosty: not with asterisk, but you can make a Dial() ring two phones |
| 13:59.47 | frk2 | not even polycom knows about them (I called today) :) |
| 13:59.53 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:59.57 | [TK]D-Fender | frk2 : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html |
| 14:00.08 | [TK]D-Fender | frk2 : You just don't know how to ask. |
| 14:00.11 | frk2 | dude im kidding :) take it easy :P |
| 14:00.11 | mishehu | [TK]D-Fender: how much did you spend on your 12 ports of PoE ? |
| 14:00.19 | techman97_andy | [TK]D-Fender - OK...I get what you're talking about with the 12 vs. 24 port PoE thing...how many watts does a IP401 need to operate? |
| 14:00.23 | SplasPood | [TK]D-Fender: whats the diff between that and the 301? |
| 14:00.27 | [TK]D-Fender | mishehu : Mine are 24 port. |
| 14:00.55 | mishehu | [TK]D-Fender: oh, misread, how much did your 24 ports of PoE run you? |
| 14:01.04 | *** join/#asterisk rogger (n=rogger@209.104.162.252) |
| 14:01.05 | [TK]D-Fender | SplasPood : 4 soft-keys, pixel disply, Full duplex speakerphone, full PoE on-board + Brick included. 5 navigation keys instead of 2. |
| 14:01.08 | mosty | i have users that want an extension at home and another at work, they want the phone to act the same but don't want to carry a single phone back and forth, i was looking for a simple way to do that without complicating my dialplan |
| 14:01.16 | [TK]D-Fender | mishehu : At the time $1000, now about $400. |
| 14:01.18 | SplasPood | [TK]D-Fender: hrm, whats the price diff? |
| 14:01.26 | SplasPood | [TK]D-Fender: not much price room between 301 and 501 |
| 14:01.31 | tzanger | ugh |
| 14:01.35 | tzanger | I need a decent wifi sip phone |
| 14:01.44 | [TK]D-Fender | mosty : set up 2 SIP entries and make your Dial command ring both at the same time. |
| 14:01.49 | mishehu | [TK]D-Fender: nice to see that the price is retreating a bit |
| 14:01.57 | tzanger | something with a loud ringer and earpiece, looks and feels like it belongs in a business, not a toy box |
| 14:02.05 | [TK]D-Fender | SplasPood : Atacomm lists at $160. IP 501 = $170. |
| 14:02.08 | coppice | decent wifi sip phone is an oxymoron |
| 14:02.10 | mosty | [TK]D-Fender, that is simple enough but they have to share a voicemailbox |
| 14:02.15 | tzanger | coppice: it sure seems to be hte case :-( |
| 14:02.16 | SplasPood | [TK]D-Fender: yea, so why bother with it? |
| 14:02.26 | [TK]D-Fender | mosty : all dialplan... extremely easy |
| 14:02.32 | SplasPood | mosty: then put the same mailbox= line in both sip.conf entries |
| 14:02.50 | [TK]D-Fender | SplasPood : INTEGRATED PoE. Smaller framer, lower cost. |
| 14:02.50 | coppice | tzanger: without QoS for 802.11 (i.e. 802.11e) it will always be the case |
| 14:02.53 | SplasPood | mosty: and after the Dial() check the status and call Voicemail(mbox@context) in the dialplan |
| 14:03.09 | mosty | and i have a general voicemail section in the dialplan that works for all single mapping ext -> voicemailboxes, but having multiple extensions use the same voicemailbox would break that |
| 14:03.13 | tzanger | coppice: I'm not worried about that right at this point, we're small enough that QoS isn't an issue |
| 14:03.14 | mishehu | frk2: I had an opteron server in my home office, and it's nice except that the supermicro chassis is noisy as hell. it's got something like 6 fans in it. |
| 14:03.20 | mishehu | s/had/have |
| 14:03.25 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:03.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:03.28 | [TK]D-Fender | mosty : Easy to adapt... |
| 14:03.28 | mosty | hrm, maybe if i use accountcode instead of extension |
| 14:03.34 | SplasPood | [TK]D-Fender: think they'll can the 301? |
| 14:03.47 | *** join/#asterisk chrismog (n=chrismog@mog.traxtech.net) |
| 14:03.50 | coppice | tzanger: with 802.11 QoS is *always* an issue. the latency is bad enough at the best of times |
| 14:03.53 | mosty | otherwise i was thinking about using symlinks to force the sharing |
| 14:03.56 | chrismog | Can asterisk do switchhooks? |
| 14:04.01 | [TK]D-Fender | SplasPood : There is still a fair price difference.... hard to say. |
| 14:04.18 | SplasPood | mosty: I suppose you could do that... I'd just rework my macro if I were you |
| 14:04.32 | mosty | spaspood: what do you mean exactly? |
| 14:04.41 | [TK]D-Fender | SplasPood : I thought they might at some point... I was also hoping the IP430's new GUI look would carry to the 501+ but that doesn't appear to be the case.. |
| 14:04.54 | *** join/#asterisk Joshaidan (n=icechat5@thunderbay-voip-4.vianet.ca) |
| 14:05.00 | [TK]D-Fender | SplasPood : From SIP 2.0 that is... |
| 14:05.03 | SplasPood | mosty: you say you can't have 2 diff phones both /w the same mbox cause of some dialing macro you have... I'm saying you should fix that |
| 14:05.31 | [TK]D-Fender | mosty : Pastebin your macro..... |
| 14:05.34 | [TK]D-Fender | ~pb |
| 14:05.36 | 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, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 14:05.51 | SplasPood | [TK]D-Fender: Still tho, if one does not need POE, I see little reason to go for the 430 over the 501 |
| 14:06.01 | mosty | actually i think i figured out a nice way to do it, let me see if that works before i hassle you guys more than i need to |
| 14:06.13 | mishehu | [TK]D-Fender: btw, speaking of macros, if I want to pass Bob Smith as a parameter to a macro, I can't seem to get it so I can send it without having it clipped at Bob or else enclosing it in quotes and the end result is that it becomes ""Bob Smith"" when logged. do you have any suggestions? |
| 14:06.37 | SplasPood | mishehu: hrm thats odd.. I have one I pass James Brinkerhoff to, without quotes, and it seems to be fine |
| 14:06.47 | [TK]D-Fender | SplasPood : Yeah, it worth a few extra bucks for the bigger screen I guess.... |
| 14:06.58 | SplasPood | [TK]D-Fender: and extra line |
| 14:07.07 | [TK]D-Fender | mishehu : dunno.... would have to look... |
| 14:07.11 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:07.22 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 14:07.24 | [TK]D-Fender | SplasPood : Yeah, no argument from me, but few people need 3 regs...... |
| 14:07.34 | SplasPood | yea but for $10 :P |
| 14:07.46 | mishehu | [TK]D-Fender: i.e. &somemacro("Bob Smith"); calls up macro somemacro ( someparam ) { CALLERID(name) = $[ $someparam ]; }; for example (same happens with or without the $[ ] ) |
| 14:08.02 | [TK]D-Fender | SplasPood : Yeah yeah! All valid points... PoE IS the big seller..... |
| 14:08.10 | mishehu | spamborg: I donno, it always clips it at Bob for me. |
| 14:08.10 | SplasPood | Yea |
| 14:08.12 | [TK]D-Fender | mishehu : Sorry.. don't do AEL :) |
| 14:08.16 | SplasPood | hrm.. that reminds me |
| 14:08.29 | mishehu | [TK]D-Fender: aw, but you did help me last time with that includes section ;-) |
| 14:09.03 | SplasPood | Should make sure someone is considering POE for our new office build-out upstairs |
| 14:09.37 | SplasPood | mishehu: Maybe it has to do with your assignment.. try calling Set() directly |
| 14:09.53 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 14:10.10 | SplasPood | or better yet |
| 14:10.13 | SplasPood | set verbose 20 |
| 14:10.14 | SplasPood | or something |
| 14:10.16 | SplasPood | then test |
| 14:10.30 | SplasPood | and see what the macro is setting for someparam based upon the ARGs |
| 14:10.40 | [TK]D-Fender | mishehu : Ok, pastebin it and I'll see if something stands out :) I learned the Include thing on the fly BECAUSE of you... |
| 14:10.55 | *** join/#asterisk apardo (n=apardo@87.217.146.210) |
| 14:11.03 | [TK]D-Fender | SplasPood : ALL CAT5E... RJ11 is DEAD. |
| 14:11.15 | mishehu | [TK]D-Fender: that was exactly it, you really want me to pastebin it? ;-) |
| 14:11.16 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 14:11.19 | [TK]D-Fender | SplasPood : Thats what I did with my new office.... |
| 14:11.19 | SplasPood | [TK]D-Fender: oh well thats a given... I just want them to source a POE switch :) |
| 14:11.30 | [TK]D-Fender | mishehu : Sure, I'll look.... |
| 14:11.46 | mishehu | SplasPood: I thought that I read on the doc page on voip-info that Set() was no longer necessary... |
| 14:11.49 | mishehu | maybe I misread. |
| 14:12.12 | SplasPood | mishehu: Maybe.. I've always had problems /wo Set() in AEL, but I wasn't using the syntax you are (I will now :P ) |
| 14:12.23 | SplasPood | but AEL/Asterisk tends to be a bit... incosistent |
| 14:12.32 | SplasPood | either in actual behavior, or documentation |
| 14:12.50 | SplasPood | mishehu: but do what I suggested instead.. watch the console with verbose set high |
| 14:12.54 | mishehu | SplasPood: in all honesty, I think most of * tends to be inconsistent |
| 14:13.13 | SplasPood | mishehu: I wasn't in the mood for an argument, so I kept it specific :P |
| 14:13.28 | mishehu | SplasPood: apparently there's no argument ;-) |
| 14:13.34 | mishehu | sec, I'll pb it |
| 14:13.42 | SplasPood | mishehu: I've had problems making comments like that in here in the past |
| 14:13.53 | SplasPood | mishehu: very lame, yes. |
| 14:13.59 | InHisName | [TK]D-Fender cleaned up and with 60 & 61 in spa and sip.conf (no sound eithe direction stillhttp://pastebin.com/765218 |
| 14:14.39 | SplasPood | whats with the bindaddr in the sip.conf stanzas? |
| 14:14.43 | tzanger | anyone here have any experience with the linksys wifi phones? or maybe the new siemens one? |
| 14:15.29 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 14:18.53 | mishehu | [TK]D-Fender: http://pastebin.com/765227 |
| 14:19.31 | [TK]D-Fender | InHisName : Says 5068 & 5069 in your pastebin.... |
| 14:19.35 | SplasPood | Set("IAX2/theforcegfi-outbound-5", "CALLERID(name)="Global Freight"") in new stack |
| 14:19.37 | SplasPood | thats not fine? |
| 14:20.08 | SplasPood | yea dude |
| 14:20.12 | SplasPood | I think thats just how it'll be |
| 14:20.18 | SplasPood | as far as the CDR |
| 14:20.27 | InHisName | Opps - trying again |
| 14:20.31 | mishehu | SplasPood: sometimes I feel like if you make any criticism in here some people react as if you are criticizing their childrearing capabilities |
| 14:20.44 | [TK]D-Fender | mishehu : stop calling your macro WITH the quotes.... |
| 14:20.47 | mishehu | SplasPood: clid gets botch |
| 14:21.00 | SplasPood | """James Brinkerhoff" |
| 14:21.00 | SplasPood | " <2122015706>" |
| 14:21.02 | SplasPood | heh |
| 14:21.15 | mishehu | [TK]D-Fender: well, when I drop the quotes it ends up only setting the Global part, it truncates the string |
| 14:21.28 | [TK]D-Fender | mishehu : show me. |
| 14:21.42 | mishehu | [TK]D-Fender: hang on |
| 14:21.44 | [TK]D-Fender | mishehu : And when in doubt just trim them yourself.... |
| 14:22.09 | *** join/#asterisk barros (n=barros@89.106.66.150) |
| 14:22.45 | barros | is there a way to ignore (just when placing call through Zap) the ring back tone comming from the ATA? |
| 14:23.09 | barros | I'm getting two ring back tones.. this is weird! |
| 14:23.24 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:23.26 | SplasPood | turn off local ring in the Dial() ? |
| 14:23.28 | SplasPood | no option r ? |
| 14:24.08 | barros | i was using the r option to ignore the ring back tone comming from the PSTN, but it is not what I want.. |
| 14:24.23 | SplasPood | option r simply supplies LOCAL ring |
| 14:24.28 | SplasPood | I don't believe it does anything else |
| 14:24.30 | barros | I want the pstn tone and not the ata.. what is this local ring? |
| 14:24.46 | SplasPood | asterisk will generate ringing to the calling party |
| 14:24.50 | SplasPood | you want the "dialtone" ? |
| 14:24.54 | SplasPood | is that what you mean? |
| 14:25.04 | barros | i'll explain better |
| 14:25.35 | barros | when I place call through Zap channel, I got the ring back tone that PSNT sends, and asterisk is sending me another ring back tone mixed |
| 14:25.54 | SplasPood | so don't use 'r' in your Dial() |
| 14:26.10 | barros | when I dont use it I get two sounds mixed.. |
| 14:26.20 | barros | when I use it I get only the asterisk generated sound |
| 14:26.21 | SplasPood | when you do? |
| 14:26.26 | SplasPood | hrm |
| 14:26.33 | SplasPood | thats not what I'd expect |
| 14:26.37 | SplasPood | but whats wrong with that then? |
| 14:26.40 | SplasPood | with r |
| 14:26.53 | InHisName | [TK]D-Fender OK here latest results with real default ports. http://pastebin.com/765235 |
| 14:27.04 | barros | well, I didnt like that so much.. especially when I call xome busy number |
| 14:27.19 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 14:27.26 | *** join/#asterisk evilrabbi (i=evilrabb@hi.onlineok.com) |
| 14:27.40 | barros | this second tone comes from my ATA (PAP2).. when I remove it, it works, but I cant get ring back tone when calling internal extensions |
| 14:28.32 | [TK]D-Fender | InHisName : add "nat=no", "canreinvite=no" to them. Think that will fix it... |
| 14:28.46 | InHisName | [TK]D-Fender will do |
| 14:29.11 | SplasPood | barros: hrm... ok well I've run out of blind guesses.. not much experience with the Zap hardware... I term my PRIs elsewhere |
| 14:29.32 | SplasPood | and never used an analog line, actually :) |
| 14:29.48 | barros | that is a PRI one, connected to one E1 |
| 14:29.52 | C4T3l | SplasPood: whom do you use for termiation? |
| 14:30.03 | mishehu | [TK]D-Fender: http://pastebin.com/765245 |
| 14:30.23 | SplasPood | C4T3l: We've some /w broadwing/focal .. But we're looking to move more of our termination elsewhere |
| 14:30.29 | *** join/#asterisk p0wr-t0c (n=powrtoc@81-86-30-78.dsl.pipex.com) |
| 14:30.37 | SplasPood | I've been testing out RNKtelecom |
| 14:30.40 | fholmes | Does anyone here use SugarCRM with Asterisk? |
| 14:30.43 | SplasPood | but so far results have been... mixed |
| 14:30.47 | SplasPood | lots of echo here and there |
| 14:30.54 | feld_ | fholmes: i will be very soon! |
| 14:31.04 | feld_ | we use it for everything else at work |
| 14:31.12 | feld_ | it's a sweet app |
| 14:31.18 | p0wr-t0c | How do you get Realtime Asterisk to support multiple databases within a single server? |
| 14:31.26 | fholmes | feld_: What field are you in? We do call center sales. |
| 14:31.29 | p0wr-t0c | what does the res_mysql.conf need to look like? |
| 14:32.18 | fholmes | feld_: I would really like to see some better integration with Sugar that is for sure. |
| 14:32.19 | SplasPood | C4T3l: Why do you ask? |
| 14:32.20 | barros | SplasPood: this is the same here.. |
| 14:32.37 | barros | wel, I'll put the r options until I found the solution |
| 14:32.43 | barros | thanks |
| 14:32.44 | SplasPood | barros: what is the same? |
| 14:33.01 | barros | 16:43 < SplasPood> but so far results have been... mixed |
| 14:33.01 | barros | 16:43 < SplasPood> lots of echo here and there |
| 14:33.02 | SplasPood | barros: when you encounter BUSY Dial will set DIALSTATUS = to BUSY |
| 14:33.11 | SplasPood | barros: you'd check for that and return the Busy() app |
| 14:33.27 | feld_ | fholmes: i'm a network engineer assigned to setup Asterisk internally and prepare to sell and implement at customer premises providing they buy into it ;) |
| 14:33.28 | barros | no, I put a Hangup just after the Dial cmd |
| 14:33.36 | SplasPood | barros: Well you *would* check for that |
| 14:33.59 | barros | hmm.. isnt a hangup the same?? |
| 14:34.11 | SplasPood | barros: hangup will hang up.. Busy() will play a busy signal |
| 14:34.24 | jarrod | is there a way to have different musiconhold music for diff sites? |
| 14:34.25 | SplasPood | barros: you said your issue was that you wanted to know when the number dialed was busy |
| 14:34.55 | [TK]D-Fender | mishehu : Ok, then just trim the "'s off |
| 14:35.08 | barros | hmmm.. when I hangup I get the busy tone.. it was a mistake here.. |
| 14:35.28 | SplasPood | mishehu: just for kicks, drop the quotes on the number too |
| 14:35.34 | SplasPood | mishehu: and see if it still chops |
| 14:35.49 | SplasPood | barros: what type of phone are you using? |
| 14:36.01 | *** join/#asterisk Modcuts (n=bob@82.133.98.155) |
| 14:36.29 | barros | SplasPood: PAP2 with normal phone |
| 14:36.49 | SplasPood | barros: ahh |
| 14:37.20 | mishehu | gah will people stop calling me? ;-) |
| 14:37.51 | mishehu | [TK]D-Fender: I don't see how that is going ot work. you saw at the bottom that it truncates Global Freight to Global |
| 14:38.17 | SplasPood | mishehu: try what I said |
| 14:38.30 | mishehu | SplasPood: the number it might work for. going to try now. |
| 14:38.48 | SplasPood | try on both |
| 14:38.53 | SplasPood | no quotes |
| 14:38.54 | SplasPood | at all |
| 14:39.02 | [TK]D-Fender | mishehu : I said put the quotes back, and chop them off yourself. |
| 14:39.15 | barros | SplasPood: PAP2 generate the ring back tone.. |
| 14:39.26 | mishehu | [TK]D-Fender: oh, you mean perform some string manipulation |
| 14:39.28 | SplasPood | barros: yea, never used one of those either |
| 14:39.33 | [TK]D-Fender | mishehu : yes |
| 14:39.41 | mishehu | [TK]D-Fender: understood |
| 14:39.46 | p0wr-t0c | Does anyone here know how to use realtime asterisk with multiple databases? I have a line in res_mysql.conf that says 'dbname= asterisk' can I do 'dbname = asterisk,other_rtdb' |
| 14:39.47 | barros | if I blcok the ring back, i couldnt here it when doing local calls |
| 14:39.48 | p0wr-t0c | ?? |
| 14:39.54 | mishehu | SplasPood: sec, goign to see what happens with teh number |
| 14:40.26 | SplasPood | besides, number doesn't need to be quoted anyway |
| 14:41.17 | mishehu | CALLERID(number)=2007630804 when I drop the quotes on the number and send it as 6302598100 |
| 14:42.08 | *** join/#asterisk zzxxcc (n=zzxxcc@221.232.2.27) |
| 14:42.27 | SplasPood | whats it getting set to within tollfree |
| 14:42.36 | mishehu | SplasPood: I shit you not. |
| 14:42.52 | SplasPood | and I'll need to bounce to work in a few |
| 14:43.03 | SplasPood | actually.. like 5min |
| 14:43.19 | mishehu | I need to head into the city |
| 14:43.35 | mishehu | going to be late for work |
| 14:43.38 | mishehu | blargh |
| 14:43.54 | SplasPood | heh |
| 14:43.55 | SplasPood | which city |
| 14:44.00 | SplasPood | thats exactly what I'm saying |
| 14:44.08 | SplasPood | I need to head into NYC, and I'm late for work :P |
| 14:46.25 | LokeshIndian | Hello People, I have a question from you, asterisk-1.2.8 does not have cdr_mysql.conf file in /etc/asterisk/....How i can configure mysql CDR although it has all sorst of other cdr conf files ?? |
| 14:46.45 | LokeshIndian | sorts* |
| 14:47.10 | [TK]D-Fender | LokeshIndian : Thats in asterisk-addons IIRC. |
| 14:47.25 | LokeshIndian | ok Thanks |
| 14:48.12 | *** join/#asterisk sunil (n=sunil@202.54.37.185) |
| 14:48.23 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 14:48.42 | sunil | hi any body tried Trixbox installation |
| 14:48.45 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 14:49.00 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 14:49.23 | [TK]D-Fender | sunil : Quite possibly, but please read the channel topic... |
| 14:49.29 | drew___ | i am trying to get 2 Wildcard X100P's to work in the same dell box... somehow it only works with one card at a time... both cards work if they are installed seperatly - if i put in both i get a error during init of zaptel: "Running ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address(6)" and /proc/zaptel only shows one card - any ideas on what i could do about it? |
| 14:50.33 | mosty | drew: i think i saw something on the wiki about that |
| 14:51.00 | mishehu | drew___: interrupts probably teh issue |
| 14:51.08 | sunil | [TK]D-Fender i have problems running Hudlite, can you help me |
| 14:51.10 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 14:51.21 | drew___ | mishehu - is there a way to check that? |
| 14:51.21 | mishehu | I don't recomment using 2 x100's on the same box in general |
| 14:51.38 | mishehu | drew___: /proc/interrupts for one. make sure htey're not using the same interrupt. |
| 14:51.42 | mishehu | anyway, I need to go |
| 14:52.06 | asda13123sd | is it possible to do international callback using * |
| 14:52.48 | uwe | hello, ive been trying to get Asterisk "CVS-v1-0-01/31/05-06:31:26" configs to work on a new 1.2.4 copy of asterisk ... the files should be backword compatable ... right? it didnt work, and i changed the SetVar to Set and the port to bindport in others, it still didnt work, now the extensions are being rebuilt from scratch ... |
| 14:52.53 | LokeshIndian | Nobody is present on asterisk-addons channel ?? Is anyone have clue about my question ? |
| 14:52.55 | [TK]D-Fender | drew___ : modprobe it 2-3 times |
| 14:53.06 | drew___ | i checked /proc/interrupts - only one wcfxo appears |
| 14:53.09 | *** part/#asterisk zzxxcc (n=zzxxcc@221.232.2.27) |
| 14:53.14 | [TK]D-Fender | sunil : No. Please read the channel topic... |
| 14:53.17 | uwe | isnt there a way to smoothly migrate ? maybe for the future ...just wondering!? |
| 14:53.17 | iq | yo |
| 14:53.24 | feld_ | uwe: u should update to 1.2.9.1 because of the vulnerability ;) |
| 14:53.44 | drew___ | Dfender - modprobe zaptel? |
| 14:53.48 | [TK]D-Fender | uwe : You will need to review your setup and make compatability changes... |
| 14:54.08 | [TK]D-Fender | drew___ : And "modprobe wcfxo" |
| 14:54.22 | uwe | [TK]D-Fender, i would if i rebuilt it from source ... but i didnt , i used xorcom package |
| 14:54.40 | [TK]D-Fender | uwu : 1.0.X generated configs? |
| 14:55.24 | uwe | yes ... with amportal i suppose |
| 14:55.28 | *** join/#asterisk Mike (n=mike@201.138.165.94) |
| 14:55.32 | drew___ | D-Fender - would i need to reboot after that? because it has no effect |
| 14:55.47 | [TK]D-Fender | uwu : EEK... ok, good luck... can't help you there... upgrade your config generator and pray. |
| 14:56.05 | [TK]D-Fender | drew___ : NO, if you do an extra modprobe the card should become visible. |
| 14:56.51 | drew___ | nope - did modprobe zaptel and wcfxo - still only one card |
| 14:56.57 | uwe | [TK]D-Fender, its already being added by hand now ... but it took me 2 days to admit for my self that i cant do it automatically |
| 14:58.03 | [TK]D-Fender | drew___ : ok, pull out the other card and test just 1 solo. Then if it works, put it in the slot the other card was in. if THAT work, then repeat with the other cord in eac slot. |
| 14:58.19 | [TK]D-Fender | uwu : How big a setup are you running? |
| 14:58.36 | drew___ | D-Fender - i checked that - each card works solo |
| 14:58.44 | uwe | [TK]D-Fender, do you suggest that it should be all done without amportal or freepbx, or are you suggesting that freepbx should be used? |
| 14:58.59 | uwe | its not big at all |
| 14:59.09 | [TK]D-Fender | drew___ : And in each slot? |
| 14:59.10 | uwe | about 30 extensions only |
| 14:59.14 | sevard | you're not big at all |
| 14:59.38 | [TK]D-Fender | uwu : I always suggest you do it from scratch..... |
| 15:00.02 | drew___ | each in its own slot - but i can check that as well - gimme couple of min's |
| 15:00.14 | uwe | i c ... |
| 15:00.18 | tzafrir | uwe, source for xorcom rapid packages is availble, BTW |
| 15:00.19 | [TK]D-Fender | drew___ : Just food for thought... have to be thorough |
| 15:01.34 | uwe | tzafrir, ive build it from sources from asterisk.org before, and i have no problem with it ... but i just though using deb packs could make things go faster |
| 15:01.35 | tzafrir | Anyway, there are some minor changes. But if you're migrating now, try a newer version of Asterisk... |
| 15:03.01 | tzafrir | e.g: deb http://rapid.dotsrc.org/rapid unstable main <==== for Sarge |
| 15:04.22 | *** part/#asterisk Joshaidan (n=icechat5@thunderbay-voip-4.vianet.ca) |
| 15:05.08 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:05.19 | *** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com) |
| 15:05.44 | key2 | is it normal that in asterisk 1.2.9 "sip show ...." doesnt work |
| 15:05.53 | InHisName | [TK]D-Fender still no go, tried phones, suspect ? context or bindaddr http://pastebin.com/765337 |
| 15:06.52 | *** join/#asterisk ghenry (i=6bb52493@81-174-212-80.pth-as5.dial.plus.net) |
| 15:07.11 | file | key2: you're about as specific as a banana |
| 15:07.16 | *** part/#asterisk ghenry (i=6bb52493@81-174-212-80.pth-as5.dial.plus.net) |
| 15:07.26 | mosty | is there a standard sound file i can play if somebody dials a non-existant extension? |
| 15:07.27 | [TK]D-Fender | InHisName : Kill the bindaddr. that only belongs in general and should be 0.0.0.0 |
| 15:08.52 | InHisName | [TK]D-Fender will do |
| 15:10.22 | feld_ | i got held up yesterday [TK]D-Fender and never got to continue my project. I have a question for you now. :) |
| 15:10.45 | *** join/#asterisk adker (n=adker@70-100-239-157.br1.glv.ny.frontiernet.net) |
| 15:11.00 | feld_ | I have my TDM400P installed and working. All modules loaded. Everything looks good according to the wiki. Lights are on the ports, lines are connected. |
| 15:11.30 | feld_ | the phone has no dialtone and I am not finding the information I need to make calling out on the outgoing line active. |
| 15:11.40 | [TK]D-Fender | feld_ : Questions are free, answers are $4.95/min ;) |
| 15:11.56 | InHisName | [TK]D-Fender same without the bindaddr, might I need a reboot rather than "reload" sometime ? I am looking into upping the sipura "volume": to extn. |
| 15:11.57 | mosty | feld: is the phone plugged into the correct port? |
| 15:12.03 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:12.16 | feld_ | yes mosty |
| 15:12.21 | [TK]D-Fender | feld_ : You mean analog phones plugged into the FXS ports? And you tried the other ones just to be sure which were which? |
| 15:12.27 | feld_ | i've tried both actually, but the original one is where there was a dialtone at one time |
| 15:12.40 | [TK]D-Fender | feld_ : Made sure to plug in the Molex connector? |
| 15:12.45 | feld_ | correct, analog to FXS |
| 15:12.50 | drew___ | [TK]D-Fender - ok i checked the cards - they work solo anywhere - but as soon as there are 2 in the box it cant find one |
| 15:13.00 | file | is teh Asterisk running? |
| 15:13.02 | [TK]D-Fender | drew___ : oK, i'M OUT OF IDEAS FOR YOU |
| 15:13.15 | [TK]D-Fender | feld_ : yES, YOU NEED * RUNNING FOR DIAL-TONE... |
| 15:13.16 | feld_ | I dont have access to the inside of the box at the moment [TK]D-Fender but my boss said they had dialtone on it at one time (they were cheating, downloading configs, and not understanding the setup at all... lol) |
| 15:13.37 | feld_ | * is running |
| 15:13.58 | feld_ | Connected to Asterisk 1.2.9.1...... |
| 15:13.58 | InHisName | [TK]D-Fender in sipura: FXS prot polarity config is forward for all three - idle, callee, caller. |
| 15:14.34 | [TK]D-Fender | feld_ : pastebin your zaptel & zapata |
| 15:14.40 | feld_ | [TK]D-Fender: ok. |
| 15:14.54 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 15:15.54 | mr_horsepower | hello again |
| 15:16.12 | mr_horsepower | ppl, do 7940 and 7960 have PoE? i cant remember. |
| 15:16.16 | eKo1 | Has anyone here tried chan_ss7? |
| 15:16.38 | [TK]D-Fender | mr_horsepower : Not 802.3af PoE, only Cisco Proprietary PoE |
| 15:16.55 | feld_ | [TK]D-Fender: http://sh.nu/p/1933 |
| 15:16.59 | mr_horsepower | [TK]D-Fender: hum, ok tks. |
| 15:17.28 | feld_ | I have kept their configs for both because they said dialtone was working. I'm not sure what needs to be changed to make this function again =/ |
| 15:17.44 | *** join/#asterisk Talmage (n=Talmage@mychoice-fw.mychoice.cc) |
| 15:21.33 | [TK]D-Fender | feld_ : Friggen AMP.... |
| 15:21.40 | InHisName | [TK]D-Fender I moved FXS port input & output gains from -3 to -1 for both. Still nothing. Maybe something wrong with my extensions.conf |
| 15:21.57 | [TK]D-Fender | feld_ : Least you could have done is attached the included files as well... |
| 15:21.58 | feld_ | [TK]D-Fender: yeah they had AMP installed on this damn thing. I have _NO_ intentions of using it |
| 15:22.18 | feld_ | ehrm let me look... sorry :P |
| 15:23.03 | feld_ | [TK]D-Fender: zapata_additional is emtpy |
| 15:23.15 | *** join/#asterisk kevinfcn (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 15:23.28 | feld_ | http://sh.nu/p/1934 |
| 15:23.32 | feld_ | and there's zapata-auto |
| 15:23.33 | [TK]D-Fender | feld_ : Well I see *1* port configured, thats it.. and its FXO. |
| 15:23.48 | [TK]D-Fender | feld_ : Your config is broken |
| 15:24.03 | feld_ | well then they lied when they said they had it working with this config then =/ |
| 15:25.01 | *** join/#asterisk Antoine67 (n=FreePBX2@212.103.11.106) |
| 15:25.05 | [TK]D-Fender | feld_ : If you have no intentions of using AMP, then start gutting that poor system out. |
| 15:25.10 | Antoine67 | Hi there |
| 15:25.34 | Antoine67 | can anyone help with an misdn problem ? |
| 15:25.37 | feld_ | [TK]D-Fender: you mean the /etc/asterisk/zapata.conf is broken? |
| 15:25.58 | feld_ | and the zaptel one is fine, right? |
| 15:26.16 | Antoine67 | unable to make a second call on an isdn line |
| 15:26.17 | [TK]D-Fender | feld_ : yes. only 1 channel defined! and the first one is FXO. |
| 15:26.34 | feld_ | that config file seems pretty strange IMO |
| 15:26.44 | feld_ | i'll have to find documentation on it...... |
| 15:26.51 | [TK]D-Fender | feld_ : And you tried overriding it in zaptel in a really screwed up way |
| 15:27.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.146) |
| 15:27.48 | Antoine67 | can anyone help me ? |
| 15:27.54 | feld_ | [TK]D-Fender: that wasnt my work. I never touched either of those configs until now |
| 15:28.06 | feld_ | i'm just now looking at them...... =/ |
| 15:28.51 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 15:29.03 | [TK]D-Fender | feld_ : PM |
| 15:29.31 | drew___ | ok i found out some stuff about the interrupts on the system - the two cards use interrupts 9 and 10 - 9 is shared with the intel usb chip on the mobo |
| 15:29.39 | Antoine67 | ? |
| 15:29.52 | drew___ | how do i reassign the interrupts? |
| 15:29.55 | [TK]D-Fender | drew___ : That is BAD |
| 15:30.05 | [TK]D-Fender | drew___ : In your BIOS if you're lucky |
| 15:31.39 | drew___ | i can only specify if a IRQ is "available" or "reserved" in the bios |
| 15:31.42 | Antoine67 | does anyone know why I'm unable to use the second channel of an ISDN line ? |
| 15:33.02 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 15:34.22 | drew___ | "reserved" interrupts would be for possible ISA devices i guess |
| 15:34.58 | [TK]D-Fender | drew___ : For PCI as well on a good MB |
| 15:35.15 | drew___ | ok - ill try to reserve two... |
| 15:35.47 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 15:35.47 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 15:36.41 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 15:37.07 | muppetmaster | Hello. Any ideas why I can not get Zaptel to compile on CentOS? http://pastebin.ca/62966 |
| 15:37.14 | Juggie | ~centosbug |
| 15:37.15 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to ... |
| 15:37.16 | muppetmaster | Asterisk v1.2.9.1 compiles just fine. |
| 15:37.30 | muppetmaster | Ah, wow,, that was fast. |
| 15:37.57 | Juggie | make sure you edit the proper version of spinlock.h |
| 15:38.16 | [TK]D-Fender | muppetmaster : Extremely well known issue with an easy fix. I'd have beaten him to it if not for chatting elsewhere ;) |
| 15:38.40 | Juggie | it affects centos & rhel |
| 15:38.42 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 15:38.53 | Juggie | ~rhelbug |
| 15:38.54 | jbot | it has been said that rhelbug is aka centosbug |
| 15:38.59 | *** join/#asterisk Jedirl (n=asdf@213.162.200.226) |
| 15:39.00 | Jedirl | Hello |
| 15:39.00 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:39.03 | muppetmaster | [TK]D-Fender The bot was quick, thanks. I had searched various forums but could not find it. |
| 15:39.08 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 15:39.13 | Jedirl | I'm having a big problem with default extensions |
| 15:39.14 | muppetmaster | I usually run on SuSE, so don't come across these RHel type issues. |
| 15:39.15 | *** join/#asterisk rogger (n=rogger@209.104.162.252) |
| 15:39.19 | muppetmaster | Not a big RHel fan anyway......... |
| 15:39.21 | InHisName | [TK]D-Fender is Ttm a valid itme for following: exten => 1021,1,Dial(${PHONES1},20,Ttm) |
| 15:39.51 | Jedirl | I have an extension called _X. in a context, but I still get "-- Extension 'fsdafsadggsda' in context 'cancelador_ecos' from 'asdfasdf' does not exist. Rejecting call on channel 0/1, span 2" |
| 15:40.04 | *** join/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146) |
| 15:40.06 | Juggie | muppetmaster, centos = rhel |
| 15:40.25 | *** part/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146) |
| 15:40.35 | frk2 | TKD-Fender --- you should be a charged service at Asterisk :) |
| 15:40.45 | frk2 | haha |
| 15:40.54 | frk2 | msg nickserv credit TKD-Fender |
| 15:40.54 | frk2 | lol |
| 15:41.06 | nettie | hey guys, I applied the jitterbuffer patch on asterisk 1.2.6 .. anyon eknow how to actually enable it in sip.conf please? or check if it's enabled? I put jitterbuffer=yes but honestly I Cant notice any difference.. uhmm any idea? thanx |
| 15:41.25 | frk2 | what patch? |
| 15:42.27 | zoa | <PROTECTED> |
| 15:42.33 | zoa | if its enabled it should show green lines |
| 15:42.35 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
| 15:42.35 | [TK]D-Fender | frk2 : that'd be "karma [TK]D-Fender++" |
| 15:42.40 | [TK]D-Fender | frk2 : that'd be "~karma [TK]D-Fender++" |
| 15:43.08 | [TK]D-Fender | frk2 : My consulting fees are very accessable for full start-finish setups :) |
| 15:43.11 | anonymouz666 | hey |
| 15:43.19 | anonymouz666 | anyone know something about varion cards? |
| 15:43.31 | Jedirl | I have an extension called _X. in a context 'cancelador_ecos', but I still get "-- Extension 'fsdafsadggsda' in context 'cancelador_ecos' from 'asdfasdf' does not exist. Rejecting call on channel 0/1, span 2" |
| 15:43.36 | anonymouz666 | it is very cheap if compared with digium cards.... |
| 15:43.51 | [TK]D-Fender | anonymouz666 : Yeah, they originally look like a cheap alternative except support sucks, and better things have come out since.... quick summary : not worth it. |
| 15:43.55 | InHisName | [TK]D-Fender I found Answer() commented out, then Dial(),Macro(vmsg), and Hangup. I uncommented Answer() and renumberd priorities. Still no audio. |
| 15:44.11 | mr_horsepower | Jedirl: offcourse, you dont have that extension. |
| 15:44.20 | Jedirl | mr_horsepower: but I have _X. |
| 15:44.22 | [TK]D-Fender | InHisName : Yeah, your scenario has just about run its course with me.... not sure what the issue is... |
| 15:44.26 | mr_horsepower | Jedirl: _X. does not match with fsdafsadggsda |
| 15:44.40 | *** join/#asterisk littlejohn (n=little@host12-254.pool8717.interbusiness.it) |
| 15:44.52 | Jedirl | mr_horsepower: I've tried with 's' too |
| 15:44.54 | anonymouz666 | [TK]D-Fender: it's jim dixon cards? |
| 15:44.56 | Jedirl | exit |
| 15:45.04 | anonymouz666 | generic alternative hehe |
| 15:45.06 | mr_horsepower | Jedirl: why should match with s? |
| 15:45.13 | InHisName | [TK]D-Fender I apperciate alll the effort that you provided, as I was running out of ideas myself. |
| 15:45.19 | Jedirl | mr_horsepower: isn't _X. the default extension? |
| 15:45.52 | anonymouz666 | there is a mofo here saying that digium makes the card specially for him....(hehe) so I opened the machine and found this card... Tormenta III |
| 15:45.55 | [TK]D-Fender | InHisName : Well you're a good distance ahead.... Keep up the good work, and paste your refined setup on the msg boards and describe your setup clearly and someone should be able to carry you on from there. |
| 15:47.01 | Jedirl | How can I make a default extension for a context??? |
| 15:47.06 | Jedirl | I thought it was _X. |
| 15:48.14 | coppice | I still have a Tormenta 1. shame i have no ISA slot to plug it into :-( |
| 15:49.16 | anonymouz666 | TE410P is about 2500-3000USD the Tormenta III 4 quad is 699USD |
| 15:49.43 | Cresl1n | coppice: lol, yeah, I think we have one or two of those kicking around too |
| 15:49.44 | anonymouz666 | that's a lot of difference |
| 15:50.22 | Jedirl | AFAIK _X. should mach any dialed number in a extension, right? |
| 15:50.56 | mosty | i have a context with a bunch of other sub-contexts included in it. once * finds the first matching extension in the sub-contexts (in order) does * then search within that sub-context for the next priority? |
| 15:52.14 | coppice | Cresl1n: my one is E1 :-) |
| 15:52.23 | Cresl1n | heh |
| 15:52.24 | mr_horsepower | Jedirl: number, not alpha-numeric. |
| 15:52.38 | Jedirl | yes, my "asdfgdsaf" is just a mask for my number |
| 15:52.48 | mr_horsepower | mask? |
| 15:53.07 | Jedirl | I've put that just to hide my phone numbers from a public chat |
| 15:54.27 | Jedirl | now I get this: |
| 15:54.27 | Jedirl | <PROTECTED> |
| 15:54.27 | Jedirl | Timed out looking for connect acknowledge |
| 15:54.40 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 15:54.44 | docelmo | Hay I got a question for the home guys in here.. When you signup with term providers do you actually download the rates? |
| 15:55.09 | salviadud | why should i upgrade immediately? |
| 15:55.24 | Talmage | I have the pap2-na adapters, I want to be able to remotely reset them via sip notify. They have the Auth_Sip-Resync parameter which if set to yes requires sip notify requests be authenticated...how do I authenticate sip notify requests? I would like to use this method, as opposed to leaving the adapter wide open for anyone to reboot. |
| 15:59.11 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 16:05.10 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released with a critical security fix for chan_iax2, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 16:13.30 | *** join/#asterisk jjjjjjj (n=jdumont@64.46.3.83.novuscom.net) |
| 16:14.07 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 16:14.42 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 16:14.50 | *** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 16:15.35 | smackus | I am seting the first priority of each extension with exten => 100x,1,Set(CDR(accountcode)=Test) for example, how do I do this globally for each context? |
| 16:16.57 | jjjjjjj | i'd like to ask a question: I already have voip with primus.ca My internet connection is fibre optic via a rj45 jack in our apartment. What benefit will using asterisk give me other than the learning curve? Is asterisk for pots or am I missing something? Thank you. |
| 16:17.07 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 16:17.12 | pythos | G-mornin! |
| 16:17.29 | salviadud | jbot, asterisk rulez maaan |
| 16:17.36 | salviadud | jjjjjjj, i mean |
| 16:17.59 | jjjjjjj | oh i agree... i saw asterisk on a systm segment |
| 16:18.26 | salviadud | well, depends on what you want to do with it |
| 16:18.35 | salviadud | if you want an answering machine for voip |
| 16:18.38 | jjjjjjj | systm.org |
| 16:18.47 | salviadud | works great |
| 16:18.51 | jjjjjjj | got that through primus.ca |
| 16:19.01 | salviadud | prank call-war dialer machine, awesome too |
| 16:19.22 | smackus | anyone? |
| 16:19.26 | pythos | I am wondering if I have something missing in my configs for getting an SIP phone working, I'd aks for help if I didn't think I'd get rocks tossed in my general direction. |
| 16:19.42 | jjjjjjj | one thing that tickled my fancy watching the systm video was the little wireless phone where I can get calls from home if I am in a wifi range. that was wicked. |
| 16:19.51 | smackus | pythos: share your configs |
| 16:19.58 | eKo1 | Has anyone here tried chan_ss7? |
| 16:20.17 | pythos | smackus: whats that pastURL? |
| 16:20.26 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.144) |
| 16:21.42 | CunningPike | Anyone else here use Colloquy? |
| 16:22.01 | salviadud | what is chan_ss7? |
| 16:22.57 | smackus | pastebin.ca |
| 16:23.04 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 16:23.07 | smackus | what kind of phones? |
| 16:23.45 | smackus | pythos: what kind of phones? |
| 16:23.55 | salviadud | you guys know of any iaxclients on freebsd? |
| 16:24.48 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 16:24.59 | zoa | iaxcomm probably |
| 16:25.13 | Dr-Linux | question, my dynamic meetme doesn't allow more than 2 users, why? here is my >> exten => 8881,1,Meetme(,dM) |
| 16:25.17 | [TK]D-Fender | jjjjjjj : Means well, but Wikif phones all SUCK right now. |
| 16:25.37 | jjjjjjj | thanks for that info. |
| 16:25.42 | pythos | smackus: Um, its a FLash updated packet8 phone... Ill get the manufacturer infor, sec. |
| 16:26.30 | uwe | <PROTECTED> |
| 16:26.35 | [TK]D-Fender | jjjjjjj : Sorry its bad news... range, batter life, the works... many hotspots require HTTP auth for access... can be a PITA. Do feel free to try one. I'd suggest you shop carefully. the newer Linksys may be better for you if you really want to give it a shot. |
| 16:26.40 | pythos | smackus: Leadtek BVA8051 |
| 16:27.04 | Dr-Linux | any idea about my conference question? |
| 16:27.17 | [TK]D-Fender | pythos : thats not a phone, its an ATA |
| 16:27.24 | pythos | OH! no wonder! |
| 16:27.42 | smackus | anyone... trying to get something like exten => X,1,Set(CDR(accountcode)=Test) to work globally over one context. |
| 16:27.43 | pythos | um... well, ok, you plug a pots phone into it, sorry |
| 16:28.20 | key2 | soneone could tell me how to use the MySQL() application ? |
| 16:28.43 | *** join/#asterisk FinboySlick (n=FinboySl@c207.134.243-64.clta.globetrotter.net) |
| 16:28.44 | trelane_ | I have just eaten my shirt on a 30 line system with an ASUS K8N motherboard. The board would not steer IRQ's around hte WCTDM24XXP, has anyone used the WCTDM24xxp, and if so what boards have they found that they work on |
| 16:28.47 | jjjjjjj | so if I already have voip via the fddi connection then I'm not really gaining anything using asterisk other than the challenge and learning curve to maybe set it up for someone who still uses pots? |
| 16:30.54 | pythos | Hmm, brb |
| 16:31.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 16:32.03 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 16:32.27 | [hC] | So, anyone have any fun ideas for my 7940 here, it requests its config from my tftp server, yet still insists "Phone Unprovisioned" |
| 16:32.34 | [hC] | Everything looks right, i dont get it. |
| 16:32.35 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 16:32.58 | pythos | ok, so anyone willing to hop me thru getting an ATA up? |
| 16:33.23 | *** part/#asterisk jjjjjjj (n=jdumont@64.46.3.83.novuscom.net) |
| 16:33.44 | *** join/#asterisk jjjjjjj (n=jdumont@64.46.3.83.novuscom.net) |
| 16:34.15 | [TK]D-Fender | trelane : how many and what kind of ports are you actually running on that server? |
| 16:34.54 | *** part/#asterisk jjjjjjj (n=jdumont@64.46.3.83.novuscom.net) |
| 16:37.31 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 16:38.56 | C4T3l | ~pb |
| 16:38.57 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 16:41.49 | salviadud | a sip ata? |
| 16:42.07 | salviadud | pythos, that's basic stuff |
| 16:42.48 | [hC] | why dont you learn the settings you download, bastard |
| 16:43.12 | pythos | sal, ok, I agree, Im less then basic, however :-) |
| 16:44.12 | `Sauron | hC: feel free to send me that 7940... ;) |
| 16:44.18 | [hC] | hah |
| 16:44.22 | [hC] | I have a ton of them |
| 16:44.27 | vader-- | hehe hc i just got 60 of these things in |
| 16:44.30 | vader-- | they work ok |
| 16:44.37 | vader-- | i don't like the dialplan.xml thing |
| 16:44.39 | vader-- | thats a pia |
| 16:44.41 | [hC] | this one is deciding that its cool to download its config from tftp, yet not actually USE it, then claim unprovisioned. |
| 16:44.51 | [hC] | Ive configured hundreds of these and never seen this |
| 16:46.35 | Dr-Linux | [TK]D-Fender: please check it http://pastebin.com/765520 |
| 16:46.42 | Dr-Linux | your thoughts? |
| 16:46.44 | Juggie | [hC], thats because it doesnt like you. |
| 16:47.04 | Juggie | i know how it feels. |
| 16:47.12 | key2 | is it possible to have a trunk in SIP or is it only for IAX ? |
| 16:47.22 | Dr-Linux | Juggie: hey there :) |
| 16:47.29 | Juggie | hey |
| 16:47.40 | Juggie | if you expect me to remember=your problem i dont |
| 16:47.42 | Juggie | but hey. |
| 16:47.46 | [TK]D-Fender | Dr-Linux : What about it? |
| 16:48.04 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
| 16:48.18 | [TK]D-Fender | key2 : Define "trunk" Any kind of line/link could be termed a "trunk". |
| 16:48.33 | Dr-Linux | Juggie: heh fixing the problem is not that much hard, but picking the problem is :) |
| 16:48.50 | [hC] | Juggie haha. |
| 16:49.07 | Dr-Linux | [TK]D-Fender: i never use PRI and T1, so in my understanding i configure the zap configs, so i need your guidness if i'm wrong |
| 16:49.37 | dlynes_home | [TK]D-Fender: btw...the A200d...I keep getting an error about it not being able to allocate memory for the card from the system |
| 16:49.38 | Dr-Linux | [TK]D-Fender: i'll use 4 ports T1 |
| 16:49.46 | [hC] | i wanna get the crap out of here. |
| 16:49.56 | dlynes_home | [TK]D-Fender: i've sent in a support ticket about it to sangoma already |
| 16:50.29 | dlynes_home | [TK]D-Fender: otoh, do you know where there's any documentation for that card? documentations for it seems to be totally nonexistent |
| 16:50.36 | [TK]D-Fender | Dr-Linux : looks mostly right. Have you considered TRYING it? |
| 16:51.01 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:51.05 | [hC] | dlynes_home: ive had to just use the sangoma wiki and voip info so far. |
| 16:51.11 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:51.12 | [TK]D-Fender | Dr-Linux : And group 3 includes 2 PRI? |
| 16:51.21 | [hC] | aah... qwell |
| 16:51.22 | dlynes_home | [hC]: are you using the pri card, or the a200 card? |
| 16:51.25 | [hC] | maybe you'd know why this is happening. |
| 16:51.32 | Qwell[] | [hC]: unlikely |
| 16:51.34 | [hC] | dlynes_home: I use the a200, a200d, and a102u's |
| 16:51.58 | [hC] | Qwell[]: :P I have a 7940 here that requests its config from the tftp server, yet still says 'phone unprovisioned' - like its ignoring it. seen that? |
| 16:52.03 | dlynes_home | [hC]: eh? aren't the 200 and the 200d the same thing? and what's the a102u? |
| 16:52.16 | Qwell[] | [hC]: got the accounts specified? |
| 16:52.23 | [TK]D-Fender | dlynes_home : "d" echo cancel, and 102 = 2 port T1 |
| 16:52.28 | [hC] | dlynes_home: the a200d = echo can, a200 = no echo can. a102u = 2 port t1 pri |
| 16:52.46 | dlynes_home | ah...ok, i've got 2 a200's, and 1 a200d |
| 16:52.51 | Qwell[] | those are analog? |
| 16:52.52 | dlynes_home | and an a101 |
| 16:52.54 | [hC] | Qwell[]: yep. copied working config from another phone |
| 16:53.09 | Qwell[] | [hC]: funky |
| 16:53.10 | Dr-Linux | [TK]D-Fender: actually i have 2 TE210P cards (2+2= 4 ports), and we will use 2 PRI lines, firt port of each card |
| 16:53.12 | dlynes_home | Qwell[]: the a200's? yes |
| 16:53.20 | [hC] | Its requesting MGCDefault.cnf which ive never noticed any others tdo before, too. |
| 16:53.59 | [hC] | sec, i'll pastebin everything, see if you see somehing funky. |
| 16:54.11 | dlynes_home | [hC]: yeah, I suppose i must just have really weird hardware |
| 16:55.04 | pythos | Ok, so on this ATA, I only have config slots for SIP settings, and when I change them, I can get the log files on the asterisk box to either gripe or NOT. But I can't get the ATA to give dial-tone on a phone. I assume that I have something wrong in my sip.conf, or else I have NOT enabled something in another conf file. Suggests? |
| 16:56.02 | *** join/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il) |
| 16:56.05 | Juggie | [hC] does the firmware name match in all the files, and the mac address match all that jazz |
| 16:56.15 | *** join/#asterisk twilson (n=twilson@dhcp-63-77-68-87.ojc.nuvio.com) |
| 16:56.25 | Juggie | maybe the phone thinks its a different mac then whats actually written on the back? |
| 16:56.25 | [TK]D-Fender | pythos : typically ATA's only give dialtone if they have successfully registered to a server. Watch your CLI when you power it up to see the attempt, including SIP debug info |
| 16:56.36 | liran_ | what is a cheap fxo card? |
| 16:56.46 | Juggie | TDM400 |
| 16:57.26 | liran_ | Juggie: uhmm, can't find that on my country's price-comparison website... maybe another model? |
| 16:57.41 | liran_ | Juggie: im looking for a card to connect asterisk to a PSTN |
| 16:57.55 | Juggie | Dr-Linux, whats the problem? |
| 16:58.36 | Juggie | liran_, http://www.digium.com/en/products/hardware/tdm400p.php |
| 16:59.12 | Dr-Linux | Juggie: didn't try yet, as i said before i never use pri/t1 stuff, so not we are going to use by tomorrow, so i configured zap configs, so just wanted to verify them. |
| 16:59.27 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 16:59.30 | Juggie | Dr-Linux, can you paste bin your zaptel and zapata.conf? |
| 16:59.52 | [hC] | http://pastebin.ca/62978 |
| 16:59.56 | [hC] | there we are. |
| 17:00.00 | [hC] | firmware all matches |
| 17:00.10 | [hC] | (ive changed my xmldefault file to reflext -2- instead of -4- as well.. |
| 17:00.17 | Dr-Linux | Juggie: friend i aready done |
| 17:00.26 | Juggie | whats the url? |
| 17:00.29 | Dr-Linux | Juggie http://pastebin.com/765520 |
| 17:00.37 | pythos | TK, only gripe is the one about no spawn_mp3, and mp3player |
| 17:00.42 | Talmage | I have the pap2-na adapters, I want to be able to remotely reset them via sip notify. They have the Auth_Sip-Resync parameter which if set to yes requires sip notify requests be authenticated...how do I authenticate sip notify requests? I would like to use this method, as opposed to leaving the adapter wide open for anyone to reboot. |
| 17:00.59 | Dr-Linux | Juggie: i have 2 TE210P digium cards |
| 17:01.08 | Juggie | are all 4 t1's in use? |
| 17:01.33 | Juggie | you are going to connect all 4? |
| 17:01.56 | Dr-Linux | Juggie: i'll have only 2 PRI lines |
| 17:02.16 | Juggie | then you dont need to configure all 4 spans |
| 17:02.20 | Dr-Linux | each pri will be on first port of each card |
| 17:02.23 | Juggie | thats one thing i noticed. |
| 17:02.26 | liran_ | Juggie: thats an overkill for me. i just need one plain old analog port to connect to the phone socket at home. |
| 17:02.46 | Juggie | liran_, digium doesnt sell a one port device anymore with the exception of the iaxy |
| 17:02.51 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 17:02.52 | salviadud | 100xp |
| 17:02.56 | Juggie | you'll have to find one elsewhere or on ebay etc. |
| 17:03.05 | Dr-Linux | Juggie: then how should i manage the channels? |
| 17:03.29 | pythos | stkn: oh, the 'sip show channels' gives info: 192.168.2.4 (none) cdfb-56cc8- 00101/00103 unknown |
| 17:03.33 | pythos | glerp! |
| 17:03.40 | Juggie | if you are only using 2 t1's then you only have to setup 2 spans not 4. |
| 17:03.46 | Juggie | so you probally need to configure span1&3 |
| 17:03.49 | pythos | [TK]D-Fender: oh, the 'sip show channels' gives info: 192.168.2.4 (none) cdfb-56cc8- 00101/00103 unknown |
| 17:03.51 | Juggie | and not 2-4 |
| 17:03.54 | Juggie | er, 2 &4 |
| 17:04.03 | Juggie | you do the channels like you have them |
| 17:04.13 | Juggie | span 3 will have 25-47 dchan=28 |
| 17:04.24 | [TK]D-Fender | pythos : Not registered... |
| 17:04.33 | Juggie | also one other thnig |
| 17:04.38 | pythos | [TK]D-Fender: the "(none)" under user/ANR seems relevant.. |
| 17:04.42 | Juggie | you have all your spans set as your primary timeing source |
| 17:04.54 | Juggie | you should only have one set to be the primary sync source |
| 17:05.00 | Juggie | i have a meeting, brb. |
| 17:05.09 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:05.19 | Dr-Linux | Juggie: and should i remove chan 25-47 ? and same for port 4 ? |
| 17:05.22 | [TK]D-Fender | pythos : put it on "qualify=yes" |
| 17:05.42 | pythos | [TK]D-Fender: where? |
| 17:05.52 | [hC] | son of a bitch |
| 17:05.55 | [hC] | junior is gonna get it |
| 17:05.59 | [hC] | tftpd was pulling configs from the wrong dir. |
| 17:06.35 | [TK]D-Fender | pythos : in your phone config in sip.conf |
| 17:06.45 | pythos | k |
| 17:11.44 | chrismog | Hello. Is it possible to have asterisk send voicemails in mp3 format when it send an email? |
| 17:11.57 | pythos | [TK]D-Fender: hmm, well ok... now it is giving me a gripe about peer is now unreachable |
| 17:12.39 | [TK]D-Fender | pythos : Not looking good for your setup... |
| 17:13.07 | pythos | heheh, as I surmised... |
| 17:13.40 | pythos | I think I just don't know enough yet. |
| 17:14.28 | [TK]D-Fender | pythos : turn up your SIP debug, power down the ATA and restart it and see what happens |
| 17:14.30 | mr_horsepower | someone in uk? |
| 17:15.56 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 17:16.55 | *** join/#asterisk JINDAL (n=root@220.226.28.164) |
| 17:17.56 | [hC] | hey, im trying to use ztmonitor to test for my proper txgain output values... if i set Zap/1 to answer and pass to milliwatt() its at the precise correct value no matter what i set it to, this is obviously not a proper test.. how should i be doing it? |
| 17:18.42 | *** join/#asterisk aze (n=aze@ACayenne-101-1-7-14.w81-248.abo.wanadoo.fr) |
| 17:19.59 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 17:20.32 | JINDAL | hey guys...... i wanna know the best softphone for |
| 17:20.45 | JINDAL | sip softphone for asterisk |
| 17:20.46 | Bullseye_Network | Has anybody elase had problems running Apache on the same server as Asterisk? |
| 17:20.59 | Bullseye_Network | JINDAL: I like sjphone |
| 17:21.09 | Bullseye_Network | JINDAL: www.sjlabs.com |
| 17:21.12 | [hC] | hrm, |
| 17:21.13 | [hC] | brb |
| 17:21.18 | JINDAL | okey |
| 17:21.31 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
| 17:21.36 | Bullseye_Network | JINDAL: I have 100+ linux and Windows PC's running sjphone |
| 17:21.53 | JINDAL | gud |
| 17:24.38 | pythos | [TK]D-Fender: hmm, I see something that catches my eye: under the :to <ip>:5060 I see transitting (no NAT) then SIP/2.0 401 unautherized |
| 17:24.42 | pythos | Im guessing thats the problem |
| 17:26.07 | *** join/#asterisk inventor_ (n=spam@static-71-121-129-61.sttlwa.dsl-w.verizon.net) |
| 17:26.20 | [TK]D-Fender | pythos : Bad user/pass.... |
| 17:26.27 | [TK]D-Fender | pythos : Stands out like a sore thumb |
| 17:26.27 | *** join/#asterisk Bert- (n=bert@i05v-87-90-132-119.d4.club-internet.fr) |
| 17:26.31 | Bert- | hello there |
| 17:26.44 | pythos | adding user/pass is done in the SIP.conf, or somewhere else? |
| 17:26.44 | Bert- | Hi [TK]D-Fender |
| 17:27.19 | Bert- | [TK]D-Fender, I'm sorry to ask you that again but can you give me the link about Asterisk book please ? |
| 17:27.45 | [TK]D-Fender | ~book |
| 17:27.51 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:27.56 | Bert- | I forgot it at desk :( and I want to try Asterisk this evening |
| 17:28.06 | [TK]D-Fender | pythos : Correct |
| 17:28.45 | inventor_ | can someone help with this error, chan_sip.c:612 __sip_xmit: sip_xmit of... returned -1: Invalid argument -- this only happens when dialing TO a polycom soundpoint.. |
| 17:28.57 | inventor_ | the soundpoint can dial outbound |
| 17:28.59 | Bert- | thx ;) |
| 17:29.29 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:29.37 | pythos | [TK]D-Fender: well it looks right there... peraps Im not seeing something in the SIP conf of the ata.. Ill keep trying, but thanks!!! |
| 17:29.40 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 17:30.21 | Seyr | What would cause the voicemail announcement to be choppy, but MOH and talking is fine? |
| 17:30.44 | [TK]D-Fender | inventor_ : Pastebin your SIP.CONF and related dial-plan for a call that fails. |
| 17:31.13 | Bullseye_Network | Seyr: is the recorded announcment choppy or the menu? |
| 17:31.39 | Seyr | menu |
| 17:31.55 | Seyr | "The person at extension XXX is unavailable" <--- that |
| 17:31.56 | Bullseye_Network | Hmmm. |
| 17:32.24 | JINDAL | okey guys can asterisk use a voice modem or the support is only for isdn |
| 17:34.36 | [TK]D-Fender | ~pb |
| 17:34.38 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 17:35.51 | Bullseye_Network | Seyr: Is it choppy for internal phones as well as calling in from outside? Are you using a VIOP provider or Digium cards? |
| 17:36.36 | *** join/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 17:38.06 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 17:39.06 | *** join/#asterisk scrubb (n=scrubb@IP-216-37-19-40.nframe.com) |
| 17:39.57 | Seyr | Bullseye_Network: I am using SIP and it does not matter if the call is from outside, or from extension to extension. MOH plays fine and voice works fine. |
| 17:40.15 | Seyr | but "The person at extension XXX is unavailable" sounds horrible |
| 17:41.40 | CunningPike | Seyr: Codec mismatch? |
| 17:41.43 | Bullseye_Network | Seyr: Have you tried different Codec's for the sip calls? |
| 17:41.54 | Bullseye_Network | lol |
| 17:42.06 | *** join/#asterisk RippPPppE (n=ripppppp@203.115.71.253) |
| 17:42.12 | RippPPppE | hi all |
| 17:43.01 | RippPPppE | facing a lot of problems lately with asterisk updated to 1.2.9.1 |
| 17:43.20 | Bullseye_Network | I upgraded last night and havnt seen a problemm.. YET |
| 17:43.21 | RippPPppE | if anyone transfers a call from one extension to another extension |
| 17:43.36 | RippPPppE | the phone on the other extension rings ones |
| 17:43.47 | RippPPppE | and then calling works for some time |
| 17:43.52 | RippPPppE | then everything just dies |
| 17:44.03 | RippPPppE | immediately the effected commands |
| 17:44.03 | RippPPppE | are |
| 17:44.10 | RippPPppE | show queues / show agents |
| 17:44.19 | RippPPppE | they do not return anything |
| 17:45.01 | RippPPppE | any ideas of debugging |
| 17:45.12 | Bullseye_Network | WOW; I just did a show queues and I get nothing. |
| 17:45.18 | Bullseye_Network | Also on show agents |
| 17:45.22 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 17:45.27 | Bullseye_Network | it says my queues do not exist |
| 17:45.30 | RippPPppE | i have 4 queues |
| 17:45.40 | RippPPppE | and 7 agents logged in |
| 17:45.42 | Bullseye_Network | I use 4 queues also |
| 17:45.49 | Bullseye_Network | I have 40+ agents logged in |
| 17:45.51 | funxion | app_addon_sql_mysql.c:273 aMYSQL_query: aMYSQL_query: mysql_store_result() failed <-- anyone seen this before? |
| 17:46.22 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:46.22 | Bullseye_Network | Its allowing them to login but I cant get info |
| 17:46.36 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:46.51 | Bullseye_Network | oh wait |
| 17:46.55 | Bullseye_Network | im an Idiot |
| 17:47.03 | Bullseye_Network | lol |
| 17:47.30 | Bullseye_Network | No problems here. |
| 17:47.40 | RippPPppE | good for you |
| 17:47.43 | Bullseye_Network | I have soo many asterisk servers I was on the wrong one |
| 17:47.44 | RippPPppE | any pointers |
| 17:47.53 | RippPPppE | you are safe |
| 17:47.56 | RippPPppE | i have one only |
| 17:47.59 | Bullseye_Network | Well. I have 40+ agents all transferring calls |
| 17:48.25 | Bullseye_Network | Hmm |
| 17:48.28 | RippPPppE | any ideas how can i debug |
| 17:49.13 | Bullseye_Network | when you say "Everything just dies" what exactally do you mean |
| 17:51.11 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 17:51.36 | lifendel | Is anyone else using Feature Group D? |
| 17:52.02 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:52.21 | Bullseye_Network | Has anybody elase had problems running Apache on the same server as Asterisk? |
| 17:52.35 | lifendel | Runs fine for me. |
| 17:52.38 | [TK]D-Fender | Bullseye_Network : nope. |
| 17:52.50 | lifendel | What problems are you having? |
| 17:52.52 | [TK]D-Fender | Bullseye_Network : I use it on my production server at work, and at home. |
| 17:53.00 | Vorondil | hey, [TK]D-Fender: i got that variable thing to work that we were talking about the other day. your ${EVAL()} works just find, but my global variables weren't being set. so that was the hangup... all because i had "[global]" instead of "[globals]" :-P |
| 17:53.03 | Vorondil | one stupid "s" |
| 17:53.15 | Vorondil | s/find/fine |
| 17:53.21 | Bullseye_Network | I have been told there are problems due to the face SIP and HTTP codes are almost identical EVEN though they are UDP vs TCP and on different prots |
| 17:53.30 | lifendel | Hahaha, it's always just one keystroke that screws up everything. |
| 17:53.55 | anonymouz666 | the multiplex chip (e1/t1) in tormenta III cards (699USD) is the same of digium cards... xilinx spartan.... |
| 17:53.56 | lifendel | No, SIP and HTTP will not interfere with eachother. |
| 17:54.08 | Bullseye_Network | But the problem only occurs when 20 or more call at make per seconds |
| 17:54.17 | Bullseye_Network | made per second |
| 17:54.25 | dlynes_office | [TK]D-Fender: even the sangoma tech is having problems trying to figure out what's wrong |
| 17:54.25 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 17:54.38 | tzafrir | anonymouz666, where can I find their drivers? |
| 17:54.46 | trelane_ | anyone from digium around? |
| 17:55.06 | lifendel | Bullseye_Network, what exact problem? Echo? Jitter? |
| 17:55.11 | salviadud | the guys at digium don't hang around here |
| 17:55.17 | trelane_ | due to a screw up w/ Allison I ended up getting a bunch of extra general purpose recordings done (They're no paid for) |
| 17:55.19 | salviadud | they're too COOL to be here maaaan |
| 17:55.20 | trelane_ | salviadud, bul. |
| 17:55.22 | trelane_ | l |
| 17:55.23 | tzafrir | trelane_, ask, just the same |
| 17:55.24 | dlynes_office | salviadud: um, yeah they do |
| 17:55.30 | salviadud | i'm joking of course |
| 17:55.40 | trelane_ | tzafrir, this would be regarding a commit to asterisk-sounds-extra |
| 17:55.44 | salviadud | i've never talked to a guy from digium though... |
| 17:55.53 | trelane_ | salviadud, there are many here :) |
| 17:55.53 | Bullseye_Network | im having deadlocks ALOT, and was told to take apache off. |
| 17:55.56 | trelane_ | though I won't name+shame them |
| 17:56.17 | dlynes_office | trelane_: if you're wanting a commit done, ask iin #asterisk-dev |
| 17:56.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:57.03 | anonymouz666 | tzafrir: zaptel? |
| 17:57.06 | lifendel | Bullseye_Network, You were probably told that because Apache will put a load on the system depending on use. |
| 17:57.24 | lifendel | Bullseye_Network, What is apache serving on the system? |
| 17:57.30 | Bullseye_Network | Im getting deadlocks on the Agent channels |
| 17:58.01 | [TK]D-Fender | dlynes_office : Did they log in to look for themselves? |
| 17:58.06 | *** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com) |
| 17:58.12 | *** mode/#asterisk [+b %anonymouz666!*@*] by russellb |
| 17:58.14 | dlynes_office | [TK]D-Fender: yeah...he's still in there |
| 17:58.32 | dlynes_office | [TK]D-Fender: he's been in there for about 1/2 hour now |
| 17:58.59 | [TK]D-Fender | Vorondil : *thwap* |
| 17:59.01 | [TK]D-Fender | :D |
| 17:59.07 | [TK]D-Fender | dlynes_home : Who is it? |
| 17:59.09 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 18:02.52 | Bullseye_Network | http://www.bullseyenetworks.com/agentdeadlock.log |
| 18:02.57 | Bullseye_Network | get these alot. |
| 18:03.12 | Vorondil | [TK]D-Fender: thanks for your help though, i really appreciate it. ^_^ |
| 18:03.31 | *** mode/#asterisk [-b %anonymouz666!*@*] by russellb |
| 18:03.59 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 18:04.35 | Bullseye_Network | lifendel:its serving a couple perl scripts not a really high volume |
| 18:04.56 | dlynes_office | [TK]D-Fender: don't know...never really asked him |
| 18:05.09 | dlynes_office | [TK]D-Fender: but he couldn't figure it out, so he's going to get a higher up to take a look at it |
| 18:05.49 | inventor_ | [TK]D-Fender: http://cpp.enisoc.com/pastebin/7013 |
| 18:06.01 | inventor_ | again, it's a polycom not ringing |
| 18:06.56 | Seyr | ok, Playback() is choppy, but MOH and talking is fine |
| 18:07.15 | Seyr | i guess the Voicemail uses Playback() or some form of it as well |
| 18:07.36 | Seyr | so what would cause the playback of gsm files to be choppy, but MOH to be ok? |
| 18:11.53 | jsharp | Sunspots |
| 18:11.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:12.35 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 18:12.43 | RippPPppE | >>> <Bullseye_Network> when you say "Everything just dies" what exactally do you mean |
| 18:12.55 | RippPPppE | people who ar on calls |
| 18:12.59 | RippPPppE | will stay on calls |
| 18:13.09 | RippPPppE | new calls do not come in |
| 18:13.11 | RippPPppE | or can be made |
| 18:13.17 | RippPPppE | AND |
| 18:13.28 | RippPPppE | queue / agent status with the show commands |
| 18:13.32 | RippPPppE | cannot be seen |
| 18:13.38 | RippPPppE | i just go back to the CLI prompt |
| 18:14.18 | Blackthorn | Hi, I am using webadmin to go over my * box. There are a few services that interest me. apmd for monitoring batteyr status, mdmonitor for monitoring raid. is it safe to turn these off on the * box if i do not have a apc on the box nor raid drives? |
| 18:16.00 | salviadud | webadmin? |
| 18:16.25 | salviadud | can't you use the CLI, best gui ever |
| 18:17.03 | Bullseye_Network | RippPPppE: I would try and reinstall asterisk. Im using agents, queues and transfers at a call center and not seeing this. |
| 18:17.06 | CunningPike | salviadud: webadmin is a general server admin package |
| 18:17.41 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 18:18.26 | Bullseye_Network | Seyr: The playback files are gsm encoded. MOH is probabally mp3. I would try changing the codec's you use for the sip devices. |
| 18:19.01 | *** join/#asterisk ness (n=Tom@pppin-39-b6.pop-kaltenengers.rz-online.NET) |
| 18:19.29 | Bullseye_Network | Seyr: If your SIP devices are using ulaw then asterisk has to convert the gsm file to ulaw as its played. |
| 18:19.38 | Seyr | i use ulaw across the board |
| 18:19.43 | Bullseye_Network | If you record a message in the mail box is the recording choppy too? |
| 18:19.55 | RippPPppE | do you mean |
| 18:19.58 | RippPPppE | UNINSTALL |
| 18:20.01 | RippPPppE | and then REINSTALL |
| 18:20.32 | Bullseye_Network | RippPPppE: I would delete all files in the modules directory and make clean; make install; for asterisk |
| 18:20.50 | RippPPppE | oh that i have not tried |
| 18:21.27 | ness | does anyone know what http://pastebin.com/764636 means and how to process it properly (the goal is to display the local status of *;I forgot to bring the logs from the office). Someone local is calling out. |
| 18:21.32 | Bullseye_Network | What version did you have on before 1.2.9.1 |
| 18:21.37 | Dr-Linux | what's this warning mean? >> http://pastebin.com/765729 |
| 18:22.08 | RippPPppE | 1.2.7 |
| 18:22.43 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 18:23.14 | [TK]D-Fender | Dr-Linux : Means your $[] condition is formatted wrong |
| 18:23.48 | Dr-Linux | [TK]D-Fender: lemme show you then |
| 18:24.25 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 18:24.30 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 18:24.33 | Bullseye_Network | Dr-Linux: I've seen that when I try to set a variable from another variable and its empty |
| 18:25.04 | [TK]D-Fender | Dr-Linux : Bullseye_Network's case is a common example... |
| 18:25.43 | Seyr | Bullseye_Network: client is checking recording to see if it is choppy. i'll know in a few mins |
| 18:25.52 | Seyr | Bullseye_Network: any idea what it could be? |
| 18:26.34 | Seyr | MOH is mp3s and play fine. talking is fine as well. seems to only be playing GSM |
| 18:26.46 | Seyr | Playback(), Background() and Voicemail() |
| 18:26.54 | Bullseye_Network | Sounds like a codec problem to me. Can you try using gsm just to see if that fixes it? |
| 18:27.05 | Seyr | zztest shows 99.9 as well |
| 18:27.20 | Seyr | well, a couple 99.8 :-) |
| 18:27.21 | Dr-Linux | [TK]D-Fender: i get this warning while: |
| 18:27.21 | Dr-Linux | exten => t,1,Set(TRIES=$[${TRIES} + 1]) |
| 18:27.22 | Dr-Linux | exten => t,2,GotoIf($["${TRIES}" = "1"]?t,3:s,5) |
| 18:28.22 | [TK]D-Fender | Dr-Linux : pASTEBIN THE WHOLE CONTEXT. |
| 18:28.39 | Dr-Linux | [TK]D-Fender: ok |
| 18:28.48 | Seyr | if it was a code problem, wouldnt that make regular calls be choppy as well? not just Playback(), etc? |
| 18:28.53 | Seyr | codec even |
| 18:29.32 | Bullseye_Network | Not if both ends are using the same codec |
| 18:29.47 | Dr-Linux | [TK]D-Fender: this is IVR context and have more than 500 lines |
| 18:30.09 | [TK]D-Fender | Dr-Linux : 500 lines? WTF for? Pastebin it anyways |
| 18:30.16 | Seyr | Bullseye_Network: gsm and ulaw both give choppy |
| 18:31.04 | Bullseye_Network | Hmmm |
| 18:31.17 | Dr-Linux | [TK]D-Fender: lemme try to pastbin as minimum as i can |
| 18:31.50 | Blackthorn | Hi, I am using webadmin to go over my * box. There are a few services that interest me. apmd for monitoring batteyr status, mdmonitor for monitoring raid. is it safe to turn these off on the * box if i do not have a apc on the box nor raid drives? |
| 18:32.04 | Seyr | calling from xlite to the box and having it do Playback, Background and WaitMusicOnHold |
| 18:32.15 | Seyr | all are choppy, except WaitMusicOnHold |
| 18:32.40 | Dr-Linux | [TK]D-Fender: check it >> http://pastebin.com/765775 |
| 18:32.48 | salviadud | bye ppl |
| 18:33.58 | Seyr | maybe i was missing a lib or something when i compiled? |
| 18:34.19 | [TK]D-Fender | Dr-Linux : you problem is you never initialize TRIES to "0". therefor it is BLANK. Theres your failure. |
| 18:35.15 | Dr-Linux | [TK]D-Fender: well, my it works fine as i want it, but i'm get that warning. so what you suggest? |
| 18:37.43 | [TK]D-Fender | Dr-Linux : You need to initialize TRIES. thats all. |
| 18:38.13 | Dr-Linux | [TK]D-Fender: hhm... i don't understand, bu thanks |
| 18:38.18 | Dr-Linux | s/bu/but |
| 18:38.52 | [TK]D-Fender | Dr-Linux : Set(TRIES=0) at the start of your IVR. when you first go in it have NO value. not 0, not 1, but BLANK. thats whats failing. |
| 18:38.58 | stephane_ | soir |
| 18:39.04 | [TK]D-Fender | stephane_ : Salut |
| 18:39.14 | lifendel | Is anyone else using Feature Group D trunks? |
| 18:39.58 | Dr-Linux | [TK]D-Fender: aww i see, now i understand :) Thanks :) |
| 18:40.16 | *** join/#asterisk _DAW (n=bob@adsl-222-35-4.msy.bellsouth.net) |
| 18:43.13 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.4.154) |
| 18:43.38 | Trojan_Hors1 | hulllo all |
| 18:43.42 | _DAW | hello mate |
| 18:45.17 | Trojan_Hors1 | am a newbie in asterisk and am trying to test asterisk....... asterisk server and sip client on d same machine and it aint working......... do i need different physical machines / should i state the exact errors |
| 18:45.46 | lunk | heh. |
| 18:46.12 | jsharp | Different machines. Asterisk and the SIP client are trying to bind to the same port. |
| 18:46.42 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.4.154) |
| 18:47.59 | Trojan_Hors1 | ya am a newbie trying to test asterisk using both asterisk server and sip client on d same machine........... and it aint working do i need seperate machines / or should i state d errors |
| 18:48.36 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 18:49.25 | Bullseye_Network | ? |
| 18:50.03 | Bullseye_Network | Guess the answer wasnt what they wanted to hear so they thought they would ask again |
| 18:50.09 | *** part/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 18:52.21 | *** join/#asterisk ManxPower (n=ewieling@207.191.118.2) |
| 18:52.31 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 18:52.59 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.4.154) |
| 18:53.03 | JASON99 | Hello, |
| 18:53.06 | ManxPower | So I arrive on site. Nobody has the circuit ID, once they get the circuit ID, we can't find it on the wall. The circuit is also not patched into our telcoms room |
| 18:53.29 | Trojan_Hors1 | sry guys i got disc |
| 18:53.35 | JASON99 | I'm using mgcp and I'm unable to make a 3-way call. Does anyone know if there is a bug with this feature? |
| 18:53.59 | chrismog | Hello. I am having some severe echo issues with Asterisk. I have a Digium TDM400P. Is there anything special I need to do? |
| 18:54.15 | X-Gen | Whooha armadilloaerospace has an update :) |
| 18:54.23 | sevard | might want to mention something about hardware, chris |
| 18:55.08 | *** join/#asterisk grabowski (i=grabowsk@i.use.efnut.com) |
| 18:56.16 | Trojan_Hors1 | hi guys i hav a query, can asterisk server and a sip client be used on d same machine at the same time... |
| 18:56.19 | chrismog | Uh, its an AMD 1800+ running CentOS 4.3 with a 4 line Digium TDM400P. |
| 18:56.32 | chrismog | Do I need a hardware echo cancelation device? |
| 18:57.25 | chrismog | fxs lines :/ |
| 18:58.06 | Talmage | I have the pap2-na adapters, I want to be able to remotely reset them via sip notify. They have the Auth_Sip-Resync parameter which if set to yes requires sip notify requests be authenticated...how do I authenticate sip notify requests? I would like to use this method, as opposed to leaving the adapter wide open for anyone to reboot. |
| 18:58.10 | MatsK | Trojan_Hors1: YES |
| 18:58.48 | MatsK | Trojan_Hors1: But enshure that the don't use the same port |
| 18:59.04 | Trojan_Hors1 | okey |
| 18:59.51 | MatsK | Trojan_Hors1: so use two machines instead, it's simpler |
| 19:00.31 | Trojan_Hors1 | will switch to two :P |
| 19:00.54 | JASON99 | does anyone here use mgcp with asterisk? |
| 19:01.03 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 19:01.57 | eBody | using Asterisk do i need hardware for every extension? or do these extension run through the ethernet and can be used through a switch?? |
| 19:04.08 | MatsK | eBody: the second answer is right |
| 19:04.51 | nextime | eBody : you need specific hw for non-voip channels, extensions are not directly linked with channels, if you use only voip channels you don't need any special hw other than your ethernet card, and if you use only local channel to do something you don't need nothing at all but you can setup how many extensions that you like to setup |
| 19:05.22 | drew___ | when installing linux for a * box (i am installing fedora) would you activate or deactivate SELinux features and the firewall? |
| 19:06.50 | *** part/#asterisk kevinfcn (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 19:06.53 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 19:07.02 | nextime | drew___ : i use selinux and firewall on all my linux box, * or not * box. A firewall is even better if your box is exposed to internet, selinux is only an "addon" for your security. |
| 19:07.33 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:10.07 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 19:10.09 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 19:10.55 | drew___ | ok thanks |
| 19:11.16 | asterboy | Can someone please look at this pastebin and explain what is going on with the "Reversed Polarity"? |
| 19:11.19 | asterboy | http://pastebin.ca/63004 |
| 19:11.58 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:12.15 | asterboy | * Disconnect Supervision seems to use that, from what I have read: "Trying asking your Telco if they can supply you with Kewlstart or Forward Disconnect Supervision on your line. Basically, all this does is momentarily reverse the polarity on the line to indicate that the line has been disconnected. The Zaptel FXO devices detect this condition to indicate to Asterisk that the line has been disconnected." |
| 19:12.28 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:13.14 | asterboy | The problem is that when the user makes a call, the call progresses and just disappears/hangsup without actually placing the call, forcing them to have to redial. |
| 19:13.39 | asterboy | It seems to happen randomly, but from the log, it looks like it happens when more than 1 person is using the system. |
| 19:13.54 | Gabriel25 | guys .... what do you think about Linksys PAP2t ? |
| 19:14.08 | asterboy | PAP2t? |
| 19:14.23 | Gabriel25 | I want to connect 2 analog phones |
| 19:14.24 | grabowski | Gabriel25: Do you mean the PAP2-NA? |
| 19:14.25 | asterboy | PAP2-NA can be used |
| 19:14.36 | Gabriel25 | yes that one |
| 19:14.39 | asterboy | otherwise the rest are locked |
| 19:14.39 | Gabriel25 | sorry ! |
| 19:14.48 | Gabriel25 | is unlook |
| 19:14.56 | *** join/#asterisk vechers (i=vechers@64.61.117.138) |
| 19:14.58 | Gabriel25 | http://cgi.ebay.com/Linksys-PAP2t-NA-2-x-FXS-VOIP-SIP-Asterisk-ATA_W0QQitemZ9735814161QQcategoryZ61840QQtcZphotoQQcmdZViewItem |
| 19:15.03 | Gabriel25 | here this one I want to buy ! |
| 19:15.07 | *** part/#asterisk vechers (i=vechers@64.61.117.138) |
| 19:15.17 | grabowski | Gabriel25: I have one, its a nice little device. |
| 19:15.28 | Gabriel25 | so is ok if I buy that one |
| 19:15.37 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:15.45 | asterboy | should be |
| 19:15.55 | Gabriel25 | and also I want to put my analog phone line tru my asterisk so I have to buy sipura 3000 |
| 19:16.00 | Gabriel25 | which one is better ? |
| 19:16.05 | asterboy | sipura 3000 |
| 19:16.36 | Gabriel25 | Linksys/Sipura SPA-3000 VOIP SIP ASTERISK PSTN FXO FXS |
| 19:16.37 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:16.39 | Gabriel25 | this one is ok ? |
| 19:16.48 | grabowski | Gabriel25: Why not just get the Sipura SPA-3000 then? You dont need a PAP2-NA as well, unless you need two analog FXS lines |
| 19:16.48 | asterboy | sure |
| 19:17.19 | asterboy | Grandstream as a not so bad ata, but the SPA is better |
| 19:17.19 | Gabriel25 | I have 3 analog phones that I want to add to my PBX box |
| 19:18.01 | Gabriel25 | So one analog phone is going to be in Sipura 3000 and also another 2 to Linksys-PAP2t-NA |
| 19:18.07 | Gabriel25 | is that OK ? |
| 19:18.20 | grabowski | Gabriel25: Do they all need independent lines because you could power 3 phones on the one FXS port. Means only one call at a time, unless you use the flash trick. |
| 19:18.46 | asterboy | sure, but you can plug all three phones into one fxs port if you don't mind line sharing. |
| 19:18.59 | Gabriel25 | I think I need sometimes some conferince |
| 19:19.08 | Gabriel25 | so I prefer to have 3 extention setup |
| 19:19.17 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:19.19 | asterboy | conference within you home? |
| 19:19.21 | grabowski | Why not just get VoIP phones then? |
| 19:19.35 | asterboy | otherwise, you can still converence with a 3 line share. |
| 19:19.44 | Gabriel25 | I want to use 3 phones at home and also for my job another phone |
| 19:19.52 | Gabriel25 | registering with the pbx at the office |
| 19:20.17 | asterboy | common guys, someone in here has the skinny on TDM dropping calls with reverse polarity |
| 19:20.38 | asterboy | how can I disable * from using reverse polarity? |
| 19:20.40 | grabowski | Gabriel25: You could get the SPA-300 and the PAP2-NA but I would say get the SPA-3000 and then 2 VoIP phones like a Grandstream |
| 19:21.01 | asterboy | Grandstream is so frustrating |
| 19:21.12 | [TK]D-Fender | GrandSUCK |
| 19:21.14 | Gabriel25 | All ready I paid a lot of money for my analog phones |
| 19:21.16 | Gabriel25 | :)) |
| 19:21.17 | asterboy | lol, ya |
| 19:21.21 | Gabriel25 | :D |
| 19:21.32 | [TK]D-Fender | Polycom = Solid business choice. |
| 19:21.39 | grabowski | Well no one said they were the best, but they are the same quality of a household analog phone? I perfer the Cisco 7960 myself :0 |
| 19:21.40 | Gabriel25 | when I send my wife to make some shoping I`m broke ! |
| 19:21.42 | Gabriel25 | :)) |
| 19:21.50 | timscott | :))) |
| 19:21.53 | timscott | smile echo. :) |
| 19:21.53 | asterboy | [TK]D-Fender, do you know how to disable * from trying reverse polarity on a line? |
| 19:22.01 | Gabriel25 | I said to her ... she shoud change her last name to hilton ! |
| 19:22.02 | Gabriel25 | :D |
| 19:22.22 | *** join/#asterisk mfdutra (n=marlon@200.208.130.16) |
| 19:22.35 | Gabriel25 | another stupid question |
| 19:22.39 | grabowski | I have yet to try any Polycom's |
| 19:22.39 | asterboy | or do you have any ideas why * would try that if other lines are in use? |
| 19:22.48 | mfdutra | when I register to an IAX server, how do I define the context of incoming calls? |
| 19:22.59 | asterboy | Polycoms and Grandstreams on the same net, do NOT get along. |
| 19:23.10 | Gabriel25 | If someone call me at my analog phone line .. and I want asterisk to redirect that to my cell I need to have 2 phone numbers ? |
| 19:23.26 | asterboy | bridging, yes |
| 19:24.02 | grabowski | Gabriel25: Well not two phone 'numbers' but two diffrent channels to call outbound on so you can bridge, yes. |
| 19:24.06 | *** join/#asterisk charlieb31 (n=oknow31@201.144.105.87) |
| 19:24.13 | charlieb31 | rules |
| 19:24.14 | asterboy | http://pastebin.ca/63004, anybody please! |
| 19:24.18 | Gabriel25 | so verizon can do that ? |
| 19:24.41 | Gabriel25 | I have to ask them to add me another channel ? |
| 19:24.48 | Gabriel25 | I have no idea |
| 19:25.07 | grabowski | asterboy: I would try on the mailing list |
| 19:25.50 | [TK]D-Fender | asterboy : You are RECEIVING a polarity reversal from the other side. Its a signalling method |
| 19:25.55 | eBody | we have analog lines coming to our Lucent PBX, do i just need an analog adapter and an Asterisk box? |
| 19:26.21 | asterboy | TK, thanks...so that means something funny is going on with the POTS line? |
| 19:26.59 | grabowski | Gabriel25: So if you have Verizon in your FXO of the SPA-3000, I would use that as your first outbound channel and then use some VoIP outbound provider for any other channels out. |
| 19:27.17 | grabowski | Gabriel25: A pay-as-you go provider. |
| 19:27.26 | Gabriel25 | ohhh I see |
| 19:27.57 | Gabriel25 | I have at home ..... now sixtel and voipjet |
| 19:28.00 | [TK]D-Fender | asterboy : notmally thats a hangup notification. |
| 19:28.05 | asterboy | What is the phone number for * tech support? |
| 19:28.11 | grabowski | Gabriel25: Yea I would stay far away from sixtel |
| 19:28.20 | asterboy | I'll have them look at the line. |
| 19:28.22 | Gabriel25 | which one is better ? |
| 19:28.38 | grabowski | Gabriel25: Use up your minutes / money and run away from sixtel |
| 19:28.44 | eKo1 | asterboy: You mean Digium tech. support. |
| 19:28.45 | asterboy | The http://asterisk.com web site is down...only a splash screen. |
| 19:28.49 | asterboy | yes |
| 19:28.57 | eKo1 | Go to digium.com |
| 19:29.19 | asterboy | doh, I keep forgetting that they are on that URL |
| 19:29.27 | Gabriel25 | grabowski which ine if better ? |
| 19:29.37 | grabowski | Gabriel25: 1 sec |
| 19:29.44 | Gabriel25 | grabowski which one is better ? |
| 19:29.53 | Gabriel25 | I`m sorry my english is not so good |
| 19:30.06 | *** join/#asterisk cmp615 (n=cmp615@fw.cmpcs.com) |
| 19:30.07 | Gabriel25 | I start learning few months ago :) |
| 19:30.25 | MikeJ[Laptop] | asterisk.com? |
| 19:30.28 | file | we don't control asterisk.com fyi |
| 19:30.58 | asterboy | Hardware Support for Digium 256.428.6000 |
| 19:31.06 | MikeJ[Laptop] | asterisk.org! |
| 19:31.14 | file | MikeJ[Laptop]: are you a .org?!? |
| 19:31.45 | blitzrage | .orgy?!? |
| 19:31.53 | cmp615 | Can someone help me with a TE110P -> Adit600 -> POTS? I can't seem to get link between TE110 and Adit600. |
| 19:31.57 | file | blitzrage: you'd like that, wouldn't you |
| 19:31.58 | file | :p |
| 19:32.08 | blitzrage | file: you'd like that if I liked that wouldn't you? |
| 19:32.18 | file | sure |
| 19:32.42 | blitzrage | lol |
| 19:32.58 | blitzrage | so yah.. for some reason my asterisk kinda locks up at app_followme.so in trunk |
| 19:33.03 | blitzrage | as of an hour ago anways |
| 19:33.24 | jsharp | cmp615: Got a T1 crossover cable between the two? |
| 19:33.43 | cmp615 | Yep, I've tried both, and don't get any lights on either side... |
| 19:33.51 | blitzrage | <CR> does nothing... a<CR> lets it continue loading |
| 19:34.03 | eKo1 | Has anyone here tried chan_ss7? |
| 19:34.21 | FinboySlick | Anybody has a link on info as to how I might have asterisk behave as a fax server? |
| 19:34.47 | blitzrage | FinboySlick: T.38 passthrough? |
| 19:34.48 | grabowski | Gabriel25: Sorry im back. I have yet to try voip-jet but I hear good things abou them. |
| 19:35.03 | russellb | blitzrage: i see it too |
| 19:35.12 | blitzrage | russellb: ok-- so I'm not crazy... |
| 19:35.12 | russellb | blitzrage: looks like it doesn't like the default config :) |
| 19:35.29 | blitzrage | russellb: yah... I didn't do a make samples after, so I'm just missing the file all together |
| 19:35.41 | blitzrage | but it should still load :) |
| 19:35.43 | russellb | oh really ... |
| 19:35.50 | russellb | well that's odd |
| 19:35.51 | blitzrage | it just stops executing when there is no file |
| 19:35.59 | russellb | hrm. |
| 19:36.00 | blitzrage | if you type a letter then hit enter, it continues to load |
| 19:36.02 | Gabriel25 | I have voip-jet |
| 19:36.05 | FinboySlick | Well, I have a sangoma A200 and I imagine it would be relatively trivial to have it 'talk' fax as well. I know you can detect if an incoming call is a fax. Hopefully you could save the fax to a .pdf if it detects a fax, or relay the call to an internal phone if it detects a voice call. |
| 19:36.09 | russellb | blitzrage: yeah, that's bizarre |
| 19:36.11 | Gabriel25 | but I need a DID number |
| 19:36.16 | blitzrage | russellb: I thought so |
| 19:36.18 | Gabriel25 | and I had this from sixtel |
| 19:36.32 | FinboySlick | I might be in dreamland there though. |
| 19:36.38 | blitzrage | russellb: it also causes another box to not load app_dial.so for some reason |
| 19:37.11 | blitzrage | FinboySlick: asterisk can't do that |
| 19:37.24 | russellb | blitzrage: i'm gdb'ing it now ... |
| 19:37.25 | blitzrage | FinboySlick: maybe if it sent it to hylafax? Not sure if that works or not though |
| 19:37.30 | blitzrage | russellb: thanks man |
| 19:37.36 | grabowski | Gabriel25: Yea, I would suggest you find another DID provider. |
| 19:38.24 | [TK]D-Fender | FinboySlick : A200 fax support is flakey right now. Work is being done on it as its a driver issue |
| 19:38.38 | FinboySlick | Allright. |
| 19:38.43 | russellb | blitzrage: found it |
| 19:38.57 | blitzrage | russellb: schweet -- what was it? |
| 19:39.07 | russellb | silly code |
| 19:39.10 | blitzrage | russellb: what just touched almost all the files in the last hour? |
| 19:39.28 | russellb | blitzrage: some header file magic from kpfleming |
| 19:39.34 | [TK]D-Fender | blitzrage : * 1.2.9.8.6.7.5.3.0.9 !!!!! |
| 19:39.36 | blitzrage | russellb: ah ok |
| 19:39.59 | grabowski | Gabriel25: You may have better luck then me with Sixtel.. but I doubt it. |
| 19:40.11 | JASON99 | I've been trying to get threeway working with an mgcp ata but it's not working. Has anyone ever got this working or is it an asterisk bug? |
| 19:40.15 | russellb | blitzrage: asterisk is actually doing exactly what it is supposed to :) |
| 19:40.25 | russellb | blitzrage: BJ used a function which reads input from stdin ... |
| 19:40.32 | *** part/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 19:40.42 | russellb | blitzrage: so whatever you type in is actually setting a variable in app_followme :) |
| 19:40.54 | Gabriel25 | grabowski I want to add my analog phone line from verizon to my PBX box and I fix all the problems |
| 19:40.55 | Gabriel25 | :)) |
| 19:41.03 | blitzrage | russellb: oh... neat :) |
| 19:41.15 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 19:42.35 | asterboy | good ol' James from Digium is checking it out now. |
| 19:42.41 | russellb | blitzrage: committed |
| 19:42.45 | JASON99 | i guess no one uses MGCP .. lol |
| 19:42.50 | blitzrage | russellb: muchos gracious!!!! |
| 19:42.57 | asterboy | I switch from MGCP to uLaw |
| 19:43.11 | blitzrage | MGCP is a protocol... uLaw is a codec... |
| 19:43.19 | *** join/#asterisk backblue (n=moo@87-196-5-13.net.novis.pt) |
| 19:43.20 | file | and I'm a person! |
| 19:43.23 | asterboy | lo |
| 19:43.27 | blitzrage | file: lies! |
| 19:43.33 | file | fine, I'm an AI |
| 19:43.49 | blitzrage | and don't you forget that! Pesky robots |
| 19:44.39 | cmp615 | jsharp - any other ideas? |
| 19:44.42 | vader-- | ok i just wrote a script to write all the damn config files out for my cisco phones |
| 19:44.57 | vader-- | now i need to write a script to write out the sip.conf stuff for them |
| 19:45.00 | vader-- | WHOOO |
| 19:45.27 | zoa | hey ho all |
| 19:45.32 | blitzrage | zoa: !!!! |
| 19:45.37 | file | zoa: ACK |
| 19:45.38 | zoa | yes hon! |
| 19:45.42 | blitzrage | zoa: I miss you |
| 19:45.43 | zoa | i dont do sip |
| 19:45.56 | zoa | ssst nobody needs to know those details |
| 19:45.58 | file | zoa: :\ AUTHREQ |
| 19:46.04 | jsharp | cmp615: Framing & line coding match on both ends? |
| 19:46.10 | zoa | vnak |
| 19:46.13 | blitzrage | zoa: lol |
| 19:46.15 | jsharp | Ports are enabled? Zaptel is loaded correctly? |
| 19:46.21 | blitzrage | zoa: coming to any of the astricon europes? |
| 19:46.28 | zoa | doesnt look like it |
| 19:46.39 | blitzrage | doh! No one in Europe likes going to conferences |
| 19:46.41 | zoa | i will try to though |
| 19:46.49 | zoa | well its very expensive |
| 19:46.52 | zoa | atm |
| 19:47.02 | file | that's silly, rob a bank! |
| 19:47.08 | cmp615 | As far as I can tell. Zaptel shows both cards - TDM24xx and the TE110P - cat/proc/zaptel/2 shows all channels available... |
| 19:47.14 | zoa | to be good i'd want to go to all 3 |
| 19:47.23 | zoa | but that would be insanely expensive for me |
| 19:47.28 | blitzrage | zoa: understand -- yah.. I hear that |
| 19:47.35 | zoa | and for just one currently its also very expensive |
| 19:47.43 | zoa | because i would meet only 1/3 of the people |
| 19:47.48 | blitzrage | true |
| 19:47.56 | zoa | my minimum flight cost is 400 euro |
| 19:47.58 | zoa | add a hotel |
| 19:48.04 | zoa | the cost of the car to get there |
| 19:48.05 | zoa | etc etc |
| 19:48.10 | zoa | its 1000 euro |
| 19:48.13 | zoa | for 2 days |
| 19:48.19 | zoa | and i have 200 on my bank account |
| 19:48.25 | blitzrage | which is not 1000 :) |
| 19:48.30 | zoa | yes |
| 19:48.31 | zoa | :) |
| 19:48.33 | blitzrage | your bank account sounds like mine :) |
| 19:48.36 | blitzrage | only better |
| 19:48.42 | blitzrage | since it's euros :) |
| 19:48.48 | zoa | haha |
| 19:48.49 | timscott | ohh snap! |
| 19:48.50 | timscott | euro ftw. |
| 19:48.58 | file | silly old people who don't save money |
| 19:49.14 | zoa | look who's talking |
| 19:49.35 | file | hey now, I save :D a lot. |
| 19:49.36 | blitzrage | gotta have money to save it |
| 19:49.45 | blitzrage | I obviously don't charge enough |
| 19:50.00 | file | blitzrage: good point, you should charge more! |
| 19:50.43 | cmp615 | jsharp - I've got the Adit sending the 8FXO as channels 1-8, fxs as 9-16, 24 is data. I've got framing/coding as ESF/B8ZS on both ends. |
| 19:51.15 | file | everybody dance, put your hands ... er no |
| 19:51.28 | lunk | down /my/ pants |
| 19:51.49 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 19:52.20 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net) |
| 19:52.36 | gambolputty | Hi. Can anyone dial me via SIP for a test call? |
| 19:53.19 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.4.154) |
| 19:56.34 | *** join/#asterisk techie (n=gus@antibala.com) |
| 19:56.41 | jsharp | cmp615: Can you get lights if you stick T1 loopback plugs in? |
| 19:57.03 | vader-- | do you guys recommend canreinvite to be on or off in your sip.conf? |
| 19:57.14 | funxion | jsharp yes |
| 19:57.23 | Hmmhesays | lets get drunk and be somebody |
| 19:57.27 | JackEstorm | this is the oddest thing, it seems like chan_agent is really buggy ...but I doubt that it could still be this buggy with everyone using it. |
| 19:59.38 | gambolputty | vader: Are your SIP phones behind a NAT firewall? |
| 20:00.34 | vader-- | na |
| 20:00.39 | vader-- | they are all internal on a network |
| 20:00.41 | *** join/#asterisk tekmaven (n=tekmaven@ool-45710bcf.dyn.optonline.net) |
| 20:00.46 | grabowski | vader--: If you can spare the bandwidth then I say always off. |
| 20:00.56 | grabowski | vader--: Is the asterisk box on the local nextwork? |
| 20:00.56 | tekmaven | hey guys |
| 20:01.01 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.150) |
| 20:01.04 | vader-- | ya |
| 20:01.15 | grabowski | vader--: Then don't worry about it. |
| 20:01.20 | gambolputty | that usually would be behind a firewall, in which case canreinvite would be off. |
| 20:01.20 | grabowski | *Off |
| 20:01.29 | vader-- | leave it on or off? |
| 20:01.35 | grabowski | vader--: off |
| 20:01.38 | vader-- | ok |
| 20:04.09 | Bert- | well |
| 20:04.47 | Bert- | as I can read in the book, Asterisk is unable to deal with MGCP VoIP provider ?? |
| 20:05.05 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 20:05.16 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 20:05.16 | grabowski | Bert-: http://www.voip-info.org/wiki/index.php?page=Asterisk+MGCP+channels |
| 20:05.17 | cmp615 | jsharp - I just made a loopback and the Adit loops up, but the TE110P doesn't... |
| 20:05.21 | eKo1 | an on T1 line, the dchan is on chan 24? |
| 20:06.30 | *** join/#asterisk moprilo (n=jjohn@201.192.107.57) |
| 20:06.54 | tzanger | on a T1 PRI, yes |
| 20:07.06 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 20:07.07 | moprilo | hi, what's the difference between puting, exten => _130,1,Macro(stdexten,SIP/130,,130) and exten => 130,1,Macro(stdexten,SIP/130,,130) |
| 20:07.22 | moprilo | exten => _XXX vs exten => XXX |
| 20:07.32 | Bert- | :( |
| 20:07.38 | jsharp | cmp615: You've got a configuration problem on your TE110P, then. |
| 20:07.43 | Bert- | fucking french exception :( |
| 20:07.53 | Bert- | 'ca pue du cul" |
| 20:08.16 | [TK]D-Fender | Bert- : lol, pauvre-toi sti! |
| 20:08.29 | Bert- | for sure :( |
| 20:08.48 | Bert- | I could code it but I've no time :( |
| 20:09.11 | Bert- | I'll add some ⏠to boundary .. :) |
| 20:09.39 | cmp615 | jsharp - that's where I'm thinking I have a problem...any ideas of what to look for? |
| 20:09.47 | Bert- | anyway your work guy is cery cool !! :) |
| 20:09.54 | Bert- | guys |
| 20:10.04 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 20:10.35 | jsharp | cmp615: The obvious stuff first. Are the modules for the TE110P loaded? Do you have the appropriate "span" lines in your zaptel.conf? |
| 20:11.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net) |
| 20:11.29 | *** part/#asterisk ness (n=Tom@pppin-39-b6.pop-kaltenengers.rz-online.NET) |
| 20:11.38 | *** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
| 20:12.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.228) |
| 20:12.17 | drew___ | does the cvs login posted on http://www.voip-info.org/wiki/view/Asterisk+Step-by-step+Installation no longer work? |
| 20:12.32 | cmp615 | jsharp - TE110P appears loaded with "cat /proc/zaptel/2" shows the card with all 24 channels... |
| 20:12.35 | timscott | http://www.voip-info.org/wiki/view/Asterisk+Crossbar+Installation |
| 20:12.56 | grabowski | drew___: any reason you don't want to use svn? |
| 20:13.09 | drew___ | need to install svn first |
| 20:13.15 | moprilo | anyone know what difference does it make to add a '_' in front of the extension in the extensions.conf? |
| 20:13.46 | jsharp | Weird |
| 20:13.52 | [TK]D-Fender | moprilo : For what you showed a leading _ does nothing because it isn't a pattern, but rather a fixed value. |
| 20:13.56 | grabowski | drew___: Yea, I don't know if Asterisk.org offers CVS anymore. The svn instructions are on the official website. |
| 20:14.10 | file | CVS was taken down a few weeks ago |
| 20:14.18 | file | no longer supported or available |
| 20:14.19 | eKo1 | Yeah, cvs is not working anymore. |
| 20:14.27 | eKo1 | which is a good thing. |
| 20:15.12 | grabowski | moprilo: _ says its going to be a pattern and your not doing any pattern matching with that other exten |
| 20:15.23 | moprilo | ok .. |
| 20:15.39 | moprilo | but what dif does it make in somethine like this.. exten => _9. ? |
| 20:15.54 | *** join/#asterisk kph100 (n=kph100@206-248-134-237.dsl.teksavvy.com) |
| 20:16.05 | grabowski | moprilo: I suggest you read http://www.voip-info.org/wiki/index.php?page=Asterisk%20Dialplan%20Patterns |
| 20:16.17 | moprilo | excelent thanks |
| 20:16.24 | [TK]D-Fender | moprilo : With = works, without = NO. |
| 20:16.42 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 20:17.12 | tekmaven | hey guys |
| 20:17.27 | tekmaven | in my dial plan, is it possible to call more then one extention at once? |
| 20:18.02 | cmp615 | jsharp - what should I be looking for in zapata/zaptel? |
| 20:19.00 | *** join/#asterisk OuterSpace (n=me1@168.226.3.194) |
| 20:19.05 | vader-- | tekmaven yes |
| 20:19.12 | vader-- | just put a & between the two |
| 20:19.17 | tekmaven | really? |
| 20:19.19 | tekmaven | awesome :) |
| 20:19.19 | vader-- | ya |
| 20:19.21 | vader-- | it's that simple |
| 20:19.24 | cmp615 | jsharp - I've got span=2,1,0,esf,b8zs |
| 20:19.26 | tekmaven | finally, something simple ;) |
| 20:19.28 | grabowski | tekmaven: Dial(Sip/kelly&IAX2/bob) |
| 20:19.32 | mcf3782 | tekmaven - sure... use something like this: exten => 123,1,Dial(Zap/1&Zap/2&Zap/3) |
| 20:19.38 | vader-- | hehe finally i was able to help someone |
| 20:19.58 | tekmaven | exten => t,1,Dial(IAX2/homeiaxy) & Dial(IAX2/ryaniaxy) |
| 20:20.03 | tekmaven | would that be right? |
| 20:20.05 | OuterSpace | hello, how can i save all calls ? (give me somthing to improve my google search because im not getting any good result) |
| 20:20.25 | vader-- | hehe i wanna write something i can do to ring every phone connected and play the sound file, weasles have eaten out telephone system |
| 20:20.33 | vader-- | out = our |
| 20:20.37 | vader-- | when they pick up |
| 20:20.52 | vader-- | tekmaven n |
| 20:20.54 | vader-- | o |
| 20:20.58 | OuterSpace | to have all calls in .wav or something like that |
| 20:21.06 | [TK]D-Fender | tekmaven : exten => t,1,Dial(IAX2/homeiaxy&IAX2/ryaniaxy) |
| 20:21.09 | vader-- | exten => t,1,Dial(IAX2/homeiaxy&IAX2/ryaniaxy) |
| 20:22.30 | cmp615 | jsharp - would bchan=1-23 or 25-47 since it's on the second span, and the first span is a tdm2400 card using channels 1-24...this is where I think I may have the issue... |
| 20:23.47 | mcf3782 | I'm reading the User's Manual that came with my TDM400 card.. There's a sample dialplan printed in there. I'm trying to follow along and understand it. I understand all but one line. |
| 20:24.03 | mcf3782 | exten => _9.,1,Dial(zap/g2/www${EXTEN:1}) |
| 20:24.48 | mcf3782 | what's the "www${EXTEN:1}" part mean? |
| 20:25.03 | mcf3782 | I can't find any docs anywhere that explain that. |
| 20:27.20 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 20:27.22 | noky | hi buddies |
| 20:27.37 | noky | how can i put a extensions that wait me 10 seconds ? |
| 20:27.41 | noky | Wait(10) ? :P |
| 20:28.04 | vader-- | you want an extension that just makes you sit there for 10 seconds? |
| 20:28.04 | grabowski | mcf3782: the www appear to tell it to wait.. thats news to me http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 20:28.16 | vader-- | trying to annoy people noky? |
| 20:28.18 | vader-- | hehe |
| 20:28.35 | noky | yes darth vader |
| 20:28.47 | Hmmhesays | hmm kustom floor monitors, i wonder if they're any good |
| 20:28.47 | noky | lord sith |
| 20:28.48 | vader-- | im going to make an extension that when you dial it on our phone system it rings every phone on the network and when you pick up it says do you wanna hear the most annoying sound in the world: AHHHHHHHHHHHHHHHHHHHHHHHH!!!!!!!!!! |
| 20:29.21 | vader-- | that would be ammusing |
| 20:29.38 | grabowski | mcf3782: You understand the ${EXTEN:1} part right? |
| 20:30.20 | grabowski | noky: Just Wait(10) in your dialplan. |
| 20:30.58 | grabowski | noky: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Wait |
| 20:32.22 | mcf3782 | grabowski - thanks for the URL to that wiki page. My googling hadn't turned that one up yet, and the text in "show application Dial" didn't mention 'w' anywhere. |
| 20:32.50 | grabowski | mcf3782: Yea, as I said that w was news to me.. :) |
| 20:33.36 | OuterSpace | how can i save all incoming/outgoing calls in .wav or something like that ? |
| 20:34.26 | grabowski | OuterSpace: http://www.voip-info.org/wiki/view/MixMonitor |
| 20:34.27 | mcf3782 | as for the ${EXTEN:1} part.. what I *think* that means is to send the digits after the '9' to the zap/g2 group.. |
| 20:34.30 | mcf3782 | Am I close? |
| 20:34.36 | noky | thanks grabowski |
| 20:35.31 | grabowski | mcf3782: Yes, so if you dialed 92345678 EXTEN:1 says remove the first digit. |
| 20:35.44 | grabowski | mcf3782: 2345678 |
| 20:36.05 | mcf3782 | Cool. I'm slowly getting my head around this then. :) |
| 20:36.09 | mcf3782 | Thank you. :) |
| 20:36.13 | Bullseye_Network | Is there a way to tell in the asterisk log which end on the call hungup first? |
| 20:36.31 | grabowski | mcf3782: np. |
| 20:40.29 | mcf3782 | And, just for my own sanity check.. the 'g2' group is what's defined in the section called "Group=2" in /etc/zapata.conf. Correct? |
| 20:44.44 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 20:45.02 | cmp615 | jsharp - you there? |
| 20:45.20 | r_evolution | wake up heads. |
| 20:47.14 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:51.20 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve) |
| 20:51.23 | grabowski | mcf3782: yes I believe so. |
| 20:51.41 | epablo | Hi guys and gals |
| 20:51.55 | mcf3782 | scary. I'm not used to being right this many times in a day. ;) |
| 20:51.59 | grabowski | mcf3782: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 20:52.51 | cmp615 | can anyone help with a TE110P? It doesn't seem to connect to a channel bank... |
| 20:53.02 | MikeJ[Laptop] | cmp615, digium support can |
| 20:53.35 | cmp615 | Wasn't sure if they'd help with channel banks... |
| 20:53.58 | r_evolution | ok wtf |
| 20:54.00 | r_evolution | Got SIP response 415 "Unsupported Media Type" back from |
| 20:54.10 | r_evolution | if it's not retarded problem |
| 20:54.12 | r_evolution | it's another... |
| 20:54.13 | grabowski | mcf3782: If your still sort of new to Asterisk I suggest you read "Asterisk: The Future of Telephony" a O'Reilly book. They have a free PDF version (under the Creative Commons license) you can get the PDF at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:54.23 | r_evolution | ~books |
| 20:54.26 | r_evolution | ~thebooks |
| 20:54.28 | r_evolution | ~thebook |
| 20:54.30 | r_evolution | which one is it today? |
| 20:54.35 | r_evolution | ~book |
| 20:54.36 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:54.42 | blitzrage | lol |
| 20:54.48 | r_evolution | you know... people keep changing that. |
| 20:54.49 | r_evolution | seriously. |
| 20:54.52 | blitzrage | why? |
| 20:54.56 | MikeJ[Laptop] | naw.. don't download it.. go buy a copy |
| 20:54.58 | blitzrage | it should just be ~thebook |
| 20:54.58 | r_evolution | i dont know why |
| 20:55.04 | blitzrage | yah -- buy it! :D |
| 20:55.04 | r_evolution | I download and buy |
| 20:55.06 | r_evolution | haha |
| 20:55.09 | r_evolution | you WOULD say that |
| 20:55.09 | mcf3782 | I've got that book. And I started there, then went to google. |
| 20:55.17 | blitzrage | r_evolution: who... little ol' me? :D |
| 20:55.20 | r_evolution | i started with the book |
| 20:55.21 | r_evolution | then i said |
| 20:55.23 | Bullseye_Network | Is there a way to tell in the asterisk log which end on the call hungup first? |
| 20:55.26 | r_evolution | man... this LIFE guy |
| 20:55.28 | r_evolution | he's crazy. |
| 20:55.29 | r_evolution | ;) |
| 20:55.31 | blitzrage | lol |
| 20:55.32 | mcf3782 | I've read it cover to cover. Just needed some clarification. |
| 20:55.35 | blitzrage | you spelled his name wrong |
| 20:55.43 | r_evolution | i spells it like i pronounces it |
| 20:55.46 | blitzrage | but pronounced it right :) |
| 20:55.52 | r_evolution | see above :) |
| 20:55.55 | blitzrage | :D |
| 20:55.56 | blitzrage | hehehe |
| 20:56.05 | blitzrage | I heard he's the COOLEST |
| 20:56.10 | r_evolution | i liked it too... i would never have done ANYTHING without that book |
| 20:56.11 | blitzrage | picks up all the girls |
| 20:56.13 | r_evolution | serious. |
| 20:56.13 | file | who is?!? |
| 20:56.16 | epablo | Is there something like the gnugk for SIP. I need a load balancer. |
| 20:56.18 | file | blitzrage: HAHAHAHAHAHA |
| 20:56.20 | r_evolution | some creepy life kid |
| 20:56.21 | r_evolution | ;) |
| 20:56.26 | blitzrage | epablo: SER |
| 20:56.31 | grabowski | Bullseye_Network: I think you will need to write a custom API for that. You may be able to log it with a custom CDR but I'm not sure. Check http://www.voip-info.org/wiki/view/Asterisk+billing |
| 20:56.34 | r_evolution | oh here's a good one for you 'life' |
| 20:56.47 | r_evolution | one of the guys here asked for the SOPs for the * box we've got here |
| 20:56.50 | r_evolution | i handed him teh book |
| 20:56.53 | r_evolution | and said... go read :) |
| 20:56.57 | Bullseye_Network | grabowski: thx |
| 20:57.05 | MikeJ[Laptop] | ~thebox |
| 20:57.06 | jbot | Set of scripts for and installing managing IP Masq and Transparent caching.. URL: http://yak.airwire.net/ |
| 20:57.09 | epablo | blitzrage: I was hoping I would get another answer.. ;) I hate SER |
| 20:57.17 | MikeJ[Laptop] | hmmm |
| 20:57.22 | MikeJ[Laptop] | ser is good |
| 20:57.27 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 20:57.33 | blitzrage | epablo: search for "Sip proxy" in google then and see what it returns :) |
| 20:57.35 | epablo | Think I'll use h323 and gnugk |
| 20:57.51 | blitzrage | I still haven't gotten my head around SER |
| 20:57.52 | Bullseye_Network | that http://yak.airwire.net/ doesnt work for me. :) |
| 20:58.00 | blitzrage | it just feeels... cheap or unstable or something |
| 20:58.00 | grabowski | Yea the Asterisk book is really good and worth buying but I like the fact you can download and read all you want before buying. |
| 20:58.10 | r_evolution | psst... i still haven't gotten my head around these piece of shit UT Starcom ATAs. |
| 20:58.15 | Hmmhesays | any audiophiles in here right now? |
| 20:58.21 | blitzrage | Hmmhesays: depends what you need |
| 20:58.23 | file | blitzrage: need to warp your head around what it really is... |
| 20:58.23 | r_evolution | SERIOUSLY. If they're not having one problem it's another ;x |
| 20:58.42 | blitzrage | file: yah... need to take a class on it or something :) |
| 20:58.44 | epablo | blitzrage, MikeJ[Laptop]: I never said it was bad. I just don't like te approch. config file, .. |
| 20:58.48 | r_evolution | some days it's just not worth chewing through the straps :-\ |
| 20:58.50 | Hmmhesays | Need to pick up some floor monitors, looking at these 15' kustoms on musiciansfriend for $120 |
| 20:59.11 | file | it doesn't route calls, it routes SIP packets ^_^ that's the thing that BlOwS the minds of teh people |
| 20:59.12 | blitzrage | epablo: I never said it was good *or* bad :D |
| 20:59.16 | r_evolution | mackie! ;x |
| 20:59.23 | blitzrage | Hmmhesays: what brand? |
| 20:59.27 | Hmmhesays | kustom |
| 20:59.35 | epablo | blitzrage: thats right.. sorry ;) |
| 20:59.37 | r_evolution | mackie? mackie anyone? mackie? |
| 20:59.43 | blitzrage | Hmmhesays: hrmmm... not really familiar with that brand |
| 20:59.49 | Hmmhesays | yeah they're kind of an off brand |
| 20:59.58 | Hmmhesays | i can't afford $400 15' jbl's |
| 21:00.04 | r_evolution | www.mackie.com |
| 21:00.05 | r_evolution | www.mackie.com |
| 21:00.07 | r_evolution | :-D |
| 21:00.09 | blitzrage | Hmmhesays: only real way to know is to test them and determine if the frequency range is good enough for your application |
| 21:00.14 | *** join/#asterisk terrapen_ (n=cjs@166.70.135.60) |
| 21:00.23 | Hmmhesays | application, live band monitors |
| 21:00.59 | r_evolution | Hmmhesays... if you REALLY want high quality... you will sell h0z until you can afford mackie |
| 21:01.05 | blitzrage | hrmmm.. yah... they probably don't need to be great... just able to handle the power you provide them, and a good amp to power them, and a good EQ to filter out the frequencies to lower the feedback |
| 21:01.07 | Bert- | hmm i've an error when trying to connect to a sip server with asterisk :( |
| 21:01.10 | drew___ | why do i get "no such command 'zap'" for "zap show channels" on the * cli ? |
| 21:01.14 | Bert- | is a way to put some log ? |
| 21:01.28 | blitzrage | drew___: chan_zap.so isn't loaded |
| 21:01.31 | r_evolution | drew... do you |
| 21:01.33 | r_evolution | yeah what he said |
| 21:01.37 | blitzrage | Bert-: sip debug |
| 21:01.39 | r_evolution | ztdummy if nothing else mang. |
| 21:01.56 | Bert- | chan_sip.c:5267 sip_reg_timeout: -- Registration for '0872354774@freephonie.net' timed out |
| 21:01.58 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 21:02.04 | r_evolution | i guess that'd be the problemo. |
| 21:02.08 | Hmmhesays | bah JBL all the way |
| 21:02.12 | r_evolution | MACKIE! |
| 21:02.12 | blitzrage | :) |
| 21:02.14 | r_evolution | :) |
| 21:02.28 | drew___ | r_evolution/blitzrage - how do i load it? |
| 21:02.29 | blitzrage | Elite! |
| 21:02.42 | blitzrage | drew___: you compile zaptel and load the ztdummy driver |
| 21:02.44 | mackie_or_die | drew -- do you HAVE a zap card? |
| 21:02.48 | Bert- | blitzrage, it's ever done. |
| 21:02.54 | drew___ | mackie - yap |
| 21:02.54 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:02.58 | Bert- | just to know I is a website to put logs |
| 21:02.59 | mackie_or_die | if not... you had to load the ztdummy at the beginning |
| 21:03.05 | mackie_or_die | which means you gotta edit the makefile |
| 21:03.08 | blitzrage | ~pb |
| 21:03.10 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 21:03.17 | Bert- | thx :) |
| 21:03.54 | blitzrage | schweet |
| 21:03.59 | blitzrage | wet tshirt contest |
| 21:04.04 | r_evolution | ... |
| 21:04.05 | mitcheloc | where? |
| 21:04.05 | r_evolution | nooooo |
| 21:04.06 | blitzrage | lol |
| 21:04.08 | Bullseye_Network | suddenly having disconnect problems with 1.2.9.1 |
| 21:04.40 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 21:05.18 | grabowski | Bullseye_Network: With? |
| 21:05.20 | *** join/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 21:05.25 | Bert- | well my pb seems to be about registration |
| 21:05.51 | Bert- | but as I can see in debug window, server returns SIP error code 401 |
| 21:06.00 | Bert- | http://pastebin.com/766131 |
| 21:06.11 | r_evolution | well |
| 21:06.11 | Bert- | if someone want to see :) |
| 21:06.14 | r_evolution | im leaving suckas |
| 21:07.05 | Bullseye_Network | here are some of the errorshttp://www.bullseyenetworks.com/1291.log |
| 21:07.51 | Bullseye_Network | its going NUTZ |
| 21:08.29 | Bullseye_Network | It cant find beep.gsm in the sounds directory and it IS there |
| 21:08.41 | grabowski | Bullseye_Network: permissions screwed up? |
| 21:08.51 | Bullseye_Network | Nothing has changed |
| 21:08.59 | Bullseye_Network | its been running all day |
| 21:09.03 | Bullseye_Network | just started this |
| 21:09.35 | Bullseye_Network | this one worries me: Failed to create pipe: Too many open files |
| 21:09.36 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:09.44 | grabowski | I'm out of ideas.. try the bugtrack? |
| 21:10.22 | Dr-Linux | Bullseye_Network: looks like you have permissions problem |
| 21:10.23 | mcf3782 | I'd think the 'Too many open files' message would be the first thing to track down. I'd bet solving that would fix the other issue. |
| 21:10.35 | mcf3782 | Is the box out of space in /tmp? |
| 21:10.43 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:10.46 | Dr-Linux | Bullseye_Network: asterisk is running as root? |
| 21:11.03 | Bullseye_Network | ummmm... Yes |
| 21:11.10 | Bullseye_Network | :) |
| 21:11.11 | Gabriel25 | if I select a plan from teliax this I have only inbund ? or I have outbound to ? |
| 21:11.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:12.19 | *** part/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 21:14.43 | lifendel | Bullseye_Network, I have seen that error before on the mailing list |
| 21:14.48 | Bullseye_Network | I up'ed the number of allowed open files |
| 21:14.52 | Bullseye_Network | and it didnt help |
| 21:15.05 | lifendel | Bullseye_Network, there's is a kernel setting for file handles |
| 21:15.30 | lifendel | have you adjusted that kernel parameter? |
| 21:15.38 | Bullseye_Network | Why would this new version cause this problem? |
| 21:15.51 | *** join/#asterisk puppet (n=puppet@1-1-3-3d.ox.mlm.bostream.se) |
| 21:15.54 | Bullseye_Network | It seems there is a problem with closing something in 1.2.9.1 |
| 21:16.13 | Bullseye_Network | Never had this problem b4 |
| 21:16.29 | Bullseye_Network | Going to have to roll back |
| 21:16.48 | Dr-Linux | Bullseye_Network: you mean going back to old version? :) |
| 21:16.53 | Bullseye_Network | ulimit -a was set to 1024 files |
| 21:17.05 | Bullseye_Network | so I set to ulimit -n 8192 |
| 21:17.08 | Bullseye_Network | and still same problem |
| 21:17.12 | Bullseye_Network | I have to.... |
| 21:17.17 | Bullseye_Network | roll back |
| 21:17.17 | tzafrir_laptop | Bullseye_Network, do you have around 125 or 250 or 500 concurrent calls? |
| 21:17.23 | Bullseye_Network | I have 60 people out there freaking out |
| 21:17.41 | lifendel | hold on a sec, I'll pulling up the solution |
| 21:17.43 | Bullseye_Network | usually 50-70 |
| 21:17.58 | *** join/#asterisk jart (n=jart@justin.ctlinc.com) |
| 21:18.16 | tzafrir_laptop | Bullseye_Network, ls /proc/`cat /varu/ruv/asterisk.pid`/fd | wc |
| 21:18.18 | Bullseye_Network | right now theres 40 people logged in to the queue |
| 21:18.26 | tzafrir_laptop | what number do you get? |
| 21:18.38 | tzafrir_laptop | This is how many open file Asterisk has |
| 21:18.45 | Bullseye_Network | says 0 0 0 |
| 21:19.10 | Bullseye_Network | ok varu? |
| 21:19.13 | tzafrir_laptop | get the correct pid instead of `cat /var/run/asterisk.pid` |
| 21:19.14 | *** join/#asterisk loud (n=ariel@omfg.wtf.no) |
| 21:19.16 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 21:19.20 | *** join/#asterisk JINDAL (n=trojan@220.226.4.154) |
| 21:19.36 | grabowski | Bullseye_Network: You did a asterisk shutdown and then started it up? The restart command no longer does a clean restart just a reload. |
| 21:19.55 | Bullseye_Network | 2405 |
| 21:20.47 | tzafrir_laptop | 2405 is the first number? (the number of lines) |
| 21:21.08 | Bullseye_Network | cat /var/run/asterisk.pid says 2405 : cat /var/run/asterisk.pid | wc says 1 1 5 |
| 21:21.15 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 21:21.19 | Bullseye_Network | 2405 is the duh. |
| 21:21.23 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 21:21.38 | Bullseye_Network | process |
| 21:21.51 | tzafrir_laptop | ls /proc/2405/fd | wc |
| 21:22.30 | Bullseye_Network | 1018 1018 3984 |
| 21:22.32 | lifendel | damn, I can't find the article.. There was actually a file in /proc you need to modify to up the global file handle limit.. I'm still searching. |
| 21:23.04 | Bullseye_Network | why do I need to change something I;ve been running with this many people for along time |
| 21:23.27 | tzafrir_laptop | But why should he be out of file descriptors for ~70 calls? |
| 21:23.48 | tzafrir_laptop | Looks like a leak. If it is a leak, increasing the limit won't fix is |
| 21:24.35 | Bullseye_Network | NOW im shoing 0 0 0 |
| 21:24.38 | Bullseye_Network | it just died |
| 21:25.05 | Bullseye_Network | asterisk completely dumped. |
| 21:25.12 | Bullseye_Network | its now on PID 8452 |
| 21:25.29 | tzafrir_laptop | Next time it gives you that error, ls -l /proc/PID/fd |
| 21:25.44 | tzafrir_laptop | and pastebin whe result |
| 21:25.51 | Bullseye_Network | its back to 372 372 1378 |
| 21:25.58 | tzafrir_laptop | maybe it could help tracing the problem |
| 21:26.18 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 21:27.21 | Dr-Linux | file table overflowed |
| 21:27.37 | Bullseye_Network | That sux... |
| 21:28.14 | Dr-Linux | Bullseye_Network: wht distro you are on? |
| 21:28.36 | Damin | ~jbot centosbug |
| 21:28.40 | jbot | i guess centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit. |
| 21:29.04 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 21:29.11 | Bullseye_Network | debian kernel 2.6.8-11-em64t-p4-smp |
| 21:30.29 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 21:31.05 | sleepy_one | hey all |
| 21:31.14 | Dr-Linux | sleepy_one: hey there |
| 21:32.08 | Dr-Linux | Bullseye_Network: did you try stopping asterisk and start again? |
| 21:32.23 | Bullseye_Network | <PROTECTED> |
| 21:32.37 | sleepy_one | Bullseye_Network, looks like you ran out of space in your kernel file table |
| 21:32.46 | sleepy_one | too many open files |
| 21:33.05 | Bullseye_Network | asterisk[7446]: segfault at 0000000000000048 rip 000000000041b890 rsp 00000000406ff5f0 error 4 |
| 21:33.37 | Bullseye_Network | yup |
| 21:33.59 | sleepy_one | what kernel version are you running? |
| 21:34.14 | Bullseye_Network | it ran with 50+ people for 5 hours and then crashed |
| 21:34.33 | Bullseye_Network | debian distro kernel 2.6.8-11-em64t-p4-smp |
| 21:34.59 | Bullseye_Network | Did NOT have a problem with 1.2.7.1 |
| 21:35.36 | sleepy_one | did you lsof to see who's hogging the file table? |
| 21:38.28 | Bullseye_Network | has to be asterisk. I upgraded to 1.2.9.1 last night |
| 21:38.34 | Bullseye_Network | never had a problem before |
| 21:39.12 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 21:39.21 | Jaxxan | hey guys |
| 21:39.37 | Jaxxan | so OMFG management sucks |
| 21:39.46 | Jaxxan | they're all, we dont like your hold music |
| 21:39.50 | feld_ | Bullseye_Network, debian's a bit bleeding edge don't you think? _I_ think it's that crazy whack distro of yours |
| 21:39.54 | feld_ | :P |
| 21:39.56 | Jaxxan | play our company single over and over again instead |
| 21:40.02 | feld_ | Jaxxan, LOL! |
| 21:40.10 | feld_ | that's terrible. |
| 21:40.11 | Jaxxan | erm... single == jingle |
| 21:40.19 | Bullseye_Network | I've been running this same server here for 10 months |
| 21:40.24 | Jaxxan | yeah |
| 21:40.27 | Bullseye_Network | just upgrading the asterisk versions |
| 21:40.38 | feld_ | start stabbing people Jaxxan . at least you'll get news coverage. |
| 21:40.52 | grabowski | lol |
| 21:40.58 | sleepy_one | Bullseye_Network, it could be asterisk or it could be something else, lsof will tell you |
| 21:41.06 | X-Rob | ?centosbug |
| 21:41.09 | X-Rob | ~centosbug |
| 21:41.11 | jbot | hmm... centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit. |
| 21:41.11 | Jaxxan | so the local radio station broadcasts to on of my servers where i gotta shoutcast setup |
| 21:41.18 | sleepy_one | find out what's keeping the files open and fix it |
| 21:41.45 | Jaxxan | i wanna use that instead and tell management to shove it up their arse |
| 21:42.03 | Bullseye_Network | Nothing has changed on this server other than I put 1.2.9.1 on it. So im gonna have to roll it back to 1.2.7 |
| 21:46.13 | Bullseye_Network | I'll roll it back to 1.2.8 that was working thats what I upgraded from |
| 21:48.13 | grabowski | FWD down for anyone else? |
| 21:48.18 | grabowski | FWD IAX2 rather |
| 21:48.25 | Sedorox | its always down |
| 21:48.27 | jarrod | sometimes during a call when on hold or initial calls i get little noises like 'psssh' |
| 21:48.35 | grabowski | Sedorox: lol |
| 21:48.37 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 21:50.14 | Bullseye_Network | the files are not being closed by the voicemail it looks like |
| 21:50.31 | Bullseye_Network | asterisk 8452 root 246u REG 8,7 0 10652315 /var/spool/asterisk/voicemail/default/600/tmp/9 |
| 21:50.31 | Bullseye_Network | QgpZA |
| 21:50.35 | X-Rob | jbot, no, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:50.36 | jbot | X-Rob: okay |
| 21:51.09 | X-Rob | jbot, redhatbug is is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:51.10 | jbot | okay, X-Rob |
| 21:51.30 | feld_ | X-Rob, you've trained him well |
| 21:51.58 | X-Rob | ~centosbug |
| 21:51.59 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:52.06 | Bullseye_Network | Found the filehandle problem |
| 21:52.28 | CunningPike | Bullseye_Network: Do tell - I'm interested |
| 21:53.18 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 21:53.34 | Bullseye_Network | I have a script that clears all the mailbox messages every few mins. Because we dont care what people put in this mailbox. BUT... |
| 21:54.04 | Bullseye_Network | Its trying to delete the files BEFORE they complete the voicemail message it looks like. So its leaving a tmp file open. |
| 21:54.13 | Bullseye_Network | Thats what I came up with |
| 21:54.23 | mitcheloc | Bullseye_Network: check the time stamp? only clear old ones out...? |
| 21:54.24 | CunningPike | Bullseye_Network: Aha - sounds reasonabe |
| 21:54.32 | jarrod | anyone have a problem with a little 1 second hiss every now and then on their calls |
| 21:54.46 | Bullseye_Network | tharight now there are 62 files open in THAT mailbox that are open. |
| 21:54.52 | Bullseye_Network | BUT theres no one leaving a message |
| 21:55.13 | feld_ | http://sh.nu/p/1948 <-sip.conf http://sh.nu/p/1947 <-zapata.conf http://sh.nu/p/1945 <- extensions.conf --------- Can someone lend me a hand??? I have a few questions that need experienced user's assistance =( |
| 21:55.21 | CunningPike | jarrod: That sounds like latency - are they SIP-to-SIP calls? Or PSTN? |
| 21:55.31 | jarrod | sip to pstn |
| 21:55.38 | jarrod | when people are talking |
| 21:55.39 | jarrod | its ok |
| 21:55.41 | CunningPike | jarrod: What card |
| 21:55.41 | jarrod | but when its silence |
| 21:55.44 | jarrod | its a hiss |
| 21:55.45 | Bullseye_Network | So right now Im just NOT taking a message on that box. |
| 21:55.55 | jarrod | well its sip -> sip(cisco gateway)pstn |
| 21:55.56 | Bullseye_Network | BUT its a new problem with this version |
| 21:56.05 | Bullseye_Network | because we have been doing it that was for months |
| 21:56.45 | bcnl | I have a intermittent problem with sip/iax calls where I get like 1 second of audio and 1 second of silence alternating. I have great latency between endpoints and can't seem to find a bandwidth related root for this. Has anyone seen/heard something similar, or has any idea on how I can start to troubleshoot? |
| 21:56.49 | jarrod | and its like little half a second 'hiss' or 'psst' |
| 21:57.00 | X-Rob | Bullseye_Network, why don't you use the 'delete=yes' option on the mailbox if you don't care whats in them. |
| 21:57.22 | CunningPike | jarrod: The only time we had something like that it was problems with a TE card |
| 21:57.40 | jarrod | yea we have a TE card |
| 21:57.45 | jarrod | well, a couple of TE cards |
| 21:57.50 | bcnl | jarrod: is it a regular pattern of hiss followed by normal call audio? |
| 21:57.50 | jarrod | but it happened with digium |
| 21:57.58 | jarrod | no pattern, just random 'psst' |
| 21:58.02 | bcnl | damn |
| 21:58.08 | jarrod | when quiet |
| 21:58.09 | bcnl | you could set a watch to mine |
| 21:58.16 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve) |
| 21:58.24 | Bullseye_Network | X-Rob that would work too... Forgot about that |
| 21:58.33 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 21:58.49 | ceeto | Hi all. What's the extensions.conf command to wait for a bunch of digits, i.e., "1234" or something? |
| 21:58.52 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:58.57 | bcnl | so I w__ __lking to t__ __stomer an_ __ _aid to go _____ with the p_ |
| 21:59.08 | bcnl | it's driving me nuts |
| 21:59.20 | jarrod | no, i dont have that problem |
| 21:59.35 | jarrod | that sounds like CoS |
| 21:59.41 | jarrod | issues |
| 21:59.45 | Dr-Linux | Bullseye_Network: did you fine any solution? or what you upto? |
| 21:59.50 | X-Rob | Church of Scientology |
| 21:59.56 | bcnl | CoS or QoS? |
| 21:59.57 | Bullseye_Network | yes I found out what it was |
| 22:00.00 | X-Rob | Damn those Body Thetans! |
| 22:00.01 | jarrod | essentially the same |
| 22:00.04 | jarrod | i speak of juniper terms |
| 22:00.22 | Dr-Linux | Bullseye_Network: did you fix it? btw, what i was? |
| 22:01.20 | Bullseye_Network | Dr-Linux: I had a script deleteing files in a voicemail box, BUT people were leaving messages at the same time so it left a file handle open. |
| 22:01.52 | Bullseye_Network | Dr-Linux: X-Rob poined out I should just use delete=yes |
| 22:02.14 | Bullseye_Network | Dr_linux: it has never caused a problem before. So I dont know why it is now. |
| 22:02.15 | mitcheloc | Bullseye_Network: why are you taking voicemails and not listening to them? |
| 22:02.35 | X-Rob | mitcheloc, I studiously avoided asking that question, because I dreaded to hear the answer. |
| 22:02.42 | X-Rob | now I'm going to have to, aren't I. |
| 22:02.55 | Dr-Linux | Bullseye_Network: i see i'm already using delete=yes in voicemail.conf |
| 22:02.58 | mitcheloc | i'm not afraid, bring it on! |
| 22:03.17 | Bullseye_Network | Ummm... To put is short its the callerid from a telemarketing company |
| 22:03.34 | Bullseye_Network | Its people calling back to say... Who's calling me. or I missed a call from this number |
| 22:03.53 | mitcheloc | heh, can't you just do "Congestion" on them? |
| 22:03.59 | Bullseye_Network | Its better than putting a FAKE number on out callerid |
| 22:04.15 | Bullseye_Network | Yes but then they would keep calling and calling |
| 22:04.27 | X-Rob | WHy not just use an announcement. |
| 22:04.37 | X-Rob | or an IVR |
| 22:04.44 | Bullseye_Network | We want them to answer next time we call them |
| 22:04.54 | X-Rob | 'You've called a telemarketer. If you'd like to be sold something, push 1.' |
| 22:04.54 | mitcheloc | use a circular ivr that never ends? |
| 22:05.09 | mitcheloc | meh, sneaky |
| 22:05.19 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 22:05.38 | hypnox | my asterisk doesnt seem to be listening on port 5060 but it works fine - is this normal? |
| 22:05.39 | Bullseye_Network | Thats what I wanted to do.. An IVR |
| 22:05.43 | Bullseye_Network | but they didnt |
| 22:06.09 | Dr-Linux | hypnox: what's your sip port in sip.conf? |
| 22:06.14 | hypnox | 5060. |
| 22:07.12 | Bullseye_Network | Anybody on here use speakeasy? I have one * server that I cant getoto 5060 on that server |
| 22:07.17 | Dr-Linux | hypnox: how you check if your server is not listing on 5060? |
| 22:07.26 | Bullseye_Network | And speakeasy SEEMS to be blocking that port |
| 22:07.28 | hypnox | well nmap, and telnet |
| 22:07.28 | Nugget | telnet is eeeeeeevil! |
| 22:07.50 | feld_ | if I have an analog phone plugged into the FXS port (which does FXO signaling), where do I configure this phone at? I get a busy signal right now and some funky stuff in the asterisk console. |
| 22:07.52 | mitcheloc | netstat -ln |
| 22:07.55 | Dr-Linux | hypnox: UDP port with telnet ? :S |
| 22:08.05 | hypnox | tried with netstat too, doesnt show |
| 22:08.14 | hypnox | Dr-Linux yeah i wasnt sure if it was udp or tcp |
| 22:08.39 | Dr-Linux | hypnox: there is some command with you can check from CLI .. something like "sip show setting" or "sip traslation" |
| 22:08.56 | Dr-Linux | hypnox: voice ports are probably UDP |
| 22:09.16 | sleepy_one | they are UDP |
| 22:09.42 | sleepy_one | you can use nc ( netcat ) |
| 22:09.59 | hypnox | settings seem ok, right port/ip are there |
| 22:10.15 | hypnox | it clearly works as my sip desk phone can talk to it just fine |
| 22:10.41 | *** join/#asterisk PaulTech2 (n=PaulTech@72.29.76.254) |
| 22:10.59 | hypnox | just confused at the port not being open (trying to diagnose guest sip users) |
| 22:11.03 | PaulTech2 | Had a quick question, Using the asterisk manager API, How can I obtain if a SIP Peer/friend is on a call and if so who they are bridged too? |
| 22:12.03 | Jaxxan | anyone have streaming audio as hold music working ? |
| 22:12.06 | PaulTech2 | sip show channel <X> shows callerid by in a weird form |
| 22:12.09 | Jaxxan | i'm not having much luck here |
| 22:12.11 | PaulTech2 | Jaxxan I did at one time |
| 22:12.15 | Bullseye_Network | PaulTech2: just 'show channels' as far as I know. Or 'Show channels concise' |
| 22:12.36 | Jaxxan | PaulTech2: any chance you could pastebin your musiconhold.conf so i can see how you did it ? |
| 22:12.57 | PaulTech2 | I dont have it any longer configured but it was on voip-info |
| 22:13.07 | Jaxxan | PaulTech2: that's easy |
| 22:13.16 | Jaxxan | just get Gastman |
| 22:13.27 | Jaxxan | you can see every call and what channels their connected too |
| 22:13.45 | Jaxxan | there's a linux and windows client of Gastman also |
| 22:13.47 | PaulTech2 | I'm doing it thru the interface to tie into our own Call Center app |
| 22:14.07 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:14.09 | PaulTech2 | show channels concise will be prefect, little regexp and we're good to go |
| 22:14.13 | PaulTech2 | Thanks Bullseye_Network |
| 22:14.20 | Bullseye_Network | np |
| 22:14.28 | sleepy_one | gnite all :-) |
| 22:15.42 | *** join/#asterisk WiredX (n=matthew@202.137.193.64) |
| 22:15.46 | Bullseye_Network | is there a reason there would be alot of /dev/zap/timer in lsof? |
| 22:16.03 | Bullseye_Network | theres 125 |
| 22:16.09 | SplasPood | So for softclients.. if I had to deploy them for a client... is x-ten's offering (eyebeam) still the best choice for pure SIP? |
| 22:18.53 | Bullseye_Network | got it |
| 22:18.54 | [TK]D-Fender | SplasPood : yup |
| 22:19.12 | SplasPood | [TK]D-Fender: Any experience with it? I've never touched eyebeam |
| 22:19.23 | SplasPood | and I'm dismayed to find that they seem to be moving away from Mac OS X support |
| 22:19.29 | SplasPood | since my client is a mixed bag... |
| 22:19.53 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 22:19.58 | [TK]D-Fender | SplasPood : its good.... not much to say. Does it all... all the codecs, video, Audio, IM, etc |
| 22:20.33 | SplasPood | yea we need straight up voice, thats it.. (or at least thats all I know about the project as of yet) |
| 22:20.50 | SplasPood | X-Lite doesn't have a mac universal binary, and chokes hard /w rosetta.. so I'm sipclientless on the macbook now |
| 22:28.18 | jarrod | dang what is up with this hiss |
| 22:29.44 | Jaxxan | hrm |
| 22:29.56 | Jaxxan | man i heard a stream for like 15 seconds and can't get it back lol |
| 22:30.52 | mitcheloc | SplasPood: i think you can use gizmo for that.. |
| 22:31.42 | WiredX | hey everyone.. |
| 22:31.44 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 22:32.23 | WiredX | is it possible to have a different ring tone for external calls coming in as opposed to internal (intercom/transfer) calls? |
| 22:34.12 | SplasPood | mitcheloc: oh? its open? |
| 22:34.21 | mitcheloc | yes |
| 22:38.37 | CunningPike | What does <ZOMBIE> mean? |
| 22:38.42 | CunningPike | ~zombie |
| 22:38.44 | jbot | Library and server for developing networked apps/games.. URL: http://www.infa.abo.fi/~chakie/zombie/ |
| 22:39.16 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:42.16 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:44.30 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:44.35 | *** join/#asterisk gnosys_ (n=gnosys_@ip68-230-150-92.ri.ri.cox.net) |
| 22:45.13 | CunningPike | Hee hee |
| 22:45.34 | [TK]D-Fender | heh |
| 22:45.38 | [TK]D-Fender | First try at that |
| 22:46.58 | CunningPike | In our dialplan, I get these messages from time to time: Spawn extension (tax-queue, s, 4) exited non-zero on 'SIP/2488-macdonap-a707<ZOMBIE>' |
| 22:47.34 | CunningPike | I was just wondering what the <ZOMBIE> meant and if I should be concerned |
| 22:48.00 | gnosys_ | any opinions/experiences (positive or negative) on WLAN VoIP phones? I'm interested in ease-of-use and quality with Asterisk and I'm also interested in WPA/WPA2 authentication. I've read a review of a Zyxel 2000W that seemed pretty positive, but no mention of WPA. |
| 22:48.29 | [TK]D-Fender | CunningPike : I suspect that implies a now-dead channel. |
| 22:48.42 | [TK]D-Fender | gnosys_ : All Wifi phones SUCK |
| 22:48.58 | CunningPike | [TK]D-Fender: Remote end hang-up? |
| 22:49.09 | gnosys_ | Thanks [TK]D-Fender. Would you elaborate? |
| 22:49.14 | JASON99 | I've been trying to figure out 3-way calling with MGCP for 4-5 days now and randomly asked here to see if anyone knows anything about it. Does anyone know if Asterisk supports 3-way with mgcp.. ?? |
| 22:49.49 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 22:49.51 | [TK]D-Fender | gnosys_ : Grap range/battery life, no browsers for those needing HTTP auth for WEP/other auth, Jitter, etc. I've had clients with nasty latency that wavers on QUALIFY=YES etc.... |
| 22:50.01 | [TK]D-Fender | CunningPike : yup |
| 22:50.24 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:51.09 | CunningPike | [TK]D-Fender: OK - thanks. Just wanted to make sure there wasn't an error in my dialplan. |
| 22:51.37 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:52.22 | WiredX | is there an easier way to upgrade from A@H 2.7 to 2.8 without having to commence a clean install?? |
| 22:53.35 | gnosys_ | [TK]D-Fender: which ones have you tried? I see 9 or 10 of them on voipsupply.com |
| 22:53.39 | JASON99 | !mgcp |
| 22:54.36 | *** join/#asterisk aetius (n=aetius@cpe-069-134-208-043.nc.res.rr.com) |
| 22:54.50 | [TK]D-Fender | gnosys_ : Personally none, slients have tried UTStartcom & Zyzel. |
| 22:55.01 | [TK]D-Fender | gnosys_ : General concensus of others isn't so great |
| 22:55.08 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:55.08 | [TK]D-Fender | clients* |
| 22:55.28 | [TK]D-Fender | WiredX : doubt it. But then again this isn't the place to ask that. |
| 22:55.31 | shmaltz | ~seen tzafrir |
| 22:55.45 | jbot | tzafrir is currently on #asterisk. Has said a total of 11 messages. Is idling for 5h 22s, last said: 'trelane_, ask, just the same'. |
| 22:56.00 | gnosys_ | ok. Thanks very much for the opinions, TK. Anybody else here who can agree or disagree? |
| 22:56.04 | *** join/#asterisk Chriss_sg (n=Chriss_S@209.172.67.146) |
| 22:56.14 | shmaltz | tzafrir ping |
| 22:56.21 | shmaltz | tzafrir_laptop ping |
| 22:56.37 | WiredX | [TK]D-Fender: Thanks, where would the appropriate place be? |
| 22:56.56 | CunningPike | gnosys_: We have a single UTStarcom that our help desk uses around the building - it seems to work OK |
| 22:57.22 | gnosys_ | Thank you CunningPike. |
| 22:58.21 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:59.05 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:59.54 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:00.51 | gnosys_ | CunningPike: was that the F1000G by UTStarCom that you mentioned? |
| 23:01.03 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:01.18 | CunningPike | gnosys_: Let me check...... |
| 23:01.38 | *** part/#asterisk Chriss_sg (n=Chriss_S@209.172.67.146) |
| 23:03.16 | [TK]D-Fender | gnosys_ : Yup, tahts the one... |
| 23:04.27 | [TK]D-Fender | justa/guy ip.withheld.toprotect.theguilty D N 1034 OK (1014 ms) |
| 23:04.39 | [TK]D-Fender | Thats qualify on a GOOD day (right now) |
| 23:04.47 | CunningPike | [TK]D-Fender: Ouch |
| 23:05.05 | [TK]D-Fender | He frequenty passes 2ms on a G router |
| 23:05.12 | [TK]D-Fender | er.... 2000ms |
| 23:05.15 | [TK]D-Fender | 2s ;) |
| 23:05.41 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 23:06.03 | *** join/#asterisk willcampos123 (n=willcamp@198.87.100.3) |
| 23:06.04 | [TK]D-Fender | atency blows... every now and again when I'm on his system and am watching CLI for debugging I have to scoll past the "unreachable" , NOW reachable" BS.... |
| 23:06.08 | *** join/#asterisk SilentValley (n=SilentVa@209.172.67.146) |
| 23:06.13 | willcampos123 | Hello... |
| 23:06.18 | JASON99 | asterisk mgcp is no good :( |
| 23:06.20 | [TK]D-Fender | YAY, down to 991ms! |
| 23:06.34 | [TK]D-Fender | JASON99 : MGCP = no good... didn't need *'s help ;) |
| 23:06.43 | CunningPike | [TK]D-Fender: Does changing verbosity help with that? I guess you'd lose a bunch of other info, too |
| 23:06.46 | *** join/#asterisk treetar1 (n=sterfabl@pool-70-20-20-128.bstnma.fios.verizon.net) |
| 23:06.56 | [TK]D-Fender | CunningPike : I always run at verbose 10 |
| 23:07.00 | *** part/#asterisk SilentValley (n=SilentVa@209.172.67.146) |
| 23:07.05 | willcampos123 | I need to change the hangup cause behavior on the Congestion application, is giving back 3f as disconnect cause, i need it to be 34, that stats for switch equipment congestion |
| 23:07.14 | CunningPike | [TK]D-Fender: Wow - I find 3 tough to keep up with sometimes |
| 23:07.17 | willcampos123 | does anyone know how to do that? |
| 23:07.18 | [TK]D-Fender | CunningPike : But thats me... I don't like flying blind and assuming I know anything. Thats what verbose is for. |
| 23:07.21 | *** join/#asterisk SilentValley (n=SilentVa@209.172.67.146) |
| 23:07.57 | *** join/#asterisk Samoied (n=Samoied@201.22.209.207.adsl.gvt.net.br) |
| 23:08.06 | CunningPike | willcampos123: I think you'd need to patch the app........ |
| 23:08.29 | willcampos123 | I know, but how? |
| 23:08.57 | willcampos123 | because I understand congestion is a pbx_buitin function |
| 23:10.10 | willcampos123 | 04946 { |
| 23:10.10 | willcampos123 | 04947 ast_indicate(chan, AST_CONTROL_CONGESTION); |
| 23:10.10 | willcampos123 | 04948 ast_setstate(chan, AST_STATE_BUSY); |
| 23:10.10 | willcampos123 | 04949 wait_for_hangup(chan, data); |
| 23:10.11 | willcampos123 | 04950 return -1; |
| 23:10.11 | willcampos123 | 04951 } |
| 23:11.10 | CunningPike | willcampos123: #asterisk-dev might be a better bet..... |
| 23:11.19 | willcampos123 | thanks Man!! |
| 23:11.48 | X-Rob | willcampos123, set PRI_HANGUP_CAUSE |
| 23:11.52 | X-Rob | or something like that |
| 23:12.07 | X-Rob | http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+PRI_CAUSE |
| 23:12.18 | X-Rob | there you go. |
| 23:13.42 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 23:14.08 | [TK]D-Fender | X-Rob : No, thats not quite worthy for a tip ;) |
| 23:14.22 | X-Rob | Hurumph 8) |
| 23:14.24 | Bullseye_Network | lol |
| 23:14.42 | Bullseye_Network | I'll give you a tip... Dont eat yellow snow. |
| 23:15.05 | [TK]D-Fender | X-Rob : Be thankful it wasn't the whole shaft ;) |
| 23:15.12 | X-Rob | I'm in tropical australia. The closes we have to snow is the frost that forms on the outside of the beer bottles! |
| 23:15.23 | X-Rob | closest |
| 23:17.18 | *** join/#asterisk b00mer (n=b00mer@ip24-255-125-65.dc.dc.cox.net) |
| 23:18.09 | *** part/#asterisk cmp615 (n=cmp615@fw.cmpcs.com) |
| 23:21.39 | [TK]D-Fender | mitcheloc : One of my clients does. |
| 23:22.03 | mitcheloc | heh, i'm working on a parser for the address book format...big pain in the ass |
| 23:22.07 | mitcheloc | i'm hoping it's worth the trouble |
| 23:25.10 | *** part/#asterisk willcampos123 (n=willcamp@198.87.100.3) |
| 23:28.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-166.cybersurf.com) |
| 23:29.06 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 23:29.32 | *** join/#asterisk iq|mobile (n=iq@71-215-55-11.omah.qwest.net) |
| 23:32.36 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 23:35.07 | CunningPike | mitcheloc: One of the things that keeps me from using Thunderbird is its lack of awareness of the Apple Address Book |
| 23:35.07 | *** join/#asterisk jeffik (n=Jeff@kns221.NetSurf.Net) |
| 23:37.13 | Mw3 | is there any windows application which can open a webpage based on information entered to an IVR (so not from caller id) ? |
| 23:38.24 | mitcheloc | CunningPike: well, thunderbird has ldap right? i don't know much about ldap..seems the best way to do an integrated address book, no? |
| 23:39.29 | CunningPike | mitcheloc: I guess - but if you're a Macphile, you use the built-in Address Book, which doesn't provide LDAP :( |
| 23:40.00 | CunningPike | mitcheloc: THere are some third-party kludges, but Thunderbird isn't better than Apple Mail anyway........... |
| 23:40.53 | mitcheloc | well, none of my work would be good for you anyway till i port to OSX, i should do it soon |
| 23:41.07 | mitcheloc | i'm trying to do an integrated search of the tb ab for searching |
| 23:44.04 | *** join/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 23:44.06 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.4.154) |
| 23:45.41 | *** join/#asterisk Samoied (n=Samoied@201.22.209.207.adsl.gvt.net.br) |
| 23:46.09 | codestr0m | I'm trying to debug a problem I'm experience with a Cisco/7 phone when it's registering.. I have say 10 sippeers that register fine, but 4 others if they try to register immidiately crash asterisk.. (It's outside my normal scope of debugging and looking for tips on how to trace and solve this.) |
| 23:48.09 | Trojan_Hors1 | hi guys, am unable to register friends dynamically ........... do i need to specify register => 1234:password@mysipprovider.com plzzz clarify |
| 23:48.31 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 23:48.49 | Freman | heyas, is there any 'easy' way to randomly select and play a sound file |
| 23:48.50 | Freman | ? |
| 23:50.00 | *** part/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 23:51.51 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 23:54.30 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 23:54.46 | ceeto | Hi all. Can someone point me in the right direction of setting up an * to * IP only call across the Internet? |
| 23:55.00 | mitcheloc | ~iax |
| 23:55.01 | jbot | methinks iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 23:57.24 | ceeto | I figured some of that out in ./iax.conf, what do I put in extensions.conf to make it call? And what port(s) do I need to open on my firewall(s)? Thanks for any help. |