00:00.39 | JT | j0: which state? there's at least 2 or 3 abbotsfords in Australia |
00:00.55 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
00:01.11 | j0 | JT: NSW |
00:01.21 | teknoprep | hey how are the cisco 7970 phones and asterisk ? |
00:01.21 | JT | right |
00:01.29 | JT | so the one in sydney :) |
00:01.34 | j0 | yep :) |
00:03.32 | jero | <PROTECTED> |
00:03.51 | jero | afaik Queue() does not return |
00:04.32 | Sputtering | it returns only if the queue is unavailable for some reason |
00:04.49 | tzanger | woo, 2 TDM400P cards, a TE405 and eight FXO modules for the TDM400 cards... I wonder if any of these qualify for HPEC licenses |
00:05.20 | jero | Sputtering: yes |
00:05.32 | jero | I'd like to run some action at the end of a Queue() call |
00:07.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
00:08.21 | |ryan| | Is there a way to get *67 to work from a SIP phone? |
00:08.30 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
00:08.51 | CunningPike | |ryan|: Define 'work' |
00:10.07 | |ryan| | I'd like it to suppress sending callerid like it does on a POTS line in the US |
00:10.50 | JT | |ryan|: what connection to the pstn? |
00:11.17 | |ryan| | I have an IAX connection. |
00:11.43 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
00:13.05 | |ryan| | the provider doesn't send caller-id on outgoing calls unless I set it, so I have it set in my outgoing context before dialing out. |
00:14.02 | |ryan| | So I guess what I would need to do is set up a *67 extension that somehow bypasses where the CallerID data gets set. |
00:14.13 | *** join/#asterisk hellojoe (n=hellojoe@c-67-160-249-95.hsd1.ca.comcast.net) |
00:14.59 | JT | that's one way i guess |
00:15.09 | JT | i'm curious what you mean by "set it" |
00:15.16 | JT | by default asterisk sets callerid |
00:16.10 | |ryan| | Well, it's set by default as my extension's userid |
00:16.20 | |ryan| | which is non-numeric |
00:16.28 | JT | ah |
00:17.02 | |ryan| | I'm trying to figure out how I'd give another dial tone and accept digits after I hit *67 |
00:18.52 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
00:18.52 | *** mode/#asterisk [+o anthm] by ChanServ |
00:19.00 | JT | oh |
00:19.02 | JT | that's easy |
00:19.04 | JT | DISA() |
00:19.07 | flenders | |ryan| DISA |
00:19.39 | |ryan| | Ok. |
00:19.43 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
00:21.38 | JT | morning flenders |
00:28.29 | ManxPower | If cell phone companies want to sell ring tones maybe it would help if they made it easy to do so. |
00:29.32 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
00:31.17 | ManxPower | Verizon wireless required me to select my phone model, let me search and find the perfect ringtone I wanted, I created an account, re-did the search, clicked on purchase, and was told that I can't purchase ringtones for this phone over the web but could but that specific ring tone by using my phone. However, I don't know which of the 4 or 5 ringtone applications (each costs money) I have to use. |
00:31.32 | JT | ManxPower: one think i really hate now is people can pay to have music play while their phone is being dialled :( |
00:31.44 | Juggie | ManxPower, sounds like you have a 1) shitty provider 2) shitty phone |
00:31.57 | ManxPower | Juggie: I have the only provider that works where I live. |
00:32.20 | Juggie | a ringtone on my phone consists of 1) connect the phone to pc using either usb cable/bluetooth/IR 2) transfer ringtone 3) enable ringtone in phone ui. |
00:32.27 | ManxPower | Juggie: the web site even has a "content ID" which in theory I can just type into the phone and get the ringtone. I can't find it. |
00:32.50 | JT | i've never downloaded a ringtone |
00:33.24 | ManxPower | JT: I've only ever downloaded 1 ringtone, now I want a different one. |
00:33.34 | JT | heh |
00:33.51 | JT | i usually look for one in the default set, that just sounds like a phone ringing |
00:33.53 | Juggie | ManxPower, your phone is totally locked cant transfer anything to it yourself? |
00:33.57 | hellojoe | guys, is there a way to just use a simple voicemail to email instead of all of the voicemail stuff in asterisk? |
00:34.01 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:34.04 | mmlj4 | hey ManxPower |
00:34.15 | ManxPower | Juggie: I'm sure I could if I spent a couple of hours figuring out how to do so. |
00:34.25 | mmlj4 | been to the new covington doctor office yet? |
00:34.26 | hellojoe | i was thinking of recording the message (Record) and then using sendmail to attach the message and ship it out? |
00:34.29 | ManxPower | hello mmlj4 |
00:34.35 | Juggie | ManxPower, CDMA providers are so anal, my phone is wide open. |
00:34.42 | Juggie | and its a locked to provider phone too. |
00:34.44 | hellojoe | anyone sees any issue with this. No need fo rme to save the vmails on the server |
00:34.49 | ManxPower | hellojoe: sure you can do that. |
00:34.54 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
00:34.55 | Juggie | i hook it up over usb, and its a flash drive. |
00:35.16 | Juggie | two flash drives actually, one is the phone memory, and the other is the addon memory card. |
00:35.20 | |ryan| | hmm |
00:35.20 | hellojoe | Ahh... thanks ManxPower. I was mainly worried about the performance/gotchas |
00:35.21 | ManxPower | Juggie: Oh it is so impressive can I hold it? |
00:35.32 | mmlj4 | ManxPower: all I need to do is screw on the wallplates, and I'm done # aside from any red phones, etc. |
00:35.38 | |ryan| | DISA is just giving me busy tones when I try to dial. |
00:35.49 | ManxPower | I spent $99 on the phone 18 months ago and it is a prepay |
00:35.56 | hellojoe | now I need to figure out the command line for sendmail with *.wav as attachements |
00:36.14 | ManxPower | |ryan|: then it is not finding a matching exten line in the context specified |
00:36.25 | ManxPower | I'm NEVER EVER doing another cellphone contract again. |
00:36.35 | Juggie | ManxPower, i told my provider someone else was offering me a better deal and i got this one for 49$ on a 3 year contrac.t |
00:36.52 | Juggie | it should/should have been 150$ |
00:36.58 | Juggie | er, would/should |
00:37.00 | JT | mmlj4: red phones? |
00:37.14 | mmlj4 | "you need a new battery? sure, just sign another contract, and you can have one" |
00:37.17 | |ryan| | ManxPower: is there any way to debug what context a line is in? |
00:37.19 | ManxPower | I basically had to lie and back up the lie with paperwork to get out of the 3 x $175 termination fee. |
00:37.33 | mmlj4 | JT: POTS bat-phone, backing up VoIP |
00:37.44 | ManxPower | |ryan|: you have to specfify the context when you run DISA. "show application dias" |
00:37.46 | mmlj4 | $client likes them to be red |
00:37.51 | ManxPower | and "show application disa" too. |
00:37.57 | JT | mmlj4: ah ok |
00:38.24 | tzanger | Juggie: GSM swap-out-the-card-to-a-new-phone is going the way of the dodo too |
00:38.29 | tzanger | CDMA was just ahead of the curve there :-( |
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00:38.40 | ManxPower | I'll replace my phone when it breaks 8-) |
00:39.21 | |ryan| | ManxPower: I did specify a context. Is it going to break it if I set it to use the same context it was accessed from? |
00:39.35 | J4k3 | sim cards are cute... but I'd rather have decent RF performance (CDMA) |
00:40.19 | ManxPower | |ryan|: paste the DISA line from extensions.conf |
00:40.22 | *** join/#asterisk elriah (n=johnny@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
00:40.52 | |ryan| | (this is in the internal context) exten => *67,n,DISA(no-password|internal) |
00:41.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
00:41.26 | ManxPower | |ryan|: did you forget the leading _ in a pattern match for the exten line you want matched? |
00:41.38 | |ryan| | no |
00:41.38 | ManxPower | elriah: you are in Birmingham?? |
00:41.48 | |ryan| | I get a dialtone when I hit *67 |
00:42.27 | ManxPower | |ryan|: does the Asterisk console showing DISA being run? If not, your phone is handleing the *67 and giving you dialtone, not Asterisk |
00:42.54 | *** join/#asterisk docelmo (n=vircuser@c-68-82-181-225.hsd1.de.comcast.net) |
00:43.04 | |ryan| | hmm |
00:43.13 | |ryan| | let me try this from my softphone |
00:43.17 | JT | anyway, DISA is probably not what you want |
00:43.22 | JT | not sure if you want a second tialtone |
00:43.24 | JT | dialtone |
00:43.43 | |ryan| | I don't really need a second dialtone |
00:44.35 | |ryan| | I wat to run some commands (setting caller id presentation to prohibited) then allow any other number accessable from the context to work. |
00:44.52 | *** join/#asterisk PaulTech85 (i=PaulTech@72.29.76.254) |
00:45.10 | PaulTech85 | So, If I want to get billsec for the number of seconds a user TALKED to a agent, How would one go about that? |
00:45.14 | PaulTech85 | Insteed of counting total time in queue |
00:45.16 | JT | yeah i was trying to work out how to do this a while back |
00:45.29 | JT | but i then realised i didn't really need it and just blocked callerid |
00:45.57 | |ryan| | well |
00:46.10 | |ryan| | I was going to do a couple other simmilar things |
00:46.20 | PaulTech85 | Is there no direct method? |
00:46.29 | |ryan| | like try and set it up to allow a code to be entered before dialing that will record the call |
00:46.44 | [TK]D-Fender | PaulTech85 : One piece of info you have in the queu log is the hold time. talk time = total time - hold time |
00:46.45 | PaulTech85 | ah |
00:46.48 | PaulTech85 | queue_log |
00:46.55 | PaulTech85 | Just say that Fender :-) |
00:47.01 | PaulTech85 | saw* |
00:47.10 | PaulTech85 | 'Asterisk does not currently support dumping queue_log data straight to a MySQL table. ' |
00:47.22 | *** part/#asterisk jero (n=jerome@modemcable046.23-81-70.mc.videotron.ca) |
00:47.52 | [TK]D-Fender | PaulTech85 : Yes, * can store queue data directly into MySQL, or another ODBC database |
00:48.04 | [TK]D-Fender | *sigh* |
00:48.11 | [TK]D-Fender | looks like a bad reference... |
00:48.12 | PaulTech85 | Maybe I'm reading old pages on wiki ;) |
00:48.33 | [TK]D-Fender | PaulTech85 : They look like current pages... only wrong ;) |
00:48.42 | PaulTech85 | Hehe |
00:48.49 | PaulTech85 | let me fire up the actual docs |
00:49.55 | PaulTech85 | Hmm being honest I dont see it anywhere.. |
00:50.15 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
00:50.21 | [TK]D-Fender | keep looking..... (hint : its in asterisk-addons) |
00:50.41 | [TK]D-Fender | anyways, must be off, back in 2 hrs or so... |
00:51.11 | PaulTech85 | I'd guess cdr_addon_mysql but I know thats not it, *looks at app_addon_sql_mysql* |
00:51.23 | hellojoe | any idea how to attach .wav file using (system /usr/bin/sendmail) command |
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01:02.49 | PaulTech85 | Hmm not seeing much |
01:03.29 | *** join/#asterisk joaovianna (n=joaovian@ool-4354d1a8.dyn.optonline.net) |
01:04.27 | joaovianna | Can someone help me ? I'm testing video in asterisk but I'm receiving a error message saying "Unknown RTP codec 126 received from ..." Anyone ? |
01:07.16 | joaovianna | Anyone using asterisk+video ? |
01:08.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
01:11.17 | Nivex | I'm building asterisk 1.2.15 on debian sarge and would like to force it to use gcc-3.4 instead of gcc. How do I go about this? |
01:13.25 | elriah | Why would you want to? (just curious) |
01:13.50 | Nivex | 3.4 > 3.3, isn't it? |
01:14.09 | Nivex | but sarge insists on keeping gcc pointed at gcc-3.3 |
01:14.22 | JT | does it even matter, if it works after compiling with 3.3? |
01:14.48 | elriah | Use the OS default compiler, save yourself some headache. |
01:15.02 | carrar | yeah, save yourself serious headache |
01:15.18 | carrar | unless you are a C coder |
01:15.28 | carrar | and know all the ins and out |
01:15.38 | carrar | but even then |
01:15.41 | carrar | I wouldn't |
01:15.54 | Nivex | ok |
01:16.15 | elriah | That's the "ok, but I'm going to try it anyway" ... |
01:16.16 | elriah | lol |
01:16.20 | carrar | hahah |
01:16.32 | carrar | and then it crashing on a dynamic module |
01:16.35 | carrar | crashes |
01:16.36 | carrar | heh |
01:17.36 | PaulTech85 | Isnt 1.4 the stanard now? |
01:17.44 | PaulTech85 | If not I'm pissed about porting 1400 line extension.conf |
01:17.45 | JT | no |
01:17.47 | PaulTech85 | ;p |
01:17.57 | JT | most people still use 1.2.x in production |
01:17.59 | JT | more stable |
01:18.04 | PaulTech85 | ahh |
01:18.56 | elriah | I hear that 1.4 has a few issues, everbody has recommended to me to wait to 1.4.1 for production upgrades. I'll probably wait until 1.4.4 or so... |
01:19.07 | elriah | 1.2.x is very stable, imho |
01:21.47 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
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01:28.31 | Juggie | elriah, not 1.4 but 1.4.0 |
01:28.35 | *** join/#asterisk Ryushin (i=chris@71.33.251.74) |
01:28.38 | Juggie | so you can run 1.4svn or wait for 1.4.1 |
01:29.15 | JT | or just keep waiting :) |
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01:34.00 | mmlj4 | grr... i'll bet ManxPower didn't recognize my nick earlier |
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01:36.21 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
01:36.52 | FuriousGeorge | hey all |
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01:36.59 | FuriousGeorge | what's shakin'? |
01:37.23 | PaulTech85 | Hehe |
01:37.28 | PaulTech85 | My SQL got interesting |
01:37.38 | PaulTech85 | Just for getting the total talk time of one dst number |
01:37.40 | PaulTech85 | after queue |
01:37.42 | PaulTech85 | 'select sum(asterisk_queue_log.arg2) from asterisk_queue_log left join cdr ON asterisk_queue_log.callid = cdr.uniqueid where asterisk_queue_log.event='COMPLETECALLER' and cdr.dst='$dst' and unix_timestamp('time) between unix_timestamp('2007-02-00 00:00:00') AND unix_timestamp('2007-02-31 00:00:00'); ' |
01:38.14 | FuriousGeorge | so i installed asterisk at a bar today. everything is working fine, they need a way to transfer calls between stations. im teaching them to transfer calls, we must have made 200 calls, everyone gets it, owner is like a kid with a new toy |
01:38.48 | PaulTech85 | Haha |
01:39.02 | PaulTech85 | Bar cant exactly be high volume |
01:39.02 | PaulTech85 | ;) |
01:39.05 | FuriousGeorge | so his buddys there and hes showing off his system, he's got a business too, he's interested asking about the price. goes to make a call to try it out. gets dropped |
01:39.09 | FuriousGeorge | tries to call again |
01:39.11 | FuriousGeorge | not found |
01:39.30 | FuriousGeorge | goes through, the third try, dropped again "well, i cant use this phone" |
01:40.02 | [hC] | whoops! |
01:40.15 | [hC] | whats the setup there? |
01:40.19 | JT | why was it dropped |
01:41.38 | FuriousGeorge | he mentioned it "normally takes a week to work these things out" owner says, "well, we just installed it today", i say "well, i dunno, ill look into that, gotta work it out". i tell owner later, "im concerned about that, and i want you guys to make a lot of calls, write down time and number if it happens again", but he tells me he thinks it was a cell phone he called or something, wasnt the system... here's the thing though, i know |
01:42.10 | FuriousGeorge | i check my messages, but my event log is blank! |
01:42.15 | FuriousGeorge | so no good info thus far |
01:42.16 | fastfeet | I am trying to use MySQL for CDR in version 1.4. I set up a database, set a set a user for the DB, but I am not sure what tables I need to create? |
01:42.22 | fastfeet | Any ideas? Hints? |
01:42.27 | FuriousGeorge | snom 360's on asterisk 1.4 |
01:42.29 | PaulTech85 | its on the wiki |
01:42.44 | FuriousGeorge | fastfeet: ~docs |
01:42.48 | FuriousGeorge | err |
01:42.49 | FuriousGeorge | ~docs |
01:42.51 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
01:42.52 | fastfeet | I;ve been through the wiki, I must be missing it |
01:42.59 | [hC] | FuriousGeorge: sip/iax/zap for outgoing lines? |
01:43.11 | FuriousGeorge | [hC]: iax every time |
01:43.23 | FuriousGeorge | i meant to say asterisk 1.2.14 |
01:43.27 | FuriousGeorge | not asterisk 1.4 |
01:43.43 | FuriousGeorge | fastfeet: ive never tried "asterisk realtime" or i'd help you |
01:43.48 | FuriousGeorge | is that what you are searching for? |
01:43.49 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
01:44.40 | FuriousGeorge | [hC]: ive used the same provider other places, not had this problem. also ive reboot his phone since then... im /thinking/ it may have something to do with this AP |
01:44.52 | fastfeet | thanks yall |
01:45.23 | FuriousGeorge | netgear cr@p between phone and asterisk/switch |
01:46.11 | FuriousGeorge | good news is guy now wants me to come back and install more phones |
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02:12.51 | foobar778 | yea all is working !!!! |
02:14.39 | FuriousGeorge | i mean, now that i think about it out loud to myself... is it bad practice to use a $30 netgear ap/switch, certainly not "business class" as the interface between my sip phone and my asterisk/switch? |
02:14.46 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
02:17.27 | foobar778 | ap/switch or router? |
02:18.14 | FuriousGeorge | the netgear ap/switch has firmware |
02:18.21 | foobar778 | and do u use an ata or ur switch/ap has phone ports? |
02:18.22 | FuriousGeorge | im not using the wan interface of course |
02:18.37 | foobar778 | ?? |
02:18.53 | FuriousGeorge | im complicating things by saying ap/switch, i just mean to imply that im using a device that is a router as a switch |
02:18.55 | foobar778 | whats ur setup modem then then?? |
02:19.03 | FuriousGeorge | ? |
02:19.30 | foobar778 | my question which is first to the moden ata or router |
02:20.38 | FuriousGeorge | linux running iptables --> switch --> (wall/jack) netgear wireless router device --> snom 360 |
02:20.55 | foobar778 | what device connects to the ethernet port of your modem???? |
02:21.18 | foobar778 | snom 360 is ur modem???? |
02:21.21 | FuriousGeorge | linux running iptables obviously, or why would i specify that |
02:21.24 | FuriousGeorge | other way |
02:21.40 | foobar778 | what connects to ur ethernet port of ur modem |
02:21.51 | foobar778 | what device? |
02:21.54 | FuriousGeorge | linux |
02:21.57 | FuriousGeorge | computer |
02:22.03 | FuriousGeorge | its a "manged switch" |
02:22.08 | foobar778 | ahh |
02:22.11 | FuriousGeorge | its not asterisk |
02:22.24 | foobar778 | so thats ur firewall |
02:22.30 | FuriousGeorge | yeah |
02:22.33 | foobar778 | ok |
02:22.50 | joaovianna | Can someone help me ? I'm testing video in asterisk but I'm receiving a error message saying "Unknown RTP codec 126 received from ..." Anyone ? |
02:22.50 | foobar778 | and what does ur phone attavh to |
02:23.09 | FuriousGeorge | netgear wireless router |
02:23.31 | foobar778 | and that router is a voip router? |
02:24.01 | FuriousGeorge | no, routers almost always have switches built into them. just dont use the wan. im thinking maybe, for some reason, the switch that is built into that (ports 1-4) is causing certain calls to drop |
02:24.44 | foobar778 | ur swich has voip phone ports?? |
02:24.53 | foobar778 | or an ata between th4em?? |
02:24.54 | FuriousGeorge | :) |
02:25.06 | FuriousGeorge | lemme link you to what im talking about |
02:25.41 | foobar778 | want to join my pbx u can then talk |
02:26.06 | FuriousGeorge | http://www.continent.com.au/images/products/WPN824AU.gif |
02:26.12 | FuriousGeorge | thats all it is |
02:26.20 | JT | FuriousGeorge: netgear is trash |
02:26.20 | JT | avoid |
02:26.23 | *** join/#asterisk n|cotine (i=nicotine@147.202.49.52) |
02:26.28 | FuriousGeorge | JT: thanks |
02:26.45 | JT | most consumer brands are fairly trashy :P |
02:26.48 | foobar778 | thats ur ata that ur analog phone goes into?? |
02:26.57 | FuriousGeorge | foobar778: its a snom360 |
02:26.59 | JT | ?!???!??!???? |
02:27.22 | FuriousGeorge | JT: me and him are having "ill communications" as the beastie boys would say |
02:27.25 | foobar778 | Im asking not familair with a snom |
02:27.36 | FuriousGeorge | just a sip phone that happens to be asterisk friendly |
02:27.48 | foobar778 | do u have the right rtp ports and sip ports fowared to that |
02:27.57 | FuriousGeorge | there is no nat there |
02:28.00 | FuriousGeorge | im using it as a switch |
02:28.09 | foobar778 | iptables -F |
02:28.17 | foobar778 | tried that as a test? |
02:28.40 | FuriousGeorge | to find out why the call dropped i would be doing a sip dump, dont you think |
02:28.55 | foobar778 | well if its a poprt issue |
02:29.03 | FuriousGeorge | its internal only no nat |
02:29.15 | foobar778 | <PROTECTED> |
02:29.25 | FuriousGeorge | :( |
02:29.29 | foobar778 | iptables -F does that |
02:30.28 | foobar778 | what linux distro is up front acting as the firewall |
02:30.39 | FuriousGeorge | i was just saying that i think the netgear thing is the culprit (even though im only using it as a switch, and no nat is involved) because only this phone does it and only this phone is behind that thing |
02:31.03 | FuriousGeorge | and besides, if the call it is getting dripped it would be the signalling that caused it, not the rtp |
02:31.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
02:31.19 | foobar778 | not my experience |
02:31.36 | FuriousGeorge | you can make a call with no rtp, why not... just wont hear anything |
02:31.40 | FuriousGeorge | it wont drop though |
02:31.49 | foobar778 | I had rtp ports not opened and my call would dreop in 30 seconds consistently |
02:32.20 | foobar778 | u have nat=no?? |
02:32.27 | FuriousGeorge | odd b/c iirc sip uses like two per call, though i could be wrong |
02:32.29 | foobar778 | localnet=???? |
02:32.56 | foobar778 | in your sip.conf globals |
02:33.05 | FuriousGeorge | its set |
02:33.15 | foobar778 | as what yes or no |
02:33.21 | FuriousGeorge | yes |
02:33.40 | foobar778 | and local net sety up |
02:33.48 | foobar778 | externalip= |
02:33.57 | FuriousGeorge | using externhost, but same idea, yeah |
02:34.59 | foobar778 | how many pcs just curious not related behind the switch |
02:35.18 | FuriousGeorge | max two wireless and the snom |
02:35.34 | foobar778 | does ur isp give u free wan ips? |
02:35.36 | puzzled | evening all |
02:35.54 | FuriousGeorge | foobar778: not feasable |
02:36.31 | foobar778 | if ur on a switch then there is no nat really and for the pcs well u said wirless |
02:36.53 | foobar778 | so the negear is a wirelss ap |
02:37.23 | FuriousGeorge | yeah |
02:37.24 | foobar778 | and the ap does network translation |
02:37.35 | FuriousGeorge | NO |
02:37.37 | FuriousGeorge | :) |
02:37.39 | foobar778 | so u only need one wan ip |
02:37.41 | puzzled | anyone know if spandsp-0.0.3pre27 works with asterisk-1.2? |
02:38.46 | FuriousGeorge | right, i set the nat part up correctly, clients can go rempote, i dont need to pay for more ips, which come at a heavy premium around here |
02:39.01 | foobar778 | wait if it doesnt do nat and its a switch every pc behind a switch must have a uniqque wan ip unless network translation is being done |
02:39.46 | foobar778 | so therte must be network translation going on |
02:39.47 | FuriousGeorge | no, linux does nat, the netgear is only being used as a switch... i feel like ive said that |
02:40.14 | FuriousGeorge | right, i set the nat part up correctly, as i just stated |
02:40.32 | foobar778 | sorry its not working for u |
02:40.43 | FuriousGeorge | well its only once in a blue moon... |
02:40.54 | FuriousGeorge | but you know its gonna happen again at the worst time |
02:41.07 | foobar778 | u dont like routers?? |
02:41.42 | foobar778 | never understood why people by switches when a trouter costs the same and can be both |
02:42.37 | puzzled | prolly performance |
02:42.40 | foobar778 | and better yet a voip router |
02:42.56 | n|cotine | foobar778: Because you can do much more interesting things if you have a linux router and a normal switch? |
02:43.25 | foobar778 | love my router wrt54G with custom linux firmaware dd-wrt |
02:43.28 | FuriousGeorge | its a restaurant that has a subnet for wireless with content filtering, subnet for wireless for back office, subnet for DMZ and several devices including security system, and by the way a voip telephone system, which may want some QoS for the first time one of them discover bit torrent in back office |
02:44.02 | foobar778 | u should see the qos on the dd-wrt |
02:44.03 | FuriousGeorge | oh AND their credit cards go out over a separate proprietary system, under mine, double natted |
02:44.19 | n|cotine | foobar778: dd-wrt has iproute2? |
02:44.51 | foobar778 | go to the ddwrt site see all the packes that can be installed\ |
02:45.04 | foobar778 | <PROTECTED> |
02:45.09 | foobar778 | <PROTECTED> |
02:45.29 | foobar778 | paackages |
02:45.41 | FuriousGeorge | as the military would say, any attempt to put a asterisk-router into production would be fubar |
02:45.59 | foobar778 | not so it runs great |
02:46.07 | foobar778 | read the success stories |
02:46.23 | FuriousGeorge | im sure its scales quite well |
02:46.56 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
02:46.56 | foobar778 | this guy had so many channels and had it up like 5 years |
02:47.16 | *** join/#asterisk Stp1800 (n=Stp1800@67-22-111-205.atlsfl.adelphia.net) |
02:47.28 | FuriousGeorge | ok, well, im not gonna argue this point with you |
02:48.27 | foobar778 | anyway turning a 40 dollar router into a 600 dollar router just by flashing the firmware is very cool |
02:49.41 | FuriousGeorge | its definitely cool, i just would use it for something "real" sparingly |
02:49.52 | FuriousGeorge | if at all |
02:49.59 | *** join/#asterisk X-Rob (n=Rob@CPE-58-169-100-13.vic.bigpond.net.au) |
02:50.07 | FuriousGeorge | its hard enough to get things just right without complicating matters for yourself |
02:50.28 | foobar778 | http://clipmarks.com/clipmark/0AC14714-0A93-4D50-82E7-8F6F6729DC36/ |
02:50.35 | foobar778 | read enjoy |
02:52.10 | FuriousGeorge | wheres the part on asterisk dimensioning |
02:52.22 | foobar778 | goto openwrt |
02:53.02 | foobar778 | http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html |
02:55.43 | FuriousGeorge | why dont you just tell me how many simultaneous channels i can run on my linksys router, and we will make an educated guess about its suitability for business use. |
02:57.14 | foobar778 | well a lot |
02:57.21 | FuriousGeorge | more or less? |
02:57.27 | FuriousGeorge | what would you say? |
02:57.30 | foobar778 | <PROTECTED> |
02:57.36 | JT | blah blah |
02:57.41 | FuriousGeorge | if you dont want to guess i will |
02:57.43 | JT | it's clear foobar778 hasn't actually done it, FuriousGeorge |
02:57.50 | JT | he's just saying how amazing it looks |
02:58.04 | foobar778 | I gave u direct links but I cant answer all I have read the sucees stories |
02:58.26 | foobar778 | <PROTECTED> |
02:58.31 | FuriousGeorge | i'd say with no compression, transcoding, pure sip, i MIGHT get 5 channels |
02:58.38 | FuriousGeorge | on a sunny day |
02:58.41 | FuriousGeorge | just a guess |
02:58.59 | JT | yeah from memory 7 g.711 sessions max |
02:59.03 | foobar778 | well if u try u will know but from what I read this guy had a huger pbx |
02:59.09 | JT | from reading the asterisk dimensioning page |
02:59.23 | FuriousGeorge | JT: its one of my many character flaws, i said i wasnt gonna argue it, and yet here i am doing it |
02:59.25 | JT | s/u/you/ |
02:59.31 | JT | heh |
03:00.16 | foobar778 | I havent put asterisk on yet because currently Im using ddwrt |
03:00.33 | foobar778 | But Im toyong with flashing openwrt |
03:00.40 | foobar778 | toying |
03:01.05 | JT | for me it's pointless |
03:01.10 | JT | if i can't terminate on it |
03:01.12 | JT | no use |
03:01.45 | foobar778 | after what I have read But sinced I resolved all my nat issues not so enthusiastic about anothert flash and a possible brick |
03:02.16 | FuriousGeorge | the warranty is long, if its covered just tellem you hosed it installing a fw update |
03:02.47 | foobar778 | But the features I mean the ddwrt really makes u have a 600 dollar type router its truly awesome asterisk or not |
03:02.48 | FuriousGeorge | its not unethical b/c you can blame the device for not appreciating the linux you tried to liberate it wityh |
03:02.56 | foobar778 | That I can tell u first hand |
03:03.18 | FuriousGeorge | its definitely cool, like i said. im all for it. been meaning to rma my POS router |
03:03.32 | JT | $600 router, i dunno about tha |
03:03.32 | FuriousGeorge | acts flakey even with openwrt |
03:03.46 | foobar778 | oh yes definiyley |
03:03.56 | foobar778 | unlimited port fowardings |
03:04.01 | foobar778 | <PROTECTED> |
03:04.02 | JT | maybe at cisco prices |
03:04.04 | JT | wow |
03:04.08 | JT | so does my linux gateway |
03:04.10 | foobar778 | <PROTECTED> |
03:04.11 | JT | it cost me nothing |
03:04.12 | JT | $0 router |
03:04.13 | foobar778 | on and on |
03:04.31 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
03:04.38 | foobar778 | go look at the ddwrt page |
03:04.43 | JT | fucks sake |
03:04.44 | JT | NO |
03:04.56 | JT | i've looked at it previously, stop advertising it every 2 seconds |
03:04.58 | JT | who cares |
03:05.05 | JT | nothing that a normal linux box cannot do |
03:05.20 | JT | no idea why i'd want samba on my gateway anyway |
03:05.27 | JT | as my gateway is not a fileserver |
03:05.34 | foobar778 | u need how many network adapyers??? |
03:05.39 | JT | ?!?!?! |
03:05.41 | FuriousGeorge | foobar778: you should check out ipcop |
03:05.48 | foobar778 | I know it |
03:05.50 | JT | foobar778: what? |
03:05.57 | JT | what about network adapters? |
03:06.00 | foobar778 | ipcop firewall |
03:06.21 | *** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com) |
03:06.25 | foobar778 | it does wirelss??????? |
03:06.31 | JT | so what's this about network adapters |
03:06.32 | foobar778 | ipcop think not |
03:06.41 | JT | and if you have a question, one question mark is sufficient |
03:06.48 | JT | ipcop can probably do wireless |
03:06.55 | foobar778 | how |
03:06.59 | FuriousGeorge | foobar778: its doable, just not supported |
03:07.00 | foobar778 | need hardware |
03:07.03 | JT | yes |
03:07.04 | *** join/#asterisk abv (n=adam@74.72.190.181) |
03:07.05 | FuriousGeorge | wireless nic |
03:07.08 | JT | omg bbq |
03:07.13 | foobar778 | well now not so free huh |
03:07.22 | JT | alternatively, you could just get an access point |
03:07.26 | JT | which is pretty cheap |
03:07.28 | JT | and solid |
03:07.29 | foobar778 | $$$$ |
03:07.32 | FuriousGeorge | it rained here the other day and a wireless nic fell out of the sky |
03:07.32 | JT | cheap |
03:07.35 | JT | they are not expensive |
03:07.45 | JT | if you want wireless, it costs money, like anything |
03:07.48 | foobar778 | well wrt54G ebay 10 dollars |
03:07.55 | JT | i use cables for everything at home |
03:08.16 | JT | sure, i've used wrt54g, but as aps only, pity you can't really switch off their router function |
03:08.20 | abv | hey. i'm an asterisk newbie. |
03:08.42 | FuriousGeorge | hey asterisk newbie |
03:08.46 | foobar778 | u can |
03:08.57 | foobar778 | turn off router function in ddwrt |
03:09.00 | JT | still not sure what your point is foobar778 |
03:09.05 | JT | yes if you use unofficial flash |
03:09.11 | foobar778 | yes |
03:09.23 | JT | i couldn't really be bothered |
03:09.28 | abv | hi. quick question. i setup asterisk on a vps today (with lylix.net). what's the easiest way to test if it working (inbound)? |
03:09.30 | foobar778 | thats my point what I have been talking about guess u havent followed |
03:09.40 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
03:09.56 | JT | s/u/you/ |
03:10.14 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
03:10.26 | JT | foobar778: no, you've been crapping on about how ddwrt is the best thing since sliced bread |
03:10.32 | JT | using wild handwaving |
03:10.39 | foobar778 | it is |
03:10.45 | FuriousGeorge | crapping is a good word to use there i think :) not to pile on |
03:10.49 | JT | it's not bad, but it's nothing that amazing |
03:10.54 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
03:10.55 | k-man | so.... after doing a make install, should I do a make samples? or should i just populate /etc/asterisk with the files as i need them? |
03:10.57 | FuriousGeorge | good verb choice |
03:11.14 | FuriousGeorge | k-man: make samples is nice when you upgrade especially |
03:11.21 | FuriousGeorge | as configs change |
03:11.22 | k-man | FuriousGeorge, why? |
03:11.32 | foobar778 | well thats ur opinion most think its great Do u create firmware?? |
03:11.33 | FuriousGeorge | i mean between major releases |
03:11.37 | k-man | it won't overwrite my existing configs? |
03:12.01 | FuriousGeorge | k-man: oh, ive always backed mine up, i assume they were overwritten |
03:12.06 | JT | foobar778: no, i generally prefer to use linux boxes to do NAT/routing |
03:12.08 | FuriousGeorge | but i think it actually makes .sample |
03:12.17 | FuriousGeorge | not responsible if im wrong about that :) |
03:12.36 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
03:13.01 | foobar778 | well good for u that make u geekier thew ddwrt in Linux my friend ssh into the os and u will see a full linux diostro at ur disposale |
03:13.16 | foobar778 | ddwrt is a Linux os |
03:13.55 | JT | ok i hardly understood that but i think i get the gist |
03:14.01 | JT | yes i know it's linux |
03:14.10 | JT | just does not make it worth $600 |
03:14.13 | foobar778 | example the v5 was vxworks and u strip that and put on a Linux ddwrt or opewrt or hyperwrt OS |
03:14.17 | FuriousGeorge | foobar778: let me put it this way, openwrt is cool, but if you want to do some serious stuff, like content filtering/web proxy etc. you just arent gonna do that with openwrt alone |
03:14.30 | foobar778 | It has it |
03:14.45 | FuriousGeorge | how am i gonna cache a million .jpgs on my linux router |
03:14.46 | foobar778 | I should let u into my setup page |
03:14.49 | JT | also, the cpus aren't very powerful, so if you have high load they can't handle it |
03:14.54 | FuriousGeorge | s/l;inux/linksys |
03:15.25 | foobar778 | U have to dedicate a whole nachine for ur firewall |
03:15.38 | foobar778 | <PROTECTED> |
03:15.40 | FuriousGeorge | you have a half a machine for a quarter of a firewall |
03:16.01 | foobar778 | wrong |
03:16.13 | JT | 10dollar, that assumes the second hand unit you buy fof ebay works |
03:16.17 | foobar778 | when u get it u will sing a differnt tune |
03:16.18 | JT | which may not be the case |
03:16.54 | foobar778 | how much does ur PC that runs ipcop cost????? |
03:16.58 | JT | umm, he's right, a linksys router is shit for a http proxy with anything but light traffic load |
03:16.59 | FuriousGeorge | foobar778: look, you obviously have done your homework, and know a lot about what openwrt can do. now you have to realize that just because it can do something, doesnt mean it can do it well, and just because it can do some THINGS doesnt mean it can do everything |
03:17.13 | JT | it cost me $0, but the market value was probably AUD$50 at the time, or USD$30 |
03:17.29 | foobar778 | well doesnt that apply to ipcop?? |
03:17.41 | JT | i don't use ipcop either |
03:17.42 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
03:17.52 | *** join/#asterisk orbikitti (n=orbitn@68-119-118-25.dhcp.jcsn.tn.charter.com) |
03:17.53 | foobar778 | so a low powerd PC is firewalling ur network |
03:18.02 | JT | no shit |
03:18.06 | JT | at home |
03:18.15 | JT | i have a much more powerful one at the office |
03:18.17 | foobar778 | well I have 6 on my lan |
03:18.40 | JT | my home low powered pc will outperform the linksys |
03:18.49 | JT | embedded machines have very limited cpu flash and hdd |
03:19.00 | JT | cpu flash and ram i mean |
03:19.28 | flenders | ahahha, I can't believe you guys are arguing about it |
03:19.42 | JT | :) |
03:19.42 | flenders | JT: just agree mate... |
03:19.46 | FuriousGeorge | foobar778: openwrt is just not viable for every application. i would not use it with or without asterisk installed because it is not appropriate for the application that started this whole conversation. im no expert but i know anyone who claims to be and says it is is wrong |
03:19.48 | JT | flenders: haha |
03:20.17 | foobar778 | i use ddwrt in frony of my asterisk machine |
03:20.23 | JT | cool |
03:20.27 | JT | i'm sure it works fine |
03:20.38 | flenders | :) |
03:20.38 | foobar778 | this started when u were talking about ur shitty netgear switch |
03:21.04 | foobar778 | my ddwrt is in front of the asterisk |
03:21.04 | JT | umm, that was FuriousGeorge |
03:21.09 | JT | i hate netgear |
03:21.20 | JT | and most consumer brands of cheap stuff |
03:21.20 | foobar778 | well that how this talk started |
03:21.22 | FuriousGeorge | foobar778: lets say you have 15 users on a moderately slow internet connection who visit basically the same 7 web pages all day. lets say those pages have content with a lot of images. guess what, openwrt is MUCH worse than a pc running linux as a firewall/proxy in that situation |
03:21.26 | FuriousGeorge | thats all there is to it |
03:21.28 | flenders | I hate the WRT54GP2 |
03:21.36 | flenders | worst router I've ever seen |
03:21.50 | JT | flenders: netcomm NB5 |
03:21.52 | JT | worst ever |
03:22.03 | JT | cannot hold PPPoE longer than 18hours |
03:22.08 | JT | then needs power cycle |
03:22.09 | foobar778 | hey there are no probs here conet filtering active x all from setup page |
03:22.20 | foobar778 | contentfiltering |
03:22.23 | FuriousGeorge | foobar778: besides flenders here we all like openwrt. its ok to like openwrt, its even ok to love openwrt, but stop *loving* openwrt |
03:22.34 | *** join/#asterisk LoneShadow (n=duh@c-24-6-162-76.hsd1.ca.comcast.net) |
03:22.34 | flenders | JT: mine needs a power cycle every 3 or 4 days |
03:22.45 | foobar778 | i love ddwrt acyually |
03:22.45 | flenders | wireless starts to get slower and slower |
03:22.46 | JT | FuriousGeorge: to be fair, flenders is probably talking about stock firmware not openwrt |
03:22.50 | JT | openwrt is aftermarket |
03:22.51 | FuriousGeorge | and by loving, of course, i mean felating at a rest stop. you barely KNOW HIM |
03:23.06 | foobar778 | openwrt is opensource |
03:23.12 | JT | FuriousGeorge: ahaha, audiable |
03:23.16 | JT | foobar778: correct |
03:23.52 | JT | flenders: netcomm has the worst tech support ever |
03:24.03 | flenders | those firmwares don't work with wrt54gp2 |
03:24.11 | flenders | I would even give it a go if it worked |
03:24.49 | foobar778 | maybe not not all wrt65g can be flashed example the v7 |
03:24.52 | flenders | I almost bought those ethernet over power things from netcomm once |
03:25.00 | flenders | just to get rid of my router |
03:25.06 | FuriousGeorge | JT: yeah ive installed openwrt to try to get rid of flakiness. the device was defective so all i got was a flaky device running linux which was much cooler, yet harder to work with :) |
03:25.18 | JT | FuriousGeorge: heh :) |
03:25.19 | foobar778 | the gs with 64mb ram would be best |
03:25.21 | *** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net) |
03:25.34 | foobar778 | i use a 2.2 with 16mb |
03:26.03 | FuriousGeorge | go price that one on ebay, then price a pentium II 500 with twice the ram and a harddrive that isnt solid state memory |
03:26.23 | foobar778 | kinda of cool to install lynkx on ur router and surf from it |
03:27.16 | JT | flenders: hey |
03:27.21 | foobar778 | I use wget from the router and drop huge files to a samb mount from the router |
03:27.40 | JT | sure, standard linux stuff |
03:27.55 | flenders | JT: yeah? |
03:27.57 | foobar778 | yes thats whats so cool the router is a Linux os |
03:28.21 | JT | flenders: when you get the PRI, you've got to tell me if optus allows setting callerid to something you don't own |
03:28.41 | flenders | JT: yeah, I asked the sales girl, but she didn't know |
03:28.45 | FuriousGeorge | linux on soekris (?) is cooler |
03:28.49 | flenders | JT: still waiting for their tech to ring me |
03:29.21 | foobar778 | solaris u mean?? |
03:29.21 | FuriousGeorge | no |
03:29.21 | JT | flenders: i think that's something you more try than ask |
03:29.26 | FuriousGeorge | turns out i spelled it right |
03:29.34 | InHisName | anyone install app_backtricks.c in their asterisk ? I have a need to read in file contents into the library |
03:29.35 | FuriousGeorge | i actually mean soekris |
03:29.38 | foobar778 | well never heared of it |
03:29.41 | flenders | JT: I'll tell you that in 6 weeks then |
03:29.45 | foobar778 | tell us |
03:29.45 | flenders | :) |
03:29.53 | FuriousGeorge | http://www.soekris.com/ |
03:30.09 | foobar778 | tell me about it one sentence |
03:30.14 | JT | no |
03:30.22 | InHisName | it is a single board computer |
03:30.24 | JT | you told us to go look everything up |
03:30.29 | JT | now you can do it :P |
03:30.45 | FuriousGeorge | foobar778: its cool |
03:30.46 | *** part/#asterisk orbikitti (n=orbitn@68-119-118-25.dhcp.jcsn.tn.charter.com) |
03:30.47 | FuriousGeorge | :) |
03:31.02 | foobar778 | yes I see |
03:31.07 | FuriousGeorge | that was my sentence |
03:31.33 | FuriousGeorge | you guys wanna know whats cooler than all this stuff though |
03:31.59 | JT | FuriousGeorge: gumstix. |
03:32.09 | JT | can't terminate PRI to them |
03:32.12 | JT | but you can do a lot of stuff witht hem |
03:32.22 | JT | and they are way smaller than soekris, linksys, etc |
03:32.37 | JT | www.gumstix.com |
03:32.39 | FuriousGeorge | and since they are EVEN cooler, they cost EVEN more |
03:32.39 | foobar778 | hey furious IIm using a dwg-1120S ur opinion has an fx0 port |
03:32.44 | FuriousGeorge | seems to stay with the theme :) |
03:32.46 | coppice | what makes the soekris more interesting than half a dozen other similar product lines? |
03:32.58 | JT | FuriousGeorge: they're actually fairly price competitive, all things considered |
03:33.53 | JT | anyone have any suggestions for a small form factor computer/embedded unit that has a 3.3v pci slot, x86 arch, and a small size? |
03:33.58 | JT | that's what i'm after |
03:34.03 | JT | found nothing suitable so far |
03:34.10 | flenders | ahhahahahahaha |
03:34.19 | flenders | the smallest flying webserver in the world! |
03:34.27 | JT | eh? :) |
03:34.32 | ping2921 | how can prevent a sip register timeout? |
03:35.02 | foobar778 | nice size gumstrix |
03:35.12 | foobar778 | pricy |
03:35.36 | JT | foobar778: actually competitively priced compared to other embedded linux solutions |
03:35.52 | foobar778 | real;tive I suppose |
03:36.39 | JT | for the size |
03:36.48 | JT | it's not for your standard router |
03:36.55 | JT | you can do much cooler things with it :) |
03:37.32 | foobar778 | still 9 times what I paid |
03:38.23 | foobar778 | imagine what u can do with a mainframe |
03:38.24 | flenders | I wanna do a router inside a robot! |
03:38.51 | foobar778 | Hey have u clusterd ur linux boxes yet?? |
03:39.51 | JT | no, but i'm planning a cluster over 2 machines, with about 16 virtual machines ;) |
03:40.08 | Nugget | I run a 100,000 computer cluster. |
03:40.21 | foobar778 | ok |
03:40.32 | foobar778 | Im President of the USA |
03:40.38 | Nugget | I do! |
03:40.42 | Nugget | distributed.net |
03:40.48 | JT | you run it? |
03:40.54 | Nugget | yes |
03:41.09 | [TK]D-Fender | JT : Get a Shuttle system, or something like an acer M-ATD desktop type case system |
03:41.31 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
03:41.40 | [TK]D-Fender | JT : Wide variety of MB's available some w/ dual NIC's etc |
03:41.43 | JT | [TK]D-Fender: i'm thinking more the motherboard than case |
03:41.51 | JT | i won't be using any standard cases |
03:42.02 | JT | most of the stuff available is VIA |
03:42.05 | flenders | JT: what do you want to cluster? |
03:42.06 | JT | via sucks |
03:42.18 | [TK]D-Fender | JT : I'm not talking VIA... VIA blows |
03:42.27 | JT | [TK]D-Fender: hrm, details? :) |
03:42.37 | JT | flenders: some voip stuff |
03:43.02 | [TK]D-Fender | http://eu.shuttle.com/en/DesktopDefault.aspx/tabid-2/ |
03:44.06 | flenders | and why does it need to be small? |
03:44.46 | [TK]D-Fender | http://usa.aopen.com/ |
03:45.00 | JT | what the hell, looks like you can't get a shuttle without a case |
03:45.14 | JT | flenders: the main cluster will be server class hardware |
03:45.51 | JT | flenders: but i want small stuff because rack space is expensive and it's a waste of space and energy using a full size pc to terminate PRIs |
03:46.14 | Hmmhesays | sweet geebus the model they used for enterprise D sold for $500,000.00 USD |
03:46.16 | [TK]D-Fender | JT : Wait... you're talking racks? So why not 1U's? |
03:46.16 | JT | i am actually thinking it's cheaper to go Redfone bridge than pri cards |
03:46.57 | JT | [TK]D-Fender: yes, the idea is for me to build a custom 1RU server which has a xeon mobo inside and 1 or 2, space depending, tiny pcs like mini ITX or similar |
03:47.12 | *** join/#asterisk Johnnie (n=jdlewis@jdlewis.org) |
03:47.37 | flenders | JT: I got 2 dells poweredge 850 for 1K AUD each here |
03:47.41 | [TK]D-Fender | JT : You can slap 2 x 8-port cards in an appropriately chosen 1U.... |
03:47.43 | flenders | they have 2 PCI slots |
03:47.48 | Johnnie | Anyone here overly familiar with func_odbc? |
03:47.54 | Hmmhesays | you can get them with pci slots |
03:47.57 | JT | [TK]D-Fender: i don't want to terminate on the same hardware |
03:48.05 | Hmmhesays | i use func_odbc |
03:48.12 | JT | i'd never go 8 port anyway |
03:48.17 | JT | too big a point of failure |
03:48.19 | Hmmhesays | get external pri gateways JT |
03:48.24 | JT | yes, |
03:48.32 | JT | i'm thinking redfone is a much better solution |
03:48.35 | Johnnie | I'm rather confused by the documentation on the VoIP Info Wiki. |
03:48.36 | Hmmhesays | I use them with no problems |
03:48.38 | JT | trying to see if it has hardware EC |
03:48.41 | Johnnie | It seems rather ambiguous to me. |
03:48.44 | JT | wonder if it has any other drawbacks |
03:48.45 | Johnnie | What am I missing? |
03:48.55 | JT | Johnnie: try the book |
03:48.58 | JT | ~thebook |
03:48.59 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
03:49.05 | Hmmhesays | I don't know Johnnie: what are you missing? |
03:49.45 | JT | Hmmhesays: does it have hardware EC? notice anything you can't do compared to native pci card? |
03:49.46 | flenders | JT: how much for the redfone? |
03:50.00 | JT | flenders: about USD$1800 for a 4 PRI model |
03:50.02 | Hmmhesays | JT yes |
03:50.06 | JT | very competitive |
03:50.09 | Hmmhesays | I use quintum pri gateways though |
03:50.10 | JT | Hmmhesays: awesome |
03:50.18 | Hmmhesays | i have no idea about redfone |
03:50.23 | Johnnie | JT: Thanks, I'll check that out. |
03:50.24 | JT | Hmmhesays: ah, was the yes for quintum or redfone? |
03:50.25 | JT | ok |
03:50.36 | JT | Hmmhesays: how much are the quintums? |
03:50.40 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
03:50.51 | JT | do quintums give you sip or tdmoe? |
03:50.56 | Hmmhesays | sip |
03:51.03 | Hmmhesays | I don't know I don't buy them |
03:51.05 | JT | hmm ok |
03:51.17 | Hmmhesays | I just make them work |
03:51.23 | JT | also, what's the RU vs. port density? |
03:51.29 | JT | sorry for all the questions :P |
03:52.09 | Hmmhesays | up to for pri's in a 1U unit |
03:52.26 | JT | 4, ok i guess |
03:52.42 | Hmmhesays | otherwise it goes 2u for 8 and 4 u for 28 |
03:53.10 | JT | yeah so maybe i'll go for 2RU of servers + 1RU of gateway |
03:53.25 | JT | and chuck a gumstix in each server for OOB/LOM |
03:54.12 | JT | seeing as for 4 pris, a pci card is almost the same cost as a whole gateway |
03:54.18 | JT | profiteering anyone? ;) |
03:55.18 | Hmmhesays | pci card doesn't have all the hardware you need to go from pri to sip |
03:55.24 | JT | i know |
03:55.30 | Hmmhesays | people seem to forget that |
03:55.34 | JT | hence why i was saying they were so much better value |
03:55.39 | Hmmhesays | ahh |
03:55.44 | JT | pci cards are overpriced |
03:56.29 | JT | most pci cards of any type that cost more than a thousand dollars each are major profit gouging |
03:56.36 | [TK]D-Fender | JT : Really? How much does a 1U 4-port PRI gateway go for? |
03:56.45 | JT | people can say "r&d", but it's really supply&demand |
03:56.50 | JT | [TK]D-Fender: USD$1800 |
03:57.27 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
03:57.41 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43638158336.htm |
03:57.44 | [TK]D-Fender | ^^^^^^^^^^^^ |
03:58.48 | [TK]D-Fender | JT : Redfone does NOT count. It is a super cheap PC running Digium's lowest end card without EC, and talk TDMoE which NOBODY cares about |
03:59.25 | bkruse_home | [TK]D-Fender: ha, i like the rundown ;] |
03:59.33 | JT | [TK]D-Fender: haha they're dreaming |
03:59.37 | JT | aww |
03:59.39 | JT | no EC? |
04:00.04 | JT | dash my hopes why don't you :P |
04:00.41 | [TK]D-Fender | but.... will it blend? ;) |
04:01.08 | JT | i thought it'd have to run asterisk |
04:01.17 | JT | didn't know it ran a low end digium card too |
04:01.27 | [TK]D-Fender | <- Ahm da pahty poopah |
04:03.08 | JT | USD$16k for a quintum 4 pri, wtf |
04:03.19 | JT | ok time to look at making my own again :P |
04:03.36 | [TK]D-Fender | Engineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, foneBRIDGE2 provides low-latency delivery of your critical voice traffic. <- Not SIP |
04:04.12 | JT | i knew that, it didn't worry me either |
04:04.25 | JT | if tdmoe works, fine |
04:04.32 | coppice | its pure E1/T1 to TDMoE. nothing more |
04:04.43 | [TK]D-Fender | JT : PRI gatways are pricy, but have a very different implementation design. they can be highly intelligent, link it with billing systems, have multiply layers of redundancy, etc. All probably overkill for your needs. |
04:04.45 | JT | coppice: is it a digium card? |
04:04.58 | [TK]D-Fender | JT : http://www.voipsupply.com/product_info.php?products_id=2026 |
04:05.02 | coppice | no. its a box from redfone |
04:05.07 | JT | [TK]D-Fender: billing isn't really what my gateway needs to do |
04:05.10 | JT | coppice: ok |
04:05.12 | bkruse_home | JT: nope |
04:05.22 | JT | [TK]D-Fender and coppice, battle! :P |
04:05.26 | puzzled | JT: for %16k you can get a MaxTNT on eBay which has a ton more functionality and redundancy than that quintum |
04:05.30 | puzzled | $16k even |
04:05.37 | JT | yeah |
04:05.51 | JT | i have a couple of 3Com totalcontrols, wonder if they do voice |
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04:06.07 | [TK]D-Fender | JT : read the specs. effectively is. |
04:06.16 | JT | [TK]D-Fender: yeah i was already on that page |
04:06.29 | [TK]D-Fender | coppice : I remember the 1st gen casing were it looked pretty damn obvious who's card was in there... |
04:06.30 | JT | [TK]D-Fender: the specs aren't low level |
04:07.02 | JT | [TK]D-Fender: i don't know how they could build it so cheap with digium card unless they got a massive discount |
04:07.09 | [TK]D-Fender | JT : Its a funnel between PRI & TDMoE. Nothing more. Now HWEC, no transcoding, no nothing. |
04:07.24 | JT | right |
04:07.43 | JT | say i could get some echo cancellers, i don't see a problem |
04:07.44 | [TK]D-Fender | JT : You DO know that the tormenta2 design wass "Open" right? there are plenty of compatable 4-port cards for about $1000 USD |
04:07.52 | JT | i don't want my gateway to transcode |
04:08.12 | JT | [TK]D-Fender: on pci, from who? |
04:08.22 | [TK]D-Fender | JT : Lemme dredge up a name... |
04:08.27 | [TK]D-Fender | JT : beena while.... |
04:08.31 | JT | heh |
04:09.43 | [TK]D-Fender | JT : For reference : http://www.zapatatelephony.org/ |
04:09.57 | JT | yeah i'm aware of the site |
04:10.05 | JT | and the original cards |
04:10.12 | JT | i thought they were relics of history |
04:10.23 | JT | not purchasable items |
04:10.37 | coppice | several people make and sell those cards |
04:11.15 | coppice | their main drawback is the lack of bus mastering |
04:11.19 | [TK]D-Fender | Damn... Can't recall the name.... one company made a fuss about 2 years ago.... |
04:11.22 | JT | hmm |
04:11.25 | JT | rhino? |
04:11.27 | [TK]D-Fender | nope |
04:11.40 | [TK]D-Fender | they were their own company before they started making shit cards ;) |
04:11.49 | coppice | they do report errors, though, which the current Digium cards hide. that makes them much easier to support remotely |
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04:11.58 | JT | in any case, i'm still in the market for a small x86 mobo |
04:12.30 | bkruse_home | coppice: things that are not zaptel debug related?? |
04:12.43 | bkruse_home | (refering to the card of course, pri debug, etc etc) |
04:12.44 | JT | via's mini itx stuff is 5v pci i believe and... VIA |
04:12.58 | coppice | E1/T1 link errors are hidden by the drivers for the current cards |
04:13.37 | bkruse_home | gotcha. |
04:13.45 | coppice | why do people always have a downer on VIA. Intel has produced most of the really troublesome chipsets |
04:13.50 | *** join/#asterisk PMantis (n=pmantis@cpe-69-207-130-14.rochester.res.rr.com) |
04:14.13 | PMantis | Is there a difference in dialing SIP/hostname/exten vs SIP/exten@hostname ? |
04:14.15 | JT | via has a phenominal habbit of sucking |
04:14.20 | [TK]D-Fender | coppice : just the C3 based uber-low-end junk. |
04:14.26 | JT | their itx stuff looks interesting |
04:14.30 | JT | just 5v pci :( |
04:14.49 | JT | [TK]D-Fender: the C3 processors are overkill for what i need anyway |
04:14.52 | coppice | can you name any 32bit PCI slot which is not 5V? |
04:15.04 | coppice | (clue: there is one) |
04:15.08 | JT | they're 3.3v these days |
04:15.10 | JT | pci2.2 |
04:15.19 | coppice | nope |
04:15.26 | JT | pci2.2 is 3.3v :) |
04:15.52 | coppice | its 3.3V signalling. the power on *every* board is 5V, except for one Dell design |
04:15.52 | JT | unless you're going for something obscure here |
04:16.03 | coppice | this is the same for VIA |
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04:16.18 | JT | right, so when someone says "3.3v pci, do they mean power or signalling"? |
04:16.43 | coppice | they are normally referring to power, because evry slot for 10 years has accepted 3.3V or 5V signalling |
04:17.02 | JT | since pci2.2? |
04:17.12 | coppice | long before that |
04:17.19 | JT | heh |
04:17.32 | JT | i know a lot of wireless cards didn't work on pre pci2.2 boards |
04:17.52 | coppice | that has nothing to do with voltages |
04:18.07 | JT | uhuh |
04:18.27 | JT | so you can buy digium pri cards in 5v or 3.3v, expand :) |
04:18.59 | [TK]D-Fender | JT : found one http://www.mapleleaf-technologies.com/webstore/openvox_voicedatacards.php |
04:19.05 | coppice | because those cards are so lousy they can't adapt to a 3.3V or 5V supply like 99% of all other PCI cards can |
04:19.31 | [TK]D-Fender | http://www.mapleleaf-technologies.com/webstore/varion_cards.php |
04:19.39 | [TK]D-Fender | Varion was the name I was looking for... |
04:19.51 | coppice | varin is tormenta 2. openvox is not |
04:19.58 | coppice | varion |
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04:20.23 | JT | coppice: heh |
04:21.00 | [TK]D-Fender | coppice : crapTASTIC aren't they? :) |
04:21.19 | JT | openvox, is there anything actually wrong with them? |
04:21.23 | coppice | if you have a TE410P card, which requires a 3.3V supply there appears to have only ever been one 32bit slot made that you can plug it into. It was a Dell. everyone else uses those cards in 64 bit slots, which are almost always 3.3V powered |
04:21.45 | coppice | people seem to say nice things about openvox, but I've never used one. |
04:21.56 | [TK]D-Fender | JT : Ancient PCI design, probably next to NIL on support. |
04:22.08 | JT | ancient pci design? |
04:22.34 | [TK]D-Fender | JT : Even Digiums lowest end has gone through a number of revisions... |
04:22.51 | JT | are you sure the openvox units are not revised? |
04:23.01 | [TK]D-Fender | JT : Would you risk it? |
04:23.06 | coppice | openvox appear to support their stuff OK |
04:23.18 | JT | well, i'd check if it was a legitimate concern first |
04:23.27 | [TK]D-Fender | JT : vs buying from a known company that is still producing cards and improving on things? |
04:23.42 | JT | heh |
04:24.01 | coppice | openvox introduces new cards pretty regularly. you are just spreading FUD |
04:24.04 | JT | looks like openvox are still making cards too |
04:24.06 | JT | so meh |
04:24.24 | [TK]D-Fender | coppice : No... I'm selling it as certainty... thats MARKETING ;) |
04:25.12 | coppice | no. marketing spread FUD. spreading false certainty gets you sued |
04:27.40 | [TK]D-Fender | coppice : If I sell MY uncertainty as "the word", I'm entitled to be wrong, its still not a lie. Also I have no economic gain, and the fact that these cards are virtually unheard of does give cause for concern. |
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04:28.18 | [TK]D-Fender | coppice : But I'll take the edge off my comedy.... it is just a little to brisk on this topic. |
04:30.51 | n|cotine | What driver do Digium TE2xxP cards use? |
04:31.58 | [TK]D-Fender | n|cotine : Zaptel. Or are you referring to the module name more specifically? |
04:32.27 | [TK]D-Fender | n|cotine : which would be : wct2xxp |
04:32.28 | n|cotine | The latter. |
04:32.52 | n|cotine | [root@voip extra]# ls -la wct2* |
04:32.52 | n|cotine | ls: wct2*: No such file or directory |
04:33.05 | n|cotine | From zaptel-1.2.13 |
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04:33.42 | n|cotine | I am slightly confused. |
04:34.07 | Juggie | have you considered the documentation |
04:34.11 | n|cotine | Aha. |
04:34.14 | n|cotine | Aliased. |
04:34.43 | n|cotine | alias wct2xxp wct4xxp in modules.alias |
04:40.39 | Snake-Eyes | any one know of write up of how to configure a box with two tdm400P cards ? (yes i have tried googleing) |
04:41.02 | Qwell | Snake-Eyes: it's the same as one, you just configure more ports |
04:43.14 | Snake-Eyes | Qwell, i tried that, but I got chan_zap errors so I figured I might be going down the wrong path as it were |
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05:13.36 | zeeesh | hi |
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05:15.13 | PMantis | Is there a difference in dialing SIP/hostname/exten vs SIP/exten@hostname ? If the first dead-wrong? |
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05:23.22 | JT | well i think you mean SIP/sip.confentry/exten to be technically correct |
05:23.28 | JT | it's not necessarily hostname |
05:24.47 | sbingner | JT, I don't suppose you actually had an answer to his question in addition to the inane comment? |
05:25.43 | JT | sbingner: inane comment, i'm sorry? |
05:26.13 | JT | i am attempting to be helpful |
05:26.27 | JT | it's a common misconception that it's SIP/hostname/exten |
05:26.37 | JT | so i was partially answering his question |
05:26.56 | JT | using a sip.conf entry, so the former method, is preferable |
05:27.20 | JT | as you can set all the options up in sip.conf on a per host basis |
05:27.39 | JT | sbingner: so exactly what was inane? |
05:32.03 | [TK]D-Fender | PMantis : For a hostname, use the latter, for a peer-name, the former, and it CAN matter and its cause DNS resolution errors when formatted improperly. |
05:33.40 | PMantis | [TK]D-Fender, Thanks... so for a peer, the "IAX format" will work. :) |
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05:35.21 | JT | PMantis: it will, iax/sip format, whatever you want to call it :) |
05:35.28 | JT | it's nicer to use sip.conf though |
05:36.34 | PMantis | Oh of course... I just recently noticed freepbx uses the SIP/peer/exten format... I always used exten@peer for my own scripts. |
05:36.50 | PMantis | well, scripts, dialplan... whatever. :) |
05:37.40 | JT | yeah i never use the exten@peer format |
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06:09.53 | FuriousGeorge | hmmm, is there a way to match any digit/character or none? |
06:10.20 | FuriousGeorge | specifically actually i want * or no * |
06:10.37 | FuriousGeorge | [ *]wouldnt work would it |
06:10.50 | JT | it should |
06:10.57 | JT | * should match * |
06:11.02 | FuriousGeorge | cool |
06:11.08 | FuriousGeorge | thanks |
06:11.20 | JT | for ANY digit/non digit it's . |
06:11.40 | JT | . uses timeout dialling when dialling digit by digit |
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06:11.59 | JT | as it means 1 or more chars, iirc |
06:14.39 | [TK]D-Fender | JT :no "." if for any NUMBER of any digits, not a SINGLE. |
06:14.48 | JT | i know |
06:14.51 | JT | that's what i said |
06:14.54 | JT | 1 or more chars |
06:15.12 | JT | and it matches non-dtmf too |
06:15.23 | [TK]D-Fender | JT : jsut took you 3 sentence fragments to peice it together ;) |
06:15.32 | [TK]D-Fender | JT : yup |
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06:41.33 | Fr0zen_ | anyone here use Cisco ip phones? |
06:53.56 | FuriousGeorge | ; Set iaxcompat to yes if you plan to use layered |
06:53.56 | FuriousGeorge | ; switches. It incurs a small performance hit to enable it |
06:53.56 | FuriousGeorge | ; |
06:53.56 | FuriousGeorge | ;iaxcompat=yes |
06:54.14 | FuriousGeorge | layered switch just means two "daisy chained" switches between you and wan, right? |
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07:13.50 | FuriousGeorge | darndest thing: i set up 3 asterisk boxes as friends of each other the exact same way. for some reason, one of them rejects the call as its trying to send it to the 'default' context even though the other box is set up as a friend in the 'inside' context |
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07:16.30 | FuriousGeorge | request '100@default' does not exist |
07:16.31 | FuriousGeorge | where is it getting 100@default, from?! |
07:19.57 | Juggie | care to provide more information |
07:20.47 | Juggie | obviously, some phone is getting into the default context |
07:20.52 | FuriousGeorge | Juggie: sure, ill pb something |
07:21.27 | Juggie | your sip/iax/extensions would be handy |
07:21.34 | Juggie | and the console output |
07:22.41 | Juggie | hurry too i'm sleepy |
07:22.44 | Juggie | its almost 2:30am :P |
07:22.59 | FuriousGeorge | http://pastebin.ca/364575 |
07:23.41 | FuriousGeorge | Juggie: let me know if you still need extensions, but from the snippets of the other confs and the cli output, i think we can agree that shouldnt happen, no |
07:25.02 | Juggie | more confs would be helpful. |
07:25.20 | Juggie | your extensions.conf plz. |
07:25.30 | FuriousGeorge | coming right up |
07:25.53 | Juggie | your dialstring from the other box would be handy doo. |
07:25.55 | Juggie | *too |
07:26.47 | Juggie | homer just finished making up with marge so hurry :) |
07:30.14 | FuriousGeorge | http://pastebin.ca/364583 |
07:30.18 | FuriousGeorge | :) |
07:30.31 | FuriousGeorge | that homer |
07:31.11 | Juggie | FuriousGeorge, iax is differnt then sip. |
07:31.29 | FuriousGeorge | you want to see my sip.conf? |
07:31.38 | Juggie | no, i already know the problem |
07:31.39 | FuriousGeorge | i should mention that this works in the other direction |
07:31.54 | Juggie | exten => _61XX,n,Dial(IAX2/Juanita/${EXTEN:1}@inside,60,,t,T) |
07:31.56 | Juggie | should do the trick |
07:32.39 | FuriousGeorge | i believe you, but i shouldnt need to do that... i dont need it on the other guys for some reason |
07:32.43 | FuriousGeorge | there are three boxes set up this way, this is the only one that doesnt accept calls |
07:32.59 | Juggie | i dont have an answer to that. |
07:33.04 | Juggie | iax is different then sip |
07:33.11 | Juggie | sip.conf is this peer has this context |
07:33.16 | Juggie | but iax can have many contexts. |
07:33.23 | Juggie | you can even do context=* |
07:34.04 | FuriousGeorge | i guess ill have to do that if i cant figure out why this one isnt acting right |
07:34.10 | FuriousGeorge | but its annoying me atm |
07:35.00 | Juggie | i am too sleepy to dive into all the configs |
07:35.12 | Juggie | you would think you would not need to specify it when there was only one |
07:35.41 | kaldemar | FuriousGeorge: on the Claudia box, do you have 'username=Claudia' in context Juanita? |
07:36.16 | FuriousGeorge | kaldemar: username= in the context? you mean the peer entry? |
07:36.18 | Juggie | FuriousGeorge, and from what i can see that is how it is designed to work |
07:36.20 | FuriousGeorge | now that i think about it |
07:36.23 | kaldemar | it may not get the right username when dialing and falls back to default context because of that. |
07:36.26 | Juggie | so, something is mucked up somewhere |
07:36.32 | FuriousGeorge | this box used to be called juanita, but now the hostname is edith |
07:36.34 | Juggie | but its most likely not a bug. |
07:36.35 | kaldemar | FuriousGeorge: yes, the peer entry. |
07:37.05 | FuriousGeorge | is it possible that changing the hostname of the box would cause this somehow? |
07:37.07 | kaldemar | FuriousGeorge: then it uses that username and the defined secret to dial the peer if the Dial application is called that way. |
07:38.40 | Juggie | FuriousGeorge, for fun add @inside to the dial |
07:38.44 | Juggie | i bet it fails some other way |
07:38.49 | Juggie | which may be more helpful |
07:39.22 | FuriousGeorge | Rejected connect attempt from blah.84.214.199, who was trying to reach '100@' |
07:39.31 | FuriousGeorge | thats what happens when i set username so were on to something |
07:39.39 | Juggie | bingo, auth problems. |
07:39.43 | Juggie | have fun, i'm going to sleep :) |
07:39.53 | FuriousGeorge | Juggie: thanks for the time |
07:39.55 | FuriousGeorge | night |
07:39.57 | Juggie | np. |
07:40.01 | Juggie | must work @ 9am :) |
07:40.03 | Juggie | and its 2:30 |
07:40.10 | FuriousGeorge | godspeed to you :) |
07:40.45 | kaldemar | FuriousGeorge: do you have friend entries in iax.conf or peer and user separately? |
07:40.51 | Juggie | yah, its some stupid auth issue.... you'll get it. |
07:41.23 | kaldemar | FuriousGeorge: if friends, are the secrets the same? |
07:42.21 | FuriousGeorge | Rejected connect attempt from 67.84.214.199, who was trying to reach '100@inside' |
07:42.32 | Juggie | also, set verbose 10 |
07:42.33 | FuriousGeorge | thats what happens when i tag @inside at the end of the dial |
07:42.35 | Juggie | to get more info |
07:42.52 | Juggie | yes, the auth is failing |
07:42.56 | FuriousGeorge | no new info there |
07:43.15 | Juggie | do you have iax setup with two peer=friend 's? |
07:43.20 | Juggie | same secret=... ? |
07:43.32 | FuriousGeorge | yeah |
07:43.44 | Juggie | look again |
07:44.42 | Juggie | pastebin me the iax.conf entries from each box. |
07:44.58 | FuriousGeorge | Juggie: one sec |
07:45.07 | FuriousGeorge | just the peer section or the whole thing? |
07:45.19 | Juggie | the section for each peer |
07:47.59 | FuriousGeorge | http://pastebin.ca/364591 |
07:48.11 | FuriousGeorge | i double and triple checked the actual values for secret |
07:48.12 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
07:48.47 | Juggie | and both dialstrings |
07:48.48 | Juggie | please |
07:49.12 | FuriousGeorge | you got it |
07:50.06 | Juggie | i have it? or you are getting it? |
07:50.39 | *** join/#asterisk shinux__ (n=shinux@196.220.27.78) |
07:51.02 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
07:51.21 | FuriousGeorge | http://pastebin.ca/364592 |
07:51.32 | FuriousGeorge | you got it, as in "your wish is my command" |
07:51.56 | FuriousGeorge | is it so straight forward and obvious that it is blinding us |
07:52.01 | FuriousGeorge | i mean, WTF :) |
07:52.43 | *** join/#asterisk ciscosurfer (n=magic@216-80-124-12.snb-bsr1.chi-snb.il.cable.rcn.com) |
07:53.00 | FuriousGeorge | Juggie: hope you see something there im missing |
07:53.10 | Juggie | in shell, if you do a 'ping Juanita' |
07:53.15 | Juggie | does it resolve to something? |
07:53.20 | Juggie | on the box that cant make calls. |
07:54.51 | FuriousGeorge | k if i ping juanita (or her gotdns address in this case) from the box that can make calls, im pinging myself |
07:54.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
07:54.54 | FuriousGeorge | and it resolves fine |
07:54.57 | FuriousGeorge | same other way |
07:55.17 | FuriousGeorge | the one thing that has changes is that claudia's hostname is claudia whereas juanitas used to be juanita and is now edith |
07:55.38 | FuriousGeorge | for some reason im scared to remotely change the hostname, though i dont see what would cause that |
07:56.05 | kaldemar | what's the deal with all the commas in the dial lines? |
07:56.15 | FuriousGeorge | rather: i dont see how changing the hostname would affect it |
07:57.06 | FuriousGeorge | kaldemar: looks like at first i didnt have seconds specified, then i added seconds but didnt remove the comma |
07:57.25 | FuriousGeorge | or is it that i need two commas anyway if i dont use (insert option here) before t,T option |
07:57.49 | *** part/#asterisk ciscosurfer (n=magic@216-80-124-12.snb-bsr1.chi-snb.il.cable.rcn.com) |
07:58.15 | kaldemar | Dial(...,60,tT) <-- i'd put it that way. |
07:59.33 | kaldemar | according to wiki there's an URL parameter after the third comma. |
07:59.33 | FuriousGeorge | ill change it but i dont think its causing my problem |
07:59.44 | FuriousGeorge | hmmmm |
08:00.00 | *** join/#asterisk topping (n=topping@204.152.96.238) |
08:00.36 | kaldemar | i don't think it directly causing the problem, but you never know if there's a bug or something.. less errors, less confusion. |
08:02.44 | Juggie | i dont see anything blatentally wrong, but i am fairly sleepy, ask me tomorow when i'm @ work and mostly awake if your still stuck. |
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08:04.01 | FuriousGeorge | thanks Juggie |
08:04.23 | FuriousGeorge | kaldemar: took out all the seconds and t and comma since the remote context handles all that anywya |
08:04.25 | FuriousGeorge | no luck |
08:04.42 | FuriousGeorge | back to seeing '100@' in remote cli |
08:05.20 | Juggie | hmmmmmmm |
08:05.30 | Juggie | the only way * has to match the incomming peer is the host |
08:05.51 | Juggie | check that |
08:06.04 | FuriousGeorge | not sure what you mean |
08:06.06 | FuriousGeorge | host= setting |
08:06.09 | FuriousGeorge | that's fine |
08:06.27 | FuriousGeorge | otherwise iax2 show peers/users would reflect |
08:07.22 | FuriousGeorge | there are some dull moments, but never a dull entire day :) |
08:08.32 | FuriousGeorge | gasp |
08:08.41 | FuriousGeorge | isnt there a dnsmanager file or something |
08:08.53 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
08:09.01 | carrar | try adding deny=0.0.0.0/0.0.0.0 & permit=1.2.3.4/255.255.255.255 on each |
08:09.18 | carrar | specify the IP /32 for each |
08:09.27 | Juggie | thats not it |
08:09.35 | carrar | then you have it working |
08:09.39 | carrar | no problem then |
08:10.28 | Juggie | http://pastebin.ca/364606 |
08:10.29 | Juggie | try that |
08:10.51 | Juggie | i would have my working configs if i was at the office, but i'm not and havnt worked on * in like a year so i'm hazey :) |
08:11.23 | Juggie | zzzz. |
08:12.13 | carrar | For every IAX peer I havem I defined the following: |
08:12.14 | carrar | type= user= host= secret= context= peercontext= accountcode= auth= deny= permit= disallow= allow= |
08:14.18 | carrar | exten => s,1,Dial(IAX2/IAX-namehere/${ARG1},100) |
08:14.22 | carrar | put that in a macro |
08:15.02 | FuriousGeorge | Juggie: looks like setting the usrname on both sides worked, homeslice |
08:15.07 | FuriousGeorge | youve earned your sleep |
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08:15.26 | Juggie | FuriousGeorge, i bet the box that coudnt be called INTO had more then one iax peer defined |
08:15.28 | Juggie | correct? |
08:15.30 | carrar | besure to set a default |
08:15.38 | FuriousGeorge | Juggie: sure did |
08:15.52 | FuriousGeorge | i actually commented out a peer that came in the default config while i was in there |
08:16.01 | Juggie | yah, and more then one with the same host= |
08:16.11 | FuriousGeorge | i dont think so |
08:16.16 | FuriousGeorge | lest i messed up and didnt notice |
08:16.22 | FuriousGeorge | i looked for Dups though |
08:16.45 | Juggie | something is causing the auth to fail, it should go oh, host=blah thats that peer, but its not. |
08:16.46 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
08:16.52 | Juggie | anyways, you can figure that out later |
08:16.57 | Juggie | i allways use username |
08:17.00 | Juggie | makes things easier |
08:17.08 | kaldemar | my guess it that the call doesn't have a proper username. |
08:17.10 | Juggie | it will never fail and doesnt depend on any guesswork |
08:17.10 | FuriousGeorge | if you want, after you sleep, ill take out username and see if taking out the demo peer did it |
08:17.38 | Juggie | you can try it, just say what happened, i'll look over last nights irc log tomorow. |
08:17.39 | FuriousGeorge | everything is identical on both ends and it only worked both ways. right down to the motherboard and the sipphone |
08:17.45 | FuriousGeorge | the isp is the only var |
08:18.03 | Juggie | but allways user username= its more foolproof. |
08:18.07 | FuriousGeorge | Juggie: its not that i want you to go to bed, im in the same timezone and dont have the energy to mess with this anymore :) |
08:18.08 | Juggie | and easier to see whats going on |
08:18.21 | FuriousGeorge | Juggie: will from now on |
08:18.45 | Juggie | instead of * guessing which peer your looking for per hostname |
08:18.49 | Juggie | it knows exactally. |
08:19.00 | Juggie | zzz |
08:19.05 | FuriousGeorge | makes sense |
08:19.14 | FuriousGeorge | later dude, see you tomorrow |
08:19.35 | Fr0zen_ | anyone here use cisco ip phones? |
08:22.09 | carrar | I do |
08:22.20 | carrar | 7940/7960 |
08:23.47 | carrar | as well has LinkSys 94X and Polycom 6XX with sidecars |
08:24.11 | carrar | and softphones |
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08:35.07 | FuriousGeorge | i have the first sipura/linksys model |
08:35.10 | FuriousGeorge | think its an 841 |
08:35.38 | FuriousGeorge | a monkey on qualudes could not have designed a less intuitive telephone |
08:38.11 | FuriousGeorge | speaking of monkies: this iaxcompat=yes option that's commented by default in iax.conf... when it says it can be disabled if i have "layered switches", what does that mean exactly |
08:39.11 | FuriousGeorge | according to wikipedia a multilayer switch just means a device switching using a cpu rather than an asic, is that what that refers to? |
08:39.22 | FuriousGeorge | or does that just mean daisy chaining regular old switches |
08:40.29 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
08:41.01 | FuriousGeorge | its not that i dont know the difference, i just dont know what that comment in iax.conf is talking about |
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08:56.11 | *** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de) |
08:56.18 | codey | hi there |
08:56.36 | codey | can somebody help me on a queue? I'm new to asterisk and I just have to add something to my queue ... but I don't get it. |
08:56.50 | codey | If someone calls our number with an ending 0 he gets into the default queue |
08:57.09 | FuriousGeorge | ive never implemented a queue, but maybe if you link me to what you're trying to do i might could help |
08:57.18 | codey | default queue calls SIP/1 and SIP/2 - and now I just want to get SIP/3 and SIP/4 called if nobody gets on the phone |
08:57.37 | codey | after 10 secs |
08:58.07 | *** join/#asterisk Dibbler_XP (n=Dibbler_@host217-45-198-229.in-addr.btopenworld.com) |
08:58.30 | codey | or just another question.. what's the second parameter of Dial()? ... maybe this is just what I need to know |
08:59.02 | codey | so i can just do exten => bleh,Dial(SIP/1&SIP/2,10) exten => bleh,2,Dial(SIP/3&SIP/4) |
08:59.30 | FuriousGeorge | i know that one off hand dial(tech/chan/exten,options) |
08:59.49 | FuriousGeorge | you can do that |
09:00.00 | FuriousGeorge | but you want to wait 10 |
09:00.17 | FuriousGeorge | there is a queue app you dont seem to be using |
09:00.25 | FuriousGeorge | or am i misunderstanding |
09:01.37 | codey | one sec, i'll paste the part of my extensions |
09:02.21 | kaldemar | Dial(<channel>,<timeout>,<options>) timeout is how many seconds you let it ring before moving on in the dialplan unless the remote end answers. |
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09:04.32 | codey | http://rafb.net/p/X4o6iA84.html |
09:05.27 | codey | if i'm understanding this right - it defines the 4 sip extensions and then goes on to dial them |
09:05.48 | codey | and if noone gets on the phone within 12 seconds it goes to "queuezentrale"? |
09:06.09 | FuriousGeorge | http://www.voip-info.org/wiki-Asterisk+call+queues |
09:06.53 | FuriousGeorge | if you use goto you have "to go somewhere". why didnt you show me context [queuezentrale] |
09:07.02 | FuriousGeorge | zentrale is italian? |
09:09.27 | codey | german |
09:09.39 | FuriousGeorge | not even close :) |
09:10.36 | codey | http://rafb.net/p/fbXJW579.html |
09:11.14 | FuriousGeorge | you are using app_queue, and you want more phones than your tutorial teaches? you probably want round robin, two extensions at a time? |
09:11.50 | codey | i didnt do the configuration, i jsut have to add something :/ |
09:12.24 | FuriousGeorge | codey: are you just trying to make a queue using dialplan logic? |
09:12.34 | FuriousGeorge | do you know what i mean by that |
09:12.59 | codey | i think so, yes ;) |
09:13.08 | FuriousGeorge | ok, wait one sec |
09:13.58 | FuriousGeorge | how about this, ill design a simple queue so you will learn how it works. you want to dial sip/1&2 then 3&4 with a 10 second pause? |
09:14.04 | FuriousGeorge | then restart? |
09:14.18 | FuriousGeorge | not necessarily in that order |
09:15.05 | codey | sounds good, why not |
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09:16.39 | FuriousGeorge | so, let's say you put someone in the queue by dialing 99 |
09:20.55 | codey | okay |
09:24.45 | FuriousGeorge | ok almost done sorta |
09:28.05 | codey | okay .. i've got to KILL someone now |
09:28.11 | codey | they've sent me the wrong configfiles |
09:28.13 | codey | ... |
09:28.20 | FuriousGeorge | http://pastebin.ca/364667 |
09:28.48 | FuriousGeorge | i started to do the complicated version where you make a terminating loop happen, but i figured i'd leave that to you |
09:29.06 | FuriousGeorge | i /think/ that will work |
09:29.17 | codey | okay, thanks :) i'll take a look at that and try it with that |
09:29.28 | FuriousGeorge | like i said though, ive never used app_queue, or tried to do it myself |
09:29.52 | FuriousGeorge | also, they wont hear music, but ringing |
09:29.58 | codey | oh my f*cking god |
09:30.03 | codey | I'm an asterisk noob - okay |
09:30.08 | codey | but their queue is ... kinda lame |
09:30.14 | codey | even *I* understand it oO |
09:30.24 | codey | its funny... |
09:30.56 | codey | http://rafb.net/p/4eqQMc14.html |
09:31.08 | codey | am I the only thinking that *this* is totally useless? :> |
09:31.42 | FuriousGeorge | arent those parts of different but related files you are pasting |
09:31.50 | codey | the first on is extensions.conf |
09:31.50 | FuriousGeorge | one is extensions.conf and the other is queue.conf |
09:31.52 | codey | the second part is queue.conf |
09:33.04 | FuriousGeorge | the only problem i see is that youll never get into the queue b/c there is no priority one |
09:33.19 | FuriousGeorge | i dont think thats useless, its the right way to do what its intended to do |
09:34.34 | FuriousGeorge | like i said i dont use queue's so im not sure as to the exact behavior, but the general idea is clear |
09:34.56 | FuriousGeorge | but you want round robin and different agents |
09:35.37 | FuriousGeorge | or send them to another queue when/if they return |
09:35.37 | FuriousGeorge | hope that helps, got to run |
09:37.42 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
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09:44.46 | BobHenk | Does anyone have a working DUNDi configuration for Trixbox 2.0? I can't get it to work, so every help is very welcome. |
09:44.50 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
09:45.13 | phearless | hello ! |
09:45.57 | phearless | when I press 00 I got call ended on my Sipura 942 phone, very strange |
09:46.09 | phearless | Linksys/Sipura SPA942 |
09:46.21 | phearless | with (0[1-9]xxxxxxxxx|4xx) as a dialplan |
09:46.47 | phearless | exten => _0XXXXXXX.,1,Dial(Zap/g1/${EXTEN}) and this for outbound calls in extensions.conf |
09:46.52 | phearless | how can I debug this? |
09:49.38 | *** join/#asterisk friedrich| (n=friedric@e177248017.adsl.alicedsl.de) |
10:00.26 | phearless | I tried with : (0[1-9]xxxxxxxxx|4xx|00xxxxxxxxxxx) |
10:00.30 | phearless | on the phone dialplan |
10:00.33 | phearless | and same problem |
10:05.13 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
10:05.15 | Stephnie | hi |
10:05.39 | Stephnie | is there any way to secure AGI PER SCRIPTS? like If I can compile or something like that? |
10:06.11 | Stephnie | PERL* |
10:08.07 | simplexio | how i get channel status from sip user, like availabe or busy(if there is call going) |
10:09.17 | simplexio | or more like sip extension. ChanIsAvail(SIP/2009|sj) returrs allways 0? and goes to next priority |
10:25.02 | kippi | hi, I have added pickup to 3 of my extenstions, two of them will pickup, the other one says nothing to pickup, any ideas? |
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10:31.41 | phearless | this "00" is crazy |
10:31.53 | Stephnie | is there any way to secure AGI PERL SCRIPTS? like If I can compile or something like that? |
10:31.56 | phearless | I can not dial any 00something number |
10:32.09 | phearless | Stephnie: I do not know! |
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10:34.09 | phearless | it does not seems to be a vertical activation code problem |
10:37.18 | *** join/#asterisk ipguy (n=ipguy@124-168-21-206.dyn.iinet.net.au) |
10:37.39 | ipguy | hi all, i just setup my first asterisk box :-) |
10:37.54 | ipguy | all working |
10:37.54 | phearless | cool ! |
10:38.01 | ipguy | except |
10:38.20 | ipguy | i can't sip call other sip users |
10:38.36 | ipguy | as in from one asterisk box to another |
10:38.51 | *** join/#asterisk XIN01OZ (n=slink@c-68-63-34-189.hsd1.al.comcast.net) |
10:38.54 | ipguy | all local accounts work fine, just no remote access |
10:39.02 | phearless | not an NAT problem ? |
10:39.11 | ipguy | no NAT or firewall |
10:39.15 | *** join/#asterisk lorinc (n=ang@217.20.136.141) |
10:39.36 | kippi | i am getting this problem with codec, how can I sort this out? http://www.pastebin.ca/364714 |
10:40.00 | XIN01OZ | Anybody know by chance why after a call is placed through a2billing it calls the number and hangs up on pickup? |
10:41.15 | ipguy | phearless: does one need to setup something in extensions.conf to allow sip call from two remote asterisk boxes ? |
10:41.29 | kippi | also getting this: http://www.pastebin.ca/364717 |
10:41.29 | phearless | ipguy: I do not know |
10:41.58 | ipguy | phearless: ok thanks |
10:44.47 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
10:44.57 | kezza491 | Hi |
10:47.27 | kezza491 | Would some one be able to explain to me if there is like a free service so you can like ring your computer to check if your astrisk set up is ok? |
10:49.50 | Poincare | kezza491: I think fwd had such a service |
10:50.02 | kezza491 | hmm k |
10:50.09 | Poincare | freeworld dialup |
10:50.23 | kezza491 | Just i want to fiddle around with asterisks before i get commited to anything |
10:50.26 | Poincare | but that's only for the sip/iax accounts |
10:50.53 | kezza491 | ?! |
10:51.05 | Poincare | how do you want to ring your computer? |
10:51.07 | kezza491 | Have'nt actual started to work on asterisks as yet |
10:51.11 | Poincare | via the PSTN line? |
10:51.27 | kezza491 | Ehh i dont mind it wont matter in the end will it that much? |
10:51.28 | Poincare | Or do you want to test an 'incomming' SIP/IAX line? |
10:51.52 | kezza491 | ok what is a SIP/IAX line? |
10:52.05 | Poincare | SIP and IAX are voice over IP protocols |
10:52.19 | kezza491 | my idea is that i connect the phone line to the comp and also have the comp dial out using the net but for learning purposes just the net for now |
10:52.55 | Poincare | in that case: register with FWD |
10:52.59 | kezza491 | Ok |
10:53.23 | kezza491 | is it possible to have asterisks work with the net and a phone line? |
10:53.31 | Poincare | if your asterisk is working, it won't matter that much if its a 'real' line or not |
10:53.33 | Poincare | sure |
10:53.42 | kezza491 | Ok |
10:54.39 | kezza491 | So no hidden costs or anything with this freeworld? |
10:56.09 | Poincare | no, as long as you don't want to call PSTN lines it's completely free |
10:56.31 | Poincare | and they have several test numbers and a 'call back test service' |
10:56.51 | kezza491 | Ohh k |
10:57.14 | kezza491 | so say if i have an asterisks box how do i interface that with say another computer like a windows machine? |
10:57.35 | Poincare | and if you ask arroun here, maybe someone will call your fwd number to test |
10:57.53 | kezza491 | ahh k |
10:58.19 | kezza491 | Hate to be a pain but what is some really bare basic easy to understand asterisk doco? |
10:58.20 | Poincare | asterisk <=> windows boxes: SIP clients or IAX clients |
10:58.39 | kezza491 | k |
10:58.41 | kezza491 | thanks |
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11:26.19 | shadebob | hi, |
11:28.34 | shadebob | I need some help with multiple registration on the same SIP provider. Problem is asterisk take the context of the last registered peer for incoming calls for this provider. I don't want to create a single context with multiple exten=>number,... because I have dev many GUI. Can asterisk can manage the good context in this case? |
11:28.57 | shadebob | I see an old project in the svn tree (oej/register) but it seem abandonned |
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11:52.32 | kezza491 | What is some good documentation for astrisks for people just starting off that is simple and easy to understand |
11:53.27 | E-bola | the book was a good starter for me |
11:53.38 | E-bola | to get a basic understanding of the different terms and concepts |
11:53.44 | E-bola | its however not enough to actualy setup a system |
11:55.08 | kezza491 | hmm |
11:55.22 | kezza491 | mabe a tutorial is my best bet then fiddle around until i get my desired effect |
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12:09.19 | XIN01OZ | whoa im having it rough this morning for some reason when a call from SIP is placed the call makes it through but once picked up by the called number asterisk hangsup after 2 secs and I get a declined |
12:09.27 | XIN01OZ | cause code 16 |
12:09.42 | XIN01OZ | anybody familiar with what is going on |
12:11.17 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
12:13.20 | XIN01OZ | RoyK: Good morning we talked awhile back .. might you know where i could get some more information about why a call is being dropped 2 seconds after it is picked up by the called number |
12:13.27 | XIN01OZ | all im seeing is cause code 16 |
12:13.57 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
12:14.21 | XIN01OZ | didnt expect this .. planned something for someone this morning thought all was well |
12:14.42 | JT | 16 is normal clearing |
12:15.00 | RoyK | XIN01OZ: no idea |
12:15.17 | *** join/#asterisk mickey9 (n=mickey9@82-69-78-118.dsl.in-addr.zen.co.uk) |
12:15.21 | JT | 16 means successful call, hanging up, from memory |
12:15.53 | XIN01OZ | yeah that was all i could find as well but my sip client gets Declined |
12:16.08 | mickey9 | Hello All, I've decided to buy ABE. I've been using Sangoma cards until now. Are they supported under the ABE support package? |
12:16.22 | XIN01OZ | as well through IAX |
12:16.38 | JT | mickey9: doubt it |
12:16.42 | JT | at least not officially |
12:17.00 | InHisName | You get cause code 16 with more than one call destination ? |
12:17.14 | InHisName | That is only 2 sec long ? |
12:18.52 | XIN01OZ | actually i now see it maybe a spawn extension error : http://pastebin.com/885158 |
12:18.54 | mickey9 | JT: What would you suggest for a digium channelized E1 with Hardware echo canc.? |
12:19.10 | JT | mickey9: have you had problems with sangoma cards? |
12:20.43 | XIN01OZ | JT: Could you please look at that pastebin and see if it says anything otherwise to u |
12:21.04 | JT | it says you're using freepbx :/ |
12:22.17 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
12:22.23 | XIN01OZ | alright most likely a freepbx issue then- im looking for more debug info |
12:22.42 | JT | i can't determine what it's done |
12:23.01 | XIN01OZ | verbose at 69 did not reveal anything more |
12:23.11 | XIN01OZ | appreciate it |
12:23.19 | JT | lol i think the max usefulness is around verbose 6 |
12:23.22 | *** join/#asterisk apardo (n=apardo@87.217.144.130) |
12:23.22 | JT | i usually use 10 |
12:24.37 | InHisName | anyone install app_backtricks.c in their asterisk ? I have a need to read in file contents into global var |
12:27.38 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
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12:43.29 | tzafrir | InHisName, there's the function SHELL in recent versions |
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13:08.51 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
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13:14.59 | *** join/#asterisk headstone (n=slice@84.90.230.119) |
13:15.32 | kippi | how comes my phone keeps on ringing and not going to voicemail? http://www.pastebin.ca/364858 |
13:16.30 | headstone | hello all. Does anyone have a B410p up & running? I'm getting insane with this card |
13:18.59 | headstone | can anyone help me ? I'v the lights blinking but ztcfg shows no channels |
13:23.53 | Nobbie | headstone: what card ? what lights/colours |
13:24.30 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
13:25.07 | Nobbie | kippi: can you paste your log file for the relevant call ? |
13:25.23 | puzzled | hi |
13:29.01 | kippi | Nobbie: http://www.pastebin.ca/364879 |
13:31.42 | Nobbie | kippi: does agent 1153 ring for 12 seconds ? |
13:31.56 | kippi | just keeps ringing |
13:31.59 | Nobbie | since that's your timeout value |
13:32.09 | Nobbie | it should say something like: nobody picked up in 12000ms |
13:32.39 | kippi | what would be stoping it from doing that |
13:32.58 | Nobbie | are you sure you're waiting for 12 seconds ? |
13:33.22 | Nobbie | _at least_ 12 seconds. |
13:33.25 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
13:34.45 | kippi | Nobbie: waited until the phone stoped rining but it did not go to voicemail, just had music on hold |
13:35.01 | headstone | Nobbie: Digium B410P 4BRI |
13:35.08 | Nobbie | and how many seconds was that ? |
13:35.31 | headstone | Nobbie: it has 4 lights on the back (one 4 each BRI channel) |
13:35.42 | Nobbie | headstone: do you have bristuff installed ? |
13:36.58 | Dr-Linux | any voicemail expert active |
13:37.11 | kippi | 44 |
13:37.12 | headstone | Nobbie: no. On all the forums I'v been (including Digium's intall intructuions for thar card, it was never mentioned |
13:37.58 | headstone | Nobbie: for all I'v read it seams to use di mISDN drivers |
13:38.13 | Dr-Linux | when i get voicemail, in subject voicemail id and in body id always differet |
13:38.21 | headstone | Nobbie: gonna try it anyway |
13:38.54 | Dr-Linux | like if message in subject is 0005 it will be voicemail 6 |
13:39.03 | Dr-Linux | always one difference |
13:39.10 | Dr-Linux | any body understand my problem? |
13:39.39 | headstone | Nobbie: question: arn't does drivers for the junghanns interfaces? |
13:39.58 | headstone | or they work as well with Digium's? |
13:45.46 | kippi | Nobbie: anyideas? |
13:48.21 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
13:48.37 | joelsolanki | Hello All. |
13:48.46 | joelsolanki | Have question regarding cdr. |
13:49.08 | joelsolanki | I have allowed my client to make calls on basis of IP. host = IP |
13:49.17 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
13:50.28 | Nobbie | headstone: i think they're for most BRI cards. worked with my Duxbury too |
13:51.02 | Nobbie | headstone: mISDN may be the way to go *shrug* |
13:51.15 | Nobbie | kippi: try removing the 'n' option |
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14:00.34 | kippi | ok |
14:03.14 | tzanger | morning |
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14:08.47 | kippi | Nobbie: done that, still nothing differnet |
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14:18.01 | *** mode/#asterisk [+o mog] by ChanServ |
14:18.46 | *** join/#asterisk Ahrimanes (n=ma@195.74.76.12) |
14:20.10 | Ahrimanes | somewhere on the great internet, i found someone who made a mysql connection pooler for asterisk, anyone here seen that? |
14:20.58 | e-ddie | there is a great internet? |
14:21.10 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
14:21.12 | ManxPower | e-ddie: Not that I'm aware of. |
14:21.22 | Ahrimanes | yeah, as opposed to the sucky one |
14:21.24 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
14:22.06 | e-ddie | doh |
14:22.19 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
14:22.29 | Ahrimanes | damnit.. need that pooler, but cant remember which software package had it |
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14:37.05 | MarkWD | Morning, we are moving our server from our RD area to our test area and we will be behind a real fire wall what are the range of ports that need to be opened in order to use sip? or is there another way... |
14:37.39 | MarkWD | should we be look at iax |
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14:39.30 | [TK]D-Fender | MarkWD: 5060-5070,10000-20000 all UDP |
14:39.43 | MarkWD | Thanks |
14:39.58 | [TK]D-Fender | MarkWD: Unless you have another server on the outside or an ITSP that uses IAX2, I wouldn't bother |
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14:52.16 | *** mode/#asterisk [+o anthm] by ChanServ |
14:54.35 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
14:54.38 | iq | Hi |
15:02.44 | *** part/#asterisk JoNate (n=noone@mail.wmelec.com) |
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15:19.18 | atn | hi everyone, im trying to see how many users a daemon can handle and how many CPU/MEM a daemon need, any help on that? maybe a redirection to a document? |
15:19.56 | ManxPower | atn: there is no way to tell because there are too many variables involved. |
15:20.19 | ManxPower | Asterisk does not use a large amount of memory, so your main concern would be CPU usage. |
15:20.22 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
15:20.47 | atn | even |
15:21.12 | ManxPower | For example, SIP to SIP on the same LAN with reinvites uses virtually no CPU. If 1 leg of a call is G729 and 1 leg of the call is iLIBC, then the call will use significant amount of CPU. |
15:21.41 | ManxPower | If you are going SIP with ulaw codec to/from PSTN line connected to the Asteirsk server in the USA then CPU usage will be very low. |
15:22.24 | atn | what about iax? |
15:22.37 | ManxPower | The protocol has almost no impact on CPU usage. |
15:22.55 | atn | so a celeron 800Mhz can handle 100 users ? :P |
15:22.58 | ManxPower | Compression (i.e. codec conversion) is what takes the CPU. |
15:23.07 | ManxPower | atn: is that 100 users or 100 calls at the same time? |
15:23.22 | atn | 1 call per user |
15:23.24 | atn | 100 calls |
15:23.54 | ManxPower | A celeron 800Mhz MIGHT be able to handle 100 calls if they are all using ulaw, reinvites are enabled, and there is no communication to the outside world, including PSTN |
15:24.03 | wunderkin | Wow. I had a package sent FedEx ground from NY to PHX. Expected delivery 5 days. Ends up being 3. Wow... They aren't screwing around anymore (hence their new ground commercials, I guess) |
15:24.16 | ManxPower | oh and if there are not many call setup/teardowns per min |
15:25.00 | ManxPower | If you need 100 calls, stop being a cheap ass and buy a decent server. |
15:25.28 | wunderkin | heh |
15:25.41 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
15:26.36 | ManxPower | You are basically asking "Can I use my Ford Escort in a race and expect to win?" The answer is "Yes, if all the other cars in the race are broken down." |
15:27.08 | ManxPower | i.e. The only situation where the answer to your question is "yes" is so unlikely to happen the answer might as well be "no" |
15:27.26 | mercestes | what if the escort has NOs? |
15:27.33 | *** join/#asterisk shinux_ (n=shinux@196.220.30.98) |
15:27.56 | ManxPower | mercestes: There's yet another variable 8-) |
15:27.58 | file | translation: "what if my system is overclocked?" |
15:27.59 | mercestes | and rocket engines |
15:28.03 | [TK]D-Fender | mercestes: The when it EXPLODES you have a chance of winning ig the judge qualifies little BITS of you arriving at the finish line first... |
15:28.30 | file | I remember overclocking a Celeron 300MHz CPU... I jumpered the wrong pins and melted the jumper |
15:28.39 | mercestes | lol@file |
15:28.43 | [TK]D-Fender | file: I remember OC'ing my 300 to 450 :) |
15:29.03 | [TK]D-Fender | file: Best bank/buck OC ever |
15:29.03 | mercestes | I remember overclocking a 1.3ghz. I had a process "freeze" and I set it to realtime and melted my proc. |
15:29.19 | file | [TK]D-Fender: quite |
15:29.46 | [TK]D-Fender | file: Celery's were stable then.... |
15:30.02 | [TK]D-Fender | file: Now I seriously can't be bothered with OCing anything... |
15:30.54 | file | 'tis already fast enough |
15:30.57 | atn | ok thanks |
15:34.09 | *** join/#asterisk Juppers (i=imagine@phantom.blueone.net) |
15:34.43 | Juppers | anyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find. |
15:40.22 | *** join/#asterisk Sputtering (n=Keelan@192.197.213.245) |
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15:51.04 | kippi | can anyone see why my calls are not going to voicemail? http://www.pastebin.ca/364999 |
15:52.09 | danp | Set(CALLERID(name))=Hall Bookings) |
15:52.18 | danp | i think you have an extra ) there |
15:52.54 | danp | but that might not be what's keeping it from going to voicemail |
15:53.46 | kippi | what could be doing this? |
15:53.59 | Sputtering | try fixing the bracket and testing again |
15:54.06 | kippi | have |
15:54.20 | kippi | so its like Set(CALLERID(name)=Hall Bookings) |
15:54.26 | mercestes | kippi: Do you have a timeout= in queues.conf?? |
15:54.41 | *** join/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de) |
15:54.43 | unice | hi |
15:54.56 | mercestes | kippi: Dump a timeout=6 in queues.conf under [hall] and do an reload app_queue.so |
15:55.07 | mercestes | hello, unice. |
15:55.10 | Sputtering | kippi: with no-o ne signed into the queue, what happens? |
15:55.24 | n|cotine | I setup DUNDi according to the documentation, and everything works fine - except if I call an extensions that does not exist, but another PBX is advertising. Then the channel stays in State Busy with application data Congestion(), and the channel never drops. Any idea what's wrong here? |
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15:57.07 | unice | i've upgrade from 1.2.10 yesterday.. asterisk 1.2.14 seems to ignore the astrundir statement in asterisk.conf - i'm unable to connect the socket using asterisk -r (Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)) |
15:57.24 | kippi | it works with the timeout in the queue |
15:57.45 | unice | astrundir => /var/run/asterisk - file is there and has the permissions set accordingly |
15:58.04 | unice | did i miss a chage? |
15:58.09 | unice | change, even |
15:58.24 | ManxPower | unice: only if you didn't read the Changelog |
15:58.44 | unice | well i grep-ed it for astrundir, socket and .ctl :) |
15:59.23 | mercestes | unice: That error message is very uninformative. It doesn't really mean that asterisk.ctl is there, or not there, or that asterisk is or is not running. |
15:59.33 | mercestes | unice: what that error message really means is "oh crap, something went wrong. sorry." |
15:59.41 | unice | yeah i know |
15:59.47 | ManxPower | I assume you can start asterisk as "asterisk -cvvv" |
16:00.06 | unice | asterisk.conf gets parsed fine, as the output of asterisk -cvvv shows |
16:00.17 | mercestes | unice: Try doing a hard shutdown on asterisk (killall -9 asterisk anyone?) and removing the .ctl file from /var/run/asterisk and do the asterisk -cvvv and check for real error messages. |
16:00.36 | *** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose) |
16:00.44 | mercestes | unice: Are you doing this via ssh? |
16:01.12 | unice | i've checked for errors, none obvious |
16:01.17 | unice | yes, ssh |
16:01.38 | mercestes | unice: do your asterisk -cvvvvv in one window, and with asterisk still up, try your asterisk -r |
16:02.10 | unice | ok haven't tried that |
16:02.50 | unice | but i have to wait... people are using the server now :) |
16:03.50 | unice | i forgot to mention: asterisk tries to write the pidfile to /var/run instead of /var/run/asterisk as configured by astrundir statement |
16:03.56 | *** join/#asterisk Johnnie (n=jdlewis@jdlewis.org) |
16:04.18 | unice | well i'll check the -cvvvv output.. thanks so far! |
16:05.08 | mercestes | unice: Check your config file permissions and make sure * can read the config file |
16:05.22 | mercestes | unice: And make sure /var/run/asterisk/ exists and has the proper permissions. |
16:05.55 | unice | the config files are owned by the asterisk user... as well as /var/run/asterisk |
16:06.12 | unice | according to -cvvvv output the files are read fine |
16:07.02 | ManxPower | unice: time to report it on bugs.digium.com |
16:07.39 | unice | yeah i'll do it after i've checked the output again tonight |
16:07.50 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
16:07.50 | *** mode/#asterisk [+o russellb] by ChanServ |
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16:12.19 | ManxPower | unice: I left the paths the default and have no problems |
16:14.45 | atn | where can i find asterisk specifications? |
16:14.51 | Qwell[] | atn: for what? |
16:15.12 | atn | installation specifications |
16:15.29 | mmlj4 | please explain |
16:15.35 | atn | what it need to be installed |
16:15.41 | atn | what software i need |
16:15.49 | mmlj4 | you mean system requirements? |
16:15.55 | mercestes | atn: Depends on what your trying to do. |
16:16.26 | atn | no i mean software requirements |
16:16.42 | mercestes | atn: an operating system. |
16:17.01 | Qwell[] | a unix/unix-like operating system |
16:17.04 | mercestes | atn: and a text editor. |
16:17.17 | mercestes | or a windows one if you *really* must. |
16:17.18 | atn | yes on linux |
16:17.20 | atn | what else |
16:17.23 | mmlj4 | you need gcc and make |
16:17.39 | Qwell[] | and a shell |
16:17.53 | mercestes | atn: And hot babe to monitor your CPU usage |
16:17.58 | file | and power |
16:19.26 | mmlj4 | hey ManxPower: you know I'm Joey, right? |
16:19.52 | [TK]D-Fender | atn : www.asterisk.org |
16:19.54 | [TK]D-Fender | ~docs |
16:19.55 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
16:19.57 | [TK]D-Fender | ~book |
16:19.58 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
16:21.05 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
16:22.09 | DrukenLPY | morning everyone |
16:22.29 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
16:22.35 | mercestes | morning |
16:23.13 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
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16:33.38 | [TK]D-Fender | Wow... I scared him right off :) |
16:34.31 | mercestes | Good job. |
16:35.20 | mercestes | bkw_: Aww....just find out she's a fraud? |
16:37.42 | [TK]D-Fender | mercestes: We can't all be psychic like you! ;) |
16:38.02 | [TK]D-Fender | .... er .... make that PSYCHOTIC ;) |
16:38.07 | mercestes | either or. |
16:38.23 | mercestes | But I do offer dirt cheap communications through Telepathy telco. |
16:38.57 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
16:41.27 | [TK]D-Fender | mercestes: I'll also take for granted that the echo is entirely natural :O |
16:43.58 | DrukenLPY | the echo makes it "magical".... |
16:44.58 | *** part/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de) |
16:45.17 | mercestes | when is the next astricon thingy? |
16:45.39 | mercestes | Yea, that's just my multiple personalities trying to speak over each other. |
16:45.49 | file | mercestes: September I do believe |
16:46.32 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
16:49.39 | mercestes | Sweet. |
16:49.58 | *** join/#asterisk VampBoi (n=VampBoi@68-187-206-043.dhcp.ahvl.nc.charter.com) |
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16:50.41 | mercestes | :D |
16:50.44 | VampBoi | :P |
16:50.56 | kippi | why would asterisk be striping off the 0 when showing the callerid when coming in? |
16:51.30 | mercestes | kippi: I don't think * is doing that, but a 0 at the beginning of a valid caller ID makes no sense. Exactly where/how are you seeing a striped 0? |
16:51.57 | mercestes | kippi: And what's wrong with stripping? |
16:51.58 | mercestes | kippi: and look for "strip most significant digit" rules. |
16:52.00 | *** join/#asterisk unice (n=tom_hens@port-83-236-223-18.static.qsc.de) |
16:52.20 | kippi | will that be on the zaptel.conf? or extenstion? |
16:52.43 | mercestes | VampBoi: That's scary. |
16:52.57 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
16:53.10 | mercestes | kippi: If your worried about zaptel.conf then ask you rtelco why they are stripping off the 0 in your spoofed and invalid callier id. |
16:53.51 | mercestes | VampBoi: Aside from loosing your CDR's and your voicemails everytime you reboot your system.....it *should* be ok. You'd have to somehow update your changes onto a slax cd or you would loose your * install and config changes every reboot oo |
16:54.01 | mercestes | VampBoi: Unless you have that flash memory crap |
16:54.26 | VampBoi | well I'm going to install Slax onto the box directally I'm going to do the server version |
16:54.35 | mercestes | VampBoi: Weird memory rules there too to avoid CDR thrashing. I wouldn't plan on running anything real intensive because your RAM is going to be dedicated to yoru kernel. |
16:54.57 | mercestes | VampBoi: oh. in that case, should be standard linux stuff |
16:55.09 | VampBoi | lol thought so just wanted to make sure :P |
16:55.13 | mercestes | VampBoi: I suggest gentoo tho |
16:55.34 | [TK]D-Fender | VampBoi: Just install Slackware and be done with it. |
16:55.40 | VampBoi | my uncle has * running on one of his servers at his place and I got addicted to the music on hold and stuff :P |
16:56.25 | cpm | multigenerational asstriks hacking, this can't really be a good sign |
16:56.36 | *** join/#asterisk bmg505 (n=leon@c1-181-4.rndf.isadsl.co.za) |
16:56.48 | kippi | or is there away I can add them? |
16:58.41 | mercestes | kippi: this is so ppl can just hit 'dial' on "missed calls" because you require a 0 for outbound calls, huh? |
17:00.25 | VampBoi | mercestes may I pvt u? |
17:00.53 | mercestes | VampBoi: Sounds lurid. Sure. |
17:01.00 | VampBoi | lol |
17:04.32 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
17:08.37 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
17:17.15 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
17:17.36 | ryant | anyone know of any nice high res wallpapers/images of asterisk logos, etc? |
17:17.57 | mercestes | do a google image search on asterisk with safe search off |
17:18.01 | mercestes | what's the worst you could get? |
17:18.08 | ryant | that didn't being back much of value |
17:18.13 | wunderkin | [TK]D-Fender: woot i finally got my ip501! hopefully the customer gets theirs today too |
17:18.29 | ryant | how about someone internally at Digium? You guys have to have high res logos somewhere!!! :) |
17:18.33 | mercestes | try digium |
17:18.39 | [TK]D-Fender | wunderkin: Good to hear. Great phone.... |
17:18.44 | Juppers | anyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find. |
17:18.48 | mercestes | or gratis6 |
17:18.55 | mercestes | oh, oops. That has nothing to do with asterisk. |
17:19.33 | kippi | what is the best way to be able to set night serice/ |
17:19.46 | kippi | want them to be able to dial a * code |
17:19.59 | mercestes | kippi: google asterisk hire a consultant |
17:20.15 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
17:20.30 | kippi | is it that hard/ |
17:20.30 | mercestes | kippi: or read the book. dial a * code is just an exten => *code,1,Do(something) |
17:20.58 | kippi | I should be able to use a string? |
17:21.03 | mercestes | kippi: night service is just a GotoIfTime,*********,1,Hangup() |
17:21.15 | mercestes | kippi: Sure! If your device can dial *code then yes. |
17:22.07 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
17:23.50 | mercestes | kippi: Otherwise I suggest something more numeric like *1234 or *812 or *6969696969 or something. |
17:25.17 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
17:25.29 | kippi | just need to work out how to get the things in a string |
17:25.56 | mercestes | kippi: I use a tin can and a button |
17:29.50 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
17:38.39 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
17:47.15 | headstone | anyone with Digium B410P? |
17:47.32 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
17:48.30 | puzzled | headstone: no but what's the problem |
17:48.36 | florz | What mechanisms are there to let a SIP -> PSTN gate know that one wishes to block caller ID presentation? |
17:49.27 | puzzled | florz: think there is a command built in. check show applications |
17:49.40 | bkruse | florz: you can match that in the dialplan of asterisk |
17:49.44 | bkruse | headstone: i have a few :] |
17:50.11 | bkruse | headstone: if it has to do with installation and configuring, i can help.....anything beyond that (errors) i have no idea |
17:50.30 | florz | puzzled: If you mean SetCallerPres (== Setting From: to "Unknown"), that doesn't work ... |
17:50.43 | bkruse | florz: what do you want to do now? |
17:50.46 | florz | bkruse: Hmm? What do you mean? |
17:50.48 | bkruse | block people with certain caller ID's? |
17:51.03 | puzzled | florz: yeah there was something with that. have you tried the 0x... values? |
17:51.30 | florz | bkruse: I want to tell a SIP -> PSTN gate that it should block CID presentation to the callee |
17:51.34 | *** join/#asterisk So3kris (n=jan-will@217.170.33.70) |
17:52.12 | florz | puzzled: Erm, no!? It does set the From header, though, the PSTN gate just doesn't care ... |
17:53.09 | bkruse | florz: easy gotoif(${CALLERID(num)} == "2564233142"? badpersoncontext : goodpersoncontext |
17:53.19 | bkruse | whoops missed my ) at the end, my apologies |
17:53.32 | puzzled | bkruse: he means outgoing, not incoming |
17:54.04 | bkruse | oh, i think the same principal can be applied |
17:54.06 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
17:55.30 | bkruse | just look at the dialplan variables, and you can use gotoif's |
17:55.48 | puzzled | florz: do you have "secallingpres=yes" in zapata.conf (found that on voip-info.org) |
17:56.14 | florz | puzzled: ?! - how is that supposed to help with SIP?! |
17:56.28 | puzzled | florz: don't ask me about the logic the developers used :) |
17:57.15 | florz | puzzled: You got some URL? |
17:57.30 | puzzled | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCallerPres |
17:57.33 | puzzled | under CLIR |
17:57.48 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
17:58.20 | florz | ; for zap channels I needed to put "usecallingpres=yes" in the zapata.conf to get <- !?! |
17:58.33 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
17:58.40 | puzzled | ah sorry, missed the "for zap channels" |
18:02.04 | *** join/#asterisk paoleela (n=paolope@217.7.206.10) |
18:02.25 | *** join/#asterisk notoriousrab (n=robert_m@207.47.34.74.static.nextweb.net) |
18:06.04 | sasch | i recompile my asterisk |
18:06.16 | sasch | and when i lunch asterisk now return Segmentation fault |
18:06.24 | sasch | why ?? |
18:06.46 | mafkees | did you do: make install while asterisk was still running ? |
18:07.17 | sasch | yes i run make clean && make && make install |
18:07.27 | *** join/#asterisk psyferre_ (n=psyferre@host-prestigemag-105-10.customer.ntelos.net) |
18:07.35 | mafkees | then asterisk will die |
18:07.39 | mafkees | can you start it now ? |
18:07.40 | *** join/#asterisk NL5124 (n=BlubBlub@port-87-234-153-49.dynamic.qsc.de) |
18:07.45 | bkruse | sasch: type this: rm /usr/lib/asterisk/modules/* |
18:07.52 | bkruse | i totally tried to tab complete that. |
18:07.57 | sasch | ok |
18:07.58 | sasch | i try |
18:08.00 | bkruse | then make install |
18:08.02 | sasch | one moment |
18:08.16 | bkruse | you dont have to make clean again, or make even |
18:08.46 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
18:08.46 | sasch | now i run root@centres:/usr/src/asterisk# rm /usr/lib/asterisk/modules/* |
18:08.47 | sasch | rm: cannot remove `/usr/lib/asterisk/modules/*': No such file or directory |
18:08.50 | sasch | ops |
18:09.02 | sasch | i post a lines ....... ops :-P |
18:09.18 | bkruse | interesting..... |
18:09.21 | psyferre_ | hey folks, i hope everyone is well today :) Would anyone have a moment to help an asterisk n00b? I registered some new codecs and all seemed to go well, but now asterisk will not start - "Illegal Instruction" right after finding the codec... |
18:09.24 | bkruse | ok, make install and try to run asterisk |
18:09.37 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
18:09.50 | bkruse | psyferre_: is it for the right architecture? and is it for the right ast version? |
18:10.03 | bkruse | sasch: try to make install |
18:10.08 | bkruse | and tell me the output, if it errors |
18:10.10 | psyferre_ | bkruse: i think so, definitely the right asterisk version |
18:10.14 | bkruse | then ls /usr/lib/asterisk/modules |
18:10.20 | bkruse | psyferre_: tell me more about the codec |
18:10.22 | bkruse | g729? |
18:10.25 | psyferre_ | yup |
18:10.36 | bkruse | what machine, and what g729 codec did you get |
18:10.40 | bkruse | and what ast version are you running |
18:10.53 | psyferre_ | one moment while I get all that |
18:10.59 | bkruse | i just downloaded the g729 codec 2 days ago, and used it |
18:10.59 | bkruse | k |
18:11.01 | bkruse | thanks |
18:12.21 | sasch | ok now run all |
18:12.24 | sasch | thanks |
18:12.35 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
18:13.06 | psyferre_ | running on an older dell poweredge 500sc with a celeron processor, codec g729a (32-bit), 32 bit register utility, and asterisk 1.2.7.1 |
18:14.07 | paoleela | Hello. Dial tones from mobile phone sometimes aren't recognized. message log looks like this: |
18:14.09 | paoleela | Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? |
18:14.13 | docelmo | Quick question for someone when asterisk says it wants -lssl it wants the libs for openssl right? |
18:14.56 | tzafrir | right |
18:15.00 | *** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com) |
18:15.02 | tzafrir | -lssl -lcrypto |
18:15.02 | paoleela | It's because I set up a callback and have to dial the number after being called by asterisk. |
18:15.17 | psyferre_ | when I try to run asterisk -rvvv I get an error that it can't connect and I should check to make sure /var/run/asterisk/asterisk.ctl exists. It does. when I run asterisk -vvv it gives illegal instruction right after "Found total of 5 g.729 licenses" |
18:15.39 | EmleyMoor | I seem to have a high echo on calls made on IP phones via my FXO port - what should I check to fix this? |
18:15.41 | psyferre_ | sorry, all that to bkruse |
18:15.45 | docelmo | What lib is crypto |
18:15.51 | EmleyMoor | Is there an echo test on a UK PSTN number? |
18:16.04 | tzafrir | psyferre_, what is your cpu? |
18:16.41 | psyferre_ | i know it's a celeron, let me see if i can find out what speed |
18:16.52 | tzafrir | cat /proc/cpuinfo |
18:17.33 | psyferre_ | tzafrir: thanks, that's a fantastic command to know - 1097.113 |
18:17.38 | mafkees | cat: /proc/cpuinfo: No such file or directory |
18:18.22 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
18:18.33 | [TK]D-Fender | EmleyMoor: What card? |
18:19.07 | EmleyMoor | TDM400P |
18:19.13 | ChicagoBud | good morning. can I have an extension like 7750 and 77XX in my dial plan, where a number like 7724 would match 77XX? Dialing 7750 is fine but as soon as I hit 772 I get an invalid extension. |
18:19.31 | ChicagoBud | what am I missing |
18:19.35 | [TK]D-Fender | paoleela: You didn't set the appropriate DTMFMODE for your channel. |
18:19.56 | [TK]D-Fender | EmleyMoor: Got "echocancel=yes"? how about "echotraining=800"? |
18:20.09 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
18:20.32 | [TK]D-Fender | ChicagoBud: Yes you can, and you have probably made a minor & silly goof |
18:20.54 | ChicagoBud | [TK]D-Fender, yeah I knew that |
18:20.55 | paoleela | [TK]D-Fender: What should be the right mode for dial tones? I mean, sometimes it works, sometimes not. |
18:21.09 | EmleyMoor | Got echotraining=400 - you recommend I try 800? |
18:21.18 | docelmo | thanks tzafrir I got it. damn ldconfig |
18:21.19 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
18:22.01 | ChicagoBud | I see 77XX added as an exten to my context when I reload |
18:22.22 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
18:23.07 | psyferre_ | bkruse, tzafrir: any idea? |
18:23.13 | paoleela | [TK]D-Fender: It's set to dtmfmode=rfc2833 |
18:23.18 | tzafrir | Echotraining is not much of a difference |
18:23.44 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
18:23.45 | tzafrir | psyferre_, file /usr/lib/asterisk/modules/* |
18:23.54 | paoleela | [TK]D-Fender: But I also tried the other modes. |
18:24.03 | tzafrir | hmm... ignore that |
18:24.22 | psyferre_ | okay |
18:24.26 | tzafrir | psyferre_, chances are you have a faulty module there |
18:24.38 | tzafrir | unless you get segfaults with other operations |
18:25.07 | psyferre_ | i haven't seen any other problems, though that's the only thing we're using this box for... |
18:26.11 | *** join/#asterisk gatuno (n=gatuno@145.red-82-158-215.user.auna.net) |
18:26.42 | psyferre_ | how can i check to see if a faulty module is the problem? |
18:26.50 | EmleyMoor | Can time ranges on GotoIfTime span overnight? |
18:26.52 | psyferre_ | redownload the g279a.so file? |
18:27.56 | wunderkin | is echotraining used on a pri? |
18:28.26 | [TK]D-Fender | EmleyMoor: I believe so, but if won't cross into the next DAY. |
18:28.44 | psyferre_ | tzafrir: I'm guessing rm /.../asterisk/modules/codecg729a.so and then do the register process over again |
18:28.53 | [TK]D-Fender | EmleyMoor: what version of Zaptel are you using? |
18:29.07 | [TK]D-Fender | EmleyMoor: You may want to consider upgrading. |
18:29.18 | EmleyMoor | [TK]D-Fender: Your answer appears to be "yes and no" - so I do need to stick with overnight in two parts |
18:29.25 | [TK]D-Fender | ChicagoBud: paste your dialplan line that isn't working. |
18:29.57 | EmleyMoor | [TK]D-Fender: 1.2.11 |
18:30.09 | [TK]D-Fender | EmleyMoor: Try an upgrade |
18:30.25 | ChicagoBud | [TK]D-Fender, just got it. _77XX not 77XX |
18:30.38 | EmleyMoor | [TK]D-Fender: That isn't easy |
18:31.03 | bkruse | [TK]D-Fender > all |
18:31.05 | bkruse | jbot: [TK]D-Fender++ |
18:31.52 | ChicagoBud | extension.conf syntax is awful |
18:32.08 | EmleyMoor | ChicagoBud: It's not bad when you get used to it |
18:32.57 | ChicagoBud | EmleyMoor, just seems it could have been better - more like a language we've seen before |
18:33.37 | EmleyMoor | I've managed to do away with numbered priorities, except of course 1. |
18:33.45 | psyferre_ | just to be sure i'm on the right track... celeron 1Ghz would be i386 architecture for the G729 codec download? |
18:34.43 | ChicagoBud | oh well, it is what is it. I think the openpbx guys are doing something different but it remains to be seen if they become mainstream |
18:36.02 | EmleyMoor | I'm just trying to see if I can implement anything in macros - but my peculiar incoming number requirements make it difficult |
18:37.27 | ChicagoBud | EmleyMoor, I try to use macros too |
18:38.32 | EmleyMoor | I managed to get my voicemail checking line down to just 10 priorities last night |
18:42.03 | *** join/#asterisk CJLinst (n=CJLinst@209-221-212-010.qnet.com) |
18:42.32 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
18:42.52 | CJLinst | Is there a specific trick to getting multiple sip registrations working from an snom to a single asterisk? |
18:43.30 | EmleyMoor | CJLinst: Each registration probably needs to be a different peer |
18:43.38 | funxion | Can anyone tell me whether its possible to get wmi working using realtime I have read articles stating yes and no, |
18:43.55 | funxion | I have rtc enabled in sip.conf |
18:43.59 | funxion | it still doesnt werk |
18:43.59 | CJLinst | That's how I have it. But only one of the sip peers can make calls, authentication fails on the other one. |
18:44.11 | psyferre_ | tzafrir, bkruse: it was a corrupted or incorrect module, after deleting it and redownloading the i686 version instead of using the generic "32-bit" from the readme file it all worked like a charm. Thanks for your help! |
18:44.20 | CJLinst | chan_sip.c:8065 check_auth: username mismatch, have <246>, digest has <226> |
18:44.23 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
18:44.58 | bkruse | psyferre_: sorry, i shoulda caught you and said use i686 |
18:45.07 | bkruse | its pretty generic in terms of which one to use :] |
18:45.09 | bkruse | gl! |
18:45.22 | CJLinst | Calls from other extensions go to the correct Identity button |
18:45.33 | psyferre_ | bkruse: no biggie :) I needed to look it up anyways to get the full deal :) Thanks again! |
18:45.38 | psyferre_ | Cheers! |
18:45.59 | sasch | to activate DND in my grandtsream ?? |
18:46.19 | [TK]D-Fender | EmleyMoor: How is it not easy? |
18:46.24 | [TK]D-Fender | bkruse: y0 |
18:46.37 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
18:46.40 | bkruse | [TK]D-Fender: wut up :] |
18:47.05 | EmleyMoor | [TK]D-Fender: I lose package management if I go to a later version |
18:47.22 | [TK]D-Fender | EmleyMoor: You say that.... as though it were a BAD thing ;) |
18:47.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net) |
18:48.08 | [TK]D-Fender | bkruse: Work, the usual.... Brought all my blades in to work as I'm going stright to martial arts after. Had some fun showing my katana to our engineers :) |
18:49.34 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-145-140.ny325.east.verizon.net) |
18:50.16 | funxion | can someone help me with WMI and realtime? |
18:51.21 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
18:52.42 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
18:52.58 | [Airwolf] | funxion, what seems to be the problem ? |
18:53.20 | funxion | I setup realtime sip exten and voicemail all seems to be working fine with teh exception of wmi |
18:53.44 | funxion | I have the rtccacchefriends or whatever set to yes in sip.conf |
18:54.03 | [Airwolf] | Can you pastebin your sip config and an sql query of one of your users ? |
18:54.05 | funxion | it will do everything but sen notification |
18:54.32 | funxion | one sec |
18:54.43 | psyferre_ | just another quick question folks :) After installing the G729 codec is there anything else to do? If I purchased a 5 channel license, will it just support that codec on 5 simultaneous connection automatically, or do i ahve to pick which 5 extensions and tell them to use that codec manually? |
18:55.02 | Qwell[] | psyferre_: it "just works" |
18:55.08 | [Airwolf] | psyferre_, it wil do that automaticly |
18:55.08 | mafkees | Qwell[]: ! |
18:55.13 | Qwell[] | mafkees: I didn't do it |
18:55.14 | mafkees | congrats with commit 55555 |
18:55.16 | psyferre_ | lol |
18:55.17 | Qwell[] | thanks :D |
18:55.25 | funxion | [Airwolf] is there a database field in realtime that turns mwi on and off? |
18:55.36 | Qwell[] | mafkees: I was TRYING to find a typo to fix, but I found a real bug instead. :P |
18:55.42 | psyferre_ | thanks, i really appreciate your help and patience. have a good one all :) |
18:55.47 | mafkees | lol Qwell[] |
18:55.53 | mafkees | sometimes luck is just with you |
18:55.55 | psyferre_ | *buys you all a beer* |
18:56.00 | [Airwolf] | funxion, not really. But you have to make sure everything is filled in correctly |
18:56.16 | *** part/#asterisk psyferre_ (n=psyferre@host-prestigemag-105-10.customer.ntelos.net) |
18:56.16 | [Airwolf] | Such as your mailbox number and voicemail context. |
18:56.23 | funxion | those are |
18:56.33 | funxion | I can call and leave a message |
18:56.36 | [Airwolf] | euh one moment |
18:56.39 | funxion | and can check messages |
18:56.50 | funxion | just no message waiting indicator |
18:57.53 | [Airwolf] | the mailbox field should be <number>@<context> (the last one if it's not default) |
18:58.24 | funxion | yes |
18:58.27 | funxion | it is |
18:59.10 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
18:59.13 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
18:59.28 | funxion | hmm |
18:59.32 | funxion | @context? |
18:59.37 | funxion | i have @voicemail |
18:59.44 | [Airwolf] | that is incorrect. |
18:59.55 | funxion | now it werx |
18:59.57 | funxion | thnx |
19:00.00 | [Airwolf] | np :) |
19:00.16 | funxion | btw do you have any experience with voicemail odbc storage? |
19:03.42 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
19:04.07 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-145-140.ny325.east.verizon.net) |
19:06.46 | EmleyMoor | What happens when a macro exits? |
19:07.38 | mmlj4 | isn't a macro like a function? |
19:08.05 | [TK]D-Fender | EmleyMoor: It returns to the point from which it is called and resumes. |
19:08.15 | froguz | it's like a C function |
19:08.21 | froguz | a sub-rutine |
19:09.38 | *** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
19:10.32 | *** join/#asterisk mega (n=mega@217.201.163.156) |
19:11.44 | CJLinst | Can anyone explain exactly what's happening when I get this: chan_sip.c:8065 check_auth: username mismatch, have <246>, digest has <226> |
19:13.01 | [TK]D-Fender | CJLinst: Check your phone. |
19:13.25 | CJLinst | Thanks, but what is happening? |
19:14.04 | mafkees | digest and username does not match |
19:14.21 | CJLinst | And the digest refers to? |
19:16.33 | [TK]D-Fender | CJLinst: Your settings are hard to swallow ;) |
19:17.13 | [TK]D-Fender | CJLinst: Just go look at your phones setup in extreme details and you're likely to see something thats "off" |
19:17.21 | *** join/#asterisk giasai68 (n=administ@ip-3-145.sn2.eutelia.it) |
19:17.29 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
19:17.54 | CJLinst | What I'm trying to do is get two Line buttons on an Snom 300 working as two different SIP peers. It's working great for calls to the phone, but outbound calls into the context are only working for onw of them. The other gets the authentication failure. |
19:21.53 | *** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net) |
19:22.43 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
19:23.22 | GaVak | I'm having problems with a PRI turn up... the provider says that they see the d-channel come up... then drop after I reload *. |
19:23.27 | GaVak | I'm getting this error: 3 |
19:23.27 | GaVak | Write to 44 failed: Unknown error 500 |
19:23.27 | GaVak | Short write: 0/5 (Unknown error 500) |
19:23.55 | GaVak | is there a command that will show the status of the PRI link? |
19:24.06 | GaVak | zap show status only says 'OK' |
19:24.27 | wunderkin | pri show span x |
19:26.19 | EmleyMoor | I've now got a macro to do my colourlist checking |
19:27.17 | *** join/#asterisk jm|laptop (n=jm@zen.jamiem.com) |
19:27.33 | GaVak | Meh, it looks like their DSL dropped. Well, thanks for the command wunderkin. |
19:28.30 | [TK]D-Fender | DSL?! |
19:29.26 | wunderkin | apparantly short loops are frequently done over hdsl |
19:29.30 | tzanger | uh |
19:29.37 | tzanger | all T1s are HDSL2 loops these days |
19:29.47 | tzanger | 5 years ago they were HDSL loops |
19:30.13 | tzanger | the real physical T1 as a long-haul transport doesn't exist anymroe |
19:31.16 | tzanger | it doesn't terminate to a DSLAM like consumer-grade DSL but the technology (coding) is all the same |
19:31.51 | tzanger | they can easily tweak the tx/rx buckets for the DMT transport in DSL to get you more than 800kbps uplink but they won't do it, even as VDSL is rolling out |
19:32.19 | cpm | tzanger, are you sure? |
19:32.28 | tzanger | cpm: about what? |
19:32.50 | cpm | realy physical T1 as a long-haul transport doesn't exist anymore |
19:33.17 | tzanger | cpm: yep, we had a 40km PTP T1 10 years ago that was actually a DS1-over-HDSL |
19:33.35 | tzanger | (two powered loops, but not T1 as in electrial+logical spec) |
19:33.57 | cpm | errr, why did I think that ds1 over hdsl , , ,ah, okay. |
19:33.57 | tzanger | our PRI upstairs is an HDSL2 (1 loop) DS1-over-HDSL2 |
19:34.02 | ManxPower | It's not suprizing. Real T-1 needed a repeater every 1,800 feet, modern "T-1s" do not. |
19:34.18 | tzanger | modern T1s are every 5000ft IIRC |
19:34.28 | tzanger | I think that's the distance, I can't remember now |
19:34.30 | ManxPower | cpm: shine a flashlight thru the smokey window on the telco T-1 box at work. |
19:34.42 | tzanger | ManxPower: don't do that |
19:34.46 | tzanger | you'll startle the hamster |
19:35.06 | cpm | I just use my key, all T1 smartcard cages use the same key I think |
19:35.13 | ManxPower | You'll see Pairgain or Paradine or Westel card with some form of DSL on the label |
19:35.17 | cpm | yup |
19:35.20 | tzanger | yup |
19:35.25 | tzanger | HRU1 in my case :-) |
19:35.32 | ManxPower | Well if you have a key why are you asking these silly questions? 8-) |
19:36.02 | cpm | all this time, and I never made the hdsl/dsl connection at all |
19:36.17 | cpm | always considered dsl a dry line alarm circuit or something |
19:36.33 | tzanger | nope |
19:36.42 | tzanger | DSL runs over powered circuits in most cases |
19:36.47 | tzanger | SDSL is typically dry copper though |
19:36.51 | tzanger | I set up a number of those over the years |
19:37.05 | EmleyMoor | Can one macro call another? If so, what happens to the MACRO_ variables? |
19:37.19 | tzafrir_laptop | hmmm... any idea why suddenly asterisk does not respect "secret" in iax.conf? (asterisk 1.2.13)? Whatever I put in "secret" I get an "empty secret" in 'iaxs show users" with Authen value of 3 |
19:37.19 | cpm | like the onces covering rotary switch and crossbar |
19:37.56 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915087.dip.t-dialin.net) |
19:38.07 | ManxPower | EmleyMoor: 1: Yes. 2: Try it |
19:38.28 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915087.dip.t-dialin.net) |
19:38.42 | tzafrir_laptop | oops, was running it from a wrong set of config |
19:40.14 | tzanger | hmm I could have sworn that ssh IP packets had low-latency bit set |
19:42.45 | Juppers | anyone familiar with using FXS ports on a cisco 2610 with asterisk? How do you register them? The 2610 doesn't support sip authentication that I can find. |
19:43.01 | *** join/#asterisk [Mr_X] (i=1000@88.118.97.205) |
19:43.05 | florz | tzanger: For interactive sessions and openssh this is usually true, yes. |
19:43.08 | mercestes | why would you want to register SIP over an FXS port? |
19:43.17 | tzanger | florz: that's what I'm staring at right now in ethereal |
19:43.22 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
19:43.23 | tzanger | flags of 0x10 |
19:43.27 | mercestes | why would you want to register an FXS port for that matter? |
19:43.31 | ManxPower | Juppers: it does not support any kind of sip authentication, you need to use IP based authen when getting calls into asterisk from a cisco like that |
19:43.55 | tzanger | 0x10 = D bit set, minimize delay... hmm |
19:44.04 | Juppers | my asterisk admin says it has to register so it can be in the same context as every other phone in our system |
19:44.15 | ManxPower | tzanger: I think you want 0xb8 |
19:44.20 | tzanger | florz: ethereal seems to misrepresent the bits |
19:44.33 | ManxPower | Juppers: your admin is an idiot |
19:44.41 | tzanger | it calls the precedence "Differentiated Services" |
19:44.56 | mercestes | Juppers: seconded. Fire his stupid ass and hire a *real* asterisk admin |
19:44.59 | ManxPower | Juppers: you can out it in whatever context you want by using permit/deny |
19:45.01 | Juppers | ManxPower - he is new to asterisk. we are trying to move away from CCM. |
19:45.07 | florz | tzanger: Well, that depends on which of the interpretations you choose ... |
19:45.21 | ManxPower | tzanger: diffserv is a more modern QoS method that builds on ToS |
19:45.33 | tzanger | ManxPower: ahh |
19:45.35 | mercestes | Then he's not an asterisk admin. he's an asterisk acolyte. not even an acolyte really...more an apprentice to an acolyte. |
19:45.42 | tzanger | heh |
19:45.56 | florz | tzanger: DS is probably state of the art. But in the end, what counts is how the routers that are to use the info so interpret it ... |
19:46.03 | ManxPower | 0xB8 SEEMS to be DiffServ "EF" and ToS Low Latency/High Priority/whatever |
19:46.06 | mercestes | juppers: you jsut set the context in zapata.conf for the fxs interface. |
19:46.13 | florz | tzanger: s/so/do/ |
19:46.14 | tzanger | I'm the receptionist's associate for the apprentice for the acolyte, studying under the great guru of all things Asterisk, ManxPower |
19:46.19 | ManxPower | mercestes: the port is on a cisco |
19:46.27 | Juppers | I'm running a 2610 with fxo and fxs so I can bring my home lines and work lines together and have my own dialplan that doesn't include 9 |
19:46.42 | ManxPower | Juppers: you need to be on a static IP |
19:46.51 | Juppers | ManxPower - I am on static |
19:47.27 | ManxPower | Juppers: other than that on the asterisk side [yourhomecisco] permit=yourip/255.255.255.255 deny=0.0.0.0/255.255.255.255 host=yourip the rest of the settings |
19:47.43 | ManxPower | Juppers: this is off the top of my head, check the sample configs that come with Asterisk |
19:48.15 | ManxPower | Juppers: that will make calls from your cisco match [yourcomecisco] and the context= line you have in that section |
19:48.24 | Juppers | so it should be set up in zapata.conf and not sip.conf |
19:48.33 | ManxPower | Juppers: no, in SIP since it is SIP |
19:48.56 | ManxPower | mercestes was not paying attention to your odd and terribly painful to thinkabout setup. |
19:49.06 | Juppers | so in sip and type peer with specific IP rules |
19:49.07 | ManxPower | I'd just use a SIPura box to do the same thing |
19:49.11 | *** join/#asterisk hellojoe (n=hellojoe@natint3.juniper.net) |
19:49.20 | ManxPower | Juppers: I would make it a type=friend in this case. |
19:49.41 | mercestes | ManxPower: probably. In every scenario I've seen fxs != sip. =/ |
19:50.46 | ManxPower | When I first started with Asterisk oh so many years ago I started out with FXO/FXS on a Cisco 1750 as my PSTN gateway device. |
19:50.56 | Juppers | what he tells my is in the default context he has to append my ip to get calls routed to me. and somehow that messes up things when trying to log me into the call queue. |
19:51.09 | tzanger | hmm, this filter should be catching it then |
19:51.13 | ManxPower | Juppers: he is confuzed. |
19:51.21 | tzanger | ethereal's seeing flags 0x10 and my tc filter is ip tos 0x10 0xff |
19:51.32 | hellojoe | Hi folks, is it true that Jajah is using Asterisk? |
19:51.46 | ManxPower | Juppers: "context" is the wrong word for sections in sip.conf unless he is talking about context] |
19:51.53 | Juppers | ManxPower - thanks a bunch. I'm trying to get our current config files so I can run a test box here and see if I run into the same as he is saying |
19:52.22 | Juppers | ManxPower - he is talking about the context sections in extensions.ael |
19:53.13 | Juppers | that instead of adressing me as Dial(SIP/224) he has to do Dial(SIP/224@xxx.xx.xxx.xxx) |
19:53.55 | ManxPower | Juppers: then he is REALLY confused. If your sip.conf is set up correctly then the call will go into the correct extensions.conf/extension.ael context with the correct destiantion number just like a real SIP phone. |
19:54.07 | ManxPower | Juppers: your set up is not very common, but should work. |
19:54.31 | ManxPower | Juppers: Nope! That is why you have a [244] section in sip.conf for your router with host=yourip |
19:54.57 | Juppers | ok.. so the whole thing is he has the sip.conf wrong for me. Thanks a bunch again |
19:55.41 | ManxPower | Juppers: He's going to have massive problems if he keeps thinking that an extension is a device. We actually use the MAC of the device as the SIP account ID |
19:55.57 | [TK]D-Fender | Juppers: if he's doing "Dial(SIP/224@xxx.xx.xxx.xxx)" then you are not authing at all. |
19:56.06 | EmleyMoor | Damn - 1 macro calling another sets the extension to s |
19:56.10 | ManxPower | Since it is easy for 4 different devices to have the same extension in a real corporate enviroment with whiney users that don't want to hear "no" |
19:56.16 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
19:56.41 | ManxPower | EmleyMoor: Set(SAVED_MACRO_EXTEN=${MACRO_EXTEN} before calling the 2nd macro |
19:56.42 | [TK]D-Fender | EmleyMoor: this is where you get to take a minute to think about the magic of PARAMETERS. |
19:57.03 | [TK]D-Fender | MACRO_EXTEN = waste. Just pass the value as a parameter |
19:57.18 | ManxPower | [TK]D-Fender: you mean the magic of if you use ARG1, ARG2, and ARG3 when calling the first macro and then when you call the end macro with ARG1 and ARG2, then ARG3 is STILL set? |
19:57.26 | funxion | anyone know why if I connect to voicemail then hangup asterisk restarts? |
19:57.32 | ManxPower | end == second |
19:57.51 | ManxPower | funxion: you are running 1.4 |
19:57.54 | funxion | no |
19:57.56 | funxion | 1-2 |
19:58.07 | [TK]D-Fender | ManxPower: Should "stack", shouldn't it? |
19:58.09 | ManxPower | funxion: never seen that problem ever |
19:58.15 | ManxPower | [TK]D-Fender: it doesn't. |
19:58.23 | funxion | last message in console is "Extension 9999, priority 1 returned normally even though call was hung up |
19:58.23 | funxion | debast-vm*CLI> |
19:58.23 | funxion | Disconnected from Asterisk server |
19:58.34 | [TK]D-Fender | ManxPower: Silly. MACRO_EXTEN should be just as vulnerable then |
19:58.55 | ManxPower | all args set when calling a macro stay set when you call any other macro from within the first macro unless you overrite them |
19:59.22 | ManxPower | funxion: hanging up a call IS normal |
19:59.34 | funxion | yes but then it restart asterisk |
19:59.35 | funxion | hmm |
19:59.51 | EmleyMoor | I knew ARG1 could be used to fix it - just wasn't sure until just now how to pass it with ExecIf |
19:59.52 | [TK]D-Fender | ManxPower: Ok, and does the nested on get ITS ARG1 where appropriate? And upon return, is the original valure reinstated? |
20:00.15 | ManxPower | [TK]D-Fender: I don't know. |
20:00.18 | [TK]D-Fender | ManxPower: (I will of course test this when I get home) |
20:00.48 | EmleyMoor | All my permission checking is now done by macros :-) |
20:00.53 | [TK]D-Fender | ManxPower: Now it #1 used ARG3, and #2 doesn't, SURE, #2 could access ARG3, but it WOULDN'T (by any sane coder) |
20:01.11 | [TK]D-Fender | s/it/if |
20:01.17 | EmleyMoor | (except checking of whitelist... that is done to bypass their calling) |
20:01.32 | ManxPower | Macro(happy-1,FRED, BARNY, WILMA) calls Macro(happy-2,FRED) then in [macro-happy-2] ARG2 is BARNY and ARG3 is WILMA, whereas I think they should not exist or be empty |
20:01.37 | [TK]D-Fender | EmleyMoor: All of your dialout's should be macro's as well |
20:02.09 | hellojoe | Guys, can anyone point me to AGI script that works like Jajah? I need this for a group of folks I am working with. Any pointers wouldl be appreciated. |
20:02.22 | EmleyMoor | [TK]D-Fender: I think some of them are a little too complicated to be macros |
20:02.41 | ManxPower | [TK]D-Fender: I got stung by this when I had a macro call itself with a different number of params and the macro did different things depending on the number of params |
20:02.43 | hellojoe | By the way, it doesn't have to be AGI. I have tried this with .call files too (not very reliable though) |
20:03.03 | hellojoe | any suggestions (AGI Vs .call?) |
20:03.07 | ManxPower | hellojoe: Does anyone here even know what the heck JaJah is? |
20:03.08 | [TK]D-Fender | ManxPower: as long as happ-2's ARG1 takes precedence and gets set back upon completion, who cares? Why would you reference a parameter not provided from within the macro? That'd be sloppy coding |
20:03.15 | hellojoe | huh! |
20:03.16 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
20:03.17 | hellojoe | jajah.com |
20:03.39 | ManxPower | [TK]D-Fender: because I don't know how many options a macro was called with. |
20:04.02 | hellojoe | basically, enter two telephone number on an online portal and then connect the two parties |
20:04.05 | hellojoe | that's all I want |
20:04.10 | [TK]D-Fender | ManxPower: I guess if you're making variable parameter macro's,but thats a strange animal.... |
20:04.14 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
20:04.22 | Juggie | then you shuold include a flag to tell the macro what it should expect |
20:04.25 | hellojoe | so a script to which I can feed 2 telephone numbers and expect it to connect |
20:04.35 | ManxPower | [TK]D-Fender: it was elegant and simple -- or would have been |
20:04.35 | hellojoe | the parties using .call or AGI |
20:04.43 | Juggie | so Macro(type of action,var,var,var) |
20:04.53 | ManxPower | hellojoe: did you check the mailing lists or the wiki? |
20:05.03 | hellojoe | yes.. no luck :-( |
20:05.06 | [TK]D-Fender | ManxPower: http://www.sofaswitch.org/d/ |
20:05.08 | ManxPower | Juggie: I coded around the issue |
20:05.21 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
20:05.28 | *** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar) |
20:05.30 | Juggie | its not so much an issue as it is a pitfall :) |
20:05.37 | [TK]D-Fender | ManxPower: Increasingly imperfect solutions for an imperfect world ;) |
20:05.58 | [TK]D-Fender | Juggie: Pitfalls wouldn't be so bad... if not for the giant spikes at the bottom ;) |
20:06.14 | BrianR___ | When I call Dial() from a zap channel, is the PRI hangup cause passed back to the calling channel? |
20:06.28 | Juggie | yes. |
20:06.35 | Juggie | well, the hangupcause |
20:06.37 | L|NUX | is there any one who do have global crossing dids |
20:07.14 | BrianR___ | Ie, if I dial a number that would result in a hangup with a "This number is not inservice..." recording, does the magic #1 code pass through automagically, or do I need to do some magic like Hangup(PRI_CAUSE) or something? |
20:07.36 | Juggie | you mean to pass it back to the dialing channel? |
20:07.51 | Juggie | so the phone will play the proper error? |
20:08.35 | BrianR___ | Juggie: Yeah. Let's take the following hypothetical: I have an asterisk box with two T1 linecards, one connected to the PSTN and the other to a second asterisk PBX. I want the second asterisk PBX to be able to see the hangup cause, but all of its calls are going via the other asterisk PBX (over T1) |
20:09.04 | ManxPower | Brian What signaling between the two asterisk's? |
20:09.16 | BrianR___ | is it enough to have in the [fromotherasterisk] context a rule like _X.,1,Dial(Zap/2/${EXTEN}) |
20:09.20 | BrianR___ | ManxPower: PRI signalling over T1 |
20:09.23 | Juggie | so far as my expierence you need to forcefully pass it |
20:09.26 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
20:09.29 | ManxPower | brian: it should pass it back |
20:09.57 | ManxPower | then you need to decide what to do based on the value of hangupcause on the originating box |
20:10.14 | BrianR___ | Juggie: Do the HANGUP_REASON, PRI_HANGUP, and Hangup(ARG1) use the same set of magic numbers? |
20:10.17 | Juggie | i have something like this, http://pastebin.ca/365231 |
20:11.03 | ManxPower | BrianR___: Hangup(number) only works in 1.2 AFIK |
20:11.11 | ManxPower | ..er... ONLY in 1.4 |
20:11.20 | *** join/#asterisk apardo (n=apardo@87.217.144.34) |
20:11.29 | BrianR___ | The local CO plays recordings for some of the messages - I'm not sure how to detect if the CO has a recording or I need to play my own also... |
20:11.36 | ManxPower | BrianR___: all of asterisk's hangupcauses are based on Q.931 hangup causes |
20:11.37 | Juggie | that macro should probally be expanded to include other dialstatus's but its what i use for sip->pstn |
20:11.53 | ManxPower | BrianR___: always assume you have to play your own on PRI |
20:12.24 | BrianR___ | The current system is a rather ancient 1.0.6... Without any special handling, it plays a reorder tone for all of the error conditions. |
20:13.03 | BrianR___ | I'm testing with a 1.2.12 system, trying to figure out if it needs any hacking or not. |
20:13.12 | ManxPower | BrianR___: yes, it would as you are supposed to handle your own audio messages when dealing with OOB signaling like SIP or PRI |
20:13.44 | *** join/#asterisk Vec (n=Vector@dsl-243-93-201.telkomadsl.co.za) |
20:13.46 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-224-149.dhcp.insightbb.com) |
20:13.54 | BrianR___ | ManxPower: How does that apply to things like SIP phones, which could potentially receive a hangupcause? |
20:14.23 | ManxPower | BrianR___: the phone won't play "The number you have dialed, 555-1212 has been disconnected". |
20:14.58 | *** join/#asterisk swamig007 (i=boom@59.92.194.37) |
20:15.04 | ManxPower | If you don't do anything the translated cause code from Q.931 should be translated into whatever the equiv SIP code is and the phone would prolly give a reorder/congestion tone |
20:15.14 | BrianR___ | Looks like the polycom phones I have just play the reorder tone for all non-zero hangup causes. |
20:15.29 | ManxPower | BrianR___: I'll bet most phones would do that |
20:15.38 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
20:15.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
20:16.12 | [[blah]asfd | has anyone experienced static on the call when no one is speaking... and then when you speak, it goes away, then comes back when you stop speaking? is there a way to correct this? |
20:16.24 | ManxPower | [[blah]asfd: no. |
20:16.28 | [[blah]asfd | phones are connected to another server via SIP |
20:16.40 | [[blah]asfd | the server that it is connected to is connected to pri t1s. |
20:16.42 | PaulTech85 | Is there anyway to get a trigger when a call goes connected in the queue to call a command |
20:16.46 | ManxPower | [[blah]asfd: the phones might have VAD/CNG enabled that might cause such a problem I guess. |
20:16.49 | [[blah]asfd | but the phones on that system do not have this. |
20:16.49 | PaulTech85 | I want to send info to the agent that answered the phone |
20:17.02 | BrianR___ | ManxPower: I suppose it may be enough to just make a dial macro that jumps to Busy 41 and lets the phone play reorder for everything else... |
20:17.08 | [[blah]asfd | they are the polycom spountpoint ip 301 |
20:17.13 | [[blah]asfd | where would i find that setting? |
20:17.14 | ManxPower | BrianR___: standby |
20:17.16 | PaulTech85 | I think mysql is the only way, constantly monitoring the queue_log |
20:17.26 | ManxPower | [[blah]asfd: it is a setting ON THE PHONE |
20:18.11 | [[blah]asfd | looking... you mean in the menu and screen on the phone as opposed to the sip.cfg or mac.cfg right? |
20:18.30 | ManxPower | BrianR___: some of this is 1.2 specific http://www.fnords.org/~eric/macros.inc |
20:18.44 | ManxPower | [[blah]asfd: what brand of phone? |
20:18.56 | ManxPower | [[blah]asfd: ah. VAD/CND is NOT on by default |
20:19.17 | ManxPower | [[blah]asfd: no, I mean it is set in the phone configuration, not the asterisk configuration |
20:19.32 | BrianR___ | Ok.. My 1.0.6 system is also patched with bristuff, which might explain why it allows hangupcause to be passed as an argument to hangup() |
20:20.10 | BrianR___ | Here, SetVar(PRI_CAUSE=1) and Hangup(1) produce the same result. |
20:20.56 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.15 (Feb. 9, 2007) Zaptel 1.2.14 (Feb. 19, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
20:21.03 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.15 (Feb. 9, 2007), Zaptel 1.2.14 (Feb. 19, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
20:21.36 | *** join/#asterisk ez` (n=ez@c207.134.229-230.clta.globetrotter.net) |
20:21.36 | [TK]D-Fender | yay, new zaptel..... |
20:21.52 | BrianR___ | Did the zaptel echo canceller ever get fixed? |
20:22.02 | [TK]D-Fender | BrianR___: Which? :) |
20:22.02 | ManxPower | BTW, for that pastebin the macro-dial-result is what you want to look at |
20:23.00 | BrianR___ | [TK]D-Fender: Uh... |
20:23.19 | BrianR___ | [TK]D-Fender: I finally broke down and bought a Sangoma A104D card with the Octasic echo canceller thing... |
20:23.38 | [TK]D-Fender | BrianR___: You say that... as though it were a BAD thing ;) |
20:23.56 | BrianR___ | [TK]D-Fender: In testing so far, it stops echo dead silent... |
20:24.15 | [TK]D-Fender | BrianR___: Ain't it grand? |
20:24.39 | BrianR___ | [TK]D-Fender: Also does the HDLC in hardware, which is nice. This Dell 1850 has only two PCI slots and the A IRQ wire in both of them is shared with other devices... |
20:25.14 | [TK]D-Fender | BrianR___: I've well versed with that card.... |
20:25.18 | [TK]D-Fender | I'm* |
20:25.40 | [TK]D-Fender | russellb: Which of thhose 3 entries in the changelog was key to such an early release? |
20:25.56 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
20:26.05 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
20:26.16 | russellb | [TK]D-Fender: the driver fix for the TDM800P |
20:26.19 | BrianR___ | [TK]D-Fender: Right now I have it mounted in a seperate PC and I'm transferring calls through it to test the echo canceller. Going to move it to a passthrough role tomorrow to stave off echo complaints until I have a chance to build up and test a new PBX around the A104D... |
20:26.35 | BrianR___ | [TK]D-Fender: Thus all of the questions about hangupcause :) |
20:26.47 | [TK]D-Fender | russellb: I sort of figured that... |
20:27.26 | russellb | BrianR___: interrupt sharing should no longer be a problem. If you have any problems related to that after zaptel 1.2.13, let us know ... |
20:27.42 | wunderkin | o rly? |
20:27.47 | russellb | yes |
20:27.51 | wunderkin | zomg |
20:27.57 | wunderkin | :d |
20:28.19 | russellb | and also, the digium quadspan and dualspan cards do hdlc in hardware, as well. |
20:28.54 | mercestes | nummeh |
20:29.28 | BrianR___ | russellb: Hmm.. Is that dependant on the zaptel driver version? |
20:29.42 | russellb | as long as it is anything not ancient ... |
20:29.58 | *** join/#asterisk tdi (n=tdi@seth.kill-9.pl) |
20:30.13 | tdi | hi, is this a channel to ask for t38 software? |
20:30.20 | russellb | o.O |
20:30.32 | [TK]D-Fender | russellb: Would that be defined as pre /topic? ;) |
20:30.40 | russellb | no |
20:30.41 | [TK]D-Fender | tdi: No. |
20:30.46 | russellb | like, over a year ago or something |
20:30.49 | BrianR___ | russellb: Ok. This machine Asterisk 1.0.6 and Zaptel 1.2.0 on it, which would probably make it sufficiently ancient to not work well |
20:30.52 | [TK]D-Fender | russellb: :O |
20:30.55 | tdi | [TK]D-Fender: aha |
20:31.06 | russellb | BrianR___: yes, that is pretty old. |
20:31.14 | *** part/#asterisk tdi (n=tdi@seth.kill-9.pl) |
20:37.30 | ChicagoBud | I'm about to dive into fax to mail via a sangoma a200d -- any advice -- looks like there are a bunch of ways to go... |
20:39.49 | n|cotine | Question on app_queue - why does the queue ring a dyanmic agent who is already on a queue call? |
20:40.00 | n|cotine | show queues even shows that the queue manager knows that the agent is (In use) |
20:40.26 | mercestes | n|cotine: I like that question. |
20:40.50 | mercestes | n|cotine: I would add to yoru question with, "why does it ring a static agent when *they* are on the phone?" |
20:40.58 | PaulTech85 | n|cotine Yes it still rings a In Use agent |
20:41.04 | PaulTech85 | Same for static |
20:41.05 | n|cotine | PaulTech85: This seems undesired. |
20:41.12 | n|cotine | Is there an easy fix in the source? |
20:41.12 | mercestes | PaulTech85: We agree. how to fix? |
20:41.30 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
20:41.32 | PaulTech85 | I wrote for a patch for mine but many people like the use of it |
20:41.40 | n|cotine | PaulTech85: Available anywhere? |
20:42.15 | PaulTech85 | It was done privately but I dont see why I cant release, What branchs are you guys patching against? |
20:42.20 | n|cotine | 1.2.15 |
20:42.23 | mercestes | 1.2.13 here |
20:42.33 | n|cotine | I can hand-apply if it's not too bad, though. |
20:42.40 | n|cotine | I can read C, just not write it. :) |
20:42.47 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
20:42.59 | PaulTech85 | let me look over it, Mine was patched against 1.4 I dont think much changed |
20:43.31 | n|cotine | PaulTech85: wouldn't it be in ring_entry - just check call status when you're checking wrapup time? |
20:43.57 | PaulTech85 | Yep, same place |
20:44.43 | PaulTech85 | When it checks 'ringinuse' against parent |
20:45.06 | `Sauron | Does anyone know offhand where the sound files are stored in openpbx? (I know this is #asterisk, bite me) |
20:45.37 | mercestes | `Sauron: no, you *BITE ME* and go crawling back to those asterisk hating source code stealing CLOWNS that can't fix their own shit. |
20:45.41 | mercestes | WTF is wrong with you? |
20:45.47 | `Sauron | Yawn. |
20:45.58 | mercestes | gah, your the type to walk into a vegan joint and go "man, where can I get some bacon?" |
20:46.00 | JerJer | <PROTECTED> |
20:46.27 | froguz | what is the "plancomment = " in the extensions.conf for? |
20:46.36 | `Sauron | JerJer: Thanks. I found that on my asterisk install at home. |
20:46.44 | JerJer | mercestes: lets feed the trolls with relevant info for Asterisk :) |
20:46.47 | `Sauron | Apparently they brainf*ed that part too. |
20:47.37 | `Sauron | mercestes: bitter? |
20:47.59 | fetcher | Is there a way to get SIP phone presence / BLF working when one of the phones is behind a 2nd Asterisk server? (linked to 1st via IAX)? |
20:48.03 | Corydon-w | `Sauron: off topic, please take it elsewhere |
20:48.33 | hellojoe | Guys, anyone has an opinion on .call Vs AGI scripts for connecting parties given their telephone numbers? I have found .call less reliable in the past |
20:48.41 | *** join/#asterisk dasenjo (n=dasenjo@190.24.24.34) |
20:48.56 | hellojoe | I used to generated .call files using a custom perl script and dump them on the same extension |
20:49.02 | ChicagoBud | Should I be looking at HylaFax or AsterFax or something else? |
20:49.21 | `Sauron | hellojoe: make sure you move .call files into the directory, so * detects them properly |
20:49.31 | `Sauron | i.e. create them elsewhere, then drop them in |
20:49.46 | froguz | AsteriskNOW has put a "plancomment = DialPlan1" on my extensions.conf and it's not documented anywhere |
20:49.52 | hellojoe | Yup.. did that. Becomes an issue when there are too many calls |
20:49.53 | fetcher | ChicagoBud: I use Hylafax, behind an AS5300 w/ t38modem and H.323. Works OK in the end, but it's a real mess to set up. |
20:49.57 | JerJer | `Sauron: we should illistrate your point |
20:49.58 | froguz | what is that for? |
20:50.20 | hellojoe | Sauron: Have you tried it in higher call volume scenarios? |
20:50.30 | `Sauron | JerJer: I use * at home. Have done so for the last 4 years. Where do I have to do the * cheerleader dance for y'all to stop being so bigoted? |
20:50.54 | `Sauron | JerJer: the day * can originate and terminate faxes, I'll switch the work stuff back to * |
20:50.59 | JerJer | when dealing with call files one should always MOVE files into the spool directory. Copying the file ~can~ lead to asterisk grabbing partically done copies |
20:51.11 | ChicagoBud | fetcher, Interesting. All I really need to do is terminte a fax call on a sangoma a200d an convert it to an email |
20:51.21 | mercestes | `Sauron: My * handles faxes just fine. |
20:51.28 | JerJer | while it is rare, it has happened to me |
20:51.31 | swamig007 | hi guys i am trying to get a SIP account configured as a trunk (this is an outgoing only account from primus india) i keep getting all circuits are busy ... |
20:51.33 | `Sauron | mercestes: origination including termination? |
20:51.38 | froguz | `Sauron, then move to * 1.4 |
20:51.39 | hellojoe | JerJer: yes, it happened to me |
20:51.40 | mercestes | yes. |
20:51.44 | `Sauron | Last I looked at 1.4, they said "use t.38" |
20:51.46 | froguz | with T.38 support |
20:51.46 | fetcher | I've used the app_rxtx FAX extension for a single line |
20:51.46 | `Sauron | I'm at 1.4 |
20:51.49 | Mavvie | !_(#_!(@_# |
20:51.53 | Mavvie | fscking IPS. |
20:51.56 | `Sauron | Mavvie |
20:51.57 | mercestes | in fact. I fail to see how switchign from * to * based derivatives "solved your fax problem." |
20:52.02 | hellojoe | JerJer: how to move files to the spool directory (Sorry, basic question) |
20:52.12 | Mavvie | The Tipping Point is blocking the Asterisk Keep Alive checks for SIP hosts. |
20:52.21 | JerJer | `Sauron: bloody tampons man - you made a good point and I was making the point you made |
20:52.23 | Mavvie | seventeen alerts on nagios right now :-) |
20:52.32 | ChicagoBud | mercestes, waht are you using for fax? |
20:52.32 | `Sauron | mercestes: openpbx put work into both analog fax origination/termination and t.38 origination/termination |
20:52.43 | mercestes | openpbx is a joke. |
20:52.46 | `Sauron | * 1.4 can't originate/terminate t.38, nor can it originate analog fax calls |
20:53.16 | JerJer | i fax almost every day via asterisk 1.2, TDM Card and a PRI |
20:53.16 | `Sauron | Mavvie: sucks, can you beat them with a black stick? |
20:53.31 | Mavvie | `Sauron: nah, will just make an exceptoin for our PABXs. |
20:54.05 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
20:54.08 | [TK]D-Fender | <`Sauron> Does anyone know offhand where the sound files are stored in openpbx? (I know this is #asterisk, bite me) <- as opening lines go, this is NOT a winner in the "how to win friends and influence people" book. YOU started the trash talk. Please sopt. just... STOP. |
20:54.08 | `Sauron | JerJer: then go flesh out the voip-info.org documentation on it. Last I looked (3-4 weeks ago) everybody was saying "Can't be done" |
20:54.15 | JerJer | Then Hylafax+Asterisk makes a functional solution - I don't like it but it does work - using PRI |
20:54.38 | JerJer | you need the right magic sause that nobody has totally figured out yet |
20:54.38 | `Sauron | Fender, I'm not in it for winning friends. I'm in it for getting shit to work. |
20:54.39 | mercestes | JerJer: Works with iaxmodem as well. |
20:54.44 | [TK]D-Fender | SHUP. Both of you. |
20:55.05 | JerJer | those that do know aren't talking |
20:55.10 | [TK]D-Fender | `Sauron: Good. Now you need to work on getting it to work in a CIVIL manner. |
20:55.16 | mercestes | wtf. I stopped talking five minutes ago...why am I getting lumped in with troll-boy? |
20:55.30 | JerJer | but faxing via g.711 over the net will simply never be reliable |
20:55.33 | funxion | anyone know of a reason that asterisk would restart when issueing a odbc show ? |
20:55.36 | `Sauron | Yawn. |
20:55.55 | J4k3 | or learning how to use the one they have. |
20:56.15 | `Sauron | JerJer: Seriously. If you can get some sort of docs together on how to both originate and terminate fax calls (to a PRI, I don't care) - I might even buy you a sixpack. |
20:56.47 | JerJer | i'll add it to my list of blog/article/whitepaper topics |
20:57.03 | BrianR___ | russellb: How is the firmware for the digium quad t1 cards handled anyway? |
20:57.03 | [TK]D-Fender | Strangly ironic how drinking 6-packs lead to a beer gut..... |
20:57.26 | `Sauron | And how well does MeetMe scale? :) |
20:57.33 | J4k3 | [TK]D-Fender: sheeit, I got a 30 pack then. |
20:57.34 | J4k3 | ;) |
20:57.39 | JerJer | `Sauron: via PRI, very nicely |
20:57.42 | file | [TK]D-Fender: crazy circles? |
20:57.46 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
20:57.49 | `Sauron | JerJer: And over sip trunks? |
20:57.51 | [TK]D-Fender | `Sauron: Not too bad, but it cooks more evenly if you toss it in the pan WHOLE ;) |
20:57.52 | JerJer | SIP maybe not so nicely - depends |
20:58.20 | J4k3 | why would you want to send 'fax tones' over voip |
20:58.33 | `Sauron | I dunno |
20:58.36 | `Sauron | I don't. :) |
20:58.37 | J4k3 | when you can just send 1's and 0's instead, and let the far end deal with the modem tones? |
20:58.49 | Corydon-w | J4k3: because people are cheap bastards |
20:59.08 | JerJer | i've had 8 full E-1 spans into one meetme |
20:59.23 | `Sauron | JerJer: the scaling, is due to the timing stuff? |
20:59.34 | `Sauron | scaling issue? |
20:59.38 | BrianR___ | for fax I've been using hylafax and of those T1 fax cards plugged into my asterisk box's quad pri card with a crossover cable... |
20:59.45 | JerJer | one issue in VoIP land is transcoding |
21:00.03 | J4k3 | Corydon-w: I'd say "shoot 'em all in the face then take their wallets" but I think the wallet thing might be mostly pointless :) |
21:00.05 | JerJer | if everyone runs ulaw then its not an issue |
21:00.12 | `Sauron | Blech, people actually still use hylafax? |
21:00.15 | JerJer | but if everyone wants to run G.729, it can become an issue |
21:00.25 | JerJer | `Sauron: many of my customers do |
21:00.27 | BrianR___ | I also have a few fax machines plugged into ATA's... |
21:00.45 | `Sauron | Last I looked at it.. it was poo. |
21:00.53 | `Sauron | then again, that might be 4-5 years ago |
21:01.04 | J4k3 | hylafax has been poo for like >10 years |
21:01.04 | JerJer | its not overly friendly |
21:01.08 | BrianR___ | `Sauron: hylafax is dirt simple faxing that "just works" |
21:01.26 | `Sauron | maybe I'll look at it when I get an extra couple minutes |
21:01.30 | JerJer | but once its setup it simply works |
21:01.39 | `Sauron | last time, it was unable to do what I needed it to do.. route faxes based on LDAP lookups |
21:01.45 | BrianR___ | We have DID's mapped to email addresses for incoming fax... And there's a email-to-fax gateway for sending faxes. |
21:01.54 | Mavvie | `Sauron: we have hylafax here too. And mgetty+sendfax. |
21:02.02 | BrianR___ | And printer driver for the windoze PC's that pops up a box asking for phone number and cover page info. |
21:02.10 | `Sauron | Mavvie: I ended up writing some perl super-glue to do the magic for me. |
21:02.23 | Mavvie | `Sauron: happily connetced to a Patton Dual E1 card, hooked up to a QUad PRI on the * server. |
21:02.30 | J4k3 | BrianR___: what you need is a terminal server that can handle fax traffic and NOT VoIP service. |
21:02.35 | J4k3 | :P |
21:02.54 | J4k3 | I mean shit, a $20 Livingston/Lucent Portmaster 3 can handle 48 (or 60) fax calls without a problem. |
21:03.00 | BrianR___ | The hylafax faxdispatch thing is just a shell script anyway.... |
21:03.01 | JerJer | BrianR___: we do very a similar operation as well |
21:03.06 | J4k3 | and yes, you can get them loaded with DSPs for about $20 off ebay. |
21:03.30 | JerJer | BrianR___: mine is perl |
21:03.42 | JerJer | like maybe 2k lines |
21:03.58 | J4k3 | now if you've got PRIs terminated to your box, thats different... although I doubt peecees handle the dsp tasks very well. |
21:04.02 | BrianR___ | I bought on3e of those divaserver T1 fax cards... Basicly a bunch of faxmodems on a PCI card... |
21:04.15 | J4k3 | brian: yuck... and you paid more than $10 for it? |
21:04.16 | JerJer | yup |
21:04.43 | BrianR___ | They're way more than $10... |
21:04.54 | JerJer | yeah, but its paid for itself time and time again |
21:05.26 | [TK]D-Fender | JerJer: Then it wasn't "too much", just "more than you'd have liked" |
21:05.38 | BrianR___ | But a lot less agrivating than futzing with half-baked soft dsp stuff or obsolete modem-pool-in-a-box appiances... |
21:05.52 | tzanger | hmm |
21:06.02 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
21:06.45 | BrianR___ | Thing just plugs in an shows up as 24 pci faxmodems... |
21:06.56 | tzanger | [incoming] has an extension that Dial(SIP/500,gt). SIP/500 picks up. Now I'm trying to use # to transfer. I hit #, I hear "Transfer" but when I go to type the extension it bombs out, saying that the extension doesn't exist in [sipphones] |
21:07.17 | JerJer | BrianR___: sorta - I got anxcious and bought one from a supplier...then a month later a customer wanted their own solution but wanted it cheap, so I found the same card on EBay for over 1/2 what I paid :( |
21:07.20 | J4k3 | BrianR___: maybe for you. |
21:07.21 | tzanger | this is true, the extension does not exist in [sipphones] but why's it looking at SIP/500's sip.conf entry? |
21:07.28 | tzanger | actually that makes some sense |
21:07.44 | Mavvie | http://lists.digium.com/pipermail/asterisk-users/2007-February/180535.html |
21:07.48 | Mavvie | so, I can go back to bed! |
21:07.55 | [TK]D-Fender | tzanger: "transfercontext" in sip.conf? |
21:07.59 | tzanger | makes it hell for me to Dial(SIP/500&Zap/g1/${CELLPHONE},,gt) and have my cell phone able to park the call so I can pick it up at home |
21:08.23 | tzanger | [TK]D-Fender: does a similar thing exist in zapata.conf? |
21:08.35 | GaVak | Status: Provisioned, Down, Active |
21:08.53 | GaVak | Does that mean that the card is good to go, but the other end isn't up? |
21:08.54 | JerJer | tzanger: there is a global value for transfer context |
21:09.00 | GaVak | (For PRI) |
21:09.11 | tzanger | JerJer: is there? |
21:09.16 | [TK]D-Fender | GaVak: You mentioned your D-chan bouncing on you. pastebin - "cat /proc/interrupts" |
21:09.32 | JerJer | on PRIs i use a global channel variable |
21:09.42 | JerJer | not sure about sip though |
21:10.43 | mercestes | ok, asterisk 1.2.13, when you get a voicemail, and then forward it with a prepended message, the "notification email" app_voicemail sends out says I have a new message of "0.00" duration. Voicemail itself is fine. Is this a known "feature?" |
21:10.46 | tzanger | [TK]D-Fender: transfercontext is not listedn in sip.conf.sample |
21:10.52 | [TK]D-Fender | GaVak: Means the card is ready, but the link is dead (no claims for fault) |
21:11.05 | [TK]D-Fender | tzanger: I ran into is somewhere, just check your file..... |
21:11.13 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
21:11.32 | GaVak | TK, I'm posting it now... The provider said they saw the D-Channel come up, then drop.. I reloaded *, and they said they saw it happen again. |
21:11.42 | elriah | Hi all. Is it possible to refer to audio files via full paths in an extensions context? i.e., background(/some/path/to/audio) instead of putting everything in the asterisk sounds dir? |
21:11.43 | tzanger | JerJer: I see ${TRANSFER_CONTEXT}, is that what you're talking about? |
21:11.43 | bkruse | GaVak: ztcfg -vv |
21:11.48 | GaVak | http://pastebin.com/885437 |
21:11.51 | JerJer | the word transfer doesn't exist in my sip.conf.sample v1.2.13 |
21:11.55 | mercestes | users are complaining that the duration in the emails are wrong. ;.; why? I don't know. |
21:11.55 | JerJer | yes |
21:12.01 | bkruse | GaVak: do that. |
21:12.07 | JerJer | mercestes: don't send the duration then :P |
21:12.37 | GaVak | Bkruse: 25 channels configured 1-12 = pots, 25-37=b, 48=d |
21:12.48 | bkruse | GaVak: ztcfg -vv gives no errors? |
21:12.54 | GaVak | none at all. |
21:13.01 | bkruse | 25 channels configured? what mode and framing are you using? |
21:13.04 | mercestes | JerJer: Nice. How to fix? |
21:13.07 | mercestes | >.> |
21:13.15 | GaVak | there are 2 cards |
21:13.21 | bkruse | mercestes: rm -f `which asterisk` ? |
21:13.25 | funxion | does anyone have odbc voicemail storage setup in here? |
21:13.33 | mercestes | GaVak: Awesome! Thanks! |
21:13.39 | bkruse | GaVak: post your zaptel.conf and zapata.conf, i have time to take quick look while im waiting for this to load |
21:13.42 | bkruse | mercestes: that was me :] |
21:13.57 | *** join/#asterisk anthonyl (n=fbfff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
21:14.00 | mercestes | bkruse: Ah, okies. gah, does it *have* to patch all these files? Do I need to recompile/restart? |
21:14.01 | bkruse | mercestes: ive seen that also, not remember what i do to change it, or i fixed it |
21:14.11 | bkruse | :P |
21:14.17 | funxion | what modules need to be loaded to get voicemail odbc working? |
21:15.29 | BrianR___ | Still interested in russellb's mention that my TE410 cards might become more usable with newer zaptel drivers... |
21:15.35 | tzanger | JerJer: that works much better :-) |
21:15.36 | elriah | Anyone: Can I use pathnames in the BackGround command for sound files? If I want to orgranize sound files into directories? |
21:15.55 | JerJer | elriah: yes |
21:16.10 | GaVak | pasting the config files now. |
21:16.11 | JerJer | i do Background(prompts/foo) |
21:16.11 | wunderkin | elriah, just dont specify the file extension |
21:16.21 | elriah | Thanks. |
21:16.42 | JerJer | then symlink /var/lib/asterisk/sounds/prompts/ to a different location (if necessary) |
21:17.02 | bkruse | BrianR___: there is a problem with your te410? |
21:17.08 | funxion | what modules need to be loaded to get voicemail odbc working? |
21:17.27 | bkruse | funxion: ls /usr/lib/asterisk/modules | grep odbc |
21:17.29 | CJLinst | Re: multiple extensions from one phone: Got it working by defining the sip devices as type=friend. But that broke subscription hints for outgoing calls. Hints still process for incoming calls (busy-level=1). |
21:17.32 | GaVak | bkruse: http://pastebin.com/885452 |
21:17.47 | funxion | bkruse do i need to preload or just load? |
21:18.03 | bkruse | try to just load, because if they depend on another module, theyll fail |
21:18.07 | bkruse | try to just load |
21:18.15 | bkruse | or even better yet, load from an asterisk already running |
21:18.15 | BrianR___ | bkruse: echo, causes PCI erros from time to time, sometimes has what appear to be timing slips. |
21:18.24 | bkruse | interesting...... |
21:18.28 | bkruse | setup? |
21:18.44 | bkruse | switchtype=5ess? |
21:18.46 | bkruse | cool |
21:19.48 | bkruse | looks fine to me GaVak, check zttool to see if they come out of alarm |
21:20.42 | funxion | if I try to do odbc show and it restart asterisk would that mean there is a problem with my dsn? |
21:20.59 | GaVak | ok |
21:22.10 | GaVak | No alarms, internally clocked. |
21:22.11 | tzanger | is there a way to reload features.conf without restarting asterisk? |
21:22.20 | tzanger | module reload res_features.so? |
21:22.30 | tzanger | yep |
21:22.31 | tzanger | heh |
21:22.50 | bkruse | gavak, so it says its configured correct, and in green? |
21:22.55 | bkruse | (pri) |
21:22.57 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
21:23.24 | GaVak | In green, I didn't see a green, but the customers DSL just went down again. *sigh* |
21:23.27 | GaVak | lost my access |
21:23.47 | GaVak | I was getting some strange errors though |
21:23.53 | GaVak | Write to 44 failed: Unknown error 500 |
21:24.10 | GaVak | Short write: 0/5 (Unknown error 500) |
21:24.14 | GaVak | not sure if it was related |
21:24.20 | bkruse | woah... |
21:24.47 | tzanger | parkcall => #72 in features.conf does not seem to work |
21:24.53 | tzanger | *3 for blindxfr does, but not that |
21:25.04 | ManxPower | tzanger: you confirmed that your phone is not eating #72 |
21:25.19 | tzanger | it doesn't eat *3 (cell phone) |
21:25.58 | funxion | anyone know asterisk would restart after issueing odbc show |
21:26.00 | ManxPower | ah, so not a SIP phone. |
21:26.13 | ManxPower | funxion: you have a serious build or hardware problem |
21:26.32 | ManxPower | and no amount of debugging asterisk is going to fix your problems. What ELSE makes asterisk crash? |
21:26.34 | funxion | its most recent stable build |
21:26.35 | funxion | 1.2 |
21:26.40 | funxion | no hardware issue |
21:26.43 | tzanger | ManxPower: nope not a sip phone |
21:26.51 | ManxPower | you already have two totally different things voicemail and show odbc which is making asterisk crash. |
21:27.07 | ManxPower | funxion: start asterisk as "asterisk -cvvv" replicate the problem, look at the error message |
21:27.21 | bkruse | turn on debugging in logger.conf |
21:27.25 | ManxPower | funxion: asterisk does not just magically crash |
21:27.25 | funxion | it worked fine until i decided to add odbc voicemail |
21:27.33 | ManxPower | bkruse: I think he's getting a segfault |
21:27.37 | bkruse | ouch |
21:27.38 | bkruse | nvm |
21:27.41 | funxion | part of checking to see if the voicemail is configured is using the odbc show |
21:27.55 | funxion | I have debugging in logger.conf |
21:28.01 | funxion | ManxPower not segfault |
21:28.06 | mercestes | funxion: Then you did something wrong. lol |
21:28.13 | funxion | obviouly |
21:28.18 | funxion | trying to figure out what |
21:28.24 | bkruse | ok. |
21:28.25 | bkruse | and? |
21:28.26 | funxion | I followed the hoto |
21:28.27 | ManxPower | funxion: what is the actual message on the console when running as "Asterisk -cvvv" |
21:28.32 | funxion | it no werkie |
21:28.47 | ManxPower | remember, you won't get all the message when running "asterisk -r" |
21:29.10 | funxion | its segfault |
21:29.40 | funxion | ur right |
21:29.40 | hellojoe | Hi folks, how do I get one of my macros to return some value? |
21:29.40 | funxion | what would cause that |
21:29.40 | ManxPower | funxion: I have a bit of experience with Asterisk |
21:29.41 | funxion | I know |
21:29.42 | ManxPower | funxion: a seg fault generates a core file. see the README.backtrace |
21:30.18 | [TK]D-Fender | tzanger: You sure TRANSFER is not eating your "#"? |
21:30.24 | ManxPower | Personally I'll bet you have a buggy ODBC library -- a version not tested with Asterisk, but I don't use ODBC |
21:31.17 | funxion | Ill tell you what Im trying to do, let me know if you can think of a better way |
21:31.22 | hellojoe | This is something I have been looking for. If asterisk macro can return a True/False, it could help in dialplan |
21:31.53 | ManxPower | funxion: that requires that I think -- I charge for thinking |
21:32.20 | funxion | I ahev an asterisk cluster with a central vm server and was hoping to make mwi werk by adding a odbc voicemail storage to all cluster nodes |
21:32.34 | ManxPower | hellojoe: it can't, but you can Set(BOOL=TRUE) before you exit the macro, then check that in your dialplan. You can do this with any variable |
21:33.01 | ManxPower | pretty much any variable you set is a global channel variable |
21:33.10 | tzanger | [TK]D-Fender: nope my featuremap has no # except for #72 |
21:33.10 | mmlj4 | hey ManxPower |
21:33.15 | tzanger | [TK]D-Fender: all others are *codes |
21:33.15 | hellojoe | i see.. thanks a lot... |
21:33.24 | ManxPower | hello mmlj4, yes I know you are joey |
21:33.24 | funxion | I know I can probably just write a script to create a msg.txt and msg.wav on each server it would prolly werk but I wanted it be built in |
21:33.28 | mmlj4 | ManxPower: um, no visits to the new covington UMC yet? |
21:34.18 | mmlj4 | you are on that job, right? |
21:34.30 | ManxPower | mmlj4: I installed the router and got everything going, then they told me about the admin office about 1 week before they moved in so I had to take the covington router for the admin office. I told JB to order a replacement router |
21:34.42 | ManxPower | I had MY stuff installed before they even had power outlets working |
21:34.42 | mmlj4 | heh |
21:35.10 | mmlj4 | well, all my stuff is done, except I haven't screwed the wallplates on yet |
21:35.12 | ManxPower | I had to use an extension cable to get power from an outside outlet for testing, etc |
21:35.22 | mmlj4 | probably done by next weekend |
21:35.42 | ManxPower | If they are not numbered I will be referring the users to you if they need to figure out which wall plate goes to which patch panel outlet |
21:36.02 | mmlj4 | they are numbered... the wires are, and the plates will be |
21:36.28 | mmlj4 | hey, freezer tape is cheap :-) |
21:36.38 | ManxPower | oh, the switches were also ordered months ago. |
21:36.49 | mmlj4 | not in yet? |
21:36.52 | ManxPower | the channel bank was installed months ago as well. |
21:37.00 | mmlj4 | yeah, that's there |
21:37.01 | ManxPower | mmlj4: They arrived months ago too |
21:37.10 | Bobthehunter | whats the order of god phones ? polycoms then waht what astra and cisco ? |
21:37.15 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
21:37.40 | mmlj4 | astara or whatever looks interesting, Bobthehunter |
21:37.58 | [TK]D-Fender | Bobthehunter: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x |
21:39.00 | mmlj4 | ManxPower: according to the soekris website, they're putting out a faster board shortly, maybe by next month... it's expected to be fast enough to run a T1 card (sangoma or whatever) and a bunch of SIP channels |
21:39.03 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
21:39.27 | mmlj4 | the existing soekris stuff can either route T1, or handle a bunch of SIP channels, but not both |
21:39.36 | thinwires | hi, could someone help me out real quick? |
21:39.43 | [TK]D-Fender | mmlj4: And support PCI 2.2? |
21:39.47 | mmlj4 | anyhow, I want to play with the new board, once it comes out |
21:39.53 | *** join/#asterisk Skarmeth (n=Skarmeth@201009014171.user.veloxzone.com.br) |
21:39.55 | mmlj4 | [TK]D-Fender: looking now |
21:39.58 | [TK]D-Fender | thinwires: Ask a specific question and you might get a specific answer. |
21:40.05 | thinwires | :-) |
21:40.42 | mmlj4 | [TK]D-Fender: not sure, it doesn't say yet |
21:40.43 | thinwires | I'm having issues when Asterisk Sends out a voicemail, in an email... it comes out all ad and the headers and message is garbleded |
21:41.46 | ManxPower | [TK]D-Fender: you need to teach that to jbot |
21:41.51 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
21:42.07 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
21:42.24 | ManxPower | thinwires: the message headers are messed up or the audio file headers are messed up? |
21:42.25 | [TK]D-Fender | ManxPower: I know how, I just hadn't repeated it enough to bother yet :) I think I may fill up the bot shortly... |
21:42.49 | ManxPower | jbot phones is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x |
21:42.51 | jbot | ...but phones is already something else... |
21:42.55 | ManxPower | ~phones |
21:42.56 | jbot | rumour has it, phones is at http://bani.anime.net/phones/ |
21:43.12 | Qwell[] | That's a good link :D |
21:43.24 | Qwell[] | and has anybody seen bani lately? |
21:43.30 | Bobthehunter | thanks |
21:43.31 | ManxPower | jbot phones is also is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x |
21:43.33 | jbot | ManxPower: okay |
21:43.35 | thinwires | both, the only thing I can read is "you have a message from userX" the header and sender don't work and the voicemail.wav is just a bunch of text |
21:43.36 | ManxPower | ~phones |
21:43.38 | jbot | well, phones is at http://bani.anime.net/phones/, or is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x |
21:43.38 | GaVak | Bkruse: you still around? |
21:43.43 | greendisease | Qwell[]: ping |
21:43.56 | Qwell[] | greendisease: I'm not really here. |
21:44.00 | ManxPower | thinwires: I've never experienced that issue |
21:44.19 | mmlj4 | anyhow: # 433 to 600 Mhz AMD Geode LX single chip processor with CS5536 companion chip |
21:44.19 | mmlj4 | <PROTECTED> |
21:44.33 | Qwell[] | well, okay, I guess I can be here for you |
21:44.36 | *** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar) |
21:44.38 | bkruse | GaVak: hey sup, if its green |
21:44.41 | Qwell[] | greendisease: what's up? |
21:45.01 | mmlj4 | the only gotcha I see is, does * require a sound card to process calls and/or voicemail? |
21:45.03 | tzanger | heh |
21:45.04 | Z_God | is anyone here using either the gtalk or jingle channel? I get only audio for 1 sec to Google Talk |
21:45.09 | tzanger | I think I found a bug, but maybe not |
21:45.30 | GaVak | bkruse: how would I know if its green. (I'm not on site.) The show command shows: Primary D-channel: 48 |
21:45.30 | GaVak | Status: Provisioned, Down, Active |
21:45.51 | tzanger | I call a DID on my PRI from a SIP phone. The DID dials my Cell phone. My cell phone picks up, I dial *3, 700 to park the call (from the cell, so the sip phone is parked) |
21:45.59 | ManxPower | GaVak: pri debug span 1 |
21:46.02 | ManxPower | ..er.. sorry |
21:46.05 | tzanger | my cell is now hung up, so I call in to my DISA and hit 701 and pick up the call just fine |
21:46.07 | bkruse | GaVak: it would say green in zttool |
21:46.18 | ManxPower | pri debug span 2 since your PRI is on span 2 |
21:46.36 | ManxPower | you should be getting messages from the telco |
21:46.42 | tzanger | I have exten => _8XX,1,Goto(parkedcalls,7${EXTEN:1},1) in my DISA context for other reasons |
21:46.51 | GaVak | zttool says no alarms |
21:46.59 | ManxPower | if you are not then you call the telco and say "I'm not getting anything on my D-Channel. Fix it or I will kill you." |
21:47.13 | ManxPower | GaVak: zztool know nothing about PRI, only the physical layer T-1 |
21:47.15 | GaVak | pri span debug 2, no logging |
21:47.26 | tzanger | I call the same DID from the SIP phone, call is picked up by cell as before. This time I dial *3, 800 to park the call (from the cell, so the sip phone is parked, as before)... now my cell phone is hung up and the SIP PHONE hears "seven oh one" but the SIP phone is parked! |
21:47.33 | tzanger | I call in on the cell again and dial 701 and it works |
21:47.38 | GaVak | Max: lol, any suggested methods of death to offer them? |
21:47.38 | ManxPower | GaVak: I assume you are in USA or Canada? |
21:47.42 | tzanger | it's playing it to the wrong end :-) |
21:47.43 | GaVak | Usa |
21:48.18 | ManxPower | GaVak: I always find the threat of being transfered to sales usually works. |
21:48.38 | GaVak | Lol, ok, I'll take it back to them. This is my first PRI install and I wasn't sure how to do the debugging my end. |
21:48.44 | GaVak | But as you say, I don't see any data. |
21:49.04 | ManxPower | GaVak: make sure you have all debugging enabled for the console in /etc/asterisk/logger.conf first |
21:49.04 | *** join/#asterisk mega (n=mega@217.201.132.7) |
21:49.11 | ManxPower | then a logger restart or logger reload |
21:50.10 | tzanger | what's funnier is that defining extension 800 as exten => 800,1,ParkAndAnnounce(PARKED) works as it should (i.e. the cell phone hears the announcement, not the phone that is about to get parked) |
21:50.45 | GaVak | uncommnted the section about console=yadda,yadda,yadda,debug |
21:50.47 | GaVak | reloaded the logger |
21:52.19 | GaVak | wow |
21:52.35 | GaVak | Feb 20 16:52:21 NOTICE[18411]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 |
21:52.46 | GaVak | spamming like crazy |
21:53.44 | Corydon-w | Sounds like a timing problem |
21:55.10 | pigpen | could someone tell me where to put the TRANSFER_CONTEXT definition for blind transfer... |
21:56.18 | pigpen | I see there is a __TRANSFER_CONTEXT...but once again...I am unsure where to stick it.... |
21:56.18 | GaVak | what do you do on timing problems? Is it a setting, or a service call to the provider? |
21:56.19 | bkw_ | GaVak, chances are its IRQ misses |
21:56.26 | bkw_ | which in turn cause frame slips |
21:56.27 | bkw_ | which are the root of the errors you're seeing |
21:56.50 | Juggie | OR, it could just be a bad cable |
21:56.57 | bkw_ | or bad card |
21:57.01 | bkw_ | or bad computer |
21:57.03 | pigpen | or low signal |
21:57.04 | GaVak | lol |
21:57.07 | bkw_ | but i doubt its the cable |
21:57.15 | pigpen | or bad smartjack |
21:57.24 | bkw_ | now I have had that happen |
21:57.32 | GaVak | wow, this is a hard error to pin down, no? |
21:57.36 | *** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net) |
21:57.53 | vader-- | does anyone know how to reload the zapata.conf in the asterisk console? |
21:57.59 | n|cotine | I noticed the discussion on asterisk-user regarding Dell servers - does anyone NOT have a problem utilizing a 2850 or a 2950 as a PBX with Asterisk? |
21:58.00 | tzanger | haha |
21:58.01 | tzanger | <PROTECTED> |
21:58.01 | tzanger | <PROTECTED> |
21:58.04 | bkw_ | vader--, just restart |
21:58.18 | vader-- | i don't wanna shut the phone system down |
21:58.19 | Corydon-w | vader--: you can do a reload chan_zap.so |
21:58.23 | bkw_ | n|cotine, buy sangoma cards and it will work |
21:58.34 | Corydon-w | vader--: that'll work as long as you aren't changing signalling |
21:58.45 | pigpen | n|cotine, hmm...I have several Dell 6850's.....and I had a dell 2850 running great too. |
21:58.59 | n|cotine | pigpen: What type of PSTN hardware? |
21:59.00 | vader-- | ya i wanted to change signalling |
21:59.02 | mafkees | n|cotine: they work great here |
21:59.04 | CrashHD | hints for local parking extensions doesn't seem to be working, any ideas? |
21:59.05 | Corydon-w | n|cotine: it can be done, you just need to isolate the IRQ |
21:59.06 | CrashHD | 1.4? |
21:59.15 | pigpen | n|cotine, digium, 2port pri/4 port pri |
21:59.26 | mafkees | Corydon-w: with quadbri I did not have to isolate IRQ at all |
21:59.37 | pigpen | n|cotine, it is helpful to have a gentoo kernel dev in house. :) |
21:59.42 | mafkees | just popin the card, boot debian, compile bristuff and it's working great |
21:59.43 | Corydon-w | mafkees: depends upon your load |
22:00.00 | bkw_ | pigpen, you mean linux kernel dev.. I wasn't aware gentoo wrote a kernel :P |
22:00.03 | *** join/#asterisk fibs- (n=chrisk@c-24-20-45-4.hsd1.mn.comcast.net) |
22:00.05 | mafkees | and the sangoma A101 works great in it as well |
22:00.10 | Corydon-w | Heavy system load needs a separate IRQ |
22:00.11 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
22:00.13 | fibs- | Hola |
22:00.21 | bkw_ | sangoma cards don't have these IRQ issues |
22:00.25 | bkw_ | they can even share |
22:00.32 | n|cotine | If I have POTS lines, are people seeing these issues with the TDM400P also in Dell hardware? |
22:00.41 | mafkees | we have 5 2850's with quadbri and 2 2850's with sangoma A101 |
22:00.45 | mafkees | all machines run fine |
22:00.57 | mafkees | no need to fidle with bios or anything |
22:00.58 | thekidrio | can i get asterisk to dial two numbers via outbound IAX2 and then connect the two calls together? |
22:01.17 | mafkees | thekidrio: yeah, look at callfiles or the manager Originate call |
22:01.36 | pigpen | bkw_, bla. |
22:01.36 | thekidrio | really i want asterisk to call one number, and if that number answers to dial out to another number |
22:01.38 | JT | n|cotine: digium cards favour dell hardware |
22:01.40 | JT | as a general rule |
22:01.51 | fibs- | I have a questions regarding incoming SIP and asterisk, I have it setup (i think) and when i call my number I hear a male voice saying "testing" and then some DTMF tones |
22:01.54 | PaulTech85 | thekidrio Use the manager interface |
22:01.56 | thekidrio | mafkees: thanks will do |
22:01.59 | pigpen | n|cotine, we also have 4 port pots and the tdm2400 |
22:02.11 | thekidrio | i assume i need to out going lines right? |
22:02.17 | Corydon-w | JT: no, Digium cards favor white boxes, really |
22:02.22 | mafkees | thekidrio: yeah |
22:02.28 | thekidrio | thanks mafkees and PaulTech85 |
22:02.37 | JT | Corydon-w: hrm, well of brand names anyway |
22:02.41 | Corydon-w | JT: Dell boxes have their own set of problems, because they like to hard assign slots to IRQs |
22:02.42 | *** join/#asterisk stack_ (n=stack@63.239.190.203) |
22:02.44 | JT | i hear dell gets the best zttest results |
22:02.47 | fibs- | Does a male voice saying "testing" and then some tones sound familiar to anyone? |
22:02.57 | thekidrio | Dell power edges are a pain in the arse |
22:03.07 | JT | thekidrio: in what respect? |
22:03.09 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
22:03.13 | thekidrio | i have had 3 power edge machines fully see my digium card |
22:03.14 | Corydon-w | JT: we get better results with zttest with white boxes with Intel motherboards |
22:03.38 | Vec | thekidrio : ahhh, I am getting a poweredge 2900, hope the thing works ? |
22:03.40 | thekidrio | not full see i mean |
22:03.42 | fibs- | I know asterisk is seeing the inbound sip "Looking for s in sipcall (domain 24.20.45.4) |
22:03.48 | mafkees | most dell stuff is intel based as well |
22:03.50 | JT | it's all fractions of a difference, are we talking 100% across the board? |
22:03.50 | stack_ | Our Grandstream 386 burned up today... should we go with another one is there something better for hooking a physical fax into our asterisk system? |
22:04.06 | JT | mafkees: based... but they do make their own boards |
22:04.07 | mafkees | stack_: cisco ata |
22:04.10 | thekidrio | i think its less of an issue in the newer dell power edges |
22:04.14 | mafkees | JT: indeed |
22:04.19 | bkw_ | haha |
22:04.22 | stack_ | mafkees: any specific model? |
22:04.32 | mafkees | stack_: nah, they all are great |
22:04.39 | thekidrio | my model numbers are 2650 and 4600 |
22:04.49 | fibs- | Anyone setup a Gizmo Voip with asterisk before? |
22:04.55 | thekidrio | 2 2650's 1 4600 |
22:05.01 | GaVak | Quick question: the timing errors I'm getting. Could it be because I'm using a TDM card in the same system? |
22:05.10 | mafkees | both rollouts with dell were on customer request. I would never buy dell machines myself |
22:05.15 | GaVak | could it be the tdm card eating the interrupt bandwidth. |
22:05.49 | bkw_ | actually you dont have to interrupt 1000 times a second |
22:06.05 | bkw_ | you can actually lower that by 95% |
22:06.12 | bkw_ | and accomplish the same results |
22:06.15 | bkw_ | and better performance |
22:06.18 | russellb | and increase latency. |
22:06.19 | mafkees | bkw_: I never understood that 1000 interrupts/second |
22:06.22 | bkw_ | no |
22:06.24 | russellb | yes. |
22:06.25 | bkw_ | no |
22:06.29 | bkw_ | youdo not |
22:06.36 | bkw_ | you're doing 20ms packets on the voip side |
22:06.43 | bkw_ | so why are you filling up 20ms packets 1ms ata time |
22:06.48 | bkw_ | slice that bad boy off there in 20ms chunks |
22:06.50 | mafkees | that 1000 interrupts are for timing right ? |
22:06.56 | bkw_ | muchmore efficient |
22:07.17 | GaVak | show do you change that? |
22:07.18 | mafkees | timing like meetme and zap trunks |
22:07.19 | GaVak | *how |
22:07.26 | bkw_ | russellb, 20ms doesn't kill anything.. ifit did then faxing from LA to London wouldn't work |
22:07.31 | thinwires | does anyone here use AsteriskNOW? |
22:07.34 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
22:07.36 | Corydon-w | GaVak: You don't. He's trying to start a flame war. |
22:07.40 | Corydon-w | bkw_: please stop. |
22:07.41 | bkw_ | no i'm not |
22:07.42 | bkruse | thinwires: join #asteriskNOW |
22:07.44 | GaVak | Roger that. |
22:07.45 | bkw_ | I'm stating a fact |
22:08.00 | thinwires | bkruse: it's dead as a doornail over there, no one is responding :-( |
22:08.07 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
22:08.21 | Corydon-w | You're stating something you you believe to be true, contrary to existing advice |
22:08.29 | bkw_ | no I know its true |
22:08.37 | bkw_ | I'm doing it right now on 16 T1's |
22:08.37 | Corydon-w | If you want to debate that, this is not the forum |
22:08.57 | bkw_ | its only the forum for that when you guys feel like it is.. |
22:09.12 | Corydon-w | Please take it elsewhere |
22:09.22 | bkw_ | I'm just stating that 1000 interrupts persecond is NOT a requirement |
22:09.42 | *** mode/#asterisk [+b %bkw_!*@*] by Corydon-w |
22:09.50 | Corydon-w | Please take it elsewhere |
22:10.01 | bkruse | thanks. |
22:10.30 | *** mode/#asterisk [-b %bkw_!*@*] by Corydon-w |
22:10.31 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-62-84.dsl.tul2ok.sbcglobal.net) |
22:10.46 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
22:13.45 | n|cotine | This isn't very clear in the cookbooks on voip-info - how exactly does call flow in DUNDi work? |
22:14.46 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
22:14.53 | *** join/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com) |
22:15.04 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
22:15.53 | Umaro | Hey guys.. I'm not looking for something free, or to circumvent the g729 patent, but is there a softphone I can buy that runs on linux and supports g729? |
22:16.03 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
22:16.24 | thinwires | can anyone tell me which is better? *1.4.0 or *1.2.15? |
22:16.47 | cpatry | 1.4.0 |
22:17.15 | J4k3 | testing and experimentation is the best method to know whats "best" |
22:17.26 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
22:17.30 | J4k3 | technically 1.4.0 is superior but as its fresh and new you may find a bug or an incompatability that you cannot live with |
22:17.38 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
22:18.08 | thinwires | I'm actually using AsteriskNOW atm but I can't find any support for it... |
22:18.12 | mogorman | Umaro, digium.com |
22:18.26 | mogorman | sells hardware and or software for g729 transcoding |
22:18.27 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
22:19.34 | Umaro | mogorman: yeah, but not a g729-enabled linux softphone |
22:19.46 | mogorman | oh i misread |
22:19.47 | mogorman | sorry |
22:19.52 | mogorman | asterisk can be a softphone |
22:19.54 | Corydon-w | Umaro: the text console of Asterisk can be used as a phone |
22:20.12 | Corydon-w | Using chan_oss or chan_alsa |
22:20.19 | J4k3 | or bluetooth |
22:20.38 | Corydon-w | The bluetooth stuff is not in there, yet, though |
22:20.49 | J4k3 | ah. |
22:21.11 | J4k3 | I need to work on that, maybe thats what I'll dedicate the rest of today to |
22:21.46 | flenders | hey, do you guys think the voice quality on a cisco 7905 would be the same as on a 7960? |
22:23.12 | Umaro | Corydon-w: Is that the only option? |
22:23.21 | Corydon-w | Umaro: I can't answer that, as I don't know |
22:23.32 | Corydon-w | It's certainly one option |
22:26.39 | Umaro | Corydon-w: yeah, just not sure how feasible it is to have 50 call center agents running asterisk on their workstations |
22:26.56 | Corydon-w | Umaro: A quick google search reveals that it's not the only option |
22:27.01 | Corydon-w | http://voxilla.com/voxilla-stories/voxilla-stories/free-linux-soft-phone-released-397.html |
22:27.07 | CrashHD | anyone have a problem with parking hints? |
22:27.17 | CrashHD | in 1.4 |
22:27.19 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-1fe7d0b17b740e8b) |
22:27.24 | CrashHD | mine seem to have stopped working |
22:29.01 | JT | Umaro: you might have to get a commercial softphone |
22:29.10 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
22:29.15 | JT | anyway softphones suck, is there any reason they must have a softphone? |
22:33.36 | [TK]D-Fender | * != softphone. Holy Jeebus |
22:34.16 | cpatry | JT: cause some compagnies dont want to buy all hardware phones? |
22:34.56 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
22:35.07 | JT | cpatry: is there a good reason or are they just being stingey? |
22:35.29 | JT | are the softphones connecting to asterisk? if so, why on earth do they need g.729? |
22:35.31 | cpatry | they probably too cheap, but i understand that. |
22:36.15 | thinwires | cheap only buy's cheap :-) |
22:36.31 | fibs- | I'm trying to do a EXEC DIAL SIP/user:secret@siphost/numbertocall, and I'm getting siphost/numbertocall hostname invalid |
22:36.40 | fibs- | which means its trying to resolve the hostname and the number im trying to dial out |
22:36.44 | fibs- | any ideas folks? |
22:37.32 | cpatry | when compiling zaptel: i see: checking for ZT_TONE_DTMF_BASE in zaptel.h... yes but theres no app_meetme.so after the make, bug? |
22:37.36 | fibs- | so basically: No such host: proxy01.sipphone.com/<phonenumber> |
22:38.26 | Umaro | JT: I don't mind buying a commercial softphone, I just can't find one that runs on linux and supports g729 |
22:38.42 | JT | that's because the codec is commercial |
22:39.04 | Umaro | right, but I don't mind paying for the softphone, AND paying for the g729 codec |
22:39.11 | Umaro | it just has to run on linux and do g729 |
22:39.29 | JT | Umaro: so the question is, if you have an asterisk server on your lan, why is it necessary to use g.729 on the softphones? |
22:39.33 | JT | not sure if any are available |
22:40.09 | Umaro | JT: I'm using g729 with the voip provider, and don't want asterisk to transcode |
22:40.22 | JT | it's the easiest option |
22:41.03 | JT | and if you do internal calling too, the staff will appreciate the superior voice quality |
22:41.24 | thekidrio | anyone have a good guide on how to get two outbound calls to be patched together? |
22:41.39 | cpatry | umaro: xten doesnt support 729? |
22:41.56 | thekidrio | i think i need to use the call manager or the fop but i am too noobish to know for certain |
22:42.36 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
22:43.09 | thekidrio | any ideas? |
22:44.58 | JT | cpatry: do they have a linux release? |
22:45.08 | *** part/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
22:45.15 | cpatry | JT: yeah |
22:45.22 | thekidrio | even just the correct terminology to search on how to connect two active outbound calls would help |
22:45.37 | JT | thekidrio: either the manager interface or .call files |
22:45.51 | cervi | <PROTECTED> |
22:45.52 | JT | thekidrio: umm, are the calls pre existing, or being set up when they require patching? |
22:46.16 | cervi | How can I start in the opposite order? |
22:46.20 | cpatry | http://www.xten.com/index.php?menu=download |
22:46.36 | thekidrio | jt, i would like the calls set up when they require patching, i am hoping to eventually use a click to call with the following method |
22:47.15 | JT | ok, so when you click, both ends are called? |
22:47.21 | thekidrio | user clicks on a link, it takes them to form where put in the number they can be reached at, after that the system calls the number attached to the link they clicked and connects them both |
22:47.25 | thekidrio | jt, yeah |
22:47.41 | e-milio | hello all |
22:47.51 | thekidrio | i am fairly certain i can get the php agi working |
22:48.03 | pigpen | I may have a DID issue. I have 400 did's. I am only getting the CID number passed to the DID's, however I am getting the entire CID (name and number) passed to our primary, published numbers. |
22:48.04 | thekidrio | just not sure the syntax to connect them |
22:48.16 | pigpen | Could asterisk be stripping this, or more likely the Telco? |
22:48.20 | e-milio | Is there any actual difference in the asterisk on AsteriskNow than just asterisk ?? |
22:48.36 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
22:48.39 | Corydon-w | e-milio: no |
22:48.54 | flenders | hey, if I change settings on zapata.conf, does a reload work? or I need to restart asterisk? |
22:49.02 | JT | thekidrio: asterisk manager interface or .callfiles |
22:49.03 | thekidrio | i could probably just write the .call files, but i don't know the syntax to get the two outbound routes to talk to each other |
22:49.09 | thekidrio | ok cool |
22:49.11 | cpatry | flenders: need to restart it. |
22:49.15 | Corydon-w | e-milio: they're built out of exactly the same source tree |
22:49.19 | e-milio | Corydon-w: For a callcenter enviroment IYHO it is the same ? |
22:49.23 | thekidrio | i will just read up on .call files as that seems the most simple solution |
22:49.25 | JT | thekidrio: there's a sample call file in the docs |
22:49.29 | JT | of the source |
22:49.41 | JT | and the book mentions them too |
22:49.45 | thekidrio | jt, great hopefully i can just template that |
22:49.53 | thekidrio | yeah i am reading the future of telephony atm heh |
22:49.58 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
22:50.06 | JT | heh, my hardcopy is shipping |
22:50.21 | JT | the second edition will be out in a couple of months |
22:50.31 | thekidrio | yeah i can't wait for that |
22:50.35 | thekidrio | mine is filled with update notes haha |
22:50.52 | JT | i use it as a reference at the moment |
22:51.52 | *** join/#asterisk flipdesk (n=flip@ip68-2-210-190.ph.ph.cox.net) |
22:54.49 | e-milio | Corydon-w:thanks |
22:55.23 | Umaro | cpatry: xten doesn't have a linux client |
22:55.32 | cpatry | for x-lite yeah. |
22:56.47 | flipdesk | so yeah, here's an interesting one |
22:57.00 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
22:57.01 | flipdesk | once in awhile, a phone will register fine |
22:57.13 | JT | Umaro: x-lite doesn't have g.729 anyway |
22:57.16 | flipdesk | passwords are fine, sip show peers shows the phone registered |
22:57.29 | flipdesk | but when the phone tries to make a call I'll get in the logs: |
22:57.40 | cpatry | when compiling zaptel: i see: checking for ZT_TONE_DTMF_BASE in zaptel.h... yes but theres no app_meetme.so after the make, bug? |
22:57.53 | flipdesk | Feb 20 15:43:34 WARNING[21087] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"That Guy" <sip:5605@67.139.179.230>;tag=as6c3eccf5' |
22:58.06 | flipdesk | with sip:5605 being the extension that is making the call |
22:58.07 | GaVak | If I install a new zaptel version, do I have to remake asterisk? |
22:58.30 | flipdesk | phone is registered though |
22:58.32 | flipdesk | weird eh? |
22:59.05 | cpatry | GaVak: yes |
22:59.09 | GaVak | thnx |
23:00.28 | flipdesk | anyone have any ideas? |
23:03.20 | flipdesk | is there a way to get the current state of a device? |
23:03.24 | flipdesk | from the CLI |
23:05.06 | *** join/#asterisk MaartenB (n=Maarten@84-105-196-31.cable.quicknet.nl) |
23:05.13 | MaartenB | hello everyone |
23:05.49 | MaartenB | I have a question about setting the outgoing callerid correctly with CALLERID(num) |
23:06.01 | MaartenB | I want to do this based on the sip account, but I have no idea how to do it right |
23:06.26 | EmleyMoor | MaartenB: Hold on, I've done that |
23:06.36 | MaartenB | great |
23:07.24 | EmleyMoor | Just put callerid="CallerID name" <CallerID number> in the relevant entry in sip.conf |
23:07.58 | MaartenB | ok, the problem is that sometimes, a call is being forwarded |
23:08.01 | flipdesk | i think he wants to use the callerid command |
23:08.07 | MaartenB | and I have to set the callerid in that cases too |
23:08.11 | flipdesk | nod |
23:08.25 | EmleyMoor | Ah, that's easy too |
23:08.47 | MaartenB | I like easy things :) |
23:09.01 | EmleyMoor | Set(CALLERID(num)=number) |
23:09.20 | MaartenB | yes, I know that |
23:09.36 | EmleyMoor | Use GotoIf or ExecIf commands to execute it conditionally |
23:09.41 | MaartenB | ah, ok |
23:10.15 | MaartenB | then I think my question is, how can I test with GotoIf or ExecIf if it is, or it is not a call initiated from the sip accounts |
23:10.18 | EmleyMoor | You'll probably have some CUTting to do to get useful information about the channel |
23:11.09 | EmleyMoor | Set(TECHNOLOGY=${CUT(CHANNEL,/,1)}) |
23:11.25 | MaartenB | ok, thanks |
23:11.26 | MaartenB | trying... |
23:11.29 | EmleyMoor | You can then check if that is SIP |
23:11.46 | EmleyMoor | If you need to check more specifically, you can do similar things |
23:12.04 | JT | MaartenB: couldn't you just use different contexts for different technologies, and set a variable? |
23:12.25 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
23:13.01 | flipdesk | wrong password on authentication for INVITE |
23:13.05 | flipdesk | man that's odd |
23:13.18 | JT | seen it before |
23:13.22 | flipdesk | how could the INVITE password be different from the REGISTER password? |
23:13.23 | JT | can't remember what caused it |
23:13.35 | JT | it might be an asterisk or provider bug |
23:13.41 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
23:13.52 | flipdesk | yeah, I found a bug report |
23:14.02 | flipdesk | but it couldn't be reproduced |
23:14.11 | JT | it occurs randomly |
23:14.16 | flipdesk | very |
23:14.54 | flipdesk | I bet if I restarted asterisk, it would go away. At least for that extension |
23:15.10 | JT | maybe |
23:15.15 | JT | or just chan_sip |
23:15.27 | flipdesk | good point |
23:15.55 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
23:16.08 | flipdesk | googling has turned up nothing productive :-/ |
23:20.37 | *** join/#asterisk pingboy12 (n=idatoo@CPE0004e24967bd-CM001692fb03c2.cpe.net.cable.rogers.com) |
23:21.07 | *** join/#asterisk X-Rob (n=rob-x@CPE-58-169-113-201.vic.bigpond.net.au) |
23:22.00 | mercestes | ok, when I forward a voicemail to another user, the emailed voicemail notification says that the duration is 0.00. Is there a way to fix this behavior?? |
23:22.32 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
23:23.38 | pigpen | If I am getting the number of the CID, and knowing that I have no special CID rewrite commands in my dialplan, would you assume that my Telco is not passing the CID Name ? |
23:23.38 | pingboy12 | having probs compiling asterisk 1.4 |
23:24.24 | pingboy12 | i get makeopts error, when doing make install |
23:25.16 | pingboy12 | says configure script must be run, before running make |
23:25.20 | pingboy12 | but i did that |
23:26.21 | ChicagoBud | pingboy12, was it successful? |
23:26.32 | ChicagoBud | pingboy12, the ./configure |
23:28.47 | ChicagoBud | has anyone built app_txfax.c and app_rxfax.c for * 1.4??? How did you do it since the build process is so different??? |
23:29.04 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
23:32.52 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
23:33.18 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
23:35.08 | Dovid | morning all |
23:35.41 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
23:38.58 | ManxPower | pigpen: on PRi or FXO? |
23:39.08 | *** join/#asterisk backblue (n=moo@87-196-109-88.net.novis.pt) |
23:39.23 | pigpen | pri. |
23:39.28 | pigpen | dam goofy. |
23:39.35 | pigpen | I have done some more testing: |
23:39.39 | ManxPower | pigpen: PRI name arrives as a message AFTER call setup |
23:39.55 | pigpen | err....continue. |
23:39.57 | ManxPower | so if you put a Wait(.5) as the first priority of whatever extens match the incoming calls, then it should wokr |
23:39.59 | pigpen | or explain... |
23:40.09 | ManxPower | Wait(.25) might work. It comes in pretty fast. |
23:40.17 | pigpen | k. |
23:40.24 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
23:40.27 | pigpen | Well, just in case...this is what I tested. |
23:40.38 | pigpen | I dial the main number, type in the exten...works. |
23:41.03 | pigpen | I redirect the main number to "Dial(IAX2/6333,20,twW)...no dice.... |
23:41.27 | pigpen | but..typically, yes...the main number goes through...well..alot of stuff. |
23:41.33 | pigpen | k..adding it now. |
23:42.47 | pigpen | Ha!. |
23:42.53 | *** join/#asterisk welby (n=welby@ivonova.whmcr.com) |
23:43.06 | pigpen | Too dam easy..thanks for the years you have been helping me. |
23:43.20 | ManxPower | feel free to send money to eric@fnords.org via PayPal |
23:43.22 | pigpen | You Qwell and many others. |
23:43.41 | pigpen | :) |
23:44.37 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
23:45.45 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
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23:47.34 | luke-jr_work | file: did 8821 re-break 7051? |
23:47.48 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
23:47.49 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
23:48.19 | blitzrage | file did it |
23:48.28 | file | luke-jr_work: I knew it would creep up when I put 7051 in, I knew it |
23:48.55 | luke-jr_work | file: 8821's configuration is broken |
23:49.16 | ManxPower | I suppose if I filed a remote on bugs.digium.com asking to remove a config option because it is not documented and nobody knows what it does would be closed? |
23:49.21 | luke-jr_work | file: the fix for 7051 is necessary to not break my setup |
23:49.39 | ManxPower | So, when is kram going to come back to IRC? |
23:50.13 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.128.38) |
23:50.20 | CrashHD | hints are not working in 1.4, any suggestions? |
23:51.40 | file | luke-jr_work: an option, that's the only way I can keep both of you happy |
23:52.01 | luke-jr_work | file: oh, there's an option now? works for me, I guess |
23:52.09 | file | no, I mean I'm going to have to code one now |
23:52.15 | luke-jr_work | oh |
23:52.23 | file | you need externip to be compared against localnet, he needs it not |
23:52.29 | luke-jr_work | or he could set localnet to include his local IP |
23:52.43 | *** part/#asterisk tuxd00d (n=tuxd00d@128.187.128.38) |
23:53.21 | ManxPower | CrashHD: remove the call-limit option from your config |
23:54.05 | CrashHD | manx none of my sip entries have a call limit |
23:54.20 | CrashHD | and show hints actually displays everything as unavabile |
23:54.26 | CrashHD | (unavailable) |
23:54.39 | CrashHD | where as working configs in 1.2 show those entries as idle |
23:54.53 | flenders | is there a huge difference on txgain if I change it from -6 to 4 on zapata.conf? |
23:55.18 | ManxPower | flenders: yes. it's a logarithmic scale |
23:56.45 | CrashHD | any other ideas ManxPower? |
23:57.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
23:57.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
23:58.39 | ManxPower | CrashHD: you know my policy on 1.4, right? |
23:58.55 | CrashHD | heh, I don't, but I have a feeling your about to enlighten me |
23:59.13 | blitzrage | hockey time!!! GO LEAFS GO! |
23:59.24 | ManxPower | CrashHD: If digium doesn't run 1.4 on their corporate PBX then I'm not going to consider running it on my client's corporate pbxs |
23:59.32 | ManxPower | and I've seen no indication that digium is doing so. |
23:59.43 | ManxPower | 1.4 == 1.4 released version |