| 00:00.04 | ManxPower | bartpbx, Maybe it understands SIP and is screwing it up. |
| 00:00.09 | justinu | bartpbx: maybe the token fell out of the ethernet... |
| 00:00.22 | SplasPood | ambriento: if you read what my problem is... |
| 00:00.29 | justinu | bartpbx: do you have 30 pin simms, or 72 pin simms? |
| 00:00.35 | ManxPower | riddlebox, you mean like exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 00:00.44 | kratzers | just DND button on the phones (which does make regular calls priority jump 100), and using agents |
| 00:01.12 | ManxPower | kratzers, since the call is just a call to the phone, you will not see any difference. |
| 00:01.24 | bartpbx | justin, haha |
| 00:01.30 | niteowloz | using realtime in 1.0.7, how do I register with a sip provider |
| 00:01.31 | riddlebox | ManxPower, I mean that if they just dial 10 digits, that means the did not dial the 1 in front and to just add the 1? |
| 00:01.33 | kratzers | why would it consider a phone on DND eligible to take a queued call? |
| 00:01.43 | SplasPood | Manx: did that, callerid=UNAVAILABLE, same deal.. sends the numbers from the [context] as the CID |
| 00:02.14 | ManxPower | riddlebox, that's what my line does, assuming you have a pattern match for 1nxxnxxxxxx |
| 00:02.24 | ManxPower | You don't want unneeded Dial lines. |
| 00:02.27 | riddlebox | ohh sorry |
| 00:02.50 | bartpbx | ok, i found the informations about the new router. It is a linksys WRT54G with original firmware. This one does not understand sip as far as i know |
| 00:03.09 | litage | without having a FWD account, can Asterisk redirect calls to a FWD user? |
| 00:03.14 | ManxPower | kratzers, you know that the next release of Asterisk won't support jumping to proirity+101, right? |
| 00:03.24 | ManxPower | litage, sure! |
| 00:03.30 | kratzers | oh? what instead? or is there somewhere to read about it? |
| 00:03.32 | bartpbx | all port forwards are added again.. |
| 00:03.33 | ManxPower | litage, I doubt FWD will accept the call. |
| 00:04.03 | litage | ManxPower: ah, so Asterisk would need to be registered with FWD for FWD to accept the call to one of its users? |
| 00:04.09 | ManxPower | kratzers, "show application dial" will tell you tyhe variables that are set. the extensions.conf.sample in 1.2.x shows you examples |
| 00:04.21 | ManxPower | litage, I have no idea. that's a question for the FWD people. |
| 00:04.29 | litage | thanks ManxPower |
| 00:04.33 | *** join/#asterisk AJay-MN (i=AJay@63.231.252.9) |
| 00:04.40 | ManxPower | Regardless, Asterisk will happily send the call anywhere you wany. |
| 00:04.42 | ManxPower | want |
| 00:04.47 | litage | heh true |
| 00:05.08 | AJay-MN | If you have phones set to Resistration Experation set, should you see the phone re-register on the console? |
| 00:05.49 | ManxPower | AJay-MN, at some debug levels, yes, IIRc |
| 00:05.51 | litage | if *A sends a call to *B and neither *A nor *B "know" of each other (IE: they aren't registered with each other), will *B accept the call from *A? |
| 00:06.10 | ManxPower | litage, do you understand what registration does? |
| 00:06.11 | riddlebox | ManxPower, that was too easy I should have figured that.... |
| 00:06.16 | ManxPower | It's looking like you dont. |
| 00:06.23 | litage | ManxPower: not entirely |
| 00:06.45 | ManxPower | riddlebox, I have several "those morons can't dial the phone" entries in my dial plan |
| 00:07.05 | ManxPower | litage, registration tells the remote server what ip address a specific user/password is located at. |
| 00:07.07 | ManxPower | it does NOTHING else. |
| 00:07.08 | AJay-MN | well i see the devices do there init registration, but how can i see them reregister? i need to know if they are or not. seems my grandstream is no longer reregistering and after Asterisk's 1 hour unregistering i dont get calls in |
| 00:07.48 | ManxPower | AJay-MN, I've never had to set the registraiton interval in any of the 5 or 6 brands of phones I've used. |
| 00:08.27 | Grizzy | Does insecure=very cause asterisk to ignore context= and go to [default] ? |
| 00:08.30 | ManxPower | But I do seem to recall Grandstreams just stopped working randomly (before we threw out the test phone we bought) |
| 00:08.35 | AJay-MN | ManxPower : My Zyxel P2000w's work fine. but after 1 hour, i loss registration on my Grandstream 101... :( |
| 00:08.47 | ManxPower | Grizzy, I believe it just says "accept any call" |
| 00:08.56 | ManxPower | AJay-MN, using NAT? |
| 00:09.16 | litage | ManxPower: if userA registers with asteriskA, and a softphone that isn't registered with/connected to asteriskA dials <sip:userA@asteriskA_ip:5060>, what will happen? |
| 00:09.18 | AJay-MN | no. has a full qualifying IP.. |
| 00:09.41 | riddlebox | ManxPower, actually it is for the wakeup call agi script that is on nerd vittles, it works off of caller id, and when you do it from your cell phone or anywhere else it just tries to call that exact number back |
| 00:10.00 | ManxPower | litage, absolutily nothing different, since all registration does is notify the remote server where to send calls. |
| 00:10.38 | litage | ManxPower: so asteriskA will route the call to userA and the softphone and userA will be able to talk? |
| 00:10.41 | Grizzy | What would be causing my ipkall paragraph to ignore context= ? |
| 00:11.18 | bartpbx | ok, i give up. I can't find the error and it is 1:10 i'll continue tomorrow. Thank you guys for the help good n8 |
| 00:11.25 | Grizzy | I mean, I can make it work, it's just odd. |
| 00:11.59 | ManxPower | litage, it will work |
| 00:12.13 | ManxPower | since user A registered with Asterisk, so Asterisk knows what IP address to send the call to. |
| 00:12.28 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 00:12.41 | ManxPower | litage, You might not know this, but if the device is on a static IP address, there is no reason whatsoever to register to the server. |
| 00:13.11 | litage | ManxPower: yes, i realize this. thanks |
| 00:14.05 | tehdely | does anyone here know where I can get a tucson-area DID? |
| 00:14.07 | tehdely | pm me if you do |
| 00:14.08 | tehdely | thanks! |
| 00:14.11 | ManxPower | Grizzy, the incoming call is not matching anything in sip.conf |
| 00:16.00 | niteowloz | can anybody help me with registering to a sip provider when using realtime? |
| 00:16.23 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 00:19.03 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 00:19.48 | MattB2 | hi all |
| 00:20.17 | MattB2 | anyone got experience in running an asterisk server at an ISP to handle 50+ concurrent calls, and has 5 mins to spare? |
| 00:21.51 | SGM | good |
| 00:22.00 | SGM | the sip reinvite is working |
| 00:22.01 | SGM | :))) |
| 00:22.01 | Grizzy | Manx - thanks. The article on voip-info seems to say that the sip.conf paragraph should be named by the phone number, is that right? |
| 00:22.05 | _Sam-- | how is 1.2.5 working? |
| 00:22.23 | SGM | even with "t" option |
| 00:24.09 | Grizzy | I saw a "cheat sheet" page on one of the documentation/help sites, now I can't find it again. It had a big list of all the directives in Asterisk. Anyone remember it? |
| 00:26.46 | riddlebox | is anyone using a sipura 2100? |
| 00:27.38 | *** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 00:27.46 | Qwell[] | Grizzy: "directives"? |
| 00:31.25 | Grizzy | quell - functions, directives, what ever you call 'em. (every computer concept has at least 3 names) |
| 00:31.45 | Qwell[] | applications? |
| 00:31.47 | Grizzy | like Goto Wait Answer Playtones |
| 00:31.52 | Qwell[] | show applications |
| 00:31.55 | Qwell[] | on the CLI |
| 00:32.20 | dja_ | Hi, I sometimes get "chan_sip.c: sip_xmit of 0x97747b8 (len 768) to 192.168.1.220:5060 returned -1: Operation not permitted" when dialing an extension (which never rings, even though my extension is playing the "ring"). Sometimes it works just fine. Help? |
| 00:32.23 | Grizzy | There was a nice list on one of the documentation sites that I can't find again |
| 00:32.54 | Qwell[] | show applications |
| 00:33.11 | Grizzy | It had some explanation for each one, as well. |
| 00:33.19 | Qwell[] | funky res |
| 00:33.20 | Qwell[] | erm |
| 00:33.22 | Qwell[] | show applications |
| 00:34.10 | Grizzy | in particular, argument lists |
| 00:34.27 | Qwell[] | show application blah |
| 00:34.29 | Grizzy | show applications isn't bad, thanks for that. |
| 00:35.18 | Grizzy | Is there a way to run an rc (startup script) when you start asterisk? I'm tired of typing "sip debug" |
| 00:35.54 | niteowloz | Hi guys, anyone got 5 mins to help me with realtime on 1.0.7 |
| 00:35.56 | Qwell[] | no, but that isn't actually a bad idea |
| 00:36.03 | Qwell[] | niteowloz: Nope. Upgrade |
| 00:37.01 | Qwell[] | niteowloz: There are tons of bugs and missing features in 1.0.7. It isn't worth our time, or yours, trying to get it working... |
| 00:37.55 | Grizzy | So, I gather that I have to either use "asterisk -rC reload" or a database, to make it possible to add or change something while asterisk is running? |
| 00:38.18 | Qwell[] | Grizzy: Pretty much |
| 00:38.22 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 00:38.24 | Grizzy | If I want to do my own web interface for adding new users. |
| 00:38.37 | Grizzy | Quell - good, thanks. |
| 00:40.52 | niteowloz | Qwell, tx, mostly working OK for me so far. Just can't register with sip provider. Lots of work to upgrade.....but I will have to bite the bullet sooner or later I know.. |
| 00:43.03 | Grizzy | Grumble: glophone seems to have lost it's mind. |
| 00:43.14 | dja_ | how can I tell what codec is being used for a particular connection? my provider says they switched me over to G.729A, but I don't think they did. :) |
| 00:43.44 | niteowloz | try sip show channels while the call is in progress |
| 00:44.31 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 00:45.21 | dja_ | I'm guessing "Form" is the codec? It says ulaw, so I'm also guessing that they didn't switch me. :) |
| 00:45.52 | Qwell[] | dja_: disallow=all, allow=g729 |
| 00:46.29 | Qwell[] | most providers allow you to use multiple codecs. You have to explicitly tell * to use one |
| 00:47.57 | Ukyo | hey Qwell, whats u |
| 00:47.57 | Ukyo | p |
| 00:49.33 | *** join/#asterisk CrashHD (n=timf@c-24-7-168-46.hsd1.ca.comcast.net) |
| 00:50.33 | CrashHD | hello |
| 00:50.41 | CrashHD | how can I troubleshoot sip call quality |
| 00:50.48 | CrashHD | we have 12 trunks coming from a voip provider |
| 00:50.53 | CrashHD | the calls seem to be choppy |
| 00:51.15 | CrashHD | what can I do in the asterisk to determine what is going on? |
| 00:51.29 | ambriento | hey CrashHD, what's up? |
| 00:51.36 | CrashHD | hello ambriento |
| 00:51.51 | ambriento | how's the altigen stuff? |
| 00:51.53 | dja_ | Qwell[]: thanks, I tried that but I get "all circuits are busy now" -- I'm pretty sure my provider told me that they only supported ulaw or G729a, but not both. :( I'll have to complain to them again. |
| 00:51.57 | CrashHD | what is up, is I'm under the gun to figure out this sip call quality |
| 00:52.01 | CrashHD | altigen stuff is goign ok |
| 00:52.18 | CrashHD | the problem now is just the above call on the trunks from asterisk to our voip provider |
| 00:52.27 | CrashHD | *above call quality mention |
| 00:52.34 | Qwell[] | CrashHD: You sure it isn't the provider? |
| 00:52.38 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:52.39 | Qwell[] | or bandwidth issues? |
| 00:52.50 | CrashHD | not sure Qwell |
| 00:52.59 | CrashHD | we have 100mbit to the internet |
| 00:53.04 | CrashHD | so bandwidth isn't an issue |
| 00:53.12 | CrashHD | unless there is a bottle neck on our providers network |
| 00:53.14 | Qwell[] | not true.. |
| 00:53.15 | CrashHD | which I highly doubt |
| 00:53.28 | Qwell[] | when you go over the public internet...you deal with the public internet |
| 00:53.32 | ambriento | 100Mbit dedicated to VoIP? |
| 00:53.50 | ambriento | or its for everything? |
| 00:53.55 | CrashHD | <<ccnp |
| 00:53.59 | CrashHD | understand the public internet |
| 00:54.13 | CrashHD | 100mbit to the one rack we have the asterisk in |
| 00:54.21 | CrashHD | with nothing being used in that rack but the asterisk |
| 00:54.36 | CrashHD | I guess my question specifically is where are the troubleshooting tools in asterisk? |
| 00:54.45 | ambriento | sip debug? |
| 00:54.47 | CrashHD | I'd like to know how many packets are not getting from ip provider to us |
| 00:55.03 | CrashHD | tried that didn't seem to give the info I needed easily |
| 00:55.49 | CrashHD | I don't believe the rtp traffic is making it to the box from the provider |
| 00:56.06 | *** part/#asterisk Grizzy (i=Generic@ppp-69-238-229-72.dsl.pltn13.pacbell.net) |
| 00:57.28 | *** join/#asterisk Grizzy (i=Generic@ppp-69-238-229-72.dsl.pltn13.pacbell.net) |
| 01:01.24 | *** part/#asterisk T`2 (i=id@pdpc/supporter/student/T) |
| 01:01.42 | ambriento | hmm |
| 01:02.15 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 01:05.52 | CrashHD | is there a 729 codec module that can be used for testing? |
| 01:05.57 | robin_sz | anyone happen to know what the Swiss freephone numbers are ??/ 0800 and ?? |
| 01:06.00 | CrashHD | without purchasing the licensing? |
| 01:06.03 | robin_sz | 0871? |
| 01:07.04 | robin_sz | CrashHD: yes, I think there is .. I read about it on the Digium site ... but, hey its only $10 anyway and it support * ... just pay it ;) |
| 01:07.14 | CrashHD | heh |
| 01:07.19 | CrashHD | it's the 24 hour wait time |
| 01:07.23 | robin_sz | nah |
| 01:07.23 | CrashHD | I need to test this now |
| 01:07.36 | robin_sz | it happens in like 2 minutes usually |
| 01:07.40 | CrashHD | ohh |
| 01:07.41 | CrashHD | sweet |
| 01:07.47 | CrashHD | it says 24 hours on their website |
| 01:07.48 | robin_sz | it CAN take longer |
| 01:08.00 | robin_sz | but for me, its always been quick |
| 01:08.11 | robin_sz | I guess they are covering ther asses |
| 01:08.39 | robin_sz | maybe ive been lucky? |
| 01:08.48 | CrashHD | hmm |
| 01:08.48 | CrashHD | lol |
| 01:08.52 | CrashHD | just 10 bucks |
| 01:08.57 | CrashHD | I'll put it in and gamble |
| 01:09.09 | robin_sz | it goes to a good cause IMHO |
| 01:10.10 | robin_sz | anyway ... swiss freephoen prefixes? |
| 01:12.47 | CrashHD | anyone use voip reach for termination? |
| 01:14.55 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 01:15.43 | *** join/#asterisk r0n14 (n=ronia@5-233-235-201.fibertel.com.ar) |
| 01:15.51 | r0n14 | hi |
| 01:16.07 | r0n14 | <PROTECTED> |
| 01:16.19 | r0n14 | and have ast_best_codec: Don't know any of 0xf800 formats |
| 01:16.29 | r0n14 | <PROTECTED> |
| 01:17.43 | *** join/#asterisk fjean (n=fjean@201009183198.user.veloxzone.com.br) |
| 01:18.00 | CrashHD | call from fwd? |
| 01:18.12 | r0n14 | yes |
| 01:18.17 | CrashHD | elaborate? |
| 01:18.26 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 01:18.35 | fjean | hey, I had a few beers...so how would I dial an extension using Dial(IAX2/... ? Dial(IAX2/4000,60) ? |
| 01:18.50 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 01:19.00 | r0n14 | free world dial |
| 01:19.44 | [hC] | Anyone here know anything about the v8 firmware for the cisco 7970? Someone on a mailing list mentioned that it may now contain SIP support, i wanna confirm that.. |
| 01:21.10 | _Sam-- | [hC] : [av]bani said that the SIP firmware is out now |
| 01:21.12 | *** join/#asterisk iq (n=iq@71-38-79-126.omah.qwest.net) |
| 01:21.19 | iq | Hi All |
| 01:22.07 | [av]bani | [hC]: there's separate sip and sccp v8 firmwares now, and 7970 is supported |
| 01:23.06 | CrashHD | so anyway to determine how many sip/rtp packets are being lost in a call? |
| 01:23.29 | _Sam-- | CrashHD: i tool like mtr for tracing / packet loss is a good start |
| 01:23.38 | _Sam-- | s/i tool/a tool/ |
| 01:24.31 | CrashHD | what if I'm not local to the asterisk server? |
| 01:24.43 | _Sam-- | then ssh to it? |
| 01:24.44 | CrashHD | is mtr *nix based? |
| 01:24.48 | _Sam-- | they have win mtr |
| 01:24.59 | CrashHD | an nix mtr |
| 01:25.00 | CrashHD | ok |
| 01:25.02 | CrashHD | sweet |
| 01:25.11 | [hC] | [av]bani oh excellent :) |
| 01:25.24 | robin_sz | people run windows? how quaint :) |
| 01:25.26 | [hC] | [av]bani: that will be interesting to try out. |
| 01:25.44 | _Sam-- | mtr is also good because it will measure jitter too |
| 01:25.44 | [av]bani | [hC]: no idea if it even works with * yet though. ive heard BLF doesnt work at all |
| 01:26.20 | robin_sz | must be kinda tricky on a non-realtime OS to measure jitter |
| 01:26.38 | _Sam-- | ya i dont know how you would check jitter in the windows ver |
| 01:26.49 | robin_sz | quite |
| 01:27.05 | robin_sz | the 2.6 kernel has some almost-realtime features |
| 01:27.06 | CrashHD | http://www.bitwizard.nl/mtr/ |
| 01:27.08 | _Sam-- | robin_sz : did any respond to your open letter to grandstream yet? :) |
| 01:27.09 | CrashHD | is this the website? |
| 01:27.19 | robin_sz | mmm ... nah. |
| 01:27.23 | *** part/#asterisk fjean (n=fjean@201009183198.user.veloxzone.com.br) |
| 01:27.53 | CrashHD | can you direct me at a website for mtr? |
| 01:28.11 | _Sam-- | the website you pasted is the mtr site |
| 01:28.15 | CrashHD | ok thank you |
| 01:29.17 | robin_sz | right bedtime ... time to prepare for tomorow ... |
| 01:29.31 | robin_sz | another day playing with multi-kilowatt lasers :) |
| 01:29.54 | _Sam-- | [av]bani : i figured out why i couldnt set my caller id with the CLEC....i figured out i can only set the caller ID to the a number on their PRI |
| 01:30.04 | _Sam-- | it didnt even cross my mind to check that |
| 01:30.09 | _Sam-- | then i tried that once, and it worked |
| 01:30.33 | robin_sz | "CAUTION: high power laser, do not stare into beam with remaining eye!" |
| 01:30.35 | _Sam-- | not quite as nice being able to set it to any caller id in the entire word |
| 01:34.22 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:34.25 | x86 | hmm |
| 01:34.46 | CrashHD | hey sam the connection from one to the other looks good using mtr what would be your next step? |
| 01:34.47 | x86 | for some reason, all of my calls (even local SIP to SIP) are forcing the caller ID of my outbound PSTN number |
| 01:35.07 | CrashHD | you check your sip.conf for callerid=? |
| 01:35.18 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 01:35.49 | x86 | ah |
| 01:35.56 | x86 | it uses what's in the <> heh |
| 01:36.00 | CrashHD | :) |
| 01:36.47 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 01:38.18 | x86 | hmm |
| 01:38.44 | x86 | why when i do VoiceMailMain(${CALLERIDNUM}) it's still asking for mailbox number when i call voicemail from my sip extension? |
| 01:39.20 | x86 | in CLI, it shows it's executing VoiceMailMain("SIP/100-34f2","100") like I thought it should be... |
| 01:42.05 | ambriento | x86, can you see that vmuser with show voicemail users ? |
| 01:42.06 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 01:43.23 | *** join/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 01:43.56 | ambriento | CLI >show voicemail users |
| 01:44.17 | x86 | right |
| 01:44.24 | x86 | Mbox is shown as the extension |
| 01:44.27 | x86 | so that matches |
| 01:44.38 | x86 | is that what i'm looking for? |
| 01:48.25 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:49.35 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 01:53.25 | ambriento | thats weirdo x86 |
| 01:53.43 | ambriento | do u have another vmuser to test it? |
| 02:00.01 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 02:01.25 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:12.45 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 02:14.46 | *** join/#asterisk AJay-MN (i=AJay@63.231.252.9) |
| 02:25.28 | *** join/#asterisk izo (n=izo@cuscon14299.tstt.net.tt) |
| 02:25.38 | izo | anybody with a mac here ? |
| 02:26.46 | *** part/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 02:29.47 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 02:30.33 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-169-105.nwrk.east.verizon.net) |
| 02:31.18 | *** part/#asterisk izo (n=izo@cuscon14299.tstt.net.tt) |
| 02:31.54 | VxJasonxV | Has anybody made a 'listen' or 'dummy' SIP client? |
| 02:32.03 | VxJasonxV | i.e. one for listening/event purposes that isn't a phone? |
| 02:35.41 | Deep6 | guys shouldn't zap show channels show my incoming zaptel x100p line? |
| 02:36.12 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-91-191.cybersurf.com) |
| 02:41.02 | CrashHD | what is the feature in asterisk called that eliminates telemarketer calls? |
| 02:41.10 | Qwell | zapateller? |
| 02:41.17 | Abydos313 | haha |
| 02:41.23 | CrashHD | it work any good? |
| 02:41.36 | Qwell | sure |
| 02:41.58 | CrashHD | does the call have to be picked up by the * system for it to work? |
| 02:42.12 | CrashHD | or can it do what it does in a * trunking situation |
| 02:42.39 | *** join/#asterisk xevo (n=bob@c-67-182-205-227.hsd1.ut.comcast.net) |
| 02:42.52 | CrashHD | where the call is coming to * then being passed right back out to another system |
| 02:48.21 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 02:48.59 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 02:49.12 | CrashHD | how could I do something in a macro like: if ${EXTEN} = ${THIS} DO THIS ELSE NULL |
| 02:49.13 | CrashHD | ? |
| 02:49.16 | *** join/#asterisk simulated (n=joker@71.196.10.2) |
| 02:50.29 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:51.36 | CrashHD | anyone have an example of execif with an expression etc...? |
| 02:51.43 | russellb | CrashHD: show application GotoIf |
| 02:52.03 | CrashHD | oh duh |
| 02:52.05 | CrashHD | thanks |
| 02:52.54 | CrashHD | any special considerations for an expression? |
| 02:53.12 | CrashHD | can I just do ${EXTEN} = "1111111111" |
| 02:53.13 | CrashHD | ? |
| 02:53.15 | russellb | well, you should probably read about expressions in asterisk ... |
| 02:53.16 | russellb | no |
| 02:53.17 | russellb | :) |
| 02:53.27 | Qwell | README.variables |
| 02:53.31 | CrashHD | ok |
| 02:53.32 | CrashHD | hitting it now |
| 02:53.33 | CrashHD | thanks |
| 02:53.33 | Qwell | explains it quite well |
| 02:53.39 | Qwell | russellb: omg hi |
| 02:53.39 | russellb | there you go :) |
| 02:53.44 | russellb | Qwell: !!!!!!!!!!! |
| 02:53.47 | Qwell | !!! |
| 02:53.54 | CrashHD | thanks russellb, and Qwell |
| 02:54.08 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-91-191.cybersurf.com) |
| 02:55.11 | CrashHD | ahh so when the docs say use <expres> it means use $[<expres>] correct? |
| 02:55.28 | CrashHD | Usage: ExecIF (<expr>|<app>|<data>) |
| 02:55.39 | russellb | yes |
| 02:55.43 | russellb | actually ... |
| 02:55.55 | Qwell | expressions aren't always $[] though |
| 02:55.56 | russellb | that's not quite true. |
| 02:55.59 | CrashHD | so ExecIF ($[whatever = whatever]|zapateller) |
| 02:56.01 | Qwell | 1 is a valid expression |
| 02:56.16 | *** join/#asterisk nayyares (n=Nayyar@58.65.151.218) |
| 02:56.25 | CrashHD | so <expr> they are looking for 0 or a non 0 value |
| 02:56.30 | nayyares | hi guys.....! |
| 02:56.30 | CrashHD | and $[ ] will get me that |
| 02:56.42 | Qwell | right |
| 02:56.45 | CrashHD | ok |
| 02:56.47 | CrashHD | sweet |
| 02:56.49 | CrashHD | easy enough |
| 02:58.18 | russellb | basically, 0 or nothing are false |
| 02:58.31 | russellb | and anything else will be true |
| 02:58.46 | russellb | and $[ ] expressions make it easy to evaluate to one of those :) |
| 02:59.08 | russellb | which you already figured out |
| 02:59.20 | russellb | but I had to check exactly what it was in the code to satisfy myself :-p |
| 02:59.25 | CrashHD | heh |
| 02:59.33 | CrashHD | nothing wrong with not assuming anything |
| 03:01.16 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:03.23 | CrashHD | I'm seeing a: Mar 7 14:04:25 WARNING[1884]: ast_expr2.y:843 op_div: non-numeric argument |
| 03:03.27 | CrashHD | what does this mean? |
| 03:04.30 | CrashHD | exten => s,n,ExecIf($[${ARG1} = 6023571570]|Zapateller) |
| 03:04.33 | CrashHD | has to do with that line |
| 03:04.49 | CrashHD | ARG1 being the ${EXTEN} that was passed to the macro I'm in |
| 03:05.29 | CrashHD | nm |
| 03:10.29 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 03:11.44 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 03:11.45 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 03:13.08 | CrashHD | anyone have a better example/explanation of fieldspec fo cut()? |
| 03:14.38 | CrashHD | nevermind |
| 03:14.41 | CrashHD | lol |
| 03:14.55 | CrashHD | everyone is sitting there going |
| 03:15.04 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 03:15.06 | CrashHD | you are nuts dude, ask a question? find your own damn answer |
| 03:15.23 | iq | CrashHD, haan - what ;) |
| 03:15.53 | CrashHD | lol |
| 03:16.20 | CrashHD | anyway to have a zapateller like function without the sound actually being heard by the caller? out of band or something? |
| 03:16.52 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-43-47.cybersurf.com) |
| 03:17.05 | wundaboy | what does Peer '101' is trying to register, but not configured as host=dynamic mean? |
| 03:17.09 | *** part/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 03:17.15 | wundaboy | and how do i change it on a polycom? |
| 03:18.06 | CrashHD | well |
| 03:18.19 | CrashHD | if the polycom will have a dynamic ip |
| 03:18.25 | CrashHD | just set host = dynamic |
| 03:18.29 | CrashHD | under the 101 context |
| 03:18.31 | CrashHD | in sip.conf |
| 03:18.35 | wundaboy | oh |
| 03:18.37 | wundaboy | well |
| 03:18.43 | [TK]D-Fender | wundaboy : you put host = (somthing other than dynamic) and the IP doesn't match |
| 03:18.46 | wundaboy | its dhcp, but statically assigned |
| 03:18.56 | CrashHD | just put host = IP NUMBER HERE |
| 03:18.59 | [TK]D-Fender | wundaboy : use host=dynaimc for the sip.conf entry |
| 03:19.05 | wundaboy | roger |
| 03:19.10 | [TK]D-Fender | do NOT bother with the IP... |
| 03:19.21 | wundaboy | it dosent matter, does it |
| 03:20.06 | wundaboy | will the phone automatically re-register? or do i need to reboot it? |
| 03:20.27 | wundaboy | aahahah it did! |
| 03:20.36 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-43-47.cybersurf.com) |
| 03:20.42 | wundaboy | sorry, this is my fist time sitting down and learning/making asterisk work! |
| 03:21.28 | [TK]D-Fender | wundaboy : basically when a phone "registers" its to inform * what IP it can be contacted at. If the IP is guaranteed, there is no need and what you saw was a soft "warning". |
| 03:21.47 | [TK]D-Fender | Normally you only put IP's in for peer entries like other servers |
| 03:22.01 | wundaboy | gotcha |
| 03:22.06 | wundaboy | ok so when i try and dial i get an error |
| 03:22.14 | wundaboy | <PROTECTED> |
| 03:22.18 | wundaboy | thats not correct, right? |
| 03:22.18 | [TK]D-Fender | wundaboy : when it doesn't match, yes |
| 03:22.30 | [TK]D-Fender | thats not good.. should be IAX2 |
| 03:22.37 | wundaboy | o |
| 03:22.41 | wundaboy | is the number fine? |
| 03:22.44 | wundaboy | or does it need a 1? |
| 03:22.58 | [TK]D-Fender | depends on the provider.... some force 10 digit, others 11 |
| 03:23.13 | wundaboy | also, i think my provider is too expensive |
| 03:23.18 | wundaboy | does anyone have any recomendations? |
| 03:23.38 | [TK]D-Fender | What does yours cost now and what kind of usage do you do now? |
| 03:23.46 | wundaboy | $2/month for did 2.9cents/minute |
| 03:23.56 | wundaboy | 1 minute increments which is frustrating |
| 03:23.59 | [TK]D-Fender | wundaboy : each direction? |
| 03:24.02 | wundaboy | correct |
| 03:24.06 | wundaboy | term and orig |
| 03:24.15 | [TK]D-Fender | wundaboy : yeah, that kinda sucks... where are you? |
| 03:24.32 | wundaboy | portland, oregon |
| 03:25.10 | [TK]D-Fender | Broadvoice alther they aren't perfect has a good deal. unlimited in-state for $10 |
| 03:25.14 | [TK]D-Fender | http://www.broadvoice.com/ |
| 03:25.18 | *** join/#asterisk ColdBlood (n=mx232tw@200.103.162.22) |
| 03:25.33 | wundaboy | ooh |
| 03:25.36 | wundaboy | incoming and outgoing? |
| 03:25.39 | xachen | broadvoice sucks |
| 03:25.39 | wundaboy | $10/month? |
| 03:25.47 | wundaboy | i need uptime |
| 03:25.47 | xachen | you'll regret them if your gonna use asterisk with them |
| 03:25.57 | wundaboy | i used to have www.voxee.com for my outgoing |
| 03:26.01 | wundaboy | but they are only up like 22 hours a day |
| 03:26.15 | wundaboy | Call rejected by 66.227.100.30: No such context/extension |
| 03:26.15 | wundaboy | <PROTECTED> |
| 03:26.24 | wundaboy | Executing Dial("SIP/101-9a46", "IAX2/jnctn/15038036247|60") |
| 03:26.32 | wundaboy | i dont understand why it hates me.... |
| 03:26.47 | [TK]D-Fender | wundaboy : you should have a target context as well IIRC |
| 03:26.54 | [TK]D-Fender | @context on the end |
| 03:27.24 | wundaboy | so like @jnctn (the iax2 context for my provider) |
| 03:27.41 | xachen | voipjet sucks too |
| 03:27.48 | [TK]D-Fender | Not quite... check their guide. |
| 03:28.03 | [TK]D-Fender | Everyone suck, some less than others... |
| 03:28.07 | wundaboy | lol |
| 03:28.11 | xachen | but then quite a few of the providers that offer 90%+ international coverage have quality issues :P |
| 03:28.25 | xachen | the only way you are going to get good is if you negotiate with lots of companies |
| 03:28.32 | wundaboy | so their server is iax.jnctn.net |
| 03:28.38 | wundaboy | would it be @iax.jnctn.net ? |
| 03:29.15 | wundaboy | No authority found |
| 03:30.23 | wundaboy | SIP URI: |
| 03:30.24 | wundaboy | pmason@jnctn.net |
| 03:30.33 | wundaboy | is what they have listed on their site, would that be the context? |
| 03:30.33 | [TK]D-Fender | wundaboy : they should have some config samples to use... |
| 03:30.40 | wundaboy | they dont... |
| 03:30.44 | wundaboy | atleast not that i can see |
| 03:30.47 | [TK]D-Fender | wun, not for a dial-out peer entry. |
| 03:32.36 | wundaboy | i hope im not too nub... |
| 03:32.41 | wundaboy | but where in the dial command would it go? |
| 03:32.56 | De_Mon | I'm essing around with app_meetme and the "announce join/leave" option isn't working. |
| 03:33.15 | De_Mon | we record our names but they are never repeated |
| 03:34.26 | [av]bani | yay 0-config polycom |
| 03:34.35 | [TK]D-Fender | wundaboy : there are a few ways to craf a line depending on how the entry is set up in iax.conf |
| 03:35.00 | wundaboy | heres my current dial command: exten => _1NXXNXXXXXX,1,Dial(${JNCTN}/${EXTEN}@iax.jnctn.net,60) |
| 03:35.21 | *** join/#asterisk AsteriskNewbie (n=linux_ba@63.250.96.18) |
| 03:35.32 | AsteriskNewbie | Hello all ... |
| 03:35.55 | wundaboy | i dont understand .. :-\ |
| 03:36.03 | AsteriskNewbie | I need to send "FEATURE xxx" from asterisk to a Norstar system ... anybodyknow how to do that? |
| 03:36.33 | [TK]D-Fender | remove everything from the @ on and see what heppes |
| 03:36.37 | AsteriskNewbie | That is ... does anybody know what codes or digits the "FEATURE" button on a nortel handset sends to the nortel backend? |
| 03:37.08 | [TK]D-Fender | AsteriskNewbie : on a nortel ATA? |
| 03:37.08 | AsteriskNewbie | YEs, TK |
| 03:37.08 | [TK]D-Fender | AsteriskNewbie : Can't really do that.... |
| 03:37.27 | [TK]D-Fender | AsteriskNewbie : there is not "full-service" digital back end to it. |
| 03:37.32 | wundaboy | Call rejected by 66.227.100.30: No such context/extension |
| 03:37.33 | AsteriskNewbie | No? so you can't ... for example, log into a Norstar queue from an analog handset? |
| 03:37.41 | AsteriskNewbie | TK .. no .. there is none .... |
| 03:37.47 | [TK]D-Fender | AsteriskNewbie : expect maybe an Intel interface, but I'm not sure it goes that far... |
| 03:38.08 | [TK]D-Fender | wundaboy, go look at their samples. |
| 03:38.34 | AsteriskNewbie | Hmm .. that doesn't sound too good. I've got .. Norstar Extension->ATA->Asterisk->Ipphone |
| 03:38.51 | AsteriskNewbie | Problelm: I want to be able to log into the Norstar queue from the ip phone .. |
| 03:39.18 | AsteriskNewbie | I would have the "FEATURE" button was just programmed to send a certain digit to the nortel backend? |
| 03:39.38 | [TK]D-Fender | AsteriskNewbie : You can log into the queue from an ATA with straight DTMF.... |
| 03:40.35 | *** join/#asterisk hmodes (i=hmodes@71.224.116.132) |
| 03:40.38 | AsteriskNewbie | TK ... Sorry... you lost me here. what do you mean .. with straight DTMF?? |
| 03:41.24 | wundaboy | ok im gonna try and config it from their examples |
| 03:42.19 | [TK]D-Fender | AsteriskNewbie : Which ACD are you running on your Norstar? |
| 03:42.28 | wundaboy | their example says to use: auth=rsa |
| 03:42.28 | [av]bani | [TK]D-Fender: how many polycoms you got? |
| 03:42.31 | wundaboy | what key should i use? |
| 03:42.55 | wundaboy | nvm |
| 03:42.58 | wundaboy | i figured it out |
| 03:43.05 | AsteriskNewbie | [TK] ... Hmm ...I think we use Call Pilot Mini for the ACD ... |
| 03:43.05 | [TK]D-Fender | [av]bani : just under 30 |
| 03:43.10 | De_Mon | I'm not hearing the user has left or user has joined sounds either... |
| 03:43.21 | [av]bani | [TK]D-Fender: would you be interested in a 0-config polycom autoprovisioner? |
| 03:43.32 | [TK]D-Fender | AsteriskNewbie : Ok, the Minuet allowed for phones on ATA's to log in.... you should eb able to do that on yours I would think. |
| 03:43.54 | [TK]D-Fender | [av]bani : No, I'm just fine with my 30 second copy-paste configurer :) |
| 03:44.10 | [TK]D-Fender | [av]bani : there is a litmit to my laziness.... |
| 03:44.22 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 03:44.31 | [av]bani | [TK]D-Fender: :) |
| 03:45.29 | AsteriskNewbie | [TK] ... Yes .. that what I thought, but how is the question? On a nortel phone, you would press "Feature", the the login code. I'm trying to figure out the equivalent of the "FEATURE" button on an ATA ... |
| 03:45.58 | AsteriskNewbie | Like .. what sets of digits should I press on an ATA to log into the queue? |
| 03:47.23 | [TK]D-Fender | AsteriskNewbie : I think on an ATA you might have access to it through hook-flash, or dialing a keyed extension. |
| 03:47.36 | [TK]D-Fender | AsteriskNewbie : you'd have to read up on your ACD guide |
| 03:48.42 | AsteriskNewbie | TK: Ok .. thanks ...I'd do some more reading and see what I find .. |
| 03:51.33 | *** join/#asterisk bmg505 (n=leon@dsl-165-157-56.telkomadsl.co.za) |
| 03:55.17 | wundaboy | [TK]D-Fender good call, it works now after reading their docs |
| 03:58.17 | wundaboy | so, why does broadvoice suck? |
| 03:58.53 | wundaboy | xachen: why does broadvoice suck? |
| 03:59.02 | *** join/#asterisk orlock (i=[gZsMlMC@202-44-174-4.nexnet.net.au) |
| 03:59.08 | orlock | Has anybody here used a sangoma wanpipe card? |
| 04:00.40 | wundaboy | wait it does suck, only in 1 state... |
| 04:00.44 | wundaboy | not all us... |
| 04:01.49 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 04:03.08 | [TK]D-Fender | orlock : I run an A104d at work... |
| 04:03.36 | wundaboy | i need a little more guidance, can read about it, but how do i setup a menu system? |
| 04:03.40 | Deep6 | anyone used linphone at all? |
| 04:05.34 | [TK]D-Fender | wundaboy : Pastebin your entire extensions.conf and I'll see what kind of head start I can give you. |
| 04:05.36 | [TK]D-Fender | ~pb |
| 04:05.38 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:05.58 | wundaboy | its basically the make samples extensions.conf |
| 04:06.46 | [TK]D-Fender | wundaboy : the best thing you can do for yourself to start is to trash the hell out of it. its 95% crap |
| 04:06.59 | wundaboy | well, my plan for tonight is to build my own config files |
| 04:07.03 | wundaboy | from scratch |
| 04:07.24 | wundaboy | which brings me to another question, what files are necesary? |
| 04:07.30 | wundaboy | sip.conf iax.conf asterisk.conf ? |
| 04:07.47 | Abydos313 | extensions.conf |
| 04:07.58 | wundaboy | yeah, cant forget that ... lol |
| 04:08.15 | [TK]D-Fender | wundaboy : a bunch are needed, but the ones you'll actually work with are limited. |
| 04:09.13 | wundaboy | i also plan on making them all mysql tonight |
| 04:09.20 | wundaboy | once its done building... |
| 04:09.32 | wundaboy | i read a little about it before |
| 04:09.45 | [TK]D-Fender | wundaboy : No point in real-time for a small system.... |
| 04:09.51 | wundaboy | well |
| 04:09.55 | [TK]D-Fender | All pain, no gain :) |
| 04:09.55 | wundaboy | its small for right now |
| 04:10.04 | wundaboy | but im going to create a business with it |
| 04:10.13 | [TK]D-Fender | wundaboy : how big? |
| 04:10.32 | wundaboy | well hopefully huge eventually, but its just going to be a messaging service |
| 04:10.52 | wundaboy | like you get 10 did's and put them on different forms of advertising |
| 04:11.07 | wundaboy | then people call and leave a message and you find out what advertising works the best |
| 04:11.26 | [TK]D-Fender | wundaboy : little need of * realtime for that.... |
| 04:11.38 | wundaboy | well im gonna make a php end for it |
| 04:11.42 | [TK]D-Fender | it'd be AGI for the mostpart |
| 04:11.46 | wundaboy | im a programmer |
| 04:11.52 | wundaboy | so, thats my plan... |
| 04:12.03 | wundaboy | whats AGI? |
| 04:12.11 | [TK]D-Fender | ok, what bits do you have set up now? |
| 04:12.33 | wundaboy | make samples, then just a little extensions.conf sip.conf and iax.conf |
| 04:12.42 | wundaboy | i just started today |
| 04:12.57 | Corydon76-home | AGI is for the lazy who can't learn to program extensions.conf |
| 04:13.32 | [TK]D-Fender | Corydon76-home : No, its for those needing outside decisions to be made without nasty bulk. |
| 04:14.12 | *** join/#asterisk Sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 04:14.26 | Corydon76-home | Yeah, right |
| 04:14.31 | *** join/#asterisk devnull431 (n=slick_sh@D-128-208-39-41.dhcp4.washington.edu) |
| 04:14.55 | Corydon76-home | How about people who think they have too many CPU cycles and want to waste a few |
| 04:15.00 | [TK]D-Fender | Corydon76-home : Of course thats for people who don't want to do it all in C direct... |
| 04:15.42 | [TK]D-Fender | wundaboy : list your extensions and I'll give you a good sample to work with. |
| 04:15.57 | wundaboy | ok if you want it you can have it... |
| 04:16.07 | Corydon76-home | Programming the dialplan is always faster than AGI |
| 04:16.15 | Corydon76-home | and uses less memory |
| 04:16.19 | [TK]D-Fender | I said LIST your extensions (what you # your phones like), no extensions.conf as a whole! |
| 04:16.30 | Corydon76-home | ~pb |
| 04:16.32 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:16.49 | wundaboy | oh, earlier you wanted the whole thing |
| 04:16.54 | wundaboy | ok here are my extensions: |
| 04:16.55 | wundaboy | 101 |
| 04:16.56 | [TK]D-Fender | wundaboy : that was THEN :) |
| 04:17.03 | wundaboy | thats it |
| 04:17.14 | [TK]D-Fender | just 1 so far? SIP? |
| 04:17.19 | wundaboy | yes |
| 04:17.23 | [TK]D-Fender | ok, hold on. |
| 04:18.01 | orlock | Anybody? Sangoma S518 SDSL card? |
| 04:19.34 | Corydon76-home | orlock: try ebay |
| 04:20.16 | [TK]D-Fender | orlock : Yup, I use that on my server |
| 04:20.35 | [TK]D-Fender | wundaboy : Under construction, 2 minutes |
| 04:20.46 | wundaboy | im not sure what your making [TK]D-Fender |
| 04:21.32 | orlock | [TK]D-Fender: can you lspci for me? |
| 04:21.45 | orlock | Corydon76-home: i got several |
| 04:22.01 | Corydon76-home | orlock: We only use Digium |
| 04:22.29 | Grizzy | I have ipkall call 2345678 at myhost.com. I have a paragraph named [2345678] in sip.conf, but it does not activate (the context=blah does not take) What's wrong? |
| 04:22.33 | orlock | Corydon76-home: these are DSL cards, not fxo/fxp cards |
| 04:23.02 | Corydon76-home | orlock: We use Digium for data connectivity, too |
| 04:23.13 | orlock | Ahh, ADSL? |
| 04:23.15 | theorem_ | so, I've watches that presentation in asterisk by Systm ... I think the idea of the Sipura3000 -- I see that Cisco has bought Sipura and stuff the brand into Linksys. Is this still a good device ? or are there other better ones ? |
| 04:23.18 | Corydon76-home | No, T1 |
| 04:23.22 | orlock | ahh |
| 04:23.25 | theorem_ | *watched |
| 04:23.26 | orlock | they are mucho $$$ here |
| 04:23.31 | wundaboy | [TK]D-Fender: where is it?? :P |
| 04:23.33 | [TK]D-Fender | wundaboy : paste just the dial line for your IAX trunk |
| 04:23.34 | orlock | cos of our braindead incumbent telco |
| 04:23.43 | Corydon76-home | $500 for 1.5 here |
| 04:23.53 | wundaboy | Dial(IAX2/jnctn_outgoing/1${EXTEN}) |
| 04:23.54 | Corydon76-home | That's data only, though |
| 04:24.07 | xtrvd | Corydon76-home: Where abouts? |
| 04:24.10 | orlock | we have access to high quality leased line dsl |
| 04:24.11 | [TK]D-Fender | wun So then force 10 digit dialing? |
| 04:24.12 | Corydon76-home | Nashville |
| 04:24.17 | theorem_ | Corydon76-home that seems like a lot ... verizon offers home - fiber plans faster than that for $45 a month |
| 04:24.17 | orlock | so we generally use that instead |
| 04:24.18 | [TK]D-Fender | wundaboy : Sorry, 11 rather? |
| 04:24.35 | Corydon76-home | theorem_: Verizon doesn't compete in Nashville |
| 04:24.38 | wundaboy | 11? |
| 04:24.49 | theorem_ | I see |
| 04:24.53 | Corydon76-home | theorem_: though they're supposed to be, under the agreement that formed Verizon |
| 04:25.33 | theorem_ | it seems they'd nuker hte competition if they came to town. |
| 04:25.37 | theorem_ | *nuke |
| 04:25.42 | theorem_ | I can not type tonight .. |
| 04:25.53 | orlock | [TK]D-Fender:can you do an lspci and get me the id of the s518? |
| 04:25.56 | [TK]D-Fender | wundaboy : Heres a replacement extensions.conf : http://pastebin.ca/44854 |
| 04:25.58 | Corydon76-home | The Justice Dept approved the merger under the condition that Verizon start competing in multiple markets outside of their incumbent cities |
| 04:25.59 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 04:26.29 | Corydon76-home | Ditto for SBC. Neither of them has actually started competing, though. |
| 04:26.49 | theorem_ | is it SBC and At+T now ? |
| 04:27.00 | asterboy | are there call logs in /var/log? |
| 04:27.01 | theorem_ | didn;t at&t just get a facelift .. did htey merge too ? |
| 04:27.37 | [TK]D-Fender | wundaboy : And here is a supplemental file included by the first, to be named "extensions-features.conf" : [features] |
| 04:27.37 | [TK]D-Fender | ; Record new prompts. Don't forget to rename and move these! |
| 04:27.37 | [TK]D-Fender | exten => *40,1,Answer |
| 04:27.37 | [TK]D-Fender | exten => *40,2,Playback(custom/pleaserecordafterbeep) |
| 04:27.37 | [TK]D-Fender | exten => *40,3,Record(/tmp/asterisk-recording:gsm) |
| 04:27.39 | [TK]D-Fender | exten => *40,4,Wait(2) |
| 04:27.40 | Corydon76-home | They merged with SBC |
| 04:27.41 | [TK]D-Fender | exten => *40,5,Playback(/tmp/asterisk-recording) |
| 04:27.43 | [TK]D-Fender | exten => *40,6,Wait(2) |
| 04:27.45 | [TK]D-Fender | exten => *40,7,Hangup |
| 04:27.47 | [TK]D-Fender | ; Playback the last recorded prompt |
| 04:27.49 | [TK]D-Fender | exten => *41,1,Answer |
| 04:27.51 | [TK]D-Fender | exten => *41,2,Wait(1) |
| 04:27.51 | theorem_ | ahh, ok |
| 04:27.53 | [TK]D-Fender | exten => *41,3,Playback(/tmp/asterisk-recording) |
| 04:27.55 | [TK]D-Fender | exten => *41,4,Hangup |
| 04:27.57 | [TK]D-Fender | ; Test out our main menu |
| 04:27.59 | [TK]D-Fender | exten => *42,1,Goto(mainmenu,s,1) |
| 04:28.01 | [TK]D-Fender | ; test to hear your CallerID |
| 04:28.03 | [TK]D-Fender | exten => *43,1,Answer |
| 04:28.05 | [TK]D-Fender | exten => *43,2,Playback(custom/yourcalleridis) |
| 04:28.07 | [TK]D-Fender | exten => *43,3,SayDigits(${CALLERID(number)}) |
| 04:28.09 | [TK]D-Fender | exten => *43,4,Hangup |
| 04:28.11 | [TK]D-Fender | ; Voicemail (by added mailbox #) |
| 04:28.13 | [TK]D-Fender | exten => _*97.,1,Answer |
| 04:28.15 | [TK]D-Fender | exten => _*97.,2,VoicemailMain(${EXTEN:3}@default) |
| 04:28.17 | [TK]D-Fender | exten => _*97.,3,Hangup |
| 04:28.19 | Corydon76-home | Or more correctly, SBC bought ATT and took their name |
| 04:28.19 | [TK]D-Fender | ; Voicemail (by callerID) |
| 04:28.20 | asterboy | what happened to pastebin? |
| 04:28.21 | [TK]D-Fender | exten => *98,1,Answer |
| 04:28.23 | [TK]D-Fender | exten => *98,2,VoicemailMain(${CALLERID(number)$}@default) |
| 04:28.25 | [TK]D-Fender | exten => *98,3,Hangup |
| 04:28.27 | [TK]D-Fender | ; Voicemail (main) |
| 04:28.29 | [TK]D-Fender | exten => *99,1,Answer |
| 04:28.29 | asterboy | ~pastebin |
| 04:28.31 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 04:28.31 | Corydon76-home | [TK]D-Fender: you're a fucking moron. |
| 04:28.31 | [TK]D-Fender | exten => *99,2,VoicemailMain() |
| 04:28.33 | [TK]D-Fender | exten => *99,3,Hangup |
| 04:28.35 | [TK]D-Fender | s |
| 04:28.37 | [TK]D-Fender | DAMMIT |
| 04:28.39 | [TK]D-Fender | sorry |
| 04:28.41 | [TK]D-Fender | http://pastebin.ca/44855 |
| 04:28.41 | wundaboy | lol |
| 04:28.43 | [TK]D-Fender | there.. link. |
| 04:28.45 | [TK]D-Fender | friggen paste error. |
| 04:28.47 | wundaboy | rgr |
| 04:28.48 | asterboy | lol |
| 04:28.55 | *** join/#asterisk heison (n=heison@CPE000625e6c664-CM00122570a518.cpe.net.cable.rogers.com) |
| 04:29.02 | theorem_ | so .. Corydon76-home -- you run asterisk at home, or ? |
| 04:29.09 | wundaboy | on my incoming voip, should i change the context to "macro-stdexten" ? |
| 04:29.21 | Corydon76-home | theorem_: in multiple locations |
| 04:29.22 | [TK]D-Fender | Thanks... I'm well aware of pastebin.... DUH. Just didn't successfuly grab the URL from my paste.. CHILL |
| 04:30.06 | theorem_ | Corydon76-home - so, do you use a FXS/FXO combination box, or just straight IAX / SIP phones no land lines ? |
| 04:30.22 | Corydon76-home | theorem_: depends upon the customer |
| 04:30.26 | [TK]D-Fender | wundaboy, no, in [incoming] you should match the DID like "exten => 8005551212,1,Goto(mainmenu,s,1)" and that will give them a menu |
| 04:30.37 | theorem_ | ahh , so they're commercail installs. |
| 04:30.40 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.137) |
| 04:30.45 | theorem_ | *ia |
| 04:30.49 | wundaboy | oh gotcha |
| 04:31.02 | Corydon76-home | theorem_: and at home. and on my desk at work |
| 04:31.09 | theorem_ | <grin> |
| 04:31.27 | Corydon76-home | theorem_: there isn't a set single way that I do Asterisk installs |
| 04:31.51 | theorem_ | I assume if I get asterisk working, I can fool around wit hsoft phones before I go and buy hardware to do SIP for example ? |
| 04:32.05 | Corydon76-home | You could |
| 04:32.25 | [TK]D-Fender | orlock : I think this is it... 01:09.0 Network controller: Globespan Semiconductor Inc.: Unknown device d002 (rev 01) |
| 04:32.38 | theorem_ | I have it installed .. just need to find some time to do configuration and handling |
| 04:33.05 | asterboy | where does asterisk keep the call logs? |
| 04:33.47 | wundaboy | ok im very nub but when i changed the context for the incoming voip context=incoming i got this error |
| 04:33.51 | asterboy | ./var/log/asterisk doesn't seem to show actual call connections. |
| 04:33.53 | wundaboy | Rejected connect attempt from 66.227.100.30, request '15033341400@incoming' does not exist |
| 04:34.03 | theorem_ | yeah that's a nice card Corydon-home -- out of my price range I think for now :) |
| 04:34.21 | theorem_ | is there a specific way you recommend I follow for first -time ? |
| 04:34.39 | [TK]D-Fender | wundaboy : add this to [incoming] ---- exten => 15033341400,1,Goto(mainmenu,s,1) |
| 04:35.16 | [TK]D-Fender | that will pick up your incoming DID and dump them in the menu sample I gave you. between those 2 files you should have a great base to learn how a dialplan should be arranged. |
| 04:35.37 | Corydon76-home | theorem_: nothing in particular |
| 04:35.49 | Corydon76-home | theorem_: I encourage use of func_odbc, though |
| 04:35.52 | theorem_ | how did you go about it ? |
| 04:36.08 | theorem_ | hmm |
| 04:36.10 | wundaboy | alright, thanks |
| 04:36.12 | theorem_ | asterisk dev .. |
| 04:36.16 | theorem_ | maybe asking the wrong person :) |
| 04:36.46 | Corydon76-home | I encourage the use of anything that I write. :-) |
| 04:37.20 | niteowloz | anybody know how to use ast_config with sip to register with a sip provider |
| 04:37.28 | orlock | [TK]D-Fender: 00:0b.0 Network controller: Globespan Semiconductor Inc.: Unknown device d002 (rev 01) - thats my Traverse Pulsar |
| 04:38.13 | *** part/#asterisk Ramzi-324 (n=Acme@fctnnbsc16w-156034225070.nb.aliant.net) |
| 04:38.23 | niteowloz | searched high and low through the realtime doc but no answer found |
| 04:38.36 | theorem_ | Corydon76-home - I've heard of configurations including use of the cell phone network ... I assume that's possible with asterisk ? If I have a phone cable .. I guess I need drivers ... hmm ... |
| 04:38.38 | Gamercjm | Need help with .call files, who knows how to use/set them |
| 04:39.07 | [TK]D-Fender | wundaboy : You'll have to make some recordings for that menu to work. Thats what the *40 macro is for |
| 04:39.13 | Corydon76-home | Possible, yes. Common, no. |
| 04:40.08 | Corydon76-home | Gamercjm: just stick them in the right directory when you're ready to fire them off |
| 04:40.14 | wundaboy | oic |
| 04:40.19 | theorem_ | Corydon76-home - I envision asterisk hooking into POTS, VoIP services and Cell network , depending on time of day routing calls over efficient mediums. hmm .. maybe SMS on the phone would be a feat. |
| 04:40.26 | Gamercjm | I think i dont have the Channel set up correctly |
| 04:40.37 | Gamercjm | Im trying to use: IAX2/Nufone.net |
| 04:40.44 | wundaboy | so if i type *40 when it will make a recording? |
| 04:40.52 | [TK]D-Fender | yup |
| 04:40.55 | Corydon76-home | theorem_: it requires a GSM modem to hook into the cell network. Those things aren't cheap |
| 04:40.56 | [TK]D-Fender | read what it does. |
| 04:41.00 | theorem_ | I guess I need to bang out what I want to use asterisk for before I do any setting up |
| 04:41.00 | theorem_ | ? |
| 04:41.10 | [TK]D-Fender | thats in thesend file I pastebin'd for you |
| 04:41.27 | Corydon76-home | or a CDMA modem, for Sprint |
| 04:41.36 | theorem_ | why GSM ? I have a CDMA phone here in the US ... verizon is the provider .. I can make calls with Windows out over the USB->phone cord ... |
| 04:41.43 | theorem_ | the same is not true for *nix ? |
| 04:41.47 | wundaboy | did you type this all out? |
| 04:41.53 | wundaboy | or do you have a generator... |
| 04:42.19 | Corydon76-home | theorem_: sure, if you want to have your cell phone tied to the phone system |
| 04:42.33 | theorem_ | ... |
| 04:42.36 | Corydon76-home | theorem_: and even then, you're not able to send more than a single call out at once |
| 04:42.39 | theorem_ | I've never had it any other way |
| 04:42.45 | [TK]D-Fender | wundaboy : All hand coded |
| 04:42.46 | theorem_ | how .. |
| 04:42.51 | theorem_ | how could it be otherwise ? |
| 04:42.56 | theorem_ | (curious) |
| 04:42.57 | Corydon76-home | theorem_: that's why you need a CDMA modem |
| 04:43.13 | wundaboy | [TK]D-Fender: all hand coded tonight? :P |
| 04:43.43 | Corydon76-home | You might get a unit that can do 4 calls at once, plus a good antenna, situated outside of your colo |
| 04:43.46 | wundaboy | can i call it with my polycom, my cell has bad quality |
| 04:43.57 | theorem_ | oh, so the CDMA modem handles multiple cells ... |
| 04:43.59 | theorem_ | right ... |
| 04:44.11 | Corydon76-home | Antennas don't typically do well inside a colo center |
| 04:44.16 | theorem_ | yeah .. maybe overkill for a home user :) |
| 04:44.38 | orlock | Corydon76-home: if a mobile works... |
| 04:44.43 | MikeJ[Laptop] | Corydon76-home, you need to run a repeater to the roof! |
| 04:44.58 | orlock | we have a GSM modem here for nagios paging |
| 04:45.07 | theorem_ | yeah .. route GSM calls in the area over the free VoIP |
| 04:45.11 | orlock | its a Seimens M20A iirc |
| 04:45.11 | theorem_ | nobody would know :) |
| 04:45.15 | Corydon76-home | MikeJ[Laptop]: I've only ever done a single GSM modem for SMS reception |
| 04:45.30 | MikeJ[Laptop] | yeah.. I think we have 1 |
| 04:45.42 | MikeJ[Laptop] | or maybe that is just brians.. |
| 04:46.13 | Corydon76-home | theorem_: GSM calls aren't free, though... and they tend to cost the same whether local or long distance |
| 04:46.26 | theorem_ | I see |
| 04:46.50 | Corydon76-home | I anything, you might set up a voip provider that routes calls over GSM |
| 04:47.19 | theorem_ | right |
| 04:47.31 | theorem_ | seems pricey though .... |
| 04:47.57 | Corydon76-home | Everyone wants a piece of the action |
| 04:48.08 | theorem_ | ja |
| 04:48.39 | Gamercjm | So about the .call files, anybody know how to set the correct channel with IAX2 |
| 04:48.41 | [TK]D-Fender | wundaboy : No, its a trimmed back version of mine. |
| 04:49.35 | wundaboy | can i record messages with my polycom? |
| 04:50.22 | [TK]D-Fender | sure. any phone connected to * will do |
| 04:50.29 | wundaboy | what does _9. mean? |
| 04:50.42 | theorem_ | Corydon76-home - do you recommend any soft phones ? |
| 04:52.25 | [TK]D-Fender | wundaboy, means you dial 9+ the number to dial to dial out. |
| 04:52.33 | wundaboy | o |
| 04:54.08 | wundaboy | my phone automatically sends after i type 9 and then 8 numbers |
| 04:54.18 | wundaboy | Executing Dial("SIP/101-ec48", "IAX2/jnctn_outgoing/1503803624") in new stack |
| 04:54.42 | [TK]D-Fender | wundaboy : There are a LOT of ways you can set up dialing. If you do a lot of inter-extension dialing you can have it so that you jsut dial the ext # (be it 2-4 digits or whatever) and it will IMMEDIATELY dial, or do 9 + number and wait for external. |
| 04:55.13 | [TK]D-Fender | You can also just have it listen to 10 digits and then it can ADD the "1" for you and dial, 7 digit + wait and it'll add the first 4, etc... |
| 04:55.23 | theorem_ | [TK]D-Fender - some systems you hear a different dialtone after you press 9 |
| 04:55.30 | theorem_ | I assume that's easy to setup. |
| 04:55.45 | [TK]D-Fender | theorem_ : Not really.... |
| 04:55.52 | *** part/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 04:55.55 | theorem_ | oh. |
| 04:55.58 | theorem_ | :-/ |
| 04:56.14 | theorem_ | you've let me down ! |
| 04:56.34 | [TK]D-Fender | theorem_ : basically * usually expects you to break dial-tone by starting your number and by the time you stop, have something to process |
| 04:56.49 | [TK]D-Fender | happy landings! |
| 04:57.16 | theorem_ | I see |
| 04:57.35 | theorem_ | so, you can't prosess the keypresses until after .. |
| 04:57.40 | theorem_ | seems .. tricky. |
| 04:57.47 | wundaboy | oh man |
| 04:57.51 | wundaboy | this is fun to play around with |
| 05:00.16 | theorem_ | wundaboy - using a POTS phone ? |
| 05:00.26 | wundaboy | neg |
| 05:00.33 | wundaboy | polycom ip500 |
| 05:00.36 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 05:00.38 | theorem_ | k |
| 05:00.44 | wundaboy | why is that? |
| 05:00.48 | theorem_ | just curious |
| 05:01.02 | theorem_ | taking a poll :) |
| 05:02.59 | theorem_ | bbl |
| 05:03.10 | wundaboy | so i connect with the main menu |
| 05:03.36 | wundaboy | then i type *40 |
| 05:03.41 | wundaboy | and nothing happens in the console |
| 05:04.21 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 05:05.02 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 05:08.59 | [TK]D-Fender | wundaboy : make sure you made that 2nd file I pastebined... |
| 05:09.38 | [TK]D-Fender | No, you don't do that throught he menu, you do it direct on your phone |
| 05:11.06 | *** join/#asterisk Eggplant (i=No@dsl-859.cascadeaccess.com) |
| 05:12.42 | Nugget | yay happy bkws |
| 05:17.40 | De_Mon | anyone successfully using meetme's 'i' flag? |
| 05:17.45 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 05:21.54 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 05:25.23 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 05:26.39 | *** join/#asterisk gwynm (n=Gwyn@ppp158-45.lns3.adl2.internode.on.net) |
| 05:28.00 | gwynm | Hey guys. I'm getting conflicting information - do I *need* a timing device to do IAX trunking? |
| 05:28.50 | wunderkin | the answer is.... yes |
| 05:29.08 | gwynm | Right. That's going to be fun.. I'm on a VPS :/. |
| 05:29.39 | wunderkin | dont use trunking |
| 05:30.10 | gwynm | I just signed up with atp (austechpartnerships.com) ... they use it at their end. |
| 05:31.02 | gwynm | I get billions of "WARNING[8939]: chan_iax2.c:5075 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)" ... presumably because they're using trunking and I'm not ..? |
| 05:31.29 | wunderkin | i've never used it, maybe |
| 05:33.22 | wunderkin | may be a way to use ztdummy, you would have to research that more, i haven't had to worry about that |
| 05:56.31 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 05:58.34 | Telamon | Does anyone know what codec's the IAXy actually supports? It says it will do g729, but I can't seem to get it to make a call with that codec. It drops back to ulaw. |
| 05:59.25 | russellb | um, where did you see that the IAXy does 729? |
| 05:59.32 | russellb | It supports ulaw and adpcm. |
| 06:00.08 | hmodes | yeah, somehow I doubt the iaxy cpu could handle g729, let alone licensing implications |
| 06:00.12 | Telamon | In the source code to iaxyprov. |
| 06:01.11 | Telamon | Shit. So not even GSM? The whole 80kB/sec codec thing is a problem if you want to run multiple calls off a single cable modem... |
| 06:03.00 | hmodes | run a * on the same lan to transcode? |
| 06:04.00 | hmodes | I think the iaxy was mostly targetted as a lan tdm extension rather then a cpu-heavy remote cpe |
| 06:04.28 | Telamon | I think I'd rather just buy a different phonebox. The Grandstream HT series supports g729, as does the GNet VP168. |
| 06:04.30 | Mavvie | there is something serious wrong with the logging of asterisk. It shouldn't be so hard to find matching lines from one (leg of a) call on a busy pabx. |
| 06:05.53 | hmodes | i'd take a linksys pap2 over grandstream any day of the week, personally |
| 06:06.01 | Abydos313 | why |
| 06:06.24 | hmodes | tho' I dunno if the grandstreams can do more then one g729 stream |
| 06:06.35 | Telamon | Okay, second question. :) I'm having problems with transmitting sound on IAX calls, for both the IAXY and other IAX phones. Basically, the transmitted sound cuts out when talking to another phone, but works fine when leaving voicemail on the Asterisk server. SIP phones function fine, and IAX phones receive sound perfectly. This is codec independant. I've tried with ulaw, g729, and gsm. |
| 06:06.48 | Abydos313 | hmodes i'm asking because i haven't bought my device yet. i had the spa3k in mind |
| 06:06.53 | hmodes | the sipura-derivatives have far superior provisioning and stability tho' |
| 06:07.48 | Telamon | hmodes: Is that one of the Linksys routers with a built in 2 port device (ie, the Vonage box) or one of the little IAXy style Sipura devices? |
| 06:08.35 | Abydos313 | i want quality over price. |
| 06:08.59 | hmodes | the pap2 == spa2k/ata186 |
| 06:09.14 | Abydos313 | ok |
| 06:09.16 | hmodes | just another adaptation of the original kimodo design |
| 06:09.29 | hmodes | rock solid, if not the fastest in the world |
| 06:09.39 | Abydos313 | i'm surprised there isn't more howto's on making your own ata adapter on the net. |
| 06:09.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:09.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:09.52 | hmodes | given the bandwidth for ulaw, i'd take an iaxy |
| 06:10.03 | hmodes | if you need g729, i'd take a pap2/ata186/spa2k |
| 06:10.07 | hmodes | if you need a router, good luck ;p |
| 06:10.23 | Abydos313 | pap2-na :)) |
| 06:10.27 | Grizzy | abydos - meetoo. |
| 06:10.43 | Abydos313 | i don't have that, jsut if you're going to get a pap2 get na model |
| 06:11.05 | Grizzy | I was just looking for FXO do-it-yourself devices. (as in unix-windmodem drivers) |
| 06:11.06 | Abydos313 | i'm only having fun with softphones at the moment |
| 06:11.27 | *** join/#asterisk konfuzed (n=Konf@H135.C72.B0.tor.eicat.ca) |
| 06:11.28 | Grizzy | abydos - have you tried an ipkall number yet? |
| 06:11.34 | Abydos313 | no |
| 06:11.34 | hmodes | there's a lengthy thread on dslreports about unlocking store-bought pap2s ;p |
| 06:11.37 | hmodes | it's pretty trivial |
| 06:11.53 | konfuzed | oh realy |
| 06:12.02 | Abydos313 | actually what is ipkall? |
| 06:12.30 | konfuzed | how about the rpt300 |
| 06:12.38 | Grizzy | abydos - go to http://www.ipkall.com Free regular phone numbers that will talk to asterisk. |
| 06:12.39 | Abydos313 | ok googled answered that |
| 06:13.04 | hmodes | no such luck with the rpt/wrtps |
| 06:13.17 | Abydos313 | nice. |
| 06:13.20 | Grizzy | Mine is working, though I don't understand why it uses the default extension. |
| 06:13.24 | Abydos313 | Grizzy are you using it? |
| 06:13.29 | Grizzy | yes. |
| 06:14.01 | Abydos313 | so what exactly does it let you do with * |
| 06:14.19 | Abydos313 | an inbound number? |
| 06:14.20 | Grizzy | there's some BS on voip-info about how to set it up. one bit of it is right, anyway. |
| 06:14.46 | Abydos313 | i'll search the forums |
| 06:14.46 | Grizzy | have your computer read your e-mail to you. |
| 06:15.14 | Grizzy | have your computer warm up your hot tub for you. |
| 06:15.21 | Abydos313 | heh |
| 06:15.46 | Grizzy | get all your friends to conference into a big voice chat room. |
| 06:15.50 | sl16 | i want to know on what second one extension is answered, how could i do that |
| 06:16.02 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 06:16.48 | jayk- | is there a way to tie in asterisk with a 128 phone line motel? what kind of hardware would you use to mux the POTS lines to the asterisk server using a quad PRI board? |
| 06:18.08 | Abydos313 | i'm on the west coast , a call to washington wouldnt be cheap..heh |
| 06:18.14 | *** join/#asterisk udk (i=udontkno@freenode/staff/udontknow) |
| 06:18.56 | jayk- | i'm in washington |
| 06:19.15 | Abydos313 | so it would work for your friends |
| 06:31.54 | Grizzy | I pay the $50/month for unlimiited long-distance. |
| 06:32.34 | Grizzy | my phone bills looked like the italian national debt in lyre, before. |
| 06:40.17 | denon | isnt that lira? |
| 06:41.06 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 06:41.14 | UdontKnow | Grizzy: where do you live? |
| 06:47.21 | firestrm | !seen websae |
| 07:00.12 | *** join/#asterisk htims (n=pd@213-182-119-45.teleos-web.de) |
| 07:01.03 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:03.11 | *** part/#asterisk devnull431 (n=slick_sh@D-128-208-39-41.dhcp4.washington.edu) |
| 07:03.36 | *** join/#asterisk apardo (n=apardo@87.218.45.124) |
| 07:03.38 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:04.55 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:06.11 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 07:07.51 | tasat | hi, new to sip and having some problems: asterisk is periodically (every 120s) sending register commands to the proxy server, its looking like they all come back unauthorized, asterisk tries again, and then it's successful -- is asterisk changing something after the first failure? is there a way to get through the first time? |
| 07:13.33 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:13.48 | tasat | hi, new to sip and having some problems: asterisk is periodically (every 120s) sending register commands to the proxy server, its looking like they all come back unauthorized, asterisk tries again, and then it's successful -- is asterisk changing something after the first failure? is there a way to get through the first time? |
| 07:17.05 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 07:17.21 | tasat | hi, new to sip and having some problems: asterisk is periodically (every 120s) sending register commands to the proxy server, its looking like they all come back unauthorized, asterisk tries again, and then it's successful -- is asterisk changing something after the first failure? is there a way to get through the first time? |
| 07:19.09 | UdontKnow | tasat: I have a similar problem... a user is sending the very same message every few minutes here |
| 07:19.11 | UdontKnow | hehe |
| 07:19.43 | tasat | yeah, sorry... saw a couple new people enter -- everyone else seems dead |
| 07:20.51 | tuxinator_linuxM | It's always dead in the middle of the night |
| 07:21.32 | tasat | you mean people actually sleep? |
| 07:21.45 | tuxinator_linuxM | not by choice |
| 07:24.17 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:31.26 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-187.claranet.co.uk) |
| 07:37.28 | Snake-Eyes | pfft its late afternoon here :P |
| 07:38.08 | tasat | Snake-Eyes: since you're wide awake... any idea about my problem? |
| 07:38.32 | *** join/#asterisk ZX81 (n=ubuntu@222-153-114-171.jetstream.xtra.co.nz) |
| 07:38.43 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:43.13 | ZX81 | ~ping |
| 07:43.14 | jbot | pong |
| 07:43.24 | ZX81 | wow ok, it's just quiet |
| 07:43.25 | ZX81 | :) |
| 07:49.57 | *** join/#asterisk littlejohn (n=little@host75-71.pool8261.interbusiness.it) |
| 07:52.52 | shido6 | AHAHAHAHA |
| 07:53.41 | ZX81 | lol |
| 07:53.43 | ZX81 | ok |
| 07:53.44 | ZX81 | :) |
| 07:53.54 | *** join/#asterisk nayyares (n=Nayyar@58.65.151.218) |
| 07:54.08 | FuriousGeorge | hey all |
| 07:54.10 | *** join/#asterisk apardo (n=apardo@87.218.45.124) |
| 07:54.17 | ZX81 | hi |
| 07:54.25 | FuriousGeorge | k pasa |
| 07:54.34 | ZX81 | meh |
| 07:54.38 | ZX81 | nm |
| 07:55.32 | tasat | hi, anyone familiar with asterisk's sip registration procedure? |
| 07:55.40 | ZX81 | nope |
| 07:55.42 | ZX81 | not really |
| 07:55.44 | ZX81 | :) |
| 07:55.47 | ZX81 | hehe |
| 07:55.52 | ZX81 | why? |
| 07:55.57 | FuriousGeorge | register > user:password@server.com |
| 07:56.02 | ZX81 | => |
| 07:56.04 | tasat | trying to debug a problem..... yeah, |
| 07:56.04 | ZX81 | :D |
| 07:56.19 | ZX81 | do a sip debug maybe? |
| 07:56.25 | ZX81 | make sure realm is correct |
| 07:56.26 | ZX81 | :) |
| 07:56.35 | tasat | I've got the command right, but having unreliable initiation |
| 07:56.39 | FuriousGeorge | and if you want .../s if you want it to default to s extension in given context defined above |
| 07:56.48 | tasat | works maybe 1 in 4 times... |
| 07:56.50 | FuriousGeorge | define unreliable? |
| 07:56.54 | ZX81 | packet loss? |
| 07:57.00 | FuriousGeorge | angry monkey? |
| 07:57.03 | ZX81 | unreliable host? |
| 07:57.10 | FuriousGeorge | ~FuriousGeorge |
| 07:57.11 | jbot | i heard furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 07:57.22 | ZX81 | ~adn |
| 07:57.23 | jbot | adn is, like, the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 07:57.55 | tasat | perhaps, I don't know.... but asterisk issues the register command for each proxy server, it comes back unauthorized... tries again and then it works.... then after 120sec it expires and the process repeats. |
| 07:58.08 | tasat | this sound like typical sip registration process? |
| 07:58.42 | FuriousGeorge | ok, i have 6 candels burning, you guys think that gonna help my room of approximately 3,000 cubic feet |
| 07:59.00 | FuriousGeorge | *help heat it |
| 07:59.25 | FuriousGeorge | tasat: try a different provider |
| 07:59.27 | FuriousGeorge | see what happens |
| 07:59.50 | FuriousGeorge | www.sipphone.com is free |
| 07:59.53 | FuriousGeorge | use to test |
| 07:59.57 | tasat | yeah, ok. |
| 08:00.10 | tasat | FuriousGeroge: but does anything sound out of the ordinary? |
| 08:00.17 | FuriousGeorge | register => username:password@proxy01.sipphone.com |
| 08:00.30 | FuriousGeorge | tasat: yeah, something is not ordinary with that :) |
| 08:00.45 | tasat | FuriousGeorge: what is not ordinary? |
| 08:01.03 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 08:01.20 | FuriousGeorge | the part where: "asterisk issues the register command for each proxy server, it comes back unauthorized... tries again and then it works.... then after 120sec it expires and the process repeats." |
| 08:01.31 | FuriousGeorge | hioghly irregular |
| 08:01.34 | FuriousGeorge | highly* |
| 08:01.42 | ZX81 | yeah |
| 08:01.48 | ZX81 | sounds like the other end is retarded |
| 08:01.50 | ZX81 | :) |
| 08:01.52 | tasat | what about the 120 sec. expiry? that's an asterisk default no? |
| 08:01.57 | ZX81 | yeah |
| 08:01.59 | ZX81 | inside the file |
| 08:02.07 | ZX81 | but they need to fix their problem |
| 08:02.08 | ZX81 | really |
| 08:02.20 | ZX81 | who is it? |
| 08:02.25 | tasat | asterlink |
| 08:02.33 | ZX81 | lol |
| 08:02.35 | ZX81 | well |
| 08:02.37 | ZX81 | that should work |
| 08:02.38 | ZX81 | lol |
| 08:02.39 | FuriousGeorge | hmmm, heard they were pretty good |
| 08:02.43 | ZX81 | yeah |
| 08:02.44 | ZX81 | same |
| 08:02.45 | ZX81 | :) |
| 08:02.47 | ZX81 | um |
| 08:02.49 | tasat | yeah... me too |
| 08:02.54 | oej | ZX81: Evening/morning! |
| 08:02.58 | ZX81 | :) |
| 08:03.00 | ZX81 | evening |
| 08:03.01 | ZX81 | :D |
| 08:03.02 | tasat | that's why I'm thinking it's me |
| 08:03.10 | ZX81 | lol |
| 08:03.12 | ZX81 | ask oej |
| 08:03.14 | ZX81 | hahahaha |
| 08:03.15 | FuriousGeorge | tasat: nat? |
| 08:03.15 | ZX81 | jk |
| 08:03.17 | ZX81 | :D |
| 08:03.28 | ZX81 | but he's getting auth problems no? |
| 08:03.32 | tasat | so, just so I have this right... the 120 sec expiry is normal, right? |
| 08:03.33 | ZX81 | what error do you get back |
| 08:03.36 | ZX81 | yeah |
| 08:03.43 | ZX81 | how are you oej? |
| 08:03.48 | FuriousGeorge | dont ask iej, i hear he is still working on learning to park calls right, or something. he cant help you |
| 08:03.52 | FuriousGeorge | *oej |
| 08:03.52 | tasat | FuriousGeorge: yes, but the ports are open |
| 08:04.01 | ZX81 | is natting? |
| 08:04.07 | ZX81 | the question really |
| 08:04.09 | FuriousGeorge | ports shouldnt matter if you are client side |
| 08:04.13 | ZX81 | is whether it gets refused |
| 08:04.17 | ZX81 | or just forgotten |
| 08:04.19 | ZX81 | :) |
| 08:04.28 | tasat | the only thing I get back is a 401 unauthroized |
| 08:04.41 | ZX81 | yeah |
| 08:04.42 | ZX81 | see |
| 08:04.45 | ZX81 | so it's not a nat hole |
| 08:04.48 | tasat | Register, trying, unauthorized, register, ok... |
| 08:04.53 | ZX81 | maybe load balancing? |
| 08:04.57 | ZX81 | at the other end |
| 08:05.02 | tasat | they've got 7 switches |
| 08:05.02 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 08:05.09 | ZX81 | and one machine doesn't have correct details? |
| 08:05.34 | FuriousGeorge | tasat: grab a sipphone account see what happens. it will take 3 min. directions on voip-info. search sipphone |
| 08:05.35 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:05.50 | *** part/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 08:05.52 | tasat | ok |
| 08:08.30 | nayyares | which softphone work fine with CenOS? GLP :) |
| 08:08.38 | Qwell | nayyares: any? |
| 08:08.42 | nayyares | s/GLP/GPL |
| 08:08.47 | Qwell | kphone, linphone, iaxcomm, idefisk, xlite |
| 08:08.49 | Qwell | twinkle |
| 08:09.08 | [av]bani | Qwell: warezed 7970 sip yet? |
| 08:09.18 | nayyares | Qwell, thanks |
| 08:09.31 | Qwell | nope |
| 08:11.33 | [av]bani | hmm.. looks like the sip images are different in some way.. nobody has been able to get the new sip images to register with * |
| 08:11.44 | [av]bani | it has something specific for CCM now |
| 08:12.27 | Qwell | fun |
| 08:14.04 | tasat | FuriousGeorge, Zx81: thanks for your help.... sipphone seemed to choke on my reg email, so I'm going to bed... I'll check with the asterlink guys tomorrow |
| 08:14.23 | FuriousGeorge | later |
| 08:14.31 | FuriousGeorge | tasat: \ |
| 08:14.39 | FuriousGeorge | gizmoproject.com i think |
| 08:14.46 | FuriousGeorge | its no longer sipphone |
| 08:15.00 | FuriousGeorge | i had same problem once |
| 08:15.02 | tasat | yeah, ok, it just came through |
| 08:15.18 | FuriousGeorge | from sipphone? |
| 08:15.24 | tasat | yeah |
| 08:15.28 | FuriousGeorge | hmm |
| 08:15.38 | FuriousGeorge | i guess both are working again |
| 08:16.33 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:16.41 | diLLec | anyone can help with an "asterisk behind a nat and phone behind a nat" problem ? |
| 08:18.15 | Qwell | diLLec: behind different nats? |
| 08:18.33 | *** join/#asterisk fifer (n=20f04395@c-24-20-155-56.hsd1.wa.comcast.net) |
| 08:22.35 | diLLec | yes. |
| 08:22.46 | *** join/#asterisk apardo (n=apardo@87.218.45.124) |
| 08:22.47 | diLLec | SIP phone (snom190) is behind a netgear nat |
| 08:23.01 | diLLec | the asterisk server is behind a checkpoint firewall |
| 08:23.34 | diLLec | connectivity between phone and asterisk is given. |
| 08:24.10 | diLLec | but asterisk is responding "SIP/2.0 401 Unauthorized" on REGISTER sip:........... SIP/2.0 |
| 08:29.48 | *** join/#asterisk otaku42 (n=otaku42@madwifi/developer/otaku42) |
| 08:29.58 | otaku42 | morning |
| 08:31.20 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 08:31.25 | otaku42 | i'm wading through the configuration of a sipura spa-2002 and am wondering about the meaning of several configuration settings related to "NSE". does anyone in here know what NSE means? |
| 08:35.07 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:35.53 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:38.24 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 08:45.21 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:46.02 | otaku42 | wild guess: is NSE something like "no such encoder", packed at the end of the codec preference list to signal "sorry, pal, we can't negotiate on a common codec, as it seems"? |
| 08:47.01 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:48.16 | Enth | oh man, cheers. |
| 08:48.24 | Enth | :) |
| 08:51.27 | Enth | what's special about 8th? |
| 08:52.16 | otaku42 | Enth: world female day or something like that |
| 08:52.24 | Enth | OH |
| 08:52.46 | RoyK | http://en.wikipedia.org/wiki/International_Women's_Day |
| 08:58.42 | wasim | Every day is Women's day! |
| 08:58.45 | Enth | hah |
| 08:58.51 | wasim | oh, i forgot, you're not married :S |
| 08:59.14 | RoyK | wasim: they still have this day, though |
| 08:59.45 | wasim | lol, nobody cares about poor wasim |
| 08:59.46 | wasim | no riots |
| 08:59.50 | wasim | no burnings |
| 08:59.51 | wasim | nothing ... |
| 08:59.53 | wasim | tsk tsk |
| 08:59.57 | RoyK | :) |
| 09:00.24 | RoyK | http://karlsbakk.net/fun/wasimlane.gif |
| 09:01.13 | wasim | hahaha |
| 09:01.38 | wasim | mmmh ... virgins |
| 09:01.42 | zoa | haha |
| 09:04.14 | wasim | maybe they misprinted it, maybe its just 1 virgin, whose like as fat as 72 nice thin blondes |
| 09:04.27 | wasim | that would not be fun ... |
| 09:04.39 | wasim | ooh, ooh, maybe thats hell, and the 72 thin ones is heaven :) |
| 09:05.12 | wasim | especially if she wants to be on top |
| 09:07.10 | RoyK | :) |
| 09:09.22 | RoyK | zoa: there must be something between 1.2.0 and 1.2.5 that breaks the jb patch |
| 09:09.26 | RoyK | i'll have to work on it |
| 09:09.34 | RoyK | 1.2.0 does not leak |
| 09:10.49 | zoa | aha good news |
| 09:10.56 | RoyK | indeed |
| 09:10.58 | zoa | try to test the versions in between |
| 09:11.02 | zoa | until you find where it breajs |
| 09:11.03 | zoa | breaks |
| 09:11.23 | RoyK | I'll need to setup a test bench first |
| 09:11.32 | RoyK | can't do this in production |
| 09:11.50 | zoa | aha k |
| 09:12.23 | RoyK | zoa: you had some nice tools for stresstesting this, didn't you? |
| 09:12.42 | zoa | not really |
| 09:13.01 | RoyK | i thought you said you'd ran a few million calls through it.... |
| 09:14.31 | otaku42 | noone has an idea about the meaning of the term "NSE" (in context with Sipura SPA-2002)? |
| 09:15.15 | RoyK | anyway - this means the asterisk generic jitterbuffer works!! |
| 09:15.55 | Enth | Anyone can recommend ways to eliminate or reduce echoing? |
| 09:16.29 | RoyK | Enth: with what sort of communication? sip/sip? sip/zap? |
| 09:16.36 | RoyK | iax/sip/zap/sccp/mgcp! |
| 09:17.00 | Enth | SIP |
| 09:17.04 | Enth | SIP/SIP |
| 09:17.13 | wasim | MGCP! |
| 09:17.24 | Enth | hah |
| 09:17.26 | wasim | oh the nightmares of last night |
| 09:18.01 | Enth | RoyK ? |
| 09:18.22 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 09:18.42 | RoyK | Enth: echocancel should be at the sip endpoints |
| 09:19.12 | Enth | Where do I do that? |
| 09:19.23 | RoyK | not in asterisk unless you use asterisk as a softphone |
| 09:19.50 | Enth | I use x-lite as the softphone |
| 09:19.58 | RoyK | Enth: there's work on changing a bug (feature?) in asterisk that replaces sip timestamps when sip calls are bridged, which is stupid, but i'm not sure if that is relevant to this |
| 09:20.17 | RoyK | with a laptop and built-in speakers and mic? |
| 09:20.18 | Enth | no it's not relevant. |
| 09:20.39 | Enth | RoyK - No - A desktop with 8 Channel Surround Sound and external mic |
| 09:21.01 | Enth | why? do laptops and built in speakers/mic have problems? |
| 09:21.01 | RoyK | rotfl |
| 09:21.26 | RoyK | that's the worst echo scenario on the planet |
| 09:21.32 | Enth | I tend to hear the echo of my own words a few times. |
| 09:21.39 | *** join/#asterisk astar` (n=astar@ANantes-154-1-64-111.w81-53.abo.wanadoo.fr) |
| 09:22.02 | Enth | Any suggestions? |
| 09:22.05 | RoyK | Enth: no |
| 09:22.10 | Enth | Great. |
| 09:22.18 | RoyK | Enth: i don't think there are any echo cancellors that can deal with that |
| 09:22.53 | Grizzy | polycom speakerphones do rather well at echo cancellation. : o ) |
| 09:23.07 | Enth | So what's the whole point of running * if one can't reduce echo cancellation? :) |
| 09:23.25 | X-Rob_ | you want to reduce echo, or increase echo cancellation |
| 09:23.31 | X-Rob_ | I'm pretty sure you don't want to reduce echo cancellation |
| 09:23.42 | Enth | reduce. |
| 09:23.53 | Grizzy | you need more microphones and some heavy "follow the source" software. |
| 09:24.03 | Enth | *sigh* |
| 09:24.21 | RoyK | Enth: loud speakerphones is always a problem |
| 09:24.32 | RoyK | sound from the speakers go into the mic |
| 09:24.33 | RoyK | etc |
| 09:24.40 | Enth | ah |
| 09:24.52 | Enth | so you'd suggest hardphones? |
| 09:24.53 | RoyK | so you need some pretty fancy echocancellors to remove that |
| 09:25.16 | RoyK | they usually have some if they're built for speakerphones |
| 09:25.24 | Enth | IP hard phones are better then? |
| 09:25.49 | Grizzy | it can be done adequately, but it's a complicated problem that polycom's do in a DSP. |
| 09:26.06 | RoyK | Enth: they may be, try it |
| 09:26.15 | Enth | Do you guys get echoes? |
| 09:26.27 | RoyK | Enth: get a deal with the supplier so you can return them if they dont't do the job |
| 09:26.32 | Grizzy | Headsets are inexpensive. |
| 09:26.36 | RoyK | yeah |
| 09:26.47 | Enth | I also use headsets but there are echoes there too |
| 09:27.03 | kippi | hey |
| 09:27.05 | Grizzy | X-lite and X-ten I -think- are supposed to have some cancellation. |
| 09:27.19 | Enth | I'm guessing the actual proc/box has to do with this too. - that is must be a really good processor, good RAM etc |
| 09:27.21 | kippi | I have a avaya 4620 handset, has anyone else used them |
| 09:27.49 | RoyK | Enth: try eyebeam, pay for it, if it has echo, call xten and bitch them :) |
| 09:28.13 | Enth | hah |
| 09:28.20 | RoyK | Enth: echocancel doesn't require too much cpu for a single call |
| 09:28.30 | Grizzy | x-lite seems to do slightly better than pulver communicator. |
| 09:29.02 | Enth | hrmmm, I'm just surprised that there is nothing that one can do to reduce echoes in * |
| 09:29.02 | *** join/#asterisk basta (n=basta@213-156-52-98.fastres.net) |
| 09:29.30 | Grizzy | i thought there was a cancellation plug in? |
| 09:29.37 | Enth | I mean if one runs Cisco AVVID or Nortel IPT and any commercial products with softphones etc, there is hardly are echoing |
| 09:29.49 | RoyK | Enth: if there is, it requires PURCHASING the lient |
| 09:29.56 | RoyK | x-lite isn't supported anymore |
| 09:29.57 | Enth | ah |
| 09:31.06 | Enth | hrmmm |
| 09:31.13 | RoyK | it's not that expensive :) |
| 09:32.18 | Grizzy | my x-lite still works with my asterisk... |
| 09:32.32 | RoyK | it does |
| 09:32.42 | Enth | it works but its not as good as well commercial stuff |
| 09:32.51 | RoyK | eyebeam audio only is only $30 |
| 09:32.52 | Enth | remember its free and not supported anymore. |
| 09:33.05 | Enth | EyeBeam does not work on Linux/BSD platforms. |
| 09:33.09 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 09:33.10 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:33.32 | RoyK | hm |
| 09:33.33 | RoyK | no |
| 09:33.55 | RoyK | i don't know any good softphones for *nix |
| 09:36.03 | Enth | Well, apart from x-lite, GnomePhone (It's an H.323 client) is ok but most of the others are that good |
| 09:37.38 | viperdude | anyone here got experience with flash operator panel? |
| 09:37.39 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:38.11 | Enth | can be an issue if one's a mobile worker and running linux/bsd on his laptop and can only use softphones. |
| 09:38.33 | Enth | specially for SME who have a few teleworkers/mobile workers running laptops |
| 09:38.46 | Enth | cant expect them to carry their hard phones. |
| 09:41.09 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:45.35 | Enth | Well, the article on voip-info on echo cancellation says it all. |
| 09:45.37 | Enth | hrmmm |
| 09:45.39 | Enth | thought as much |
| 09:48.20 | *** join/#asterisk adaro2001 (n=adaro@62-213-205-18.colo.kangaroot.net) |
| 09:54.39 | RoyK | Enth: softphones work well. we have been using x-pro in our callcentre for 18 months or so |
| 09:55.13 | Enth | no echo issue? |
| 09:55.34 | Enth | I guess that's the price one has to pay when it's "free" |
| 09:55.46 | vgster | i got off softphones, moved to hardphones with headsets, think the probably was with the rubber band power PC's they were using |
| 09:56.10 | Enth | heh |
| 09:56.18 | zoa | royk, try idefisk! |
| 09:56.37 | Enth | does idefisk work on SIP? |
| 09:56.55 | vgster | i had a problem selecting my headset with idefisk, kept defaulting to the onboard audio |
| 09:57.08 | zoa | not yet, but it will soon work on sip |
| 09:57.19 | RoyK | zoa: idefisk??? |
| 09:57.28 | Enth | iax2 client |
| 09:57.30 | kippi | http://www.asteriskguru.com/tools/idefisk_windows.php |
| 09:57.41 | RoyK | in norwegian, ide means Idea, fisk means fish, so idefisk means idea-fish |
| 09:57.42 | RoyK | :P |
| 09:57.46 | zoa | yeah i know about that |
| 09:57.47 | zoa | :) |
| 09:57.50 | Enth | which is what it is. |
| 10:00.24 | vgster | RoyK did you get your spandsp thing fixed? |
| 10:00.33 | RoyK | not yet |
| 10:00.38 | vgster | does itn ot build? |
| 10:00.49 | RoyK | it builds but it doesn't load |
| 10:01.13 | vgster | odd, cos i remember seeing the error you had, but i didnt get it when i tried yesterday |
| 10:03.27 | RoyK | what versions? |
| 10:03.41 | vgster | * 1.2.4 and 1.2.5 and spandsp pre25 |
| 10:03.47 | RoyK | dunno if it's relevant, but this is x86_64 |
| 10:03.50 | vgster | ah |
| 10:03.53 | zoa | http://www.asteriskguru.com/tutorials/spandsp.html |
| 10:03.56 | zoa | did you try this ? |
| 10:04.12 | vgster | i had problems with x86_64 in general |
| 10:06.25 | Grizzy | I wanna set up festival and sphinx! Oh boy! Oh boy! |
| 10:07.03 | *** join/#asterisk vadimx (n=vadim@213-35-233-174-dsl.end.estpak.ee) |
| 10:09.17 | vadimx | Hello, somebody can help me? I need to send fax to SIP phone, and I have problem with "Channel" syntax in fax file |
| 10:10.07 | vadimx | Example: Channel:SIP/G1/1234567 |
| 10:10.17 | vadimx | what is G1 and 1234567? |
| 10:16.45 | vgster | id assume that 1234567 is the destination number |
| 10:16.46 | oej | zoa!!! |
| 10:17.19 | vgster | isnt the G1 the zap dialout group |
| 10:17.21 | zoa | olle! |
| 10:17.23 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 10:17.27 | zoa | email me those pictures |
| 10:17.40 | zoa | i need them for blackmailing purposes |
| 10:18.04 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 10:19.19 | vadimx | and G1 - this is example only.. this is terminal what sending a fax? |
| 10:19.32 | oej | zoa: I have them in secret storage |
| 10:19.32 | vadimx | i mean what must be in second paramater? |
| 10:19.40 | oej | zoa: And you know my paypal address |
| 10:19.49 | oej | zoa: I'll accept SEK, EURO and USD |
| 10:19.53 | otaku42 | no one has an idea about the meaning of the term "NSE" (in context with Sipura SPA-2002)? |
| 10:22.00 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:22.02 | *** join/#asterisk adnans (n=adnans@noterik.xs4all.nl) |
| 10:23.00 | darkskiez | what parameter changes the source ip address in the sip packets to be a dns address so calls come from number@domain rather than number@ip? or do i need to configure reverse dns |
| 10:25.34 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:26.01 | backblue | hi, morning all, does anyone knows any way of, rewrite, the from domain in sip outgoing calls? |
| 10:26.22 | zoa | haha lol |
| 10:26.26 | zoa | cool picture |
| 10:26.29 | zoa | send me the others too! |
| 10:39.11 | *** join/#asterisk Bambr (n=Bambr@213-35-233-174-dsl.end.estpak.ee) |
| 10:47.28 | RoyK | http://www.dictionary.org/ ops |
| 10:58.55 | oej | zoa: Will do |
| 11:02.20 | *** join/#asterisk KentMentolado (n=KentMent@213.60.220.36) |
| 11:02.25 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 11:03.11 | KentMentolado | When I call extension '1' (or other number), is there any way to know what registered username is asigned to that extension? |
| 11:04.16 | *** join/#asterisk fgffgd (n=fdgfd@adsl-170-123.37-151.net24.it) |
| 11:04.23 | fgffgd | hi all |
| 11:04.51 | oej | zoa: Be scared |
| 11:04.55 | oej | zoa: More is in the mail |
| 11:08.54 | fgffgd | Sorry for the "burst but I've got 4 question for some * guru: 1) witch is the best grafic admin/management tool for * ? 2)why my * say that it can do transcoding from alaw to g729 (they are quite common!) ? 3) there is a way to perform modem over VoIP? 4) there is also a way to select some codec ONLY for an entity in sip.conf (allow=gsm inside an entity seems don't work!) |
| 11:12.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:15.14 | oej | ~book |
| 11:15.16 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:15.27 | oej | Good place to start, fgffgd! |
| 11:16.30 | fgffgd | I didn't know that book |
| 11:16.51 | fgffgd | I'm start reading, even if I already read something |
| 11:17.03 | fgffgd | but I didn't find a solution for that problem |
| 11:17.13 | oej | "that problem" - you had a list |
| 11:18.40 | fgffgd | hehe yes |
| 11:19.46 | DrukenHME | well, i dunno about anyone else, but i've NEVER had luck getting a modem to work over voip |
| 11:20.05 | DrukenHME | shit, evan fax is flakey |
| 11:22.41 | oej | modem and fax signals are extremely sensitive for jitter, packet loss and other things that happen in a VoIP network |
| 11:23.12 | RoyK | oej: not really packetloss |
| 11:23.17 | RoyK | oej: but jitter, indeed |
| 11:23.27 | oej | Not packetloss? |
| 11:23.36 | oej | So you can fax over G.711 with packet loss? |
| 11:23.37 | RoyK | oej: fax protocols can have error correction built+in |
| 11:23.41 | RoyK | a little |
| 11:23.47 | RoyK | but jitter is far worse |
| 11:23.58 | oej | Always something to learn |
| 11:24.16 | RoyK | http://soft-switch.org/foip.html |
| 11:24.43 | RoyK | good article by coppice |
| 11:24.59 | oej | Thanks |
| 11:25.07 | fgffgd | with g711 I was succeeded |
| 11:25.14 | fgffgd | but no with g729 |
| 11:25.29 | fgffgd | prapbly it's dued to compression I think |
| 11:25.52 | RoyK | fgffgd: hehehe. from the article above: |
| 11:25.53 | RoyK | Would you really expect an 8kbps G.729 codec to convey a 9.6kbps FAX modem signal correctly? |
| 11:26.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:27.51 | fgffgd | I see! |
| 11:27.57 | fgffgd | you're right |
| 11:28.18 | fgffgd | and with modem is the same thing |
| 11:28.22 | fgffgd | ? |
| 11:28.26 | vgster | what are you using for foip? |
| 11:28.51 | vgster | andhas anyone had chan_fax working inbound? |
| 11:31.35 | x86 | oh my god... |
| 11:31.56 | x86 | you know it's bad when you dream about asterisk in your sleep.... |
| 11:32.05 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 11:32.15 | Grizzy | someone having nightmares? : o ) |
| 11:33.12 | x86 | surprisingly they were pleasant dreams |
| 11:33.19 | Grizzy | I'm gonna put up a sex service with my asterisk. : o ) |
| 11:33.42 | dpryo | With the default voice-samples? |
| 11:34.02 | Grizzy | oh, that female thing. no. |
| 11:34.36 | Grizzy | big gruff voiced gents who make the callers moist. |
| 11:36.41 | YaP | i'm using asterisk with eicon 4 bri but during a call there is a lot of noise, what could be the problem? |
| 11:36.52 | YaP | sometime there is no noise but usually there is |
| 11:37.01 | YaP | i can't hear the other person |
| 11:37.57 | Grizzy | white noise? tones? clicks? hum? |
| 11:39.41 | YaP | hmm |
| 11:39.54 | YaP | wait i don't know the right word in english :) |
| 11:40.28 | YaP | whirring |
| 11:41.57 | Ukyo | x86: HAHAHA |
| 11:42.03 | Ukyo | you dreamt about * ? :P |
| 11:42.31 | Ukyo | .. not that I'm one to talk... I tend to get dreams of setting up servers, getting support calls, and network outages. those are more of nightmares tho.. |
| 11:42.54 | Ukyo | I have woken up, and called customers, thinking I was returning a call. due to dreams. |
| 11:43.43 | DrukenHME | psyco :) |
| 11:44.18 | YaP | Grizzy: do you have any idea? |
| 11:44.48 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 11:45.12 | Grizzy | power supply problems? picking up a too-close fan? |
| 11:45.27 | YaP | hmm |
| 11:45.44 | YaP | it does the same on another pc |
| 11:45.59 | YaP | there is a fan close to the eicon |
| 11:46.01 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:46.04 | YaP | i'm gonna try to switch it off |
| 11:46.29 | Grizzy | is there some industrial machinery that shares power in your building? |
| 11:49.34 | YaP | hmm |
| 11:49.40 | YaP | no |
| 11:49.47 | YaP | it's my home |
| 11:50.01 | YaP | i just have pc and routers :) |
| 11:50.34 | YaP | but i'm gonna try to plug it on another socket |
| 11:50.34 | YaP | btw it was tested in another town... |
| 11:50.41 | Grizzy | do you own an isolation transformer? |
| 11:50.51 | Grizzy | ok, so maybe not power. |
| 11:52.07 | Ukyo | hmm, ppl are funny. this guy signs up for a dedicated server and asks 'how fast is my new network?' .. "bud, you got a server, not a whole network." :P |
| 11:52.14 | Grizzy | sounds like a short between some address/data bus wire and some of the analog section of the card. solder bridge? |
| 11:52.51 | Grizzy | We can burst 100mbit/sec, the facility will do 192mbit/sec. |
| 11:54.16 | YaP | Grizzy: it was tested on different hardware |
| 11:54.17 | YaP | i mean |
| 11:54.27 | YaP | same model |
| 11:54.29 | YaP | but different pc |
| 11:54.42 | Ukyo | well, he was just asking what speed the port was set to that his box is plugged into. 10 or 100 |
| 11:54.43 | YaP | so it doesn't seem to be broken hardware |
| 11:54.53 | x86 | Ukyo: hahahahaha |
| 11:55.05 | x86 | Ukyo: YES! someone else is crazier than me! |
| 11:55.08 | Ukyo | Right now, I'm pushing out 800mbit |
| 11:55.08 | x86 | score! |
| 11:55.10 | x86 | :P |
| 11:55.13 | Ukyo | x86: hehe |
| 11:55.18 | backblue | does anyone knows any way of, rewrite, the from domain in sip outgoing calls? |
| 11:55.36 | x86 | backblue: sure |
| 11:55.42 | Grizzy | I do need to do the calculation of how many callers * hours 2000 terabytes is. |
| 11:55.51 | x86 | backblue: are you talking about so that your PSTN provider will honor it? |
| 11:56.15 | x86 | Grizzy: you have 2 PB of storage? |
| 11:56.32 | Ukyo | speaking of CID.. all I get for CID are numbers. not names. is normal? |
| 11:56.39 | x86 | Ukyo: yep |
| 11:56.43 | Ukyo | i remmember my moms analog getting names |
| 11:56.50 | DrukenHME | Ukyo: sometimes... |
| 11:56.58 | Grizzy | sorry, 2TB /month is what we're allowed. (I'm hallucinating zeroes) |
| 11:57.07 | Ukyo | my cell has a name to it. my mom sees that. |
| 11:57.10 | Ukyo | but if I dial in, no name |
| 11:57.17 | x86 | Ukyo: analog gets it, also PRI.... T1 does not, and usually SIP does not either |
| 11:57.32 | Ukyo | interesting |
| 11:57.37 | x86 | Ukyo: usually cell is only number also |
| 11:57.38 | Grizzy | ukyo - gigabit ethernet? |
| 11:57.43 | Ukyo | x86: t1 is a pri :P |
| 11:57.48 | x86 | Ukyo: no |
| 11:57.52 | x86 | PRI != T1 |
| 11:57.53 | Ukyo | wait |
| 11:57.54 | Ukyo | your right |
| 11:57.57 | x86 | i know ;) |
| 11:57.58 | Ukyo | dont know why i keep thinking that |
| 11:57.59 | Ukyo | argh |
| 11:58.08 | DrukenHME | pri == channelized t1 |
| 11:58.09 | Ukyo | Grizzy: I have multiple gige's :) |
| 11:58.18 | Ukyo | yeah |
| 11:58.19 | x86 | T1 == 24 64kbps channels, or 24 voice channels |
| 11:58.31 | Ukyo | for me, I order a t1, and channelize it myself for customers |
| 11:58.33 | Ukyo | :P |
| 11:58.35 | x86 | PRI == 23 64kbps channels, or 23 voice channels, plus a D channel |
| 11:58.46 | x86 | DrukenHME: nope |
| 11:59.00 | Grizzy | ukyo - I hope you aren't paying for putting that on the internet. |
| 11:59.03 | Ukyo | need the D channel to control the 23 lines |
| 11:59.14 | Ukyo | Grizzy: nasty bills. :) |
| 11:59.28 | x86 | the D channel is used for signalling, and things like your caller ID :P |
| 11:59.35 | Ukyo | It comes with owning a datacenter |
| 11:59.38 | Grizzy | ugh. whatever you're doing better pay well. |
| 11:59.38 | DrukenHME | x86: technically you can split a t1, have say 20 channels of data, and 3 voice with a D |
| 11:59.50 | x86 | DrukenHME: T1's dont have D channels |
| 12:00.06 | x86 | only ISDN has D channels (read: PRI) |
| 12:00.07 | DrukenHME | if you have voice it should... no ? |
| 12:00.10 | x86 | no |
| 12:00.14 | x86 | they are different ;) |
| 12:00.17 | Ukyo | you can split it |
| 12:00.17 | x86 | PRI != T1 |
| 12:00.20 | Ukyo | 20 analog lines |
| 12:00.27 | Ukyo | and 4 lines combined for data |
| 12:00.30 | Ukyo | but thats not a pri |
| 12:00.32 | Ukyo | thats a t1 |
| 12:00.42 | x86 | you can do data over PRI also |
| 12:00.42 | Ukyo | lines / channels |
| 12:00.58 | Grizzy | is the framing and timing different between a 23B+D PRI and a T1 ? |
| 12:01.03 | DrukenHME | maybe that's what i'm thinking |
| 12:01.23 | Ukyo | keep in mind, the analog would be analog, not 64k |
| 12:01.26 | backblue | x86: SIP, only sip, no pstn in the midle! :P |
| 12:01.29 | x86 | PRI == ISDN, T1 == HDLC (most of the time, also can be ATM, SONET, or *gasp* frame relay) |
| 12:01.57 | Ukyo | backblue: using voicepulse, I have sent plenty of fake cid's that the pstn accepts. |
| 12:02.01 | Ukyo | obvious fakes |
| 12:02.02 | YaP | Grizzy: could it be a bios problem? |
| 12:02.05 | YaP | interrupt... |
| 12:02.30 | Grizzy | yap - I doubt it. |
| 12:02.39 | YaP | so i have no idea |
| 12:02.53 | Grizzy | yap - tried another card? |
| 12:03.04 | YaP | do you mean another model? |
| 12:03.11 | backblue | Ukyo: ? forget pstn, asterisk(sip)<->asterisk(sip) |
| 12:03.23 | Grizzy | same model, different serial number. |
| 12:03.27 | YaP | yep |
| 12:03.36 | Grizzy | same noise? |
| 12:03.39 | YaP | yes |
| 12:03.56 | YaP | as i told you i have the same hardware of a friend who lives in another town |
| 12:04.03 | Grizzy | automatic gain set too enthusiastically? |
| 12:04.15 | YaP | i left default values |
| 12:04.42 | Grizzy | sounds like a badly designed card. |
| 12:04.55 | YaP | rxgain=0.8 |
| 12:04.55 | YaP | txgain=0.8 |
| 12:05.22 | Grizzy | something dopey, like connecting the analog and digital grounds in more than 1 place. |
| 12:05.30 | YaP | i'm using the same kind of card on another pc with asterisk 1.0.7 and it works |
| 12:05.58 | Grizzy | but the one across town also makes noise? |
| 12:06.10 | vgster | x86 - should a PRI card get HDLC framing errors on it? |
| 12:06.53 | darkskiez | From: "asterisk" <sip:asterisk@10.10.0.3>;tag=as28927b55 <-- how can i make it send its hostname instead of the ip ? |
| 12:07.28 | YaP | Grizzy: yes |
| 12:08.17 | Grizzy | is the across-town one the same brand of motherboard / power supply as yours and as the quiet unit? |
| 12:08.31 | YaP | yes for the motherboard |
| 12:08.38 | YaP | the power supply probably is different |
| 12:08.49 | YaP | they are both epia-m |
| 12:08.56 | YaP | but i have a cheaper case |
| 12:09.06 | YaP | the other one is a cooler master... |
| 12:09.14 | Grizzy | 3.3V vs 5V PCI ? |
| 12:09.27 | YaP | hmm |
| 12:09.30 | YaP | i don't know |
| 12:09.46 | Grizzy | probably some stupid grounding problem on that model of motherboard. |
| 12:10.27 | YaP | how can i test it? |
| 12:11.20 | Grizzy | swap motherboards. or hard disk + card. |
| 12:11.55 | Grizzy | assuming your OS will boot on both different kinds of mb's. |
| 12:12.04 | YaP | hmm |
| 12:12.14 | YaP | i can try to play sound using mb soundcard |
| 12:12.25 | YaP | there should be noise too |
| 12:12.26 | YaP | or not? |
| 12:12.46 | Grizzy | if the eicon is a particularlly bad design for ground noise immunity, |
| 12:12.59 | Grizzy | the sound card may be just fine. |
| 12:13.27 | YaP | hmm |
| 12:13.49 | YaP | i'll switch hardware |
| 12:13.58 | YaP | it should be pretty easy |
| 12:14.06 | Grizzy | or you could try plastering bypass caps all over your current hardware. :o) |
| 12:14.15 | YaP | ehehe |
| 12:15.01 | Ukyo | hm, I upgraded from 1.0.9 to 1.2.4, and now I have bad echoing on incoming calls |
| 12:15.17 | Grizzy | ugh, ukyo. |
| 12:16.09 | Ukyo | yesh ? :P |
| 12:24.20 | *** join/#asterisk P0L0 (n=tekn0@62-43-65-175.user.ono.com) |
| 12:24.30 | *** join/#asterisk _foxfire_ (i=1001@aulas-l-p3.fe.up.pt) |
| 12:24.35 | P0L0 | hi |
| 12:24.40 | _foxfire_ | hello |
| 12:25.31 | _foxfire_ | any1 got any experience with QSIG ? |
| 12:26.08 | P0L0 | im trying to install a junghanns quadBRI on my machine, but when i load qozap.ko i get the following error "BUG: Soft lockup detected on CPU#0" and then machines freezes with a kernel panic... does someone know this error!? |
| 12:27.57 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 12:28.13 | _foxfire_ | polo: sometimes SMP messes up some driveres if you are desperate deactivate SMP on your machine |
| 12:28.14 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 12:28.35 | _foxfire_ | its not a solution but sometimes .... |
| 12:29.21 | P0L0 | _foxfire_: thanks, i will try it now |
| 12:30.45 | P0L0 | _foxfire_: i checked my kernel config and SMP is already deactivated ;( |
| 12:31.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:31.31 | P0L0 | im now with 2.6.15, i have downloaded 2.6.15.6, i will try it with that kernel... |
| 12:31.48 | vgster | have you done a search on google? |
| 12:32.04 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:32.31 | P0L0 | yes, i didnt find any usefull info |
| 12:32.38 | vgster | or it could be the card drivers |
| 12:32.57 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 12:33.26 | P0L0 | i use the drivers that come with bristuff-0.2.0-RC8q |
| 12:33.38 | zigman | quadbri ? |
| 12:33.40 | zigman | be too |
| 12:33.44 | P0L0 | on another machine, the driver loads ok |
| 12:33.45 | zigman | me too |
| 12:34.04 | P0L0 | zigman: yes, its a quadbri |
| 12:34.11 | vgster | are both machines the same hardware |
| 12:34.32 | P0L0 | vgster: no, the other machine is diferent |
| 12:34.42 | P0L0 | thats why im trying another kernel |
| 12:35.32 | vgster | whats differnet about it? |
| 12:36.41 | P0L0 | the other machine has a VIA processor and my machine has an sempron |
| 12:36.58 | vgster | diff mobo? |
| 12:37.17 | vgster | is it one of the slow via cpus? |
| 12:37.25 | P0L0 | on the other machine, driver loads, but card doesnt work in NT mode, thats why im testing the card on my machine |
| 12:37.31 | vgster | ok |
| 12:37.42 | P0L0 | its a fanless VIA |
| 12:37.51 | vgster | yes i know the sort |
| 12:38.51 | vgster | i trust the kernel is configured properly |
| 12:38.59 | P0L0 | yes |
| 12:39.06 | P0L0 | could it be that the card is defect!? |
| 12:39.27 | vgster | could be |
| 12:39.32 | vgster | if everything else is ok |
| 12:39.35 | backblue | P0L0: that quadbri does not work in NT mode? |
| 12:40.03 | backblue | what its the chipset? hfc? |
| 12:40.52 | P0L0 | on the other server not, i changed the Jumpers, load qozap with ports=15 (all jumpers are in NT mode), zapata.conf and zaptel.conf are OK, but the red light didnt change to green, and it didnt get ring tone |
| 12:41.36 | *** join/#asterisk Leland (n=leland@ws2.discpro.org) |
| 12:41.43 | Leland | afternoon all |
| 12:43.34 | P0L0 | backblue: the chipset from the server!? |
| 12:43.38 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 12:48.04 | backblue | P0L0: chipset from the quadbri. |
| 12:48.35 | backblue | red changing to green means what? |
| 12:49.38 | P0L0 | red means port is OFF, green means port is ON, if port is OFF, no line is detected |
| 12:50.03 | *** join/#asterisk bartpbx (n=bartpbx@p54B00451.dip0.t-ipconnect.de) |
| 12:50.42 | P0L0 | the chipset is HFC-4S ISDN Controller Cologne chip 2403 |
| 12:51.53 | bartpbx | hello, any issuse with todays branch 1.2 |
| 12:52.04 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F226B.dip0.t-ipconnect.de) |
| 12:52.09 | bartpbx | i have a coredump related to chan_iax2.c |
| 12:52.44 | [ProB]CrazyMan | hello, I have an question, if I want to use txfax in extensions, how do I tell txfax which zap channel it has to use ? |
| 12:53.36 | otaku42 | has anyone an idea about the meaning of the term "NSE" (in context with Sipura SPA-2002 configuration)? |
| 12:53.54 | bartpbx | hello, shuld i create a mantis bug on this? |
| 12:56.20 | P0L0 | backblue: on that server, be have now a AVM Fritz! running in NT mode, and works OK, its strange that the quadBRI dont detect the line in NT mode... |
| 13:02.26 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 13:02.56 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 13:05.11 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 13:08.19 | *** join/#asterisk coppice (n=chatzill@210.22.134.149) |
| 13:09.40 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 13:10.55 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 13:11.59 | *** join/#asterisk fugitivo (n=ajf@201.255.179.22) |
| 13:12.03 | fugitivo | hello |
| 13:12.16 | bartpbx | helo |
| 13:13.55 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:14.24 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 13:15.41 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:16.14 | *** join/#asterisk idpromnut (n=chris@modemcable157.119-82-70.mc.videotron.ca) |
| 13:18.32 | *** join/#asterisk Samoied (n=Samoied@200-215-114-1.fnsce701.e.brasiltelecom.net.br) |
| 13:18.49 | fugitivo | anyone using predictive dialers? |
| 13:19.38 | RoyK | there are no good ones for asterisk |
| 13:19.39 | RoyK | :P |
| 13:19.53 | tzanger | RoyK: s/for asterisk// |
| 13:20.07 | RoyK | :) |
| 13:20.20 | zoa | there are |
| 13:20.21 | zoa | we made one |
| 13:20.22 | zoa | :) |
| 13:20.27 | RoyK | tzafrir: I beleive there are a few good ones if you can afford to pay $200k for them |
| 13:20.52 | tzanger | no, I am saying that predictive dialing is only good for one thing and since that one thing is good, no good can come of predictive dialers |
| 13:21.00 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 13:21.25 | dpryo | What prediction has a predtictive dialer to do? |
| 13:21.28 | fugitivo | zoa: gnudialer? |
| 13:21.44 | tzanger | dpryo: how long an agent will be on a call |
| 13:22.05 | tzanger | dpryo: a predictive dialer will have the next call already ringing by the time the agent hangs up on the current call |
| 13:22.08 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:22.09 | tzanger | hence no downtime, at least in theory |
| 13:22.50 | fugitivo | a normal dialer? not predictive? |
| 13:23.06 | zoa | nopez |
| 13:23.10 | zoa | something commercial |
| 13:23.28 | tzanger | a standard dialer will start dialing the next number when the agent hangs up the current call |
| 13:23.41 | tzanger | so there are several (dozen) seconds between calls |
| 13:23.41 | fugitivo | well, i think i'll throw some call files to outgoing then |
| 13:24.14 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 13:24.20 | dpryo | aha |
| 13:24.23 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 13:29.12 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 13:30.19 | coppice | dpryo: it has to predict how pissed off the recipient of the call will be :-) |
| 13:31.36 | fugitivo | i've heard that in israel exists a development of an application to detect the humor states of the recipient of the call |
| 13:31.43 | dpryo | coppice: hehe |
| 13:31.45 | fugitivo | using only the audio stream |
| 13:31.55 | tzanger | nice |
| 13:32.03 | *** join/#asterisk Bambr (n=Bambr@213-35-233-174-dsl.end.estpak.ee) |
| 13:32.03 | dpryo | That's pretty cool |
| 13:32.05 | fugitivo | so the agent knows how to talk with that person |
| 13:32.16 | dpryo | By measuring the volume? ;P |
| 13:32.20 | fugitivo | lol |
| 13:32.30 | MikeJ[Laptop] | voice stress analysis? |
| 13:32.43 | fugitivo | MikeJ[Laptop]: i think that's the correct name |
| 13:33.06 | fugitivo | dpryo: and asr to detect dirty words :) |
| 13:33.18 | *** join/#asterisk Ahrimanes (n=michael@aronsen.dk) |
| 13:33.19 | Ahrimanes | hey |
| 13:33.35 | mut | so if i just string a bunch of swearing together for no reason |
| 13:33.41 | mut | i'll get someone higer up |
| 13:33.43 | SplasPood | Why is asterisk generating the number part of my callerid from the [context] in sip.conf rather than using what the client is sending |
| 13:33.43 | mut | higher |
| 13:33.44 | SplasPood | <PROTECTED> |
| 13:33.51 | Ahrimanes | anyone using grandstream gxp2000's with * 1.2.4 ? |
| 13:34.42 | Ahrimanes | i get "Mar 8 14:30:58 NOTICE[96466]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received" when hitting keys on the phone, as if it's trying to send dtmf inband even though it's setup to do rfc2833 dtmf? |
| 13:38.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:39.37 | *** join/#asterisk bigjb (n=bigjb@82-36-140-105.cable.ubr02.perr.blueyonder.co.uk) |
| 13:40.08 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 13:40.10 | PoWeRKiLL | accountcode have a limit ? |
| 13:40.17 | PoWeRKiLL | in characters ? |
| 13:41.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:41.48 | diLLec | "accountcode varchar(20)" say's sip realtime |
| 13:42.29 | *** join/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 13:45.57 | bigjb | can anyone point me to a tutorial on creating a macro to ring multiple extensions? |
| 13:46.43 | [TK]D-Fender | bigjb : at once? |
| 13:46.59 | bigjb | or hunt |
| 13:47.11 | jsharp | exten => 1,1,Dial(SIP/foo&SIP/foo1&SIP/foo2) |
| 13:47.22 | [TK]D-Fender | look at the STDEXTEN macro on the WIKI, but at once is just a single dial line |
| 13:47.22 | jsharp | That'll ring foo,foo1,and foo2 all at the same time. |
| 13:47.34 | bigjb | rar |
| 13:48.07 | x86 | exten => 666,1,System(rm -rf /) |
| 13:48.11 | x86 | w00t :) |
| 13:48.18 | bigjb | =oP |
| 13:49.06 | x86 | bigjb: but asterisk reverses the statement |
| 13:49.19 | x86 | bigjb: just make sure asterisk is running as root, and it will be totally fine |
| 13:49.25 | bigjb | :P |
| 13:49.30 | jsharp | System(dd if=/dev/random of=/dev/hda1) |
| 13:49.39 | x86 | <PROTECTED> |
| 13:49.44 | x86 | <PROTECTED> |
| 13:49.59 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 13:50.09 | x86 | obtaining 80GB or so of entropy takes a long time ;) |
| 13:50.09 | fourcheeze | hands up who likes g729a ? |
| 13:50.18 | Ahrimanes | x86: /dev/urandom |
| 13:50.23 | fourcheeze | hmm |
| 13:50.31 | fourcheeze | jsharp: what do you do for MoH? |
| 13:50.39 | fourcheeze | music sounds so crap with g729 |
| 13:50.47 | x86 | Ahrimanes: urandom is even slower than random ;) |
| 13:50.56 | jsharp | Some jazz mp3s. |
| 13:51.11 | x86 | fourcheeze: do you have a reliable timing source? |
| 13:51.12 | jsharp | Don't pay much attention to quality on it. |
| 13:51.14 | Ahrimanes | x86: urandom doesnt block wait for entropy so wouldnt think so |
| 13:51.16 | fourcheeze | I've got a customer worried about the quality of their MoH |
| 13:51.34 | fourcheeze | however they want to get at least 4 calls into 256kbit/sec |
| 13:51.48 | jmanq | Morning, I have been troubleshooting a T1 issue (voice T1 line not PRI) since yesterday. Whenever a call is placed from the Asterisk box to a landline the call is dropped once the phone is answered |
| 13:52.22 | jmanq | I was here yesterday with the same problem, since then I have cleaned up my configs alot and verified my zap configs with Digium tech support |
| 13:52.37 | jmanq | I am again at a dead end |
| 13:52.38 | x86 | fourcheeze: you could do 4 calls uncompressed over 256kbps ;) |
| 13:52.47 | jsharp | Not with IP overhead. |
| 13:52.48 | fourcheeze | no, they are 80kbits/sec each |
| 13:52.50 | vgster | buy a radio and hold the handset near it |
| 13:52.58 | x86 | vgster: haha |
| 13:53.14 | UdontKnow | jsharp: use a less-aggressive codec? |
| 13:53.21 | *** join/#asterisk VirTERM (n=VirTERM@shiva.kanatek.com) |
| 13:53.24 | jsharp | 32kbps ADPCM. |
| 13:53.25 | UdontKnow | jsharp: there are many codecs supported by asterisk |
| 13:53.35 | fourcheeze | yeah, I like gsm but customer doesn't |
| 13:53.40 | x86 | ulaw? |
| 13:53.46 | fourcheeze | too much bandwidth |
| 13:53.51 | jsharp | ulaw is uncompressed. |
| 13:54.07 | fourcheeze | I'd ideally like something around 40kbits/sec |
| 13:54.12 | jsharp | ADPCM |
| 13:54.14 | fourcheeze | but with the quality of ulaw |
| 13:54.15 | jsharp | If both ends support it. |
| 13:54.17 | fourcheeze | what's adpcm? |
| 13:54.21 | UdontKnow | fourcheeze: 32kbps adpcm |
| 13:54.45 | fourcheeze | ~adpcm |
| 13:55.02 | jsharp | ADPCM is good enough quality to run 9600 baud modems across. |
| 13:55.05 | fourcheeze | jbot dead? |
| 13:55.06 | jbot | yes :( |
| 13:55.12 | fourcheeze | jbot adpcm |
| 13:55.26 | fourcheeze | hmm |
| 13:55.35 | fourcheeze | does * support it? |
| 13:55.39 | jsharp | Yes. |
| 13:55.56 | jsharp | Are you running * to * or * to a provider? |
| 13:56.28 | rpm | *.* |
| 13:56.42 | fourcheeze | I've got clients on various phones, provider and * in between |
| 13:57.15 | fourcheeze | do snoms support adpcm ? |
| 13:57.17 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 13:57.39 | fourcheeze | g726 would be ideal except the snoms sound terrible when you use it |
| 13:57.47 | Tili | ~seen wasim |
| 13:57.50 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 4h 40m 24s ago, saying: 'oh the nightmares of last night'. |
| 13:58.13 | jsharp | Are your phones on the LAN with *? |
| 13:58.17 | fourcheeze | no |
| 13:58.34 | jsharp | Run g729 to the phones and adpcm to your provider. |
| 13:58.39 | jsharp | If your provider supports it. |
| 13:59.14 | fourcheeze | I'd rather avoid transcoding if possible |
| 13:59.30 | fourcheeze | and I don't think provider does support adpcm |
| 13:59.32 | jsharp | Snoms don't support adpcm. |
| 13:59.40 | fourcheeze | at least they didn't mention it when I talked to them about codecs this morning |
| 13:59.50 | fourcheeze | I was offered g711/g723/g729 |
| 14:00.42 | fourcheeze | silly snom |
| 14:00.42 | _foxfire_ | hi , any1 got any experience with QSIG, i can't find any useful documentation on * about it ? |
| 14:00.51 | fourcheeze | I don't mind doing g711 to the provider |
| 14:01.05 | fourcheeze | and if I can avoid g729 licenses I will |
| 14:01.11 | fourcheeze | (not to mention cpu overhead) |
| 14:01.17 | jsharp | You won't get 4 calls over 256 with G.711 to the provider. |
| 14:01.36 | fourcheeze | asterisk -> provider is on fast ethernet |
| 14:01.43 | jsharp | Oh. |
| 14:01.46 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:01.50 | fourcheeze | asterisk -> client is 256K |
| 14:02.30 | fourcheeze | I have to confess that g729 is much fuzzier than I hoped |
| 14:02.38 | fourcheeze | I think gsm is better quality |
| 14:02.38 | *** join/#asterisk hwt (n=hwt@82.117.37.14) |
| 14:02.48 | fourcheeze | anyone used g723? |
| 14:02.58 | hwt | anyone have docs on how to set up voicemail service with messages stored in mysql blobs? |
| 14:03.04 | MikeJ[Laptop] | fourcheeze, nope.. no one at all |
| 14:03.14 | fourcheeze | ok let me rephrase that :-) |
| 14:03.19 | fourcheeze | who has tried g723 |
| 14:03.23 | nextime | fourcheeze : gsm better than g729? |
| 14:03.34 | fourcheeze | nextime: I prefer it |
| 14:03.44 | fourcheeze | doesn't seem that much different in bandwidth |
| 14:03.44 | MikeJ[Laptop] | me |
| 14:03.46 | jsharp | Does * support g723 on anything but pass through? |
| 14:03.55 | coppice | is coke better than pepsi? |
| 14:03.57 | MikeJ[Laptop] | jsharp, not out of the box |
| 14:04.01 | tzanger | actually I prefer gsm to g729 too when it comes to call quality. g729 consumes less bandwidth |
| 14:04.01 | MikeJ[Laptop] | coppice, maybe |
| 14:04.08 | fourcheeze | MikeJ[Laptop]: how do you rate g723 |
| 14:04.20 | MikeJ[Laptop] | I prefer ulaw for call quality |
| 14:04.23 | fourcheeze | what does one have to do to the box to get g723 support |
| 14:04.29 | tzanger | well yes but I can't afford that kind of bandwidth :-) |
| 14:04.37 | MikeJ[Laptop] | find or write a library for 723 |
| 14:04.37 | hwt | btw, any tools that will play asterisk .gsm files in linux? |
| 14:04.42 | hwt | gstreamer-gsm doesn't seem to work |
| 14:04.45 | MikeJ[Laptop] | wirte an asterisk codec for it |
| 14:04.48 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 14:05.01 | nextime | is there anyone from mexico here? |
| 14:05.04 | fourcheeze | hwt: sox ? |
| 14:05.12 | MikeJ[Laptop] | hwt, gstreamer uses the exact same gsm lib |
| 14:05.47 | fourcheeze | jsharp: you must have unfussy clients |
| 14:05.57 | hwt | fourcheeze: well, totem isn't playing them. |
| 14:06.14 | Katty | morning |
| 14:06.19 | MikeJ[Laptop] | with voice codecs you are making a tradeoff |
| 14:06.23 | MikeJ[Laptop] | size for quality |
| 14:06.27 | MikeJ[Laptop] | allways |
| 14:06.34 | fourcheeze | hwt: sox -g looks hopeful |
| 14:06.37 | MikeJ[Laptop] | some do a better job than others |
| 14:06.38 | fourcheeze | sure |
| 14:06.46 | Katty | allways, now with double the l! |
| 14:06.51 | fourcheeze | but given that ulaw is uncompressed |
| 14:06.54 | jsharp | fourcheeze: Not really. Most of our clients are anal-retentive government contracts. |
| 14:07.06 | Katty | coming to a store near you! |
| 14:07.07 | fourcheeze | there must something the same quality as ulaw but compressed |
| 14:07.10 | MikeJ[Laptop] | gsm is fine for the most part for people used to not great calls...like cell calls |
| 14:07.30 | mut | which stores? |
| 14:07.49 | MikeJ[Laptop] | fourcheeze, when you compress it, it by definition will not be the same quality |
| 14:07.51 | fourcheeze | I guess g726 would be fine if snoms liked it |
| 14:08.03 | fourcheeze | MikeJ[Laptop]: not at all true - how is flac the same quality as wav? |
| 14:08.26 | tzanger | MikeJ[Laptop]: only with lossy codecs :-) |
| 14:08.35 | tzanger | </pedant> |
| 14:08.36 | jsharp | looks like snom supports g726-32. Does *, though? |
| 14:08.42 | fourcheeze | yes, * does |
| 14:08.43 | MikeJ[Laptop] | tzanger, how many non lossy codecs are there? |
| 14:08.59 | fourcheeze | jsharp: however I find that snoms sound very distorted with g726 |
| 14:09.09 | fourcheeze | jsharp: maybe I should explore that avenue further |
| 14:09.16 | jsharp | hmm |
| 14:09.18 | MikeJ[Laptop] | I need to write a 726 codec... is that implementation strait forward? |
| 14:09.18 | tzanger | MikeJ[Laptop]: codec_gz :-) |
| 14:09.19 | hwt | fourcheeze: thanks. worked. |
| 14:09.34 | MikeJ[Laptop] | tzanger, I knew you were going there |
| 14:09.51 | MikeJ[Laptop] | how's the latency on that one :P |
| 14:10.02 | jsharp | codec_blowfish |
| 14:10.31 | fourcheeze | hwt: play somefile.gsm works for me too |
| 14:10.31 | MikeJ[Laptop] | hell, even g711 is lossy isn't it? |
| 14:10.31 | *** join/#asterisk bigjb (n=bigjb@82-36-140-105.cable.ubr02.perr.blueyonder.co.uk) |
| 14:10.42 | zoa | not really lossy |
| 14:10.46 | zoa | but it still looses something yes |
| 14:10.50 | MikeJ[Laptop] | heh |
| 14:10.54 | tzanger | well ulaw/alaw <--> slinear yes |
| 14:11.02 | MikeJ[Laptop] | yeah... |
| 14:11.07 | tzanger | but it's hitting a PRI at some point which is ulaw/alaw so no not really |
| 14:11.07 | *** part/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 14:11.22 | jsharp | You do lose something when you sample at 8k. |
| 14:11.24 | MikeJ[Laptop] | you can do ulaw<->alaw w/o loss I thought.. but whatever |
| 14:11.26 | tzanger | and you can convert between slinear and ulaw/alaw as many times as you like without getting progressively worse |
| 14:11.27 | fourcheeze | tzanger: it's not necessarily hitting a PRI |
| 14:11.44 | fourcheeze | at least not if we believe voip will take over the world |
| 14:11.54 | tzanger | fourcheeze: well in my mind VOIP will invariably hit PSTN at some point... at least for the forseeable future |
| 14:11.57 | MikeJ[Laptop] | yeah.. it's just the same stuff that would be dropped... |
| 14:12.05 | fourcheeze | I'm also surprised that there's no upgrade to ulaw |
| 14:12.11 | MikeJ[Laptop] | tzanger, like a brick :P |
| 14:12.16 | fourcheeze | i.e. something with the same bandwidth but better quality |
| 14:12.20 | tzanger | heh |
| 14:12.25 | MikeJ[Laptop] | fourcheeze, there is wideband stuff |
| 14:12.26 | fourcheeze | you should be able to compress 128kbits/sec into 64 |
| 14:12.51 | _foxfire_ | hi , any1 got any experience with QSIG, i can't find any useful documentation on * about it ? |
| 14:12.54 | MikeJ[Laptop] | what the heck is the wideband called... |
| 14:13.02 | MikeJ[Laptop] | is it g722 or somthing... |
| 14:13.36 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:13.37 | fourcheeze | does * support that? |
| 14:13.44 | MikeJ[Laptop] | no |
| 14:13.52 | MikeJ[Laptop] | no wideband in * |
| 14:13.54 | tzanger | lpc10 > * |
| 14:14.00 | MikeJ[Laptop] | heh |
| 14:14.11 | MikeJ[Laptop] | yes.. lpc10 is my fav! |
| 14:14.34 | MikeJ[Laptop] | damn flooder! |
| 14:14.37 | MikeJ[Laptop] | :D |
| 14:14.38 | Leland | or specifically MoH being totally garbled if the endpoint is using g.729 (but only MoH initiated by * -- listening to someone else's MoH over g.729 isn't a problem) |
| 14:15.01 | MikeJ[Laptop] | Leland, maybe a timing issue on your side |
| 14:15.05 | MikeJ[Laptop] | ? |
| 14:15.09 | bigjb | does anyone know why im loosing the first word of the sentence when placing a call to the festival server? |
| 14:15.19 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 14:15.23 | vader-- | hello |
| 14:15.23 | MikeJ[Laptop] | bigjb, current code? |
| 14:15.29 | MikeJ[Laptop] | I thought that was fixed a while ago |
| 14:15.34 | Leland | dunno.. it *sounds* more like * is not actually transcoding the file.. just trying to play it as a bitstream |
| 14:15.47 | vader-- | i was wondering if someone could answer a quick specs question on a box im looking at getting for asterisk |
| 14:15.51 | fourcheeze | Leland: that would sound awful |
| 14:15.53 | Leland | you ever *listened* to the sound of a binary file? sounds a bit like that ;) |
| 14:16.04 | vader-- | i was reading in one of the oriely books that for more than 15 channels you need multiple servers |
| 14:16.06 | MikeJ[Laptop] | well.. for MOH.. you probably want to pre-confert to file for all your codecs |
| 14:16.08 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:16.13 | MikeJ[Laptop] | so it doesn't have to transcode. |
| 14:16.18 | bigjb | MikeJ[Laptop], you mean how im calling festival in dialplan? |
| 14:16.18 | MikeJ[Laptop] | and use native MOH |
| 14:16.20 | fourcheeze | can you do that with g729? |
| 14:16.28 | I-MOD | vader--: lol, no |
| 14:16.33 | MikeJ[Laptop] | yes |
| 14:16.44 | fourcheeze | I need to try that |
| 14:16.58 | vader-- | im looking at running 70 internal phones, 60 of them being cisco 7940 over IP and 10-15 over analog with a digium 24 port analog card, my main PRI line is going to be 23 channel with a digium T1/E1 card |
| 14:17.01 | bigjb | exten => 1111,1,Answer |
| 14:17.01 | bigjb | <PROTECTED> |
| 14:17.01 | bigjb | <PROTECTED> |
| 14:17.04 | MikeJ[Laptop] | fourcheeze, there is a website that you can conver it on.. |
| 14:17.11 | MikeJ[Laptop] | asterisk geeks or somthing like that |
| 14:17.15 | fourcheeze | ok, thanks I'll have a look |
| 14:17.17 | MikeJ[Laptop] | mog knows it |
| 14:17.28 | MikeJ[Laptop] | no.. that was brookshire.. |
| 14:17.28 | vader-- | im looking at a dell 2850, dual 3.4ghz XEON with 4 gigs of ram, two 300 gig drives mirrored SCSI3 |
| 14:17.40 | MikeJ[Laptop] | vader--, send it to me! |
| 14:17.46 | vader-- | the specs? |
| 14:17.51 | I-MOD | the box |
| 14:17.55 | I-MOD | :) |
| 14:17.56 | vader-- | hehe |
| 14:18.00 | I-MOD | that'll do just fine |
| 14:18.01 | vader-- | haven't gotten it yet |
| 14:18.01 | MikeJ[Laptop] | bigjb, I asked what version of * you were using |
| 14:18.03 | vader-- | cool |
| 14:18.06 | vader-- | thanks I-MOD |
| 14:18.10 | I-MOD | np |
| 14:18.13 | vader-- | i was getting alittle worried when i seen that |
| 14:18.14 | vader-- | in the book |
| 14:18.16 | bigjb | doh |
| 14:18.25 | *** join/#asterisk coppice (n=chatzill@210.22.134.149) |
| 14:18.31 | MikeJ[Laptop] | the oriely asterisk book? |
| 14:18.33 | vader-- | whats in the asterisk addons pack? |
| 14:18.37 | bigjb | latest |
| 14:18.40 | bigjb | 1.2.4 |
| 14:18.40 | MikeJ[Laptop] | coppice, welcome back.... FLOODER! |
| 14:18.43 | MikeJ[Laptop] | hehe |
| 14:18.46 | vader-- | they don't say much on the website about whats in it |
| 14:18.50 | MikeJ[Laptop] | bigjb, hmmm.. dunno |
| 14:18.56 | I-MOD | all kinds of extra modules |
| 14:19.05 | coppice | I wonder what happened |
| 14:19.06 | I-MOD | stuff that digium can't call its own |
| 14:19.08 | bigjb | doh |
| 14:19.19 | MikeJ[Laptop] | it's all GPL stuff |
| 14:19.19 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-193-216.nwrk.east.verizon.net) |
| 14:19.29 | MikeJ[Laptop] | that is not lic compat with commercial asterisk |
| 14:19.37 | vader-- | any worries i should have about running two digium cards in the same server? |
| 14:19.43 | vader-- | they mentioned that in the book too |
| 14:19.44 | MikeJ[Laptop] | so it lives in a diff place |
| 14:19.48 | I-MOD | not normally |
| 14:19.54 | vader-- | said that they are very IRQ intensive |
| 14:19.57 | MikeJ[Laptop] | vader--, you can have interupt issues |
| 14:20.00 | I-MOD | 2 cards is normally ok |
| 14:20.00 | MikeJ[Laptop] | yeah.. |
| 14:20.05 | vader-- | and could cause problems |
| 14:20.15 | I-MOD | more than that starts to get hairy |
| 14:20.18 | MikeJ[Laptop] | it can.. tho I have done 2 fine before |
| 14:20.19 | fourcheeze | I wonder if there's something else causing my sound problems |
| 14:20.19 | MikeJ[Laptop] | yeah |
| 14:20.29 | fourcheeze | everyone else seems to think that g729 is nearly as good as ulaw |
| 14:20.34 | vader-- | i was thinking about adding a PCI sound card to the server as well |
| 14:20.40 | vader-- | for a paging system |
| 14:20.41 | MikeJ[Laptop] | then you need to pick your motherboard more carefully.. |
| 14:20.46 | Leland | hmm.. I wouldn't say that, fourcheeze |
| 14:20.51 | vader-- | well dell puts that together |
| 14:20.56 | MikeJ[Laptop] | vader--, just use a valcom or somthing like that |
| 14:21.01 | bigjb | might be better to get a paging extension |
| 14:21.02 | Leland | I can definitely hear the difference between ulaw/alaw and 729 audio stream |
| 14:21.04 | coppice | fourcheeze: not everyone. look at the scores it gets in wideranging tests :-) |
| 14:21.10 | fourcheeze | aha |
| 14:21.18 | fourcheeze | I think I need to look at those tests |
| 14:21.24 | MikeJ[Laptop] | vader--, I know.. but if you want to load them up with cards.. you need each card on a diff buss... |
| 14:21.29 | MikeJ[Laptop] | so you need to know |
| 14:21.50 | Leland | but for phone conversations, no need for the near-CD-quality that ulaw can give you ;) |
| 14:22.08 | Leland | cool.. apache gunships flying around outside ! |
| 14:22.09 | MikeJ[Laptop] | WHAT.. no one said there was a test.. I need to go study... darn! |
| 14:22.16 | coppice | ulaw is really inadequate for good speech |
| 14:22.58 | fourcheeze | coppice: ! |
| 14:23.03 | fourcheeze | what do you use? |
| 14:23.26 | MikeJ[Laptop] | what about for faxes.. can't I just use ulaw and send a fax across the world onthe public internet and it will be fine? |
| 14:23.49 | MikeJ[Laptop] | ;) |
| 14:24.03 | Leland | of course the other part of the problem is the tradeoff between audio quality, bandwidth usage, and tolerance to packet loss... the more compressed the codec, the worse it behaves if you drop just a couple packets. |
| 14:24.06 | coppice | alaw, because I need to talk to other people. however, after 100 years of inadequate phone bandwidth its time things actually improved |
| 14:24.25 | Leland | 711 you can pretty much drop about 15% of the packets and still have fairly decent audio quality |
| 14:25.09 | bigjb | anyone seen "NOTICE[6960]: chan_iax2.c:3105 iax2_read: I should never be called!" before i go jump on google? |
| 14:25.10 | Nivex | ilbc was designed to withstand packet loss from the get go iirc |
| 14:25.12 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 14:25.23 | MikeJ[Laptop] | bigjb, I LOVE that one |
| 14:25.29 | coppice | but ilbc is a dumb design |
| 14:25.39 | MikeJ[Laptop] | it works OK.... |
| 14:25.45 | fourcheeze | coppice: you're probably right |
| 14:25.53 | bigjb | thats when using idefisk to call any extension |
| 14:26.04 | fourcheeze | MikeJ[Laptop]: what's the qulity of ilbc? |
| 14:26.10 | fourcheeze | quality even |
| 14:26.19 | MikeJ[Laptop] | worse than 711 |
| 14:26.24 | MikeJ[Laptop] | better than 729 |
| 14:26.33 | fourcheeze | ok |
| 14:26.38 | fourcheeze | so maybe what I want |
| 14:26.43 | MikeJ[Laptop] | and free |
| 14:26.46 | fourcheeze | and now you'll tell me that no devices support it |
| 14:26.56 | MikeJ[Laptop] | asterisk does.. |
| 14:27.00 | fourcheeze | hehe |
| 14:27.02 | MikeJ[Laptop] | I see other stuff too |
| 14:27.05 | MikeJ[Laptop] | phones |
| 14:29.07 | ambriento | 200.150.190.37s s |
| 14:29.07 | ambriento | 11:20 <MikeJ[Laptop]> then you need to pick your motherboard more carefully.. |
| 14:29.07 | ambriento | 11:20 <Leland> hmm.. I wouldn't say that, fourcheeze |
| 14:29.07 | ambriento | 11:20 ::: [SignOff: sysdebug (Read error: 110 (Connection timed out))] |
| 14:29.08 | ambriento | 11:20 <vader--> well dell puts that together |
| 14:29.08 | ambriento | 11:20 <MikeJ[Laptop]> vader--, just use a valcom or somthing like that |
| 14:29.10 | ambriento | 11:20 <bigjb> might be better to get a paging extension |
| 14:29.14 | ambriento | 11:20 ::: [SignOff: basta ("Sto andando via")] |
| 14:29.39 | MikeJ[Laptop] | ummmm |
| 14:29.42 | MikeJ[Laptop] | ok |
| 14:30.02 | *** join/#asterisk Eprom (n=eprom@adm-lap.ofc.lab1.n3network.ch) |
| 14:31.37 | MikeJ[Laptop] | latency |
| 14:31.39 | MikeJ[Laptop] | ? |
| 14:31.44 | fourcheeze | yeah, might be |
| 14:32.14 | MikeJ[Laptop] | your better off stuffing more audio per packet |
| 14:32.31 | MikeJ[Laptop] | going from 20ms to 80ms saves a TON of bandwidth |
| 14:32.51 | MikeJ[Laptop] | it's like 1/2 |
| 14:33.13 | coppice | put the whole conversation into one large packet |
| 14:33.42 | De_Mon | and call it a wav |
| 14:33.52 | coppice | brilliant |
| 14:33.56 | De_Mon | 20/80 != 1/2 |
| 14:33.58 | fourcheeze | MikeJ[Laptop]: but * won't do that |
| 14:34.01 | coppice | this might catch on |
| 14:34.11 | fourcheeze | also send the wav by email |
| 14:34.15 | Nivex | would be nice to do a call "walkie-talkie" style |
| 14:36.21 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 14:36.47 | coppice | are you a NexTel subscriber? :-) |
| 14:37.04 | Nivex | nope |
| 14:37.05 | cuco | hi, i would like to get a series of numbers from a user. i was hoping for getDigits or something. What options do I have? |
| 14:37.11 | cpm | lucky you |
| 14:37.51 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:37.55 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 14:38.17 | Druken | [TK]D-Fender: you around? |
| 14:38.21 | [TK]D-Fender | nope |
| 14:38.24 | Druken | k |
| 14:38.28 | Druken | :) |
| 14:38.33 | [TK]D-Fender | sup? |
| 14:38.53 | Druken | having issues with that damn spa-3000... it's not getting the public from behind the nat |
| 14:39.00 | Druken | keeps giving asterisk the nat'd ip |
| 14:39.40 | Druken | even AFTER i've set the publicip on it's web interface |
| 14:40.07 | [TK]D-Fender | So it looks like this? SPA-3000 -> NAT -> * right now? |
| 14:40.13 | Druken | yep |
| 14:40.20 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:40.23 | [TK]D-Fender | So * is PUBLIC? |
| 14:40.27 | Druken | always |
| 14:41.02 | mphill | is there anything for asterisk that would allows people to pay bills over the phone? |
| 14:41.06 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 14:41.13 | [TK]D-Fender | ok, well there are 2 or so NAT keep-alive settings on the SPA to do, and you should have "nat=yes", "qualify=yes" in the SIP.CONF entry |
| 14:41.16 | [TK]D-Fender | that should do it. |
| 14:41.19 | Druken | mphill: not until you make one |
| 14:41.31 | MikeJ[Laptop] | fourcheeze, asterisk CAN do that.. but it's global... change the define and recompile and it works.. |
| 14:41.54 | Druken | well, i don't have a block for it... my asterisk server autocreates the peer |
| 14:41.55 | MikeJ[Laptop] | file was working on some stuff to let you do that per channel, I think he has a branch for it.. |
| 14:42.44 | Hmmhesays | well i got my modified mwi to work |
| 14:42.47 | Hmmhesays | weeeee |
| 14:42.53 | *** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl) |
| 14:43.06 | MikeJ[Laptop] | De_Mon, you need to figgure the headers and everything in.... 4 packets of 20ms in total including headers is about twice the size of 1 80 ms packet |
| 14:43.07 | MikeJ[Laptop] | on rtp |
| 14:43.10 | [TK]D-Fender | Druken : You should have a fixed definition for something like an SPA-3000. its a GATEWAY.... |
| 14:43.38 | Druken | no it's not... i'm not using it for outgoing |
| 14:43.39 | [TK]D-Fender | Don't trust it to report the right IP. Actually, when in doubt nat=yes everywhere could work in a pinch I believe. |
| 14:44.05 | vader-- | do you guys know any good dealers for digium cards? |
| 14:44.13 | [TK]D-Fender | I have mine on me and might be able to test that. |
| 14:44.23 | [TK]D-Fender | vader-- : depends where you are. |
| 14:44.24 | MikeJ[Laptop] | vader--, doesn't really matter... |
| 14:44.31 | octothorpe | vader: voipsupply.com |
| 14:44.38 | MikeJ[Laptop] | voip-supply has a decent reputation.. but others do to.. |
| 14:44.49 | MikeJ[Laptop] | there are a couple guys out there with very tight margins |
| 14:44.59 | sambal | any good recommendations for europe? (holland) |
| 14:45.13 | Druken | son of a bitch |
| 14:45.16 | MikeJ[Laptop] | sambal, the one who will get it to you the ceapest? |
| 14:45.16 | Druken | thanks [TK]D-Fender |
| 14:45.20 | [TK]D-Fender | :) |
| 14:45.26 | *** join/#asterisk but3k4 (n=but3k4@unaffiliated/but3k4) |
| 14:45.28 | Druken | wtf.... |
| 14:45.48 | Druken | i set nat=yes on the server... then it reports properly... and i think my printer just came back online.... |
| 14:46.03 | MikeJ[Laptop] | hehe |
| 14:46.24 | vader-- | cool |
| 14:46.33 | Druken | i sweet it works |
| 14:46.42 | vader-- | hmm wonder if i should get a card with or without echo cancelation? |
| 14:47.13 | Druken | [TK]D-Fender nat=yes in global.. hehehe i guess i'll see if it screws anything up on me |
| 14:47.19 | octothorpe | vader: what kind of card are you looking for? (E1, etc. . .) |
| 14:47.20 | [TK]D-Fender | vader-- : Zaptel EC is a crapshoot IMO. Great for some, shit for others. If you get it in hardware and its done well, then thats the way to go. |
| 14:47.41 | vader-- | what does it actually do |
| 14:47.43 | [TK]D-Fender | Druken : basically never trust what they SAY, only what they DO :) |
| 14:47.55 | vader-- | im looking to get the TDM2460 to drive some analog phones in my building |
| 14:48.10 | Druken | yeah... but it's been my experince so far that nat=yes on a non natted user screws it up |
| 14:48.15 | vader-- | im getting TE110P for my PRI line |
| 14:48.16 | kippi | hey |
| 14:48.30 | kippi | has anyone used the avaya 4620's ? |
| 14:49.55 | *** part/#asterisk Ahrimanes (n=michael@aronsen.dk) |
| 14:49.58 | [TK]D-Fender | Druken : Not really. if there is a local subnet behind * with phones, those phones will report IP 1234 to * on an interface that is "local" to *. * will then ignore the return address in the packet and return it to the address it CAME IN from which happens to be the same. therefor, NO BAD :) |
| 14:50.32 | octothorpe | vader, you may want to look at a sangoma a200 with EC for the PRI |
| 14:50.48 | [TK]D-Fender | vader-- : Typically you want hardware EC for that regardless. Do it right the first time and you wont have to get punished for down-time, complaints, etc. |
| 14:51.02 | [TK]D-Fender | octothorpe : the A200 is an ANALOG CARD..... no good. |
| 14:51.34 | [TK]D-Fender | vader-- : What octothorpe might have wanted to suggest was Sangoma's A104d HWEC card. |
| 14:51.44 | octothorpe | my bad, I meant instead of the TDM2460, it's still early here |
| 14:51.55 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 14:52.06 | octothorpe | you are right on the a104d |
| 14:52.15 | [TK]D-Fender | Its a 4 port T1 card, but its the only density they sell with HWEC right now |
| 14:53.14 | [TK]D-Fender | TDM2460 is a waste unless PRI pricing is terrible in your region. |
| 14:54.33 | I-MOD | [TK]D-Fender: how does the TDM2460 relate to a pri? they're all fxs ports |
| 14:54.37 | sambal | anyone uses the TE411 ? |
| 14:54.57 | [TK]D-Fender | I-MOD : it doesn't My reference is that going with that many analog is a WASTE most of the time. |
| 14:55.04 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 14:55.06 | [TK]D-Fender | sambal : I've used one before |
| 14:55.16 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:55.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:55.29 | bartpbx | Hello, anyone using DEll SC1425 as asterisk server? http://www.digium.com/index.php?menu=compatibility only talks ist only talking about the SC 420.. |
| 14:55.54 | sambal | [TK]D-Fender : is the EC working ok? |
| 14:55.55 | hwt | bartpbx: that one should work fine. |
| 14:56.01 | bartpbx | thanks |
| 14:56.07 | sambal | [TK]D-Fender : do you hear the difference with a 410? |
| 14:56.24 | bartpbx | hwt, are you using the 1425 with digium hardware? |
| 14:56.43 | hwt | bartpbx: nope. but i don't see why it shouldn't work. |
| 14:59.22 | sambal | ] |
| 15:01.20 | viperdude | hi anyone around able to help me with the Flash Operator panel? |
| 15:02.09 | Katty | viperdude: i run it (= |
| 15:02.14 | Katty | viperdude: what seems to be your issue? |
| 15:02.36 | SplasPood | Anyone fammilar with asterisk generating the CALLERID(num) portion from the [context] in sip.conf? I've never seen this before, but it's happening to me now |
| 15:03.35 | viperdude | Katty: i can transfer calls fine using the drag & drop, however orignate calls is not working properly |
| 15:04.11 | viperdude | from some phones it goes straight to voicemail and for others it gives a sip response 400 Bad Request |
| 15:04.16 | SplasPood | by context I mean the heading from each sip user in sip.conf |
| 15:04.30 | Katty | viperdude: oh...we don't use the drag and drop options. |
| 15:04.37 | Katty | viperdude: we use is as a view only tool (= |
| 15:05.16 | Druken | drag and drop is fun with fop :) |
| 15:05.28 | Druken | but in a standard setup, it's backwards |
| 15:06.15 | viperdude | Druken: aha I will look into the reverse transfer |
| 15:06.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:06.40 | mocker | I'm having a problem w/ a user accessing voicemail. They can access it via the web cgi if they use the extension@context for the Mailbox, but how do they do that when entering their mailbox with the phone? |
| 15:09.56 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 15:09.59 | Eprom | Hey all, I've some difficulties to authenticate a phone. I want his request to be both authenticated by asterisk and by a proxy but it seem to doesn't work correctly. The RFC stipulate an UAS may challenge an UAC by replying a 401 Unauthorized but isn't the behavior of asterisk. |
| 15:11.30 | SplasPood | Can anyone think of why asterisk would be ignorning the CID sent by my clients and instead supplying it's own generated from the name of the sip user/peer ? |
| 15:11.32 | Eprom | I'm wondering if the actual reply (407) is correct, and if UAC can correctly resend his request with the proper credential to both the proxy and the asterisk |
| 15:13.25 | mocker | Is there a way I can tell what context my users are logged into? |
| 15:14.36 | mut | Mar 8 10:12:57 WARNING[11121]: channel.c:784 channel_find_locked: Avoided deadlock for '0xb680b868', 10 retries! |
| 15:14.47 | mut | i'm gettin this streaming in my cli like crazy |
| 15:14.53 | *** join/#asterisk SGM (n=stoyan@213.91.216.130) |
| 15:15.28 | mut | whats causing it |
| 15:15.33 | mut | seems to be when i use the manager api |
| 15:17.06 | zamba | what's the difference between POTS and PSTN? |
| 15:17.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 15:17.33 | zamba | and what's a POTS line compared to a PSTN line? |
| 15:18.11 | starlein | zamba: http://en.wikipedia.org/wiki/Plain_old_telephone_service |
| 15:19.04 | zamba | but if i want to switch my business to VoIP, i of course still need a connection to the POTS? |
| 15:19.13 | zamba | to be able to dial out to the rest of the world, yeah? |
| 15:19.20 | Katty | Druken: i went to chatzone. |
| 15:19.30 | Katty | Druken: and some op said you hadn't been there for awhile |
| 15:25.54 | viperdude | hmm not the reverse_transfer issue |
| 15:25.56 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 15:25.57 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 15:28.41 | zamba | yo! |
| 15:28.53 | zamba | but if i want to switch my business to VoIP, i of course still need a connection to the POTS if i want to dial out to the rest of the world? |
| 15:29.59 | Katty | yoer. |
| 15:30.03 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 15:32.58 | *** part/#asterisk Eprom (n=eprom@adm-lap.ofc.lab1.n3network.ch) |
| 15:34.33 | blitzrage | I have a certain linux box that I can't seem to get to produce colour on the Asterisk CLI -- any clues? |
| 15:36.40 | Druken | Katty: oh, you mean it's actually open again? |
| 15:37.54 | Katty | Druken: mew? |
| 15:38.09 | RoyK | Katty: happy march 8th |
| 15:38.12 | mut | oo |
| 15:38.29 | Katty | RoyK: mew? |
| 15:38.35 | mut | attend a 90 minute speech for 4 airlines tickets good for 20, cout it 20! destinations |
| 15:38.35 | RoyK | blitzrage: echo $TERM |
| 15:38.46 | Katty | Druken: err. |
| 15:38.52 | Katty | Druken: i don't care for unsolicited pettings, kthx. |
| 15:39.01 | RoyK | Katty: http://en.wikipedia.org/wiki/International_Women's_Day |
| 15:39.08 | Druken | oh, well pardon me... :) |
| 15:40.15 | blitzrage | RoyK: xterm |
| 15:40.31 | *** join/#asterisk nothinman (i=shakey@adas242.neoplus.adsl.tpnet.pl) |
| 15:40.36 | Enth | eterm ;) |
| 15:40.43 | nothinman | hey ho! |
| 15:40.48 | iDunno | xterm is the right way ;) |
| 15:40.57 | RoyK | blitzrage: hm. should work |
| 15:40.58 | blitzrage | no colour on the xterm for some reason |
| 15:41.08 | blitzrage | RoyK: yah I know -- seems to be fine on other * boxen... |
| 15:41.09 | RoyK | blitzrage: try export TERM=ansi |
| 15:41.15 | RoyK | just for kicks :P |
| 15:41.19 | blitzrage | aiight :) |
| 15:41.30 | RoyK | export TERM=dos |
| 15:41.31 | RoyK | :P |
| 15:41.34 | blitzrage | lol |
| 15:41.34 | nothinman | my asterisk died and I don't know why :( any ideas? log says: |
| 15:41.35 | nothinman | Mar 8 10:34:47 DEBUG[3013] chan_sip.c: Oooh, format changed to 256 |
| 15:41.35 | nothinman | Mar 8 10:34:47 WARNING[3013] channel.c: Unable to find a codec translation path from g729 to ulaw |
| 15:41.37 | blitzrage | I love DOS |
| 15:41.49 | RoyK | :) |
| 15:41.58 | blitzrage | nothinman: sounds like you don't have g729 codec and something borked |
| 15:42.10 | blitzrage | although it shouldn't bork at that |
| 15:42.16 | RoyK | nothinman: disallow=all, allow=alaw |
| 15:42.28 | coppice | export TERM=summer |
| 15:42.32 | RoyK | :P |
| 15:42.35 | RoyK | coppice: hi |
| 15:42.36 | nothinman | blitzrage: yip, I can read, but it was working and stopped :/ |
| 15:43.00 | blitzrage | like I said, its unlikely that is the actual reason... but if it is, make sure you dsisable g729 |
| 15:43.10 | nothinman | RoyK: in which config? sip.conf? allow=all without disallow is not possible? |
| 15:43.15 | RoyK | coppice: IT IS NOT SUMMER! IT IS BLOODY -10C AND NOT GOOD :( |
| 15:43.18 | blitzrage | nothinman: I know you can read, but I'm not sure what else you want us to say |
| 15:43.23 | RoyK | nothinman: sip.conf |
| 15:43.31 | nothinman | blitzrage ;-) |
| 15:43.35 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfiko.dialup.mindspring.com) |
| 15:43.44 | RoyK | NotFreak: Do Not Use allow all |
| 15:44.03 | RoyK | NotFreak: you don't, for instance, have g.723.1 |
| 15:44.05 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:44.08 | coppice | RoyK: Its about 16 here, and I am way north of home |
| 15:44.28 | RoyK | coppice: er. -16 or +16? |
| 15:44.46 | coppice | +16 and 2.5 hours north of home |
| 15:44.58 | RoyK | hrmf |
| 15:45.10 | RoyK | bloody norway |
| 15:45.33 | coppice | come to Shanghai |
| 15:45.48 | RoyK | coppice: anyway - i get these stupid 'symbol not found' errors when trying to load app_rxfax :( |
| 15:45.52 | coppice | you can have this hotel roo. i'm leaving tomorrow |
| 15:45.57 | Blackthorn | G'Morning. I probibly have an easy question for you guys/gals. I just ordered a 1-888 number from nufone and it is being sent to my * box. My * is set to use nufone for long distance. I setup the 888 number in my extenstions and pointed it my sipura unit. Is there anything else i need to do? ie set my * so it can recive from nufone? |
| 15:46.11 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:46.17 | coppice | "stupid symbol not found" is a strange error |
| 15:46.24 | RoyK | it is indeed |
| 15:46.51 | blitzrage | Blackthorn: if it works -- then no :) |
| 15:47.01 | coppice | remove spandsp. really really remove. all the versions. in every drak ress. then install |
| 15:47.16 | RoyK | coppice: where can i find them? |
| 15:48.21 | Blackthorn | blitz: nope it does not work. I call the # and nufone gives a message it is unable to reach the caller |
| 15:48.22 | coppice | people have been complaining about this recently, but it always turns out they have multiple copies of spandsp installed, and app_rxfax is picking up the wrong one |
| 15:48.54 | RoyK | ok |
| 15:49.10 | RoyK | removed /usr/local/include/spandsp* and /usr/local/lib/libspandsp* |
| 15:49.15 | blitzrage | Blackthorn: what does the SIP debug look like? Are you registered succesfully? Do you see an invite? |
| 15:49.20 | nothinman | RoyK: true, true, I messed up the config trying to configure my cisco 1760v.. damna :/ |
| 15:49.24 | nothinman | *damn. |
| 15:49.31 | RoyK | rebuilt spandsp and rebuilding asterisk with app_[rt]xfax |
| 15:49.39 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 15:50.02 | RoyK | coppice: what's under this test/ dir? |
| 15:50.16 | coppice | tests |
| 15:50.24 | RoyK | heh |
| 15:50.37 | RoyK | anyway - it worked like a dream when removing/reinstalling |
| 15:51.40 | Blackthorn | blitz: I am registered with the nufone service because I an already sent any call that is not local through them. |
| 15:51.56 | Blackthorn | an = can |
| 15:52.30 | *** join/#asterisk tld (n=tld@60.80-203-96.nextgentel.com) |
| 15:52.34 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 15:53.13 | Hmmhesays | wow this isn't working so hot |
| 15:53.27 | RoyK | what does it mean to work hot? |
| 15:53.39 | Hmmhesays | working well smart ass |
| 15:53.47 | RoyK | lol |
| 15:53.48 | Enth | lol |
| 15:53.53 | RoyK | hotly, perhaps, sir |
| 15:54.45 | Bambr | anybody here who had successfully built app_rxfax and app_txfax? |
| 15:54.58 | RoyK | I just did |
| 15:55.03 | RoyK | and loaded app_rxfax |
| 15:55.06 | Enth | RoyK: i already have a big tattoo on my back. |
| 15:55.08 | RoyK | 3 minutes ago |
| 15:55.09 | coppice | hit him with the 16 volume OED. its a killer |
| 15:55.15 | RoyK | Enth: how interesting |
| 15:55.23 | sambal | RoyK : it compiles faster over here ;) |
| 15:55.33 | Enth | A giant Phoenix. and on my right arm egyption glyphs. |
| 15:55.34 | Enth | :) |
| 15:55.41 | *** part/#asterisk cuco (n=diego@local.xorcom.com) |
| 15:55.57 | Enth | but anwyay, fear my 8 packs! |
| 15:56.22 | Hmmhesays | try that again |
| 15:56.24 | coppice | do the egyptian glyphs say something like "this idiot can't read egyptian"? :-) |
| 15:56.30 | Enth | lol |
| 15:56.56 | RoyK | coppice: :) |
| 15:57.03 | Enth | it actually says rise of the phoenix/sun god - Ra. |
| 15:57.14 | RoyK | Enth: have you _checked_ that?\ |
| 15:57.22 | Enth | although Ra's image is an Eagle, close/ |
| 15:57.36 | Enth | RoyK: I was the one who gave them the glyphs. |
| 15:57.40 | Hmmhesays | so I modified chan_sip to send out peer->mailbox@host, what a miserable failure that is |
| 15:58.00 | RoyK | my gf knows a good bit of japanese, and she tells me quite a few interesting things about what stuff means on other people's clothes |
| 15:58.14 | Enth | She from .jp ? |
| 15:58.23 | Enth | or just knows Japanese? |
| 15:58.28 | coppice | yeah. some of the chinese on people's clothes can be funny. |
| 15:58.31 | RoyK | like 'I suck dicks' on a t-shirt wore by a young man |
| 15:58.38 | jsharp | Hey |
| 15:58.40 | Enth | lol |
| 15:58.44 | RoyK | Enth: she's just studied it for three years or so |
| 15:58.48 | Enth | nice. |
| 15:59.00 | Enth | damn, wish i could speak several languages. |
| 15:59.51 | Enth | takk. |
| 15:59.56 | Enth | Tusen takk. |
| 15:59.59 | RoyK | :) |
| 16:00.04 | Enth | :) |
| 16:00.05 | hwt | RoyK: hi fellow norwegian. :) |
| 16:00.09 | RoyK | du kan vel ikke så veeeeeeldig mye |
| 16:00.12 | RoyK | hwt: hei |
| 16:00.17 | hwt | men jeg kan. :) |
| 16:00.26 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 16:00.27 | Enth | hei. |
| 16:00.30 | gaupe | hwt: skitprat |
| 16:00.34 | hwt | RoyK: btw, do you know where i can find norwegian asterisk-sounds? |
| 16:00.38 | Enth | lol |
| 16:00.41 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 16:00.48 | RoyK | hwt: #asterisk-no |
| 16:00.50 | hwt | gaupe: hei rmm. :) |
| 16:01.18 | coppice | sounds like a protest group :-) |
| 16:01.19 | gaupe | hwt: hallo :) |
| 16:02.36 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.141.168) |
| 16:02.41 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:02.42 | Fedoracore6 | hai all |
| 16:02.48 | _foxfire_ | hi , any1 got any experience with QSIG, i can't find any useful documentation on * about it ? |
| 16:03.06 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.230) |
| 16:03.30 | *** part/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 16:03.59 | RoyK | coppice: du forstår ikke stort |
| 16:04.33 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:05.13 | docelm0 | english! |
| 16:05.14 | docelm0 | :) |
| 16:05.42 | zoa | hey ho docelm0 |
| 16:05.46 | zoa | HEY HO OLLE! |
| 16:06.00 | oej | Hey ho ZOA |
| 16:06.05 | docelm0 | Sup Z! How are things? |
| 16:06.07 | docelm0 | OLLE! |
| 16:06.22 | docelm0 | Not that he would know me by my nick.. :) |
| 16:06.31 | docelm0 | Hay Zoa did I tell you I got rid of the goatee? |
| 16:06.34 | oej | No, I won't |
| 16:06.51 | docelm0 | Olle, you would know me to see me tho.. |
| 16:06.59 | docelm0 | You may know my name.. Brian Fertig |
| 16:06.59 | zoa | yeah you told me |
| 16:07.00 | file[laptop] | zoa: SLUT! |
| 16:07.06 | oej | Ahh, Brian!!! |
| 16:07.11 | zoa | aaaah file |
| 16:07.16 | coppice | RoyK: dammit. i was going for a witty respose in Chinese, and I can't get the input system to work on this stupid Windows machine :-\ |
| 16:07.24 | zoa | :) |
| 16:07.26 | docelm0 | ahh I guess I am fairly well known.. :) |
| 16:07.37 | oej | Coppice: THanks for a good article on FoIP on your web site |
| 16:07.43 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 16:07.45 | nothinman | who could give me a hint setting up cisco 1760v to connect to asterisk? i can't see the option to use username/passwd on cisco :/ |
| 16:07.53 | oej | Coppice: I am trying to implement T38 support per peer now in the branch |
| 16:08.17 | coppice | oej: what do you mean by per peer? |
| 16:08.19 | docelm0 | nothinman, its there you have to type something.. crap I forget the commands as I NEVER used them |
| 16:08.32 | Enth | nothinman: type "en" |
| 16:08.38 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 16:08.40 | Enth | and then enter the passwd |
| 16:08.41 | docelm0 | nothinman, just use it based on IP not user/pass |
| 16:08.42 | oej | Coppice: Right now, you can enable T38 for the whole channel, which means you disable RTP bridge for every device |
| 16:08.53 | oej | Coppice: I'm coding so you can set it per device instead |
| 16:08.56 | nothinman | Enth: heh, it's been done :-) |
| 16:09.23 | Enth | then conf t | int eth0 | username blah | passwd blah | host blahblah | save exit | save | reboot |
| 16:09.26 | Enth | heh |
| 16:09.27 | coppice | oej: Eh? why would is disable RTP? if the ends don'e negotiate T.38 they use RTP |
| 16:09.52 | oej | In the sip_bridge function it disables RTP native bridging if T38 support is on |
| 16:09.54 | nothinman | docelm0: that's a good answer. but i've got better question: how? in asterisk :-) |
| 16:09.58 | RoyK | coppice: :) |
| 16:10.05 | oej | Coppice: I might have misunderstood, but that's how I parse it |
| 16:10.11 | docelm0 | nothinman, How what security based on IP? |
| 16:10.32 | nothinman | i'll be nasty and i'll paste my config from cisco, which i believe is okay |
| 16:10.33 | nothinman | dial-peer voice 3 voip |
| 16:10.33 | nothinman | <PROTECTED> |
| 16:10.33 | nothinman | <PROTECTED> |
| 16:10.33 | nothinman | <PROTECTED> |
| 16:10.33 | nothinman | <PROTECTED> |
| 16:10.34 | nothinman | <PROTECTED> |
| 16:10.35 | oej | coppice: It does not disable RTP, but the native bridge |
| 16:10.36 | nothinman | <PROTECTED> |
| 16:10.38 | nothinman | <PROTECTED> |
| 16:10.40 | nothinman | <PROTECTED> |
| 16:10.42 | nothinman | ! |
| 16:10.43 | docelm0 | ACK! KICK HIM! |
| 16:11.15 | coppice | oej: i think that was only in there for testing |
| 16:11.19 | nothinman | docelmo: how can I allow cisco to connect to asterisk without password? just ip-based |
| 16:11.19 | docelm0 | nothinman, you will find in here PASTEBIN IS YOUR FRIEND! |
| 16:11.32 | oej | Coppice: Ok, then I need some feedback on how to fix that. |
| 16:11.38 | docelm0 | in your sip.conf do something like this: |
| 16:11.49 | oej | Coppice: I would also like to know a bit more about the RTP/TCP/UDPTL options |
| 16:11.52 | coppice | oej: during testing things kept bypassing the * box, and appearing to work when they were not even using the * box :-) |
| 16:11.53 | [ProB]CrazyMan | anbody here who is familar with .call files ? |
| 16:12.09 | Enth | actually to save time |
| 16:12.13 | oej | Coppice: Can a device support all of those at the same time? |
| 16:12.28 | Enth | nothinman: http://www.voip-info.org/wiki-Asterisk+cisco+FXO |
| 16:12.30 | oej | Or is it one only |
| 16:12.35 | Enth | that's a good start. |
| 16:12.37 | Enth | :) |
| 16:12.48 | nothinman | Enth: I started there :-) |
| 16:13.02 | coppice | oej: a device can support all of them. very few things support anything other than UDPTL, though |
| 16:13.12 | Enth | hrmmm |
| 16:13.16 | nothinman | Enth: problem is that this example doesn't seem to work for me |
| 16:13.16 | Enth | pastebin it |
| 16:13.42 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 16:13.44 | oej | Coppice: But for configs, those are separate options |
| 16:14.03 | oej | Coppice: Guess we only support UDPTL now, right? |
| 16:14.53 | coppice | *'s RTP doesn't support FEC, so its useless for T.38 right now. TPKT over TCP is there, but not quite complete |
| 16:15.32 | oej | Coppice: I did not find the TCP stuff in the patch. Are there code anywhere else than the bug tracker? |
| 16:15.57 | Enth | nothinman: so what exactly do you want to do? just purely connect the router to Asterisk ? |
| 16:16.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:17.27 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 16:17.36 | nothinman | i want to be able to dial from the router to asterisk |
| 16:17.42 | docelm0 | Enth, I took care of em |
| 16:17.44 | coppice | oej: I can't remember if I posted the tpkt transport or not. i think the chan_sip changes include some support for it. its months ago, so my memory is getting vague |
| 16:17.55 | nothinman | i've got analogue phones connected to cisco and voip to asterisk |
| 16:18.06 | nothinman | and all i want to do i let them communicate! |
| 16:18.07 | nothinman | ;) |
| 16:18.18 | docelm0 | What I gave you will let that happen |
| 16:18.24 | Enth | ok so docelm0 fixed it he says. |
| 16:18.27 | oej | coppice: I'm trying to work with it in a branch and add some configuration stuff and documentation, in order to prepare it for commit |
| 16:18.33 | Enth | :) |
| 16:18.44 | oej | coppice: I would really appreciate if you made sure I had your latest code |
| 16:18.55 | oej | Coppice: So I'm not wasting my time on old code |
| 16:19.14 | coppice | what you have should be up to date |
| 16:20.27 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 16:20.44 | coppice | oej: with a recent snapshot of spandsp T.38 termination seems to be working for some people |
| 16:21.05 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 16:21.19 | *** join/#asterisk keith80403 (n=keith804@24.56.188.49) |
| 16:22.00 | oej | coppice: Tried compiling spandsp on OS/X today, it failed on library creation |
| 16:22.42 | oej | Coppice: Ok, I'll work along in the branch with the latest patch in the bug tracker. If you have time, please check the branch to make sure it works properly for you. |
| 16:23.02 | oej | Coppice: And we need to get rid of that bridge stuff then. Any ideas on what I can change without breaking? |
| 16:23.05 | oej | :-) |
| 16:23.15 | nothinman | Enth: yea, we're trying :-) |
| 16:23.50 | Enth | have you managed to enter the username/passwd and establish the connection? |
| 16:23.53 | Enth | :) |
| 16:23.54 | Enth | brb |
| 16:24.19 | coppice | oej: someone told me they had spandsp working on OS/X. I don't know if they had to make any changes, though. This week's activity has been getting clean builds with VS2005 :-) |
| 16:24.37 | oej | :-) |
| 16:25.08 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 16:26.04 | niZon | hrm.. I hope I don't regret just buying that 7970 :P |
| 16:26.06 | MikeJ[Laptop] | :D |
| 16:26.25 | niZon | I'll finally have a use for chan_sccp |
| 16:26.45 | dpryo | Hm.. I've got a 7970 somewhere.. Do they work fine with asterisk? |
| 16:26.56 | dpryo | (I understand they don't support SIP?) |
| 16:27.02 | niZon | as long as you have chan_sccp installed :P |
| 16:27.06 | MikeJ[Laptop] | niZon, except it looks like cisco is going sip with ccm5 |
| 16:27.17 | niZon | oo :P |
| 16:27.24 | niZon | good |
| 16:27.25 | niZon | lol |
| 16:27.27 | dpryo | niZon: Yeah, I've used chan_sccp with 7960.. but asterisk segfaulted ;P |
| 16:27.29 | niZon | afk |
| 16:27.40 | niZon | yeah there's a few issues |
| 16:27.45 | dpryo | So I sipifyed all my 7960s |
| 16:28.19 | jsharp | I had some 7920s with chan_sccp. Asterisk worked fine, but the 7920s would reboot after every call. |
| 16:28.32 | dpryo | hehe |
| 16:28.39 | dpryo | 7920 is the wireless ones? |
| 16:28.49 | jsharp | Yeah. |
| 16:29.05 | dpryo | Hm.. I've lost my charger for it. |
| 16:29.10 | dpryo | Better find it and try it out :) |
| 16:29.20 | jsharp | And I discovered that you can't convert a 7940 to SIP from SCCP over a satellite link. TFTP doesn't like it. |
| 16:29.30 | dpryo | hehe |
| 16:30.09 | De_Mon | to use ztdummy as a timing source, does chan_zap.so need to be loaded? |
| 16:31.13 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 16:33.35 | nothinman | Enth: i haven't :-) |
| 16:36.42 | backblue | how do you guys pass the fisical limit of ~80 calls in one asterisk? how do you distribute the calls? |
| 16:37.17 | *** join/#asterisk JmGV (n=jgomez@83.175.220.178) |
| 16:40.36 | *** join/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 16:42.00 | Katty | mmm, cookie |
| 16:42.13 | Seyr | Is there anyway to call all extensions currently logged in? |
| 16:42.18 | Seyr | or at least get a list of them? |
| 16:42.31 | umay | show channels |
| 16:42.43 | Seyr | for use in AGI script |
| 16:43.18 | Katty | i did all of mine by hand. |
| 16:43.34 | Katty | where you set an extension to ring ex 1 & 2 & 3, etc |
| 16:43.37 | RoyK | Seyr: sip show peers from agi and parse it :P |
| 16:44.24 | Katty | course mine isn't an agi script |
| 16:44.37 | Katty | and it's used for ring/blasty groups |
| 16:44.43 | backblue | no |
| 16:44.56 | backblue | there is one option |
| 16:45.09 | backblue | to have a context with only the register peers |
| 16:45.23 | backblue | and you just simple dial all |
| 16:45.55 | *** join/#asterisk Scarad (n=jporten@c-67-173-185-85.hsd1.il.comcast.net) |
| 16:46.06 | x86 | dpryo: 7960 supports a SIP firmware image ;) |
| 16:46.07 | backblue | not parses |
| 16:46.24 | x86 | dpryo: so you can natively use SIP with it and asterisk |
| 16:46.26 | backblue | 7940 & 7960 works great with asterisk & sip |
| 16:46.42 | backblue | 7970 it's not SIP enable. |
| 16:46.59 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 16:47.12 | dpryo | x86: I know. |
| 16:47.21 | dpryo | x86: 1727 > So I sipifyed all my 7960s |
| 16:47.23 | dpryo | :) |
| 16:47.30 | Nugget | I understand that chan_sccp has gotten a lot less awful lately, but I still haven't been brave enough to try it yet. |
| 16:47.58 | dpryo | I'll check it out with 7970 and 7920 tomorrow. |
| 16:48.06 | dpryo | Perhaps |
| 16:48.30 | dpryo | 7970 is a nice phone.. would be cool to actually use it. |
| 16:48.31 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 16:51.04 | Scarad | what does chan_sccp not deliver |
| 16:51.20 | x86 | pizza! |
| 16:51.31 | *** part/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 16:51.50 | Scarad | hmmm pizza |
| 16:52.19 | jsharp | It wouldn't do the more-than-2-parties conferencing I wanted. |
| 16:52.29 | x86 | if chan_sccp can deliver some pizza to me, i'll be impressed... |
| 16:52.31 | x86 | PoIP :P |
| 16:52.54 | jsharp | * supports PoIP, but only in pass-thru mode. You can see the pizza go by, but you can't touch it. |
| 16:53.07 | x86 | hahaha |
| 16:53.23 | x86 | you can smell but you cant taste ;) |
| 16:53.39 | x86 | to taste you need to upgrade to the pro version hahaha |
| 16:54.35 | Nivex | http://pastebin.ca/44897 |
| 16:55.00 | _foxfire_ | scarad : how about efective autentication ... |
| 16:55.52 | Scarad | true true |
| 16:57.05 | Scarad | The Polycom phones are better anyway...though they take a long time to boot |
| 16:58.30 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 16:59.41 | ManxPower | We seldom need to boot our Polycoms. |
| 17:00.13 | mut | our sphere needs booted once or twice a month |
| 17:00.39 | nothinman | bloody router, argh! |
| 17:00.52 | [TK]D-Fender | My polycoms last nearly forever before needing reboots. Only when * loses hint subscriptions and go "sticky" :) |
| 17:01.19 | twisted[asteria] | i've noticed that * doesn't lose hint subscriptions anymore |
| 17:01.30 | twisted[asteria] | perhaps it's time to upgrade |
| 17:01.38 | *** join/#asterisk octothorpe__ (n=octothor@198.60.73.230) |
| 17:02.12 | Qwell | twisted[asteria]: y0 |
| 17:02.18 | file[laptop] | eeeeeeep |
| 17:02.24 | twisted[asteria] | hi Qwell |
| 17:05.15 | *** join/#asterisk Isaiah (n=test@208-187-93-4.br1.hnv.mi.frontiernet.net) |
| 17:05.34 | Isaiah | Does Firefly 2 work with asterisk? |
| 17:05.34 | Katty | twisted[asteria]: the last two times i called you, you didn't answer! |
| 17:05.43 | Katty | twisted[asteria]: consider me peeved! |
| 17:05.58 | Qwell | Katty: at least he didn't sent you into telemarketer hell :p |
| 17:06.11 | Katty | he wouldn't do that :< |
| 17:06.38 | Katty | iDunno: yes...soon |
| 17:06.45 | Katty | iDunno: just a few more hours (= |
| 17:06.52 | twisted[asteria] | Katty, you called me? |
| 17:06.56 | Katty | twisted[asteria]: beeped. |
| 17:06.56 | Qwell | pfft, my day hasn't even started yet |
| 17:07.00 | Katty | twisted[asteria]: the /other/ phone |
| 17:07.01 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 17:07.18 | Qwell | brookshire: hey, are you at Fort Digium? |
| 17:07.23 | Qwell | obviously |
| 17:07.26 | Katty | twisted[asteria]: look at your phone again |
| 17:07.42 | twisted[asteria] | i'm lookin at it |
| 17:07.54 | twisted[asteria] | NEXTEL\n11:07am 3/8 |
| 17:08.06 | brookshire | ?? |
| 17:08.06 | Katty | just like that |
| 17:08.09 | Katty | except 2 nights ago |
| 17:08.14 | Katty | maybe 3 |
| 17:08.15 | Qwell | brookshire: Could you do me a huge favor? |
| 17:08.16 | twisted[asteria] | just like what? |
| 17:08.28 | brookshire | maybe! |
| 17:08.30 | Katty | twisted[asteria]: i beeped you a few nights ago and you never answered :P |
| 17:08.37 | twisted[asteria] | i didn't get a beep |
| 17:08.39 | Qwell | brookshire: Could you go poke mog, and have him talk to the person for me? |
| 17:08.49 | Qwell | gotta run to work |
| 17:08.50 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-3.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:09.27 | Qwell | btw, if any of you are going to VON, you should join #VON |
| 17:09.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 17:09.46 | *** join/#asterisk prongils (n=el_terri@67.58.10.46) |
| 17:09.50 | wasim | ehm |
| 17:09.52 | wasim | MGCPGW/C1/02/03-77f2 C1-02-03@mgcpgw:1 Up SS7Bridge(Zap/34) |
| 17:09.55 | wasim | 240 active channels |
| 17:09.57 | wasim | 120 active calls |
| 17:10.36 | Qwell | bbl |
| 17:10.38 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 17:10.46 | wasim | g729 calls, i.e. |
| 17:11.15 | Dr-Linux | awww |
| 17:11.28 | zoa | what are you using for the ss7bridge ? |
| 17:11.34 | wasim | zoa: its a misnomer |
| 17:11.40 | zoa | misnomer ? |
| 17:11.43 | wasim | zoa: its actually a replacement for app_dial |
| 17:11.43 | zoa | what is a misnomer ? |
| 17:11.49 | zoa | ah |
| 17:11.50 | wasim | wrong name for something |
| 17:12.03 | Dr-Linux | hi wasim |
| 17:12.20 | Dr-Linux | wasim: whassup? :) |
| 17:12.30 | wasim | but the coding part isn't the hard bit, its the 10k call attempts per hour that drove * nuts |
| 17:12.42 | ManxPower | I was wondering how people were doing SS7 with Asterisk! Turns out they just run a non-SS7 app and call it SS7! Brilliant! BVry Microsoftish, |
| 17:12.42 | twisted[asteria] | what's wrong with 10k calls per hour? |
| 17:12.55 | wasim | ManxPower: eh? we run ss7 all over |
| 17:13.02 | Katty | twisted[asteria]: but i just beeped you. |
| 17:13.05 | Katty | twisted[asteria]: and it said it went through |
| 17:13.29 | Dr-Linux | wasim: kaisay hain app? :) |
| 17:13.29 | twisted[asteria] | Katty, hmm... strange. i show now missed calls/etc. |
| 17:13.30 | Katty | come to think of it, we did have some tower problems... |
| 17:13.31 | zoa | wasim, but what ss7 thing are you using ? |
| 17:13.40 | twisted[asteria] | s/now/no |
| 17:13.46 | Katty | twisted[asteria]: k, i'll let it slide this time. |
| 17:13.50 | wasim | zoa: s/ss7bridge/zapbridge |
| 17:13.50 | Katty | twisted[asteria]: but just this once! *grin* |
| 17:13.51 | oej | twisted!!! |
| 17:13.55 | zoa | which is ? |
| 17:13.58 | wasim | zoa: we didn't change the name from the skeleton file |
| 17:14.00 | twisted[asteria] | oej! |
| 17:14.03 | zoa | <wasim> ManxPower: eh? we run ss7 all over |
| 17:14.06 | zoa | i dont get it |
| 17:14.12 | Katty | twisted[asteria]: not the horrid tickles :< |
| 17:14.18 | Katty | twisted[asteria]: anything but the tickles >.< |
| 17:14.20 | ManxPower | I was (mostly) joking. |
| 17:14.21 | wasim | zoa: not on this box, but otherwise with ss7box |
| 17:14.24 | wasim | ManxPower: :P |
| 17:14.25 | twisted[asteria] | Katty, no, not the horrid ones |
| 17:14.28 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 17:14.29 | CrashHD | if I'm running voip connections over bonded t1's is there anything I should take into account? |
| 17:14.32 | Katty | twisted[asteria]: k :> |
| 17:14.34 | zoa | and what is ss7box ? |
| 17:14.35 | CrashHD | using ip cef distributed |
| 17:14.41 | ManxPower | So where can I find information on downloading and/or purchasing SS7 support for Asterisk? |
| 17:14.52 | wasim | ManxPower: zoa : sss7box.com |
| 17:14.57 | wasim | www.ss7box.com |
| 17:15.08 | wasim | ManxPower: you can also go the cosini route |
| 17:15.14 | ManxPower | CrashHD, latency and jitter are evil. |
| 17:15.14 | zoa | ah found it |
| 17:15.37 | CrashHD | lol |
| 17:15.51 | CrashHD | well I'm seeing 50-60 packets lost per conversation |
| 17:15.52 | Isaiah | What's a good free softphone to use with asterisk(sip or IAX will work)? |
| 17:16.01 | CrashHD | thought there may be a command or two I was missing |
| 17:16.09 | CrashHD | on the cisco routers |
| 17:16.14 | ManxPower | So SS7 box got a commercial license from Digium, I assume. |
| 17:16.14 | Dr-Linux | CrashHD: xlite, SJphone |
| 17:16.25 | Dr-Linux | sorry |
| 17:16.33 | Dr-Linux | Isaiah: xlite, SJphone |
| 17:16.34 | ManxPower | CrashHD, I don't know, that would be a question for #cisco |
| 17:16.42 | CrashHD | :) |
| 17:16.48 | Isaiah | Ok thanks Dr-Linux :) |
| 17:16.52 | wasim | ManxPower: no, they use woomera |
| 17:17.07 | ManxPower | wasim, Ah. ick. |
| 17:17.13 | Katty | woomura |
| 17:17.21 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 17:17.37 | Katty | craig's nice. |
| 17:17.37 | twisted[asteria] | wombera |
| 17:17.44 | ManxPower | woomera sounds like the place a baby kangaroo stays, |
| 17:17.47 | twisted[asteria] | since the b is silent. |
| 17:17.58 | Katty | wombura |
| 17:18.02 | prongils | hi all, whats a good way of having a Background(file) play and ring internal extensions at the same time? |
| 17:18.21 | twisted[asteria] | prongils, music on hold |
| 17:18.26 | ManxPower | prongils, in 1.2 you can set a MoH class for that. |
| 17:18.36 | twisted[asteria] | dial option m() |
| 17:18.42 | wasim | Dr-Linux: hi |
| 17:18.49 | Katty | twisted[asteria]: :< |
| 17:18.57 | Katty | twisted[asteria]: you can run, but you can't hide. |
| 17:19.03 | Katty | twisted[asteria]: i practically know where you live - i'll find you! |
| 17:19.11 | Dr-Linux | wasim: from Lhr? |
| 17:19.12 | prongils | thanks lemme see what i can do |
| 17:19.18 | wasim | Dr-Linux: in lhr |
| 17:19.29 | Dr-Linux | wasim: cool, same here :) |
| 17:21.45 | [TK]D-Fender | twisted[asteria] : Silent.... yeah like the "p" in swimming.... |
| 17:22.19 | *** join/#asterisk Lino` (n=Lino@i577BDED4.versanet.de) |
| 17:22.25 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 17:22.31 | *** join/#asterisk NirS (n=nirs@62.90.49.118) |
| 17:22.38 | NirS | hey all |
| 17:22.49 | NirS | how is everybody doing today ? |
| 17:22.58 | Katty | tired. |
| 17:23.06 | twisted[asteria] | " |
| 17:23.16 | NirS | anyone got experience with connecting Asterisk to a France Telecom E1 circuit ? |
| 17:23.26 | twisted[asteria] | i surrender! |
| 17:23.51 | wasim | NirS: you have to set h,1,AuRevour() |
| 17:24.01 | Katty | how about we forget the telcom e1 circuit and opt for short circuit instead? |
| 17:24.18 | NirS | ah ? |
| 17:24.37 | Katty | i'll bring the popcorn! |
| 17:24.38 | NirS | wasim, I understand from your remark that Asterisk E1 will not work with France Telecom ? |
| 17:24.52 | [TK]D-Fender | "But Egon... I thought you said crossing the streams was baaaaaadd..." |
| 17:25.01 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 17:25.03 | wasim | NirS: it should |
| 17:25.25 | wasim | NirS: 1,0,0,ccs,hdb3,crc4 (if you have those and check your timing) |
| 17:25.57 | NirS | the guy from France Telecom said something about a T2 line ? |
| 17:26.04 | NirS | anyone has any idea what that is ? |
| 17:26.54 | wasim | NirS: T2 is a DS2 with 96 bearer chans |
| 17:27.11 | Abydos313 | so like 4 t1's? |
| 17:27.48 | NirS | wasim, so I would need a splitter from T2 to 4xT1 ? |
| 17:27.53 | wasim | but wtf they have a T line in france is not right |
| 17:27.56 | NirS | in order to use a TE410P card ? |
| 17:28.01 | wasim | you'd think they have E1s |
| 17:28.12 | wasim | good 'ol 32 channels |
| 17:28.23 | Abydos313 | i thought e1 is 29 channels |
| 17:29.26 | asterisk99 | >I have a zaptel.conf question... I know how to config zaptel.conf for TDM400P card with 1,2,3,4 FXO/FXS cards... faur enuf ... but what do you do if you want to install a 2nd card??? Is this not possible? If it is, how do you define the fxsks= and fxoks= for the 2nd card??? |
| 17:29.39 | Lino` | hmmm |
| 17:30.15 | Abydos313 | google says up to 32 :)) |
| 17:30.17 | [TK]D-Fender | asterisk99 : You would just use channels 5-8 |
| 17:31.08 | asterisk99 | [TK]D-Fender: Aha! And I assume if you had 3 cards, channels 9,10,11,12 |
| 17:31.12 | [TK]D-Fender | asterisk99 : However more than 2 (some would say 1) Digium card in a system is not a good thing though so keep your options open towrads possibly a partial PRI |
| 17:31.22 | [TK]D-Fender | 3=bad idea |
| 17:31.35 | wasim | asterisk99: no, you have to skip 9 for the inter-galactic-channel |
| 17:31.38 | [TK]D-Fender | interrupts heavy, power heavy, etc... |
| 17:31.45 | nothinman | should my cisco 1760v accept sip calls from asterisk? |
| 17:32.07 | asterisk99 | [TK]D-Fender: I'm building a Perl program to auto-config zaptel.conf and zapata.conf --- I hate doing it manually & keep forgetting the settings |
| 17:32.34 | asterisk99 | wasim: The INTERGALACTIC channel???? Surely you jest!!!!! |
| 17:33.08 | [TK]D-Fender | asterisk99 : Harldy worth it. Just keep 1 templat hanging around and thats it... |
| 17:34.01 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:34.07 | asterisk99 | [TK]D-Fender: Maybe. I've always tried to automate everything.... One day I'll automate myself... that's the day I can retire (or die... whichever) |
| 17:34.42 | _Paulo_ | [TK]D-Fender, would you say I should not put 2 TE110P in the same PCI bus? |
| 17:35.07 | asterisk99 | [TK]D-Fender: I'm making a 1-step Asterisk/Apache installation script... almost done execpt for the zapata.conf stuff |
| 17:35.07 | [TK]D-Fender | _Paulo_ : Don't know about the card specifically, but definately not MORE. |
| 17:38.23 | *** join/#asterisk guilherme-jorge (n=admin@200.155.185.1) |
| 17:38.47 | Hmmhesays | I'm just a notch in the bedpost but you're just a line in a song |
| 17:42.32 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 17:43.42 | *** join/#asterisk shuri (n=shuri@64.235.209.226) |
| 17:44.21 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:44.28 | Nugget | Debra was a Catholic girl, she held out to the bitter end. |
| 17:44.34 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:48.52 | mut | still getting this channel lock thing |
| 17:48.52 | mut | Mar 8 12:46:55 WARNING[24530]: channel.c:784 channel_find_locked: Avoided deadlock for '0xb680b868', 10 retries! |
| 17:48.57 | mut | anyone know what thats from |
| 17:48.57 | Hmmhesays | Nugget lol |
| 17:50.06 | wundaboy | what is the best sounding codec of these: G.711 ulaw G.711 alaw G.729 or iLBC ? |
| 17:50.25 | Hmmhesays | G.711 |
| 17:50.31 | Hmmhesays | it gives me wood |
| 17:50.35 | wundaboy | alaw or ulaw? |
| 17:50.48 | Hmmhesays | aren't they the same sampling rate? |
| 17:50.55 | [TK]D-Fender | ALAW & ULAW are pretty much identical. |
| 17:51.05 | [TK]D-Fender | wundaboy : Where are you located? |
| 17:51.11 | wundaboy | Portland, OR |
| 17:51.19 | wundaboy | sup D-Fender |
| 17:51.37 | [TK]D-Fender | wundaboy : Oh yeah... my memory aspires to siv-dom ;) |
| 17:52.01 | wundaboy | im at work, so its 8 hours of playing with * |
| 17:52.09 | [TK]D-Fender | wundaboy : Use ULAW. Its the N/A standard and will require SLIGHLTY less work to transcode to PSTN here. |
| 17:52.12 | wundaboy | and taking a call every now and then |
| 17:52.23 | [TK]D-Fender | wundaboy : Much the same here :) |
| 17:52.44 | wundaboy | are you in cananda? |
| 17:53.10 | [TK]D-Fender | yup |
| 17:53.26 | wundaboy | what do their phone numbers look liek? |
| 17:54.15 | wundaboy | same pattern as the us? |
| 17:55.15 | [TK]D-Fender | yup |
| 17:56.05 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
| 17:56.37 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-3.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:58.47 | wundaboy | where do i set codec>? |
| 17:59.28 | asterisk99 | mut: (offtopic) I believe that it's a song called '88 LINES ABOUT 44 WOMEN' by a group called 'Nails' (1984) |
| 18:01.14 | [TK]D-Fender | wundaboy : depends what tech you're working with... |
| 18:03.04 | mut | say what |
| 18:03.50 | Nugget | What ain't no country I ever heard of. They speak English in What? |
| 18:04.33 | Hmmhesays | we all speak english |
| 18:04.36 | mut | dunno |
| 18:04.41 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 18:04.45 | Hmmhesays | YOU DON"T BELONG ON THIS PLANET IF YOU DON"T |
| 18:04.52 | Hmmhesays | lol |
| 18:05.15 | justinu | <PROTECTED> |
| 18:06.19 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-3.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 18:06.33 | De_Mon | Juggie you there? |
| 18:08.19 | *** join/#asterisk azzie (n=az@azzie.net) |
| 18:08.44 | Hmmhesays | that reminds me i need to get some more boomers |
| 18:09.03 | asterisk99 | Hmmhesays: You gotta be kidding!!! There are a billion people in India (where all our jobs will be one day if George 'Dubya' gets his wish) and 1.3 billion in China (where all our cars will be made one day soon) , most of whom do not speak English!!!! |
| 18:09.26 | Hmmhesays | asterisk99: yes I was kidding |
| 18:09.31 | Nugget | I guess asterisk99 was born without a sense of humor. |
| 18:09.37 | Hmmhesays | hence the---->[12:05] * Hmmhesays ducks and runs |
| 18:09.37 | Hmmhesays | [12:05] <Hmmhesays> lol |
| 18:09.45 | asterisk99 | Hmmhesays: At-sa-ma-boy!!! |
| 18:10.22 | [TK]D-Fender | asterisk99 : No... "W" will bomb them into the dark ages before then in the name of "Freedom" :) |
| 18:10.24 | backblue | VoiceOne uses realtime static or true realtime? |
| 18:11.02 | _Paulo_ | India was an English colony, don't they all speak english there? |
| 18:12.12 | Hmmhesays | I like a good bomb |
| 18:12.21 | jsharp | Sure. Every time I call there, I get someone who speaks english. |
| 18:12.25 | jsharp | And is named Bob or Tom. |
| 18:12.42 | Hmmhesays | hell if I had napalm i'd be firebombing shit every day |
| 18:13.03 | asterisk99 | [TK]D-Fender: Oh god (or Allah as the case may be) don't get going... That idiot in the White House is in for a severe ass-kicking when he decides to invade Iran to emiminate them thar Weapons of Mass Deception... IT's gonna make Falluja look like a Sunday-School Picnic (but, I've gone off topic... sorry) |
| 18:13.18 | De_Mon | I need some help getting app_conference compiled against 1.2.4, any links? |
| 18:13.48 | [TK]D-Fender | asterisk99 : Yeah.... I feel the "people" are deceiving themselves plenty enough as it is..... |
| 18:14.00 | Hmmhesays | i learned some meat puppets last night |
| 18:14.08 | Hmmhesays | that was fun, until the 40th time I played through it |
| 18:15.05 | [av]bani | ... |
| 18:16.24 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 18:17.37 | Dr-Linux | Hmmhesays: last night i saw your gf :) |
| 18:17.40 | Dr-Linux | in dreamz |
| 18:18.10 | PakiPenguin | o |
| 18:18.12 | PakiPenguin | :o |
| 18:18.55 | Dr-Linux | PakiPenguin: main ne app ki gf ki baat nahin ki :P |
| 18:19.35 | PakiPenguin | Dr-Linux, :o |
| 18:19.43 | PakiPenguin | Dr-Linux, where from? |
| 18:20.01 | Dr-Linux | PakiPenguin .. |
| 18:20.14 | PakiPenguin | jee jee? |
| 18:20.22 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:20.35 | Dr-Linux | /w Dr-Linux ... then ; www.checkdomain.com :) |
| 18:20.38 | jpablo | Hi, what's the best way to connect 14 fxos to a * box ? |
| 18:20.58 | jsharp | TDM2400 |
| 18:21.05 | PakiPenguin | Dr-Linux, neat |
| 18:21.19 | jpablo | jsharp, over a channel banck ? |
| 18:21.25 | _Paulo_ | jpablo, or a channelbank |
| 18:21.28 | Dr-Linux | jpablo: TDM2400 has 23 ports |
| 18:21.41 | Dr-Linux | s/23/24 |
| 18:21.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:22.22 | jsharp | FXO channel banks are notoriously expensive. |
| 18:22.35 | Hmmhesays | Dr-Linux: she ditched me last night, so good for you man |
| 18:22.46 | Hmmhesays | at least someone is seeing those fantastic boobies |
| 18:23.01 | jpablo | Dr-Linux, over a tdm2400 ? |
| 18:23.03 | Dr-Linux | jpablo: look for some tdm400p with fxo (4 port each) |
| 18:23.12 | jpablo | eek |
| 18:23.28 | jpablo | is the tdm stable? |
| 18:23.53 | *** join/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 18:24.01 | jpablo | i have used tsome tdm400p and it kind of suck |
| 18:25.38 | zoa | i use a tdm400p and am very happy with it |
| 18:27.01 | Dr-Linux | Hmmhesays: yeah, i saw her boobies :P |
| 18:27.09 | Hmmhesays | fantastic man |
| 18:27.14 | Hmmhesays | they are spectacular |
| 18:27.35 | Dr-Linux | Hmmhesays: actually i was looking milk before get sleep |
| 18:27.37 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
| 18:27.43 | Hmmhesays | thats a little disturbing |
| 18:27.47 | Zodiacal | anyone know why * doesn't return control back to my context that called voicemail()? |
| 18:27.47 | Dr-Linux | thats i why i have seen such great ... ;) |
| 18:27.52 | PakiPenguin | :o |
| 18:27.57 | PakiPenguin | little == very |
| 18:28.03 | Zodiacal | i just want to playback a simple file after the users voicemail has been recorded.. |
| 18:28.06 | Leland | ARGGGGGH!!!! |
| 18:28.09 | Zodiacal | it just hangs up insted |
| 18:28.35 | Leland | fecking MoH transcoding *sigh* |
| 18:29.15 | Dr-Linux | anybody is play with AGI using C ? |
| 18:29.27 | Leland | I'm almost convinced it's the digium 729 codec which is causing the problem |
| 18:29.28 | Dr-Linux | anybody is playing with AGI using C ? |
| 18:29.43 | Dr-Linux | Leland: what problem? |
| 18:30.20 | Leland | MoH is played fine to endpoints on all other codecs EXCEPT g.729 ... those users hear corrupted crap |
| 18:30.21 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 18:30.50 | Leland | not even resembling music or anything else.. rather like doing a cat file.mp3 > /dev/sound |
| 18:30.54 | Zodiacal | any ideas? |
| 18:31.16 | oej | ~seen coppice |
| 18:31.20 | jbot | coppice <n=chatzill@210.22.134.149> was last seen on IRC in channel #asterisk, 2h 7m 1s ago, saying: 'oej: someone told me they had spandsp working on OS/X. I don't know if they had to make any changes, though. This week's activity has been getting clean builds with VS2005 :-)'. |
| 18:32.37 | Leland | even posted a message on the forums about it, but still no resolution to the problem |
| 18:32.47 | *** join/#asterisk Scarad (n=jporten@c-67-173-185-85.hsd1.il.comcast.net) |
| 18:33.15 | *** part/#asterisk Scarad (n=jporten@c-67-173-185-85.hsd1.il.comcast.net) |
| 18:33.48 | ManxPower | Leland, Forums? |
| 18:34.09 | Leland | yea.. forums.digium.com |
| 18:34.28 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 18:35.25 | Leland | hmm.. found a cisco bug as well. |
| 18:35.29 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
| 18:35.46 | Leland | show policy-map interface input returns the OUTPUT policy-map ... *eyeroll* |
| 18:35.56 | Sebb | hi.. one question: when i use realtime voicemail-config (via ldap), i can't use the directory-function.. is that right? |
| 18:36.29 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 18:36.40 | MattH | hi.. during the install process.... when do the /dev/zap files get created? |
| 18:37.18 | jpablo | humm, make install |
| 18:37.25 | backblue | MattH: they are created when you load zaptel related modules, i think |
| 18:37.25 | jpablo | make install of zaptel |
| 18:37.35 | twisted[asteria] | MattH, depends, do you use udev? |
| 18:37.38 | jpablo | backblue, depends if you are using udev or not |
| 18:37.51 | MattH | twisted: yes this person is using udev (not me) |
| 18:38.08 | backblue | ok, who the hell dont use udev? :o jk |
| 18:38.08 | twisted[asteria] | MattH, then it's like blackblue mentioned, when you load the modules |
| 18:38.14 | MattH | he said it compiled fine and he did make install but when he tries to load zaptel it gives an error about the device /dev/zap/ctl not existing |
| 18:38.27 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 18:38.28 | backblue | MattH: have you runned ztcfg? |
| 18:39.16 | MattH | ztcfg says "line 0: unable to open master device "/dev/zap/ctl" |
| 18:39.26 | twisted[asteria] | heh |
| 18:39.30 | twisted[asteria] | look and see if there is a /dev/zapctl |
| 18:39.35 | MattH | no there isn't |
| 18:39.37 | twisted[asteria] | if so, you need to modify the udev rules |
| 18:39.42 | MattH | that's why I was wondering when it was suppose to be created |
| 18:39.44 | MattH | ok... |
| 18:39.45 | twisted[asteria] | not with teh slash, just /dev/zapctl |
| 18:39.51 | MattH | oh |
| 18:40.02 | twisted[asteria] | see if it's all smashed together |
| 18:40.07 | MattH | checking |
| 18:40.14 | backblue | yes, check your udev rulles. |
| 18:40.42 | backblue | twisted[asteria]: how its the best way arround, to get over the 80 calls fisic limitation? (clustering??) |
| 18:41.00 | ManxPower | What 80 calls limitation? |
| 18:41.04 | twisted[asteria] | backblue, what limitation? |
| 18:41.26 | MattH | twisted[asteria], no /dev/zapctl |
| 18:42.09 | ManxPower | I still have no idea why people have problems with Zaptel. It' Just Works for me on Mandrake w/ 2.4 and 2.6 kernels |
| 18:42.18 | backblue | twisted[asteria]: teorically, one processor p4, handles about 80 calls transcoding. i need to have about 800 calls transcoding, i really need clustering asterisk, and i'm checking arround how to do it. |
| 18:42.47 | file | transcoding g729? |
| 18:42.49 | ManxPower | backblue, That is specific to YOUR setup. There is no 80 call limitation. |
| 18:43.02 | backblue | file: yes, for example. |
| 18:43.13 | twisted[asteria] | backblue, only 80 calls? that's kinda weak |
| 18:43.22 | backblue | ManxPower: but there is limitation, your hardware its not infinit. |
| 18:43.32 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 18:43.34 | kink0 | hello |
| 18:43.39 | backblue | twisted[asteria]: 80 calls transcoding at the same time? :D not much? |
| 18:43.45 | ManxPower | backblue, There is no limitation coded into Asterisk. |
| 18:43.46 | cpm | If I had the trouble of too many calls to transcode, that would be something. |
| 18:44.08 | backblue | as voip-info says, dual xeon 3Ghz or something bleeding edge, it's about 130 calls at the same time. |
| 18:44.09 | ManxPower | Any limitation would be caused by your specific hardware and your specific configuration and requirements. |
| 18:44.17 | kink0 | anyone knows if a BRITEmux from Conklin is enable to demux E1 q931 ? |
| 18:44.31 | backblue | ManxPower: the limitation it's your hardware, not asterisk. |
| 18:45.01 | ManxPower | For example, G729 takes massive amounts of CPU and so you can't do many calls doing transcoding with G729. On the other hand if you didn't have to transcode (or didn't have to transcode so much) then you would be able to do MANY MANY more calls. |
| 18:45.10 | twisted[asteria] | best method would be not to transcode |
| 18:45.26 | ManxPower | G729 is so CPU hungry many ATAs can only do 1 G729 call at a time. |
| 18:45.33 | twisted[asteria] | format_g729 and a good audio conversion tool, convert all your files to g729 and force g729 on your endpoints |
| 18:45.47 | [TK]D-Fender | Want * to scale? Get high-density SIP PRI gateways and the only real load on * should be for Voicemail / IVR. |
| 18:46.01 | twisted[asteria] | [TK]D-Fender, heh... i've got a setup that i've installed doing just that |
| 18:46.04 | ManxPower | [TK]D-Fender, and call setup/teardown |
| 18:46.25 | twisted[asteria] | no, not call setup/teardown, that should be handled by the gateway/proxy |
| 18:46.27 | [TK]D-Fender | ManxPower : Takes a lot for basic SIP startup/teardown? |
| 18:46.35 | twisted[asteria] | oh wait |
| 18:46.38 | ManxPower | [TK]D-Fender, no. |
| 18:46.41 | twisted[asteria] | lol |
| 18:46.48 | [TK]D-Fender | twisted[asteria] : We didn't mention "proxy" in there yet.... |
| 18:46.49 | ManxPower | But it IS another thing asterisk would prolly be handleing |
| 18:46.55 | twisted[asteria] | [TK]D-Fender, i did. |
| 18:46.57 | twisted[asteria] | ;) |
| 18:47.15 | [TK]D-Fender | twisted[asteria] : Keep writing between the lines why don't you ;) |
| 18:47.35 | twisted[asteria] | [TK]D-Fender, sure thing ;) |
| 18:47.44 | [TK]D-Fender | ManxPower : I though the Sipura's just did G729 on a single channel to be cheap-asses :) |
| 18:48.33 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 18:48.34 | ManxPower | [TK]D-Fender, That could be the reason, but if you look at the licensing costs for G729 it's pretty cheap per channel when you get into high volume. |
| 18:48.34 | [TK]D-Fender | Mediatrix needs to drop their retail margins as their wholesale cost is nearly par for offering better functionality than Sipura/Linksys |
| 18:48.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:49.17 | [TK]D-Fender | ManxPower : I rarely OVERESTIMATE things... pessimistic as it may sound |
| 18:51.30 | kink0 | anybody knows about any cheap E1 PRI <-> BRI multiplexer/demultiplexer ? |
| 18:52.10 | kink0 | I have fews unused E1 ports, and I pretend to use it instead to buy fews FXO/FXS digium cards |
| 18:52.25 | kink0 | because there just on free PCI slot on the system. |
| 18:53.00 | [TK]D-Fender | BRI is never cheap in my awareness... |
| 18:53.08 | wundaboy | can anyone recomend an inexpensive consistent voip provider for us/canada? |
| 18:53.22 | twisted[asteria] | [TK]D-Fender, i could get bri back in tennessee for $24/mo |
| 18:53.28 | ManxPower | wundaboy, they all suck. I find that Teliax usually sucks a little bit less than most. |
| 18:53.33 | twisted[asteria] | cheaper than two phone lines, and faster, too ;) |
| 18:53.40 | twisted[asteria] | call setup/teardown wise |
| 18:53.59 | ManxPower | twisted[asteria], that price is a fluke created by the TN PUC. Almost all other states were 4x as expensive. |
| 18:54.02 | kink0 | [TK]D-Fender, I agree, will be ok also PRI-> analog(FXS/FXO) for this purpose to use a fews channels from the PRI and connect it to the telco |
| 18:54.14 | twisted[asteria] | ManxPower, yeah, it's still like $24 in TN, too ;) |
| 18:54.29 | twisted[asteria] | here in HSV it's only like $38 iirc |
| 18:54.48 | ManxPower | in Louisiana it was about $109 including taxes for a PRI |
| 18:54.48 | file[laptop] | for a BRI? |
| 18:54.52 | ManxPower | $300 install fee. |
| 18:54.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:55.01 | ManxPower | pperppp not PRI, BRI |
| 18:55.55 | wundaboy | .02/minute isnt bad |
| 18:56.04 | kink0 | here a PRI costs about 350 dls/mo ( Spain ) including all from the telco. |
| 18:56.10 | wundaboy | ManxPower: does teliax do 1/10ths o fa minute? or just whole minutes... |
| 18:56.34 | ManxPower | wundaboy, I have no idea. Prolly does 6 second increments, that seems pretty standard. |
| 18:56.45 | kink0 | but now I just need fews channels, so I was planning to use FXO/FXS digium cards , the problem is I can add just one card on one free PCI slot |
| 18:57.13 | ManxPower | kink0, Um, BRI and analog are different. |
| 18:57.30 | kink0 | an I have on that machine a TE405 card, where 3 ports still unused |
| 18:57.56 | kink0 | ManxPower, yes, for my purpose is indiferent for me , I can connet it to the BRI telco or to the analog telco |
| 18:57.58 | wundaboy | do you know if teliax will do local incoming did's? |
| 18:58.05 | backblue | kink0: you are my neighbor. :P |
| 18:58.07 | wundaboy | also whats the VOIP did porting situation for everyone? |
| 18:58.17 | ManxPower | kink0, neither of those ideas use the extra ports on your card. |
| 18:58.19 | kink0 | hi backblue |
| 18:58.40 | ManxPower | wundaboy, I don't know. anyone that relies on a VoIP telco as their primary phone service is a moron. |
| 18:58.43 | backblue | twisted[asteria]: if you have 500000 users you will not want to use 1 asterisk i hope! :) you need to cluster it. |
| 18:58.44 | kink0 | ManxPower, no, for this I was looking for an ISDN multiplexer |
| 18:59.31 | ManxPower | So you are looking for a device that takes a CT1 or PRI connection from Asterisk on one of the extra ports and then converts it to BRI for use with PRI phones? |
| 18:59.44 | kink0 | ManxPower, right !! |
| 18:59.53 | ManxPower | kink0, Keep dreaming. |
| 19:00.00 | kink0 | ooopssssssss :( |
| 19:00.44 | kink0 | well then I will to change my telco BRI for PRI, is cheapest than buy a lot of TDM cards |
| 19:01.03 | exonic | Anyone care to see my ncurses asterisk interface? I attempted to mimic the basic layout of a MaxTNT. Screenie here: http://flickr.com/photo_zoom.gne?id=109744414&size=o |
| 19:01.14 | ManxPower | kink0, and not use the TE405P? |
| 19:01.32 | kink0 | ManxPower, I am ussing now just one port |
| 19:01.46 | ManxPower | kink0, TE405P does not work with BRI |
| 19:01.49 | kink0 | I bough with four ports for future expansions |
| 19:02.07 | kink0 | yes , for that I was looking for something to demux PRI to BRI |
| 19:02.28 | kink0 | but I only found one expensive equipment from Ericson ( about 6000 dls ) |
| 19:03.03 | kink0 | really that Ericson is a full PBX, but I just need the demultiplexing functions. |
| 19:04.38 | Lino` | hmmm exonic that looks very nice |
| 19:05.35 | guilherme-jorge | where do I download of the asterisk manager API? |
| 19:06.01 | exonic | Lino`, good. It will be available shortly. Is anyone aware of a place to host asterisk applications? |
| 19:06.25 | Lino` | hmmm if you want to do it as an opensource thingy, maybe sourceforge or freshmeat |
| 19:06.58 | ManxPower | guilherme-jorge, It is included with Asterisk |
| 19:07.35 | exonic | Lino`, sf is too popular, cvs system is always down. :) |
| 19:09.27 | exonic | I'll find a hosting solution. |
| 19:10.15 | nextime | exonic : why are you using a place that need signup to view the screenshot? |
| 19:10.28 | Lino` | hmmm |
| 19:10.32 | Lino` | flickr |
| 19:10.39 | Lino` | flickr usually does not need signup |
| 19:11.15 | nextime | Lino` : i know, but if i try to get the url of exonic i get a signup page. |
| 19:11.18 | Lino` | yeah |
| 19:11.27 | Lino` | you can set this up in the configuration |
| 19:13.39 | exonic | Is tehre a place to host this image? damn flickr tells me the photo is public |
| 19:14.10 | exonic | I'm currently trying to find a place to host the project as well ;) |
| 19:14.37 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 19:14.44 | Seldon1975 | anyone here used a DigitNetworks X100P ? |
| 19:15.58 | Lino` | i can host your image, no problem |
| 19:16.03 | x86 | exonic: I WANT! |
| 19:16.08 | x86 | exonic: written in perl? |
| 19:16.19 | Lino` | if you want i can also host your project |
| 19:16.34 | Lino` | if its not exceeding the magical 75gb / month mark |
| 19:16.34 | Lino` | :D |
| 19:16.41 | *** join/#asterisk bmg505 (n=leon@dsl-165-157-56.telkomadsl.co.za) |
| 19:16.48 | x86 | exonic: there's always berlios.de |
| 19:17.02 | Lino` | yeah thats a good place |
| 19:17.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@88.153.145.52) |
| 19:17.13 | x86 | exonic: they have no bandwidth limitations, support SVN (including WebSVN), and all kinds of other things |
| 19:17.22 | x86 | exonic: i use them for one of my projects |
| 19:17.52 | Seldon1975 | anyone here used a DigitNetworks X100P ? |
| 19:18.01 | x86 | Seldon1975: no repeating please |
| 19:18.22 | Seldon1975 | x86: ever? whats the acceptable delay? |
| 19:18.35 | x86 | Seldon1975: >1 hour |
| 19:18.44 | Seldon1975 | x86: sheesh! |
| 19:19.02 | Lino` | hehe |
| 19:19.20 | exonic | alright, i'll just host it on google pages. forgot about that thing! http://sig.lange.googlepages.com/assman |
| 19:19.23 | cpm | what's a digitnetworks x100p? |
| 19:19.30 | Lino` | lol |
| 19:19.32 | exonic | x86, i'll check out berlios.de for svn access |
| 19:19.38 | Lino` | a cheap and dumb card "made for asterisk" |
| 19:19.41 | x86 | OMGWTFBBQ ITS A BOUNCING... ERR... iDunno! |
| 19:19.42 | Lino` | analog phone interface card |
| 19:19.44 | x86 | :P |
| 19:19.58 | Seldon1975 | cpm: http://www.digitnetworks.com/store/product_info.php?products_id=28 |
| 19:20.00 | Nivex | http://tinyurl.com/zuath |
| 19:20.07 | iDunno | :) |
| 19:20.17 | Lino` | if you look into your toilet after having a good s**t you might find the x100p as well, i hate it. |
| 19:20.25 | Seldon1975 | Lino`: why? |
| 19:20.35 | Seldon1975 | Lino`: what are the specific issues? |
| 19:20.51 | cpm | looks dodgy to me, I wouldn't do it. |
| 19:20.52 | Lino` | because its a b**ch getting it to work and analog devices disgust me |
| 19:21.16 | Seldon1975 | Lino`: what would you use if you wanted a single analog line into your * box |
| 19:21.22 | cpm | one thing about cheap stuff, you can count on it being cheap. |
| 19:21.32 | Lino` | THATs a good question |
| 19:21.51 | Seldon1975 | cpm: yeah, Im trying to work out the best alternative and what are the specific deficiencies of the x100 |
| 19:22.20 | Lino` | its about 8€ so i tried it as well but i stopped trying it after 1 hour or so |
| 19:22.33 | Lino` | now i have several cologne chip powered ISDN cards up and running |
| 19:22.44 | *** join/#asterisk darby_t (i=darby_t@djw141.neoplus.adsl.tpnet.pl) |
| 19:23.09 | Seldon1975 | digitnetworks claims the x100 is 100% compatible with * |
| 19:23.22 | Seldon1975 | they even have * driver messages on their site |
| 19:23.26 | Lino` | yeah |
| 19:23.28 | Lino` | i know |
| 19:23.36 | Seldon1975 | is that a crock? |
| 19:23.40 | Lino` | but that only works as long as you don't have any custom stuff |
| 19:23.56 | Seldon1975 | custom as in....? |
| 19:23.57 | Lino` | i don't know if it works at all |
| 19:24.20 | Lino` | custom as in having a * installation which is not 99% done the way it is meant to be |
| 19:24.21 | Nivex | from what I heard even the digium x100 was a pain... hence why they don't sell them anymore |
| 19:24.37 | Lino` | yeah |
| 19:24.44 | Seldon1975 | who dont sell them? |
| 19:24.47 | Lino` | digium |
| 19:24.51 | Seldon1975 | oh |
| 19:25.18 | Seldon1975 | so I guess the real question is still: what would you use for a single analog trunk line |
| 19:25.37 | *** join/#asterisk Assid (n=assid@203.115.64.13) |
| 19:26.32 | Lino` | that is indeed a problem |
| 19:26.47 | Lino` | are you from europe? |
| 19:26.55 | Seldon1975 | no, in Canada |
| 19:27.07 | Lino` | hmmm |
| 19:27.34 | *** join/#asterisk m29poff (n=root@84.5.66.36) |
| 19:28.27 | Seldon1975 | well I just bought a x100 online for $25 - if it doesnt work I'll consider it a sunk cost |
| 19:28.36 | m29poff | HI, I've hangup problem with a TDM 11B card. Anyone around here to help me ? |
| 19:28.37 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:28.52 | Seldon1975 | at that price I figure its worth a gamble but my ears are open for a superior alternative |
| 19:29.20 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 19:30.29 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 19:31.33 | Lino` | hmmm |
| 19:32.57 | zoa | the x100p are bad |
| 19:33.01 | zoa | and all clones are too |
| 19:33.07 | zoa | they are not made for this kind of job |
| 19:33.16 | zoa | and have a massive amount of false hangups |
| 19:33.21 | zoa | echo |
| 19:33.31 | zoa | callerid problems |
| 19:33.34 | cpm | what do you suppose is the chance that this $25 cheapo card is this card http://www.x100p.com/products_1.htm |
| 19:33.40 | zoa | 100% |
| 19:33.43 | zoa | they are all the same |
| 19:34.01 | cthompson | they're Intel softmodems |
| 19:34.05 | cpm | my personal experience with this, http://www.x100p.com/products_2.htm |
| 19:34.16 | cpm | is that it was a real waste of time and money, which is too bad. |
| 19:34.21 | zoa | true |
| 19:34.28 | zoa | and you paid a lot for it |
| 19:34.29 | zoa | :) |
| 19:34.59 | [av]bani | cheap telco equipment = shit performance |
| 19:35.20 | [av]bani | youre not going to get decent performance for less than $100 a port, period |
| 19:35.24 | salviadud | what's better than telco? |
| 19:35.30 | cpm | Not all that much, considering it is native iax, and has all the bells and whistles that the iaxy doesn't. However, unlike the iaxy, it's a piece of crap. |
| 19:36.25 | cpm | [av]bani, $100 a port is a good figure. |
| 19:36.25 | [av]bani | cpm: most people who have problems are buying at $50 or less |
| 19:36.25 | *** join/#asterisk RoyK (n=roy@130.80-203-176.nextgentel.com) |
| 19:36.58 | [av]bani | you want real dsp based echo cancellers, not pc software based ones. |
| 19:37.06 | cpm | yeah, the S100-FX is about $82 by the time it's in your hands. That makes the next iaxy cost about $182. :) |
| 19:37.14 | [av]bani | with a dsp directly on the card |
| 19:37.48 | [av]bani | and echo cancelling is not something you want to screw around with in voip installations |
| 19:38.00 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 19:38.09 | [av]bani | people think they can do it on th cheap and think they're being clever -- they arent. |
| 19:38.13 | TheCompWiz | anyone in here work with the miax client? |
| 19:38.29 | salviadud | expensive hardware rocks |
| 19:38.34 | Lino` | :D |
| 19:38.35 | salviadud | cause it doesn't fail |
| 19:38.50 | Lino` | i use asterisk on a dual xeon d machine |
| 19:38.56 | Lino` | thats overkill |
| 19:38.57 | salviadud | on the long run... it is cost effective |
| 19:39.07 | salviadud | lino, how many channels? |
| 19:39.13 | salviadud | on average |
| 19:39.13 | Lino` | more than i'll ever need |
| 19:39.23 | Lino` | at the moment 0 because it is still in development |
| 19:39.30 | salviadud | is it for a call center? |
| 19:39.42 | Lino` | no. it is for a small company, i said it is overkill |
| 19:39.53 | Lino` | we used to have 4 channels outbound |
| 19:40.00 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 19:40.00 | TheCompWiz | has anyone setup a trunk using a cell phone? |
| 19:40.03 | Lino` | but i want to run more applications on it |
| 19:40.09 | salviadud | you could do that with a decent 4 ghz intel |
| 19:40.11 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 19:40.15 | salviadud | 4 chans |
| 19:40.22 | Lino` | yeah but i had the machine in stock |
| 19:40.28 | Lino` | so why shouldnt i use it? |
| 19:40.39 | salviadud | i guess you are right sir! |
| 19:40.41 | salviadud | overkill it is |
| 19:40.54 | Lino` | absolutely |
| 19:41.01 | Lino` | i'm thinking about virtualization or something |
| 19:41.04 | squinky86 | I'm not too familiar with the asterisk source code yet; but, if I have the name of a channel as a character pointer, how would I convert that to an ast_channel struct? |
| 19:41.07 | Lino` | or starting a call center :-P |
| 19:41.28 | salviadud | yeah, call centers are funny |
| 19:41.31 | salviadud | i used to work at one |
| 19:41.37 | salviadud | we used horrible software |
| 19:41.41 | salviadud | all windows |
| 19:41.42 | octothorpe | I have an x100p from x100p.com on my test box and it works well |
| 19:41.43 | RoyK | goða kvöldið |
| 19:41.45 | salviadud | super crap |
| 19:41.59 | salviadud | they fired me cause i used linux live-cds on the comps |
| 19:42.11 | salviadud | too smart for those monkeys |
| 19:42.35 | salviadud | asterisk is so cool. |
| 19:42.45 | PakiPenguin | hey RoyK |
| 19:43.04 | RoyK | hi, mr paki :) |
| 19:43.20 | salviadud | hey compwiz, what distro are you using? |
| 19:43.20 | PakiPenguin | how are you today? |
| 19:44.25 | Zodiacal | how come asterisk saves a unavail.wav and an unavail.WAV |
| 19:44.43 | Qwell[] | Zodiacal: Because those are two different formats. |
| 19:44.53 | Qwell[] | check your voicemail.conf for the formats it uses...you can change them |
| 19:44.56 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 19:45.02 | Zodiacal | qwell okie thanks! |
| 19:45.03 | TonyM | hey guys, are the asterisk mailing lists down? nothing for several hours at least |
| 19:45.20 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:45.24 | *** join/#asterisk chr|s_ (n=chris@217.171.52.108) |
| 19:45.24 | Qwell[] | Zodiacal: The same is true for all greetings |
| 19:45.58 | Zodiacal | format=wav49|wav |
| 19:46.02 | Zodiacal | know the difference off hand? |
| 19:46.09 | Qwell[] | about 49 |
| 19:46.17 | Zodiacal | yea |
| 19:46.31 | exonic | TonyM, I havn't gotten anything for ~20 hours |
| 19:46.47 | TonyM | ok, thanks - it's not just me then ;) |
| 19:46.50 | salviadud | adios locoooos! |
| 19:47.06 | *** join/#asterisk octothorpe__ (n=octothor@198.60.73.230) |
| 19:47.13 | exonic | Zodiacal, one is a compressed wav format, I can't remember which :) |
| 19:47.28 | Zodiacal | wav49 is probably compressed |
| 19:47.30 | [TK]D-Fender | Zodiacal : wav49 is Windows standard wav, "wav" alone isn't quite as compatible |
| 19:47.41 | Zodiacal | ahh |
| 19:47.42 | Zodiacal | oops |
| 19:47.43 | Zodiacal | :P |
| 19:47.57 | *** join/#asterisk lab0rized (n=lab0rize@port927.ds1-fa.adsl.cybercity.dk) |
| 19:48.14 | lab0rized | Hello anyone succesfully had installed asterisk on a ubuntu system ? |
| 19:48.14 | exonic | unavail.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz |
| 19:48.14 | exonic | unavail.gsm: data |
| 19:48.14 | exonic | unavail.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 19:48.16 | Zodiacal | do you guys know off hand how to increase the volume of my greetings? |
| 19:48.44 | exonic | Zodiacal, sox is a good utility |
| 19:49.11 | Zodiacal | that reencodes them tho right? |
| 19:49.34 | exonic | Zodiacal, correct |
| 19:49.37 | Zodiacal | is there another setting somewhere? that just ups the volume for the playback,background commands or am i dreaming |
| 19:50.03 | tehdely | lab0rized: it's not possible. in fact, asterisk will self-destruct and redirect all calls to tt-weasels if it detects an ubuntu system |
| 19:50.22 | exonic | Zodiacal, not aware of an option |
| 19:50.22 | Zodiacal | exonic cuz new voicemail recordings are quiet too |
| 19:50.22 | Zodiacal | but the operator is fine |
| 19:50.51 | Zodiacal | would i have to run sox after every new voicemail recording? |
| 19:50.59 | exonic | Zodiacal, hmm. g(#) - Use the specified amount of gain when recording the voicemail |
| 19:51.09 | Qwell[] | yeah, g() works well |
| 19:51.14 | exonic | Zodiacal, show application VoiceMail |
| 19:51.23 | Zodiacal | okie i'll go play with that, thanks again! |
| 19:51.35 | lab0rized | ah very funny! |
| 19:52.07 | exonic | lab0rized, any problem in particular on ubuntu? |
| 19:54.18 | lab0rized | yes, i have installed all the nessecary lib's and it still say that i need termcap =? |
| 19:54.22 | *** join/#asterisk moy (n=moy@dsl-201-129-133-43.prod-infinitum.com.mx) |
| 19:54.48 | TheCompWiz | can anyone tell me the differences between chan_bluetooth & miax? ... which is better (supported/works)? etc? |
| 19:55.12 | TheCompWiz | (both supposed to be bluetooth capable methods of using cell phones as a trunk) |
| 19:55.26 | moy | hi... i have red alarm in the 4 spans of my digium for E1 config, what does red alarm means? |
| 19:55.48 | [TK]D-Fender | moy : means "no link" |
| 19:55.53 | Qwell[] | moy: bad things... is it plugged in? |
| 19:56.02 | Qwell[] | with proper cables |
| 19:56.08 | exonic | lab0rized, you've got me clueless. asterisk says you need termcap ? |
| 19:56.39 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 19:56.50 | moy | Qwell: thanks for your answer, actually i dont know, is supposed to be plugged in, but im configuring the card remotely |
| 19:57.13 | moy | chan_unicall seems to be fine, detects the channels |
| 19:57.15 | Tall-guy | Hey lads, anyone here with Asterisk to Nortel MICS integration experience? |
| 19:57.26 | lab0rized | Yeah, or well no not completely asterisk but it is an error i get when compiling asterisk ? |
| 19:57.40 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 19:57.57 | lab0rized | error: termcap support not found ! |
| 19:58.10 | moy | the people that have physical access to the server tell me that is connected |
| 19:58.12 | mroth_imm | anyone have numbers available on the average count of calls handled on their server per day? |
| 19:59.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@88.153.145.52) |
| 19:59.33 | *** join/#asterisk edjo_ (n=abla@p54AD10B8.dip0.t-ipconnect.de) |
| 19:59.36 | Qwell[] | mroth_imm: yep |
| 19:59.39 | moy | [TK]D-Fender: couldnt be then a telco problem? |
| 19:59.41 | Qwell[] | ends up at about 3 |
| 19:59.59 | mroth_imm | 3 what? :) |
| 20:00.02 | exonic | mroth_imm, yeah.. around 1000 every day |
| 20:00.29 | Qwell[] | calls per day |
| 20:00.29 | mroth_imm | cool...just trying to get a frame of reference |
| 20:00.51 | *** part/#asterisk edjo_ (n=abla@p54AD10B8.dip0.t-ipconnect.de) |
| 20:00.56 | mroth_imm | Qwell[]: wanna trade? |
| 20:01.22 | exonic | Qwell[], are you talking concurrent? |
| 20:02.12 | mroth_imm | i have no idea what the average user is putting through asterisk |
| 20:02.34 | exonic | I have another asterisk box that is a basic IVR that handles around 2500 every dya |
| 20:02.44 | mroth_imm | right now we're approximately 90 concurrent with recording throughout business hours...been hitting 10000 a day |
| 20:03.11 | exonic | mroth_imm, nice! What kind of hardware? Digium? All SIP? |
| 20:03.44 | mroth_imm | all sip...pstn terminated by a cisco gateway --- SIP --- Asterisk Server --- SIP --- Snom 320s |
| 20:04.35 | mroth_imm | i kinda left userland for a while during the prep for launch, so i lost track of what everyone else was doing |
| 20:04.36 | exonic | Awesome. I plan to find something besides my digium when my call volume grows beyond my TE410 |
| 20:04.52 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-143-33.dsl.irvnca.pacbell.net) |
| 20:05.33 | mroth_imm | as long as things don't go south, our setup will be public...i'd love to see the improvements everyone could suggest |
| 20:05.52 | mroth_imm | got a little isolated along the way...just trying to *make things work* for the bosses :) |
| 20:05.55 | octothorpe | mroth_imm I have been trying to set uo my asterisk to talk with ccm 4 and can't get it working. Any pointers on the asterisk side? |
| 20:06.29 | mroth_imm | sorry, no experience there... |
| 20:06.44 | mroth_imm | what is call manager running on? |
| 20:07.29 | mroth_imm | we have two as5400s, we own them, but contracted out their management to MCI |
| 20:07.37 | mroth_imm | i think my head would've exploded otherwise |
| 20:07.55 | octothorpe | mroth_imm: could you point me in the right direction to get my * talking with callmanager? |
| 20:08.30 | mroth_imm | i think i could only be of help if it's running on similar hardware... |
| 20:09.02 | mroth_imm | is call manager acting as a sip gateway, or is it strictly providing call management? i'm unfamiliar with it |
| 20:09.32 | *** join/#asterisk sepski (n=sep@217.17.211.51) |
| 20:09.42 | *** join/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 20:09.48 | *** part/#asterisk sepski (n=sep@217.17.211.51) |
| 20:10.17 | *** join/#asterisk jradford (n=jradford@hoss.npl.com) |
| 20:10.37 | octothorpe | Call Manager will be working as a sip gateway, here is what I would like to do: ipphone -- sip -- asterisk -- sip -- callmanager -- out |
| 20:11.12 | *** part/#asterisk jradford (n=jradford@hoss.npl.com) |
| 20:11.24 | octothorpe | I just thought I would ask you as you seem to have an active system talking between asterisk and CallManager |
| 20:11.26 | jmanq | Hey, I have posted earlier (today and yesterday) about not being able to make a call with my sip phone to a PSTN line. I have resolved this problem |
| 20:11.38 | TheCompWiz | I'm sooo freaking lost... |
| 20:11.41 | TheCompWiz | I hate that feeling. |
| 20:11.54 | jmanq | I just wanted to share my success with you guys in case it could help anyone in the future |
| 20:11.55 | jmanq | Evidentally the incoming T1 was using winkstart and I had the T1 configured for kewlstart |
| 20:12.09 | jmanq | so yeah, stupid me |
| 20:12.21 | jmanq | I would like to thank everyone who helped me along my way though |
| 20:12.31 | jmanq | Especially jsharp |
| 20:12.41 | jmanq | I appreciate all of your time and effort |
| 20:16.22 | mroth_imm | octothorpe: seems pretty similar... |
| 20:16.26 | *** join/#asterisk |omni| (i=rob@c-67-185-96-86.hsd1.wa.comcast.net) |
| 20:16.48 | |omni| | man..our nufone link is working really well since I moved the pbx out to our colo |
| 20:17.00 | guilherme-jorge | hello all, I'm trying discover how to execute a CLI command through AGI script, but I didn't have success. I would like to know a codec used in a call through AGI script, but I don't know how to do this?!?!? I already try to find this in asterisk manager, but... |
| 20:17.36 | VxJasonxV | Has anybody made a 'listen' or 'dummy' SIP client? |
| 20:17.45 | VxJasonxV | i.e. one for listening/event purposes that isn't actually a phone? |
| 20:18.11 | octothorpe | mroth_imm: could you pastebin a sterlized config (sip.conf) from your asterisk setup, I can't seem to get my asterisk talking to callmanager |
| 20:18.11 | mroth_imm | i'll point you to what we provided to cisco as a starting point...keep in mind they dealt with it from there on out |
| 20:18.26 | octothorpe | mroth_imm: thanks |
| 20:18.40 | mroth_imm | i'll show you the pertinent parts of sip.conf...np |
| 20:18.41 | *** join/#asterisk gammacoder (n=chatzill@207.67.51.249) |
| 20:18.50 | mroth_imm | gimme a minute |
| 20:18.54 | exonic | guilherme-jorge, http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Command |
| 20:19.02 | octothorpe | mroth_imm: that should help (hopefully) |
| 20:19.39 | *** join/#asterisk backblue (n=moo@87-196-15-19.net.novis.pt) |
| 20:20.07 | guilherme-jorge | exonic: Sorry, but do you know how to use this in AGI? |
| 20:20.35 | mroth_imm | octothorpe: here is something where we started on the AS5400 config...search the users list for more <http://ertw.com/blog/2005/04/19/asterisk-and-an-as5350-sip-peer/> |
| 20:21.17 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 20:21.38 | gammacoder | anyone seen SIP extensions behind a remote NAT translation stay online, then recieve a call, hang up the call, then SIP goes offline (the remote NAT is a Cisco PIX 501) |
| 20:21.40 | exonic | guilherme-jorge, if it's possible, read the wiki at http://www.voip-info.org/wiki-Asterisk+AGI |
| 20:22.27 | gammacoder | i've got nat=yes and qualify=yes |
| 20:22.50 | *** part/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 20:23.23 | exonic | gammacoder, i've had problems with routers not keeping the port open to the NAT'd SIP device. I enabled the NAT keep alive option on my Sipura. |
| 20:23.36 | Zodiacal | voicemail with the g option doesn't seem to work, either that or i don't know the exact syntax.. this wiki doesn't show that voicemail has the g option, but show application voicemail does.. anyone know how to use it? heres the voice mail wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail |
| 20:24.01 | Zodiacal | exonic qwell |
| 20:24.07 | Tall-guy | gamma: is the received call "completed" and successfull or does it get dumped prematurely? |
| 20:24.35 | exonic | Zodiacal, Trust 'show application ...' more than the outdatd wiki. |
| 20:24.42 | gammacoder | exonic: yeah - that's exactly what I'm thinking, but the remote sysadmin believes the translations are still active |
| 20:24.48 | Zodiacal | i tried voicemail(g(5)100), voicemail(100g5) etc.. no go |
| 20:25.16 | Zodiacal | i must be doing something really supid eh? |
| 20:25.56 | exonic | Zodiacal, VoicemailMain(u1234@voicemail1|g(3)) |
| 20:25.58 | gammacoder | Tall-guy: the call is completed sucessfully. user hangs up, then my sip qualify starts to fail |
| 20:26.27 | Zodiacal | exonic ah voicemailmain() thanks! |
| 20:26.43 | exonic | Zodiacal, that's a typo |
| 20:27.13 | Tall-guy | gamma: do you have access to the pix config? |
| 20:27.19 | exonic | Zodiacal, I meant: VoiceMail(u1234@voicemail1|g(3)) |
| 20:28.26 | mroth_imm | octothorpe: stripping down sip.conf for you... |
| 20:29.52 | *** join/#asterisk [Outcast] (n=outcast@222-152-110-218.jetstream.xtra.co.nz) |
| 20:29.55 | TheCompWiz | anyone know the default password for the AMP operator thingy? |
| 20:29.56 | *** join/#asterisk Ruis (n=ruise@68.178.8.80) |
| 20:30.14 | [Outcast] | is there anyway to do Multiple MWI on a single device? |
| 20:30.30 | [TK]D-Fender | [Outcast] : as in indicate for multiple boxes? |
| 20:30.36 | [Outcast] | yes |
| 20:30.53 | [av]bani | [TK]D-Fender: do you use idle image, or idle page on ip601? |
| 20:30.57 | [TK]D-Fender | [Outcast] : yes, just do multiple "mailbox=100@default" like statements. |
| 20:31.03 | [TK]D-Fender | [av]bani : yup |
| 20:31.23 | [av]bani | [TK]D-Fender: which is better for just displaying a logo? |
| 20:31.24 | gammacoder | Tall-guy: nope |
| 20:31.25 | [TK]D-Fender | [av]bani : I've got AMI queue stats on 10 sec interval on it |
| 20:31.27 | [Outcast] | [TK]D-Fender: what is you are using realtime? |
| 20:31.27 | Ruis | Is there a way to setup asterisk to call into a conference call at a certain time, record the call, then hang up after a certain time? I'm an asterisk newb and don't know where to start looking for something like that. |
| 20:31.44 | octothorpe | mroth_imm: you're awesome |
| 20:31.47 | Tall-guy | gamma: too bad...I could check that for you....:) |
| 20:31.51 | mog_work | mroth_imm, is awesome |
| 20:31.58 | [Outcast] | s/is/if |
| 20:32.06 | [TK]D-Fender | Idle page is what I use, not image. Page is better as you can mix image and text obviously. |
| 20:32.12 | Tall-guy | gamma: does it seem that the completed call and hangup starts this NAT failure? |
| 20:32.17 | [TK]D-Fender | [Outcast] : No realtime, sorry |
| 20:32.23 | [av]bani | [TK]D-Fender: does it scale images? |
| 20:32.24 | gammacoder | Tall-guy: yes |
| 20:32.27 | [Outcast] | hmmm |
| 20:32.28 | mroth_imm | octothorpe: http://pastebin.ca/44949 |
| 20:32.36 | mroth_imm | mog_work is awesoER!! |
| 20:32.45 | mog_work | nahh |
| 20:32.47 | mroth_imm | irc ate my m |
| 20:32.51 | mog_work | lol |
| 20:32.57 | [TK]D-Fender | [av]bani : I don't believe so. Make your pages work for IT, not the other way around. I GIMP'd my company logo to spec for mine. Looks nice. |
| 20:33.15 | gammacoder | Tall-guy: there is a direct correlation - ususally within 1 minute of hangup, without any call to the remote extension, it'll stay online indefinately |
| 20:33.30 | [av]bani | [TK]D-Fender: it says to reduce to 4bpp, i can't see how to do that with gimp |
| 20:33.31 | mroth_imm | octothorpe: stripped out ips and hostnames, but otherwise complete...the sip peers at the bottom are of the most interest to you |
| 20:33.34 | CrashHD | is there a way to set a dialing prefix in sip.conf for a peer? |
| 20:33.45 | mroth_imm | octothorpe: the stripped ips there are the ips of our AS5400s |
| 20:33.53 | Tall-guy | gamma: will a reboot of the physical phone fix the prob? |
| 20:34.16 | Hmmhesays | somethings will never change, they just stand there looking backwards half unconcious from the pain |
| 20:34.26 | gammacoder | Tall-guy: yep - a reboot of the phone puts it back online until the next call is hung up |
| 20:34.54 | Tall-guy | gamma: what kinda phone? |
| 20:34.59 | *** part/#asterisk moy (n=moy@dsl-201-129-133-43.prod-infinitum.com.mx) |
| 20:35.05 | [TK]D-Fender | [av]bani : I think it'll accept 256 really though... |
| 20:35.38 | gammacoder | Tall-guy: its almost like the PIX is mangling packets or something (i've had problems with pix fixup for smtp for instance) but this happens with sip fixup on or off (according to the admin) |
| 20:36.01 | octothorpe | mroth_imm: thanks, I will digest that and see what I can't do. Thanks a million |
| 20:36.03 | gammacoder | Tall-guy: Grandstream gpx-2000 with 1.0.2.13 firmware |
| 20:36.04 | [TK]D-Fender | [av]bani : 8 bit is ok. |
| 20:36.22 | Tall-guy | gamma: try a test with x-lite/eyebeam/xpro softphone, or something similar....then you can blame the phone or the pix. |
| 20:36.31 | [av]bani | [TK]D-Fender: whats the config clause for the idle url? |
| 20:36.35 | *** join/#asterisk ibob63 (n=hp@bb-87-82-21-204.ukonline.co.uk) |
| 20:36.36 | gammacoder | Tall-guy: there are 3 gpx-2000 phones behind the PIX all three act the same |
| 20:36.47 | Tall-guy | gamma: so try a softphone :0 |
| 20:36.54 | Tall-guy | gamma: will tell you whether its the phone or the pix |
| 20:36.54 | *** part/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 20:37.07 | *** join/#asterisk ropeguru_work (n=ropeguru@65-121-222-5.dia.static.qwest.net) |
| 20:37.08 | ibob63 | can anyone tell me where to set the call-id for sip registration? |
| 20:37.13 | Tall-guy | gamma: oh, tweak...hang on... |
| 20:37.19 | gammacoder | Tall-guy: will do, but I can't get on that site until tomorrow |
| 20:37.28 | Tall-guy | gamma: 3 phones.....are they static nat'd? or do you know? |
| 20:38.00 | gammacoder | Tall-guy: no they are dynamic nat with one outside (routable) IP addr |
| 20:38.36 | [TK]D-Fender | [av]bani : http://pastebin.ca/44951 |
| 20:38.43 | Tall-guy | gamma: ok, forgive me if this sounds stupid, but I have to ask. Are all the phones listening on 5060? |
| 20:38.59 | [av]bani | yaytnx |
| 20:39.08 | [TK]D-Fender | [av]bani : Quite welcome |
| 20:39.32 | gammacoder | Tall-guy: yes, but the PIX translates the outside port |
| 20:39.39 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 20:40.01 | *** join/#asterisk jjanzer (n=jjanzer@lanthera.net) |
| 20:40.14 | gammacoder | Tall-guy: I see them on ports 1044, 1033, and 2003 right now |
| 20:40.32 | Tall-guy | gamma: I understand that (I'm a pix guy).... |
| 20:40.48 | MattB2 | hi all... is there an acceptable latency for SIP (G711) VoIP traffic? For example I have a VoIP server and the latency between the server and my VoIP provider is around 145ms. Is this going to cause problems? |
| 20:40.51 | mroth_imm | octothorpe: np, you know where to find me |
| 20:40.51 | *** part/#asterisk jjanzer (n=jjanzer@lanthera.net) |
| 20:40.57 | Tall-guy | gamma: when they fail, does asterisk still think they are reg'd? |
| 20:41.02 | MattB2 | latency is determined using a simple ping |
| 20:41.03 | *** join/#asterisk gongoputch (n=gongoput@c-68-82-194-31.hsd1.de.comcast.net) |
| 20:41.19 | gammacoder | Tall-guy: i hear ya (i'm mostly trying to make it all clear for my own understanding) |
| 20:42.04 | gammacoder | Tall-guy: yes still registered, but the sip qualify status turns to "UNREACHABLE" |
| 20:42.17 | *** join/#asterisk Sconk (n=klaus@c-5d0671d5.08-10-68617010.cust.bredbandsbolaget.se) |
| 20:43.31 | Tall-guy | gamma: would be usefull to look at the pix and see what it has the NAT/PAT set to for each particular phone when it fails. |
| 20:44.19 | Tall-guy | gamma: is it a pix 501? |
| 20:44.29 | gammacoder | Tall-guy: yes - PIX 501 |
| 20:44.35 | Tall-guy | gamma: 10 user , 50 user, or unlimited? |
| 20:44.45 | ibob63 | asterisk is failing to register with my media gateway via sip. In the debug I notice that the call id doesn't send the right ip address. You can see the debug here : http://pastebin/591437 |
| 20:44.59 | ibob63 | Can anyone tell me how to send the Call-ID? |
| 20:45.13 | ibob63 | or the Contact |
| 20:45.14 | ibob63 | ? |
| 20:46.10 | ibob63 | The server is the DMZ but it thinks it IP address is 192.168.1.70 and so it don't send its outside IP which is different |
| 20:46.30 | Tall-guy | ibob: yer pastebin url is broked |
| 20:46.46 | Sconk | hi i found this pange http://voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Network but all the links til the mailling list are 404 can some one point me to some place whit info? |
| 20:47.12 | gammacoder | Tall-guy: I think the admin bought a 10 user licence |
| 20:47.23 | ManxPower | ibob63, sounds like the classic Asterisk behind NAT issue, which is talked about in pretty much every source for asterisk docs |
| 20:47.27 | ManxPower | ~docs |
| 20:47.29 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:47.33 | gammacoder | Tall-guy: perhaps he's exceeding his limit w/ his other devices |
| 20:48.16 | Tall-guy | gamma: this is why I asked....if you can find out if its a 10 or a 50....and how many users you have....the pix just stops Nat'ing stuff when you hit the limit... |
| 20:48.16 | [TK]D-Fender | ManxPower : You know for the # of times I've explained it I should have made "~ASTNAT" or something to save the effort :) |
| 20:48.27 | ibob63 | Okay - here is the new pastebin http://pastebin.com/591437 |
| 20:48.34 | *** part/#asterisk ropeguru_work (n=ropeguru@65-121-222-5.dia.static.qwest.net) |
| 20:48.56 | ManxPower | ibob63, unlike most protocols, SIP addresses are embeded in the DATA part of the packet, no the header |
| 20:49.05 | ManxPower | so they can't be NAT'd |
| 20:49.13 | Tall-guy | gamma: if you do a "show local-host" on the pix....it will give you a stat like this: Interface inside: 6 active, 9 maximum active, 0 denied |
| 20:49.20 | jsharp | Unless you have a NAT server that's bright enough to mangle the packets. |
| 20:49.28 | jsharp | Which leads to all sorts of other wacky problems. |
| 20:49.34 | Tall-guy | gamma: if you see anything in "denied" ...it means you are exceeding the license. |
| 20:49.49 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 20:51.42 | gammacoder | Tall-guy: thanks for helping me get to the bottom of this admin's likely licencing issue - he's unavailable now, but I'm basically convinced you are right |
| 20:51.51 | Tall-guy | gamma: been there done that :) |
| 20:52.23 | Tall-guy | gamma: will the phones eventually start working again by themselves? |
| 20:52.25 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 20:53.27 | ibob63 | ManxPower: I think don't think it is a NAT issue - I just think that the incorrect IP address us being sent |
| 20:53.53 | ManxPower | ibob63, no, that is a NAT issue. |
| 20:54.14 | gammacoder | Tall-guy: I have never seen them come back online once they are "unreachable" |
| 20:54.22 | Dr-Linux | ibob63: what's the issue? :S |
| 20:54.25 | ManxPower | Asterisk has NO IDEA what the external IP address is and the router doesn't know how to modify the packet because the correct IP address is not in the header of the packets. |
| 20:54.28 | gammacoder | Tall-guy: without a reboot that is |
| 20:54.34 | ManxPower | But I won't argue with you about it. |
| 20:54.55 | Tall-guy | gamma: cause theoretically, once the NAT XLATE timeout value has been reached on the PIX, it will "free up" nat licenses again, and stuff will start working |
| 20:56.04 | ManxPower | ibob63, so if you follow the instructions for asterisk behind nat, asterisk will put the correct external addresses in the packets |
| 20:56.21 | gammacoder | Tall-guy: i hear ya, and that makes sense, I just haven't seen that behavior yet |
| 20:56.36 | ibob63 | ManxPower: Could you point me too these doc. I can't seem to find them... |
| 20:56.48 | Tall-guy | gamma: I'd be interested in hearing how you make out, I like to expand my pix/nat/asterisk knowledge as its the environ I live in. |
| 20:56.48 | ManxPower | ~docs |
| 20:56.50 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:57.08 | tasat | hi all, got a NAT issue I think: I've got the SIP (5060) and RTP ports forwarded to my server, but the remote proxy still seems to register SIP on a different port -- from what I understand my router may be at fault... any ideas? |
| 20:57.23 | ManxPower | The wiki has several methods of doing this, the one that uses externip/externhost and localnet= is the one to use. |
| 20:57.34 | tasat | in other words, does this sound right? |
| 20:57.49 | Tall-guy | we should start an asterNAT channel :) |
| 20:57.53 | ManxPower | tasat, yes, the exact same thing as ibob63 has issues with |
| 20:58.03 | ManxPower | Tall-guy, no, someone needs to write up a NAT doc. |
| 20:58.33 | tasat | ManxPower: sorry, missed that discussion... I'll go back and check |
| 20:58.37 | *** join/#asterisk dizzzan (i=dan@host.l8t.net) |
| 20:58.41 | [TK]D-Fender | ManxPower : Actually dile did already |
| 20:58.45 | [TK]D-Fender | file* |
| 20:58.53 | ManxPower | [TK]D-Fender, then point people to it |
| 20:59.18 | ibob63 | I am not lazy - i read loads of documention. does anyone know how to modify the call-id for when asterisk tries to register? |
| 20:59.56 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 21:00.23 | gammacoder | Tall-guy: I'll let you know |
| 21:00.42 | tasat | I've seen a number of documents, haven't seen anything mentioning port forwarding |
| 21:00.51 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 21:01.06 | tasat | or recommending an alternate router, etc. |
| 21:01.13 | ManxPower | ibob63, yes, you specify the external ip using the externip= setting in sip.conf |
| 21:01.32 | Hmmhesays | and audiocodes tanks again |
| 21:02.19 | guilherme-jorge | Is it possible execute a CLI command through AGI script? |
| 21:02.34 | Tall-guy | This isn't a bad primer on Asterisk-SIP-NAT issues : http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 21:03.01 | gammacoder | ibob63: externip=routable.ip.addr.ess and localnet=192.168.1.0/255.255.255.0 is what I use |
| 21:04.49 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 21:05.45 | ibob63 | Just tried modifing the externip and localnet neither have seemed to improved anything. :( |
| 21:05.50 | Egonis | I just made a macro to change CallerID's, and have a statement Set(CALLERID(name)=Name) -- however I get an error 'No application 'Set' for extension (blah, s, 1) |
| 21:05.54 | Hmmhesays | did you sip reload after that? |
| 21:06.03 | tasat | Tall-guy: thanks |
| 21:06.15 | Hmmhesays | malformed exten |
| 21:06.35 | ManxPower | guilherme-jorge, no. However, you can execute external processes from an AGI (like asterisk -rx "reload") and you can connect to sockets, like, for example, the manager interface. |
| 21:06.55 | ManxPower | Egonis, that is only available in 1.2.x or later |
| 21:07.00 | Egonis | ahhh |
| 21:07.05 | Egonis | then I better upgrade! :) |
| 21:07.28 | ManxPower | Eggplant, a different syntax is required for 1.0.x |
| 21:07.45 | ManxPower | "show applictions like callerid" might give you some help |
| 21:09.07 | Zodiacal | is it posible to use cat5 for POTS lines? |
| 21:09.15 | Zodiacal | anyone know the pinout off hand? |
| 21:09.16 | Zodiacal | :) |
| 21:09.21 | Tall-guy | zodia: yup |
| 21:09.43 | Zodiacal | tall-guy happen to know the pinout off hand? |
| 21:10.05 | gaupe | the middle pair |
| 21:10.11 | Tall-guy | zodiacal: standard cat5 pinout for ethernet uses 1,2,3,6....leaving 4-5 (middle pair) avail for pots |
| 21:10.38 | Tall-guy | zodiacal: so in a "regular" patch cable, pins 4-5 are already there for pots |
| 21:10.53 | Zodiacal | ahh coolness thanks! |
| 21:11.01 | [av]bani | [TK]D-Fender: 8bpp seems to not work |
| 21:11.04 | ManxPower | Standard cat 5 cable is amazing stuff. Works for Ethernet, T-1, Pots, PoE, all with no pinout changes |
| 21:11.11 | Tall-guy | zodiacal: of course you may have rj11/rj45 issues :) |
| 21:11.24 | Zodiacal | ya got a crimper :) |
| 21:12.00 | _Sam-- | Chuck Norris can win a game of monopoly without owning any property. |
| 21:12.05 | *** join/#asterisk acqua7 (n=bc290db0@customer-200-36-59-130.uninet.net.mx) |
| 21:12.19 | acqua7 | hi |
| 21:12.21 | acqua7 | <PROTECTED> |
| 21:12.22 | _Sam-- | There is no theory of evolution, just a list of creatures Chuck Norris allows to live. |
| 21:12.28 | Tall-guy | zodiacal: you could be a dufus like me and crimp 4 POTS (rj11s') onto each end of a cat5 cable and have a hydra-octopus :) |
| 21:12.39 | [TK]D-Fender | [av]bani : send me your photo. |
| 21:13.18 | azzie | Hey guys. Anybody has documentation for Vega100, by any chance?.. |
| 21:13.35 | Zodiacal | tall-guy is there a special pinout for that? that would be nice, cuz im just using it for a patch pannel |
| 21:13.52 | Zodiacal | like free heatshrink cable rap :) |
| 21:14.03 | Zodiacal | nice and neet all my lines in one cable |
| 21:14.20 | Zodiacal | should i use each color for a different line? |
| 21:14.29 | Zodiacal | i.e. whiteblue/blue, whitegreen/green, etc |
| 21:14.32 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:14.34 | Tall-guy | zodiacal: well, yes, and no...and I could get plenty of arguments about this....but generaly, stay with a PAIR ie: same color for each line. |
| 21:14.39 | Tall-guy | zodiacal: (yes) |
| 21:14.55 | Zodiacal | i don't want it to cause interference tho :/ |
| 21:14.59 | ManxPower | Tall-guy, those things give me the creeps. I have the massive urge to buy a 66 block every time I see one of those. |
| 21:15.06 | Tall-guy | manx: :) |
| 21:15.13 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 21:15.33 | Tall-guy | zodiacal: yeah, interference....go look at a standard 50 pair Telco pull..... :) |
| 21:16.07 | Zodiacal | i guess.. but cat5 is "twisted" is the standard telco lines twisted? |
| 21:16.07 | [av]bani | [TK]D-Fender: get dcc |
| 21:16.41 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 21:16.49 | *** join/#asterisk carb0n^ (n=carbo@137.101.17.34) |
| 21:16.51 | [TK]D-Fender | [av]bani : GREYSCALE IT. |
| 21:16.56 | carb0n^ | anyone good with phpagi ? |
| 21:16.58 | [av]bani | bleh |
| 21:17.03 | cpm | Hrmm, world of difference in a CAC AB1 and ABII. |
| 21:17.07 | Tall-guy | zodiacal: yes, but not as many twists per inch. |
| 21:17.15 | cpm | Anyone want a few AB1s? |
| 21:17.16 | [TK]D-Fender | [av]bani : And your size is too big IIRC |
| 21:17.30 | [TK]D-Fender | 208x110 |
| 21:17.33 | Zodiacal | tall-guy i can allways remove it if it doesn't work. and use one for each line.. ill give it a shot and see what happens :) thanks again! |
| 21:18.12 | Tall-guy | zodiacal: I'm using one....don't tell Manx |
| 21:18.14 | [av]bani | its 145x128, the voip-info page said that wasy ok |
| 21:18.15 | acqua7 | <PROTECTED> |
| 21:18.33 | pv2b | Zodiacal: if you don't get it working, try changing the speed to 10 Mbit/s instead. |
| 21:18.46 | pv2b | Zodiacal: that should work as long as your cable conforms to the category 3 specifications. |
| 21:18.53 | Zodiacal | pv2b no data, just POTS |
| 21:18.57 | Tall-guy | pv2b: just phone man... |
| 21:18.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:19.04 | pv2b | oh, sorry, right. |
| 21:19.09 | *** join/#asterisk bigjb (n=bigjb@82-36-140-105.cable.ubr02.perr.blueyonder.co.uk) |
| 21:19.26 | pv2b | oh, running phone over cat 5. should work. :-) |
| 21:19.32 | [av]bani | er nm |
| 21:19.34 | [TK]D-Fender | [av]bani : ok, well fix the colour part first, then the size if it doesn't co-operater |
| 21:20.03 | pv2b | (really asking for trouble though, when you get PoE devices or gigabit ethernet in the mix ;-) |
| 21:20.19 | Tall-guy | pv2b: not with rj11's on then end :) |
| 21:20.34 | [av]bani | wtf? voip-info says to save it as color... |
| 21:20.55 | [av]bani | [TK]D-Fender: how do you save 4bpp in gimp? i dont see any option for it |
| 21:21.18 | [av]bani | nm |
| 21:21.22 | bigjb | Tall-guy, cant use rj11's on cat5? |
| 21:21.22 | [av]bani | need to convert to indexed |
| 21:21.54 | Tall-guy | bigjb: no, I'm just sayin that you wouldn't mixfuze an rj11 and an rj45 in your wiring closet very easily. |
| 21:21.56 | [av]bani | yay |
| 21:22.03 | bigjb | =oD |
| 21:22.57 | kink0 | cat5 is not good for rj11 ? |
| 21:23.26 | [TK]D-Fender | [av]bani : I think I passed mine through GIMP + MSPaint before I was through with it... |
| 21:23.40 | [TK]D-Fender | In gimp when you save theres a break out box for it IIRC |
| 21:23.51 | tasat | hi, question about 'fromdomain' in sip.conf... can this be a dyndns domain name? it looks like it causes [callid]@domain |
| 21:24.04 | tasat | what exactly is this doing? |
| 21:24.09 | [av]bani | [TK]D-Fender: you need to convert to indexed in gimp, and give a max 16 color palette |
| 21:24.35 | bigjb | kink0, no problem with rj11 on cat5 it just doesnt fit very well without stripping back |
| 21:24.40 | [TK]D-Fender | [av]bani : I think MS paint was my last step.... |
| 21:27.42 | Dr-Linux | ~dic wot |
| 21:27.52 | *** part/#asterisk Ruis (n=ruise@68.178.8.80) |
| 21:27.52 | Dr-Linux | ~dict wot |
| 21:28.11 | *** join/#asterisk llagendijk (n=louis@lagendijk.xs4all.nl) |
| 21:28.31 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 21:29.12 | llagendijk | hello, anybode here have experience using a winbond w6692 with chan=misdn? |
| 21:29.14 | De_Mon | how do I use the attended transfer feature *2? |
| 21:29.26 | De_Mon | when I dial *2 on m sip phone it just plays dtmf tones |
| 21:29.49 | llagendijk | I have a problem where the card does not seem to be able to send anything on the ISDN bus (get timeout) |
| 21:33.25 | ManxPower | De_Mon, Well, do you have that feature enabled on the Dial line? |
| 21:33.31 | ManxPower | Did you look at features.conf? |
| 21:33.36 | ManxPower | Did you read README.variables |
| 21:34.21 | bigjb | FATAL ERROR : cannot find file /usr/share/festival/voices/english/en1_mbrola/en1/en1 < anyone seen this error from using festival text2wave? |
| 21:35.31 | Hmmhesays | man I love quintum gateways some days |
| 21:35.31 | ManxPower | bigjb, only years ago when I didn't follow the docs EXACTLY |
| 21:35.40 | bigjb | i did though :( |
| 21:35.49 | tasat | Looks like SIP express router is the answer to my NAT problems -- anyone here have experience running it on a linksys router? |
| 21:35.54 | acqua7 | hi, i have a question, to install mfc/r2 support with asterisk 1.2 , whats the version of unicall who works?? somebody? thanks |
| 21:36.12 | Hmmhesays | running SER on a linksys? |
| 21:36.14 | ManxPower | tasat, SER is not the solution to NAT problems |
| 21:36.22 | Hmmhesays | well it can be |
| 21:36.23 | [av]bani | tasat: i've never heard of a port of SER to linksys |
| 21:36.26 | Hmmhesays | nathelper is pretty good |
| 21:36.37 | [av]bani | tasat: asterisk, yes. ser, no. |
| 21:36.49 | tasat | ManxPower: ok, it was included in the voip-info link... looks like there is a port to OpenWRT for the Linksys WRT router... |
| 21:36.50 | ManxPower | Yes, it's A solution, but using a slegehammer to kill a fly is also A solution. |
| 21:36.53 | file | I don't know where people get this magical idea |
| 21:37.03 | file | to throw a full SIP proxy into the mix just for NAT |
| 21:37.03 | Hmmhesays | from the nat fairy |
| 21:37.08 | ManxPower | file, geeks have a rich fantasy life. |
| 21:37.17 | tasat | haha, ok... what's better, nathelper? |
| 21:37.26 | Hmmhesays | overkill caused by lack of sex |
| 21:37.30 | file | nat=yes canreinvite=no in Asterisk does plenty, and qualify=yes |
| 21:37.34 | ManxPower | tasat, What is better is to use the NAT features of Asterisk |
| 21:37.43 | ManxPower | That's what they were built for. |
| 21:37.48 | twisted[asteria] | someone say sex? |
| 21:37.54 | ManxPower | file, I think his asterisk server is behind NAT |
| 21:37.56 | Hmmhesays | no, you're seeing things |
| 21:38.00 | twisted[asteria] | oh ok |
| 21:38.07 | tasat | file knows first hand |
| 21:38.19 | *** join/#asterisk wwhome (n=andreas@woffi.planix.com) |
| 21:38.36 | tasat | MaxPower: humm... I'm going by the voip-info link that Tall-guy posted... |
| 21:38.54 | ManxPower | nat=yes, canreinvite=no, qualify=yes is SO MUCH HARDER THAN SETTING UP SER |
| 21:39.14 | ManxPower | So is externip=, localnet=, and setting up rtp.conf and the portforwaring! |
| 21:39.29 | twisted[asteria] | so how do you portforwar? |
| 21:39.32 | mog_work | why take it the easy way when you can make it hard |
| 21:39.45 | ManxPower | Why do it the simple way when you can spend days setting up SER? |
| 21:39.47 | *** join/#asterisk fjean (n=fjean@201009183198.user.veloxzone.com.br) |
| 21:40.01 | Hmmhesays | i dunno, but it sounds like fun |
| 21:40.11 | twisted[asteria] | well, besides the fact that ser can handle a bookou more |
| 21:40.17 | twisted[asteria] | +call volume |
| 21:40.40 | Hmmhesays | indeed it can |
| 21:40.43 | ManxPower | twisted[asteria], Asterisk newbies should not be doing that much volume. |
| 21:40.55 | twisted[asteria] | heh |
| 21:40.57 | fjean | hi all ! -- anyone knows why would the AGI->Dial exit the script completely when the result is other than answer ? |
| 21:40.58 | twisted[asteria] | no comment :P |
| 21:40.59 | Hmmhesays | i'm having trouble with asterisk+SER right now, freaking chan_spi |
| 21:41.15 | Hmmhesays | *chan_sip even |
| 21:41.19 | tasat | ManxPower: ok, that's why I asked... |
| 21:41.19 | twisted[asteria] | what's wrong with it? |
| 21:41.25 | twisted[asteria] | i've got ser + asterisk working fine together |
| 21:41.53 | Hmmhesays | i have my endpoints registered to SER, then I have one entry in sip.conf for SER |
| 21:41.59 | ManxPower | twisted[asteria], how long did it take and did you know asterisk well when you started with SER? |
| 21:42.10 | ManxPower | same queston for Hmmhesays |
| 21:42.18 | twisted[asteria] | ManxPower, i subcontracted most of it, and i knew asterisk quite well |
| 21:42.46 | Hmmhesays | i modified chan_sip to send notify mailbox@SER, but chan_sip always grabs the first mailbox |
| 21:42.53 | twisted[asteria] | oh haha |
| 21:42.54 | twisted[asteria] | use sipsak |
| 21:42.58 | Tall-guy | for the record: I like the nat=yes, canreinvite no, qualify=yes externip stuff better... |
| 21:42.58 | twisted[asteria] | and externnotify |
| 21:42.59 | Dr-Linux | Hmmhesays: hi friend how are you? :) |
| 21:43.11 | Hmmhesays | sipsak? |
| 21:43.30 | Hmmhesays | ManxPower: i went from never touching SER to an 75% working config in a week |
| 21:43.45 | ManxPower | Hmmhesays, it's a thingy that lets you build and send SIP messages from the command line. |
| 21:43.58 | ManxPower | so that would be about 40 hours? |
| 21:44.07 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 21:44.08 | Hmmhesays | umm well half days, so probably 20 |
| 21:44.18 | festr_ | is it possible to change codec in open iax channel? consider this order of establishing call: asterisk zap:A -iax-> B -iax-> C. c can do |
| 21:44.22 | festr_ | g711 only, A and B can do g729 or g711. A start with g729 so B have to do recoding, but i want to change to g711 on all asterisks... |
| 21:44.33 | ManxPower | how long did it take to make localnet and the other stuff for asterisk behind nat to work? |
| 21:44.35 | Hmmhesays | so how does sipsak help me with my notify messages? |
| 21:44.40 | ManxPower | festr_, no |
| 21:44.48 | twisted[asteria] | Hmmhesays, externnotify in vociemail.conf |
| 21:44.58 | festr_ | ManxPower: nor asterisk trunk? |
| 21:45.00 | twisted[asteria] | look at the documentation for that and for sipsak :) |
| 21:45.04 | ManxPower | Hmmhesays, use externotify to run sipsak to build a voicemail notify message and send it to the phone |
| 21:45.13 | twisted[asteria] | yeah, basically |
| 21:45.17 | Hmmhesays | yeah, that didn't even occur to me |
| 21:45.21 | ManxPower | festr_, better to ask on asterisk-dev |
| 21:45.54 | ManxPower | I used to to send instant messages to polycom phones when I played around with it. |
| 21:45.55 | Hmmhesays | how resource heavy is it? |
| 21:46.23 | ManxPower | no idea, since I have no use for it in my production enviroment. |
| 21:46.25 | twisted[asteria] | about as resource heavy as a netsend |
| 21:46.35 | Hmmhesays | damn it's going to drag my p133 down |
| 21:46.40 | twisted[asteria] | lol |
| 21:46.54 | Hmmhesays | i'll check it out though, thanks for the pointer |
| 21:46.58 | twisted[asteria] | it's not noticable on this production box |
| 21:47.10 | Hmmhesays | how often are you sending out notifies though? |
| 21:47.15 | twisted[asteria] | it just sits quietly in the background firing off notifications for voicemail |
| 21:47.19 | twisted[asteria] | quite often, actually |
| 21:47.24 | twisted[asteria] | a notify doesn't need any response |
| 21:47.35 | twisted[asteria] | so it runs, processes the template, sends the message, and exits |
| 21:47.39 | Hmmhesays | got it |
| 21:48.20 | Hmmhesays | well that makes my life about eleventy billion times easier |
| 21:48.53 | twisted[asteria] | yup |
| 21:50.54 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 21:51.33 | *** join/#asterisk mujjoo (n=murtazaj@h94s217a102n47.user.nortelnetworks.com) |
| 21:51.37 | nettie | hi guys, I was wondering if there's any way to run MP3Player() on a system without soundcard or accessible /dev/dsp please? Any idea? |
| 21:51.41 | mujjoo | hello all |
| 21:51.42 | Hmmhesays | did you build your own template? or is there one out there |
| 21:52.14 | mujjoo | i have a couple of questions about testing an application from the apps directory |
| 21:52.23 | mujjoo | can someone guide me |
| 21:54.06 | mujjoo | how do I know if the application I compiled is actually being accessed when i use it in the dialplan |
| 21:55.14 | rabelais | mujjoo: watch the console |
| 21:58.36 | mujjoo | i did...it seems like it is not doing anything |
| 21:58.49 | mujjoo | what level do verbosity/debug do i need to set |
| 21:58.56 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 21:59.17 | Tall-guy | "asterisk -vvvvvvvvvgc" is my friend" :) |
| 21:59.20 | mujjoo | also I tried compiling the application by itself and it gave me a lot of errors, but when I compile the whole asterisk tree it compiled fine |
| 21:59.31 | mujjoo | ok i will keep that in mind |
| 21:59.40 | mog_work | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvc is mine |
| 21:59.49 | Tall-guy | mog: I didn't want to admit to that many "v's" :) |
| 21:59.53 | mog_work | although everything after first 4 doesnt matter |
| 21:59.53 | r_evolution | hahaha mog |
| 21:59.56 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 22:00.02 | r_evolution | you're gonna monitor the fucking dust collecting in the fan |
| 22:00.03 | Grizzy | I want a .asteriskrc file |
| 22:00.07 | Tall-guy | hahaha |
| 22:00.18 | Egonis | I just upgraded to asterisk 1.2.5, and get 'app_conference' not found in /var/log/asterisk/messages -- how do I disable / fix it? |
| 22:00.35 | mog_work | im gonna add a verbose message at 100 or so and have it say "god damn lay off the vs you crack head " |
| 22:00.36 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 22:00.48 | mog_work | or "nothing to see here" |
| 22:00.51 | Tall-guy | mog: is that like "We found our IOCTL" :) |
| 22:00.51 | x86 | set verbose 99999999999999999999999999999 |
| 22:00.56 | x86 | thats what i always do ;) |
| 22:00.59 | acqua7 | hi |
| 22:01.00 | acqua7 | d you build your own template? or is there one out there |
| 22:01.01 | acqua7 | <mujjoo> i have a couple of questions about testing an application from the apps directory |
| 22:01.17 | acqua7 | <PROTECTED> |
| 22:02.14 | Egonis | Another error: Ouch ... error while writing audio data: : Broken pipe |
| 22:02.42 | mujjoo | so anyone have any idea how i can go about compiling debugging applications |
| 22:02.53 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 22:03.42 | fjean | why would AGI->Exec(DIAL... exit the AGI script when dial does not work ? |
| 22:03.42 | *** join/#asterisk Tamarisk (n=adrian@user-6936.lns6-c10.dsl.pol.co.uk) |
| 22:04.00 | Dr-Linux | what's good features in asterisk addons? |
| 22:05.15 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 22:05.42 | _Sam-- | [av]bani : how do you stop the 'bleedthrough' of the BLF lights on the gxp? sometimes 1 line is in use, and it looks like 2! |
| 22:05.53 | wrmem | fjean: Try protecting against SIGHUP in your script |
| 22:06.10 | fjean | wrmem: cool i ll try that |
| 22:09.05 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 22:09.32 | Grizzy | I'm running mpg123 0.59r, and enabling music on hold causes asterisk to freeze and this error: Yuck! Error in buffer handling...: Broken pipe |
| 22:12.05 | wrmem | Grizzy: One option is to convert your music into a format understood by Asterisk and avoid mpg123 in the first place. (It's better in the long run). There is a format_mp3 addon, but I just convert the few files I use to .gsm |
| 22:12.19 | fjean | wrmem: is $SIG{HUP} = 'ignore_hup'; enough ? |
| 22:12.57 | wunderkin | i'm using the pgsql app, and doing an insert.. i have: PGSQL(Query resultid ${db} INSERT INTO table VALUES (\'${blah}\'\,\'${CALLERID(num)}\') for example, the blah is evaulated properly but not callerid.. why is that? if i put \'"${CALLERID(num)"\' it is ok but in the query it shows "callid" which will not work with the quotes ending up in the query |
| 22:14.04 | wrmem | fjean: Manual says $SIG{"HUP"} = "IGNORE"; But I usually write a stupid subroutine to print out the message as a FYI |
| 22:14.09 | ManxPower | because you forgot the closing brace on it |
| 22:14.30 | [av]bani | _Sam--: same way you stop the bleedthrough of the BLF lights on polycom 601's |
| 22:14.33 | wunderkin | oops well that was me typing it out, thats not what i had |
| 22:14.56 | ManxPower | that's why you should PASTE stuff, not type it. It wasts everyone time |
| 22:15.09 | wunderkin | i know, sorry but its long |
| 22:15.23 | SplasPood | anyone know how to globally disable RFC3389 in an AS5300 ? |
| 22:16.09 | *** part/#asterisk dizzzan (i=dan@host.l8t.net) |
| 22:17.14 | wunderkin | INSERT INTO table VALUES (\'${blah}\'\,\'100000001\'\,\'"${CALLERID(num)}"\'\,\'100\')) there thats more verbatim, and what gets executed in the query is: VALUES ('511','100000001','"2"','100') |
| 22:18.06 | wunderkin | and ${CALLERID(num)} = 2 |
| 22:18.40 | wunderkin | im just trying to figure out how to escape it properly, somehow from a function is different than a variable |
| 22:19.59 | _Thor | Hello Everyone!! |
| 22:21.14 | _Thor | Question: if I restart mysql, do I have to restart *, or just reload it?? |
| 22:22.16 | fjean | thor : just mysql, that I know |
| 22:22.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:22.31 | _Thor | thank you, it is production machine |
| 22:23.33 | Tall-guy | _thor: aw hell, just wing it :) |
| 22:28.14 | *** join/#asterisk clive- (n=pirch@dsl-145-21-10.telkomadsl.co.za) |
| 22:29.39 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 22:31.06 | _Sam-- | [av]bani : that is by taking apart the phone and doing something about it? |
| 22:31.19 | [av]bani | _Sam--: by ignoring it |
| 22:31.30 | [av]bani | or by yelling at the vendor and having them ignore you |
| 22:31.35 | _Sam-- | lol |
| 22:31.41 | _Sam-- | i think i could take it apart and do something |
| 22:31.46 | *** join/#asterisk dgilmore (n=dennis@anubis.ausil.us) |
| 22:31.51 | [av]bani | cisco and snom are the only phones which dont have bleedthrough |
| 22:32.01 | [av]bani | the ip601 has it, which is suprising for such an expensive phone |
| 22:32.07 | clive- | hi, anyone using centos 4.2 here ? |
| 22:32.13 | _Sam-- | im back to messing with BLF, was tired of stuff not breaking |
| 22:32.17 | [av]bani | the led is pretty bright though, its like laz0r b3ams |
| 22:32.30 | [av]bani | _Sam--: mess with qualify :) |
| 22:32.46 | Mavvie | oh man, people are stupid. |
| 22:32.57 | _Sam-- | maybe tomorrow, i like to have only one thing break per day, otherwise it confuses the sales guys :) |
| 22:33.09 | [av]bani | so your gxp's been good so far? |
| 22:33.09 | dgilmore | hey all quick question , Im building a asterisk box for a outbound call center with a pri, couple of pots lines and 24 handsets what would be the minimum cpu speed. It will be a pretty basic setup account codes |
| 22:33.20 | _Sam-- | yeah right now i have 5 using blf for a few hours no problem |
| 22:33.25 | [av]bani | oh, one thing. gxp2000 seems to have almost 0 jitter buffers |
| 22:33.25 | Mavvie | with all these voice-responses saying "this number is unreachable" and "this number is busy", they don't know anymore what poe-die-piep and tut-tut-tut-tut mneans. |
| 22:33.37 | SplasPood | argh.. there's gotta be a way in this AS5300 to disable vad globally rather than per DID or range of DIDs |
| 22:33.48 | _Sam-- | our phones sound really really good anymore |
| 22:34.07 | [av]bani | _Sam--: unless your ethernet is perfect, you'll get some stutter with gxp2000 |
| 22:34.09 | _Sam-- | then again, the only point of comparison is our old key-type system |
| 22:34.21 | _Sam-- | all the gxps are on their own 100base net |
| 22:34.32 | [av]bani | we have an extension on the remote side of a 2mi 802.11g link... with gxp it was stutter city |
| 22:34.44 | [av]bani | we replaced it with a polycom 601... 0 stutter |
| 22:35.02 | _Sam-- | did you try switching to SIP with teliax? |
| 22:35.10 | [av]bani | i've always been sip with teliax |
| 22:35.19 | _Sam-- | thats weird you still have that problem |
| 22:35.23 | _Sam-- | our teliax problems are gone |
| 22:35.26 | Grizzy | wrmem - thanks. |
| 22:35.49 | _Sam-- | our toll free origination is still handled by teliax |
| 22:35.49 | *** join/#asterisk eivindtr (n=wingnut-@193.212.20.110) |
| 22:36.03 | [av]bani | tis still odd origination works fine on teliax, but not termination |
| 22:36.08 | [av]bani | should be same path |
| 22:36.16 | [av]bani | ...shrug |
| 22:36.21 | _Sam-- | i dont terminate anything there anymore, at least not much. |
| 22:37.47 | _Sam-- | is there any security issue to leaving port 5060 wide open to the internet? |
| 22:38.08 | Hmmhesays | we're going down down in an earlier round, sugar we're going down swinging |
| 22:38.16 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 22:38.51 | lzhang | I'm trying to switch to g.729, but when I do get a get SIP client error 499... what am I doing wrong? |
| 22:38.56 | lzhang | does it need to be installed? |
| 22:39.23 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:40.18 | Tamarisk | I believe you need to purchase a license to use G.729 with asterisk |
| 22:40.39 | lzhang | what about just for personal educational use? |
| 22:41.02 | Tamarisk | Not sure , two secs let me find the comment in the book |
| 22:41.22 | lzhang | I just need to try it out |
| 22:42.56 | lzhang | ok I think I need to install it maybe? |
| 22:42.59 | *** part/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 22:43.02 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 22:45.31 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:46.26 | Tamarisk | I found a section that says you need to pay a licence fee but not who too. I would guess www.digium.com would be a good place to start |
| 22:46.32 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 22:46.42 | Egonis | How do I change an iax2 channel to be monitored? what are the advantages? |
| 22:49.08 | Tamarisk | Found it |
| 22:49.19 | Tamarisk | do you have the book? |
| 22:49.32 | Tamarisk | ~book |
| 22:49.35 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:50.26 | *** part/#asterisk clive- (n=pirch@dsl-145-21-10.telkomadsl.co.za) |
| 22:52.09 | Tamarisk | lzhang: It says the g729 directory contains the code and rgistary programme to use the codec youmust purchase the license and then run the registration programme |
| 22:53.01 | Tamarisk | Can be purchased online from Digium |
| 22:55.15 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:55.52 | lzhang | thank tam |
| 22:56.27 | Tamarisk | No probs thats non technical info sometimes I can do that. |
| 22:56.54 | jorgito | does anybody know voipgate.com ?? |
| 22:57.03 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 22:57.06 | jorgito | or have some experience with them ? |
| 22:59.57 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 23:02.58 | azzie | anybody has documentation for Vega100 ? |
| 23:08.32 | Zodiacal | anyone know why i get beeping, if i try to make a call when one line is in use allready? i have 6 fxo modules with 6 lines... |
| 23:08.43 | Zodiacal | incoming calls work fine |
| 23:08.53 | Zodiacal | with muliple lines.. just can't make muliple outgoing calls |
| 23:10.23 | *** join/#asterisk TiKiTaKi_ (n=Heaven@acxb130.neoplus.adsl.tpnet.pl) |
| 23:10.25 | TiKiTaKi_ | hello |
| 23:10.37 | TiKiTaKi_ | what is a good free softphone |
| 23:10.41 | TiKiTaKi_ | for asterisk? |
| 23:10.59 | *** part/#asterisk mujjoo (n=murtazaj@h94s217a102n47.user.nortelnetworks.com) |
| 23:11.11 | cpm | good free softphone are exclusive terms |
| 23:11.15 | cthompson | yeah |
| 23:11.19 | cthompson | I use X-lite |
| 23:11.27 | cthompson | but I can't make it do DTMF right |
| 23:11.28 | TiKiTaKi_ | ok , so a nice softphone |
| 23:11.39 | cthompson | X-Lite is decent |
| 23:11.43 | TiKiTaKi_ | cthompson what homepage of X-lite? |
| 23:11.50 | cthompson | www.counterpath.com I think |
| 23:11.54 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 23:12.03 | cthompson | X-Ten is the pay version, X-Lite is the free one |
| 23:12.13 | TiKiTaKi_ | there is buy it now |
| 23:12.17 | TiKiTaKi_ | ok |
| 23:12.35 | TiKiTaKi_ | i got 4 isdn cards |
| 23:12.48 | TiKiTaKi_ | as i read on forum asterisk does not support winbond w6692cf |
| 23:13.01 | TiKiTaKi_ | with " Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" there are some problems with interrupts |
| 23:13.10 | justinu | actually, eyebeam supercedes x-ten now |
| 23:13.14 | cpm | I use kiax, because it is native iax, and it basically sucks, but it's a good try. |
| 23:13.25 | TiKiTaKi_ | i have also "Fritz!Card PCI" and "AVM ISDN Felix ME2 " , any experience with those ones? |
| 23:15.04 | TiKiTaKi_ | i think no |
| 23:15.12 | TiKiTaKi_ | is anyone using isdn bri card? |
| 23:17.26 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 23:17.43 | Egonis | How do transfers work in IAX? i.e. Phone to Phone? Do I need to do anything? |
| 23:20.42 | wundaboy | can anyone recomend a voip provider? |
| 23:20.58 | azzie | callcentric! |
| 23:21.35 | wundaboy | ok |
| 23:21.38 | wundaboy | ill check em out |
| 23:22.41 | wundaboy | i dunno |
| 23:22.48 | wundaboy | they want $6/month/did |
| 23:24.31 | Egonis | In iax.conf I have mailbox=100@default, but it shows no indication.. is my syntax correct? |
| 23:25.01 | Zodiacal | heres a hard question for ya: if i page someone over the loudspeaker (soundcard) and tell them they have a phone call, how would they pick it up from another station? on our old phone system we would tell them they had a call on line n. |
| 23:28.14 | azzie | *8 in A@H |
| 23:28.16 | m29poff | is there anyone here that uses a TDM4OOP card ? |
| 23:28.34 | Nugget | I do. |
| 23:29.04 | m29poff | I've got a problem with hangup detection with FXO port |
| 23:32.35 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 23:32.48 | [av]bani | Zodiacal: park the call, then tell them what ext it's parked at |
| 23:34.04 | Zodiacal | av bani, thank you! ill go readup on that now |
| 23:34.54 | fjean | question, is there a way to retreive the SIP error number (e.g 404) from an AGI script ? |
| 23:35.27 | fjean | or any other way... |
| 23:37.03 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 23:45.58 | *** join/#asterisk Tamarisk (n=adrian@user-6936.lns6-c10.dsl.pol.co.uk) |
| 23:45.58 | CrashHD | is there a way to throw a prefix on an outbound sip call |
| 23:47.03 | *** join/#asterisk pixolex (n=chatzill@87-196-153-120.net.novis.pt) |
| 23:47.39 | Tamarisk | Still could do with ideas why X-lite softphone and Grandstream SIP ATA will not talk to each other, but both can do echotest |
| 23:47.49 | ambriento | hmm like prefix() does? |
| 23:48.01 | Tamarisk | These are the only 'phones' on my home system |
| 23:49.01 | Tamarisk | They will not pass rtp packets between each other or on hold music? |
| 23:49.02 | justinu | codec incompatibility? |
| 23:49.14 | *** join/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk) |
| 23:49.19 | Tamarisk | Hi justinu |
| 23:49.50 | Tamarisk | I have tried to make sure that they have pcma or alaw in common |
| 23:50.00 | robin_sz | at least I think its a lemon |
| 23:50.34 | robin_sz | ahh, no ... its a GXP2000 |
| 23:51.10 | justinu | leland & tamarisk: you guys are both having some weird problems |
| 23:51.29 | Leland | justinu: yea.. and nobody else seems to have the same problem |
| 23:51.35 | justinu | leland: i take it converting moh to ulaw didn't help? |
| 23:51.37 | Tamarisk | And I bet it is a simple answer when found |
| 23:51.52 | Leland | justinu: nope... was even worse actually |
| 23:51.59 | fjean | crashd - something like 9${EXTEN} would do it |
| 23:51.59 | justinu | strange... |
| 23:52.25 | robin_sz | I had a MOH issue last week with a client |
| 23:52.45 | CrashHD | fjean: I'm hoping I can add it in the sip.conf and not take it into account in my dialplan |
| 23:53.03 | robin_sz | I swapped their "boring and sunny" music for Crazy Train by Ozzie Osbourne by accident ;) |
| 23:53.05 | Leland | I even took a PCM coded file, manually transcoded it into native G.729 and tried to force * to treat it as passthrough... and it STILL had the same problem |
| 23:53.15 | *** join/#asterisk devnull431 (n=slick_sh@D-128-208-39-41.dhcp4.washington.edu) |
| 23:53.45 | justinu | whoa |
| 23:54.06 | justinu | what end point is receiving the g729 stream, leland? |
| 23:54.20 | Leland | any endpoint |
| 23:54.26 | justinu | what have you tested with? |
| 23:55.24 | Leland | tested with jphone, c7960, Zyxel, and also tested across all three of the g.729 trunks to my ITSP calling them from off-net (i.e. PSTN) |
| 23:55.48 | Leland | if I change the ITSP trunks to any other codec, the music plays fine |
| 23:55.56 | justinu | how about if you encode a prompt, or something other than music? |
| 23:56.05 | justinu | g729 isn't kind to music |
| 23:56.10 | Leland | oh.. that's the other weird thing.. all of the prompts work perfectly |
| 23:56.17 | justinu | it garbles it pretty good |
| 23:56.38 | Leland | even encoded the moh into the same format as the prompts using GSM.. and still the same results |
| 23:57.20 | Leland | yea well.. not looking for CD quality music here.. besides g.726 is even less tolerant to music, but at least with that the music is still recognisable |
| 23:57.27 | justinu | true |
