| 00:00.03 | *** part/#asterisk andio (n=andio@port-195-158-165-243.dynamic.qsc.de) |
| 00:00.05 | SibRphrek | so like i said - i did all that - now i'm bored |
| 00:00.09 | SibRphrek | i dunno what else to do |
| 00:00.17 | SibRphrek | maybe i can make set it up to call my cell if no asnwer |
| 00:00.19 | pifiu | sib but is yours dynamicaly created? |
| 00:00.23 | pifiu | in real time autromatically? |
| 00:00.42 | SibRphrek | pifiu: when you open the FM database, it pulls the data from the sql upon load |
| 00:00.49 | Qwell[] | Zodiacal: Tried anything besides gs? |
| 00:00.50 | pifiu | oh |
| 00:01.09 | pifiu | interesting got a screenshot? |
| 00:01.19 | SibRphrek | gimme a sec |
| 00:01.21 | pifiu | ok |
| 00:01.30 | pifiu | PM the link |
| 00:01.35 | pifiu | brb in 5 minutes |
| 00:01.56 | SibRphrek | k |
| 00:02.10 | Qwell[] | Zodiacal: And did you reload the module after making the change? I believe you need to |
| 00:03.03 | Zodiacal | i did modprobe zaptel, modprobe wcfxs then ztcfg |
| 00:03.10 | Zodiacal | and i tried rebooting |
| 00:03.26 | Zodiacal | qwell is that what you mean by reload? |
| 00:03.29 | Qwell[] | try ls... |
| 00:03.36 | Qwell[] | well, a reboot would reload the modules, so yes |
| 00:04.28 | Zodiacal | kewlstart and loopstart work.. just not groundstart.. |
| 00:04.30 | Zodiacal | any ideas? |
| 00:04.36 | Qwell[] | don't use gs? :p |
| 00:04.40 | Zodiacal | :P |
| 00:04.59 | Zodiacal | im trying to speed up my outgoing calls still.. someone suggested i try that.. |
| 00:05.02 | *** join/#asterisk asskicker (i=asskicke@207.34.197.90) |
| 00:05.29 | Zodiacal | heres my orginal problem: http://forums.digium.com/viewtopic.php?p=13097 |
| 00:05.30 | pifiu | ok im back |
| 00:05.43 | SibRphrek | pifiu: it will take me a few minutes, my odbc driver got fucked up |
| 00:07.22 | *** join/#asterisk family (n=family@c-67-163-169-5.hsd1.ct.comcast.net) |
| 00:08.08 | family | anyone here use astcc? i used to sample dia plan., and the numbers either tell me the length the callign card msut be and hangs up, or just says goodbye ..... i tested the dabase conenction and its fine...so waht gives? |
| 00:10.49 | *** part/#asterisk tamp4x (n=family@c-67-163-169-5.hsd1.ct.comcast.net) |
| 00:10.52 | *** join/#asterisk tamp4x (n=family@c-67-163-169-5.hsd1.ct.comcast.net) |
| 00:11.02 | tamp4x | anyone here use astcc? i used to sample dia plan., and the numbers either tell me the length the callign card msut be and hangs up, or just says goodbye ..... i tested the database conenction and its fine...so waht gives? |
| 00:11.09 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 00:11.12 | pifiu | ok well just PM so i dont overlook it here |
| 00:11.43 | SibRphrek | ok |
| 00:11.51 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 00:11.57 | SibRphrek | i'm just editing the pic now (taking out important #s) |
| 00:12.51 | st3v | Hi, I want to make asterisk do a 3-way call on an analog line on a zap channel. I made a macro in the dialplan to read both numbers from the user, then first dial the first number. I want to be able to let the user press the number 5 when they are ready to start dialing the second number. How can I do that> |
| 00:13.43 | *** join/#asterisk FastJack (i=fastjack@p5091DD66.dip.t-dialin.net) |
| 00:13.58 | st3v | so when they're already on a call with the first number, to dial the 2nd number |
| 00:14.40 | Ariel_ | st3v, you talking about mega flash hell |
| 00:15.04 | SibRphrek | pifiu: you got PM'd! |
| 00:15.13 | st3v | I can do Flash(), Dial(..), Flash(), correct? |
| 00:15.33 | st3v | I just want to know how to make it execute those lines while in the call |
| 00:16.14 | asskicker | for IMMEDIATE sale 2 high performance laptops price 500 each includes shipping carry case and wireless router. message me if interested on aim at ogd443, msn at mcsltd@telusmail.net, or yahoo at mcsltd2. specs p4m 2.26, 1gb ram, 80gb hard drive sata, built in modem, ethernet, firewire, usb2.0, 802.11b/g wireless, 256mb nvidia geforce go 6800 pcie, 17.1" uwxga display. |
| 00:16.38 | *** join/#asterisk asskicker (i=asskicke@207.34.197.90) |
| 00:16.47 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 00:16.56 | De_Mon | wow what a great deal! I,think,I,will,contact,him |
| 00:17.00 | rob0 | haha |
| 00:17.09 | *** join/#asterisk kn0x (i=atlantic@71.194.235.251) |
| 00:17.48 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-244.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 00:17.49 | De_Mon | where can I find debs for 1.2.4 :) |
| 00:17.54 | file[laptop] | asskicker: go away |
| 00:17.56 | Qwell[] | De_Mon: In the future |
| 00:18.15 | Qwell[] | file[laptop]: old ep...next week though! |
| 00:18.34 | file[laptop] | I'm optimistic |
| 00:18.37 | Ariel_ | now we have laptops that have fallen off a trunk.... |
| 00:18.39 | Ariel_ | hummm |
| 00:18.39 | De_Mon | they are too slow and I don't want to build my own. Sucky |
| 00:18.48 | sevard | hey Ariel_ |
| 00:18.51 | Qwell[] | De_Mon: Just compile from source |
| 00:19.00 | Ariel_ | sevard, hello |
| 00:19.12 | Ariel_ | saw your post on the user list. |
| 00:20.03 | sevard | I didn't |
| 00:20.03 | SibRphrek | pifiu: you see the image? |
| 00:20.03 | Ariel_ | really now.. looks just like your problem with slackware and all |
| 00:20.03 | [TK]D-Fender | We need +e again :/ |
| 00:20.23 | SibRphrek | +e? |
| 00:20.23 | sevard | Ariel_: a forum somewhere? I hadn't posted anything anywhere yet. Can I have the URL? |
| 00:20.31 | Qwell[] | [TK]D-Fender: restrict joins to registered users? |
| 00:20.41 | [TK]D-Fender | SibRphrek : Channel mode that blocks out unregistered people..... |
| 00:20.42 | Qwell[] | /msg nickserv info asskicker |
| 00:20.47 | SibRphrek | ah |
| 00:20.49 | Qwell[] | [TK]D-Fender: ^ |
| 00:21.07 | Qwell[] | Doesn't do much good, now does it? :p |
| 00:21.07 | sevard | heh |
| 00:21.12 | Ariel_ | It's in the amportal user list. |
| 00:21.22 | sevard | Checking |
| 00:21.29 | [TK]D-Fender | Qwell : Shup yuo! |
| 00:21.41 | [TK]D-Fender | I like my delusions...... |
| 00:21.48 | Qwell[] | indeed |
| 00:22.31 | rene- | indeed |
| 00:22.50 | De_Mon | INdeed |
| 00:23.11 | Ariel_ | strange the top of my seciton for here says +tncf #asterisk-unregistered |
| 00:23.20 | Qwell[] | Ariel_: join floods |
| 00:23.22 | Ariel_ | secition/section |
| 00:23.35 | SibRphrek | to setup a 911 call do i really need to configure zaptel? |
| 00:23.46 | sevard | I forgot how to start asterisk with a CLI but still be able to detach :| |
| 00:24.01 | Qwell[] | sevard: Just start it, then connect with -r |
| 00:24.04 | sevard | oh DUHUHGUEHGIEHGE |
| 00:24.07 | sevard | god hit me |
| 00:24.18 | sevard | jesus, that means i need coffee, brb |
| 00:24.25 | Ariel_ | asterisk -vvvvgc |
| 00:24.36 | Ariel_ | or safe_asterisk |
| 00:24.40 | arcy | is there a way to Dial() more than one extensions? (queues seem to require that i Answer() first ) |
| 00:24.43 | Ariel_ | then asterisk -rvvvv |
| 00:25.00 | Qwell[] | arcy: Dial(SIP/1&SIP/2) |
| 00:25.05 | Ariel_ | & |
| 00:25.20 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 00:25.21 | arcy | hey, thank you. |
| 00:25.27 | sevard | safe_asterisk kicks out to a tty, and when i disable that, restart safe_asterisk, and asterick -r it says it can't connect to remote asterisk |
| 00:25.36 | Ariel_ | SibRphrek, well unless you have a E911 provider you have to setup the zap to pstn or a sipura 3000 fxo sip gateway. |
| 00:25.38 | arcy | lightining fast responses ... :) |
| 00:25.57 | Ariel_ | sevard, how about asterisk -vvvvgc |
| 00:26.09 | Qwell[] | Ariel_: That doesn't let you detach though |
| 00:26.10 | sevard | after safe_asterisk? one second. |
| 00:26.14 | Ariel_ | that is not detached |
| 00:26.19 | rene- | Ariel: it soesnt |
| 00:26.20 | SibRphrek | i havn't had to setup zapdata.conf yet |
| 00:26.21 | rene- | doesnt |
| 00:26.25 | rene- | ive tried it |
| 00:26.29 | rene- | just a sec ago |
| 00:26.44 | rene- | it does gives you a cli tho |
| 00:26.51 | Ariel_ | asterisk -vvvvgc will let you know if all runs correctly but if you exit it shuts down. |
| 00:27.00 | Ariel_ | another way is also service asterisk start |
| 00:28.47 | s34n | where is a nice example of time-based routing? |
| 00:29.09 | Qwell[] | s34n: The wiki |
| 00:29.10 | Qwell[] | ~wikis |
| 00:29.12 | jbot | i guess wikis is http://www.voip-info.org |
| 00:29.26 | Ariel_ | s34n, have you check the /usr/src/asterisk/config/extensions.conf.sample |
| 00:32.47 | s34n | Ariel_: thx. time-based includes is interesting |
| 00:32.59 | SibRphrek | if i wanna make it so when a call comes in, and the person is busy it goes to another external phone number, what should i be looking for? |
| 00:33.16 | Ariel_ | macro |
| 00:34.12 | Ariel_ | SibRphrek, you can create a macro to try other devices or even to to voicemail. look at the macro's which are in the extensions.conf.sample |
| 00:35.01 | sevard | Ariel_: when I do /usr/sbin/asterisk -U asterisk -G asterisk |
| 00:35.09 | sevard | it starts, and runs as user asterisk |
| 00:35.19 | sevard | but when user asterisk does /usr/sbin/asterisk -rvvv i get |
| 00:35.26 | sevard | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 00:35.35 | Qwell[] | sevard: Does that file exist? |
| 00:35.52 | sch19 | maybe check permissions, if asterisk can't make the file |
| 00:35.58 | Qwell[] | ie; was user asterisk able to make it? |
| 00:35.58 | sevard | It does, but, I think I found the problem. |
| 00:37.06 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 00:37.23 | sevard | 0 srwxr-xr-x 1 asterisk asterisk 0 2006-02-01 18:31 /var/run/asterisk.ctl |
| 00:38.11 | sevard | asterisk@host:/$ /usr/sbin/asterisk -r |
| 00:38.13 | sevard | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 00:38.58 | sevard | why would that be? |
| 00:39.03 | Qwell[] | is it actually running? |
| 00:39.05 | sevard | yes |
| 00:39.17 | sevard | many processes |
| 00:39.29 | Qwell[] | run it with -c, find out if it's stopping or anything |
| 00:39.39 | Sedorox | 1.2.4...... damn I'm behind! |
| 00:39.54 | Ariel_ | Sedorox, yes actually in just a day |
| 00:39.55 | BlueDevi1 | sevard: sudo asterisk -r :) |
| 00:40.17 | Sedorox | lol |
| 00:40.23 | Qwell[] | user asterisk defnitely shouldn't be able to sudo :p |
| 00:40.25 | Sedorox | well I knew 1.2.3 was released... |
| 00:40.29 | Qwell[] | if it can...what's the point? |
| 00:40.30 | sevard | Asterisk Ready. |
| 00:40.30 | sevard | *CLI> |
| 00:40.33 | Sedorox | but I'm not even out of 1.2.0... |
| 00:40.36 | Sedorox | er... 1.2.1 |
| 00:40.53 | BlueDevi1 | Qwell[]: lalala :) |
| 00:42.54 | sevard | heh. |
| 00:43.00 | st3v | is there a way to be able to 3way call in asterisk? |
| 00:43.03 | *** join/#asterisk ke4qqq_ (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 00:45.46 | Ariel_ | st3v, lots of ways meetme, phones that do it. But zap one line is a flash problem... but it can be done. But It's not that reliable. |
| 00:46.28 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 00:47.46 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 00:48.49 | st3v | is there a way to get an outside line connected by pressing an extension? |
| 00:48.55 | *** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 00:49.10 | st3v | so I can do the 3way call myself |
| 00:49.38 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 00:50.23 | Ariel_ | st3v, yes but problem is when you use flash or in a sip phnoe asterisk will do something with it. and not send it out the phone line. Look at the wiki for flash |
| 00:50.26 | Ariel_ | ~docs |
| 00:50.28 | jbot | i heard 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 |
| 00:51.34 | rene- | ~seen jerjer |
| 00:51.37 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 14d 3h 20m 43s ago, saying: 'thanks again'. |
| 00:51.41 | *** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com) |
| 00:51.53 | Ariel_ | wow he as been away for 14 days... |
| 00:51.55 | *** join/#asterisk ke4qqq_ (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 00:52.32 | justinu | are we having fun yet? |
| 00:52.52 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 00:57.08 | litage | Qwell[]: i've been looking through Ranch Networks' products, but am not quite sure what they're used for. are they just glorified switches "designed for use with Asterisk"? |
| 00:58.09 | Qwell[] | litage: firewalls and such |
| 00:58.12 | *** join/#asterisk p0g0_ (n=p0g0@69.176.63.217) |
| 00:58.33 | litage | ah, just found a page on their site explaining MidCom |
| 00:59.05 | Dr-Linux | justinu: you know what |
| 00:59.07 | De_Mon | my queue is going to an agents voicemail after it times out, but I don't see that anywhere in the dialplan, what gives? |
| 00:59.12 | *** join/#asterisk p0g0__ (n=pogo@69.176.63.217) |
| 00:59.56 | litage | De_Mon: shouldn't you be examining the queue rather than the dialplan? |
| 01:00.09 | Dr-Linux | i have congiured cisco phone remotely, everything is fine. but there is a very strange problem |
| 01:00.17 | SibRphrek | i can't make zaptel cuz i'm on an OS X box |
| 01:00.22 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 01:00.23 | SibRphrek | what should i use instead of zaptel |
| 01:00.25 | SibRphrek | or ztdummy |
| 01:00.26 | znoG | what exactly does faxdetect=outgoing do in zapata.conf? |
| 01:00.31 | Dr-Linux | the phone doesn't send request to the asterisk server |
| 01:00.35 | De_Mon | litage no.. |
| 01:00.43 | justinu | Dr-Linux: no, i don't |
| 01:00.52 | znoG | ie. if I dial out with a modem to the Internet via a FXS port on my TDM card, do I need faxdetect=outgoing? |
| 01:00.54 | Dr-Linux | even phone and the asterisk server are on same LAN |
| 01:01.00 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 01:01.25 | Dr-Linux | justinu: i'm trying to reslove this problem since 7 hours .. but ... still no luck :S |
| 01:02.29 | Dr-Linux | justinu: i know there will a small thing missing, but we can't figure out :S |
| 01:03.05 | Dr-Linux | justinu: on the same lan other 8 cisco phones are working fine .. and we match all the settings are same .. |
| 01:03.39 | tronix | does anyone maintain any publicly distributed list of numbers blacklisted due to telemarketers or fax spam? |
| 01:04.00 | Dr-Linux | but phone doesn't send request to the asterisk :S even i tried to register the phone with another server, but same problem :S |
| 01:04.37 | tronix | Dr-Linux: firmware version same on all 9 phones? |
| 01:04.44 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 01:04.58 | Dr-Linux | tronix: yes each and everything is the same |
| 01:05.15 | Dr-Linux | sip firmware 7.4 |
| 01:05.22 | tronix | Dr-Linux: if you enable sip debug on asterisk console, |
| 01:05.28 | tronix | you don't see the phone send a request at all? |
| 01:06.05 | Dr-Linux | tronix: i can do that, but once i saw the request from the phone .. but its very strange |
| 01:06.09 | Dr-Linux | i checked everything |
| 01:06.16 | dumb---me | Do i need to install the JVM to execute Java code with asterisk |
| 01:06.22 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 01:06.23 | Dr-Linux | even i changed SIP port |
| 01:06.33 | tronix | if no registration or call setup request come into asterisk, then it's a phone problem with firmware. |
| 01:06.50 | *** join/#asterisk chop1 (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 01:07.03 | Dr-Linux | i checked it with differnt servers .. |
| 01:07.33 | tronix | when you enable sip debug on asterisk console, you see no registration requests from phone, no call setup requests, nothing. |
| 01:07.44 | tronix | means phone is not even trying or traffic is getting lost somewhere |
| 01:07.48 | tronix | not an asterisk problem. |
| 01:08.12 | tronix | means you will need to debug network or phone |
| 01:08.25 | Dr-Linux | tronix: what do you mean? first i installed sip firware 6.3 and then i upgrade it to 7.4 and same i had for all other phones |
| 01:08.46 | Dr-Linux | tronix: all the things are on the same LAN |
| 01:08.52 | s34n | My voicemail password is rejected from a sip phone, accepted from PSTN |
| 01:08.53 | Dr-Linux | and there is no firewall |
| 01:08.57 | tronix | on asterisk console, do this command: 'sip debug'. then pick up the phone. do you see any registration attempts? |
| 01:09.59 | Dr-Linux | tronix: i'm in Pakistan and the phone is in US, i nver seen these phone in real |
| 01:10.02 | tronix | if yes, you see registration attempts from phone, look to see if it succeeded or failed. |
| 01:10.03 | Dr-Linux | but all i did well |
| 01:10.03 | s34n | the sip phone dials the vm extension just fine, but * rejects the password I key in |
| 01:10.29 | De_Mon | the the agents extension is using a voicemail macro |
| 01:10.44 | Dr-Linux | tronix: only thing i didn't check :S |
| 01:10.48 | {zombie} | s34n: I suspect you don't have your DTMF settings right |
| 01:11.02 | Dr-Linux | what if i assign public IP to the phone |
| 01:11.08 | {zombie} | make sure both asterisk and the phone are using rfc2833 |
| 01:11.15 | s34n | {zombie}: but the tones work to dial the extension... |
| 01:11.22 | tronix | Dr-Linux: here's another idea... turn on sip debug, then look for any messages relating to that phone. see if success or failure |
| 01:11.23 | {zombie} | you don't dial the extension with tones |
| 01:11.25 | {zombie} | not with SIP |
| 01:11.27 | De_Mon | hrm... |
| 01:11.31 | tronix | Dr-Linux: phone will attempt to re-register periodically. |
| 01:11.45 | s34n | {zombie}: huh? |
| 01:11.48 | sch19 | does anyone in here have some experience with aastra phones? I purchased two 9133i models last week and while I'm having some trouble with many things, mostly BLF is where my issues are. |
| 01:12.00 | {zombie} | the phone sends a SIP packet to asterisk containing the destination address in the headers |
| 01:12.01 | {zombie} | as text |
| 01:12.06 | {zombie} | not as dtmf tones |
| 01:12.09 | tronix | Dr-Linux: save messages to a file then grep for phone's IP and look at each message. |
| 01:12.16 | {zombie} | you may HEAR dtmf when you dial, but that's not what is sent to asterisk |
| 01:12.17 | SibRphrek | if i turn on features in the features.conf and reload asterisk, how come they aren't activated |
| 01:12.38 | Dr-Linux | tronix: but i doesn't show anything |
| 01:13.00 | tronix | Dr-Linux: wait one hour or more. it might take a while. |
| 01:13.11 | sch19 | I've got hint setup and working in asterisk, and this is probably off topic in here, just thought someone might have some experience, that's all. |
| 01:13.25 | s34n | {zombie}: I couldn't dial extensions when the dmtf was wrong before |
| 01:13.40 | tronix | Dr-Linux: you will need to debug this problem, and that means you need to gather detailed information about what phone is doing. this will help. |
| 01:13.46 | tronix | Dr-Linux: or easy way out: replace phone. |
| 01:14.19 | {zombie} | s34n: trust me on this, SIP phones do NOT dial with dtmf |
| 01:14.21 | Dr-Linux | awwwwwwwww |
| 01:14.23 | {zombie} | it's a digital connection |
| 01:14.29 | Dr-Linux | tronix: i got something |
| 01:14.56 | justinu | sip phones do what's called "en block" dialing |
| 01:15.03 | Dr-Linux | tronix: i can see now sip debug log |
| 01:15.12 | {zombie} | if you are using an ordinary phone connected to an ATA (like a sipura) then you use dtmf to talk to the ATA and the ATA sends it as a sip packet to asterisk |
| 01:15.17 | {zombie} | but that's different |
| 01:16.13 | sch19 | well, was worth a shot i guess :( |
| 01:17.00 | s34n | {zombie}: I trust you, but I don't know how to fix the dtmf. I've tried all three modes |
| 01:18.05 | {zombie} | what options does the phone give? |
| 01:20.00 | s34n | it's a grandstream gxp-2000 |
| 01:20.07 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:20.27 | sch19 | popular model, do you like it? I'm thinking since nobody is using these aastra things, I should swap 'em out |
| 01:20.31 | Dr-Linux | http://pastebin.com/534741 << anybody tell me whats the problem? :S |
| 01:20.37 | Dr-Linux | oo justinu is here |
| 01:20.38 | justinu | people use aastra |
| 01:21.14 | s34n | {zombie}: errrrg. I got it |
| 01:21.22 | {zombie} | s34n: so set the phone to "via RTP (RFC2833)" and asterisk to dtmfmode=rfc2833 |
| 01:21.49 | [TK]D-Fender | {zombie} : What version of *? |
| 01:21.49 | sch19 | the docs are poor, I can't seem to make BLF work properly, i can get either a blf indicator light, or a functional button, but not both. |
| 01:22.16 | {zombie} | d-fender: I dunno, ask s34n :) |
| 01:22.20 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 01:22.26 | s34n | sch19: I have 1 gxp-2000, and one polycom 501 |
| 01:22.30 | [TK]D-Fender | uhhh yeah s34n <- |
| 01:22.49 | s34n | [TK]D-Fender: 1.2.3 |
| 01:23.11 | sch19 | if the aastra is sending the number i have programmed for blf to asterisk with a special prefix (as in the grandstream) i'm not seeing it logged on the console, how would I be able to tell it is doing that? |
| 01:23.48 | st3v | I am trying to make a meetme configuration, but it says Unable to open '/dev/zap/pseudo': No such file or directory. This device file does exist in this location though. what could be the problem? |
| 01:24.07 | s34n | [TK]D-Fender: I got it working inband |
| 01:24.41 | sch19 | s34n, yes, I have a poly 501 scheduled to be ordered for receptionist desk, and the rest were going to be aastra, for simpliciy, and price. |
| 01:24.42 | [TK]D-Fender | st3v : My guess.... no ZTDUMMY, no 2.6 kernel, and no Zaptel hardware..... |
| 01:24.56 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 01:25.03 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:25.21 | SibRphrek | argh |
| 01:25.25 | SibRphrek | i wanna make this work |
| 01:25.40 | SibRphrek | how can i make it so if my ext. is busy asterisk fowards the call to my cell phone? |
| 01:25.47 | s34n | sch19: price? what pricing did you get on aastra? |
| 01:25.50 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 01:26.09 | sch19 | comparable to cisco and poly i meant, grandstream should be cheaper, no ? |
| 01:26.12 | st3v | [TK]D-Fender: I have all of them |
| 01:26.47 | *** join/#asterisk Eggplant (i=No@dsl-125.cascadeaccess.com) |
| 01:26.52 | st3v | I am using a 2.6 kernel, I have ztdummy loaded, and I am using a digium TE210P |
| 01:27.29 | sch19 | lsmod shows zaptel/ztdummy ? |
| 01:27.55 | Qwell[] | wait |
| 01:27.58 | s34n | sch19: I've seen aastra's over $200 |
| 01:28.02 | Qwell[] | why are you using ztdummy if you have hardware? |
| 01:28.16 | Qwell[] | how silly |
| 01:28.25 | sch19 | i have hardware and ztdummy is loaded, i don't think it's in use though. :) |
| 01:28.30 | Qwell[] | well, unload it |
| 01:28.37 | s34n | sch19: their lowest is around $140? |
| 01:28.55 | De_Mon | 2.6 doesn't need ztdummy? |
| 01:29.10 | Qwell[] | De_Mon: Not if you have hardware |
| 01:29.16 | st3v | well he said I needed ztdummy, so I just loaded it |
| 01:29.20 | st3v | and tried again |
| 01:29.22 | Qwell[] | well, unload it.. |
| 01:29.31 | De_Mon | Qwell[] how is 2.6 part of the equation then? |
| 01:29.31 | sch19 | s34n, $169 is what I paid for the 9133i's, I probably got raped, but I'm n00b so.. expected i guess. |
| 01:29.38 | Qwell[] | De_Mon: I never said it was |
| 01:29.51 | sch19 | noob* |
| 01:29.57 | De_Mon | Qwell[] you never said it wasn't either |
| 01:30.10 | I-MOD | the diff between 2.4 and 2.6 is that 2.6 can provide timing where as 2.4 required a uhci usb controller for timing |
| 01:30.12 | Qwell[] | De_Mon: Well then, where does cheese come into the equation? |
| 01:30.18 | Qwell[] | You never said it wasn't part of it |
| 01:30.19 | s34n | sch19: that's about right. and $199 for the polycoms? |
| 01:30.23 | De_Mon | [TK]D-Fender how does 2.6 fit in to that problem? |
| 01:30.30 | Qwell[] | Don't argue semantics with me, I'll win every time. :p |
| 01:31.34 | denon | he's right - there comes a point when we always just give up on logic, and let Qwell believe he won |
| 01:31.35 | st3v | so what else could be the problem.. it also says chandup: Unable to dup channel: |
| 01:31.41 | Qwell[] | I-MOD: BOTH require timing. The difference is that you can use rtc with 2.6...you still need ztdummy though |
| 01:31.44 | De_Mon | Qwell[] I was responding to TK's inclusion of the kernel version in his list, you piped up with reply that didn't address the question and started arguing -- whatever! |
| 01:31.45 | sch19 | s34n, didn't order yet, I've only got the 2 phones atm, for test-run before i feel confident running this in production. at the moment, my confidence level is still very low, as I break this thing every day. |
| 01:32.03 | Qwell[] | De_Mon: Because he has hardware. ztdummy isn't needed, period |
| 01:32.20 | Qwell[] | denon :p |
| 01:32.26 | sch19 | and some soft phones here and there |
| 01:32.29 | s34n | sch19: I haven't pounded the grandstream with a lot of use, so we'll see how it works out |
| 01:32.41 | mzo | i've abused my grandstreams a bunch and they work okay. |
| 01:32.45 | De_Mon | Qwell[] Still not helpful. Pretend he didn't ask the question and MY question was all that was there. I-MOD thankfully, knows more about it than you do |
| 01:32.49 | De_Mon | (burn) |
| 01:32.56 | s34n | sch19: so far, it has been more user-friendly than the polycom |
| 01:33.02 | Qwell[] | no, he doesn't, and no, I answered what I saw |
| 01:33.28 | De_Mon | I can't keep up ::boggle:: |
| 01:33.50 | De_Mon | so its 2.4 + usb + ztdummy or 2.6 + rtc + ztdummy |
| 01:34.02 | Qwell[] | unless they have hardware |
| 01:34.19 | Qwell[] | which in this case, was true |
| 01:34.29 | De_Mon | or hardware ok, got it. The case, was my own setup. |
| 01:34.52 | Qwell[] | De_Mon: Great, I wasn't correcting you. I was correcting whoever said they were using ztdummy with a digium card in the box |
| 01:35.22 | *** join/#asterisk Cresl1n (n=matt@m515e36d0.tmodns.net) |
| 01:35.35 | Qwell[] | So, if you'd like to flame me for pointing out somebodys flaw, go right ahead, but don't expect help from me in the future |
| 01:35.35 | De_Mon | yeah, you were talking to me about his problem when I was talking about my own.. assuming too much can be dangerous |
| 01:35.36 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:36.00 | Qwell[] | I never once mentioned your name, until you tried to start on me. :) |
| 01:36.04 | sch19 | atm, I'm just running on a *@home install, because I know next to nothing, and it loaded ztdummy *duck* |
| 01:36.18 | SibRphrek | nice |
| 01:36.22 | SibRphrek | got 911 and 411 to work |
| 01:36.25 | sch19 | yay |
| 01:36.27 | mzo | i thought @home didn't load zdummy? |
| 01:36.35 | sch19 | did 911 appreciate your test call ? |
| 01:36.38 | tronix | :-) |
| 01:36.38 | Qwell[] | sch19: If you have hardware, then AMP is wrong, and you should correct it :p |
| 01:36.50 | SibRphrek | sch19: i hung up as soon as i realized it worked |
| 01:36.59 | sch19 | mzo, apparently it does, because i installed it with the hardware in the computer, and it loaded it. |
| 01:36.59 | Qwell[] | SibRphrek: bad move... |
| 01:37.00 | mzo | um |
| 01:37.05 | st3v | so does anyone know why asterisk cant open /dev/zap/pseudo although its there and I have a hardware digium card? |
| 01:37.06 | mzo | you always tell 911 that it's a test call |
| 01:37.06 | SibRphrek | Qwell it's NYC they don't care |
| 01:37.07 | SibRphrek | :) |
| 01:37.10 | mzo | they DO care |
| 01:37.11 | mzo | dumbass |
| 01:37.16 | tronix | sibrphrek: careful... they often will send police out to check "hangup calls" since possible domestic violence |
| 01:37.17 | Qwell[] | They *ALWAYS* dispatch an officer if you hangup on a 911 call |
| 01:37.18 | SibRphrek | mzo - relaax |
| 01:37.19 | mzo | you're required to tell them it's a test call |
| 01:37.26 | SibRphrek | ok ok |
| 01:37.27 | Ariel_ | you can dial and tell them your testing to make sure it works. on a new pbx they allow one or 2 calls |
| 01:37.27 | SibRphrek | reloax |
| 01:37.29 | SibRphrek | it was one call |
| 01:37.30 | SibRphrek | it's fine |
| 01:37.33 | SibRphrek | besides |
| 01:37.34 | Qwell[] | heh |
| 01:37.34 | SibRphrek | they |
| 01:37.40 | sch19 | Qwell[], amp manipulates modprobe.conf ? |
| 01:37.40 | SibRphrek | they'll go to my office or our colo |
| 01:37.41 | SibRphrek | :) |
| 01:37.44 | SibRphrek | and they haven't called back |
| 01:37.44 | RoyK | -.-. --.- |
| 01:37.46 | Ariel_ | if you just hang up they sometimes call you back |
| 01:37.49 | Qwell[] | sch19: it probably mods something else |
| 01:38.15 | Qwell[] | I'll ammend my statement. They *always* dispatch an officer, if (and this only happens sometimes) they call you back, and you don't answer. |
| 01:38.27 | Qwell[] | Ariel_: Thanks...forgot that part |
| 01:39.14 | Qwell[] | sometimes, even if you say everything is okay, they'll still have an officer drive by |
| 01:39.27 | sch19 | assuming they picked up before you hung up. |
| 01:39.35 | sch19 | here if you dial 911, they put you on hold half the time. |
| 01:40.14 | sch19 | nobdoy has a clue on my blf/aastra problem, eh? |
| 01:40.24 | De_Mon | bah, any "flaming" was unintentional, just tryin to get a straight answer, which I got in a round about way |
| 01:40.31 | sch19 | imagine i spelled nobody correctly ^ |
| 01:40.41 | Qwell[] | De_Mon: Because you asked in a roundabout way... |
| 01:41.10 | De_Mon | true enough! :D |
| 01:41.37 | De_Mon | my brain has turned to mush after staring at this queue stuff for the past 3 hours |
| 01:42.27 | sch19 | I was having problems with the queue feature yesterday, no matter how i set it up, it exited the queue after 15 seconds. |
| 01:42.29 | st3v | could my problem be because I'm using version 1.2.0 |
| 01:42.49 | De_Mon | sch19 I'm on 1.2.1 and can't get it to ever exit! |
| 01:42.51 | st3v | does that have any problems with conferencing |
| 01:43.07 | Qwell[] | st3v: Only one way to find out... |
| 01:43.08 | sch19 | De_Mon, I wish I had that problem, that was my desire at the time |
| 01:43.23 | st3v | well I am having problems with conferencing |
| 01:43.36 | Qwell[] | st3v: Try 1.2.4 |
| 01:43.37 | sch19 | I'm not up to that yet, sorry st3v |
| 01:43.53 | De_Mon | sch19 ever figure it out? I'll bet you had a [global] timeout set |
| 01:44.07 | De_Mon | errr [general] |
| 01:44.14 | tamp4x | anyone here use astcc? i used to sample dia plan., and the numbers either tell me the length the callign card msut be and hangs up, or just says goodbye ..... i tested the database conenction and its fine...so waht gives? |
| 01:44.22 | tamp4x | dial plan |
| 01:44.54 | sch19 | De_Mon, most probably, but like i say, I'm newb and I am trying to learn slowly.. AMP is no great help there.. when i touch something in the confs AMP breaks it next time i make a change thru it |
| 01:45.18 | mzo | amp rox |
| 01:45.41 | sch19 | how can i make it so that there is a 15 second timeout for extensions -> voicemail, but no timeout for the queue then? don't i need it to be a global timeout? |
| 01:45.55 | De_Mon | sch19 yikes. I decided to figure out how manage it manually before going for any helper apps. glad I did now. |
| 01:46.28 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 01:46.29 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 01:46.49 | sch19 | De_Mon, yes, good way to do it. I think I've got the syntax and things down, I am still learning about AGI and where everything is though. |
| 01:46.53 | De_Mon | all this queue stuff sure likes to generate crashes |
| 01:46.58 | sch19 | I think I'm doing ok for being at this a couple of days :P |
| 01:47.30 | De_Mon | core debug and verbose arn't giving me any meaningfull errors either :/ |
| 01:48.20 | dudes | De_Mon - app_queue tends to be buggy |
| 01:48.31 | sch19 | great, just what i wanted to hear |
| 01:49.01 | De_Mon | dudes that means it's not just me, I feel better already :P |
| 01:49.47 | sch19 | lol |
| 01:49.49 | sch19 | the thing that worries me, is I had *@home 2.2 installed, and wanted to upgrade the asterisk on it to yesterdays release, but i broke it somehow and it would no longer run. :( |
| 01:50.42 | sch19 | makes me worry greatly. I think fresh install of some linux, and manual install of asterisk + friends is in order, so i know what I'm doing. |
| 01:50.42 | dudes | I don't use app_queue, but in the past I have. In any event, it's easier handling queues outside of *. |
| 01:51.32 | De_Mon | how so? |
| 01:51.41 | De_Mon | I'm ready for a new approach |
| 01:51.51 | sch19 | that's how i felt yesterday |
| 01:52.00 | sch19 | today, haven't tried it again yet. |
| 01:52.06 | Qwell[] | De_Mon: Is it actually crashing? |
| 01:52.09 | dudes | What are you doing? |
| 01:52.21 | Qwell[] | sch19: AMP and *@~ mask problems...you can never tell what causes them |
| 01:52.51 | De_Mon | Qwell[] yup no more asterisk processes, no console |
| 01:52.57 | _Sam-- | never seen it that way *@! |
| 01:52.58 | _Sam-- | er ~ |
| 01:53.01 | justinu | sch19: i think you're on the right track |
| 01:53.04 | Qwell[] | De_Mon: Get a backtrace |
| 01:53.41 | sch19 | Qwell[],I just figured it is probably a good idea to know my way around * backwards and forwards before I'm dependant on it, pulling all-nighters with huge downtimes, because I used some disk 4 dummies |
| 01:54.02 | Qwell[] | sch19: indeed |
| 01:54.03 | sch19 | so, guess I'll go do that, tomorrow. late atm |
| 01:54.16 | sch19 | what are you all reccomending for base os? just any flavor will do? |
| 01:54.21 | Qwell[] | pretty much |
| 01:54.24 | sch19 | (please don't say gentoo -- hah) |
| 01:54.28 | Qwell[] | whatever you're familiar with should be fine |
| 01:54.33 | dudes | Debian works fine |
| 01:54.34 | Qwell[] | I prefer to run it on gentoo :p |
| 01:54.57 | sch19 | you wouldn't happen to have a 12" spoiler bolted on the back of your pento, would you? |
| 01:55.03 | dudes | Just search debian asterisk on the wiki |
| 01:55.12 | Qwell[] | sch19: 16" |
| 01:55.16 | sch19 | ahh :P |
| 01:55.23 | justinu | pinto? |
| 01:55.24 | _Sam-- | pento? |
| 01:55.26 | _Sam-- | ah |
| 01:55.29 | tronix | sch19: i'll second Qwell[]'s comment -- whatever you're comfortable with will do great |
| 01:55.45 | sch19 | justinu, probably, I grew up on irc, we can't spell. |
| 01:55.49 | Qwell[] | unless it's like Linspire or something |
| 01:55.51 | De_Mon | aptitude install backtrace |
| 01:56.00 | Qwell[] | De_Mon: less doc/README.backtrace |
| 01:56.03 | De_Mon | gah *smacks himself* |
| 01:56.25 | sch19 | I kind of fond of the redhat ones, rhel, centos, fedora |
| 01:56.36 | sch19 | just need to get rid of the bloat is all |
| 01:57.02 | dudes | I like Debian cause it's a small 100MB netISO and you just apt-get what you need. |
| 01:57.08 | De_Mon | ment to install gdb ;) |
| 01:57.27 | sch19 | dudes, the thing I hate most about debian is that it's different from all the others, "su" annoys the heck out of me. |
| 01:57.29 | dudes | But Gentoo is a small ISO too |
| 01:57.41 | justinu | bah, iso's are for wussies |
| 01:57.48 | justinu | pxelinux |
| 01:57.51 | dudes | sch19 - su? |
| 01:57.52 | sch19 | haha |
| 01:57.53 | Qwell[] | real men punch in their bootloaders in binary |
| 01:57.58 | Qwell[] | with dipswitches |
| 01:58.07 | dudes | you don't have to sudo in Debian, you get root. Perhaps you're thinking of Ubuntu |
| 01:58.11 | sch19 | dudes, yes, different syntax on debian than any other *nix-like os I've ever seen. |
| 01:58.34 | sch19 | no, su. |
| 01:58.58 | dudes | yea, what's wrong with su? |
| 01:59.01 | Qwell[] | every distro has su |
| 01:59.03 | De_Mon | huh? what? |
| 01:59.09 | sch19 | i "su -l" a lot, from users, debian makes me type "su -", annoying :P |
| 01:59.19 | justinu | uh |
| 01:59.19 | Qwell[] | "makes" you? no |
| 01:59.22 | De_Mon | whats different about the syntax? |
| 01:59.22 | justinu | su - is pretty standard |
| 01:59.23 | dudes | what are you talking about |
| 01:59.30 | De_Mon | oookay |
| 01:59.30 | Qwell[] | the - gives you the env of the new user |
| 01:59.35 | sch19 | -l does |
| 01:59.36 | Qwell[] | man su |
| 01:59.39 | Qwell[] | no, - does |
| 01:59.40 | justinu | it like works on other unix flavors too |
| 01:59.51 | justinu | like solari |
| 01:59.52 | justinu | s |
| 01:59.56 | Qwell[] | solarii? |
| 02:00.02 | *** join/#asterisk carex (n=widiasto@202.159.57.221) |
| 02:00.13 | justinu | my solari boxen |
| 02:00.15 | sch19 | then it was expanded to allow - too, it used to be -l for as long as i can remember.. and I am set in my ways :( |
| 02:00.34 | Qwell[] | bbl |
| 02:00.53 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.62.185.Dial1.Chicago1.Level3.net) |
| 02:01.12 | sch19 | service <x> start, and service <x> stop is nice too, debian doesn't seem to have that sort of scripting. |
| 02:01.21 | sch19 | but I'm not expert. |
| 02:01.22 | dudes | sch19 - get the su source and change the - arg to -l =) |
| 02:01.28 | sch19 | :) |
| 02:01.49 | sch19 | there was some other things that bothered me, I just cant' remember them atm, it's been a while. |
| 02:02.24 | dudes | If you're referring to the pre-woody days perhaps, but I think Debian has taken some nice steps over the last year or two. |
| 02:02.29 | sch19 | those too are probably just my being used to typing in the wrong crap, and debian is less forgiving |
| 02:02.59 | De_Mon | sch19 invoke-rc.d <x> start/stop |
| 02:03.08 | sch19 | good to know. |
| 02:03.21 | SibRphrek | http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
| 02:03.23 | SibRphrek | anyone try that? |
| 02:03.28 | De_Mon | I found that one last week, i like /etc/init.d/<tab> more tho :) |
| 02:03.33 | dudes | It really comes down to preferences ;p |
| 02:03.40 | p0g0_ | hi, 'CLI> zap show status' shows one of two X101P's clones (the first) with "ALARMS RED", the second has "OK"... what's that mean? |
| 02:03.44 | Nugget | Moo. |
| 02:03.45 | sch19 | since I'm most famililar with the redhat way, maybe it wouldn't be such a bad idea to learn something new :D |
| 02:03.48 | De_Mon | ln -s invoke-rc.d service |
| 02:04.23 | sch19 | will take some getting used to to convert my thinking from rpm to debpkg (?) |
| 02:04.42 | De_Mon | are you planning to build packages? |
| 02:04.50 | De_Mon | I hated rpms and yum |
| 02:04.54 | dudes | sch19 - just apt-get install |
| 02:05.02 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
| 02:05.04 | dudes | yum is dumb |
| 02:05.09 | dudes | I think anyway |
| 02:05.16 | De_Mon | yum dumb um hum |
| 02:05.29 | sch19 | rpm -q pkgname -i, and rpm -q pkgname -l i've found to be useful in the past, is all.. |
| 02:05.40 | sch19 | isn't the "installed debs" package manager called debpkg or something liek that? |
| 02:05.50 | dudes | dpkg? |
| 02:05.55 | sch19 | that's it |
| 02:05.57 | dudes | I think anyway |
| 02:06.21 | sch19 | maybe i just don't know how to use apt-cache |
| 02:06.26 | sch19 | and it will give me what i want |
| 02:06.36 | dudes | apt-cache search package |
| 02:06.39 | De_Mon | I duno what -i and -l |
| 02:06.46 | sch19 | -i is info, like a description |
| 02:06.57 | sch19 | -l is list of files installed by the pkg, and their path |
| 02:07.41 | sch19 | we're so far off topic here though, I should be in #debian-newbs, only.. they'll just flame + kick me I'm sure :P |
| 02:08.30 | De_Mon | apt-get is just for installing stuff, apt-cache show <pkg> will show a desc and dpkg -S(or is it s) lists files in a package. |
| 02:08.50 | rene- | what does -lidn represents |
| 02:08.51 | sch19 | so, just a matter of getting used to |
| 02:09.07 | De_Mon | aptitude combines apt-get apt-cache and a nice GUI/CLI all in one. but back on topic |
| 02:09.33 | sch19 | aptitude is a gui frontend I assume ? |
| 02:09.51 | De_Mon | or cli |
| 02:10.03 | SibRphrek | why is it that all my extentions that start with * give me a busy signal |
| 02:10.20 | De_Mon | isn't * reserved? |
| 02:10.40 | sch19 | SibRphrek, I'd try to help, but I'd be wrong :P |
| 02:10.41 | SibRphrek | well if i remove the * the shit don't work eitheraay |
| 02:10.51 | SibRphrek | i'm trying to setup wake up calls and DND |
| 02:10.53 | sch19 | going home for the day, thanks for info guys, be back tomorrow |
| 02:13.20 | De_Mon | I can't find the page but I remember having problems with *7 and *8 because they were used internally for parking and blind transfers or somesuch |
| 02:13.42 | SibRphrek | De_Mon: http://danielwebb.us/software/asterisk/ |
| 02:13.51 | SibRphrek | i'm trying to do his DND and wake up setups |
| 02:13.54 | SibRphrek | from his extentions.conf |
| 02:14.02 | SibRphrek | i copied pretty much everything, including the other conf's |
| 02:14.09 | SibRphrek | and the extensions show when i reload asterisk |
| 02:14.12 | SibRphrek | i just get a busy signal |
| 02:14.22 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 02:14.58 | De_Mon | SibRphrek set verbose to 3 and watch the CLI and see what's going on |
| 02:15.15 | De_Mon | it may be a 404 errror where the phones arent setup correctly |
| 02:15.17 | SibRphrek | the CLI isn't moving |
| 02:15.27 | De_Mon | then asterisk isn't talking to your phone |
| 02:15.32 | SibRphrek | but it is |
| 02:15.35 | SibRphrek | cuz i can make regular calls |
| 02:16.15 | SibRphrek | but yeah i'm getting 404 not found from my x-lte |
| 02:16.35 | De_Mon | well, if the cli isn't moving when you dial an extension it's because there is no extension configured in asterisk that your phone is allowed to use |
| 02:16.55 | De_Mon | paste your extensions.conf somewhere |
| 02:16.58 | SibRphrek | ok hang on |
| 02:17.13 | *** join/#asterisk mmlj4 (n=looseduk@ip70-171-92-106.no.no.cox.net) |
| 02:17.14 | De_Mon | if their sip phones include that too |
| 02:17.20 | De_Mon | they're |
| 02:18.00 | SibRphrek | http://pastebin.com/534798 |
| 02:18.06 | SibRphrek | that's extentions |
| 02:18.31 | SibRphrek | from that tell me what else you need |
| 02:18.58 | [av]bani | a blank check would be nice |
| 02:19.01 | *** part/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 02:19.02 | eyz | heh |
| 02:20.22 | *** join/#asterisk danitza (n=dpizarro@libra.infostar.com.pe) |
| 02:20.30 | De_Mon | SibRphrek what context do your sip phones go into? I'm guessing internal by the looks |
| 02:20.49 | SibRphrek | uhm - i don't understand the ? - sorry i'm not that good with this stuff |
| 02:20.53 | infinity2 | does voxee have a problem with callerID ? i set callerID before dialing and i get "unavailable" ...i swear it use to work also. |
| 02:21.04 | De_Mon | context is a variable in sip.conf |
| 02:21.16 | De_Mon | it tells asterisk where to dump the caller when he starts dialing |
| 02:21.24 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 02:21.25 | SibRphrek | De_Mon: context=default |
| 02:21.34 | SibRphrek | oh |
| 02:21.41 | SibRphrek | for my ext. context=internal |
| 02:22.07 | De_Mon | SibRphrek then you need to include => <other context> in internal |
| 02:22.27 | SibRphrek | ?? |
| 02:22.30 | De_Mon | so include => app-dnd and any other context with extensions you want to use internally |
| 02:22.43 | SibRphrek | where? in sip.conf? |
| 02:22.49 | De_Mon | see line 22 extensions.conf |
| 02:23.02 | De_Mon | thats why you're able to make outgoing calls |
| 02:23.09 | SibRphrek | ok |
| 02:23.25 | De_Mon | so include => app-dnd and |
| 02:23.31 | De_Mon | include => wakeupcalls |
| 02:24.08 | SibRphrek | anywhere in extentions.conf? |
| 02:24.19 | SibRphrek | or under default? |
| 02:24.31 | De_Mon | under line 22 |
| 02:24.32 | danitza | hello |
| 02:24.56 | SibRphrek | ok |
| 02:24.58 | SibRphrek | reload? |
| 02:25.02 | De_Mon | put them under [default] if you want people calling into the box to have access to those extensions |
| 02:25.06 | De_Mon | extensions reload, ya |
| 02:25.08 | *** join/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 02:25.33 | SibRphrek | De_Mon: still didn't work |
| 02:25.36 | SibRphrek | but |
| 02:25.40 | SibRphrek | wakeupcall CLI gave me something |
| 02:25.47 | SibRphrek | oh |
| 02:25.48 | SibRphrek | oops |
| 02:25.49 | SibRphrek | hang on |
| 02:26.29 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 02:26.31 | znoG | can one make all FXO modules in a TDM card "detect" an incoming fax and call the fax priority? |
| 02:27.29 | De_Mon | look at http://danielwebb.us/software/asterisk/examples/extensions.conf again |
| 02:27.42 | De_Mon | it has the include lines |
| 02:28.26 | SibRphrek | i have those on the top |
| 02:29.07 | SibRphrek | De_Mon: ok - wake up calls work |
| 02:29.09 | De_Mon | #include includes files, include => puts them within other contexts |
| 02:29.10 | SibRphrek | but DND still errors |
| 02:29.29 | De_Mon | busy? |
| 02:29.32 | De_Mon | err 404 |
| 02:29.46 | SibRphrek | i dunno |
| 02:29.47 | SibRphrek | hang on |
| 02:29.49 | SibRphrek | GF on the phone |
| 02:29.50 | SibRphrek | crying |
| 02:30.13 | justinu | high maintenence women... boo |
| 02:30.24 | SibRphrek | she's not high maintenance - she's sick |
| 02:30.24 | De_Mon | girls cry too much hehehe |
| 02:30.35 | De_Mon | I broke a nail boo hoo |
| 02:30.42 | De_Mon | I barfed boohoo |
| 02:31.18 | justinu | whatever - if she's calling you in tears, she's high maintenance |
| 02:31.35 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 02:31.45 | SibRphrek | damn - DND still 404's |
| 02:31.59 | De_Mon | SibRphrek is do_not_disturb.conf exactly the same as on the site? |
| 02:32.07 | SibRphrek | yes |
| 02:32.09 | SibRphrek | copy and pasted |
| 02:32.12 | De_Mon | and is it in your asterisk config dir with all the other files |
| 02:32.56 | SibRphrek | yeah i put the do_not_disturb.conf and the stdextensions.conf are in my /etc/asterisk |
| 02:33.05 | De_Mon | paste your new extensions.conf |
| 02:33.26 | De_Mon | provided you include => dnd-app under internal, it should be workin' |
| 02:33.39 | De_Mon | err app-dnd --whatever |
| 02:35.15 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 02:35.16 | SibRphrek | http://pastebin.com/534814 |
| 02:35.21 | SibRphrek | extensions.conf |
| 02:35.57 | De_Mon | you got exten 363 twice |
| 02:36.06 | De_Mon | when you reloaded it would have complained about that |
| 02:36.14 | SibRphrek | one should have a * and the other a # |
| 02:36.24 | De_Mon | I see neither |
| 02:36.45 | SibRphrek | hang on |
| 02:36.50 | SibRphrek | i musta taken them out for something |
| 02:36.50 | De_Mon | you sure they're there? |
| 02:36.53 | SibRphrek | i put them back and i'm reloading |
| 02:37.20 | De_Mon | if it's still screwey, you have to atleast give them different numbers |
| 02:37.22 | SibRphrek | as soon as i get to *36 i get a busy signal |
| 02:37.33 | SibRphrek | i don't even get to put in the last 3 |
| 02:38.27 | De_Mon | something else is taking over |
| 02:38.41 | De_Mon | try a different series of numbers without * or # |
| 02:38.55 | justinu | that's your sip phone |
| 02:38.56 | SibRphrek | ok |
| 02:39.06 | De_Mon | justinu dtmf problem? |
| 02:39.13 | justinu | some of them are programmed with * as a vertical service activation code |
| 02:39.15 | justinu | sipura is like that |
| 02:39.30 | SibRphrek | yeah and i have a sipura |
| 02:39.34 | justinu | so there you |
| 02:39.35 | justinu | go |
| 02:39.39 | justinu | you can turn it off |
| 02:40.00 | De_Mon | or juse something like 1363 and 2363 |
| 02:40.02 | SibRphrek | i think it worked De_Mon |
| 02:40.17 | De_Mon | SibRphrek grats! |
| 02:40.18 | SibRphrek | i heard the DND work |
| 02:40.24 | SibRphrek | but i just tried to call it and it got thru |
| 02:40.27 | SibRphrek | so it's not working |
| 02:40.52 | SibRphrek | ooh |
| 02:40.53 | SibRphrek | erros |
| 02:40.56 | SibRphrek | hang on pasting |
| 02:41.02 | De_Mon | atleast it's getting to the extension now you can debug everything else |
| 02:41.14 | *** join/#asterisk YARICK (n=spiderma@pool-151-196-17-195.balt.east.verizon.net) |
| 02:41.22 | SibRphrek | http://pastebin.com/534818 |
| 02:43.12 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 02:43.28 | _-Jon-_ | hey everyone |
| 02:43.55 | _-Jon-_ | I'm having a very strange problem with my phone all of a sudden, hoping someone here right be able to help |
| 02:44.32 | De_Mon | ask and see |
| 02:44.45 | _-Jon-_ | Basically if I get a call in through any of my providers (sixtel and broadvoice), it won't show CID, but if I call from another extention in my house it shows up |
| 02:45.08 | SibRphrek | De_Mon: my wakeup call isn't calling the right person - so it's buggen out |
| 02:45.21 | _-Jon-_ | But, if I check the cdr logs, it shows the number, and when i run sip debug it shows it as well |
| 02:45.24 | SibRphrek | Feb 1 21:46:02 NOTICE[29822]: chan_local.c:494 local_alloc: No such extension/context 5163480146@internal creating local channel |
| 02:45.52 | De_Mon | SibRphrek so create that extension in the context |
| 02:46.32 | SibRphrek | oh the wakeup.php ? |
| 02:46.45 | De_Mon | 5163480146@internal |
| 02:46.54 | De_Mon | internal is the context [internal] in extensions.conf |
| 02:47.07 | De_Mon | it should be dialing out |
| 02:47.29 | De_Mon | how do you make an outgoing call, will just dialing that number work? |
| 02:47.41 | *** part/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 02:47.49 | SibRphrek | to make an outgoing call i dial 9+1+phonenumber |
| 02:48.25 | De_Mon | then your app should be trying to dial 915163480146@internal instead of what it's dialing |
| 02:48.27 | SibRphrek | i don't understand why wakeup isn't trying to dial my ext. which is 200, instead of 5163480146@internal |
| 02:48.47 | SibRphrek | 5163480146 is my number - so if you called me from outside my phone will ring |
| 02:49.02 | SibRphrek | i think wakeup should be trying to call 200 |
| 02:49.06 | SibRphrek | not the full thing |
| 02:50.26 | _-Jon-_ | Is there any reason why a copy of asterisk wouldn't show CID? Basically my computer died, so I compiled asterisk on a new machine and put all my config files in place exactly like they were, but it won't show CID |
| 02:50.42 | De_Mon | it's setup in the database or the #included wakeup config |
| 02:50.44 | De_Mon | shrug |
| 02:50.44 | lithi | Anyone care to recommend a user-friendly softphone? Firefly 3rd party is nice but its got a few problems like insted of busy it sends Call rejected by xxx.xxx.xxx.xxx: In call. |
| 02:50.53 | justinu | no jon, it's a configuration issue |
| 02:51.26 | znoG | lithi: i use DIAX, its not bad. Otherwise IAXComm is OK too |
| 02:51.33 | _-Jon-_ | justinu, but i just don't get it.. the configs are EXACLY the same. And if i run 'sip debug', it clearly shows my cell # in there |
| 02:52.12 | De_Mon | lithi i'm using xlite & eyebeam |
| 02:52.26 | znoG | _-Jon-_: so you've tried NoOp(${CALLERID}) and didn't show a thing? |
| 02:52.28 | justinu | unless it's a bug in asterisk |
| 02:52.58 | znoG | _-Jon-_: actually from what you've said I would expect it to show it. What phone are you using? |
| 02:52.58 | _-Jon-_ | znoG, never used NoOp before. What does it do? |
| 02:53.19 | _-Jon-_ | znoG, I have a Sipura 1001 with a cordless |
| 02:53.24 | znoG | <PROTECTED> |
| 02:53.24 | znoG | purposes. Any text that is provided as arguments to this application can be |
| 02:53.24 | znoG | viewed at the Asterisk CLI. This method can be used to see the evaluations of |
| 02:53.24 | znoG | variables or functions without having any effect. |
| 02:53.24 | lithi | De_Mon: Xlite is ok but the interface is a little clumsy |
| 02:53.44 | lithi | znoG: Ill give DIAX a try |
| 02:53.51 | _-Jon-_ | ohhh okay, I'll put that into my extentions.conf to see what happens |
| 02:53.57 | znoG | _-Jon-_: right, you could get ethereal going and see the SIP packets, to see if it's really being sent or not (the caller ID info) |
| 02:55.00 | *** join/#asterisk _cleric_ (n=dacleric@84.245.166.165) |
| 02:55.21 | _-Jon-_ | hmm let me try the NoOp first, and then I'll try ethereal |
| 02:55.38 | znoG | _-Jon-_: well you're saying from other extensions in your house the CID shows up, so that can't be it. |
| 02:56.25 | _-Jon-_ | That's true. Okay get this.. |
| 02:56.31 | mdave | how can I have a particular extension, go to another context and then accept additional dialing? |
| 02:56.41 | mdave | like a second dialtone? |
| 02:56.59 | mdave | eg how can I use goto(), but specify a context only, without any digits |
| 02:57.37 | _-Jon-_ | I have a line like this: exten => _242X,1,Dial(SIP/${EXTEN},30,Tr) and if I call 2423 => 2425 it shows CID, but 2425 => 2523 no CID |
| 02:57.39 | znoG | Goto(context,s,1) ?? |
| 02:57.51 | _-Jon-_ | BUT, if I connect my Sipura to a different server completely and try the same experiment it works fine |
| 02:57.55 | mdave | thats what i tried |
| 02:58.01 | mdave | and it said 'invalid extension s' |
| 02:58.24 | znoG | _-Jon-_: i presume the sip.conf entries for both sipura lines are identical? |
| 02:58.24 | mdave | <PROTECTED> |
| 02:58.44 | znoG | then you don't have: exten => s,1,Something in [menu1] |
| 02:59.04 | mdave | i dont want it to do anything immediately, i want it to then wait for additional dialing |
| 02:59.12 | _-Jon-_ | znoG, yeah, except obvious things like 'host' are different. but i copied the lines from the non-working one to the working one |
| 02:59.17 | mdave | should I add a s,2,noop() ? |
| 02:59.18 | mdave | er |
| 02:59.22 | mdave | s,1,noop() ? |
| 02:59.25 | znoG | mdave: not sure how you're going to provide a new dialtone within the context |
| 02:59.40 | mdave | it doesnt even need a new dialtone, as long as it accepts more dialing |
| 02:59.56 | mdave | how does one do submenus? |
| 02:59.59 | znoG | _-Jon-_: all I'd suggest is comparing SIP packets from 2423 and 2425 to see if there are any obvious differences |
| 03:00.26 | znoG | mdave: you could just do: exten => s,1,WaitExten |
| 03:00.32 | mdave | alright |
| 03:00.34 | znoG | and: exten => 55,1,Something |
| 03:00.40 | znoG | (for extension 55) |
| 03:00.42 | _-Jon-_ | znoG, alright. ethereal it is! |
| 03:00.45 | _-Jon-_ | Thanks for your help |
| 03:01.08 | znoG | wasn't much, but hopefully lead you onto the right track |
| 03:01.08 | znoG | but you're welcome :) |
| 03:01.43 | znoG | _-Jon-_: just for the heck of it, have you tried switching phones between 2423 and 2425? |
| 03:01.56 | mdave | hrm.. that didnt seem to work.. lemme go look at docs for that command |
| 03:02.23 | _-Jon-_ | znoG, actually i never even thought about that. might be worth a shot though |
| 03:02.24 | znoG | mdave: you'd have to www.pastebin.com your extensions.conf |
| 03:02.50 | znoG | _-Jon-_: well it takes 2 seconds and it's worthwile to kill the possibility that its the phone causing it |
| 03:03.15 | _-Jon-_ | znoG, that's true. I'll try that out and see what happens |
| 03:03.27 | znoG | good luck |
| 03:03.34 | znoG | let me know if you figure it out |
| 03:03.40 | znoG | for the sake of knowing what it was :) |
| 03:03.45 | _-Jon-_ | Will do. I'll get to the bottom of this :) |
| 03:07.55 | *** join/#asterisk J_- (n=raz@32-pool1.ras14.floca.alerondial.net) |
| 03:08.59 | mdave | basically I want to send inbound calls from my voip provider to a menu, rather than ringing them thru directly |
| 03:09.25 | J_- | hy all |
| 03:09.29 | mdave | originally the registration line had /s at the end, and I had a context that had s that just rang thru, then that stopped working at some point, not sure why |
| 03:09.42 | mdave | i changed the register to /mynumber, and changed the extensions to match that too |
| 03:10.15 | mdave | but cant figure out how after voip sends the original set of digits, to then (after playing a menu prompt) accept dialing after ive picked ip the call |
| 03:10.26 | mdave | after * has picked up, that is |
| 03:10.38 | J_- | anyone else notice that weighting static queues is broken along with ackcall |
| 03:10.46 | mdave | wondering if there is even an example of that somewhere |
| 03:10.58 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 03:11.32 | znoG | mdave: seems like a standard menu setup |
| 03:11.41 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 03:11.47 | mdave | yeah, but how to i specify something like that? |
| 03:11.53 | znoG | mdave: ie. i have a voip provider who sends me calls, and it presents the caller with a menu and a bunch of options they can dial |
| 03:12.18 | mdave | do I remove the 'extension' from the end of the register line? |
| 03:12.20 | *** join/#asterisk YARICK (n=spiderma@pool-151-196-17-195.balt.east.verizon.net) |
| 03:12.56 | mdave | is there an example config somewhere I can look at for doing something like that |
| 03:13.43 | dumb---me | hey |
| 03:14.06 | dumb---me | does asterisk have built in JVM |
| 03:14.10 | dumb---me | anyone? |
| 03:14.53 | Math[laptop] | no |
| 03:15.16 | dumb---me | so I have to run the java SDK in order to be able to run java scripts? |
| 03:15.30 | mishehu | javascript != java |
| 03:15.45 | mishehu | they are completely different languages. |
| 03:15.55 | Math[laptop] | and, if you want to run *JAVA* stuff, you only need the JRE (Java Runtime Env) |
| 03:16.32 | dumb---me | but i can use java language to built agi scripts? |
| 03:16.32 | znoG | it'd have to be a pretty beefy box to run smoothly! |
| 03:17.02 | znoG | dumb---me: check voip-info.org - search for java agi |
| 03:17.51 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 03:17.55 | *** join/#asterisk asteriskmonkey (n=phil@67.71.177.202) |
| 03:17.58 | *** join/#asterisk dfgas (n=dfgas@adsl-69-210-88-233.dsl.milwwi.ameritech.net) |
| 03:18.02 | asteriskmonkey | hey all |
| 03:18.06 | dfgas | where do i setup dir info? |
| 03:18.25 | dfgas | so when someone tried to get directory of extentions |
| 03:18.38 | asteriskmonkey | is there an internal command that lets me write information to a database on a different server in asterisk? |
| 03:19.35 | SibRphrek | can someone help me with this DND extention i found |
| 03:19.45 | SibRphrek | http://danielwebb.us/software/asterisk/examples/do_not_disturb.conf |
| 03:19.51 | SibRphrek | every time i dial the extention |
| 03:19.59 | SibRphrek | CLI reports Feb 1 22:20:54 WARNING[29822]: app_macro.c:149 macro_exec: No such context 'macro-return_to_top' for macro 'return_to_top' |
| 03:20.02 | SibRphrek | i dunno what i'm doing wrong |
| 03:20.43 | znoG | chan_zap.c: In function `load_module': |
| 03:20.43 | znoG | chan_zap.c:11116: error: `inuse' undeclared (first use in this function) |
| 03:20.44 | znoG | chan_zap.c:11116: error: (Each undeclared identifier is reported only once |
| 03:20.44 | znoG | chan_zap.c:11116: error: for each function it appears in.) |
| 03:20.50 | znoG | anyone else had issues compiling SVN Asterisk? |
| 03:21.13 | *** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-51.cst.dccnet.com) |
| 03:21.20 | Math[laptop] | if you're running svn you should probably be able to diagnose that |
| 03:23.01 | mdave | ok, I *do* have a context [frombv]. It *does* have an exten => s,1,something entry. the context=frombv is set in the entry for broadvoice. yet when it passes me a call, * is sending 414 not found back to * |
| 03:23.03 | mdave | wtf |
| 03:23.14 | mdave | er 404 not found |
| 03:23.17 | znoG | Math[laptop]: ..? should I not be running SVN? |
| 03:23.24 | Qwell | mdave: 404 what? Check the sip debug |
| 03:23.37 | J_- | I remember seing a broadvoice patch on the bug tracker |
| 03:23.50 | austinnichols10 | anyone got a good source for an spa3k? |
| 03:24.02 | mdave | SIP/2.0 404 Not Found |
| 03:24.04 | *** join/#asterisk FastJack_ (i=fastjack@p5091F8AF.dip.t-dialin.net) |
| 03:24.12 | Qwell | mdave: Why is it giving a 404? |
| 03:24.17 | mdave | thats my question |
| 03:24.22 | Qwell | read the sip debug |
| 03:24.29 | mdave | ? |
| 03:24.36 | mdave | as far as I can see, it doesnt say *why* |
| 03:24.47 | mdave | actually |
| 03:24.49 | mdave | waitaminute |
| 03:24.52 | mdave | its not looking for s |
| 03:24.56 | mdave | its looking for my bv number |
| 03:24.58 | mdave | grr |
| 03:25.01 | Qwell | there you go |
| 03:25.05 | mdave | well I want it to go to s |
| 03:25.11 | Qwell | why? |
| 03:25.19 | mdave | so that the context can accept menu selections from the caller |
| 03:25.25 | J_- | do a sip debug ip bvip |
| 03:25.34 | Math[laptop] | mdave: _X. => goto s|1; |
| 03:25.45 | Qwell | s doesn't mean "let the user type shit" |
| 03:25.46 | mdave | Math[laptop], alright, i'll try that |
| 03:25.49 | Math[laptop] | uhm thats in ael |
| 03:25.53 | J_- | If the Request-URI does not identify an address that the UAS is willing to accept requests for, it SHOULD reject the request with a 404 (Not Found) response |
| 03:25.58 | mdave | ok, then waht DOES mean let the caller make a selection? |
| 03:26.09 | Math[laptop] | exten => _X.,1,Goto(s|1); <-- use that for extensions.conf |
| 03:26.10 | Qwell | mdave: reaching the dialplan |
| 03:26.16 | mdave | Math[laptop], ok, lemme try that |
| 03:26.16 | Qwell | make it do whatever you like |
| 03:26.59 | *** join/#asterisk ms345 (n=mike_sim@dsl027-163-193.atl1.dsl.speakeasy.net) |
| 03:28.07 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 03:28.20 | mdave | Math[laptop], it went to s, did the one thing, but then didnt accept any further digits.. how can I 'throw away' the digits bv is sending, and wait for a selection by the caller *after* * has picked up? |
| 03:28.40 | mdave | how can I send the call to a specific context, but *without* any extension |
| 03:28.45 | Qwell | mdave: You don't NEED to do that |
| 03:28.52 | Qwell | Just make your dialplan accept digits |
| 03:29.21 | mdave | yes, by putting *waht* in it? it appears to be accepting the digits of my number from bv, and not waiting for anything else |
| 03:29.39 | Qwell | I don't know...it's your dialplan |
| 03:29.55 | Qwell | What do you want it to do? |
| 03:30.10 | J_- | Qwell: I rather do the dial plans in agi so I dont have to type the exten=> it gets a bit repetitive :) |
| 03:30.10 | mdave | how do you create a dialplan, that after accepting digits sent by a voip provider, answers the call, and then waits for dialing from the user |
| 03:30.15 | mdave | thats what I want it to do |
| 03:30.17 | J_- | err s/agi/ael |
| 03:30.44 | Qwell | mdave: There are several ways. Background, WaitExten, etc |
| 03:30.53 | *** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 03:31.02 | mdave | it would be really nice if there was an example somewhere |
| 03:31.08 | mdave | i tried waitexten, it didnt work |
| 03:31.09 | Qwell | You mean like on the wiki? |
| 03:31.11 | Qwell | ~wikis |
| 03:31.12 | jbot | wikis is, like, http://www.voip-info.org |
| 03:31.23 | Qwell | like the 20+ examples of each? |
| 03:31.42 | mdave | do you by any chance have a url directly to some examples? becuase i havent found anything that actually gives a complete example for how to make that work |
| 03:31.59 | Qwell | .google asterisk waitexten site:voip-info.org |
| 03:32.49 | Math[laptop] | would be fun to have something pop-up google automaticly on the user's screen |
| 03:33.04 | mdave | that seems to be an example for how to cope with users dialing slowly |
| 03:33.11 | mdave | not for how to create a dialable menu |
| 03:33.14 | mdave | and id already see it |
| 03:33.17 | Qwell | mdave: Do you need somebody to hold your hand? |
| 03:33.30 | mdave | no, but id like to see an example that actually works |
| 03:33.43 | Qwell | The one on the wiki works |
| 03:33.53 | mdave | i cant imagine that setting up a dialable menu isnt a pretty damn common thing that there might be a good idea to have a working example of |
| 03:34.02 | mdave | sure, it probably works for 'how to handle users dialing too slowly' |
| 03:34.06 | mdave | but thats now what im trying to do |
| 03:34.10 | Qwell | Have you tried it? |
| 03:35.10 | mdave | yes, I added a 'waitexten' to my plan, and it didnt work |
| 03:35.17 | De_Mon | hrm... asterisk isn't producing a core. ulimit -c is set to 102400 and I'm running "asterisk -U asterisk -g" as root |
| 03:35.27 | mdave | i ignored all the parts of the example that have to do with 'waiting for slow dialing users', since that isnt what im trying to do |
| 03:35.29 | Qwell | mdave: okay, and what did you do with it? |
| 03:35.37 | Qwell | mdave: give me the link |
| 03:35.43 | *** part/#asterisk kb1_kanobe (n=jsmith@h24-207-96-51.cst.dccnet.com) |
| 03:35.51 | *** join/#asterisk ivanfm[wrk] (n=ivanfm@cumulus.saisp.br) |
| 03:37.54 | mdave | that also seems to be about dialing the digits they entered to a channel, which also isnt want I want.. I want a menu.. 1 for this, 2 for that.. etc |
| 03:38.10 | Qwell | What do you think a menu is? |
| 03:38.16 | Qwell | It's "digits dialed in a channel" |
| 03:38.22 | _-Jon-_ | znoG, you still around? |
| 03:38.32 | mdave | I want exten 1,1,dothis(), exten 2,1,dothat().. where the digits they dial are used to decide which line to use |
| 03:38.41 | Qwell | okay, so do that |
| 03:38.57 | dudes | this is rocket science damnit |
| 03:39.02 | mdave | btw Qwell need not answer, since your answers are useless im ignoring you |
| 03:39.09 | Qwell | yay |
| 03:39.13 | Qwell | next? |
| 03:39.14 | ms345 | hahaha! |
| 03:39.43 | _-Jon-_ | mdave, you're trying to make 1 + number dial out through one line, and 2 + number dial out another? |
| 03:39.55 | *** join/#asterisk Cresl1n (n=matt@146.229.187.7) |
| 03:40.00 | mdave | i need to enter a context, *after* receiving a call, without any digits having been accepting, |
| 03:40.11 | mdave | or play different greetings, dialing different destinations |
| 03:40.18 | tronix | AGI script. can set context, do dialling, etc |
| 03:40.18 | dudes | then goto(context,s,1) |
| 03:40.21 | mdave | or execute any number of different commands |
| 03:40.29 | mdave | yeah, it did the s stuff and terminated |
| 03:40.33 | Qwell | or use WaitExten |
| 03:40.34 | mdave | never waiting for any additional entries |
| 03:40.36 | Qwell | or background |
| 03:40.42 | Qwell | oh, right, ignore |
| 03:40.46 | dudes | s,1,Background(fucking-message||s) |
| 03:40.51 | *** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net) |
| 03:41.02 | Qwell | "but it isn't going to s! *whine*whine*whine*" |
| 03:41.07 | mdave | there is no prompt yet.. i just want to get the dialing part working |
| 03:41.16 | jijgeh | Anyone know how I can override telecom company CID when using DISA? |
| 03:41.30 | dudes | goto(context,${EXTEN},1) |
| 03:41.57 | Qwell | dudes: I have a feeling that even if your wrote the dialplan for him...it wouldn't work |
| 03:42.06 | danitza | hello |
| 03:42.12 | asteriskmonkey | can i use the DBPut command to put variables to any database? |
| 03:42.14 | znoG | _-Jon-_: yep, how'd you go? |
| 03:42.22 | danitza | I have a problem configuring a TDM2406 |
| 03:42.25 | mdave | yes, but i dont know the etension yet until it gos there and gets it |
| 03:42.25 | Qwell | asteriskmonkey: no, that's for the astdb |
| 03:42.27 | dudes | Qwell - Are you saying I'm stupid? |
| 03:42.28 | dudes | heh |
| 03:42.31 | mdave | it hasnt been dialed |
| 03:42.34 | Qwell | dudes: No, I'm saying he's stupid. :) |
| 03:42.39 | jijgeh | When I dial out, Asterisk sets CID to "my CID" <Mynumber>, but it still shows old CID from CO |
| 03:42.41 | dudes | haha |
| 03:42.44 | _-Jon-_ | znoG, reset the Sipura and it works like a charm now :P |
| 03:43.02 | dudes | jijgeh - that's probably cause they don't let you set your callerid |
| 03:43.04 | asteriskmonkey | Qwell: i want to be able to dump so variables to a serperate database.. any ideas on how to do that :P ? use a sytem command? |
| 03:43.14 | danitza | could anybody help me? |
| 03:43.16 | Qwell | asteriskmonkey: What version of *? |
| 03:43.24 | asteriskmonkey | Qwell: 1.2x |
| 03:43.28 | jijgeh | Do all CO block setting own CID? |
| 03:43.33 | Qwell | You could probably backport func_odbc |
| 03:43.42 | Qwell | jijgeh: name can't usually be set |
| 03:43.43 | dudes | jijgeh - no |
| 03:43.51 | *** join/#asterisk _cleric_ (n=dacleric@84.245.174.181) |
| 03:44.10 | dudes | You can send Name to some WIreless phones |
| 03:44.21 | asteriskmonkey | Qwell : gah , dont want to have odbc installed.. proply just a system command directly connect sql might be better |
| 03:44.28 | dudes | But I've never ran across a teleco that allows you to spoof ANI |
| 03:44.36 | Qwell | asteriskmonkey: try the MYSQL function. It's kinda junk, but... |
| 03:44.43 | jijgeh | if it is sent through VOIP over Internet, then better luck probably then? |
| 03:44.48 | Qwell | jijgeh: no |
| 03:44.54 | Qwell | cid name is looked up on the remote side |
| 03:44.56 | J_- | dudes: on some business ones they will let you |
| 03:45.06 | asteriskmonkey | Qwell : there is a mysql function ? cool .. whats it called so i can look it up :) |
| 03:45.10 | jijgeh | It is going to be for business, just doing testing at home |
| 03:45.10 | Qwell | MYSQL |
| 03:45.12 | J_- | sometimes pri's too |
| 03:45.35 | dudes | I've never used a PRI that let you send ANI |
| 03:45.39 | jijgeh | So if I had PRI, maybe be able to set CID then, depending on Telco |
| 03:45.40 | Nugget | mysql is ucky. |
| 03:45.48 | jijgeh | oh |
| 03:45.51 | Qwell | Nugget: write a func_postgresql |
| 03:45.51 | jijgeh | hmm.... |
| 03:45.58 | dudes | What's wrong with MySQL |
| 03:46.11 | Qwell | Nugget: or, for you, func_bsdisbetterthanlinux_postgresql :P |
| 03:46.28 | jijgeh | Trying to set up asterisk for my business, but didn't want to *(&&)* up the business phone system, so am working on testing at home |
| 03:46.34 | austinnichols10 | jijgeh: I have my disa set up so that I dial in with my mobile, system sees me and dials back, gives me dialtone and set outbound CID to another number. CID Name comes from the telco. |
| 03:46.43 | Nugget | mysql is the windows me of databases. |
| 03:47.04 | Nugget | it looks like a real system, but underneath it's an ill-designed and totally wrongheaded mess |
| 03:47.23 | mzo | omg nugget ;p |
| 03:47.27 | rob0 | No, Windows ME was never popular :) |
| 03:47.36 | Nugget | and there's no shortage of mouthbreathing fans who don't know any better, think it's the bees' knees, and rate it two thumbs up because they've never used anythng else. |
| 03:47.49 | jijgeh | austinnichols10: so what does your final CID on the reciever's end look like finally? |
| 03:47.53 | *** join/#asterisk iq (n=iq@71-214-5-158.omah.qwest.net) |
| 03:48.06 | mzo | nugget, make my zyxel phone suck less? :P |
| 03:48.13 | Nugget | man, I wish I could. :) |
| 03:48.20 | austinnichols10 | jijgeh: I'm using the CID for my office phone (versus my mobile) |
| 03:48.59 | jijgeh | So it ends up being the Office phone's CID... That is what I am after: make call from cell phone while on road, but have work CID |
| 03:49.23 | austinnichols10 | jijgeh: so when the PSTN bill hits at the end of the month the calls show up under my did |
| 03:49.49 | jijgeh | for your business? |
| 03:49.52 | mzo | it sits on the charger the entire week, and i use it like once for a minute tops before it annoys me too much. |
| 03:49.58 | austinnichols10 | jijgeh: correct |
| 03:50.04 | [TK]D-Fender | rob0 : Have you the latest from M$? They are taking all the most popular features from Windows CE, Windows ME, and Widows NT, and creating CEMENT :D |
| 03:50.15 | [av]bani | \o/ |
| 03:50.41 | J_- | yea I have not heard a windows machines shooting down missles lately :) |
| 03:50.42 | [av]bani | [TK]D-Fender: http://bani.anime.net/xp.jpg |
| 03:50.48 | [av]bani | actual ad sent to me in the mail by m$ |
| 03:50.48 | jijgeh | thanks for answering my q! |
| 03:50.50 | asteriskmonkey | Qwell do you have to use \ everwhere there is suppost to be a space? |
| 03:51.52 | rob0 | haha yes I heard that one |
| 03:51.52 | austinnichols10 | jijgeh: check your pm |
| 03:53.01 | jijgeh | sorry, don't understand pm |
| 03:53.21 | austinnichols10 | jijgeh: you should have another message window (private message) |
| 03:54.03 | jijgeh | no, i don't... at least I don't see it readily |
| 03:54.30 | austinnichols10 | what irc client you using? |
| 03:54.52 | J_- | austinnichols10: just do a ctcp version |
| 03:54.59 | *** join/#asterisk techie (i=gus@antibala.com) |
| 03:55.05 | jijgeh | X-Chat |
| 03:55.10 | jijgeh | v2.0.4 |
| 03:56.02 | Qwell | austinnichols10: Are you registered? |
| 03:56.22 | austinnichols10 | should be |
| 03:56.31 | De_Mon | :| asterisk deb is compiled without debug symbols |
| 03:56.35 | De_Mon | oh well |
| 03:56.36 | austinnichols10 | nope |
| 03:56.44 | *** join/#asterisk pauly (n=pauly@84.133.233.220.exetel.com.au) |
| 03:56.45 | Qwell | austinnichols10: Then you can't send msgs |
| 03:57.04 | pauly | Anyone know of a hardware phone which supports IAX2 ? |
| 03:57.06 | austinnichols10 | k - sounds like I should register. tks |
| 03:57.17 | pauly | I prefer not to use adapters with analog phones |
| 03:57.18 | austinnichols10 | jijgeh: http://pastebin.ca/39551 |
| 03:57.29 | Qwell | pauldy: Why IAX2? |
| 03:57.41 | pauly | I prefer to use it over SIP |
| 03:57.49 | Qwell | why? |
| 03:58.08 | pauly | Less overhead seems to work better behind nat etc |
| 03:58.51 | Qwell | I don't think there is less overhead |
| 03:59.20 | pauly | I find there is plus my phone server is on a net4801 and when I have a few IAX connections it seems to use less CPU power |
| 04:00.39 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 04:04.51 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 04:06.36 | [av]bani | soekris :) |
| 04:07.44 | De_Mon | well, it's not much but http://pastebin.com/534882 if the user hangs up while in queue or the queue timeout is reached 1/8 times asterisk crashes. If I time it just right, I've managed to crash it 1/3 times... |
| 04:08.05 | De_Mon | (manual hangup) |
| 04:09.32 | znoG | is there any way to know if distinctive ring detection has been enabled as per zapata.conf? |
| 04:10.17 | znoG | apart from asterisk -vvvvvvvvc, are there any other ways to get maximum debug output from Asterisk? |
| 04:10.55 | Qwell | znoG: turn on debug in logger.conf |
| 04:13.27 | *** join/#asterisk db48x (n=db48x@66.25.39.5) |
| 04:14.23 | *** part/#asterisk db48x (n=db48x@66.25.39.5) |
| 04:15.11 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 04:15.24 | *** part/#asterisk twisted[mobile] (n=twisted@24.96.140.230) |
| 04:17.18 | znoG | Qwell: thanks, still didn't see what I was hoping to see. There is stuff like this in chan_zap.c: |
| 04:17.51 | znoG | <PROTECTED> |
| 04:17.51 | znoG | <PROTECTED> |
| 04:18.03 | znoG | i don't see that in the logs |
| 04:18.06 | Qwell | znoG: set debug 10 |
| 04:18.12 | PMantis | I have an older wctdm device, what do I need to modify in zaptel before compiling? |
| 04:20.14 | znoG | Qwell: wow, out of pure luck, that was the first time Asterisk displayed: Feb 2 01:19:42 NOTICE[7010]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... |
| 04:20.58 | znoG | Qwell: that's usually a sign that distinctive ring detection is going to work |
| 04:20.58 | znoG | Qwell: too bad the next 50 rings don't do that |
| 04:21.15 | austinnichols10 | can an spa-3000 do anything with distinctive ring from the PSTN? |
| 04:24.31 | znoG | Qwell: do you know what exactly is ZT_IOMUX_SIGEVENT ? |
| 04:24.41 | Qwell | no |
| 04:25.33 | znoG | i give up on this distinctivering stuff for now |
| 04:25.43 | znoG | my guess is that i need to do the detection after the CID comes in |
| 04:26.02 | znoG | but, not having played with Asterisk code, makes it a little hard |
| 04:28.25 | PMantis | I have two wctdm cards in my server. only one is recognized after upgrading. I forget what I need to uncomment in a .c file before compiling. My card IIRC, has a problem with identifying itself to the system. |
| 04:28.34 | asteriskmonkey | if i have the mysql command it should be i see it in the show functions command? |
| 04:29.15 | *** join/#asterisk YARICK (n=spiderma@pool-71-255-198-81.bltmmd.east.verizon.net) |
| 04:31.23 | JunK-Y | show functions like foo . |
| 04:32.18 | asteriskmonkey | just wanted to see if i had the mysql command instaleld |
| 04:32.35 | PMantis | I think what I need is called the REV H fix, or something like that |
| 04:32.49 | troyb | you can always check by putting mysqld in /dev/null :) |
| 04:33.18 | asteriskmonkey | really... /me throws troyb in /dev/null |
| 04:33.49 | asteriskmonkey | think ill stick with the lovly system escape command :P tied in with an awk/sed script 4tw |
| 04:34.08 | JunK-Y | myysql command? u mean mysql applications? |
| 04:34.18 | *** join/#asterisk gopherspidey (n=spidey@12-216-165-134.client.mchsi.com) |
| 04:34.24 | troyb | asr__ :) |
| 04:34.25 | PMantis | ??? ZT_CHANCONFIG failed on channel 5: No such device or address (6) |
| 04:34.28 | Qwell | function |
| 04:34.29 | troyb | err asteriskmonkey :P |
| 04:35.09 | asteriskmonkey | asterisk apperently has a mysql command you can use in your extensions.conf :P its nasty with all the \ to escape everything |
| 04:35.25 | troyb | ack |
| 04:35.26 | Qwell | asteriskmonkey: use func_odbc |
| 04:35.32 | gopherspidey | Does any know where I can find the polycom configuartion zipfile. They meantion it in the Admin guide, but I can not find it. |
| 04:35.37 | troyb | Qwell you must live here? |
| 04:35.41 | dudes | the MySQL addon isn't that bad |
| 04:35.50 | Qwell | troyb: something like that |
| 04:35.56 | JunK-Y | gopherspidey: www.voip-info.org has a lot of infos. |
| 04:36.02 | file | !!!!!!!!!!!!!! |
| 04:36.24 | JunK-Y | file: outch, baby ur hurting me! |
| 04:36.38 | troyb | Qwell what does a room cost? |
| 04:36.40 | gopherspidey | JunK-Y, I have been there. I just would like to start from scratch. |
| 04:36.49 | Qwell | tronix: your soul... |
| 04:37.08 | troyb | tronix soul or my soul :P |
| 04:37.08 | asteriskmonkey | Qwell: func_odbc requries and odbc database connection though right? |
| 04:37.35 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:37.36 | JunK-Y | gopherspidey: try it, u'll be surprise. |
| 04:37.53 | asteriskmonkey | dudes: i can live witht he mysql command in asterisk .. im just trying to find out if i have it installed though |
| 04:38.00 | asteriskmonkey | its not showing up in show functions |
| 04:38.06 | dudes | you need asterisk-addons |
| 04:38.19 | asteriskmonkey | yes i installed addons a while ago |
| 04:38.50 | dudes | Then, it should be in unless your compiled failed cause it couldn't find something (ie, mysql_config) |
| 04:39.08 | asteriskmonkey | i can just make clean && make install the addons right i dont need to recompile asterisk? |
| 04:39.12 | dudes | The compile doesn't fail it just ignore installing it |
| 04:39.28 | asteriskmonkey | ah |
| 04:39.37 | asteriskmonkey | ill do a find see if i see it anywhere first |
| 04:41.21 | gopherspidey | JunK-Y, I have and it works find, but I would like to learn it. Not just copy it |
| 04:41.54 | JunK-Y | admin guide is like 200 pages, user guide, almost the same thing, have fun :) |
| 04:42.30 | gopherspidey | JunK-Y, I am |
| 04:42.49 | gopherspidey | JunK-Y, I love the phone. |
| 04:42.55 | JunK-Y | 501s? |
| 04:43.23 | [TK]D-Fender | I like my 600 here :) |
| 04:43.24 | asteriskmonkey | grrr.. cant get the app_addon_sql_mysql.so to load |
| 04:43.28 | file | yay IP600 |
| 04:43.31 | asteriskmonkey | guess im missing config stuff soemwhere |
| 04:44.05 | Qwell | troyb: whichever |
| 04:45.11 | znoG | Qwell: what do you suggest I can do to grab the attention of * developers? to discuss a distinctive ring problem |
| 04:45.23 | znoG | i tried to email the list twice about this, but didn't get a reply. |
| 04:45.28 | znoG | twice over a period of time, of course |
| 04:49.50 | robbie2 | hey znoG |
| 04:50.02 | robbie2 | znoG: you in .au atm ? |
| 04:50.12 | robbie2 | there is good asterisk consultant in melbourne |
| 04:51.29 | gopherspidey | JunK-Y, 601 |
| 04:51.50 | znoG | robbie2: nope |
| 04:53.13 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 05:03.42 | asteriskmonkey | why does asterisk keep turning my , in |'s |
| 05:03.55 | asteriskmonkey | im trying to do a system command and its anoying as heck |
| 05:04.14 | mdave | is the template * uses for sending an email about voicemail editable? if so what file is it in? |
| 05:04.19 | *** join/#asterisk korptix (i=akiru@d58-104-239-208.dsl.nsw.optusnet.com.au) |
| 05:04.24 | *** part/#asterisk korptix (i=akiru@d58-104-239-208.dsl.nsw.optusnet.com.au) |
| 05:04.34 | mdave | nvm |
| 05:04.39 | mdave | its right in voicemail.conf |
| 05:06.40 | mdave | reload |
| 05:07.31 | robbie2 | when i try to dial out through my TE110P i get |
| 05:07.33 | robbie2 | -- Channel 0/1, span 1 got hangup request |
| 05:07.42 | robbie2 | -- Hungup 'Zap/1-1' |
| 05:07.42 | robbie2 | <PROTECTED> |
| 05:07.47 | robbie2 | i can get incoming calls ok |
| 05:07.50 | *** join/#asterisk joonysam (n=a@brom-245-243.flexabit.net) |
| 05:07.54 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 05:08.13 | joonysam | simple question, what is the conference number for a 2002 extension in asterisk@home? |
| 05:08.23 | joonysam | I tried adding 8000 to make 10002, but that didn't work |
| 05:11.28 | danitza | hello |
| 05:11.49 | danitza | maybe anybody could help me |
| 05:12.43 | *** join/#asterisk SwK (n=SwK@12-219-147-107.client.mchsi.com) |
| 05:15.03 | brookshire[home] | maybe |
| 05:15.35 | danitza | I have a CPU with a TDM2406 and a TDM2402 boards |
| 05:15.54 | danitza | running CentOS 4.2 |
| 05:16.14 | brookshire[home] | k |
| 05:16.39 | danitza | everytime the machine restarted the tdm configuration los |
| 05:16.54 | brookshire[home] | intresting |
| 05:16.58 | brookshire[home] | have you called digium support? |
| 05:17.00 | file[laptop] | Matttttttt |
| 05:17.01 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 05:17.04 | *** join/#asterisk websae (i=websae@CPE-24-167-204-30.wi.res.rr.com) |
| 05:17.08 | danitza | yeah |
| 05:17.27 | websae | anyone know of a good E911 provider, I am having a hard time finding one where I don't have to invest $10,000 for a startup cost |
| 05:17.28 | danitza | first they suggested to change the motherboard and I did |
| 05:18.14 | danitza | now my cpu has a Intel 865 GVSL |
| 05:18.18 | *** join/#asterisk _mistral (i=mistral@jstevenson.plus.com) |
| 05:19.42 | asteriskmonkey | i cant get app_mysql going can anyone help |
| 05:19.43 | brookshire[home] | but one card keeps the configuration? |
| 05:19.53 | slan | Is Yum a part of CentOS or Asterisk? |
| 05:19.55 | brookshire[home] | so like if you only configure one, does it keep the configuration? |
| 05:19.55 | websae | any suggestions for e911 providers anyone? |
| 05:20.11 | brookshire[home] | yum is part of centos |
| 05:20.18 | danitza | brookshire: No, they lost it completly |
| 05:20.33 | brookshire[home] | danitza: even with only one card? |
| 05:20.43 | slan | Thanks. Too bad, I want to use another distro and wonder how many things in Asterisk depend on Yum |
| 05:20.58 | danitza | I have to reload the modules manually |
| 05:21.10 | brookshire[home] | slan: asterisk isn't dependent on yum |
| 05:21.26 | brookshire[home] | yum is a package management tool |
| 05:21.32 | danitza | yum is an utility of RPM based systems |
| 05:21.59 | brookshire[home] | instead of yum, there is apt-get on debian |
| 05:22.00 | danitza | as apt-get for DEbian |
| 05:22.05 | brookshire[home] | and emerge on gentoo |
| 05:22.15 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 05:22.40 | *** join/#asterisk flush (n=flush@ip216-239-68-39.vif.net) |
| 05:22.47 | flush | yo |
| 05:22.53 | slan | brookshire: Problem is I want to use Slackware which has no package mgt - they use tgz files. |
| 05:23.06 | brookshire[home] | slan: well just download the tar.gz from asterisk.org |
| 05:23.07 | danitza | brookshire: In this CPU I configured a TE110P successfully |
| 05:23.35 | slan | brookshire: Are you saying that any updates I need from Asterisk will be available in tgz form? |
| 05:23.46 | danitza | but the TDM2406B config is a nightmare |
| 05:24.07 | brookshire[home] | slan: no.. you'll have to manually update them with slackware |
| 05:24.17 | brookshire[home] | until slackware decides to maintain the project |
| 05:24.27 | slan | brookshire: How would I manually update? |
| 05:24.35 | flush | i got asterisk installed, but i can only start it as root |
| 05:24.39 | brookshire[home] | slan: you download the source and complie |
| 05:24.41 | brookshire[home] | compile |
| 05:24.46 | flush | why that |
| 05:24.54 | flush | flush@h4xb0x:~$ /usr/sbin/asterisk |
| 05:24.54 | flush | Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 05:25.12 | slan | brookshire: OK compile it is. Frequently necessary with Slackware. |
| 05:25.52 | websae | anyone know of a good e911 provider? |
| 05:25.56 | slan | brookshire: Although I like Slack maybe I should use CentOS but I'm not sure how to get it without all the X and extra stuff. |
| 05:26.00 | brookshire[home] | flush: you have to make certian files read/write for whatever user you want to run asterisk as |
| 05:26.37 | danitza | I hate TDM2400!!!!!!! |
| 05:26.38 | flush | the fact is that the /var/run/asterisk.pid file doesnt exist at all |
| 05:26.44 | slan | brookshire: I mean get a stripped down CLI version of CentOS |
| 05:26.54 | brookshire[home] | slan: try gentoo :D |
| 05:27.01 | brookshire[home] | they have a package for asterisk |
| 05:27.02 | brookshire[home] | :D |
| 05:27.04 | flush | how can i not have permissions to something that doesnt exist |
| 05:27.06 | websae | SwK: are you there? |
| 05:27.13 | [TK]D-Fender | slan : Just compile from source on slack... its dead easy stuff.... |
| 05:27.31 | brookshire[home] | flush: then add read/write to /var/run/ |
| 05:27.36 | brookshire[home] | for maybe a group |
| 05:27.39 | danitza | has anybody experienced IRQ conflicts? |
| 05:27.39 | brookshire[home] | or something :) |
| 05:27.40 | slan | [TK]D-Fender: I wonder if the BSD inits in Slack would cause problems. |
| 05:27.41 | [TK]D-Fender | slan : All my *nix boxes that I set up are running it... |
| 05:27.54 | brookshire[home] | or! run in chroot |
| 05:27.56 | [TK]D-Fender | slan : Just shove it in rc.local and you'll be fine |
| 05:27.57 | brookshire[home] | problem solved |
| 05:28.18 | slan | [TK]D-Fender: Happy to hear from another Slacker. Do you run * on Slack? |
| 05:28.28 | [TK]D-Fender | slan : Several servers |
| 05:28.32 | [TK]D-Fender | Including my home one |
| 05:28.47 | slan | [TK]D-Fender: Yes but are they the Asterisk servers? |
| 05:29.16 | slan | [TK]D-Fender: If so, which version? 10.2? |
| 05:29.56 | [TK]D-Fender | slan : That was implicit in my answer and the fact I'm HERE :) Yes I run * on multiple Slackware servers. just DL the source,compile and away you go. trouble free. |
| 05:30.03 | danitza | I don't know how Asterisk could run in production environment for a real application |
| 05:30.28 | slan | [TK]D-Fender: That gives me a lot of hope because I'm very comfortable with Slack. |
| 05:30.29 | [TK]D-Fender | I've been using Slackware exclusvely since 9.0. My servers are either 10. ro 10.2 updated to -current |
| 05:31.25 | brookshire[home] | heh.. slackware was one of the first distros of linux i ever used |
| 05:31.35 | websae | anyone know anything about e911 at all? |
| 05:31.37 | slan | [TK]D-Fender: I'm running 10.1 for my main home machine right now but going to 10.2 soon. A little problem with Win4lin patches but maybe it's time to dump w9x entirely. |
| 05:31.52 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 05:32.15 | slan | brookshire: Heck it was one of the first distros available. |
| 05:32.48 | [TK]D-Fender | slan : I Run WinXP on my primary PC but Slackware on everything else (my home server does X, KDE, hooked to my 52" HDTV, *, SAMBA, NAT gateway, etc) |
| 05:32.53 | asteriskmonkey | anyone have experience with bash script? |
| 05:33.48 | slan | [TK]D-Fender: Pretty much the same with me. Win98SE for occasional use but everything gets controlled from the Slack box, KDE, Samba and ssh. |
| 05:34.17 | [TK]D-Fender | <PROTECTED> |
| 05:34.36 | brookshire[home] | slan: it was the first, lol |
| 05:35.00 | brookshire[home] | i started out with it 11 years ago |
| 05:35.06 | slan | [TK]D-Fender: We have an active Asterisk UG here in Austin and an installfest on the 20th. I'm going to bring a Slack box and let the experts have their way with it. |
| 05:35.07 | YARICK | probably because of some silly game or other software |
| 05:35.12 | YARICK | that is dependant on win98SE |
| 05:35.14 | asteriskmonkey | why does asterisk change my ,'s to |'s |
| 05:35.15 | brookshire[home] | and ran it up until i started college, heh |
| 05:35.38 | asteriskmonkey | im trying to do a system sql command and it keeps changeing the commas on me :P |
| 05:35.42 | YARICK | otherwise, LIKE a real ninja he would be running GENTOO LINUX !!! |
| 05:35.43 | [TK]D-Fender | slan : You are concerned about the difficulty of installing * on it? |
| 05:35.54 | YARICK | 98SE is for pussies |
| 05:36.12 | slan | brookshire: 11 years! Long time ago. I _think_ I started with a 7 something version but that was only about 4-5 years ago. |
| 05:36.56 | brookshire[home] | hehe.. it ran something like kernel 1.0.99 or something |
| 05:36.57 | slan | [TK]D-Fender: Yes I was concerned, mainly because of the difference in init structures. |
| 05:37.00 | websae | if anyone needs unlimited incoming with DID for $8/month let me know!!! can do LNP as well, just $8/month for DID and FREE INCOMING! |
| 05:37.16 | brookshire[home] | and my window manager was afterstep |
| 05:37.20 | [TK]D-Fender | slan : NO issue at all then. Just shove it anywhere in rc.local and you'll be fine. |
| 05:37.36 | [TK]D-Fender | * is the most brain-dean simple job on Slack... |
| 05:37.40 | [TK]D-Fender | dead* |
| 05:37.42 | brookshire[home] | i remember when kde came out in beta |
| 05:37.46 | asteriskmonkey | if anyone needs unlimited incoming with DID for 5$/month let me know!!! |
| 05:37.49 | brookshire[home] | and i had to compile to source to get it to run |
| 05:37.50 | brookshire[home] | lol |
| 05:37.55 | brookshire[home] | on a 486! |
| 05:38.01 | slan | [TK]D-Fender: That's great news for me. I have lots in rc.local now <g> |
| 05:38.18 | brookshire[home] | actually.. it was probably more like a pentium 133 |
| 05:38.18 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 05:38.27 | [TK]D-Fender | slan : So get you hands off your nuts and seize the day! |
| 05:38.49 | slan | brookshire: When I compile from source, the messages flash by so fast I wonder if something went wrong. |
| 05:39.16 | brookshire[home] | haha |
| 05:39.24 | brookshire[home] | it's not so fast on a pentium |
| 05:39.30 | *** part/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com) |
| 05:39.42 | brookshire[home] | i think it took 3 or 4 hours to compile kde |
| 05:39.42 | slan | [TK]D-Fender: It won't happen tonight. I'm preparing a hard disk - wipe an old W9x, test the disk, then I'll do a minimal Slack install. |
| 05:39.42 | [TK]D-Fender | slan : typically Slack installs ALL the proper devel stuff so unless you manually selected all your pacgages one by one Yyou're almost assured a trouble-free install |
| 05:40.58 | slan | [TK]D-Fender: I really don't like to use anything but Slack. Started with some other distros but once tried Slack it just seemed to fit. I like editing config files <g> |
| 05:41.31 | slan | [TK]D-Fender: and I abolutely hate gui configs that don't let me know what's going on. |
| 05:41.57 | brookshire[home] | heh.. i remember writting my ppp config file for my 28.8 modem for slack.. |
| 05:42.02 | brookshire[home] | it tooke me a week to get working |
| 05:42.10 | brookshire[home] | s/tooke/took |
| 05:42.12 | brookshire[home] | lol |
| 05:42.28 | slan | brookshire: Well that's a little easier with KPPP. |
| 05:42.49 | brookshire[home] | slan: kde didn't exist back then |
| 05:42.52 | brookshire[home] | lol |
| 05:43.10 | Qwell | This from the guy who said he doesn't like gui configs |
| 05:43.11 | [TK]D-Fender | For some reason Slack just seemed easier for me too coming from a strong DOS background. The fact is had MC allowed me to become useful almost immediately knowing next to nothing :) |
| 05:43.20 | brookshire[home] | plus.. there was no standard for auth of ppp |
| 05:43.33 | brookshire[home] | so you had to script around whatever script your isp made to authenticate you |
| 05:44.01 | slan | [TK]D-Fender: I've been writing 4dos batch files for many years - shell scripts were easy and better after that. |
| 05:44.21 | slan | brookshire: The bad old days. |
| 05:44.23 | brookshire[home] | but then windows 95 came out.. and made a standard |
| 05:44.47 | [TK]D-Fender | slan : GUI configs aren't always that bad. For the install of most Linux Distro's its not different than the text based ones like Slack/Debian. |
| 05:45.09 | slan | brookshire: I live for the day when I'm Windows free. All except my ham radio programs that can't be replaced easily in Linux. |
| 05:45.22 | brookshire[home] | [TK]D-Fender: i think kppp is the only gui config i've ever liked.. actually ;) |
| 05:45.23 | [TK]D-Fender | brookshire : Makes you want to run back in time and give a big hug to Trumpet Winsock doesn't it? :! |
| 05:45.29 | *** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 05:45.48 | propagandhi | hi |
| 05:45.50 | techie | haha, podcasting & asterisk, how cool |
| 05:45.58 | brookshire[home] | [TK]D-Fender: oh yeah.. thank god for trumpet 16! |
| 05:46.03 | [TK]D-Fender | brookshire : I've only done RPPPOE and striaght DHCP on mine... never had to touch dial-up. |
| 05:46.04 | propagandhi | can somebody give me some insight on a caller id problem |
| 05:46.11 | brookshire[home] | big improvement over the 8bit version |
| 05:46.11 | brookshire[home] | lol |
| 05:46.26 | [TK]D-Fender | Yeah.. this shit is way better than our old crap! |
| 05:46.47 | slan | [TK]D-Fender: Dialup is handy when I want to test my system from a foreign IP. |
| 05:46.58 | brookshire[home] | speaking of the old days |
| 05:47.08 | brookshire[home] | does anyone remember the first mp3 they ever got?? |
| 05:47.30 | [TK]D-Fender | brookshire : Ok... no more horror stories about the swooping pteradactyls ok?!?! |
| 05:47.55 | slan | [TK]D-Fender: Nice spelling. |
| 05:48.13 | brookshire[home] | hehe |
| 05:48.14 | [TK]D-Fender | slan : Did I get it right? Did it fro memory.... |
| 05:48.24 | slan | [TK]D-Fender: I believe so. |
| 05:48.25 | [TK]D-Fender | (no comment on my TYPING) |
| 05:48.25 | justinu | pterodactykl, iirc |
| 05:48.26 | techie | brookshire[home]: I think it was a file that said 'oh hell, I hate this shit' |
| 05:48.48 | [TK]D-Fender | justinu : Was that 2... or *3* "q"'s in "hello" ;) |
| 05:48.50 | brookshire[home] | mine was cake - the distance |
| 05:49.01 | justinu | lol |
| 05:49.05 | brookshire[home] | hopefully it wasn't the lama |
| 05:49.07 | brookshire[home] | llama |
| 05:49.13 | *** join/#asterisk NayNay (n=someone@c-24-10-162-48.hsd1.ut.comcast.net) |
| 05:49.14 | justinu | [TK]D-Fender: you asked :P |
| 05:49.41 | justinu | i'm trying to debug openser and it's NOT cooperating |
| 05:50.06 | slan | brookshire: What about the swooping pterodactyls? |
| 05:50.19 | brookshire[home] | slan: they died off |
| 05:50.23 | brookshire[home] | :( |
| 05:50.28 | justinu | they got too big and fat, and they all died |
| 05:50.36 | brookshire[home] | it was a comet |
| 05:50.47 | justinu | striker striker striker! |
| 05:50.53 | justinu | ted striker! |
| 05:51.04 | Nugget | it's a damn good think you don't know how much he hates your guts. |
| 05:51.18 | websae | anyone familiar with e911? any providers here for that? |
| 05:51.35 | justinu | well, it's very pretty... it's big and it's white.... it looks like a big tylenol!! |
| 05:51.47 | Nugget | ok boys, let's get some pictures. |
| 05:52.03 | justinu | i need to buy that |
| 05:52.07 | Nugget | yes |
| 05:52.10 | justinu | classic cinema |
| 05:52.15 | Nugget | unparalleled. |
| 05:53.32 | asteriskmonkey | ok so no one knows bash scripting or why asterisk turns commas in to pipes when using system command |
| 05:53.57 | Nugget | don't script with bash. either write it for sh or use a real language. |
| 05:54.01 | brookshire[home] | asteriskmonkey: did you try using a \' |
| 05:54.04 | justinu | asteriskmonkey: do you like gladiator movies? |
| 05:54.16 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 05:54.30 | asteriskmonkey | justinu: ? where did that come from lol |
| 05:54.39 | justinu | airplane, the movie |
| 05:54.55 | asteriskmonkey | ah playing on tv somewhere/ |
| 05:55.12 | websae | $8/month DID and UNLIMITED INCOMING...that's pretty good right |
| 05:55.21 | justinu | oh knock if off with your white red zone shit again tom... we know what this is really about... you want me to get an abortion! |
| 05:55.34 | justinu | no, it just popped into my head as brilliant for some strange reason |
| 05:55.54 | justinu | cuz they were talking bout dinosaurs dying off |
| 05:55.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:56.53 | slan | [TK]D-Fender: Do you think Slack and Asterisk will live without crowding on a 4 gig disk? Want to put that oldie back into service. |
| 05:56.56 | Nugget | don't pirate it. just buy the damn thing. |
| 05:56.56 | justinu | lol |
| 05:56.59 | Nugget | it's entertainment. |
| 05:57.05 | justinu | yeah, it's worth the money |
| 05:57.09 | justinu | hours and hours of laughs |
| 05:57.47 | Nugget | I mean, jeez, the dvd is like 12.99 |
| 05:57.48 | slan | [TK]D-Fender: No X or emacs of course <g> |
| 05:57.53 | dokhench | it's not like i'm actually going to watch the whole thing.. i'll just scan bits here and there, then in 15 mins i'll be sick of it. |
| 05:57.56 | asteriskmonkey | brookshire[home]: yep put so many \'s in there and still | |
| 05:58.00 | asteriskmonkey | so what the heck |
| 06:00.27 | justinu | this looks like a future hobby for me: |
| 06:00.27 | justinu | http://www.msnbc.msn.com/id/11101951/ |
| 06:01.11 | asteriskmonkey | gah! |
| 06:01.23 | asteriskmonkey | i reloaded configs and it too away all the /'s grrrr |
| 06:01.28 | brookshire[home] | did you try using a " instead of ' |
| 06:01.38 | asteriskmonkey | yes |
| 06:02.13 | brookshire[home] | is this in bash agi? |
| 06:02.54 | asteriskmonkey | no this is exten => yadda,1,System ( echo insert into yadaa... |
| 06:03.11 | brookshire[home] | bah |
| 06:03.23 | brookshire[home] | try doing it in bash agi ;) |
| 06:04.08 | asteriskmonkey | i dont know how :( |
| 06:04.39 | brookshire[home] | http://www.google.com/search?q=bash+agi |
| 06:04.39 | brookshire[home] | :D |
| 06:05.10 | asteriskmonkey | im trying to do something simple like this echo "INSERT INTO logtable (val1,val2) VALUES ('${DATETIME}','${CALLERIDNUM}');" | mysql -uusername -ppassword -ddatabase |
| 06:05.16 | brookshire[home] | yeah |
| 06:05.20 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 06:05.28 | brookshire[home] | you can just pass the two vars to agi |
| 06:05.39 | brookshire[home] | and it can run and do that for you |
| 06:06.49 | asteriskmonkey | yes i tried to make one earlier with no success :( |
| 06:06.50 | *** join/#asterisk jfincher (n=jfincher@cpe-66-68-179-146.austin.res.rr.com) |
| 06:06.51 | *** join/#asterisk mattwj2006 (n=Matt@dialup-4.159.62.222.Dial1.Chicago1.Level3.net) |
| 06:07.34 | brookshire[home] | asteriskmonkey: another good read |
| 06:07.35 | brookshire[home] | http://forums.digium.com/viewtopic.php?t=3148 |
| 06:08.04 | *** part/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 06:09.03 | asteriskmonkey | have you dont php agis? |
| 06:09.29 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 06:09.49 | brookshire[home] | there is phpagi yes! |
| 06:09.59 | mattwj2006 | any ideas on how to set up an extension that uses a shoutcast feed? |
| 06:10.04 | brookshire[home] | i hear it's not as advanced as perl agi though |
| 06:10.13 | watchy | mattwj2006: what would be the point |
| 06:10.28 | brookshire[home] | watchy: music on hold |
| 06:10.28 | asteriskmonkey | i wrote a simple php agi for stuffing calls but nothing to be called from asterisk itself |
| 06:10.36 | watchy | hrm |
| 06:10.38 | brookshire[home] | mattwj2006: it can be done |
| 06:10.41 | brookshire[home] | use the wiki ;) |
| 06:10.49 | mattwj2006 | lol |
| 06:10.51 | watchy | i tried to get shoutcast to work for music on hold |
| 06:10.58 | watchy | but i couldnt get it to work. |
| 06:11.00 | mattwj2006 | voip-info.org ? |
| 06:11.06 | brookshire[home] | yeah |
| 06:11.15 | mattwj2006 | go figure |
| 06:11.18 | mattwj2006 | lol |
| 06:11.29 | mattwj2006 | thanks |
| 06:11.52 | watchy | boyz in the hood is on |
| 06:11.56 | watchy | im white and i love that flick |
| 06:12.41 | newl | that's because you can't jump. |
| 06:12.52 | watchy | haha |
| 06:13.53 | watchy | the internet is for porn |
| 06:14.57 | Assid | is there any difference between 'im going online' and 'im watching porn' ? |
| 06:15.57 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 06:17.01 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 06:17.07 | brookshire[home] | matt! |
| 06:18.21 | watchy | nope |
| 06:19.02 | mattwj2006 | yes brook |
| 06:19.31 | brookshire[home] | well.. cresl1n |
| 06:19.37 | brookshire[home] | but hey back! |
| 06:19.45 | brookshire[home] | i mean.. hey too |
| 06:19.46 | Cresl1n | hey! |
| 06:19.48 | Cresl1n | :-) |
| 06:19.52 | brookshire[home] | what's up matt? |
| 06:19.53 | Cresl1n | brookshire[home]: !!! |
| 06:20.00 | Cresl1n | not much, just about asleep |
| 06:20.06 | Cresl1n | what're you up to? |
| 06:20.07 | brookshire[home] | lame! |
| 06:20.08 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 06:20.13 | mattwj2006 | lame |
| 06:20.14 | mattwj2006 | hmmm |
| 06:20.26 | brookshire[home] | just chatting, lol |
| 06:20.31 | mattwj2006 | lame ain't mp3 encoder? |
| 06:20.35 | mattwj2006 | lol |
| 06:20.59 | tuxinator_linux | Cresl1n, you should have been sleeping a while ago |
| 06:20.59 | mattwj2006 | I am a bum right now....so I am trying to learn something new |
| 06:21.08 | mattwj2006 | I picked Asterisk |
| 06:21.11 | Cresl1n | heh |
| 06:21.20 | Cresl1n | just a little distracted |
| 06:22.07 | tuxinator_linux | porn is bad |
| 06:22.27 | tuxinator_linux | waste all your time, keeps you from sleeping |
| 06:22.36 | tuxinator_linux | and you end up with nothing |
| 06:22.50 | mattwj2006 | I agree tux |
| 06:22.50 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 06:23.12 | *** join/#asterisk bloch (n=lol@c-67-171-8-154.hsd1.wa.comcast.net) |
| 06:23.17 | mattwj2006 | on the other hand, the world can be quite lonely |
| 06:23.34 | tuxinator_linux | I've been married to a very good girl for 4 years |
| 06:23.50 | tuxinator_linux | don't need no stinkin porn |
| 06:23.59 | tuxinator_linux | ;-) |
| 06:24.06 | mattwj2006 | lol |
| 06:24.16 | mattwj2006 | I had my first real gf |
| 06:24.22 | mattwj2006 | 1 year and goodbye |
| 06:24.42 | mattwj2006 | oh well at least I got a college degree and an associate degree |
| 06:25.00 | tuxinator_linux | that's you're problem, too busy with school |
| 06:25.02 | asteriskmonkey | almost got it .. |
| 06:25.22 | asteriskmonkey | i need to use the ; but asterisk keeps choppign eveythign off after it int he System command |
| 06:25.26 | asteriskmonkey | ... |
| 06:25.27 | mattwj2006 | yeah tux, your probably right |
| 06:25.53 | tuxinator_linux | flashnet[BNC], BNC? |
| 06:26.01 | tuxinator_linux | What's BNC? |
| 06:26.13 | flashnet[BNC] | Bouncer |
| 06:26.36 | tuxinator_linux | ah, story behind that? |
| 06:26.36 | flashnet[BNC] | im using a psybouncer |
| 06:26.58 | flashnet[BNC] | i dont understan ur questiom |
| 06:27.08 | tuxinator_linux | no problem |
| 06:27.14 | flashnet[BNC] | ok bye |
| 06:28.20 | tuxinator_linux | I just googled psybouncer, I learned something new today, thanks |
| 06:28.36 | dokhench | tuxinator_linux, i did the same. :) |
| 06:29.12 | asteriskmonkey | ok need to know how to be able to use a ; in a command line without asterisk eating it :) |
| 06:29.21 | dokhench | i was like psybouncer... whazzat? still not sure what it is good for.. bitchx and screen accomplish the same thing it looks like. |
| 06:30.17 | tuxinator_linux | asteriskmonkey, hungry little * isn't it? |
| 06:31.27 | asteriskmonkey | yes .. id say its almost a bug |
| 06:31.40 | asteriskmonkey | you should be able to use ; inside of System() |
| 06:31.51 | tuxinator_linux | Just installed ubuntu on this laptop (my wifes), seems to be working well |
| 06:31.59 | asteriskmonkey | ive tried escapign it with a \ no love there |
| 06:32.18 | tuxinator_linux | asteriskmonkey, read the source code for that part? |
| 06:32.36 | tuxinator_linux | I haven't looked at it, but there may be a clue |
| 06:32.41 | asteriskmonkey | ? |
| 06:32.43 | tuxinator_linux | or a patch to be written |
| 06:32.50 | asteriskmonkey | ah .. |
| 06:33.04 | asteriskmonkey | was looking for a more immiadte patch :P lol |
| 06:33.36 | tuxinator_linux | trying to do multiple system calls? |
| 06:34.21 | asteriskmonkey | trying to do an syste sql insert statement piped to mysql but the ; keeps chopping of my system command before the pipe to mysql |
| 06:34.46 | tuxinator_linux | hmm |
| 06:36.12 | asteriskmonkey | trying to do System (echo insert into yadda (field) value ('yadda') ; | mysql ... |
| 06:36.18 | asteriskmonkey | you get the idea |
| 06:36.18 | tuxinator_linux | asteriskmonkey, is it recognising it as a comment, after the semi? |
| 06:36.30 | asteriskmonkey | yes |
| 06:36.40 | asteriskmonkey | its stupid and frustrating |
| 06:36.49 | asteriskmonkey | ive tried to escape it using a / to no aveal |
| 06:38.31 | *** join/#asterisk alphaque (n=alphaque@218.111.26.246) |
| 06:38.45 | tuxinator_linux | let me check a few places |
| 06:39.00 | asteriskmonkey | thanks ive googled with no good refs :P |
| 06:40.25 | [av]bani | :> |
| 06:40.33 | tuxinator_linux | I found out that they didn't use # (as the commenter) as it would most like be used in the dialplan |
| 06:41.35 | mattwj2006 | I know this might be a dumb question, but how do you get googled? |
| 06:41.36 | mattwj2006 | :P |
| 06:42.03 | asteriskmonkey | damn... ; is required to complete and sql command too |
| 06:42.59 | tuxinator_linux | asteriskmonkey, is using the system() the best way to do what you want to do? |
| 06:43.18 | tuxinator_linux | is there not another command where you don't have to pass a semi? |
| 06:43.28 | tuxinator_linux | I'm still learning, so I don't know |
| 06:43.38 | asteriskmonkey | tuxinator_linux: not that i know of.. i have tried to get app_mysql installed with no luch |
| 06:44.25 | Qwell | MYSQL, it's a function |
| 06:44.30 | Qwell | comes with asterisk-addons, no doubt |
| 06:44.35 | asteriskmonkey | Qwell: i know ive tried installing it |
| 06:44.36 | tuxinator_linux | what about inserting a charage using....ummm.... something like does for HTML |
| 06:44.43 | asteriskmonkey | Qwell i would love to use it |
| 06:44.47 | asteriskmonkey | i just cant get it loaded |
| 06:44.53 | Qwell | then just use func_odbc ;P |
| 06:45.11 | tuxinator_linux | Qwell, Aren't you suppose to be sleeping |
| 06:45.12 | asteriskmonkey | i know nothing of odbc |
| 06:45.25 | tuxinator_linux | Qwell, finally, someone with answers ;-) |
| 06:45.30 | tuxinator_linux | ODBC is not hard |
| 06:45.40 | asteriskmonkey | do i need to setup odbc |
| 06:45.40 | eyz | asteriskmoney, I just skimmed over stuff from a few minutes ago.. escape characters aren't coming up? |
| 06:45.49 | eyz | did you try echo -e ? |
| 06:45.53 | Qwell | asteriskmonkey: yeah, but it takes like 5 minutes |
| 06:46.06 | eyz | sorry if I'm not on-track with what you're doing, but it seems like perhaps the answer |
| 06:46.22 | Qwell | any real distro will set it up for you too |
| 06:46.26 | asteriskmonkey | no.. issue is asterisk sees the ; and craps out thinking its a comment |
| 06:46.32 | eyz | ok |
| 06:46.52 | asteriskmonkey | qwell id rather someone help me get the mysql module loaded |
| 06:47.47 | asteriskmonkey | ah it is loaded |
| 06:47.50 | asteriskmonkey | app_addon_sql_mysql.so |
| 06:47.53 | asteriskmonkey | thats the one right? |
| 06:47.58 | tuxinator_linux | asteriskmonkey, mysql module would be the route I would take, but I am not know for taking the easy route |
| 06:49.46 | tuxinator_linux | asteriskmonkey, I assume you found http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
| 06:50.52 | asteriskmonkey | yes |
| 06:51.00 | asteriskmonkey | no worries i got basic mysql working now |
| 06:51.04 | tuxinator_linux | looks like they persistant connections needs to be added to that package |
| 06:51.05 | asteriskmonkey | wasnt workign before |
| 06:51.23 | asteriskmonkey | damn this has been a frustrating experience |
| 06:51.48 | tuxinator_linux | I wish my fingers would type what I was thinking, some stuff sure types out funny |
| 06:51.58 | asteriskmonkey | lol |
| 06:52.01 | *** join/#asterisk DataBookBoy (n=datacomp@217.8.236.1) |
| 06:52.09 | DataBookBoy | Hello, people! |
| 06:52.15 | tuxinator_linux | hey DataBookBoy |
| 06:52.34 | tuxinator_linux | err... hey DataCompBoy |
| 06:52.41 | DataCompBoy | :) hello |
| 06:52.53 | DataCompBoy | have someone played with asterisk AMI interface? |
| 06:52.58 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:53.05 | DataCompBoy | I'm getting on any Originate request error. |
| 06:53.14 | DataCompBoy | My phone rings, i'm pickup |
| 06:53.28 | DataCompBoy | And get "Loop detected" and break. |
| 06:53.40 | DataCompBoy | if I call by myself -- everything going fine |
| 06:53.51 | *** join/#asterisk ChaiNX (n=ChaiNX@212.44.194.4) |
| 06:54.14 | tuxinator_linux | DataCompBoy, whish you luck, but I need to get back to work |
| 06:54.39 | DataCompBoy | at "asterisk -RTvvvvvvvdddddddddddd" shows no errors nor information lines :( |
| 06:54.59 | watchy | hug me |
| 06:56.23 | *** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
| 06:56.46 | DataCompBoy | have tried also in way |
| 06:56.47 | DataCompBoy | <PROTECTED> |
| 06:56.47 | DataCompBoy | <PROTECTED> |
| 06:56.49 | DataCompBoy | same result |
| 06:57.19 | DataCompBoy | nothing hear after line pickup, and no reaction at asterisk... |
| 06:58.12 | mattwj2006 | I found something great....anyone want to see it? |
| 06:58.27 | tuxinator_linux | hide the wife and children |
| 06:58.51 | Krill | does anyone have ooh323 working with an avaya system and can they possible send their ooh323.conf - I'm having DTMF issues |
| 06:58.56 | mattwj2006 | http://lame.sourceforge.net/download/riaa.html |
| 06:59.32 | DataCompBoy | hee hee |
| 06:59.36 | DataCompBoy | communism forever! |
| 06:59.45 | mattwj2006 | lol |
| 07:00.20 | DataCompBoy | so, nobody tried to originate?! |
| 07:00.30 | *** part/#asterisk ChaiNX (n=ChaiNX@212.44.194.4) |
| 07:01.24 | *** join/#asterisk pengyong (n=lala@218.93.158.116) |
| 07:01.48 | pengyong | any one use gmail to receive asterisk maillist? |
| 07:02.01 | pengyong | i can't get emails from gmail pop3 server |
| 07:02.29 | pengyong | when i logined into gmail web mail, all mails are in inbox |
| 07:02.39 | DataCompBoy | gmail sux |
| 07:02.39 | Krill | any ideas/help re: ooh323 & avaya ? |
| 07:02.51 | DataCompBoy | best way is own mailserver |
| 07:03.02 | DataCompBoy | Krill: ooh323 works itself? |
| 07:03.25 | Krill | DataCompBoy, yes |
| 07:03.30 | Krill | but i cannot get DTMF to pass through |
| 07:03.33 | DataCompBoy | if you have problems with DTMF, first try is switch DFTM to inband. |
| 07:03.43 | Krill | tried inband :( |
| 07:03.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:03.54 | DataCompBoy | without luck?! |
| 07:03.55 | DataCompBoy | :/ hmm |
| 07:04.01 | Krill | yep |
| 07:04.13 | DataCompBoy | what softphone you using before asterisk? |
| 07:04.20 | DataCompBoy | i mean phone -> asterisk -> avaya |
| 07:04.49 | Krill | at the moment im just connectinv avaya pabx's to the asterisk as a convergence point |
| 07:04.59 | Math[laptop] | what kind of dtmf signalling |
| 07:05.12 | DataCompBoy | err |
| 07:05.19 | DataCompBoy | describe what are you really doing |
| 07:05.33 | Krill | i've tried a few different types, currently using dtmfmode=h245signal |
| 07:06.16 | Krill | I want to link the avaya's to asterisk to route calls via an IAX2 wholesaler for cheaper calls |
| 07:06.44 | Krill | i can get the demo greeting no problems, but it doesnt register any prompts when i try to use it |
| 07:07.42 | mattwj2006 | go source code go |
| 07:08.42 | Krill | ultimately i'm not sure what exact settings i need on both the avaya & asterisk end points to make it work properly |
| 07:12.06 | *** join/#asterisk Data (n=datacomp@217.8.236.1) |
| 07:12.23 | DataBookBoy | err |
| 07:12.50 | DataBookBoy | so, what can I do to check what problem is -- AMI or asterisk? |
| 07:13.03 | mattwj2006 | avaya is an IP phone? |
| 07:13.25 | Krill | PABX with IP phones hanging off it |
| 07:13.26 | DataBookBoy | agh |
| 07:13.26 | Math[laptop] | avaya is a company |
| 07:13.49 | Krill | so in my situtation no IP phones are directly connecting to Asterisk, they are going via the Avaya unit |
| 07:14.08 | mattwj2006 | okay |
| 07:14.10 | DataBookBoy | have any dtmf configuration avaya have itself? |
| 07:14.25 | Krill | yes |
| 07:14.40 | *** join/#asterisk jgomata (n=jgomata@red-corp-201.143.78.76.telnor.net) |
| 07:14.41 | Krill | just out of band setting |
| 07:14.43 | DataBookBoy | any DTMF configuration here? |
| 07:19.30 | DataBookBoy | ... umm same problem, if I trying .call file |
| 07:19.39 | DataBookBoy | i'm got at asterisk console: |
| 07:20.02 | DataBookBoy | "Attempting call on SIP/4100 for 600@demo:1 (retry 1)" |
| 07:20.07 | DataBookBoy | my xten rings |
| 07:20.15 | DataBookBoy | i'm pickup line |
| 07:20.19 | DataBookBoy | and hear nothing |
| 07:20.35 | *** join/#asterisk alk (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 07:20.38 | DataBookBoy | and no lines like executing playing demo- message. |
| 07:20.39 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 07:20.56 | DataBookBoy | what I need to check? |
| 07:21.25 | [av]bani | DataBookBoy: nat |
| 07:21.42 | DataBookBoy | i have enabled RTP DEBUG |
| 07:21.46 | DataBookBoy | and see packets when I |
| 07:21.49 | DataBookBoy | and see packets when I'm call |
| 07:21.56 | DataBookBoy | but see nothing on incomming call |
| 07:21.57 | DataBookBoy | |
| 07:22.06 | DataBookBoy | hmm... what kind of NAT ? |
| 07:22.21 | [av]bani | enable nat, disable invite |
| 07:22.31 | DataCompBoy | reinvite disabled |
| 07:22.31 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 07:22.31 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) [NETSPLIT VICTIM] |
| 07:22.32 | *** join/#asterisk Cazper (n=cazper@c5100A229.sdsl.catch.no) [NETSPLIT VICTIM] |
| 07:22.32 | *** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM] |
| 07:22.33 | [av]bani | nat=yes |
| 07:22.33 | [av]bani | canreinvite=no |
| 07:22.34 | DataCompBoy | asterisk in vmware console |
| 07:22.52 | [av]bani | vmware? all bets are off :D |
| 07:22.53 | DataCompBoy | so we are really near :) |
| 07:23.09 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-235-036.pools.arcor-ip.net) |
| 07:23.11 | DataCompBoy | but there problem -- XTen won't bind onto vmware interface |
| 07:23.27 | DataCompBoy | but all packets routed excellent |
| 07:23.36 | DataCompBoy | I can ping from vmware any host interface |
| 07:23.46 | Math[laptop] | asterisk in vmware.... |
| 07:23.48 | Math[laptop] | that wont sound good |
| 07:24.03 | DataCompBoy | i'm don't need sound quality :) i'm just playing with AMI |
| 07:24.04 | Math[laptop] | I had that before... u'll get choppy sound on playback because of the architecture |
| 07:24.27 | DataCompBoy | for real application there real server :D |
| 07:24.50 | Math[laptop] | I'd just get a dev server but... it can do in the meanwhile |
| 07:25.08 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 07:25.41 | DataCompBoy | quality doesn't mean anything now |
| 07:25.46 | DataCompBoy | i just need to make calls work... |
| 07:26.19 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:26.19 | *** join/#asterisk asr (i=asr@pimpbox.latency.net) |
| 07:26.21 | Math[laptop] | why don't u bridge the interfaces? |
| 07:26.33 | Math[laptop] | or create a host-only network with a different addressing scheme |
| 07:26.42 | DataCompBoy | that is what done |
| 07:26.55 | DataCompBoy | but xten binds onto external iface :( |
| 07:27.19 | Math[laptop] | you can control that |
| 07:27.39 | Math[laptop] | it should bind on 0.0.0.0 or you can override the reported ip in its settings |
| 07:28.11 | DataCompBoy | I have tried |
| 07:28.26 | DataCompBoy | at diagnostic log it write |
| 07:28.27 | DataCompBoy | SIP: 172.24.1.61:5060 |
| 07:28.28 | DataCompBoy | RTP: 172.24.1.61:8000 |
| 07:28.29 | DataCompBoy | NAT: 217.8.236.1 |
| 07:28.34 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 07:29.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:30.32 | *** join/#asterisk _grey_ (n=grey@host54-106.bol.co.tz) |
| 07:30.39 | *** join/#asterisk EriSan (n=erisan@151.8.109.118) |
| 07:31.00 | DataCompBoy | oooo |
| 07:31.04 | DataCompBoy | nat=yes |
| 07:31.09 | DataCompBoy | canreinvite=no |
| 07:31.17 | DataCompBoy | uncommented local networks |
| 07:31.25 | DataCompBoy | change net settings at xten |
| 07:31.31 | DataCompBoy | and looks like it start to work! |
| 07:31.57 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 07:32.25 | DataCompBoy | thanks [av]bani Math[laptop] :)) |
| 07:33.46 | *** part/#asterisk jgomata (n=jgomata@red-corp-201.143.78.76.telnor.net) |
| 07:35.01 | infobox | hell all |
| 07:35.05 | infobox | hello all |
| 07:35.06 | infobox | sorry |
| 07:35.13 | DataCompBoy | olleh |
| 07:35.50 | infobox | have anybody tested the TDM240X boards? |
| 07:36.22 | DataCompBoy | i think, digium :D |
| 07:37.40 | infobox | hahaha |
| 07:37.53 | infobox | I bought two |
| 07:38.23 | infobox | in order to build a gateway with an ISDN card |
| 07:39.47 | infobox | I wonder if anyone has did it before |
| 07:40.11 | infobox | maybe some DIgium technical guy... |
| 07:40.48 | infobox | I have an article from MArk Spencer recommended those boards to small companies |
| 07:41.08 | infobox | but they don't work how I expected |
| 07:43.54 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 07:44.16 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:44.36 | mattwj2006 | my music on hold sounds weird?! |
| 07:46.30 | mattwj2006 | It sounds like I am watching Star Wars if you know what I mean |
| 07:47.26 | DataCompBoy | Yoda: the master of Forth language? |
| 07:47.39 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 07:48.18 | mattwj2006 | I feel like the Sith have just invaded |
| 07:48.19 | mattwj2006 | :P |
| 07:48.45 | mattwj2006 | what I was getting at is my mp3 sounds like Dark Vader |
| 07:56.03 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-231.claranet.co.uk) |
| 07:56.10 | *** join/#asterisk angler (n=angler@gateway.digium.com) |
| 07:58.26 | mattwj2006 | yeah it is 2 am here |
| 07:58.33 | mattwj2006 | time for bed....goodnight everyone |
| 07:58.46 | DataCompBoy | 2 pm here |
| 07:58.50 | DataCompBoy | good day! |
| 07:59.35 | DataCompBoy | good luck, man from past |
| 07:59.37 | DataCompBoy | -1 |
| 07:59.38 | *** part/#asterisk DataCompBoy (n=datacomp@217.8.236.1) |
| 07:59.53 | *** join/#asterisk JunK-Y_ (n=junky@Toronto-HSE-ppp3742231.sympatico.ca) |
| 08:00.47 | mattwj2006 | just wondering..... |
| 08:00.53 | mattwj2006 | what day you got data? |
| 08:02.04 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 08:02.32 | mattwj2006 | Today is Thursday |
| 08:02.35 | mattwj2006 | here |
| 08:03.03 | mattwj2006 | nm....I am going to sleep bye |
| 08:03.08 | *** part/#asterisk mattwj2006 (n=Matt@dialup-4.159.62.222.Dial1.Chicago1.Level3.net) |
| 08:06.36 | asteriskmonkey | slut ho bag wooo !!!!! |
| 08:06.57 | asteriskmonkey | I now know why it was eating my /'s |
| 08:06.57 | *** join/#asterisk JunK-Y__ (n=junky@Toronto-HSE-ppp3742560.sympatico.ca) |
| 08:07.16 | asteriskmonkey | damn AA@HOME WAS SCOFFING MY /'s |
| 08:07.24 | asteriskmonkey | dont use the webgui |
| 08:07.33 | asteriskmonkey | always use vi instead :P |
| 08:08.40 | thazza | ~amp |
| 08:08.42 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 08:09.50 | iDunno | maybe that should be added to the /topic |
| 08:12.10 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:16.37 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 08:16.42 | jozsab1 | Hy al |
| 08:17.24 | asteriskmonkey | yes DEATH TO AMP |
| 08:17.45 | asteriskmonkey | ok of to go get 1.2 hrs sleep before work now hehehe |
| 08:17.47 | jozsab1 | Is it a known bug that you can not turn off RFC2833 (use inband instead) on asterisk 1.2.1 ?? |
| 08:18.10 | asteriskmonkey | jozsab1 move to 1.2.4 |
| 08:18.27 | asteriskmonkey | there was all sorts of whack errors in 1.2x before it |
| 08:18.52 | jozsab1 | thanks <asteriskmonkey> |
| 08:19.33 | jozsab1 | I have problems using g729 codec. Is there a no comercial version (downloadable) wich works for sure ? |
| 08:19.59 | *** part/#asterisk xeet2 (n=xeet3@bwi1-br1-gig2-1.jsci.net) |
| 08:21.00 | *** join/#asterisk moshe (n=moshe@212.29.200.10) |
| 08:22.06 | moshe | Hi all. I have some burning issue. This morning, my phone company called me and said that my pbx is unreachable. Checking with zttool, I see that it said the card is OK, so I told them "it's your end". Meanwhile, I arrived on site, and my Digium T110P has NO LED lit on it. Either flashing red, or green. Do I assume right that the card is dead? |
| 08:23.12 | *** join/#asterisk DataCompBoy (n=datacomp@217.8.236.1) |
| 08:23.29 | DataCompBoy | umm... one more question about AMI |
| 08:23.33 | DataCompBoy | strange events |
| 08:23.54 | DataCompBoy | sent newchannel when new channel created, with status: down |
| 08:24.09 | DataCompBoy | then, when it going to ring, sent event "newchannel" ! |
| 08:24.19 | DataCompBoy | when it connected sent "newstatus, status up". |
| 08:24.58 | DataCompBoy | but why "newchannel, status ringing", not "newstate" ?! |
| 08:28.09 | *** join/#asterisk SuperX (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za) |
| 08:28.52 | moshe | Hi all. I have some burning issue. This morning, my phone company called me and said that my pbx is unreachable. Checking with zttool, I see that it said the card is OK, so I told them "it's your end". Meanwhile, I arrived on site, and my Digium T110P has NO LED lit on it. Either flashing red, or green. Do I assume right that the card is dead? |
| 08:33.18 | tuxinator_linux | moshe, make sure you check the cable also, and maybe try reloading the driver? |
| 08:33.47 | tuxinator_linux | moshe, sorry, can't help you any more, too sleepy, off to bed for me |
| 08:33.54 | moshe | tried everything... |
| 08:34.13 | tuxinator_linux | restarted the machine also? |
| 08:34.32 | moshe | even took the card to a different slot and a different computer |
| 08:34.53 | DataCompBoy | if you tried everything, and nothing helps, you should tried anything that you have not tried. |
| 08:34.56 | tuxinator_linux | oh, well, is there anyway you can verify that the line is good? |
| 08:35.02 | moshe | the weird thing is that the drive loads, I see the card in /etc/zaptel/1 and... |
| 08:35.02 | DataCompBoy | for example: replace card |
| 08:35.06 | *** join/#asterisk OnuR (n=emineee@62.220.217.169) |
| 08:35.11 | moshe | I don't have another card, that's the problem |
| 08:35.19 | moshe | I came here wandering maybe it is a known problem |
| 08:35.37 | DataCompBoy | that looks like hardware problem |
| 08:35.41 | tuxinator_linux | do you have a surge protector on the line? |
| 08:35.44 | moshe | the phone company sent a technician with hopefully a PRI slave emulator... |
| 08:35.48 | moshe | the machine sits on UPS |
| 08:35.56 | moshe | (on-line) |
| 08:35.58 | tuxinator_linux | what about the PRI line? |
| 08:36.17 | moshe | the phone company managed to get to the media gateway on the other side of the wire |
| 08:36.21 | tuxinator_linux | is there surge protection on it? |
| 08:36.25 | moshe | and they said they see the link is down |
| 08:36.35 | moshe | that equipment is UPS powered too |
| 08:36.56 | tuxinator_linux | no, surges can come over the T1 line |
| 08:36.57 | moshe | and the PRI itself arrives on fiber optic |
| 08:37.02 | tuxinator_linux | oh, okay |
| 08:37.08 | *** join/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 08:37.17 | *** part/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 08:37.29 | *** join/#asterisk Abbas__ (n=Abbas@203.81.196.140) |
| 08:37.30 | moshe | do you know digium's policy on replacements of broken hardware? |
| 08:38.01 | moshe | or do I need to send the card to my reseller (which is not in my country) and then wait two weeks or I don't know how much for replacement... |
| 08:38.46 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.230.177.telnor.net) |
| 08:39.06 | tuxinator_linux | positive your media gateway is functional? |
| 08:39.28 | tuxinator_linux | how old is the card? |
| 08:39.33 | moshe | about 4 months |
| 08:40.12 | moshe | after the phone company will be here he'll test with a pri client emulator the media gateway... but I have a good reason to believe that it is working [they logged in to it remotely successfully] |
| 08:40.34 | tuxinator_linux | well, you zt'd successfully also |
| 08:40.43 | tuxinator_linux | so, it's still possible |
| 08:41.00 | moshe | BUT, the red led should be going on and off when the cable is disconnected |
| 08:41.01 | tuxinator_linux | Call digium in a few hours |
| 08:41.07 | tuxinator_linux | http://www.digium.com/index.php?menu=return_policy |
| 08:41.46 | tuxinator_linux | looks like they will cross-ship, so you can get a card quickly |
| 08:43.01 | moshe | problem is that company doesn't have phones now :\ |
| 08:43.05 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 08:44.26 | [av]bani | should keep spares :() |
| 08:44.43 | moshe | Iif it didn't cost so much, I would have :) |
| 08:44.49 | [av]bani | always have backup for your critical systems |
| 08:45.04 | tuxinator_linux | moshe, I guess all you can do is budget in extra spare equipment. With any phone system, stuff can fail at any time. Even my toshiba phone system died after a few years of use. |
| 08:45.25 | moshe | few years is more acceptible than 3 :) |
| 08:45.29 | moshe | [months] |
| 08:45.35 | Mw3 | moshe: this failure might cost you more than a spare card |
| 08:45.37 | [av]bani | ask the phone co to call forward your DIDs to some landlines |
| 08:45.48 | tuxinator_linux | or a cell phone |
| 08:45.49 | [av]bani | yes, you have to figure the cost of downtime into a spare |
| 08:46.11 | moshe | yeah, I know that. tell it to management... :> |
| 08:46.23 | [av]bani | its easier if you show them a balance sheet |
| 08:46.24 | *** join/#asterisk asr__ (i=asr@pimpbox.latency.net) |
| 08:46.32 | [av]bani | cost of spare: $$$ cost of downtime: $$$$$$$ |
| 08:46.37 | moshe | heh :) |
| 08:47.18 | tuxinator_linux | moshe, I wish you luck, good night |
| 08:47.37 | [av]bani | israel? you might be abel to call forward your DIDs to an ITSP who can give you incoming SIP trunks |
| 08:47.45 | [av]bani | assuming your internet connection doesnt suck |
| 08:47.50 | Mw3 | downtime has a negative impact on your company's image too |
| 08:48.02 | moshe | we got nothing here in israel |
| 08:48.08 | moshe | we waited 3 months for the PRI |
| 08:48.11 | [av]bani | :() |
| 08:49.11 | [av]bani | you might be able to find asterisk consultant in israel who has hardware |
| 08:49.16 | [av]bani | http://www.voip-info.org/wiki-Asterisk+consultants+EMEA |
| 08:51.12 | [av]bani | http://www.voice.co.il/eng/contact.aspx?count=voipinfo they sell israel DIDs |
| 08:51.15 | [av]bani | so there you go :) |
| 08:51.16 | litage | if you direct CDRs to be sent to the Manager interface, how do you grab each CDR as it comes in? |
| 08:51.16 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 08:51.20 | mazzanet | roar. |
| 08:51.33 | mazzanet | Feb 2 19:51:00 NOTICE[31376]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 203.161.128.253, request 's@from-sip' does not exist |
| 08:51.45 | mazzanet | i'm getting this message when i get incoming calls from one of my iax trunks |
| 08:52.27 | [av]bani | http://www.zitsp.com/row/ claims to have israel DIDs |
| 08:53.49 | moshe | thanks for all the help so far, guys |
| 08:53.55 | [av]bani | good luck! |
| 08:54.37 | mazzanet | i can't seem to see where the "s@" is coming |
| 08:54.49 | mazzanet | ...from |
| 09:00.27 | mazzanet | i'm getting |
| 09:00.28 | mazzanet | Feb 2 20:00:04 NOTICE[31376]: chan_iax2.c:6775 socket_read: Rejected connect attempt from 192.246.69.186, who was trying to reach '[snip number]@' |
| 09:00.32 | mazzanet | from another iax trunk too |
| 09:04.07 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
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| 09:19.50 | mazzanet | CAUSE : No authority found |
| 09:19.51 | mazzanet | hmm |
| 09:19.54 | mazzanet | what does that mean? |
| 09:20.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:23.48 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-235-036.pools.arcor-ip.net) |
| 09:24.01 | Makenshi | yay, i got my itad assignment from iana this morning |
| 09:24.11 | Makenshi | first one in the uk :o) |
| 09:24.17 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:28.19 | mazzanet | ah ha |
| 09:28.27 | mazzanet | i fixed one of my trunks |
| 09:28.58 | mazzanet | it seems the section header name in iax.conf actually matters |
| 09:29.33 | *** join/#asterisk sim (n=simon@stp25-1-82-226-150-87.fbx.proxad.net) |
| 09:29.40 | sim | hop |
| 09:30.02 | *** join/#asterisk Pegger (n=peg@pool-68-163-180-64.bos.east.verizon.net) |
| 09:30.16 | Pegger | anyone here using t1 with there asterisk server? |
| 09:30.30 | mazzanet | what speed is T1? |
| 09:30.48 | moshe | 1.544 |
| 09:30.59 | Pegger | and 24 channels |
| 09:31.10 | mazzanet | well i have 1.5mbit adsl here |
| 09:31.18 | Pegger | or 23 voice and one singaling |
| 09:31.26 | Pegger | mazzanet, very diffrent |
| 09:32.39 | knobo | Im trying to configure ringgroups with odbc, but I'm having some truble. I have followd the different guides from www.voip-info.org. Then I tryed to put some dummy data into the table, but when i do a "show queues" it does not show anything |
| 09:32.47 | knobo | any way to debug this? |
| 09:33.05 | knobo | No queues.CLI> |
| 09:33.47 | knobo | Maybe it's the dummy data that phuks it up? |
| 09:33.48 | Pegger | knobo, i am still trying to get real time working |
| 09:34.10 | knobo | Pegger: how far are you? |
| 09:34.45 | Pegger | well the other night I was getting compile errors for zaptel |
| 09:35.25 | knobo | Pegger: that does not have so much with realtime config. |
| 09:35.39 | Pegger | i know but you need it as part of the install |
| 09:36.02 | Pegger | and if i can not install asterisk with real time comiled in that it is useless |
| 09:36.07 | knobo | but, you havent lookt at the realtime part yet? |
| 09:36.34 | knobo | Pegger: which version are you compiling? |
| 09:36.37 | Pegger | a little not enough to be usefulll |
| 09:37.12 | Pegger | 1.2 |
| 09:37.56 | Pegger | ok on the real time stuff it is supos to be in the asterisk-addons but when i look at the README all i see is libpri stuff |
| 09:38.56 | *** join/#asterisk okkum (n=okkum@62-43-137-94.user.ono.com) |
| 09:42.41 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 09:43.28 | okkum | Hi? |
| 09:43.31 | *** join/#asterisk svenna_ (n=svenna@p548D2087.dip0.t-ipconnect.de) |
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| 09:46.18 | *** join/#asterisk okkum (n=okkum@62-43-137-94.user.ono.com) |
| 09:46.57 | knobo | Pegger: I'm a litle bit unshure what you mean about realtime <-> asteriks-addons |
| 09:47.44 | knobo | I thought it was in by default. |
| 09:48.34 | oej | Pegger: realtime is in the standard asterisk distribution |
| 09:48.44 | oej | Pegger: The realtime MySQL driver is in the addons |
| 09:49.17 | knobo | anyway, is there some way to debug the odbc connection |
| 09:49.42 | knobo | to se if it is trying to get the callgroups from odbc? |
| 09:51.32 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
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| 09:54.41 | The_LightSide | hi all |
| 09:54.46 | *** part/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za) |
| 09:55.08 | The_LightSide | has anyone got the grandstream handy tone 488 to work with aah? |
| 09:57.30 | okkum | What can cause a: Cause No. 100 - Invalid information element contents. when dialling through a PRI? |
| 09:58.04 | *** join/#asterisk soundguy (n=soundguy@mel-gw-pt1536.blendtek.com.au) |
| 09:59.52 | soundguy | Hi. Is there someone here who can give me a hand with configuring the TDM400P (one FXO module) |
| 10:00.00 | soundguy | I've read voip-info.org and it doesnt say too much about it |
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| 10:05.04 | jonathh | hey can anyone tell me if it is possible to pass stdin to a forked application? |
| 10:07.24 | *** join/#asterisk zoa (n=zoa@87.215.18.236) |
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| 10:08.50 | lme | hi guys |
| 10:10.00 | fourcheeze | can someone point me at where the numeric disposition values in CDR are documented? |
| 10:10.56 | fourcheeze | or else does anyone know what a disposition of 8 is? |
| 10:12.12 | oej | Good question... |
| 10:12.50 | oej | DOc file says 15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY |
| 10:13.13 | fourcheeze | yes |
| 10:13.19 | fourcheeze | Answered seems to be 4 |
| 10:13.32 | oej | I get text strings like that, but you get numbers, like "8" ? |
| 10:13.43 | fourcheeze | this is what gets logged to sql back ends |
| 10:13.54 | oej | Which driver? |
| 10:14.10 | fourcheeze | cdr_odbc.so |
| 10:14.34 | oej | cdr_csv calls ast_cdr_disp2str |
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| 10:15.25 | oej | It is a setting in the config, called "dispositionstring=yes or no" |
| 10:15.44 | fourcheeze | ahh |
| 10:15.46 | fourcheeze | ok |
| 10:15.59 | oej | cdr_odbc.conf |
| 10:16.41 | oej | The values are defined in cdr.h as |
| 10:16.43 | oej | #define AST_CDR_NOANSWER (1 << 0) |
| 10:16.43 | oej | #define AST_CDR_BUSY (1 << 1) |
| 10:16.43 | oej | #define AST_CDR_ANSWERED (1 << 2) |
| 10:16.43 | oej | #define AST_CDR_FAILED (1 << 3) |
| 10:16.53 | oej | I guess 8 means FAILED |
| 10:16.56 | fourcheeze | hmmm |
| 10:17.00 | fourcheeze | starnge |
| 10:17.11 | fourcheeze | not quite sure what that means |
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| 10:23.51 | knobo | It actualy looks like it works, except that "show queues" does not work. |
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| 10:34.46 | knobo | "show queues" should have looked in my db. But it does no. |
| 10:35.54 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 10:47.51 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 10:48.02 | kink0 | good morning |
| 10:48.26 | kink0 | a newbie question: how to limit Asterisk use only to registered users ? |
| 10:48.50 | kink0 | that is, they need a login:password to carry calls thorugh this pbx |
| 10:49.48 | {zombie} | don't allow anyone in your [default] context to dial out |
| 10:50.01 | {zombie} | and pick sane usernames & passwords in your sip.conf |
| 10:50.59 | kink0 | {zombie}: then use only the [local] context ? |
| 10:51.12 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 10:51.27 | kink0 | I have username and pass in my sip.conf , but users are able to dial out even if not registered or register fails ( due to passwd change ) |
| 10:53.10 | {zombie} | in the [general] section of your sip.conf you will have something like |
| 10:53.11 | {zombie} | context = default ; Default context for incoming calls |
| 10:53.25 | {zombie} | make sure the [default] context in your extensions.conf has no permission to do anything |
| 10:53.36 | {zombie} | other than perhaps ring local extensions if that's what you want |
| 10:53.48 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 10:54.53 | kink0 | {zombie}: ok, I will try then to move [default] extensions to [local context, and try it |
| 10:56.39 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 10:56.50 | {zombie} | you can create any context you like |
| 10:56.55 | {zombie} | I call mine [sipphones] |
| 10:57.20 | {zombie} | you just need to make sure that the context you specify in sip.conf exists in extensions.conf |
| 10:57.34 | {zombie} | you can set different phones to different contexts, so you can give some people more access than others |
| 10:57.37 | BugKham | anyone knows of the best providers for making long distance calls? |
| 10:58.04 | {zombie} | BugKham: I use www.siphone.com.au |
| 10:58.26 | {zombie} | but you don't say what country you are in, and I'm guesing you're not in .au :) |
| 10:59.13 | BugKham | {zombie}: does it work well with *? |
| 10:59.41 | BugKham | {zombie}: i'm in Thailand |
| 10:59.57 | {zombie} | works excellent with * |
| 11:00.01 | BugKham | {zombie}: but asking informiend who;s in japanation for my fr |
| 11:00.10 | BugKham | sorry |
| 11:00.11 | {zombie} | they support IAX connections too if you ask them |
| 11:00.27 | BugKham | <PROTECTED> |
| 11:00.57 | BugKham | <PROTECTED> |
| 11:01.10 | {zombie} | there's a list of VoIP providers on voip-info.org - I would suggest looking there for provides in japan & thailand |
| 11:02.04 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 11:02.15 | BugKham | {zombie}: okay |
| 11:04.46 | kink0 | {zombie}: I got Failed to authenticate on INVITE even I see is registered |
| 11:05.01 | RoyK | hi |
| 11:05.22 | RoyK | ztdummy relies on a 1000Hz scheduler timer, right? |
| 11:05.33 | kink0 | I set in special context called peer-registrations , and register peers are now in that exten, where that exten contents de allowed dial out rules |
| 11:09.11 | *** join/#asterisk cpm (n=Chip@border1.avitecture.net) |
| 11:10.12 | RoyK | ~netsec |
| 11:10.31 | RoyK | ~lart himself |
| 11:15.55 | tzafrir | RoyK, do you mean to ask if ztdummy relies on HZ=1000? |
| 11:16.17 | tzafrir | Not as of 2.6.13 |
| 11:16.24 | tzafrir | (at least on i386) |
| 11:18.31 | RoyK | it doesn't? |
| 11:18.32 | RoyK | hm |
| 11:18.38 | RoyK | then what timers does it use? |
| 11:18.48 | RoyK | i thought someone said you needed HZ1000 |
| 11:21.56 | *** join/#asterisk prtxti[ePlus] (n=e152488@80-219-152-9.dclient.hispeed.ch) |
| 11:22.02 | *** join/#asterisk p_bateman (n=10337D33@80-219-157-1.dclient.hispeed.ch) |
| 11:22.31 | prtxti[ePlus] | servus |
| 11:22.49 | p_bateman | servus |
| 11:26.32 | kink0 | {zombie}: would I need an auth=name:pass@realm in my sip.conf to allow registered friends to call out ussing this Asterisk ? |
| 11:33.20 | *** join/#asterisk CaViCcHi (n=matteo@81.208.84.216) |
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| 11:40.52 | franck | How can I change the volume level on a zaptel digium card? |
| 11:41.44 | X-Rob | rxgain and txgain |
| 11:42.49 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:43.05 | RoyK | tzafrir: ping |
| 11:45.35 | prtxti[ePlus] | Should I update the firmware of my Grandstream GXP 2000 (1.0.1.9) to the latest version 1.0.2.3. ???? Any risks??? |
| 11:45.36 | CaViCcHi | what is the correct way to route a phone call from an FXO to an FXS module? Dial(Zap/${MODULEFXS}); right ?? |
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| 11:52.50 | CaViCcHi | what is the correct way to route a phone call from an FXO to an FXS module? Dial(Zap/${MODULEFXS}); right ?? |
| 11:53.06 | *** join/#asterisk Tiger (n=navajodu@adsl-69-106-231-140.dsl.pltn13.pacbell.net) |
| 11:53.45 | {zombie} | prtxti[ePlus]: have a gooood read of the voip-info wiki page |
| 11:53.54 | {zombie} | there's no way back to the previous version |
| 11:54.02 | {zombie} | and a few people reporting bugs that have me wary |
| 11:54.37 | *** part/#asterisk Tiger (n=navajodu@adsl-69-106-231-140.dsl.pltn13.pacbell.net) |
| 11:56.10 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
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| 12:01.27 | tzafrir | RoyK, pong |
| 12:02.06 | prtxti[ePlus] | @zombie have read the wiki. most of the features sound pretty cool (phonebook, miss called list...) but i am just not convinced. |
| 12:02.18 | tzafrir | RoyK, basically, set USE_RTC to use RTC and avoid the need to rely on HZ |
| 12:02.59 | RoyK | ah |
| 12:05.24 | *** join/#asterisk zoa (n=zoa@87.215.17.12) |
| 12:06.41 | {zombie} | prtxti: yeah, I figure I'll wait a little while.. I am most disappointed they didn't fix any of the previous little minor issues |
| 12:07.06 | {zombie} | like the fact speeddials don't work when you are transferring |
| 12:11.49 | skeffling | we're using asterisk with SNOM phones, is there a way, when transfering a call, for it to beep to indicate the call has now been transfered? |
| 12:13.52 | prtxti[ePlus] | ok, i think i´ll wait as well |
| 12:13.55 | prtxti[ePlus] | thanx |
| 12:24.33 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 12:31.43 | corruptor | hi, where can i read about automatic voip provider choice in *. For example i dial number X, * looks DB or some file and choose the cheapest. Or should i do this manually in dialplan? |
| 12:32.32 | t0ke | corruptor, check about LCR |
| 12:37.32 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 12:37.42 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 12:39.01 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 12:40.53 | *** join/#asterisk ukh (n=ukh@130.226.34.10) |
| 12:43.16 | ukh | dumb question: would asterisk benefit from hyperthreading? or should I just leave it off? |
| 12:44.59 | bigjb | i think the concensus from the conversation had by some people the other day is that although asterisk wont benefit from it theres no reason to turn it off |
| 12:45.47 | ukh | ok |
| 12:46.37 | cpm | seems like there is related stuff that can benefit though, like mp3player, that kinda thing. |
| 12:48.06 | cpm | Okay, What's the deal? |
| 12:48.18 | cpm | Isn't postfix supposed to attempt delivery down the line? |
| 12:48.48 | kink0 | SIP/2.0 407 Proxy Authentication Required |
| 12:48.48 | kink0 | <PROTECTED> |
| 12:48.56 | cpm | First to MX(s), then to primary dns, then secondary dns, then to A record? Isn't that the "rule" ? |
| 12:49.03 | kink0 | Failed to authenticate on INVITE to ... |
| 12:49.13 | cpm | Ooops, sorry, wrong window |
| 12:50.03 | darkskiez | cpm: if that is the rule i didnt know about trying the prim/secondary dns's |
| 12:50.39 | darkskiez | cpm: and i thought the A record was only tried if there were no MX's at all |
| 12:51.27 | *** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 12:51.35 | oej | ~seen zoa |
| 12:51.49 | jbot | zoa is currently on #asterisk (46m 26s), last said: 'sevard: then change the permission'. |
| 12:51.49 | _Paulo_ | Hi everybody. |
| 12:52.05 | _Paulo_ | I'm using *+unicall+spamdsp |
| 12:52.44 | _Paulo_ | I'm making a FAX machine with 30 lines... :-) |
| 12:53.28 | RoyK | spamdsp :) |
| 12:53.29 | _Paulo_ | I've installed a bunch of debian binary packets. |
| 12:53.34 | RoyK | dsp-spammerr |
| 12:53.36 | RoyK | tamtitam |
| 12:53.49 | _Paulo_ | s/spamdsp/spandsp/ |
| 12:53.52 | RoyK | :) |
| 12:54.14 | _Paulo_ | Well, I'm receiving faxes very well |
| 12:54.34 | _Paulo_ | But when sending... |
| 12:54.35 | RoyK | does transmitting them work as well? |
| 12:54.42 | _Paulo_ | No... |
| 12:54.47 | _Paulo_ | Half-pages |
| 12:54.59 | fubster | finally asterisk is in portage |
| 12:55.07 | _Paulo_ | Half-pages in the best cases. |
| 12:55.16 | RoyK | fubster: i only use the source..... |
| 12:55.50 | oej | What is portage? |
| 12:56.00 | _Paulo_ | Yesterday I made from sources. |
| 12:56.37 | _Paulo_ | now, app_?xfax are segfaulting... |
| 12:57.36 | _Paulo_ | :'( |
| 12:59.18 | bigjb | is that while sending a single fax to a single number on a single line? |
| 13:00.41 | TallAndy | If u make a call from asterisk manager will it get picked up by extensions.conf for a say custom agi script? |
| 13:03.15 | *** join/#asterisk coppice (n=chatzill@223.203.17.210.dyn.pacific.net.hk) |
| 13:03.23 | *** join/#asterisk arcy (n=arcanum@ppp66-adsl-244.ath.forthnet.gr) |
| 13:03.29 | *** join/#asterisk glnx (n=chatzill@62.37.237.17) |
| 13:04.25 | sivana | TallAndy: I think you might need to use the Local channel for it to get picked up |
| 13:05.44 | _Paulo_ | bigjb, sending or receiving does the same... |
| 13:06.27 | _Paulo_ | bigjb, to load app_rxfax or app_txfax triggers a Segmentation fault. |
| 13:08.02 | _Paulo_ | I think these apps are not working with the snapshot versions. |
| 13:08.22 | bigjb | _Paulo_, http://lists.digium.com/pipermail/asterisk-users/2004-June/049414.html |
| 13:08.29 | bigjb | issue with libtiff? |
| 13:08.42 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:10.13 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 13:10.57 | coppice | _Paulo_ which snapshot are you using? |
| 13:11.04 | *** part/#asterisk ukh (n=ukh@130.226.34.10) |
| 13:12.47 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:12.53 | *** join/#asterisk pengyong (n=lala@218.93.158.116) |
| 13:14.24 | _Paulo_ | coppice, spandsp-0.0.2pre23.tar.gz |
| 13:14.39 | _Paulo_ | coppice, libunicall-20060131.tar.gz |
| 13:14.44 | coppice | that should work OK |
| 13:15.11 | *** join/#asterisk keith (n=keith@82.152.131.65) |
| 13:15.14 | coppice | are you using R2? |
| 13:15.21 | _Paulo_ | I will restart from scratch. |
| 13:15.33 | CaViCcHi | what is the correct way to route a phone call from an FXO to an FXS module? Dial(Zap/${MODULEFXS}); right ?? |
| 13:15.38 | _Paulo_ | coppice, Yes, using r2 |
| 13:15.56 | CaViCcHi | or not? |
| 13:16.23 | coppice | it should work OK, as long as you use the app_rxfax.c and app_txfax.c that go with 0.0.2pre23 |
| 13:16.40 | *** join/#asterisk elephantMan (n=elephant@46.205.103-84.rev.gaoland.net) |
| 13:19.04 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:20.05 | *** join/#asterisk keith (n=keith@82.152.131.65) |
| 13:21.14 | _Paulo_ | <PROTECTED> |
| 13:21.15 | _Paulo_ | Segmentation fault |
| 13:21.35 | dpryo | owned |
| 13:22.51 | coppice | is this your frist time using rxfax, or have older versions worked OK? |
| 13:23.03 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 13:23.09 | mut | mornin all |
| 13:23.11 | _Paulo_ | coppice, older version worked well to receive |
| 13:23.21 | mut | i'm havin a lil problem with peer qualification |
| 13:23.27 | RoyK | vm24:/usr/src/linux# insmod /lib/modules/2.6.15.2/kernel/drivers/char/rtc.ko |
| 13:23.27 | RoyK | insmod: error inserting '/lib/modules/2.6.15.2/kernel/drivers/char/rtc.ko': -1 No such device |
| 13:23.28 | RoyK | wtf? |
| 13:23.30 | _Paulo_ | coppice, sending half pages at best. |
| 13:23.35 | mut | i have this customer on adsl, and he seems to timeout quite often |
| 13:23.58 | X-Rob | Woo. |
| 13:24.02 | mut | http://pastebin.ca/39607 |
| 13:24.06 | X-Rob | I found a fucker of a bug |
| 13:24.15 | X-Rob | http://bugs.digium.com/view.php?id=6399 |
| 13:25.00 | _Paulo_ | coppice, thats why I compiled a newer version, to see if the txfax improved. |
| 13:25.13 | RoyK | genrtc is a bitch... |
| 13:25.55 | oej | X-Rob: Well, the anything AGI did not exist, so Asterisk returned -1 |
| 13:26.04 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 13:26.08 | mut | anyone know why they'de qualify at 32ms |
| 13:26.08 | oej | We possibly should not execute the dial |
| 13:26.16 | X-Rob | oej, note, it doesn't matter wether or not it exists |
| 13:26.22 | mut | then 2 min later dissapear |
| 13:26.53 | mut | even tho it's there, and i can login to it |
| 13:27.02 | MattH | Anyone have an idea why I might have gotten this error, and why it then dropped the call? channel.c: Didn't get a frame from channel: SIP/5706011097-9fda |
| 13:27.03 | oej | I see |
| 13:27.04 | mut | spa-2002 ata |
| 13:27.25 | oej | x-rob: Yes, that is weird. How can the agi affect the dial timeout? |
| 13:27.33 | X-Rob | fiik 8-\ |
| 13:27.36 | oej | Test with resetcdr between the agi and dial, just for fun |
| 13:27.41 | X-Rob | k |
| 13:27.49 | oej | And see if there's something in the cdr that actually messes with dial |
| 13:27.56 | oej | Just shooting from the hip |
| 13:28.15 | X-Rob | Nup, same thing |
| 13:28.18 | coppice | ah, the foul stench of corruption in high locations :-) |
| 13:28.37 | X-Rob | OOh. Corruption. Can I have some? |
| 13:28.47 | oej | X-Rob - turn on SIP debugging and see if you see something there |
| 13:28.56 | oej | Seems like SIP is doing something at least |
| 13:29.05 | X-Rob | It is, it's dialling and then rejecting the call |
| 13:29.19 | oej | And this is trunk |
| 13:29.24 | oej | Cool beans |
| 13:29.29 | oej | Let's see if I can reproduce it |
| 13:31.15 | coppice | _Paulo_ txfax will not have improved. for most people it works OK. some have the same trouble as you. I think its a timing issue, but I haven't really followed up on it properly |
| 13:32.01 | oej | X-rob: Deadlocks... On my powerbook with only two phones |
| 13:32.25 | X-Rob | Deadlocks? *boggle* |
| 13:32.34 | MattH | Anyone have an idea why I might have gotten this error, and why it then dropped the call? channel.c: Didn't get a frame from channel: SIP/5706011097-9fda |
| 13:32.41 | X-Rob | I'm reading 9060 atm, seeing if there's anything in there that might have done it |
| 13:32.47 | CaViCcHi | what is the correct way to route a phone call from an FXO to an FXS module? Dial(Zap/${MODULEFXS}); right ?? |
| 13:33.27 | _Paulo_ | coppice, thank you. I will try with a 32bit kernel to see if the timing issue goes way. |
| 13:34.12 | X-Rob | CaViCcHi, depends if you've got a group set up. You can dial Zap/12 or whataver, to connect to a channel number, or you can dial Zap/g0 for a group |
| 13:34.32 | tzafrir | CaViCcHi, "from" depends on where in the dialplan you're at. Only "to" depends on the argument to Dial() |
| 13:34.51 | coppice | _Paulo_ are using using an AMD64? |
| 13:34.53 | CaViCcHi | okok i can answer... |
| 13:34.55 | X-Rob | CaViCcHi, read /usr/src/asterisk/configs/zapata.conf.sample |
| 13:35.23 | CaViCcHi | X-Rob I think there is no difference between dialling a group or a channel directly |
| 13:35.32 | X-Rob | well you think incorrectly |
| 13:35.33 | _Paulo_ | coppice, Intel(R) Pentium(R) 4 CPU 3.20GHz with 64bit intruction set |
| 13:35.37 | oej | X-Rob : Phone is ringing properly |
| 13:35.39 | X-Rob | a group is a group of channels. a channel is a channel. |
| 13:35.44 | X-Rob | oej, grr. |
| 13:35.47 | oej | X-Rob: for 15 secs |
| 13:35.52 | X-Rob | *swear* |
| 13:35.55 | *** join/#asterisk comunicatorx (n=comunica@82.152.131.65) |
| 13:36.08 | X-Rob | svn is meant to stop this! |
| 13:36.12 | CaViCcHi | X-Rob ... yes.. thats why i said so... |
| 13:36.26 | CaViCcHi | Can I ask you in pvt? |
| 13:36.30 | X-Rob | no |
| 13:36.40 | coppice | _Paulo_ I mostly test on a 32bit dual xeon, and a 64bit dual core AMD. they both work OK |
| 13:36.55 | CaViCcHi | mmm |
| 13:36.56 | CaViCcHi | ok |
| 13:37.50 | CaViCcHi | The problem is... if I call My FXO it answers correctly... after browsing thru his context... ad a certain point it... Dial(Zap/3,15) |
| 13:38.04 | CaViCcHi | at channel 3 I have an FXS module |
| 13:38.38 | CaViCcHi | from cli I watch at debug and it says that's ringing... |
| 13:39.02 | CaViCcHi | and a certain kind of phone rings (just a cordless... i dont know why) |
| 13:39.09 | X-Rob | ah. |
| 13:39.13 | X-Rob | you want 'boostringer' |
| 13:39.14 | CaViCcHi | but if you answer nothing happens |
| 13:39.18 | X-Rob | oh |
| 13:39.20 | X-Rob | maybe you don't. |
| 13:39.25 | X-Rob | possibly a broken board. |
| 13:39.34 | CaViCcHi | oh god :( |
| 13:39.37 | X-Rob | is it a digium board? |
| 13:39.40 | CaViCcHi | yes |
| 13:39.47 | _Paulo_ | coppice, At this point I have no clue, shooting blindly. |
| 13:39.54 | RoyK | CaViCcHi: call support :) |
| 13:39.55 | CaViCcHi | tdm400p with... mmm green module i cant remember |
| 13:39.57 | X-Rob | Call 'em. They give support for their hardware. |
| 13:40.31 | X-Rob | oej, just redownloaded |
| 13:40.35 | X-Rob | compiling again now |
| 13:41.02 | RoyK | argh |
| 13:41.10 | MikeJ[Laptop] | argh indeed... |
| 13:41.13 | RoyK | ztdummy doesn't seem to work with rtc.ko |
| 13:41.15 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 13:41.18 | RoyK | or genrtc.ko |
| 13:41.19 | RoyK | any ideas? |
| 13:41.22 | MikeJ[Laptop] | prevent global warming, become a pirate |
| 13:41.27 | RoyK | lol |
| 13:41.29 | MikeJ[Laptop] | nope |
| 13:41.41 | st3v | is there a way to modify the directory to use Festival to pronounce the user's names? |
| 13:41.45 | MikeJ[Laptop] | flying speghetti monster! |
| 13:41.54 | st3v | It is just spelling out the names right now |
| 13:41.56 | MikeJ[Laptop] | st3v, yes |
| 13:42.23 | X-Rob | Royk, is it returning ummm. I/O Error? |
| 13:42.25 | st3v | how can I do that |
| 13:42.54 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:42.58 | oej | Interesting - after timeout SIP sends 603 declined |
| 13:43.06 | MikeJ[Laptop] | I don't recall.. I think I had issues with names that it didn't know how to pronounce |
| 13:43.08 | coppice | the FSM must have been cooked up by an engineer, cos years ago I heard someone call a finite state machine a flying spaghetti monster |
| 13:43.09 | RoyK | X-Rob: yes |
| 13:43.11 | RoyK | insmod: error inserting '/lib/modules/2.6.15.2/misc/ztdummy.ko': -1 Input/output error |
| 13:43.19 | RoyK | ztdummy: Unable to register zaptel rtc driver |
| 13:43.29 | RoyK | but rtc.ko is indeed loaded |
| 13:43.36 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:43.57 | RoyK | genrtc lacks some symbols |
| 13:44.01 | RoyK | X-Rob: any idea? |
| 13:44.07 | X-Rob | Royk, disable HPET_RTC_IRQ |
| 13:44.17 | X-Rob | then recompile rtc |
| 13:44.23 | RoyK | ah |
| 13:45.03 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 13:45.34 | X-Rob | oej, still did the same thing for me, with a clean checkout. |
| 13:45.38 | X-Rob | ok. Must be something else |
| 13:45.46 | RoyK | danke sehr |
| 13:45.54 | oej | X-Rob: good luck hunting |
| 13:46.05 | X-Rob | this is wierd |
| 13:46.13 | X-Rob | removed everythign from -addons in modules |
| 13:46.56 | X-Rob | oej, arch/compiler? |
| 13:47.27 | oej | Powerbook, OS/X Tiger, GCC |
| 13:48.15 | RoyK | oej: gcc4? |
| 13:49.20 | CaViCcHi | Sorry, how can I verify if it's a real hardware problem? |
| 13:50.26 | X-Rob | I s'pose my next step is to nuke everything from /etc/asterisk and see if I can narrow the problem down there. |
| 13:51.32 | X-Rob | rm -rf asterisk ; svn co http://svn.digium.com/svn/asterisk/trunk asterisk |
| 13:51.34 | RoyK | how can i check if asterisk really sees the timing device? ztdummy or whatnot? |
| 13:51.52 | RoyK | X-Rob: any reason you use trunk? |
| 13:52.03 | oej | X-rob: Yes, gcc 4 |
| 13:52.10 | X-Rob | RoyK, coz I like living on the edge 8) |
| 13:52.29 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 13:52.32 | oej | We need testers on the edge :-) |
| 13:52.51 | RoyK | heh |
| 13:52.57 | X-Rob | RoyK, cat /proc/zaptel/1 |
| 13:53.02 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:53.03 | RoyK | but how can i have asterisk tell me if it has a timing device or not? |
| 13:53.04 | X-Rob | if it says 'ZTDUMMY' then it's working. |
| 13:53.19 | RoyK | where? |
| 13:53.29 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 13:53.31 | X-Rob | cat /proc/zaptel/2 |
| 13:53.31 | X-Rob | Span 2: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 13:53.38 | X-Rob | (that's 2, because I have zaptel hardware as 1) |
| 13:53.48 | RoyK | Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 13:53.49 | RoyK | ok |
| 13:53.59 | RoyK | but should asterisk be configured to use it anyhow? |
| 13:54.10 | X-Rob | it'll use it if it's there. If it's not there, it won't use it. |
| 13:54.28 | RoyK | but it won't ever tell me if it tries and doesn't find it? |
| 13:54.42 | RoyK | perhaps it likes that privacy..... |
| 13:54.54 | X-Rob | I think everything that needs a timing device will bitch if it doesn't find it. |
| 13:55.27 | dpryo | You can always run it through strace :) |
| 13:57.23 | iCEBrkr | Hey yo! |
| 13:59.20 | X-Rob | ok. Definately broken. |
| 13:59.25 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:59.39 | oej | X-Rob: Still have the problem? |
| 13:59.43 | X-Rob | yep |
| 13:59.51 | X-Rob | Going to try 89something |
| 13:59.55 | X-Rob | it was working then |
| 14:00.05 | X-Rob | just to prove I'm not totally mental 8) |
| 14:00.18 | oej | What is your incoming channel? |
| 14:00.27 | X-Rob | it's sip->sip |
| 14:00.34 | oej | Me too |
| 14:01.07 | X-Rob | uh. OK. How do I roll back with svn? do I have to do a checkout or can I just update with a revision number? |
| 14:01.30 | oej | svn checkout --revision 1234 |
| 14:01.33 | X-Rob | ..-r8926 update |
| 14:01.35 | X-Rob | seemed to work |
| 14:01.40 | oej | Guess so |
| 14:01.48 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:02.26 | X-Rob | which, looking at it, was probably a bad one to pick |
| 14:02.36 | X-Rob | 'issue 5892: Set a minimum T1 timer for calls' |
| 14:03.23 | X-Rob | ok, I'm at 8906 now. compiling. |
| 14:04.03 | X-Rob | which is what I _think_ I was running before I upgraded. |
| 14:04.44 | oej | Wow. svn is open for commits again. |
| 14:06.12 | *** join/#asterisk FastJack (n=fastjack@reverse-82-141-61-154.dialin.kamp-dsl.de) |
| 14:06.19 | X-Rob | aaaaalmost there. |
| 14:06.46 | X-Rob | wish I was at work. Dual Xeon's make short work of compiles. |
| 14:07.08 | coppice | dual xeon PIIs don't -) |
| 14:07.20 | X-Rob | heh |
| 14:07.33 | coppice | my AMD X2 slaughters by 2.4GHz xeons for compiles |
| 14:07.49 | oej | And Damin's P66 was really slow |
| 14:08.09 | Cresl1n | oej!!!!! |
| 14:08.13 | Cresl1n | coppice!!!!! |
| 14:08.14 | Cresl1n | :-) |
| 14:08.18 | Cresl1n | X-Rob: !!!!!! |
| 14:08.41 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:08.42 | coppice | good morrow |
| 14:08.48 | Cresl1n | howdy |
| 14:08.58 | comunicatorx | My experience with amd 64 X2 is a delight |
| 14:09.07 | *** join/#asterisk jmacz (n=jmacz@201.244.199.176) |
| 14:09.13 | oej | Cresl1n: Welcome back! |
| 14:09.35 | coppice | running the spandsp tests on my X2 is quite impressive compared to a xeon |
| 14:10.14 | Cresl1n | coppice: I think the AMDs have more processing power than the Xeons do |
| 14:10.26 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflnd.dialup.mindspring.com) |
| 14:10.41 | asteriskmonkey | Amds do have more power :D |
| 14:10.46 | Cresl1n | coppice: I heard they have a few more processing units on the pipeline than intels |
| 14:10.57 | asteriskmonkey | its to do with instruction sets |
| 14:11.01 | coppice | well the difference is DSP speed is way out of line with that. it must be the extra registers avoiding bottlenecks or something |
| 14:11.04 | asteriskmonkey | and smp meshing |
| 14:11.14 | X-Rob | ok, back to 8877 and I can still hear the other phone ringing. |
| 14:11.17 | X-Rob | but it stops |
| 14:11.26 | Cresl1n | coppice: I'd assume you're compiling specifically for the AMD, right? |
| 14:11.44 | coppice | it seems like the xeons should be hotter for DSP, as the FP runs at the clock rate |
| 14:12.07 | coppice | yeah. I use FC4 for AMD64 on the X2 |
| 14:12.26 | X-Rob | oooh |
| 14:13.22 | coppice | they're getting cheaper :-) |
| 14:14.11 | coppice | I've had this one since last June, so it was less cheap (it would make me worry if I called it expensive) |
| 14:14.24 | X-Rob | hehehe |
| 14:14.30 | X-Rob | significantly less cheap 8) |
| 14:14.31 | *** join/#asterisk _8ball (n=spunk@64-13-46-242.eug.clearwire-dns.net) |
| 14:15.32 | coppice | they run nice and cool, so the X2 machine makes a lot less noise than the stinky xeon |
| 14:16.06 | X-Rob | YAY |
| 14:16.10 | X-Rob | Found a revision that works |
| 14:16.14 | X-Rob | r8571 _works_ |
| 14:16.32 | coppice | the core duo chips don't sound too bad for general use, but 32 bits and lousy FP really doesn't cut it these days |
| 14:20.36 | coppice | asteriskmonkey: i have no idea what that means :-\ |
| 14:21.56 | coppice | those military boxes someone is selling with the cell processor seem to be a big box with just as few processors. I thought they would be packing them in. |
| 14:21.57 | coppice | i want a PCI card with several cell processors |
| 14:22.04 | RoyK | :) |
| 14:22.15 | RoyK | octo cell on pci express.... |
| 14:22.25 | RoyK | with a fan the size of a small car |
| 14:22.31 | X-Rob | good to 8712...trying 8809 |
| 14:22.47 | coppice | 8 might be a lot, but I assume 3 or 4 is realistic |
| 14:23.14 | X-Rob | a lot is far more fun than not enough |
| 14:23.33 | coppice | 8809 should be able to decode DTMF well :-) |
| 14:25.51 | TallAndy | sivana: Sorted it! If you use exten => s it picks up calls from the asterisk manager |
| 14:25.53 | mut | is it not possible to compile the MARK2 echo canceller anymore? |
| 14:25.56 | oej | This is like Star Trek stardates |
| 14:26.12 | oej | Asterisk logbook by Mark Spencer, Astdate 8809, Good date for DTMF |
| 14:26.14 | asteriskmonkey | coppice: sos was thinking of dealer x2 :P |
| 14:26.23 | mut | In file included from /usr/src/zaptel/zaptel.c:111: |
| 14:26.26 | mut | /usr/src/zaptel/mec2.h: In function 'echo_can_create': |
| 14:26.29 | mut | and dies |
| 14:26.31 | asteriskmonkey | is there anyway to cancel absoutetmeout with a keystroke? |
| 14:26.56 | coppice | the 8809 was an old DTMF decoder chip |
| 14:27.30 | *** join/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 14:27.34 | X-Rob | coppice, ahha. I googled for 8809, but I probably should have been slightly more selective |
| 14:28.15 | oej | asteriskmonkey - three: ctrl-alt-delete |
| 14:28.31 | oej | coppice :-) |
| 14:28.42 | asteriskmonkey | no seriously |
| 14:29.03 | asteriskmonkey | on a telephone if you have set in your exten => an absoulte time value |
| 14:29.14 | asteriskmonkey | probably not right as that negates its purpose |
| 14:30.26 | X-Rob | Ahha! between r8835 r8883 it's broken |
| 14:35.05 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivflnd.dialup.mindspring.com) |
| 14:37.42 | X-Rob | 8852 works.. |
| 14:39.08 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 14:40.24 | X-Rob | OOOH |
| 14:40.27 | X-Rob | I _bet_ this is it |
| 14:40.42 | X-Rob | URL: http://svn.digium.com/view/asterisk?rev=8877&view=rev |
| 14:40.42 | X-Rob | Log: |
| 14:40.42 | X-Rob | Merge Rizzo's waitfor update (bug #4584) |
| 14:41.13 | X-Rob | Bingo! |
| 14:42.25 | *** join/#asterisk jmacz (n=jmacz@201.244.199.176) |
| 14:44.12 | X-Rob | Hey, is OSX 64bit? |
| 14:44.38 | X-Rob | (eg, what's the size of time_t?) |
| 14:44.39 | jaiger | X-Rob, I'm pretty sure it's 23 bit |
| 14:44.45 | jaiger | I mean 32 bit |
| 14:44.49 | X-Rob | heh |
| 14:44.50 | *** join/#asterisk santiago (n=santiago@63.245.86.182) |
| 14:44.57 | X-Rob | Looking through the change there to see if I Can find a problem |
| 14:45.02 | X-Rob | going to have a smoke first though. |
| 14:45.18 | oej | Wow. Only 23 bits. I need to update to a new laptop |
| 14:45.37 | oej | X-Rob: Your system is 64? |
| 14:45.39 | *** part/#asterisk santiago (n=santiago@63.245.86.182) |
| 14:45.59 | file[laptop] | oej: my system is 128 bit, take ethat! |
| 14:46.22 | jaiger | AFAIK IBM's high-end equipment is the only 64 bit PPC out there |
| 14:46.27 | oej | I need the new Mac laptops, I bet they are 256 bit with extra muffinpower |
| 14:46.39 | jaiger | oej, do you know the muffin man? |
| 14:46.49 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 14:47.06 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 14:48.05 | cypromis | the ones with muffinpower are the ones tuned by file |
| 14:48.22 | file[laptop] | you just have to install the muffinpower module |
| 14:49.38 | X-Rob | oej, nope, 32. |
| 14:49.42 | X-Rob | was thinking that could have been the problem |
| 14:49.48 | X-Rob | turning off compiler optimizations now, just to check |
| 14:50.13 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:50.48 | *** join/#asterisk blint (n=blint@adsl-669.mirage.euroweb.hu) |
| 14:50.54 | blint | hi everyone |
| 14:52.27 | Katty | allo, blint |
| 14:52.35 | X-Rob | nope, that didn't fix it. |
| 14:52.40 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 14:53.08 | kink0 | hello, I still having problems while try to allow registered peers to place outbound calls, I got Proxy Auth Req |
| 14:53.59 | blint | i use debian's asterisk 1.0.7 with 2 tdm400p cards in russia |
| 14:53.59 | X-Rob | ooh. There's some modified stuff in ZAPTEL_OPTIMIZATIONS, oej, which you wouldn't be using. |
| 14:54.23 | oej | X-rob: Not at all. No zaptel on this laptop |
| 14:54.37 | blint | as russia has no tone dialing my Zap interfaces have pulsedial=yes |
| 14:54.50 | kink0 | <PROTECTED> |
| 14:55.16 | kink0 | what I need to authenticate INVITEs from a registered peer ? |
| 14:55.20 | blint | and it works all right, but for some reason if I put a w in the dialstring (8w10) is the international prefix, i just get silence in the line |
| 14:55.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:56.47 | blint | i tried everything from different wait times to sending the rest as dtmf but nothing helped |
| 14:56.53 | coppice | OSX is going backwards - 64 bit to 32 bit. why didn't they wait for the 64 bit intel chips later this year? |
| 14:57.25 | *** join/#asterisk Johan (n=kvirc@194.151.113.2) |
| 14:57.26 | Johan | Hi all |
| 14:57.28 | MikeJ[Laptop] | I hear the intel chips they are using are actually performing better than the old chips |
| 14:57.50 | MikeJ[Laptop] | so 4 bits or 2 million.. if it works better! |
| 14:58.10 | coppice | but what happens to the 64 bit native apps? |
| 14:58.16 | Johan | Is it possible to make asterisk jump to an context when someone hangsup? So a sort of OnHangup(context) in extension.conf? |
| 14:58.30 | X-Rob | Johan, use the 'h' context. |
| 14:58.36 | kink0 | Johan, maybe deadagi |
| 14:58.44 | X-Rob | exten => xxx,h,Goto(foo,s,1) |
| 14:59.04 | Johan | h context? Lemme search for some docs on that. Thanx. |
| 14:59.38 | brad_mssw | coppice: yeah, didn't make sense to me either ... they had the whole 64bit marketting push with G5, then they come out with a 32bit-only platform ... odd |
| 14:59.43 | kink0 | I get registered peers, but they are unable to send a call out ussing this asterisk, anybody can give me some help about peer authentication ? |
| 14:59.55 | darkskiez | johan: its the h extension, not the h context |
| 14:59.55 | X-Rob | Nup. Still broken w/o ZAPTEL_OPTIMIZATIONS |
| 14:59.56 | kink0 | appear is ok for REGISTER but no for INVITE |
| 15:00.38 | Johan | darkskiez: ok, thanx |
| 15:00.44 | *** part/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:01.24 | coppice | 32bits - more compact than 64bits, for a lighter notebook :-) |
| 15:01.42 | MikeJ[Laptop] | see.... |
| 15:02.15 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 15:03.57 | X-Rob | Hrm. Some assumptions are being made here. |
| 15:04.07 | X-Rob | long whentohangup = 0, diff |
| 15:04.17 | X-Rob | but it should be time_t, which is what it's being compared against. |
| 15:04.52 | mut | k, right now i have an 'unreachable' client according to my qualify |
| 15:05.23 | mut | but i can login to the ata, says it's registered, next registration in 500 seconds |
| 15:05.23 | mut | why does it become inreachable |
| 15:05.23 | mut | unreachable |
| 15:05.30 | mut | it does it a lot.. |
| 15:05.47 | mut | qualify is set to 3000ms, usual time for the user is 32ms |
| 15:07.03 | Ahrimanes | hm sip notify messages arent reaching the ataq? |
| 15:07.21 | Ahrimanes | mut: what ata? |
| 15:07.29 | mut | spa-2002 |
| 15:07.37 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 15:08.02 | mut | it's behind nat, but port 5060 is forwarded to the ata as is the rtp range it uses |
| 15:08.03 | Ahrimanes | mut: i have seen it with a 1001 recently.. sent the user a grandstream handytone 286.. it works flawlessly under the exact some conditions.. |
| 15:08.21 | masonf | is there any way to surpress the first ring with my zaptel driver? |
| 15:08.26 | mut | well i have quite a few (200) or so spa's |
| 15:08.32 | mut | and it only happens on a select few |
| 15:08.37 | mut | but happens constantly on them |
| 15:08.46 | Ahrimanes | hm what kind of nat devices? |
| 15:08.54 | mut | zoom adsl modem |
| 15:08.56 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 15:09.11 | Ahrimanes | hm ok, dont know them |
| 15:09.27 | Ahrimanes | why not use stun instead of forwarding ports? |
| 15:09.50 | nestar | anyone using the Grandstream GXP2000? |
| 15:10.03 | nestar | opinions before i buy them? |
| 15:10.54 | mut | because for the most part people work fine |
| 15:11.17 | coppice | nestar: you'll get opinions between excellent value for money, and it stinks |
| 15:11.38 | mut | dunno whats all involved in running stun either |
| 15:11.44 | mut | have to take time that i don't have to do that |
| 15:11.53 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 15:12.15 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 15:12.38 | nestar | can't be as bad as the Budgettones |
| 15:12.45 | nestar | those are pretty ghetto |
| 15:12.51 | _Sam-- | i like the gxps myself |
| 15:12.54 | _Sam-- | new firmware is aight |
| 15:13.00 | _Sam-- | price is right |
| 15:13.12 | _Sam-- | all in all...i would buy it for 85 bucks...i dont know what else at that price you would consider that would be any bettter |
| 15:13.23 | shmaltz | ~seen JerJer |
| 15:13.27 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 14d 17h 42m 33s ago, saying: 'thanks again'. |
| 15:13.29 | shmaltz | JerJer, Ping |
| 15:13.34 | coppice | an analogue phoen for $1? :-) |
| 15:13.42 | _Sam-- | you can think that. |
| 15:13.48 | _Sam-- | but it would be a silly thought |
| 15:14.13 | _Sam-- | the firmware that came out last week for the gxp really moved it up a notch or two |
| 15:14.22 | coppice | OK, the $1 phone's don't have an LCD display. other than that they outperform most VoIP phones |
| 15:14.55 | _Sam-- | 1 dollar phone with a 70 dollar ata is still no better than a gxp :) |
| 15:15.04 | _Sam-- | and costs only a few bucks less |
| 15:15.06 | nestar | well, for the price we paid for all our Polycom's, i'm not that impressed |
| 15:15.21 | _Sam-- | nestar: talk to [av]bani |
| 15:15.27 | _Sam-- | he's got both polycoms and gxps |
| 15:15.30 | shmaltz | anybody here has a teleconferencing server available for the next 6 months for 10 conferences 1.5 hours per week? |
| 15:15.31 | _Sam-- | he'll tell ya like it is |
| 15:16.01 | _Sam-- | [av]bani: wake up |
| 15:16.18 | coppice | Sam-: is the case of the GXP as nasty as the bodgitones? |
| 15:16.34 | _Sam-- | never owned a budgetone so i cant say |
| 15:16.58 | coppice | OK, do your consider the GXP well made and attractive? |
| 15:17.25 | _Sam-- | compared the a merlin analog phone? yeah! |
| 15:17.32 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:17.39 | nestar | i guess i really need to upgrade my polycom firmware |
| 15:17.42 | _Sam-- | its a cheap phone that works well for a cheap phone |
| 15:17.47 | nestar | i'm still running like 1.3.4 |
| 15:17.51 | coppice | if the bodgitones had a nice case and the same software, people would have a totally different opinion of them |
| 15:17.53 | _Sam-- | there is no other 85 dollar phone with more features that works any better |
| 15:18.36 | _Sam-- | in MY opinion of course! |
| 15:19.08 | pb__ | nestar: we have a pile of GXPs and they seem to work well enough. |
| 15:19.09 | coppice | their new videophone looks pretty nasty in the pictures. they really should find themselves a good case designer. there are plenty within a kilometre of grandstream's place |
| 15:19.29 | pb__ | the other phones we have here are snom 360s, which cost about twice as much as the GXPs. The snoms are definitely better phones, but they're not twice as good. |
| 15:19.44 | _Sam-- | some people are more price sensitive and care more about the price than about looking pretty |
| 15:19.52 | _Sam-- | in fact, every one of my clients fits that statement |
| 15:19.59 | _Sam-- | but i get all the cheap guys :) |
| 15:20.03 | coppice | only twice? i thought snoms were more expensive than that |
| 15:20.39 | coppice | a smart case costs no more, but can really drive up sales |
| 15:20.46 | _Sam-- | thats true |
| 15:21.22 | coppice | these days everyone uses plastics that crumble in 3-4 years |
| 15:21.53 | _Sam-- | some of the discussions on this channel have revealed really that the weakest link on the GXP is the way that it handles configurations via tftp...you can set cfg-mac file in a directory...but you still have to tell the phone how to get the stuff |
| 15:22.16 | _Sam-- | and that has been about the weakest link anyone has been able to find in comparing the gxp to other more costly phones |
| 15:22.33 | pb__ | yeah, that's annoying. you only have to do it once, though. |
| 15:22.38 | coppice | people have complained about far more than that |
| 15:22.48 | _Sam-- | most of the other issues are being resolved |
| 15:22.53 | TallAndy | hi, anyone here with phpagi experience? |
| 15:22.56 | coppice | dunno how justified those complaints are, though |
| 15:22.57 | nestar | yeah, i don't really care about that, i'm only setting up 4 |
| 15:22.58 | _Sam-- | that was the WEAKEST link in discussing it against other phones |
| 15:23.13 | pb__ | coppice: I think we pay about 75 GBP for the grandstreams, and about 160GBP for the snoms. |
| 15:23.30 | asteriskmonkey | damn... |
| 15:23.40 | coppice | well 75 for the GXP is way too much |
| 15:23.50 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:24.02 | asteriskmonkey | you can get aastra 9113's for 150cad thats like 70gbp and its better phone |
| 15:24.36 | pb__ | aastra 9113s are about 100GBP over here. |
| 15:24.50 | pb__ | you might be able to get them for 150CAD in Canada, but that isn't exactly the same thing. |
| 15:25.40 | coppice | its really hard to find voip phones for sale here, yet most are made not far away :-\ |
| 15:27.04 | Hmmhesays | little pig little pig, let me in |
| 15:27.44 | coppice | are you talking about blow jobs? :-) |
| 15:28.04 | asteriskmonkey | sweet i should open up show in uk then |
| 15:28.10 | nestar | oh fawk |
| 15:28.12 | nestar | hahaha |
| 15:28.22 | nestar | this new polycom is *downgrading* itself |
| 15:28.30 | nestar | from 1.6.x to 1.3.x |
| 15:28.40 | nestar | damnit |
| 15:28.48 | *** join/#asterisk ordej (i=TRAX_@62.162.229.95) |
| 15:29.16 | coppice | many things are way overpriced in the UK. many are way overpriced here. its amazing how much companies can get away with loading up prices in specific markets |
| 15:29.37 | nestar | errrr |
| 15:29.39 | nestar | 1.3.1 |
| 15:29.41 | nestar | damnit |
| 15:29.44 | *** join/#asterisk clint_ (n=clint@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 15:29.44 | many | thats capitalism! |
| 15:29.46 | nestar | stupid phone |
| 15:30.06 | *** join/#asterisk orde1 (i=TRAX_@62.162.228.130) |
| 15:30.40 | coppice | well, a lot of it is illegal in the EU, but little gets done about it |
| 15:31.17 | [TK]D-Fender | nestar : You can set it to not accept upgrades in the provisioning files... |
| 15:32.02 | orde1 | where to download asterisk live cd with calling card billing? |
| 15:32.21 | nestar | [TK]D-Fender: I'm sure, but i wouldn't expect it to downgrade |
| 15:33.14 | [TK]D-Fender | nestar : Polycom does what you tell it, not necessarily what you WANT :) |
| 15:33.28 | [TK]D-Fender | I find that to be a good policy. |
| 15:34.18 | malverian[work] | Why does Snom break things with every single firmware release? |
| 15:34.30 | malverian[work] | One thing gets fixed, and another break.s |
| 15:34.41 | nestar | bleh |
| 15:34.45 | nestar | polycom is junk! |
| 15:34.49 | nestar | just like dsm's |
| 15:34.58 | [TK]D-Fender | malverian[work] : Don't think of them as bugs, rater as "features". And ones requiring the complete reboot of the phone or re-flashing as "special features" :) |
| 15:35.06 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 15:35.13 | [TK]D-Fender | nestar : LOL.... junk... thats funny.... |
| 15:35.16 | coppice | see, snom sucks as badly as grandstream, but with a nicer case people have a much higher opinion of it |
| 15:35.30 | [TK]D-Fender | DSM's? |
| 15:35.33 | *** join/#asterisk Elgi (n=Elgi@vigor.solutions-systemes.com) |
| 15:35.34 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 15:35.35 | TallAndy | With an agi script 'input.php' set chmod 444 i get 'Permission denied' obviously. But when the file is chmod 755 i get 'No such file or directory' any ideas? |
| 15:35.43 | coppice | Dumb Sales Men |
| 15:36.08 | *** join/#asterisk MatsK (n=mk@84-217-5-20.tn.glocalnet.net) |
| 15:36.21 | *** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 15:36.23 | arkanis | hi |
| 15:36.33 | nestar | sorry, dsm is a type of car |
| 15:36.34 | nestar | ;) |
| 15:37.03 | *** join/#asterisk Donkcyde (i=jk@packetsexchange.net) |
| 15:38.18 | coppice | haha, the IE7 beta has so many problems, testers are considered at serious risk :-) |
| 15:38.24 | mover | wait i use another client |
| 15:38.42 | [TK]D-Fender | nestar : Any other complaints about them? Basically allowing you to downgrade isn't normally a BAD thing. Imagine if a newer release had bugs you could not live with and couldn't go back... |
| 15:39.09 | *** part/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 15:39.39 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 15:39.40 | blint | TallAndy: if you try to execute it from the command line as the user that is running asterisk, what happens? |
| 15:39.50 | corruptor | have any1 had experience with MeetMe? What's the max number of participants in one room? |
| 15:39.51 | [TK]D-Fender | coppice : Well Firefox is pretty "heavy" these days and need to clean up their act. I still use it for the value of "extensions" and the better rendering for most things (greater resizing, etc) |
| 15:39.56 | TallAndy | I'll give it a try |
| 15:40.14 | nestar | [TK]D-Fender: their corporate attitude sucks, their support sucks, their config is cumbersome, inability to download firmware directly from their site.. |
| 15:40.28 | nestar | downgrading is nice, but it shouldn't downgrade without prompting me |
| 15:40.30 | coppice | firefox seems to work a lot better on linux than windows |
| 15:40.36 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 15:40.36 | TallAndy | blint: -bash: ./input.php: /usr/local/bin/php: bad interpreter: No such file or directory |
| 15:40.58 | TallAndy | Its the location to php then |
| 15:41.08 | TallAndy | its a SuSE 9.2 pro box |
| 15:41.11 | blint | indeed :D |
| 15:41.15 | TallAndy | .... :) |
| 15:41.22 | blint | should be in /usr/bin then I guess |
| 15:41.25 | mover | TallAndy: ls -s /usr/sbin/php /usr/local/bin/php |
| 15:41.34 | TallAndy | long day :P cheers blint! |
| 15:41.48 | blint | np |
| 15:41.49 | [TK]D-Fender | nestar : Well there are reasons why they do what they do, and while that means you should try to buy them from an authorized and supporting dealer I've got nothing bad to say of them so far. There are a few bugs I'd like to see fixed, but to date the product just keeps getting better. ACD support coming soon to * as well... |
| 15:42.57 | nestar | [TK]D-Fender: I did buy them from a authorized dealer.. and they told me to go download firmware from the freedomphone's site.. :boggle: |
| 15:43.09 | nestar | to their credit, they now have their own ftp site |
| 15:43.22 | nestar | but for the first 6 months, i had to rely on some random site |
| 15:43.38 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:43.58 | nestar | ACD support would be nice. |
| 15:44.01 | [TK]D-Fender | nestar : For shame.... my guys jump when I call :) |
| 15:44.15 | [TK]D-Fender | nestar : there's a patch in SVN awaiting full mergability. |
| 15:44.49 | nestar | :) |
| 15:44.53 | coppice | so, another 5 years and it might get in :-) |
| 15:46.10 | coppice | hey, that isreali USB channel bank has finally been launched |
| 15:46.47 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 15:47.02 | _Sam-- | i was just saying its crazy that channel banks still interface over t1 |
| 15:48.01 | coppice | dunno. audio over USB has a somewhat problematic history |
| 15:48.03 | fulgas | humm |
| 15:48.06 | fulgas | asterisk started to give me "AGI Rx << Usage: php [options] [-f] <file> [--] [args...]" |
| 15:48.11 | fulgas | on the php scripts |
| 15:48.12 | *** join/#asterisk Defraz_ (i=t0tal@tim.mychoice.cc) |
| 15:48.13 | [TK]D-Fender | TDMoE :) |
| 15:49.13 | tzafrir | coppice, well, if you have any questions about it, just ask |
| 15:49.20 | Makenshi | is there any Skype to SIP gateway, so I can route incoming Skype calls to Asterisk? |
| 15:49.28 | Makenshi | cause i'm fed up of having to run that crappy client |
| 15:49.29 | _Sam-- | i think nuggett was testing one |
| 15:49.46 | _Sam-- | nugget you got that skype adapter yet? |
| 15:50.07 | coppice | tzafrir: do you have much trouble with quirky audio? It seems a lot of machines still don't run smoothly with USB audio devices |
| 15:50.07 | blint | i saw one on the asterisk-users list, windows-based for $50 |
| 15:51.02 | tzafrir | coppice, it's not a USB audio device. It's a separate USB device. We don't use isochronious transfers, btw |
| 15:51.07 | *** join/#asterisk CleanerX (n=nix@p54A3A704.dip0.t-ipconnect.de) |
| 15:51.29 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.7) |
| 15:51.34 | coppice | is that because you had trouble with isochronous, or by free choice? |
| 15:51.46 | tzafrir | coppice, by free choice, IIRC |
| 15:51.54 | coppice | USB was originally designed to carry E1s almost transparently |
| 15:52.18 | tzafrir | USB or USB2? |
| 15:52.20 | mdave | reload |
| 15:52.22 | mdave | er |
| 15:52.35 | asteriskmonkey | does asterisk support qsig |
| 15:52.52 | coppice | tzafrir: the original USB specs, which people like Nortel had a big hand in |
| 15:53.39 | Makenshi | coppice, that's a a lot of overhead, given usb1 is 12mbs and e1 is 2mbs |
| 15:53.56 | coppice | USB1 is really 6M |
| 15:54.58 | coppice | just as USB2 is really 240M |
| 15:56.08 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:58.01 | *** join/#asterisk alliance (n=alliance@69.158.1.248) |
| 15:58.02 | jaiger | coppice, how is usb1 6M? do they count 6M up + 6M down? |
| 15:58.40 | coppice | yep. the speeds they quote are the aggregates. nothing can stream data > half those rates |
| 15:59.14 | jaiger | lame |
| 15:59.55 | file | BUNNIES |
| 16:00.00 | file | IT MUST BE BUNNIES |
| 16:00.05 | file | ...or maybe midgets |
| 16:00.16 | coppice | i guess their defence would be that ethernet quoted aggregates in those days, but at least ethernet could burst up to 10M or 100M |
| 16:00.42 | coppice | file: are you looking at playboy? |
| 16:00.59 | file | no, singing |
| 16:02.07 | *** join/#asterisk lorinc (n=ang@caracas-3168.adsl.interware.hu) |
| 16:03.40 | *** join/#asterisk crich1999 (n=crich@p54BFA60E.dip0.t-ipconnect.de) |
| 16:06.07 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 16:06.32 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 16:07.22 | asteriskmonkey | anyone failiar with the winpop stuff? |
| 16:07.33 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
| 16:07.36 | Seldon1975 | hey all |
| 16:07.49 | Seldon1975 | Im installing Debian for my new Asterisk box |
| 16:08.00 | Seldon1975 | should I use Kernel 2.4 or 2.6? |
| 16:08.08 | rob0 | asteriskmonkey: like "winpopup" or "smbclient -M"? |
| 16:08.17 | mut | 2.6... |
| 16:08.29 | Seldon1975 | mut: ok so it's stable enough |
| 16:08.30 | Seldon1975 | ? |
| 16:08.42 | mut | ya |
| 16:08.44 | asteriskmonkey | rob0: thats where i get lost you need smb got winpop do you not? |
| 16:09.34 | rob0 | Winpopup is a SMB thing, yes. You want to receive winpopup on Unix? |
| 16:12.14 | cpm | It's doable, I've done it, was like 10 years ago, and I think I used ytalk. |
| 16:12.55 | ctooley | http://linpopup2.sourceforge.net/ |
| 16:12.58 | asteriskmonkey | <PROTECTED> |
| 16:13.50 | asteriskmonkey | can i winpop on windows? |
| 16:13.53 | *** join/#asterisk CleanerX1 (n=nix@p54A3A704.dip0.t-ipconnect.de) |
| 16:14.21 | ctooley | asteriskmonkey, you need SMB/CIFS for Winpopup messages. On Windows this is provided by a lot of dfferent things, depending on your version of windows |
| 16:15.07 | asteriskmonkey | ctooley: so its nothing that comes with windows? |
| 16:15.28 | ctooley | If you want to do a Winpopup when you receive a call or something look at either linpopup2.sf.net or http://aplawrence.com/Unix/perlwinpopup.html |
| 16:15.37 | ctooley | asteriskmonkey, it is built into windows |
| 16:15.45 | asteriskmonkey | ok cool :) |
| 16:15.51 | ctooley | but it depends on which version of Windows you have as to what provides it. |
| 16:15.52 | austinnichols101 | asteriskmonkey: net send from the command line |
| 16:16.21 | asteriskmonkey | ah but isnt that stuff blocked not by default |
| 16:16.49 | austinnichols101 | asteriskmonkey: beware opening ports for that - there's a TON of stuff out there that targets winpopup |
| 16:16.51 | ctooley | asteriskmonkey, not a clue. I haven't used windows in years and I haven't used winpopup in a far longer time |
| 16:16.53 | austinnichols101 | asteriskmonkey: yes |
| 16:17.21 | austinnichols101 | asteriskmonkey: you couldn't pay me to enable it unless you could really control the senders |
| 16:17.21 | *** join/#asterisk FastJack (i=fastjack@p5091F8AF.dip.t-dialin.net) |
| 16:17.28 | asteriskmonkey | so is there some safe little proggi i can install on office machines that does winpop niceley and securly? |
| 16:17.35 | ctooley | asteriskmonkey, as austinnichols101 says though, there are quite a few virii and attacks out there that target winpopup |
| 16:18.02 | ctooley | asteriskmonkey, what is the problem you're actually trying to solve? |
| 16:18.13 | asteriskmonkey | ok winpop out the windows, is there a messanging system you can have installed in the tray? |
| 16:18.32 | ctooley | asteriskmonkey, you mean to send messages to users? |
| 16:18.33 | acehunky | hi |
| 16:18.35 | austinnichols101 | asteriskmonkey: our IPS equipment picks up one ever few seconds trying to enter the network - thousands per day |
| 16:18.37 | asteriskmonkey | im trying to have some form of live messaging to let people know a)soemone is calling and b)who it is |
| 16:19.04 | asteriskmonkey | yes damn people are hitting the broadcast address hoping to spam people |
| 16:19.08 | *** join/#asterisk kio (n=kio@195-11.customer.cloud9.net) |
| 16:19.27 | ctooley | asteriskmonkey, install a jabber server. install one of a hundred jabber clients, use the jabber client libraries to send messages to the user(s) |
| 16:19.35 | asteriskmonkey | mmm how uber would i be if i made one for msn |
| 16:19.43 | austinnichols101 | aasteriskmonkey: what about something like asterisk jabber. http://www.voip-info.org/wiki-Asterisk+Jabber |
| 16:19.49 | austinnichols101 | (haven't tried it) |
| 16:19.58 | ctooley | I use Jabber a lot |
| 16:20.17 | austinnichols101 | then you would get a built-in message log |
| 16:21.10 | austinnichols101 | I may try it this afternoon :-) |
| 16:21.10 | ctooley | asteriskmonkey, if you use Asterisk+Jabber you can use the jabber gateway modules to talk to MSN, AIM, Yahoo, IRC, or just about anything else. |
| 16:21.10 | austinnichols101 | asterksmonkey: found this link too: http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 16:21.10 | asteriskmonkey | youd need to have a jabber server installed though |
| 16:21.10 | ctooley | Winpopup is fugly, Jabber is good |
| 16:21.17 | ctooley | and? |
| 16:21.19 | asteriskmonkey | i agree thanks for the links guys |
| 16:21.23 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 16:21.32 | austinnichols101 | and get yourself a copy of trillian http://www.trillian.cc for the win boxen |
| 16:21.41 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 16:21.43 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 16:21.54 | austinnichols101 | ctooley: now all he has to do is worry about IM spam! |
| 16:22.30 | ctooley | austinnichols101, not if he's using Jabber... it not allow "internet" traffic to his Jabber server by not turning on S2S |
| 16:22.30 | *** join/#asterisk Darth`Tater (i=No@dsl-337.cascadeaccess.com) |
| 16:22.36 | nestar | mmm im spam |
| 16:22.52 | *** part/#asterisk kio (n=kio@195-11.customer.cloud9.net) |
| 16:23.02 | asteriskmonkey | yes i think for internall office jabber might be cool, but what about wan stuff |
| 16:23.03 | ctooley | It creates a private IM network |
| 16:23.36 | ctooley | asteriskmonkey, they can still use the Jabber server, you just have to be a user of that particular server to be able to send people on that server messages |
| 16:23.53 | ctooley | So WAN stuff still works, you just set them up accounts on the Jabber server and they log in remotely. |
| 16:23.54 | *** join/#asterisk websae (n=root@206.132.218.42) |
| 16:23.59 | asteriskmonkey | cool ill look into it more.. is the server stable... not going to eat memory up like pacman |
| 16:24.13 | ctooley | Our jabber server isn't even hosted in a state where we have any employees |
| 16:24.13 | austinnichols101 | I wonder if you could point it to your google account so you don't have to set up your own Jabber server |
| 16:24.25 | websae | anyone here familiar with e911 services and know of a good provider? very much appreciated! |
| 16:24.34 | ctooley | austinnichols101, you can if you wanted to |
| 16:24.40 | austinnichols101 | kewl |
| 16:24.45 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:24.54 | ctooley | asteriskmonkey, our jabber server has been running for months without restart the process |
| 16:24.57 | asteriskmonkey | damn it ill have to setup an e911 service for people |
| 16:25.01 | asteriskmonkey | cool |
| 16:25.12 | asteriskmonkey | ctooley: do you know much about centos? |
| 16:25.12 | websae | are you looking into it asteriskmonkey? |
| 16:25.30 | asteriskmonkey | yes i am goign to offer it and even offer it to others if i get it locked nicley |
| 16:25.36 | ctooley | asteriskmonkey, we run CentOS and RHEL version 4 |
| 16:25.48 | kink0 | my asterisk is just running in the inverse way i want, non registered peers/users are allowed to send calls, while registered users no. What to see ? |
| 16:25.50 | austinnichols101 | ctooley: I did an app a couple of months ago that lets me build IM 'bots' in ColdFusion 7. You could ask the bot questions or have it do stuff, etc. |
| 16:25.55 | asteriskmonkey | ctooley: I mainly run freebsd and the memory readout in top bothers me |
| 16:26.16 | asteriskmonkey | ctooley: it always says i have 7megs free the rest is cached .. is that normal of centos? |
| 16:26.17 | *** join/#asterisk evilbit (n=brad@interlink-gw1.ilsw.com) |
| 16:26.18 | austinnichols101 | ctooley: might be cool to build an IM realtime interface to asterisk CLI |
| 16:26.20 | websae | how do you guys like jabber server? |
| 16:27.00 | ctooley | asteriskmonkey, there's nothing wrong with it having all of the memory in "cache" that's a good thing. Cache is effectively "free" memory |
| 16:27.20 | ctooley | austinnichols101, for what? |
| 16:27.34 | ctooley | websae, we like it, it's convienent and the price is right. |
| 16:27.42 | asteriskmonkey | ctooley: ah thats good then .. sp what controls its shift to stay in cache vs stay free |
| 16:27.42 | austinnichols101 | ctooley: I dunno :-) |
| 16:27.50 | websae | free is a great price |
| 16:27.56 | websae | what client is good for windows? |
| 16:28.41 | darkskiez | Free is a better price :) |
| 16:29.04 | ctooley | asteriskmonkey, things that are "cache"d are something that's been read into memory for some reason (mostly from disk). It will keep it in memory until you need the memory for something else. That way if you need to read that part of the disk again and it hasn't changed, you don't have to do a read from disk, which makes the process much faster |
| 16:29.12 | cpm | someone elses client isgood for windows |
| 16:29.32 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:29.42 | evilbit | Anyone aware of a telecom (not asterisk specific) irc server/channel? I have q's regarding T1 demark extension. |
| 16:29.48 | ctooley | websae, gaim is what we use for windows and linux. I don't use windows so I don't know how well it works, but it's great for linux |
| 16:30.12 | websae | i wonder if there is a client for Palm OS |
| 16:30.17 | websae | i have a treo 650 |
| 16:30.20 | ctooley | webmind, yep |
| 16:30.38 | websae | do you know what it's called? |
| 16:30.42 | ctooley | almost every platform available has some kind of Jabber client |
| 16:30.57 | ctooley | websae, no but jabber.org has a link to "Clients" |
| 16:31.22 | websae | looking at it now, thank you |
| 16:33.29 | websae | $8/month ---DID and free incoming --that's pretty good right? |
| 16:34.06 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 16:36.11 | ctooley | That would be 1000 minutes of inbound for me. So it's probably decent. What's your outbound rate? |
| 16:36.51 | Makenshi | do you know of any java sip clients, similar to Xten's old X-Web? |
| 16:37.17 | darkskiez | http://www.google.co.uk/search?hl=en&q=java+sip |
| 16:37.30 | ctooley | http://www.voip-info.org/wiki/view/VOIP+Phones |
| 16:37.59 | Makenshi | i ask in the hope that someone here knows, since i've spent a long time trawling the web for one |
| 16:38.04 | Makenshi | and come up with nothing |
| 16:38.06 | *** join/#asterisk sch19 (n=sch19@adsl-223-233-112.mia.bellsouth.net) |
| 16:38.17 | Makenshi | i've looked in all the usual places |
| 16:38.28 | websae | 1.2-1.3cents/ min |
| 16:38.37 | websae | have two different routes |
| 16:39.22 | ctooley | A Java phone that does SIP? |
| 16:39.46 | ctooley | Like this one https://sip-communicator.dev.java.net/ that has sip in the name and lives at Java.net? |
| 16:40.11 | Makenshi | yes, it's not suitable for web use |
| 16:40.38 | darkskiez | Oh, now u want an applet |
| 16:40.38 | Makenshi | it's a standalone client |
| 16:40.40 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 16:40.50 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 16:41.12 | darkskiez | use java web start to load it then |
| 16:41.12 | Makenshi | darkskiez, that's what i said in the first place |
| 16:41.21 | Makenshi | have you ever seen x-web? |
| 16:41.30 | darkskiez | Makenshi: no, but u didnt say applet |
| 16:41.37 | darkskiez | Makenshi: would a java web start not do? |
| 16:42.31 | Makenshi | darkskiez, no, not really, unless you know of any java web start sip client that needs a minimal amount of configuration for the end user (ie microphone/speaker setup) |
| 16:43.53 | mut | does a polycom 601 have a web config interface? |
| 16:43.54 | masonf | is there any way to make the hold button on my phone play music on hold? |
| 16:44.04 | masonf | 501 does |
| 16:44.47 | blint | Has anyone here used asterisk in russia? I have a problem with international calling there. The prefix for that should be 8(wait for dialtone)10 (country code) (city code) (number). I did a 8w10, 8p10, even just dialing 8 and sending a rest as DTMF but it doesn't seem to work. All I get is silence. Any suggestions? |
| 16:46.45 | mut | whats the default username/password? |
| 16:47.16 | blint | I almost forgot - Russia has no DTMF, so I have pulsedial=yes on my Zap interface (TDM400P) |
| 16:47.37 | websae | anyone interested in termination $8/month unlimited incoming and one DID---1.3cents/min outgoing |
| 16:48.17 | mut | masonf? |
| 16:48.27 | redax | hi |
| 16:48.29 | mut | anyone know the default user/pass to polycom web config? |
| 16:48.51 | file | websae: I don't mean to sound rude, but other companies rarely advertise in here as this is more of a help channel... if you're offering services use the asterisk-biz mailing list |
| 16:48.56 | redax | is there a way to "Silently" record the channels to files? |
| 16:48.57 | [Atlas] | websae: iax? |
| 16:49.33 | websae | sip and iax |
| 16:50.07 | masonf | mut: Polycom 456 |
| 16:50.25 | *** join/#asterisk mroth_janitor (n=mroth@63.65.26.220) |
| 16:50.30 | mut | ah thx |
| 16:50.36 | mroth_janitor | we're using the rawplayer method of moh, and i see 311 zombies related to it...any thoughts on that? |
| 16:50.52 | `lyme | is it possible to give each extension their own queue? and be able to transfer calls into that queue? |
| 16:54.16 | RoyK | ~seen oej |
| 16:54.26 | jbot | oej <n=oej@apollo.webway.se> was last seen on IRC in channel #asterisk-doc, 1h 16m 8s ago, saying: 'He's very asian in regards to his photography'. |
| 16:54.41 | *** join/#asterisk gvag11 (n=gvag11@ipa229.5.tellas.gr) |
| 16:55.44 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 16:56.04 | wunderkin | mroth_janitor, like kevin just pretty much suggested, use native moh if you are on 1.2 |
| 16:56.37 | wunderkin | small tier 12 provider has NPA 570 DIDs availible. <-- tier 12 provider.. wow.. nice! thats a BIG number! must be good! |
| 16:56.45 | file | ha... |
| 16:56.49 | file | A+ on that |
| 16:57.11 | [TK]D-Fender | `lyme : Sure... though why you'd want to annoy people like that is beyond me... |
| 16:57.41 | Beirdo | because it's fun :) |
| 16:57.53 | `lyme | [TK]D-Fender: how would that be an annoyance? |
| 16:57.59 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 16:58.03 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 16:58.57 | wunderkin | he also said no refunds |
| 17:00.19 | mroth_janitor | wunderkin: any scalability (number of concurrent calls) impact of going to native moh? |
| 17:00.30 | mroth_janitor | we have plenty of memory, i believe the box will be cpu bound |
| 17:01.21 | *** join/#asterisk ^Howler_ (n=wwolfe@68-250-139-209.ded.ameritech.net) |
| 17:01.27 | docelm0 | wunderkin, I saw that earlier.. too funny |
| 17:01.58 | [TK]D-Fender | `lyme : Having people queueing up to bug you... |
| 17:02.04 | heison | ~seen JerJEr |
| 17:02.07 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 14d 19h 31m 12s ago, saying: 'thanks again'. |
| 17:02.07 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 17:02.07 | wunderkin | hehe :) |
| 17:02.17 | wunderkin | mroth_janitor, all of the calls are ulaw right? |
| 17:02.22 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:02.31 | mroth_janitor | yes, all ulaw calls |
| 17:02.36 | `lyme | [TK]D-Fender: lol its a trucking company, often we have 2/3 people call at once for the same person... |
| 17:02.37 | gvag11 | Anybody here who have managed to send ok with spandsp ? |
| 17:02.39 | mroth_janitor | all pcm digital recordings |
| 17:02.48 | mroth_janitor | all other codecs noloaded |
| 17:02.52 | wunderkin | i think abe is based on 1.0.9 if thats what you are using (thats what i heard at least) |
| 17:03.09 | mroth_janitor | yes, what else do you know about me <puts on tinfoil hat> |
| 17:03.16 | jarrod | is there a way to implement solid faxing with a legacy fax machine thru asterisk yet |
| 17:03.31 | *** join/#asterisk deb_user (n=Hypnotis@71-36-56-107.albq.qwest.net) |
| 17:03.36 | deb_user | hello |
| 17:03.45 | mroth_janitor | hi |
| 17:03.51 | deb_user | can anybody recommend a good iax client to run on a windows machine? |
| 17:04.06 | mroth_janitor | client = softphone? |
| 17:04.06 | wunderkin | native would be the best to use if you can, convert all of the moh files to ulaw, no external processes, no transcoding |
| 17:04.09 | MRH2 | hi can you mixmonitor to g729 format? |
| 17:04.20 | deb_user | yes... |
| 17:04.22 | mroth_janitor | wunderkin, i will look into it...thanks!! |
| 17:04.27 | deb_user | i mean a softphone |
| 17:05.08 | mroth_janitor | looks like firefly supports it |
| 17:05.19 | mroth_janitor | http://216.239.51.104/search?q=cache:Ecblx4-uBQAJ:www.voip-info.org/wiki-VOIP%2BPhones+iax+windows+softphone&hl=en&gl=us&ct=clnk&cd=1 |
| 17:05.48 | mroth_janitor | there are others there, but firefly is the only one we've played with |
| 17:06.24 | deb_user | there's a bunch |
| 17:06.26 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:06.27 | deb_user | and I knew about this page |
| 17:06.36 | deb_user | I was hoping someone could recommend one they've used |
| 17:06.53 | mroth_janitor | oh, firefly is pretty good in our experience |
| 17:07.09 | bjames | is there a tool I can use for the Operater on my Asterisk system? |
| 17:07.23 | bjames | so she could do transfers and drop calls as needed? |
| 17:07.39 | mroth_janitor | flash operator panel, maybe |
| 17:07.59 | deb_user | is it free? |
| 17:08.02 | deb_user | doesn't look free |
| 17:08.15 | mroth_janitor | my phone man just said: it should be |
| 17:08.16 | bjames | flash doesn't like my FreeBSD desktop |
| 17:08.25 | deb_user | nevermind |
| 17:08.29 | deb_user | it is free |
| 17:08.34 | mroth_janitor | aaaah. strange, no homestar runner for you |
| 17:08.36 | bjames | that might work though |
| 17:09.01 | mroth_janitor | i believe it does what you asked, so try it on a windows box... |
| 17:09.05 | bjames | they're using Windows in the fron office |
| 17:09.17 | bjames | I guess I'll have to |
| 17:09.29 | mroth_janitor | it's a pretty friendly interface if they aren't big techies |
| 17:09.41 | bjames | that's good, yeah they're secretaries |
| 17:09.52 | mroth_janitor | i think it installs with AMP, which is nice if you are running something like an office pbx |
| 17:09.53 | bjames | but we got about 70 phones here |
| 17:10.28 | sivana | what's the command to apply a patch? patch -p0 < file.patch ? |
| 17:10.31 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) |
| 17:11.01 | Math` | yeah |
| 17:11.32 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:11.37 | mroth_janitor | arrrrr |
| 17:12.05 | mroth_pirate | stop looking at me patch! |
| 17:16.18 | *** join/#asterisk dijit0 (n=eric@adsl-68-127-136-155.dsl.pltn13.pacbell.net) |
| 17:16.57 | *** join/#asterisk lpires (n=lpires@200.243.188.2) |
| 17:18.09 | dijit0 | i have this strange issue with my caller id, i am using idefisk with asterisk, but in iax.conf i set the caller id to what i want it to be, and if a caller id is defined in idefisk, the one in iax.conf will be dominant... but if i leave the caller id in idefisk blank, when i dial a number through nufone, it shows up as 9999991234... any suggestions??? |
| 17:18.57 | dijit0 | and the thing is, it works when i define a caller id in idefisk, although that caller id isnt being used, the one in iax.conf will then be used... |
| 17:18.59 | _Sam-- | why cant you just set the caller id in idefisk |
| 17:19.07 | deb_user | is there a web-based softphone? |
| 17:19.18 | websae | anyone here have e911 implemented--trying to find a good service provider |
| 17:19.38 | deb_user | you know, that I could use to interface an apache server with iax or sip and make calls via a web page? |
| 17:19.47 | dijit0 | well, thats the thing, i don't want the caller id to be dependent upon idefisk, whether i set the caller id there or not, i want the caller id i set in iax.conf to work no matter what |
| 17:19.50 | _Sam-- | or rather, what is the problem with setting the caller ID in idefisk as opposed to iax.conf? |
| 17:20.09 | _Sam-- | i think the callerid in the conf files will always be used last |
| 17:20.17 | _Sam-- | the phone caller id will always be used first, i think. |
| 17:20.22 | docelm0 | deb_user, you mean like a webcall back? |
| 17:20.46 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 17:20.51 | dijit0 | if a caller id is set in idefisk, then whatever is in iax.conf for the caller id will be set on an outgoing call |
| 17:20.53 | dijit0 | its weird |
| 17:21.04 | zoa | dijit0: how do you like idefisk ? |
| 17:21.06 | Flauto | Feb 2 11:01:12 WARNING[3706]: chan_sip.c:9599 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 17:21.09 | deb_user | docel: no, I mean instead of using x-lite or firefly or something, just visit www.example.com/phone and I could make all my phone calls via the webpage |
| 17:21.11 | Flauto | i always get this |
| 17:21.13 | Flauto | what does it mean |
| 17:21.20 | dijit0 | so far so good... :) |
| 17:21.27 | dijit0 | id just like to fix this problem |
| 17:21.54 | _Sam-- | zoa wake up |
| 17:22.00 | dijit0 | cause if i want my friends to connect to my asterisk, i dont want htem being able to set the caller id to whatever they want, lol... theyd be up to no good |
| 17:22.01 | Flauto | i noticed that the message is showing when my internet traffic is busy |
| 17:22.03 | _Sam-- | help dijit with ideifsk |
| 17:22.04 | MRH2 | hmmm e911 why don't they just let you put the geolocation in the callerid or something. |
| 17:22.07 | docelm0 | deb_user, ohh ok.. I dont know unless you can find a java softphone that can run from a website.. other wise your looking at a activeX control |
| 17:22.32 | deb_user | a java softphone would be cool |
| 17:22.50 | _Sam-- | broadvoice has a web based softphone if you use them |
| 17:23.03 | deb_user | sam: I want to host it on my apache server |
| 17:23.09 | RoyK | ~sip |
| 17:23.10 | jbot | i guess sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 17:23.10 | _Sam-- | you will have to write your own then |
| 17:23.33 | dijit0 | i dont know if its a Dial issue or what... because what i set in iax.conf for setcallerid= will be what the callerid says within my own network... but, once it dials out through nufone, it will only work if there is SOMETHING in idefisks callerid, doesnt matter what is there, but if it is blank, the one in iax.conf will not work |
| 17:23.37 | dijit0 | i mean, does that make sense? |
| 17:23.47 | _Sam-- | you use setcallerid in iax.conf? |
| 17:23.58 | dijit0 | yes... |
| 17:24.00 | deb_user | https://sip-communicator.dev.java.net/ |
| 17:24.07 | deb_user | says its not ready to go yet |
| 17:24.09 | *** join/#asterisk MatsK (n=mk@84-217-5-20.tn.glocalnet.net) |
| 17:24.10 | deb_user | still alpha |
| 17:24.17 | deb_user | cool idea though |
| 17:24.35 | _Sam-- | that could be part of it |
| 17:24.40 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 17:24.43 | _Sam-- | i think the proper syntax for iax.conf is |
| 17:24.43 | _Sam-- | ;callerid="Some Host" <(256) 428-6011> |
| 17:25.12 | dijit0 | umm, are the () and the - in the number part required? |
| 17:25.20 | _Sam-- | no |
| 17:25.37 | _Sam-- | this is mine for iax.conf: |
| 17:25.40 | websae | e911 services--anyone? |
| 17:25.49 | _Sam-- | callerid="Sam" <877294XXXX> |
| 17:25.56 | dijit0 | yea, thast more like mine |
| 17:26.05 | _Sam-- | but you said you use SETCALLERID |
| 17:26.07 | _Sam-- | that is no go |
| 17:26.08 | _Sam-- | i think |
| 17:26.14 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:26.20 | zoa | sam, it doesnt sound like an idefisk problem |
| 17:26.26 | zoa | it sounds like a problem on asterisk |
| 17:26.30 | dijit0 | well, mine seems to WORK... but ONLY if something is defined in the idefisk caller id |
| 17:26.32 | _Sam-- | hi! |
| 17:26.35 | dijit0 | even a space can be there |
| 17:26.39 | dijit0 | then it will all work... |
| 17:26.53 | _Sam-- | dijit0: zoa is one of the main idefisk developers...if anyone can help you it will be him! |
| 17:26.56 | _Sam-- | but not me! |
| 17:27.14 | *** join/#asterisk roulduke_ (i=2qxmef1w@p508D1477.dip0.t-ipconnect.de) |
| 17:27.17 | _Sam-- | but i do believe setcallerid is no good in iax.conf |
| 17:27.24 | _Sam-- | needs to be callerid |
| 17:27.35 | dijit0 | oh wait, i used callerid, nm, lol |
| 17:27.58 | dijit0 | yes, i have callerid, thats why it works on my network, but i need it to work when dialing out through nufone |
| 17:28.17 | dijit0 | not depending on whether the iaxclient or whatever sets a caller id or not |
| 17:28.35 | _Sam-- | dunno...i use iax w/ outgoing caller id all the time just fine |
| 17:28.48 | _Sam-- | and i use idefisk also sometimes |
| 17:28.53 | dijit0 | i even did NoOp to check all my callerid variables |
| 17:28.59 | _Sam-- | w/ nothing set in the caller id of idefisk |
| 17:29.04 | _Sam-- | and my caller id from iax.conf works fine |
| 17:29.06 | SplasPood | Is there any way to group sip users so they can only pick up calls users within their group parked? |
| 17:29.20 | *** join/#asterisk j4m3s_ (n=j4m3s@146.229.189.146) |
| 17:29.36 | *** join/#asterisk tero (n=cool_man@85.112.65.72) |
| 17:29.40 | dijit0 | well, it must be somehow sending an invalid callerid out to nufone or something |
| 17:30.16 | _Sam-- | maybe with nufone you can only set your caller id to the real phone number you are registered with them |
| 17:30.19 | _Sam-- | i have no idea really |
| 17:30.29 | dijit0 | nah... i tried iax.cc |
| 17:30.37 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:31.03 | _Sam-- | zoa: if caller id is set in fisk, and in iax.conf...which takes precedence? |
| 17:31.29 | dijit0 | thats what im saying... if theyre both set,,, the one in iax.conf is the one that is used |
| 17:31.37 | tero | hi |
| 17:31.47 | dijit0 | but if idefisks ISNT set, then the iax.conf one has no effect once i dial out |
| 17:32.07 | _Sam-- | i can confirm that my idefisk/asterisk over IAX for outgoing calls doesnt work that |
| 17:32.08 | tero | hi |
| 17:32.15 | _Sam-- | er doesnt work that way |
| 17:32.27 | _Sam-- | my idefisk is blank for caller id, and my outgoing calls show my iax.conf callerid fine |
| 17:32.31 | _Sam-- | just called my cell |
| 17:32.46 | dijit0 | really.... hmm |
| 17:32.52 | _Sam-- | what asterisk version |
| 17:32.56 | dijit0 | 1.2.3 |
| 17:33.11 | dijit0 | the local phone network around here seems to set 9999991234 if the number is not valid in length |
| 17:33.37 | dijit0 | ill try this real quick, then check my nufone records and see what it says |
| 17:33.39 | websae | e911 is a pain in the butt |
| 17:33.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:33.48 | websae | anyone have any suggestions? |
| 17:34.00 | zoa | the iax.conf takes precendese |
| 17:34.06 | zoa | or however you spell that |
| 17:34.16 | zoa | hi oej, i saw you looked for me ? |
| 17:34.19 | websae | exit |
| 17:34.19 | _Sam-- | thanks |
| 17:34.19 | zoa | precedence |
| 17:34.29 | _Sam-- | hey zoa, someone yesterday was looking for some call cetner apps |
| 17:34.37 | _Sam-- | i told them you were the man, but i forget where they went |
| 17:34.45 | mdave | BV CAN KISS MY FAT WHITE ASS |
| 17:34.48 | zoa | aha, send em to me then or to aheeva |
| 17:34.49 | mdave | their network is SUCH a POS |
| 17:35.04 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 17:35.05 | [TK]D-Fender | Aheeza? ... $$$$ |
| 17:35.16 | [TK]D-Fender | Aheeva* |
| 17:35.21 | _Sam-- | zoa: is your call center stuff ready for me to sell? for example, if i had a client that wanted certain things from it, its ready to go? |
| 17:35.33 | zoa | no not yet, but it depends on what he needs currently |
| 17:35.48 | _Sam-- | skill is done? |
| 17:35.51 | zoa | yes |
| 17:35.59 | zoa | but the gui is needed, but i could finish that faster |
| 17:35.59 | mdave | id like to use dialpad for outbound, but i need to find a number for inbound, ideally that doesnt charge per-minute |
| 17:36.05 | tero | hiiiiiiiiiiii |
| 17:36.07 | oej | zoa: That is over... talked with your secretary. Don't need you any more. |
| 17:36.08 | sivana | anyone know if asterisk.conf, rtp.conf, cdr.conf are mandatory? |
| 17:36.10 | mdave | in the +1-616 area |
| 17:36.20 | _Paulo_ | hi tero. |
| 17:36.35 | zoa | ah k |
| 17:36.40 | tero | hi |
| 17:36.45 | zoa | she's lovely, isnt she ? :) |
| 17:37.22 | *** join/#asterisk Delvar (n=irc@host-83-146-53-40.bulldogdsl.com) |
| 17:37.45 | deb_user | anybody here know how I would interface sugarcrm with an asterisk installation? |
| 17:39.38 | [TK]D-Fender | "Not a chance" |
| 17:39.55 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 17:42.48 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
| 17:43.05 | mut | is there any documentation on how to use the polycom web interface? |
| 17:43.06 | *** join/#asterisk ToTo (n=ToTo@host94-157.pool874.interbusiness.it) |
| 17:43.47 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:44.21 | BlueDevi1 | feature request: agents from realtime database |
| 17:46.14 | [TK]D-Fender | mut : to be avoided.... provisioning is the way to go.... |
| 17:46.16 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 17:51.39 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 17:52.16 | kink0 | how is possible that non-registered users/peers can send calls while registered users are reported as INVITE Proxy Authentication and are not allowed ? |
| 17:52.34 | oej | kink0: bad configuration |
| 17:52.39 | kink0 | I was pretending just the inverse... |
| 17:52.39 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 17:53.14 | kink0 | oej: yeahhh I know, but I have changed my config several times, and still no running as I was expecting, so I think is time to go for some help |
| 17:53.16 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 17:53.34 | oej | kink0: have you read the asterisk book? |
| 17:53.36 | oej | :-) |
| 17:54.00 | oej | Hint: Check what you can access in the default context under [general] in sip.conf |
| 17:54.03 | kink0 | oej, yes, sure. I have edited several times my sip.conf at both asterisk's box |
| 17:54.08 | oej | I guess that is pretty open |
| 17:54.22 | oej | And that is why unknown people can call without auth |
| 17:54.28 | kink0 | oej, yes, now I set my extensions again in default context |
| 17:54.36 | oej | Right |
| 17:54.49 | oej | And that's propably what you have in the [generral] section as well |
| 17:54.52 | oej | An open hole |
| 17:54.55 | kink0 | yes, that is right, but how to allow INVITE to registered users ? |
| 17:54.59 | oej | Saying "welcome, feel free to call" |
| 17:55.21 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 17:55.23 | kink0 | oej, yes, I know I have open that hole now, I set so to test if no other thing was wrong |
| 17:55.23 | oej | Check that username and auth user name are the same, then make sure they have the same secret |
| 17:55.27 | oej | file: Peek |
| 17:55.33 | file | eep |
| 17:55.35 | oej | file: Booking tickets tomorrow! |
| 17:55.49 | oej | file: You booked hotel already |
| 17:56.13 | kink0 | oej: I set registry=user:pass in sip.conf and same user:pass on [user] friend on the other asterisk |
| 17:56.25 | kink0 | registration then goes ok, but INVITE no. |
| 17:56.28 | file | oej: I'm with Kevin |
| 17:56.35 | file | so I didn't have to :P |
| 17:57.16 | kink0 | oej: what also is required in sip.conf to allow INVITE for the peers/friend ? |
| 17:57.22 | blitzrage | oej: secret metting? |
| 17:57.26 | blitzrage | meeting* |
| 17:57.44 | oej | blitzrage: Oh yes, very secret - von2006 ... |
| 17:57.47 | sivana | anyone know if asterisk.conf, rtp.conf, cdr.conf are mandatory config files? |
| 17:57.57 | blitzrage | oej: ahhhh yes -- is that soon? |
| 17:58.10 | oej | If you have no asterisk.conf - we will use the compile time defaults, so it's not mandatory |
| 17:58.20 | file | blitzrage: March |
| 17:58.24 | blitzrage | file: ah |
| 17:59.04 | kink0 | will I need to set domain, real, or so in my sip.conf to allow registered peers to do calls ? ( well as I configured it, dont work ) |
| 17:59.38 | kink0 | I set the same realm, set both domains, and so... but INVITES doesn't work and report just until Proxy Authetication Required message. |
| 17:59.45 | kink0 | ( I am ussing SIP ) |
| 18:03.02 | file | fiddle sticks... |
| 18:06.39 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:06.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:08.45 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:09.08 | *** join/#asterisk ncjp_ (n=switch@61.206.115.5.user.ad.il24.net) |
| 18:11.28 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 18:11.28 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 18:11.28 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 18:11.28 | *** join/#asterisk Triffid_Hunter (n=Splat@funkmunch.net) |
| 18:11.28 | *** join/#asterisk wilymage (n=wily@funkmunch.net) |
| 18:11.28 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 18:11.29 | *** join/#asterisk artichoke (n=artichok@c220-237-68-248.kelvn1.qld.optusnet.com.au) |
| 18:11.29 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 18:11.29 | *** join/#asterisk angler_ (n=angler@24.214.255.222) |
| 18:12.36 | *** join/#asterisk RazorJack (i=RazorJac@205.211.153.20) |
| 18:13.31 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 18:13.35 | RazorJack | I am using asterisk 1.2.3 with same version of zaptel. Have a WCTDM connecting to an asterisk pap2 and have to use fxs_ls to make it work. If I use fxs_Ks, asterisk works fine but no dialtone on that channel.... fxs_ls should work, but crashes asterisk... any ideas? |
| 18:14.06 | *** join/#asterisk switch_ (n=switch@61.206.115.5.user.ad.il24.net) |
| 18:14.44 | justinu | RazorJack: make sure your signalling in zaptel.conf matches zapata.conf |
| 18:15.13 | elg | anyone doing spa provisioning with XML? |
| 18:15.25 | RazorJack | justinu; hmmmm, 1sec, never though to dbl check that, im not sure that someone hasnt changed in the make install |
| 18:15.30 | *** join/#asterisk Prival (n=someone@64.235.216.178) |
| 18:16.33 | RazorJack | justinu; I have to restart the zaptel right? |
| 18:16.48 | justinu | no, just run ztcfg |
| 18:17.01 | Prival | Hi, anyone got the following. Asterisk 1.09, PRI connection, when you call to a specific number, you can't navigate the menus on the other PBX (unknown model/make). Tried from an asterisk 1.09 with an analog connection with the same phone model and it works... |
| 18:17.59 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 18:18.20 | RazorJack | justinu; thx, its working now! something modified my zaptel.conf and didnt notice the l/k difference .... |
| 18:18.26 | justinu | cool |
| 18:19.19 | Makenshi | great i have my dns record for TRIP set up |
| 18:19.25 | Makenshi | test number is 505*302 |
| 18:19.30 | *** join/#asterisk thosa (n=thosa@p54879ADE.dip0.t-ipconnect.de) |
| 18:20.28 | *** join/#asterisk Shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 18:22.11 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 18:23.17 | [TK]D-Fender | Prival : How is the phone connecting to *? |
| 18:24.12 | sivana | is adsi obsolete? |
| 18:24.36 | [TK]D-Fender | sivana : All things analog are obsolete... |
| 18:24.46 | sivana | good point :) |
| 18:25.20 | Prival | TK: Aastra 9133i, through a Cisco switch |
| 18:26.11 | [TK]D-Fender | Prival : Are you sure your DTMFMODE settings are right for it? |
| 18:26.51 | *** join/#asterisk redondos (n=redondos@12-207-132-99.client.mchsi.com) |
| 18:27.16 | *** join/#asterisk secure75 (n=mic@dslb-084-057-024-106.pools.arcor-ip.net) |
| 18:27.18 | Prival | TK: rfc2833 |
| 18:27.41 | Prival | TK: When you press the keypad keys to access the menus, there is no dtmf issues, almost like the other PBX sends a blip to disable the keypad. But from my cell phone it works. Also from the same network, other external calls works... |
| 18:28.11 | Prival | TK: Only call to this perticular number fait and from any phone on this network. |
| 18:28.47 | Prival | TK: also, we tried the same number from the same phone model from the office (we have an analog connection instead of a PRI) and it works. |
| 18:29.27 | *** join/#asterisk YARICK (n=spiderma@pool-71-246-95-171.bltmmd.east.verizon.net) |
| 18:29.31 | redondos | What degradation/bad performance could I expect from a clone of a Digium TE205P? Is the original worth the extra $300? |
| 18:29.41 | [TK]D-Fender | :/ |
| 18:31.30 | Qwell[] | redondos: I'd say so |
| 18:31.43 | Qwell[] | The support alone is worth that |
| 18:31.51 | thazza | yyyyyyyyyyyyyyyyyyyyyyyyyyyyy7uhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhh;['']\[ |
| 18:32.02 | [Atlas] | compiling asterisk on slackware and it fails - I have the message at http://pastebin.com/535669 can someone take a look and help me see what im missing? |
| 18:32.20 | justinu | install openssl |
| 18:32.34 | [Atlas] | omg that was fast! |
| 18:32.50 | [Atlas] | im pretty sure we have it installed though |
| 18:32.51 | justinu | thank you, come again |
| 18:32.59 | Netgeeks | lol |
| 18:32.59 | [Atlas] | ill check |
| 18:33.02 | justinu | if it's installed, it's not finding it |
| 18:33.08 | rob0 | FWIW I have no trouble on various Slackwares. |
| 18:33.09 | [TK]D-Fender | NEXT!!!! (c) BKW |
| 18:33.13 | justinu | heh |
| 18:33.23 | [TK]D-Fender | same here..... |
| 18:33.25 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 18:33.39 | [TK]D-Fender | Then again I like the "everything" button a lot on the install :) |
| 18:33.49 | rob0 | definitely ... fastest that way |
| 18:34.01 | fiber0pti | how do I truncate an extension that's dialed? I want to have users dial a star followed by the extension to page that extension so _*XXXX but i need to pull out the XXXX how do I do that? |
| 18:34.08 | [TK]D-Fender | rob0 : So rare to get a decent distro with 2 CD's these days.... |
| 18:34.10 | rob0 | (leave off kdei and x* on servers) |
| 18:34.12 | [Atlas] | ls /var/log/packages/ | grep -i ssl returns openssl-0.9.7b-i386-1 |
| 18:34.21 | Netgeeks | did you know the skill required to configure asterisk for a basic office PBX is about the same level of skill required to perform an 'ice-pick' lobotomy?..... |
| 18:34.30 | [TK]D-Fender | fiber0pti : ${EXTEN:1} |
| 18:34.44 | fiber0pti | thanks! |
| 18:34.52 | [TK]D-Fender | rob0 : Whatcha talking about" I run KDE, SAMBA, and a ton of other shit on my server :) |
| 18:35.12 | rob0 | And actually the 'ice-pick' lobotomy helps enhance the skills ;) |
| 18:35.23 | [TK]D-Fender | Netgeeks : I'd rather have a bottle infront of me than a frontal lobotomy :) |
| 18:36.07 | Netgeeks | now writing dynamic html code is much more like having a lobotomy... |
| 18:36.59 | [av]bani | \o/ |
| 18:37.32 | justinu | how erotic |
| 18:37.44 | [TK]D-Fender | sivana : With HD's these days (and since many years) I'd hardly worth thinking about not installing almost everything. At least everything non-X related. |
| 18:38.06 | dpryo | What the hell? |
| 18:38.09 | dpryo | Are you serious? |
| 18:38.23 | [TK]D-Fender | Yeah.... whats the "\o/" mean anyways? |
| 18:38.33 | _Sam-- | looks like a guy with his arms up |
| 18:38.48 | justinu | i saw something entirely different :P |
| 18:38.51 | rob0 | haha |
| 18:39.07 | [TK]D-Fender | justinu : Get your foot out of the gutter.... you're standing on my head :) |
| 18:39.24 | justinu | heh |
| 18:39.41 | Netgeeks | I saw 'escape code' o forward slash! |
| 18:39.58 | *** join/#asterisk Flyboy-SR22 (i=rsears@gateway.adnc.com) |
| 18:40.04 | [Atlas] | sivana: where might i come across those package lists :) ? |
| 18:40.26 | Flyboy-SR22 | Hey Everyone |
| 18:40.28 | Qwell[] | offtopic: How much does L2 cache help on an athlon64? |
| 18:40.32 | *** join/#asterisk none (n=nix@p54A3A704.dip0.t-ipconnect.de) |
| 18:40.33 | [Atlas] | i really think i can make it work if there is a way to pass to asterisk where to look for ssl libs |
| 18:40.50 | justinu | you'd have to edit the make file |
| 18:40.56 | [TK]D-Fender | [Atlas] : You installed yours serperately somehow? |
| 18:41.13 | [Atlas] | TK: eh? |
| 18:41.27 | Qwell[] | is it worth an extra $100 for 1MB per core instead of 512KB per core? |
| 18:41.27 | *** join/#asterisk antoniofcano (n=antoniof@194.Red-83-38-44.dynamicIP.rima-tde.net) |
| 18:41.28 | [TK]D-Fender | Your SSL libs aren't in the normal place? |
| 18:41.46 | [Atlas] | not sure at one point someone here made thier own package |
| 18:41.56 | justinu | i'd go for it, L2 cache is l337 |
| 18:41.59 | [Atlas] | its very rare that our sytems (if ever) have default slack packages |
| 18:42.02 | [TK]D-Fender | [Atlas] : And you can't install the standard one on top? |
| 18:42.09 | Qwell[] | only problem, is the one I'd be getting is a first gen |
| 18:42.35 | [TK]D-Fender | Qwell[] : Putting the "bleed" back into bleeding edge :D |
| 18:42.41 | [Atlas] | i suppose i can -- think i would much rather have one lib and just edit the makefile though |
| 18:42.48 | Qwell[] | It's either 2nd gen Manchester with 512, or first gen Toledo with 1mb |
| 18:43.05 | Qwell[] | or an extra $300 for a 2nd gen Toledo :D |
| 18:45.14 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 18:46.31 | Netgeeks | I've got a X2 4800+ system, but don't have any 512MB L2 to compare it against... |
| 18:46.55 | Qwell[] | They lowered the L1 on Opterons? wtf? |
| 18:47.15 | justinu | 512mb L2 would be insane |
| 18:47.28 | justinu | most I ever worked with was 8MB :P |
| 18:47.39 | [Atlas] | sparc? |
| 18:48.32 | justinu | yep |
| 18:48.42 | tracinet | recently upgraded to 1.2x from 1 year-old 1.0X CVS and so far so good - just noticed that when using the # sign to do a transfer and you dial the extension, you can no longer press the # sign at the end to make the tranfer go (you must wait for the digit timeout) - anyone know of a workaround to re-enable the ability to press #? |
| 18:48.44 | Netgeeks | hehe, I've got a Sun E4500 with 12 procs (8MB cache) and 12 G of ram in shippment to me, gonna see how asterisk runs on it and what kind of load it can handle |
| 18:48.56 | *** join/#asterisk groogs (n=greg@d226-27-136.home.cgocable.net) |
| 18:49.31 | justinu | yeah, i've got a 4 CPU E4500 |
| 18:49.36 | justinu | solid machine |
| 18:50.01 | Netgeeks | solid is right, shipping company says this one weighs in at just under 200 lbs |
| 18:50.04 | mzo | but does it play world of warcraft? :P |
| 18:50.06 | asteriskmonkey | is there a command to see how many bogomips a cpu can do in linux? |
| 18:50.17 | Netgeeks | world of warcraft is sooo yesterday |
| 18:50.23 | Beirdo | cat /proc/cpuinfo |
| 18:50.30 | mzo | i know, every geek plays nethack now anyway |
| 18:50.30 | Flyboy-SR22 | cat /proc/cpuinfo |
| 18:50.37 | Netgeeks | :) |
| 18:50.50 | justinu | wait till you have a look at the CPU boards, netgeeks |
| 18:51.08 | asteriskmonkey | mzo: whats nethack? |
| 18:51.18 | Netgeeks | I'm familiar with them, we had two 10ks at a company I worked at... same boards |
| 18:51.20 | mzo | heh google it :P it's good |
| 18:51.24 | Beirdo | Netgeeks: I hope you have a rack in a machine room for that puppy |
| 18:51.44 | [Atlas] | im like so last week then b/c im still addicted to DAoC ;p |
| 18:51.45 | Netgeeks | Beirdo? rack in a machine room? I got a cement floor in my garage |
| 18:51.50 | Beirdo | heh |
| 18:51.55 | Qwell[] | asteriskmonkey: bogomips are a bogus number |
| 18:51.55 | Beirdo | that's a good start, I guess |
| 18:52.03 | Qwell[] | on x86, it's almost always 2x the speed of the proc |
| 18:52.15 | Qwell[] | bogomips = "bogus mips" |
| 18:52.28 | asteriskmonkey | ah ok :D |
| 18:52.32 | Beirdo | Qwell[] past a PentiumPro or so, I think |
| 18:52.45 | twisted[asteria] | Qwell, not EXACTLY right, but close. |
| 18:52.46 | twisted[asteria] | http://www.catb.org/jargon/html/B/BogoMIPS.html |
| 18:52.47 | Beirdo | before that, the multiplier varied. Or was it Pentium? |
| 18:53.08 | Beirdo | anything we'd be using these days anyways :) |
| 18:53.29 | Qwell[] | twisted[asteria]: yeah...I just meant that it didn't really mean anything |
| 18:53.59 | [Atlas] | ok so im perusing the Makefile and the only 3 references to ssl are in an if block that goes ifeq ($(OSARCH),SunOS),, thats not what im looking for is it? |
| 18:54.11 | fiber0pti | Anyone have experience with the auto answer functionality on the polycoms? |
| 18:54.18 | asteriskmonkey | so it cant be used to compare which system is faster |
| 18:54.22 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:54.25 | Qwell[] | asteriskmonkey: right |
| 18:54.40 | Beirdo | heh |
| 18:54.40 | Netgeeks | kinda like knowing the horsepower of an engine in a car without knowing torque, powerband, and weight of the car... it gives you some information, but not enough to fully estimate performance? |
| 18:54.46 | Beirdo | it CAN, but it's meaningless for that |
| 18:54.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:54.57 | Beirdo | if they aren't the same architecture, it's meaningless |
| 18:55.01 | mzo | processor stats are like penis length :P |
| 18:55.02 | fiber0pti | I've followed the directions at http://www.voip-info.org/wiki/view/Polycom+auto-answer+config. In the CLI I see the variable Alert_Info being set to the right context in the sip.cfg but the phone just rings |
| 18:55.20 | *** join/#asterisk razu_ (n=razu@ip59.cab62.mus.starman.ee) |
| 18:55.25 | asteriskmonkey | so is there a linux dyno |
| 18:55.30 | Qwell[] | mzo: They only matter when you run Gentoo? |
| 18:55.36 | mzo | hahaha probably |
| 18:55.41 | Netgeeks | there are quite a few linux dynos |
| 18:55.42 | bsdfreak | linux dynamo |
| 18:55.47 | bsdfreak | powered by wind |
| 18:56.20 | Qwell[] | Why the crap would AMD lower the L1 cache on their opterons? |
| 18:56.42 | justinu | cheaping out |
| 18:57.03 | twisted[asteria] | Qwell, http://www.clifton.nl/ar01s03.html#id2449461 |
| 18:57.06 | twisted[asteria] | heh |
| 18:57.08 | _Sam-- | hey brad_mssw you around bud? |
| 18:57.28 | Qwell[] | twisted[asteria]: slow proxy |
| 18:57.39 | twisted[asteria] | Qwell, ? |
| 18:57.40 | niZon | Using realtime, if I add a SIP device to the queue_members_table, why do I still have to reload asterisk to get it to recognize the entry? |
| 18:57.46 | Qwell[] | taking forever to load |
| 18:57.58 | Qwell[] | niZon: shouldn't, no |
| 18:58.09 | Qwell[] | twisted[asteria]: ahh, heh |
| 18:58.13 | niZon | it doesn't seem to work unless I do |
| 18:58.26 | [Atlas] | anyone know why the only references to SSL in the Makefile are in ifeq ($(OSARCH),SunOS) |
| 18:58.28 | [Atlas] | ? |
| 18:59.07 | Netgeeks | did they lower the prices of the opterons as well? they are probably trying to create an 'entry' level opteron option... just like you have the 512K x2's and the 1M x2's |
| 19:00.07 | asteriskmonkey | i want an itanium with 16meg cache |
| 19:00.21 | *** join/#asterisk glnx (n=chatzill@62.37.237.17) |
| 19:00.28 | Beirdo | I want a 96-way Sparc box |
| 19:00.39 | Beirdo | anyone want to pay the cooling and electric bills? |
| 19:00.40 | asteriskmonkey | those are cool |
| 19:00.42 | Beirdo | and buy it for me? |
| 19:00.55 | [Atlas] | Guess no one knows... |
| 19:00.58 | [Atlas] | :D |
| 19:01.11 | Qwell[] | Netgeeks: Well, they made 939 opterons |
| 19:01.18 | Netgeeks | 110W per cpu chip |
| 19:01.19 | Qwell[] | They lowered the cache at the 170 though...bastards |
| 19:01.28 | Netgeeks | hrm |
| 19:01.41 | tracinet | when doing a blind transfer, anyone know how to allow the pound sign at the end of the ext number to make the call go rather that wait for the timeout? |
| 19:01.44 | Netgeeks | Qwell: check the fab location |
| 19:02.27 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
| 19:04.10 | cnet2 | hi.. I have 2 offices connected through asterisk (trhough internet) iax channel. And the sound is awful, big delays,.. I monitored ping times to yahoo, and it settles at 140ms, but when i start a conversation, the ping times go up to 3000ms!, anyone know why could this be? (internet connection is of about 512 each side) |
| 19:04.45 | tracinet | cnet2 - what verion are you running? |
| 19:04.46 | cpm | sounds like congestion. and 140 isn't very good. |
| 19:04.47 | Netgeeks | who is your provider for each office, what type of connection is it (dsl, cable, T1, etc.)? |
| 19:04.52 | cnet2 | 1.02 |
| 19:05.24 | cnet2 | 1.09, i don't know.. i know is not 1.2 |
| 19:05.24 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 19:05.24 | cnet2 | dsl each side |
| 19:05.26 | tracinet | i had bad iax quality as well with my version of asterisk prior to upgrading to 1.2x |
| 19:05.46 | tracinet | since upgrading to 1.2x quality has improved in a big way |
| 19:05.48 | cnet2 | ok. I'll do the upgrading |
| 19:05.52 | _Sam-- | bkw_: how long should it take to get the confirmation email from asterlink after signing up |
| 19:05.56 | Netgeeks | what is your normal traffic like? do you keep track (like using mrtg, or some other monitoring system) of your capacity usage? |
| 19:05.57 | cnet2 | thanks .. |
| 19:06.04 | tracinet | only problem is that you need to make sure notransfer=yes |
| 19:06.27 | cnet2 | transfer=es yes? |
| 19:06.31 | cnet2 | ohh in the new version? |
| 19:06.34 | tracinet | yes |
| 19:06.36 | tracinet | in iax.conf |
| 19:06.36 | cnet2 | ok |
| 19:06.44 | tracinet | found a bug when set to no |
| 19:06.53 | Hmmhesays | here goes nothing |
| 19:06.55 | tracinet | need to do more testing before opening up bug report |
| 19:07.11 | Netgeeks | get your polycom to stop ringing, Hmm? |
| 19:07.18 | Hmmhesays | cnet, modem buffer is filling up |
| 19:07.20 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 19:07.23 | Hmmhesays | Netgeeks, nope |
| 19:07.30 | cnet2 | ..you think? |
| 19:07.34 | Hmmhesays | I posted on the mailing list, waiting for some logs from the customer |
| 19:07.39 | cnet2 | that would me too much traffic |
| 19:07.42 | Hmmhesays | cnet2, yes |
| 19:07.53 | cnet2 | ..but we have almost no traffic at all |
| 19:07.59 | Hmmhesays | do a throughput test for upload |
| 19:07.59 | cnet2 | i'll do some monitoring |
| 19:08.18 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) |
| 19:08.19 | Hmmhesays | slam that bi@tch with udp packets |
| 19:08.47 | Hmmhesays | while pinging google |
| 19:09.01 | t0ke | anyone know any ip phone which support IAX protocol? tnks |
| 19:09.01 | Hmmhesays | or yahoo, or brazilianwax.com |
| 19:09.07 | justinu | heh |
| 19:09.27 | Netgeeks | or even better since you control both ends get one of the server/client mode bandwidth analyzers and you can see the exacp performance of your udp throughput |
| 19:09.39 | Netgeeks | with the right tool, you can even measure jitter |
| 19:09.55 | *** join/#asterisk Hymie (i=hymie@L8R.net) |
| 19:10.26 | Hmmhesays | yeah I was just giving the poor mans test |
| 19:10.46 | bsdfreak | heh |
| 19:10.53 | Hmmhesays | for those of us who don't have those fancy tools like, a keyboard |
| 19:11.01 | Netgeeks | ! |
| 19:11.29 | Hymie | does anyone know why I'm getting names in CALLERIDNUM, in Asterisk 1.2.0? |
| 19:11.58 | Hymie | CALLERID is supposed to be name and number, CALLIDNAME is supposed to be name, and CALLERIDNUM is supposed to be the number, yes? |
| 19:12.22 | Netgeeks | assuming you didn't mean the mispelling, yes |
| 19:12.36 | Qwell[] | those are deprecated anyhow |
| 19:12.39 | Hymie | hmm.. it's weird |
| 19:12.41 | Hymie | eh? |
| 19:12.43 | Qwell[] | use CALLERID(num) |
| 19:12.45 | Hymie | in 1.2.0? |
| 19:12.56 | Hymie | I don't know what the number is |
| 19:13.02 | Hymie | I'll trying to get it, not set it... |
| 19:13.03 | Hmmhesays | Set(CALLERID(num)) |
| 19:13.10 | Qwell[] | ${CALLERID(num)} |
| 19:13.54 | Hymie | hmm, wiki has no info I can find yet.. is this valid in 1.2.0? |
| 19:13.55 | Netgeeks | Asterisk is going through puberty, you'llhave to forgive the changes |
| 19:13.58 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 19:14.01 | Hymie | I guess name is name? |
| 19:14.07 | Hmmhesays | yes |
| 19:14.13 | Hymie | ok, thanks.. I'll give it a try |
| 19:14.17 | Hmmhesays | you win the prize |
| 19:15.03 | hypnox | im looking at an example that uses Goto(number) to changed the dialled number (EXTEN) to that number - does that work? |
| 19:15.06 | *** part/#asterisk Shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 19:15.08 | _Sam-- | thanks file :) |
| 19:15.14 | file | _Sam--: there you are |
| 19:15.25 | Hmmhesays | hypnox: punctuation man |
| 19:15.34 | _Sam-- | i cant wait to hear how good this gonna sound. |
| 19:15.39 | hypnox | yeah yeah.. :p |
| 19:15.40 | _Sam-- | you guys are 9ms away |
| 19:16.22 | file | _Sam--: BGP is gooooood |
| 19:16.27 | Hymie | thanks guys, now I can perform correct surgury on the callidnumber after I get it |
| 19:16.40 | Hymie | in this area, the telco provides you with the proper info for returning the call |
| 19:16.42 | Hymie | the phones |
| 19:16.49 | _Sam-- | the only i will need bgp to you is if that cogent router we both connect to dies :) |
| 19:16.53 | _Sam-- | er the only WAY |
| 19:17.06 | Hymie | don't.. so, if the number is 1666-666-6666, then you need to cut it to 666-6666 for the phone, if the phone is local to that number ;P |
| 19:17.47 | _Sam-- | file: i know you already explained this...but hoe come for iax, you just need to register to one server, but sip is 7 |
| 19:18.02 | Prival | Anyone experienced a lot of echo problems with Sangoma PRI card? |
| 19:18.11 | file | _Sam--: well, our database server was slow and in chan_sip if it didn't get a result fast enough - it would send back a codec error |
| 19:18.17 | hypnox | Hymie can you msg me your config for chopping up numbers like that? |
| 19:18.20 | file | _Sam--: but on chan_iax2 it didn't |
| 19:18.26 | _Sam-- | i see, thanks! |
| 19:19.07 | Hymie | hypnox: once I've got it re-written for 1.2.0, in a few secs, sure |
| 19:19.09 | _Sam-- | based on your personal opinion, in a small environment is there any advantage to using one protocol over the other? iax vs. sip (firewall stuff notwithstanding) |
| 19:19.17 | hypnox | thanks Hymie |
| 19:19.28 | file | _Sam--: I'm a SIP person, I just like it better... and it's worked better for us |
| 19:19.34 | _Sam-- | ok then, thanks. |
| 19:20.36 | file | _Sam--: your welcome letter just came into my email, haha |
| 19:20.37 | Netgeeks | IAX at one point had some kind of problem dealing with a single handler for the port, meaning there could be some 'blocking' I only read about it and never experienced anything that could be attributed to it. Don't know if it's been changed or not |
| 19:20.40 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 19:20.46 | Hymie | hmm, Set(CALLERID(num){555-5555)))?? |
| 19:20.52 | t0ke | anyone know any ip phone which support IAX protocol? tnks |
| 19:21.43 | Qwell[] | Set(CALLERID(num)=5555555) |
| 19:21.47 | _Sam-- | toke: IDEfisk (softphone) |
| 19:21.55 | *** join/#asterisk telcoDEJ (n=ed@c-68-82-75-221.hsd1.de.comcast.net) |
| 19:21.57 | _Sam-- | there is one hardphone but i dont think we can get it in the usa, maybe we can..not sure |
| 19:22.04 | Hymie | thanks |
| 19:22.09 | BlueDevi1 | t0ke: http://voip-info.org/wiki/index.php?page=VOIP+Phones |
| 19:22.18 | _Sam-- | telcoDEJ: are you in delaware? |
| 19:22.34 | telcoDEJ | hello, yes in New Castle DE |
| 19:22.41 | _Sam-- | im sitting in new castle right now |
| 19:22.43 | _Sam-- | by the post office |
| 19:22.45 | _Sam-- | on quigley |
| 19:23.05 | _Sam-- | lol |
| 19:23.27 | _Sam-- | telcoDEJ: do you do telephony work? |
| 19:23.30 | telcoDEJ | right down the road |
| 19:23.40 | Qwell[] | "down the road"? |
| 19:23.45 | Qwell[] | DE is the size of my living room |
| 19:23.53 | _Sam-- | its not as small as you'd think |
| 19:23.56 | Qwell[] | :p |
| 19:23.56 | telcoDEJ | Yes, Asterisk is new to me. I'm trying to get as much info as possible |
| 19:24.02 | _Sam-- | its narrow, but to go from north to south is a few hours |
| 19:24.11 | telcoDEJ | figured logging the IRC chan would help |
| 19:24.16 | file | Qwell[]: are you on... uh... Google Talk? |
| 19:24.23 | _Sam-- | if you need any help with asterisk...im right down the road :) |
| 19:24.25 | Qwell[] | file: nope, at work |
| 19:24.27 | Hmmhesays | the spanish ivr girl for this gateways sounds super sexay |
| 19:24.31 | file | Qwell[]: bad Qwell bad! |
| 19:24.33 | Qwell[] | _Sam--: free lunch ;] |
| 19:24.36 | telcoDEJ | many thanks |
| 19:24.50 | Hmmhesays | isn't everyone on google talk now? |
| 19:24.50 | telcoDEJ | looks like the potential is huge |
| 19:24.50 | Qwell[] | file: Why, I miss something fun? |
| 19:25.03 | _Sam-- | the only limit to its potential is...your imagination? |
| 19:25.06 | _Sam-- | and coding ability. |
| 19:25.15 | Hmmhesays | and patience |
| 19:25.19 | Hmmhesays | and beer supply |
| 19:25.21 | Qwell[] | and beer |
| 19:25.23 | Qwell[] | indeed |
| 19:25.24 | telcoDEJ | and time |
| 19:25.28 | file | Qwell[]: nah |
| 19:25.33 | Netgeeks | aspirin helps too |
| 19:25.34 | _Sam-- | it has a steep learning curve, in my opinion |
| 19:25.38 | _Sam-- | but once you get it, you 'get it' |
| 19:25.40 | Hmmhesays | beer in place of aspirin |
| 19:25.40 | Qwell[] | file: I didn't like the sound of that "nah"... |
| 19:25.47 | Netgeeks | beer + aspirin |
| 19:26.01 | _Sam-- | telcoDEJ: you've never set up an asterisk pbx yet? |
| 19:26.06 | Hmmhesays | not beer + tylenol... if you value your liver |
| 19:26.06 | kink0 | when I try to send a call from a registered peer ussing SIP, this sends the first INVITE, then the other part do Proxy Auth Require... but not second INVITE is sent, what may be wrong ? |
| 19:26.31 | telcoDEJ | well you'll se me here, and no doubt will have questions. Logging will let me go back and read questions/answers from others. |
| 19:26.35 | Netgeeks | I take two execedrin extra-strength before any asterisk work session |
| 19:26.37 | kink0 | I even set reinvite=yes, but same. |
| 19:26.52 | telcoDEJ | I best get back to my primary chores though |
| 19:26.53 | Hmmhesays | i make sure i have a healthy supply of budweiser and pizza |
| 19:27.01 | file | hot'n'sexy Hmmhesays!~ |
| 19:27.07 | Netgeeks | Hmm: that would kill me |
| 19:27.17 | hypnox | Hmmhesays add some beer and some food and then you're talking.. |
| 19:27.37 | _Sam-- | hey file, you said if i had a static ip i only need to register to one sip proxy...which one |
| 19:27.44 | _Sam-- | <sorry to ask the same thing twice> |
| 19:27.46 | file | _Sam--: any of them |
| 19:27.56 | _Sam-- | k thanks |
| 19:27.57 | Hmmhesays | yo file |
| 19:28.00 | file | my favorite switch today is... |
| 19:28.01 | file | 4! |
| 19:28.08 | Qwell[] | 04 or 4? |
| 19:28.09 | _Sam-- | hah then i am going N+1 to 5 |
| 19:28.11 | file | 04 |
| 19:28.14 | Qwell[] | because 04 is way cooler |
| 19:28.22 | file | core switches only go up to 3 |
| 19:28.34 | file | regular switches go up to 7 |
| 19:28.34 | Netgeeks | kink0: so let me get this straight, you have a registered UA, that UA sends an invite to your server, your server answers with a challenge, and the UA never responds? |
| 19:28.38 | Qwell[] | 07! |
| 19:29.04 | Netgeeks | 007! it has a license to fragment! |
| 19:29.15 | *** join/#asterisk santiago (n=santiago@63.245.86.215) |
| 19:29.52 | Qwell[] | file: So, if somebody has static, no need to use more than one peer? |
| 19:30.08 | file | Qwell[]: what? |
| 19:30.16 | Qwell[] | dunno, that's what _Sam-- said |
| 19:30.20 | Qwell[] | or something |
| 19:30.32 | file | get a grip on reality! |
| 19:30.39 | *** part/#asterisk santiago (n=santiago@63.245.86.215) |
| 19:31.06 | Qwell[] | Why don't you guys have SER on the front? |
| 19:31.11 | Qwell[] | or do you? |
| 19:31.20 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 19:31.23 | _Sam-- | they list 7 registration hosts to register to in the instructions |
| 19:31.26 | file | we use SER for other purposes, but I've been contemplating that actually |
| 19:31.30 | _Sam-- | if you have a static ip, you odnt need to register to all 7 |
| 19:31.36 | justinu | ser out front |
| 19:31.36 | Qwell[] | would make configs a lot easier, no? |
| 19:31.38 | justinu | intel inside |
| 19:31.47 | websae | anyone here familiar with e911 or h ave it implemented at all ? |
| 19:31.54 | Qwell[] | justinu: BZZT, that phrase is now dead |
| 19:31.56 | file | Qwell[]: well, no |
| 19:32.03 | Qwell[] | file: user configs, that is |
| 19:32.21 | file | the 7 registrations were to keep the information in memory |
| 19:32.33 | justinu | ser out front, asterisk inside? |
| 19:32.55 | file | but I've been pondering using SER as a load balancer... since right now we just have switch that's round robin DNS |
| 19:33.19 | Netgeeks | use asterisk and Transfer() as a load balancer |
| 19:33.50 | Qwell[] | Answer() , Transfer(${RAND(1,7)}) |
| 19:34.10 | *** join/#asterisk cassio- (i=cassio-@200216087155.user.veloxzone.com.br) |
| 19:34.32 | Netgeeks | or better yet, use an AGI that keeps track of load and allows you to shut down a node by not allowing new sessions... would make management much easier |
| 19:34.35 | _Sam-- | file sorry to keep asking ?s...it will stop soon...but how will i know when my prepaid minutes become authorized |
| 19:34.47 | file | _Sam--: transaction history will show your payment approved |
| 19:34.55 | cassio- | when I do make clean on asterisk I get an error on /root/asterisk/funcs can some1 help me on that?I am running FC2 |
| 19:34.56 | Qwell[] | it must be bug file day |
| 19:35.06 | Qwell[] | cassio-: pastebin the error |
| 19:35.13 | *** join/#asterisk loick (n=loick@APuteaux-151-1-57-122.w82-120.abo.wanadoo.fr) |
| 19:35.22 | cassio- | qwell ok |
| 19:35.30 | Hymie | <PROTECTED> |
| 19:35.30 | Hymie | <PROTECTED> |
| 19:35.33 | Hymie | weird |
| 19:35.40 | Hymie | it's not true, so why is it going to 4? |
| 19:35.52 | Qwell[] | Hymie: That isn't an eval |
| 19:35.53 | Qwell[] | $[] |
| 19:36.08 | file | uh tangent... what was it |
| 19:36.08 | file | OH |
| 19:36.13 | justinu | snom360 boots pretty damn fast |
| 19:36.14 | *** part/#asterisk dijit0 (n=eric@adsl-68-127-136-155.dsl.pltn13.pacbell.net) |
| 19:36.15 | file | I've sort of run out of servers in the datacenter |
| 19:36.20 | justinu | i forgot it was that quick |
| 19:36.22 | file | the last one I had sort of... went brain dead |
| 19:36.33 | Qwell[] | file: get more? :p |
| 19:36.42 | file | trying to get more boxes from David is... not easy |
| 19:36.50 | Hymie | Qwell[]: hmm, ok... used to work before, damn.. the wiki needs some updating, I'm trying to fix anything like this when I see it |
| 19:36.56 | Qwell[] | file: Go above David! :p |
| 19:37.23 | Qwell[] | Hymie: I assure you it didn't work before, if you didn't have the $ |
| 19:37.28 | cassio- | Qwell here http://pastebin.com/535788 |
| 19:37.35 | Qwell[] | [] isn't valid, and never has been... $[] is |
| 19:38.00 | _Sam-- | file, this is pure ball busting now...but i own a company that does more than 10mil a year in credit card sales...how long does it take to get an auth number and post my minutes? |
| 19:38.03 | Qwell[] | cassio-: Those are warnings |
| 19:38.08 | Hymie | heh, I guess not.. my other gotoifs have it, these two don't.. funny ;) |
| 19:38.09 | _Sam-- | i mean, it should take about 5 seconds to get the auth number |
| 19:38.23 | Netgeeks | GotoIf($["${MY-VAR}" = "${SOME-VAR}"]?yes,1:no,1) |
| 19:38.26 | cassio- | but this warning keeps getting all the time, nothing happens more, I saw this warning for 30 mins |
| 19:38.28 | Hymie | it's funny how the end user does not report callerid being broken ;) |
| 19:38.38 | Hymie | there's only a dozen of them, and nary a word! |
| 19:38.52 | Hymie | maybe I'm too much of a BOFH heh |
| 19:38.53 | Qwell[] | cassio-: well...touch them |
| 19:38.59 | file | _Sam--: our system is *NOT* real time, it's batched |
| 19:39.03 | Qwell[] | touch func_md5.c |
| 19:39.06 | _Sam-- | lame |
| 19:39.17 | _Sam-- | i guess for small charges..it makes sense |
| 19:39.20 | _Sam-- | our avg ticket is 200 |
| 19:39.21 | cassio- | Qwell what do you mean? |
| 19:39.22 | _Sam-- | not 2 dollars |
| 19:39.26 | file | lemme login |
| 19:39.27 | Qwell[] | cassio-: type that |
| 19:39.44 | *** join/#asterisk jmacz (n=jmacz@201.244.199.176) |
| 19:39.53 | cassio- | Qwell ok |
| 19:40.00 | Hymie | slick, ok thanks guys |
| 19:40.14 | cassio- | Qwell I did, now make clean again? |
| 19:40.17 | Qwell[] | file: Hey, while you're logged in... |
| 19:40.24 | Qwell[] | cassio-: Do that to all of the offending files |
| 19:40.29 | Hymie | hypnox: join #bleh |
| 19:40.30 | cassio- | ok |
| 19:40.40 | *** join/#asterisk Tili (i=Tili@202-133-67-118-dialup.sat.net.pk) |
| 19:40.59 | file | _Sam--: your $1.95 was approved, that's for a monthly recurring fee for the toll-free you ohave |
| 19:41.14 | justinu | i'd buy that for a dollar |
| 19:41.37 | _Sam-- | i just put in for 50 bucks |
| 19:41.40 | _Sam-- | about 10 minutes ago |
| 19:41.56 | Hmmhesays | what do you guys charge for incoming 1800 calls file |
| 19:41.58 | kink0 | what if peer after receive the nonce does not sends the second INVITE ? ( SIP INVITE Auth fail ) |
| 19:42.04 | file | _Sam--: I'm not even seeing it |
| 19:42.09 | Qwell[] | Hmmhesays: $6.50/minute |
| 19:42.16 | file | Hmmhesays: 2 cents per minute, $1.95/mth per toll-free |
| 19:42.17 | Hmmhesays | hot |
| 19:42.23 | file | oh |
| 19:42.24 | file | there it is |
| 19:42.34 | file | it wasn't in transactions yet |
| 19:42.37 | Hmmhesays | not bad, how long does it take you to put a number in service? |
| 19:42.40 | file | yeah, hasn't been processed yet - give it time |
| 19:42.46 | file | Hmmhesays: what type? |
| 19:42.49 | Netgeeks | kink0 if you are sending a Auth challenege to an invite and the far end doesn't respond, it could be a few things... |
| 19:42.54 | Hmmhesays | tollfree |
| 19:42.59 | _Sam-- | ok, by the time you help me with this next part...i bet the card will be auth'd :) |
| 19:43.03 | _Sam-- | what do i need to do: |
| 19:43.04 | websae | we offer $8/month DID and free inbound |
| 19:43.10 | justinu | Netgeeks: i've had that problem many times, but can never remember the solution. |
| 19:43.18 | file | Hmmhesays: a pool number takes about 5 minutes |
| 19:43.18 | _Sam-- | (sorry going for pastebin instead) |
| 19:43.34 | Hmmhesays | I shall pass that along |
| 19:43.41 | Netgeeks | kink0, 1. It doesn't have the required information to respond.. i.e. you challenge me and I don't have a secret to respond with |
| 19:43.48 | [av]bani | there you are _Sam-- !! |
| 19:43.59 | _Sam-- | file: http://pastebin.com/535795 |
| 19:44.08 | Netgeeks | kink0 2. ...... hrm, prolly just 1 reason |
| 19:44.12 | file | you need a 1 in front |
| 19:44.19 | cassio- | Qwell I keep getting that and doing touch on many files, what does that mean? |
| 19:44.27 | _Sam-- | damn i thought i normally added it...thanks. |
| 19:44.44 | Qwell[] | cassio-: your clock was probably hosed when you got the files, or it's hosed now |
| 19:45.12 | cassio- | Qwell what do you mean hosed?sorry for my english |
| 19:45.23 | Qwell[] | broken, busted, wrong, incorrect |
| 19:45.49 | cassio- | hmmmm |
| 19:46.06 | websae | has anyone had to implement e911 here yet? |
| 19:46.11 | Hmmhesays | whoa asterisklink is seriously b0rken |
| 19:46.19 | Hmmhesays | er.. this page on the site |
| 19:46.26 | file | I didn't do the site |
| 19:46.31 | file | if I knew who, they'd be dead |
| 19:46.32 | justinu | bs |
| 19:46.34 | _Sam-- | the pbix link at the bottom was bad a few days ago |
| 19:46.35 | justinu | it's all your fault |
| 19:46.46 | Hmmhesays | yeah someone should probably fix it, you notice the 800 number selection page? |
| 19:46.47 | file | it exists before my employment start time! |
| 19:46.53 | Netgeeks | hey I was just at asterlink... I wanted to see what they do |
| 19:46.57 | *** join/#asterisk Vladimirr (i=vlad@rapwap.razor.dk) |
| 19:47.01 | Netgeeks | apprently they don't shave |
| 19:47.03 | justinu | with an attitude like that young man, you won't make it very far! :P |
| 19:47.13 | file | Hmmhesays: which one? |
| 19:47.42 | cassio- | Qwell it worked, now its going on, thanks |
| 19:48.26 | Hmmhesays | mainpage -->signup-->basic package--> this page |
| 19:48.50 | `lyme | does each zap channel have to be unique? or only unique within it's group? |
| 19:49.03 | file | looks fine to me |
| 19:49.07 | Qwell[] | `lyme: Each is unique, and it's a specific order |
| 19:49.14 | EriSan | Does * support trunk fallback ? |
| 19:49.14 | Hmmhesays | must have been a fluke, i got half a page of script |
| 19:49.34 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 19:49.35 | Hmmhesays | yeap, that was strange |
| 19:49.40 | _Sam-- | file...more pastebin |
| 19:49.51 | `lyme | Qwell[]: so i cant have 2 channels 1s even though they are in different groups? |
| 19:50.02 | Qwell[] | `lyme no |
| 19:50.07 | kink0 | Netgeeks, I have set in both the same secret, and register is suscefull, but when server side sends to peer the noon, then the peer only answer with one ACK, and stop the SIP dialog |
| 19:50.13 | _Sam-- | file: http://pastebin.com/535810 |
| 19:50.14 | `lyme | thanks, would explain why that line wont work |
| 19:50.24 | Qwell[] | ooo, I got the script too |
| 19:50.30 | _Sam-- | the calls just stop with attempting bridge |
| 19:50.42 | Dr-Linux | why is this doing, any help |
| 19:50.43 | Dr-Linux | Feb 2 11:39:14 NOTICE[16794]: chan_sip.c:9514 handle_response_invite: Failed to authenticate on INVITE to '"Saud Shah 4092" <sip:4092@70.89.66.123>;tag=as3914db06' |
| 19:51.05 | file | _Sam--: what are the two devices? |
| 19:51.07 | _Sam-- | http://pastebin.com/535812 |
| 19:51.10 | kink0 | Netgeeks, sorry, I means the "nonce" not the noon |
| 19:51.12 | _Sam-- | asterisk / gxp2000 |
| 19:51.19 | _Sam-- | and they are calling out through you |
| 19:51.37 | kink0 | Dr-Linux, hey !! that is my same problem today !! |
| 19:51.40 | file | canreinvite=no on both entries in sip.conf? |
| 19:51.47 | Netgeeks | kink0: they are both asterisk systems? |
| 19:51.54 | kink0 | Netgeeks, yes |
| 19:51.59 | _Sam-- | lemme check it |
| 19:52.08 | Netgeeks | do you have the config set up as type=friend on both? |
| 19:52.20 | Netgeeks | kink0: are you using realtime? |
| 19:52.34 | kink0 | Netgeeks, only in one side, where is the server |
| 19:52.36 | _Sam-- | i think the gxp2000 in sip.conf = canreinvite = yes |
| 19:52.39 | cassio- | Qwell look : |
| 19:52.40 | cassio- | m |
| 19:52.45 | Qwell[] | m? |
| 19:53.10 | cassio- | wrong url :) |
| 19:53.20 | _Sam-- | changing them all to no |
| 19:53.25 | kink0 | realtime ? yes, I am on realtime now, debuging SIP, even I can add your entry to my sip.conf if you want to try |
| 19:53.25 | cassio- | http://pastebin.com/535813 |
| 19:53.41 | Netgeeks | kink0: the server is using realtime? or you have type=friend set up only on the server.... |
| 19:53.59 | kink0 | only in the server |
| 19:54.18 | Netgeeks | you should probably break the type=friend setting into type=user and type=peer |
| 19:54.22 | *** part/#asterisk secure75 (n=mic@dslb-084-057-024-106.pools.arcor-ip.net) |
| 19:54.32 | Qwell[] | cassio-: yes, clock skew is bad, you need to make the timestamps of all the files current |
| 19:54.42 | Netgeeks | you could also use the insecure=invite directive to not challenge on invite once the peer is registered |
| 19:54.43 | kink0 | I also try with type=user, but the same. |
| 19:54.44 | cassio- | Qwell what do you mean? |
| 19:54.49 | Qwell[] | touch every file... |
| 19:55.07 | Qwell[] | put it in a find loop or something |
| 19:55.10 | cassio- | on /funcs? |
| 19:55.15 | Qwell[] | everywhere |
| 19:55.26 | kink0 | Netgeeks, I set insecure=invite also, but same. Let me try again one more time with insecure=invite in the server. |
| 19:55.36 | cassio- | damn, is there any other way to fix it?like redownloading |
| 19:55.42 | Qwell[] | cassio-: You can try |
| 19:56.12 | *** join/#asterisk apardo (n=apardo@87.218.45.104) |
| 19:56.13 | cassio- | I did already, and same thing happened |
| 19:56.22 | Qwell[] | then your clock is broken... |
| 19:56.30 | cassio- | computer clock? |
| 19:56.37 | Qwell[] | yes |
| 19:56.44 | cassio- | what can I do to fix it? |
| 19:56.48 | kink0 | Netgeeks, the same with insecure=invite, the server sends Proxy-Authenticate: Digest realm="asterisk", nonce="035499d2" and the peer just answer "ACK" |
| 19:56.50 | Qwell[] | dunno |
| 19:56.54 | cassio- | its a dedicated server from godaddy.com |
| 19:57.13 | Netgeeks | kink0: something is wrong, with insecure=invite it is not supposed to challenge an invite |
| 19:57.26 | *** join/#asterisk pigpen2 (n=mark@65.91.220.162) |
| 19:57.46 | kink0 | Netgeeks, yes, that is what I read from the docs, but I am sure I am doing some mistake. |
| 19:57.57 | Netgeeks | it may be that the peer/user search routine is finding an entry other than the one you think it is |
| 19:58.35 | *** join/#asterisk infobox (n=dpizarro@libra.infostar.com.pe) |
| 19:58.36 | kink0 | hmmmm... I have only two peers registered, with differents names and info |
| 19:58.44 | _Sam-- | file: all sip clients/peers set to canreinvite = no and same problem |
| 19:58.56 | file | which switch is it going to? |
| 19:59.03 | *** join/#asterisk scon_ (n=scon@dslb-084-057-003-233.pools.arcor-ip.net) |
| 20:00.08 | _Sam-- | instead of trying it on a clients running production let me dick around with it on my own running production box |
| 20:00.17 | file | good idea |
| 20:00.42 | Netgeeks | kink0: are both peers using the same ip address? |
| 20:00.50 | Netgeeks | or even the same secret? |
| 20:01.00 | Netgeeks | are they both asterisk devices? |
| 20:01.06 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 20:01.10 | cassio- | Qwell is there a way to fix it? |
| 20:01.11 | kink0 | Netgeeks, no, there are even in differents nets, one is from outside, and the other is in the local lan |
| 20:01.19 | Hmmhesays | argh this auth_radius is killing me, binaries, no. compile from source, same error that tells me nothing |
| 20:01.24 | kink0 | no, no the same secret, user, callerid, or so |
| 20:01.40 | Netgeeks | kink0: hrm, okay, all the easy answers are gone :( |
| 20:02.21 | kink0 | Netgeeks, if was any peer/user confusion, I suppuse I will detect because I did a "sip debug" without ip or peer option |
| 20:05.38 | Netgeeks | kink0: I ran through all the simple issues I know of that would cause that |
| 20:05.41 | *** join/#asterisk crich1999 (n=crich@port-212-202-201-26.dynamic.qsc.de) |
| 20:05.47 | *** join/#asterisk The_LightSide (n=Frog@wbs-196-2-115-254.wbs.co.za) |
| 20:06.21 | Hmmhesays | thanks file, this is incredibly annoying |
| 20:06.28 | The_LightSide | hi all... has anyone got the FXO port of a HT488 working with AAH? |
| 20:06.40 | bkw_ | Hmmhesays, its ok dear |
| 20:06.47 | Hmmhesays | textops - initializing |
| 20:06.47 | Hmmhesays | ERROR: error while initializing modules |
| 20:06.50 | Hmmhesays | ARGH |
| 20:06.55 | infobox | hello everybody |
| 20:06.56 | Hmmhesays | hey bkw_ been awhile |
| 20:07.06 | infobox | Please, I have a terrible problem |
| 20:07.18 | Hmmhesays | does it burn when you .... |
| 20:07.26 | Dr-Linux | <file> canreinvite=no on both entries in sip.conf? what should i do |
| 20:07.30 | Dr-Linux | Feb 2 11:39:14 NOTICE[16794]: chan_sip.c:9514 handle_response_invite: Failed to authenticate on INVITE to '"Saud Shah 4092" <sip:4092@70.89.66.123>;tag=as3914db06' |
| 20:07.59 | Dr-Linux | one petren is working fine .. but the other short patren doesn't work and gives this error |
| 20:08.05 | `lyme | is there a way to manually close a channel thats stuck open? |
| 20:08.06 | Dr-Linux | what could be happend :S |
| 20:08.27 | Hmmhesays | has anyone ever gotten auth_radius to work with SER? |
| 20:08.29 | Netgeeks | CLI> soft hangup <channel> |
| 20:08.40 | _Sam-- | file: im trying iax instead of sip ...and now my iax registration times out to your server |
| 20:08.51 | _Sam-- | (i really am not that clueless...this box has 5 other providers set up fine on it) |
| 20:08.59 | _Sam-- | but yours so far has been the hardest to get going |
| 20:09.06 | *** part/#asterisk Vladimirr (i=vlad@rapwap.razor.dk) |
| 20:09.07 | file | is it switch-1? |
| 20:09.12 | Dr-Linux | file: |
| 20:09.20 | _Sam-- | switch-4 |
| 20:09.24 | _Sam-- | is that my problem |
| 20:09.27 | file | yes |
| 20:09.28 | file | it doesn't exist |
| 20:09.34 | Netgeeks | infobox: you should ask your question |
| 20:09.35 | _Sam-- | lol |
| 20:09.38 | _Sam-- | sorry for being a tool |
| 20:09.41 | file | switch-04 exists though |
| 20:09.41 | _Sam-- | fixed |
| 20:09.43 | infobox | ok. Netgeeks |
| 20:09.49 | dogtanian | lol |
| 20:10.00 | `lyme | Netgeeks: like soft hangup zap/3-1 (group 1, chan 3) ? |
| 20:10.00 | infobox | I want to deploy a gateway |
| 20:10.03 | dogtanian | <PROTECTED> |
| 20:10.20 | Hmmhesays | good for you infobox that shows some real motivation |
| 20:10.20 | infobox | ISDN/FXS |
| 20:10.25 | file | Dr-Linux: where are you trying to send the call, that you get that error message |
| 20:10.27 | dogtanian | <PROTECTED> |
| 20:10.28 | _Sam-- | dogtanian: im near philadelphia, how about yourself? |
| 20:10.34 | Netgeeks | 'lyme, yes, if you type soft hangup and then hit TAB, it should list all the channel arguments that are optional |
| 20:10.36 | dogtanian | london :D |
| 20:10.44 | Hmmhesays | i have an asterisk box in london |
| 20:10.50 | dogtanian | me too :) |
| 20:11.03 | infobox | I bought three boards: TE110P, TDM2406 and TDM2402 |
| 20:11.07 | dogtanian | <PROTECTED> |
| 20:11.24 | `lyme | Netgeeks: i do that and it tells me it is not a known channel, but i see it in the conf file |
| 20:11.26 | Hmmhesays | because the old pos alcatel pbx i'm interfaced with likes dass2 |
| 20:11.35 | infobox | But I can´t install the three in my PC |
| 20:11.46 | Netgeeks | 'lyme: does this channel stick all the time? |
| 20:11.50 | infobox | my motherboards is INtel865 GVSL |
| 20:11.52 | dogtanian | ahh i see. I use IAX2 trunks to voiptalk for all my pstn terminated numbers :) |
| 20:12.06 | dogtanian | voiptalk are great |
| 20:12.07 | Dr-Linux | file: i'm sending this call to my another server, |
| 20:12.27 | file | Dr-Linux: well it needs either a user entry [4092] with the same password, or a peer entry with the IP address and insecure=very |
| 20:12.35 | Dr-Linux | file: in same setting, one pertrn is working fine, but this one doesn't work :S |
| 20:12.41 | infobox | maybe anybody coul help me , please |
| 20:12.54 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 20:13.07 | `lyme | Netgeeks: kinda... but im still testing, and people keep picking up hardlines, thats what im blaming it on |
| 20:13.14 | Dr-Linux | file: all these settings are there |
| 20:13.43 | websae | a T1 can handle how many g.729 connections? |
| 20:13.52 | Netgeeks | 'lyme: it's a Zap channel? you could try running ztcfg -v at the system prompts (warning this will possibly reset all zap channels) |
| 20:14.08 | file | Dr-Linux: sip debug on both machines, and pastebin it |
| 20:14.15 | justinu | websae: a lot |
| 20:14.26 | `lyme | Netgeeks: i will try that is i continue to have issues |
| 20:14.32 | crich1999 | infobox, what kind of problem happens ? |
| 20:14.41 | Netgeeks | 'lyme do you see the channel in the list when you type soft hangup <TAB>? |
| 20:15.19 | The_LightSide | would anyone be able to help me with a grandstream handytone488? |
| 20:15.39 | `lyme | Netgeeks: nothing happens when i hit tab? |
| 20:15.50 | *** join/#asterisk backblue (n=moo@87-196-1-50.net.novis.pt) |
| 20:16.13 | Netgeeks | localhost*CLI> soft hangup |
| 20:16.14 | Netgeeks | SIP/8459435661-3ca1 Zap/23-1 |
| 20:16.21 | Netgeeks | you should see something like that |
| 20:16.43 | `lyme | its not there |
| 20:16.47 | infobox | Netgeeks: do you have experienced with TDM24XX ? |
| 20:16.52 | Dr-Linux | file: here is two lines, the 1st one is working and 2nd dosn't work |
| 20:16.53 | Dr-Linux | exten => _92XXXXXXXXXX,2,Dial(SIP/serverA/0${EXTEN:2},30,r) |
| 20:16.53 | Dr-Linux | exten => _92XX.,1,Dial(SIP/serverA/${EXTEN:2},30,r) |
| 20:17.04 | Netgeeks | if you don't see any channel when you hit tab after soft hangup, then your asterisk doesn't think the channel is there/active |
| 20:17.16 | file | well for one thing, the top one has priority 2 |
| 20:17.21 | Netgeeks | it's really stuck... |
| 20:17.30 | Netgeeks | infobox: no I don't |
| 20:17.40 | `lyme | no joke. |
| 20:17.41 | Dr-Linux | file: bcoz it has caller id in 1st pirority, that i didn't paste |
| 20:17.49 | *** join/#asterisk PoWeRKiLL (i=PoWeRKiL@212.43.222.138) |
| 20:18.00 | file | okay so do what I said before |
| 20:18.00 | Dr-Linux | that one is able to dial properly |
| 20:18.02 | file | sip debug |
| 20:18.08 | Dr-Linux | ok |
| 20:18.15 | infobox | ok thanks netgeeks |
| 20:18.17 | `lyme | so that command should show all open connections right? |
| 20:19.19 | Netgeeks | it should show all the channels that asterisk consideres active |
| 20:19.19 | file | Dr-Linux: I think I know why, but I doubt I could explain it so you could understand it |
| 20:19.19 | Netgeeks | considers |
| 20:19.20 | file | but I'll try anyway |
| 20:19.20 | `lyme | uhrm... now it shows up?! |
| 20:19.20 | Netgeeks | sorry, new fingers |
| 20:19.20 | file | Dr-Linux: when you change the callerid it changes the From header in the INVITE, unless fromuser is specified - is fromuser specified for the serverA peer entry in sip.conf? |
| 20:19.20 | Netgeeks | old brain, incompatibility issues between the two |
| 20:19.28 | *** join/#asterisk Norm (n=Norm@normmac.net.wm.edu) |
| 20:19.32 | Hmmhesays | dogtanian, i poach numbers off of outgoing calls that are supposed to hit BT and send them out to a termination provider |
| 20:19.47 | file | Dr-Linux: what's happening is that the other server is matching the _92XX. call to an entry in sip.conf, and is challenging for credentials - which can't be provided because you have this setup wrong |
| 20:20.28 | Netgeeks | 'lyme: welcome to the wonderful world of asterisk! |
| 20:20.33 | `lyme | yeah... |
| 20:20.44 | websae | g.729 operates at 8kbps correct? |
| 20:20.44 | `lyme | and i hung it up... and the physical line is still open. |
| 20:20.55 | Netgeeks | now you know why I take two extra-strength excedrin before doing any work with asterisk |
| 20:21.08 | justinu | more like 10-12kbps with overhead, iirc |
| 20:21.08 | Hmmhesays | wow my milliwatt test sounds like crap even on this lan |
| 20:21.11 | Hmmhesays | somethin gis farked |
| 20:21.28 | `lyme | see whasts weird is i see where is keeps starting the simple switch on that channel, but.... it should show as unavil? |
| 20:21.29 | justinu | netgeeks: try ser |
| 20:21.39 | justinu | netgeeks: you'll need vicodin, not excedrin :P |
| 20:21.44 | kink0 | Netgeeks, now is working !!! I just change the name from a number to alfa in both : register => Joaquin:pass instead register => 1234:pass |
| 20:21.57 | Netgeeks | moz is playing DAoC on your LAN, there is a big Raid for merlin's staff or soemthing |
| 20:22.00 | file | Dr-Linux: http://pastebin.com/535862 that's using username and password authentication |
| 20:22.02 | *** part/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:22.22 | Netgeeks | justinu: SER = Send Emergency Response |
| 20:22.45 | justinu | heh |
| 20:22.48 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:22.53 | Dr-Linux | file: here is debug from destination server |
| 20:22.58 | Dr-Linux | http://pastebin.com/535861 |
| 20:23.15 | file | Dr-Linux: it's using user entry 4092 in sip.conf |
| 20:23.17 | dogtanian | Hmmhesays: yeah - best way to do it |
| 20:23.29 | file | Dr-Linux: and your other machine can't authenticate it |
| 20:23.30 | Netgeeks | kink0: great to hear you got it working! Bad to hear asterisk has an issue with one versus the other |
| 20:23.43 | infobox | I am beginning to hate Digium/asterisk... |
| 20:23.52 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 20:24.08 | Hmmhesays | did you wake up one morning and wonder what you did with them last night infobox? |
| 20:24.09 | iCEBrkr | infobox: Y U B H8 N? |
| 20:24.13 | Netgeeks | justinu: unless you want the register many UAs to a single asterisk SIP account, I see zero value in using SER with asterisk |
| 20:24.20 | Dr-Linux | file: but why it authenticate with other patren? |
| 20:24.30 | Hmmhesays | Netgeeks: media server |
| 20:24.33 | file | Dr-Linux: because when you change the callerid, it changes the part that asterisk uses to find a user entry |
| 20:24.37 | file | so it no longer finds one |
| 20:24.40 | Hmmhesays | and let ser take care of the p2p calls |
| 20:24.45 | justinu | netgeeks: i use it for clustering |
| 20:24.49 | file | and lets the call through, probably using your default specified context |
| 20:25.00 | websae | does anyone here know about anyone offering E911 services to asterisk? |
| 20:25.17 | Dr-Linux | hhmmmmmmm..... |
| 20:25.18 | Netgeeks | justinu: with a little ingenuity, you can do clustering without having to use SER |
| 20:25.48 | Dr-Linux | file: i think i should use an caller ID before this dialplan line? :S |
| 20:26.01 | file | or find the real problem, and decide how you're going to do authentication and stuff |
| 20:26.10 | EriSan | Does * support trunk fallback ? ie, if trunk1 is busy, move to trunk2 for outbound calls ? |
| 20:26.15 | Hmmhesays | justinu, have you ever used auth_radius? |
| 20:26.20 | Dr-Linux | file: the one is working i'm using with same user, |
| 20:26.28 | Netgeeks | EriSan if you set the dialplan up to do so, yes |
| 20:26.30 | justinu | Hmmhesays: not yet. |
| 20:26.41 | Dr-Linux | file: look there is not much difference |
| 20:26.42 | Dr-Linux | exten => _92XXXXXXXXXX,1,Set(CALLERID(NUM)=16505935400) |
| 20:26.42 | Dr-Linux | exten => _92XXXXXXXXXX,2,Dial(SIP/serverA/0${EXTEN:2},30,r) |
| 20:26.42 | Dr-Linux | exten => _92XX.,1,Dial(SIP/serverA/${EXTEN:2},30,r) |
| 20:26.47 | justinu | netgeeks: i need a real sip proxy for what I do |
| 20:26.49 | Netgeeks | ~pb |
| 20:26.55 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:26.55 | file | Dr-Linux: no, there is a BIG difference |
| 20:27.01 | Netgeeks | justinu: understood |
| 20:27.01 | EriSan | Netgeeks, any example online somewhere ? |
| 20:27.26 | Netgeeks | EriSan: I'm sure there is, but off the top of my head, I don't have a url for you |
| 20:27.29 | file | Dr-Linux: you need to decide on how you're going to authenticate on that other box, because I suspect right now - it's wide open |
| 20:27.35 | Dr-Linux | file: and that difference causes the prbolem? |
| 20:27.35 | *** join/#asterisk Zodiacal (n=hehe@bdsl.66.14.242.199.gte.net) |
| 20:27.39 | file | Dr-Linux: yes |
| 20:27.59 | EriSan | Netgeeks,thx, will google |
| 20:27.59 | file | Dr-Linux: when you change the callerid, you change the part that Asterisk uses to figure out who to challenge for for authentication |
| 20:28.06 | Netgeeks | you would basically try a tunk, check the DIALSTATUS, and try another trunk if the result was something other than BUSY, NO ANSWER, or ANSWER |
| 20:28.17 | Hmmhesays | CALLERID(NUM) is not valid is it? |
| 20:28.18 | file | Dr-Linux: since you change it, it doesn't match any entries - and goes to the default context most likely |
| 20:28.34 | Netgeeks | not sure if those are the correct spelling/phrasing of dialstatus results, but you get the idea |
| 20:28.49 | Zodiacal | how come i get this error when putting a .call file in asterisk outgoing directory. someone told me to put a simple call file there that dials a phone to figure out my delay issue. heres the asterisk error: pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/test.call: Permission denied, deleting |
| 20:28.53 | *** join/#asterisk pablasso (n=pablasso@201.135.98.244) |
| 20:28.56 | Zodiacal | how do i give it the right permissions? |
| 20:28.58 | Netgeeks | Dial(SIP/blah/${EXTEN}) |
| 20:29.19 | file | ummm oh right |
| 20:29.20 | file | Hmmhesays: yes |
| 20:29.29 | Netgeeks | GotoIf($["${DIALSTATUS}" = "CONGESTION"]?next-trunk,s,1) |
| 20:29.30 | file | it's valid... I think |
| 20:29.31 | Seldon1975 | do I want to co http://svn.digium.com/svn/asterisk/tags/1.2.4/ for latest |
| 20:29.31 | Netgeeks | etc... |
| 20:29.38 | Seldon1975 | or something else |
| 20:29.55 | file | yeah it's valid |
| 20:29.57 | Qwell[] | Seldon1975: The 1.2.4 tag is the latest release version |
| 20:30.10 | Seldon1975 | Qwell: thx |
| 20:30.32 | Hmmhesays | ok, every freaking thing I try doesn't work and this gives me 0 info about why |
| 20:30.37 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 20:30.47 | Dr-Linux | file: problem is fixed. but i just wanna understand the logic. even my sip.conf things are also fine |
| 20:31.01 | *** join/#asterisk heath__ (n=heath__@12-215-32-56.client.mchsi.com) |
| 20:31.32 | file | Dr-Linux: I've explained it 3 times I think, see above and try to understand |
| 20:31.38 | Netgeeks | don't throw your gauntlet at SER's feet, it's a nasty thing to piss off |
| 20:31.50 | Zodiacal | any ideas? |
| 20:32.49 | Hmmhesays | Netgeeks, no doubt |
| 20:33.17 | Netgeeks | you just trying to get it to work as a registry using radius? |
| 20:33.45 | Dr-Linux | file: as you said, you suspect its wide open. hmm.. lemme show you, maybe there is something wrong |
| 20:34.36 | Hmmhesays | Netgeeks, at this point i'm just trying to get auth_radius to load |
| 20:34.43 | Hmmhesays | it errors with no debugging info |
| 20:34.53 | Netgeeks | :( |
| 20:34.54 | Seldon1975 | I'm having trouble getting the latest from SVN: |
| 20:34.55 | *** join/#asterisk ToTo (n=ToTo@host94-157.pool874.interbusiness.it) |
| 20:34.56 | Seldon1975 | I get: svn: PROPFIND of '/svn/asterisk/trunk': 400 Bad request: request method denied ( |
| 20:34.56 | Seldon1975 | http://svn.digium.com) |
| 20:35.06 | Seldon1975 | any ideas? |
| 20:35.14 | Qwell[] | Seldon1975: Behind a proxy? |
| 20:35.25 | Seldon1975 | Qwell: firewall but no proxy |
| 20:35.35 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 20:35.38 | Seldon1975 | should this command work: |
| 20:35.54 | Seldon1975 | svn co http://svn.digium.com/svn/asterisk/tags/1.2.4/ asterisk |
| 20:35.57 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 20:36.02 | Qwell[] | should work fine |
| 20:36.14 | Seldon1975 | Qwell: thx Ill talk to my sysadmin |
| 20:36.42 | Dr-Linux | file: this my sip.conf setting from where i dial ... pvt paste |
| 20:37.07 | Dr-Linux | file: at the other end same |
| 20:37.46 | *** part/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:38.54 | *** join/#asterisk rexykik (n=karl@c-24-11-146-21.hsd1.mi.comcast.net) |
| 20:39.15 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:39.35 | *** join/#asterisk freat (n=rscnyk@h-72-244-84-43.chcgilgm.covad.net) |
| 20:39.46 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 20:39.49 | rexykik | anyone have any idea where i can get the new firmware for a Eeezeephone netweb-301? The eezeephone site doesn't appear to exist anymore, and I haven't been able to find it cached somewhere through google |
| 20:39.50 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-32-116.dsl.irvnca.pacbell.net) |
| 20:39.57 | tainted_ | did local channel change in 1.2.4? |
| 20:40.18 | freat | hello all.... I gots problems with AAH. I'm used to doing it all command-line style, and still can't get trunks to work like I want |
| 20:40.37 | freat | the from-internal context is not recognized as a valid context |
| 20:40.55 | justinu | ~amportal |
| 20:40.58 | justinu | ~amp |
| 20:41.00 | jbot | i heard amp is NOT supported here! people using it should join #amportal |
| 20:41.16 | freat | oh thanks jbot heh |
| 20:41.40 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:42.29 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:43.13 | evilbit | Anyone aware of a telecom (not necessarily asterisk specific) irc server/channel? I have q's regarding T1 demark extension. |
| 20:43.36 | *** part/#asterisk mroth_imm (n=mroth@63.65.26.220) |
| 20:43.57 | justinu | we can probably help you |
| 20:44.16 | evilbit | That'd be great if you don't mind ... |
| 20:44.22 | justinu | go ahead |
| 20:45.10 | [Atlas] | lesson #1 have the telco extend the demark so you dont get into telco / 3rd party cable monkey battles when your t1 doesnt work * :) |
| 20:45.12 | evilbit | I have 4 T1's, 4 100Mbit connections, and 12 voice lines I need to move from the telco closet in my building, to a suite 70 feet or so in distance. I'd like to use a 50 pair cable with 110 blocks at either end but from what I'm reading, the t1's might not be so good in this bundle. |
| 20:45.38 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 20:46.02 | justinu | bah |
| 20:46.03 | evilbit | i've heard the *right* way to do the t1's is with individually shielded twisted pair but also hear a lot of posts along the lines of "cat5 works for me" |
| 20:46.05 | justinu | it'll be fine |
| 20:46.15 | justinu | telco runs t1's on 50 pair all the time |
| 20:46.16 | austinnichols101 | evilbit: imo, t1's aren't going to care other than possibly having to make adjustments to the csu for distnace. 100mb connections are another story |
| 20:46.26 | Dr-Linux | file: what if my other server has non-SIP standard port? like other than 6050 ? |
| 20:46.44 | [av]bani | justinu: it's crosstalk that's the worry. rj11 picking up from the t1, which i've seen in experience |
| 20:46.46 | file | port=6050? |
| 20:46.52 | Dr-Linux | sorry |
| 20:46.57 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:46.58 | [Atlas] | ive always done it with cat5 -- 50 pair is fine dude :D |
| 20:46.58 | Dr-Linux | sip port 5060 |
| 20:47.00 | [av]bani | he's not worried about t1's being corrupted, he's worried about rj11 picking up t1 interference |
| 20:47.14 | justinu | i haven't had that problem |
| 20:47.14 | [av]bani | 100mb is fine |
| 20:47.20 | evilbit | I'm worried about any of them interfering with any of the others, really. I just don't want to have to go back and do it again later if i run a 50 pair now. |
| 20:47.28 | justinu | so run 2 50 pairs |
| 20:47.31 | justinu | er 25 pairs |
| 20:47.40 | [av]bani | keep your t1 in different bundles |
| 20:47.43 | [av]bani | t1 with t1 is fine |
| 20:47.55 | evilbit | tx and rx pairs in the same bundle? |
| 20:47.57 | [av]bani | they dont interfere with each other, you can also throw sdsl/hdsl into the bundle |
| 20:48.10 | [av]bani | just avoid mixing your rj11 voice with the t1s |
| 20:48.14 | [av]bani | 100mb should also be fine with t1 |
| 20:48.43 | Netgeeks | 70 feet of non-twisted pair + 100M ethernet = BAD |
| 20:48.52 | justinu | maybe it's HDSL that can live in the same bundle as voice |
| 20:48.57 | [av]bani | you cant do non twisted pair with 100m ether really |
| 20:49.01 | [av]bani | at any distance |
| 20:49.02 | *** join/#asterisk aqua26 (n=aqua@59.182.15.148) |
| 20:49.11 | [av]bani | justinu: yeah, hdsl + voice = death. |
| 20:49.14 | evilbit | twisted pair is what i was looking at |
| 20:49.21 | [av]bani | sdsl is generally ok |
| 20:49.26 | justinu | hmm |
| 20:49.36 | t0ke | Anyone use any ip phone supporting IAX2 protocol? |
| 20:49.43 | Netgeeks | just make sure you get the correct signals onto the right twisted pair and you will be fine then |
| 20:49.51 | justinu | i thought telco ran T1s on HDSL in huge bundle cables |
| 20:49.52 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 20:49.57 | [av]bani | if you use cat5 it's already twisted |
| 20:50.01 | justinu | they go thru the same b-box and all that |
| 20:50.01 | ctooley | Why is it that one 1.2.1 of asterisk I could drop a call file with a channel line of "Channel: SIP/202" and it would go to peer 202 on the local machine, but in 1.2.3 it tells me it failed to go throug with reason 1? |
| 20:50.02 | t0ke | I looked many but I will like to know from anyone |
| 20:50.11 | [av]bani | justinu: they do, all together with other T1s. they dont mix voice with them |
| 20:50.18 | [av]bani | t1 has strict rules on bundling |
| 20:50.26 | [av]bani | you can mix hdsl, sdsl and t1 |
| 20:50.28 | evilbit | that's what i thought bani, due to the 6v on the line |
| 20:50.39 | evilbit | which got me wondering about shielding. |
| 20:50.40 | [av]bani | they are designed to not interfere with each other |
| 20:50.57 | [av]bani | evilbit: cat5, cat5e should be fine with t1. its all twisted and balanced |
| 20:51.02 | justinu | [av]bani: so they have a seperate bundle running to those b-boxes for data? |
| 20:51.16 | justinu | what if for example, I ask for a T1 at my house. |
| 20:51.17 | [av]bani | justinu: they have them separated from rj11 bundles by a specific distance |
| 20:51.32 | justinu | the HDSL pairs will have to run on the same bundle that serves my entire neighborhood |
| 20:51.35 | evilbit | ok so general consensus is that i should be ok for 70 ft of 50 twisted pair mixing t1, voice, and 100mbit ethernet? |
| 20:51.35 | [av]bani | and they dont run them in the same groups |
| 20:51.37 | hypnox | Any UK PSTN guys here? I want to know if subscriber numbers always start with 2 or higher. |
| 20:52.26 | [av]bani | evilbit: dont mix voice with the t1 if you can avoid it |
| 20:52.34 | [av]bani | put it in a separate bundle |
| 20:52.35 | benjk | hypnox: no |
| 20:52.55 | benjk | you may want to check out the UK numbering plan |
| 20:52.58 | [av]bani | t1 is pretty noisy, voice bundles may pick it up. 100mbit should be pretty much immune to anything |
| 20:53.03 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:53.10 | rexykik | anyone have any idea where i can get the new firmware for a Eeezeephone netweb-301? The eezeephone site doesn't appear to exist anymore, and I haven't been able to find it cached somewhere through google |
| 20:53.12 | [av]bani | and 100mbit doesnt generate noise |
| 20:53.46 | evilbit | Can they all be punched down on the same 110 block then if I'm running the t1's in a separate bundle? |
| 20:53.51 | [av]bani | yes |
| 20:53.56 | *** join/#asterisk acqua7 (n=bc290db0@customer-200-36-59-130.uninet.net.mx) |
| 20:54.14 | hypnox | benjk am doing.. ive just read about "national dialing only" ranges which informs me that there are subscriber numbers that begin with 0 or 1, but the STD must always also be dialled |
| 20:54.34 | [av]bani | its just the long runs of cable next to each other that causes the issues |
| 20:54.44 | benjk | http://en.wikipedia.org/wiki/UK_telephone_numbering_plan |
| 20:54.52 | Seldon1975 | Cat: "I got it! Don't fish swim south for the winter?" Kryten: "No, that's birds, sir" Cat: "Birds swim south for the winter !? How do they breate?" |
| 20:54.53 | hypnox | that's the one :) |
| 20:55.05 | Seldon1975 | breath* |
| 20:55.08 | Seldon1975 | breathe* |
| 20:55.08 | evilbit | Great, I'll run one bunle then for the 100mbit and voice, and a separate 8 pair for the t1's, punch everything down onto the same blocks and be in business. |
| 20:55.12 | evilbit | thanks a ton for the help guys. |
| 20:56.27 | [av]bani | taht should work |
| 20:56.37 | hypnox | benjk given that, would you say it's safe to assume that if a number starts with 2 and is between 6-8 digits long, its a local number? |
| 20:56.45 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:56.52 | [av]bani | though i'd convert everything to ip and run them over gbe :) |
| 20:56.57 | hypnox | 2 or above i mean of course |
| 20:56.58 | [av]bani | 1 pair |
| 20:56.59 | [av]bani | :) |
| 20:57.06 | Seyr | Is there any way to timestamp messages in the CLI? |
| 20:57.46 | acqua7 | hi, i'm having a problem.... to integrate asterisk with an avaya sip server, i have to edit iax.conf ? or i have to define parameters in sip.conf? |
| 20:57.51 | pifiu | i need to dial out of the country to 1+58+261+xxxxxxx |
| 20:57.56 | pifiu | how would i enable that in the dialplan? |
| 20:58.12 | file | acqua7: since it's SIP, one would think sip.conf |
| 20:58.16 | evilbit | mmm gbe |
| 20:58.55 | tainted_ | did call files change in 1.2.4? |
| 20:58.56 | evilbit | that would be a task though, involving moving my routing equipment (builtin csu) from the rack 70 ft away to the telco closet |
| 20:59.04 | Seyr | pifu: Dial(SIP/MyProvider/158261${EXTEN}) |
| 20:59.05 | Seyr | i believe |
| 20:59.06 | acqua7 | file thanks |
| 20:59.09 | file | okay people, who wants my lesson of the day? |
| 20:59.23 | Qwell[] | file: go for it |
| 20:59.28 | Qwell[] | then I'll do mine |
| 20:59.29 | file | lemme get it ready |
| 20:59.37 | Qwell[] | okay, I'll do mine first then :p |
| 20:59.48 | Qwell[] | When writing a web form to purchase something... |
| 20:59.49 | [TK]D-Fender | "How to STFU in 1 easy step" ? |
| 20:59.57 | Qwell[] | don't do your freaking CC checking in javascript |
| 21:00.06 | Qwell[] | file: I might have a full pass to VON now :P |
| 21:00.07 | Netgeeks | ! |
| 21:00.20 | file | Qwell[]: great |
| 21:00.27 | Qwell[] | under a fake name, of course |
| 21:00.38 | file | svn diff http://svn.digium.com/svn/asterisk/tags/1.2.3 http://svn.digium.com/svn/asterisk/tags/1.2.4 | more |
| 21:00.38 | Netgeeks | Abby Normal? |
| 21:00.42 | file | will tell you all the code changes between 1.2.3 and 1.2.4 |
| 21:00.42 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:00.48 | file | using that you can figure out what changed |
| 21:00.51 | file | or! |
| 21:00.53 | file | read the changelog :P |
| 21:00.58 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 21:00.59 | Qwell[] | file: nah! |
| 21:02.32 | *** join/#asterisk bluecrow76 (n=Mitch@ip24-250-77-177.br.br.cox.net) |
| 21:02.34 | pifiu | seyr |
| 21:02.41 | pifiu | do i have to put XXXXXX after the 158261? |
| 21:02.45 | tainted_ | file thx |
| 21:03.04 | file | Qwell[]: if you do show up, we must chill |
| 21:04.11 | pifiu | exten => Dial(IAX2/ipcomms/158261${EXTEN}) |
| 21:04.15 | pifiu | are you sure thats right? |
| 21:04.22 | rexykik | anyone have any idea where i can get the new firmware for a Eeezeephone netweb-301? The eezeephone site doesn't appear to exist anymore, and I haven't been able to find it cached somewhere through google |
| 21:04.28 | Netgeeks | </div></span> |
| 21:04.33 | Netgeeks | gah, wrong window |
| 21:05.12 | nestar | lol |
| 21:05.53 | Netgeeks | now I'm wondering if my web page has some smart ass comment I meant to make here |
| 21:05.56 | MikeJ[Laptop] | yay fun with voicexml |
| 21:06.28 | Qwell[] | MikeJ[Laptop]: eww |
| 21:06.36 | Hmmhesays | ugh voicexml is killing me |
| 21:06.44 | Hmmhesays | that and auth_radius |
| 21:07.05 | MikeJ[Laptop] | voicexml is fun... |
| 21:07.13 | MikeJ[Laptop] | getting ASR working at the house.. |
| 21:07.23 | MikeJ[Laptop] | so my kids friends can ask for her by name |
| 21:07.24 | Qwell[] | sphinx? |
| 21:07.27 | MikeJ[Laptop] | no. |
| 21:07.28 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 21:07.31 | MikeJ[Laptop] | voxeo |
| 21:07.35 | Qwell[] | nuance rebrand? heh |
| 21:07.47 | MikeJ[Laptop] | yeah, somthing like that |
| 21:07.53 | asr__ | getting me working? |
| 21:07.58 | MikeJ[Laptop] | but 2 port free.. all I need for the house |
| 21:08.11 | MikeJ[Laptop] | basic ivr and a little ccxml |
| 21:08.11 | Qwell[] | I finally got my cepstral stuff |
| 21:08.14 | Qwell[] | I don't like it |
| 21:08.14 | Netgeeks | neat |
| 21:08.25 | rpm | [2006-02-02 14:04:24] WARNING[30057]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! |
| 21:08.26 | MikeJ[Laptop] | Qwell, new voices for cepstral out today |
| 21:08.31 | pifiu | qwell my dear buddy does this seem right? |
| 21:08.33 | rpm | why do i get that when i make a call? |
| 21:08.33 | pifiu | exten => _1NXXNXXXXXXXX,1,Dial(IAX2/ipcomms/158261${EXTEN}) |
| 21:08.35 | Qwell[] | better than the beta 4? |
| 21:08.46 | Qwell[] | pifiu: Sure, try it |
| 21:08.51 | Netgeeks | rpm: are you perhaps out of G.729 decoder licenses? |
| 21:08.52 | rpm | I have 1 license, I make 1 call. |
| 21:08.53 | MikeJ[Laptop] | Qwell, it's fine |
| 21:09.03 | MikeJ[Laptop] | its certainly a big step from 3.5 |
| 21:09.04 | Qwell[] | MikeJ[Laptop]: The only one that sounded decent was William |
| 21:09.13 | MikeJ[Laptop] | and a WHOLE lot better than festival |
| 21:09.20 | MikeJ[Laptop] | and it's cheap.. |
| 21:09.29 | MikeJ[Laptop] | well... for me it's free.. but even normally it's cheap |
| 21:09.32 | Qwell[] | I'm still not impressed. ;/ |
| 21:09.46 | MikeJ[Laptop] | then pay somone to record prompts : |
| 21:09.47 | sch19 | hmm.. Qwell[], now that I've installed debian and felt my way around a bit so that I'm kind of comfortable, I should not be messing with trunk any, but using the tagged 1.2.4, or maybe the 1.2 branch if it is stable ? |
| 21:09.48 | MikeJ[Laptop] | :P |
| 21:09.51 | Netgeeks | 1 license will decode one leg.... if you have two legs that need to be encoded/decoded (for example using meetme to join a call forces conversion of both legs to ulaw) |
| 21:09.55 | Qwell[] | yeah...probably far better |
| 21:10.15 | Qwell[] | sch19: Sure, 1.2.4 works |
| 21:10.20 | MikeJ[Laptop] | It will be fine for my purposes.. |
| 21:10.31 | MikeJ[Laptop] | but we need somone to write a real mrcp stack... |
| 21:10.51 | MikeJ[Laptop] | we need that anyways... can do cool stuff with distributing calls across gateways with that.. |
| 21:11.00 | sch19 | hehe, is that the short, "go away" answer? :P |
| 21:11.10 | tainted_ | did call files change? |
| 21:11.10 | MikeJ[Laptop] | it is one of the missing pieces to load balancing across asterisk gateways... |
| 21:11.13 | Hmmhesays | heh, this milliwatt test sounds like crap and i'm on the same network segment |
| 21:11.15 | Qwell[] | sch19: It's the "try it yourself, and see what works" answer |
| 21:11.38 | sch19 | yup, guess I'm just trying to save time, I know better i suppose. |
| 21:11.57 | Qwell[] | if you need bleeding edge, go for it...if not...get the release versions |
| 21:12.13 | Flyboy-SR22 | anyone here use FOP..? |
| 21:12.16 | hypa7ia | hey Qwell[]! |
| 21:12.20 | Netgeeks | time and land on the california coast... two of the most expensive things |
| 21:12.21 | pifiu | qwell i tried it and it said: == Everyone is busy/congested at this time (1:0/1/0) |
| 21:12.23 | Qwell[] | hypa7ia: hey |
| 21:12.32 | pifiu | could it be my provider blocks international calls? |
| 21:12.32 | hypa7ia | i actually came on with a question this time :) |
| 21:12.37 | hypa7ia | On a system with FXO's connected to PSTN lines, and FXS's connected to analog phones, when we do "flash" on the phone sets a second line is taken up on the trunks out. We have threewaycalling=yes and transfer=yes in zapata.conf; does anything else need to be set in zapata.conf or elsewhere to enable the "flash" on the handset to be sent out the Bell line? |
| 21:12.49 | sch19 | yeah, i hear ya.. I'm prematuerly diving into this anyway, I'm not even on the mailing lists yet. need to devote more time to it. |
| 21:12.54 | sch19 | brb |
| 21:13.09 | hypa7ia | my googling didn't turn anything up :/ |
| 21:14.09 | sch19 | hypa7ia, that seems kinda wierd to want to use 3-way on zapata, to me at least.. |
| 21:14.21 | sch19 | *zap |
| 21:14.24 | hypa7ia | sch19: we only have 4 lines total, so it would be useful |
| 21:14.56 | hypa7ia | there are like 20 people in the office, we're a little underprovisioned :) |
| 21:15.05 | sch19 | haha, sounds like it |
| 21:15.06 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 21:15.21 | hypa7ia | plus remote users :s |
| 21:15.41 | sch19 | call volume high ? |
| 21:15.45 | hypa7ia | not really |
| 21:16.20 | hypa7ia | just gets busy once in a while, and it'd be better if we could get around the three-way thing |
| 21:17.05 | rob0 | It's OT here, but I'm curious, has anyone here ever set up VLANs on Linux? |
| 21:17.09 | sch19 | that seems just so .. wrong..to me. but i guess milk them for all you can, and good luck :P |
| 21:18.01 | hypa7ia | we'll eventually go to VOIP trunks instead |
| 21:18.19 | rob0 | Specifically I'm wondering if you can activate the vlan interface while the physical interface is still up and has an IP? |
| 21:19.00 | Netgeeks | rob0: thats pretty far OT |
| 21:19.05 | *** join/#asterisk freezer (i=leetiden@ACB0D6E3.ipt.aol.com) |
| 21:19.09 | freezer | hi |
| 21:19.20 | rob0 | Not as bad as the politics which comes up here often :) |
| 21:19.36 | Netgeeks | however, the answer would be yes, I would think |
| 21:19.46 | _Sam-- | politics day? |
| 21:19.48 | sch19 | and asterisk is commonly run on linux, and I would imagine vlans. |
| 21:19.56 | sch19 | so not too far fetched :P |
| 21:20.06 | rob0 | thx sch19 :) |
| 21:20.09 | Netgeeks | you pretty much need the physical interface up and active, otherwise how are your vlan packets going to get to the far end of the tunnel? |
| 21:20.09 | _Sam-- | that was some good politickin' a few days ago |
| 21:20.32 | tainted_ | anyone know why call files that were working before now don't work? |
| 21:20.49 | *** join/#asterisk UberDuper (n=jsatter@elrond.uberduper.com) |
| 21:21.16 | rob0 | Up and promisc, I suppose, but with no IP? I'm guessing it's a lot like bridging, which I *have* done. |
| 21:21.49 | heath__ | tainted_: try generating a random filename for all your call files and then see if you still have a problem |
| 21:22.10 | rpm | whats a good frontend for asterisk which i can login and check my voicemail? |
| 21:22.19 | _Sam-- | a phone? |
| 21:22.22 | tainted_ | heath__ all call file file names are random |
| 21:22.25 | denon | rpm: vmail.cgi |
| 21:22.27 | denon | or whatever it's called |
| 21:22.40 | tainted_ | heath__ generated dynamically by script based on timestamp |
| 21:23.11 | Katty | hi lads. |
| 21:23.21 | _Sam-- | hi katt...how those molars feeling |
| 21:23.27 | rob0 | I'll try it, see if I get cut off :) |
| 21:23.28 | Netgeeks | Greets Katty |
| 21:23.29 | Katty | _Sam--: currently not in pain (= |
| 21:23.34 | Katty | herro, Netgeeks |
| 21:23.40 | rob0 | hi Katty did you have the surgery? |
| 21:23.41 | tainted_ | it used to drop immediately into the into the context i specify but now it doesn't work |
| 21:23.46 | Katty | rob0: surgery is the 17th. |
| 21:23.46 | heath__ | tainted_ that's your problem, if you send calls during the same second, they'll have the same filename and it'll get all messed up |
| 21:23.53 | rob0 | oh good luck then. |
| 21:23.59 | heath__ | use the manager instead if at all possible |
| 21:24.08 | UberDuper | Are there any particular downsides to running * on a fbsd machine? |
| 21:24.12 | _Sam-- | wait til you feel the big holes in your gums where those teeth were...its fun when food gets stuck in there |
| 21:24.13 | Katty | rob0: thanks. |
| 21:24.16 | tainted_ | heath__ i'm just trying to send one call though |
| 21:24.31 | tainted_ | and the timestamp is based on unix time |
| 21:24.47 | tainted_ | Katty what surgery are u having |
| 21:24.53 | tainted_ | i just got some done |
| 21:26.21 | Netgeeks | tainted: she's losing her wisdom teetch |
| 21:26.24 | Netgeeks | teeth |
| 21:27.14 | rob0 | <== no wisdom teeth, and not much wisdom either! |
| 21:27.38 | docelm0 | Katty, wisedom teeth? Does this mean she will be less wise? |
| 21:27.48 | docelm0 | haha.. I had mine took out and I got dumber.. |
| 21:27.56 | UberDuper | =/ |
| 21:28.01 | *** part/#asterisk evilbit (n=brad@interlink-gw1.ilsw.com) |
| 21:28.26 | docelm0 | I liked the perc's they gave me.. GOOD drugs |
| 21:28.50 | *** join/#asterisk jsolares (n=jsolares@200.6.238.195) |
| 21:30.56 | Katty | tainted_: wisdom teeth (= |
| 21:31.40 | _Sam-- | file: you there |
| 21:31.44 | Katty | docelm0: i'll always be a smrt girl. |
| 21:31.45 | _Sam-- | still do not see my 50 posted |
| 21:32.09 | hmodes | hrmm |
| 21:32.26 | hmodes | anyone know why console audio would be extremely garbled with an i810 sound card? |
| 21:32.31 | hmodes | sounds like the sample rate is off or something |
| 21:32.44 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:32.56 | _Sam-- | also file, the only way to get origination is via SIP? Deliver inbound via SIP |
| 21:34.05 | *** join/#asterisk rnovotny22 (n=rnovonty@198.57.19.112) |
| 21:34.27 | file | _Sam--: yes |
| 21:34.48 | file | if you don't attach that service, it goes via IAX2 |
| 21:34.52 | file | to remove it - you have to talk/email me |
| 21:35.06 | _Sam-- | i see...so by default my inbound will right now be coming in via iax2? |
| 21:35.06 | Netgeeks | true automation! |
| 21:35.11 | _Sam-- | or i have to do something to make that happen? |
| 21:35.11 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 21:35.28 | file | if it's registered, it'll go via IAX2 |
| 21:35.32 | _Sam-- | ok then, thank you. |
| 21:35.46 | file | if you need anything else - email support or leave me a private message |
| 21:35.50 | file | I'm going to do make food, do dishes, clean |
| 21:35.54 | _Sam-- | thank you! |
| 21:35.54 | file | standard pedestrian house work |
| 21:35.59 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:37.06 | Hmmhesays | wow its just a rotten day |
| 21:37.07 | docelm0 | Katty, already forgetting your a's.. |
| 21:37.16 | Netgeeks | yay! 1/3 done with converting an interactive web site to CSS support... *puke* |
| 21:37.31 | Katty | docelm0: hmm? |
| 21:37.40 | Katty | docelm0: oh. |
| 21:37.45 | Katty | docelm0: you clearly missed my reference. |
| 21:37.59 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:38.23 | rob0 | <== Still here, despite vlan experimentation :) |
| 21:38.25 | Katty | Hmmhesays: did you break a string? |
| 21:38.35 | Netgeeks | I heard your reference fly overhead... ;) decided not to expose my lack of understanding until now |
| 21:38.39 | Hmmhesays | Katty: 2 things I need to work are not working |
| 21:38.50 | Netgeeks | and he's missing beer and pizza |
| 21:38.51 | Hmmhesays | and I have nowhere to turn for support |
| 21:39.24 | Katty | Hmmhesays: uh oh :< |
| 21:39.39 | Katty | Hmmhesays: i can only offer emotional support. |
| 21:39.50 | Katty | Hmmhesays: so, i can hold your paw if you want. |
| 21:40.15 | *** join/#asterisk Samoied (n=Samoied@200.193.91.136) |
| 21:40.44 | *** join/#asterisk slak- (i=slak@rewted.biz) |
| 21:40.49 | Netgeeks | have you posted a question to the ser mail list? or is it less helpful that asterisk-users? |
| 21:40.54 | slak- | sup |
| 21:41.11 | slak- | need a sip provider that does international calls (to the UK from usa) |
| 21:41.15 | slak- | any suggestions? |
| 21:44.28 | dudes | http://binfone.com/ ? |
| 21:44.42 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:45.03 | Hmmhesays | woooot emotional support |
| 21:45.11 | Hmmhesays | does a pack of smokes and a beer come with that? |
| 21:47.49 | Hmmhesays | hmmm 96 simultaneous calls on a dual proc xeon 3.0ghz, no transcoding, yes I do believe so |
| 21:48.51 | Katty | Hmmhesays: you know i don't drink or smoke. |
| 21:48.58 | Netgeeks | 96 on a dial proc Xeon 3.0 is cake |
| 21:49.09 | Katty | Hmmhesays: but i will hug you. |
| 21:49.23 | Hmmhesays | that's ok Katty: i do enough for both of us |
| 21:49.26 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 21:49.37 | Renacor | anybody point me to a good doc on how to setup asterisk cdr? |
| 21:50.37 | Netgeeks | I had the opportunity to test some super-micro dual 3.0G Xeons with 4 4port sangomas and was able to sustain all 16 PRIs fully loaded with calls with no noticable issues |
| 21:51.10 | _Sam-- | no transcoding? |
| 21:51.11 | slak- | i wish there was a way to test the latency between me and some of these sip providers |
| 21:51.14 | twisted[asteria] | Hmmhesays, what you need man? |
| 21:51.19 | Netgeeks | PRI to ulaw |
| 21:51.25 | _Sam-- | slak-: isnt that ping? or mtr or something? |
| 21:51.29 | slak- | ya |
| 21:51.33 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:51.35 | Hmmhesays | auth_radius on SER |
| 21:51.42 | slak- | but i would need to know their sip servers ip |
| 21:51.51 | _Sam-- | i have 5 providers...who you looking for |
| 21:51.53 | twisted[asteria] | Hmmhesays, hmm.... I seem to remember that working... does it not? |
| 21:51.57 | slak- | [TK]D-Fender: sup bud, hey man im looking for a sip provider that supports International calls |
| 21:52.05 | slak- | Sam: international support |
| 21:52.17 | Hmmhesays | that is my current hang up, well everything compiles fine, i have all my libraries, when I load the module I get a generic error with no debugging info |
| 21:52.19 | slak- | something reputable |
| 21:52.21 | _Sam-- | if you have a company who's sip/iax server you want to trace to, just let me know the company |
| 21:52.26 | _Sam-- | and i may have an ip/name |
| 21:52.29 | twisted[asteria] | Hmmhesays, strange. what build of ser? |
| 21:52.31 | [TK]D-Fender | slak- : No idea... Haven't ever used a real SIP>PSTN provider before |
| 21:52.52 | slak- | sam: binfone.com |
| 21:53.00 | slak- | id like to test the latency between us and them |
| 21:53.15 | Hmmhesays | ser 0.9.6 and openser1.0 behave the same |
| 21:53.21 | twisted[asteria] | ahhh |
| 21:53.22 | _Sam-- | sorry my list is teliax, asterlink, broadvoice, ,sixtel and junction networks |
| 21:53.30 | slak- | okay ill check those |
| 21:53.39 | slak- | i have asterlink now..exeptional latency..17ms |
| 21:53.42 | Hmmhesays | hell I even tried the binaries for openSER |
| 21:53.43 | slak- | but no internatrional support |
| 21:53.49 | _Sam-- | my asterlink is 9ms away! |
| 21:53.51 | _Sam-- | incredible really |
| 21:54.09 | twisted[asteria] | i've seen it work on 0.10.99-dev5 |
| 21:54.38 | Hmmhesays | you happend to have a config file? |
| 21:55.39 | twisted[asteria] | not handy, but this link: http://www.iptel.org/ser/doc/ser_radius/ser_radius.html was used as reference |
| 21:55.43 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:56.05 | Hmmhesays | I followed that to a T, except I didn't set up freeradius |
| 21:56.20 | twisted[asteria] | try with ser 0.10.99-dev5 |
| 21:56.34 | twisted[asteria] | or one of the development sources |
| 21:56.45 | twisted[asteria] | the 'stable' stuff seems to blow up more often |
| 21:56.45 | file | twisted[asteria]: is that the one I gave you? |
| 21:56.50 | twisted[asteria] | file, i think so, ya |
| 21:57.00 | file | I wonder where I put it... |
| 21:57.09 | twisted[asteria] | i can make it a link in a jiffy |
| 21:57.20 | *** join/#asterisk drega (n=drega@p54A09D26.dip0.t-ipconnect.de) |
| 21:57.47 | file | http://quark.file-radio.com/ser/ser-gemini.tgz or http://quark.file-radio.com/ser/ser-jcolp.tgz |
| 21:57.48 | file | one of those |
| 21:58.14 | _Sam-- | file i felt bad interuppting your household duties...but when i call the 800 number assigned to me...it doesnt ring anywhere |
| 21:58.15 | Hmmhesays | how far did you get on gemini file? |
| 21:58.20 | _Sam-- | it doesnt make it to my asterisk |
| 21:58.25 | _Sam-- | even though im registered to you on iax |
| 21:58.30 | file | _Sam--: what's the number? |
| 21:58.46 | _Sam-- | pm... |
| 21:59.00 | twisted[asteria] | i also have it at http://www.indigent-networks.com/ser-0.10.99-dev5.tar.gz |
| 21:59.01 | file | Hmmhesays: well, the module in that tarball is enough of a base that you could do a full fledged load balancing, authentication, and billing |
| 21:59.23 | Hmmhesays | cool |
| 21:59.37 | file | so if you wanna play with it - power to you1 |
| 21:59.46 | Hmmhesays | danke |
| 22:00.20 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 22:01.10 | slak- | _Sam--: how about teliax |
| 22:01.11 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:01.18 | slak- | whats their sip server ip |
| 22:01.21 | _Sam-- | slak-: your mileage will vary with teliax |
| 22:01.28 | _Sam-- | some people are fine with their routes there, some arent |
| 22:01.38 | _Sam-- | bani, for example, has perfect routes to teliax, no packet loss... |
| 22:01.39 | slak- | _Sam--: okay? out of the list you posted whats the best one besides asterlink |
| 22:01.44 | _Sam-- | and his stuff doesnt sound good |
| 22:01.50 | *** join/#asterisk Mother_ (i=Mother@62.87.119.95) |
| 22:01.53 | Mother_ | greetings |
| 22:01.53 | _Sam-- | my stuff is further away, more hops...and sounds fine to teliax |
| 22:02.01 | slak- | it has to sound good |
| 22:02.07 | slak- | or ill get my as chewed |
| 22:02.09 | slak- | ass |
| 22:02.10 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:02.11 | _Sam-- | for ME, the next best one IS teliax...but not for many others |
| 22:02.15 | Mother_ | is there a way to disable call waiting on SIP clients from within *? |
| 22:02.21 | *** join/#asterisk beebz (n=bbz@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 22:02.26 | _Sam-- | the teliax host is voip-co2.teliax.com |
| 22:02.48 | Mother_ | I want attended transfers to get a busy tone if the callee is already in a call |
| 22:02.48 | _Sam-- | the best part about teliax, for me...is that the main guy there is really good to em. |
| 22:02.50 | _Sam-- | er to me. |
| 22:02.57 | slak- | rtt min/avg/max/mdev = 57.763/62.513/64.695/2.162 ms |
| 22:03.02 | slak- | i dont like that too much |
| 22:03.49 | Mother_ | anyone? |
| 22:04.11 | _Sam-- | my times to teliax are the same as the ones you just pasted...and my calls are perfect |
| 22:04.21 | slak- | okay |
| 22:04.24 | slak- | ill give them a shot |
| 22:04.35 | _Sam-- | what backbone network are you on? |
| 22:04.40 | slak- | speakeasy |
| 22:04.59 | _Sam-- | when you traceroute to teliax, what networks do you go across |
| 22:05.37 | slak- | speakeasy, level3, pnap, rockynet, teliax |
| 22:05.45 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 22:06.13 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 22:06.27 | JunK-Y | some1 has played a lot with waitforsilence? |
| 22:07.44 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 22:07.46 | austinnichols101 | rockynet.demarc.cogentco.com seems like it has a lot of latency right now |
| 22:08.11 | slak- | <PROTECTED> |
| 22:08.11 | slak- | 10 border3.ge3-0-bbnet2.den.pnap.net (216.52.40.71) 60.462 ms 59.259 ms 60.420 ms |
| 22:08.14 | slak- | 11 rockynet-1.border3.den.pnap.net (63.251.181.222) 65.452 ms 78.377 ms 61.688 ms |
| 22:08.18 | slak- | 12 f-5-0-0-cd2.rockynet.com (199.45.239.148) 65.954 ms 72.163 ms 75.647 ms |
| 22:08.19 | slak- | next hop is teliax |
| 22:08.34 | Qwell[] | That's a lot of jitter... |
| 22:08.57 | tainted_ | i've been having some problems with teliax to certain rate centers |
| 22:09.03 | slak- | it begins with this: |
| 22:09.04 | slak- | <PROTECTED> |
| 22:09.09 | *** part/#asterisk MatsK (n=mk@84-217-5-20.tn.glocalnet.net) |
| 22:09.10 | slak- | before that its like 20ms |
| 22:09.29 | slak- | nyc-denver route |
| 22:10.02 | SkramX | anyone written a script to check IMAP mail via Asterisk? That would be pretty cool ;) |
| 22:10.32 | De_Mon | can multiple people register to the same extension? |
| 22:10.44 | tronix | SkramX: it'd sound funny if they read off subject lines of pr0n spam ;) |
| 22:11.06 | De_Mon | tronix you still get spam? |
| 22:11.18 | SkramX | tronix: would be. |
| 22:11.19 | tronix | De_Mon: occasionally. not a lot. |
| 22:11.34 | SkramX | I do, I have one email account which I do not filter, as Id rather not loose a message or two. |
| 22:11.37 | SkramX | ;( |
| 22:12.09 | De_Mon | SkramX 99.98% isn't good enough eh? |
| 22:12.17 | Hmmhesays | I get the exact same problem wiht the dev version |
| 22:13.30 | *** join/#asterisk Triple1243 (n=Triple12@modemcable171.79-70-69.mc.videotron.ca) |
| 22:13.31 | twisted[asteria] | Hmmhesays, what does ser -c say? |
| 22:13.54 | *** join/#asterisk Simon- (i=fictitio@80.193.211.68) |
| 22:16.01 | *** join/#asterisk badboyz (n=bbz@adsl-70-128-78-21.dsl.stlsmo.swbell.net) |
| 22:16.27 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:16.42 | Hmmhesays | config file good |
| 22:16.49 | Hmmhesays | I don't freaking get it |
| 22:17.08 | Hmmhesays | stateless - initializing |
| 22:17.08 | Hmmhesays | textops - initializing |
| 22:17.08 | Hmmhesays | ERROR: error while initializing modules |
| 22:17.34 | shido6 | :) |
| 22:17.46 | Qwell[] | shido6: Get the email I sent you? |
| 22:17.55 | shido6 | yes, sir. |
| 22:17.59 | Qwell[] | k |
| 22:19.56 | twisted[asteria] | Hmmhesays, and it starts up fine with radius module omitted? |
| 22:20.03 | Hmmhesays | yessah |
| 22:20.21 | Hmmhesays | starts up, i put in a simple stateless forward, that works fine |
| 22:20.36 | twisted[asteria] | hrm. odd. |
| 22:21.02 | Seldon1975 | can someone please remind me how to pipe 'find' result to 'rm' |
| 22:21.23 | twisted[asteria] | rtfm |
| 22:21.23 | tainted_ | twisted[asteria] |
| 22:21.28 | twisted[asteria] | lol |
| 22:21.31 | twisted[asteria] | j/k |
| 22:21.39 | *** part/#asterisk Seyr (n=Seyr_@64.242.187.194) |
| 22:21.39 | *** join/#asterisk bluecrow76 (n=Mitch@ip24-250-77-177.br.br.cox.net) |
| 22:21.41 | tainted_ | twisted[asteria] that app_bridge is so awesome |
| 22:21.50 | Damin | twisted[asteria]: Hey there.. |
| 22:22.01 | Damin | twisted[asteria]: Any chance of getting that Astricon voicemail? |
| 22:22.02 | tainted_ | thank you thank you thank you and heath and mattf |
| 22:22.17 | Cresl1n | ??? |
| 22:22.21 | twisted[asteria] | tainted_, heh.. np. |
| 22:22.23 | twisted[asteria] | Cresl1n, wrong matt. |
| 22:22.38 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 22:22.46 | twisted[asteria] | Cresl1n, he's talking about matt florell |
| 22:23.39 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 22:23.46 | wunderkin | matt florell is on irc? |
| 22:25.24 | twisted[asteria] | Damin, possibly, i don't know when |
| 22:25.38 | Cresl1n | oooh.... bad mistake |
| 22:25.52 | twisted[asteria] | ? |
| 22:26.01 | Cresl1n | me and matt florell |
| 22:26.14 | *** join/#asterisk davidj (n=davidj@cpe-024-074-080-163.carolina.res.rr.com) |
| 22:26.18 | twisted[asteria] | sitting in a tree? |
| 22:26.24 | wunderkin | kissing? |
| 22:26.35 | Hmmhesays | f-*-$-%-ing? |
| 22:27.19 | Damin | Essobi: Yes.. caveats is that you must be disclaimed and it must go up on the bugtracker.. |
| 22:27.26 | tracinet | exit |
| 22:27.36 | tracinet | oops - wrong window LOL |
| 22:28.39 | *** join/#asterisk mut (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 22:29.22 | Hmmhesays | i'll be your #1 with a bullet, loaded gun complex, cock it and pull it |
| 22:29.25 | *** join/#asterisk cassio- (i=cassio-@200216087155.user.veloxzone.com.br) |
| 22:30.14 | Dr-Linux | Hmmhesays: wtf did you just said? :S |
| 22:30.24 | wunderkin | he said cock |
| 22:30.37 | Hmmhesays | haha |
| 22:30.44 | Hmmhesays | a song i have in my head |
| 22:31.11 | Dr-Linux | then wash your head :) |
| 22:31.17 | Hmmhesays | good song though |
| 22:31.32 | Hmmhesays | the funny thing about the minds eye/ear... you can't erase things |
| 22:31.42 | Dr-Linux | you like cock? :S |
| 22:31.46 | cassio- | I have installed asterisk as digium says, but after the last make clean ; make install when I type asterisk -vvvvc I get unknown command |
| 22:31.47 | Dr-Linux | or song? |
| 22:31.54 | file | sadly he doesn't |
| 22:32.01 | Hmmhesays | how the fark did you go from that song lyric to "i like cock" |
| 22:32.31 | *** join/#asterisk geoffb1974 (n=gbodeman@mail.televerde.com) |
| 22:32.33 | Hmmhesays | yeah chicks are more my thing |
| 22:32.39 | geoffb1974 | hey all |
| 22:32.49 | file | doesn't mean I won't pick on you Hmmhesays |
| 22:32.52 | geoffb1974 | any chance someone could help a newbie |
| 22:32.54 | file | nice spikey hair |
| 22:33.01 | Dr-Linux | ok i'm sorry |
| 22:33.08 | Hmmhesays | i chopped it all off when I was on my voip hiatus |
| 22:33.10 | Dr-Linux | i don't know much english words |
| 22:33.16 | *** join/#asterisk clive- (n=pirch@dsl-165-157-205.telkomadsl.co.za) |
| 22:33.17 | file | Hmmhesays: it'll grow back. |
| 22:33.20 | file | eventually |
| 22:33.21 | Hmmhesays | oh its back |
| 22:33.27 | file | excellent |
| 22:33.29 | Hmmhesays | and pitch black |
| 22:33.29 | geoffb1974 | setting up an asterisk box |
| 22:33.34 | file | very good |
| 22:33.39 | Hmmhesays | and need the look for the band |
| 22:33.41 | geoffb1974 | have registered with FWD |
| 22:33.51 | geoffb1974 | but cannot run echo test |
| 22:34.04 | geoffb1974 | can anybody help? |
| 22:34.21 | *** join/#asterisk bjames_ (n=bjames@67-102-228-17.adsl.lbdsl.net) |
| 22:34.44 | dudes | cassio- - why would you make clean ... make install ? |
| 22:35.01 | cassio- | dudes yes, I did like the tutorial says |
| 22:35.05 | cassio- | I will pastebin |
| 22:35.31 | dudes | perhaps try ... make clean && make && make install |
| 22:35.34 | geoffb1974 | can call internal extensions only right now |
| 22:35.49 | geoffb1974 | and even then they cannot hear me but I hear them |
| 22:35.52 | dudes | that's not * specific since it's pretty standard stuff on *nix |
| 22:36.01 | cassio- | dudes take a look |
| 22:36.02 | cassio- | http://pastebin.com/536082 |
| 22:36.51 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 22:37.02 | dudes | perhaps /usr/bin/ isn't in your profile |
| 22:37.03 | De_Mon | asterisknix :) |
| 22:37.08 | geoffb1974 | can anyone help me? |
| 22:37.16 | dudes | try /usr/bin/asterisk |
| 22:37.36 | cassio- | [root@cassiorozelli src]# /usr/bin/asterisk |
| 22:37.37 | cassio- | bash: /usr/bin/asterisk: No such file or directory |
| 22:37.48 | dudes | then do as I mentioned |
| 22:37.52 | geoffb1974 | running into very frustrating troubles |
| 22:38.03 | cassio- | ok |
| 22:38.04 | wunderkin | it would be /usr/sbin/asterisk or /usr/local/sbin/asterisk |
| 22:38.29 | dudes | opps, my bad |
| 22:38.47 | cassio- | I started recompiling again :) |
| 22:39.04 | geoffb1974 | anyone have a memory leak issue with 1.2.1??? |
| 22:39.20 | Qwell[] | geoffb1974: See channel topic |
| 22:40.07 | geoffb1974 | I understand channel topic, the info I have seen online refers to 1.2.2 and above |
| 22:40.12 | Qwell[] | 1.2.x |
| 22:40.15 | geoffb1974 | was wondering if it also apllied to 1.2.1 |
| 22:40.19 | Qwell[] | yes |
| 22:40.21 | geoffb1974 | thanks |
| 22:40.53 | geoffb1974 | does it seem that 1.2.4 helps this issue? |
| 22:40.57 | Qwell[] | yes |
| 22:41.12 | Qwell[] | That's why the topic says it was fixed.. |
| 22:41.15 | cassio- | about load balance between lines, does anyone know about it? |
| 22:41.28 | *** part/#asterisk telcoDEJ (n=ed@c-68-82-75-221.hsd1.de.comcast.net) |
| 22:41.37 | Qwell[] | brb |
| 22:41.46 | Seldon1975 | all your base |
| 22:41.46 | Seldon1975 | base |
| 22:41.54 | geoffb1974 | .2.3 and .4 came out so quickly |
| 22:42.06 | geoffb1974 | did they all apply to memory leak issue |
| 22:42.37 | *** join/#asterisk Soul (n=Soul@87-196-39-16.net.novis.pt) |
| 22:43.01 | geoffb1974 | hey qwell |
| 22:43.26 | geoffb1974 | anyone in here have a setup that is behind a NAT's router and using SIP |
| 22:44.32 | cassio- | [root@cassiorozelli root]# /usr/sbin/asterisk -vvvvvvvvvvvvgcr |
| 22:44.33 | cassio- | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 22:44.54 | cassio- | what does that mean? |
| 22:45.10 | JonR800 | there is no remote console |
| 22:45.17 | clive- | cassio you need to make config..I think... |
| 22:45.21 | JonR800 | c tells it to start in a local console |
| 22:45.28 | JonR800 | remove the r and it'll start |
| 22:45.33 | cassio- | clint_ I already did |
| 22:45.47 | cassio- | JonR800 thanks |
| 22:45.54 | geoffb1974 | hey cassio |
| 22:45.56 | JonR800 | np |
| 22:46.00 | geoffb1974 | you have a moment? |
| 22:46.47 | [TK]D-Fender | geoffb1974 : plenty of people do.. Just ask your real qustion. |
| 22:47.13 | geoffb1974 | thanks |
| 22:47.40 | cassio- | yes geoffb1974 |
| 22:47.59 | geoffb1974 | my real question is this...I have setup a very new very basic server. I can currently make calls between internal SIP extensions only |
| 22:48.07 | geoffb1974 | just added FWD IAX trunk |
| 22:48.17 | geoffb1974 | but I fail my echo tests |
| 22:48.26 | geoffb1974 | and no one can hear me |
| 22:48.30 | geoffb1974 | I can hear them |
| 22:48.41 | geoffb1974 | I am behind a linksys router |
| 22:49.03 | geoffb1974 | but have port forwarded what i beleive are all needed UDP and TCP ports |
| 22:49.13 | [TK]D-Fender | geoffb1974 : You need to add on of "EXTERNHOST" or "EXTERNIP", and "LOCALNET" clauses to your SIP.conf and forward the appropriate ports to you * box (5060,10000-20000 all UDP) |
| 22:50.02 | geoffb1974 | done that |
| 22:50.03 | JonR800 | that doesn't answer the FWD IAX question. |
| 22:50.15 | JonR800 | people on FWD can't hear you right? |
| 22:50.23 | geoffb1974 | right |
| 22:50.32 | geoffb1974 | even loacl extensions can't hear |
| 22:50.40 | geoffb1974 | echo test fails |
| 22:50.44 | JonR800 | oh |
| 22:50.51 | geoffb1974 | I have simple eccho set to ext 611 locally |
| 22:50.53 | jsolares | your atas have a busted mic |
| 22:50.56 | jsolares | :D |
| 22:51.19 | geoffb1974 | tested mic, using softphones |
| 22:51.35 | geoffb1974 | tried from 2 pc's one mac |
| 22:52.00 | JonR800 | geoffb1974: are you sure you specified externip/localnet correctly? |
| 22:52.03 | jsolares | the problem is to FWD, anything to FWD and they cant hear you? |
| 22:52.13 | jsolares | maybe your firewall blocks outgoing ports |
| 22:52.16 | *** join/#asterisk Darkness (n=ArmyOf@authority.lolice.org) |
| 22:52.22 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 22:52.25 | JonR800 | he said local aren't working either now. |
| 22:52.30 | Darkness | hello |
| 22:52.43 | jsolares | kick the server and restart it |
| 22:52.47 | jsolares | oh wait it isnt windows |
| 22:52.52 | geoffb1974 | :) |
| 22:53.14 | geoffb1974 | [general] |
| 22:53.14 | geoffb1974 | port =5060 |
| 22:53.14 | geoffb1974 | bindaddr=0.0.0.0 |
| 22:53.14 | geoffb1974 | externip=68.2.52.70 |
| 22:53.14 | geoffb1974 | localnet=192.168.1.0/255.255.255.0 |
| 22:53.14 | geoffb1974 | srvlookup=yes |
| 22:53.16 | geoffb1974 | nat=yes |
| 22:53.18 | geoffb1974 | qualify=2000 |
| 22:53.36 | JonR800 | turn nat=yes off |
| 22:53.47 | JonR800 | specify nat=yes only in users that are nat'd |
| 22:54.04 | jsolares | nat'd to the asterisk box |
| 22:54.16 | geoffb1974 | box is nat'd |
| 22:54.17 | JonR800 | yes |
| 22:54.27 | JonR800 | but local users aren't going through nat to reach the asterisk box |
| 22:54.50 | JonR800 | only specify nat=yes in users that you have to traverse the nat to reach your asterisk. |
| 22:54.57 | geoffb1974 | ok |
| 22:54.58 | geoffb1974 | done |
| 22:55.37 | jsolares | and since you're forwarding ports, anyone with a public IP is not going thru nat per se, only SIP phones behind another nat |
| 22:55.56 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 22:55.58 | geoffb1974 | so nat=no or nat=off |
| 22:56.06 | JonR800 | jsolares: but asterisk won't send the correct externip without nat=yes.. correct? |
| 22:56.12 | JonR800 | nat=no i think |
| 22:56.47 | geoffb1974 | ok |
| 22:56.53 | geoffb1974 | so current issue with echo |
| 22:57.02 | jsolares | try it |
| 22:57.04 | geoffb1974 | have tried other boxes, so I know the mic works |
| 22:58.21 | JonR800 | im willing to bet the nat setting was the cause |
| 22:58.26 | JonR800 | and if not then i bet nothing :) |
| 22:58.49 | geoffb1974 | restarting the box |
| 22:59.40 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:59.44 | geoffb1974 | no luck |
| 23:00.12 | geoffb1974 | same thing internal echo test not working |
| 23:00.23 | geoffb1974 | as well as dialing 613 for FWD echo test |
| 23:00.27 | JonR800 | can users call each other and hear each other? |
| 23:00.32 | JonR800 | internal that is |
| 23:00.47 | geoffb1974 | I can hear them |
| 23:00.52 | geoffb1974 | they can't hear me |
| 23:01.15 | slak- | hey |
| 23:01.36 | geoffb1974 | I am offsite now using port forwarding, but tested locally 30 minutes ago |
| 23:01.42 | geoffb1974 | same thing |
| 23:01.53 | geoffb1974 | so I don't think NAT is the cause |
| 23:02.13 | JonR800 | well that nat setting would have caused an issue internally. |
| 23:02.46 | Darkness | i've a problem detecting hangup with my TDM400P (2 FXS, 1 FXO) when fxo-fxs bridged, my defaultzone is set to FR, I try hanguponpolarityswitch=yes without success, does anybody have an idea ? (soft version : asterisk 1.2.4, zaptel 1.2.3) |
| 23:03.03 | Ariel_ | one way audio is caused by nat issue and firewall, IP address need to be setup correctly with sip externip= localnet= settings |
| 23:04.24 | geoffb1974 | I have that set ariel |
| 23:04.45 | geoffb1974 | not sure what else to do |
| 23:05.29 | Ariel_ | so you can hear them which means there rtp sound stream is leaving there network. but yours in not getting to them. |
| 23:05.29 | jsolares | you removed the nat=yes from [general], if so restart and retry |
| 23:05.43 | geoffb1974 | I did |
| 23:05.50 | geoffb1974 | currently have nat=no |
| 23:05.54 | geoffb1974 | should I just take it out |
| 23:05.59 | jsolares | then goto kick server, restart windows |
| 23:06.01 | *** join/#asterisk MstlyHrm1s (n=mh@melbourne.mostly-harmless.ca) |
| 23:06.03 | jsolares | yeah try taking it out |
| 23:06.14 | JonR800 | geoffb1974: no is default. |
| 23:06.28 | JonR800 | geoffb1974: from the examples no = Use NAT mode only according to RFC3581 |
| 23:06.42 | JonR800 | yes = Always ignore info and assume NAT |
| 23:06.52 | ahattar | hi all |
| 23:07.04 | jsolares | ebay sucks |
| 23:07.05 | cassio- | how can I make the /usr/sbin/asterisk able to type just asterisk -vvvv in any dir in the system? |
| 23:07.21 | jsolares | cassio- : add /usr/sbin to your path |
| 23:07.28 | JonR800 | geoffb1974: on your client do you have nat set to yes? also maybe try putting nat=yes under the user you're registered as. |
| 23:07.34 | cassio- | jsolares how do I do that? |
| 23:07.35 | jsolares | export PATH=$PATH:/usr/sbin or something |
| 23:07.38 | cassio- | ok |
| 23:07.49 | jsolares | that depends on your shell |
| 23:07.52 | JonR800 | cassio-: use safe_asterisk if possible |
| 23:07.57 | *** join/#asterisk lehel (i=lehel@86.125.98.100) |
| 23:08.49 | Darkness | nobody for my hangup detection problem ? :( |
| 23:09.21 | cassio- | great, I got in asterisk, but when I typed stop it didnt do anything, or stop now, then I did ctrl+c and said Yuck! Error in buffer handling...: Broken pipe |
| 23:10.05 | jsolares | odd that stop now did nothing |
| 23:10.08 | geoffb1974 | well progress a bit |
| 23:10.10 | JonR800 | sometimes stop now means stop in a minute |
| 23:10.12 | geoffb1974 | internal echo test works now |
| 23:10.32 | geoffb1974 | anyone know if 613 at FWD actually works |
| 23:10.40 | JonR800 | geoffb1974: nice, glad to hear at least one thing is working. |
| 23:10.41 | cassio- | JonR800 what can I do to stop at the moment? |
| 23:10.52 | jsolares | killall -9 asterisk? |
| 23:10.59 | Qwell[] | JonR800: now means now |
| 23:11.12 | JonR800 | Qwell[]: yes but sometimes it doesn't listen so well. |
| 23:11.13 | Qwell[] | s/w//g |
| 23:11.15 | jsolares | he claims it did nothing |
| 23:11.17 | Qwell[] | :D |
| 23:11.19 | JonR800 | haha |
| 23:11.35 | *** join/#asterisk scon__ (n=scon@dslb-084-057-000-058.pools.arcor-ip.net) |
| 23:12.50 | cassio- | but it did not |
| 23:14.45 | De_Mon | Qwell[] lol |
| 23:15.55 | cassio- | so the command is stop now to stop asterisk?or is there another command? |
| 23:16.12 | _Sam-- | stop |
| 23:16.20 | _Paulo_ | help stop |
| 23:16.21 | _Sam-- | stop gracefully |
| 23:16.35 | austinnichols101 | geoffb197: yup - echo test is working |
| 23:16.55 | slak- | Feb 2 18:16:42 WARNING[20033]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 23:16.58 | slak- | :D |
| 23:17.01 | slak- | whats that mean |
| 23:17.17 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 23:17.23 | geoffb1974 | so echo tests works |
| 23:18.05 | geoffb1974 | but another user who is NATstill can't hear |
| 23:18.11 | cassio- | Yuck! Error in buffer handling...: Broken pipe what does it mean? |
| 23:18.28 | slak- | thats just me Hacking the Planet |
| 23:18.30 | slak- | dont worry |
| 23:18.34 | *** join/#asterisk mauilion (n=Rocky@tor/session/x-3ef692117a8752fe) |
| 23:18.36 | geoffb1974 | opened 4569 for IAX |
| 23:18.37 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 23:18.51 | geoffb1974 | 5004-5082 on TCP and UDP |
| 23:19.01 | geoffb1974 | and 10000-20000 on TCP UDP |
| 23:19.10 | jsolares | geoffb1974: if the user behind nat is conecting to your asterisk box, then you need to set nat=yes on that user's definition |
| 23:19.12 | geoffb1974 | am i missing something |
| 23:19.13 | JonR800 | geoffb1974: same thing... do they have nat specified in their client, and if that doesn't help specify nat=yes in their entry in sip.conf |
| 23:19.48 | geoffb1974 | I did |
| 23:19.51 | geoffb1974 | on both of us |
| 23:19.53 | JonR800 | oh |
| 23:20.01 | geoffb1974 | we are at work now, behind router |
| 23:20.13 | geoffb1974 | connecting to my system at home, behinf router |
| 23:20.21 | jsolares | try setting canreinvite=no on both users |
| 23:21.04 | jsolares | asterisk might be trying to conect both on their own |
| 23:21.39 | JonR800 | good call jsolares |
| 23:21.42 | jsolares | or, ditch sip and move to IAX phones/ata's/softphones |
| 23:22.32 | jsolares | rather IAX2 |
| 23:23.28 | geoffb1974 | that's what I keep reading |
| 23:24.53 | JonR800 | SIP is doable. it's just trickier through a nat. IMHO sip.conf settings for nat make it harder than it should be, but that's besides the point. |
| 23:25.12 | geoffb1974 | smae thing still happening |
| 23:25.28 | geoffb1974 | I did a test this morning with a friend in NY |
| 23:25.37 | geoffb1974 | now doing a test with someone sitting next to me |
| 23:25.37 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 23:25.40 | geoffb1974 | I hear them fine |
| 23:25.53 | Umaro | hey guys.. anyone using a zhone zplex10A channel bank? |
| 23:26.45 | jsolares | softphones? |
| 23:26.50 | *** join/#asterisk chaos1 (i=anonymou@klover226.bitnet.nu) |
| 23:26.52 | chaos1 | hi |
| 23:26.52 | jsolares | hardphones? |
| 23:26.57 | Ariel_ | zhone wow have not seen one in a long time. |
| 23:27.10 | geoffb1974 | alright |
| 23:27.13 | jsolares | if softphones, may you have a firewall on your pc? |
| 23:27.18 | geoffb1974 | will see you all tomorrow |
| 23:27.25 | geoffb1974 | at least one thing is working |
| 23:27.27 | *** join/#asterisk mauilion (i=debian-t@tor/session/x-d3b203a9489a49c3) |
| 23:27.39 | *** part/#asterisk chaos1 (i=anonymou@klover226.bitnet.nu) |
| 23:27.45 | *** join/#asterisk chaos1 (i=anonymou@klover226.bitnet.nu) |
| 23:27.45 | *** join/#asterisk areski (n=areski@250.Red-83-60-91.dynamicIP.rima-tde.net) |
| 23:27.54 | chaos1 | hi |
| 23:28.11 | geoffb1974 | trying that |
| 23:28.21 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com) |
| 23:28.29 | Umaro | Ariel_: oh? zplex10b or zplex10a? :) |
| 23:28.45 | Umaro | Ariel_: zplex10b are great.. zplex10a is turning out to suck pretty bad. |
| 23:28.52 | Ariel_ | Umaro, don't know have nto used them for some time now 3 years |
| 23:29.18 | Umaro | Ariel_: oic |
| 23:29.47 | chaos1 | i have some users registered with SER coming into asterisk box. I would like them to be queued in asterisk based on priority |
| 23:29.54 | chaos1 | is it possible |
| 23:29.59 | Ariel_ | Umaro, I know it's the zplex10 but did not see any a or b in the model number |
| 23:30.16 | *** part/#asterisk lehel (i=lehel@86.125.98.100) |
| 23:30.20 | jsolares | everything is possible, it's just a question on how much coding/money you are willing to put in ;) |
| 23:30.37 | jsolares | that being said, i have not done it, nor have i used SER |
| 23:30.39 | wunderkin | any sphinx3 users here? |
| 23:30.55 | chaos1 | jsolaresL:that was pretty much an abstract statement |
| 23:31.33 | jsolares | that is true |
| 23:31.37 | chaos1 | what if the agents are not registered on asterisk, and you have to check whether they are busy or not before forwarding a call to them |
| 23:31.40 | chaos1 | how to do that |
| 23:32.03 | chaos1 | secondly, is there a secure asterisk implementation available in form of distribution as well. I kmow AMP |
| 23:33.00 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:34.19 | chaos1 | darn |
| 23:34.25 | chaos1 | times up |
| 23:34.29 | chaos1 | bye |
| 23:34.33 | Umaro | has anyone used the lucent connectreach channel banks? |
| 23:35.05 | *** join/#asterisk shinta_kenshin (n=jsolares@200.6.238.195) |
| 23:35.26 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 23:35.42 | danzig | EHLO all! |
| 23:36.07 | jsolares | ah my isp is dying again |
| 23:36.43 | rob0 | danzig: no, Asterisk doesn't speak ESMTP |
| 23:37.11 | rob0 | And aren't you supposed to be Gdansk these days? |
| 23:37.28 | danzig | Can someone explain how CDR and ForkCDR work - I want to do something likeexten => _XXXXX.,5,Dial(SIP/${t}-${ACC}/${EXTEN},30) |
| 23:37.29 | danzig | exten => _XXXXX.,6,Goto(_XXXXX.-${DIALSTATUS},1) ; |
| 23:37.31 | danzig | exten => _XXXXX.,7,Congestion |
| 23:37.32 | danzig | exten => _XXXXX.-CHANUNAVAIL,1,BackGround(channel) |
| 23:37.34 | danzig | exten => _XXXXX.-CHANUNAVAIL,2,BackGround(is-curntly-unavail) |
| 23:37.35 | danzig | exten => _XXXXX.-CHANUNAVAIL,3,Hangup |
| 23:37.37 | danzig | But then I cannot see in the CDR file what number the person tried to call. How do I get * to write that instead of "_XXXXX.-CHANUNAVAI"? |
| 23:37.43 | litage | when someone says "this is the H.323 stack code", what exactly do they mean? |
| 23:38.06 | ahattar | hi all |
| 23:38.21 | Ariel_ | danzig, next time use pastebin.ca |
| 23:38.54 | Ariel_ | litage, there sending you a call via h323 |
| 23:38.56 | danzig | ok, sorry |
| 23:39.37 | litage | Ariel_: i was referring to the actual code, not sending/receiving calls, etc |
| 23:40.06 | litage | the H.323 channel driver uses the H.323 stack code. i'm wondering what exactly a channel driver and stack code are |
| 23:40.37 | rnovotny22 | Does anyone know if Broadvoice has been having audio problems the last couple of days? |
| 23:41.03 | *** part/#asterisk cassio- (i=cassio-@200216087155.user.veloxzone.com.br) |
| 23:42.22 | wunderkin | rnovotny22, i did yesterday... but i dont get many calls, i think that was the only one |
| 23:42.23 | *** join/#asterisk YARICK (i=YARICK@usg-206-135-187-99.portnetworks.net) |
| 23:43.10 | rnovotny22 | wunderkin, Thanks, we've been getting intermittant receive for 2 days now. |
| 23:43.35 | *** join/#asterisk santiago (n=santiago@63.245.86.215) |
| 23:43.35 | rnovotny22 | Was beginning to think it was our system |
| 23:43.41 | *** join/#asterisk bluecrow76 (n=Mitch@ip24-250-77-177.br.br.cox.net) |
| 23:44.23 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 23:45.11 | *** part/#asterisk santiago (n=santiago@63.245.86.215) |
| 23:45.43 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:47.13 | *** join/#asterisk brian_ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 23:47.29 | Skkip | inexpesive channel banks?? I need to put a few analog lines around our building. |
| 23:49.09 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:49.27 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 23:49.50 | Alric | Can you remap a hard button on a Polycom phone to dial an extension? |
| 23:58.48 | *** join/#asterisk Garak_ (n=garak@stjhnf0111w-142162156203.dhcp-dynamic.nl.aliant.net) |
| 23:59.31 | Garak_ | Are their any text based sip clients that will work with asterisk? |
